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Commit 34af0264 authored by Glenn Kasten's avatar Glenn Kasten
Browse files

Add resampler comments and fix a typo 

Change-Id: Ie071673875f663de4212eed4a4dff89d51a5a915
parent 39a60318

services/audioflinger/AudioFlinger.cpp

+1 −1
Original line number Diff line number Diff line
@@ -1714,7 +1714,7 @@ audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 
        AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);



        // Start record thread

        // RecorThread require both input and output device indication to forward to audio

        // RecordThread requires both input and output device indication to forward to audio

        // pre processing modules

        thread = new RecordThread(this,

                                  input,

services/audioflinger/AudioResampler.h

+8 −0
Original line number Diff line number Diff line
@@ -56,6 +56,14 @@ public: 
    // set the PTS of the next buffer output by the resampler

    virtual void setPTS(int64_t pts);



    // Resample int16_t samples from provider and accumulate into 'out'.

    // A mono provider delivers a sequence of samples.

    // A stereo provider delivers a sequence of interleaved pairs of samples.

    // Multi-channel providers are not supported.

    // In either case, 'out' holds interleaved pairs of fixed-point signed Q19.12.

    // That is, for a mono provider, there is an implicit up-channeling.

    // Since this method accumulates, the caller is responsible for clearing 'out' initially.

    // FIXME assumes provider is always successful; it should return the actual frame count.

    virtual void resample(int32_t* out, size_t outFrameCount,

            AudioBufferProvider* provider) = 0;

services/audioflinger/Threads.cpp

+5 −2
Original line number Diff line number Diff line
@@ -3787,7 +3787,8 @@ bool AudioFlinger::RecordThread::threadLoop() 
                } else {

                    // resampling



                    memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t));

                    // resampler accumulates, but we only have one source track

                    memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t));

                    // alter output frame count as if we were expecting stereo samples

                    if (mChannelCount == 1 && mReqChannelCount == 1) {

                        framesOut >>= 1;
@@ -3797,6 +3798,7 @@ bool AudioFlinger::RecordThread::threadLoop() 
                    // 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) {

                        // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t

                        ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);

                        // the resampler always outputs stereo samples:

                        // do post stereo to mono conversion
@@ -3805,6 +3807,7 @@ bool AudioFlinger::RecordThread::threadLoop() 
                    } else {

                        ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);

                    }

                    // now done with mRsmpOutBuffer



                }

                if (mFramestoDrop == 0) {
@@ -4385,7 +4388,7 @@ void AudioFlinger::RecordThread::readInputParameters() 
        mResampler = AudioResampler::create(16, channelCount, mReqSampleRate);

        mResampler->setSampleRate(mSampleRate);

        mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);

        mRsmpOutBuffer = new int32_t[mFrameCount * 2];

        mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2];



        // optmization: if mono to mono, alter input frame count as if we were inputing

        // stereo samples

services/audioflinger/Threads.h

+2 −1
Original line number Diff line number Diff line
@@ -794,8 +794,9 @@ private: 
            sp<RecordTrack>                     mActiveTrack;

            Condition                           mStartStopCond;

            AudioResampler                      *mResampler;

            // interleaved stereo pairs of fixed-point signed Q19.12

            int32_t                             *mRsmpOutBuffer;

            int16_t                             *mRsmpInBuffer;

            int16_t                             *mRsmpInBuffer; // [mFrameCount * mChannelCount]

            size_t                              mRsmpInIndex;

            size_t                              mInputBytes;

            const uint32_t                      mReqChannelCount;