Audio笔记之SoftAAC2

SoftOMXComponent::SoftOMXComponent(
        const char *name,
        const OMX_CALLBACKTYPE *callbacks,
        OMX_PTR appData,
        OMX_COMPONENTTYPE **component)
    : mName(name),
      mCallbacks(callbacks),
      mComponent(new OMX_COMPONENTTYPE),
      mLibHandle(NULL) {
    mComponent->nSize = sizeof(*mComponent);
    mComponent->nVersion.s.nVersionMajor = 1;
    mComponent->nVersion.s.nVersionMinor = 0;
    mComponent->nVersion.s.nRevision = 0;
    mComponent->nVersion.s.nStep = 0;
    mComponent->pComponentPrivate = this;
    mComponent->pApplicationPrivate = appData;

    mComponent->GetComponentVersion = NULL;
    mComponent->SendCommand = SendCommandWrapper;
    mComponent->GetParameter = GetParameterWrapper;
    mComponent->SetParameter = SetParameterWrapper;
    mComponent->GetConfig = GetConfigWrapper;
    mComponent->SetConfig = SetConfigWrapper;
    mComponent->GetExtensionIndex = GetExtensionIndexWrapper;
    mComponent->GetState = GetStateWrapper;
    mComponent->ComponentTunnelRequest = NULL;
    mComponent->UseBuffer = UseBufferWrapper;
    mComponent->AllocateBuffer = AllocateBufferWrapper;
    mComponent->FreeBuffer = FreeBufferWrapper;
    mComponent->EmptyThisBuffer = EmptyThisBufferWrapper;
    mComponent->FillThisBuffer = FillThisBufferWrapper;
    mComponent->SetCallbacks = NULL;
    mComponent->ComponentDeInit = NULL;
    mComponent->UseEGLImage = NULL;
    mComponent->ComponentRoleEnum = NULL;

    *component = mComponent;
}
SimpleSoftOMXComponent::SimpleSoftOMXComponent(
        const char *name,
        const OMX_CALLBACKTYPE *callbacks,
        OMX_PTR appData,
        OMX_COMPONENTTYPE **component)
    : SoftOMXComponent(name, callbacks, appData, component),
      mLooper(new ALooper),
      mHandler(new AHandlerReflector<SimpleSoftOMXComponent>(this)),
      mState(OMX_StateLoaded),
      mTargetState(OMX_StateLoaded) {
    mLooper->setName(name);
    mLooper->registerHandler(mHandler);

    mLooper->start(
            false, // runOnCallingThread
            false, // canCallJava
            ANDROID_PRIORITY_FOREGROUND);
}
SoftAAC2::SoftAAC2(
        const char *name,
        const OMX_CALLBACKTYPE *callbacks,
        OMX_PTR appData,
        OMX_COMPONENTTYPE **component)
    : SimpleSoftOMXComponent(name, callbacks, appData, component),
      mAACDecoder(NULL),
      mStreamInfo(NULL),
      mIsADTS(false),
      mInputBufferCount(0),
      mSignalledError(false),
      mSawInputEos(false),
      mSignalledOutputEos(false),
      mAnchorTimeUs(0),
      mNumSamplesOutput(0),
      mOutputPortSettingsChange(NONE) {
    initPorts();
    CHECK_EQ(initDecoder(), (status_t)OK);
}
void SoftAAC2::initPorts() {
    OMX_PARAM_PORTDEFINITIONTYPE def;
    InitOMXParams(&def);

    def.nPortIndex = 0;
    def.eDir = OMX_DirInput;
    def.nBufferCountMin = kNumInputBuffers;
    def.nBufferCountActual = def.nBufferCountMin;
    def.nBufferSize = 8192;
    def.bEnabled = OMX_TRUE;
    def.bPopulated = OMX_FALSE;
    def.eDomain = OMX_PortDomainAudio;
    def.bBuffersContiguous = OMX_FALSE;
    def.nBufferAlignment = 1;

    def.format.audio.cMIMEType = const_cast<char *>("audio/aac");
    def.format.audio.pNativeRender = NULL;
    def.format.audio.bFlagErrorConcealment = OMX_FALSE;
    def.format.audio.eEncoding = OMX_AUDIO_CodingAAC;

    addPort(def);

