| /* |
| * Copyright (c) 2004 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 "media/engine/webrtc_video_engine.h" |
| |
| #include <algorithm> |
| #include <map> |
| #include <memory> |
| #include <string> |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/algorithm/container.h" |
| #include "absl/memory/memory.h" |
| #include "absl/strings/match.h" |
| #include "api/rtc_event_log/rtc_event_log.h" |
| #include "api/rtp_parameters.h" |
| #include "api/task_queue/default_task_queue_factory.h" |
| #include "api/test/mock_video_bitrate_allocator.h" |
| #include "api/test/mock_video_bitrate_allocator_factory.h" |
| #include "api/test/mock_video_decoder_factory.h" |
| #include "api/test/mock_video_encoder_factory.h" |
| #include "api/test/video/function_video_decoder_factory.h" |
| #include "api/transport/field_trial_based_config.h" |
| #include "api/units/time_delta.h" |
| #include "api/video/builtin_video_bitrate_allocator_factory.h" |
| #include "api/video/i420_buffer.h" |
| #include "api/video/video_bitrate_allocation.h" |
| #include "api/video_codecs/builtin_video_decoder_factory.h" |
| #include "api/video_codecs/builtin_video_encoder_factory.h" |
| #include "api/video_codecs/h264_profile_level_id.h" |
| #include "api/video_codecs/sdp_video_format.h" |
| #include "api/video_codecs/video_decoder_factory.h" |
| #include "api/video_codecs/video_encoder.h" |
| #include "api/video_codecs/video_encoder_factory.h" |
| #include "call/flexfec_receive_stream.h" |
| #include "media/base/fake_frame_source.h" |
| #include "media/base/fake_network_interface.h" |
| #include "media/base/fake_video_renderer.h" |
| #include "media/base/media_constants.h" |
| #include "media/base/rtp_utils.h" |
| #include "media/base/test_utils.h" |
| #include "media/engine/fake_webrtc_call.h" |
| #include "media/engine/fake_webrtc_video_engine.h" |
| #include "media/engine/simulcast.h" |
| #include "media/engine/webrtc_voice_engine.h" |
| #include "modules/rtp_rtcp/source/rtp_packet.h" |
| #include "rtc_base/arraysize.h" |
| #include "rtc_base/event.h" |
| #include "rtc_base/experiments/min_video_bitrate_experiment.h" |
| #include "rtc_base/fake_clock.h" |
| #include "rtc_base/gunit.h" |
| #include "rtc_base/numerics/safe_conversions.h" |
| #include "rtc_base/time_utils.h" |
| #include "system_wrappers/include/field_trial.h" |
| #include "test/fake_decoder.h" |
| #include "test/field_trial.h" |
| #include "test/frame_forwarder.h" |
| #include "test/gmock.h" |
| |
| using ::testing::_; |
| using ::testing::Contains; |
| using ::testing::Each; |
| using ::testing::ElementsAreArray; |
| using ::testing::Eq; |
| using ::testing::Field; |
| using ::testing::IsEmpty; |
| using ::testing::Pair; |
| using ::testing::Return; |
| using ::testing::SizeIs; |
| using ::testing::StrNe; |
| using ::testing::Values; |
| using ::webrtc::BitrateConstraints; |
| using ::webrtc::RtpExtension; |
| using ::webrtc::RtpPacket; |
| |
| namespace { |
| static const int kDefaultQpMax = 56; |
| |
| static const uint8_t kRedRtxPayloadType = 125; |
| |
| static const uint32_t kTimeout = 5000U; |
| static const uint32_t kSsrc = 1234u; |
| static const uint32_t kSsrcs4[] = {1, 2, 3, 4}; |
| static const int kVideoWidth = 640; |
| static const int kVideoHeight = 360; |
| static const int kFramerate = 30; |
| |
| static const uint32_t kSsrcs1[] = {1}; |
| static const uint32_t kSsrcs3[] = {1, 2, 3}; |
| static const uint32_t kRtxSsrcs1[] = {4}; |
| static const uint32_t kFlexfecSsrc = 5; |
| static const uint32_t kIncomingUnsignalledSsrc = 0xC0FFEE; |
| static const int64_t kUnsignalledReceiveStreamCooldownMs = 500; |
| |
| constexpr uint32_t kRtpHeaderSize = 12; |
| |
| static const char kUnsupportedExtensionName[] = |
| "urn:ietf:params:rtp-hdrext:unsupported"; |
| |
| cricket::VideoCodec RemoveFeedbackParams(cricket::VideoCodec&& codec) { |
| codec.feedback_params = cricket::FeedbackParams(); |
| return std::move(codec); |
| } |
| |
| void VerifyCodecHasDefaultFeedbackParams(const cricket::VideoCodec& codec, |
| bool lntf_expected) { |
| EXPECT_EQ(lntf_expected, |
| codec.HasFeedbackParam(cricket::FeedbackParam( |
| cricket::kRtcpFbParamLntf, cricket::kParamValueEmpty))); |
| EXPECT_TRUE(codec.HasFeedbackParam(cricket::FeedbackParam( |
| cricket::kRtcpFbParamNack, cricket::kParamValueEmpty))); |
| EXPECT_TRUE(codec.HasFeedbackParam(cricket::FeedbackParam( |
| cricket::kRtcpFbParamNack, cricket::kRtcpFbNackParamPli))); |
| EXPECT_TRUE(codec.HasFeedbackParam(cricket::FeedbackParam( |
| cricket::kRtcpFbParamRemb, cricket::kParamValueEmpty))); |
| EXPECT_TRUE(codec.HasFeedbackParam(cricket::FeedbackParam( |
| cricket::kRtcpFbParamTransportCc, cricket::kParamValueEmpty))); |
| EXPECT_TRUE(codec.HasFeedbackParam(cricket::FeedbackParam( |
| cricket::kRtcpFbParamCcm, cricket::kRtcpFbCcmParamFir))); |
| } |
| |
| // Return true if any codec in `codecs` is an RTX codec with associated payload |
| // type `payload_type`. |
| bool HasRtxCodec(const std::vector<cricket::VideoCodec>& codecs, |
| int payload_type) { |
| for (const cricket::VideoCodec& codec : codecs) { |
| int associated_payload_type; |
| if (absl::EqualsIgnoreCase(codec.name.c_str(), "rtx") && |
| codec.GetParam(cricket::kCodecParamAssociatedPayloadType, |
| &associated_payload_type) && |
| associated_payload_type == payload_type) { |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| // TODO(nisse): Duplicated in call.cc. |
| const int* FindKeyByValue(const std::map<int, int>& m, int v) { |
| for (const auto& kv : m) { |
| if (kv.second == v) |
| return &kv.first; |
| } |
| return nullptr; |
| } |
| |
| bool HasRtxReceiveAssociation(const webrtc::VideoReceiveStream::Config& config, |
| int payload_type) { |
| return FindKeyByValue(config.rtp.rtx_associated_payload_types, |
| payload_type) != nullptr; |
| } |
| |
| // Check that there's an Rtx payload type for each decoder. |
| bool VerifyRtxReceiveAssociations( |
| const webrtc::VideoReceiveStream::Config& config) { |
| for (const auto& decoder : config.decoders) { |
| if (!HasRtxReceiveAssociation(config, decoder.payload_type)) |
| return false; |
| } |
| return true; |
| } |
| |
| rtc::scoped_refptr<webrtc::VideoFrameBuffer> CreateBlackFrameBuffer( |
| int width, |
| int height) { |
| rtc::scoped_refptr<webrtc::I420Buffer> buffer = |
| webrtc::I420Buffer::Create(width, height); |
| webrtc::I420Buffer::SetBlack(buffer); |
| return buffer; |
| } |
| |
| void VerifySendStreamHasRtxTypes(const webrtc::VideoSendStream::Config& config, |
| const std::map<int, int>& rtx_types) { |
| std::map<int, int>::const_iterator it; |
| it = rtx_types.find(config.rtp.payload_type); |
| EXPECT_TRUE(it != rtx_types.end() && |
| it->second == config.rtp.rtx.payload_type); |
| |
| if (config.rtp.ulpfec.red_rtx_payload_type != -1) { |
| it = rtx_types.find(config.rtp.ulpfec.red_payload_type); |
| EXPECT_TRUE(it != rtx_types.end() && |
| it->second == config.rtp.ulpfec.red_rtx_payload_type); |
| } |
| } |
| |
| cricket::MediaConfig GetMediaConfig() { |
| cricket::MediaConfig media_config; |
| media_config.video.enable_cpu_adaptation = false; |
| return media_config; |
| } |
| |
| // Values from GetMaxDefaultVideoBitrateKbps in webrtcvideoengine.cc. |
| int GetMaxDefaultBitrateBps(size_t width, size_t height) { |
| if (width * height <= 320 * 240) { |
| return 600000; |
| } else if (width * height <= 640 * 480) { |
| return 1700000; |
| } else if (width * height <= 960 * 540) { |
| return 2000000; |
| } else { |
| return 2500000; |
| } |
| } |
| |
| class MockVideoSource : public rtc::VideoSourceInterface<webrtc::VideoFrame> { |
| public: |
| MOCK_METHOD(void, |
| AddOrUpdateSink, |
| (rtc::VideoSinkInterface<webrtc::VideoFrame> * sink, |
| const rtc::VideoSinkWants& wants), |
| (override)); |
| MOCK_METHOD(void, |
| RemoveSink, |
| (rtc::VideoSinkInterface<webrtc::VideoFrame> * sink), |
| (override)); |
| }; |
| |
| } // namespace |
| |
| #define EXPECT_FRAME_WAIT(c, w, h, t) \ |
| EXPECT_EQ_WAIT((c), renderer_.num_rendered_frames(), (t)); \ |
| EXPECT_EQ((w), renderer_.width()); \ |
| EXPECT_EQ((h), renderer_.height()); \ |
| EXPECT_EQ(0, renderer_.errors()); |
| |
| #define EXPECT_FRAME_ON_RENDERER_WAIT(r, c, w, h, t) \ |
| EXPECT_EQ_WAIT((c), (r).num_rendered_frames(), (t)); \ |
| EXPECT_EQ((w), (r).width()); \ |
| EXPECT_EQ((h), (r).height()); \ |
| EXPECT_EQ(0, (r).errors()); |
| |
| namespace cricket { |
| class WebRtcVideoEngineTest : public ::testing::Test { |
| public: |
| WebRtcVideoEngineTest() : WebRtcVideoEngineTest("") {} |
| explicit WebRtcVideoEngineTest(const std::string& field_trials) |
| : override_field_trials_( |
| field_trials.empty() |
| ? nullptr |
| : std::make_unique<webrtc::test::ScopedFieldTrials>( |
| field_trials)), |
| task_queue_factory_(webrtc::CreateDefaultTaskQueueFactory()), |
| call_(webrtc::Call::Create([&] { |
| webrtc::Call::Config call_config(&event_log_); |
| call_config.task_queue_factory = task_queue_factory_.get(); |
| call_config.trials = &field_trials_; |
| return call_config; |
| }())), |
| encoder_factory_(new cricket::FakeWebRtcVideoEncoderFactory), |
| decoder_factory_(new cricket::FakeWebRtcVideoDecoderFactory), |
| video_bitrate_allocator_factory_( |
| webrtc::CreateBuiltinVideoBitrateAllocatorFactory()), |
| engine_(std::unique_ptr<cricket::FakeWebRtcVideoEncoderFactory>( |
| encoder_factory_), |
| std::unique_ptr<cricket::FakeWebRtcVideoDecoderFactory>( |
| decoder_factory_), |
| field_trials_) { |
| // Ensure fake clock doesn't return 0, which will cause some initializations |
| // fail inside RTP senders. |
| fake_clock_.AdvanceTime(webrtc::TimeDelta::Micros(1)); |
| } |
| |
| protected: |
| void AssignDefaultAptRtxTypes(); |
| void AssignDefaultCodec(); |
| |
| // Find the index of the codec in the engine with the given name. The codec |
| // must be present. |
| size_t GetEngineCodecIndex(const std::string& name) const; |
| |
| // Find the codec in the engine with the given name. The codec must be |
| // present. |
| cricket::VideoCodec GetEngineCodec(const std::string& name) const; |
| void AddSupportedVideoCodecType(const std::string& name); |
| VideoMediaChannel* SetSendParamsWithAllSupportedCodecs(); |
| |
| VideoMediaChannel* SetRecvParamsWithSupportedCodecs( |
| const std::vector<VideoCodec>& codecs); |
| |
| void ExpectRtpCapabilitySupport(const char* uri, bool supported) const; |
| |
| // Has to be the first one, so it is initialized before the call or there is a |
| // race condition in the clock access. |
| rtc::ScopedFakeClock fake_clock_; |
| std::unique_ptr<webrtc::test::ScopedFieldTrials> override_field_trials_; |
| webrtc::FieldTrialBasedConfig field_trials_; |
| webrtc::RtcEventLogNull event_log_; |
| std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory_; |
| // Used in WebRtcVideoEngineVoiceTest, but defined here so it's properly |
| // initialized when the constructor is called. |
| std::unique_ptr<webrtc::Call> call_; |
| cricket::FakeWebRtcVideoEncoderFactory* encoder_factory_; |
| cricket::FakeWebRtcVideoDecoderFactory* decoder_factory_; |
| std::unique_ptr<webrtc::VideoBitrateAllocatorFactory> |
| video_bitrate_allocator_factory_; |
| WebRtcVideoEngine engine_; |
| VideoCodec default_codec_; |
| std::map<int, int> default_apt_rtx_types_; |
| }; |
| |
| TEST_F(WebRtcVideoEngineTest, DefaultRtxCodecHasAssociatedPayloadTypeSet) { |
| encoder_factory_->AddSupportedVideoCodecType("VP8"); |
| AssignDefaultCodec(); |
| |
| std::vector<VideoCodec> engine_codecs = engine_.send_codecs(); |
| for (size_t i = 0; i < engine_codecs.size(); ++i) { |
| if (engine_codecs[i].name != kRtxCodecName) |
| continue; |
| int associated_payload_type; |
| EXPECT_TRUE(engine_codecs[i].GetParam(kCodecParamAssociatedPayloadType, |
| &associated_payload_type)); |
| EXPECT_EQ(default_codec_.id, associated_payload_type); |
| return; |
| } |
| FAIL() << "No RTX codec found among default codecs."; |
| } |
| |
| TEST_F(WebRtcVideoEngineTest, SupportsTimestampOffsetHeaderExtension) { |
| ExpectRtpCapabilitySupport(RtpExtension::kTimestampOffsetUri, true); |
| } |
| |
| TEST_F(WebRtcVideoEngineTest, SupportsAbsoluteSenderTimeHeaderExtension) { |
| ExpectRtpCapabilitySupport(RtpExtension::kAbsSendTimeUri, true); |
| } |
| |
| TEST_F(WebRtcVideoEngineTest, SupportsTransportSequenceNumberHeaderExtension) { |
| ExpectRtpCapabilitySupport(RtpExtension::kTransportSequenceNumberUri, true); |
| } |
| |
| TEST_F(WebRtcVideoEngineTest, SupportsVideoRotationHeaderExtension) { |
| ExpectRtpCapabilitySupport(RtpExtension::kVideoRotationUri, true); |
| } |
| |
| TEST_F(WebRtcVideoEngineTest, SupportsPlayoutDelayHeaderExtension) { |
| ExpectRtpCapabilitySupport(RtpExtension::kPlayoutDelayUri, true); |
| } |
| |
| TEST_F(WebRtcVideoEngineTest, SupportsVideoContentTypeHeaderExtension) { |
| ExpectRtpCapabilitySupport(RtpExtension::kVideoContentTypeUri, true); |
| } |
| |
| TEST_F(WebRtcVideoEngineTest, SupportsVideoTimingHeaderExtension) { |
| ExpectRtpCapabilitySupport(RtpExtension::kVideoTimingUri, true); |
| } |
| |
| TEST_F(WebRtcVideoEngineTest, SupportsColorSpaceHeaderExtension) { |
| ExpectRtpCapabilitySupport(RtpExtension::kColorSpaceUri, true); |
| } |
| |
| TEST_F(WebRtcVideoEngineTest, AdvertiseGenericDescriptor00) { |
| ExpectRtpCapabilitySupport(RtpExtension::kGenericFrameDescriptorUri00, false); |
| } |
| |
| class WebRtcVideoEngineTestWithGenericDescriptor |
| : public WebRtcVideoEngineTest { |
| public: |
| WebRtcVideoEngineTestWithGenericDescriptor() |
| : WebRtcVideoEngineTest("WebRTC-GenericDescriptorAdvertised/Enabled/") {} |
| }; |
| |
| TEST_F(WebRtcVideoEngineTestWithGenericDescriptor, |
| AdvertiseGenericDescriptor00) { |
| ExpectRtpCapabilitySupport(RtpExtension::kGenericFrameDescriptorUri00, true); |
| } |
| |
| class WebRtcVideoEngineTestWithDependencyDescriptor |
| : public WebRtcVideoEngineTest { |
| public: |
| WebRtcVideoEngineTestWithDependencyDescriptor() |
| : WebRtcVideoEngineTest( |
| "WebRTC-DependencyDescriptorAdvertised/Enabled/") {} |
| }; |
| |
| TEST_F(WebRtcVideoEngineTestWithDependencyDescriptor, |
| AdvertiseDependencyDescriptor) { |
| ExpectRtpCapabilitySupport(RtpExtension::kDependencyDescriptorUri, true); |
| } |
| |
| TEST_F(WebRtcVideoEngineTest, AdvertiseVideoLayersAllocation) { |
| ExpectRtpCapabilitySupport(RtpExtension::kVideoLayersAllocationUri, false); |
| } |
| |
| class WebRtcVideoEngineTestWithVideoLayersAllocation |
| : public WebRtcVideoEngineTest { |
| public: |
| WebRtcVideoEngineTestWithVideoLayersAllocation() |
| : WebRtcVideoEngineTest( |
| "WebRTC-VideoLayersAllocationAdvertised/Enabled/") {} |
| }; |
| |
| TEST_F(WebRtcVideoEngineTestWithVideoLayersAllocation, |
| AdvertiseVideoLayersAllocation) { |
| ExpectRtpCapabilitySupport(RtpExtension::kVideoLayersAllocationUri, true); |
| } |
| |
| class WebRtcVideoFrameTrackingId : public WebRtcVideoEngineTest { |
| public: |
| WebRtcVideoFrameTrackingId() |
| : WebRtcVideoEngineTest( |
| "WebRTC-VideoFrameTrackingIdAdvertised/Enabled/") {} |
| }; |
| |
| TEST_F(WebRtcVideoFrameTrackingId, AdvertiseVideoFrameTrackingId) { |
| ExpectRtpCapabilitySupport(RtpExtension::kVideoFrameTrackingIdUri, true); |
| } |
| |
| TEST_F(WebRtcVideoEngineTest, CVOSetHeaderExtensionBeforeCapturer) { |
| // Allocate the source first to prevent early destruction before channel's |
| // dtor is called. |
| ::testing::NiceMock<MockVideoSource> video_source; |
| |
| AddSupportedVideoCodecType("VP8"); |
| |
| std::unique_ptr<VideoMediaChannel> channel( |
| SetSendParamsWithAllSupportedCodecs()); |
| EXPECT_TRUE(channel->AddSendStream(StreamParams::CreateLegacy(kSsrc))); |
| |
| // Add CVO extension. |
| const int id = 1; |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| parameters.extensions.push_back( |
| RtpExtension(RtpExtension::kVideoRotationUri, id)); |
| EXPECT_TRUE(channel->SetSendParameters(parameters)); |
| |
| EXPECT_CALL( |
| video_source, |
| AddOrUpdateSink(_, Field(&rtc::VideoSinkWants::rotation_applied, false))); |
| // Set capturer. |
| EXPECT_TRUE(channel->SetVideoSend(kSsrc, nullptr, &video_source)); |
| |
| // Verify capturer has turned off applying rotation. |
| ::testing::Mock::VerifyAndClear(&video_source); |
| |
| // Verify removing header extension turns on applying rotation. |
| parameters.extensions.clear(); |
| EXPECT_CALL( |
| video_source, |
| AddOrUpdateSink(_, Field(&rtc::VideoSinkWants::rotation_applied, true))); |
| |
| EXPECT_TRUE(channel->SetSendParameters(parameters)); |
| } |
| |
| TEST_F(WebRtcVideoEngineTest, CVOSetHeaderExtensionBeforeAddSendStream) { |
| // Allocate the source first to prevent early destruction before channel's |
| // dtor is called. |
| ::testing::NiceMock<MockVideoSource> video_source; |
| |
| AddSupportedVideoCodecType("VP8"); |
| |
| std::unique_ptr<VideoMediaChannel> channel( |
| SetSendParamsWithAllSupportedCodecs()); |
| // Add CVO extension. |
| const int id = 1; |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| parameters.extensions.push_back( |
| RtpExtension(RtpExtension::kVideoRotationUri, id)); |
| EXPECT_TRUE(channel->SetSendParameters(parameters)); |
| EXPECT_TRUE(channel->AddSendStream(StreamParams::CreateLegacy(kSsrc))); |
| |
| // Set source. |
| EXPECT_CALL( |
| video_source, |
| AddOrUpdateSink(_, Field(&rtc::VideoSinkWants::rotation_applied, false))); |
| EXPECT_TRUE(channel->SetVideoSend(kSsrc, nullptr, &video_source)); |
| } |
| |
| TEST_F(WebRtcVideoEngineTest, CVOSetHeaderExtensionAfterCapturer) { |
| ::testing::NiceMock<MockVideoSource> video_source; |
| |
| AddSupportedVideoCodecType("VP8"); |
| AddSupportedVideoCodecType("VP9"); |
| |
| std::unique_ptr<VideoMediaChannel> channel( |
| SetSendParamsWithAllSupportedCodecs()); |
| EXPECT_TRUE(channel->AddSendStream(StreamParams::CreateLegacy(kSsrc))); |
| |
| // Set capturer. |
| EXPECT_CALL( |
| video_source, |
| AddOrUpdateSink(_, Field(&rtc::VideoSinkWants::rotation_applied, true))); |
| EXPECT_TRUE(channel->SetVideoSend(kSsrc, nullptr, &video_source)); |
| |
| // Verify capturer has turned on applying rotation. |
| ::testing::Mock::VerifyAndClear(&video_source); |
| |
| // Add CVO extension. |
| const int id = 1; |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| parameters.codecs.push_back(GetEngineCodec("VP9")); |
| parameters.extensions.push_back( |
| RtpExtension(RtpExtension::kVideoRotationUri, id)); |
| // Also remove the first codec to trigger a codec change as well. |
| parameters.codecs.erase(parameters.codecs.begin()); |
| EXPECT_CALL( |
| video_source, |
| AddOrUpdateSink(_, Field(&rtc::VideoSinkWants::rotation_applied, false))); |
| EXPECT_TRUE(channel->SetSendParameters(parameters)); |
| |
| // Verify capturer has turned off applying rotation. |
| ::testing::Mock::VerifyAndClear(&video_source); |
| |
| // Verify removing header extension turns on applying rotation. |
| parameters.extensions.clear(); |
| EXPECT_CALL( |
| video_source, |
| AddOrUpdateSink(_, Field(&rtc::VideoSinkWants::rotation_applied, true))); |
| EXPECT_TRUE(channel->SetSendParameters(parameters)); |
| } |
| |
| TEST_F(WebRtcVideoEngineTest, SetSendFailsBeforeSettingCodecs) { |
| AddSupportedVideoCodecType("VP8"); |
| |
| std::unique_ptr<VideoMediaChannel> channel(engine_.CreateMediaChannel( |
| call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(), |
| video_bitrate_allocator_factory_.get())); |
| |
| EXPECT_TRUE(channel->AddSendStream(StreamParams::CreateLegacy(123))); |
| |
| EXPECT_FALSE(channel->SetSend(true)) |
| << "Channel should not start without codecs."; |
| EXPECT_TRUE(channel->SetSend(false)) |
| << "Channel should be stoppable even without set codecs."; |
| } |
| |
| TEST_F(WebRtcVideoEngineTest, GetStatsWithoutSendCodecsSetDoesNotCrash) { |
| AddSupportedVideoCodecType("VP8"); |
| |
| std::unique_ptr<VideoMediaChannel> channel(engine_.CreateMediaChannel( |
| call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(), |
| video_bitrate_allocator_factory_.get())); |
| EXPECT_TRUE(channel->AddSendStream(StreamParams::CreateLegacy(123))); |
| VideoMediaInfo info; |
| channel->GetStats(&info); |
| } |
| |
| TEST_F(WebRtcVideoEngineTest, UseFactoryForVp8WhenSupported) { |
| AddSupportedVideoCodecType("VP8"); |
| |
| std::unique_ptr<VideoMediaChannel> channel( |
| SetSendParamsWithAllSupportedCodecs()); |
| channel->OnReadyToSend(true); |
| |
| EXPECT_TRUE( |
| channel->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc))); |
| EXPECT_EQ(0, encoder_factory_->GetNumCreatedEncoders()); |
| EXPECT_TRUE(channel->SetSend(true)); |
| webrtc::test::FrameForwarder frame_forwarder; |
| cricket::FakeFrameSource frame_source(1280, 720, |
| rtc::kNumMicrosecsPerSec / 30); |
| EXPECT_TRUE(channel->SetVideoSend(kSsrc, nullptr, &frame_forwarder)); |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| // Sending one frame will have allocate the encoder. |
| ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(1)); |
| EXPECT_TRUE_WAIT(encoder_factory_->encoders()[0]->GetNumEncodedFrames() > 0, |
| kTimeout); |
| |
| int num_created_encoders = encoder_factory_->GetNumCreatedEncoders(); |
| EXPECT_EQ(num_created_encoders, 1); |
| |
| // Setting codecs of the same type should not reallocate any encoders |
| // (expecting a no-op). |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| EXPECT_TRUE(channel->SetSendParameters(parameters)); |
| EXPECT_EQ(num_created_encoders, encoder_factory_->GetNumCreatedEncoders()); |
| |
| // Remove stream previously added to free the external encoder instance. |
| EXPECT_TRUE(channel->RemoveSendStream(kSsrc)); |
| EXPECT_EQ(0u, encoder_factory_->encoders().size()); |
| } |
| |
| // Test that when an encoder factory supports H264, we add an RTX |
| // codec for it. |
| // TODO(deadbeef): This test should be updated if/when we start |
| // adding RTX codecs for unrecognized codec names. |
| TEST_F(WebRtcVideoEngineTest, RtxCodecAddedForH264Codec) { |
| using webrtc::H264Level; |
| using webrtc::H264Profile; |
| using webrtc::H264ProfileLevelId; |
| using webrtc::H264ProfileLevelIdToString; |
| webrtc::SdpVideoFormat h264_constrained_baseline("H264"); |
| h264_constrained_baseline.parameters[kH264FmtpProfileLevelId] = |
| *H264ProfileLevelIdToString(H264ProfileLevelId( |
| H264Profile::kProfileConstrainedBaseline, H264Level::kLevel1)); |
| webrtc::SdpVideoFormat h264_constrained_high("H264"); |
| h264_constrained_high.parameters[kH264FmtpProfileLevelId] = |
| *H264ProfileLevelIdToString(H264ProfileLevelId( |
| H264Profile::kProfileConstrainedHigh, H264Level::kLevel1)); |
| webrtc::SdpVideoFormat h264_high("H264"); |
| h264_high.parameters[kH264FmtpProfileLevelId] = *H264ProfileLevelIdToString( |
| H264ProfileLevelId(H264Profile::kProfileHigh, H264Level::kLevel1)); |
| |
| encoder_factory_->AddSupportedVideoCodec(h264_constrained_baseline); |
| encoder_factory_->AddSupportedVideoCodec(h264_constrained_high); |
| encoder_factory_->AddSupportedVideoCodec(h264_high); |
| |
| // First figure out what payload types the test codecs got assigned. |
| const std::vector<cricket::VideoCodec> codecs = engine_.send_codecs(); |
| // Now search for RTX codecs for them. Expect that they all have associated |
| // RTX codecs. |
| EXPECT_TRUE(HasRtxCodec( |
| codecs, |
| FindMatchingCodec(codecs, cricket::VideoCodec(h264_constrained_baseline)) |
| ->id)); |
| EXPECT_TRUE(HasRtxCodec( |
| codecs, |
| FindMatchingCodec(codecs, cricket::VideoCodec(h264_constrained_high)) |
| ->id)); |
| EXPECT_TRUE(HasRtxCodec( |
| codecs, FindMatchingCodec(codecs, cricket::VideoCodec(h264_high))->id)); |
| } |
| |
| #if defined(RTC_ENABLE_VP9) |
| TEST_F(WebRtcVideoEngineTest, CanConstructDecoderForVp9EncoderFactory) { |
| AddSupportedVideoCodecType("VP9"); |
| |
| std::unique_ptr<VideoMediaChannel> channel( |
| SetSendParamsWithAllSupportedCodecs()); |
| |
| EXPECT_TRUE( |
| channel->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc))); |
| } |
| #endif // defined(RTC_ENABLE_VP9) |
| |
| TEST_F(WebRtcVideoEngineTest, PropagatesInputFrameTimestamp) { |
| AddSupportedVideoCodecType("VP8"); |
| FakeCall* fake_call = new FakeCall(); |
| call_.reset(fake_call); |
| std::unique_ptr<VideoMediaChannel> channel( |
| SetSendParamsWithAllSupportedCodecs()); |
| |
| EXPECT_TRUE( |
| channel->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc))); |
| |
| webrtc::test::FrameForwarder frame_forwarder; |
| cricket::FakeFrameSource frame_source(1280, 720, |
| rtc::kNumMicrosecsPerSec / 60); |
| EXPECT_TRUE(channel->SetVideoSend(kSsrc, nullptr, &frame_forwarder)); |
| channel->SetSend(true); |
| |
| FakeVideoSendStream* stream = fake_call->GetVideoSendStreams()[0]; |
| |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| int64_t last_timestamp = stream->GetLastTimestamp(); |
| for (int i = 0; i < 10; i++) { |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| int64_t timestamp = stream->GetLastTimestamp(); |
| int64_t interval = timestamp - last_timestamp; |
| |
| // Precision changes from nanosecond to millisecond. |
| // Allow error to be no more than 1. |
| EXPECT_NEAR(cricket::VideoFormat::FpsToInterval(60) / 1E6, interval, 1); |
| |
| last_timestamp = timestamp; |
| } |
| |
| frame_forwarder.IncomingCapturedFrame( |
| frame_source.GetFrame(1280, 720, webrtc::VideoRotation::kVideoRotation_0, |
| rtc::kNumMicrosecsPerSec / 30)); |
| last_timestamp = stream->GetLastTimestamp(); |
| for (int i = 0; i < 10; i++) { |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame( |
| 1280, 720, webrtc::VideoRotation::kVideoRotation_0, |
| rtc::kNumMicrosecsPerSec / 30)); |
| int64_t timestamp = stream->GetLastTimestamp(); |
| int64_t interval = timestamp - last_timestamp; |
| |
| // Precision changes from nanosecond to millisecond. |
| // Allow error to be no more than 1. |
| EXPECT_NEAR(cricket::VideoFormat::FpsToInterval(30) / 1E6, interval, 1); |
| |
| last_timestamp = timestamp; |
| } |
| |
| // Remove stream previously added to free the external encoder instance. |
| EXPECT_TRUE(channel->RemoveSendStream(kSsrc)); |
| } |
| |
| void WebRtcVideoEngineTest::AssignDefaultAptRtxTypes() { |
| std::vector<VideoCodec> engine_codecs = engine_.send_codecs(); |
| RTC_DCHECK(!engine_codecs.empty()); |
| for (const cricket::VideoCodec& codec : engine_codecs) { |
| if (codec.name == "rtx") { |
| int associated_payload_type; |
| if (codec.GetParam(kCodecParamAssociatedPayloadType, |
| &associated_payload_type)) { |
| default_apt_rtx_types_[associated_payload_type] = codec.id; |
| } |
| } |
| } |
| } |
| |
| void WebRtcVideoEngineTest::AssignDefaultCodec() { |
| std::vector<VideoCodec> engine_codecs = engine_.send_codecs(); |
| RTC_DCHECK(!engine_codecs.empty()); |
| bool codec_set = false; |
| for (const cricket::VideoCodec& codec : engine_codecs) { |
| if (!codec_set && codec.name != "rtx" && codec.name != "red" && |
| codec.name != "ulpfec" && codec.name != "flexfec-03") { |
| default_codec_ = codec; |
| codec_set = true; |
| } |
| } |
| |
| RTC_DCHECK(codec_set); |
| } |
| |
| size_t WebRtcVideoEngineTest::GetEngineCodecIndex( |
| const std::string& name) const { |
| const std::vector<cricket::VideoCodec> codecs = engine_.send_codecs(); |
| for (size_t i = 0; i < codecs.size(); ++i) { |
| const cricket::VideoCodec engine_codec = codecs[i]; |
| if (!absl::EqualsIgnoreCase(name, engine_codec.name)) |
| continue; |
| // The tests only use H264 Constrained Baseline. Make sure we don't return |
| // an internal H264 codec from the engine with a different H264 profile. |
| if (absl::EqualsIgnoreCase(name.c_str(), kH264CodecName)) { |
| const absl::optional<webrtc::H264ProfileLevelId> profile_level_id = |
| webrtc::ParseSdpForH264ProfileLevelId(engine_codec.params); |
| if (profile_level_id->profile != |
| webrtc::H264Profile::kProfileConstrainedBaseline) { |
| continue; |
| } |
| } |
| return i; |
| } |
| // This point should never be reached. |
| ADD_FAILURE() << "Unrecognized codec name: " << name; |
| return -1; |
| } |
| |
| cricket::VideoCodec WebRtcVideoEngineTest::GetEngineCodec( |
| const std::string& name) const { |
| return engine_.send_codecs()[GetEngineCodecIndex(name)]; |
| } |
| |
| void WebRtcVideoEngineTest::AddSupportedVideoCodecType( |
| const std::string& name) { |
| encoder_factory_->AddSupportedVideoCodecType(name); |
| decoder_factory_->AddSupportedVideoCodecType(name); |
| } |
| |
| VideoMediaChannel* |
| WebRtcVideoEngineTest::SetSendParamsWithAllSupportedCodecs() { |
| VideoMediaChannel* channel = engine_.CreateMediaChannel( |
| call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(), |
| video_bitrate_allocator_factory_.get()); |
| cricket::VideoSendParameters parameters; |
| // We need to look up the codec in the engine to get the correct payload type. |
| for (const webrtc::SdpVideoFormat& format : |
| encoder_factory_->GetSupportedFormats()) { |
| cricket::VideoCodec engine_codec = GetEngineCodec(format.name); |
| if (!absl::c_linear_search(parameters.codecs, engine_codec)) { |
| parameters.codecs.push_back(engine_codec); |
| } |
| } |
| |
| EXPECT_TRUE(channel->SetSendParameters(parameters)); |
| |
| return channel; |
| } |
| |
| VideoMediaChannel* WebRtcVideoEngineTest::SetRecvParamsWithSupportedCodecs( |
| const std::vector<VideoCodec>& codecs) { |
| VideoMediaChannel* channel = engine_.CreateMediaChannel( |
| call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(), |
| video_bitrate_allocator_factory_.get()); |
| cricket::VideoRecvParameters parameters; |
| parameters.codecs = codecs; |
| EXPECT_TRUE(channel->SetRecvParameters(parameters)); |
| |
| return channel; |
| } |
| |
| void WebRtcVideoEngineTest::ExpectRtpCapabilitySupport(const char* uri, |
| bool supported) const { |
| const std::vector<webrtc::RtpExtension> header_extensions = |
| GetDefaultEnabledRtpHeaderExtensions(engine_); |
| if (supported) { |
| EXPECT_THAT(header_extensions, Contains(Field(&RtpExtension::uri, uri))); |
| } else { |
| EXPECT_THAT(header_extensions, Each(Field(&RtpExtension::uri, StrNe(uri)))); |
| } |
| } |
| |
| TEST_F(WebRtcVideoEngineTest, UsesSimulcastAdapterForVp8Factories) { |
| AddSupportedVideoCodecType("VP8"); |
| |
| std::unique_ptr<VideoMediaChannel> channel( |
| SetSendParamsWithAllSupportedCodecs()); |
| |
| std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3); |
| |
| EXPECT_TRUE(channel->AddSendStream(CreateSimStreamParams("cname", ssrcs))); |
| EXPECT_TRUE(channel->SetSend(true)); |
| |
| webrtc::test::FrameForwarder frame_forwarder; |
| cricket::FakeFrameSource frame_source(1280, 720, |
| rtc::kNumMicrosecsPerSec / 60); |
| EXPECT_TRUE(channel->SetVideoSend(ssrcs.front(), nullptr, &frame_forwarder)); |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| |
| ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(2)); |
| |
| // Verify that encoders are configured for simulcast through adapter |
| // (increasing resolution and only configured to send one stream each). |
| int prev_width = -1; |
| for (size_t i = 0; i < encoder_factory_->encoders().size(); ++i) { |
| ASSERT_TRUE(encoder_factory_->encoders()[i]->WaitForInitEncode()); |
| webrtc::VideoCodec codec_settings = |
| encoder_factory_->encoders()[i]->GetCodecSettings(); |
| EXPECT_EQ(0, codec_settings.numberOfSimulcastStreams); |
| EXPECT_GT(codec_settings.width, prev_width); |
| prev_width = codec_settings.width; |
| } |
| |
| EXPECT_TRUE(channel->SetVideoSend(ssrcs.front(), nullptr, nullptr)); |
| |
| channel.reset(); |
| ASSERT_EQ(0u, encoder_factory_->encoders().size()); |
| } |
| |
| TEST_F(WebRtcVideoEngineTest, ChannelWithH264CanChangeToVp8) { |
| AddSupportedVideoCodecType("VP8"); |
| AddSupportedVideoCodecType("H264"); |
| |
| // Frame source. |
| webrtc::test::FrameForwarder frame_forwarder; |
| cricket::FakeFrameSource frame_source(1280, 720, |
| rtc::kNumMicrosecsPerSec / 30); |
| |
| std::unique_ptr<VideoMediaChannel> channel(engine_.CreateMediaChannel( |
| call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(), |
| video_bitrate_allocator_factory_.get())); |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("H264")); |
| EXPECT_TRUE(channel->SetSendParameters(parameters)); |
| |
| EXPECT_TRUE( |
| channel->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc))); |
| EXPECT_TRUE(channel->SetVideoSend(kSsrc, nullptr, &frame_forwarder)); |
| // Sending one frame will have allocate the encoder. |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| |
| ASSERT_EQ_WAIT(1u, encoder_factory_->encoders().size(), kTimeout); |
| |
| cricket::VideoSendParameters new_parameters; |
| new_parameters.codecs.push_back(GetEngineCodec("VP8")); |
| EXPECT_TRUE(channel->SetSendParameters(new_parameters)); |
| |
| // Sending one frame will switch encoder. |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| |
| EXPECT_EQ_WAIT(1u, encoder_factory_->encoders().size(), kTimeout); |
| } |
| |
| TEST_F(WebRtcVideoEngineTest, |
| UsesSimulcastAdapterForVp8WithCombinedVP8AndH264Factory) { |
| AddSupportedVideoCodecType("VP8"); |
| AddSupportedVideoCodecType("H264"); |
| |
| std::unique_ptr<VideoMediaChannel> channel(engine_.CreateMediaChannel( |
| call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(), |
| video_bitrate_allocator_factory_.get())); |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| EXPECT_TRUE(channel->SetSendParameters(parameters)); |
| |
| std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3); |
| |
| EXPECT_TRUE(channel->AddSendStream(CreateSimStreamParams("cname", ssrcs))); |
| EXPECT_TRUE(channel->SetSend(true)); |
| |
| // Send a fake frame, or else the media engine will configure the simulcast |
| // encoder adapter at a low-enough size that it'll only create a single |
| // encoder layer. |
| webrtc::test::FrameForwarder frame_forwarder; |
| cricket::FakeFrameSource frame_source(1280, 720, |
| rtc::kNumMicrosecsPerSec / 30); |
| EXPECT_TRUE(channel->SetVideoSend(ssrcs.front(), nullptr, &frame_forwarder)); |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| |
| ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(2)); |
| ASSERT_TRUE(encoder_factory_->encoders()[0]->WaitForInitEncode()); |
| EXPECT_EQ(webrtc::kVideoCodecVP8, |
| encoder_factory_->encoders()[0]->GetCodecSettings().codecType); |
| |
| channel.reset(); |
| // Make sure DestroyVideoEncoder was called on the factory. |
| EXPECT_EQ(0u, encoder_factory_->encoders().size()); |
| } |
| |
| TEST_F(WebRtcVideoEngineTest, |
| DestroysNonSimulcastEncoderFromCombinedVP8AndH264Factory) { |
| AddSupportedVideoCodecType("VP8"); |
| AddSupportedVideoCodecType("H264"); |
| |
| std::unique_ptr<VideoMediaChannel> channel(engine_.CreateMediaChannel( |
| call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(), |
| video_bitrate_allocator_factory_.get())); |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("H264")); |
| EXPECT_TRUE(channel->SetSendParameters(parameters)); |
| |
| EXPECT_TRUE( |
| channel->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc))); |
| |
| // Send a frame of 720p. This should trigger a "real" encoder initialization. |
| webrtc::test::FrameForwarder frame_forwarder; |
| cricket::FakeFrameSource frame_source(1280, 720, |
| rtc::kNumMicrosecsPerSec / 30); |
| EXPECT_TRUE(channel->SetVideoSend(kSsrc, nullptr, &frame_forwarder)); |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(1)); |
| ASSERT_EQ(1u, encoder_factory_->encoders().size()); |
| ASSERT_TRUE(encoder_factory_->encoders()[0]->WaitForInitEncode()); |
| EXPECT_EQ(webrtc::kVideoCodecH264, |
| encoder_factory_->encoders()[0]->GetCodecSettings().codecType); |
| |
| channel.reset(); |
| // Make sure DestroyVideoEncoder was called on the factory. |
| ASSERT_EQ(0u, encoder_factory_->encoders().size()); |
| } |
| |
| TEST_F(WebRtcVideoEngineTest, SimulcastEnabledForH264BehindFieldTrial) { |
| RTC_DCHECK(!override_field_trials_); |
| override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>( |
| "WebRTC-H264Simulcast/Enabled/"); |
| AddSupportedVideoCodecType("H264"); |
| |
| std::unique_ptr<VideoMediaChannel> channel(engine_.CreateMediaChannel( |
| call_.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(), |
| video_bitrate_allocator_factory_.get())); |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("H264")); |
| EXPECT_TRUE(channel->SetSendParameters(parameters)); |
| |
| const std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3); |
| EXPECT_TRUE( |
| channel->AddSendStream(cricket::CreateSimStreamParams("cname", ssrcs))); |
| |
| // Send a frame of 720p. This should trigger a "real" encoder initialization. |
| webrtc::test::FrameForwarder frame_forwarder; |
| cricket::FakeFrameSource frame_source(1280, 720, |
| rtc::kNumMicrosecsPerSec / 30); |
| EXPECT_TRUE(channel->SetVideoSend(ssrcs[0], nullptr, &frame_forwarder)); |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| |
| ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(1)); |
| ASSERT_EQ(1u, encoder_factory_->encoders().size()); |
| FakeWebRtcVideoEncoder* encoder = encoder_factory_->encoders()[0]; |
| ASSERT_TRUE(encoder_factory_->encoders()[0]->WaitForInitEncode()); |
| EXPECT_EQ(webrtc::kVideoCodecH264, encoder->GetCodecSettings().codecType); |
| EXPECT_LT(1u, encoder->GetCodecSettings().numberOfSimulcastStreams); |
| EXPECT_TRUE(channel->SetVideoSend(ssrcs[0], nullptr, nullptr)); |
| } |
| |
| // Test that FlexFEC is not supported as a send video codec by default. |
| // Only enabling field trial should allow advertising FlexFEC send codec. |
| TEST_F(WebRtcVideoEngineTest, Flexfec03SendCodecEnablesWithFieldTrial) { |
| encoder_factory_->AddSupportedVideoCodecType("VP8"); |
| |
| auto flexfec = Field("name", &VideoCodec::name, "flexfec-03"); |
| |
| EXPECT_THAT(engine_.send_codecs(), Not(Contains(flexfec))); |
| |
| RTC_DCHECK(!override_field_trials_); |
| override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>( |
| "WebRTC-FlexFEC-03-Advertised/Enabled/"); |
| EXPECT_THAT(engine_.send_codecs(), Contains(flexfec)); |
| } |
| |
| // Test that FlexFEC is supported as a receive video codec by default. |
| // Disabling field trial should prevent advertising FlexFEC receive codec. |
| TEST_F(WebRtcVideoEngineTest, Flexfec03ReceiveCodecDisablesWithFieldTrial) { |
| decoder_factory_->AddSupportedVideoCodecType("VP8"); |
| |
| auto flexfec = Field("name", &VideoCodec::name, "flexfec-03"); |
| |
| EXPECT_THAT(engine_.recv_codecs(), Contains(flexfec)); |
| |
| RTC_DCHECK(!override_field_trials_); |
| override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>( |
| "WebRTC-FlexFEC-03-Advertised/Disabled/"); |
| EXPECT_THAT(engine_.recv_codecs(), Not(Contains(flexfec))); |
| } |
| |
| // Test that the FlexFEC "codec" gets assigned to the lower payload type range |
| TEST_F(WebRtcVideoEngineTest, Flexfec03LowerPayloadTypeRange) { |
| encoder_factory_->AddSupportedVideoCodecType("VP8"); |
| |
| auto flexfec = Field("name", &VideoCodec::name, "flexfec-03"); |
| |
| // FlexFEC is active with field trial. |
| RTC_DCHECK(!override_field_trials_); |
| override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>( |
| "WebRTC-FlexFEC-03-Advertised/Enabled/"); |
| auto send_codecs = engine_.send_codecs(); |
| auto it = std::find_if(send_codecs.begin(), send_codecs.end(), |
| [](const cricket::VideoCodec& codec) { |
| return codec.name == "flexfec-03"; |
| }); |
| ASSERT_NE(it, send_codecs.end()); |
| EXPECT_LE(35, it->id); |
| EXPECT_GE(65, it->id); |
| } |
| |
| // Test that codecs are added in the order they are reported from the factory. |
| TEST_F(WebRtcVideoEngineTest, ReportSupportedCodecs) { |
| encoder_factory_->AddSupportedVideoCodecType("VP8"); |
| const char* kFakeCodecName = "FakeCodec"; |
| encoder_factory_->AddSupportedVideoCodecType(kFakeCodecName); |
| |
| // The last reported codec should appear after the first codec in the vector. |
| const size_t vp8_index = GetEngineCodecIndex("VP8"); |
| const size_t fake_codec_index = GetEngineCodecIndex(kFakeCodecName); |
| EXPECT_LT(vp8_index, fake_codec_index); |
| } |
| |
| // Test that a codec that was added after the engine was initialized |
| // does show up in the codec list after it was added. |
| TEST_F(WebRtcVideoEngineTest, ReportSupportedAddedCodec) { |
| const char* kFakeExternalCodecName1 = "FakeExternalCodec1"; |
| const char* kFakeExternalCodecName2 = "FakeExternalCodec2"; |
| |
| // Set up external encoder factory with first codec, and initialize engine. |
| encoder_factory_->AddSupportedVideoCodecType(kFakeExternalCodecName1); |
| |
| std::vector<cricket::VideoCodec> codecs_before(engine_.send_codecs()); |
| |
| // Add second codec. |
| encoder_factory_->AddSupportedVideoCodecType(kFakeExternalCodecName2); |
| std::vector<cricket::VideoCodec> codecs_after(engine_.send_codecs()); |
| // The codec itself and RTX should have been added. |
| EXPECT_EQ(codecs_before.size() + 2, codecs_after.size()); |
| |
| // Check that both fake codecs are present and that the second fake codec |
| // appears after the first fake codec. |
| const size_t fake_codec_index1 = GetEngineCodecIndex(kFakeExternalCodecName1); |
| const size_t fake_codec_index2 = GetEngineCodecIndex(kFakeExternalCodecName2); |
| EXPECT_LT(fake_codec_index1, fake_codec_index2); |
| } |
| |
| TEST_F(WebRtcVideoEngineTest, ReportRtxForExternalCodec) { |
| const char* kFakeCodecName = "FakeCodec"; |
| encoder_factory_->AddSupportedVideoCodecType(kFakeCodecName); |
| |
| const size_t fake_codec_index = GetEngineCodecIndex(kFakeCodecName); |
| EXPECT_EQ("rtx", engine_.send_codecs().at(fake_codec_index + 1).name); |
| } |
| |
| TEST_F(WebRtcVideoEngineTest, RegisterDecodersIfSupported) { |
| AddSupportedVideoCodecType("VP8"); |
| cricket::VideoRecvParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| |
| std::unique_ptr<VideoMediaChannel> channel( |
| SetRecvParamsWithSupportedCodecs(parameters.codecs)); |
| |
| EXPECT_TRUE( |
| channel->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc))); |
| ASSERT_EQ(1u, decoder_factory_->decoders().size()); |
| |
| // Setting codecs of the same type should not reallocate the decoder. |
| EXPECT_TRUE(channel->SetRecvParameters(parameters)); |
| EXPECT_EQ(1, decoder_factory_->GetNumCreatedDecoders()); |
| |
| // Remove stream previously added to free the external decoder instance. |
| EXPECT_TRUE(channel->RemoveRecvStream(kSsrc)); |
| EXPECT_EQ(0u, decoder_factory_->decoders().size()); |
| } |
| |
| // Verifies that we can set up decoders. |
| TEST_F(WebRtcVideoEngineTest, RegisterH264DecoderIfSupported) { |
| // TODO(pbos): Do not assume that encoder/decoder support is symmetric. We |
| // can't even query the WebRtcVideoDecoderFactory for supported codecs. |
| // For now we add a FakeWebRtcVideoEncoderFactory to add H264 to supported |
| // codecs. |
| AddSupportedVideoCodecType("H264"); |
| std::vector<cricket::VideoCodec> codecs; |
| codecs.push_back(GetEngineCodec("H264")); |
| |
| std::unique_ptr<VideoMediaChannel> channel( |
| SetRecvParamsWithSupportedCodecs(codecs)); |
| |
| EXPECT_TRUE( |
| channel->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc))); |
| ASSERT_EQ(1u, decoder_factory_->decoders().size()); |
| } |
| |
| // Tests when GetSources is called with non-existing ssrc, it will return an |
| // empty list of RtpSource without crashing. |
| TEST_F(WebRtcVideoEngineTest, GetSourcesWithNonExistingSsrc) { |
| // Setup an recv stream with `kSsrc`. |
| AddSupportedVideoCodecType("VP8"); |
| cricket::VideoRecvParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| std::unique_ptr<VideoMediaChannel> channel( |
| SetRecvParamsWithSupportedCodecs(parameters.codecs)); |
| |
| EXPECT_TRUE( |
| channel->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc))); |
| |
| // Call GetSources with |kSsrc + 1| which doesn't exist. |
| std::vector<webrtc::RtpSource> sources = channel->GetSources(kSsrc + 1); |
| EXPECT_EQ(0u, sources.size()); |
| } |
| |
| TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, NullFactories) { |
| std::unique_ptr<webrtc::VideoEncoderFactory> encoder_factory; |
| std::unique_ptr<webrtc::VideoDecoderFactory> decoder_factory; |
| webrtc::FieldTrialBasedConfig trials; |
| WebRtcVideoEngine engine(std::move(encoder_factory), |
| std::move(decoder_factory), trials); |
| EXPECT_EQ(0u, engine.send_codecs().size()); |
| EXPECT_EQ(0u, engine.recv_codecs().size()); |
| } |
| |
| TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, EmptyFactories) { |
| // `engine` take ownership of the factories. |
| webrtc::MockVideoEncoderFactory* encoder_factory = |
| new webrtc::MockVideoEncoderFactory(); |
| webrtc::MockVideoDecoderFactory* decoder_factory = |
| new webrtc::MockVideoDecoderFactory(); |
| webrtc::FieldTrialBasedConfig trials; |
| WebRtcVideoEngine engine( |
| (std::unique_ptr<webrtc::VideoEncoderFactory>(encoder_factory)), |
| (std::unique_ptr<webrtc::VideoDecoderFactory>(decoder_factory)), trials); |
| // TODO(kron): Change to Times(1) once send and receive codecs are changed |
| // to be treated independently. |
| EXPECT_CALL(*encoder_factory, GetSupportedFormats()).Times(1); |
| EXPECT_EQ(0u, engine.send_codecs().size()); |
| EXPECT_EQ(0u, engine.recv_codecs().size()); |
| EXPECT_CALL(*encoder_factory, Die()); |
| EXPECT_CALL(*decoder_factory, Die()); |
| } |
| |
| // Test full behavior in the video engine when video codec factories of the new |
| // type are injected supporting the single codec Vp8. Check the returned codecs |
| // from the engine and that we will create a Vp8 encoder and decoder using the |
| // new factories. |
| TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, Vp8) { |
| // `engine` take ownership of the factories. |
| webrtc::MockVideoEncoderFactory* encoder_factory = |
| new webrtc::MockVideoEncoderFactory(); |
| webrtc::MockVideoDecoderFactory* decoder_factory = |
| new webrtc::MockVideoDecoderFactory(); |
| std::unique_ptr<webrtc::MockVideoBitrateAllocatorFactory> |
| rate_allocator_factory = |
| std::make_unique<webrtc::MockVideoBitrateAllocatorFactory>(); |
| EXPECT_CALL(*rate_allocator_factory, |
| CreateVideoBitrateAllocator(Field(&webrtc::VideoCodec::codecType, |
| webrtc::kVideoCodecVP8))) |
| .WillOnce( |
| [] { return std::make_unique<webrtc::MockVideoBitrateAllocator>(); }); |
| webrtc::FieldTrialBasedConfig trials; |
| WebRtcVideoEngine engine( |
| (std::unique_ptr<webrtc::VideoEncoderFactory>(encoder_factory)), |
| (std::unique_ptr<webrtc::VideoDecoderFactory>(decoder_factory)), trials); |
| const webrtc::SdpVideoFormat vp8_format("VP8"); |
| const std::vector<webrtc::SdpVideoFormat> supported_formats = {vp8_format}; |
| EXPECT_CALL(*encoder_factory, GetSupportedFormats()) |
| .WillRepeatedly(Return(supported_formats)); |
| EXPECT_CALL(*decoder_factory, GetSupportedFormats()) |
| .WillRepeatedly(Return(supported_formats)); |
| |
| // Verify the codecs from the engine. |
| const std::vector<VideoCodec> engine_codecs = engine.send_codecs(); |
| // Verify default codecs has been added correctly. |
| EXPECT_EQ(5u, engine_codecs.size()); |
| EXPECT_EQ("VP8", engine_codecs.at(0).name); |
| |
| // RTX codec for VP8. |
| EXPECT_EQ("rtx", engine_codecs.at(1).name); |
| int vp8_associated_payload; |
| EXPECT_TRUE(engine_codecs.at(1).GetParam(kCodecParamAssociatedPayloadType, |
| &vp8_associated_payload)); |
| EXPECT_EQ(vp8_associated_payload, engine_codecs.at(0).id); |
| |
| EXPECT_EQ(kRedCodecName, engine_codecs.at(2).name); |
| |
| // RTX codec for RED. |
| EXPECT_EQ("rtx", engine_codecs.at(3).name); |
| int red_associated_payload; |
| EXPECT_TRUE(engine_codecs.at(3).GetParam(kCodecParamAssociatedPayloadType, |
| &red_associated_payload)); |
| EXPECT_EQ(red_associated_payload, engine_codecs.at(2).id); |
| |
| EXPECT_EQ(kUlpfecCodecName, engine_codecs.at(4).name); |
| |
| int associated_payload_type; |
| EXPECT_TRUE(engine_codecs.at(1).GetParam( |
| cricket::kCodecParamAssociatedPayloadType, &associated_payload_type)); |
| EXPECT_EQ(engine_codecs.at(0).id, associated_payload_type); |
| // Verify default parameters has been added to the VP8 codec. |
| VerifyCodecHasDefaultFeedbackParams(engine_codecs.at(0), |
| /*lntf_expected=*/false); |
| |
| // Mock encoder creation. `engine` take ownership of the encoder. |
| webrtc::VideoEncoderFactory::CodecInfo codec_info; |
| codec_info.has_internal_source = false; |
| const webrtc::SdpVideoFormat format("VP8"); |
| EXPECT_CALL(*encoder_factory, QueryVideoEncoder(format)) |
| .WillRepeatedly(Return(codec_info)); |
