| /* |
| * 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/common_video/libyuv/include/webrtc_libyuv.h" |
| #include "webrtc/engine_configurations.h" |
| #include "webrtc/modules/media_file/interface/media_file.h" |
| #include "webrtc/modules/utility/source/file_recorder_impl.h" |
| #include "webrtc/system_wrappers/interface/trace.h" |
| |
| #ifdef WEBRTC_MODULE_UTILITY_VIDEO |
| #include "critical_section_wrapper.h" |
| #include "frame_scaler.h" |
| #include "video_coder.h" |
| #include "video_frames_queue.h" |
| #endif |
| |
| namespace webrtc { |
| FileRecorder* FileRecorder::CreateFileRecorder(uint32_t instanceID, |
| FileFormats fileFormat) |
| { |
| switch(fileFormat) |
| { |
| case kFileFormatWavFile: |
| case kFileFormatCompressedFile: |
| case kFileFormatPreencodedFile: |
| case kFileFormatPcm16kHzFile: |
| case kFileFormatPcm8kHzFile: |
| case kFileFormatPcm32kHzFile: |
| return new FileRecorderImpl(instanceID, fileFormat); |
| case kFileFormatAviFile: |
| #ifdef WEBRTC_MODULE_UTILITY_VIDEO |
| return new AviRecorder(instanceID, fileFormat); |
| #else |
| WEBRTC_TRACE(kTraceError, kTraceFile, -1, |
| "Invalid file format: %d", kFileFormatAviFile); |
| assert(false); |
| return NULL; |
| #endif |
| } |
| assert(false); |
| return NULL; |
| } |
| |
| void FileRecorder::DestroyFileRecorder(FileRecorder* recorder) |
| { |
| delete recorder; |
| } |
| |
| FileRecorderImpl::FileRecorderImpl(uint32_t instanceID, |
| FileFormats fileFormat) |
| : _instanceID(instanceID), |
| _fileFormat(fileFormat), |
| _moduleFile(MediaFile::CreateMediaFile(_instanceID)), |
| codec_info_(), |
| _amrFormat(AMRFileStorage), |
| _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, |
| ACMAMRPackingFormat amrFormat) |
| { |
| if(_moduleFile == NULL) |
| { |
| return -1; |
| } |
| codec_info_ = codecInst; |
| _amrFormat = amrFormat; |
| |
| int32_t retVal = 0; |
| if(_fileFormat != kFileFormatAviFile) |
| { |
| // AVI files should be started using StartRecordingVideoFile(..) all |
| // other formats should use this API. |
| retVal =_moduleFile->StartRecordingAudioFile(fileName, _fileFormat, |
| codecInst, |
| notificationTimeMs); |
| } |
| |
| if( retVal == 0) |
| { |
| retVal = SetUpAudioEncoder(); |
| } |
| if( retVal != 0) |
| { |
| WEBRTC_TRACE( |
| kTraceWarning, |
| kTraceVoice, |
| _instanceID, |
| "FileRecorder::StartRecording() failed to initialize file %s for\ |
| recording.", |
| fileName); |
| |
| if(IsRecording()) |
| { |
| StopRecording(); |
| } |
| } |
| return retVal; |
| } |
| |
| int32_t FileRecorderImpl::StartRecordingAudioFile( |
| OutStream& destStream, |
| const CodecInst& codecInst, |
| uint32_t notificationTimeMs, |
| ACMAMRPackingFormat amrFormat) |
| { |
| codec_info_ = codecInst; |
| _amrFormat = amrFormat; |
| |
| int32_t retVal = _moduleFile->StartRecordingAudioStream( |
| destStream, |
| _fileFormat, |
| codecInst, |
| notificationTimeMs); |
| |
| if( retVal == 0) |
| { |
| retVal = SetUpAudioEncoder(); |
| } |
| if( retVal != 0) |
| { |
| WEBRTC_TRACE( |
| kTraceWarning, |
| kTraceVoice, |
| _instanceID, |
| "FileRecorder::StartRecording() 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, |
| const TickTime* playoutTS) |
| { |
| if (codec_info_.plfreq == 0) |
| { |
| WEBRTC_TRACE( |
| kTraceWarning, |
| kTraceVoice, |
| _instanceID, |
| "FileRecorder::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 (uint16_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 (uint16_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. |
