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Commit 4b34662b authored by Andreas Huber's avatar Andreas Huber Committed by android-build SharedAccount
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Squashed commit of the following: 

commit 4abf16bb04dc9695fedf4007a84f903074312ccd
Author: Andreas Huber <andih@google.com>
Date:   Tue Jul 20 09:21:17 2010 -0700

    Support a single format change at the beginning of audio playback. This way the AAC+ decoder may change its output format from what is originally encoded in the audio stream and we'll still play it back correctly.

    Change-Id: Icc790122744745e9a88099788d4818ca1e265a82
    related-to-bug: 2826841

commit 09c74da63e6ad5cb5dafb70f62696d75d2978967
Author: James Dong <jdong@google.com>
Date:   Sun Jul 18 17:57:01 2010 -0700

    Fix MPEG4Extractor to extract sampling frequency correctly when SBR is enabled.

    Change-Id: I883c81dad3ea465e71cb5590e89d763671a90ff8

commit f672bf2a782dc7d5fb6325d611a7fe17045dfe9a
Author: James Dong <jdong@google.com>
Date:   Thu Jul 8 20:56:13 2010 -0700

    Enable the support for decoding audio with AAC+ and eAAC+ features

    bug - 282684

    Change-Id: I73c8377af3cc4edd3ee7cea86dc3b1c369fbd78b

Change-Id: I012f1179e933b6d1345d2368f357576c722485f7
parent eb2e0954

include/media/stagefright/AudioPlayer.h

+4 −0
Original line number Diff line number Diff line
@@ -86,6 +86,10 @@ private: 


    bool mStarted;



    bool mIsFirstBuffer;

    status_t mFirstBufferResult;

    MediaBuffer *mFirstBuffer;



    sp<MediaPlayerBase::AudioSink> mAudioSink;



    static void AudioCallback(int event, void *user, void *info);

media/libstagefright/AudioPlayer.cpp

+63 −3
Original line number Diff line number Diff line
@@ -23,6 +23,7 @@ 
#include <media/stagefright/AudioPlayer.h>

#include <media/stagefright/MediaDebug.h>

#include <media/stagefright/MediaDefs.h>

#include <media/stagefright/MediaErrors.h>

#include <media/stagefright/MediaSource.h>

#include <media/stagefright/MetaData.h>
@@ -41,6 +42,9 @@ AudioPlayer::AudioPlayer(const sp<MediaPlayerBase::AudioSink> &audioSink) 
      mReachedEOS(false),

      mFinalStatus(OK),

      mStarted(false),

      mIsFirstBuffer(false),

      mFirstBufferResult(OK),

      mFirstBuffer(NULL),

      mAudioSink(audioSink) {

}
@@ -68,6 +72,24 @@ status_t AudioPlayer::start(bool sourceAlreadyStarted) { 
        }

    }



    // We allow an optional INFO_FORMAT_CHANGED at the very beginning

    // of playback, if there is one, getFormat below will retrieve the

    // updated format, if there isn't, we'll stash away the valid buffer

    // of data to be used on the first audio callback.



    CHECK(mFirstBuffer == NULL);



    mFirstBufferResult = mSource->read(&mFirstBuffer);

    if (mFirstBufferResult == INFO_FORMAT_CHANGED) {

        LOGV("INFO_FORMAT_CHANGED!!!");



        CHECK(mFirstBuffer == NULL);

        mFirstBufferResult = OK;

        mIsFirstBuffer = false;

    } else {

        mIsFirstBuffer = true;

    }



    sp<MetaData> format = mSource->getFormat();

    const char *mime;

    bool success = format->findCString(kKeyMIMEType, &mime);
@@ -87,7 +109,14 @@ status_t AudioPlayer::start(bool sourceAlreadyStarted) { 
                DEFAULT_AUDIOSINK_BUFFERCOUNT,

                &AudioPlayer::AudioSinkCallback, this);

        if (err != OK) {

            if (mFirstBuffer != NULL) {

                mFirstBuffer->release();

                mFirstBuffer = NULL;

            }



            if (!sourceAlreadyStarted) {

                mSource->stop();

            }



            return err;

        }
@@ -108,7 +137,14 @@ status_t AudioPlayer::start(bool sourceAlreadyStarted) { 
            delete mAudioTrack;

            mAudioTrack = NULL;



            if (mFirstBuffer != NULL) {

                mFirstBuffer->release();

                mFirstBuffer = NULL;

            }



            if (!sourceAlreadyStarted) {

                mSource->stop();

