Merge changes Iec1011a9,I64249876,Icb348a4a,Ibd46b7de into udc-qpr-dev-plus-aosp
* changes:
AudioFlinger: Do not dereference a nullptr for a reference
AudioFlinger: Extract inner Thread classes
AudioFlinger: Add MmapThread interfaces
AudioFlinger: Add more Thread interfaces
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 280281e..b49c284 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -433,8 +433,8 @@
for (const auto& [trackId, secondaryOutputs] : trackSecondaryOutputs) {
size_t i = 0;
for (; i < mPlaybackThreads.size(); ++i) {
- PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
- Mutex::Autolock _tl(thread->mLock);
+ IAfPlaybackThread* thread = mPlaybackThreads.valueAt(i).get();
+ Mutex::Autolock _tl(thread->mutex());
sp<IAfTrack> track = thread->getTrackById_l(trackId);
if (track != nullptr) {
ALOGD("%s trackId: %u", __func__, trackId);
@@ -678,9 +678,9 @@
// at this stage, a MmapThread was created when openOutput() or openInput() was called by
// audio policy manager and we can retrieve it
- sp<MmapThread> thread = mMmapThreads.valueFor(io);
+ const sp<IAfMmapThread> thread = mMmapThreads.valueFor(io);
if (thread != 0) {
- interface = new MmapThreadHandle(thread);
+ interface = IAfMmapThread::createMmapStreamInterfaceAdapter(thread);
thread->configure(&localAttr, streamType, actualSessionId, callback, *deviceId, portId);
*handle = portId;
*sessionId = actualSessionId;
@@ -1191,7 +1191,7 @@
{
Mutex::Autolock _l(mLock);
- PlaybackThread *thread = checkPlaybackThread_l(output.outputId);
+ IAfPlaybackThread* thread = checkPlaybackThread_l(output.outputId);
if (thread == NULL) {
ALOGE("no playback thread found for output handle %d", output.outputId);
lStatus = BAD_VALUE;
@@ -1200,14 +1200,14 @@
client = registerPid(clientPid);
- PlaybackThread *effectThread = NULL;
+ IAfPlaybackThread* effectThread = nullptr;
// 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);
+ sp<IAfPlaybackThread> t = mPlaybackThreads.valueAt(i);
if (mPlaybackThreads.keyAt(i) != output.outputId) {
uint32_t sessions = t->hasAudioSession(sessionId);
- if (sessions & ThreadBase::EFFECT_SESSION) {
+ if (sessions & IAfThreadBase::EFFECT_SESSION) {
effectThread = t.get();
break;
}
@@ -1242,7 +1242,7 @@
if (lStatus == NO_ERROR) {
// no risk of deadlock because AudioFlinger::mLock is held
- Mutex::Autolock _dl(thread->mLock);
+ Mutex::Autolock _dl(thread->mutex());
// Connect secondary outputs. Failure on a secondary output must not imped the primary
// Any secondary output setup failure will lead to a desync between the AP and AF until
// the track is destroyed.
@@ -1250,7 +1250,7 @@
// move effect chain to this output thread if an effect on same session was waiting
// for a track to be created
if (effectThread != nullptr) {
- Mutex::Autolock _sl(effectThread->mLock);
+ Mutex::Autolock _sl(effectThread->mutex());
if (moveEffectChain_l(sessionId, effectThread, thread) == NO_ERROR) {
effectThreadId = thread->id();
effectIds = thread->getEffectIds_l(sessionId);
@@ -1310,7 +1310,7 @@
uint32_t AudioFlinger::sampleRate(audio_io_handle_t ioHandle) const
{
Mutex::Autolock _l(mLock);
- ThreadBase *thread = checkThread_l(ioHandle);
+ IAfThreadBase* const thread = checkThread_l(ioHandle);
if (thread == NULL) {
ALOGW("sampleRate() unknown thread %d", ioHandle);
return 0;
@@ -1321,7 +1321,7 @@
audio_format_t AudioFlinger::format(audio_io_handle_t output) const
{
Mutex::Autolock _l(mLock);
- PlaybackThread *thread = checkPlaybackThread_l(output);
+ IAfPlaybackThread* const thread = checkPlaybackThread_l(output);
if (thread == NULL) {
ALOGW("format() unknown thread %d", output);
return AUDIO_FORMAT_INVALID;
@@ -1332,7 +1332,7 @@
size_t AudioFlinger::frameCount(audio_io_handle_t ioHandle) const
{
Mutex::Autolock _l(mLock);
- ThreadBase *thread = checkThread_l(ioHandle);
+ IAfThreadBase* const thread = checkThread_l(ioHandle);
if (thread == NULL) {
ALOGW("frameCount() unknown thread %d", ioHandle);
return 0;
@@ -1345,7 +1345,7 @@
size_t AudioFlinger::frameCountHAL(audio_io_handle_t ioHandle) const
{
Mutex::Autolock _l(mLock);
- ThreadBase *thread = checkThread_l(ioHandle);
+ IAfThreadBase* const thread = checkThread_l(ioHandle);
if (thread == NULL) {
ALOGW("frameCountHAL() unknown thread %d", ioHandle);
return 0;
@@ -1356,7 +1356,7 @@
uint32_t AudioFlinger::latency(audio_io_handle_t output) const
{
Mutex::Autolock _l(mLock);
- PlaybackThread *thread = checkPlaybackThread_l(output);
+ IAfPlaybackThread* const thread = checkPlaybackThread_l(output);
if (thread == NULL) {
ALOGW("latency(): no playback thread found for output handle %d", output);
return 0;
@@ -1676,7 +1676,7 @@
return BAD_VALUE;
}
AutoMutex lock(mLock);
- PlaybackThread *thread = checkPlaybackThread_l(output);
+ IAfPlaybackThread* const thread = checkPlaybackThread_l(output);
if (thread == nullptr) {
return BAD_VALUE;
}
@@ -1689,7 +1689,7 @@
return BAD_VALUE;
}
AutoMutex lock(mLock);
- PlaybackThread *thread = checkPlaybackThread_l(output);
+ IAfPlaybackThread* const thread = checkPlaybackThread_l(output);
if (thread == nullptr) {
return BAD_VALUE;
}
@@ -1820,14 +1820,15 @@
// forwardAudioHwSyncToDownstreamPatches_l() must be called with AudioFlinger::mLock held
void AudioFlinger::forwardParametersToDownstreamPatches_l(
audio_io_handle_t upStream, const String8& keyValuePairs,
- const std::function<bool(const sp<PlaybackThread>&)>& useThread)
+ const std::function<bool(const sp<IAfPlaybackThread>&)>& useThread)
{
std::vector<PatchPanel::SoftwarePatch> swPatches;
if (mPatchPanel.getDownstreamSoftwarePatches(upStream, &swPatches) != OK) return;
ALOGV_IF(!swPatches.empty(), "%s found %zu downstream patches for stream ID %d",
__func__, swPatches.size(), upStream);
for (const auto& swPatch : swPatches) {
- sp<PlaybackThread> downStream = checkPlaybackThread_l(swPatch.getPlaybackThreadHandle());
+ const sp<IAfPlaybackThread> downStream =
+ checkPlaybackThread_l(swPatch.getPlaybackThreadHandle());
if (downStream != NULL && (useThread == nullptr || useThread(downStream))) {
downStream->setParameters(keyValuePairs);
}
@@ -1839,7 +1840,7 @@
const std::set<audio_io_handle_t>& streams)
{
for (const audio_io_handle_t stream : streams) {
- PlaybackThread *playbackThread = checkPlaybackThread_l(stream);
+ IAfPlaybackThread* const playbackThread = checkPlaybackThread_l(stream);
if (playbackThread == nullptr || !playbackThread->isMsdDevice()) {
continue;
}
@@ -1962,7 +1963,7 @@
// 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;
+ sp<IAfThreadBase> thread;
{
Mutex::Autolock _l(mLock);
thread = checkPlaybackThread_l(ioHandle);
@@ -2011,11 +2012,11 @@
return out_s8;
}
- ThreadBase *thread = (ThreadBase *)checkPlaybackThread_l(ioHandle);
+ IAfThreadBase* thread = checkPlaybackThread_l(ioHandle);
if (thread == NULL) {
- thread = (ThreadBase *)checkRecordThread_l(ioHandle);
+ thread = checkRecordThread_l(ioHandle);
if (thread == NULL) {
- thread = (ThreadBase *)checkMmapThread_l(ioHandle);
+ thread = checkMmapThread_l(ioHandle);
if (thread == NULL) {
return String8("");
}
@@ -2111,7 +2112,7 @@
{
Mutex::Autolock _l(mLock);
- RecordThread *recordThread = checkRecordThread_l(ioHandle);
+ IAfRecordThread* const recordThread = checkRecordThread_l(ioHandle);
if (recordThread != NULL) {
return recordThread->getInputFramesLost();
}
@@ -2151,7 +2152,7 @@
{
Mutex::Autolock _l(mLock);
- PlaybackThread *playbackThread = checkPlaybackThread_l(output);
+ IAfPlaybackThread* const playbackThread = checkPlaybackThread_l(output);
if (playbackThread != NULL) {
return playbackThread->getRenderPosition(halFrames, dspFrames);
}
@@ -2274,10 +2275,10 @@
}
// getEffectThread_l() must be called with AudioFlinger::mLock held
-sp<AudioFlinger::ThreadBase> AudioFlinger::getEffectThread_l(audio_session_t sessionId,
+sp<IAfThreadBase> AudioFlinger::getEffectThread_l(audio_session_t sessionId,
int effectId)
{
- sp<ThreadBase> thread;
+ sp<IAfThreadBase> thread;
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, effectId) != 0) {
@@ -2480,7 +2481,7 @@
{
Mutex::Autolock _l(mLock);
- RecordThread *thread = checkRecordThread_l(output.inputId);
+ IAfRecordThread* const thread = checkRecordThread_l(output.inputId);
if (thread == NULL) {
ALOGW("createRecord() checkRecordThread_l failed, input handle %d", output.inputId);
lStatus = FAILED_TRANSACTION;
@@ -2536,7 +2537,7 @@
// session and move it to this thread.
sp<IAfEffectChain> chain = getOrphanEffectChain_l(sessionId);
if (chain != 0) {
- Mutex::Autolock _l2(thread->mLock);
+ Mutex::Autolock _l2(thread->mutex());
thread->addEffectChain_l(chain);
}
break;
@@ -2738,14 +2739,14 @@
uint32_t AudioFlinger::getPrimaryOutputSamplingRate()
{
Mutex::Autolock _l(mLock);
- PlaybackThread *thread = fastPlaybackThread_l();
+ IAfPlaybackThread* const thread = fastPlaybackThread_l();
return thread != NULL ? thread->sampleRate() : 0;
}
size_t AudioFlinger::getPrimaryOutputFrameCount()
{
Mutex::Autolock _l(mLock);
- PlaybackThread *thread = fastPlaybackThread_l();
+ IAfPlaybackThread* const thread = fastPlaybackThread_l();
return thread != NULL ? thread->frameCountHAL() : 0;
}
@@ -2870,15 +2871,15 @@
mHwAvSyncIds.add(sessionId, value);
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
- sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i);
+ const sp<IAfPlaybackThread> thread = mPlaybackThreads.valueAt(i);
uint32_t sessions = thread->hasAudioSession(sessionId);
- if (sessions & ThreadBase::TRACK_SESSION) {
+ if (sessions & IAfThreadBase::TRACK_SESSION) {
AudioParameter param = AudioParameter();
param.addInt(String8(AudioParameter::keyStreamHwAvSync), value);
String8 keyValuePairs = param.toString();
thread->setParameters(keyValuePairs);
forwardParametersToDownstreamPatches_l(thread->id(), keyValuePairs,
- [](const sp<PlaybackThread>& thread) { return thread->usesHwAvSync(); });
+ [](const sp<IAfPlaybackThread>& thread) { return thread->usesHwAvSync(); });
break;
}
}
@@ -2897,15 +2898,15 @@
}
mSystemReady = true;
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
- ThreadBase *thread = (ThreadBase *)mPlaybackThreads.valueAt(i).get();
+ IAfThreadBase* const thread = mPlaybackThreads.valueAt(i).get();
thread->systemReady();
}
for (size_t i = 0; i < mRecordThreads.size(); i++) {
- ThreadBase *thread = (ThreadBase *)mRecordThreads.valueAt(i).get();
+ IAfThreadBase* const thread = mRecordThreads.valueAt(i).get();
thread->systemReady();
}
for (size_t i = 0; i < mMmapThreads.size(); i++) {
- ThreadBase *thread = (ThreadBase *)mMmapThreads.valueAt(i).get();
+ IAfThreadBase* const thread = mMmapThreads.valueAt(i).get();
thread->systemReady();
}
@@ -2957,7 +2958,8 @@
}
// setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId)
+void AudioFlinger::setAudioHwSyncForSession_l(
+ IAfPlaybackThread* const thread, audio_session_t sessionId)
{
ssize_t index = mHwAvSyncIds.indexOfKey(sessionId);
if (index >= 0) {
@@ -2968,7 +2970,7 @@
String8 keyValuePairs = param.toString();
thread->setParameters(keyValuePairs);
forwardParametersToDownstreamPatches_l(thread->id(), keyValuePairs,
- [](const sp<PlaybackThread>& thread) { return thread->usesHwAvSync(); });
+ [](const sp<IAfPlaybackThread>& thread) { return thread->usesHwAvSync(); });
}
}
@@ -2976,7 +2978,7 @@
// ----------------------------------------------------------------------------
-sp<AudioFlinger::ThreadBase> AudioFlinger::openOutput_l(audio_module_handle_t module,
+sp<IAfThreadBase> AudioFlinger::openOutput_l(audio_module_handle_t module,
audio_io_handle_t *output,
audio_config_t *halConfig,
audio_config_base_t *mixerConfig,
@@ -3034,37 +3036,39 @@
if (status == NO_ERROR) {
if (flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) {
- sp<MmapPlaybackThread> thread =
- new MmapPlaybackThread(this, *output, outHwDev, outputStream, mSystemReady);
+ const sp<IAfMmapPlaybackThread> thread = IAfMmapPlaybackThread::create(
+ this, *output, outHwDev, outputStream, mSystemReady);
mMmapThreads.add(*output, thread);
ALOGV("openOutput_l() created mmap playback thread: ID %d thread %p",
*output, thread.get());
return thread;
} else {
- sp<PlaybackThread> thread;
+ sp<IAfPlaybackThread> thread;
if (flags & AUDIO_OUTPUT_FLAG_BIT_PERFECT) {
- thread = sp<BitPerfectThread>::make(this, outputStream, *output, mSystemReady);
+ thread = IAfPlaybackThread::createBitPerfectThread(
+ this, outputStream, *output, mSystemReady);
ALOGV("%s() created bit-perfect output: ID %d thread %p",
__func__, *output, thread.get());
} else if (flags & AUDIO_OUTPUT_FLAG_SPATIALIZER) {
- thread = new SpatializerThread(this, outputStream, *output,
+ thread = IAfPlaybackThread::createSpatializerThread(this, outputStream, *output,
mSystemReady, mixerConfig);
ALOGV("openOutput_l() created spatializer output: ID %d thread %p",
*output, thread.get());
} else if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
- thread = new OffloadThread(this, outputStream, *output,
+ thread = IAfPlaybackThread::createOffloadThread(this, outputStream, *output,
mSystemReady, halConfig->offload_info);
ALOGV("openOutput_l() created offload output: ID %d thread %p",
*output, thread.get());
} else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT)
|| !isValidPcmSinkFormat(halConfig->format)
|| !isValidPcmSinkChannelMask(halConfig->channel_mask)) {
- thread = new DirectOutputThread(this, outputStream, *output,
+ thread = IAfPlaybackThread::createDirectOutputThread(this, outputStream, *output,
mSystemReady, halConfig->offload_info);
ALOGV("openOutput_l() created direct output: ID %d thread %p",
*output, thread.get());
} else {
- thread = new MixerThread(this, outputStream, *output, mSystemReady);
+ thread = IAfPlaybackThread::createMixerThread(
+ this, outputStream, *output, mSystemReady);
ALOGV("openOutput_l() created mixer output: ID %d thread %p",
*output, thread.get());
}
@@ -3116,12 +3120,12 @@
Mutex::Autolock _l(mLock);
- sp<ThreadBase> thread = openOutput_l(module, &output, &halConfig,
+ const sp<IAfThreadBase> thread = openOutput_l(module, &output, &halConfig,
&mixerConfig, deviceType, address, flags);
if (thread != 0) {
uint32_t latencyMs = 0;
if ((flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) == 0) {
- PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+ const auto playbackThread = thread->asIAfPlaybackThread();
latencyMs = playbackThread->latency();
// notify client processes of the new output creation
@@ -3139,8 +3143,7 @@
mHardwareStatus = AUDIO_HW_IDLE;
}
} else {
- MmapThread *mmapThread = (MmapThread *)thread.get();
- mmapThread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
+ thread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
}
response->output = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(output));
response->config = VALUE_OR_RETURN_STATUS(
@@ -3158,8 +3161,8 @@
audio_io_handle_t output2)
{
Mutex::Autolock _l(mLock);
- MixerThread *thread1 = checkMixerThread_l(output1);
- MixerThread *thread2 = checkMixerThread_l(output2);
+ IAfPlaybackThread* const thread1 = checkMixerThread_l(output1);
+ IAfPlaybackThread* const thread2 = checkMixerThread_l(output2);
if (thread1 == NULL || thread2 == NULL) {
ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1,
@@ -3168,7 +3171,8 @@
}
audio_io_handle_t id = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT);
- DuplicatingThread *thread = new DuplicatingThread(this, thread1, id, mSystemReady);
+ const sp<IAfDuplicatingThread> thread = IAfDuplicatingThread::create(
+ this, thread1, id, mSystemReady);
thread->addOutputTrack(thread2);
mPlaybackThreads.add(id, thread);
// notify client processes of the new output creation
@@ -3185,8 +3189,8 @@
{
// keep strong reference on the playback thread so that
// it is not destroyed while exit() is executed
- sp<PlaybackThread> playbackThread;
- sp<MmapPlaybackThread> mmapThread;
+ sp<IAfPlaybackThread> playbackThread;
+ sp<IAfMmapPlaybackThread> mmapThread;
{
Mutex::Autolock _l(mLock);
playbackThread = checkPlaybackThread_l(output);
@@ -3195,12 +3199,12 @@
dumpToThreadLog_l(playbackThread);
- if (playbackThread->type() == ThreadBase::MIXER) {
+ if (playbackThread->type() == IAfThreadBase::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 *)playbackThread.get());
+ IAfDuplicatingThread* const dupThread =
+ mPlaybackThreads.valueAt(i)->asIAfDuplicatingThread().get();
+ dupThread->removeOutputTrack(playbackThread.get());
}
}
}
@@ -3209,11 +3213,12 @@
mPlaybackThreads.removeItem(output);
// save all effects to the default thread
if (mPlaybackThreads.size()) {
- PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
+ IAfPlaybackThread* const dstThread =
+ checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
if (dstThread != NULL) {
// audioflinger lock is held so order of thread lock acquisition doesn't matter
- Mutex::Autolock _dl(dstThread->mLock);
- Mutex::Autolock _sl(playbackThread->mLock);
+ Mutex::Autolock _dl(dstThread->mutex());
+ Mutex::Autolock _sl(playbackThread->mutex());
Vector<sp<IAfEffectChain>> effectChains = playbackThread->getEffectChains_l();
for (size_t i = 0; i < effectChains.size(); i ++) {
moveEffectChain_l(effectChains[i]->sessionId(), playbackThread.get(),
@@ -3222,7 +3227,8 @@
}
}
} else {
- mmapThread = (MmapPlaybackThread *)checkMmapThread_l(output);
+ const sp<IAfMmapThread> mt = checkMmapThread_l(output);
+ mmapThread = mt ? mt->asIAfMmapPlaybackThread().get() : nullptr;
if (mmapThread == 0) {
return BAD_VALUE;
}
@@ -3234,7 +3240,7 @@
mPatchPanel.notifyStreamClosed(output);
}
// The thread entity (active unit of execution) is no longer running here,
- // but the ThreadBase container still exists.
+ // but the IAfThreadBase container still exists.
if (playbackThread != 0) {
playbackThread->exit();
@@ -3252,7 +3258,7 @@
return NO_ERROR;
}
-void AudioFlinger::closeOutputFinish(const sp<PlaybackThread>& thread)
+void AudioFlinger::closeOutputFinish(const sp<IAfPlaybackThread>& thread)
{
AudioStreamOut *out = thread->clearOutput();
ALOG_ASSERT(out != NULL, "out shouldn't be NULL");
@@ -3260,9 +3266,9 @@
delete out;
}
-void AudioFlinger::closeThreadInternal_l(const sp<PlaybackThread>& thread)
+void AudioFlinger::closeThreadInternal_l(const sp<IAfPlaybackThread>& thread)
{
- mPlaybackThreads.removeItem(thread->mId);
+ mPlaybackThreads.removeItem(thread->id());
thread->exit();
closeOutputFinish(thread);
}
@@ -3270,7 +3276,7 @@
status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
{
Mutex::Autolock _l(mLock);
- PlaybackThread *thread = checkPlaybackThread_l(output);
+ IAfPlaybackThread* const thread = checkPlaybackThread_l(output);
if (thread == NULL) {
return BAD_VALUE;
@@ -3285,7 +3291,7 @@
status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
{
Mutex::Autolock _l(mLock);
- PlaybackThread *thread = checkPlaybackThread_l(output);
+ IAfPlaybackThread* const thread = checkPlaybackThread_l(output);
if (thread == NULL) {
return BAD_VALUE;
@@ -3314,7 +3320,7 @@
audio_config_t config = VALUE_OR_RETURN_STATUS(
aidl2legacy_AudioConfig_audio_config_t(request.config, true /*isInput*/));
- sp<ThreadBase> thread = openInput_l(
+ const sp<IAfThreadBase> thread = openInput_l(
VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_module_handle_t(request.module)),
&input,
&config,
@@ -3338,7 +3344,7 @@
return NO_INIT;
}
-sp<AudioFlinger::ThreadBase> AudioFlinger::openInput_l(audio_module_handle_t module,
+sp<IAfThreadBase> AudioFlinger::openInput_l(audio_module_handle_t module,
audio_io_handle_t *input,
audio_config_t *config,
audio_devices_t devices,
@@ -3404,17 +3410,18 @@
if (status == NO_ERROR && inStream != 0) {
AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream, flags);
if ((flags & AUDIO_INPUT_FLAG_MMAP_NOIRQ) != 0) {
- sp<MmapCaptureThread> thread =
- new MmapCaptureThread(this, *input, inHwDev, inputStream, mSystemReady);
+ const sp<IAfMmapCaptureThread> thread =
+ IAfMmapCaptureThread::create(this, *input, inHwDev, inputStream, mSystemReady);
mMmapThreads.add(*input, thread);
ALOGV("openInput_l() created mmap capture thread: ID %d thread %p", *input,
thread.get());
return thread;
} else {
// 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, mSystemReady);
+ // IAfRecordThread requires both input and output device indication
+ // to forward to audio pre processing modules
+ const sp<IAfRecordThread> thread =
+ IAfRecordThread::create(this, inputStream, *input, mSystemReady);
mRecordThreads.add(*input, thread);
ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get());
return thread;
@@ -3434,8 +3441,8 @@
{
// keep strong reference on the record thread so that
// it is not destroyed while exit() is executed
- sp<RecordThread> recordThread;
- sp<MmapCaptureThread> mmapThread;
+ sp<IAfRecordThread> recordThread;
+ sp<IAfMmapCaptureThread> mmapThread;
{
Mutex::Autolock _l(mLock);
recordThread = checkRecordThread_l(input);
@@ -3450,8 +3457,8 @@
// new capture on the same session
sp<IAfEffectChain> chain;
{
- Mutex::Autolock _sl(recordThread->mLock);
- Vector< sp<IAfEffectChain> > effectChains = recordThread->getEffectChains_l();
+ Mutex::Autolock _sl(recordThread->mutex());
+ const Vector<sp<IAfEffectChain>> effectChains = recordThread->getEffectChains_l();
// Note: maximum one chain per record thread
if (effectChains.size() != 0) {
chain = effectChains[0];
@@ -3463,12 +3470,12 @@
// creation of its replacement
size_t i;
for (i = 0; i < mRecordThreads.size(); i++) {
- sp<RecordThread> t = mRecordThreads.valueAt(i);
+ const sp<IAfRecordThread> t = mRecordThreads.valueAt(i);
if (t == recordThread) {
continue;
}
if (t->hasAudioSession(chain->sessionId()) != 0) {
- Mutex::Autolock _l2(t->mLock);
+ Mutex::Autolock _l2(t->mutex());
ALOGV("closeInput() found thread %d for effect session %d",
t->id(), chain->sessionId());
t->addEffectChain_l(chain);
@@ -3482,7 +3489,8 @@
}
mRecordThreads.removeItem(input);
} else {
- mmapThread = (MmapCaptureThread *)checkMmapThread_l(input);
+ const sp<IAfMmapThread> mt = checkMmapThread_l(input);
+ mmapThread = mt ? mt->asIAfMmapCaptureThread().get() : nullptr;
if (mmapThread == 0) {
return BAD_VALUE;
}
@@ -3505,7 +3513,7 @@
return NO_ERROR;
}
-void AudioFlinger::closeInputFinish(const sp<RecordThread>& thread)
+void AudioFlinger::closeInputFinish(const sp<IAfRecordThread>& thread)
{
thread->exit();
AudioStreamIn *in = thread->clearInput();
@@ -3514,9 +3522,9 @@
delete in;
}
-void AudioFlinger::closeThreadInternal_l(const sp<RecordThread>& thread)
+void AudioFlinger::closeThreadInternal_l(const sp<IAfRecordThread>& thread)
{
- mRecordThreads.removeItem(thread->mId);
+ mRecordThreads.removeItem(thread->id());
closeInputFinish(thread);
}
@@ -3526,7 +3534,7 @@
std::set<audio_port_handle_t> portIdSet(portIds.begin(), portIds.end());
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
- PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
+ IAfPlaybackThread* const thread = mPlaybackThreads.valueAt(i).get();
thread->invalidateTracks(portIdSet);
if (portIdSet.empty()) {
return NO_ERROR;
@@ -3646,14 +3654,15 @@
ALOGV("purging stale effects");
- Vector< sp<IAfEffectChain> > chains;
+ Vector<sp<IAfEffectChain>> chains;
std::vector< sp<IAfEffectModule> > removedEffects;
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
- sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
- Mutex::Autolock _l(t->mLock);
- for (size_t j = 0; j < t->mEffectChains.size(); j++) {
- sp<IAfEffectChain> ec = t->mEffectChains[j];
+ sp<IAfPlaybackThread> t = mPlaybackThreads.valueAt(i);
+ Mutex::Autolock _l(t->mutex());
+ const Vector<sp<IAfEffectChain>> threadChains = t->getEffectChains_l();
+ for (size_t j = 0; j < threadChains.size(); j++) {
+ sp<IAfEffectChain> ec = threadChains[j];
if (!audio_is_global_session(ec->sessionId())) {
chains.push(ec);
}
@@ -3661,19 +3670,21 @@
}
for (size_t i = 0; i < mRecordThreads.size(); i++) {
- sp<RecordThread> t = mRecordThreads.valueAt(i);
- Mutex::Autolock _l(t->mLock);
- for (size_t j = 0; j < t->mEffectChains.size(); j++) {
- sp<IAfEffectChain> ec = t->mEffectChains[j];
+ sp<IAfRecordThread> t = mRecordThreads.valueAt(i);
+ Mutex::Autolock _l(t->mutex());
+ const Vector<sp<IAfEffectChain>> threadChains = t->getEffectChains_l();
+ for (size_t j = 0; j < threadChains.size(); j++) {
+ sp<IAfEffectChain> ec = threadChains[j];
chains.push(ec);
}
}
for (size_t i = 0; i < mMmapThreads.size(); i++) {
- sp<MmapThread> t = mMmapThreads.valueAt(i);
- Mutex::Autolock _l(t->mLock);
- for (size_t j = 0; j < t->mEffectChains.size(); j++) {
- sp<IAfEffectChain> ec = t->mEffectChains[j];
+ const sp<IAfMmapThread> t = mMmapThreads.valueAt(i);
+ Mutex::Autolock _l(t->mutex());
+ const Vector<sp<IAfEffectChain>> threadChains = t->getEffectChains_l();
+ for (size_t j = 0; j < threadChains.size(); j++) {
+ sp<IAfEffectChain> ec = threadChains[j];
chains.push(ec);
}
}
@@ -3682,7 +3693,7 @@
// clang-tidy suggests const ref
sp<IAfEffectChain> ec = chains[i]; // NOLINT(performance-unnecessary-copy-initialization)
int sessionid = ec->sessionId();
- sp<ThreadBase> t = sp<ThreadBase>::cast(ec->thread().promote()); // TODO(b/288339104)
+ const auto t = sp<IAfThreadBase>::cast(ec->thread().promote()); // TODO(b/288339104)
if (t == 0) {
continue;
}
@@ -3698,7 +3709,7 @@
}
}
if (!found) {
- Mutex::Autolock _l(t->mLock);
+ Mutex::Autolock _l(t->mutex());
// remove all effects from the chain
while (ec->numberOfEffects()) {
sp<IAfEffectModule> effect = ec->getEffectModule(0);
@@ -3715,7 +3726,7 @@
}
// dumpToThreadLog_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::dumpToThreadLog_l(const sp<ThreadBase> &thread)
+void AudioFlinger::dumpToThreadLog_l(const sp<IAfThreadBase> &thread)
{
constexpr int THREAD_DUMP_TIMEOUT_MS = 2;
audio_utils::FdToString fdToString("- ", THREAD_DUMP_TIMEOUT_MS);
@@ -3727,9 +3738,9 @@
}
// checkThread_l() must be called with AudioFlinger::mLock held
-AudioFlinger::ThreadBase *AudioFlinger::checkThread_l(audio_io_handle_t ioHandle) const
+IAfThreadBase* AudioFlinger::checkThread_l(audio_io_handle_t ioHandle) const
{
- ThreadBase *thread = checkMmapThread_l(ioHandle);
+ IAfThreadBase* thread = checkMmapThread_l(ioHandle);
if (thread == 0) {
switch (audio_unique_id_get_use(ioHandle)) {
case AUDIO_UNIQUE_ID_USE_OUTPUT:
@@ -3746,13 +3757,13 @@
}
// checkOutputThread_l() must be called with AudioFlinger::mLock held
-sp<AudioFlinger::ThreadBase> AudioFlinger::checkOutputThread_l(audio_io_handle_t ioHandle) const
+sp<IAfThreadBase> AudioFlinger::checkOutputThread_l(audio_io_handle_t ioHandle) const
{
if (audio_unique_id_get_use(ioHandle) != AUDIO_UNIQUE_ID_USE_OUTPUT) {
return nullptr;
}
- sp<AudioFlinger::ThreadBase> thread = mPlaybackThreads.valueFor(ioHandle);
+ sp<IAfThreadBase> thread = mPlaybackThreads.valueFor(ioHandle);
if (thread == nullptr) {
thread = mMmapThreads.valueFor(ioHandle);
}
@@ -3760,26 +3771,26 @@
}
// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
-AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
+IAfPlaybackThread* 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
+IAfPlaybackThread* AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
{
- PlaybackThread *thread = checkPlaybackThread_l(output);
- return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
+ IAfPlaybackThread * const thread = checkPlaybackThread_l(output);
+ return thread != nullptr && thread->type() != IAfThreadBase::DIRECT ? thread : nullptr;
}
// checkRecordThread_l() must be called with AudioFlinger::mLock held
-AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
+IAfRecordThread* AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
{
return mRecordThreads.valueFor(input).get();
}
// checkMmapThread_l() must be called with AudioFlinger::mLock held
-AudioFlinger::MmapThread *AudioFlinger::checkMmapThread_l(audio_io_handle_t io) const
+IAfMmapThread* AudioFlinger::checkMmapThread_l(audio_io_handle_t io) const
{
return mMmapThreads.valueFor(io).get();
}
@@ -3790,11 +3801,11 @@
{
sp<VolumeInterface> volumeInterface = mPlaybackThreads.valueFor(output).get();
if (volumeInterface == nullptr) {
- MmapThread *mmapThread = mMmapThreads.valueFor(output).get();
+ IAfMmapThread* const mmapThread = mMmapThreads.valueFor(output).get();
if (mmapThread != nullptr) {
if (mmapThread->isOutput()) {
- MmapPlaybackThread *mmapPlaybackThread =
- static_cast<MmapPlaybackThread *>(mmapThread);
+ IAfMmapPlaybackThread* const mmapPlaybackThread =
+ mmapThread->asIAfMmapPlaybackThread().get();
volumeInterface = mmapPlaybackThread;
}
}
@@ -3810,8 +3821,8 @@
}
for (size_t i = 0; i < mMmapThreads.size(); i++) {
if (mMmapThreads.valueAt(i)->isOutput()) {
- MmapPlaybackThread *mmapPlaybackThread =
- static_cast<MmapPlaybackThread *>(mMmapThreads.valueAt(i).get());
+ IAfMmapPlaybackThread* const mmapPlaybackThread =
+ mMmapThreads.valueAt(i)->asIAfMmapPlaybackThread().get();
volumeInterfaces.push_back(mmapPlaybackThread);
}
}
@@ -3839,14 +3850,14 @@
// TODO Use a floor after wraparound. This may need a mutex.
