| /* |
| * Copyright 2023 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 <string> |
| #include <vector> |
| |
| #include "absl/strings/match.h" |
| #include "absl/types/optional.h" |
| #include "api/audio_codecs/builtin_audio_decoder_factory.h" |
| #include "api/audio_codecs/builtin_audio_encoder_factory.h" |
| #include "api/audio_codecs/opus_audio_decoder_factory.h" |
| #include "api/audio_codecs/opus_audio_encoder_factory.h" |
| #include "api/media_types.h" |
| #include "api/rtc_error.h" |
| #include "api/rtp_parameters.h" |
| #include "api/rtp_transceiver_direction.h" |
| #include "api/rtp_transceiver_interface.h" |
| #include "api/stats/rtcstats_objects.h" |
| #include "api/units/data_rate.h" |
| #include "api/video_codecs/video_decoder_factory_template.h" |
| #include "api/video_codecs/video_decoder_factory_template_dav1d_adapter.h" |
| #include "api/video_codecs/video_decoder_factory_template_libvpx_vp8_adapter.h" |
| #include "api/video_codecs/video_decoder_factory_template_libvpx_vp9_adapter.h" |
| #include "api/video_codecs/video_decoder_factory_template_open_h264_adapter.h" |
| #include "api/video_codecs/video_encoder_factory_template.h" |
| #include "api/video_codecs/video_encoder_factory_template_libaom_av1_adapter.h" |
| #include "api/video_codecs/video_encoder_factory_template_libvpx_vp8_adapter.h" |
| #include "api/video_codecs/video_encoder_factory_template_libvpx_vp9_adapter.h" |
| #include "api/video_codecs/video_encoder_factory_template_open_h264_adapter.h" |
| #include "pc/sdp_utils.h" |
| #include "pc/simulcast_description.h" |
| #include "pc/test/mock_peer_connection_observers.h" |
| #include "pc/test/peer_connection_test_wrapper.h" |
| #include "pc/test/simulcast_layer_util.h" |
| #include "rtc_base/gunit.h" |
| #include "rtc_base/physical_socket_server.h" |
| #include "test/gmock.h" |
| #include "test/gtest.h" |
| |
| using ::testing::Eq; |
| using ::testing::Optional; |
| using ::testing::SizeIs; |
| using ::testing::StrCaseEq; |
| using ::testing::StrEq; |
| |
| namespace webrtc { |
| |
| namespace { |
| |
| constexpr TimeDelta kDefaultTimeout = TimeDelta::Seconds(5); |
| // Most tests pass in 20-30 seconds, but some tests take longer such as AV1 |
| // requiring additional ramp-up time (https://crbug.com/webrtc/15006) or SVC |
| // (LxTx_KEY) being slower than simulcast to send top spatial layer. |
| // TODO(https://crbug.com/webrtc/15076): Remove need for long rampup timeouts by |
| // using simulated time. |
| constexpr TimeDelta kLongTimeoutForRampingUp = TimeDelta::Minutes(1); |
| |
| // The max bitrate 1500 kbps may be subject to change in the future. What we're |
| // interested in here is that all code paths that result in L1T3 result in the |
| // same target bitrate which does not exceed this limit. |
| constexpr DataRate kVp9ExpectedMaxBitrateForL1T3 = |
| DataRate::KilobitsPerSec(1500); |
| |
| struct StringParamToString { |
| std::string operator()(const ::testing::TestParamInfo<std::string>& info) { |
| return info.param; |
| } |
| }; |
| |
| // RTX, RED and FEC are reliability mechanisms used in combinations with other |
| // codecs, but are not themselves a specific codec. Typically you don't want to |
| // filter these out of the list of codec preferences. |
| bool IsReliabilityMechanism(const webrtc::RtpCodecCapability& codec) { |
| return absl::EqualsIgnoreCase(codec.name, cricket::kRtxCodecName) || |
| absl::EqualsIgnoreCase(codec.name, cricket::kRedCodecName) || |
| absl::EqualsIgnoreCase(codec.name, cricket::kUlpfecCodecName); |
| } |
| |
| std::string GetCurrentCodecMimeType( |
