| /* |
| * 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. |
| */ |
| #include <assert.h> |
| |
| #include <map> |
| |
| #include "testing/gtest/include/gtest/gtest.h" |
| |
| #include "webrtc/call.h" |
| #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" |
| #include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h" |
| #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" |
| #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" |
| #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
| #include "webrtc/system_wrappers/interface/event_wrapper.h" |
| #include "webrtc/system_wrappers/interface/scoped_ptr.h" |
| #include "webrtc/test/direct_transport.h" |
| #include "webrtc/test/fake_decoder.h" |
| #include "webrtc/test/fake_encoder.h" |
| #include "webrtc/test/frame_generator_capturer.h" |
| #include "webrtc/test/generate_ssrcs.h" |
| |
| namespace webrtc { |
| |
| namespace { |
| static const int kTOffsetExtensionId = 7; |
| } |
| |
| class StreamObserver : public newapi::Transport, public RemoteBitrateObserver { |
| public: |
| typedef std::map<uint32_t, int> BytesSentMap; |
| StreamObserver(int num_expected_ssrcs, |
| newapi::Transport* feedback_transport, |
| Clock* clock) |
| : critical_section_(CriticalSectionWrapper::CreateCriticalSection()), |
| all_ssrcs_sent_(EventWrapper::Create()), |
| rtp_parser_(RtpHeaderParser::Create()), |
| feedback_transport_(new TransportWrapper(feedback_transport)), |
| receive_stats_(ReceiveStatistics::Create(clock)), |
| clock_(clock), |
| num_expected_ssrcs_(num_expected_ssrcs) { |
| // Ideally we would only have to instantiate an RtcpSender, an |
| // RtpHeaderParser and a RemoteBitrateEstimator here, but due to the current |
| // state of the RTP module we need a full module and receive statistics to |
| // be able to produce an RTCP with REMB. |
| RtpRtcp::Configuration config; |
| config.receive_statistics = receive_stats_.get(); |
| config.outgoing_transport = feedback_transport_.get(); |
| rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config)); |
| rtp_rtcp_->SetREMBStatus(true); |
| rtp_rtcp_->SetRTCPStatus(kRtcpNonCompound); |
| rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset, |
| kTOffsetExtensionId); |
| AbsoluteSendTimeRemoteBitrateEstimatorFactory rbe_factory; |
| remote_bitrate_estimator_.reset(rbe_factory.Create(this, clock)); |
| } |
| |
| virtual void OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs, |
| unsigned int bitrate) { |
| CriticalSectionScoped lock(critical_section_.get()); |
| if (ssrcs.size() == num_expected_ssrcs_ && bitrate >= kExpectedBitrateBps) |
| all_ssrcs_sent_->Set(); |
| rtp_rtcp_->SetREMBData( |
| bitrate, static_cast<uint8_t>(ssrcs.size()), &ssrcs[0]); |
| rtp_rtcp_->Process(); |
| } |
| |
| virtual bool SendRTP(const uint8_t* packet, size_t length) OVERRIDE { |
| CriticalSectionScoped lock(critical_section_.get()); |
| RTPHeader header; |
| EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast<int>(length), &header)); |
| receive_stats_->IncomingPacket(header, length, false); |
| rtp_rtcp_->SetRemoteSSRC(header.ssrc); |
| remote_bitrate_estimator_->IncomingPacket( |
| clock_->TimeInMilliseconds(), static_cast<int>(length - 12), header); |
| if (remote_bitrate_estimator_->TimeUntilNextProcess() <= 0) { |
| remote_bitrate_estimator_->Process(); |
| } |
| return true; |
| } |
| |
