blob: a9dd3a88633afe44bbd059cba056bcdad58dac61 [file] [log] [blame]
/*
* 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 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/base/platform_thread.h"
#include "webrtc/common_audio/resampler/include/resampler.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.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/modules/utility/include/file_recorder.h"
#include "webrtc/modules/utility/source/coder.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/typedefs.h"
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.
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;
protected:
int32_t WriteEncodedAudioData(const int8_t* audioBuffer,
size_t bufferLength);
int32_t SetUpAudioEncoder();
uint32_t _instanceID;
FileFormats _fileFormat;
MediaFile* _moduleFile;
private:
CodecInst codec_info_;
int8_t _audioBuffer[MAX_AUDIO_BUFFER_IN_BYTES];
AudioCoder _audioEncoder;
Resampler _audioResampler;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_