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Commit 9b58f63e authored by Glenn Kasten's avatar Glenn Kasten
Browse files

Move members from ThreadBase to PlaybackThread 

Move mNormalFrameCount and frameCountHAL(), since they're not used by
record threads.  Also comment which fields are updated by readParameters().

Change-Id: I5fc0a8a89cc637976f22d49271a5a3e136dab4e1
parent 2314e90e

services/audioflinger/Threads.cpp

+6 −7
Original line number Diff line number Diff line
@@ -266,10 +266,9 @@ AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio 
        audio_devices_t outDevice, audio_devices_t inDevice, type_t type)

    :   Thread(false /*canCallJava*/),

        mType(type),

        mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0),

        // mChannelMask

        mChannelCount(0),

        mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID),

        mAudioFlinger(audioFlinger),

        // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, and mFormat are

        // set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters()

        mParamStatus(NO_ERROR),

        mStandby(false), mOutDevice(outDevice), mInDevice(inDevice),

        mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id),
@@ -425,8 +424,6 @@ void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 
    result.append(buffer);

    snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount);

    result.append(buffer);

    snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount);

    result.append(buffer);

    snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount);

    result.append(buffer);

    snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask);
@@ -932,6 +929,7 @@ AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinge 
                                             audio_devices_t device,

                                             type_t type)

    :   ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type),

        mNormalFrameCount(0), mMixBuffer(NULL),

        mAllocMixBuffer(NULL), mSuspended(0), mBytesWritten(0),

        // mStreamTypes[] initialized in constructor body

        mOutput(output),
@@ -1054,6 +1052,8 @@ void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& 


    snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);

    result.append(buffer);

    snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount);

    result.append(buffer);

    snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n",

            ns2ms(systemTime() - mLastWriteTime));

    result.append(buffer);
@@ -4920,7 +4920,6 @@ void AudioFlinger::RecordThread::readInputParameters() 
    mFrameSize = audio_stream_frame_size(&mInput->stream->common);

    mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common);

    mFrameCount = mInputBytes / mFrameSize;

    mNormalFrameCount = mFrameCount; // not used by record, but used by input effects

    mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];



    if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2)

services/audioflinger/Threads.h

+16 −5
Original line number Diff line number Diff line
@@ -126,10 +126,8 @@ public: 
                audio_channel_mask_t channelMask() const { return mChannelMask; }

                audio_format_t format() const { return mFormat; }

                // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects,

                // and returns the normal mix buffer's frame count.

                size_t      frameCount() const { return mNormalFrameCount; }

                // Return's the HAL's frame count i.e. fast mixer buffer size.

                size_t      frameCountHAL() const { return mFrameCount; }

                // and returns the [normal mix] buffer's frame count.

    virtual     size_t      frameCount() const = 0;

                size_t      frameSize() const { return mFrameSize; }



    // Should be "virtual status_t requestExitAndWait()" and override same
@@ -263,9 +261,11 @@ protected: 
                Condition               mWaitWorkCV;



                const sp<AudioFlinger>  mAudioFlinger;



                // updated by PlaybackThread::readOutputParameters() or

                // RecordThread::readInputParameters()

                uint32_t                mSampleRate;

                size_t                  mFrameCount;       // output HAL, direct output, record

                size_t                  mNormalFrameCount; // normal mixer and effects

                audio_channel_mask_t    mChannelMask;

                uint32_t                mChannelCount;

                size_t                  mFrameSize;
@@ -461,8 +461,15 @@ public: 
                // called with AudioFlinger lock held

                        void     invalidateTracks(audio_stream_type_t streamType);



    virtual     size_t      frameCount() const { return mNormalFrameCount; }



                // Return's the HAL's frame count i.e. fast mixer buffer size.

                size_t      frameCountHAL() const { return mFrameCount; }



protected:

    // updated by readOutputParameters()

    size_t                          mNormalFrameCount;  // normal mixer and effects



    int16_t*                        mMixBuffer;         // frame size aligned mix buffer

    int8_t*                         mAllocMixBuffer;    // mixer buffer allocation address
@@ -871,6 +878,8 @@ public: 
    static void syncStartEventCallback(const wp<SyncEvent>& event);

           void handleSyncStartEvent(const sp<SyncEvent>& event);



    virtual size_t      frameCount() const { return mFrameCount; }



private:

            void clearSyncStartEvent();
@@ -886,6 +895,8 @@ private: 
            // is used together with mStartStopCond to indicate start()/stop() progress

            sp<RecordTrack>                     mActiveTrack;

            Condition                           mStartStopCond;



            // updated by RecordThread::readInputParameters()

            AudioResampler                      *mResampler;

            int32_t                             *mRsmpOutBuffer;

            int16_t                             *mRsmpInBuffer;