Gitiles
Code Review Sign In
LeafOS / LeafOS-Project / android_frameworks_av / 442a2ee35a2c3922237540aed8c16481e03dfb0d / . / services / audioflinger / AudioFlinger.cpp
blob: c6d4fe38226fbeb61464eacdf319ecf9f4b742ff [file] [log] [blame]
/*
**
** Copyright 2007, 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 "AudioFlinger"
//#define LOG_NDEBUG 0
#include "Configuration.h"
#include <dirent.h>
#include <math.h>
#include <signal.h>
#include <sys/time.h>
#include <sys/resource.h>
#include <binder/IPCThreadState.h>
#include <binder/IServiceManager.h>
#include <utils/Log.h>
#include <utils/Trace.h>
#include <binder/Parcel.h>
#include <memunreachable/memunreachable.h>
#include <utils/String16.h>
#include <utils/threads.h>
#include <utils/Atomic.h>
#include <cutils/bitops.h>
#include <cutils/properties.h>
#include <system/audio.h>
#include <hardware/audio.h>
#include "AudioMixer.h"
#include "AudioFlinger.h"
#include "ServiceUtilities.h"
#include <media/AudioResamplerPublic.h>
#include <media/EffectsFactoryApi.h>
#include <audio_effects/effect_visualizer.h>
#include <audio_effects/effect_ns.h>
#include <audio_effects/effect_aec.h>
#include <audio_utils/primitives.h>
#include <powermanager/PowerManager.h>
#include <media/IMediaLogService.h>
#include <media/MemoryLeakTrackUtil.h>
#include <media/nbaio/Pipe.h>
#include <media/nbaio/PipeReader.h>
#include <media/AudioParameter.h>
#include <mediautils/BatteryNotifier.h>
#include <private/android_filesystem_config.h>
// ----------------------------------------------------------------------------
// Note: the following macro is used for extremely verbose logging message. In
// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to
// 0; but one side effect of this is to turn all LOGV's as well. Some messages
// are so verbose that we want to suppress them even when we have ALOG_ASSERT
// turned on. Do not uncomment the #def below unless you really know what you
// are doing and want to see all of the extremely verbose messages.
//#define VERY_VERY_VERBOSE_LOGGING
#ifdef VERY_VERY_VERBOSE_LOGGING
#define ALOGVV ALOGV
#else
#define ALOGVV(a...) do { } while(0)
#endif
namespace android {
static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n";
static const char kHardwareLockedString[] = "Hardware lock is taken\n";
static const char kClientLockedString[] = "Client lock is taken\n";
nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
uint32_t AudioFlinger::mScreenState;
#ifdef TEE_SINK
bool AudioFlinger::mTeeSinkInputEnabled = false;
bool AudioFlinger::mTeeSinkOutputEnabled = false;
bool AudioFlinger::mTeeSinkTrackEnabled = false;
size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault;
size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault;
size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault;
#endif
// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off
// we define a minimum time during which a global effect is considered enabled.
static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200);
// ----------------------------------------------------------------------------
const char *formatToString(audio_format_t format) {
switch (audio_get_main_format(format)) {
case AUDIO_FORMAT_PCM:
switch (format) {
case AUDIO_FORMAT_PCM_16_BIT: return "pcm16";
case AUDIO_FORMAT_PCM_8_BIT: return "pcm8";
case AUDIO_FORMAT_PCM_32_BIT: return "pcm32";
case AUDIO_FORMAT_PCM_8_24_BIT: return "pcm8.24";
case AUDIO_FORMAT_PCM_FLOAT: return "pcmfloat";
case AUDIO_FORMAT_PCM_24_BIT_PACKED: return "pcm24";
default:
break;
}
break;
case AUDIO_FORMAT_MP3: return "mp3";
case AUDIO_FORMAT_AMR_NB: return "amr-nb";
case AUDIO_FORMAT_AMR_WB: return "amr-wb";
case AUDIO_FORMAT_AAC: return "aac";
case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1";
case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2";
case AUDIO_FORMAT_VORBIS: return "vorbis";
case AUDIO_FORMAT_OPUS: return "opus";
case AUDIO_FORMAT_AC3: return "ac-3";
case AUDIO_FORMAT_E_AC3: return "e-ac-3";
case AUDIO_FORMAT_IEC61937: return "iec61937";
default:
break;
}
return "unknown";
}
static int load_audio_interface(const char *if_name, audio_hw_device_t **dev)
{
const hw_module_t *mod;
int rc;
rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod);
ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__,
AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
if (rc) {
goto out;
}
rc = audio_hw_device_open(mod, dev);
ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__,
AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
if (rc) {
goto out;
}
if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) {
ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version);
rc = BAD_VALUE;
goto out;
}
return 0;
out:
*dev = NULL;
return rc;
}
// ----------------------------------------------------------------------------
AudioFlinger::AudioFlinger()
: BnAudioFlinger(),
mPrimaryHardwareDev(NULL),
mAudioHwDevs(NULL),
mHardwareStatus(AUDIO_HW_IDLE),
mMasterVolume(1.0f),
mMasterMute(false),
// mNextUniqueId(AUDIO_UNIQUE_ID_USE_MAX),
mMode(AUDIO_MODE_INVALID),
mBtNrecIsOff(false),
mIsLowRamDevice(true),
mIsDeviceTypeKnown(false),
mGlobalEffectEnableTime(0),
mSystemReady(false)
{
// unsigned instead of audio_unique_id_use_t, because ++ operator is unavailable for enum
for (unsigned use = AUDIO_UNIQUE_ID_USE_UNSPECIFIED; use < AUDIO_UNIQUE_ID_USE_MAX; use++) {
// zero ID has a special meaning, so unavailable
mNextUniqueIds[use] = AUDIO_UNIQUE_ID_USE_MAX;
}
getpid_cached = getpid();
const bool doLog = property_get_bool("ro.test_harness", false);
if (doLog) {
mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters",
MemoryHeapBase::READ_ONLY);
}
// reset battery stats.
// if the audio service has crashed, battery stats could be left
// in bad state, reset the state upon service start.
BatteryNotifier::getInstance().noteResetAudio();
#ifdef TEE_SINK
char value[PROPERTY_VALUE_MAX];
(void) property_get("ro.debuggable", value, "0");
int debuggable = atoi(value);
int teeEnabled = 0;
if (debuggable) {
(void) property_get("af.tee", value, "0");
teeEnabled = atoi(value);
}
// FIXME symbolic constants here
if (teeEnabled & 1) {
mTeeSinkInputEnabled = true;
}
if (teeEnabled & 2) {
mTeeSinkOutputEnabled = true;
}
if (teeEnabled & 4) {
mTeeSinkTrackEnabled = true;
}
#endif
}
void AudioFlinger::onFirstRef()
{
Mutex::Autolock _l(mLock);
/* TODO: move all this work into an Init() function */
char val_str[PROPERTY_VALUE_MAX] = { 0 };
if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) {
uint32_t int_val;
if (1 == sscanf(val_str, "%u", &int_val)) {
mStandbyTimeInNsecs = milliseconds(int_val);
ALOGI("Using %u mSec as standby time.", int_val);
} else {
mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
ALOGI("Using default %u mSec as standby time.",
(uint32_t)(mStandbyTimeInNsecs / 1000000));
}
}
mPatchPanel = new PatchPanel(this);
mMode = AUDIO_MODE_NORMAL;
}
AudioFlinger::~AudioFlinger()
{
while (!mRecordThreads.isEmpty()) {
// closeInput_nonvirtual() will remove specified entry from mRecordThreads
closeInput_nonvirtual(mRecordThreads.keyAt(0));
}
while (!mPlaybackThreads.isEmpty()) {
// closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads
closeOutput_nonvirtual(mPlaybackThreads.keyAt(0));
}
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
// no mHardwareLock needed, as there are no other references to this
audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice());
delete mAudioHwDevs.valueAt(i);
}
// Tell media.log service about any old writers that still need to be unregistered
if (mLogMemoryDealer != 0) {
sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
if (binder != 0) {
sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory());
mUnregisteredWriters.pop();
mediaLogService->unregisterWriter(iMemory);
}
}
}
}
static const char * const audio_interfaces[] = {
AUDIO_HARDWARE_MODULE_ID_PRIMARY,
AUDIO_HARDWARE_MODULE_ID_A2DP,
AUDIO_HARDWARE_MODULE_ID_USB,
};
#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0])))
AudioHwDevice* AudioFlinger::findSuitableHwDev_l(
audio_module_handle_t module,
audio_devices_t devices)
{
// if module is 0, the request comes from an old policy manager and we should load
// well known modules
if (module == 0) {
ALOGW("findSuitableHwDev_l() loading well know audio hw modules");
for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
loadHwModule_l(audio_interfaces[i]);
}
// then try to find a module supporting the requested device.
