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Commit 3348e36c authored by Andy Hung's avatar Andy Hung
Browse files

Replace int bitDepth with audio_format_t in Resampler 

Remove mBitDepth from class (not used).
Replace with audio_format_t in factory method to distinguish
between float and pcm 16-bit.

Change-Id: I166860796c68285077ef4458d8758d19b82523f9
parent 075abae2

services/audioflinger/AudioMixer.cpp

+1 −13
Original line number Diff line number Diff line
@@ -763,21 +763,9 @@ bool AudioMixer::track_t::setResampler(uint32_t value, uint32_t devSampleRate) 
                    quality = AudioResampler::DEFAULT_QUALITY;

                }



                int bits;

                switch (mMixerInFormat) {

                case AUDIO_FORMAT_PCM_16_BIT:

                    bits = 16;

                    break;

                case AUDIO_FORMAT_PCM_FLOAT:

                    bits = 32; // 32 bits to the AudioResampler::create() indicates float.

                    break;

                default:

                    LOG_ALWAYS_FATAL("Invalid mMixerInFormat: %#x", mMixerInFormat);

                    break;

                }

                ALOGVV("Creating resampler with %d bits\n", bits);

                resampler = AudioResampler::create(

                        bits,

                        mMixerInFormat,

                        // the resampler sees the number of channels after the downmixer, if any

                        (int) (downmixerBufferProvider != NULL ? MAX_NUM_CHANNELS : channelCount),

                        devSampleRate, quality);

services/audioflinger/AudioResampler.cpp

+30 −25
Original line number Diff line number Diff line
@@ -40,8 +40,8 @@ namespace android { 


class AudioResamplerOrder1 : public AudioResampler {

public:

    AudioResamplerOrder1(int bitDepth, int inChannelCount, int32_t sampleRate) :

        AudioResampler(bitDepth, inChannelCount, sampleRate, LOW_QUALITY), mX0L(0), mX0R(0) {

    AudioResamplerOrder1(int inChannelCount, int32_t sampleRate) :

        AudioResampler(inChannelCount, sampleRate, LOW_QUALITY), mX0L(0), mX0R(0) {

    }

    virtual void resample(int32_t* out, size_t outFrameCount,

            AudioBufferProvider* provider);
@@ -145,7 +145,7 @@ static const uint32_t maxMHz = 130; // an arbitrary number that permits 3 VHQ, s 
static pthread_mutex_t mutex = PTHREAD_MUTEX_INITIALIZER;

static uint32_t currentMHz = 0;



AudioResampler* AudioResampler::create(int bitDepth, int inChannelCount,

AudioResampler* AudioResampler::create(audio_format_t format, int inChannelCount,

        int32_t sampleRate, src_quality quality) {



    bool atFinalQuality;
@@ -216,34 +216,41 @@ AudioResampler* AudioResampler::create(int bitDepth, int inChannelCount, 
    default:

    case LOW_QUALITY:

        ALOGV("Create linear Resampler");

        resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate);

        LOG_ALWAYS_FATAL_IF(format != AUDIO_FORMAT_PCM_16_BIT);

        resampler = new AudioResamplerOrder1(inChannelCount, sampleRate);

        break;

    case MED_QUALITY:

        ALOGV("Create cubic Resampler");

        resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate);

        LOG_ALWAYS_FATAL_IF(format != AUDIO_FORMAT_PCM_16_BIT);

        resampler = new AudioResamplerCubic(inChannelCount, sampleRate);

        break;

    case HIGH_QUALITY:

        ALOGV("Create HIGH_QUALITY sinc Resampler");

        resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate);

        LOG_ALWAYS_FATAL_IF(format != AUDIO_FORMAT_PCM_16_BIT);

        resampler = new AudioResamplerSinc(inChannelCount, sampleRate);

        break;

    case VERY_HIGH_QUALITY:

        ALOGV("Create VERY_HIGH_QUALITY sinc Resampler = %d", quality);

        resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate, quality);

        LOG_ALWAYS_FATAL_IF(format != AUDIO_FORMAT_PCM_16_BIT);

        resampler = new AudioResamplerSinc(inChannelCount, sampleRate, quality);

        break;

    case DYN_LOW_QUALITY:

    case DYN_MED_QUALITY:

    case DYN_HIGH_QUALITY:

        ALOGV("Create dynamic Resampler = %d", quality);

        if (bitDepth == 32) { /* bitDepth == 32 signals float precision */

            resampler = new AudioResamplerDyn<float, float, float>(bitDepth, inChannelCount,

        if (format == AUDIO_FORMAT_PCM_FLOAT) {

            resampler = new AudioResamplerDyn<float, float, float>(inChannelCount,

                    sampleRate, quality);

        } else if (quality == DYN_HIGH_QUALITY) {

            resampler = new AudioResamplerDyn<int32_t, int16_t, int32_t>(bitDepth, inChannelCount,

