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Commit f2816fd4 authored by Andy Hung's avatar Andy Hung Committed by Android (Google) Code Review
Browse files

Merge changes Ib67fc35a,If2cdda0f 

* changes:
  AudioFlinger: Remove conditional effect types
  AudioFlinger: Clean up Effect configuration macros
parents 4436976d 181d96f1

services/audioflinger/AudioFlinger.cpp

+0 −8
Original line number Diff line number Diff line
@@ -3002,14 +3002,6 @@ sp<AudioFlinger::ThreadBase> AudioFlinger::openOutput_l(audio_module_handle_t mo 
        return nullptr;

    }



#ifndef MULTICHANNEL_EFFECT_CHAIN

    if (flags & AUDIO_OUTPUT_FLAG_SPATIALIZER) {

        ALOGE("openOutput_l() cannot create spatializer thread "

                "without #define MULTICHANNEL_EFFECT_CHAIN");

        return nullptr;

    }

#endif



    mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;



    // FOR TESTING ONLY:

services/audioflinger/AudioFlinger.h

+0 −7
Original line number Diff line number Diff line
@@ -608,13 +608,6 @@ private: 
    };



    // --- PlaybackThread ---

#ifdef FLOAT_EFFECT_CHAIN

#define EFFECT_BUFFER_FORMAT AUDIO_FORMAT_PCM_FLOAT

using effect_buffer_t = float;

#else

#define EFFECT_BUFFER_FORMAT AUDIO_FORMAT_PCM_16_BIT

using effect_buffer_t = int16_t;

#endif



#include "Threads.h"

services/audioflinger/Configuration.h

+0 −11
Original line number Diff line number Diff line
@@ -41,15 +41,4 @@ 
// uncomment to log CPU statistics every n wall clock seconds

//#define DEBUG_CPU_USAGE 10



// 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



// define MULTICHANNEL_EFFECT_CHAIN to allow multichannel effects (FLOAT_EFFECT_CHAIN defined)

#define MULTICHANNEL_EFFECT_CHAIN

#endif



#endif // ANDROID_AUDIOFLINGER_CONFIGURATION_H

services/audioflinger/Effects.cpp

+6 −86
Original line number Diff line number Diff line
@@ -573,9 +573,7 @@ AudioFlinger::EffectModule::EffectModule(const sp<AudioFlinger::EffectCallbackIn 
      mDisableWaitCnt(0),    // set by process() and updateState()

      mOffloaded(false),

      mIsOutput(false)

#ifdef FLOAT_EFFECT_CHAIN

      , mSupportsFloat(false)

#endif

{

    ALOGV("Constructor %p pinned %d", this, pinned);

    int lStatus;
@@ -694,31 +692,16 @@ void AudioFlinger::EffectModule::process() 
                            mConfig.inputCfg.buffer.frameCount,

                            mConfig.outputCfg.buffer.frameCount);

    const auto accumulateInputToOutput = [this, safeInputOutputSampleCount]() {

#ifdef FLOAT_EFFECT_CHAIN

        accumulate_float(

                mConfig.outputCfg.buffer.f32,

                mConfig.inputCfg.buffer.f32,

                safeInputOutputSampleCount);

#else

        accumulate_i16(

                mConfig.outputCfg.buffer.s16,

                mConfig.inputCfg.buffer.s16,

                safeInputOutputSampleCount);

#endif

    };

    const auto copyInputToOutput = [this, safeInputOutputSampleCount]() {

#ifdef FLOAT_EFFECT_CHAIN

        memcpy(

                mConfig.outputCfg.buffer.f32,

                mConfig.inputCfg.buffer.f32,

                safeInputOutputSampleCount * sizeof(*mConfig.outputCfg.buffer.f32));



#else

        memcpy(

                mConfig.outputCfg.buffer.s16,

                mConfig.inputCfg.buffer.s16,

                safeInputOutputSampleCount * sizeof(*mConfig.outputCfg.buffer.s16));

#endif

    };



    if (isProcessEnabled()) {
@@ -727,35 +710,14 @@ void AudioFlinger::EffectModule::process() 
            if (auxType) {

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

                // aux input is always mono.