| uint32_t encodedLenInBytes = 0; |
| if (_fileFormat == kFileFormatPreencodedFile || |
| STR_CASE_CMP(codec_info_.plname, "L16") != 0) |
| { |
| if (_audioEncoder.Encode(*ptrAudioFrame, _audioBuffer, |
| encodedLenInBytes) == -1) |
| { |
| WEBRTC_TRACE( |
| kTraceWarning, |
| kTraceVoice, |
| _instanceID, |
| "FileRecorder::RecordAudioToFile() codec %s not supported or\ |
| failed to encode stream", |
| codec_info_.plname); |
| return -1; |
| } |
| } else { |
| int outLen = 0; |
| if(ptrAudioFrame->num_channels_ == 2) |
| { |
| // ptrAudioFrame contains interleaved stereo audio. |
| _audioResampler.ResetIfNeeded(ptrAudioFrame->sample_rate_hz_, |
| codec_info_.plfreq, |
| kResamplerSynchronousStereo); |
| _audioResampler.Push(ptrAudioFrame->data_, |
| ptrAudioFrame->samples_per_channel_ * |
| ptrAudioFrame->num_channels_, |
| (int16_t*)_audioBuffer, |
| MAX_AUDIO_BUFFER_IN_BYTES, outLen); |
| } else { |
| _audioResampler.ResetIfNeeded(ptrAudioFrame->sample_rate_hz_, |
| codec_info_.plfreq, |
| kResamplerSynchronous); |
| _audioResampler.Push(ptrAudioFrame->data_, |
| ptrAudioFrame->samples_per_channel_, |
| (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) |
| { |
| uint16_t msOfData = |
| ptrAudioFrame->samples_per_channel_ / |
| uint16_t(ptrAudioFrame->sample_rate_hz_ / 1000); |
| if (WriteEncodedAudioData(_audioBuffer, |
| (uint16_t)encodedLenInBytes, |
| msOfData, playoutTS) == -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_,_amrFormat) == -1) |
| { |
| WEBRTC_TRACE( |
| kTraceError, |
| kTraceVoice, |
| _instanceID, |
| "FileRecorder::StartRecording() codec %s not supported", |
| codec_info_.plname); |
| 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, |
| uint16_t bufferLength, |
| uint16_t /*millisecondsOfData*/, |
| const TickTime* /*playoutTS*/) |
| { |
| return _moduleFile->IncomingAudioData(audioBuffer, bufferLength); |
| } |
| |
| |
| #ifdef WEBRTC_MODULE_UTILITY_VIDEO |
| class AudioFrameFileInfo |
| { |
| public: |
| AudioFrameFileInfo(const int8_t* audioData, |
| const uint16_t audioSize, |
| const uint16_t audioMS, |
| const TickTime& playoutTS) |
| : _audioData(), _audioSize(audioSize), _audioMS(audioMS), |
| _playoutTS(playoutTS) |
| { |
| if(audioSize > MAX_AUDIO_BUFFER_IN_BYTES) |
| { |
| assert(false); |
| _audioSize = 0; |
| return; |
| } |
| memcpy(_audioData, audioData, audioSize); |
| }; |
| // TODO (hellner): either turn into a struct or provide get/set functions. |
| int8_t _audioData[MAX_AUDIO_BUFFER_IN_BYTES]; |
| uint16_t _audioSize; |
| uint16_t _audioMS; |
| TickTime _playoutTS; |
| }; |
| |
| AviRecorder::AviRecorder(uint32_t instanceID, FileFormats fileFormat) |
| : FileRecorderImpl(instanceID, fileFormat), |
| _videoOnly(false), |
| _thread( 0), |
| _timeEvent(*EventWrapper::Create()), |
| _critSec(CriticalSectionWrapper::CreateCriticalSection()), |
| _writtenVideoFramesCounter(0), |
| _writtenAudioMS(0), |
| _writtenVideoMS(0) |
| { |
| _videoEncoder = new VideoCoder(instanceID); |
| _frameScaler = new FrameScaler(); |
| _videoFramesQueue = new VideoFramesQueue(); |
| _thread = ThreadWrapper::CreateThread(Run, this, kNormalPriority, |
| "AviRecorder()"); |
| } |
| |
| AviRecorder::~AviRecorder( ) |
| { |
| StopRecording( ); |
| |
| delete _videoEncoder; |
| delete _frameScaler; |
| delete _videoFramesQueue; |
| delete _thread; |