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i);
audio_hw_device_t *dev = audioHwDevice->hwDevice();
if ((dev->get_supported_devices != NULL) &&
(dev->get_supported_devices(dev) & devices) == devices)
return audioHwDevice;
}
} else {
// check a match for the requested module handle
AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module);
if (audioHwDevice != NULL) {
return audioHwDevice;
}
}
return NULL;
}
void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
result.append("Clients:\n");
for (size_t i = 0; i < mClients.size(); ++i) {
sp<Client> client = mClients.valueAt(i).promote();
if (client != 0) {
snprintf(buffer, SIZE, " pid: %d\n", client->pid());
result.append(buffer);
}
}
result.append("Notification Clients:\n");
for (size_t i = 0; i < mNotificationClients.size(); ++i) {
snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i));
result.append(buffer);
}
result.append("Global session refs:\n");
result.append(" session pid count\n");
for (size_t i = 0; i < mAudioSessionRefs.size(); i++) {
AudioSessionRef *r = mAudioSessionRefs[i];
snprintf(buffer, SIZE, " %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt);
result.append(buffer);
}
write(fd, result.string(), result.size());
}
void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
hardware_call_state hardwareStatus = mHardwareStatus;
snprintf(buffer, SIZE, "Hardware status: %d\n"
"Standby Time mSec: %u\n",
hardwareStatus,
(uint32_t)(mStandbyTimeInNsecs / 1000000));
result.append(buffer);
write(fd, result.string(), result.size());
}
void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused)
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
snprintf(buffer, SIZE, "Permission Denial: "
"can't dump AudioFlinger from pid=%d, uid=%d\n",
IPCThreadState::self()->getCallingPid(),
IPCThreadState::self()->getCallingUid());
result.append(buffer);
write(fd, result.string(), result.size());
}
bool AudioFlinger::dumpTryLock(Mutex& mutex)
{
bool locked = false;
for (int i = 0; i < kDumpLockRetries; ++i) {
if (mutex.tryLock() == NO_ERROR) {
locked = true;
break;
}
usleep(kDumpLockSleepUs);
}
return locked;
}
status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
{
if (!dumpAllowed()) {
dumpPermissionDenial(fd, args);
} else {
// get state of hardware lock
bool hardwareLocked = dumpTryLock(mHardwareLock);
if (!hardwareLocked) {
String8 result(kHardwareLockedString);
write(fd, result.string(), result.size());
} else {
mHardwareLock.unlock();
}
bool locked = dumpTryLock(mLock);
// failed to lock - AudioFlinger is probably deadlocked
if (!locked) {
String8 result(kDeadlockedString);
write(fd, result.string(), result.size());
}
bool clientLocked = dumpTryLock(mClientLock);
if (!clientLocked) {
String8 result(kClientLockedString);
write(fd, result.string(), result.size());
}
EffectDumpEffects(fd);
dumpClients(fd, args);
if (clientLocked) {
mClientLock.unlock();
}
dumpInternals(fd, args);
// dump playback threads
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
mPlaybackThreads.valueAt(i)->dump(fd, args);
}
// dump record threads
for (size_t i = 0; i < mRecordThreads.size(); i++) {
mRecordThreads.valueAt(i)->dump(fd, args);
}
// dump orphan effect chains
if (mOrphanEffectChains.size() != 0) {
write(fd, " Orphan Effect Chains\n", strlen(" Orphan Effect Chains\n"));
for (size_t i = 0; i < mOrphanEffectChains.size(); i++) {
mOrphanEffectChains.valueAt(i)->dump(fd, args);
}
}
// dump all hardware devs
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
dev->dump(dev, fd);
}
#ifdef TEE_SINK
// dump the serially shared record tee sink
if (mRecordTeeSource != 0) {
dumpTee(fd, mRecordTeeSource);
}
#endif
if (locked) {
mLock.unlock();
}
// append a copy of media.log here by forwarding fd to it, but don't attempt
// to lookup the service if it's not running, as it will block for a second
if (mLogMemoryDealer != 0) {
sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
if (binder != 0) {
dprintf(fd, "\nmedia.log:\n");
Vector<String16> args;
binder->dump(fd, args);
}
}
// check for optional arguments
bool dumpMem = false;
bool unreachableMemory = false;
for (const auto &arg : args) {
if (arg == String16("-m")) {
dumpMem = true;
} else if (arg == String16("--unreachable")) {
unreachableMemory = true;
}
}
if (dumpMem) {
dprintf(fd, "\nDumping memory:\n");
std::string s = dumpMemoryAddresses(100 /* limit */);
write(fd, s.c_str(), s.size());
}
if (unreachableMemory) {
dprintf(fd, "\nDumping unreachable memory:\n");
// TODO - should limit be an argument parameter?
std::string s = GetUnreachableMemoryString(true /* contents */, 100 /* limit */);
write(fd, s.c_str(), s.size());
}
}
return NO_ERROR;
}
sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid)
{
Mutex::Autolock _cl(mClientLock);
// If pid is already in the mClients wp<> map, then use that entry
// (for which promote() is always != 0), otherwise create a new entry and Client.
sp<Client> client = mClients.valueFor(pid).promote();
if (client == 0) {
client = new Client(this, pid);
mClients.add(pid, client);
}
return client;
}
sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name)
{
// If there is no memory allocated for logs, return a dummy writer that does nothing
if (mLogMemoryDealer == 0) {
return new NBLog::Writer();
}
sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log"));
// Similarly if we can't contact the media.log service, also return a dummy writer
if (binder == 0) {
return new NBLog::Writer();
}
sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder));
sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
// If allocation fails, consult the vector of previously unregistered writers
// and garbage-collect one or more them until an allocation succeeds
if (shared == 0) {
Mutex::Autolock _l(mUnregisteredWritersLock);
for (size_t count = mUnregisteredWriters.size(); count > 0; count--) {
{
// Pick the oldest stale writer to garbage-collect
sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory());
mUnregisteredWriters.removeAt(0);
mediaLogService->unregisterWriter(iMemory);
// Now the media.log remote reference to IMemory is gone. When our last local
// reference to IMemory also drops to zero at end of this block,
// the IMemory destructor will deallocate the region from mLogMemoryDealer.
}
// Re-attempt the allocation
shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size));
if (shared != 0) {
goto success;
}
}
// Even after garbage-collecting all old writers, there is still not enough memory,
// so return a dummy writer
return new NBLog::Writer();
}
success:
mediaLogService->registerWriter(shared, size, name);
return new NBLog::Writer(size, shared);
}
void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer)
{
if (writer == 0) {
return;
}
sp<IMemory> iMemory(writer->getIMemory());
if (iMemory == 0) {
return;
}
// Rather than removing the writer immediately, append it to a queue of old writers to
// be garbage-collected later. This allows us to continue to view old logs for a while.
Mutex::Autolock _l(mUnregisteredWritersLock);
mUnregisteredWriters.push(writer);
}
// IAudioFlinger interface
sp<IAudioTrack> AudioFlinger::createTrack(
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t *frameCount,
IAudioFlinger::track_flags_t *flags,
const sp<IMemory>& sharedBuffer,
audio_io_handle_t output,
pid_t pid,
pid_t tid,
audio_session_t *sessionId,
int clientUid,
status_t *status)
{
sp<PlaybackThread::Track> track;
sp<TrackHandle> trackHandle;
sp<Client> client;
status_t lStatus;
audio_session_t lSessionId;
const uid_t callingUid = IPCThreadState::self()->getCallingUid();
if (pid == -1 || !isTrustedCallingUid(callingUid)) {
const pid_t callingPid = IPCThreadState::self()->getCallingPid();
ALOGW_IF(pid != -1 && pid != callingPid,
"%s uid %d pid %d tried to pass itself off as pid %d",
__func__, callingUid, callingPid, pid);
pid = callingPid;
}
// client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC,
// but if someone uses binder directly they could bypass that and cause us to crash
if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
ALOGE("createTrack() invalid stream type %d", streamType);
lStatus = BAD_VALUE;
goto Exit;
}
// further sample rate checks are performed by createTrack_l() depending on the thread type
if (sampleRate == 0) {
ALOGE("createTrack() invalid sample rate %u", sampleRate);
lStatus = BAD_VALUE;
goto Exit;
}
// further channel mask checks are performed by createTrack_l() depending on the thread type
if (!audio_is_output_channel(channelMask)) {
ALOGE("createTrack() invalid channel mask %#x", channelMask);
lStatus = BAD_VALUE;
goto Exit;
}
// further format checks are performed by createTrack_l() depending on the thread type
if (!audio_is_valid_format(format)) {
ALOGE("createTrack() invalid format %#x", format);
lStatus = BAD_VALUE;
goto Exit;
}
if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) {
ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()");
lStatus = BAD_VALUE;
goto Exit;
}
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
ALOGE("no playback thread found for output handle %d", output);
lStatus = BAD_VALUE;
goto Exit;
}
client = registerPid(pid);
PlaybackThread *effectThread = NULL;
if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) {
ALOGE("createTrack() invalid session ID %d", *sessionId);
lStatus = BAD_VALUE;
goto Exit;
}
lSessionId = *sessionId;
// check if an effect chain with the same session ID is present on another
// output thread and move it here.
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
if (mPlaybackThreads.keyAt(i) != output) {
uint32_t sessions = t->hasAudioSession(lSessionId);
if (sessions & PlaybackThread::EFFECT_SESSION) {
effectThread = t.get();
break;
}
}
}
} else {
// if no audio session id is provided, create one here
lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
if (sessionId != NULL) {
*sessionId = lSessionId;
}
}
ALOGV("createTrack() lSessionId: %d", lSessionId);
track = thread->createTrack_l(client, streamType, sampleRate, format,
channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus);
LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0));
// we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
// move effect chain to this output thread if an effect on same session was waiting
// for a track to be created
if (lStatus == NO_ERROR && effectThread != NULL) {
// no risk of deadlock because AudioFlinger::mLock is held
Mutex::Autolock _dl(thread->mLock);
Mutex::Autolock _sl(effectThread->mLock);
moveEffectChain_l(lSessionId, effectThread, thread, true);
}
// Look for sync events awaiting for a session to be used.
for (size_t i = 0; i < mPendingSyncEvents.size(); i++) {
if (mPendingSyncEvents[i]->triggerSession() == lSessionId) {
if (thread->isValidSyncEvent(mPendingSyncEvents[i])) {
if (lStatus == NO_ERROR) {
(void) track->setSyncEvent(mPendingSyncEvents[i]);
} else {
mPendingSyncEvents[i]->cancel();
}
mPendingSyncEvents.removeAt(i);
i--;
}
}
}
setAudioHwSyncForSession_l(thread, lSessionId);
}
if (lStatus != NO_ERROR) {
// remove local strong reference to Client before deleting the Track so that the
// Client destructor is called by the TrackBase destructor with mClientLock held
// Don't hold mClientLock when releasing the reference on the track as the
// destructor will acquire it.
