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-rw-r--r--services/audioflinger/AudioFlinger.cpp99
-rw-r--r--services/audioflinger/AudioFlinger.h6
-rw-r--r--services/audioflinger/AudioMixer.cpp125
-rw-r--r--services/audioflinger/AudioMixer.h4
-rw-r--r--services/audioflinger/AudioPolicyService.cpp1
-rw-r--r--services/audioflinger/AudioPolicyService.h3
6 files changed, 123 insertions, 115 deletions
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index dcc0bd0..a13fddb 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -63,8 +63,8 @@
namespace android {
-static const char* kDeadlockedString = "AudioFlinger may be deadlocked\n";
-static const char* kHardwareLockedString = "Hardware lock is taken\n";
+static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n";
+static const char kHardwareLockedString[] = "Hardware lock is taken\n";
//static const nsecs_t kStandbyTimeInNsecs = seconds(3);
static const float MAX_GAIN = 4096.0f;
@@ -80,14 +80,16 @@ static const int8_t kMaxTrackStartupRetries = 50;
static const int8_t kMaxTrackRetriesDirect = 2;
static const int kDumpLockRetries = 50;
-static const int kDumpLockSleep = 20000;
+static const int kDumpLockSleepUs = 20000;
-static const nsecs_t kWarningThrottle = seconds(5);
+// don't warn about blocked writes or record buffer overflows more often than this
+static const nsecs_t kWarningThrottleNs = seconds(5);
// RecordThread loop sleep time upon application overrun or audio HAL read error
static const int kRecordThreadSleepUs = 5000;
-static const nsecs_t kSetParametersTimeout = seconds(2);
+// maximum time to wait for setParameters to complete
+static const nsecs_t kSetParametersTimeoutNs = seconds(2);
// minimum sleep time for the mixer thread loop when tracks are active but in underrun
static const uint32_t kMinThreadSleepTimeUs = 5000;
@@ -147,7 +149,7 @@ out:
return rc;
}
-static const char *audio_interfaces[] = {
+static const char * const audio_interfaces[] = {
"primary",
"a2dp",
"usb",
@@ -158,7 +160,7 @@ static const char *audio_interfaces[] = {
AudioFlinger::AudioFlinger()
: BnAudioFlinger(),
- mPrimaryHardwareDev(0), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1),
+ mPrimaryHardwareDev(NULL), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1),
mBtNrecIsOff(false)
{
}
@@ -320,7 +322,7 @@ static bool tryLock(Mutex& mutex)
locked = true;
break;
}
- usleep(kDumpLockSleep);
+ usleep(kDumpLockSleepUs);
}
return locked;
}
@@ -395,7 +397,7 @@ sp<IAudioTrack> AudioFlinger::createTrack(
int lSessionId;
if (streamType >= AUDIO_STREAM_CNT) {
- LOGE("invalid stream type");
+ LOGE("createTrack() invalid stream type %d", streamType);
lStatus = BAD_VALUE;
goto Exit;
}
@@ -427,6 +429,7 @@ sp<IAudioTrack> AudioFlinger::createTrack(
// prevent same audio session on different output threads
uint32_t sessions = t->hasAudioSession(*sessionId);
if (sessions & PlaybackThread::TRACK_SESSION) {
+ LOGE("createTrack() session ID %d already in use", *sessionId);
lStatus = BAD_VALUE;
goto Exit;
}
@@ -657,6 +660,7 @@ status_t AudioFlinger::setStreamVolume(int stream, float value, int output)
}
if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) {
+ LOGE("setStreamVolume() invalid stream %d", stream);
return BAD_VALUE;
}
@@ -691,6 +695,7 @@ status_t AudioFlinger::setStreamMute(int stream, bool muted)
if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT ||
uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
+ LOGE("setStreamMute() invalid stream %d", stream);
return BAD_VALUE;
}
@@ -988,7 +993,6 @@ AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int i
AudioFlinger::ThreadBase::~ThreadBase()
{
mParamCond.broadcast();
- mNewParameters.clear();
// do not lock the mutex in destructor
releaseWakeLock_l();
if (mPowerManager != 0) {
@@ -999,7 +1003,7 @@ AudioFlinger::ThreadBase::~ThreadBase()