| delete &_timeEvent; |
| delete _critSec; |
| } |
| |
| int32_t AviRecorder::StartRecordingVideoFile( |
| const char* fileName, |
| const CodecInst& audioCodecInst, |
| const VideoCodec& videoCodecInst, |
| ACMAMRPackingFormat amrFormat, |
| bool videoOnly) |
| { |
| _firstAudioFrameReceived = false; |
| _videoCodecInst = videoCodecInst; |
| _videoOnly = videoOnly; |
| |
| if(_moduleFile->StartRecordingVideoFile(fileName, _fileFormat, |
| audioCodecInst, videoCodecInst, |
| videoOnly) != 0) |
| { |
| return -1; |
| } |
| |
| if(!videoOnly) |
| { |
| if(FileRecorderImpl::StartRecordingAudioFile(fileName,audioCodecInst, 0, |
| amrFormat) !=0) |
| { |
| StopRecording(); |
| return -1; |
| } |
| } |
| if( SetUpVideoEncoder() != 0) |
| { |
| StopRecording(); |
| return -1; |
| } |
| if(_videoOnly) |
| { |
| // Writing to AVI file is non-blocking. |
| // Start non-blocking timer if video only. If recording both video and |
| // audio let the pushing of audio frames be the timer. |
| _timeEvent.StartTimer(true, 1000 / _videoCodecInst.maxFramerate); |
| } |
| StartThread(); |
| return 0; |
| } |
| |
| int32_t AviRecorder::StopRecording() |
| { |
| _timeEvent.StopTimer(); |
| |
| StopThread(); |
| return FileRecorderImpl::StopRecording(); |
| } |
| |
| int32_t AviRecorder::CalcI420FrameSize( ) const |
| { |
| return 3 * _videoCodecInst.width * _videoCodecInst.height / 2; |
| } |
| |
| int32_t AviRecorder::SetUpVideoEncoder() |
| { |
| // Size of unencoded data (I420) should be the largest possible frame size |
| // in a file. |
| _videoMaxPayloadSize = CalcI420FrameSize(); |
| _videoEncodedData.VerifyAndAllocate(_videoMaxPayloadSize); |
| |
| _videoCodecInst.plType = _videoEncoder->DefaultPayloadType( |
| _videoCodecInst.plName); |
| |
| int32_t useNumberOfCores = 1; |
| // Set the max payload size to 16000. This means that the codec will try to |
| // create slices that will fit in 16000 kByte packets. However, the |
| // Encode() call will still generate one full frame. |
| if(_videoEncoder->SetEncodeCodec(_videoCodecInst, useNumberOfCores, |
| 16000)) |
| { |
| return -1; |
| } |
| return 0; |
| } |
| |
| int32_t AviRecorder::RecordVideoToFile(const I420VideoFrame& videoFrame) |
| { |
| CriticalSectionScoped lock(_critSec); |
| if(!IsRecording() || videoFrame.IsZeroSize()) |
| { |
| return -1; |
| } |
| // The frame is written to file in AviRecorder::Process(). |
| int32_t retVal = _videoFramesQueue->AddFrame(videoFrame); |
| if(retVal != 0) |
| { |
| StopRecording(); |
| } |
| return retVal; |
| } |
| |
| bool AviRecorder::StartThread() |
| { |
| unsigned int id; |
| if( _thread == 0) |
| { |
| return false; |
| } |
| |
| return _thread->Start(id); |
| } |
| |
| bool AviRecorder::StopThread() |
| { |
| _critSec->Enter(); |
| |
| if(_thread) |
| { |
| _thread->SetNotAlive(); |
| |
| ThreadWrapper* thread = _thread; |
| _thread = NULL; |
| |
| _timeEvent.Set(); |
| |
| _critSec->Leave(); |
| |
| if(thread->Stop()) |
| { |
| delete thread; |
| } else { |
| return false; |
| } |
| } else { |
| _critSec->Leave(); |
| } |
| return true; |
| } |
| |
| bool AviRecorder::Run( ThreadObj threadObj) |
| { |
| return static_cast<AviRecorder*>( threadObj)->Process(); |
| } |
| |
| int32_t AviRecorder::ProcessAudio() |
| { |
| if (_writtenVideoFramesCounter == 0) |
| { |
| // Get the most recent frame that is due for writing to file. Since |
| // frames are unencoded it's safe to throw away frames if necessary |
