Files
platform-external-webrtc/webrtc/modules/utility/source/coder.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

66 lines
2.0 KiB
C++

/*
* Copyright (c) 2011 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_CODER_H_
#define WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class AudioFrame;
class AudioCoder : public AudioPacketizationCallback
{
public:
AudioCoder(uint32_t instanceID);
~AudioCoder();
int32_t SetEncodeCodec(
const CodecInst& codecInst,
ACMAMRPackingFormat amrFormat = AMRBandwidthEfficient);
int32_t SetDecodeCodec(
const CodecInst& codecInst,
ACMAMRPackingFormat amrFormat = AMRBandwidthEfficient);
int32_t Decode(AudioFrame& decodedAudio, uint32_t sampFreqHz,
const int8_t* incomingPayload, size_t payloadLength);
int32_t PlayoutData(AudioFrame& decodedAudio, uint16_t& sampFreqHz);
int32_t Encode(const AudioFrame& audio, int8_t* encodedData,
size_t& encodedLengthInBytes);
protected:
int32_t SendData(FrameType frameType,
uint8_t payloadType,
uint32_t timeStamp,
const uint8_t* payloadData,
size_t payloadSize,
const RTPFragmentationHeader* fragmentation) override;
private:
rtc::scoped_ptr<AudioCodingModule> _acm;
CodecInst _receiveCodec;
uint32_t _encodeTimestamp;
int8_t* _encodedData;
size_t _encodedLengthInBytes;
uint32_t _decodeTimestamp;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_