webrtc_m130/webrtc/test/rtp_rtcp_observer.h
Henrik Kjellander ff761fba82 modules: more interface -> include renames
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface

To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
                "use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)

BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc

R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1417683006 .

Cr-Commit-Position: refs/heads/master@{#10500}
2015-11-04 07:32:04 +00:00

142 lines
4.2 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/base/criticalsection.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/test/constants.h"
#include "webrtc/test/direct_transport.h"
#include "webrtc/typedefs.h"
#include "webrtc/video_send_stream.h"
namespace webrtc {
namespace test {
class PacketTransport;
class RtpRtcpObserver {
public:
enum Action {
SEND_PACKET,
DROP_PACKET,
};
virtual ~RtpRtcpObserver() {}
virtual EventTypeWrapper Wait() {
EventTypeWrapper result = observation_complete_->Wait(timeout_ms_);
return result;
}
virtual Action OnSendRtp(const uint8_t* packet, size_t length) {
return SEND_PACKET;
}
virtual Action OnSendRtcp(const uint8_t* packet, size_t length) {
return SEND_PACKET;
}
virtual Action OnReceiveRtp(const uint8_t* packet, size_t length) {
return SEND_PACKET;
}
virtual Action OnReceiveRtcp(const uint8_t* packet, size_t length) {
return SEND_PACKET;
}
protected:
explicit RtpRtcpObserver(unsigned int event_timeout_ms)
: observation_complete_(EventWrapper::Create()),
parser_(RtpHeaderParser::Create()),
timeout_ms_(event_timeout_ms) {
parser_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
kTOffsetExtensionId);
parser_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
kAbsSendTimeExtensionId);
parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId);
}
const rtc::scoped_ptr<EventWrapper> observation_complete_;
const rtc::scoped_ptr<RtpHeaderParser> parser_;
private:
unsigned int timeout_ms_;
};
class PacketTransport : public test::DirectTransport {
public:
enum TransportType { kReceiver, kSender };
PacketTransport(Call* send_call,
RtpRtcpObserver* observer,
TransportType transport_type,
const FakeNetworkPipe::Config& configuration)
: test::DirectTransport(configuration, send_call),
observer_(observer),
transport_type_(transport_type) {}
private:
bool SendRtp(const uint8_t* packet,
size_t length,
const PacketOptions& options) override {
EXPECT_FALSE(RtpHeaderParser::IsRtcp(packet, length));
RtpRtcpObserver::Action action;
{
if (transport_type_ == kSender) {
action = observer_->OnSendRtp(packet, length);
} else {
action = observer_->OnReceiveRtp(packet, length);
}
}
switch (action) {
case RtpRtcpObserver::DROP_PACKET:
// Drop packet silently.
return true;
case RtpRtcpObserver::SEND_PACKET:
return test::DirectTransport::SendRtp(packet, length, options);
}
return true; // Will never happen, makes compiler happy.
}
bool SendRtcp(const uint8_t* packet, size_t length) override {
EXPECT_TRUE(RtpHeaderParser::IsRtcp(packet, length));
RtpRtcpObserver::Action action;
{
if (transport_type_ == kSender) {
action = observer_->OnSendRtcp(packet, length);
} else {
action = observer_->OnReceiveRtcp(packet, length);
}
}
switch (action) {
case RtpRtcpObserver::DROP_PACKET:
// Drop packet silently.
return true;
case RtpRtcpObserver::SEND_PACKET:
return test::DirectTransport::SendRtcp(packet, length);
}
return true; // Will never happen, makes compiler happy.
}
RtpRtcpObserver* const observer_;
TransportType transport_type_;
};
} // namespace test
} // namespace webrtc
#endif // WEBRTC_VIDEO_ENGINE_TEST_COMMON_RTP_RTCP_OBSERVER_H_