| // for synchronizing audio and video. |
| I420VideoFrame* frameToProcess = _videoFramesQueue->FrameToRecord(); |
| if(frameToProcess) |
| { |
| // Syncronize audio to the current frame to process by throwing away |
| // audio samples with older timestamp than the video frame. |
| uint32_t numberOfAudioElements = |
| _audioFramesToWrite.GetSize(); |
| for (uint32_t i = 0; i < numberOfAudioElements; ++i) |
| { |
| AudioFrameFileInfo* frameInfo = |
| (AudioFrameFileInfo*)_audioFramesToWrite.First()->GetItem(); |
| if(frameInfo) |
| { |
| if(TickTime::TicksToMilliseconds( |
| frameInfo->_playoutTS.Ticks()) < |
| frameToProcess->render_time_ms()) |
| { |
| delete frameInfo; |
| _audioFramesToWrite.PopFront(); |
| } else |
| { |
| break; |
| } |
| } |
| } |
| } |
| } |
| // Write all audio up to current timestamp. |
| int32_t error = 0; |
| uint32_t numberOfAudioElements = _audioFramesToWrite.GetSize(); |
| for (uint32_t i = 0; i < numberOfAudioElements; ++i) |
| { |
| AudioFrameFileInfo* frameInfo = |
| (AudioFrameFileInfo*)_audioFramesToWrite.First()->GetItem(); |
| if(frameInfo) |
| { |
| if((TickTime::Now() - frameInfo->_playoutTS).Milliseconds() > 0) |
| { |
| _moduleFile->IncomingAudioData(frameInfo->_audioData, |
| frameInfo->_audioSize); |
| _writtenAudioMS += frameInfo->_audioMS; |
| delete frameInfo; |
| _audioFramesToWrite.PopFront(); |
| } else { |
| break; |
| } |
| } else { |
| _audioFramesToWrite.PopFront(); |
| } |
| } |
| return error; |
| } |
| |
| bool AviRecorder::Process() |
| { |
| switch(_timeEvent.Wait(500)) |
| { |
| case kEventSignaled: |
| if(_thread == NULL) |
| { |
| return false; |
| } |
| break; |
| case kEventError: |
| return false; |
| case kEventTimeout: |
| // No events triggered. No work to do. |
| return true; |
| } |
| CriticalSectionScoped lock( _critSec); |
| |
| // Get the most recent frame to write to file (if any). Synchronize it with |
| // the audio stream (if any). Synchronization the video based on its render |
| // timestamp (i.e. VideoFrame::RenderTimeMS()) |
| I420VideoFrame* frameToProcess = _videoFramesQueue->FrameToRecord(); |
| if( frameToProcess == NULL) |
| { |
| return true; |
| } |
| int32_t error = 0; |
| if(!_videoOnly) |
| { |
| if(!_firstAudioFrameReceived) |
| { |
| // Video and audio can only be synchronized if both have been |
| // received. |
| return true; |
| } |
| error = ProcessAudio(); |
| |
| while (_writtenAudioMS > _writtenVideoMS) |
| { |
| error = EncodeAndWriteVideoToFile( *frameToProcess); |
| if( error != 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceVideo, _instanceID, |
| "AviRecorder::Process() error writing to file."); |
| break; |
| } else { |
| uint32_t frameLengthMS = 1000 / |
| _videoCodecInst.maxFramerate; |
| _writtenVideoFramesCounter++; |
| _writtenVideoMS += frameLengthMS; |
| // A full seconds worth of frames have been written. |
| if(_writtenVideoFramesCounter%_videoCodecInst.maxFramerate == 0) |
| { |
| // Frame rate is in frames per seconds. Frame length is |
| // calculated as an integer division which means it may |
| // be rounded down. Compensate for this every second. |
| uint32_t rest = 1000 % frameLengthMS; |
| _writtenVideoMS += rest; |
| } |
| } |
| } |
| } else { |
| // Frame rate is in frames per seconds. Frame length is calculated as an |
| // integer division which means it may be rounded down. This introduces |
