| /* |
| * 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 <algorithm> |
| #include <sstream> |
| #include <string> |
| |
| #include "testing/gtest/include/gtest/gtest.h" |
| |
| #include "webrtc/call.h" |
| #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h" |
| #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" |
| #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
| #include "webrtc/system_wrappers/interface/rtp_to_ntp.h" |
| #include "webrtc/system_wrappers/interface/scoped_ptr.h" |
| #include "webrtc/system_wrappers/interface/thread_annotations.h" |
| #include "webrtc/test/direct_transport.h" |
| #include "webrtc/test/encoder_settings.h" |
| #include "webrtc/test/fake_audio_device.h" |
| #include "webrtc/test/fake_decoder.h" |
| #include "webrtc/test/fake_encoder.h" |
| #include "webrtc/test/frame_generator.h" |
| #include "webrtc/test/frame_generator_capturer.h" |
| #include "webrtc/test/rtp_rtcp_observer.h" |
| #include "webrtc/test/testsupport/fileutils.h" |
| #include "webrtc/test/testsupport/perf_test.h" |
| #include "webrtc/video/transport_adapter.h" |
| #include "webrtc/voice_engine/include/voe_base.h" |
| #include "webrtc/voice_engine/include/voe_codec.h" |
| #include "webrtc/voice_engine/include/voe_network.h" |
| #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" |
| #include "webrtc/voice_engine/include/voe_video_sync.h" |
| |
| namespace webrtc { |
| |
| static unsigned int kLongTimeoutMs = 120 * 1000; |
| static const uint32_t kSendSsrc = 0x654321; |
| static const uint32_t kReceiverLocalSsrc = 0x123456; |
| static const uint8_t kSendPayloadType = 125; |
| |
| class CallPerfTest : public ::testing::Test { |
| public: |
| CallPerfTest() |
| : send_stream_(NULL), fake_encoder_(Clock::GetRealTimeClock()) {} |
| |
| protected: |
| VideoSendStream::Config GetSendTestConfig(Call* call) { |
| VideoSendStream::Config config = call->GetDefaultSendConfig(); |
| config.rtp.ssrcs.push_back(kSendSsrc); |
| config.encoder_settings = test::CreateEncoderSettings( |
| &fake_encoder_, "FAKE", kSendPayloadType, 1); |
| return config; |
| } |
| |
| void RunVideoSendTest(Call* call, |
| const VideoSendStream::Config& config, |
| test::RtpRtcpObserver* observer) { |
| send_stream_ = call->CreateVideoSendStream(config); |
| scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer( |
| test::FrameGeneratorCapturer::Create( |
| send_stream_->Input(), 320, 240, 30, Clock::GetRealTimeClock())); |
| send_stream_->Start(); |
| frame_generator_capturer->Start(); |
| |
| EXPECT_EQ(kEventSignaled, observer->Wait()); |
| |
| observer->StopSending(); |
| frame_generator_capturer->Stop(); |
| send_stream_->Stop(); |
| call->DestroyVideoSendStream(send_stream_); |
| } |
| |
| void TestMinTransmitBitrate(bool pad_to_min_bitrate); |
| |
| void TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config, |
| int threshold_ms, |
| int start_time_ms, |
| int run_time_ms); |
| |
| VideoSendStream* send_stream_; |
| test::FakeEncoder fake_encoder_; |
| }; |
| |
| class SyncRtcpObserver : public test::RtpRtcpObserver { |
| public: |
| explicit SyncRtcpObserver(const FakeNetworkPipe::Config& config) |
| : test::RtpRtcpObserver(kLongTimeoutMs, config), |
| crit_(CriticalSectionWrapper::CreateCriticalSection()) {} |
| |
| virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE { |
| RTCPUtility::RTCPParserV2 parser(packet, length, true); |
| EXPECT_TRUE(parser.IsValid()); |
| |
| for (RTCPUtility::RTCPPacketTypes packet_type = parser.Begin(); |
| packet_type != RTCPUtility::kRtcpNotValidCode; |
