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-rw-r--r--libs/audioflinger/AudioFlinger.cpp133
1 files changed, 60 insertions, 73 deletions
diff --git a/libs/audioflinger/AudioFlinger.cpp b/libs/audioflinger/AudioFlinger.cpp
index 43df7dd..b8e5bd0 100644
--- a/libs/audioflinger/AudioFlinger.cpp
+++ b/libs/audioflinger/AudioFlinger.cpp
@@ -202,8 +202,8 @@ void AudioFlinger::setA2dpEnabled_l(bool enable)
SortedVector < sp<MixerThread::Track> > tracks;
SortedVector < wp<MixerThread::Track> > activeTracks;
- LOGV_IF(enable, "set output to A2DP\n");
- LOGV_IF(!enable, "set output to hardware audio\n");
+ LOGD_IF(enable, "set output to A2DP\n");
+ LOGD_IF(!enable, "set output to hardware audio\n");
// Transfer tracks playing on MUSIC stream from one mixer to the other
if (enable) {
@@ -212,6 +212,7 @@ void AudioFlinger::setA2dpEnabled_l(bool enable)
} else {
mA2dpMixerThread->getTracks_l(tracks, activeTracks);
mHardwareMixerThread->putTracks_l(tracks, activeTracks);
+ mA2dpMixerThread->mOutput->standby();
}
mA2dpEnabled = enable;
mNotifyA2dpChange = true;
@@ -499,7 +500,8 @@ status_t AudioFlinger::setRouting(int mode, uint32_t routes, uint32_t mask)
}
#ifdef WITH_A2DP
- LOGD("setRouting %d %d %d, tid %d, calling tid %d\n", mode, routes, mask, gettid(), IPCThreadState::self()->getCallingPid());
+ LOGV("setRouting %d %d %d, tid %d, calling tid %d\n", mode, routes, mask, gettid(),
+ IPCThreadState::self()->getCallingPid());
if (mode == AudioSystem::MODE_NORMAL &&
(mask & AudioSystem::ROUTE_BLUETOOTH_A2DP)) {
AutoMutex lock(&mLock);
@@ -655,16 +657,12 @@ status_t AudioFlinger::setStreamVolume(int stream, float value)
if (stream == AudioSystem::VOICE_CALL ||
stream == AudioSystem::BLUETOOTH_SCO) {
- float hwValue = value;
+ float hwValue;
if (stream == AudioSystem::VOICE_CALL) {
hwValue = (float)AudioSystem::logToLinear(value)/100.0f;
- // FIXME: This is a temporary fix to re-base the internally
- // generated in-call audio so that it is never muted, which is
- // already the case for the hardware routed in-call audio.
- // When audio stream handling is reworked, this should be
- // addressed more cleanly. Fixes #1324; see discussion at
- // http://review.source.android.com/8224
- value = value * 0.99 + 0.01;
+ // offset value to reflect actual hardware volume that never reaches 0
+ // 1% corresponds roughly to first step in VOICE_CALL stream volume setting (see AudioService.java)
+ value = 0.01 + 0.99 * value;
} else { // (type == AudioSystem::BLUETOOTH_SCO)
hwValue = 1.0f;
}
@@ -681,6 +679,11 @@ status_t AudioFlinger::setStreamVolume(int stream, float value)
mA2dpMixerThread->setStreamVolume(stream, value);
#endif
+ mHardwareMixerThread->setStreamVolume(stream, value);
+#ifdef WITH_A2DP
+ mA2dpMixerThread->setStreamVolume(stream, value);
+#endif
+
return ret;
}
@@ -718,15 +721,14 @@ float AudioFlinger::streamVolume(int stream) const
if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
return 0.0f;
}
- float value = mHardwareMixerThread->streamVolume(stream);
+ float volume = mHardwareMixerThread->streamVolume(stream);
+ // remove correction applied by setStreamVolume()
if (stream == AudioSystem::VOICE_CALL) {
- // FIXME: Re-base internally generated in-call audio,
- // reverse of above in setStreamVolume.
