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/*
* 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 <list>
#include "webrtc/common_audio/resampler/include/resampler.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/media_file/interface/media_file.h"
#include "webrtc/modules/media_file/interface/media_file_defines.h"
#include "webrtc/modules/utility/interface/file_recorder.h"
#include "webrtc/modules/utility/source/coder.h"
#include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "webrtc/system_wrappers/interface/thread_wrapper.h"
#include "webrtc/system_wrappers/interface/tick_util.h"
#include "webrtc/typedefs.h"
#ifdef WEBRTC_MODULE_UTILITY_VIDEO
#include "webrtc/modules/utility/source/frame_scaler.h"
#include "webrtc/modules/utility/source/video_coder.h"
#include "webrtc/modules/utility/source/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 CriticalSectionWrapper;
class FileRecorderImpl : public FileRecorder
{
public:
FileRecorderImpl(uint32_t instanceID, FileFormats fileFormat);
virtual ~FileRecorderImpl();
// FileRecorder functions.
virtual int32_t RegisterModuleFileCallback(FileCallback* callback);
virtual FileFormats RecordingFileFormat() const;
virtual int32_t StartRecordingAudioFile(
const char* fileName,
const CodecInst& codecInst,
uint32_t notificationTimeMs,
ACMAMRPackingFormat amrFormat = AMRFileStorage);
virtual int32_t StartRecordingAudioFile(
OutStream& destStream,
const CodecInst& codecInst,
uint32_t notificationTimeMs,
ACMAMRPackingFormat amrFormat = AMRFileStorage);
virtual int32_t StopRecording();
virtual bool IsRecording() const;
virtual int32_t codec_info(CodecInst& codecInst) const;
virtual int32_t RecordAudioToFile(
const AudioFrame& frame,
const TickTime* playoutTS = NULL);
virtual int32_t StartRecordingVideoFile(
const char* fileName,
const CodecInst& audioCodecInst,
const VideoCodec& videoCodecInst,
ACMAMRPackingFormat amrFormat = AMRFileStorage,
bool videoOnly = false)
{
return -1;
}
virtual int32_t RecordVideoToFile(const I420VideoFrame& videoFrame)
{
return -1;
}
protected:
virtual int32_t WriteEncodedAudioData(
const int8_t* audioBuffer,
size_t bufferLength,
uint16_t millisecondsOfData,
const TickTime* playoutTS);
int32_t SetUpAudioEncoder();
uint32_t _instanceID;
FileFormats _fileFormat;
MediaFile* _moduleFile;
private:
CodecInst codec_info_;
ACMAMRPackingFormat _amrFormat;
int8_t _audioBuffer[MAX_AUDIO_BUFFER_IN_BYTES];
AudioCoder _audioEncoder;
Resampler _audioResampler;
};
#ifdef WEBRTC_MODULE_UTILITY_VIDEO
class AudioFrameFileInfo
{
public:
AudioFrameFileInfo(const int8_t* audioData,
const size_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];
size_t _audioSize;
uint16_t _audioMS;
TickTime _playoutTS;
};
class AviRecorder : public FileRecorderImpl
{
public:
AviRecorder(uint32_t instanceID, FileFormats fileFormat);
virtual ~AviRecorder();
// FileRecorder functions.
virtual int32_t StartRecordingVideoFile(
const char* fileName,
const CodecInst& audioCodecInst,
const VideoCodec& videoCodecInst,
ACMAMRPackingFormat amrFormat = AMRFileStorage,
bool videoOnly = false);
virtual int32_t StopRecording();
virtual int32_t RecordVideoToFile(const I420VideoFrame& videoFrame);
protected:
virtual int32_t WriteEncodedAudioData(
const int8_t* audioBuffer,
size_t bufferLength,
uint16_t millisecondsOfData,
const TickTime* playoutTS);
private:
typedef std::list<AudioFrameFileInfo*> AudioInfoList;
static bool Run(ThreadObj threadObj);
bool Process();
bool StartThread();
bool StopThread();
int32_t EncodeAndWriteVideoToFile(I420VideoFrame& videoFrame);
int32_t ProcessAudio();
size_t CalcI420FrameSize() const;
int32_t SetUpVideoEncoder();
VideoCodec _videoCodecInst;
bool _videoOnly;
AudioInfoList _audioFramesToWrite;
bool _firstAudioFrameReceived;
VideoFramesQueue* _videoFramesQueue;
FrameScaler* _frameScaler;
VideoCoder* _videoEncoder;
size_t _videoMaxPayloadSize;
EncodedVideoData _videoEncodedData;
ThreadWrapper* _thread;
EventWrapper& _timeEvent;
CriticalSectionWrapper* _critSec;
int64_t _writtenVideoFramesCounter;
int64_t _writtenAudioMS;
int64_t _writtenVideoMS;
};
#endif // WEBRTC_MODULE_UTILITY_VIDEO
} // namespace webrtc
#endif // WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_