| packet_type = parser.Iterate()) { |
| if (packet_type == RTCPUtility::kRtcpSrCode) { |
| const RTCPUtility::RTCPPacket& packet = parser.Packet(); |
| RtcpMeasurement ntp_rtp_pair( |
| packet.SR.NTPMostSignificant, |
| packet.SR.NTPLeastSignificant, |
| packet.SR.RTPTimestamp); |
| StoreNtpRtpPair(ntp_rtp_pair); |
| } |
| } |
| return SEND_PACKET; |
| } |
| |
| int64_t RtpTimestampToNtp(uint32_t timestamp) const { |
| CriticalSectionScoped lock(crit_.get()); |
| int64_t timestamp_in_ms = -1; |
| if (ntp_rtp_pairs_.size() == 2) { |
| // TODO(stefan): We can't EXPECT_TRUE on this call due to a bug in the |
| // RTCP sender where it sends RTCP SR before any RTP packets, which leads |
| // to a bogus NTP/RTP mapping. |
| RtpToNtpMs(timestamp, ntp_rtp_pairs_, ×tamp_in_ms); |
| return timestamp_in_ms; |
| } |
| return -1; |
| } |
| |
| private: |
| void StoreNtpRtpPair(RtcpMeasurement ntp_rtp_pair) { |
| CriticalSectionScoped lock(crit_.get()); |
| for (RtcpList::iterator it = ntp_rtp_pairs_.begin(); |
| it != ntp_rtp_pairs_.end(); |
| ++it) { |
| if (ntp_rtp_pair.ntp_secs == it->ntp_secs && |
| ntp_rtp_pair.ntp_frac == it->ntp_frac) { |
| // This RTCP has already been added to the list. |
| return; |
| } |
| } |
| // We need two RTCP SR reports to map between RTP and NTP. More than two |
| // will not improve the mapping. |
| if (ntp_rtp_pairs_.size() == 2) { |
| ntp_rtp_pairs_.pop_back(); |
| } |
| ntp_rtp_pairs_.push_front(ntp_rtp_pair); |
| } |
| |
| const scoped_ptr<CriticalSectionWrapper> crit_; |
| RtcpList ntp_rtp_pairs_ GUARDED_BY(crit_); |
| }; |
| |
| class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer { |
| static const int kInSyncThresholdMs = 50; |
| static const int kStartupTimeMs = 2000; |
| static const int kMinRunTimeMs = 30000; |
| |
| public: |
| VideoRtcpAndSyncObserver(Clock* clock, |
| int voe_channel, |
| VoEVideoSync* voe_sync, |
| SyncRtcpObserver* audio_observer) |
| : SyncRtcpObserver(FakeNetworkPipe::Config()), |
| clock_(clock), |
| voe_channel_(voe_channel), |
| voe_sync_(voe_sync), |
| audio_observer_(audio_observer), |
| creation_time_ms_(clock_->TimeInMilliseconds()), |
| first_time_in_sync_(-1) {} |
| |
| virtual void RenderFrame(const I420VideoFrame& video_frame, |
| int time_to_render_ms) OVERRIDE { |
| int64_t now_ms = clock_->TimeInMilliseconds(); |
| uint32_t playout_timestamp = 0; |
| if (voe_sync_->GetPlayoutTimestamp(voe_channel_, playout_timestamp) != 0) |
| return; |
| int64_t latest_audio_ntp = |
| audio_observer_->RtpTimestampToNtp(playout_timestamp); |
| int64_t latest_video_ntp = RtpTimestampToNtp(video_frame.timestamp()); |
| if (latest_audio_ntp < 0 || latest_video_ntp < 0) |
| return; |
| int time_until_render_ms = |
| std::max(0, static_cast<int>(video_frame.render_time_ms() - now_ms)); |
| latest_video_ntp += time_until_render_ms; |
| int64_t stream_offset = latest_audio_ntp - latest_video_ntp; |
| std::stringstream ss; |
| ss << stream_offset; |
| webrtc::test::PrintResult("stream_offset", |
| "", |
| "synchronization", |
| ss.str(), |
| "ms", |
| false); |
| int64_t time_since_creation = now_ms - creation_time_ms_; |
| // During the first couple of seconds audio and video can falsely be |
| // estimated as being synchronized. We don't want to trigger on those. |
| if (time_since_creation < kStartupTimeMs) |
| return; |
| if (std::abs(latest_audio_ntp - latest_video_ntp) < kInSyncThresholdMs) { |
| if (first_time_in_sync_ == -1) { |
| first_time_in_sync_ = now_ms; |
| webrtc::test::PrintResult("sync_convergence_time", |