| rtc::Event encoder_created; |
| EXPECT_CALL(*encoder_factory, CreateVideoEncoder(format)).WillOnce([&] { |
| encoder_created.Set(); |
| return std::make_unique<FakeWebRtcVideoEncoder>(nullptr); |
| }); |
| |
| // Mock decoder creation. `engine` take ownership of the decoder. |
| EXPECT_CALL(*decoder_factory, CreateVideoDecoder(format)).WillOnce([] { |
| return std::make_unique<FakeWebRtcVideoDecoder>(nullptr); |
| }); |
| |
| // Create a call. |
| webrtc::RtcEventLogNull event_log; |
| auto task_queue_factory = webrtc::CreateDefaultTaskQueueFactory(); |
| webrtc::Call::Config call_config(&event_log); |
| webrtc::FieldTrialBasedConfig field_trials; |
| call_config.trials = &field_trials; |
| call_config.task_queue_factory = task_queue_factory.get(); |
| const auto call = absl::WrapUnique(webrtc::Call::Create(call_config)); |
| |
| // Create send channel. |
| const int send_ssrc = 123; |
| std::unique_ptr<VideoMediaChannel> send_channel(engine.CreateMediaChannel( |
| call.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(), |
| rate_allocator_factory.get())); |
| cricket::VideoSendParameters send_parameters; |
| send_parameters.codecs.push_back(engine_codecs.at(0)); |
| EXPECT_TRUE(send_channel->SetSendParameters(send_parameters)); |
| send_channel->OnReadyToSend(true); |
| EXPECT_TRUE( |
| send_channel->AddSendStream(StreamParams::CreateLegacy(send_ssrc))); |
| EXPECT_TRUE(send_channel->SetSend(true)); |
| |
| // Set capturer. |
| webrtc::test::FrameForwarder frame_forwarder; |
| cricket::FakeFrameSource frame_source(1280, 720, |
| rtc::kNumMicrosecsPerSec / 30); |
| EXPECT_TRUE(send_channel->SetVideoSend(send_ssrc, nullptr, &frame_forwarder)); |
| // Sending one frame will allocate the encoder. |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| encoder_created.Wait(kTimeout); |
| |
| // Create recv channel. |
| const int recv_ssrc = 321; |
| std::unique_ptr<VideoMediaChannel> recv_channel(engine.CreateMediaChannel( |
| call.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(), |
| rate_allocator_factory.get())); |
| cricket::VideoRecvParameters recv_parameters; |
| recv_parameters.codecs.push_back(engine_codecs.at(0)); |
| EXPECT_TRUE(recv_channel->SetRecvParameters(recv_parameters)); |
| EXPECT_TRUE(recv_channel->AddRecvStream( |
| cricket::StreamParams::CreateLegacy(recv_ssrc))); |
| |
| // Remove streams previously added to free the encoder and decoder instance. |
| EXPECT_CALL(*encoder_factory, Die()); |
| EXPECT_CALL(*decoder_factory, Die()); |
| EXPECT_CALL(*rate_allocator_factory, Die()); |
| EXPECT_TRUE(send_channel->RemoveSendStream(send_ssrc)); |
| EXPECT_TRUE(recv_channel->RemoveRecvStream(recv_ssrc)); |
| } |
| |
| // Test behavior when decoder factory fails to create a decoder (returns null). |
| TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, NullDecoder) { |
| // `engine` take ownership of the factories. |
| webrtc::MockVideoEncoderFactory* encoder_factory = |
| new webrtc::MockVideoEncoderFactory(); |
| webrtc::MockVideoDecoderFactory* decoder_factory = |
| new webrtc::MockVideoDecoderFactory(); |
| std::unique_ptr<webrtc::MockVideoBitrateAllocatorFactory> |
| rate_allocator_factory = |
| std::make_unique<webrtc::MockVideoBitrateAllocatorFactory>(); |
| webrtc::FieldTrialBasedConfig trials; |
| WebRtcVideoEngine engine( |
| (std::unique_ptr<webrtc::VideoEncoderFactory>(encoder_factory)), |
| (std::unique_ptr<webrtc::VideoDecoderFactory>(decoder_factory)), trials); |
| const webrtc::SdpVideoFormat vp8_format("VP8"); |
| const std::vector<webrtc::SdpVideoFormat> supported_formats = {vp8_format}; |
| EXPECT_CALL(*encoder_factory, GetSupportedFormats()) |
| .WillRepeatedly(Return(supported_formats)); |
| |
| // Decoder creation fails. |
| EXPECT_CALL(*decoder_factory, CreateVideoDecoder).WillOnce([] { |
| return nullptr; |
| }); |
| |
| // Create a call. |
| webrtc::RtcEventLogNull event_log; |
| auto task_queue_factory = webrtc::CreateDefaultTaskQueueFactory(); |
| webrtc::Call::Config call_config(&event_log); |
| webrtc::FieldTrialBasedConfig field_trials; |
| call_config.trials = &field_trials; |
| call_config.task_queue_factory = task_queue_factory.get(); |
| const auto call = absl::WrapUnique(webrtc::Call::Create(call_config)); |
| |
| // Create recv channel. |
| EXPECT_CALL(*decoder_factory, GetSupportedFormats()) |
| .WillRepeatedly(::testing::Return(supported_formats)); |
| const int recv_ssrc = 321; |
| std::unique_ptr<VideoMediaChannel> recv_channel(engine.CreateMediaChannel( |
| call.get(), GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(), |
| rate_allocator_factory.get())); |
| cricket::VideoRecvParameters recv_parameters; |
| recv_parameters.codecs.push_back(engine.recv_codecs().front()); |
| EXPECT_TRUE(recv_channel->SetRecvParameters(recv_parameters)); |
| EXPECT_TRUE(recv_channel->AddRecvStream( |
| cricket::StreamParams::CreateLegacy(recv_ssrc))); |
| |
| // Remove streams previously added to free the encoder and decoder instance. |
| EXPECT_TRUE(recv_channel->RemoveRecvStream(recv_ssrc)); |
| } |
| |
| TEST_F(WebRtcVideoEngineTest, DISABLED_RecreatesEncoderOnContentTypeChange) { |
| encoder_factory_->AddSupportedVideoCodecType("VP8"); |
| std::unique_ptr<FakeCall> fake_call(new FakeCall()); |
| std::unique_ptr<VideoMediaChannel> channel( |
| SetSendParamsWithAllSupportedCodecs()); |
| ASSERT_TRUE( |
| channel->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc))); |
| cricket::VideoCodec codec = GetEngineCodec("VP8"); |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(codec); |
| channel->OnReadyToSend(true); |
| channel->SetSend(true); |
| ASSERT_TRUE(channel->SetSendParameters(parameters)); |
| |
| webrtc::test::FrameForwarder frame_forwarder; |
| cricket::FakeFrameSource frame_source(1280, 720, |
| rtc::kNumMicrosecsPerSec / 30); |
| VideoOptions options; |
| EXPECT_TRUE(channel->SetVideoSend(kSsrc, &options, &frame_forwarder)); |
| |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(1)); |
| EXPECT_EQ(webrtc::VideoCodecMode::kRealtimeVideo, |
| encoder_factory_->encoders().back()->GetCodecSettings().mode); |
| |
| EXPECT_TRUE(channel->SetVideoSend(kSsrc, &options, &frame_forwarder)); |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| // No change in content type, keep current encoder. |
| EXPECT_EQ(1, encoder_factory_->GetNumCreatedEncoders()); |
| |
| options.is_screencast.emplace(true); |
| EXPECT_TRUE(channel->SetVideoSend(kSsrc, &options, &frame_forwarder)); |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| // Change to screen content, recreate encoder. For the simulcast encoder |
| // adapter case, this will result in two calls since InitEncode triggers a |
| // a new instance. |
| ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(2)); |
| EXPECT_EQ(webrtc::VideoCodecMode::kScreensharing, |
| encoder_factory_->encoders().back()->GetCodecSettings().mode); |
| |
| EXPECT_TRUE(channel->SetVideoSend(kSsrc, &options, &frame_forwarder)); |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| // Still screen content, no need to update encoder. |
| EXPECT_EQ(2, encoder_factory_->GetNumCreatedEncoders()); |
| |
| options.is_screencast.emplace(false); |
| options.video_noise_reduction.emplace(false); |
| EXPECT_TRUE(channel->SetVideoSend(kSsrc, &options, &frame_forwarder)); |
| // Change back to regular video content, update encoder. Also change |
| // a non `is_screencast` option just to verify it doesn't affect recreation. |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(3)); |
| EXPECT_EQ(webrtc::VideoCodecMode::kRealtimeVideo, |
| encoder_factory_->encoders().back()->GetCodecSettings().mode); |
| |
| // Remove stream previously added to free the external encoder instance. |
| EXPECT_TRUE(channel->RemoveSendStream(kSsrc)); |
| EXPECT_EQ(0u, encoder_factory_->encoders().size()); |
| } |
| |
| class WebRtcVideoChannelEncodedFrameCallbackTest : public ::testing::Test { |
| protected: |
| webrtc::Call::Config GetCallConfig( |
| webrtc::RtcEventLogNull* event_log, |
| webrtc::TaskQueueFactory* task_queue_factory) { |
| webrtc::Call::Config call_config(event_log); |
| call_config.task_queue_factory = task_queue_factory; |
| call_config.trials = &field_trials_; |
| return call_config; |
| } |
| |
| WebRtcVideoChannelEncodedFrameCallbackTest() |
| : task_queue_factory_(webrtc::CreateDefaultTaskQueueFactory()), |
| call_(absl::WrapUnique(webrtc::Call::Create( |
| GetCallConfig(&event_log_, task_queue_factory_.get())))), |
| video_bitrate_allocator_factory_( |
| webrtc::CreateBuiltinVideoBitrateAllocatorFactory()), |
| engine_( |
| webrtc::CreateBuiltinVideoEncoderFactory(), |
| std::make_unique<webrtc::test::FunctionVideoDecoderFactory>( |
| []() { return std::make_unique<webrtc::test::FakeDecoder>(); }, |
| kSdpVideoFormats), |
| field_trials_), |
| channel_(absl::WrapUnique(static_cast<cricket::WebRtcVideoChannel*>( |
| engine_.CreateMediaChannel( |
| call_.get(), |
| cricket::MediaConfig(), |
| cricket::VideoOptions(), |
| webrtc::CryptoOptions(), |
| video_bitrate_allocator_factory_.get())))) { |
| network_interface_.SetDestination(channel_.get()); |
| channel_->SetInterface(&network_interface_); |
| cricket::VideoRecvParameters parameters; |
| parameters.codecs = engine_.recv_codecs(); |
| channel_->SetRecvParameters(parameters); |
| } |
| |
| ~WebRtcVideoChannelEncodedFrameCallbackTest() override { |
| channel_->SetInterface(nullptr); |
| } |
| |
| void DeliverKeyFrame(uint32_t ssrc) { |
| RtpPacket packet; |
| packet.SetMarker(true); |
| packet.SetPayloadType(96); // VP8 |
| packet.SetSsrc(ssrc); |
| |
| // VP8 Keyframe + 1 byte payload |
| uint8_t* buf_ptr = packet.AllocatePayload(11); |
| memset(buf_ptr, 0, 11); // Pass MSAN (don't care about bytes 1-9) |
| buf_ptr[0] = 0x10; // Partition ID 0 + beginning of partition. |
| constexpr unsigned width = 1080; |
| constexpr unsigned height = 720; |
| buf_ptr[6] = width & 255; |
| buf_ptr[7] = width >> 8; |
| buf_ptr[8] = height & 255; |
| buf_ptr[9] = height >> 8; |
| |
| call_->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO, packet.Buffer(), |
| /*packet_time_us=*/0); |
| } |
| |
| void DeliverKeyFrameAndWait(uint32_t ssrc) { |
| DeliverKeyFrame(ssrc); |
| EXPECT_EQ_WAIT(1, renderer_.num_rendered_frames(), kTimeout); |
| EXPECT_EQ(0, renderer_.errors()); |
| } |
| |
| static const std::vector<webrtc::SdpVideoFormat> kSdpVideoFormats; |
| webrtc::FieldTrialBasedConfig field_trials_; |
| webrtc::RtcEventLogNull event_log_; |
| std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory_; |
| std::unique_ptr<webrtc::Call> call_; |
| std::unique_ptr<webrtc::VideoBitrateAllocatorFactory> |
| video_bitrate_allocator_factory_; |
| WebRtcVideoEngine engine_; |
| std::unique_ptr<WebRtcVideoChannel> channel_; |
| cricket::FakeNetworkInterface network_interface_; |
| cricket::FakeVideoRenderer renderer_; |
| }; |
| |
| const std::vector<webrtc::SdpVideoFormat> |
| WebRtcVideoChannelEncodedFrameCallbackTest::kSdpVideoFormats = { |
| webrtc::SdpVideoFormat("VP8")}; |
| |
| TEST_F(WebRtcVideoChannelEncodedFrameCallbackTest, |
| SetEncodedFrameBufferFunction_DefaultStream) { |
| testing::MockFunction<void(const webrtc::RecordableEncodedFrame&)> callback; |
| EXPECT_CALL(callback, Call); |
| EXPECT_TRUE(channel_->AddRecvStream( |
| cricket::StreamParams::CreateLegacy(kSsrc), /*is_default_stream=*/true)); |
| channel_->SetRecordableEncodedFrameCallback(/*ssrc=*/0, |
| callback.AsStdFunction()); |
| EXPECT_TRUE(channel_->SetSink(kSsrc, &renderer_)); |
| DeliverKeyFrame(kSsrc); |
| EXPECT_EQ_WAIT(1, renderer_.num_rendered_frames(), kTimeout); |
| EXPECT_EQ(0, renderer_.errors()); |
| channel_->RemoveRecvStream(kSsrc); |
| } |
| |
| TEST_F(WebRtcVideoChannelEncodedFrameCallbackTest, |
| SetEncodedFrameBufferFunction_MatchSsrcWithDefaultStream) { |
| testing::MockFunction<void(const webrtc::RecordableEncodedFrame&)> callback; |
| EXPECT_CALL(callback, Call); |
| EXPECT_TRUE(channel_->AddRecvStream( |
| cricket::StreamParams::CreateLegacy(kSsrc), /*is_default_stream=*/true)); |
| EXPECT_TRUE(channel_->SetSink(kSsrc, &renderer_)); |
| channel_->SetRecordableEncodedFrameCallback(kSsrc, callback.AsStdFunction()); |
| DeliverKeyFrame(kSsrc); |
| EXPECT_EQ_WAIT(1, renderer_.num_rendered_frames(), kTimeout); |
| EXPECT_EQ(0, renderer_.errors()); |
| channel_->RemoveRecvStream(kSsrc); |
| } |
| |
| TEST_F(WebRtcVideoChannelEncodedFrameCallbackTest, |
| SetEncodedFrameBufferFunction_MatchSsrc) { |
| testing::MockFunction<void(const webrtc::RecordableEncodedFrame&)> callback; |
| EXPECT_CALL(callback, Call); |
| EXPECT_TRUE(channel_->AddRecvStream( |
| cricket::StreamParams::CreateLegacy(kSsrc), /*is_default_stream=*/false)); |
| EXPECT_TRUE(channel_->SetSink(kSsrc, &renderer_)); |
| channel_->SetRecordableEncodedFrameCallback(kSsrc, callback.AsStdFunction()); |
| DeliverKeyFrame(kSsrc); |
| EXPECT_EQ_WAIT(1, renderer_.num_rendered_frames(), kTimeout); |
| EXPECT_EQ(0, renderer_.errors()); |
| channel_->RemoveRecvStream(kSsrc); |
| } |
| |
| TEST_F(WebRtcVideoChannelEncodedFrameCallbackTest, |
| SetEncodedFrameBufferFunction_MismatchSsrc) { |
| testing::StrictMock< |
| testing::MockFunction<void(const webrtc::RecordableEncodedFrame&)>> |
| callback; |
| EXPECT_TRUE( |
| channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc + 1), |
| /*is_default_stream=*/false)); |
| EXPECT_TRUE(channel_->SetSink(kSsrc + 1, &renderer_)); |
| channel_->SetRecordableEncodedFrameCallback(kSsrc, callback.AsStdFunction()); |
| DeliverKeyFrame(kSsrc); // Expected to not cause function to fire. |
| DeliverKeyFrameAndWait(kSsrc + 1); |
| channel_->RemoveRecvStream(kSsrc + 1); |
| } |
| |
| TEST_F(WebRtcVideoChannelEncodedFrameCallbackTest, |
| SetEncodedFrameBufferFunction_MismatchSsrcWithDefaultStream) { |
| testing::StrictMock< |
| testing::MockFunction<void(const webrtc::RecordableEncodedFrame&)>> |
| callback; |
| EXPECT_TRUE( |
| channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc + 1), |
| /*is_default_stream=*/true)); |
| EXPECT_TRUE(channel_->SetSink(kSsrc + 1, &renderer_)); |
| channel_->SetRecordableEncodedFrameCallback(kSsrc, callback.AsStdFunction()); |
| DeliverKeyFrame(kSsrc); // Expected to not cause function to fire. |
| DeliverKeyFrameAndWait(kSsrc + 1); |
| channel_->RemoveRecvStream(kSsrc + 1); |
| } |
| |
| class WebRtcVideoChannelBaseTest : public ::testing::Test { |
| protected: |
| WebRtcVideoChannelBaseTest() |
| : task_queue_factory_(webrtc::CreateDefaultTaskQueueFactory()), |
| video_bitrate_allocator_factory_( |
| webrtc::CreateBuiltinVideoBitrateAllocatorFactory()), |
| engine_(webrtc::CreateBuiltinVideoEncoderFactory(), |
| webrtc::CreateBuiltinVideoDecoderFactory(), |
| field_trials_) {} |
| |
| void SetUp() override { |
| // One testcase calls SetUp in a loop, only create call_ once. |
| if (!call_) { |
| webrtc::Call::Config call_config(&event_log_); |
| call_config.task_queue_factory = task_queue_factory_.get(); |
| call_config.trials = &field_trials_; |
| call_.reset(webrtc::Call::Create(call_config)); |
| } |
| cricket::MediaConfig media_config; |
| // Disabling cpu overuse detection actually disables quality scaling too; it |
| // implies DegradationPreference kMaintainResolution. Automatic scaling |
| // needs to be disabled, otherwise, tests which check the size of received |
| // frames become flaky. |
| media_config.video.enable_cpu_adaptation = false; |
| channel_.reset( |
| static_cast<cricket::WebRtcVideoChannel*>(engine_.CreateMediaChannel( |
| call_.get(), media_config, cricket::VideoOptions(), |
| webrtc::CryptoOptions(), video_bitrate_allocator_factory_.get()))); |
| channel_->OnReadyToSend(true); |
| EXPECT_TRUE(channel_.get() != NULL); |
| network_interface_.SetDestination(channel_.get()); |
| channel_->SetInterface(&network_interface_); |
| cricket::VideoRecvParameters parameters; |
| parameters.codecs = engine_.send_codecs(); |
| channel_->SetRecvParameters(parameters); |
| EXPECT_TRUE(channel_->AddSendStream(DefaultSendStreamParams())); |
| frame_forwarder_ = std::make_unique<webrtc::test::FrameForwarder>(); |
| frame_source_ = std::make_unique<cricket::FakeFrameSource>( |
| 640, 480, rtc::kNumMicrosecsPerSec / kFramerate); |
| EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, frame_forwarder_.get())); |
| } |
| |
| // Utility method to setup an additional stream to send and receive video. |
| // Used to test send and recv between two streams. |
| void SetUpSecondStream() { |
| SetUpSecondStreamWithNoRecv(); |
| // Setup recv for second stream. |
| EXPECT_TRUE(channel_->AddRecvStream( |
| cricket::StreamParams::CreateLegacy(kSsrc + 2))); |
| // Make the second renderer available for use by a new stream. |
| EXPECT_TRUE(channel_->SetSink(kSsrc + 2, &renderer2_)); |
| } |
| |
| // Setup an additional stream just to send video. Defer add recv stream. |
| // This is required if you want to test unsignalled recv of video rtp packets. |
| void SetUpSecondStreamWithNoRecv() { |
| // SetUp() already added kSsrc make sure duplicate SSRCs cant be added. |
| EXPECT_TRUE( |
| channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc))); |
| EXPECT_TRUE(channel_->SetSink(kSsrc, &renderer_)); |
| EXPECT_FALSE( |
| channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc))); |
| EXPECT_TRUE(channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(kSsrc + 2))); |
| // We dont add recv for the second stream. |
| |
| // Setup the receive and renderer for second stream after send. |
| frame_forwarder_2_ = std::make_unique<webrtc::test::FrameForwarder>(); |
| EXPECT_TRUE( |
| channel_->SetVideoSend(kSsrc + 2, nullptr, frame_forwarder_2_.get())); |
| } |
| |
| void TearDown() override { |
| channel_->SetInterface(nullptr); |
| channel_.reset(); |
| } |
| |
| void ResetTest() { |
| TearDown(); |
| SetUp(); |
| } |
| |
| bool SetDefaultCodec() { return SetOneCodec(DefaultCodec()); } |
| |
| bool SetOneCodec(const cricket::VideoCodec& codec) { |
| frame_source_ = std::make_unique<cricket::FakeFrameSource>( |
| kVideoWidth, kVideoHeight, rtc::kNumMicrosecsPerSec / kFramerate); |
| |
| bool sending = channel_->sending(); |
| bool success = SetSend(false); |
| if (success) { |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(codec); |
| success = channel_->SetSendParameters(parameters); |
| } |
| if (success) { |
| success = SetSend(sending); |
| } |
| return success; |
| } |
| bool SetSend(bool send) { return channel_->SetSend(send); } |
| void SendFrame() { |
| if (frame_forwarder_2_) { |
| frame_forwarder_2_->IncomingCapturedFrame(frame_source_->GetFrame()); |
| } |
| frame_forwarder_->IncomingCapturedFrame(frame_source_->GetFrame()); |
| } |
| bool WaitAndSendFrame(int wait_ms) { |
| bool ret = rtc::Thread::Current()->ProcessMessages(wait_ms); |
| SendFrame(); |
| return ret; |
| } |
| int NumRtpBytes() { return network_interface_.NumRtpBytes(); } |
| int NumRtpBytes(uint32_t ssrc) { |
| return network_interface_.NumRtpBytes(ssrc); |
| } |
| int NumRtpPackets() { return network_interface_.NumRtpPackets(); } |
| int NumRtpPackets(uint32_t ssrc) { |
| return network_interface_.NumRtpPackets(ssrc); |
| } |
| int NumSentSsrcs() { return network_interface_.NumSentSsrcs(); } |
| rtc::CopyOnWriteBuffer GetRtpPacket(int index) { |
| return network_interface_.GetRtpPacket(index); |
| } |
| static int GetPayloadType(rtc::CopyOnWriteBuffer p) { |
| RtpPacket header; |
| EXPECT_TRUE(header.Parse(std::move(p))); |
| return header.PayloadType(); |
| } |
| |
| // Tests that we can send and receive frames. |
| void SendAndReceive(const cricket::VideoCodec& codec) { |
| EXPECT_TRUE(SetOneCodec(codec)); |
| EXPECT_TRUE(SetSend(true)); |
| channel_->SetDefaultSink(&renderer_); |
| EXPECT_EQ(0, renderer_.num_rendered_frames()); |
| SendFrame(); |
| EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout); |
| EXPECT_EQ(codec.id, GetPayloadType(GetRtpPacket(0))); |
| } |
| |
| void SendReceiveManyAndGetStats(const cricket::VideoCodec& codec, |
| int duration_sec, |
| int fps) { |
| EXPECT_TRUE(SetOneCodec(codec)); |
| EXPECT_TRUE(SetSend(true)); |
| channel_->SetDefaultSink(&renderer_); |
| EXPECT_EQ(0, renderer_.num_rendered_frames()); |
| for (int i = 0; i < duration_sec; ++i) { |
| for (int frame = 1; frame <= fps; ++frame) { |
| EXPECT_TRUE(WaitAndSendFrame(1000 / fps)); |
| EXPECT_FRAME_WAIT(frame + i * fps, kVideoWidth, kVideoHeight, kTimeout); |
| } |
| } |
| EXPECT_EQ(codec.id, GetPayloadType(GetRtpPacket(0))); |
| } |
| |
| cricket::VideoSenderInfo GetSenderStats(size_t i) { |
| cricket::VideoMediaInfo info; |
| EXPECT_TRUE(channel_->GetStats(&info)); |
| return info.senders[i]; |
| } |
| |
| cricket::VideoReceiverInfo GetReceiverStats(size_t i) { |
| cricket::VideoMediaInfo info; |
| EXPECT_TRUE(channel_->GetStats(&info)); |
| return info.receivers[i]; |
| } |
| |
| // Two streams one channel tests. |
| |
| // Tests that we can send and receive frames. |
| void TwoStreamsSendAndReceive(const cricket::VideoCodec& codec) { |
| SetUpSecondStream(); |
| // Test sending and receiving on first stream. |
| SendAndReceive(codec); |
| // Test sending and receiving on second stream. |
| EXPECT_EQ_WAIT(1, renderer2_.num_rendered_frames(), kTimeout); |
| EXPECT_GT(NumRtpPackets(), 0); |
| EXPECT_EQ(1, renderer2_.num_rendered_frames()); |
| } |
| |
| cricket::VideoCodec GetEngineCodec(const std::string& name) { |
| for (const cricket::VideoCodec& engine_codec : engine_.send_codecs()) { |
| if (absl::EqualsIgnoreCase(name, engine_codec.name)) |
| return engine_codec; |
| } |
| // This point should never be reached. |
| ADD_FAILURE() << "Unrecognized codec name: " << name; |
| return cricket::VideoCodec(); |
| } |
| |
| cricket::VideoCodec DefaultCodec() { return GetEngineCodec("VP8"); } |
| |
| cricket::StreamParams DefaultSendStreamParams() { |
| return cricket::StreamParams::CreateLegacy(kSsrc); |
| } |
| |
| webrtc::RtcEventLogNull event_log_; |
| webrtc::FieldTrialBasedConfig field_trials_; |
| std::unique_ptr<webrtc::test::ScopedFieldTrials> override_field_trials_; |
| std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory_; |
| std::unique_ptr<webrtc::Call> call_; |
| std::unique_ptr<webrtc::VideoBitrateAllocatorFactory> |
| video_bitrate_allocator_factory_; |
| WebRtcVideoEngine engine_; |
| |
| std::unique_ptr<cricket::FakeFrameSource> frame_source_; |
| std::unique_ptr<webrtc::test::FrameForwarder> frame_forwarder_; |
| std::unique_ptr<webrtc::test::FrameForwarder> frame_forwarder_2_; |
| |
| std::unique_ptr<WebRtcVideoChannel> channel_; |
| cricket::FakeNetworkInterface network_interface_; |
| cricket::FakeVideoRenderer renderer_; |
| |
| // Used by test cases where 2 streams are run on the same channel. |
| cricket::FakeVideoRenderer renderer2_; |
| }; |
| |
| // Test that SetSend works. |
| TEST_F(WebRtcVideoChannelBaseTest, SetSend) { |
| EXPECT_FALSE(channel_->sending()); |
| EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, frame_forwarder_.get())); |
| EXPECT_TRUE(SetOneCodec(DefaultCodec())); |
| EXPECT_FALSE(channel_->sending()); |
| EXPECT_TRUE(SetSend(true)); |
| EXPECT_TRUE(channel_->sending()); |
| SendFrame(); |
| EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout); |
| EXPECT_TRUE(SetSend(false)); |
| EXPECT_FALSE(channel_->sending()); |
| } |
| |
| // Test that SetSend fails without codecs being set. |
| TEST_F(WebRtcVideoChannelBaseTest, SetSendWithoutCodecs) { |
| EXPECT_FALSE(channel_->sending()); |
| EXPECT_FALSE(SetSend(true)); |
| EXPECT_FALSE(channel_->sending()); |
| } |
| |
| // Test that we properly set the send and recv buffer sizes by the time |
| // SetSend is called. |
| TEST_F(WebRtcVideoChannelBaseTest, SetSendSetsTransportBufferSizes) { |
| EXPECT_TRUE(SetOneCodec(DefaultCodec())); |
| EXPECT_TRUE(SetSend(true)); |
| EXPECT_EQ(64 * 1024, network_interface_.sendbuf_size()); |
| EXPECT_EQ(256 * 1024, network_interface_.recvbuf_size()); |
| } |
| |
| // Test that we properly set the send and recv buffer sizes when overriding |
| // via field trials. |
| TEST_F(WebRtcVideoChannelBaseTest, OverridesRecvBufferSize) { |
| // Set field trial to override the default recv buffer size, and then re-run |
| // setup where the interface is created and configured. |
| const int kCustomRecvBufferSize = 123456; |
| RTC_DCHECK(!override_field_trials_); |
| override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>( |
| "WebRTC-IncreasedReceivebuffers/123456/"); |
| |
| ResetTest(); |
| |
| EXPECT_TRUE(SetOneCodec(DefaultCodec())); |
| EXPECT_TRUE(SetSend(true)); |
| EXPECT_EQ(64 * 1024, network_interface_.sendbuf_size()); |
| EXPECT_EQ(kCustomRecvBufferSize, network_interface_.recvbuf_size()); |
| } |
| |
| // Test that we properly set the send and recv buffer sizes when overriding |
| // via field trials with suffix. |
| TEST_F(WebRtcVideoChannelBaseTest, OverridesRecvBufferSizeWithSuffix) { |
| // Set field trial to override the default recv buffer size, and then re-run |
| // setup where the interface is created and configured. |
| const int kCustomRecvBufferSize = 123456; |
| RTC_DCHECK(!override_field_trials_); |
| override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>( |
| "WebRTC-IncreasedReceivebuffers/123456_Dogfood/"); |
| ResetTest(); |
| |
| EXPECT_TRUE(SetOneCodec(DefaultCodec())); |
| EXPECT_TRUE(SetSend(true)); |
| EXPECT_EQ(64 * 1024, network_interface_.sendbuf_size()); |
| EXPECT_EQ(kCustomRecvBufferSize, network_interface_.recvbuf_size()); |
| } |
| |
| // Test that we properly set the send and recv buffer sizes when overriding |
| // via field trials that don't make any sense. |
| TEST_F(WebRtcVideoChannelBaseTest, InvalidRecvBufferSize) { |
| // Set bogus field trial values to override the default recv buffer size, and |
| // then re-run setup where the interface is created and configured. The |
| // default value should still be used. |
| |
| const char* prev_field_trials = webrtc::field_trial::GetFieldTrialString(); |
| |
| std::string field_trial_string; |
| for (std::string group : {" ", "NotANumber", "-1", "0"}) { |
| std::string trial_string = "WebRTC-IncreasedReceivebuffers/"; |
| trial_string += group; |
| trial_string += "/"; |
| |
| // Dear reader. Sorry for this... it's a bit of a mess. |
| // TODO(bugs.webrtc.org/12854): This test needs to be rewritten to not use |
| // ResetTest and changing global field trials in a loop. |
| TearDown(); |
| // This is a hack to appease tsan. Because of the way the test is written |
| // active state within Call, including running task queues may race with |
| // the test changing the global field trial variable. |
| // This particular hack, pauses the transport controller TQ while we |
| // change the field trial. |
| rtc::TaskQueue* tq = call_->GetTransportControllerSend()->GetWorkerQueue(); |
| rtc::Event waiting, resume; |
| tq->PostTask([&waiting, &resume]() { |
| waiting.Set(); |
| resume.Wait(rtc::Event::kForever); |
| }); |
| |
| waiting.Wait(rtc::Event::kForever); |
| field_trial_string = std::move(trial_string); |
| webrtc::field_trial::InitFieldTrialsFromString(field_trial_string.c_str()); |
| |
| SetUp(); |
| resume.Set(); |
| |
| // OK, now the test can carry on. |
| |
| EXPECT_TRUE(SetOneCodec(DefaultCodec())); |
| EXPECT_TRUE(SetSend(true)); |
| EXPECT_EQ(64 * 1024, network_interface_.sendbuf_size()); |
| EXPECT_EQ(256 * 1024, network_interface_.recvbuf_size()); |
| } |
| |
| webrtc::field_trial::InitFieldTrialsFromString(prev_field_trials); |
| } |
| |
| // Test that stats work properly for a 1-1 call. |
| TEST_F(WebRtcVideoChannelBaseTest, GetStats) { |
| const int kDurationSec = 3; |
| const int kFps = 10; |
| SendReceiveManyAndGetStats(DefaultCodec(), kDurationSec, kFps); |
| |
| cricket::VideoMediaInfo info; |
| EXPECT_TRUE(channel_->GetStats(&info)); |
| |
| ASSERT_EQ(1U, info.senders.size()); |
| // TODO(whyuan): bytes_sent and bytes_rcvd are different. Are both payload? |
| // For webrtc, bytes_sent does not include the RTP header length. |
| EXPECT_EQ(info.senders[0].payload_bytes_sent, |
| NumRtpBytes() - kRtpHeaderSize * NumRtpPackets()); |
| EXPECT_EQ(NumRtpPackets(), info.senders[0].packets_sent); |
| EXPECT_EQ(0.0, info.senders[0].fraction_lost); |
| ASSERT_TRUE(info.senders[0].codec_payload_type); |
| EXPECT_EQ(DefaultCodec().id, *info.senders[0].codec_payload_type); |
| EXPECT_EQ(0, info.senders[0].firs_rcvd); |
| EXPECT_EQ(0, info.senders[0].plis_rcvd); |
| EXPECT_EQ(0u, info.senders[0].nacks_rcvd); |
| EXPECT_EQ(kVideoWidth, info.senders[0].send_frame_width); |
| EXPECT_EQ(kVideoHeight, info.senders[0].send_frame_height); |
| EXPECT_GT(info.senders[0].framerate_input, 0); |
| EXPECT_GT(info.senders[0].framerate_sent, 0); |
| |
| EXPECT_EQ(1U, info.send_codecs.count(DefaultCodec().id)); |
| EXPECT_EQ(DefaultCodec().ToCodecParameters(), |
| info.send_codecs[DefaultCodec().id]); |
| |
| ASSERT_EQ(1U, info.receivers.size()); |
| EXPECT_EQ(1U, info.senders[0].ssrcs().size()); |
| EXPECT_EQ(1U, info.receivers[0].ssrcs().size()); |
| EXPECT_EQ(info.senders[0].ssrcs()[0], info.receivers[0].ssrcs()[0]); |
| ASSERT_TRUE(info.receivers[0].codec_payload_type); |
| EXPECT_EQ(DefaultCodec().id, *info.receivers[0].codec_payload_type); |
| EXPECT_EQ(NumRtpBytes() - kRtpHeaderSize * NumRtpPackets(), |
| info.receivers[0].payload_bytes_rcvd); |
| EXPECT_EQ(NumRtpPackets(), info.receivers[0].packets_rcvd); |
| EXPECT_EQ(0, info.receivers[0].packets_lost); |
| // TODO(asapersson): Not set for webrtc. Handle missing stats. |
| // EXPECT_EQ(0, info.receivers[0].packets_concealed); |
| EXPECT_EQ(0, info.receivers[0].firs_sent); |
| EXPECT_EQ(0, info.receivers[0].plis_sent); |
| EXPECT_EQ(0U, info.receivers[0].nacks_sent); |
| EXPECT_EQ(kVideoWidth, info.receivers[0].frame_width); |
| EXPECT_EQ(kVideoHeight, info.receivers[0].frame_height); |
| EXPECT_GT(info.receivers[0].framerate_rcvd, 0); |
| EXPECT_GT(info.receivers[0].framerate_decoded, 0); |
| EXPECT_GT(info.receivers[0].framerate_output, 0); |
| |
| EXPECT_EQ(1U, info.receive_codecs.count(DefaultCodec().id)); |
| EXPECT_EQ(DefaultCodec().ToCodecParameters(), |
| info.receive_codecs[DefaultCodec().id]); |
| } |
| |
| // Test that stats work properly for a conf call with multiple recv streams. |
| TEST_F(WebRtcVideoChannelBaseTest, GetStatsMultipleRecvStreams) { |
| cricket::FakeVideoRenderer renderer1, renderer2; |
| EXPECT_TRUE(SetOneCodec(DefaultCodec())); |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(DefaultCodec()); |
| parameters.conference_mode = true; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_TRUE(SetSend(true)); |
| EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(1))); |
| EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(2))); |
| EXPECT_TRUE(channel_->SetSink(1, &renderer1)); |
| EXPECT_TRUE(channel_->SetSink(2, &renderer2)); |
| EXPECT_EQ(0, renderer1.num_rendered_frames()); |
| EXPECT_EQ(0, renderer2.num_rendered_frames()); |
| std::vector<uint32_t> ssrcs; |
| ssrcs.push_back(1); |
| ssrcs.push_back(2); |
| network_interface_.SetConferenceMode(true, ssrcs); |
| SendFrame(); |
| EXPECT_FRAME_ON_RENDERER_WAIT(renderer1, 1, kVideoWidth, kVideoHeight, |
| kTimeout); |
| EXPECT_FRAME_ON_RENDERER_WAIT(renderer2, 1, kVideoWidth, kVideoHeight, |
| kTimeout); |
| |
| EXPECT_TRUE(channel_->SetSend(false)); |
| |
| cricket::VideoMediaInfo info; |
| EXPECT_TRUE(channel_->GetStats(&info)); |
| ASSERT_EQ(1U, info.senders.size()); |
| // TODO(whyuan): bytes_sent and bytes_rcvd are different. Are both payload? |
| // For webrtc, bytes_sent does not include the RTP header length. |
| EXPECT_EQ_WAIT(NumRtpBytes() - kRtpHeaderSize * NumRtpPackets(), |
| GetSenderStats(0).payload_bytes_sent, kTimeout); |
| EXPECT_EQ_WAIT(NumRtpPackets(), GetSenderStats(0).packets_sent, kTimeout); |
| EXPECT_EQ(kVideoWidth, GetSenderStats(0).send_frame_width); |
| EXPECT_EQ(kVideoHeight, GetSenderStats(0).send_frame_height); |
| |
| ASSERT_EQ(2U, info.receivers.size()); |
| for (size_t i = 0; i < info.receivers.size(); ++i) { |
| EXPECT_EQ(1U, GetReceiverStats(i).ssrcs().size()); |
| EXPECT_EQ(i + 1, GetReceiverStats(i).ssrcs()[0]); |
| EXPECT_EQ_WAIT(NumRtpBytes() - kRtpHeaderSize * NumRtpPackets(), |
| GetReceiverStats(i).payload_bytes_rcvd, kTimeout); |
| EXPECT_EQ_WAIT(NumRtpPackets(), GetReceiverStats(i).packets_rcvd, kTimeout); |
| EXPECT_EQ_WAIT(kVideoWidth, GetReceiverStats(i).frame_width, kTimeout); |
| EXPECT_EQ_WAIT(kVideoHeight, GetReceiverStats(i).frame_height, kTimeout); |
| } |
| } |
| |
| // Test that stats work properly for a conf call with multiple send streams. |
| TEST_F(WebRtcVideoChannelBaseTest, GetStatsMultipleSendStreams) { |
| // Normal setup; note that we set the SSRC explicitly to ensure that |
| // it will come first in the senders map. |
| EXPECT_TRUE(SetOneCodec(DefaultCodec())); |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(DefaultCodec()); |
| parameters.conference_mode = true; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_TRUE( |
| channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc))); |
| EXPECT_TRUE(channel_->SetSink(kSsrc, &renderer_)); |
| EXPECT_TRUE(SetSend(true)); |
| SendFrame(); |
| EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout); |
| EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout); |
| |
| // Add an additional capturer, and hook up a renderer to receive it. |
| cricket::FakeVideoRenderer renderer2; |
| webrtc::test::FrameForwarder frame_forwarder; |
| const int kTestWidth = 160; |
| const int kTestHeight = 120; |
| cricket::FakeFrameSource frame_source(kTestWidth, kTestHeight, |
| rtc::kNumMicrosecsPerSec / 5); |
| EXPECT_TRUE( |
| channel_->AddSendStream(cricket::StreamParams::CreateLegacy(5678))); |
| EXPECT_TRUE(channel_->SetVideoSend(5678, nullptr, &frame_forwarder)); |
| EXPECT_TRUE( |
| channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(5678))); |
| EXPECT_TRUE(channel_->SetSink(5678, &renderer2)); |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| EXPECT_FRAME_ON_RENDERER_WAIT(renderer2, 1, kTestWidth, kTestHeight, |
| kTimeout); |
| |
| // Get stats, and make sure they are correct for two senders. We wait until |
| // the number of expected packets have been sent to avoid races where we |
| // check stats before it has been updated. |
| cricket::VideoMediaInfo info; |
| for (uint32_t i = 0; i < kTimeout; ++i) { |
| rtc::Thread::Current()->ProcessMessages(1); |
| EXPECT_TRUE(channel_->GetStats(&info)); |
| ASSERT_EQ(2U, info.senders.size()); |
| if (info.senders[0].packets_sent + info.senders[1].packets_sent == |
| NumRtpPackets()) { |
| // Stats have been updated for both sent frames, expectations can be |
| // checked now. |
| break; |
| } |
| } |
| EXPECT_EQ(NumRtpPackets(), |
| info.senders[0].packets_sent + info.senders[1].packets_sent) |
| << "Timed out while waiting for packet counts for all sent packets."; |
| EXPECT_EQ(1U, info.senders[0].ssrcs().size()); |
| EXPECT_EQ(1234U, info.senders[0].ssrcs()[0]); |
| EXPECT_EQ(kVideoWidth, info.senders[0].send_frame_width); |
| EXPECT_EQ(kVideoHeight, info.senders[0].send_frame_height); |
| EXPECT_EQ(1U, info.senders[1].ssrcs().size()); |
| EXPECT_EQ(5678U, info.senders[1].ssrcs()[0]); |
| EXPECT_EQ(kTestWidth, info.senders[1].send_frame_width); |
| EXPECT_EQ(kTestHeight, info.senders[1].send_frame_height); |
| // The capturer must be unregistered here as it runs out of it's scope next. |
| channel_->SetVideoSend(5678, nullptr, nullptr); |
| } |
| |
| // Test that we can set the bandwidth. |
| TEST_F(WebRtcVideoChannelBaseTest, SetSendBandwidth) { |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(DefaultCodec()); |
| parameters.max_bandwidth_bps = -1; // <= 0 means unlimited. |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| parameters.max_bandwidth_bps = 128 * 1024; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| } |
| |
| // Test that we can set the SSRC for the default send source. |
| TEST_F(WebRtcVideoChannelBaseTest, SetSendSsrc) { |
| EXPECT_TRUE(SetDefaultCodec()); |
| EXPECT_TRUE(SetSend(true)); |
| SendFrame(); |
| EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout); |
| RtpPacket header; |
| EXPECT_TRUE(header.Parse(GetRtpPacket(0))); |
| EXPECT_EQ(kSsrc, header.Ssrc()); |
| |
| // Packets are being paced out, so these can mismatch between the first and |
| // second call to NumRtpPackets until pending packets are paced out. |
| EXPECT_EQ_WAIT(NumRtpPackets(), NumRtpPackets(header.Ssrc()), kTimeout); |
| EXPECT_EQ_WAIT(NumRtpBytes(), NumRtpBytes(header.Ssrc()), kTimeout); |
| EXPECT_EQ(1, NumSentSsrcs()); |
| EXPECT_EQ(0, NumRtpPackets(kSsrc - 1)); |
| EXPECT_EQ(0, NumRtpBytes(kSsrc - 1)); |
| } |
| |
| // Test that we can set the SSRC even after codecs are set. |
| TEST_F(WebRtcVideoChannelBaseTest, SetSendSsrcAfterSetCodecs) { |
| // Remove stream added in Setup. |
| EXPECT_TRUE(channel_->RemoveSendStream(kSsrc)); |
| EXPECT_TRUE(SetDefaultCodec()); |
| EXPECT_TRUE( |
| channel_->AddSendStream(cricket::StreamParams::CreateLegacy(999))); |
| EXPECT_TRUE(channel_->SetVideoSend(999u, nullptr, frame_forwarder_.get())); |
| EXPECT_TRUE(SetSend(true)); |
| EXPECT_TRUE(WaitAndSendFrame(0)); |
| EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout); |
| RtpPacket header; |
| EXPECT_TRUE(header.Parse(GetRtpPacket(0))); |
| EXPECT_EQ(999u, header.Ssrc()); |
| // Packets are being paced out, so these can mismatch between the first and |
| // second call to NumRtpPackets until pending packets are paced out. |
| EXPECT_EQ_WAIT(NumRtpPackets(), NumRtpPackets(header.Ssrc()), kTimeout); |
| EXPECT_EQ_WAIT(NumRtpBytes(), NumRtpBytes(header.Ssrc()), kTimeout); |
| EXPECT_EQ(1, NumSentSsrcs()); |
| EXPECT_EQ(0, NumRtpPackets(kSsrc)); |
| EXPECT_EQ(0, NumRtpBytes(kSsrc)); |
| } |
| |
| // Test that we can set the default video renderer before and after |
| // media is received. |
| TEST_F(WebRtcVideoChannelBaseTest, SetSink) { |
| RtpPacket packet; |
| packet.SetSsrc(kSsrc); |
| channel_->SetDefaultSink(NULL); |
| EXPECT_TRUE(SetDefaultCodec()); |
| EXPECT_TRUE(SetSend(true)); |
| EXPECT_EQ(0, renderer_.num_rendered_frames()); |
| channel_->OnPacketReceived(packet.Buffer(), /* packet_time_us */ -1); |
| channel_->SetDefaultSink(&renderer_); |
| SendFrame(); |
| EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout); |
| } |
| |
| // Tests setting up and configuring a send stream. |
| TEST_F(WebRtcVideoChannelBaseTest, AddRemoveSendStreams) { |
| EXPECT_TRUE(SetOneCodec(DefaultCodec())); |
| EXPECT_TRUE(SetSend(true)); |
| channel_->SetDefaultSink(&renderer_); |
| SendFrame(); |
| EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout); |
| EXPECT_GT(NumRtpPackets(), 0); |
| RtpPacket header; |
| size_t last_packet = NumRtpPackets() - 1; |
| EXPECT_TRUE(header.Parse(GetRtpPacket(static_cast<int>(last_packet)))); |
| EXPECT_EQ(kSsrc, header.Ssrc()); |
| |
| // Remove the send stream that was added during Setup. |
| EXPECT_TRUE(channel_->RemoveSendStream(kSsrc)); |
| int rtp_packets = NumRtpPackets(); |
| |
| EXPECT_TRUE( |
| channel_->AddSendStream(cricket::StreamParams::CreateLegacy(789u))); |
| EXPECT_TRUE(channel_->SetVideoSend(789u, nullptr, frame_forwarder_.get())); |
| EXPECT_EQ(rtp_packets, NumRtpPackets()); |
| // Wait 30ms to guarantee the engine does not drop the frame. |
| EXPECT_TRUE(WaitAndSendFrame(30)); |
| EXPECT_TRUE_WAIT(NumRtpPackets() > rtp_packets, kTimeout); |
| |
| last_packet = NumRtpPackets() - 1; |
| EXPECT_TRUE(header.Parse(GetRtpPacket(static_cast<int>(last_packet)))); |
| EXPECT_EQ(789u, header.Ssrc()); |
| } |
| |
| // Tests the behavior of incoming streams in a conference scenario. |
| TEST_F(WebRtcVideoChannelBaseTest, SimulateConference) { |
| cricket::FakeVideoRenderer renderer1, renderer2; |
| EXPECT_TRUE(SetDefaultCodec()); |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(DefaultCodec()); |
| parameters.conference_mode = true; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_TRUE(SetSend(true)); |
| EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(1))); |
| EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(2))); |
| EXPECT_TRUE(channel_->SetSink(1, &renderer1)); |
| EXPECT_TRUE(channel_->SetSink(2, &renderer2)); |
| EXPECT_EQ(0, renderer1.num_rendered_frames()); |
| EXPECT_EQ(0, renderer2.num_rendered_frames()); |
| std::vector<uint32_t> ssrcs; |
| ssrcs.push_back(1); |
| ssrcs.push_back(2); |
| network_interface_.SetConferenceMode(true, ssrcs); |
| SendFrame(); |
| EXPECT_FRAME_ON_RENDERER_WAIT(renderer1, 1, kVideoWidth, kVideoHeight, |
| kTimeout); |
| EXPECT_FRAME_ON_RENDERER_WAIT(renderer2, 1, kVideoWidth, kVideoHeight, |
| kTimeout); |
| |
| EXPECT_EQ(DefaultCodec().id, GetPayloadType(GetRtpPacket(0))); |
| EXPECT_EQ(kVideoWidth, renderer1.width()); |
| EXPECT_EQ(kVideoHeight, renderer1.height()); |
| EXPECT_EQ(kVideoWidth, renderer2.width()); |
| EXPECT_EQ(kVideoHeight, renderer2.height()); |
| EXPECT_TRUE(channel_->RemoveRecvStream(2)); |
| EXPECT_TRUE(channel_->RemoveRecvStream(1)); |
| } |
| |
| // Tests that we can add and remove capturers and frames are sent out properly |
| TEST_F(WebRtcVideoChannelBaseTest, DISABLED_AddRemoveCapturer) { |
| using cricket::FOURCC_I420; |
| using cricket::VideoCodec; |
| using cricket::VideoFormat; |
| using cricket::VideoOptions; |
| |
| VideoCodec codec = DefaultCodec(); |
| const int time_between_send_ms = VideoFormat::FpsToInterval(kFramerate); |
| EXPECT_TRUE(SetOneCodec(codec)); |
| EXPECT_TRUE(SetSend(true)); |
| channel_->SetDefaultSink(&renderer_); |
| EXPECT_EQ(0, renderer_.num_rendered_frames()); |
| SendFrame(); |
| EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout); |
| |
| webrtc::test::FrameForwarder frame_forwarder; |
| cricket::FakeFrameSource frame_source(480, 360, rtc::kNumMicrosecsPerSec / 30, |
| rtc::kNumMicrosecsPerSec / 30); |
| |
| // TODO(nisse): This testcase fails if we don't configure |
| // screencast. It's unclear why, I see nothing obvious in this |
| // test which is related to screencast logic. |
| VideoOptions video_options; |
| video_options.is_screencast = true; |
| channel_->SetVideoSend(kSsrc, &video_options, nullptr); |
| |
| int captured_frames = 1; |
| for (int iterations = 0; iterations < 2; ++iterations) { |
| EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, &frame_forwarder)); |
| rtc::Thread::Current()->ProcessMessages(time_between_send_ms); |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| |
| ++captured_frames; |
| // Wait until frame of right size is captured. |
| EXPECT_TRUE_WAIT(renderer_.num_rendered_frames() >= captured_frames && |
| 480 == renderer_.width() && |
| 360 == renderer_.height() && !renderer_.black_frame(), |
| kTimeout); |
| EXPECT_GE(renderer_.num_rendered_frames(), captured_frames); |
| EXPECT_EQ(480, renderer_.width()); |
| EXPECT_EQ(360, renderer_.height()); |
| captured_frames = renderer_.num_rendered_frames() + 1; |
| EXPECT_FALSE(renderer_.black_frame()); |
| EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, nullptr)); |
| // Make sure a black frame is generated within the specified timeout. |
| // The black frame should be the resolution of the previous frame to |
| // prevent expensive encoder reconfigurations. |
| EXPECT_TRUE_WAIT(renderer_.num_rendered_frames() >= captured_frames && |
| 480 == renderer_.width() && |
| 360 == renderer_.height() && renderer_.black_frame(), |
| kTimeout); |
| EXPECT_GE(renderer_.num_rendered_frames(), captured_frames); |
| EXPECT_EQ(480, renderer_.width()); |
| EXPECT_EQ(360, renderer_.height()); |
| EXPECT_TRUE(renderer_.black_frame()); |
| |
| // The black frame has the same timestamp as the next frame since it's |
| // timestamp is set to the last frame's timestamp + interval. WebRTC will |
| // not render a frame with the same timestamp so capture another frame |
| // with the frame capturer to increment the next frame's timestamp. |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| } |
| } |
| |
| // Tests that if SetVideoSend is called with a NULL capturer after the |
| // capturer was already removed, the application doesn't crash (and no black |
| // frame is sent). |
| TEST_F(WebRtcVideoChannelBaseTest, RemoveCapturerWithoutAdd) { |
| EXPECT_TRUE(SetOneCodec(DefaultCodec())); |
| EXPECT_TRUE(SetSend(true)); |
| channel_->SetDefaultSink(&renderer_); |
| EXPECT_EQ(0, renderer_.num_rendered_frames()); |
| SendFrame(); |
| EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout); |
| // Wait for one frame so they don't get dropped because we send frames too |
| // tightly. |
| rtc::Thread::Current()->ProcessMessages(30); |
| // Remove the capturer. |
| EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, nullptr)); |
| |
| // No capturer was added, so this SetVideoSend shouldn't do anything. |
| EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, nullptr)); |
| rtc::Thread::Current()->ProcessMessages(300); |
| // Verify no more frames were sent. |
| EXPECT_EQ(1, renderer_.num_rendered_frames()); |
| } |
| |
| // Tests that we can add and remove capturer as unique sources. |
| TEST_F(WebRtcVideoChannelBaseTest, AddRemoveCapturerMultipleSources) { |
| // WebRTC implementation will drop frames if pushed to quickly. Wait the |
| // interval time to avoid that. |
| // WebRTC implementation will drop frames if pushed to quickly. Wait the |
| // interval time to avoid that. |
| // Set up the stream associated with the engine. |
| EXPECT_TRUE( |
| channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc))); |
| EXPECT_TRUE(channel_->SetSink(kSsrc, &renderer_)); |
| cricket::VideoFormat capture_format( |
| kVideoWidth, kVideoHeight, |
| cricket::VideoFormat::FpsToInterval(kFramerate), cricket::FOURCC_I420); |
| // Set up additional stream 1. |
| cricket::FakeVideoRenderer renderer1; |
| EXPECT_FALSE(channel_->SetSink(1, &renderer1)); |
| EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(1))); |
| EXPECT_TRUE(channel_->SetSink(1, &renderer1)); |
| EXPECT_TRUE(channel_->AddSendStream(cricket::StreamParams::CreateLegacy(1))); |
| |
| webrtc::test::FrameForwarder frame_forwarder1; |
| cricket::FakeFrameSource frame_source(kVideoWidth, kVideoHeight, |
| rtc::kNumMicrosecsPerSec / kFramerate); |
| |
| // Set up additional stream 2. |
| cricket::FakeVideoRenderer renderer2; |
| EXPECT_FALSE(channel_->SetSink(2, &renderer2)); |
| EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(2))); |
| EXPECT_TRUE(channel_->SetSink(2, &renderer2)); |
| EXPECT_TRUE(channel_->AddSendStream(cricket::StreamParams::CreateLegacy(2))); |
| webrtc::test::FrameForwarder frame_forwarder2; |
| |
| // State for all the streams. |
| EXPECT_TRUE(SetOneCodec(DefaultCodec())); |
| // A limitation in the lmi implementation requires that SetVideoSend() is |
| // called after SetOneCodec(). |
| // TODO(hellner): this seems like an unnecessary constraint, fix it. |
| EXPECT_TRUE(channel_->SetVideoSend(1, nullptr, &frame_forwarder1)); |
| EXPECT_TRUE(channel_->SetVideoSend(2, nullptr, &frame_forwarder2)); |
| EXPECT_TRUE(SetSend(true)); |
| // Test capturer associated with engine. |
| const int kTestWidth = 160; |
| const int kTestHeight = 120; |
| frame_forwarder1.IncomingCapturedFrame(frame_source.GetFrame( |
| kTestWidth, kTestHeight, webrtc::VideoRotation::kVideoRotation_0, |
| rtc::kNumMicrosecsPerSec / kFramerate)); |
| EXPECT_FRAME_ON_RENDERER_WAIT(renderer1, 1, kTestWidth, kTestHeight, |
| kTimeout); |
| // Capture a frame with additional capturer2, frames should be received |
| frame_forwarder2.IncomingCapturedFrame(frame_source.GetFrame( |
| kTestWidth, kTestHeight, webrtc::VideoRotation::kVideoRotation_0, |
| rtc::kNumMicrosecsPerSec / kFramerate)); |
| EXPECT_FRAME_ON_RENDERER_WAIT(renderer2, 1, kTestWidth, kTestHeight, |
| kTimeout); |
| // Successfully remove the capturer. |
| EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, nullptr)); |
| // The capturers must be unregistered here as it runs out of it's scope |
| // next. |
| EXPECT_TRUE(channel_->SetVideoSend(1, nullptr, nullptr)); |
| EXPECT_TRUE(channel_->SetVideoSend(2, nullptr, nullptr)); |
| } |
| |
| // Tests empty StreamParams is rejected. |
| TEST_F(WebRtcVideoChannelBaseTest, RejectEmptyStreamParams) { |
| // Remove the send stream that was added during Setup. |
| EXPECT_TRUE(channel_->RemoveSendStream(kSsrc)); |
| |
| cricket::StreamParams empty; |
| EXPECT_FALSE(channel_->AddSendStream(empty)); |
