| /* |
| * 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. |
| */ |
| |
| // This file contains a class that can write audio and/or video to file in |
| // multiple file formats. The unencoded input data is written to file in the |
| // encoded format specified. |
| |
| #ifndef WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_ |
| #define WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_ |
| |
| #include "coder.h" |
| #include "common_types.h" |
| #include "engine_configurations.h" |
| #include "event_wrapper.h" |
| #include "file_recorder.h" |
| #include "media_file_defines.h" |
| #include "media_file.h" |
| #include "module_common_types.h" |
| #include "resampler.h" |
| #include "thread_wrapper.h" |
| #include "tick_util.h" |
| #include "typedefs.h" |
| |
| #ifdef WEBRTC_MODULE_UTILITY_VIDEO |
| #include "frame_scaler.h" |
| #include "video_coder.h" |
| #include "video_frames_queue.h" |
| #endif |
| |
| namespace webrtc { |
| // 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(WebRtc_UWord32 instanceID, FileFormats fileFormat); |
| virtual ~FileRecorderImpl(); |
| |
| // FileRecorder functions. |
| virtual WebRtc_Word32 RegisterModuleFileCallback(FileCallback* callback); |
| virtual FileFormats RecordingFileFormat() const; |
| virtual WebRtc_Word32 StartRecordingAudioFile( |
| const char* fileName, |
| const CodecInst& codecInst, |
| WebRtc_UWord32 notificationTimeMs, |
| ACMAMRPackingFormat amrFormat = AMRFileStorage); |
| virtual WebRtc_Word32 StartRecordingAudioFile( |
| OutStream& destStream, |
| const CodecInst& codecInst, |
| WebRtc_UWord32 notificationTimeMs, |
| ACMAMRPackingFormat amrFormat = AMRFileStorage); |
| virtual WebRtc_Word32 StopRecording(); |
| virtual bool IsRecording() const; |
| virtual WebRtc_Word32 codec_info(CodecInst& codecInst) const; |
| virtual WebRtc_Word32 RecordAudioToFile( |
| const AudioFrame& frame, |
| const TickTime* playoutTS = NULL); |
| virtual WebRtc_Word32 StartRecordingVideoFile( |
| const char* fileName, |
| const CodecInst& audioCodecInst, |
| const VideoCodec& videoCodecInst, |
| ACMAMRPackingFormat amrFormat = AMRFileStorage, |
| bool videoOnly = false) |
| { |
| return -1; |
| } |
| virtual WebRtc_Word32 RecordVideoToFile(const VideoFrame& videoFrame) |
| { |
| return -1; |
| } |
| |
| protected: |
| virtual WebRtc_Word32 WriteEncodedAudioData( |
| const WebRtc_Word8* audioBuffer, |
| WebRtc_UWord16 bufferLength, |
| WebRtc_UWord16 millisecondsOfData, |
| const TickTime* playoutTS); |
| |
| WebRtc_Word32 SetUpAudioEncoder(); |
| |
| WebRtc_UWord32 _instanceID; |
| FileFormats _fileFormat; |
| MediaFile* _moduleFile; |
| |
| private: |
| CodecInst codec_info_; |
| ACMAMRPackingFormat _amrFormat; |
| |
| WebRtc_Word8 _audioBuffer[MAX_AUDIO_BUFFER_IN_BYTES]; |
| AudioCoder _audioEncoder; |
| Resampler _audioResampler; |
| }; |
| |
| |
| #ifdef WEBRTC_MODULE_UTILITY_VIDEO |
| class AviRecorder : public FileRecorderImpl |
| { |
| public: |
| AviRecorder(WebRtc_UWord32 instanceID, FileFormats fileFormat); |
| virtual ~AviRecorder(); |
| |
| // FileRecorder functions. |
| virtual WebRtc_Word32 StartRecordingVideoFile( |
| const char* fileName, |
| const CodecInst& audioCodecInst, |
| const VideoCodec& videoCodecInst, |
| ACMAMRPackingFormat amrFormat = AMRFileStorage, |
| bool videoOnly = false); |
| virtual WebRtc_Word32 StopRecording(); |
| virtual WebRtc_Word32 RecordVideoToFile(const VideoFrame& videoFrame); |
| |
| protected: |
| virtual WebRtc_Word32 WriteEncodedAudioData( |
| const WebRtc_Word8* audioBuffer, |
| WebRtc_UWord16 bufferLength, |
| WebRtc_UWord16 millisecondsOfData, |
| const TickTime* playoutTS); |
| private: |
| static bool Run(ThreadObj threadObj); |
| bool Process(); |
| |
| bool StartThread(); |
| bool StopThread(); |
| |
| WebRtc_Word32 EncodeAndWriteVideoToFile(VideoFrame& videoFrame); |
| WebRtc_Word32 ProcessAudio(); |
| |
| WebRtc_Word32 CalcI420FrameSize() const; |
| WebRtc_Word32 SetUpVideoEncoder(); |
| |
| VideoCodec _videoCodecInst; |
| bool _videoOnly; |
| |
| ListWrapper _audioFramesToWrite; |
| bool _firstAudioFrameReceived; |
| |
| VideoFramesQueue* _videoFramesQueue; |
| |
| FrameScaler* _frameScaler; |
| VideoCoder* _videoEncoder; |
| WebRtc_Word32 _videoMaxPayloadSize; |
| EncodedVideoData _videoEncodedData; |
| |
| ThreadWrapper* _thread; |
| EventWrapper& _timeEvent; |
| CriticalSectionWrapper* _critSec; |
| WebRtc_Word64 _writtenVideoFramesCounter; |
| WebRtc_Word64 _writtenAudioMS; |
| WebRtc_Word64 _writtenVideoMS; |
| }; |
| #endif // WEBRTC_MODULE_UTILITY_VIDEO |
| } // namespace webrtc |
| #endif // WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_ |