blob: d9b75da46f07a85f9e91cfdec4a7adab0fe52ccd [file] [log] [blame]
/*
* Copyright (C) 2017 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
#include <utils/Log.h>
#include <algorithm>
#include <audio_utils/format.h>
#include <aaudio/AAudio.h>
#include <media/MediaMetricsItem.h>
#include "client/AudioStreamInternalCapture.h"
#include "utility/AudioClock.h"
#undef ATRACE_TAG
#define ATRACE_TAG ATRACE_TAG_AUDIO
#include <utils/Trace.h>
// We do this after the #includes because if a header uses ALOG.
// it would fail on the reference to mInService.
#undef LOG_TAG
// This file is used in both client and server processes.
// This is needed to make sense of the logs more easily.
#define LOG_TAG (mInService ? "AudioStreamInternalCapture_Service" \
: "AudioStreamInternalCapture_Client")
using android::WrappingBuffer;
using namespace aaudio;
AudioStreamInternalCapture::AudioStreamInternalCapture(AAudioServiceInterface &serviceInterface,
bool inService)
: AudioStreamInternal(serviceInterface, inService) {
}
aaudio_result_t AudioStreamInternalCapture::open(const AudioStreamBuilder &builder) {
aaudio_result_t result = AudioStreamInternal::open(builder);
if (result == AAUDIO_OK) {
result = mFlowGraph.configure(getDeviceFormat(),
getDeviceSamplesPerFrame(),
getDeviceSampleRate(),
getFormat(),
getSamplesPerFrame(),
getSampleRate(),
getRequireMonoBlend(),
false /* useVolumeRamps */,
getAudioBalance(),
aaudio::resampler::MultiChannelResampler::Quality::Medium);
if (result != AAUDIO_OK) {
safeReleaseClose();
}
}
return result;
}
void AudioStreamInternalCapture::advanceClientToMatchServerPosition(int32_t serverMargin) {
int64_t readCounter = mAudioEndpoint->getDataReadCounter();
int64_t writeCounter = mAudioEndpoint->getDataWriteCounter() + serverMargin;
// Bump offset so caller does not see the retrograde motion in getFramesRead().
int64_t offset = readCounter - writeCounter;
mFramesOffsetFromService += offset;
ALOGD("advanceClientToMatchServerPosition() readN = %lld, writeN = %lld, offset = %lld",
(long long)readCounter, (long long)writeCounter, (long long)mFramesOffsetFromService);
// Force readCounter to match writeCounter.
// This is because we cannot change the write counter in the hardware.
mAudioEndpoint->setDataReadCounter(writeCounter);
}
// Write the data, block if needed and timeoutMillis > 0
aaudio_result_t AudioStreamInternalCapture::read(void *buffer, int32_t numFrames,
int64_t timeoutNanoseconds)
{
return processData(buffer, numFrames, timeoutNanoseconds);
}
// Read as much data as we can without blocking.
aaudio_result_t AudioStreamInternalCapture::processDataNow(void *buffer, int32_t numFrames,
int64_t currentNanoTime, int64_t *wakeTimePtr) {
aaudio_result_t result = processCommands();
if (result != AAUDIO_OK) {
return result;
}
const char *traceName = "aaRdNow";
ATRACE_BEGIN(traceName);
if (mClockModel.isStarting()) {
// Still haven't got any timestamps from server.
// Keep waiting until we get some valid timestamps then start writing to the
// current buffer position.
ALOGD("processDataNow() wait for valid timestamps");
// Sleep very briefly and hope we get a timestamp soon.
*wakeTimePtr = currentNanoTime + (2000 * AAUDIO_NANOS_PER_MICROSECOND);
ATRACE_END();
return 0;
}
// If we have gotten this far then we have at least one timestamp from server.
if (mAudioEndpoint->isFreeRunning()) {
//ALOGD("AudioStreamInternalCapture::processDataNow() - update remote counter");
// Update data queue based on the timing model.
// Jitter in the DSP can cause late writes to the FIFO.
// This might be caused by resampling.