| EXPECT_TRUE( |
| channel_->AddSendStream(cricket::StreamParams::CreateLegacy(789u))); |
| } |
| |
| // Test that multiple send streams can be created and deleted properly. |
| TEST_F(WebRtcVideoChannelBaseTest, MultipleSendStreams) { |
| // Remove stream added in Setup. I.e. remove stream corresponding to default |
| // channel. |
| EXPECT_TRUE(channel_->RemoveSendStream(kSsrc)); |
| const unsigned int kSsrcsSize = sizeof(kSsrcs4) / sizeof(kSsrcs4[0]); |
| for (unsigned int i = 0; i < kSsrcsSize; ++i) { |
| EXPECT_TRUE(channel_->AddSendStream( |
| cricket::StreamParams::CreateLegacy(kSsrcs4[i]))); |
| } |
| // Delete one of the non default channel streams, let the destructor delete |
| // the remaining ones. |
| EXPECT_TRUE(channel_->RemoveSendStream(kSsrcs4[kSsrcsSize - 1])); |
| // Stream should already be deleted. |
| EXPECT_FALSE(channel_->RemoveSendStream(kSsrcs4[kSsrcsSize - 1])); |
| } |
| |
| TEST_F(WebRtcVideoChannelBaseTest, SendAndReceiveVp8Vga) { |
| SendAndReceive(GetEngineCodec("VP8")); |
| } |
| |
| TEST_F(WebRtcVideoChannelBaseTest, SendAndReceiveVp8Qvga) { |
| SendAndReceive(GetEngineCodec("VP8")); |
| } |
| |
| TEST_F(WebRtcVideoChannelBaseTest, SendAndReceiveVp8SvcQqvga) { |
| SendAndReceive(GetEngineCodec("VP8")); |
| } |
| |
| TEST_F(WebRtcVideoChannelBaseTest, TwoStreamsSendAndReceive) { |
| // Set a high bitrate to not be downscaled by VP8 due to low initial start |
| // bitrates. This currently happens at <250k, and two streams sharing 300k |
| // initially will use QVGA instead of VGA. |
| // TODO(pbos): Set up the quality scaler so that both senders reliably start |
| // at QVGA, then verify that instead. |
| cricket::VideoCodec codec = GetEngineCodec("VP8"); |
| codec.params[kCodecParamStartBitrate] = "1000000"; |
| TwoStreamsSendAndReceive(codec); |
| } |
| |
| #if defined(RTC_ENABLE_VP9) |
| |
| TEST_F(WebRtcVideoChannelBaseTest, RequestEncoderFallback) { |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP9")); |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| |
| VideoCodec codec; |
| ASSERT_TRUE(channel_->GetSendCodec(&codec)); |
| EXPECT_EQ("VP9", codec.name); |
| |
| // RequestEncoderFallback will post a task to the worker thread (which is also |
| // the current thread), hence the ProcessMessages call. |
| channel_->RequestEncoderFallback(); |
| rtc::Thread::Current()->ProcessMessages(30); |
| ASSERT_TRUE(channel_->GetSendCodec(&codec)); |
| EXPECT_EQ("VP8", codec.name); |
| |
| // No other codec to fall back to, keep using VP8. |
| channel_->RequestEncoderFallback(); |
| rtc::Thread::Current()->ProcessMessages(30); |
| ASSERT_TRUE(channel_->GetSendCodec(&codec)); |
| EXPECT_EQ("VP8", codec.name); |
| } |
| |
| TEST_F(WebRtcVideoChannelBaseTest, RequestEncoderSwitchWithConfig) { |
| const std::string kParam = "the-param"; |
| const std::string kPing = "ping"; |
| const std::string kPong = "pong"; |
| |
| cricket::VideoSendParameters parameters; |
| VideoCodec vp9 = GetEngineCodec("VP9"); |
| vp9.params[kParam] = kPong; |
| parameters.codecs.push_back(vp9); |
| |
| VideoCodec vp8 = GetEngineCodec("VP8"); |
| vp8.params[kParam] = kPing; |
| parameters.codecs.push_back(vp8); |
| |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| channel_->SetVideoCodecSwitchingEnabled(true); |
| |
| VideoCodec codec; |
| ASSERT_TRUE(channel_->GetSendCodec(&codec)); |
| EXPECT_THAT(codec.name, Eq("VP9")); |
| |
| // RequestEncoderSwitch will post a task to the worker thread (which is also |
| // the current thread), hence the ProcessMessages call. |
| webrtc::EncoderSwitchRequestCallback::Config conf1{"VP8", kParam, kPing}; |
| channel_->RequestEncoderSwitch(conf1); |
| rtc::Thread::Current()->ProcessMessages(30); |
| ASSERT_TRUE(channel_->GetSendCodec(&codec)); |
| EXPECT_THAT(codec.name, Eq("VP8")); |
| EXPECT_THAT(codec.params, Contains(Pair(kParam, kPing))); |
| |
| webrtc::EncoderSwitchRequestCallback::Config conf2{"VP9", kParam, kPong}; |
| channel_->RequestEncoderSwitch(conf2); |
| rtc::Thread::Current()->ProcessMessages(30); |
| ASSERT_TRUE(channel_->GetSendCodec(&codec)); |
| EXPECT_THAT(codec.name, Eq("VP9")); |
| EXPECT_THAT(codec.params, Contains(Pair(kParam, kPong))); |
| } |
| |
| TEST_F(WebRtcVideoChannelBaseTest, RequestEncoderSwitchIncorrectParam) { |
| const std::string kParam = "the-param"; |
| const std::string kPing = "ping"; |
| const std::string kPong = "pong"; |
| |
| cricket::VideoSendParameters parameters; |
| VideoCodec vp9 = GetEngineCodec("VP9"); |
| vp9.params[kParam] = kPong; |
| parameters.codecs.push_back(vp9); |
| |
| VideoCodec vp8 = GetEngineCodec("VP8"); |
| vp8.params[kParam] = kPing; |
| parameters.codecs.push_back(vp8); |
| |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| channel_->SetVideoCodecSwitchingEnabled(true); |
| |
| VideoCodec codec; |
| ASSERT_TRUE(channel_->GetSendCodec(&codec)); |
| EXPECT_THAT(codec.name, Eq("VP9")); |
| |
| // RequestEncoderSwitch will post a task to the worker thread (which is also |
| // the current thread), hence the ProcessMessages call. |
| webrtc::EncoderSwitchRequestCallback::Config conf1{"VP8", kParam, kPing}; |
| channel_->RequestEncoderSwitch(conf1); |
| rtc::Thread::Current()->ProcessMessages(30); |
| ASSERT_TRUE(channel_->GetSendCodec(&codec)); |
| EXPECT_THAT(codec.name, Eq("VP8")); |
| EXPECT_THAT(codec.params, Contains(Pair(kParam, kPing))); |
| |
| // Incorrect conf2.value, expect no codec switch. |
| webrtc::EncoderSwitchRequestCallback::Config conf2{"VP9", kParam, kPing}; |
| channel_->RequestEncoderSwitch(conf2); |
| rtc::Thread::Current()->ProcessMessages(30); |
| ASSERT_TRUE(channel_->GetSendCodec(&codec)); |
| EXPECT_THAT(codec.name, Eq("VP8")); |
| EXPECT_THAT(codec.params, Contains(Pair(kParam, kPing))); |
| } |
| |
| TEST_F(WebRtcVideoChannelBaseTest, |
| RequestEncoderSwitchWithConfigBeforeEnabling) { |
| const std::string kParam = "the-param"; |
| const std::string kPing = "ping"; |
| const std::string kPong = "pong"; |
| |
| cricket::VideoSendParameters parameters; |
| VideoCodec vp9 = GetEngineCodec("VP9"); |
| vp9.params[kParam] = kPong; |
| parameters.codecs.push_back(vp9); |
| |
| VideoCodec vp8 = GetEngineCodec("VP8"); |
| vp8.params[kParam] = kPing; |
| parameters.codecs.push_back(vp8); |
| |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| |
| VideoCodec codec; |
| ASSERT_TRUE(channel_->GetSendCodec(&codec)); |
| EXPECT_THAT(codec.name, Eq("VP9")); |
| |
| webrtc::EncoderSwitchRequestCallback::Config conf{"VP8", kParam, kPing}; |
| channel_->RequestEncoderSwitch(conf); |
| |
| // Enable codec switching after it has been requested. |
| channel_->SetVideoCodecSwitchingEnabled(true); |
| |
| // RequestEncoderSwitch will post a task to the worker thread (which is also |
| // the current thread), hence the ProcessMessages call. |
| rtc::Thread::Current()->ProcessMessages(30); |
| ASSERT_TRUE(channel_->GetSendCodec(&codec)); |
| EXPECT_THAT(codec.name, Eq("VP8")); |
| EXPECT_THAT(codec.params, Contains(Pair(kParam, kPing))); |
| } |
| |
| #endif // defined(RTC_ENABLE_VP9) |
| |
| class WebRtcVideoChannelTest : public WebRtcVideoEngineTest { |
| public: |
| WebRtcVideoChannelTest() : WebRtcVideoChannelTest("") {} |
| explicit WebRtcVideoChannelTest(const char* field_trials) |
| : WebRtcVideoEngineTest(field_trials), |
| frame_source_(1280, 720, rtc::kNumMicrosecsPerSec / 30), |
| last_ssrc_(0) {} |
| void SetUp() override { |
| AddSupportedVideoCodecType("VP8"); |
| AddSupportedVideoCodecType("VP9"); |
| #if defined(WEBRTC_USE_H264) |
| AddSupportedVideoCodecType("H264"); |
| #endif |
| |
| fake_call_.reset(new FakeCall()); |
| channel_.reset(engine_.CreateMediaChannel( |
| fake_call_.get(), GetMediaConfig(), VideoOptions(), |
| webrtc::CryptoOptions(), video_bitrate_allocator_factory_.get())); |
| channel_->OnReadyToSend(true); |
| last_ssrc_ = 123; |
| send_parameters_.codecs = engine_.send_codecs(); |
| recv_parameters_.codecs = engine_.recv_codecs(); |
| ASSERT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| } |
| |
| void TearDown() override { |
| channel_->SetInterface(nullptr); |
| channel_ = nullptr; |
| } |
| |
| void ResetTest() { |
| TearDown(); |
| SetUp(); |
| } |
| |
| cricket::VideoCodec GetEngineCodec(const std::string& name) { |
| for (const cricket::VideoCodec& engine_codec : engine_.send_codecs()) { |
| if (absl::EqualsIgnoreCase(name, engine_codec.name)) |
| return engine_codec; |
| } |
| // This point should never be reached. |
| ADD_FAILURE() << "Unrecognized codec name: " << name; |
| return cricket::VideoCodec(); |
| } |
| |
| cricket::VideoCodec DefaultCodec() { return GetEngineCodec("VP8"); } |
| |
| // After receciving and processing the packet, enough time is advanced that |
| // the unsignalled receive stream cooldown is no longer in effect. |
| void ReceivePacketAndAdvanceTime(rtc::CopyOnWriteBuffer packet, |
| int64_t packet_time_us) { |
| channel_->OnPacketReceived(packet, packet_time_us); |
| rtc::Thread::Current()->ProcessMessages(0); |
| fake_clock_.AdvanceTime( |
| webrtc::TimeDelta::Millis(kUnsignalledReceiveStreamCooldownMs)); |
| } |
| |
| protected: |
| FakeVideoSendStream* AddSendStream() { |
| return AddSendStream(StreamParams::CreateLegacy(++last_ssrc_)); |
| } |
| |
| FakeVideoSendStream* AddSendStream(const StreamParams& sp) { |
| size_t num_streams = fake_call_->GetVideoSendStreams().size(); |
| EXPECT_TRUE(channel_->AddSendStream(sp)); |
| std::vector<FakeVideoSendStream*> streams = |
| fake_call_->GetVideoSendStreams(); |
| EXPECT_EQ(num_streams + 1, streams.size()); |
| return streams[streams.size() - 1]; |
| } |
| |
| std::vector<FakeVideoSendStream*> GetFakeSendStreams() { |
| return fake_call_->GetVideoSendStreams(); |
| } |
| |
| FakeVideoReceiveStream* AddRecvStream() { |
| return AddRecvStream(StreamParams::CreateLegacy(++last_ssrc_)); |
| } |
| |
| FakeVideoReceiveStream* AddRecvStream(const StreamParams& sp) { |
| size_t num_streams = fake_call_->GetVideoReceiveStreams().size(); |
| EXPECT_TRUE(channel_->AddRecvStream(sp)); |
| std::vector<FakeVideoReceiveStream*> streams = |
| fake_call_->GetVideoReceiveStreams(); |
| EXPECT_EQ(num_streams + 1, streams.size()); |
| return streams[streams.size() - 1]; |
| } |
| |
| void SetSendCodecsShouldWorkForBitrates(const char* min_bitrate_kbps, |
| int expected_min_bitrate_bps, |
| const char* start_bitrate_kbps, |
| int expected_start_bitrate_bps, |
| const char* max_bitrate_kbps, |
| int expected_max_bitrate_bps) { |
| ExpectSetBitrateParameters(expected_min_bitrate_bps, |
| expected_start_bitrate_bps, |
| expected_max_bitrate_bps); |
| auto& codecs = send_parameters_.codecs; |
| codecs.clear(); |
| codecs.push_back(GetEngineCodec("VP8")); |
| codecs[0].params[kCodecParamMinBitrate] = min_bitrate_kbps; |
| codecs[0].params[kCodecParamStartBitrate] = start_bitrate_kbps; |
| codecs[0].params[kCodecParamMaxBitrate] = max_bitrate_kbps; |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| } |
| |
| void ExpectSetBitrateParameters(int min_bitrate_bps, |
| int start_bitrate_bps, |
| int max_bitrate_bps) { |
| EXPECT_CALL( |
| *fake_call_->GetMockTransportControllerSend(), |
| SetSdpBitrateParameters(AllOf( |
| Field(&BitrateConstraints::min_bitrate_bps, min_bitrate_bps), |
| Field(&BitrateConstraints::start_bitrate_bps, start_bitrate_bps), |
| Field(&BitrateConstraints::max_bitrate_bps, max_bitrate_bps)))); |
| } |
| |
| void ExpectSetMaxBitrate(int max_bitrate_bps) { |
| EXPECT_CALL(*fake_call_->GetMockTransportControllerSend(), |
| SetSdpBitrateParameters(Field( |
| &BitrateConstraints::max_bitrate_bps, max_bitrate_bps))); |
| } |
| |
| void TestExtmapAllowMixedCaller(bool extmap_allow_mixed) { |
| // For a caller, the answer will be applied in set remote description |
| // where SetSendParameters() is called. |
| EXPECT_TRUE( |
| channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc))); |
| send_parameters_.extmap_allow_mixed = extmap_allow_mixed; |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| const webrtc::VideoSendStream::Config& config = |
| fake_call_->GetVideoSendStreams()[0]->GetConfig(); |
| EXPECT_EQ(extmap_allow_mixed, config.rtp.extmap_allow_mixed); |
| } |
| |
| void TestExtmapAllowMixedCallee(bool extmap_allow_mixed) { |
| // For a callee, the answer will be applied in set local description |
| // where SetExtmapAllowMixed() and AddSendStream() are called. |
| channel_->SetExtmapAllowMixed(extmap_allow_mixed); |
| EXPECT_TRUE( |
| channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc))); |
| const webrtc::VideoSendStream::Config& config = |
| fake_call_->GetVideoSendStreams()[0]->GetConfig(); |
| EXPECT_EQ(extmap_allow_mixed, config.rtp.extmap_allow_mixed); |
| } |
| |
| void TestSetSendRtpHeaderExtensions(const std::string& ext_uri) { |
| // Enable extension. |
| const int id = 1; |
| cricket::VideoSendParameters parameters = send_parameters_; |
| parameters.extensions.push_back(RtpExtension(ext_uri, id)); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| FakeVideoSendStream* send_stream = |
| AddSendStream(cricket::StreamParams::CreateLegacy(123)); |
| |
| // Verify the send extension id. |
| ASSERT_EQ(1u, send_stream->GetConfig().rtp.extensions.size()); |
| EXPECT_EQ(id, send_stream->GetConfig().rtp.extensions[0].id); |
| EXPECT_EQ(ext_uri, send_stream->GetConfig().rtp.extensions[0].uri); |
| // Verify call with same set of extensions returns true. |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| // Verify that SetSendRtpHeaderExtensions doesn't implicitly add them for |
| // receivers. |
| EXPECT_TRUE(AddRecvStream(cricket::StreamParams::CreateLegacy(123)) |
| ->GetConfig() |
| .rtp.extensions.empty()); |
| |
| // Verify that existing RTP header extensions can be removed. |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| ASSERT_EQ(1u, fake_call_->GetVideoSendStreams().size()); |
| send_stream = fake_call_->GetVideoSendStreams()[0]; |
| EXPECT_TRUE(send_stream->GetConfig().rtp.extensions.empty()); |
| |
| // Verify that adding receive RTP header extensions adds them for existing |
| // streams. |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| send_stream = fake_call_->GetVideoSendStreams()[0]; |
| ASSERT_EQ(1u, send_stream->GetConfig().rtp.extensions.size()); |
| EXPECT_EQ(id, send_stream->GetConfig().rtp.extensions[0].id); |
| EXPECT_EQ(ext_uri, send_stream->GetConfig().rtp.extensions[0].uri); |
| } |
| |
| void TestSetRecvRtpHeaderExtensions(const std::string& ext_uri) { |
| // Enable extension. |
| const int id = 1; |
| cricket::VideoRecvParameters parameters = recv_parameters_; |
| parameters.extensions.push_back(RtpExtension(ext_uri, id)); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| |
| FakeVideoReceiveStream* recv_stream = |
| AddRecvStream(cricket::StreamParams::CreateLegacy(123)); |
| |
| // Verify the recv extension id. |
| ASSERT_EQ(1u, recv_stream->GetConfig().rtp.extensions.size()); |
| EXPECT_EQ(id, recv_stream->GetConfig().rtp.extensions[0].id); |
| EXPECT_EQ(ext_uri, recv_stream->GetConfig().rtp.extensions[0].uri); |
| // Verify call with same set of extensions returns true. |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| |
| // Verify that SetRecvRtpHeaderExtensions doesn't implicitly add them for |
| // senders. |
| EXPECT_TRUE(AddSendStream(cricket::StreamParams::CreateLegacy(123)) |
| ->GetConfig() |
| .rtp.extensions.empty()); |
| |
| // Verify that existing RTP header extensions can be removed. |
| EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_)); |
| ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size()); |
| recv_stream = fake_call_->GetVideoReceiveStreams()[0]; |
| EXPECT_TRUE(recv_stream->GetConfig().rtp.extensions.empty()); |
| |
| // Verify that adding receive RTP header extensions adds them for existing |
| // streams. |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| recv_stream = fake_call_->GetVideoReceiveStreams()[0]; |
| ASSERT_EQ(1u, recv_stream->GetConfig().rtp.extensions.size()); |
| EXPECT_EQ(id, recv_stream->GetConfig().rtp.extensions[0].id); |
| EXPECT_EQ(ext_uri, recv_stream->GetConfig().rtp.extensions[0].uri); |
| } |
| |
| void TestLossNotificationState(bool expect_lntf_enabled) { |
| AssignDefaultCodec(); |
| VerifyCodecHasDefaultFeedbackParams(default_codec_, expect_lntf_enabled); |
| |
| cricket::VideoSendParameters parameters; |
| parameters.codecs = engine_.send_codecs(); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_TRUE(channel_->SetSend(true)); |
| |
| // Send side. |
| FakeVideoSendStream* send_stream = |
| AddSendStream(cricket::StreamParams::CreateLegacy(1)); |
| EXPECT_EQ(send_stream->GetConfig().rtp.lntf.enabled, expect_lntf_enabled); |
| |
| // Receiver side. |
| FakeVideoReceiveStream* recv_stream = |
| AddRecvStream(cricket::StreamParams::CreateLegacy(1)); |
| EXPECT_EQ(recv_stream->GetConfig().rtp.lntf.enabled, expect_lntf_enabled); |
| } |
| |
| void TestExtensionFilter(const std::vector<std::string>& extensions, |
| const std::string& expected_extension) { |
| cricket::VideoSendParameters parameters = send_parameters_; |
| int expected_id = -1; |
| int id = 1; |
| for (const std::string& extension : extensions) { |
| if (extension == expected_extension) |
| expected_id = id; |
| parameters.extensions.push_back(RtpExtension(extension, id++)); |
| } |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| FakeVideoSendStream* send_stream = |
| AddSendStream(cricket::StreamParams::CreateLegacy(123)); |
| |
| // Verify that only one of them has been set, and that it is the one with |
| // highest priority (transport sequence number). |
| ASSERT_EQ(1u, send_stream->GetConfig().rtp.extensions.size()); |
| EXPECT_EQ(expected_id, send_stream->GetConfig().rtp.extensions[0].id); |
| EXPECT_EQ(expected_extension, |
| send_stream->GetConfig().rtp.extensions[0].uri); |
| } |
| |
| void TestDegradationPreference(bool resolution_scaling_enabled, |
| bool fps_scaling_enabled); |
| |
| void TestCpuAdaptation(bool enable_overuse, bool is_screenshare); |
| void TestReceiverLocalSsrcConfiguration(bool receiver_first); |
| void TestReceiveUnsignaledSsrcPacket(uint8_t payload_type, |
| bool expect_created_receive_stream); |
| |
| FakeVideoSendStream* SetDenoisingOption( |
| uint32_t ssrc, |
| webrtc::test::FrameForwarder* frame_forwarder, |
| bool enabled) { |
| cricket::VideoOptions options; |
| options.video_noise_reduction = enabled; |
| EXPECT_TRUE(channel_->SetVideoSend(ssrc, &options, frame_forwarder)); |
| // Options only take effect on the next frame. |
| frame_forwarder->IncomingCapturedFrame(frame_source_.GetFrame()); |
| |
| return fake_call_->GetVideoSendStreams().back(); |
| } |
| |
| FakeVideoSendStream* SetUpSimulcast(bool enabled, bool with_rtx) { |
| const int kRtxSsrcOffset = 0xDEADBEEF; |
| last_ssrc_ += 3; |
| std::vector<uint32_t> ssrcs; |
| std::vector<uint32_t> rtx_ssrcs; |
| uint32_t num_streams = enabled ? 3 : 1; |
| for (uint32_t i = 0; i < num_streams; ++i) { |
| uint32_t ssrc = last_ssrc_ + i; |
| ssrcs.push_back(ssrc); |
| if (with_rtx) { |
| rtx_ssrcs.push_back(ssrc + kRtxSsrcOffset); |
| } |
| } |
| if (with_rtx) { |
| return AddSendStream( |
| cricket::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs)); |
| } |
| return AddSendStream(CreateSimStreamParams("cname", ssrcs)); |
| } |
| |
| int GetMaxEncoderBitrate() { |
| std::vector<FakeVideoSendStream*> streams = |
| fake_call_->GetVideoSendStreams(); |
| EXPECT_EQ(1u, streams.size()); |
| FakeVideoSendStream* stream = streams[streams.size() - 1]; |
| EXPECT_EQ(1u, stream->GetEncoderConfig().number_of_streams); |
| return stream->GetVideoStreams()[0].max_bitrate_bps; |
| } |
| |
| void SetAndExpectMaxBitrate(int global_max, |
| int stream_max, |
| int expected_encoder_bitrate) { |
| VideoSendParameters limited_send_params = send_parameters_; |
| limited_send_params.max_bandwidth_bps = global_max; |
| EXPECT_TRUE(channel_->SetSendParameters(limited_send_params)); |
| webrtc::RtpParameters parameters = |
| channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_EQ(1UL, parameters.encodings.size()); |
| parameters.encodings[0].max_bitrate_bps = stream_max; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| // Read back the parameteres and verify they have the correct value |
| parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_EQ(1UL, parameters.encodings.size()); |
| EXPECT_EQ(stream_max, parameters.encodings[0].max_bitrate_bps); |
| // Verify that the new value propagated down to the encoder |
| EXPECT_EQ(expected_encoder_bitrate, GetMaxEncoderBitrate()); |
| } |
| |
| // Values from kSimulcastConfigs in simulcast.cc. |
| const std::vector<webrtc::VideoStream> GetSimulcastBitrates720p() const { |
| std::vector<webrtc::VideoStream> layers(3); |
| layers[0].min_bitrate_bps = 30000; |
| layers[0].target_bitrate_bps = 150000; |
| layers[0].max_bitrate_bps = 200000; |
| layers[1].min_bitrate_bps = 150000; |
| layers[1].target_bitrate_bps = 500000; |
| layers[1].max_bitrate_bps = 700000; |
| layers[2].min_bitrate_bps = 600000; |
| layers[2].target_bitrate_bps = 2500000; |
| layers[2].max_bitrate_bps = 2500000; |
| return layers; |
| } |
| |
| cricket::FakeFrameSource frame_source_; |
| std::unique_ptr<FakeCall> fake_call_; |
| std::unique_ptr<VideoMediaChannel> channel_; |
| cricket::VideoSendParameters send_parameters_; |
| cricket::VideoRecvParameters recv_parameters_; |
| uint32_t last_ssrc_; |
| }; |
| |
| TEST_F(WebRtcVideoChannelTest, SetsSyncGroupFromSyncLabel) { |
| const uint32_t kVideoSsrc = 123; |
| const std::string kSyncLabel = "AvSyncLabel"; |
| |
| cricket::StreamParams sp = cricket::StreamParams::CreateLegacy(kVideoSsrc); |
| sp.set_stream_ids({kSyncLabel}); |
| EXPECT_TRUE(channel_->AddRecvStream(sp)); |
| |
| EXPECT_EQ(1u, fake_call_->GetVideoReceiveStreams().size()); |
| EXPECT_EQ(kSyncLabel, |
| fake_call_->GetVideoReceiveStreams()[0]->GetConfig().sync_group) |
| << "SyncGroup should be set based on sync_label"; |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, RecvStreamWithSimAndRtx) { |
| cricket::VideoSendParameters parameters; |
| parameters.codecs = engine_.send_codecs(); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_TRUE(channel_->SetSend(true)); |
| parameters.conference_mode = true; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| |
| // Send side. |
| const std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs1); |
| const std::vector<uint32_t> rtx_ssrcs = MAKE_VECTOR(kRtxSsrcs1); |
| FakeVideoSendStream* send_stream = AddSendStream( |
| cricket::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs)); |
| |
| ASSERT_EQ(rtx_ssrcs.size(), send_stream->GetConfig().rtp.rtx.ssrcs.size()); |
| for (size_t i = 0; i < rtx_ssrcs.size(); ++i) |
| EXPECT_EQ(rtx_ssrcs[i], send_stream->GetConfig().rtp.rtx.ssrcs[i]); |
| |
| // Receiver side. |
| FakeVideoReceiveStream* recv_stream = AddRecvStream( |
| cricket::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs)); |
| EXPECT_FALSE( |
| recv_stream->GetConfig().rtp.rtx_associated_payload_types.empty()); |
| EXPECT_TRUE(VerifyRtxReceiveAssociations(recv_stream->GetConfig())) |
| << "RTX should be mapped for all decoders/payload types."; |
| EXPECT_TRUE(HasRtxReceiveAssociation(recv_stream->GetConfig(), |
| GetEngineCodec("red").id)) |
| << "RTX should be mapped for the RED payload type"; |
| |
| EXPECT_EQ(rtx_ssrcs[0], recv_stream->GetConfig().rtp.rtx_ssrc); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, RecvStreamWithRtx) { |
| // Setup one channel with an associated RTX stream. |
| cricket::StreamParams params = |
| cricket::StreamParams::CreateLegacy(kSsrcs1[0]); |
| params.AddFidSsrc(kSsrcs1[0], kRtxSsrcs1[0]); |
| FakeVideoReceiveStream* recv_stream = AddRecvStream(params); |
| EXPECT_EQ(kRtxSsrcs1[0], recv_stream->GetConfig().rtp.rtx_ssrc); |
| |
| EXPECT_TRUE(VerifyRtxReceiveAssociations(recv_stream->GetConfig())) |
| << "RTX should be mapped for all decoders/payload types."; |
| EXPECT_TRUE(HasRtxReceiveAssociation(recv_stream->GetConfig(), |
| GetEngineCodec("red").id)) |
| << "RTX should be mapped for the RED payload type"; |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, RecvStreamNoRtx) { |
| // Setup one channel without an associated RTX stream. |
| cricket::StreamParams params = |
| cricket::StreamParams::CreateLegacy(kSsrcs1[0]); |
| FakeVideoReceiveStream* recv_stream = AddRecvStream(params); |
| ASSERT_EQ(0U, recv_stream->GetConfig().rtp.rtx_ssrc); |
| } |
| |
| // Test propagation of extmap allow mixed setting. |
| TEST_F(WebRtcVideoChannelTest, SetExtmapAllowMixedAsCaller) { |
| TestExtmapAllowMixedCaller(/*extmap_allow_mixed=*/true); |
| } |
| TEST_F(WebRtcVideoChannelTest, SetExtmapAllowMixedDisabledAsCaller) { |
| TestExtmapAllowMixedCaller(/*extmap_allow_mixed=*/false); |
| } |
| TEST_F(WebRtcVideoChannelTest, SetExtmapAllowMixedAsCallee) { |
| TestExtmapAllowMixedCallee(/*extmap_allow_mixed=*/true); |
| } |
| TEST_F(WebRtcVideoChannelTest, SetExtmapAllowMixedDisabledAsCallee) { |
| TestExtmapAllowMixedCallee(/*extmap_allow_mixed=*/false); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, NoHeaderExtesionsByDefault) { |
| FakeVideoSendStream* send_stream = |
| AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcs1[0])); |
| ASSERT_TRUE(send_stream->GetConfig().rtp.extensions.empty()); |
| |
| FakeVideoReceiveStream* recv_stream = |
| AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrcs1[0])); |
| ASSERT_TRUE(recv_stream->GetConfig().rtp.extensions.empty()); |
| } |
| |
| // Test support for RTP timestamp offset header extension. |
| TEST_F(WebRtcVideoChannelTest, SendRtpTimestampOffsetHeaderExtensions) { |
| TestSetSendRtpHeaderExtensions(RtpExtension::kTimestampOffsetUri); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, RecvRtpTimestampOffsetHeaderExtensions) { |
| TestSetRecvRtpHeaderExtensions(RtpExtension::kTimestampOffsetUri); |
| } |
| |
| // Test support for absolute send time header extension. |
| TEST_F(WebRtcVideoChannelTest, SendAbsoluteSendTimeHeaderExtensions) { |
| TestSetSendRtpHeaderExtensions(RtpExtension::kAbsSendTimeUri); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, RecvAbsoluteSendTimeHeaderExtensions) { |
| TestSetRecvRtpHeaderExtensions(RtpExtension::kAbsSendTimeUri); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, FiltersExtensionsPicksTransportSeqNum) { |
| webrtc::test::ScopedFieldTrials override_field_trials( |
| "WebRTC-FilterAbsSendTimeExtension/Enabled/"); |
| // Enable three redundant extensions. |
| std::vector<std::string> extensions; |
| extensions.push_back(RtpExtension::kAbsSendTimeUri); |
| extensions.push_back(RtpExtension::kTimestampOffsetUri); |
| extensions.push_back(RtpExtension::kTransportSequenceNumberUri); |
| TestExtensionFilter(extensions, RtpExtension::kTransportSequenceNumberUri); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, FiltersExtensionsPicksAbsSendTime) { |
| // Enable two redundant extensions. |
| std::vector<std::string> extensions; |
| extensions.push_back(RtpExtension::kAbsSendTimeUri); |
| extensions.push_back(RtpExtension::kTimestampOffsetUri); |
| TestExtensionFilter(extensions, RtpExtension::kAbsSendTimeUri); |
| } |
| |
| // Test support for transport sequence number header extension. |
| TEST_F(WebRtcVideoChannelTest, SendTransportSequenceNumberHeaderExtensions) { |
| TestSetSendRtpHeaderExtensions(RtpExtension::kTransportSequenceNumberUri); |
| } |
| TEST_F(WebRtcVideoChannelTest, RecvTransportSequenceNumberHeaderExtensions) { |
| TestSetRecvRtpHeaderExtensions(RtpExtension::kTransportSequenceNumberUri); |
| } |
| |
| // Test support for video rotation header extension. |
| TEST_F(WebRtcVideoChannelTest, SendVideoRotationHeaderExtensions) { |
| TestSetSendRtpHeaderExtensions(RtpExtension::kVideoRotationUri); |
| } |
| TEST_F(WebRtcVideoChannelTest, RecvVideoRotationHeaderExtensions) { |
| TestSetRecvRtpHeaderExtensions(RtpExtension::kVideoRotationUri); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, IdenticalSendExtensionsDoesntRecreateStream) { |
| const int kAbsSendTimeId = 1; |
| const int kVideoRotationId = 2; |
| send_parameters_.extensions.push_back( |
| RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId)); |
| send_parameters_.extensions.push_back( |
| RtpExtension(RtpExtension::kVideoRotationUri, kVideoRotationId)); |
| |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| FakeVideoSendStream* send_stream = |
| AddSendStream(cricket::StreamParams::CreateLegacy(123)); |
| |
| EXPECT_EQ(1, fake_call_->GetNumCreatedSendStreams()); |
| ASSERT_EQ(2u, send_stream->GetConfig().rtp.extensions.size()); |
| |
| // Setting the same extensions (even if in different order) shouldn't |
| // reallocate the stream. |
| absl::c_reverse(send_parameters_.extensions); |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| |
| EXPECT_EQ(1, fake_call_->GetNumCreatedSendStreams()); |
| |
| // Setting different extensions should recreate the stream. |
| send_parameters_.extensions.resize(1); |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| |
| EXPECT_EQ(2, fake_call_->GetNumCreatedSendStreams()); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, IdenticalRecvExtensionsDoesntRecreateStream) { |
| const int kTOffsetId = 1; |
| const int kAbsSendTimeId = 2; |
| const int kVideoRotationId = 3; |
| recv_parameters_.extensions.push_back( |
| RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId)); |
| recv_parameters_.extensions.push_back( |
| RtpExtension(RtpExtension::kTimestampOffsetUri, kTOffsetId)); |
| recv_parameters_.extensions.push_back( |
| RtpExtension(RtpExtension::kVideoRotationUri, kVideoRotationId)); |
| |
| EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_)); |
| FakeVideoReceiveStream* recv_stream = |
| AddRecvStream(cricket::StreamParams::CreateLegacy(123)); |
| |
| EXPECT_EQ(1, fake_call_->GetNumCreatedReceiveStreams()); |
| ASSERT_EQ(3u, recv_stream->GetConfig().rtp.extensions.size()); |
| |
| // Setting the same extensions (even if in different order) shouldn't |
| // reallocate the stream. |
| absl::c_reverse(recv_parameters_.extensions); |
| EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_)); |
| |
| EXPECT_EQ(1, fake_call_->GetNumCreatedReceiveStreams()); |
| |
| // Setting different extensions should not require the stream to be recreated. |
| recv_parameters_.extensions.resize(1); |
| EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_)); |
| |
| EXPECT_EQ(1, fake_call_->GetNumCreatedReceiveStreams()); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| SetSendRtpHeaderExtensionsExcludeUnsupportedExtensions) { |
| const int kUnsupportedId = 1; |
| const int kTOffsetId = 2; |
| |
| send_parameters_.extensions.push_back( |
| RtpExtension(kUnsupportedExtensionName, kUnsupportedId)); |
| send_parameters_.extensions.push_back( |
| RtpExtension(RtpExtension::kTimestampOffsetUri, kTOffsetId)); |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| FakeVideoSendStream* send_stream = |
| AddSendStream(cricket::StreamParams::CreateLegacy(123)); |
| |
| // Only timestamp offset extension is set to send stream, |
| // unsupported rtp extension is ignored. |
| ASSERT_EQ(1u, send_stream->GetConfig().rtp.extensions.size()); |
| EXPECT_STREQ(RtpExtension::kTimestampOffsetUri, |
| send_stream->GetConfig().rtp.extensions[0].uri.c_str()); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| SetRecvRtpHeaderExtensionsExcludeUnsupportedExtensions) { |
| const int kUnsupportedId = 1; |
| const int kTOffsetId = 2; |
| |
| recv_parameters_.extensions.push_back( |
| RtpExtension(kUnsupportedExtensionName, kUnsupportedId)); |
| recv_parameters_.extensions.push_back( |
| RtpExtension(RtpExtension::kTimestampOffsetUri, kTOffsetId)); |
| EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_)); |
| FakeVideoReceiveStream* recv_stream = |
| AddRecvStream(cricket::StreamParams::CreateLegacy(123)); |
| |
| // Only timestamp offset extension is set to receive stream, |
| // unsupported rtp extension is ignored. |
| ASSERT_EQ(1u, recv_stream->GetConfig().rtp.extensions.size()); |
| EXPECT_STREQ(RtpExtension::kTimestampOffsetUri, |
| recv_stream->GetConfig().rtp.extensions[0].uri.c_str()); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetSendRtpHeaderExtensionsRejectsIncorrectIds) { |
| const int kIncorrectIds[] = {-2, -1, 0, 15, 16}; |
| for (size_t i = 0; i < arraysize(kIncorrectIds); ++i) { |
| send_parameters_.extensions.push_back( |
| RtpExtension(RtpExtension::kTimestampOffsetUri, kIncorrectIds[i])); |
| EXPECT_FALSE(channel_->SetSendParameters(send_parameters_)) |
| << "Bad extension id '" << kIncorrectIds[i] << "' accepted."; |
| } |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetRecvRtpHeaderExtensionsRejectsIncorrectIds) { |
| const int kIncorrectIds[] = {-2, -1, 0, 15, 16}; |
| for (size_t i = 0; i < arraysize(kIncorrectIds); ++i) { |
| recv_parameters_.extensions.push_back( |
| RtpExtension(RtpExtension::kTimestampOffsetUri, kIncorrectIds[i])); |
| EXPECT_FALSE(channel_->SetRecvParameters(recv_parameters_)) |
| << "Bad extension id '" << kIncorrectIds[i] << "' accepted."; |
| } |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetSendRtpHeaderExtensionsRejectsDuplicateIds) { |
| const int id = 1; |
| send_parameters_.extensions.push_back( |
| RtpExtension(RtpExtension::kTimestampOffsetUri, id)); |
| send_parameters_.extensions.push_back( |
| RtpExtension(RtpExtension::kAbsSendTimeUri, id)); |
| EXPECT_FALSE(channel_->SetSendParameters(send_parameters_)); |
| |
| // Duplicate entries are also not supported. |
| send_parameters_.extensions.clear(); |
| send_parameters_.extensions.push_back( |
| RtpExtension(RtpExtension::kTimestampOffsetUri, id)); |
| send_parameters_.extensions.push_back(send_parameters_.extensions.back()); |
| EXPECT_FALSE(channel_->SetSendParameters(send_parameters_)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetRecvRtpHeaderExtensionsRejectsDuplicateIds) { |
| const int id = 1; |
| recv_parameters_.extensions.push_back( |
| RtpExtension(RtpExtension::kTimestampOffsetUri, id)); |
| recv_parameters_.extensions.push_back( |
| RtpExtension(RtpExtension::kAbsSendTimeUri, id)); |
| EXPECT_FALSE(channel_->SetRecvParameters(recv_parameters_)); |
| |
| // Duplicate entries are also not supported. |
| recv_parameters_.extensions.clear(); |
| recv_parameters_.extensions.push_back( |
| RtpExtension(RtpExtension::kTimestampOffsetUri, id)); |
| recv_parameters_.extensions.push_back(recv_parameters_.extensions.back()); |
| EXPECT_FALSE(channel_->SetRecvParameters(recv_parameters_)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, AddRecvStreamOnlyUsesOneReceiveStream) { |
| EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(1))); |
| EXPECT_EQ(1u, fake_call_->GetVideoReceiveStreams().size()); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, RtcpIsCompoundByDefault) { |
| FakeVideoReceiveStream* stream = AddRecvStream(); |
| EXPECT_EQ(webrtc::RtcpMode::kCompound, stream->GetConfig().rtp.rtcp_mode); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, TransportCcIsEnabledByDefault) { |
| FakeVideoReceiveStream* stream = AddRecvStream(); |
| EXPECT_TRUE(stream->GetConfig().rtp.transport_cc); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, TransportCcCanBeEnabledAndDisabled) { |
| FakeVideoReceiveStream* stream = AddRecvStream(); |
| EXPECT_TRUE(stream->GetConfig().rtp.transport_cc); |
| |
| // Verify that transport cc feedback is turned off when send(!) codecs without |
| // transport cc feedback are set. |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(RemoveFeedbackParams(GetEngineCodec("VP8"))); |
| EXPECT_TRUE(parameters.codecs[0].feedback_params.params().empty()); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| stream = fake_call_->GetVideoReceiveStreams()[0]; |
| EXPECT_FALSE(stream->GetConfig().rtp.transport_cc); |
| |
| // Verify that transport cc feedback is turned on when setting default codecs |
| // since the default codecs have transport cc feedback enabled. |
| parameters.codecs = engine_.send_codecs(); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| stream = fake_call_->GetVideoReceiveStreams()[0]; |
| EXPECT_TRUE(stream->GetConfig().rtp.transport_cc); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, LossNotificationIsDisabledByDefault) { |
| TestLossNotificationState(false); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, LossNotificationIsEnabledByFieldTrial) { |
| RTC_DCHECK(!override_field_trials_); |
| override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>( |
| "WebRTC-RtcpLossNotification/Enabled/"); |
| ResetTest(); |
| TestLossNotificationState(true); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, LossNotificationCanBeEnabledAndDisabled) { |
| RTC_DCHECK(!override_field_trials_); |
| override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>( |
| "WebRTC-RtcpLossNotification/Enabled/"); |
| ResetTest(); |
| |
| AssignDefaultCodec(); |
| VerifyCodecHasDefaultFeedbackParams(default_codec_, true); |
| |
| { |
| cricket::VideoSendParameters parameters; |
| parameters.codecs = engine_.send_codecs(); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_TRUE(channel_->SetSend(true)); |
| } |
| |
| // Start with LNTF enabled. |
| FakeVideoSendStream* send_stream = |
| AddSendStream(cricket::StreamParams::CreateLegacy(1)); |
| ASSERT_TRUE(send_stream->GetConfig().rtp.lntf.enabled); |
| FakeVideoReceiveStream* recv_stream = |
| AddRecvStream(cricket::StreamParams::CreateLegacy(1)); |
| ASSERT_TRUE(recv_stream->GetConfig().rtp.lntf.enabled); |
| |
| // Verify that LNTF is turned off when send(!) codecs without LNTF are set. |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(RemoveFeedbackParams(GetEngineCodec("VP8"))); |
| EXPECT_TRUE(parameters.codecs[0].feedback_params.params().empty()); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| recv_stream = fake_call_->GetVideoReceiveStreams()[0]; |
| EXPECT_FALSE(recv_stream->GetConfig().rtp.lntf.enabled); |
| send_stream = fake_call_->GetVideoSendStreams()[0]; |
| EXPECT_FALSE(send_stream->GetConfig().rtp.lntf.enabled); |
| |
| // Setting the default codecs again, including VP8, turns LNTF back on. |
| parameters.codecs = engine_.send_codecs(); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| recv_stream = fake_call_->GetVideoReceiveStreams()[0]; |
| EXPECT_TRUE(recv_stream->GetConfig().rtp.lntf.enabled); |
| send_stream = fake_call_->GetVideoSendStreams()[0]; |
| EXPECT_TRUE(send_stream->GetConfig().rtp.lntf.enabled); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, NackIsEnabledByDefault) { |
| AssignDefaultCodec(); |
| VerifyCodecHasDefaultFeedbackParams(default_codec_, false); |
| |
| cricket::VideoSendParameters parameters; |
| parameters.codecs = engine_.send_codecs(); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_TRUE(channel_->SetSend(true)); |
| |
| // Send side. |
| FakeVideoSendStream* send_stream = |
| AddSendStream(cricket::StreamParams::CreateLegacy(1)); |
| EXPECT_GT(send_stream->GetConfig().rtp.nack.rtp_history_ms, 0); |
| |
| // Receiver side. |
| FakeVideoReceiveStream* recv_stream = |
| AddRecvStream(cricket::StreamParams::CreateLegacy(1)); |
| EXPECT_GT(recv_stream->GetConfig().rtp.nack.rtp_history_ms, 0); |
| |
| // Nack history size should match between sender and receiver. |
| EXPECT_EQ(send_stream->GetConfig().rtp.nack.rtp_history_ms, |
| recv_stream->GetConfig().rtp.nack.rtp_history_ms); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, NackCanBeEnabledAndDisabled) { |
| FakeVideoSendStream* send_stream = AddSendStream(); |
| FakeVideoReceiveStream* recv_stream = AddRecvStream(); |
| |
| EXPECT_GT(recv_stream->GetConfig().rtp.nack.rtp_history_ms, 0); |
| EXPECT_GT(send_stream->GetConfig().rtp.nack.rtp_history_ms, 0); |
| |
| // Verify that NACK is turned off when send(!) codecs without NACK are set. |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(RemoveFeedbackParams(GetEngineCodec("VP8"))); |
| EXPECT_TRUE(parameters.codecs[0].feedback_params.params().empty()); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| recv_stream = fake_call_->GetVideoReceiveStreams()[0]; |
| EXPECT_EQ(0, recv_stream->GetConfig().rtp.nack.rtp_history_ms); |
| send_stream = fake_call_->GetVideoSendStreams()[0]; |
| EXPECT_EQ(0, send_stream->GetConfig().rtp.nack.rtp_history_ms); |
| |
| // Verify that NACK is turned on when setting default codecs since the |
| // default codecs have NACK enabled. |
| parameters.codecs = engine_.send_codecs(); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| recv_stream = fake_call_->GetVideoReceiveStreams()[0]; |
| EXPECT_GT(recv_stream->GetConfig().rtp.nack.rtp_history_ms, 0); |
| send_stream = fake_call_->GetVideoSendStreams()[0]; |
| EXPECT_GT(send_stream->GetConfig().rtp.nack.rtp_history_ms, 0); |
| } |
| |
| // This test verifies that new frame sizes reconfigures encoders even though not |
| // (yet) sending. The purpose of this is to permit encoding as quickly as |
| // possible once we start sending. Likely the frames being input are from the |
| // same source that will be sent later, which just means that we're ready |
| // earlier. |
| TEST_F(WebRtcVideoChannelTest, ReconfiguresEncodersWhenNotSending) { |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| ASSERT_TRUE(channel_->SetSendParameters(parameters)); |
| channel_->SetSend(false); |
| |
| FakeVideoSendStream* stream = AddSendStream(); |
| |
| // No frames entered. |
| std::vector<webrtc::VideoStream> streams = stream->GetVideoStreams(); |
| EXPECT_EQ(0u, streams[0].width); |
| EXPECT_EQ(0u, streams[0].height); |
| |
| webrtc::test::FrameForwarder frame_forwarder; |
| cricket::FakeFrameSource frame_source(1280, 720, |
| rtc::kNumMicrosecsPerSec / 30); |
| |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder)); |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| |
| // Frame entered, should be reconfigured to new dimensions. |
| streams = stream->GetVideoStreams(); |
| EXPECT_EQ(rtc::checked_cast<size_t>(1280), streams[0].width); |
| EXPECT_EQ(rtc::checked_cast<size_t>(720), streams[0].height); |
| |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, UsesCorrectSettingsForScreencast) { |
| static const int kScreenshareMinBitrateKbps = 800; |
| cricket::VideoCodec codec = GetEngineCodec("VP8"); |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(codec); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| AddSendStream(); |
| |
| webrtc::test::FrameForwarder frame_forwarder; |
| cricket::FakeFrameSource frame_source(1280, 720, |
| rtc::kNumMicrosecsPerSec / 30); |
| VideoOptions min_bitrate_options; |
| min_bitrate_options.screencast_min_bitrate_kbps = kScreenshareMinBitrateKbps; |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &min_bitrate_options, |
| &frame_forwarder)); |
| |
| EXPECT_TRUE(channel_->SetSend(true)); |
| |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| ASSERT_EQ(1u, fake_call_->GetVideoSendStreams().size()); |
| FakeVideoSendStream* send_stream = fake_call_->GetVideoSendStreams().front(); |
| |
| EXPECT_EQ(1, send_stream->GetNumberOfSwappedFrames()); |
| |
| // Verify non-screencast settings. |
| webrtc::VideoEncoderConfig encoder_config = |
| send_stream->GetEncoderConfig().Copy(); |
| EXPECT_EQ(webrtc::VideoEncoderConfig::ContentType::kRealtimeVideo, |
| encoder_config.content_type); |
| std::vector<webrtc::VideoStream> streams = send_stream->GetVideoStreams(); |
| EXPECT_EQ(rtc::checked_cast<size_t>(1280), streams.front().width); |
| EXPECT_EQ(rtc::checked_cast<size_t>(720), streams.front().height); |
| EXPECT_EQ(0, encoder_config.min_transmit_bitrate_bps) |
| << "Non-screenshare shouldn't use min-transmit bitrate."; |
| |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr)); |
| EXPECT_EQ(1, send_stream->GetNumberOfSwappedFrames()); |
| VideoOptions screencast_options; |
| screencast_options.is_screencast = true; |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &screencast_options, |
| &frame_forwarder)); |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| // Send stream recreated after option change. |
| ASSERT_EQ(2, fake_call_->GetNumCreatedSendStreams()); |
| send_stream = fake_call_->GetVideoSendStreams().front(); |
| EXPECT_EQ(1, send_stream->GetNumberOfSwappedFrames()); |
| |
| // Verify screencast settings. |
| encoder_config = send_stream->GetEncoderConfig().Copy(); |
| EXPECT_EQ(webrtc::VideoEncoderConfig::ContentType::kScreen, |
| encoder_config.content_type); |
| EXPECT_EQ(kScreenshareMinBitrateKbps * 1000, |
| encoder_config.min_transmit_bitrate_bps); |
| |
| streams = send_stream->GetVideoStreams(); |
| EXPECT_EQ(rtc::checked_cast<size_t>(1280), streams.front().width); |
| EXPECT_EQ(rtc::checked_cast<size_t>(720), streams.front().height); |
| EXPECT_FALSE(streams[0].num_temporal_layers.has_value()); |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| ConferenceModeScreencastConfiguresTemporalLayer) { |
| static const int kConferenceScreencastTemporalBitrateBps = 200 * 1000; |
| send_parameters_.conference_mode = true; |
| channel_->SetSendParameters(send_parameters_); |
| |
| AddSendStream(); |
| VideoOptions options; |
| options.is_screencast = true; |
| webrtc::test::FrameForwarder frame_forwarder; |
| cricket::FakeFrameSource frame_source(1280, 720, |
| rtc::kNumMicrosecsPerSec / 30); |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder)); |
| EXPECT_TRUE(channel_->SetSend(true)); |
| |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| ASSERT_EQ(1u, fake_call_->GetVideoSendStreams().size()); |
| FakeVideoSendStream* send_stream = fake_call_->GetVideoSendStreams().front(); |
| |
| webrtc::VideoEncoderConfig encoder_config = |
| send_stream->GetEncoderConfig().Copy(); |
| |
| // Verify screencast settings. |
| encoder_config = send_stream->GetEncoderConfig().Copy(); |
| EXPECT_EQ(webrtc::VideoEncoderConfig::ContentType::kScreen, |
| encoder_config.content_type); |
| |
| std::vector<webrtc::VideoStream> streams = send_stream->GetVideoStreams(); |
| ASSERT_EQ(1u, streams.size()); |
| ASSERT_EQ(2u, streams[0].num_temporal_layers); |
| EXPECT_EQ(kConferenceScreencastTemporalBitrateBps, |
| streams[0].target_bitrate_bps); |
| |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SuspendBelowMinBitrateDisabledByDefault) { |
| FakeVideoSendStream* stream = AddSendStream(); |
| EXPECT_FALSE(stream->GetConfig().suspend_below_min_bitrate); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetMediaConfigSuspendBelowMinBitrate) { |
| MediaConfig media_config = GetMediaConfig(); |
| media_config.video.suspend_below_min_bitrate = true; |
| |
| channel_.reset(engine_.CreateMediaChannel( |
| fake_call_.get(), media_config, VideoOptions(), webrtc::CryptoOptions(), |
| video_bitrate_allocator_factory_.get())); |
| channel_->OnReadyToSend(true); |
| |
| channel_->SetSendParameters(send_parameters_); |
| |
| FakeVideoSendStream* stream = AddSendStream(); |
| EXPECT_TRUE(stream->GetConfig().suspend_below_min_bitrate); |
| |
| media_config.video.suspend_below_min_bitrate = false; |
| channel_.reset(engine_.CreateMediaChannel( |
| fake_call_.get(), media_config, VideoOptions(), webrtc::CryptoOptions(), |
| video_bitrate_allocator_factory_.get())); |
| channel_->OnReadyToSend(true); |
| |
| channel_->SetSendParameters(send_parameters_); |
| |
| stream = AddSendStream(); |
| EXPECT_FALSE(stream->GetConfig().suspend_below_min_bitrate); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, Vp8DenoisingEnabledByDefault) { |
| FakeVideoSendStream* stream = AddSendStream(); |
| webrtc::VideoCodecVP8 vp8_settings; |
| ASSERT_TRUE(stream->GetVp8Settings(&vp8_settings)) << "No VP8 config set."; |
| EXPECT_TRUE(vp8_settings.denoisingOn); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, VerifyVp8SpecificSettings) { |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| ASSERT_TRUE(channel_->SetSendParameters(parameters)); |
| |
| // Single-stream settings should apply with RTX as well (verifies that we |
| // check number of regular SSRCs and not StreamParams::ssrcs which contains |
| // both RTX and regular SSRCs). |
| FakeVideoSendStream* stream = SetUpSimulcast(false, true); |
| |
| webrtc::test::FrameForwarder frame_forwarder; |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder)); |
| channel_->SetSend(true); |
| |
| frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame()); |
| |
