voice_engine/file_recorder.cc - src/webrtc - Git at Google

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */

 #include "webrtc/voice_engine/file_recorder.h"

 #include <list>

 #include "webrtc/base/logging.h"
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/common_audio/resampler/include/resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/media_file/media_file.h"
 #include "webrtc/modules/media_file/media_file_defines.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/voice_engine/coder.h"

 namespace webrtc {

 namespace {

 // The largest decoded frame size in samples (60ms with 32kHz sample rate).
 enum { MAX_AUDIO_BUFFER_IN_SAMPLES = 60 * 32 };
 enum { MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES * 2 };
 enum { kMaxAudioBufferQueueLength = 100 };

 class FileRecorderImpl : public FileRecorder {
  public:
   FileRecorderImpl(uint32_t instanceID, FileFormats fileFormat);
   ~FileRecorderImpl() override;

   // FileRecorder functions.
   int32_t RegisterModuleFileCallback(FileCallback* callback) override;
   FileFormats RecordingFileFormat() const override;
   int32_t StartRecordingAudioFile(const char* fileName,
                                   const CodecInst& codecInst,
                                   uint32_t notificationTimeMs) override;
   int32_t StartRecordingAudioFile(OutStream* destStream,
                                   const CodecInst& codecInst,
                                   uint32_t notificationTimeMs) override;
   int32_t StopRecording() override;
   bool IsRecording() const override;
   int32_t codec_info(CodecInst* codecInst) const override;
   int32_t RecordAudioToFile(const AudioFrame& frame) override;

  private:
   int32_t WriteEncodedAudioData(const int8_t* audioBuffer, size_t bufferLength);

   int32_t SetUpAudioEncoder();

   uint32_t _instanceID;
   FileFormats _fileFormat;
   MediaFile* _moduleFile;

   CodecInst codec_info_;
   int8_t _audioBuffer[MAX_AUDIO_BUFFER_IN_BYTES];
   AudioCoder _audioEncoder;
   Resampler _audioResampler;
 };

 FileRecorderImpl::FileRecorderImpl(uint32_t instanceID, FileFormats fileFormat)
     : _instanceID(instanceID),
       _fileFormat(fileFormat),
       _moduleFile(MediaFile::CreateMediaFile(_instanceID)),
       codec_info_(),
       _audioBuffer(),
       _audioEncoder(instanceID),
       _audioResampler() {}

 FileRecorderImpl::~FileRecorderImpl() {
   MediaFile::DestroyMediaFile(_moduleFile);
 }

 FileFormats FileRecorderImpl::RecordingFileFormat() const {
   return _fileFormat;
 }

 int32_t FileRecorderImpl::RegisterModuleFileCallback(FileCallback* callback) {
   if (_moduleFile == NULL) {
     return -1;
   }
   return _moduleFile->SetModuleFileCallback(callback);
 }

 int32_t FileRecorderImpl::StartRecordingAudioFile(const char* fileName,
                                                   const CodecInst& codecInst,
                                                   uint32_t notificationTimeMs) {
   if (_moduleFile == NULL) {
     return -1;
   }
   codec_info_ = codecInst;
   int32_t retVal = 0;
   retVal = _moduleFile->StartRecordingAudioFile(fileName, _fileFormat,
                                                 codecInst, notificationTimeMs);

   if (retVal == 0) {
     retVal = SetUpAudioEncoder();
   }
   if (retVal != 0) {
     LOG(LS_WARNING) << "Failed to initialize file " << fileName
                     << " for recording.";

     if (IsRecording()) {
       StopRecording();
     }
   }
   return retVal;
 }

 int32_t FileRecorderImpl::StartRecordingAudioFile(OutStream* destStream,
                                                   const CodecInst& codecInst,
                                                   uint32_t notificationTimeMs) {
   codec_info_ = codecInst;
   int32_t retVal = _moduleFile->StartRecordingAudioStream(
       *destStream, _fileFormat, codecInst, notificationTimeMs);