| rtc::scoped_refptr<const webrtc::RTCStatsReport> report, |
| const webrtc::RTCOutboundRtpStreamStats& outbound_rtp) { |
| return outbound_rtp.codec_id.is_defined() |
| ? *report->GetAs<webrtc::RTCCodecStats>(*outbound_rtp.codec_id) |
| ->mime_type |
| : ""; |
| } |
| |
| struct RidAndResolution { |
| std::string rid; |
| uint32_t width; |
| uint32_t height; |
| }; |
| |
| const webrtc::RTCOutboundRtpStreamStats* FindOutboundRtpByRid( |
| const std::vector<const webrtc::RTCOutboundRtpStreamStats*>& outbound_rtps, |
| const absl::string_view& rid) { |
| for (const auto* outbound_rtp : outbound_rtps) { |
| if (outbound_rtp->rid.is_defined() && *outbound_rtp->rid == rid) { |
| return outbound_rtp; |
| } |
| } |
| return nullptr; |
| } |
| |
| } // namespace |
| |
| class PeerConnectionEncodingsIntegrationTest : public ::testing::Test { |
| public: |
| PeerConnectionEncodingsIntegrationTest() |
| : background_thread_(std::make_unique<rtc::Thread>(&pss_)) { |
| RTC_CHECK(background_thread_->Start()); |
| } |
| |
| rtc::scoped_refptr<PeerConnectionTestWrapper> CreatePc() { |
| auto pc_wrapper = rtc::make_ref_counted<PeerConnectionTestWrapper>( |
| "pc", &pss_, background_thread_.get(), background_thread_.get()); |
| pc_wrapper->CreatePc({}, webrtc::CreateBuiltinAudioEncoderFactory(), |
| webrtc::CreateBuiltinAudioDecoderFactory()); |
| return pc_wrapper; |
| } |
| |
| rtc::scoped_refptr<RtpTransceiverInterface> AddTransceiverWithSimulcastLayers( |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local, |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote, |
| std::vector<cricket::SimulcastLayer> init_layers) { |
| rtc::scoped_refptr<webrtc::MediaStreamInterface> stream = |
| local->GetUserMedia( |
| /*audio=*/false, cricket::AudioOptions(), /*video=*/true, |
| {.width = 1280, .height = 720}); |
| rtc::scoped_refptr<VideoTrackInterface> track = stream->GetVideoTracks()[0]; |
| |
| RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> |
| transceiver_or_error = local->pc()->AddTransceiver( |
| track, CreateTransceiverInit(init_layers)); |
| EXPECT_TRUE(transceiver_or_error.ok()); |
| return transceiver_or_error.value(); |
| } |
| |
| bool HasSenderVideoCodecCapability( |
| rtc::scoped_refptr<PeerConnectionTestWrapper> pc_wrapper, |
| absl::string_view codec_name) { |
| std::vector<RtpCodecCapability> codecs = |
| pc_wrapper->pc_factory() |
| ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO) |
| .codecs; |
| return std::find_if(codecs.begin(), codecs.end(), |
| [&codec_name](const RtpCodecCapability& codec) { |
| return absl::EqualsIgnoreCase(codec.name, codec_name); |
| }) != codecs.end(); |
| } |
| |
| std::vector<RtpCodecCapability> GetCapabilitiesAndRestrictToCodec( |
| rtc::scoped_refptr<PeerConnectionTestWrapper> pc_wrapper, |
| absl::string_view codec_name) { |
| std::vector<RtpCodecCapability> codecs = |
| pc_wrapper->pc_factory() |
| ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO) |
| .codecs; |
| codecs.erase(std::remove_if(codecs.begin(), codecs.end(), |
| [&codec_name](const RtpCodecCapability& codec) { |
| return !IsReliabilityMechanism(codec) && |
| !absl::EqualsIgnoreCase(codec.name, |
| codec_name); |
| }), |
| codecs.end()); |
| RTC_DCHECK(std::find_if(codecs.begin(), codecs.end(), |
| [&codec_name](const RtpCodecCapability& codec) { |
| return absl::EqualsIgnoreCase(codec.name, |
| codec_name); |
| }) != codecs.end()); |
| return codecs; |
| } |
| |
| void ExchangeIceCandidates( |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper, |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper) { |