| "", |
| "synchronization", |
| time_since_creation, |
| "ms", |
| false); |
| } |
| if (time_since_creation > kMinRunTimeMs) |
| observation_complete_->Set(); |
| } |
| } |
| |
| private: |
| Clock* const clock_; |
| int voe_channel_; |
| VoEVideoSync* voe_sync_; |
| SyncRtcpObserver* audio_observer_; |
| int64_t creation_time_ms_; |
| int64_t first_time_in_sync_; |
| }; |
| |
| TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSync) { |
| VoiceEngine* voice_engine = VoiceEngine::Create(); |
| VoEBase* voe_base = VoEBase::GetInterface(voice_engine); |
| VoECodec* voe_codec = VoECodec::GetInterface(voice_engine); |
| VoENetwork* voe_network = VoENetwork::GetInterface(voice_engine); |
| VoEVideoSync* voe_sync = VoEVideoSync::GetInterface(voice_engine); |
| const std::string audio_filename = |
| test::ResourcePath("voice_engine/audio_long16", "pcm"); |
| ASSERT_STRNE("", audio_filename.c_str()); |
| test::FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(), |
| audio_filename); |
| EXPECT_EQ(0, voe_base->Init(&fake_audio_device, NULL)); |
| int channel = voe_base->CreateChannel(); |
| |
| FakeNetworkPipe::Config net_config; |
| net_config.queue_delay_ms = 500; |
| SyncRtcpObserver audio_observer(net_config); |
| VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock(), |
| channel, |
| voe_sync, |
| &audio_observer); |
| |
| Call::Config receiver_config(observer.ReceiveTransport()); |
| receiver_config.voice_engine = voice_engine; |
| scoped_ptr<Call> sender_call( |
| Call::Create(Call::Config(observer.SendTransport()))); |
| scoped_ptr<Call> receiver_call(Call::Create(receiver_config)); |
| CodecInst isac = {103, "ISAC", 16000, 480, 1, 32000}; |
| EXPECT_EQ(0, voe_codec->SetSendCodec(channel, isac)); |
| |
| class VoicePacketReceiver : public PacketReceiver { |
| public: |
| VoicePacketReceiver(int channel, VoENetwork* voe_network) |
| : channel_(channel), |
| voe_network_(voe_network), |
| parser_(RtpHeaderParser::Create()) {} |
| virtual DeliveryStatus DeliverPacket(const uint8_t* packet, |
| size_t length) OVERRIDE { |
| int ret; |
| if (parser_->IsRtcp(packet, static_cast<int>(length))) { |
| ret = voe_network_->ReceivedRTCPPacket( |
| channel_, packet, static_cast<unsigned int>(length)); |
| } else { |
| ret = voe_network_->ReceivedRTPPacket( |
| channel_, packet, static_cast<unsigned int>(length), PacketTime()); |
| } |
| return ret == 0 ? DELIVERY_OK : DELIVERY_PACKET_ERROR; |
| } |
| |
| private: |
| int channel_; |
| VoENetwork* voe_network_; |
| scoped_ptr<RtpHeaderParser> parser_; |
| } voe_packet_receiver(channel, voe_network); |
| |
| audio_observer.SetReceivers(&voe_packet_receiver, &voe_packet_receiver); |
| |
| internal::TransportAdapter transport_adapter(audio_observer.SendTransport()); |
| transport_adapter.Enable(); |
| EXPECT_EQ(0, |
| voe_network->RegisterExternalTransport(channel, transport_adapter)); |
| |
| observer.SetReceivers(receiver_call->Receiver(), sender_call->Receiver()); |
| |
| test::FakeDecoder fake_decoder; |
| |
| VideoSendStream::Config send_config = GetSendTestConfig(sender_call.get()); |
| |
| VideoReceiveStream::Config receive_config = |
| receiver_call->GetDefaultReceiveConfig(); |
| assert(receive_config.codecs.empty()); |
| VideoCodec codec = |
| test::CreateDecoderVideoCodec(send_config.encoder_settings); |
| receive_config.codecs.push_back(codec); |
| assert(receive_config.external_decoders.empty()); |
| ExternalVideoDecoder decoder; |
| decoder.decoder = &fake_decoder; |
| decoder.payload_type = send_config.encoder_settings.payload_type; |