}
-AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const
+IAfPlaybackThread* AudioFlinger::primaryPlaybackThread_l() const
{
AutoMutex lock(mHardwareLock);
if (mPrimaryHardwareDev == nullptr) {
return nullptr;
}
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
- PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
+ IAfPlaybackThread* const thread = mPlaybackThreads.valueAt(i).get();
if(thread->isDuplicating()) {
continue;
}
@@ -3860,7 +3871,7 @@
DeviceTypeSet AudioFlinger::primaryOutputDevice_l() const
{
- PlaybackThread *thread = primaryPlaybackThread_l();
+ IAfPlaybackThread* const thread = primaryPlaybackThread_l();
if (thread == NULL) {
return {};
@@ -3869,12 +3880,12 @@
return thread->outDeviceTypes();
}
-AudioFlinger::PlaybackThread *AudioFlinger::fastPlaybackThread_l() const
+IAfPlaybackThread* AudioFlinger::fastPlaybackThread_l() const
{
size_t minFrameCount = 0;
- PlaybackThread *minThread = NULL;
+ IAfPlaybackThread* minThread = nullptr;
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
- PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
+ IAfPlaybackThread* const thread = mPlaybackThreads.valueAt(i).get();
if (!thread->isDuplicating()) {
size_t frameCount = thread->frameCountHAL();
if (frameCount != 0 && (minFrameCount == 0 || frameCount < minFrameCount ||
@@ -3888,9 +3899,9 @@
return minThread;
}
-AudioFlinger::ThreadBase *AudioFlinger::hapticPlaybackThread_l() const {
+IAfThreadBase* AudioFlinger::hapticPlaybackThread_l() const {
for (size_t i = 0; i < mPlaybackThreads.size(); ++i) {
- PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
+ IAfPlaybackThread* const thread = mPlaybackThreads.valueAt(i).get();
if (thread->hapticChannelMask() != AUDIO_CHANNEL_NONE) {
return thread;
}
@@ -3900,11 +3911,11 @@
void AudioFlinger::updateSecondaryOutputsForTrack_l(
IAfTrack* track,
- PlaybackThread* thread,
+ IAfPlaybackThread* thread,
const std::vector<audio_io_handle_t> &secondaryOutputs) const {
TeePatches teePatches;
for (audio_io_handle_t secondaryOutput : secondaryOutputs) {
- PlaybackThread *secondaryThread = checkPlaybackThread_l(secondaryOutput);
+ IAfPlaybackThread* const secondaryThread = checkPlaybackThread_l(secondaryOutput);
if (secondaryThread == nullptr) {
ALOGE("no playback thread found for secondary output %d", thread->id());
continue;
@@ -3930,10 +3941,10 @@
// The frameCount should also not be smaller than the secondary thread min frame
// count
size_t minFrameCount = AudioSystem::calculateMinFrameCount(
- [&] { Mutex::Autolock _l(secondaryThread->mLock);
+ [&] { Mutex::Autolock _l(secondaryThread->mutex());
return secondaryThread->latency_l(); }(),
- secondaryThread->mNormalFrameCount,
- secondaryThread->mSampleRate,
+ secondaryThread->frameCount(), // normal frame count
+ secondaryThread->sampleRate(),
track->sampleRate(),
track->getSpeed());
frameCount = std::max(frameCount, minFrameCount);
@@ -4182,7 +4193,7 @@
lStatus = BAD_VALUE;
goto Exit;
}
- PlaybackThread *thread = checkPlaybackThread_l(io);
+ IAfPlaybackThread* const thread = checkPlaybackThread_l(io);
if (thread == nullptr) {
ALOGE("%s: invalid output %d specified for AUDIO_SESSION_OUTPUT_STAGE", __func__, io);
lStatus = BAD_VALUE;
@@ -4351,7 +4362,7 @@
}
const uint32_t sessionType =
mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId);
- if ((sessionType & ThreadBase::EFFECT_SESSION) != 0) {
+ if ((sessionType & IAfThreadBase::EFFECT_SESSION) != 0) {
ALOGE("%s: effect %s io %d denied because session %d effect exists on io %d",
__func__, descOut.name, (int) io, (int) sessionId, (int) checkIo);
android_errorWriteLog(0x534e4554, "123237974");
@@ -4360,7 +4371,7 @@
}
}
}
- ThreadBase *thread = checkRecordThread_l(io);
+ IAfThreadBase* thread = checkRecordThread_l(io);
if (thread == NULL) {
thread = checkPlaybackThread_l(io);
if (thread == NULL) {
@@ -4376,7 +4387,7 @@
// session and used it instead of creating a new one.
sp<IAfEffectChain> chain = getOrphanEffectChain_l(sessionId);
if (chain != 0) {
- Mutex::Autolock _l2(thread->mLock);
+ Mutex::Autolock _l2(thread->mutex());
thread->addEffectChain_l(chain);
}
}
@@ -4385,9 +4396,9 @@
// create effect on selected output thread
bool pinned = !audio_is_global_session(sessionId) && isSessionAcquired_l(sessionId);
- ThreadBase *oriThread = nullptr;
+ IAfThreadBase* oriThread = nullptr;
if (hapticPlaybackRequired && thread->hapticChannelMask() == AUDIO_CHANNEL_NONE) {
- ThreadBase *hapticThread = hapticPlaybackThread_l();
+ IAfThreadBase* const hapticThread = hapticPlaybackThread_l();
if (hapticThread == nullptr) {
ALOGE("%s haptic thread not found while it is required", __func__);
lStatus = INVALID_OPERATION;
@@ -4450,26 +4461,26 @@
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);
+ ALOGV("%s() session %d, srcOutput %d, dstOutput %d",
+ __func__, sessionId, srcOutput, dstOutput);
Mutex::Autolock _l(mLock);
if (srcOutput == dstOutput) {
- ALOGW("moveEffects() same dst and src outputs %d", dstOutput);
+ ALOGW("%s() same dst and src outputs %d", __func__, dstOutput);
return NO_ERROR;
}
- PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput);
- if (srcThread == NULL) {
- ALOGW("moveEffects() bad srcOutput %d", srcOutput);
+ IAfPlaybackThread* const srcThread = checkPlaybackThread_l(srcOutput);
+ if (srcThread == nullptr) {
+ ALOGW("%s() bad srcOutput %d", __func__, srcOutput);
return BAD_VALUE;
}
- PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput);
- if (dstThread == NULL) {
- ALOGW("moveEffects() bad dstOutput %d", dstOutput);
+ IAfPlaybackThread* const dstThread = checkPlaybackThread_l(dstOutput);
+ if (dstThread == nullptr) {
+ ALOGW("%s() bad dstOutput %d", __func__, dstOutput);
return BAD_VALUE;
}
- Mutex::Autolock _dl(dstThread->mLock);
- Mutex::Autolock _sl(srcThread->mLock);
+ Mutex::Autolock _dl(dstThread->mutex());
+ Mutex::Autolock _sl(srcThread->mutex());
return moveEffectChain_l(sessionId, srcThread, dstThread);
}
@@ -4480,11 +4491,11 @@
{
Mutex::Autolock _l(mLock);
- sp<ThreadBase> thread = getEffectThread_l(sessionId, effectId);
+ sp<IAfThreadBase> thread = getEffectThread_l(sessionId, effectId);
if (thread == nullptr) {
return;
}
- Mutex::Autolock _sl(thread->mLock);
+ Mutex::Autolock _sl(thread->mutex());
sp<IAfEffectModule> effect = thread->getEffect_l(sessionId, effectId);
thread->setEffectSuspended_l(&effect->desc().type, suspended, sessionId);
}
@@ -4492,8 +4503,7 @@
// 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)
+ IAfPlaybackThread* srcThread, IAfPlaybackThread* dstThread)
NO_THREAD_SAFETY_ANALYSIS // requires srcThread and dstThread locks
{
ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
@@ -4603,17 +4613,16 @@
}
status_t AudioFlinger::moveAuxEffectToIo(int EffectId,
- const sp<PlaybackThread>& dstThread,
- sp<PlaybackThread> *srcThread)
+ const sp<IAfPlaybackThread>& dstThread, sp<IAfPlaybackThread>* srcThread)
{
status_t status = NO_ERROR;
Mutex::Autolock _l(mLock);
- sp<PlaybackThread> thread =
- static_cast<PlaybackThread *>(getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId).get());
+ const sp<IAfThreadBase> threadBase = getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId);
+ const sp<IAfPlaybackThread> thread = threadBase ? threadBase->asIAfPlaybackThread() : nullptr;
if (EffectId != 0 && thread != 0 && dstThread != thread.get()) {
- Mutex::Autolock _dl(dstThread->mLock);
- Mutex::Autolock _sl(thread->mLock);
+ Mutex::Autolock _dl(dstThread->mutex());
+ Mutex::Autolock _sl(thread->mutex());
sp<IAfEffectChain> srcChain = thread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX);
sp<IAfEffectChain> dstChain;
if (srcChain == 0) {
@@ -4677,8 +4686,8 @@
mGlobalEffectEnableTime = systemTime();
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
- sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
- if (t->mType == ThreadBase::OFFLOAD) {
+ const sp<IAfPlaybackThread> t = mPlaybackThreads.valueAt(i);
+ if (t->type() == IAfThreadBase::OFFLOAD) {
t->invalidateTracks(AUDIO_STREAM_MUSIC);
}
}
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 02ebbc2..4562d3e 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -150,10 +150,24 @@
using android::content::AttributionSourceState;
+struct stream_type_t {
+ float volume = 1.f;
+ bool mute = false;
+};
+
class AudioFlinger : public AudioFlingerServerAdapter::Delegate
{
friend class sp<AudioFlinger>;
friend class Client; // removeClient_l();
+ // TODO(b/291012167) replace the Thread friends with an interface.
+ friend class DirectOutputThread;
+ friend class MixerThread;
+ friend class MmapPlaybackThread;
+ friend class MmapThread;
+ friend class PlaybackThread;
+ friend class RecordThread;
+ friend class ThreadBase;
+
public:
static void instantiate() ANDROID_API;
@@ -499,11 +513,7 @@
SimpleLog mThreadLog{16}; // 16 Thread history limit
-public:
- // TODO(b/288339104)
- class ThreadBase;
-private:
- void dumpToThreadLog_l(const sp<ThreadBase> &thread);
+ void dumpToThreadLog_l(const sp<IAfThreadBase>& thread);
// --- Notification Client ---
class NotificationClient : public IBinder::DeathRecipient {
@@ -575,20 +585,6 @@
using TeePatches = std::vector<TeePatch>;
private:
- struct stream_type_t {
- stream_type_t()
- : volume(1.0f),
- mute(false)
- {
- }
- float volume;
- bool mute;
- };
-
- // --- PlaybackThread ---
-
-#include "Threads.h"
-
#include "PatchPanel.h"
#include "PatchCommandThread.h"
@@ -609,7 +605,7 @@
const uint32_t sessionType = threads.valueAt(i)->hasAudioSession(sessionId);
if (sessionType != 0) {
io = threads.keyAt(i);
- if ((sessionType & AudioFlinger::ThreadBase::EFFECT_SESSION) != 0) {
+ if ((sessionType & IAfThreadBase::EFFECT_SESSION) != 0) {
break; // effect chain here.
}
}
@@ -617,40 +613,16 @@
return io;
}
- // Mmap stream control interface implementation. Each MmapThreadHandle controls one
- // MmapPlaybackThread or MmapCaptureThread instance.
- class MmapThreadHandle : public MmapStreamInterface {
- public:
- explicit MmapThreadHandle(const sp<MmapThread>& thread);
- virtual ~MmapThreadHandle();
-
- // MmapStreamInterface virtuals
- virtual status_t createMmapBuffer(int32_t minSizeFrames,
- struct audio_mmap_buffer_info *info);
- virtual status_t getMmapPosition(struct audio_mmap_position *position);
- virtual status_t getExternalPosition(uint64_t *position, int64_t *timeNanos);
- virtual status_t start(const AudioClient& client,
- const audio_attributes_t *attr,
- audio_port_handle_t *handle);
- virtual status_t stop(audio_port_handle_t handle);
- virtual status_t standby();
- status_t reportData(const void* buffer, size_t frameCount) override;
-
- private:
- const sp<MmapThread> mThread;
- };
-
- ThreadBase *checkThread_l(audio_io_handle_t ioHandle) const;
- sp<AudioFlinger::ThreadBase> checkOutputThread_l(audio_io_handle_t ioHandle) const
- REQUIRES(mLock);
- PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const;
- MixerThread *checkMixerThread_l(audio_io_handle_t output) const;
- RecordThread *checkRecordThread_l(audio_io_handle_t input) const;
- MmapThread *checkMmapThread_l(audio_io_handle_t io) const;
+ IAfThreadBase* checkThread_l(audio_io_handle_t ioHandle) const;
+ sp<IAfThreadBase> checkOutputThread_l(audio_io_handle_t ioHandle) const REQUIRES(mLock);
+ IAfPlaybackThread* checkPlaybackThread_l(audio_io_handle_t output) const;
+ IAfPlaybackThread* checkMixerThread_l(audio_io_handle_t output) const;
+ IAfRecordThread* checkRecordThread_l(audio_io_handle_t input) const;
+ IAfMmapThread* checkMmapThread_l(audio_io_handle_t io) const;
sp<VolumeInterface> getVolumeInterface_l(audio_io_handle_t output) const;
std::vector<sp<VolumeInterface>> getAllVolumeInterfaces_l() const;
- sp<ThreadBase> openInput_l(audio_module_handle_t module,
+ sp<IAfThreadBase> openInput_l(audio_module_handle_t module,
audio_io_handle_t *input,
audio_config_t *config,
audio_devices_t device,
@@ -659,7 +631,7 @@
audio_input_flags_t flags,
audio_devices_t outputDevice,
const String8& outputDeviceAddress);
- sp<ThreadBase> openOutput_l(audio_module_handle_t module,
+ sp<IAfThreadBase> openOutput_l(audio_module_handle_t module,
audio_io_handle_t *output,
audio_config_t *halConfig,
audio_config_base_t *mixerConfig,
@@ -667,8 +639,8 @@
const String8& address,
audio_output_flags_t flags);
- void closeOutputFinish(const sp<PlaybackThread>& thread);
- void closeInputFinish(const sp<RecordThread>& thread);
+ void closeOutputFinish(const sp<IAfPlaybackThread>& thread);
+ void closeInputFinish(const sp<IAfRecordThread>& thread);
// no range check, AudioFlinger::mLock held
bool streamMute_l(audio_stream_type_t stream) const
@@ -693,30 +665,28 @@
audio_unique_id_t nextUniqueId(audio_unique_id_use_t use);
status_t moveEffectChain_l(audio_session_t sessionId,
- PlaybackThread *srcThread,
- PlaybackThread *dstThread);
+ IAfPlaybackThread* srcThread, IAfPlaybackThread* dstThread);
public:
// TODO(b/288339104) cluster together
status_t moveAuxEffectToIo(int EffectId,
- const sp<PlaybackThread>& dstThread,
- sp<PlaybackThread> *srcThread);
+ const sp<IAfPlaybackThread>& dstThread, sp<IAfPlaybackThread>* srcThread);
private:
// return thread associated with primary hardware device, or NULL
- PlaybackThread *primaryPlaybackThread_l() const;
+ IAfPlaybackThread* primaryPlaybackThread_l() const;
DeviceTypeSet primaryOutputDevice_l() const;
// return the playback thread with smallest HAL buffer size, and prefer fast
- PlaybackThread *fastPlaybackThread_l() const;
+ IAfPlaybackThread* fastPlaybackThread_l() const;
- sp<ThreadBase> getEffectThread_l(audio_session_t sessionId, int effectId);
+ sp<IAfThreadBase> getEffectThread_l(audio_session_t sessionId, int effectId);
- ThreadBase *hapticPlaybackThread_l() const;
+ IAfThreadBase* hapticPlaybackThread_l() const;
void updateSecondaryOutputsForTrack_l(
IAfTrack* track,
- PlaybackThread* thread,
+ IAfPlaybackThread* thread,
const std::vector<audio_io_handle_t>& secondaryOutputs) const;
@@ -754,7 +724,7 @@
void updateOutDevicesForRecordThreads_l(const DeviceDescriptorBaseVector& devices);
void forwardParametersToDownstreamPatches_l(
audio_io_handle_t upStream, const String8& keyValuePairs,
- const std::function<bool(const sp<PlaybackThread>&)>& useThread = nullptr);
+ const std::function<bool(const sp<IAfPlaybackThread>&)>& useThread = nullptr);
struct TeePatch {
sp<IAfPatchRecord> patchRecord;
@@ -827,7 +797,7 @@
mutable hardware_call_state mHardwareStatus; // for dump only
- DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads;
+ DefaultKeyedVector<audio_io_handle_t, sp<IAfPlaybackThread>> mPlaybackThreads;
stream_type_t mStreamTypes[AUDIO_STREAM_CNT];
// member variables below are protected by mLock
@@ -836,7 +806,7 @@
float mMasterBalance = 0.f;
// end of variables protected by mLock
- DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads;
+ DefaultKeyedVector<audio_io_handle_t, sp<IAfRecordThread>> mRecordThreads;
// protected by mClientLock
DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients;
@@ -867,17 +837,17 @@
// list of MMAP stream control threads. Those threads allow for wake lock, routing
// and volume control for activity on the associated MMAP stream at the HAL.
// Audio data transfer is directly handled by the client creating the MMAP stream
- DefaultKeyedVector< audio_io_handle_t, sp<MmapThread> > mMmapThreads;
+ DefaultKeyedVector<audio_io_handle_t, sp<IAfMmapThread>> mMmapThreads;
private:
sp<Client> registerPid(pid_t pid); // always returns non-0
// for use from destructor
status_t closeOutput_nonvirtual(audio_io_handle_t output);
- void closeThreadInternal_l(const sp<PlaybackThread>& thread);
+ void closeThreadInternal_l(const sp<IAfPlaybackThread>& thread);
status_t closeInput_nonvirtual(audio_io_handle_t input);
- void closeThreadInternal_l(const sp<RecordThread>& thread);
- void setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId);
+ void closeThreadInternal_l(const sp<IAfRecordThread>& thread);
+ void setAudioHwSyncForSession_l(IAfPlaybackThread* thread, audio_session_t sessionId);
status_t checkStreamType(audio_stream_type_t stream) const;
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index 1f26cb0..6b8e905 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -2116,21 +2116,21 @@
/* static */
sp<IAfEffectChain> IAfEffectChain::create(
- const wp<Thread /*ThreadBase*/>& wThread, // TODO(b/288339104) update type
+ const wp<IAfThreadBase>& wThread,
audio_session_t sessionId)
{
// TODO(b/288339104) no weak pointer cast.
- return sp<EffectChain>::make(sp<AudioFlinger::ThreadBase>::cast(wThread.promote()), sessionId);
+ return sp<EffectChain>::make(wThread, sessionId);
}
-EffectChain::EffectChain(const wp<AudioFlinger::ThreadBase>& thread,
+EffectChain::EffectChain(const wp<IAfThreadBase>& thread,
audio_session_t sessionId)
: mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0),
mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX),
mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX),
mEffectCallback(new EffectCallback(wp<EffectChain>(this), thread))
{
- sp<AudioFlinger::ThreadBase> p = thread.promote();
+ const sp<IAfThreadBase> p = thread.promote();
if (p == nullptr) {
return;
}
@@ -2143,7 +2143,7 @@
{
}
-// getEffectFromDesc_l() must be called with AudioFlinger::ThreadBase::mLock held
+// getEffectFromDesc_l() must be called with IAfThreadBase::mutex() held
sp<IAfEffectModule> EffectChain::getEffectFromDesc_l(
effect_descriptor_t *descriptor) const
{
@@ -2157,7 +2157,7 @@
return 0;
}
-// getEffectFromId_l() must be called with AudioFlinger::ThreadBase::mLock held
+// getEffectFromId_l() must be called with IAfThreadBase::mutex() held
sp<IAfEffectModule> EffectChain::getEffectFromId_l(int id) const
{
size_t size = mEffects.size();
@@ -2171,7 +2171,7 @@
return 0;
}
-// getEffectFromType_l() must be called with AudioFlinger::ThreadBase::mLock held
+// getEffectFromType_l() must be called with IAfThreadBase::mutex() held
sp<IAfEffectModule> EffectChain::getEffectFromType_l(
const effect_uuid_t *type) const
{
@@ -2266,7 +2266,7 @@
}
}
-// createEffect_l() must be called with AudioFlinger::ThreadBase::mLock held
+// createEffect_l() must be called with IAfThreadBase::mutex() held
status_t EffectChain::createEffect_l(sp<IAfEffectModule>& effect,
effect_descriptor_t *desc,
int id,
@@ -2285,13 +2285,13 @@
return lStatus;
}
-// addEffect_l() must be called with AudioFlinger::ThreadBase::mLock held
+// addEffect_l() must be called with IAfThreadBase::mutex() held
status_t EffectChain::addEffect_l(const sp<IAfEffectModule>& effect)
{
Mutex::Autolock _l(mLock);
return addEffect_ll(effect);
}
-// addEffect_l() must be called with AudioFlinger::ThreadBase::mLock and EffectChain::mLock held
+// addEffect_l() must be called with IAfThreadBase::mLock and EffectChain::mutex() held
status_t EffectChain::addEffect_ll(const sp<IAfEffectModule>& effect)
{
effect->setCallback(mEffectCallback);
@@ -2445,7 +2445,7 @@
return idx_insert;
}
-// removeEffect_l() must be called with AudioFlinger::ThreadBase::mLock held
+// removeEffect_l() must be called with IAfThreadBase::mutex() held
size_t EffectChain::removeEffect_l(const sp<IAfEffectModule>& effect,
bool release)
{
@@ -2493,7 +2493,7 @@
return mEffects.size();
}
-// setDevices_l() must be called with AudioFlinger::ThreadBase::mLock held
+// setDevices_l() must be called with IAfThreadBase::mutex() held
void EffectChain::setDevices_l(const AudioDeviceTypeAddrVector &devices)
{
size_t size = mEffects.size();
@@ -2502,7 +2502,7 @@
}
}
-// setInputDevice_l() must be called with AudioFlinger::ThreadBase::mLock held
+// setInputDevice_l() must be called with IAfThreadBase::mutex() held
void EffectChain::setInputDevice_l(const AudioDeviceTypeAddr &device)
{
size_t size = mEffects.size();
@@ -2511,7 +2511,7 @@
}
}
-// setMode_l() must be called with AudioFlinger::ThreadBase::mLock held
+// setMode_l() must be called with IAfThreadBase::mutex() held
void EffectChain::setMode_l(audio_mode_t mode)
{
size_t size = mEffects.size();
@@ -2520,7 +2520,7 @@
}
}
-// setAudioSource_l() must be called with AudioFlinger::ThreadBase::mLock held
+// setAudioSource_l() must be called with IAfThreadBase::mutex() held
void EffectChain::setAudioSource_l(audio_source_t source)
{
size_t size = mEffects.size();
@@ -2536,7 +2536,7 @@
return false;
}
-// setVolume_l() must be called with AudioFlinger::ThreadBase::mLock or EffectChain::mLock held
+// setVolume_l() must be called with IAfThreadBase::mLock or EffectChain::mLock held
bool EffectChain::setVolume_l(uint32_t *left, uint32_t *right, bool force)
{
uint32_t newLeft = *left;
@@ -2603,7 +2603,7 @@
return hasControl;
}
-// resetVolume_l() must be called with AudioFlinger::ThreadBase::mLock or EffectChain::mLock held
+// resetVolume_l() must be called with IAfThreadBase::mutex() or EffectChain::mLock held
void EffectChain::resetVolume_l()
{
if ((mLeftVolume != UINT_MAX) && (mRightVolume != UINT_MAX)) {
@@ -2614,7 +2614,7 @@
}
// containsHapticGeneratingEffect_l must be called with
-// AudioFlinger::ThreadBase::mLock or EffectChain::mLock held
+// IAfThreadBase::mutex() or EffectChain::mLock held
bool EffectChain::containsHapticGeneratingEffect_l()
{
for (size_t i = 0; i < mEffects.size(); ++i) {
@@ -2683,7 +2683,7 @@
}
}
-// must be called with AudioFlinger::ThreadBase::mLock held
+// must be called with IAfThreadBase::mutex() held
void EffectChain::setEffectSuspended_l(
const effect_uuid_t *type, bool suspend)
{
@@ -2739,7 +2739,7 @@
}
}
-// must be called with AudioFlinger::ThreadBase::mLock held
+// must be called with IAfThreadBase::mutex() held
void EffectChain::setEffectSuspendedAll_l(bool suspend)
{
sp<SuspendedEffectDesc> desc;
@@ -2895,7 +2895,7 @@
return false;
}
-void EffectChain::setThread(const sp<AudioFlinger::ThreadBase>& thread)
+void EffectChain::setThread(const sp<IAfThreadBase>& thread)
{
Mutex::Autolock _l(mLock);
mEffectCallback->setThread(thread);
@@ -2962,7 +2962,7 @@
}
// isCompatibleWithThread_l() must be called with thread->mLock held
-bool EffectChain::isCompatibleWithThread_l(const sp<AudioFlinger::ThreadBase>& thread) const
+bool EffectChain::isCompatibleWithThread_l(const sp<IAfThreadBase>& thread) const
{
Mutex::Autolock _l(mLock);
for (size_t i = 0; i < mEffects.size(); i++) {
@@ -3000,7 +3000,7 @@
status_t EffectChain::EffectCallback::addEffectToHal(
const sp<EffectHalInterface>& effect) {
status_t result = NO_INIT;
- sp<AudioFlinger::ThreadBase> t = thread().promote();
+ const sp<IAfThreadBase> t = thread().promote();
if (t == nullptr) {
return result;
}
@@ -3016,7 +3016,7 @@
status_t EffectChain::EffectCallback::removeEffectFromHal(
const sp<EffectHalInterface>& effect) {
status_t result = NO_INIT;
- sp<AudioFlinger::ThreadBase> t = thread().promote();
+ const sp<IAfThreadBase> t = thread().promote();
if (t == nullptr) {
return result;
}
@@ -3030,7 +3030,7 @@
}
audio_io_handle_t EffectChain::EffectCallback::io() const {
- sp<AudioFlinger::ThreadBase> t = thread().promote();
+ const sp<IAfThreadBase> t = thread().promote();
if (t == nullptr) {
return AUDIO_IO_HANDLE_NONE;
}
@@ -3038,7 +3038,7 @@
}
bool EffectChain::EffectCallback::isOutput() const {
- sp<AudioFlinger::ThreadBase> t = thread().promote();
+ const sp<IAfThreadBase> t = thread().promote();
if (t == nullptr) {
return true;
}
@@ -3046,19 +3046,19 @@
}
bool EffectChain::EffectCallback::isOffload() const {
- return mThreadType == AudioFlinger::ThreadBase::OFFLOAD;
+ return mThreadType == IAfThreadBase::OFFLOAD;
}
bool EffectChain::EffectCallback::isOffloadOrDirect() const {
- return mThreadType == AudioFlinger::ThreadBase::OFFLOAD
- || mThreadType == AudioFlinger::ThreadBase::DIRECT;
+ return mThreadType == IAfThreadBase::OFFLOAD
+ || mThreadType == IAfThreadBase::DIRECT;
}
bool EffectChain::EffectCallback::isOffloadOrMmap() const {
switch (mThreadType) {
- case AudioFlinger::ThreadBase::OFFLOAD:
- case AudioFlinger::ThreadBase::MMAP_PLAYBACK:
- case AudioFlinger::ThreadBase::MMAP_CAPTURE:
+ case IAfThreadBase::OFFLOAD:
+ case IAfThreadBase::MMAP_PLAYBACK:
+ case IAfThreadBase::MMAP_CAPTURE:
return true;
default:
return false;
@@ -3066,11 +3066,11 @@
}
bool EffectChain::EffectCallback::isSpatializer() const {
- return mThreadType == AudioFlinger::ThreadBase::SPATIALIZER;
+ return mThreadType == IAfThreadBase::SPATIALIZER;
}
uint32_t EffectChain::EffectCallback::sampleRate() const {
- sp<AudioFlinger::ThreadBase> t = thread().promote();
+ const sp<IAfThreadBase> t = thread().promote();
if (t == nullptr) {
return 0;
}
@@ -3078,7 +3078,7 @@
}
audio_channel_mask_t EffectChain::EffectCallback::inChannelMask(int id) const {
- sp<AudioFlinger::ThreadBase> t = thread().promote();
+ const sp<IAfThreadBase> t = thread().promote();
if (t == nullptr) {
return AUDIO_CHANNEL_NONE;
}
@@ -3087,7 +3087,7 @@
return AUDIO_CHANNEL_NONE;
}
- if (mThreadType == AudioFlinger::ThreadBase::SPATIALIZER) {
+ if (mThreadType == IAfThreadBase::SPATIALIZER) {
if (c->sessionId() == AUDIO_SESSION_OUTPUT_STAGE) {
if (c->isFirstEffect(id)) {
return t->mixerChannelMask();
@@ -3096,7 +3096,7 @@
}
} else if (!audio_is_global_session(c->sessionId())) {
if ((t->hasAudioSession_l(c->sessionId())
- & AudioFlinger::ThreadBase::SPATIALIZED_SESSION) != 0) {
+ & IAfThreadBase::SPATIALIZED_SESSION) != 0) {
return t->mixerChannelMask();
} else {
return t->channelMask();
@@ -3114,7 +3114,7 @@
}
audio_channel_mask_t EffectChain::EffectCallback::outChannelMask() const {
- sp<AudioFlinger::ThreadBase> t = thread().promote();
+ const sp<IAfThreadBase> t = thread().promote();
if (t == nullptr) {
return AUDIO_CHANNEL_NONE;
}
@@ -3123,10 +3123,10 @@
return AUDIO_CHANNEL_NONE;
}
- if (mThreadType == AudioFlinger::ThreadBase::SPATIALIZER) {
+ if (mThreadType == IAfThreadBase::SPATIALIZER) {
if (!audio_is_global_session(c->sessionId())) {
if ((t->hasAudioSession_l(c->sessionId())
- & AudioFlinger::ThreadBase::SPATIALIZED_SESSION) != 0) {
+ & IAfThreadBase::SPATIALIZED_SESSION) != 0) {
return t->mixerChannelMask();
} else {
return t->channelMask();
@@ -3144,7 +3144,7 @@
}
audio_channel_mask_t EffectChain::EffectCallback::hapticChannelMask() const {
- sp<AudioFlinger::ThreadBase> t = thread().promote();
+ const sp<IAfThreadBase> t = thread().promote();
if (t == nullptr) {
return AUDIO_CHANNEL_NONE;
}
@@ -3152,7 +3152,7 @@
}
size_t EffectChain::EffectCallback::frameCount() const {
- sp<AudioFlinger::ThreadBase> t = thread().promote();
+ const sp<IAfThreadBase> t = thread().promote();
if (t == nullptr) {
return 0;
}
@@ -3162,7 +3162,7 @@
uint32_t EffectChain::EffectCallback::latency() const
NO_THREAD_SAFETY_ANALYSIS // latency_l() access
{
- sp<AudioFlinger::ThreadBase> t = thread().promote();
+ const sp<IAfThreadBase> t = thread().promote();
if (t == nullptr) {
return 0;
}
@@ -3173,7 +3173,7 @@
void EffectChain::EffectCallback::setVolumeForOutput(float left, float right) const
NO_THREAD_SAFETY_ANALYSIS // setVolumeForOutput_l() access
{
- sp<AudioFlinger::ThreadBase> t = thread().promote();
+ const sp<IAfThreadBase> t = thread().promote();
if (t == nullptr) {
return;
}
@@ -3182,7 +3182,7 @@
void EffectChain::EffectCallback::checkSuspendOnEffectEnabled(
const sp<IAfEffectBase>& effect, bool enabled, bool threadLocked) {
- sp<AudioFlinger::ThreadBase> t = thread().promote();
+ const sp<IAfThreadBase> t = thread().promote();
if (t == nullptr) {
return;
}
@@ -3197,7 +3197,7 @@
}
void EffectChain::EffectCallback::onEffectEnable(const sp<IAfEffectBase>& effect) {
- sp<AudioFlinger::ThreadBase> t = thread().promote();
+ const sp<IAfThreadBase> t = thread().promote();
if (t == nullptr) {
return;
}
@@ -3208,7 +3208,7 @@
void EffectChain::EffectCallback::onEffectDisable(const sp<IAfEffectBase>& effect) {
checkSuspendOnEffectEnabled(effect, false, false /*threadLocked*/);
- sp<AudioFlinger::ThreadBase> t = thread().promote();
+ const sp<IAfThreadBase> t = thread().promote();
if (t == nullptr) {
return;
}
@@ -3217,7 +3217,7 @@
bool EffectChain::EffectCallback::disconnectEffectHandle(IAfEffectHandle *handle,
bool unpinIfLast) {
- sp<AudioFlinger::ThreadBase> t = thread().promote();
+ const sp<IAfThreadBase> t = thread().promote();
if (t == nullptr) {
return false;
}
@@ -3376,7 +3376,7 @@
mDevicePort.id = AUDIO_PORT_HANDLE_NONE;
}
} else if (patch.isSoftware() || patch.thread().promote() != nullptr) {
- sp <AudioFlinger::ThreadBase> thread;
+ sp<IAfThreadBase> thread;
if (audio_port_config_has_input_direction(port)) {
if (patch.isSoftware()) {
thread = patch.mRecord.thread();
diff --git a/services/audioflinger/Effects.h b/services/audioflinger/Effects.h
index 07790be..ae87346 100644
--- a/services/audioflinger/Effects.h
+++ b/services/audioflinger/Effects.h
@@ -382,7 +382,7 @@
// it also provide it's own input buffer used by the track as accumulation buffer.
class EffectChain : public IAfEffectChain {
public:
- EffectChain(const wp<AudioFlinger::ThreadBase>& wThread, audio_session_t sessionId);
+ EffectChain(const wp<IAfThreadBase>& wThread, audio_session_t sessionId);
~EffectChain() override;
void process_l() final;
@@ -479,12 +479,7 @@
bool isBitPerfectCompatible() const final;
// isCompatibleWithThread_l() must be called with thread->mLock held
- // TODO(b/288339104) type
- bool isCompatibleWithThread_l(const sp<Thread>& thread) const final {
- return isCompatibleWithThread_l(sp<AudioFlinger::ThreadBase>::cast(thread));
- }
-
- bool isCompatibleWithThread_l(const sp<AudioFlinger::ThreadBase>& thread) const;
+ bool isCompatibleWithThread_l(const sp<IAfThreadBase>& thread) const final;
bool containsHapticGeneratingEffect_l() final;
@@ -492,8 +487,7 @@
sp<EffectCallbackInterface> effectCallback() const final { return mEffectCallback; }
- // TODO(b/288339104) type
- wp<Thread> thread() const final { return mEffectCallback->thread(); }
+ wp<IAfThreadBase> thread() const final { return mEffectCallback->thread(); }
bool isFirstEffect(int id) const final {
return !mEffects.isEmpty() && id == mEffects[0]->id();
@@ -507,12 +501,7 @@
return mEffects[index];
}
- // TODO(b/288339104) type
- void setThread(const sp<Thread>& thread) final {
- setThread(sp<AudioFlinger::ThreadBase>::cast(thread));
- }
-
- void setThread(const sp<AudioFlinger::ThreadBase>& thread);
+ void setThread(const sp<IAfThreadBase>& thread) final;
private:
@@ -527,15 +516,15 @@
// Note: ctors taking a weak pointer to their owner must not promote it
// during construction (but may keep a reference for later promotion).
