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Commit 116a4988 authored by Andy Hung's avatar Andy Hung
Browse files

AudioEffect: Convert aux buffer format from q4.27 to float 

Test: Solo Tester aux reverb
Bug: 69853407
Change-Id: I72ca2377036e5ccfd5f4631ac6cc5e86d2e40433
parent fa69ca30

media/libaudioclient/include/media/AudioMixer.h

+3 −0
Original line number Diff line number Diff line
@@ -33,6 +33,9 @@ 
// FIXME This is actually unity gain, which might not be max in future, expressed in U.12

#define MAX_GAIN_INT AudioMixer::UNITY_GAIN_INT



// This must match frameworks/av/services/audioflinger/Configuration.h

#define FLOAT_AUX



namespace android {



// ----------------------------------------------------------------------------

media/libaudioprocessing/AudioMixer.cpp

+67 −38
Original line number Diff line number Diff line
@@ -71,12 +71,20 @@ T max(const T& x, const T& y) { 


// Set kUseNewMixer to true to use the new mixer engine always. Otherwise the

// original code will be used for stereo sinks, the new mixer for multichannel.

static const bool kUseNewMixer = true;

static constexpr bool kUseNewMixer = true;



// Set kUseFloat to true to allow floating input into the mixer engine.

// If kUseNewMixer is false, this is ignored or may be overridden internally

// because of downmix/upmix support.

static const bool kUseFloat = true;

static constexpr bool kUseFloat = true;



#ifdef FLOAT_AUX

using TYPE_AUX = float;

static_assert(kUseNewMixer && kUseFloat,

        "kUseNewMixer and kUseFloat must be true for FLOAT_AUX option");

#else

using TYPE_AUX = int32_t; // q4.27

#endif



// Set to default copy buffer size in frames for input processing.

static const size_t kCopyBufferFrameCount = 256;
@@ -861,16 +869,25 @@ inline void AudioMixer::track_t::adjustVolumeRamp(bool aux, bool useFloat) 
            }

        }

    }

    /* TODO: aux is always integer regardless of output buffer type */



    if (aux) {

        if (((auxInc>0) && (((prevAuxLevel+auxInc)>>16) >= auxLevel)) ||

                ((auxInc<0) && (((prevAuxLevel+auxInc)>>16) <= auxLevel))) {

#ifdef FLOAT_AUX

        if (useFloat) {

            if ((mAuxInc > 0.f && mPrevAuxLevel + mAuxInc >= mAuxLevel) ||

                    (mAuxInc < 0.f && mPrevAuxLevel + mAuxInc <= mAuxLevel)) {

                auxInc = 0;

                prevAuxLevel = auxLevel << 16;

            mAuxInc = 0.;

                mAuxInc = 0.f;

                mPrevAuxLevel = mAuxLevel;

            }

        } else

#endif

        if ((auxInc > 0 && ((prevAuxLevel + auxInc) >> 16) >= auxLevel) ||

                (auxInc < 0 && ((prevAuxLevel + auxInc) >> 16) <= auxLevel)) {

            auxInc = 0;

            prevAuxLevel = auxLevel << 16;

            mAuxInc = 0.f;

            mPrevAuxLevel = mAuxLevel;

        } else {

            //ALOGV("aux ramp: %d %d %d", auxLevel << 16, prevAuxLevel, auxInc);

        }

    }

}
@@ -1694,7 +1711,7 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state) 
/* MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)

 * TO: int32_t (Q4.27) or float

 * TI: int32_t (Q4.27) or int16_t (Q0.15) or float

 * TA: int32_t (Q4.27)

 * TA: int32_t (Q4.27) or float

 */

template <int MIXTYPE,

        typename TO, typename TI, typename TV, typename TA, typename TAV>
@@ -1738,7 +1755,7 @@ static void volumeRampMulti(uint32_t channels, TO* out, size_t frameCount, 
/* MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)

 * TO: int32_t (Q4.27) or float

 * TI: int32_t (Q4.27) or int16_t (Q0.15) or float

 * TA: int32_t (Q4.27)

 * TA: int32_t (Q4.27) or float

 */

template <int MIXTYPE,

        typename TO, typename TI, typename TV, typename TA, typename TAV>
@@ -1778,7 +1795,7 @@ static void volumeMulti(uint32_t channels, TO* out, size_t frameCount, 
 * ADJUSTVOL   (set to true if volume ramp parameters needs adjustment afterwards)

