141 lines
4.1 KiB
C++
141 lines
4.1 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_TEST_RTP_RTCP_OBSERVER_H_
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#define WEBRTC_TEST_RTP_RTCP_OBSERVER_H_
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#include <map>
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#include <memory>
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#include <vector>
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/base/event.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
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#include "webrtc/test/constants.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 PacketTransport;
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class RtpRtcpObserver {
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public:
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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 ~RtpRtcpObserver() {}
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virtual bool Wait() { return observation_complete_.Wait(timeout_ms_); }
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virtual Action OnSendRtp(const uint8_t* packet, size_t length) {
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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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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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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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return SEND_PACKET;
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}
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protected:
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explicit RtpRtcpObserver(int event_timeout_ms)
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: observation_complete_(false, false),
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parser_(RtpHeaderParser::Create()),
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timeout_ms_(event_timeout_ms) {
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parser_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
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kTOffsetExtensionId);
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parser_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
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kAbsSendTimeExtensionId);
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parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
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kTransportSequenceNumberExtensionId);
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}
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rtc::Event observation_complete_;
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const std::unique_ptr<RtpHeaderParser> parser_;
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private:
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const int timeout_ms_;
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};
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class PacketTransport : public test::DirectTransport {
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public:
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enum TransportType { kReceiver, kSender };
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PacketTransport(Call* send_call,
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RtpRtcpObserver* observer,
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TransportType transport_type,
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const FakeNetworkPipe::Config& configuration)
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: test::DirectTransport(configuration, send_call),
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observer_(observer),
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transport_type_(transport_type) {}
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private:
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bool SendRtp(const uint8_t* packet,
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size_t length,
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const PacketOptions& options) override {
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EXPECT_FALSE(RtpHeaderParser::IsRtcp(packet, length));
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RtpRtcpObserver::Action action;
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{
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if (transport_type_ == kSender) {
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action = observer_->OnSendRtp(packet, length);
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} else {
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action = observer_->OnReceiveRtp(packet, length);
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}
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}
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switch (action) {
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case RtpRtcpObserver::DROP_PACKET:
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// Drop packet silently.
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return true;
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case RtpRtcpObserver::SEND_PACKET:
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return test::DirectTransport::SendRtp(packet, length, options);
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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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RtpRtcpObserver::Action action;
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{
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if (transport_type_ == kSender) {
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action = observer_->OnSendRtcp(packet, length);
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} else {
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action = observer_->OnReceiveRtcp(packet, length);
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}
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}
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switch (action) {
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case RtpRtcpObserver::DROP_PACKET:
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// Drop packet silently.
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return true;
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case RtpRtcpObserver::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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RtpRtcpObserver* const observer_;
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TransportType transport_type_;
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};
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} // namespace test
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} // namespace webrtc
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#endif // WEBRTC_TEST_RTP_RTCP_OBSERVER_H_
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