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

64 lines
2.6 KiB
C++

/*
* Copyright (c) 2013 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_AUDIO_CODING_NETEQ_DECISION_LOGIC_FAX_H_
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_FAX_H_
#include "webrtc/base/constructormagic.h"
#include "webrtc/modules/audio_coding/neteq/decision_logic.h"
#include "webrtc/typedefs.h"
namespace webrtc {
// Implementation of the DecisionLogic class for playout modes kPlayoutFax and
// kPlayoutOff.
class DecisionLogicFax : public DecisionLogic {
public:
// Constructor.
DecisionLogicFax(int fs_hz,
int output_size_samples,
NetEqPlayoutMode playout_mode,
DecoderDatabase* decoder_database,
const PacketBuffer& packet_buffer,
DelayManager* delay_manager,
BufferLevelFilter* buffer_level_filter)
: DecisionLogic(fs_hz, output_size_samples, playout_mode,
decoder_database, packet_buffer, delay_manager,
buffer_level_filter) {
}
// Destructor.
virtual ~DecisionLogicFax() {}
protected:
// Returns the operation that should be done next. |sync_buffer| and |expand|
// are provided for reference. |decoder_frame_length| is the number of samples
// obtained from the last decoded frame. If there is a packet available, the
// packet header should be supplied in |packet_header|; otherwise it should
// be NULL. The mode resulting form the last call to NetEqImpl::GetAudio is
// supplied in |prev_mode|. If there is a DTMF event to play, |play_dtmf|
// should be set to true. The output variable |reset_decoder| will be set to
// true if a reset is required; otherwise it is left unchanged (i.e., it can
// remain true if it was true before the call).
Operations GetDecisionSpecialized(const SyncBuffer& sync_buffer,
const Expand& expand,
int decoder_frame_length,
const RTPHeader* packet_header,
Modes prev_mode,
bool play_dtmf,
bool* reset_decoder) override;
private:
DISALLOW_COPY_AND_ASSIGN(DecisionLogicFax);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_FAX_H_