Merge "Remove the no longer used PV software aac decoder." into jb-mr1-dev (a9e0ea58) · Commits · e / os / android_frameworks_av

media/libstagefright/codecs/aacdec/Android.mk

+19 −204

Original line number	Diff line number	Diff line
		LOCAL_PATH:= $(call my-dir)

		AAC_LIBRARY = fraunhofer

		ifeq ($(AAC_LIBRARY), fraunhofer)
		include $(CLEAR_VARS)

		LOCAL_SRC_FILES := \
		@@ -29,185 +26,3 @@ ifeq ($(AAC_LIBRARY), fraunhofer)
		LOCAL_MODULE_TAGS := optional

		include $(BUILD_SHARED_LIBRARY)

		else # pv

		LOCAL_SRC_FILES := \
		analysis_sub_band.cpp \
		apply_ms_synt.cpp \
		apply_tns.cpp \
		buf_getbits.cpp \
		byte_align.cpp \
		calc_auto_corr.cpp \
		calc_gsfb_table.cpp \
		calc_sbr_anafilterbank.cpp \
		calc_sbr_envelope.cpp \
		calc_sbr_synfilterbank.cpp \
		check_crc.cpp \
		dct16.cpp \
		dct64.cpp \
		decode_huff_cw_binary.cpp \
		decode_noise_floorlevels.cpp \
		deinterleave.cpp \
		digit_reversal_tables.cpp \
		dst16.cpp \
		dst32.cpp \
		dst8.cpp \
		esc_iquant_scaling.cpp \
		extractframeinfo.cpp \
		fft_rx4_long.cpp \
		fft_rx4_short.cpp \
		fft_rx4_tables_fxp.cpp \
		find_adts_syncword.cpp \
		fwd_long_complex_rot.cpp \
		fwd_short_complex_rot.cpp \
		gen_rand_vector.cpp \
		get_adif_header.cpp \
		get_adts_header.cpp \
		get_audio_specific_config.cpp \
		get_dse.cpp \
		get_ele_list.cpp \
		get_ga_specific_config.cpp \
		get_ics_info.cpp \
		get_prog_config.cpp \
		get_pulse_data.cpp \
		get_sbr_bitstream.cpp \
		get_sbr_startfreq.cpp \
		get_sbr_stopfreq.cpp \
		get_tns.cpp \
		getfill.cpp \
		getgroup.cpp \
		getics.cpp \
		getmask.cpp \
		hcbtables_binary.cpp \
		huffcb.cpp \
		huffdecode.cpp \
		hufffac.cpp \
		huffspec_fxp.cpp \
		idct16.cpp \
		idct32.cpp \
		idct8.cpp \
		imdct_fxp.cpp \
		infoinit.cpp \
		init_sbr_dec.cpp \
		intensity_right.cpp \
		inv_long_complex_rot.cpp \
		inv_short_complex_rot.cpp \
		iquant_table.cpp \
		long_term_prediction.cpp \
		long_term_synthesis.cpp \
		lt_decode.cpp \
		mdct_fxp.cpp \
		mdct_tables_fxp.cpp \
		mdst.cpp \
		mix_radix_fft.cpp \
		ms_synt.cpp \
		pns_corr.cpp \
		pns_intensity_right.cpp \
		pns_left.cpp \
		ps_all_pass_filter_coeff.cpp \
		ps_all_pass_fract_delay_filter.cpp \
		ps_allocate_decoder.cpp \
		ps_applied.cpp \
		ps_bstr_decoding.cpp \
		ps_channel_filtering.cpp \
		ps_decode_bs_utils.cpp \
		ps_decorrelate.cpp \
		ps_fft_rx8.cpp \
		ps_hybrid_analysis.cpp \
		ps_hybrid_filter_bank_allocation.cpp \
		ps_hybrid_synthesis.cpp \
		ps_init_stereo_mixing.cpp \
		ps_pwr_transient_detection.cpp \
		ps_read_data.cpp \
		ps_stereo_processing.cpp \
		pulse_nc.cpp \
		pv_div.cpp \
		pv_log2.cpp \
		pv_normalize.cpp \
		pv_pow2.cpp \
		pv_sine.cpp \
		pv_sqrt.cpp \
		pvmp4audiodecoderconfig.cpp \
		pvmp4audiodecoderframe.cpp \
		pvmp4audiodecodergetmemrequirements.cpp \
		pvmp4audiodecoderinitlibrary.cpp \
		pvmp4audiodecoderresetbuffer.cpp \
		q_normalize.cpp \
		qmf_filterbank_coeff.cpp \
		sbr_aliasing_reduction.cpp \
		sbr_applied.cpp \
		sbr_code_book_envlevel.cpp \
		sbr_crc_check.cpp \
		sbr_create_limiter_bands.cpp \
		sbr_dec.cpp \
		sbr_decode_envelope.cpp \
		sbr_decode_huff_cw.cpp \
		sbr_downsample_lo_res.cpp \
		sbr_envelope_calc_tbl.cpp \
		sbr_envelope_unmapping.cpp \
		sbr_extract_extended_data.cpp \
		sbr_find_start_andstop_band.cpp \
		sbr_generate_high_freq.cpp \
		sbr_get_additional_data.cpp \
		sbr_get_cpe.cpp \
		sbr_get_dir_control_data.cpp \
		sbr_get_envelope.cpp \
		sbr_get_header_data.cpp \
		sbr_get_noise_floor_data.cpp \
		sbr_get_sce.cpp \
		sbr_inv_filt_levelemphasis.cpp \
		sbr_open.cpp \
		sbr_read_data.cpp \
		sbr_requantize_envelope_data.cpp \
		sbr_reset_dec.cpp \
		sbr_update_freq_scale.cpp \
		set_mc_info.cpp \
		sfb.cpp \
		shellsort.cpp \
		synthesis_sub_band.cpp \
		tns_ar_filter.cpp \
		tns_decode_coef.cpp \
		tns_inv_filter.cpp \
		trans4m_freq_2_time_fxp.cpp \
		trans4m_time_2_freq_fxp.cpp \
		unpack_idx.cpp \
		window_tables_fxp.cpp \
		pvmp4setaudioconfig.cpp \