void AudioFlinger::ThreadBase::exit()
{
- // keep a strong ref on ourself so that we wont get
+ // keep a strong ref on ourself so that we won't get
// destroyed in the middle of requestExitAndWait()
sp <ThreadBase> strongMe = this;
@@ -1044,7 +1048,7 @@ status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
mWaitWorkCV.signal();
// wait condition with timeout in case the thread loop has exited
// before the request could be processed
- if (mParamCond.waitRelative(mLock, kSetParametersTimeout) == NO_ERROR) {
+ if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) {
status = mParamStatus;
mWaitWorkCV.signal();
} else {
@@ -1062,9 +1066,9 @@ void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param)
// sendConfigEvent_l() must be called with ThreadBase::mLock held
void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param)
{
- ConfigEvent *configEvent = new ConfigEvent();
- configEvent->mEvent = event;
- configEvent->mParam = param;
+ ConfigEvent configEvent;
+ configEvent.mEvent = event;
+ configEvent.mParam = param;
mConfigEvents.add(configEvent);
ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param);
mWaitWorkCV.signal();
@@ -1075,15 +1079,14 @@ void AudioFlinger::ThreadBase::processConfigEvents()
mLock.lock();
while(!mConfigEvents.isEmpty()) {
ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
- ConfigEvent *configEvent = mConfigEvents[0];
+ ConfigEvent configEvent = mConfigEvents[0];
mConfigEvents.removeAt(0);
// release mLock before locking AudioFlinger mLock: lock order is always
// AudioFlinger then ThreadBase to avoid cross deadlock
mLock.unlock();
mAudioFlinger->mLock.lock();
- audioConfigChanged_l(configEvent->mEvent, configEvent->mParam);
+ audioConfigChanged_l(configEvent.mEvent, configEvent.mParam);
mAudioFlinger->mLock.unlock();
- delete configEvent;
mLock.lock();
}
mLock.unlock();
@@ -1130,7 +1133,7 @@ status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args
snprintf(buffer, SIZE, " Index event param\n");
result.append(buffer);
for (size_t i = 0; i < mConfigEvents.size(); i++) {
- snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i]->mEvent, mConfigEvents[i]->mParam);
+ snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam);
result.append(buffer);
}
result.append("\n");
@@ -1367,7 +1370,7 @@ AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinge
int id,
uint32_t device)
: ThreadBase(audioFlinger, id, device),
- mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output),
+ mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), mOutput(output),
mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false)
{
snprintf(mName, kNameLength, "AudioOut_%d", id);
@@ -1526,8 +1529,10 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTra
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> t = mTracks[i];
if (t != 0) {
- if (sessionId == t->sessionId() &&
- strategy != AudioSystem::getStrategyForStream((audio_stream_type_t)t->type())) {
+ uint32_t actual = AudioSystem::getStrategyForStream((audio_stream_type_t)t->type());
+ if (sessionId == t->sessionId() && strategy != actual) {
+ LOGE("createTrack_l() mismatched strategy; expected %u but found %u",
+ strategy, actual);
lStatus = BAD_VALUE;
goto Exit;
}
@@ -1832,7 +1837,7 @@ uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs()
AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
: PlaybackThread(audioFlinger, output, id, device),
- mAudioMixer(0)
+ mAudioMixer(NULL)
{
mType = ThreadBase::MIXER;
mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
@@ -1967,7 +1972,7 @@ bool AudioFlinger::MixerThread::threadLoop()
// during mixing and effect process as the audio buffers could be deleted
// or modified if an effect is created or deleted
lockEffectChains_l(effectChains);
- }
+ }
if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
// mix buffers...