{
Mutex::Autolock _cl(mClientLock);
client.clear();
}
track.clear();
goto Exit;
}
// return handle to client
trackHandle = new TrackHandle(track);
Exit:
*status = lStatus;
return trackHandle;
}
uint32_t AudioFlinger::sampleRate(audio_io_handle_t ioHandle) const
{
Mutex::Autolock _l(mLock);
ThreadBase *thread = checkThread_l(ioHandle);
if (thread == NULL) {
ALOGW("sampleRate() unknown thread %d", ioHandle);
return 0;
}
return thread->sampleRate();
}
audio_format_t AudioFlinger::format(audio_io_handle_t output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
ALOGW("format() unknown thread %d", output);
return AUDIO_FORMAT_INVALID;
}
return thread->format();
}
size_t AudioFlinger::frameCount(audio_io_handle_t ioHandle) const
{
Mutex::Autolock _l(mLock);
ThreadBase *thread = checkThread_l(ioHandle);
if (thread == NULL) {
ALOGW("frameCount() unknown thread %d", ioHandle);
return 0;
}
// FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers;
// should examine all callers and fix them to handle smaller counts
return thread->frameCount();
}
size_t AudioFlinger::frameCountHAL(audio_io_handle_t ioHandle) const
{
Mutex::Autolock _l(mLock);
ThreadBase *thread = checkThread_l(ioHandle);
if (thread == NULL) {
ALOGW("frameCountHAL() unknown thread %d", ioHandle);
return 0;
}
return thread->frameCountHAL();
}
uint32_t AudioFlinger::latency(audio_io_handle_t output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
ALOGW("latency(): no playback thread found for output handle %d", output);
return 0;
}
return thread->latency();
}
status_t AudioFlinger::setMasterVolume(float value)
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return ret;
}
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
Mutex::Autolock _l(mLock);
mMasterVolume = value;
// Set master volume in the HALs which support it.
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
AutoMutex lock(mHardwareLock);
AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
if (dev->canSetMasterVolume()) {
dev->hwDevice()->set_master_volume(dev->hwDevice(), value);
}
mHardwareStatus = AUDIO_HW_IDLE;
}
// Now set the master volume in each playback thread. Playback threads
// assigned to HALs which do not have master volume support will apply
// master volume during the mix operation. Threads with HALs which do
// support master volume will simply ignore the setting.
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
if (mPlaybackThreads.valueAt(i)->isDuplicating()) {
continue;
}
mPlaybackThreads.valueAt(i)->setMasterVolume(value);
}
return NO_ERROR;
}
status_t AudioFlinger::setMode(audio_mode_t mode)
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return ret;
}
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
if (uint32_t(mode) >= AUDIO_MODE_CNT) {
ALOGW("Illegal value: setMode(%d)", mode);
return BAD_VALUE;
}
{ // scope for the lock
AutoMutex lock(mHardwareLock);
audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
mHardwareStatus = AUDIO_HW_SET_MODE;
ret = dev->set_mode(dev, mode);
mHardwareStatus = AUDIO_HW_IDLE;
}
if (NO_ERROR == ret) {
Mutex::Autolock _l(mLock);
mMode = mode;
for (size_t i = 0; i < mPlaybackThreads.size(); i++)
mPlaybackThreads.valueAt(i)->setMode(mode);
}
return ret;
}
status_t AudioFlinger::setMicMute(bool state)
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return ret;
}
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
status_t result = dev->set_mic_mute(dev, state);
if (result != NO_ERROR) {
ret = result;
}
}
mHardwareStatus = AUDIO_HW_IDLE;
return ret;
}
bool AudioFlinger::getMicMute() const
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return false;
}
bool mute = true;
bool state = AUDIO_MODE_INVALID;
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
status_t result = dev->get_mic_mute(dev, &state);
if (result == NO_ERROR) {
mute = mute && state;
}
}
mHardwareStatus = AUDIO_HW_IDLE;
return mute;
}
status_t AudioFlinger::setMasterMute(bool muted)
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return ret;
}
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
Mutex::Autolock _l(mLock);
mMasterMute = muted;
// Set master mute in the HALs which support it.
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
AutoMutex lock(mHardwareLock);
AudioHwDevice *dev = mAudioHwDevs.valueAt(i);
mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
if (dev->canSetMasterMute()) {
dev->hwDevice()->set_master_mute(dev->hwDevice(), muted);
}
mHardwareStatus = AUDIO_HW_IDLE;
}
// Now set the master mute in each playback thread. Playback threads
// assigned to HALs which do not have master mute support will apply master
// mute during the mix operation. Threads with HALs which do support master
// mute will simply ignore the setting.
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
if (mPlaybackThreads.valueAt(i)->isDuplicating()) {
continue;
}
mPlaybackThreads.valueAt(i)->setMasterMute(muted);
}
return NO_ERROR;
}
float AudioFlinger::masterVolume() const
{
Mutex::Autolock _l(mLock);
return masterVolume_l();
}
bool AudioFlinger::masterMute() const
{
Mutex::Autolock _l(mLock);
return masterMute_l();
}
float AudioFlinger::masterVolume_l() const
{
return mMasterVolume;
}
bool AudioFlinger::masterMute_l() const
{
return mMasterMute;
}
status_t AudioFlinger::checkStreamType(audio_stream_type_t stream) const
{
if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
ALOGW("setStreamVolume() invalid stream %d", stream);
return BAD_VALUE;
}
pid_t caller = IPCThreadState::self()->getCallingPid();
if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT && caller != getpid_cached) {
ALOGW("setStreamVolume() pid %d cannot use internal stream type %d", caller, stream);
return PERMISSION_DENIED;
}
return NO_ERROR;
}
status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
audio_io_handle_t output)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
status_t status = checkStreamType(stream);
if (status != NO_ERROR) {
return status;
}
ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to change AUDIO_STREAM_PATCH volume");
AutoMutex lock(mLock);
PlaybackThread *thread = NULL;
if (output != AUDIO_IO_HANDLE_NONE) {
thread = checkPlaybackThread_l(output);
if (thread == NULL) {
return BAD_VALUE;
}
}
mStreamTypes[stream].volume = value;
if (thread == NULL) {
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
}
} else {
thread->setStreamVolume(stream, value);
}
return NO_ERROR;
}
status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
status_t status = checkStreamType(stream);
if (status != NO_ERROR) {
return status;
}
ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to mute AUDIO_STREAM_PATCH");
if (uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
ALOGE("setStreamMute() invalid stream %d", stream);
return BAD_VALUE;
}
AutoMutex lock(mLock);
mStreamTypes[stream].mute = muted;
for (size_t i = 0; i < mPlaybackThreads.size(); i++)
mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
return NO_ERROR;
}
float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
{
status_t status = checkStreamType(stream);
if (status != NO_ERROR) {
return 0.0f;
}
AutoMutex lock(mLock);
float volume;
if (output != AUDIO_IO_HANDLE_NONE) {
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
return 0.0f;
}
volume = thread->streamVolume(stream);
} else {
volume = streamVolume_l(stream);
}
return volume;
}
bool AudioFlinger::streamMute(audio_stream_type_t stream) const
{
status_t status = checkStreamType(stream);
if (status != NO_ERROR) {
return true;
}
AutoMutex lock(mLock);
return streamMute_l(stream);
}
void AudioFlinger::broacastParametersToRecordThreads_l(const String8& keyValuePairs)
{
for (size_t i = 0; i < mRecordThreads.size(); i++) {
mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
}
}
status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
{
ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d",
ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid());
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
// AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface
if (ioHandle == AUDIO_IO_HANDLE_NONE) {
Mutex::Autolock _l(mLock);
status_t final_result = NO_ERROR;
{
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_PARAMETER;
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
status_t result = dev->set_parameters(dev, keyValuePairs.string());
final_result = result ?: final_result;
}
mHardwareStatus = AUDIO_HW_IDLE;
}
// disable AEC and NS if the device is a BT SCO headset supporting those pre processings
AudioParameter param = AudioParameter(keyValuePairs);
String8 value;
if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) {
bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF);
if (mBtNrecIsOff != btNrecIsOff) {
for (size_t i = 0; i < mRecordThreads.size(); i++) {
sp<RecordThread> thread = mRecordThreads.valueAt(i);
audio_devices_t device = thread->inDevice();
bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff;
// collect all of the thread's session IDs
KeyedVector<audio_session_t, bool> ids = thread->sessionIds();
// suspend effects associated with those session IDs
for (size_t j = 0; j < ids.size(); ++j) {
audio_session_t sessionId = ids.keyAt(j);
thread->setEffectSuspended(FX_IID_AEC,
suspend,
sessionId);
thread->setEffectSuspended(FX_IID_NS,
suspend,
sessionId);
}
}
mBtNrecIsOff = btNrecIsOff;
}
}
String8 screenState;
if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) {
bool isOff = screenState == "off";
if (isOff != (AudioFlinger::mScreenState & 1)) {
AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff;
}
}
return final_result;
}
// hold a strong ref on thread in case closeOutput() or closeInput() is called
// and the thread is exited once the lock is released
sp<ThreadBase> thread;
{
Mutex::Autolock _l(mLock);
thread = checkPlaybackThread_l(ioHandle);
if (thread == 0) {
thread = checkRecordThread_l(ioHandle);
} else if (thread == primaryPlaybackThread_l()) {
// indicate output device change to all input threads for pre processing
AudioParameter param = AudioParameter(keyValuePairs);
int value;
if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) &&
(value != 0)) {
broacastParametersToRecordThreads_l(keyValuePairs);
}
}
}
if (thread != 0) {
return thread->setParameters(keyValuePairs);
}
return BAD_VALUE;
}
String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
{
ALOGVV("getParameters() io %d, keys %s, calling pid %d",
ioHandle, keys.string(), IPCThreadState::self()->getCallingPid());
Mutex::Autolock _l(mLock);
if (ioHandle == AUDIO_IO_HANDLE_NONE) {
String8 out_s8;
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
char *s;
{
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_GET_PARAMETER;
audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice();
s = dev->get_parameters(dev, keys.string());
mHardwareStatus = AUDIO_HW_IDLE;
}
out_s8 += String8(s ? s : "");
free(s);
}
return out_s8;
}
PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle);
if (playbackThread != NULL) {
return playbackThread->getParameters(keys);
}
RecordThread *recordThread = checkRecordThread_l(ioHandle);
if (recordThread != NULL) {
return recordThread->getParameters(keys);
}
return String8("");
}
size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
audio_channel_mask_t channelMask) const
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return 0;
}
if ((sampleRate == 0) ||
!audio_is_valid_format(format) || !audio_has_proportional_frames(format) ||
!audio_is_input_channel(channelMask)) {
return 0;
}
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
audio_config_t config, proposed;
memset(&proposed, 0, sizeof(proposed));
proposed.sample_rate = sampleRate;
proposed.channel_mask = channelMask;
proposed.format = format;
audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
size_t frames;
for (;;) {
// Note: config is currently a const parameter for get_input_buffer_size()
// but we use a copy from proposed in case config changes from the call.
config = proposed;
frames = dev->get_input_buffer_size(dev, &config);
if (frames != 0) {
break; // hal success, config is the result
}
// change one parameter of the configuration each iteration to a more "common" value
// to see if the device will support it.
if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) {
proposed.format = AUDIO_FORMAT_PCM_16_BIT;
} else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as
proposed.sample_rate = 44100; // legacy AudioRecord.java. TODO: Query hw?
