am cf09e292: am 2d9ded1d: am d607e5a5: Merge "Fixes for GSM decoder"
* commit 'cf09e2926e3f76ac54a6efb6626a7eb256d2aaff':
Fixes for GSM decoder
diff --git a/camera/ICameraService.cpp b/camera/ICameraService.cpp
index fc3e437..a75cb48 100644
--- a/camera/ICameraService.cpp
+++ b/camera/ICameraService.cpp
@@ -209,6 +209,20 @@
return status;
}
+ virtual status_t setTorchMode(const String16& cameraId, bool enabled,
+ const sp<IBinder>& clientBinder)
+ {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICameraService::getInterfaceDescriptor());
+ data.writeString16(cameraId);
+ data.writeInt32(enabled ? 1 : 0);
+ data.writeStrongBinder(clientBinder);
+ remote()->transact(BnCameraService::SET_TORCH_MODE, data, &reply);
+
+ if (readExceptionCode(reply)) return -EPROTO;
+ return reply.readInt32();
+ }
+
// connect to camera service (pro client)
virtual status_t connectPro(const sp<IProCameraCallbacks>& cameraCb, int cameraId,
const String16 &clientPackageName, int clientUid,
@@ -490,6 +504,16 @@
}
return NO_ERROR;
} break;
+ case SET_TORCH_MODE: {
+ CHECK_INTERFACE(ICameraService, data, reply);
+ String16 cameraId = data.readString16();
+ bool enabled = data.readInt32() != 0 ? true : false;
+ const sp<IBinder> clientBinder = data.readStrongBinder();
+ status_t status = setTorchMode(cameraId, enabled, clientBinder);
+ reply->writeNoException();
+ reply->writeInt32(status);
+ return NO_ERROR;
+ } break;
default:
return BBinder::onTransact(code, data, reply, flags);
}
diff --git a/camera/ICameraServiceListener.cpp b/camera/ICameraServiceListener.cpp
index b2f1729..90a8bc2 100644
--- a/camera/ICameraServiceListener.cpp
+++ b/camera/ICameraServiceListener.cpp
@@ -29,6 +29,7 @@
namespace {
enum {
STATUS_CHANGED = IBinder::FIRST_CALL_TRANSACTION,
+ TORCH_STATUS_CHANGED,
};
}; // namespace anonymous
@@ -54,8 +55,21 @@
data,
&reply,
IBinder::FLAG_ONEWAY);
+ }
- reply.readExceptionCode();
+ virtual void onTorchStatusChanged(TorchStatus status, const String16 &cameraId)
+ {
+ Parcel data, reply;
+ data.writeInterfaceToken(
+ ICameraServiceListener::getInterfaceDescriptor());
+
+ data.writeInt32(static_cast<int32_t>(status));
+ data.writeString16(cameraId);
+
+ remote()->transact(TORCH_STATUS_CHANGED,
+ data,
+ &reply,
+ IBinder::FLAG_ONEWAY);
}
};
@@ -75,7 +89,16 @@
int32_t cameraId = data.readInt32();
onStatusChanged(status, cameraId);
- reply->writeNoException();
+
+ return NO_ERROR;
+ } break;
+ case TORCH_STATUS_CHANGED: {
+ CHECK_INTERFACE(ICameraServiceListener, data, reply);
+
+ TorchStatus status = static_cast<TorchStatus>(data.readInt32());
+ String16 cameraId = data.readString16();
+
+ onTorchStatusChanged(status, cameraId);
return NO_ERROR;
} break;
diff --git a/camera/tests/ProCameraTests.cpp b/camera/tests/ProCameraTests.cpp
index 1f5867a..6212678 100644
--- a/camera/tests/ProCameraTests.cpp
+++ b/camera/tests/ProCameraTests.cpp
@@ -89,6 +89,12 @@
mCondition.broadcast();
}
+ void onTorchStatusChanged(TorchStatus status, const String16& cameraId) {
+ dout << "On torch status changed: 0x" << std::hex
+ << (unsigned int) status << " cameraId " << cameraId.string()
+ << std::endl;
+ }
+
status_t waitForStatusChange(Status& newStatus) {
Mutex::Autolock al(mMutex);
diff --git a/cmds/screenrecord/screenrecord.cpp b/cmds/screenrecord/screenrecord.cpp
index 02df1d2..36a7e73 100644
--- a/cmds/screenrecord/screenrecord.cpp
+++ b/cmds/screenrecord/screenrecord.cpp
@@ -23,7 +23,10 @@
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
+#include <sys/stat.h>
+#include <sys/types.h>
#include <sys/wait.h>
+
#include <termios.h>
#include <unistd.h>
@@ -637,7 +640,13 @@
case FORMAT_MP4: {
// Configure muxer. We have to wait for the CSD blob from the encoder
// before we can start it.
- muxer = new MediaMuxer(fileName, MediaMuxer::OUTPUT_FORMAT_MPEG_4);
+ int fd = open(fileName, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
+ if (fd < 0) {
+ fprintf(stderr, "ERROR: couldn't open file\n");
+ abort();
+ }
+ muxer = new MediaMuxer(fd, MediaMuxer::OUTPUT_FORMAT_MPEG_4);
+ close(fd);
if (gRotate) {
muxer->setOrientationHint(90); // TODO: does this do anything?
}
diff --git a/cmds/stagefright/Android.mk b/cmds/stagefright/Android.mk
index 561ce02..0e3bc68 100644
--- a/cmds/stagefright/Android.mk
+++ b/cmds/stagefright/Android.mk
@@ -169,6 +169,48 @@
include $(CLEAR_VARS)
+LOCAL_SRC_FILES:= \
+ filters/argbtorgba.rs \
+ filters/nightvision.rs \
+ filters/saturation.rs \
+ mediafilter.cpp \
+
+LOCAL_SHARED_LIBRARIES := \
+ libstagefright \
+ liblog \
+ libutils \
+ libbinder \
+ libstagefright_foundation \
+ libmedia \
+ libgui \
+ libcutils \
+ libui \
+ libRScpp \
+
+LOCAL_C_INCLUDES:= \
+ $(TOP)/frameworks/av/media/libstagefright \
+ $(TOP)/frameworks/native/include/media/openmax \
+ $(TOP)/frameworks/rs/cpp \
+ $(TOP)/frameworks/rs \
+
+intermediates := $(call intermediates-dir-for,STATIC_LIBRARIES,libRS,TARGET,)
+LOCAL_C_INCLUDES += $(intermediates)
+
+LOCAL_STATIC_LIBRARIES:= \
+ libstagefright_mediafilter
+
+LOCAL_CFLAGS += -Wno-multichar
+
+LOCAL_MODULE_TAGS := optional
+
+LOCAL_MODULE:= mediafilter
+
+include $(BUILD_EXECUTABLE)
+
+################################################################################
+
+include $(CLEAR_VARS)
+
LOCAL_SRC_FILES:= \
muxer.cpp \
diff --git a/cmds/stagefright/audioloop.cpp b/cmds/stagefright/audioloop.cpp
index 96073f1..7b0de24 100644
--- a/cmds/stagefright/audioloop.cpp
+++ b/cmds/stagefright/audioloop.cpp
@@ -14,6 +14,10 @@
* limitations under the License.
*/
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <fcntl.h>
+
#include <binder/ProcessState.h>
#include <media/mediarecorder.h>
#include <media/stagefright/foundation/ADebug.h>
@@ -109,7 +113,12 @@
if (fileOut != NULL) {
// target file specified, write encoded AMR output
- sp<AMRWriter> writer = new AMRWriter(fileOut);
+ int fd = open(fileOut, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
+ if (fd < 0) {
+ return 1;
+ }
+ sp<AMRWriter> writer = new AMRWriter(fd);
+ close(fd);
writer->addSource(encoder);
writer->start();
sleep(duration);
diff --git a/cmds/stagefright/codec.cpp b/cmds/stagefright/codec.cpp
index fd02bcc..d987250 100644
--- a/cmds/stagefright/codec.cpp
+++ b/cmds/stagefright/codec.cpp
@@ -45,9 +45,10 @@
fprintf(stderr, "usage: %s [-a] use audio\n"
"\t\t[-v] use video\n"
"\t\t[-p] playback\n"
- "\t\t[-S] allocate buffers from a surface\n",
+ "\t\t[-S] allocate buffers from a surface\n"
+ "\t\t[-R] render output to surface (enables -S)\n"
+ "\t\t[-T] use render timestamps (enables -R)\n",
me);
-
exit(1);
}
@@ -71,7 +72,9 @@
const char *path,
bool useAudio,
bool useVideo,
- const android::sp<android::Surface> &surface) {
+ const android::sp<android::Surface> &surface,
+ bool renderSurface,
+ bool useTimestamp) {
using namespace android;
static int64_t kTimeout = 500ll;
@@ -136,6 +139,7 @@
CHECK(!stateByTrack.isEmpty());
int64_t startTimeUs = ALooper::GetNowUs();
+ int64_t startTimeRender = -1;
for (size_t i = 0; i < stateByTrack.size(); ++i) {
CodecState *state = &stateByTrack.editValueAt(i);
@@ -260,7 +264,23 @@
++state->mNumBuffersDecoded;
state->mNumBytesDecoded += size;
- err = state->mCodec->releaseOutputBuffer(index);
+ if (surface == NULL || !renderSurface) {
+ err = state->mCodec->releaseOutputBuffer(index);
+ } else if (useTimestamp) {
+ if (startTimeRender == -1) {
+ // begin rendering 2 vsyncs (~33ms) after first decode
+ startTimeRender =
+ systemTime(SYSTEM_TIME_MONOTONIC) + 33000000
+ - (presentationTimeUs * 1000);
+ }
+ presentationTimeUs =
+ (presentationTimeUs * 1000) + startTimeRender;
+ err = state->mCodec->renderOutputBufferAndRelease(
+ index, presentationTimeUs);
+ } else {
+ err = state->mCodec->renderOutputBufferAndRelease(index);
+ }
+
CHECK_EQ(err, (status_t)OK);
if (flags & MediaCodec::BUFFER_FLAG_EOS) {
@@ -320,34 +340,42 @@
bool useVideo = false;
bool playback = false;
bool useSurface = false;
+ bool renderSurface = false;
+ bool useTimestamp = false;
int res;
- while ((res = getopt(argc, argv, "havpSD")) >= 0) {
+ while ((res = getopt(argc, argv, "havpSDRT")) >= 0) {
switch (res) {
case 'a':
{
useAudio = true;
break;
}
-
case 'v':
{
useVideo = true;
break;
}
-
case 'p':
{
playback = true;
break;
}
-
+ case 'T':
+ {
+ useTimestamp = true;
+ }
+ // fall through
+ case 'R':
+ {
+ renderSurface = true;
+ }
+ // fall through
case 'S':
{
useSurface = true;
break;
}
-
case '?':
case 'h':
default:
@@ -422,7 +450,8 @@
player->stop();
player->reset();
} else {
- decode(looper, argv[0], useAudio, useVideo, surface);
+ decode(looper, argv[0], useAudio, useVideo, surface, renderSurface,
+ useTimestamp);
}
if (playback || (useSurface && useVideo)) {
diff --git a/include/media/nbaio/roundup.h b/cmds/stagefright/filters/argbtorgba.rs
similarity index 65%
rename from include/media/nbaio/roundup.h
rename to cmds/stagefright/filters/argbtorgba.rs
index 4c3cc25..229ff8c 100644
--- a/include/media/nbaio/roundup.h
+++ b/cmds/stagefright/filters/argbtorgba.rs
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2012 The Android Open Source Project
+ * Copyright (C) 2014 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
@@ -14,18 +14,13 @@
* limitations under the License.
*/
-#ifndef ROUNDUP_H
-#define ROUNDUP_H
+#pragma version(1)
+#pragma rs java_package_name(com.android.rs.cppbasic)
+#pragma rs_fp_relaxed
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-// Round up to the next highest power of 2
-unsigned roundup(unsigned v);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif // ROUNDUP_H
+void root(const uchar4 *v_in, uchar4 *v_out) {
+ v_out->x = v_in->y;
+ v_out->y = v_in->z;
+ v_out->z = v_in->w;
+ v_out->w = v_in->x;
+}
\ No newline at end of file
diff --git a/cmds/stagefright/filters/nightvision.rs b/cmds/stagefright/filters/nightvision.rs
new file mode 100644
index 0000000..f61413c
--- /dev/null
+++ b/cmds/stagefright/filters/nightvision.rs
@@ -0,0 +1,38 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#pragma version(1)
+#pragma rs java_package_name(com.android.rs.cppbasic)
+#pragma rs_fp_relaxed
+
+const static float3 gMonoMult = {0.299f, 0.587f, 0.114f};
+const static float3 gNightVisionMult = {0.5f, 1.f, 0.5f};
+
+// calculates luminance of pixel, then biases color balance toward green
+void root(const uchar4 *v_in, uchar4 *v_out) {
+ v_out->x = v_in->x; // don't modify A
+
+ // get RGB, scale 0-255 uchar to 0-1.0 float
+ float3 rgb = {v_in->y * 0.003921569f, v_in->z * 0.003921569f,
+ v_in->w * 0.003921569f};
+
+ // apply filter
+ float3 result = dot(rgb, gMonoMult) * gNightVisionMult;
+
+ v_out->y = (uchar)clamp((result.r * 255.f + 0.5f), 0.f, 255.f);
+ v_out->z = (uchar)clamp((result.g * 255.f + 0.5f), 0.f, 255.f);
+ v_out->w = (uchar)clamp((result.b * 255.f + 0.5f), 0.f, 255.f);
+}
diff --git a/cmds/stagefright/filters/saturation.rs b/cmds/stagefright/filters/saturation.rs
new file mode 100644
index 0000000..1de9dd8
--- /dev/null
+++ b/cmds/stagefright/filters/saturation.rs
@@ -0,0 +1,40 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#pragma version(1)
+#pragma rs java_package_name(com.android.rs.cppbasic)
+#pragma rs_fp_relaxed
+
+const static float3 gMonoMult = {0.299f, 0.587f, 0.114f};
+
+// global variables (parameters accessible to application code)
+float gSaturation = 1.0f;
+
+void root(const uchar4 *v_in, uchar4 *v_out) {
+ v_out->x = v_in->x; // don't modify A
+
+ // get RGB, scale 0-255 uchar to 0-1.0 float
+ float3 rgb = {v_in->y * 0.003921569f, v_in->z * 0.003921569f,
+ v_in->w * 0.003921569f};
+
+ // apply saturation filter
+ float3 result = dot(rgb, gMonoMult);
+ result = mix(result, rgb, gSaturation);
+
+ v_out->y = (uchar)clamp((result.r * 255.f + 0.5f), 0.f, 255.f);
+ v_out->z = (uchar)clamp((result.g * 255.f + 0.5f), 0.f, 255.f);
+ v_out->w = (uchar)clamp((result.b * 255.f + 0.5f), 0.f, 255.f);
+}
diff --git a/cmds/stagefright/mediafilter.cpp b/cmds/stagefright/mediafilter.cpp
new file mode 100644
index 0000000..f77b38b
--- /dev/null
+++ b/cmds/stagefright/mediafilter.cpp
@@ -0,0 +1,785 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "mediafilterTest"
+
+#include <inttypes.h>
+
+#include <binder/ProcessState.h>
+#include <filters/ColorConvert.h>
+#include <gui/ISurfaceComposer.h>
+#include <gui/SurfaceComposerClient.h>
+#include <gui/Surface.h>
+#include <media/ICrypto.h>
+#include <media/IMediaHTTPService.h>
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/DataSource.h>
+#include <media/stagefright/MediaCodec.h>
+#include <media/stagefright/NuMediaExtractor.h>
+#include <media/stagefright/RenderScriptWrapper.h>
+#include <OMX_IVCommon.h>
+#include <ui/DisplayInfo.h>
+
+#include "RenderScript.h"
+#include "ScriptC_argbtorgba.h"
+#include "ScriptC_nightvision.h"
+#include "ScriptC_saturation.h"
+
+// test parameters
+static const bool kTestFlush = true; // Note: true will drop 1 out of
+static const int kFlushAfterFrames = 25; // kFlushAfterFrames output frames
+static const int64_t kTimeout = 500ll;
+
+// built-in filter parameters
+static const int32_t kInvert = false; // ZeroFilter param
+static const float kBlurRadius = 15.0f; // IntrinsicBlurFilter param
+static const float kSaturation = 0.0f; // SaturationFilter param
+
+static void usage(const char *me) {
+ fprintf(stderr, "usage: [flags] %s\n"
+ "\t[-b] use IntrinsicBlurFilter\n"
+ "\t[-c] use argb to rgba conversion RSFilter\n"
+ "\t[-n] use night vision RSFilter\n"
+ "\t[-r] use saturation RSFilter\n"
+ "\t[-s] use SaturationFilter\n"
+ "\t[-z] use ZeroFilter (copy filter)\n"
+ "\t[-R] render output to surface (enables -S)\n"
+ "\t[-S] allocate buffers from a surface\n"
+ "\t[-T] use render timestamps (enables -R)\n",
+ me);
+ exit(1);
+}
+
+namespace android {
+
+struct SaturationRSFilter : RenderScriptWrapper::RSFilterCallback {
+ void init(RSC::sp<RSC::RS> context) {
+ mScript = new ScriptC_saturation(context);
+ mScript->set_gSaturation(3.f);
+ }
+
+ virtual status_t processBuffers(
+ RSC::Allocation *inBuffer, RSC::Allocation *outBuffer) {
+ mScript->forEach_root(inBuffer, outBuffer);
+
+ return OK;
+ }
+
+ status_t handleSetParameters(const sp<AMessage> &msg) {
+ return OK;
+ }
+
+private:
+ RSC::sp<ScriptC_saturation> mScript;
+};
+
+struct NightVisionRSFilter : RenderScriptWrapper::RSFilterCallback {
+ void init(RSC::sp<RSC::RS> context) {
+ mScript = new ScriptC_nightvision(context);
+ }
+
+ virtual status_t processBuffers(
+ RSC::Allocation *inBuffer, RSC::Allocation *outBuffer) {
+ mScript->forEach_root(inBuffer, outBuffer);
+
+ return OK;
+ }
+
+ status_t handleSetParameters(const sp<AMessage> &msg) {
+ return OK;
+ }
+
+private:
+ RSC::sp<ScriptC_nightvision> mScript;
+};
+
+struct ARGBToRGBARSFilter : RenderScriptWrapper::RSFilterCallback {
+ void init(RSC::sp<RSC::RS> context) {
+ mScript = new ScriptC_argbtorgba(context);
+ }
+
+ virtual status_t processBuffers(
+ RSC::Allocation *inBuffer, RSC::Allocation *outBuffer) {
+ mScript->forEach_root(inBuffer, outBuffer);
+
+ return OK;
+ }
+
+ status_t handleSetParameters(const sp<AMessage> &msg) {
+ return OK;
+ }
+
+private:
+ RSC::sp<ScriptC_argbtorgba> mScript;
+};
+
+struct CodecState {
+ sp<MediaCodec> mCodec;
+ Vector<sp<ABuffer> > mInBuffers;
+ Vector<sp<ABuffer> > mOutBuffers;
+ bool mSignalledInputEOS;
+ bool mSawOutputEOS;
+ int64_t mNumBuffersDecoded;
+};
+
+struct DecodedFrame {
+ size_t index;
+ size_t offset;
+ size_t size;
+ int64_t presentationTimeUs;
+ uint32_t flags;
+};
+
+enum FilterType {
+ FILTERTYPE_ZERO,
+ FILTERTYPE_INTRINSIC_BLUR,
+ FILTERTYPE_SATURATION,
+ FILTERTYPE_RS_SATURATION,
+ FILTERTYPE_RS_NIGHT_VISION,
+ FILTERTYPE_RS_ARGB_TO_RGBA,
+};
+
+size_t inputFramesSinceFlush = 0;
+void tryCopyDecodedBuffer(
+ List<DecodedFrame> *decodedFrameIndices,
+ CodecState *filterState,
+ CodecState *vidState) {
+ if (decodedFrameIndices->empty()) {
+ return;
+ }
+
+ size_t filterIndex;
+ status_t err = filterState->mCodec->dequeueInputBuffer(
+ &filterIndex, kTimeout);
+ if (err != OK) {
+ return;
+ }
+
+ ++inputFramesSinceFlush;
+
+ DecodedFrame frame = *decodedFrameIndices->begin();
+
+ // only consume a buffer if we are not going to flush, since we expect
+ // the dequeue -> flush -> queue operation to cause an error and
+ // not produce an output frame
+ if (!kTestFlush || inputFramesSinceFlush < kFlushAfterFrames) {
+ decodedFrameIndices->erase(decodedFrameIndices->begin());
+ }
+ size_t outIndex = frame.index;
+
+ const sp<ABuffer> &srcBuffer =
+ vidState->mOutBuffers.itemAt(outIndex);
+ const sp<ABuffer> &destBuffer =
+ filterState->mInBuffers.itemAt(filterIndex);
+
+ sp<AMessage> srcFormat, destFormat;
+ vidState->mCodec->getOutputFormat(&srcFormat);
+ filterState->mCodec->getInputFormat(&destFormat);
+
+ int32_t srcWidth, srcHeight, srcStride, srcSliceHeight;
+ int32_t srcColorFormat, destColorFormat;
+ int32_t destWidth, destHeight, destStride, destSliceHeight;
+ CHECK(srcFormat->findInt32("stride", &srcStride)
+ && srcFormat->findInt32("slice-height", &srcSliceHeight)
+ && srcFormat->findInt32("width", &srcWidth)
+ && srcFormat->findInt32("height", & srcHeight)
+ && srcFormat->findInt32("color-format", &srcColorFormat));
+ CHECK(destFormat->findInt32("stride", &destStride)
+ && destFormat->findInt32("slice-height", &destSliceHeight)
+ && destFormat->findInt32("width", &destWidth)
+ && destFormat->findInt32("height", & destHeight)
+ && destFormat->findInt32("color-format", &destColorFormat));
+
+ CHECK(srcWidth <= destStride && srcHeight <= destSliceHeight);
+
+ convertYUV420spToARGB(
+ srcBuffer->data(),
+ srcBuffer->data() + srcStride * srcSliceHeight,
+ srcWidth,
+ srcHeight,
+ destBuffer->data());
+
+ // copy timestamp
+ int64_t timeUs;
+ CHECK(srcBuffer->meta()->findInt64("timeUs", &timeUs));
+ destBuffer->meta()->setInt64("timeUs", timeUs);
+
+ if (kTestFlush && inputFramesSinceFlush >= kFlushAfterFrames) {
+ inputFramesSinceFlush = 0;
+
+ // check that queueing a buffer that was dequeued before flush
+ // fails with expected error EACCES
+ filterState->mCodec->flush();
+
+ err = filterState->mCodec->queueInputBuffer(
+ filterIndex, 0 /* offset */, destBuffer->size(),
+ timeUs, frame.flags);
+
+ if (err == OK) {
+ ALOGE("FAIL: queue after flush returned OK");
+ } else if (err != -EACCES) {
+ ALOGE("queueInputBuffer after flush returned %d, "
+ "expected -EACCES (-13)", err);
+ }
+ } else {
+ err = filterState->mCodec->queueInputBuffer(
+ filterIndex, 0 /* offset */, destBuffer->size(),
+ timeUs, frame.flags);
+ CHECK(err == OK);
+
+ err = vidState->mCodec->releaseOutputBuffer(outIndex);
+ CHECK(err == OK);
+ }
+}
+
+size_t outputFramesSinceFlush = 0;
+void tryDrainOutputBuffer(
+ CodecState *filterState,
+ const sp<Surface> &surface, bool renderSurface,
+ bool useTimestamp, int64_t *startTimeRender) {
+ size_t index;
+ size_t offset;
+ size_t size;
+ int64_t presentationTimeUs;
+ uint32_t flags;
+ status_t err = filterState->mCodec->dequeueOutputBuffer(
+ &index, &offset, &size, &presentationTimeUs, &flags,
+ kTimeout);
+
+ if (err != OK) {
+ return;
+ }
+
+ ++outputFramesSinceFlush;
+
+ if (kTestFlush && outputFramesSinceFlush >= kFlushAfterFrames) {
+ filterState->mCodec->flush();
+ }
+
+ if (surface == NULL || !renderSurface) {
+ err = filterState->mCodec->releaseOutputBuffer(index);
+ } else if (useTimestamp) {
+ if (*startTimeRender == -1) {
+ // begin rendering 2 vsyncs after first decode
+ *startTimeRender = systemTime(SYSTEM_TIME_MONOTONIC)
+ + 33000000 - (presentationTimeUs * 1000);
+ }
+ presentationTimeUs =
+ (presentationTimeUs * 1000) + *startTimeRender;
+ err = filterState->mCodec->renderOutputBufferAndRelease(
+ index, presentationTimeUs);
+ } else {
+ err = filterState->mCodec->renderOutputBufferAndRelease(index);
+ }
+
+ if (kTestFlush && outputFramesSinceFlush >= kFlushAfterFrames) {
+ outputFramesSinceFlush = 0;
+
+ // releasing the buffer dequeued before flush should cause an error
+ // if so, the frame will also be skipped in output stream
+ if (err == OK) {
+ ALOGE("FAIL: release after flush returned OK");
+ } else if (err != -EACCES) {
+ ALOGE("releaseOutputBuffer after flush returned %d, "
+ "expected -EACCES (-13)", err);
+ }
+ } else {
+ CHECK(err == OK);
+ }
+
+ if (flags & MediaCodec::BUFFER_FLAG_EOS) {
+ ALOGV("reached EOS on output.");
+ filterState->mSawOutputEOS = true;
+ }
+}
+
+static int decode(
+ const sp<ALooper> &looper,
+ const char *path,
+ const sp<Surface> &surface,
+ bool renderSurface,
+ bool useTimestamp,
+ FilterType filterType) {
+
+ static int64_t kTimeout = 500ll;
+
+ sp<NuMediaExtractor> extractor = new NuMediaExtractor;
+ if (extractor->setDataSource(NULL /* httpService */, path) != OK) {
+ fprintf(stderr, "unable to instantiate extractor.\n");
+ return 1;
+ }
+
+ KeyedVector<size_t, CodecState> stateByTrack;
+
+ CodecState *vidState = NULL;
+ for (size_t i = 0; i < extractor->countTracks(); ++i) {
+ sp<AMessage> format;
+ status_t err = extractor->getTrackFormat(i, &format);
+ CHECK(err == OK);
+
+ AString mime;
+ CHECK(format->findString("mime", &mime));
+ bool isVideo = !strncasecmp(mime.c_str(), "video/", 6);
+ if (!isVideo) {
+ continue;
+ }
+
+ ALOGV("selecting track %zu", i);
+
+ err = extractor->selectTrack(i);
+ CHECK(err == OK);
+
+ CodecState *state =
+ &stateByTrack.editValueAt(stateByTrack.add(i, CodecState()));
+
+ vidState = state;
+
+ state->mNumBuffersDecoded = 0;
+
+ state->mCodec = MediaCodec::CreateByType(
+ looper, mime.c_str(), false /* encoder */);
+
+ CHECK(state->mCodec != NULL);
+
+ err = state->mCodec->configure(
+ format, NULL /* surface */, NULL /* crypto */, 0 /* flags */);
+
+ CHECK(err == OK);
+
+ state->mSignalledInputEOS = false;
+ state->mSawOutputEOS = false;
+
+ break;
+ }
+ CHECK(!stateByTrack.isEmpty());
+ CHECK(vidState != NULL);
+ sp<AMessage> vidFormat;
+ vidState->mCodec->getOutputFormat(&vidFormat);
+
+ // set filter to use ARGB8888
+ vidFormat->setInt32("color-format", OMX_COLOR_Format32bitARGB8888);
+ // set app cache directory path
+ vidFormat->setString("cacheDir", "/system/bin");
+
+ // create RenderScript context for RSFilters
+ RSC::sp<RSC::RS> context = new RSC::RS();
+ context->init("/system/bin");
+
+ sp<RenderScriptWrapper::RSFilterCallback> rsFilter;
+
+ // create renderscript wrapper for RSFilters
+ sp<RenderScriptWrapper> rsWrapper = new RenderScriptWrapper;
+ rsWrapper->mContext = context.get();
+
+ CodecState *filterState = new CodecState();
+ filterState->mNumBuffersDecoded = 0;
+
+ sp<AMessage> params = new AMessage();
+
+ switch (filterType) {
+ case FILTERTYPE_ZERO:
+ {
+ filterState->mCodec = MediaCodec::CreateByComponentName(
+ looper, "android.filter.zerofilter");
+ params->setInt32("invert", kInvert);
+ break;
+ }
+ case FILTERTYPE_INTRINSIC_BLUR:
+ {
+ filterState->mCodec = MediaCodec::CreateByComponentName(
+ looper, "android.filter.intrinsicblur");
+ params->setFloat("blur-radius", kBlurRadius);
+ break;
+ }
+ case FILTERTYPE_SATURATION:
+ {
+ filterState->mCodec = MediaCodec::CreateByComponentName(
+ looper, "android.filter.saturation");
+ params->setFloat("saturation", kSaturation);
+ break;
+ }
+ case FILTERTYPE_RS_SATURATION:
+ {
+ SaturationRSFilter *satFilter = new SaturationRSFilter;
+ satFilter->init(context);
+ rsFilter = satFilter;
+ rsWrapper->mCallback = rsFilter;
+ vidFormat->setObject("rs-wrapper", rsWrapper);
+
+ filterState->mCodec = MediaCodec::CreateByComponentName(
+ looper, "android.filter.RenderScript");
+ break;
+ }
+ case FILTERTYPE_RS_NIGHT_VISION:
+ {
+ NightVisionRSFilter *nightVisionFilter = new NightVisionRSFilter;
+ nightVisionFilter->init(context);
+ rsFilter = nightVisionFilter;
+ rsWrapper->mCallback = rsFilter;
+ vidFormat->setObject("rs-wrapper", rsWrapper);
+
+ filterState->mCodec = MediaCodec::CreateByComponentName(
+ looper, "android.filter.RenderScript");
+ break;
+ }
+ case FILTERTYPE_RS_ARGB_TO_RGBA:
+ {
+ ARGBToRGBARSFilter *argbToRgbaFilter = new ARGBToRGBARSFilter;
+ argbToRgbaFilter->init(context);
+ rsFilter = argbToRgbaFilter;
+ rsWrapper->mCallback = rsFilter;
+ vidFormat->setObject("rs-wrapper", rsWrapper);
+
+ filterState->mCodec = MediaCodec::CreateByComponentName(
+ looper, "android.filter.RenderScript");
+ break;
+ }
+ default:
+ {
+ LOG_ALWAYS_FATAL("mediacodec.cpp error: unrecognized FilterType");
+ break;
+ }
+ }
+ CHECK(filterState->mCodec != NULL);
+
+ status_t err = filterState->mCodec->configure(
+ vidFormat /* format */, surface, NULL /* crypto */, 0 /* flags */);
+ CHECK(err == OK);
+
+ filterState->mSignalledInputEOS = false;
+ filterState->mSawOutputEOS = false;
+
+ int64_t startTimeUs = ALooper::GetNowUs();
+ int64_t startTimeRender = -1;
+
+ for (size_t i = 0; i < stateByTrack.size(); ++i) {
+ CodecState *state = &stateByTrack.editValueAt(i);
+
+ sp<MediaCodec> codec = state->mCodec;
+
+ CHECK_EQ((status_t)OK, codec->start());
+
+ CHECK_EQ((status_t)OK, codec->getInputBuffers(&state->mInBuffers));
+ CHECK_EQ((status_t)OK, codec->getOutputBuffers(&state->mOutBuffers));
+
+ ALOGV("got %zu input and %zu output buffers",
+ state->mInBuffers.size(), state->mOutBuffers.size());
+ }
+
+ CHECK_EQ((status_t)OK, filterState->mCodec->setParameters(params));
+
+ if (kTestFlush) {
+ status_t flushErr = filterState->mCodec->flush();
+ if (flushErr == OK) {
+ ALOGE("FAIL: Flush before start returned OK");
+ } else {
+ ALOGV("Flush before start returned status %d, usually ENOSYS (-38)",
+ flushErr);
+ }
+ }
+
+ CHECK_EQ((status_t)OK, filterState->mCodec->start());
+ CHECK_EQ((status_t)OK, filterState->mCodec->getInputBuffers(
+ &filterState->mInBuffers));
+ CHECK_EQ((status_t)OK, filterState->mCodec->getOutputBuffers(
+ &filterState->mOutBuffers));
+
+ if (kTestFlush) {
+ status_t flushErr = filterState->mCodec->flush();
+ if (flushErr != OK) {
+ ALOGE("FAIL: Flush after start returned %d, expect OK (0)",
+ flushErr);
+ } else {
+ ALOGV("Flush immediately after start OK");
+ }
+ }
+
+ List<DecodedFrame> decodedFrameIndices;
+
+ // loop until decoder reaches EOS
+ bool sawInputEOS = false;
+ bool sawOutputEOSOnAllTracks = false;
+ while (!sawOutputEOSOnAllTracks) {
+ if (!sawInputEOS) {
+ size_t trackIndex;
+ status_t err = extractor->getSampleTrackIndex(&trackIndex);
+
+ if (err != OK) {
+ ALOGV("saw input eos");
+ sawInputEOS = true;
+ } else {
+ CodecState *state = &stateByTrack.editValueFor(trackIndex);
+
+ size_t index;
+ err = state->mCodec->dequeueInputBuffer(&index, kTimeout);
+
+ if (err == OK) {
+ ALOGV("filling input buffer %zu", index);
+
+ const sp<ABuffer> &buffer = state->mInBuffers.itemAt(index);
+
+ err = extractor->readSampleData(buffer);
+ CHECK(err == OK);
+
+ int64_t timeUs;
+ err = extractor->getSampleTime(&timeUs);
+ CHECK(err == OK);
+
+ uint32_t bufferFlags = 0;
+
+ err = state->mCodec->queueInputBuffer(
+ index, 0 /* offset */, buffer->size(),
+ timeUs, bufferFlags);
+
+ CHECK(err == OK);
+
+ extractor->advance();
+ } else {
+ CHECK_EQ(err, -EAGAIN);
+ }
+ }
+ } else {
+ for (size_t i = 0; i < stateByTrack.size(); ++i) {
+ CodecState *state = &stateByTrack.editValueAt(i);
+
+ if (!state->mSignalledInputEOS) {
+ size_t index;
+ status_t err =
+ state->mCodec->dequeueInputBuffer(&index, kTimeout);
+
+ if (err == OK) {
+ ALOGV("signalling input EOS on track %zu", i);
+
+ err = state->mCodec->queueInputBuffer(
+ index, 0 /* offset */, 0 /* size */,
+ 0ll /* timeUs */, MediaCodec::BUFFER_FLAG_EOS);
+
+ CHECK(err == OK);
+
+ state->mSignalledInputEOS = true;
+ } else {
+ CHECK_EQ(err, -EAGAIN);
+ }
+ }
+ }
+ }
+
+ sawOutputEOSOnAllTracks = true;
+ for (size_t i = 0; i < stateByTrack.size(); ++i) {
+ CodecState *state = &stateByTrack.editValueAt(i);
+
+ if (state->mSawOutputEOS) {
+ continue;
+ } else {
+ sawOutputEOSOnAllTracks = false;
+ }
+
+ DecodedFrame frame;
+ status_t err = state->mCodec->dequeueOutputBuffer(
+ &frame.index, &frame.offset, &frame.size,
+ &frame.presentationTimeUs, &frame.flags, kTimeout);
+
+ if (err == OK) {
+ ALOGV("draining decoded buffer %zu, time = %lld us",
+ frame.index, frame.presentationTimeUs);
+
+ ++(state->mNumBuffersDecoded);
+
+ decodedFrameIndices.push_back(frame);
+
+ if (frame.flags & MediaCodec::BUFFER_FLAG_EOS) {
+ ALOGV("reached EOS on decoder output.");
+ state->mSawOutputEOS = true;
+ }
+
+ } else if (err == INFO_OUTPUT_BUFFERS_CHANGED) {
+ ALOGV("INFO_OUTPUT_BUFFERS_CHANGED");
+ CHECK_EQ((status_t)OK, state->mCodec->getOutputBuffers(
+ &state->mOutBuffers));
+
+ ALOGV("got %zu output buffers", state->mOutBuffers.size());
+ } else if (err == INFO_FORMAT_CHANGED) {
+ sp<AMessage> format;
+ CHECK_EQ((status_t)OK, state->mCodec->getOutputFormat(&format));
+
+ ALOGV("INFO_FORMAT_CHANGED: %s",
+ format->debugString().c_str());
+ } else {
+ CHECK_EQ(err, -EAGAIN);
+ }
+
+ tryCopyDecodedBuffer(&decodedFrameIndices, filterState, vidState);
+
+ tryDrainOutputBuffer(
+ filterState, surface, renderSurface,
+ useTimestamp, &startTimeRender);
+ }
+ }
+
+ // after EOS on decoder, let filter reach EOS
+ while (!filterState->mSawOutputEOS) {
+ tryCopyDecodedBuffer(&decodedFrameIndices, filterState, vidState);
+
+ tryDrainOutputBuffer(
+ filterState, surface, renderSurface,
+ useTimestamp, &startTimeRender);
+ }
+
+ int64_t elapsedTimeUs = ALooper::GetNowUs() - startTimeUs;
+
+ for (size_t i = 0; i < stateByTrack.size(); ++i) {
+ CodecState *state = &stateByTrack.editValueAt(i);
+
+ CHECK_EQ((status_t)OK, state->mCodec->release());
+
+ printf("track %zu: %" PRId64 " frames decoded and filtered, "
+ "%.2f fps.\n", i, state->mNumBuffersDecoded,
+ state->mNumBuffersDecoded * 1E6 / elapsedTimeUs);
+ }
+
+ return 0;
+}
+
+} // namespace android
+
+int main(int argc, char **argv) {
+ using namespace android;
+
+ const char *me = argv[0];
+
+ bool useSurface = false;
+ bool renderSurface = false;
+ bool useTimestamp = false;
+ FilterType filterType = FILTERTYPE_ZERO;
+
+ int res;
+ while ((res = getopt(argc, argv, "bcnrszTRSh")) >= 0) {
+ switch (res) {
+ case 'b':
+ {
+ filterType = FILTERTYPE_INTRINSIC_BLUR;
+ break;
+ }
+ case 'c':
+ {
+ filterType = FILTERTYPE_RS_ARGB_TO_RGBA;
+ break;
+ }
+ case 'n':
+ {
+ filterType = FILTERTYPE_RS_NIGHT_VISION;
+ break;
+ }
+ case 'r':
+ {
+ filterType = FILTERTYPE_RS_SATURATION;
+ break;
+ }
+ case 's':
+ {
+ filterType = FILTERTYPE_SATURATION;
+ break;
+ }
+ case 'z':
+ {
+ filterType = FILTERTYPE_ZERO;
+ break;
+ }
+ case 'T':
+ {
+ useTimestamp = true;
+ }
+ // fall through
+ case 'R':
+ {
+ renderSurface = true;
+ }
+ // fall through
+ case 'S':
+ {
+ useSurface = true;
+ break;
+ }
+ case '?':
+ case 'h':
+ default:
+ {
+ usage(me);
+ break;
+ }
+ }
+ }
+
+ argc -= optind;
+ argv += optind;
+
+ if (argc != 1) {
+ usage(me);
+ }
+
+ ProcessState::self()->startThreadPool();
+
+ DataSource::RegisterDefaultSniffers();
+
+ android::sp<ALooper> looper = new ALooper;
+ looper->start();
+
+ android::sp<SurfaceComposerClient> composerClient;
+ android::sp<SurfaceControl> control;
+ android::sp<Surface> surface;
+
+ if (useSurface) {
+ composerClient = new SurfaceComposerClient;
+ CHECK_EQ((status_t)OK, composerClient->initCheck());
+
+ android::sp<IBinder> display(SurfaceComposerClient::getBuiltInDisplay(
+ ISurfaceComposer::eDisplayIdMain));
+ DisplayInfo info;
+ SurfaceComposerClient::getDisplayInfo(display, &info);
+ ssize_t displayWidth = info.w;
+ ssize_t displayHeight = info.h;
+
+ ALOGV("display is %zd x %zd", displayWidth, displayHeight);
+
+ control = composerClient->createSurface(
+ String8("A Surface"), displayWidth, displayHeight,
+ PIXEL_FORMAT_RGBA_8888, 0);
+
+ CHECK(control != NULL);
+ CHECK(control->isValid());
+
+ SurfaceComposerClient::openGlobalTransaction();
+ CHECK_EQ((status_t)OK, control->setLayer(INT_MAX));
+ CHECK_EQ((status_t)OK, control->show());
+ SurfaceComposerClient::closeGlobalTransaction();
+
+ surface = control->getSurface();
+ CHECK(surface != NULL);
+ }
+
+ decode(looper, argv[0], surface, renderSurface, useTimestamp, filterType);
+
+ if (useSurface) {
+ composerClient->dispose();
+ }
+
+ looper->stop();
+
+ return 0;
+}
diff --git a/cmds/stagefright/muxer.cpp b/cmds/stagefright/muxer.cpp
index f4a33e8..461b56c 100644
--- a/cmds/stagefright/muxer.cpp
+++ b/cmds/stagefright/muxer.cpp
@@ -17,6 +17,9 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "muxer"
#include <inttypes.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <fcntl.h>
#include <utils/Log.h>
#include <binder/ProcessState.h>
@@ -72,8 +75,15 @@
ALOGV("input file %s, output file %s", path, outputFileName);
ALOGV("useAudio %d, useVideo %d", useAudio, useVideo);
- sp<MediaMuxer> muxer = new MediaMuxer(outputFileName,
+ int fd = open(outputFileName, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
+
+ if (fd < 0) {
+ ALOGE("couldn't open file");
+ return fd;
+ }
+ sp<MediaMuxer> muxer = new MediaMuxer(fd,
MediaMuxer::OUTPUT_FORMAT_MPEG_4);
+ close(fd);
size_t trackCount = extractor->countTracks();
// Map the extractor's track index to the muxer's track index.
diff --git a/cmds/stagefright/recordvideo.cpp b/cmds/stagefright/recordvideo.cpp
index 9f547c7..2ad40bd 100644
--- a/cmds/stagefright/recordvideo.cpp
+++ b/cmds/stagefright/recordvideo.cpp
@@ -17,6 +17,10 @@
#include "SineSource.h"
#include <inttypes.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <fcntl.h>
+
#include <binder/ProcessState.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/AudioPlayer.h>
@@ -300,7 +304,13 @@
client.interface(), enc_meta, true /* createEncoder */, source,
0, preferSoftwareCodec ? OMXCodec::kPreferSoftwareCodecs : 0);
- sp<MPEG4Writer> writer = new MPEG4Writer(fileName);
+ int fd = open(fileName, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
+ if (fd < 0) {
+ fprintf(stderr, "couldn't open file");
+ return 1;
+ }
+ sp<MPEG4Writer> writer = new MPEG4Writer(fd);
+ close(fd);
writer->addSource(encoder);
int64_t start = systemTime();
CHECK_EQ((status_t)OK, writer->start());
diff --git a/cmds/stagefright/stagefright.cpp b/cmds/stagefright/stagefright.cpp
index 81edcb4..318b56d 100644
--- a/cmds/stagefright/stagefright.cpp
+++ b/cmds/stagefright/stagefright.cpp
@@ -19,6 +19,8 @@
#include <stdlib.h>
#include <string.h>
#include <sys/time.h>
+#include <sys/types.h>
+#include <sys/stat.h>
//#define LOG_NDEBUG 0
#define LOG_TAG "stagefright"
@@ -506,8 +508,13 @@
sp<MPEG4Writer> writer =
new MPEG4Writer(gWriteMP4Filename.string());
#else
+ int fd = open(gWriteMP4Filename.string(), O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
+ if (fd < 0) {
+ fprintf(stderr, "couldn't open file");
+ return;
+ }
sp<MPEG2TSWriter> writer =
- new MPEG2TSWriter(gWriteMP4Filename.string());
+ new MPEG2TSWriter(fd);
#endif
// at most one minute.
diff --git a/include/camera/ICameraService.h b/include/camera/ICameraService.h
index f7f06bb..cc41efe 100644
--- a/include/camera/ICameraService.h
+++ b/include/camera/ICameraService.h
@@ -53,6 +53,7 @@
GET_LEGACY_PARAMETERS,
SUPPORTS_CAMERA_API,
CONNECT_LEGACY,
+ SET_TORCH_MODE,
};
enum {
@@ -142,6 +143,12 @@
int clientUid,
/*out*/
sp<ICamera>& device) = 0;
+
+ /**
+ * Turn on or off a camera's torch mode.
+ */
+ virtual status_t setTorchMode(const String16& cameraId, bool enabled,
+ const sp<IBinder>& clientBinder) = 0;
};
// ----------------------------------------------------------------------------
diff --git a/include/camera/ICameraServiceListener.h b/include/camera/ICameraServiceListener.h
index 0a0e43a..9e8b912 100644
--- a/include/camera/ICameraServiceListener.h
+++ b/include/camera/ICameraServiceListener.h
@@ -66,9 +66,33 @@
STATUS_UNKNOWN = 0xFFFFFFFF,
};
+ /**
+ * The torch mode status of a camera.
+ *
+ * Initial status will be transmitted with onTorchStatusChanged immediately
+ * after this listener is added to the service listener list.
+ */
+ enum TorchStatus {
+ // The camera's torch mode has become available to use via
+ // setTorchMode().
+ TORCH_STATUS_AVAILABLE = TORCH_MODE_STATUS_AVAILABLE,
+ // The camera's torch mode has become not available to use via
+ // setTorchMode().
+ TORCH_STATUS_NOT_AVAILABLE = TORCH_MODE_STATUS_RESOURCE_BUSY,
+ // The camera's torch mode has been turned off by setTorchMode().
+ TORCH_STATUS_OFF = TORCH_MODE_STATUS_OFF,
+ // The camera's torch mode has been turned on by setTorchMode().
+ TORCH_STATUS_ON = 0x80000000,
+
+ // Use to initialize variables only
+ TORCH_STATUS_UNKNOWN = 0xFFFFFFFF,
+ };
+
DECLARE_META_INTERFACE(CameraServiceListener);
virtual void onStatusChanged(Status status, int32_t cameraId) = 0;
+
+ virtual void onTorchStatusChanged(TorchStatus status, const String16& cameraId) = 0;
};
// ----------------------------------------------------------------------------
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index 843a354..2ab3dd6 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -201,7 +201,7 @@
// IAudioPolicyService interface (see AudioPolicyInterface for method descriptions)
//
static status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state,
- const char *device_address);
+ const char *device_address, const char *device_name);
static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
const char *device_address);
static status_t setPhoneState(audio_mode_t state);
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index fd51b8f..2e1ed6c 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -79,9 +79,7 @@
size_t size; // input/output in bytes == frameCount * frameSize
// on input it is unused
// on output is the number of bytes actually filled
- // FIXME this is redundant with respect to frameCount,
- // and TRANSFER_OBTAIN mode is broken for 8-bit data
- // since we don't define the frame format
+ // FIXME this is redundant with respect to frameCount.
union {
void* raw;
@@ -154,9 +152,9 @@
* streamType: Select the type of audio stream this track is attached to
* (e.g. AUDIO_STREAM_MUSIC).
* sampleRate: Data source sampling rate in Hz.
- * format: Audio format. For mixed tracks, any PCM format supported by server is OK
- * or AUDIO_FORMAT_PCM_8_BIT which is handled on client side. For direct
- * and offloaded tracks, the possible format(s) depends on the output sink.
+ * format: Audio format. For mixed tracks, any PCM format supported by server is OK.
+ * For direct and offloaded tracks, the possible format(s) depends on the
+ * output sink.
* channelMask: Channel mask, such that audio_is_output_channel(channelMask) is true.
* frameCount: Minimum size of track PCM buffer in frames. This defines the
* application's contribution to the
@@ -193,7 +191,6 @@
/* Creates an audio track and registers it with AudioFlinger.
* With this constructor, the track is configured for static buffer mode.
- * The format must not be 8-bit linear PCM.
* Data to be rendered is passed in a shared memory buffer
* identified by the argument sharedBuffer, which must be non-0.
* The memory should be initialized to the desired data before calling start().
@@ -614,6 +611,7 @@
void pause(); // suspend thread from execution at next loop boundary
void resume(); // allow thread to execute, if not requested to exit
+ void wake(); // wake to handle changed notification conditions.
private:
void pauseInternal(nsecs_t ns = 0LL);
@@ -628,7 +626,9 @@
bool mPaused; // whether thread is requested to pause at next loop entry
bool mPausedInt; // whether thread internally requests pause
nsecs_t mPausedNs; // if mPausedInt then associated timeout, otherwise ignored
- bool mIgnoreNextPausedInt; // whether to ignore next mPausedInt request
+ bool mIgnoreNextPausedInt; // skip any internal pause and go immediately
+ // to processAudioBuffer() as state may have changed
+ // since pause time calculated.
};
// body of AudioTrackThread::threadLoop()
@@ -680,7 +680,7 @@
float mVolume[2];
float mSendLevel;
- mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it.
+ mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it
size_t mFrameCount; // corresponds to current IAudioTrack, value is
// reported back by AudioFlinger to the client
size_t mReqFrameCount; // frame count to request the first or next time
@@ -698,10 +698,7 @@
const audio_offload_info_t* mOffloadInfo;
audio_attributes_t mAttributes;
- // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data. For 8-bit PCM data, it's
- // twice as large as mFrameSize because data is expanded to 16-bit before it's stored in buffer.
- size_t mFrameSize; // app-level frame size
- size_t mFrameSizeAF; // AudioFlinger frame size
+ size_t mFrameSize; // frame size in bytes
status_t mStatus;
@@ -732,13 +729,20 @@
bool mRefreshRemaining; // processAudioBuffer() should refresh
// mRemainingFrames and mRetryOnPartialBuffer
+ // used for static track cbf and restoration
+ int32_t mLoopCount; // last setLoop loopCount; zero means disabled
+ uint32_t mLoopStart; // last setLoop loopStart
+ uint32_t mLoopEnd; // last setLoop loopEnd
+ int32_t mLoopCountNotified; // the last loopCount notified by callback.
+ // mLoopCountNotified counts down, matching
+ // the remaining loop count for static track
+ // playback.
+
// These are private to processAudioBuffer(), and are not protected by a lock
uint32_t mRemainingFrames; // number of frames to request in obtainBuffer()
bool mRetryOnPartialBuffer; // sleep and retry after partial obtainBuffer()
uint32_t mObservedSequence; // last observed value of mSequence
- uint32_t mLoopPeriod; // in frames, zero means looping is disabled
-
uint32_t mMarkerPosition; // in wrapping (overflow) frame units
bool mMarkerReached;
uint32_t mNewPosition; // in frames
diff --git a/include/media/EffectsFactoryApi.h b/include/media/EffectsFactoryApi.h
index b1ed7b0..64a3212 100644
--- a/include/media/EffectsFactoryApi.h
+++ b/include/media/EffectsFactoryApi.h
@@ -171,6 +171,8 @@
////////////////////////////////////////////////////////////////////////////////
int EffectIsNullUuid(const effect_uuid_t *pEffectUuid);
+int EffectDumpEffects(int fd);
+
#if __cplusplus
} // extern "C"
#endif
diff --git a/include/media/IAudioPolicyService.h b/include/media/IAudioPolicyService.h
index c98c475..fecc6f1 100644
--- a/include/media/IAudioPolicyService.h
+++ b/include/media/IAudioPolicyService.h
@@ -44,7 +44,8 @@
//
virtual status_t setDeviceConnectionState(audio_devices_t device,
audio_policy_dev_state_t state,
- const char *device_address) = 0;
+ const char *device_address,
+ const char *device_name) = 0;
virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
const char *device_address) = 0;
virtual status_t setPhoneState(audio_mode_t state) = 0;
diff --git a/include/media/IMediaPlayerService.h b/include/media/IMediaPlayerService.h
index 67b599a..49a3d61 100644
--- a/include/media/IMediaPlayerService.h
+++ b/include/media/IMediaPlayerService.h
@@ -49,7 +49,8 @@
virtual sp<IMediaRecorder> createMediaRecorder() = 0;
virtual sp<IMediaMetadataRetriever> createMetadataRetriever() = 0;
- virtual sp<IMediaPlayer> create(const sp<IMediaPlayerClient>& client, int audioSessionId = 0) = 0;
+ virtual sp<IMediaPlayer> create(const sp<IMediaPlayerClient>& client, int audioSessionId = 0)
+ = 0;
virtual sp<IOMX> getOMX() = 0;
virtual sp<ICrypto> makeCrypto() = 0;
diff --git a/include/media/IMediaRecorder.h b/include/media/IMediaRecorder.h
index 3e67550..509c06b 100644
--- a/include/media/IMediaRecorder.h
+++ b/include/media/IMediaRecorder.h
@@ -41,7 +41,6 @@
virtual status_t setOutputFormat(int of) = 0;
virtual status_t setVideoEncoder(int ve) = 0;
virtual status_t setAudioEncoder(int ae) = 0;
- virtual status_t setOutputFile(const char* path) = 0;
virtual status_t setOutputFile(int fd, int64_t offset, int64_t length) = 0;
virtual status_t setVideoSize(int width, int height) = 0;
virtual status_t setVideoFrameRate(int frames_per_second) = 0;
diff --git a/include/media/IOMX.h b/include/media/IOMX.h
index 627f23b..6def65b 100644
--- a/include/media/IOMX.h
+++ b/include/media/IOMX.h
@@ -147,6 +147,7 @@
INTERNAL_OPTION_SUSPEND, // data is a bool
INTERNAL_OPTION_REPEAT_PREVIOUS_FRAME_DELAY, // data is an int64_t
INTERNAL_OPTION_MAX_TIMESTAMP_GAP, // data is int64_t
+ INTERNAL_OPTION_MAX_FPS, // data is float
INTERNAL_OPTION_START_TIME, // data is an int64_t
INTERNAL_OPTION_TIME_LAPSE, // data is an int64_t[2]
};
diff --git a/include/media/MediaRecorderBase.h b/include/media/MediaRecorderBase.h
index d7ac302..f55063e 100644
--- a/include/media/MediaRecorderBase.h
+++ b/include/media/MediaRecorderBase.h
@@ -43,7 +43,6 @@
virtual status_t setCamera(const sp<ICamera>& camera,
const sp<ICameraRecordingProxy>& proxy) = 0;
virtual status_t setPreviewSurface(const sp<IGraphicBufferProducer>& surface) = 0;
- virtual status_t setOutputFile(const char *path) = 0;
virtual status_t setOutputFile(int fd, int64_t offset, int64_t length) = 0;
virtual status_t setOutputFileAuxiliary(int fd) {return INVALID_OPERATION;}
virtual status_t setParameters(const String8& params) = 0;
diff --git a/include/media/SingleStateQueue.h b/include/media/SingleStateQueue.h
index 04c5fd0..d423962 100644
--- a/include/media/SingleStateQueue.h
+++ b/include/media/SingleStateQueue.h
@@ -21,6 +21,7 @@
// Non-blocking single-reader / single-writer multi-word atomic load / store
#include <stdint.h>
+#include <cutils/atomic.h>
namespace android {
@@ -31,6 +32,12 @@
class Mutator;
class Observer;
+ enum SSQ_STATUS {
+ SSQ_PENDING, /* = 0 */
+ SSQ_READ,
+ SSQ_DONE,
+ };
+
struct Shared {
// needs to be part of a union so don't define constructor or destructor
@@ -41,28 +48,56 @@
void init() { mAck = 0; mSequence = 0; }
volatile int32_t mAck;
-#if 0
- int mPad[7];
- // cache line boundary
-#endif
volatile int32_t mSequence;
T mValue;
};
class Mutator {
public:
- Mutator(Shared *shared);
- /*virtual*/ ~Mutator() { }
+ Mutator(Shared *shared)
+ : mSequence(0), mShared(shared)
+ {
+ // exactly one of Mutator and Observer must initialize, currently it is Observer
+ // shared->init();
+ }
// push new value onto state queue, overwriting previous value;
// returns a sequence number which can be used with ack()
- int32_t push(const T& value);
+ int32_t push(const T& value)
+ {
+ Shared *shared = mShared;
+ int32_t sequence = mSequence;
+ sequence++;
+ android_atomic_acquire_store(sequence, &shared->mSequence);
+ shared->mValue = value;
+ sequence++;
+ android_atomic_release_store(sequence, &shared->mSequence);
+ mSequence = sequence;
+ // consider signalling a futex here, if we know that observer is waiting
+ return sequence;
+ }
- // return true if most recent push has been observed
- bool ack();
+ // returns the status of the last state push. This may be a stale value.
+ //
+ // SSQ_PENDING, or 0, means it has not been observed
+ // SSQ_READ means it has been read
+ // SSQ_DONE means it has been acted upon, after Observer::done() is called
+ enum SSQ_STATUS ack() const
+ {
+ // in the case of SSQ_DONE, prevent any subtle data-races of subsequent reads
+ // being performed (out-of-order) before the ack read, should the caller be
+ // depending on sequentiality of reads.
+ const int32_t ack = android_atomic_acquire_load(&mShared->mAck);
+ return ack - mSequence & ~1 ? SSQ_PENDING /* seq differ */ :
+ ack & 1 ? SSQ_DONE : SSQ_READ;
+ }
// return true if a push with specified sequence number or later has been observed
- bool ack(int32_t sequence);
+ bool ack(int32_t sequence) const
+ {
+ // this relies on 2's complement rollover to detect an ancient sequence number
+ return mShared->mAck - sequence >= 0;
+ }
private:
int32_t mSequence;
@@ -71,11 +106,54 @@
class Observer {
public:
- Observer(Shared *shared);
- /*virtual*/ ~Observer() { }
+ Observer(Shared *shared)
+ : mSequence(0), mSeed(1), mShared(shared)
+ {
+ // exactly one of Mutator and Observer must initialize, currently it is Observer
+ shared->init();
+ }
// return true if value has changed
- bool poll(T& value);
+ bool poll(T& value)
+ {
+ Shared *shared = mShared;
+ int32_t before = shared->mSequence;
+ if (before == mSequence) {
+ return false;
+ }
+ for (int tries = 0; ; ) {
+ const int MAX_TRIES = 5;
+ if (before & 1) {
+ if (++tries >= MAX_TRIES) {
+ return false;
+ }
+ before = shared->mSequence;
+ } else {
+ android_memory_barrier();
+ T temp = shared->mValue;
+ int32_t after = android_atomic_release_load(&shared->mSequence);
+ if (after == before) {
+ value = temp;
+ shared->mAck = before;
+ mSequence = before; // mSequence is even after poll success
+ return true;
+ }
+ if (++tries >= MAX_TRIES) {
+ return false;
+ }
+ before = after;
+ }
+ }
+ }
+
+ // (optional) used to indicate to the Mutator that the state that has been polled
+ // has also been acted upon.
+ void done()
+ {
+ const int32_t ack = mShared->mAck + 1;
+ // ensure all previous writes have been performed.
+ android_atomic_release_store(ack, &mShared->mAck); // mSequence is odd after "done"
+ }
private:
int32_t mSequence;
diff --git a/include/media/StringArray.h b/include/media/StringArray.h
index ae47085..48d98bf 100644
--- a/include/media/StringArray.h
+++ b/include/media/StringArray.h
@@ -16,7 +16,7 @@
//
// Sortable array of strings. STL-ish, but STL-free.
-//
+//
#ifndef _LIBS_MEDIA_STRING_ARRAY_H
#define _LIBS_MEDIA_STRING_ARRAY_H
diff --git a/include/media/mediarecorder.h b/include/media/mediarecorder.h
index b0a62a7..74a6469 100644
--- a/include/media/mediarecorder.h
+++ b/include/media/mediarecorder.h
@@ -221,7 +221,6 @@
status_t setOutputFormat(int of);
status_t setVideoEncoder(int ve);
status_t setAudioEncoder(int ae);
- status_t setOutputFile(const char* path);
status_t setOutputFile(int fd, int64_t offset, int64_t length);
status_t setVideoSize(int width, int height);
status_t setVideoFrameRate(int frames_per_second);
diff --git a/include/media/nbaio/NBAIO.h b/include/media/nbaio/NBAIO.h
index d422576..d9bbc8d 100644
--- a/include/media/nbaio/NBAIO.h
+++ b/include/media/nbaio/NBAIO.h
@@ -231,7 +231,8 @@
virtual status_t getTimestamp(AudioTimestamp& timestamp) { return INVALID_OPERATION; }
protected:
- NBAIO_Sink(const NBAIO_Format& format = Format_Invalid) : NBAIO_Port(format), mFramesWritten(0) { }
+ NBAIO_Sink(const NBAIO_Format& format = Format_Invalid) : NBAIO_Port(format), mFramesWritten(0)
+ { }
virtual ~NBAIO_Sink() { }
// Implementations are free to ignore these if they don't need them
@@ -322,7 +323,8 @@
virtual void onTimestamp(const AudioTimestamp& timestamp) { }
protected:
- NBAIO_Source(const NBAIO_Format& format = Format_Invalid) : NBAIO_Port(format), mFramesRead(0) { }
+ NBAIO_Source(const NBAIO_Format& format = Format_Invalid) : NBAIO_Port(format), mFramesRead(0)
+ { }
virtual ~NBAIO_Source() { }
// Implementations are free to ignore these if they don't need them
diff --git a/include/media/nbaio/NBLog.h b/include/media/nbaio/NBLog.h
index bcbbc04..1297b51 100644
--- a/include/media/nbaio/NBLog.h
+++ b/include/media/nbaio/NBLog.h
@@ -21,7 +21,7 @@
#include <binder/IMemory.h>
#include <utils/Mutex.h>
-#include <media/nbaio/roundup.h>
+#include <audio_utils/roundup.h>
namespace android {
diff --git a/include/media/stagefright/AACWriter.h b/include/media/stagefright/AACWriter.h
index d22707a..86417a5 100644
--- a/include/media/stagefright/AACWriter.h
+++ b/include/media/stagefright/AACWriter.h
@@ -27,7 +27,6 @@
struct MetaData;
struct AACWriter : public MediaWriter {
- AACWriter(const char *filename);
AACWriter(int fd);
status_t initCheck() const;
diff --git a/include/media/stagefright/ACodec.h b/include/media/stagefright/ACodec.h
index 595ace8..442c861 100644
--- a/include/media/stagefright/ACodec.h
+++ b/include/media/stagefright/ACodec.h
@@ -214,6 +214,7 @@
int64_t mRepeatFrameDelayUs;
int64_t mMaxPtsGapUs;
+ float mMaxFps;
int64_t mTimePerFrameUs;
int64_t mTimePerCaptureUs;
diff --git a/include/media/stagefright/AMRWriter.h b/include/media/stagefright/AMRWriter.h
index 392f968..bac878b 100644
--- a/include/media/stagefright/AMRWriter.h
+++ b/include/media/stagefright/AMRWriter.h
@@ -29,7 +29,6 @@
struct MetaData;
struct AMRWriter : public MediaWriter {
- AMRWriter(const char *filename);
AMRWriter(int fd);
status_t initCheck() const;
diff --git a/include/media/stagefright/MPEG2TSWriter.h b/include/media/stagefright/MPEG2TSWriter.h
index 2e2922e..3d7960b 100644
--- a/include/media/stagefright/MPEG2TSWriter.h
+++ b/include/media/stagefright/MPEG2TSWriter.h
@@ -29,7 +29,6 @@
struct MPEG2TSWriter : public MediaWriter {
MPEG2TSWriter(int fd);
- MPEG2TSWriter(const char *filename);
MPEG2TSWriter(
void *cookie,
diff --git a/include/media/stagefright/MPEG4Writer.h b/include/media/stagefright/MPEG4Writer.h
index 26ce5f9..899b324 100644
--- a/include/media/stagefright/MPEG4Writer.h
+++ b/include/media/stagefright/MPEG4Writer.h
@@ -32,7 +32,6 @@
class MPEG4Writer : public MediaWriter {
public:
- MPEG4Writer(const char *filename);
MPEG4Writer(int fd);
// Limitations
diff --git a/include/media/stagefright/MediaFilter.h b/include/media/stagefright/MediaFilter.h
new file mode 100644
index 0000000..7b3f700
--- /dev/null
+++ b/include/media/stagefright/MediaFilter.h
@@ -0,0 +1,167 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef MEDIA_FILTER_H_
+#define MEDIA_FILTER_H_
+
+#include <media/stagefright/CodecBase.h>
+
+namespace android {
+
+struct ABuffer;
+struct GraphicBufferListener;
+struct MemoryDealer;
+struct SimpleFilter;
+
+struct MediaFilter : public CodecBase {
+ MediaFilter();
+
+ virtual void setNotificationMessage(const sp<AMessage> &msg);
+
+ virtual void initiateAllocateComponent(const sp<AMessage> &msg);
+ virtual void initiateConfigureComponent(const sp<AMessage> &msg);
+ virtual void initiateCreateInputSurface();
+ virtual void initiateStart();
+ virtual void initiateShutdown(bool keepComponentAllocated = false);
+
+ virtual void signalFlush();
+ virtual void signalResume();
+
+ virtual void signalRequestIDRFrame();
+ virtual void signalSetParameters(const sp<AMessage> &msg);
+ virtual void signalEndOfInputStream();
+
+ virtual void onMessageReceived(const sp<AMessage> &msg);
+
+ struct PortDescription : public CodecBase::PortDescription {
+ virtual size_t countBuffers();
+ virtual IOMX::buffer_id bufferIDAt(size_t index) const;
+ virtual sp<ABuffer> bufferAt(size_t index) const;
+
+ protected:
+ PortDescription();
+
+ private:
+ friend struct MediaFilter;
+
+ Vector<IOMX::buffer_id> mBufferIDs;
+ Vector<sp<ABuffer> > mBuffers;
+
+ void addBuffer(IOMX::buffer_id id, const sp<ABuffer> &buffer);
+
+ DISALLOW_EVIL_CONSTRUCTORS(PortDescription);
+ };
+
+protected:
+ virtual ~MediaFilter();
+
+private:
+ struct BufferInfo {
+ enum Status {
+ OWNED_BY_US,
+ OWNED_BY_UPSTREAM,
+ };
+
+ IOMX::buffer_id mBufferID;
+ int32_t mGeneration;
+ int32_t mOutputFlags;
+ Status mStatus;
+
+ sp<ABuffer> mData;
+ };
+
+ enum State {
+ UNINITIALIZED,
+ INITIALIZED,
+ CONFIGURED,
+ STARTED,
+ };
+
+ enum {
+ kWhatInputBufferFilled = 'inpF',
+ kWhatOutputBufferDrained = 'outD',
+ kWhatShutdown = 'shut',
+ kWhatFlush = 'flus',
+ kWhatResume = 'resm',
+ kWhatAllocateComponent = 'allo',
+ kWhatConfigureComponent = 'conf',
+ kWhatCreateInputSurface = 'cisf',
+ kWhatSignalEndOfInputStream = 'eois',
+ kWhatStart = 'star',
+ kWhatSetParameters = 'setP',
+ kWhatProcessBuffers = 'proc',
+ };
+
+ enum {
+ kPortIndexInput = 0,
+ kPortIndexOutput = 1
+ };
+
+ // member variables
+ AString mComponentName;
+ State mState;
+ status_t mInputEOSResult;
+ int32_t mWidth, mHeight;
+ int32_t mStride, mSliceHeight;
+ int32_t mColorFormatIn, mColorFormatOut;
+ size_t mMaxInputSize, mMaxOutputSize;
+ int32_t mGeneration;
+ sp<AMessage> mNotify;
+ sp<AMessage> mInputFormat;
+ sp<AMessage> mOutputFormat;
+
+ sp<MemoryDealer> mDealer[2];
+ Vector<BufferInfo> mBuffers[2];
+ Vector<BufferInfo*> mAvailableInputBuffers;
+ Vector<BufferInfo*> mAvailableOutputBuffers;
+ bool mPortEOS[2];
+
+ sp<SimpleFilter> mFilter;
+ sp<GraphicBufferListener> mGraphicBufferListener;
+
+ // helper functions
+ void signalProcessBuffers();
+ void signalError(status_t error);
+
+ status_t allocateBuffersOnPort(OMX_U32 portIndex);
+ BufferInfo *findBufferByID(
+ uint32_t portIndex, IOMX::buffer_id bufferID,
+ ssize_t *index = NULL);
+ void postFillThisBuffer(BufferInfo *info);
+ void postDrainThisBuffer(BufferInfo *info);
+ void postEOS();
+ void sendFormatChange();
+ void requestFillEmptyInput();
+ void processBuffers();
+
+ void onAllocateComponent(const sp<AMessage> &msg);
+ void onConfigureComponent(const sp<AMessage> &msg);
+ void onStart();
+ void onInputBufferFilled(const sp<AMessage> &msg);
+ void onOutputBufferDrained(const sp<AMessage> &msg);
+ void onShutdown(const sp<AMessage> &msg);
+ void onFlush();
+ void onSetParameters(const sp<AMessage> &msg);
+ void onCreateInputSurface();
+ void onInputFrameAvailable();
+ void onSignalEndOfInputStream();
+
+ DISALLOW_EVIL_CONSTRUCTORS(MediaFilter);
+};
+
+} // namespace android
+
+#endif // MEDIA_FILTER_H_
diff --git a/include/media/stagefright/MediaMuxer.h b/include/media/stagefright/MediaMuxer.h
index 9da98d9..e6538d1 100644
--- a/include/media/stagefright/MediaMuxer.h
+++ b/include/media/stagefright/MediaMuxer.h
@@ -50,9 +50,6 @@
OUTPUT_FORMAT_LIST_END // must be last - used to validate format type
};
- // Construct the muxer with the output file path.
- MediaMuxer(const char *path, OutputFormat format);
-
// Construct the muxer with the file descriptor. Note that the MediaMuxer
// will close this file at stop().
MediaMuxer(int fd, OutputFormat format);
diff --git a/include/media/stagefright/RenderScriptWrapper.h b/include/media/stagefright/RenderScriptWrapper.h
new file mode 100644
index 0000000..b42649e
--- /dev/null
+++ b/include/media/stagefright/RenderScriptWrapper.h
@@ -0,0 +1,42 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef RENDERSCRIPT_WRAPPER_H_
+#define RENDERSCRIPT_WRAPPER_H_
+
+#include <RenderScript.h>
+
+namespace android {
+
+struct RenderScriptWrapper : public RefBase {
+public:
+ struct RSFilterCallback : public RefBase {
+ public:
+ // called by RSFilter to process each input buffer
+ virtual status_t processBuffers(
+ RSC::Allocation* inBuffer,
+ RSC::Allocation* outBuffer) = 0;
+
+ virtual status_t handleSetParameters(const sp<AMessage> &msg) = 0;
+ };
+
+ sp<RSFilterCallback> mCallback;
+ RSC::sp<RSC::RS> mContext;
+};
+
+} // namespace android
+
+#endif // RENDERSCRIPT_WRAPPER_H_
diff --git a/include/ndk/NdkMediaCodec.h b/include/ndk/NdkMediaCodec.h
index c07f4c9..4f6a1ef 100644
--- a/include/ndk/NdkMediaCodec.h
+++ b/include/ndk/NdkMediaCodec.h
@@ -142,7 +142,8 @@
/**
* Get the index of the next available buffer of processed data.
*/
-ssize_t AMediaCodec_dequeueOutputBuffer(AMediaCodec*, AMediaCodecBufferInfo *info, int64_t timeoutUs);
+ssize_t AMediaCodec_dequeueOutputBuffer(AMediaCodec*, AMediaCodecBufferInfo *info,
+ int64_t timeoutUs);
AMediaFormat* AMediaCodec_getOutputFormat(AMediaCodec*);
/**
diff --git a/include/ndk/NdkMediaExtractor.h b/include/ndk/NdkMediaExtractor.h
index 7a4e702..7324d31 100644
--- a/include/ndk/NdkMediaExtractor.h
+++ b/include/ndk/NdkMediaExtractor.h
@@ -55,12 +55,14 @@
/**
* Set the file descriptor from which the extractor will read.
*/
-media_status_t AMediaExtractor_setDataSourceFd(AMediaExtractor*, int fd, off64_t offset, off64_t length);
+media_status_t AMediaExtractor_setDataSourceFd(AMediaExtractor*, int fd, off64_t offset,
+ off64_t length);
/**
* Set the URI from which the extractor will read.
*/
-media_status_t AMediaExtractor_setDataSource(AMediaExtractor*, const char *location); // TODO support headers
+media_status_t AMediaExtractor_setDataSource(AMediaExtractor*, const char *location);
+ // TODO support headers
/**
* Return the number of tracks in the previously specified media file
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 31dff36..7143f1a 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -24,9 +24,8 @@
#include <utils/threads.h>
#include <utils/Log.h>
#include <utils/RefBase.h>
-#include <media/nbaio/roundup.h>
+#include <audio_utils/roundup.h>
#include <media/SingleStateQueue.h>
-#include <private/media/StaticAudioTrackState.h>
namespace android {
@@ -61,15 +60,57 @@
volatile uint32_t mUnderrunFrames; // server increments for each unavailable but desired frame
};
+// Represents a single state of an AudioTrack that was created in static mode (shared memory buffer
+// supplied by the client). This state needs to be communicated from the client to server. As this
+// state is too large to be updated atomically without a mutex, and mutexes aren't allowed here, the
+// state is wrapped by a SingleStateQueue.
+struct StaticAudioTrackState {
+ // Do not define constructors, destructors, or virtual methods as this is part of a
+ // union in shared memory and they will not get called properly.
+
+ // These fields should both be size_t, but since they are located in shared memory we
+ // force to 32-bit. The client and server may have different typedefs for size_t.
+
+ // The state has a sequence counter to indicate whether changes are made to loop or position.
+ // The sequence counter also currently indicates whether loop or position is first depending
+ // on which is greater; it jumps by max(mLoopSequence, mPositionSequence) + 1.
+
+ uint32_t mLoopStart;
+ uint32_t mLoopEnd;
+ int32_t mLoopCount;
+ uint32_t mLoopSequence; // a sequence counter to indicate changes to loop
+ uint32_t mPosition;
+ uint32_t mPositionSequence; // a sequence counter to indicate changes to position
+};
+
typedef SingleStateQueue<StaticAudioTrackState> StaticAudioTrackSingleStateQueue;
+struct StaticAudioTrackPosLoop {
+ // Do not define constructors, destructors, or virtual methods as this is part of a
+ // union in shared memory and will not get called properly.
+
+ // These fields should both be size_t, but since they are located in shared memory we
+ // force to 32-bit. The client and server may have different typedefs for size_t.
+
+ // This struct information is stored in a single state queue to communicate the
+ // static AudioTrack server state to the client while data is consumed.
+ // It is smaller than StaticAudioTrackState to prevent unnecessary information from
+ // being sent.
+
+ uint32_t mBufferPosition;
+ int32_t mLoopCount;
+};
+
+typedef SingleStateQueue<StaticAudioTrackPosLoop> StaticAudioTrackPosLoopQueue;
+
struct AudioTrackSharedStatic {
+ // client requests to the server for loop or position changes.
StaticAudioTrackSingleStateQueue::Shared
mSingleStateQueue;
- // This field should be a size_t, but since it is located in shared memory we
- // force to 32-bit. The client and server may have different typedefs for size_t.
- uint32_t mBufferPosition; // updated asynchronously by server,
- // "for entertainment purposes only"
+ // position info updated asynchronously by server and read by client,
+ // "for entertainment purposes only"
+ StaticAudioTrackPosLoopQueue::Shared
+ mPosLoopQueue;
};
// ----------------------------------------------------------------------------
@@ -96,7 +137,8 @@
uint32_t mServer; // Number of filled frames consumed by server (mIsOut),
// or filled frames provided by server (!mIsOut).
// It is updated asynchronously by server without a barrier.
- // The value should be used "for entertainment purposes only",
+ // The value should be used
+ // "for entertainment purposes only",
// which means don't make important decisions based on it.
uint32_t mPad1; // unused
@@ -313,8 +355,28 @@
virtual void flush();
#define MIN_LOOP 16 // minimum length of each loop iteration in frames
+
+ // setLoop(), setBufferPosition(), and setBufferPositionAndLoop() set the
+ // static buffer position and looping parameters. These commands are not
+ // synchronous (they do not wait or block); instead they take effect at the
+ // next buffer data read from the server side. However, the client side
+ // getters will read a cached version of the position and loop variables
+ // until the setting takes effect.
+ //
+ // setBufferPositionAndLoop() is equivalent to calling, in order, setLoop() and
+ // setBufferPosition().
+ //
+ // The functions should not be relied upon to do parameter or state checking.
+ // That is done at the AudioTrack level.
+
void setLoop(size_t loopStart, size_t loopEnd, int loopCount);
+ void setBufferPosition(size_t position);
+ void setBufferPositionAndLoop(size_t position, size_t loopStart, size_t loopEnd,
+ int loopCount);
size_t getBufferPosition();
+ // getBufferPositionAndLoopCount() provides the proper snapshot of
+ // position and loopCount together.
+ void getBufferPositionAndLoopCount(size_t *position, int *loopCount);
virtual size_t getMisalignment() {
return 0;
@@ -326,7 +388,9 @@
private:
StaticAudioTrackSingleStateQueue::Mutator mMutator;
- size_t mBufferPosition; // so that getBufferPosition() appears to be synchronous
+ StaticAudioTrackPosLoopQueue::Observer mPosLoopObserver;
+ StaticAudioTrackState mState; // last communicated state to server
+ StaticAudioTrackPosLoop mPosLoop; // snapshot of position and loop.
};
// ----------------------------------------------------------------------------
@@ -447,10 +511,13 @@
virtual uint32_t getUnderrunFrames() const { return 0; }
private:
+ status_t updateStateWithLoop(StaticAudioTrackState *localState,
+ const StaticAudioTrackState &update) const;
+ status_t updateStateWithPosition(StaticAudioTrackState *localState,
+ const StaticAudioTrackState &update) const;
ssize_t pollPosition(); // poll for state queue update, and return current position
StaticAudioTrackSingleStateQueue::Observer mObserver;
- size_t mPosition; // server's current play position in frames, relative to 0
-
+ StaticAudioTrackPosLoopQueue::Mutator mPosLoopMutator;
size_t mFramesReadySafe; // Assuming size_t read/writes are atomic on 32 / 64 bit
// processors, this is a thread-safe version of
// mFramesReady.
@@ -459,7 +526,8 @@
// can cause a track to appear to have a large number
// of frames. INT64_MAX means an infinite loop.
bool mFramesReadyIsCalledByMultipleThreads;
- StaticAudioTrackState mState;
+ StaticAudioTrackState mState; // Server side state. Any updates from client must be
+ // passed by the mObserver SingleStateQueue.
};
// Proxy used by AudioFlinger for servicing AudioRecord
diff --git a/include/private/media/StaticAudioTrackState.h b/include/private/media/StaticAudioTrackState.h
deleted file mode 100644
index d483061..0000000
--- a/include/private/media/StaticAudioTrackState.h
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
- * Copyright (C) 2012 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef STATIC_AUDIO_TRACK_STATE_H
-#define STATIC_AUDIO_TRACK_STATE_H
-
-namespace android {
-
-// Represents a single state of an AudioTrack that was created in static mode (shared memory buffer
-// supplied by the client). This state needs to be communicated from the client to server. As this
-// state is too large to be updated atomically without a mutex, and mutexes aren't allowed here, the
-// state is wrapped by a SingleStateQueue.
-struct StaticAudioTrackState {
- // do not define constructors, destructors, or virtual methods
-
- // These fields should both be size_t, but since they are located in shared memory we
- // force to 32-bit. The client and server may have different typedefs for size_t.
- uint32_t mLoopStart;
- uint32_t mLoopEnd;
-
- int mLoopCount;
-};
-
-} // namespace android
-
-#endif // STATIC_AUDIO_TRACK_STATE_H
diff --git a/media/libeffects/factory/EffectsFactory.c b/media/libeffects/factory/EffectsFactory.c
index 6d30d64..c310fe2 100644
--- a/media/libeffects/factory/EffectsFactory.c
+++ b/media/libeffects/factory/EffectsFactory.c
@@ -28,6 +28,7 @@
static list_elem_t *gEffectList; // list of effect_entry_t: all currently created effects
static list_elem_t *gLibraryList; // list of lib_entry_t: all currently loaded libraries
+static list_elem_t *gSkippedEffects; // list of effects skipped because of duplicate uuid
// list of effect_descriptor and list of sub effects : all currently loaded
// It does not contain effects without sub effects.
static list_sub_elem_t *gSubEffectList;
@@ -63,10 +64,10 @@
lib_entry_t **lib,
effect_descriptor_t **desc);
// To search a subeffect in the gSubEffectList
-int findSubEffect(const effect_uuid_t *uuid,
+static int findSubEffect(const effect_uuid_t *uuid,
lib_entry_t **lib,
effect_descriptor_t **desc);
-static void dumpEffectDescriptor(effect_descriptor_t *desc, char *str, size_t len);
+static void dumpEffectDescriptor(effect_descriptor_t *desc, char *str, size_t len, int indent);
static int stringToUuid(const char *str, effect_uuid_t *uuid);
static int uuidToString(const effect_uuid_t *uuid, char *str, size_t maxLen);
@@ -237,8 +238,8 @@
}
#if (LOG_NDEBUG == 0)
- char str[256];
- dumpEffectDescriptor(pDescriptor, str, 256);
+ char str[512];
+ dumpEffectDescriptor(pDescriptor, str, sizeof(str), 0 /* indent */);
ALOGV("EffectQueryEffect() desc:%s", str);
#endif
pthread_mutex_unlock(&gLibLock);
@@ -503,15 +504,31 @@
audio_effect_library_t *desc;
list_elem_t *e;
lib_entry_t *l;
+ char path[PATH_MAX];
+ char *str;
+ size_t len;
node = config_find(root, PATH_TAG);
if (node == NULL) {
return -EINVAL;
}
+ // audio_effects.conf always specifies 32 bit lib path: convert to 64 bit path if needed
+ strlcpy(path, node->value, PATH_MAX);
+#ifdef __LP64__
+ str = strstr(path, "/lib/");
+ if (str == NULL)
+ return -EINVAL;
+ len = str - path;
+ path[len] = '\0';
+ strlcat(path, "/lib64/", PATH_MAX);
+ strlcat(path, node->value + len + strlen("/lib/"), PATH_MAX);
+#endif
+ if (strlen(path) >= PATH_MAX - 1)
+ return -EINVAL;
- hdl = dlopen(node->value, RTLD_NOW);
+ hdl = dlopen(path, RTLD_NOW);
if (hdl == NULL) {
- ALOGW("loadLibrary() failed to open %s", node->value);
+ ALOGW("loadLibrary() failed to open %s", path);
goto error;
}
@@ -535,7 +552,7 @@
// add entry for library in gLibraryList
l = malloc(sizeof(lib_entry_t));
l->name = strndup(name, PATH_MAX);
- l->path = strndup(node->value, PATH_MAX);
+ l->path = strndup(path, PATH_MAX);
l->handle = hdl;
l->desc = desc;
l->effects = NULL;
@@ -547,7 +564,7 @@
e->next = gLibraryList;
gLibraryList = e;
pthread_mutex_unlock(&gLibLock);
- ALOGV("getLibrary() linked library %p for path %s", l, node->value);
+ ALOGV("getLibrary() linked library %p for path %s", l, path);
return 0;
@@ -595,8 +612,8 @@
return -EINVAL;
}
#if (LOG_NDEBUG==0)
- char s[256];
- dumpEffectDescriptor(d, s, 256);
+ char s[512];
+ dumpEffectDescriptor(d, s, sizeof(s), 0 /* indent */);
ALOGV("addSubEffect() read descriptor %p:%s",d, s);
#endif
if (EFFECT_API_VERSION_MAJOR(d->apiVersion) !=
@@ -660,6 +677,13 @@
ALOGW("loadEffect() invalid uuid %s", node->value);
return -EINVAL;
}
+ lib_entry_t *tmp;
+ bool skip = false;
+ if (findEffect(NULL, &uuid, &tmp, NULL) == 0) {
+ ALOGW("skipping duplicate uuid %s %s", node->value,
+ node->next ? "and its sub-effects" : "");
+ skip = true;
+ }
d = malloc(sizeof(effect_descriptor_t));
if (l->desc->get_descriptor(&uuid, d) != 0) {
@@ -670,8 +694,8 @@
return -EINVAL;
}
#if (LOG_NDEBUG==0)
- char s[256];
- dumpEffectDescriptor(d, s, 256);
+ char s[512];
+ dumpEffectDescriptor(d, s, sizeof(s), 0 /* indent */);
ALOGV("loadEffect() read descriptor %p:%s",d, s);
#endif
if (EFFECT_API_VERSION_MAJOR(d->apiVersion) !=
@@ -682,8 +706,14 @@
}
e = malloc(sizeof(list_elem_t));
e->object = d;
- e->next = l->effects;
- l->effects = e;
+ if (skip) {
+ e->next = gSkippedEffects;
+ gSkippedEffects = e;
+ return -EINVAL;
+ } else {
+ e->next = l->effects;
+ l->effects = e;
+ }
// After the UUID node in the config_tree, if node->next is valid,
// that would be sub effect node.
