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/*
* 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/interface/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_
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