| webrtc::VideoCodecVP8 vp8_settings; |
| ASSERT_TRUE(stream->GetVp8Settings(&vp8_settings)) << "No VP8 config set."; |
| EXPECT_TRUE(vp8_settings.denoisingOn) |
| << "VP8 denoising should be on by default."; |
| |
| stream = SetDenoisingOption(last_ssrc_, &frame_forwarder, false); |
| |
| ASSERT_TRUE(stream->GetVp8Settings(&vp8_settings)) << "No VP8 config set."; |
| EXPECT_FALSE(vp8_settings.denoisingOn); |
| EXPECT_TRUE(vp8_settings.automaticResizeOn); |
| EXPECT_TRUE(vp8_settings.frameDroppingOn); |
| |
| stream = SetDenoisingOption(last_ssrc_, &frame_forwarder, true); |
| |
| ASSERT_TRUE(stream->GetVp8Settings(&vp8_settings)) << "No VP8 config set."; |
| EXPECT_TRUE(vp8_settings.denoisingOn); |
| EXPECT_TRUE(vp8_settings.automaticResizeOn); |
| EXPECT_TRUE(vp8_settings.frameDroppingOn); |
| |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr)); |
| stream = SetUpSimulcast(true, false); |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder)); |
| channel_->SetSend(true); |
| frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame()); |
| |
| EXPECT_EQ(3u, stream->GetVideoStreams().size()); |
| ASSERT_TRUE(stream->GetVp8Settings(&vp8_settings)) << "No VP8 config set."; |
| // Autmatic resize off when using simulcast. |
| EXPECT_FALSE(vp8_settings.automaticResizeOn); |
| EXPECT_TRUE(vp8_settings.frameDroppingOn); |
| |
| // In screen-share mode, denoising is forced off. |
| VideoOptions options; |
| options.is_screencast = true; |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder)); |
| |
| stream = SetDenoisingOption(last_ssrc_, &frame_forwarder, false); |
| |
| EXPECT_EQ(3u, stream->GetVideoStreams().size()); |
| ASSERT_TRUE(stream->GetVp8Settings(&vp8_settings)) << "No VP8 config set."; |
| EXPECT_FALSE(vp8_settings.denoisingOn); |
| // Resizing and frame dropping always off for screen sharing. |
| EXPECT_FALSE(vp8_settings.automaticResizeOn); |
| EXPECT_FALSE(vp8_settings.frameDroppingOn); |
| |
| stream = SetDenoisingOption(last_ssrc_, &frame_forwarder, true); |
| |
| ASSERT_TRUE(stream->GetVp8Settings(&vp8_settings)) << "No VP8 config set."; |
| EXPECT_FALSE(vp8_settings.denoisingOn); |
| EXPECT_FALSE(vp8_settings.automaticResizeOn); |
| EXPECT_FALSE(vp8_settings.frameDroppingOn); |
| |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr)); |
| } |
| |
| // Test that setting the same options doesn't result in the encoder being |
| // reconfigured. |
| TEST_F(WebRtcVideoChannelTest, SetIdenticalOptionsDoesntReconfigureEncoder) { |
| VideoOptions options; |
| webrtc::test::FrameForwarder frame_forwarder; |
| |
| AddSendStream(); |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| ASSERT_TRUE(channel_->SetSendParameters(parameters)); |
| FakeVideoSendStream* send_stream = fake_call_->GetVideoSendStreams().front(); |
| |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder)); |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder)); |
| frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame()); |
| // Expect 1 reconfigurations at this point from the initial configuration. |
| EXPECT_EQ(1, send_stream->num_encoder_reconfigurations()); |
| |
| // Set the options one more time and expect no additional reconfigurations. |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder)); |
| EXPECT_EQ(1, send_stream->num_encoder_reconfigurations()); |
| |
| // Change `options` and expect 2 reconfigurations. |
| options.video_noise_reduction = true; |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder)); |
| EXPECT_EQ(2, send_stream->num_encoder_reconfigurations()); |
| |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr)); |
| } |
| |
| class Vp9SettingsTest : public WebRtcVideoChannelTest { |
| public: |
| Vp9SettingsTest() : Vp9SettingsTest("") {} |
| explicit Vp9SettingsTest(const char* field_trials) |
| : WebRtcVideoChannelTest(field_trials) { |
| encoder_factory_->AddSupportedVideoCodecType("VP9"); |
| } |
| virtual ~Vp9SettingsTest() {} |
| |
| protected: |
| void TearDown() override { |
| // Remove references to encoder_factory_ since this will be destroyed |
| // before channel_ and engine_. |
| ASSERT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| } |
| }; |
| |
| TEST_F(Vp9SettingsTest, VerifyVp9SpecificSettings) { |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP9")); |
| ASSERT_TRUE(channel_->SetSendParameters(parameters)); |
| |
| FakeVideoSendStream* stream = SetUpSimulcast(false, false); |
| |
| webrtc::test::FrameForwarder frame_forwarder; |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder)); |
| channel_->SetSend(true); |
| |
| frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame()); |
| |
| webrtc::VideoCodecVP9 vp9_settings; |
| ASSERT_TRUE(stream->GetVp9Settings(&vp9_settings)) << "No VP9 config set."; |
| EXPECT_TRUE(vp9_settings.denoisingOn) |
| << "VP9 denoising should be on by default."; |
| |
| stream = SetDenoisingOption(last_ssrc_, &frame_forwarder, false); |
| |
| ASSERT_TRUE(stream->GetVp9Settings(&vp9_settings)) << "No VP9 config set."; |
| EXPECT_FALSE(vp9_settings.denoisingOn); |
| // Frame dropping always on for real time video. |
| EXPECT_TRUE(vp9_settings.frameDroppingOn); |
| |
| stream = SetDenoisingOption(last_ssrc_, &frame_forwarder, true); |
| |
| ASSERT_TRUE(stream->GetVp9Settings(&vp9_settings)) << "No VP9 config set."; |
| EXPECT_TRUE(vp9_settings.denoisingOn); |
| EXPECT_TRUE(vp9_settings.frameDroppingOn); |
| |
| // In screen-share mode, denoising is forced off. |
| VideoOptions options; |
| options.is_screencast = true; |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder)); |
| |
| stream = SetDenoisingOption(last_ssrc_, &frame_forwarder, false); |
| |
| ASSERT_TRUE(stream->GetVp9Settings(&vp9_settings)) << "No VP9 config set."; |
| EXPECT_FALSE(vp9_settings.denoisingOn); |
| // Frame dropping always on for screen sharing. |
| EXPECT_TRUE(vp9_settings.frameDroppingOn); |
| |
| stream = SetDenoisingOption(last_ssrc_, &frame_forwarder, false); |
| |
| ASSERT_TRUE(stream->GetVp9Settings(&vp9_settings)) << "No VP9 config set."; |
| EXPECT_FALSE(vp9_settings.denoisingOn); |
| EXPECT_TRUE(vp9_settings.frameDroppingOn); |
| |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr)); |
| } |
| |
| TEST_F(Vp9SettingsTest, MultipleSsrcsEnablesSvc) { |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP9")); |
| ASSERT_TRUE(channel_->SetSendParameters(parameters)); |
| |
| std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3); |
| |
| FakeVideoSendStream* stream = |
| AddSendStream(CreateSimStreamParams("cname", ssrcs)); |
| |
| webrtc::VideoSendStream::Config config = stream->GetConfig().Copy(); |
| |
| webrtc::test::FrameForwarder frame_forwarder; |
| EXPECT_TRUE(channel_->SetVideoSend(ssrcs[0], nullptr, &frame_forwarder)); |
| channel_->SetSend(true); |
| |
| frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame()); |
| |
| webrtc::VideoCodecVP9 vp9_settings; |
| ASSERT_TRUE(stream->GetVp9Settings(&vp9_settings)) << "No VP9 config set."; |
| |
| const size_t kNumSpatialLayers = ssrcs.size(); |
| const size_t kNumTemporalLayers = 3; |
| EXPECT_EQ(vp9_settings.numberOfSpatialLayers, kNumSpatialLayers); |
| EXPECT_EQ(vp9_settings.numberOfTemporalLayers, kNumTemporalLayers); |
| |
| EXPECT_TRUE(channel_->SetVideoSend(ssrcs[0], nullptr, nullptr)); |
| } |
| |
| TEST_F(Vp9SettingsTest, SvcModeCreatesSingleRtpStream) { |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP9")); |
| ASSERT_TRUE(channel_->SetSendParameters(parameters)); |
| |
| std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3); |
| |
| FakeVideoSendStream* stream = |
| AddSendStream(CreateSimStreamParams("cname", ssrcs)); |
| |
| webrtc::VideoSendStream::Config config = stream->GetConfig().Copy(); |
| |
| // Despite 3 ssrcs provided, single layer is used. |
| EXPECT_EQ(1u, config.rtp.ssrcs.size()); |
| |
| webrtc::test::FrameForwarder frame_forwarder; |
| EXPECT_TRUE(channel_->SetVideoSend(ssrcs[0], nullptr, &frame_forwarder)); |
| channel_->SetSend(true); |
| |
| frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame()); |
| |
| webrtc::VideoCodecVP9 vp9_settings; |
| ASSERT_TRUE(stream->GetVp9Settings(&vp9_settings)) << "No VP9 config set."; |
| |
| const size_t kNumSpatialLayers = ssrcs.size(); |
| EXPECT_EQ(vp9_settings.numberOfSpatialLayers, kNumSpatialLayers); |
| |
| EXPECT_TRUE(channel_->SetVideoSend(ssrcs[0], nullptr, nullptr)); |
| } |
| |
| TEST_F(Vp9SettingsTest, AllEncodingParametersCopied) { |
| cricket::VideoSendParameters send_parameters; |
| send_parameters.codecs.push_back(GetEngineCodec("VP9")); |
| ASSERT_TRUE(channel_->SetSendParameters(send_parameters)); |
| |
| const size_t kNumSpatialLayers = 3; |
| std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3); |
| |
| FakeVideoSendStream* stream = |
| AddSendStream(CreateSimStreamParams("cname", ssrcs)); |
| |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(ssrcs[0]); |
| ASSERT_EQ(kNumSpatialLayers, parameters.encodings.size()); |
| ASSERT_TRUE(parameters.encodings[0].active); |
| ASSERT_TRUE(parameters.encodings[1].active); |
| ASSERT_TRUE(parameters.encodings[2].active); |
| // Invert value to verify copying. |
| parameters.encodings[1].active = false; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(ssrcs[0], parameters).ok()); |
| |
| webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy(); |
| |
| // number_of_streams should be 1 since all spatial layers are sent on the |
| // same SSRC. But encoding parameters of all layers is supposed to be copied |
| // and stored in simulcast_layers[]. |
| EXPECT_EQ(1u, encoder_config.number_of_streams); |
| EXPECT_EQ(encoder_config.simulcast_layers.size(), kNumSpatialLayers); |
| EXPECT_TRUE(encoder_config.simulcast_layers[0].active); |
| EXPECT_FALSE(encoder_config.simulcast_layers[1].active); |
| EXPECT_TRUE(encoder_config.simulcast_layers[2].active); |
| } |
| |
| class Vp9SettingsTestWithFieldTrial |
| : public Vp9SettingsTest, |
| public ::testing::WithParamInterface< |
| ::testing::tuple<const char*, int, int, webrtc::InterLayerPredMode>> { |
| protected: |
| Vp9SettingsTestWithFieldTrial() |
| : Vp9SettingsTest(::testing::get<0>(GetParam())), |
| num_spatial_layers_(::testing::get<1>(GetParam())), |
| num_temporal_layers_(::testing::get<2>(GetParam())), |
| inter_layer_pred_mode_(::testing::get<3>(GetParam())) {} |
| |
| void VerifySettings(int num_spatial_layers, |
| int num_temporal_layers, |
| webrtc::InterLayerPredMode interLayerPred) { |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP9")); |
| ASSERT_TRUE(channel_->SetSendParameters(parameters)); |
| |
| FakeVideoSendStream* stream = SetUpSimulcast(false, false); |
| |
| webrtc::test::FrameForwarder frame_forwarder; |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder)); |
| channel_->SetSend(true); |
| |
| frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame()); |
| |
| webrtc::VideoCodecVP9 vp9_settings; |
| ASSERT_TRUE(stream->GetVp9Settings(&vp9_settings)) << "No VP9 config set."; |
| EXPECT_EQ(num_spatial_layers, vp9_settings.numberOfSpatialLayers); |
| EXPECT_EQ(num_temporal_layers, vp9_settings.numberOfTemporalLayers); |
| EXPECT_EQ(inter_layer_pred_mode_, vp9_settings.interLayerPred); |
| |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr)); |
| } |
| |
| const uint8_t num_spatial_layers_; |
| const uint8_t num_temporal_layers_; |
| const webrtc::InterLayerPredMode inter_layer_pred_mode_; |
| }; |
| |
| TEST_P(Vp9SettingsTestWithFieldTrial, VerifyCodecSettings) { |
| VerifySettings(num_spatial_layers_, num_temporal_layers_, |
| inter_layer_pred_mode_); |
| } |
| |
| INSTANTIATE_TEST_SUITE_P( |
| All, |
| Vp9SettingsTestWithFieldTrial, |
| Values( |
| std::make_tuple("", 1, 1, webrtc::InterLayerPredMode::kOnKeyPic), |
| std::make_tuple("WebRTC-SupportVP9SVC/Default/", |
| 1, |
| 1, |
| webrtc::InterLayerPredMode::kOnKeyPic), |
| std::make_tuple("WebRTC-SupportVP9SVC/EnabledByFlag_2SL3TL/", |
| 2, |
| 3, |
| webrtc::InterLayerPredMode::kOnKeyPic), |
| std::make_tuple("WebRTC-Vp9InterLayerPred/Default/", |
| 1, |
| 1, |
| webrtc::InterLayerPredMode::kOnKeyPic), |
| std::make_tuple("WebRTC-Vp9InterLayerPred/Disabled/", |
| 1, |
| 1, |
| webrtc::InterLayerPredMode::kOnKeyPic), |
| std::make_tuple( |
| "WebRTC-Vp9InterLayerPred/Enabled,inter_layer_pred_mode:off/", |
| 1, |
| 1, |
| webrtc::InterLayerPredMode::kOff), |
| std::make_tuple( |
| "WebRTC-Vp9InterLayerPred/Enabled,inter_layer_pred_mode:on/", |
| 1, |
| 1, |
| webrtc::InterLayerPredMode::kOn), |
| std::make_tuple( |
| "WebRTC-Vp9InterLayerPred/Enabled,inter_layer_pred_mode:onkeypic/", |
| 1, |
| 1, |
| webrtc::InterLayerPredMode::kOnKeyPic))); |
| |
| TEST_F(WebRtcVideoChannelTest, VerifyMinBitrate) { |
| std::vector<webrtc::VideoStream> streams = AddSendStream()->GetVideoStreams(); |
| ASSERT_EQ(1u, streams.size()); |
| EXPECT_EQ(webrtc::kDefaultMinVideoBitrateBps, streams[0].min_bitrate_bps); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, VerifyMinBitrateWithForcedFallbackFieldTrial) { |
| RTC_DCHECK(!override_field_trials_); |
| override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>( |
| "WebRTC-VP8-Forced-Fallback-Encoder-v2/Enabled-1,2,34567/"); |
| std::vector<webrtc::VideoStream> streams = AddSendStream()->GetVideoStreams(); |
| ASSERT_EQ(1u, streams.size()); |
| EXPECT_EQ(34567, streams[0].min_bitrate_bps); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| BalancedDegradationPreferenceNotSupportedWithoutFieldtrial) { |
| RTC_DCHECK(!override_field_trials_); |
| override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>( |
| "WebRTC-Video-BalancedDegradation/Disabled/"); |
| const bool kResolutionScalingEnabled = true; |
| const bool kFpsScalingEnabled = false; |
| TestDegradationPreference(kResolutionScalingEnabled, kFpsScalingEnabled); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| BalancedDegradationPreferenceSupportedBehindFieldtrial) { |
| RTC_DCHECK(!override_field_trials_); |
| override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>( |
| "WebRTC-Video-BalancedDegradation/Enabled/"); |
| const bool kResolutionScalingEnabled = true; |
| const bool kFpsScalingEnabled = true; |
| TestDegradationPreference(kResolutionScalingEnabled, kFpsScalingEnabled); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, AdaptsOnOveruse) { |
| TestCpuAdaptation(true, false); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, DoesNotAdaptOnOveruseWhenDisabled) { |
| TestCpuAdaptation(false, false); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, DoesNotAdaptWhenScreeensharing) { |
| TestCpuAdaptation(false, true); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, DoesNotAdaptOnOveruseWhenScreensharing) { |
| TestCpuAdaptation(true, true); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, PreviousAdaptationDoesNotApplyToScreenshare) { |
| cricket::VideoCodec codec = GetEngineCodec("VP8"); |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(codec); |
| |
| MediaConfig media_config = GetMediaConfig(); |
| media_config.video.enable_cpu_adaptation = true; |
| channel_.reset(engine_.CreateMediaChannel( |
| fake_call_.get(), media_config, VideoOptions(), webrtc::CryptoOptions(), |
| video_bitrate_allocator_factory_.get())); |
| channel_->OnReadyToSend(true); |
| ASSERT_TRUE(channel_->SetSendParameters(parameters)); |
| |
| AddSendStream(); |
| webrtc::test::FrameForwarder frame_forwarder; |
| |
| ASSERT_TRUE(channel_->SetSend(true)); |
| cricket::VideoOptions camera_options; |
| camera_options.is_screencast = false; |
| channel_->SetVideoSend(last_ssrc_, &camera_options, &frame_forwarder); |
| |
| ASSERT_EQ(1u, fake_call_->GetVideoSendStreams().size()); |
| FakeVideoSendStream* send_stream = fake_call_->GetVideoSendStreams().front(); |
| |
| EXPECT_TRUE(send_stream->resolution_scaling_enabled()); |
| // Dont' expect anything on framerate_scaling_enabled, since the default is |
| // transitioning from MAINTAIN_FRAMERATE to BALANCED. |
| |
| // Switch to screen share. Expect no resolution scaling. |
| cricket::VideoOptions screenshare_options; |
| screenshare_options.is_screencast = true; |
| channel_->SetVideoSend(last_ssrc_, &screenshare_options, &frame_forwarder); |
| ASSERT_EQ(2, fake_call_->GetNumCreatedSendStreams()); |
| send_stream = fake_call_->GetVideoSendStreams().front(); |
| EXPECT_FALSE(send_stream->resolution_scaling_enabled()); |
| |
| // Switch back to the normal capturer. Expect resolution scaling to be |
| // reenabled. |
| channel_->SetVideoSend(last_ssrc_, &camera_options, &frame_forwarder); |
| send_stream = fake_call_->GetVideoSendStreams().front(); |
| ASSERT_EQ(3, fake_call_->GetNumCreatedSendStreams()); |
| send_stream = fake_call_->GetVideoSendStreams().front(); |
| EXPECT_TRUE(send_stream->resolution_scaling_enabled()); |
| |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr)); |
| } |
| |
| // TODO(asapersson): Remove this test when the balanced field trial is removed. |
| void WebRtcVideoChannelTest::TestDegradationPreference( |
| bool resolution_scaling_enabled, |
| bool fps_scaling_enabled) { |
| cricket::VideoCodec codec = GetEngineCodec("VP8"); |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(codec); |
| |
| MediaConfig media_config = GetMediaConfig(); |
| media_config.video.enable_cpu_adaptation = true; |
| channel_.reset(engine_.CreateMediaChannel( |
| fake_call_.get(), media_config, VideoOptions(), webrtc::CryptoOptions(), |
| video_bitrate_allocator_factory_.get())); |
| channel_->OnReadyToSend(true); |
| |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| |
| AddSendStream(); |
| |
| webrtc::test::FrameForwarder frame_forwarder; |
| VideoOptions options; |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder)); |
| |
| EXPECT_TRUE(channel_->SetSend(true)); |
| |
| FakeVideoSendStream* send_stream = fake_call_->GetVideoSendStreams().front(); |
| EXPECT_EQ(resolution_scaling_enabled, |
| send_stream->resolution_scaling_enabled()); |
| EXPECT_EQ(fps_scaling_enabled, send_stream->framerate_scaling_enabled()); |
| |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr)); |
| } |
| |
| void WebRtcVideoChannelTest::TestCpuAdaptation(bool enable_overuse, |
| bool is_screenshare) { |
| cricket::VideoCodec codec = GetEngineCodec("VP8"); |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(codec); |
| |
| MediaConfig media_config = GetMediaConfig(); |
| if (enable_overuse) { |
| media_config.video.enable_cpu_adaptation = true; |
| } |
| channel_.reset(engine_.CreateMediaChannel( |
| fake_call_.get(), media_config, VideoOptions(), webrtc::CryptoOptions(), |
| video_bitrate_allocator_factory_.get())); |
| channel_->OnReadyToSend(true); |
| |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| |
| AddSendStream(); |
| |
| webrtc::test::FrameForwarder frame_forwarder; |
| VideoOptions options; |
| options.is_screencast = is_screenshare; |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder)); |
| |
| EXPECT_TRUE(channel_->SetSend(true)); |
| |
| FakeVideoSendStream* send_stream = fake_call_->GetVideoSendStreams().front(); |
| |
| if (!enable_overuse) { |
| EXPECT_FALSE(send_stream->resolution_scaling_enabled()); |
| EXPECT_FALSE(send_stream->framerate_scaling_enabled()); |
| } else if (is_screenshare) { |
| EXPECT_FALSE(send_stream->resolution_scaling_enabled()); |
| EXPECT_TRUE(send_stream->framerate_scaling_enabled()); |
| } else { |
| EXPECT_TRUE(send_stream->resolution_scaling_enabled()); |
| EXPECT_FALSE(send_stream->framerate_scaling_enabled()); |
| } |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, EstimatesNtpStartTimeCorrectly) { |
| // Start at last timestamp to verify that wraparounds are estimated correctly. |
| static const uint32_t kInitialTimestamp = 0xFFFFFFFFu; |
| static const int64_t kInitialNtpTimeMs = 1247891230; |
| static const int kFrameOffsetMs = 20; |
| EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_)); |
| |
| FakeVideoReceiveStream* stream = AddRecvStream(); |
| cricket::FakeVideoRenderer renderer; |
| EXPECT_TRUE(channel_->SetSink(last_ssrc_, &renderer)); |
| |
| webrtc::VideoFrame video_frame = |
| webrtc::VideoFrame::Builder() |
| .set_video_frame_buffer(CreateBlackFrameBuffer(4, 4)) |
| .set_timestamp_rtp(kInitialTimestamp) |
| .set_timestamp_us(0) |
| .set_rotation(webrtc::kVideoRotation_0) |
| .build(); |
| // Initial NTP time is not available on the first frame, but should still be |
| // able to be estimated. |
| stream->InjectFrame(video_frame); |
| |
| EXPECT_EQ(1, renderer.num_rendered_frames()); |
| |
| // This timestamp is kInitialTimestamp (-1) + kFrameOffsetMs * 90, which |
| // triggers a constant-overflow warning, hence we're calculating it explicitly |
| // here. |
| fake_clock_.AdvanceTime(webrtc::TimeDelta::Millis(kFrameOffsetMs)); |
| video_frame.set_timestamp(kFrameOffsetMs * 90 - 1); |
| video_frame.set_ntp_time_ms(kInitialNtpTimeMs + kFrameOffsetMs); |
| stream->InjectFrame(video_frame); |
| |
| EXPECT_EQ(2, renderer.num_rendered_frames()); |
| |
| // Verify that NTP time has been correctly deduced. |
| cricket::VideoMediaInfo info; |
| ASSERT_TRUE(channel_->GetStats(&info)); |
| ASSERT_EQ(1u, info.receivers.size()); |
| EXPECT_EQ(kInitialNtpTimeMs, info.receivers[0].capture_start_ntp_time_ms); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetDefaultSendCodecs) { |
| AssignDefaultAptRtxTypes(); |
| ASSERT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| |
| VideoCodec codec; |
| EXPECT_TRUE(channel_->GetSendCodec(&codec)); |
| EXPECT_TRUE(codec.Matches(engine_.send_codecs()[0])); |
| |
| // Using a RTX setup to verify that the default RTX payload type is good. |
| const std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs1); |
| const std::vector<uint32_t> rtx_ssrcs = MAKE_VECTOR(kRtxSsrcs1); |
| FakeVideoSendStream* stream = AddSendStream( |
| cricket::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs)); |
| webrtc::VideoSendStream::Config config = stream->GetConfig().Copy(); |
| |
| // Make sure NACK and FEC are enabled on the correct payload types. |
| EXPECT_EQ(1000, config.rtp.nack.rtp_history_ms); |
| EXPECT_EQ(GetEngineCodec("ulpfec").id, config.rtp.ulpfec.ulpfec_payload_type); |
| EXPECT_EQ(GetEngineCodec("red").id, config.rtp.ulpfec.red_payload_type); |
| |
| EXPECT_EQ(1u, config.rtp.rtx.ssrcs.size()); |
| EXPECT_EQ(kRtxSsrcs1[0], config.rtp.rtx.ssrcs[0]); |
| VerifySendStreamHasRtxTypes(config, default_apt_rtx_types_); |
| // TODO(juberti): Check RTCP, PLI, TMMBR. |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithoutPacketization) { |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| |
| FakeVideoSendStream* stream = AddSendStream(); |
| const webrtc::VideoSendStream::Config config = stream->GetConfig().Copy(); |
| EXPECT_FALSE(config.rtp.raw_payload); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithPacketization) { |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| parameters.codecs.back().packetization = kPacketizationParamRaw; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| |
| FakeVideoSendStream* stream = AddSendStream(); |
| const webrtc::VideoSendStream::Config config = stream->GetConfig().Copy(); |
| EXPECT_TRUE(config.rtp.raw_payload); |
| } |
| |
| // The following four tests ensures that FlexFEC is not activated by default |
| // when the field trials are not enabled. |
| // TODO(brandtr): Remove or update these tests when FlexFEC _is_ enabled by |
| // default. |
| TEST_F(WebRtcVideoChannelTest, FlexfecSendCodecWithoutSsrcNotExposedByDefault) { |
| FakeVideoSendStream* stream = AddSendStream(); |
| webrtc::VideoSendStream::Config config = stream->GetConfig().Copy(); |
| |
| EXPECT_EQ(-1, config.rtp.flexfec.payload_type); |
| EXPECT_EQ(0U, config.rtp.flexfec.ssrc); |
| EXPECT_TRUE(config.rtp.flexfec.protected_media_ssrcs.empty()); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, FlexfecSendCodecWithSsrcNotExposedByDefault) { |
| FakeVideoSendStream* stream = AddSendStream( |
| CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc)); |
| webrtc::VideoSendStream::Config config = stream->GetConfig().Copy(); |
| |
| EXPECT_EQ(-1, config.rtp.flexfec.payload_type); |
| EXPECT_EQ(0U, config.rtp.flexfec.ssrc); |
| EXPECT_TRUE(config.rtp.flexfec.protected_media_ssrcs.empty()); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, FlexfecRecvCodecWithoutSsrcNotExposedByDefault) { |
| AddRecvStream(); |
| |
| const std::vector<FakeFlexfecReceiveStream*>& streams = |
| fake_call_->GetFlexfecReceiveStreams(); |
| EXPECT_TRUE(streams.empty()); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, FlexfecRecvCodecWithSsrcExposedByDefault) { |
| AddRecvStream( |
| CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc)); |
| |
| const std::vector<FakeFlexfecReceiveStream*>& streams = |
| fake_call_->GetFlexfecReceiveStreams(); |
| EXPECT_EQ(1U, streams.size()); |
| } |
| |
| // TODO(brandtr): When FlexFEC is no longer behind a field trial, merge all |
| // tests that use this test fixture into the corresponding "non-field trial" |
| // tests. |
| class WebRtcVideoChannelFlexfecRecvTest : public WebRtcVideoChannelTest { |
| public: |
| WebRtcVideoChannelFlexfecRecvTest() |
| : WebRtcVideoChannelTest("WebRTC-FlexFEC-03-Advertised/Enabled/") {} |
| }; |
| |
| TEST_F(WebRtcVideoChannelFlexfecRecvTest, |
| DefaultFlexfecCodecHasTransportCcAndRembFeedbackParam) { |
| EXPECT_TRUE(cricket::HasTransportCc(GetEngineCodec("flexfec-03"))); |
| EXPECT_TRUE(cricket::HasRemb(GetEngineCodec("flexfec-03"))); |
| } |
| |
| TEST_F(WebRtcVideoChannelFlexfecRecvTest, SetDefaultRecvCodecsWithoutSsrc) { |
| AddRecvStream(); |
| |
| const std::vector<FakeFlexfecReceiveStream*>& streams = |
| fake_call_->GetFlexfecReceiveStreams(); |
| EXPECT_TRUE(streams.empty()); |
| |
| const std::vector<FakeVideoReceiveStream*>& video_streams = |
| fake_call_->GetVideoReceiveStreams(); |
| ASSERT_EQ(1U, video_streams.size()); |
| const FakeVideoReceiveStream& video_stream = *video_streams.front(); |
| const webrtc::VideoReceiveStream::Config& video_config = |
| video_stream.GetConfig(); |
| EXPECT_FALSE(video_config.rtp.protected_by_flexfec); |
| EXPECT_EQ(video_config.rtp.packet_sink_, nullptr); |
| } |
| |
| TEST_F(WebRtcVideoChannelFlexfecRecvTest, SetDefaultRecvCodecsWithSsrc) { |
| AddRecvStream( |
| CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc)); |
| |
| const std::vector<FakeFlexfecReceiveStream*>& streams = |
| fake_call_->GetFlexfecReceiveStreams(); |
| ASSERT_EQ(1U, streams.size()); |
| const auto* stream = streams.front(); |
| const webrtc::FlexfecReceiveStream::Config& config = stream->GetConfig(); |
| EXPECT_EQ(GetEngineCodec("flexfec-03").id, config.payload_type); |
| EXPECT_EQ(kFlexfecSsrc, config.rtp.remote_ssrc); |
| ASSERT_EQ(1U, config.protected_media_ssrcs.size()); |
| EXPECT_EQ(kSsrcs1[0], config.protected_media_ssrcs[0]); |
| |
| const std::vector<FakeVideoReceiveStream*>& video_streams = |
| fake_call_->GetVideoReceiveStreams(); |
| ASSERT_EQ(1U, video_streams.size()); |
| const FakeVideoReceiveStream& video_stream = *video_streams.front(); |
| const webrtc::VideoReceiveStream::Config& video_config = |
| video_stream.GetConfig(); |
| EXPECT_TRUE(video_config.rtp.protected_by_flexfec); |
| EXPECT_NE(video_config.rtp.packet_sink_, nullptr); |
| } |
| |
| // Test changing the configuration after a video stream has been created and |
| // turn on flexfec. This will result in the video stream being recreated because |
| // the flexfec stream pointer is injected to the video stream at construction. |
| TEST_F(WebRtcVideoChannelFlexfecRecvTest, |
| EnablingFlexfecRecreatesVideoReceiveStream) { |
| cricket::VideoRecvParameters recv_parameters; |
| recv_parameters.codecs.push_back(GetEngineCodec("VP8")); |
| ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters)); |
| |
| AddRecvStream( |
| CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc)); |
| EXPECT_EQ(1, fake_call_->GetNumCreatedReceiveStreams()); |
| const std::vector<FakeVideoReceiveStream*>& video_streams = |
| fake_call_->GetVideoReceiveStreams(); |
| ASSERT_EQ(1U, video_streams.size()); |
| const FakeVideoReceiveStream* video_stream = video_streams.front(); |
| const webrtc::VideoReceiveStream::Config* video_config = |
| &video_stream->GetConfig(); |
| EXPECT_FALSE(video_config->rtp.protected_by_flexfec); |
| EXPECT_EQ(video_config->rtp.packet_sink_, nullptr); |
| |
| // Enable FlexFEC. |
| recv_parameters.codecs.push_back(GetEngineCodec("flexfec-03")); |
| ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters)); |
| |
| // Now the count of created streams will be 3 since the video stream was |
| // recreated and a flexfec stream was created. |
| EXPECT_EQ(3, fake_call_->GetNumCreatedReceiveStreams()) |
| << "Enabling FlexFEC should create FlexfecReceiveStream."; |
| |
| EXPECT_EQ(1U, fake_call_->GetVideoReceiveStreams().size()) |
| << "Enabling FlexFEC should not create VideoReceiveStream."; |
| EXPECT_EQ(1U, fake_call_->GetFlexfecReceiveStreams().size()) |
| << "Enabling FlexFEC should create a single FlexfecReceiveStream."; |
| video_stream = video_streams.front(); |
| video_config = &video_stream->GetConfig(); |
| EXPECT_TRUE(video_config->rtp.protected_by_flexfec); |
| EXPECT_NE(video_config->rtp.packet_sink_, nullptr); |
| } |
| |
| // Test changing the configuration after a video stream has been created with |
| // flexfec enabled and then turn off flexfec. This will result in the video |
| // stream being recreated because the flexfec stream pointer is injected to the |
| // video stream at construction and that config needs to be torn down. |
| TEST_F(WebRtcVideoChannelFlexfecRecvTest, |
| DisablingFlexfecRecreatesVideoReceiveStream) { |
| cricket::VideoRecvParameters recv_parameters; |
| recv_parameters.codecs.push_back(GetEngineCodec("VP8")); |
| recv_parameters.codecs.push_back(GetEngineCodec("flexfec-03")); |
| ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters)); |
| |
| AddRecvStream( |
| CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc)); |
| EXPECT_EQ(2, fake_call_->GetNumCreatedReceiveStreams()); |
| EXPECT_EQ(1U, fake_call_->GetFlexfecReceiveStreams().size()); |
| const std::vector<FakeVideoReceiveStream*>& video_streams = |
| fake_call_->GetVideoReceiveStreams(); |
| ASSERT_EQ(1U, video_streams.size()); |
| const FakeVideoReceiveStream* video_stream = video_streams.front(); |
| const webrtc::VideoReceiveStream::Config* video_config = |
| &video_stream->GetConfig(); |
| EXPECT_TRUE(video_config->rtp.protected_by_flexfec); |
| EXPECT_NE(video_config->rtp.packet_sink_, nullptr); |
| |
| // Disable FlexFEC. |
| recv_parameters.codecs.clear(); |
| recv_parameters.codecs.push_back(GetEngineCodec("VP8")); |
| ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters)); |
| // Now the count of created streams will be 3 since the video stream had to |
| // be recreated on account of the flexfec stream being deleted. |
| EXPECT_EQ(3, fake_call_->GetNumCreatedReceiveStreams()) |
| << "Disabling FlexFEC should not recreate VideoReceiveStream."; |
| EXPECT_EQ(1U, fake_call_->GetVideoReceiveStreams().size()) |
| << "Disabling FlexFEC should not destroy VideoReceiveStream."; |
| EXPECT_TRUE(fake_call_->GetFlexfecReceiveStreams().empty()) |
| << "Disabling FlexFEC should destroy FlexfecReceiveStream."; |
| video_stream = video_streams.front(); |
| video_config = &video_stream->GetConfig(); |
| EXPECT_FALSE(video_config->rtp.protected_by_flexfec); |
| EXPECT_EQ(video_config->rtp.packet_sink_, nullptr); |
| } |
| |
| TEST_F(WebRtcVideoChannelFlexfecRecvTest, DuplicateFlexfecCodecIsDropped) { |
| constexpr int kUnusedPayloadType1 = 127; |
| |
| cricket::VideoRecvParameters recv_parameters; |
| recv_parameters.codecs.push_back(GetEngineCodec("VP8")); |
| recv_parameters.codecs.push_back(GetEngineCodec("flexfec-03")); |
| cricket::VideoCodec duplicate = GetEngineCodec("flexfec-03"); |
| duplicate.id = kUnusedPayloadType1; |
| recv_parameters.codecs.push_back(duplicate); |
| ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters)); |
| |
| AddRecvStream( |
| CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc)); |
| |
| const std::vector<FakeFlexfecReceiveStream*>& streams = |
| fake_call_->GetFlexfecReceiveStreams(); |
| ASSERT_EQ(1U, streams.size()); |
| const auto* stream = streams.front(); |
| const webrtc::FlexfecReceiveStream::Config& config = stream->GetConfig(); |
| EXPECT_EQ(GetEngineCodec("flexfec-03").id, config.payload_type); |
| } |
| |
| // TODO(brandtr): When FlexFEC is no longer behind a field trial, merge all |
| // tests that use this test fixture into the corresponding "non-field trial" |
| // tests. |
| class WebRtcVideoChannelFlexfecSendRecvTest : public WebRtcVideoChannelTest { |
| public: |
| WebRtcVideoChannelFlexfecSendRecvTest() |
| : WebRtcVideoChannelTest( |
| "WebRTC-FlexFEC-03-Advertised/Enabled/WebRTC-FlexFEC-03/Enabled/") { |
| } |
| }; |
| |
| TEST_F(WebRtcVideoChannelFlexfecSendRecvTest, SetDefaultSendCodecsWithoutSsrc) { |
| FakeVideoSendStream* stream = AddSendStream(); |
| webrtc::VideoSendStream::Config config = stream->GetConfig().Copy(); |
| |
| EXPECT_EQ(GetEngineCodec("flexfec-03").id, config.rtp.flexfec.payload_type); |
| EXPECT_EQ(0U, config.rtp.flexfec.ssrc); |
| EXPECT_TRUE(config.rtp.flexfec.protected_media_ssrcs.empty()); |
| } |
| |
| TEST_F(WebRtcVideoChannelFlexfecSendRecvTest, SetDefaultSendCodecsWithSsrc) { |
| FakeVideoSendStream* stream = AddSendStream( |
| CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc)); |
| webrtc::VideoSendStream::Config config = stream->GetConfig().Copy(); |
| |
| EXPECT_EQ(GetEngineCodec("flexfec-03").id, config.rtp.flexfec.payload_type); |
| EXPECT_EQ(kFlexfecSsrc, config.rtp.flexfec.ssrc); |
| ASSERT_EQ(1U, config.rtp.flexfec.protected_media_ssrcs.size()); |
| EXPECT_EQ(kSsrcs1[0], config.rtp.flexfec.protected_media_ssrcs[0]); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithoutFec) { |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| ASSERT_TRUE(channel_->SetSendParameters(parameters)); |
| |
| FakeVideoSendStream* stream = AddSendStream(); |
| webrtc::VideoSendStream::Config config = stream->GetConfig().Copy(); |
| |
| EXPECT_EQ(-1, config.rtp.ulpfec.ulpfec_payload_type); |
| EXPECT_EQ(-1, config.rtp.ulpfec.red_payload_type); |
| } |
| |
| TEST_F(WebRtcVideoChannelFlexfecSendRecvTest, SetSendCodecsWithoutFec) { |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| ASSERT_TRUE(channel_->SetSendParameters(parameters)); |
| |
| FakeVideoSendStream* stream = AddSendStream(); |
| webrtc::VideoSendStream::Config config = stream->GetConfig().Copy(); |
| |
| EXPECT_EQ(-1, config.rtp.flexfec.payload_type); |
| } |
| |
| TEST_F(WebRtcVideoChannelFlexfecRecvTest, SetRecvCodecsWithFec) { |
| AddRecvStream( |
| CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc)); |
| |
| cricket::VideoRecvParameters recv_parameters; |
| recv_parameters.codecs.push_back(GetEngineCodec("VP8")); |
| recv_parameters.codecs.push_back(GetEngineCodec("flexfec-03")); |
| ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters)); |
| |
| const std::vector<FakeFlexfecReceiveStream*>& flexfec_streams = |
| fake_call_->GetFlexfecReceiveStreams(); |
| ASSERT_EQ(1U, flexfec_streams.size()); |
| const FakeFlexfecReceiveStream* flexfec_stream = flexfec_streams.front(); |
| const webrtc::FlexfecReceiveStream::Config& flexfec_stream_config = |
| flexfec_stream->GetConfig(); |
| EXPECT_EQ(GetEngineCodec("flexfec-03").id, |
| flexfec_stream_config.payload_type); |
| EXPECT_EQ(kFlexfecSsrc, flexfec_stream_config.rtp.remote_ssrc); |
| ASSERT_EQ(1U, flexfec_stream_config.protected_media_ssrcs.size()); |
| EXPECT_EQ(kSsrcs1[0], flexfec_stream_config.protected_media_ssrcs[0]); |
| const std::vector<FakeVideoReceiveStream*>& video_streams = |
| fake_call_->GetVideoReceiveStreams(); |
| const FakeVideoReceiveStream* video_stream = video_streams.front(); |
| const webrtc::VideoReceiveStream::Config& video_stream_config = |
| video_stream->GetConfig(); |
| EXPECT_EQ(video_stream_config.rtp.local_ssrc, |
| flexfec_stream_config.rtp.local_ssrc); |
| EXPECT_EQ(video_stream_config.rtp.rtcp_mode, flexfec_stream_config.rtcp_mode); |
| EXPECT_EQ(video_stream_config.rtcp_send_transport, |
| flexfec_stream_config.rtcp_send_transport); |
| // TODO(brandtr): Update this EXPECT when we set `transport_cc` in a |
| // spec-compliant way. |
| EXPECT_EQ(video_stream_config.rtp.transport_cc, |
| flexfec_stream_config.rtp.transport_cc); |
| EXPECT_EQ(video_stream_config.rtp.rtcp_mode, flexfec_stream_config.rtcp_mode); |
| EXPECT_EQ(video_stream_config.rtp.extensions, |
| flexfec_stream_config.rtp.extensions); |
| } |
| |
| // We should not send FlexFEC, even if we advertise it, unless the right |
| // field trial is set. |
| // TODO(brandtr): Remove when FlexFEC is enabled by default. |
| TEST_F(WebRtcVideoChannelFlexfecRecvTest, |
| SetSendCodecsWithoutSsrcWithFecDoesNotEnableFec) { |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| parameters.codecs.push_back(GetEngineCodec("flexfec-03")); |
| ASSERT_TRUE(channel_->SetSendParameters(parameters)); |
| |
| FakeVideoSendStream* stream = AddSendStream(); |
| webrtc::VideoSendStream::Config config = stream->GetConfig().Copy(); |
| |
| EXPECT_EQ(-1, config.rtp.flexfec.payload_type); |
| EXPECT_EQ(0u, config.rtp.flexfec.ssrc); |
| EXPECT_TRUE(config.rtp.flexfec.protected_media_ssrcs.empty()); |
| } |
| |
| TEST_F(WebRtcVideoChannelFlexfecRecvTest, |
| SetSendCodecsWithSsrcWithFecDoesNotEnableFec) { |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| parameters.codecs.push_back(GetEngineCodec("flexfec-03")); |
| ASSERT_TRUE(channel_->SetSendParameters(parameters)); |
| |
| FakeVideoSendStream* stream = AddSendStream( |
| CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc)); |
| webrtc::VideoSendStream::Config config = stream->GetConfig().Copy(); |
| |
| EXPECT_EQ(-1, config.rtp.flexfec.payload_type); |
| EXPECT_EQ(0u, config.rtp.flexfec.ssrc); |
| EXPECT_TRUE(config.rtp.flexfec.protected_media_ssrcs.empty()); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| SetSendCodecRejectsRtxWithoutAssociatedPayloadType) { |
| const int kUnusedPayloadType = 127; |
| EXPECT_FALSE(FindCodecById(engine_.send_codecs(), kUnusedPayloadType)); |
| |
| cricket::VideoSendParameters parameters; |
| cricket::VideoCodec rtx_codec(kUnusedPayloadType, "rtx"); |
| parameters.codecs.push_back(rtx_codec); |
| EXPECT_FALSE(channel_->SetSendParameters(parameters)) |
| << "RTX codec without associated payload type should be rejected."; |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| SetSendCodecRejectsRtxWithoutMatchingVideoCodec) { |
| const int kUnusedPayloadType1 = 126; |
| const int kUnusedPayloadType2 = 127; |
| EXPECT_FALSE(FindCodecById(engine_.send_codecs(), kUnusedPayloadType1)); |
| EXPECT_FALSE(FindCodecById(engine_.send_codecs(), kUnusedPayloadType2)); |
| { |
| cricket::VideoCodec rtx_codec = cricket::VideoCodec::CreateRtxCodec( |
| kUnusedPayloadType1, GetEngineCodec("VP8").id); |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| parameters.codecs.push_back(rtx_codec); |
| ASSERT_TRUE(channel_->SetSendParameters(parameters)); |
| } |
| { |
| cricket::VideoCodec rtx_codec = cricket::VideoCodec::CreateRtxCodec( |
| kUnusedPayloadType1, kUnusedPayloadType2); |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| parameters.codecs.push_back(rtx_codec); |
| EXPECT_FALSE(channel_->SetSendParameters(parameters)) |
| << "RTX without matching video codec should be rejected."; |
| } |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithChangedRtxPayloadType) { |
| const int kUnusedPayloadType1 = 126; |
| const int kUnusedPayloadType2 = 127; |
| EXPECT_FALSE(FindCodecById(engine_.send_codecs(), kUnusedPayloadType1)); |
| EXPECT_FALSE(FindCodecById(engine_.send_codecs(), kUnusedPayloadType2)); |
| |
| // SSRCs for RTX. |
| cricket::StreamParams params = |
| cricket::StreamParams::CreateLegacy(kSsrcs1[0]); |
| params.AddFidSsrc(kSsrcs1[0], kRtxSsrcs1[0]); |
| AddSendStream(params); |
| |
| // Original payload type for RTX. |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| cricket::VideoCodec rtx_codec(kUnusedPayloadType1, "rtx"); |
| rtx_codec.SetParam("apt", GetEngineCodec("VP8").id); |
| parameters.codecs.push_back(rtx_codec); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| ASSERT_EQ(1U, fake_call_->GetVideoSendStreams().size()); |
| const webrtc::VideoSendStream::Config& config_before = |
| fake_call_->GetVideoSendStreams()[0]->GetConfig(); |
| EXPECT_EQ(kUnusedPayloadType1, config_before.rtp.rtx.payload_type); |
| ASSERT_EQ(1U, config_before.rtp.rtx.ssrcs.size()); |
| EXPECT_EQ(kRtxSsrcs1[0], config_before.rtp.rtx.ssrcs[0]); |
| |
| // Change payload type for RTX. |
| parameters.codecs[1].id = kUnusedPayloadType2; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| ASSERT_EQ(1U, fake_call_->GetVideoSendStreams().size()); |
| const webrtc::VideoSendStream::Config& config_after = |
| fake_call_->GetVideoSendStreams()[0]->GetConfig(); |
| EXPECT_EQ(kUnusedPayloadType2, config_after.rtp.rtx.payload_type); |
| ASSERT_EQ(1U, config_after.rtp.rtx.ssrcs.size()); |
| EXPECT_EQ(kRtxSsrcs1[0], config_after.rtp.rtx.ssrcs[0]); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithoutFecDisablesFec) { |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| parameters.codecs.push_back(GetEngineCodec("ulpfec")); |
| ASSERT_TRUE(channel_->SetSendParameters(parameters)); |
| |
| FakeVideoSendStream* stream = AddSendStream(); |
| webrtc::VideoSendStream::Config config = stream->GetConfig().Copy(); |
| |
| EXPECT_EQ(GetEngineCodec("ulpfec").id, config.rtp.ulpfec.ulpfec_payload_type); |
| |
| parameters.codecs.pop_back(); |
| ASSERT_TRUE(channel_->SetSendParameters(parameters)); |
| stream = fake_call_->GetVideoSendStreams()[0]; |
| ASSERT_TRUE(stream != nullptr); |
| config = stream->GetConfig().Copy(); |
| EXPECT_EQ(-1, config.rtp.ulpfec.ulpfec_payload_type) |
| << "SetSendCodec without ULPFEC should disable current ULPFEC."; |
| } |
| |
| TEST_F(WebRtcVideoChannelFlexfecSendRecvTest, |
| SetSendCodecsWithoutFecDisablesFec) { |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| parameters.codecs.push_back(GetEngineCodec("flexfec-03")); |
| ASSERT_TRUE(channel_->SetSendParameters(parameters)); |
| |
| FakeVideoSendStream* stream = AddSendStream( |
| CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc)); |
| webrtc::VideoSendStream::Config config = stream->GetConfig().Copy(); |
| |
| EXPECT_EQ(GetEngineCodec("flexfec-03").id, config.rtp.flexfec.payload_type); |
| EXPECT_EQ(kFlexfecSsrc, config.rtp.flexfec.ssrc); |
| ASSERT_EQ(1U, config.rtp.flexfec.protected_media_ssrcs.size()); |
| EXPECT_EQ(kSsrcs1[0], config.rtp.flexfec.protected_media_ssrcs[0]); |
| |
| parameters.codecs.pop_back(); |
| ASSERT_TRUE(channel_->SetSendParameters(parameters)); |
| stream = fake_call_->GetVideoSendStreams()[0]; |
| ASSERT_TRUE(stream != nullptr); |
| config = stream->GetConfig().Copy(); |
| EXPECT_EQ(-1, config.rtp.flexfec.payload_type) |
| << "SetSendCodec without FlexFEC should disable current FlexFEC."; |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetSendCodecsChangesExistingStreams) { |
| cricket::VideoSendParameters parameters; |
| cricket::VideoCodec codec(100, "VP8"); |
| codec.SetParam(kCodecParamMaxQuantization, kDefaultQpMax); |
| parameters.codecs.push_back(codec); |
| |
| ASSERT_TRUE(channel_->SetSendParameters(parameters)); |
| channel_->SetSend(true); |
| |
| FakeVideoSendStream* stream = AddSendStream(); |
| webrtc::test::FrameForwarder frame_forwarder; |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder)); |
| |
| std::vector<webrtc::VideoStream> streams = stream->GetVideoStreams(); |
| EXPECT_EQ(kDefaultQpMax, streams[0].max_qp); |
| |
| parameters.codecs.clear(); |
| codec.SetParam(kCodecParamMaxQuantization, kDefaultQpMax + 1); |
| parameters.codecs.push_back(codec); |
| ASSERT_TRUE(channel_->SetSendParameters(parameters)); |
| streams = fake_call_->GetVideoSendStreams()[0]->GetVideoStreams(); |
| EXPECT_EQ(kDefaultQpMax + 1, streams[0].max_qp); |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithBitrates) { |
| SetSendCodecsShouldWorkForBitrates("100", 100000, "150", 150000, "200", |
| 200000); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithHighMaxBitrate) { |
| SetSendCodecsShouldWorkForBitrates("", 0, "", -1, "10000", 10000000); |
| std::vector<webrtc::VideoStream> streams = AddSendStream()->GetVideoStreams(); |
| ASSERT_EQ(1u, streams.size()); |
| EXPECT_EQ(10000000, streams[0].max_bitrate_bps); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| SetSendCodecsWithoutBitratesUsesCorrectDefaults) { |
| SetSendCodecsShouldWorkForBitrates("", 0, "", -1, "", -1); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetSendCodecsCapsMinAndStartBitrate) { |
| SetSendCodecsShouldWorkForBitrates("-1", 0, "-100", -1, "", -1); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetSendCodecsRejectsMaxLessThanMinBitrate) { |
| send_parameters_.codecs[0].params[kCodecParamMinBitrate] = "300"; |
| send_parameters_.codecs[0].params[kCodecParamMaxBitrate] = "200"; |
| EXPECT_FALSE(channel_->SetSendParameters(send_parameters_)); |
| } |
| |
| // Test that when both the codec-specific bitrate params and max_bandwidth_bps |
| // are present in the same send parameters, the settings are combined correctly. |
| TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithBitratesAndMaxSendBandwidth) { |
| send_parameters_.codecs[0].params[kCodecParamMinBitrate] = "100"; |
| send_parameters_.codecs[0].params[kCodecParamStartBitrate] = "200"; |
| send_parameters_.codecs[0].params[kCodecParamMaxBitrate] = "300"; |
| send_parameters_.max_bandwidth_bps = 400000; |
| // We expect max_bandwidth_bps to take priority, if set. |
| ExpectSetBitrateParameters(100000, 200000, 400000); |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| // Since the codec isn't changing, start_bitrate_bps should be -1. |
| ExpectSetBitrateParameters(100000, -1, 350000); |
| |
| // Decrease max_bandwidth_bps. |
| send_parameters_.max_bandwidth_bps = 350000; |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| |
| // Now try again with the values flipped around. |
| send_parameters_.codecs[0].params[kCodecParamMaxBitrate] = "400"; |
| send_parameters_.max_bandwidth_bps = 300000; |
| ExpectSetBitrateParameters(100000, 200000, 300000); |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| |
| // If we change the codec max, max_bandwidth_bps should still apply. |
| send_parameters_.codecs[0].params[kCodecParamMaxBitrate] = "350"; |
| ExpectSetBitrateParameters(100000, 200000, 300000); |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetMaxSendBandwidthShouldPreserveOtherBitrates) { |
| SetSendCodecsShouldWorkForBitrates("100", 100000, "150", 150000, "200", |
| 200000); |
| send_parameters_.max_bandwidth_bps = 300000; |
| // Setting max bitrate should keep previous min bitrate. |
| // Setting max bitrate should not reset start bitrate. |
| ExpectSetBitrateParameters(100000, -1, 300000); |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetMaxSendBandwidthShouldBeRemovable) { |
| send_parameters_.max_bandwidth_bps = 300000; |
| ExpectSetMaxBitrate(300000); |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| // -1 means to disable max bitrate (set infinite). |
| send_parameters_.max_bandwidth_bps = -1; |
| ExpectSetMaxBitrate(-1); |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetMaxSendBandwidthAndAddSendStream) { |
| send_parameters_.max_bandwidth_bps = 99999; |
| FakeVideoSendStream* stream = AddSendStream(); |
| ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps); |
| ASSERT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| ASSERT_EQ(1u, stream->GetVideoStreams().size()); |
| EXPECT_EQ(send_parameters_.max_bandwidth_bps, |
| stream->GetVideoStreams()[0].max_bitrate_bps); |
| |
| send_parameters_.max_bandwidth_bps = 77777; |
| ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps); |
| ASSERT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| EXPECT_EQ(send_parameters_.max_bandwidth_bps, |
| stream->GetVideoStreams()[0].max_bitrate_bps); |
| } |
| |
| // Tests that when the codec specific max bitrate and VideoSendParameters |
| // max_bandwidth_bps are used, that it sets the VideoStream's max bitrate |
| // appropriately. |
| TEST_F(WebRtcVideoChannelTest, |