// We want to read the FIFO after the latest possible time
// that the DSP could have written the data.
int64_t estimatedRemoteCounter = mClockModel.convertLatestTimeToPosition(currentNanoTime);
// TODO refactor, maybe use setRemoteCounter()
mAudioEndpoint->setDataWriteCounter(estimatedRemoteCounter);
}
// This code assumes that we have already received valid timestamps.
if (mNeedCatchUp.isRequested()) {
// Catch an MMAP pointer that is already advancing.
// This will avoid initial underruns caused by a slow cold start.
advanceClientToMatchServerPosition(0 /*serverMargin*/);
mNeedCatchUp.acknowledge();
}
// If the capture buffer is full beyond capacity then consider it an overrun.
// For shared streams, the xRunCount is passed up from the service.
if (mAudioEndpoint->isFreeRunning()
&& mAudioEndpoint->getFullFramesAvailable() > mAudioEndpoint->getBufferCapacityInFrames()) {
mXRunCount++;
if (ATRACE_ENABLED()) {
ATRACE_INT("aaOverRuns", mXRunCount);
}
}
// Read some data from the buffer.
//ALOGD("AudioStreamInternalCapture::processDataNow() - readNowWithConversion(%d)", numFrames);
int32_t framesProcessed = readNowWithConversion(buffer, numFrames);
//ALOGD("AudioStreamInternalCapture::processDataNow() - tried to read %d frames, read %d",
// numFrames, framesProcessed);
if (ATRACE_ENABLED()) {
ATRACE_INT("aaRead", framesProcessed);
}
// Calculate an ideal time to wake up.
if (wakeTimePtr != nullptr && framesProcessed >= 0) {
// By default wake up a few milliseconds from now. // TODO review
int64_t wakeTime = currentNanoTime + (1 * AAUDIO_NANOS_PER_MILLISECOND);
aaudio_stream_state_t state = getState();
//ALOGD("AudioStreamInternalCapture::processDataNow() - wakeTime based on %s",
// AAudio_convertStreamStateToText(state));
switch (state) {
case AAUDIO_STREAM_STATE_OPEN:
case AAUDIO_STREAM_STATE_STARTING:
break;
case AAUDIO_STREAM_STATE_STARTED:
{
// When do we expect the next write burst to occur?
// Calculate frame position based off of the readCounter because
// the writeCounter might have just advanced in the background,
// causing us to sleep until a later burst.
const int64_t nextPosition = mAudioEndpoint->getDataReadCounter() +
getDeviceFramesPerBurst();
wakeTime = mClockModel.convertPositionToLatestTime(nextPosition);
}
break;
default:
break;
}
*wakeTimePtr = wakeTime;
}
ATRACE_END();
return framesProcessed;
}
aaudio_result_t AudioStreamInternalCapture::readNowWithConversion(void *buffer,
int32_t numFrames) {
WrappingBuffer wrappingBuffer;
uint8_t *byteBuffer = (uint8_t *) buffer;
int32_t framesLeftInByteBuffer = numFrames;
if (framesLeftInByteBuffer > 0) {
// Pull data from the flowgraph in case there is residual data.
const int32_t framesActuallyWrittenToByteBuffer = mFlowGraph.pull(
(void *)byteBuffer,
framesLeftInByteBuffer);
const int32_t numBytesActuallyWrittenToByteBuffer =
framesActuallyWrittenToByteBuffer * getBytesPerFrame();
byteBuffer += numBytesActuallyWrittenToByteBuffer;
framesLeftInByteBuffer -= framesActuallyWrittenToByteBuffer;
}
mAudioEndpoint->getFullFramesAvailable(&wrappingBuffer);
// Write data in one or two parts.
int partIndex = 0;
int framesReadFromAudioEndpoint = 0;
while (framesLeftInByteBuffer > 0 && partIndex < WrappingBuffer::SIZE) {
const int32_t totalFramesInWrappingBuffer = wrappingBuffer.numFrames[partIndex];
int32_t framesAvailableInWrappingBuffer = totalFramesInWrappingBuffer;
uint8_t *currentWrappingBuffer = (uint8_t *) wrappingBuffer.data[partIndex];
if (framesAvailableInWrappingBuffer <= 0) break;
// Put data from the wrapping buffer into the flowgraph 8 frames at a time.
// Continuously pull as much data as possible from the flowgraph into the byte buffer.