 * TO: int32_t (Q4.27) or float

 * TI: int32_t (Q4.27) or int16_t (Q0.15) or float

 * TA: int32_t (Q4.27)

 * TA: int32_t (Q4.27) or float

 */

template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,

    typename TO, typename TI, typename TA>
@@ -1788,13 +1805,25 @@ void AudioMixer::volumeMix(TO *out, size_t outFrames, 
    if (USEFLOATVOL) {

        if (ramp) {

            volumeRampMulti<MIXTYPE>(t->mMixerChannelCount, out, outFrames, in, aux,

                    t->mPrevVolume, t->mVolumeInc, &t->prevAuxLevel, t->auxInc);

                    t->mPrevVolume, t->mVolumeInc,

#ifdef FLOAT_AUX

                    &t->mPrevAuxLevel, t->mAuxInc

#else

                    &t->prevAuxLevel, t->auxInc

#endif

                );

            if (ADJUSTVOL) {

                t->adjustVolumeRamp(aux != NULL, true);

            }

        } else {

            volumeMulti<MIXTYPE>(t->mMixerChannelCount, out, outFrames, in, aux,

                    t->mVolume, t->auxLevel);

                    t->mVolume,

#ifdef FLOAT_AUX

                    t->mAuxLevel

#else

                    t->auxLevel

#endif

            );

        }

    } else {

        if (ramp) {
@@ -1851,7 +1880,7 @@ void AudioMixer::process_NoResampleOneTrack(state_t* state) 
        }



        const size_t outFrames = b.frameCount;

        volumeMix<MIXTYPE, is_same<TI, float>::value, false> (

        volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, false /* ADJUSTVOL */> (

                out, outFrames, in, aux, ramp, t);



        out += outFrames * channels;
@@ -1874,7 +1903,7 @@ void AudioMixer::process_NoResampleOneTrack(state_t* state) 
 * MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)

 * TO: int32_t (Q4.27) or float

 * TI: int32_t (Q4.27) or int16_t (Q0.15) or float

 * TA: int32_t (Q4.27)

 * TA: int32_t (Q4.27) or float

 */

template <int MIXTYPE, typename TO, typename TI, typename TA>

void AudioMixer::track__Resample(track_t* t, TO* out, size_t outFrameCount, TO* temp, TA* aux)
@@ -1890,7 +1919,7 @@ void AudioMixer::track__Resample(track_t* t, TO* out, size_t outFrameCount, TO* 
        memset(temp, 0, outFrameCount * t->mMixerChannelCount * sizeof(TO));

        t->resampler->resample((int32_t*)temp, outFrameCount, t->bufferProvider);



        volumeMix<MIXTYPE, is_same<TI, float>::value, true>(

        volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, true /* ADJUSTVOL */>(

                out, outFrameCount, temp, aux, ramp, t);



    } else { // constant volume gain
@@ -1905,7 +1934,7 @@ void AudioMixer::track__Resample(track_t* t, TO* out, size_t outFrameCount, TO* 
 * MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)

 * TO: int32_t (Q4.27) or float

 * TI: int32_t (Q4.27) or int16_t (Q0.15) or float

 * TA: int32_t (Q4.27)

 * TA: int32_t (Q4.27) or float

 */

template <int MIXTYPE, typename TO, typename TI, typename TA>

void AudioMixer::track__NoResample(track_t* t, TO* out, size_t frameCount,
@@ -1914,7 +1943,7 @@ void AudioMixer::track__NoResample(track_t* t, TO* out, size_t frameCount, 
    ALOGVV("track__NoResample\n");

    const TI *in = static_cast<const TI *>(t->in);



    volumeMix<MIXTYPE, is_same<TI, float>::value, true>(

    volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, true /* ADJUSTVOL */>(

            out, frameCount, in, aux, t->needsRamp(), t);



    // MIXTYPE_MONOEXPAND reads a single input channel and expands to NCHAN output channels.
@@ -1990,11 +2019,11 @@ AudioMixer::hook_t AudioMixer::getTrackHook(int trackType, uint32_t channelCount 
    case TRACKTYPE_RESAMPLE:

        switch (mixerInFormat) {

        case AUDIO_FORMAT_PCM_FLOAT:

            return (AudioMixer::hook_t)

                    track__Resample<MIXTYPE_MULTI, float /*TO*/, float /*TI*/, int32_t /*TA*/>;

            return (AudioMixer::hook_t)track__Resample<

                    MIXTYPE_MULTI, float /*TO*/, float /*TI*/, TYPE_AUX>;

        case AUDIO_FORMAT_PCM_16_BIT:

            return (AudioMixer::hook_t)\

                    track__Resample<MIXTYPE_MULTI, int32_t, int16_t, int32_t>;

            return (AudioMixer::hook_t)track__Resample<

                    MIXTYPE_MULTI, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;

        default:

            LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);

            break;
@@ -2003,11 +2032,11 @@ AudioMixer::hook_t AudioMixer::getTrackHook(int trackType, uint32_t channelCount 
    case TRACKTYPE_NORESAMPLEMONO:

        switch (mixerInFormat) {

        case AUDIO_FORMAT_PCM_FLOAT:

            return (AudioMixer::hook_t)

                    track__NoResample<MIXTYPE_MONOEXPAND, float, float, int32_t>;

            return (AudioMixer::hook_t)track__NoResample<

                            MIXTYPE_MONOEXPAND, float /*TO*/, float /*TI*/, TYPE_AUX>;

        case AUDIO_FORMAT_PCM_16_BIT:

            return (AudioMixer::hook_t)

                    track__NoResample<MIXTYPE_MONOEXPAND, int32_t, int16_t, int32_t>;

            return (AudioMixer::hook_t)track__NoResample<

                            MIXTYPE_MONOEXPAND, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;

        default:

            LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);

            break;
@@ -2016,11 +2045,11 @@ AudioMixer::hook_t AudioMixer::getTrackHook(int trackType, uint32_t channelCount 
    case TRACKTYPE_NORESAMPLE:

        switch (mixerInFormat) {

        case AUDIO_FORMAT_PCM_FLOAT:

            return (AudioMixer::hook_t)

                    track__NoResample<MIXTYPE_MULTI, float, float, int32_t>;

            return (AudioMixer::hook_t)track__NoResample<

                    MIXTYPE_MULTI, float /*TO*/, float /*TI*/, TYPE_AUX>;

        case AUDIO_FORMAT_PCM_16_BIT:

            return (AudioMixer::hook_t)

                    track__NoResample<MIXTYPE_MULTI, int32_t, int16_t, int32_t>;

            return (AudioMixer::hook_t)track__NoResample<

                    MIXTYPE_MULTI, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;

        default:

            LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);

            break;
@@ -2055,11 +2084,11 @@ AudioMixer::process_hook_t AudioMixer::getProcessHook(int processType, uint32_t 
    case AUDIO_FORMAT_PCM_FLOAT:

        switch (mixerOutFormat) {

        case AUDIO_FORMAT_PCM_FLOAT:

            return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,

                    float /*TO*/, float /*TI*/, int32_t /*TA*/>;

            return process_NoResampleOneTrack<

                    MIXTYPE_MULTI_SAVEONLY, float /*TO*/, float /*TI*/, TYPE_AUX>;

        case AUDIO_FORMAT_PCM_16_BIT:

            return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,

                    int16_t, float, int32_t>;

            return process_NoResampleOneTrack<

                    MIXTYPE_MULTI_SAVEONLY, int16_t /*TO*/, float /*TI*/, TYPE_AUX>;

        default:

            LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);

            break;
@@ -2068,11 +2097,11 @@ AudioMixer::process_hook_t AudioMixer::getProcessHook(int processType, uint32_t 
    case AUDIO_FORMAT_PCM_16_BIT:

        switch (mixerOutFormat) {

        case AUDIO_FORMAT_PCM_FLOAT:

            return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,

                    float, int16_t, int32_t>;

            return process_NoResampleOneTrack<

                    MIXTYPE_MULTI_SAVEONLY, float /*TO*/, int16_t /*TI*/, TYPE_AUX>;

        case AUDIO_FORMAT_PCM_16_BIT:

            return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,

                    int16_t, int16_t, int32_t>;

            return process_NoResampleOneTrack<

                    MIXTYPE_MULTI_SAVEONLY, int16_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;

        default:

            LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);

            break;

media/libaudioprocessing/AudioMixerOps.h

+7 −4
Original line number Diff line number Diff line
@@ -188,13 +188,13 @@ inline void MixAccum(TO *auxaccum, TI value) { 


template<>

inline void MixAccum<float, int16_t>(float *auxaccum, int16_t value) {

    static const float norm = 1. / (1 << 15);

    static constexpr float norm = 1. / (1 << 15);

    *auxaccum += norm * value;

}



template<>

inline void MixAccum<float, int32_t>(float *auxaccum, int32_t value) {

    static const float norm = 1. / (1 << 27);

    static constexpr float norm = 1. / (1 << 27);

    *auxaccum += norm * value;

}
@@ -238,6 +238,7 @@ enum { 
 *   NCHAN represents number of input and output channels.