@@ -876,22 +906,30 @@
return ret;
}
-void dumpEffectDescriptor(effect_descriptor_t *desc, char *str, size_t len) {
+void dumpEffectDescriptor(effect_descriptor_t *desc, char *str, size_t len, int indent) {
char s[256];
+ char ss[256];
+ char idt[indent + 1];
- snprintf(str, len, "\nEffect Descriptor %p:\n", desc);
- strncat(str, "- TYPE: ", len);
- uuidToString(&desc->uuid, s, 256);
- snprintf(str, len, "- UUID: %s\n", s);
- uuidToString(&desc->type, s, 256);
- snprintf(str, len, "- TYPE: %s\n", s);
- sprintf(s, "- apiVersion: %08X\n- flags: %08X\n",
- desc->apiVersion, desc->flags);
- strncat(str, s, len);
- sprintf(s, "- name: %s\n", desc->name);
- strncat(str, s, len);
- sprintf(s, "- implementor: %s\n", desc->implementor);
- strncat(str, s, len);
+ memset(idt, ' ', indent);
+ idt[indent] = 0;
+
+ str[0] = 0;
+
+ snprintf(s, sizeof(s), "%s%s / %s\n", idt, desc->name, desc->implementor);
+ strlcat(str, s, len);
+
+ uuidToString(&desc->uuid, s, sizeof(s));
+ snprintf(ss, sizeof(ss), "%s UUID: %s\n", idt, s);
+ strlcat(str, ss, len);
+
+ uuidToString(&desc->type, s, sizeof(s));
+ snprintf(ss, sizeof(ss), "%s TYPE: %s\n", idt, s);
+ strlcat(str, ss, len);
+
+ sprintf(s, "%s apiVersion: %08X\n%s flags: %08X\n", idt,
+ desc->apiVersion, idt, desc->flags);
+ strlcat(str, s, len);
}
int stringToUuid(const char *str, effect_uuid_t *uuid)
@@ -934,3 +972,40 @@
return 0;
}
+int EffectDumpEffects(int fd) {
+ char s[512];
+ list_elem_t *e = gLibraryList;
+ lib_entry_t *l = NULL;
+ effect_descriptor_t *d = NULL;
+ int found = 0;
+ int ret = 0;
+
+ while (e) {
+ l = (lib_entry_t *)e->object;
+ list_elem_t *efx = l->effects;
+ dprintf(fd, "Library %s\n", l->name);
+ if (!efx) {
+ dprintf(fd, " (no effects)\n");
+ }
+ while (efx) {
+ d = (effect_descriptor_t *)efx->object;
+ dumpEffectDescriptor(d, s, sizeof(s), 2);
+ dprintf(fd, "%s", s);
+ efx = efx->next;
+ }
+ e = e->next;
+ }
+
+ e = gSkippedEffects;
+ if (e) {
+ dprintf(fd, "Skipped effects\n");
+ while(e) {
+ d = (effect_descriptor_t *)e->object;
+ dumpEffectDescriptor(d, s, sizeof(s), 2 /* indent */);
+ dprintf(fd, "%s", s);
+ e = e->next;
+ }
+ }
+ return ret;
+}
+
diff --git a/media/libmedia/Android.mk b/media/libmedia/Android.mk
index 6c585fb..5378bf2 100644
--- a/media/libmedia/Android.mk
+++ b/media/libmedia/Android.mk
@@ -61,15 +61,11 @@
StringArray.cpp \
AudioPolicy.cpp
-LOCAL_SRC_FILES += ../libnbaio/roundup.c
-
LOCAL_SHARED_LIBRARIES := \
libui liblog libcutils libutils libbinder libsonivox libicuuc libicui18n libexpat \
libcamera_client libstagefright_foundation \
libgui libdl libaudioutils libnbaio
-LOCAL_STATIC_LIBRARIES += libinstantssq
-
LOCAL_WHOLE_STATIC_LIBRARIES := libmedia_helper
LOCAL_MODULE:= libmedia
@@ -85,12 +81,3 @@
include $(BUILD_SHARED_LIBRARY)
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES += SingleStateQueue.cpp
-LOCAL_CFLAGS += -DSINGLE_STATE_QUEUE_INSTANTIATIONS='"SingleStateQueueInstantiations.cpp"'
-
-LOCAL_MODULE := libinstantssq
-LOCAL_MODULE_TAGS := optional
-
-include $(BUILD_STATIC_LIBRARY)
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 9cae21c..f5a5712 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -499,8 +499,8 @@
OutputDescriptor *outputDesc = new OutputDescriptor(*desc);
gOutputs.add(ioHandle, outputDesc);
- ALOGV("ioConfigChanged() new output samplingRate %u, format %#x channel mask %#x frameCount %zu "
- "latency %d",
+ ALOGV("ioConfigChanged() new output samplingRate %u, format %#x channel mask %#x "
+ "frameCount %zu latency %d",
outputDesc->samplingRate, outputDesc->format, outputDesc->channelMask,
outputDesc->frameCount, outputDesc->latency);
} break;
@@ -523,8 +523,8 @@
if (param2 == NULL) break;
desc = (const OutputDescriptor *)param2;
- ALOGV("ioConfigChanged() new config for output %d samplingRate %u, format %#x channel mask %#x "
- "frameCount %zu latency %d",
+ ALOGV("ioConfigChanged() new config for output %d samplingRate %u, format %#x "
+ "channel mask %#x frameCount %zu latency %d",
ioHandle, desc->samplingRate, desc->format,
desc->channelMask, desc->frameCount, desc->latency);
OutputDescriptor *outputDesc = gOutputs.valueAt(index);
@@ -590,18 +590,22 @@
status_t AudioSystem::setDeviceConnectionState(audio_devices_t device,
audio_policy_dev_state_t state,
- const char *device_address)
+ const char *device_address,
+ const char *device_name)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
const char *address = "";
+ const char *name = "";
if (aps == 0) return PERMISSION_DENIED;
if (device_address != NULL) {
address = device_address;
}
-
- return aps->setDeviceConnectionState(device, state, address);
+ if (device_name != NULL) {
+ name = device_name;
+ }
+ return aps->setDeviceConnectionState(device, state, address, name);
}
audio_policy_dev_state_t AudioSystem::getDeviceConnectionState(audio_devices_t device,
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 735db5c..d4bacc0 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -33,11 +33,16 @@
#define WAIT_PERIOD_MS 10
#define WAIT_STREAM_END_TIMEOUT_SEC 120
-
+static const int kMaxLoopCountNotifications = 32;
namespace android {
// ---------------------------------------------------------------------------
+template <typename T>
+const T &min(const T &x, const T &y) {
+ return x < y ? x : y;
+}
+
static int64_t convertTimespecToUs(const struct timespec &tv)
{
return tv.tv_sec * 1000000ll + tv.tv_nsec / 1000;
@@ -317,12 +322,6 @@
uint32_t channelCount = audio_channel_count_from_out_mask(channelMask);
mChannelCount = channelCount;
- // AudioFlinger does not currently support 8-bit data in shared memory
- if (format == AUDIO_FORMAT_PCM_8_BIT && sharedBuffer != 0) {
- ALOGE("8-bit data in shared memory is not supported");
- return BAD_VALUE;
- }
-
// force direct flag if format is not linear PCM
// or offload was requested
if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
@@ -346,12 +345,9 @@
} else {
mFrameSize = sizeof(uint8_t);
}
- mFrameSizeAF = mFrameSize;
} else {
ALOG_ASSERT(audio_is_linear_pcm(format));
mFrameSize = channelCount * audio_bytes_per_sample(format);
- mFrameSizeAF = channelCount * audio_bytes_per_sample(
- format == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : format);
// createTrack will return an error if PCM format is not supported by server,
// so no need to check for specific PCM formats here
}
@@ -420,7 +416,10 @@
mStatus = NO_ERROR;
mState = STATE_STOPPED;
mUserData = user;
- mLoopPeriod = 0;
+ mLoopCount = 0;
+ mLoopStart = 0;
+ mLoopEnd = 0;
+ mLoopCountNotified = 0;
mMarkerPosition = 0;
mMarkerReached = false;
mNewPosition = 0;
@@ -531,14 +530,12 @@
// the playback head position will reset to 0, so if a marker is set, we need
// to activate it again
mMarkerReached = false;
-#if 0
- // Force flush if a shared buffer is used otherwise audioflinger
- // will not stop before end of buffer is reached.
- // It may be needed to make sure that we stop playback, likely in case looping is on.
+
if (mSharedBuffer != 0) {
- flush_l();
+ // clear buffer position and loop count.
+ mStaticProxy->setBufferPositionAndLoop(0 /* position */,
+ 0 /* loopStart */, 0 /* loopEnd */, 0 /* loopCount */);
}
-#endif
sp<AudioTrackThread> t = mAudioTrackThread;
if (t != 0) {
@@ -740,10 +737,15 @@
void AudioTrack::setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount)
{
- // Setting the loop will reset next notification update period (like setPosition).
- mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
- mLoopPeriod = loopCount != 0 ? loopEnd - loopStart : 0;
+ // We do not update the periodic notification point.
+ // mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
+ mLoopCount = loopCount;
+ mLoopEnd = loopEnd;
+ mLoopStart = loopStart;
+ mLoopCountNotified = loopCount;
mStaticProxy->setLoop(loopStart, loopEnd, loopCount);
+
+ // Waking the AudioTrackThread is not needed as this cannot be called when active.
}
status_t AudioTrack::setMarkerPosition(uint32_t marker)
@@ -757,6 +759,10 @@
mMarkerPosition = marker;
mMarkerReached = false;
+ sp<AudioTrackThread> t = mAudioTrackThread;
+ if (t != 0) {
+ t->wake();
+ }
return NO_ERROR;
}
@@ -786,6 +792,10 @@
mNewPosition = updateAndGetPosition_l() + updatePeriod;
mUpdatePeriod = updatePeriod;
+ sp<AudioTrackThread> t = mAudioTrackThread;
+ if (t != 0) {
+ t->wake();
+ }
return NO_ERROR;
}
@@ -823,12 +833,11 @@
if (mState == STATE_ACTIVE) {
return INVALID_OPERATION;
}
+ // After setting the position, use full update period before notification.
mNewPosition = updateAndGetPosition_l() + mUpdatePeriod;
- mLoopPeriod = 0;
- // FIXME Check whether loops and setting position are incompatible in old code.
- // If we use setLoop for both purposes we lose the capability to set the position while looping.
- mStaticProxy->setLoop(position, mFrameCount, 0);
+ mStaticProxy->setBufferPosition(position);
+ // Waking the AudioTrackThread is not needed as this cannot be called when active.
return NO_ERROR;
}
@@ -893,10 +902,18 @@
return INVALID_OPERATION;
}
mNewPosition = mUpdatePeriod;
- mLoopPeriod = 0;
- // FIXME The new code cannot reload while keeping a loop specified.
- // Need to check how the old code handled this, and whether it's a significant change.
- mStaticProxy->setLoop(0, mFrameCount, 0);
+ (void) updateAndGetPosition_l();
+ mPosition = 0;
+#if 0
+ // The documentation is not clear on the behavior of reload() and the restoration
+ // of loop count. Historically we have not restored loop count, start, end,
+ // but it makes sense if one desires to repeat playing a particular sound.
+ if (mLoopCount != 0) {
+ mLoopCountNotified = mLoopCount;
+ mStaticProxy->setLoop(mLoopStart, mLoopEnd, mLoopCount);
+ }
+#endif
+ mStaticProxy->setBufferPosition(0);
return NO_ERROR;
}
@@ -1019,12 +1036,12 @@
mNotificationFramesAct = frameCount;
}
} else if (mSharedBuffer != 0) {
-
- // Ensure that buffer alignment matches channel count
- // 8-bit data in shared memory is not currently supported by AudioFlinger
- size_t alignment = audio_bytes_per_sample(
- mFormat == AUDIO_FORMAT_PCM_8_BIT ? AUDIO_FORMAT_PCM_16_BIT : mFormat);
+ // FIXME: Ensure client side memory buffers need
+ // not have additional alignment beyond sample
+ // (e.g. 16 bit stereo accessed as 32 bit frame).
+ size_t alignment = audio_bytes_per_sample(mFormat);
if (alignment & 1) {
+ // for AUDIO_FORMAT_PCM_24_BIT_PACKED (not exposed through Java).
alignment = 1;
}
if (mChannelCount > 1) {
@@ -1042,7 +1059,7 @@
// there's no frameCount parameter.
// But when initializing a shared buffer AudioTrack via set(),
// there _is_ a frameCount parameter. We silently ignore it.
- frameCount = mSharedBuffer->size() / mFrameSizeAF;
+ frameCount = mSharedBuffer->size() / mFrameSize;
} else if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) {
@@ -1103,10 +1120,7 @@
// but we will still need the original value also
sp<IAudioTrack> track = audioFlinger->createTrack(streamType,
mSampleRate,
- // AudioFlinger only sees 16-bit PCM
- mFormat == AUDIO_FORMAT_PCM_8_BIT &&
- !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT) ?
- AUDIO_FORMAT_PCM_16_BIT : mFormat,
+ mFormat,
mChannelMask,
&temp,
&trackFlags,
@@ -1230,9 +1244,9 @@
// update proxy
if (mSharedBuffer == 0) {
mStaticProxy.clear();
- mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF);
+ mProxy = new AudioTrackClientProxy(cblk, buffers, frameCount, mFrameSize);
} else {
- mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSizeAF);
+ mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, frameCount, mFrameSize);
mProxy = mStaticProxy;
}
@@ -1352,7 +1366,7 @@
} while ((status == DEAD_OBJECT) && (tryCounter-- > 0));
audioBuffer->frameCount = buffer.mFrameCount;
- audioBuffer->size = buffer.mFrameCount * mFrameSizeAF;
+ audioBuffer->size = buffer.mFrameCount * mFrameSize;
audioBuffer->raw = buffer.mRaw;
if (nonContig != NULL) {
*nonContig = buffer.mNonContig;
@@ -1366,7 +1380,7 @@
return;
}
- size_t stepCount = audioBuffer->size / mFrameSizeAF;
+ size_t stepCount = audioBuffer->size / mFrameSize;
if (stepCount == 0) {
return;
}
@@ -1432,14 +1446,8 @@
}
size_t toWrite;
- if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
- // Divide capacity by 2 to take expansion into account
- toWrite = audioBuffer.size >> 1;
- memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite);
- } else {
- toWrite = audioBuffer.size;
- memcpy(audioBuffer.i8, buffer, toWrite);
- }
+ toWrite = audioBuffer.size;
+ memcpy(audioBuffer.i8, buffer, toWrite);
buffer = ((const char *) buffer) + toWrite;
userSize -= toWrite;
written += toWrite;
@@ -1559,9 +1567,8 @@
// that the upper layers can recreate the track
if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) {
status_t status = restoreTrack_l("processAudioBuffer");
- mLock.unlock();
- // Run again immediately, but with a new IAudioTrack
- return 0;
+ // after restoration, continue below to make sure that the loop and buffer events
+ // are notified because they have been cleared from mCblk->mFlags above.
}
}
@@ -1610,7 +1617,6 @@
}
// Cache other fields that will be needed soon
- uint32_t loopPeriod = mLoopPeriod;
uint32_t sampleRate = mSampleRate;
uint32_t notificationFrames = mNotificationFramesAct;
if (mRefreshRemaining) {
@@ -1622,8 +1628,30 @@
uint32_t sequence = mSequence;
sp<AudioTrackClientProxy> proxy = mProxy;
+ // Determine the number of new loop callback(s) that will be needed, while locked.
+ int loopCountNotifications = 0;
+ uint32_t loopPeriod = 0; // time in frames for next EVENT_LOOP_END or EVENT_BUFFER_END
+
+ if (mLoopCount > 0) {
+ int loopCount;
+ size_t bufferPosition;
+ mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
+ loopPeriod = ((loopCount > 0) ? mLoopEnd : mFrameCount) - bufferPosition;
+ loopCountNotifications = min(mLoopCountNotified - loopCount, kMaxLoopCountNotifications);
+ mLoopCountNotified = loopCount; // discard any excess notifications
+ } else if (mLoopCount < 0) {
+ // FIXME: We're not accurate with notification count and position with infinite looping
+ // since loopCount from server side will always return -1 (we could decrement it).
+ size_t bufferPosition = mStaticProxy->getBufferPosition();
+ loopCountNotifications = int((flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) != 0);
+ loopPeriod = mLoopEnd - bufferPosition;
+ } else if (/* mLoopCount == 0 && */ mSharedBuffer != 0) {
+ size_t bufferPosition = mStaticProxy->getBufferPosition();
+ loopPeriod = mFrameCount - bufferPosition;
+ }
+
// These fields don't need to be cached, because they are assigned only by set():
- // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFrameSizeAF, mFlags
+ // mTransfer, mCbf, mUserData, mFormat, mFrameSize, mFlags
// mFlags is also assigned by createTrack_l(), but not the bit we care about.
mLock.unlock();
@@ -1662,10 +1690,9 @@
if (newUnderrun) {
mCbf(EVENT_UNDERRUN, mUserData, NULL);
}
- // FIXME we will miss loops if loop cycle was signaled several times since last call
- // to processAudioBuffer()
- if (flags & (CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL)) {
+ while (loopCountNotifications > 0) {
mCbf(EVENT_LOOP_END, mUserData, NULL);
+ --loopCountNotifications;
}
if (flags & CBLK_BUFFER_END) {
mCbf(EVENT_BUFFER_END, mUserData, NULL);
@@ -1701,10 +1728,11 @@
minFrames = markerPosition - position;
}
if (loopPeriod > 0 && loopPeriod < minFrames) {
+ // loopPeriod is already adjusted for actual position.
minFrames = loopPeriod;
}
- if (updatePeriod > 0 && updatePeriod < minFrames) {
- minFrames = updatePeriod;
+ if (updatePeriod > 0) {
+ minFrames = min(minFrames, uint32_t(newPosition - position));
}
// If > 0, poll periodically to recover from a stuck server. A good value is 2.
@@ -1767,13 +1795,6 @@
}
}
- // Divide buffer size by 2 to take into account the expansion
- // due to 8 to 16 bit conversion: the callback must fill only half
- // of the destination buffer
- if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
- audioBuffer.size >>= 1;
- }
-
size_t reqSize = audioBuffer.size;
mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
size_t writtenSize = audioBuffer.size;
@@ -1793,13 +1814,7 @@
return WAIT_PERIOD_MS * 1000000LL;
}
- if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
- // 8 to 16 bit conversion, note that source and destination are the same address
- memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize);
- audioBuffer.size <<= 1;
- }
-
- size_t releasedFrames = audioBuffer.size / mFrameSizeAF;
+ size_t releasedFrames = audioBuffer.size / mFrameSize;
audioBuffer.frameCount = releasedFrames;
mRemainingFrames -= releasedFrames;
if (misalignment >= releasedFrames) {
@@ -1856,7 +1871,11 @@
}
// save the old static buffer position
- size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0;
+ size_t bufferPosition = 0;
+ int loopCount = 0;
+ if (mStaticProxy != 0) {
+ mStaticProxy->getBufferPositionAndLoopCount(&bufferPosition, &loopCount);
+ }
// If a new IAudioTrack is successfully created, createTrack_l() will modify the
// following member variables: mAudioTrack, mCblkMemory and mCblk.
@@ -1865,30 +1884,26 @@
result = createTrack_l();
// take the frames that will be lost by track recreation into account in saved position
+ // For streaming tracks, this is the amount we obtained from the user/client
+ // (not the number actually consumed at the server - those are already lost).
(void) updateAndGetPosition_l();
- mPosition = mReleased;
+ if (mStaticProxy != 0) {
+ mPosition = mReleased;
+ }
if (result == NO_ERROR) {
- // continue playback from last known position, but
- // don't attempt to restore loop after invalidation; it's difficult and not worthwhile
- if (mStaticProxy != NULL) {
- mLoopPeriod = 0;
- mStaticProxy->setLoop(bufferPosition, mFrameCount, 0);
- }
- // FIXME How do we simulate the fact that all frames present in the buffer at the time of
- // track destruction have been played? This is critical for SoundPool implementation
- // This must be broken, and needs to be tested/debugged.
-#if 0
- // restore write index and set other indexes to reflect empty buffer status
- if (!strcmp(from, "start")) {
- // Make sure that a client relying on callback events indicating underrun or
- // the actual amount of audio frames played (e.g SoundPool) receives them.
- if (mSharedBuffer == 0) {
- // restart playback even if buffer is not completely filled.
- android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags);
+ // Continue playback from last known position and restore loop.
+ if (mStaticProxy != 0) {
+ if (loopCount != 0) {
+ mStaticProxy->setBufferPositionAndLoop(bufferPosition,
+ mLoopStart, mLoopEnd, loopCount);
+ } else {
+ mStaticProxy->setBufferPosition(bufferPosition);
+ if (bufferPosition == mFrameCount) {
+ ALOGD("restoring track at end of static buffer");
+ }
}
}
-#endif
if (mState == STATE_ACTIVE) {
result = mAudioTrack->start();
}
@@ -2148,8 +2163,8 @@
case NS_NEVER:
return false;
case NS_WHENEVER:
- // FIXME increase poll interval, or make event-driven
- ns = 1000000000LL;
+ // Event driven: call wake() when callback notifications conditions change.
+ ns = INT64_MAX;
// fall through
default:
LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns);
@@ -2182,6 +2197,17 @@
}
}
+void AudioTrack::AudioTrackThread::wake()
+{
+ AutoMutex _l(mMyLock);
+ if (!mPaused && mPausedInt && mPausedNs > 0) {
+ // audio track is active and internally paused with timeout.
+ mIgnoreNextPausedInt = true;
+ mPausedInt = false;
+ mMyCond.signal();
+ }
+}
+
void AudioTrack::AudioTrackThread::pauseInternal(nsecs_t ns)
{
AutoMutex _l(mMyLock);
diff --git a/media/libmedia/AudioTrackShared.cpp b/media/libmedia/AudioTrackShared.cpp
index ff24475..08241e2 100644
--- a/media/libmedia/AudioTrackShared.cpp
+++ b/media/libmedia/AudioTrackShared.cpp
@@ -28,7 +28,21 @@
// used to clamp a value to size_t. TODO: move to another file.
template <typename T>
size_t clampToSize(T x) {
- return x > SIZE_MAX ? SIZE_MAX : x < 0 ? 0 : (size_t) x;
+ return sizeof(T) > sizeof(size_t) && x > (T) SIZE_MAX ? SIZE_MAX : x < 0 ? 0 : (size_t) x;
+}
+
+// incrementSequence is used to determine the next sequence value
+// for the loop and position sequence counters. It should return
+// a value between "other" + 1 and "other" + INT32_MAX, the choice of
+// which needs to be the "least recently used" sequence value for "self".
+// In general, this means (new_self) returned is max(self, other) + 1.
+
+static uint32_t incrementSequence(uint32_t self, uint32_t other) {
+ int32_t diff = self - other;
+ if (diff >= 0 && diff < INT32_MAX) {
+ return self + 1; // we're already ahead of other.
+ }
+ return other + 1; // we're behind, so move just ahead of other.
}
audio_track_cblk_t::audio_track_cblk_t()
@@ -485,8 +499,11 @@
StaticAudioTrackClientProxy::StaticAudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers,
size_t frameCount, size_t frameSize)
: AudioTrackClientProxy(cblk, buffers, frameCount, frameSize),
- mMutator(&cblk->u.mStatic.mSingleStateQueue), mBufferPosition(0)
+ mMutator(&cblk->u.mStatic.mSingleStateQueue),
+ mPosLoopObserver(&cblk->u.mStatic.mPosLoopQueue)
{
+ memset(&mState, 0, sizeof(mState));
+ memset(&mPosLoop, 0, sizeof(mPosLoop));
}
void StaticAudioTrackClientProxy::flush()
@@ -501,30 +518,72 @@
// FIXME Should return an error status
return;
}
- StaticAudioTrackState newState;
- newState.mLoopStart = (uint32_t) loopStart;
- newState.mLoopEnd = (uint32_t) loopEnd;
- newState.mLoopCount = loopCount;
- size_t bufferPosition;
- if (loopCount == 0 || (bufferPosition = getBufferPosition()) >= loopEnd) {
- bufferPosition = loopStart;
+ mState.mLoopStart = (uint32_t) loopStart;
+ mState.mLoopEnd = (uint32_t) loopEnd;
+ mState.mLoopCount = loopCount;
+ mState.mLoopSequence = incrementSequence(mState.mLoopSequence, mState.mPositionSequence);
+ // set patch-up variables until the mState is acknowledged by the ServerProxy.
+ // observed buffer position and loop count will freeze until then to give the
+ // illusion of a synchronous change.
+ getBufferPositionAndLoopCount(NULL, NULL);
+ // preserve behavior to restart at mState.mLoopStart if position exceeds mState.mLoopEnd.
+ if (mState.mLoopCount != 0 && mPosLoop.mBufferPosition >= mState.mLoopEnd) {
+ mPosLoop.mBufferPosition = mState.mLoopStart;
}
- mBufferPosition = bufferPosition; // snapshot buffer position until loop is acknowledged.
- (void) mMutator.push(newState);
+ mPosLoop.mLoopCount = mState.mLoopCount;
+ (void) mMutator.push(mState);
+}
+
+void StaticAudioTrackClientProxy::setBufferPosition(size_t position)
+{
+ // This can only happen on a 64-bit client
+ if (position > UINT32_MAX) {
+ // FIXME Should return an error status
+ return;
+ }
+ mState.mPosition = (uint32_t) position;
+ mState.mPositionSequence = incrementSequence(mState.mPositionSequence, mState.mLoopSequence);
+ // set patch-up variables until the mState is acknowledged by the ServerProxy.
+ // observed buffer position and loop count will freeze until then to give the
+ // illusion of a synchronous change.
+ if (mState.mLoopCount > 0) { // only check if loop count is changing
+ getBufferPositionAndLoopCount(NULL, NULL); // get last position
+ }
+ mPosLoop.mBufferPosition = position;
+ if (position >= mState.mLoopEnd) {
+ // no ongoing loop is possible if position is greater than loopEnd.
+ mPosLoop.mLoopCount = 0;
+ }
+ (void) mMutator.push(mState);
+}
+
+void StaticAudioTrackClientProxy::setBufferPositionAndLoop(size_t position, size_t loopStart,
+ size_t loopEnd, int loopCount)
+{
+ setLoop(loopStart, loopEnd, loopCount);
+ setBufferPosition(position);
}
size_t StaticAudioTrackClientProxy::getBufferPosition()
{
- size_t bufferPosition;
- if (mMutator.ack()) {
- bufferPosition = (size_t) mCblk->u.mStatic.mBufferPosition;
- if (bufferPosition > mFrameCount) {
- bufferPosition = mFrameCount;
- }
- } else {
- bufferPosition = mBufferPosition;
+ getBufferPositionAndLoopCount(NULL, NULL);
+ return mPosLoop.mBufferPosition;
+}
+
+void StaticAudioTrackClientProxy::getBufferPositionAndLoopCount(
+ size_t *position, int *loopCount)
+{
+ if (mMutator.ack() == StaticAudioTrackSingleStateQueue::SSQ_DONE) {
+ if (mPosLoopObserver.poll(mPosLoop)) {
+ ; // a valid mPosLoop should be available if ackDone is true.
+ }
}
- return bufferPosition;
+ if (position != NULL) {
+ *position = mPosLoop.mBufferPosition;
+ }
+ if (loopCount != NULL) {
+ *loopCount = mPosLoop.mLoopCount;
+ }
}
// ---------------------------------------------------------------------------
@@ -560,7 +619,8 @@
ssize_t filled = rear - newFront;
// Rather than shutting down on a corrupt flush, just treat it as a full flush
if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
- ALOGE("mFlush %#x -> %#x, front %#x, rear %#x, mask %#x, newFront %#x, filled %d=%#x",
+ ALOGE("mFlush %#x -> %#x, front %#x, rear %#x, mask %#x, newFront %#x, "
+ "filled %d=%#x",
mFlush, flush, front, rear, mask, newFront, filled, filled);
newFront = rear;
}
@@ -739,13 +799,12 @@
StaticAudioTrackServerProxy::StaticAudioTrackServerProxy(audio_track_cblk_t* cblk, void *buffers,
size_t frameCount, size_t frameSize)
: AudioTrackServerProxy(cblk, buffers, frameCount, frameSize),
- mObserver(&cblk->u.mStatic.mSingleStateQueue), mPosition(0),
+ mObserver(&cblk->u.mStatic.mSingleStateQueue),
+ mPosLoopMutator(&cblk->u.mStatic.mPosLoopQueue),
mFramesReadySafe(frameCount), mFramesReady(frameCount),
mFramesReadyIsCalledByMultipleThreads(false)
{
- mState.mLoopStart = 0;
- mState.mLoopEnd = 0;
- mState.mLoopCount = 0;
+ memset(&mState, 0, sizeof(mState));
}
void StaticAudioTrackServerProxy::framesReadyIsCalledByMultipleThreads()
@@ -762,55 +821,97 @@
return mFramesReadySafe;
}
-ssize_t StaticAudioTrackServerProxy::pollPosition()
+status_t StaticAudioTrackServerProxy::updateStateWithLoop(
+ StaticAudioTrackState *localState, const StaticAudioTrackState &update) const
{
- size_t position = mPosition;
- StaticAudioTrackState state;
- if (mObserver.poll(state)) {
+ if (localState->mLoopSequence != update.mLoopSequence) {
bool valid = false;
- size_t loopStart = state.mLoopStart;
- size_t loopEnd = state.mLoopEnd;
- if (state.mLoopCount == 0) {
- if (loopStart > mFrameCount) {
- loopStart = mFrameCount;
- }
- // ignore loopEnd
- mPosition = position = loopStart;
- mFramesReady = mFrameCount - mPosition;
- mState.mLoopCount = 0;
+ const size_t loopStart = update.mLoopStart;
+ const size_t loopEnd = update.mLoopEnd;
+ size_t position = localState->mPosition;
+ if (update.mLoopCount == 0) {
valid = true;
- } else if (state.mLoopCount >= -1) {
+ } else if (update.mLoopCount >= -1) {
if (loopStart < loopEnd && loopEnd <= mFrameCount &&
loopEnd - loopStart >= MIN_LOOP) {
// If the current position is greater than the end of the loop
// we "wrap" to the loop start. This might cause an audible pop.
if (position >= loopEnd) {
- mPosition = position = loopStart;
+ position = loopStart;
}
- if (state.mLoopCount == -1) {
- mFramesReady = INT64_MAX;
- } else {
- // mFramesReady is 64 bits to handle the effective number of frames
- // that the static audio track contains, including loops.
- // TODO: Later consider fixing overflow, but does not seem needed now
- // as will not overflow if loopStart and loopEnd are Java "ints".
- mFramesReady = int64_t(state.mLoopCount) * (loopEnd - loopStart)
- + mFrameCount - mPosition;
- }
- mState = state;
valid = true;
}
}
- if (!valid || mPosition > mFrameCount) {
+ if (!valid || position > mFrameCount) {
+ return NO_INIT;
+ }
+ localState->mPosition = position;
+ localState->mLoopCount = update.mLoopCount;
+ localState->mLoopEnd = loopEnd;
+ localState->mLoopStart = loopStart;
+ localState->mLoopSequence = update.mLoopSequence;
+ }
+ return OK;
+}
+
+status_t StaticAudioTrackServerProxy::updateStateWithPosition(
+ StaticAudioTrackState *localState, const StaticAudioTrackState &update) const
+{
+ if (localState->mPositionSequence != update.mPositionSequence) {
+ if (update.mPosition > mFrameCount) {
+ return NO_INIT;
+ } else if (localState->mLoopCount != 0 && update.mPosition >= localState->mLoopEnd) {
+ localState->mLoopCount = 0; // disable loop count if position is beyond loop end.
+ }
+ localState->mPosition = update.mPosition;
+ localState->mPositionSequence = update.mPositionSequence;
+ }
+ return OK;
+}
+
+ssize_t StaticAudioTrackServerProxy::pollPosition()
+{
+ StaticAudioTrackState state;
+ if (mObserver.poll(state)) {
+ StaticAudioTrackState trystate = mState;
+ bool result;
+ const int32_t diffSeq = state.mLoopSequence - state.mPositionSequence;
+
+ if (diffSeq < 0) {
+ result = updateStateWithLoop(&trystate, state) == OK &&
+ updateStateWithPosition(&trystate, state) == OK;
+ } else {
+ result = updateStateWithPosition(&trystate, state) == OK &&
+ updateStateWithLoop(&trystate, state) == OK;
+ }
+ if (!result) {
+ mObserver.done();
+ // caution: no update occurs so server state will be inconsistent with client state.
ALOGE("%s client pushed an invalid state, shutting down", __func__);
mIsShutdown = true;
return (ssize_t) NO_INIT;
}
+ mState = trystate;
+ if (mState.mLoopCount == -1) {
+ mFramesReady = INT64_MAX;
+ } else if (mState.mLoopCount == 0) {
+ mFramesReady = mFrameCount - mState.mPosition;
+ } else if (mState.mLoopCount > 0) {
+ // TODO: Later consider fixing overflow, but does not seem needed now
+ // as will not overflow if loopStart and loopEnd are Java "ints".
+ mFramesReady = int64_t(mState.mLoopCount) * (mState.mLoopEnd - mState.mLoopStart)
+ + mFrameCount - mState.mPosition;
+ }
mFramesReadySafe = clampToSize(mFramesReady);
// This may overflow, but client is not supposed to rely on it
- mCblk->u.mStatic.mBufferPosition = (uint32_t) position;
+ StaticAudioTrackPosLoop posLoop;
+
+ posLoop.mLoopCount = (int32_t) mState.mLoopCount;
+ posLoop.mBufferPosition = (uint32_t) mState.mPosition;
+ mPosLoopMutator.push(posLoop);
+ mObserver.done(); // safe to read mStatic variables.
}
- return (ssize_t) position;
+ return (ssize_t) mState.mPosition;
}
status_t StaticAudioTrackServerProxy::obtainBuffer(Buffer* buffer, bool ackFlush __unused)
@@ -849,7 +950,7 @@
}
// As mFramesReady is the total remaining frames in the static audio track,
// it is always larger or equal to avail.
- LOG_ALWAYS_FATAL_IF(mFramesReady < avail);
+ LOG_ALWAYS_FATAL_IF(mFramesReady < (int64_t) avail);
buffer->mNonContig = mFramesReady == INT64_MAX ? SIZE_MAX : clampToSize(mFramesReady - avail);
mUnreleased = avail;
return NO_ERROR;
@@ -858,7 +959,7 @@
void StaticAudioTrackServerProxy::releaseBuffer(Buffer* buffer)
{
size_t stepCount = buffer->mFrameCount;
- LOG_ALWAYS_FATAL_IF(!(stepCount <= mFramesReady));
+ LOG_ALWAYS_FATAL_IF(!((int64_t) stepCount <= mFramesReady));
LOG_ALWAYS_FATAL_IF(!(stepCount <= mUnreleased));
if (stepCount == 0) {
// prevent accidental re-use of buffer
@@ -868,11 +969,12 @@
}
mUnreleased -= stepCount;
audio_track_cblk_t* cblk = mCblk;
- size_t position = mPosition;
+ size_t position = mState.mPosition;
size_t newPosition = position + stepCount;
int32_t setFlags = 0;
if (!(position <= newPosition && newPosition <= mFrameCount)) {
- ALOGW("%s newPosition %zu outside [%zu, %zu]", __func__, newPosition, position, mFrameCount);
+ ALOGW("%s newPosition %zu outside [%zu, %zu]", __func__, newPosition, position,
+ mFrameCount);
newPosition = mFrameCount;
} else if (mState.mLoopCount != 0 && newPosition == mState.mLoopEnd) {
newPosition = mState.mLoopStart;
@@ -885,7 +987,7 @@
if (newPosition == mFrameCount) {
setFlags |= CBLK_BUFFER_END;
}
- mPosition = newPosition;
+ mState.mPosition = newPosition;
if (mFramesReady != INT64_MAX) {
mFramesReady -= stepCount;
}
@@ -893,7 +995,10 @@
cblk->mServer += stepCount;
// This may overflow, but client is not supposed to rely on it
- cblk->u.mStatic.mBufferPosition = (uint32_t) newPosition;
+ StaticAudioTrackPosLoop posLoop;
+ posLoop.mBufferPosition = mState.mPosition;
+ posLoop.mLoopCount = mState.mLoopCount;
+ mPosLoopMutator.push(posLoop);
if (setFlags != 0) {
(void) android_atomic_or(setFlags, &cblk->mFlags);
// this would be a good place to wake a futex
diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp
index dbc7a9e..f2ff27b 100644
--- a/media/libmedia/IAudioPolicyService.cpp
+++ b/media/libmedia/IAudioPolicyService.cpp
@@ -73,6 +73,8 @@
REGISTER_POLICY_MIXES,
};
+#define MAX_ITEMS_PER_LIST 1024
+
class BpAudioPolicyService : public BpInterface<IAudioPolicyService>
{
public:
@@ -84,13 +86,15 @@
virtual status_t setDeviceConnectionState(
audio_devices_t device,
audio_policy_dev_state_t state,
- const char *device_address)
+ const char *device_address,
+ const char *device_name)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
data.writeInt32(static_cast <uint32_t>(device));
data.writeInt32(static_cast <uint32_t>(state));
data.writeCString(device_address);
+ data.writeCString(device_name);
remote()->transact(SET_DEVICE_CONNECTION_STATE, data, &reply);
return static_cast <status_t> (reply.readInt32());
}
@@ -726,9 +730,11 @@
audio_policy_dev_state_t state =
static_cast <audio_policy_dev_state_t>(data.readInt32());
const char *device_address = data.readCString();
+ const char *device_name = data.readCString();
reply->writeInt32(static_cast<uint32_t> (setDeviceConnectionState(device,
state,
- device_address)));
+ device_address,
+ device_name)));
return NO_ERROR;
} break;
@@ -1054,10 +1060,18 @@
audio_port_role_t role = (audio_port_role_t)data.readInt32();
audio_port_type_t type = (audio_port_type_t)data.readInt32();
unsigned int numPortsReq = data.readInt32();
+ if (numPortsReq > MAX_ITEMS_PER_LIST) {
+ numPortsReq = MAX_ITEMS_PER_LIST;
+ }
unsigned int numPorts = numPortsReq;
- unsigned int generation;
struct audio_port *ports =
(struct audio_port *)calloc(numPortsReq, sizeof(struct audio_port));
+ if (ports == NULL) {
+ reply->writeInt32(NO_MEMORY);
+ reply->writeInt32(0);
+ return NO_ERROR;
+ }
+ unsigned int generation;
status_t status = listAudioPorts(role, type, &numPorts, ports, &generation);
reply->writeInt32(status);
reply->writeInt32(numPorts);
@@ -1111,11 +1125,19 @@
case LIST_AUDIO_PATCHES: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
unsigned int numPatchesReq = data.readInt32();
+ if (numPatchesReq > MAX_ITEMS_PER_LIST) {
+ numPatchesReq = MAX_ITEMS_PER_LIST;
+ }
unsigned int numPatches = numPatchesReq;
- unsigned int generation;
struct audio_patch *patches =
(struct audio_patch *)calloc(numPatchesReq,
sizeof(struct audio_patch));
+ if (patches == NULL) {
+ reply->writeInt32(NO_MEMORY);
+ reply->writeInt32(0);
+ return NO_ERROR;
+ }
+ unsigned int generation;
status_t status = listAudioPatches(&numPatches, patches, &generation);
reply->writeInt32(status);
reply->writeInt32(numPatches);
diff --git a/media/libmedia/IMediaRecorder.cpp b/media/libmedia/IMediaRecorder.cpp
index a733b68..9181f86 100644
--- a/media/libmedia/IMediaRecorder.cpp
+++ b/media/libmedia/IMediaRecorder.cpp
@@ -46,7 +46,6 @@
SET_OUTPUT_FORMAT,
SET_VIDEO_ENCODER,
SET_AUDIO_ENCODER,
- SET_OUTPUT_FILE_PATH,
SET_OUTPUT_FILE_FD,
SET_VIDEO_SIZE,
SET_VIDEO_FRAMERATE,
@@ -158,16 +157,6 @@
return reply.readInt32();
}
- status_t setOutputFile(const char* path)
- {
- ALOGV("setOutputFile(%s)", path);
- Parcel data, reply;
- data.writeInterfaceToken(IMediaRecorder::getInterfaceDescriptor());
- data.writeCString(path);
- remote()->transact(SET_OUTPUT_FILE_PATH, data, &reply);
- return reply.readInt32();
- }
-
status_t setOutputFile(int fd, int64_t offset, int64_t length) {
ALOGV("setOutputFile(%d, %" PRId64 ", %" PRId64 ")", fd, offset, length);
Parcel data, reply;
@@ -300,7 +289,8 @@
// ----------------------------------------------------------------------
status_t BnMediaRecorder::onTransact(
- uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
+ uint32_t code, const Parcel& data, Parcel* reply,
+ uint32_t flags)
{
switch (code) {
case RELEASE: {
@@ -390,13 +380,6 @@
return NO_ERROR;
} break;
- case SET_OUTPUT_FILE_PATH: {
- ALOGV("SET_OUTPUT_FILE_PATH");
- CHECK_INTERFACE(IMediaRecorder, data, reply);
- const char* path = data.readCString();
- reply->writeInt32(setOutputFile(path));
- return NO_ERROR;
- } break;
case SET_OUTPUT_FILE_FD: {
ALOGV("SET_OUTPUT_FILE_FD");
CHECK_INTERFACE(IMediaRecorder, data, reply);
@@ -445,7 +428,8 @@
case SET_PREVIEW_SURFACE: {
ALOGV("SET_PREVIEW_SURFACE");
CHECK_INTERFACE(IMediaRecorder, data, reply);
- sp<IGraphicBufferProducer> surface = interface_cast<IGraphicBufferProducer>(data.readStrongBinder());
+ sp<IGraphicBufferProducer> surface = interface_cast<IGraphicBufferProducer>(
+ data.readStrongBinder());
reply->writeInt32(setPreviewSurface(surface));
return NO_ERROR;
} break;
diff --git a/media/libmedia/JetPlayer.cpp b/media/libmedia/JetPlayer.cpp
index 721d8d7..271be0c 100644
--- a/media/libmedia/JetPlayer.cpp
+++ b/media/libmedia/JetPlayer.cpp
@@ -408,7 +408,8 @@
ALOGV("JetPlayer::queueSegment segmentNum=%d, libNum=%d, repeatCount=%d, transpose=%d",
segmentNum, libNum, repeatCount, transpose);
Mutex::Autolock lock(mMutex);
- return JET_QueueSegment(mEasData, segmentNum, libNum, repeatCount, transpose, muteFlags, userID);
+ return JET_QueueSegment(mEasData, segmentNum, libNum, repeatCount, transpose, muteFlags,
+ userID);
}
//-------------------------------------------------------------------------------------------------
@@ -449,7 +450,8 @@
void JetPlayer::dumpJetStatus(S_JET_STATUS* pJetStatus)
{
if (pJetStatus!=NULL)
- ALOGV(">> current JET player status: userID=%d segmentRepeatCount=%d numQueuedSegments=%d paused=%d",
+ ALOGV(">> current JET player status: userID=%d segmentRepeatCount=%d numQueuedSegments=%d "
+ "paused=%d",
pJetStatus->currentUserID, pJetStatus->segmentRepeatCount,
pJetStatus->numQueuedSegments, pJetStatus->paused);
else
diff --git a/media/libmedia/MediaProfiles.cpp b/media/libmedia/MediaProfiles.cpp
index e2e6042..47f9258 100644
--- a/media/libmedia/MediaProfiles.cpp
+++ b/media/libmedia/MediaProfiles.cpp
@@ -163,7 +163,8 @@
}
/*static*/ int
-MediaProfiles::findTagForName(const MediaProfiles::NameToTagMap *map, size_t nMappings, const char *name)
+MediaProfiles::findTagForName(const MediaProfiles::NameToTagMap *map, size_t nMappings,
+ const char *name)
{
int tag = -1;
for (size_t i = 0; i < nMappings; ++i) {
@@ -295,9 +296,8 @@
CHECK(codec != -1);
MediaProfiles::AudioEncoderCap *cap =
- new MediaProfiles::AudioEncoderCap(static_cast<audio_encoder>(codec), atoi(atts[5]), atoi(atts[7]),
- atoi(atts[9]), atoi(atts[11]), atoi(atts[13]),
- atoi(atts[15]));
+ new MediaProfiles::AudioEncoderCap(static_cast<audio_encoder>(codec), atoi(atts[5]),
+ atoi(atts[7]), atoi(atts[9]), atoi(atts[11]), atoi(atts[13]), atoi(atts[15]));
logAudioEncoderCap(*cap);
return cap;
}
@@ -330,7 +330,8 @@
!strcmp("fileFormat", atts[2]) &&
!strcmp("duration", atts[4]));
- const size_t nProfileMappings = sizeof(sCamcorderQualityNameMap)/sizeof(sCamcorderQualityNameMap[0]);
+ const size_t nProfileMappings = sizeof(sCamcorderQualityNameMap)/
+ sizeof(sCamcorderQualityNameMap[0]);
const int quality = findTagForName(sCamcorderQualityNameMap, nProfileMappings, atts[1]);
CHECK(quality != -1);
@@ -722,16 +723,20 @@
MediaProfiles::createDefaultCamcorderTimeLapseLowProfiles(
MediaProfiles::CamcorderProfile **lowTimeLapseProfile,
MediaProfiles::CamcorderProfile **lowSpecificTimeLapseProfile) {
- *lowTimeLapseProfile = createDefaultCamcorderTimeLapseQcifProfile(CAMCORDER_QUALITY_TIME_LAPSE_LOW);
- *lowSpecificTimeLapseProfile = createDefaultCamcorderTimeLapseQcifProfile(CAMCORDER_QUALITY_TIME_LAPSE_QCIF);
+ *lowTimeLapseProfile = createDefaultCamcorderTimeLapseQcifProfile(
+ CAMCORDER_QUALITY_TIME_LAPSE_LOW);
+ *lowSpecificTimeLapseProfile = createDefaultCamcorderTimeLapseQcifProfile(
+ CAMCORDER_QUALITY_TIME_LAPSE_QCIF);
}
/*static*/ void
MediaProfiles::createDefaultCamcorderTimeLapseHighProfiles(
MediaProfiles::CamcorderProfile **highTimeLapseProfile,
MediaProfiles::CamcorderProfile **highSpecificTimeLapseProfile) {
- *highTimeLapseProfile = createDefaultCamcorderTimeLapse480pProfile(CAMCORDER_QUALITY_TIME_LAPSE_HIGH);
- *highSpecificTimeLapseProfile = createDefaultCamcorderTimeLapse480pProfile(CAMCORDER_QUALITY_TIME_LAPSE_480P);
+ *highTimeLapseProfile = createDefaultCamcorderTimeLapse480pProfile(
+ CAMCORDER_QUALITY_TIME_LAPSE_HIGH);
+ *highSpecificTimeLapseProfile = createDefaultCamcorderTimeLapse480pProfile(
+ CAMCORDER_QUALITY_TIME_LAPSE_480P);
}
/*static*/ MediaProfiles::CamcorderProfile*
@@ -809,7 +814,8 @@
// high camcorder time lapse profiles.
MediaProfiles::CamcorderProfile *highTimeLapseProfile, *highSpecificTimeLapseProfile;
- createDefaultCamcorderTimeLapseHighProfiles(&highTimeLapseProfile, &highSpecificTimeLapseProfile);
+ createDefaultCamcorderTimeLapseHighProfiles(&highTimeLapseProfile,
+ &highSpecificTimeLapseProfile);
profiles->mCamcorderProfiles.add(highTimeLapseProfile);
profiles->mCamcorderProfiles.add(highSpecificTimeLapseProfile);
diff --git a/media/libmedia/SingleStateQueue.cpp b/media/libmedia/SingleStateQueue.cpp
deleted file mode 100644
index c241184..0000000
--- a/media/libmedia/SingleStateQueue.cpp
+++ /dev/null
@@ -1,106 +0,0 @@
-/*
- * Copyright (C) 2012 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include <new>
-#include <cutils/atomic.h>
-#include <media/SingleStateQueue.h>
-
-namespace android {
-
-template<typename T> SingleStateQueue<T>::Mutator::Mutator(Shared *shared)
- : mSequence(0), mShared((Shared *) shared)
-{
- // exactly one of Mutator and Observer must initialize, currently it is Observer
- //shared->init();
-}
-
-template<typename T> int32_t SingleStateQueue<T>::Mutator::push(const T& value)
-{
- Shared *shared = mShared;
- int32_t sequence = mSequence;
- sequence++;
- android_atomic_acquire_store(sequence, &shared->mSequence);
- shared->mValue = value;
- sequence++;
- android_atomic_release_store(sequence, &shared->mSequence);
- mSequence = sequence;
- // consider signalling a futex here, if we know that observer is waiting
- return sequence;
-}
-
-template<typename T> bool SingleStateQueue<T>::Mutator::ack()
-{
- return mShared->mAck - mSequence == 0;
-}
-
-template<typename T> bool SingleStateQueue<T>::Mutator::ack(int32_t sequence)
-{
- // this relies on 2's complement rollover to detect an ancient sequence number
- return mShared->mAck - sequence >= 0;
-}
-
-template<typename T> SingleStateQueue<T>::Observer::Observer(Shared *shared)
- : mSequence(0), mSeed(1), mShared((Shared *) shared)
-{
- // exactly one of Mutator and Observer must initialize, currently it is Observer
- shared->init();
-}
-
-template<typename T> bool SingleStateQueue<T>::Observer::poll(T& value)
-{
- Shared *shared = mShared;
- int32_t before = shared->mSequence;
- if (before == mSequence) {
- return false;
- }
- for (int tries = 0; ; ) {
- const int MAX_TRIES = 5;
- if (before & 1) {
- if (++tries >= MAX_TRIES) {
- return false;
- }
- before = shared->mSequence;
- } else {
- android_memory_barrier();
- T temp = shared->mValue;
- int32_t after = android_atomic_release_load(&shared->mSequence);
- if (after == before) {
- value = temp;
- shared->mAck = before;
- mSequence = before;
- return true;
- }
- if (++tries >= MAX_TRIES) {
- return false;
- }
- before = after;
- }
- }
-}
-
-#if 0
-template<typename T> SingleStateQueue<T>::SingleStateQueue(void /*Shared*/ *shared)
-{
- ((Shared *) shared)->init();
-}
-#endif
-
-} // namespace android
-
-// hack for gcc
-#ifdef SINGLE_STATE_QUEUE_INSTANTIATIONS
-#include SINGLE_STATE_QUEUE_INSTANTIATIONS
-#endif
diff --git a/media/libmedia/SingleStateQueueInstantiations.cpp b/media/libmedia/SingleStateQueueInstantiations.cpp
deleted file mode 100644
index 0265c8c..0000000
--- a/media/libmedia/SingleStateQueueInstantiations.cpp
+++ /dev/null
@@ -1,28 +0,0 @@
-/*
- * Copyright (C) 2012 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include <media/SingleStateQueue.h>
-#include <private/media/StaticAudioTrackState.h>
-#include <media/AudioTimestamp.h>
-
-// FIXME hack for gcc
-
-namespace android {
-
-template class SingleStateQueue<StaticAudioTrackState>; // typedef StaticAudioTrackSingleStateQueue
-template class SingleStateQueue<AudioTimestamp>; // typedef AudioTimestampSingleStateQueue
-
-}
diff --git a/media/libmedia/StringArray.cpp b/media/libmedia/StringArray.cpp
index 5f5b57a..477e3fd 100644
--- a/media/libmedia/StringArray.cpp
+++ b/media/libmedia/StringArray.cpp
@@ -16,7 +16,7 @@
//
// Sortable array of strings. STL-ish, but STL-free.
-//
+//
#include <stdlib.h>
#include <string.h>
diff --git a/media/libmedia/docs/Makefile b/media/libmedia/docs/Makefile
new file mode 100644
index 0000000..bddbc9b
--- /dev/null
+++ b/media/libmedia/docs/Makefile
@@ -0,0 +1,2 @@
+paused.png : paused.dot
+ dot -Tpng < $< > $@
diff --git a/media/libmedia/docs/paused.dot b/media/libmedia/docs/paused.dot
new file mode 100644
index 0000000..11e1777
--- /dev/null
+++ b/media/libmedia/docs/paused.dot
@@ -0,0 +1,85 @@
+digraph paused {
+initial [label="INITIAL\n\
+mIgnoreNextPausedInt = false\n\
+mPaused = false\n\
+mPausedInt = false"];
+
+resume_body [label="mIgnoreNextPausedInt = true\nif (mPaused || mPausedInt)"];
+resume_paused [label="mPaused = false\nmPausedInt = false\nsignal()"];
+resume_paused -> resume_merged;
+resume_merged [label="return"];
+
+Application -> ATstop;
+ATstop [label="AudioTrack::stop()"];
+ATstop -> pause;
+Application -> ATpause;
+ATpause [label="AudioTrack::pause()"];
+ATpause -> pause;
+ATstart -> resume;
+ATstart [label="AudioTrack::start()"];
+destructor [label="~AudioTrack()"];
+destructor -> requestExit;
+requestExit [label="AudioTrackThread::requestExit()"];
+requestExit -> resume;
+Application -> ATsetMarkerPosition
+ATsetMarkerPosition [label="AudioTrack::setMarkerPosition()\n[sets marker variables]"];
+ATsetMarkerPosition -> ATTwake
+Application -> ATsetPositionUpdatePeriod
+ATsetPositionUpdatePeriod [label="AudioTrack::setPositionUpdatePeriod()\n[sets update period variables]"];
+ATsetPositionUpdatePeriod -> ATTwake
+Application -> ATstart;
+
+resume [label="AudioTrackThread::resume()"];
+resume -> resume_body;
+
+resume_body -> resume_paused [label="true"];
+resume_body -> resume_merged [label="false"];
+
+ATTwake [label="AudioTrackThread::wake()\nif (!mPaused && mPausedInt && mPausedNs > 0)"];
+ATTwake-> ATTWake_wakeable [label="true"];
+ATTWake_wakeable [label="mIgnoreNextPausedInt = true\nmPausedInt = false\nsignal()"];
+ATTwake-> ATTWake_cannotwake [label="false"]
+ATTWake_cannotwake [label="ignore"];
+
+pause [label="mPaused = true"];
+pause -> return;
+
+threadLoop [label="AudioTrackThread::threadLoop()\nENTRY"];
+threadLoop -> threadLoop_1;
+threadLoop_1 [label="if (mPaused)"];
+threadLoop_1 -> threadLoop_1_true [label="true"];
+threadLoop_1 -> threadLoop_2 [label="false"];
+threadLoop_1_true [label="wait()\nreturn true"];
+threadLoop_2 [label="if (mIgnoreNextPausedInt)"];
+threadLoop_2 -> threadLoop_2_true [label="true"];
+threadLoop_2 -> threadLoop_3 [label="false"];
+threadLoop_2_true [label="mIgnoreNextPausedInt = false\nmPausedInt = false"];
+threadLoop_2_true -> threadLoop_3;
+threadLoop_3 [label="if (mPausedInt)"];
+threadLoop_3 -> threadLoop_3_true [label="true"];
+threadLoop_3 -> threadLoop_4 [label="false"];
+threadLoop_3_true [label="wait()\nmPausedInt = false\nreturn true"];
+threadLoop_4 [label="if (exitPending)"];
+threadLoop_4 -> threadLoop_4_true [label="true"];
+threadLoop_4 -> threadLoop_5 [label="false"];
+threadLoop_4_true [label="return false"];
+threadLoop_5 [label="ns = processAudioBuffer()"];
+threadLoop_5 -> threadLoop_6;
+threadLoop_6 [label="case ns"];
+threadLoop_6 -> threadLoop_6_0 [label="0"];
+threadLoop_6 -> threadLoop_6_NS_INACTIVE [label="NS_INACTIVE"];
+threadLoop_6 -> threadLoop_6_NS_NEVER [label="NS_NEVER"];
+threadLoop_6 -> threadLoop_6_NS_WHENEVER [label="NS_WHENEVER"];
+threadLoop_6 -> threadLoop_6_default [label="default"];
+threadLoop_6_default [label="if (ns < 0)"];
+threadLoop_6_default -> threadLoop_6_default_true [label="true"];
+threadLoop_6_default -> threadLoop_6_default_false [label="false"];
+threadLoop_6_default_true [label="FATAL"];
+threadLoop_6_default_false [label="pauseInternal(ns) [wake()-able]\nmPausedInternal = true\nmPausedNs = ns\nreturn true"];
+threadLoop_6_0 [label="return true"];
+threadLoop_6_NS_INACTIVE [label="pauseInternal()\nmPausedInternal = true\nmPausedNs = 0\nreturn true"];
+threadLoop_6_NS_NEVER [label="return false"];
+threadLoop_6_NS_WHENEVER [label="ns = 1s"];
+threadLoop_6_NS_WHENEVER -> threadLoop_6_default_false;
+
+}
diff --git a/media/libmedia/mediaplayer.cpp b/media/libmedia/mediaplayer.cpp
index 05c89ed..432ecda 100644
--- a/media/libmedia/mediaplayer.cpp
+++ b/media/libmedia/mediaplayer.cpp
@@ -240,7 +240,7 @@
// must call with lock held
status_t MediaPlayer::prepareAsync_l()
{
- if ( (mPlayer != 0) && ( mCurrentState & ( MEDIA_PLAYER_INITIALIZED | MEDIA_PLAYER_STOPPED) ) ) {
+ if ( (mPlayer != 0) && ( mCurrentState & (MEDIA_PLAYER_INITIALIZED | MEDIA_PLAYER_STOPPED) ) ) {
mPlayer->setAudioStreamType(mStreamType);
if (mAudioAttributesParcel != NULL) {
mPlayer->setParameter(KEY_PARAMETER_AUDIO_ATTRIBUTES, *mAudioAttributesParcel);
@@ -414,7 +414,8 @@
status_t MediaPlayer::getDuration_l(int *msec)
{
ALOGV("getDuration_l");
- bool isValidState = (mCurrentState & (MEDIA_PLAYER_PREPARED | MEDIA_PLAYER_STARTED | MEDIA_PLAYER_PAUSED | MEDIA_PLAYER_STOPPED | MEDIA_PLAYER_PLAYBACK_COMPLETE));
+ bool isValidState = (mCurrentState & (MEDIA_PLAYER_PREPARED | MEDIA_PLAYER_STARTED |
+ MEDIA_PLAYER_PAUSED | MEDIA_PLAYER_STOPPED | MEDIA_PLAYER_PLAYBACK_COMPLETE));
if (mPlayer != 0 && isValidState) {
int durationMs;
status_t ret = mPlayer->getDuration(&durationMs);
@@ -443,7 +444,8 @@
status_t MediaPlayer::seekTo_l(int msec)
{
ALOGV("seekTo %d", msec);
- if ((mPlayer != 0) && ( mCurrentState & ( MEDIA_PLAYER_STARTED | MEDIA_PLAYER_PREPARED | MEDIA_PLAYER_PAUSED | MEDIA_PLAYER_PLAYBACK_COMPLETE) ) ) {
+ if ((mPlayer != 0) && ( mCurrentState & ( MEDIA_PLAYER_STARTED | MEDIA_PLAYER_PREPARED |
+ MEDIA_PLAYER_PAUSED | MEDIA_PLAYER_PLAYBACK_COMPLETE) ) ) {
if ( msec < 0 ) {
ALOGW("Attempt to seek to invalid position: %d", msec);
msec = 0;
@@ -477,7 +479,8 @@
return NO_ERROR;
}
}
- ALOGE("Attempt to perform seekTo in wrong state: mPlayer=%p, mCurrentState=%u", mPlayer.get(), mCurrentState);
+ ALOGE("Attempt to perform seekTo in wrong state: mPlayer=%p, mCurrentState=%u", mPlayer.get(),
+ mCurrentState);
return INVALID_OPERATION;
}
diff --git a/media/libmedia/mediarecorder.cpp b/media/libmedia/mediarecorder.cpp
index 1952b86..973e156 100644
--- a/media/libmedia/mediarecorder.cpp
+++ b/media/libmedia/mediarecorder.cpp
@@ -264,32 +264,6 @@
return ret;
}
-status_t MediaRecorder::setOutputFile(const char* path)
-{
- ALOGV("setOutputFile(%s)", path);
- if (mMediaRecorder == NULL) {
- ALOGE("media recorder is not initialized yet");
- return INVALID_OPERATION;
- }
- if (mIsOutputFileSet) {
- ALOGE("output file has already been set");
- return INVALID_OPERATION;
- }
- if (!(mCurrentState & MEDIA_RECORDER_DATASOURCE_CONFIGURED)) {
- ALOGE("setOutputFile called in an invalid state(%d)", mCurrentState);
- return INVALID_OPERATION;
- }
-
- status_t ret = mMediaRecorder->setOutputFile(path);
- if (OK != ret) {
- ALOGV("setOutputFile failed: %d", ret);
- mCurrentState = MEDIA_RECORDER_ERROR;
- return ret;
- }
- mIsOutputFileSet = true;
- return ret;
-}
-
status_t MediaRecorder::setOutputFile(int fd, int64_t offset, int64_t length)
{
ALOGV("setOutputFile(%d, %" PRId64 ", %" PRId64 ")", fd, offset, length);
diff --git a/media/libmediaplayerservice/MediaRecorderClient.cpp b/media/libmediaplayerservice/MediaRecorderClient.cpp
index 194abbb..4d4de9b 100644
--- a/media/libmediaplayerservice/MediaRecorderClient.cpp
+++ b/media/libmediaplayerservice/MediaRecorderClient.cpp
@@ -154,17 +154,6 @@
return mRecorder->setAudioEncoder((audio_encoder)ae);
}
-status_t MediaRecorderClient::setOutputFile(const char* path)
-{
- ALOGV("setOutputFile(%s)", path);
- Mutex::Autolock lock(mLock);
- if (mRecorder == NULL) {
- ALOGE("recorder is not initialized");
- return NO_INIT;
- }
- return mRecorder->setOutputFile(path);
-}
-
status_t MediaRecorderClient::setOutputFile(int fd, int64_t offset, int64_t length)
{
ALOGV("setOutputFile(%d, %lld, %lld)", fd, offset, length);
diff --git a/media/libmediaplayerservice/MediaRecorderClient.h b/media/libmediaplayerservice/MediaRecorderClient.h
index a65ec9f..a444b6c 100644
--- a/media/libmediaplayerservice/MediaRecorderClient.h
+++ b/media/libmediaplayerservice/MediaRecorderClient.h
@@ -38,7 +38,6 @@
virtual status_t setOutputFormat(int of);
virtual status_t setVideoEncoder(int ve);
virtual status_t setAudioEncoder(int ae);
- virtual status_t setOutputFile(const char* path);
virtual status_t setOutputFile(int fd, int64_t offset,
int64_t length);
virtual status_t setVideoSize(int width, int height);
diff --git a/media/libmediaplayerservice/StagefrightRecorder.cpp b/media/libmediaplayerservice/StagefrightRecorder.cpp
index 86639cb..2551040 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.cpp
+++ b/media/libmediaplayerservice/StagefrightRecorder.cpp
@@ -206,7 +206,7 @@
status_t StagefrightRecorder::setVideoFrameRate(int frames_per_second) {
ALOGV("setVideoFrameRate: %d", frames_per_second);
if ((frames_per_second <= 0 && frames_per_second != -1) ||
- frames_per_second > 120) {
+ frames_per_second > kMaxHighSpeedFps) {
ALOGE("Invalid video frame rate: %d", frames_per_second);
return BAD_VALUE;
}
@@ -241,14 +241,6 @@
return OK;
}
-status_t StagefrightRecorder::setOutputFile(const char * /* path */) {
- ALOGE("setOutputFile(const char*) must not be called");
- // We don't actually support this at all, as the media_server process
- // no longer has permissions to create files.
-
- return -EPERM;
-}
-
status_t StagefrightRecorder::setOutputFile(int fd, int64_t offset, int64_t length) {
ALOGV("setOutputFile: %d, %lld, %lld", fd, offset, length);
// These don't make any sense, do they?
diff --git a/media/libmediaplayerservice/StagefrightRecorder.h b/media/libmediaplayerservice/StagefrightRecorder.h
index 54c38d3..b5a49d3 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.h
+++ b/media/libmediaplayerservice/StagefrightRecorder.h
@@ -53,7 +53,6 @@
virtual status_t setVideoFrameRate(int frames_per_second);
virtual status_t setCamera(const sp<ICamera>& camera, const sp<ICameraRecordingProxy>& proxy);
virtual status_t setPreviewSurface(const sp<IGraphicBufferProducer>& surface);
- virtual status_t setOutputFile(const char *path);
virtual status_t setOutputFile(int fd, int64_t offset, int64_t length);
virtual status_t setParameters(const String8& params);
virtual status_t setListener(const sp<IMediaRecorderClient>& listener);
@@ -127,6 +126,8 @@
sp<IGraphicBufferProducer> mGraphicBufferProducer;
sp<ALooper> mLooper;
+ static const int kMaxHighSpeedFps = 1000;
+
status_t prepareInternal();
status_t setupMPEG4orWEBMRecording();
void setupMPEG4orWEBMMetaData(sp<MetaData> *meta);
diff --git a/media/libmediaplayerservice/nuplayer/Android.mk b/media/libmediaplayerservice/nuplayer/Android.mk
index 6609874..e2c72ed 100644
--- a/media/libmediaplayerservice/nuplayer/Android.mk
+++ b/media/libmediaplayerservice/nuplayer/Android.mk
@@ -4,6 +4,7 @@
LOCAL_SRC_FILES:= \
GenericSource.cpp \
HTTPLiveSource.cpp \
+ MediaClock.cpp \
NuPlayer.cpp \
NuPlayerCCDecoder.cpp \
NuPlayerDecoder.cpp \
diff --git a/media/libmediaplayerservice/nuplayer/MediaClock.cpp b/media/libmediaplayerservice/nuplayer/MediaClock.cpp
new file mode 100644
index 0000000..7bfff13
--- /dev/null
+++ b/media/libmediaplayerservice/nuplayer/MediaClock.cpp
@@ -0,0 +1,135 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "MediaClock"
+#include <utils/Log.h>
+
+#include "MediaClock.h"
+
+#include <media/stagefright/foundation/ALooper.h>
+
+namespace android {
+
+// Maximum time change between two updates.
+static const int64_t kMaxAnchorFluctuationUs = 1000ll;
+
+MediaClock::MediaClock()
+ : mAnchorTimeMediaUs(-1),
+ mAnchorTimeRealUs(-1),
+ mMaxTimeMediaUs(INT64_MAX),
+ mStartingTimeMediaUs(-1),
+ mPaused(false) {
+}
+
+MediaClock::~MediaClock() {
+}
+
+void MediaClock::setStartingTimeMedia(int64_t startingTimeMediaUs) {
+ Mutex::Autolock autoLock(mLock);
+ mStartingTimeMediaUs = startingTimeMediaUs;
+}
+
+void MediaClock::clearAnchor() {
+ Mutex::Autolock autoLock(mLock);
+ mAnchorTimeMediaUs = -1;
+ mAnchorTimeRealUs = -1;
+}
+
+void MediaClock::updateAnchor(
+ int64_t anchorTimeMediaUs,
+ int64_t anchorTimeRealUs,
+ int64_t maxTimeMediaUs) {
+ if (anchorTimeMediaUs < 0 || anchorTimeRealUs < 0) {
+ ALOGW("reject anchor time since it is negative.");
+ return;
+ }
+
+ int64_t nowUs = ALooper::GetNowUs();
+ int64_t nowMediaUs = anchorTimeMediaUs + nowUs - anchorTimeRealUs;
+ if (nowMediaUs < 0) {
+ ALOGW("reject anchor time since it leads to negative media time.");
+ return;
+ }
+
+ Mutex::Autolock autoLock(mLock);
+ mAnchorTimeRealUs = nowUs;
+ mAnchorTimeMediaUs = nowMediaUs;
+ mMaxTimeMediaUs = maxTimeMediaUs;
+}
+
+void MediaClock::updateMaxTimeMedia(int64_t maxTimeMediaUs) {
+ Mutex::Autolock autoLock(mLock);
+ mMaxTimeMediaUs = maxTimeMediaUs;
+}
+
+void MediaClock::pause() {
+ Mutex::Autolock autoLock(mLock);
+ if (mPaused) {
+ return;
+ }
+
+ mPaused = true;
+ if (mAnchorTimeRealUs == -1) {
+ return;
+ }
+
+ int64_t nowUs = ALooper::GetNowUs();
+ mAnchorTimeMediaUs += nowUs - mAnchorTimeRealUs;
+ if (mAnchorTimeMediaUs < 0) {
+ ALOGW("anchor time should not be negative, set to 0.");
+ mAnchorTimeMediaUs = 0;
+ }
+ mAnchorTimeRealUs = nowUs;
+}
+
+void MediaClock::resume() {
+ Mutex::Autolock autoLock(mLock);
+ if (!mPaused) {
+ return;
+ }
+
+ mPaused = false;
+ if (mAnchorTimeRealUs == -1) {
+ return;
+ }
+
+ mAnchorTimeRealUs = ALooper::GetNowUs();
+}
+
+int64_t MediaClock::getTimeMedia(int64_t realUs, bool allowPastMaxTime) {
+ Mutex::Autolock autoLock(mLock);
+ if (mAnchorTimeRealUs == -1) {
+ return -1ll;
+ }
+
+ if (mPaused) {
+ realUs = mAnchorTimeRealUs;
+ }
+ int64_t currentMediaUs = mAnchorTimeMediaUs + realUs - mAnchorTimeRealUs;
+ if (currentMediaUs > mMaxTimeMediaUs && !allowPastMaxTime) {
+ currentMediaUs = mMaxTimeMediaUs;
+ }
+ if (currentMediaUs < mStartingTimeMediaUs) {
+ currentMediaUs = mStartingTimeMediaUs;
+ }
+ if (currentMediaUs < 0) {
+ currentMediaUs = 0;
+ }
+ return currentMediaUs;
+}
+
+} // namespace android
diff --git a/media/libmediaplayerservice/nuplayer/MediaClock.h b/media/libmediaplayerservice/nuplayer/MediaClock.h
new file mode 100644
index 0000000..d005993
--- /dev/null
+++ b/media/libmediaplayerservice/nuplayer/MediaClock.h
@@ -0,0 +1,68 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef MEDIA_CLOCK_H_
+
+#define MEDIA_CLOCK_H_
+
+#include <media/stagefright/foundation/ABase.h>
+#include <utils/Mutex.h>
+#include <utils/RefBase.h>
+
+namespace android {
+
+struct AMessage;
+
+struct MediaClock : public RefBase {
+ MediaClock();
+
+ void setStartingTimeMedia(int64_t startingTimeMediaUs);
+
+ void clearAnchor();
+ // It's highly recommended to use timestamp of just rendered frame as
+ // anchor time, especially in paused state. Such restriction will be
+ // required when dynamic playback rate is supported in the future.
+ void updateAnchor(
+ int64_t anchorTimeMediaUs,
+ int64_t anchorTimeRealUs,
+ int64_t maxTimeMediaUs = INT64_MAX);
+
+ void updateMaxTimeMedia(int64_t maxTimeMediaUs);
+
+ void pause();
+ void resume();
+
+ int64_t getTimeMedia(int64_t realUs, bool allowPastMaxTime = false);
+
+protected:
+ virtual ~MediaClock();
+
+private:
+ Mutex mLock;
+
+ int64_t mAnchorTimeMediaUs;
+ int64_t mAnchorTimeRealUs;
+ int64_t mMaxTimeMediaUs;
+ int64_t mStartingTimeMediaUs;
+
+ bool mPaused;
+
+ DISALLOW_EVIL_CONSTRUCTORS(MediaClock);
+};
+
+} // namespace android
+
+#endif // MEDIA_CLOCK_H_
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.cpp
index 9229704..cf3e8ad 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.cpp
@@ -19,6 +19,7 @@
#include <utils/Log.h>
#include <inttypes.h>
+#include "avc_utils.h"
#include "NuPlayerCCDecoder.h"
#include <media/stagefright/foundation/ABitReader.h>
@@ -185,17 +186,38 @@
// returns true if a new CC track is found
bool NuPlayer::CCDecoder::extractFromSEI(const sp<ABuffer> &accessUnit) {
- int64_t timeUs;
- CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs));
-
sp<ABuffer> sei;
if (!accessUnit->meta()->findBuffer("sei", &sei) || sei == NULL) {
return false;
}
+ int64_t timeUs;
+ CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs));
+
bool trackAdded = false;
- NALBitReader br(sei->data() + 1, sei->size() - 1);
+ const NALPosition *nal = (NALPosition *) sei->data();
+
+ for (size_t i = 0; i < sei->size() / sizeof(NALPosition); ++i, ++nal) {
+ trackAdded |= parseSEINalUnit(
+ timeUs, accessUnit->data() + nal->nalOffset, nal->nalSize);
+ }
+
+ return trackAdded;
+}
+
+// returns true if a new CC track is found
+bool NuPlayer::CCDecoder::parseSEINalUnit(
+ int64_t timeUs, const uint8_t *nalStart, size_t nalSize) {
+ unsigned nalType = nalStart[0] & 0x1f;
+
+ // the buffer should only have SEI in it
+ if (nalType != 6) {
+ return false;
+ }
+
+ bool trackAdded = false;
+ NALBitReader br(nalStart + 1, nalSize - 1);
// sei_message()
while (br.atLeastNumBitsLeft(16)) { // at least 16-bit for sei_message()
uint32_t payload_type = 0;
@@ -214,20 +236,25 @@
// sei_payload()
if (payload_type == 4) {
- // user_data_registered_itu_t_t35()
+ bool isCC = false;
+ if (payload_size > 1 + 2 + 4 + 1) {
+ // user_data_registered_itu_t_t35()
- // ATSC A/72: 6.4.2
- uint8_t itu_t_t35_country_code = br.getBits(8);
- uint16_t itu_t_t35_provider_code = br.getBits(16);
- uint32_t user_identifier = br.getBits(32);
- uint8_t user_data_type_code = br.getBits(8);
+ // ATSC A/72: 6.4.2
+ uint8_t itu_t_t35_country_code = br.getBits(8);
+ uint16_t itu_t_t35_provider_code = br.getBits(16);
+ uint32_t user_identifier = br.getBits(32);
+ uint8_t user_data_type_code = br.getBits(8);
- payload_size -= 1 + 2 + 4 + 1;
+ payload_size -= 1 + 2 + 4 + 1;
- if (itu_t_t35_country_code == 0xB5
- && itu_t_t35_provider_code == 0x0031
- && user_identifier == 'GA94'
- && user_data_type_code == 0x3) {
+ isCC = itu_t_t35_country_code == 0xB5
+ && itu_t_t35_provider_code == 0x0031
+ && user_identifier == 'GA94'
+ && user_data_type_code == 0x3;
+ }
+
+ if (isCC && payload_size > 2) {
// MPEG_cc_data()
// ATSC A/53 Part 4: 6.2.3.1
br.skipBits(1); //process_em_data_flag
@@ -243,7 +270,7 @@
sp<ABuffer> ccBuf = new ABuffer(cc_count * sizeof(CCData));
ccBuf->setRange(0, 0);
- for (size_t i = 0; i < cc_count; i++) {
+ for (size_t i = 0; i < cc_count && payload_size >= 3; i++) {
uint8_t marker = br.getBits(5);
CHECK_EQ(marker, 0x1f);
@@ -253,6 +280,8 @@
uint8_t cc_data_1 = br.getBits(8) & 0x7f;
uint8_t cc_data_2 = br.getBits(8) & 0x7f;
+ payload_size -= 3;
+
if (cc_valid
&& (cc_type == 0 || cc_type == 1)) {
CCData cc(cc_type, cc_data_1, cc_data_2);
@@ -269,7 +298,6 @@
}
}
}
- payload_size -= cc_count * 3;
mCCMap.add(timeUs, ccBuf);
break;
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.h b/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.h
index 5e06f4e..77fb0fe 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.h
@@ -49,6 +49,7 @@
bool isTrackValid(size_t index) const;
int32_t getTrackIndex(size_t channel) const;
bool extractFromSEI(const sp<ABuffer> &accessUnit);
+ bool parseSEINalUnit(int64_t timeUs, const uint8_t *nalStart, size_t nalSize);
sp<ABuffer> filterCCBuf(const sp<ABuffer> &ccBuf, size_t index);
DISALLOW_EVIL_CONSTRUCTORS(CCDecoder);
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
index bc79fdb..abfa4d3 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
@@ -351,6 +351,14 @@
case STATE_PREPARED:
case STATE_STOPPED_AND_PREPARED:
{
+ int curpos = 0;
+ if (mPositionUs > 0) {
+ curpos = (mPositionUs + 500ll) / 1000;
+ }
+ if (curpos == msec) {
+ // nothing to do, and doing something anyway could result in deadlock (b/15323063)
+ break;
+ }
mStartupSeekTimeUs = seekTimeUs;
// pretend that the seek completed. It will actually happen when starting playback.
// TODO: actually perform the seek here, so the player is ready to go at the new
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
index 25225a8..7f8680d 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
@@ -20,6 +20,8 @@
#include "NuPlayerRenderer.h"
+#include "MediaClock.h"
+
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/AMessage.h>
@@ -63,22 +65,18 @@
mDrainVideoQueuePending(false),
mAudioQueueGeneration(0),
mVideoQueueGeneration(0),
+ mAudioDrainGeneration(0),
+ mVideoDrainGeneration(0),
mAudioFirstAnchorTimeMediaUs(-1),
mAnchorTimeMediaUs(-1),
- mAnchorTimeRealUs(-1),
mAnchorNumFramesWritten(-1),
- mAnchorMaxMediaUs(-1),
mVideoLateByUs(0ll),
mHasAudio(false),
mHasVideo(false),
- mPauseStartedTimeRealUs(-1),
- mFlushingAudio(false),
- mFlushingVideo(false),
mNotifyCompleteAudio(false),
mNotifyCompleteVideo(false),
mSyncQueues(false),
mPaused(false),
- mPausePositionMediaTimeUs(-1),
mVideoSampleReceived(false),
mVideoRenderingStarted(false),
mVideoRenderingStartGeneration(0),
@@ -90,7 +88,7 @@
mTotalBuffersQueued(0),
mLastAudioBufferDrained(0),
mWakeLock(new AWakeLock()) {
-
+ mMediaClock = new MediaClock;
}
NuPlayer::Renderer::~Renderer() {
@@ -106,6 +104,7 @@
const sp<ABuffer> &buffer,
const sp<AMessage> ¬ifyConsumed) {
sp<AMessage> msg = new AMessage(kWhatQueueBuffer, id());
+ msg->setInt32("queueGeneration", getQueueGeneration(audio));
msg->setInt32("audio", static_cast<int32_t>(audio));
msg->setBuffer("buffer", buffer);
msg->setMessage("notifyConsumed", notifyConsumed);
@@ -116,6 +115,7 @@
CHECK_NE(finalResult, (status_t)OK);
sp<AMessage> msg = new AMessage(kWhatQueueEOS, id());
+ msg->setInt32("queueGeneration", getQueueGeneration(audio));
msg->setInt32("audio", static_cast<int32_t>(audio));
msg->setInt32("finalResult", finalResult);
msg->post();
@@ -123,20 +123,21 @@
void NuPlayer::Renderer::flush(bool audio, bool notifyComplete) {
{
- Mutex::Autolock autoLock(mFlushLock);
+ Mutex::Autolock autoLock(mLock);
if (audio) {
mNotifyCompleteAudio |= notifyComplete;
- if (mFlushingAudio) {
- return;
- }
- mFlushingAudio = true;
+ ++mAudioQueueGeneration;
+ ++mAudioDrainGeneration;
} else {
mNotifyCompleteVideo |= notifyComplete;
- if (mFlushingVideo) {
- return;
- }
- mFlushingVideo = true;
+ ++mVideoQueueGeneration;
+ ++mVideoDrainGeneration;
}
+
+ clearAnchorTime_l();
+ clearAudioFirstAnchorTime_l();
+ mVideoLateByUs = 0;
+ mSyncQueues = false;
}
sp<AMessage> msg = new AMessage(kWhatFlush, id());
@@ -145,17 +146,6 @@
}
void NuPlayer::Renderer::signalTimeDiscontinuity() {
- Mutex::Autolock autoLock(mLock);
- // CHECK(mAudioQueue.empty());
- // CHECK(mVideoQueue.empty());
- setAudioFirstAnchorTime(-1);
- setAnchorTime(-1, -1);
- setVideoLateByUs(0);
- mSyncQueues = false;
-}
-
-void NuPlayer::Renderer::signalAudioSinkChanged() {
- (new AMessage(kWhatAudioSinkChanged, id()))->post();
}
void NuPlayer::Renderer::signalDisableOffloadAudio() {
@@ -180,127 +170,44 @@
msg->post();
}
-// Called on any threads, except renderer's thread.
-status_t NuPlayer::Renderer::getCurrentPosition(int64_t *mediaUs) {
- {
- Mutex::Autolock autoLock(mLock);
- int64_t currentPositionUs;
- if (getCurrentPositionIfPaused_l(¤tPositionUs)) {
- *mediaUs = currentPositionUs;
- return OK;
- }
- }
- return getCurrentPositionFromAnchor(mediaUs, ALooper::GetNowUs());
-}
-
-// Called on only renderer's thread.
-status_t NuPlayer::Renderer::getCurrentPositionOnLooper(int64_t *mediaUs) {
- return getCurrentPositionOnLooper(mediaUs, ALooper::GetNowUs());
-}
-
-// Called on only renderer's thread.
-// Since mPaused and mPausePositionMediaTimeUs are changed only on renderer's
-// thread, no need to acquire mLock.
-status_t NuPlayer::Renderer::getCurrentPositionOnLooper(
- int64_t *mediaUs, int64_t nowUs, bool allowPastQueuedVideo) {
- int64_t currentPositionUs;
- if (getCurrentPositionIfPaused_l(¤tPositionUs)) {
- *mediaUs = currentPositionUs;
- return OK;
- }
- return getCurrentPositionFromAnchor(mediaUs, nowUs, allowPastQueuedVideo);
-}
-
-// Called either with mLock acquired or on renderer's thread.
-bool NuPlayer::Renderer::getCurrentPositionIfPaused_l(int64_t *mediaUs) {
- if (!mPaused || mPausePositionMediaTimeUs < 0ll) {
- return false;
- }
- *mediaUs = mPausePositionMediaTimeUs;
- return true;
-}
-
// Called on any threads.
-status_t NuPlayer::Renderer::getCurrentPositionFromAnchor(
- int64_t *mediaUs, int64_t nowUs, bool allowPastQueuedVideo) {
- Mutex::Autolock autoLock(mTimeLock);
- if (!mHasAudio && !mHasVideo) {
+status_t NuPlayer::Renderer::getCurrentPosition(int64_t *mediaUs) {
+ int64_t currentTimeUs = mMediaClock->getTimeMedia(ALooper::GetNowUs());
+ if (currentTimeUs == -1) {
return NO_INIT;
}
-
- if (mAnchorTimeMediaUs < 0) {
- return NO_INIT;
- }
-
- int64_t positionUs = (nowUs - mAnchorTimeRealUs) + mAnchorTimeMediaUs;
-
- if (mPauseStartedTimeRealUs != -1) {
- positionUs -= (nowUs - mPauseStartedTimeRealUs);
- }
-
- // limit position to the last queued media time (for video only stream
- // position will be discrete as we don't know how long each frame lasts)
- if (mAnchorMaxMediaUs >= 0 && !allowPastQueuedVideo) {
- if (positionUs > mAnchorMaxMediaUs) {
- positionUs = mAnchorMaxMediaUs;
- }
- }
-
- if (positionUs < mAudioFirstAnchorTimeMediaUs) {
- positionUs = mAudioFirstAnchorTimeMediaUs;
- }
-
- *mediaUs = (positionUs <= 0) ? 0 : positionUs;
+ *mediaUs = currentTimeUs;
return OK;
}
-void NuPlayer::Renderer::setHasMedia(bool audio) {
- Mutex::Autolock autoLock(mTimeLock);
- if (audio) {
- mHasAudio = true;
- } else {
- mHasVideo = true;
- }
+void NuPlayer::Renderer::clearAudioFirstAnchorTime_l() {
+ mAudioFirstAnchorTimeMediaUs = -1;
+ mMediaClock->setStartingTimeMedia(-1);
}
-void NuPlayer::Renderer::setAudioFirstAnchorTime(int64_t mediaUs) {
- Mutex::Autolock autoLock(mTimeLock);
- mAudioFirstAnchorTimeMediaUs = mediaUs;
-}
-
-void NuPlayer::Renderer::setAudioFirstAnchorTimeIfNeeded(int64_t mediaUs) {
- Mutex::Autolock autoLock(mTimeLock);
+void NuPlayer::Renderer::setAudioFirstAnchorTimeIfNeeded_l(int64_t mediaUs) {
if (mAudioFirstAnchorTimeMediaUs == -1) {
mAudioFirstAnchorTimeMediaUs = mediaUs;
+ mMediaClock->setStartingTimeMedia(mediaUs);
}
}
-void NuPlayer::Renderer::setAnchorTime(
- int64_t mediaUs, int64_t realUs, int64_t numFramesWritten, bool resume) {
- Mutex::Autolock autoLock(mTimeLock);
- mAnchorTimeMediaUs = mediaUs;
- mAnchorTimeRealUs = realUs;
- mAnchorNumFramesWritten = numFramesWritten;
- if (resume) {
- mPauseStartedTimeRealUs = -1;
- }
+void NuPlayer::Renderer::clearAnchorTime_l() {
+ mMediaClock->clearAnchor();
+ mAnchorTimeMediaUs = -1;
+ mAnchorNumFramesWritten = -1;
}
void NuPlayer::Renderer::setVideoLateByUs(int64_t lateUs) {
- Mutex::Autolock autoLock(mTimeLock);
+ Mutex::Autolock autoLock(mLock);
mVideoLateByUs = lateUs;
}
int64_t NuPlayer::Renderer::getVideoLateByUs() {
- Mutex::Autolock autoLock(mTimeLock);
+ Mutex::Autolock autoLock(mLock);
return mVideoLateByUs;
}
-void NuPlayer::Renderer::setPauseStartedTimeRealUs(int64_t realUs) {
- Mutex::Autolock autoLock(mTimeLock);
- mPauseStartedTimeRealUs = realUs;
-}
-
status_t NuPlayer::Renderer::openAudioSink(
const sp<AMessage> &format,
bool offloadOnly,
@@ -384,8 +291,8 @@
case kWhatDrainAudioQueue:
{
int32_t generation;
- CHECK(msg->findInt32("generation", &generation));
- if (generation != mAudioQueueGeneration) {
+ CHECK(msg->findInt32("drainGeneration", &generation));
+ if (generation != getDrainGeneration(true /* audio */)) {
break;
}
@@ -407,9 +314,7 @@
// Let's give it more data after about half that time
// has elapsed.
- // kWhatDrainAudioQueue is used for non-offloading mode,
- // and mLock is used only for offloading mode. Therefore,
- // no need to acquire mLock here.
+ Mutex::Autolock autoLock(mLock);
postDrainAudioQueue_l(delayUs / 2);
}
break;
@@ -418,8 +323,8 @@
case kWhatDrainVideoQueue:
{
int32_t generation;
- CHECK(msg->findInt32("generation", &generation));
- if (generation != mVideoQueueGeneration) {
+ CHECK(msg->findInt32("drainGeneration", &generation));
+ if (generation != getDrainGeneration(false /* audio */)) {
break;
}
@@ -427,22 +332,20 @@
onDrainVideoQueue();
- Mutex::Autolock autoLock(mLock);
- postDrainVideoQueue_l();
+ postDrainVideoQueue();
break;
}
case kWhatPostDrainVideoQueue:
{
int32_t generation;
- CHECK(msg->findInt32("generation", &generation));
- if (generation != mVideoQueueGeneration) {
+ CHECK(msg->findInt32("drainGeneration", &generation));
+ if (generation != getDrainGeneration(false /* audio */)) {
break;
}
mDrainVideoQueuePending = false;
- Mutex::Autolock autoLock(mLock);
- postDrainVideoQueue_l();
+ postDrainVideoQueue();
break;
}
@@ -464,12 +367,6 @@
break;
}
- case kWhatAudioSinkChanged:
- {
- onAudioSinkChanged();
- break;
- }
-
case kWhatDisableOffloadAudio:
{
onDisableOffloadAudio();
@@ -511,7 +408,7 @@
case kWhatAudioOffloadPauseTimeout:
{
int32_t generation;
- CHECK(msg->findInt32("generation", &generation));
+ CHECK(msg->findInt32("drainGeneration", &generation));
if (generation != mAudioOffloadPauseTimeoutGeneration) {
break;
}
@@ -539,18 +436,18 @@
mDrainAudioQueuePending = true;
sp<AMessage> msg = new AMessage(kWhatDrainAudioQueue, id());
- msg->setInt32("generation", mAudioQueueGeneration);
+ msg->setInt32("drainGeneration", mAudioDrainGeneration);
msg->post(delayUs);
}
-void NuPlayer::Renderer::prepareForMediaRenderingStart() {
- mAudioRenderingStartGeneration = mAudioQueueGeneration;
- mVideoRenderingStartGeneration = mVideoQueueGeneration;
+void NuPlayer::Renderer::prepareForMediaRenderingStart_l() {
+ mAudioRenderingStartGeneration = mAudioDrainGeneration;
+ mVideoRenderingStartGeneration = mVideoDrainGeneration;
}
-void NuPlayer::Renderer::notifyIfMediaRenderingStarted() {
- if (mVideoRenderingStartGeneration == mVideoQueueGeneration &&
- mAudioRenderingStartGeneration == mAudioQueueGeneration) {
+void NuPlayer::Renderer::notifyIfMediaRenderingStarted_l() {
+ if (mVideoRenderingStartGeneration == mVideoDrainGeneration &&
+ mAudioRenderingStartGeneration == mAudioDrainGeneration) {
mVideoRenderingStartGeneration = -1;
mAudioRenderingStartGeneration = -1;
@@ -618,7 +515,7 @@
int64_t mediaTimeUs;
CHECK(entry->mBuffer->meta()->findInt64("timeUs", &mediaTimeUs));
ALOGV("rendering audio at media time %.2f secs", mediaTimeUs / 1E6);
- setAudioFirstAnchorTimeIfNeeded(mediaTimeUs);
+ setAudioFirstAnchorTimeIfNeeded_l(mediaTimeUs);
}
size_t copy = entry->mBuffer->size() - entry->mOffset;
@@ -638,17 +535,18 @@
entry = NULL;
}
sizeCopied += copy;
- notifyIfMediaRenderingStarted();
+
+ notifyIfMediaRenderingStarted_l();
}
if (mAudioFirstAnchorTimeMediaUs >= 0) {
int64_t nowUs = ALooper::GetNowUs();
- setAnchorTime(mAudioFirstAnchorTimeMediaUs, nowUs - getPlayedOutAudioDurationUs(nowUs));
+ // we don't know how much data we are queueing for offloaded tracks.
+ mMediaClock->updateAnchor(mAudioFirstAnchorTimeMediaUs,
+ nowUs - getPlayedOutAudioDurationUs(nowUs),
+ INT64_MAX);
}
- // we don't know how much data we are queueing for offloaded tracks
- mAnchorMaxMediaUs = -1;
-
if (hasEOS) {
(new AMessage(kWhatStopAudioSink, id()))->post();
}
@@ -733,7 +631,10 @@
size_t copiedFrames = written / mAudioSink->frameSize();
mNumFramesWritten += copiedFrames;
- notifyIfMediaRenderingStarted();
+ {
+ Mutex::Autolock autoLock(mLock);
+ notifyIfMediaRenderingStarted_l();
+ }
if (written != (ssize_t)copy) {
// A short count was received from AudioSink::write()
@@ -756,10 +657,15 @@
break;
}
}
- mAnchorMaxMediaUs =
- mAnchorTimeMediaUs +
- (int64_t)(max((long long)mNumFramesWritten - mAnchorNumFramesWritten, 0LL)
- * 1000LL * mAudioSink->msecsPerFrame());
+ int64_t maxTimeMedia;
+ {
+ Mutex::Autolock autoLock(mLock);
+ maxTimeMedia =
+ mAnchorTimeMediaUs +
+ (int64_t)(max((long long)mNumFramesWritten - mAnchorNumFramesWritten, 0LL)
+ * 1000LL * mAudioSink->msecsPerFrame());
+ }
+ mMediaClock->updateMaxTimeMedia(maxTimeMedia);
return !mAudioQueue.empty();
}
@@ -771,31 +677,35 @@
}
int64_t NuPlayer::Renderer::getRealTimeUs(int64_t mediaTimeUs, int64_t nowUs) {
- int64_t currentPositionUs;
- if (mPaused || getCurrentPositionOnLooper(
- ¤tPositionUs, nowUs, true /* allowPastQueuedVideo */) != OK) {
- // If failed to get current position, e.g. due to audio clock is not ready, then just
- // play out video immediately without delay.