EffectCallback(const wp<EffectChain>& owner,
- const wp<AudioFlinger::ThreadBase>& thread)
+ const wp<IAfThreadBase>& thread)
: mChain(owner)
, mThread(thread)
, mAudioFlinger(*AudioFlinger::gAudioFlinger) {
- sp<AudioFlinger::ThreadBase> base = thread.promote();
+ const sp<IAfThreadBase> base = thread.promote();
if (base != nullptr) {
mThreadType = base->type();
} else {
- mThreadType = AudioFlinger::ThreadBase::MIXER; // assure a consistent value.
+ mThreadType = IAfThreadBase::MIXER; // assure a consistent value.
}
}
@@ -580,18 +569,18 @@
return mAudioFlinger.isAudioPolicyReady();
}
- wp<AudioFlinger::ThreadBase> thread() const { return mThread.load(); }
+ wp<IAfThreadBase> thread() const { return mThread.load(); }
- void setThread(const sp<AudioFlinger::ThreadBase>& thread) {
+ void setThread(const sp<IAfThreadBase>& thread) {
mThread = thread;
mThreadType = thread->type();
}
private:
const wp<IAfEffectChain> mChain;
- mediautils::atomic_wp<AudioFlinger::ThreadBase> mThread;
+ mediautils::atomic_wp<IAfThreadBase> mThread;
AudioFlinger &mAudioFlinger; // implementation detail: outer instance always exists.
- AudioFlinger::ThreadBase::type_t mThreadType;
+ IAfThreadBase::type_t mThreadType;
};
DISALLOW_COPY_AND_ASSIGN(EffectChain);
diff --git a/services/audioflinger/IAfEffect.h b/services/audioflinger/IAfEffect.h
index 75112ca..cff0f73 100644
--- a/services/audioflinger/IAfEffect.h
+++ b/services/audioflinger/IAfEffect.h
@@ -23,6 +23,7 @@
class IAfEffectChain;
class IAfEffectHandle;
class IAfEffectModule;
+class IAfThreadBase;
// Interface implemented by the EffectModule parent or owner (e.g an EffectChain) to abstract
// interactions between the EffectModule and the reset of the audio framework.
@@ -190,7 +191,7 @@
// Most of these methods are accessed from AudioFlinger::Thread
public:
static sp<IAfEffectChain> create(
- const wp<Thread /*ThreadBase*/>& wThread, // TODO(b/288339104) type
+ const wp<IAfThreadBase>& wThread,
audio_session_t sessionId);
// special key used for an entry in mSuspendedEffects keyed vector
@@ -279,8 +280,7 @@
virtual bool isBitPerfectCompatible() const = 0;
// isCompatibleWithThread_l() must be called with thread->mLock held
- // TODO(b/288339104) type
- virtual bool isCompatibleWithThread_l(const sp<Thread>& thread) const = 0;
+ virtual bool isCompatibleWithThread_l(const sp<IAfThreadBase>& thread) const = 0;
virtual bool containsHapticGeneratingEffect_l() = 0;
@@ -288,8 +288,8 @@
virtual sp<EffectCallbackInterface> effectCallback() const = 0;
- virtual wp<Thread> thread() const = 0; // TODO(b/288339104) type
- virtual void setThread(const sp<Thread>& thread) = 0; // TODO(b/288339104) type
+ virtual wp<IAfThreadBase> thread() const = 0;
+ virtual void setThread(const sp<IAfThreadBase>& thread) = 0;
virtual bool isFirstEffect(int id) const = 0;
@@ -364,4 +364,4 @@
virtual status_t removeEffectFromHal(const sp<EffectHalInterface>& effect) = 0;
};
-} // namespace android
+} // namespace android
diff --git a/services/audioflinger/IAfThread.h b/services/audioflinger/IAfThread.h
index 449ed90..8b87ed7 100644
--- a/services/audioflinger/IAfThread.h
+++ b/services/audioflinger/IAfThread.h
@@ -20,6 +20,13 @@
namespace android {
+class IAfDirectOutputThread;
+class IAfDuplicatingThread;
+class IAfMmapCaptureThread;
+class IAfMmapPlaybackThread;
+class IAfPlaybackThread;
+class IAfRecordThread;
+
class IAfThreadBase : public virtual RefBase {
public:
enum type_t {
@@ -52,6 +59,7 @@
// and returns the [normal mix] buffer's frame count.
virtual size_t frameCount() const = 0;
virtual audio_channel_mask_t hapticChannelMask() const = 0;
+ virtual uint32_t hapticChannelCount() const = 0;
virtual uint32_t latency_l() const = 0;
virtual void setVolumeForOutput_l(float left, float right) const = 0;
@@ -233,10 +241,42 @@
virtual void stopMelComputation_l() = 0;
virtual product_strategy_t getStrategyForStream(audio_stream_type_t stream) const = 0;
+
+ virtual void setEffectSuspended_l(
+ const effect_uuid_t* type, bool suspend, audio_session_t sessionId) = 0;
+
+ // Dynamic cast to derived interface
+ virtual sp<IAfDirectOutputThread> asIAfDirectOutputThread() { return nullptr; }
+ virtual sp<IAfDuplicatingThread> asIAfDuplicatingThread() { return nullptr; }
+ virtual sp<IAfPlaybackThread> asIAfPlaybackThread() { return nullptr; }
+ virtual sp<IAfRecordThread> asIAfRecordThread() { return nullptr; }
+ virtual AudioFlinger* audioFlinger() const = 0;
};
-class IAfPlaybackThread : public virtual IAfThreadBase {
+class IAfPlaybackThread : public virtual IAfThreadBase, public virtual VolumeInterface {
public:
+ static sp<IAfPlaybackThread> createBitPerfectThread(
+ const sp<AudioFlinger>& audioflinger, AudioStreamOut* output, audio_io_handle_t id,
+ bool systemReady);
+
+ static sp<IAfPlaybackThread> createDirectOutputThread(
+ const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, audio_io_handle_t id,
+ bool systemReady, const audio_offload_info_t& offloadInfo);
+
+ static sp<IAfPlaybackThread> createMixerThread(
+ const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, audio_io_handle_t id,
+ bool systemReady, type_t type = MIXER, audio_config_base_t* mixerConfig = nullptr);
+
+ static sp<IAfPlaybackThread> createOffloadThread(
+ const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, audio_io_handle_t id,
+ bool systemReady, const audio_offload_info_t& offloadInfo);
+
+ static sp<IAfPlaybackThread> createSpatializerThread(
+ const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, audio_io_handle_t id,
+ bool systemReady, audio_config_base_t* mixerConfig);
+
+ static constexpr int8_t kMaxTrackStopRetriesOffload = 2;
+
enum mixer_state {
MIXER_IDLE, // no active tracks
MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready
@@ -250,6 +290,8 @@
// return estimated latency in milliseconds, as reported by HAL
virtual uint32_t latency() const = 0; // should be in IAfThreadBase?
+ virtual uint32_t& fastTrackAvailMask_l() = 0;
+
virtual sp<IAfTrack> createTrack_l(
const sp<Client>& client,
audio_stream_type_t streamType,
@@ -273,9 +315,14 @@
bool isSpatialized,
bool isBitPerfect) = 0;
+ virtual status_t addTrack_l(const sp<IAfTrack>& track) = 0;
+ virtual bool destroyTrack_l(const sp<IAfTrack>& track) = 0;
+ virtual bool isTrackActive(const sp<IAfTrack>& track) const = 0;
+ virtual void addOutputTrack_l(const sp<IAfTrack>& track) = 0;
+
+ virtual AudioStreamOut* getOutput_l() const = 0;
virtual AudioStreamOut* getOutput() const = 0;
virtual AudioStreamOut* clearOutput() = 0;
- virtual sp<StreamHalInterface> stream() const = 0;
// a very large number of suspend() will eventually wraparound, but unlikely
virtual void suspend() = 0;
@@ -329,6 +376,142 @@
virtual bool hasFastMixer() const = 0;
virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const = 0;
virtual const std::atomic<int64_t>& framesWritten() const = 0;
+
+ virtual bool usesHwAvSync() const = 0;
+};
+
+class IAfDirectOutputThread : public virtual IAfPlaybackThread {
+public:
+ virtual status_t selectPresentation(int presentationId, int programId) = 0;
+};
+
+class IAfDuplicatingThread : public virtual IAfPlaybackThread {
+public:
+ static sp<IAfDuplicatingThread> create(
+ const sp<AudioFlinger>& audioFlinger, IAfPlaybackThread* mainThread,
+ audio_io_handle_t id, bool systemReady);
+
+ virtual void addOutputTrack(IAfPlaybackThread* thread) = 0;
+ virtual uint32_t waitTimeMs() const = 0;
+ virtual void removeOutputTrack(IAfPlaybackThread* thread) = 0;
+};
+
+class IAfRecordThread : public virtual IAfThreadBase {
+public:
+ static sp<IAfRecordThread> create(
+ const sp<AudioFlinger>& audioFlinger, AudioStreamIn* input, audio_io_handle_t id,
+ bool systemReady);
+
+ virtual sp<IAfRecordTrack> createRecordTrack_l(
+ const sp<Client>& client,
+ const audio_attributes_t& attr,
+ uint32_t* pSampleRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ size_t* pFrameCount,
+ audio_session_t sessionId,
+ size_t* pNotificationFrameCount,
+ pid_t creatorPid,
+ const AttributionSourceState& attributionSource,
+ audio_input_flags_t* flags,
+ pid_t tid,
+ status_t* status /*non-NULL*/,
+ audio_port_handle_t portId,
+ int32_t maxSharedAudioHistoryMs) = 0;
+ virtual void destroyTrack_l(const sp<IAfRecordTrack>& track) = 0;
+ virtual void removeTrack_l(const sp<IAfRecordTrack>& track) = 0;
+
+ virtual status_t start(
+ IAfRecordTrack* recordTrack, AudioSystem::sync_event_t event,
+ audio_session_t triggerSession) = 0;
+
+ // ask the thread to stop the specified track, and
+ // return true if the caller should then do it's part of the stopping process
+ virtual bool stop(IAfRecordTrack* recordTrack) = 0;
+
+ virtual AudioStreamIn* getInput() const = 0;
+ virtual AudioStreamIn* clearInput() = 0;
+
+ virtual status_t getActiveMicrophones(
+ std::vector<media::MicrophoneInfoFw>* activeMicrophones) const = 0;
+ virtual status_t setPreferredMicrophoneDirection(audio_microphone_direction_t direction) = 0;
+ virtual status_t setPreferredMicrophoneFieldDimension(float zoom) = 0;
+
+ virtual void addPatchTrack(const sp<IAfPatchRecord>& record) = 0;
+ virtual void deletePatchTrack(const sp<IAfPatchRecord>& record) = 0;
+ virtual bool fastTrackAvailable() const = 0;
+ virtual void setFastTrackAvailable(bool available) = 0;
+
+ virtual void setRecordSilenced(audio_port_handle_t portId, bool silenced) = 0;
+ virtual bool hasFastCapture() const = 0;
+
+ virtual void checkBtNrec() = 0;
+ virtual uint32_t getInputFramesLost() const = 0;
+
+ virtual status_t shareAudioHistory(
+ const std::string& sharedAudioPackageName,
+ audio_session_t sharedSessionId = AUDIO_SESSION_NONE,
+ int64_t sharedAudioStartMs = -1) = 0;
+ virtual void resetAudioHistory_l() = 0;
+};
+
+class IAfMmapThread : public virtual IAfThreadBase {
+public:
+ // createIAudioTrackAdapter() is a static constructor which creates an
+ // MmapStreamInterface AIDL interface adapter from the MmapThread object that
+ // may be passed back to the client.
+ //
+ // Only one AIDL MmapStreamInterface interface adapter should be created per MmapThread.
+ static sp<MmapStreamInterface> createMmapStreamInterfaceAdapter(
+ const sp<IAfMmapThread>& mmapThread);
+
+ virtual void configure(
+ const audio_attributes_t* attr,
+ audio_stream_type_t streamType,
+ audio_session_t sessionId,
+ const sp<MmapStreamCallback>& callback,
+ audio_port_handle_t deviceId,
+ audio_port_handle_t portId) = 0;
+ virtual void disconnect() = 0;
+
+ // MmapStreamInterface handling (see adapter)
+ virtual status_t createMmapBuffer(
+ int32_t minSizeFrames, struct audio_mmap_buffer_info* info) = 0;
+ virtual status_t getMmapPosition(struct audio_mmap_position* position) const = 0;
+ virtual status_t start(
+ const AudioClient& client, const audio_attributes_t* attr,
+ audio_port_handle_t* handle) = 0;
+ virtual status_t stop(audio_port_handle_t handle) = 0;
+ virtual status_t standby() = 0;
+ virtual status_t getExternalPosition(uint64_t* position, int64_t* timeNanos) const = 0;
+ virtual status_t reportData(const void* buffer, size_t frameCount) = 0;
+
+ // TODO(b/288339104) move to IAfThreadBase?
+ virtual void invalidateTracks(std::set<audio_port_handle_t>& portIds) = 0;
+
+ // Sets the UID records silence - TODO(b/288339104) move to IAfMmapCaptureThread
+ virtual void setRecordSilenced(audio_port_handle_t portId, bool silenced) = 0;
+
+ virtual sp<IAfMmapPlaybackThread> asIAfMmapPlaybackThread() { return nullptr; }
+ virtual sp<IAfMmapCaptureThread> asIAfMmapCaptureThread() { return nullptr; }
+};
+
+class IAfMmapPlaybackThread : public virtual IAfMmapThread, public virtual VolumeInterface {
+public:
+ static sp<IAfMmapPlaybackThread> create(
+ const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, AudioHwDevice* hwDev,
+ AudioStreamOut* output, bool systemReady);
+
+ virtual AudioStreamOut* clearOutput() = 0;
+};
+
+class IAfMmapCaptureThread : public virtual IAfMmapThread {
+public:
+ static sp<IAfMmapCaptureThread> create(
+ const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, AudioHwDevice* hwDev,
+ AudioStreamIn* input, bool systemReady);
+
+ virtual AudioStreamIn* clearInput() = 0;
};
} // namespace android
diff --git a/services/audioflinger/IAfTrack.h b/services/audioflinger/IAfTrack.h
index 4718474..9ca13ca 100644
--- a/services/audioflinger/IAfTrack.h
+++ b/services/audioflinger/IAfTrack.h
@@ -18,6 +18,11 @@
namespace android {
+class IAfDuplicatingThread;
+class IAfPlaybackThread;
+class IAfRecordThread;
+class IAfThreadBase;
+
// Common interface to all Playback and Record tracks.
class IAfTrackBase : public virtual RefBase {
public:
@@ -97,8 +102,7 @@
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer) = 0;
// Added for RecordTrack and OutputTrack
- // TODO(b/288339104) type
- virtual wp<Thread> thread() const = 0;
+ virtual wp<IAfThreadBase> thread() const = 0;
virtual const sp<ServerProxy>& serverProxy() const = 0;
// TEE_SINK
@@ -233,8 +237,8 @@
// Only one AIDL IAudioTrack interface adapter should be created per Track.
static sp<media::IAudioTrack> createIAudioTrackAdapter(const sp<IAfTrack>& track);
- static sp<IAfTrack> create( // TODO(b/288339104) void*
- void* /* AudioFlinger::PlaybackThread */ thread,
+ static sp<IAfTrack> create(
+ IAfPlaybackThread* thread,
const sp<Client>& client,
audio_stream_type_t streamType,
const audio_attributes_t& attr,
@@ -399,8 +403,8 @@
public:
// TODO(b/288339104) void*
static sp<IAfOutputTrack> create(
- void* /* AudioFlinger::PlaybackThread */ playbackThread,
- void* /* AudioFlinger::DuplicatingThread */ sourceThread, uint32_t sampleRate,
+ IAfPlaybackThread* playbackThread,
+ IAfDuplicatingThread* sourceThread, uint32_t sampleRate,
audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount,
const AttributionSourceState& attributionSource);
@@ -417,7 +421,7 @@
class IAfMmapTrack : public virtual IAfTrackBase {
public:
// TODO(b/288339104) void*
- static sp<IAfMmapTrack> create(void* /*AudioFlinger::ThreadBase */ thread,
+ static sp<IAfMmapTrack> create(IAfThreadBase* thread,
const audio_attributes_t& attr,
uint32_t sampleRate,
audio_format_t format,
@@ -455,7 +459,7 @@
static sp<media::IAudioRecord> createIAudioRecordAdapter(const sp<IAfRecordTrack>& recordTrack);
// TODO(b/288339104) void*
- static sp<IAfRecordTrack> create(void* /* AudioFlinger::RecordThread */ thread,
+ static sp<IAfRecordTrack> create(IAfRecordThread* thread,
const sp<Client>& client,
const audio_attributes_t& attr,
uint32_t sampleRate,
@@ -533,7 +537,7 @@
class IAfPatchTrack : public virtual IAfTrack, public virtual IAfPatchTrackBase {
public:
static sp<IAfPatchTrack> create(
- void * /* PlaybackThread */ playbackThread, // TODO(b/288339104)
+ IAfPlaybackThread* playbackThread,
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_channel_mask_t channelMask,
@@ -552,7 +556,7 @@
class IAfPatchRecord : public virtual IAfRecordTrack, public virtual IAfPatchTrackBase {
public:
static sp<IAfPatchRecord> create(
- void* /* RecordThread */ recordThread, // TODO(b/288339104)
+ IAfRecordThread* recordThread,
uint32_t sampleRate,
audio_channel_mask_t channelMask,
audio_format_t format,
@@ -564,7 +568,7 @@
audio_source_t source = AUDIO_SOURCE_DEFAULT);
static sp<IAfPatchRecord> createPassThru(
- void* /* RecordThread */ recordThread, // TODO(b/288339104)
+ IAfRecordThread* recordThread,
uint32_t sampleRate,
audio_channel_mask_t channelMask,
audio_format_t format,
diff --git a/services/audioflinger/MelReporter.cpp b/services/audioflinger/MelReporter.cpp
index 3af8828..5589ff5 100644
--- a/services/audioflinger/MelReporter.cpp
+++ b/services/audioflinger/MelReporter.cpp
@@ -184,9 +184,9 @@
mSoundDoseManager->getOrCreateProcessorForDevice(
device.first,
patch.streamHandle,
- outputThread->mSampleRate,
- outputThread->mChannelCount,
- outputThread->mFormat));
+ outputThread->sampleRate(),
+ outputThread->channelCount(),
+ outputThread->format()));
}
}
}
diff --git a/services/audioflinger/MmapTracks.h b/services/audioflinger/MmapTracks.h
index 081af74..16c141f 100644
--- a/services/audioflinger/MmapTracks.h
+++ b/services/audioflinger/MmapTracks.h
@@ -22,7 +22,7 @@
// playback track
class MmapTrack : public TrackBase, public IAfMmapTrack {
public:
- MmapTrack(AudioFlinger::ThreadBase* thread,
+ MmapTrack(IAfThreadBase* thread,
const audio_attributes_t& attr,
uint32_t sampleRate,
audio_format_t format,
@@ -60,7 +60,7 @@
*/
void processMuteEvent_l(const sp<IAudioManager>& audioManager,
mute_state_t muteState)
- REQUIRES(AudioFlinger::MmapPlaybackThread::mLock) final;
+ /* REQUIRES(MmapPlaybackThread::mLock) */ final;
private:
friend class MmapThread;
@@ -82,9 +82,9 @@
// TODO: replace PersistableBundle with own struct
// access these two variables only when holding player thread lock.
std::unique_ptr<os::PersistableBundle> mMuteEventExtras
- GUARDED_BY(AudioFlinger::MmapPlaybackThread::mLock);
+ /* GUARDED_BY(MmapPlaybackThread::mLock) */;
mute_state_t mMuteState
- GUARDED_BY(AudioFlinger::MmapPlaybackThread::mLock);
+ /* GUARDED_BY(MmapPlaybackThread::mLock) */;
}; // end of Track
} // namespace android
\ No newline at end of file
diff --git a/services/audioflinger/PatchPanel.cpp b/services/audioflinger/PatchPanel.cpp
index cb74292..9de9dc5 100644
--- a/services/audioflinger/PatchPanel.cpp
+++ b/services/audioflinger/PatchPanel.cpp
@@ -135,7 +135,7 @@
status_t AudioFlinger::PatchPanel::createAudioPatch(const struct audio_patch *patch,
audio_patch_handle_t *handle,
bool endpointPatch)
- //unlocks AudioFlinger::mLock when calling ThreadBase::sendCreateAudioPatchConfigEvent
+ //unlocks AudioFlinger::mLock when calling IAfThreadBase::sendCreateAudioPatchConfigEvent
//to avoid deadlocks if the thread loop needs to acquire AudioFlinger::mLock
//before processing the create patch request.
NO_THREAD_SAFETY_ANALYSIS
@@ -249,7 +249,7 @@
status = INVALID_OPERATION;
goto exit;
}
- sp<ThreadBase> thread =
+ const sp<IAfThreadBase> thread =
mAudioFlinger.checkPlaybackThread_l(patch->sources[1].ext.mix.handle);
if (thread == 0) {
ALOGW("%s() cannot get playback thread", __func__);
@@ -258,7 +258,7 @@
}
// existing playback thread is reused, so it is not closed when patch is cleared
newPatch.mPlayback.setThread(
- reinterpret_cast<PlaybackThread*>(thread.get()), false /*closeThread*/);
+ thread->asIAfPlaybackThread().get(), false /*closeThread*/);
} else {
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
audio_config_base_t mixerConfig = AUDIO_CONFIG_BASE_INITIALIZER;
@@ -276,7 +276,7 @@
if (patch->sinks[0].config_mask & AUDIO_PORT_CONFIG_FLAGS) {
flags = patch->sinks[0].flags.output;
}
- sp<ThreadBase> thread = mAudioFlinger.openOutput_l(
+ const sp<IAfThreadBase> thread = mAudioFlinger.openOutput_l(
patch->sinks[0].ext.device.hw_module,
&output,
&config,
@@ -289,7 +289,7 @@
status = NO_MEMORY;
goto exit;
}
- newPatch.mPlayback.setThread(reinterpret_cast<PlaybackThread*>(thread.get()));
+ newPatch.mPlayback.setThread(thread->asIAfPlaybackThread().get());
}
audio_devices_t device = patch->sources[0].ext.device.type;
String8 address = String8(patch->sources[0].ext.device.address);
@@ -323,7 +323,7 @@
== AUDIO_STREAM_VOICE_CALL) {
source = AUDIO_SOURCE_VOICE_COMMUNICATION;
}
- sp<ThreadBase> thread = mAudioFlinger.openInput_l(srcModule,
+ const sp<IAfThreadBase> thread = mAudioFlinger.openInput_l(srcModule,
&input,
&config,
device,
@@ -338,7 +338,7 @@
status = NO_MEMORY;
goto exit;
}
- newPatch.mRecord.setThread(reinterpret_cast<RecordThread*>(thread.get()));
+ newPatch.mRecord.setThread(thread->asIAfRecordThread().get());
status = newPatch.createConnections(this);
if (status != NO_ERROR) {
goto exit;
@@ -348,7 +348,7 @@
}
} else {
if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) {
- sp<ThreadBase> thread = mAudioFlinger.checkRecordThread_l(
+ sp<IAfThreadBase> thread = mAudioFlinger.checkRecordThread_l(
patch->sinks[0].ext.mix.handle);
if (thread == 0) {
thread = mAudioFlinger.checkMmapThread_l(patch->sinks[0].ext.mix.handle);
@@ -411,7 +411,7 @@
device->applyAudioPortConfig(&patch->sinks[i]);
devices.push_back(device);
}
- sp<ThreadBase> thread =
+ sp<IAfThreadBase> thread =
mAudioFlinger.checkPlaybackThread_l(patch->sources[0].ext.mix.handle);
if (thread == 0) {
thread = mAudioFlinger.checkMmapThread_l(patch->sources[0].ext.mix.handle);
@@ -735,7 +735,7 @@
/* Disconnect a patch */
status_t AudioFlinger::PatchPanel::releaseAudioPatch(audio_patch_handle_t handle)
- //unlocks AudioFlinger::mLock when calling ThreadBase::sendReleaseAudioPatchConfigEvent
+ //unlocks AudioFlinger::mLock when calling IAfThreadBase::sendReleaseAudioPatchConfigEvent
//to avoid deadlocks if the thread loop needs to acquire AudioFlinger::mLock
//before processing the release patch request.
NO_THREAD_SAFETY_ANALYSIS
@@ -767,7 +767,7 @@
if (patch.sinks[0].type == AUDIO_PORT_TYPE_MIX) {
audio_io_handle_t ioHandle = patch.sinks[0].ext.mix.handle;
- sp<ThreadBase> thread = mAudioFlinger.checkRecordThread_l(ioHandle);
+ sp<IAfThreadBase> thread = mAudioFlinger.checkRecordThread_l(ioHandle);
if (thread == 0) {
thread = mAudioFlinger.checkMmapThread_l(ioHandle);
if (thread == 0) {
@@ -790,7 +790,7 @@
break;
}
audio_io_handle_t ioHandle = src.ext.mix.handle;
- sp<ThreadBase> thread = mAudioFlinger.checkPlaybackThread_l(ioHandle);
+ sp<IAfThreadBase> thread = mAudioFlinger.checkPlaybackThread_l(ioHandle);
if (thread == 0) {
thread = mAudioFlinger.checkMmapThread_l(ioHandle);
if (thread == 0) {
diff --git a/services/audioflinger/PatchPanel.h b/services/audioflinger/PatchPanel.h
index e693486..4bb11b0 100644
--- a/services/audioflinger/PatchPanel.h
+++ b/services/audioflinger/PatchPanel.h
@@ -199,8 +199,8 @@
return mRecord.handle() != AUDIO_PATCH_HANDLE_NONE ||
mPlayback.handle() != AUDIO_PATCH_HANDLE_NONE; }
- void setThread(const sp<ThreadBase>& thread) { mThread = thread; }
- wp<ThreadBase> thread() const { return mThread; }
+ void setThread(const sp<IAfThreadBase>& thread) { mThread = thread; }
+ wp<IAfThreadBase> thread() const { return mThread; }
// returns the latency of the patch (from record to playback).
status_t getLatencyMs(double *latencyMs) const;
@@ -216,11 +216,11 @@
// the objects are created by createConnections() and released by clearConnections()
// playback thread is created if no existing playback thread can be used
// connects playback thread output to sink device
- Endpoint<PlaybackThread, IAfPatchTrack> mPlayback;
+ Endpoint<IAfPlaybackThread, IAfPatchTrack> mPlayback;
// connects source device to record thread input
- Endpoint<RecordThread, IAfPatchRecord> mRecord;
+ Endpoint<IAfRecordThread, IAfPatchRecord> mRecord;
- wp<ThreadBase> mThread;
+ wp<IAfThreadBase> mThread;
bool mIsEndpointPatch;
};
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index c549f3f..6a2887d 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -29,13 +29,13 @@
bool hasOpPlayAudio() const;
static sp<OpPlayAudioMonitor> createIfNeeded(
- AudioFlinger::ThreadBase* thread,
+ IAfThreadBase* thread,
const AttributionSourceState& attributionSource,
const audio_attributes_t& attr, int id,
audio_stream_type_t streamType);
private:
- OpPlayAudioMonitor(AudioFlinger::ThreadBase* thread,
+ OpPlayAudioMonitor(IAfThreadBase* thread,
const AttributionSourceState& attributionSource,
audio_usage_t usage, int id, uid_t uid);
void onFirstRef() override;
@@ -56,7 +56,7 @@
// called by PlayAudioOpCallback when OP_PLAY_AUDIO is updated in AppOp callback
void checkPlayAudioForUsage();
- wp<AudioFlinger::ThreadBase> mThread;
+ wp<IAfThreadBase> mThread;
std::atomic_bool mHasOpPlayAudio;
const AttributionSourceState mAttributionSource;
const int32_t mUsage; // on purpose not audio_usage_t because always checked in appOps as int32_t
@@ -68,7 +68,7 @@
// playback track
class Track : public TrackBase, public virtual IAfTrack, public VolumeProvider {
public:
- Track(AudioFlinger::PlaybackThread* thread,
+ Track(IAfPlaybackThread* thread,
const sp<Client>& client,
audio_stream_type_t streamType,
const audio_attributes_t& attr,
@@ -310,7 +310,7 @@
mutable FillingStatus mFillingStatus;
int8_t mRetryCount;
- // see comment at AudioFlinger::PlaybackThread::Track::~Track for why this can't be const
+ // see comment at ~Track for why this can't be const
sp<IMemory> mSharedBuffer;
bool mResetDone;
@@ -411,8 +411,8 @@
void *mBuffer;
};
- OutputTrack(AudioFlinger::PlaybackThread* thread,
- AudioFlinger::DuplicatingThread* sourceThread,
+ OutputTrack(IAfPlaybackThread* thread,
+ IAfDuplicatingThread* sourceThread,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
@@ -457,7 +457,7 @@
Vector < Buffer* > mBufferQueue;
AudioBufferProvider::Buffer mOutBuffer;
bool mActive;
- AudioFlinger::DuplicatingThread* const mSourceThread; // for waitTimeMs() in write()
+ IAfDuplicatingThread* const mSourceThread; // for waitTimeMs() in write()
sp<AudioTrackClientProxy> mClientProxy;
/** Attributes of the source tracks.