// The return value of mFlowGraph.process is the number of frames actually pulled.
while (framesAvailableInWrappingBuffer > 0 && framesLeftInByteBuffer > 0) {
const int32_t framesToReadFromWrappingBuffer = std::min(flowgraph::kDefaultBufferSize,
framesAvailableInWrappingBuffer);
const int32_t numBytesToReadFromWrappingBuffer = getBytesPerDeviceFrame() *
framesToReadFromWrappingBuffer;
// If framesActuallyWrittenToByteBuffer < framesLeftInByteBuffer, it is guaranteed
// that all the data is pulled. If there is no more space in the byteBuffer, the
// remaining data will be pulled in the following readNowWithConversion().
const int32_t framesActuallyWrittenToByteBuffer = mFlowGraph.process(
(void *)currentWrappingBuffer,
framesToReadFromWrappingBuffer,
(void *)byteBuffer,
framesLeftInByteBuffer);
const int32_t numBytesActuallyWrittenToByteBuffer =
framesActuallyWrittenToByteBuffer * getBytesPerFrame();
byteBuffer += numBytesActuallyWrittenToByteBuffer;
framesLeftInByteBuffer -= framesActuallyWrittenToByteBuffer;
currentWrappingBuffer += numBytesToReadFromWrappingBuffer;
framesAvailableInWrappingBuffer -= framesToReadFromWrappingBuffer;
//ALOGD("%s() numBytesActuallyWrittenToByteBuffer %d, framesLeftInByteBuffer %d"
// "framesAvailableInWrappingBuffer %d, framesReadFromAudioEndpoint %d"
// , __func__, numBytesActuallyWrittenToByteBuffer, framesLeftInByteBuffer,
// framesAvailableInWrappingBuffer, framesReadFromAudioEndpoint);
}
framesReadFromAudioEndpoint += totalFramesInWrappingBuffer -
framesAvailableInWrappingBuffer;
partIndex++;
}
// The audio endpoint should reference the number of frames written to the wrapping buffer.
mAudioEndpoint->advanceReadIndex(framesReadFromAudioEndpoint);
// The internal code should use the number of frames read from the app.
return numFrames - framesLeftInByteBuffer;
}
int64_t AudioStreamInternalCapture::getFramesWritten() {
if (mAudioEndpoint) {
const int64_t framesWrittenHardware = isClockModelInControl()
? mClockModel.convertTimeToPosition(AudioClock::getNanoseconds())
: mAudioEndpoint->getDataWriteCounter();
// Add service offset and prevent retrograde motion.
mLastFramesWritten = std::max(mLastFramesWritten,
framesWrittenHardware + mFramesOffsetFromService);
}
return mLastFramesWritten;
}
int64_t AudioStreamInternalCapture::getFramesRead() {
if (mAudioEndpoint) {
mLastFramesRead = mAudioEndpoint->getDataReadCounter() + mFramesOffsetFromService;
}
return mLastFramesRead;
}
// Read data from the stream and pass it to the callback for processing.
void *AudioStreamInternalCapture::callbackLoop() {
aaudio_result_t result = AAUDIO_OK;
aaudio_data_callback_result_t callbackResult = AAUDIO_CALLBACK_RESULT_CONTINUE;
if (!isDataCallbackSet()) return nullptr;
// result might be a frame count
while (mCallbackEnabled.load() && isActive() && (result >= 0)) {
// Read audio data from stream.
int64_t timeoutNanos = calculateReasonableTimeout(mCallbackFrames);
// This is a BLOCKING READ!
result = read(mCallbackBuffer.get(), mCallbackFrames, timeoutNanos);
if ((result != mCallbackFrames)) {
ALOGE("callbackLoop: read() returned %d", result);
if (result >= 0) {
// Only read some of the frames requested. Must have timed out.
result = AAUDIO_ERROR_TIMEOUT;
}
maybeCallErrorCallback(result);
break;
}
// Call application using the AAudio callback interface.
callbackResult = maybeCallDataCallback(mCallbackBuffer.get(), mCallbackFrames);
if (callbackResult == AAUDIO_CALLBACK_RESULT_STOP) {
ALOGD("%s(): callback returned AAUDIO_CALLBACK_RESULT_STOP", __func__);
result = systemStopInternal();
break;
}
}
ALOGD("callbackLoop() exiting, result = %d, isActive() = %d",
result, (int) isActive());
return nullptr;
}