| receive_config.external_decoders.push_back(decoder); |
| receive_config.rtp.remote_ssrc = send_config.rtp.ssrcs[0]; |
| receive_config.rtp.local_ssrc = kReceiverLocalSsrc; |
| receive_config.renderer = &observer; |
| receive_config.audio_channel_id = channel; |
| |
| VideoSendStream* send_stream = |
| sender_call->CreateVideoSendStream(send_config); |
| VideoReceiveStream* receive_stream = |
| receiver_call->CreateVideoReceiveStream(receive_config); |
| scoped_ptr<test::FrameGeneratorCapturer> capturer( |
| test::FrameGeneratorCapturer::Create( |
| send_stream->Input(), |
| send_config.encoder_settings.streams[0].width, |
| send_config.encoder_settings.streams[0].height, |
| 30, |
| Clock::GetRealTimeClock())); |
| receive_stream->Start(); |
| send_stream->Start(); |
| capturer->Start(); |
| |
| fake_audio_device.Start(); |
| EXPECT_EQ(0, voe_base->StartPlayout(channel)); |
| EXPECT_EQ(0, voe_base->StartReceive(channel)); |
| EXPECT_EQ(0, voe_base->StartSend(channel)); |
| |
| EXPECT_EQ(kEventSignaled, observer.Wait()) |
| << "Timed out while waiting for audio and video to be synchronized."; |
| |
| EXPECT_EQ(0, voe_base->StopSend(channel)); |
| EXPECT_EQ(0, voe_base->StopReceive(channel)); |
| EXPECT_EQ(0, voe_base->StopPlayout(channel)); |
| fake_audio_device.Stop(); |
| |
| capturer->Stop(); |
| send_stream->Stop(); |
| receive_stream->Stop(); |
| observer.StopSending(); |
| audio_observer.StopSending(); |
| |
| voe_base->DeleteChannel(channel); |
| voe_base->Release(); |
| voe_codec->Release(); |
| voe_network->Release(); |
| voe_sync->Release(); |
| sender_call->DestroyVideoSendStream(send_stream); |
| receiver_call->DestroyVideoReceiveStream(receive_stream); |
| VoiceEngine::Delete(voice_engine); |
| } |
| |
| class CaptureNtpTimeObserver : public test::RtpRtcpObserver, |
| public VideoRenderer { |
| public: |
| CaptureNtpTimeObserver(Clock* clock, |
| const FakeNetworkPipe::Config& config, |
| int threshold_ms, |
| int start_time_ms, |
| int run_time_ms) |
| : RtpRtcpObserver(kLongTimeoutMs, config), |
| clock_(clock), |
| threshold_ms_(threshold_ms), |
| start_time_ms_(start_time_ms), |
| run_time_ms_(run_time_ms), |
| creation_time_ms_(clock_->TimeInMilliseconds()), |
| capturer_(NULL), |
| rtp_start_timestamp_set_(false), |
| rtp_start_timestamp_(0) {} |
| |
| virtual void RenderFrame(const I420VideoFrame& video_frame, |
| int time_to_render_ms) OVERRIDE { |
| if (video_frame.ntp_time_ms() <= 0) { |
| // Haven't got enough RTCP SR in order to calculate the capture ntp time. |
| return; |
| } |
| |
| int64_t now_ms = clock_->TimeInMilliseconds(); |
| int64_t time_since_creation = now_ms - creation_time_ms_; |
| if (time_since_creation < start_time_ms_) { |
| // Wait for |start_time_ms_| before start measuring. |
| return; |
| } |
| |
| if (time_since_creation > run_time_ms_) { |
| observation_complete_->Set(); |
| } |
| |
| FrameCaptureTimeList::iterator iter = |
| capture_time_list_.find(video_frame.timestamp()); |
| EXPECT_TRUE(iter != capture_time_list_.end()); |
| |
| // The real capture time has been wrapped to uint32_t before converted |
| // to rtp timestamp in the sender side. So here we convert the estimated |
| // capture time to a uint32_t 90k timestamp also for comparing. |
| uint32_t estimated_capture_timestamp = |
| 90 * static_cast<uint32_t>(video_frame.ntp_time_ms()); |
| uint32_t real_capture_timestamp = iter->second; |
| int time_offset_ms = real_capture_timestamp - estimated_capture_timestamp; |
| time_offset_ms = time_offset_ms / 90; |