@@ -2012,11 +2017,11 @@ bool AudioFlinger::MixerThread::threadLoop()
}
// sleepTime == 0 means we must write to audio hardware
if (sleepTime == 0) {
- for (size_t i = 0; i < effectChains.size(); i ++) {
- effectChains[i]->process_l();
- }
- // enable changes in effect chain
- unlockEffectChains(effectChains);
+ for (size_t i = 0; i < effectChains.size(); i ++) {
+ effectChains[i]->process_l();
+ }
+ // enable changes in effect chain
+ unlockEffectChains(effectChains);
mLastWriteTime = systemTime();
mInWrite = true;
mBytesWritten += mixBufferSize;
@@ -2029,7 +2034,7 @@ bool AudioFlinger::MixerThread::threadLoop()
nsecs_t delta = now - mLastWriteTime;
if (!mStandby && delta > maxPeriod) {
mNumDelayedWrites++;
- if ((now - lastWarning) > kWarningThrottle) {
+ if ((now - lastWarning) > kWarningThrottleNs) {
LOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
ns2ms(delta), mNumDelayedWrites, this);
lastWarning = now;
@@ -2344,7 +2349,7 @@ bool AudioFlinger::MixerThread::checkForNewParameters_l()
}
if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
// do not accept frame count changes if tracks are open as the track buffer
- // size depends on frame count and correct behavior would not be garantied
+ // size depends on frame count and correct behavior would not be guaranteed
// if frame count is changed after track creation
if (!mTracks.isEmpty()) {
status = INVALID_OPERATION;
@@ -2415,7 +2420,7 @@ bool AudioFlinger::MixerThread::checkForNewParameters_l()
mParamCond.signal();
// wait for condition with time out in case the thread calling ThreadBase::setParameters()
// already timed out waiting for the status and will never signal the condition.
- mWaitWorkCV.waitRelative(mLock, kSetParametersTimeout);
+ mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
}
return reconfig;
}
@@ -2766,7 +2771,7 @@ bool AudioFlinger::DirectOutputThread::threadLoop()
while (frameCount) {
buffer.frameCount = frameCount;
activeTrack->getNextBuffer(&buffer);
- if (UNLIKELY(buffer.raw == 0)) {
+ if (UNLIKELY(buffer.raw == NULL)) {
memset(curBuf, 0, frameCount * mFrameSize);
break;
}
@@ -2891,7 +2896,7 @@ bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
mParamCond.signal();
// wait for condition with time out in case the thread calling ThreadBase::setParameters()
// already timed out waiting for the status and will never signal the condition.
- mWaitWorkCV.waitRelative(mLock, kSetParametersTimeout);
+ mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
}
return reconfig;
}
@@ -3264,7 +3269,7 @@ AudioFlinger::ThreadBase::TrackBase::~TrackBase()
void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
{
- buffer->raw = 0;
+ buffer->raw = NULL;
mFrameCount = buffer->frameCount;
step();
buffer->frameCount = 0;
@@ -3457,14 +3462,14 @@ status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider:
}
buffer->raw = getBuffer(s, framesReq);
- if (buffer->raw == 0) goto getNextBuffer_exit;
+ if (buffer->raw == NULL) goto getNextBuffer_exit;
buffer->frameCount = framesReq;
return NO_ERROR;
}
getNextBuffer_exit:
- buffer->raw = 0;
+ buffer->raw = NULL;
buffer->frameCount = 0;
ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
return NOT_ENOUGH_DATA;
@@ -3705,14 +3710,14 @@ status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvi
}
buffer->raw = getBuffer(s, framesReq);
- if (buffer->raw == 0) goto getNextBuffer_exit;
+ if (buffer->raw == NULL) goto getNextBuffer_exit;
buffer->frameCount = framesReq;
return NO_ERROR;
}
getNextBuffer_exit:
- buffer->raw = 0;
+ buffer->raw = NULL;
buffer->frameCount = 0;
return NOT_ENOUGH_DATA;