+ int64_t currentPositionUs =
+ mMediaClock->getTimeMedia(nowUs, true /* allowPastMaxTime */);
+ if (currentPositionUs == -1) {
+ // If failed to get current position, e.g. due to audio clock is
+ // not ready, then just play out video immediately without delay.
return nowUs;
}
return (mediaTimeUs - currentPositionUs) + nowUs;
}
void NuPlayer::Renderer::onNewAudioMediaTime(int64_t mediaTimeUs) {
+ Mutex::Autolock autoLock(mLock);
// TRICKY: vorbis decoder generates multiple frames with the same
// timestamp, so only update on the first frame with a given timestamp
if (mediaTimeUs == mAnchorTimeMediaUs) {
return;
}
- setAudioFirstAnchorTimeIfNeeded(mediaTimeUs);
+ setAudioFirstAnchorTimeIfNeeded_l(mediaTimeUs);
int64_t nowUs = ALooper::GetNowUs();
- setAnchorTime(
- mediaTimeUs, nowUs + getPendingAudioPlayoutDurationUs(nowUs), mNumFramesWritten);
+ mMediaClock->updateAnchor(mediaTimeUs,
+ nowUs + getPendingAudioPlayoutDurationUs(nowUs),
+ mediaTimeUs);
+ mAnchorTimeMediaUs = mediaTimeUs;
}
-void NuPlayer::Renderer::postDrainVideoQueue_l() {
+// Called without mLock acquired.
+void NuPlayer::Renderer::postDrainVideoQueue() {
if (mDrainVideoQueuePending
- || mSyncQueues
+ || getSyncQueues()
|| (mPaused && mVideoSampleReceived)) {
return;
}
@@ -807,7 +717,7 @@
QueueEntry &entry = *mVideoQueue.begin();
sp<AMessage> msg = new AMessage(kWhatDrainVideoQueue, id());
- msg->setInt32("generation", mVideoQueueGeneration);
+ msg->setInt32("drainGeneration", getDrainGeneration(false /* audio */));
if (entry.mBuffer == NULL) {
// EOS doesn't carry a timestamp.
@@ -827,16 +737,19 @@
int64_t mediaTimeUs;
CHECK(entry.mBuffer->meta()->findInt64("timeUs", &mediaTimeUs));
- if (mAnchorTimeMediaUs < 0) {
- setAnchorTime(mediaTimeUs, nowUs);
- mPausePositionMediaTimeUs = mediaTimeUs;
- mAnchorMaxMediaUs = mediaTimeUs;
- realTimeUs = nowUs;
- } else {
- realTimeUs = getRealTimeUs(mediaTimeUs, nowUs);
+ {
+ Mutex::Autolock autoLock(mLock);
+ if (mAnchorTimeMediaUs < 0) {
+ mMediaClock->updateAnchor(mediaTimeUs, nowUs, mediaTimeUs);
+ mAnchorTimeMediaUs = mediaTimeUs;
+ realTimeUs = nowUs;
+ } else {
+ realTimeUs = getRealTimeUs(mediaTimeUs, nowUs);
+ }
}
if (!mHasAudio) {
- mAnchorMaxMediaUs = mediaTimeUs + 100000; // smooth out videos >= 10fps
+ // smooth out videos >= 10fps
+ mMediaClock->updateMaxTimeMedia(mediaTimeUs + 100000);
}
// Heuristics to handle situation when media time changed without a
@@ -917,14 +830,15 @@
} else {
ALOGV("rendering video at media time %.2f secs",
(mFlags & FLAG_REAL_TIME ? realTimeUs :
- (realTimeUs + mAnchorTimeMediaUs - mAnchorTimeRealUs)) / 1E6);
+ mMediaClock->getTimeMedia(realTimeUs)) / 1E6);
}
} else {
setVideoLateByUs(0);
if (!mVideoSampleReceived && !mHasAudio) {
// This will ensure that the first frame after a flush won't be used as anchor
// when renderer is in paused state, because resume can happen any time after seek.
- setAnchorTime(-1, -1);
+ Mutex::Autolock autoLock(mLock);
+ clearAnchorTime_l();
}
}
@@ -941,7 +855,8 @@
mVideoRenderingStarted = true;
notifyVideoRenderingStart();
}
- notifyIfMediaRenderingStarted();
+ Mutex::Autolock autoLock(mLock);
+ notifyIfMediaRenderingStarted_l();
}
}
@@ -967,7 +882,15 @@
int32_t audio;
CHECK(msg->findInt32("audio", &audio));
- setHasMedia(audio);
+ if (dropBufferIfStale(audio, msg)) {
+ return;
+ }
+
+ if (audio) {
+ mHasAudio = true;
+ } else {
+ mHasVideo = true;
+ }
if (mHasVideo) {
if (mVideoScheduler == NULL) {
@@ -976,10 +899,6 @@
}
}
- if (dropBufferWhileFlushing(audio, msg)) {
- return;
- }
-
sp<ABuffer> buffer;
CHECK(msg->findBuffer("buffer", &buffer));
@@ -993,15 +912,16 @@
entry.mFinalResult = OK;
entry.mBufferOrdinal = ++mTotalBuffersQueued;
- Mutex::Autolock autoLock(mLock);
if (audio) {
+ Mutex::Autolock autoLock(mLock);
mAudioQueue.push_back(entry);
postDrainAudioQueue_l();
} else {
mVideoQueue.push_back(entry);
- postDrainVideoQueue_l();
+ postDrainVideoQueue();
}
+ Mutex::Autolock autoLock(mLock);
if (!mSyncQueues || mAudioQueue.empty() || mVideoQueue.empty()) {
return;
}
@@ -1050,7 +970,9 @@
}
if (!mVideoQueue.empty()) {
- postDrainVideoQueue_l();
+ mLock.unlock();
+ postDrainVideoQueue();
+ mLock.lock();
}
}
@@ -1058,7 +980,7 @@
int32_t audio;
CHECK(msg->findInt32("audio", &audio));
- if (dropBufferWhileFlushing(audio, msg)) {
+ if (dropBufferIfStale(audio, msg)) {
return;
}
@@ -1069,19 +991,20 @@
entry.mOffset = 0;
entry.mFinalResult = finalResult;
- Mutex::Autolock autoLock(mLock);
if (audio) {
+ Mutex::Autolock autoLock(mLock);
if (mAudioQueue.empty() && mSyncQueues) {
syncQueuesDone_l();
}
mAudioQueue.push_back(entry);
postDrainAudioQueue_l();
} else {
- if (mVideoQueue.empty() && mSyncQueues) {
+ if (mVideoQueue.empty() && getSyncQueues()) {
+ Mutex::Autolock autoLock(mLock);
syncQueuesDone_l();
}
mVideoQueue.push_back(entry);
- postDrainVideoQueue_l();
+ postDrainVideoQueue();
}
}
@@ -1090,31 +1013,25 @@
CHECK(msg->findInt32("audio", &audio));
{
- Mutex::Autolock autoLock(mFlushLock);
+ Mutex::Autolock autoLock(mLock);
if (audio) {
- mFlushingAudio = false;
notifyComplete = mNotifyCompleteAudio;
mNotifyCompleteAudio = false;
} else {
- mFlushingVideo = false;
notifyComplete = mNotifyCompleteVideo;
mNotifyCompleteVideo = false;
}
- }
- // If we're currently syncing the queues, i.e. dropping audio while
- // aligning the first audio/video buffer times and only one of the
- // two queues has data, we may starve that queue by not requesting
- // more buffers from the decoder. If the other source then encounters
- // a discontinuity that leads to flushing, we'll never find the
- // corresponding discontinuity on the other queue.
- // Therefore we'll stop syncing the queues if at least one of them
- // is flushed.
- {
- Mutex::Autolock autoLock(mLock);
- syncQueuesDone_l();
- setPauseStartedTimeRealUs(-1);
- setAnchorTime(-1, -1);
+ // If we're currently syncing the queues, i.e. dropping audio while
+ // aligning the first audio/video buffer times and only one of the
+ // two queues has data, we may starve that queue by not requesting
+ // more buffers from the decoder. If the other source then encounters
+ // a discontinuity that leads to flushing, we'll never find the
+ // corresponding discontinuity on the other queue.
+ // Therefore we'll stop syncing the queues if at least one of them
+ // is flushed.
+ syncQueuesDone_l();
+ clearAnchorTime_l();
}
ALOGV("flushing %s", audio ? "audio" : "video");
@@ -1123,11 +1040,11 @@
Mutex::Autolock autoLock(mLock);
flushQueue(&mAudioQueue);
- ++mAudioQueueGeneration;
- prepareForMediaRenderingStart();
+ ++mAudioDrainGeneration;
+ prepareForMediaRenderingStart_l();
if (offloadingAudio()) {
- setAudioFirstAnchorTime(-1);
+ clearAudioFirstAnchorTime_l();
}
}
@@ -1142,13 +1059,14 @@
flushQueue(&mVideoQueue);
mDrainVideoQueuePending = false;
- ++mVideoQueueGeneration;
if (mVideoScheduler != NULL) {
mVideoScheduler->restart();
}
- prepareForMediaRenderingStart();
+ Mutex::Autolock autoLock(mLock);
+ ++mVideoDrainGeneration;
+ prepareForMediaRenderingStart_l();
}
mVideoSampleReceived = false;
@@ -1178,20 +1096,12 @@
notify->post();
}
-bool NuPlayer::Renderer::dropBufferWhileFlushing(
+bool NuPlayer::Renderer::dropBufferIfStale(
bool audio, const sp<AMessage> &msg) {
- bool flushing = false;
+ int32_t queueGeneration;
+ CHECK(msg->findInt32("queueGeneration", &queueGeneration));
- {
- Mutex::Autolock autoLock(mFlushLock);
- if (audio) {
- flushing = mFlushingAudio;
- } else {
- flushing = mFlushingVideo;
- }
- }
-
- if (!flushing) {
+ if (queueGeneration == getQueueGeneration(audio)) {
return false;
}
@@ -1209,7 +1119,10 @@
}
CHECK(!mDrainAudioQueuePending);
mNumFramesWritten = 0;
- mAnchorNumFramesWritten = -1;
+ {
+ Mutex::Autolock autoLock(mLock);
+ mAnchorNumFramesWritten = -1;
+ }
uint32_t written;
if (mAudioSink->getFramesWritten(&written) == OK) {
mNumFramesWritten = written;
@@ -1219,13 +1132,13 @@
void NuPlayer::Renderer::onDisableOffloadAudio() {
Mutex::Autolock autoLock(mLock);
mFlags &= ~FLAG_OFFLOAD_AUDIO;
- ++mAudioQueueGeneration;
+ ++mAudioDrainGeneration;
}
void NuPlayer::Renderer::onEnableOffloadAudio() {
Mutex::Autolock autoLock(mLock);
mFlags |= FLAG_OFFLOAD_AUDIO;
- ++mAudioQueueGeneration;
+ ++mAudioDrainGeneration;
}
void NuPlayer::Renderer::onPause() {
@@ -1234,25 +1147,13 @@
return;
}
int64_t currentPositionUs;
- int64_t pausePositionMediaTimeUs;
- if (getCurrentPositionFromAnchor(
- ¤tPositionUs, ALooper::GetNowUs()) == OK) {
- pausePositionMediaTimeUs = currentPositionUs;
- } else {
- // Set paused position to -1 (unavailabe) if we don't have anchor time
- // This could happen if client does a seekTo() immediately followed by
- // pause(). Renderer will be flushed with anchor time cleared. We don't
- // want to leave stale value in mPausePositionMediaTimeUs.
- pausePositionMediaTimeUs = -1;
- }
{
Mutex::Autolock autoLock(mLock);
- mPausePositionMediaTimeUs = pausePositionMediaTimeUs;
- ++mAudioQueueGeneration;
- ++mVideoQueueGeneration;
- prepareForMediaRenderingStart();
+ ++mAudioDrainGeneration;
+ ++mVideoDrainGeneration;
+ prepareForMediaRenderingStart_l();
mPaused = true;
- setPauseStartedTimeRealUs(ALooper::GetNowUs());
+ mMediaClock->pause();
}
mDrainAudioQueuePending = false;
@@ -1277,21 +1178,18 @@
mAudioSink->start();
}
- Mutex::Autolock autoLock(mLock);
- mPaused = false;
- if (mPauseStartedTimeRealUs != -1) {
- int64_t newAnchorRealUs =
- mAnchorTimeRealUs + ALooper::GetNowUs() - mPauseStartedTimeRealUs;
- setAnchorTime(
- mAnchorTimeMediaUs, newAnchorRealUs, mAnchorNumFramesWritten, true /* resume */);
- }
+ {
+ Mutex::Autolock autoLock(mLock);
+ mPaused = false;
+ mMediaClock->resume();
- if (!mAudioQueue.empty()) {
- postDrainAudioQueue_l();
+ if (!mAudioQueue.empty()) {
+ postDrainAudioQueue_l();
+ }
}
if (!mVideoQueue.empty()) {
- postDrainVideoQueue_l();
+ postDrainVideoQueue();
}
}
@@ -1302,6 +1200,21 @@
mVideoScheduler->init(fps);
}
+int32_t NuPlayer::Renderer::getQueueGeneration(bool audio) {
+ Mutex::Autolock autoLock(mLock);
+ return (audio ? mAudioQueueGeneration : mVideoQueueGeneration);
+}
+
+int32_t NuPlayer::Renderer::getDrainGeneration(bool audio) {
+ Mutex::Autolock autoLock(mLock);
+ return (audio ? mAudioDrainGeneration : mVideoDrainGeneration);
+}
+
+bool NuPlayer::Renderer::getSyncQueues() {
+ Mutex::Autolock autoLock(mLock);
+ return mSyncQueues;
+}
+
// TODO: Remove unnecessary calls to getPlayedOutAudioDurationUs()
// as it acquires locks and may query the audio driver.
//
@@ -1373,7 +1286,7 @@
mAudioOffloadTornDown = true;
int64_t currentPositionUs;
- if (getCurrentPositionOnLooper(¤tPositionUs) != OK) {
+ if (getCurrentPosition(¤tPositionUs) != OK) {
currentPositionUs = 0;
}
@@ -1391,7 +1304,7 @@
if (offloadingAudio()) {
mWakeLock->acquire();
sp<AMessage> msg = new AMessage(kWhatAudioOffloadPauseTimeout, id());
- msg->setInt32("generation", mAudioOffloadPauseTimeoutGeneration);
+ msg->setInt32("drainGeneration", mAudioOffloadPauseTimeoutGeneration);
msg->post(kOffloadPauseMaxUs);
}
}
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.h b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.h
index 003d1d0..faf3b3f 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.h
@@ -24,6 +24,7 @@
struct ABuffer;
class AWakeLock;
+struct MediaClock;
struct VideoFrameScheduler;
struct NuPlayer::Renderer : public AHandler {
@@ -61,16 +62,8 @@
void setVideoFrameRate(float fps);
- // Following setters and getters are protected by mTimeLock.
status_t getCurrentPosition(int64_t *mediaUs);
- void setHasMedia(bool audio);
- void setAudioFirstAnchorTime(int64_t mediaUs);
- void setAudioFirstAnchorTimeIfNeeded(int64_t mediaUs);
- void setAnchorTime(
- int64_t mediaUs, int64_t realUs, int64_t numFramesWritten = -1, bool resume = false);
- void setVideoLateByUs(int64_t lateUs);
int64_t getVideoLateByUs();
- void setPauseStartedTimeRealUs(int64_t realUs);
status_t openAudioSink(
const sp<AMessage> &format,
@@ -108,7 +101,6 @@
kWhatQueueBuffer = 'queB',
kWhatQueueEOS = 'qEOS',
kWhatFlush = 'flus',
- kWhatAudioSinkChanged = 'auSC',
kWhatPause = 'paus',
kWhatResume = 'resm',
kWhatOpenAudioSink = 'opnA',
@@ -142,26 +134,17 @@
bool mDrainVideoQueuePending;
int32_t mAudioQueueGeneration;
int32_t mVideoQueueGeneration;
+ int32_t mAudioDrainGeneration;
+ int32_t mVideoDrainGeneration;
- Mutex mTimeLock;
- // |mTimeLock| protects the following 7 member vars that are related to time.
- // Note: those members are only written on Renderer thread, so reading on Renderer thread
- // doesn't need to be protected. Otherwise accessing those members must be protected by
- // |mTimeLock|.
- // TODO: move those members to a seperated media clock class.
+ sp<MediaClock> mMediaClock;
int64_t mAudioFirstAnchorTimeMediaUs;
int64_t mAnchorTimeMediaUs;
- int64_t mAnchorTimeRealUs;
int64_t mAnchorNumFramesWritten;
- int64_t mAnchorMaxMediaUs;
int64_t mVideoLateByUs;
bool mHasAudio;
bool mHasVideo;
- int64_t mPauseStartedTimeRealUs;
- Mutex mFlushLock; // protects the following 2 member vars.
- bool mFlushingAudio;
- bool mFlushingVideo;
bool mNotifyCompleteAudio;
bool mNotifyCompleteVideo;
@@ -169,7 +152,6 @@
// modified on only renderer's thread.
bool mPaused;
- int64_t mPausePositionMediaTimeUs;
bool mVideoSampleReceived;
bool mVideoRenderingStarted;
@@ -211,14 +193,19 @@
int64_t getPlayedOutAudioDurationUs(int64_t nowUs);
void postDrainAudioQueue_l(int64_t delayUs = 0);
+ void clearAnchorTime_l();
+ void clearAudioFirstAnchorTime_l();
+ void setAudioFirstAnchorTimeIfNeeded_l(int64_t mediaUs);
+ void setVideoLateByUs(int64_t lateUs);
+
void onNewAudioMediaTime(int64_t mediaTimeUs);
int64_t getRealTimeUs(int64_t mediaTimeUs, int64_t nowUs);
void onDrainVideoQueue();
- void postDrainVideoQueue_l();
+ void postDrainVideoQueue();
- void prepareForMediaRenderingStart();
- void notifyIfMediaRenderingStarted();
+ void prepareForMediaRenderingStart_l();
+ void notifyIfMediaRenderingStarted_l();
void onQueueBuffer(const sp<AMessage> &msg);
void onQueueEOS(const sp<AMessage> &msg);
@@ -229,6 +216,9 @@
void onPause();
void onResume();
void onSetVideoFrameRate(float fps);
+ int32_t getQueueGeneration(bool audio);
+ int32_t getDrainGeneration(bool audio);
+ bool getSyncQueues();
void onAudioOffloadTearDown(AudioOffloadTearDownReason reason);
status_t onOpenAudioSink(
const sp<AMessage> &format,
@@ -245,7 +235,7 @@
void notifyAudioOffloadTearDown();
void flushQueue(List<QueueEntry> *queue);
- bool dropBufferWhileFlushing(bool audio, const sp<AMessage> &msg);
+ bool dropBufferIfStale(bool audio, const sp<AMessage> &msg);
void syncQueuesDone_l();
bool offloadingAudio() const { return (mFlags & FLAG_OFFLOAD_AUDIO) != 0; }
diff --git a/media/libnbaio/Android.mk b/media/libnbaio/Android.mk
index 9707c4a..1353f28 100644
--- a/media/libnbaio/Android.mk
+++ b/media/libnbaio/Android.mk
@@ -11,7 +11,6 @@
MonoPipeReader.cpp \
Pipe.cpp \
PipeReader.cpp \
- roundup.c \
SourceAudioBufferProvider.cpp
LOCAL_SRC_FILES += NBLog.cpp
@@ -27,12 +26,13 @@
LOCAL_MODULE := libnbaio
LOCAL_SHARED_LIBRARIES := \
+ libaudioutils \
libbinder \
libcommon_time_client \
libcutils \
libutils \
liblog
-LOCAL_STATIC_LIBRARIES += libinstantssq
+LOCAL_C_INCLUDES := $(call include-path-for, audio-utils)
include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libnbaio/MonoPipe.cpp b/media/libnbaio/MonoPipe.cpp
index 0b65861..129e9ef 100644
--- a/media/libnbaio/MonoPipe.cpp
+++ b/media/libnbaio/MonoPipe.cpp
@@ -27,7 +27,7 @@
#include <utils/Trace.h>
#include <media/AudioBufferProvider.h>
#include <media/nbaio/MonoPipe.h>
-#include <media/nbaio/roundup.h>
+#include <audio_utils/roundup.h>
namespace android {
diff --git a/media/libnbaio/MonoPipeReader.cpp b/media/libnbaio/MonoPipeReader.cpp
index de82229..e4d3ed8 100644
--- a/media/libnbaio/MonoPipeReader.cpp
+++ b/media/libnbaio/MonoPipeReader.cpp
@@ -39,7 +39,7 @@
return NEGOTIATE;
}
ssize_t ret = android_atomic_acquire_load(&mPipe->mRear) - mPipe->mFront;
- ALOG_ASSERT((0 <= ret) && (ret <= mMaxFrames));
+ ALOG_ASSERT((0 <= ret) && ((size_t) ret <= mPipe->mMaxFrames));
return ret;
}
diff --git a/media/libnbaio/Pipe.cpp b/media/libnbaio/Pipe.cpp
index 6e0ec8c..13f211d 100644
--- a/media/libnbaio/Pipe.cpp
+++ b/media/libnbaio/Pipe.cpp
@@ -21,7 +21,7 @@
#include <cutils/compiler.h>
#include <utils/Log.h>
#include <media/nbaio/Pipe.h>
-#include <media/nbaio/roundup.h>
+#include <audio_utils/roundup.h>
namespace android {
diff --git a/media/libnbaio/roundup.c b/media/libnbaio/roundup.c
deleted file mode 100644
index 1d552d1..0000000
--- a/media/libnbaio/roundup.c
+++ /dev/null
@@ -1,32 +0,0 @@
-/*
- * Copyright (C) 2012 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include <media/nbaio/roundup.h>
-
-unsigned roundup(unsigned v)
-{
- // __builtin_clz is undefined for zero input
- if (v == 0) {
- v = 1;
- }
- int lz = __builtin_clz((int) v);
- unsigned rounded = ((unsigned) 0x80000000) >> lz;
- // 0x800000001 and higher are actually rounded _down_ to prevent overflow
- if (v > rounded && lz > 0) {
- rounded <<= 1;
- }
- return rounded;
-}
diff --git a/media/libstagefright/AACWriter.cpp b/media/libstagefright/AACWriter.cpp
index 2e41d80..9d90dbd 100644
--- a/media/libstagefright/AACWriter.cpp
+++ b/media/libstagefright/AACWriter.cpp
@@ -36,33 +36,19 @@
namespace android {
-AACWriter::AACWriter(const char *filename)
- : mFd(-1),
- mInitCheck(NO_INIT),
- mStarted(false),
- mPaused(false),
- mResumed(false),
- mChannelCount(-1),
- mSampleRate(-1),
- mAACProfile(OMX_AUDIO_AACObjectLC) {
-
- ALOGV("AACWriter Constructor");
-
- mFd = open(filename, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
- if (mFd >= 0) {
- mInitCheck = OK;
- }
-}
-
AACWriter::AACWriter(int fd)
: mFd(dup(fd)),
mInitCheck(mFd < 0? NO_INIT: OK),
mStarted(false),
mPaused(false),
mResumed(false),
+ mThread(0),
+ mEstimatedSizeBytes(0),
+ mEstimatedDurationUs(0),
mChannelCount(-1),
mSampleRate(-1),
- mAACProfile(OMX_AUDIO_AACObjectLC) {
+ mAACProfile(OMX_AUDIO_AACObjectLC),
+ mFrameDurationUs(0) {
}
AACWriter::~AACWriter() {
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index c8806ae..7d313e0 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -419,6 +419,7 @@
mMetaDataBuffersToSubmit(0),
mRepeatFrameDelayUs(-1ll),
mMaxPtsGapUs(-1ll),
+ mMaxFps(-1),
mTimePerFrameUs(-1ll),
mTimePerCaptureUs(-1ll),
mCreateInputBuffersSuspended(false),
@@ -1259,6 +1260,10 @@
mMaxPtsGapUs = -1ll;
}
+ if (!msg->findFloat("max-fps-to-encoder", &mMaxFps)) {
+ mMaxFps = -1;
+ }
+
if (!msg->findInt64("time-lapse", &mTimePerCaptureUs)) {
mTimePerCaptureUs = -1ll;
}
@@ -5110,6 +5115,21 @@
}
}
+ if (err == OK && mCodec->mMaxFps > 0) {
+ err = mCodec->mOMX->setInternalOption(
+ mCodec->mNode,
+ kPortIndexInput,
+ IOMX::INTERNAL_OPTION_MAX_FPS,
+ &mCodec->mMaxFps,
+ sizeof(mCodec->mMaxFps));
+
+ if (err != OK) {
+ ALOGE("[%s] Unable to configure max fps (err %d)",
+ mCodec->mComponentName.c_str(),
+ err);
+ }
+ }
+
if (err == OK && mCodec->mTimePerCaptureUs > 0ll
&& mCodec->mTimePerFrameUs > 0ll) {
int64_t timeLapse[2];
diff --git a/media/libstagefright/AMRWriter.cpp b/media/libstagefright/AMRWriter.cpp
index 9aa7d95..f53d7f0 100644
--- a/media/libstagefright/AMRWriter.cpp
+++ b/media/libstagefright/AMRWriter.cpp
@@ -31,19 +31,6 @@
namespace android {
-AMRWriter::AMRWriter(const char *filename)
- : mFd(-1),
- mInitCheck(NO_INIT),
- mStarted(false),
- mPaused(false),
- mResumed(false) {
-
- mFd = open(filename, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
- if (mFd >= 0) {
- mInitCheck = OK;
- }
-}
-
AMRWriter::AMRWriter(int fd)
: mFd(dup(fd)),
mInitCheck(mFd < 0? NO_INIT: OK),
diff --git a/media/libstagefright/Android.mk b/media/libstagefright/Android.mk
index 2629afc..6d9bbae 100644
--- a/media/libstagefright/Android.mk
+++ b/media/libstagefright/Android.mk
@@ -101,6 +101,7 @@
libstagefright_color_conversion \
libstagefright_aacenc \
libstagefright_matroska \
+ libstagefright_mediafilter \
libstagefright_webm \
libstagefright_timedtext \
libvpx \
@@ -108,13 +109,14 @@
libstagefright_mpeg2ts \
libstagefright_id3 \
libFLAC \
- libmedia_helper
+ libmedia_helper \
LOCAL_SHARED_LIBRARIES += \
libstagefright_enc_common \
libstagefright_avc_common \
libstagefright_foundation \
- libdl
+ libdl \
+ libRScpp \
LOCAL_CFLAGS += -Wno-multichar
diff --git a/media/libstagefright/FileSource.cpp b/media/libstagefright/FileSource.cpp
index a7ca3da..f0db76b 100644
--- a/media/libstagefright/FileSource.cpp
+++ b/media/libstagefright/FileSource.cpp
@@ -14,6 +14,10 @@
* limitations under the License.
*/
+//#define LOG_NDEBUG 0
+#define LOG_TAG "FileSource"
+#include <utils/Log.h>
+
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/FileSource.h>
#include <sys/types.h>
diff --git a/media/libstagefright/MPEG2TSWriter.cpp b/media/libstagefright/MPEG2TSWriter.cpp
index 9856f92..4359fb9 100644
--- a/media/libstagefright/MPEG2TSWriter.cpp
+++ b/media/libstagefright/MPEG2TSWriter.cpp
@@ -480,19 +480,6 @@
init();
}
-MPEG2TSWriter::MPEG2TSWriter(const char *filename)
- : mFile(fopen(filename, "wb")),
- mWriteCookie(NULL),
- mWriteFunc(NULL),
- mStarted(false),
- mNumSourcesDone(0),
- mNumTSPacketsWritten(0),
- mNumTSPacketsBeforeMeta(0),
- mPATContinuityCounter(0),
- mPMTContinuityCounter(0) {
- init();
-}
-
MPEG2TSWriter::MPEG2TSWriter(
void *cookie,
ssize_t (*write)(void *cookie, const void *data, size_t size))
diff --git a/media/libstagefright/MPEG4Extractor.cpp b/media/libstagefright/MPEG4Extractor.cpp
index 8bf7f63..d0f42cc 100644
--- a/media/libstagefright/MPEG4Extractor.cpp
+++ b/media/libstagefright/MPEG4Extractor.cpp
@@ -354,6 +354,8 @@
MPEG4Extractor::MPEG4Extractor(const sp<DataSource> &source)
: mMoofOffset(0),
+ mMoofFound(false),
+ mMdatFound(false),
mDataSource(source),
mInitCheck(NO_INIT),
mHasVideo(false),
@@ -490,7 +492,9 @@
off64_t offset = 0;
status_t err;
- while (true) {
+ bool sawMoovOrSidx = false;
+
+ while (!(sawMoovOrSidx && (mMdatFound || mMoofFound))) {
off64_t orig_offset = offset;
err = parseChunk(&offset, 0);
@@ -502,23 +506,9 @@
ALOGE("did not advance: 0x%lld->0x%lld", orig_offset, offset);
err = ERROR_MALFORMED;
break;
- } else if (err == OK) {
- continue;
+ } else if (err == UNKNOWN_ERROR) {
+ sawMoovOrSidx = true;
}
-
- uint32_t hdr[2];
- if (mDataSource->readAt(offset, hdr, 8) < 8) {
- break;
- }
- uint32_t chunk_type = ntohl(hdr[1]);
- if (chunk_type == FOURCC('m', 'o', 'o', 'f')) {
- // store the offset of the first segment
- mMoofOffset = offset;
- } else if (chunk_type != FOURCC('m', 'd', 'a', 't')) {
- // keep parsing until we get to the data
- continue;
- }
- break;
}
if (mInitCheck == OK) {
@@ -864,6 +854,12 @@
case FOURCC('s', 'c', 'h', 'i'):
case FOURCC('e', 'd', 't', 's'):
{
+ if (chunk_type == FOURCC('m', 'o', 'o', 'f') && !mMoofFound) {
+ // store the offset of the first segment
+ mMoofFound = true;
+ mMoofOffset = *offset;
+ }
+
if (chunk_type == FOURCC('s', 't', 'b', 'l')) {
ALOGV("sampleTable chunk is %" PRIu64 " bytes long.", chunk_size);
@@ -1830,6 +1826,9 @@
case FOURCC('m', 'd', 'a', 't'):
{
ALOGV("mdat chunk, drm: %d", mIsDrm);
+
+ mMdatFound = true;
+
if (!mIsDrm) {
*offset += chunk_size;
break;
diff --git a/media/libstagefright/MPEG4Writer.cpp b/media/libstagefright/MPEG4Writer.cpp
index 9f20b1d..beb6f20 100644
--- a/media/libstagefright/MPEG4Writer.cpp
+++ b/media/libstagefright/MPEG4Writer.cpp
@@ -345,31 +345,6 @@
Track &operator=(const Track &);
};
-MPEG4Writer::MPEG4Writer(const char *filename)
- : mFd(-1),
- mInitCheck(NO_INIT),
- mIsRealTimeRecording(true),
- mUse4ByteNalLength(true),
- mUse32BitOffset(true),
- mIsFileSizeLimitExplicitlyRequested(false),
- mPaused(false),
- mStarted(false),
- mWriterThreadStarted(false),
- mOffset(0),
- mMdatOffset(0),
- mEstimatedMoovBoxSize(0),
- mInterleaveDurationUs(1000000),
- mLatitudex10000(0),
- mLongitudex10000(0),
- mAreGeoTagsAvailable(false),
- mStartTimeOffsetMs(-1) {
-
- mFd = open(filename, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
- if (mFd >= 0) {
- mInitCheck = OK;
- }
-}
-
MPEG4Writer::MPEG4Writer(int fd)
: mFd(dup(fd)),
mInitCheck(mFd < 0? NO_INIT: OK),
diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
index 6ca123a..50e6bd0 100644
--- a/media/libstagefright/MediaCodec.cpp
+++ b/media/libstagefright/MediaCodec.cpp
@@ -36,6 +36,7 @@
#include <media/stagefright/MediaCodecList.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MediaFilter.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/NativeWindowWrapper.h>
#include <private/android_filesystem_config.h>
@@ -189,7 +190,16 @@
// quickly, violating the OpenMAX specs, until that is remedied
// we need to invest in an extra looper to free the main event
// queue.
- mCodec = new ACodec;
+
+ if (nameIsType || !strncasecmp(name.c_str(), "omx.", 4)) {
+ mCodec = new ACodec;
+ } else if (!nameIsType
+ && !strncasecmp(name.c_str(), "android.filter.", 15)) {
+ mCodec = new MediaFilter;
+ } else {
+ return NAME_NOT_FOUND;
+ }
+
bool needDedicatedLooper = false;
if (nameIsType && !strncasecmp(name.c_str(), "video/", 6)) {
needDedicatedLooper = true;
diff --git a/media/libstagefright/MediaMuxer.cpp b/media/libstagefright/MediaMuxer.cpp
index c7c6f34..b13877d 100644
--- a/media/libstagefright/MediaMuxer.cpp
+++ b/media/libstagefright/MediaMuxer.cpp
@@ -38,21 +38,6 @@
namespace android {
-MediaMuxer::MediaMuxer(const char *path, OutputFormat format)
- : mFormat(format),
- mState(UNINITIALIZED) {
- if (format == OUTPUT_FORMAT_MPEG_4) {
- mWriter = new MPEG4Writer(path);
- } else if (format == OUTPUT_FORMAT_WEBM) {
- mWriter = new WebmWriter(path);
- }
-
- if (mWriter != NULL) {
- mFileMeta = new MetaData;
- mState = INITIALIZED;
- }
-}
-
MediaMuxer::MediaMuxer(int fd, OutputFormat format)
: mFormat(format),
mState(UNINITIALIZED) {
diff --git a/media/libstagefright/Utils.cpp b/media/libstagefright/Utils.cpp
index b3a79a0..c0be136 100644
--- a/media/libstagefright/Utils.cpp
+++ b/media/libstagefright/Utils.cpp
@@ -344,6 +344,28 @@
buffer->meta()->setInt32("csd", true);
buffer->meta()->setInt64("timeUs", 0);
msg->setBuffer("csd-0", buffer);
+
+ if (!meta->findData(kKeyOpusCodecDelay, &type, &data, &size)) {
+ return -EINVAL;
+ }
+
+ buffer = new ABuffer(size);
+ memcpy(buffer->data(), data, size);
+
+ buffer->meta()->setInt32("csd", true);
+ buffer->meta()->setInt64("timeUs", 0);
+ msg->setBuffer("csd-1", buffer);
+
+ if (!meta->findData(kKeyOpusSeekPreRoll, &type, &data, &size)) {
+ return -EINVAL;
+ }
+
+ buffer = new ABuffer(size);
+ memcpy(buffer->data(), data, size);
+
+ buffer->meta()->setInt32("csd", true);
+ buffer->meta()->setInt64("timeUs", 0);
+ msg->setBuffer("csd-2", buffer);
}
*format = msg;
diff --git a/media/libstagefright/codecs/on2/dec/SoftVPX.cpp b/media/libstagefright/codecs/on2/dec/SoftVPX.cpp
index 8a95643..6e6a78a 100644
--- a/media/libstagefright/codecs/on2/dec/SoftVPX.cpp
+++ b/media/libstagefright/codecs/on2/dec/SoftVPX.cpp
@@ -38,7 +38,10 @@
NULL /* profileLevels */, 0 /* numProfileLevels */,
320 /* width */, 240 /* height */, callbacks, appData, component),
mMode(codingType == OMX_VIDEO_CodingVP8 ? MODE_VP8 : MODE_VP9),
+ mEOSStatus(INPUT_DATA_AVAILABLE),
mCtx(NULL),
+ mFrameParallelMode(false),
+ mTimeStampIdx(0),
mImg(NULL) {
// arbitrary from avc/hevc as vpx does not specify a min compression ratio
const size_t kMinCompressionRatio = mMode == MODE_VP8 ? 2 : 4;
@@ -51,9 +54,7 @@
}
SoftVPX::~SoftVPX() {
- vpx_codec_destroy((vpx_codec_ctx_t *)mCtx);
- delete (vpx_codec_ctx_t *)mCtx;
- mCtx = NULL;
+ destroyDecoder();
}
static int GetCPUCoreCount() {
@@ -73,12 +74,19 @@
mCtx = new vpx_codec_ctx_t;
vpx_codec_err_t vpx_err;
vpx_codec_dec_cfg_t cfg;
+ vpx_codec_flags_t flags;
memset(&cfg, 0, sizeof(vpx_codec_dec_cfg_t));
+ memset(&flags, 0, sizeof(vpx_codec_flags_t));
cfg.threads = GetCPUCoreCount();
+
+ if (mFrameParallelMode) {
+ flags |= VPX_CODEC_USE_FRAME_THREADING;
+ }
+
if ((vpx_err = vpx_codec_dec_init(
(vpx_codec_ctx_t *)mCtx,
mMode == MODE_VP8 ? &vpx_codec_vp8_dx_algo : &vpx_codec_vp9_dx_algo,
- &cfg, 0))) {
+ &cfg, flags))) {
ALOGE("on2 decoder failed to initialize. (%d)", vpx_err);
return UNKNOWN_ERROR;
}
@@ -86,86 +94,155 @@
return OK;
}
+status_t SoftVPX::destroyDecoder() {
+ vpx_codec_destroy((vpx_codec_ctx_t *)mCtx);
+ delete (vpx_codec_ctx_t *)mCtx;
+ mCtx = NULL;
+ return OK;
+}
+
+bool SoftVPX::outputBuffers(bool flushDecoder, bool display, bool eos, bool *portWillReset) {
+ List<BufferInfo *> &inQueue = getPortQueue(0);
+ List<BufferInfo *> &outQueue = getPortQueue(1);
+ BufferInfo *outInfo = NULL;
+ OMX_BUFFERHEADERTYPE *outHeader = NULL;
+ vpx_codec_iter_t iter = NULL;
+
+ if (flushDecoder && mFrameParallelMode) {
+ // Flush decoder by passing NULL data ptr and 0 size.
+ // Ideally, this should never fail.
+ if (vpx_codec_decode((vpx_codec_ctx_t *)mCtx, NULL, 0, NULL, 0)) {
+ ALOGE("Failed to flush on2 decoder.");
+ return false;
+ }
+ }
+
+ if (!display) {
+ if (!flushDecoder) {
+ ALOGE("Invalid operation.");
+ return false;
+ }
+ // Drop all the decoded frames in decoder.
+ while ((mImg = vpx_codec_get_frame((vpx_codec_ctx_t *)mCtx, &iter))) {
+ }
+ return true;
+ }
+
+ while (!outQueue.empty()) {
+ if (mImg == NULL) {
+ mImg = vpx_codec_get_frame((vpx_codec_ctx_t *)mCtx, &iter);
+ if (mImg == NULL) {
+ break;
+ }
+ }
+ uint32_t width = mImg->d_w;
+ uint32_t height = mImg->d_h;
+ outInfo = *outQueue.begin();
+ outHeader = outInfo->mHeader;
+ CHECK_EQ(mImg->fmt, IMG_FMT_I420);
+ handlePortSettingsChange(portWillReset, width, height);
+ if (*portWillReset) {
+ return true;
+ }
+
+ outHeader->nOffset = 0;
+ outHeader->nFilledLen = (width * height * 3) / 2;
+ outHeader->nFlags = 0;
+ outHeader->nTimeStamp = *(OMX_TICKS *)mImg->user_priv;
+
+ uint8_t *dst = outHeader->pBuffer;
+ const uint8_t *srcY = (const uint8_t *)mImg->planes[PLANE_Y];
+ const uint8_t *srcU = (const uint8_t *)mImg->planes[PLANE_U];
+ const uint8_t *srcV = (const uint8_t *)mImg->planes[PLANE_V];
+ size_t srcYStride = mImg->stride[PLANE_Y];
+ size_t srcUStride = mImg->stride[PLANE_U];
+ size_t srcVStride = mImg->stride[PLANE_V];
+ copyYV12FrameToOutputBuffer(dst, srcY, srcU, srcV, srcYStride, srcUStride, srcVStride);
+
+ mImg = NULL;
+ outInfo->mOwnedByUs = false;
+ outQueue.erase(outQueue.begin());
+ outInfo = NULL;
+ notifyFillBufferDone(outHeader);
+ outHeader = NULL;
+ }
+
+ if (!eos) {
+ return true;
+ }
+
+ if (!outQueue.empty()) {
+ outInfo = *outQueue.begin();
+ outQueue.erase(outQueue.begin());
+ outHeader = outInfo->mHeader;
+ outHeader->nTimeStamp = 0;
+ outHeader->nFilledLen = 0;
+ outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+ outInfo->mOwnedByUs = false;
+ notifyFillBufferDone(outHeader);
+ mEOSStatus = OUTPUT_FRAMES_FLUSHED;
+ }
+ return true;
+}
+
void SoftVPX::onQueueFilled(OMX_U32 /* portIndex */) {
- if (mOutputPortSettingsChange != NONE) {
+ if (mOutputPortSettingsChange != NONE || mEOSStatus == OUTPUT_FRAMES_FLUSHED) {
return;
}
List<BufferInfo *> &inQueue = getPortQueue(0);
List<BufferInfo *> &outQueue = getPortQueue(1);
bool EOSseen = false;
+ vpx_codec_err_t err;
+ bool portWillReset = false;
- while (!inQueue.empty() && !outQueue.empty()) {
- BufferInfo *inInfo = *inQueue.begin();
- OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
-
- BufferInfo *outInfo = *outQueue.begin();
- OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
-
- if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
- EOSseen = true;
- if (inHeader->nFilledLen == 0) {
- inQueue.erase(inQueue.begin());
- inInfo->mOwnedByUs = false;
- notifyEmptyBufferDone(inHeader);
-
- outHeader->nFilledLen = 0;
- outHeader->nFlags = OMX_BUFFERFLAG_EOS;
-
- outQueue.erase(outQueue.begin());
- outInfo->mOwnedByUs = false;
- notifyFillBufferDone(outHeader);
- return;
- }
- }
-
- if (mImg == NULL) {
- if (vpx_codec_decode(
- (vpx_codec_ctx_t *)mCtx,
- inHeader->pBuffer + inHeader->nOffset,
- inHeader->nFilledLen,
- NULL,
- 0)) {
- ALOGE("on2 decoder failed to decode frame.");
-
+ while ((mEOSStatus == INPUT_EOS_SEEN || !inQueue.empty())
+ && !outQueue.empty()) {
+ // Output the pending frames that left from last port reset or decoder flush.
+ if (mEOSStatus == INPUT_EOS_SEEN || mImg != NULL) {
+ if (!outputBuffers(
+ mEOSStatus == INPUT_EOS_SEEN, true /* display */,
+ mEOSStatus == INPUT_EOS_SEEN, &portWillReset)) {
+ ALOGE("on2 decoder failed to output frame.");
notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
return;
}
- vpx_codec_iter_t iter = NULL;
- mImg = vpx_codec_get_frame((vpx_codec_ctx_t *)mCtx, &iter);
- }
-
- if (mImg != NULL) {
- CHECK_EQ(mImg->fmt, IMG_FMT_I420);
-
- uint32_t width = mImg->d_w;
- uint32_t height = mImg->d_h;
- bool portWillReset = false;
- handlePortSettingsChange(&portWillReset, width, height);
- if (portWillReset) {
+ if (portWillReset || mEOSStatus == OUTPUT_FRAMES_FLUSHED ||
+ mEOSStatus == INPUT_EOS_SEEN) {
return;
}
+ }
- outHeader->nOffset = 0;
- outHeader->nFilledLen = (width * height * 3) / 2;
- outHeader->nFlags = EOSseen ? OMX_BUFFERFLAG_EOS : 0;
- outHeader->nTimeStamp = inHeader->nTimeStamp;
+ BufferInfo *inInfo = *inQueue.begin();
+ OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
+ mTimeStamps[mTimeStampIdx] = inHeader->nTimeStamp;
- uint8_t *dst = outHeader->pBuffer;
- const uint8_t *srcY = (const uint8_t *)mImg->planes[PLANE_Y];
- const uint8_t *srcU = (const uint8_t *)mImg->planes[PLANE_U];
- const uint8_t *srcV = (const uint8_t *)mImg->planes[PLANE_V];
- size_t srcYStride = mImg->stride[PLANE_Y];
- size_t srcUStride = mImg->stride[PLANE_U];
- size_t srcVStride = mImg->stride[PLANE_V];
- copyYV12FrameToOutputBuffer(dst, srcY, srcU, srcV, srcYStride, srcUStride, srcVStride);
+ BufferInfo *outInfo = *outQueue.begin();
+ OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
+ if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
+ mEOSStatus = INPUT_EOS_SEEN;
+ EOSseen = true;
+ }
- mImg = NULL;
- outInfo->mOwnedByUs = false;
- outQueue.erase(outQueue.begin());
- outInfo = NULL;
- notifyFillBufferDone(outHeader);
- outHeader = NULL;
+ if (inHeader->nFilledLen > 0 &&
+ vpx_codec_decode((vpx_codec_ctx_t *)mCtx,
+ inHeader->pBuffer + inHeader->nOffset,
+ inHeader->nFilledLen,
+ &mTimeStamps[mTimeStampIdx], 0)) {
+ ALOGE("on2 decoder failed to decode frame.");
+ notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
+ return;
+ }
+ mTimeStampIdx = (mTimeStampIdx + 1) % kNumBuffers;
+
+ if (!outputBuffers(
+ EOSseen /* flushDecoder */, true /* display */, EOSseen, &portWillReset)) {
+ ALOGE("on2 decoder failed to output frame.");
+ notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
+ return;
+ }
+ if (portWillReset) {
+ return;
}
inInfo->mOwnedByUs = false;
@@ -176,6 +253,30 @@
}
}
+void SoftVPX::onPortFlushCompleted(OMX_U32 portIndex) {
+ if (portIndex == kInputPortIndex) {
+ bool portWillReset = false;
+ if (!outputBuffers(
+ true /* flushDecoder */, false /* display */, false /* eos */, &portWillReset)) {
+ ALOGE("Failed to flush decoder.");
+ notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
+ return;
+ }
+ mEOSStatus = INPUT_DATA_AVAILABLE;
+ }
+}
+
+void SoftVPX::onReset() {
+ bool portWillReset = false;
+ if (!outputBuffers(
+ true /* flushDecoder */, false /* display */, false /* eos */, &portWillReset)) {
+ ALOGW("Failed to flush decoder. Try to hard reset decoder");
+ destroyDecoder();
+ initDecoder();
+ }
+ mEOSStatus = INPUT_DATA_AVAILABLE;
+}
+
} // namespace android
android::SoftOMXComponent *createSoftOMXComponent(
diff --git a/media/libstagefright/codecs/on2/dec/SoftVPX.h b/media/libstagefright/codecs/on2/dec/SoftVPX.h
index 8f68693..8ccbae2 100644
--- a/media/libstagefright/codecs/on2/dec/SoftVPX.h
+++ b/media/libstagefright/codecs/on2/dec/SoftVPX.h
@@ -38,6 +38,8 @@
virtual ~SoftVPX();
virtual void onQueueFilled(OMX_U32 portIndex);
+ virtual void onPortFlushCompleted(OMX_U32 portIndex);
+ virtual void onReset();
private:
enum {
@@ -49,11 +51,21 @@
MODE_VP9
} mMode;
- void *mCtx;
+ enum {
+ INPUT_DATA_AVAILABLE, // VPX component is ready to decode data.
+ INPUT_EOS_SEEN, // VPX component saw EOS and is flushing On2 decoder.
+ OUTPUT_FRAMES_FLUSHED // VPX component finished flushing On2 decoder.
+ } mEOSStatus;
+ void *mCtx;
+ bool mFrameParallelMode; // Frame parallel is only supported by VP9 decoder.
+ OMX_TICKS mTimeStamps[kNumBuffers];
+ uint8_t mTimeStampIdx;
vpx_image_t *mImg;
status_t initDecoder();
+ status_t destroyDecoder();
+ bool outputBuffers(bool flushDecoder, bool display, bool eos, bool *portWillReset);
DISALLOW_EVIL_CONSTRUCTORS(SoftVPX);
};
diff --git a/media/libstagefright/codecs/opus/dec/SoftOpus.cpp b/media/libstagefright/codecs/opus/dec/SoftOpus.cpp
index b8084ae..6322dc2 100644
--- a/media/libstagefright/codecs/opus/dec/SoftOpus.cpp
+++ b/media/libstagefright/codecs/opus/dec/SoftOpus.cpp
@@ -345,9 +345,15 @@
}
uint8_t channel_mapping[kMaxChannels] = {0};
- memcpy(&channel_mapping,
- kDefaultOpusChannelLayout,
- kMaxChannelsWithDefaultLayout);
+ if (mHeader->channels <= kMaxChannelsWithDefaultLayout) {
+ memcpy(&channel_mapping,
+ kDefaultOpusChannelLayout,
+ kMaxChannelsWithDefaultLayout);
+ } else {
+ memcpy(&channel_mapping,
+ mHeader->stream_map,
+ mHeader->channels);
+ }
int status = OPUS_INVALID_STATE;
mDecoder = opus_multistream_decoder_create(kRate,
diff --git a/media/libstagefright/colorconversion/SoftwareRenderer.cpp b/media/libstagefright/colorconversion/SoftwareRenderer.cpp
index 4e75250..21da707 100644
--- a/media/libstagefright/colorconversion/SoftwareRenderer.cpp
+++ b/media/libstagefright/colorconversion/SoftwareRenderer.cpp
@@ -98,33 +98,49 @@
mCropWidth = mCropRight - mCropLeft + 1;
mCropHeight = mCropBottom - mCropTop + 1;
- int halFormat;
- size_t bufWidth, bufHeight;
+ // by default convert everything to RGB565
+ int halFormat = HAL_PIXEL_FORMAT_RGB_565;
+ size_t bufWidth = mCropWidth;
+ size_t bufHeight = mCropHeight;
- switch (mColorFormat) {
- case OMX_COLOR_FormatYUV420Planar:
- case OMX_TI_COLOR_FormatYUV420PackedSemiPlanar:
- case OMX_COLOR_FormatYUV420SemiPlanar:
- {
- if (!runningInEmulator()) {
+ // hardware has YUV12 and RGBA8888 support, so convert known formats
+ if (!runningInEmulator()) {
+ switch (mColorFormat) {
+ case OMX_COLOR_FormatYUV420Planar:
+ case OMX_COLOR_FormatYUV420SemiPlanar:
+ case OMX_TI_COLOR_FormatYUV420PackedSemiPlanar:
+ {
halFormat = HAL_PIXEL_FORMAT_YV12;
bufWidth = (mCropWidth + 1) & ~1;
bufHeight = (mCropHeight + 1) & ~1;
break;
}
-
- // fall through.
+ case OMX_COLOR_Format24bitRGB888:
+ {
+ halFormat = HAL_PIXEL_FORMAT_RGB_888;
+ bufWidth = (mCropWidth + 1) & ~1;
+ bufHeight = (mCropHeight + 1) & ~1;
+ break;
+ }
+ case OMX_COLOR_Format32bitARGB8888:
+ case OMX_COLOR_Format32BitRGBA8888:
+ {
+ halFormat = HAL_PIXEL_FORMAT_RGBA_8888;
+ bufWidth = (mCropWidth + 1) & ~1;
+ bufHeight = (mCropHeight + 1) & ~1;
+ break;
+ }
+ default:
+ {
+ break;
+ }
}
+ }
- default:
- halFormat = HAL_PIXEL_FORMAT_RGB_565;
- bufWidth = mCropWidth;
- bufHeight = mCropHeight;
-
- mConverter = new ColorConverter(
- mColorFormat, OMX_COLOR_Format16bitRGB565);
- CHECK(mConverter->isValid());
- break;
+ if (halFormat == HAL_PIXEL_FORMAT_RGB_565) {
+ mConverter = new ColorConverter(
+ mColorFormat, OMX_COLOR_Format16bitRGB565);
+ CHECK(mConverter->isValid());
}
CHECK(mNativeWindow != NULL);
@@ -201,6 +217,8 @@
CHECK_EQ(0, mapper.lock(
buf->handle, GRALLOC_USAGE_SW_WRITE_OFTEN, bounds, &dst));
+ // TODO move the other conversions also into ColorConverter, and
+ // fix cropping issues (when mCropLeft/Top != 0 or mWidth != mCropWidth)
if (mConverter) {
mConverter->convert(
data,
@@ -211,7 +229,8 @@
0, 0, mCropWidth - 1, mCropHeight - 1);
} else if (mColorFormat == OMX_COLOR_FormatYUV420Planar) {
const uint8_t *src_y = (const uint8_t *)data;
- const uint8_t *src_u = (const uint8_t *)data + mWidth * mHeight;
+ const uint8_t *src_u =
+ (const uint8_t *)data + mWidth * mHeight;
const uint8_t *src_v = src_u + (mWidth / 2 * mHeight / 2);
uint8_t *dst_y = (uint8_t *)dst;
@@ -239,11 +258,9 @@
}
} else if (mColorFormat == OMX_TI_COLOR_FormatYUV420PackedSemiPlanar
|| mColorFormat == OMX_COLOR_FormatYUV420SemiPlanar) {
- const uint8_t *src_y =
- (const uint8_t *)data;
-
- const uint8_t *src_uv =
- (const uint8_t *)data + mWidth * (mHeight - mCropTop / 2);
+ const uint8_t *src_y = (const uint8_t *)data;
+ const uint8_t *src_uv = (const uint8_t *)data
+ + mWidth * (mHeight - mCropTop / 2);
uint8_t *dst_y = (uint8_t *)dst;
@@ -271,6 +288,38 @@
dst_u += dst_c_stride;
dst_v += dst_c_stride;
}
+ } else if (mColorFormat == OMX_COLOR_Format24bitRGB888) {
+ uint8_t* srcPtr = (uint8_t*)data;
+ uint8_t* dstPtr = (uint8_t*)dst;
+
+ for (size_t y = 0; y < (size_t)mCropHeight; ++y) {
+ memcpy(dstPtr, srcPtr, mCropWidth * 3);
+ srcPtr += mWidth * 3;
+ dstPtr += buf->stride * 3;
+ }
+ } else if (mColorFormat == OMX_COLOR_Format32bitARGB8888) {
+ uint8_t *srcPtr, *dstPtr;
+
+ for (size_t y = 0; y < (size_t)mCropHeight; ++y) {
+ srcPtr = (uint8_t*)data + mWidth * 4 * y;
+ dstPtr = (uint8_t*)dst + buf->stride * 4 * y;
+ for (size_t x = 0; x < (size_t)mCropWidth; ++x) {
+ uint8_t a = *srcPtr++;
+ for (size_t i = 0; i < 3; ++i) { // copy RGB
+ *dstPtr++ = *srcPtr++;
+ }
+ *dstPtr++ = a; // alpha last (ARGB to RGBA)
+ }
+ }
+ } else if (mColorFormat == OMX_COLOR_Format32BitRGBA8888) {
+ uint8_t* srcPtr = (uint8_t*)data;
+ uint8_t* dstPtr = (uint8_t*)dst;
+
+ for (size_t y = 0; y < (size_t)mCropHeight; ++y) {
+ memcpy(dstPtr, srcPtr, mCropWidth * 4);
+ srcPtr += mWidth * 4;
+ dstPtr += buf->stride * 4;
+ }
} else {
LOG_ALWAYS_FATAL("bad color format %#x", mColorFormat);
}
diff --git a/media/libstagefright/filters/Android.mk b/media/libstagefright/filters/Android.mk
new file mode 100644
index 0000000..36ab444
--- /dev/null
+++ b/media/libstagefright/filters/Android.mk
@@ -0,0 +1,27 @@
+LOCAL_PATH:= $(call my-dir)
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES := \
+ ColorConvert.cpp \
+ GraphicBufferListener.cpp \
+ IntrinsicBlurFilter.cpp \
+ MediaFilter.cpp \
+ RSFilter.cpp \
+ SaturationFilter.cpp \
+ saturationARGB.rs \
+ SimpleFilter.cpp \
+ ZeroFilter.cpp
+
+LOCAL_C_INCLUDES := \
+ $(TOP)/frameworks/native/include/media/openmax \
+ $(TOP)/frameworks/rs/cpp \
+ $(TOP)/frameworks/rs \
+
+intermediates := $(call intermediates-dir-for,STATIC_LIBRARIES,libRS,TARGET,)
+LOCAL_C_INCLUDES += $(intermediates)
+
+LOCAL_CFLAGS += -Wno-multichar
+
+LOCAL_MODULE:= libstagefright_mediafilter
+
+include $(BUILD_STATIC_LIBRARY)
diff --git a/media/libstagefright/filters/ColorConvert.cpp b/media/libstagefright/filters/ColorConvert.cpp
new file mode 100644
index 0000000..a5039f9
--- /dev/null
+++ b/media/libstagefright/filters/ColorConvert.cpp
@@ -0,0 +1,111 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include "ColorConvert.h"
+
+#ifndef max
+#define max(a,b) ((a) > (b) ? (a) : (b))
+#endif
+#ifndef min
+#define min(a,b) ((a) < (b) ? (a) : (b))
+#endif
+
+namespace android {
+
+void YUVToRGB(
+ int32_t y, int32_t u, int32_t v,
+ int32_t* r, int32_t* g, int32_t* b) {
+ y -= 16;
+ u -= 128;
+ v -= 128;
+
+ *b = 1192 * y + 2066 * u;
+ *g = 1192 * y - 833 * v - 400 * u;
+ *r = 1192 * y + 1634 * v;
+
+ *r = min(262143, max(0, *r));
+ *g = min(262143, max(0, *g));
+ *b = min(262143, max(0, *b));
+
+ *r >>= 10;
+ *g >>= 10;
+ *b >>= 10;
+}
+
+void convertYUV420spToARGB(
+ uint8_t *pY, uint8_t *pUV, int32_t width, int32_t height,
+ uint8_t *dest) {
+ const int32_t bytes_per_pixel = 2;
+
+ for (int32_t i = 0; i < height; i++) {
+ for (int32_t j = 0; j < width; j++) {
+ int32_t y = *(pY + i * width + j);
+ int32_t u = *(pUV + (i/2) * width + bytes_per_pixel * (j/2));
+ int32_t v = *(pUV + (i/2) * width + bytes_per_pixel * (j/2) + 1);
+
+ int32_t r, g, b;
+ YUVToRGB(y, u, v, &r, &g, &b);
+
+ *dest++ = 0xFF;
+ *dest++ = r;
+ *dest++ = g;
+ *dest++ = b;
+ }
+ }
+}
+
+void convertYUV420spToRGB888(
+ uint8_t *pY, uint8_t *pUV, int32_t width, int32_t height,
+ uint8_t *dest) {
+ const int32_t bytes_per_pixel = 2;
+
+ for (int32_t i = 0; i < height; i++) {
+ for (int32_t j = 0; j < width; j++) {
+ int32_t y = *(pY + i * width + j);
+ int32_t u = *(pUV + (i/2) * width + bytes_per_pixel * (j/2));
+ int32_t v = *(pUV + (i/2) * width + bytes_per_pixel * (j/2) + 1);
+
+ int32_t r, g, b;
+ YUVToRGB(y, u, v, &r, &g, &b);
+
+ *dest++ = r;
+ *dest++ = g;
+ *dest++ = b;
+ }
+ }
+}
+
+// HACK - not even slightly optimized
+// TODO: remove when RGBA support is added to SoftwareRenderer
+void convertRGBAToARGB(
+ uint8_t *src, int32_t width, int32_t height, uint32_t stride,
+ uint8_t *dest) {
+ for (size_t i = 0; i < height; ++i) {
+ for (size_t j = 0; j < width; ++j) {
+ uint8_t r = *src++;
+ uint8_t g = *src++;
+ uint8_t b = *src++;
+ uint8_t a = *src++;
+ *dest++ = a;
+ *dest++ = r;
+ *dest++ = g;
+ *dest++ = b;
+ }
+ src += (stride - width) * 4;
+ }
+}
+
+} // namespace android
diff --git a/media/libstagefright/filters/ColorConvert.h b/media/libstagefright/filters/ColorConvert.h
new file mode 100644
index 0000000..13faa02
--- /dev/null
+++ b/media/libstagefright/filters/ColorConvert.h
@@ -0,0 +1,43 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef COLOR_CONVERT_H_
+#define COLOR_CONVERT_H_
+
+#include <inttypes.h>
+
+namespace android {
+
+void YUVToRGB(
+ int32_t y, int32_t u, int32_t v,
+ int32_t* r, int32_t* g, int32_t* b);
+
+void convertYUV420spToARGB(
+ uint8_t *pY, uint8_t *pUV, int32_t width, int32_t height,
+ uint8_t *dest);
+
+void convertYUV420spToRGB888(
+ uint8_t *pY, uint8_t *pUV, int32_t width, int32_t height,
+ uint8_t *dest);
+
+// TODO: remove when RGBA support is added to SoftwareRenderer
+void convertRGBAToARGB(
+ uint8_t *src, int32_t width, int32_t height, uint32_t stride,
+ uint8_t *dest);
+
+} // namespace android
+
+#endif // COLOR_CONVERT_H_
diff --git a/media/libstagefright/filters/GraphicBufferListener.cpp b/media/libstagefright/filters/GraphicBufferListener.cpp
new file mode 100644
index 0000000..fa38192
--- /dev/null
+++ b/media/libstagefright/filters/GraphicBufferListener.cpp
@@ -0,0 +1,154 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "GraphicBufferListener"
+
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/MediaErrors.h>
+
+#include "GraphicBufferListener.h"
+
+namespace android {
+
+status_t GraphicBufferListener::init(
+ const sp<AMessage> ¬ify,
+ size_t bufferWidth, size_t bufferHeight, size_t bufferCount) {
+ mNotify = notify;
+
+ String8 name("GraphicBufferListener");
+ BufferQueue::createBufferQueue(&mProducer, &mConsumer);
+ mConsumer->setConsumerName(name);
+ mConsumer->setDefaultBufferSize(bufferWidth, bufferHeight);
+ mConsumer->setConsumerUsageBits(GRALLOC_USAGE_SW_READ_OFTEN);
+
+ status_t err = mConsumer->setMaxAcquiredBufferCount(bufferCount);
+ if (err != NO_ERROR) {
+ ALOGE("Unable to set BQ max acquired buffer count to %u: %d",
+ bufferCount, err);
+ return err;
+ }
+
+ wp<BufferQueue::ConsumerListener> listener =
+ static_cast<BufferQueue::ConsumerListener*>(this);
+ sp<BufferQueue::ProxyConsumerListener> proxy =
+ new BufferQueue::ProxyConsumerListener(listener);
+
+ err = mConsumer->consumerConnect(proxy, false);
+ if (err != NO_ERROR) {
+ ALOGE("Error connecting to BufferQueue: %s (%d)",
+ strerror(-err), err);
+ return err;
+ }
+
+ ALOGV("init() successful.");
+
+ return OK;
+}
+
+void GraphicBufferListener::onFrameAvailable(const BufferItem& /* item */) {
+ ALOGV("onFrameAvailable() called");
+
+ {
+ Mutex::Autolock autoLock(mMutex);
+ mNumFramesAvailable++;
+ }
+
+ sp<AMessage> notify = mNotify->dup();
+ mNotify->setWhat(kWhatFrameAvailable);
+ mNotify->post();
+}
+
+void GraphicBufferListener::onBuffersReleased() {
+ ALOGV("onBuffersReleased() called");
+ // nothing to do
+}
+
+void GraphicBufferListener::onSidebandStreamChanged() {
+ ALOGW("GraphicBufferListener cannot consume sideband streams.");
+ // nothing to do
+}
+
+BufferQueue::BufferItem GraphicBufferListener::getBufferItem() {
+ BufferQueue::BufferItem item;
+
+ {
+ Mutex::Autolock autoLock(mMutex);
+ if (mNumFramesAvailable <= 0) {
+ ALOGE("getBuffer() called with no frames available");
+ return item;
+ }
+ mNumFramesAvailable--;
+ }
+
+ status_t err = mConsumer->acquireBuffer(&item, 0);
+ if (err == BufferQueue::NO_BUFFER_AVAILABLE) {
+ // shouldn't happen, since we track num frames available
+ ALOGE("frame was not available");
+ item.mBuf = -1;
+ return item;
+ } else if (err != OK) {
+ ALOGE("acquireBuffer returned err=%d", err);
+ item.mBuf = -1;
+ return item;
+ }
+
+ // Wait for it to become available.
+ err = item.mFence->waitForever("GraphicBufferListener::getBufferItem");
+ if (err != OK) {
+ ALOGW("failed to wait for buffer fence: %d", err);
+ // keep going
+ }
+
+ // If this is the first time we're seeing this buffer, add it to our
+ // slot table.
+ if (item.mGraphicBuffer != NULL) {
+ ALOGV("setting mBufferSlot %d", item.mBuf);
+ mBufferSlot[item.mBuf] = item.mGraphicBuffer;
+ }
+
+ return item;
+}
+
+sp<GraphicBuffer> GraphicBufferListener::getBuffer(
+ BufferQueue::BufferItem item) {
+ sp<GraphicBuffer> buf;
+ if (item.mBuf < 0 || item.mBuf >= BufferQueue::NUM_BUFFER_SLOTS) {
+ ALOGE("getBuffer() received invalid BufferItem: mBuf==%d", item.mBuf);
+ return buf;
+ }
+
+ buf = mBufferSlot[item.mBuf];
+ CHECK(buf.get() != NULL);
+
+ return buf;
+}
+
+status_t GraphicBufferListener::releaseBuffer(
+ BufferQueue::BufferItem item) {
+ if (item.mBuf < 0 || item.mBuf >= BufferQueue::NUM_BUFFER_SLOTS) {
+ ALOGE("getBuffer() received invalid BufferItem: mBuf==%d", item.mBuf);
+ return ERROR_OUT_OF_RANGE;
+ }
+
+ mConsumer->releaseBuffer(item.mBuf, item.mFrameNumber,
+ EGL_NO_DISPLAY, EGL_NO_SYNC_KHR, Fence::NO_FENCE);
+
+ return OK;
+}
+
+} // namespace android
diff --git a/media/libstagefright/filters/GraphicBufferListener.h b/media/libstagefright/filters/GraphicBufferListener.h
new file mode 100644
index 0000000..b3e0ee3
--- /dev/null
+++ b/media/libstagefright/filters/GraphicBufferListener.h
@@ -0,0 +1,70 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef GRAPHIC_BUFFER_LISTENER_H_
+#define GRAPHIC_BUFFER_LISTENER_H_
+
+#include <gui/BufferQueue.h>
+
+namespace android {
+
+struct AMessage;
+
+struct GraphicBufferListener : public BufferQueue::ConsumerListener {
+public:
+ GraphicBufferListener() {};
+
+ status_t init(
+ const sp<AMessage> ¬ify,
+ size_t bufferWidth, size_t bufferHeight, size_t bufferCount);
+
+ virtual void onFrameAvailable(const BufferItem& item);
+ virtual void onBuffersReleased();
+ virtual void onSidebandStreamChanged();
+
+ // Returns the handle to the producer side of the BufferQueue. Buffers
+ // queued on this will be received by GraphicBufferListener.
+ sp<IGraphicBufferProducer> getIGraphicBufferProducer() const {
+ return mProducer;
+ }
+
+ BufferQueue::BufferItem getBufferItem();
+ sp<GraphicBuffer> getBuffer(BufferQueue::BufferItem item);
+ status_t releaseBuffer(BufferQueue::BufferItem item);
+
+ enum {
+ kWhatFrameAvailable = 'frav',
+ };
+
+private:
+ sp<AMessage> mNotify;
+ size_t mNumFramesAvailable;
+
+ mutable Mutex mMutex;
+
+ // Our BufferQueue interfaces. mProducer is passed to the producer through
+ // getIGraphicBufferProducer, and mConsumer is used internally to retrieve
+ // the buffers queued by the producer.
+ sp<IGraphicBufferProducer> mProducer;
+ sp<IGraphicBufferConsumer> mConsumer;
+
+ // Cache of GraphicBuffers from the buffer queue.
+ sp<GraphicBuffer> mBufferSlot[BufferQueue::NUM_BUFFER_SLOTS];
+};
+
+} // namespace android
+
+#endif // GRAPHIC_BUFFER_LISTENER_H
diff --git a/media/libstagefright/filters/IntrinsicBlurFilter.cpp b/media/libstagefright/filters/IntrinsicBlurFilter.cpp
new file mode 100644
index 0000000..cbcf699
--- /dev/null
+++ b/media/libstagefright/filters/IntrinsicBlurFilter.cpp
@@ -0,0 +1,99 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "IntrinsicBlurFilter"
+
+#include <utils/Log.h>
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+
+#include "IntrinsicBlurFilter.h"
+
+namespace android {
+
+status_t IntrinsicBlurFilter::configure(const sp<AMessage> &msg) {
+ status_t err = SimpleFilter::configure(msg);
+ if (err != OK) {
+ return err;
+ }
+
+ if (!msg->findString("cacheDir", &mCacheDir)) {
+ ALOGE("Failed to find cache directory in config message.");
+ return NAME_NOT_FOUND;
+ }
+
+ return OK;
+}
+
+status_t IntrinsicBlurFilter::start() {
+ // TODO: use a single RS context object for entire application
+ mRS = new RSC::RS();
+
+ if (!mRS->init(mCacheDir.c_str())) {
+ ALOGE("Failed to initialize RenderScript context.");
+ return NO_INIT;
+ }
+
+ // 32-bit elements for ARGB8888
+ RSC::sp<const RSC::Element> e = RSC::Element::U8_4(mRS);
+
+ RSC::Type::Builder tb(mRS, e);
+ tb.setX(mWidth);
+ tb.setY(mHeight);
+ RSC::sp<const RSC::Type> t = tb.create();
+
+ mAllocIn = RSC::Allocation::createTyped(mRS, t);
+ mAllocOut = RSC::Allocation::createTyped(mRS, t);
+
+ mBlur = RSC::ScriptIntrinsicBlur::create(mRS, e);
+ mBlur->setRadius(mBlurRadius);
+ mBlur->setInput(mAllocIn);
+
+ return OK;
+}
+
+void IntrinsicBlurFilter::reset() {
+ mBlur.clear();
+ mAllocOut.clear();
+ mAllocIn.clear();
+ mRS.clear();
+}
+
+status_t IntrinsicBlurFilter::setParameters(const sp<AMessage> &msg) {
+ sp<AMessage> params;
+ CHECK(msg->findMessage("params", ¶ms));
+
+ float blurRadius;
+ if (params->findFloat("blur-radius", &blurRadius)) {
+ mBlurRadius = blurRadius;
+ }
+
+ return OK;
+}
+
+status_t IntrinsicBlurFilter::processBuffers(
+ const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer) {
+ mAllocIn->copy1DRangeFrom(0, mWidth * mHeight, srcBuffer->data());
+ mBlur->forEach(mAllocOut);
+ mAllocOut->copy1DRangeTo(0, mWidth * mHeight, outBuffer->data());
+
+ return OK;
+}
+
+} // namespace android
diff --git a/media/libstagefright/filters/IntrinsicBlurFilter.h b/media/libstagefright/filters/IntrinsicBlurFilter.h
new file mode 100644
index 0000000..4707ab7
--- /dev/null
+++ b/media/libstagefright/filters/IntrinsicBlurFilter.h
@@ -0,0 +1,50 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef INTRINSIC_BLUR_FILTER_H_
+#define INTRINSIC_BLUR_FILTER_H_
+
+#include "RenderScript.h"
+#include "SimpleFilter.h"
+
+namespace android {
+
+struct IntrinsicBlurFilter : public SimpleFilter {
+public:
+ IntrinsicBlurFilter() : mBlurRadius(1.f) {};
+
+ virtual status_t configure(const sp<AMessage> &msg);
+ virtual status_t start();
+ virtual void reset();
+ virtual status_t setParameters(const sp<AMessage> &msg);
+ virtual status_t processBuffers(
+ const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer);
+
+protected:
+ virtual ~IntrinsicBlurFilter() {};
+
+private:
+ AString mCacheDir;
+ RSC::sp<RSC::RS> mRS;
+ RSC::sp<RSC::Allocation> mAllocIn;
+ RSC::sp<RSC::Allocation> mAllocOut;
+ RSC::sp<RSC::ScriptIntrinsicBlur> mBlur;
+ float mBlurRadius;
+};
+
+} // namespace android
+
+#endif // INTRINSIC_BLUR_FILTER_H_
diff --git a/media/libstagefright/filters/MediaFilter.cpp b/media/libstagefright/filters/MediaFilter.cpp
new file mode 100644
index 0000000..c5289b6
--- /dev/null
+++ b/media/libstagefright/filters/MediaFilter.cpp
@@ -0,0 +1,816 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "MediaFilter"
+
+#include <inttypes.h>
+#include <utils/Trace.h>
+
+#include <binder/MemoryDealer.h>
+
+#include <media/stagefright/BufferProducerWrapper.h>
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MediaFilter.h>
+
+#include "ColorConvert.h"
+#include "GraphicBufferListener.h"
+#include "IntrinsicBlurFilter.h"
+#include "RSFilter.h"
+#include "SaturationFilter.h"
+#include "ZeroFilter.h"
+
+namespace android {
+
+// parameter: number of input and output buffers
+static const size_t kBufferCountActual = 4;
+
+MediaFilter::MediaFilter()
+ : mState(UNINITIALIZED),
+ mGeneration(0),
+ mGraphicBufferListener(NULL) {
+}
+
+MediaFilter::~MediaFilter() {
+}
+
+//////////////////// PUBLIC FUNCTIONS //////////////////////////////////////////
+
+void MediaFilter::setNotificationMessage(const sp<AMessage> &msg) {
+ mNotify = msg;
+}
+
+void MediaFilter::initiateAllocateComponent(const sp<AMessage> &msg) {
+ msg->setWhat(kWhatAllocateComponent);
+ msg->setTarget(id());
+ msg->post();
+}
+
+void MediaFilter::initiateConfigureComponent(const sp<AMessage> &msg) {
+ msg->setWhat(kWhatConfigureComponent);
+ msg->setTarget(id());
+ msg->post();
+}
+
+void MediaFilter::initiateCreateInputSurface() {
+ (new AMessage(kWhatCreateInputSurface, id()))->post();
+}
+
+void MediaFilter::initiateStart() {
+ (new AMessage(kWhatStart, id()))->post();
+}
+
+void MediaFilter::initiateShutdown(bool keepComponentAllocated) {
+ sp<AMessage> msg = new AMessage(kWhatShutdown, id());
+ msg->setInt32("keepComponentAllocated", keepComponentAllocated);
+ msg->post();
+}
+
+void MediaFilter::signalFlush() {
+ (new AMessage(kWhatFlush, id()))->post();
+}
+
+void MediaFilter::signalResume() {
+ (new AMessage(kWhatResume, id()))->post();
+}
+
+// nothing to do
+void MediaFilter::signalRequestIDRFrame() {
+ return;
+}
+
+void MediaFilter::signalSetParameters(const sp<AMessage> ¶ms) {
+ sp<AMessage> msg = new AMessage(kWhatSetParameters, id());
+ msg->setMessage("params", params);
+ msg->post();
+}
+
+void MediaFilter::signalEndOfInputStream() {
+ (new AMessage(kWhatSignalEndOfInputStream, id()))->post();
+}
+
+void MediaFilter::onMessageReceived(const sp<AMessage> &msg) {
+ switch (msg->what()) {
+ case kWhatAllocateComponent:
+ {
+ onAllocateComponent(msg);
+ break;
+ }
+ case kWhatConfigureComponent:
+ {
+ onConfigureComponent(msg);
+ break;
+ }
+ case kWhatStart:
+ {
+ onStart();
+ break;
+ }
+ case kWhatProcessBuffers:
+ {
+ processBuffers();
+ break;
+ }
+ case kWhatInputBufferFilled:
+ {
+ onInputBufferFilled(msg);
+ break;
+ }
+ case kWhatOutputBufferDrained:
+ {
+ onOutputBufferDrained(msg);
+ break;
+ }
+ case kWhatShutdown:
+ {
+ onShutdown(msg);
+ break;
+ }
+ case kWhatFlush:
+ {
+ onFlush();
+ break;
+ }
+ case kWhatResume:
+ {
+ // nothing to do
+ break;
+ }
+ case kWhatSetParameters:
+ {
+ onSetParameters(msg);
+ break;
+ }
+ case kWhatCreateInputSurface:
+ {
+ onCreateInputSurface();
+ break;
+ }
+ case GraphicBufferListener::kWhatFrameAvailable:
+ {
+ onInputFrameAvailable();
+ break;
+ }
+ case kWhatSignalEndOfInputStream:
+ {
+ onSignalEndOfInputStream();
+ break;
+ }
+ default:
+ {
+ ALOGE("Message not handled:\n%s", msg->debugString().c_str());
+ break;
+ }
+ }
+}
+
+//////////////////// PORT DESCRIPTION //////////////////////////////////////////
+
+MediaFilter::PortDescription::PortDescription() {
+}
+
+void MediaFilter::PortDescription::addBuffer(
+ IOMX::buffer_id id, const sp<ABuffer> &buffer) {
+ mBufferIDs.push_back(id);
+ mBuffers.push_back(buffer);
+}
+
+size_t MediaFilter::PortDescription::countBuffers() {
+ return mBufferIDs.size();
+}
+
+IOMX::buffer_id MediaFilter::PortDescription::bufferIDAt(size_t index) const {
+ return mBufferIDs.itemAt(index);
+}
+
+sp<ABuffer> MediaFilter::PortDescription::bufferAt(size_t index) const {
+ return mBuffers.itemAt(index);
+}
+
+//////////////////// HELPER FUNCTIONS //////////////////////////////////////////
+
+void MediaFilter::signalProcessBuffers() {
+ (new AMessage(kWhatProcessBuffers, id()))->post();
+}
+
+void MediaFilter::signalError(status_t error) {
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", CodecBase::kWhatError);
+ notify->setInt32("err", error);
+ notify->post();
+}
+
+status_t MediaFilter::allocateBuffersOnPort(OMX_U32 portIndex) {
+ CHECK(portIndex == kPortIndexInput || portIndex == kPortIndexOutput);
+ const bool isInput = portIndex == kPortIndexInput;
+ const size_t bufferSize = isInput ? mMaxInputSize : mMaxOutputSize;
+
+ CHECK(mDealer[portIndex] == NULL);
+ CHECK(mBuffers[portIndex].isEmpty());
+
+ ALOGV("Allocating %zu buffers of size %zu on %s port",
+ kBufferCountActual, bufferSize,
+ isInput ? "input" : "output");
+
+ size_t totalSize = kBufferCountActual * bufferSize;
+
+ mDealer[portIndex] = new MemoryDealer(totalSize, "MediaFilter");
+
+ for (size_t i = 0; i < kBufferCountActual; ++i) {
+ sp<IMemory> mem = mDealer[portIndex]->allocate(bufferSize);
+ CHECK(mem.get() != NULL);
+
+ BufferInfo info;
+ info.mStatus = BufferInfo::OWNED_BY_US;
+ info.mBufferID = i;
+ info.mGeneration = mGeneration;
+ info.mOutputFlags = 0;
+ info.mData = new ABuffer(mem->pointer(), bufferSize);
+ info.mData->meta()->setInt64("timeUs", 0);
+
+ mBuffers[portIndex].push_back(info);
+
+ if (!isInput) {
+ mAvailableOutputBuffers.push(
+ &mBuffers[portIndex].editItemAt(i));
+ }
+ }
+
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", CodecBase::kWhatBuffersAllocated);
+
+ notify->setInt32("portIndex", portIndex);
+
+ sp<PortDescription> desc = new PortDescription;
+
+ for (size_t i = 0; i < mBuffers[portIndex].size(); ++i) {
+ const BufferInfo &info = mBuffers[portIndex][i];
+
+ desc->addBuffer(info.mBufferID, info.mData);
+ }
+
+ notify->setObject("portDesc", desc);
+ notify->post();
+
+ return OK;
+}
+
+MediaFilter::BufferInfo* MediaFilter::findBufferByID(
+ uint32_t portIndex, IOMX::buffer_id bufferID,
+ ssize_t *index) {
+ for (size_t i = 0; i < mBuffers[portIndex].size(); ++i) {
+ BufferInfo *info = &mBuffers[portIndex].editItemAt(i);
+
+ if (info->mBufferID == bufferID) {
+ if (index != NULL) {
+ *index = i;
+ }
+ return info;
+ }
+ }
+
+ TRESPASS();
+
+ return NULL;
+}
+
+void MediaFilter::postFillThisBuffer(BufferInfo *info) {
+ ALOGV("postFillThisBuffer on buffer %d", info->mBufferID);
+ if (mPortEOS[kPortIndexInput]) {
+ return;
+ }
+
+ CHECK_EQ((int)info->mStatus, (int)BufferInfo::OWNED_BY_US);
+
+ info->mGeneration = mGeneration;
+
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", CodecBase::kWhatFillThisBuffer);
+ notify->setInt32("buffer-id", info->mBufferID);
+
+ info->mData->meta()->clear();
+ notify->setBuffer("buffer", info->mData);
+
+ sp<AMessage> reply = new AMessage(kWhatInputBufferFilled, id());
+ reply->setInt32("buffer-id", info->mBufferID);
+
+ notify->setMessage("reply", reply);
+
+ info->mStatus = BufferInfo::OWNED_BY_UPSTREAM;
+ notify->post();
+}
+
+void MediaFilter::postDrainThisBuffer(BufferInfo *info) {
+ CHECK_EQ((int)info->mStatus, (int)BufferInfo::OWNED_BY_US);
+
+ info->mGeneration = mGeneration;
+
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", CodecBase::kWhatDrainThisBuffer);
+ notify->setInt32("buffer-id", info->mBufferID);
+ notify->setInt32("flags", info->mOutputFlags);
+ notify->setBuffer("buffer", info->mData);
+
+ sp<AMessage> reply = new AMessage(kWhatOutputBufferDrained, id());
+ reply->setInt32("buffer-id", info->mBufferID);
+
+ notify->setMessage("reply", reply);
+
+ notify->post();
+
+ info->mStatus = BufferInfo::OWNED_BY_UPSTREAM;
+}
+
+void MediaFilter::postEOS() {
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", CodecBase::kWhatEOS);
+ notify->setInt32("err", ERROR_END_OF_STREAM);
+ notify->post();
+
+ ALOGV("Sent kWhatEOS.");
+}
+
+void MediaFilter::sendFormatChange() {
+ sp<AMessage> notify = mNotify->dup();
+
+ notify->setInt32("what", kWhatOutputFormatChanged);
+
+ AString mime;
+ CHECK(mOutputFormat->findString("mime", &mime));
+ notify->setString("mime", mime.c_str());
+
+ notify->setInt32("stride", mStride);
+ notify->setInt32("slice-height", mSliceHeight);
+ notify->setInt32("color-format", mColorFormatOut);
+ notify->setRect("crop", 0, 0, mStride - 1, mSliceHeight - 1);
+ notify->setInt32("width", mWidth);
+ notify->setInt32("height", mHeight);
+
+ notify->post();
+}
+
+void MediaFilter::requestFillEmptyInput() {
+ if (mPortEOS[kPortIndexInput]) {
+ return;
+ }
+
+ for (size_t i = 0; i < mBuffers[kPortIndexInput].size(); ++i) {
+ BufferInfo *info = &mBuffers[kPortIndexInput].editItemAt(i);
+
+ if (info->mStatus == BufferInfo::OWNED_BY_US) {
+ postFillThisBuffer(info);
+ }
+ }
+}
+
+void MediaFilter::processBuffers() {
+ if (mAvailableInputBuffers.empty() || mAvailableOutputBuffers.empty()) {
+ ALOGV("Skipping process (buffers unavailable)");
+ return;
+ }
+
+ if (mPortEOS[kPortIndexOutput]) {
+ // TODO notify caller of queueInput error when it is supported
+ // in MediaCodec
+ ALOGW("Tried to process a buffer after EOS.");
+ return;
+ }
+
+ BufferInfo *inputInfo = mAvailableInputBuffers[0];
+ mAvailableInputBuffers.removeAt(0);
+ BufferInfo *outputInfo = mAvailableOutputBuffers[0];
+ mAvailableOutputBuffers.removeAt(0);
+
+ status_t err;
+ err = mFilter->processBuffers(inputInfo->mData, outputInfo->mData);
+ if (err != (status_t)OK) {
+ outputInfo->mData->meta()->setInt32("err", err);
+ }
+
+ int64_t timeUs;
+ CHECK(inputInfo->mData->meta()->findInt64("timeUs", &timeUs));
+ outputInfo->mData->meta()->setInt64("timeUs", timeUs);
+ outputInfo->mOutputFlags = 0;
+ int32_t eos = 0;
+ if (inputInfo->mData->meta()->findInt32("eos", &eos) && eos != 0) {
+ outputInfo->mOutputFlags |= OMX_BUFFERFLAG_EOS;
+ mPortEOS[kPortIndexOutput] = true;
+ outputInfo->mData->meta()->setInt32("eos", eos);
+ postEOS();
+ ALOGV("Output stream saw EOS.");
+ }
+
+ ALOGV("Processed input buffer %u [%zu], output buffer %u [%zu]",
+ inputInfo->mBufferID, inputInfo->mData->size(),
+ outputInfo->mBufferID, outputInfo->mData->size());
+
+ if (mGraphicBufferListener != NULL) {
+ delete inputInfo;
+ } else {
+ postFillThisBuffer(inputInfo);
+ }
+ postDrainThisBuffer(outputInfo);
+
+ // prevent any corner case where buffers could get stuck in queue
+ signalProcessBuffers();
+}
+
+void MediaFilter::onAllocateComponent(const sp<AMessage> &msg) {
+ CHECK_EQ(mState, UNINITIALIZED);
+
+ CHECK(msg->findString("componentName", &mComponentName));
+ const char* name = mComponentName.c_str();
+ if (!strcasecmp(name, "android.filter.zerofilter")) {
+ mFilter = new ZeroFilter;
+ } else if (!strcasecmp(name, "android.filter.saturation")) {
+ mFilter = new SaturationFilter;
+ } else if (!strcasecmp(name, "android.filter.intrinsicblur")) {
+ mFilter = new IntrinsicBlurFilter;
+ } else if (!strcasecmp(name, "android.filter.RenderScript")) {
+ mFilter = new RSFilter;
+ } else {
+ ALOGE("Unrecognized filter name: %s", name);
+ signalError(NAME_NOT_FOUND);
+ return;
+ }
+
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", kWhatComponentAllocated);
+ // HACK - need "OMX.google" to use MediaCodec's software renderer
+ notify->setString("componentName", "OMX.google.MediaFilter");
+ notify->post();
+ mState = INITIALIZED;
+ ALOGV("Handled kWhatAllocateComponent.");
+}
+
+void MediaFilter::onConfigureComponent(const sp<AMessage> &msg) {
+ // TODO: generalize to allow audio filters as well as video
+
+ CHECK_EQ(mState, INITIALIZED);
+
+ // get params - at least mime, width & height
+ AString mime;
+ CHECK(msg->findString("mime", &mime));
+ if (strcasecmp(mime.c_str(), MEDIA_MIMETYPE_VIDEO_RAW)) {
+ ALOGE("Bad mime: %s", mime.c_str());
+ signalError(BAD_VALUE);
+ return;
+ }
+
+ CHECK(msg->findInt32("width", &mWidth));
+ CHECK(msg->findInt32("height", &mHeight));
+ if (!msg->findInt32("stride", &mStride)) {
+ mStride = mWidth;
+ }
+ if (!msg->findInt32("slice-height", &mSliceHeight)) {
+ mSliceHeight = mHeight;
+ }
+
+ mMaxInputSize = mWidth * mHeight * 4; // room for ARGB8888
+ int32_t maxInputSize;
+ if (msg->findInt32("max-input-size", &maxInputSize)
+ && (size_t)maxInputSize > mMaxInputSize) {
+ mMaxInputSize = maxInputSize;
+ }
+
+ if (!msg->findInt32("color-format", &mColorFormatIn)) {
+ // default to OMX_COLOR_Format32bitARGB8888
+ mColorFormatIn = OMX_COLOR_Format32bitARGB8888;
+ msg->setInt32("color-format", mColorFormatIn);
+ }
+ mColorFormatOut = mColorFormatIn;
+
+ mMaxOutputSize = mWidth * mHeight * 4; // room for ARGB8888
+
+ AString cacheDir;
+ if (!msg->findString("cacheDir", &cacheDir)) {
+ ALOGE("Failed to find cache directory in config message.");
+ signalError(NAME_NOT_FOUND);
+ return;
+ }
+
+ status_t err;
+ err = mFilter->configure(msg);
+ if (err != (status_t)OK) {
+ ALOGE("Failed to configure filter component, err %d", err);
+ signalError(err);
+ return;
+ }
+
+ mInputFormat = new AMessage();
+ mInputFormat->setString("mime", mime.c_str());
+ mInputFormat->setInt32("stride", mStride);
+ mInputFormat->setInt32("slice-height", mSliceHeight);
+ mInputFormat->setInt32("color-format", mColorFormatIn);
+ mInputFormat->setRect("crop", 0, 0, mStride, mSliceHeight);
+ mInputFormat->setInt32("width", mWidth);
+ mInputFormat->setInt32("height", mHeight);
+
+ mOutputFormat = new AMessage();
+ mOutputFormat->setString("mime", mime.c_str());
+ mOutputFormat->setInt32("stride", mStride);
+ mOutputFormat->setInt32("slice-height", mSliceHeight);
+ mOutputFormat->setInt32("color-format", mColorFormatOut);
+ mOutputFormat->setRect("crop", 0, 0, mStride, mSliceHeight);
+ mOutputFormat->setInt32("width", mWidth);
+ mOutputFormat->setInt32("height", mHeight);
+
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", kWhatComponentConfigured);
+ notify->setString("componentName", "MediaFilter");
+ notify->setMessage("input-format", mInputFormat);
+ notify->setMessage("output-format", mOutputFormat);
+ notify->post();
+ mState = CONFIGURED;
+ ALOGV("Handled kWhatConfigureComponent.");
+
+ sendFormatChange();
+}
+
+void MediaFilter::onStart() {
+ CHECK_EQ(mState, CONFIGURED);
+
+ allocateBuffersOnPort(kPortIndexInput);
+
+ allocateBuffersOnPort(kPortIndexOutput);
+
+ status_t err = mFilter->start();
+ if (err != (status_t)OK) {
+ ALOGE("Failed to start filter component, err %d", err);
+ signalError(err);
+ return;
+ }
+
+ mPortEOS[kPortIndexInput] = false;
+ mPortEOS[kPortIndexOutput] = false;
+ mInputEOSResult = OK;
+ mState = STARTED;
+
+ requestFillEmptyInput();
+ ALOGV("Handled kWhatStart.");
+}
+
+void MediaFilter::onInputBufferFilled(const sp<AMessage> &msg) {
+ IOMX::buffer_id bufferID;
+ CHECK(msg->findInt32("buffer-id", (int32_t*)&bufferID));
+ BufferInfo *info = findBufferByID(kPortIndexInput, bufferID);
+
+ if (mState != STARTED) {
+ // we're not running, so we'll just keep that buffer...
