ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
80 lines
2.6 KiB
C++
80 lines
2.6 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_
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#define WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_
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#include "webrtc/common_audio/resampler/include/resampler.h"
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#include "webrtc/common_types.h"
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#include "webrtc/engine_configurations.h"
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#include "webrtc/modules/media_file/include/media_file.h"
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#include "webrtc/modules/media_file/include/media_file_defines.h"
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#include "webrtc/modules/utility/include/file_player.h"
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#include "webrtc/modules/utility/source/coder.h"
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#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
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#include "webrtc/system_wrappers/include/tick_util.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class FilePlayerImpl : public FilePlayer
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{
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public:
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FilePlayerImpl(uint32_t instanceID, FileFormats fileFormat);
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~FilePlayerImpl();
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virtual int Get10msAudioFromFile(
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int16_t* outBuffer,
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size_t& lengthInSamples,
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int frequencyInHz);
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virtual int32_t RegisterModuleFileCallback(FileCallback* callback);
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virtual int32_t StartPlayingFile(
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const char* fileName,
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bool loop,
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uint32_t startPosition,
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float volumeScaling,
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uint32_t notification,
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uint32_t stopPosition = 0,
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const CodecInst* codecInst = NULL);
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virtual int32_t StartPlayingFile(
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InStream& sourceStream,
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uint32_t startPosition,
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float volumeScaling,
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uint32_t notification,
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uint32_t stopPosition = 0,
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const CodecInst* codecInst = NULL);
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virtual int32_t StopPlayingFile();
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virtual bool IsPlayingFile() const;
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virtual int32_t GetPlayoutPosition(uint32_t& durationMs);
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virtual int32_t AudioCodec(CodecInst& audioCodec) const;
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virtual int32_t Frequency() const;
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virtual int32_t SetAudioScaling(float scaleFactor);
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protected:
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int32_t SetUpAudioDecoder();
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uint32_t _instanceID;
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const FileFormats _fileFormat;
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MediaFile& _fileModule;
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uint32_t _decodedLengthInMS;
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private:
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AudioCoder _audioDecoder;
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CodecInst _codec;
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int32_t _numberOf10MsPerFrame;
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int32_t _numberOf10MsInDecoder;
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Resampler _resampler;
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float _scaling;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_
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