}
@@ -4217,7 +4222,7 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
int id,
uint32_t device) :
ThreadBase(audioFlinger, id, device),
- mInput(input), mTrack(NULL), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0)
+ mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL)
{
mType = ThreadBase::RECORD;
@@ -4232,7 +4237,7 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
AudioFlinger::RecordThread::~RecordThread()
{
delete[] mRsmpInBuffer;
- if (mResampler != 0) {
+ if (mResampler != NULL) {
delete mResampler;
delete[] mRsmpOutBuffer;
}
@@ -4326,7 +4331,7 @@ bool AudioFlinger::RecordThread::threadLoop()
buffer.frameCount = mFrameCount;
if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
size_t framesOut = buffer.frameCount;
- if (mResampler == 0) {
+ if (mResampler == NULL) {
// no resampling
while (framesOut) {
size_t framesIn = mFrameCount - mRsmpInIndex;
@@ -4411,7 +4416,7 @@ bool AudioFlinger::RecordThread::threadLoop()
else {
if (!mActiveTrack->setOverflow()) {
nsecs_t now = systemTime();
- if ((now - lastWarning) > kWarningThrottle) {
+ if ((now - lastWarning) > kWarningThrottleNs) {
LOGW("RecordThread: buffer overflow");
lastWarning = now;
}
@@ -4584,7 +4589,7 @@ status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
result.append(buffer);
snprintf(buffer, SIZE, "In size: %d\n", mInputBytes);
result.append(buffer);
- snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0));
+ snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL));
result.append(buffer);
snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount);
result.append(buffer);
@@ -4619,7 +4624,7 @@ status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer*
mInput->stream->common.standby(&mInput->stream->common);
usleep(kRecordThreadSleepUs);
}
- buffer->raw = 0;
+ buffer->raw = NULL;
buffer->frameCount = 0;
return NOT_ENOUGH_DATA;
}
@@ -4734,7 +4739,7 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l()
mParamCond.signal();
// wait for condition with time out in case the thread calling ThreadBase::setParameters()
// already timed out waiting for the status and will never signal the condition.
- mWaitWorkCV.waitRelative(mLock, kSetParametersTimeout);
+ mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
}
return reconfig;
}
@@ -4782,7 +4787,7 @@ void AudioFlinger::RecordThread::readInputParameters()
if (mRsmpInBuffer) delete mRsmpInBuffer;
if (mRsmpOutBuffer) delete mRsmpOutBuffer;
if (mResampler) delete mResampler;
- mResampler = 0;
+ mResampler = NULL;
mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 6cafa7e..9707cf4 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -547,7 +547,7 @@ private:
Condition mParamCond;
Vector<String8> mNewParameters;
status_t mParamStatus;
- Vector<ConfigEvent *> mConfigEvents;
+ Vector<ConfigEvent> mConfigEvents;
bool mStandby;
int mId;
bool mExiting;
@@ -703,7 +703,7 @@ private:
virtual status_t readyToRun();
virtual void onFirstRef();
- virtual status_t initCheck() const { return (mOutput == 0) ? NO_INIT : NO_ERROR; }
+ virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; }
virtual uint32_t latency() const;
@@ -980,7 +980,7 @@ private:
virtual status_t readyToRun();
virtual void onFirstRef();
- virtual status_t initCheck() const { return (mInput == 0) ? NO_INIT : NO_ERROR; }
+ virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; }
sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l(
const sp<AudioFlinger::Client>& client,
uint32_t sampleRate,
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 8996816..f302b2f 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -50,47 +50,53 @@ AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate)