| MaxBitratePrioritizesVideoSendParametersOverCodecMaxBitrate) { |
| send_parameters_.codecs[0].params[kCodecParamMinBitrate] = "100"; |
| send_parameters_.codecs[0].params[kCodecParamStartBitrate] = "200"; |
| send_parameters_.codecs[0].params[kCodecParamMaxBitrate] = "300"; |
| send_parameters_.max_bandwidth_bps = -1; |
| AddSendStream(); |
| ExpectSetMaxBitrate(300000); |
| ASSERT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| |
| std::vector<FakeVideoSendStream*> video_send_streams = GetFakeSendStreams(); |
| ASSERT_EQ(1u, video_send_streams.size()); |
| FakeVideoSendStream* video_send_stream = video_send_streams[0]; |
| ASSERT_EQ(1u, video_send_streams[0]->GetVideoStreams().size()); |
| // First the max bitrate is set based upon the codec param. |
| EXPECT_EQ(300000, |
| video_send_streams[0]->GetVideoStreams()[0].max_bitrate_bps); |
| |
| // The VideoSendParameters max bitrate overrides the codec's. |
| send_parameters_.max_bandwidth_bps = 500000; |
| ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps); |
| ASSERT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| ASSERT_EQ(1u, video_send_stream->GetVideoStreams().size()); |
| EXPECT_EQ(500000, video_send_stream->GetVideoStreams()[0].max_bitrate_bps); |
| } |
| |
| // Tests that when the codec specific max bitrate and RtpParameters |
| // max_bitrate_bps are used, that it sets the VideoStream's max bitrate |
| // appropriately. |
| TEST_F(WebRtcVideoChannelTest, |
| MaxBitratePrioritizesRtpParametersOverCodecMaxBitrate) { |
| send_parameters_.codecs[0].params[kCodecParamMinBitrate] = "100"; |
| send_parameters_.codecs[0].params[kCodecParamStartBitrate] = "200"; |
| send_parameters_.codecs[0].params[kCodecParamMaxBitrate] = "300"; |
| send_parameters_.max_bandwidth_bps = -1; |
| AddSendStream(); |
| ExpectSetMaxBitrate(300000); |
| ASSERT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| |
| std::vector<FakeVideoSendStream*> video_send_streams = GetFakeSendStreams(); |
| ASSERT_EQ(1u, video_send_streams.size()); |
| FakeVideoSendStream* video_send_stream = video_send_streams[0]; |
| ASSERT_EQ(1u, video_send_stream->GetVideoStreams().size()); |
| // First the max bitrate is set based upon the codec param. |
| EXPECT_EQ(300000, video_send_stream->GetVideoStreams()[0].max_bitrate_bps); |
| |
| // The RtpParameter max bitrate overrides the codec's. |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| ASSERT_EQ(1u, parameters.encodings.size()); |
| parameters.encodings[0].max_bitrate_bps = 500000; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| ASSERT_EQ(1u, video_send_stream->GetVideoStreams().size()); |
| EXPECT_EQ(parameters.encodings[0].max_bitrate_bps, |
| video_send_stream->GetVideoStreams()[0].max_bitrate_bps); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| MaxBitrateIsMinimumOfMaxSendBandwidthAndMaxEncodingBitrate) { |
| send_parameters_.max_bandwidth_bps = 99999; |
| FakeVideoSendStream* stream = AddSendStream(); |
| ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps); |
| ASSERT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| ASSERT_EQ(1u, stream->GetVideoStreams().size()); |
| EXPECT_EQ(send_parameters_.max_bandwidth_bps, |
| stream->GetVideoStreams()[0].max_bitrate_bps); |
| |
| // Get and set the rtp encoding parameters. |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_EQ(1u, parameters.encodings.size()); |
| |
| parameters.encodings[0].max_bitrate_bps = 99999 - 1; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| EXPECT_EQ(parameters.encodings[0].max_bitrate_bps, |
| stream->GetVideoStreams()[0].max_bitrate_bps); |
| |
| parameters.encodings[0].max_bitrate_bps = 99999 + 1; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| EXPECT_EQ(send_parameters_.max_bandwidth_bps, |
| stream->GetVideoStreams()[0].max_bitrate_bps); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetMaxSendBitrateCanIncreaseSenderBitrate) { |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| ASSERT_TRUE(channel_->SetSendParameters(parameters)); |
| channel_->SetSend(true); |
| |
| FakeVideoSendStream* stream = AddSendStream(); |
| |
| webrtc::test::FrameForwarder frame_forwarder; |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder)); |
| |
| std::vector<webrtc::VideoStream> streams = stream->GetVideoStreams(); |
| int initial_max_bitrate_bps = streams[0].max_bitrate_bps; |
| EXPECT_GT(initial_max_bitrate_bps, 0); |
| |
| parameters.max_bandwidth_bps = initial_max_bitrate_bps * 2; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| // Insert a frame to update the encoder config. |
| frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame()); |
| streams = stream->GetVideoStreams(); |
| EXPECT_EQ(initial_max_bitrate_bps * 2, streams[0].max_bitrate_bps); |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| SetMaxSendBitrateCanIncreaseSimulcastSenderBitrate) { |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| ASSERT_TRUE(channel_->SetSendParameters(parameters)); |
| channel_->SetSend(true); |
| |
| FakeVideoSendStream* stream = AddSendStream( |
| cricket::CreateSimStreamParams("cname", MAKE_VECTOR(kSsrcs3))); |
| |
| // Send a frame to make sure this scales up to >1 stream (simulcast). |
| webrtc::test::FrameForwarder frame_forwarder; |
| EXPECT_TRUE(channel_->SetVideoSend(kSsrcs3[0], nullptr, &frame_forwarder)); |
| frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame()); |
| |
| std::vector<webrtc::VideoStream> streams = stream->GetVideoStreams(); |
| ASSERT_GT(streams.size(), 1u) |
| << "Without simulcast this test doesn't make sense."; |
| int initial_max_bitrate_bps = GetTotalMaxBitrate(streams).bps(); |
| EXPECT_GT(initial_max_bitrate_bps, 0); |
| |
| parameters.max_bandwidth_bps = initial_max_bitrate_bps * 2; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| // Insert a frame to update the encoder config. |
| frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame()); |
| streams = stream->GetVideoStreams(); |
| int increased_max_bitrate_bps = GetTotalMaxBitrate(streams).bps(); |
| EXPECT_EQ(initial_max_bitrate_bps * 2, increased_max_bitrate_bps); |
| |
| EXPECT_TRUE(channel_->SetVideoSend(kSsrcs3[0], nullptr, nullptr)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithMaxQuantization) { |
| static const char* kMaxQuantization = "21"; |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| parameters.codecs[0].params[kCodecParamMaxQuantization] = kMaxQuantization; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| EXPECT_EQ(atoi(kMaxQuantization), |
| AddSendStream()->GetVideoStreams().back().max_qp); |
| |
| VideoCodec codec; |
| EXPECT_TRUE(channel_->GetSendCodec(&codec)); |
| EXPECT_EQ(kMaxQuantization, codec.params[kCodecParamMaxQuantization]); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetSendCodecsRejectBadPayloadTypes) { |
| // TODO(pbos): Should we only allow the dynamic range? |
| static const int kIncorrectPayloads[] = {-2, -1, 128, 129}; |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| for (size_t i = 0; i < arraysize(kIncorrectPayloads); ++i) { |
| parameters.codecs[0].id = kIncorrectPayloads[i]; |
| EXPECT_FALSE(channel_->SetSendParameters(parameters)) |
| << "Bad payload type '" << kIncorrectPayloads[i] << "' accepted."; |
| } |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetSendCodecsAcceptAllValidPayloadTypes) { |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| for (int payload_type = 96; payload_type <= 127; ++payload_type) { |
| parameters.codecs[0].id = payload_type; |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)) |
| << "Payload type '" << payload_type << "' rejected."; |
| } |
| } |
| |
| // Test that setting the a different set of codecs but with an identical front |
| // codec doesn't result in the stream being recreated. |
| // This may happen when a subsequent negotiation includes fewer codecs, as a |
| // result of one of the codecs being rejected. |
| TEST_F(WebRtcVideoChannelTest, |
| SetSendCodecsIdenticalFirstCodecDoesntRecreateStream) { |
| cricket::VideoSendParameters parameters1; |
| parameters1.codecs.push_back(GetEngineCodec("VP8")); |
| parameters1.codecs.push_back(GetEngineCodec("VP9")); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters1)); |
| |
| AddSendStream(); |
| EXPECT_EQ(1, fake_call_->GetNumCreatedSendStreams()); |
| |
| cricket::VideoSendParameters parameters2; |
| parameters2.codecs.push_back(GetEngineCodec("VP8")); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters2)); |
| EXPECT_EQ(1, fake_call_->GetNumCreatedSendStreams()); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetRecvCodecsWithOnlyVp8) { |
| cricket::VideoRecvParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| } |
| |
| // Test that we set our inbound RTX codecs properly. |
| TEST_F(WebRtcVideoChannelTest, SetRecvCodecsWithRtx) { |
| const int kUnusedPayloadType1 = 126; |
| const int kUnusedPayloadType2 = 127; |
| EXPECT_FALSE(FindCodecById(engine_.recv_codecs(), kUnusedPayloadType1)); |
| EXPECT_FALSE(FindCodecById(engine_.recv_codecs(), kUnusedPayloadType2)); |
| |
| cricket::VideoRecvParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| cricket::VideoCodec rtx_codec(kUnusedPayloadType1, "rtx"); |
| parameters.codecs.push_back(rtx_codec); |
| EXPECT_FALSE(channel_->SetRecvParameters(parameters)) |
| << "RTX codec without associated payload should be rejected."; |
| |
| parameters.codecs[1].SetParam("apt", kUnusedPayloadType2); |
| EXPECT_FALSE(channel_->SetRecvParameters(parameters)) |
| << "RTX codec with invalid associated payload type should be rejected."; |
| |
| parameters.codecs[1].SetParam("apt", GetEngineCodec("VP8").id); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| |
| cricket::VideoCodec rtx_codec2(kUnusedPayloadType2, "rtx"); |
| rtx_codec2.SetParam("apt", rtx_codec.id); |
| parameters.codecs.push_back(rtx_codec2); |
| |
| EXPECT_FALSE(channel_->SetRecvParameters(parameters)) |
| << "RTX codec with another RTX as associated payload type should be " |
| "rejected."; |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetRecvCodecsWithPacketization) { |
| cricket::VideoCodec vp8_codec = GetEngineCodec("VP8"); |
| vp8_codec.packetization = kPacketizationParamRaw; |
| |
| cricket::VideoRecvParameters parameters; |
| parameters.codecs = {vp8_codec, GetEngineCodec("VP9")}; |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| |
| const cricket::StreamParams params = |
| cricket::StreamParams::CreateLegacy(kSsrcs1[0]); |
| AddRecvStream(params); |
| ASSERT_THAT(fake_call_->GetVideoReceiveStreams(), testing::SizeIs(1)); |
| |
| const webrtc::VideoReceiveStream::Config& config = |
| fake_call_->GetVideoReceiveStreams()[0]->GetConfig(); |
| ASSERT_THAT(config.rtp.raw_payload_types, testing::SizeIs(1)); |
| EXPECT_EQ(config.rtp.raw_payload_types.count(vp8_codec.id), 1U); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetRecvCodecsWithPacketizationRecreatesStream) { |
| cricket::VideoRecvParameters parameters; |
| parameters.codecs = {GetEngineCodec("VP8"), GetEngineCodec("VP9")}; |
| parameters.codecs.back().packetization = kPacketizationParamRaw; |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| |
| const cricket::StreamParams params = |
| cricket::StreamParams::CreateLegacy(kSsrcs1[0]); |
| AddRecvStream(params); |
| ASSERT_THAT(fake_call_->GetVideoReceiveStreams(), testing::SizeIs(1)); |
| EXPECT_EQ(fake_call_->GetNumCreatedReceiveStreams(), 1); |
| |
| parameters.codecs.back().packetization.reset(); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| EXPECT_EQ(fake_call_->GetNumCreatedReceiveStreams(), 2); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, DuplicateUlpfecCodecIsDropped) { |
| constexpr int kFirstUlpfecPayloadType = 126; |
| constexpr int kSecondUlpfecPayloadType = 127; |
| |
| cricket::VideoRecvParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| parameters.codecs.push_back( |
| cricket::VideoCodec(kFirstUlpfecPayloadType, cricket::kUlpfecCodecName)); |
| parameters.codecs.push_back( |
| cricket::VideoCodec(kSecondUlpfecPayloadType, cricket::kUlpfecCodecName)); |
| ASSERT_TRUE(channel_->SetRecvParameters(parameters)); |
| |
| FakeVideoReceiveStream* recv_stream = AddRecvStream(); |
| EXPECT_EQ(kFirstUlpfecPayloadType, |
| recv_stream->GetConfig().rtp.ulpfec_payload_type); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, DuplicateRedCodecIsDropped) { |
| constexpr int kFirstRedPayloadType = 126; |
| constexpr int kSecondRedPayloadType = 127; |
| |
| cricket::VideoRecvParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| parameters.codecs.push_back( |
| cricket::VideoCodec(kFirstRedPayloadType, cricket::kRedCodecName)); |
| parameters.codecs.push_back( |
| cricket::VideoCodec(kSecondRedPayloadType, cricket::kRedCodecName)); |
| ASSERT_TRUE(channel_->SetRecvParameters(parameters)); |
| |
| FakeVideoReceiveStream* recv_stream = AddRecvStream(); |
| EXPECT_EQ(kFirstRedPayloadType, |
| recv_stream->GetConfig().rtp.red_payload_type); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetRecvCodecsWithChangedRtxPayloadType) { |
| const int kUnusedPayloadType1 = 126; |
| const int kUnusedPayloadType2 = 127; |
| EXPECT_FALSE(FindCodecById(engine_.recv_codecs(), kUnusedPayloadType1)); |
| EXPECT_FALSE(FindCodecById(engine_.recv_codecs(), kUnusedPayloadType2)); |
| |
| // SSRCs for RTX. |
| cricket::StreamParams params = |
| cricket::StreamParams::CreateLegacy(kSsrcs1[0]); |
| params.AddFidSsrc(kSsrcs1[0], kRtxSsrcs1[0]); |
| AddRecvStream(params); |
| |
| // Original payload type for RTX. |
| cricket::VideoRecvParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| cricket::VideoCodec rtx_codec(kUnusedPayloadType1, "rtx"); |
| rtx_codec.SetParam("apt", GetEngineCodec("VP8").id); |
| parameters.codecs.push_back(rtx_codec); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| ASSERT_EQ(1U, fake_call_->GetVideoReceiveStreams().size()); |
| const webrtc::VideoReceiveStream::Config& config_before = |
| fake_call_->GetVideoReceiveStreams()[0]->GetConfig(); |
| EXPECT_EQ(1U, config_before.rtp.rtx_associated_payload_types.size()); |
| const int* payload_type_before = FindKeyByValue( |
| config_before.rtp.rtx_associated_payload_types, GetEngineCodec("VP8").id); |
| ASSERT_NE(payload_type_before, nullptr); |
| EXPECT_EQ(kUnusedPayloadType1, *payload_type_before); |
| EXPECT_EQ(kRtxSsrcs1[0], config_before.rtp.rtx_ssrc); |
| |
| // Change payload type for RTX. |
| parameters.codecs[1].id = kUnusedPayloadType2; |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| ASSERT_EQ(1U, fake_call_->GetVideoReceiveStreams().size()); |
| const webrtc::VideoReceiveStream::Config& config_after = |
| fake_call_->GetVideoReceiveStreams()[0]->GetConfig(); |
| EXPECT_EQ(1U, config_after.rtp.rtx_associated_payload_types.size()); |
| const int* payload_type_after = FindKeyByValue( |
| config_after.rtp.rtx_associated_payload_types, GetEngineCodec("VP8").id); |
| ASSERT_NE(payload_type_after, nullptr); |
| EXPECT_EQ(kUnusedPayloadType2, *payload_type_after); |
| EXPECT_EQ(kRtxSsrcs1[0], config_after.rtp.rtx_ssrc); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetRecvCodecsRtxWithRtxTime) { |
| const int kUnusedPayloadType1 = 126; |
| const int kUnusedPayloadType2 = 127; |
| EXPECT_FALSE(FindCodecById(engine_.recv_codecs(), kUnusedPayloadType1)); |
| EXPECT_FALSE(FindCodecById(engine_.recv_codecs(), kUnusedPayloadType2)); |
| |
| // SSRCs for RTX. |
| cricket::StreamParams params = |
| cricket::StreamParams::CreateLegacy(kSsrcs1[0]); |
| params.AddFidSsrc(kSsrcs1[0], kRtxSsrcs1[0]); |
| AddRecvStream(params); |
| |
| // Payload type for RTX. |
| cricket::VideoRecvParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| cricket::VideoCodec rtx_codec(kUnusedPayloadType1, "rtx"); |
| rtx_codec.SetParam("apt", GetEngineCodec("VP8").id); |
| parameters.codecs.push_back(rtx_codec); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| ASSERT_EQ(1U, fake_call_->GetVideoReceiveStreams().size()); |
| const webrtc::VideoReceiveStream::Config& config = |
| fake_call_->GetVideoReceiveStreams()[0]->GetConfig(); |
| |
| const int kRtxTime = 343; |
| // Assert that the default value is different from the ones we test |
| // and store the default value. |
| EXPECT_NE(config.rtp.nack.rtp_history_ms, kRtxTime); |
| int default_history_ms = config.rtp.nack.rtp_history_ms; |
| |
| // Set rtx-time. |
| parameters.codecs[1].SetParam(kCodecParamRtxTime, kRtxTime); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| EXPECT_EQ(fake_call_->GetVideoReceiveStreams()[0] |
| ->GetConfig() |
| .rtp.nack.rtp_history_ms, |
| kRtxTime); |
| |
| // Negative values are ignored so the default value applies. |
| parameters.codecs[1].SetParam(kCodecParamRtxTime, -1); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| EXPECT_NE(fake_call_->GetVideoReceiveStreams()[0] |
| ->GetConfig() |
| .rtp.nack.rtp_history_ms, |
| -1); |
| EXPECT_EQ(fake_call_->GetVideoReceiveStreams()[0] |
| ->GetConfig() |
| .rtp.nack.rtp_history_ms, |
| default_history_ms); |
| |
| // 0 is ignored so the default applies. |
| parameters.codecs[1].SetParam(kCodecParamRtxTime, 0); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| EXPECT_NE(fake_call_->GetVideoReceiveStreams()[0] |
| ->GetConfig() |
| .rtp.nack.rtp_history_ms, |
| 0); |
| EXPECT_EQ(fake_call_->GetVideoReceiveStreams()[0] |
| ->GetConfig() |
| .rtp.nack.rtp_history_ms, |
| default_history_ms); |
| |
| // Values larger than the default are clamped to the default. |
| parameters.codecs[1].SetParam(kCodecParamRtxTime, default_history_ms + 100); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| EXPECT_EQ(fake_call_->GetVideoReceiveStreams()[0] |
| ->GetConfig() |
| .rtp.nack.rtp_history_ms, |
| default_history_ms); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetRecvCodecsDifferentPayloadType) { |
| cricket::VideoRecvParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| parameters.codecs[0].id = 99; |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetRecvCodecsAcceptDefaultCodecs) { |
| cricket::VideoRecvParameters parameters; |
| parameters.codecs = engine_.recv_codecs(); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| |
| FakeVideoReceiveStream* stream = AddRecvStream(); |
| const webrtc::VideoReceiveStream::Config& config = stream->GetConfig(); |
| EXPECT_EQ(engine_.recv_codecs()[0].name, |
| config.decoders[0].video_format.name); |
| EXPECT_EQ(engine_.recv_codecs()[0].id, config.decoders[0].payload_type); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetRecvCodecsRejectUnsupportedCodec) { |
| cricket::VideoRecvParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| parameters.codecs.push_back(VideoCodec(101, "WTF3")); |
| EXPECT_FALSE(channel_->SetRecvParameters(parameters)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetRecvCodecsAcceptsMultipleVideoCodecs) { |
| cricket::VideoRecvParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| parameters.codecs.push_back(GetEngineCodec("VP9")); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetRecvCodecsWithoutFecDisablesFec) { |
| cricket::VideoSendParameters send_parameters; |
| send_parameters.codecs.push_back(GetEngineCodec("VP8")); |
| send_parameters.codecs.push_back(GetEngineCodec("red")); |
| send_parameters.codecs.push_back(GetEngineCodec("ulpfec")); |
| ASSERT_TRUE(channel_->SetSendParameters(send_parameters)); |
| |
| FakeVideoReceiveStream* stream = AddRecvStream(); |
| |
| EXPECT_EQ(GetEngineCodec("ulpfec").id, |
| stream->GetConfig().rtp.ulpfec_payload_type); |
| |
| cricket::VideoRecvParameters recv_parameters; |
| recv_parameters.codecs.push_back(GetEngineCodec("VP8")); |
| ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters)); |
| stream = fake_call_->GetVideoReceiveStreams()[0]; |
| ASSERT_TRUE(stream != nullptr); |
| EXPECT_EQ(-1, stream->GetConfig().rtp.ulpfec_payload_type) |
| << "SetSendCodec without ULPFEC should disable current ULPFEC."; |
| } |
| |
| TEST_F(WebRtcVideoChannelFlexfecRecvTest, SetRecvParamsWithoutFecDisablesFec) { |
| AddRecvStream( |
| CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc)); |
| const std::vector<FakeFlexfecReceiveStream*>& streams = |
| fake_call_->GetFlexfecReceiveStreams(); |
| |
| ASSERT_EQ(1U, streams.size()); |
| const FakeFlexfecReceiveStream* stream = streams.front(); |
| EXPECT_EQ(GetEngineCodec("flexfec-03").id, stream->GetConfig().payload_type); |
| EXPECT_EQ(kFlexfecSsrc, stream->rtp_config().remote_ssrc); |
| ASSERT_EQ(1U, stream->GetConfig().protected_media_ssrcs.size()); |
| EXPECT_EQ(kSsrcs1[0], stream->GetConfig().protected_media_ssrcs[0]); |
| |
| cricket::VideoRecvParameters recv_parameters; |
| recv_parameters.codecs.push_back(GetEngineCodec("VP8")); |
| ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters)); |
| EXPECT_TRUE(streams.empty()) |
| << "SetSendCodec without FlexFEC should disable current FlexFEC."; |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetSendParamsWithFecEnablesFec) { |
| FakeVideoReceiveStream* stream = AddRecvStream(); |
| EXPECT_EQ(GetEngineCodec("ulpfec").id, |
| stream->GetConfig().rtp.ulpfec_payload_type); |
| |
| cricket::VideoRecvParameters recv_parameters; |
| recv_parameters.codecs.push_back(GetEngineCodec("VP8")); |
| recv_parameters.codecs.push_back(GetEngineCodec("red")); |
| recv_parameters.codecs.push_back(GetEngineCodec("ulpfec")); |
| ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters)); |
| stream = fake_call_->GetVideoReceiveStreams()[0]; |
| ASSERT_TRUE(stream != nullptr); |
| EXPECT_EQ(GetEngineCodec("ulpfec").id, |
| stream->GetConfig().rtp.ulpfec_payload_type) |
| << "ULPFEC should be enabled on the receive stream."; |
| |
| cricket::VideoSendParameters send_parameters; |
| send_parameters.codecs.push_back(GetEngineCodec("VP8")); |
| send_parameters.codecs.push_back(GetEngineCodec("red")); |
| send_parameters.codecs.push_back(GetEngineCodec("ulpfec")); |
| ASSERT_TRUE(channel_->SetSendParameters(send_parameters)); |
| stream = fake_call_->GetVideoReceiveStreams()[0]; |
| EXPECT_EQ(GetEngineCodec("ulpfec").id, |
| stream->GetConfig().rtp.ulpfec_payload_type) |
| << "ULPFEC should be enabled on the receive stream."; |
| } |
| |
| TEST_F(WebRtcVideoChannelFlexfecSendRecvTest, |
| SetSendRecvParamsWithFecEnablesFec) { |
| AddRecvStream( |
| CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc)); |
| const std::vector<FakeFlexfecReceiveStream*>& streams = |
| fake_call_->GetFlexfecReceiveStreams(); |
| |
| cricket::VideoRecvParameters recv_parameters; |
| recv_parameters.codecs.push_back(GetEngineCodec("VP8")); |
| recv_parameters.codecs.push_back(GetEngineCodec("flexfec-03")); |
| ASSERT_TRUE(channel_->SetRecvParameters(recv_parameters)); |
| ASSERT_EQ(1U, streams.size()); |
| const FakeFlexfecReceiveStream* stream_with_recv_params = streams.front(); |
| EXPECT_EQ(GetEngineCodec("flexfec-03").id, |
| stream_with_recv_params->GetConfig().payload_type); |
| EXPECT_EQ(kFlexfecSsrc, stream_with_recv_params->GetConfig().rtp.remote_ssrc); |
| EXPECT_EQ(1U, |
| stream_with_recv_params->GetConfig().protected_media_ssrcs.size()); |
| EXPECT_EQ(kSsrcs1[0], |
| stream_with_recv_params->GetConfig().protected_media_ssrcs[0]); |
| |
| cricket::VideoSendParameters send_parameters; |
| send_parameters.codecs.push_back(GetEngineCodec("VP8")); |
| send_parameters.codecs.push_back(GetEngineCodec("flexfec-03")); |
| ASSERT_TRUE(channel_->SetSendParameters(send_parameters)); |
| ASSERT_EQ(1U, streams.size()); |
| const FakeFlexfecReceiveStream* stream_with_send_params = streams.front(); |
| EXPECT_EQ(GetEngineCodec("flexfec-03").id, |
| stream_with_send_params->GetConfig().payload_type); |
| EXPECT_EQ(kFlexfecSsrc, stream_with_send_params->GetConfig().rtp.remote_ssrc); |
| EXPECT_EQ(1U, |
| stream_with_send_params->GetConfig().protected_media_ssrcs.size()); |
| EXPECT_EQ(kSsrcs1[0], |
| stream_with_send_params->GetConfig().protected_media_ssrcs[0]); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetRecvCodecsRejectDuplicateFecPayloads) { |
| cricket::VideoRecvParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| parameters.codecs.push_back(GetEngineCodec("red")); |
| parameters.codecs[1].id = parameters.codecs[0].id; |
| EXPECT_FALSE(channel_->SetRecvParameters(parameters)); |
| } |
| |
| TEST_F(WebRtcVideoChannelFlexfecRecvTest, |
| SetRecvCodecsRejectDuplicateFecPayloads) { |
| cricket::VideoRecvParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| parameters.codecs.push_back(GetEngineCodec("flexfec-03")); |
| parameters.codecs[1].id = parameters.codecs[0].id; |
| EXPECT_FALSE(channel_->SetRecvParameters(parameters)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetRecvCodecsRejectDuplicateCodecPayloads) { |
| cricket::VideoRecvParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| parameters.codecs.push_back(GetEngineCodec("VP9")); |
| parameters.codecs[1].id = parameters.codecs[0].id; |
| EXPECT_FALSE(channel_->SetRecvParameters(parameters)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| SetRecvCodecsAcceptSameCodecOnMultiplePayloadTypes) { |
| cricket::VideoRecvParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| parameters.codecs[1].id += 1; |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| } |
| |
| // Test that setting the same codecs but with a different order |
| // doesn't result in the stream being recreated. |
| TEST_F(WebRtcVideoChannelTest, |
| SetRecvCodecsDifferentOrderDoesntRecreateStream) { |
| cricket::VideoRecvParameters parameters1; |
| parameters1.codecs.push_back(GetEngineCodec("VP8")); |
| parameters1.codecs.push_back(GetEngineCodec("red")); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters1)); |
| |
| AddRecvStream(cricket::StreamParams::CreateLegacy(123)); |
| EXPECT_EQ(1, fake_call_->GetNumCreatedReceiveStreams()); |
| |
| cricket::VideoRecvParameters parameters2; |
| parameters2.codecs.push_back(GetEngineCodec("red")); |
| parameters2.codecs.push_back(GetEngineCodec("VP8")); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters2)); |
| EXPECT_EQ(1, fake_call_->GetNumCreatedReceiveStreams()); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SendStreamNotSendingByDefault) { |
| EXPECT_FALSE(AddSendStream()->IsSending()); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, ReceiveStreamReceivingByDefault) { |
| EXPECT_TRUE(AddRecvStream()->IsReceiving()); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetSend) { |
| FakeVideoSendStream* stream = AddSendStream(); |
| EXPECT_FALSE(stream->IsSending()); |
| |
| // false->true |
| EXPECT_TRUE(channel_->SetSend(true)); |
| EXPECT_TRUE(stream->IsSending()); |
| // true->true |
| EXPECT_TRUE(channel_->SetSend(true)); |
| EXPECT_TRUE(stream->IsSending()); |
| // true->false |
| EXPECT_TRUE(channel_->SetSend(false)); |
| EXPECT_FALSE(stream->IsSending()); |
| // false->false |
| EXPECT_TRUE(channel_->SetSend(false)); |
| EXPECT_FALSE(stream->IsSending()); |
| |
| EXPECT_TRUE(channel_->SetSend(true)); |
| FakeVideoSendStream* new_stream = AddSendStream(); |
| EXPECT_TRUE(new_stream->IsSending()) |
| << "Send stream created after SetSend(true) not sending initially."; |
| } |
| |
| // This test verifies DSCP settings are properly applied on video media channel. |
| TEST_F(WebRtcVideoChannelTest, TestSetDscpOptions) { |
| std::unique_ptr<cricket::FakeNetworkInterface> network_interface( |
| new cricket::FakeNetworkInterface); |
| MediaConfig config; |
| std::unique_ptr<cricket::WebRtcVideoChannel> channel; |
| webrtc::RtpParameters parameters; |
| |
| channel.reset( |
| static_cast<cricket::WebRtcVideoChannel*>(engine_.CreateMediaChannel( |
| call_.get(), config, VideoOptions(), webrtc::CryptoOptions(), |
| video_bitrate_allocator_factory_.get()))); |
| channel->SetInterface(network_interface.get()); |
| // Default value when DSCP is disabled should be DSCP_DEFAULT. |
| EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface->dscp()); |
| channel->SetInterface(nullptr); |
| |
| // Default value when DSCP is enabled is also DSCP_DEFAULT, until it is set |
| // through rtp parameters. |
| config.enable_dscp = true; |
| channel.reset( |
| static_cast<cricket::WebRtcVideoChannel*>(engine_.CreateMediaChannel( |
| call_.get(), config, VideoOptions(), webrtc::CryptoOptions(), |
| video_bitrate_allocator_factory_.get()))); |
| channel->SetInterface(network_interface.get()); |
| EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface->dscp()); |
| |
| // Create a send stream to configure |
| EXPECT_TRUE(channel->AddSendStream(StreamParams::CreateLegacy(kSsrc))); |
| parameters = channel->GetRtpSendParameters(kSsrc); |
| ASSERT_FALSE(parameters.encodings.empty()); |
| |
| // Various priorities map to various dscp values. |
| parameters.encodings[0].network_priority = webrtc::Priority::kHigh; |
| ASSERT_TRUE(channel->SetRtpSendParameters(kSsrc, parameters).ok()); |
| EXPECT_EQ(rtc::DSCP_AF41, network_interface->dscp()); |
| parameters.encodings[0].network_priority = webrtc::Priority::kVeryLow; |
| ASSERT_TRUE(channel->SetRtpSendParameters(kSsrc, parameters).ok()); |
| EXPECT_EQ(rtc::DSCP_CS1, network_interface->dscp()); |
| |
| // Packets should also self-identify their dscp in PacketOptions. |
| const uint8_t kData[10] = {0}; |
| EXPECT_TRUE(static_cast<webrtc::Transport*>(channel.get()) |
| ->SendRtcp(kData, sizeof(kData))); |
| EXPECT_EQ(rtc::DSCP_CS1, network_interface->options().dscp); |
| channel->SetInterface(nullptr); |
| |
| // Verify that setting the option to false resets the |
| // DiffServCodePoint. |
| config.enable_dscp = false; |
| channel.reset( |
| static_cast<cricket::WebRtcVideoChannel*>(engine_.CreateMediaChannel( |
| call_.get(), config, VideoOptions(), webrtc::CryptoOptions(), |
| video_bitrate_allocator_factory_.get()))); |
| channel->SetInterface(network_interface.get()); |
| EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface->dscp()); |
| channel->SetInterface(nullptr); |
| } |
| |
| // This test verifies that the RTCP reduced size mode is properly applied to |
| // send video streams. |
| TEST_F(WebRtcVideoChannelTest, TestSetSendRtcpReducedSize) { |
| // Create stream, expecting that default mode is "compound". |
| FakeVideoSendStream* stream1 = AddSendStream(); |
| EXPECT_EQ(webrtc::RtcpMode::kCompound, stream1->GetConfig().rtp.rtcp_mode); |
| webrtc::RtpParameters rtp_parameters = |
| channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_FALSE(rtp_parameters.rtcp.reduced_size); |
| |
| // Now enable reduced size mode. |
| send_parameters_.rtcp.reduced_size = true; |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| stream1 = fake_call_->GetVideoSendStreams()[0]; |
| EXPECT_EQ(webrtc::RtcpMode::kReducedSize, stream1->GetConfig().rtp.rtcp_mode); |
| rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_TRUE(rtp_parameters.rtcp.reduced_size); |
| |
| // Create a new stream and ensure it picks up the reduced size mode. |
| FakeVideoSendStream* stream2 = AddSendStream(); |
| EXPECT_EQ(webrtc::RtcpMode::kReducedSize, stream2->GetConfig().rtp.rtcp_mode); |
| } |
| |
| // This test verifies that the RTCP reduced size mode is properly applied to |
| // receive video streams. |
| TEST_F(WebRtcVideoChannelTest, TestSetRecvRtcpReducedSize) { |
| // Create stream, expecting that default mode is "compound". |
| FakeVideoReceiveStream* stream1 = AddRecvStream(); |
| EXPECT_EQ(webrtc::RtcpMode::kCompound, stream1->GetConfig().rtp.rtcp_mode); |
| |
| // Now enable reduced size mode. |
| // TODO(deadbeef): Once "recv_parameters" becomes "receiver_parameters", |
| // the reduced_size flag should come from that. |
| send_parameters_.rtcp.reduced_size = true; |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| stream1 = fake_call_->GetVideoReceiveStreams()[0]; |
| EXPECT_EQ(webrtc::RtcpMode::kReducedSize, stream1->GetConfig().rtp.rtcp_mode); |
| |
| // Create a new stream and ensure it picks up the reduced size mode. |
| FakeVideoReceiveStream* stream2 = AddRecvStream(); |
| EXPECT_EQ(webrtc::RtcpMode::kReducedSize, stream2->GetConfig().rtp.rtcp_mode); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, OnReadyToSendSignalsNetworkState) { |
| EXPECT_EQ(webrtc::kNetworkUp, |
| fake_call_->GetNetworkState(webrtc::MediaType::VIDEO)); |
| EXPECT_EQ(webrtc::kNetworkUp, |
| fake_call_->GetNetworkState(webrtc::MediaType::AUDIO)); |
| |
| channel_->OnReadyToSend(false); |
| EXPECT_EQ(webrtc::kNetworkDown, |
| fake_call_->GetNetworkState(webrtc::MediaType::VIDEO)); |
| EXPECT_EQ(webrtc::kNetworkUp, |
| fake_call_->GetNetworkState(webrtc::MediaType::AUDIO)); |
| |
| channel_->OnReadyToSend(true); |
| EXPECT_EQ(webrtc::kNetworkUp, |
| fake_call_->GetNetworkState(webrtc::MediaType::VIDEO)); |
| EXPECT_EQ(webrtc::kNetworkUp, |
| fake_call_->GetNetworkState(webrtc::MediaType::AUDIO)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, GetStatsReportsSentCodecName) { |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| |
| AddSendStream(); |
| |
| cricket::VideoMediaInfo info; |
| ASSERT_TRUE(channel_->GetStats(&info)); |
| EXPECT_EQ("VP8", info.senders[0].codec_name); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, GetStatsReportsEncoderImplementationName) { |
| FakeVideoSendStream* stream = AddSendStream(); |
| webrtc::VideoSendStream::Stats stats; |
| stats.encoder_implementation_name = "encoder_implementation_name"; |
| stream->SetStats(stats); |
| |
| cricket::VideoMediaInfo info; |
| ASSERT_TRUE(channel_->GetStats(&info)); |
| EXPECT_EQ(stats.encoder_implementation_name, |
| info.senders[0].encoder_implementation_name); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, GetStatsReportsCpuOveruseMetrics) { |
| FakeVideoSendStream* stream = AddSendStream(); |
| webrtc::VideoSendStream::Stats stats; |
| stats.avg_encode_time_ms = 13; |
| stats.encode_usage_percent = 42; |
| stream->SetStats(stats); |
| |
| cricket::VideoMediaInfo info; |
| ASSERT_TRUE(channel_->GetStats(&info)); |
| EXPECT_EQ(stats.avg_encode_time_ms, info.senders[0].avg_encode_ms); |
| EXPECT_EQ(stats.encode_usage_percent, info.senders[0].encode_usage_percent); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, GetStatsReportsFramesEncoded) { |
| FakeVideoSendStream* stream = AddSendStream(); |
| webrtc::VideoSendStream::Stats stats; |
| stats.frames_encoded = 13; |
| stream->SetStats(stats); |
| |
| cricket::VideoMediaInfo info; |
| ASSERT_TRUE(channel_->GetStats(&info)); |
| EXPECT_EQ(stats.frames_encoded, info.senders[0].frames_encoded); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, GetStatsReportsKeyFramesEncoded) { |
| FakeVideoSendStream* stream = AddSendStream(); |
| webrtc::VideoSendStream::Stats stats; |
| stats.substreams[123].frame_counts.key_frames = 10; |
| stats.substreams[456].frame_counts.key_frames = 87; |
| stream->SetStats(stats); |
| |
| cricket::VideoMediaInfo info; |
| ASSERT_TRUE(channel_->GetStats(&info)); |
| EXPECT_EQ(info.senders.size(), 2u); |
| EXPECT_EQ(10u, info.senders[0].key_frames_encoded); |
| EXPECT_EQ(87u, info.senders[1].key_frames_encoded); |
| EXPECT_EQ(97u, info.aggregated_senders[0].key_frames_encoded); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, GetStatsReportsPerLayerQpSum) { |
| FakeVideoSendStream* stream = AddSendStream(); |
| webrtc::VideoSendStream::Stats stats; |
| stats.substreams[123].qp_sum = 15; |
| stats.substreams[456].qp_sum = 11; |
| stream->SetStats(stats); |
| |
| cricket::VideoMediaInfo info; |
| ASSERT_TRUE(channel_->GetStats(&info)); |
| EXPECT_EQ(info.senders.size(), 2u); |
| EXPECT_EQ(stats.substreams[123].qp_sum, info.senders[0].qp_sum); |
| EXPECT_EQ(stats.substreams[456].qp_sum, info.senders[1].qp_sum); |
| EXPECT_EQ(*info.aggregated_senders[0].qp_sum, 26u); |
| } |
| |
| webrtc::VideoSendStream::Stats GetInitialisedStats() { |
| webrtc::VideoSendStream::Stats stats; |
| stats.encoder_implementation_name = "vp"; |
| stats.input_frame_rate = 1; |
| stats.encode_frame_rate = 2; |
| stats.avg_encode_time_ms = 3; |
| stats.encode_usage_percent = 4; |
| stats.frames_encoded = 5; |
| stats.total_encode_time_ms = 6; |
| stats.frames_dropped_by_capturer = 7; |
| stats.frames_dropped_by_encoder_queue = 8; |
| stats.frames_dropped_by_rate_limiter = 9; |
| stats.frames_dropped_by_congestion_window = 10; |
| stats.frames_dropped_by_encoder = 11; |
| stats.target_media_bitrate_bps = 13; |
| stats.media_bitrate_bps = 14; |
| stats.suspended = true; |
| stats.bw_limited_resolution = true; |
| stats.cpu_limited_resolution = true; |
| // Not wired. |
| stats.bw_limited_framerate = true; |
| // Not wired. |
| stats.cpu_limited_framerate = true; |
| stats.quality_limitation_reason = webrtc::QualityLimitationReason::kCpu; |
| stats.quality_limitation_durations_ms[webrtc::QualityLimitationReason::kCpu] = |
| 15; |
| stats.quality_limitation_resolution_changes = 16; |
| stats.number_of_cpu_adapt_changes = 17; |
| stats.number_of_quality_adapt_changes = 18; |
| stats.has_entered_low_resolution = true; |
| stats.content_type = webrtc::VideoContentType::SCREENSHARE; |
| stats.frames_sent = 19; |
| stats.huge_frames_sent = 20; |
| |
| return stats; |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, GetAggregatedStatsReportWithoutSubStreams) { |
| FakeVideoSendStream* stream = AddSendStream(); |
| auto stats = GetInitialisedStats(); |
| stream->SetStats(stats); |
| cricket::VideoMediaInfo video_media_info; |
| ASSERT_TRUE(channel_->GetStats(&video_media_info)); |
| EXPECT_EQ(video_media_info.aggregated_senders.size(), 1u); |
| auto& sender = video_media_info.aggregated_senders[0]; |
| |
| // MediaSenderInfo |
| |
| EXPECT_EQ(sender.payload_bytes_sent, 0); |
| EXPECT_EQ(sender.header_and_padding_bytes_sent, 0); |
| EXPECT_EQ(sender.retransmitted_bytes_sent, 0u); |
| EXPECT_EQ(sender.packets_sent, 0); |
| EXPECT_EQ(sender.retransmitted_packets_sent, 0u); |
| EXPECT_EQ(sender.packets_lost, 0); |
| EXPECT_EQ(sender.fraction_lost, 0.0f); |
| EXPECT_EQ(sender.rtt_ms, 0); |
| EXPECT_EQ(sender.codec_name, DefaultCodec().name); |
| EXPECT_EQ(sender.codec_payload_type, DefaultCodec().id); |
| EXPECT_EQ(sender.local_stats.size(), 1u); |
| EXPECT_EQ(sender.local_stats[0].ssrc, last_ssrc_); |
| EXPECT_EQ(sender.local_stats[0].timestamp, 0.0f); |
| EXPECT_EQ(sender.remote_stats.size(), 0u); |
| EXPECT_EQ(sender.report_block_datas.size(), 0u); |
| |
| // VideoSenderInfo |
| |
| EXPECT_EQ(sender.ssrc_groups.size(), 0u); |
| EXPECT_EQ(sender.encoder_implementation_name, |
| stats.encoder_implementation_name); |
| // Comes from substream only. |
| EXPECT_EQ(sender.firs_rcvd, 0); |
| EXPECT_EQ(sender.plis_rcvd, 0); |
| EXPECT_EQ(sender.nacks_rcvd, 0u); |
| EXPECT_EQ(sender.send_frame_width, 0); |
| EXPECT_EQ(sender.send_frame_height, 0); |
| |
| EXPECT_EQ(sender.framerate_input, stats.input_frame_rate); |
| EXPECT_EQ(sender.framerate_sent, stats.encode_frame_rate); |
| EXPECT_EQ(sender.nominal_bitrate, stats.media_bitrate_bps); |
| EXPECT_NE(sender.adapt_reason & WebRtcVideoChannel::ADAPTREASON_CPU, 0); |
| EXPECT_NE(sender.adapt_reason & WebRtcVideoChannel::ADAPTREASON_BANDWIDTH, 0); |
| EXPECT_EQ(sender.adapt_changes, stats.number_of_cpu_adapt_changes); |
| EXPECT_EQ(sender.quality_limitation_reason, stats.quality_limitation_reason); |
| EXPECT_EQ(sender.quality_limitation_durations_ms, |
| stats.quality_limitation_durations_ms); |
| EXPECT_EQ(sender.quality_limitation_resolution_changes, |
| stats.quality_limitation_resolution_changes); |
| EXPECT_EQ(sender.avg_encode_ms, stats.avg_encode_time_ms); |
| EXPECT_EQ(sender.encode_usage_percent, stats.encode_usage_percent); |
| EXPECT_EQ(sender.frames_encoded, stats.frames_encoded); |
| // Comes from substream only. |
| EXPECT_EQ(sender.key_frames_encoded, 0u); |
| |
| EXPECT_EQ(sender.total_encode_time_ms, stats.total_encode_time_ms); |
| EXPECT_EQ(sender.total_encoded_bytes_target, |
| stats.total_encoded_bytes_target); |
| // Comes from substream only. |
| EXPECT_EQ(sender.total_packet_send_delay_ms, 0u); |
| EXPECT_EQ(sender.qp_sum, absl::nullopt); |
| |
| EXPECT_EQ(sender.has_entered_low_resolution, |
| stats.has_entered_low_resolution); |
| EXPECT_EQ(sender.content_type, webrtc::VideoContentType::SCREENSHARE); |
| EXPECT_EQ(sender.frames_sent, stats.frames_encoded); |
| EXPECT_EQ(sender.huge_frames_sent, stats.huge_frames_sent); |
| EXPECT_EQ(sender.rid, absl::nullopt); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, GetAggregatedStatsReportForSubStreams) { |
| FakeVideoSendStream* stream = AddSendStream(); |
| auto stats = GetInitialisedStats(); |
| |
| const uint32_t ssrc_1 = 123u; |
| const uint32_t ssrc_2 = 456u; |
| |
| auto& substream = stats.substreams[ssrc_1]; |
| substream.frame_counts.key_frames = 1; |
| substream.frame_counts.delta_frames = 2; |
| substream.width = 3; |
| substream.height = 4; |
| substream.total_bitrate_bps = 5; |
| substream.retransmit_bitrate_bps = 6; |
| substream.avg_delay_ms = 7; |
| substream.max_delay_ms = 8; |
| substream.total_packet_send_delay_ms = 9; |
| substream.rtp_stats.transmitted.header_bytes = 10; |
| substream.rtp_stats.transmitted.padding_bytes = 11; |
| substream.rtp_stats.retransmitted.payload_bytes = 12; |
| substream.rtp_stats.retransmitted.packets = 13; |
| substream.rtcp_packet_type_counts.fir_packets = 14; |
| substream.rtcp_packet_type_counts.nack_packets = 15; |
| substream.rtcp_packet_type_counts.pli_packets = 16; |
| webrtc::RTCPReportBlock report_block; |
| report_block.packets_lost = 17; |
| report_block.fraction_lost = 18; |
| webrtc::ReportBlockData report_block_data; |
| report_block_data.SetReportBlock(report_block, 0); |
| report_block_data.AddRoundTripTimeSample(19); |
| substream.report_block_data = report_block_data; |
| substream.encode_frame_rate = 20.0; |
| substream.frames_encoded = 21; |
| substream.qp_sum = 22; |
| substream.total_encode_time_ms = 23; |
| substream.total_encoded_bytes_target = 24; |
| substream.huge_frames_sent = 25; |
| |
| stats.substreams[ssrc_2] = substream; |
| |
| stream->SetStats(stats); |
| |
| cricket::VideoMediaInfo video_media_info; |
| ASSERT_TRUE(channel_->GetStats(&video_media_info)); |
| EXPECT_EQ(video_media_info.aggregated_senders.size(), 1u); |
| auto& sender = video_media_info.aggregated_senders[0]; |
| |
| // MediaSenderInfo |
| |
| EXPECT_EQ( |
| sender.payload_bytes_sent, |
| static_cast<int64_t>(2u * substream.rtp_stats.transmitted.payload_bytes)); |
| EXPECT_EQ(sender.header_and_padding_bytes_sent, |
| static_cast<int64_t>( |
| 2u * (substream.rtp_stats.transmitted.header_bytes + |
| substream.rtp_stats.transmitted.padding_bytes))); |
| EXPECT_EQ(sender.retransmitted_bytes_sent, |
| 2u * substream.rtp_stats.retransmitted.payload_bytes); |
| EXPECT_EQ(sender.packets_sent, |
| static_cast<int>(2 * substream.rtp_stats.transmitted.packets)); |
| EXPECT_EQ(sender.retransmitted_packets_sent, |
| 2u * substream.rtp_stats.retransmitted.packets); |
| EXPECT_EQ(sender.packets_lost, |
| 2 * substream.report_block_data->report_block().packets_lost); |
| EXPECT_EQ(sender.fraction_lost, |
| static_cast<float>( |
| substream.report_block_data->report_block().fraction_lost) / |
| (1 << 8)); |
| EXPECT_EQ(sender.rtt_ms, 0); |
| EXPECT_EQ(sender.codec_name, DefaultCodec().name); |
| EXPECT_EQ(sender.codec_payload_type, DefaultCodec().id); |
| EXPECT_EQ(sender.local_stats.size(), 1u); |
| EXPECT_EQ(sender.local_stats[0].ssrc, last_ssrc_); |
| EXPECT_EQ(sender.local_stats[0].timestamp, 0.0f); |
| EXPECT_EQ(sender.remote_stats.size(), 0u); |
| EXPECT_EQ(sender.report_block_datas.size(), 2u * 1); |
| |
| // VideoSenderInfo |
| |
| EXPECT_EQ(sender.ssrc_groups.size(), 0u); |
| EXPECT_EQ(sender.encoder_implementation_name, |
| stats.encoder_implementation_name); |
| EXPECT_EQ( |
| sender.firs_rcvd, |
| static_cast<int>(2 * substream.rtcp_packet_type_counts.fir_packets)); |
| EXPECT_EQ( |
| sender.plis_rcvd, |
| static_cast<int>(2 * substream.rtcp_packet_type_counts.pli_packets)); |
| EXPECT_EQ(sender.nacks_rcvd, |
| 2 * substream.rtcp_packet_type_counts.nack_packets); |
| EXPECT_EQ(sender.send_frame_width, substream.width); |
| EXPECT_EQ(sender.send_frame_height, substream.height); |
| |
| EXPECT_EQ(sender.framerate_input, stats.input_frame_rate); |
| EXPECT_EQ(sender.framerate_sent, stats.encode_frame_rate); |
| EXPECT_EQ(sender.nominal_bitrate, stats.media_bitrate_bps); |
| EXPECT_NE(sender.adapt_reason & WebRtcVideoChannel::ADAPTREASON_CPU, 0); |
| EXPECT_NE(sender.adapt_reason & WebRtcVideoChannel::ADAPTREASON_BANDWIDTH, 0); |
| EXPECT_EQ(sender.adapt_changes, stats.number_of_cpu_adapt_changes); |
| EXPECT_EQ(sender.quality_limitation_reason, stats.quality_limitation_reason); |
| EXPECT_EQ(sender.quality_limitation_durations_ms, |
| stats.quality_limitation_durations_ms); |
| EXPECT_EQ(sender.quality_limitation_resolution_changes, |
| stats.quality_limitation_resolution_changes); |
| EXPECT_EQ(sender.avg_encode_ms, stats.avg_encode_time_ms); |
| EXPECT_EQ(sender.encode_usage_percent, stats.encode_usage_percent); |
| EXPECT_EQ(sender.frames_encoded, 2u * substream.frames_encoded); |
| EXPECT_EQ(sender.key_frames_encoded, 2u * substream.frame_counts.key_frames); |
| EXPECT_EQ(sender.total_encode_time_ms, 2u * substream.total_encode_time_ms); |
| EXPECT_EQ(sender.total_encoded_bytes_target, |
| 2u * substream.total_encoded_bytes_target); |
| EXPECT_EQ(sender.total_packet_send_delay_ms, |
| 2u * substream.total_packet_send_delay_ms); |
| EXPECT_EQ(sender.has_entered_low_resolution, |
| stats.has_entered_low_resolution); |
| EXPECT_EQ(sender.qp_sum, 2u * *substream.qp_sum); |
| EXPECT_EQ(sender.content_type, webrtc::VideoContentType::SCREENSHARE); |
| EXPECT_EQ(sender.frames_sent, 2u * substream.frames_encoded); |
| EXPECT_EQ(sender.huge_frames_sent, stats.huge_frames_sent); |
| EXPECT_EQ(sender.rid, absl::nullopt); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, GetPerLayerStatsReportForSubStreams) { |
| FakeVideoSendStream* stream = AddSendStream(); |
| auto stats = GetInitialisedStats(); |
| |
| const uint32_t ssrc_1 = 123u; |
| const uint32_t ssrc_2 = 456u; |
| |
| auto& substream = stats.substreams[ssrc_1]; |
| substream.frame_counts.key_frames = 1; |
| substream.frame_counts.delta_frames = 2; |
| substream.width = 3; |
| substream.height = 4; |
| substream.total_bitrate_bps = 5; |
| substream.retransmit_bitrate_bps = 6; |
| substream.avg_delay_ms = 7; |
| substream.max_delay_ms = 8; |
| substream.total_packet_send_delay_ms = 9; |
| substream.rtp_stats.transmitted.header_bytes = 10; |
| substream.rtp_stats.transmitted.padding_bytes = 11; |
| substream.rtp_stats.retransmitted.payload_bytes = 12; |
| substream.rtp_stats.retransmitted.packets = 13; |
| substream.rtcp_packet_type_counts.fir_packets = 14; |
| substream.rtcp_packet_type_counts.nack_packets = 15; |
| substream.rtcp_packet_type_counts.pli_packets = 16; |
| webrtc::RTCPReportBlock report_block; |
| report_block.packets_lost = 17; |
| report_block.fraction_lost = 18; |
| webrtc::ReportBlockData report_block_data; |
| report_block_data.SetReportBlock(report_block, 0); |
| report_block_data.AddRoundTripTimeSample(19); |
| substream.report_block_data = report_block_data; |
| substream.encode_frame_rate = 20.0; |
| substream.frames_encoded = 21; |
| substream.qp_sum = 22; |
| substream.total_encode_time_ms = 23; |
| substream.total_encoded_bytes_target = 24; |
| substream.huge_frames_sent = 25; |
| |
| stats.substreams[ssrc_2] = substream; |
| |
| stream->SetStats(stats); |
| |
| cricket::VideoMediaInfo video_media_info; |
| ASSERT_TRUE(channel_->GetStats(&video_media_info)); |
| EXPECT_EQ(video_media_info.senders.size(), 2u); |
| auto& sender = video_media_info.senders[0]; |
| |
| // MediaSenderInfo |
| |
| EXPECT_EQ( |
| sender.payload_bytes_sent, |
| static_cast<int64_t>(substream.rtp_stats.transmitted.payload_bytes)); |
| EXPECT_EQ( |
| sender.header_and_padding_bytes_sent, |
| static_cast<int64_t>(substream.rtp_stats.transmitted.header_bytes + |
| substream.rtp_stats.transmitted.padding_bytes)); |
| EXPECT_EQ(sender.retransmitted_bytes_sent, |
| substream.rtp_stats.retransmitted.payload_bytes); |
| EXPECT_EQ(sender.packets_sent, |
| static_cast<int>(substream.rtp_stats.transmitted.packets)); |
| EXPECT_EQ(sender.retransmitted_packets_sent, |
| substream.rtp_stats.retransmitted.packets); |
| EXPECT_EQ(sender.packets_lost, |
| substream.report_block_data->report_block().packets_lost); |
| EXPECT_EQ(sender.fraction_lost, |
| static_cast<float>( |