- value = (value - 0.01) / 0.99;
+ volume = (volume - 0.01) / 0.99 ;
}
- return value;
+ return volume;
}
bool AudioFlinger::streamMute(int stream) const
@@ -744,12 +746,13 @@ bool AudioFlinger::streamMute(int stream) const
bool AudioFlinger::isMusicActive() const
{
+ Mutex::Autolock _l(mLock);
#ifdef WITH_A2DP
if (isA2dpEnabled()) {
- return mA2dpMixerThread->isMusicActive();
+ return mA2dpMixerThread->isMusicActive_l();
}
#endif
- return mHardwareMixerThread->isMusicActive();
+ return mHardwareMixerThread->isMusicActive_l();
}
status_t AudioFlinger::setParameter(const char* key, const char* value)
@@ -824,24 +827,22 @@ void AudioFlinger::handleForcedSpeakerRoute(int command)
{
AutoMutex lock(mHardwareLock);
if (mForcedSpeakerCount++ == 0) {
- mRouteRestoreTime = 0;
- mMusicMuteSaved = mHardwareMixerThread->streamMute(AudioSystem::MUSIC);
- if (mForcedRoute == 0 && !(mSavedRoute & AudioSystem::ROUTE_SPEAKER)) {
- LOGV("Route forced to Speaker ON %08x", mSavedRoute | AudioSystem::ROUTE_SPEAKER);
- mHardwareMixerThread->setStreamMute(AudioSystem::MUSIC, true);
- mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
- mAudioHardware->setMasterVolume(0);
- usleep(mHardwareMixerThread->latency()*1000);
- mHardwareStatus = AUDIO_HW_SET_ROUTING;
- mAudioHardware->setRouting(AudioSystem::MODE_NORMAL, mSavedRoute | AudioSystem::ROUTE_SPEAKER);
- mHardwareStatus = AUDIO_HW_IDLE;
- // delay track start so that audio hardware has time to siwtch routes
- usleep(kStartSleepTime);
- mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
- mAudioHardware->setMasterVolume(mHardwareMixerThread->masterVolume());
- mHardwareStatus = AUDIO_HW_IDLE;
+ if (mForcedRoute == 0) {
+ mMusicMuteSaved = mHardwareMixerThread->streamMute(AudioSystem::MUSIC);
+ LOGV("++mForcedSpeakerCount == 0, mMusicMuteSaved = %d, mRouteRestoreTime = %d", mMusicMuteSaved, mRouteRestoreTime);
+ if (!(mSavedRoute & AudioSystem::ROUTE_SPEAKER)) {
+ LOGV("Route forced to Speaker ON %08x", mSavedRoute | AudioSystem::ROUTE_SPEAKER);
+ mHardwareMixerThread->setStreamMute(AudioSystem::MUSIC, true);
+ usleep(mHardwareMixerThread->latency()*1000);
+ mHardwareStatus = AUDIO_HW_SET_ROUTING;
+ mAudioHardware->setRouting(AudioSystem::MODE_NORMAL, mSavedRoute | AudioSystem::ROUTE_SPEAKER);
+ mHardwareStatus = AUDIO_HW_IDLE;
+ // delay track start so that audio hardware has time to siwtch routes
+ usleep(kStartSleepTime);
+ }
}
mForcedRoute = AudioSystem::ROUTE_SPEAKER;
+ mRouteRestoreTime = 0;
}
LOGV("mForcedSpeakerCount incremented to %d", mForcedSpeakerCount);
}
@@ -902,7 +903,7 @@ void AudioFlinger::handleRouteDisablesA2dp_l(int routes)
}
LOGV("mA2dpDisableCount decremented to %d", mA2dpDisableCount);
} else {
- LOGE("mA2dpDisableCount is already zero");
+ LOGV("mA2dpDisableCount is already zero");
}
}
}
@@ -1289,7 +1290,7 @@ sp<AudioFlinger::MixerThread::Track> AudioFlinger::MixerThread::createTrack_l(
status_t lStatus;
// Resampler implementation limits input sampling rate to 2 x output sampling rate.