@@ -479,7 +479,7 @@
// playback track, used by PatchPanel
class PatchTrack : public Track, public PatchTrackBase, public IAfPatchTrack {
public:
- PatchTrack(AudioFlinger::PlaybackThread* playbackThread,
+ PatchTrack(IAfPlaybackThread* playbackThread,
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_channel_mask_t channelMask,
diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h
index 9d25ba4..5cf09c5 100644
--- a/services/audioflinger/RecordTracks.h
+++ b/services/audioflinger/RecordTracks.h
@@ -24,7 +24,7 @@
// record track
class RecordTrack : public TrackBase, public virtual IAfRecordTrack {
public:
- RecordTrack(AudioFlinger::RecordThread* thread,
+ RecordTrack(IAfRecordThread* thread,
const sp<Client>& client,
const audio_attributes_t& attr,
uint32_t sampleRate,
@@ -133,7 +133,7 @@
// playback track, used by PatchPanel
class PatchRecord : public RecordTrack, public PatchTrackBase, public IAfPatchRecord {
public:
- PatchRecord(AudioFlinger::RecordThread* recordThread,
+ PatchRecord(IAfRecordThread* recordThread,
uint32_t sampleRate,
audio_channel_mask_t channelMask,
audio_format_t format,
@@ -169,7 +169,7 @@
class PassthruPatchRecord : public PatchRecord, public Source {
public:
- PassthruPatchRecord(AudioFlinger::RecordThread* recordThread,
+ PassthruPatchRecord(IAfRecordThread* recordThread,
uint32_t sampleRate,
audio_channel_mask_t channelMask,
audio_format_t format,
@@ -212,7 +212,7 @@
PassthruPatchRecord& mPassthru;
};
- sp<StreamInHalInterface> obtainStream(sp<AudioFlinger::ThreadBase>* thread);
+ sp<StreamInHalInterface> obtainStream(sp<IAfThreadBase>* thread);
PatchRecordAudioBufferProvider mPatchRecordAudioBufferProvider;
std::unique_ptr<void, decltype(free)*> mSinkBuffer; // frame size aligned continuous buffer
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 1688af4..67b8493 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -76,6 +76,8 @@
#include <media/audiohal/StreamHalInterface.h>
#include "AudioFlinger.h"
+#include "Threads.h"
+
#include <mediautils/SchedulingPolicyService.h>
#include <mediautils/ServiceUtilities.h>
@@ -120,6 +122,7 @@
namespace android {
+using audioflinger::SyncEvent;
using media::IEffectClient;
using content::AttributionSourceState;
@@ -515,7 +518,7 @@
// ----------------------------------------------------------------------------
// static
-const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type)
+const char* ThreadBase::threadTypeToString(ThreadBase::type_t type)
{
switch (type) {
case MIXER:
@@ -541,7 +544,7 @@
}
}
-AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
+ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
type_t type, bool systemReady, bool isOut)
: Thread(false /*canCallJava*/),
mType(type),
@@ -564,7 +567,7 @@
memset(&mPatch, 0, sizeof(struct audio_patch));
}
-AudioFlinger::ThreadBase::~ThreadBase()
+ThreadBase::~ThreadBase()
{
// mConfigEvents should be empty, but just in case it isn't, free the memory it owns
mConfigEvents.clear();
@@ -579,7 +582,7 @@
sendStatistics(true /* force */);
}
-status_t AudioFlinger::ThreadBase::readyToRun()
+status_t ThreadBase::readyToRun()
{
status_t status = initCheck();
if (status == NO_ERROR) {
@@ -590,7 +593,7 @@
return status;
}
-void AudioFlinger::ThreadBase::exit()
+void ThreadBase::exit()
{
ALOGV("ThreadBase::exit");
// do any cleanup required for exit to succeed
@@ -614,7 +617,7 @@
requestExitAndWait();
}
-status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
+status_t ThreadBase::setParameters(const String8& keyValuePairs)
{
ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string());
Mutex::Autolock _l(mLock);
@@ -624,7 +627,7 @@
// sendConfigEvent_l() must be called with ThreadBase::mLock held
// Can temporarily release the lock if waiting for a reply from processConfigEvents_l().
-status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event)
+status_t ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event)
NO_THREAD_SAFETY_ANALYSIS // condition variable
{
status_t status = NO_ERROR;
@@ -652,7 +655,7 @@
return status;
}
-void AudioFlinger::ThreadBase::sendIoConfigEvent(audio_io_config_event_t event, pid_t pid,
+void ThreadBase::sendIoConfigEvent(audio_io_config_event_t event, pid_t pid,
audio_port_handle_t portId)
{
Mutex::Autolock _l(mLock);
@@ -660,7 +663,7 @@
}
// sendIoConfigEvent_l() must be called with ThreadBase::mLock held
-void AudioFlinger::ThreadBase::sendIoConfigEvent_l(audio_io_config_event_t event, pid_t pid,
+void ThreadBase::sendIoConfigEvent_l(audio_io_config_event_t event, pid_t pid,
audio_port_handle_t portId)
{
// The audio statistics history is exponentially weighted to forget events
@@ -677,14 +680,14 @@
sendConfigEvent_l(configEvent);
}
-void AudioFlinger::ThreadBase::sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio, bool forApp)
+void ThreadBase::sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio, bool forApp)
{
Mutex::Autolock _l(mLock);
sendPrioConfigEvent_l(pid, tid, prio, forApp);
}
// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held
-void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(
+void ThreadBase::sendPrioConfigEvent_l(
pid_t pid, pid_t tid, int32_t prio, bool forApp)
{
sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio, forApp);
@@ -692,7 +695,7 @@
}
// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held
-status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair)
+status_t ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair)
{
sp<ConfigEvent> configEvent;
AudioParameter param(keyValuePair);
@@ -710,7 +713,7 @@
return sendConfigEvent_l(configEvent);
}
-status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent(
+status_t ThreadBase::sendCreateAudioPatchConfigEvent(
const struct audio_patch *patch,
audio_patch_handle_t *handle)
{
@@ -725,7 +728,7 @@
return status;
}
-status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent(
+status_t ThreadBase::sendReleaseAudioPatchConfigEvent(
const audio_patch_handle_t handle)
{
Mutex::Autolock _l(mLock);
@@ -733,7 +736,7 @@
return sendConfigEvent_l(configEvent);
}
-status_t AudioFlinger::ThreadBase::sendUpdateOutDeviceConfigEvent(
+status_t ThreadBase::sendUpdateOutDeviceConfigEvent(
const DeviceDescriptorBaseVector& outDevices)
{
if (type() != RECORD) {
@@ -745,7 +748,7 @@
return sendConfigEvent_l(configEvent);
}
-void AudioFlinger::ThreadBase::sendResizeBufferConfigEvent_l(int32_t maxSharedAudioHistoryMs)
+void ThreadBase::sendResizeBufferConfigEvent_l(int32_t maxSharedAudioHistoryMs)
{
ALOG_ASSERT(type() == RECORD, "sendResizeBufferConfigEvent_l() called on non record thread");
sp<ConfigEvent> configEvent =
@@ -753,27 +756,27 @@
sendConfigEvent_l(configEvent);
}
-void AudioFlinger::ThreadBase::sendCheckOutputStageEffectsEvent()
+void ThreadBase::sendCheckOutputStageEffectsEvent()
{
Mutex::Autolock _l(mLock);
sendCheckOutputStageEffectsEvent_l();
}
-void AudioFlinger::ThreadBase::sendCheckOutputStageEffectsEvent_l()
+void ThreadBase::sendCheckOutputStageEffectsEvent_l()
{
sp<ConfigEvent> configEvent =
(ConfigEvent *)new CheckOutputStageEffectsEvent();
sendConfigEvent_l(configEvent);
}
-void AudioFlinger::ThreadBase::sendHalLatencyModesChangedEvent_l()
+void ThreadBase::sendHalLatencyModesChangedEvent_l()
{
sp<ConfigEvent> configEvent = sp<HalLatencyModesChangedEvent>::make();
sendConfigEvent_l(configEvent);
}
// post condition: mConfigEvents.isEmpty()
-void AudioFlinger::ThreadBase::processConfigEvents_l()
+void ThreadBase::processConfigEvents_l()
{
bool configChanged = false;
@@ -940,7 +943,7 @@
}
}
-void AudioFlinger::ThreadBase::dump(int fd, const Vector<String16>& args)
+void ThreadBase::dump(int fd, const Vector<String16>& args)
NO_THREAD_SAFETY_ANALYSIS // conditional try lock
{
dprintf(fd, "\n%s thread %p, name %s, tid %d, type %d (%s):\n", isOutput() ? "Output" : "Input",
@@ -978,7 +981,7 @@
}
}
-void AudioFlinger::ThreadBase::dumpBase_l(int fd, const Vector<String16>& args __unused)
+void ThreadBase::dumpBase_l(int fd, const Vector<String16>& /* args */)
{
dprintf(fd, " I/O handle: %d\n", mId);
dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no");
@@ -1051,7 +1054,7 @@
}
}
-void AudioFlinger::ThreadBase::dumpEffectChains_l(int fd, const Vector<String16>& args)
+void ThreadBase::dumpEffectChains_l(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
@@ -1068,13 +1071,13 @@
}
}
-void AudioFlinger::ThreadBase::acquireWakeLock()
+void ThreadBase::acquireWakeLock()
{
Mutex::Autolock _l(mLock);
acquireWakeLock_l();
}
-String16 AudioFlinger::ThreadBase::getWakeLockTag()
+String16 ThreadBase::getWakeLockTag()
{
switch (mType) {
case MIXER:
@@ -1099,7 +1102,7 @@
}
}
-void AudioFlinger::ThreadBase::acquireWakeLock_l()
+void ThreadBase::acquireWakeLock_l()
{
getPowerManager_l();
if (mPowerManager != 0) {
@@ -1122,13 +1125,13 @@
gBoottime.getBoottimeOffset();
}
-void AudioFlinger::ThreadBase::releaseWakeLock()
+void ThreadBase::releaseWakeLock()
{
Mutex::Autolock _l(mLock);
releaseWakeLock_l();
}
-void AudioFlinger::ThreadBase::releaseWakeLock_l()
+void ThreadBase::releaseWakeLock_l()
{
gBoottime.release(mWakeLockToken);
if (mWakeLockToken != 0) {
@@ -1140,7 +1143,7 @@
}
}
-void AudioFlinger::ThreadBase::getPowerManager_l() {
+void ThreadBase::getPowerManager_l() {
if (mSystemReady && mPowerManager == 0) {
// use checkService() to avoid blocking if power service is not up yet
sp<IBinder> binder =
@@ -1154,7 +1157,7 @@
}
}
-void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<uid_t> &uids) {
+void ThreadBase::updateWakeLockUids_l(const SortedVector<uid_t>& uids) {
getPowerManager_l();
#if !LOG_NDEBUG
@@ -1181,25 +1184,25 @@
}
}
-void AudioFlinger::ThreadBase::clearPowerManager()
+void ThreadBase::clearPowerManager()
{
Mutex::Autolock _l(mLock);
releaseWakeLock_l();
mPowerManager.clear();
}
-void AudioFlinger::ThreadBase::updateOutDevices(
+void ThreadBase::updateOutDevices(
const DeviceDescriptorBaseVector& outDevices __unused)
{
ALOGE("%s should only be called in RecordThread", __func__);
}
-void AudioFlinger::ThreadBase::resizeInputBuffer_l(int32_t maxSharedAudioHistoryMs __unused)
+void ThreadBase::resizeInputBuffer_l(int32_t /* maxSharedAudioHistoryMs */)
{
ALOGE("%s should only be called in RecordThread", __func__);
}
-void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused)
+void ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& /* who */)
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
@@ -1208,7 +1211,7 @@
ALOGW("power manager service died !!!");
}
-void AudioFlinger::ThreadBase::setEffectSuspended_l(
+void ThreadBase::setEffectSuspended_l(
const effect_uuid_t *type, bool suspend, audio_session_t sessionId)
{
sp<IAfEffectChain> chain = getEffectChain_l(sessionId);
@@ -1223,7 +1226,7 @@
updateSuspendedSessions_l(type, suspend, sessionId);
}
-void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<IAfEffectChain>& chain)
+void ThreadBase::checkSuspendOnAddEffectChain_l(const sp<IAfEffectChain>& chain)
{
ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId());
if (index < 0) {
@@ -1247,7 +1250,7 @@
}
}
-void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type,
+void ThreadBase::updateSuspendedSessions_l(const effect_uuid_t* type,
bool suspend,
audio_session_t sessionId)
{
@@ -1308,7 +1311,7 @@
}
}
-void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(bool enabled,
+void ThreadBase::checkSuspendOnEffectEnabled(bool enabled,
audio_session_t sessionId,
bool threadLocked)
NO_THREAD_SAFETY_ANALYSIS // manual locking
@@ -1334,7 +1337,7 @@
}
// checkEffectCompatibility_l() must be called with ThreadBase::mLock held
-status_t AudioFlinger::RecordThread::checkEffectCompatibility_l(
+status_t RecordThread::checkEffectCompatibility_l(
const effect_descriptor_t *desc, audio_session_t sessionId)
{
// No global output effect sessions on record threads
@@ -1378,7 +1381,7 @@
}
// checkEffectCompatibility_l() must be called with ThreadBase::mLock held
-status_t AudioFlinger::PlaybackThread::checkEffectCompatibility_l(
+status_t PlaybackThread::checkEffectCompatibility_l(
const effect_descriptor_t *desc, audio_session_t sessionId)
{
// no preprocessing on playback threads
@@ -1533,7 +1536,7 @@
}
// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held
-sp<IAfEffectHandle> AudioFlinger::ThreadBase::createEffect_l(
+sp<IAfEffectHandle> ThreadBase::createEffect_l(
const sp<Client>& client,
const sp<IEffectClient>& effectClient,
int32_t priority,
@@ -1638,7 +1641,7 @@
return handle;
}
-void AudioFlinger::ThreadBase::disconnectEffectHandle(IAfEffectHandle *handle,
+void ThreadBase::disconnectEffectHandle(IAfEffectHandle* handle,
bool unpinIfLast)
{
bool remove = false;
@@ -1668,7 +1671,7 @@
}
}
-void AudioFlinger::ThreadBase::onEffectEnable(const sp<IAfEffectModule>& effect) {
+void ThreadBase::onEffectEnable(const sp<IAfEffectModule>& effect) {
if (isOffloadOrMmap()) {
Mutex::Autolock _l(mLock);
broadcast_l();
@@ -1684,28 +1687,28 @@
}
}
-void AudioFlinger::ThreadBase::onEffectDisable() {
+void ThreadBase::onEffectDisable() {
if (isOffloadOrMmap()) {
Mutex::Autolock _l(mLock);
broadcast_l();
}
}
-sp<IAfEffectModule> AudioFlinger::ThreadBase::getEffect(audio_session_t sessionId,
+sp<IAfEffectModule> ThreadBase::getEffect(audio_session_t sessionId,
int effectId) const
{
Mutex::Autolock _l(mLock);
return getEffect_l(sessionId, effectId);
}
-sp<IAfEffectModule> AudioFlinger::ThreadBase::getEffect_l(audio_session_t sessionId,
+sp<IAfEffectModule> ThreadBase::getEffect_l(audio_session_t sessionId,
int effectId) const
{
sp<IAfEffectChain> chain = getEffectChain_l(sessionId);
return chain != 0 ? chain->getEffectFromId_l(effectId) : 0;
}
-std::vector<int> AudioFlinger::ThreadBase::getEffectIds_l(audio_session_t sessionId) const
+std::vector<int> ThreadBase::getEffectIds_l(audio_session_t sessionId) const
{
sp<IAfEffectChain> chain = getEffectChain_l(sessionId);
return chain != nullptr ? chain->getEffectIds() : std::vector<int>{};
@@ -1713,7 +1716,7 @@
// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and
// PlaybackThread::mLock held
-status_t AudioFlinger::ThreadBase::addEffect_l(const sp<IAfEffectModule>& effect)
+status_t ThreadBase::addEffect_l(const sp<IAfEffectModule>& effect)
{
// check for existing effect chain with the requested audio session
audio_session_t sessionId = effect->sessionId();
@@ -1758,7 +1761,7 @@
return NO_ERROR;
}
-void AudioFlinger::ThreadBase::removeEffect_l(const sp<IAfEffectModule>& effect, bool release) {
+void ThreadBase::removeEffect_l(const sp<IAfEffectModule>& effect, bool release) {
ALOGV("%s %p effect %p", __FUNCTION__, this, effect.get());
effect_descriptor_t desc = effect->desc();
@@ -1777,7 +1780,7 @@
}
}
-void AudioFlinger::ThreadBase::lockEffectChains_l(
+void ThreadBase::lockEffectChains_l(
Vector<sp<IAfEffectChain>>& effectChains)
NO_THREAD_SAFETY_ANALYSIS // calls EffectChain::lock()
{
@@ -1787,7 +1790,7 @@
}
}
-void AudioFlinger::ThreadBase::unlockEffectChains(
+void ThreadBase::unlockEffectChains(
const Vector<sp<IAfEffectChain>>& effectChains)
NO_THREAD_SAFETY_ANALYSIS // calls EffectChain::unlock()
{
@@ -1796,13 +1799,13 @@
}
}
-sp<IAfEffectChain> AudioFlinger::ThreadBase::getEffectChain(audio_session_t sessionId) const
+sp<IAfEffectChain> ThreadBase::getEffectChain(audio_session_t sessionId) const
{
Mutex::Autolock _l(mLock);
return getEffectChain_l(sessionId);
}
-sp<IAfEffectChain> AudioFlinger::ThreadBase::getEffectChain_l(audio_session_t sessionId)
+sp<IAfEffectChain> ThreadBase::getEffectChain_l(audio_session_t sessionId)
const
{
size_t size = mEffectChains.size();
@@ -1814,7 +1817,7 @@
return 0;
}
-void AudioFlinger::ThreadBase::setMode(audio_mode_t mode)
+void ThreadBase::setMode(audio_mode_t mode)
{
Mutex::Autolock _l(mLock);
size_t size = mEffectChains.size();
@@ -1823,7 +1826,7 @@
}
}
-void AudioFlinger::ThreadBase::toAudioPortConfig(struct audio_port_config *config)
+void ThreadBase::toAudioPortConfig(struct audio_port_config* config)
{
config->type = AUDIO_PORT_TYPE_MIX;
config->ext.mix.handle = mId;
@@ -1834,7 +1837,7 @@
AUDIO_PORT_CONFIG_FORMAT;
}
-void AudioFlinger::ThreadBase::systemReady()
+void ThreadBase::systemReady()
{
Mutex::Autolock _l(mLock);
if (mSystemReady) {
@@ -1849,7 +1852,7 @@
}
template <typename T>
-ssize_t AudioFlinger::ThreadBase::ActiveTracks<T>::add(const sp<T> &track) {
+ssize_t ThreadBase::ActiveTracks<T>::add(const sp<T>& track) {
ssize_t index = mActiveTracks.indexOf(track);
if (index >= 0) {
ALOGW("ActiveTracks<T>::add track %p already there", track.get());
@@ -1864,7 +1867,7 @@
}
template <typename T>
-ssize_t AudioFlinger::ThreadBase::ActiveTracks<T>::remove(const sp<T> &track) {
+ssize_t ThreadBase::ActiveTracks<T>::remove(const sp<T>& track) {
ssize_t index = mActiveTracks.remove(track);
if (index < 0) {
ALOGW("ActiveTracks<T>::remove nonexistent track %p", track.get());
@@ -1883,7 +1886,7 @@
}
template <typename T>
-void AudioFlinger::ThreadBase::ActiveTracks<T>::clear() {
+void ThreadBase::ActiveTracks<T>::clear() {
for (const sp<T> &track : mActiveTracks) {
track->endBatteryAttribution();
logTrack("clear", track);
@@ -1895,7 +1898,7 @@
}
template <typename T>
-void AudioFlinger::ThreadBase::ActiveTracks<T>::updatePowerState(
+void ThreadBase::ActiveTracks<T>::updatePowerState(
const sp<ThreadBase>& thread, bool force) {
// Updates ActiveTracks client uids to the thread wakelock.
if (mActiveTracksGeneration != mLastActiveTracksGeneration || force) {
@@ -1905,7 +1908,7 @@
}
template <typename T>
-bool AudioFlinger::ThreadBase::ActiveTracks<T>::readAndClearHasChanged() {
+bool ThreadBase::ActiveTracks<T>::readAndClearHasChanged() {
bool hasChanged = mHasChanged;
mHasChanged = false;
@@ -1918,7 +1921,7 @@
}
template <typename T>
-void AudioFlinger::ThreadBase::ActiveTracks<T>::logTrack(
+void ThreadBase::ActiveTracks<T>::logTrack(
const char *funcName, const sp<T> &track) const {
if (mLocalLog != nullptr) {
String8 result;
@@ -1927,7 +1930,7 @@
}
}
-void AudioFlinger::ThreadBase::broadcast_l()
+void ThreadBase::broadcast_l()
{
// Thread could be blocked waiting for async
// so signal it to handle state changes immediately
@@ -1939,7 +1942,7 @@
// Call only from threadLoop() or when it is idle.
// Do not call from high performance code as this may do binder rpc to the MediaMetrics service.
-void AudioFlinger::ThreadBase::sendStatistics(bool force)
+void ThreadBase::sendStatistics(bool force)
{
// Do not log if we have no stats.
// We choose the timestamp verifier because it is the most likely item to be present.
@@ -2002,7 +2005,7 @@
item->selfrecord();
}
-product_strategy_t AudioFlinger::ThreadBase::getStrategyForStream(audio_stream_type_t stream) const
+product_strategy_t ThreadBase::getStrategyForStream(audio_stream_type_t stream) const
{
if (!mAudioFlinger->isAudioPolicyReady()) {
return PRODUCT_STRATEGY_NONE;
@@ -2011,7 +2014,7 @@
}
// startMelComputation_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::ThreadBase::startMelComputation_l(
+void ThreadBase::startMelComputation_l(
const sp<audio_utils::MelProcessor>& /*processor*/)
{
// Do nothing
@@ -2019,7 +2022,7 @@
}
// stopMelComputation_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::ThreadBase::stopMelComputation_l()
+void ThreadBase::stopMelComputation_l()
{
// Do nothing
ALOGW("%s: ThreadBase does not support CSD", __func__);
@@ -2029,7 +2032,7 @@
// Playback
// ----------------------------------------------------------------------------
-AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
+PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamOut* output,
audio_io_handle_t id,
type_t type,
@@ -2129,7 +2132,7 @@
mStreamTypes[AUDIO_STREAM_CALL_ASSISTANT].mute = false;
}
-AudioFlinger::PlaybackThread::~PlaybackThread()
+PlaybackThread::~PlaybackThread()
{
mAudioFlinger->unregisterWriter(mNBLogWriter);
free(mSinkBuffer);
@@ -2140,7 +2143,7 @@
// Thread virtuals
-void AudioFlinger::PlaybackThread::onFirstRef()
+void PlaybackThread::onFirstRef()
{
if (!isStreamInitialized()) {
ALOGE("The stream is not open yet"); // This should not happen.
@@ -2155,7 +2158,7 @@
if (mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING &&
mOutput->stream->setCallback(this) == OK) {
mUseAsyncWrite = true;
- mCallbackThread = new AudioFlinger::AsyncCallbackThread(this);
+ mCallbackThread = sp<AsyncCallbackThread>::make(this);
}
if (mOutput->stream->setEventCallback(this) != OK) {
@@ -2167,14 +2170,14 @@
}
// ThreadBase virtuals
-void AudioFlinger::PlaybackThread::preExit()
+void PlaybackThread::preExit()
{
ALOGV(" preExit()");
status_t result = mOutput->stream->exit();
ALOGE_IF(result != OK, "Error when calling exit(): %d", result);
}
-void AudioFlinger::PlaybackThread::dumpTracks_l(int fd, const Vector<String16>& args __unused)
+void PlaybackThread::dumpTracks_l(int fd, const Vector<String16>& /* args */)
{
String8 result;
@@ -2239,7 +2242,7 @@
write(fd, result.string(), result.size());
}
-void AudioFlinger::PlaybackThread::dumpInternals_l(int fd, const Vector<String16>& args)
+void PlaybackThread::dumpInternals_l(int fd, const Vector<String16>& args)
{
dprintf(fd, " Master volume: %f\n", mMasterVolume);
dprintf(fd, " Master mute: %s\n", mMasterMute ? "on" : "off");
@@ -2275,7 +2278,7 @@
}
// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
-sp<IAfTrack> AudioFlinger::PlaybackThread::createTrack_l(
+sp<IAfTrack> PlaybackThread::createTrack_l(
const sp<Client>& client,
audio_stream_type_t streamType,
const audio_attributes_t& attr,
@@ -2660,7 +2663,7 @@
}
template<typename T>
-ssize_t AudioFlinger::PlaybackThread::Tracks<T>::remove(const sp<T> &track)
+ssize_t PlaybackThread::Tracks<T>::remove(const sp<T>& track)
{
const int trackId = track->id();
const ssize_t index = mTracks.remove(track);
@@ -2675,17 +2678,17 @@
return index;
}
-uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const
+uint32_t PlaybackThread::correctLatency_l(uint32_t latency) const
{
return latency;
}
-uint32_t AudioFlinger::PlaybackThread::latency() const
+uint32_t PlaybackThread::latency() const
{
Mutex::Autolock _l(mLock);
return latency_l();
}
-uint32_t AudioFlinger::PlaybackThread::latency_l() const
+uint32_t PlaybackThread::latency_l() const
{
uint32_t latency;
if (initCheck() == NO_ERROR && mOutput->stream->getLatency(&latency) == OK) {
@@ -2694,7 +2697,7 @@
return 0;
}
-void AudioFlinger::PlaybackThread::setMasterVolume(float value)
+void PlaybackThread::setMasterVolume(float value)
{
Mutex::Autolock _l(mLock);
// Don't apply master volume in SW if our HAL can do it for us.
@@ -2706,12 +2709,12 @@
}
}
-void AudioFlinger::PlaybackThread::setMasterBalance(float balance)
+void PlaybackThread::setMasterBalance(float balance)
{
mMasterBalance.store(balance);
}
-void AudioFlinger::PlaybackThread::setMasterMute(bool muted)
+void PlaybackThread::setMasterMute(bool muted)
{
if (isDuplicating()) {
return;
@@ -2726,33 +2729,33 @@
}
}
-void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value)
+void PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value)
{
Mutex::Autolock _l(mLock);
mStreamTypes[stream].volume = value;
broadcast_l();
}
-void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted)
+void PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted)
{
Mutex::Autolock _l(mLock);
mStreamTypes[stream].mute = muted;
broadcast_l();
}
-float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const
+float PlaybackThread::streamVolume(audio_stream_type_t stream) const
{
Mutex::Autolock _l(mLock);
return mStreamTypes[stream].volume;
}
-void AudioFlinger::PlaybackThread::setVolumeForOutput_l(float left, float right) const
+void PlaybackThread::setVolumeForOutput_l(float left, float right) const
{
mOutput->stream->setVolume(left, right);
}
// addTrack_l() must be called with ThreadBase::mLock held
-status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<IAfTrack>& track)
+status_t PlaybackThread::addTrack_l(const sp<IAfTrack>& track)
NO_THREAD_SAFETY_ANALYSIS // release and re-acquire mLock
{
status_t status = ALREADY_EXISTS;
@@ -2858,7 +2861,7 @@
return status;
}
-bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<IAfTrack>& track)
+bool PlaybackThread::destroyTrack_l(const sp<IAfTrack>& track)
{
track->terminate();
// active tracks are removed by threadLoop()
@@ -2876,7 +2879,7 @@
return trackActive;
}
-void AudioFlinger::PlaybackThread::removeTrack_l(const sp<IAfTrack>& track)
+void PlaybackThread::removeTrack_l(const sp<IAfTrack>& track)
{
track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
@@ -2903,7 +2906,7 @@
}
}
-String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
+String8 PlaybackThread::getParameters(const String8& keys)
{
Mutex::Autolock _l(mLock);
String8 out_s8;
@@ -2913,7 +2916,7 @@
return {};
}
-status_t AudioFlinger::DirectOutputThread::selectPresentation(int presentationId, int programId) {
+status_t DirectOutputThread::selectPresentation(int presentationId, int programId) {
Mutex::Autolock _l(mLock);
if (!isStreamInitialized()) {
return NO_INIT;
@@ -2921,7 +2924,7 @@
return mOutput->stream->selectPresentation(presentationId, programId);
}
-void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event_t event, pid_t pid,
+void PlaybackThread::ioConfigChanged(audio_io_config_event_t event, pid_t pid,
audio_port_handle_t portId) {
ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event);
sp<AudioIoDescriptor> desc;
@@ -2946,27 +2949,27 @@
mAudioFlinger->ioConfigChanged(event, desc, pid);
}
-void AudioFlinger::PlaybackThread::onWriteReady()
+void PlaybackThread::onWriteReady()
{
mCallbackThread->resetWriteBlocked();
}
-void AudioFlinger::PlaybackThread::onDrainReady()
+void PlaybackThread::onDrainReady()
{
mCallbackThread->resetDraining();
}
-void AudioFlinger::PlaybackThread::onError()
+void PlaybackThread::onError()
{
mCallbackThread->setAsyncError();
}
-void AudioFlinger::PlaybackThread::onCodecFormatChanged(
+void PlaybackThread::onCodecFormatChanged(
const std::basic_string<uint8_t>& metadataBs)
{
- wp<AudioFlinger::PlaybackThread> weakPointerThis = this;
+ const auto weakPointerThis = wp<PlaybackThread>::fromExisting(this);
std::thread([this, metadataBs, weakPointerThis]() {
- sp<AudioFlinger::PlaybackThread> playbackThread = weakPointerThis.promote();
+ const sp<PlaybackThread> playbackThread = weakPointerThis.promote();
if (playbackThread == nullptr) {
ALOGW("PlaybackThread was destroyed, skip codec format change event");
return;
@@ -2991,7 +2994,7 @@
}).detach();
}
-void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence)
+void PlaybackThread::resetWriteBlocked(uint32_t sequence)
{
Mutex::Autolock _l(mLock);
// reject out of sequence requests
@@ -3001,7 +3004,7 @@
}
}
-void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence)
+void PlaybackThread::resetDraining(uint32_t sequence)
{
Mutex::Autolock _l(mLock);
// reject out of sequence requests
@@ -3016,7 +3019,7 @@
}
}
-void AudioFlinger::PlaybackThread::readOutputParameters_l()
+void PlaybackThread::readOutputParameters_l()
{
// unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL
const audio_config_base_t audioConfig = mOutput->getAudioProperties();
@@ -3025,7 +3028,7 @@
if (!audio_is_output_channel(mChannelMask)) {
LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask);
}
- if (hasMixer() && !isValidPcmSinkChannelMask(mChannelMask)) {
+ if (hasMixer() && !AudioFlinger::isValidPcmSinkChannelMask(mChannelMask)) {
LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output",
mChannelMask);
}
@@ -3048,7 +3051,7 @@
if (!audio_is_valid_format(mFormat)) {
LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat);
}
- if (hasMixer() && !isValidPcmSinkFormat(mFormat)) {
+ if (hasMixer() && !AudioFlinger::isValidPcmSinkFormat(mFormat)) {
LOG_FATAL("HAL format %#x not supported for mixed output",
mFormat);
}
@@ -3216,7 +3219,7 @@
item.record();
}
-AudioFlinger::ThreadBase::MetadataUpdate AudioFlinger::PlaybackThread::updateMetadata_l()
+ThreadBase::MetadataUpdate PlaybackThread::updateMetadata_l()
{
if (!isStreamInitialized() || !mActiveTracks.readAndClearHasChanged()) {
return {}; // nothing to do
@@ -3233,13 +3236,13 @@
return change;
}
-void AudioFlinger::PlaybackThread::sendMetadataToBackend_l(
+void PlaybackThread::sendMetadataToBackend_l(
const StreamOutHalInterface::SourceMetadata& metadata)
{
mOutput->stream->updateSourceMetadata(metadata);
};
-status_t AudioFlinger::PlaybackThread::getRenderPosition(
+status_t PlaybackThread::getRenderPosition(
uint32_t* halFrames, uint32_t* dspFrames) const
{
if (halFrames == NULL || dspFrames == NULL) {
@@ -3267,8 +3270,7 @@
}
}
-product_strategy_t AudioFlinger::PlaybackThread::getStrategyForSession_l(
- audio_session_t sessionId) const
+product_strategy_t PlaybackThread::getStrategyForSession_l(audio_session_t sessionId) const
{
// session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
// it is moved to correct output by audio policy manager when A2DP is connected or disconnected
@@ -3285,13 +3287,13 @@
}
-AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const
+AudioStreamOut* PlaybackThread::getOutput() const
{
Mutex::Autolock _l(mLock);
return mOutput;
}
-AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput()
+AudioStreamOut* PlaybackThread::clearOutput()
{
Mutex::Autolock _l(mLock);
AudioStreamOut *output = mOutput;
@@ -3305,7 +3307,7 @@
}
// this method must always be called either with ThreadBase mLock held or inside the thread loop
-sp<StreamHalInterface> AudioFlinger::PlaybackThread::stream() const
+sp<StreamHalInterface> PlaybackThread::stream() const
{
if (mOutput == NULL) {
return NULL;
@@ -3313,12 +3315,12 @@
return mOutput->stream;
}
-uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const
+uint32_t PlaybackThread::activeSleepTimeUs() const
{
return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000);
}
-status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<audioflinger::SyncEvent>& event)
+status_t PlaybackThread::setSyncEvent(const sp<SyncEvent>& event)
{
if (!isValidSyncEvent(event)) {
return BAD_VALUE;
@@ -3337,13 +3339,12 @@
return NAME_NOT_FOUND;
}
-bool AudioFlinger::PlaybackThread::isValidSyncEvent(
- const sp<audioflinger::SyncEvent>& event) const
+bool PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const
{
return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE;
}
-void AudioFlinger::PlaybackThread::threadLoop_removeTracks(
+void PlaybackThread::threadLoop_removeTracks(
[[maybe_unused]] const Vector<sp<IAfTrack>>& tracksToRemove)
{
// Miscellaneous track cleanup when removed from the active list,
@@ -3358,7 +3359,7 @@
#endif
}
-void AudioFlinger::PlaybackThread::checkSilentMode_l()
+void PlaybackThread::checkSilentMode_l()
{
if (!mMasterMute) {
char value[PROPERTY_VALUE_MAX];
@@ -3384,7 +3385,7 @@
}
// shared by MIXER and DIRECT, overridden by DUPLICATING
-ssize_t AudioFlinger::PlaybackThread::threadLoop_write()
+ssize_t PlaybackThread::threadLoop_write()
{
LOG_HIST_TS();
mInWrite = true;
@@ -3456,7 +3457,7 @@
}
// startMelComputation_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::PlaybackThread::startMelComputation_l(
+void PlaybackThread::startMelComputation_l(
const sp<audio_utils::MelProcessor>& processor)
{
auto outputSink = static_cast<AudioStreamOutSink*>(mOutputSink.get());
@@ -3466,7 +3467,7 @@
}
// stopMelComputation_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::PlaybackThread::stopMelComputation_l()
+void PlaybackThread::stopMelComputation_l()
{
auto outputSink = static_cast<AudioStreamOutSink*>(mOutputSink.get());
if (outputSink != nullptr) {
@@ -3474,7 +3475,7 @@
}
}
-void AudioFlinger::PlaybackThread::threadLoop_drain()
+void PlaybackThread::threadLoop_drain()
{
bool supportsDrain = false;
if (mOutput->stream->supportsDrain(&supportsDrain) == OK && supportsDrain) {
@@ -3490,7 +3491,7 @@
}
}
-void AudioFlinger::PlaybackThread::threadLoop_exit()
+void PlaybackThread::threadLoop_exit()
{
{
Mutex::Autolock _l(mLock);
@@ -3526,7 +3527,7 @@
- idle sleep time
*/
-void AudioFlinger::PlaybackThread::cacheParameters_l()
+void PlaybackThread::cacheParameters_l()
{
mSinkBufferSize = mNormalFrameCount * mFrameSize;
mActiveSleepTimeUs = activeSleepTimeUs();
@@ -3543,7 +3544,7 @@
}
}
-bool AudioFlinger::PlaybackThread::invalidateTracks_l(audio_stream_type_t streamType)
+bool PlaybackThread::invalidateTracks_l(audio_stream_type_t streamType)
{
ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %zu",
this, streamType, mTracks.size());
@@ -3559,18 +3560,18 @@
return trackMatch;
}
-void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType)
+void PlaybackThread::invalidateTracks(audio_stream_type_t streamType)
{
Mutex::Autolock _l(mLock);
invalidateTracks_l(streamType);
}
-void AudioFlinger::PlaybackThread::invalidateTracks(std::set<audio_port_handle_t>& portIds) {
+void PlaybackThread::invalidateTracks(std::set<audio_port_handle_t>& portIds) {
Mutex::Autolock _l(mLock);
invalidateTracks_l(portIds);
}
-bool AudioFlinger::PlaybackThread::invalidateTracks_l(std::set<audio_port_handle_t>& portIds) {
+bool PlaybackThread::invalidateTracks_l(std::set<audio_port_handle_t>& portIds) {
bool trackMatch = false;
const size_t size = mTracks.size();
for (size_t i = 0; i < size; i++) {
@@ -3588,7 +3589,7 @@
}
// getTrackById_l must be called with holding thread lock
-IAfTrack* AudioFlinger::PlaybackThread::getTrackById_l(
+IAfTrack* PlaybackThread::getTrackById_l(
audio_port_handle_t trackPortId) {
for (size_t i = 0; i < mTracks.size(); i++) {
if (mTracks[i]->portId() == trackPortId) {
@@ -3598,7 +3599,7 @@
return nullptr;
}
-status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<IAfEffectChain>& chain)
+status_t PlaybackThread::addEffectChain_l(const sp<IAfEffectChain>& chain)
{
audio_session_t session = chain->sessionId();
sp<EffectBufferHalInterface> halInBuffer, halOutBuffer;
@@ -3734,7 +3735,7 @@
return NO_ERROR;
}
-size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<IAfEffectChain>& chain)
+size_t PlaybackThread::removeEffectChain_l(const sp<IAfEffectChain>& chain)
{
audio_session_t session = chain->sessionId();
@@ -3766,14 +3767,14 @@
return mEffectChains.size();
}
-status_t AudioFlinger::PlaybackThread::attachAuxEffect(
+status_t PlaybackThread::attachAuxEffect(
const sp<IAfTrack>& track, int EffectId)
{
Mutex::Autolock _l(mLock);
return attachAuxEffect_l(track, EffectId);
}
-status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(
+status_t PlaybackThread::attachAuxEffect_l(
const sp<IAfTrack>& track, int EffectId)
{
status_t status = NO_ERROR;
@@ -3796,7 +3797,7 @@
return status;
}
-void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
+void PlaybackThread::detachAuxEffect_l(int effectId)
{
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<IAfTrack> track = mTracks[i];
@@ -3806,7 +3807,7 @@
}
}
-bool AudioFlinger::PlaybackThread::threadLoop()
+bool PlaybackThread::threadLoop()
NO_THREAD_SAFETY_ANALYSIS // manual locking of AudioFlinger
{
aflog::setThreadWriter(mNBLogWriter.get());
@@ -3875,7 +3876,7 @@
// Here, we try for the AF lock, but do not block on it as the latency
// is more informational.