| std::stringstream ss; |
| ss << time_offset_ms; |
| |
| webrtc::test::PrintResult("capture_ntp_time", |
| "", |
| "real - estimated", |
| ss.str(), |
| "ms", |
| true); |
| EXPECT_TRUE(std::abs(time_offset_ms) < threshold_ms_); |
| } |
| |
| virtual Action OnSendRtp(const uint8_t* packet, size_t length) { |
| RTPHeader header; |
| EXPECT_TRUE(parser_->Parse(packet, static_cast<int>(length), &header)); |
| |
| if (!rtp_start_timestamp_set_) { |
| // Calculate the rtp timestamp offset in order to calculate the real |
| // capture time. |
| uint32_t first_capture_timestamp = |
| 90 * static_cast<uint32_t>(capturer_->first_frame_capture_time()); |
| rtp_start_timestamp_ = header.timestamp - first_capture_timestamp; |
| rtp_start_timestamp_set_ = true; |
| } |
| |
| uint32_t capture_timestamp = header.timestamp - rtp_start_timestamp_; |
| capture_time_list_.insert(capture_time_list_.end(), |
| std::make_pair(header.timestamp, |
| capture_timestamp)); |
| return SEND_PACKET; |
| } |
| |
| void SetCapturer(test::FrameGeneratorCapturer* capturer) { |
| capturer_ = capturer; |
| } |
| |
| private: |
| Clock* clock_; |
| int threshold_ms_; |
| int start_time_ms_; |
| int run_time_ms_; |
| int64_t creation_time_ms_; |
| test::FrameGeneratorCapturer* capturer_; |
| bool rtp_start_timestamp_set_; |
| uint32_t rtp_start_timestamp_; |
| typedef std::map<uint32_t, uint32_t> FrameCaptureTimeList; |
| FrameCaptureTimeList capture_time_list_; |
| }; |
| |
| void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config, |
| int threshold_ms, |
| int start_time_ms, |
| int run_time_ms) { |
| CaptureNtpTimeObserver observer(Clock::GetRealTimeClock(), |
| net_config, |
| threshold_ms, |
| start_time_ms, |
| run_time_ms); |
| |
| // Sender/receiver call. |
| Call::Config receiver_config(observer.ReceiveTransport()); |
| scoped_ptr<Call> receiver_call(Call::Create(receiver_config)); |
| scoped_ptr<Call> sender_call( |
| Call::Create(Call::Config(observer.SendTransport()))); |
| observer.SetReceivers(receiver_call->Receiver(), sender_call->Receiver()); |
| |
| // Configure send stream. |
| VideoSendStream::Config send_config = GetSendTestConfig(sender_call.get()); |
| VideoSendStream* send_stream = |
| sender_call->CreateVideoSendStream(send_config); |
| scoped_ptr<test::FrameGeneratorCapturer> capturer( |
| test::FrameGeneratorCapturer::Create( |
| send_stream->Input(), |
| send_config.encoder_settings.streams[0].width, |
| send_config.encoder_settings.streams[0].height, |
| 30, |
| Clock::GetRealTimeClock())); |
| observer.SetCapturer(capturer.get()); |
| |
| // Configure receive stream. |
| VideoReceiveStream::Config receive_config = |
| receiver_call->GetDefaultReceiveConfig(); |
| assert(receive_config.codecs.empty()); |
| VideoCodec codec = |
| test::CreateDecoderVideoCodec(send_config.encoder_settings); |
| receive_config.codecs.push_back(codec); |
| assert(receive_config.external_decoders.empty()); |
| ExternalVideoDecoder decoder; |
| test::FakeDecoder fake_decoder; |
| decoder.decoder = &fake_decoder; |
| decoder.payload_type = send_config.encoder_settings.payload_type; |
| receive_config.external_decoders.push_back(decoder); |
| receive_config.rtp.remote_ssrc = send_config.rtp.ssrcs[0]; |
| receive_config.rtp.local_ssrc = kReceiverLocalSsrc; |
| receive_config.renderer = &observer; |
| // Enable the receiver side rtt calculation. |
| receive_config.rtp.rtcp_xr.receiver_reference_time_report = true; |
| VideoReceiveStream* receive_stream = |
| receiver_call->CreateVideoReceiveStream(receive_config); |
| |