		LOCAL_CFLAGS := -DAAC_PLUS -DHQ_SBR -DPARAMETRICSTEREO -DOSCL_IMPORT_REF= -DOSCL_EXPORT_REF= -DOSCL_UNUSED_ARG=

		LOCAL_C_INCLUDES := \
		frameworks/av/media/libstagefright/include \

		LOCAL_ARM_MODE := arm

		LOCAL_MODULE := libstagefright_aacdec

		include $(BUILD_STATIC_LIBRARY)

		################################################################################

		include $(CLEAR_VARS)

		LOCAL_SRC_FILES := \
		SoftAAC.cpp

		LOCAL_C_INCLUDES := \
		frameworks/av/media/libstagefright/include \
		frameworks/native/include/media/openmax

		LOCAL_CFLAGS := -DOSCL_IMPORT_REF=

		LOCAL_STATIC_LIBRARIES := \
		libstagefright_aacdec

		LOCAL_SHARED_LIBRARIES := \
		libstagefright_omx libstagefright_foundation libutils

		LOCAL_MODULE := libstagefright_soft_aacdec
		LOCAL_MODULE_TAGS := optional

		include $(BUILD_SHARED_LIBRARY)

		endif # $(AAC_LIBRARY)

media/libstagefright/codecs/aacdec/SoftAAC.cpp

deleted100644 → 0

+0 −553

Original line number	Diff line number	Diff line
		/*
		* Copyright (C) 2011 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.
		*/

		//#define LOG_NDEBUG 0
		#define LOG_TAG "SoftAAC"
		#include <utils/Log.h>

		#include "SoftAAC.h"

		#include "pvmp4audiodecoder_api.h"

		#include <media/stagefright/foundation/ADebug.h>
		#include <media/stagefright/foundation/hexdump.h>
		#include <media/stagefright/MediaErrors.h>

		namespace android {

		template<class T>
		static void InitOMXParams(T *params) {
		params->nSize = sizeof(T);
		params->nVersion.s.nVersionMajor = 1;
		params->nVersion.s.nVersionMinor = 0;
		params->nVersion.s.nRevision = 0;
		params->nVersion.s.nStep = 0;
		}