   if (retVal == 0) {
     retVal = SetUpAudioEncoder();
   }
   if (retVal != 0) {
     LOG(LS_WARNING) << "Failed to initialize outStream for recording.";

     if (IsRecording()) {
       StopRecording();
     }
   }
   return retVal;
 }

 int32_t FileRecorderImpl::StopRecording() {
   memset(&codec_info_, 0, sizeof(CodecInst));
   return _moduleFile->StopRecording();
 }

 bool FileRecorderImpl::IsRecording() const {
   return _moduleFile->IsRecording();
 }

 int32_t FileRecorderImpl::RecordAudioToFile(
     const AudioFrame& incomingAudioFrame) {
   if (codec_info_.plfreq == 0) {
     LOG(LS_WARNING) << "RecordAudioToFile() recording audio is not "
                     << "turned on.";
     return -1;
   }
   AudioFrame tempAudioFrame;
   tempAudioFrame.samples_per_channel_ = 0;
   if (incomingAudioFrame.num_channels_ == 2 && !_moduleFile->IsStereo()) {
     // Recording mono but incoming audio is (interleaved) stereo.
     tempAudioFrame.num_channels_ = 1;
     tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_;
     tempAudioFrame.samples_per_channel_ =
         incomingAudioFrame.samples_per_channel_;
     for (size_t i = 0; i < (incomingAudioFrame.samples_per_channel_); i++) {
       // Sample value is the average of left and right buffer rounded to
       // closest integer value. Note samples can be either 1 or 2 byte.
       tempAudioFrame.data_[i] = ((incomingAudioFrame.data_[2 * i] +
                                   incomingAudioFrame.data_[(2 * i) + 1] + 1) >>
                                  1);
     }
   } else if (incomingAudioFrame.num_channels_ == 1 && _moduleFile->IsStereo()) {
     // Recording stereo but incoming audio is mono.
     tempAudioFrame.num_channels_ = 2;
     tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_;
     tempAudioFrame.samples_per_channel_ =
         incomingAudioFrame.samples_per_channel_;
     for (size_t i = 0; i < (incomingAudioFrame.samples_per_channel_); i++) {
       // Duplicate sample to both channels
       tempAudioFrame.data_[2 * i] = incomingAudioFrame.data_[i];
       tempAudioFrame.data_[2 * i + 1] = incomingAudioFrame.data_[i];
     }
   }

   const AudioFrame* ptrAudioFrame = &incomingAudioFrame;
   if (tempAudioFrame.samples_per_channel_ != 0) {
     // If ptrAudioFrame is not empty it contains the audio to be recorded.
     ptrAudioFrame = &tempAudioFrame;
   }

   // Encode the audio data before writing to file. Don't encode if the codec
   // is PCM.
   // NOTE: stereo recording is only supported for WAV files.
   // TODO(hellner): WAV expect PCM in little endian byte order. Not
   // "encoding" with PCM coder should be a problem for big endian systems.
   size_t encodedLenInBytes = 0;
   if (_fileFormat == kFileFormatPreencodedFile ||
       STR_CASE_CMP(codec_info_.plname, "L16") != 0) {
     if (_audioEncoder.Encode(*ptrAudioFrame, _audioBuffer,
                              &encodedLenInBytes) == -1) {
       LOG(LS_WARNING) << "RecordAudioToFile() codec " << codec_info_.plname
                       << " not supported or failed to encode stream.";
       return -1;
     }
   } else {
     size_t outLen = 0;
     _audioResampler.ResetIfNeeded(ptrAudioFrame->sample_rate_hz_,
                                   codec_info_.plfreq,
                                   ptrAudioFrame->num_channels_);
     _audioResampler.Push(
         ptrAudioFrame->data_,
         ptrAudioFrame->samples_per_channel_ * ptrAudioFrame->num_channels_,
         reinterpret_cast<int16_t*>(_audioBuffer), MAX_AUDIO_BUFFER_IN_BYTES,
         outLen);
     encodedLenInBytes = outLen * sizeof(int16_t);
   }

