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-rw-r--r--services/audioflinger/AudioFlinger.cpp79
1 files changed, 38 insertions, 41 deletions
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index d83d19a..c5ad0f5 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -442,11 +442,9 @@ sp<IAudioTrack> AudioFlinger::createTrack(
audio_format_t format,
uint32_t channelMask,
int frameCount,
- // FIXME dead, remove from IAudioFlinger
- uint32_t flags,
+ IAudioFlinger::track_flags_t flags,
const sp<IMemory>& sharedBuffer,
audio_io_handle_t output,
- bool isTimed,
int *sessionId,
status_t *status)
{
@@ -504,6 +502,7 @@ sp<IAudioTrack> AudioFlinger::createTrack(
}
ALOGV("createTrack() lSessionId: %d", lSessionId);
+ bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0;
track = thread->createTrack_l(client, streamType, sampleRate, format,
channelMask, frameCount, sharedBuffer, lSessionId, isTimed, &lStatus);
@@ -894,7 +893,8 @@ status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8&
// indicate output device change to all input threads for pre processing
AudioParameter param = AudioParameter(keyValuePairs);
int value;
- if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
+ if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) &&
+ (value != 0)) {
for (size_t i = 0; i < mRecordThreads.size(); i++) {
mRecordThreads.valueAt(i)->setParameters(keyValuePairs);
}
@@ -1588,7 +1588,7 @@ void AudioFlinger::PlaybackThread::onFirstRef()
}
// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
-sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(
+sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(
const sp<AudioFlinger::Client>& client,
audio_stream_type_t streamType,
uint32_t sampleRate,
@@ -2337,7 +2337,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
size_t tracksWithEffect = 0;
float masterVolume = mMasterVolume;
- bool masterMute = mMasterMute;
+ bool masterMute = mMasterMute;
if (masterMute) {
masterVolume = 0;
@@ -2376,7 +2376,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
// +1 for rounding and +1 for additional sample needed for interpolation
minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1;
// add frames already consumed but not yet released by the resampler
- // because cblk->framesReady() will include these frames
+ // because cblk->framesReady() will include these frames
minFrames += mAudioMixer->getUnreleasedFrames(track->name());
// the minimum track buffer size is normally twice the number of frames necessary
// to fill one buffer and the resampler should not leave more than one buffer worth
@@ -2514,6 +2514,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
// reset retry count
track->mRetryCount = kMaxTrackRetries;
+
// If one track is ready, set the mixer ready if:
// - the mixer was not ready during previous round OR
// - no other track is not ready
@@ -3372,19 +3373,19 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
}
} else {
mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
- // construct the shared structure in-place.
- new(mCblk) audio_track_cblk_t();
- // clear all buffers
- mCblk->frameCount = frameCount;
- mCblk->sampleRate = sampleRate;
- mChannelCount = channelCount;
- mChannelMask = channelMask;
- 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
- // written to buffer (other flags are cleared)
- mCblk->flags = CBLK_UNDERRUN_ON;
- mBufferEnd = (uint8_t *)mBuffer + bufferSize;
+ // construct the shared structure in-place.
+ new(mCblk) audio_track_cblk_t();
+ // clear all buffers
+ mCblk->frameCount = frameCount;
+ mCblk->sampleRate = sampleRate;
+ mChannelCount = channelCount;
+ mChannelMask = channelMask;
+ 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
+ // written to buffer (other flags are cleared)
+ mCblk->flags = CBLK_UNDERRUN_ON;
+ mBufferEnd = (uint8_t *)mBuffer + bufferSize;
}
}
@@ -3479,23 +3480,27 @@ AudioFlinger::PlaybackThread::Track::Track(
const sp<IMemory>& sharedBuffer,
int sessionId)
: TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId),
- mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL),
+ mMute(false),
+ // mFillingUpStatus ?
+ // mRetryCount initialized later when needed
+ mSharedBuffer(sharedBuffer),
+ mStreamType(streamType),
+ mName(-1), // see note below
+ mMainBuffer(thread->mixBuffer()),
+ mAuxBuffer(NULL),
mAuxEffectId(0), mHasVolumeController(false)
{
if (mCblk != NULL) {
- if (thread != NULL) {
- mName = thread->getTrackName_l();
- mMainBuffer = thread->mixBuffer();
- }
- ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid());
- if (mName < 0) {
- ALOGE("no more track names available");
- }
- mStreamType = streamType;
// NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of
// 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t);
+ // to avoid leaking a track name, do not allocate one unless there is an mCblk
+ mName = thread->getTrackName_l();
+ if (mName < 0) {
+ ALOGE("no more track names available");
+ }
}
+ ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid());
}
AudioFlinger::PlaybackThread::Track::~Track()
@@ -3792,16 +3797,9 @@ AudioFlinger::PlaybackThread::TimedTrack::create(
if (!client->reserveTimedTrack())
return NULL;
- sp<TimedTrack> track = new TimedTrack(
+ return new TimedTrack(
thread, client, streamType, sampleRate, format, channelMask, frameCount,
sharedBuffer, sessionId);
-
- if (track == NULL) {
- client->releaseTimedTrack();
- return NULL;
- }
-
- return track;
}
AudioFlinger::PlaybackThread::TimedTrack::TimedTrack(
@@ -4261,7 +4259,7 @@ void AudioFlinger::RecordThread::RecordTrack::stop()
RecordThread *recordThread = (RecordThread *)thread.get();
recordThread->stop(this);
TrackBase::reset();
- // Force overerrun condition to avoid false overrun callback until first data is
+ // Force overrun condition to avoid false overrun callback until first data is
// read from buffer
android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags);
}
@@ -4677,8 +4675,7 @@ sp<IAudioRecord> AudioFlinger::openRecord(
audio_format_t format,
uint32_t channelMask,
int frameCount,
- // FIXME dead, remove from IAudioFlinger
- uint32_t flags,
+ IAudioFlinger::track_flags_t flags,
int *sessionId,
status_t *status)
{
generated by cgit v1.1 at 2025-11-11 05:23:57 +0000