+ info->mStatus = BufferInfo::OWNED_BY_US;
+ return;
+ }
+
+ if (info->mGeneration != mGeneration) {
+ ALOGV("Caught a stale input buffer [ID %d]", bufferID);
+ // buffer is stale (taken before a flush/shutdown) - repost it
+ CHECK_EQ(info->mStatus, BufferInfo::OWNED_BY_US);
+ postFillThisBuffer(info);
+ return;
+ }
+
+ CHECK_EQ(info->mStatus, BufferInfo::OWNED_BY_UPSTREAM);
+ info->mStatus = BufferInfo::OWNED_BY_US;
+
+ sp<ABuffer> buffer;
+ int32_t err = OK;
+ bool eos = false;
+
+ if (!msg->findBuffer("buffer", &buffer)) {
+ // these are unfilled buffers returned by client
+ CHECK(msg->findInt32("err", &err));
+
+ if (err == OK) {
+ // buffers with no errors are returned on MediaCodec.flush
+ ALOGV("saw unfilled buffer (MediaCodec.flush)");
+ postFillThisBuffer(info);
+ return;
+ } else {
+ ALOGV("saw error %d instead of an input buffer", err);
+ eos = true;
+ }
+
+ buffer.clear();
+ }
+
+ int32_t isCSD;
+ if (buffer != NULL && buffer->meta()->findInt32("csd", &isCSD)
+ && isCSD != 0) {
+ // ignore codec-specific data buffers
+ ALOGW("MediaFilter received a codec-specific data buffer");
+ postFillThisBuffer(info);
+ return;
+ }
+
+ int32_t tmp;
+ if (buffer != NULL && buffer->meta()->findInt32("eos", &tmp) && tmp) {
+ eos = true;
+ err = ERROR_END_OF_STREAM;
+ }
+
+ mAvailableInputBuffers.push_back(info);
+ processBuffers();
+
+ if (eos) {
+ mPortEOS[kPortIndexInput] = true;
+ mInputEOSResult = err;
+ }
+
+ ALOGV("Handled kWhatInputBufferFilled. [ID %u]", bufferID);
+}
+
+void MediaFilter::onOutputBufferDrained(const sp<AMessage> &msg) {
+ IOMX::buffer_id bufferID;
+ CHECK(msg->findInt32("buffer-id", (int32_t*)&bufferID));
+ BufferInfo *info = findBufferByID(kPortIndexOutput, bufferID);
+
+ if (mState != STARTED) {
+ // we're not running, so we'll just keep that buffer...
+ info->mStatus = BufferInfo::OWNED_BY_US;
+ return;
+ }
+
+ if (info->mGeneration != mGeneration) {
+ ALOGV("Caught a stale output buffer [ID %d]", bufferID);
+ // buffer is stale (taken before a flush/shutdown) - keep it
+ CHECK_EQ(info->mStatus, BufferInfo::OWNED_BY_US);
+ return;
+ }
+
+ CHECK_EQ(info->mStatus, BufferInfo::OWNED_BY_UPSTREAM);
+ info->mStatus = BufferInfo::OWNED_BY_US;
+
+ mAvailableOutputBuffers.push_back(info);
+
+ processBuffers();
+
+ ALOGV("Handled kWhatOutputBufferDrained. [ID %u]",
+ bufferID);
+}
+
+void MediaFilter::onShutdown(const sp<AMessage> &msg) {
+ mGeneration++;
+
+ if (mState != UNINITIALIZED) {
+ mFilter->reset();
+ }
+
+ int32_t keepComponentAllocated;
+ CHECK(msg->findInt32("keepComponentAllocated", &keepComponentAllocated));
+ if (!keepComponentAllocated || mState == UNINITIALIZED) {
+ mState = UNINITIALIZED;
+ } else {
+ mState = INITIALIZED;
+ }
+
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", CodecBase::kWhatShutdownCompleted);
+ notify->post();
+}
+
+void MediaFilter::onFlush() {
+ mGeneration++;
+
+ mAvailableInputBuffers.clear();
+ for (size_t i = 0; i < mBuffers[kPortIndexInput].size(); ++i) {
+ BufferInfo *info = &mBuffers[kPortIndexInput].editItemAt(i);
+ info->mStatus = BufferInfo::OWNED_BY_US;
+ }
+ mAvailableOutputBuffers.clear();
+ for (size_t i = 0; i < mBuffers[kPortIndexOutput].size(); ++i) {
+ BufferInfo *info = &mBuffers[kPortIndexOutput].editItemAt(i);
+ info->mStatus = BufferInfo::OWNED_BY_US;
+ mAvailableOutputBuffers.push_back(info);
+ }
+
+ mPortEOS[kPortIndexInput] = false;
+ mPortEOS[kPortIndexOutput] = false;
+ mInputEOSResult = OK;
+
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", CodecBase::kWhatFlushCompleted);
+ notify->post();
+ ALOGV("Posted kWhatFlushCompleted");
+
+ // MediaCodec returns all input buffers after flush, so in
+ // onInputBufferFilled we call postFillThisBuffer on them
+}
+
+void MediaFilter::onSetParameters(const sp<AMessage> &msg) {
+ CHECK(mState != STARTED);
+
+ status_t err = mFilter->setParameters(msg);
+ if (err != (status_t)OK) {
+ ALOGE("setParameters returned err %d", err);
+ }
+}
+
+void MediaFilter::onCreateInputSurface() {
+ CHECK(mState == CONFIGURED);
+
+ mGraphicBufferListener = new GraphicBufferListener;
+
+ sp<AMessage> notify = new AMessage();
+ notify->setTarget(id());
+ status_t err = mGraphicBufferListener->init(
+ notify, mStride, mSliceHeight, kBufferCountActual);
+
+ if (err != OK) {
+ ALOGE("Failed to init mGraphicBufferListener: %d", err);
+ signalError(err);
+ return;
+ }
+
+ sp<AMessage> reply = mNotify->dup();
+ reply->setInt32("what", CodecBase::kWhatInputSurfaceCreated);
+ reply->setObject(
+ "input-surface",
+ new BufferProducerWrapper(
+ mGraphicBufferListener->getIGraphicBufferProducer()));
+ reply->post();
+}
+
+void MediaFilter::onInputFrameAvailable() {
+ BufferQueue::BufferItem item = mGraphicBufferListener->getBufferItem();
+ sp<GraphicBuffer> buf = mGraphicBufferListener->getBuffer(item);
+
+ // get pointer to graphic buffer
+ void* bufPtr;
+ buf->lock(GraphicBuffer::USAGE_SW_READ_OFTEN, &bufPtr);
+
+ // HACK - there is no OMX_COLOR_FORMATTYPE value for RGBA, so the format
+ // conversion is hardcoded until we add this.
+ // TODO: check input format and convert only if necessary
+ // copy RGBA graphic buffer into temporary ARGB input buffer
+ BufferInfo *inputInfo = new BufferInfo;
+ inputInfo->mData = new ABuffer(buf->getWidth() * buf->getHeight() * 4);
+ ALOGV("Copying surface data into temp buffer.");
+ convertRGBAToARGB(
+ (uint8_t*)bufPtr, buf->getWidth(), buf->getHeight(),
+ buf->getStride(), inputInfo->mData->data());
+ inputInfo->mBufferID = item.mBuf;
+ inputInfo->mGeneration = mGeneration;
+ inputInfo->mOutputFlags = 0;
+ inputInfo->mStatus = BufferInfo::OWNED_BY_US;
+ inputInfo->mData->meta()->setInt64("timeUs", item.mTimestamp / 1000);
+
+ mAvailableInputBuffers.push_back(inputInfo);
+
+ mGraphicBufferListener->releaseBuffer(item);
+
+ signalProcessBuffers();
+}
+
+void MediaFilter::onSignalEndOfInputStream() {
+ // if using input surface, need to send an EOS output buffer
+ if (mGraphicBufferListener != NULL) {
+ Vector<BufferInfo> *outputBufs = &mBuffers[kPortIndexOutput];
+ BufferInfo* eosBuf;
+ bool foundBuf = false;
+ for (size_t i = 0; i < kBufferCountActual; i++) {
+ eosBuf = &outputBufs->editItemAt(i);
+ if (eosBuf->mStatus == BufferInfo::OWNED_BY_US) {
+ foundBuf = true;
+ break;
+ }
+ }
+
+ if (!foundBuf) {
+ ALOGE("onSignalEndOfInputStream failed to find an output buffer");
+ return;
+ }
+
+ eosBuf->mOutputFlags = OMX_BUFFERFLAG_EOS;
+ eosBuf->mGeneration = mGeneration;
+ eosBuf->mData->setRange(0, 0);
+ postDrainThisBuffer(eosBuf);
+ ALOGV("Posted EOS on output buffer %zu", eosBuf->mBufferID);
+ }
+
+ mPortEOS[kPortIndexOutput] = true;
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", CodecBase::kWhatSignaledInputEOS);
+ notify->post();
+
+ ALOGV("Output stream saw EOS.");
+}
+
+} // namespace android
diff --git a/media/libstagefright/filters/RSFilter.cpp b/media/libstagefright/filters/RSFilter.cpp
new file mode 100644
index 0000000..b569945
--- /dev/null
+++ b/media/libstagefright/filters/RSFilter.cpp
@@ -0,0 +1,96 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "RSFilter"
+
+#include <utils/Log.h>
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+
+#include "RSFilter.h"
+
+namespace android {
+
+RSFilter::RSFilter() {
+
+}
+
+RSFilter::~RSFilter() {
+
+}
+
+status_t RSFilter::configure(const sp<AMessage> &msg) {
+ status_t err = SimpleFilter::configure(msg);
+ if (err != OK) {
+ return err;
+ }
+
+ if (!msg->findString("cacheDir", &mCacheDir)) {
+ ALOGE("Failed to find cache directory in config message.");
+ return NAME_NOT_FOUND;
+ }
+
+ sp<RenderScriptWrapper> wrapper;
+ if (!msg->findObject("rs-wrapper", (sp<RefBase>*)&wrapper)) {
+ ALOGE("Failed to find RenderScriptWrapper in config message.");
+ return NAME_NOT_FOUND;
+ }
+
+ mRS = wrapper->mContext;
+ mCallback = wrapper->mCallback;
+
+ return OK;
+}
+
+status_t RSFilter::start() {
+ // 32-bit elements for ARGB8888
+ RSC::sp<const RSC::Element> e = RSC::Element::U8_4(mRS);
+
+ RSC::Type::Builder tb(mRS, e);
+ tb.setX(mWidth);
+ tb.setY(mHeight);
+ RSC::sp<const RSC::Type> t = tb.create();
+
+ mAllocIn = RSC::Allocation::createTyped(mRS, t);
+ mAllocOut = RSC::Allocation::createTyped(mRS, t);
+
+ return OK;
+}
+
+void RSFilter::reset() {
+ mCallback.clear();
+ mAllocOut.clear();
+ mAllocIn.clear();
+ mRS.clear();
+}
+
+status_t RSFilter::setParameters(const sp<AMessage> &msg) {
+ return mCallback->handleSetParameters(msg);
+}
+
+status_t RSFilter::processBuffers(
+ const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer) {
+ mAllocIn->copy1DRangeFrom(0, mWidth * mHeight, srcBuffer->data());
+ mCallback->processBuffers(mAllocIn.get(), mAllocOut.get());
+ mAllocOut->copy1DRangeTo(0, mWidth * mHeight, outBuffer->data());
+
+ return OK;
+}
+
+} // namespace android
diff --git a/media/libstagefright/filters/RSFilter.h b/media/libstagefright/filters/RSFilter.h
new file mode 100644
index 0000000..c5b5074
--- /dev/null
+++ b/media/libstagefright/filters/RSFilter.h
@@ -0,0 +1,53 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef RS_FILTER_H_
+#define RS_FILTER_H_
+
+#include <media/stagefright/RenderScriptWrapper.h>
+#include <RenderScript.h>
+
+#include "SimpleFilter.h"
+
+namespace android {
+
+struct AString;
+
+struct RSFilter : public SimpleFilter {
+public:
+ RSFilter();
+
+ virtual status_t configure(const sp<AMessage> &msg);
+ virtual status_t start();
+ virtual void reset();
+ virtual status_t setParameters(const sp<AMessage> &msg);
+ virtual status_t processBuffers(
+ const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer);
+
+protected:
+ virtual ~RSFilter();
+
+private:
+ AString mCacheDir;
+ sp<RenderScriptWrapper::RSFilterCallback> mCallback;
+ RSC::sp<RSC::RS> mRS;
+ RSC::sp<RSC::Allocation> mAllocIn;
+ RSC::sp<RSC::Allocation> mAllocOut;
+};
+
+} // namespace android
+
+#endif // RS_FILTER_H_
diff --git a/media/libstagefright/filters/SaturationFilter.cpp b/media/libstagefright/filters/SaturationFilter.cpp
new file mode 100644
index 0000000..ba5f75a
--- /dev/null
+++ b/media/libstagefright/filters/SaturationFilter.cpp
@@ -0,0 +1,99 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "SaturationFilter"
+
+#include <utils/Log.h>
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+
+#include "SaturationFilter.h"
+
+namespace android {
+
+status_t SaturationFilter::configure(const sp<AMessage> &msg) {
+ status_t err = SimpleFilter::configure(msg);
+ if (err != OK) {
+ return err;
+ }
+
+ if (!msg->findString("cacheDir", &mCacheDir)) {
+ ALOGE("Failed to find cache directory in config message.");
+ return NAME_NOT_FOUND;
+ }
+
+ return OK;
+}
+
+status_t SaturationFilter::start() {
+ // TODO: use a single RS context object for entire application
+ mRS = new RSC::RS();
+
+ if (!mRS->init(mCacheDir.c_str())) {
+ ALOGE("Failed to initialize RenderScript context.");
+ return NO_INIT;
+ }
+
+ // 32-bit elements for ARGB8888
+ RSC::sp<const RSC::Element> e = RSC::Element::U8_4(mRS);
+
+ RSC::Type::Builder tb(mRS, e);
+ tb.setX(mWidth);
+ tb.setY(mHeight);
+ RSC::sp<const RSC::Type> t = tb.create();
+
+ mAllocIn = RSC::Allocation::createTyped(mRS, t);
+ mAllocOut = RSC::Allocation::createTyped(mRS, t);
+
+ mScript = new ScriptC_saturationARGB(mRS);
+
+ mScript->set_gSaturation(mSaturation);
+
+ return OK;
+}
+
+void SaturationFilter::reset() {
+ mScript.clear();
+ mAllocOut.clear();
+ mAllocIn.clear();
+ mRS.clear();
+}
+
+status_t SaturationFilter::setParameters(const sp<AMessage> &msg) {
+ sp<AMessage> params;
+ CHECK(msg->findMessage("params", ¶ms));
+
+ float saturation;
+ if (params->findFloat("saturation", &saturation)) {
+ mSaturation = saturation;
+ }
+
+ return OK;
+}
+
+status_t SaturationFilter::processBuffers(
+ const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer) {
+ mAllocIn->copy1DRangeFrom(0, mWidth * mHeight, srcBuffer->data());
+ mScript->forEach_root(mAllocIn, mAllocOut);
+ mAllocOut->copy1DRangeTo(0, mWidth * mHeight, outBuffer->data());
+
+ return OK;
+}
+
+} // namespace android
diff --git a/media/libstagefright/filters/SaturationFilter.h b/media/libstagefright/filters/SaturationFilter.h
new file mode 100644
index 0000000..0545021
--- /dev/null
+++ b/media/libstagefright/filters/SaturationFilter.h
@@ -0,0 +1,52 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef SATURATION_FILTER_H_
+#define SATURATION_FILTER_H_
+
+#include <RenderScript.h>
+
+#include "ScriptC_saturationARGB.h"
+#include "SimpleFilter.h"
+
+namespace android {
+
+struct SaturationFilter : public SimpleFilter {
+public:
+ SaturationFilter() : mSaturation(1.f) {};
+
+ virtual status_t configure(const sp<AMessage> &msg);
+ virtual status_t start();
+ virtual void reset();
+ virtual status_t setParameters(const sp<AMessage> &msg);
+ virtual status_t processBuffers(
+ const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer);
+
+protected:
+ virtual ~SaturationFilter() {};
+
+private:
+ AString mCacheDir;
+ RSC::sp<RSC::RS> mRS;
+ RSC::sp<RSC::Allocation> mAllocIn;
+ RSC::sp<RSC::Allocation> mAllocOut;
+ RSC::sp<ScriptC_saturationARGB> mScript;
+ float mSaturation;
+};
+
+} // namespace android
+
+#endif // SATURATION_FILTER_H_
diff --git a/media/libstagefright/filters/SimpleFilter.cpp b/media/libstagefright/filters/SimpleFilter.cpp
new file mode 100644
index 0000000..6c1ca2c
--- /dev/null
+++ b/media/libstagefright/filters/SimpleFilter.cpp
@@ -0,0 +1,39 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+
+#include "SimpleFilter.h"
+
+namespace android {
+
+status_t SimpleFilter::configure(const sp<AMessage> &msg) {
+ CHECK(msg->findInt32("width", &mWidth));
+ CHECK(msg->findInt32("height", &mHeight));
+ if (!msg->findInt32("stride", &mStride)) {
+ mStride = mWidth;
+ }
+ if (!msg->findInt32("slice-height", &mSliceHeight)) {
+ mSliceHeight = mHeight;
+ }
+ CHECK(msg->findInt32("color-format", &mColorFormatIn));
+ mColorFormatOut = mColorFormatIn;
+
+ return OK;
+}
+
+} // namespace android
diff --git a/media/libstagefright/filters/SimpleFilter.h b/media/libstagefright/filters/SimpleFilter.h
new file mode 100644
index 0000000..4cd37ef
--- /dev/null
+++ b/media/libstagefright/filters/SimpleFilter.h
@@ -0,0 +1,52 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef SIMPLE_FILTER_H_
+#define SIMPLE_FILTER_H_
+
+#include <stdint.h>
+#include <utils/Errors.h>
+#include <utils/RefBase.h>
+
+struct ABuffer;
+struct AMessage;
+
+namespace android {
+
+struct SimpleFilter : public RefBase {
+public:
+ SimpleFilter() : mWidth(0), mHeight(0), mStride(0), mSliceHeight(0),
+ mColorFormatIn(0), mColorFormatOut(0) {};
+
+ virtual status_t configure(const sp<AMessage> &msg);
+
+ virtual status_t start() = 0;
+ virtual void reset() = 0;
+ virtual status_t setParameters(const sp<AMessage> &msg) = 0;
+ virtual status_t processBuffers(
+ const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer) = 0;
+
+protected:
+ int32_t mWidth, mHeight;
+ int32_t mStride, mSliceHeight;
+ int32_t mColorFormatIn, mColorFormatOut;
+
+ virtual ~SimpleFilter() {};
+};
+
+} // namespace android
+
+#endif // SIMPLE_FILTER_H_
diff --git a/media/libstagefright/filters/ZeroFilter.cpp b/media/libstagefright/filters/ZeroFilter.cpp
new file mode 100644
index 0000000..3f1243c
--- /dev/null
+++ b/media/libstagefright/filters/ZeroFilter.cpp
@@ -0,0 +1,57 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "ZeroFilter"
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+
+#include "ZeroFilter.h"
+
+namespace android {
+
+status_t ZeroFilter::setParameters(const sp<AMessage> &msg) {
+ sp<AMessage> params;
+ CHECK(msg->findMessage("params", ¶ms));
+
+ int32_t invert;
+ if (params->findInt32("invert", &invert)) {
+ mInvertData = (invert != 0);
+ }
+
+ return OK;
+}
+
+status_t ZeroFilter::processBuffers(
+ const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer) {
+ // assuming identical input & output buffers, since we're a copy filter
+ if (mInvertData) {
+ uint32_t* src = (uint32_t*)srcBuffer->data();
+ uint32_t* dest = (uint32_t*)outBuffer->data();
+ for (size_t i = 0; i < srcBuffer->size() / 4; ++i) {
+ *(dest++) = *(src++) ^ 0xFFFFFFFF;
+ }
+ } else {
+ memcpy(outBuffer->data(), srcBuffer->data(), srcBuffer->size());
+ }
+ outBuffer->setRange(0, srcBuffer->size());
+
+ return OK;
+}
+
+} // namespace android
diff --git a/media/libstagefright/filters/ZeroFilter.h b/media/libstagefright/filters/ZeroFilter.h
new file mode 100644
index 0000000..bd34dfb
--- /dev/null
+++ b/media/libstagefright/filters/ZeroFilter.h
@@ -0,0 +1,43 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ZERO_FILTER_H_
+#define ZERO_FILTER_H_
+
+#include "SimpleFilter.h"
+
+namespace android {
+
+struct ZeroFilter : public SimpleFilter {
+public:
+ ZeroFilter() : mInvertData(false) {};
+
+ virtual status_t start() { return OK; };
+ virtual void reset() {};
+ virtual status_t setParameters(const sp<AMessage> &msg);
+ virtual status_t processBuffers(
+ const sp<ABuffer> &srcBuffer, const sp<ABuffer> &outBuffer);
+
+protected:
+ virtual ~ZeroFilter() {};
+
+private:
+ bool mInvertData;
+};
+
+} // namespace android
+
+#endif // ZERO_FILTER_H_
diff --git a/media/libstagefright/filters/saturation.rs b/media/libstagefright/filters/saturation.rs
new file mode 100644
index 0000000..2c867ac
--- /dev/null
+++ b/media/libstagefright/filters/saturation.rs
@@ -0,0 +1,40 @@
+// Sample script for RGB888 support (compare to saturationARGB.rs)
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#pragma version(1)
+#pragma rs java_package_name(com.android.rs.cppbasic)
+#pragma rs_fp_relaxed
+
+const static float3 gMonoMult = {0.299f, 0.587f, 0.114f};
+
+// global variables (parameters accessible to application code)
+float gSaturation = 1.0f;
+
+void root(const uchar3 *v_in, uchar3 *v_out) {
+ // scale 0-255 uchar to 0-1.0 float
+ float3 in = {v_in->r * 0.003921569f, v_in->g * 0.003921569f,
+ v_in->b * 0.003921569f};
+
+ // apply saturation filter
+ float3 result = dot(in, gMonoMult);
+ result = mix(result, in, gSaturation);
+
+ // convert to uchar, copied from rsPackColorTo8888
+ v_out->x = (uchar)clamp((result.r * 255.f + 0.5f), 0.f, 255.f);
+ v_out->y = (uchar)clamp((result.g * 255.f + 0.5f), 0.f, 255.f);
+ v_out->z = (uchar)clamp((result.b * 255.f + 0.5f), 0.f, 255.f);
+}
diff --git a/media/libstagefright/filters/saturationARGB.rs b/media/libstagefright/filters/saturationARGB.rs
new file mode 100644
index 0000000..1de9dd8
--- /dev/null
+++ b/media/libstagefright/filters/saturationARGB.rs
@@ -0,0 +1,40 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#pragma version(1)
+#pragma rs java_package_name(com.android.rs.cppbasic)
+#pragma rs_fp_relaxed
+
+const static float3 gMonoMult = {0.299f, 0.587f, 0.114f};
+
+// global variables (parameters accessible to application code)
+float gSaturation = 1.0f;
+
+void root(const uchar4 *v_in, uchar4 *v_out) {
+ v_out->x = v_in->x; // don't modify A
+
+ // get RGB, scale 0-255 uchar to 0-1.0 float
+ float3 rgb = {v_in->y * 0.003921569f, v_in->z * 0.003921569f,
+ v_in->w * 0.003921569f};
+
+ // apply saturation filter
+ float3 result = dot(rgb, gMonoMult);
+ result = mix(result, rgb, gSaturation);
+
+ v_out->y = (uchar)clamp((result.r * 255.f + 0.5f), 0.f, 255.f);
+ v_out->z = (uchar)clamp((result.g * 255.f + 0.5f), 0.f, 255.f);
+ v_out->w = (uchar)clamp((result.b * 255.f + 0.5f), 0.f, 255.f);
+}
diff --git a/media/libstagefright/include/MPEG4Extractor.h b/media/libstagefright/include/MPEG4Extractor.h
index 1fe6fcf..8c16251 100644
--- a/media/libstagefright/include/MPEG4Extractor.h
+++ b/media/libstagefright/include/MPEG4Extractor.h
@@ -83,6 +83,8 @@
Vector<SidxEntry> mSidxEntries;
off64_t mMoofOffset;
+ bool mMoofFound;
+ bool mMdatFound;
Vector<PsshInfo> mPssh;
diff --git a/media/libstagefright/include/avc_utils.h b/media/libstagefright/include/avc_utils.h
index c270bc1..dafa07e 100644
--- a/media/libstagefright/include/avc_utils.h
+++ b/media/libstagefright/include/avc_utils.h
@@ -36,6 +36,11 @@
kAVCProfileCAVLC444Intra = 0x2c
};
+struct NALPosition {
+ size_t nalOffset;
+ size_t nalSize;
+};
+
// Optionally returns sample aspect ratio as well.
void FindAVCDimensions(
const sp<ABuffer> &seqParamSet,
diff --git a/media/libstagefright/mpeg2ts/ESQueue.cpp b/media/libstagefright/mpeg2ts/ESQueue.cpp
index a605595..88da275 100644
--- a/media/libstagefright/mpeg2ts/ESQueue.cpp
+++ b/media/libstagefright/mpeg2ts/ESQueue.cpp
@@ -617,8 +617,6 @@
// having to interpolate.
// The final AAC frame may well extend into the next RangeInfo but
// that's ok.
- // TODO: the logic commented above is skipped because codec cannot take
- // arbitrary sized input buffers;
size_t offset = 0;
while (offset < info.mLength) {
if (offset + 7 > mBuffer->size()) {
@@ -683,12 +681,9 @@
size_t headerSize __unused = protection_absent ? 7 : 9;
offset += aac_frame_length;
- // TODO: move back to concatenation when codec can support arbitrary input buffers.
- // For now only queue a single buffer
- break;
}
- int64_t timeUs = fetchTimestampAAC(offset);
+ int64_t timeUs = fetchTimestamp(offset);
sp<ABuffer> accessUnit = new ABuffer(offset);
memcpy(accessUnit->data(), mBuffer->data(), offset);
@@ -735,50 +730,6 @@
return timeUs;
}
-// TODO: avoid interpolating timestamps once codec supports arbitrary sized input buffers
-int64_t ElementaryStreamQueue::fetchTimestampAAC(size_t size) {
- int64_t timeUs = -1;
- bool first = true;
-
- size_t samplesize = size;
- while (size > 0) {
- CHECK(!mRangeInfos.empty());
-
- RangeInfo *info = &*mRangeInfos.begin();
-
- if (first) {
- timeUs = info->mTimestampUs;
- first = false;
- }
-
- if (info->mLength > size) {
- int32_t sampleRate;
- CHECK(mFormat->findInt32(kKeySampleRate, &sampleRate));
- info->mLength -= size;
- size_t numSamples = 1024 * size / samplesize;
- info->mTimestampUs += numSamples * 1000000ll / sampleRate;
- size = 0;
- } else {
- size -= info->mLength;
-
- mRangeInfos.erase(mRangeInfos.begin());
- info = NULL;
- }
-
- }
-
- if (timeUs == 0ll) {
- ALOGV("Returning 0 timestamp");
- }
-
- return timeUs;
-}
-
-struct NALPosition {
- size_t nalOffset;
- size_t nalSize;
-};
-
sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnitH264() {
const uint8_t *data = mBuffer->data();
@@ -786,6 +737,7 @@
Vector<NALPosition> nals;
size_t totalSize = 0;
+ size_t seiCount = 0;
status_t err;
const uint8_t *nalStart;
@@ -815,6 +767,9 @@
// next frame.
flush = true;
+ } else if (nalType == 6 && nalSize > 0) {
+ // found non-zero sized SEI
+ ++seiCount;
}
if (flush) {
@@ -823,21 +778,29 @@
size_t auSize = 4 * nals.size() + totalSize;
sp<ABuffer> accessUnit = new ABuffer(auSize);
+ sp<ABuffer> sei;
+
+ if (seiCount > 0) {
+ sei = new ABuffer(seiCount * sizeof(NALPosition));
+ accessUnit->meta()->setBuffer("sei", sei);
+ }
#if !LOG_NDEBUG
AString out;
#endif
size_t dstOffset = 0;
+ size_t seiIndex = 0;
for (size_t i = 0; i < nals.size(); ++i) {
const NALPosition &pos = nals.itemAt(i);
unsigned nalType = mBuffer->data()[pos.nalOffset] & 0x1f;
- if (nalType == 6) {
- sp<ABuffer> sei = new ABuffer(pos.nalSize);
- memcpy(sei->data(), mBuffer->data() + pos.nalOffset, pos.nalSize);
- accessUnit->meta()->setBuffer("sei", sei);
+ if (nalType == 6 && pos.nalSize > 0) {
+ CHECK_LT(seiIndex, sei->size() / sizeof(NALPosition));
+ NALPosition &seiPos = ((NALPosition *)sei->data())[seiIndex++];
+ seiPos.nalOffset = dstOffset + 4;
+ seiPos.nalSize = pos.nalSize;
}
#if !LOG_NDEBUG
diff --git a/media/libstagefright/mpeg2ts/ESQueue.h b/media/libstagefright/mpeg2ts/ESQueue.h
index eb4b1c9..45b4624 100644
--- a/media/libstagefright/mpeg2ts/ESQueue.h
+++ b/media/libstagefright/mpeg2ts/ESQueue.h
@@ -77,7 +77,6 @@
// consume a logical (compressed) access unit of size "size",
// returns its timestamp in us (or -1 if no time information).
int64_t fetchTimestamp(size_t size);
- int64_t fetchTimestampAAC(size_t size);
DISALLOW_EVIL_CONSTRUCTORS(ElementaryStreamQueue);
};
diff --git a/media/libstagefright/omx/Android.mk b/media/libstagefright/omx/Android.mk
index aaa8334..be8cf46 100644
--- a/media/libstagefright/omx/Android.mk
+++ b/media/libstagefright/omx/Android.mk
@@ -6,6 +6,7 @@
endif
LOCAL_SRC_FILES:= \
+ FrameDropper.cpp \
GraphicBufferSource.cpp \
OMX.cpp \
OMXMaster.cpp \
diff --git a/media/libstagefright/omx/FrameDropper.cpp b/media/libstagefright/omx/FrameDropper.cpp
new file mode 100644
index 0000000..9fba0b7
--- /dev/null
+++ b/media/libstagefright/omx/FrameDropper.cpp
@@ -0,0 +1,70 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "FrameDropper"
+#include <utils/Log.h>
+
+#include "FrameDropper.h"
+
+#include <media/stagefright/foundation/ADebug.h>
+
+namespace android {
+
+static const int64_t kMaxJitterUs = 2000;
+
+FrameDropper::FrameDropper()
+ : mDesiredMinTimeUs(-1),
+ mMinIntervalUs(0) {
+}
+
+FrameDropper::~FrameDropper() {
+}
+
+status_t FrameDropper::setMaxFrameRate(float maxFrameRate) {
+ if (maxFrameRate <= 0) {
+ ALOGE("framerate should be positive but got %f.", maxFrameRate);
+ return BAD_VALUE;
+ }
+ mMinIntervalUs = (int64_t) (1000000.0f / maxFrameRate);
+ return OK;
+}
+
+bool FrameDropper::shouldDrop(int64_t timeUs) {
+ if (mMinIntervalUs <= 0) {
+ return false;
+ }
+
+ if (mDesiredMinTimeUs < 0) {
+ mDesiredMinTimeUs = timeUs + mMinIntervalUs;
+ ALOGV("first frame %lld, next desired frame %lld", timeUs, mDesiredMinTimeUs);
+ return false;
+ }
+
+ if (timeUs < (mDesiredMinTimeUs - kMaxJitterUs)) {
+ ALOGV("drop frame %lld, desired frame %lld, diff %lld",
+ timeUs, mDesiredMinTimeUs, mDesiredMinTimeUs - timeUs);
+ return true;
+ }
+
+ int64_t n = (timeUs - mDesiredMinTimeUs + kMaxJitterUs) / mMinIntervalUs;
+ mDesiredMinTimeUs += (n + 1) * mMinIntervalUs;
+ ALOGV("keep frame %lld, next desired frame %lld, diff %lld",
+ timeUs, mDesiredMinTimeUs, mDesiredMinTimeUs - timeUs);
+ return false;
+}
+
+} // namespace android
diff --git a/media/libstagefright/omx/FrameDropper.h b/media/libstagefright/omx/FrameDropper.h
new file mode 100644
index 0000000..c5a6d4b
--- /dev/null
+++ b/media/libstagefright/omx/FrameDropper.h
@@ -0,0 +1,50 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef FRAME_DROPPER_H_
+
+#define FRAME_DROPPER_H_
+
+#include <utils/Errors.h>
+#include <utils/RefBase.h>
+
+#include <media/stagefright/foundation/ABase.h>
+
+namespace android {
+
+struct FrameDropper : public RefBase {
+ // No frames will be dropped until a valid max frame rate is set.
+ FrameDropper();
+
+ // maxFrameRate required to be positive.
+ status_t setMaxFrameRate(float maxFrameRate);
+
+ // Returns false if max frame rate has not been set via setMaxFrameRate.
+ bool shouldDrop(int64_t timeUs);
+
+protected:
+ virtual ~FrameDropper();
+
+private:
+ int64_t mDesiredMinTimeUs;
+ int64_t mMinIntervalUs;
+
+ DISALLOW_EVIL_CONSTRUCTORS(FrameDropper);
+};
+
+} // namespace android
+
+#endif // FRAME_DROPPER_H_
diff --git a/media/libstagefright/omx/GraphicBufferSource.cpp b/media/libstagefright/omx/GraphicBufferSource.cpp
index 44c7edc..7afe699 100644
--- a/media/libstagefright/omx/GraphicBufferSource.cpp
+++ b/media/libstagefright/omx/GraphicBufferSource.cpp
@@ -30,6 +30,7 @@
#include <ui/GraphicBuffer.h>
#include <inttypes.h>
+#include "FrameDropper.h"
namespace android {
@@ -53,9 +54,9 @@
mRepeatAfterUs(-1ll),
mRepeatLastFrameGeneration(0),
mRepeatLastFrameTimestamp(-1ll),
- mLatestSubmittedBufferId(-1),
- mLatestSubmittedBufferFrameNum(0),
- mLatestSubmittedBufferUseCount(0),
+ mLatestBufferId(-1),
+ mLatestBufferFrameNum(0),
+ mLatestBufferUseCount(0),
mRepeatBufferDeferred(false),
mTimePerCaptureUs(-1ll),
mTimePerFrameUs(-1ll),
@@ -152,7 +153,7 @@
mLooper->registerHandler(mReflector);
mLooper->start();
- if (mLatestSubmittedBufferId >= 0) {
+ if (mLatestBufferId >= 0) {
sp<AMessage> msg =
new AMessage(kWhatRepeatLastFrame, mReflector->id());
@@ -287,8 +288,8 @@
ALOGV("cbi %d matches bq slot %d, handle=%p",
cbi, id, mBufferSlot[id]->handle);
- if (id == mLatestSubmittedBufferId) {
- CHECK_GT(mLatestSubmittedBufferUseCount--, 0);
+ if (id == mLatestBufferId) {
+ CHECK_GT(mLatestBufferUseCount--, 0);
} else {
mConsumer->releaseBuffer(id, codecBuffer.mFrameNumber,
EGL_NO_DISPLAY, EGL_NO_SYNC_KHR, Fence::NO_FENCE);
@@ -313,11 +314,11 @@
ALOGV("buffer freed, EOS pending");
submitEndOfInputStream_l();
} else if (mRepeatBufferDeferred) {
- bool success = repeatLatestSubmittedBuffer_l();
+ bool success = repeatLatestBuffer_l();
if (success) {
- ALOGV("deferred repeatLatestSubmittedBuffer_l SUCCESS");
+ ALOGV("deferred repeatLatestBuffer_l SUCCESS");
} else {
- ALOGV("deferred repeatLatestSubmittedBuffer_l FAILURE");
+ ALOGV("deferred repeatLatestBuffer_l FAILURE");
}
mRepeatBufferDeferred = false;
}
@@ -382,12 +383,12 @@
mSuspended = false;
if (mExecuting && mNumFramesAvailable == 0 && mRepeatBufferDeferred) {
- if (repeatLatestSubmittedBuffer_l()) {
- ALOGV("suspend/deferred repeatLatestSubmittedBuffer_l SUCCESS");
+ if (repeatLatestBuffer_l()) {
+ ALOGV("suspend/deferred repeatLatestBuffer_l SUCCESS");
mRepeatBufferDeferred = false;
} else {
- ALOGV("suspend/deferred repeatLatestSubmittedBuffer_l FAILURE");
+ ALOGV("suspend/deferred repeatLatestBuffer_l FAILURE");
}
}
}
@@ -441,12 +442,22 @@
// only submit sample if start time is unspecified, or sample
// is queued after the specified start time
+ bool dropped = false;
if (mSkipFramesBeforeNs < 0ll || item.mTimestamp >= mSkipFramesBeforeNs) {
// if start time is set, offset time stamp by start time
if (mSkipFramesBeforeNs > 0) {
item.mTimestamp -= mSkipFramesBeforeNs;
}
- err = submitBuffer_l(item, cbi);
+
+ int64_t timeUs = item.mTimestamp / 1000;
+ if (mFrameDropper != NULL && mFrameDropper->shouldDrop(timeUs)) {
+ ALOGV("skipping frame (%lld) to meet max framerate", static_cast<long long>(timeUs));
+ // set err to OK so that the skipped frame can still be saved as the lastest frame
+ err = OK;
+ dropped = true;
+ } else {
+ err = submitBuffer_l(item, cbi);
+ }
}
if (err != OK) {
@@ -455,46 +466,46 @@
EGL_NO_DISPLAY, EGL_NO_SYNC_KHR, Fence::NO_FENCE);
} else {
ALOGV("buffer submitted (bq %d, cbi %d)", item.mBuf, cbi);
- setLatestSubmittedBuffer_l(item);
+ setLatestBuffer_l(item, dropped);
}
return true;
}
-bool GraphicBufferSource::repeatLatestSubmittedBuffer_l() {
+bool GraphicBufferSource::repeatLatestBuffer_l() {
CHECK(mExecuting && mNumFramesAvailable == 0);
- if (mLatestSubmittedBufferId < 0 || mSuspended) {
+ if (mLatestBufferId < 0 || mSuspended) {
return false;
}
- if (mBufferSlot[mLatestSubmittedBufferId] == NULL) {
+ if (mBufferSlot[mLatestBufferId] == NULL) {
// This can happen if the remote side disconnects, causing
// onBuffersReleased() to NULL out our copy of the slots. The
// buffer is gone, so we have nothing to show.
//
// To be on the safe side we try to release the buffer.
- ALOGD("repeatLatestSubmittedBuffer_l: slot was NULL");
+ ALOGD("repeatLatestBuffer_l: slot was NULL");
mConsumer->releaseBuffer(
- mLatestSubmittedBufferId,
- mLatestSubmittedBufferFrameNum,
+ mLatestBufferId,
+ mLatestBufferFrameNum,
EGL_NO_DISPLAY,
EGL_NO_SYNC_KHR,
Fence::NO_FENCE);
- mLatestSubmittedBufferId = -1;
- mLatestSubmittedBufferFrameNum = 0;
+ mLatestBufferId = -1;
+ mLatestBufferFrameNum = 0;
return false;
}
int cbi = findAvailableCodecBuffer_l();
if (cbi < 0) {
// No buffers available, bail.
- ALOGV("repeatLatestSubmittedBuffer_l: no codec buffers.");
+ ALOGV("repeatLatestBuffer_l: no codec buffers.");
return false;
}
BufferQueue::BufferItem item;
- item.mBuf = mLatestSubmittedBufferId;
- item.mFrameNumber = mLatestSubmittedBufferFrameNum;
+ item.mBuf = mLatestBufferId;
+ item.mFrameNumber = mLatestBufferFrameNum;
item.mTimestamp = mRepeatLastFrameTimestamp;
status_t err = submitBuffer_l(item, cbi);
@@ -503,7 +514,7 @@
return false;
}
- ++mLatestSubmittedBufferUseCount;
+ ++mLatestBufferUseCount;
/* repeat last frame up to kRepeatLastFrameCount times.
* in case of static scene, a single repeat might not get rid of encoder
@@ -522,26 +533,26 @@
return true;
}
-void GraphicBufferSource::setLatestSubmittedBuffer_l(
- const BufferQueue::BufferItem &item) {
- ALOGV("setLatestSubmittedBuffer_l");
+void GraphicBufferSource::setLatestBuffer_l(
+ const BufferQueue::BufferItem &item, bool dropped) {
+ ALOGV("setLatestBuffer_l");
- if (mLatestSubmittedBufferId >= 0) {
- if (mLatestSubmittedBufferUseCount == 0) {
+ if (mLatestBufferId >= 0) {
+ if (mLatestBufferUseCount == 0) {
mConsumer->releaseBuffer(
- mLatestSubmittedBufferId,
- mLatestSubmittedBufferFrameNum,
+ mLatestBufferId,
+ mLatestBufferFrameNum,
EGL_NO_DISPLAY,
EGL_NO_SYNC_KHR,
Fence::NO_FENCE);
}
}
- mLatestSubmittedBufferId = item.mBuf;
- mLatestSubmittedBufferFrameNum = item.mFrameNumber;
+ mLatestBufferId = item.mBuf;
+ mLatestBufferFrameNum = item.mFrameNumber;
mRepeatLastFrameTimestamp = item.mTimestamp + mRepeatAfterUs * 1000;
- mLatestSubmittedBufferUseCount = 1;
+ mLatestBufferUseCount = dropped ? 0 : 1;
mRepeatBufferDeferred = false;
mRepeatLastFrameCount = kRepeatLastFrameCount;
@@ -841,6 +852,23 @@
return OK;
}
+status_t GraphicBufferSource::setMaxFps(float maxFps) {
+ Mutex::Autolock autoLock(mMutex);
+
+ if (mExecuting) {
+ return INVALID_OPERATION;
+ }
+
+ mFrameDropper = new FrameDropper();
+ status_t err = mFrameDropper->setMaxFrameRate(maxFps);
+ if (err != OK) {
+ mFrameDropper.clear();
+ return err;
+ }
+
+ return OK;
+}
+
void GraphicBufferSource::setSkipFramesBeforeUs(int64_t skipFramesBeforeUs) {
Mutex::Autolock autoLock(mMutex);
@@ -879,12 +907,12 @@
break;
}
- bool success = repeatLatestSubmittedBuffer_l();
+ bool success = repeatLatestBuffer_l();
if (success) {
- ALOGV("repeatLatestSubmittedBuffer_l SUCCESS");
+ ALOGV("repeatLatestBuffer_l SUCCESS");
} else {
- ALOGV("repeatLatestSubmittedBuffer_l FAILURE");
+ ALOGV("repeatLatestBuffer_l FAILURE");
mRepeatBufferDeferred = true;
}
break;
diff --git a/media/libstagefright/omx/GraphicBufferSource.h b/media/libstagefright/omx/GraphicBufferSource.h
index c8e3775..ce3881e 100644
--- a/media/libstagefright/omx/GraphicBufferSource.h
+++ b/media/libstagefright/omx/GraphicBufferSource.h
@@ -30,6 +30,8 @@
namespace android {
+class FrameDropper;
+
/*
* This class is used to feed OMX codecs from a Surface via BufferQueue.
*
@@ -119,6 +121,9 @@
// of suspension on input.
status_t setMaxTimestampGapUs(int64_t maxGapUs);
+ // When set, the max frame rate fed to the encoder will be capped at maxFps.
+ status_t setMaxFps(float maxFps);
+
// Sets the time lapse (or slow motion) parameters.
// data[0] is the time (us) between two frames for playback
// data[1] is the time (us) between two frames for capture
@@ -193,8 +198,8 @@
// doing anything if we don't have a codec buffer available.
void submitEndOfInputStream_l();
- void setLatestSubmittedBuffer_l(const BufferQueue::BufferItem &item);
- bool repeatLatestSubmittedBuffer_l();
+ void setLatestBuffer_l(const BufferQueue::BufferItem &item, bool dropped);
+ bool repeatLatestBuffer_l();
int64_t getTimestamp(const BufferQueue::BufferItem &item);
// Lock, covers all member variables.
@@ -250,6 +255,8 @@
int64_t mPrevModifiedTimeUs;
int64_t mSkipFramesBeforeNs;
+ sp<FrameDropper> mFrameDropper;
+
sp<ALooper> mLooper;
sp<AHandlerReflector<GraphicBufferSource> > mReflector;
@@ -258,11 +265,11 @@
int64_t mRepeatLastFrameTimestamp;
int32_t mRepeatLastFrameCount;
- int mLatestSubmittedBufferId;
- uint64_t mLatestSubmittedBufferFrameNum;
- int32_t mLatestSubmittedBufferUseCount;
+ int mLatestBufferId;
+ uint64_t mLatestBufferFrameNum;
+ int32_t mLatestBufferUseCount;
- // The previously submitted buffer should've been repeated but
+ // The previous buffer should've been repeated but
// no codec buffer was available at the time.
bool mRepeatBufferDeferred;
diff --git a/media/libstagefright/omx/OMXNodeInstance.cpp b/media/libstagefright/omx/OMXNodeInstance.cpp
index c04d95f..bf59460 100644
--- a/media/libstagefright/omx/OMXNodeInstance.cpp
+++ b/media/libstagefright/omx/OMXNodeInstance.cpp
@@ -1075,6 +1075,7 @@
case IOMX::INTERNAL_OPTION_REPEAT_PREVIOUS_FRAME_DELAY:
return "REPEAT_PREVIOUS_FRAME_DELAY";
case IOMX::INTERNAL_OPTION_MAX_TIMESTAMP_GAP: return "MAX_TIMESTAMP_GAP";
+ case IOMX::INTERNAL_OPTION_MAX_FPS: return "MAX_FPS";
case IOMX::INTERNAL_OPTION_START_TIME: return "START_TIME";
case IOMX::INTERNAL_OPTION_TIME_LAPSE: return "TIME_LAPSE";
default: return def;
@@ -1092,6 +1093,7 @@
case IOMX::INTERNAL_OPTION_SUSPEND:
case IOMX::INTERNAL_OPTION_REPEAT_PREVIOUS_FRAME_DELAY:
case IOMX::INTERNAL_OPTION_MAX_TIMESTAMP_GAP:
+ case IOMX::INTERNAL_OPTION_MAX_FPS:
case IOMX::INTERNAL_OPTION_START_TIME:
case IOMX::INTERNAL_OPTION_TIME_LAPSE:
{
@@ -1129,6 +1131,14 @@
int64_t maxGapUs = *(int64_t *)data;
CLOG_CONFIG(setInternalOption, "gapUs=%lld", (long long)maxGapUs);
return bufferSource->setMaxTimestampGapUs(maxGapUs);
+ } else if (type == IOMX::INTERNAL_OPTION_MAX_FPS) {
+ if (size != sizeof(float)) {
+ return INVALID_OPERATION;
+ }
+
+ float maxFps = *(float *)data;
+ CLOG_CONFIG(setInternalOption, "maxFps=%f", maxFps);
+ return bufferSource->setMaxFps(maxFps);
} else if (type == IOMX::INTERNAL_OPTION_START_TIME) {
if (size != sizeof(int64_t)) {
return INVALID_OPERATION;
diff --git a/media/libstagefright/omx/tests/Android.mk b/media/libstagefright/omx/tests/Android.mk
index 447b29e..9be637a 100644
--- a/media/libstagefright/omx/tests/Android.mk
+++ b/media/libstagefright/omx/tests/Android.mk
@@ -20,3 +20,21 @@
LOCAL_32_BIT_ONLY := true
include $(BUILD_EXECUTABLE)
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE := FrameDropper_test
+
+LOCAL_MODULE_TAGS := tests
+
+LOCAL_SRC_FILES := \
+ FrameDropper_test.cpp \
+
+LOCAL_SHARED_LIBRARIES := \
+ libstagefright_omx \
+ libutils \
+
+LOCAL_C_INCLUDES := \
+ frameworks/av/media/libstagefright/omx \
+
+include $(BUILD_NATIVE_TEST)
diff --git a/media/libstagefright/omx/tests/FrameDropper_test.cpp b/media/libstagefright/omx/tests/FrameDropper_test.cpp
new file mode 100644
index 0000000..4ac72c4
--- /dev/null
+++ b/media/libstagefright/omx/tests/FrameDropper_test.cpp
@@ -0,0 +1,136 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "FrameDropper_test"
+#include <utils/Log.h>
+
+#include <gtest/gtest.h>
+
+#include "FrameDropper.h"
+#include <media/stagefright/foundation/ADebug.h>
+
+namespace android {
+
+struct TestFrame {
+ int64_t timeUs;
+ bool shouldDrop;
+};
+
+static const TestFrame testFrames20Fps[] = {
+ {1000000, false}, {1050000, false}, {1100000, false}, {1150000, false},
+ {1200000, false}, {1250000, false}, {1300000, false}, {1350000, false},
+ {1400000, false}, {1450000, false}, {1500000, false}, {1550000, false},
+ {1600000, false}, {1650000, false}, {1700000, false}, {1750000, false},
+ {1800000, false}, {1850000, false}, {1900000, false}, {1950000, false},
+};
+
+static const TestFrame testFrames30Fps[] = {
+ {1000000, false}, {1033333, false}, {1066667, false}, {1100000, false},
+ {1133333, false}, {1166667, false}, {1200000, false}, {1233333, false},
+ {1266667, false}, {1300000, false}, {1333333, false}, {1366667, false},
+ {1400000, false}, {1433333, false}, {1466667, false}, {1500000, false},
+ {1533333, false}, {1566667, false}, {1600000, false}, {1633333, false},
+};
+
+static const TestFrame testFrames40Fps[] = {
+ {1000000, false}, {1025000, true}, {1050000, false}, {1075000, false},
+ {1100000, false}, {1125000, true}, {1150000, false}, {1175000, false},
+ {1200000, false}, {1225000, true}, {1250000, false}, {1275000, false},
+ {1300000, false}, {1325000, true}, {1350000, false}, {1375000, false},
+ {1400000, false}, {1425000, true}, {1450000, false}, {1475000, false},
+};
+
+static const TestFrame testFrames60Fps[] = {
+ {1000000, false}, {1016667, true}, {1033333, false}, {1050000, true},
+ {1066667, false}, {1083333, true}, {1100000, false}, {1116667, true},
+ {1133333, false}, {1150000, true}, {1166667, false}, {1183333, true},
+ {1200000, false}, {1216667, true}, {1233333, false}, {1250000, true},
+ {1266667, false}, {1283333, true}, {1300000, false}, {1316667, true},
+};
+
+static const TestFrame testFramesVariableFps[] = {
+ // 40fps
+ {1000000, false}, {1025000, true}, {1050000, false}, {1075000, false},
+ {1100000, false}, {1125000, true}, {1150000, false}, {1175000, false},
+ {1200000, false}, {1225000, true}, {1250000, false}, {1275000, false},
+ {1300000, false}, {1325000, true}, {1350000, false}, {1375000, false},
+ {1400000, false}, {1425000, true}, {1450000, false}, {1475000, false},
+ // a timestamp jump plus switch to 20fps
+ {2000000, false}, {2050000, false}, {2100000, false}, {2150000, false},
+ {2200000, false}, {2250000, false}, {2300000, false}, {2350000, false},
+ {2400000, false}, {2450000, false}, {2500000, false}, {2550000, false},
+ {2600000, false}, {2650000, false}, {2700000, false}, {2750000, false},
+ {2800000, false}, {2850000, false}, {2900000, false}, {2950000, false},
+ // 60fps
+ {2966667, false}, {2983333, true}, {3000000, false}, {3016667, true},
+ {3033333, false}, {3050000, true}, {3066667, false}, {3083333, true},
+ {3100000, false}, {3116667, true}, {3133333, false}, {3150000, true},
+ {3166667, false}, {3183333, true}, {3200000, false}, {3216667, true},
+ {3233333, false}, {3250000, true}, {3266667, false}, {3283333, true},
+};
+
+static const int kMaxTestJitterUs = 2000;
+// return one of 1000, 0, -1000 as jitter.
+static int GetJitter(size_t i) {
+ return (1 - (i % 3)) * (kMaxTestJitterUs / 2);
+}
+
+class FrameDropperTest : public ::testing::Test {
+public:
+ FrameDropperTest() : mFrameDropper(new FrameDropper()) {
+ EXPECT_EQ(OK, mFrameDropper->setMaxFrameRate(30.0));
+ }
+
+protected:
+ void RunTest(const TestFrame* frames, size_t size) {
+ for (size_t i = 0; i < size; ++i) {
+ int jitter = GetJitter(i);
+ int64_t testTimeUs = frames[i].timeUs + jitter;
+ printf("time %lld, testTime %lld, jitter %d\n", frames[i].timeUs, testTimeUs, jitter);
+ EXPECT_EQ(frames[i].shouldDrop, mFrameDropper->shouldDrop(testTimeUs));
+ }
+ }
+
+ sp<FrameDropper> mFrameDropper;
+};
+
+TEST_F(FrameDropperTest, TestInvalidMaxFrameRate) {
+ EXPECT_NE(OK, mFrameDropper->setMaxFrameRate(-1.0));
+ EXPECT_NE(OK, mFrameDropper->setMaxFrameRate(0));
+}
+
+TEST_F(FrameDropperTest, Test20Fps) {
+ RunTest(testFrames20Fps, ARRAY_SIZE(testFrames20Fps));
+}
+
+TEST_F(FrameDropperTest, Test30Fps) {
+ RunTest(testFrames30Fps, ARRAY_SIZE(testFrames30Fps));
+}
+
+TEST_F(FrameDropperTest, Test40Fps) {
+ RunTest(testFrames40Fps, ARRAY_SIZE(testFrames40Fps));
+}
+
+TEST_F(FrameDropperTest, Test60Fps) {
+ RunTest(testFrames60Fps, ARRAY_SIZE(testFrames60Fps));
+}
+
+TEST_F(FrameDropperTest, TestVariableFps) {
+ RunTest(testFramesVariableFps, ARRAY_SIZE(testFramesVariableFps));
+}
+
+} // namespace android
diff --git a/media/libstagefright/webm/WebmWriter.cpp b/media/libstagefright/webm/WebmWriter.cpp
index 069961b..737f144 100644
--- a/media/libstagefright/webm/WebmWriter.cpp
+++ b/media/libstagefright/webm/WebmWriter.cpp
@@ -80,38 +80,6 @@
mCuePoints);
}
-WebmWriter::WebmWriter(const char *filename)
- : mInitCheck(NO_INIT),
- mTimeCodeScale(1000000),
- mStartTimestampUs(0),
- mStartTimeOffsetMs(0),
- mSegmentOffset(0),
- mSegmentDataStart(0),
- mInfoOffset(0),
- mInfoSize(0),
- mTracksOffset(0),
- mCuesOffset(0),
- mPaused(false),
- mStarted(false),
- mIsFileSizeLimitExplicitlyRequested(false),
- mIsRealTimeRecording(false),
- mStreamableFile(true),
- mEstimatedCuesSize(0) {
- mFd = open(filename, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
- if (mFd >= 0) {
- ALOGV("fd %d; flags: %o", mFd, fcntl(mFd, F_GETFL, 0));
- mInitCheck = OK;
- }
- mStreams[kAudioIndex] = WebmStream(kAudioType, "Audio", &WebmWriter::audioTrack);
- mStreams[kVideoIndex] = WebmStream(kVideoType, "Video", &WebmWriter::videoTrack);
- mSinkThread = new WebmFrameSinkThread(
- mFd,
- mSegmentDataStart,
- mStreams[kVideoIndex].mSink,
- mStreams[kAudioIndex].mSink,
- mCuePoints);
-}
-
// static
sp<WebmElement> WebmWriter::videoTrack(const sp<MetaData>& md) {
int32_t width, height;
diff --git a/media/libstagefright/webm/WebmWriter.h b/media/libstagefright/webm/WebmWriter.h
index 36b6965..4ad770e 100644
--- a/media/libstagefright/webm/WebmWriter.h
+++ b/media/libstagefright/webm/WebmWriter.h
@@ -37,7 +37,6 @@
class WebmWriter : public MediaWriter {
public:
WebmWriter(int fd);
- WebmWriter(const char *filename);
~WebmWriter() { reset(); }
diff --git a/media/mediaserver/Android.mk b/media/mediaserver/Android.mk
index 3a280f0..f1b84ad 100644
--- a/media/mediaserver/Android.mk
+++ b/media/mediaserver/Android.mk
@@ -11,7 +11,7 @@
include $(CLEAR_VARS)
LOCAL_SRC_FILES:= \
- main_mediaserver.cpp
+ main_mediaserver.cpp
LOCAL_SHARED_LIBRARIES := \
libaudioflinger \
diff --git a/media/mediaserver/main_mediaserver.cpp b/media/mediaserver/main_mediaserver.cpp
index af1c9e6..263dd32 100644
--- a/media/mediaserver/main_mediaserver.cpp
+++ b/media/mediaserver/main_mediaserver.cpp
@@ -33,7 +33,7 @@
#include "CameraService.h"
#include "MediaLogService.h"
#include "MediaPlayerService.h"
-#include "AudioPolicyService.h"
+#include "service/AudioPolicyService.h"
#include "SoundTriggerHwService.h"
using namespace android;
diff --git a/media/ndk/NdkMediaCodec.cpp b/media/ndk/NdkMediaCodec.cpp
index ed00b72..3124e4a 100644
--- a/media/ndk/NdkMediaCodec.cpp
+++ b/media/ndk/NdkMediaCodec.cpp
@@ -352,7 +352,8 @@
}
//EXPORT
-media_status_t AMediaCodec_setNotificationCallback(AMediaCodec *mData, OnCodecEvent callback, void *userdata) {
+media_status_t AMediaCodec_setNotificationCallback(AMediaCodec *mData, OnCodecEvent callback,
+ void *userdata) {
mData->mCallback = callback;
mData->mCallbackUserData = userdata;
return AMEDIA_OK;
diff --git a/media/ndk/NdkMediaExtractor.cpp b/media/ndk/NdkMediaExtractor.cpp
index db57d0b..0ecd64f 100644
--- a/media/ndk/NdkMediaExtractor.cpp
+++ b/media/ndk/NdkMediaExtractor.cpp
@@ -70,7 +70,8 @@
}
EXPORT
-media_status_t AMediaExtractor_setDataSourceFd(AMediaExtractor *mData, int fd, off64_t offset, off64_t length) {
+media_status_t AMediaExtractor_setDataSourceFd(AMediaExtractor *mData, int fd, off64_t offset,
+ off64_t length) {
ALOGV("setDataSource(%d, %lld, %lld)", fd, offset, length);
return translate_error(mData->mImpl->setDataSource(fd, offset, length));
}
diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk
index 44d2553..f029333 100644
--- a/services/audioflinger/Android.mk
+++ b/services/audioflinger/Android.mk
@@ -74,9 +74,15 @@
LOCAL_MODULE:= libaudioflinger
LOCAL_32_BIT_ONLY := true
-LOCAL_SRC_FILES += FastMixer.cpp FastMixerState.cpp AudioWatchdog.cpp
-LOCAL_SRC_FILES += FastThread.cpp FastThreadState.cpp
-LOCAL_SRC_FILES += FastCapture.cpp FastCaptureState.cpp
+LOCAL_SRC_FILES += \
+ AudioWatchdog.cpp \
+ FastCapture.cpp \
+ FastCaptureState.cpp \
+ FastMixer.cpp \
+ FastMixerState.cpp \
+ FastThread.cpp \
+ FastThreadDumpState.cpp \
+ FastThreadState.cpp
LOCAL_CFLAGS += -DSTATE_QUEUE_INSTANTIATIONS='"StateQueueInstantiations.cpp"'
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 993db73..9ad437a 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -185,7 +185,8 @@
char value[PROPERTY_VALUE_MAX];
bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1);
if (doLog) {
- mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", MemoryHeapBase::READ_ONLY);
+ mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters",
+ MemoryHeapBase::READ_ONLY);
}
#ifdef TEE_SINK
@@ -401,6 +402,9 @@
String8 result(kClientLockedString);
write(fd, result.string(), result.size());
}
+
+ EffectDumpEffects(fd);
+
dumpClients(fd, args);
if (clientLocked) {
mClientLock.unlock();
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index fd28ea1..0d4b358 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -430,6 +430,10 @@
mState.mLog = log;
}
+static inline audio_format_t selectMixerInFormat(audio_format_t inputFormat __unused) {
+ return kUseFloat && kUseNewMixer ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
+}
+
int AudioMixer::getTrackName(audio_channel_mask_t channelMask,
audio_format_t format, int sessionId)
{
@@ -492,24 +496,23 @@
t->mInputBufferProvider = NULL;
t->mReformatBufferProvider = NULL;
t->downmixerBufferProvider = NULL;
+ t->mPostDownmixReformatBufferProvider = NULL;
t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
t->mFormat = format;
- t->mMixerInFormat = kUseFloat && kUseNewMixer
- ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
+ t->mMixerInFormat = selectMixerInFormat(format);
+ t->mDownmixRequiresFormat = AUDIO_FORMAT_INVALID; // no format required
t->mMixerChannelMask = audio_channel_mask_from_representation_and_bits(
AUDIO_CHANNEL_REPRESENTATION_POSITION, AUDIO_CHANNEL_OUT_STEREO);
t->mMixerChannelCount = audio_channel_count_from_out_mask(t->mMixerChannelMask);
// Check the downmixing (or upmixing) requirements.
- status_t status = initTrackDownmix(t, n);
+ status_t status = t->prepareForDownmix();
if (status != OK) {
ALOGE("AudioMixer::getTrackName invalid channelMask (%#x)", channelMask);
return -1;
}
- // initTrackDownmix() may change the input format requirement.
- // If you desire floating point input to the mixer, it may change
- // to integer because the downmixer requires integer to process.
+ // prepareForDownmix() may change mDownmixRequiresFormat
ALOGVV("mMixerFormat:%#x mMixerInFormat:%#x\n", t->mMixerFormat, t->mMixerInFormat);
- prepareTrackForReformat(t, n);
+ t->prepareForReformat();
mTrackNames |= 1 << n;
return TRACK0 + n;
}
@@ -526,7 +529,7 @@
}
// Called when channel masks have changed for a track name
-// TODO: Fix Downmixbufferprofider not to (possibly) change mixer input format,
+// TODO: Fix DownmixerBufferProvider not to (possibly) change mixer input format,
// which will simplify this logic.
bool AudioMixer::setChannelMasks(int name,
audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask) {
@@ -551,21 +554,18 @@
// channel masks have changed, does this track need a downmixer?
// update to try using our desired format (if we aren't already using it)
- const audio_format_t prevMixerInFormat = track.mMixerInFormat;
- track.mMixerInFormat = kUseFloat && kUseNewMixer
- ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
- const status_t status = initTrackDownmix(&mState.tracks[name], name);
+ const audio_format_t prevDownmixerFormat = track.mDownmixRequiresFormat;
+ const status_t status = mState.tracks[name].prepareForDownmix();
ALOGE_IF(status != OK,
- "initTrackDownmix error %d, track channel mask %#x, mixer channel mask %#x",
+ "prepareForDownmix error %d, track channel mask %#x, mixer channel mask %#x",
status, track.channelMask, track.mMixerChannelMask);
- const bool mixerInFormatChanged = prevMixerInFormat != track.mMixerInFormat;
- if (mixerInFormatChanged) {
- prepareTrackForReformat(&track, name); // because of downmixer, track format may change!
+ if (prevDownmixerFormat != track.mDownmixRequiresFormat) {
+ track.prepareForReformat(); // because of downmixer, track format may change!
}
- if (track.resampler && (mixerInFormatChanged || mixerChannelCountChanged)) {
- // resampler input format or channels may have changed.
+ if (track.resampler && mixerChannelCountChanged) {
+ // resampler channels may have changed.
const uint32_t resetToSampleRate = track.sampleRate;
delete track.resampler;
track.resampler = NULL;
@@ -576,99 +576,122 @@
return true;
}
-status_t AudioMixer::initTrackDownmix(track_t* pTrack, int trackName)
-{
- // Only remix (upmix or downmix) if the track and mixer/device channel masks
- // are not the same and not handled internally, as mono -> stereo currently is.
- if (pTrack->channelMask != pTrack->mMixerChannelMask
- && !(pTrack->channelMask == AUDIO_CHANNEL_OUT_MONO
- && pTrack->mMixerChannelMask == AUDIO_CHANNEL_OUT_STEREO)) {
- return prepareTrackForDownmix(pTrack, trackName);
- }
- // no remix necessary
- unprepareTrackForDownmix(pTrack, trackName);
- return NO_ERROR;
-}
+void AudioMixer::track_t::unprepareForDownmix() {
+ ALOGV("AudioMixer::unprepareForDownmix(%p)", this);
-void AudioMixer::unprepareTrackForDownmix(track_t* pTrack, int trackName __unused) {
- ALOGV("AudioMixer::unprepareTrackForDownmix(%d)", trackName);
-
- if (pTrack->downmixerBufferProvider != NULL) {
+ mDownmixRequiresFormat = AUDIO_FORMAT_INVALID;
+ if (downmixerBufferProvider != NULL) {
// this track had previously been configured with a downmixer, delete it
ALOGV(" deleting old downmixer");
- delete pTrack->downmixerBufferProvider;
- pTrack->downmixerBufferProvider = NULL;
- reconfigureBufferProviders(pTrack);
+ delete downmixerBufferProvider;
+ downmixerBufferProvider = NULL;
+ reconfigureBufferProviders();
} else {
ALOGV(" nothing to do, no downmixer to delete");
}
}
-status_t AudioMixer::prepareTrackForDownmix(track_t* pTrack, int trackName)
+status_t AudioMixer::track_t::prepareForDownmix()
{
- ALOGV("AudioMixer::prepareTrackForDownmix(%d) with mask 0x%x", trackName, pTrack->channelMask);
+ ALOGV("AudioMixer::prepareForDownmix(%p) with mask 0x%x",
+ this, channelMask);
// discard the previous downmixer if there was one
- unprepareTrackForDownmix(pTrack, trackName);
+ unprepareForDownmix();
+ // Only remix (upmix or downmix) if the track and mixer/device channel masks
+ // are not the same and not handled internally, as mono -> stereo currently is.
+ if (channelMask == mMixerChannelMask
+ || (channelMask == AUDIO_CHANNEL_OUT_MONO
+ && mMixerChannelMask == AUDIO_CHANNEL_OUT_STEREO)) {
+ return NO_ERROR;
+ }
if (DownmixerBufferProvider::isMultichannelCapable()) {
- DownmixerBufferProvider* pDbp = new DownmixerBufferProvider(pTrack->channelMask,
- pTrack->mMixerChannelMask,
- AUDIO_FORMAT_PCM_16_BIT /* TODO: use pTrack->mMixerInFormat, now only PCM 16 */,
- pTrack->sampleRate, pTrack->sessionId, kCopyBufferFrameCount);
+ DownmixerBufferProvider* pDbp = new DownmixerBufferProvider(channelMask,
+ mMixerChannelMask,
+ AUDIO_FORMAT_PCM_16_BIT /* TODO: use mMixerInFormat, now only PCM 16 */,
+ sampleRate, sessionId, kCopyBufferFrameCount);
if (pDbp->isValid()) { // if constructor completed properly
- pTrack->mMixerInFormat = AUDIO_FORMAT_PCM_16_BIT; // PCM 16 bit required for downmix
- pTrack->downmixerBufferProvider = pDbp;
- reconfigureBufferProviders(pTrack);
+ mDownmixRequiresFormat = AUDIO_FORMAT_PCM_16_BIT; // PCM 16 bit required for downmix
+ downmixerBufferProvider = pDbp;
+ reconfigureBufferProviders();
return NO_ERROR;
}
delete pDbp;
}
// Effect downmixer does not accept the channel conversion. Let's use our remixer.
- RemixBufferProvider* pRbp = new RemixBufferProvider(pTrack->channelMask,
- pTrack->mMixerChannelMask, pTrack->mMixerInFormat, kCopyBufferFrameCount);
+ RemixBufferProvider* pRbp = new RemixBufferProvider(channelMask,
+ mMixerChannelMask, mMixerInFormat, kCopyBufferFrameCount);
// Remix always finds a conversion whereas Downmixer effect above may fail.
- pTrack->downmixerBufferProvider = pRbp;
- reconfigureBufferProviders(pTrack);
+ downmixerBufferProvider = pRbp;
+ reconfigureBufferProviders();
return NO_ERROR;
}
-void AudioMixer::unprepareTrackForReformat(track_t* pTrack, int trackName __unused) {
- ALOGV("AudioMixer::unprepareTrackForReformat(%d)", trackName);
- if (pTrack->mReformatBufferProvider != NULL) {
- delete pTrack->mReformatBufferProvider;
- pTrack->mReformatBufferProvider = NULL;
- reconfigureBufferProviders(pTrack);
+void AudioMixer::track_t::unprepareForReformat() {
+ ALOGV("AudioMixer::unprepareForReformat(%p)", this);
+ bool requiresReconfigure = false;
+ if (mReformatBufferProvider != NULL) {
+ delete mReformatBufferProvider;
+ mReformatBufferProvider = NULL;
+ requiresReconfigure = true;
+ }
+ if (mPostDownmixReformatBufferProvider != NULL) {
+ delete mPostDownmixReformatBufferProvider;
+ mPostDownmixReformatBufferProvider = NULL;
+ requiresReconfigure = true;
+ }
+ if (requiresReconfigure) {
+ reconfigureBufferProviders();
}
}
-status_t AudioMixer::prepareTrackForReformat(track_t* pTrack, int trackName)
+status_t AudioMixer::track_t::prepareForReformat()
{
- ALOGV("AudioMixer::prepareTrackForReformat(%d) with format %#x", trackName, pTrack->mFormat);
- // discard the previous reformatter if there was one
- unprepareTrackForReformat(pTrack, trackName);
- // only configure reformatter if needed
- if (pTrack->mFormat != pTrack->mMixerInFormat) {
- pTrack->mReformatBufferProvider = new ReformatBufferProvider(
- audio_channel_count_from_out_mask(pTrack->channelMask),
- pTrack->mFormat, pTrack->mMixerInFormat,
+ ALOGV("AudioMixer::prepareForReformat(%p) with format %#x", this, mFormat);
+ // discard previous reformatters
+ unprepareForReformat();
+ // only configure reformatters as needed
+ const audio_format_t targetFormat = mDownmixRequiresFormat != AUDIO_FORMAT_INVALID
+ ? mDownmixRequiresFormat : mMixerInFormat;
+ bool requiresReconfigure = false;
+ if (mFormat != targetFormat) {
+ mReformatBufferProvider = new ReformatBufferProvider(
+ audio_channel_count_from_out_mask(channelMask),
+ mFormat,
+ targetFormat,
kCopyBufferFrameCount);
- reconfigureBufferProviders(pTrack);
+ requiresReconfigure = true;
+ }
+ if (targetFormat != mMixerInFormat) {
+ mPostDownmixReformatBufferProvider = new ReformatBufferProvider(
+ audio_channel_count_from_out_mask(mMixerChannelMask),
+ targetFormat,
+ mMixerInFormat,
+ kCopyBufferFrameCount);
+ requiresReconfigure = true;
+ }
+ if (requiresReconfigure) {
+ reconfigureBufferProviders();
}
return NO_ERROR;
}
-void AudioMixer::reconfigureBufferProviders(track_t* pTrack)
+void AudioMixer::track_t::reconfigureBufferProviders()
{
- pTrack->bufferProvider = pTrack->mInputBufferProvider;
- if (pTrack->mReformatBufferProvider) {
- pTrack->mReformatBufferProvider->setBufferProvider(pTrack->bufferProvider);
- pTrack->bufferProvider = pTrack->mReformatBufferProvider;
+ bufferProvider = mInputBufferProvider;
+ if (mReformatBufferProvider) {
+ mReformatBufferProvider->setBufferProvider(bufferProvider);
+ bufferProvider = mReformatBufferProvider;
}
- if (pTrack->downmixerBufferProvider) {
- pTrack->downmixerBufferProvider->setBufferProvider(pTrack->bufferProvider);
- pTrack->bufferProvider = pTrack->downmixerBufferProvider;
+ if (downmixerBufferProvider) {
+ downmixerBufferProvider->setBufferProvider(bufferProvider);
+ bufferProvider = downmixerBufferProvider;
+ }
+ if (mPostDownmixReformatBufferProvider) {
+ mPostDownmixReformatBufferProvider->setBufferProvider(bufferProvider);
+ bufferProvider = mPostDownmixReformatBufferProvider;
}
}
@@ -687,9 +710,9 @@
delete track.resampler;
track.resampler = NULL;
// delete the downmixer
- unprepareTrackForDownmix(&mState.tracks[name], name);
+ mState.tracks[name].unprepareForDownmix();
// delete the reformatter
- unprepareTrackForReformat(&mState.tracks[name], name);
+ mState.tracks[name].unprepareForReformat();
mTrackNames &= ~(1<<name);
}
@@ -828,7 +851,7 @@
ALOG_ASSERT(audio_is_linear_pcm(format), "Invalid format %#x", format);
track.mFormat = format;
ALOGV("setParameter(TRACK, FORMAT, %#x)", format);
- prepareTrackForReformat(&track, name);
+ track.prepareForReformat();
invalidateState(1 << name);
}
} break;
@@ -1032,10 +1055,13 @@
if (mState.tracks[name].mReformatBufferProvider != NULL) {
mState.tracks[name].mReformatBufferProvider->reset();
} else if (mState.tracks[name].downmixerBufferProvider != NULL) {
+ mState.tracks[name].downmixerBufferProvider->reset();
+ } else if (mState.tracks[name].mPostDownmixReformatBufferProvider != NULL) {
+ mState.tracks[name].mPostDownmixReformatBufferProvider->reset();
}
mState.tracks[name].mInputBufferProvider = bufferProvider;
- reconfigureBufferProviders(&mState.tracks[name]);
+ mState.tracks[name].reconfigureBufferProviders();
}
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index f4f142b..c5df08a 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -127,10 +127,16 @@
size_t getUnreleasedFrames(int name) const;
static inline bool isValidPcmTrackFormat(audio_format_t format) {
- return format == AUDIO_FORMAT_PCM_16_BIT ||
- format == AUDIO_FORMAT_PCM_24_BIT_PACKED ||
- format == AUDIO_FORMAT_PCM_32_BIT ||
- format == AUDIO_FORMAT_PCM_FLOAT;
+ switch (format) {
+ case AUDIO_FORMAT_PCM_8_BIT:
+ case AUDIO_FORMAT_PCM_16_BIT:
+ case AUDIO_FORMAT_PCM_24_BIT_PACKED:
+ case AUDIO_FORMAT_PCM_32_BIT:
+ case AUDIO_FORMAT_PCM_FLOAT:
+ return true;
+ default:
+ return false;
+ }
}
private:
@@ -205,17 +211,34 @@
int32_t* auxBuffer;
// 16-byte boundary
+
+ /* Buffer providers are constructed to translate the track input data as needed.
+ *
+ * 1) mInputBufferProvider: The AudioTrack buffer provider.
+ * 2) mReformatBufferProvider: If not NULL, performs the audio reformat to
+ * match either mMixerInFormat or mDownmixRequiresFormat, if the downmixer
+ * requires reformat. For example, it may convert floating point input to
+ * PCM_16_bit if that's required by the downmixer.
+ * 3) downmixerBufferProvider: If not NULL, performs the channel remixing to match
+ * the number of channels required by the mixer sink.
+ * 4) mPostDownmixReformatBufferProvider: If not NULL, performs reformatting from
+ * the downmixer requirements to the mixer engine input requirements.
+ */
AudioBufferProvider* mInputBufferProvider; // externally provided buffer provider.
CopyBufferProvider* mReformatBufferProvider; // provider wrapper for reformatting.
CopyBufferProvider* downmixerBufferProvider; // wrapper for channel conversion.
-
- int32_t sessionId;
+ CopyBufferProvider* mPostDownmixReformatBufferProvider;
// 16-byte boundary
+ int32_t sessionId;
+
audio_format_t mMixerFormat; // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
audio_format_t mFormat; // input track format
audio_format_t mMixerInFormat; // mix internal format AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
// each track must be converted to this format.
+ audio_format_t mDownmixRequiresFormat; // required downmixer format
+ // AUDIO_FORMAT_PCM_16_BIT if 16 bit necessary
+ // AUDIO_FORMAT_INVALID if no required format
float mVolume[MAX_NUM_VOLUMES]; // floating point set volume
float mPrevVolume[MAX_NUM_VOLUMES]; // floating point previous volume
@@ -225,7 +248,6 @@
float mPrevAuxLevel; // floating point prev aux level
float mAuxInc; // floating point aux increment
- // 16-byte boundary
audio_channel_mask_t mMixerChannelMask;
uint32_t mMixerChannelCount;
@@ -236,6 +258,12 @@
void adjustVolumeRamp(bool aux, bool useFloat = false);
size_t getUnreleasedFrames() const { return resampler != NULL ?
resampler->getUnreleasedFrames() : 0; };
+
+ status_t prepareForDownmix();
+ void unprepareForDownmix();
+ status_t prepareForReformat();
+ void unprepareForReformat();
+ void reconfigureBufferProviders();
};
typedef void (*process_hook_t)(state_t* state, int64_t pts);
@@ -382,14 +410,6 @@
bool setChannelMasks(int name,
audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask);
- // TODO: remove unused trackName/trackNum from functions below.
- static status_t initTrackDownmix(track_t* pTrack, int trackName);
- static status_t prepareTrackForDownmix(track_t* pTrack, int trackNum);
- static void unprepareTrackForDownmix(track_t* pTrack, int trackName);
- static status_t prepareTrackForReformat(track_t* pTrack, int trackNum);
- static void unprepareTrackForReformat(track_t* pTrack, int trackName);
- static void reconfigureBufferProviders(track_t* pTrack);
-
static void track__genericResample(track_t* t, int32_t* out, size_t numFrames, int32_t* temp,
int32_t* aux);
static void track__nop(track_t* t, int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
diff --git a/services/audioflinger/AudioResamplerFirGen.h b/services/audioflinger/AudioResamplerFirGen.h
index f3718b6..a9c84de 100644
--- a/services/audioflinger/AudioResamplerFirGen.h
+++ b/services/audioflinger/AudioResamplerFirGen.h
@@ -204,7 +204,8 @@
template <>
struct I0ATerm<0> { // 1/sqrt(2*PI);
- static const CONSTEXPR double value = 0.398942280401432677939946059934381868475858631164934657665925;
+ static const CONSTEXPR double value =
+ 0.398942280401432677939946059934381868475858631164934657665925;
};
#if USE_HORNERS_METHOD
diff --git a/services/audioflinger/AudioResamplerFirProcess.h b/services/audioflinger/AudioResamplerFirProcess.h
index efc8055..1118bf8 100644
--- a/services/audioflinger/AudioResamplerFirProcess.h
+++ b/services/audioflinger/AudioResamplerFirProcess.h
@@ -174,7 +174,8 @@
* Process() calls ProcessBase() with TFUNC = InterpCompute, for interpolated phase.
*/
-template <int CHANNELS, int STRIDE, typename TFUNC, typename TC, typename TI, typename TO, typename TINTERP>
+template <int CHANNELS, int STRIDE, typename TFUNC, typename TC, typename TI, typename TO,
+ typename TINTERP>
static inline
void ProcessBase(TO* const out,
size_t count,
@@ -268,7 +269,8 @@
TINTERP lerpP,
const TO* const volumeLR)
{
- ProcessBase<CHANNELS, STRIDE, InterpCompute>(out, count, coefsP, coefsN, sP, sN, lerpP, volumeLR);
+ ProcessBase<CHANNELS, STRIDE, InterpCompute>(out, count, coefsP, coefsN, sP, sN, lerpP,
+ volumeLR);
}
/*
diff --git a/services/audioflinger/FastCapture.h b/services/audioflinger/FastCapture.h
index e535b9d..e61e026 100644
--- a/services/audioflinger/FastCapture.h
+++ b/services/audioflinger/FastCapture.h
@@ -20,6 +20,7 @@
#include "FastThread.h"
#include "StateQueue.h"
#include "FastCaptureState.h"
+#include "FastThreadDumpState.h"
namespace android {
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index 2678cbf..141a79e 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -629,7 +629,8 @@
left.sample(tail[i]);
right.sample(tail[n - (i + 1)]);
}
- dprintf(fd, " Distribution of mix cycle times in ms for the tails (> ~3 stddev outliers):\n"
+ dprintf(fd, " Distribution of mix cycle times in ms for the tails "
+ "(> ~3 stddev outliers):\n"
" left tail: mean=%.2f min=%.2f max=%.2f stddev=%.2f\n"
" right tail: mean=%.2f min=%.2f max=%.2f stddev=%.2f\n",
left.mean()*1e-6, left.minimum()*1e-6, left.maximum()*1e-6, left.stddev()*1e-6,
diff --git a/services/audioflinger/FastMixerDumpState.h b/services/audioflinger/FastMixerDumpState.h
index 6a1e4649..1ae191c 100644
--- a/services/audioflinger/FastMixerDumpState.h
+++ b/services/audioflinger/FastMixerDumpState.h
@@ -18,6 +18,7 @@
#define ANDROID_AUDIO_FAST_MIXER_DUMP_STATE_H
#include "Configuration.h"
+#include "FastThreadDumpState.h"
namespace android {
diff --git a/services/audioflinger/FastThread.cpp b/services/audioflinger/FastThread.cpp
index 216dace..3e12cca 100644
--- a/services/audioflinger/FastThread.cpp
+++ b/services/audioflinger/FastThread.cpp
@@ -25,6 +25,7 @@
#include <utils/Log.h>
#include <utils/Trace.h>
#include "FastThread.h"
+#include "FastThreadDumpState.h"
#define FAST_DEFAULT_NS 999999999L // ~1 sec: default time to sleep
#define FAST_HOT_IDLE_NS 1000000L // 1 ms: time to sleep while hot idling
diff --git a/services/audiopolicy/AudioPolicyFactory.cpp b/services/audioflinger/FastThreadDumpState.cpp
similarity index 61%
copy from services/audiopolicy/AudioPolicyFactory.cpp
copy to services/audioflinger/FastThreadDumpState.cpp
index 2ae7bc1..d7b825d 100644
--- a/services/audiopolicy/AudioPolicyFactory.cpp
+++ b/services/audioflinger/FastThreadDumpState.cpp
@@ -14,19 +14,24 @@
* limitations under the License.
*/
-#include "AudioPolicyManager.h"
+#include "FastThreadDumpState.h"
namespace android {
-extern "C" AudioPolicyInterface* createAudioPolicyManager(
- AudioPolicyClientInterface *clientInterface)
+FastThreadDumpState::FastThreadDumpState() :
+ mCommand(FastThreadState::INITIAL), mUnderruns(0), mOverruns(0),
+ /* mMeasuredWarmupTs({0, 0}), */
+ mWarmupCycles(0)
+#ifdef FAST_MIXER_STATISTICS
+ , mSamplingN(1), mBounds(0)
+#endif
{
- return new AudioPolicyManager(clientInterface);
+ mMeasuredWarmupTs.tv_sec = 0;
+ mMeasuredWarmupTs.tv_nsec = 0;
}
-extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface)
+FastThreadDumpState::~FastThreadDumpState()
{
- delete interface;
}
-}; // namespace android
+} // android
diff --git a/services/audioflinger/FastThreadDumpState.h b/services/audioflinger/FastThreadDumpState.h
new file mode 100644
index 0000000..17afbe5
--- /dev/null
+++ b/services/audioflinger/FastThreadDumpState.h
@@ -0,0 +1,61 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_AUDIO_FAST_THREAD_DUMP_STATE_H
+#define ANDROID_AUDIO_FAST_THREAD_DUMP_STATE_H
+
+#include "Configuration.h"
+#include "FastThreadState.h"
+
+namespace android {
+
+// FIXME extract common part of comment at FastMixerDumpState
+struct FastThreadDumpState {
+ FastThreadDumpState();
+ /*virtual*/ ~FastThreadDumpState();
+
+ FastThreadState::Command mCommand; // current command
+ uint32_t mUnderruns; // total number of underruns
+ uint32_t mOverruns; // total number of overruns
+ struct timespec mMeasuredWarmupTs; // measured warmup time
+ uint32_t mWarmupCycles; // number of loop cycles required to warmup
+
+#ifdef FAST_MIXER_STATISTICS
+ // Recently collected samples of per-cycle monotonic time, thread CPU time, and CPU frequency.
+ // kSamplingN is max size of sampling frame (statistics), and must be a power of 2 <= 0x8000.
+ // The sample arrays are virtually allocated based on this compile-time constant,
+ // but are only initialized and used based on the runtime parameter mSamplingN.
+ static const uint32_t kSamplingN = 0x8000;
+ // Corresponding runtime maximum size of sample arrays, must be a power of 2 <= kSamplingN.
+ uint32_t mSamplingN;
+ // The bounds define the interval of valid samples, and are represented as follows:
+ // newest open (excluded) endpoint = lower 16 bits of bounds, modulo N
+ // oldest closed (included) endpoint = upper 16 bits of bounds, modulo N
+ // Number of valid samples is newest - oldest.
+ uint32_t mBounds; // bounds for mMonotonicNs, mThreadCpuNs, and mCpukHz
+ // The elements in the *Ns arrays are in units of nanoseconds <= 3999999999.