    def.nPortIndex = 1;
    def.eDir = OMX_DirOutput;
    def.nBufferCountMin = kNumOutputBuffers;
    def.nBufferCountActual = def.nBufferCountMin;
    def.nBufferSize = 4096 * MAX_CHANNEL_COUNT;
    def.bEnabled = OMX_TRUE;
    def.bPopulated = OMX_FALSE;
    def.eDomain = OMX_PortDomainAudio;
    def.bBuffersContiguous = OMX_FALSE;
    def.nBufferAlignment = 2;

    def.format.audio.cMIMEType = const_cast<char *>("audio/raw");
    def.format.audio.pNativeRender = NULL;
    def.format.audio.bFlagErrorConcealment = OMX_FALSE;
    def.format.audio.eEncoding = OMX_AUDIO_CodingPCM;

    addPort(def);
}
//初始化解码器，
status_t SoftAAC2::initDecoder() {
    status_t status = UNKNOWN_ERROR;
    //获得解码库的句柄mAACDecoder
    mAACDecoder = aacDecoder_Open(TT_MP4_ADIF, /* num layers */ 1);
    if (mAACDecoder != NULL) {
        //获取aac的一些信息，保存在结构体CStreamInfo中
        mStreamInfo = aacDecoder_GetStreamInfo(mAACDecoder);
        if (mStreamInfo != NULL) {
            status = OK;
        }
    }
    mDecoderHasData = false;

    // for streams that contain metadata, use the mobile profile DRC settings unless overridden
    // by platform properties:
    char value[PROPERTY_VALUE_MAX];
    //  * AAC_DRC_REFERENCE_LEVEL
    if (property_get(PROP_DRC_OVERRIDE_REF_LEVEL, value, NULL)) {
        unsigned refLevel = atoi(value);
        ALOGV("AAC decoder using AAC_DRC_REFERENCE_LEVEL of %d instead of %d",
                refLevel, DRC_DEFAULT_MOBILE_REF_LEVEL);
        aacDecoder_SetParam(mAACDecoder, AAC_DRC_REFERENCE_LEVEL, refLevel);
    } else {
        aacDecoder_SetParam(mAACDecoder, AAC_DRC_REFERENCE_LEVEL, DRC_DEFAULT_MOBILE_REF_LEVEL);
    }
    //  * AAC_DRC_ATTENUATION_FACTOR
    if (property_get(PROP_DRC_OVERRIDE_CUT, value, NULL)) {
        unsigned cut = atoi(value);
        ALOGV("AAC decoder using AAC_DRC_ATTENUATION_FACTOR of %d instead of %d",
                        cut, DRC_DEFAULT_MOBILE_DRC_CUT);
        aacDecoder_SetParam(mAACDecoder, AAC_DRC_ATTENUATION_FACTOR, cut);
    } else {
        aacDecoder_SetParam(mAACDecoder, AAC_DRC_ATTENUATION_FACTOR, DRC_DEFAULT_MOBILE_DRC_CUT);
    }
    //  * AAC_DRC_BOOST_FACTOR (note: no default, using cut)
    if (property_get(PROP_DRC_OVERRIDE_BOOST, value, NULL)) {
        unsigned boost = atoi(value);
        ALOGV("AAC decoder using AAC_DRC_BOOST_FACTOR of %d", boost);
        aacDecoder_SetParam(mAACDecoder, AAC_DRC_BOOST_FACTOR, boost);
    } else {
        aacDecoder_SetParam(mAACDecoder, AAC_DRC_BOOST_FACTOR, DRC_DEFAULT_MOBILE_DRC_BOOST);
    }

    return status;
}
void SoftAAC2::onQueueFilled(OMX_U32 portIndex) {
    if (mSignalledError || mOutputPortSettingsChange != NONE) {
        return;
    }

    UCHAR* inBuffer[FILEREAD_MAX_LAYERS];
    UINT inBufferLength[FILEREAD_MAX_LAYERS] = {0};
    UINT bytesValid[FILEREAD_MAX_LAYERS] = {0};

    List<BufferInfo *> &inQueue = getPortQueue(0);
    List<BufferInfo *> &outQueue = getPortQueue(1);

    if (portIndex == 0 && mInputBufferCount == 0) {
        ++mInputBufferCount;
        BufferInfo *info = *inQueue.begin();
        OMX_BUFFERHEADERTYPE *header = info->mHeader;

        inBuffer[0] = header->pBuffer + header->nOffset;
        inBufferLength[0] = header->nFilledLen;

