webrtc_m130/webrtc/modules/utility/source/file_player_impl.h
Henrik Kjellander 0b9e29c87d Remove include dirs from modules/{media_file,pacing}
Also move files out of media_file/source.

BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
R=asapersson@webrtc.org, perkj@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1435093002 .

Cr-Commit-Position: refs/heads/master@{#10647}
2015-11-16 10:12:32 +00:00

80 lines
2.6 KiB
C++

/*
* 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.
*/
#ifndef WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_
#define WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_
#include "webrtc/common_audio/resampler/include/resampler.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/media_file/media_file.h"
#include "webrtc/modules/media_file/media_file_defines.h"
#include "webrtc/modules/utility/include/file_player.h"
#include "webrtc/modules/utility/source/coder.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/tick_util.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class FilePlayerImpl : public FilePlayer
{
public:
FilePlayerImpl(uint32_t instanceID, FileFormats fileFormat);
~FilePlayerImpl();
virtual int Get10msAudioFromFile(
int16_t* outBuffer,
size_t& lengthInSamples,
int frequencyInHz);
virtual int32_t RegisterModuleFileCallback(FileCallback* callback);
virtual int32_t StartPlayingFile(
const char* fileName,
bool loop,
uint32_t startPosition,
float volumeScaling,
uint32_t notification,
uint32_t stopPosition = 0,
const CodecInst* codecInst = NULL);
virtual int32_t StartPlayingFile(
InStream& sourceStream,
uint32_t startPosition,
float volumeScaling,
uint32_t notification,
uint32_t stopPosition = 0,
const CodecInst* codecInst = NULL);
virtual int32_t StopPlayingFile();
virtual bool IsPlayingFile() const;
virtual int32_t GetPlayoutPosition(uint32_t& durationMs);
virtual int32_t AudioCodec(CodecInst& audioCodec) const;
virtual int32_t Frequency() const;
virtual int32_t SetAudioScaling(float scaleFactor);
protected:
int32_t SetUpAudioDecoder();
uint32_t _instanceID;
const FileFormats _fileFormat;
MediaFile& _fileModule;
uint32_t _decodedLengthInMS;
private:
AudioCoder _audioDecoder;
CodecInst _codec;
int32_t _numberOf10MsPerFrame;
int32_t _numberOf10MsInDecoder;
Resampler _resampler;
float _scaling;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_