#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)");



                    memcpy_to_float_from_q4_27(

                            mConfig.inputCfg.buffer.f32,

                            mConfig.inputCfg.buffer.s32,

                            mConfig.inputCfg.buffer.frameCount);

#endif // !FLOAT_AUX

                } else

#endif // FLOAT_EFFECT_CHAIN

                {

#ifdef FLOAT_AUX



                if (!mSupportsFloat) {

                    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

            sp<EffectBufferHalInterface> inBuffer = mInBuffer;

            sp<EffectBufferHalInterface> outBuffer = mOutBuffer;
@@ -802,9 +764,7 @@ void AudioFlinger::EffectModule::process() 
                    outBuffer = mOutConversionBuffer;

                }

            }

#endif

            ret = mEffectInterface->process();

#ifdef FLOAT_EFFECT_CHAIN

            if (!mSupportsFloat) { // convert output int16_t back to float.

                sp<EffectBufferHalInterface> target =

                        mOutChannelCountRequested != outChannelCount
@@ -821,11 +781,8 @@ void AudioFlinger::EffectModule::process() 
                        sizeof(float),

                        sizeof(float) * outChannelCount * mConfig.outputCfg.buffer.frameCount);

            }

#endif

        } else {

#ifdef FLOAT_EFFECT_CHAIN

            data_bypass:

#endif

            if (!auxType  /* aux effects do not require data bypass */

                    && mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {

                if (mConfig.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
@@ -844,13 +801,8 @@ void AudioFlinger::EffectModule::process() 


        // clear auxiliary effect input buffer for next accumulation

        if (auxType) {

#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 &&
@@ -905,23 +857,6 @@ status_t AudioFlinger::EffectModule::configure() 
            ALOGV("Overriding auxiliary effect input channels %#x as MONO",

                    mConfig.inputCfg.channels);

        }

#ifndef MULTICHANNEL_EFFECT_CHAIN

        if (mConfig.outputCfg.channels != AUDIO_CHANNEL_OUT_STEREO) {

            mConfig.outputCfg.channels = AUDIO_CHANNEL_OUT_STEREO;

            ALOGV("Overriding auxiliary effect output channels %#x as STEREO",

                    mConfig.outputCfg.channels);

        }

#endif

    } else {

#ifndef MULTICHANNEL_EFFECT_CHAIN

        // TODO: Update this logic when multichannel effects are implemented.

        // For offloaded tracks consider mono output as stereo for proper effect initialization

        if (channelMask == AUDIO_CHANNEL_OUT_MONO) {

            mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_STEREO;

            mConfig.outputCfg.channels = AUDIO_CHANNEL_OUT_STEREO;

            ALOGV("Overriding effect input and output as STEREO");

        }

#endif

    }

    if (isHapticGenerator()) {

        audio_channel_mask_t hapticChannelMask = callback->hapticChannelMask();
@@ -933,8 +868,8 @@ status_t AudioFlinger::EffectModule::configure() 
    mOutChannelCountRequested =

            audio_channel_count_from_out_mask(mConfig.outputCfg.channels);



    mConfig.inputCfg.format = EFFECT_BUFFER_FORMAT;

    mConfig.outputCfg.format = EFFECT_BUFFER_FORMAT;

    mConfig.inputCfg.format = AUDIO_FORMAT_PCM_FLOAT;

    mConfig.outputCfg.format = AUDIO_FORMAT_PCM_FLOAT;



    // Don't use sample rate for thread if effect isn't offloadable.

    if (callback->isOffloadOrDirect() && !isOffloaded()) {
@@ -982,7 +917,6 @@ status_t AudioFlinger::EffectModule::configure() 
        status = cmdStatus;

    }



#ifdef MULTICHANNEL_EFFECT_CHAIN

    if (status != NO_ERROR &&

            EffectConfiguration::isHidl() && // only HIDL effects support channel conversion

            mIsOutput &&
@@ -1008,9 +942,7 @@ status_t AudioFlinger::EffectModule::configure() 
            status = cmdStatus;

        }

    }

#endif



#ifdef FLOAT_EFFECT_CHAIN

    if (status == NO_ERROR) {

        mSupportsFloat = true;

    }
@@ -1036,7 +968,6 @@ status_t AudioFlinger::EffectModule::configure() 
            ALOGE("%s failed %d with int16_t (as well as float)", __func__, status);

        }

    }

#endif



    if (status == NO_ERROR) {

        // Establish Buffer strategy
@@ -1349,7 +1280,6 @@ void AudioFlinger::EffectModule::setInBuffer(const sp<EffectBufferHalInterface>& 
    mInBuffer = buffer;

    mEffectInterface->setInBuffer(buffer);



#ifdef FLOAT_EFFECT_CHAIN

    // aux effects do in place conversion to float - we don't allocate mInConversionBuffer.