        AAC_DECODER_ERROR decoderErr =
            aacDecoder_ConfigRaw(mAACDecoder,
                                 inBuffer,
                                 inBufferLength);

        if (decoderErr != AAC_DEC_OK) {
            mSignalledError = true;
            notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, NULL);
            return;
        }

        inQueue.erase(inQueue.begin());
        info->mOwnedByUs = false;
        notifyEmptyBufferDone(header);

        // Only send out port settings changed event if both sample rate
        // and numChannels are valid.
        if (mStreamInfo->sampleRate && mStreamInfo->numChannels) {
            maybeConfigureDownmix();
            ALOGI("Initially configuring decoder: %d Hz, %d channels",
                mStreamInfo->sampleRate,
                mStreamInfo->numChannels);

            notify(OMX_EventPortSettingsChanged, 1, 0, NULL);
            mOutputPortSettingsChange = AWAITING_DISABLED;
        }

        return;
    }

    while ((!inQueue.empty() || (mSawInputEos && !mSignalledOutputEos)) && !outQueue.empty()) {
        BufferInfo *inInfo = NULL;
        OMX_BUFFERHEADERTYPE *inHeader = NULL;
        if (!inQueue.empty()) {
            inInfo = *inQueue.begin();
            inHeader = inInfo->mHeader;
        }

        BufferInfo *outInfo = *outQueue.begin();
        OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
        outHeader->nFlags = 0;

        if (inHeader) {
            if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
                mSawInputEos = true;
            }

            if (inHeader->nOffset == 0 && inHeader->nFilledLen) {
                mAnchorTimeUs = inHeader->nTimeStamp;
                mNumSamplesOutput = 0;
            }

            if (mIsADTS) {
                size_t adtsHeaderSize = 0;
                // skip 30 bits, aac_frame_length follows.
                // ssssssss ssssiiip ppffffPc ccohCCll llllllll lll?????

                const uint8_t *adtsHeader = inHeader->pBuffer + inHeader->nOffset;

                bool signalError = false;
                if (inHeader->nFilledLen < 7) {
                    ALOGE("Audio data too short to contain even the ADTS header. "
                          "Got %d bytes.", inHeader->nFilledLen);
                    hexdump(adtsHeader, inHeader->nFilledLen);
                    signalError = true;
                } else {
                    bool protectionAbsent = (adtsHeader[1] & 1);

                    unsigned aac_frame_length =
                        ((adtsHeader[3] & 3) << 11)
                        | (adtsHeader[4] << 3)
                        | (adtsHeader[5] >> 5);

                    if (inHeader->nFilledLen < aac_frame_length) {
                        ALOGE("Not enough audio data for the complete frame. "
                              "Got %d bytes, frame size according to the ADTS "
                              "header is %u bytes.",
                              inHeader->nFilledLen, aac_frame_length);
                        hexdump(adtsHeader, inHeader->nFilledLen);
                        signalError = true;
                    } else {
                        adtsHeaderSize = (protectionAbsent ? 7 : 9);

                        inBuffer[0] = (UCHAR *)adtsHeader + adtsHeaderSize;
                        inBufferLength[0] = aac_frame_length - adtsHeaderSize;

                        inHeader->nOffset += adtsHeaderSize;
                        inHeader->nFilledLen -= adtsHeaderSize;
                    }
                }

                if (signalError) {
                    mSignalledError = true;

                    notify(OMX_EventError,
                           OMX_ErrorStreamCorrupt,
                           ERROR_MALFORMED,
                           NULL);

                    return;
                }
            } else {
                inBuffer[0] = inHeader->pBuffer + inHeader->nOffset;
                inBufferLength[0] = inHeader->nFilledLen;
            }
        } else {
            inBufferLength[0] = 0;
        }

        // Fill and decode
        INT_PCM *outBuffer = reinterpret_cast<INT_PCM *>(
                outHeader->pBuffer + outHeader->nOffset);

        bytesValid[0] = inBufferLength[0];

        int prevSampleRate = mStreamInfo->sampleRate;
        int prevNumChannels = mStreamInfo->numChannels;