| local_pc_wrapper->SignalOnIceCandidateReady.connect( |
| remote_pc_wrapper.get(), &PeerConnectionTestWrapper::AddIceCandidate); |
| remote_pc_wrapper->SignalOnIceCandidateReady.connect( |
| local_pc_wrapper.get(), &PeerConnectionTestWrapper::AddIceCandidate); |
| } |
| |
| void NegotiateWithSimulcastTweaks( |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper, |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper) { |
| // Create and set offer for `local_pc_wrapper`. |
| std::unique_ptr<SessionDescriptionInterface> offer = |
| CreateOffer(local_pc_wrapper); |
| rtc::scoped_refptr<MockSetSessionDescriptionObserver> p1 = |
| SetLocalDescription(local_pc_wrapper, offer.get()); |
| // Modify the offer before handoff because `remote_pc_wrapper` only supports |
| // receiving singlecast. |
| cricket::SimulcastDescription simulcast_description = |
| RemoveSimulcast(offer.get()); |
| rtc::scoped_refptr<MockSetSessionDescriptionObserver> p2 = |
| SetRemoteDescription(remote_pc_wrapper, offer.get()); |
| EXPECT_TRUE(Await({p1, p2})); |
| |
| // Create and set answer for `remote_pc_wrapper`. |
| std::unique_ptr<SessionDescriptionInterface> answer = |
| CreateAnswer(remote_pc_wrapper); |
| p1 = SetLocalDescription(remote_pc_wrapper, answer.get()); |
| // Modify the answer before handoff because `local_pc_wrapper` should still |
| // send simulcast. |
| cricket::MediaContentDescription* mcd_answer = |
| answer->description()->contents()[0].media_description(); |
| mcd_answer->mutable_streams().clear(); |
| std::vector<cricket::SimulcastLayer> simulcast_layers = |
| simulcast_description.send_layers().GetAllLayers(); |
| cricket::SimulcastLayerList& receive_layers = |
| mcd_answer->simulcast_description().receive_layers(); |
| for (const auto& layer : simulcast_layers) { |
| receive_layers.AddLayer(layer); |
| } |
| p2 = SetRemoteDescription(local_pc_wrapper, answer.get()); |
| EXPECT_TRUE(Await({p1, p2})); |
| } |
| |
| rtc::scoped_refptr<const RTCStatsReport> GetStats( |
| rtc::scoped_refptr<PeerConnectionTestWrapper> pc_wrapper) { |
| auto callback = rtc::make_ref_counted<MockRTCStatsCollectorCallback>(); |
| pc_wrapper->pc()->GetStats(callback.get()); |
| EXPECT_TRUE_WAIT(callback->called(), kDefaultTimeout.ms()); |
| return callback->report(); |
| } |
| |
| bool IsCodecIdDifferent( |
| rtc::scoped_refptr<PeerConnectionTestWrapper> pc_wrapper, |
| size_t index, |
| const std::string& codec_id) { |
| return IsCodecIdDifferentWithScalabilityMode(pc_wrapper, index, codec_id, |
| absl::nullopt); |
| } |
| |
| bool IsCodecIdDifferentWithScalabilityMode( |
| rtc::scoped_refptr<PeerConnectionTestWrapper> pc_wrapper, |
| size_t index, |
| const std::string& codec_id, |
| absl::optional<std::string> wanted_scalability_mode) { |
| rtc::scoped_refptr<const RTCStatsReport> report = GetStats(pc_wrapper); |
| std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps = |
| report->GetStatsOfType<RTCOutboundRtpStreamStats>(); |
| return outbound_rtps[index]->codec_id.value() != codec_id && |
| (!wanted_scalability_mode || |
| (outbound_rtps[index]->scalability_mode.has_value() && |
| outbound_rtps[index]->scalability_mode.value() == |
| wanted_scalability_mode)); |
| } |
| |
| bool HasOutboundRtpBytesSent( |
| rtc::scoped_refptr<PeerConnectionTestWrapper> pc_wrapper, |
| size_t num_layers) { |
| return HasOutboundRtpBytesSent(pc_wrapper, num_layers, num_layers); |
| } |
| |
| bool HasOutboundRtpBytesSent( |
| rtc::scoped_refptr<PeerConnectionTestWrapper> pc_wrapper, |
| size_t num_layers, |
| size_t num_active_layers) { |
| rtc::scoped_refptr<const RTCStatsReport> report = GetStats(pc_wrapper); |
| std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps = |
| report->GetStatsOfType<RTCOutboundRtpStreamStats>(); |
| if (outbound_rtps.size() != num_layers) { |
| return false; |
| } |
| size_t num_sending_layers = 0; |
| for (const auto* outbound_rtp : outbound_rtps) { |
| if (outbound_rtp->bytes_sent.is_defined() && |
| *outbound_rtp->bytes_sent > 0u) { |
| ++num_sending_layers; |
| } |
| } |
| return num_sending_layers == num_active_layers; |
| } |
| |
| bool HasOutboundRtpWithRidAndScalabilityMode( |
| rtc::scoped_refptr<PeerConnectionTestWrapper> pc_wrapper, |
| absl::string_view rid, |
| absl::string_view expected_scalability_mode, |
| uint32_t frame_height) { |
| rtc::scoped_refptr<const RTCStatsReport> report = GetStats(pc_wrapper); |
| std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps = |
| report->GetStatsOfType<RTCOutboundRtpStreamStats>(); |
| auto* outbound_rtp = FindOutboundRtpByRid(outbound_rtps, rid); |
| if (!outbound_rtp || !outbound_rtp->scalability_mode.is_defined() || |
| *outbound_rtp->scalability_mode != expected_scalability_mode) { |
| return false; |
| } |
| if (outbound_rtp->frame_height.is_defined()) { |
| RTC_LOG(LS_INFO) << "Waiting for target resolution (" << frame_height |
| << "p). Currently at " << *outbound_rtp->frame_height |
| << "p..."; |
| } else { |
| RTC_LOG(LS_INFO) |
| << "Waiting for target resolution. No frames encoded yet..."; |
| } |
| if (!outbound_rtp->frame_height.is_defined() || |
| *outbound_rtp->frame_height != frame_height) { |
| // Sleep to avoid log spam when this is used in ASSERT_TRUE_WAIT(). |
| rtc::Thread::Current()->SleepMs(1000); |
| return false; |
| } |
| return true; |
| } |
| |
| bool OutboundRtpResolutionsAreLessThanOrEqualToExpectations( |
| rtc::scoped_refptr<PeerConnectionTestWrapper> pc_wrapper, |
| std::vector<RidAndResolution> resolutions) { |
| rtc::scoped_refptr<const RTCStatsReport> report = GetStats(pc_wrapper); |
| std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps = |
| report->GetStatsOfType<RTCOutboundRtpStreamStats>(); |
| for (const RidAndResolution& resolution : resolutions) { |
| const RTCOutboundRtpStreamStats* outbound_rtp = nullptr; |
| if (!resolution.rid.empty()) { |
| outbound_rtp = FindOutboundRtpByRid(outbound_rtps, resolution.rid); |
| } else if (outbound_rtps.size() == 1u) { |
| outbound_rtp = outbound_rtps[0]; |
| } |
| if (!outbound_rtp || !outbound_rtp->frame_width.is_defined() || |
| !outbound_rtp->frame_height.is_defined()) { |
| // RTP not found by rid or has not encoded a frame yet. |
| RTC_LOG(LS_ERROR) << "rid=" << resolution.rid << " does not have " |
| << "resolution metrics"; |
| return false; |
| } |
| if (*outbound_rtp->frame_width > resolution.width || |
| *outbound_rtp->frame_height > resolution.height) { |
| RTC_LOG(LS_ERROR) << "rid=" << resolution.rid << " is " |
| << *outbound_rtp->frame_width << "x" |
| << *outbound_rtp->frame_height |
| << ", this is greater than the " |
| << "expected " << resolution.width << "x" |
| << resolution.height; |
| return false; |
| } |
| } |
| return true; |
| } |
| |
| protected: |
| std::unique_ptr<SessionDescriptionInterface> CreateOffer( |
| rtc::scoped_refptr<PeerConnectionTestWrapper> pc_wrapper) { |
| auto observer = |
| rtc::make_ref_counted<MockCreateSessionDescriptionObserver>(); |
| pc_wrapper->pc()->CreateOffer(observer.get(), {}); |
| EXPECT_EQ_WAIT(true, observer->called(), kDefaultTimeout.ms()); |
| return observer->MoveDescription(); |
| } |
| |
| std::unique_ptr<SessionDescriptionInterface> CreateAnswer( |