| substream.report_block_data->report_block().fraction_lost) / |
| (1 << 8)); |
| EXPECT_EQ(sender.rtt_ms, 0); |
| EXPECT_EQ(sender.codec_name, DefaultCodec().name); |
| EXPECT_EQ(sender.codec_payload_type, DefaultCodec().id); |
| EXPECT_EQ(sender.local_stats.size(), 1u); |
| EXPECT_EQ(sender.local_stats[0].ssrc, ssrc_1); |
| EXPECT_EQ(sender.local_stats[0].timestamp, 0.0f); |
| EXPECT_EQ(sender.remote_stats.size(), 0u); |
| EXPECT_EQ(sender.report_block_datas.size(), 1u); |
| |
| // VideoSenderInfo |
| |
| EXPECT_EQ(sender.ssrc_groups.size(), 0u); |
| EXPECT_EQ(sender.encoder_implementation_name, |
| stats.encoder_implementation_name); |
| EXPECT_EQ(sender.firs_rcvd, |
| static_cast<int>(substream.rtcp_packet_type_counts.fir_packets)); |
| EXPECT_EQ(sender.plis_rcvd, |
| static_cast<int>(substream.rtcp_packet_type_counts.pli_packets)); |
| EXPECT_EQ(sender.nacks_rcvd, substream.rtcp_packet_type_counts.nack_packets); |
| EXPECT_EQ(sender.send_frame_width, substream.width); |
| EXPECT_EQ(sender.send_frame_height, substream.height); |
| |
| EXPECT_EQ(sender.framerate_input, stats.input_frame_rate); |
| EXPECT_EQ(sender.framerate_sent, substream.encode_frame_rate); |
| EXPECT_EQ(sender.nominal_bitrate, stats.media_bitrate_bps); |
| EXPECT_NE(sender.adapt_reason & WebRtcVideoChannel::ADAPTREASON_CPU, 0); |
| EXPECT_NE(sender.adapt_reason & WebRtcVideoChannel::ADAPTREASON_BANDWIDTH, 0); |
| EXPECT_EQ(sender.adapt_changes, stats.number_of_cpu_adapt_changes); |
| EXPECT_EQ(sender.quality_limitation_reason, stats.quality_limitation_reason); |
| EXPECT_EQ(sender.quality_limitation_durations_ms, |
| stats.quality_limitation_durations_ms); |
| EXPECT_EQ(sender.quality_limitation_resolution_changes, |
| stats.quality_limitation_resolution_changes); |
| EXPECT_EQ(sender.avg_encode_ms, stats.avg_encode_time_ms); |
| EXPECT_EQ(sender.encode_usage_percent, stats.encode_usage_percent); |
| EXPECT_EQ(sender.frames_encoded, |
| static_cast<uint32_t>(substream.frames_encoded)); |
| EXPECT_EQ(sender.key_frames_encoded, |
| static_cast<uint32_t>(substream.frame_counts.key_frames)); |
| EXPECT_EQ(sender.total_encode_time_ms, substream.total_encode_time_ms); |
| EXPECT_EQ(sender.total_encoded_bytes_target, |
| substream.total_encoded_bytes_target); |
| EXPECT_EQ(sender.total_packet_send_delay_ms, |
| substream.total_packet_send_delay_ms); |
| EXPECT_EQ(sender.has_entered_low_resolution, |
| stats.has_entered_low_resolution); |
| EXPECT_EQ(sender.qp_sum, *substream.qp_sum); |
| EXPECT_EQ(sender.content_type, webrtc::VideoContentType::SCREENSHARE); |
| EXPECT_EQ(sender.frames_sent, |
| static_cast<uint32_t>(substream.frames_encoded)); |
| EXPECT_EQ(sender.huge_frames_sent, substream.huge_frames_sent); |
| EXPECT_EQ(sender.rid, absl::nullopt); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, MediaSubstreamMissingProducesEmpyStats) { |
| FakeVideoSendStream* stream = AddSendStream(); |
| |
| const uint32_t kRtxSsrc = 123u; |
| const uint32_t kMissingMediaSsrc = 124u; |
| |
| // Set up a scenarios where we have a substream that is not kMedia (in this |
| // case: kRtx) but its associated kMedia stream does not exist yet. This |
| // results in zero GetPerLayerVideoSenderInfos despite non-empty substreams. |
| // Covers https://crbug.com/1090712. |
| auto stats = GetInitialisedStats(); |
| auto& substream = stats.substreams[kRtxSsrc]; |
| substream.type = webrtc::VideoSendStream::StreamStats::StreamType::kRtx; |
| substream.referenced_media_ssrc = kMissingMediaSsrc; |
| stream->SetStats(stats); |
| |
| cricket::VideoMediaInfo video_media_info; |
| ASSERT_TRUE(channel_->GetStats(&video_media_info)); |
| EXPECT_TRUE(video_media_info.senders.empty()); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, GetStatsReportsUpperResolution) { |
| FakeVideoSendStream* stream = AddSendStream(); |
| webrtc::VideoSendStream::Stats stats; |
| stats.substreams[17].width = 123; |
| stats.substreams[17].height = 40; |
| stats.substreams[42].width = 80; |
| stats.substreams[42].height = 31; |
| stats.substreams[11].width = 20; |
| stats.substreams[11].height = 90; |
| stream->SetStats(stats); |
| |
| cricket::VideoMediaInfo info; |
| ASSERT_TRUE(channel_->GetStats(&info)); |
| ASSERT_EQ(1u, info.aggregated_senders.size()); |
| ASSERT_EQ(3u, info.senders.size()); |
| EXPECT_EQ(123, info.senders[1].send_frame_width); |
| EXPECT_EQ(40, info.senders[1].send_frame_height); |
| EXPECT_EQ(80, info.senders[2].send_frame_width); |
| EXPECT_EQ(31, info.senders[2].send_frame_height); |
| EXPECT_EQ(20, info.senders[0].send_frame_width); |
| EXPECT_EQ(90, info.senders[0].send_frame_height); |
| EXPECT_EQ(123, info.aggregated_senders[0].send_frame_width); |
| EXPECT_EQ(90, info.aggregated_senders[0].send_frame_height); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, GetStatsReportsCpuAdaptationStats) { |
| FakeVideoSendStream* stream = AddSendStream(); |
| webrtc::VideoSendStream::Stats stats; |
| stats.number_of_cpu_adapt_changes = 2; |
| stats.cpu_limited_resolution = true; |
| stream->SetStats(stats); |
| |
| cricket::VideoMediaInfo info; |
| EXPECT_TRUE(channel_->GetStats(&info)); |
| ASSERT_EQ(1U, info.senders.size()); |
| EXPECT_EQ(WebRtcVideoChannel::ADAPTREASON_CPU, info.senders[0].adapt_reason); |
| EXPECT_EQ(stats.number_of_cpu_adapt_changes, info.senders[0].adapt_changes); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, GetStatsReportsAdaptationAndBandwidthStats) { |
| FakeVideoSendStream* stream = AddSendStream(); |
| webrtc::VideoSendStream::Stats stats; |
| stats.number_of_cpu_adapt_changes = 2; |
| stats.cpu_limited_resolution = true; |
| stats.bw_limited_resolution = true; |
| stream->SetStats(stats); |
| |
| cricket::VideoMediaInfo info; |
| EXPECT_TRUE(channel_->GetStats(&info)); |
| ASSERT_EQ(1U, info.senders.size()); |
| EXPECT_EQ(WebRtcVideoChannel::ADAPTREASON_CPU | |
| WebRtcVideoChannel::ADAPTREASON_BANDWIDTH, |
| info.senders[0].adapt_reason); |
| EXPECT_EQ(stats.number_of_cpu_adapt_changes, info.senders[0].adapt_changes); |
| } |
| |
| TEST(WebRtcVideoChannelHelperTest, MergeInfoAboutOutboundRtpSubstreams) { |
| const uint32_t kFirstMediaStreamSsrc = 10; |
| const uint32_t kSecondMediaStreamSsrc = 20; |
| const uint32_t kRtxSsrc = 30; |
| const uint32_t kFlexfecSsrc = 40; |
| std::map<uint32_t, webrtc::VideoSendStream::StreamStats> substreams; |
| // First kMedia stream. |
| substreams[kFirstMediaStreamSsrc].type = |
| webrtc::VideoSendStream::StreamStats::StreamType::kMedia; |
| substreams[kFirstMediaStreamSsrc].rtp_stats.transmitted.header_bytes = 1; |
| substreams[kFirstMediaStreamSsrc].rtp_stats.transmitted.padding_bytes = 2; |
| substreams[kFirstMediaStreamSsrc].rtp_stats.transmitted.payload_bytes = 3; |
| substreams[kFirstMediaStreamSsrc].rtp_stats.transmitted.packets = 4; |
| substreams[kFirstMediaStreamSsrc].rtp_stats.retransmitted.header_bytes = 5; |
| substreams[kFirstMediaStreamSsrc].rtp_stats.retransmitted.padding_bytes = 6; |
| substreams[kFirstMediaStreamSsrc].rtp_stats.retransmitted.payload_bytes = 7; |
| substreams[kFirstMediaStreamSsrc].rtp_stats.retransmitted.packets = 8; |
| substreams[kFirstMediaStreamSsrc].referenced_media_ssrc = absl::nullopt; |
| substreams[kFirstMediaStreamSsrc].width = 1280; |
| substreams[kFirstMediaStreamSsrc].height = 720; |
| // Second kMedia stream. |
| substreams[kSecondMediaStreamSsrc].type = |
| webrtc::VideoSendStream::StreamStats::StreamType::kMedia; |
| substreams[kSecondMediaStreamSsrc].rtp_stats.transmitted.header_bytes = 10; |
| substreams[kSecondMediaStreamSsrc].rtp_stats.transmitted.padding_bytes = 11; |
| substreams[kSecondMediaStreamSsrc].rtp_stats.transmitted.payload_bytes = 12; |
| substreams[kSecondMediaStreamSsrc].rtp_stats.transmitted.packets = 13; |
| substreams[kSecondMediaStreamSsrc].rtp_stats.retransmitted.header_bytes = 14; |
| substreams[kSecondMediaStreamSsrc].rtp_stats.retransmitted.padding_bytes = 15; |
| substreams[kSecondMediaStreamSsrc].rtp_stats.retransmitted.payload_bytes = 16; |
| substreams[kSecondMediaStreamSsrc].rtp_stats.retransmitted.packets = 17; |
| substreams[kSecondMediaStreamSsrc].referenced_media_ssrc = absl::nullopt; |
| substreams[kSecondMediaStreamSsrc].width = 640; |
| substreams[kSecondMediaStreamSsrc].height = 480; |
| // kRtx stream referencing the first kMedia stream. |
| substreams[kRtxSsrc].type = |
| webrtc::VideoSendStream::StreamStats::StreamType::kRtx; |
| substreams[kRtxSsrc].rtp_stats.transmitted.header_bytes = 19; |
| substreams[kRtxSsrc].rtp_stats.transmitted.padding_bytes = 20; |
| substreams[kRtxSsrc].rtp_stats.transmitted.payload_bytes = 21; |
| substreams[kRtxSsrc].rtp_stats.transmitted.packets = 22; |
| substreams[kRtxSsrc].rtp_stats.retransmitted.header_bytes = 23; |
| substreams[kRtxSsrc].rtp_stats.retransmitted.padding_bytes = 24; |
| substreams[kRtxSsrc].rtp_stats.retransmitted.payload_bytes = 25; |
| substreams[kRtxSsrc].rtp_stats.retransmitted.packets = 26; |
| substreams[kRtxSsrc].referenced_media_ssrc = kFirstMediaStreamSsrc; |
| // kFlexfec stream referencing the second kMedia stream. |
| substreams[kFlexfecSsrc].type = |
| webrtc::VideoSendStream::StreamStats::StreamType::kFlexfec; |
| substreams[kFlexfecSsrc].rtp_stats.transmitted.header_bytes = 19; |
| substreams[kFlexfecSsrc].rtp_stats.transmitted.padding_bytes = 20; |
| substreams[kFlexfecSsrc].rtp_stats.transmitted.payload_bytes = 21; |
| substreams[kFlexfecSsrc].rtp_stats.transmitted.packets = 22; |
| substreams[kFlexfecSsrc].rtp_stats.retransmitted.header_bytes = 23; |
| substreams[kFlexfecSsrc].rtp_stats.retransmitted.padding_bytes = 24; |
| substreams[kFlexfecSsrc].rtp_stats.retransmitted.payload_bytes = 25; |
| substreams[kFlexfecSsrc].rtp_stats.retransmitted.packets = 26; |
| substreams[kFlexfecSsrc].referenced_media_ssrc = kSecondMediaStreamSsrc; |
| |
| auto merged_substreams = |
| MergeInfoAboutOutboundRtpSubstreamsForTesting(substreams); |
| // Only kMedia substreams remain. |
| EXPECT_TRUE(merged_substreams.find(kFirstMediaStreamSsrc) != |
| merged_substreams.end()); |
| EXPECT_EQ(merged_substreams[kFirstMediaStreamSsrc].type, |
| webrtc::VideoSendStream::StreamStats::StreamType::kMedia); |
| EXPECT_TRUE(merged_substreams.find(kSecondMediaStreamSsrc) != |
| merged_substreams.end()); |
| EXPECT_EQ(merged_substreams[kSecondMediaStreamSsrc].type, |
| webrtc::VideoSendStream::StreamStats::StreamType::kMedia); |
| EXPECT_FALSE(merged_substreams.find(kRtxSsrc) != merged_substreams.end()); |
| EXPECT_FALSE(merged_substreams.find(kFlexfecSsrc) != merged_substreams.end()); |
| // Expect kFirstMediaStreamSsrc's rtp_stats to be merged with kRtxSsrc. |
| webrtc::StreamDataCounters first_media_expected_rtp_stats = |
| substreams[kFirstMediaStreamSsrc].rtp_stats; |
| first_media_expected_rtp_stats.Add(substreams[kRtxSsrc].rtp_stats); |
| EXPECT_EQ(merged_substreams[kFirstMediaStreamSsrc].rtp_stats.transmitted, |
| first_media_expected_rtp_stats.transmitted); |
| EXPECT_EQ(merged_substreams[kFirstMediaStreamSsrc].rtp_stats.retransmitted, |
| first_media_expected_rtp_stats.retransmitted); |
| // Expect kSecondMediaStreamSsrc' rtp_stats to be merged with kFlexfecSsrc. |
| webrtc::StreamDataCounters second_media_expected_rtp_stats = |
| substreams[kSecondMediaStreamSsrc].rtp_stats; |
| second_media_expected_rtp_stats.Add(substreams[kFlexfecSsrc].rtp_stats); |
| EXPECT_EQ(merged_substreams[kSecondMediaStreamSsrc].rtp_stats.transmitted, |
| second_media_expected_rtp_stats.transmitted); |
| EXPECT_EQ(merged_substreams[kSecondMediaStreamSsrc].rtp_stats.retransmitted, |
| second_media_expected_rtp_stats.retransmitted); |
| // Expect other metrics to come from the original kMedia stats. |
| EXPECT_EQ(merged_substreams[kFirstMediaStreamSsrc].width, |
| substreams[kFirstMediaStreamSsrc].width); |
| EXPECT_EQ(merged_substreams[kFirstMediaStreamSsrc].height, |
| substreams[kFirstMediaStreamSsrc].height); |
| EXPECT_EQ(merged_substreams[kSecondMediaStreamSsrc].width, |
| substreams[kSecondMediaStreamSsrc].width); |
| EXPECT_EQ(merged_substreams[kSecondMediaStreamSsrc].height, |
| substreams[kSecondMediaStreamSsrc].height); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| GetStatsReportsTransmittedAndRetransmittedBytesAndPacketsCorrectly) { |
| FakeVideoSendStream* stream = AddSendStream(); |
| webrtc::VideoSendStream::Stats stats; |
| // Simulcast layer 1, RTP stream. header+padding=10, payload=20, packets=3. |
| stats.substreams[101].type = |
| webrtc::VideoSendStream::StreamStats::StreamType::kMedia; |
| stats.substreams[101].rtp_stats.transmitted.header_bytes = 5; |
| stats.substreams[101].rtp_stats.transmitted.padding_bytes = 5; |
| stats.substreams[101].rtp_stats.transmitted.payload_bytes = 20; |
| stats.substreams[101].rtp_stats.transmitted.packets = 3; |
| stats.substreams[101].rtp_stats.retransmitted.header_bytes = 0; |
| stats.substreams[101].rtp_stats.retransmitted.padding_bytes = 0; |
| stats.substreams[101].rtp_stats.retransmitted.payload_bytes = 0; |
| stats.substreams[101].rtp_stats.retransmitted.packets = 0; |
| stats.substreams[101].referenced_media_ssrc = absl::nullopt; |
| // Simulcast layer 1, RTX stream. header+padding=5, payload=10, packets=1. |
| stats.substreams[102].type = |
| webrtc::VideoSendStream::StreamStats::StreamType::kRtx; |
| stats.substreams[102].rtp_stats.retransmitted.header_bytes = 3; |
| stats.substreams[102].rtp_stats.retransmitted.padding_bytes = 2; |
| stats.substreams[102].rtp_stats.retransmitted.payload_bytes = 10; |
| stats.substreams[102].rtp_stats.retransmitted.packets = 1; |
| stats.substreams[102].rtp_stats.transmitted = |
| stats.substreams[102].rtp_stats.retransmitted; |
| stats.substreams[102].referenced_media_ssrc = 101; |
| // Simulcast layer 2, RTP stream. header+padding=20, payload=40, packets=7. |
| stats.substreams[201].type = |
| webrtc::VideoSendStream::StreamStats::StreamType::kMedia; |
| stats.substreams[201].rtp_stats.transmitted.header_bytes = 10; |
| stats.substreams[201].rtp_stats.transmitted.padding_bytes = 10; |
| stats.substreams[201].rtp_stats.transmitted.payload_bytes = 40; |
| stats.substreams[201].rtp_stats.transmitted.packets = 7; |
| stats.substreams[201].rtp_stats.retransmitted.header_bytes = 0; |
| stats.substreams[201].rtp_stats.retransmitted.padding_bytes = 0; |
| stats.substreams[201].rtp_stats.retransmitted.payload_bytes = 0; |
| stats.substreams[201].rtp_stats.retransmitted.packets = 0; |
| stats.substreams[201].referenced_media_ssrc = absl::nullopt; |
| // Simulcast layer 2, RTX stream. header+padding=10, payload=20, packets=4. |
| stats.substreams[202].type = |
| webrtc::VideoSendStream::StreamStats::StreamType::kRtx; |
| stats.substreams[202].rtp_stats.retransmitted.header_bytes = 6; |
| stats.substreams[202].rtp_stats.retransmitted.padding_bytes = 4; |
| stats.substreams[202].rtp_stats.retransmitted.payload_bytes = 20; |
| stats.substreams[202].rtp_stats.retransmitted.packets = 4; |
| stats.substreams[202].rtp_stats.transmitted = |
| stats.substreams[202].rtp_stats.retransmitted; |
| stats.substreams[202].referenced_media_ssrc = 201; |
| // FlexFEC stream associated with the Simulcast layer 2. |
| // header+padding=15, payload=17, packets=5. |
| stats.substreams[301].type = |
| webrtc::VideoSendStream::StreamStats::StreamType::kFlexfec; |
| stats.substreams[301].rtp_stats.transmitted.header_bytes = 13; |
| stats.substreams[301].rtp_stats.transmitted.padding_bytes = 2; |
| stats.substreams[301].rtp_stats.transmitted.payload_bytes = 17; |
| stats.substreams[301].rtp_stats.transmitted.packets = 5; |
| stats.substreams[301].rtp_stats.retransmitted.header_bytes = 0; |
| stats.substreams[301].rtp_stats.retransmitted.padding_bytes = 0; |
| stats.substreams[301].rtp_stats.retransmitted.payload_bytes = 0; |
| stats.substreams[301].rtp_stats.retransmitted.packets = 0; |
| stats.substreams[301].referenced_media_ssrc = 201; |
| stream->SetStats(stats); |
| |
| cricket::VideoMediaInfo info; |
| ASSERT_TRUE(channel_->GetStats(&info)); |
| EXPECT_EQ(info.senders.size(), 2u); |
| EXPECT_EQ(15u, info.senders[0].header_and_padding_bytes_sent); |
| EXPECT_EQ(30u, info.senders[0].payload_bytes_sent); |
| EXPECT_EQ(4, info.senders[0].packets_sent); |
| EXPECT_EQ(10u, info.senders[0].retransmitted_bytes_sent); |
| EXPECT_EQ(1u, info.senders[0].retransmitted_packets_sent); |
| |
| EXPECT_EQ(45u, info.senders[1].header_and_padding_bytes_sent); |
| EXPECT_EQ(77u, info.senders[1].payload_bytes_sent); |
| EXPECT_EQ(16, info.senders[1].packets_sent); |
| EXPECT_EQ(20u, info.senders[1].retransmitted_bytes_sent); |
| EXPECT_EQ(4u, info.senders[1].retransmitted_packets_sent); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| GetStatsTranslatesBandwidthLimitedResolutionCorrectly) { |
| FakeVideoSendStream* stream = AddSendStream(); |
| webrtc::VideoSendStream::Stats stats; |
| stats.bw_limited_resolution = true; |
| stream->SetStats(stats); |
| |
| cricket::VideoMediaInfo info; |
| EXPECT_TRUE(channel_->GetStats(&info)); |
| ASSERT_EQ(1U, info.senders.size()); |
| EXPECT_EQ(WebRtcVideoChannel::ADAPTREASON_BANDWIDTH, |
| info.senders[0].adapt_reason); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, GetStatsTranslatesSendRtcpPacketTypesCorrectly) { |
| FakeVideoSendStream* stream = AddSendStream(); |
| webrtc::VideoSendStream::Stats stats; |
| stats.substreams[17].rtcp_packet_type_counts.fir_packets = 2; |
| stats.substreams[17].rtcp_packet_type_counts.nack_packets = 3; |
| stats.substreams[17].rtcp_packet_type_counts.pli_packets = 4; |
| |
| stats.substreams[42].rtcp_packet_type_counts.fir_packets = 5; |
| stats.substreams[42].rtcp_packet_type_counts.nack_packets = 7; |
| stats.substreams[42].rtcp_packet_type_counts.pli_packets = 9; |
| |
| stream->SetStats(stats); |
| |
| cricket::VideoMediaInfo info; |
| ASSERT_TRUE(channel_->GetStats(&info)); |
| EXPECT_EQ(2, info.senders[0].firs_rcvd); |
| EXPECT_EQ(3u, info.senders[0].nacks_rcvd); |
| EXPECT_EQ(4, info.senders[0].plis_rcvd); |
| |
| EXPECT_EQ(5, info.senders[1].firs_rcvd); |
| EXPECT_EQ(7u, info.senders[1].nacks_rcvd); |
| EXPECT_EQ(9, info.senders[1].plis_rcvd); |
| |
| EXPECT_EQ(7, info.aggregated_senders[0].firs_rcvd); |
| EXPECT_EQ(10u, info.aggregated_senders[0].nacks_rcvd); |
| EXPECT_EQ(13, info.aggregated_senders[0].plis_rcvd); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| GetStatsTranslatesReceiveRtcpPacketTypesCorrectly) { |
| FakeVideoReceiveStream* stream = AddRecvStream(); |
| webrtc::VideoReceiveStream::Stats stats; |
| stats.rtcp_packet_type_counts.fir_packets = 2; |
| stats.rtcp_packet_type_counts.nack_packets = 3; |
| stats.rtcp_packet_type_counts.pli_packets = 4; |
| stream->SetStats(stats); |
| |
| cricket::VideoMediaInfo info; |
| ASSERT_TRUE(channel_->GetStats(&info)); |
| EXPECT_EQ(stats.rtcp_packet_type_counts.fir_packets, |
| rtc::checked_cast<unsigned int>(info.receivers[0].firs_sent)); |
| EXPECT_EQ(stats.rtcp_packet_type_counts.nack_packets, |
| info.receivers[0].nacks_sent); |
| EXPECT_EQ(stats.rtcp_packet_type_counts.pli_packets, |
| rtc::checked_cast<unsigned int>(info.receivers[0].plis_sent)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, GetStatsTranslatesDecodeStatsCorrectly) { |
| FakeVideoReceiveStream* stream = AddRecvStream(); |
| webrtc::VideoReceiveStream::Stats stats; |
| stats.decoder_implementation_name = "decoder_implementation_name"; |
| stats.decode_ms = 2; |
| stats.max_decode_ms = 3; |
| stats.current_delay_ms = 4; |
| stats.target_delay_ms = 5; |
| stats.jitter_buffer_ms = 6; |
| stats.jitter_buffer_delay_seconds = 60; |
| stats.jitter_buffer_emitted_count = 6; |
| stats.min_playout_delay_ms = 7; |
| stats.render_delay_ms = 8; |
| stats.width = 9; |
| stats.height = 10; |
| stats.frame_counts.key_frames = 11; |
| stats.frame_counts.delta_frames = 12; |
| stats.frames_rendered = 13; |
| stats.frames_decoded = 14; |
| stats.qp_sum = 15; |
| stats.total_decode_time_ms = 16; |
| stream->SetStats(stats); |
| |
| cricket::VideoMediaInfo info; |
| ASSERT_TRUE(channel_->GetStats(&info)); |
| EXPECT_EQ(stats.decoder_implementation_name, |
| info.receivers[0].decoder_implementation_name); |
| EXPECT_EQ(stats.decode_ms, info.receivers[0].decode_ms); |
| EXPECT_EQ(stats.max_decode_ms, info.receivers[0].max_decode_ms); |
| EXPECT_EQ(stats.current_delay_ms, info.receivers[0].current_delay_ms); |
| EXPECT_EQ(stats.target_delay_ms, info.receivers[0].target_delay_ms); |
| EXPECT_EQ(stats.jitter_buffer_ms, info.receivers[0].jitter_buffer_ms); |
| EXPECT_EQ(stats.jitter_buffer_delay_seconds, |
| info.receivers[0].jitter_buffer_delay_seconds); |
| EXPECT_EQ(stats.jitter_buffer_emitted_count, |
| info.receivers[0].jitter_buffer_emitted_count); |
| EXPECT_EQ(stats.min_playout_delay_ms, info.receivers[0].min_playout_delay_ms); |
| EXPECT_EQ(stats.render_delay_ms, info.receivers[0].render_delay_ms); |
| EXPECT_EQ(stats.width, info.receivers[0].frame_width); |
| EXPECT_EQ(stats.height, info.receivers[0].frame_height); |
| EXPECT_EQ(rtc::checked_cast<unsigned int>(stats.frame_counts.key_frames + |
| stats.frame_counts.delta_frames), |
| info.receivers[0].frames_received); |
| EXPECT_EQ(stats.frames_rendered, info.receivers[0].frames_rendered); |
| EXPECT_EQ(stats.frames_decoded, info.receivers[0].frames_decoded); |
| EXPECT_EQ(rtc::checked_cast<unsigned int>(stats.frame_counts.key_frames), |
| info.receivers[0].key_frames_decoded); |
| EXPECT_EQ(stats.qp_sum, info.receivers[0].qp_sum); |
| EXPECT_EQ(stats.total_decode_time_ms, info.receivers[0].total_decode_time_ms); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| GetStatsTranslatesInterFrameDelayStatsCorrectly) { |
| FakeVideoReceiveStream* stream = AddRecvStream(); |
| webrtc::VideoReceiveStream::Stats stats; |
| stats.total_inter_frame_delay = 0.123; |
| stats.total_squared_inter_frame_delay = 0.00456; |
| stream->SetStats(stats); |
| |
| cricket::VideoMediaInfo info; |
| ASSERT_TRUE(channel_->GetStats(&info)); |
| EXPECT_EQ(stats.total_inter_frame_delay, |
| info.receivers[0].total_inter_frame_delay); |
| EXPECT_EQ(stats.total_squared_inter_frame_delay, |
| info.receivers[0].total_squared_inter_frame_delay); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, GetStatsTranslatesReceivePacketStatsCorrectly) { |
| FakeVideoReceiveStream* stream = AddRecvStream(); |
| webrtc::VideoReceiveStream::Stats stats; |
| stats.rtp_stats.packet_counter.payload_bytes = 2; |
| stats.rtp_stats.packet_counter.header_bytes = 3; |
| stats.rtp_stats.packet_counter.padding_bytes = 4; |
| stats.rtp_stats.packet_counter.packets = 5; |
| stats.rtp_stats.packets_lost = 6; |
| stream->SetStats(stats); |
| |
| cricket::VideoMediaInfo info; |
| ASSERT_TRUE(channel_->GetStats(&info)); |
| EXPECT_EQ(stats.rtp_stats.packet_counter.payload_bytes, |
| rtc::checked_cast<size_t>(info.receivers[0].payload_bytes_rcvd)); |
| EXPECT_EQ(stats.rtp_stats.packet_counter.packets, |
| rtc::checked_cast<unsigned int>(info.receivers[0].packets_rcvd)); |
| EXPECT_EQ(stats.rtp_stats.packets_lost, info.receivers[0].packets_lost); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, TranslatesCallStatsCorrectly) { |
| AddSendStream(); |
| AddSendStream(); |
| webrtc::Call::Stats stats; |
| stats.rtt_ms = 123; |
| fake_call_->SetStats(stats); |
| |
| cricket::VideoMediaInfo info; |
| ASSERT_TRUE(channel_->GetStats(&info)); |
| ASSERT_EQ(2u, info.senders.size()); |
| EXPECT_EQ(stats.rtt_ms, info.senders[0].rtt_ms); |
| EXPECT_EQ(stats.rtt_ms, info.senders[1].rtt_ms); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, TranslatesSenderBitrateStatsCorrectly) { |
| FakeVideoSendStream* stream = AddSendStream(); |
| webrtc::VideoSendStream::Stats stats; |
| stats.target_media_bitrate_bps = 156; |
| stats.media_bitrate_bps = 123; |
| stats.substreams[17].total_bitrate_bps = 1; |
| stats.substreams[17].retransmit_bitrate_bps = 2; |
| stats.substreams[42].total_bitrate_bps = 3; |
| stats.substreams[42].retransmit_bitrate_bps = 4; |
| stream->SetStats(stats); |
| |
| FakeVideoSendStream* stream2 = AddSendStream(); |
| webrtc::VideoSendStream::Stats stats2; |
| stats2.target_media_bitrate_bps = 200; |
| stats2.media_bitrate_bps = 321; |
| stats2.substreams[13].total_bitrate_bps = 5; |
| stats2.substreams[13].retransmit_bitrate_bps = 6; |
| stats2.substreams[21].total_bitrate_bps = 7; |
| stats2.substreams[21].retransmit_bitrate_bps = 8; |
| stream2->SetStats(stats2); |
| |
| cricket::VideoMediaInfo info; |
| ASSERT_TRUE(channel_->GetStats(&info)); |
| ASSERT_EQ(2u, info.aggregated_senders.size()); |
| ASSERT_EQ(4u, info.senders.size()); |
| BandwidthEstimationInfo bwe_info; |
| channel_->FillBitrateInfo(&bwe_info); |
| // Assuming stream and stream2 corresponds to senders[0] and [1] respectively |
| // is OK as std::maps are sorted and AddSendStream() gives increasing SSRCs. |
| EXPECT_EQ(stats.media_bitrate_bps, |
| info.aggregated_senders[0].nominal_bitrate); |
| EXPECT_EQ(stats2.media_bitrate_bps, |
| info.aggregated_senders[1].nominal_bitrate); |
| EXPECT_EQ(stats.target_media_bitrate_bps + stats2.target_media_bitrate_bps, |
| bwe_info.target_enc_bitrate); |
| EXPECT_EQ(stats.media_bitrate_bps + stats2.media_bitrate_bps, |
| bwe_info.actual_enc_bitrate); |
| EXPECT_EQ(1 + 3 + 5 + 7, bwe_info.transmit_bitrate) |
| << "Bandwidth stats should take all streams into account."; |
| EXPECT_EQ(2 + 4 + 6 + 8, bwe_info.retransmit_bitrate) |
| << "Bandwidth stats should take all streams into account."; |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, DefaultReceiveStreamReconfiguresToUseRtx) { |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| |
| const std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs1); |
| const std::vector<uint32_t> rtx_ssrcs = MAKE_VECTOR(kRtxSsrcs1); |
| |
| ASSERT_EQ(0u, fake_call_->GetVideoReceiveStreams().size()); |
| RtpPacket packet; |
| packet.SetSsrc(ssrcs[0]); |
| ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1); |
| |
| ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size()) |
| << "No default receive stream created."; |
| FakeVideoReceiveStream* recv_stream = fake_call_->GetVideoReceiveStreams()[0]; |
| EXPECT_EQ(0u, recv_stream->GetConfig().rtp.rtx_ssrc) |
| << "Default receive stream should not have configured RTX"; |
| |
| EXPECT_TRUE(channel_->AddRecvStream( |
| cricket::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs))); |
| ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size()) |
| << "AddRecvStream should have reconfigured, not added a new receiver."; |
| recv_stream = fake_call_->GetVideoReceiveStreams()[0]; |
| EXPECT_FALSE( |
| recv_stream->GetConfig().rtp.rtx_associated_payload_types.empty()); |
| EXPECT_TRUE(VerifyRtxReceiveAssociations(recv_stream->GetConfig())) |
| << "RTX should be mapped for all decoders/payload types."; |
| EXPECT_TRUE(HasRtxReceiveAssociation(recv_stream->GetConfig(), |
| GetEngineCodec("red").id)) |
| << "RTX should be mapped also for the RED payload type"; |
| EXPECT_EQ(rtx_ssrcs[0], recv_stream->GetConfig().rtp.rtx_ssrc); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, RejectsAddingStreamsWithMissingSsrcsForRtx) { |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| |
| const std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs1); |
| const std::vector<uint32_t> rtx_ssrcs = MAKE_VECTOR(kRtxSsrcs1); |
| |
| StreamParams sp = |
| cricket::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs); |
| sp.ssrcs = ssrcs; // Without RTXs, this is the important part. |
| |
| EXPECT_FALSE(channel_->AddSendStream(sp)); |
| EXPECT_FALSE(channel_->AddRecvStream(sp)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, RejectsAddingStreamsWithOverlappingRtxSsrcs) { |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| |
| const std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs1); |
| const std::vector<uint32_t> rtx_ssrcs = MAKE_VECTOR(kRtxSsrcs1); |
| |
| StreamParams sp = |
| cricket::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs); |
| |
| EXPECT_TRUE(channel_->AddSendStream(sp)); |
| EXPECT_TRUE(channel_->AddRecvStream(sp)); |
| |
| // The RTX SSRC is already used in previous streams, using it should fail. |
| sp = cricket::StreamParams::CreateLegacy(rtx_ssrcs[0]); |
| EXPECT_FALSE(channel_->AddSendStream(sp)); |
| EXPECT_FALSE(channel_->AddRecvStream(sp)); |
| |
| // After removing the original stream this should be fine to add (makes sure |
| // that RTX ssrcs are not forever taken). |
| EXPECT_TRUE(channel_->RemoveSendStream(ssrcs[0])); |
| EXPECT_TRUE(channel_->RemoveRecvStream(ssrcs[0])); |
| EXPECT_TRUE(channel_->AddSendStream(sp)); |
| EXPECT_TRUE(channel_->AddRecvStream(sp)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| RejectsAddingStreamsWithOverlappingSimulcastSsrcs) { |
| static const uint32_t kFirstStreamSsrcs[] = {1, 2, 3}; |
| static const uint32_t kOverlappingStreamSsrcs[] = {4, 3, 5}; |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| |
| StreamParams sp = |
| cricket::CreateSimStreamParams("cname", MAKE_VECTOR(kFirstStreamSsrcs)); |
| |
| EXPECT_TRUE(channel_->AddSendStream(sp)); |
| EXPECT_TRUE(channel_->AddRecvStream(sp)); |
| |
| // One of the SSRCs is already used in previous streams, using it should fail. |
| sp = cricket::CreateSimStreamParams("cname", |
| MAKE_VECTOR(kOverlappingStreamSsrcs)); |
| EXPECT_FALSE(channel_->AddSendStream(sp)); |
| EXPECT_FALSE(channel_->AddRecvStream(sp)); |
| |
| // After removing the original stream this should be fine to add (makes sure |
| // that RTX ssrcs are not forever taken). |
| EXPECT_TRUE(channel_->RemoveSendStream(kFirstStreamSsrcs[0])); |
| EXPECT_TRUE(channel_->RemoveRecvStream(kFirstStreamSsrcs[0])); |
| EXPECT_TRUE(channel_->AddSendStream(sp)); |
| EXPECT_TRUE(channel_->AddRecvStream(sp)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, ReportsSsrcGroupsInStats) { |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| |
| static const uint32_t kSenderSsrcs[] = {4, 7, 10}; |
| static const uint32_t kSenderRtxSsrcs[] = {5, 8, 11}; |
| |
| StreamParams sender_sp = cricket::CreateSimWithRtxStreamParams( |
| "cname", MAKE_VECTOR(kSenderSsrcs), MAKE_VECTOR(kSenderRtxSsrcs)); |
| |
| EXPECT_TRUE(channel_->AddSendStream(sender_sp)); |
| |
| static const uint32_t kReceiverSsrcs[] = {3}; |
| static const uint32_t kReceiverRtxSsrcs[] = {2}; |
| |
| StreamParams receiver_sp = cricket::CreateSimWithRtxStreamParams( |
| "cname", MAKE_VECTOR(kReceiverSsrcs), MAKE_VECTOR(kReceiverRtxSsrcs)); |
| EXPECT_TRUE(channel_->AddRecvStream(receiver_sp)); |
| |
| cricket::VideoMediaInfo info; |
| ASSERT_TRUE(channel_->GetStats(&info)); |
| |
| ASSERT_EQ(1u, info.senders.size()); |
| ASSERT_EQ(1u, info.receivers.size()); |
| |
| EXPECT_NE(sender_sp.ssrc_groups, receiver_sp.ssrc_groups); |
| EXPECT_EQ(sender_sp.ssrc_groups, info.senders[0].ssrc_groups); |
| EXPECT_EQ(receiver_sp.ssrc_groups, info.receivers[0].ssrc_groups); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, MapsReceivedPayloadTypeToCodecName) { |
| FakeVideoReceiveStream* stream = AddRecvStream(); |
| webrtc::VideoReceiveStream::Stats stats; |
| cricket::VideoMediaInfo info; |
| |
| // Report no codec name before receiving. |
| stream->SetStats(stats); |
| ASSERT_TRUE(channel_->GetStats(&info)); |
| EXPECT_STREQ("", info.receivers[0].codec_name.c_str()); |
| |
| // Report VP8 if we're receiving it. |
| stats.current_payload_type = GetEngineCodec("VP8").id; |
| stream->SetStats(stats); |
| ASSERT_TRUE(channel_->GetStats(&info)); |
| EXPECT_STREQ(kVp8CodecName, info.receivers[0].codec_name.c_str()); |
| |
| // Report no codec name for unknown playload types. |
| stats.current_payload_type = 3; |
| stream->SetStats(stats); |
| ASSERT_TRUE(channel_->GetStats(&info)); |
| EXPECT_STREQ("", info.receivers[0].codec_name.c_str()); |
| } |
| |
| // Tests that when we add a stream without SSRCs, but contains a stream_id |
| // that it is stored and its stream id is later used when the first packet |
| // arrives to properly create a receive stream with a sync label. |
| TEST_F(WebRtcVideoChannelTest, RecvUnsignaledSsrcWithSignaledStreamId) { |
| const char kSyncLabel[] = "sync_label"; |
| cricket::StreamParams unsignaled_stream; |
| unsignaled_stream.set_stream_ids({kSyncLabel}); |
| ASSERT_TRUE(channel_->AddRecvStream(unsignaled_stream)); |
| channel_->OnDemuxerCriteriaUpdatePending(); |
| channel_->OnDemuxerCriteriaUpdateComplete(); |
| rtc::Thread::Current()->ProcessMessages(0); |
| // The stream shouldn't have been created at this point because it doesn't |
| // have any SSRCs. |
| EXPECT_EQ(0u, fake_call_->GetVideoReceiveStreams().size()); |
| |
| // Create and deliver packet. |
| RtpPacket packet; |
| packet.SetSsrc(kIncomingUnsignalledSsrc); |
| ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1); |
| |
| // The stream should now be created with the appropriate sync label. |
| EXPECT_EQ(1u, fake_call_->GetVideoReceiveStreams().size()); |
| EXPECT_EQ(kSyncLabel, |
| fake_call_->GetVideoReceiveStreams()[0]->GetConfig().sync_group); |
| |
| // Reset the unsignaled stream to clear the cache. This deletes the receive |
| // stream. |
| channel_->ResetUnsignaledRecvStream(); |
| channel_->OnDemuxerCriteriaUpdatePending(); |
| EXPECT_EQ(0u, fake_call_->GetVideoReceiveStreams().size()); |
| |
| // Until the demuxer criteria has been updated, we ignore in-flight ssrcs of |
| // the recently removed unsignaled receive stream. |
| ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1); |
| EXPECT_EQ(0u, fake_call_->GetVideoReceiveStreams().size()); |
| |
| // After the demuxer criteria has been updated, we should proceed to create |
| // unsignalled receive streams. This time when a default video receive stream |
| // is created it won't have a sync_group. |
| channel_->OnDemuxerCriteriaUpdateComplete(); |
| ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1); |
| EXPECT_EQ(1u, fake_call_->GetVideoReceiveStreams().size()); |
| EXPECT_TRUE( |
| fake_call_->GetVideoReceiveStreams()[0]->GetConfig().sync_group.empty()); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| ResetUnsignaledRecvStreamDeletesAllDefaultStreams) { |
| // No receive streams to start with. |
| EXPECT_TRUE(fake_call_->GetVideoReceiveStreams().empty()); |
| |
| // Packet with unsignaled SSRC is received. |
| RtpPacket packet; |
| packet.SetSsrc(kIncomingUnsignalledSsrc); |
| ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1); |
| |
| // Default receive stream created. |
| const auto& receivers1 = fake_call_->GetVideoReceiveStreams(); |
| ASSERT_EQ(receivers1.size(), 1u); |
| EXPECT_EQ(receivers1[0]->GetConfig().rtp.remote_ssrc, |
| kIncomingUnsignalledSsrc); |
| |
| // Stream with another SSRC gets signaled. |
| channel_->ResetUnsignaledRecvStream(); |
| constexpr uint32_t kIncomingSignalledSsrc = kIncomingUnsignalledSsrc + 1; |
| ASSERT_TRUE(channel_->AddRecvStream( |
| cricket::StreamParams::CreateLegacy(kIncomingSignalledSsrc))); |
| |
| // New receiver is for the signaled stream. |
| const auto& receivers2 = fake_call_->GetVideoReceiveStreams(); |
| ASSERT_EQ(receivers2.size(), 1u); |
| EXPECT_EQ(receivers2[0]->GetConfig().rtp.remote_ssrc, kIncomingSignalledSsrc); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| RecentlyAddedSsrcsDoNotCreateUnsignalledRecvStreams) { |
| const uint32_t kSsrc1 = 1; |
| const uint32_t kSsrc2 = 2; |
| |
| // Starting point: receiving kSsrc1. |
| EXPECT_TRUE(channel_->AddRecvStream(StreamParams::CreateLegacy(kSsrc1))); |
| channel_->OnDemuxerCriteriaUpdatePending(); |
| channel_->OnDemuxerCriteriaUpdateComplete(); |
| rtc::Thread::Current()->ProcessMessages(0); |
| EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 1u); |
| |
| // If this is the only m= section the demuxer might be configure to forward |
| // all packets, regardless of ssrc, to this channel. When we go to multiple m= |
| // sections, there can thus be a window of time where packets that should |
| // never have belonged to this channel arrive anyway. |
| |
| // Emulate a second m= section being created by updating the demuxer criteria |
| // without adding any streams. |
| channel_->OnDemuxerCriteriaUpdatePending(); |
| |
| // Emulate there being in-flight packets for kSsrc1 and kSsrc2 arriving before |
| // the demuxer is updated. |
| { |
| // Receive a packet for kSsrc1. |
| RtpPacket packet; |
| packet.SetSsrc(kSsrc1); |
| ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1); |
| } |
| { |
| // Receive a packet for kSsrc2. |
| RtpPacket packet; |
| packet.SetSsrc(kSsrc2); |
| ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1); |
| } |
| |
| // No unsignaled ssrc for kSsrc2 should have been created, but kSsrc1 should |
| // arrive since it already has a stream. |
| EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 1u); |
| EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc1), 1u); |
| EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc2), 0u); |
| |
| // Signal that the demuxer update is complete. Because there are no more |
| // pending demuxer updates, receiving unknown ssrcs (kSsrc2) should again |
| // result in unsignalled receive streams being created. |
| channel_->OnDemuxerCriteriaUpdateComplete(); |
| rtc::Thread::Current()->ProcessMessages(0); |
| |
| // Receive packets for kSsrc1 and kSsrc2 again. |
| { |
| // Receive a packet for kSsrc1. |
| RtpPacket packet; |
| packet.SetSsrc(kSsrc1); |
| ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1); |
| } |
| { |
| // Receive a packet for kSsrc2. |
| RtpPacket packet; |
| packet.SetSsrc(kSsrc2); |
| ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1); |
| } |
| |
| // An unsignalled ssrc for kSsrc2 should be created and the packet counter |
| // should increase for both ssrcs. |
| EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 2u); |
| EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc1), 2u); |
| EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc2), 1u); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| RecentlyRemovedSsrcsDoNotCreateUnsignalledRecvStreams) { |
| const uint32_t kSsrc1 = 1; |
| const uint32_t kSsrc2 = 2; |
| |
| // Starting point: receiving kSsrc1 and kSsrc2. |
| EXPECT_TRUE(channel_->AddRecvStream(StreamParams::CreateLegacy(kSsrc1))); |
| EXPECT_TRUE(channel_->AddRecvStream(StreamParams::CreateLegacy(kSsrc2))); |
| channel_->OnDemuxerCriteriaUpdatePending(); |
| channel_->OnDemuxerCriteriaUpdateComplete(); |
| rtc::Thread::Current()->ProcessMessages(0); |
| EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 2u); |
| EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc1), 0u); |
| EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc2), 0u); |
| |
| // Remove kSsrc1, signal that a demuxer criteria update is pending, but not |
| // completed yet. |
| EXPECT_TRUE(channel_->RemoveRecvStream(kSsrc1)); |
| channel_->OnDemuxerCriteriaUpdatePending(); |
| |
| // We only have a receiver for kSsrc2 now. |
| EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 1u); |
| |
| // Emulate there being in-flight packets for kSsrc1 and kSsrc2 arriving before |
| // the demuxer is updated. |
| { |
| // Receive a packet for kSsrc1. |
| RtpPacket packet; |
| packet.SetSsrc(kSsrc1); |
| ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1); |
| } |
| { |
| // Receive a packet for kSsrc2. |
| RtpPacket packet; |
| packet.SetSsrc(kSsrc2); |
| ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1); |
| } |
| |
| // No unsignaled ssrc for kSsrc1 should have been created, but the packet |
| // count for kSsrc2 should increase. |
| EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 1u); |
| EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc1), 0u); |
| EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc2), 1u); |
| |
| // Signal that the demuxer update is complete. This means we should stop |
| // ignorning kSsrc1. |
| channel_->OnDemuxerCriteriaUpdateComplete(); |
| rtc::Thread::Current()->ProcessMessages(0); |
| |
| // Receive packets for kSsrc1 and kSsrc2 again. |
| { |
| // Receive a packet for kSsrc1. |
| RtpPacket packet; |
| packet.SetSsrc(kSsrc1); |
| ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1); |
| } |
| { |
| // Receive a packet for kSsrc2. |
| RtpPacket packet; |
| packet.SetSsrc(kSsrc2); |
| ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1); |
| } |
| |
| // An unsignalled ssrc for kSsrc1 should be created and the packet counter |
| // should increase for both ssrcs. |
| EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 2u); |
| EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc1), 1u); |
| EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc2), 2u); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, MultiplePendingDemuxerCriteriaUpdates) { |
| const uint32_t kSsrc = 1; |
| |
| // Starting point: receiving kSsrc. |
| EXPECT_TRUE(channel_->AddRecvStream(StreamParams::CreateLegacy(kSsrc))); |
| channel_->OnDemuxerCriteriaUpdatePending(); |
| channel_->OnDemuxerCriteriaUpdateComplete(); |
| rtc::Thread::Current()->ProcessMessages(0); |
| ASSERT_EQ(fake_call_->GetVideoReceiveStreams().size(), 1u); |
| |
| // Remove kSsrc... |
| EXPECT_TRUE(channel_->RemoveRecvStream(kSsrc)); |
| channel_->OnDemuxerCriteriaUpdatePending(); |
| EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 0u); |
| // And then add it back again, before the demuxer knows about the new |
| // criteria! |
| EXPECT_TRUE(channel_->AddRecvStream(StreamParams::CreateLegacy(kSsrc))); |
| channel_->OnDemuxerCriteriaUpdatePending(); |
| EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 1u); |
| |
| // In-flight packets should arrive because the stream was recreated, even |
| // though demuxer criteria updates are pending... |
| { |
| RtpPacket packet; |
| packet.SetSsrc(kSsrc); |
| ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1); |
| } |
| EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc), 1u); |
| |
| // Signal that the demuxer knows about the first update: the removal. |
| channel_->OnDemuxerCriteriaUpdateComplete(); |
| rtc::Thread::Current()->ProcessMessages(0); |
| |
| // This still should not prevent in-flight packets from arriving because we |
| // have a receive stream for it. |
| { |
| RtpPacket packet; |
| packet.SetSsrc(kSsrc); |
| ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1); |
| } |
| EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc), 2u); |
| |
| // Remove the kSsrc again while previous demuxer updates are still pending. |
| EXPECT_TRUE(channel_->RemoveRecvStream(kSsrc)); |
| channel_->OnDemuxerCriteriaUpdatePending(); |
| EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 0u); |
| |
| // Now the packet should be dropped and not create an unsignalled receive |
| // stream. |
| { |
| RtpPacket packet; |
| packet.SetSsrc(kSsrc); |
| ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1); |
| } |
| EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 0u); |
| EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc), 2u); |
| |
| // Signal that the demuxer knows about the second update: adding it back. |
| channel_->OnDemuxerCriteriaUpdateComplete(); |
| rtc::Thread::Current()->ProcessMessages(0); |
| |
| // The packets should continue to be dropped because removal happened after |
| // the most recently completed demuxer update. |
| { |
| RtpPacket packet; |
| packet.SetSsrc(kSsrc); |
| ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1); |
| } |
| EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 0u); |
| EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc), 2u); |
| |
| // Signal that the demuxer knows about the last update: the second removal. |
| channel_->OnDemuxerCriteriaUpdateComplete(); |
| rtc::Thread::Current()->ProcessMessages(0); |
| |
| // If packets still arrive after the demuxer knows about the latest removal we |
| // should finally create an unsignalled receive stream. |
| { |
| RtpPacket packet; |
| packet.SetSsrc(kSsrc); |
| ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1); |
| } |
| EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 1u); |
| EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc), 3u); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, UnsignalledSsrcHasACooldown) { |
| const uint32_t kSsrc1 = 1; |
| const uint32_t kSsrc2 = 2; |
| |
| // Send packets for kSsrc1, creating an unsignalled receive stream. |
| { |
| // Receive a packet for kSsrc1. |
| RtpPacket packet; |
| packet.SetSsrc(kSsrc1); |
| channel_->OnPacketReceived(packet.Buffer(), /* packet_time_us */ -1); |
| } |
| rtc::Thread::Current()->ProcessMessages(0); |
| fake_clock_.AdvanceTime( |
| webrtc::TimeDelta::Millis(kUnsignalledReceiveStreamCooldownMs - 1)); |
| |
| // We now have an unsignalled receive stream for kSsrc1. |
| EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 1u); |
| EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc1), 1u); |
| EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc2), 0u); |
| |
| { |
| // Receive a packet for kSsrc2. |
| RtpPacket packet; |
| packet.SetSsrc(kSsrc2); |
| channel_->OnPacketReceived(packet.Buffer(), /* packet_time_us */ -1); |
| } |
| rtc::Thread::Current()->ProcessMessages(0); |
| |
| // Not enough time has passed to replace the unsignalled receive stream, so |
| // the kSsrc2 should be ignored. |
| EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 1u); |
| EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc1), 1u); |
| EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc2), 0u); |
| |
| // After 500 ms, kSsrc2 should trigger a new unsignalled receive stream that |
| // replaces the old one. |
| fake_clock_.AdvanceTime(webrtc::TimeDelta::Millis(1)); |
| { |
| // Receive a packet for kSsrc2. |
| RtpPacket packet; |
| packet.SetSsrc(kSsrc2); |
| channel_->OnPacketReceived(packet.Buffer(), /* packet_time_us */ -1); |
| } |
| rtc::Thread::Current()->ProcessMessages(0); |
| |
| // The old unsignalled receive stream was destroyed and replaced, so we still |
| // only have one unsignalled receive stream. But tha packet counter for kSsrc2 |
| // has now increased. |
| EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 1u); |
| EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc1), 1u); |
| EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc2), 1u); |
| } |
| |
| // Test BaseMinimumPlayoutDelayMs on receive streams. |
| TEST_F(WebRtcVideoChannelTest, BaseMinimumPlayoutDelayMs) { |
| // Test that set won't work for non-existing receive streams. |
| EXPECT_FALSE(channel_->SetBaseMinimumPlayoutDelayMs(kSsrc + 2, 200)); |
| // Test that get won't work for non-existing receive streams. |
| EXPECT_FALSE(channel_->GetBaseMinimumPlayoutDelayMs(kSsrc + 2)); |
| |
| EXPECT_TRUE(AddRecvStream()); |
| // Test that set works for the existing receive stream. |
| EXPECT_TRUE(channel_->SetBaseMinimumPlayoutDelayMs(last_ssrc_, 200)); |
| auto* recv_stream = fake_call_->GetVideoReceiveStream(last_ssrc_); |
| EXPECT_TRUE(recv_stream); |
| EXPECT_EQ(recv_stream->base_mininum_playout_delay_ms(), 200); |
| EXPECT_EQ(channel_->GetBaseMinimumPlayoutDelayMs(last_ssrc_).value_or(0), |
| 200); |
| } |
| |
| // Test BaseMinimumPlayoutDelayMs on unsignaled receive streams. |
| TEST_F(WebRtcVideoChannelTest, BaseMinimumPlayoutDelayMsUnsignaledRecvStream) { |
| absl::optional<int> delay_ms; |
| const FakeVideoReceiveStream* recv_stream; |
| |
| // Set default stream with SSRC 0 |
| EXPECT_TRUE(channel_->SetBaseMinimumPlayoutDelayMs(0, 200)); |
| EXPECT_EQ(200, channel_->GetBaseMinimumPlayoutDelayMs(0).value_or(0)); |
| |
| // Spawn an unsignaled stream by sending a packet, it should inherit |
| // default delay 200. |
| RtpPacket packet; |
| packet.SetSsrc(kIncomingUnsignalledSsrc); |
| ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1); |
| |
| recv_stream = fake_call_->GetVideoReceiveStream(kIncomingUnsignalledSsrc); |
| EXPECT_EQ(recv_stream->base_mininum_playout_delay_ms(), 200); |
| delay_ms = channel_->GetBaseMinimumPlayoutDelayMs(kIncomingUnsignalledSsrc); |
| EXPECT_EQ(200, delay_ms.value_or(0)); |
| |
| // Check that now if we change delay for SSRC 0 it will change delay for the |
| // default receiving stream as well. |
| EXPECT_TRUE(channel_->SetBaseMinimumPlayoutDelayMs(0, 300)); |
| EXPECT_EQ(300, channel_->GetBaseMinimumPlayoutDelayMs(0).value_or(0)); |
| delay_ms = channel_->GetBaseMinimumPlayoutDelayMs(kIncomingUnsignalledSsrc); |
| EXPECT_EQ(300, delay_ms.value_or(0)); |