- if (sampleRate > MAX_SAMPLE_RATE || sampleRate > mSampleRate*2) {
+ if (sampleRate > mSampleRate*2) {
LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
lStatus = BAD_VALUE;
goto Exit;
@@ -1452,7 +1453,8 @@ bool AudioFlinger::MixerThread::streamMute(int stream) const
return mStreamTypes[stream].mute;
}
-bool AudioFlinger::MixerThread::isMusicActive() const
+// isMusicActive_l() must be called with AudioFlinger::mLock held
+bool AudioFlinger::MixerThread::isMusicActive_l() const
{
size_t count = mActiveTracks.size();
for (size_t i = 0 ; i < count ; ++i) {
@@ -1497,18 +1499,6 @@ status_t AudioFlinger::MixerThread::addTrack_l(const sp<Track>& track)
return status;
}
-// removeTrack_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::MixerThread::removeTrack_l(wp<Track> track, int name)
-{
- sp<Track> t = track.promote();
- if (t!=NULL && (t->mState <= TrackBase::STOPPED)) {
- t->reset();
- deleteTrackName_l(name);
- removeActiveTrack_l(track);
- mAudioFlinger->mWaitWorkCV.broadcast();
- }
-}
-
// destroyTrack_l() must be called with AudioFlinger::mLock held
void AudioFlinger::MixerThread::destroyTrack_l(const sp<Track>& track)
{
@@ -1577,7 +1567,6 @@ size_t AudioFlinger::MixerThread::getOutputFrameCount()
AudioFlinger::MixerThread::TrackBase::TrackBase(
const sp<MixerThread>& mixerThread,
const sp<Client>& client,
- int streamType,
uint32_t sampleRate,
int format,
int channelCount,
@@ -1587,7 +1576,6 @@ AudioFlinger::MixerThread::TrackBase::TrackBase(
: RefBase(),
mMixerThread(mixerThread),
mClient(client),
- mStreamType(streamType),
mFrameCount(0),
mState(IDLE),
mClientTid(-1),
@@ -1618,8 +1606,8 @@ AudioFlinger::MixerThread::TrackBase::TrackBase(
new(mCblk) audio_track_cblk_t();
// clear all buffers
mCblk->frameCount = frameCount;
- mCblk->sampleRate = (uint16_t)sampleRate;
- mCblk->channels = (uint16_t)channelCount;
+ mCblk->sampleRate = sampleRate;
+ mCblk->channels = (uint8_t)channelCount;
if (sharedBuffer == 0) {
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
@@ -1642,8 +1630,8 @@ AudioFlinger::MixerThread::TrackBase::TrackBase(
new(mCblk) audio_track_cblk_t();
// clear all buffers
mCblk->frameCount = frameCount;
- mCblk->sampleRate = (uint16_t)sampleRate;
- mCblk->channels = (uint16_t)channelCount;
+ mCblk->sampleRate = sampleRate;
+ mCblk->channels = (uint8_t)channelCount;
mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
// Force underrun condition to avoid false underrun callback until first data is
@@ -1704,7 +1692,7 @@ int AudioFlinger::MixerThread::TrackBase::sampleRate() const {
}
int AudioFlinger::MixerThread::TrackBase::channelCount() const {
- return mCblk->channels;
+ return (int)mCblk->channels;
}
void* AudioFlinger::MixerThread::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
@@ -1714,7 +1702,7 @@ void* AudioFlinger::MixerThread::TrackBase::getBuffer(uint32_t offset, uint32_t
// Check validity of returned pointer in case the track control block would have been corrupted.
