src/android/AacBqToPcmCbRenderer.cpp - platform/frameworks/wilhelm - Git at Google

 /*
  * 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 USE_LOG SLAndroidLogLevel_Debug

 #include "sles_allinclusive.h"
 #include "android/include/AacBqToPcmCbRenderer.h"
 #include "android/channels.h"
 #include <media/stagefright/SimpleDecodingSource.h>

 namespace android {

 // ADTS header size is 7, but frame size information ends on byte 6 (when counting from byte 1)
 #define ADTS_HEADER_SIZE_UP_TO_FRAMESIZE 6

 /**
  * Returns the size of an AAC ADTS frame.
  * Note that if the returned value + offset > size, it means that a partial frame starts at that
  *   offset, but this function will still return the size of the full frame.
  * @param data pointer to the compressed audio data
  * @param offset offset in bytes relative to data of where the frame is supposed to start
  * @param size the size in bytes of the data block starting at data
  * @return the size in bytes of the AAC ADTS frame starting at the given offset of the given
  *    memory address, 0 if the frame couldn't be parsed.
  */
 static size_t getAdtsFrameSize(const uint8_t *data, off64_t offset, size_t size) {
     size_t frameSize = 0;

     if (!(offset + ADTS_HEADER_SIZE_UP_TO_FRAMESIZE < (off64_t) size)) {
         SL_LOGE("AacBqToPcmCbRenderer::getAdtsFrameSize() returns 0 (can't read syncword or header)"
                 );
         return 0;
     }

     const uint8_t *syncword = data + offset;
     if ((syncword[0] != 0xff) || ((syncword[1] & 0xf6) != 0xf0)) {
         SL_LOGE("AacBqToPcmCbRenderer::getAdtsFrameSize() returns 0 (wrong syncword)");
         return 0;
     }

     const uint8_t protectionAbsent = data[offset+1] & 0x1;

     const uint8_t* header = data + offset + 3;
     frameSize = (header[0] & 0x3) << 11 | header[1] << 3 | header[2] >> 5;
     // the frame size read already contains the size of the header, so no need to add it here

     // protectionAbsent is 0 if there is CRC
     static const size_t kAdtsHeaderLengthNoCrc = 7;
     static const size_t kAdtsHeaderLengthWithCrc = 9;
     size_t headSize = protectionAbsent ? kAdtsHeaderLengthNoCrc : kAdtsHeaderLengthWithCrc;
     if (headSize > frameSize) {
         SL_LOGE("AacBqToPcmCbRenderer::getAdtsFrameSize() returns 0 (frameSize %zu < headSize %zu)",
                 frameSize, headSize);
         return 0;
     }

     SL_LOGV("AacBqToPcmCbRenderer::getAdtsFrameSize() returns %u", frameSize);

     return frameSize;
 }

 /**
  * Returns whether a block of memory starts and ends on AAC ADTS frame boundaries
  * @param data pointer to the compressed audio data
  * @param size the size in bytes of the data block to validate
  * @return SL_RESULT_SUCCESS if there is AAC ADTS data, and it starts and ends on frame boundaries,
  *    or an appropriate error code otherwise:
  *      SL_RESULT_PARAMETER_INVALID if not possible to attempt validation of even one frame
  *      SL_RESULT_CONTENT_CORRUPTED if the frame contents are otherwise invalid
  */
 SLresult AacBqToPcmCbRenderer::validateBufferStartEndOnFrameBoundaries(void* data, size_t size)
 {
     off64_t offset = 0;
     size_t frameSize = 0;

     if ((NULL == data) || (size == 0)) {
         SL_LOGE("No ADTS to validate");
         return SL_RESULT_PARAMETER_INVALID;
     }

     while (offset < (off64_t) size) {
         if ((frameSize = getAdtsFrameSize((uint8_t *)data, offset, size)) == 0) {
             SL_LOGE("found ADTS frame of size 0 at offset %lld", (long long) offset);
             return SL_RESULT_CONTENT_CORRUPTED;
         }
         //SL_LOGV("last good offset %llu", offset);
         offset += frameSize;
         if (offset > (off64_t) size) {
             SL_LOGE("found incomplete ADTS frame at end of data");
             return SL_RESULT_CONTENT_CORRUPTED;
         }
     }
     if (offset != (off64_t) size) {
         SL_LOGE("ADTS parsing error: reached end of incomplete frame");
     }
     assert(offset == (off64_t) size);
     return SL_RESULT_SUCCESS;
 }

 //--------------------------------------------------------------------------------------------------
 AacBqToPcmCbRenderer::AacBqToPcmCbRenderer(const AudioPlayback_Parameters* params,
         IAndroidBufferQueue *androidBufferQueue) :
         AudioToCbRenderer(params),
         mBqSource(new BufferQueueSource(androidBufferQueue))
 {
     SL_LOGD("AacBqToPcmCbRenderer::AacBqToPcmCbRenderer()");
 }


 AacBqToPcmCbRenderer::~AacBqToPcmCbRenderer() {
     SL_LOGD("AacBqToPcmCbRenderer::~AacBqToPcmCbRenderer()");

 }


 //--------------------------------------------------
 // Event handlers
 void AacBqToPcmCbRenderer::onPrepare() {
     SL_LOGD("AacBqToPcmCbRenderer::onPrepare()");
     Mutex::Autolock _l(mBufferSourceLock);

     // Initialize the PCM format info with the known parameters before the start of the decode
     {
         android::Mutex::Autolock autoLock(mPcmFormatLock);
         mPcmFormatValues[ANDROID_KEY_INDEX_PCMFORMAT_BITSPERSAMPLE] = SL_PCMSAMPLEFORMAT_FIXED_16;
         mPcmFormatValues[ANDROID_KEY_INDEX_PCMFORMAT_CONTAINERSIZE] = 16;
         //FIXME not true on all platforms
         mPcmFormatValues[ANDROID_KEY_INDEX_PCMFORMAT_ENDIANNESS] = SL_BYTEORDER_LITTLEENDIAN;
         //    initialization with the default values: they will be replaced by the actual values
         //      once the decoder has figured them out
         mPcmFormatValues[ANDROID_KEY_INDEX_PCMFORMAT_NUMCHANNELS] = UNKNOWN_NUMCHANNELS;
         mPcmFormatValues[ANDROID_KEY_INDEX_PCMFORMAT_SAMPLERATE] = UNKNOWN_SAMPLERATE;
         mPcmFormatValues[ANDROID_KEY_INDEX_PCMFORMAT_CHANNELMASK] = UNKNOWN_CHANNELMASK;
     }

     sp<MediaExtractor> extractor = new AacAdtsExtractor(mBqSource);

     // only decoding a single track of data
     const size_t kTrackToDecode = 0;

     sp<MediaSource> source = extractor->getTrack(kTrackToDecode);
     if (source == 0) {
         SL_LOGE("AacBqToPcmCbRenderer::onPrepare: error getting source from extractor");
         notifyPrepared(ERROR_UNSUPPORTED);
         return;
     }

     // the audio content is not raw PCM, so we need a decoder
     source = SimpleDecodingSource::Create(source);

     if (source == NULL) {
         SL_LOGE("AacBqToPcmCbRenderer::onPrepare: Could not instantiate decoder.");
         notifyPrepared(ERROR_UNSUPPORTED);
         return;
     }

     sp<MetaData> meta = source->getFormat();

     SL_LOGD("AacBqToPcmCbRenderer::onPrepare() after instantiating decoder");

     if (source->start() != OK) {
         SL_LOGE("AacBqToPcmCbRenderer::onPrepare() Failed to start source/decoder.");
         notifyPrepared(MEDIA_ERROR_BASE);
         return;
     }

     //---------------------------------
     int32_t channelCount;
     CHECK(meta->findInt32(kKeyChannelCount, &channelCount));
     int32_t sr;
     CHECK(meta->findInt32(kKeySampleRate, &sr));
     // FIXME similar to AudioSfDecoder::onPrepare()

     // already "good to go" (compare to AudioSfDecoder::onPrepare)
     mCacheStatus = kStatusHigh;
     mCacheFill = 1000;
     notifyStatus();
     notifyCacheFill();

     {
         android::Mutex::Autolock autoLock(mPcmFormatLock);
         mPcmFormatValues[ANDROID_KEY_INDEX_PCMFORMAT_SAMPLERATE] = sr;
         mPcmFormatValues[ANDROID_KEY_INDEX_PCMFORMAT_NUMCHANNELS] = channelCount;
         mPcmFormatValues[ANDROID_KEY_INDEX_PCMFORMAT_CHANNELMASK] =
                 channelCountToMask(channelCount);
     }
     SL_LOGV("AacBqToPcmCbRenderer::onPrepare() channel count=%d SR=%d",
             channelCount, sr);

     //---------------------------------
     // The data source, and audio source (a decoder) are ready to be used
     mDataSource = mBqSource;
     mAudioSource = source;
     mAudioSourceStarted = true;

     //-------------------------------------
     // signal successful completion of prepare
     mStateFlags |= kFlagPrepared;

     // skipping past AudioToCbRenderer and AudioSfDecoder
     GenericPlayer::onPrepare();

     SL_LOGD("AacBqToPcmCbRenderer::onPrepare() done, mStateFlags=0x%x", mStateFlags);
 }

 } // namespace android
	/*
	* 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 USE_LOG SLAndroidLogLevel_Debug

	#include "sles_allinclusive.h"
	#include "android/include/AacBqToPcmCbRenderer.h"
	#include "android/channels.h"
	#include <media/stagefright/SimpleDecodingSource.h>

	namespace android {

	// ADTS header size is 7, but frame size information ends on byte 6 (when counting from byte 1)
	#define ADTS_HEADER_SIZE_UP_TO_FRAMESIZE 6

	/**
	* Returns the size of an AAC ADTS frame.
	* Note that if the returned value + offset > size, it means that a partial frame starts at that
	* offset, but this function will still return the size of the full frame.
	* @param data pointer to the compressed audio data
	* @param offset offset in bytes relative to data of where the frame is supposed to start
	* @param size the size in bytes of the data block starting at data
	* @return the size in bytes of the AAC ADTS frame starting at the given offset of the given
	* memory address, 0 if the frame couldn't be parsed.
	*/
	static size_t getAdtsFrameSize(const uint8_t *data, off64_t offset, size_t size) {
	size_t frameSize = 0;

	if (!(offset + ADTS_HEADER_SIZE_UP_TO_FRAMESIZE < (off64_t) size)) {
	SL_LOGE("AacBqToPcmCbRenderer::getAdtsFrameSize() returns 0 (can't read syncword or header)"
	);
	return 0;
	}

	const uint8_t *syncword = data + offset;
	if ((syncword[0] != 0xff) \|\| ((syncword[1] & 0xf6) != 0xf0)) {
	SL_LOGE("AacBqToPcmCbRenderer::getAdtsFrameSize() returns 0 (wrong syncword)");
	return 0;
	}

	const uint8_t protectionAbsent = data[offset+1] & 0x1;

	const uint8_t* header = data + offset + 3;
	frameSize = (header[0] & 0x3) << 11 \| header[1] << 3 \| header[2] >> 5;
	// the frame size read already contains the size of the header, so no need to add it here

	// protectionAbsent is 0 if there is CRC
	static const size_t kAdtsHeaderLengthNoCrc = 7;
	static const size_t kAdtsHeaderLengthWithCrc = 9;
	size_t headSize = protectionAbsent ? kAdtsHeaderLengthNoCrc : kAdtsHeaderLengthWithCrc;
	if (headSize > frameSize) {
	SL_LOGE("AacBqToPcmCbRenderer::getAdtsFrameSize() returns 0 (frameSize %zu < headSize %zu)",
	frameSize, headSize);
	return 0;
	}

	SL_LOGV("AacBqToPcmCbRenderer::getAdtsFrameSize() returns %u", frameSize);

	return frameSize;
	}

	/**
	* Returns whether a block of memory starts and ends on AAC ADTS frame boundaries
	* @param data pointer to the compressed audio data
	* @param size the size in bytes of the data block to validate
	* @return SL_RESULT_SUCCESS if there is AAC ADTS data, and it starts and ends on frame boundaries,
	* or an appropriate error code otherwise:
	* SL_RESULT_PARAMETER_INVALID if not possible to attempt validation of even one frame
	* SL_RESULT_CONTENT_CORRUPTED if the frame contents are otherwise invalid
	*/
	SLresult AacBqToPcmCbRenderer::validateBufferStartEndOnFrameBoundaries(void* data, size_t size)
	{
	off64_t offset = 0;
	size_t frameSize = 0;

	if ((NULL == data) \|\| (size == 0)) {
	SL_LOGE("No ADTS to validate");
	return SL_RESULT_PARAMETER_INVALID;
	}

	while (offset < (off64_t) size) {
	if ((frameSize = getAdtsFrameSize((uint8_t *)data, offset, size)) == 0) {
	SL_LOGE("found ADTS frame of size 0 at offset %lld", (long long) offset);
	return SL_RESULT_CONTENT_CORRUPTED;
	}
	//SL_LOGV("last good offset %llu", offset);
	offset += frameSize;
	if (offset > (off64_t) size) {
	SL_LOGE("found incomplete ADTS frame at end of data");
	return SL_RESULT_CONTENT_CORRUPTED;
	}
	}
	if (offset != (off64_t) size) {
	SL_LOGE("ADTS parsing error: reached end of incomplete frame");
	}
	assert(offset == (off64_t) size);
	return SL_RESULT_SUCCESS;
	}

	//--------------------------------------------------------------------------------------------------
	AacBqToPcmCbRenderer::AacBqToPcmCbRenderer(const AudioPlayback_Parameters* params,
	IAndroidBufferQueue *androidBufferQueue) :
	AudioToCbRenderer(params),
	mBqSource(new BufferQueueSource(androidBufferQueue))
	{
	SL_LOGD("AacBqToPcmCbRenderer::AacBqToPcmCbRenderer()");
	}


	AacBqToPcmCbRenderer::~AacBqToPcmCbRenderer() {
	SL_LOGD("AacBqToPcmCbRenderer::~AacBqToPcmCbRenderer()");

	}


	//--------------------------------------------------
	// Event handlers
	void AacBqToPcmCbRenderer::onPrepare() {
	SL_LOGD("AacBqToPcmCbRenderer::onPrepare()");
	Mutex::Autolock _l(mBufferSourceLock);

	// Initialize the PCM format info with the known parameters before the start of the decode
	{
	android::Mutex::Autolock autoLock(mPcmFormatLock);
	mPcmFormatValues[ANDROID_KEY_INDEX_PCMFORMAT_BITSPERSAMPLE] = SL_PCMSAMPLEFORMAT_FIXED_16;
	mPcmFormatValues[ANDROID_KEY_INDEX_PCMFORMAT_CONTAINERSIZE] = 16;
	//FIXME not true on all platforms
	mPcmFormatValues[ANDROID_KEY_INDEX_PCMFORMAT_ENDIANNESS] = SL_BYTEORDER_LITTLEENDIAN;
	// initialization with the default values: they will be replaced by the actual values
	// once the decoder has figured them out
	mPcmFormatValues[ANDROID_KEY_INDEX_PCMFORMAT_NUMCHANNELS] = UNKNOWN_NUMCHANNELS;
	mPcmFormatValues[ANDROID_KEY_INDEX_PCMFORMAT_SAMPLERATE] = UNKNOWN_SAMPLERATE;
	mPcmFormatValues[ANDROID_KEY_INDEX_PCMFORMAT_CHANNELMASK] = UNKNOWN_CHANNELMASK;
	}

	sp<MediaExtractor> extractor = new AacAdtsExtractor(mBqSource);

	// only decoding a single track of data
	const size_t kTrackToDecode = 0;

	sp<MediaSource> source = extractor->getTrack(kTrackToDecode);
	if (source == 0) {
	SL_LOGE("AacBqToPcmCbRenderer::onPrepare: error getting source from extractor");
	notifyPrepared(ERROR_UNSUPPORTED);
	return;
	}

	// the audio content is not raw PCM, so we need a decoder
	source = SimpleDecodingSource::Create(source);

	if (source == NULL) {
	SL_LOGE("AacBqToPcmCbRenderer::onPrepare: Could not instantiate decoder.");
	notifyPrepared(ERROR_UNSUPPORTED);
	return;
	}

	sp<MetaData> meta = source->getFormat();

	SL_LOGD("AacBqToPcmCbRenderer::onPrepare() after instantiating decoder");

	if (source->start() != OK) {
	SL_LOGE("AacBqToPcmCbRenderer::onPrepare() Failed to start source/decoder.");
	notifyPrepared(MEDIA_ERROR_BASE);
	return;
	}

	//---------------------------------
	int32_t channelCount;
	CHECK(meta->findInt32(kKeyChannelCount, &channelCount));
	int32_t sr;
	CHECK(meta->findInt32(kKeySampleRate, &sr));
	// FIXME similar to AudioSfDecoder::onPrepare()

	// already "good to go" (compare to AudioSfDecoder::onPrepare)
	mCacheStatus = kStatusHigh;
	mCacheFill = 1000;
	notifyStatus();
	notifyCacheFill();

	{
	android::Mutex::Autolock autoLock(mPcmFormatLock);
	mPcmFormatValues[ANDROID_KEY_INDEX_PCMFORMAT_SAMPLERATE] = sr;
	mPcmFormatValues[ANDROID_KEY_INDEX_PCMFORMAT_NUMCHANNELS] = channelCount;
	mPcmFormatValues[ANDROID_KEY_INDEX_PCMFORMAT_CHANNELMASK] =
	channelCountToMask(channelCount);
	}
	SL_LOGV("AacBqToPcmCbRenderer::onPrepare() channel count=%d SR=%d",
	channelCount, sr);

	//---------------------------------
	// The data source, and audio source (a decoder) are ready to be used
	mDataSource = mBqSource;
	mAudioSource = source;
	mAudioSourceStarted = true;

	//-------------------------------------
	// signal successful completion of prepare
	mStateFlags \|= kFlagPrepared;

	// skipping past AudioToCbRenderer and AudioSfDecoder
	GenericPlayer::onPrepare();

	SL_LOGD("AacBqToPcmCbRenderer::onPrepare() done, mStateFlags=0x%x", mStateFlags);
	}

	} // namespace android