} else {
ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, "
"format %#x, channelMask 0x%X",
sampleRate, format, channelMask);
break; // retries failed, break out of loop with frames == 0.
}
}
mHardwareStatus = AUDIO_HW_IDLE;
if (frames > 0 && config.sample_rate != sampleRate) {
frames = destinationFramesPossible(frames, sampleRate, config.sample_rate);
}
return frames; // may be converted to bytes at the Java level.
}
uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
{
Mutex::Autolock _l(mLock);
RecordThread *recordThread = checkRecordThread_l(ioHandle);
if (recordThread != NULL) {
return recordThread->getInputFramesLost();
}
return 0;
}
status_t AudioFlinger::setVoiceVolume(float value)
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return ret;
}
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
AutoMutex lock(mHardwareLock);
audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME;
ret = dev->set_voice_volume(dev, value);
mHardwareStatus = AUDIO_HW_IDLE;
return ret;
}
status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
audio_io_handle_t output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *playbackThread = checkPlaybackThread_l(output);
if (playbackThread != NULL) {
return playbackThread->getRenderPosition(halFrames, dspFrames);
}
return BAD_VALUE;
}
void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
{
Mutex::Autolock _l(mLock);
if (client == 0) {
return;
}
pid_t pid = IPCThreadState::self()->getCallingPid();
{
Mutex::Autolock _cl(mClientLock);
if (mNotificationClients.indexOfKey(pid) < 0) {
sp<NotificationClient> notificationClient = new NotificationClient(this,
client,
pid);
ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid);
mNotificationClients.add(pid, notificationClient);
sp<IBinder> binder = IInterface::asBinder(client);
binder->linkToDeath(notificationClient);
}
}
// mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the
// ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock.
// the config change is always sent from playback or record threads to avoid deadlock
// with AudioSystem::gLock
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AUDIO_OUTPUT_OPENED, pid);
}
for (size_t i = 0; i < mRecordThreads.size(); i++) {
mRecordThreads.valueAt(i)->sendIoConfigEvent(AUDIO_INPUT_OPENED, pid);
}
}
void AudioFlinger::removeNotificationClient(pid_t pid)
{
Mutex::Autolock _l(mLock);
{
Mutex::Autolock _cl(mClientLock);
mNotificationClients.removeItem(pid);
}
ALOGV("%d died, releasing its sessions", pid);
size_t num = mAudioSessionRefs.size();
bool removed = false;
for (size_t i = 0; i< num; ) {
AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
ALOGV(" pid %d @ %zu", ref->mPid, i);
if (ref->mPid == pid) {
ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid);
mAudioSessionRefs.removeAt(i);
delete ref;
removed = true;
num--;
} else {
i++;
}
}
if (removed) {
purgeStaleEffects_l();
}
}
void AudioFlinger::ioConfigChanged(audio_io_config_event event,
const sp<AudioIoDescriptor>& ioDesc,
pid_t pid)
{
Mutex::Autolock _l(mClientLock);
size_t size = mNotificationClients.size();
for (size_t i = 0; i < size; i++) {
if ((pid == 0) || (mNotificationClients.keyAt(i) == pid)) {
mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioDesc);
}
}
}
// removeClient_l() must be called with AudioFlinger::mClientLock held
void AudioFlinger::removeClient_l(pid_t pid)
{
ALOGV("removeClient_l() pid %d, calling pid %d", pid,
IPCThreadState::self()->getCallingPid());
mClients.removeItem(pid);
}
// getEffectThread_l() must be called with AudioFlinger::mLock held
sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(audio_session_t sessionId,
int EffectId)
{
sp<PlaybackThread> thread;
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) {
ALOG_ASSERT(thread == 0);
thread = mPlaybackThreads.valueAt(i);
}
}
return thread;
}
// ----------------------------------------------------------------------------
AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
: RefBase(),
mAudioFlinger(audioFlinger),
mPid(pid)
{
size_t heapSize = property_get_int32("ro.af.client_heap_size_kbyte", 0);
heapSize *= 1024;
if (!heapSize) {
heapSize = kClientSharedHeapSizeBytes;
// Increase heap size on non low ram devices to limit risk of reconnection failure for
// invalidated tracks
if (!audioFlinger->isLowRamDevice()) {
heapSize *= kClientSharedHeapSizeMultiplier;
}
}
mMemoryDealer = new MemoryDealer(heapSize, "AudioFlinger::Client");
}
// Client destructor must be called with AudioFlinger::mClientLock held
AudioFlinger::Client::~Client()
{
mAudioFlinger->removeClient_l(mPid);
}
sp<MemoryDealer> AudioFlinger::Client::heap() const
{
return mMemoryDealer;
}
// ----------------------------------------------------------------------------
AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
const sp<IAudioFlingerClient>& client,
pid_t pid)
: mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
{
}
AudioFlinger::NotificationClient::~NotificationClient()
{
}
void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused)
{
sp<NotificationClient> keep(this);
mAudioFlinger->removeNotificationClient(mPid);
}
// ----------------------------------------------------------------------------
sp<IAudioRecord> AudioFlinger::openRecord(
audio_io_handle_t input,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
const String16& opPackageName,
size_t *frameCount,
IAudioFlinger::track_flags_t *flags,
pid_t pid,
pid_t tid,
int clientUid,
audio_session_t *sessionId,
size_t *notificationFrames,
sp<IMemory>& cblk,
sp<IMemory>& buffers,
status_t *status)
{
sp<RecordThread::RecordTrack> recordTrack;
sp<RecordHandle> recordHandle;
sp<Client> client;
status_t lStatus;
audio_session_t lSessionId;
cblk.clear();
buffers.clear();
bool updatePid = (pid == -1);
const uid_t callingUid = IPCThreadState::self()->getCallingUid();
if (!isTrustedCallingUid(callingUid)) {
ALOGW_IF((uid_t)clientUid != callingUid,
"%s uid %d tried to pass itself off as %d", __FUNCTION__, callingUid, clientUid);
clientUid = callingUid;
updatePid = true;
}
if (updatePid) {
const pid_t callingPid = IPCThreadState::self()->getCallingPid();
ALOGW_IF(pid != -1 && pid != callingPid,
"%s uid %d pid %d tried to pass itself off as pid %d",
__func__, callingUid, callingPid, pid);
pid = callingPid;
}
// check calling permissions
if (!recordingAllowed(opPackageName, tid, clientUid)) {
ALOGE("openRecord() permission denied: recording not allowed");
lStatus = PERMISSION_DENIED;
goto Exit;
}
// further sample rate checks are performed by createRecordTrack_l()
if (sampleRate == 0) {
ALOGE("openRecord() invalid sample rate %u", sampleRate);
lStatus = BAD_VALUE;
goto Exit;
}
// we don't yet support anything other than linear PCM
if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) {
ALOGE("openRecord() invalid format %#x", format);
lStatus = BAD_VALUE;
goto Exit;
}
// further channel mask checks are performed by createRecordTrack_l()
if (!audio_is_input_channel(channelMask)) {
ALOGE("openRecord() invalid channel mask %#x", channelMask);
lStatus = BAD_VALUE;
goto Exit;
}
{
Mutex::Autolock _l(mLock);
RecordThread *thread = checkRecordThread_l(input);
if (thread == NULL) {
ALOGE("openRecord() checkRecordThread_l failed");
lStatus = BAD_VALUE;
goto Exit;
}
client = registerPid(pid);
if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) {
lStatus = BAD_VALUE;
goto Exit;
}
lSessionId = *sessionId;
} else {
// if no audio session id is provided, create one here
lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
if (sessionId != NULL) {
*sessionId = lSessionId;
}
}
ALOGV("openRecord() lSessionId: %d input %d", lSessionId, input);
recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
frameCount, lSessionId, notificationFrames,
clientUid, flags, tid, &lStatus);
LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0));
if (lStatus == NO_ERROR) {
// Check if one effect chain was awaiting for an AudioRecord to be created on this
// session and move it to this thread.
sp<EffectChain> chain = getOrphanEffectChain_l(lSessionId);
if (chain != 0) {
Mutex::Autolock _l(thread->mLock);
thread->addEffectChain_l(chain);
}
}
}
if (lStatus != NO_ERROR) {
// remove local strong reference to Client before deleting the RecordTrack so that the
// Client destructor is called by the TrackBase destructor with mClientLock held
// Don't hold mClientLock when releasing the reference on the track as the
// destructor will acquire it.
{
Mutex::Autolock _cl(mClientLock);
client.clear();
}
recordTrack.clear();
goto Exit;
}
cblk = recordTrack->getCblk();
buffers = recordTrack->getBuffers();
// return handle to client
recordHandle = new RecordHandle(recordTrack);
Exit:
*status = lStatus;
return recordHandle;
}
// ----------------------------------------------------------------------------
audio_module_handle_t AudioFlinger::loadHwModule(const char *name)
{
if (name == NULL) {
return AUDIO_MODULE_HANDLE_NONE;
}
if (!settingsAllowed()) {
return AUDIO_MODULE_HANDLE_NONE;
}
Mutex::Autolock _l(mLock);
return loadHwModule_l(name);
}
// loadHwModule_l() must be called with AudioFlinger::mLock held
audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name)
{
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) {
ALOGW("loadHwModule() module %s already loaded", name);
return mAudioHwDevs.keyAt(i);
}
}
audio_hw_device_t *dev;
int rc = load_audio_interface(name, &dev);
if (rc) {
ALOGE("loadHwModule() error %d loading module %s", rc, name);
return AUDIO_MODULE_HANDLE_NONE;
}
mHardwareStatus = AUDIO_HW_INIT;
rc = dev->init_check(dev);
mHardwareStatus = AUDIO_HW_IDLE;
if (rc) {
ALOGE("loadHwModule() init check error %d for module %s", rc, name);
return AUDIO_MODULE_HANDLE_NONE;
}
// Check and cache this HAL's level of support for master mute and master
// volume. If this is the first HAL opened, and it supports the get
// methods, use the initial values provided by the HAL as the current
// master mute and volume settings.
AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0);
{ // scope for auto-lock pattern
AutoMutex lock(mHardwareLock);
if (0 == mAudioHwDevs.size()) {
mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
if (NULL != dev->get_master_volume) {
float mv;
if (OK == dev->get_master_volume(dev, &mv)) {
mMasterVolume = mv;
}
}
mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE;
if (NULL != dev->get_master_mute) {
bool mm;
if (OK == dev->get_master_mute(dev, &mm)) {
mMasterMute = mm;
}
}
}
mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
if ((NULL != dev->set_master_volume) &&
(OK == dev->set_master_volume(dev, mMasterVolume))) {
flags = static_cast<AudioHwDevice::Flags>(flags |
AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME);
}
mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE;
if ((NULL != dev->set_master_mute) &&
(OK == dev->set_master_mute(dev, mMasterMute))) {
flags = static_cast<AudioHwDevice::Flags>(flags |
AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE);
}
mHardwareStatus = AUDIO_HW_IDLE;
}
audio_module_handle_t handle = (audio_module_handle_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_MODULE);
mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags));
ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d",
name, dev->common.module->name, dev->common.module->id, handle);
return handle;
}
// ----------------------------------------------------------------------------
uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = primaryPlaybackThread_l();
return thread != NULL ? thread->sampleRate() : 0;
}
size_t AudioFlinger::getPrimaryOutputFrameCount()
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = primaryPlaybackThread_l();
return thread != NULL ? thread->frameCountHAL() : 0;
}
// ----------------------------------------------------------------------------
status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice)
{
uid_t uid = IPCThreadState::self()->getCallingUid();
if (uid != AID_SYSTEM) {
return PERMISSION_DENIED;
}
Mutex::Autolock _l(mLock);
if (mIsDeviceTypeKnown) {
return INVALID_OPERATION;
}
mIsLowRamDevice = isLowRamDevice;
mIsDeviceTypeKnown = true;
return NO_ERROR;
}
audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId)
{
Mutex::Autolock _l(mLock);
ssize_t index = mHwAvSyncIds.indexOfKey(sessionId);
if (index >= 0) {
ALOGV("getAudioHwSyncForSession found ID %d for session %d",
mHwAvSyncIds.valueAt(index), sessionId);
return mHwAvSyncIds.valueAt(index);
}
audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
if (dev == NULL) {
return AUDIO_HW_SYNC_INVALID;
}
char *reply = dev->get_parameters(dev, AUDIO_PARAMETER_HW_AV_SYNC);
AudioParameter param = AudioParameter(String8(reply));
free(reply);
int value;
if (param.getInt(String8(AUDIO_PARAMETER_HW_AV_SYNC), value) != NO_ERROR) {
ALOGW("getAudioHwSyncForSession error getting sync for session %d", sessionId);
return AUDIO_HW_SYNC_INVALID;
}
// allow only one session for a given HW A/V sync ID.
for (size_t i = 0; i < mHwAvSyncIds.size(); i++) {
if (mHwAvSyncIds.valueAt(i) == (audio_hw_sync_t)value) {
ALOGV("getAudioHwSyncForSession removing ID %d for session %d",
value, mHwAvSyncIds.keyAt(i));
mHwAvSyncIds.removeItemsAt(i);
break;
}
}
mHwAvSyncIds.add(sessionId, value);
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i);
uint32_t sessions = thread->hasAudioSession(sessionId);
if (sessions & PlaybackThread::TRACK_SESSION) {
AudioParameter param = AudioParameter();
param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), value);
thread->setParameters(param.toString());
break;
}
}
ALOGV("getAudioHwSyncForSession adding ID %d for session %d", value, sessionId);
return (audio_hw_sync_t)value;
}
status_t AudioFlinger::systemReady()
{
Mutex::Autolock _l(mLock);
ALOGI("%s", __FUNCTION__);
if (mSystemReady) {
ALOGW("%s called twice", __FUNCTION__);
return NO_ERROR;
}
mSystemReady = true;
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
ThreadBase *thread = (ThreadBase *)mPlaybackThreads.valueAt(i).get();
thread->systemReady();
}
for (size_t i = 0; i < mRecordThreads.size(); i++) {
ThreadBase *thread = (ThreadBase *)mRecordThreads.valueAt(i).get();
thread->systemReady();
}
return NO_ERROR;
}
// setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held
void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId)
{
ssize_t index = mHwAvSyncIds.indexOfKey(sessionId);
if (index >= 0) {
audio_hw_sync_t syncId = mHwAvSyncIds.valueAt(index);
ALOGV("setAudioHwSyncForSession_l found ID %d for session %d", syncId, sessionId);
AudioParameter param = AudioParameter();
param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), syncId);
thread->setParameters(param.toString());
}
}
// ----------------------------------------------------------------------------
sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_t module,
audio_io_handle_t *output,
audio_config_t *config,
audio_devices_t devices,
const String8& address,
audio_output_flags_t flags)
{
AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices);
if (outHwDev == NULL) {
return 0;
}
if (*output == AUDIO_IO_HANDLE_NONE) {
*output = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT);
} else {
// Audio Policy does not currently request a specific output handle.
// If this is ever needed, see openInput_l() for example code.
ALOGE("openOutput_l requested output handle %d is not AUDIO_IO_HANDLE_NONE", *output);
return 0;
}
mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
// FOR TESTING ONLY:
// This if statement allows overriding the audio policy settings
// and forcing a specific format or channel mask to the HAL/Sink device for testing.
if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) {
// Check only for Normal Mixing mode
if (kEnableExtendedPrecision) {
// Specify format (uncomment one below to choose)
//config->format = AUDIO_FORMAT_PCM_FLOAT;
//config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED;
//config->format = AUDIO_FORMAT_PCM_32_BIT;
//config->format = AUDIO_FORMAT_PCM_8_24_BIT;
// ALOGV("openOutput_l() upgrading format to %#08x", config->format);
}
if (kEnableExtendedChannels) {
// Specify channel mask (uncomment one below to choose)
//config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch
//config->channel_mask = audio_channel_mask_from_representation_and_bits(
// AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example
}
}
AudioStreamOut *outputStream = NULL;
status_t status = outHwDev->openOutputStream(
&outputStream,
*output,
devices,
flags,
config,
address.string());
mHardwareStatus = AUDIO_HW_IDLE;
if (status == NO_ERROR) {
PlaybackThread *thread;
if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
thread = new OffloadThread(this, outputStream, *output, devices, mSystemReady);
ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread);
} else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT)
|| !isValidPcmSinkFormat(config->format)
|| !isValidPcmSinkChannelMask(config->channel_mask)) {
thread = new DirectOutputThread(this, outputStream, *output, devices, mSystemReady);
ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread);
} else {
thread = new MixerThread(this, outputStream, *output, devices, mSystemReady);
ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread);
}
mPlaybackThreads.add(*output, thread);
return thread;
}
return 0;
}
status_t AudioFlinger::openOutput(audio_module_handle_t module,
audio_io_handle_t *output,
audio_config_t *config,
audio_devices_t *devices,
const String8& address,
uint32_t *latencyMs,
audio_output_flags_t flags)
{
ALOGI("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x",
module,
(devices != NULL) ? *devices : 0,
config->sample_rate,
config->format,
config->channel_mask,
flags);
if (*devices == AUDIO_DEVICE_NONE) {
return BAD_VALUE;
}
Mutex::Autolock _l(mLock);
sp<PlaybackThread> thread = openOutput_l(module, output, config, *devices, address, flags);
if (thread != 0) {
*latencyMs = thread->latency();
// notify client processes of the new output creation
thread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
// the first primary output opened designates the primary hw device
if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
ALOGI("Using module %d has the primary audio interface", module);
mPrimaryHardwareDev = thread->getOutput()->audioHwDev;
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_SET_MODE;
mPrimaryHardwareDev->hwDevice()->set_mode(mPrimaryHardwareDev->hwDevice(), mMode);
mHardwareStatus = AUDIO_HW_IDLE;
}
return NO_ERROR;
}
return NO_INIT;
}
audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
audio_io_handle_t output2)
{
Mutex::Autolock _l(mLock);
MixerThread *thread1 = checkMixerThread_l(output1);
MixerThread *thread2 = checkMixerThread_l(output2);
if (thread1 == NULL || thread2 == NULL) {
ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1,
output2);
return AUDIO_IO_HANDLE_NONE;
}
audio_io_handle_t id = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT);
DuplicatingThread *thread = new DuplicatingThread(this, thread1, id, mSystemReady);
thread->addOutputTrack(thread2);
mPlaybackThreads.add(id, thread);
// notify client processes of the new output creation
thread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
return id;
}
status_t AudioFlinger::closeOutput(audio_io_handle_t output)
{
return closeOutput_nonvirtual(output);
}
status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output)
{
// keep strong reference on the playback thread so that
// it is not destroyed while exit() is executed
sp<PlaybackThread> thread;
{
Mutex::Autolock _l(mLock);
thread = checkPlaybackThread_l(output);
if (thread == NULL) {
return BAD_VALUE;
}
ALOGV("closeOutput() %d", output);
if (thread->type() == ThreadBase::MIXER) {
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
if (mPlaybackThreads.valueAt(i)->isDuplicating()) {
DuplicatingThread *dupThread =
(DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
dupThread->removeOutputTrack((MixerThread *)thread.get());
}
}
}
mPlaybackThreads.removeItem(output);
// save all effects to the default thread
if (mPlaybackThreads.size()) {
PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
if (dstThread != NULL) {
// audioflinger lock is held here so the acquisition order of thread locks does not
// matter
Mutex::Autolock _dl(dstThread->mLock);
Mutex::Autolock _sl(thread->mLock);
Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l();
for (size_t i = 0; i < effectChains.size(); i ++) {
moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true);
}
}
}
const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor();
ioDesc->mIoHandle = output;
ioConfigChanged(AUDIO_OUTPUT_CLOSED, ioDesc);
}
thread->exit();
// The thread entity (active unit of execution) is no longer running here,
// but the ThreadBase container still exists.
if (!thread->isDuplicating()) {
closeOutputFinish(thread);
}
return NO_ERROR;
}
void AudioFlinger::closeOutputFinish(const sp<PlaybackThread>& thread)
{
AudioStreamOut *out = thread->clearOutput();
ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
// from now on thread->mOutput is NULL
out->hwDev()->close_output_stream(out->hwDev(), out->stream);
delete out;
}
void AudioFlinger::closeOutputInternal_l(const sp<PlaybackThread>& thread)
{
mPlaybackThreads.removeItem(thread->mId);
thread->exit();
closeOutputFinish(thread);
}
status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
return BAD_VALUE;
}
ALOGV("suspendOutput() %d", output);
thread->suspend();
return NO_ERROR;
}
status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
return BAD_VALUE;
}
ALOGV("restoreOutput() %d", output);
thread->restore();
return NO_ERROR;
}
status_t AudioFlinger::openInput(audio_module_handle_t module,
audio_io_handle_t *input,
audio_config_t *config,
audio_devices_t *devices,
const String8& address,
audio_source_t source,
audio_input_flags_t flags)
{
Mutex::Autolock _l(mLock);
if (*devices == AUDIO_DEVICE_NONE) {
return BAD_VALUE;
}
sp<RecordThread> thread = openInput_l(module, input, config, *devices, address, source, flags);
if (thread != 0) {
// notify client processes of the new input creation
thread->ioConfigChanged(AUDIO_INPUT_OPENED);
return NO_ERROR;
}
return NO_INIT;
}
sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module,
audio_io_handle_t *input,
audio_config_t *config,
audio_devices_t devices,
const String8& address,
audio_source_t source,
audio_input_flags_t flags)
{
AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices);
if (inHwDev == NULL) {
*input = AUDIO_IO_HANDLE_NONE;
return 0;
}
// Audio Policy can request a specific handle for hardware hotword.