   // Codec may not be operating at a frame rate of 10 ms. Whenever enough
   // 10 ms chunks of data has been pushed to the encoder an encoded frame
   // will be available. Wait until then.
   if (encodedLenInBytes) {
     if (WriteEncodedAudioData(_audioBuffer, encodedLenInBytes) == -1) {
       return -1;
     }
   }
   return 0;
 }

 int32_t FileRecorderImpl::SetUpAudioEncoder() {
   if (_fileFormat == kFileFormatPreencodedFile ||
       STR_CASE_CMP(codec_info_.plname, "L16") != 0) {
     if (_audioEncoder.SetEncodeCodec(codec_info_) == -1) {
       LOG(LS_ERROR) << "SetUpAudioEncoder() codec " << codec_info_.plname
                     << " not supported.";
       return -1;
     }
   }
   return 0;
 }

 int32_t FileRecorderImpl::codec_info(CodecInst* codecInst) const {
   if (codec_info_.plfreq == 0) {
     return -1;
   }
   *codecInst = codec_info_;
   return 0;
 }

 int32_t FileRecorderImpl::WriteEncodedAudioData(const int8_t* audioBuffer,
                                                 size_t bufferLength) {
   return _moduleFile->IncomingAudioData(audioBuffer, bufferLength);
 }

 }  // namespace

 std::unique_ptr<FileRecorder> FileRecorder::CreateFileRecorder(
     uint32_t instanceID,
     FileFormats fileFormat) {
   return std::unique_ptr<FileRecorder>(
       new FileRecorderImpl(instanceID, fileFormat));
 }

 }  // namespace webrtc
	/*
	* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
	*
	* Use of this source code is governed by a BSD-style license
	* that can be found in the LICENSE file in the root of the source
	* tree. An additional intellectual property rights grant can be found
	* in the file PATENTS. All contributing project authors may
	* be found in the AUTHORS file in the root of the source tree.
	*/

	#include "webrtc/voice_engine/file_recorder.h"

	#include <list>

	#include "webrtc/base/logging.h"
	#include "webrtc/base/platform_thread.h"
	#include "webrtc/common_audio/resampler/include/resampler.h"
	#include "webrtc/common_types.h"
	#include "webrtc/modules/include/module_common_types.h"
	#include "webrtc/modules/media_file/media_file.h"
	#include "webrtc/modules/media_file/media_file_defines.h"
	#include "webrtc/system_wrappers/include/event_wrapper.h"
	#include "webrtc/typedefs.h"
	#include "webrtc/voice_engine/coder.h"

	namespace webrtc {

	namespace {

	// The largest decoded frame size in samples (60ms with 32kHz sample rate).
	enum { MAX_AUDIO_BUFFER_IN_SAMPLES = 60 * 32 };
	enum { MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES * 2 };
	enum { kMaxAudioBufferQueueLength = 100 };

	class FileRecorderImpl : public FileRecorder {
	public:
	FileRecorderImpl(uint32_t instanceID, FileFormats fileFormat);
	~FileRecorderImpl() override;

	// FileRecorder functions.
	int32_t RegisterModuleFileCallback(FileCallback* callback) override;
	FileFormats RecordingFileFormat() const override;
	int32_t StartRecordingAudioFile(const char* fileName,
	const CodecInst& codecInst,
	uint32_t notificationTimeMs) override;
	int32_t StartRecordingAudioFile(OutStream* destStream,
	const CodecInst& codecInst,
	uint32_t notificationTimeMs) override;
	int32_t StopRecording() override;
	bool IsRecording() const override;
	int32_t codec_info(CodecInst* codecInst) const override;
	int32_t RecordAudioToFile(const AudioFrame& frame) override;

	private:
	int32_t WriteEncodedAudioData(const int8_t* audioBuffer, size_t bufferLength);

	int32_t SetUpAudioEncoder();

	uint32_t _instanceID;
	FileFormats _fileFormat;
	MediaFile* _moduleFile;