+ uint32_t mMonotonicNs[kSamplingN]; // delta monotonic (wall clock) time
+ uint32_t mLoadNs[kSamplingN]; // delta CPU load in time
+#ifdef CPU_FREQUENCY_STATISTICS
+ uint32_t mCpukHz[kSamplingN]; // absolute CPU clock frequency in kHz, bits 0-3 are CPU#
+#endif
+#endif
+
+}; // struct FastThreadDumpState
+
+} // android
+
+#endif // ANDROID_AUDIO_FAST_THREAD_DUMP_STATE_H
diff --git a/services/audioflinger/FastThreadState.cpp b/services/audioflinger/FastThreadState.cpp
index 6994872..e6cf85c 100644
--- a/services/audioflinger/FastThreadState.cpp
+++ b/services/audioflinger/FastThreadState.cpp
@@ -29,21 +29,4 @@
{
}
-
-FastThreadDumpState::FastThreadDumpState() :
- mCommand(FastThreadState::INITIAL), mUnderruns(0), mOverruns(0),
- /* mMeasuredWarmupTs({0, 0}), */
- mWarmupCycles(0)
-#ifdef FAST_MIXER_STATISTICS
- , mSamplingN(1), mBounds(0)
-#endif
-{
- mMeasuredWarmupTs.tv_sec = 0;
- mMeasuredWarmupTs.tv_nsec = 0;
-}
-
-FastThreadDumpState::~FastThreadDumpState()
-{
-}
-
} // namespace android
diff --git a/services/audioflinger/FastThreadState.h b/services/audioflinger/FastThreadState.h
index 1ab8a0a..011921d 100644
--- a/services/audioflinger/FastThreadState.h
+++ b/services/audioflinger/FastThreadState.h
@@ -48,41 +48,6 @@
}; // struct FastThreadState
-
-// FIXME extract common part of comment at FastMixerDumpState
-struct FastThreadDumpState {
- FastThreadDumpState();
- /*virtual*/ ~FastThreadDumpState();
-
- FastThreadState::Command mCommand; // current command
- uint32_t mUnderruns; // total number of underruns
- uint32_t mOverruns; // total number of overruns
- struct timespec mMeasuredWarmupTs; // measured warmup time
- uint32_t mWarmupCycles; // number of loop cycles required to warmup
-
-#ifdef FAST_MIXER_STATISTICS
- // Recently collected samples of per-cycle monotonic time, thread CPU time, and CPU frequency.
- // kSamplingN is max size of sampling frame (statistics), and must be a power of 2 <= 0x8000.
- // The sample arrays are virtually allocated based on this compile-time constant,
- // but are only initialized and used based on the runtime parameter mSamplingN.
- static const uint32_t kSamplingN = 0x8000;
- // Corresponding runtime maximum size of sample arrays, must be a power of 2 <= kSamplingN.
- uint32_t mSamplingN;
- // The bounds define the interval of valid samples, and are represented as follows:
- // newest open (excluded) endpoint = lower 16 bits of bounds, modulo N
- // oldest closed (included) endpoint = upper 16 bits of bounds, modulo N
- // Number of valid samples is newest - oldest.
- uint32_t mBounds; // bounds for mMonotonicNs, mThreadCpuNs, and mCpukHz
- // The elements in the *Ns arrays are in units of nanoseconds <= 3999999999.
- uint32_t mMonotonicNs[kSamplingN]; // delta monotonic (wall clock) time
- uint32_t mLoadNs[kSamplingN]; // delta CPU load in time
-#ifdef CPU_FREQUENCY_STATISTICS
- uint32_t mCpukHz[kSamplingN]; // absolute CPU clock frequency in kHz, bits 0-3 are CPU#
-#endif
-#endif
-
-}; // struct FastThreadDumpState
-
} // android
#endif // ANDROID_AUDIO_FAST_THREAD_STATE_H
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index ee48276..902d5e4 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -255,7 +255,7 @@
class Buffer : public AudioBufferProvider::Buffer {
public:
- int16_t *mBuffer;
+ void *mBuffer;
};
OutputTrack(PlaybackThread *thread,
@@ -271,7 +271,7 @@
AudioSystem::SYNC_EVENT_NONE,
int triggerSession = 0);
virtual void stop();
- bool write(int16_t* data, uint32_t frames);
+ bool write(void* data, uint32_t frames);
bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; }
bool isActive() const { return mActive; }
const wp<ThreadBase>& thread() const { return mThread; }
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 51025fe..15dd408 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -172,6 +172,18 @@
// and that all "fast" AudioRecord clients read from. In either case, the size can be small.
static const size_t kRecordThreadReadOnlyHeapSize = 0x2000;
+// Returns the source frames needed to resample to destination frames. This is not a precise
+// value and depends on the resampler (and possibly how it handles rounding internally).
+// If srcSampleRate and dstSampleRate are equal, then it returns destination frames, which
+// may not be a true if the resampler is asynchronous.
+static inline size_t sourceFramesNeeded(
+ uint32_t srcSampleRate, size_t dstFramesRequired, uint32_t dstSampleRate) {
+ // +1 for rounding - always do this even if matched ratio
+ // +1 for additional sample needed for interpolation
+ return srcSampleRate == dstSampleRate ? dstFramesRequired :
+ size_t((uint64_t)dstFramesRequired * srcSampleRate / dstSampleRate + 1 + 1);
+}
+
// ----------------------------------------------------------------------------
static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT;
@@ -314,6 +326,64 @@
// ThreadBase
// ----------------------------------------------------------------------------
+// static
+const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type)
+{
+ switch (type) {
+ case MIXER:
+ return "MIXER";
+ case DIRECT:
+ return "DIRECT";
+ case DUPLICATING:
+ return "DUPLICATING";
+ case RECORD:
+ return "RECORD";
+ case OFFLOAD:
+ return "OFFLOAD";
+ default:
+ return "unknown";
+ }
+}
+
+static String8 outputFlagsToString(audio_output_flags_t flags)
+{
+ static const struct mapping {
+ audio_output_flags_t mFlag;
+ const char * mString;
+ } mappings[] = {
+ AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT",
+ AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY",
+ AUDIO_OUTPUT_FLAG_FAST, "FAST",
+ AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER",
+ AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD, "COMPRESS_OFFLOAAD",
+ AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING",
+ AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC",
+ AUDIO_OUTPUT_FLAG_NONE, "NONE", // must be last
+ };
+ String8 result;
+ audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE;
+ const mapping *entry;
+ for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) {
+ allFlags = (audio_output_flags_t) (allFlags | entry->mFlag);
+ if (flags & entry->mFlag) {
+ if (!result.isEmpty()) {
+ result.append("|");
+ }
+ result.append(entry->mString);
+ }
+ }
+ if (flags & ~allFlags) {
+ if (!result.isEmpty()) {
+ result.append("|");
+ }
+ result.appendFormat("0x%X", flags & ~allFlags);
+ }
+ if (result.isEmpty()) {
+ result.append(entry->mString);
+ }
+ return result;
+}
+
AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
audio_devices_t outDevice, audio_devices_t inDevice, type_t type)
: Thread(false /*canCallJava*/),
@@ -577,20 +647,21 @@
bool locked = AudioFlinger::dumpTryLock(mLock);
if (!locked) {
- dprintf(fd, "thread %p maybe dead locked\n", this);
+ dprintf(fd, "thread %p may be deadlocked\n", this);
}
dprintf(fd, " I/O handle: %d\n", mId);
dprintf(fd, " TID: %d\n", getTid());
dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no");
- dprintf(fd, " Sample rate: %u\n", mSampleRate);
+ dprintf(fd, " Sample rate: %u Hz\n", mSampleRate);
dprintf(fd, " HAL frame count: %zu\n", mFrameCount);
+ dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat));
dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize);
- dprintf(fd, " Channel Count: %u\n", mChannelCount);
- dprintf(fd, " Channel Mask: 0x%08x (%s)\n", mChannelMask,
+ dprintf(fd, " Channel count: %u\n", mChannelCount);
+ dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask,
channelMaskToString(mChannelMask, mType != RECORD).string());
- dprintf(fd, " Format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat));
- dprintf(fd, " Frame size: %zu\n", mFrameSize);
+ dprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat));
+ dprintf(fd, " Frame size: %zu bytes\n", mFrameSize);
dprintf(fd, " Pending config events:");
size_t numConfig = mConfigEvents.size();
if (numConfig) {
@@ -1315,7 +1386,7 @@
void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
{
- dprintf(fd, "\nOutput thread %p:\n", this);
+ dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type()));
dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount);
dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
dprintf(fd, " Total writes: %d\n", mNumWrites);
@@ -1326,6 +1397,10 @@
dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer);
dprintf(fd, " Effect buffer: %p\n", mEffectBuffer);
dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask);
+ AudioStreamOut *output = mOutput;
+ audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE;
+ String8 flagsAsString = outputFlagsToString(flags);
+ dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string());
dumpBase(fd, args);
}
@@ -1861,6 +1936,22 @@
}
}
+ if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) {
+ // For best precision, we use float instead of the associated output
+ // device format (typically PCM 16 bit).
+
+ mFormat = AUDIO_FORMAT_PCM_FLOAT;
+ mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat);
+ mBufferSize = mFrameSize * mFrameCount;
+
+ // TODO: We currently use the associated output device channel mask and sample rate.
+ // (1) Perhaps use the ORed channel mask of all downstream MixerThreads
+ // (if a valid mask) to avoid premature downmix.
+ // (2) Perhaps use the maximum sample rate of all downstream MixerThreads
+ // instead of the output device sample rate to avoid loss of high frequency information.
+ // This may need to be updated as MixerThread/OutputTracks are added and not here.
+ }
+
// Calculate size of normal sink buffer relative to the HAL output buffer size
double multiplier = 1.0;
if (mType == MIXER && (kUseFastMixer == FastMixer_Static ||
@@ -2137,6 +2228,7 @@
} else {
bytesWritten = framesWritten;
}
+ mLatchDValid = false;
status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp);
if (status == NO_ERROR) {
size_t totalFramesWritten = mNormalSink->framesWritten();
@@ -2640,7 +2732,9 @@
}
} else {
+ ATRACE_BEGIN("sleep");
usleep(sleepTime);
+ ATRACE_END();
}
}
@@ -2800,6 +2894,12 @@
mNormalFrameCount);
mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
+ if (type == DUPLICATING) {
+ // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks
+ // (downstream MixerThreads) in DuplicatingThread::threadLoop_write().
+ // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink.
+ return;
+ }
// create an NBAIO sink for the HAL output stream, and negotiate
mOutputSink = new AudioStreamOutSink(output->stream);
size_t numCounterOffers = 0;
@@ -2841,6 +2941,7 @@
NBAIO_Format format = mOutputSink->format();
NBAIO_Format origformat = format;
// adjust format to match that of the Fast Mixer
+ ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat);
format.mFormat = fastMixerFormat;
format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount;
@@ -3386,8 +3487,7 @@
if (sr == mSampleRate) {
desiredFrames = mNormalFrameCount;
} else {
- // +1 for rounding and +1 for additional sample needed for interpolation
- desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1;
+ desiredFrames = sourceFramesNeeded(sr, mNormalFrameCount, mSampleRate);
// add frames already consumed but not yet released by the resampler
// because mAudioTrackServerProxy->framesReady() will include these frames
desiredFrames += mAudioMixer->getUnreleasedFrames(track->name());
@@ -3405,6 +3505,23 @@
}
size_t framesReady = track->framesReady();
+ if (ATRACE_ENABLED()) {
+ // I wish we had formatted trace names
+ char traceName[16];
+ strcpy(traceName, "nRdy");
+ int name = track->name();
+ if (AudioMixer::TRACK0 <= name &&
+ name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) {
+ name -= AudioMixer::TRACK0;
+ traceName[4] = (name / 10) + '0';
+ traceName[5] = (name % 10) + '0';
+ } else {
+ traceName[4] = '?';
+ traceName[5] = '?';
+ }
+ traceName[6] = '\0';
+ ATRACE_INT(traceName, framesReady);
+ }
if ((framesReady >= minFrames) && track->isReady() &&
!track->isPaused() && !track->isTerminated())
{
@@ -4797,16 +4914,8 @@
ssize_t AudioFlinger::DuplicatingThread::threadLoop_write()
{
- // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT
- // for delivery downstream as needed. This in-place conversion is safe as
- // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format
- // (AUDIO_FORMAT_PCM_8_BIT is not allowed here).
- if (mFormat != AUDIO_FORMAT_PCM_16_BIT) {
- memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT,
- mSinkBuffer, mFormat, writeFrames * mChannelCount);
- }
for (size_t i = 0; i < outputTracks.size(); i++) {
- outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames);
+ outputTracks[i]->write(mSinkBuffer, writeFrames);
}
mStandby = false;
return (ssize_t)mSinkBufferSize;
@@ -4833,25 +4942,26 @@
void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
{
Mutex::Autolock _l(mLock);
- // FIXME explain this formula
- size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate();
- // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat
- // due to current usage case and restrictions on the AudioBufferProvider.
- // Actual buffer conversion is done in threadLoop_write().
- //
- // TODO: This may change in the future, depending on multichannel
- // (and non int16_t*) support on AF::PlaybackThread::OutputTrack
- OutputTrack *outputTrack = new OutputTrack(thread,
+ // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass.
+ // Adjust for thread->sampleRate() to determine minimum buffer frame count.
+ // Then triple buffer because Threads do not run synchronously and may not be clock locked.
+ const size_t frameCount =
+ 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate());
+ // TODO: Consider asynchronous sample rate conversion to handle clock disparity
+ // from different OutputTracks and their associated MixerThreads (e.g. one may
+ // nearly empty and the other may be dropping data).
+
+ sp<OutputTrack> outputTrack = new OutputTrack(thread,
this,
mSampleRate,
- AUDIO_FORMAT_PCM_16_BIT,
+ mFormat,
mChannelMask,
frameCount,
IPCThreadState::self()->getCallingUid());
if (outputTrack->cblk() != NULL) {
thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f);
mOutputTracks.add(outputTrack);
- ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
+ ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread);
updateWaitTime_l();
}
}
@@ -5135,7 +5245,9 @@
// sleep with mutex unlocked
if (sleepUs > 0) {
+ ATRACE_BEGIN("sleep");
usleep(sleepUs);
+ ATRACE_END();
sleepUs = 0;
}
@@ -5279,7 +5391,8 @@
state->mCommand = FastCaptureState::READ_WRITE;
#if 0 // FIXME
mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ?
- FastCaptureDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN);
+ FastCaptureDumpState::kSamplingNforLowRamDevice :
+ FastMixerDumpState::kSamplingN);
#endif
didModify = true;
}
@@ -5427,8 +5540,8 @@
upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (const int16_t *)src,
part1);
} else {
- downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (const int16_t *)src,
- part1);
+ downmix_to_mono_i16_from_stereo_i16((int16_t *)dst,
+ (const int16_t *)src, part1);
}
dst += part1 * activeTrack->mFrameSize;
front += part1;
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 1088843..a1ac42c 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -32,6 +32,8 @@
OFFLOAD // Thread class is OffloadThread
};
+ static const char *threadTypeToString(type_t type);
+
ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
audio_devices_t outDevice, audio_devices_t inDevice, type_t type);
virtual ~ThreadBase();
@@ -406,6 +408,7 @@
audio_channel_mask_t mChannelMask;
uint32_t mChannelCount;
size_t mFrameSize;
+ // not HAL frame size, this is for output sink (to pipe to fast mixer)
audio_format_t mFormat; // Source format for Recording and
// Sink format for Playback.
// Sink format may be different than
@@ -1167,7 +1170,8 @@
const sp<MemoryDealer> mReadOnlyHeap;
// one-time initialization, no locks required
- sp<FastCapture> mFastCapture; // non-0 if there is also a fast capture
+ sp<FastCapture> mFastCapture; // non-0 if there is also
+ // a fast capture
// FIXME audio watchdog thread
// contents are not guaranteed to be consistent, no locks required
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index e970036..7757ea2 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -859,6 +859,7 @@
if (mState == FLUSHED) {
mState = IDLE;
}
+ mPreviousValid = false;
}
}
@@ -1709,14 +1710,13 @@
mActive = false;
}
-bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
+bool AudioFlinger::PlaybackThread::OutputTrack::write(void* data, uint32_t frames)
{
Buffer *pInBuffer;
Buffer inBuffer;
- uint32_t channelCount = mChannelCount;
bool outputBufferFull = false;
inBuffer.frameCount = frames;
- inBuffer.i16 = data;
+ inBuffer.raw = data;
uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
@@ -1726,13 +1726,17 @@
if (thread != 0) {
MixerThread *mixerThread = (MixerThread *)thread.get();
if (mFrameCount > frames) {
+ // For the first write after being inactive, ensure that we have
+ // enough frames to fill mFrameCount (which should be multiples of
+ // the minimum buffer requirements of the downstream MixerThread).
+ // This provides enough frames for the downstream mixer to begin
+ // (see AudioFlinger::PlaybackThread::Track::isReady()).
if (mBufferQueue.size() < kMaxOverFlowBuffers) {
uint32_t startFrames = (mFrameCount - frames);
pInBuffer = new Buffer;
- pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
+ pInBuffer->mBuffer = calloc(1, startFrames * mFrameSize);
pInBuffer->frameCount = startFrames;
- pInBuffer->i16 = pInBuffer->mBuffer;
- memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
+ pInBuffer->raw = pInBuffer->mBuffer;
mBufferQueue.add(pInBuffer);
} else {
ALOGW("OutputTrack::write() %p no more buffers in queue", this);
@@ -1773,20 +1777,20 @@
uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount :
pInBuffer->frameCount;
- memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
+ memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * mFrameSize);
Proxy::Buffer buf;
buf.mFrameCount = outFrames;
buf.mRaw = NULL;
mClientProxy->releaseBuffer(&buf);
pInBuffer->frameCount -= outFrames;
- pInBuffer->i16 += outFrames * channelCount;
+ pInBuffer->raw = (int8_t *)pInBuffer->raw + outFrames * mFrameSize;
mOutBuffer.frameCount -= outFrames;
- mOutBuffer.i16 += outFrames * channelCount;
+ mOutBuffer.raw = (int8_t *)mOutBuffer.raw + outFrames * mFrameSize;
if (pInBuffer->frameCount == 0) {
if (mBufferQueue.size()) {
mBufferQueue.removeAt(0);
- delete [] pInBuffer->mBuffer;
+ free(pInBuffer->mBuffer);
delete pInBuffer;
ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this,
mThread.unsafe_get(), mBufferQueue.size());
@@ -1802,11 +1806,10 @@
if (thread != 0 && !thread->standby()) {
if (mBufferQueue.size() < kMaxOverFlowBuffers) {
pInBuffer = new Buffer;
- pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
+ pInBuffer->mBuffer = malloc(inBuffer.frameCount * mFrameSize);
pInBuffer->frameCount = inBuffer.frameCount;
- pInBuffer->i16 = pInBuffer->mBuffer;
- memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount *
- sizeof(int16_t));
+ pInBuffer->raw = pInBuffer->mBuffer;
+ memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * mFrameSize);
mBufferQueue.add(pInBuffer);
ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this,
mThread.unsafe_get(), mBufferQueue.size());
@@ -1817,23 +1820,10 @@
}
}
- // Calling write() with a 0 length buffer, means that no more data will be written:
- // If no more buffers are pending, fill output track buffer to make sure it is started
- // by output mixer.
- if (frames == 0 && mBufferQueue.size() == 0) {
- // FIXME borken, replace by getting framesReady() from proxy
- size_t user = 0; // was mCblk->user
- if (user < mFrameCount) {
- frames = mFrameCount - user;
- pInBuffer = new Buffer;
- pInBuffer->mBuffer = new int16_t[frames * channelCount];
- pInBuffer->frameCount = frames;
- pInBuffer->i16 = pInBuffer->mBuffer;
- memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
- mBufferQueue.add(pInBuffer);
- } else if (mActive) {
- stop();
- }
+ // Calling write() with a 0 length buffer means that no more data will be written:
+ // We rely on stop() to set the appropriate flags to allow the remaining frames to play out.
+ if (frames == 0 && mBufferQueue.size() == 0 && mActive) {
+ stop();
}
return outputBufferFull;
@@ -1859,7 +1849,7 @@
for (size_t i = 0; i < size; i++) {
Buffer *pBuffer = mBufferQueue.itemAt(i);
- delete [] pBuffer->mBuffer;
+ free(pBuffer->mBuffer);
delete pBuffer;
}
mBufferQueue.clear();
diff --git a/services/audioflinger/tests/mixer_to_wav_tests.sh b/services/audioflinger/tests/mixer_to_wav_tests.sh
index 9b39e77..e60e6d5 100755
--- a/services/audioflinger/tests/mixer_to_wav_tests.sh
+++ b/services/audioflinger/tests/mixer_to_wav_tests.sh
@@ -63,8 +63,18 @@
# process__genericResampling
# track__Resample / track__genericResample
adb shell test-mixer $1 -s 48000 \
+ -o /sdcard/tm48000grif.wav \
+ sine:2,4000,7520 chirp:2,9200 sine:1,3000,18000 \
+ sine:f,6,6000,19000 chirp:i,4,30000
+ adb pull /sdcard/tm48000grif.wav $2
+
+# Test:
+# process__genericResampling
+# track__Resample / track__genericResample
+ adb shell test-mixer $1 -s 48000 \
-o /sdcard/tm48000gr.wav \
- sine:2,4000,7520 chirp:2,9200 sine:1,3000,18000
+ sine:2,4000,7520 chirp:2,9200 sine:1,3000,18000 \
+ sine:6,6000,19000
adb pull /sdcard/tm48000gr.wav $2
# Test:
diff --git a/services/audioflinger/tests/test-mixer.cpp b/services/audioflinger/tests/test-mixer.cpp
index 9a4fad6..8da6245 100644
--- a/services/audioflinger/tests/test-mixer.cpp
+++ b/services/audioflinger/tests/test-mixer.cpp
@@ -39,7 +39,7 @@
fprintf(stderr, "Usage: %s [-f] [-m] [-c channels]"
" [-s sample-rate] [-o <output-file>] [-a <aux-buffer-file>] [-P csv]"
" (<input-file> | <command>)+\n", name);
- fprintf(stderr, " -f enable floating point input track\n");
+ fprintf(stderr, " -f enable floating point input track by default\n");
fprintf(stderr, " -m enable floating point mixer output\n");
fprintf(stderr, " -c number of mixer output channels\n");
fprintf(stderr, " -s mixer sample-rate\n");
@@ -47,8 +47,8 @@
fprintf(stderr, " -a <aux-buffer-file>\n");
fprintf(stderr, " -P # frames provided per call to resample() in CSV format\n");
fprintf(stderr, " <input-file> is a WAV file\n");
- fprintf(stderr, " <command> can be 'sine:<channels>,<frequency>,<samplerate>'\n");
- fprintf(stderr, " 'chirp:<channels>,<samplerate>'\n");
+ fprintf(stderr, " <command> can be 'sine:[(i|f),]<channels>,<frequency>,<samplerate>'\n");
+ fprintf(stderr, " 'chirp:[(i|f),]<channels>,<samplerate>'\n");
}
static int writeFile(const char *filename, const void *buffer,
@@ -78,6 +78,18 @@
return EXIT_SUCCESS;
}
+const char *parseFormat(const char *s, bool *useFloat) {
+ if (!strncmp(s, "f,", 2)) {
+ *useFloat = true;
+ return s + 2;
+ }
+ if (!strncmp(s, "i,", 2)) {
+ *useFloat = false;
+ return s + 2;
+ }
+ return s;
+}
+
int main(int argc, char* argv[]) {
const char* const progname = argv[0];
bool useInputFloat = false;
@@ -88,8 +100,9 @@
std::vector<int> Pvalues;
const char* outputFilename = NULL;
const char* auxFilename = NULL;
- std::vector<int32_t> Names;
- std::vector<SignalProvider> Providers;
+ std::vector<int32_t> names;
+ std::vector<SignalProvider> providers;
+ std::vector<audio_format_t> formats;
for (int ch; (ch = getopt(argc, argv, "fmc:s:o:a:P:")) != -1;) {
switch (ch) {
@@ -138,54 +151,65 @@
size_t outputFrames = 0;
// create providers for each track
- Providers.resize(argc);
+ names.resize(argc);
+ providers.resize(argc);
+ formats.resize(argc);
for (int i = 0; i < argc; ++i) {
static const char chirp[] = "chirp:";
static const char sine[] = "sine:";
static const double kSeconds = 1;
+ bool useFloat = useInputFloat;
if (!strncmp(argv[i], chirp, strlen(chirp))) {
std::vector<int> v;
+ const char *s = parseFormat(argv[i] + strlen(chirp), &useFloat);
- parseCSV(argv[i] + strlen(chirp), v);
+ parseCSV(s, v);
if (v.size() == 2) {
printf("creating chirp(%d %d)\n", v[0], v[1]);
- if (useInputFloat) {
- Providers[i].setChirp<float>(v[0], 0, v[1]/2, v[1], kSeconds);
+ if (useFloat) {
+ providers[i].setChirp<float>(v[0], 0, v[1]/2, v[1], kSeconds);
+ formats[i] = AUDIO_FORMAT_PCM_FLOAT;
} else {
- Providers[i].setChirp<int16_t>(v[0], 0, v[1]/2, v[1], kSeconds);
+ providers[i].setChirp<int16_t>(v[0], 0, v[1]/2, v[1], kSeconds);
+ formats[i] = AUDIO_FORMAT_PCM_16_BIT;
}
- Providers[i].setIncr(Pvalues);
+ providers[i].setIncr(Pvalues);
} else {
fprintf(stderr, "malformed input '%s'\n", argv[i]);
}
} else if (!strncmp(argv[i], sine, strlen(sine))) {
std::vector<int> v;
+ const char *s = parseFormat(argv[i] + strlen(sine), &useFloat);
- parseCSV(argv[i] + strlen(sine), v);
+ parseCSV(s, v);
if (v.size() == 3) {
printf("creating sine(%d %d %d)\n", v[0], v[1], v[2]);
- if (useInputFloat) {
- Providers[i].setSine<float>(v[0], v[1], v[2], kSeconds);
+ if (useFloat) {
+ providers[i].setSine<float>(v[0], v[1], v[2], kSeconds);
+ formats[i] = AUDIO_FORMAT_PCM_FLOAT;
} else {
- Providers[i].setSine<int16_t>(v[0], v[1], v[2], kSeconds);
+ providers[i].setSine<int16_t>(v[0], v[1], v[2], kSeconds);
+ formats[i] = AUDIO_FORMAT_PCM_16_BIT;
}
- Providers[i].setIncr(Pvalues);
+ providers[i].setIncr(Pvalues);
} else {
fprintf(stderr, "malformed input '%s'\n", argv[i]);
}
} else {
printf("creating filename(%s)\n", argv[i]);
if (useInputFloat) {
- Providers[i].setFile<float>(argv[i]);
+ providers[i].setFile<float>(argv[i]);
+ formats[i] = AUDIO_FORMAT_PCM_FLOAT;
} else {
- Providers[i].setFile<short>(argv[i]);
+ providers[i].setFile<short>(argv[i]);
+ formats[i] = AUDIO_FORMAT_PCM_16_BIT;
}
- Providers[i].setIncr(Pvalues);
+ providers[i].setIncr(Pvalues);
}
// calculate the number of output frames
- size_t nframes = (int64_t) Providers[i].getNumFrames() * outputSampleRate
- / Providers[i].getSampleRate();
+ size_t nframes = (int64_t) providers[i].getNumFrames() * outputSampleRate
+ / providers[i].getSampleRate();
if (i == 0 || outputFrames > nframes) { // choose minimum for outputFrames
outputFrames = nframes;
}
@@ -213,22 +237,20 @@
// create the mixer.
const size_t mixerFrameCount = 320; // typical numbers may range from 240 or 960
AudioMixer *mixer = new AudioMixer(mixerFrameCount, outputSampleRate);
- audio_format_t inputFormat = useInputFloat
- ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
audio_format_t mixerFormat = useMixerFloat
? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
- float f = AudioMixer::UNITY_GAIN_FLOAT / Providers.size(); // normalize volume by # tracks
+ float f = AudioMixer::UNITY_GAIN_FLOAT / providers.size(); // normalize volume by # tracks
static float f0; // zero
// set up the tracks.
- for (size_t i = 0; i < Providers.size(); ++i) {
- //printf("track %d out of %d\n", i, Providers.size());
- uint32_t channelMask = audio_channel_out_mask_from_count(Providers[i].getNumChannels());
+ for (size_t i = 0; i < providers.size(); ++i) {
+ //printf("track %d out of %d\n", i, providers.size());
+ uint32_t channelMask = audio_channel_out_mask_from_count(providers[i].getNumChannels());
int32_t name = mixer->getTrackName(channelMask,
- inputFormat, AUDIO_SESSION_OUTPUT_MIX);
+ formats[i], AUDIO_SESSION_OUTPUT_MIX);
ALOG_ASSERT(name >= 0);
- Names.push_back(name);
- mixer->setBufferProvider(name, &Providers[i]);
+ names[i] = name;
+ mixer->setBufferProvider(name, &providers[i]);
mixer->setParameter(name, AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
(void *)outputAddr);
mixer->setParameter(
@@ -240,7 +262,7 @@
name,
AudioMixer::TRACK,
AudioMixer::FORMAT,
- (void *)(uintptr_t)inputFormat);
+ (void *)(uintptr_t)formats[i]);
mixer->setParameter(
name,
AudioMixer::TRACK,
@@ -255,7 +277,7 @@
name,
AudioMixer::RESAMPLE,
AudioMixer::SAMPLE_RATE,
- (void *)(uintptr_t)Providers[i].getSampleRate());
+ (void *)(uintptr_t)providers[i].getSampleRate());
if (useRamp) {
mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME0, &f0);
mixer->setParameter(name, AudioMixer::VOLUME, AudioMixer::VOLUME1, &f0);
@@ -277,11 +299,11 @@
// pump the mixer to process data.
size_t i;
for (i = 0; i < outputFrames - mixerFrameCount; i += mixerFrameCount) {
- for (size_t j = 0; j < Names.size(); ++j) {
- mixer->setParameter(Names[j], AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
+ for (size_t j = 0; j < names.size(); ++j) {
+ mixer->setParameter(names[j], AudioMixer::TRACK, AudioMixer::MAIN_BUFFER,
(char *) outputAddr + i * outputFrameSize);
if (auxFilename) {
- mixer->setParameter(Names[j], AudioMixer::TRACK, AudioMixer::AUX_BUFFER,
+ mixer->setParameter(names[j], AudioMixer::TRACK, AudioMixer::AUX_BUFFER,
(char *) auxAddr + i * auxFrameSize);
}
}
diff --git a/services/audiopolicy/Android.mk b/services/audiopolicy/Android.mk
index 188fc89..351ed79 100644
--- a/services/audiopolicy/Android.mk
+++ b/services/audiopolicy/Android.mk
@@ -3,19 +3,19 @@
include $(CLEAR_VARS)
LOCAL_SRC_FILES:= \
- AudioPolicyService.cpp \
- AudioPolicyEffects.cpp
+ service/AudioPolicyService.cpp \
+ service/AudioPolicyEffects.cpp
ifeq ($(USE_LEGACY_AUDIO_POLICY), 1)
LOCAL_SRC_FILES += \
- AudioPolicyInterfaceImplLegacy.cpp \
- AudioPolicyClientImplLegacy.cpp
+ service/AudioPolicyInterfaceImplLegacy.cpp \
+ service/AudioPolicyClientImplLegacy.cpp
LOCAL_CFLAGS += -DUSE_LEGACY_AUDIO_POLICY
else
LOCAL_SRC_FILES += \
- AudioPolicyInterfaceImpl.cpp \
- AudioPolicyClientImpl.cpp
+ service/AudioPolicyInterfaceImpl.cpp \
+ service/AudioPolicyClientImpl.cpp
endif
LOCAL_C_INCLUDES := \
@@ -53,7 +53,15 @@
include $(CLEAR_VARS)
LOCAL_SRC_FILES:= \
- AudioPolicyManager.cpp
+ managerdefault/AudioPolicyManager.cpp \
+ managerdefault/ConfigParsingUtils.cpp \
+ managerdefault/Devices.cpp \
+ managerdefault/Gains.cpp \
+ managerdefault/HwModule.cpp \
+ managerdefault/IOProfile.cpp \
+ managerdefault/Ports.cpp \
+ managerdefault/AudioInputDescriptor.cpp \
+ managerdefault/AudioOutputDescriptor.cpp
LOCAL_SHARED_LIBRARIES := \
libcutils \
@@ -73,7 +81,7 @@
include $(CLEAR_VARS)
LOCAL_SRC_FILES:= \
- AudioPolicyFactory.cpp
+ manager/AudioPolicyFactory.cpp
LOCAL_SHARED_LIBRARIES := \
libaudiopolicymanagerdefault
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
index 4508fa7..116d0d6 100644
--- a/services/audiopolicy/AudioPolicyInterface.h
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -75,7 +75,8 @@
// indicate a change in device connection status
virtual status_t setDeviceConnectionState(audio_devices_t device,
audio_policy_dev_state_t state,
- const char *device_address) = 0;
+ const char *device_address,
+ const char *device_name) = 0;
// retrieve a device connection status
virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
const char *device_address) = 0;
diff --git a/services/audiopolicy/AudioPolicyManager.h b/services/audiopolicy/AudioPolicyManager.h
deleted file mode 100644
index cbdafa6..0000000
--- a/services/audiopolicy/AudioPolicyManager.h
+++ /dev/null
@@ -1,937 +0,0 @@
-/*
- * Copyright (C) 2009 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-
-#include <stdint.h>
-#include <sys/types.h>
-#include <cutils/config_utils.h>
-#include <cutils/misc.h>
-#include <utils/Timers.h>
-#include <utils/Errors.h>
-#include <utils/KeyedVector.h>
-#include <utils/SortedVector.h>
-#include <media/AudioPolicy.h>
-#include "AudioPolicyInterface.h"
-
-
-namespace android {
-
-// ----------------------------------------------------------------------------
-
-// Attenuation applied to STRATEGY_SONIFICATION streams when a headset is connected: 6dB
-#define SONIFICATION_HEADSET_VOLUME_FACTOR 0.5
-// Min volume for STRATEGY_SONIFICATION streams when limited by music volume: -36dB
-#define SONIFICATION_HEADSET_VOLUME_MIN 0.016
-// Time in milliseconds during which we consider that music is still active after a music
-// track was stopped - see computeVolume()
-#define SONIFICATION_HEADSET_MUSIC_DELAY 5000
-// Time in milliseconds after media stopped playing during which we consider that the
-// sonification should be as unobtrusive as during the time media was playing.
-#define SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY 5000
-// Time in milliseconds during witch some streams are muted while the audio path
-// is switched
-#define MUTE_TIME_MS 2000
-
-#define NUM_TEST_OUTPUTS 5
-
-#define NUM_VOL_CURVE_KNEES 2
-
-// Default minimum length allowed for offloading a compressed track
-// Can be overridden by the audio.offload.min.duration.secs property
-#define OFFLOAD_DEFAULT_MIN_DURATION_SECS 60
-
-#define MAX_MIXER_SAMPLING_RATE 48000
-#define MAX_MIXER_CHANNEL_COUNT 8
-
-// ----------------------------------------------------------------------------
-// AudioPolicyManager implements audio policy manager behavior common to all platforms.
-// ----------------------------------------------------------------------------
-
-class AudioPolicyManager: public AudioPolicyInterface
-#ifdef AUDIO_POLICY_TEST
- , public Thread
-#endif //AUDIO_POLICY_TEST
-{
-
-public:
- AudioPolicyManager(AudioPolicyClientInterface *clientInterface);
- virtual ~AudioPolicyManager();
-
- // AudioPolicyInterface
- virtual status_t setDeviceConnectionState(audio_devices_t device,
- audio_policy_dev_state_t state,
- const char *device_address);
- virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
- const char *device_address);
- virtual void setPhoneState(audio_mode_t state);
- virtual void setForceUse(audio_policy_force_use_t usage,
- audio_policy_forced_cfg_t config);
- virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
- virtual void setSystemProperty(const char* property, const char* value);
- virtual status_t initCheck();
- virtual audio_io_handle_t getOutput(audio_stream_type_t stream,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_output_flags_t flags,
- const audio_offload_info_t *offloadInfo);
- virtual status_t getOutputForAttr(const audio_attributes_t *attr,
- audio_io_handle_t *output,
- audio_session_t session,
- audio_stream_type_t *stream,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_output_flags_t flags,
- const audio_offload_info_t *offloadInfo);
- virtual status_t startOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session);
- virtual status_t stopOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session);
- virtual void releaseOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session);
- virtual status_t getInputForAttr(const audio_attributes_t *attr,
- audio_io_handle_t *input,
- audio_session_t session,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_input_flags_t flags,
- input_type_t *inputType);
-
- // indicates to the audio policy manager that the input starts being used.
- virtual status_t startInput(audio_io_handle_t input,
- audio_session_t session);
-
- // indicates to the audio policy manager that the input stops being used.
- virtual status_t stopInput(audio_io_handle_t input,
- audio_session_t session);
- virtual void releaseInput(audio_io_handle_t input,
- audio_session_t session);
- virtual void closeAllInputs();
- virtual void initStreamVolume(audio_stream_type_t stream,
- int indexMin,
- int indexMax);
- virtual status_t setStreamVolumeIndex(audio_stream_type_t stream,
- int index,
- audio_devices_t device);
- virtual status_t getStreamVolumeIndex(audio_stream_type_t stream,
- int *index,
- audio_devices_t device);
-
- // return the strategy corresponding to a given stream type
- virtual uint32_t getStrategyForStream(audio_stream_type_t stream);
- // return the strategy corresponding to the given audio attributes
- virtual uint32_t getStrategyForAttr(const audio_attributes_t *attr);
-
- // return the enabled output devices for the given stream type
- virtual audio_devices_t getDevicesForStream(audio_stream_type_t stream);
-
- virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc = NULL);
- virtual status_t registerEffect(const effect_descriptor_t *desc,
- audio_io_handle_t io,
- uint32_t strategy,
- int session,
- int id);
- virtual status_t unregisterEffect(int id);
- virtual status_t setEffectEnabled(int id, bool enabled);
-
- virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const;
- // return whether a stream is playing remotely, override to change the definition of
- // local/remote playback, used for instance by notification manager to not make
- // media players lose audio focus when not playing locally
- // For the base implementation, "remotely" means playing during screen mirroring which
- // uses an output for playback with a non-empty, non "0" address.
- virtual bool isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs = 0) const;
- virtual bool isSourceActive(audio_source_t source) const;
-
- virtual status_t dump(int fd);
-
- virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo);
-
- virtual status_t listAudioPorts(audio_port_role_t role,
- audio_port_type_t type,
- unsigned int *num_ports,
- struct audio_port *ports,
- unsigned int *generation);
- virtual status_t getAudioPort(struct audio_port *port);
- virtual status_t createAudioPatch(const struct audio_patch *patch,
- audio_patch_handle_t *handle,
- uid_t uid);
- virtual status_t releaseAudioPatch(audio_patch_handle_t handle,
- uid_t uid);
- virtual status_t listAudioPatches(unsigned int *num_patches,
- struct audio_patch *patches,
- unsigned int *generation);
- virtual status_t setAudioPortConfig(const struct audio_port_config *config);
- virtual void clearAudioPatches(uid_t uid);
-
- virtual status_t acquireSoundTriggerSession(audio_session_t *session,
- audio_io_handle_t *ioHandle,
- audio_devices_t *device);
-
- virtual status_t releaseSoundTriggerSession(audio_session_t session);
-
- virtual status_t registerPolicyMixes(Vector<AudioMix> mixes);
- virtual status_t unregisterPolicyMixes(Vector<AudioMix> mixes);
-
-protected:
-
- enum routing_strategy {
- STRATEGY_MEDIA,
- STRATEGY_PHONE,
- STRATEGY_SONIFICATION,
- STRATEGY_SONIFICATION_RESPECTFUL,
- STRATEGY_DTMF,
- STRATEGY_ENFORCED_AUDIBLE,
- STRATEGY_TRANSMITTED_THROUGH_SPEAKER,
- STRATEGY_ACCESSIBILITY,
- STRATEGY_REROUTING,
- NUM_STRATEGIES
- };
-
- // 4 points to define the volume attenuation curve, each characterized by the volume
- // index (from 0 to 100) at which they apply, and the attenuation in dB at that index.
- // we use 100 steps to avoid rounding errors when computing the volume in volIndexToAmpl()
-
- enum { VOLMIN = 0, VOLKNEE1 = 1, VOLKNEE2 = 2, VOLMAX = 3, VOLCNT = 4};
-
- class VolumeCurvePoint
- {
- public:
- int mIndex;
- float mDBAttenuation;
- };
-
- // device categories used for volume curve management.
- enum device_category {
- DEVICE_CATEGORY_HEADSET,
- DEVICE_CATEGORY_SPEAKER,
- DEVICE_CATEGORY_EARPIECE,
- DEVICE_CATEGORY_EXT_MEDIA,
- DEVICE_CATEGORY_CNT
- };
-
- class HwModule;
-
- class AudioGain: public RefBase
- {
- public:
- AudioGain(int index, bool useInChannelMask);
- virtual ~AudioGain() {}
-
- void dump(int fd, int spaces, int index) const;
-
- void getDefaultConfig(struct audio_gain_config *config);
- status_t checkConfig(const struct audio_gain_config *config);
- int mIndex;
- struct audio_gain mGain;
- bool mUseInChannelMask;
- };
-
- class AudioPort: public virtual RefBase
- {
- public:
- AudioPort(const String8& name, audio_port_type_t type,
- audio_port_role_t role, const sp<HwModule>& module);
- virtual ~AudioPort() {}
-
- virtual void toAudioPort(struct audio_port *port) const;
-
- void importAudioPort(const sp<AudioPort> port);
- void clearCapabilities();
-
- void loadSamplingRates(char *name);
- void loadFormats(char *name);
- void loadOutChannels(char *name);
- void loadInChannels(char *name);
-
- audio_gain_mode_t loadGainMode(char *name);
- void loadGain(cnode *root, int index);
- virtual void loadGains(cnode *root);
-
- // searches for an exact match
- status_t checkExactSamplingRate(uint32_t samplingRate) const;
- // searches for a compatible match, and returns the best match via updatedSamplingRate
- status_t checkCompatibleSamplingRate(uint32_t samplingRate,
- uint32_t *updatedSamplingRate) const;
- // searches for an exact match
- status_t checkExactChannelMask(audio_channel_mask_t channelMask) const;
- // searches for a compatible match, currently implemented for input channel masks only
- status_t checkCompatibleChannelMask(audio_channel_mask_t channelMask) const;
- status_t checkFormat(audio_format_t format) const;
- status_t checkGain(const struct audio_gain_config *gainConfig, int index) const;
-
- uint32_t pickSamplingRate() const;
- audio_channel_mask_t pickChannelMask() const;
- audio_format_t pickFormat() const;
-
- static const audio_format_t sPcmFormatCompareTable[];
- static int compareFormats(audio_format_t format1, audio_format_t format2);
-
- void dump(int fd, int spaces) const;
-
- String8 mName;
- audio_port_type_t mType;
- audio_port_role_t mRole;
- bool mUseInChannelMask;
- // by convention, "0' in the first entry in mSamplingRates, mChannelMasks or mFormats
- // indicates the supported parameters should be read from the output stream
- // after it is opened for the first time
- Vector <uint32_t> mSamplingRates; // supported sampling rates
- Vector <audio_channel_mask_t> mChannelMasks; // supported channel masks
- Vector <audio_format_t> mFormats; // supported audio formats
- Vector < sp<AudioGain> > mGains; // gain controllers
- sp<HwModule> mModule; // audio HW module exposing this I/O stream
- uint32_t mFlags; // attribute flags (e.g primary output,
- // direct output...).
- };
-
- class AudioPortConfig: public virtual RefBase
- {
- public:
- AudioPortConfig();
- virtual ~AudioPortConfig() {}
-
- status_t applyAudioPortConfig(const struct audio_port_config *config,
- struct audio_port_config *backupConfig = NULL);
- virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig = NULL) const = 0;
- virtual sp<AudioPort> getAudioPort() const = 0;
- uint32_t mSamplingRate;
- audio_format_t mFormat;
- audio_channel_mask_t mChannelMask;
- struct audio_gain_config mGain;
- };
-
-
- class AudioPatch: public RefBase
- {
- public:
- AudioPatch(audio_patch_handle_t handle,
- const struct audio_patch *patch, uid_t uid) :
- mHandle(handle), mPatch(*patch), mUid(uid), mAfPatchHandle(0) {}
-
- status_t dump(int fd, int spaces, int index) const;
-
- audio_patch_handle_t mHandle;
- struct audio_patch mPatch;
- uid_t mUid;
- audio_patch_handle_t mAfPatchHandle;
- };
-
- class DeviceDescriptor: public AudioPort, public AudioPortConfig
- {
- public:
- DeviceDescriptor(const String8& name, audio_devices_t type);
-
- virtual ~DeviceDescriptor() {}
-
- bool equals(const sp<DeviceDescriptor>& other) const;
-
- // AudioPortConfig
- virtual sp<AudioPort> getAudioPort() const { return (AudioPort*) this; }
- virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig = NULL) const;
-
- // AudioPort
- virtual void loadGains(cnode *root);
- virtual void toAudioPort(struct audio_port *port) const;
-
- status_t dump(int fd, int spaces, int index) const;
-
- audio_devices_t mDeviceType;
- String8 mAddress;
- audio_port_handle_t mId;
- };
-
- class DeviceVector : public SortedVector< sp<DeviceDescriptor> >
- {
- public:
- DeviceVector() : SortedVector(), mDeviceTypes(AUDIO_DEVICE_NONE) {}
-
- ssize_t add(const sp<DeviceDescriptor>& item);
- ssize_t remove(const sp<DeviceDescriptor>& item);
- ssize_t indexOf(const sp<DeviceDescriptor>& item) const;
-
- audio_devices_t types() const { return mDeviceTypes; }
-
- void loadDevicesFromType(audio_devices_t types);
- void loadDevicesFromName(char *name, const DeviceVector& declaredDevices);
-
- sp<DeviceDescriptor> getDevice(audio_devices_t type, String8 address) const;
- DeviceVector getDevicesFromType(audio_devices_t types) const;
- sp<DeviceDescriptor> getDeviceFromId(audio_port_handle_t id) const;
- sp<DeviceDescriptor> getDeviceFromName(const String8& name) const;
- DeviceVector getDevicesFromTypeAddr(audio_devices_t type, String8 address)
- const;
-
- private:
- void refreshTypes();
- audio_devices_t mDeviceTypes;
- };
-
- // the IOProfile class describes the capabilities of an output or input stream.
- // It is currently assumed that all combination of listed parameters are supported.
- // It is used by the policy manager to determine if an output or input is suitable for
- // a given use case, open/close it accordingly and connect/disconnect audio tracks
- // to/from it.
- class IOProfile : public AudioPort
- {
- public:
- IOProfile(const String8& name, audio_port_role_t role, const sp<HwModule>& module);
- virtual ~IOProfile();
-
- // This method is used for both output and input.
- // If parameter updatedSamplingRate is non-NULL, it is assigned the actual sample rate.
- // For input, flags is interpreted as audio_input_flags_t.
- // TODO: merge audio_output_flags_t and audio_input_flags_t.
- bool isCompatibleProfile(audio_devices_t device,
- String8 address,
- uint32_t samplingRate,
- uint32_t *updatedSamplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- uint32_t flags) const;
-
- void dump(int fd);
- void log();
-
- DeviceVector mSupportedDevices; // supported devices
- // (devices this output can be routed to)
- };
-
- class HwModule : public RefBase
- {
- public:
- HwModule(const char *name);
- ~HwModule();
-
- status_t loadOutput(cnode *root);
- status_t loadInput(cnode *root);
- status_t loadDevice(cnode *root);
-
- status_t addOutputProfile(String8 name, const audio_config_t *config,
- audio_devices_t device, String8 address);
- status_t removeOutputProfile(String8 name);
- status_t addInputProfile(String8 name, const audio_config_t *config,
- audio_devices_t device, String8 address);
- status_t removeInputProfile(String8 name);
-
- void dump(int fd);
-
- const char *const mName; // base name of the audio HW module (primary, a2dp ...)
- uint32_t mHalVersion; // audio HAL API version
- audio_module_handle_t mHandle;
- Vector < sp<IOProfile> > mOutputProfiles; // output profiles exposed by this module
- Vector < sp<IOProfile> > mInputProfiles; // input profiles exposed by this module
- DeviceVector mDeclaredDevices; // devices declared in audio_policy.conf
-
- };
-
- // default volume curve
- static const VolumeCurvePoint sDefaultVolumeCurve[AudioPolicyManager::VOLCNT];
- // default volume curve for media strategy
- static const VolumeCurvePoint sDefaultMediaVolumeCurve[AudioPolicyManager::VOLCNT];
- // volume curve for non-media audio on ext media outputs (HDMI, Line, etc)
- static const VolumeCurvePoint sExtMediaSystemVolumeCurve[AudioPolicyManager::VOLCNT];
- // volume curve for media strategy on speakers
- static const VolumeCurvePoint sSpeakerMediaVolumeCurve[AudioPolicyManager::VOLCNT];
- static const VolumeCurvePoint sSpeakerMediaVolumeCurveDrc[AudioPolicyManager::VOLCNT];
- // volume curve for sonification strategy on speakers
- static const VolumeCurvePoint sSpeakerSonificationVolumeCurve[AudioPolicyManager::VOLCNT];
- static const VolumeCurvePoint sSpeakerSonificationVolumeCurveDrc[AudioPolicyManager::VOLCNT];
- static const VolumeCurvePoint sDefaultSystemVolumeCurve[AudioPolicyManager::VOLCNT];
- static const VolumeCurvePoint sDefaultSystemVolumeCurveDrc[AudioPolicyManager::VOLCNT];
- static const VolumeCurvePoint sHeadsetSystemVolumeCurve[AudioPolicyManager::VOLCNT];
- static const VolumeCurvePoint sDefaultVoiceVolumeCurve[AudioPolicyManager::VOLCNT];
- static const VolumeCurvePoint sSpeakerVoiceVolumeCurve[AudioPolicyManager::VOLCNT];
- static const VolumeCurvePoint sLinearVolumeCurve[AudioPolicyManager::VOLCNT];
- static const VolumeCurvePoint sSilentVolumeCurve[AudioPolicyManager::VOLCNT];
- static const VolumeCurvePoint sFullScaleVolumeCurve[AudioPolicyManager::VOLCNT];
- // default volume curves per stream and device category. See initializeVolumeCurves()
- static const VolumeCurvePoint *sVolumeProfiles[AUDIO_STREAM_CNT][DEVICE_CATEGORY_CNT];
-
- // descriptor for audio outputs. Used to maintain current configuration of each opened audio output
- // and keep track of the usage of this output by each audio stream type.
- class AudioOutputDescriptor: public AudioPortConfig
- {
- public:
- AudioOutputDescriptor(const sp<IOProfile>& profile);
-
- status_t dump(int fd);
-
- audio_devices_t device() const;
- void changeRefCount(audio_stream_type_t stream, int delta);
-
- bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); }
- audio_devices_t supportedDevices();
- uint32_t latency();
- bool sharesHwModuleWith(const sp<AudioOutputDescriptor> outputDesc);
- bool isActive(uint32_t inPastMs = 0) const;
- bool isStreamActive(audio_stream_type_t stream,
- uint32_t inPastMs = 0,
- nsecs_t sysTime = 0) const;
- bool isStrategyActive(routing_strategy strategy,
- uint32_t inPastMs = 0,
- nsecs_t sysTime = 0) const;
-
- virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig = NULL) const;
- virtual sp<AudioPort> getAudioPort() const { return mProfile; }
- void toAudioPort(struct audio_port *port) const;
-
- audio_port_handle_t mId;
- audio_io_handle_t mIoHandle; // output handle
- uint32_t mLatency; //
- audio_output_flags_t mFlags; //
- audio_devices_t mDevice; // current device this output is routed to
- AudioMix *mPolicyMix; // non NULL when used by a dynamic policy
- audio_patch_handle_t mPatchHandle;
- uint32_t mRefCount[AUDIO_STREAM_CNT]; // number of streams of each type using this output
- nsecs_t mStopTime[AUDIO_STREAM_CNT];
- sp<AudioOutputDescriptor> mOutput1; // used by duplicated outputs: first output
- sp<AudioOutputDescriptor> mOutput2; // used by duplicated outputs: second output
- float mCurVolume[AUDIO_STREAM_CNT]; // current stream volume
- int mMuteCount[AUDIO_STREAM_CNT]; // mute request counter
- const sp<IOProfile> mProfile; // I/O profile this output derives from
- bool mStrategyMutedByDevice[NUM_STRATEGIES]; // strategies muted because of incompatible
- // device selection. See checkDeviceMuteStrategies()
- uint32_t mDirectOpenCount; // number of clients using this output (direct outputs only)
- };
-
- // descriptor for audio inputs. Used to maintain current configuration of each opened audio input
- // and keep track of the usage of this input.
- class AudioInputDescriptor: public AudioPortConfig
- {
- public:
- AudioInputDescriptor(const sp<IOProfile>& profile);
-
- status_t dump(int fd);
-
- audio_port_handle_t mId;
- audio_io_handle_t mIoHandle; // input handle
- audio_devices_t mDevice; // current device this input is routed to
- AudioMix *mPolicyMix; // non NULL when used by a dynamic policy
- audio_patch_handle_t mPatchHandle;
- uint32_t mRefCount; // number of AudioRecord clients using
- // this input
- uint32_t mOpenRefCount;
- audio_source_t mInputSource; // input source selected by application
- //(mediarecorder.h)
- const sp<IOProfile> mProfile; // I/O profile this output derives from
- SortedVector<audio_session_t> mSessions; // audio sessions attached to this input
- bool mIsSoundTrigger; // used by a soundtrigger capture
-
- virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig = NULL) const;
- virtual sp<AudioPort> getAudioPort() const { return mProfile; }
- void toAudioPort(struct audio_port *port) const;
- };
-
- // stream descriptor used for volume control
- class StreamDescriptor
- {
- public:
- StreamDescriptor();
-
- int getVolumeIndex(audio_devices_t device);
- void dump(int fd);
-
- int mIndexMin; // min volume index
- int mIndexMax; // max volume index
- KeyedVector<audio_devices_t, int> mIndexCur; // current volume index per device
- bool mCanBeMuted; // true is the stream can be muted
-
- const VolumeCurvePoint *mVolumeCurve[DEVICE_CATEGORY_CNT];
- };
-
- // stream descriptor used for volume control
- class EffectDescriptor : public RefBase
- {
- public:
-
- status_t dump(int fd);
-
- int mIo; // io the effect is attached to
- routing_strategy mStrategy; // routing strategy the effect is associated to
- int mSession; // audio session the effect is on
- effect_descriptor_t mDesc; // effect descriptor
- bool mEnabled; // enabled state: CPU load being used or not
- };
-
- void addOutput(audio_io_handle_t output, sp<AudioOutputDescriptor> outputDesc);
- void addInput(audio_io_handle_t input, sp<AudioInputDescriptor> inputDesc);
-
- // return the strategy corresponding to a given stream type
- static routing_strategy getStrategy(audio_stream_type_t stream);
-
- // return appropriate device for streams handled by the specified strategy according to current
- // phone state, connected devices...
- // if fromCache is true, the device is returned from mDeviceForStrategy[],
- // otherwise it is determine by current state
- // (device connected,phone state, force use, a2dp output...)
- // This allows to:
- // 1 speed up process when the state is stable (when starting or stopping an output)
- // 2 access to either current device selection (fromCache == true) or
- // "future" device selection (fromCache == false) when called from a context
- // where conditions are changing (setDeviceConnectionState(), setPhoneState()...) AND
- // before updateDevicesAndOutputs() is called.
- virtual audio_devices_t getDeviceForStrategy(routing_strategy strategy,
- bool fromCache);
-
- // change the route of the specified output. Returns the number of ms we have slept to
- // allow new routing to take effect in certain cases.
- virtual uint32_t setOutputDevice(audio_io_handle_t output,
- audio_devices_t device,
- bool force = false,
- int delayMs = 0,
- audio_patch_handle_t *patchHandle = NULL,
- const char* address = NULL);
- status_t resetOutputDevice(audio_io_handle_t output,
- int delayMs = 0,
- audio_patch_handle_t *patchHandle = NULL);
- status_t setInputDevice(audio_io_handle_t input,
- audio_devices_t device,
- bool force = false,
- audio_patch_handle_t *patchHandle = NULL);
- status_t resetInputDevice(audio_io_handle_t input,
- audio_patch_handle_t *patchHandle = NULL);
-
- // select input device corresponding to requested audio source
- virtual audio_devices_t getDeviceForInputSource(audio_source_t inputSource);
-
- // return io handle of active input or 0 if no input is active
- // Only considers inputs from physical devices (e.g. main mic, headset mic) when
- // ignoreVirtualInputs is true.
- audio_io_handle_t getActiveInput(bool ignoreVirtualInputs = true);
-
- uint32_t activeInputsCount() const;
-
- // initialize volume curves for each strategy and device category
- void initializeVolumeCurves();
-
- // compute the actual volume for a given stream according to the requested index and a particular
- // device
- virtual float computeVolume(audio_stream_type_t stream, int index,
- audio_io_handle_t output, audio_devices_t device);
-
- // check that volume change is permitted, compute and send new volume to audio hardware
- virtual status_t checkAndSetVolume(audio_stream_type_t stream, int index,
- audio_io_handle_t output,
- audio_devices_t device,
- int delayMs = 0, bool force = false);
-
- // apply all stream volumes to the specified output and device
- void applyStreamVolumes(audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false);
-
- // Mute or unmute all streams handled by the specified strategy on the specified output
- void setStrategyMute(routing_strategy strategy,
- bool on,
- audio_io_handle_t output,
- int delayMs = 0,
- audio_devices_t device = (audio_devices_t)0);
-
- // Mute or unmute the stream on the specified output
- void setStreamMute(audio_stream_type_t stream,
- bool on,
- audio_io_handle_t output,
- int delayMs = 0,
- audio_devices_t device = (audio_devices_t)0);
-
- // handle special cases for sonification strategy while in call: mute streams or replace by
- // a special tone in the device used for communication
- void handleIncallSonification(audio_stream_type_t stream, bool starting, bool stateChange);
-
- // true if device is in a telephony or VoIP call
- virtual bool isInCall();
-
- // true if given state represents a device in a telephony or VoIP call
- virtual bool isStateInCall(int state);
-
- // when a device is connected, checks if an open output can be routed
- // to this device. If none is open, tries to open one of the available outputs.
- // Returns an output suitable to this device or 0.
- // when a device is disconnected, checks if an output is not used any more and
- // returns its handle if any.
- // transfers the audio tracks and effects from one output thread to another accordingly.
- status_t checkOutputsForDevice(const sp<DeviceDescriptor> devDesc,
- audio_policy_dev_state_t state,
- SortedVector<audio_io_handle_t>& outputs,
- const String8 address);
-
- status_t checkInputsForDevice(audio_devices_t device,
- audio_policy_dev_state_t state,
- SortedVector<audio_io_handle_t>& inputs,
- const String8 address);
-
- // close an output and its companion duplicating output.
- void closeOutput(audio_io_handle_t output);
-
- // close an input.
- void closeInput(audio_io_handle_t input);
-
- // checks and if necessary changes outputs used for all strategies.
- // must be called every time a condition that affects the output choice for a given strategy
- // changes: connected device, phone state, force use...
- // Must be called before updateDevicesAndOutputs()
- void checkOutputForStrategy(routing_strategy strategy);
-
- // Same as checkOutputForStrategy() but for a all strategies in order of priority
- void checkOutputForAllStrategies();
-
- // manages A2DP output suspend/restore according to phone state and BT SCO usage
- void checkA2dpSuspend();
-
- // returns the A2DP output handle if it is open or 0 otherwise
- audio_io_handle_t getA2dpOutput();
-
- // selects the most appropriate device on output for current state
- // must be called every time a condition that affects the device choice for a given output is
- // changed: connected device, phone state, force use, output start, output stop..
- // see getDeviceForStrategy() for the use of fromCache parameter
- audio_devices_t getNewOutputDevice(audio_io_handle_t output, bool fromCache);
-
- // updates cache of device used by all strategies (mDeviceForStrategy[])
- // must be called every time a condition that affects the device choice for a given strategy is
- // changed: connected device, phone state, force use...
- // cached values are used by getDeviceForStrategy() if parameter fromCache is true.
- // Must be called after checkOutputForAllStrategies()
- void updateDevicesAndOutputs();
-
- // selects the most appropriate device on input for current state
- audio_devices_t getNewInputDevice(audio_io_handle_t input);
-
- virtual uint32_t getMaxEffectsCpuLoad();
- virtual uint32_t getMaxEffectsMemory();
-#ifdef AUDIO_POLICY_TEST
- virtual bool threadLoop();
- void exit();
- int testOutputIndex(audio_io_handle_t output);
-#endif //AUDIO_POLICY_TEST
-
- status_t setEffectEnabled(const sp<EffectDescriptor>& effectDesc, bool enabled);
-
- // returns the category the device belongs to with regard to volume curve management
- static device_category getDeviceCategory(audio_devices_t device);
-
- // extract one device relevant for volume control from multiple device selection
- static audio_devices_t getDeviceForVolume(audio_devices_t device);
-
- SortedVector<audio_io_handle_t> getOutputsForDevice(audio_devices_t device,
- DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > openOutputs);
- bool vectorsEqual(SortedVector<audio_io_handle_t>& outputs1,
- SortedVector<audio_io_handle_t>& outputs2);
-
- // mute/unmute strategies using an incompatible device combination
- // if muting, wait for the audio in pcm buffer to be drained before proceeding
- // if unmuting, unmute only after the specified delay
- // Returns the number of ms waited
- virtual uint32_t checkDeviceMuteStrategies(sp<AudioOutputDescriptor> outputDesc,
- audio_devices_t prevDevice,
- uint32_t delayMs);
-
- audio_io_handle_t selectOutput(const SortedVector<audio_io_handle_t>& outputs,
- audio_output_flags_t flags,
- audio_format_t format);
- // samplingRate parameter is an in/out and so may be modified
- sp<IOProfile> getInputProfile(audio_devices_t device,
- String8 address,
- uint32_t& samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_input_flags_t flags);
- sp<IOProfile> getProfileForDirectOutput(audio_devices_t device,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_output_flags_t flags);
-
- audio_io_handle_t selectOutputForEffects(const SortedVector<audio_io_handle_t>& outputs);
-
- bool isNonOffloadableEffectEnabled();
-
- virtual status_t addAudioPatch(audio_patch_handle_t handle,
- const sp<AudioPatch>& patch);
- virtual status_t removeAudioPatch(audio_patch_handle_t handle);
-
- sp<AudioOutputDescriptor> getOutputFromId(audio_port_handle_t id) const;
- sp<AudioInputDescriptor> getInputFromId(audio_port_handle_t id) const;
- sp<HwModule> getModuleForDevice(audio_devices_t device) const;
- sp<HwModule> getModuleFromName(const char *name) const;
- audio_devices_t availablePrimaryOutputDevices();
- audio_devices_t availablePrimaryInputDevices();
-
- void updateCallRouting(audio_devices_t rxDevice, int delayMs = 0);
-
- //
- // Audio policy configuration file parsing (audio_policy.conf)
- //
- static uint32_t stringToEnum(const struct StringToEnum *table,
- size_t size,
- const char *name);
- static const char *enumToString(const struct StringToEnum *table,
- size_t size,
- uint32_t value);
- static bool stringToBool(const char *value);
- static uint32_t parseOutputFlagNames(char *name);
- static uint32_t parseInputFlagNames(char *name);
- static audio_devices_t parseDeviceNames(char *name);
- void loadHwModule(cnode *root);
- void loadHwModules(cnode *root);
- void loadGlobalConfig(cnode *root, const sp<HwModule>& module);
- status_t loadAudioPolicyConfig(const char *path);
- void defaultAudioPolicyConfig(void);
-
-
- uid_t mUidCached;
- AudioPolicyClientInterface *mpClientInterface; // audio policy client interface
- audio_io_handle_t mPrimaryOutput; // primary output handle
- // list of descriptors for outputs currently opened
- DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > mOutputs;
- // copy of mOutputs before setDeviceConnectionState() opens new outputs
- // reset to mOutputs when updateDevicesAndOutputs() is called.
- DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > mPreviousOutputs;
- DefaultKeyedVector<audio_io_handle_t, sp<AudioInputDescriptor> > mInputs; // list of input descriptors
- DeviceVector mAvailableOutputDevices; // all available output devices
- DeviceVector mAvailableInputDevices; // all available input devices
- int mPhoneState; // current phone state
- audio_policy_forced_cfg_t mForceUse[AUDIO_POLICY_FORCE_USE_CNT]; // current forced use configuration
-
- StreamDescriptor mStreams[AUDIO_STREAM_CNT]; // stream descriptors for volume control
- bool mLimitRingtoneVolume; // limit ringtone volume to music volume if headset connected
- audio_devices_t mDeviceForStrategy[NUM_STRATEGIES];
- float mLastVoiceVolume; // last voice volume value sent to audio HAL
-
- // Maximum CPU load allocated to audio effects in 0.1 MIPS (ARMv5TE, 0 WS memory) units
- static const uint32_t MAX_EFFECTS_CPU_LOAD = 1000;
- // Maximum memory allocated to audio effects in KB
- static const uint32_t MAX_EFFECTS_MEMORY = 512;
- uint32_t mTotalEffectsCpuLoad; // current CPU load used by effects
- uint32_t mTotalEffectsMemory; // current memory used by effects
- KeyedVector<int, sp<EffectDescriptor> > mEffects; // list of registered audio effects
- bool mA2dpSuspended; // true if A2DP output is suspended
- sp<DeviceDescriptor> mDefaultOutputDevice; // output device selected by default at boot time
- bool mSpeakerDrcEnabled;// true on devices that use DRC on the DEVICE_CATEGORY_SPEAKER path
- // to boost soft sounds, used to adjust volume curves accordingly
-
- Vector < sp<HwModule> > mHwModules;
- volatile int32_t mNextUniqueId;
- volatile int32_t mAudioPortGeneration;
-
- DefaultKeyedVector<audio_patch_handle_t, sp<AudioPatch> > mAudioPatches;
-
- DefaultKeyedVector<audio_session_t, audio_io_handle_t> mSoundTriggerSessions;
-
- sp<AudioPatch> mCallTxPatch;
- sp<AudioPatch> mCallRxPatch;
-
- // for supporting "beacon" streams, i.e. streams that only play on speaker, and never
- // when something other than STREAM_TTS (a.k.a. "Transmitted Through Speaker") is playing
- enum {
- STARTING_OUTPUT,
- STARTING_BEACON,
- STOPPING_OUTPUT,
- STOPPING_BEACON
- };
- uint32_t mBeaconMuteRefCount; // ref count for stream that would mute beacon
- uint32_t mBeaconPlayingRefCount;// ref count for the playing beacon streams
- bool mBeaconMuted; // has STREAM_TTS been muted
-
- // custom mix entry in mPolicyMixes
- class AudioPolicyMix : public RefBase {
- public:
- AudioPolicyMix() {}
-
- AudioMix mMix; // Audio policy mix descriptor
- sp<AudioOutputDescriptor> mOutput; // Corresponding output stream
- };
- DefaultKeyedVector<String8, sp<AudioPolicyMix> > mPolicyMixes; // list of registered mixes
-
-
-#ifdef AUDIO_POLICY_TEST
- Mutex mLock;
- Condition mWaitWorkCV;
-
- int mCurOutput;
- bool mDirectOutput;
- audio_io_handle_t mTestOutputs[NUM_TEST_OUTPUTS];
- int mTestInput;
- uint32_t mTestDevice;
- uint32_t mTestSamplingRate;
- uint32_t mTestFormat;
- uint32_t mTestChannels;
- uint32_t mTestLatencyMs;
-#endif //AUDIO_POLICY_TEST
- static float volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
- int indexInUi);
- static bool isVirtualInputDevice(audio_devices_t device);
- uint32_t nextUniqueId();
- uint32_t nextAudioPortGeneration();
-private:
- // updates device caching and output for streams that can influence the
- // routing of notifications
- void handleNotificationRoutingForStream(audio_stream_type_t stream);
- static bool deviceDistinguishesOnAddress(audio_devices_t device);
- // find the outputs on a given output descriptor that have the given address.
- // to be called on an AudioOutputDescriptor whose supported devices (as defined
- // in mProfile->mSupportedDevices) matches the device whose address is to be matched.
- // see deviceDistinguishesOnAddress(audio_devices_t) for whether the device type is one
- // where addresses are used to distinguish between one connected device and another.
- void findIoHandlesByAddress(sp<AudioOutputDescriptor> desc /*in*/,
- const audio_devices_t device /*in*/,
- const String8 address /*in*/,
- SortedVector<audio_io_handle_t>& outputs /*out*/);
- uint32_t curAudioPortGeneration() const { return mAudioPortGeneration; }
- // internal method to return the output handle for the given device and format
- audio_io_handle_t getOutputForDevice(
- audio_devices_t device,
- audio_session_t session,
- audio_stream_type_t stream,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_output_flags_t flags,
- const audio_offload_info_t *offloadInfo);
- // internal function to derive a stream type value from audio attributes
- audio_stream_type_t streamTypefromAttributesInt(const audio_attributes_t *attr);
- // return true if any output is playing anything besides the stream to ignore
- bool isAnyOutputActive(audio_stream_type_t streamToIgnore);
- // event is one of STARTING_OUTPUT, STARTING_BEACON, STOPPING_OUTPUT, STOPPING_BEACON
- // returns 0 if no mute/unmute event happened, the largest latency of the device where
- // the mute/unmute happened
- uint32_t handleEventForBeacon(int event);
- uint32_t setBeaconMute(bool mute);
- bool isValidAttributes(const audio_attributes_t *paa);
-
- // select input device corresponding to requested audio source and return associated policy
- // mix if any. Calls getDeviceForInputSource().
- audio_devices_t getDeviceAndMixForInputSource(audio_source_t inputSource,
- AudioMix **policyMix = NULL);
-
- // Called by setDeviceConnectionState().
- status_t setDeviceConnectionStateInt(audio_devices_t device,
- audio_policy_dev_state_t state,
- const char *device_address);
- sp<DeviceDescriptor> getDeviceDescriptor(const audio_devices_t device,
- const char *device_address);
-
-};
-
-};
diff --git a/services/audiopolicy/AudioPolicyFactory.cpp b/services/audiopolicy/manager/AudioPolicyFactory.cpp
similarity index 94%
rename from services/audiopolicy/AudioPolicyFactory.cpp
rename to services/audiopolicy/manager/AudioPolicyFactory.cpp
index 2ae7bc1..9910a1f 100644
--- a/services/audiopolicy/AudioPolicyFactory.cpp
+++ b/services/audiopolicy/manager/AudioPolicyFactory.cpp
@@ -14,7 +14,7 @@
* limitations under the License.
*/
-#include "AudioPolicyManager.h"
+#include "managerdefault/AudioPolicyManager.h"
namespace android {
diff --git a/services/audiopolicy/AudioPolicyFactory.cpp b/services/audiopolicy/managerdefault/ApmImplDefinitions.h
similarity index 60%
copy from services/audiopolicy/AudioPolicyFactory.cpp
copy to services/audiopolicy/managerdefault/ApmImplDefinitions.h
index 2ae7bc1..620979b 100644
--- a/services/audiopolicy/AudioPolicyFactory.cpp
+++ b/services/audiopolicy/managerdefault/ApmImplDefinitions.h
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2014 The Android Open Source Project
+ * Copyright (C) 2015 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
@@ -14,19 +14,19 @@
* limitations under the License.
*/
-#include "AudioPolicyManager.h"
-
namespace android {
-extern "C" AudioPolicyInterface* createAudioPolicyManager(
- AudioPolicyClientInterface *clientInterface)
-{
- return new AudioPolicyManager(clientInterface);
-}
+enum routing_strategy {
+ STRATEGY_MEDIA,
+ STRATEGY_PHONE,
+ STRATEGY_SONIFICATION,
+ STRATEGY_SONIFICATION_RESPECTFUL,
+ STRATEGY_DTMF,
+ STRATEGY_ENFORCED_AUDIBLE,
+ STRATEGY_TRANSMITTED_THROUGH_SPEAKER,
+ STRATEGY_ACCESSIBILITY,
+ STRATEGY_REROUTING,
+ NUM_STRATEGIES
+};
-extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface)
-{
- delete interface;
-}
-
-}; // namespace android
+}; //namespace android
diff --git a/services/audiopolicy/managerdefault/AudioInputDescriptor.cpp b/services/audiopolicy/managerdefault/AudioInputDescriptor.cpp
new file mode 100644
index 0000000..f4054c8
--- /dev/null
+++ b/services/audiopolicy/managerdefault/AudioInputDescriptor.cpp
@@ -0,0 +1,100 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM::AudioInputDescriptor"
+//#define LOG_NDEBUG 0
+
+#include "AudioPolicyManager.h"
+
+namespace android {
+
+AudioInputDescriptor::AudioInputDescriptor(const sp<IOProfile>& profile)
+ : mId(0), mIoHandle(0),
+ mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL), mPatchHandle(0), mRefCount(0),
+ mInputSource(AUDIO_SOURCE_DEFAULT), mProfile(profile), mIsSoundTrigger(false)
+{
+ if (profile != NULL) {
+ mSamplingRate = profile->pickSamplingRate();
+ mFormat = profile->pickFormat();
+ mChannelMask = profile->pickChannelMask();
+ if (profile->mGains.size() > 0) {
+ profile->mGains[0]->getDefaultConfig(&mGain);
+ }
+ }
+}
+
+void AudioInputDescriptor::toAudioPortConfig(
+ struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig) const
+{
+ ALOG_ASSERT(mProfile != 0,
+ "toAudioPortConfig() called on input with null profile %d", mIoHandle);
+ dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
+ AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN;
+ if (srcConfig != NULL) {
+ dstConfig->config_mask |= srcConfig->config_mask;
+ }
+
+ AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
+
+ dstConfig->id = mId;
+ dstConfig->role = AUDIO_PORT_ROLE_SINK;
+ dstConfig->type = AUDIO_PORT_TYPE_MIX;
+ dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle;
+ dstConfig->ext.mix.handle = mIoHandle;
+ dstConfig->ext.mix.usecase.source = mInputSource;
+}
+
+void AudioInputDescriptor::toAudioPort(
+ struct audio_port *port) const
+{
+ ALOG_ASSERT(mProfile != 0, "toAudioPort() called on input with null profile %d", mIoHandle);
+
+ mProfile->toAudioPort(port);
+ port->id = mId;
+ toAudioPortConfig(&port->active_config);
+ port->ext.mix.hw_module = mProfile->mModule->mHandle;
+ port->ext.mix.handle = mIoHandle;
+ port->ext.mix.latency_class = AUDIO_LATENCY_NORMAL;
+}
+
+status_t AudioInputDescriptor::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, " ID: %d\n", mId);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Format: %d\n", mFormat);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Devices %08x\n", mDevice);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Ref Count %d\n", mRefCount);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Open Ref Count %d\n", mOpenRefCount);
+ result.append(buffer);
+
+ write(fd, result.string(), result.size());
+
+ return NO_ERROR;
+}
+
+}; //namespace android
diff --git a/services/audiopolicy/managerdefault/AudioInputDescriptor.h b/services/audiopolicy/managerdefault/AudioInputDescriptor.h
new file mode 100644
index 0000000..02579e6
--- /dev/null
+++ b/services/audiopolicy/managerdefault/AudioInputDescriptor.h
@@ -0,0 +1,48 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+namespace android {
+
+// descriptor for audio inputs. Used to maintain current configuration of each opened audio input
+// and keep track of the usage of this input.
+class AudioInputDescriptor: public AudioPortConfig
+{
+public:
+ AudioInputDescriptor(const sp<IOProfile>& profile);
+
+ status_t dump(int fd);
+
+ audio_port_handle_t mId;
+ audio_io_handle_t mIoHandle; // input handle
+ audio_devices_t mDevice; // current device this input is routed to
+ AudioMix *mPolicyMix; // non NULL when used by a dynamic policy
+ audio_patch_handle_t mPatchHandle;
+ uint32_t mRefCount; // number of AudioRecord clients using
+ // this input
+ uint32_t mOpenRefCount;
+ audio_source_t mInputSource; // input source selected by application
+ //(mediarecorder.h)
+ const sp<IOProfile> mProfile; // I/O profile this output derives from
+ SortedVector<audio_session_t> mSessions; // audio sessions attached to this input
+ bool mIsSoundTrigger; // used by a soundtrigger capture
+
+ virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig = NULL) const;
+ virtual sp<AudioPort> getAudioPort() const { return mProfile; }
+ void toAudioPort(struct audio_port *port) const;
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/AudioOutputDescriptor.cpp b/services/audiopolicy/managerdefault/AudioOutputDescriptor.cpp
new file mode 100644
index 0000000..4b85972
--- /dev/null
+++ b/services/audiopolicy/managerdefault/AudioOutputDescriptor.cpp
@@ -0,0 +1,221 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM::AudioOutputDescriptor"
+//#define LOG_NDEBUG 0
+
+#include "AudioPolicyManager.h"
+
+namespace android {
+
+AudioOutputDescriptor::AudioOutputDescriptor(
+ const sp<IOProfile>& profile)
+ : mId(0), mIoHandle(0), mLatency(0),
+ mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL),
+ mPatchHandle(0),
+ mOutput1(0), mOutput2(0), mProfile(profile), mDirectOpenCount(0)
+{
+ // clear usage count for all stream types
+ for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
+ mRefCount[i] = 0;
+ mCurVolume[i] = -1.0;
+ mMuteCount[i] = 0;
+ mStopTime[i] = 0;
+ }
+ for (int i = 0; i < NUM_STRATEGIES; i++) {
+ mStrategyMutedByDevice[i] = false;
+ }
+ if (profile != NULL) {
+ mFlags = (audio_output_flags_t)profile->mFlags;
+ mSamplingRate = profile->pickSamplingRate();
+ mFormat = profile->pickFormat();
+ mChannelMask = profile->pickChannelMask();
+ if (profile->mGains.size() > 0) {
+ profile->mGains[0]->getDefaultConfig(&mGain);
+ }
+ }
+}
+
+audio_devices_t AudioOutputDescriptor::device() const
+{
+ if (isDuplicated()) {
+ return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice);
+ } else {
+ return mDevice;
+ }
+}
+
+uint32_t AudioOutputDescriptor::latency()
+{
+ if (isDuplicated()) {
+ return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency;
+ } else {
+ return mLatency;
+ }
+}
+
+bool AudioOutputDescriptor::sharesHwModuleWith(
+ const sp<AudioOutputDescriptor> outputDesc)
+{
+ if (isDuplicated()) {
+ return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc);
+ } else if (outputDesc->isDuplicated()){
+ return sharesHwModuleWith(outputDesc->mOutput1) || sharesHwModuleWith(outputDesc->mOutput2);
+ } else {
+ return (mProfile->mModule == outputDesc->mProfile->mModule);
+ }
+}
+
+void AudioOutputDescriptor::changeRefCount(audio_stream_type_t stream,
+ int delta)
+{
+ // forward usage count change to attached outputs
+ if (isDuplicated()) {
+ mOutput1->changeRefCount(stream, delta);
+ mOutput2->changeRefCount(stream, delta);
+ }
+ if ((delta + (int)mRefCount[stream]) < 0) {
+ ALOGW("changeRefCount() invalid delta %d for stream %d, refCount %d",
+ delta, stream, mRefCount[stream]);
+ mRefCount[stream] = 0;
+ return;
+ }
+ mRefCount[stream] += delta;
+ ALOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]);
+}
+
+audio_devices_t AudioOutputDescriptor::supportedDevices()
+{
+ if (isDuplicated()) {
+ return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices());
+ } else {
+ return mProfile->mSupportedDevices.types() ;
+ }
+}
+
+bool AudioOutputDescriptor::isActive(uint32_t inPastMs) const
+{
+ return isStrategyActive(NUM_STRATEGIES, inPastMs);
+}
+
+bool AudioOutputDescriptor::isStrategyActive(routing_strategy strategy,
+ uint32_t inPastMs,
+ nsecs_t sysTime) const
+{
+ if ((sysTime == 0) && (inPastMs != 0)) {
+ sysTime = systemTime();
+ }
+ for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
+ if (i == AUDIO_STREAM_PATCH) {
+ continue;
+ }
+ if (((AudioPolicyManager::getStrategy((audio_stream_type_t)i) == strategy) ||
+ (NUM_STRATEGIES == strategy)) &&
+ isStreamActive((audio_stream_type_t)i, inPastMs, sysTime)) {
+ return true;
+ }
+ }
+ return false;
+}
+
+bool AudioOutputDescriptor::isStreamActive(audio_stream_type_t stream,
+ uint32_t inPastMs,
+ nsecs_t sysTime) const
+{
+ if (mRefCount[stream] != 0) {
+ return true;
+ }
+ if (inPastMs == 0) {
+ return false;
+ }
+ if (sysTime == 0) {
+ sysTime = systemTime();
+ }
+ if (ns2ms(sysTime - mStopTime[stream]) < inPastMs) {
+ return true;
+ }
+ return false;
+}
+
+void AudioOutputDescriptor::toAudioPortConfig(
+ struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig) const
+{
+ ALOG_ASSERT(!isDuplicated(), "toAudioPortConfig() called on duplicated output %d", mIoHandle);
+
+ dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
+ AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN;
+ if (srcConfig != NULL) {
+ dstConfig->config_mask |= srcConfig->config_mask;
+ }
+ AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
+
+ dstConfig->id = mId;
+ dstConfig->role = AUDIO_PORT_ROLE_SOURCE;
+ dstConfig->type = AUDIO_PORT_TYPE_MIX;
+ dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle;
+ dstConfig->ext.mix.handle = mIoHandle;
+ dstConfig->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT;
+}
+
+void AudioOutputDescriptor::toAudioPort(
+ struct audio_port *port) const
+{
+ ALOG_ASSERT(!isDuplicated(), "toAudioPort() called on duplicated output %d", mIoHandle);
+ mProfile->toAudioPort(port);
+ port->id = mId;
+ toAudioPortConfig(&port->active_config);
+ port->ext.mix.hw_module = mProfile->mModule->mHandle;
+ port->ext.mix.handle = mIoHandle;
+ port->ext.mix.latency_class =
+ mFlags & AUDIO_OUTPUT_FLAG_FAST ? AUDIO_LATENCY_LOW : AUDIO_LATENCY_NORMAL;
+}
+
+status_t AudioOutputDescriptor::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, " ID: %d\n", mId);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Format: %08x\n", mFormat);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Latency: %d\n", mLatency);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Flags %08x\n", mFlags);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Devices %08x\n", device());
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Stream volume refCount muteCount\n");
+ result.append(buffer);
+ for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
+ snprintf(buffer, SIZE, " %02d %.03f %02d %02d\n",
+ i, mCurVolume[i], mRefCount[i], mMuteCount[i]);
+ result.append(buffer);
+ }
+ write(fd, result.string(), result.size());
+
+ return NO_ERROR;
+}
+
+
+
+}; //namespace android
diff --git a/services/audiopolicy/managerdefault/AudioOutputDescriptor.h b/services/audiopolicy/managerdefault/AudioOutputDescriptor.h
new file mode 100644
index 0000000..32f46e4
--- /dev/null
+++ b/services/audiopolicy/managerdefault/AudioOutputDescriptor.h
@@ -0,0 +1,69 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include "ApmImplDefinitions.h"
+
+namespace android {
+
+// descriptor for audio outputs. Used to maintain current configuration of each opened audio output
+// and keep track of the usage of this output by each audio stream type.
+class AudioOutputDescriptor: public AudioPortConfig
+{
+public:
+ AudioOutputDescriptor(const sp<IOProfile>& profile);
+
+ status_t dump(int fd);
+
+ audio_devices_t device() const;
+ void changeRefCount(audio_stream_type_t stream, int delta);
+
+ bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); }
+ audio_devices_t supportedDevices();
+ uint32_t latency();
+ bool sharesHwModuleWith(const sp<AudioOutputDescriptor> outputDesc);
+ bool isActive(uint32_t inPastMs = 0) const;
+ bool isStreamActive(audio_stream_type_t stream,
+ uint32_t inPastMs = 0,
+ nsecs_t sysTime = 0) const;
+ bool isStrategyActive(routing_strategy strategy,
+ uint32_t inPastMs = 0,
+ nsecs_t sysTime = 0) const;
+
+ virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig = NULL) const;
+ virtual sp<AudioPort> getAudioPort() const { return mProfile; }
+ void toAudioPort(struct audio_port *port) const;
+
+ audio_port_handle_t mId;
+ audio_io_handle_t mIoHandle; // output handle
+ uint32_t mLatency; //
+ audio_output_flags_t mFlags; //
+ audio_devices_t mDevice; // current device this output is routed to
+ AudioMix *mPolicyMix; // non NULL when used by a dynamic policy
+ audio_patch_handle_t mPatchHandle;
+ uint32_t mRefCount[AUDIO_STREAM_CNT]; // number of streams of each type using this output
+ nsecs_t mStopTime[AUDIO_STREAM_CNT];
+ sp<AudioOutputDescriptor> mOutput1; // used by duplicated outputs: first output
+ sp<AudioOutputDescriptor> mOutput2; // used by duplicated outputs: second output
+ float mCurVolume[AUDIO_STREAM_CNT]; // current stream volume
+ int mMuteCount[AUDIO_STREAM_CNT]; // mute request counter
+ const sp<IOProfile> mProfile; // I/O profile this output derives from
+ bool mStrategyMutedByDevice[NUM_STRATEGIES]; // strategies muted because of incompatible
+ // device selection. See checkDeviceMuteStrategies()
+ uint32_t mDirectOpenCount; // number of clients using this output (direct outputs only)
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
similarity index 72%
rename from services/audiopolicy/AudioPolicyManager.cpp
rename to services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 7f27659..b48dc80 100644
--- a/services/audiopolicy/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -14,7 +14,7 @@
* limitations under the License.