 *   TO: int32_t (Q4.27) or float

 *   TI: int32_t (Q4.27) or int16_t (Q0.15) or float

 *   TA: int32_t (Q4.27) or float

 *   TV: int32_t (U4.28) or int16_t (U4.12) or float

 *   vol: represents a volume array.

 *
@@ -247,7 +248,8 @@ enum { 
 *   Single input channel. NCHAN represents number of output channels.

 *   TO: int32_t (Q4.27) or float

 *   TI: int32_t (Q4.27) or int16_t (Q0.15) or float

 *   TV: int32_t (U4.28) or int16_t (U4.12) or float

 *   TA: int32_t (Q4.27) or float

 *   TV/TAV: int32_t (U4.28) or int16_t (U4.12) or float

 *   Input channel count is 1.

 *   vol: represents volume array.

 *
@@ -257,7 +259,8 @@ enum { 
 *   NCHAN represents number of input and output channels.

 *   TO: int16_t (Q.15) or float

 *   TI: int32_t (Q4.27) or int16_t (Q0.15) or float

 *   TV: int32_t (U4.28) or int16_t (U4.12) or float

 *   TA: int32_t (Q4.27) or float

 *   TV/TAV: int32_t (U4.28) or int16_t (U4.12) or float

 *   vol: represents a volume array.

 *

 *   MIXTYPE_MULTI_SAVEONLY does not accumulate into the out pointer.

services/audioflinger/Configuration.h

+5 −0
Original line number Diff line number Diff line
@@ -44,4 +44,9 @@ 
// define FLOAT_EFFECT_CHAIN to request float effects (falls back to int16_t if unavailable)

#define FLOAT_EFFECT_CHAIN



#ifdef FLOAT_EFFECT_CHAIN

// define FLOAT_AUX to process aux effect buffers in float (FLOAT_EFFECT_CHAIN must be defined)

#define FLOAT_AUX

#endif



#endif // ANDROID_AUDIOFLINGER_CONFIGURATION_H

services/audioflinger/Effects.cpp

+15 −3
Original line number Diff line number Diff line
@@ -338,9 +338,9 @@ void AudioFlinger::EffectModule::process() 
        if (isProcessImplemented()) {

            if (auxType) {

                // We overwrite the aux input buffer here and clear after processing.

                // Note that aux input buffers are format q4_27.

#ifdef FLOAT_EFFECT_CHAIN

                if (mSupportsFloat) {

#ifndef FLOAT_AUX

                    // Do in-place float conversion for auxiliary effect input buffer.

                    static_assert(sizeof(float) <= sizeof(int32_t),

                            "in-place conversion requires sizeof(float) <= sizeof(int32_t)");
@@ -349,13 +349,21 @@ void AudioFlinger::EffectModule::process() 
                            mConfig.inputCfg.buffer.f32,

                            mConfig.inputCfg.buffer.s32,

                            mConfig.inputCfg.buffer.frameCount);

#endif // !FLOAT_AUX

                } else

#endif

#endif // FLOAT_EFFECT_CHAIN

                {

#ifdef FLOAT_AUX

                    memcpy_to_i16_from_float(

                            mConfig.inputCfg.buffer.s16,

                            mConfig.inputCfg.buffer.f32,

                            mConfig.inputCfg.buffer.frameCount);

#else

                    memcpy_to_i16_from_q4_27(

                            mConfig.inputCfg.buffer.s16,

                            mConfig.inputCfg.buffer.s32,

                            mConfig.inputCfg.buffer.frameCount);

#endif

                }

            }

#ifdef FLOAT_EFFECT_CHAIN
@@ -415,9 +423,13 @@ void AudioFlinger::EffectModule::process() 


        // clear auxiliary effect input buffer for next accumulation

        if (auxType) {

            // input always q4_27 regardless of FLOAT_EFFECT_CHAIN.

#ifdef FLOAT_AUX

            const size_t size =

                    mConfig.inputCfg.buffer.frameCount * inChannelCount * sizeof(float);

#else

            const size_t size =

                    mConfig.inputCfg.buffer.frameCount * inChannelCount * sizeof(int32_t);

#endif

            memset(mConfig.inputCfg.buffer.raw, 0, size);

        }

    } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT &&