media/libstagefright/AACWriter.cpp - third_party/android/platform/frameworks/av - 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.
  */

 #include <fcntl.h>
 #include <inttypes.h>
 #include <sys/prctl.h>
 #include <sys/stat.h>
 #include <sys/types.h>

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

 #include <media/openmax/OMX_Audio.h>
 #include <media/stagefright/AACWriter.h>
 #include <media/stagefright/MediaBuffer.h>
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/MediaDefs.h>
 #include <media/stagefright/MediaErrors.h>
 #include <media/stagefright/MediaSource.h>
 #include <media/stagefright/MetaData.h>
 #include <media/mediarecorder.h>

 namespace android {

 AACWriter::AACWriter(int fd)
     : mFd(dup(fd)),
       mInitCheck(mFd < 0? NO_INIT: OK),
       mStarted(false),
       mPaused(false),
       mResumed(false),
       mThread(0),
       mEstimatedSizeBytes(0),
       mEstimatedDurationUs(0),
       mChannelCount(-1),
       mSampleRate(-1),
       mAACProfile(OMX_AUDIO_AACObjectLC),
       mFrameDurationUs(0) {
 }

 AACWriter::~AACWriter() {
     if (mStarted) {
         reset();
     }

     if (mFd != -1) {
         close(mFd);
         mFd = -1;
     }
 }

 status_t AACWriter::initCheck() const {
     return mInitCheck;
 }


 status_t AACWriter::addSource(const sp<IMediaSource> &source) {
     if (mInitCheck != OK) {
         return mInitCheck;
     }

     if (mSource != NULL) {
         ALOGE("AAC files only support a single track of audio.");
         return UNKNOWN_ERROR;
     }

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

     const char *mime;
     CHECK(meta->findCString(kKeyMIMEType, &mime));

     CHECK(!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_AAC));
     CHECK(meta->findInt32(kKeyChannelCount, &mChannelCount));
     CHECK(meta->findInt32(kKeySampleRate, &mSampleRate));
     CHECK(mChannelCount >= 1 && mChannelCount <= 2);

     // Optionally, we want to check whether AACProfile is also set.
     if (meta->findInt32(kKeyAACProfile, &mAACProfile)) {
         ALOGI("AAC profile is changed to %d", mAACProfile);
     }

     mSource = source;
     return OK;
 }

 status_t AACWriter::start(MetaData * /* params */) {
     if (mInitCheck != OK) {
         return mInitCheck;
     }

     if (mSource == NULL) {
         return UNKNOWN_ERROR;
     }

     if (mStarted && mPaused) {
         mPaused = false;
         mResumed = true;
         return OK;
     } else if (mStarted) {
         // Already started, does nothing
         return OK;
     }

     mFrameDurationUs = (kSamplesPerFrame * 1000000LL + (mSampleRate >> 1))
                             / mSampleRate;

     status_t err = mSource->start();

     if (err != OK) {
         return err;
     }

     pthread_attr_t attr;
     pthread_attr_init(&attr);
     pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_JOINABLE);

     mReachedEOS = false;
     mDone = false;

     pthread_create(&mThread, &attr, ThreadWrapper, this);
     pthread_attr_destroy(&attr);

     mStarted = true;

     return OK;
 }

 status_t AACWriter::pause() {
     if (!mStarted) {
         return OK;
     }
     mPaused = true;
     return OK;
 }

 status_t AACWriter::reset() {
     if (!mStarted) {
         return OK;
     }

     mDone = true;

     void *dummy;
     pthread_join(mThread, &dummy);

     status_t err = static_cast<status_t>(reinterpret_cast<uintptr_t>(dummy));
     {
         status_t status = mSource->stop();
         if (err == OK &&
             (status != OK && status != ERROR_END_OF_STREAM)) {
             err = status;
         }
     }

     mStarted = false;
     return err;
 }

 bool AACWriter::exceedsFileSizeLimit() {
     if (mMaxFileSizeLimitBytes == 0) {
         return false;
     }
     return mEstimatedSizeBytes >= mMaxFileSizeLimitBytes;
 }

 bool AACWriter::exceedsFileDurationLimit() {
     if (mMaxFileDurationLimitUs == 0) {
         return false;
     }
     return mEstimatedDurationUs >= mMaxFileDurationLimitUs;
 }

 // static
 void *AACWriter::ThreadWrapper(void *me) {
     return (void *)(uintptr_t)static_cast<AACWriter *>(me)->threadFunc();
 }

 /*
 * Returns an index into the sample rate table if the
 * given sample rate is found; otherwise, returns -1.
 */
 static bool getSampleRateTableIndex(int sampleRate, uint8_t* tableIndex) {
     static const int kSampleRateTable[] = {
         96000, 88200, 64000, 48000, 44100, 32000,
         24000, 22050, 16000, 12000, 11025, 8000
     };
     const int tableSize =
         sizeof(kSampleRateTable) / sizeof(kSampleRateTable[0]);

     *tableIndex = 0;
     for (int index = 0; index < tableSize; ++index) {
         if (sampleRate == kSampleRateTable[index]) {
             ALOGV("Sample rate: %d and index: %d",
                 sampleRate, index);
             *tableIndex = index;
             return true;
         }
     }

     ALOGE("Sampling rate %d bps is not supported", sampleRate);
     return false;
 }

 /*
  * ADTS (Audio data transport stream) header structure.
  * It consists of 7 or 9 bytes (with or without CRC):
  * 12 bits of syncword 0xFFF, all bits must be 1
  * 1 bit of field ID. 0 for MPEG-4, and 1 for MPEG-2
  * 2 bits of MPEG layer. If in MPEG-TS, set to 0
  * 1 bit of protection absense. Set to 1 if no CRC.
  * 2 bits of profile code. Set to 1 (The MPEG-4 Audio
  *   object type minus 1. We are using AAC-LC = 2)
  * 4 bits of sampling frequency index code (15 is not allowed)
  * 1 bit of private stream. Set to 0.
  * 3 bits of channel configuration code. 0 resevered for inband PCM
  * 1 bit of originality. Set to 0.
  * 1 bit of home. Set to 0.
  * 1 bit of copyrighted steam. Set to 0.
  * 1 bit of copyright start. Set to 0.
  * 13 bits of frame length. It included 7 ot 9 bytes header length.
  *   it is set to (protection absense? 7: 9) + size(AAC frame)
  * 11 bits of buffer fullness. 0x7FF for VBR.
  * 2 bits of frames count in one packet. Set to 0.
  */
 status_t AACWriter::writeAdtsHeader(uint32_t frameLength) {
     uint8_t data = 0xFF;
     write(mFd, &data, 1);

     const uint8_t kFieldId = 0;
     const uint8_t kMpegLayer = 0;
     const uint8_t kProtectionAbsense = 1;  // 1: kAdtsHeaderLength = 7
     data = 0xF0;
     data |= (kFieldId << 3);
     data |= (kMpegLayer << 1);
     data |= kProtectionAbsense;
     write(mFd, &data, 1);

     const uint8_t kProfileCode = mAACProfile - 1;
     uint8_t kSampleFreqIndex;
     CHECK(getSampleRateTableIndex(mSampleRate, &kSampleFreqIndex));
     const uint8_t kPrivateStream = 0;
     const uint8_t kChannelConfigCode = mChannelCount;
     data = (kProfileCode << 6);
     data |= (kSampleFreqIndex << 2);
     data |= (kPrivateStream << 1);
     data |= (kChannelConfigCode >> 2);
     write(mFd, &data, 1);

     // 4 bits from originality to copyright start
     const uint8_t kCopyright = 0;
     const uint32_t kFrameLength = frameLength;
     data = ((kChannelConfigCode & 3) << 6);
     data |= (kCopyright << 2);
     data |= ((kFrameLength & 0x1800) >> 11);
     write(mFd, &data, 1);

     data = ((kFrameLength & 0x07F8) >> 3);
     write(mFd, &data, 1);

     const uint32_t kBufferFullness = 0x7FF;  // VBR
     data = ((kFrameLength & 0x07) << 5);
     data |= ((kBufferFullness & 0x07C0) >> 6);
     write(mFd, &data, 1);

     const uint8_t kFrameCount = 0;
     data = ((kBufferFullness & 0x03F) << 2);
     data |= kFrameCount;
     write(mFd, &data, 1);

     return OK;
 }

 status_t AACWriter::threadFunc() {
     mEstimatedDurationUs = 0;
     mEstimatedSizeBytes = 0;
     int64_t previousPausedDurationUs = 0;
     int64_t maxTimestampUs = 0;
     status_t err = OK;
     bool stoppedPrematurely = true;

     prctl(PR_SET_NAME, (unsigned long)"AACWriterThread", 0, 0, 0);

     while (!mDone && err == OK) {
         MediaBuffer *buffer;
         err = mSource->read(&buffer);

         if (err != OK) {
             break;
         }

         if (mPaused) {
             buffer->release();
             buffer = NULL;
             continue;
         }

         mEstimatedSizeBytes += kAdtsHeaderLength + buffer->range_length();
         if (exceedsFileSizeLimit()) {
             buffer->release();
             buffer = NULL;
             notify(MEDIA_RECORDER_EVENT_INFO, MEDIA_RECORDER_INFO_MAX_FILESIZE_REACHED, 0);
             break;
         }

         int32_t isCodecSpecific = 0;
         if (buffer->meta_data()->findInt32(kKeyIsCodecConfig, &isCodecSpecific) && isCodecSpecific) {
             ALOGV("Drop codec specific info buffer");
             buffer->release();
             buffer = NULL;
             continue;
         }

         int64_t timestampUs;
         CHECK(buffer->meta_data()->findInt64(kKeyTime, &timestampUs));
         if (timestampUs > mEstimatedDurationUs) {
             mEstimatedDurationUs = timestampUs;
         }
         if (mResumed) {
             previousPausedDurationUs += (timestampUs - maxTimestampUs - mFrameDurationUs);
             mResumed = false;
         }
         timestampUs -= previousPausedDurationUs;
         ALOGV("time stamp: %" PRId64 ", previous paused duration: %" PRId64,
             timestampUs, previousPausedDurationUs);
         if (timestampUs > maxTimestampUs) {
             maxTimestampUs = timestampUs;
         }

         if (exceedsFileDurationLimit()) {
             buffer->release();
             buffer = NULL;
             notify(MEDIA_RECORDER_EVENT_INFO, MEDIA_RECORDER_INFO_MAX_DURATION_REACHED, 0);
             break;
         }

         // Each output AAC audio frame to the file contains
         // 1. an ADTS header, followed by
         // 2. the compressed audio data.
         ssize_t dataLength = buffer->range_length();
         uint8_t *data = (uint8_t *)buffer->data() + buffer->range_offset();
         if (writeAdtsHeader(kAdtsHeaderLength + dataLength) != OK ||
             dataLength != write(mFd, data, dataLength)) {
             err = ERROR_IO;
         }

         buffer->release();
         buffer = NULL;

         if (err != OK) {
             break;
         }

         if (stoppedPrematurely) {
             stoppedPrematurely = false;
         }
     }

     if ((err == OK || err == ERROR_END_OF_STREAM) && stoppedPrematurely) {
         err = ERROR_MALFORMED;
     }

     close(mFd);
     mFd = -1;
     mReachedEOS = true;
     if (err == ERROR_END_OF_STREAM) {
         return OK;
     }
     return err;
 }

 bool AACWriter::reachedEOS() {
     return mReachedEOS;
 }

 }  // 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.
	*/

	#include <fcntl.h>
	#include <inttypes.h>
	#include <sys/prctl.h>
	#include <sys/stat.h>
	#include <sys/types.h>

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

	#include <media/openmax/OMX_Audio.h>
	#include <media/stagefright/AACWriter.h>
	#include <media/stagefright/MediaBuffer.h>
	#include <media/stagefright/foundation/ADebug.h>
	#include <media/stagefright/MediaDefs.h>
	#include <media/stagefright/MediaErrors.h>
	#include <media/stagefright/MediaSource.h>
	#include <media/stagefright/MetaData.h>
	#include <media/mediarecorder.h>

	namespace android {

	AACWriter::AACWriter(int fd)
	: mFd(dup(fd)),
	mInitCheck(mFd < 0? NO_INIT: OK),
	mStarted(false),
	mPaused(false),
	mResumed(false),
	mThread(0),
	mEstimatedSizeBytes(0),
	mEstimatedDurationUs(0),
	mChannelCount(-1),
	mSampleRate(-1),
	mAACProfile(OMX_AUDIO_AACObjectLC),
	mFrameDurationUs(0) {
	}

	AACWriter::~AACWriter() {
	if (mStarted) {
	reset();
	}

	if (mFd != -1) {
	close(mFd);
	mFd = -1;
	}
	}

	status_t AACWriter::initCheck() const {
	return mInitCheck;
	}


	status_t AACWriter::addSource(const sp<IMediaSource> &source) {
	if (mInitCheck != OK) {
	return mInitCheck;
	}

	if (mSource != NULL) {
	ALOGE("AAC files only support a single track of audio.");
	return UNKNOWN_ERROR;
	}

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

	const char *mime;
	CHECK(meta->findCString(kKeyMIMEType, &mime));

	CHECK(!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_AAC));
	CHECK(meta->findInt32(kKeyChannelCount, &mChannelCount));
	CHECK(meta->findInt32(kKeySampleRate, &mSampleRate));
	CHECK(mChannelCount >= 1 && mChannelCount <= 2);

	// Optionally, we want to check whether AACProfile is also set.
	if (meta->findInt32(kKeyAACProfile, &mAACProfile)) {
	ALOGI("AAC profile is changed to %d", mAACProfile);
	}

	mSource = source;
	return OK;
	}

	status_t AACWriter::start(MetaData * /* params */) {
	if (mInitCheck != OK) {
	return mInitCheck;
	}

	if (mSource == NULL) {
	return UNKNOWN_ERROR;
	}

	if (mStarted && mPaused) {
	mPaused = false;
	mResumed = true;
	return OK;
	} else if (mStarted) {
	// Already started, does nothing
	return OK;
	}

	mFrameDurationUs = (kSamplesPerFrame * 1000000LL + (mSampleRate >> 1))
	/ mSampleRate;

	status_t err = mSource->start();

	if (err != OK) {
	return err;
	}

	pthread_attr_t attr;
	pthread_attr_init(&attr);
	pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_JOINABLE);

	mReachedEOS = false;
	mDone = false;

	pthread_create(&mThread, &attr, ThreadWrapper, this);
	pthread_attr_destroy(&attr);

	mStarted = true;

	return OK;
	}

	status_t AACWriter::pause() {
	if (!mStarted) {
	return OK;
	}
	mPaused = true;
	return OK;
	}

	status_t AACWriter::reset() {
	if (!mStarted) {
	return OK;
	}

	mDone = true;

	void *dummy;
	pthread_join(mThread, &dummy);

	status_t err = static_cast<status_t>(reinterpret_cast<uintptr_t>(dummy));
	{
	status_t status = mSource->stop();
	if (err == OK &&
	(status != OK && status != ERROR_END_OF_STREAM)) {
	err = status;
	}
	}

	mStarted = false;
	return err;
	}

	bool AACWriter::exceedsFileSizeLimit() {
	if (mMaxFileSizeLimitBytes == 0) {
	return false;
	}
	return mEstimatedSizeBytes >= mMaxFileSizeLimitBytes;
	}

	bool AACWriter::exceedsFileDurationLimit() {
	if (mMaxFileDurationLimitUs == 0) {
	return false;
	}
	return mEstimatedDurationUs >= mMaxFileDurationLimitUs;
	}