            }



            return err;

        }
@@ -159,6 +195,12 @@ void AudioPlayer::stop() { 


    // Make sure to release any buffer we hold onto so that the

    // source is able to stop().



    if (mFirstBuffer != NULL) {

        mFirstBuffer->release();

        mFirstBuffer = NULL;

    }



    if (mInputBuffer != NULL) {

        LOGV("AudioPlayer releasing input buffer.");
@@ -243,6 +285,14 @@ size_t AudioPlayer::fillBuffer(void *data, size_t size) { 
            Mutex::Autolock autoLock(mLock);



            if (mSeeking) {

                if (mIsFirstBuffer) {

                    if (mFirstBuffer != NULL) {

                        mFirstBuffer->release();

                        mFirstBuffer = NULL;

                    }

                    mIsFirstBuffer = false;

                }



                options.setSeekTo(mSeekTimeUs);



                if (mInputBuffer != NULL) {
@@ -255,7 +305,17 @@ size_t AudioPlayer::fillBuffer(void *data, size_t size) { 
        }



        if (mInputBuffer == NULL) {

            status_t err = mSource->read(&mInputBuffer, &options);

            status_t err;



            if (mIsFirstBuffer) {

                mInputBuffer = mFirstBuffer;

                mFirstBuffer = NULL;

                err = mFirstBufferResult;



                mIsFirstBuffer = false;

            } else {

                err = mSource->read(&mInputBuffer, &options);

            }



            CHECK((err == OK && mInputBuffer != NULL)

                   || (err != OK && mInputBuffer == NULL));

media/libstagefright/codecs/aacdec/AACDecoder.cpp

+65 −45
Original line number Diff line number Diff line
@@ -15,6 +15,7 @@ 
 */



#include "AACDecoder.h"

#define LOG_TAG "AACDecoder"



#include "../../include/ESDS.h"
@@ -36,26 +37,33 @@ AACDecoder::AACDecoder(const sp<MediaSource> &source) 
      mAnchorTimeUs(0),

      mNumSamplesOutput(0),

      mInputBuffer(NULL) {

}



AACDecoder::~AACDecoder() {

    if (mStarted) {

        stop();

    }

    sp<MetaData> srcFormat = mSource->getFormat();



    delete mConfig;

    mConfig = NULL;

}

    int32_t sampleRate;

    CHECK(srcFormat->findInt32(kKeySampleRate, &sampleRate));



status_t AACDecoder::start(MetaData *params) {

    CHECK(!mStarted);

    mMeta = new MetaData;

    mMeta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_RAW);



    mBufferGroup = new MediaBufferGroup;

    mBufferGroup->add_buffer(new MediaBuffer(2048 * 2));

    // We'll always output stereo, regardless of how many channels are

    // present in the input due to decoder limitations.

    mMeta->setInt32(kKeyChannelCount, 2);

    mMeta->setInt32(kKeySampleRate, sampleRate);



    int64_t durationUs;

    if (srcFormat->findInt64(kKeyDuration, &durationUs)) {

        mMeta->setInt64(kKeyDuration, durationUs);

    }

    mMeta->setCString(kKeyDecoderComponent, "AACDecoder");



    mInitCheck = initCheck();

}



status_t AACDecoder::initCheck() {

    memset(mConfig, 0, sizeof(tPVMP4AudioDecoderExternal));

    mConfig->outputFormat = OUTPUTFORMAT_16PCM_INTERLEAVED;

    mConfig->aacPlusUpsamplingFactor = 0;

    mConfig->aacPlusEnabled = false;

    mConfig->aacPlusEnabled = 1;



    // The software decoder doesn't properly support mono output on

    // AACplus files. Always output stereo.
@@ -64,8 +72,11 @@ status_t AACDecoder::start(MetaData *params) { 
    UInt32 memRequirements = PVMP4AudioDecoderGetMemRequirements();

    mDecoderBuf = malloc(memRequirements);



    CHECK_EQ(PVMP4AudioDecoderInitLibrary(mConfig, mDecoderBuf),

             MP4AUDEC_SUCCESS);

    status_t err = PVMP4AudioDecoderInitLibrary(mConfig, mDecoderBuf);

    if (err != MP4AUDEC_SUCCESS) {

        LOGE("Failed to initialize MP4 audio decoder");

        return UNKNOWN_ERROR;