| rtc::scoped_refptr<PeerConnectionTestWrapper> pc_wrapper) { |
| auto observer = |
| rtc::make_ref_counted<MockCreateSessionDescriptionObserver>(); |
| pc_wrapper->pc()->CreateAnswer(observer.get(), {}); |
| EXPECT_EQ_WAIT(true, observer->called(), kDefaultTimeout.ms()); |
| return observer->MoveDescription(); |
| } |
| |
| rtc::scoped_refptr<MockSetSessionDescriptionObserver> SetLocalDescription( |
| rtc::scoped_refptr<PeerConnectionTestWrapper> pc_wrapper, |
| SessionDescriptionInterface* sdp) { |
| auto observer = rtc::make_ref_counted<MockSetSessionDescriptionObserver>(); |
| pc_wrapper->pc()->SetLocalDescription( |
| observer.get(), CloneSessionDescription(sdp).release()); |
| return observer; |
| } |
| |
| rtc::scoped_refptr<MockSetSessionDescriptionObserver> SetRemoteDescription( |
| rtc::scoped_refptr<PeerConnectionTestWrapper> pc_wrapper, |
| SessionDescriptionInterface* sdp) { |
| auto observer = rtc::make_ref_counted<MockSetSessionDescriptionObserver>(); |
| pc_wrapper->pc()->SetRemoteDescription( |
| observer.get(), CloneSessionDescription(sdp).release()); |
| return observer; |
| } |
| |
| // To avoid ICE candidates arriving before the remote endpoint has received |
| // the offer it is important to SetLocalDescription() and |
| // SetRemoteDescription() are kicked off without awaiting in-between. This |
| // helper is used to await multiple observers. |
| bool Await(std::vector<rtc::scoped_refptr<MockSetSessionDescriptionObserver>> |
| observers) { |
| for (auto& observer : observers) { |
| EXPECT_EQ_WAIT(true, observer->called(), kDefaultTimeout.ms()); |
| if (!observer->result()) { |
| return false; |
| } |
| } |
| return true; |
| } |
| |
| rtc::PhysicalSocketServer pss_; |
| std::unique_ptr<rtc::Thread> background_thread_; |
| }; |
| |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| VP8_SingleEncodingDefaultsToL1T1) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); |
| ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); |
| |
| std::vector<cricket::SimulcastLayer> layers = |
| CreateLayers({"f"}, /*active=*/true); |
| rtc::scoped_refptr<RtpTransceiverInterface> transceiver = |
| AddTransceiverWithSimulcastLayers(local_pc_wrapper, remote_pc_wrapper, |
| layers); |
| std::vector<RtpCodecCapability> codecs = |
| GetCapabilitiesAndRestrictToCodec(local_pc_wrapper, "VP8"); |
| transceiver->SetCodecPreferences(codecs); |
| |
| NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); |
| local_pc_wrapper->WaitForConnection(); |
| remote_pc_wrapper->WaitForConnection(); |
| |
| // Wait until media is flowing. |
| ASSERT_TRUE_WAIT(HasOutboundRtpBytesSent(local_pc_wrapper, 1u), |
| kDefaultTimeout.ms()); |
| EXPECT_TRUE(OutboundRtpResolutionsAreLessThanOrEqualToExpectations( |
| local_pc_wrapper, {{"", 1280, 720}})); |
| // Verify codec and scalability mode. |
| rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper); |
| std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps = |
| report->GetStatsOfType<RTCOutboundRtpStreamStats>(); |
| ASSERT_THAT(outbound_rtps, SizeIs(1u)); |
| EXPECT_THAT(GetCurrentCodecMimeType(report, *outbound_rtps[0]), |
| StrCaseEq("video/VP8")); |
| EXPECT_THAT(*outbound_rtps[0]->scalability_mode, StrEq("L1T1")); |
| } |
| |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| VP8_RejectsSvcAndDefaultsToL1T1) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); |
| ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); |
| |
| std::vector<cricket::SimulcastLayer> layers = |
| CreateLayers({"f"}, /*active=*/true); |
| rtc::scoped_refptr<RtpTransceiverInterface> transceiver = |
