webrtc_m130/webrtc/test/rtp_rtcp_observer.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

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