/* * 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 #include #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 observation_complete_; const rtc::scoped_ptr 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_