		SoftAAC::SoftAAC(
		const char *name,
		const OMX_CALLBACKTYPE *callbacks,
		OMX_PTR appData,
		OMX_COMPONENTTYPE **component)
		: SimpleSoftOMXComponent(name, callbacks, appData, component),
		mConfig(new tPVMP4AudioDecoderExternal),
		mIsADTS(false),
		mDecoderBuf(NULL),
		mInputBufferCount(0),
		mUpsamplingFactor(2),
		mAnchorTimeUs(0),
		mNumSamplesOutput(0),
		mSignalledError(false),
		mOutputPortSettingsChange(NONE) {
		initPorts();
		CHECK_EQ(initDecoder(), (status_t)OK);
		}

		SoftAAC::~SoftAAC() {
		free(mDecoderBuf);
		mDecoderBuf = NULL;

		delete mConfig;
		mConfig = NULL;
		}

		void SoftAAC::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 = 8192;
		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);
		}

		status_t SoftAAC::initDecoder() {
		memset(mConfig, 0, sizeof(tPVMP4AudioDecoderExternal));
		mConfig->outputFormat = OUTPUTFORMAT_16PCM_INTERLEAVED;
		mConfig->aacPlusEnabled = 1;

		// The software decoder doesn't properly support mono output on
		// AACplus files. Always output stereo.
		mConfig->desiredChannels = 2;

		UInt32 memRequirements = PVMP4AudioDecoderGetMemRequirements();
		mDecoderBuf = malloc(memRequirements);

		Int err = PVMP4AudioDecoderInitLibrary(mConfig, mDecoderBuf);
		if (err != MP4AUDEC_SUCCESS) {
		ALOGE("Failed to initialize MP4 audio decoder");
		return UNKNOWN_ERROR;
		}

		return OK;
		}

		OMX_ERRORTYPE SoftAAC::internalGetParameter(
		OMX_INDEXTYPE index, OMX_PTR params) {
		switch (index) {
		case OMX_IndexParamAudioAac:
		{
		OMX_AUDIO_PARAM_AACPROFILETYPE *aacParams =
		(OMX_AUDIO_PARAM_AACPROFILETYPE *)params;

		if (aacParams->nPortIndex != 0) {
		return OMX_ErrorUndefined;
		}

		aacParams->nBitRate = 0;
		aacParams->nAudioBandWidth = 0;
		aacParams->nAACtools = 0;
		aacParams->nAACERtools = 0;
		aacParams->eAACProfile = OMX_AUDIO_AACObjectMain;

		aacParams->eAACStreamFormat =
		mIsADTS
		? OMX_AUDIO_AACStreamFormatMP4ADTS
		: OMX_AUDIO_AACStreamFormatMP4FF;

		aacParams->eChannelMode = OMX_AUDIO_ChannelModeStereo;

		if (!isConfigured()) {
		aacParams->nChannels = 1;
		aacParams->nSampleRate = 44100;
		aacParams->nFrameLength = 0;
		} else {
		aacParams->nChannels = mConfig->encodedChannels;
		aacParams->nSampleRate = mConfig->samplingRate;
		aacParams->nFrameLength = mConfig->frameLength;
		}

		return OMX_ErrorNone;
		}

		case OMX_IndexParamAudioPcm:
		{
		OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
		(OMX_AUDIO_PARAM_PCMMODETYPE *)params;

		if (pcmParams->nPortIndex != 1) {
		return OMX_ErrorUndefined;
		}

		pcmParams->eNumData = OMX_NumericalDataSigned;
		pcmParams->eEndian = OMX_EndianBig;
		pcmParams->bInterleaved = OMX_TRUE;
		pcmParams->nBitPerSample = 16;
		pcmParams->ePCMMode = OMX_AUDIO_PCMModeLinear;
		pcmParams->eChannelMapping[0] = OMX_AUDIO_ChannelLF;
		pcmParams->eChannelMapping[1] = OMX_AUDIO_ChannelRF;

		if (!isConfigured()) {
		pcmParams->nChannels = 1;
		pcmParams->nSamplingRate = 44100;
		} else {
		pcmParams->nChannels = mConfig->desiredChannels;
		pcmParams->nSamplingRate = mConfig->samplingRate;
		}

		return OMX_ErrorNone;
		}

		default:
		return SimpleSoftOMXComponent::internalGetParameter(index, params);
		}
		}

