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Commit 15d5d7c1 authored by Yamit Mehta's avatar Yamit Mehta Committed by Linux Build Service Account
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audio: Audio resampler support for 192Khz playback 

Add support for QTI audio resampler

audio: Audio resampler support for 192Khz playback
Change-Id: Ia8f24a0874ebf6e16ef7bd1f2759a14f47149875

audio: Add audio mixing support for qti resampler
Change-Id: Ib657aa12b2a72323564148c302ff8891e1bb7433

AudioMixer: Extend use of QTI resampler for 44.1Khz sampling rate
Change-Id: I2a819dbc9f1e3e280cb4fa79328e331883a3e981

AudioMixer: fill 0s at right place when no more buffers available
Change-Id: I50504c5a02eb0c69abfc9b047792b0f6f85b9ce8

audioflinger: add channel count check to use QTI resampler
Change-Id: I8f76dd82b72a0dd8b77343e77e0d0545e1be2114

audio: QTI resampler supports for 32 bit input format
Change-Id: I7e1b8582cd6bb106ab0bd25f9bc1bd9e4092318b

Change-Id: Ia8f24a0874ebf6e16ef7bd1f2759a14f47149875
parent 97071e78

services/audioflinger/Android.mk

+21 −1
Original line number Diff line number Diff line
@@ -64,6 +64,14 @@ LOCAL_STATIC_LIBRARIES := \ 


LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)



#QTI Resampler

ifeq ($(call is-vendor-board-platform,QCOM), true)

ifeq ($(strip $(AUDIO_FEATURE_ENABLED_EXTN_RESAMPLER)), true)

LOCAL_CFLAGS += -DQTI_RESAMPLER

endif

endif

#QTI Resampler



LOCAL_MODULE:= libaudioflinger



LOCAL_SRC_FILES += \
@@ -130,7 +138,19 @@ LOCAL_C_INCLUDES := \ 
LOCAL_SHARED_LIBRARIES := \

    libcutils \

    libdl \

    liblog

    liblog \

    libaudioutils



#QTI Resampler

ifeq ($(call is-vendor-board-platform,QCOM), true)

ifeq ($(strip $(AUDIO_FEATURE_ENABLED_EXTN_RESAMPLER)), true)

LOCAL_SRC_FILES_$(TARGET_2ND_ARCH) += AudioResamplerQTI.cpp.arm

LOCAL_C_INCLUDES_$(TARGET_2ND_ARCH) += $(TARGET_OUT_HEADERS)/mm-audio/audio-src

LOCAL_SHARED_LIBRARIES_$(TARGET_2ND_ARCH) += libqct_resampler

LOCAL_CFLAGS_$(TARGET_2ND_ARCH) += -DQTI_RESAMPLER

endif

endif

#QTI Resampler



LOCAL_MODULE := libaudioresampler

services/audioflinger/AudioMixer.cpp

+14 −0
Original line number Diff line number Diff line
@@ -83,6 +83,9 @@ static const bool kUseFloat = true; 
// Set to default copy buffer size in frames for input processing.

static const size_t kCopyBufferFrameCount = 256;



#ifdef QTI_RESAMPLER

#define QTI_RESAMPLER_MAX_SAMPLERATE 192000

#endif

namespace android {



// ----------------------------------------------------------------------------
@@ -777,6 +780,14 @@ bool AudioMixer::track_t::setResampler(uint32_t trackSampleRate, uint32_t devSam 
                // but if none exists, it is the channel count (1 for mono).

                const int resamplerChannelCount = downmixerBufferProvider != NULL

                        ? mMixerChannelCount : channelCount;

#ifdef QTI_RESAMPLER

                if ((trackSampleRate <= QTI_RESAMPLER_MAX_SAMPLERATE) &&

                       (trackSampleRate > devSampleRate * 2) &&

                       ((devSampleRate == 48000)||(devSampleRate == 44100)) &&

                       (resamplerChannelCount <= 2)) {

                    quality = AudioResampler::QTI_QUALITY;

                }

#endif

                ALOGVV("Creating resampler:"

                        " format(%#x) channels(%d) devSampleRate(%u) quality(%d)\n",

                        mMixerInFormat, resamplerChannelCount, devSampleRate, quality);
@@ -1633,6 +1644,9 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state) 
                // Note: In case of later int16_t sink output,

                // conversion and clamping is done by memcpy_to_i16_from_float().