    // Theoretically insert effects can also do in-place conversions (destroying

    // the original buffer) when the output buffer is identical to the input buffer,
@@ -1381,7 +1311,6 @@ void AudioFlinger::EffectModule::setInBuffer(const sp<EffectBufferHalInterface>& 
            ALOGE("%s cannot create mInConversionBuffer", __func__);

        }

    }

#endif

}



void AudioFlinger::EffectModule::setOutBuffer(const sp<EffectBufferHalInterface>& buffer) {
@@ -1397,7 +1326,6 @@ void AudioFlinger::EffectModule::setOutBuffer(const sp<EffectBufferHalInterface> 
    mOutBuffer = buffer;

    mEffectInterface->setOutBuffer(buffer);



#ifdef FLOAT_EFFECT_CHAIN

    // Note: Any effect that does not accumulate does not need mOutConversionBuffer and

    // can do in-place conversion from int16_t to float.  We don't optimize here.

    const uint32_t outChannelCount =
@@ -1425,7 +1353,6 @@ void AudioFlinger::EffectModule::setOutBuffer(const sp<EffectBufferHalInterface> 
            ALOGE("%s cannot create mOutConversionBuffer", __func__);

        }

    }

#endif

}



status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller)
@@ -1721,15 +1648,12 @@ NO_THREAD_SAFETY_ANALYSIS // conditional try lock 
            mConfig.outputCfg.format,

            formatToString((audio_format_t)mConfig.outputCfg.format).c_str());



#ifdef FLOAT_EFFECT_CHAIN



    result.appendFormat("\t\t- HAL buffers:\n"

            "\t\t\tIn(%s) InConversion(%s) Out(%s) OutConversion(%s)\n",

            dumpInOutBuffer(true /* isInput */, mInBuffer).c_str(),

            dumpInOutBuffer(true /* isInput */, mInConversionBuffer).c_str(),

            dumpInOutBuffer(false /* isInput */, mOutBuffer).c_str(),

            dumpInOutBuffer(false /* isInput */, mOutConversionBuffer).c_str());

#endif



    write(fd, result.string(), result.length());
@@ -2262,7 +2186,7 @@ void AudioFlinger::EffectChain::clearInputBuffer_l() 
    if (mInBuffer == NULL) {

        return;

    }

    const size_t frameSize = audio_bytes_per_sample(EFFECT_BUFFER_FORMAT)

    const size_t frameSize = audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT)

            * mEffectCallback->inChannelCount(mEffects[0]->id());



    memset(mInBuffer->audioBuffer()->raw, 0, mEffectCallback->frameCount() * frameSize);
@@ -2363,13 +2287,9 @@ status_t AudioFlinger::EffectChain::addEffect_ll(const sp<EffectModule>& effect) 
        // calling the process in effect engine

        size_t numSamples = mEffectCallback->frameCount();

        sp<EffectBufferHalInterface> halBuffer;

#ifdef FLOAT_EFFECT_CHAIN



        status_t result = mEffectCallback->allocateHalBuffer(

                numSamples * sizeof(float), &halBuffer);

#else

        status_t result = mEffectCallback->allocateHalBuffer(

                numSamples * sizeof(int32_t), &halBuffer);

#endif

        if (result != OK) return result;



        effect->configure();

services/audioflinger/Effects.h

+4 −6
Original line number Diff line number Diff line
@@ -323,13 +323,11 @@ private: 
    audio_io_handle_t mCurrentHalStream = AUDIO_IO_HANDLE_NONE;

    bool     mIsOutput;             // direction of the AF thread



#ifdef FLOAT_EFFECT_CHAIN

    bool    mSupportsFloat;         // effect supports float processing

    sp<EffectBufferHalInterface> mInConversionBuffer;  // Buffers for HAL conversion if needed.

    sp<EffectBufferHalInterface> mOutConversionBuffer;

    uint32_t mInChannelCountRequested;

    uint32_t mOutChannelCountRequested;

#endif



    class AutoLockReentrant {

    public:
@@ -494,14 +492,14 @@ public: 
    void setInBuffer(const sp<EffectBufferHalInterface>& buffer) {

        mInBuffer = buffer;

    }

    effect_buffer_t *inBuffer() const {

        return mInBuffer != 0 ? reinterpret_cast<effect_buffer_t*>(mInBuffer->ptr()) : NULL;

    float *inBuffer() const {

        return mInBuffer != 0 ? reinterpret_cast<float*>(mInBuffer->ptr()) : NULL;

    }

    void setOutBuffer(const sp<EffectBufferHalInterface>& buffer) {

        mOutBuffer = buffer;

    }

    effect_buffer_t *outBuffer() const {

        return mOutBuffer != 0 ? reinterpret_cast<effect_buffer_t*>(mOutBuffer->ptr()) : NULL;

    float *outBuffer() const {

        return mOutBuffer != 0 ? reinterpret_cast<float*>(mOutBuffer->ptr()) : NULL;

    }



    void incTrackCnt() { android_atomic_inc(&mTrackCnt); }