| AddTransceiverWithSimulcastLayers(local_pc_wrapper, remote_pc_wrapper, |
| layers); |
| // Restricting codecs restricts what SetParameters() will accept or reject. |
| std::vector<RtpCodecCapability> codecs = |
| GetCapabilitiesAndRestrictToCodec(local_pc_wrapper, "VP8"); |
| transceiver->SetCodecPreferences(codecs); |
| // Attempt SVC (L3T3_KEY). This is not possible because only VP8 is up for |
| // negotiation and VP8 does not support it. |
| rtc::scoped_refptr<RtpSenderInterface> sender = transceiver->sender(); |
| RtpParameters parameters = sender->GetParameters(); |
| ASSERT_EQ(parameters.encodings.size(), 1u); |
| parameters.encodings[0].scalability_mode = "L3T3_KEY"; |
| parameters.encodings[0].scale_resolution_down_by = 1; |
| EXPECT_FALSE(sender->SetParameters(parameters).ok()); |
| // `scalability_mode` remains unset because SetParameters() failed. |
| parameters = sender->GetParameters(); |
| ASSERT_EQ(parameters.encodings.size(), 1u); |
| EXPECT_THAT(parameters.encodings[0].scalability_mode, Eq(absl::nullopt)); |
| |
| NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); |
| local_pc_wrapper->WaitForConnection(); |
| remote_pc_wrapper->WaitForConnection(); |
| |
| // Wait until media is flowing. |
| ASSERT_TRUE_WAIT(HasOutboundRtpBytesSent(local_pc_wrapper, 1u), |
| kDefaultTimeout.ms()); |
| // When `scalability_mode` is not set, VP8 defaults to L1T1. |
| rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper); |
| std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps = |
| report->GetStatsOfType<RTCOutboundRtpStreamStats>(); |
| ASSERT_THAT(outbound_rtps, SizeIs(1u)); |
| EXPECT_THAT(GetCurrentCodecMimeType(report, *outbound_rtps[0]), |
| StrCaseEq("video/VP8")); |
| EXPECT_THAT(*outbound_rtps[0]->scalability_mode, StrEq("L1T1")); |
| // GetParameters() confirms `scalability_mode` is still not set. |
| parameters = sender->GetParameters(); |
| ASSERT_EQ(parameters.encodings.size(), 1u); |
| EXPECT_THAT(parameters.encodings[0].scalability_mode, Eq(absl::nullopt)); |
| } |
| |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| VP8_FallbackFromSvcResultsInL1T2) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); |
| ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); |
| |
| std::vector<cricket::SimulcastLayer> layers = |
| CreateLayers({"f"}, /*active=*/true); |
| rtc::scoped_refptr<RtpTransceiverInterface> transceiver = |
| AddTransceiverWithSimulcastLayers(local_pc_wrapper, remote_pc_wrapper, |
| layers); |
| // Verify test assumption that VP8 is first in the list, but don't modify the |
| // codec preferences because we want the sender to think SVC is a possibility. |
| std::vector<RtpCodecCapability> codecs = |
| local_pc_wrapper->pc_factory() |
| ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO) |
| .codecs; |
| EXPECT_THAT(codecs[0].name, StrCaseEq("VP8")); |
| // Attempt SVC (L3T3_KEY), which is not possible with VP8, but the sender does |
| // not yet know which codec we'll use so the parameters will be accepted. |
| rtc::scoped_refptr<RtpSenderInterface> sender = transceiver->sender(); |
| RtpParameters parameters = sender->GetParameters(); |
| ASSERT_EQ(parameters.encodings.size(), 1u); |
| parameters.encodings[0].scalability_mode = "L3T3_KEY"; |
| parameters.encodings[0].scale_resolution_down_by = 1; |
| EXPECT_TRUE(sender->SetParameters(parameters).ok()); |
| // Verify fallback has not happened yet. |
| parameters = sender->GetParameters(); |
| ASSERT_EQ(parameters.encodings.size(), 1u); |
| EXPECT_THAT(parameters.encodings[0].scalability_mode, |