		OMX_ERRORTYPE SoftAAC::internalSetParameter(
		OMX_INDEXTYPE index, const OMX_PTR params) {
		switch (index) {
		case OMX_IndexParamStandardComponentRole:
		{
		const OMX_PARAM_COMPONENTROLETYPE *roleParams =
		(const OMX_PARAM_COMPONENTROLETYPE *)params;

		if (strncmp((const char *)roleParams->cRole,
		"audio_decoder.aac",
		OMX_MAX_STRINGNAME_SIZE - 1)) {
		return OMX_ErrorUndefined;
		}

		return OMX_ErrorNone;
		}

		case OMX_IndexParamAudioAac:
		{
		const OMX_AUDIO_PARAM_AACPROFILETYPE *aacParams =
		(const OMX_AUDIO_PARAM_AACPROFILETYPE *)params;

		if (aacParams->nPortIndex != 0) {
		return OMX_ErrorUndefined;
		}

		if (aacParams->eAACStreamFormat == OMX_AUDIO_AACStreamFormatMP4FF) {
		mIsADTS = false;
		} else if (aacParams->eAACStreamFormat
		== OMX_AUDIO_AACStreamFormatMP4ADTS) {
		mIsADTS = true;
		} else {
		return OMX_ErrorUndefined;
		}

		return OMX_ErrorNone;
		}

		case OMX_IndexParamAudioPcm:
		{
		const OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
		(OMX_AUDIO_PARAM_PCMMODETYPE *)params;

		if (pcmParams->nPortIndex != 1) {
		return OMX_ErrorUndefined;
		}

		return OMX_ErrorNone;
		}

		default:
		return SimpleSoftOMXComponent::internalSetParameter(index, params);
		}
		}

		bool SoftAAC::isConfigured() const {
		return mInputBufferCount > 0;
		}

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

		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;

		mConfig->pInputBuffer = header->pBuffer + header->nOffset;
		mConfig->inputBufferCurrentLength = header->nFilledLen;
		mConfig->inputBufferMaxLength = 0;

		Int err = PVMP4AudioDecoderConfig(mConfig, mDecoderBuf);
		if (err != MP4AUDEC_SUCCESS) {
		mSignalledError = true;
		notify(OMX_EventError, OMX_ErrorUndefined, err, NULL);
		return;
		}

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

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

		while (!inQueue.empty() && !outQueue.empty()) {
		BufferInfo inInfo = inQueue.begin();
		OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;

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

		if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
		inQueue.erase(inQueue.begin());
		inInfo->mOwnedByUs = false;
		notifyEmptyBufferDone(inHeader);

		outHeader->nFilledLen = 0;
		outHeader->nFlags = OMX_BUFFERFLAG_EOS;

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

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

		size_t adtsHeaderSize = 0;
		if (mIsADTS) {
		// 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 %ld 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 %ld 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);

		mConfig->pInputBuffer =
		(UChar *)adtsHeader + adtsHeaderSize;

		mConfig->inputBufferCurrentLength =
		aac_frame_length - adtsHeaderSize;

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

		if (signalError) {
		mSignalledError = true;

		notify(OMX_EventError, OMX_ErrorStreamCorrupt,
		ERROR_MALFORMED, NULL);

		return;
		}
		} else {
		mConfig->pInputBuffer = inHeader->pBuffer + inHeader->nOffset;
		mConfig->inputBufferCurrentLength = inHeader->nFilledLen;
		}

		mConfig->inputBufferMaxLength = 0;
		mConfig->inputBufferUsedLength = 0;
		mConfig->remainderBits = 0;

		mConfig->pOutputBuffer =
		reinterpret_cast<Int16 *>(outHeader->pBuffer + outHeader->nOffset);

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

		Int32 prevSamplingRate = mConfig->samplingRate;
		Int decoderErr = PVMP4AudioDecodeFrame(mConfig, mDecoderBuf);

		/*
		* 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 (decoderErr == MP4AUDEC_SUCCESS && mInputBufferCount <= 2) {
		ALOGV("audio/extended audio object type: %d + %d",
		mConfig->audioObjectType, mConfig->extendedAudioObjectType);
		ALOGV("aac+ upsampling factor: %d desired channels: %d",
		mConfig->aacPlusUpsamplingFactor, mConfig->desiredChannels);

		if (mInputBufferCount == 1) {
		mUpsamplingFactor = mConfig->aacPlusUpsamplingFactor;
		// Check on the sampling rate to see whether it is changed.
		if (mConfig->samplingRate != prevSamplingRate) {
		ALOGW("Sample rate was %d Hz, but now is %d Hz",
		prevSamplingRate, mConfig->samplingRate);

		// We'll hold onto the input buffer and will decode
		// it again once the output port has been reconfigured.