	// static
	void AACWriter::ThreadWrapper(void me) {
	return (void )(uintptr_t)static_cast<AACWriter >(me)->threadFunc();
	}

	/*
	* Returns an index into the sample rate table if the
	* given sample rate is found; otherwise, returns -1.
	*/
	static bool getSampleRateTableIndex(int sampleRate, uint8_t* tableIndex) {
	static const int kSampleRateTable[] = {
	96000, 88200, 64000, 48000, 44100, 32000,
	24000, 22050, 16000, 12000, 11025, 8000
	};
	const int tableSize =
	sizeof(kSampleRateTable) / sizeof(kSampleRateTable[0]);

	*tableIndex = 0;
	for (int index = 0; index < tableSize; ++index) {
	if (sampleRate == kSampleRateTable[index]) {
	ALOGV("Sample rate: %d and index: %d",
	sampleRate, index);
	*tableIndex = index;
	return true;
	}
	}

	ALOGE("Sampling rate %d bps is not supported", sampleRate);
	return false;
	}

	/*
	* ADTS (Audio data transport stream) header structure.
	* It consists of 7 or 9 bytes (with or without CRC):
	* 12 bits of syncword 0xFFF, all bits must be 1
	* 1 bit of field ID. 0 for MPEG-4, and 1 for MPEG-2
	* 2 bits of MPEG layer. If in MPEG-TS, set to 0
	* 1 bit of protection absense. Set to 1 if no CRC.
	* 2 bits of profile code. Set to 1 (The MPEG-4 Audio
	* object type minus 1. We are using AAC-LC = 2)
	* 4 bits of sampling frequency index code (15 is not allowed)
	* 1 bit of private stream. Set to 0.
	* 3 bits of channel configuration code. 0 resevered for inband PCM
	* 1 bit of originality. Set to 0.
	* 1 bit of home. Set to 0.
	* 1 bit of copyrighted steam. Set to 0.
	* 1 bit of copyright start. Set to 0.
	* 13 bits of frame length. It included 7 ot 9 bytes header length.
	* it is set to (protection absense? 7: 9) + size(AAC frame)
	* 11 bits of buffer fullness. 0x7FF for VBR.
	* 2 bits of frames count in one packet. Set to 0.
	*/
	status_t AACWriter::writeAdtsHeader(uint32_t frameLength) {
	uint8_t data = 0xFF;
	write(mFd, &data, 1);

	const uint8_t kFieldId = 0;
	const uint8_t kMpegLayer = 0;
	const uint8_t kProtectionAbsense = 1; // 1: kAdtsHeaderLength = 7
	data = 0xF0;
	data \|= (kFieldId << 3);
	data \|= (kMpegLayer << 1);
	data \|= kProtectionAbsense;
	write(mFd, &data, 1);

	const uint8_t kProfileCode = mAACProfile - 1;
	uint8_t kSampleFreqIndex;
	CHECK(getSampleRateTableIndex(mSampleRate, &kSampleFreqIndex));
	const uint8_t kPrivateStream = 0;
	const uint8_t kChannelConfigCode = mChannelCount;
	data = (kProfileCode << 6);
	data \|= (kSampleFreqIndex << 2);
	data \|= (kPrivateStream << 1);
	data \|= (kChannelConfigCode >> 2);
	write(mFd, &data, 1);