| Optional(std::string("L3T3_KEY"))); |
| |
| // Negotiate, this results in VP8 being picked and fallback happening. |
| NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); |
| local_pc_wrapper->WaitForConnection(); |
| remote_pc_wrapper->WaitForConnection(); |
| // `scalaiblity_mode` is assigned the fallback value "L1T2" which is different |
| // than the default of absl::nullopt. |
| parameters = sender->GetParameters(); |
| ASSERT_EQ(parameters.encodings.size(), 1u); |
| EXPECT_THAT(parameters.encodings[0].scalability_mode, |
| Optional(std::string("L1T2"))); |
| |
| // Wait until media is flowing, no significant time needed because we only |
| // have one layer. |
| ASSERT_TRUE_WAIT(HasOutboundRtpBytesSent(local_pc_wrapper, 1u), |
| kDefaultTimeout.ms()); |
| // GetStats() confirms "L1T2" is used which is different than the "L1T1" |
| // default or the "L3T3_KEY" that was attempted. |
| rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper); |
| std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps = |
| report->GetStatsOfType<RTCOutboundRtpStreamStats>(); |
| ASSERT_THAT(outbound_rtps, SizeIs(1u)); |
| EXPECT_THAT(GetCurrentCodecMimeType(report, *outbound_rtps[0]), |
| StrCaseEq("video/VP8")); |
| EXPECT_THAT(*outbound_rtps[0]->scalability_mode, StrEq("L1T2")); |
| } |
| |
| // The legacy SVC path is triggered when VP9 us used, but `scalability_mode` has |
| // not been specified. |
| // TODO(https://crbug.com/webrtc/14889): When legacy VP9 SVC path has been |
| // deprecated and removed, update this test to assert that simulcast is used |
| // (i.e. VP9 is not treated differently than VP8). |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| VP9_LegacySvcWhenScalabilityModeNotSpecified) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); |
| ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); |
| |
| std::vector<cricket::SimulcastLayer> layers = |
| CreateLayers({"f", "h", "q"}, /*active=*/true); |
| rtc::scoped_refptr<RtpTransceiverInterface> transceiver = |
| AddTransceiverWithSimulcastLayers(local_pc_wrapper, remote_pc_wrapper, |
| layers); |
| std::vector<RtpCodecCapability> codecs = |
| GetCapabilitiesAndRestrictToCodec(local_pc_wrapper, "VP9"); |
| transceiver->SetCodecPreferences(codecs); |
| |
| NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); |
| local_pc_wrapper->WaitForConnection(); |
| remote_pc_wrapper->WaitForConnection(); |
| |
| // Wait until media is flowing. We only expect a single RTP stream. |
| // We expect to see bytes flowing almost immediately on the lowest layer. |
| ASSERT_TRUE_WAIT(HasOutboundRtpBytesSent(local_pc_wrapper, 1u), |
| kDefaultTimeout.ms()); |
| // Wait until scalability mode is reported and expected resolution reached. |
| // Ramp up time may be significant. |
| ASSERT_TRUE_WAIT(HasOutboundRtpWithRidAndScalabilityMode( |
| local_pc_wrapper, "f", "L3T3_KEY", 720), |
| kLongTimeoutForRampingUp.ms()); |
| |
| // Despite SVC being used on a single RTP stream, GetParameters() returns the |
| // three encodings that we configured earlier (this is not spec-compliant but |
| // it is how legacy SVC behaves). |
| rtc::scoped_refptr<RtpSenderInterface> sender = transceiver->sender(); |
| std::vector<RtpEncodingParameters> encodings = |
| sender->GetParameters().encodings; |
| ASSERT_EQ(encodings.size(), 3u); |
| // When legacy SVC is used, `scalability_mode` is not specified. |
| EXPECT_FALSE(encodings[0].scalability_mode.has_value()); |
| EXPECT_FALSE(encodings[1].scalability_mode.has_value()); |
| EXPECT_FALSE(encodings[2].scalability_mode.has_value()); |
| } |