| // Start the test |
| receive_stream->Start(); |
| send_stream->Start(); |
| capturer->Start(); |
| |
| EXPECT_EQ(kEventSignaled, observer.Wait()) |
| << "Timed out while waiting for estimated capture ntp time to be " |
| << "within bounds."; |
| |
| capturer->Stop(); |
| send_stream->Stop(); |
| receive_stream->Stop(); |
| observer.StopSending(); |
| |
| sender_call->DestroyVideoSendStream(send_stream); |
| receiver_call->DestroyVideoReceiveStream(receive_stream); |
| } |
| |
| TEST_F(CallPerfTest, CaptureNtpTimeWithNetworkDelay) { |
| FakeNetworkPipe::Config net_config; |
| net_config.queue_delay_ms = 100; |
| // TODO(wu): lower the threshold as the calculation/estimatation becomes more |
| // accurate. |
| const int kThresholdMs = 30; |
| const int kStartTimeMs = 10000; |
| const int kRunTimeMs = 20000; |
| TestCaptureNtpTime(net_config, kThresholdMs, kStartTimeMs, kRunTimeMs); |
| } |
| |
| TEST_F(CallPerfTest, CaptureNtpTimeWithNetworkJitter) { |
| FakeNetworkPipe::Config net_config; |
| net_config.queue_delay_ms = 100; |
| net_config.delay_standard_deviation_ms = 10; |
| // TODO(wu): lower the threshold as the calculation/estimatation becomes more |
| // accurate. |
| const int kThresholdMs = 100; |
| const int kStartTimeMs = 10000; |
| const int kRunTimeMs = 20000; |
| TestCaptureNtpTime(net_config, kThresholdMs, kStartTimeMs, kRunTimeMs); |
| } |
| |
| TEST_F(CallPerfTest, RegisterCpuOveruseObserver) { |
| // Verifies that either a normal or overuse callback is triggered. |
| class OveruseCallbackObserver : public test::RtpRtcpObserver, |
| public webrtc::OveruseCallback { |
| public: |
| OveruseCallbackObserver() : RtpRtcpObserver(kLongTimeoutMs) {} |
| |
| virtual void OnOveruse() OVERRIDE { |
| observation_complete_->Set(); |
| } |
| virtual void OnNormalUse() OVERRIDE { |
| observation_complete_->Set(); |
| } |
| }; |
| |
| OveruseCallbackObserver observer; |
| Call::Config call_config(observer.SendTransport()); |
| call_config.overuse_callback = &observer; |
| scoped_ptr<Call> call(Call::Create(call_config)); |
| |
| VideoSendStream::Config send_config = GetSendTestConfig(call.get()); |
| RunVideoSendTest(call.get(), send_config, &observer); |
| } |
| |
| void CallPerfTest::TestMinTransmitBitrate(bool pad_to_min_bitrate) { |
| static const int kMaxEncodeBitrateKbps = 30; |
| static const int kMinTransmitBitrateBps = 150000; |
| static const int kMinAcceptableTransmitBitrate = 130; |
| static const int kMaxAcceptableTransmitBitrate = 170; |
| static const int kNumBitrateObservationsInRange = 100; |
| class BitrateObserver : public test::RtpRtcpObserver, public PacketReceiver { |
| public: |
| explicit BitrateObserver(bool using_min_transmit_bitrate) |
| : test::RtpRtcpObserver(kLongTimeoutMs), |
| send_stream_(NULL), |
| send_transport_receiver_(NULL), |
| using_min_transmit_bitrate_(using_min_transmit_bitrate), |
| num_bitrate_observations_in_range_(0) {} |
| |
| virtual void SetReceivers(PacketReceiver* send_transport_receiver, |
| PacketReceiver* receive_transport_receiver) |
| OVERRIDE { |
| send_transport_receiver_ = send_transport_receiver; |
| test::RtpRtcpObserver::SetReceivers(this, receive_transport_receiver); |
| } |
| |
| void SetSendStream(VideoSendStream* send_stream) { |
| send_stream_ = send_stream; |
| } |
| |
| private: |
| virtual DeliveryStatus DeliverPacket(const uint8_t* packet, |
| size_t length) OVERRIDE { |
| VideoSendStream::Stats stats = send_stream_->GetStats(); |
| if (stats.substreams.size() > 0) { |
| assert(stats.substreams.size() == 1); |
