| Index: webrtc/modules/utility/source/file_recorder_impl.h
|
| diff --git a/webrtc/modules/utility/source/file_recorder_impl.h b/webrtc/modules/utility/source/file_recorder_impl.h
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..a9dd3a88633afe44bbd059cba056bcdad58dac61
|
| --- /dev/null
|
| +++ b/webrtc/modules/utility/source/file_recorder_impl.h
|
| @@ -0,0 +1,79 @@
|
| +/*
|
| + * 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_
|
|
|