if (mAudioFlinger->mLock.tryLock() == NO_ERROR) {
- std::vector<PatchPanel::SoftwarePatch> swPatches;
+ std::vector<AudioFlinger::PatchPanel::SoftwarePatch> swPatches;
double latencyMs = 0.; // not required; initialized for clang-tidy
status_t status = INVALID_OPERATION;
audio_patch_handle_t downstreamPatchHandle = AUDIO_PATCH_HANDLE_NONE;
@@ -4462,7 +4463,7 @@
return false;
}
-void AudioFlinger::PlaybackThread::collectTimestamps_l()
+void PlaybackThread::collectTimestamps_l()
{
if (mStandby) {
mTimestampVerifier.discontinuity(discontinuityForStandbyOrFlush());
@@ -4598,7 +4599,7 @@
}
// removeTracks_l() must be called with ThreadBase::mLock held
-void AudioFlinger::PlaybackThread::removeTracks_l(const Vector<sp<IAfTrack>>& tracksToRemove)
+void PlaybackThread::removeTracks_l(const Vector<sp<IAfTrack>>& tracksToRemove)
NO_THREAD_SAFETY_ANALYSIS // release and re-acquire mLock
{
for (const auto& track : tracksToRemove) {
@@ -4640,7 +4641,7 @@
}
}
-status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp)
+status_t PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp)
{
if (mNormalSink != 0) {
ExtendedTimestamp ets;
@@ -4669,7 +4670,7 @@
// All tracks attached to a mixer with flag VOIP_RX are tied to the same
// stream type STREAM_VOICE_CALL so this will only change the HAL volume once even
// if more than one track are active
-status_t AudioFlinger::PlaybackThread::handleVoipVolume_l(float *volume)
+status_t PlaybackThread::handleVoipVolume_l(float* volume)
{
status_t result = NO_ERROR;
if ((mOutput->flags & AUDIO_OUTPUT_FLAG_VOIP_RX) != 0) {
@@ -4691,7 +4692,7 @@
return result;
}
-status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch,
+status_t MixerThread::createAudioPatch_l(const struct audio_patch* patch,
audio_patch_handle_t *handle)
{
status_t status;
@@ -4708,7 +4709,7 @@
return status;
}
-status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch,
+status_t PlaybackThread::createAudioPatch_l(const struct audio_patch *patch,
audio_patch_handle_t *handle)
{
status_t status = NO_ERROR;
@@ -4792,7 +4793,7 @@
return status;
}
-status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
+status_t MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
{
status_t status;
if (property_get_bool("af.patch_park", false /* default_value */)) {
@@ -4806,7 +4807,7 @@
return status;
}
-status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
+status_t PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
{
status_t status = NO_ERROR;
@@ -4825,19 +4826,19 @@
return status;
}
-void AudioFlinger::PlaybackThread::addPatchTrack(const sp<IAfPatchTrack>& track)
+void PlaybackThread::addPatchTrack(const sp<IAfPatchTrack>& track)
{
Mutex::Autolock _l(mLock);
mTracks.add(track);
}
-void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<IAfPatchTrack>& track)
+void PlaybackThread::deletePatchTrack(const sp<IAfPatchTrack>& track)
{
Mutex::Autolock _l(mLock);
destroyTrack_l(track);
}
-void AudioFlinger::PlaybackThread::toAudioPortConfig(struct audio_port_config *config)
+void PlaybackThread::toAudioPortConfig(struct audio_port_config* config)
{
ThreadBase::toAudioPortConfig(config);
config->role = AUDIO_PORT_ROLE_SOURCE;
@@ -4851,7 +4852,14 @@
// ----------------------------------------------------------------------------
-AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
+/* static */
+sp<IAfPlaybackThread> IAfPlaybackThread::createMixerThread(
+ const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
+ audio_io_handle_t id, bool systemReady, type_t type, audio_config_base_t* mixerConfig) {
+ return sp<MixerThread>::make(audioFlinger, output, id, systemReady, type, mixerConfig);
+}
+
+MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
audio_io_handle_t id, bool systemReady, type_t type, audio_config_base_t *mixerConfig)
: PlaybackThread(audioFlinger, output, id, type, systemReady, mixerConfig),
// mAudioMixer below
@@ -5036,7 +5044,7 @@
}
}
-AudioFlinger::MixerThread::~MixerThread()
+MixerThread::~MixerThread()
{
if (mFastMixer != 0) {
FastMixerStateQueue *sq = mFastMixer->sq();
@@ -5073,7 +5081,7 @@
delete mAudioMixer;
}
-void AudioFlinger::MixerThread::onFirstRef() {
+void MixerThread::onFirstRef() {
PlaybackThread::onFirstRef();
Mutex::Autolock _l(mLock);
@@ -5089,7 +5097,7 @@
}
}
-uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const
+uint32_t MixerThread::correctLatency_l(uint32_t latency) const
{
if (mFastMixer != 0) {
MonoPipe *pipe = (MonoPipe *)mPipeSink.get();
@@ -5098,7 +5106,7 @@
return latency;
}
-ssize_t AudioFlinger::MixerThread::threadLoop_write()
+ssize_t MixerThread::threadLoop_write()
{
// FIXME we should only do one push per cycle; confirm this is true
// Start the fast mixer if it's not already running
@@ -5141,7 +5149,7 @@
return PlaybackThread::threadLoop_write();
}
-void AudioFlinger::MixerThread::threadLoop_standby()
+void MixerThread::threadLoop_standby()
{
// Idle the fast mixer if it's currently running
if (mFastMixer != 0) {
@@ -5179,24 +5187,24 @@
PlaybackThread::threadLoop_standby();
}
-bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l()
+bool PlaybackThread::waitingAsyncCallback_l()
{
return false;
}
-bool AudioFlinger::PlaybackThread::shouldStandby_l()
+bool PlaybackThread::shouldStandby_l()
{
return !mStandby;
}
-bool AudioFlinger::PlaybackThread::waitingAsyncCallback()
+bool PlaybackThread::waitingAsyncCallback()
{
Mutex::Autolock _l(mLock);
return waitingAsyncCallback_l();
}
// shared by MIXER and DIRECT, overridden by DUPLICATING
-void AudioFlinger::PlaybackThread::threadLoop_standby()
+void PlaybackThread::threadLoop_standby()
{
ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended);
mOutput->standby();
@@ -5212,20 +5220,20 @@
setHalLatencyMode_l();
}
-void AudioFlinger::PlaybackThread::onAddNewTrack_l()
+void PlaybackThread::onAddNewTrack_l()
{
ALOGV("signal playback thread");
broadcast_l();
}
-void AudioFlinger::PlaybackThread::onAsyncError()
+void PlaybackThread::onAsyncError()
{
for (int i = AUDIO_STREAM_SYSTEM; i < (int)AUDIO_STREAM_CNT; i++) {
invalidateTracks((audio_stream_type_t)i);
}
}
-void AudioFlinger::MixerThread::threadLoop_mix()
+void MixerThread::threadLoop_mix()
{
// mix buffers...
mAudioMixer->process();
@@ -5243,7 +5251,7 @@
}
-void AudioFlinger::MixerThread::threadLoop_sleepTime()
+void MixerThread::threadLoop_sleepTime()
{
// If no tracks are ready, sleep once for the duration of an output
// buffer size, then write 0s to the output
@@ -5297,7 +5305,7 @@
}
// prepareTracks_l() must be called with ThreadBase::mLock held
-AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l(
+PlaybackThread::mixer_state MixerThread::prepareTracks_l(
Vector<sp<IAfTrack>>* tracksToRemove)
{
// clean up deleted track ids in AudioMixer before allocating new tracks
@@ -6093,7 +6101,7 @@
}
// trackCountForUid_l() must be called with ThreadBase::mLock held
-uint32_t AudioFlinger::PlaybackThread::trackCountForUid_l(uid_t uid) const
+uint32_t PlaybackThread::trackCountForUid_l(uid_t uid) const
{
uint32_t trackCount = 0;
for (size_t i = 0; i < mTracks.size() ; i++) {
@@ -6104,7 +6112,7 @@
return trackCount;
}
-bool AudioFlinger::PlaybackThread::IsTimestampAdvancing::check(AudioStreamOut * output)
+bool PlaybackThread::IsTimestampAdvancing::check(AudioStreamOut* output)
{
// Check the timestamp to see if it's advancing once every 150ms. If we check too frequently, we
// could falsely detect that the frame position has stalled due to underrun because we haven't
@@ -6128,7 +6136,7 @@
return mLatchedValue;
}
-void AudioFlinger::PlaybackThread::IsTimestampAdvancing::clear()
+void PlaybackThread::IsTimestampAdvancing::clear()
{
mLatchedValue = true;
mPreviousPosition = 0;
@@ -6136,7 +6144,7 @@
}
// isTrackAllowed_l() must be called with ThreadBase::mLock held
-bool AudioFlinger::MixerThread::isTrackAllowed_l(
+bool MixerThread::isTrackAllowed_l(
audio_channel_mask_t channelMask, audio_format_t format,
audio_session_t sessionId, uid_t uid) const
{
@@ -6156,7 +6164,7 @@
}
// checkForNewParameter_l() must be called with ThreadBase::mLock held
-bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair,
+bool MixerThread::checkForNewParameter_l(const String8& keyValuePair,
status_t& status)
{
bool reconfig = false;
@@ -6170,7 +6178,7 @@
reconfig = true;
}
if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
- if (!isValidPcmSinkFormat((audio_format_t) value)) {
+ if (!AudioFlinger::isValidPcmSinkFormat(static_cast<audio_format_t>(value))) {
status = BAD_VALUE;
} else {
// no need to save value, since it's constant
@@ -6178,7 +6186,7 @@
}
}
if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
- if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) {
+ if (!AudioFlinger::isValidPcmSinkChannelMask(static_cast<audio_channel_mask_t>(value))) {
status = BAD_VALUE;
} else {
// no need to save value, since it's constant
@@ -6236,7 +6244,7 @@
}
-void AudioFlinger::MixerThread::dumpInternals_l(int fd, const Vector<String16>& args)
+void MixerThread::dumpInternals_l(int fd, const Vector<String16>& args)
{
PlaybackThread::dumpInternals_l(fd, args);
dprintf(fd, " Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs);
@@ -6283,17 +6291,17 @@
dprintf(fd, "Supported latency modes: %s\n", toString(mSupportedLatencyModes).c_str());
}
-uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const
+uint32_t MixerThread::idleSleepTimeUs() const
{
return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2;
}
-uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const
+uint32_t MixerThread::suspendSleepTimeUs() const
{
return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000);
}
-void AudioFlinger::MixerThread::cacheParameters_l()
+void MixerThread::cacheParameters_l()
{
PlaybackThread::cacheParameters_l();
@@ -6304,11 +6312,11 @@
maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15;
}
-void AudioFlinger::MixerThread::onHalLatencyModesChanged_l() {
+void MixerThread::onHalLatencyModesChanged_l() {
mAudioFlinger->onSupportedLatencyModesChanged(mId, mSupportedLatencyModes);
}
-void AudioFlinger::MixerThread::setHalLatencyMode_l() {
+void MixerThread::setHalLatencyMode_l() {
// Only handle latency mode if:
// - mBluetoothLatencyModesEnabled is true
// - the HAL supports latency modes
@@ -6350,7 +6358,7 @@
}
}
-void AudioFlinger::MixerThread::updateHalSupportedLatencyModes_l() {
+void MixerThread::updateHalSupportedLatencyModes_l() {
if (mOutput == nullptr || mOutput->stream == nullptr) {
return;
@@ -6368,7 +6376,7 @@
}
}
-status_t AudioFlinger::MixerThread::getSupportedLatencyModes(
+status_t MixerThread::getSupportedLatencyModes(
std::vector<audio_latency_mode_t>* modes) {
if (modes == nullptr) {
return BAD_VALUE;
@@ -6378,7 +6386,7 @@
return NO_ERROR;
}
-void AudioFlinger::MixerThread::onRecommendedLatencyModeChanged(
+void MixerThread::onRecommendedLatencyModeChanged(
std::vector<audio_latency_mode_t> modes) {
Mutex::Autolock _l(mLock);
if (modes != mSupportedLatencyModes) {
@@ -6389,7 +6397,7 @@
}
}
-status_t AudioFlinger::MixerThread::setBluetoothVariableLatencyEnabled(bool enabled) {
+status_t MixerThread::setBluetoothVariableLatencyEnabled(bool enabled) {
if (mOutput == nullptr || mOutput->audioHwDev == nullptr
|| !mOutput->audioHwDev->supportsBluetoothVariableLatency()) {
return INVALID_OPERATION;
@@ -6400,7 +6408,16 @@
// ----------------------------------------------------------------------------
-AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger,
+/* static */
+sp<IAfPlaybackThread> IAfPlaybackThread::createDirectOutputThread(
+ const sp<AudioFlinger>& audioFlinger,
+ AudioStreamOut* output, audio_io_handle_t id, bool systemReady,
+ const audio_offload_info_t& offloadInfo) {
+ return sp<DirectOutputThread>::make(
+ audioFlinger, output, id, systemReady, offloadInfo);
+}
+
+DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamOut* output, audio_io_handle_t id, ThreadBase::type_t type, bool systemReady,
const audio_offload_info_t& offloadInfo)
: PlaybackThread(audioFlinger, output, id, type, systemReady)
@@ -6409,18 +6426,18 @@
setMasterBalance(audioFlinger->getMasterBalance_l());
}
-AudioFlinger::DirectOutputThread::~DirectOutputThread()
+DirectOutputThread::~DirectOutputThread()
{
}
-void AudioFlinger::DirectOutputThread::dumpInternals_l(int fd, const Vector<String16>& args)
+void DirectOutputThread::dumpInternals_l(int fd, const Vector<String16>& args)
{
PlaybackThread::dumpInternals_l(fd, args);
dprintf(fd, " Master balance: %f Left: %f Right: %f\n",
mMasterBalance.load(), mMasterBalanceLeft, mMasterBalanceRight);
}
-void AudioFlinger::DirectOutputThread::setMasterBalance(float balance)
+void DirectOutputThread::setMasterBalance(float balance)
{
Mutex::Autolock _l(mLock);
if (mMasterBalance != balance) {
@@ -6430,7 +6447,7 @@
}
}
-void AudioFlinger::DirectOutputThread::processVolume_l(IAfTrack *track, bool lastTrack)
+void DirectOutputThread::processVolume_l(IAfTrack* track, bool lastTrack)
{
float left, right;
@@ -6509,7 +6526,7 @@
}
}
-void AudioFlinger::DirectOutputThread::onAddNewTrack_l()
+void DirectOutputThread::onAddNewTrack_l()
{
sp<IAfTrack> previousTrack = mPreviousTrack.promote();
sp<IAfTrack> latestTrack = mActiveTracks.getLatest();
@@ -6534,7 +6551,7 @@
PlaybackThread::onAddNewTrack_l();
}
-AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l(
+PlaybackThread::mixer_state DirectOutputThread::prepareTracks_l(
Vector<sp<IAfTrack>>* tracksToRemove
)
{
@@ -6757,7 +6774,7 @@
return mixerStatus;
}
-void AudioFlinger::DirectOutputThread::threadLoop_mix()
+void DirectOutputThread::threadLoop_mix()
{
size_t frameCount = mFrameCount;
int8_t *curBuf = (int8_t *)mSinkBuffer;
@@ -6784,7 +6801,7 @@
mActiveTrack.clear();
}
-void AudioFlinger::DirectOutputThread::threadLoop_sleepTime()
+void DirectOutputThread::threadLoop_sleepTime()
{
// do not write to HAL when paused
if (mHwPaused || (usesHwAvSync() && mStandby)) {
@@ -6800,7 +6817,7 @@
// linear or proportional PCM direct tracks in underrun.
}
-void AudioFlinger::DirectOutputThread::threadLoop_exit()
+void DirectOutputThread::threadLoop_exit()
{
{
Mutex::Autolock _l(mLock);
@@ -6818,7 +6835,7 @@
}
// must be called with thread mutex locked
-bool AudioFlinger::DirectOutputThread::shouldStandby_l()
+bool DirectOutputThread::shouldStandby_l()
{
bool trackPaused = false;
bool trackStopped = false;
@@ -6835,7 +6852,7 @@
}
// checkForNewParameter_l() must be called with ThreadBase::mLock held
-bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair,
+bool DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair,
status_t& status)
{
bool reconfig = false;
@@ -6877,7 +6894,7 @@
return reconfig;
}
-uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const
+uint32_t DirectOutputThread::activeSleepTimeUs() const
{
uint32_t time;
if (audio_has_proportional_frames(mFormat)) {
@@ -6888,7 +6905,7 @@
return time;
}
-uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const
+uint32_t DirectOutputThread::idleSleepTimeUs() const
{
uint32_t time;
if (audio_has_proportional_frames(mFormat)) {
@@ -6899,7 +6916,7 @@
return time;
}
-uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const
+uint32_t DirectOutputThread::suspendSleepTimeUs() const
{
uint32_t time;
if (audio_has_proportional_frames(mFormat)) {
@@ -6910,7 +6927,7 @@
return time;
}
-void AudioFlinger::DirectOutputThread::cacheParameters_l()
+void DirectOutputThread::cacheParameters_l()
{
PlaybackThread::cacheParameters_l();
@@ -6926,7 +6943,7 @@
}
}
-void AudioFlinger::DirectOutputThread::flushHw_l()
+void DirectOutputThread::flushHw_l()
{
PlaybackThread::flushHw_l();
mOutput->flush();
@@ -6937,7 +6954,7 @@
mMonotonicFrameCounter.onFlush();
}
-int64_t AudioFlinger::DirectOutputThread::computeWaitTimeNs_l() const {
+int64_t DirectOutputThread::computeWaitTimeNs_l() const {
// If a VolumeShaper is active, we must wake up periodically to update volume.
const int64_t NS_PER_MS = 1000000;
return mVolumeShaperActive ?
@@ -6946,8 +6963,8 @@
// ----------------------------------------------------------------------------
-AudioFlinger::AsyncCallbackThread::AsyncCallbackThread(
- const wp<AudioFlinger::PlaybackThread>& playbackThread)
+AsyncCallbackThread::AsyncCallbackThread(
+ const wp<PlaybackThread>& playbackThread)
: Thread(false /*canCallJava*/),
mPlaybackThread(playbackThread),
mWriteAckSequence(0),
@@ -6956,16 +6973,12 @@
{
}
-AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread()
-{
-}
-
-void AudioFlinger::AsyncCallbackThread::onFirstRef()
+void AsyncCallbackThread::onFirstRef()
{
run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO);
}
-bool AudioFlinger::AsyncCallbackThread::threadLoop()
+bool AsyncCallbackThread::threadLoop()
{
while (!exitPending()) {
uint32_t writeAckSequence;
@@ -6994,7 +7007,7 @@
mAsyncError = false;
}
{
- sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote();
+ const sp<PlaybackThread> playbackThread = mPlaybackThread.promote();
if (playbackThread != 0) {
if (writeAckSequence & 1) {
playbackThread->resetWriteBlocked(writeAckSequence >> 1);
@@ -7011,7 +7024,7 @@
return false;
}
-void AudioFlinger::AsyncCallbackThread::exit()
+void AsyncCallbackThread::exit()
{
ALOGV("AsyncCallbackThread::exit");
Mutex::Autolock _l(mLock);
@@ -7019,14 +7032,14 @@
mWaitWorkCV.broadcast();
}
-void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence)
+void AsyncCallbackThread::setWriteBlocked(uint32_t sequence)
{
Mutex::Autolock _l(mLock);
// bit 0 is cleared
mWriteAckSequence = sequence << 1;
}
-void AudioFlinger::AsyncCallbackThread::resetWriteBlocked()
+void AsyncCallbackThread::resetWriteBlocked()
{
Mutex::Autolock _l(mLock);
// ignore unexpected callbacks
@@ -7036,14 +7049,14 @@
}
}
-void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence)
+void AsyncCallbackThread::setDraining(uint32_t sequence)
{
Mutex::Autolock _l(mLock);
// bit 0 is cleared
mDrainSequence = sequence << 1;
}
-void AudioFlinger::AsyncCallbackThread::resetDraining()
+void AsyncCallbackThread::resetDraining()
{
Mutex::Autolock _l(mLock);
// ignore unexpected callbacks
@@ -7053,7 +7066,7 @@
}
}
-void AudioFlinger::AsyncCallbackThread::setAsyncError()
+void AsyncCallbackThread::setAsyncError()
{
Mutex::Autolock _l(mLock);
mAsyncError = true;
@@ -7062,7 +7075,16 @@
// ----------------------------------------------------------------------------
-AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger,
+
+/* static */
+sp<IAfPlaybackThread> IAfPlaybackThread::createOffloadThread(
+ const sp<AudioFlinger>& audioFlinger,
+ AudioStreamOut* output, audio_io_handle_t id, bool systemReady,
+ const audio_offload_info_t& offloadInfo) {
+ return sp<OffloadThread>::make(audioFlinger, output, id, systemReady, offloadInfo);
+}
+
+OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamOut* output, audio_io_handle_t id, bool systemReady,
const audio_offload_info_t& offloadInfo)
: DirectOutputThread(audioFlinger, output, id, OFFLOAD, systemReady, offloadInfo),
@@ -7073,7 +7095,7 @@
mKeepWakeLock = property_get_bool("ro.audio.offload_wakelock", true /* default_value */);
}
-void AudioFlinger::OffloadThread::threadLoop_exit()
+void OffloadThread::threadLoop_exit()
{
if (mFlushPending || mHwPaused) {
// If a flush is pending or track was paused, just discard buffered data
@@ -7089,7 +7111,7 @@
PlaybackThread::threadLoop_exit();
}
-AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l(
+PlaybackThread::mixer_state OffloadThread::prepareTracks_l(
Vector<sp<IAfTrack>>* tracksToRemove
)
{
@@ -7332,7 +7354,7 @@
}
// must be called with thread mutex locked
-bool AudioFlinger::OffloadThread::waitingAsyncCallback_l()
+bool OffloadThread::waitingAsyncCallback_l()
{
ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d",
mWriteAckSequence, mDrainSequence);
@@ -7342,13 +7364,13 @@
return false;
}
-bool AudioFlinger::OffloadThread::waitingAsyncCallback()
+bool OffloadThread::waitingAsyncCallback()
{
Mutex::Autolock _l(mLock);
return waitingAsyncCallback_l();
}
-void AudioFlinger::OffloadThread::flushHw_l()
+void OffloadThread::flushHw_l()
{
DirectOutputThread::flushHw_l();
// Flush anything still waiting in the mixbuffer
@@ -7369,7 +7391,7 @@
}
}
-void AudioFlinger::OffloadThread::invalidateTracks(audio_stream_type_t streamType)
+void OffloadThread::invalidateTracks(audio_stream_type_t streamType)
{
Mutex::Autolock _l(mLock);
if (PlaybackThread::invalidateTracks_l(streamType)) {
@@ -7377,7 +7399,7 @@
}
}
-void AudioFlinger::OffloadThread::invalidateTracks(std::set<audio_port_handle_t>& portIds) {
+void OffloadThread::invalidateTracks(std::set<audio_port_handle_t>& portIds) {
Mutex::Autolock _l(mLock);
if (PlaybackThread::invalidateTracks_l(portIds)) {
mFlushPending = true;
@@ -7386,8 +7408,15 @@
// ----------------------------------------------------------------------------
-AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger,
- AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady)
+/* static */
+sp<IAfDuplicatingThread> IAfDuplicatingThread::create(
+ const sp<AudioFlinger>& audioFlinger,
+ IAfPlaybackThread* mainThread, audio_io_handle_t id, bool systemReady) {
+ return sp<DuplicatingThread>::make(audioFlinger, mainThread, id, systemReady);
+}
+
+DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger,
+ IAfPlaybackThread* mainThread, audio_io_handle_t id, bool systemReady)
: MixerThread(audioFlinger, mainThread->getOutput(), id,
systemReady, DUPLICATING),
mWaitTimeMs(UINT_MAX)
@@ -7395,14 +7424,14 @@
addOutputTrack(mainThread);
}
-AudioFlinger::DuplicatingThread::~DuplicatingThread()
+DuplicatingThread::~DuplicatingThread()
{
for (size_t i = 0; i < mOutputTracks.size(); i++) {
mOutputTracks[i]->destroy();
}
}
-void AudioFlinger::DuplicatingThread::threadLoop_mix()
+void DuplicatingThread::threadLoop_mix()
{
// mix buffers...
if (outputsReady()) {
@@ -7420,7 +7449,7 @@
mStandbyTimeNs = systemTime() + mStandbyDelayNs;
}
-void AudioFlinger::DuplicatingThread::threadLoop_sleepTime()
+void DuplicatingThread::threadLoop_sleepTime()
{
if (mSleepTimeUs == 0) {
if (mMixerStatus == MIXER_TRACKS_ENABLED) {
@@ -7440,7 +7469,7 @@
}
}
-ssize_t AudioFlinger::DuplicatingThread::threadLoop_write()
+ssize_t DuplicatingThread::threadLoop_write()
{
for (size_t i = 0; i < outputTracks.size(); i++) {
const ssize_t actualWritten = outputTracks[i]->write(mSinkBuffer, writeFrames);
@@ -7468,7 +7497,7 @@
return (ssize_t)mSinkBufferSize;
}
-void AudioFlinger::DuplicatingThread::threadLoop_standby()
+void DuplicatingThread::threadLoop_standby()
{
// DuplicatingThread implements standby by stopping all tracks
for (size_t i = 0; i < outputTracks.size(); i++) {
@@ -7476,7 +7505,7 @@
}
}
-void AudioFlinger::DuplicatingThread::dumpInternals_l(int fd, const Vector<String16>& args)
+void DuplicatingThread::dumpInternals_l(int fd, const Vector<String16>& args)
{
MixerThread::dumpInternals_l(fd, args);
@@ -7487,7 +7516,7 @@
ss << ":";
for (const auto &track : mOutputTracks) {
// TODO(b/288339104) type
- const auto thread = sp<ThreadBase>::cast(track->thread().promote());
+ const auto thread = track->thread().promote();
ss << " (" << track->id() << " : ";
if (thread.get() != nullptr) {
ss << thread.get() << ", " << thread->id();
@@ -7502,17 +7531,17 @@
write(fd, result.c_str(), result.size());
}
-void AudioFlinger::DuplicatingThread::saveOutputTracks()
+void DuplicatingThread::saveOutputTracks()
{
outputTracks = mOutputTracks;
}
-void AudioFlinger::DuplicatingThread::clearOutputTracks()
+void DuplicatingThread::clearOutputTracks()
{
outputTracks.clear();
}
-void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
+void DuplicatingThread::addOutputTrack(IAfPlaybackThread* thread)
{
Mutex::Autolock _l(mLock);
// The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass.
