14665ff7d4
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
68 lines
2.3 KiB
C++
68 lines
2.3 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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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/common_types.h"
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#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
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namespace webrtc {
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// This class sends all its packet straight to the provided RtpRtcp module.
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// with optional packet loss.
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class LoopBackTransport : public webrtc::Transport {
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public:
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LoopBackTransport()
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: count_(0),
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packet_loss_(0),
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rtp_payload_registry_(NULL),
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rtp_receiver_(NULL),
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rtp_rtcp_module_(NULL) {}
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void SetSendModule(RtpRtcp* rtp_rtcp_module,
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RTPPayloadRegistry* payload_registry,
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RtpReceiver* receiver,
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ReceiveStatistics* receive_statistics);
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void DropEveryNthPacket(int n);
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int SendPacket(int channel, const void* data, size_t len) override;
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int SendRTCPPacket(int channel, const void* data, size_t len) override;
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private:
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int count_;
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int packet_loss_;
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ReceiveStatistics* receive_statistics_;
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RTPPayloadRegistry* rtp_payload_registry_;
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RtpReceiver* rtp_receiver_;
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RtpRtcp* rtp_rtcp_module_;
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};
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class TestRtpReceiver : public NullRtpData {
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public:
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int32_t OnReceivedPayloadData(
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const uint8_t* payload_data,
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const size_t payload_size,
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const webrtc::WebRtcRTPHeader* rtp_header) override;
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const uint8_t* payload_data() const { return payload_data_; }
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size_t payload_size() const { return payload_size_; }
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webrtc::WebRtcRTPHeader rtp_header() const { return rtp_header_; }
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private:
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uint8_t payload_data_[1500];
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size_t payload_size_;
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webrtc::WebRtcRTPHeader rtp_header_;
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};
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} // namespace webrtc
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