| int bitrate_kbps = stats.substreams.begin()->second.bitrate_bps / 1000; |
| if (bitrate_kbps > 0) { |
| test::PrintResult( |
| "bitrate_stats_", |
| (using_min_transmit_bitrate_ ? "min_transmit_bitrate" |
| : "without_min_transmit_bitrate"), |
| "bitrate_kbps", |
| static_cast<size_t>(bitrate_kbps), |
| "kbps", |
| false); |
| if (using_min_transmit_bitrate_) { |
| if (bitrate_kbps > kMinAcceptableTransmitBitrate && |
| bitrate_kbps < kMaxAcceptableTransmitBitrate) { |
| ++num_bitrate_observations_in_range_; |
| } |
| } else { |
| // Expect bitrate stats to roughly match the max encode bitrate. |
| if (bitrate_kbps > kMaxEncodeBitrateKbps - 5 && |
| bitrate_kbps < kMaxEncodeBitrateKbps + 5) { |
| ++num_bitrate_observations_in_range_; |
| } |
| } |
| if (num_bitrate_observations_in_range_ == |
| kNumBitrateObservationsInRange) |
| observation_complete_->Set(); |
| } |
| } |
| return send_transport_receiver_->DeliverPacket(packet, length); |
| } |
| |
| VideoSendStream* send_stream_; |
| PacketReceiver* send_transport_receiver_; |
| const bool using_min_transmit_bitrate_; |
| int num_bitrate_observations_in_range_; |
| } observer(pad_to_min_bitrate); |
| |
| scoped_ptr<Call> sender_call( |
| Call::Create(Call::Config(observer.SendTransport()))); |
| scoped_ptr<Call> receiver_call( |
| Call::Create(Call::Config(observer.ReceiveTransport()))); |
| |
| VideoSendStream::Config send_config = GetSendTestConfig(sender_call.get()); |
| fake_encoder_.SetMaxBitrate(kMaxEncodeBitrateKbps); |
| |
| observer.SetReceivers(receiver_call->Receiver(), sender_call->Receiver()); |
| |
| send_config.pacing = true; |
| if (pad_to_min_bitrate) { |
| send_config.rtp.min_transmit_bitrate_bps = kMinTransmitBitrateBps; |
| } else { |
| assert(send_config.rtp.min_transmit_bitrate_bps == 0); |
| } |
| |
| VideoReceiveStream::Config receive_config = |
| receiver_call->GetDefaultReceiveConfig(); |
| receive_config.codecs.clear(); |
| VideoCodec codec = |
| test::CreateDecoderVideoCodec(send_config.encoder_settings); |
| receive_config.codecs.push_back(codec); |
| test::FakeDecoder fake_decoder; |
| ExternalVideoDecoder decoder; |
| decoder.decoder = &fake_decoder; |
| decoder.payload_type = send_config.encoder_settings.payload_type; |
| receive_config.external_decoders.push_back(decoder); |
| receive_config.rtp.remote_ssrc = send_config.rtp.ssrcs[0]; |
| receive_config.rtp.local_ssrc = kReceiverLocalSsrc; |
| |
| VideoSendStream* send_stream = |
| sender_call->CreateVideoSendStream(send_config); |
| VideoReceiveStream* receive_stream = |
| receiver_call->CreateVideoReceiveStream(receive_config); |
| scoped_ptr<test::FrameGeneratorCapturer> capturer( |
| test::FrameGeneratorCapturer::Create( |
| send_stream->Input(), |
| send_config.encoder_settings.streams[0].width, |
| send_config.encoder_settings.streams[0].height, |
| 30, |
| Clock::GetRealTimeClock())); |
| observer.SetSendStream(send_stream); |
| receive_stream->Start(); |
| send_stream->Start(); |
| capturer->Start(); |
| |
| EXPECT_EQ(kEventSignaled, observer.Wait()) |
| << "Timeout while waiting for send-bitrate stats."; |
| |
| send_stream->Stop(); |
| receive_stream->Stop(); |
| observer.StopSending(); |
| capturer->Stop(); |
| sender_call->DestroyVideoSendStream(send_stream); |
| receiver_call->DestroyVideoReceiveStream(receive_stream); |
| } |
| |
| TEST_F(CallPerfTest, PadsToMinTransmitBitrate) { TestMinTransmitBitrate(true); } |
| |
| TEST_F(CallPerfTest, NoPadWithoutMinTransmitBitrate) { |
| TestMinTransmitBitrate(false); |
| } |
| |
| } // namespace webrtc |
