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-rw-r--r--libs/audioflinger/AudioFlinger.cpp69
1 files changed, 34 insertions, 35 deletions
diff --git a/libs/audioflinger/AudioFlinger.cpp b/libs/audioflinger/AudioFlinger.cpp
index b56221f..8a19fbd 100644
--- a/libs/audioflinger/AudioFlinger.cpp
+++ b/libs/audioflinger/AudioFlinger.cpp
@@ -499,7 +499,8 @@ 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);
@@ -817,19 +818,22 @@ 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);
- 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);
+ 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);
}
@@ -890,7 +894,7 @@ void AudioFlinger::handleRouteDisablesA2dp_l(int routes)
}
LOGV("mA2dpDisableCount decremented to %d", mA2dpDisableCount);
} else {
- LOGE("mA2dpDisableCount is already zero");
+ LOGV("mA2dpDisableCount is already zero");
}
}
}
@@ -1277,7 +1281,7 @@ 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;
@@ -1553,7 +1557,6 @@ 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,
@@ -1563,7 +1566,6 @@ AudioFlinger::MixerThread::TrackBase::TrackBase(
: RefBase(),
mMixerThread(mixerThread),
mClient(client),
- mStreamType(streamType),
mFrameCount(0),
mState(IDLE),
mClientTid(-1),
@@ -1594,8 +1596,8 @@ 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));
@@ -1618,8 +1620,8 @@ 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
@@ -1680,7 +1682,7 @@ 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 {
@@ -1713,12 +1715,13 @@ 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()
@@ -1902,15 +1905,15 @@ 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)
{
}
@@ -2235,7 +2238,7 @@ status_t AudioFlinger::TrackHandle::onTransact(
sp<IAudioRecord> AudioFlinger::openRecord(
pid_t pid,
- int streamType,
+ int inputSource,
uint32_t sampleRate,
int format,
int channelCount,
@@ -2258,18 +2261,12 @@ 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;
@@ -2301,7 +2298,7 @@ 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) {
@@ -2407,7 +2404,9 @@ 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 2025-08-15 13:41:51 +0000