| recv_stream = fake_call_->GetVideoReceiveStream(kIncomingUnsignalledSsrc); |
| EXPECT_EQ(recv_stream->base_mininum_playout_delay_ms(), 300); |
| } |
| |
| void WebRtcVideoChannelTest::TestReceiveUnsignaledSsrcPacket( |
| uint8_t payload_type, |
| bool expect_created_receive_stream) { |
| // kRedRtxPayloadType must currently be unused. |
| EXPECT_FALSE(FindCodecById(engine_.recv_codecs(), kRedRtxPayloadType)); |
| |
| // Add a RED RTX codec. |
| VideoCodec red_rtx_codec = |
| VideoCodec::CreateRtxCodec(kRedRtxPayloadType, GetEngineCodec("red").id); |
| recv_parameters_.codecs.push_back(red_rtx_codec); |
| EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_)); |
| |
| ASSERT_EQ(0u, fake_call_->GetVideoReceiveStreams().size()); |
| RtpPacket packet; |
| packet.SetPayloadType(payload_type); |
| packet.SetSsrc(kIncomingUnsignalledSsrc); |
| ReceivePacketAndAdvanceTime(packet.Buffer(), /* packet_time_us */ -1); |
| |
| if (expect_created_receive_stream) { |
| EXPECT_EQ(1u, fake_call_->GetVideoReceiveStreams().size()) |
| << "Should have created a receive stream for payload type: " |
| << payload_type; |
| } else { |
| EXPECT_EQ(0u, fake_call_->GetVideoReceiveStreams().size()) |
| << "Shouldn't have created a receive stream for payload type: " |
| << payload_type; |
| } |
| } |
| |
| class WebRtcVideoChannelDiscardUnknownSsrcTest : public WebRtcVideoChannelTest { |
| public: |
| WebRtcVideoChannelDiscardUnknownSsrcTest() |
| : WebRtcVideoChannelTest( |
| "WebRTC-Video-DiscardPacketsWithUnknownSsrc/Enabled/") {} |
| }; |
| |
| TEST_F(WebRtcVideoChannelDiscardUnknownSsrcTest, NoUnsignalledStreamCreated) { |
| TestReceiveUnsignaledSsrcPacket(GetEngineCodec("VP8").id, |
| false /* expect_created_receive_stream */); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, Vp8PacketCreatesUnsignalledStream) { |
| TestReceiveUnsignaledSsrcPacket(GetEngineCodec("VP8").id, |
| true /* expect_created_receive_stream */); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, Vp9PacketCreatesUnsignalledStream) { |
| TestReceiveUnsignaledSsrcPacket(GetEngineCodec("VP9").id, |
| true /* expect_created_receive_stream */); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, RtxPacketDoesntCreateUnsignalledStream) { |
| AssignDefaultAptRtxTypes(); |
| const cricket::VideoCodec vp8 = GetEngineCodec("VP8"); |
| const int rtx_vp8_payload_type = default_apt_rtx_types_[vp8.id]; |
| TestReceiveUnsignaledSsrcPacket(rtx_vp8_payload_type, |
| false /* expect_created_receive_stream */); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, UlpfecPacketDoesntCreateUnsignalledStream) { |
| TestReceiveUnsignaledSsrcPacket(GetEngineCodec("ulpfec").id, |
| false /* expect_created_receive_stream */); |
| } |
| |
| TEST_F(WebRtcVideoChannelFlexfecRecvTest, |
| FlexfecPacketDoesntCreateUnsignalledStream) { |
| TestReceiveUnsignaledSsrcPacket(GetEngineCodec("flexfec-03").id, |
| false /* expect_created_receive_stream */); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, RedRtxPacketDoesntCreateUnsignalledStream) { |
| TestReceiveUnsignaledSsrcPacket(kRedRtxPayloadType, |
| false /* expect_created_receive_stream */); |
| } |
| |
| // Test that receiving any unsignalled SSRC works even if it changes. |
| // The first unsignalled SSRC received will create a default receive stream. |
| // Any different unsignalled SSRC received will replace the default. |
| TEST_F(WebRtcVideoChannelTest, ReceiveDifferentUnsignaledSsrc) { |
| // Allow receiving VP8, VP9, H264 (if enabled). |
| cricket::VideoRecvParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| parameters.codecs.push_back(GetEngineCodec("VP9")); |
| |
| #if defined(WEBRTC_USE_H264) |
| cricket::VideoCodec H264codec(126, "H264"); |
| parameters.codecs.push_back(H264codec); |
| #endif |
| |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| // No receive streams yet. |
| ASSERT_EQ(0u, fake_call_->GetVideoReceiveStreams().size()); |
| cricket::FakeVideoRenderer renderer; |
| channel_->SetDefaultSink(&renderer); |
| |
| // Receive VP8 packet on first SSRC. |
| RtpPacket rtp_packet; |
| rtp_packet.SetPayloadType(GetEngineCodec("VP8").id); |
| rtp_packet.SetSsrc(kIncomingUnsignalledSsrc + 1); |
| ReceivePacketAndAdvanceTime(rtp_packet.Buffer(), /* packet_time_us */ -1); |
| // VP8 packet should create default receive stream. |
| ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size()); |
| FakeVideoReceiveStream* recv_stream = fake_call_->GetVideoReceiveStreams()[0]; |
| EXPECT_EQ(rtp_packet.Ssrc(), recv_stream->GetConfig().rtp.remote_ssrc); |
| // Verify that the receive stream sinks to a renderer. |
| webrtc::VideoFrame video_frame = |
| webrtc::VideoFrame::Builder() |
| .set_video_frame_buffer(CreateBlackFrameBuffer(4, 4)) |
| .set_timestamp_rtp(100) |
| .set_timestamp_us(0) |
| .set_rotation(webrtc::kVideoRotation_0) |
| .build(); |
| recv_stream->InjectFrame(video_frame); |
| EXPECT_EQ(1, renderer.num_rendered_frames()); |
| |
| // Receive VP9 packet on second SSRC. |
| rtp_packet.SetPayloadType(GetEngineCodec("VP9").id); |
| rtp_packet.SetSsrc(kIncomingUnsignalledSsrc + 2); |
| ReceivePacketAndAdvanceTime(rtp_packet.Buffer(), /* packet_time_us */ -1); |
| // VP9 packet should replace the default receive SSRC. |
| ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size()); |
| recv_stream = fake_call_->GetVideoReceiveStreams()[0]; |
| EXPECT_EQ(rtp_packet.Ssrc(), recv_stream->GetConfig().rtp.remote_ssrc); |
| // Verify that the receive stream sinks to a renderer. |
| webrtc::VideoFrame video_frame2 = |
| webrtc::VideoFrame::Builder() |
| .set_video_frame_buffer(CreateBlackFrameBuffer(4, 4)) |
| .set_timestamp_rtp(200) |
| .set_timestamp_us(0) |
| .set_rotation(webrtc::kVideoRotation_0) |
| .build(); |
| recv_stream->InjectFrame(video_frame2); |
| EXPECT_EQ(2, renderer.num_rendered_frames()); |
| |
| #if defined(WEBRTC_USE_H264) |
| // Receive H264 packet on third SSRC. |
| rtp_packet.SetPayloadType(126); |
| rtp_packet.SetSsrc(kIncomingUnsignalledSsrc + 3); |
| ReceivePacketAndAdvanceTime(rtp_packet.Buffer(), /* packet_time_us */ -1); |
| // H264 packet should replace the default receive SSRC. |
| ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size()); |
| recv_stream = fake_call_->GetVideoReceiveStreams()[0]; |
| EXPECT_EQ(rtp_packet.Ssrc(), recv_stream->GetConfig().rtp.remote_ssrc); |
| // Verify that the receive stream sinks to a renderer. |
| webrtc::VideoFrame video_frame3 = |
| webrtc::VideoFrame::Builder() |
| .set_video_frame_buffer(CreateBlackFrameBuffer(4, 4)) |
| .set_timestamp_rtp(300) |
| .set_timestamp_us(0) |
| .set_rotation(webrtc::kVideoRotation_0) |
| .build(); |
| recv_stream->InjectFrame(video_frame3); |
| EXPECT_EQ(3, renderer.num_rendered_frames()); |
| #endif |
| } |
| |
| // This test verifies that when a new default stream is created for a new |
| // unsignaled SSRC, the new stream does not overwrite any old stream that had |
| // been the default receive stream before being properly signaled. |
| TEST_F(WebRtcVideoChannelTest, |
| NewUnsignaledStreamDoesNotDestroyPreviouslyUnsignaledStream) { |
| cricket::VideoRecvParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| ASSERT_TRUE(channel_->SetRecvParameters(parameters)); |
| |
| // No streams signaled and no packets received, so we should not have any |
| // stream objects created yet. |
| EXPECT_EQ(0u, fake_call_->GetVideoReceiveStreams().size()); |
| |
| // Receive packet on an unsignaled SSRC. |
| RtpPacket rtp_packet; |
| rtp_packet.SetPayloadType(GetEngineCodec("VP8").id); |
| rtp_packet.SetSsrc(kSsrcs3[0]); |
| ReceivePacketAndAdvanceTime(rtp_packet.Buffer(), /* packet_time_us */ -1); |
| // Default receive stream should be created. |
| ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size()); |
| FakeVideoReceiveStream* recv_stream0 = |
| fake_call_->GetVideoReceiveStreams()[0]; |
| EXPECT_EQ(kSsrcs3[0], recv_stream0->GetConfig().rtp.remote_ssrc); |
| |
| // Signal the SSRC. |
| EXPECT_TRUE( |
| channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrcs3[0]))); |
| ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size()); |
| recv_stream0 = fake_call_->GetVideoReceiveStreams()[0]; |
| EXPECT_EQ(kSsrcs3[0], recv_stream0->GetConfig().rtp.remote_ssrc); |
| |
| // Receive packet on a different unsignaled SSRC. |
| rtp_packet.SetSsrc(kSsrcs3[1]); |
| ReceivePacketAndAdvanceTime(rtp_packet.Buffer(), /* packet_time_us */ -1); |
| // New default receive stream should be created, but old stream should remain. |
| ASSERT_EQ(2u, fake_call_->GetVideoReceiveStreams().size()); |
| EXPECT_EQ(recv_stream0, fake_call_->GetVideoReceiveStreams()[0]); |
| FakeVideoReceiveStream* recv_stream1 = |
| fake_call_->GetVideoReceiveStreams()[1]; |
| EXPECT_EQ(kSsrcs3[1], recv_stream1->GetConfig().rtp.remote_ssrc); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, CanSetMaxBitrateForExistingStream) { |
| AddSendStream(); |
| |
| webrtc::test::FrameForwarder frame_forwarder; |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder)); |
| EXPECT_TRUE(channel_->SetSend(true)); |
| frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame()); |
| |
| int default_encoder_bitrate = GetMaxEncoderBitrate(); |
| EXPECT_GT(default_encoder_bitrate, 1000); |
| |
| // TODO(skvlad): Resolve the inconsistency between the interpretation |
| // of the global bitrate limit for audio and video: |
| // - Audio: max_bandwidth_bps = 0 - fail the operation, |
| // max_bandwidth_bps = -1 - remove the bandwidth limit |
| // - Video: max_bandwidth_bps = 0 - remove the bandwidth limit, |
| // max_bandwidth_bps = -1 - remove the bandwidth limit |
| |
| SetAndExpectMaxBitrate(1000, 0, 1000); |
| SetAndExpectMaxBitrate(1000, 800, 800); |
| SetAndExpectMaxBitrate(600, 800, 600); |
| SetAndExpectMaxBitrate(0, 800, 800); |
| SetAndExpectMaxBitrate(0, 0, default_encoder_bitrate); |
| |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, CannotSetMaxBitrateForNonexistentStream) { |
| webrtc::RtpParameters nonexistent_parameters = |
| channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_EQ(0u, nonexistent_parameters.encodings.size()); |
| |
| nonexistent_parameters.encodings.push_back(webrtc::RtpEncodingParameters()); |
| EXPECT_FALSE( |
| channel_->SetRtpSendParameters(last_ssrc_, nonexistent_parameters).ok()); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| SetLowMaxBitrateOverwritesVideoStreamMinBitrate) { |
| FakeVideoSendStream* stream = AddSendStream(); |
| |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_EQ(1UL, parameters.encodings.size()); |
| EXPECT_FALSE(parameters.encodings[0].max_bitrate_bps.has_value()); |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| |
| // Note that this is testing the behavior of the FakeVideoSendStream, which |
| // also calls to CreateEncoderStreams to get the VideoStreams, so essentially |
| // we are just testing the behavior of |
| // EncoderStreamFactory::CreateEncoderStreams. |
| ASSERT_EQ(1UL, stream->GetVideoStreams().size()); |
| EXPECT_EQ(webrtc::kDefaultMinVideoBitrateBps, |
| stream->GetVideoStreams()[0].min_bitrate_bps); |
| |
| // Set a low max bitrate & check that VideoStream.min_bitrate_bps is limited |
| // by this amount. |
| parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| int low_max_bitrate_bps = webrtc::kDefaultMinVideoBitrateBps - 1000; |
| parameters.encodings[0].max_bitrate_bps = low_max_bitrate_bps; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| |
| ASSERT_EQ(1UL, stream->GetVideoStreams().size()); |
| EXPECT_EQ(low_max_bitrate_bps, stream->GetVideoStreams()[0].min_bitrate_bps); |
| EXPECT_EQ(low_max_bitrate_bps, stream->GetVideoStreams()[0].max_bitrate_bps); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| SetHighMinBitrateOverwritesVideoStreamMaxBitrate) { |
| FakeVideoSendStream* stream = AddSendStream(); |
| |
| // Note that this is testing the behavior of the FakeVideoSendStream, which |
| // also calls to CreateEncoderStreams to get the VideoStreams, so essentially |
| // we are just testing the behavior of |
| // EncoderStreamFactory::CreateEncoderStreams. |
| ASSERT_EQ(1UL, stream->GetVideoStreams().size()); |
| int high_min_bitrate_bps = stream->GetVideoStreams()[0].max_bitrate_bps + 1; |
| |
| // Set a high min bitrate and check that max_bitrate_bps is adjusted up. |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_EQ(1UL, parameters.encodings.size()); |
| parameters.encodings[0].min_bitrate_bps = high_min_bitrate_bps; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| |
| ASSERT_EQ(1UL, stream->GetVideoStreams().size()); |
| EXPECT_EQ(high_min_bitrate_bps, stream->GetVideoStreams()[0].min_bitrate_bps); |
| EXPECT_EQ(high_min_bitrate_bps, stream->GetVideoStreams()[0].max_bitrate_bps); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| SetMinBitrateAboveMaxBitrateLimitAdjustsMinBitrateDown) { |
| send_parameters_.max_bandwidth_bps = 99999; |
| FakeVideoSendStream* stream = AddSendStream(); |
| ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps); |
| ASSERT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| ASSERT_EQ(1UL, stream->GetVideoStreams().size()); |
| EXPECT_EQ(webrtc::kDefaultMinVideoBitrateBps, |
| stream->GetVideoStreams()[0].min_bitrate_bps); |
| EXPECT_EQ(send_parameters_.max_bandwidth_bps, |
| stream->GetVideoStreams()[0].max_bitrate_bps); |
| |
| // Set min bitrate above global max bitrate and check that min_bitrate_bps is |
| // adjusted down. |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_EQ(1UL, parameters.encodings.size()); |
| parameters.encodings[0].min_bitrate_bps = 99999 + 1; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| ASSERT_EQ(1UL, stream->GetVideoStreams().size()); |
| EXPECT_EQ(send_parameters_.max_bandwidth_bps, |
| stream->GetVideoStreams()[0].min_bitrate_bps); |
| EXPECT_EQ(send_parameters_.max_bandwidth_bps, |
| stream->GetVideoStreams()[0].max_bitrate_bps); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetMaxFramerateOneStream) { |
| FakeVideoSendStream* stream = AddSendStream(); |
| |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_EQ(1UL, parameters.encodings.size()); |
| EXPECT_FALSE(parameters.encodings[0].max_framerate.has_value()); |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| |
| // Note that this is testing the behavior of the FakeVideoSendStream, which |
| // also calls to CreateEncoderStreams to get the VideoStreams, so essentially |
| // we are just testing the behavior of |
| // EncoderStreamFactory::CreateEncoderStreams. |
| ASSERT_EQ(1UL, stream->GetVideoStreams().size()); |
| EXPECT_EQ(kDefaultVideoMaxFramerate, |
| stream->GetVideoStreams()[0].max_framerate); |
| |
| // Set max framerate and check that VideoStream.max_framerate is set. |
| const int kNewMaxFramerate = kDefaultVideoMaxFramerate - 1; |
| parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| parameters.encodings[0].max_framerate = kNewMaxFramerate; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| |
| ASSERT_EQ(1UL, stream->GetVideoStreams().size()); |
| EXPECT_EQ(kNewMaxFramerate, stream->GetVideoStreams()[0].max_framerate); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetNumTemporalLayersForSingleStream) { |
| FakeVideoSendStream* stream = AddSendStream(); |
| |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_EQ(1UL, parameters.encodings.size()); |
| EXPECT_FALSE(parameters.encodings[0].num_temporal_layers.has_value()); |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| |
| // Note that this is testing the behavior of the FakeVideoSendStream, which |
| // also calls to CreateEncoderStreams to get the VideoStreams, so essentially |
| // we are just testing the behavior of |
| // EncoderStreamFactory::CreateEncoderStreams. |
| ASSERT_EQ(1UL, stream->GetVideoStreams().size()); |
| EXPECT_FALSE(stream->GetVideoStreams()[0].num_temporal_layers.has_value()); |
| |
| // Set temporal layers and check that VideoStream.num_temporal_layers is set. |
| parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| parameters.encodings[0].num_temporal_layers = 2; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| |
| ASSERT_EQ(1UL, stream->GetVideoStreams().size()); |
| EXPECT_EQ(2UL, stream->GetVideoStreams()[0].num_temporal_layers); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| CannotSetRtpSendParametersWithIncorrectNumberOfEncodings) { |
| AddSendStream(); |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| // Two or more encodings should result in failure. |
| parameters.encodings.push_back(webrtc::RtpEncodingParameters()); |
| EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| // Zero encodings should also fail. |
| parameters.encodings.clear(); |
| EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| CannotSetSimulcastRtpSendParametersWithIncorrectNumberOfEncodings) { |
| std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3); |
| StreamParams sp = CreateSimStreamParams("cname", ssrcs); |
| AddSendStream(sp); |
| |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| |
| // Additional encodings should result in failure. |
| parameters.encodings.push_back(webrtc::RtpEncodingParameters()); |
| EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| // Zero encodings should also fail. |
| parameters.encodings.clear(); |
| EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| } |
| |
| // Changing the SSRC through RtpParameters is not allowed. |
| TEST_F(WebRtcVideoChannelTest, CannotSetSsrcInRtpSendParameters) { |
| AddSendStream(); |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| parameters.encodings[0].ssrc = 0xdeadbeef; |
| EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| } |
| |
| // Tests that when RTCRtpEncodingParameters.bitrate_priority gets set to |
| // a value <= 0, setting the parameters returns false. |
| TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersInvalidBitratePriority) { |
| AddSendStream(); |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_EQ(1UL, parameters.encodings.size()); |
| EXPECT_EQ(webrtc::kDefaultBitratePriority, |
| parameters.encodings[0].bitrate_priority); |
| |
| parameters.encodings[0].bitrate_priority = 0; |
| EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| parameters.encodings[0].bitrate_priority = -2; |
| EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| } |
| |
| // Tests when the the RTCRtpEncodingParameters.bitrate_priority gets set |
| // properly on the VideoChannel and propogates down to the video encoder. |
| TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersPriorityOneStream) { |
| AddSendStream(); |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_EQ(1UL, parameters.encodings.size()); |
| EXPECT_EQ(webrtc::kDefaultBitratePriority, |
| parameters.encodings[0].bitrate_priority); |
| |
| // Change the value and set it on the VideoChannel. |
| double new_bitrate_priority = 2.0; |
| parameters.encodings[0].bitrate_priority = new_bitrate_priority; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| |
| // Verify that the encoding parameters bitrate_priority is set for the |
| // VideoChannel. |
| parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_EQ(1UL, parameters.encodings.size()); |
| EXPECT_EQ(new_bitrate_priority, parameters.encodings[0].bitrate_priority); |
| |
| // Verify that the new value propagated down to the encoder. |
| std::vector<FakeVideoSendStream*> video_send_streams = |
| fake_call_->GetVideoSendStreams(); |
| EXPECT_EQ(1UL, video_send_streams.size()); |
| FakeVideoSendStream* video_send_stream = video_send_streams.front(); |
| // Check that the WebRtcVideoSendStream updated the VideoEncoderConfig |
| // appropriately. |
| EXPECT_EQ(new_bitrate_priority, |
| video_send_stream->GetEncoderConfig().bitrate_priority); |
| // Check that the vector of VideoStreams also was propagated correctly. Note |
| // that this is testing the behavior of the FakeVideoSendStream, which mimics |
| // the calls to CreateEncoderStreams to get the VideoStreams. |
| EXPECT_EQ(absl::optional<double>(new_bitrate_priority), |
| video_send_stream->GetVideoStreams()[0].bitrate_priority); |
| } |
| |
| // Tests that the RTCRtpEncodingParameters.bitrate_priority is set for the |
| // VideoChannel and the value propogates to the video encoder with all simulcast |
| // streams. |
| TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersPrioritySimulcastStreams) { |
| // Create the stream params with multiple ssrcs for simulcast. |
| const size_t kNumSimulcastStreams = 3; |
| std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3); |
| StreamParams stream_params = CreateSimStreamParams("cname", ssrcs); |
| AddSendStream(stream_params); |
| uint32_t primary_ssrc = stream_params.first_ssrc(); |
| |
| // Using the FrameForwarder, we manually send a full size |
| // frame. This creates multiple VideoStreams for all simulcast layers when |
| // reconfiguring, and allows us to test this behavior. |
| webrtc::test::FrameForwarder frame_forwarder; |
| VideoOptions options; |
| EXPECT_TRUE(channel_->SetVideoSend(primary_ssrc, &options, &frame_forwarder)); |
| channel_->SetSend(true); |
| frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame( |
| 1920, 1080, webrtc::VideoRotation::kVideoRotation_0, |
| rtc::kNumMicrosecsPerSec / 30)); |
| |
| // Get and set the rtp encoding parameters. |
| webrtc::RtpParameters parameters = |
| channel_->GetRtpSendParameters(primary_ssrc); |
| EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size()); |
| EXPECT_EQ(webrtc::kDefaultBitratePriority, |
| parameters.encodings[0].bitrate_priority); |
| // Change the value and set it on the VideoChannel. |
| double new_bitrate_priority = 2.0; |
| parameters.encodings[0].bitrate_priority = new_bitrate_priority; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(primary_ssrc, parameters).ok()); |
| |
| // Verify that the encoding parameters priority is set on the VideoChannel. |
| parameters = channel_->GetRtpSendParameters(primary_ssrc); |
| EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size()); |
| EXPECT_EQ(new_bitrate_priority, parameters.encodings[0].bitrate_priority); |
| |
| // Verify that the new value propagated down to the encoder. |
| std::vector<FakeVideoSendStream*> video_send_streams = |
| fake_call_->GetVideoSendStreams(); |
| EXPECT_EQ(1UL, video_send_streams.size()); |
| FakeVideoSendStream* video_send_stream = video_send_streams.front(); |
| // Check that the WebRtcVideoSendStream updated the VideoEncoderConfig |
| // appropriately. |
| EXPECT_EQ(kNumSimulcastStreams, |
| video_send_stream->GetEncoderConfig().number_of_streams); |
| EXPECT_EQ(new_bitrate_priority, |
| video_send_stream->GetEncoderConfig().bitrate_priority); |
| // Check that the vector of VideoStreams also propagated correctly. The |
| // FakeVideoSendStream calls CreateEncoderStreams, and we are testing that |
| // these are created appropriately for the simulcast case. |
| EXPECT_EQ(kNumSimulcastStreams, video_send_stream->GetVideoStreams().size()); |
| EXPECT_EQ(absl::optional<double>(new_bitrate_priority), |
| video_send_stream->GetVideoStreams()[0].bitrate_priority); |
| // Since we are only setting bitrate priority per-sender, the other |
| // VideoStreams should have a bitrate priority of 0. |
| EXPECT_EQ(absl::nullopt, |
| video_send_stream->GetVideoStreams()[1].bitrate_priority); |
| EXPECT_EQ(absl::nullopt, |
| video_send_stream->GetVideoStreams()[2].bitrate_priority); |
| EXPECT_TRUE(channel_->SetVideoSend(primary_ssrc, nullptr, nullptr)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| GetAndSetRtpSendParametersScaleResolutionDownByVP8) { |
| VideoSendParameters parameters; |
| parameters.codecs.push_back(VideoCodec(kVp8CodecName)); |
| ASSERT_TRUE(channel_->SetSendParameters(parameters)); |
| FakeVideoSendStream* stream = SetUpSimulcast(true, false); |
| |
| webrtc::test::FrameForwarder frame_forwarder; |
| FakeFrameSource frame_source(1280, 720, rtc::kNumMicrosecsPerSec / 30); |
| |
| VideoOptions options; |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder)); |
| channel_->SetSend(true); |
| |
| // Try layers in natural order (smallest to largest). |
| { |
| auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| ASSERT_EQ(3u, rtp_parameters.encodings.size()); |
| rtp_parameters.encodings[0].scale_resolution_down_by = 4.0; |
| rtp_parameters.encodings[1].scale_resolution_down_by = 2.0; |
| rtp_parameters.encodings[2].scale_resolution_down_by = 1.0; |
| auto result = channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters); |
| ASSERT_TRUE(result.ok()); |
| |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| |
| std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams(); |
| ASSERT_EQ(3u, video_streams.size()); |
| EXPECT_EQ(320u, video_streams[0].width); |
| EXPECT_EQ(180u, video_streams[0].height); |
| EXPECT_EQ(640u, video_streams[1].width); |
| EXPECT_EQ(360u, video_streams[1].height); |
| EXPECT_EQ(1280u, video_streams[2].width); |
| EXPECT_EQ(720u, video_streams[2].height); |
| } |
| |
| // Try layers in reverse natural order (largest to smallest). |
| { |
| auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| ASSERT_EQ(3u, rtp_parameters.encodings.size()); |
| rtp_parameters.encodings[0].scale_resolution_down_by = 1.0; |
| rtp_parameters.encodings[1].scale_resolution_down_by = 2.0; |
| rtp_parameters.encodings[2].scale_resolution_down_by = 4.0; |
| auto result = channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters); |
| ASSERT_TRUE(result.ok()); |
| |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| |
| std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams(); |
| ASSERT_EQ(3u, video_streams.size()); |
| EXPECT_EQ(1280u, video_streams[0].width); |
| EXPECT_EQ(720u, video_streams[0].height); |
| EXPECT_EQ(640u, video_streams[1].width); |
| EXPECT_EQ(360u, video_streams[1].height); |
| EXPECT_EQ(320u, video_streams[2].width); |
| EXPECT_EQ(180u, video_streams[2].height); |
| } |
| |
| // Try layers in mixed order. |
| { |
| auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| ASSERT_EQ(3u, rtp_parameters.encodings.size()); |
| rtp_parameters.encodings[0].scale_resolution_down_by = 10.0; |
| rtp_parameters.encodings[1].scale_resolution_down_by = 2.0; |
| rtp_parameters.encodings[2].scale_resolution_down_by = 4.0; |
| auto result = channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters); |
| ASSERT_TRUE(result.ok()); |
| |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| |
| std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams(); |
| ASSERT_EQ(3u, video_streams.size()); |
| EXPECT_EQ(128u, video_streams[0].width); |
| EXPECT_EQ(72u, video_streams[0].height); |
| EXPECT_EQ(640u, video_streams[1].width); |
| EXPECT_EQ(360u, video_streams[1].height); |
| EXPECT_EQ(320u, video_streams[2].width); |
| EXPECT_EQ(180u, video_streams[2].height); |
| } |
| |
| // Try with a missing scale setting, defaults to 1.0 if any other is set. |
| { |
| auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| ASSERT_EQ(3u, rtp_parameters.encodings.size()); |
| rtp_parameters.encodings[0].scale_resolution_down_by = 1.0; |
| rtp_parameters.encodings[1].scale_resolution_down_by.reset(); |
| rtp_parameters.encodings[2].scale_resolution_down_by = 4.0; |
| auto result = channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters); |
| ASSERT_TRUE(result.ok()); |
| |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| |
| std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams(); |
| ASSERT_EQ(3u, video_streams.size()); |
| EXPECT_EQ(1280u, video_streams[0].width); |
| EXPECT_EQ(720u, video_streams[0].height); |
| EXPECT_EQ(1280u, video_streams[1].width); |
| EXPECT_EQ(720u, video_streams[1].height); |
| EXPECT_EQ(320u, video_streams[2].width); |
| EXPECT_EQ(180u, video_streams[2].height); |
| } |
| |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| GetAndSetRtpSendParametersScaleResolutionDownByVP8WithOddResolution) { |
| // Ensure that the top layer has width and height divisible by 2^3, |
| // so that the bottom layer has width and height divisible by 2. |
| // TODO(bugs.webrtc.org/8785): Remove this field trial when we fully trust |
| // the number of simulcast layers set by the app. |
| webrtc::test::ScopedFieldTrials field_trial( |
| "WebRTC-NormalizeSimulcastResolution/Enabled-3/"); |
| |
| // Set up WebRtcVideoChannel for 3-layer VP8 simulcast. |
| VideoSendParameters parameters; |
| parameters.codecs.push_back(VideoCodec(kVp8CodecName)); |
| ASSERT_TRUE(channel_->SetSendParameters(parameters)); |
| FakeVideoSendStream* stream = SetUpSimulcast(true, false); |
| webrtc::test::FrameForwarder frame_forwarder; |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, /*options=*/nullptr, |
| &frame_forwarder)); |
| channel_->SetSend(true); |
| |
| // Set `scale_resolution_down_by`'s. |
| auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| ASSERT_EQ(rtp_parameters.encodings.size(), 3u); |
| rtp_parameters.encodings[0].scale_resolution_down_by = 1.0; |
| rtp_parameters.encodings[1].scale_resolution_down_by = 2.0; |
| rtp_parameters.encodings[2].scale_resolution_down_by = 4.0; |
| const auto result = |
| channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters); |
| ASSERT_TRUE(result.ok()); |
| |
| // Use a capture resolution whose width and height are not divisible by 2^3. |
| // (See field trial set at the top of the test.) |
| FakeFrameSource frame_source(2007, 1207, rtc::kNumMicrosecsPerSec / 30); |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| |
| // Ensure the scaling is correct. |
| const auto video_streams = stream->GetVideoStreams(); |
| ASSERT_EQ(video_streams.size(), 3u); |
| // Ensure that we round the capture resolution down for the top layer... |
| EXPECT_EQ(video_streams[0].width, 2000u); |
| EXPECT_EQ(video_streams[0].height, 1200u); |
| EXPECT_EQ(video_streams[1].width, 1000u); |
| EXPECT_EQ(video_streams[1].height, 600u); |
| // ...and that the bottom layer has a width/height divisible by 2. |
| EXPECT_EQ(video_streams[2].width, 500u); |
| EXPECT_EQ(video_streams[2].height, 300u); |
| |
| // Tear down. |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| GetAndSetRtpSendParametersScaleResolutionDownByH264) { |
| encoder_factory_->AddSupportedVideoCodecType(kH264CodecName); |
| VideoSendParameters parameters; |
| parameters.codecs.push_back(VideoCodec(kH264CodecName)); |
| ASSERT_TRUE(channel_->SetSendParameters(parameters)); |
| FakeVideoSendStream* stream = SetUpSimulcast(true, false); |
| |
| webrtc::test::FrameForwarder frame_forwarder; |
| FakeFrameSource frame_source(1280, 720, rtc::kNumMicrosecsPerSec / 30); |
| |
| VideoOptions options; |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder)); |
| channel_->SetSend(true); |
| |
| // Try layers in natural order (smallest to largest). |
| { |
| auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| ASSERT_EQ(3u, rtp_parameters.encodings.size()); |
| rtp_parameters.encodings[0].scale_resolution_down_by = 4.0; |
| rtp_parameters.encodings[1].scale_resolution_down_by = 2.0; |
| rtp_parameters.encodings[2].scale_resolution_down_by = 1.0; |
| auto result = channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters); |
| ASSERT_TRUE(result.ok()); |
| |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| |
| std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams(); |
| ASSERT_EQ(3u, video_streams.size()); |
| EXPECT_EQ(320u, video_streams[0].width); |
| EXPECT_EQ(180u, video_streams[0].height); |
| EXPECT_EQ(640u, video_streams[1].width); |
| EXPECT_EQ(360u, video_streams[1].height); |
| EXPECT_EQ(1280u, video_streams[2].width); |
| EXPECT_EQ(720u, video_streams[2].height); |
| } |
| |
| // Try layers in reverse natural order (largest to smallest). |
| { |
| auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| ASSERT_EQ(3u, rtp_parameters.encodings.size()); |
| rtp_parameters.encodings[0].scale_resolution_down_by = 1.0; |
| rtp_parameters.encodings[1].scale_resolution_down_by = 2.0; |
| rtp_parameters.encodings[2].scale_resolution_down_by = 4.0; |
| auto result = channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters); |
| ASSERT_TRUE(result.ok()); |
| |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| |
| std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams(); |
| ASSERT_EQ(3u, video_streams.size()); |
| EXPECT_EQ(1280u, video_streams[0].width); |
| EXPECT_EQ(720u, video_streams[0].height); |
| EXPECT_EQ(640u, video_streams[1].width); |
| EXPECT_EQ(360u, video_streams[1].height); |
| EXPECT_EQ(320u, video_streams[2].width); |
| EXPECT_EQ(180u, video_streams[2].height); |
| } |
| |
| // Try layers in mixed order. |
| { |
| auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| ASSERT_EQ(3u, rtp_parameters.encodings.size()); |
| rtp_parameters.encodings[0].scale_resolution_down_by = 10.0; |
| rtp_parameters.encodings[1].scale_resolution_down_by = 2.0; |
| rtp_parameters.encodings[2].scale_resolution_down_by = 4.0; |
| auto result = channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters); |
| ASSERT_TRUE(result.ok()); |
| |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| |
| std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams(); |
| ASSERT_EQ(3u, video_streams.size()); |
| EXPECT_EQ(128u, video_streams[0].width); |
| EXPECT_EQ(72u, video_streams[0].height); |
| EXPECT_EQ(640u, video_streams[1].width); |
| EXPECT_EQ(360u, video_streams[1].height); |
| EXPECT_EQ(320u, video_streams[2].width); |
| EXPECT_EQ(180u, video_streams[2].height); |
| } |
| |
| // Try with a missing scale setting, defaults to 1.0 if any other is set. |
| { |
| auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| ASSERT_EQ(3u, rtp_parameters.encodings.size()); |
| rtp_parameters.encodings[0].scale_resolution_down_by = 1.0; |
| rtp_parameters.encodings[1].scale_resolution_down_by.reset(); |
| rtp_parameters.encodings[2].scale_resolution_down_by = 4.0; |
| auto result = channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters); |
| ASSERT_TRUE(result.ok()); |
| |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| |
| std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams(); |
| ASSERT_EQ(3u, video_streams.size()); |
| EXPECT_EQ(1280u, video_streams[0].width); |
| EXPECT_EQ(720u, video_streams[0].height); |
| EXPECT_EQ(1280u, video_streams[1].width); |
| EXPECT_EQ(720u, video_streams[1].height); |
| EXPECT_EQ(320u, video_streams[2].width); |
| EXPECT_EQ(180u, video_streams[2].height); |
| } |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| GetAndSetRtpSendParametersScaleResolutionDownByH264WithOddResolution) { |
| // Ensure that the top layer has width and height divisible by 2^3, |
| // so that the bottom layer has width and height divisible by 2. |
| // TODO(bugs.webrtc.org/8785): Remove this field trial when we fully trust |
| // the number of simulcast layers set by the app. |
| webrtc::test::ScopedFieldTrials field_trial( |
| "WebRTC-NormalizeSimulcastResolution/Enabled-3/"); |
| |
| // Set up WebRtcVideoChannel for 3-layer H264 simulcast. |
| encoder_factory_->AddSupportedVideoCodecType(kH264CodecName); |
| VideoSendParameters parameters; |
| parameters.codecs.push_back(VideoCodec(kH264CodecName)); |
| ASSERT_TRUE(channel_->SetSendParameters(parameters)); |
| FakeVideoSendStream* stream = SetUpSimulcast(true, false); |
| webrtc::test::FrameForwarder frame_forwarder; |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, /*options=*/nullptr, |
| &frame_forwarder)); |
| channel_->SetSend(true); |
| |
| // Set `scale_resolution_down_by`'s. |
| auto rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| ASSERT_EQ(rtp_parameters.encodings.size(), 3u); |
| rtp_parameters.encodings[0].scale_resolution_down_by = 1.0; |
| rtp_parameters.encodings[1].scale_resolution_down_by = 2.0; |
| rtp_parameters.encodings[2].scale_resolution_down_by = 4.0; |
| const auto result = |
| channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters); |
| ASSERT_TRUE(result.ok()); |
| |
| // Use a capture resolution whose width and height are not divisible by 2^3. |
| // (See field trial set at the top of the test.) |
| FakeFrameSource frame_source(2007, 1207, rtc::kNumMicrosecsPerSec / 30); |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| |
| // Ensure the scaling is correct. |
| const auto video_streams = stream->GetVideoStreams(); |
| ASSERT_EQ(video_streams.size(), 3u); |
| // Ensure that we round the capture resolution down for the top layer... |
| EXPECT_EQ(video_streams[0].width, 2000u); |
| EXPECT_EQ(video_streams[0].height, 1200u); |
| EXPECT_EQ(video_streams[1].width, 1000u); |
| EXPECT_EQ(video_streams[1].height, 600u); |
| // ...and that the bottom layer has a width/height divisible by 2. |
| EXPECT_EQ(video_streams[2].width, 500u); |
| EXPECT_EQ(video_streams[2].height, 300u); |
| |
| // Tear down. |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, GetAndSetRtpSendParametersMaxFramerate) { |
| const size_t kNumSimulcastStreams = 3; |
| SetUpSimulcast(true, false); |
| |
| // Get and set the rtp encoding parameters. |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size()); |
| for (const auto& encoding : parameters.encodings) { |
| EXPECT_FALSE(encoding.max_framerate); |
| } |
| |
| // Change the value and set it on the VideoChannel. |
| parameters.encodings[0].max_framerate = 10; |
| parameters.encodings[1].max_framerate = 20; |
| parameters.encodings[2].max_framerate = 25; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| |
| // Verify that the bitrates are set on the VideoChannel. |
| parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size()); |
| EXPECT_EQ(10, parameters.encodings[0].max_framerate); |
| EXPECT_EQ(20, parameters.encodings[1].max_framerate); |
| EXPECT_EQ(25, parameters.encodings[2].max_framerate); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| SetRtpSendParametersNumTemporalLayersFailsForInvalidRange) { |
| const size_t kNumSimulcastStreams = 3; |
| SetUpSimulcast(true, false); |
| |
| // Get and set the rtp encoding parameters. |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size()); |
| |
| // Num temporal layers should be in the range [1, kMaxTemporalStreams]. |
| parameters.encodings[0].num_temporal_layers = 0; |
| EXPECT_EQ(webrtc::RTCErrorType::INVALID_RANGE, |
| channel_->SetRtpSendParameters(last_ssrc_, parameters).type()); |
| parameters.encodings[0].num_temporal_layers = webrtc::kMaxTemporalStreams + 1; |
| EXPECT_EQ(webrtc::RTCErrorType::INVALID_RANGE, |
| channel_->SetRtpSendParameters(last_ssrc_, parameters).type()); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, GetAndSetRtpSendParametersNumTemporalLayers) { |
| const size_t kNumSimulcastStreams = 3; |
| SetUpSimulcast(true, false); |
| |
| // Get and set the rtp encoding parameters. |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size()); |
| for (const auto& encoding : parameters.encodings) |
| EXPECT_FALSE(encoding.num_temporal_layers); |
| |
| // Change the value and set it on the VideoChannel. |
| parameters.encodings[0].num_temporal_layers = 3; |
| parameters.encodings[1].num_temporal_layers = 3; |
| parameters.encodings[2].num_temporal_layers = 3; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| |
| // Verify that the number of temporal layers are set on the VideoChannel. |
| parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size()); |
| EXPECT_EQ(3, parameters.encodings[0].num_temporal_layers); |
| EXPECT_EQ(3, parameters.encodings[1].num_temporal_layers); |
| EXPECT_EQ(3, parameters.encodings[2].num_temporal_layers); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, NumTemporalLayersPropagatedToEncoder) { |
| const size_t kNumSimulcastStreams = 3; |
| FakeVideoSendStream* stream = SetUpSimulcast(true, false); |
| |
| // Send a full size frame so all simulcast layers are used when reconfiguring. |
| webrtc::test::FrameForwarder frame_forwarder; |
| VideoOptions options; |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder)); |
| channel_->SetSend(true); |
| frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame()); |
| |
| // Get and set the rtp encoding parameters. |
| // Change the value and set it on the VideoChannel. |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size()); |
| parameters.encodings[0].num_temporal_layers = 3; |
| parameters.encodings[1].num_temporal_layers = 2; |
| parameters.encodings[2].num_temporal_layers = 1; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| |
| // Verify that the new value is propagated down to the encoder. |
| // Check that WebRtcVideoSendStream updates VideoEncoderConfig correctly. |
| EXPECT_EQ(2, stream->num_encoder_reconfigurations()); |
| webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy(); |
| EXPECT_EQ(kNumSimulcastStreams, encoder_config.number_of_streams); |
| EXPECT_EQ(kNumSimulcastStreams, encoder_config.simulcast_layers.size()); |
| EXPECT_EQ(3UL, encoder_config.simulcast_layers[0].num_temporal_layers); |
| EXPECT_EQ(2UL, encoder_config.simulcast_layers[1].num_temporal_layers); |
| EXPECT_EQ(1UL, encoder_config.simulcast_layers[2].num_temporal_layers); |
| |
| // FakeVideoSendStream calls CreateEncoderStreams, test that the vector of |
| // VideoStreams are created appropriately for the simulcast case. |
| EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size()); |
| EXPECT_EQ(3UL, stream->GetVideoStreams()[0].num_temporal_layers); |
| EXPECT_EQ(2UL, stream->GetVideoStreams()[1].num_temporal_layers); |
| EXPECT_EQ(1UL, stream->GetVideoStreams()[2].num_temporal_layers); |
| |
| // No parameter changed, encoder should not be reconfigured. |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| EXPECT_EQ(2, stream->num_encoder_reconfigurations()); |
| |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| DefaultValuePropagatedToEncoderForUnsetNumTemporalLayers) { |
| const size_t kDefaultNumTemporalLayers = 3; |
| const size_t kNumSimulcastStreams = 3; |
| FakeVideoSendStream* stream = SetUpSimulcast(true, false); |
| |
| // Send a full size frame so all simulcast layers are used when reconfiguring. |
| webrtc::test::FrameForwarder frame_forwarder; |
| VideoOptions options; |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder)); |
| channel_->SetSend(true); |
| frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame()); |
| |
| // Change rtp encoding parameters. |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size()); |
| parameters.encodings[0].num_temporal_layers = 2; |
| parameters.encodings[2].num_temporal_layers = 1; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| |
| // Verify that no value is propagated down to the encoder. |
| webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy(); |
| EXPECT_EQ(kNumSimulcastStreams, encoder_config.number_of_streams); |
| EXPECT_EQ(kNumSimulcastStreams, encoder_config.simulcast_layers.size()); |
| EXPECT_EQ(2UL, encoder_config.simulcast_layers[0].num_temporal_layers); |
| EXPECT_FALSE(encoder_config.simulcast_layers[1].num_temporal_layers); |
| EXPECT_EQ(1UL, encoder_config.simulcast_layers[2].num_temporal_layers); |
| |
| // FakeVideoSendStream calls CreateEncoderStreams, test that the vector of |
| // VideoStreams are created appropriately for the simulcast case. |
| EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size()); |
| EXPECT_EQ(2UL, stream->GetVideoStreams()[0].num_temporal_layers); |
| EXPECT_EQ(kDefaultNumTemporalLayers, |
| stream->GetVideoStreams()[1].num_temporal_layers); |
| EXPECT_EQ(1UL, stream->GetVideoStreams()[2].num_temporal_layers); |
| |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| DefaultValuePropagatedToEncoderForUnsetFramerate) { |
| const size_t kNumSimulcastStreams = 3; |
| const std::vector<webrtc::VideoStream> kDefault = GetSimulcastBitrates720p(); |
| FakeVideoSendStream* stream = SetUpSimulcast(true, false); |
| |
| // Send a full size frame so all simulcast layers are used when reconfiguring. |
| webrtc::test::FrameForwarder frame_forwarder; |
| VideoOptions options; |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder)); |
| channel_->SetSend(true); |
| frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame()); |
| |
| // Get and set the rtp encoding parameters. |
| // Change the value and set it on the VideoChannel. |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size()); |
| parameters.encodings[0].max_framerate = 15; |
| parameters.encodings[2].max_framerate = 20; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| |
| // Verify that the new value propagated down to the encoder. |
| // Check that WebRtcVideoSendStream updates VideoEncoderConfig correctly. |
| webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy(); |
| EXPECT_EQ(kNumSimulcastStreams, encoder_config.number_of_streams); |
| EXPECT_EQ(kNumSimulcastStreams, encoder_config.simulcast_layers.size()); |
| EXPECT_EQ(15, encoder_config.simulcast_layers[0].max_framerate); |
| EXPECT_EQ(-1, encoder_config.simulcast_layers[1].max_framerate); |
| EXPECT_EQ(20, encoder_config.simulcast_layers[2].max_framerate); |
| |
| // FakeVideoSendStream calls CreateEncoderStreams, test that the vector of |
| // VideoStreams are created appropriately for the simulcast case. |
| // The maximum `max_framerate` is used, kDefaultVideoMaxFramerate: 60. |
| EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size()); |
| EXPECT_EQ(15, stream->GetVideoStreams()[0].max_framerate); |
| EXPECT_EQ(kDefaultVideoMaxFramerate, |
| stream->GetVideoStreams()[1].max_framerate); |
| EXPECT_EQ(20, stream->GetVideoStreams()[2].max_framerate); |
| |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, GetAndSetRtpSendParametersMinAndMaxBitrate) { |
| const size_t kNumSimulcastStreams = 3; |
| SetUpSimulcast(true, false); |
| |
| // Get and set the rtp encoding parameters. |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size()); |
| for (const auto& encoding : parameters.encodings) { |
| EXPECT_FALSE(encoding.min_bitrate_bps); |
| EXPECT_FALSE(encoding.max_bitrate_bps); |
| } |
| |
| // Change the value and set it on the VideoChannel. |
| parameters.encodings[0].min_bitrate_bps = 100000; |
| parameters.encodings[0].max_bitrate_bps = 200000; |
| parameters.encodings[1].min_bitrate_bps = 300000; |
| parameters.encodings[1].max_bitrate_bps = 400000; |
| parameters.encodings[2].min_bitrate_bps = 500000; |
| parameters.encodings[2].max_bitrate_bps = 600000; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| |
| // Verify that the bitrates are set on the VideoChannel. |
| parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size()); |
| EXPECT_EQ(100000, parameters.encodings[0].min_bitrate_bps); |
| EXPECT_EQ(200000, parameters.encodings[0].max_bitrate_bps); |
| EXPECT_EQ(300000, parameters.encodings[1].min_bitrate_bps); |
| EXPECT_EQ(400000, parameters.encodings[1].max_bitrate_bps); |
| EXPECT_EQ(500000, parameters.encodings[2].min_bitrate_bps); |
| EXPECT_EQ(600000, parameters.encodings[2].max_bitrate_bps); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersFailsWithIncorrectBitrate) { |
| const size_t kNumSimulcastStreams = 3; |
| SetUpSimulcast(true, false); |
| |
| // Get and set the rtp encoding parameters. |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size()); |
| |
| // Max bitrate lower than min bitrate should fail. |
| parameters.encodings[2].min_bitrate_bps = 100000; |
| parameters.encodings[2].max_bitrate_bps = 100000 - 1; |
| EXPECT_EQ(webrtc::RTCErrorType::INVALID_RANGE, |
| channel_->SetRtpSendParameters(last_ssrc_, parameters).type()); |
| } |
| |
| // Test that min and max bitrate values set via RtpParameters are correctly |
| // propagated to the underlying encoder, and that the target is set to 3/4 of |
| // the maximum (3/4 was chosen because it's similar to the simulcast defaults |
| // that are used if no min/max are specified). |
| TEST_F(WebRtcVideoChannelTest, MinAndMaxSimulcastBitratePropagatedToEncoder) { |
| const size_t kNumSimulcastStreams = 3; |
| FakeVideoSendStream* stream = SetUpSimulcast(true, false); |
| |
| // Send a full size frame so all simulcast layers are used when reconfiguring. |
| webrtc::test::FrameForwarder frame_forwarder; |
| VideoOptions options; |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder)); |
| channel_->SetSend(true); |
| frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame()); |
| |
| // Get and set the rtp encoding parameters. |
