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
184 lines
5.9 KiB
C++
184 lines
5.9 KiB
C++
/*
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* Copyright (c) 2013 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_VIDEO_ENGINE_TEST_COMMON_RTP_RTCP_OBSERVER_H_
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#define WEBRTC_VIDEO_ENGINE_TEST_COMMON_RTP_RTCP_OBSERVER_H_
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#include <map>
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#include <vector>
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
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#include "webrtc/test/direct_transport.h"
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#include "webrtc/typedefs.h"
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#include "webrtc/video_send_stream.h"
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namespace webrtc {
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namespace test {
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class RtpRtcpObserver {
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public:
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virtual ~RtpRtcpObserver() {}
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newapi::Transport* SendTransport() {
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return &send_transport_;
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}
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newapi::Transport* ReceiveTransport() {
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return &receive_transport_;
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}
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virtual void SetReceivers(PacketReceiver* send_transport_receiver,
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PacketReceiver* receive_transport_receiver) {
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send_transport_.SetReceiver(send_transport_receiver);
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receive_transport_.SetReceiver(receive_transport_receiver);
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}
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void StopSending() {
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send_transport_.StopSending();
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receive_transport_.StopSending();
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}
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virtual EventTypeWrapper Wait() {
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EventTypeWrapper result = observation_complete_->Wait(timeout_ms_);
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return result;
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}
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protected:
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RtpRtcpObserver(unsigned int event_timeout_ms,
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const FakeNetworkPipe::Config& configuration)
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: crit_(CriticalSectionWrapper::CreateCriticalSection()),
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observation_complete_(EventWrapper::Create()),
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parser_(RtpHeaderParser::Create()),
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send_transport_(crit_.get(),
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this,
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&RtpRtcpObserver::OnSendRtp,
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&RtpRtcpObserver::OnSendRtcp,
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configuration),
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receive_transport_(crit_.get(),
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this,
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&RtpRtcpObserver::OnReceiveRtp,
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&RtpRtcpObserver::OnReceiveRtcp,
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configuration),
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timeout_ms_(event_timeout_ms) {}
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explicit RtpRtcpObserver(unsigned int event_timeout_ms)
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: crit_(CriticalSectionWrapper::CreateCriticalSection()),
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observation_complete_(EventWrapper::Create()),
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parser_(RtpHeaderParser::Create()),
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send_transport_(crit_.get(),
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this,
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&RtpRtcpObserver::OnSendRtp,
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&RtpRtcpObserver::OnSendRtcp,
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FakeNetworkPipe::Config()),
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receive_transport_(crit_.get(),
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this,
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&RtpRtcpObserver::OnReceiveRtp,
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&RtpRtcpObserver::OnReceiveRtcp,
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FakeNetworkPipe::Config()),
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timeout_ms_(event_timeout_ms) {}
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enum Action {
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SEND_PACKET,
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DROP_PACKET,
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};
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virtual Action OnSendRtp(const uint8_t* packet, size_t length)
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EXCLUSIVE_LOCKS_REQUIRED(crit_) {
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return SEND_PACKET;
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}
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virtual Action OnSendRtcp(const uint8_t* packet, size_t length)
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EXCLUSIVE_LOCKS_REQUIRED(crit_) {
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return SEND_PACKET;
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}
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virtual Action OnReceiveRtp(const uint8_t* packet, size_t length)
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EXCLUSIVE_LOCKS_REQUIRED(crit_) {
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return SEND_PACKET;
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}
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virtual Action OnReceiveRtcp(const uint8_t* packet, size_t length)
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EXCLUSIVE_LOCKS_REQUIRED(crit_) {
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return SEND_PACKET;
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}
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private:
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class PacketTransport : public test::DirectTransport {
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public:
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typedef Action (RtpRtcpObserver::*PacketTransportAction)(const uint8_t*,
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size_t);
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PacketTransport(CriticalSectionWrapper* lock,
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RtpRtcpObserver* observer,
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PacketTransportAction on_rtp,
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PacketTransportAction on_rtcp,
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const FakeNetworkPipe::Config& configuration)
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: test::DirectTransport(configuration),
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crit_(lock),
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observer_(observer),
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on_rtp_(on_rtp),
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on_rtcp_(on_rtcp) {}
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private:
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bool SendRtp(const uint8_t* packet, size_t length) override {
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EXPECT_FALSE(RtpHeaderParser::IsRtcp(packet, length));
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Action action;
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{
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CriticalSectionScoped lock(crit_);
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action = (observer_->*on_rtp_)(packet, length);
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}
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switch (action) {
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case DROP_PACKET:
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// Drop packet silently.
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return true;
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case SEND_PACKET:
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return test::DirectTransport::SendRtp(packet, length);
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}
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return true; // Will never happen, makes compiler happy.
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}
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bool SendRtcp(const uint8_t* packet, size_t length) override {
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EXPECT_TRUE(RtpHeaderParser::IsRtcp(packet, length));
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Action action;
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{
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CriticalSectionScoped lock(crit_);
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action = (observer_->*on_rtcp_)(packet, length);
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}
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switch (action) {
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case DROP_PACKET:
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// Drop packet silently.
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return true;
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case SEND_PACKET:
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return test::DirectTransport::SendRtcp(packet, length);
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}
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return true; // Will never happen, makes compiler happy.
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}
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// Pointer to shared lock instance protecting on_rtp_/on_rtcp_ calls.
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CriticalSectionWrapper* const crit_;
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RtpRtcpObserver* const observer_;
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const PacketTransportAction on_rtp_, on_rtcp_;
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};
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protected:
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const rtc::scoped_ptr<CriticalSectionWrapper> crit_;
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const rtc::scoped_ptr<EventWrapper> observation_complete_;
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const rtc::scoped_ptr<RtpHeaderParser> parser_;
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private:
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PacketTransport send_transport_, receive_transport_;
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unsigned int timeout_ms_;
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};
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} // namespace test
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} // namespace webrtc
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#endif // WEBRTC_VIDEO_ENGINE_TEST_COMMON_RTP_RTCP_OBSERVER_H_
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