	CodecInst codec_info_;
	int8_t _audioBuffer[MAX_AUDIO_BUFFER_IN_BYTES];
	AudioCoder _audioEncoder;
	Resampler _audioResampler;
	};

	FileRecorderImpl::FileRecorderImpl(uint32_t instanceID, FileFormats fileFormat)
	: _instanceID(instanceID),
	_fileFormat(fileFormat),
	_moduleFile(MediaFile::CreateMediaFile(_instanceID)),
	codec_info_(),
	_audioBuffer(),
	_audioEncoder(instanceID),
	_audioResampler() {}

	FileRecorderImpl::~FileRecorderImpl() {
	MediaFile::DestroyMediaFile(_moduleFile);
	}

	FileFormats FileRecorderImpl::RecordingFileFormat() const {
	return _fileFormat;
	}

	int32_t FileRecorderImpl::RegisterModuleFileCallback(FileCallback* callback) {
	if (_moduleFile == NULL) {
	return -1;
	}
	return _moduleFile->SetModuleFileCallback(callback);
	}

	int32_t FileRecorderImpl::StartRecordingAudioFile(const char* fileName,
	const CodecInst& codecInst,
	uint32_t notificationTimeMs) {
	if (_moduleFile == NULL) {
	return -1;
	}
	codec_info_ = codecInst;
	int32_t retVal = 0;
	retVal = _moduleFile->StartRecordingAudioFile(fileName, _fileFormat,
	codecInst, notificationTimeMs);

	if (retVal == 0) {
	retVal = SetUpAudioEncoder();
	}
	if (retVal != 0) {
	LOG(LS_WARNING) << "Failed to initialize file " << fileName
	<< " for recording.";

	if (IsRecording()) {
	StopRecording();
	}
	}
	return retVal;
	}

	int32_t FileRecorderImpl::StartRecordingAudioFile(OutStream* destStream,
	const CodecInst& codecInst,
	uint32_t notificationTimeMs) {
	codec_info_ = codecInst;
	int32_t retVal = _moduleFile->StartRecordingAudioStream(
	*destStream, _fileFormat, codecInst, notificationTimeMs);

	if (retVal == 0) {
	retVal = SetUpAudioEncoder();
	}
	if (retVal != 0) {
	LOG(LS_WARNING) << "Failed to initialize outStream for recording.";

	if (IsRecording()) {
	StopRecording();
	}
	}
	return retVal;
	}

	int32_t FileRecorderImpl::StopRecording() {
	memset(&codec_info_, 0, sizeof(CodecInst));
	return _moduleFile->StopRecording();
	}

	bool FileRecorderImpl::IsRecording() const {
	return _moduleFile->IsRecording();
	}

	int32_t FileRecorderImpl::RecordAudioToFile(
	const AudioFrame& incomingAudioFrame) {
	if (codec_info_.plfreq == 0) {
	LOG(LS_WARNING) << "RecordAudioToFile() recording audio is not "
	<< "turned on.";
	return -1;
	}
	AudioFrame tempAudioFrame;
	tempAudioFrame.samples_per_channel_ = 0;
	if (incomingAudioFrame.num_channels_ == 2 && !_moduleFile->IsStereo()) {
	// Recording mono but incoming audio is (interleaved) stereo.
	tempAudioFrame.num_channels_ = 1;
	tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_;
	tempAudioFrame.samples_per_channel_ =
	incomingAudioFrame.samples_per_channel_;
	for (size_t i = 0; i < (incomingAudioFrame.samples_per_channel_); i++) {
	// Sample value is the average of left and right buffer rounded to
	// closest integer value. Note samples can be either 1 or 2 byte.
	tempAudioFrame.data_[i] = ((incomingAudioFrame.data_[2 * i] +
	incomingAudioFrame.data_[(2 * i) + 1] + 1) >>
	1);
	}
	} else if (incomingAudioFrame.num_channels_ == 1 && _moduleFile->IsStereo()) {
	// Recording stereo but incoming audio is mono.
	tempAudioFrame.num_channels_ = 2;
	tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_;
	tempAudioFrame.samples_per_channel_ =
	incomingAudioFrame.samples_per_channel_;
	for (size_t i = 0; i < (incomingAudioFrame.samples_per_channel_); i++) {
	// Duplicate sample to both channels
	tempAudioFrame.data_[2 * i] = incomingAudioFrame.data_[i];
	tempAudioFrame.data_[2 * i + 1] = incomingAudioFrame.data_[i];
	}
	}