| // Change the value and set it on the VideoChannel. |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size()); |
| parameters.encodings[0].min_bitrate_bps = 100000; |
| parameters.encodings[0].max_bitrate_bps = 200000; |
| parameters.encodings[1].min_bitrate_bps = 300000; |
| parameters.encodings[1].max_bitrate_bps = 400000; |
| parameters.encodings[2].min_bitrate_bps = 500000; |
| parameters.encodings[2].max_bitrate_bps = 600000; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| |
| // Verify that the new value propagated down to the encoder. |
| // Check that WebRtcVideoSendStream updates VideoEncoderConfig correctly. |
| EXPECT_EQ(2, stream->num_encoder_reconfigurations()); |
| webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy(); |
| EXPECT_EQ(kNumSimulcastStreams, encoder_config.number_of_streams); |
| EXPECT_EQ(kNumSimulcastStreams, encoder_config.simulcast_layers.size()); |
| EXPECT_EQ(100000, encoder_config.simulcast_layers[0].min_bitrate_bps); |
| EXPECT_EQ(200000, encoder_config.simulcast_layers[0].max_bitrate_bps); |
| EXPECT_EQ(300000, encoder_config.simulcast_layers[1].min_bitrate_bps); |
| EXPECT_EQ(400000, encoder_config.simulcast_layers[1].max_bitrate_bps); |
| EXPECT_EQ(500000, encoder_config.simulcast_layers[2].min_bitrate_bps); |
| EXPECT_EQ(600000, encoder_config.simulcast_layers[2].max_bitrate_bps); |
| |
| // FakeVideoSendStream calls CreateEncoderStreams, test that the vector of |
| // VideoStreams are created appropriately for the simulcast case. |
| EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size()); |
| // Target bitrate: 200000 * 3 / 4 = 150000. |
| EXPECT_EQ(100000, stream->GetVideoStreams()[0].min_bitrate_bps); |
| EXPECT_EQ(150000, stream->GetVideoStreams()[0].target_bitrate_bps); |
| EXPECT_EQ(200000, stream->GetVideoStreams()[0].max_bitrate_bps); |
| // Target bitrate: 400000 * 3 / 4 = 300000. |
| EXPECT_EQ(300000, stream->GetVideoStreams()[1].min_bitrate_bps); |
| EXPECT_EQ(300000, stream->GetVideoStreams()[1].target_bitrate_bps); |
| EXPECT_EQ(400000, stream->GetVideoStreams()[1].max_bitrate_bps); |
| // Target bitrate: 600000 * 3 / 4 = 450000, less than min -> max. |
| EXPECT_EQ(500000, stream->GetVideoStreams()[2].min_bitrate_bps); |
| EXPECT_EQ(600000, stream->GetVideoStreams()[2].target_bitrate_bps); |
| EXPECT_EQ(600000, stream->GetVideoStreams()[2].max_bitrate_bps); |
| |
| // No parameter changed, encoder should not be reconfigured. |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| EXPECT_EQ(2, stream->num_encoder_reconfigurations()); |
| |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr)); |
| } |
| |
| // Test to only specify the min or max bitrate value for a layer via |
| // RtpParameters. The unspecified min/max and target value should be set to the |
| // simulcast default that is used if no min/max are specified. |
| TEST_F(WebRtcVideoChannelTest, MinOrMaxSimulcastBitratePropagatedToEncoder) { |
| const size_t kNumSimulcastStreams = 3; |
| const std::vector<webrtc::VideoStream> kDefault = GetSimulcastBitrates720p(); |
| FakeVideoSendStream* stream = SetUpSimulcast(true, false); |
| |
| // Send a full size frame so all simulcast layers are used when reconfiguring. |
| webrtc::test::FrameForwarder frame_forwarder; |
| VideoOptions options; |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder)); |
| channel_->SetSend(true); |
| frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame()); |
| |
| // Get and set the rtp encoding parameters. |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size()); |
| |
| // Change the value and set it on the VideoChannel. |
| // Layer 0: only configure min bitrate. |
| const int kMinBpsLayer0 = kDefault[0].min_bitrate_bps + 1; |
| parameters.encodings[0].min_bitrate_bps = kMinBpsLayer0; |
| // Layer 1: only configure max bitrate. |
| const int kMaxBpsLayer1 = kDefault[1].max_bitrate_bps - 1; |
| parameters.encodings[1].max_bitrate_bps = kMaxBpsLayer1; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| |
| // Verify that the new value propagated down to the encoder. |
| // Check that WebRtcVideoSendStream updates VideoEncoderConfig correctly. |
| webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy(); |
| EXPECT_EQ(kNumSimulcastStreams, encoder_config.number_of_streams); |
| EXPECT_EQ(kNumSimulcastStreams, encoder_config.simulcast_layers.size()); |
| EXPECT_EQ(kMinBpsLayer0, encoder_config.simulcast_layers[0].min_bitrate_bps); |
| EXPECT_EQ(-1, encoder_config.simulcast_layers[0].max_bitrate_bps); |
| EXPECT_EQ(-1, encoder_config.simulcast_layers[1].min_bitrate_bps); |
| EXPECT_EQ(kMaxBpsLayer1, encoder_config.simulcast_layers[1].max_bitrate_bps); |
| EXPECT_EQ(-1, encoder_config.simulcast_layers[2].min_bitrate_bps); |
| EXPECT_EQ(-1, encoder_config.simulcast_layers[2].max_bitrate_bps); |
| |
| // FakeVideoSendStream calls CreateEncoderStreams, test that the vector of |
| // VideoStreams are created appropriately for the simulcast case. |
| EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size()); |
| // Layer 0: min configured bitrate should overwrite min default. |
| EXPECT_EQ(kMinBpsLayer0, stream->GetVideoStreams()[0].min_bitrate_bps); |
| EXPECT_EQ(kDefault[0].target_bitrate_bps, |
| stream->GetVideoStreams()[0].target_bitrate_bps); |
| EXPECT_EQ(kDefault[0].max_bitrate_bps, |
| stream->GetVideoStreams()[0].max_bitrate_bps); |
| // Layer 1: max configured bitrate should overwrite max default. |
| // And target bitrate should be 3/4 * max bitrate or default target |
| // which is larger. |
| EXPECT_EQ(kDefault[1].min_bitrate_bps, |
| stream->GetVideoStreams()[1].min_bitrate_bps); |
| const int kTargetBpsLayer1 = |
| std::max(kDefault[1].target_bitrate_bps, kMaxBpsLayer1 * 3 / 4); |
| EXPECT_EQ(kTargetBpsLayer1, stream->GetVideoStreams()[1].target_bitrate_bps); |
| EXPECT_EQ(kMaxBpsLayer1, stream->GetVideoStreams()[1].max_bitrate_bps); |
| // Layer 2: min and max bitrate not configured, default expected. |
| EXPECT_EQ(kDefault[2].min_bitrate_bps, |
| stream->GetVideoStreams()[2].min_bitrate_bps); |
| EXPECT_EQ(kDefault[2].target_bitrate_bps, |
| stream->GetVideoStreams()[2].target_bitrate_bps); |
| EXPECT_EQ(kDefault[2].max_bitrate_bps, |
| stream->GetVideoStreams()[2].max_bitrate_bps); |
| |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr)); |
| } |
| |
| // Test that specifying the min (or max) bitrate value for a layer via |
| // RtpParameters above (or below) the simulcast default max (or min) adjusts the |
| // unspecified values accordingly. |
| TEST_F(WebRtcVideoChannelTest, SetMinAndMaxSimulcastBitrateAboveBelowDefault) { |
| const size_t kNumSimulcastStreams = 3; |
| const std::vector<webrtc::VideoStream> kDefault = GetSimulcastBitrates720p(); |
| FakeVideoSendStream* stream = SetUpSimulcast(true, false); |
| |
| // Send a full size frame so all simulcast layers are used when reconfiguring. |
| webrtc::test::FrameForwarder frame_forwarder; |
| VideoOptions options; |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder)); |
| channel_->SetSend(true); |
| frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame()); |
| |
| // Get and set the rtp encoding parameters. |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size()); |
| |
| // Change the value and set it on the VideoChannel. |
| // For layer 0, set the min bitrate above the default max. |
| const int kMinBpsLayer0 = kDefault[0].max_bitrate_bps + 1; |
| parameters.encodings[0].min_bitrate_bps = kMinBpsLayer0; |
| // For layer 1, set the max bitrate below the default min. |
| const int kMaxBpsLayer1 = kDefault[1].min_bitrate_bps - 1; |
| parameters.encodings[1].max_bitrate_bps = kMaxBpsLayer1; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| |
| // Verify that the new value propagated down to the encoder. |
| // FakeVideoSendStream calls CreateEncoderStreams, test that the vector of |
| // VideoStreams are created appropriately for the simulcast case. |
| EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size()); |
| // Layer 0: Min bitrate above default max (target/max should be adjusted). |
| EXPECT_EQ(kMinBpsLayer0, stream->GetVideoStreams()[0].min_bitrate_bps); |
| EXPECT_EQ(kMinBpsLayer0, stream->GetVideoStreams()[0].target_bitrate_bps); |
| EXPECT_EQ(kMinBpsLayer0, stream->GetVideoStreams()[0].max_bitrate_bps); |
| // Layer 1: Max bitrate below default min (min/target should be adjusted). |
| EXPECT_EQ(kMaxBpsLayer1, stream->GetVideoStreams()[1].min_bitrate_bps); |
| EXPECT_EQ(kMaxBpsLayer1, stream->GetVideoStreams()[1].target_bitrate_bps); |
| EXPECT_EQ(kMaxBpsLayer1, stream->GetVideoStreams()[1].max_bitrate_bps); |
| // Layer 2: min and max bitrate not configured, default expected. |
| EXPECT_EQ(kDefault[2].min_bitrate_bps, |
| stream->GetVideoStreams()[2].min_bitrate_bps); |
| EXPECT_EQ(kDefault[2].target_bitrate_bps, |
| stream->GetVideoStreams()[2].target_bitrate_bps); |
| EXPECT_EQ(kDefault[2].max_bitrate_bps, |
| stream->GetVideoStreams()[2].max_bitrate_bps); |
| |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, BandwidthAboveTotalMaxBitrateGivenToMaxLayer) { |
| const size_t kNumSimulcastStreams = 3; |
| const std::vector<webrtc::VideoStream> kDefault = GetSimulcastBitrates720p(); |
| FakeVideoSendStream* stream = SetUpSimulcast(true, false); |
| |
| // Send a full size frame so all simulcast layers are used when reconfiguring. |
| webrtc::test::FrameForwarder frame_forwarder; |
| VideoOptions options; |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder)); |
| channel_->SetSend(true); |
| frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame()); |
| |
| // Set max bitrate for all but the highest layer. |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size()); |
| parameters.encodings[0].max_bitrate_bps = kDefault[0].max_bitrate_bps; |
| parameters.encodings[1].max_bitrate_bps = kDefault[1].max_bitrate_bps; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| |
| // Set max bandwidth equal to total max bitrate. |
| send_parameters_.max_bandwidth_bps = |
| GetTotalMaxBitrate(stream->GetVideoStreams()).bps(); |
| ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps); |
| ASSERT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| |
| // No bitrate above the total max to give to the highest layer. |
| EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size()); |
| EXPECT_EQ(kDefault[2].max_bitrate_bps, |
| stream->GetVideoStreams()[2].max_bitrate_bps); |
| |
| // Set max bandwidth above the total max bitrate. |
| send_parameters_.max_bandwidth_bps = |
| GetTotalMaxBitrate(stream->GetVideoStreams()).bps() + 1; |
| ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps); |
| ASSERT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| |
| // The highest layer has no max bitrate set -> the bitrate above the total |
| // max should be given to the highest layer. |
| EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size()); |
| EXPECT_EQ(send_parameters_.max_bandwidth_bps, |
| GetTotalMaxBitrate(stream->GetVideoStreams()).bps()); |
| EXPECT_EQ(kDefault[2].max_bitrate_bps + 1, |
| stream->GetVideoStreams()[2].max_bitrate_bps); |
| |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| BandwidthAboveTotalMaxBitrateNotGivenToMaxLayerIfMaxBitrateSet) { |
| const size_t kNumSimulcastStreams = 3; |
| const std::vector<webrtc::VideoStream> kDefault = GetSimulcastBitrates720p(); |
| EXPECT_EQ(kNumSimulcastStreams, kDefault.size()); |
| FakeVideoSendStream* stream = SetUpSimulcast(true, false); |
| |
| // Send a full size frame so all simulcast layers are used when reconfiguring. |
| webrtc::test::FrameForwarder frame_forwarder; |
| VideoOptions options; |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder)); |
| channel_->SetSend(true); |
| frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame()); |
| |
| // Set max bitrate for the highest layer. |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size()); |
| parameters.encodings[2].max_bitrate_bps = kDefault[2].max_bitrate_bps; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| |
| // Set max bandwidth above the total max bitrate. |
| send_parameters_.max_bandwidth_bps = |
| GetTotalMaxBitrate(stream->GetVideoStreams()).bps() + 1; |
| ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps); |
| ASSERT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| |
| // The highest layer has the max bitrate set -> the bitrate above the total |
| // max should not be given to the highest layer. |
| EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size()); |
| EXPECT_EQ(*parameters.encodings[2].max_bitrate_bps, |
| stream->GetVideoStreams()[2].max_bitrate_bps); |
| |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr)); |
| } |
| |
| // Test that min and max bitrate values set via RtpParameters are correctly |
| // propagated to the underlying encoder for a single stream. |
| TEST_F(WebRtcVideoChannelTest, MinAndMaxBitratePropagatedToEncoder) { |
| FakeVideoSendStream* stream = AddSendStream(); |
| EXPECT_TRUE(channel_->SetSend(true)); |
| EXPECT_TRUE(stream->IsSending()); |
| |
| // Set min and max bitrate. |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_EQ(1u, parameters.encodings.size()); |
| parameters.encodings[0].min_bitrate_bps = 80000; |
| parameters.encodings[0].max_bitrate_bps = 150000; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| |
| // Check that WebRtcVideoSendStream updates VideoEncoderConfig correctly. |
| webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy(); |
| EXPECT_EQ(1u, encoder_config.number_of_streams); |
| EXPECT_EQ(1u, encoder_config.simulcast_layers.size()); |
| EXPECT_EQ(80000, encoder_config.simulcast_layers[0].min_bitrate_bps); |
| EXPECT_EQ(150000, encoder_config.simulcast_layers[0].max_bitrate_bps); |
| |
| // FakeVideoSendStream calls CreateEncoderStreams, test that the vector of |
| // VideoStreams are created appropriately. |
| EXPECT_EQ(1u, stream->GetVideoStreams().size()); |
| EXPECT_EQ(80000, stream->GetVideoStreams()[0].min_bitrate_bps); |
| EXPECT_EQ(150000, stream->GetVideoStreams()[0].target_bitrate_bps); |
| EXPECT_EQ(150000, stream->GetVideoStreams()[0].max_bitrate_bps); |
| } |
| |
| // Test the default min and max bitrate value are correctly propagated to the |
| // underlying encoder for a single stream (when the values are not set via |
| // RtpParameters). |
| TEST_F(WebRtcVideoChannelTest, DefaultMinAndMaxBitratePropagatedToEncoder) { |
| FakeVideoSendStream* stream = AddSendStream(); |
| EXPECT_TRUE(channel_->SetSend(true)); |
| EXPECT_TRUE(stream->IsSending()); |
| |
| // Check that WebRtcVideoSendStream updates VideoEncoderConfig correctly. |
| webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy(); |
| EXPECT_EQ(1u, encoder_config.number_of_streams); |
| EXPECT_EQ(1u, encoder_config.simulcast_layers.size()); |
| EXPECT_EQ(-1, encoder_config.simulcast_layers[0].min_bitrate_bps); |
| EXPECT_EQ(-1, encoder_config.simulcast_layers[0].max_bitrate_bps); |
| |
| // FakeVideoSendStream calls CreateEncoderStreams, test that the vector of |
| // VideoStreams are created appropriately. |
| EXPECT_EQ(1u, stream->GetVideoStreams().size()); |
| EXPECT_EQ(webrtc::kDefaultMinVideoBitrateBps, |
| stream->GetVideoStreams()[0].min_bitrate_bps); |
| EXPECT_GT(stream->GetVideoStreams()[0].max_bitrate_bps, |
| stream->GetVideoStreams()[0].min_bitrate_bps); |
| EXPECT_EQ(stream->GetVideoStreams()[0].max_bitrate_bps, |
| stream->GetVideoStreams()[0].target_bitrate_bps); |
| } |
| |
| // Test that a stream will not be sending if its encoding is made inactive |
| // through SetRtpSendParameters. |
| TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersOneEncodingActive) { |
| FakeVideoSendStream* stream = AddSendStream(); |
| EXPECT_TRUE(channel_->SetSend(true)); |
| EXPECT_TRUE(stream->IsSending()); |
| |
| // Get current parameters and change "active" to false. |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| ASSERT_EQ(1u, parameters.encodings.size()); |
| ASSERT_TRUE(parameters.encodings[0].active); |
| parameters.encodings[0].active = false; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| EXPECT_FALSE(stream->IsSending()); |
| |
| // Now change it back to active and verify we resume sending. |
| parameters.encodings[0].active = true; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| EXPECT_TRUE(stream->IsSending()); |
| } |
| |
| // Tests that when active is updated for any simulcast layer then the send |
| // stream's sending state will be updated and it will be reconfigured with the |
| // new appropriate active simulcast streams. |
| TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersMultipleEncodingsActive) { |
| // Create the stream params with multiple ssrcs for simulcast. |
| const size_t kNumSimulcastStreams = 3; |
| std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3); |
| StreamParams stream_params = CreateSimStreamParams("cname", ssrcs); |
| FakeVideoSendStream* fake_video_send_stream = AddSendStream(stream_params); |
| uint32_t primary_ssrc = stream_params.first_ssrc(); |
| |
| // Using the FrameForwarder, we manually send a full size |
| // frame. This allows us to test that ReconfigureEncoder is called |
| // appropriately. |
| webrtc::test::FrameForwarder frame_forwarder; |
| VideoOptions options; |
| EXPECT_TRUE(channel_->SetVideoSend(primary_ssrc, &options, &frame_forwarder)); |
| channel_->SetSend(true); |
| frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame( |
| 1920, 1080, webrtc::VideoRotation::kVideoRotation_0, |
| rtc::kNumMicrosecsPerSec / 30)); |
| |
| // Check that all encodings are initially active. |
| webrtc::RtpParameters parameters = |
| channel_->GetRtpSendParameters(primary_ssrc); |
| EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size()); |
| EXPECT_TRUE(parameters.encodings[0].active); |
| EXPECT_TRUE(parameters.encodings[1].active); |
| EXPECT_TRUE(parameters.encodings[2].active); |
| EXPECT_TRUE(fake_video_send_stream->IsSending()); |
| |
| // Only turn on only the middle stream. |
| parameters.encodings[0].active = false; |
| parameters.encodings[1].active = true; |
| parameters.encodings[2].active = false; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(primary_ssrc, parameters).ok()); |
| // Verify that the active fields are set on the VideoChannel. |
| parameters = channel_->GetRtpSendParameters(primary_ssrc); |
| EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size()); |
| EXPECT_FALSE(parameters.encodings[0].active); |
| EXPECT_TRUE(parameters.encodings[1].active); |
| EXPECT_FALSE(parameters.encodings[2].active); |
| // Check that the VideoSendStream is updated appropriately. This means its |
| // send state was updated and it was reconfigured. |
| EXPECT_TRUE(fake_video_send_stream->IsSending()); |
| std::vector<webrtc::VideoStream> simulcast_streams = |
| fake_video_send_stream->GetVideoStreams(); |
| EXPECT_EQ(kNumSimulcastStreams, simulcast_streams.size()); |
| EXPECT_FALSE(simulcast_streams[0].active); |
| EXPECT_TRUE(simulcast_streams[1].active); |
| EXPECT_FALSE(simulcast_streams[2].active); |
| |
| // Turn off all streams. |
| parameters.encodings[0].active = false; |
| parameters.encodings[1].active = false; |
| parameters.encodings[2].active = false; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(primary_ssrc, parameters).ok()); |
| // Verify that the active fields are set on the VideoChannel. |
| parameters = channel_->GetRtpSendParameters(primary_ssrc); |
| EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size()); |
| EXPECT_FALSE(parameters.encodings[0].active); |
| EXPECT_FALSE(parameters.encodings[1].active); |
| EXPECT_FALSE(parameters.encodings[2].active); |
| // Check that the VideoSendStream is off. |
| EXPECT_FALSE(fake_video_send_stream->IsSending()); |
| simulcast_streams = fake_video_send_stream->GetVideoStreams(); |
| EXPECT_EQ(kNumSimulcastStreams, simulcast_streams.size()); |
| EXPECT_FALSE(simulcast_streams[0].active); |
| EXPECT_FALSE(simulcast_streams[1].active); |
| EXPECT_FALSE(simulcast_streams[2].active); |
| |
| EXPECT_TRUE(channel_->SetVideoSend(primary_ssrc, nullptr, nullptr)); |
| } |
| |
| // Tests that when some streams are disactivated then the lowest |
| // stream min_bitrate would be reused for the first active stream. |
| TEST_F(WebRtcVideoChannelTest, |
| SetRtpSendParametersSetsMinBitrateForFirstActiveStream) { |
| // Create the stream params with multiple ssrcs for simulcast. |
| const size_t kNumSimulcastStreams = 3; |
| std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3); |
| StreamParams stream_params = CreateSimStreamParams("cname", ssrcs); |
| FakeVideoSendStream* fake_video_send_stream = AddSendStream(stream_params); |
| uint32_t primary_ssrc = stream_params.first_ssrc(); |
| |
| // Using the FrameForwarder, we manually send a full size |
| // frame. This allows us to test that ReconfigureEncoder is called |
| // appropriately. |
| webrtc::test::FrameForwarder frame_forwarder; |
| VideoOptions options; |
| EXPECT_TRUE(channel_->SetVideoSend(primary_ssrc, &options, &frame_forwarder)); |
| channel_->SetSend(true); |
| frame_forwarder.IncomingCapturedFrame(frame_source_.GetFrame( |
| 1920, 1080, webrtc::VideoRotation::kVideoRotation_0, |
| rtc::kNumMicrosecsPerSec / 30)); |
| |
| // Check that all encodings are initially active. |
| webrtc::RtpParameters parameters = |
| channel_->GetRtpSendParameters(primary_ssrc); |
| EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size()); |
| EXPECT_TRUE(parameters.encodings[0].active); |
| EXPECT_TRUE(parameters.encodings[1].active); |
| EXPECT_TRUE(parameters.encodings[2].active); |
| EXPECT_TRUE(fake_video_send_stream->IsSending()); |
| |
| // Only turn on the highest stream. |
| parameters.encodings[0].active = false; |
| parameters.encodings[1].active = false; |
| parameters.encodings[2].active = true; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(primary_ssrc, parameters).ok()); |
| |
| // Check that the VideoSendStream is updated appropriately. This means its |
| // send state was updated and it was reconfigured. |
| EXPECT_TRUE(fake_video_send_stream->IsSending()); |
| std::vector<webrtc::VideoStream> simulcast_streams = |
| fake_video_send_stream->GetVideoStreams(); |
| EXPECT_EQ(kNumSimulcastStreams, simulcast_streams.size()); |
| EXPECT_FALSE(simulcast_streams[0].active); |
| EXPECT_FALSE(simulcast_streams[1].active); |
| EXPECT_TRUE(simulcast_streams[2].active); |
| |
| EXPECT_EQ(simulcast_streams[2].min_bitrate_bps, |
| simulcast_streams[0].min_bitrate_bps); |
| |
| EXPECT_TRUE(channel_->SetVideoSend(primary_ssrc, nullptr, nullptr)); |
| } |
| |
| // Test that if a stream is reconfigured (due to a codec change or other |
| // change) while its encoding is still inactive, it doesn't start sending. |
| TEST_F(WebRtcVideoChannelTest, |
| InactiveStreamDoesntStartSendingWhenReconfigured) { |
| // Set an initial codec list, which will be modified later. |
| cricket::VideoSendParameters parameters1; |
| parameters1.codecs.push_back(GetEngineCodec("VP8")); |
| parameters1.codecs.push_back(GetEngineCodec("VP9")); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters1)); |
| |
| FakeVideoSendStream* stream = AddSendStream(); |
| EXPECT_TRUE(channel_->SetSend(true)); |
| EXPECT_TRUE(stream->IsSending()); |
| |
| // Get current parameters and change "active" to false. |
| webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_); |
| ASSERT_EQ(1u, parameters.encodings.size()); |
| ASSERT_TRUE(parameters.encodings[0].active); |
| parameters.encodings[0].active = false; |
| EXPECT_EQ(1u, GetFakeSendStreams().size()); |
| EXPECT_EQ(1, fake_call_->GetNumCreatedSendStreams()); |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok()); |
| EXPECT_FALSE(stream->IsSending()); |
| |
| // Reorder the codec list, causing the stream to be reconfigured. |
| cricket::VideoSendParameters parameters2; |
| parameters2.codecs.push_back(GetEngineCodec("VP9")); |
| parameters2.codecs.push_back(GetEngineCodec("VP8")); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters2)); |
| auto new_streams = GetFakeSendStreams(); |
| // Assert that a new underlying stream was created due to the codec change. |
| // Otherwise, this test isn't testing what it set out to test. |
| EXPECT_EQ(1u, GetFakeSendStreams().size()); |
| EXPECT_EQ(2, fake_call_->GetNumCreatedSendStreams()); |
| |
| // Verify that we still are not sending anything, due to the inactive |
| // encoding. |
| EXPECT_FALSE(new_streams[0]->IsSending()); |
| } |
| |
| // Test that GetRtpSendParameters returns the currently configured codecs. |
| TEST_F(WebRtcVideoChannelTest, GetRtpSendParametersCodecs) { |
| AddSendStream(); |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| parameters.codecs.push_back(GetEngineCodec("VP9")); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| |
| webrtc::RtpParameters rtp_parameters = |
| channel_->GetRtpSendParameters(last_ssrc_); |
| ASSERT_EQ(2u, rtp_parameters.codecs.size()); |
| EXPECT_EQ(GetEngineCodec("VP8").ToCodecParameters(), |
| rtp_parameters.codecs[0]); |
| EXPECT_EQ(GetEngineCodec("VP9").ToCodecParameters(), |
| rtp_parameters.codecs[1]); |
| } |
| |
| // Test that GetRtpSendParameters returns the currently configured RTCP CNAME. |
| TEST_F(WebRtcVideoChannelTest, GetRtpSendParametersRtcpCname) { |
| StreamParams params = StreamParams::CreateLegacy(kSsrc); |
| params.cname = "rtcpcname"; |
| AddSendStream(params); |
| |
| webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrc); |
| EXPECT_STREQ("rtcpcname", rtp_parameters.rtcp.cname.c_str()); |
| } |
| |
| // Test that RtpParameters for send stream has one encoding and it has |
| // the correct SSRC. |
| TEST_F(WebRtcVideoChannelTest, GetRtpSendParametersSsrc) { |
| AddSendStream(); |
| |
| webrtc::RtpParameters rtp_parameters = |
| channel_->GetRtpSendParameters(last_ssrc_); |
| ASSERT_EQ(1u, rtp_parameters.encodings.size()); |
| EXPECT_EQ(last_ssrc_, rtp_parameters.encodings[0].ssrc); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, DetectRtpSendParameterHeaderExtensionsChange) { |
| AddSendStream(); |
| |
| webrtc::RtpParameters rtp_parameters = |
| channel_->GetRtpSendParameters(last_ssrc_); |
| rtp_parameters.header_extensions.emplace_back(); |
| |
| EXPECT_NE(0u, rtp_parameters.header_extensions.size()); |
| |
| webrtc::RTCError result = |
| channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters); |
| EXPECT_EQ(webrtc::RTCErrorType::INVALID_MODIFICATION, result.type()); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, GetRtpSendParametersDegradationPreference) { |
| AddSendStream(); |
| |
| webrtc::test::FrameForwarder frame_forwarder; |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &frame_forwarder)); |
| |
| webrtc::RtpParameters rtp_parameters = |
| channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_FALSE(rtp_parameters.degradation_preference.has_value()); |
| rtp_parameters.degradation_preference = |
| webrtc::DegradationPreference::MAINTAIN_FRAMERATE; |
| |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters).ok()); |
| |
| webrtc::RtpParameters updated_rtp_parameters = |
| channel_->GetRtpSendParameters(last_ssrc_); |
| EXPECT_EQ(updated_rtp_parameters.degradation_preference, |
| webrtc::DegradationPreference::MAINTAIN_FRAMERATE); |
| |
| // Remove the source since it will be destroyed before the channel |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr)); |
| } |
| |
| // Test that if we set/get parameters multiple times, we get the same results. |
| TEST_F(WebRtcVideoChannelTest, SetAndGetRtpSendParameters) { |
| AddSendStream(); |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| parameters.codecs.push_back(GetEngineCodec("VP9")); |
| EXPECT_TRUE(channel_->SetSendParameters(parameters)); |
| |
| webrtc::RtpParameters initial_params = |
| channel_->GetRtpSendParameters(last_ssrc_); |
| |
| // We should be able to set the params we just got. |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, initial_params).ok()); |
| |
| // ... And this shouldn't change the params returned by GetRtpSendParameters. |
| EXPECT_EQ(initial_params, channel_->GetRtpSendParameters(last_ssrc_)); |
| } |
| |
| // Test that GetRtpReceiveParameters returns the currently configured codecs. |
| TEST_F(WebRtcVideoChannelTest, GetRtpReceiveParametersCodecs) { |
| AddRecvStream(); |
| cricket::VideoRecvParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| parameters.codecs.push_back(GetEngineCodec("VP9")); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| |
| webrtc::RtpParameters rtp_parameters = |
| channel_->GetRtpReceiveParameters(last_ssrc_); |
| ASSERT_EQ(2u, rtp_parameters.codecs.size()); |
| EXPECT_EQ(GetEngineCodec("VP8").ToCodecParameters(), |
| rtp_parameters.codecs[0]); |
| EXPECT_EQ(GetEngineCodec("VP9").ToCodecParameters(), |
| rtp_parameters.codecs[1]); |
| } |
| |
| #if defined(WEBRTC_USE_H264) |
| TEST_F(WebRtcVideoChannelTest, GetRtpReceiveFmtpSprop) { |
| #else |
| TEST_F(WebRtcVideoChannelTest, DISABLED_GetRtpReceiveFmtpSprop) { |
| #endif |
| cricket::VideoRecvParameters parameters; |
| cricket::VideoCodec kH264sprop1(101, "H264"); |
| kH264sprop1.SetParam(kH264FmtpSpropParameterSets, "uvw"); |
| parameters.codecs.push_back(kH264sprop1); |
| cricket::VideoCodec kH264sprop2(102, "H264"); |
| kH264sprop2.SetParam(kH264FmtpSpropParameterSets, "xyz"); |
| parameters.codecs.push_back(kH264sprop2); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| |
| FakeVideoReceiveStream* recv_stream = AddRecvStream(); |
| const webrtc::VideoReceiveStream::Config& cfg = recv_stream->GetConfig(); |
| webrtc::RtpParameters rtp_parameters = |
| channel_->GetRtpReceiveParameters(last_ssrc_); |
| ASSERT_EQ(2u, rtp_parameters.codecs.size()); |
| EXPECT_EQ(kH264sprop1.ToCodecParameters(), rtp_parameters.codecs[0]); |
| ASSERT_EQ(2u, cfg.decoders.size()); |
| EXPECT_EQ(101, cfg.decoders[0].payload_type); |
| EXPECT_EQ("H264", cfg.decoders[0].video_format.name); |
| const auto it0 = |
| cfg.decoders[0].video_format.parameters.find(kH264FmtpSpropParameterSets); |
| ASSERT_TRUE(it0 != cfg.decoders[0].video_format.parameters.end()); |
| EXPECT_EQ("uvw", it0->second); |
| |
| EXPECT_EQ(102, cfg.decoders[1].payload_type); |
| EXPECT_EQ("H264", cfg.decoders[1].video_format.name); |
| const auto it1 = |
| cfg.decoders[1].video_format.parameters.find(kH264FmtpSpropParameterSets); |
| ASSERT_TRUE(it1 != cfg.decoders[1].video_format.parameters.end()); |
| EXPECT_EQ("xyz", it1->second); |
| } |
| |
| // Test that RtpParameters for receive stream has one encoding and it has |
| // the correct SSRC. |
| TEST_F(WebRtcVideoChannelTest, GetRtpReceiveParametersSsrc) { |
| AddRecvStream(); |
| |
| webrtc::RtpParameters rtp_parameters = |
| channel_->GetRtpReceiveParameters(last_ssrc_); |
| ASSERT_EQ(1u, rtp_parameters.encodings.size()); |
| EXPECT_EQ(last_ssrc_, rtp_parameters.encodings[0].ssrc); |
| } |
| |
| // Test that if we set/get parameters multiple times, we get the same results. |
| TEST_F(WebRtcVideoChannelTest, SetAndGetRtpReceiveParameters) { |
| AddRecvStream(); |
| cricket::VideoRecvParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| parameters.codecs.push_back(GetEngineCodec("VP9")); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| |
| webrtc::RtpParameters initial_params = |
| channel_->GetRtpReceiveParameters(last_ssrc_); |
| |
| // ... And this shouldn't change the params returned by |
| // GetRtpReceiveParameters. |
| EXPECT_EQ(initial_params, channel_->GetRtpReceiveParameters(last_ssrc_)); |
| } |
| |
| // Test that GetDefaultRtpReceiveParameters returns parameters correctly when |
| // SSRCs aren't signaled. It should always return an empty |
| // "RtpEncodingParameters", even after a packet is received and the unsignaled |
| // SSRC is known. |
| TEST_F(WebRtcVideoChannelTest, |
| GetDefaultRtpReceiveParametersWithUnsignaledSsrc) { |
| // Call necessary methods to configure receiving a default stream as |
| // soon as it arrives. |
| cricket::VideoRecvParameters parameters; |
| parameters.codecs.push_back(GetEngineCodec("VP8")); |
| parameters.codecs.push_back(GetEngineCodec("VP9")); |
| EXPECT_TRUE(channel_->SetRecvParameters(parameters)); |
| |
| // Call GetRtpReceiveParameters before configured to receive an unsignaled |
| // stream. Should return nothing. |
| EXPECT_EQ(webrtc::RtpParameters(), |
| channel_->GetDefaultRtpReceiveParameters()); |
| |
| // Set a sink for an unsignaled stream. |
| cricket::FakeVideoRenderer renderer; |
| channel_->SetDefaultSink(&renderer); |
| |
| // Call GetDefaultRtpReceiveParameters before the SSRC is known. |
| webrtc::RtpParameters rtp_parameters = |
| channel_->GetDefaultRtpReceiveParameters(); |
| ASSERT_EQ(1u, rtp_parameters.encodings.size()); |
| EXPECT_FALSE(rtp_parameters.encodings[0].ssrc); |
| |
| // Receive VP8 packet. |
| RtpPacket rtp_packet; |
| rtp_packet.SetPayloadType(GetEngineCodec("VP8").id); |
| rtp_packet.SetSsrc(kIncomingUnsignalledSsrc); |
| ReceivePacketAndAdvanceTime(rtp_packet.Buffer(), /* packet_time_us */ -1); |
| |
| // The `ssrc` member should still be unset. |
| rtp_parameters = channel_->GetDefaultRtpReceiveParameters(); |
| ASSERT_EQ(1u, rtp_parameters.encodings.size()); |
| EXPECT_FALSE(rtp_parameters.encodings[0].ssrc); |
| } |
| |
| // Test that if a default stream is created for a non-primary stream (for |
| // example, RTX before we know it's RTX), we are still able to explicitly add |
| // the stream later. |
| TEST_F(WebRtcVideoChannelTest, |
| AddReceiveStreamAfterReceivingNonPrimaryUnsignaledSsrc) { |
| // Receive VP8 RTX packet. |
| RtpPacket rtp_packet; |
| const cricket::VideoCodec vp8 = GetEngineCodec("VP8"); |
| rtp_packet.SetPayloadType(default_apt_rtx_types_[vp8.id]); |
| rtp_packet.SetSsrc(2); |
| ReceivePacketAndAdvanceTime(rtp_packet.Buffer(), /* packet_time_us */ -1); |
| EXPECT_EQ(1u, fake_call_->GetVideoReceiveStreams().size()); |
| |
| cricket::StreamParams params = cricket::StreamParams::CreateLegacy(1); |
| params.AddFidSsrc(1, 2); |
| EXPECT_TRUE(channel_->AddRecvStream(params)); |
| } |
| |
| void WebRtcVideoChannelTest::TestReceiverLocalSsrcConfiguration( |
| bool receiver_first) { |
| EXPECT_TRUE(channel_->SetSendParameters(send_parameters_)); |
| |
| const uint32_t kSenderSsrc = 0xC0FFEE; |
| const uint32_t kSecondSenderSsrc = 0xBADCAFE; |
| const uint32_t kReceiverSsrc = 0x4711; |
| const uint32_t kExpectedDefaultReceiverSsrc = 1; |
| |
| if (receiver_first) { |
| AddRecvStream(StreamParams::CreateLegacy(kReceiverSsrc)); |
| std::vector<FakeVideoReceiveStream*> receive_streams = |
| fake_call_->GetVideoReceiveStreams(); |
| ASSERT_EQ(1u, receive_streams.size()); |
| // Default local SSRC when we have no sender. |
| EXPECT_EQ(kExpectedDefaultReceiverSsrc, |
| receive_streams[0]->GetConfig().rtp.local_ssrc); |
| } |
| AddSendStream(StreamParams::CreateLegacy(kSenderSsrc)); |
| if (!receiver_first) |
| AddRecvStream(StreamParams::CreateLegacy(kReceiverSsrc)); |
| std::vector<FakeVideoReceiveStream*> receive_streams = |
| fake_call_->GetVideoReceiveStreams(); |
| ASSERT_EQ(1u, receive_streams.size()); |
| EXPECT_EQ(kSenderSsrc, receive_streams[0]->GetConfig().rtp.local_ssrc); |
| |
| // Removing first sender should fall back to another (in this case the second) |
| // local send stream's SSRC. |
| AddSendStream(StreamParams::CreateLegacy(kSecondSenderSsrc)); |
| ASSERT_TRUE(channel_->RemoveSendStream(kSenderSsrc)); |
| receive_streams = fake_call_->GetVideoReceiveStreams(); |
| ASSERT_EQ(1u, receive_streams.size()); |
| EXPECT_EQ(kSecondSenderSsrc, receive_streams[0]->GetConfig().rtp.local_ssrc); |
| |
| // Removing the last sender should fall back to default local SSRC. |
| ASSERT_TRUE(channel_->RemoveSendStream(kSecondSenderSsrc)); |
| receive_streams = fake_call_->GetVideoReceiveStreams(); |
| ASSERT_EQ(1u, receive_streams.size()); |
| EXPECT_EQ(kExpectedDefaultReceiverSsrc, |
| receive_streams[0]->GetConfig().rtp.local_ssrc); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, ConfiguresLocalSsrc) { |
| TestReceiverLocalSsrcConfiguration(false); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, ConfiguresLocalSsrcOnExistingReceivers) { |
| TestReceiverLocalSsrcConfiguration(true); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, Simulcast_QualityScalingNotAllowed) { |
| FakeVideoSendStream* stream = SetUpSimulcast(true, true); |
| EXPECT_FALSE(stream->GetEncoderConfig().is_quality_scaling_allowed); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, Singlecast_QualityScalingAllowed) { |
| FakeVideoSendStream* stream = SetUpSimulcast(false, true); |
| EXPECT_TRUE(stream->GetEncoderConfig().is_quality_scaling_allowed); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| SinglecastScreenSharing_QualityScalingNotAllowed) { |
| SetUpSimulcast(false, true); |
| |
| webrtc::test::FrameForwarder frame_forwarder; |
| VideoOptions options; |
| options.is_screencast = true; |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &frame_forwarder)); |
| // Fetch the latest stream since SetVideoSend() may recreate it if the |
| // screen content setting is changed. |
| FakeVideoSendStream* stream = fake_call_->GetVideoSendStreams().front(); |
| |
| EXPECT_FALSE(stream->GetEncoderConfig().is_quality_scaling_allowed); |
| EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr)); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, |
| SimulcastSingleActiveStream_QualityScalingAllowed) { |
| FakeVideoSendStream* stream = SetUpSimulcast(true, false); |
| |
| webrtc::RtpParameters rtp_parameters = |
| channel_->GetRtpSendParameters(last_ssrc_); |
| ASSERT_EQ(3u, rtp_parameters.encodings.size()); |
| ASSERT_TRUE(rtp_parameters.encodings[0].active); |
| ASSERT_TRUE(rtp_parameters.encodings[1].active); |
| ASSERT_TRUE(rtp_parameters.encodings[2].active); |
| rtp_parameters.encodings[0].active = false; |
| rtp_parameters.encodings[1].active = false; |
| EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters).ok()); |
| EXPECT_TRUE(stream->GetEncoderConfig().is_quality_scaling_allowed); |
| } |
| |
| class WebRtcVideoChannelSimulcastTest : public ::testing::Test { |
| public: |
| WebRtcVideoChannelSimulcastTest() |
| : fake_call_(), |
| encoder_factory_(new cricket::FakeWebRtcVideoEncoderFactory), |
| decoder_factory_(new cricket::FakeWebRtcVideoDecoderFactory), |
| mock_rate_allocator_factory_( |
| std::make_unique<webrtc::MockVideoBitrateAllocatorFactory>()), |
| engine_(std::unique_ptr<cricket::FakeWebRtcVideoEncoderFactory>( |
| encoder_factory_), |
| std::unique_ptr<cricket::FakeWebRtcVideoDecoderFactory>( |
| decoder_factory_), |
| field_trials_), |
| last_ssrc_(0) {} |
| |
| void SetUp() override { |
| encoder_factory_->AddSupportedVideoCodecType("VP8"); |
| decoder_factory_->AddSupportedVideoCodecType("VP8"); |
| channel_.reset(engine_.CreateMediaChannel( |
| &fake_call_, GetMediaConfig(), VideoOptions(), webrtc::CryptoOptions(), |
| mock_rate_allocator_factory_.get())); |
| channel_->OnReadyToSend(true); |
| last_ssrc_ = 123; |
| } |
| |
| protected: |
| void VerifySimulcastSettings(const VideoCodec& codec, |
| int capture_width, |
| int capture_height, |
| size_t num_configured_streams, |
| size_t expected_num_streams, |
| bool screenshare, |
| bool conference_mode) { |
| cricket::VideoSendParameters parameters; |
| parameters.codecs.push_back(codec); |
| parameters.conference_mode = conference_mode; |
| ASSERT_TRUE(channel_->SetSendParameters(parameters)); |
| |
| std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3); |
| RTC_DCHECK(num_configured_streams <= ssrcs.size()); |
| ssrcs.resize(num_configured_streams); |
| |
| AddSendStream(CreateSimStreamParams("cname", ssrcs)); |
| // Send a full-size frame to trigger a stream reconfiguration to use all |
| // expected simulcast layers. |
| webrtc::test::FrameForwarder frame_forwarder; |
| cricket::FakeFrameSource frame_source(capture_width, capture_height, |
| rtc::kNumMicrosecsPerSec / 30); |
| |
| VideoOptions options; |
| if (screenshare) |
| options.is_screencast = screenshare; |
| EXPECT_TRUE( |
| channel_->SetVideoSend(ssrcs.front(), &options, &frame_forwarder)); |
| // Fetch the latest stream since SetVideoSend() may recreate it if the |
| // screen content setting is changed. |
| FakeVideoSendStream* stream = fake_call_.GetVideoSendStreams().front(); |
| channel_->SetSend(true); |
| frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame()); |
| |
| auto rtp_parameters = channel_->GetRtpSendParameters(kSsrcs3[0]); |
| EXPECT_EQ(num_configured_streams, rtp_parameters.encodings.size()); |
| |
| std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams(); |
| ASSERT_EQ(expected_num_streams, video_streams.size()); |
| EXPECT_LE(expected_num_streams, stream->GetConfig().rtp.ssrcs.size()); |
| |
| std::vector<webrtc::VideoStream> expected_streams; |
| if (num_configured_streams > 1 || conference_mode) { |
| expected_streams = GetSimulcastConfig( |
| /*min_layers=*/1, num_configured_streams, capture_width, |
| capture_height, webrtc::kDefaultBitratePriority, kDefaultQpMax, |
| screenshare && conference_mode, true, field_trials_); |
| if (screenshare && conference_mode) { |
| for (const webrtc::VideoStream& stream : expected_streams) { |
| // Never scale screen content. |
| EXPECT_EQ(stream.width, rtc::checked_cast<size_t>(capture_width)); |
| EXPECT_EQ(stream.height, rtc::checked_cast<size_t>(capture_height)); |
| } |
| } |
| } else { |
| webrtc::VideoStream stream; |
| stream.width = capture_width; |
| stream.height = capture_height; |
| stream.max_framerate = kDefaultVideoMaxFramerate; |
| stream.min_bitrate_bps = webrtc::kDefaultMinVideoBitrateBps; |
| stream.target_bitrate_bps = stream.max_bitrate_bps = |
| GetMaxDefaultBitrateBps(capture_width, capture_height); |
| stream.max_qp = kDefaultQpMax; |
| expected_streams.push_back(stream); |
| } |
| |
| ASSERT_EQ(expected_streams.size(), video_streams.size()); |
| |
| size_t num_streams = video_streams.size(); |
| int total_max_bitrate_bps = 0; |
| for (size_t i = 0; i < num_streams; ++i) { |
| EXPECT_EQ(expected_streams[i].width, video_streams[i].width); |
| EXPECT_EQ(expected_streams[i].height, video_streams[i].height); |
| |
| EXPECT_GT(video_streams[i].max_framerate, 0); |
| EXPECT_EQ(expected_streams[i].max_framerate, |
| video_streams[i].max_framerate); |
| |
| EXPECT_GT(video_streams[i].min_bitrate_bps, 0); |
| EXPECT_EQ(expected_streams[i].min_bitrate_bps, |
| video_streams[i].min_bitrate_bps); |
| |
| EXPECT_GT(video_streams[i].target_bitrate_bps, 0); |
| EXPECT_EQ(expected_streams[i].target_bitrate_bps, |
| video_streams[i].target_bitrate_bps); |
| |
| EXPECT_GT(video_streams[i].max_bitrate_bps, 0); |
| EXPECT_EQ(expected_streams[i].max_bitrate_bps, |
| video_streams[i].max_bitrate_bps); |
| |
| EXPECT_GT(video_streams[i].max_qp, 0); |
| EXPECT_EQ(expected_streams[i].max_qp, video_streams[i].max_qp); |
| |
| EXPECT_EQ(num_configured_streams > 1 || conference_mode, |
| expected_streams[i].num_temporal_layers.has_value()); |
| |
| if (conference_mode) { |
| EXPECT_EQ(expected_streams[i].num_temporal_layers, |
| video_streams[i].num_temporal_layers); |
| } |
| |
| if (i == num_streams - 1) { |
| total_max_bitrate_bps += video_streams[i].max_bitrate_bps; |
| } else { |
| total_max_bitrate_bps += video_streams[i].target_bitrate_bps; |
| } |
| } |
| |
| EXPECT_TRUE(channel_->SetVideoSend(ssrcs.front(), nullptr, nullptr)); |
| } |
| |
| FakeVideoSendStream* AddSendStream() { |
| return AddSendStream(StreamParams::CreateLegacy(last_ssrc_++)); |
| } |
| |
| FakeVideoSendStream* AddSendStream(const StreamParams& sp) { |
| size_t num_streams = fake_call_.GetVideoSendStreams().size(); |
| EXPECT_TRUE(channel_->AddSendStream(sp)); |
| std::vector<FakeVideoSendStream*> streams = |
| fake_call_.GetVideoSendStreams(); |
| EXPECT_EQ(num_streams + 1, streams.size()); |
| return streams[streams.size() - 1]; |
| } |
| |
| std::vector<FakeVideoSendStream*> GetFakeSendStreams() { |
| return fake_call_.GetVideoSendStreams(); |
| } |
| |
| FakeVideoReceiveStream* AddRecvStream() { |
| return AddRecvStream(StreamParams::CreateLegacy(last_ssrc_++)); |
| } |
| |
| FakeVideoReceiveStream* AddRecvStream(const StreamParams& sp) { |
| size_t num_streams = fake_call_.GetVideoReceiveStreams().size(); |
| EXPECT_TRUE(channel_->AddRecvStream(sp)); |
| std::vector<FakeVideoReceiveStream*> streams = |
| fake_call_.GetVideoReceiveStreams(); |
| EXPECT_EQ(num_streams + 1, streams.size()); |
| return streams[streams.size() - 1]; |
| } |
| |
| webrtc::RtcEventLogNull event_log_; |
| FakeCall fake_call_; |
| cricket::FakeWebRtcVideoEncoderFactory* encoder_factory_; |
| cricket::FakeWebRtcVideoDecoderFactory* decoder_factory_; |
| std::unique_ptr<webrtc::MockVideoBitrateAllocatorFactory> |
| mock_rate_allocator_factory_; |
| webrtc::FieldTrialBasedConfig field_trials_; |
| WebRtcVideoEngine engine_; |
| std::unique_ptr<VideoMediaChannel> channel_; |
| uint32_t last_ssrc_; |
| }; |
| |
| TEST_F(WebRtcVideoChannelSimulcastTest, SetSendCodecsWith2SimulcastStreams) { |
| VerifySimulcastSettings(cricket::VideoCodec("VP8"), 640, 360, 2, 2, false, |
| true); |
| } |
| |
| TEST_F(WebRtcVideoChannelSimulcastTest, SetSendCodecsWith3SimulcastStreams) { |
| VerifySimulcastSettings(cricket::VideoCodec("VP8"), 1280, 720, 3, 3, false, |
| true); |
| } |
| |
| // Test that we normalize send codec format size in simulcast. |
| TEST_F(WebRtcVideoChannelSimulcastTest, SetSendCodecsWithOddSizeInSimulcast) { |
| VerifySimulcastSettings(cricket::VideoCodec("VP8"), 541, 271, 2, 2, false, |
| true); |
| } |
| |
| TEST_F(WebRtcVideoChannelSimulcastTest, SetSendCodecsForScreenshare) { |
| VerifySimulcastSettings(cricket::VideoCodec("VP8"), 1280, 720, 3, 3, true, |
| false); |
| } |
| |
| TEST_F(WebRtcVideoChannelSimulcastTest, SetSendCodecsForSimulcastScreenshare) { |
| VerifySimulcastSettings(cricket::VideoCodec("VP8"), 1280, 720, 3, 2, true, |
| true); |
| } |
| |
| TEST_F(WebRtcVideoChannelSimulcastTest, SimulcastScreenshareWithoutConference) { |
| VerifySimulcastSettings(cricket::VideoCodec("VP8"), 1280, 720, 3, 3, true, |
| false); |
| } |
| |
| TEST_F(WebRtcVideoChannelBaseTest, GetSources) { |
| EXPECT_THAT(channel_->GetSources(kSsrc), IsEmpty()); |
| |
| channel_->SetDefaultSink(&renderer_); |
| EXPECT_TRUE(SetDefaultCodec()); |
| EXPECT_TRUE(SetSend(true)); |
| EXPECT_EQ(renderer_.num_rendered_frames(), 0); |
| |
| // Send and receive one frame. |
| SendFrame(); |
| EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout); |
| |
| EXPECT_THAT(channel_->GetSources(kSsrc - 1), IsEmpty()); |
| EXPECT_THAT(channel_->GetSources(kSsrc), SizeIs(1)); |
| EXPECT_THAT(channel_->GetSources(kSsrc + 1), IsEmpty()); |
| |
| webrtc::RtpSource source = channel_->GetSources(kSsrc)[0]; |
| EXPECT_EQ(source.source_id(), kSsrc); |
| EXPECT_EQ(source.source_type(), webrtc::RtpSourceType::SSRC); |
| int64_t rtp_timestamp_1 = source.rtp_timestamp(); |
| int64_t timestamp_ms_1 = source.timestamp_ms(); |
| |
| // Send and receive another frame. |
| SendFrame(); |
| EXPECT_FRAME_WAIT(2, kVideoWidth, kVideoHeight, kTimeout); |
| |
| EXPECT_THAT(channel_->GetSources(kSsrc - 1), IsEmpty()); |
| EXPECT_THAT(channel_->GetSources(kSsrc), SizeIs(1)); |
| EXPECT_THAT(channel_->GetSources(kSsrc + 1), IsEmpty()); |
| |
| source = channel_->GetSources(kSsrc)[0]; |
| EXPECT_EQ(source.source_id(), kSsrc); |
| EXPECT_EQ(source.source_type(), webrtc::RtpSourceType::SSRC); |
| int64_t rtp_timestamp_2 = source.rtp_timestamp(); |
| int64_t timestamp_ms_2 = source.timestamp_ms(); |
| |
| EXPECT_GT(rtp_timestamp_2, rtp_timestamp_1); |
| EXPECT_GT(timestamp_ms_2, timestamp_ms_1); |
| } |
| |
| TEST_F(WebRtcVideoChannelTest, SetsRidsOnSendStream) { |
| StreamParams sp = CreateSimStreamParams("cname", {123, 456, 789}); |
| |
| std::vector<std::string> rids = {"f", "h", "q"}; |
| std::vector<cricket::RidDescription> rid_descriptions; |
| for (const auto& rid : rids) { |
| rid_descriptions.emplace_back(rid, cricket::RidDirection::kSend); |
| } |
| sp.set_rids(rid_descriptions); |
| |
| ASSERT_TRUE(channel_->AddSendStream(sp)); |
| const auto& streams = fake_call_->GetVideoSendStreams(); |
| ASSERT_EQ(1u, streams.size()); |
| auto stream = streams[0]; |
| ASSERT_NE(stream, nullptr); |
| const auto& config = stream->GetConfig(); |
| EXPECT_THAT(config.rtp.rids, ElementsAreArray(rids)); |
| } |
| |
| } // namespace cricket |