        AAC_DECODER_ERROR decoderErr = AAC_DEC_NOT_ENOUGH_BITS;
        while ((bytesValid[0] > 0 || mSawInputEos) && decoderErr == AAC_DEC_NOT_ENOUGH_BITS) {
            mDecoderHasData |= (bytesValid[0] > 0);
            aacDecoder_Fill(mAACDecoder,
                            inBuffer,
                            inBufferLength,
                            bytesValid);

            decoderErr = aacDecoder_DecodeFrame(mAACDecoder,
                                                outBuffer,
                                                outHeader->nAllocLen,
                                                0 /* flags */);
            if (decoderErr == AAC_DEC_NOT_ENOUGH_BITS) {
                if (mSawInputEos && bytesValid[0] <= 0) {
                    if (mDecoderHasData) {
                        // flush out the decoder's delayed data by calling DecodeFrame
                        // one more time, with the AACDEC_FLUSH flag set
                        decoderErr = aacDecoder_DecodeFrame(mAACDecoder,
                                                            outBuffer,
                                                            outHeader->nAllocLen,
                                                            AACDEC_FLUSH);
                        mDecoderHasData = false;
                    }
                    outHeader->nFlags = OMX_BUFFERFLAG_EOS;
                    mSignalledOutputEos = true;
                    break;
                } else {
                    ALOGW("Not enough bits, bytesValid %d", bytesValid[0]);
                }
            }
        }

        size_t numOutBytes =
            mStreamInfo->frameSize * sizeof(int16_t) * mStreamInfo->numChannels;

        if (inHeader) {
            if (decoderErr == AAC_DEC_OK) {
                UINT inBufferUsedLength = inBufferLength[0] - bytesValid[0];
                inHeader->nFilledLen -= inBufferUsedLength;
                inHeader->nOffset += inBufferUsedLength;
            } else {
                ALOGW("AAC decoder returned error %d, substituting silence",
                      decoderErr);

                memset(outHeader->pBuffer + outHeader->nOffset, 0, numOutBytes);

                // Discard input buffer.
                inHeader->nFilledLen = 0;

                aacDecoder_SetParam(mAACDecoder, AAC_TPDEC_CLEAR_BUFFER, 1);

                // fall through
            }

            if (inHeader->nFilledLen == 0) {
                inInfo->mOwnedByUs = false;
                inQueue.erase(inQueue.begin());
                inInfo = NULL;
                notifyEmptyBufferDone(inHeader);
                inHeader = NULL;
            }
        }

        /*
         * AAC+/eAAC+ streams can be signalled in two ways: either explicitly
         * or implicitly, according to MPEG4 spec. AAC+/eAAC+ is a dual
         * rate system and the sampling rate in the final output is actually
         * doubled compared with the core AAC decoder sampling rate.
         *
         * Explicit signalling is done by explicitly defining SBR audio object
         * type in the bitstream. Implicit signalling is done by embedding
         * SBR content in AAC extension payload specific to SBR, and hence
         * requires an AAC decoder to perform pre-checks on actual audio frames.
         *
         * Thus, we could not say for sure whether a stream is
         * AAC+/eAAC+ until the first data frame is decoded.
         */
        if (mInputBufferCount <= 2) {
            if (mStreamInfo->sampleRate != prevSampleRate ||
                mStreamInfo->numChannels != prevNumChannels) {
                maybeConfigureDownmix();
                ALOGI("Reconfiguring decoder: %d->%d Hz, %d->%d channels",
                      prevSampleRate, mStreamInfo->sampleRate,
                      prevNumChannels, mStreamInfo->numChannels);

                notify(OMX_EventPortSettingsChanged, 1, 0, NULL);
                mOutputPortSettingsChange = AWAITING_DISABLED;
                return;
            }
        } else if (!mStreamInfo->sampleRate || !mStreamInfo->numChannels) {
            ALOGW("Invalid AAC stream");
            mSignalledError = true;
            notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, NULL);
            return;
        }

        if (decoderErr == AAC_DEC_OK || mNumSamplesOutput > 0) {
            // We'll only output data if we successfully decoded it or
            // we've previously decoded valid data, in the latter case
            // (decode failed) we'll output a silent frame.
            outHeader->nFilledLen = numOutBytes;

            outHeader->nTimeStamp =
                mAnchorTimeUs
                    + (mNumSamplesOutput * 1000000ll) / mStreamInfo->sampleRate;

            mNumSamplesOutput += mStreamInfo->frameSize;

            outInfo->mOwnedByUs = false;
            outQueue.erase(outQueue.begin());
            outInfo = NULL;
            notifyFillBufferDone(outHeader);
            outHeader = NULL;
        }

        if (decoderErr == AAC_DEC_OK) {
            ++mInputBufferCount;
        }
    }
}