*/
-#define LOG_TAG "AudioPolicyManager"
+#define LOG_TAG "APM::AudioPolicyManager"
//#define LOG_NDEBUG 0
//#define VERY_VERBOSE_LOGGING
@@ -51,184 +51,29 @@
namespace android {
// ----------------------------------------------------------------------------
-// Definitions for audio_policy.conf file parsing
-// ----------------------------------------------------------------------------
-
-struct StringToEnum {
- const char *name;
- uint32_t value;
-};
-
-#define STRING_TO_ENUM(string) { #string, string }
-#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
-
-const StringToEnum sDeviceNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_EARPIECE),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER_SAFE),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADPHONE),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_SCO),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_A2DP),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_DIGITAL),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_ACCESSORY),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_DEVICE),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_USB),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_REMOTE_SUBMIX),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_TELEPHONY_TX),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_LINE),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI_ARC),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPDIF),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_FM),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_LINE),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_AMBIENT),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_BUILTIN_MIC),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_ALL_SCO),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_WIRED_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_AUX_DIGITAL),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_HDMI),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_TELEPHONY_RX),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_VOICE_CALL),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_BACK_MIC),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_REMOTE_SUBMIX),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_ACCESSORY),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_DEVICE),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_FM_TUNER),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_TV_TUNER),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_LINE),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_SPDIF),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_A2DP),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_LOOPBACK),
-};
-
-const StringToEnum sOutputFlagNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DIRECT),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_PRIMARY),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_FAST),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DEEP_BUFFER),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_NON_BLOCKING),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_HW_AV_SYNC),
-};
-
-const StringToEnum sInputFlagNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_INPUT_FLAG_FAST),
- STRING_TO_ENUM(AUDIO_INPUT_FLAG_HW_HOTWORD),
-};
-
-const StringToEnum sFormatNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_BIT),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_32_BIT),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_24_BIT),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_FLOAT),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_PACKED),
- STRING_TO_ENUM(AUDIO_FORMAT_MP3),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_MAIN),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_LC),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_SSR),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_LTP),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V1),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_SCALABLE),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_ERLC),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_LD),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V2),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_ELD),
- STRING_TO_ENUM(AUDIO_FORMAT_VORBIS),
- STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V1),
- STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V2),
- STRING_TO_ENUM(AUDIO_FORMAT_OPUS),
- STRING_TO_ENUM(AUDIO_FORMAT_AC3),
- STRING_TO_ENUM(AUDIO_FORMAT_E_AC3),
-};
-
-const StringToEnum sOutChannelsNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_CHANNEL_OUT_MONO),
- STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO),
- STRING_TO_ENUM(AUDIO_CHANNEL_OUT_QUAD),
- STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1),
- STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1),
-};
-
-const StringToEnum sInChannelsNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_CHANNEL_IN_MONO),
- STRING_TO_ENUM(AUDIO_CHANNEL_IN_STEREO),
- STRING_TO_ENUM(AUDIO_CHANNEL_IN_FRONT_BACK),
-};
-
-const StringToEnum sGainModeNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_GAIN_MODE_JOINT),
- STRING_TO_ENUM(AUDIO_GAIN_MODE_CHANNELS),
- STRING_TO_ENUM(AUDIO_GAIN_MODE_RAMP),
-};
-
-
-uint32_t AudioPolicyManager::stringToEnum(const struct StringToEnum *table,
- size_t size,
- const char *name)
-{
- for (size_t i = 0; i < size; i++) {
- if (strcmp(table[i].name, name) == 0) {
- ALOGV("stringToEnum() found %s", table[i].name);
- return table[i].value;
- }
- }
- return 0;
-}
-
-const char *AudioPolicyManager::enumToString(const struct StringToEnum *table,
- size_t size,
- uint32_t value)
-{
- for (size_t i = 0; i < size; i++) {
- if (table[i].value == value) {
- return table[i].name;
- }
- }
- return "";
-}
-
-bool AudioPolicyManager::stringToBool(const char *value)
-{
- return ((strcasecmp("true", value) == 0) || (strcmp("1", value) == 0));
-}
-
-
-// ----------------------------------------------------------------------------
// AudioPolicyInterface implementation
// ----------------------------------------------------------------------------
status_t AudioPolicyManager::setDeviceConnectionState(audio_devices_t device,
- audio_policy_dev_state_t state,
- const char *device_address)
+ audio_policy_dev_state_t state,
+ const char *device_address,
+ const char *device_name)
{
- return setDeviceConnectionStateInt(device, state, device_address);
+ return setDeviceConnectionStateInt(device, state, device_address, device_name);
}
status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device,
audio_policy_dev_state_t state,
- const char *device_address)
+ const char *device_address,
+ const char *device_name)
{
- ALOGV("setDeviceConnectionState() device: %x, state %d, address %s",
- device, state, device_address != NULL ? device_address : "");
+ ALOGV("setDeviceConnectionStateInt() device: 0x%X, state %d, address %s name %s",
+- device, state, device_address, device_name);
// connect/disconnect only 1 device at a time
if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE;
- sp<DeviceDescriptor> devDesc = getDeviceDescriptor(device, device_address);
+ sp<DeviceDescriptor> devDesc = getDeviceDescriptor(device, device_address, device_name);
// handle output devices
if (audio_is_output_device(device)) {
@@ -259,8 +104,7 @@
mAvailableOutputDevices.remove(devDesc);
return INVALID_OPERATION;
}
- mAvailableOutputDevices[index]->mId = nextUniqueId();
- mAvailableOutputDevices[index]->mModule = module;
+ mAvailableOutputDevices[index]->attach(module);
} else {
return NO_MEMORY;
}
@@ -275,8 +119,7 @@
ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %zu outputs",
outputs.size());
-
- // Set connect to HALs
+ // Send connect to HALs
AudioParameter param = AudioParameter(devDesc->mAddress);
param.addInt(String8(AUDIO_PARAMETER_DEVICE_CONNECT), device);
mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
@@ -291,7 +134,7 @@
ALOGV("setDeviceConnectionState() disconnecting output device %x", device);
- // Set Disconnect to HALs
+ // Send Disconnect to HALs
AudioParameter param = AudioParameter(devDesc->mAddress);
param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device);
mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
@@ -377,8 +220,7 @@
index = mAvailableInputDevices.add(devDesc);
if (index >= 0) {
- mAvailableInputDevices[index]->mId = nextUniqueId();
- mAvailableInputDevices[index]->mModule = module;
+ mAvailableInputDevices[index]->attach(module);
} else {
return NO_MEMORY;
}
@@ -432,7 +274,7 @@
audio_policy_dev_state_t AudioPolicyManager::getDeviceConnectionState(audio_devices_t device,
const char *device_address)
{
- sp<DeviceDescriptor> devDesc = getDeviceDescriptor(device, device_address);
+ sp<DeviceDescriptor> devDesc = getDeviceDescriptor(device, device_address, "");
DeviceVector *deviceVector;
if (audio_is_output_device(device)) {
@@ -452,9 +294,9 @@
}
}
-sp<AudioPolicyManager::DeviceDescriptor> AudioPolicyManager::getDeviceDescriptor(
- const audio_devices_t device,
- const char *device_address)
+sp<DeviceDescriptor> AudioPolicyManager::getDeviceDescriptor(const audio_devices_t device,
+ const char *device_address,
+ const char *device_name)
{
String8 address = (device_address == NULL) ? String8("") : String8(device_address);
// handle legacy remote submix case where the address was not always specified
@@ -477,7 +319,8 @@
}
}
- sp<DeviceDescriptor> devDesc = new DeviceDescriptor(String8(""), device);
+ sp<DeviceDescriptor> devDesc =
+ new DeviceDescriptor(String8(device_name != NULL ? device_name : ""), device);
devDesc->mAddress = address;
return devDesc;
}
@@ -640,18 +483,18 @@
// force routing command to audio hardware when starting a call
// even if no device change is needed
force = true;
- for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
+ for (int j = 0; j < ApmGains::DEVICE_CATEGORY_CNT; j++) {
mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] =
- sVolumeProfiles[AUDIO_STREAM_VOICE_CALL][j];
+ ApmGains::sVolumeProfiles[AUDIO_STREAM_VOICE_CALL][j];
}
} else if (isStateInCall(oldState) && !isStateInCall(state)) {
ALOGV(" Exiting call in setPhoneState()");
// force routing command to audio hardware when exiting a call
// even if no device change is needed
force = true;
- for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
+ for (int j = 0; j < ApmGains::DEVICE_CATEGORY_CNT; j++) {
mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] =
- sVolumeProfiles[AUDIO_STREAM_DTMF][j];
+ ApmGains::sVolumeProfiles[AUDIO_STREAM_DTMF][j];
}
} else if (isStateInCall(state) && (state != oldState)) {
ALOGV(" Switching between telephony and VoIP in setPhoneState()");
@@ -842,7 +685,7 @@
// Find a direct output profile compatible with the parameters passed, even if the input flags do
// not explicitly request a direct output
-sp<AudioPolicyManager::IOProfile> AudioPolicyManager::getProfileForDirectOutput(
+sp<IOProfile> AudioPolicyManager::getProfileForDirectOutput(
audio_devices_t device,
uint32_t samplingRate,
audio_format_t format,
@@ -1130,6 +973,10 @@
if (audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE) {
goto non_direct_output;
}
+ // fall back to mixer output if possible when the direct output could not be open
+ if (audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE) {
+ goto non_direct_output;
+ }
return AUDIO_IO_HANDLE_NONE;
}
outputDesc->mSamplingRate = config.sample_rate;
@@ -1322,7 +1169,8 @@
outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) {
setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
- outputDesc->mPolicyMix->mRegistrationId);
+ outputDesc->mPolicyMix->mRegistrationId,
+ "remote-submix");
}
// force reevaluating accessibility routing when ringtone or alarm starts
@@ -1371,7 +1219,8 @@
outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) {
setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
- outputDesc->mPolicyMix->mRegistrationId);
+ outputDesc->mPolicyMix->mRegistrationId,
+ "remote-submix");
}
outputDesc->mStopTime[stream] = systemTime();
@@ -1672,7 +1521,7 @@
if (address != "") {
setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
- address);
+ address, "remote-submix");
}
}
}
@@ -1720,7 +1569,7 @@
if (address != "") {
setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
- address);
+ address, "remote-submix");
}
}
@@ -1849,7 +1698,7 @@
status_t status = NO_ERROR;
for (size_t i = 0; i < mOutputs.size(); i++) {
audio_devices_t curDevice =
- getDeviceForVolume(mOutputs.valueAt(i)->device());
+ ApmGains::getDeviceForVolume(mOutputs.valueAt(i)->device());
if ((device == AUDIO_DEVICE_OUT_DEFAULT) || ((curDevice & strategyDevice) != 0)) {
status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), curDevice);
if (volStatus != NO_ERROR) {
@@ -1879,7 +1728,7 @@
if (device == AUDIO_DEVICE_OUT_DEFAULT) {
device = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/);
}
- device = getDeviceForVolume(device);
+ device = ApmGains::getDeviceForVolume(device);
*index = mStreams[stream].getVolumeIndex(device);
ALOGV("getStreamVolumeIndex() stream %d device %08x index %d", stream, device, *index);
@@ -2177,11 +2026,11 @@
if (mixes[i].mMixType == MIX_TYPE_PLAYERS) {
setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
- address.string());
+ address.string(), "remote-submix");
} else {
setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
- address.string());
+ address.string(), "remote-submix");
}
}
return NO_ERROR;
@@ -2219,7 +2068,7 @@
{
setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
- address.string());
+ address.string(), "remote-submix");
}
if (getDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address.string()) ==
@@ -2227,7 +2076,7 @@
{
setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
- address.string());
+ address.string(), "remote-submix");
}
module->removeOutputProfile(address);
module->removeInputProfile(address);
@@ -2463,7 +2312,7 @@
return NO_ERROR;
}
-sp<AudioPolicyManager::AudioOutputDescriptor> AudioPolicyManager::getOutputFromId(
+sp<AudioOutputDescriptor> AudioPolicyManager::getOutputFromId(
audio_port_handle_t id) const
{
sp<AudioOutputDescriptor> outputDesc = NULL;
@@ -2476,7 +2325,7 @@
return outputDesc;
}
-sp<AudioPolicyManager::AudioInputDescriptor> AudioPolicyManager::getInputFromId(
+sp<AudioInputDescriptor> AudioPolicyManager::getInputFromId(
audio_port_handle_t id) const
{
sp<AudioInputDescriptor> inputDesc = NULL;
@@ -2489,7 +2338,7 @@
return inputDesc;
}
-sp <AudioPolicyManager::HwModule> AudioPolicyManager::getModuleForDevice(
+sp <HwModule> AudioPolicyManager::getModuleForDevice(
audio_devices_t device) const
{
sp <HwModule> module;
@@ -2517,7 +2366,7 @@
return module;
}
-sp <AudioPolicyManager::HwModule> AudioPolicyManager::getModuleFromName(const char *name) const
+sp <HwModule> AudioPolicyManager::getModuleFromName(const char *name) const
{
sp <HwModule> module;
@@ -3042,6 +2891,8 @@
return android_atomic_inc(&mAudioPortGeneration);
}
+int32_t volatile AudioPolicyManager::mNextUniqueId = 1;
+
AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterface)
:
#ifdef AUDIO_POLICY_TEST
@@ -3052,7 +2903,7 @@
mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f),
mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0),
mA2dpSuspended(false),
- mSpeakerDrcEnabled(false), mNextUniqueId(1),
+ mSpeakerDrcEnabled(false),
mAudioPortGeneration(1),
mBeaconMuteRefCount(0),
mBeaconPlayingRefCount(0),
@@ -3065,7 +2916,7 @@
mForceUse[i] = AUDIO_POLICY_FORCE_NONE;
}
- mDefaultOutputDevice = new DeviceDescriptor(String8(""), AUDIO_DEVICE_OUT_SPEAKER);
+ mDefaultOutputDevice = new DeviceDescriptor(String8("Speaker"), AUDIO_DEVICE_OUT_SPEAKER);
if (loadAudioPolicyConfig(AUDIO_POLICY_VENDOR_CONFIG_FILE) != NO_ERROR) {
if (loadAudioPolicyConfig(AUDIO_POLICY_CONFIG_FILE) != NO_ERROR) {
ALOGE("could not load audio policy configuration file, setting defaults");
@@ -3148,9 +2999,8 @@
ssize_t index =
mAvailableOutputDevices.indexOf(outProfile->mSupportedDevices[k]);
// give a valid ID to an attached device once confirmed it is reachable
- if ((index >= 0) && (mAvailableOutputDevices[index]->mId == 0)) {
- mAvailableOutputDevices[index]->mId = nextUniqueId();
- mAvailableOutputDevices[index]->mModule = mHwModules[i];
+ if (index >= 0 && !mAvailableOutputDevices[index]->isAttached()) {
+ mAvailableOutputDevices[index]->attach(mHwModules[i]);
}
}
if (mPrimaryOutput == 0 &&
@@ -3217,9 +3067,8 @@
ssize_t index =
mAvailableInputDevices.indexOf(inProfile->mSupportedDevices[k]);
// give a valid ID to an attached device once confirmed it is reachable
- if ((index >= 0) && (mAvailableInputDevices[index]->mId == 0)) {
- mAvailableInputDevices[index]->mId = nextUniqueId();
- mAvailableInputDevices[index]->mModule = mHwModules[i];
+ if (index >= 0 && !mAvailableInputDevices[index]->isAttached()) {
+ mAvailableInputDevices[index]->attach(mHwModules[i]);
}
}
mpClientInterface->closeInput(input);
@@ -3232,7 +3081,7 @@
}
// make sure all attached devices have been allocated a unique ID
for (size_t i = 0; i < mAvailableOutputDevices.size();) {
- if (mAvailableOutputDevices[i]->mId == 0) {
+ if (!mAvailableOutputDevices[i]->isAttached()) {
ALOGW("Input device %08x unreachable", mAvailableOutputDevices[i]->mDeviceType);
mAvailableOutputDevices.remove(mAvailableOutputDevices[i]);
continue;
@@ -3240,7 +3089,7 @@
i++;
}
for (size_t i = 0; i < mAvailableInputDevices.size();) {
- if (mAvailableInputDevices[i]->mId == 0) {
+ if (!mAvailableInputDevices[i]->isAttached()) {
ALOGW("Input device %08x unreachable", mAvailableInputDevices[i]->mDeviceType);
mAvailableInputDevices.remove(mAvailableInputDevices[i]);
continue;
@@ -4328,7 +4177,7 @@
return AUDIO_DEVICE_NONE;
}
audio_devices_t devices;
- AudioPolicyManager::routing_strategy strategy = getStrategy(stream);
+ routing_strategy strategy = getStrategy(stream);
devices = getDeviceForStrategy(strategy, true /*fromCache*/);
SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(devices, mOutputs);
for (size_t i = 0; i < outputs.size(); i++) {
@@ -4349,7 +4198,7 @@
return devices;
}
-AudioPolicyManager::routing_strategy AudioPolicyManager::getStrategy(
+routing_strategy AudioPolicyManager::getStrategy(
audio_stream_type_t stream) {
ALOG_ASSERT(stream != AUDIO_STREAM_PATCH,"getStrategy() called for AUDIO_STREAM_PATCH");
@@ -5128,7 +4977,7 @@
return status;
}
-sp<AudioPolicyManager::IOProfile> AudioPolicyManager::getInputProfile(audio_devices_t device,
+sp<IOProfile> AudioPolicyManager::getInputProfile(audio_devices_t device,
String8 address,
uint32_t& samplingRate,
audio_format_t format,
@@ -5338,305 +5187,29 @@
}
-audio_devices_t AudioPolicyManager::getDeviceForVolume(audio_devices_t device)
-{
- if (device == AUDIO_DEVICE_NONE) {
- // this happens when forcing a route update and no track is active on an output.
- // In this case the returned category is not important.
- device = AUDIO_DEVICE_OUT_SPEAKER;
- } else if (popcount(device) > 1) {
- // Multiple device selection is either:
- // - speaker + one other device: give priority to speaker in this case.
- // - one A2DP device + another device: happens with duplicated output. In this case
- // retain the device on the A2DP output as the other must not correspond to an active
- // selection if not the speaker.
- // - HDMI-CEC system audio mode only output: give priority to available item in order.
- if (device & AUDIO_DEVICE_OUT_SPEAKER) {
- device = AUDIO_DEVICE_OUT_SPEAKER;
- } else if (device & AUDIO_DEVICE_OUT_HDMI_ARC) {
- device = AUDIO_DEVICE_OUT_HDMI_ARC;
- } else if (device & AUDIO_DEVICE_OUT_AUX_LINE) {
- device = AUDIO_DEVICE_OUT_AUX_LINE;
- } else if (device & AUDIO_DEVICE_OUT_SPDIF) {
- device = AUDIO_DEVICE_OUT_SPDIF;
- } else {
- device = (audio_devices_t)(device & AUDIO_DEVICE_OUT_ALL_A2DP);
- }
- }
-
- /*SPEAKER_SAFE is an alias of SPEAKER for purposes of volume control*/
- if (device == AUDIO_DEVICE_OUT_SPEAKER_SAFE)
- device = AUDIO_DEVICE_OUT_SPEAKER;
-
- ALOGW_IF(popcount(device) != 1,
- "getDeviceForVolume() invalid device combination: %08x",
- device);
-
- return device;
-}
-
-AudioPolicyManager::device_category AudioPolicyManager::getDeviceCategory(audio_devices_t device)
-{
- switch(getDeviceForVolume(device)) {
- case AUDIO_DEVICE_OUT_EARPIECE:
- return DEVICE_CATEGORY_EARPIECE;
- case AUDIO_DEVICE_OUT_WIRED_HEADSET:
- case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
- case AUDIO_DEVICE_OUT_BLUETOOTH_SCO:
- case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
- case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP:
- case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES:
- return DEVICE_CATEGORY_HEADSET;
- case AUDIO_DEVICE_OUT_LINE:
- case AUDIO_DEVICE_OUT_AUX_DIGITAL:
- /*USB? Remote submix?*/
- return DEVICE_CATEGORY_EXT_MEDIA;
- case AUDIO_DEVICE_OUT_SPEAKER:
- case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
- case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER:
- case AUDIO_DEVICE_OUT_USB_ACCESSORY:
- case AUDIO_DEVICE_OUT_USB_DEVICE:
- case AUDIO_DEVICE_OUT_REMOTE_SUBMIX:
- default:
- return DEVICE_CATEGORY_SPEAKER;
- }
-}
-
-/* static */
-float AudioPolicyManager::volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
- int indexInUi)
-{
- device_category deviceCategory = getDeviceCategory(device);
- const VolumeCurvePoint *curve = streamDesc.mVolumeCurve[deviceCategory];
-
- // the volume index in the UI is relative to the min and max volume indices for this stream type
- int nbSteps = 1 + curve[VOLMAX].mIndex -
- curve[VOLMIN].mIndex;
- int volIdx = (nbSteps * (indexInUi - streamDesc.mIndexMin)) /
- (streamDesc.mIndexMax - streamDesc.mIndexMin);
-
- // find what part of the curve this index volume belongs to, or if it's out of bounds
- int segment = 0;
- if (volIdx < curve[VOLMIN].mIndex) { // out of bounds
- return 0.0f;
- } else if (volIdx < curve[VOLKNEE1].mIndex) {
- segment = 0;
- } else if (volIdx < curve[VOLKNEE2].mIndex) {
- segment = 1;
- } else if (volIdx <= curve[VOLMAX].mIndex) {
- segment = 2;
- } else { // out of bounds
- return 1.0f;
- }
-
- // linear interpolation in the attenuation table in dB
- float decibels = curve[segment].mDBAttenuation +
- ((float)(volIdx - curve[segment].mIndex)) *
- ( (curve[segment+1].mDBAttenuation -
- curve[segment].mDBAttenuation) /
- ((float)(curve[segment+1].mIndex -
- curve[segment].mIndex)) );
-
- float amplification = exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 )
-
- ALOGVV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f] ampl=%.5f",
- curve[segment].mIndex, volIdx,
- curve[segment+1].mIndex,
- curve[segment].mDBAttenuation,
- decibels,
- curve[segment+1].mDBAttenuation,
- amplification);
-
- return amplification;
-}
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sDefaultVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {1, -49.5f}, {33, -33.5f}, {66, -17.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sDefaultMediaVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {1, -58.0f}, {20, -40.0f}, {60, -17.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sExtMediaSystemVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {1, -58.0f}, {20, -40.0f}, {60, -21.0f}, {100, -10.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sSpeakerMediaVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {1, -56.0f}, {20, -34.0f}, {60, -11.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sSpeakerMediaVolumeCurveDrc[AudioPolicyManager::VOLCNT] = {
- {1, -55.0f}, {20, -43.0f}, {86, -12.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sSpeakerSonificationVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {1, -29.7f}, {33, -20.1f}, {66, -10.2f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sSpeakerSonificationVolumeCurveDrc[AudioPolicyManager::VOLCNT] = {
- {1, -35.7f}, {33, -26.1f}, {66, -13.2f}, {100, 0.0f}
-};
-
-// AUDIO_STREAM_SYSTEM, AUDIO_STREAM_ENFORCED_AUDIBLE and AUDIO_STREAM_DTMF volume tracks
-// AUDIO_STREAM_RING on phones and AUDIO_STREAM_MUSIC on tablets.
-// AUDIO_STREAM_DTMF tracks AUDIO_STREAM_VOICE_CALL while in call (See AudioService.java).
-// The range is constrained between -24dB and -6dB over speaker and -30dB and -18dB over headset.
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sDefaultSystemVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {1, -24.0f}, {33, -18.0f}, {66, -12.0f}, {100, -6.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sDefaultSystemVolumeCurveDrc[AudioPolicyManager::VOLCNT] = {
- {1, -34.0f}, {33, -24.0f}, {66, -15.0f}, {100, -6.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sHeadsetSystemVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {1, -30.0f}, {33, -26.0f}, {66, -22.0f}, {100, -18.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sDefaultVoiceVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {0, -42.0f}, {33, -28.0f}, {66, -14.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sSpeakerVoiceVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {0, -24.0f}, {33, -16.0f}, {66, -8.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sLinearVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {0, -96.0f}, {33, -68.0f}, {66, -34.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sSilentVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {0, -96.0f}, {1, -96.0f}, {2, -96.0f}, {100, -96.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sFullScaleVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {0, 0.0f}, {1, 0.0f}, {2, 0.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- *AudioPolicyManager::sVolumeProfiles[AUDIO_STREAM_CNT]
- [AudioPolicyManager::DEVICE_CATEGORY_CNT] = {
- { // AUDIO_STREAM_VOICE_CALL
- sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_SYSTEM
- sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_RING
- sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_MUSIC
- sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_ALARM
- sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_NOTIFICATION
- sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_BLUETOOTH_SCO
- sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_ENFORCED_AUDIBLE
- sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_DTMF
- sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_TTS
- // "Transmitted Through Speaker": always silent except on DEVICE_CATEGORY_SPEAKER
- sSilentVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sLinearVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sSilentVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sSilentVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_ACCESSIBILITY
- sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_REROUTING
- sFullScaleVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sFullScaleVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sFullScaleVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sFullScaleVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_PATCH
- sFullScaleVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sFullScaleVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sFullScaleVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sFullScaleVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
-};
-
void AudioPolicyManager::initializeVolumeCurves()
{
for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
- for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
+ for (int j = 0; j < ApmGains::DEVICE_CATEGORY_CNT; j++) {
mStreams[i].mVolumeCurve[j] =
- sVolumeProfiles[i][j];
+ ApmGains::sVolumeProfiles[i][j];
}
}
// Check availability of DRC on speaker path: if available, override some of the speaker curves
if (mSpeakerDrcEnabled) {
- mStreams[AUDIO_STREAM_SYSTEM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
- sDefaultSystemVolumeCurveDrc;
- mStreams[AUDIO_STREAM_RING].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
- sSpeakerSonificationVolumeCurveDrc;
- mStreams[AUDIO_STREAM_ALARM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
- sSpeakerSonificationVolumeCurveDrc;
- mStreams[AUDIO_STREAM_NOTIFICATION].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
- sSpeakerSonificationVolumeCurveDrc;
- mStreams[AUDIO_STREAM_MUSIC].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
- sSpeakerMediaVolumeCurveDrc;
- mStreams[AUDIO_STREAM_ACCESSIBILITY].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
- sSpeakerMediaVolumeCurveDrc;
+ mStreams[AUDIO_STREAM_SYSTEM].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] =
+ ApmGains::sDefaultSystemVolumeCurveDrc;
+ mStreams[AUDIO_STREAM_RING].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] =
+ ApmGains::sSpeakerSonificationVolumeCurveDrc;
+ mStreams[AUDIO_STREAM_ALARM].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] =
+ ApmGains::sSpeakerSonificationVolumeCurveDrc;
+ mStreams[AUDIO_STREAM_NOTIFICATION].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] =
+ ApmGains::sSpeakerSonificationVolumeCurveDrc;
+ mStreams[AUDIO_STREAM_MUSIC].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] =
+ ApmGains::sSpeakerMediaVolumeCurveDrc;
+ mStreams[AUDIO_STREAM_ACCESSIBILITY].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] =
+ ApmGains::sSpeakerMediaVolumeCurveDrc;
}
}
@@ -5653,7 +5226,7 @@
device = outputDesc->device();
}
- volume = volIndexToAmpl(device, streamDesc, index);
+ volume = ApmGains::volIndexToAmpl(device, streamDesc, index);
// if a headset is connected, apply the following rules to ring tones and notifications
// to avoid sound level bursts in user's ears:
@@ -5909,319 +5482,6 @@
}
-// --- AudioOutputDescriptor class implementation
-
-AudioPolicyManager::AudioOutputDescriptor::AudioOutputDescriptor(
- const sp<IOProfile>& profile)
- : mId(0), mIoHandle(0), mLatency(0),
- mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL),
- mPatchHandle(0),
- mOutput1(0), mOutput2(0), mProfile(profile), mDirectOpenCount(0)
-{
- // clear usage count for all stream types
- for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
- mRefCount[i] = 0;
- mCurVolume[i] = -1.0;
- mMuteCount[i] = 0;
- mStopTime[i] = 0;
- }
- for (int i = 0; i < NUM_STRATEGIES; i++) {
- mStrategyMutedByDevice[i] = false;
- }
- if (profile != NULL) {
- mFlags = (audio_output_flags_t)profile->mFlags;
- mSamplingRate = profile->pickSamplingRate();
- mFormat = profile->pickFormat();
- mChannelMask = profile->pickChannelMask();
- if (profile->mGains.size() > 0) {
- profile->mGains[0]->getDefaultConfig(&mGain);
- }
- }
-}
-
-audio_devices_t AudioPolicyManager::AudioOutputDescriptor::device() const
-{
- if (isDuplicated()) {
- return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice);
- } else {
- return mDevice;
- }
-}
-
-uint32_t AudioPolicyManager::AudioOutputDescriptor::latency()
-{
- if (isDuplicated()) {
- return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency;
- } else {
- return mLatency;
- }
-}
-
-bool AudioPolicyManager::AudioOutputDescriptor::sharesHwModuleWith(
- const sp<AudioOutputDescriptor> outputDesc)
-{
- if (isDuplicated()) {
- return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc);
- } else if (outputDesc->isDuplicated()){
- return sharesHwModuleWith(outputDesc->mOutput1) || sharesHwModuleWith(outputDesc->mOutput2);
- } else {
- return (mProfile->mModule == outputDesc->mProfile->mModule);
- }
-}
-
-void AudioPolicyManager::AudioOutputDescriptor::changeRefCount(audio_stream_type_t stream,
- int delta)
-{
- // forward usage count change to attached outputs
- if (isDuplicated()) {
- mOutput1->changeRefCount(stream, delta);
- mOutput2->changeRefCount(stream, delta);
- }
- if ((delta + (int)mRefCount[stream]) < 0) {
- ALOGW("changeRefCount() invalid delta %d for stream %d, refCount %d",
- delta, stream, mRefCount[stream]);
- mRefCount[stream] = 0;
- return;
- }
- mRefCount[stream] += delta;
- ALOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]);
-}
-
-audio_devices_t AudioPolicyManager::AudioOutputDescriptor::supportedDevices()
-{
- if (isDuplicated()) {
- return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices());
- } else {
- return mProfile->mSupportedDevices.types() ;
- }
-}
-
-bool AudioPolicyManager::AudioOutputDescriptor::isActive(uint32_t inPastMs) const
-{
- return isStrategyActive(NUM_STRATEGIES, inPastMs);
-}
-
-bool AudioPolicyManager::AudioOutputDescriptor::isStrategyActive(routing_strategy strategy,
- uint32_t inPastMs,
- nsecs_t sysTime) const
-{
- if ((sysTime == 0) && (inPastMs != 0)) {
- sysTime = systemTime();
- }
- for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
- if (i == AUDIO_STREAM_PATCH) {
- continue;
- }
- if (((getStrategy((audio_stream_type_t)i) == strategy) ||
- (NUM_STRATEGIES == strategy)) &&
- isStreamActive((audio_stream_type_t)i, inPastMs, sysTime)) {
- return true;
- }
- }
- return false;
-}
-
-bool AudioPolicyManager::AudioOutputDescriptor::isStreamActive(audio_stream_type_t stream,
- uint32_t inPastMs,
- nsecs_t sysTime) const
-{
- if (mRefCount[stream] != 0) {
- return true;
- }
- if (inPastMs == 0) {
- return false;
- }
- if (sysTime == 0) {
- sysTime = systemTime();
- }
- if (ns2ms(sysTime - mStopTime[stream]) < inPastMs) {
- return true;
- }
- return false;
-}
-
-void AudioPolicyManager::AudioOutputDescriptor::toAudioPortConfig(
- struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig) const
-{
- ALOG_ASSERT(!isDuplicated(), "toAudioPortConfig() called on duplicated output %d", mIoHandle);
-
- dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
- AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN;
- if (srcConfig != NULL) {
- dstConfig->config_mask |= srcConfig->config_mask;
- }
- AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
-
- dstConfig->id = mId;
- dstConfig->role = AUDIO_PORT_ROLE_SOURCE;
- dstConfig->type = AUDIO_PORT_TYPE_MIX;
- dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle;
- dstConfig->ext.mix.handle = mIoHandle;
- dstConfig->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT;
-}
-
-void AudioPolicyManager::AudioOutputDescriptor::toAudioPort(
- struct audio_port *port) const
-{
- ALOG_ASSERT(!isDuplicated(), "toAudioPort() called on duplicated output %d", mIoHandle);
- mProfile->toAudioPort(port);
- port->id = mId;
- toAudioPortConfig(&port->active_config);
- port->ext.mix.hw_module = mProfile->mModule->mHandle;
- port->ext.mix.handle = mIoHandle;
- port->ext.mix.latency_class =
- mFlags & AUDIO_OUTPUT_FLAG_FAST ? AUDIO_LATENCY_LOW : AUDIO_LATENCY_NORMAL;
-}
-
-status_t AudioPolicyManager::AudioOutputDescriptor::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, " ID: %d\n", mId);
- result.append(buffer);
- snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
- result.append(buffer);
- snprintf(buffer, SIZE, " Format: %08x\n", mFormat);
- result.append(buffer);
- snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
- result.append(buffer);
- snprintf(buffer, SIZE, " Latency: %d\n", mLatency);
- result.append(buffer);
- snprintf(buffer, SIZE, " Flags %08x\n", mFlags);
- result.append(buffer);
- snprintf(buffer, SIZE, " Devices %08x\n", device());
- result.append(buffer);
- snprintf(buffer, SIZE, " Stream volume refCount muteCount\n");
- result.append(buffer);
- for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
- snprintf(buffer, SIZE, " %02d %.03f %02d %02d\n",
- i, mCurVolume[i], mRefCount[i], mMuteCount[i]);
- result.append(buffer);
- }
- write(fd, result.string(), result.size());
-
- return NO_ERROR;
-}
-
-// --- AudioInputDescriptor class implementation
-
-AudioPolicyManager::AudioInputDescriptor::AudioInputDescriptor(const sp<IOProfile>& profile)
- : mId(0), mIoHandle(0),
- mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL), mPatchHandle(0), mRefCount(0),
- mInputSource(AUDIO_SOURCE_DEFAULT), mProfile(profile), mIsSoundTrigger(false)
-{
- if (profile != NULL) {
- mSamplingRate = profile->pickSamplingRate();
- mFormat = profile->pickFormat();
- mChannelMask = profile->pickChannelMask();
- if (profile->mGains.size() > 0) {
- profile->mGains[0]->getDefaultConfig(&mGain);
- }
- }
-}
-
-void AudioPolicyManager::AudioInputDescriptor::toAudioPortConfig(
- struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig) const
-{
- ALOG_ASSERT(mProfile != 0,
- "toAudioPortConfig() called on input with null profile %d", mIoHandle);
- dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
- AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN;
- if (srcConfig != NULL) {
- dstConfig->config_mask |= srcConfig->config_mask;
- }
-
- AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
-
- dstConfig->id = mId;
- dstConfig->role = AUDIO_PORT_ROLE_SINK;
- dstConfig->type = AUDIO_PORT_TYPE_MIX;
- dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle;
- dstConfig->ext.mix.handle = mIoHandle;
- dstConfig->ext.mix.usecase.source = mInputSource;
-}
-
-void AudioPolicyManager::AudioInputDescriptor::toAudioPort(
- struct audio_port *port) const
-{
- ALOG_ASSERT(mProfile != 0, "toAudioPort() called on input with null profile %d", mIoHandle);
-
- mProfile->toAudioPort(port);
- port->id = mId;
- toAudioPortConfig(&port->active_config);
- port->ext.mix.hw_module = mProfile->mModule->mHandle;
- port->ext.mix.handle = mIoHandle;
- port->ext.mix.latency_class = AUDIO_LATENCY_NORMAL;
-}
-
-status_t AudioPolicyManager::AudioInputDescriptor::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, " ID: %d\n", mId);
- result.append(buffer);
- snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
- result.append(buffer);
- snprintf(buffer, SIZE, " Format: %d\n", mFormat);
- result.append(buffer);
- snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
- result.append(buffer);
- snprintf(buffer, SIZE, " Devices %08x\n", mDevice);
- result.append(buffer);
- snprintf(buffer, SIZE, " Ref Count %d\n", mRefCount);
- result.append(buffer);
- snprintf(buffer, SIZE, " Open Ref Count %d\n", mOpenRefCount);
- result.append(buffer);
-
- write(fd, result.string(), result.size());
-
- return NO_ERROR;
-}
-
-// --- StreamDescriptor class implementation
-
-AudioPolicyManager::StreamDescriptor::StreamDescriptor()
- : mIndexMin(0), mIndexMax(1), mCanBeMuted(true)
-{
- mIndexCur.add(AUDIO_DEVICE_OUT_DEFAULT, 0);
-}
-
-int AudioPolicyManager::StreamDescriptor::getVolumeIndex(audio_devices_t device)
-{
- device = AudioPolicyManager::getDeviceForVolume(device);
- // there is always a valid entry for AUDIO_DEVICE_OUT_DEFAULT
- if (mIndexCur.indexOfKey(device) < 0) {
- device = AUDIO_DEVICE_OUT_DEFAULT;
- }
- return mIndexCur.valueFor(device);
-}
-
-void AudioPolicyManager::StreamDescriptor::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, "%s %02d %02d ",
- mCanBeMuted ? "true " : "false", mIndexMin, mIndexMax);
- result.append(buffer);
- for (size_t i = 0; i < mIndexCur.size(); i++) {
- snprintf(buffer, SIZE, "%04x : %02d, ",
- mIndexCur.keyAt(i),
- mIndexCur.valueAt(i));
- result.append(buffer);
- }
- result.append("\n");
-
- write(fd, result.string(), result.size());
-}
-
// --- EffectDescriptor class implementation
status_t AudioPolicyManager::EffectDescriptor::dump(int fd)
@@ -6245,1601 +5505,9 @@
return NO_ERROR;
}
-// --- HwModule class implementation
-
-AudioPolicyManager::HwModule::HwModule(const char *name)
- : mName(strndup(name, AUDIO_HARDWARE_MODULE_ID_MAX_LEN)),
- mHalVersion(AUDIO_DEVICE_API_VERSION_MIN), mHandle(0)
-{
-}
-
-AudioPolicyManager::HwModule::~HwModule()
-{
- for (size_t i = 0; i < mOutputProfiles.size(); i++) {
- mOutputProfiles[i]->mSupportedDevices.clear();
- }
- for (size_t i = 0; i < mInputProfiles.size(); i++) {
- mInputProfiles[i]->mSupportedDevices.clear();
- }
- free((void *)mName);
-}
-
-status_t AudioPolicyManager::HwModule::loadInput(cnode *root)
-{
- cnode *node = root->first_child;
-
- sp<IOProfile> profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SINK, this);
-
- while (node) {
- if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
- profile->loadSamplingRates((char *)node->value);
- } else if (strcmp(node->name, FORMATS_TAG) == 0) {
- profile->loadFormats((char *)node->value);
- } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
- profile->loadInChannels((char *)node->value);
- } else if (strcmp(node->name, DEVICES_TAG) == 0) {
- profile->mSupportedDevices.loadDevicesFromName((char *)node->value,
- mDeclaredDevices);
- } else if (strcmp(node->name, FLAGS_TAG) == 0) {
- profile->mFlags = parseInputFlagNames((char *)node->value);
- } else if (strcmp(node->name, GAINS_TAG) == 0) {
- profile->loadGains(node);
- }
- node = node->next;
- }
- ALOGW_IF(profile->mSupportedDevices.isEmpty(),
- "loadInput() invalid supported devices");
- ALOGW_IF(profile->mChannelMasks.size() == 0,
- "loadInput() invalid supported channel masks");
- ALOGW_IF(profile->mSamplingRates.size() == 0,
- "loadInput() invalid supported sampling rates");
- ALOGW_IF(profile->mFormats.size() == 0,
- "loadInput() invalid supported formats");
- if (!profile->mSupportedDevices.isEmpty() &&
- (profile->mChannelMasks.size() != 0) &&
- (profile->mSamplingRates.size() != 0) &&
- (profile->mFormats.size() != 0)) {
-
- ALOGV("loadInput() adding input Supported Devices %04x",
- profile->mSupportedDevices.types());
-
- mInputProfiles.add(profile);
- return NO_ERROR;
- } else {
- return BAD_VALUE;
- }
-}
-
-status_t AudioPolicyManager::HwModule::loadOutput(cnode *root)
-{
- cnode *node = root->first_child;
-
- sp<IOProfile> profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SOURCE, this);
-
- while (node) {
- if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
- profile->loadSamplingRates((char *)node->value);
- } else if (strcmp(node->name, FORMATS_TAG) == 0) {
- profile->loadFormats((char *)node->value);
- } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
- profile->loadOutChannels((char *)node->value);
- } else if (strcmp(node->name, DEVICES_TAG) == 0) {
- profile->mSupportedDevices.loadDevicesFromName((char *)node->value,
- mDeclaredDevices);
- } else if (strcmp(node->name, FLAGS_TAG) == 0) {
- profile->mFlags = parseOutputFlagNames((char *)node->value);
- } else if (strcmp(node->name, GAINS_TAG) == 0) {
- profile->loadGains(node);
- }
- node = node->next;
- }
- ALOGW_IF(profile->mSupportedDevices.isEmpty(),
- "loadOutput() invalid supported devices");
- ALOGW_IF(profile->mChannelMasks.size() == 0,
- "loadOutput() invalid supported channel masks");
- ALOGW_IF(profile->mSamplingRates.size() == 0,
- "loadOutput() invalid supported sampling rates");
- ALOGW_IF(profile->mFormats.size() == 0,
- "loadOutput() invalid supported formats");
- if (!profile->mSupportedDevices.isEmpty() &&
- (profile->mChannelMasks.size() != 0) &&
- (profile->mSamplingRates.size() != 0) &&
- (profile->mFormats.size() != 0)) {
-
- ALOGV("loadOutput() adding output Supported Devices %04x, mFlags %04x",
- profile->mSupportedDevices.types(), profile->mFlags);
-
- mOutputProfiles.add(profile);
- return NO_ERROR;
- } else {
- return BAD_VALUE;
- }
-}
-
-status_t AudioPolicyManager::HwModule::loadDevice(cnode *root)
-{
- cnode *node = root->first_child;
-
- audio_devices_t type = AUDIO_DEVICE_NONE;
- while (node) {
- if (strcmp(node->name, DEVICE_TYPE) == 0) {
- type = parseDeviceNames((char *)node->value);
- break;
- }
- node = node->next;
- }
- if (type == AUDIO_DEVICE_NONE ||
- (!audio_is_input_device(type) && !audio_is_output_device(type))) {
- ALOGW("loadDevice() bad type %08x", type);
- return BAD_VALUE;
- }
- sp<DeviceDescriptor> deviceDesc = new DeviceDescriptor(String8(root->name), type);
- deviceDesc->mModule = this;
-
- node = root->first_child;
- while (node) {
- if (strcmp(node->name, DEVICE_ADDRESS) == 0) {
- deviceDesc->mAddress = String8((char *)node->value);
- } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
- if (audio_is_input_device(type)) {
- deviceDesc->loadInChannels((char *)node->value);
- } else {
- deviceDesc->loadOutChannels((char *)node->value);
- }
- } else if (strcmp(node->name, GAINS_TAG) == 0) {
- deviceDesc->loadGains(node);
- }
- node = node->next;
- }
-
- ALOGV("loadDevice() adding device name %s type %08x address %s",
- deviceDesc->mName.string(), type, deviceDesc->mAddress.string());
-
- mDeclaredDevices.add(deviceDesc);
-
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::HwModule::addOutputProfile(String8 name, const audio_config_t *config,
- audio_devices_t device, String8 address)
-{
- sp<IOProfile> profile = new IOProfile(name, AUDIO_PORT_ROLE_SOURCE, this);
-
- profile->mSamplingRates.add(config->sample_rate);
- profile->mChannelMasks.add(config->channel_mask);
- profile->mFormats.add(config->format);
-
- sp<DeviceDescriptor> devDesc = new DeviceDescriptor(String8(""), device);
- devDesc->mAddress = address;
- profile->mSupportedDevices.add(devDesc);
-
- mOutputProfiles.add(profile);
-
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::HwModule::removeOutputProfile(String8 name)
-{
- for (size_t i = 0; i < mOutputProfiles.size(); i++) {
- if (mOutputProfiles[i]->mName == name) {
- mOutputProfiles.removeAt(i);
- break;
- }
- }
-
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::HwModule::addInputProfile(String8 name, const audio_config_t *config,
- audio_devices_t device, String8 address)
-{
- sp<IOProfile> profile = new IOProfile(name, AUDIO_PORT_ROLE_SINK, this);
-
- profile->mSamplingRates.add(config->sample_rate);
- profile->mChannelMasks.add(config->channel_mask);
- profile->mFormats.add(config->format);
-
- sp<DeviceDescriptor> devDesc = new DeviceDescriptor(String8(""), device);
- devDesc->mAddress = address;
- profile->mSupportedDevices.add(devDesc);
-
- ALOGV("addInputProfile() name %s rate %d mask 0x08", name.string(), config->sample_rate, config->channel_mask);
-
- mInputProfiles.add(profile);
-
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::HwModule::removeInputProfile(String8 name)
-{
- for (size_t i = 0; i < mInputProfiles.size(); i++) {
- if (mInputProfiles[i]->mName == name) {
- mInputProfiles.removeAt(i);
- break;
- }
- }
-
- return NO_ERROR;
-}
-
-
-void AudioPolicyManager::HwModule::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, " - name: %s\n", mName);
- result.append(buffer);
- snprintf(buffer, SIZE, " - handle: %d\n", mHandle);
- result.append(buffer);
- snprintf(buffer, SIZE, " - version: %u.%u\n", mHalVersion >> 8, mHalVersion & 0xFF);
- result.append(buffer);
- write(fd, result.string(), result.size());
- if (mOutputProfiles.size()) {
- write(fd, " - outputs:\n", strlen(" - outputs:\n"));
- for (size_t i = 0; i < mOutputProfiles.size(); i++) {
- snprintf(buffer, SIZE, " output %zu:\n", i);
- write(fd, buffer, strlen(buffer));
- mOutputProfiles[i]->dump(fd);
- }
- }
- if (mInputProfiles.size()) {
- write(fd, " - inputs:\n", strlen(" - inputs:\n"));
- for (size_t i = 0; i < mInputProfiles.size(); i++) {
- snprintf(buffer, SIZE, " input %zu:\n", i);
- write(fd, buffer, strlen(buffer));
- mInputProfiles[i]->dump(fd);
- }
- }
- if (mDeclaredDevices.size()) {
- write(fd, " - devices:\n", strlen(" - devices:\n"));
- for (size_t i = 0; i < mDeclaredDevices.size(); i++) {
- mDeclaredDevices[i]->dump(fd, 4, i);
- }
- }
-}
-
-// --- AudioPort class implementation
-
-
-AudioPolicyManager::AudioPort::AudioPort(const String8& name, audio_port_type_t type,
- audio_port_role_t role, const sp<HwModule>& module) :
- mName(name), mType(type), mRole(role), mModule(module), mFlags(0)
-{
- mUseInChannelMask = ((type == AUDIO_PORT_TYPE_DEVICE) && (role == AUDIO_PORT_ROLE_SOURCE)) ||
- ((type == AUDIO_PORT_TYPE_MIX) && (role == AUDIO_PORT_ROLE_SINK));
-}
-
-void AudioPolicyManager::AudioPort::toAudioPort(struct audio_port *port) const
-{
- port->role = mRole;
- port->type = mType;
- unsigned int i;
- for (i = 0; i < mSamplingRates.size() && i < AUDIO_PORT_MAX_SAMPLING_RATES; i++) {
- if (mSamplingRates[i] != 0) {
- port->sample_rates[i] = mSamplingRates[i];
- }
- }
- port->num_sample_rates = i;
- for (i = 0; i < mChannelMasks.size() && i < AUDIO_PORT_MAX_CHANNEL_MASKS; i++) {
- if (mChannelMasks[i] != 0) {
- port->channel_masks[i] = mChannelMasks[i];
- }
- }
- port->num_channel_masks = i;
- for (i = 0; i < mFormats.size() && i < AUDIO_PORT_MAX_FORMATS; i++) {
- if (mFormats[i] != 0) {
- port->formats[i] = mFormats[i];
- }
- }
- port->num_formats = i;
-
- ALOGV("AudioPort::toAudioPort() num gains %zu", mGains.size());
-
- for (i = 0; i < mGains.size() && i < AUDIO_PORT_MAX_GAINS; i++) {
- port->gains[i] = mGains[i]->mGain;
- }
- port->num_gains = i;
-}
-
-void AudioPolicyManager::AudioPort::importAudioPort(const sp<AudioPort> port) {
- for (size_t k = 0 ; k < port->mSamplingRates.size() ; k++) {
- const uint32_t rate = port->mSamplingRates.itemAt(k);
- if (rate != 0) { // skip "dynamic" rates
- bool hasRate = false;
- for (size_t l = 0 ; l < mSamplingRates.size() ; l++) {
- if (rate == mSamplingRates.itemAt(l)) {
- hasRate = true;
- break;
- }
- }
- if (!hasRate) { // never import a sampling rate twice
- mSamplingRates.add(rate);
- }
- }
- }
- for (size_t k = 0 ; k < port->mChannelMasks.size() ; k++) {
- const audio_channel_mask_t mask = port->mChannelMasks.itemAt(k);
- if (mask != 0) { // skip "dynamic" masks
- bool hasMask = false;
- for (size_t l = 0 ; l < mChannelMasks.size() ; l++) {
- if (mask == mChannelMasks.itemAt(l)) {
- hasMask = true;
- break;
- }
- }
- if (!hasMask) { // never import a channel mask twice
- mChannelMasks.add(mask);
- }
- }
- }
- for (size_t k = 0 ; k < port->mFormats.size() ; k++) {
- const audio_format_t format = port->mFormats.itemAt(k);
- if (format != 0) { // skip "dynamic" formats
- bool hasFormat = false;
- for (size_t l = 0 ; l < mFormats.size() ; l++) {
- if (format == mFormats.itemAt(l)) {
- hasFormat = true;
- break;
- }
- }
- if (!hasFormat) { // never import a channel mask twice
- mFormats.add(format);
- }
- }
- }
- for (size_t k = 0 ; k < port->mGains.size() ; k++) {
- sp<AudioGain> gain = port->mGains.itemAt(k);
- if (gain != 0) {
- bool hasGain = false;
- for (size_t l = 0 ; l < mGains.size() ; l++) {
- if (gain == mGains.itemAt(l)) {
- hasGain = true;
- break;
- }
- }
- if (!hasGain) { // never import a gain twice
- mGains.add(gain);
- }
- }
- }
-}
-
-void AudioPolicyManager::AudioPort::clearCapabilities() {
- mChannelMasks.clear();
- mFormats.clear();
- mSamplingRates.clear();
- mGains.clear();
-}
-
-void AudioPolicyManager::AudioPort::loadSamplingRates(char *name)
-{
- char *str = strtok(name, "|");
-
- // by convention, "0' in the first entry in mSamplingRates indicates the supported sampling
- // rates should be read from the output stream after it is opened for the first time
- if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
- mSamplingRates.add(0);
- return;
- }
-
- while (str != NULL) {
- uint32_t rate = atoi(str);
- if (rate != 0) {
- ALOGV("loadSamplingRates() adding rate %d", rate);
- mSamplingRates.add(rate);
- }
- str = strtok(NULL, "|");
- }
-}
-
-void AudioPolicyManager::AudioPort::loadFormats(char *name)
-{
- char *str = strtok(name, "|");
-
- // by convention, "0' in the first entry in mFormats indicates the supported formats
- // should be read from the output stream after it is opened for the first time
- if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
- mFormats.add(AUDIO_FORMAT_DEFAULT);
- return;
- }
-
- while (str != NULL) {
- audio_format_t format = (audio_format_t)stringToEnum(sFormatNameToEnumTable,
- ARRAY_SIZE(sFormatNameToEnumTable),
- str);
- if (format != AUDIO_FORMAT_DEFAULT) {
- mFormats.add(format);
- }
- str = strtok(NULL, "|");
- }
-}
-
-void AudioPolicyManager::AudioPort::loadInChannels(char *name)
-{
- const char *str = strtok(name, "|");
-
- ALOGV("loadInChannels() %s", name);
-
- if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
- mChannelMasks.add(0);
- return;
- }
-
- while (str != NULL) {
- audio_channel_mask_t channelMask =
- (audio_channel_mask_t)stringToEnum(sInChannelsNameToEnumTable,
- ARRAY_SIZE(sInChannelsNameToEnumTable),
- str);
- if (channelMask != 0) {
- ALOGV("loadInChannels() adding channelMask %04x", channelMask);
- mChannelMasks.add(channelMask);
- }
- str = strtok(NULL, "|");
- }
-}
-
-void AudioPolicyManager::AudioPort::loadOutChannels(char *name)
-{
- const char *str = strtok(name, "|");
-
- ALOGV("loadOutChannels() %s", name);
-
- // by convention, "0' in the first entry in mChannelMasks indicates the supported channel
- // masks should be read from the output stream after it is opened for the first time
- if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
- mChannelMasks.add(0);
- return;
- }
-
- while (str != NULL) {
- audio_channel_mask_t channelMask =
- (audio_channel_mask_t)stringToEnum(sOutChannelsNameToEnumTable,
- ARRAY_SIZE(sOutChannelsNameToEnumTable),
- str);
- if (channelMask != 0) {
- mChannelMasks.add(channelMask);
- }
- str = strtok(NULL, "|");
- }
- return;
-}
-
-audio_gain_mode_t AudioPolicyManager::AudioPort::loadGainMode(char *name)
-{
- const char *str = strtok(name, "|");
-
- ALOGV("loadGainMode() %s", name);
- audio_gain_mode_t mode = 0;
- while (str != NULL) {
- mode |= (audio_gain_mode_t)stringToEnum(sGainModeNameToEnumTable,
- ARRAY_SIZE(sGainModeNameToEnumTable),
- str);
- str = strtok(NULL, "|");
- }
- return mode;
-}
-
-void AudioPolicyManager::AudioPort::loadGain(cnode *root, int index)
-{
- cnode *node = root->first_child;
-
- sp<AudioGain> gain = new AudioGain(index, mUseInChannelMask);
-
- while (node) {
- if (strcmp(node->name, GAIN_MODE) == 0) {
- gain->mGain.mode = loadGainMode((char *)node->value);
- } else if (strcmp(node->name, GAIN_CHANNELS) == 0) {
- if (mUseInChannelMask) {
- gain->mGain.channel_mask =
- (audio_channel_mask_t)stringToEnum(sInChannelsNameToEnumTable,
- ARRAY_SIZE(sInChannelsNameToEnumTable),
- (char *)node->value);
- } else {
- gain->mGain.channel_mask =
- (audio_channel_mask_t)stringToEnum(sOutChannelsNameToEnumTable,
- ARRAY_SIZE(sOutChannelsNameToEnumTable),
- (char *)node->value);
- }
- } else if (strcmp(node->name, GAIN_MIN_VALUE) == 0) {
- gain->mGain.min_value = atoi((char *)node->value);
- } else if (strcmp(node->name, GAIN_MAX_VALUE) == 0) {
- gain->mGain.max_value = atoi((char *)node->value);
- } else if (strcmp(node->name, GAIN_DEFAULT_VALUE) == 0) {
- gain->mGain.default_value = atoi((char *)node->value);
- } else if (strcmp(node->name, GAIN_STEP_VALUE) == 0) {
- gain->mGain.step_value = atoi((char *)node->value);
- } else if (strcmp(node->name, GAIN_MIN_RAMP_MS) == 0) {
- gain->mGain.min_ramp_ms = atoi((char *)node->value);
- } else if (strcmp(node->name, GAIN_MAX_RAMP_MS) == 0) {
- gain->mGain.max_ramp_ms = atoi((char *)node->value);
- }
- node = node->next;
- }
-
- ALOGV("loadGain() adding new gain mode %08x channel mask %08x min mB %d max mB %d",
- gain->mGain.mode, gain->mGain.channel_mask, gain->mGain.min_value, gain->mGain.max_value);
-
- if (gain->mGain.mode == 0) {
- return;
- }
- mGains.add(gain);
-}
-
-void AudioPolicyManager::AudioPort::loadGains(cnode *root)
-{
- cnode *node = root->first_child;
- int index = 0;
- while (node) {
- ALOGV("loadGains() loading gain %s", node->name);
- loadGain(node, index++);
- node = node->next;
- }
-}
-
-status_t AudioPolicyManager::AudioPort::checkExactSamplingRate(uint32_t samplingRate) const
-{
- if (mSamplingRates.isEmpty()) {
- return NO_ERROR;
- }
-
- for (size_t i = 0; i < mSamplingRates.size(); i ++) {
- if (mSamplingRates[i] == samplingRate) {
- return NO_ERROR;
- }
- }
- return BAD_VALUE;
-}
-
-status_t AudioPolicyManager::AudioPort::checkCompatibleSamplingRate(uint32_t samplingRate,
- uint32_t *updatedSamplingRate) const
-{
- if (mSamplingRates.isEmpty()) {
- return NO_ERROR;
- }
-
- // Search for the closest supported sampling rate that is above (preferred)
- // or below (acceptable) the desired sampling rate, within a permitted ratio.
- // The sampling rates do not need to be sorted in ascending order.
- ssize_t maxBelow = -1;
- ssize_t minAbove = -1;
- uint32_t candidate;
- for (size_t i = 0; i < mSamplingRates.size(); i++) {
- candidate = mSamplingRates[i];
- if (candidate == samplingRate) {
- if (updatedSamplingRate != NULL) {
- *updatedSamplingRate = candidate;
- }
- return NO_ERROR;
- }
- // candidate < desired
- if (candidate < samplingRate) {
- if (maxBelow < 0 || candidate > mSamplingRates[maxBelow]) {
- maxBelow = i;
- }
- // candidate > desired
- } else {
- if (minAbove < 0 || candidate < mSamplingRates[minAbove]) {
- minAbove = i;
- }
- }
- }
- // This uses hard-coded knowledge about AudioFlinger resampling ratios.
- // TODO Move these assumptions out.
- static const uint32_t kMaxDownSampleRatio = 6; // beyond this aliasing occurs
- static const uint32_t kMaxUpSampleRatio = 256; // beyond this sample rate inaccuracies occur
- // due to approximation by an int32_t of the
- // phase increments
- // Prefer to down-sample from a higher sampling rate, as we get the desired frequency spectrum.
- if (minAbove >= 0) {
- candidate = mSamplingRates[minAbove];
- if (candidate / kMaxDownSampleRatio <= samplingRate) {
- if (updatedSamplingRate != NULL) {
- *updatedSamplingRate = candidate;
- }
- return NO_ERROR;
- }
- }
- // But if we have to up-sample from a lower sampling rate, that's OK.
- if (maxBelow >= 0) {
- candidate = mSamplingRates[maxBelow];
- if (candidate * kMaxUpSampleRatio >= samplingRate) {
- if (updatedSamplingRate != NULL) {
- *updatedSamplingRate = candidate;
- }
- return NO_ERROR;
- }
- }
- // leave updatedSamplingRate unmodified
- return BAD_VALUE;
-}
-
-status_t AudioPolicyManager::AudioPort::checkExactChannelMask(audio_channel_mask_t channelMask) const
-{
- if (mChannelMasks.isEmpty()) {
- return NO_ERROR;
- }
-
- for (size_t i = 0; i < mChannelMasks.size(); i++) {
- if (mChannelMasks[i] == channelMask) {
- return NO_ERROR;
- }
- }
- return BAD_VALUE;
-}
-
-status_t AudioPolicyManager::AudioPort::checkCompatibleChannelMask(audio_channel_mask_t channelMask)
- const
-{
- if (mChannelMasks.isEmpty()) {
- return NO_ERROR;
- }
-
- const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK;
- for (size_t i = 0; i < mChannelMasks.size(); i ++) {
- // FIXME Does not handle multi-channel automatic conversions yet
- audio_channel_mask_t supported = mChannelMasks[i];
- if (supported == channelMask) {
- return NO_ERROR;
- }
- if (isRecordThread) {
- // This uses hard-coded knowledge that AudioFlinger can silently down-mix and up-mix.
- // FIXME Abstract this out to a table.
- if (((supported == AUDIO_CHANNEL_IN_FRONT_BACK || supported == AUDIO_CHANNEL_IN_STEREO)
- && channelMask == AUDIO_CHANNEL_IN_MONO) ||
- (supported == AUDIO_CHANNEL_IN_MONO && (channelMask == AUDIO_CHANNEL_IN_FRONT_BACK
- || channelMask == AUDIO_CHANNEL_IN_STEREO))) {
- return NO_ERROR;
- }
- }
- }
- return BAD_VALUE;
-}
-
-status_t AudioPolicyManager::AudioPort::checkFormat(audio_format_t format) const
-{
- if (mFormats.isEmpty()) {
- return NO_ERROR;
- }
-
- for (size_t i = 0; i < mFormats.size(); i ++) {
- if (mFormats[i] == format) {
- return NO_ERROR;
- }
- }
- return BAD_VALUE;
-}
-
-
-uint32_t AudioPolicyManager::AudioPort::pickSamplingRate() const
-{
- // special case for uninitialized dynamic profile
- if (mSamplingRates.size() == 1 && mSamplingRates[0] == 0) {
- return 0;
- }
-
- // For direct outputs, pick minimum sampling rate: this helps ensuring that the
- // channel count / sampling rate combination chosen will be supported by the connected
- // sink
- if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
- (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) {
- uint32_t samplingRate = UINT_MAX;
- for (size_t i = 0; i < mSamplingRates.size(); i ++) {
- if ((mSamplingRates[i] < samplingRate) && (mSamplingRates[i] > 0)) {
- samplingRate = mSamplingRates[i];
- }
- }
- return (samplingRate == UINT_MAX) ? 0 : samplingRate;
- }
-
- uint32_t samplingRate = 0;
- uint32_t maxRate = MAX_MIXER_SAMPLING_RATE;
-
- // For mixed output and inputs, use max mixer sampling rates. Do not
- // limit sampling rate otherwise
- if (mType != AUDIO_PORT_TYPE_MIX) {
- maxRate = UINT_MAX;
- }
- for (size_t i = 0; i < mSamplingRates.size(); i ++) {
- if ((mSamplingRates[i] > samplingRate) && (mSamplingRates[i] <= maxRate)) {
- samplingRate = mSamplingRates[i];
- }
- }
- return samplingRate;
-}
-
-audio_channel_mask_t AudioPolicyManager::AudioPort::pickChannelMask() const
-{
- // special case for uninitialized dynamic profile
- if (mChannelMasks.size() == 1 && mChannelMasks[0] == 0) {
- return AUDIO_CHANNEL_NONE;
- }
- audio_channel_mask_t channelMask = AUDIO_CHANNEL_NONE;
-
- // For direct outputs, pick minimum channel count: this helps ensuring that the
- // channel count / sampling rate combination chosen will be supported by the connected
- // sink
- if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
- (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) {
- uint32_t channelCount = UINT_MAX;
- for (size_t i = 0; i < mChannelMasks.size(); i ++) {
- uint32_t cnlCount;
- if (mUseInChannelMask) {
- cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]);
- } else {
- cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]);
- }
- if ((cnlCount < channelCount) && (cnlCount > 0)) {
- channelMask = mChannelMasks[i];
- channelCount = cnlCount;
- }
- }
- return channelMask;
- }
-
- uint32_t channelCount = 0;
- uint32_t maxCount = MAX_MIXER_CHANNEL_COUNT;
-
- // For mixed output and inputs, use max mixer channel count. Do not
- // limit channel count otherwise
- if (mType != AUDIO_PORT_TYPE_MIX) {
- maxCount = UINT_MAX;
- }
- for (size_t i = 0; i < mChannelMasks.size(); i ++) {
- uint32_t cnlCount;
- if (mUseInChannelMask) {
- cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]);
- } else {
- cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]);
- }
- if ((cnlCount > channelCount) && (cnlCount <= maxCount)) {
- channelMask = mChannelMasks[i];
- channelCount = cnlCount;
- }
- }
- return channelMask;
-}
-
-/* format in order of increasing preference */
-const audio_format_t AudioPolicyManager::AudioPort::sPcmFormatCompareTable[] = {
- AUDIO_FORMAT_DEFAULT,
- AUDIO_FORMAT_PCM_16_BIT,
- AUDIO_FORMAT_PCM_8_24_BIT,
- AUDIO_FORMAT_PCM_24_BIT_PACKED,
- AUDIO_FORMAT_PCM_32_BIT,
- AUDIO_FORMAT_PCM_FLOAT,
-};
-
-int AudioPolicyManager::AudioPort::compareFormats(audio_format_t format1,
- audio_format_t format2)
-{
- // NOTE: AUDIO_FORMAT_INVALID is also considered not PCM and will be compared equal to any
- // compressed format and better than any PCM format. This is by design of pickFormat()
- if (!audio_is_linear_pcm(format1)) {
- if (!audio_is_linear_pcm(format2)) {
- return 0;
- }
- return 1;
- }
- if (!audio_is_linear_pcm(format2)) {
- return -1;
- }
-
- int index1 = -1, index2 = -1;
- for (size_t i = 0;
- (i < ARRAY_SIZE(sPcmFormatCompareTable)) && ((index1 == -1) || (index2 == -1));
- i ++) {
- if (sPcmFormatCompareTable[i] == format1) {
- index1 = i;
- }
- if (sPcmFormatCompareTable[i] == format2) {
- index2 = i;
- }
- }
- // format1 not found => index1 < 0 => format2 > format1
- // format2 not found => index2 < 0 => format2 < format1
- return index1 - index2;
-}
-
-audio_format_t AudioPolicyManager::AudioPort::pickFormat() const
-{
- // special case for uninitialized dynamic profile
- if (mFormats.size() == 1 && mFormats[0] == 0) {
- return AUDIO_FORMAT_DEFAULT;
- }
-
- audio_format_t format = AUDIO_FORMAT_DEFAULT;
- audio_format_t bestFormat =
- AudioPolicyManager::AudioPort::sPcmFormatCompareTable[
- ARRAY_SIZE(AudioPolicyManager::AudioPort::sPcmFormatCompareTable) - 1];
- // For mixed output and inputs, use best mixer output format. Do not
- // limit format otherwise
- if ((mType != AUDIO_PORT_TYPE_MIX) ||
- ((mRole == AUDIO_PORT_ROLE_SOURCE) &&
- (((mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) != 0)))) {
- bestFormat = AUDIO_FORMAT_INVALID;
- }
-
- for (size_t i = 0; i < mFormats.size(); i ++) {
- if ((compareFormats(mFormats[i], format) > 0) &&
- (compareFormats(mFormats[i], bestFormat) <= 0)) {
- format = mFormats[i];
- }
- }
- return format;
-}
-
-status_t AudioPolicyManager::AudioPort::checkGain(const struct audio_gain_config *gainConfig,
- int index) const
-{
- if (index < 0 || (size_t)index >= mGains.size()) {
- return BAD_VALUE;
- }
- return mGains[index]->checkConfig(gainConfig);
-}
-
-void AudioPolicyManager::AudioPort::dump(int fd, int spaces) const
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- if (mName.size() != 0) {
- snprintf(buffer, SIZE, "%*s- name: %s\n", spaces, "", mName.string());
- result.append(buffer);
- }
-
- if (mSamplingRates.size() != 0) {
- snprintf(buffer, SIZE, "%*s- sampling rates: ", spaces, "");
- result.append(buffer);
- for (size_t i = 0; i < mSamplingRates.size(); i++) {
- if (i == 0 && mSamplingRates[i] == 0) {
- snprintf(buffer, SIZE, "Dynamic");
- } else {
- snprintf(buffer, SIZE, "%d", mSamplingRates[i]);
- }
- result.append(buffer);
- result.append(i == (mSamplingRates.size() - 1) ? "" : ", ");
- }
- result.append("\n");
- }
-
- if (mChannelMasks.size() != 0) {
- snprintf(buffer, SIZE, "%*s- channel masks: ", spaces, "");
- result.append(buffer);
- for (size_t i = 0; i < mChannelMasks.size(); i++) {
- ALOGV("AudioPort::dump mChannelMasks %zu %08x", i, mChannelMasks[i]);
-
- if (i == 0 && mChannelMasks[i] == 0) {
- snprintf(buffer, SIZE, "Dynamic");
- } else {
- snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]);
- }
- result.append(buffer);
- result.append(i == (mChannelMasks.size() - 1) ? "" : ", ");
- }
- result.append("\n");
- }
-
- if (mFormats.size() != 0) {
- snprintf(buffer, SIZE, "%*s- formats: ", spaces, "");
- result.append(buffer);
- for (size_t i = 0; i < mFormats.size(); i++) {
- const char *formatStr = enumToString(sFormatNameToEnumTable,
- ARRAY_SIZE(sFormatNameToEnumTable),
- mFormats[i]);
- if (i == 0 && strcmp(formatStr, "") == 0) {
- snprintf(buffer, SIZE, "Dynamic");
- } else {
- snprintf(buffer, SIZE, "%s", formatStr);
- }
- result.append(buffer);
- result.append(i == (mFormats.size() - 1) ? "" : ", ");
- }
- result.append("\n");
- }
- write(fd, result.string(), result.size());
- if (mGains.size() != 0) {
- snprintf(buffer, SIZE, "%*s- gains:\n", spaces, "");
- write(fd, buffer, strlen(buffer) + 1);
- result.append(buffer);
- for (size_t i = 0; i < mGains.size(); i++) {
- mGains[i]->dump(fd, spaces + 2, i);
- }
- }
-}
-
-// --- AudioGain class implementation
-
-AudioPolicyManager::AudioGain::AudioGain(int index, bool useInChannelMask)
-{
- mIndex = index;
- mUseInChannelMask = useInChannelMask;
- memset(&mGain, 0, sizeof(struct audio_gain));
-}
-
-void AudioPolicyManager::AudioGain::getDefaultConfig(struct audio_gain_config *config)
-{
- config->index = mIndex;
- config->mode = mGain.mode;
- config->channel_mask = mGain.channel_mask;
- if ((mGain.mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
- config->values[0] = mGain.default_value;
- } else {
- uint32_t numValues;
- if (mUseInChannelMask) {
- numValues = audio_channel_count_from_in_mask(mGain.channel_mask);
- } else {
- numValues = audio_channel_count_from_out_mask(mGain.channel_mask);
- }
- for (size_t i = 0; i < numValues; i++) {
- config->values[i] = mGain.default_value;
- }
- }
- if ((mGain.mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
- config->ramp_duration_ms = mGain.min_ramp_ms;
- }
-}
-
-status_t AudioPolicyManager::AudioGain::checkConfig(const struct audio_gain_config *config)
-{
- if ((config->mode & ~mGain.mode) != 0) {
- return BAD_VALUE;
- }
- if ((config->mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
- if ((config->values[0] < mGain.min_value) ||
- (config->values[0] > mGain.max_value)) {
- return BAD_VALUE;
- }
- } else {
- if ((config->channel_mask & ~mGain.channel_mask) != 0) {
- return BAD_VALUE;
- }
- uint32_t numValues;
- if (mUseInChannelMask) {
- numValues = audio_channel_count_from_in_mask(config->channel_mask);
- } else {
- numValues = audio_channel_count_from_out_mask(config->channel_mask);
- }
- for (size_t i = 0; i < numValues; i++) {
- if ((config->values[i] < mGain.min_value) ||
- (config->values[i] > mGain.max_value)) {
- return BAD_VALUE;
- }
- }
- }
- if ((config->mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
- if ((config->ramp_duration_ms < mGain.min_ramp_ms) ||
- (config->ramp_duration_ms > mGain.max_ramp_ms)) {
- return BAD_VALUE;
- }
- }
- return NO_ERROR;
-}
-
-void AudioPolicyManager::AudioGain::dump(int fd, int spaces, int index) const
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, "%*sGain %d:\n", spaces, "", index+1);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- mode: %08x\n", spaces, "", mGain.mode);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- channel_mask: %08x\n", spaces, "", mGain.channel_mask);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- min_value: %d mB\n", spaces, "", mGain.min_value);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- max_value: %d mB\n", spaces, "", mGain.max_value);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- default_value: %d mB\n", spaces, "", mGain.default_value);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- step_value: %d mB\n", spaces, "", mGain.step_value);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- min_ramp_ms: %d ms\n", spaces, "", mGain.min_ramp_ms);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- max_ramp_ms: %d ms\n", spaces, "", mGain.max_ramp_ms);
- result.append(buffer);
-
- write(fd, result.string(), result.size());
-}
-
-// --- AudioPortConfig class implementation
-
-AudioPolicyManager::AudioPortConfig::AudioPortConfig()
-{
- mSamplingRate = 0;
- mChannelMask = AUDIO_CHANNEL_NONE;
- mFormat = AUDIO_FORMAT_INVALID;
- mGain.index = -1;
-}
-
-status_t AudioPolicyManager::AudioPortConfig::applyAudioPortConfig(
- const struct audio_port_config *config,
- struct audio_port_config *backupConfig)
-{
- struct audio_port_config localBackupConfig;
- status_t status = NO_ERROR;
-
- localBackupConfig.config_mask = config->config_mask;
- toAudioPortConfig(&localBackupConfig);
-
- sp<AudioPort> audioport = getAudioPort();
- if (audioport == 0) {
- status = NO_INIT;
- goto exit;
- }
- if (config->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
- status = audioport->checkExactSamplingRate(config->sample_rate);
- if (status != NO_ERROR) {
- goto exit;
- }
- mSamplingRate = config->sample_rate;
- }
- if (config->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
- status = audioport->checkExactChannelMask(config->channel_mask);
- if (status != NO_ERROR) {
- goto exit;
- }
- mChannelMask = config->channel_mask;
- }
- if (config->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
- status = audioport->checkFormat(config->format);
- if (status != NO_ERROR) {
- goto exit;
- }
- mFormat = config->format;
- }
- if (config->config_mask & AUDIO_PORT_CONFIG_GAIN) {
- status = audioport->checkGain(&config->gain, config->gain.index);
- if (status != NO_ERROR) {
- goto exit;
- }
- mGain = config->gain;
- }
-
-exit:
- if (status != NO_ERROR) {
- applyAudioPortConfig(&localBackupConfig);
- }
- if (backupConfig != NULL) {
- *backupConfig = localBackupConfig;
- }
- return status;
-}
-
-void AudioPolicyManager::AudioPortConfig::toAudioPortConfig(
- struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig) const
-{
- if (dstConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
- dstConfig->sample_rate = mSamplingRate;
- if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE)) {
- dstConfig->sample_rate = srcConfig->sample_rate;
- }
- } else {
- dstConfig->sample_rate = 0;
- }
- if (dstConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
- dstConfig->channel_mask = mChannelMask;
- if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK)) {
- dstConfig->channel_mask = srcConfig->channel_mask;
- }
- } else {
- dstConfig->channel_mask = AUDIO_CHANNEL_NONE;
- }
- if (dstConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
- dstConfig->format = mFormat;
- if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT)) {
- dstConfig->format = srcConfig->format;
- }
- } else {
- dstConfig->format = AUDIO_FORMAT_INVALID;
- }
- if (dstConfig->config_mask & AUDIO_PORT_CONFIG_GAIN) {
- dstConfig->gain = mGain;
- if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_GAIN)) {
- dstConfig->gain = srcConfig->gain;
- }
- } else {
- dstConfig->gain.index = -1;
- }
- if (dstConfig->gain.index != -1) {
- dstConfig->config_mask |= AUDIO_PORT_CONFIG_GAIN;
- } else {
- dstConfig->config_mask &= ~AUDIO_PORT_CONFIG_GAIN;
- }
-}
-
-// --- IOProfile class implementation
-
-AudioPolicyManager::IOProfile::IOProfile(const String8& name, audio_port_role_t role,
- const sp<HwModule>& module)
- : AudioPort(name, AUDIO_PORT_TYPE_MIX, role, module)
-{
-}
-
-AudioPolicyManager::IOProfile::~IOProfile()
-{
-}
-
-// checks if the IO profile is compatible with specified parameters.
-// Sampling rate, format and channel mask must be specified in order to
-// get a valid a match
-bool AudioPolicyManager::IOProfile::isCompatibleProfile(audio_devices_t device,
- String8 address,
- uint32_t samplingRate,
- uint32_t *updatedSamplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- uint32_t flags) const
-{
- const bool isPlaybackThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SOURCE;
- const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK;
- ALOG_ASSERT(isPlaybackThread != isRecordThread);
-
- if (device != AUDIO_DEVICE_NONE && mSupportedDevices.getDevice(device, address) == 0) {
- return false;
- }
-
- if (samplingRate == 0) {
- return false;
- }
- uint32_t myUpdatedSamplingRate = samplingRate;
- if (isPlaybackThread && checkExactSamplingRate(samplingRate) != NO_ERROR) {
- return false;
- }
- if (isRecordThread && checkCompatibleSamplingRate(samplingRate, &myUpdatedSamplingRate) !=
- NO_ERROR) {
- return false;
- }
-
- if (!audio_is_valid_format(format) || checkFormat(format) != NO_ERROR) {
- return false;
- }
-
- if (isPlaybackThread && (!audio_is_output_channel(channelMask) ||
- checkExactChannelMask(channelMask) != NO_ERROR)) {
- return false;
- }
- if (isRecordThread && (!audio_is_input_channel(channelMask) ||
- checkCompatibleChannelMask(channelMask) != NO_ERROR)) {
- return false;
- }
-
- if (isPlaybackThread && (mFlags & flags) != flags) {
- return false;
- }
- // The only input flag that is allowed to be different is the fast flag.
- // An existing fast stream is compatible with a normal track request.
- // An existing normal stream is compatible with a fast track request,
- // but the fast request will be denied by AudioFlinger and converted to normal track.
- if (isRecordThread && ((mFlags ^ flags) &
- ~AUDIO_INPUT_FLAG_FAST)) {
- return false;
- }
-
- if (updatedSamplingRate != NULL) {
- *updatedSamplingRate = myUpdatedSamplingRate;
- }
- return true;
-}
-
-void AudioPolicyManager::IOProfile::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- AudioPort::dump(fd, 4);
-
- snprintf(buffer, SIZE, " - flags: 0x%04x\n", mFlags);
- result.append(buffer);
- snprintf(buffer, SIZE, " - devices:\n");
- result.append(buffer);
- write(fd, result.string(), result.size());
- for (size_t i = 0; i < mSupportedDevices.size(); i++) {
- mSupportedDevices[i]->dump(fd, 6, i);
- }
-}
-
-void AudioPolicyManager::IOProfile::log()
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- ALOGV(" - sampling rates: ");
- for (size_t i = 0; i < mSamplingRates.size(); i++) {
- ALOGV(" %d", mSamplingRates[i]);
- }
-
- ALOGV(" - channel masks: ");
- for (size_t i = 0; i < mChannelMasks.size(); i++) {
- ALOGV(" 0x%04x", mChannelMasks[i]);
- }
-
- ALOGV(" - formats: ");
- for (size_t i = 0; i < mFormats.size(); i++) {
- ALOGV(" 0x%08x", mFormats[i]);
- }
-
- ALOGV(" - devices: 0x%04x\n", mSupportedDevices.types());
- ALOGV(" - flags: 0x%04x\n", mFlags);
-}
-
-
-// --- DeviceDescriptor implementation
-
-
-AudioPolicyManager::DeviceDescriptor::DeviceDescriptor(const String8& name, audio_devices_t type) :
- AudioPort(name, AUDIO_PORT_TYPE_DEVICE,
- audio_is_output_device(type) ? AUDIO_PORT_ROLE_SINK :
- AUDIO_PORT_ROLE_SOURCE,
- NULL),
- mDeviceType(type), mAddress(""), mId(0)
-{
-}
-
-bool AudioPolicyManager::DeviceDescriptor::equals(const sp<DeviceDescriptor>& other) const
-{
- // Devices are considered equal if they:
- // - are of the same type (a device type cannot be AUDIO_DEVICE_NONE)
- // - have the same address or one device does not specify the address
- // - have the same channel mask or one device does not specify the channel mask
- return (mDeviceType == other->mDeviceType) &&
- (mAddress == "" || other->mAddress == "" || mAddress == other->mAddress) &&
- (mChannelMask == 0 || other->mChannelMask == 0 ||
- mChannelMask == other->mChannelMask);
-}
-
-void AudioPolicyManager::DeviceDescriptor::loadGains(cnode *root)
-{
- AudioPort::loadGains(root);
- if (mGains.size() > 0) {
- mGains[0]->getDefaultConfig(&mGain);
- }
-}
-
-
-void AudioPolicyManager::DeviceVector::refreshTypes()
-{
- mDeviceTypes = AUDIO_DEVICE_NONE;
- for(size_t i = 0; i < size(); i++) {
- mDeviceTypes |= itemAt(i)->mDeviceType;
- }
- ALOGV("DeviceVector::refreshTypes() mDeviceTypes %08x", mDeviceTypes);
-}
-
-ssize_t AudioPolicyManager::DeviceVector::indexOf(const sp<DeviceDescriptor>& item) const
-{
- for(size_t i = 0; i < size(); i++) {
- if (item->equals(itemAt(i))) {
- return i;
- }
- }
- return -1;
-}
-
-ssize_t AudioPolicyManager::DeviceVector::add(const sp<DeviceDescriptor>& item)
-{
- ssize_t ret = indexOf(item);
-
- if (ret < 0) {
- ret = SortedVector::add(item);
- if (ret >= 0) {
- refreshTypes();
- }
- } else {
- ALOGW("DeviceVector::add device %08x already in", item->mDeviceType);
- ret = -1;
- }
- return ret;
-}
-
-ssize_t AudioPolicyManager::DeviceVector::remove(const sp<DeviceDescriptor>& item)
-{
- size_t i;
- ssize_t ret = indexOf(item);
-
- if (ret < 0) {
- ALOGW("DeviceVector::remove device %08x not in", item->mDeviceType);
- } else {
- ret = SortedVector::removeAt(ret);
- if (ret >= 0) {
- refreshTypes();
- }
- }
- return ret;
-}
-
-void AudioPolicyManager::DeviceVector::loadDevicesFromType(audio_devices_t types)
-{
- DeviceVector deviceList;
-
- uint32_t role_bit = AUDIO_DEVICE_BIT_IN & types;
- types &= ~role_bit;
-
- while (types) {
- uint32_t i = 31 - __builtin_clz(types);
- uint32_t type = 1 << i;
- types &= ~type;
- add(new DeviceDescriptor(String8(""), type | role_bit));
- }
-}
-
-void AudioPolicyManager::DeviceVector::loadDevicesFromName(char *name,
- const DeviceVector& declaredDevices)
-{
- char *devName = strtok(name, "|");
- while (devName != NULL) {
- if (strlen(devName) != 0) {
- audio_devices_t type = stringToEnum(sDeviceNameToEnumTable,
- ARRAY_SIZE(sDeviceNameToEnumTable),
- devName);
- if (type != AUDIO_DEVICE_NONE) {
- sp<DeviceDescriptor> dev = new DeviceDescriptor(String8(""), type);
- if (type == AUDIO_DEVICE_IN_REMOTE_SUBMIX ||
- type == AUDIO_DEVICE_OUT_REMOTE_SUBMIX ) {
- dev->mAddress = String8("0");
- }
- add(dev);
- } else {
- sp<DeviceDescriptor> deviceDesc =
- declaredDevices.getDeviceFromName(String8(devName));
- if (deviceDesc != 0) {
- add(deviceDesc);
- }
- }
- }
- devName = strtok(NULL, "|");
- }
-}
-
-sp<AudioPolicyManager::DeviceDescriptor> AudioPolicyManager::DeviceVector::getDevice(
- audio_devices_t type, String8 address) const
-{
- sp<DeviceDescriptor> device;
- for (size_t i = 0; i < size(); i++) {
- if (itemAt(i)->mDeviceType == type) {
- if (address == "" || itemAt(i)->mAddress == address) {
- device = itemAt(i);
- if (itemAt(i)->mAddress == address) {
- break;
- }
- }
- }
- }
- ALOGV("DeviceVector::getDevice() for type %08x address %s found %p",
- type, address.string(), device.get());
- return device;
-}
-
-sp<AudioPolicyManager::DeviceDescriptor> AudioPolicyManager::DeviceVector::getDeviceFromId(
- audio_port_handle_t id) const
-{
- sp<DeviceDescriptor> device;
- for (size_t i = 0; i < size(); i++) {
- ALOGV("DeviceVector::getDeviceFromId(%d) itemAt(%zu)->mId %d", id, i, itemAt(i)->mId);
- if (itemAt(i)->mId == id) {
- device = itemAt(i);
- break;
- }
- }
- return device;
-}
-
-AudioPolicyManager::DeviceVector AudioPolicyManager::DeviceVector::getDevicesFromType(
- audio_devices_t type) const
-{
- DeviceVector devices;
- for (size_t i = 0; (i < size()) && (type != AUDIO_DEVICE_NONE); i++) {
- if (itemAt(i)->mDeviceType & type & ~AUDIO_DEVICE_BIT_IN) {
- devices.add(itemAt(i));
- type &= ~itemAt(i)->mDeviceType;
- ALOGV("DeviceVector::getDevicesFromType() for type %x found %p",
- itemAt(i)->mDeviceType, itemAt(i).get());
- }
- }
- return devices;
-}
-
-AudioPolicyManager::DeviceVector AudioPolicyManager::DeviceVector::getDevicesFromTypeAddr(
- audio_devices_t type, String8 address) const
-{
- DeviceVector devices;
- for (size_t i = 0; i < size(); i++) {
- if (itemAt(i)->mDeviceType == type) {
- if (itemAt(i)->mAddress == address) {
- devices.add(itemAt(i));
- }
- }
- }
- return devices;
-}
-
-sp<AudioPolicyManager::DeviceDescriptor> AudioPolicyManager::DeviceVector::getDeviceFromName(
- const String8& name) const
-{
- sp<DeviceDescriptor> device;
- for (size_t i = 0; i < size(); i++) {
- if (itemAt(i)->mName == name) {
- device = itemAt(i);
- break;
- }
- }
- return device;
-}
-
-void AudioPolicyManager::DeviceDescriptor::toAudioPortConfig(
- struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig) const
-{
- dstConfig->config_mask = AUDIO_PORT_CONFIG_CHANNEL_MASK|AUDIO_PORT_CONFIG_GAIN;
- if (srcConfig != NULL) {
- dstConfig->config_mask |= srcConfig->config_mask;
- }
-
- AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
-
- dstConfig->id = mId;
- dstConfig->role = audio_is_output_device(mDeviceType) ?
- AUDIO_PORT_ROLE_SINK : AUDIO_PORT_ROLE_SOURCE;
- dstConfig->type = AUDIO_PORT_TYPE_DEVICE;
- dstConfig->ext.device.type = mDeviceType;
- dstConfig->ext.device.hw_module = mModule->mHandle;
- strncpy(dstConfig->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN);
-}
-
-void AudioPolicyManager::DeviceDescriptor::toAudioPort(struct audio_port *port) const
-{
- ALOGV("DeviceDescriptor::toAudioPort() handle %d type %x", mId, mDeviceType);
- AudioPort::toAudioPort(port);
- port->id = mId;
- toAudioPortConfig(&port->active_config);
- port->ext.device.type = mDeviceType;
- port->ext.device.hw_module = mModule->mHandle;
- strncpy(port->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN);
-}
-
-status_t AudioPolicyManager::DeviceDescriptor::dump(int fd, int spaces, int index) const
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, "%*sDevice %d:\n", spaces, "", index+1);
- result.append(buffer);
- if (mId != 0) {
- snprintf(buffer, SIZE, "%*s- id: %2d\n", spaces, "", mId);
- result.append(buffer);
- }
- snprintf(buffer, SIZE, "%*s- type: %-48s\n", spaces, "",
- enumToString(sDeviceNameToEnumTable,
- ARRAY_SIZE(sDeviceNameToEnumTable),
- mDeviceType));
- result.append(buffer);
- if (mAddress.size() != 0) {
- snprintf(buffer, SIZE, "%*s- address: %-32s\n", spaces, "", mAddress.string());
- result.append(buffer);
- }
- write(fd, result.string(), result.size());
- AudioPort::dump(fd, spaces);
-
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::AudioPatch::dump(int fd, int spaces, int index) const
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
-
- snprintf(buffer, SIZE, "%*sAudio patch %d:\n", spaces, "", index+1);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- handle: %2d\n", spaces, "", mHandle);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- audio flinger handle: %2d\n", spaces, "", mAfPatchHandle);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- owner uid: %2d\n", spaces, "", mUid);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- %d sources:\n", spaces, "", mPatch.num_sources);
- result.append(buffer);
- for (size_t i = 0; i < mPatch.num_sources; i++) {
- if (mPatch.sources[i].type == AUDIO_PORT_TYPE_DEVICE) {
- snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "",
- mPatch.sources[i].id, enumToString(sDeviceNameToEnumTable,
- ARRAY_SIZE(sDeviceNameToEnumTable),
- mPatch.sources[i].ext.device.type));
- } else {
- snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "",
- mPatch.sources[i].id, mPatch.sources[i].ext.mix.handle);
- }
- result.append(buffer);
- }
- snprintf(buffer, SIZE, "%*s- %d sinks:\n", spaces, "", mPatch.num_sinks);
- result.append(buffer);
- for (size_t i = 0; i < mPatch.num_sinks; i++) {
- if (mPatch.sinks[i].type == AUDIO_PORT_TYPE_DEVICE) {
- snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "",
- mPatch.sinks[i].id, enumToString(sDeviceNameToEnumTable,
- ARRAY_SIZE(sDeviceNameToEnumTable),
- mPatch.sinks[i].ext.device.type));
- } else {
- snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "",
- mPatch.sinks[i].id, mPatch.sinks[i].ext.mix.handle);
- }
- result.append(buffer);
- }
-
- write(fd, result.string(), result.size());
- return NO_ERROR;
-}
// --- audio_policy.conf file parsing
-
-uint32_t AudioPolicyManager::parseOutputFlagNames(char *name)
-{
- uint32_t flag = 0;
-
- // it is OK to cast name to non const here as we are not going to use it after
- // strtok() modifies it
- char *flagName = strtok(name, "|");
- while (flagName != NULL) {
- if (strlen(flagName) != 0) {
- flag |= stringToEnum(sOutputFlagNameToEnumTable,
- ARRAY_SIZE(sOutputFlagNameToEnumTable),
- flagName);
- }
- flagName = strtok(NULL, "|");
- }
- //force direct flag if offload flag is set: offloading implies a direct output stream
- // and all common behaviors are driven by checking only the direct flag
- // this should normally be set appropriately in the policy configuration file
- if ((flag & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
- flag |= AUDIO_OUTPUT_FLAG_DIRECT;
- }
-
- return flag;
-}
-
-uint32_t AudioPolicyManager::parseInputFlagNames(char *name)
-{
- uint32_t flag = 0;
-
- // it is OK to cast name to non const here as we are not going to use it after
- // strtok() modifies it
- char *flagName = strtok(name, "|");
- while (flagName != NULL) {
- if (strlen(flagName) != 0) {
- flag |= stringToEnum(sInputFlagNameToEnumTable,
- ARRAY_SIZE(sInputFlagNameToEnumTable),
- flagName);
- }
- flagName = strtok(NULL, "|");
- }
- return flag;
-}
-
-audio_devices_t AudioPolicyManager::parseDeviceNames(char *name)
-{
- uint32_t device = 0;
-
- char *devName = strtok(name, "|");
- while (devName != NULL) {
- if (strlen(devName) != 0) {
- device |= stringToEnum(sDeviceNameToEnumTable,
- ARRAY_SIZE(sDeviceNameToEnumTable),
- devName);
- }
- devName = strtok(NULL, "|");
- }
- return device;
-}
-
+// TODO candidate to be moved to ConfigParsingUtils
void AudioPolicyManager::loadHwModule(cnode *root)
{
status_t status = NAME_NOT_FOUND;
@@ -7889,6 +5557,7 @@
}
}
+// TODO candidate to be moved to ConfigParsingUtils
void AudioPolicyManager::loadHwModules(cnode *root)
{
cnode *node = config_find(root, AUDIO_HW_MODULE_TAG);
@@ -7904,6 +5573,7 @@
}
}
+// TODO candidate to be moved to ConfigParsingUtils
void AudioPolicyManager::loadGlobalConfig(cnode *root, const sp<HwModule>& module)
{
cnode *node = config_find(root, GLOBAL_CONFIG_TAG);
@@ -7924,11 +5594,12 @@
ALOGV("loadGlobalConfig() Attached Output Devices %08x",
mAvailableOutputDevices.types());
} else if (strcmp(DEFAULT_OUTPUT_DEVICE_TAG, node->name) == 0) {
- audio_devices_t device = (audio_devices_t)stringToEnum(sDeviceNameToEnumTable,
- ARRAY_SIZE(sDeviceNameToEnumTable),
- (char *)node->value);
+ audio_devices_t device = (audio_devices_t)ConfigParsingUtils::stringToEnum(
+ sDeviceNameToEnumTable,
+ ARRAY_SIZE(sDeviceNameToEnumTable),
+ (char *)node->value);
if (device != AUDIO_DEVICE_NONE) {
- mDefaultOutputDevice = new DeviceDescriptor(String8(""), device);
+ mDefaultOutputDevice = new DeviceDescriptor(String8("default-output"), device);
} else {
ALOGW("loadGlobalConfig() default device not specified");
}
@@ -7938,7 +5609,7 @@
declaredDevices);
ALOGV("loadGlobalConfig() Available InputDevices %08x", mAvailableInputDevices.types());
} else if (strcmp(SPEAKER_DRC_ENABLED_TAG, node->name) == 0) {
- mSpeakerDrcEnabled = stringToBool((char *)node->value);
+ mSpeakerDrcEnabled = ConfigParsingUtils::stringToBool((char *)node->value);
ALOGV("loadGlobalConfig() mSpeakerDrcEnabled = %d", mSpeakerDrcEnabled);
} else if (strcmp(AUDIO_HAL_VERSION_TAG, node->name) == 0) {
uint32_t major, minor;
@@ -7951,6 +5622,7 @@
}
}
+// TODO candidate to be moved to ConfigParsingUtils
status_t AudioPolicyManager::loadAudioPolicyConfig(const char *path)
{
cnode *root;
@@ -7979,8 +5651,8 @@
{
sp<HwModule> module;
sp<IOProfile> profile;
- sp<DeviceDescriptor> defaultInputDevice = new DeviceDescriptor(String8(""),
- AUDIO_DEVICE_IN_BUILTIN_MIC);
+ sp<DeviceDescriptor> defaultInputDevice =
+ new DeviceDescriptor(String8("builtin-mic"), AUDIO_DEVICE_IN_BUILTIN_MIC);
mAvailableOutputDevices.add(mDefaultOutputDevice);
mAvailableInputDevices.add(defaultInputDevice);
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
new file mode 100644
index 0000000..61ea6f2
--- /dev/null
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -0,0 +1,560 @@
+/*
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+
+#include <stdint.h>
+#include <sys/types.h>
+#include <cutils/config_utils.h>
+#include <cutils/misc.h>
+#include <utils/Timers.h>
+#include <utils/Errors.h>
+#include <utils/KeyedVector.h>
+#include <utils/SortedVector.h>
+#include <media/AudioPolicy.h>
+#include "AudioPolicyInterface.h"
+
+#include "Gains.h"
+#include "Ports.h"
+#include "ConfigParsingUtils.h"
+#include "Devices.h"
+#include "IOProfile.h"
+#include "HwModule.h"
+#include "AudioInputDescriptor.h"
+#include "AudioOutputDescriptor.h"
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+// Attenuation applied to STRATEGY_SONIFICATION streams when a headset is connected: 6dB
+#define SONIFICATION_HEADSET_VOLUME_FACTOR 0.5
+// Min volume for STRATEGY_SONIFICATION streams when limited by music volume: -36dB
+#define SONIFICATION_HEADSET_VOLUME_MIN 0.016
+// Time in milliseconds during which we consider that music is still active after a music
+// track was stopped - see computeVolume()
+#define SONIFICATION_HEADSET_MUSIC_DELAY 5000
+// Time in milliseconds after media stopped playing during which we consider that the
+// sonification should be as unobtrusive as during the time media was playing.
