Files
platform-external-webrtc/webrtc/modules/media_file/source/media_file_impl.h
kjellander@webrtc.org 14665ff7d4 Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
Clang version changed 223108:230914
Details: e144d30..6fdb142/tools/clang/scripts/update.sh

Removes the OVERRIDE macro defined in:
* webrtc/base/common.h
* webrtc/typedefs.h

The majority of the source changes were done by running this in src/:
perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h"  -o -name "*.cc*" -o -name "*.mm*"`

which converted all:
virtual Foo() OVERRIDE
functions to:
Foo() override

Then I manually edited:
* talk/media/webrtc/fakewebrtccommon.h
* webrtc/test/fake_common.h

Remaining uses of OVERRIDE was fixed by search+replace.

Manual edits were done to fix virtual destructors that were
overriding inherited ones.

Finally a build error related to the pure virtual definitions of
Read, Write and Rewind in common_types.h required a bit of
refactoring in:
* webrtc/common_types.cc
* webrtc/common_types.h
* webrtc/system_wrappers/interface/file_wrapper.h
* webrtc/system_wrappers/source/file_impl.cc

This roll should make it possible for us to finally re-enable deadlock
detection for TSan on the buildbots.

BUG=4106
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41069004

Cr-Commit-Position: refs/heads/master@{#8596}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 13:04:54 +00:00

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