	const AudioFrame* ptrAudioFrame = &incomingAudioFrame;
	if (tempAudioFrame.samples_per_channel_ != 0) {
	// If ptrAudioFrame is not empty it contains the audio to be recorded.
	ptrAudioFrame = &tempAudioFrame;
	}

	// Encode the audio data before writing to file. Don't encode if the codec
	// is PCM.
	// NOTE: stereo recording is only supported for WAV files.
	// TODO(hellner): WAV expect PCM in little endian byte order. Not
	// "encoding" with PCM coder should be a problem for big endian systems.
	size_t encodedLenInBytes = 0;
	if (_fileFormat == kFileFormatPreencodedFile \|\|
	STR_CASE_CMP(codec_info_.plname, "L16") != 0) {
	if (_audioEncoder.Encode(*ptrAudioFrame, _audioBuffer,
	&encodedLenInBytes) == -1) {
	LOG(LS_WARNING) << "RecordAudioToFile() codec " << codec_info_.plname
	<< " not supported or failed to encode stream.";
	return -1;
	}
	} else {
	size_t outLen = 0;
	_audioResampler.ResetIfNeeded(ptrAudioFrame->sample_rate_hz_,
	codec_info_.plfreq,
	ptrAudioFrame->num_channels_);
	_audioResampler.Push(
	ptrAudioFrame->data_,
	ptrAudioFrame->samples_per_channel_ * ptrAudioFrame->num_channels_,
	reinterpret_cast<int16_t*>(_audioBuffer), MAX_AUDIO_BUFFER_IN_BYTES,
	outLen);
	encodedLenInBytes = outLen * sizeof(int16_t);
	}

	// Codec may not be operating at a frame rate of 10 ms. Whenever enough
	// 10 ms chunks of data has been pushed to the encoder an encoded frame
	// will be available. Wait until then.
	if (encodedLenInBytes) {
	if (WriteEncodedAudioData(_audioBuffer, encodedLenInBytes) == -1) {
	return -1;
	}
	}
	return 0;
	}

	int32_t FileRecorderImpl::SetUpAudioEncoder() {
	if (_fileFormat == kFileFormatPreencodedFile \|\|
	STR_CASE_CMP(codec_info_.plname, "L16") != 0) {
	if (_audioEncoder.SetEncodeCodec(codec_info_) == -1) {
	LOG(LS_ERROR) << "SetUpAudioEncoder() codec " << codec_info_.plname
	<< " not supported.";
	return -1;
	}
	}
	return 0;
	}

	int32_t FileRecorderImpl::codec_info(CodecInst* codecInst) const {
	if (codec_info_.plfreq == 0) {
	return -1;
	}
	*codecInst = codec_info_;
	return 0;
	}

	int32_t FileRecorderImpl::WriteEncodedAudioData(const int8_t* audioBuffer,
	size_t bufferLength) {
	return _moduleFile->IncomingAudioData(audioBuffer, bufferLength);
	}

	} // namespace

	std::unique_ptr<FileRecorder> FileRecorder::CreateFileRecorder(
	uint32_t instanceID,
	FileFormats fileFormat) {
	return std::unique_ptr<FileRecorder>(
	new FileRecorderImpl(instanceID, fileFormat));
	}

	} // namespace webrtc