        } else {

            LOG_ALWAYS_FATAL_IF(format != AUDIO_FORMAT_PCM_16_BIT);

            if (quality == DYN_HIGH_QUALITY) {

                resampler = new AudioResamplerDyn<int32_t, int16_t, int32_t>(inChannelCount,

                        sampleRate, quality);

            } else {

            resampler = new AudioResamplerDyn<int16_t, int16_t, int32_t>(bitDepth, inChannelCount,

                resampler = new AudioResamplerDyn<int16_t, int16_t, int32_t>(inChannelCount,

                        sampleRate, quality);

            }

        }

        break;

    }
@@ -252,18 +259,17 @@ AudioResampler* AudioResampler::create(int bitDepth, int inChannelCount, 
    return resampler;

}



AudioResampler::AudioResampler(int bitDepth, int inChannelCount,

AudioResampler::AudioResampler(int inChannelCount,

        int32_t sampleRate, src_quality quality) :

    mBitDepth(bitDepth), mChannelCount(inChannelCount),

        mChannelCount(inChannelCount),

        mSampleRate(sampleRate), mInSampleRate(sampleRate), mInputIndex(0),

        mPhaseFraction(0), mLocalTimeFreq(0),

        mPTS(AudioBufferProvider::kInvalidPTS), mQuality(quality) {

    // sanity check on format

    if ((bitDepth != 16 && (quality < DYN_LOW_QUALITY || bitDepth != 32))

            || inChannelCount < 1



    if (inChannelCount < 1

            || inChannelCount > (quality < DYN_LOW_QUALITY ? 2 : 8)) {

        LOG_ALWAYS_FATAL("Unsupported sample format %d quality %d bits, %d channels",

                quality, bitDepth, inChannelCount);

        LOG_ALWAYS_FATAL("Unsupported sample format %d quality %d channels",

                quality, inChannelCount);

    }

    if (sampleRate <= 0) {

        LOG_ALWAYS_FATAL("Unsupported sample rate %d Hz", sampleRate);
@@ -272,7 +278,6 @@ AudioResampler::AudioResampler(int bitDepth, int inChannelCount, 
    // initialize common members

    mVolume[0] = mVolume[1] = 0;

    mBuffer.frameCount = 0;



}



AudioResampler::~AudioResampler() {

services/audioflinger/AudioResampler.h

+3 −3
Original line number Diff line number Diff line
@@ -22,6 +22,7 @@ 
#include <cutils/compiler.h>



#include <media/AudioBufferProvider.h>

#include <system/audio.h>



namespace android {

// ----------------------------------------------------------------------------
@@ -46,7 +47,7 @@ public: 
        DYN_HIGH_QUALITY=7,

    };



    static AudioResampler* create(int bitDepth, int inChannelCount,

    static AudioResampler* create(audio_format_t format, int inChannelCount,

            int32_t sampleRate, src_quality quality=DEFAULT_QUALITY);



    virtual ~AudioResampler();
@@ -86,7 +87,7 @@ protected: 
    // multiplier to calculate fixed point phase increment

    static const double kPhaseMultiplier;



    AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate, src_quality quality);

    AudioResampler(int inChannelCount, int32_t sampleRate, src_quality quality);



    // prevent copying

    AudioResampler(const AudioResampler&);
@@ -94,7 +95,6 @@ protected: 


    int64_t calculateOutputPTS(int outputFrameIndex);



    const int32_t mBitDepth;

    const int32_t mChannelCount;

    const int32_t mSampleRate;

    int32_t mInSampleRate;

services/audioflinger/AudioResamplerCubic.h

+2 −2
Original line number Diff line number Diff line
@@ -28,8 +28,8 @@ namespace android { 


class AudioResamplerCubic : public AudioResampler {

public:

    AudioResamplerCubic(int bitDepth, int inChannelCount, int32_t sampleRate) :

        AudioResampler(bitDepth, inChannelCount, sampleRate, MED_QUALITY) {

    AudioResamplerCubic(int inChannelCount, int32_t sampleRate) :

        AudioResampler(inChannelCount, sampleRate, MED_QUALITY) {

    }

    virtual void resample(int32_t* out, size_t outFrameCount,

            AudioBufferProvider* provider);

services/audioflinger/AudioResamplerDyn.cpp

+2 −2
Original line number Diff line number Diff line
@@ -162,9 +162,9 @@ void AudioResamplerDyn<TC, TI, TO>::Constants::set( 
}



template<typename TC, typename TI, typename TO>

AudioResamplerDyn<TC, TI, TO>::AudioResamplerDyn(int bitDepth,

AudioResamplerDyn<TC, TI, TO>::AudioResamplerDyn(

        int inChannelCount, int32_t sampleRate, src_quality quality)

    : AudioResampler(bitDepth, inChannelCount, sampleRate, quality),

    : AudioResampler(inChannelCount, sampleRate, quality),

      mResampleFunc(0), mFilterSampleRate(0), mFilterQuality(DEFAULT_QUALITY),

    mCoefBuffer(NULL)

{