diff options
Diffstat (limited to 'services/audioflinger')
| -rw-r--r-- | services/audioflinger/AudioFlinger.cpp | 13 | ||||
| -rw-r--r-- | services/audioflinger/AudioFlinger.h | 11 |
2 files changed, 14 insertions, 10 deletions
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp index 1a37f4f..14f74b5 100644 --- a/services/audioflinger/AudioFlinger.cpp +++ b/services/audioflinger/AudioFlinger.cpp @@ -169,8 +169,8 @@ static const int kPriorityFastMixer = 3; // for the track. The client then sub-divides this into smaller buffers for its use. // Currently the client uses double-buffering by default, but doesn't tell us about that. // So for now we just assume that client is double-buffered. -// FIXME It would be better for client to tell us whether it wants double-buffering or N-buffering, -// so we could allocate the right amount of memory. +// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or +// N-buffering, so AudioFlinger could allocate the right amount of memory. // See the client's minBufCount and mNotificationFramesAct calculations for details. static const int kFastTrackMultiplier = 2; @@ -258,11 +258,11 @@ void AudioFlinger::onFirstRef() AudioFlinger::~AudioFlinger() { while (!mRecordThreads.isEmpty()) { - // closeInput() will remove first entry from mRecordThreads + // closeInput_nonvirtual() will remove specified entry from mRecordThreads closeInput_nonvirtual(mRecordThreads.keyAt(0)); } while (!mPlaybackThreads.isEmpty()) { - // closeOutput() will remove first entry from mPlaybackThreads + // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); } @@ -1134,7 +1134,7 @@ sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, audio_devices_t device, type_t type) - : Thread(false), + : Thread(false /*canCallJava*/), mType(type), mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), // mChannelMask @@ -1142,6 +1142,7 @@ AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), mParamStatus(NO_ERROR), mStandby(false), mDevice(device), mId(id), + // mName will be set by concrete (non-virtual) subclass mDeathRecipient(new PMDeathRecipient(this)) { } @@ -6097,7 +6098,7 @@ bool AudioFlinger::RecordThread::threadLoop() if (mChannelCount == 1 && mReqChannelCount == 1) { framesOut >>= 1; } - mResampler->resample(mRsmpOutBuffer, framesOut, this); + mResampler->resample(mRsmpOutBuffer, framesOut, this /* AudioBufferProvider* */); // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() // are 32 bit aligned which should be always true. if (mChannelCount == 2 && mReqChannelCount == 1) { diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h index b4aefc1..4723cd9 100644 --- a/services/audioflinger/AudioFlinger.h +++ b/services/audioflinger/AudioFlinger.h @@ -454,8 +454,9 @@ private: /*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const sp<IMemory> mCblkMemory; audio_track_cblk_t* mCblk; - void* mBuffer; - void* mBufferEnd; + void* mBuffer; // start of track buffer, typically in shared memory + void* mBufferEnd; // &mBuffer[mFrameCount * frameSize], where frameSize + // is based on mChannelCount and 16-bit samples uint32_t mFrameCount; // we don't really need a lock for these track_state mState; @@ -1364,6 +1365,7 @@ private: // record thread class RecordThread : public ThreadBase, public AudioBufferProvider + // derives from AudioBufferProvider interface for use by resampler { public: @@ -1420,7 +1422,7 @@ private: void dumpInternals(int fd, const Vector<String16>& args); void dumpTracks(int fd, const Vector<String16>& args); - // Thread + // Thread virtuals virtual bool threadLoop(); virtual status_t readyToRun(); @@ -1968,9 +1970,10 @@ mutable Mutex mLock; // mutex for process, commands and handl DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; - // both are protected by mLock + // member variables below are protected by mLock float mMasterVolume; bool mMasterMute; + // end of variables protected by mLock DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; |