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}
149 lines
4.8 KiB
C++
149 lines
4.8 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_MEDIA_FILE_SOURCE_MEDIA_FILE_IMPL_H_
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#define WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_IMPL_H_
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#include "webrtc/common_types.h"
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#include "webrtc/modules/include/module_common_types.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/media_file/source/media_file_utility.h"
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#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
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namespace webrtc {
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class MediaFileImpl : public MediaFile
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{
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public:
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MediaFileImpl(const int32_t id);
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~MediaFileImpl();
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int32_t Process() override;
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int64_t TimeUntilNextProcess() override;
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// MediaFile functions
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int32_t PlayoutAudioData(int8_t* audioBuffer,
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size_t& dataLengthInBytes) override;
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int32_t PlayoutStereoData(int8_t* audioBufferLeft,
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int8_t* audioBufferRight,
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size_t& dataLengthInBytes) override;
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int32_t StartPlayingAudioFile(
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const char* fileName,
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const uint32_t notificationTimeMs = 0,
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const bool loop = false,
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const FileFormats format = kFileFormatPcm16kHzFile,
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const CodecInst* codecInst = NULL,
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const uint32_t startPointMs = 0,
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const uint32_t stopPointMs = 0) override;
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int32_t StartPlayingAudioStream(
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InStream& stream,
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const uint32_t notificationTimeMs = 0,
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const FileFormats format = kFileFormatPcm16kHzFile,
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const CodecInst* codecInst = NULL,
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const uint32_t startPointMs = 0,
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const uint32_t stopPointMs = 0) override;
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int32_t StopPlaying() override;
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bool IsPlaying() override;
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int32_t PlayoutPositionMs(uint32_t& positionMs) const override;
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int32_t IncomingAudioData(const int8_t* audioBuffer,
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const size_t bufferLength) override;
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int32_t StartRecordingAudioFile(const char* fileName,
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const FileFormats format,
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const CodecInst& codecInst,
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const uint32_t notificationTimeMs = 0,
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const uint32_t maxSizeBytes = 0) override;
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int32_t StartRecordingAudioStream(
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OutStream& stream,
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const FileFormats format,
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const CodecInst& codecInst,
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const uint32_t notificationTimeMs = 0) override;
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int32_t StopRecording() override;
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bool IsRecording() override;
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int32_t RecordDurationMs(uint32_t& durationMs) override;
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bool IsStereo() override;
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int32_t SetModuleFileCallback(FileCallback* callback) override;
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int32_t FileDurationMs(const char* fileName,
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uint32_t& durationMs,
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const FileFormats format,
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const uint32_t freqInHz = 16000) override;
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int32_t codec_info(CodecInst& codecInst) const override;
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private:
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// Returns true if the combination of format and codecInst is valid.
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static bool ValidFileFormat(const FileFormats format,
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const CodecInst* codecInst);
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// Returns true if the filename is valid
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static bool ValidFileName(const char* fileName);
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// Returns true if the combination of startPointMs and stopPointMs is valid.
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static bool ValidFilePositions(const uint32_t startPointMs,
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const uint32_t stopPointMs);
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// Returns true if frequencyInHz is a supported frequency.
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static bool ValidFrequency(const uint32_t frequencyInHz);
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void HandlePlayCallbacks(int32_t bytesRead);
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int32_t StartPlayingStream(
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InStream& stream,
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bool loop,
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const uint32_t notificationTimeMs,
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const FileFormats format,
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const CodecInst* codecInst,
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const uint32_t startPointMs,
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const uint32_t stopPointMs);
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int32_t _id;
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CriticalSectionWrapper* _crit;
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CriticalSectionWrapper* _callbackCrit;
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ModuleFileUtility* _ptrFileUtilityObj;
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CodecInst codec_info_;
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InStream* _ptrInStream;
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OutStream* _ptrOutStream;
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FileFormats _fileFormat;
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uint32_t _recordDurationMs;
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uint32_t _playoutPositionMs;
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uint32_t _notificationMs;
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bool _playingActive;
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bool _recordingActive;
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bool _isStereo;
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bool _openFile;
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char _fileName[512];
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FileCallback* _ptrCallback;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_IMPL_H_
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