            } while (--outFrames);

            //assign fout to out, when no more frames are available, so that 0s

            //can be filled at the right place

            out = (int32_t *)fout;

            break;

        case AUDIO_FORMAT_PCM_16_BIT:

            if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN_INT || uint32_t(vr) > UNITY_GAIN_INT)) {

services/audioflinger/AudioResampler.cpp

+25 −0
Original line number Diff line number Diff line
@@ -28,6 +28,10 @@ 
#include "AudioResamplerCubic.h"

#include "AudioResamplerDyn.h"



#ifdef QTI_RESAMPLER

#include "AudioResamplerQTI.h"

#endif



#ifdef __arm__

    // bug 13102576

    //#define ASM_ARM_RESAMP1 // enable asm optimisation for ResamplerOrder1
@@ -91,6 +95,9 @@ bool AudioResampler::qualityIsSupported(src_quality quality) 
    case DYN_LOW_QUALITY:

    case DYN_MED_QUALITY:

    case DYN_HIGH_QUALITY:

#ifdef QTI_RESAMPLER

    case QTI_QUALITY:

#endif

        return true;

    default:

        return false;
@@ -111,7 +118,11 @@ void AudioResampler::init_routine() 
        if (*endptr == '\0') {

            defaultQuality = (src_quality) l;

            ALOGD("forcing AudioResampler quality to %d", defaultQuality);

#ifdef QTI_RESAMPLER

            if (defaultQuality < DEFAULT_QUALITY || defaultQuality > QTI_QUALITY) {

#else

            if (defaultQuality < DEFAULT_QUALITY || defaultQuality > DYN_HIGH_QUALITY) {

#endif

                defaultQuality = DEFAULT_QUALITY;

            }

        }
@@ -130,6 +141,9 @@ uint32_t AudioResampler::qualityMHz(src_quality quality) 
    case HIGH_QUALITY:

        return 20;

    case VERY_HIGH_QUALITY:

#ifdef QTI_RESAMPLER

    case QTI_QUALITY: //for QTI_QUALITY, currently assuming same as VHQ

#endif

        return 34;

    case DYN_LOW_QUALITY:

        return 4;
@@ -205,6 +219,11 @@ AudioResampler* AudioResampler::create(audio_format_t format, int inChannelCount 
        case DYN_HIGH_QUALITY:

            quality = DYN_MED_QUALITY;

            break;

#ifdef QTI_RESAMPLER

        case QTI_QUALITY:

            quality = DYN_HIGH_QUALITY;

            break;

#endif

        }

    }

    pthread_mutex_unlock(&mutex);
@@ -251,6 +270,12 @@ AudioResampler* AudioResampler::create(audio_format_t format, int inChannelCount 
            }

        }

        break;

#ifdef QTI_RESAMPLER

    case QTI_QUALITY:

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

        resampler = new AudioResamplerQTI(format, inChannelCount, sampleRate);

        break;

#endif

    }



    // initialize resampler

services/audioflinger/AudioResampler.h

+3 −0
Original line number Diff line number Diff line
@@ -47,6 +47,9 @@ public: 
        DYN_LOW_QUALITY=5,

        DYN_MED_QUALITY=6,

        DYN_HIGH_QUALITY=7,

#ifdef QTI_RESAMPLER

        QTI_QUALITY=8,

#endif

    };



    static const CONSTEXPR float UNITY_GAIN_FLOAT = 1.0f;

services/audioflinger/AudioResamplerQTI.cpp

0 → 100644
+173 −0
Original line number Diff line number Diff line 
/*

 * Copyright (C) 2014, The Linux Foundation. All rights reserved.

 * Not a Contribution.

 * Copyright (C) 2007 The Android Open Source Project

 *

 * Licensed under the Apache License, Version 2.0 (the "License");

 * you may not use this file except in compliance with the License.

 * You may obtain a copy of the License at

 *

 *      http://www.apache.org/licenses/LICENSE-2.0

 *

 * Unless required by applicable law or agreed to in writing, software

 * distributed under the License is distributed on an "AS IS" BASIS,

 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.

 * See the License for the specific language governing permissions and

 * limitations under the License.

 */



#include "AudioResamplerQTI.h"

#include "QCT_Resampler.h"

#include <sys/time.h>

#include <audio_utils/primitives.h>



namespace android {

AudioResamplerQTI::AudioResamplerQTI(int format,

        int inChannelCount, int32_t sampleRate)

    :AudioResampler(inChannelCount, sampleRate, QTI_QUALITY),

     mTmpBuf(0), mResamplerOutBuf(0), mFrameIndex(0),mOutFrameCount(0)

{

    stateSize = QCT_Resampler::MemAlloc(format, inChannelCount, sampleRate, sampleRate);

    mState = new int16_t[stateSize];

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

    mBuffer.frameCount = 0;

}



AudioResamplerQTI::~AudioResamplerQTI()

{

    if (mState) {

        delete [] mState;