+#define SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY 5000
+// Time in milliseconds during witch some streams are muted while the audio path
+// is switched
+#define MUTE_TIME_MS 2000
+
+#define NUM_TEST_OUTPUTS 5
+
+#define NUM_VOL_CURVE_KNEES 2
+
+// Default minimum length allowed for offloading a compressed track
+// Can be overridden by the audio.offload.min.duration.secs property
+#define OFFLOAD_DEFAULT_MIN_DURATION_SECS 60
+
+#define MAX_MIXER_SAMPLING_RATE 48000
+#define MAX_MIXER_CHANNEL_COUNT 8
+
+// ----------------------------------------------------------------------------
+// AudioPolicyManager implements audio policy manager behavior common to all platforms.
+// ----------------------------------------------------------------------------
+
+class AudioPolicyManager: public AudioPolicyInterface
+#ifdef AUDIO_POLICY_TEST
+ , public Thread
+#endif //AUDIO_POLICY_TEST
+{
+
+public:
+ AudioPolicyManager(AudioPolicyClientInterface *clientInterface);
+ virtual ~AudioPolicyManager();
+
+ // AudioPolicyInterface
+ virtual status_t setDeviceConnectionState(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ const char *device_address,
+ const char *device_name);
+ virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
+ const char *device_address);
+ virtual void setPhoneState(audio_mode_t state);
+ virtual void setForceUse(audio_policy_force_use_t usage,
+ audio_policy_forced_cfg_t config);
+ virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
+ virtual void setSystemProperty(const char* property, const char* value);
+ virtual status_t initCheck();
+ virtual audio_io_handle_t getOutput(audio_stream_type_t stream,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo);
+ virtual status_t getOutputForAttr(const audio_attributes_t *attr,
+ audio_io_handle_t *output,
+ audio_session_t session,
+ audio_stream_type_t *stream,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo);
+ virtual status_t startOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ audio_session_t session);
+ virtual status_t stopOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ audio_session_t session);
+ virtual void releaseOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ audio_session_t session);
+ virtual status_t getInputForAttr(const audio_attributes_t *attr,
+ audio_io_handle_t *input,
+ audio_session_t session,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_input_flags_t flags,
+ input_type_t *inputType);
+
+ // indicates to the audio policy manager that the input starts being used.
+ virtual status_t startInput(audio_io_handle_t input,
+ audio_session_t session);
+
+ // indicates to the audio policy manager that the input stops being used.
+ virtual status_t stopInput(audio_io_handle_t input,
+ audio_session_t session);
+ virtual void releaseInput(audio_io_handle_t input,
+ audio_session_t session);
+ virtual void closeAllInputs();
+ virtual void initStreamVolume(audio_stream_type_t stream,
+ int indexMin,
+ int indexMax);
+ virtual status_t setStreamVolumeIndex(audio_stream_type_t stream,
+ int index,
+ audio_devices_t device);
+ virtual status_t getStreamVolumeIndex(audio_stream_type_t stream,
+ int *index,
+ audio_devices_t device);
+
+ // return the strategy corresponding to a given stream type
+ virtual uint32_t getStrategyForStream(audio_stream_type_t stream);
+ // return the strategy corresponding to the given audio attributes
+ virtual uint32_t getStrategyForAttr(const audio_attributes_t *attr);
+
+ // return the enabled output devices for the given stream type
+ virtual audio_devices_t getDevicesForStream(audio_stream_type_t stream);
+
+ virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc = NULL);
+ virtual status_t registerEffect(const effect_descriptor_t *desc,
+ audio_io_handle_t io,
+ uint32_t strategy,
+ int session,
+ int id);
+ virtual status_t unregisterEffect(int id);
+ virtual status_t setEffectEnabled(int id, bool enabled);
+
+ virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const;
+ // return whether a stream is playing remotely, override to change the definition of
+ // local/remote playback, used for instance by notification manager to not make
+ // media players lose audio focus when not playing locally
+ // For the base implementation, "remotely" means playing during screen mirroring which
+ // uses an output for playback with a non-empty, non "0" address.
+ virtual bool isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs = 0) const;
+ virtual bool isSourceActive(audio_source_t source) const;
+
+ virtual status_t dump(int fd);
+
+ virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo);
+
+ virtual status_t listAudioPorts(audio_port_role_t role,
+ audio_port_type_t type,
+ unsigned int *num_ports,
+ struct audio_port *ports,
+ unsigned int *generation);
+ virtual status_t getAudioPort(struct audio_port *port);
+ virtual status_t createAudioPatch(const struct audio_patch *patch,
+ audio_patch_handle_t *handle,
+ uid_t uid);
+ virtual status_t releaseAudioPatch(audio_patch_handle_t handle,
+ uid_t uid);
+ virtual status_t listAudioPatches(unsigned int *num_patches,
+ struct audio_patch *patches,
+ unsigned int *generation);
+ virtual status_t setAudioPortConfig(const struct audio_port_config *config);
+ virtual void clearAudioPatches(uid_t uid);
+
+ virtual status_t acquireSoundTriggerSession(audio_session_t *session,
+ audio_io_handle_t *ioHandle,
+ audio_devices_t *device);
+
+ virtual status_t releaseSoundTriggerSession(audio_session_t session);
+
+ virtual status_t registerPolicyMixes(Vector<AudioMix> mixes);
+ virtual status_t unregisterPolicyMixes(Vector<AudioMix> mixes);
+
+ // Audio policy configuration file parsing (audio_policy.conf)
+ // TODO candidates to be moved to ConfigParsingUtils
+ void loadHwModule(cnode *root);
+ void loadHwModules(cnode *root);
+ void loadGlobalConfig(cnode *root, const sp<HwModule>& module);
+ status_t loadAudioPolicyConfig(const char *path);
+ void defaultAudioPolicyConfig(void);
+
+ // return the strategy corresponding to a given stream type
+ static routing_strategy getStrategy(audio_stream_type_t stream);
+
+ static uint32_t nextUniqueId();
+protected:
+
+ class EffectDescriptor : public RefBase
+ {
+ public:
+
+ status_t dump(int fd);
+
+ int mIo; // io the effect is attached to
+ routing_strategy mStrategy; // routing strategy the effect is associated to
+ int mSession; // audio session the effect is on
+ effect_descriptor_t mDesc; // effect descriptor
+ bool mEnabled; // enabled state: CPU load being used or not
+ };
+
+ void addOutput(audio_io_handle_t output, sp<AudioOutputDescriptor> outputDesc);
+ void addInput(audio_io_handle_t input, sp<AudioInputDescriptor> inputDesc);
+
+ // return appropriate device for streams handled by the specified strategy according to current
+ // phone state, connected devices...
+ // if fromCache is true, the device is returned from mDeviceForStrategy[],
+ // otherwise it is determine by current state
+ // (device connected,phone state, force use, a2dp output...)
+ // This allows to:
+ // 1 speed up process when the state is stable (when starting or stopping an output)
+ // 2 access to either current device selection (fromCache == true) or
+ // "future" device selection (fromCache == false) when called from a context
+ // where conditions are changing (setDeviceConnectionState(), setPhoneState()...) AND
+ // before updateDevicesAndOutputs() is called.
+ virtual audio_devices_t getDeviceForStrategy(routing_strategy strategy,
+ bool fromCache);
+
+ // change the route of the specified output. Returns the number of ms we have slept to
+ // allow new routing to take effect in certain cases.
+ virtual uint32_t setOutputDevice(audio_io_handle_t output,
+ audio_devices_t device,
+ bool force = false,
+ int delayMs = 0,
+ audio_patch_handle_t *patchHandle = NULL,
+ const char* address = NULL);
+ status_t resetOutputDevice(audio_io_handle_t output,
+ int delayMs = 0,
+ audio_patch_handle_t *patchHandle = NULL);
+ status_t setInputDevice(audio_io_handle_t input,
+ audio_devices_t device,
+ bool force = false,
+ audio_patch_handle_t *patchHandle = NULL);
+ status_t resetInputDevice(audio_io_handle_t input,
+ audio_patch_handle_t *patchHandle = NULL);
+
+ // select input device corresponding to requested audio source
+ virtual audio_devices_t getDeviceForInputSource(audio_source_t inputSource);
+
+ // return io handle of active input or 0 if no input is active
+ // Only considers inputs from physical devices (e.g. main mic, headset mic) when
+ // ignoreVirtualInputs is true.
+ audio_io_handle_t getActiveInput(bool ignoreVirtualInputs = true);
+
+ uint32_t activeInputsCount() const;
+
+ // initialize volume curves for each strategy and device category
+ void initializeVolumeCurves();
+
+ // compute the actual volume for a given stream according to the requested index and a particular
+ // device
+ virtual float computeVolume(audio_stream_type_t stream, int index,
+ audio_io_handle_t output, audio_devices_t device);
+
+ // check that volume change is permitted, compute and send new volume to audio hardware
+ virtual status_t checkAndSetVolume(audio_stream_type_t stream, int index,
+ audio_io_handle_t output,
+ audio_devices_t device,
+ int delayMs = 0, bool force = false);
+
+ // apply all stream volumes to the specified output and device
+ void applyStreamVolumes(audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false);
+
+ // Mute or unmute all streams handled by the specified strategy on the specified output
+ void setStrategyMute(routing_strategy strategy,
+ bool on,
+ audio_io_handle_t output,
+ int delayMs = 0,
+ audio_devices_t device = (audio_devices_t)0);
+
+ // Mute or unmute the stream on the specified output
+ void setStreamMute(audio_stream_type_t stream,
+ bool on,
+ audio_io_handle_t output,
+ int delayMs = 0,
+ audio_devices_t device = (audio_devices_t)0);
+
+ // handle special cases for sonification strategy while in call: mute streams or replace by
+ // a special tone in the device used for communication
+ void handleIncallSonification(audio_stream_type_t stream, bool starting, bool stateChange);
+
+ // true if device is in a telephony or VoIP call
+ virtual bool isInCall();
+
+ // true if given state represents a device in a telephony or VoIP call
+ virtual bool isStateInCall(int state);
+
+ // when a device is connected, checks if an open output can be routed
+ // to this device. If none is open, tries to open one of the available outputs.
+ // Returns an output suitable to this device or 0.
+ // when a device is disconnected, checks if an output is not used any more and
+ // returns its handle if any.
+ // transfers the audio tracks and effects from one output thread to another accordingly.
+ status_t checkOutputsForDevice(const sp<DeviceDescriptor> devDesc,
+ audio_policy_dev_state_t state,
+ SortedVector<audio_io_handle_t>& outputs,
+ const String8 address);
+
+ status_t checkInputsForDevice(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ SortedVector<audio_io_handle_t>& inputs,
+ const String8 address);
+
+ // close an output and its companion duplicating output.
+ void closeOutput(audio_io_handle_t output);
+
+ // close an input.
+ void closeInput(audio_io_handle_t input);
+
+ // checks and if necessary changes outputs used for all strategies.
+ // must be called every time a condition that affects the output choice for a given strategy
+ // changes: connected device, phone state, force use...
+ // Must be called before updateDevicesAndOutputs()
+ void checkOutputForStrategy(routing_strategy strategy);
+
+ // Same as checkOutputForStrategy() but for a all strategies in order of priority
+ void checkOutputForAllStrategies();
+
+ // manages A2DP output suspend/restore according to phone state and BT SCO usage
+ void checkA2dpSuspend();
+
+ // returns the A2DP output handle if it is open or 0 otherwise
+ audio_io_handle_t getA2dpOutput();
+
+ // selects the most appropriate device on output for current state
+ // must be called every time a condition that affects the device choice for a given output is
+ // changed: connected device, phone state, force use, output start, output stop..
+ // see getDeviceForStrategy() for the use of fromCache parameter
+ audio_devices_t getNewOutputDevice(audio_io_handle_t output, bool fromCache);
+
+ // updates cache of device used by all strategies (mDeviceForStrategy[])
+ // must be called every time a condition that affects the device choice for a given strategy is
+ // changed: connected device, phone state, force use...
+ // cached values are used by getDeviceForStrategy() if parameter fromCache is true.
+ // Must be called after checkOutputForAllStrategies()
+ void updateDevicesAndOutputs();
+
+ // selects the most appropriate device on input for current state
+ audio_devices_t getNewInputDevice(audio_io_handle_t input);
+
+ virtual uint32_t getMaxEffectsCpuLoad();
+ virtual uint32_t getMaxEffectsMemory();
+#ifdef AUDIO_POLICY_TEST
+ virtual bool threadLoop();
+ void exit();
+ int testOutputIndex(audio_io_handle_t output);
+#endif //AUDIO_POLICY_TEST
+
+ status_t setEffectEnabled(const sp<EffectDescriptor>& effectDesc, bool enabled);
+
+ SortedVector<audio_io_handle_t> getOutputsForDevice(audio_devices_t device,
+ DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > openOutputs);
+ bool vectorsEqual(SortedVector<audio_io_handle_t>& outputs1,
+ SortedVector<audio_io_handle_t>& outputs2);
+
+ // mute/unmute strategies using an incompatible device combination
+ // if muting, wait for the audio in pcm buffer to be drained before proceeding
+ // if unmuting, unmute only after the specified delay
+ // Returns the number of ms waited
+ virtual uint32_t checkDeviceMuteStrategies(sp<AudioOutputDescriptor> outputDesc,
+ audio_devices_t prevDevice,
+ uint32_t delayMs);
+
+ audio_io_handle_t selectOutput(const SortedVector<audio_io_handle_t>& outputs,
+ audio_output_flags_t flags,
+ audio_format_t format);
+ // samplingRate parameter is an in/out and so may be modified
+ sp<IOProfile> getInputProfile(audio_devices_t device,
+ String8 address,
+ uint32_t& samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_input_flags_t flags);
+ sp<IOProfile> getProfileForDirectOutput(audio_devices_t device,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags);
+
+ audio_io_handle_t selectOutputForEffects(const SortedVector<audio_io_handle_t>& outputs);
+
+ bool isNonOffloadableEffectEnabled();
+
+ virtual status_t addAudioPatch(audio_patch_handle_t handle,
+ const sp<AudioPatch>& patch);
+ virtual status_t removeAudioPatch(audio_patch_handle_t handle);
+
+ sp<AudioOutputDescriptor> getOutputFromId(audio_port_handle_t id) const;
+ sp<AudioInputDescriptor> getInputFromId(audio_port_handle_t id) const;
+ sp<HwModule> getModuleForDevice(audio_devices_t device) const;
+ sp<HwModule> getModuleFromName(const char *name) const;
+ audio_devices_t availablePrimaryOutputDevices();
+ audio_devices_t availablePrimaryInputDevices();
+
+ void updateCallRouting(audio_devices_t rxDevice, int delayMs = 0);
+
+
+ uid_t mUidCached;
+ AudioPolicyClientInterface *mpClientInterface; // audio policy client interface
+ audio_io_handle_t mPrimaryOutput; // primary output handle
+ // list of descriptors for outputs currently opened
+ DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > mOutputs;
+ // copy of mOutputs before setDeviceConnectionState() opens new outputs
+ // reset to mOutputs when updateDevicesAndOutputs() is called.
+ DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > mPreviousOutputs;
+ DefaultKeyedVector<audio_io_handle_t, sp<AudioInputDescriptor> > mInputs; // list of input descriptors
+ DeviceVector mAvailableOutputDevices; // all available output devices
+ DeviceVector mAvailableInputDevices; // all available input devices
+ int mPhoneState; // current phone state
+ audio_policy_forced_cfg_t mForceUse[AUDIO_POLICY_FORCE_USE_CNT]; // current forced use configuration
+
+ StreamDescriptor mStreams[AUDIO_STREAM_CNT]; // stream descriptors for volume control
+ bool mLimitRingtoneVolume; // limit ringtone volume to music volume if headset connected
+ audio_devices_t mDeviceForStrategy[NUM_STRATEGIES];
+ float mLastVoiceVolume; // last voice volume value sent to audio HAL
+
+ // Maximum CPU load allocated to audio effects in 0.1 MIPS (ARMv5TE, 0 WS memory) units
+ static const uint32_t MAX_EFFECTS_CPU_LOAD = 1000;
+ // Maximum memory allocated to audio effects in KB
+ static const uint32_t MAX_EFFECTS_MEMORY = 512;
+ uint32_t mTotalEffectsCpuLoad; // current CPU load used by effects
+ uint32_t mTotalEffectsMemory; // current memory used by effects
+ KeyedVector<int, sp<EffectDescriptor> > mEffects; // list of registered audio effects
+ bool mA2dpSuspended; // true if A2DP output is suspended
+ sp<DeviceDescriptor> mDefaultOutputDevice; // output device selected by default at boot time
+ bool mSpeakerDrcEnabled;// true on devices that use DRC on the DEVICE_CATEGORY_SPEAKER path
+ // to boost soft sounds, used to adjust volume curves accordingly
+
+ Vector < sp<HwModule> > mHwModules;
+ static volatile int32_t mNextUniqueId;
+ volatile int32_t mAudioPortGeneration;
+
+ DefaultKeyedVector<audio_patch_handle_t, sp<AudioPatch> > mAudioPatches;
+
+ DefaultKeyedVector<audio_session_t, audio_io_handle_t> mSoundTriggerSessions;
+
+ sp<AudioPatch> mCallTxPatch;
+ sp<AudioPatch> mCallRxPatch;
+
+ // for supporting "beacon" streams, i.e. streams that only play on speaker, and never
+ // when something other than STREAM_TTS (a.k.a. "Transmitted Through Speaker") is playing
+ enum {
+ STARTING_OUTPUT,
+ STARTING_BEACON,
+ STOPPING_OUTPUT,
+ STOPPING_BEACON
+ };
+ uint32_t mBeaconMuteRefCount; // ref count for stream that would mute beacon
+ uint32_t mBeaconPlayingRefCount;// ref count for the playing beacon streams
+ bool mBeaconMuted; // has STREAM_TTS been muted
+
+ // custom mix entry in mPolicyMixes
+ class AudioPolicyMix : public RefBase {
+ public:
+ AudioPolicyMix() {}
+
+ AudioMix mMix; // Audio policy mix descriptor
+ sp<AudioOutputDescriptor> mOutput; // Corresponding output stream
+ };
+ DefaultKeyedVector<String8, sp<AudioPolicyMix> > mPolicyMixes; // list of registered mixes
+
+
+#ifdef AUDIO_POLICY_TEST
+ Mutex mLock;
+ Condition mWaitWorkCV;
+
+ int mCurOutput;
+ bool mDirectOutput;
+ audio_io_handle_t mTestOutputs[NUM_TEST_OUTPUTS];
+ int mTestInput;
+ uint32_t mTestDevice;
+ uint32_t mTestSamplingRate;
+ uint32_t mTestFormat;
+ uint32_t mTestChannels;
+ uint32_t mTestLatencyMs;
+#endif //AUDIO_POLICY_TEST
+
+ static bool isVirtualInputDevice(audio_devices_t device);
+
+ uint32_t nextAudioPortGeneration();
+private:
+ // updates device caching and output for streams that can influence the
+ // routing of notifications
+ void handleNotificationRoutingForStream(audio_stream_type_t stream);
+ static bool deviceDistinguishesOnAddress(audio_devices_t device);
+ // find the outputs on a given output descriptor that have the given address.
+ // to be called on an AudioOutputDescriptor whose supported devices (as defined
+ // in mProfile->mSupportedDevices) matches the device whose address is to be matched.
+ // see deviceDistinguishesOnAddress(audio_devices_t) for whether the device type is one
+ // where addresses are used to distinguish between one connected device and another.
+ void findIoHandlesByAddress(sp<AudioOutputDescriptor> desc /*in*/,
+ const audio_devices_t device /*in*/,
+ const String8 address /*in*/,
+ SortedVector<audio_io_handle_t>& outputs /*out*/);
+ uint32_t curAudioPortGeneration() const { return mAudioPortGeneration; }
+ // internal method to return the output handle for the given device and format
+ audio_io_handle_t getOutputForDevice(
+ audio_devices_t device,
+ audio_session_t session,
+ audio_stream_type_t stream,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo);
+ // internal function to derive a stream type value from audio attributes
+ audio_stream_type_t streamTypefromAttributesInt(const audio_attributes_t *attr);
+ // return true if any output is playing anything besides the stream to ignore
+ bool isAnyOutputActive(audio_stream_type_t streamToIgnore);
+ // event is one of STARTING_OUTPUT, STARTING_BEACON, STOPPING_OUTPUT, STOPPING_BEACON
+ // returns 0 if no mute/unmute event happened, the largest latency of the device where
+ // the mute/unmute happened
+ uint32_t handleEventForBeacon(int event);
+ uint32_t setBeaconMute(bool mute);
+ bool isValidAttributes(const audio_attributes_t *paa);
+
+ // select input device corresponding to requested audio source and return associated policy
+ // mix if any. Calls getDeviceForInputSource().
+ audio_devices_t getDeviceAndMixForInputSource(audio_source_t inputSource,
+ AudioMix **policyMix = NULL);
+
+ // Called by setDeviceConnectionState().
+ status_t setDeviceConnectionStateInt(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ const char *device_address,
+ const char *device_name);
+ sp<DeviceDescriptor> getDeviceDescriptor(const audio_devices_t device,
+ const char *device_address,
+ const char *device_name);
+};
+
+};
diff --git a/services/audiopolicy/managerdefault/ConfigParsingUtils.cpp b/services/audiopolicy/managerdefault/ConfigParsingUtils.cpp
new file mode 100644
index 0000000..1afd487
--- /dev/null
+++ b/services/audiopolicy/managerdefault/ConfigParsingUtils.cpp
@@ -0,0 +1,121 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM::ConfigParsingUtils"
+//#define LOG_NDEBUG 0
+
+#include "AudioPolicyManager.h"
+
+namespace android {
+
+//static
+uint32_t ConfigParsingUtils::stringToEnum(const struct StringToEnum *table,
+ size_t size,
+ const char *name)
+{
+ for (size_t i = 0; i < size; i++) {
+ if (strcmp(table[i].name, name) == 0) {
+ ALOGV("stringToEnum() found %s", table[i].name);
+ return table[i].value;
+ }
+ }
+ return 0;
+}
+
+//static
+const char *ConfigParsingUtils::enumToString(const struct StringToEnum *table,
+ size_t size,
+ uint32_t value)
+{
+ for (size_t i = 0; i < size; i++) {
+ if (table[i].value == value) {
+ return table[i].name;
+ }
+ }
+ return "";
+}
+
+//static
+bool ConfigParsingUtils::stringToBool(const char *value)
+{
+ return ((strcasecmp("true", value) == 0) || (strcmp("1", value) == 0));
+}
+
+
+// --- audio_policy.conf file parsing
+//static
+uint32_t ConfigParsingUtils::parseOutputFlagNames(char *name)
+{
+ uint32_t flag = 0;
+
+ // it is OK to cast name to non const here as we are not going to use it after
+ // strtok() modifies it
+ char *flagName = strtok(name, "|");
+ while (flagName != NULL) {
+ if (strlen(flagName) != 0) {
+ flag |= ConfigParsingUtils::stringToEnum(sOutputFlagNameToEnumTable,
+ ARRAY_SIZE(sOutputFlagNameToEnumTable),
+ flagName);
+ }
+ flagName = strtok(NULL, "|");
+ }
+ //force direct flag if offload flag is set: offloading implies a direct output stream
+ // and all common behaviors are driven by checking only the direct flag
+ // this should normally be set appropriately in the policy configuration file
+ if ((flag & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+ flag |= AUDIO_OUTPUT_FLAG_DIRECT;
+ }
+
+ return flag;
+}
+
+//static
+uint32_t ConfigParsingUtils::parseInputFlagNames(char *name)
+{
+ uint32_t flag = 0;
+
+ // it is OK to cast name to non const here as we are not going to use it after
+ // strtok() modifies it
+ char *flagName = strtok(name, "|");
+ while (flagName != NULL) {
+ if (strlen(flagName) != 0) {
+ flag |= stringToEnum(sInputFlagNameToEnumTable,
+ ARRAY_SIZE(sInputFlagNameToEnumTable),
+ flagName);
+ }
+ flagName = strtok(NULL, "|");
+ }
+ return flag;
+}
+
+//static
+audio_devices_t ConfigParsingUtils::parseDeviceNames(char *name)
+{
+ uint32_t device = 0;
+
+ char *devName = strtok(name, "|");
+ while (devName != NULL) {
+ if (strlen(devName) != 0) {
+ device |= stringToEnum(sDeviceNameToEnumTable,
+ ARRAY_SIZE(sDeviceNameToEnumTable),
+ devName);
+ }
+ devName = strtok(NULL, "|");
+ }
+ return device;
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/ConfigParsingUtils.h b/services/audiopolicy/managerdefault/ConfigParsingUtils.h
new file mode 100644
index 0000000..7969661
--- /dev/null
+++ b/services/audiopolicy/managerdefault/ConfigParsingUtils.h
@@ -0,0 +1,159 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+// Definitions for audio_policy.conf file parsing
+// ----------------------------------------------------------------------------
+
+struct StringToEnum {
+ const char *name;
+ uint32_t value;
+};
+
+#define STRING_TO_ENUM(string) { #string, string }
+#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
+
+const StringToEnum sDeviceNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_EARPIECE),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER_SAFE),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADPHONE),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_SCO),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_A2DP),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_DIGITAL),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_ACCESSORY),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_DEVICE),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_USB),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_REMOTE_SUBMIX),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_TELEPHONY_TX),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_LINE),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI_ARC),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPDIF),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_FM),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_LINE),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_AMBIENT),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_BUILTIN_MIC),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_ALL_SCO),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_WIRED_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_AUX_DIGITAL),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_HDMI),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_TELEPHONY_RX),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_VOICE_CALL),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_BACK_MIC),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_REMOTE_SUBMIX),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_ACCESSORY),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_DEVICE),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_FM_TUNER),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_TV_TUNER),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_LINE),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_SPDIF),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_A2DP),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_LOOPBACK),
+};
+
+const StringToEnum sOutputFlagNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DIRECT),
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_PRIMARY),
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_FAST),
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DEEP_BUFFER),
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD),
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_NON_BLOCKING),
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_HW_AV_SYNC),
+};
+
+const StringToEnum sInputFlagNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_INPUT_FLAG_FAST),
+ STRING_TO_ENUM(AUDIO_INPUT_FLAG_HW_HOTWORD),
+};
+
+const StringToEnum sFormatNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT),
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_BIT),
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_32_BIT),
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_24_BIT),
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_FLOAT),
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_PACKED),
+ STRING_TO_ENUM(AUDIO_FORMAT_MP3),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_MAIN),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_LC),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_SSR),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_LTP),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V1),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_SCALABLE),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_ERLC),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_LD),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V2),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_ELD),
+ STRING_TO_ENUM(AUDIO_FORMAT_VORBIS),
+ STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V1),
+ STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V2),
+ STRING_TO_ENUM(AUDIO_FORMAT_OPUS),
+ STRING_TO_ENUM(AUDIO_FORMAT_AC3),
+ STRING_TO_ENUM(AUDIO_FORMAT_E_AC3),
+};
+
+const StringToEnum sOutChannelsNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_CHANNEL_OUT_MONO),
+ STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO),
+ STRING_TO_ENUM(AUDIO_CHANNEL_OUT_QUAD),
+ STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1),
+ STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1),
+};
+
+const StringToEnum sInChannelsNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_CHANNEL_IN_MONO),
+ STRING_TO_ENUM(AUDIO_CHANNEL_IN_STEREO),
+ STRING_TO_ENUM(AUDIO_CHANNEL_IN_FRONT_BACK),
+};
+
+const StringToEnum sGainModeNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_GAIN_MODE_JOINT),
+ STRING_TO_ENUM(AUDIO_GAIN_MODE_CHANNELS),
+ STRING_TO_ENUM(AUDIO_GAIN_MODE_RAMP),
+};
+
+class ConfigParsingUtils
+{
+public:
+ static uint32_t stringToEnum(const struct StringToEnum *table,
+ size_t size,
+ const char *name);
+ static const char *enumToString(const struct StringToEnum *table,
+ size_t size,
+ uint32_t value);
+ static bool stringToBool(const char *value);
+ static uint32_t parseOutputFlagNames(char *name);
+ static uint32_t parseInputFlagNames(char *name);
+ static audio_devices_t parseDeviceNames(char *name);
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/Devices.cpp b/services/audiopolicy/managerdefault/Devices.cpp
new file mode 100644
index 0000000..13c8bbc
--- /dev/null
+++ b/services/audiopolicy/managerdefault/Devices.cpp
@@ -0,0 +1,282 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM::Devices"
+//#define LOG_NDEBUG 0
+
+#include "AudioPolicyManager.h"
+
+namespace android {
+
+String8 DeviceDescriptor::emptyNameStr = String8("");
+
+DeviceDescriptor::DeviceDescriptor(const String8& name, audio_devices_t type) :
+ AudioPort(name, AUDIO_PORT_TYPE_DEVICE,
+ audio_is_output_device(type) ? AUDIO_PORT_ROLE_SINK :
+ AUDIO_PORT_ROLE_SOURCE,
+ NULL),
+ mDeviceType(type), mAddress("")
+{
+
+}
+
+bool DeviceDescriptor::equals(const sp<DeviceDescriptor>& other) const
+{
+ // Devices are considered equal if they:
+ // - are of the same type (a device type cannot be AUDIO_DEVICE_NONE)
+ // - have the same address or one device does not specify the address
+ // - have the same channel mask or one device does not specify the channel mask
+ return (mDeviceType == other->mDeviceType) &&
+ (mAddress == "" || other->mAddress == "" || mAddress == other->mAddress) &&
+ (mChannelMask == 0 || other->mChannelMask == 0 ||
+ mChannelMask == other->mChannelMask);
+}
+
+void DeviceDescriptor::loadGains(cnode *root)
+{
+ AudioPort::loadGains(root);
+ if (mGains.size() > 0) {
+ mGains[0]->getDefaultConfig(&mGain);
+ }
+}
+
+void DeviceVector::refreshTypes()
+{
+ mDeviceTypes = AUDIO_DEVICE_NONE;
+ for(size_t i = 0; i < size(); i++) {
+ mDeviceTypes |= itemAt(i)->mDeviceType;
+ }
+ ALOGV("DeviceVector::refreshTypes() mDeviceTypes %08x", mDeviceTypes);
+}
+
+ssize_t DeviceVector::indexOf(const sp<DeviceDescriptor>& item) const
+{
+ for(size_t i = 0; i < size(); i++) {
+ if (item->equals(itemAt(i))) {
+ return i;
+ }
+ }
+ return -1;
+}
+
+ssize_t DeviceVector::add(const sp<DeviceDescriptor>& item)
+{
+ ssize_t ret = indexOf(item);
+
+ if (ret < 0) {
+ ret = SortedVector::add(item);
+ if (ret >= 0) {
+ refreshTypes();
+ }
+ } else {
+ ALOGW("DeviceVector::add device %08x already in", item->mDeviceType);
+ ret = -1;
+ }
+ return ret;
+}
+
+ssize_t DeviceVector::remove(const sp<DeviceDescriptor>& item)
+{
+ size_t i;
+ ssize_t ret = indexOf(item);
+
+ if (ret < 0) {
+ ALOGW("DeviceVector::remove device %08x not in", item->mDeviceType);
+ } else {
+ ret = SortedVector::removeAt(ret);
+ if (ret >= 0) {
+ refreshTypes();
+ }
+ }
+ return ret;
+}
+
+void DeviceVector::loadDevicesFromType(audio_devices_t types)
+{
+ DeviceVector deviceList;
+
+ uint32_t role_bit = AUDIO_DEVICE_BIT_IN & types;
+ types &= ~role_bit;
+
+ while (types) {
+ uint32_t i = 31 - __builtin_clz(types);
+ uint32_t type = 1 << i;
+ types &= ~type;
+ add(new DeviceDescriptor(String8("device_type"), type | role_bit));
+ }
+}
+
+void DeviceVector::loadDevicesFromName(char *name,
+ const DeviceVector& declaredDevices)
+{
+ char *devName = strtok(name, "|");
+ while (devName != NULL) {
+ if (strlen(devName) != 0) {
+ audio_devices_t type = ConfigParsingUtils::stringToEnum(sDeviceNameToEnumTable,
+ ARRAY_SIZE(sDeviceNameToEnumTable),
+ devName);
+ if (type != AUDIO_DEVICE_NONE) {
+ sp<DeviceDescriptor> dev = new DeviceDescriptor(String8(name), type);
+ if (type == AUDIO_DEVICE_IN_REMOTE_SUBMIX ||
+ type == AUDIO_DEVICE_OUT_REMOTE_SUBMIX ) {
+ dev->mAddress = String8("0");
+ }
+ add(dev);
+ } else {
+ sp<DeviceDescriptor> deviceDesc =
+ declaredDevices.getDeviceFromName(String8(devName));
+ if (deviceDesc != 0) {
+ add(deviceDesc);
+ }
+ }
+ }
+ devName = strtok(NULL, "|");
+ }
+}
+
+sp<DeviceDescriptor> DeviceVector::getDevice(audio_devices_t type, String8 address) const
+{
+ sp<DeviceDescriptor> device;
+ for (size_t i = 0; i < size(); i++) {
+ if (itemAt(i)->mDeviceType == type) {
+ if (address == "" || itemAt(i)->mAddress == address) {
+ device = itemAt(i);
+ if (itemAt(i)->mAddress == address) {
+ break;
+ }
+ }
+ }
+ }
+ ALOGV("DeviceVector::getDevice() for type %08x address %s found %p",
+ type, address.string(), device.get());
+ return device;
+}
+
+sp<DeviceDescriptor> DeviceVector::getDeviceFromId(audio_port_handle_t id) const
+{
+ sp<DeviceDescriptor> device;
+ for (size_t i = 0; i < size(); i++) {
+ if (itemAt(i)->getHandle() == id) {
+ device = itemAt(i);
+ break;
+ }
+ }
+ return device;
+}
+
+DeviceVector DeviceVector::getDevicesFromType(audio_devices_t type) const
+{
+ DeviceVector devices;
+ for (size_t i = 0; (i < size()) && (type != AUDIO_DEVICE_NONE); i++) {
+ if (itemAt(i)->mDeviceType & type & ~AUDIO_DEVICE_BIT_IN) {
+ devices.add(itemAt(i));
+ type &= ~itemAt(i)->mDeviceType;
+ ALOGV("DeviceVector::getDevicesFromType() for type %x found %p",
+ itemAt(i)->mDeviceType, itemAt(i).get());
+ }
+ }
+ return devices;
+}
+
+DeviceVector DeviceVector::getDevicesFromTypeAddr(
+ audio_devices_t type, String8 address) const
+{
+ DeviceVector devices;
+ for (size_t i = 0; i < size(); i++) {
+ if (itemAt(i)->mDeviceType == type) {
+ if (itemAt(i)->mAddress == address) {
+ devices.add(itemAt(i));
+ }
+ }
+ }
+ return devices;
+}
+
+sp<DeviceDescriptor> DeviceVector::getDeviceFromName(const String8& name) const
+{
+ sp<DeviceDescriptor> device;
+ for (size_t i = 0; i < size(); i++) {
+ if (itemAt(i)->mName == name) {
+ device = itemAt(i);
+ break;
+ }
+ }
+ return device;
+}
+
+void DeviceDescriptor::toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig) const
+{
+ dstConfig->config_mask = AUDIO_PORT_CONFIG_CHANNEL_MASK|AUDIO_PORT_CONFIG_GAIN;
+ if (srcConfig != NULL) {
+ dstConfig->config_mask |= srcConfig->config_mask;
+ }
+
+ AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
+
+ dstConfig->id = mId;
+ dstConfig->role = audio_is_output_device(mDeviceType) ?
+ AUDIO_PORT_ROLE_SINK : AUDIO_PORT_ROLE_SOURCE;
+ dstConfig->type = AUDIO_PORT_TYPE_DEVICE;
+ dstConfig->ext.device.type = mDeviceType;
+
+ //TODO Understand why this test is necessary. i.e. why at boot time does it crash
+ // without the test?
+ // This has been demonstrated to NOT be true (at start up)
+ // ALOG_ASSERT(mModule != NULL);
+ dstConfig->ext.device.hw_module = mModule != NULL ? mModule->mHandle : NULL;
+ strncpy(dstConfig->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN);
+}
+
+void DeviceDescriptor::toAudioPort(struct audio_port *port) const
+{
+ ALOGV("DeviceDescriptor::toAudioPort() handle %d type %x", mId, mDeviceType);
+ AudioPort::toAudioPort(port);
+ port->id = mId;
+ toAudioPortConfig(&port->active_config);
+ port->ext.device.type = mDeviceType;
+ port->ext.device.hw_module = mModule->mHandle;
+ strncpy(port->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN);
+}
+
+status_t DeviceDescriptor::dump(int fd, int spaces, int index) const
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "%*sDevice %d:\n", spaces, "", index+1);
+ result.append(buffer);
+ if (mId != 0) {
+ snprintf(buffer, SIZE, "%*s- id: %2d\n", spaces, "", mId);
+ result.append(buffer);
+ }
+ snprintf(buffer, SIZE, "%*s- type: %-48s\n", spaces, "",
+ ConfigParsingUtils::enumToString(sDeviceNameToEnumTable,
+ ARRAY_SIZE(sDeviceNameToEnumTable),
+ mDeviceType));
+ result.append(buffer);
+ if (mAddress.size() != 0) {
+ snprintf(buffer, SIZE, "%*s- address: %-32s\n", spaces, "", mAddress.string());
+ result.append(buffer);
+ }
+ write(fd, result.string(), result.size());
+ AudioPort::dump(fd, spaces);
+
+ return NO_ERROR;
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/Devices.h b/services/audiopolicy/managerdefault/Devices.h
new file mode 100644
index 0000000..65e1416
--- /dev/null
+++ b/services/audiopolicy/managerdefault/Devices.h
@@ -0,0 +1,75 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+namespace android {
+
+class AudioPort;
+class AudioPortConfig;
+
+class DeviceDescriptor: public AudioPort, public AudioPortConfig
+{
+public:
+ DeviceDescriptor(const String8& name, audio_devices_t type);
+
+ virtual ~DeviceDescriptor() {}
+
+ bool equals(const sp<DeviceDescriptor>& other) const;
+
+ // AudioPortConfig
+ virtual sp<AudioPort> getAudioPort() const { return (AudioPort*) this; }
+ virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig = NULL) const;
+
+ // AudioPort
+ virtual void loadGains(cnode *root);
+ virtual void toAudioPort(struct audio_port *port) const;
+
+ status_t dump(int fd, int spaces, int index) const;
+
+ audio_devices_t mDeviceType;
+ String8 mAddress;
+ audio_port_handle_t mId;
+
+ static String8 emptyNameStr;
+};
+
+class DeviceVector : public SortedVector< sp<DeviceDescriptor> >
+{
+public:
+ DeviceVector() : SortedVector(), mDeviceTypes(AUDIO_DEVICE_NONE) {}
+
+ ssize_t add(const sp<DeviceDescriptor>& item);
+ ssize_t remove(const sp<DeviceDescriptor>& item);
+ ssize_t indexOf(const sp<DeviceDescriptor>& item) const;
+
+ audio_devices_t types() const { return mDeviceTypes; }
+
+ void loadDevicesFromType(audio_devices_t types);
+ void loadDevicesFromName(char *name, const DeviceVector& declaredDevices);
+
+ sp<DeviceDescriptor> getDevice(audio_devices_t type, String8 address) const;
+ DeviceVector getDevicesFromType(audio_devices_t types) const;
+ sp<DeviceDescriptor> getDeviceFromId(audio_port_handle_t id) const;
+ sp<DeviceDescriptor> getDeviceFromName(const String8& name) const;
+ DeviceVector getDevicesFromTypeAddr(audio_devices_t type, String8 address)
+ const;
+
+private:
+ void refreshTypes();
+ audio_devices_t mDeviceTypes;
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/Gains.cpp b/services/audiopolicy/managerdefault/Gains.cpp
new file mode 100644
index 0000000..4aca26d
--- /dev/null
+++ b/services/audiopolicy/managerdefault/Gains.cpp
@@ -0,0 +1,446 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM::Gains"
+//#define LOG_NDEBUG 0
+
+//#define VERY_VERBOSE_LOGGING
+#ifdef VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+#include "AudioPolicyManager.h"
+
+#include <math.h>
+
+namespace android {
+
+const VolumeCurvePoint
+ApmGains::sDefaultVolumeCurve[ApmGains::VOLCNT] = {
+ {1, -49.5f}, {33, -33.5f}, {66, -17.0f}, {100, 0.0f}
+};
+
+
+const VolumeCurvePoint
+ApmGains::sDefaultMediaVolumeCurve[ApmGains::VOLCNT] = {
+ {1, -58.0f}, {20, -40.0f}, {60, -17.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sExtMediaSystemVolumeCurve[ApmGains::VOLCNT] = {
+ {1, -58.0f}, {20, -40.0f}, {60, -21.0f}, {100, -10.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sSpeakerMediaVolumeCurve[ApmGains::VOLCNT] = {
+ {1, -56.0f}, {20, -34.0f}, {60, -11.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sSpeakerMediaVolumeCurveDrc[ApmGains::VOLCNT] = {
+ {1, -55.0f}, {20, -43.0f}, {86, -12.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sSpeakerSonificationVolumeCurve[ApmGains::VOLCNT] = {
+ {1, -29.7f}, {33, -20.1f}, {66, -10.2f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sSpeakerSonificationVolumeCurveDrc[ApmGains::VOLCNT] = {
+ {1, -35.7f}, {33, -26.1f}, {66, -13.2f}, {100, 0.0f}
+};
+
+// AUDIO_STREAM_SYSTEM, AUDIO_STREAM_ENFORCED_AUDIBLE and AUDIO_STREAM_DTMF volume tracks
+// AUDIO_STREAM_RING on phones and AUDIO_STREAM_MUSIC on tablets.
+// AUDIO_STREAM_DTMF tracks AUDIO_STREAM_VOICE_CALL while in call (See AudioService.java).
+// The range is constrained between -24dB and -6dB over speaker and -30dB and -18dB over headset.
+
+const VolumeCurvePoint
+ApmGains::sDefaultSystemVolumeCurve[ApmGains::VOLCNT] = {
+ {1, -24.0f}, {33, -18.0f}, {66, -12.0f}, {100, -6.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sDefaultSystemVolumeCurveDrc[ApmGains::VOLCNT] = {
+ {1, -34.0f}, {33, -24.0f}, {66, -15.0f}, {100, -6.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sHeadsetSystemVolumeCurve[ApmGains::VOLCNT] = {
+ {1, -30.0f}, {33, -26.0f}, {66, -22.0f}, {100, -18.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sDefaultVoiceVolumeCurve[ApmGains::VOLCNT] = {
+ {0, -42.0f}, {33, -28.0f}, {66, -14.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sSpeakerVoiceVolumeCurve[ApmGains::VOLCNT] = {
+ {0, -24.0f}, {33, -16.0f}, {66, -8.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sLinearVolumeCurve[ApmGains::VOLCNT] = {
+ {0, -96.0f}, {33, -68.0f}, {66, -34.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sSilentVolumeCurve[ApmGains::VOLCNT] = {
+ {0, -96.0f}, {1, -96.0f}, {2, -96.0f}, {100, -96.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sFullScaleVolumeCurve[ApmGains::VOLCNT] = {
+ {0, 0.0f}, {1, 0.0f}, {2, 0.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint *ApmGains::sVolumeProfiles[AUDIO_STREAM_CNT]
+ [ApmGains::DEVICE_CATEGORY_CNT] = {
+ { // AUDIO_STREAM_VOICE_CALL
+ ApmGains::sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_SYSTEM
+ ApmGains::sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_RING
+ ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_MUSIC
+ ApmGains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_ALARM
+ ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_NOTIFICATION
+ ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_BLUETOOTH_SCO
+ ApmGains::sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_ENFORCED_AUDIBLE
+ ApmGains::sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_DTMF
+ ApmGains::sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_TTS
+ // "Transmitted Through Speaker": always silent except on DEVICE_CATEGORY_SPEAKER
+ ApmGains::sSilentVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sLinearVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sSilentVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sSilentVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_ACCESSIBILITY
+ ApmGains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_REROUTING
+ ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sFullScaleVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_PATCH
+ ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sFullScaleVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+};
+
+//static
+audio_devices_t ApmGains::getDeviceForVolume(audio_devices_t device)
+{
+ if (device == AUDIO_DEVICE_NONE) {
+ // this happens when forcing a route update and no track is active on an output.
+ // In this case the returned category is not important.
+ device = AUDIO_DEVICE_OUT_SPEAKER;
+ } else if (popcount(device) > 1) {
+ // Multiple device selection is either:
+ // - speaker + one other device: give priority to speaker in this case.
+ // - one A2DP device + another device: happens with duplicated output. In this case
+ // retain the device on the A2DP output as the other must not correspond to an active
+ // selection if not the speaker.
+ // - HDMI-CEC system audio mode only output: give priority to available item in order.
+ if (device & AUDIO_DEVICE_OUT_SPEAKER) {
+ device = AUDIO_DEVICE_OUT_SPEAKER;
+ } else if (device & AUDIO_DEVICE_OUT_HDMI_ARC) {
+ device = AUDIO_DEVICE_OUT_HDMI_ARC;
+ } else if (device & AUDIO_DEVICE_OUT_AUX_LINE) {
+ device = AUDIO_DEVICE_OUT_AUX_LINE;
+ } else if (device & AUDIO_DEVICE_OUT_SPDIF) {
+ device = AUDIO_DEVICE_OUT_SPDIF;
+ } else {
+ device = (audio_devices_t)(device & AUDIO_DEVICE_OUT_ALL_A2DP);
+ }
+ }
+
+ /*SPEAKER_SAFE is an alias of SPEAKER for purposes of volume control*/
+ if (device == AUDIO_DEVICE_OUT_SPEAKER_SAFE)
+ device = AUDIO_DEVICE_OUT_SPEAKER;
+
+ ALOGW_IF(popcount(device) != 1,
+ "getDeviceForVolume() invalid device combination: %08x",
+ device);
+
+ return device;
+}
+
+//static
+ApmGains::device_category ApmGains::getDeviceCategory(audio_devices_t device)
+{
+ switch(getDeviceForVolume(device)) {
+ case AUDIO_DEVICE_OUT_EARPIECE:
+ return ApmGains::DEVICE_CATEGORY_EARPIECE;
+ case AUDIO_DEVICE_OUT_WIRED_HEADSET:
+ case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_SCO:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES:
+ return ApmGains::DEVICE_CATEGORY_HEADSET;
+ case AUDIO_DEVICE_OUT_LINE:
+ case AUDIO_DEVICE_OUT_AUX_DIGITAL:
+ /*USB? Remote submix?*/
+ return ApmGains::DEVICE_CATEGORY_EXT_MEDIA;
+ case AUDIO_DEVICE_OUT_SPEAKER:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER:
+ case AUDIO_DEVICE_OUT_USB_ACCESSORY:
+ case AUDIO_DEVICE_OUT_USB_DEVICE:
+ case AUDIO_DEVICE_OUT_REMOTE_SUBMIX:
+ default:
+ return ApmGains::DEVICE_CATEGORY_SPEAKER;
+ }
+}
+
+//static
+float ApmGains::volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
+ int indexInUi)
+{
+ ApmGains::device_category deviceCategory = ApmGains::getDeviceCategory(device);
+ const VolumeCurvePoint *curve = streamDesc.mVolumeCurve[deviceCategory];
+
+ // the volume index in the UI is relative to the min and max volume indices for this stream type
+ int nbSteps = 1 + curve[ApmGains::VOLMAX].mIndex -
+ curve[ApmGains::VOLMIN].mIndex;
+ int volIdx = (nbSteps * (indexInUi - streamDesc.mIndexMin)) /
+ (streamDesc.mIndexMax - streamDesc.mIndexMin);
+
+ // find what part of the curve this index volume belongs to, or if it's out of bounds
+ int segment = 0;
+ if (volIdx < curve[ApmGains::VOLMIN].mIndex) { // out of bounds
+ return 0.0f;
+ } else if (volIdx < curve[ApmGains::VOLKNEE1].mIndex) {
+ segment = 0;
+ } else if (volIdx < curve[ApmGains::VOLKNEE2].mIndex) {
+ segment = 1;
+ } else if (volIdx <= curve[ApmGains::VOLMAX].mIndex) {
+ segment = 2;
+ } else { // out of bounds
+ return 1.0f;
+ }
+
+ // linear interpolation in the attenuation table in dB
+ float decibels = curve[segment].mDBAttenuation +
+ ((float)(volIdx - curve[segment].mIndex)) *
+ ( (curve[segment+1].mDBAttenuation -
+ curve[segment].mDBAttenuation) /
+ ((float)(curve[segment+1].mIndex -
+ curve[segment].mIndex)) );
+
+ float amplification = exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 )
+
+ ALOGVV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f] ampl=%.5f",
+ curve[segment].mIndex, volIdx,
+ curve[segment+1].mIndex,
+ curve[segment].mDBAttenuation,
+ decibels,
+ curve[segment+1].mDBAttenuation,
+ amplification);
+
+ return amplification;
+}
+
+
+
+AudioGain::AudioGain(int index, bool useInChannelMask)
+{
+ mIndex = index;
+ mUseInChannelMask = useInChannelMask;
+ memset(&mGain, 0, sizeof(struct audio_gain));
+}
+
+void AudioGain::getDefaultConfig(struct audio_gain_config *config)
+{
+ config->index = mIndex;
+ config->mode = mGain.mode;
+ config->channel_mask = mGain.channel_mask;
+ if ((mGain.mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
+ config->values[0] = mGain.default_value;
+ } else {
+ uint32_t numValues;
+ if (mUseInChannelMask) {
+ numValues = audio_channel_count_from_in_mask(mGain.channel_mask);
+ } else {
+ numValues = audio_channel_count_from_out_mask(mGain.channel_mask);
+ }
+ for (size_t i = 0; i < numValues; i++) {
+ config->values[i] = mGain.default_value;
+ }
+ }
+ if ((mGain.mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
+ config->ramp_duration_ms = mGain.min_ramp_ms;
+ }
+}
+
+status_t AudioGain::checkConfig(const struct audio_gain_config *config)
+{
+ if ((config->mode & ~mGain.mode) != 0) {
+ return BAD_VALUE;
+ }
+ if ((config->mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
+ if ((config->values[0] < mGain.min_value) ||
+ (config->values[0] > mGain.max_value)) {
+ return BAD_VALUE;
+ }
+ } else {
+ if ((config->channel_mask & ~mGain.channel_mask) != 0) {
+ return BAD_VALUE;
+ }
+ uint32_t numValues;
+ if (mUseInChannelMask) {
+ numValues = audio_channel_count_from_in_mask(config->channel_mask);
+ } else {
+ numValues = audio_channel_count_from_out_mask(config->channel_mask);
+ }
+ for (size_t i = 0; i < numValues; i++) {
+ if ((config->values[i] < mGain.min_value) ||
+ (config->values[i] > mGain.max_value)) {
+ return BAD_VALUE;
+ }
+ }
+ }
+ if ((config->mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
+ if ((config->ramp_duration_ms < mGain.min_ramp_ms) ||
+ (config->ramp_duration_ms > mGain.max_ramp_ms)) {
+ return BAD_VALUE;
+ }
+ }
+ return NO_ERROR;
+}
+
+void AudioGain::dump(int fd, int spaces, int index) const
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "%*sGain %d:\n", spaces, "", index+1);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- mode: %08x\n", spaces, "", mGain.mode);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- channel_mask: %08x\n", spaces, "", mGain.channel_mask);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- min_value: %d mB\n", spaces, "", mGain.min_value);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- max_value: %d mB\n", spaces, "", mGain.max_value);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- default_value: %d mB\n", spaces, "", mGain.default_value);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- step_value: %d mB\n", spaces, "", mGain.step_value);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- min_ramp_ms: %d ms\n", spaces, "", mGain.min_ramp_ms);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- max_ramp_ms: %d ms\n", spaces, "", mGain.max_ramp_ms);
+ result.append(buffer);
+
+ write(fd, result.string(), result.size());
+}
+
+
+// --- StreamDescriptor class implementation
+
+StreamDescriptor::StreamDescriptor()
+ : mIndexMin(0), mIndexMax(1), mCanBeMuted(true)
+{
+ mIndexCur.add(AUDIO_DEVICE_OUT_DEFAULT, 0);
+}
+
+int StreamDescriptor::getVolumeIndex(audio_devices_t device)
+{
+ device = ApmGains::getDeviceForVolume(device);
+ // there is always a valid entry for AUDIO_DEVICE_OUT_DEFAULT
+ if (mIndexCur.indexOfKey(device) < 0) {
+ device = AUDIO_DEVICE_OUT_DEFAULT;
+ }
+ return mIndexCur.valueFor(device);
+}
+
+void StreamDescriptor::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "%s %02d %02d ",
+ mCanBeMuted ? "true " : "false", mIndexMin, mIndexMax);
+ result.append(buffer);
+ for (size_t i = 0; i < mIndexCur.size(); i++) {
+ snprintf(buffer, SIZE, "%04x : %02d, ",
+ mIndexCur.keyAt(i),
+ mIndexCur.valueAt(i));
+ result.append(buffer);
+ }
+ result.append("\n");
+
+ write(fd, result.string(), result.size());
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/Gains.h b/services/audiopolicy/managerdefault/Gains.h
new file mode 100644
index 0000000..b4ab129
--- /dev/null
+++ b/services/audiopolicy/managerdefault/Gains.h
@@ -0,0 +1,112 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+namespace android {
+
+class VolumeCurvePoint
+{
+public:
+ int mIndex;
+ float mDBAttenuation;
+};
+
+class StreamDescriptor;
+
+class ApmGains
+{
+public :
+ // 4 points to define the volume attenuation curve, each characterized by the volume
+ // index (from 0 to 100) at which they apply, and the attenuation in dB at that index.
+ // we use 100 steps to avoid rounding errors when computing the volume in volIndexToAmpl()
+ enum { VOLMIN = 0, VOLKNEE1 = 1, VOLKNEE2 = 2, VOLMAX = 3, VOLCNT = 4};
+
+ // device categories used for volume curve management.
+ enum device_category {
+ DEVICE_CATEGORY_HEADSET,
+ DEVICE_CATEGORY_SPEAKER,
+ DEVICE_CATEGORY_EARPIECE,
+ DEVICE_CATEGORY_EXT_MEDIA,
+ DEVICE_CATEGORY_CNT
+ };
+
+ // returns the category the device belongs to with regard to volume curve management
+ static ApmGains::device_category getDeviceCategory(audio_devices_t device);
+
+ // extract one device relevant for volume control from multiple device selection
+ static audio_devices_t getDeviceForVolume(audio_devices_t device);
+
+ static float volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
+ int indexInUi);
+
+ // default volume curve
+ static const VolumeCurvePoint sDefaultVolumeCurve[ApmGains::VOLCNT];
+ // default volume curve for media strategy
+ static const VolumeCurvePoint sDefaultMediaVolumeCurve[ApmGains::VOLCNT];
+ // volume curve for non-media audio on ext media outputs (HDMI, Line, etc)
+ static const VolumeCurvePoint sExtMediaSystemVolumeCurve[ApmGains::VOLCNT];
+ // volume curve for media strategy on speakers
+ static const VolumeCurvePoint sSpeakerMediaVolumeCurve[ApmGains::VOLCNT];
+ static const VolumeCurvePoint sSpeakerMediaVolumeCurveDrc[ApmGains::VOLCNT];
+ // volume curve for sonification strategy on speakers
+ static const VolumeCurvePoint sSpeakerSonificationVolumeCurve[ApmGains::VOLCNT];
+ static const VolumeCurvePoint sSpeakerSonificationVolumeCurveDrc[ApmGains::VOLCNT];
+ static const VolumeCurvePoint sDefaultSystemVolumeCurve[ApmGains::VOLCNT];
+ static const VolumeCurvePoint sDefaultSystemVolumeCurveDrc[ApmGains::VOLCNT];
+ static const VolumeCurvePoint sHeadsetSystemVolumeCurve[ApmGains::VOLCNT];
+ static const VolumeCurvePoint sDefaultVoiceVolumeCurve[ApmGains::VOLCNT];
+ static const VolumeCurvePoint sSpeakerVoiceVolumeCurve[ApmGains::VOLCNT];
+ static const VolumeCurvePoint sLinearVolumeCurve[ApmGains::VOLCNT];
+ static const VolumeCurvePoint sSilentVolumeCurve[ApmGains::VOLCNT];
+ static const VolumeCurvePoint sFullScaleVolumeCurve[ApmGains::VOLCNT];
+ // default volume curves per stream and device category. See initializeVolumeCurves()
+ static const VolumeCurvePoint *sVolumeProfiles[AUDIO_STREAM_CNT][ApmGains::DEVICE_CATEGORY_CNT];
+};
+
+
+class AudioGain: public RefBase
+{
+public:
+ AudioGain(int index, bool useInChannelMask);
+ virtual ~AudioGain() {}
+
+ void dump(int fd, int spaces, int index) const;
+
+ void getDefaultConfig(struct audio_gain_config *config);
+ status_t checkConfig(const struct audio_gain_config *config);
+ int mIndex;
+ struct audio_gain mGain;
+ bool mUseInChannelMask;
+};
+
+
+// stream descriptor used for volume control
+class StreamDescriptor
+{
+public:
+ StreamDescriptor();
+
+ int getVolumeIndex(audio_devices_t device);
+ void dump(int fd);
+
+ int mIndexMin; // min volume index
+ int mIndexMax; // max volume index
+ KeyedVector<audio_devices_t, int> mIndexCur; // current volume index per device
+ bool mCanBeMuted; // true is the stream can be muted
+
+ const VolumeCurvePoint *mVolumeCurve[ApmGains::DEVICE_CATEGORY_CNT];
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/HwModule.cpp b/services/audiopolicy/managerdefault/HwModule.cpp
new file mode 100644
index 0000000..a04bdc8
--- /dev/null
+++ b/services/audiopolicy/managerdefault/HwModule.cpp
@@ -0,0 +1,279 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM::HwModule"
+//#define LOG_NDEBUG 0
+
+#include "AudioPolicyManager.h"
+#include "audio_policy_conf.h"
+#include <hardware/audio.h>
+
+namespace android {
+
+HwModule::HwModule(const char *name)
+ : mName(strndup(name, AUDIO_HARDWARE_MODULE_ID_MAX_LEN)),
+ mHalVersion(AUDIO_DEVICE_API_VERSION_MIN), mHandle(0)
+{
+}
+
+HwModule::~HwModule()
+{
+ for (size_t i = 0; i < mOutputProfiles.size(); i++) {
+ mOutputProfiles[i]->mSupportedDevices.clear();
+ }
+ for (size_t i = 0; i < mInputProfiles.size(); i++) {
+ mInputProfiles[i]->mSupportedDevices.clear();
+ }
+ free((void *)mName);
+}
+
+status_t HwModule::loadInput(cnode *root)
+{
+ cnode *node = root->first_child;
+
+ sp<IOProfile> profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SINK, this);
+
+ while (node) {
+ if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
+ profile->loadSamplingRates((char *)node->value);
+ } else if (strcmp(node->name, FORMATS_TAG) == 0) {
+ profile->loadFormats((char *)node->value);
+ } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
+ profile->loadInChannels((char *)node->value);
+ } else if (strcmp(node->name, DEVICES_TAG) == 0) {
+ profile->mSupportedDevices.loadDevicesFromName((char *)node->value,
+ mDeclaredDevices);
+ } else if (strcmp(node->name, FLAGS_TAG) == 0) {
+ profile->mFlags = ConfigParsingUtils::parseInputFlagNames((char *)node->value);
+ } else if (strcmp(node->name, GAINS_TAG) == 0) {
+ profile->loadGains(node);
+ }
+ node = node->next;
+ }
+ ALOGW_IF(profile->mSupportedDevices.isEmpty(),
+ "loadInput() invalid supported devices");
+ ALOGW_IF(profile->mChannelMasks.size() == 0,
+ "loadInput() invalid supported channel masks");
+ ALOGW_IF(profile->mSamplingRates.size() == 0,
+ "loadInput() invalid supported sampling rates");
+ ALOGW_IF(profile->mFormats.size() == 0,
+ "loadInput() invalid supported formats");
+ if (!profile->mSupportedDevices.isEmpty() &&
+ (profile->mChannelMasks.size() != 0) &&
+ (profile->mSamplingRates.size() != 0) &&
+ (profile->mFormats.size() != 0)) {
+
+ ALOGV("loadInput() adding input Supported Devices %04x",
+ profile->mSupportedDevices.types());
+
+ mInputProfiles.add(profile);
+ return NO_ERROR;
+ } else {
+ return BAD_VALUE;
+ }
+}
+
+status_t HwModule::loadOutput(cnode *root)
+{
+ cnode *node = root->first_child;
+
+ sp<IOProfile> profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SOURCE, this);
+
+ while (node) {
+ if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
+ profile->loadSamplingRates((char *)node->value);
+ } else if (strcmp(node->name, FORMATS_TAG) == 0) {
+ profile->loadFormats((char *)node->value);
+ } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
+ profile->loadOutChannels((char *)node->value);
+ } else if (strcmp(node->name, DEVICES_TAG) == 0) {
+ profile->mSupportedDevices.loadDevicesFromName((char *)node->value,
+ mDeclaredDevices);
+ } else if (strcmp(node->name, FLAGS_TAG) == 0) {
+ profile->mFlags = ConfigParsingUtils::parseOutputFlagNames((char *)node->value);
+ } else if (strcmp(node->name, GAINS_TAG) == 0) {
+ profile->loadGains(node);
+ }
+ node = node->next;
+ }
+ ALOGW_IF(profile->mSupportedDevices.isEmpty(),
+ "loadOutput() invalid supported devices");
+ ALOGW_IF(profile->mChannelMasks.size() == 0,
+ "loadOutput() invalid supported channel masks");
+ ALOGW_IF(profile->mSamplingRates.size() == 0,
+ "loadOutput() invalid supported sampling rates");
+ ALOGW_IF(profile->mFormats.size() == 0,
+ "loadOutput() invalid supported formats");
+ if (!profile->mSupportedDevices.isEmpty() &&
+ (profile->mChannelMasks.size() != 0) &&
+ (profile->mSamplingRates.size() != 0) &&
+ (profile->mFormats.size() != 0)) {
+
+ ALOGV("loadOutput() adding output Supported Devices %04x, mFlags %04x",
+ profile->mSupportedDevices.types(), profile->mFlags);
+
+ mOutputProfiles.add(profile);
+ return NO_ERROR;
+ } else {
+ return BAD_VALUE;
+ }
+}
+
+status_t HwModule::loadDevice(cnode *root)
+{
+ cnode *node = root->first_child;
+
+ audio_devices_t type = AUDIO_DEVICE_NONE;
+ while (node) {
+ if (strcmp(node->name, DEVICE_TYPE) == 0) {
+ type = ConfigParsingUtils::parseDeviceNames((char *)node->value);
+ break;
+ }
+ node = node->next;
+ }
+ if (type == AUDIO_DEVICE_NONE ||
+ (!audio_is_input_device(type) && !audio_is_output_device(type))) {
+ ALOGW("loadDevice() bad type %08x", type);
+ return BAD_VALUE;
+ }
+ sp<DeviceDescriptor> deviceDesc = new DeviceDescriptor(String8(root->name), type);
+ deviceDesc->mModule = this;
+
+ node = root->first_child;
+ while (node) {
+ if (strcmp(node->name, DEVICE_ADDRESS) == 0) {
+ deviceDesc->mAddress = String8((char *)node->value);
+ } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
+ if (audio_is_input_device(type)) {
+ deviceDesc->loadInChannels((char *)node->value);
+ } else {
+ deviceDesc->loadOutChannels((char *)node->value);
+ }
+ } else if (strcmp(node->name, GAINS_TAG) == 0) {
+ deviceDesc->loadGains(node);
+ }
+ node = node->next;
+ }
+
+ ALOGV("loadDevice() adding device name %s type %08x address %s",
+ deviceDesc->mName.string(), type, deviceDesc->mAddress.string());
+
+ mDeclaredDevices.add(deviceDesc);
+
+ return NO_ERROR;
+}
+
+status_t HwModule::addOutputProfile(String8 name, const audio_config_t *config,
+ audio_devices_t device, String8 address)
+{
+ sp<IOProfile> profile = new IOProfile(name, AUDIO_PORT_ROLE_SOURCE, this);
+
+ profile->mSamplingRates.add(config->sample_rate);
+ profile->mChannelMasks.add(config->channel_mask);
+ profile->mFormats.add(config->format);
+
+ sp<DeviceDescriptor> devDesc = new DeviceDescriptor(name, device);
+ devDesc->mAddress = address;
+ profile->mSupportedDevices.add(devDesc);
+
+ mOutputProfiles.add(profile);
+
+ return NO_ERROR;
+}
+
+status_t HwModule::removeOutputProfile(String8 name)
+{
+ for (size_t i = 0; i < mOutputProfiles.size(); i++) {
+ if (mOutputProfiles[i]->mName == name) {
+ mOutputProfiles.removeAt(i);
+ break;
+ }
+ }
+
+ return NO_ERROR;
+}
+
+status_t HwModule::addInputProfile(String8 name, const audio_config_t *config,
+ audio_devices_t device, String8 address)
+{
+ sp<IOProfile> profile = new IOProfile(name, AUDIO_PORT_ROLE_SINK, this);
+
+ profile->mSamplingRates.add(config->sample_rate);
+ profile->mChannelMasks.add(config->channel_mask);
+ profile->mFormats.add(config->format);
+
+ sp<DeviceDescriptor> devDesc = new DeviceDescriptor(name, device);
+ devDesc->mAddress = address;
+ profile->mSupportedDevices.add(devDesc);
+
+ ALOGV("addInputProfile() name %s rate %d mask 0x08", name.string(), config->sample_rate, config->channel_mask);
+
+ mInputProfiles.add(profile);
+
+ return NO_ERROR;
+}
+
+status_t HwModule::removeInputProfile(String8 name)
+{
+ for (size_t i = 0; i < mInputProfiles.size(); i++) {
+ if (mInputProfiles[i]->mName == name) {
+ mInputProfiles.removeAt(i);
+ break;
+ }
+ }
+
+ return NO_ERROR;
+}
+
+
+void HwModule::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, " - name: %s\n", mName);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " - handle: %d\n", mHandle);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " - version: %u.%u\n", mHalVersion >> 8, mHalVersion & 0xFF);
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+ if (mOutputProfiles.size()) {
+ write(fd, " - outputs:\n", strlen(" - outputs:\n"));
+ for (size_t i = 0; i < mOutputProfiles.size(); i++) {
+ snprintf(buffer, SIZE, " output %zu:\n", i);
+ write(fd, buffer, strlen(buffer));
+ mOutputProfiles[i]->dump(fd);
+ }
+ }
+ if (mInputProfiles.size()) {
+ write(fd, " - inputs:\n", strlen(" - inputs:\n"));
+ for (size_t i = 0; i < mInputProfiles.size(); i++) {
+ snprintf(buffer, SIZE, " input %zu:\n", i);
+ write(fd, buffer, strlen(buffer));
+ mInputProfiles[i]->dump(fd);
+ }
+ }
+ if (mDeclaredDevices.size()) {
+ write(fd, " - devices:\n", strlen(" - devices:\n"));
+ for (size_t i = 0; i < mDeclaredDevices.size(); i++) {
+ mDeclaredDevices[i]->dump(fd, 4, i);
+ }
+ }
+}
+
+} //namespace android
diff --git a/services/audiopolicy/managerdefault/HwModule.h b/services/audiopolicy/managerdefault/HwModule.h
new file mode 100644
index 0000000..f814dd9
--- /dev/null
+++ b/services/audiopolicy/managerdefault/HwModule.h
@@ -0,0 +1,46 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+namespace android {
+
+class HwModule : public RefBase
+{
+public:
+ HwModule(const char *name);
+ ~HwModule();
+
+ status_t loadOutput(cnode *root);
+ status_t loadInput(cnode *root);
+ status_t loadDevice(cnode *root);
+
+ status_t addOutputProfile(String8 name, const audio_config_t *config,
+ audio_devices_t device, String8 address);
+ status_t removeOutputProfile(String8 name);
+ status_t addInputProfile(String8 name, const audio_config_t *config,
+ audio_devices_t device, String8 address);
+ status_t removeInputProfile(String8 name);
+
+ void dump(int fd);
+
+ const char *const mName; // base name of the audio HW module (primary, a2dp ...)
+ uint32_t mHalVersion; // audio HAL API version
+ audio_module_handle_t mHandle;
+ Vector < sp<IOProfile> > mOutputProfiles; // output profiles exposed by this module
+ Vector < sp<IOProfile> > mInputProfiles; // input profiles exposed by this module
+ DeviceVector mDeclaredDevices; // devices declared in audio_policy.conf
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/IOProfile.cpp b/services/audiopolicy/managerdefault/IOProfile.cpp
new file mode 100644
index 0000000..538ac1a
--- /dev/null
+++ b/services/audiopolicy/managerdefault/IOProfile.cpp
@@ -0,0 +1,139 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM::IOProfile"
+//#define LOG_NDEBUG 0
+
+#include "AudioPolicyManager.h"
+
+namespace android {
+
+IOProfile::IOProfile(const String8& name, audio_port_role_t role,
+ const sp<HwModule>& module)
+ : AudioPort(name, AUDIO_PORT_TYPE_MIX, role, module)
+{
+}
+
+IOProfile::~IOProfile()
+{
+}
+
+// checks if the IO profile is compatible with specified parameters.
+// Sampling rate, format and channel mask must be specified in order to
+// get a valid a match
+bool IOProfile::isCompatibleProfile(audio_devices_t device,
+ String8 address,
+ uint32_t samplingRate,
+ uint32_t *updatedSamplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ uint32_t flags) const
+{
+ const bool isPlaybackThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SOURCE;
+ const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK;
+ ALOG_ASSERT(isPlaybackThread != isRecordThread);
+
+ if (device != AUDIO_DEVICE_NONE && mSupportedDevices.getDevice(device, address) == 0) {
+ return false;
+ }
+
+ if (samplingRate == 0) {
+ return false;
+ }
+ uint32_t myUpdatedSamplingRate = samplingRate;
+ if (isPlaybackThread && checkExactSamplingRate(samplingRate) != NO_ERROR) {
+ return false;
+ }
+ if (isRecordThread && checkCompatibleSamplingRate(samplingRate, &myUpdatedSamplingRate) !=
+ NO_ERROR) {
+ return false;
+ }
+
+ if (!audio_is_valid_format(format) || checkFormat(format) != NO_ERROR) {
+ return false;
+ }
+
+ if (isPlaybackThread && (!audio_is_output_channel(channelMask) ||
+ checkExactChannelMask(channelMask) != NO_ERROR)) {
+ return false;
+ }
+ if (isRecordThread && (!audio_is_input_channel(channelMask) ||
+ checkCompatibleChannelMask(channelMask) != NO_ERROR)) {
+ return false;
+ }
+
+ if (isPlaybackThread && (mFlags & flags) != flags) {
+ return false;
+ }
+ // The only input flag that is allowed to be different is the fast flag.
+ // An existing fast stream is compatible with a normal track request.
+ // An existing normal stream is compatible with a fast track request,
+ // but the fast request will be denied by AudioFlinger and converted to normal track.
+ if (isRecordThread && ((mFlags ^ flags) &
+ ~AUDIO_INPUT_FLAG_FAST)) {
+ return false;
+ }
+
+ if (updatedSamplingRate != NULL) {
+ *updatedSamplingRate = myUpdatedSamplingRate;
+ }
+ return true;
+}
+
+void IOProfile::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ AudioPort::dump(fd, 4);
+
+ snprintf(buffer, SIZE, " - flags: 0x%04x\n", mFlags);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " - devices:\n");
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+ for (size_t i = 0; i < mSupportedDevices.size(); i++) {
+ mSupportedDevices[i]->dump(fd, 6, i);
+ }
+}
+
+void IOProfile::log()
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ ALOGV(" - sampling rates: ");
+ for (size_t i = 0; i < mSamplingRates.size(); i++) {
+ ALOGV(" %d", mSamplingRates[i]);
+ }
+
+ ALOGV(" - channel masks: ");
+ for (size_t i = 0; i < mChannelMasks.size(); i++) {
+ ALOGV(" 0x%04x", mChannelMasks[i]);
+ }
+
+ ALOGV(" - formats: ");
+ for (size_t i = 0; i < mFormats.size(); i++) {
+ ALOGV(" 0x%08x", mFormats[i]);
+ }
+
+ ALOGV(" - devices: 0x%04x\n", mSupportedDevices.types());
+ ALOGV(" - flags: 0x%04x\n", mFlags);
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/IOProfile.h b/services/audiopolicy/managerdefault/IOProfile.h
new file mode 100644
index 0000000..3317969
--- /dev/null
+++ b/services/audiopolicy/managerdefault/IOProfile.h
@@ -0,0 +1,51 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+namespace android {
+
+class HwModule;
+
+// the IOProfile class describes the capabilities of an output or input stream.
+// It is currently assumed that all combination of listed parameters are supported.
+// It is used by the policy manager to determine if an output or input is suitable for
+// a given use case, open/close it accordingly and connect/disconnect audio tracks
+// to/from it.
+class IOProfile : public AudioPort
+{
+public:
+ IOProfile(const String8& name, audio_port_role_t role, const sp<HwModule>& module);
+ virtual ~IOProfile();
+
+ // This method is used for both output and input.
+ // If parameter updatedSamplingRate is non-NULL, it is assigned the actual sample rate.
+ // For input, flags is interpreted as audio_input_flags_t.
+ // TODO: merge audio_output_flags_t and audio_input_flags_t.
+ bool isCompatibleProfile(audio_devices_t device,
+ String8 address,
+ uint32_t samplingRate,
+ uint32_t *updatedSamplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ uint32_t flags) const;
+
+ void dump(int fd);
+ void log();
+
+ DeviceVector mSupportedDevices; // supported devices
+ // (devices this output can be routed to)
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/Ports.cpp b/services/audiopolicy/managerdefault/Ports.cpp
new file mode 100644
index 0000000..3e55cee
--- /dev/null
+++ b/services/audiopolicy/managerdefault/Ports.cpp
@@ -0,0 +1,844 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM::Ports"
+//#define LOG_NDEBUG 0
+
+#include "AudioPolicyManager.h"
+
+#include "audio_policy_conf.h"
+
+namespace android {
+
+// --- AudioPort class implementation
+
+AudioPort::AudioPort(const String8& name, audio_port_type_t type,
+ audio_port_role_t role, const sp<HwModule>& module) :
+ mName(name), mType(type), mRole(role), mModule(module), mFlags(0), mId(0)
+{
+ mUseInChannelMask = ((type == AUDIO_PORT_TYPE_DEVICE) && (role == AUDIO_PORT_ROLE_SOURCE)) ||
+ ((type == AUDIO_PORT_TYPE_MIX) && (role == AUDIO_PORT_ROLE_SINK));
+}
+
+void AudioPort::attach(const sp<HwModule>& module) {
+ mId = AudioPolicyManager::nextUniqueId();
+ mModule = module;
+}
+
+void AudioPort::toAudioPort(struct audio_port *port) const
+{
+ port->role = mRole;
+ port->type = mType;
+ strlcpy(port->name, mName, AUDIO_PORT_MAX_NAME_LEN);
+ unsigned int i;
+ for (i = 0; i < mSamplingRates.size() && i < AUDIO_PORT_MAX_SAMPLING_RATES; i++) {
+ if (mSamplingRates[i] != 0) {
+ port->sample_rates[i] = mSamplingRates[i];
+ }
+ }
+ port->num_sample_rates = i;
+ for (i = 0; i < mChannelMasks.size() && i < AUDIO_PORT_MAX_CHANNEL_MASKS; i++) {
+ if (mChannelMasks[i] != 0) {
+ port->channel_masks[i] = mChannelMasks[i];
+ }
+ }
+ port->num_channel_masks = i;
+ for (i = 0; i < mFormats.size() && i < AUDIO_PORT_MAX_FORMATS; i++) {
+ if (mFormats[i] != 0) {
+ port->formats[i] = mFormats[i];
+ }
+ }
+ port->num_formats = i;
+
+ ALOGV("AudioPort::toAudioPort() num gains %zu", mGains.size());
+
+ for (i = 0; i < mGains.size() && i < AUDIO_PORT_MAX_GAINS; i++) {
+ port->gains[i] = mGains[i]->mGain;
+ }
+ port->num_gains = i;
+}
+
+void AudioPort::importAudioPort(const sp<AudioPort> port) {
+ for (size_t k = 0 ; k < port->mSamplingRates.size() ; k++) {
+ const uint32_t rate = port->mSamplingRates.itemAt(k);
+ if (rate != 0) { // skip "dynamic" rates
+ bool hasRate = false;
+ for (size_t l = 0 ; l < mSamplingRates.size() ; l++) {
+ if (rate == mSamplingRates.itemAt(l)) {
+ hasRate = true;
+ break;
+ }
+ }
+ if (!hasRate) { // never import a sampling rate twice
+ mSamplingRates.add(rate);
+ }
+ }
+ }
+ for (size_t k = 0 ; k < port->mChannelMasks.size() ; k++) {
+ const audio_channel_mask_t mask = port->mChannelMasks.itemAt(k);
+ if (mask != 0) { // skip "dynamic" masks
+ bool hasMask = false;
+ for (size_t l = 0 ; l < mChannelMasks.size() ; l++) {
+ if (mask == mChannelMasks.itemAt(l)) {
+ hasMask = true;
+ break;
+ }
+ }
+ if (!hasMask) { // never import a channel mask twice
+ mChannelMasks.add(mask);
+ }
+ }
+ }
+ for (size_t k = 0 ; k < port->mFormats.size() ; k++) {
+ const audio_format_t format = port->mFormats.itemAt(k);
+ if (format != 0) { // skip "dynamic" formats
+ bool hasFormat = false;
+ for (size_t l = 0 ; l < mFormats.size() ; l++) {
+ if (format == mFormats.itemAt(l)) {
+ hasFormat = true;
+ break;
+ }
+ }
+ if (!hasFormat) { // never import a channel mask twice
+ mFormats.add(format);
+ }
+ }
+ }
+ for (size_t k = 0 ; k < port->mGains.size() ; k++) {
+ sp<AudioGain> gain = port->mGains.itemAt(k);
+ if (gain != 0) {
+ bool hasGain = false;
+ for (size_t l = 0 ; l < mGains.size() ; l++) {
+ if (gain == mGains.itemAt(l)) {
+ hasGain = true;
+ break;
+ }
+ }
+ if (!hasGain) { // never import a gain twice
+ mGains.add(gain);
+ }
+ }
+ }
+}
+
+void AudioPort::clearCapabilities() {
+ mChannelMasks.clear();
+ mFormats.clear();
+ mSamplingRates.clear();
+ mGains.clear();
+}
+
+void AudioPort::loadSamplingRates(char *name)
+{
+ char *str = strtok(name, "|");
+
+ // by convention, "0' in the first entry in mSamplingRates indicates the supported sampling
+ // rates should be read from the output stream after it is opened for the first time
+ if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+ mSamplingRates.add(0);
+ return;
+ }
+
+ while (str != NULL) {
+ uint32_t rate = atoi(str);
+ if (rate != 0) {
+ ALOGV("loadSamplingRates() adding rate %d", rate);
+ mSamplingRates.add(rate);
+ }
+ str = strtok(NULL, "|");
+ }
+}
+
+void AudioPort::loadFormats(char *name)
+{
+ char *str = strtok(name, "|");
+
+ // by convention, "0' in the first entry in mFormats indicates the supported formats
+ // should be read from the output stream after it is opened for the first time
+ if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+ mFormats.add(AUDIO_FORMAT_DEFAULT);
+ return;
+ }
+
+ while (str != NULL) {
+ audio_format_t format = (audio_format_t)ConfigParsingUtils::stringToEnum(sFormatNameToEnumTable,
+ ARRAY_SIZE(sFormatNameToEnumTable),
+ str);
+ if (format != AUDIO_FORMAT_DEFAULT) {
+ mFormats.add(format);
+ }
+ str = strtok(NULL, "|");
+ }
+}
+
+void AudioPort::loadInChannels(char *name)
+{
+ const char *str = strtok(name, "|");
+
+ ALOGV("loadInChannels() %s", name);
+
+ if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+ mChannelMasks.add(0);
+ return;
+ }
+
+ while (str != NULL) {
+ audio_channel_mask_t channelMask =
+ (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sInChannelsNameToEnumTable,
+ ARRAY_SIZE(sInChannelsNameToEnumTable),
+ str);
+ if (channelMask != 0) {
+ ALOGV("loadInChannels() adding channelMask %04x", channelMask);
+ mChannelMasks.add(channelMask);
+ }
+ str = strtok(NULL, "|");
+ }
+}
+
+void AudioPort::loadOutChannels(char *name)
+{
+ const char *str = strtok(name, "|");
+
+ ALOGV("loadOutChannels() %s", name);
+
+ // by convention, "0' in the first entry in mChannelMasks indicates the supported channel
+ // masks should be read from the output stream after it is opened for the first time
+ if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+ mChannelMasks.add(0);
+ return;
+ }
+
+ while (str != NULL) {
+ audio_channel_mask_t channelMask =
+ (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sOutChannelsNameToEnumTable,
+ ARRAY_SIZE(sOutChannelsNameToEnumTable),
+ str);
+ if (channelMask != 0) {
+ mChannelMasks.add(channelMask);
+ }
+ str = strtok(NULL, "|");
+ }
+ return;
+}
+
+audio_gain_mode_t AudioPort::loadGainMode(char *name)
+{
+ const char *str = strtok(name, "|");
+
+ ALOGV("loadGainMode() %s", name);
+ audio_gain_mode_t mode = 0;
+ while (str != NULL) {
+ mode |= (audio_gain_mode_t)ConfigParsingUtils::stringToEnum(sGainModeNameToEnumTable,
+ ARRAY_SIZE(sGainModeNameToEnumTable),
+ str);
+ str = strtok(NULL, "|");
+ }
+ return mode;
+}
+
+void AudioPort::loadGain(cnode *root, int index)
+{
+ cnode *node = root->first_child;
+
+ sp<AudioGain> gain = new AudioGain(index, mUseInChannelMask);
+
+ while (node) {
+ if (strcmp(node->name, GAIN_MODE) == 0) {
+ gain->mGain.mode = loadGainMode((char *)node->value);
+ } else if (strcmp(node->name, GAIN_CHANNELS) == 0) {
+ if (mUseInChannelMask) {
+ gain->mGain.channel_mask =
+ (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sInChannelsNameToEnumTable,
+ ARRAY_SIZE(sInChannelsNameToEnumTable),
+ (char *)node->value);
+ } else {
+ gain->mGain.channel_mask =
+ (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sOutChannelsNameToEnumTable,
+ ARRAY_SIZE(sOutChannelsNameToEnumTable),
+ (char *)node->value);
+ }
+ } else if (strcmp(node->name, GAIN_MIN_VALUE) == 0) {
+ gain->mGain.min_value = atoi((char *)node->value);
+ } else if (strcmp(node->name, GAIN_MAX_VALUE) == 0) {
+ gain->mGain.max_value = atoi((char *)node->value);
+ } else if (strcmp(node->name, GAIN_DEFAULT_VALUE) == 0) {
+ gain->mGain.default_value = atoi((char *)node->value);
+ } else if (strcmp(node->name, GAIN_STEP_VALUE) == 0) {
+ gain->mGain.step_value = atoi((char *)node->value);
+ } else if (strcmp(node->name, GAIN_MIN_RAMP_MS) == 0) {
+ gain->mGain.min_ramp_ms = atoi((char *)node->value);
+ } else if (strcmp(node->name, GAIN_MAX_RAMP_MS) == 0) {
+ gain->mGain.max_ramp_ms = atoi((char *)node->value);
+ }
+ node = node->next;
+ }
+
+ ALOGV("loadGain() adding new gain mode %08x channel mask %08x min mB %d max mB %d",
+ gain->mGain.mode, gain->mGain.channel_mask, gain->mGain.min_value, gain->mGain.max_value);
+
+ if (gain->mGain.mode == 0) {
+ return;
+ }
+ mGains.add(gain);
+}
+
+void AudioPort::loadGains(cnode *root)
+{
+ cnode *node = root->first_child;
+ int index = 0;
+ while (node) {
+ ALOGV("loadGains() loading gain %s", node->name);
+ loadGain(node, index++);
+ node = node->next;
+ }
+}
+
+status_t AudioPort::checkExactSamplingRate(uint32_t samplingRate) const
+{
+ if (mSamplingRates.isEmpty()) {
+ return NO_ERROR;
+ }
+
+ for (size_t i = 0; i < mSamplingRates.size(); i ++) {
+ if (mSamplingRates[i] == samplingRate) {
+ return NO_ERROR;
+ }
+ }
+ return BAD_VALUE;
+}
+
+status_t AudioPort::checkCompatibleSamplingRate(uint32_t samplingRate,
+ uint32_t *updatedSamplingRate) const
+{
+ if (mSamplingRates.isEmpty()) {
+ return NO_ERROR;
+ }
+
+ // Search for the closest supported sampling rate that is above (preferred)
+ // or below (acceptable) the desired sampling rate, within a permitted ratio.