	// 4 bits from originality to copyright start
	const uint8_t kCopyright = 0;
	const uint32_t kFrameLength = frameLength;
	data = ((kChannelConfigCode & 3) << 6);
	data \|= (kCopyright << 2);
	data \|= ((kFrameLength & 0x1800) >> 11);
	write(mFd, &data, 1);

	data = ((kFrameLength & 0x07F8) >> 3);
	write(mFd, &data, 1);

	const uint32_t kBufferFullness = 0x7FF; // VBR
	data = ((kFrameLength & 0x07) << 5);
	data \|= ((kBufferFullness & 0x07C0) >> 6);
	write(mFd, &data, 1);

	const uint8_t kFrameCount = 0;
	data = ((kBufferFullness & 0x03F) << 2);
	data \|= kFrameCount;
	write(mFd, &data, 1);

	return OK;
	}

	status_t AACWriter::threadFunc() {
	mEstimatedDurationUs = 0;
	mEstimatedSizeBytes = 0;
	int64_t previousPausedDurationUs = 0;
	int64_t maxTimestampUs = 0;
	status_t err = OK;
	bool stoppedPrematurely = true;

	prctl(PR_SET_NAME, (unsigned long)"AACWriterThread", 0, 0, 0);

	while (!mDone && err == OK) {
	MediaBuffer *buffer;
	err = mSource->read(&buffer);

	if (err != OK) {
	break;
	}

	if (mPaused) {
	buffer->release();
	buffer = NULL;
	continue;
	}

	mEstimatedSizeBytes += kAdtsHeaderLength + buffer->range_length();
	if (exceedsFileSizeLimit()) {
	buffer->release();
	buffer = NULL;
	notify(MEDIA_RECORDER_EVENT_INFO, MEDIA_RECORDER_INFO_MAX_FILESIZE_REACHED, 0);
	break;
	}

	int32_t isCodecSpecific = 0;
	if (buffer->meta_data()->findInt32(kKeyIsCodecConfig, &isCodecSpecific) && isCodecSpecific) {
	ALOGV("Drop codec specific info buffer");
	buffer->release();
	buffer = NULL;
	continue;
	}

	int64_t timestampUs;
	CHECK(buffer->meta_data()->findInt64(kKeyTime, &timestampUs));
	if (timestampUs > mEstimatedDurationUs) {
	mEstimatedDurationUs = timestampUs;
	}
	if (mResumed) {
	previousPausedDurationUs += (timestampUs - maxTimestampUs - mFrameDurationUs);
	mResumed = false;
	}
	timestampUs -= previousPausedDurationUs;
	ALOGV("time stamp: %" PRId64 ", previous paused duration: %" PRId64,
	timestampUs, previousPausedDurationUs);
	if (timestampUs > maxTimestampUs) {
	maxTimestampUs = timestampUs;
	}

	if (exceedsFileDurationLimit()) {
	buffer->release();
	buffer = NULL;
	notify(MEDIA_RECORDER_EVENT_INFO, MEDIA_RECORDER_INFO_MAX_DURATION_REACHED, 0);
	break;
	}

	// Each output AAC audio frame to the file contains
	// 1. an ADTS header, followed by
	// 2. the compressed audio data.
	ssize_t dataLength = buffer->range_length();
	uint8_t data = (uint8_t )buffer->data() + buffer->range_offset();
	if (writeAdtsHeader(kAdtsHeaderLength + dataLength) != OK \|\|
	dataLength != write(mFd, data, dataLength)) {
	err = ERROR_IO;
	}

	buffer->release();
	buffer = NULL;

	if (err != OK) {
	break;
	}

	if (stoppedPrematurely) {
	stoppedPrematurely = false;
	}
	}

	if ((err == OK \|\| err == ERROR_END_OF_STREAM) && stoppedPrematurely) {
	err = ERROR_MALFORMED;
	}

	close(mFd);
	mFd = -1;
	mReachedEOS = true;
	if (err == ERROR_END_OF_STREAM) {
	return OK;
	}
	return err;
	}

	bool AACWriter::reachedEOS() {
	return mReachedEOS;
	}

	} // namespace android