    }

    if (mTmpBuf) {

        delete [] mTmpBuf;

    }

    if(mResamplerOutBuf) {

        delete [] mResamplerOutBuf;

    }

}



size_t AudioResamplerQTI::resample(int32_t* out, size_t outFrameCount,

            AudioBufferProvider* provider)

{

    int16_t vl = mVolume[0];

    int16_t vr = mVolume[1];

    int32_t *pBuf;



    int64_t tempL, tempR;

    size_t inFrameRequest;

    size_t inFrameCount = getNumInSample(outFrameCount);

    size_t index = 0;

    size_t frameIndex = mFrameIndex;

    size_t out_count = outFrameCount * 2;

    float *fout = reinterpret_cast<float *>(out);



    if (mChannelCount == 1) {

        inFrameRequest = inFrameCount;

    } else {

        inFrameRequest = inFrameCount * 2;

    }



    if (mOutFrameCount < outFrameCount) {

        mOutFrameCount = outFrameCount;

        if (mTmpBuf) {

            delete [] mTmpBuf;

        }

        if(mResamplerOutBuf) {

            delete [] mResamplerOutBuf;

        }

        mTmpBuf = new int32_t[inFrameRequest + 16];

        mResamplerOutBuf = new int32_t[out_count];

    }



    if (mChannelCount == 1) {

        // buffer is empty, fetch a new one

        while (index < inFrameCount) {

            if (!mBuffer.frameCount) {

                mBuffer.frameCount = inFrameCount;

                provider->getNextBuffer(&mBuffer);

                frameIndex = 0;

            }



            if (mBuffer.raw == NULL) {

                while (index < inFrameCount) {

                    mTmpBuf[index++] = 0;

                }

                QCT_Resampler::Resample90dB(mState, mTmpBuf, mResamplerOutBuf, inFrameCount, outFrameCount);

                goto resample_exit;

            }



            mTmpBuf[index++] = clampq4_27_from_float(*((float *)mBuffer.raw + frameIndex++));



            if (frameIndex >= mBuffer.frameCount) {

                provider->releaseBuffer(&mBuffer);

            }

        }



        QCT_Resampler::Resample90dB(mState, mTmpBuf, mResamplerOutBuf, inFrameCount, outFrameCount);

    } else {

        pBuf = &mTmpBuf[inFrameCount];

        // buffer is empty, fetch a new one

        while (index < inFrameCount) {

            if (!mBuffer.frameCount) {

                mBuffer.frameCount = inFrameCount;

                provider->getNextBuffer(&mBuffer);

                frameIndex = 0;

            }

            if (mBuffer.raw == NULL) {

                while (index < inFrameCount) {

                    mTmpBuf[index] = 0;

                    pBuf[index++] = 0;

                }

                QCT_Resampler::Resample90dB(mState, mTmpBuf, mResamplerOutBuf, inFrameCount, outFrameCount);

                goto resample_exit;

            }



            mTmpBuf[index] = clampq4_27_from_float(*((float *)mBuffer.raw + frameIndex++));

            pBuf[index++] = clampq4_27_from_float(*((float *)mBuffer.raw + frameIndex++));

            if (frameIndex >= mBuffer.frameCount * 2) {

                provider->releaseBuffer(&mBuffer);

            }

       }



       QCT_Resampler::Resample90dB(mState, mTmpBuf, mResamplerOutBuf, inFrameCount, outFrameCount);

    }



resample_exit:

    for (uint32_t i = 0; i < out_count; i += 2) {

        // Multiplying q4.27 data with u4.12 gain could result in 39 fractional bit data(27+12)

        // To get back the 27 fractional bit format output data, do right shift by 12

        tempL = (int64_t)mResamplerOutBuf[i] * vl;

        tempR = (int64_t)mResamplerOutBuf[i+1] * vr;

        fout[i] += float_from_q4_27((int32_t)(tempL>>12));

        fout[i+1] += float_from_q4_27((int32_t)(tempR>>12));

    }



    mFrameIndex = frameIndex;

    return index;

}



void AudioResamplerQTI::setSampleRate(int32_t inSampleRate)

{

    if (mInSampleRate != inSampleRate) {

        mInSampleRate = inSampleRate;

        init();

    }

}



void AudioResamplerQTI::init()

{

    QCT_Resampler::Init(mState, mChannelCount, mInSampleRate, mSampleRate, 1/*32bit in*/);

}



size_t AudioResamplerQTI::getNumInSample(size_t outFrameCount)

{

    size_t size = (size_t)QCT_Resampler::GetNumInSamp(mState, outFrameCount);

    return size;

}



void AudioResamplerQTI::reset()

{

    AudioResampler::reset();

}



}; // namespace android