// The goal here is not to re-open an already opened input.
// It is to use a pre-assigned I/O handle.
if (*input == AUDIO_IO_HANDLE_NONE) {
*input = nextUniqueId(AUDIO_UNIQUE_ID_USE_INPUT);
} else if (audio_unique_id_get_use(*input) != AUDIO_UNIQUE_ID_USE_INPUT) {
ALOGE("openInput_l() requested input handle %d is invalid", *input);
return 0;
} else if (mRecordThreads.indexOfKey(*input) >= 0) {
// This should not happen in a transient state with current design.
ALOGE("openInput_l() requested input handle %d is already assigned", *input);
return 0;
}
audio_config_t halconfig = *config;
audio_hw_device_t *inHwHal = inHwDev->hwDevice();
audio_stream_in_t *inStream = NULL;
status_t status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig,
&inStream, flags, address.string(), source);
ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d"
", Format %#x, Channels %x, flags %#x, status %d addr %s",
inStream,
halconfig.sample_rate,
halconfig.format,
halconfig.channel_mask,
flags,
status, address.string());
// If the input could not be opened with the requested parameters and we can handle the
// conversion internally, try to open again with the proposed parameters.
if (status == BAD_VALUE &&
audio_is_linear_pcm(config->format) &&
audio_is_linear_pcm(halconfig.format) &&
(halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) &&
(audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_8) &&
(audio_channel_count_from_in_mask(config->channel_mask) <= FCC_8)) {
// FIXME describe the change proposed by HAL (save old values so we can log them here)
ALOGV("openInput_l() reopening with proposed sampling rate and channel mask");
inStream = NULL;
status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig,
&inStream, flags, address.string(), source);
// FIXME log this new status; HAL should not propose any further changes
}
if (status == NO_ERROR && inStream != NULL) {
#ifdef TEE_SINK
// Try to re-use most recently used Pipe to archive a copy of input for dumpsys,
// or (re-)create if current Pipe is idle and does not match the new format
sp<NBAIO_Sink> teeSink;
enum {
TEE_SINK_NO, // don't copy input
TEE_SINK_NEW, // copy input using a new pipe
TEE_SINK_OLD, // copy input using an existing pipe
} kind;
NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate,
audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format);
if (!mTeeSinkInputEnabled) {
kind = TEE_SINK_NO;
} else if (!Format_isValid(format)) {
kind = TEE_SINK_NO;
} else if (mRecordTeeSink == 0) {
kind = TEE_SINK_NEW;
} else if (mRecordTeeSink->getStrongCount() != 1) {
kind = TEE_SINK_NO;
} else if (Format_isEqual(format, mRecordTeeSink->format())) {
kind = TEE_SINK_OLD;
} else {
kind = TEE_SINK_NEW;
}
switch (kind) {
case TEE_SINK_NEW: {
Pipe *pipe = new Pipe(mTeeSinkInputFrames, format);
size_t numCounterOffers = 0;
const NBAIO_Format offers[1] = {format};
ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
ALOG_ASSERT(index == 0);
PipeReader *pipeReader = new PipeReader(*pipe);
numCounterOffers = 0;
index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
ALOG_ASSERT(index == 0);
mRecordTeeSink = pipe;
mRecordTeeSource = pipeReader;
teeSink = pipe;
}
break;
case TEE_SINK_OLD:
teeSink = mRecordTeeSink;
break;
case TEE_SINK_NO:
default:
break;
}
#endif
AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream);
// Start record thread
// RecordThread requires both input and output device indication to forward to audio
// pre processing modules
sp<RecordThread> thread = new RecordThread(this,
inputStream,
*input,
primaryOutputDevice_l(),
devices,
mSystemReady
#ifdef TEE_SINK
, teeSink
#endif
);
mRecordThreads.add(*input, thread);
ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get());
return thread;
}
*input = AUDIO_IO_HANDLE_NONE;
return 0;
}
status_t AudioFlinger::closeInput(audio_io_handle_t input)
{
return closeInput_nonvirtual(input);
}
status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input)
{
// keep strong reference on the record thread so that
// it is not destroyed while exit() is executed
sp<RecordThread> thread;
{
Mutex::Autolock _l(mLock);
thread = checkRecordThread_l(input);
if (thread == 0) {
return BAD_VALUE;
}
ALOGV("closeInput() %d", input);
// If we still have effect chains, it means that a client still holds a handle
// on at least one effect. We must either move the chain to an existing thread with the
// same session ID or put it aside in case a new record thread is opened for a
// new capture on the same session
sp<EffectChain> chain;
{
Mutex::Autolock _sl(thread->mLock);
Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l();
// Note: maximum one chain per record thread
if (effectChains.size() != 0) {
chain = effectChains[0];
}
}
if (chain != 0) {
// first check if a record thread is already opened with a client on the same session.
// This should only happen in case of overlap between one thread tear down and the
// creation of its replacement
size_t i;
for (i = 0; i < mRecordThreads.size(); i++) {
sp<RecordThread> t = mRecordThreads.valueAt(i);
if (t == thread) {
continue;
}
if (t->hasAudioSession(chain->sessionId()) != 0) {
Mutex::Autolock _l(t->mLock);
ALOGV("closeInput() found thread %d for effect session %d",
t->id(), chain->sessionId());
t->addEffectChain_l(chain);
break;
}
}
// put the chain aside if we could not find a record thread with the same session id.
if (i == mRecordThreads.size()) {
putOrphanEffectChain_l(chain);
}
}
const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor();
ioDesc->mIoHandle = input;
ioConfigChanged(AUDIO_INPUT_CLOSED, ioDesc);
mRecordThreads.removeItem(input);
}
// FIXME: calling thread->exit() without mLock held should not be needed anymore now that
// we have a different lock for notification client
closeInputFinish(thread);
return NO_ERROR;
}
void AudioFlinger::closeInputFinish(const sp<RecordThread>& thread)
{
thread->exit();
AudioStreamIn *in = thread->clearInput();
ALOG_ASSERT(in != NULL, "in shouldn't be NULL");
// from now on thread->mInput is NULL
in->hwDev()->close_input_stream(in->hwDev(), in->stream);
delete in;
}
void AudioFlinger::closeInputInternal_l(const sp<RecordThread>& thread)
{
mRecordThreads.removeItem(thread->mId);
closeInputFinish(thread);
}
status_t AudioFlinger::invalidateStream(audio_stream_type_t stream)
{
Mutex::Autolock _l(mLock);
ALOGV("invalidateStream() stream %d", stream);
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
thread->invalidateTracks(stream);
}
return NO_ERROR;
}
audio_unique_id_t AudioFlinger::newAudioUniqueId(audio_unique_id_use_t use)
{
// This is a binder API, so a malicious client could pass in a bad parameter.
// Check for that before calling the internal API nextUniqueId().
if ((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX) {
ALOGE("newAudioUniqueId invalid use %d", use);
return AUDIO_UNIQUE_ID_ALLOCATE;
}
return nextUniqueId(use);
}
void AudioFlinger::acquireAudioSessionId(audio_session_t audioSession, pid_t pid)
{
Mutex::Autolock _l(mLock);
pid_t caller = IPCThreadState::self()->getCallingPid();
ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid);
if (pid != -1 && (caller == getpid_cached)) {
caller = pid;
}
{
Mutex::Autolock _cl(mClientLock);
// Ignore requests received from processes not known as notification client. The request
// is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be
// called from a different pid leaving a stale session reference. Also we don't know how
// to clear this reference if the client process dies.
if (mNotificationClients.indexOfKey(caller) < 0) {
ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession);
return;
}
}
size_t num = mAudioSessionRefs.size();
for (size_t i = 0; i< num; i++) {
AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
if (ref->mSessionid == audioSession && ref->mPid == caller) {
ref->mCnt++;
ALOGV(" incremented refcount to %d", ref->mCnt);
return;
}
}
mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
ALOGV(" added new entry for %d", audioSession);
}
void AudioFlinger::releaseAudioSessionId(audio_session_t audioSession, pid_t pid)
{
Mutex::Autolock _l(mLock);
pid_t caller = IPCThreadState::self()->getCallingPid();
ALOGV("releasing %d from %d for %d", audioSession, caller, pid);
if (pid != -1 && (caller == getpid_cached)) {
caller = pid;
}
size_t num = mAudioSessionRefs.size();
for (size_t i = 0; i< num; i++) {
AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
if (ref->mSessionid == audioSession && ref->mPid == caller) {
ref->mCnt--;
ALOGV(" decremented refcount to %d", ref->mCnt);
if (ref->mCnt == 0) {
mAudioSessionRefs.removeAt(i);
delete ref;
purgeStaleEffects_l();
}
return;
}
}
// If the caller is mediaserver it is likely that the session being released was acquired
// on behalf of a process not in notification clients and we ignore the warning.
ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller);
}
void AudioFlinger::purgeStaleEffects_l() {
ALOGV("purging stale effects");
Vector< sp<EffectChain> > chains;
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
for (size_t j = 0; j < t->mEffectChains.size(); j++) {
sp<EffectChain> ec = t->mEffectChains[j];
if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
chains.push(ec);
}
}
}
for (size_t i = 0; i < mRecordThreads.size(); i++) {
sp<RecordThread> t = mRecordThreads.valueAt(i);
for (size_t j = 0; j < t->mEffectChains.size(); j++) {
sp<EffectChain> ec = t->mEffectChains[j];
chains.push(ec);
}
}
for (size_t i = 0; i < chains.size(); i++) {
sp<EffectChain> ec = chains[i];
int sessionid = ec->sessionId();
sp<ThreadBase> t = ec->mThread.promote();
if (t == 0) {
continue;
}
size_t numsessionrefs = mAudioSessionRefs.size();
bool found = false;
for (size_t k = 0; k < numsessionrefs; k++) {
AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
if (ref->mSessionid == sessionid) {
ALOGV(" session %d still exists for %d with %d refs",
sessionid, ref->mPid, ref->mCnt);
found = true;
break;
}
}
if (!found) {
Mutex::Autolock _l(t->mLock);
// remove all effects from the chain
while (ec->mEffects.size()) {
sp<EffectModule> effect = ec->mEffects[0];
effect->unPin();
t->removeEffect_l(effect);
if (effect->purgeHandles()) {
t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
}
AudioSystem::unregisterEffect(effect->id());
}
}
}
return;
}
// checkThread_l() must be called with AudioFlinger::mLock held
AudioFlinger::ThreadBase *AudioFlinger::checkThread_l(audio_io_handle_t ioHandle) const
{
ThreadBase *thread = NULL;
switch (audio_unique_id_get_use(ioHandle)) {
case AUDIO_UNIQUE_ID_USE_OUTPUT:
thread = checkPlaybackThread_l(ioHandle);
break;
case AUDIO_UNIQUE_ID_USE_INPUT:
thread = checkRecordThread_l(ioHandle);
break;
default:
break;
}
return thread;
}
// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
{
return mPlaybackThreads.valueFor(output).get();
}
// checkMixerThread_l() must be called with AudioFlinger::mLock held
AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
{
PlaybackThread *thread = checkPlaybackThread_l(output);
return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
}
// checkRecordThread_l() must be called with AudioFlinger::mLock held
AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
{
return mRecordThreads.valueFor(input).get();
}
audio_unique_id_t AudioFlinger::nextUniqueId(audio_unique_id_use_t use)
{
// This is the internal API, so it is OK to assert on bad parameter.
LOG_ALWAYS_FATAL_IF((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX);
const int maxRetries = use == AUDIO_UNIQUE_ID_USE_SESSION ? 3 : 1;
for (int retry = 0; retry < maxRetries; retry++) {
// The cast allows wraparound from max positive to min negative instead of abort
uint32_t base = (uint32_t) atomic_fetch_add_explicit(&mNextUniqueIds[use],
(uint_fast32_t) AUDIO_UNIQUE_ID_USE_MAX, memory_order_acq_rel);
ALOG_ASSERT(audio_unique_id_get_use(base) == AUDIO_UNIQUE_ID_USE_UNSPECIFIED);
// allow wrap by skipping 0 and -1 for session ids
if (!(base == 0 || base == (~0u & ~AUDIO_UNIQUE_ID_USE_MASK))) {
ALOGW_IF(retry != 0, "unique ID overflow for use %d", use);
return (audio_unique_id_t) (base | use);
}
}
// We have no way of recovering from wraparound
LOG_ALWAYS_FATAL("unique ID overflow for use %d", use);
// TODO Use a floor after wraparound. This may need a mutex.
}
AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const
{
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
if(thread->isDuplicating()) {
continue;
}
AudioStreamOut *output = thread->getOutput();
if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) {
return thread;
}
}
return NULL;
}
audio_devices_t AudioFlinger::primaryOutputDevice_l() const
{
PlaybackThread *thread = primaryPlaybackThread_l();
if (thread == NULL) {
return 0;
}
return thread->outDevice();
}
sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
audio_session_t triggerSession,
audio_session_t listenerSession,
sync_event_callback_t callBack,
const wp<RefBase>& cookie)
{
Mutex::Autolock _l(mLock);
sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie);
status_t playStatus = NAME_NOT_FOUND;
status_t recStatus = NAME_NOT_FOUND;
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event);
if (playStatus == NO_ERROR) {
return event;
}
}
for (size_t i = 0; i < mRecordThreads.size(); i++) {
recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event);
if (recStatus == NO_ERROR) {
return event;
}
}
if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) {
mPendingSyncEvents.add(event);
} else {
ALOGV("createSyncEvent() invalid event %d", event->type());
event.clear();
}
return event;
}
// ----------------------------------------------------------------------------
// Effect management
// ----------------------------------------------------------------------------
status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
{
Mutex::Autolock _l(mLock);
return EffectQueryNumberEffects(numEffects);
}
status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
{
Mutex::Autolock _l(mLock);
return EffectQueryEffect(index, descriptor);
}
status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
effect_descriptor_t *descriptor) const
{
Mutex::Autolock _l(mLock);
return EffectGetDescriptor(pUuid, descriptor);
}
sp<IEffect> AudioFlinger::createEffect(
effect_descriptor_t *pDesc,
const sp<IEffectClient>& effectClient,
int32_t priority,
audio_io_handle_t io,
audio_session_t sessionId,
const String16& opPackageName,
status_t *status,
int *id,
int *enabled)
{
status_t lStatus = NO_ERROR;
sp<EffectHandle> handle;
effect_descriptor_t desc;
pid_t pid = IPCThreadState::self()->getCallingPid();
ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d",
pid, effectClient.get(), priority, sessionId, io);
if (pDesc == NULL) {
lStatus = BAD_VALUE;
goto Exit;
}
// check audio settings permission for global effects
if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) {
lStatus = PERMISSION_DENIED;
goto Exit;
}
// Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
// that can only be created by audio policy manager (running in same process)
if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) {
lStatus = PERMISSION_DENIED;
goto Exit;
}
{
if (!EffectIsNullUuid(&pDesc->uuid)) {
// if uuid is specified, request effect descriptor
lStatus = EffectGetDescriptor(&pDesc->uuid, &desc);
if (lStatus < 0) {
ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
goto Exit;
}
} else {
// if uuid is not specified, look for an available implementation
// of the required type in effect factory
if (EffectIsNullUuid(&pDesc->type)) {
ALOGW("createEffect() no effect type");
lStatus = BAD_VALUE;
goto Exit;
}
uint32_t numEffects = 0;
effect_descriptor_t d;
d.flags = 0; // prevent compiler warning
bool found = false;
lStatus = EffectQueryNumberEffects(&numEffects);
if (lStatus < 0) {
ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
goto Exit;
}
for (uint32_t i = 0; i < numEffects; i++) {
lStatus = EffectQueryEffect(i, &desc);
if (lStatus < 0) {
ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
continue;
}
if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
// If matching type found save effect descriptor. If the session is
// 0 and the effect is not auxiliary, continue enumeration in case
// an auxiliary version of this effect type is available
found = true;
d = desc;
if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
(desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
break;
}
}
}
if (!found) {
lStatus = BAD_VALUE;
ALOGW("createEffect() effect not found");
goto Exit;
}
// For same effect type, chose auxiliary version over insert version if
// connect to output mix (Compliance to OpenSL ES)
if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
(d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
desc = d;
}
}
// Do not allow auxiliary effects on a session different from 0 (output mix)
if (sessionId != AUDIO_SESSION_OUTPUT_MIX &&
(desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
lStatus = INVALID_OPERATION;
goto Exit;
}
// check recording permission for visualizer
if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
!recordingAllowed(opPackageName, pid, IPCThreadState::self()->getCallingUid())) {
lStatus = PERMISSION_DENIED;
goto Exit;
}
// return effect descriptor
*pDesc = desc;
if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) {
// if the output returned by getOutputForEffect() is removed before we lock the
// mutex below, the call to checkPlaybackThread_l(io) below will detect it
// and we will exit safely
io = AudioSystem::getOutputForEffect(&desc);
ALOGV("createEffect got output %d", io);
}
Mutex::Autolock _l(mLock);
// If output is not specified try to find a matching audio session ID in one of the
// output threads.
// If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
// because of code checking output when entering the function.
// Note: io is never 0 when creating an effect on an input
if (io == AUDIO_IO_HANDLE_NONE) {
if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
// output must be specified by AudioPolicyManager when using session
// AUDIO_SESSION_OUTPUT_STAGE
lStatus = BAD_VALUE;
goto Exit;
}
// look for the thread where the specified audio session is present
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
io = mPlaybackThreads.keyAt(i);
break;
}
}
if (io == 0) {
for (size_t i = 0; i < mRecordThreads.size(); i++) {
if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
io = mRecordThreads.keyAt(i);
break;
}
}
}
// If no output thread contains the requested session ID, default to
// first output. The effect chain will be moved to the correct output
// thread when a track with the same session ID is created
if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) {
io = mPlaybackThreads.keyAt(0);
}
ALOGV("createEffect() got io %d for effect %s", io, desc.name);
}
ThreadBase *thread = checkRecordThread_l(io);
if (thread == NULL) {
thread = checkPlaybackThread_l(io);
if (thread == NULL) {
ALOGE("createEffect() unknown output thread");
lStatus = BAD_VALUE;
goto Exit;
}
} else {
// Check if one effect chain was awaiting for an effect to be created on this
// session and used it instead of creating a new one.