if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
- cblk->channels == 2 && ((unsigned long)bufferStart & 3) ) {
+ (cblk->channels == 2 && ((unsigned long)bufferStart & 3))) {
LOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \
server %d, serverBase %d, user %d, userBase %d, channels %d",
bufferStart, bufferEnd, mBuffer, mBufferEnd,
@@ -1737,12 +1725,13 @@ AudioFlinger::MixerThread::Track::Track(
int channelCount,
int frameCount,
const sp<IMemory>& sharedBuffer)
- : TrackBase(mixerThread, client, streamType, sampleRate, format, channelCount, frameCount, 0, sharedBuffer)
+ : TrackBase(mixerThread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer)
{
mVolume[0] = 1.0f;
mVolume[1] = 1.0f;
mMute = false;
mSharedBuffer = sharedBuffer;
+ mStreamType = streamType;
}
AudioFlinger::MixerThread::Track::~Track()
@@ -1750,7 +1739,6 @@ AudioFlinger::MixerThread::Track::~Track()
wp<Track> weak(this); // never create a strong ref from the dtor
Mutex::Autolock _l(mMixerThread->mAudioFlinger->mLock);
mState = TERMINATED;
- mMixerThread->removeTrack_l(weak, mName);
}
void AudioFlinger::MixerThread::Track::destroy()
@@ -1927,15 +1915,15 @@ void AudioFlinger::MixerThread::Track::setVolume(float left, float right)
AudioFlinger::MixerThread::RecordTrack::RecordTrack(
const sp<MixerThread>& mixerThread,
const sp<Client>& client,
- int streamType,
+ int inputSource,
uint32_t sampleRate,
int format,
int channelCount,
int frameCount,
uint32_t flags)
- : TrackBase(mixerThread, client, streamType, sampleRate, format,
+ : TrackBase(mixerThread, client, sampleRate, format,
channelCount, frameCount, flags, 0),
- mOverflow(false)
+ mOverflow(false), mInputSource(inputSource)
{
}
@@ -2052,7 +2040,10 @@ void AudioFlinger::MixerThread::OutputTrack::write(int16_t* data, uint32_t frame
inBuffer.i16 = data;
if (mCblk->user == 0) {
- if (mOutputMixerThread->isMusicActive()) {
+ mOutputMixerThread->mAudioFlinger->mLock.lock();
+ bool isMusicActive = mOutputMixerThread->isMusicActive_l();
+ mOutputMixerThread->mAudioFlinger->mLock.unlock();
+ if (isMusicActive) {
mCblk->forceReady = 1;
LOGV("OutputTrack::start() force ready");
} else if (mCblk->frameCount > frames){
@@ -2260,7 +2251,7 @@ status_t AudioFlinger::TrackHandle::onTransact(
sp<IAudioRecord> AudioFlinger::openRecord(
pid_t pid,
- int streamType,
+ int inputSource,
uint32_t sampleRate,
int format,
int channelCount,
@@ -2283,18 +2274,12 @@ sp<IAudioRecord> AudioFlinger::openRecord(
goto Exit;
}
- if (uint32_t(streamType) >= AudioRecord::NUM_STREAM_TYPES) {
+ if (uint32_t(inputSource) >= AudioRecord::NUM_INPUT_SOURCES) {
LOGE("invalid stream type");
lStatus = BAD_VALUE;
goto Exit;
}
- if (sampleRate > MAX_SAMPLE_RATE) {
- LOGE("Sample rate out of range");
- lStatus = BAD_VALUE;
- goto Exit;
- }
-
if (mAudioRecordThread == 0) {
LOGE("Audio record thread not started");
lStatus = NO_INIT;
@@ -2326,7 +2311,7 @@ sp<IAudioRecord> AudioFlinger::openRecord(
frameCount = ((frameCount - 1)/inFrameCount + 1) * inFrameCount;
// create new record track. The record track uses one track in mHardwareMixerThread by convention.
- recordTrack = new MixerThread::RecordTrack(mHardwareMixerThread, client, streamType, sampleRate,
+ recordTrack = new MixerThread::RecordTrack(mHardwareMixerThread, client, inputSource, sampleRate,
format, channelCount, frameCount, flags);
}
if (recordTrack->getCblk() == NULL) {
@@ -2432,7 +2417,9 @@ bool AudioFlinger::AudioRecordThread::threadLoop()
LOGV("AudioRecordThread: loop starting");
if (mRecordTrack != 0) {
- input = mAudioHardware->openInputStream(mRecordTrack->format(),
+ input = mAudioHardware->openInputStream(
+ mRecordTrack->inputSource(),
+ mRecordTrack->format(),
mRecordTrack->channelCount(),
mRecordTrack->sampleRate(),
&mStartStatus,
generated by cgit v1.1 at 2025-06-03 18:09:44 +0000