@@ -7549,7 +7578,7 @@
updateWaitTime_l();
}
-void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread)
+void DuplicatingThread::removeOutputTrack(IAfPlaybackThread* thread)
{
Mutex::Autolock _l(mLock);
for (size_t i = 0; i < mOutputTracks.size(); i++) {
@@ -7567,12 +7596,12 @@
}
// caller must hold mLock
-void AudioFlinger::DuplicatingThread::updateWaitTime_l()
+void DuplicatingThread::updateWaitTime_l()
{
mWaitTimeMs = UINT_MAX;
for (size_t i = 0; i < mOutputTracks.size(); i++) {
// TODO(b/288339104) type
- const auto strong = sp<ThreadBase>::cast(mOutputTracks[i]->thread().promote());
+ const auto strong = mOutputTracks[i]->thread().promote();
if (strong != 0) {
uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
if (waitTimeMs < mWaitTimeMs) {
@@ -7582,17 +7611,16 @@
}
}
-bool AudioFlinger::DuplicatingThread::outputsReady()
+bool DuplicatingThread::outputsReady()
{
for (size_t i = 0; i < outputTracks.size(); i++) {
- // TODO(b/288339104) type
- const auto thread = sp<ThreadBase>::cast(outputTracks[i]->thread().promote());
+ const auto thread = outputTracks[i]->thread().promote();
if (thread == 0) {
ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p",
outputTracks[i].get());
return false;
}
- PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+ IAfPlaybackThread* const playbackThread = thread->asIAfPlaybackThread().get();
// see note at standby() declaration
if (playbackThread->inStandby() && !playbackThread->isSuspended()) {
ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(),
@@ -7603,7 +7631,7 @@
return true;
}
-void AudioFlinger::DuplicatingThread::sendMetadataToBackend_l(
+void DuplicatingThread::sendMetadataToBackend_l(
const StreamOutHalInterface::SourceMetadata& metadata)
{
for (auto& outputTrack : outputTracks) { // not mOutputTracks
@@ -7611,12 +7639,12 @@
}
}
-uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const
+uint32_t DuplicatingThread::activeSleepTimeUs() const
{
return (mWaitTimeMs * 1000) / 2;
}
-void AudioFlinger::DuplicatingThread::cacheParameters_l()
+void DuplicatingThread::cacheParameters_l()
{
// updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first
updateWaitTime_l();
@@ -7626,7 +7654,17 @@
// ----------------------------------------------------------------------------
-AudioFlinger::SpatializerThread::SpatializerThread(const sp<AudioFlinger>& audioFlinger,
+/* static */
+sp<IAfPlaybackThread> IAfPlaybackThread::createSpatializerThread(
+ const sp<AudioFlinger>& audioFlinger,
+ AudioStreamOut* output,
+ audio_io_handle_t id,
+ bool systemReady,
+ audio_config_base_t* mixerConfig) {
+ return sp<SpatializerThread>::make(audioFlinger, output, id, systemReady, mixerConfig);
+}
+
+SpatializerThread::SpatializerThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamOut* output,
audio_io_handle_t id,
bool systemReady,
@@ -7635,7 +7673,7 @@
{
}
-void AudioFlinger::SpatializerThread::onFirstRef() {
+void SpatializerThread::onFirstRef() {
MixerThread::onFirstRef();
const pid_t tid = getTid();
@@ -7650,7 +7688,7 @@
}
}
-void AudioFlinger::SpatializerThread::setHalLatencyMode_l() {
+void SpatializerThread::setHalLatencyMode_l() {
// if mSupportedLatencyModes is empty, the HAL stream does not support
// latency mode control and we can exit.
if (mSupportedLatencyModes.empty()) {
@@ -7688,7 +7726,7 @@
}
}
-status_t AudioFlinger::SpatializerThread::setRequestedLatencyMode(audio_latency_mode_t mode) {
+status_t SpatializerThread::setRequestedLatencyMode(audio_latency_mode_t mode) {
if (mode != AUDIO_LATENCY_MODE_LOW && mode != AUDIO_LATENCY_MODE_FREE) {
return BAD_VALUE;
}
@@ -7697,7 +7735,7 @@
return NO_ERROR;
}
-void AudioFlinger::SpatializerThread::checkOutputStageEffects()
+void SpatializerThread::checkOutputStageEffects()
{
bool hasVirtualizer = false;
bool hasDownMixer = false;
@@ -7753,7 +7791,14 @@
// Record
// ----------------------------------------------------------------------------
-AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
+sp<IAfRecordThread> IAfRecordThread::create(const sp<AudioFlinger>& audioFlinger,
+ AudioStreamIn* input,
+ audio_io_handle_t id,
+ bool systemReady) {
+ return sp<RecordThread>::make(audioFlinger, input, id, systemReady);
+}
+
+RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamIn *input,
audio_io_handle_t id,
bool systemReady
@@ -7912,7 +7957,7 @@
// FIXME mNormalSource
}
-AudioFlinger::RecordThread::~RecordThread()
+RecordThread::~RecordThread()
{
if (mFastCapture != 0) {
FastCaptureStateQueue *sq = mFastCapture->sq();
@@ -7934,12 +7979,12 @@
free(mRsmpInBuffer);
}
-void AudioFlinger::RecordThread::onFirstRef()
+void RecordThread::onFirstRef()
{
run(mThreadName, PRIORITY_URGENT_AUDIO);
}
-void AudioFlinger::RecordThread::preExit()
+void RecordThread::preExit()
{
ALOGV(" preExit()");
Mutex::Autolock _l(mLock);
@@ -7951,7 +7996,7 @@
mStartStopCond.broadcast();
}
-bool AudioFlinger::RecordThread::threadLoop()
+bool RecordThread::threadLoop()
{
nsecs_t lastWarning = 0;
@@ -8516,7 +8561,7 @@
return false;
}
-void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby()
+void RecordThread::standbyIfNotAlreadyInStandby()
{
if (!mStandby) {
inputStandBy();
@@ -8526,7 +8571,7 @@
}
}
-void AudioFlinger::RecordThread::inputStandBy()
+void RecordThread::inputStandBy()
{
// Idle the fast capture if it's currently running
if (mFastCapture != 0) {
@@ -8567,7 +8612,7 @@
}
// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held
-sp<IAfRecordTrack> AudioFlinger::RecordThread::createRecordTrack_l(
+sp<IAfRecordTrack> RecordThread::createRecordTrack_l(
const sp<Client>& client,
const audio_attributes_t& attr,
uint32_t *pSampleRate,
@@ -8765,7 +8810,7 @@
return track;
}
-status_t AudioFlinger::RecordThread::start(IAfRecordTrack* recordTrack,
+status_t RecordThread::start(IAfRecordTrack* recordTrack,
AudioSystem::sync_event_t event,
audio_session_t triggerSession)
{
@@ -8861,9 +8906,9 @@
}
}
-void AudioFlinger::RecordThread::syncStartEventCallback(const wp<audioflinger::SyncEvent>& event)
+void RecordThread::syncStartEventCallback(const wp<SyncEvent>& event)
{
- sp<audioflinger::SyncEvent> strongEvent = event.promote();
+ const sp<SyncEvent> strongEvent = event.promote();
if (strongEvent != 0) {
sp<IAfTrackBase> ptr =
@@ -8875,7 +8920,7 @@
}
}
-bool AudioFlinger::RecordThread::stop(IAfRecordTrack* recordTrack) {
+bool RecordThread::stop(IAfRecordTrack* recordTrack) {
ALOGV("RecordThread::stop");
AutoMutex _l(mLock);
// if we're invalid, we can't be on the ActiveTracks.
@@ -8903,14 +8948,12 @@
return false;
}
-bool AudioFlinger::RecordThread::isValidSyncEvent(
- const sp<audioflinger::SyncEvent>& /* event */) const
+bool RecordThread::isValidSyncEvent(const sp<SyncEvent>& /* event */) const
{
return false;
}
-status_t AudioFlinger::RecordThread::setSyncEvent(
- const sp<audioflinger::SyncEvent>& event __unused)
+status_t RecordThread::setSyncEvent(const sp<SyncEvent>& /* event */)
{
#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future
if (!isValidSyncEvent(event)) {
@@ -8935,8 +8978,8 @@
#endif
}
-status_t AudioFlinger::RecordThread::getActiveMicrophones(
- std::vector<media::MicrophoneInfoFw>* activeMicrophones)
+status_t RecordThread::getActiveMicrophones(
+ std::vector<media::MicrophoneInfoFw>* activeMicrophones) const
{
ALOGV("RecordThread::getActiveMicrophones");
AutoMutex _l(mLock);
@@ -8947,7 +8990,7 @@
return status;
}
-status_t AudioFlinger::RecordThread::setPreferredMicrophoneDirection(
+status_t RecordThread::setPreferredMicrophoneDirection(
audio_microphone_direction_t direction)
{
ALOGV("setPreferredMicrophoneDirection(%d)", direction);
@@ -8958,7 +9001,7 @@
return mInput->stream->setPreferredMicrophoneDirection(direction);
}
-status_t AudioFlinger::RecordThread::setPreferredMicrophoneFieldDimension(float zoom)
+status_t RecordThread::setPreferredMicrophoneFieldDimension(float zoom)
{
ALOGV("setPreferredMicrophoneFieldDimension(%f)", zoom);
AutoMutex _l(mLock);
@@ -8968,14 +9011,14 @@
return mInput->stream->setPreferredMicrophoneFieldDimension(zoom);
}
-status_t AudioFlinger::RecordThread::shareAudioHistory(
+status_t RecordThread::shareAudioHistory(
const std::string& sharedAudioPackageName, audio_session_t sharedSessionId,
int64_t sharedAudioStartMs) {
AutoMutex _l(mLock);
return shareAudioHistory_l(sharedAudioPackageName, sharedSessionId, sharedAudioStartMs);
}
-status_t AudioFlinger::RecordThread::shareAudioHistory_l(
+status_t RecordThread::shareAudioHistory_l(
const std::string& sharedAudioPackageName, audio_session_t sharedSessionId,
int64_t sharedAudioStartMs) {
@@ -9015,13 +9058,13 @@
return NO_ERROR;
}
-void AudioFlinger::RecordThread::resetAudioHistory_l() {
+void RecordThread::resetAudioHistory_l() {
mSharedAudioSessionId = AUDIO_SESSION_NONE;
mSharedAudioStartFrames = -1;
mSharedAudioPackageName = "";
}
-AudioFlinger::ThreadBase::MetadataUpdate AudioFlinger::RecordThread::updateMetadata_l()
+ThreadBase::MetadataUpdate RecordThread::updateMetadata_l()
{
if (!isStreamInitialized() || !mActiveTracks.readAndClearHasChanged()) {
return {}; // nothing to do
@@ -9038,7 +9081,7 @@
}
// destroyTrack_l() must be called with ThreadBase::mLock held
-void AudioFlinger::RecordThread::destroyTrack_l(const sp<IAfRecordTrack>& track)
+void RecordThread::destroyTrack_l(const sp<IAfRecordTrack>& track)
{
track->terminate();
track->setState(IAfTrackBase::STOPPED);
@@ -9049,7 +9092,7 @@
}
}
-void AudioFlinger::RecordThread::removeTrack_l(const sp<IAfRecordTrack>& track)
+void RecordThread::removeTrack_l(const sp<IAfRecordTrack>& track)
{
String8 result;
track->appendDump(result, false /* active */);
@@ -9063,7 +9106,7 @@
}
}
-void AudioFlinger::RecordThread::dumpInternals_l(int fd, const Vector<String16>& args __unused)
+void RecordThread::dumpInternals_l(int fd, const Vector<String16>& /* args */)
{
AudioStreamIn *input = mInput;
audio_input_flags_t flags = input != NULL ? input->flags : AUDIO_INPUT_FLAG_NONE;
@@ -9091,7 +9134,7 @@
copy->dump(fd);
}
-void AudioFlinger::RecordThread::dumpTracks_l(int fd, const Vector<String16>& args __unused)
+void RecordThread::dumpTracks_l(int fd, const Vector<String16>& /* args */)
{
String8 result;
size_t numtracks = mTracks.size();
@@ -9135,7 +9178,7 @@
write(fd, result.string(), result.size());
}
-void AudioFlinger::RecordThread::setRecordSilenced(audio_port_handle_t portId, bool silenced)
+void RecordThread::setRecordSilenced(audio_port_handle_t portId, bool silenced)
{
Mutex::Autolock _l(mLock);
for (size_t i = 0; i < mTracks.size() ; i++) {
@@ -9148,8 +9191,8 @@
void ResamplerBufferProvider::reset()
{
- const auto threadBase = sp<AudioFlinger::ThreadBase>::cast(mRecordTrack->thread().promote());
- auto* const recordThread = static_cast<AudioFlinger::RecordThread *>(threadBase.get());
+ const auto threadBase = mRecordTrack->thread().promote();
+ auto* const recordThread = static_cast<RecordThread *>(threadBase->asIAfRecordThread().get());
mRsmpInUnrel = 0;
const int32_t rear = recordThread->mRsmpInRear;
ssize_t deltaFrames = 0;
@@ -9172,8 +9215,8 @@
void ResamplerBufferProvider::sync(
size_t *framesAvailable, bool *hasOverrun)
{
- const auto threadBase = sp<AudioFlinger::ThreadBase>::cast(mRecordTrack->thread().promote());
- auto* const recordThread = static_cast<AudioFlinger::RecordThread *>(threadBase.get());
+ const auto threadBase = mRecordTrack->thread().promote();
+ auto* const recordThread = static_cast<RecordThread *>(threadBase->asIAfRecordThread().get());
const int32_t rear = recordThread->mRsmpInRear;
const int32_t front = mRsmpInFront;
const ssize_t filled = audio_utils::safe_sub_overflow(rear, front);
@@ -9206,13 +9249,13 @@
status_t ResamplerBufferProvider::getNextBuffer(
AudioBufferProvider::Buffer* buffer)
{
- const auto threadBase = sp<AudioFlinger::ThreadBase>::cast(mRecordTrack->thread().promote());
+ const auto threadBase = mRecordTrack->thread().promote();
if (threadBase == 0) {
buffer->frameCount = 0;
buffer->raw = NULL;
return NOT_ENOUGH_DATA;
}
- auto* const recordThread = static_cast<AudioFlinger::RecordThread *>(threadBase.get());
+ auto* const recordThread = static_cast<RecordThread *>(threadBase->asIAfRecordThread().get());
int32_t rear = recordThread->mRsmpInRear;
int32_t front = mRsmpInFront;
ssize_t filled = audio_utils::safe_sub_overflow(rear, front);
@@ -9260,13 +9303,13 @@
buffer->frameCount = 0;
}
-void AudioFlinger::RecordThread::checkBtNrec()
+void RecordThread::checkBtNrec()
{
Mutex::Autolock _l(mLock);
checkBtNrec_l();
}
-void AudioFlinger::RecordThread::checkBtNrec_l()
+void RecordThread::checkBtNrec_l()
{
// disable AEC and NS if the device is a BT SCO headset supporting those
// pre processings
@@ -9281,7 +9324,7 @@
}
-bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair,
+bool RecordThread::checkForNewParameter_l(const String8& keyValuePair,
status_t& status)
{
bool reconfig = false;
@@ -9369,7 +9412,7 @@
return reconfig;
}
-String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
+String8 RecordThread::getParameters(const String8& keys)
{
Mutex::Autolock _l(mLock);
if (initCheck() == NO_ERROR) {
@@ -9381,7 +9424,7 @@
return {};
}
-void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event_t event, pid_t pid,
+void RecordThread::ioConfigChanged(audio_io_config_event_t event, pid_t pid,
audio_port_handle_t portId) {
sp<AudioIoDescriptor> desc;
switch (event) {
@@ -9402,7 +9445,7 @@
mAudioFlinger->ioConfigChanged(event, desc, pid);
}
-void AudioFlinger::RecordThread::readInputParameters_l()
+void RecordThread::readInputParameters_l()
{
status_t result = mInput->stream->getAudioProperties(&mSampleRate, &mChannelMask, &mHALFormat);
LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving audio properties from HAL: %d", result);
@@ -9445,7 +9488,7 @@
.record();
}
-uint32_t AudioFlinger::RecordThread::getInputFramesLost()
+uint32_t RecordThread::getInputFramesLost() const
{
Mutex::Autolock _l(mLock);
uint32_t result;
@@ -9455,7 +9498,7 @@
return 0;
}
-KeyedVector<audio_session_t, bool> AudioFlinger::RecordThread::sessionIds() const
+KeyedVector<audio_session_t, bool> RecordThread::sessionIds() const
{
KeyedVector<audio_session_t, bool> ids;
Mutex::Autolock _l(mLock);
@@ -9469,7 +9512,7 @@
return ids;
}
-AudioStreamIn* AudioFlinger::RecordThread::clearInput()
+AudioStreamIn* RecordThread::clearInput()
{
Mutex::Autolock _l(mLock);
AudioStreamIn *input = mInput;
@@ -9478,7 +9521,7 @@
}
// this method must always be called either with ThreadBase mLock held or inside the thread loop
-sp<StreamHalInterface> AudioFlinger::RecordThread::stream() const
+sp<StreamHalInterface> RecordThread::stream() const
{
if (mInput == NULL) {
return NULL;
@@ -9486,7 +9529,7 @@
return mInput->stream;
}
-status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<IAfEffectChain>& chain)
+status_t RecordThread::addEffectChain_l(const sp<IAfEffectChain>& chain)
{
ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this);
chain->setThread(this);
@@ -9504,7 +9547,7 @@
return NO_ERROR;
}
-size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<IAfEffectChain>& chain)
+size_t RecordThread::removeEffectChain_l(const sp<IAfEffectChain>& chain)
{
ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this);
@@ -9517,7 +9560,7 @@
return mEffectChains.size();
}
-status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch,
+status_t RecordThread::createAudioPatch_l(const struct audio_patch* patch,
audio_patch_handle_t *handle)
{
status_t status = NO_ERROR;
@@ -9574,7 +9617,7 @@
return status;
}
-status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
+status_t RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
{
status_t status = NO_ERROR;
@@ -9593,7 +9636,7 @@
return status;
}
-void AudioFlinger::RecordThread::updateOutDevices(const DeviceDescriptorBaseVector& outDevices)
+void RecordThread::updateOutDevices(const DeviceDescriptorBaseVector& outDevices)
{
Mutex::Autolock _l(mLock);
mOutDevices = outDevices;
@@ -9603,7 +9646,7 @@
}
}
-int32_t AudioFlinger::RecordThread::getOldestFront_l()
+int32_t RecordThread::getOldestFront_l()
{
if (mTracks.size() == 0) {
return mRsmpInRear;
@@ -9625,7 +9668,7 @@
return oldestFront;
}
-void AudioFlinger::RecordThread::updateFronts_l(int32_t offset)
+void RecordThread::updateFronts_l(int32_t offset)
{
if (offset == 0) {
return;
@@ -9637,7 +9680,7 @@
}
}
-void AudioFlinger::RecordThread::resizeInputBuffer_l(int32_t maxSharedAudioHistoryMs)
+void RecordThread::resizeInputBuffer_l(int32_t maxSharedAudioHistoryMs)
{
// This is the formula for calculating the temporary buffer size.
// With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to
@@ -9661,7 +9704,7 @@
mRsmpInRear = 0;
ALOG_ASSERT(maxSharedAudioHistoryMs >= 0
- && maxSharedAudioHistoryMs <= AudioFlinger::kMaxSharedAudioHistoryMs,
+ && maxSharedAudioHistoryMs <= kMaxSharedAudioHistoryMs,
"resizeInputBuffer_l() called with invalid max shared history %d",
maxSharedAudioHistoryMs);
if (maxSharedAudioHistoryMs != 0) {
@@ -9730,7 +9773,7 @@
mRsmpInBuffer = rsmpInBuffer;
}
-void AudioFlinger::RecordThread::addPatchTrack(const sp<IAfPatchRecord>& record)
+void RecordThread::addPatchTrack(const sp<IAfPatchRecord>& record)
{
Mutex::Autolock _l(mLock);
mTracks.add(record);
@@ -9739,7 +9782,7 @@
}
}
-void AudioFlinger::RecordThread::deletePatchTrack(const sp<IAfPatchRecord>& record)
+void RecordThread::deletePatchTrack(const sp<IAfPatchRecord>& record)
{
Mutex::Autolock _l(mLock);
if (mSource == record->getSource()) {
@@ -9748,7 +9791,7 @@
destroyTrack_l(record);
}
-void AudioFlinger::RecordThread::toAudioPortConfig(struct audio_port_config *config)
+void RecordThread::toAudioPortConfig(struct audio_port_config* config)
{
ThreadBase::toAudioPortConfig(config);
config->role = AUDIO_PORT_ROLE_SINK;
@@ -9764,56 +9807,85 @@
// Mmap
// ----------------------------------------------------------------------------
-AudioFlinger::MmapThreadHandle::MmapThreadHandle(const sp<MmapThread>& thread)
+// Mmap stream control interface implementation. Each MmapThreadHandle controls one
+// MmapPlaybackThread or MmapCaptureThread instance.
+class MmapThreadHandle : public MmapStreamInterface {
+public:
+ explicit MmapThreadHandle(const sp<IAfMmapThread>& thread);
+ ~MmapThreadHandle() override;
+
+ // MmapStreamInterface virtuals
+ status_t createMmapBuffer(int32_t minSizeFrames,
+ struct audio_mmap_buffer_info* info) final;
+ status_t getMmapPosition(struct audio_mmap_position* position) final;
+ status_t getExternalPosition(uint64_t* position, int64_t* timeNanos) final;
+ status_t start(const AudioClient& client,
+ const audio_attributes_t* attr, audio_port_handle_t* handle) final;
+ status_t stop(audio_port_handle_t handle) final;
+ status_t standby() final;
+ status_t reportData(const void* buffer, size_t frameCount) final;
+private:
+ const sp<IAfMmapThread> mThread;
+};
+
+/* static */
+sp<MmapStreamInterface> IAfMmapThread::createMmapStreamInterfaceAdapter(
+ const sp<IAfMmapThread>& mmapThread) {
+ return sp<MmapThreadHandle>::make(mmapThread);
+}
+
+MmapThreadHandle::MmapThreadHandle(const sp<IAfMmapThread>& thread)
: mThread(thread)
{
assert(thread != 0); // thread must start non-null and stay non-null
}
-AudioFlinger::MmapThreadHandle::~MmapThreadHandle()
+// MmapStreamInterface could be directly implemented by MmapThread excepting this
+// special handling on adapter dtor.
+MmapThreadHandle::~MmapThreadHandle()
{
mThread->disconnect();
}
-status_t AudioFlinger::MmapThreadHandle::createMmapBuffer(int32_t minSizeFrames,
+status_t MmapThreadHandle::createMmapBuffer(int32_t minSizeFrames,
struct audio_mmap_buffer_info *info)
{
return mThread->createMmapBuffer(minSizeFrames, info);
}
-status_t AudioFlinger::MmapThreadHandle::getMmapPosition(struct audio_mmap_position *position)
+status_t MmapThreadHandle::getMmapPosition(struct audio_mmap_position* position)
{
return mThread->getMmapPosition(position);
}
-status_t AudioFlinger::MmapThreadHandle::getExternalPosition(uint64_t *position,
+status_t MmapThreadHandle::getExternalPosition(uint64_t* position,
int64_t *timeNanos) {
return mThread->getExternalPosition(position, timeNanos);
}
-status_t AudioFlinger::MmapThreadHandle::start(const AudioClient& client,
+status_t MmapThreadHandle::start(const AudioClient& client,
const audio_attributes_t *attr, audio_port_handle_t *handle)
-
{
return mThread->start(client, attr, handle);
}
-status_t AudioFlinger::MmapThreadHandle::stop(audio_port_handle_t handle)
+status_t MmapThreadHandle::stop(audio_port_handle_t handle)
{
return mThread->stop(handle);
}
-status_t AudioFlinger::MmapThreadHandle::standby()
+status_t MmapThreadHandle::standby()
{
return mThread->standby();
}
-status_t AudioFlinger::MmapThreadHandle::reportData(const void* buffer, size_t frameCount) {
+status_t MmapThreadHandle::reportData(const void* buffer, size_t frameCount)
+{
return mThread->reportData(buffer, frameCount);
}
-AudioFlinger::MmapThread::MmapThread(
+MmapThread::MmapThread(
const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
AudioHwDevice *hwDev, const sp<StreamHalInterface>& stream, bool systemReady, bool isOut)
: ThreadBase(audioFlinger, id, (isOut ? MMAP_PLAYBACK : MMAP_CAPTURE), systemReady, isOut),
@@ -9828,16 +9900,12 @@
readHalParameters_l();
}
-AudioFlinger::MmapThread::~MmapThread()
-{
-}
-
-void AudioFlinger::MmapThread::onFirstRef()
+void MmapThread::onFirstRef()
{
run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO);
}
-void AudioFlinger::MmapThread::disconnect()
+void MmapThread::disconnect()
{
ActiveTracks<IAfMmapTrack> activeTracks;
{
@@ -9858,7 +9926,7 @@
}
-void AudioFlinger::MmapThread::configure(const audio_attributes_t *attr,
+void MmapThread::configure(const audio_attributes_t* attr,
audio_stream_type_t streamType __unused,
audio_session_t sessionId,
const sp<MmapStreamCallback>& callback,
@@ -9872,7 +9940,7 @@
mPortId = portId;
}
-status_t AudioFlinger::MmapThread::createMmapBuffer(int32_t minSizeFrames,
+status_t MmapThread::createMmapBuffer(int32_t minSizeFrames,
struct audio_mmap_buffer_info *info)
{
if (mHalStream == 0) {
@@ -9882,7 +9950,7 @@
return mHalStream->createMmapBuffer(minSizeFrames, info);
}
-status_t AudioFlinger::MmapThread::getMmapPosition(struct audio_mmap_position *position)
+status_t MmapThread::getMmapPosition(struct audio_mmap_position* position) const
{
if (mHalStream == 0) {
return NO_INIT;
@@ -9890,7 +9958,7 @@
return mHalStream->getMmapPosition(position);
}
-status_t AudioFlinger::MmapThread::exitStandby_l()
+status_t MmapThread::exitStandby_l()
{
// The HAL must receive track metadata before starting the stream
updateMetadata_l();
@@ -9907,7 +9975,7 @@
return NO_ERROR;
}
-status_t AudioFlinger::MmapThread::start(const AudioClient& client,
+status_t MmapThread::start(const AudioClient& client,
const audio_attributes_t *attr,
audio_port_handle_t *handle)
{
@@ -10054,7 +10122,7 @@
return ret;
}
-status_t AudioFlinger::MmapThread::stop(audio_port_handle_t handle)
+status_t MmapThread::stop(audio_port_handle_t handle)
{
ALOGV("%s handle %d", __FUNCTION__, handle);
@@ -10108,7 +10176,7 @@
return NO_ERROR;
}
-status_t AudioFlinger::MmapThread::standby()
+status_t MmapThread::standby()
{
ALOGV("%s", __FUNCTION__);
@@ -10128,12 +10196,12 @@
return NO_ERROR;
}
-status_t AudioFlinger::MmapThread::reportData(const void* /*buffer*/, size_t /*frameCount*/) {
+status_t MmapThread::reportData(const void* /*buffer*/, size_t /*frameCount*/) {
// This is a stub implementation. The MmapPlaybackThread overrides this function.
return INVALID_OPERATION;
}
-void AudioFlinger::MmapThread::readHalParameters_l()
+void MmapThread::readHalParameters_l()
{
status_t result = mHalStream->getAudioProperties(&mSampleRate, &mChannelMask, &mHALFormat);
LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving audio properties from HAL: %d", result);
@@ -10169,7 +10237,7 @@
.record();
}
-bool AudioFlinger::MmapThread::threadLoop()
+bool MmapThread::threadLoop()
{
checkSilentMode_l();
@@ -10240,7 +10308,7 @@
}
// checkForNewParameter_l() must be called with ThreadBase::mLock held
-bool AudioFlinger::MmapThread::checkForNewParameter_l(const String8& keyValuePair,
+bool MmapThread::checkForNewParameter_l(const String8& keyValuePair,
status_t& status)
{
AudioParameter param = AudioParameter(keyValuePair);
@@ -10258,7 +10326,7 @@
return false;
}
-String8 AudioFlinger::MmapThread::getParameters(const String8& keys)
+String8 MmapThread::getParameters(const String8& keys)
{
Mutex::Autolock _l(mLock);
String8 out_s8;
@@ -10268,7 +10336,7 @@
return {};
}
-void AudioFlinger::MmapThread::ioConfigChanged(audio_io_config_event_t event, pid_t pid,
+void MmapThread::ioConfigChanged(audio_io_config_event_t event, pid_t pid,
audio_port_handle_t portId __unused) {
sp<AudioIoDescriptor> desc;
bool isInput = false;
@@ -10293,7 +10361,7 @@
mAudioFlinger->ioConfigChanged(event, desc, pid);
}
-status_t AudioFlinger::MmapThread::createAudioPatch_l(const struct audio_patch *patch,
+status_t MmapThread::createAudioPatch_l(const struct audio_patch* patch,
audio_patch_handle_t *handle)
NO_THREAD_SAFETY_ANALYSIS // elease and re-acquire mLock
{
@@ -10384,7 +10452,7 @@
return status;
}
-status_t AudioFlinger::MmapThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
+status_t MmapThread::releaseAudioPatch_l(const audio_patch_handle_t handle)
{
status_t status = NO_ERROR;
@@ -10406,7 +10474,7 @@
return status;
}
-void AudioFlinger::MmapThread::toAudioPortConfig(struct audio_port_config *config)
+void MmapThread::toAudioPortConfig(struct audio_port_config* config)
{
ThreadBase::toAudioPortConfig(config);
if (isOutput()) {
@@ -10420,7 +10488,7 @@
}
}
-status_t AudioFlinger::MmapThread::addEffectChain_l(const sp<IAfEffectChain>& chain)
+status_t MmapThread::addEffectChain_l(const sp<IAfEffectChain>& chain)
{
audio_session_t session = chain->sessionId();
@@ -10444,7 +10512,7 @@
return NO_ERROR;
}
-size_t AudioFlinger::MmapThread::removeEffectChain_l(const sp<IAfEffectChain>& chain)
+size_t MmapThread::removeEffectChain_l(const sp<IAfEffectChain>& chain)
{
audio_session_t session = chain->sessionId();
@@ -10467,29 +10535,29 @@
return mEffectChains.size();
}
-void AudioFlinger::MmapThread::threadLoop_standby()
+void MmapThread::threadLoop_standby()
{
mHalStream->standby();
}
-void AudioFlinger::MmapThread::threadLoop_exit()
+void MmapThread::threadLoop_exit()
{
// Do not call callback->onTearDown() because it is redundant for thread exit
// and because it can cause a recursive mutex lock on stop().
}
-status_t AudioFlinger::MmapThread::setSyncEvent(const sp<audioflinger::SyncEvent>& /* event */)
+status_t MmapThread::setSyncEvent(const sp<SyncEvent>& /* event */)
{
return BAD_VALUE;
}
-bool AudioFlinger::MmapThread::isValidSyncEvent(
- const sp<audioflinger::SyncEvent>& /* event */) const
+bool MmapThread::isValidSyncEvent(
+ const sp<SyncEvent>& /* event */) const
{
return false;
}
-status_t AudioFlinger::MmapThread::checkEffectCompatibility_l(
+status_t MmapThread::checkEffectCompatibility_l(
const effect_descriptor_t *desc, audio_session_t sessionId)
{
// No global effect sessions on mmap threads
@@ -10523,7 +10591,7 @@
return NO_ERROR;
}
-void AudioFlinger::MmapThread::checkInvalidTracks_l()
+void MmapThread::checkInvalidTracks_l()
NO_THREAD_SAFETY_ANALYSIS // release and re-acquire mLock
{
sp<MmapStreamCallback> callback;
@@ -10544,7 +10612,7 @@
}
}
-void AudioFlinger::MmapThread::dumpInternals_l(int fd, const Vector<String16>& args __unused)
+void MmapThread::dumpInternals_l(int fd, const Vector<String16>& /* args */)
{
dprintf(fd, " Attributes: content type %d usage %d source %d\n",
mAttr.content_type, mAttr.usage, mAttr.source);
@@ -10554,7 +10622,7 @@
}
}
-void AudioFlinger::MmapThread::dumpTracks_l(int fd, const Vector<String16>& args __unused)
+void MmapThread::dumpTracks_l(int fd, const Vector<String16>& /* args */)
{
String8 result;
size_t numtracks = mActiveTracks.size();
@@ -10574,7 +10642,14 @@
write(fd, result.string(), result.size());
}
-AudioFlinger::MmapPlaybackThread::MmapPlaybackThread(
+/* static */
+sp<IAfMmapPlaybackThread> IAfMmapPlaybackThread::create(
+ const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
+ AudioHwDevice* hwDev, AudioStreamOut* output, bool systemReady) {
+ return sp<MmapPlaybackThread>::make(audioFlinger, id, hwDev, output, systemReady);
+}
+
+MmapPlaybackThread::MmapPlaybackThread(
const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
AudioHwDevice *hwDev, AudioStreamOut *output, bool systemReady)
: MmapThread(audioFlinger, id, hwDev, output->stream, systemReady, true /* isOut */),
@@ -10598,7 +10673,7 @@
}
}
-void AudioFlinger::MmapPlaybackThread::configure(const audio_attributes_t *attr,
+void MmapPlaybackThread::configure(const audio_attributes_t* attr,
audio_stream_type_t streamType,
audio_session_t sessionId,
const sp<MmapStreamCallback>& callback,
@@ -10609,7 +10684,7 @@
mStreamType = streamType;
}
-AudioStreamOut* AudioFlinger::MmapPlaybackThread::clearOutput()
+AudioStreamOut* MmapPlaybackThread::clearOutput()
{
Mutex::Autolock _l(mLock);
AudioStreamOut *output = mOutput;
@@ -10617,7 +10692,7 @@
return output;
}
-void AudioFlinger::MmapPlaybackThread::setMasterVolume(float value)
+void MmapPlaybackThread::setMasterVolume(float value)
{
Mutex::Autolock _l(mLock);
// Don't apply master volume in SW if our HAL can do it for us.
@@ -10629,7 +10704,7 @@
}
}
-void AudioFlinger::MmapPlaybackThread::setMasterMute(bool muted)
+void MmapPlaybackThread::setMasterMute(bool muted)
{
Mutex::Autolock _l(mLock);
// Don't apply master mute in SW if our HAL can do it for us.