+ // The sampling rates do not need to be sorted in ascending order.
+ ssize_t maxBelow = -1;
+ ssize_t minAbove = -1;
+ uint32_t candidate;
+ for (size_t i = 0; i < mSamplingRates.size(); i++) {
+ candidate = mSamplingRates[i];
+ if (candidate == samplingRate) {
+ if (updatedSamplingRate != NULL) {
+ *updatedSamplingRate = candidate;
+ }
+ return NO_ERROR;
+ }
+ // candidate < desired
+ if (candidate < samplingRate) {
+ if (maxBelow < 0 || candidate > mSamplingRates[maxBelow]) {
+ maxBelow = i;
+ }
+ // candidate > desired
+ } else {
+ if (minAbove < 0 || candidate < mSamplingRates[minAbove]) {
+ minAbove = i;
+ }
+ }
+ }
+ // This uses hard-coded knowledge about AudioFlinger resampling ratios.
+ // TODO Move these assumptions out.
+ static const uint32_t kMaxDownSampleRatio = 6; // beyond this aliasing occurs
+ static const uint32_t kMaxUpSampleRatio = 256; // beyond this sample rate inaccuracies occur
+ // due to approximation by an int32_t of the
+ // phase increments
+ // Prefer to down-sample from a higher sampling rate, as we get the desired frequency spectrum.
+ if (minAbove >= 0) {
+ candidate = mSamplingRates[minAbove];
+ if (candidate / kMaxDownSampleRatio <= samplingRate) {
+ if (updatedSamplingRate != NULL) {
+ *updatedSamplingRate = candidate;
+ }
+ return NO_ERROR;
+ }
+ }
+ // But if we have to up-sample from a lower sampling rate, that's OK.
+ if (maxBelow >= 0) {
+ candidate = mSamplingRates[maxBelow];
+ if (candidate * kMaxUpSampleRatio >= samplingRate) {
+ if (updatedSamplingRate != NULL) {
+ *updatedSamplingRate = candidate;
+ }
+ return NO_ERROR;
+ }
+ }
+ // leave updatedSamplingRate unmodified
+ return BAD_VALUE;
+}
+
+status_t AudioPort::checkExactChannelMask(audio_channel_mask_t channelMask) const
+{
+ if (mChannelMasks.isEmpty()) {
+ return NO_ERROR;
+ }
+
+ for (size_t i = 0; i < mChannelMasks.size(); i++) {
+ if (mChannelMasks[i] == channelMask) {
+ return NO_ERROR;
+ }
+ }
+ return BAD_VALUE;
+}
+
+status_t AudioPort::checkCompatibleChannelMask(audio_channel_mask_t channelMask)
+ const
+{
+ if (mChannelMasks.isEmpty()) {
+ return NO_ERROR;
+ }
+
+ const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK;
+ for (size_t i = 0; i < mChannelMasks.size(); i ++) {
+ // FIXME Does not handle multi-channel automatic conversions yet
+ audio_channel_mask_t supported = mChannelMasks[i];
+ if (supported == channelMask) {
+ return NO_ERROR;
+ }
+ if (isRecordThread) {
+ // This uses hard-coded knowledge that AudioFlinger can silently down-mix and up-mix.
+ // FIXME Abstract this out to a table.
+ if (((supported == AUDIO_CHANNEL_IN_FRONT_BACK || supported == AUDIO_CHANNEL_IN_STEREO)
+ && channelMask == AUDIO_CHANNEL_IN_MONO) ||
+ (supported == AUDIO_CHANNEL_IN_MONO && (channelMask == AUDIO_CHANNEL_IN_FRONT_BACK
+ || channelMask == AUDIO_CHANNEL_IN_STEREO))) {
+ return NO_ERROR;
+ }
+ }
+ }
+ return BAD_VALUE;
+}
+
+status_t AudioPort::checkFormat(audio_format_t format) const
+{
+ if (mFormats.isEmpty()) {
+ return NO_ERROR;
+ }
+
+ for (size_t i = 0; i < mFormats.size(); i ++) {
+ if (mFormats[i] == format) {
+ return NO_ERROR;
+ }
+ }
+ return BAD_VALUE;
+}
+
+
+uint32_t AudioPort::pickSamplingRate() const
+{
+ // special case for uninitialized dynamic profile
+ if (mSamplingRates.size() == 1 && mSamplingRates[0] == 0) {
+ return 0;
+ }
+
+ // For direct outputs, pick minimum sampling rate: this helps ensuring that the
+ // channel count / sampling rate combination chosen will be supported by the connected
+ // sink
+ if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
+ (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) {
+ uint32_t samplingRate = UINT_MAX;
+ for (size_t i = 0; i < mSamplingRates.size(); i ++) {
+ if ((mSamplingRates[i] < samplingRate) && (mSamplingRates[i] > 0)) {
+ samplingRate = mSamplingRates[i];
+ }
+ }
+ return (samplingRate == UINT_MAX) ? 0 : samplingRate;
+ }
+
+ uint32_t samplingRate = 0;
+ uint32_t maxRate = MAX_MIXER_SAMPLING_RATE;
+
+ // For mixed output and inputs, use max mixer sampling rates. Do not
+ // limit sampling rate otherwise
+ if (mType != AUDIO_PORT_TYPE_MIX) {
+ maxRate = UINT_MAX;
+ }
+ for (size_t i = 0; i < mSamplingRates.size(); i ++) {
+ if ((mSamplingRates[i] > samplingRate) && (mSamplingRates[i] <= maxRate)) {
+ samplingRate = mSamplingRates[i];
+ }
+ }
+ return samplingRate;
+}
+
+audio_channel_mask_t AudioPort::pickChannelMask() const
+{
+ // special case for uninitialized dynamic profile
+ if (mChannelMasks.size() == 1 && mChannelMasks[0] == 0) {
+ return AUDIO_CHANNEL_NONE;
+ }
+ audio_channel_mask_t channelMask = AUDIO_CHANNEL_NONE;
+
+ // For direct outputs, pick minimum channel count: this helps ensuring that the
+ // channel count / sampling rate combination chosen will be supported by the connected
+ // sink
+ if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
+ (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) {
+ uint32_t channelCount = UINT_MAX;
+ for (size_t i = 0; i < mChannelMasks.size(); i ++) {
+ uint32_t cnlCount;
+ if (mUseInChannelMask) {
+ cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]);
+ } else {
+ cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]);
+ }
+ if ((cnlCount < channelCount) && (cnlCount > 0)) {
+ channelMask = mChannelMasks[i];
+ channelCount = cnlCount;
+ }
+ }
+ return channelMask;
+ }
+
+ uint32_t channelCount = 0;
+ uint32_t maxCount = MAX_MIXER_CHANNEL_COUNT;
+
+ // For mixed output and inputs, use max mixer channel count. Do not
+ // limit channel count otherwise
+ if (mType != AUDIO_PORT_TYPE_MIX) {
+ maxCount = UINT_MAX;
+ }
+ for (size_t i = 0; i < mChannelMasks.size(); i ++) {
+ uint32_t cnlCount;
+ if (mUseInChannelMask) {
+ cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]);
+ } else {
+ cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]);
+ }
+ if ((cnlCount > channelCount) && (cnlCount <= maxCount)) {
+ channelMask = mChannelMasks[i];
+ channelCount = cnlCount;
+ }
+ }
+ return channelMask;
+}
+
+/* format in order of increasing preference */
+const audio_format_t AudioPort::sPcmFormatCompareTable[] = {
+ AUDIO_FORMAT_DEFAULT,
+ AUDIO_FORMAT_PCM_16_BIT,
+ AUDIO_FORMAT_PCM_8_24_BIT,
+ AUDIO_FORMAT_PCM_24_BIT_PACKED,
+ AUDIO_FORMAT_PCM_32_BIT,
+ AUDIO_FORMAT_PCM_FLOAT,
+};
+
+int AudioPort::compareFormats(audio_format_t format1,
+ audio_format_t format2)
+{
+ // NOTE: AUDIO_FORMAT_INVALID is also considered not PCM and will be compared equal to any
+ // compressed format and better than any PCM format. This is by design of pickFormat()
+ if (!audio_is_linear_pcm(format1)) {
+ if (!audio_is_linear_pcm(format2)) {
+ return 0;
+ }
+ return 1;
+ }
+ if (!audio_is_linear_pcm(format2)) {
+ return -1;
+ }
+
+ int index1 = -1, index2 = -1;
+ for (size_t i = 0;
+ (i < ARRAY_SIZE(sPcmFormatCompareTable)) && ((index1 == -1) || (index2 == -1));
+ i ++) {
+ if (sPcmFormatCompareTable[i] == format1) {
+ index1 = i;
+ }
+ if (sPcmFormatCompareTable[i] == format2) {
+ index2 = i;
+ }
+ }
+ // format1 not found => index1 < 0 => format2 > format1
+ // format2 not found => index2 < 0 => format2 < format1
+ return index1 - index2;
+}
+
+audio_format_t AudioPort::pickFormat() const
+{
+ // special case for uninitialized dynamic profile
+ if (mFormats.size() == 1 && mFormats[0] == 0) {
+ return AUDIO_FORMAT_DEFAULT;
+ }
+
+ audio_format_t format = AUDIO_FORMAT_DEFAULT;
+ audio_format_t bestFormat =
+ AudioPort::sPcmFormatCompareTable[
+ ARRAY_SIZE(AudioPort::sPcmFormatCompareTable) - 1];
+ // For mixed output and inputs, use best mixer output format. Do not
+ // limit format otherwise
+ if ((mType != AUDIO_PORT_TYPE_MIX) ||
+ ((mRole == AUDIO_PORT_ROLE_SOURCE) &&
+ (((mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) != 0)))) {
+ bestFormat = AUDIO_FORMAT_INVALID;
+ }
+
+ for (size_t i = 0; i < mFormats.size(); i ++) {
+ if ((compareFormats(mFormats[i], format) > 0) &&
+ (compareFormats(mFormats[i], bestFormat) <= 0)) {
+ format = mFormats[i];
+ }
+ }
+ return format;
+}
+
+status_t AudioPort::checkGain(const struct audio_gain_config *gainConfig,
+ int index) const
+{
+ if (index < 0 || (size_t)index >= mGains.size()) {
+ return BAD_VALUE;
+ }
+ return mGains[index]->checkConfig(gainConfig);
+}
+
+void AudioPort::dump(int fd, int spaces) const
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ if (mName.size() != 0) {
+ snprintf(buffer, SIZE, "%*s- name: %s\n", spaces, "", mName.string());
+ result.append(buffer);
+ }
+
+ if (mSamplingRates.size() != 0) {
+ snprintf(buffer, SIZE, "%*s- sampling rates: ", spaces, "");
+ result.append(buffer);
+ for (size_t i = 0; i < mSamplingRates.size(); i++) {
+ if (i == 0 && mSamplingRates[i] == 0) {
+ snprintf(buffer, SIZE, "Dynamic");
+ } else {
+ snprintf(buffer, SIZE, "%d", mSamplingRates[i]);
+ }
+ result.append(buffer);
+ result.append(i == (mSamplingRates.size() - 1) ? "" : ", ");
+ }
+ result.append("\n");
+ }
+
+ if (mChannelMasks.size() != 0) {
+ snprintf(buffer, SIZE, "%*s- channel masks: ", spaces, "");
+ result.append(buffer);
+ for (size_t i = 0; i < mChannelMasks.size(); i++) {
+ ALOGV("AudioPort::dump mChannelMasks %zu %08x", i, mChannelMasks[i]);
+
+ if (i == 0 && mChannelMasks[i] == 0) {
+ snprintf(buffer, SIZE, "Dynamic");
+ } else {
+ snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]);
+ }
+ result.append(buffer);
+ result.append(i == (mChannelMasks.size() - 1) ? "" : ", ");
+ }
+ result.append("\n");
+ }
+
+ if (mFormats.size() != 0) {
+ snprintf(buffer, SIZE, "%*s- formats: ", spaces, "");
+ result.append(buffer);
+ for (size_t i = 0; i < mFormats.size(); i++) {
+ const char *formatStr = ConfigParsingUtils::enumToString(sFormatNameToEnumTable,
+ ARRAY_SIZE(sFormatNameToEnumTable),
+ mFormats[i]);
+ if (i == 0 && strcmp(formatStr, "") == 0) {
+ snprintf(buffer, SIZE, "Dynamic");
+ } else {
+ snprintf(buffer, SIZE, "%s", formatStr);
+ }
+ result.append(buffer);
+ result.append(i == (mFormats.size() - 1) ? "" : ", ");
+ }
+ result.append("\n");
+ }
+ write(fd, result.string(), result.size());
+ if (mGains.size() != 0) {
+ snprintf(buffer, SIZE, "%*s- gains:\n", spaces, "");
+ write(fd, buffer, strlen(buffer) + 1);
+ result.append(buffer);
+ for (size_t i = 0; i < mGains.size(); i++) {
+ mGains[i]->dump(fd, spaces + 2, i);
+ }
+ }
+}
+
+
+// --- AudioPortConfig class implementation
+
+AudioPortConfig::AudioPortConfig()
+{
+ mSamplingRate = 0;
+ mChannelMask = AUDIO_CHANNEL_NONE;
+ mFormat = AUDIO_FORMAT_INVALID;
+ mGain.index = -1;
+}
+
+status_t AudioPortConfig::applyAudioPortConfig(
+ const struct audio_port_config *config,
+ struct audio_port_config *backupConfig)
+{
+ struct audio_port_config localBackupConfig;
+ status_t status = NO_ERROR;
+
+ localBackupConfig.config_mask = config->config_mask;
+ toAudioPortConfig(&localBackupConfig);
+
+ sp<AudioPort> audioport = getAudioPort();
+ if (audioport == 0) {
+ status = NO_INIT;
+ goto exit;
+ }
+ if (config->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
+ status = audioport->checkExactSamplingRate(config->sample_rate);
+ if (status != NO_ERROR) {
+ goto exit;
+ }
+ mSamplingRate = config->sample_rate;
+ }
+ if (config->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
+ status = audioport->checkExactChannelMask(config->channel_mask);
+ if (status != NO_ERROR) {
+ goto exit;
+ }
+ mChannelMask = config->channel_mask;
+ }
+ if (config->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
+ status = audioport->checkFormat(config->format);
+ if (status != NO_ERROR) {
+ goto exit;
+ }
+ mFormat = config->format;
+ }
+ if (config->config_mask & AUDIO_PORT_CONFIG_GAIN) {
+ status = audioport->checkGain(&config->gain, config->gain.index);
+ if (status != NO_ERROR) {
+ goto exit;
+ }
+ mGain = config->gain;
+ }
+
+exit:
+ if (status != NO_ERROR) {
+ applyAudioPortConfig(&localBackupConfig);
+ }
+ if (backupConfig != NULL) {
+ *backupConfig = localBackupConfig;
+ }
+ return status;
+}
+
+void AudioPortConfig::toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig) const
+{
+ if (dstConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
+ dstConfig->sample_rate = mSamplingRate;
+ if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE)) {
+ dstConfig->sample_rate = srcConfig->sample_rate;
+ }
+ } else {
+ dstConfig->sample_rate = 0;
+ }
+ if (dstConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
+ dstConfig->channel_mask = mChannelMask;
+ if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK)) {
+ dstConfig->channel_mask = srcConfig->channel_mask;
+ }
+ } else {
+ dstConfig->channel_mask = AUDIO_CHANNEL_NONE;
+ }
+ if (dstConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
+ dstConfig->format = mFormat;
+ if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT)) {
+ dstConfig->format = srcConfig->format;
+ }
+ } else {
+ dstConfig->format = AUDIO_FORMAT_INVALID;
+ }
+ if (dstConfig->config_mask & AUDIO_PORT_CONFIG_GAIN) {
+ dstConfig->gain = mGain;
+ if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_GAIN)) {
+ dstConfig->gain = srcConfig->gain;
+ }
+ } else {
+ dstConfig->gain.index = -1;
+ }
+ if (dstConfig->gain.index != -1) {
+ dstConfig->config_mask |= AUDIO_PORT_CONFIG_GAIN;
+ } else {
+ dstConfig->config_mask &= ~AUDIO_PORT_CONFIG_GAIN;
+ }
+}
+
+
+// --- AudioPatch class implementation
+
+AudioPatch::AudioPatch(audio_patch_handle_t handle,
+ const struct audio_patch *patch, uid_t uid) :
+ mHandle(handle), mPatch(*patch), mUid(uid), mAfPatchHandle(0)
+{}
+
+status_t AudioPatch::dump(int fd, int spaces, int index) const
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "%*sAudio patch %d:\n", spaces, "", index+1);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- handle: %2d\n", spaces, "", mHandle);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- audio flinger handle: %2d\n", spaces, "", mAfPatchHandle);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- owner uid: %2d\n", spaces, "", mUid);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- %d sources:\n", spaces, "", mPatch.num_sources);
+ result.append(buffer);
+ for (size_t i = 0; i < mPatch.num_sources; i++) {
+ if (mPatch.sources[i].type == AUDIO_PORT_TYPE_DEVICE) {
+ snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "",
+ mPatch.sources[i].id, ConfigParsingUtils::enumToString(sDeviceNameToEnumTable,
+ ARRAY_SIZE(sDeviceNameToEnumTable),
+ mPatch.sources[i].ext.device.type));
+ } else {
+ snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "",
+ mPatch.sources[i].id, mPatch.sources[i].ext.mix.handle);
+ }
+ result.append(buffer);
+ }
+ snprintf(buffer, SIZE, "%*s- %d sinks:\n", spaces, "", mPatch.num_sinks);
+ result.append(buffer);
+ for (size_t i = 0; i < mPatch.num_sinks; i++) {
+ if (mPatch.sinks[i].type == AUDIO_PORT_TYPE_DEVICE) {
+ snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "",
+ mPatch.sinks[i].id, ConfigParsingUtils::enumToString(sDeviceNameToEnumTable,
+ ARRAY_SIZE(sDeviceNameToEnumTable),
+ mPatch.sinks[i].ext.device.type));
+ } else {
+ snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "",
+ mPatch.sinks[i].id, mPatch.sinks[i].ext.mix.handle);
+ }
+ result.append(buffer);
+ }
+
+ write(fd, result.string(), result.size());
+ return NO_ERROR;
+}
+
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/Ports.h b/services/audiopolicy/managerdefault/Ports.h
new file mode 100644
index 0000000..f6e0e93
--- /dev/null
+++ b/services/audiopolicy/managerdefault/Ports.h
@@ -0,0 +1,122 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+namespace android {
+
+class HwModule;
+
+class AudioPort: public virtual RefBase
+{
+public:
+ AudioPort(const String8& name, audio_port_type_t type,
+ audio_port_role_t role, const sp<HwModule>& module);
+ virtual ~AudioPort() {}
+
+ audio_port_handle_t getHandle() { return mId; }
+
+ void attach(const sp<HwModule>& module);
+ bool isAttached() { return mId != 0; }
+
+ virtual void toAudioPort(struct audio_port *port) const;
+
+ void importAudioPort(const sp<AudioPort> port);
+ void clearCapabilities();
+
+ void loadSamplingRates(char *name);
+ void loadFormats(char *name);
+ void loadOutChannels(char *name);
+ void loadInChannels(char *name);
+
+ audio_gain_mode_t loadGainMode(char *name);
+ void loadGain(cnode *root, int index);
+ virtual void loadGains(cnode *root);
+
+ // searches for an exact match
+ status_t checkExactSamplingRate(uint32_t samplingRate) const;
+ // searches for a compatible match, and returns the best match via updatedSamplingRate
+ status_t checkCompatibleSamplingRate(uint32_t samplingRate,
+ uint32_t *updatedSamplingRate) const;
+ // searches for an exact match
+ status_t checkExactChannelMask(audio_channel_mask_t channelMask) const;
+ // searches for a compatible match, currently implemented for input channel masks only
+ status_t checkCompatibleChannelMask(audio_channel_mask_t channelMask) const;
+ status_t checkFormat(audio_format_t format) const;
+ status_t checkGain(const struct audio_gain_config *gainConfig, int index) const;
+
+ uint32_t pickSamplingRate() const;
+ audio_channel_mask_t pickChannelMask() const;
+ audio_format_t pickFormat() const;
+
+ static const audio_format_t sPcmFormatCompareTable[];
+ static int compareFormats(audio_format_t format1, audio_format_t format2);
+
+ void dump(int fd, int spaces) const;
+
+ String8 mName;
+ audio_port_type_t mType;
+ audio_port_role_t mRole;
+ bool mUseInChannelMask;
+ // by convention, "0' in the first entry in mSamplingRates, mChannelMasks or mFormats
+ // indicates the supported parameters should be read from the output stream
+ // after it is opened for the first time
+ Vector <uint32_t> mSamplingRates; // supported sampling rates
+ Vector <audio_channel_mask_t> mChannelMasks; // supported channel masks
+ Vector <audio_format_t> mFormats; // supported audio formats
+ Vector < sp<AudioGain> > mGains; // gain controllers
+ sp<HwModule> mModule; // audio HW module exposing this I/O stream
+ uint32_t mFlags; // attribute flags (e.g primary output,
+ // direct output...).
+
+
+protected:
+ //TODO - clarify the role of mId in this case, both an "attached" indicator
+ // and a unique ID for identifying a port to the (upcoming) selection API,
+ // and its relationship to the mId in AudioOutputDescriptor and AudioInputDescriptor.
+ audio_port_handle_t mId;
+};
+
+class AudioPortConfig: public virtual RefBase
+{
+public:
+ AudioPortConfig();
+ virtual ~AudioPortConfig() {}
+
+ status_t applyAudioPortConfig(const struct audio_port_config *config,
+ struct audio_port_config *backupConfig = NULL);
+ virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig = NULL) const = 0;
+ virtual sp<AudioPort> getAudioPort() const = 0;
+ uint32_t mSamplingRate;
+ audio_format_t mFormat;
+ audio_channel_mask_t mChannelMask;
+ struct audio_gain_config mGain;
+};
+
+
+class AudioPatch: public RefBase
+{
+public:
+ AudioPatch(audio_patch_handle_t handle, const struct audio_patch *patch, uid_t uid);
+
+ status_t dump(int fd, int spaces, int index) const;
+
+ audio_patch_handle_t mHandle;
+ struct audio_patch mPatch;
+ uid_t mUid;
+ audio_patch_handle_t mAfPatchHandle;
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/audio_policy_conf.h b/services/audiopolicy/managerdefault/audio_policy_conf.h
similarity index 100%
rename from services/audiopolicy/audio_policy_conf.h
rename to services/audiopolicy/managerdefault/audio_policy_conf.h
diff --git a/services/audiopolicy/AudioPolicyClientImpl.cpp b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
similarity index 100%
rename from services/audiopolicy/AudioPolicyClientImpl.cpp
rename to services/audiopolicy/service/AudioPolicyClientImpl.cpp
diff --git a/services/audiopolicy/AudioPolicyClientImplLegacy.cpp b/services/audiopolicy/service/AudioPolicyClientImplLegacy.cpp
similarity index 100%
rename from services/audiopolicy/AudioPolicyClientImplLegacy.cpp
rename to services/audiopolicy/service/AudioPolicyClientImplLegacy.cpp
diff --git a/services/audiopolicy/AudioPolicyEffects.cpp b/services/audiopolicy/service/AudioPolicyEffects.cpp
similarity index 100%
rename from services/audiopolicy/AudioPolicyEffects.cpp
rename to services/audiopolicy/service/AudioPolicyEffects.cpp
diff --git a/services/audiopolicy/AudioPolicyEffects.h b/services/audiopolicy/service/AudioPolicyEffects.h
similarity index 100%
rename from services/audiopolicy/AudioPolicyEffects.h
rename to services/audiopolicy/service/AudioPolicyEffects.h
diff --git a/services/audiopolicy/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
similarity index 98%
rename from services/audiopolicy/AudioPolicyInterfaceImpl.cpp
rename to services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
index a45dbb3..e9ff8389 100644
--- a/services/audiopolicy/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
@@ -28,7 +28,8 @@
status_t AudioPolicyService::setDeviceConnectionState(audio_devices_t device,
audio_policy_dev_state_t state,
- const char *device_address)
+ const char *device_address,
+ const char *device_name)
{
if (mAudioPolicyManager == NULL) {
return NO_INIT;
@@ -46,8 +47,8 @@
ALOGV("setDeviceConnectionState()");
Mutex::Autolock _l(mLock);
- return mAudioPolicyManager->setDeviceConnectionState(device,
- state, device_address);
+ return mAudioPolicyManager->setDeviceConnectionState(device, state,
+ device_address, device_name);
}
audio_policy_dev_state_t AudioPolicyService::getDeviceConnectionState(
diff --git a/services/audiopolicy/AudioPolicyInterfaceImplLegacy.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImplLegacy.cpp
similarity index 99%
rename from services/audiopolicy/AudioPolicyInterfaceImplLegacy.cpp
rename to services/audiopolicy/service/AudioPolicyInterfaceImplLegacy.cpp
index b8846c6..5a91192 100644
--- a/services/audiopolicy/AudioPolicyInterfaceImplLegacy.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImplLegacy.cpp
@@ -33,7 +33,8 @@
status_t AudioPolicyService::setDeviceConnectionState(audio_devices_t device,
audio_policy_dev_state_t state,
- const char *device_address)
+ const char *device_address,
+ const char *device_name __unused)
{
if (mpAudioPolicy == NULL) {
return NO_INIT;
diff --git a/services/audiopolicy/AudioPolicyService.cpp b/services/audiopolicy/service/AudioPolicyService.cpp
similarity index 100%
rename from services/audiopolicy/AudioPolicyService.cpp
rename to services/audiopolicy/service/AudioPolicyService.cpp
diff --git a/services/audiopolicy/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h
similarity index 99%
rename from services/audiopolicy/AudioPolicyService.h
rename to services/audiopolicy/service/AudioPolicyService.h
index 80284a4..0378384 100644
--- a/services/audiopolicy/AudioPolicyService.h
+++ b/services/audiopolicy/service/AudioPolicyService.h
@@ -35,7 +35,7 @@
#include <hardware_legacy/AudioPolicyInterface.h>
#endif
#include "AudioPolicyEffects.h"
-#include "AudioPolicyManager.h"
+#include "managerdefault/AudioPolicyManager.h"
namespace android {
@@ -61,7 +61,8 @@
virtual status_t setDeviceConnectionState(audio_devices_t device,
audio_policy_dev_state_t state,
- const char *device_address);
+ const char *device_address,
+ const char *device_name);
virtual audio_policy_dev_state_t getDeviceConnectionState(
audio_devices_t device,
const char *device_address);
diff --git a/services/camera/libcameraservice/Android.mk b/services/camera/libcameraservice/Android.mk
index e184d97..5d6423a 100644
--- a/services/camera/libcameraservice/Android.mk
+++ b/services/camera/libcameraservice/Android.mk
@@ -23,8 +23,10 @@
LOCAL_SRC_FILES:= \
CameraService.cpp \
CameraDeviceFactory.cpp \
+ CameraFlashlight.cpp \
common/Camera2ClientBase.cpp \
common/CameraDeviceBase.cpp \
+ common/CameraModule.cpp \
common/FrameProcessorBase.cpp \
api1/CameraClient.cpp \
api1/Camera2Client.cpp \
diff --git a/services/camera/libcameraservice/CameraFlashlight.cpp b/services/camera/libcameraservice/CameraFlashlight.cpp
new file mode 100644
index 0000000..00a70eb
--- /dev/null
+++ b/services/camera/libcameraservice/CameraFlashlight.cpp
@@ -0,0 +1,520 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "CameraFlashlight"
+#define ATRACE_TAG ATRACE_TAG_CAMERA
+#define LOG_NDEBUG 0
+
+#include <utils/Log.h>
+#include <utils/Trace.h>
+#include <cutils/properties.h>
+
+#include "camera/CameraMetadata.h"
+#include "CameraFlashlight.h"
+#include "gui/IGraphicBufferConsumer.h"
+#include "gui/BufferQueue.h"
+#include "camera/camera2/CaptureRequest.h"
+#include "CameraDeviceFactory.h"
+
+
+namespace android {
+
+CameraFlashlight::CameraFlashlight(CameraModule& cameraModule,
+ const camera_module_callbacks_t& callbacks) :
+ mCameraModule(&cameraModule),
+ mCallbacks(&callbacks) {
+}
+
+CameraFlashlight::~CameraFlashlight() {
+}
+
+status_t CameraFlashlight::createFlashlightControl(const String16& cameraId) {
+ ALOGV("%s: creating a flash light control for camera %s", __FUNCTION__,
+ cameraId.string());
+ if (mFlashControl != NULL) {
+ return INVALID_OPERATION;
+ }
+
+ status_t res = OK;
+
+ if (mCameraModule->getRawModule()->module_api_version >=
+ CAMERA_MODULE_API_VERSION_2_4) {
+ mFlashControl = new FlashControl(*mCameraModule, *mCallbacks);
+ if (mFlashControl == NULL) {
+ ALOGV("%s: cannot create flash control for module api v2.4+",
+ __FUNCTION__);
+ return NO_MEMORY;
+ }
+ } else {
+ uint32_t deviceVersion = CAMERA_DEVICE_API_VERSION_1_0;
+
+ if (mCameraModule->getRawModule()->module_api_version >=
+ CAMERA_MODULE_API_VERSION_2_0) {
+ camera_info info;
+ res = mCameraModule->getCameraInfo(
+ atoi(String8(cameraId).string()), &info);
+ if (res) {
+ ALOGV("%s: failed to get camera info for camera %s",
+ __FUNCTION__, cameraId.string());
+ return res;
+ }
+ deviceVersion = info.device_version;
+ }
+
+ if (deviceVersion >= CAMERA_DEVICE_API_VERSION_2_0) {
+ CameraDeviceClientFlashControl *flashControl =
+ new CameraDeviceClientFlashControl(*mCameraModule,
+ *mCallbacks);
+ if (!flashControl) {
+ return NO_MEMORY;
+ }
+
+ mFlashControl = flashControl;
+ }
+ else {
+ // todo: implement for device api 1
+ return INVALID_OPERATION;
+ }
+ }
+
+ return OK;
+}
+
+status_t CameraFlashlight::setTorchMode(const String16& cameraId, bool enabled) {
+ if (!mCameraModule) {
+ return NO_INIT;
+ }
+
+ ALOGV("%s: set torch mode of camera %s to %d", __FUNCTION__,
+ cameraId.string(), enabled);
+
+ status_t res = OK;
+ Mutex::Autolock l(mLock);
+
+ if (mFlashControl == NULL) {
+ res = createFlashlightControl(cameraId);
+ if (res) {
+ return res;
+ }
+ res = mFlashControl->setTorchMode(cameraId, enabled);
+ return res;
+ }
+
+ // if flash control already exists, turning on torch mode may fail if it's
+ // tied to another camera device for module v2.3 and below.
+ res = mFlashControl->setTorchMode(cameraId, enabled);
+ if (res == BAD_INDEX) {
+ // flash control is tied to another camera device, need to close it and
+ // try again.
+ mFlashControl.clear();
+ res = createFlashlightControl(cameraId);
+ if (res) {
+ return res;
+ }
+ res = mFlashControl->setTorchMode(cameraId, enabled);
+ }
+
+ return res;
+}
+
+bool CameraFlashlight::hasFlashUnit(const String16& cameraId) {
+ status_t res;
+
+ Mutex::Autolock l(mLock);
+
+ if (mFlashControl == NULL) {
+ res = createFlashlightControl(cameraId);
+ if (res) {
+ ALOGE("%s: failed to create flash control for %s ",
+ __FUNCTION__, cameraId.string());
+ return false;
+ }
+ }
+
+ bool flashUnit = false;
+
+ // if flash control already exists, querying if a camera device has a flash
+ // unit may fail if it's module v1
+ res = mFlashControl->hasFlashUnit(cameraId, &flashUnit);
+ if (res == BAD_INDEX) {
+ // need to close the flash control before query.
+ mFlashControl.clear();
+ res = createFlashlightControl(cameraId);
+ if (res) {
+ ALOGE("%s: failed to create flash control for %s ", __FUNCTION__,
+ cameraId.string());
+ return false;
+ }
+ res = mFlashControl->hasFlashUnit(cameraId, &flashUnit);
+ if (res) {
+ flashUnit = false;
+ }
+ }
+
+ return flashUnit;
+}
+
+status_t CameraFlashlight::prepareDeviceOpen() {
+ ALOGV("%s: prepare for device open", __FUNCTION__);
+
+ Mutex::Autolock l(mLock);
+
+ if (mCameraModule && mCameraModule->getRawModule()->module_api_version <
+ CAMERA_MODULE_API_VERSION_2_4) {
+ // framework is going to open a camera device, all flash light control
+ // should be closed for backward compatible support.
+ if (mFlashControl != NULL) {
+ mFlashControl.clear();
+ }
+ }
+
+ return OK;
+}
+
+
+FlashControlBase::~FlashControlBase() {
+}
+
+
+FlashControl::FlashControl(CameraModule& cameraModule,
+ const camera_module_callbacks_t& callbacks) :
+ mCameraModule(&cameraModule) {
+}
+
+FlashControl::~FlashControl() {
+}
+
+status_t FlashControl::hasFlashUnit(const String16& cameraId, bool *hasFlash) {
+ if (!hasFlash) {
+ return BAD_VALUE;
+ }
+
+ *hasFlash = false;
+
+ Mutex::Autolock l(mLock);
+
+ if (!mCameraModule) {
+ return NO_INIT;
+ }
+
+ camera_info info;
+ status_t res = mCameraModule->getCameraInfo(atoi(String8(cameraId).string()),
+ &info);
+ if (res != 0) {
+ return res;
+ }
+
+ CameraMetadata metadata;
+ metadata = info.static_camera_characteristics;
+ camera_metadata_entry flashAvailable =
+ metadata.find(ANDROID_FLASH_INFO_AVAILABLE);
+ if (flashAvailable.count == 1 && flashAvailable.data.u8[0] == 1) {
+ *hasFlash = true;
+ }
+
+ return OK;
+}
+
+status_t FlashControl::setTorchMode(const String16& cameraId, bool enabled) {
+ ALOGV("%s: set camera %s torch mode to %d", __FUNCTION__,
+ cameraId.string(), enabled);
+
+ Mutex::Autolock l(mLock);
+ if (!mCameraModule) {
+ return NO_INIT;
+ }
+
+ return mCameraModule->setTorchMode(String8(cameraId).string(), enabled);
+}
+
+CameraDeviceClientFlashControl::CameraDeviceClientFlashControl(
+ CameraModule& cameraModule,
+ const camera_module_callbacks_t& callbacks) :
+ mCameraModule(&cameraModule),
+ mCallbacks(&callbacks),
+ mTorchEnabled(false),
+ mMetadata(NULL) {
+}
+
+CameraDeviceClientFlashControl::~CameraDeviceClientFlashControl() {
+ if (mDevice != NULL) {
+ mDevice->flush();
+ mDevice->deleteStream(mStreamId);
+ mDevice.clear();
+ }
+ if (mMetadata) {
+ delete mMetadata;
+ }
+
+ mAnw.clear();
+ mSurfaceTexture.clear();
+ mProducer.clear();
+ mConsumer.clear();
+
+ if (mTorchEnabled) {
+ if (mCallbacks) {
+ ALOGV("%s: notify the framework that torch was turned off",
+ __FUNCTION__);
+ mCallbacks->torch_mode_status_change(mCallbacks,
+ String8(mCameraId).string(), TORCH_MODE_STATUS_OFF);
+ }
+ }
+}
+
+status_t CameraDeviceClientFlashControl::initializeSurface(int32_t width,
+ int32_t height) {
+ status_t res;
+ BufferQueue::createBufferQueue(&mProducer, &mConsumer);
+
+ mSurfaceTexture = new GLConsumer(mConsumer, 0, GLConsumer::TEXTURE_EXTERNAL,
+ true, true);
+ if (mSurfaceTexture == NULL) {
+ return NO_MEMORY;
+ }
+
+ int32_t format = HAL_PIXEL_FORMAT_IMPLEMENTATION_DEFINED;
+ res = mSurfaceTexture->setDefaultBufferSize(width, height);
+ if (res) {
+ return res;
+ }
+ res = mSurfaceTexture->setDefaultBufferFormat(format);
+ if (res) {
+ return res;
+ }
+
+ bool useAsync = false;
+ int32_t consumerUsage;
+ res = mProducer->query(NATIVE_WINDOW_CONSUMER_USAGE_BITS, &consumerUsage);
+ if (res) {
+ return res;
+ }
+
+ if (consumerUsage & GraphicBuffer::USAGE_HW_TEXTURE) {
+ useAsync = true;
+ }
+
+ mAnw = new Surface(mProducer, useAsync);
+ if (mAnw == NULL) {
+ return NO_MEMORY;
+ }
+ res = mDevice->createStream(mAnw, width, height, format, &mStreamId);
+ if (res) {
+ return res;
+ }
+
+ res = mDevice->configureStreams();
+ if (res) {
+ return res;
+ }
+
+ return res;
+}
+
+status_t CameraDeviceClientFlashControl::getSmallestSurfaceSize(
+ const camera_info& info, int32_t *width, int32_t *height) {
+ if (!width || !height) {
+ return BAD_VALUE;
+ }
+
+ int32_t w = INT32_MAX;
+ int32_t h = 1;
+
+ CameraMetadata metadata;
+ metadata = info.static_camera_characteristics;
+ camera_metadata_entry streamConfigs =
+ metadata.find(ANDROID_SCALER_AVAILABLE_STREAM_CONFIGURATIONS);
+ for (size_t i = 0; i < streamConfigs.count; i += 4) {
+ int32_t fmt = streamConfigs.data.i32[i];
+ if (fmt == ANDROID_SCALER_AVAILABLE_FORMATS_IMPLEMENTATION_DEFINED) {
+ int32_t ww = streamConfigs.data.i32[i + 1];
+ int32_t hh = streamConfigs.data.i32[i + 2];
+
+ if (w* h > ww * hh) {
+ w = ww;
+ h = hh;
+ }
+ }
+ }
+
+ if (w == INT32_MAX) {
+ return NAME_NOT_FOUND;
+ }
+
+ *width = w;
+ *height = h;
+
+ return OK;
+}
+
+status_t CameraDeviceClientFlashControl::connectCameraDevice(
+ const String16& cameraId) {
+ String8 id = String8(cameraId);
+ camera_info info;
+ status_t res = mCameraModule->getCameraInfo(atoi(id.string()), &info);
+ if (res != 0) {
+ ALOGE("%s: failed to get camera info for camera %s", __FUNCTION__,
+ mCameraId.string());
+ return res;
+ }
+
+ mDevice = CameraDeviceFactory::createDevice(atoi(id.string()));
+ if (mDevice == NULL) {
+ return NO_MEMORY;
+ }
+
+ res = mDevice->initialize(mCameraModule);
+ if (res) {
+ goto fail;
+ }
+
+ int32_t width, height;
+ res = getSmallestSurfaceSize(info, &width, &height);
+ if (res) {
+ return res;
+ }
+ res = initializeSurface(width, height);
+ if (res) {
+ goto fail;
+ }
+
+ mCameraId = cameraId;
+
+ return OK;
+
+fail:
+ mDevice.clear();
+ return res;
+}
+
+
+status_t CameraDeviceClientFlashControl::hasFlashUnit(const String16& cameraId,
+ bool *hasFlash) {
+ ALOGV("%s: checking if camera %s has a flash unit", __FUNCTION__,
+ cameraId.string());
+
+ Mutex::Autolock l(mLock);
+ return hasFlashUnitLocked(cameraId, hasFlash);
+
+}
+
+status_t CameraDeviceClientFlashControl::hasFlashUnitLocked(
+ const String16& cameraId, bool *hasFlash) {
+ if (!mCameraModule) {
+ ALOGE("%s: camera module is NULL", __FUNCTION__);
+ return NO_INIT;
+ }
+
+ if (!hasFlash) {
+ return BAD_VALUE;
+ }
+
+ camera_info info;
+ status_t res = mCameraModule->getCameraInfo(
+ atoi(String8(cameraId).string()), &info);
+ if (res != 0) {
+ ALOGE("%s: failed to get camera info for camera %s", __FUNCTION__,
+ cameraId.string());
+ return res;
+ }
+
+ CameraMetadata metadata;
+ metadata = info.static_camera_characteristics;
+ camera_metadata_entry flashAvailable =
+ metadata.find(ANDROID_FLASH_INFO_AVAILABLE);
+ if (flashAvailable.count == 1 && flashAvailable.data.u8[0] == 1) {
+ *hasFlash = true;
+ }
+
+ return OK;
+}
+
+status_t CameraDeviceClientFlashControl::submitTorchRequest(bool enabled) {
+ status_t res;
+
+ if (mMetadata == NULL) {
+ mMetadata = new CameraMetadata();
+ if (mMetadata == NULL) {
+ return NO_MEMORY;
+ }
+ res = mDevice->createDefaultRequest(
+ CAMERA3_TEMPLATE_PREVIEW, mMetadata);
+ if (res) {
+ return res;
+ }
+ }
+
+ uint8_t torchOn = enabled ? ANDROID_FLASH_MODE_TORCH :
+ ANDROID_FLASH_MODE_OFF;
+
+ mMetadata->update(ANDROID_FLASH_MODE, &torchOn, 1);
+ mMetadata->update(ANDROID_REQUEST_OUTPUT_STREAMS, &mStreamId, 1);
+
+ int32_t requestId = 0;
+ mMetadata->update(ANDROID_REQUEST_ID, &requestId, 1);
+
+ List<const CameraMetadata> metadataRequestList;
+ metadataRequestList.push_back(*mMetadata);
+
+ int64_t lastFrameNumber = 0;
+ res = mDevice->captureList(metadataRequestList, &lastFrameNumber);
+
+ return res;
+}
+
+
+status_t CameraDeviceClientFlashControl::setTorchMode(
+ const String16& cameraId, bool enabled) {
+ bool hasFlash = false;
+
+ Mutex::Autolock l(mLock);
+ status_t res = hasFlashUnitLocked(cameraId, &hasFlash);
+
+ // pre-check
+ if (enabled) {
+ // invalid camera?
+ if (res) {
+ return -EINVAL;
+ }
+ // no flash unit?
+ if (!hasFlash) {
+ return -ENOSYS;
+ }
+ // already opened for a different device?
+ if (mDevice != NULL && cameraId != mCameraId) {
+ return BAD_INDEX;
+ }
+ } else if (mDevice == NULL || cameraId != mCameraId) {
+ // disabling the torch mode of an un-opened or different device.
+ return OK;
+ }
+
+ if (mDevice == NULL) {
+ res = connectCameraDevice(cameraId);
+ if (res) {
+ return res;
+ }
+ }
+
+ res = submitTorchRequest(enabled);
+ if (res) {
+ return res;
+ }
+
+ mTorchEnabled = enabled;
+ return OK;
+}
+
+}
diff --git a/services/camera/libcameraservice/CameraFlashlight.h b/services/camera/libcameraservice/CameraFlashlight.h
new file mode 100644
index 0000000..a0de0b0
--- /dev/null
+++ b/services/camera/libcameraservice/CameraFlashlight.h
@@ -0,0 +1,149 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_SERVERS_CAMERA_CAMERAFLASHLIGHT_H
+#define ANDROID_SERVERS_CAMERA_CAMERAFLASHLIGHT_H
+
+#include "hardware/camera_common.h"
+#include "utils/KeyedVector.h"
+#include "gui/GLConsumer.h"
+#include "gui/Surface.h"
+#include "common/CameraDeviceBase.h"
+
+namespace android {
+
+/**
+ * FlashControlBase is a base class for flash control. It defines the functions
+ * that a flash control for each camera module/device version should implement.
+ */
+class FlashControlBase : public virtual VirtualLightRefBase {
+ public:
+ virtual ~FlashControlBase();
+
+ // Whether a camera device has a flash unit. Calling this function may
+ // cause the torch mode to be turned off in HAL v1 devices. If
+ // previously-on torch mode is turned off,
+ // callbacks.torch_mode_status_change() should be invoked.
+ virtual status_t hasFlashUnit(const String16& cameraId,
+ bool *hasFlash) = 0;
+
+ // set the torch mode to on or off.
+ virtual status_t setTorchMode(const String16& cameraId,
+ bool enabled) = 0;
+};
+
+/**
+ * CameraFlashlight can be used by camera service to control flashflight.
+ */
+class CameraFlashlight : public virtual VirtualLightRefBase {
+ public:
+ CameraFlashlight(CameraModule& cameraModule,
+ const camera_module_callbacks_t& callbacks);
+ virtual ~CameraFlashlight();
+
+ // set the torch mode to on or off.
+ status_t setTorchMode(const String16& cameraId, bool enabled);
+
+ // Whether a camera device has a flash unit. Calling this function may
+ // cause the torch mode to be turned off in HAL v1 devices.
+ bool hasFlashUnit(const String16& cameraId);
+
+ // Notify CameraFlashlight that camera service is going to open a camera
+ // device. CameraFlashlight will free the resources that may cause the
+ // camera open to fail. Camera service must call this function before
+ // opening a camera device.
+ status_t prepareDeviceOpen();
+
+ private:
+ // create flashlight control based on camera module API and camera
+ // device API versions.
+ status_t createFlashlightControl(const String16& cameraId);
+
+ sp<FlashControlBase> mFlashControl;
+ CameraModule *mCameraModule;
+ const camera_module_callbacks_t *mCallbacks;
+
+ Mutex mLock;
+};
+
+/**
+ * Flash control for camera module v2.4 and above.
+ */
+class FlashControl : public FlashControlBase {
+ public:
+ FlashControl(CameraModule& cameraModule,
+ const camera_module_callbacks_t& callbacks);
+ virtual ~FlashControl();
+
+ // FlashControlBase
+ status_t hasFlashUnit(const String16& cameraId, bool *hasFlash);
+ status_t setTorchMode(const String16& cameraId, bool enabled);
+
+ private:
+ CameraModule *mCameraModule;
+
+ Mutex mLock;
+};
+
+/**
+ * Flash control for camera module <= v2.3 and camera HAL v2-v3
+ */
+class CameraDeviceClientFlashControl : public FlashControlBase {
+ public:
+ CameraDeviceClientFlashControl(CameraModule& cameraModule,
+ const camera_module_callbacks_t& callbacks);
+ virtual ~CameraDeviceClientFlashControl();
+
+ // FlashControlBase
+ status_t setTorchMode(const String16& cameraId, bool enabled);
+ status_t hasFlashUnit(const String16& cameraId, bool *hasFlash);
+
+ private:
+ // connect to a camera device
+ status_t connectCameraDevice(const String16& cameraId);
+
+ // initialize a surface
+ status_t initializeSurface(int32_t width, int32_t height);
+
+ // submit a request with the given torch mode
+ status_t submitTorchRequest(bool enabled);
+
+ // get the smallest surface size of IMPLEMENTATION_DEFINED
+ status_t getSmallestSurfaceSize(const camera_info& info, int32_t *width,
+ int32_t *height);
+
+ status_t hasFlashUnitLocked(const String16& cameraId, bool *hasFlash);
+
+ CameraModule *mCameraModule;
+ const camera_module_callbacks_t *mCallbacks;
+ String16 mCameraId;
+ bool mTorchEnabled;
+ CameraMetadata *mMetadata;
+
+ sp<CameraDeviceBase> mDevice;
+
+ sp<IGraphicBufferProducer> mProducer;
+ sp<IGraphicBufferConsumer> mConsumer;
+ sp<GLConsumer> mSurfaceTexture;
+ sp<ANativeWindow> mAnw;
+ int32_t mStreamId;
+
+ Mutex mLock;
+};
+
+} // namespace android
+
+#endif
diff --git a/services/camera/libcameraservice/CameraService.cpp b/services/camera/libcameraservice/CameraService.cpp
index 1232c32..d65ac21 100644
--- a/services/camera/libcameraservice/CameraService.cpp
+++ b/services/camera/libcameraservice/CameraService.cpp
@@ -29,6 +29,7 @@
#include <binder/MemoryHeapBase.h>
#include <cutils/atomic.h>
#include <cutils/properties.h>
+#include <cutils/multiuser.h>
#include <gui/Surface.h>
#include <hardware/hardware.h>
#include <media/AudioSystem.h>
@@ -86,6 +87,38 @@
camera_id,
new_status);
}
+
+static void torch_mode_status_change(
+ const struct camera_module_callbacks* callbacks,
+ const char* camera_id,
+ int new_status) {
+ if (!callbacks || !camera_id) {
+ ALOGE("%s invalid parameters. callbacks %p, camera_id %p", __FUNCTION__,
+ callbacks, camera_id);
+ }
+ sp<CameraService> cs = const_cast<CameraService*>(
+ static_cast<const CameraService*>(callbacks));
+
+ ICameraServiceListener::TorchStatus status;
+ switch (new_status) {
+ case TORCH_MODE_STATUS_AVAILABLE:
+ status = ICameraServiceListener::TORCH_STATUS_AVAILABLE;
+ break;
+ case TORCH_MODE_STATUS_RESOURCE_BUSY:
+ status = ICameraServiceListener::TORCH_STATUS_NOT_AVAILABLE;
+ break;
+ case TORCH_MODE_STATUS_OFF:
+ status = ICameraServiceListener::TORCH_STATUS_OFF;
+ break;
+ default:
+ ALOGE("Unknown torch status %d", new_status);
+ return;
+ }
+
+ cs->onTorchStatusChanged(
+ String16(camera_id),
+ status);
+}
} // extern "C"
// ----------------------------------------------------------------------------
@@ -95,7 +128,7 @@
static CameraService *gCameraService;
CameraService::CameraService()
- :mSoundRef(0), mModule(0)
+ :mSoundRef(0), mModule(0), mFlashlight(0)
{
ALOGI("CameraService started (pid=%d)", getpid());
gCameraService = this;
@@ -105,6 +138,8 @@
}
this->camera_device_status_change = android::camera_device_status_change;
+ this->torch_mode_status_change = android::torch_mode_status_change;
+
}
void CameraService::onFirstRef()
@@ -113,14 +148,19 @@
BnCameraService::onFirstRef();
+ camera_module_t *rawModule;
if (hw_get_module(CAMERA_HARDWARE_MODULE_ID,
- (const hw_module_t **)&mModule) < 0) {
+ (const hw_module_t **)&rawModule) < 0) {
ALOGE("Could not load camera HAL module");
mNumberOfCameras = 0;
}
else {
- ALOGI("Loaded \"%s\" camera module", mModule->common.name);
- mNumberOfCameras = mModule->get_number_of_cameras();
+ mModule = new CameraModule(rawModule);
+ mFlashlight = new CameraFlashlight(*mModule, *this);
+
+ const hw_module_t *common = mModule->getRawModule();
+ ALOGI("Loaded \"%s\" camera module", common->name);
+ mNumberOfCameras = mModule->getNumberOfCameras();
if (mNumberOfCameras > MAX_CAMERAS) {
ALOGE("Number of cameras(%d) > MAX_CAMERAS(%d).",
mNumberOfCameras, MAX_CAMERAS);
@@ -128,16 +168,21 @@
}
for (int i = 0; i < mNumberOfCameras; i++) {
setCameraFree(i);
+
+ String16 cameraName = String16(String8::format("%d", i));
+ if (mFlashlight->hasFlashUnit(cameraName)) {
+ mTorchStatusMap.add(cameraName,
+ ICameraServiceListener::TORCH_STATUS_AVAILABLE);
+ }
}
- if (mModule->common.module_api_version >=
- CAMERA_MODULE_API_VERSION_2_1) {
- mModule->set_callbacks(this);
+ if (common->module_api_version >= CAMERA_MODULE_API_VERSION_2_1) {
+ mModule->setCallbacks(this);
}
VendorTagDescriptor::clearGlobalVendorTagDescriptor();
- if (mModule->common.module_api_version >= CAMERA_MODULE_API_VERSION_2_2) {
+ if (common->module_api_version >= CAMERA_MODULE_API_VERSION_2_2) {
setUpVendorTags();
}
@@ -152,6 +197,9 @@
}
}
+ if (mModule) {
+ delete mModule;
+ }
VendorTagDescriptor::clearGlobalVendorTagDescriptor();
gCameraService = NULL;
}
@@ -220,6 +268,37 @@
}
+void CameraService::onTorchStatusChanged(const String16& cameraId,
+ ICameraServiceListener::TorchStatus newStatus) {
+ Mutex::Autolock al(mTorchStatusMutex);
+ onTorchStatusChangedLocked(cameraId, newStatus);
+}
+
+void CameraService::onTorchStatusChangedLocked(const String16& cameraId,
+ ICameraServiceListener::TorchStatus newStatus) {
+ ALOGI("%s: Torch status changed for cameraId=%s, newStatus=%d",
+ __FUNCTION__, cameraId.string(), newStatus);
+
+ if (getTorchStatusLocked(cameraId) == newStatus) {
+ ALOGE("%s: Torch state transition to the same status 0x%x not allowed",
+ __FUNCTION__, (uint32_t)newStatus);
+ return;
+ }
+
+ status_t res = setTorchStatusLocked(cameraId, newStatus);
+ if (res) {
+ ALOGE("%s: Failed to set the torch status", __FUNCTION__,
+ (uint32_t)newStatus);
+ return;
+ }
+
+ Vector<sp<ICameraServiceListener> >::const_iterator it;
+ for (it = mListenerList.begin(); it != mListenerList.end(); ++it) {
+ (*it)->onTorchStatusChanged(newStatus, cameraId);
+ }
+}
+
+
int32_t CameraService::getNumberOfCameras() {
return mNumberOfCameras;
}
@@ -236,7 +315,7 @@
struct camera_info info;
status_t rc = filterGetInfoErrorCode(
- mModule->get_camera_info(cameraId, &info));
+ mModule->getCameraInfo(cameraId, &info));
cameraInfo->facing = info.facing;
cameraInfo->orientation = info.orientation;
return rc;
@@ -347,7 +426,7 @@
int facing;
status_t ret = OK;
- if (mModule->common.module_api_version < CAMERA_MODULE_API_VERSION_2_0 ||
+ if (mModule->getRawModule()->module_api_version < CAMERA_MODULE_API_VERSION_2_0 ||
getDeviceVersion(cameraId, &facing) <= CAMERA_DEVICE_API_VERSION_2_1 ) {
/**
* Backwards compatibility mode for old HALs:
@@ -368,7 +447,7 @@
* Normal HAL 2.1+ codepath.
*/
struct camera_info info;
- ret = filterGetInfoErrorCode(mModule->get_camera_info(cameraId, &info));
+ ret = filterGetInfoErrorCode(mModule->getCameraInfo(cameraId, &info));
*cameraInfo = info.static_camera_characteristics;
}
@@ -387,12 +466,12 @@
int CameraService::getDeviceVersion(int cameraId, int* facing) {
struct camera_info info;
- if (mModule->get_camera_info(cameraId, &info) != OK) {
+ if (mModule->getCameraInfo(cameraId, &info) != OK) {
return -1;
}
int deviceVersion;
- if (mModule->common.module_api_version >= CAMERA_MODULE_API_VERSION_2_0) {
+ if (mModule->getRawModule()->module_api_version >= CAMERA_MODULE_API_VERSION_2_0) {
deviceVersion = info.device_version;
} else {
deviceVersion = CAMERA_DEVICE_API_VERSION_1_0;
@@ -433,13 +512,13 @@
vendor_tag_ops_t vOps = vendor_tag_ops_t();
// Check if vendor operations have been implemented
- if (mModule->get_vendor_tag_ops == NULL) {
+ if (!mModule->isVendorTagDefined()) {
ALOGI("%s: No vendor tags defined for this device.", __FUNCTION__);
return false;
}
ATRACE_BEGIN("camera3->get_metadata_vendor_tag_ops");
- mModule->get_vendor_tag_ops(&vOps);
+ mModule->getVendorTagOps(&vOps);
ATRACE_END();
// Ensure all vendor operations are present
@@ -592,7 +671,10 @@
}
char value[PROPERTY_VALUE_MAX];
- property_get("sys.secpolicy.camera.disabled", value, "0");
+ char key[PROPERTY_KEY_MAX];
+ int clientUserId = multiuser_get_user_id(clientUid);
+ snprintf(key, PROPERTY_KEY_MAX, "sys.secpolicy.camera.off_%d", clientUserId);
+ property_get(key, value, "0");
if (strcmp(value, "1") == 0) {
// Camera is disabled by DevicePolicyManager.
ALOGI("Camera is disabled. connect X (pid %d) rejected", callingPid);
@@ -671,6 +753,9 @@
int halVersion,
bool legacyMode) {
+ // give flashlight a chance to close devices if necessary.
+ mFlashlight->prepareDeviceOpen();
+
int facing = -1;
int deviceVersion = getDeviceVersion(cameraId, &facing);
@@ -789,8 +874,9 @@
/*out*/
sp<ICamera>& device) {
+ int apiVersion = mModule->getRawModule()->module_api_version;
if (halVersion != CAMERA_HAL_API_VERSION_UNSPECIFIED &&
- mModule->common.module_api_version < CAMERA_MODULE_API_VERSION_2_3) {
+ apiVersion < CAMERA_MODULE_API_VERSION_2_3) {
/*
* Either the HAL version is unspecified in which case this just creates
* a camera client selected by the latest device version, or
@@ -798,7 +884,7 @@
* the open_legacy call
*/
ALOGE("%s: camera HAL module version %x doesn't support connecting to legacy HAL devices!",
- __FUNCTION__, mModule->common.module_api_version);
+ __FUNCTION__, apiVersion);
return INVALID_OPERATION;
}
@@ -846,6 +932,47 @@
return OK;
}
+status_t CameraService::setTorchMode(const String16& cameraId, bool enabled,
+ const sp<IBinder>& clientBinder) {
+ if (enabled && clientBinder == NULL) {
+ ALOGE("%s: torch client binder is NULL", __FUNCTION__);
+ return -ENOSYS;
+ }
+
+ Mutex::Autolock al(mTorchStatusMutex);
+ status_t res = mFlashlight->setTorchMode(cameraId, enabled);
+ if (res) {
+ ALOGE("%s: setting torch mode of camera %s to %d failed", __FUNCTION__,
+ cameraId.string(), enabled);
+ return res;
+ }
+
+ // update the link to client's death
+ ssize_t index = mTorchClientMap.indexOfKey(cameraId);
+ if (enabled) {
+ if (index == NAME_NOT_FOUND) {
+ mTorchClientMap.add(cameraId, clientBinder);
+ } else {
+ const sp<IBinder> oldBinder = mTorchClientMap.valueAt(index);
+ oldBinder->unlinkToDeath(this);
+
+ mTorchClientMap.replaceValueAt(index, clientBinder);
+ }
+ clientBinder->linkToDeath(this);
+ } else if (index != NAME_NOT_FOUND) {
+ sp<IBinder> oldBinder = mTorchClientMap.valueAt(index);
+ oldBinder->unlinkToDeath(this);
+ }
+
+ // notify the listeners the change.
+ ICameraServiceListener::TorchStatus status = enabled ?
+ ICameraServiceListener::TORCH_STATUS_ON :
+ ICameraServiceListener::TORCH_STATUS_OFF;
+ onTorchStatusChangedLocked(cameraId, status);
+
+ return OK;
+}
+
status_t CameraService::connectFinishUnsafe(const sp<BasicClient>& client,
const sp<IBinder>& remoteCallback) {
status_t status = client->initialize(mModule);
@@ -971,6 +1098,9 @@
int facing = -1;
int deviceVersion = getDeviceVersion(cameraId, &facing);
+ // give flashlight a chance to close devices if necessary.
+ mFlashlight->prepareDeviceOpen();
+
switch(deviceVersion) {
case CAMERA_DEVICE_API_VERSION_1_0:
ALOGW("Camera using old HAL version: %d", deviceVersion);
@@ -1042,6 +1172,16 @@
}
}
+ /* Immediately signal current torch status to this listener only */
+ {
+ Mutex::Autolock al(mTorchStatusMutex);
+ for (size_t i = 0; i < mTorchStatusMap.size(); i++ ) {
+ listener->onTorchStatusChanged(mTorchStatusMap.valueAt(i),
+ mTorchStatusMap.keyAt(i));
+ }
+
+ }
+
return OK;
}
status_t CameraService::removeListener(
@@ -1633,14 +1773,11 @@
return NO_ERROR;
}
- result = String8::format("Camera module HAL API version: 0x%x\n",
- mModule->common.hal_api_version);
- result.appendFormat("Camera module API version: 0x%x\n",
- mModule->common.module_api_version);
- result.appendFormat("Camera module name: %s\n",
- mModule->common.name);
- result.appendFormat("Camera module author: %s\n",
- mModule->common.author);
+ const hw_module_t* common = mModule->getRawModule();
+ result = String8::format("Camera module HAL API version: 0x%x\n", common->hal_api_version);
+ result.appendFormat("Camera module API version: 0x%x\n", common->module_api_version);
+ result.appendFormat("Camera module name: %s\n", common->name);
+ result.appendFormat("Camera module author: %s\n", common->author);
result.appendFormat("Number of camera devices: %d\n\n", mNumberOfCameras);
sp<VendorTagDescriptor> desc = VendorTagDescriptor::getGlobalVendorTagDescriptor();
@@ -1660,7 +1797,7 @@
result = String8::format("Camera %d static information:\n", i);
camera_info info;
- status_t rc = mModule->get_camera_info(i, &info);
+ status_t rc = mModule->getCameraInfo(i, &info);
if (rc != OK) {
result.appendFormat(" Error reading static information!\n");
write(fd, result.string(), result.size());
@@ -1669,8 +1806,7 @@
info.facing == CAMERA_FACING_BACK ? "BACK" : "FRONT");
result.appendFormat(" Orientation: %d\n", info.orientation);
int deviceVersion;
- if (mModule->common.module_api_version <
- CAMERA_MODULE_API_VERSION_2_0) {
+ if (common->module_api_version < CAMERA_MODULE_API_VERSION_2_0) {
deviceVersion = CAMERA_DEVICE_API_VERSION_1_0;
} else {
deviceVersion = info.device_version;
@@ -1725,6 +1861,23 @@
return NO_ERROR;
}
+void CameraService::handleTorchClientBinderDied(const wp<IBinder> &who) {
+ Mutex::Autolock al(mTorchStatusMutex);
+ for (size_t i = 0; i < mTorchClientMap.size(); i++) {
+ if (mTorchClientMap[i] == who) {
+ // turn off the torch mode that was turned on by dead client
+ String16 cameraId = mTorchClientMap.keyAt(i);
+ mFlashlight->setTorchMode(cameraId, false);
+ mTorchClientMap.removeItemsAt(i);
+
+ // notify torch mode was turned off
+ onTorchStatusChangedLocked(cameraId,
+ ICameraServiceListener::TORCH_STATUS_OFF);
+ break;
+ }
+ }
+}
+
/*virtual*/void CameraService::binderDied(
const wp<IBinder> &who) {
@@ -1735,6 +1888,10 @@
ALOGV("java clients' binder died");
+ // check torch client
+ handleTorchClientBinderDied(who);
+
+ // check camera device client
sp<BasicClient> cameraClient = getClientByRemote(who);
if (cameraClient == 0) {
@@ -1828,4 +1985,27 @@
return mStatusList[cameraId];
}
+ICameraServiceListener::TorchStatus CameraService::getTorchStatusLocked(
+ const String16& cameraId) const {
+ ssize_t index = mTorchStatusMap.indexOfKey(cameraId);
+ if (index == NAME_NOT_FOUND) {
+ return ICameraServiceListener::TORCH_STATUS_NOT_AVAILABLE;
+ }
+
+ return mTorchStatusMap.valueAt(index);
+}
+
+status_t CameraService::setTorchStatusLocked(const String16& cameraId,
+ ICameraServiceListener::TorchStatus status) {
+ ssize_t index = mTorchStatusMap.indexOfKey(cameraId);
+ if (index == NAME_NOT_FOUND) {
+ return BAD_VALUE;
+ }
+ ICameraServiceListener::TorchStatus& item =
+ mTorchStatusMap.editValueAt(index);
+ item = status;
+
+ return OK;
+}
+
}; // namespace android
diff --git a/services/camera/libcameraservice/CameraService.h b/services/camera/libcameraservice/CameraService.h
index 126d8d9..84bcdb8 100644
--- a/services/camera/libcameraservice/CameraService.h
+++ b/services/camera/libcameraservice/CameraService.h
@@ -36,6 +36,10 @@
#include <camera/CameraParameters.h>
#include <camera/ICameraServiceListener.h>
+#include "CameraFlashlight.h"
+
+
+#include "common/CameraModule.h"
/* This needs to be increased if we can have more cameras */
#define MAX_CAMERAS 2
@@ -68,6 +72,9 @@
// HAL Callbacks
virtual void onDeviceStatusChanged(int cameraId,
int newStatus);
+ virtual void onTorchStatusChanged(const String16& cameraId,
+ ICameraServiceListener::TorchStatus
+ newStatus);
/////////////////////////////////////////////////////////////////////
// ICameraService
@@ -110,6 +117,9 @@
/*out*/
String16* parameters);
+ virtual status_t setTorchMode(const String16& cameraId, bool enabled,
+ const sp<IBinder>& clientBinder);
+
// OK = supports api of that version, -EOPNOTSUPP = does not support
virtual status_t supportsCameraApi(
int cameraId, int apiVersion);
@@ -153,7 +163,7 @@
class BasicClient : public virtual RefBase {
public:
- virtual status_t initialize(camera_module_t *module) = 0;
+ virtual status_t initialize(CameraModule *module) = 0;
virtual void disconnect();
// because we can't virtually inherit IInterface, which breaks
@@ -385,7 +395,7 @@
sp<MediaPlayer> mSoundPlayer[NUM_SOUNDS];
int mSoundRef; // reference count (release all MediaPlayer when 0)
- camera_module_t *mModule;
+ CameraModule* mModule;
Vector<sp<ICameraServiceListener> >
mListenerList;
@@ -406,6 +416,32 @@
int32_t cameraId,
const StatusVector *rejectSourceStates = NULL);
+ // flashlight control
+ sp<CameraFlashlight> mFlashlight;
+ // guard mTorchStatusMap and mTorchClientMap
+ Mutex mTorchStatusMutex;
+ // camera id -> torch status
+ KeyedVector<String16, ICameraServiceListener::TorchStatus> mTorchStatusMap;
+ // camera id -> torch client binder
+ // only store the last client that turns on each camera's torch mode
+ KeyedVector<String16, sp<IBinder> > mTorchClientMap;
+
+ // check and handle if torch client's process has died
+ void handleTorchClientBinderDied(const wp<IBinder> &who);
+
+ // handle torch mode status change and invoke callbacks. mTorchStatusMutex
+ // should be locked.
+ void onTorchStatusChangedLocked(const String16& cameraId,
+ ICameraServiceListener::TorchStatus newStatus);
+
+ // get a camera's torch status. mTorchStatusMutex should be locked.
+ ICameraServiceListener::TorchStatus getTorchStatusLocked(
+ const String16 &cameraId) const;
+
+ // set a camera's torch status. mTorchStatusMutex should be locked.
+ status_t setTorchStatusLocked(const String16 &cameraId,
+ ICameraServiceListener::TorchStatus status);
+
// IBinder::DeathRecipient implementation
virtual void binderDied(const wp<IBinder> &who);
diff --git a/services/camera/libcameraservice/api1/Camera2Client.cpp b/services/camera/libcameraservice/api1/Camera2Client.cpp
index 0ed5586..4ac5166 100644
--- a/services/camera/libcameraservice/api1/Camera2Client.cpp
+++ b/services/camera/libcameraservice/api1/Camera2Client.cpp
@@ -67,7 +67,7 @@
mLegacyMode = legacyMode;
}
-status_t Camera2Client::initialize(camera_module_t *module)
+status_t Camera2Client::initialize(CameraModule *module)
{
ATRACE_CALL();
ALOGV("%s: Initializing client for camera %d", __FUNCTION__, mCameraId);
diff --git a/services/camera/libcameraservice/api1/Camera2Client.h b/services/camera/libcameraservice/api1/Camera2Client.h
index d68bb29..5a8241f 100644
--- a/services/camera/libcameraservice/api1/Camera2Client.h
+++ b/services/camera/libcameraservice/api1/Camera2Client.h
@@ -94,7 +94,7 @@
virtual ~Camera2Client();
- status_t initialize(camera_module_t *module);
+ status_t initialize(CameraModule *module);
virtual status_t dump(int fd, const Vector<String16>& args);
diff --git a/services/camera/libcameraservice/api1/CameraClient.cpp b/services/camera/libcameraservice/api1/CameraClient.cpp
index bbb2fe0..6bea3b6 100644
--- a/services/camera/libcameraservice/api1/CameraClient.cpp
+++ b/services/camera/libcameraservice/api1/CameraClient.cpp
@@ -59,7 +59,7 @@
LOG1("CameraClient::CameraClient X (pid %d, id %d)", callingPid, cameraId);
}
-status_t CameraClient::initialize(camera_module_t *module) {
+status_t CameraClient::initialize(CameraModule *module) {
int callingPid = getCallingPid();
status_t res;
@@ -75,7 +75,7 @@
snprintf(camera_device_name, sizeof(camera_device_name), "%d", mCameraId);
mHardware = new CameraHardwareInterface(camera_device_name);
- res = mHardware->initialize(&module->common);
+ res = mHardware->initialize(module);
if (res != OK) {
ALOGE("%s: Camera %d: unable to initialize device: %s (%d)",
__FUNCTION__, mCameraId, strerror(-res), res);
diff --git a/services/camera/libcameraservice/api1/CameraClient.h b/services/camera/libcameraservice/api1/CameraClient.h
index 63a9d0f..95616b2 100644
--- a/services/camera/libcameraservice/api1/CameraClient.h
+++ b/services/camera/libcameraservice/api1/CameraClient.h
@@ -68,7 +68,7 @@
bool legacyMode = false);
~CameraClient();
- status_t initialize(camera_module_t *module);
+ status_t initialize(CameraModule *module);
status_t dump(int fd, const Vector<String16>& args);
diff --git a/services/camera/libcameraservice/api2/CameraDeviceClient.cpp b/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
index 6a1ee44..acc092c 100644
--- a/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
+++ b/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
@@ -71,7 +71,7 @@
ALOGI("CameraDeviceClient %d: Opened", cameraId);
}
-status_t CameraDeviceClient::initialize(camera_module_t *module)
+status_t CameraDeviceClient::initialize(CameraModule *module)
{
ATRACE_CALL();
status_t res;
diff --git a/services/camera/libcameraservice/api2/CameraDeviceClient.h b/services/camera/libcameraservice/api2/CameraDeviceClient.h
index 84e46b7..e687175 100644
--- a/services/camera/libcameraservice/api2/CameraDeviceClient.h
+++ b/services/camera/libcameraservice/api2/CameraDeviceClient.h
@@ -119,7 +119,7 @@
int servicePid);
virtual ~CameraDeviceClient();
- virtual status_t initialize(camera_module_t *module);
+ virtual status_t initialize(CameraModule *module);
virtual status_t dump(int fd, const Vector<String16>& args);
diff --git a/services/camera/libcameraservice/api_pro/ProCamera2Client.cpp b/services/camera/libcameraservice/api_pro/ProCamera2Client.cpp
index 59e5083..30a89c2 100644
--- a/services/camera/libcameraservice/api_pro/ProCamera2Client.cpp
+++ b/services/camera/libcameraservice/api_pro/ProCamera2Client.cpp
@@ -50,7 +50,7 @@
mExclusiveLock = false;
}
-status_t ProCamera2Client::initialize(camera_module_t *module)
+status_t ProCamera2Client::initialize(CameraModule *module)
{
ATRACE_CALL();
status_t res;
diff --git a/services/camera/libcameraservice/api_pro/ProCamera2Client.h b/services/camera/libcameraservice/api_pro/ProCamera2Client.h
index 9d83122..7f5f6ac 100644
--- a/services/camera/libcameraservice/api_pro/ProCamera2Client.h
+++ b/services/camera/libcameraservice/api_pro/ProCamera2Client.h
@@ -85,7 +85,7 @@
int servicePid);
virtual ~ProCamera2Client();
- virtual status_t initialize(camera_module_t *module);
+ virtual status_t initialize(CameraModule *module);
virtual status_t dump(int fd, const Vector<String16>& args);
diff --git a/services/camera/libcameraservice/common/Camera2ClientBase.cpp b/services/camera/libcameraservice/common/Camera2ClientBase.cpp
index 453c8bd..0415d67 100644
--- a/services/camera/libcameraservice/common/Camera2ClientBase.cpp
+++ b/services/camera/libcameraservice/common/Camera2ClientBase.cpp
@@ -78,7 +78,7 @@
}
template <typename TClientBase>
-status_t Camera2ClientBase<TClientBase>::initialize(camera_module_t *module) {
+status_t Camera2ClientBase<TClientBase>::initialize(CameraModule *module) {
ATRACE_CALL();
ALOGV("%s: Initializing client for camera %d", __FUNCTION__,
TClientBase::mCameraId);
diff --git a/services/camera/libcameraservice/common/Camera2ClientBase.h b/services/camera/libcameraservice/common/Camera2ClientBase.h
index e09c1b5..eb21d55 100644
--- a/services/camera/libcameraservice/common/Camera2ClientBase.h
+++ b/services/camera/libcameraservice/common/Camera2ClientBase.h
@@ -18,6 +18,7 @@
#define ANDROID_SERVERS_CAMERA_CAMERA2CLIENT_BASE_H
#include "common/CameraDeviceBase.h"
+#include "common/CameraModule.h"
#include "camera/CaptureResult.h"
namespace android {
@@ -55,7 +56,7 @@
int servicePid);
virtual ~Camera2ClientBase();
- virtual status_t initialize(camera_module_t *module);
+ virtual status_t initialize(CameraModule *module);
virtual status_t dump(int fd, const Vector<String16>& args);
/**
diff --git a/services/camera/libcameraservice/common/CameraDeviceBase.h b/services/camera/libcameraservice/common/CameraDeviceBase.h
index d26e20c..06615f6 100644
--- a/services/camera/libcameraservice/common/CameraDeviceBase.h
+++ b/services/camera/libcameraservice/common/CameraDeviceBase.h
@@ -29,6 +29,7 @@
#include "hardware/camera3.h"
#include "camera/CameraMetadata.h"
#include "camera/CaptureResult.h"
+#include "common/CameraModule.h"
namespace android {
@@ -45,7 +46,7 @@
*/
virtual int getId() const = 0;
- virtual status_t initialize(camera_module_t *module) = 0;
+ virtual status_t initialize(CameraModule *module) = 0;
virtual status_t disconnect() = 0;
virtual status_t dump(int fd, const Vector<String16> &args) = 0;
diff --git a/services/camera/libcameraservice/common/CameraModule.cpp b/services/camera/libcameraservice/common/CameraModule.cpp
new file mode 100644
index 0000000..bbf47e8
--- /dev/null
+++ b/services/camera/libcameraservice/common/CameraModule.cpp
@@ -0,0 +1,130 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "CameraModule"
+//#define LOG_NDEBUG 0
+
+#include "CameraModule.h"
+
+namespace android {
+
+void CameraModule::deriveCameraCharacteristicsKeys(
+ uint32_t deviceVersion, CameraMetadata &chars) {
+ // HAL1 devices should not reach here
+ if (deviceVersion < CAMERA_DEVICE_API_VERSION_2_0) {
+ ALOGV("%s: Cannot derive keys for HAL version < 2.0");
+ return;
+ }
+
+ // Keys added in HAL3.3
+ if (deviceVersion < CAMERA_DEVICE_API_VERSION_3_3) {
+ Vector<uint8_t> controlModes;
+ uint8_t data = ANDROID_CONTROL_AE_LOCK_AVAILABLE_TRUE;
+ chars.update(ANDROID_CONTROL_AE_LOCK_AVAILABLE, &data, /*count*/1);
+ data = ANDROID_CONTROL_AWB_LOCK_AVAILABLE_TRUE;
+ chars.update(ANDROID_CONTROL_AWB_LOCK_AVAILABLE, &data, /*count*/1);
+ controlModes.push(ANDROID_CONTROL_MODE_OFF);
+ controlModes.push(ANDROID_CONTROL_MODE_AUTO);
+ camera_metadata_entry entry = chars.find(ANDROID_CONTROL_AVAILABLE_SCENE_MODES);
+ if (entry.count > 1 || entry.data.u8[0] != ANDROID_CONTROL_SCENE_MODE_DISABLED) {
+ controlModes.push(ANDROID_CONTROL_MODE_USE_SCENE_MODE);
+ }
+ chars.update(ANDROID_CONTROL_AVAILABLE_MODES, controlModes);
+ }
+ return;
+}
+
+CameraModule::CameraModule(camera_module_t *module) {
+ if (module == NULL) {
+ ALOGE("%s: camera hardware module must not be null", __FUNCTION__);
+ assert(0);
+ }
+
+ mModule = module;
+ for (int i = 0; i < MAX_CAMERAS_PER_MODULE; i++) {
+ mCameraInfoCached[i] = false;
+ }
+}
+
+int CameraModule::getCameraInfo(int cameraId, struct camera_info *info) {
+ Mutex::Autolock lock(mCameraInfoLock);
+ if (cameraId < 0 || cameraId >= MAX_CAMERAS_PER_MODULE) {
+ ALOGE("%s: Invalid camera ID %d", __FUNCTION__, cameraId);
+ return -EINVAL;
+ }
+
+ camera_info &wrappedInfo = mCameraInfo[cameraId];
+ if (!mCameraInfoCached[cameraId]) {
+ camera_info rawInfo;
+ int ret = mModule->get_camera_info(cameraId, &rawInfo);
+ if (ret != 0) {
+ return ret;
+ }
+ CameraMetadata &m = mCameraCharacteristics[cameraId];
+ m = rawInfo.static_camera_characteristics;
+ int deviceVersion;
+ int apiVersion = mModule->common.module_api_version;
+ if (apiVersion >= CAMERA_MODULE_API_VERSION_2_0) {
+ deviceVersion = rawInfo.device_version;
+ } else {
+ deviceVersion = CAMERA_DEVICE_API_VERSION_1_0;
+ }
+ deriveCameraCharacteristicsKeys(deviceVersion, m);
+ wrappedInfo = rawInfo;
+ wrappedInfo.static_camera_characteristics = m.getAndLock();
+ mCameraInfoCached[cameraId] = true;
+ }
+ *info = wrappedInfo;
+ return 0;
+}
+
+int CameraModule::open(const char* id, struct hw_device_t** device) {
+ return mModule->common.methods->open(&mModule->common, id, device);
+}
+
+int CameraModule::openLegacy(
+ const char* id, uint32_t halVersion, struct hw_device_t** device) {
+ return mModule->open_legacy(&mModule->common, id, halVersion, device);
+}
+
+const hw_module_t* CameraModule::getRawModule() {
+ return &mModule->common;
+}
+
+int CameraModule::getNumberOfCameras() {
+ return mModule->get_number_of_cameras();
+}
+
+int CameraModule::setCallbacks(const camera_module_callbacks_t *callbacks) {
+ return mModule->set_callbacks(callbacks);
+}
+
+bool CameraModule::isVendorTagDefined() {
+ return mModule->get_vendor_tag_ops != NULL;
+}
+
+void CameraModule::getVendorTagOps(vendor_tag_ops_t* ops) {
+ if (mModule->get_vendor_tag_ops) {
+ mModule->get_vendor_tag_ops(ops);
+ }
+}
+
+int CameraModule::setTorchMode(const char* camera_id, bool enable) {
+ return mModule->set_torch_mode(camera_id, enable);
+}
+
+}; // namespace android
+
diff --git a/services/camera/libcameraservice/common/CameraModule.h b/services/camera/libcameraservice/common/CameraModule.h
new file mode 100644
index 0000000..31b9ae2
--- /dev/null
+++ b/services/camera/libcameraservice/common/CameraModule.h
@@ -0,0 +1,63 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_SERVERS_CAMERA_CAMERAMODULE_H
+#define ANDROID_SERVERS_CAMERA_CAMERAMODULE_H
+
+#include <hardware/camera.h>
+#include <camera/CameraMetadata.h>
+#include <utils/Mutex.h>
+
+/* This needs to be increased if we can have more cameras */
+#define MAX_CAMERAS_PER_MODULE 2
+
+
+namespace android {
+/**
+ * A wrapper class for HAL camera module.
+ *
+ * This class wraps camera_module_t returned from HAL to provide a wrapped
+ * get_camera_info implementation which CameraService generates some
+ * camera characteristics keys defined in newer HAL version on an older HAL.
+ */
+class CameraModule {
+public:
+ CameraModule(camera_module_t *module);
+
+ const hw_module_t* getRawModule();
+ int getCameraInfo(int cameraId, struct camera_info *info);
+ int getNumberOfCameras(void);
+ int open(const char* id, struct hw_device_t** device);
+ int openLegacy(const char* id, uint32_t halVersion, struct hw_device_t** device);
+ int setCallbacks(const camera_module_callbacks_t *callbacks);
+ bool isVendorTagDefined();
+ void getVendorTagOps(vendor_tag_ops_t* ops);
+ int setTorchMode(const char* camera_id, bool enable);
+
+private:
+ // Derive camera characteristics keys defined after HAL device version
+ static void deriveCameraCharacteristicsKeys(uint32_t deviceVersion, CameraMetadata &chars);
+ camera_module_t *mModule;
+ CameraMetadata mCameraCharacteristics[MAX_CAMERAS_PER_MODULE];
+ camera_info mCameraInfo[MAX_CAMERAS_PER_MODULE];
+ bool mCameraInfoCached[MAX_CAMERAS_PER_MODULE];
+ Mutex mCameraInfoLock;
+};
+
+} // namespace android
+
+#endif
+
diff --git a/services/camera/libcameraservice/device1/CameraHardwareInterface.h b/services/camera/libcameraservice/device1/CameraHardwareInterface.h
index 1935c2b..9e1cdc9 100644
--- a/services/camera/libcameraservice/device1/CameraHardwareInterface.h
+++ b/services/camera/libcameraservice/device1/CameraHardwareInterface.h
@@ -89,24 +89,23 @@
}
}
- status_t initialize(hw_module_t *module)
+ status_t initialize(CameraModule *module)
{
ALOGI("Opening camera %s", mName.string());
- camera_module_t *cameraModule = reinterpret_cast<camera_module_t *>(module);
camera_info info;
- status_t res = cameraModule->get_camera_info(atoi(mName.string()), &info);
+ status_t res = module->getCameraInfo(atoi(mName.string()), &info);
if (res != OK) return res;
int rc = OK;
- if (module->module_api_version >= CAMERA_MODULE_API_VERSION_2_3 &&
+ if (module->getRawModule()->module_api_version >= CAMERA_MODULE_API_VERSION_2_3 &&
info.device_version > CAMERA_DEVICE_API_VERSION_1_0) {
// Open higher version camera device as HAL1.0 device.
- rc = cameraModule->open_legacy(module, mName.string(),
- CAMERA_DEVICE_API_VERSION_1_0,
- (hw_device_t **)&mDevice);
+ rc = module->openLegacy(mName.string(),
+ CAMERA_DEVICE_API_VERSION_1_0,
+ (hw_device_t **)&mDevice);
} else {
- rc = CameraService::filterOpenErrorCode(module->methods->open(
- module, mName.string(), (hw_device_t **)&mDevice));
+ rc = CameraService::filterOpenErrorCode(module->open(
+ mName.string(), (hw_device_t **)&mDevice));
}
if (rc != OK) {
ALOGE("Could not open camera %s: %d", mName.string(), rc);
diff --git a/services/camera/libcameraservice/device2/Camera2Device.cpp b/services/camera/libcameraservice/device2/Camera2Device.cpp
index d1158d6..be66c4d 100644
--- a/services/camera/libcameraservice/device2/Camera2Device.cpp
+++ b/services/camera/libcameraservice/device2/Camera2Device.cpp
@@ -53,7 +53,7 @@
return mId;
}
-status_t Camera2Device::initialize(camera_module_t *module)
+status_t Camera2Device::initialize(CameraModule *module)
{
ATRACE_CALL();
ALOGV("%s: Initializing device for camera %d", __FUNCTION__, mId);
@@ -68,8 +68,8 @@
camera2_device_t *device;
- res = CameraService::filterOpenErrorCode(module->common.methods->open(
- &module->common, name, reinterpret_cast<hw_device_t**>(&device)));
+ res = CameraService::filterOpenErrorCode(module->open(
+ name, reinterpret_cast<hw_device_t**>(&device)));
if (res != OK) {
ALOGE("%s: Could not open camera %d: %s (%d)", __FUNCTION__,
@@ -87,7 +87,7 @@
}
camera_info info;
- res = module->get_camera_info(mId, &info);
+ res = module->getCameraInfo(mId, &info);
if (res != OK ) return res;
if (info.device_version != device->common.version) {
diff --git a/services/camera/libcameraservice/device2/Camera2Device.h b/services/camera/libcameraservice/device2/Camera2Device.h
index 4def8ae..1cc5482 100644
--- a/services/camera/libcameraservice/device2/Camera2Device.h
+++ b/services/camera/libcameraservice/device2/Camera2Device.h
@@ -43,7 +43,7 @@
* CameraDevice interface
*/
virtual int getId() const;
- virtual status_t initialize(camera_module_t *module);
+ virtual status_t initialize(CameraModule *module);
virtual status_t disconnect();
virtual status_t dump(int fd, const Vector<String16>& args);
virtual const CameraMetadata& info() const;
diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp
index 53e6fa9..9a4e5ac 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Device.cpp
@@ -86,7 +86,7 @@
* CameraDeviceBase interface
*/
-status_t Camera3Device::initialize(camera_module_t *module)
+status_t Camera3Device::initialize(CameraModule *module)
{
ATRACE_CALL();
Mutex::Autolock il(mInterfaceLock);
@@ -106,9 +106,8 @@
camera3_device_t *device;
ATRACE_BEGIN("camera3->open");
- res = CameraService::filterOpenErrorCode(module->common.methods->open(
- &module->common, deviceName.string(),
- reinterpret_cast<hw_device_t**>(&device)));
+ res = CameraService::filterOpenErrorCode(module->open(
+ deviceName.string(), reinterpret_cast<hw_device_t**>(&device)));
ATRACE_END();
if (res != OK) {
@@ -127,7 +126,7 @@
}
camera_info info;
- res = CameraService::filterGetInfoErrorCode(module->get_camera_info(
+ res = CameraService::filterGetInfoErrorCode(module->getCameraInfo(
mId, &info));
if (res != OK) return res;
diff --git a/services/camera/libcameraservice/device3/Camera3Device.h b/services/camera/libcameraservice/device3/Camera3Device.h
index ec8dc10..de10cfe 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.h
+++ b/services/camera/libcameraservice/device3/Camera3Device.h
@@ -73,7 +73,7 @@
virtual int getId() const;
// Transitions to idle state on success.
- virtual status_t initialize(camera_module_t *module);
+ virtual status_t initialize(CameraModule *module);
virtual status_t disconnect();
virtual status_t dump(int fd, const Vector<String16> &args);
virtual const CameraMetadata& info() const;
diff --git a/services/medialog/Android.mk b/services/medialog/Android.mk
index 95f2fef..03438bf 100644
--- a/services/medialog/Android.mk
+++ b/services/medialog/Android.mk
@@ -10,4 +10,6 @@
LOCAL_32_BIT_ONLY := true
+LOCAL_C_INCLUDES := $(call include-path-for, audio-utils)
+
include $(BUILD_SHARED_LIBRARY)
diff --git a/tools/resampler_tools/Android.mk b/tools/resampler_tools/Android.mk
index e8cbe39..b58e4cd 100644
--- a/tools/resampler_tools/Android.mk
+++ b/tools/resampler_tools/Android.mk
@@ -1,6 +1,6 @@
# Copyright 2005 The Android Open Source Project
#
-# Android.mk for resampler_tools
+# Android.mk for resampler_tools
#