| // drift. Once a full frame worth of drift has happened, skip writing |
| // one frame. Note that frame rate is in frames per second so the |
| // drift is completely compensated for. |
| uint32_t frameLengthMS = 1000/_videoCodecInst.maxFramerate; |
| uint32_t restMS = 1000 % frameLengthMS; |
| uint32_t frameSkip = (_videoCodecInst.maxFramerate * |
| frameLengthMS) / restMS; |
| |
| _writtenVideoFramesCounter++; |
| if(_writtenVideoFramesCounter % frameSkip == 0) |
| { |
| _writtenVideoMS += frameLengthMS; |
| return true; |
| } |
| |
| error = EncodeAndWriteVideoToFile( *frameToProcess); |
| if(error != 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceVideo, _instanceID, |
| "AviRecorder::Process() error writing to file."); |
| } else { |
| _writtenVideoMS += frameLengthMS; |
| } |
| } |
| return error == 0; |
| } |
| |
| int32_t AviRecorder::EncodeAndWriteVideoToFile(I420VideoFrame& videoFrame) |
| { |
| if (!IsRecording() || videoFrame.IsZeroSize()) |
| { |
| return -1; |
| } |
| |
| if(_frameScaler->ResizeFrameIfNeeded(&videoFrame, _videoCodecInst.width, |
| _videoCodecInst.height) != 0) |
| { |
| return -1; |
| } |
| |
| _videoEncodedData.payloadSize = 0; |
| |
| if( STR_CASE_CMP(_videoCodecInst.plName, "I420") == 0) |
| { |
| int length = CalcBufferSize(kI420, videoFrame.width(), |
| videoFrame.height()); |
| _videoEncodedData.VerifyAndAllocate(length); |
| |
| // I420 is raw data. No encoding needed (each sample is represented by |
| // 1 byte so there is no difference depending on endianness). |
| int ret_length = ExtractBuffer(videoFrame, length, |
| _videoEncodedData.payloadData); |
| if (ret_length < 0) |
| return -1; |
| |
| _videoEncodedData.payloadSize = ret_length; |
| _videoEncodedData.frameType = kVideoFrameKey; |
| }else { |
| if( _videoEncoder->Encode(videoFrame, _videoEncodedData) != 0) |
| { |
| return -1; |
| } |
| } |
| |
| if(_videoEncodedData.payloadSize > 0) |
| { |
| if(_moduleFile->IncomingAVIVideoData( |
| (int8_t*)(_videoEncodedData.payloadData), |
| _videoEncodedData.payloadSize)) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceVideo, _instanceID, |
| "Error writing AVI file"); |
| return -1; |
| } |
| } else { |
| WEBRTC_TRACE( |
| kTraceError, |
| kTraceVideo, |
| _instanceID, |
| "FileRecorder::RecordVideoToFile() frame dropped by encoder bitrate\ |
| likely to low."); |
| } |
| return 0; |
| } |
| |
| // Store audio frame in the _audioFramesToWrite buffer. The writing to file |
| // happens in AviRecorder::Process(). |
| int32_t AviRecorder::WriteEncodedAudioData( |
| const int8_t* audioBuffer, |
| uint16_t bufferLength, |
| uint16_t millisecondsOfData, |
| const TickTime* playoutTS) |
| { |
| if (!IsRecording()) |
| { |
| return -1; |
| } |
| if (bufferLength > MAX_AUDIO_BUFFER_IN_BYTES) |
| { |
| return -1; |
| } |
| if (_videoOnly) |
| { |
| return -1; |
| } |
| if (_audioFramesToWrite.GetSize() > kMaxAudioBufferQueueLength) |
| { |
| StopRecording(); |
| return -1; |
| } |
| _firstAudioFrameReceived = true; |
| |
| if(playoutTS) |
| { |
| _audioFramesToWrite.PushBack(new AudioFrameFileInfo(audioBuffer, |
| bufferLength, |
| millisecondsOfData, |
| *playoutTS)); |
| } else { |
| _audioFramesToWrite.PushBack(new AudioFrameFileInfo(audioBuffer, |
| bufferLength, |
| millisecondsOfData, |
| TickTime::Now())); |
| } |
| _timeEvent.Set(); |
| return 0; |
| } |
| |
| #endif // WEBRTC_MODULE_UTILITY_VIDEO |
| } // namespace webrtc |