mState.enabledTracks= 0;
mState.needsChanged = 0;
mState.frameCount = frameCount;
- mState.outputTemp = 0;
- mState.resampleTemp = 0;
+ mState.outputTemp = NULL;
+ mState.resampleTemp = NULL;
mState.hook = process__nop;
track_t* t = mState.tracks;
for (int i=0 ; i<32 ; i++) {
t->needs = 0;
t->volume[0] = UNITY_GAIN;
t->volume[1] = UNITY_GAIN;
+ // no initialization needed
+ // t->prevVolume[0]
+ // t->prevVolume[1]
t->volumeInc[0] = 0;
t->volumeInc[1] = 0;
t->auxLevel = 0;
t->auxInc = 0;
+ // no initialization needed
+ // t->prevAuxLevel
+ // t->frameCount
t->channelCount = 2;
t->enabled = 0;
t->format = 16;
t->channelMask = AUDIO_CHANNEL_OUT_STEREO;
t->buffer.raw = 0;
- t->bufferProvider = 0;
- t->hook = 0;
- t->resampler = 0;
+ t->bufferProvider = NULL;
+ t->hook = NULL;
+ t->resampler = NULL;
t->sampleRate = mSampleRate;
- t->in = 0;
+ t->in = NULL;
t->mainBuffer = NULL;
t->auxBuffer = NULL;
t++;
}
}
- AudioMixer::~AudioMixer()
- {
- track_t* t = mState.tracks;
- for (int i=0 ; i<32 ; i++) {
- delete t->resampler;
- t++;
- }
- delete [] mState.outputTemp;
- delete [] mState.resampleTemp;
- }
+AudioMixer::~AudioMixer()
+{
+ track_t* t = mState.tracks;
+ for (int i=0 ; i<32 ; i++) {
+ delete t->resampler;
+ t++;
+ }
+ delete [] mState.outputTemp;
+ delete [] mState.resampleTemp;
+}
- int AudioMixer::getTrackName()
- {
+int AudioMixer::getTrackName()
+{
uint32_t names = mTrackNames;
uint32_t mask = 1;
int n = 0;
@@ -104,18 +110,18 @@ AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate)
return TRACK0 + n;
}
return -1;
- }
+}
- void AudioMixer::invalidateState(uint32_t mask)
- {
+void AudioMixer::invalidateState(uint32_t mask)
+{
if (mask) {
mState.needsChanged |= mask;
mState.hook = process__validate;
}
}
- void AudioMixer::deleteTrackName(int name)
- {
+void AudioMixer::deleteTrackName(int name)
+{
name -= TRACK0;
if (uint32_t(name) < MAX_NUM_TRACKS) {
ALOGV("deleteTrackName(%d)", name);
@@ -127,7 +133,7 @@ AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate)
if (track.resampler) {
// delete the resampler
delete track.resampler;
- track.resampler = 0;
+ track.resampler = NULL;
track.sampleRate = mSampleRate;
invalidateState(1<<name);
}
@@ -135,7 +141,7 @@ AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate)
track.volumeInc[1] = 0;
mTrackNames &= ~(1<<name);
}
- }
+}
void AudioMixer::enable()
{
@@ -276,7 +282,7 @@ bool AudioMixer::track_t::setResampler(uint32_t value, uint32_t devSampleRate)
if (value!=devSampleRate || resampler) {
if (sampleRate != value) {
sampleRate = value;
- if (resampler == 0) {
+ if (resampler == NULL) {
resampler = AudioResampler::create(
format, channelCount, devSampleRate);
}
@@ -288,12 +294,12 @@ bool AudioMixer::track_t::setResampler(uint32_t value, uint32_t devSampleRate)
bool AudioMixer::track_t::doesResample() const
{
- return resampler != 0;
+ return resampler != NULL;
}
void AudioMixer::track_t::resetResampler()
{
- if (resampler != 0) {
+ if (resampler != NULL) {
resampler->reset();
}
}
@@ -416,11 +422,11 @@ void AudioMixer::process__validate(state_t* state)
} else {
if (state->outputTemp) {
delete [] state->outputTemp;
- state->outputTemp = 0;
+ state->outputTemp = NULL;
}
if (state->resampleTemp) {
delete [] state->resampleTemp;
- state->resampleTemp = 0;
+ state->resampleTemp = NULL;
}
state->hook = process__genericNoResampling;
if (all16BitsStereoNoResample && !volumeRamp) {
@@ -436,33 +442,33 @@ void AudioMixer::process__validate(state_t* state)