| virtual bool SendRTCP(const uint8_t* packet, size_t length) OVERRIDE { |
| return true; |
| } |
| |
| EventTypeWrapper Wait() { return all_ssrcs_sent_->Wait(120 * 1000); } |
| |
| private: |
| class TransportWrapper : public webrtc::Transport { |
| public: |
| explicit TransportWrapper(newapi::Transport* new_transport) |
| : new_transport_(new_transport) {} |
| |
| virtual int SendPacket(int channel, const void* data, int len) OVERRIDE { |
| return new_transport_->SendRTP(static_cast<const uint8_t*>(data), len) |
| ? len |
| : -1; |
| } |
| |
| virtual int SendRTCPPacket(int channel, |
| const void* data, |
| int len) OVERRIDE { |
| return new_transport_->SendRTCP(static_cast<const uint8_t*>(data), len) |
| ? len |
| : -1; |
| } |
| |
| private: |
| newapi::Transport* new_transport_; |
| }; |
| |
| static const unsigned int kExpectedBitrateBps = 1200000; |
| |
| scoped_ptr<CriticalSectionWrapper> critical_section_; |
| scoped_ptr<EventWrapper> all_ssrcs_sent_; |
| scoped_ptr<RtpHeaderParser> rtp_parser_; |
| scoped_ptr<RtpRtcp> rtp_rtcp_; |
| scoped_ptr<TransportWrapper> feedback_transport_; |
| scoped_ptr<ReceiveStatistics> receive_stats_; |
| scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_; |
| Clock* clock_; |
| const size_t num_expected_ssrcs_; |
| }; |
| |
| class RampUpTest : public ::testing::TestWithParam<bool> { |
| public: |
| virtual void SetUp() { reserved_ssrcs_.clear(); } |
| |
| protected: |
| std::map<uint32_t, bool> reserved_ssrcs_; |
| }; |
| |
| TEST_P(RampUpTest, RampUpWithPadding) { |
| test::DirectTransport receiver_transport; |
| StreamObserver stream_observer( |
| 3, &receiver_transport, Clock::GetRealTimeClock()); |
| Call::Config call_config(&stream_observer); |
| scoped_ptr<Call> call(Call::Create(call_config)); |
| VideoSendStream::Config send_config = call->GetDefaultSendConfig(); |
| |
| receiver_transport.SetReceiver(call->Receiver()); |
| |
| test::FakeEncoder encoder(Clock::GetRealTimeClock()); |
| send_config.encoder = &encoder; |
| send_config.internal_source = false; |
| test::FakeEncoder::SetCodecSettings(&send_config.codec, 3); |
| send_config.codec.plType = 125; |
| send_config.pacing = GetParam(); |
| send_config.rtp.extensions.push_back( |
| RtpExtension("toffset", kTOffsetExtensionId)); |
| |
| test::GenerateRandomSsrcs(&send_config, &reserved_ssrcs_); |
| |
| VideoSendStream* send_stream = call->CreateSendStream(send_config); |
| |
| VideoReceiveStream::Config receive_config; |
| receive_config.rtp.ssrc = send_config.rtp.ssrcs[0]; |
| receive_config.rtp.nack.rtp_history_ms = send_config.rtp.nack.rtp_history_ms; |
| VideoReceiveStream* receive_stream = |
| call->CreateReceiveStream(receive_config); |
| |
| scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer( |
| test::FrameGeneratorCapturer::Create(send_stream->Input(), |
| send_config.codec.width, |
| send_config.codec.height, |
| 30, |
| Clock::GetRealTimeClock())); |
| |
| receive_stream->StartReceive(); |
| send_stream->StartSend(); |
| frame_generator_capturer->Start(); |
| |
| EXPECT_EQ(kEventSignaled, stream_observer.Wait()); |
| |
| frame_generator_capturer->Stop(); |
| send_stream->StopSend(); |
| receive_stream->StopReceive(); |
| |
| call->DestroyReceiveStream(receive_stream); |
| call->DestroySendStream(send_stream); |
| } |
| |
| INSTANTIATE_TEST_CASE_P(RampUpTest, RampUpTest, ::testing::Bool()); |
| |
| } // namespace webrtc |