sp<EffectChain> chain = getOrphanEffectChain_l(sessionId);
if (chain != 0) {
Mutex::Autolock _l(thread->mLock);
thread->addEffectChain_l(chain);
}
}
sp<Client> client = registerPid(pid);
// create effect on selected output thread
handle = thread->createEffect_l(client, effectClient, priority, sessionId,
&desc, enabled, &lStatus);
if (handle != 0 && id != NULL) {
*id = handle->id();
}
if (handle == 0) {
// remove local strong reference to Client with mClientLock held
Mutex::Autolock _cl(mClientLock);
client.clear();
}
}
Exit:
*status = lStatus;
return handle;
}
status_t AudioFlinger::moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput,
audio_io_handle_t dstOutput)
{
ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
sessionId, srcOutput, dstOutput);
Mutex::Autolock _l(mLock);
if (srcOutput == dstOutput) {
ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
return NO_ERROR;
}
PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
if (srcThread == NULL) {
ALOGW("moveEffects() bad srcOutput %d", srcOutput);
return BAD_VALUE;
}
PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
if (dstThread == NULL) {
ALOGW("moveEffects() bad dstOutput %d", dstOutput);
return BAD_VALUE;
}
Mutex::Autolock _dl(dstThread->mLock);
Mutex::Autolock _sl(srcThread->mLock);
return moveEffectChain_l(sessionId, srcThread, dstThread, false);
}
// moveEffectChain_l must be called with both srcThread and dstThread mLocks held
status_t AudioFlinger::moveEffectChain_l(audio_session_t sessionId,
AudioFlinger::PlaybackThread *srcThread,
AudioFlinger::PlaybackThread *dstThread,
bool reRegister)
{
ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
sessionId, srcThread, dstThread);
sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId);
if (chain == 0) {
ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p",
sessionId, srcThread);
return INVALID_OPERATION;
}
// Check whether the destination thread has a channel count of FCC_2, which is
// currently required for (most) effects. Prevent moving the effect chain here rather
// than disabling the addEffect_l() call in dstThread below.
if ((dstThread->type() == ThreadBase::MIXER || dstThread->isDuplicating()) &&
dstThread->mChannelCount != FCC_2) {
ALOGW("moveEffectChain_l() effect chain failed because"
" destination thread %p channel count(%u) != %u",
dstThread, dstThread->mChannelCount, FCC_2);
return INVALID_OPERATION;
}
// remove chain first. This is useful only if reconfiguring effect chain on same output thread,
// so that a new chain is created with correct parameters when first effect is added. This is
// otherwise unnecessary as removeEffect_l() will remove the chain when last effect is
// removed.
srcThread->removeEffectChain_l(chain);
// transfer all effects one by one so that new effect chain is created on new thread with
// correct buffer sizes and audio parameters and effect engines reconfigured accordingly
sp<EffectChain> dstChain;
uint32_t strategy = 0; // prevent compiler warning
sp<EffectModule> effect = chain->getEffectFromId_l(0);
Vector< sp<EffectModule> > removed;
status_t status = NO_ERROR;
while (effect != 0) {
srcThread->removeEffect_l(effect);
removed.add(effect);
status = dstThread->addEffect_l(effect);
if (status != NO_ERROR) {
break;
}
// removeEffect_l() has stopped the effect if it was active so it must be restarted
if (effect->state() == EffectModule::ACTIVE ||
effect->state() == EffectModule::STOPPING) {
effect->start();
}
// if the move request is not received from audio policy manager, the effect must be
// re-registered with the new strategy and output
if (dstChain == 0) {
dstChain = effect->chain().promote();
if (dstChain == 0) {
ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
status = NO_INIT;
break;
}
strategy = dstChain->strategy();
}
if (reRegister) {
AudioSystem::unregisterEffect(effect->id());
AudioSystem::registerEffect(&effect->desc(),
dstThread->id(),
strategy,
sessionId,
effect->id());
AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
}
effect = chain->getEffectFromId_l(0);
}
if (status != NO_ERROR) {
for (size_t i = 0; i < removed.size(); i++) {
srcThread->addEffect_l(removed[i]);
if (dstChain != 0 && reRegister) {
AudioSystem::unregisterEffect(removed[i]->id());
AudioSystem::registerEffect(&removed[i]->desc(),
srcThread->id(),
strategy,
sessionId,
removed[i]->id());
AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
}
}
}
return status;
}
bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l()
{
if (mGlobalEffectEnableTime != 0 &&
((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) {
return true;
}
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
sp<EffectChain> ec =
mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
if (ec != 0 && ec->isNonOffloadableEnabled()) {
return true;
}
}
return false;
}
void AudioFlinger::onNonOffloadableGlobalEffectEnable()
{
Mutex::Autolock _l(mLock);
mGlobalEffectEnableTime = systemTime();
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
if (t->mType == ThreadBase::OFFLOAD) {
t->invalidateTracks(AUDIO_STREAM_MUSIC);
}
}
}
status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain)
{
audio_session_t session = chain->sessionId();
ssize_t index = mOrphanEffectChains.indexOfKey(session);
ALOGV("putOrphanEffectChain_l session %d index %zd", session, index);
if (index >= 0) {
ALOGW("putOrphanEffectChain_l chain for session %d already present", session);
return ALREADY_EXISTS;
}
mOrphanEffectChains.add(session, chain);
return NO_ERROR;
}
sp<AudioFlinger::EffectChain> AudioFlinger::getOrphanEffectChain_l(audio_session_t session)
{
sp<EffectChain> chain;
ssize_t index = mOrphanEffectChains.indexOfKey(session);
ALOGV("getOrphanEffectChain_l session %d index %zd", session, index);
if (index >= 0) {
chain = mOrphanEffectChains.valueAt(index);
mOrphanEffectChains.removeItemsAt(index);
}
return chain;
}
bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>& effect)
{
Mutex::Autolock _l(mLock);
audio_session_t session = effect->sessionId();
ssize_t index = mOrphanEffectChains.indexOfKey(session);
ALOGV("updateOrphanEffectChains session %d index %zd", session, index);
if (index >= 0) {
sp<EffectChain> chain = mOrphanEffectChains.valueAt(index);
if (chain->removeEffect_l(effect) == 0) {
ALOGV("updateOrphanEffectChains removing effect chain at index %zd", index);
mOrphanEffectChains.removeItemsAt(index);
}
return true;
}
return false;
}
struct Entry {
#define TEE_MAX_FILENAME 32 // %Y%m%d%H%M%S_%d.wav = 4+2+2+2+2+2+1+1+4+1 = 21
char mFileName[TEE_MAX_FILENAME];
};
int comparEntry(const void *p1, const void *p2)
{
return strcmp(((const Entry *) p1)->mFileName, ((const Entry *) p2)->mFileName);
}
#ifdef TEE_SINK
void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id)
{
NBAIO_Source *teeSource = source.get();
if (teeSource != NULL) {
// .wav rotation
// There is a benign race condition if 2 threads call this simultaneously.
// They would both traverse the directory, but the result would simply be
// failures at unlink() which are ignored. It's also unlikely since
// normally dumpsys is only done by bugreport or from the command line.
char teePath[32+256];
strcpy(teePath, "/data/misc/audioserver");
size_t teePathLen = strlen(teePath);
DIR *dir = opendir(teePath);
teePath[teePathLen++] = '/';
if (dir != NULL) {
#define TEE_MAX_SORT 20 // number of entries to sort
#define TEE_MAX_KEEP 10 // number of entries to keep
struct Entry entries[TEE_MAX_SORT];
size_t entryCount = 0;
while (entryCount < TEE_MAX_SORT) {
struct dirent de;
struct dirent *result = NULL;
int rc = readdir_r(dir, &de, &result);
if (rc != 0) {
ALOGW("readdir_r failed %d", rc);
break;
}
if (result == NULL) {
break;
}
if (result != &de) {
ALOGW("readdir_r returned unexpected result %p != %p", result, &de);
break;
}
// ignore non .wav file entries
size_t nameLen = strlen(de.d_name);
if (nameLen <= 4 || nameLen >= TEE_MAX_FILENAME ||
strcmp(&de.d_name[nameLen - 4], ".wav")) {
continue;
}
strcpy(entries[entryCount++].mFileName, de.d_name);
}
(void) closedir(dir);
if (entryCount > TEE_MAX_KEEP) {
qsort(entries, entryCount, sizeof(Entry), comparEntry);
for (size_t i = 0; i < entryCount - TEE_MAX_KEEP; ++i) {
strcpy(&teePath[teePathLen], entries[i].mFileName);
(void) unlink(teePath);
}
}
} else {
if (fd >= 0) {
dprintf(fd, "unable to rotate tees in %.*s: %s\n", teePathLen, teePath,
strerror(errno));
}
}
char teeTime[16];
struct timeval tv;
gettimeofday(&tv, NULL);
struct tm tm;
localtime_r(&tv.tv_sec, &tm);
strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm);
snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id);
// if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd
int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR);
if (teeFd >= 0) {
// FIXME use libsndfile
char wavHeader[44];
memcpy(wavHeader,
"RIFF\0\0\0\0WAVEfmt \20\0\0\0\1\0\2\0\104\254\0\0\0\0\0\0\4\0\20\0data\0\0\0\0",
sizeof(wavHeader));
NBAIO_Format format = teeSource->format();
unsigned channelCount = Format_channelCount(format);
uint32_t sampleRate = Format_sampleRate(format);
size_t frameSize = Format_frameSize(format);
wavHeader[22] = channelCount; // number of channels
wavHeader[24] = sampleRate; // sample rate
wavHeader[25] = sampleRate >> 8;
wavHeader[32] = frameSize; // block alignment
wavHeader[33] = frameSize >> 8;
write(teeFd, wavHeader, sizeof(wavHeader));
size_t total = 0;
bool firstRead = true;
#define TEE_SINK_READ 1024 // frames per I/O operation
void *buffer = malloc(TEE_SINK_READ * frameSize);
for (;;) {
size_t count = TEE_SINK_READ;
ssize_t actual = teeSource->read(buffer, count);
bool wasFirstRead = firstRead;
firstRead = false;
if (actual <= 0) {
if (actual == (ssize_t) OVERRUN && wasFirstRead) {
continue;
}
break;
}
ALOG_ASSERT(actual <= (ssize_t)count);
write(teeFd, buffer, actual * frameSize);
total += actual;
}
free(buffer);
lseek(teeFd, (off_t) 4, SEEK_SET);
uint32_t temp = 44 + total * frameSize - 8;
// FIXME not big-endian safe
write(teeFd, &temp, sizeof(temp));
lseek(teeFd, (off_t) 40, SEEK_SET);
temp = total * frameSize;
// FIXME not big-endian safe
write(teeFd, &temp, sizeof(temp));
close(teeFd);
if (fd >= 0) {
dprintf(fd, "tee copied to %s\n", teePath);
}
} else {
if (fd >= 0) {
dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno));
}
}
}
}
#endif
// ----------------------------------------------------------------------------
status_t AudioFlinger::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
return BnAudioFlinger::onTransact(code, data, reply, flags);
}
} // namespace android