    }



    uint32_t type;

    const void *data;
@@ -83,18 +94,29 @@ status_t AACDecoder::start(MetaData *params) { 
        mConfig->pInputBuffer = (UChar *)codec_specific_data;

        mConfig->inputBufferCurrentLength = codec_specific_data_size;

        mConfig->inputBufferMaxLength = 0;

        mConfig->inputBufferUsedLength = 0;

        mConfig->remainderBits = 0;



        mConfig->pOutputBuffer = NULL;

        mConfig->pOutputBuffer_plus = NULL;

        mConfig->repositionFlag = false;



        if (PVMP4AudioDecoderConfig(mConfig, mDecoderBuf)

                != MP4AUDEC_SUCCESS) {

            return ERROR_UNSUPPORTED;

        }

    }

    return OK;

}



AACDecoder::~AACDecoder() {

    if (mStarted) {

        stop();

    }



    delete mConfig;

    mConfig = NULL;

}



status_t AACDecoder::start(MetaData *params) {

    CHECK(!mStarted);



    mBufferGroup = new MediaBufferGroup;

    mBufferGroup->add_buffer(new MediaBuffer(4096 * 2));



    mSource->start();
@@ -127,28 +149,7 @@ status_t AACDecoder::stop() { 
}



sp<MetaData> AACDecoder::getFormat() {

    sp<MetaData> srcFormat = mSource->getFormat();



    int32_t sampleRate;

    CHECK(srcFormat->findInt32(kKeySampleRate, &sampleRate));



    sp<MetaData> meta = new MetaData;

    meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_RAW);



    // We'll always output stereo, regardless of how many channels are

    // present in the input due to decoder limitations.

    meta->setInt32(kKeyChannelCount, 2);



    meta->setInt32(kKeySampleRate, sampleRate);



    int64_t durationUs;

    if (srcFormat->findInt64(kKeyDuration, &durationUs)) {

        meta->setInt64(kKeyDuration, durationUs);

    }



    meta->setCString(kKeyDecoderComponent, "AACDecoder");



    return meta;

    return mMeta;

}



status_t AACDecoder::read(
@@ -200,13 +201,32 @@ status_t AACDecoder::read( 
    mConfig->remainderBits = 0;



    mConfig->pOutputBuffer = static_cast<Int16 *>(buffer->data());

    mConfig->pOutputBuffer_plus = NULL;

    mConfig->pOutputBuffer_plus = &mConfig->pOutputBuffer[2048];

    mConfig->repositionFlag = false;



    Int decoderErr = PVMP4AudioDecodeFrame(mConfig, mDecoderBuf);



    // Check on the sampling rate to see whether it is changed.

    int32_t sampleRate;

    CHECK(mMeta->findInt32(kKeySampleRate, &sampleRate));

    if (mConfig->samplingRate != sampleRate) {

        mMeta->setInt32(kKeySampleRate, mConfig->samplingRate);

        LOGW("Sample rate was %d, but now is %d",

                sampleRate, mConfig->samplingRate);

        buffer->release();

        mInputBuffer->release();

        mInputBuffer = NULL;

        return INFO_FORMAT_CHANGED;

    }



    size_t numOutBytes =

        mConfig->frameLength * sizeof(int16_t) * mConfig->desiredChannels;

    if (mConfig->aacPlusUpsamplingFactor == 2) {

        if (mConfig->desiredChannels == 1) {

            memcpy(&mConfig->pOutputBuffer[1024], &mConfig->pOutputBuffer[2048], numOutBytes * 2);

        }

        numOutBytes *= 2;

    }



    if (decoderErr != MP4AUDEC_SUCCESS) {

        LOGW("AAC decoder returned error %d, substituting silence", decoderErr);

media/libstagefright/include/AACDecoder.h

+4 −0
Original line number Diff line number Diff line
@@ -25,6 +25,7 @@ struct tPVMP4AudioDecoderExternal; 
namespace android {



struct MediaBufferGroup;

struct MetaData;



struct AACDecoder : public MediaSource {

    AACDecoder(const sp<MediaSource> &source);
@@ -41,6 +42,7 @@ protected: 
    virtual ~AACDecoder();



private:

    sp<MetaData>    mMeta;

    sp<MediaSource> mSource;

    bool mStarted;
@@ -50,9 +52,11 @@ private: 
    void *mDecoderBuf;

    int64_t mAnchorTimeUs;

    int64_t mNumSamplesOutput;

    status_t mInitCheck;



    MediaBuffer *mInputBuffer;



    status_t initCheck();

    AACDecoder(const AACDecoder &);

    AACDecoder &operator=(const AACDecoder &);

};