@@ -10640,7 +10715,7 @@
}
}
-void AudioFlinger::MmapPlaybackThread::setStreamVolume(audio_stream_type_t stream, float value)
+void MmapPlaybackThread::setStreamVolume(audio_stream_type_t stream, float value)
{
Mutex::Autolock _l(mLock);
if (stream == mStreamType) {
@@ -10649,7 +10724,7 @@
}
}
-float AudioFlinger::MmapPlaybackThread::streamVolume(audio_stream_type_t stream) const
+float MmapPlaybackThread::streamVolume(audio_stream_type_t stream) const
{
Mutex::Autolock _l(mLock);
if (stream == mStreamType) {
@@ -10658,7 +10733,7 @@
return 0.0f;
}
-void AudioFlinger::MmapPlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted)
+void MmapPlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted)
{
Mutex::Autolock _l(mLock);
if (stream == mStreamType) {
@@ -10667,7 +10742,7 @@
}
}
-void AudioFlinger::MmapPlaybackThread::invalidateTracks(audio_stream_type_t streamType)
+void MmapPlaybackThread::invalidateTracks(audio_stream_type_t streamType)
{
Mutex::Autolock _l(mLock);
if (streamType == mStreamType) {
@@ -10678,7 +10753,7 @@
}
}
-void AudioFlinger::MmapPlaybackThread::invalidateTracks(std::set<audio_port_handle_t>& portIds)
+void MmapPlaybackThread::invalidateTracks(std::set<audio_port_handle_t>& portIds)
{
Mutex::Autolock _l(mLock);
bool trackMatch = false;
@@ -10697,7 +10772,7 @@
}
}
-void AudioFlinger::MmapPlaybackThread::processVolume_l()
+void MmapPlaybackThread::processVolume_l()
NO_THREAD_SAFETY_ANALYSIS // access of track->processMuteEvent_l
{
float volume;
@@ -10753,7 +10828,7 @@
}
}
-AudioFlinger::ThreadBase::MetadataUpdate AudioFlinger::MmapPlaybackThread::updateMetadata_l()
+ThreadBase::MetadataUpdate MmapPlaybackThread::updateMetadata_l()
{
if (!isStreamInitialized() || !mActiveTracks.readAndClearHasChanged()) {
return {}; // nothing to do
@@ -10778,7 +10853,7 @@
return change;
};
-void AudioFlinger::MmapPlaybackThread::checkSilentMode_l()
+void MmapPlaybackThread::checkSilentMode_l()
{
if (!mMasterMute) {
char value[PROPERTY_VALUE_MAX];
@@ -10795,7 +10870,7 @@
}
}
-void AudioFlinger::MmapPlaybackThread::toAudioPortConfig(struct audio_port_config *config)
+void MmapPlaybackThread::toAudioPortConfig(struct audio_port_config* config)
{
MmapThread::toAudioPortConfig(config);
if (mOutput && mOutput->flags != AUDIO_OUTPUT_FLAG_NONE) {
@@ -10804,7 +10879,7 @@
}
}
-status_t AudioFlinger::MmapPlaybackThread::getExternalPosition(uint64_t *position,
+status_t MmapPlaybackThread::getExternalPosition(uint64_t* position,
int64_t* timeNanos) const
{
if (mOutput == nullptr) {
@@ -10818,7 +10893,7 @@
return status;
}
-status_t AudioFlinger::MmapPlaybackThread::reportData(const void* buffer, size_t frameCount) {
+status_t MmapPlaybackThread::reportData(const void* buffer, size_t frameCount) {
// Send to MelProcessor for sound dose measurement.
auto processor = mMelProcessor.load();
if (processor) {
@@ -10829,7 +10904,7 @@
}
// startMelComputation_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::MmapPlaybackThread::startMelComputation_l(
+void MmapPlaybackThread::startMelComputation_l(
const sp<audio_utils::MelProcessor>& processor)
{
ALOGV("%s: starting mel processor for thread %d", __func__, id());
@@ -10843,7 +10918,7 @@
}
// stopMelComputation_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::MmapPlaybackThread::stopMelComputation_l()
+void MmapPlaybackThread::stopMelComputation_l()
{
ALOGV("%s: pausing mel processor for thread %d", __func__, id());
auto melProcessor = mMelProcessor.load();
@@ -10852,7 +10927,7 @@
}
}
-void AudioFlinger::MmapPlaybackThread::dumpInternals_l(int fd, const Vector<String16>& args)
+void MmapPlaybackThread::dumpInternals_l(int fd, const Vector<String16>& args)
{
MmapThread::dumpInternals_l(fd, args);
@@ -10861,7 +10936,14 @@
dprintf(fd, " Master volume: %f Master mute %d\n", mMasterVolume, mMasterMute);
}
-AudioFlinger::MmapCaptureThread::MmapCaptureThread(
+/* static */
+sp<IAfMmapCaptureThread> IAfMmapCaptureThread::create(
+ const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
+ AudioHwDevice* hwDev, AudioStreamIn* input, bool systemReady) {
+ return sp<MmapCaptureThread>::make(audioFlinger, id, hwDev, input, systemReady);
+}
+
+MmapCaptureThread::MmapCaptureThread(
const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
AudioHwDevice *hwDev, AudioStreamIn *input, bool systemReady)
: MmapThread(audioFlinger, id, hwDev, input->stream, systemReady, false /* isOut */),
@@ -10871,7 +10953,7 @@
mChannelCount = audio_channel_count_from_in_mask(mChannelMask);
}
-status_t AudioFlinger::MmapCaptureThread::exitStandby_l()
+status_t MmapCaptureThread::exitStandby_l()
{
{
// mInput might have been cleared by clearInput()
@@ -10882,7 +10964,7 @@
return MmapThread::exitStandby_l();
}
-AudioStreamIn* AudioFlinger::MmapCaptureThread::clearInput()
+AudioStreamIn* MmapCaptureThread::clearInput()
{
Mutex::Autolock _l(mLock);
AudioStreamIn *input = mInput;
@@ -10890,7 +10972,7 @@
return input;
}
-void AudioFlinger::MmapCaptureThread::processVolume_l()
+void MmapCaptureThread::processVolume_l()
{
bool changed = false;
bool silenced = false;
@@ -10917,7 +10999,7 @@
}
}
-AudioFlinger::ThreadBase::MetadataUpdate AudioFlinger::MmapCaptureThread::updateMetadata_l()
+ThreadBase::MetadataUpdate MmapCaptureThread::updateMetadata_l()
{
if (!isStreamInitialized() || !mActiveTracks.readAndClearHasChanged()) {
return {}; // nothing to do
@@ -10940,7 +11022,7 @@
return change;
}
-void AudioFlinger::MmapCaptureThread::setRecordSilenced(audio_port_handle_t portId, bool silenced)
+void MmapCaptureThread::setRecordSilenced(audio_port_handle_t portId, bool silenced)
{
Mutex::Autolock _l(mLock);
for (size_t i = 0; i < mActiveTracks.size() ; i++) {
@@ -10952,7 +11034,7 @@
setClientSilencedIfExists_l(portId, silenced);
}
-void AudioFlinger::MmapCaptureThread::toAudioPortConfig(struct audio_port_config *config)
+void MmapCaptureThread::toAudioPortConfig(struct audio_port_config* config)
{
MmapThread::toAudioPortConfig(config);
if (mInput && mInput->flags != AUDIO_INPUT_FLAG_NONE) {
@@ -10961,7 +11043,7 @@
}
}
-status_t AudioFlinger::MmapCaptureThread::getExternalPosition(
+status_t MmapCaptureThread::getExternalPosition(
uint64_t* position, int64_t* timeNanos) const
{
if (mInput == nullptr) {
@@ -10972,11 +11054,18 @@
// ----------------------------------------------------------------------------
-AudioFlinger::BitPerfectThread::BitPerfectThread(const sp<AudioFlinger> &audioflinger,
+/* static */
+sp<IAfPlaybackThread> IAfPlaybackThread::createBitPerfectThread(
+ const sp<AudioFlinger>& audioflinger,
+ AudioStreamOut* output, audio_io_handle_t id, bool systemReady) {
+ return sp<BitPerfectThread>::make(audioflinger, output, id, systemReady);
+}
+
+BitPerfectThread::BitPerfectThread(const sp<AudioFlinger> &audioflinger,
AudioStreamOut *output, audio_io_handle_t id, bool systemReady)
: MixerThread(audioflinger, output, id, systemReady, BIT_PERFECT) {}
-AudioFlinger::PlaybackThread::mixer_state AudioFlinger::BitPerfectThread::prepareTracks_l(
+PlaybackThread::mixer_state BitPerfectThread::prepareTracks_l(
Vector<sp<IAfTrack>>* tracksToRemove) {
mixer_state result = MixerThread::prepareTracks_l(tracksToRemove);
// If there is only one active track and it is bit-perfect, enable tee buffer.
@@ -11009,7 +11098,7 @@
return result;
}
-void AudioFlinger::BitPerfectThread::threadLoop_mix() {
+void BitPerfectThread::threadLoop_mix() {
MixerThread::threadLoop_mix();
mHasDataCopiedToSinkBuffer = mIsBitPerfect;
}
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index eaee663..eba8a40 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -15,11 +15,9 @@
** limitations under the License.
*/
-#ifndef INCLUDING_FROM_AUDIOFLINGER_H
- #error This header file should only be included from AudioFlinger.h
-#endif
+#pragma once
-public: // TODO(b/288339104) extract out of AudioFlinger class
+namespace android {
class AsyncCallbackThread;
@@ -31,6 +29,8 @@
public:
static const char *threadTypeToString(type_t type);
+ AudioFlinger* audioFlinger() const final { return mAudioFlinger.get(); }
+
ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
type_t type, bool systemReady, bool isOut);
~ThreadBase() override;
@@ -294,6 +294,7 @@
audio_format_t format() const final { return mHALFormat; }
uint32_t channelCount() const final { return mChannelCount; }
audio_channel_mask_t hapticChannelMask() const override { return AUDIO_CHANNEL_NONE; }
+ uint32_t hapticChannelCount() const override { return 0; }
uint32_t latency_l() const override { return 0; }
void setVolumeForOutput_l(float /* left */, float /* right */) const override {}
@@ -543,7 +544,7 @@
// occurs when all suspend requests are cancelled.
void setEffectSuspended_l(const effect_uuid_t *type,
bool suspend,
- audio_session_t sessionId);
+ audio_session_t sessionId) final;
// updated mSuspendedSessions when an effect is suspended or restored
void updateSuspendedSessions_l(const effect_uuid_t *type,
bool suspend,
@@ -725,7 +726,7 @@
bool isEmpty() const {
return mActiveTracks.isEmpty();
}
- ssize_t indexOf(const sp<T>& item) {
+ ssize_t indexOf(const sp<T>& item) const {
return mActiveTracks.indexOf(item);
}
sp<T> operator[](size_t index) const {
@@ -789,11 +790,14 @@
// --- PlaybackThread ---
class PlaybackThread : public ThreadBase, public virtual IAfPlaybackThread,
public StreamOutHalInterfaceCallback,
- public VolumeInterface, public StreamOutHalInterfaceEventCallback {
+ public virtual VolumeInterface, public StreamOutHalInterfaceEventCallback {
// TODO(b/288339104) remove friends
friend class OutputTrack;
friend class Track;
public:
+ sp<IAfPlaybackThread> asIAfPlaybackThread() final {
+ return sp<IAfPlaybackThread>::fromExisting(this);
+ }
// retry count before removing active track in case of underrun on offloaded thread:
// we need to make sure that AudioTrack client has enough time to send large buffers
@@ -801,7 +805,6 @@
// handled for offloaded tracks
static const int8_t kMaxTrackRetriesOffload = 20;
static const int8_t kMaxTrackStartupRetriesOffload = 100;
- static const int8_t kMaxTrackStopRetriesOffload = 2;
static constexpr uint32_t kMaxTracksPerUid = 40;
static constexpr size_t kMaxTracks = 256;
@@ -825,6 +828,10 @@
status_t checkEffectCompatibility_l(
const effect_descriptor_t* desc, audio_session_t sessionId) final;
+ void addOutputTrack_l(const sp<IAfTrack>& track) final {
+ mTracks.add(track);
+ }
+
protected:
// Code snippets that were lifted up out of threadLoop()
virtual void threadLoop_mix() = 0;
@@ -848,15 +855,18 @@
virtual void onDrainReady();
virtual void onError();
+public: // AsyncCallbackThread
void resetWriteBlocked(uint32_t sequence);
void resetDraining(uint32_t sequence);
+protected:
virtual bool waitingAsyncCallback();
virtual bool waitingAsyncCallback_l();
virtual bool shouldStandby_l();
virtual void onAddNewTrack_l();
+public: // AsyncCallbackThread
void onAsyncError(); // error reported by AsyncCallbackThread
-
+protected:
// StreamHalInterfaceCodecFormatCallback implementation
void onCodecFormatChanged(
const std::basic_string<uint8_t>& metadataBs) final;
@@ -918,6 +928,11 @@
bool isSpatialized,
bool isBitPerfect) final;
+ bool isTrackActive(const sp<IAfTrack>& track) const final {
+ return mActiveTracks.indexOf(track) >= 0;
+ }
+
+ AudioStreamOut* getOutput_l() const final { return mOutput; }
AudioStreamOut* getOutput() const final;
AudioStreamOut* clearOutput() final;
sp<StreamHalInterface> stream() const final;
@@ -968,7 +983,7 @@
// the given set if the corresponding track is found and invalidated.
void invalidateTracks(std::set<audio_port_handle_t>& portIds) override;
- size_t frameCount() const final{ return mNormalFrameCount; }
+ size_t frameCount() const final { return mNormalFrameCount; }
audio_channel_mask_t mixerChannelMask() const final {
return mMixerChannelMask;
@@ -1005,6 +1020,11 @@
audio_channel_mask_t hapticChannelMask() const final {
return mHapticChannelMask;
}
+
+ uint32_t hapticChannelCount() const final {
+ return mHapticChannelCount;
+ }
+
bool supportsHapticPlayback() const final {
return (mHapticChannelMask & AUDIO_CHANNEL_HAPTIC_ALL) != AUDIO_CHANNEL_NONE;
}
@@ -1162,7 +1182,6 @@
audio_channel_mask_t mMixerChannelMask = AUDIO_CHANNEL_NONE;
-private:
// mMasterMute is in both PlaybackThread and in AudioFlinger. When a
// PlaybackThread needs to find out if master-muted, it checks it's local
// copy rather than the one in AudioFlinger. This optimization saves a lock.
@@ -1176,7 +1195,6 @@
: mTimestampVerifier.DISCONTINUITY_MODE_CONTINUOUS;
}
-protected:
ActiveTracks<IAfTrack> mActiveTracks;
// Time to sleep between cycles when:
@@ -1205,7 +1223,7 @@
audio_patch_handle_t *handle);
virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle);
- bool usesHwAvSync() const { return (mType == DIRECT) && (mOutput != NULL)
+ bool usesHwAvSync() const final { return mType == DIRECT && mOutput != nullptr
&& mHwSupportsPause
&& (mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC); }
@@ -1216,14 +1234,13 @@
ThreadBase::invalidateTracksForAudioSession_l(sessionId, mTracks);
}
-private:
-
friend class AudioFlinger; // for numerous
DISALLOW_COPY_AND_ASSIGN(PlaybackThread);
- status_t addTrack_l(const sp<IAfTrack>& track);
- bool destroyTrack_l(const sp<IAfTrack>& track);
+ status_t addTrack_l(const sp<IAfTrack>& track) final;
+ bool destroyTrack_l(const sp<IAfTrack>& track) final;
+
void removeTrack_l(const sp<IAfTrack>& track);
void readOutputParameters_l();
@@ -1285,6 +1302,7 @@
Tracks<IAfTrack> mTracks;
stream_type_t mStreamTypes[AUDIO_STREAM_CNT];
+
AudioStreamOut *mOutput;
float mMasterVolume;
@@ -1339,19 +1357,20 @@
// Bit 0 is reset by the async callback thread calling resetDraining(). Out of sequence
// callbacks are ignored.
uint32_t mDrainSequence;
+
sp<AsyncCallbackThread> mCallbackThread;
Mutex mAudioTrackCbLock;
// Record of IAudioTrackCallback
std::map<sp<IAfTrack>, sp<media::IAudioTrackCallback>> mAudioTrackCallbacks;
-private:
// The HAL output sink is treated as non-blocking, but current implementation is blocking
sp<NBAIO_Sink> mOutputSink;
// If a fast mixer is present, the blocking pipe sink, otherwise clear
sp<NBAIO_Sink> mPipeSink;
// The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink
sp<NBAIO_Sink> mNormalSink;
+
uint32_t mScreenState; // cached copy of gScreenState
// TODO: add comment and adjust size as needed
static const size_t kFastMixerLogSize = 8 * 1024;
@@ -1375,7 +1394,8 @@
protected:
// accessed by both binder threads and within threadLoop(), lock on mutex needed
- unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available
+ uint32_t& fastTrackAvailMask_l() final { return mFastTrackAvailMask; }
+ uint32_t mFastTrackAvailMask; // bit i set if fast track [i] is available
bool mHwSupportsPause;
bool mHwPaused;
bool mFlushPending;
@@ -1549,9 +1569,13 @@
void setHalLatencyMode_l() override;
};
-class DirectOutputThread : public PlaybackThread {
+class DirectOutputThread : public PlaybackThread, public virtual IAfDirectOutputThread {
public:
+ sp<IAfDirectOutputThread> asIAfDirectOutputThread() final {
+ return sp<IAfDirectOutputThread>::fromExisting(this);
+ }
+
DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
audio_io_handle_t id, bool systemReady,
const audio_offload_info_t& offloadInfo)
@@ -1559,7 +1583,7 @@
virtual ~DirectOutputThread();
- status_t selectPresentation(int presentationId, int programId);
+ status_t selectPresentation(int presentationId, int programId) final;
// Thread virtuals
@@ -1659,11 +1683,8 @@
class AsyncCallbackThread : public Thread {
public:
-
explicit AsyncCallbackThread(const wp<PlaybackThread>& playbackThread);
- virtual ~AsyncCallbackThread();
-
// Thread virtuals
virtual bool threadLoop();
@@ -1692,16 +1713,20 @@
bool mAsyncError;
};
-class DuplicatingThread : public MixerThread {
+class DuplicatingThread : public MixerThread, public IAfDuplicatingThread {
public:
- DuplicatingThread(const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread,
+ DuplicatingThread(const sp<AudioFlinger>& audioFlinger, IAfPlaybackThread* mainThread,
audio_io_handle_t id, bool systemReady);
- virtual ~DuplicatingThread();
+ ~DuplicatingThread() override;
+
+ sp<IAfDuplicatingThread> asIAfDuplicatingThread() final {
+ return sp<IAfDuplicatingThread>::fromExisting(this);
+ }
// Thread virtuals
- void addOutputTrack(MixerThread* thread);
- void removeOutputTrack(MixerThread* thread);
- uint32_t waitTimeMs() const { return mWaitTimeMs; }
+ void addOutputTrack(IAfPlaybackThread* thread) final;
+ void removeOutputTrack(IAfPlaybackThread* thread) final;
+ uint32_t waitTimeMs() const final { return mWaitTimeMs; }
void sendMetadataToBackend_l(
const StreamOutHalInterface::SourceMetadata& metadata) override;
@@ -1777,14 +1802,16 @@
};
// record thread
-class RecordThread : public ThreadBase
+class RecordThread : public IAfRecordThread, public ThreadBase
{
// TODO(b/288339104) remove friends
friend class PassthruPatchRecord;
friend class RecordTrack;
friend class ResamplerBufferProvider;
public:
-
+ sp<IAfRecordThread> asIAfRecordThread() final {
+ return sp<IAfRecordThread>::fromExisting(this);
+ }
RecordThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamIn *input,
@@ -1794,8 +1821,8 @@
~RecordThread() override;
// no addTrack_l ?
- void destroyTrack_l(const sp<IAfRecordTrack>& track);
- void removeTrack_l(const sp<IAfRecordTrack>& track);
+ void destroyTrack_l(const sp<IAfRecordTrack>& track) final;
+ void removeTrack_l(const sp<IAfRecordTrack>& track) final;
// Thread virtuals
bool threadLoop() final;
@@ -1810,7 +1837,7 @@
sp<IMemory> pipeMemory() const final { return mPipeMemory; }
- sp<IAfRecordTrack> createRecordTrack_l(
+ sp<IAfRecordTrack> createRecordTrack_l(
const sp<Client>& client,
const audio_attributes_t& attr,
uint32_t *pSampleRate,
@@ -1825,17 +1852,19 @@
pid_t tid,
status_t *status /*non-NULL*/,
audio_port_handle_t portId,
- int32_t maxSharedAudioHistoryMs);
+ int32_t maxSharedAudioHistoryMs) final;
status_t start(IAfRecordTrack* recordTrack,
AudioSystem::sync_event_t event,
- audio_session_t triggerSession);
+ audio_session_t triggerSession) final;
// ask the thread to stop the specified track, and
// return true if the caller should then do it's part of the stopping process
- bool stop(IAfRecordTrack* recordTrack);
+ bool stop(IAfRecordTrack* recordTrack) final;
+ AudioStreamIn* getInput() const final { return mInput; }
+ AudioStreamIn* clearInput() final;
- AudioStreamIn* clearInput();
+ // TODO(b/288339104) Unify with IAfThreadBase
virtual sp<StreamHalInterface> stream() const;
@@ -1843,19 +1872,19 @@
status_t& status);
virtual void cacheParameters_l() {}
virtual String8 getParameters(const String8& keys);
- virtual void ioConfigChanged(audio_io_config_event_t event, pid_t pid = 0,
- audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
+ void ioConfigChanged(audio_io_config_event_t event, pid_t pid = 0,
+ audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) final;
virtual status_t createAudioPatch_l(const struct audio_patch *patch,
audio_patch_handle_t *handle);
virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle);
void updateOutDevices(const DeviceDescriptorBaseVector& outDevices) override;
void resizeInputBuffer_l(int32_t maxSharedAudioHistoryMs) override;
- void addPatchTrack(const sp<IAfPatchRecord>& record);
- void deletePatchTrack(const sp<IAfPatchRecord>& record);
+ void addPatchTrack(const sp<IAfPatchRecord>& record) final;
+ void deletePatchTrack(const sp<IAfPatchRecord>& record) final;
void readInputParameters_l();
- virtual uint32_t getInputFramesLost();
+ uint32_t getInputFramesLost() const final;
virtual status_t addEffectChain_l(const sp<IAfEffectChain>& chain);
virtual size_t removeEffectChain_l(const sp<IAfEffectChain>& chain);
@@ -1874,7 +1903,7 @@
static void syncStartEventCallback(const wp<audioflinger::SyncEvent>& event);
virtual size_t frameCount() const { return mFrameCount; }
- bool hasFastCapture() const { return mFastCapture != 0; }
+ bool hasFastCapture() const final { return mFastCapture != 0; }
virtual void toAudioPortConfig(struct audio_port_config *config);
virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc,
@@ -1885,20 +1914,20 @@
mActiveTracks.updatePowerState(this, true /* force */);
}
- void checkBtNrec();
+ void checkBtNrec() final;
// Sets the UID records silence
- void setRecordSilenced(audio_port_handle_t portId, bool silenced);
+ void setRecordSilenced(audio_port_handle_t portId, bool silenced) final;
- status_t getActiveMicrophones(
- std::vector<media::MicrophoneInfoFw>* activeMicrophones);
-
- status_t setPreferredMicrophoneDirection(audio_microphone_direction_t direction);
- status_t setPreferredMicrophoneFieldDimension(float zoom);
+ status_t getActiveMicrophones(
+ std::vector<media::MicrophoneInfoFw>* activeMicrophones) const final;
+ status_t setPreferredMicrophoneDirection(audio_microphone_direction_t direction) final;
+ status_t setPreferredMicrophoneFieldDimension(float zoom) final;
MetadataUpdate updateMetadata_l() override;
- bool fastTrackAvailable() const { return mFastTrackAvail; }
+ bool fastTrackAvailable() const final { return mFastTrackAvail; }
+ void setFastTrackAvailable(bool available) final { mFastTrackAvail = available; }
bool isTimestampCorrectionEnabled() const override {
// checks popcount for exactly one device.
@@ -1908,13 +1937,13 @@
&& inDeviceType() == mTimestampCorrectedDevice;
}
- status_t shareAudioHistory(const std::string& sharedAudioPackageName,
+ status_t shareAudioHistory(const std::string& sharedAudioPackageName,
audio_session_t sharedSessionId = AUDIO_SESSION_NONE,
- int64_t sharedAudioStartMs = -1);
+ int64_t sharedAudioStartMs = -1) final;
status_t shareAudioHistory_l(const std::string& sharedAudioPackageName,
audio_session_t sharedSessionId = AUDIO_SESSION_NONE,
int64_t sharedAudioStartMs = -1);
- void resetAudioHistory_l();
+ void resetAudioHistory_l() final;
bool isStreamInitialized() const final {
return !(mInput == nullptr || mInput->stream == nullptr);
@@ -2008,34 +2037,32 @@
audio_session_t mSharedAudioSessionId = AUDIO_SESSION_NONE;
};
-class MmapThread : public ThreadBase
+class MmapThread : public ThreadBase, public virtual IAfMmapThread
{
public:
MmapThread(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
AudioHwDevice *hwDev, const sp<StreamHalInterface>& stream, bool systemReady,
bool isOut);
- ~MmapThread() override;
- virtual void configure(const audio_attributes_t *attr,
+ void configure(const audio_attributes_t* attr,
audio_stream_type_t streamType,
audio_session_t sessionId,
const sp<MmapStreamCallback>& callback,
audio_port_handle_t deviceId,
- audio_port_handle_t portId);
+ audio_port_handle_t portId) override;
- void disconnect();
+ void disconnect() final;
// MmapStreamInterface for adapter.
- virtual status_t createMmapBuffer(int32_t minSizeFrames,
- struct audio_mmap_buffer_info *info);
- virtual status_t getMmapPosition(struct audio_mmap_position* position);
- virtual status_t start(const AudioClient& client,
+ status_t createMmapBuffer(int32_t minSizeFrames, struct audio_mmap_buffer_info* info) final;
+ status_t getMmapPosition(struct audio_mmap_position* position) const override;
+ status_t start(const AudioClient& client,
const audio_attributes_t *attr,
- audio_port_handle_t *handle);
- virtual status_t stop(audio_port_handle_t handle);
- virtual status_t standby();
- virtual status_t getExternalPosition(uint64_t* position, int64_t* timeNanos) const = 0;
- virtual status_t reportData(const void* buffer, size_t frameCount);
+ audio_port_handle_t* handle) final;
+ status_t stop(audio_port_handle_t handle) final;
+ status_t standby() final;
+ status_t getExternalPosition(uint64_t* position, int64_t* timeNanos) const = 0;
+ status_t reportData(const void* buffer, size_t frameCount) override;
// RefBase
void onFirstRef() final;
@@ -2082,11 +2109,11 @@
// Not in ThreadBase
virtual audio_stream_type_t streamType() const { return AUDIO_STREAM_DEFAULT; }
virtual void invalidateTracks(audio_stream_type_t /* streamType */) {}
- virtual void invalidateTracks(std::set<audio_port_handle_t>& /* portIds */) {}
+ void invalidateTracks(std::set<audio_port_handle_t>& /* portIds */) override {}
// Sets the UID records silence
- virtual void setRecordSilenced(audio_port_handle_t portId __unused,
- bool silenced __unused) {}
+ void setRecordSilenced(
+ audio_port_handle_t /* portId */, bool /* silenced */) override {}
bool isStreamInitialized() const override { return false; }
@@ -2135,12 +2162,16 @@
static constexpr int32_t kMaxNoCallbackWarnings = 5;
};
-class MmapPlaybackThread : public MmapThread, public VolumeInterface
-{
+class MmapPlaybackThread : public MmapThread, public IAfMmapPlaybackThread,
+ public virtual VolumeInterface {
public:
MmapPlaybackThread(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
AudioHwDevice *hwDev, AudioStreamOut *output, bool systemReady);
+ sp<IAfMmapPlaybackThread> asIAfMmapPlaybackThread() final {
+ return sp<IAfMmapPlaybackThread>::fromExisting(this);
+ }
+
void configure(const audio_attributes_t* attr,
audio_stream_type_t streamType,
audio_session_t sessionId,
@@ -2148,7 +2179,7 @@
audio_port_handle_t deviceId,
audio_port_handle_t portId) final;
- AudioStreamOut* clearOutput();
+ AudioStreamOut* clearOutput() final;
// VolumeInterface
void setMasterVolume(float value) final;
@@ -2195,13 +2226,17 @@
mediautils::atomic_sp<audio_utils::MelProcessor> mMelProcessor;
};
-class MmapCaptureThread : public MmapThread
+class MmapCaptureThread : public MmapThread, public IAfMmapCaptureThread
{
public:
MmapCaptureThread(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
AudioHwDevice *hwDev, AudioStreamIn *input, bool systemReady);
- AudioStreamIn* clearInput();
+ sp<IAfMmapCaptureThread> asIAfMmapCaptureThread() final {
+ return sp<IAfMmapCaptureThread>::fromExisting(this);
+ }
+
+ AudioStreamIn* clearInput() final;
status_t exitStandby_l() REQUIRES(mLock) final;
@@ -2237,4 +2272,4 @@
float mVolumeRight = 0.f;
};
-private:
+} // namespace android
diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h
index 8f31468..194a515 100644
--- a/services/audioflinger/TrackBase.h
+++ b/services/audioflinger/TrackBase.h
@@ -22,7 +22,7 @@
// base for record and playback
class TrackBase : public ExtendedAudioBufferProvider, public virtual IAfTrackBase {
public:
- TrackBase(AudioFlinger::ThreadBase* thread,
+ TrackBase(IAfThreadBase* thread,
const sp<Client>& client,
const audio_attributes_t& mAttr,
uint32_t sampleRate,
@@ -69,8 +69,7 @@
bool isSpatialized() const override { return false; }
bool isBitPerfect() const override { return false; }
- // TODO(b/288339104) type
- wp<Thread> thread() const final { return mThread; }
+ wp<IAfThreadBase> thread() const final { return mThread; }
const sp<ServerProxy>& serverProxy() const final { return mServerProxy; }
@@ -322,7 +321,7 @@
// true for Track, false for RecordTrack,
// this could be a track type if needed later
- const wp<AudioFlinger::ThreadBase> mThread;
+ const wp<IAfThreadBase> mThread;
const alloc_type mAllocType;
/*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const
sp<IMemory> mCblkMemory;
@@ -392,7 +391,7 @@
{
public:
PatchTrackBase(const sp<ClientProxy>& proxy,
- const AudioFlinger::ThreadBase& thread,
+ IAfThreadBase* thread,
const Timeout& timeout);
void setPeerTimeout(std::chrono::nanoseconds timeout) final;
void setPeerProxy(const sp<IAfPatchTrackBase>& proxy, bool holdReference) final {
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 6b16a01..2a59315 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -81,7 +81,7 @@
// TrackBase constructor must be called with AudioFlinger::mLock held
TrackBase::TrackBase(
- AudioFlinger::ThreadBase *thread,
+ IAfThreadBase *thread,
const sp<Client>& client,
const audio_attributes_t& attr,
uint32_t sampleRate,
@@ -315,15 +315,15 @@
}
PatchTrackBase::PatchTrackBase(const sp<ClientProxy>& proxy,
- const AudioFlinger::ThreadBase& thread, const Timeout& timeout)
+ IAfThreadBase* thread, const Timeout& timeout)
: mProxy(proxy)
{
if (timeout) {
setPeerTimeout(*timeout);
} else {
// Double buffer mixer
- uint64_t mixBufferNs = ((uint64_t)2 * thread.frameCount() * 1000000000) /
- thread.sampleRate();
+ uint64_t mixBufferNs = ((uint64_t)2 * thread->frameCount() * 1000000000) /
+ thread->sampleRate();
setPeerTimeout(std::chrono::nanoseconds{mixBufferNs});
}
}
@@ -559,7 +559,7 @@
// static
sp<OpPlayAudioMonitor> OpPlayAudioMonitor::createIfNeeded(
- AudioFlinger::ThreadBase* thread,
+ IAfThreadBase* thread,
const AttributionSourceState& attributionSource, const audio_attributes_t& attr, int id,
audio_stream_type_t streamType)
{
@@ -589,11 +589,10 @@
return sp<OpPlayAudioMonitor>::make(thread, attributionSource, attr.usage, id, uid);
}
-OpPlayAudioMonitor::OpPlayAudioMonitor(
- AudioFlinger::ThreadBase* thread,
- const AttributionSourceState& attributionSource,
- audio_usage_t usage, int id, uid_t uid)
- : mThread(wp<AudioFlinger::ThreadBase>::fromExisting(thread)),
+OpPlayAudioMonitor::OpPlayAudioMonitor(IAfThreadBase* thread,
+ const AttributionSourceState& attributionSource,
+ audio_usage_t usage, int id, uid_t uid)
+ : mThread(wp<IAfThreadBase>::fromExisting(thread)),
mHasOpPlayAudio(true),
mAttributionSource(attributionSource),
mUsage((int32_t)usage),
@@ -638,9 +637,9 @@
if (mHasOpPlayAudio.compare_exchange_strong(shouldChange, hasAppOps)) {
ALOGD("OpPlayAudio: track:%d usage:%d %smuted", mId, mUsage, hasAppOps ? "not " : "");
auto thread = mThread.promote();
- if (thread != nullptr && thread->type() == AudioFlinger::ThreadBase::OFFLOAD) {
+ if (thread != nullptr && thread->type() == IAfThreadBase::OFFLOAD) {
// Wake up Thread if offloaded, otherwise it may be several seconds for update.