		// We're going to want to revisit this input buffer, but
		// may have already advanced the offset. Undo that if
		// necessary.
		inHeader->nOffset -= adtsHeaderSize;
		inHeader->nFilledLen += adtsHeaderSize;

		notify(OMX_EventPortSettingsChanged, 1, 0, NULL);
		mOutputPortSettingsChange = AWAITING_DISABLED;
		return;
		}
		} else { // mInputBufferCount == 2
		if (mConfig->extendedAudioObjectType == MP4AUDIO_AAC_LC \|\|
		mConfig->extendedAudioObjectType == MP4AUDIO_LTP) {
		if (mUpsamplingFactor == 2) {
		// The stream turns out to be not aacPlus mode anyway
		ALOGW("Disable AAC+/eAAC+ since extended audio object "
		"type is %d",
		mConfig->extendedAudioObjectType);
		mConfig->aacPlusEnabled = 0;
		}
		} else {
		if (mUpsamplingFactor == 1) {
		// aacPlus mode does not buy us anything, but to cause
		// 1. CPU load to increase, and
		// 2. a half speed of decoding
		ALOGW("Disable AAC+/eAAC+ since upsampling factor is 1");
		mConfig->aacPlusEnabled = 0;
		}
		}
		}
		}

		size_t numOutBytes =
		mConfig->frameLength * sizeof(int16_t) * mConfig->desiredChannels;

		if (decoderErr == MP4AUDEC_SUCCESS) {
		CHECK_LE(mConfig->inputBufferUsedLength, inHeader->nFilledLen);

		inHeader->nFilledLen -= mConfig->inputBufferUsedLength;
		inHeader->nOffset += mConfig->inputBufferUsedLength;
		} else {
		ALOGW("AAC decoder returned error %d, substituting silence",
		decoderErr);

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

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

		// fall through
		}

		if (decoderErr == MP4AUDEC_SUCCESS \|\| 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.

		if (mUpsamplingFactor == 2) {
		if (mConfig->desiredChannels == 1) {
		memcpy(&mConfig->pOutputBuffer[1024],
		&mConfig->pOutputBuffer[2048],
		numOutBytes * 2);
		}
		numOutBytes *= 2;
		}

		outHeader->nFilledLen = numOutBytes;
		outHeader->nFlags = 0;

		outHeader->nTimeStamp =
		mAnchorTimeUs
		+ (mNumSamplesOutput * 1000000ll) / mConfig->samplingRate;

		mNumSamplesOutput += mConfig->frameLength * mUpsamplingFactor;

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

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

		if (decoderErr == MP4AUDEC_SUCCESS) {
		++mInputBufferCount;
		}
		}
		}

		void SoftAAC::onPortFlushCompleted(OMX_U32 portIndex) {
		if (portIndex == 0) {
		// Make sure that the next buffer output does not still
		// depend on fragments from the last one decoded.
		PVMP4AudioDecoderResetBuffer(mDecoderBuf);
		}
		}

		void SoftAAC::onPortEnableCompleted(OMX_U32 portIndex, bool enabled) {
		if (portIndex != 1) {
		return;
		}

		switch (mOutputPortSettingsChange) {
		case NONE:
		break;

		case AWAITING_DISABLED:
		{
		CHECK(!enabled);
		mOutputPortSettingsChange = AWAITING_ENABLED;
		break;
		}

		default:
		{
		CHECK_EQ((int)mOutputPortSettingsChange, (int)AWAITING_ENABLED);
		CHECK(enabled);
		mOutputPortSettingsChange = NONE;
		break;
		}
		}
		}

		} // namespace android

		android::SoftOMXComponent *createSoftOMXComponent(
		const char name, const OMX_CALLBACKTYPE callbacks,
		OMX_PTR appData, OMX_COMPONENTTYPE **component) {
		return new android::SoftAAC(name, callbacks, appData, component);
		}

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