countActiveTracks, state->enabledTracks,
all16BitsStereoNoResample, resampling, volumeRamp);
- state->hook(state);
-
- // Now that the volume ramp has been done, set optimal state and
- // track hooks for subsequent mixer process
- if (countActiveTracks) {
- int allMuted = 1;
- uint32_t en = state->enabledTracks;
- while (en) {
- const int i = 31 - __builtin_clz(en);
- en &= ~(1<<i);
- track_t& t = state->tracks[i];
- if (!t.doesResample() && t.volumeRL == 0)
- {
- t.needs |= NEEDS_MUTE_ENABLED;
- t.hook = track__nop;
- } else {
- allMuted = 0;
- }
- }
- if (allMuted) {
- state->hook = process__nop;
- } else if (all16BitsStereoNoResample) {
- if (countActiveTracks == 1) {
- state->hook = process__OneTrack16BitsStereoNoResampling;
- }
- }
- }
+ state->hook(state);
+
+ // Now that the volume ramp has been done, set optimal state and
+ // track hooks for subsequent mixer process
+ if (countActiveTracks) {
+ int allMuted = 1;
+ uint32_t en = state->enabledTracks;
+ while (en) {
+ const int i = 31 - __builtin_clz(en);
+ en &= ~(1<<i);
+ track_t& t = state->tracks[i];
+ if (!t.doesResample() && t.volumeRL == 0)
+ {
+ t.needs |= NEEDS_MUTE_ENABLED;
+ t.hook = track__nop;
+ } else {
+ allMuted = 0;
+ }
+ }
+ if (allMuted) {
+ state->hook = process__nop;
+ } else if (all16BitsStereoNoResample) {
+ if (countActiveTracks == 1) {
+ state->hook = process__OneTrack16BitsStereoNoResampling;
+ }
+ }
+ }
}
static inline
@@ -979,7 +985,7 @@ void AudioMixer::process__genericNoResampling(state_t* state)
}
- // generic code with resampling
+// generic code with resampling
void AudioMixer::process__genericResampling(state_t* state)
{
int32_t* const outTemp = state->outputTemp;
@@ -1159,7 +1165,7 @@ void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state)
}
in1 = buff;
b1.frameCount = numFrames;
- } else {
+ } else {
in1 = b1.i16;
}
frameCount1 = b1.frameCount;
@@ -1201,4 +1207,3 @@ void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state)
// ----------------------------------------------------------------------------
}; // namespace android
-
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index 0dc918d..02722d7 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -64,10 +64,10 @@ public:
FORMAT = 0x4001,
MAIN_BUFFER = 0x4002,
AUX_BUFFER = 0x4003,
- // for TARGET RESAMPLE
+ // for target RESAMPLE
SAMPLE_RATE = 0x4100,
RESET = 0x4101,
- // for TARGET VOLUME (8 channels max)
+ // for target RAMP_VOLUME and VOLUME (8 channels max)
VOLUME0 = 0x4200,
VOLUME1 = 0x4201,
AUXLEVEL = 0x4210,
diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp
index 840d70d..bc4c90c 100644
--- a/services/audioflinger/AudioPolicyService.cpp
+++ b/services/audioflinger/AudioPolicyService.cpp
@@ -31,7 +31,6 @@
#include <utils/threads.h>
#include "AudioPolicyService.h"
#include <cutils/properties.h>
-#include <dlfcn.h>
#include <hardware_legacy/power.h>
#include <media/AudioEffect.h>
#include <media/EffectsFactoryApi.h>
diff --git a/services/audioflinger/AudioPolicyService.h b/services/audioflinger/AudioPolicyService.h
index d898a53..88cb1e9 100644
--- a/services/audioflinger/AudioPolicyService.h
+++ b/services/audioflinger/AudioPolicyService.h
@@ -19,6 +19,7 @@
#include <cutils/misc.h>
#include <cutils/config_utils.h>
+#include <utils/String8.h>
#include <utils/Vector.h>
#include <utils/SortedVector.h>
#include <binder/BinderService.h>
@@ -31,8 +32,6 @@
namespace android {
-class String8;
-
// ----------------------------------------------------------------------------
class AudioPolicyService :
generated by cgit v1.1 at 2024-06-09 22:59:40 +0000