- Mutex::Autolock _l(thread->mLock);
+ Mutex::Autolock _l(thread->mutex());
thread->broadcast_l();
}
}
@@ -676,8 +675,8 @@
#define LOG_TAG "AF::Track"
/* static */
-sp<IAfTrack> IAfTrack::create( // TODO(b/288339104) void*
- void * /* AudioFlinger::PlaybackThread */ thread,
+sp<IAfTrack> IAfTrack::create(
+ IAfPlaybackThread* thread,
const sp<Client>& client,
audio_stream_type_t streamType,
const audio_attributes_t& attr,
@@ -700,7 +699,7 @@
float speed,
bool isSpatialized,
bool isBitPerfect) {
- return sp<Track>::make(reinterpret_cast<AudioFlinger::PlaybackThread*>(thread),
+ return sp<Track>::make(thread,
client,
streamType,
attr,
@@ -725,7 +724,7 @@
// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
Track::Track(
- AudioFlinger::PlaybackThread *thread,
+ IAfPlaybackThread* thread,
const sp<Client>& client,
audio_stream_type_t streamType,
const audio_attributes_t& attr,
@@ -816,15 +815,15 @@
// race with setSyncEvent(). However, if we call it, we cannot properly start
// static fast tracks (SoundPool) immediately after stopping.
//mAudioTrackServerProxy->framesReadyIsCalledByMultipleThreads();
- ALOG_ASSERT(thread->mFastTrackAvailMask != 0);
- int i = __builtin_ctz(thread->mFastTrackAvailMask);
+ ALOG_ASSERT(thread->fastTrackAvailMask_l() != 0);
+ const int i = __builtin_ctz(thread->fastTrackAvailMask_l());
ALOG_ASSERT(0 < i && i < (int)FastMixerState::sMaxFastTracks);
// FIXME This is too eager. We allocate a fast track index before the
// fast track becomes active. Since fast tracks are a scarce resource,
// this means we are potentially denying other more important fast tracks from
// being created. It would be better to allocate the index dynamically.
mFastIndex = i;
- thread->mFastTrackAvailMask &= ~(1 << i);
+ thread->fastTrackAvailMask_l() &= ~(1 << i);
}
mServerLatencySupported = checkServerLatencySupported(format, flags);
@@ -884,10 +883,10 @@
sp<Track> keep(this);
{ // scope for mLock
bool wasActive = false;
- sp<AudioFlinger::ThreadBase> thread = mThread.promote();
+ const sp<IAfThreadBase> thread = mThread.promote();
if (thread != 0) {
- Mutex::Autolock _l(thread->mLock);
- auto* const playbackThread = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
+ Mutex::Autolock _l(thread->mutex());
+ auto* const playbackThread = thread->asIAfPlaybackThread().get();
wasActive = playbackThread->destroyTrack_l(this);
forEachTeePatchTrack_l([](const auto& patchTrack) { patchTrack->destroy(); });
}
@@ -1163,19 +1162,19 @@
ALOGV("%s(%d): calling pid %d session %d",
__func__, mId, IPCThreadState::self()->getCallingPid(), mSessionId);
- sp<AudioFlinger::ThreadBase> thread = mThread.promote();
+ const sp<IAfThreadBase> thread = mThread.promote();
if (thread != 0) {
if (isOffloaded()) {
- Mutex::Autolock _laf(thread->mAudioFlinger->mLock);
- Mutex::Autolock _lth(thread->mLock);
+ Mutex::Autolock _laf(thread->audioFlinger()->mLock);
+ Mutex::Autolock _lth(thread->mutex());
sp<IAfEffectChain> ec = thread->getEffectChain_l(mSessionId);
- if (thread->mAudioFlinger->isNonOffloadableGlobalEffectEnabled_l() ||
+ if (thread->audioFlinger()->isNonOffloadableGlobalEffectEnabled_l() ||
(ec != 0 && ec->isNonOffloadableEnabled())) {
invalidate();
return PERMISSION_DENIED;
}
}
- Mutex::Autolock _lth(thread->mLock);
+ Mutex::Autolock _lth(thread->mutex());
track_state state = mState;
// here the track could be either new, or restarted
// in both cases "unstop" the track
@@ -1207,7 +1206,7 @@
__func__, mId, (int)mThreadIoHandle);
}
- auto* const playbackThread = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
+ auto* const playbackThread = thread->asIAfPlaybackThread().get();
// states to reset position info for pcm tracks
if (audio_is_linear_pcm(mFormat)
@@ -1275,7 +1274,7 @@
}
if (status == NO_ERROR) {
// send format to AudioManager for playback activity monitoring
- sp<IAudioManager> audioManager = thread->mAudioFlinger->getOrCreateAudioManager();
+ const sp<IAudioManager> audioManager = thread->audioFlinger()->getOrCreateAudioManager();
if (audioManager && mPortId != AUDIO_PORT_HANDLE_NONE) {
std::unique_ptr<os::PersistableBundle> bundle =
std::make_unique<os::PersistableBundle>();
@@ -1297,14 +1296,14 @@
void Track::stop()
{
ALOGV("%s(%d): calling pid %d", __func__, mId, IPCThreadState::self()->getCallingPid());
- sp<AudioFlinger::ThreadBase> thread = mThread.promote();
+ const sp<IAfThreadBase> thread = mThread.promote();
if (thread != 0) {
- Mutex::Autolock _l(thread->mLock);
+ Mutex::Autolock _l(thread->mutex());
track_state state = mState;
if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) {
// If the track is not active (PAUSED and buffers full), flush buffers
- auto* const playbackThread = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
- if (playbackThread->mActiveTracks.indexOf(this) < 0) {
+ auto* const playbackThread = thread->asIAfPlaybackThread().get();
+ if (!playbackThread->isTrackActive(this)) {
reset();
mState = STOPPED;
} else if (!isFastTrack() && !isOffloaded() && !isDirect()) {
@@ -1316,7 +1315,7 @@
// move to STOPPING_2 when drain completes and then STOPPED
mState = STOPPING_1;
if (isOffloaded()) {
- mRetryCount = AudioFlinger::PlaybackThread::kMaxTrackStopRetriesOffload;
+ mRetryCount = IAfPlaybackThread::kMaxTrackStopRetriesOffload;
}
}
playbackThread->broadcast_l();
@@ -1330,10 +1329,10 @@
void Track::pause()
{
ALOGV("%s(%d): calling pid %d", __func__, mId, IPCThreadState::self()->getCallingPid());
- sp<AudioFlinger::ThreadBase> thread = mThread.promote();
+ const sp<IAfThreadBase> thread = mThread.promote();
if (thread != 0) {
- Mutex::Autolock _l(thread->mLock);
- auto* const playbackThread = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
+ Mutex::Autolock _l(thread->mutex());
+ auto* const playbackThread = thread->asIAfPlaybackThread().get();
switch (mState) {
case STOPPING_1:
case STOPPING_2:
@@ -1367,15 +1366,15 @@
void Track::flush()
{
ALOGV("%s(%d)", __func__, mId);
- sp<AudioFlinger::ThreadBase> thread = mThread.promote();
+ const sp<IAfThreadBase> thread = mThread.promote();
if (thread != 0) {
- Mutex::Autolock _l(thread->mLock);
- auto* const playbackThread = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
+ Mutex::Autolock _l(thread->mutex());
+ auto* const playbackThread = thread->asIAfPlaybackThread().get();
// Flush the ring buffer now if the track is not active in the PlaybackThread.
// Otherwise the flush would not be done until the track is resumed.
// Requires FastTrack removal be BLOCK_UNTIL_ACKED
- if (playbackThread->mActiveTracks.indexOf(this) < 0) {
+ if (!playbackThread->isTrackActive(this)) {
(void)mServerProxy->flushBufferIfNeeded();
}
@@ -1414,7 +1413,7 @@
if (isDirect()) {
mFlushHwPending = true;
}
- if (playbackThread->mActiveTracks.indexOf(this) < 0) {
+ if (!playbackThread->isTrackActive(this)) {
reset();
}
}
@@ -1465,12 +1464,12 @@
status_t Track::setParameters(const String8& keyValuePairs)
{
- sp<AudioFlinger::ThreadBase> thread = mThread.promote();
+ const sp<IAfThreadBase> thread = mThread.promote();
if (thread == 0) {
ALOGE("%s(%d): thread is dead", __func__, mId);
return FAILED_TRANSACTION;
- } else if ((thread->type() == AudioFlinger::ThreadBase::DIRECT) ||
- (thread->type() == AudioFlinger::ThreadBase::OFFLOAD)) {
+ } else if (thread->type() == IAfThreadBase::DIRECT
+ || thread->type() == IAfThreadBase::OFFLOAD) {
return thread->setParameters(keyValuePairs);
} else {
return PERMISSION_DENIED;
@@ -1479,13 +1478,13 @@
status_t Track::selectPresentation(int presentationId,
int programId) {
- sp<AudioFlinger::ThreadBase> thread = mThread.promote();
+ const sp<IAfThreadBase> thread = mThread.promote();
if (thread == 0) {
ALOGE("thread is dead");
return FAILED_TRANSACTION;
- } else if ((thread->type() == AudioFlinger::ThreadBase::DIRECT)
- || (thread->type() == AudioFlinger::ThreadBase::OFFLOAD)) {
- auto directOutputThread = static_cast<AudioFlinger::DirectOutputThread*>(thread.get());
+ } else if (thread->type() == IAfThreadBase::DIRECT
+ || thread->type() == IAfThreadBase::OFFLOAD) {
+ auto directOutputThread = thread->asIAfDirectOutputThread().get();
return directOutputThread->selectPresentation(presentationId, programId);
}
return INVALID_OPERATION;
@@ -1499,9 +1498,9 @@
if (isOffloadedOrDirect()) {
// Signal thread to fetch new volume.
- sp<AudioFlinger::ThreadBase> thread = mThread.promote();
+ const sp<IAfThreadBase> thread = mThread.promote();
if (thread != 0) {
- Mutex::Autolock _l(thread->mLock);
+ Mutex::Autolock _l(thread->mutex());
thread->broadcast_l();
}
}
@@ -1660,26 +1659,26 @@
if (!isOffloaded() && !isDirect()) {
return INVALID_OPERATION; // normal tracks handled through SSQ
}
- sp<AudioFlinger::ThreadBase> thread = mThread.promote();
+ const sp<IAfThreadBase> thread = mThread.promote();
if (thread == 0) {
return INVALID_OPERATION;
}
- Mutex::Autolock _l(thread->mLock);
- auto* const playbackThread = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
+ Mutex::Autolock _l(thread->mutex());
+ auto* const playbackThread = thread->asIAfPlaybackThread().get();
return playbackThread->getTimestamp_l(timestamp);
}
status_t Track::attachAuxEffect(int EffectId)
{
- sp<AudioFlinger::ThreadBase> thread = mThread.promote();
+ const sp<IAfThreadBase> thread = mThread.promote();
if (thread == nullptr) {
return DEAD_OBJECT;
}
- auto dstThread = sp<AudioFlinger::PlaybackThread>::cast(thread);
+ auto dstThread = thread->asIAfPlaybackThread();
// srcThread is initialized by call to moveAuxEffectToIo()
- sp<AudioFlinger::PlaybackThread> srcThread;
+ sp<IAfPlaybackThread> srcThread;
sp<AudioFlinger> af = mClient->audioFlinger();
status_t status = af->moveAuxEffectToIo(EffectId, dstThread, &srcThread);
@@ -1865,10 +1864,10 @@
void Track::signal()
{
- sp<AudioFlinger::ThreadBase> thread = mThread.promote();
+ const sp<IAfThreadBase> thread = mThread.promote();
if (thread != 0) {
- auto* const t = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
- Mutex::Autolock _l(t->mLock);
+ auto* const t = thread->asIAfPlaybackThread().get();
+ Mutex::Autolock _l(t->mutex());
t->broadcast_l();
}
}
@@ -1877,11 +1876,11 @@
{
status_t status = INVALID_OPERATION;
if (isOffloadedOrDirect()) {
- sp<AudioFlinger::ThreadBase> thread = mThread.promote();
+ const sp<IAfThreadBase> thread = mThread.promote();
if (thread != nullptr) {
- auto* const t = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
- Mutex::Autolock _l(t->mLock);
- status = t->mOutput->stream->getDualMonoMode(mode);
+ auto* const t = thread->asIAfPlaybackThread().get();
+ Mutex::Autolock _l(t->mutex());
+ status = t->getOutput_l()->stream->getDualMonoMode(mode);
ALOGD_IF((status == NO_ERROR) && (mDualMonoMode != *mode),
"%s: mode %d inconsistent", __func__, mDualMonoMode);
}
@@ -1893,11 +1892,11 @@
{
status_t status = INVALID_OPERATION;
if (isOffloadedOrDirect()) {
- sp<AudioFlinger::ThreadBase> thread = mThread.promote();
+ const sp<IAfThreadBase> thread = mThread.promote();
if (thread != nullptr) {
- auto* const t = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
- Mutex::Autolock lock(t->mLock);
- status = t->mOutput->stream->setDualMonoMode(mode);
+ auto* const t = thread->asIAfPlaybackThread().get();
+ Mutex::Autolock lock(t->mutex());
+ status = t->getOutput_l()->stream->setDualMonoMode(mode);
if (status == NO_ERROR) {
mDualMonoMode = mode;
}
@@ -1910,11 +1909,11 @@
{
status_t status = INVALID_OPERATION;
if (isOffloadedOrDirect()) {
- sp<AudioFlinger::ThreadBase> thread = mThread.promote();
+ sp<IAfThreadBase> thread = mThread.promote();
if (thread != nullptr) {
- auto* const t = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
- Mutex::Autolock lock(t->mLock);
- status = t->mOutput->stream->getAudioDescriptionMixLevel(leveldB);
+ auto* const t = thread->asIAfPlaybackThread().get();
+ Mutex::Autolock lock(t->mutex());
+ status = t->getOutput_l()->stream->getAudioDescriptionMixLevel(leveldB);
ALOGD_IF((status == NO_ERROR) && (mAudioDescriptionMixLevel != *leveldB),
"%s: level %.3f inconsistent", __func__, mAudioDescriptionMixLevel);
}
@@ -1926,11 +1925,11 @@
{
status_t status = INVALID_OPERATION;
if (isOffloadedOrDirect()) {
- sp<AudioFlinger::ThreadBase> thread = mThread.promote();
+ const sp<IAfThreadBase> thread = mThread.promote();
if (thread != nullptr) {
- auto* const t = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
- Mutex::Autolock lock(t->mLock);
- status = t->mOutput->stream->setAudioDescriptionMixLevel(leveldB);
+ auto* const t = thread->asIAfPlaybackThread().get();
+ Mutex::Autolock lock(t->mutex());
+ status = t->getOutput_l()->stream->setAudioDescriptionMixLevel(leveldB);
if (status == NO_ERROR) {
mAudioDescriptionMixLevel = leveldB;
}
@@ -1944,11 +1943,11 @@
{
status_t status = INVALID_OPERATION;
if (isOffloadedOrDirect()) {
- sp<AudioFlinger::ThreadBase> thread = mThread.promote();
+ const sp<IAfThreadBase> thread = mThread.promote();
if (thread != nullptr) {
- auto* const t = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
- Mutex::Autolock lock(t->mLock);
- status = t->mOutput->stream->getPlaybackRateParameters(playbackRate);
+ auto* const t = thread->asIAfPlaybackThread().get();
+ Mutex::Autolock lock(t->mutex());
+ status = t->getOutput_l()->stream->getPlaybackRateParameters(playbackRate);
ALOGD_IF((status == NO_ERROR) &&
!isAudioPlaybackRateEqual(mPlaybackRateParameters, *playbackRate),
"%s: playbackRate inconsistent", __func__);
@@ -1962,11 +1961,11 @@
{
status_t status = INVALID_OPERATION;
if (isOffloadedOrDirect()) {
- sp<AudioFlinger::ThreadBase> thread = mThread.promote();
+ const sp<IAfThreadBase> thread = mThread.promote();
if (thread != nullptr) {
- auto* const t = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
- Mutex::Autolock lock(t->mLock);
- status = t->mOutput->stream->setPlaybackRateParameters(playbackRate);
+ auto* const t = thread->asIAfPlaybackThread().get();
+ Mutex::Autolock lock(t->mutex());
+ status = t->getOutput_l()->stream->setPlaybackRateParameters(playbackRate);
if (status == NO_ERROR) {
mPlaybackRateParameters = playbackRate;
}
@@ -2085,13 +2084,13 @@
}
bool Track::AudioVibrationController::setMute(bool muted) {
- sp<AudioFlinger::ThreadBase> thread = mTrack->mThread.promote();
+ const sp<IAfThreadBase> thread = mTrack->mThread.promote();
if (thread != 0) {
// Lock for updating mHapticPlaybackEnabled.
- Mutex::Autolock _l(thread->mLock);
- auto* const playbackThread = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
+ Mutex::Autolock _l(thread->mutex());
+ auto* const playbackThread = thread->asIAfPlaybackThread().get();
if ((mTrack->channelMask() & AUDIO_CHANNEL_HAPTIC_ALL) != AUDIO_CHANNEL_NONE
- && playbackThread->mHapticChannelCount > 0) {
+ && playbackThread->hapticChannelCount() > 0) {
ALOGD("%s, haptic playback was %s for track %d",
__func__, muted ? "muted" : "unmuted", mTrack->id());
mTrack->setHapticPlaybackEnabled(!muted);
@@ -2118,17 +2117,17 @@
#define LOG_TAG "AF::OutputTrack"
/* static */
-sp<IAfOutputTrack> IAfOutputTrack::create( // TODO(b/288339104) void*
- void* /* AudioFlinger::PlaybackThread */ playbackThread,
- void* /* AudioFlinger::DuplicatingThread */ sourceThread,
+sp<IAfOutputTrack> IAfOutputTrack::create(
+ IAfPlaybackThread* playbackThread,
+ IAfDuplicatingThread* sourceThread,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
size_t frameCount,
const AttributionSourceState& attributionSource) {
return sp<OutputTrack>::make(
- reinterpret_cast<AudioFlinger::PlaybackThread*>(playbackThread),
- reinterpret_cast<AudioFlinger::DuplicatingThread*>(sourceThread),
+ playbackThread,
+ sourceThread,
sampleRate,
format,
channelMask,
@@ -2137,8 +2136,8 @@
}
OutputTrack::OutputTrack(
- AudioFlinger::PlaybackThread *playbackThread,
- AudioFlinger::DuplicatingThread *sourceThread,
+ IAfPlaybackThread* playbackThread,
+ IAfDuplicatingThread* sourceThread,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
@@ -2155,7 +2154,7 @@
if (mCblk != NULL) {
mOutBuffer.frameCount = 0;
- playbackThread->mTracks.add(this);
+ playbackThread->addOutputTrack_l(this);
ALOGV("%s(): mCblk %p, mBuffer %p, "
"frameCount %zu, mChannelMask 0x%08x",
__func__, mCblk, mBuffer,
@@ -2203,7 +2202,7 @@
ssize_t OutputTrack::write(void* data, uint32_t frames)
{
if (!mActive && frames != 0) {
- sp<AudioFlinger::ThreadBase> thread = mThread.promote();
+ const sp<IAfThreadBase> thread = mThread.promote();
if (thread != nullptr && thread->inStandby()) {
// preload one silent buffer to trigger mixer on start()
ClientProxy::Buffer buf { .mFrameCount = mClientProxy->getStartThresholdInFrames() };
@@ -2222,7 +2221,7 @@
// If another OutputTrack has already started it can underrun but this is OK
// as only silence has been played so far and the retry count is very high on
// OutputTrack.
- auto* const pt = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
+ auto* const pt = thread->asIAfPlaybackThread().get();
if (!pt->waitForHalStart()) {
ALOGW("%s(%d): timeout waiting for thread to exit standby", __func__, mId);
stop();
@@ -2311,7 +2310,7 @@
// If we could not write all frames, allocate a buffer and queue it for next time.
if (inBuffer.frameCount) {
- sp<AudioFlinger::ThreadBase> thread = mThread.promote();
+ const sp<IAfThreadBase> thread = mThread.promote();
if (thread != nullptr && !thread->inStandby()) {
queueBuffer(inBuffer);
}
@@ -2404,7 +2403,7 @@
/* static */
sp<IAfPatchTrack> IAfPatchTrack::create(
- void* /* PlaybackThread */ playbackThread, // TODO(b/288339104)
+ IAfPlaybackThread* playbackThread,
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_channel_mask_t channelMask,
@@ -2420,7 +2419,7 @@
* even if it might glitch. */)
{
return sp<PatchTrack>::make(
- reinterpret_cast<AudioFlinger::PlaybackThread*>(playbackThread),
+ playbackThread,
streamType,
sampleRate,
channelMask,
@@ -2433,7 +2432,7 @@
frameCountToBeReady);
}
-PatchTrack::PatchTrack(AudioFlinger::PlaybackThread *playbackThread,
+PatchTrack::PatchTrack(IAfPlaybackThread* playbackThread,
audio_stream_type_t streamType,
uint32_t sampleRate,
audio_channel_mask_t channelMask,
@@ -2452,7 +2451,7 @@
TYPE_PATCH, AUDIO_PORT_HANDLE_NONE, frameCountToBeReady),
PatchTrackBase(mCblk ? new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, true, true)
: nullptr,
- *playbackThread, timeout)
+ playbackThread, timeout)
{
ALOGV("%s(%d): sampleRate %d mPeerTimeout %d.%03d sec",
__func__, mId, sampleRate,
@@ -2551,9 +2550,9 @@
if (mFillingStatus == FS_ACTIVE
&& audio_is_linear_pcm(mFormat)
&& !isOffloadedOrDirect()) {
- if (sp<AudioFlinger::ThreadBase> thread = mThread.promote();
+ if (const sp<IAfThreadBase> thread = mThread.promote();
thread != 0) {
- auto* const playbackThread = static_cast<AudioFlinger::PlaybackThread*>(thread.get());
+ auto* const playbackThread = thread->asIAfPlaybackThread().get();
const size_t frameCount = playbackThread->frameCount() * sampleRate()
/ playbackThread->sampleRate();
if (framesReady() < frameCount) {
@@ -2669,7 +2668,7 @@
/* static */ // TODO(b/288339104)
-sp<IAfRecordTrack> IAfRecordTrack::create(void* /*AudioFlinger::RecordThread */ thread,
+sp<IAfRecordTrack> IAfRecordTrack::create(IAfRecordThread* thread,
const sp<Client>& client,
const audio_attributes_t& attr,
uint32_t sampleRate,
@@ -2687,7 +2686,7 @@
int32_t startFrames)
{
return sp<RecordTrack>::make(
- reinterpret_cast<AudioFlinger::RecordThread*>(thread),
+ thread,
client,
attr,
sampleRate,
@@ -2707,7 +2706,7 @@
// RecordTrack constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
RecordTrack::RecordTrack(
- AudioFlinger::RecordThread* thread,
+ IAfRecordThread* thread,
const sp<Client>& client,
const audio_attributes_t& attr,
uint32_t sampleRate,
@@ -2746,7 +2745,7 @@
if (!isDirect()) {
mRecordBufferConverter = new RecordBufferConverter(
- thread->mChannelMask, thread->mFormat, thread->mSampleRate,
+ thread->channelMask(), thread->format(), thread->sampleRate(),
channelMask, format, sampleRate);
// Check if the RecordBufferConverter construction was successful.
// If not, don't continue with construction.
@@ -2766,8 +2765,8 @@
mResamplerBufferProvider = new ResamplerBufferProvider(this);
if (flags & AUDIO_INPUT_FLAG_FAST) {
- ALOG_ASSERT(thread->mFastTrackAvail);
- thread->mFastTrackAvail = false;
+ ALOG_ASSERT(thread->fastTrackAvailable());
+ thread->setFastTrackAvailable(false);
} else {
// TODO: only Normal Record has timestamps (Fast Record does not).
mServerLatencySupported = checkServerLatencySupported(mFormat, flags);
@@ -2816,9 +2815,9 @@
status_t RecordTrack::start(AudioSystem::sync_event_t event,
audio_session_t triggerSession)
{
- sp<AudioFlinger::ThreadBase> thread = mThread.promote();
+ const sp<IAfThreadBase> thread = mThread.promote();
if (thread != 0) {
- auto* const recordThread = static_cast<AudioFlinger::RecordThread*>(thread.get());
+ auto* const recordThread = thread->asIAfRecordThread().get();
return recordThread->start(this, event, triggerSession);
} else {
ALOGW("%s track %d: thread was destroyed", __func__, portId());
@@ -2828,9 +2827,9 @@
void RecordTrack::stop()
{
- sp<AudioFlinger::ThreadBase> thread = mThread.promote();
+ const sp<IAfThreadBase> thread = mThread.promote();
if (thread != 0) {
- auto* const recordThread = static_cast<AudioFlinger::RecordThread*>(thread.get());
+ auto* const recordThread = thread->asIAfRecordThread().get();
if (recordThread->stop(this) && isExternalTrack()) {
AudioSystem::stopInput(mPortId);
}
@@ -2843,10 +2842,10 @@
sp<RecordTrack> keep(this);
{
track_state priorState = mState;
- sp<AudioFlinger::ThreadBase> thread = mThread.promote();
+ const sp<IAfThreadBase> thread = mThread.promote();
if (thread != 0) {
- Mutex::Autolock _l(thread->mLock);
- auto* const recordThread = static_cast<AudioFlinger::RecordThread*>(thread.get());
+ Mutex::Autolock _l(thread->mutex());
+ auto* const recordThread = thread->asIAfRecordThread().get();
priorState = mState;
if (!mSharedAudioPackageName.empty()) {
recordThread->resetAudioHistory_l();
@@ -2940,11 +2939,11 @@
const sp<audioflinger::SyncEvent>& event)
{
size_t framesToDrop = 0;
- sp<AudioFlinger::ThreadBase> threadBase = mThread.promote();
+ const sp<IAfThreadBase> threadBase = mThread.promote();
if (threadBase != 0) {
// TODO: use actual buffer filling status instead of 2 buffers when info is available
// from audio HAL
- framesToDrop = threadBase->mFrameCount * 2;
+ framesToDrop = threadBase->frameCount() * 2;
}
mSynchronizedRecordState.onPlaybackFinished(event, framesToDrop);
@@ -2998,9 +2997,9 @@
status_t RecordTrack::getActiveMicrophones(
std::vector<media::MicrophoneInfoFw>* activeMicrophones) const
{
- sp<AudioFlinger::ThreadBase> thread = mThread.promote();
+ const sp<IAfThreadBase> thread = mThread.promote();
if (thread != 0) {
- auto* const recordThread = static_cast<AudioFlinger::RecordThread*>(thread.get());
+ auto* const recordThread = thread->asIAfRecordThread().get();
return recordThread->getActiveMicrophones(activeMicrophones);
} else {
return BAD_VALUE;
@@ -3009,9 +3008,9 @@
status_t RecordTrack::setPreferredMicrophoneDirection(
audio_microphone_direction_t direction) {
- sp<AudioFlinger::ThreadBase> thread = mThread.promote();
+ const sp<IAfThreadBase> thread = mThread.promote();
if (thread != 0) {
- auto* const recordThread = static_cast<AudioFlinger::RecordThread*>(thread.get());
+ auto* const recordThread = thread->asIAfRecordThread().get();
return recordThread->setPreferredMicrophoneDirection(direction);
} else {
return BAD_VALUE;
@@ -3019,9 +3018,9 @@
}
status_t RecordTrack::setPreferredMicrophoneFieldDimension(float zoom) {
- sp<AudioFlinger::ThreadBase> thread = mThread.promote();
+ const sp<IAfThreadBase> thread = mThread.promote();
if (thread != 0) {
- auto* const recordThread = static_cast<AudioFlinger::RecordThread*>(thread.get());
+ auto* const recordThread = thread->asIAfRecordThread().get();
return recordThread->setPreferredMicrophoneFieldDimension(zoom);
} else {
return BAD_VALUE;
@@ -3045,9 +3044,9 @@
return PERMISSION_DENIED;
}
- sp<AudioFlinger::ThreadBase> thread = mThread.promote();
+ const sp<IAfThreadBase> thread = mThread.promote();
if (thread != 0) {
- auto* const recordThread = static_cast<AudioFlinger::RecordThread*>(thread.get());
+ auto* const recordThread = thread->asIAfRecordThread().get();
status_t status = recordThread->shareAudioHistory(
sharedAudioPackageName, mSessionId, sharedAudioStartMs);
if (status == NO_ERROR) {
@@ -3085,7 +3084,7 @@
/* static */
sp<IAfPatchRecord> IAfPatchRecord::create(
- void* /* RecordThread */ recordThread, // TODO(b/288339104)
+ IAfRecordThread* recordThread,
uint32_t sampleRate,
audio_channel_mask_t channelMask,
audio_format_t format,
@@ -3097,7 +3096,7 @@
audio_source_t source)
{
return sp<PatchRecord>::make(
- reinterpret_cast<AudioFlinger::RecordThread*>(recordThread),
+ recordThread,
sampleRate,
channelMask,
format,
@@ -3109,7 +3108,7 @@
source);
}
-PatchRecord::PatchRecord(AudioFlinger::RecordThread *recordThread,
+PatchRecord::PatchRecord(IAfRecordThread* recordThread,
uint32_t sampleRate,
audio_channel_mask_t channelMask,
audio_format_t format,
@@ -3126,7 +3125,7 @@
audioServerAttributionSource(getpid()), flags, TYPE_PATCH),
PatchTrackBase(mCblk ? new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, false, true)
: nullptr,
- *recordThread, timeout)
+ recordThread, timeout)
{
ALOGV("%s(%d): sampleRate %d mPeerTimeout %d.%03d sec",
__func__, mId, sampleRate,
@@ -3228,7 +3227,7 @@
/* static */
sp<IAfPatchRecord> IAfPatchRecord::createPassThru(
- void* /* RecordThread */ recordThread, // TODO(b/288339104)
+ IAfRecordThread* recordThread,
uint32_t sampleRate,
audio_channel_mask_t channelMask,
audio_format_t format,
@@ -3237,7 +3236,7 @@
audio_source_t source)
{
return sp<PassthruPatchRecord>::make(
- reinterpret_cast<AudioFlinger::RecordThread*>(recordThread),
+ recordThread,
sampleRate,
channelMask,
format,
@@ -3247,7 +3246,7 @@
}
PassthruPatchRecord::PassthruPatchRecord(
- AudioFlinger::RecordThread* recordThread,
+ IAfRecordThread* recordThread,
uint32_t sampleRate,
audio_channel_mask_t channelMask,
audio_format_t format,
@@ -3264,13 +3263,13 @@
}
sp<StreamInHalInterface> PassthruPatchRecord::obtainStream(
- sp<AudioFlinger::ThreadBase>* thread)
+ sp<IAfThreadBase>* thread)
{
*thread = mThread.promote();
if (!*thread) return nullptr;
- auto* const recordThread = static_cast<AudioFlinger::RecordThread*>((*thread).get());
- Mutex::Autolock _l(recordThread->mLock);
- return recordThread->mInput ? recordThread->mInput->stream : nullptr;
+ auto* const recordThread = (*thread)->asIAfRecordThread().get();
+ Mutex::Autolock _l(recordThread->mutex());
+ return recordThread->getInput() ? recordThread->getInput()->stream : nullptr;
}
// PatchProxyBufferProvider methods are called on DirectOutputThread
@@ -3292,7 +3291,7 @@
const size_t framesToRead = std::min(buffer->mFrameCount, mFrameCount);
buffer->mFrameCount = 0;
buffer->mRaw = nullptr;
- sp<AudioFlinger::ThreadBase> thread;
+ sp<IAfThreadBase> thread;
sp<StreamInHalInterface> stream = obtainStream(&thread);
if (!stream) return NO_INIT; // If there is no stream, RecordThread is not reading.
@@ -3379,7 +3378,7 @@
status_t PassthruPatchRecord::getCapturePosition(
int64_t* frames, int64_t* time)
{
- sp<AudioFlinger::ThreadBase> thread;
+ sp<IAfThreadBase> thread;
sp<StreamInHalInterface> stream = obtainStream(&thread);
return stream ? stream->getCapturePosition(frames, time) : NO_INIT;
}
@@ -3416,7 +3415,7 @@
#define LOG_TAG "AF::MmapTrack"
/* static */
-sp<IAfMmapTrack> IAfMmapTrack::create(void* /* AudioFlinger::ThreadBase */ thread,
+sp<IAfMmapTrack> IAfMmapTrack::create(IAfThreadBase* thread,
const audio_attributes_t& attr,
uint32_t sampleRate,
audio_format_t format,
@@ -3428,7 +3427,7 @@
audio_port_handle_t portId)
{
return sp<MmapTrack>::make(
- reinterpret_cast<AudioFlinger::ThreadBase*>(thread),
+ thread,
attr,
sampleRate,
format,
@@ -3440,7 +3439,7 @@
portId);
}
-MmapTrack::MmapTrack(AudioFlinger::ThreadBase* thread,
+MmapTrack::MmapTrack(IAfThreadBase* thread,
const audio_attributes_t& attr,
uint32_t sampleRate,
audio_format_t format,