| |
| // The spec-compliant way to configure SVC for a single stream. The expected |
| // outcome is the same as for the legacy SVC case except that we only have one |
| // encoding in GetParameters(). |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| VP9_StandardSvcWithOnlyOneEncoding) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); |
| ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); |
| |
| std::vector<cricket::SimulcastLayer> layers = |
| CreateLayers({"f"}, /*active=*/true); |
| rtc::scoped_refptr<RtpTransceiverInterface> transceiver = |
| AddTransceiverWithSimulcastLayers(local_pc_wrapper, remote_pc_wrapper, |
| layers); |
| std::vector<RtpCodecCapability> codecs = |
| GetCapabilitiesAndRestrictToCodec(local_pc_wrapper, "VP9"); |
| transceiver->SetCodecPreferences(codecs); |
| // Configure SVC, a.k.a. "L3T3_KEY". |
| rtc::scoped_refptr<RtpSenderInterface> sender = transceiver->sender(); |
| RtpParameters parameters = sender->GetParameters(); |
| ASSERT_EQ(parameters.encodings.size(), 1u); |
| parameters.encodings[0].scalability_mode = "L3T3_KEY"; |
| parameters.encodings[0].scale_resolution_down_by = 1; |
| EXPECT_TRUE(sender->SetParameters(parameters).ok()); |
| |
| NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); |
| local_pc_wrapper->WaitForConnection(); |
| remote_pc_wrapper->WaitForConnection(); |
| |
| // Wait until media is flowing. We only expect a single RTP stream. |
| // We expect to see bytes flowing almost immediately on the lowest layer. |
| ASSERT_TRUE_WAIT(HasOutboundRtpBytesSent(local_pc_wrapper, 1u), |
| kDefaultTimeout.ms()); |
| EXPECT_TRUE(OutboundRtpResolutionsAreLessThanOrEqualToExpectations( |
| local_pc_wrapper, {{"", 1280, 720}})); |
| // Verify codec and scalability mode. |
| rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper); |
| std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps = |
| report->GetStatsOfType<RTCOutboundRtpStreamStats>(); |
| ASSERT_THAT(outbound_rtps, SizeIs(1u)); |
| EXPECT_THAT(GetCurrentCodecMimeType(report, *outbound_rtps[0]), |
| StrCaseEq("video/VP9")); |
| EXPECT_THAT(*outbound_rtps[0]->scalability_mode, StrEq("L3T3_KEY")); |
| |
| // GetParameters() is consistent with what we asked for and got. |
| parameters = sender->GetParameters(); |
| ASSERT_EQ(parameters.encodings.size(), 1u); |
| EXPECT_THAT(parameters.encodings[0].scalability_mode, |
| Optional(std::string("L3T3_KEY"))); |
| } |
| |
| // The {active,inactive,inactive} case is technically simulcast but since we |
| // only have one active stream, we're able to do SVC (multiple spatial layers |
| // is not supported if multiple encodings are active). The expected outcome is |
| // the same as above except we end up with two inactive RTP streams which are |
| // observable in GetStats(). |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| VP9_StandardSvcWithSingleActiveEncoding) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); |
| ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); |
| |
| std::vector<cricket::SimulcastLayer> layers = |
| CreateLayers({"f", "h", "q"}, /*active=*/true); |
| rtc::scoped_refptr<RtpTransceiverInterface> transceiver = |
| AddTransceiverWithSimulcastLayers(local_pc_wrapper, remote_pc_wrapper, |
| layers); |
| std::vector<RtpCodecCapability> codecs = |
| GetCapabilitiesAndRestrictToCodec(local_pc_wrapper, "VP9"); |
| transceiver->SetCodecPreferences(codecs); |
| // Configure SVC, a.k.a. "L3T3_KEY". |
| rtc::scoped_refptr<RtpSenderInterface> sender = transceiver->sender(); |
| RtpParameters parameters = sender->GetParameters(); |
| ASSERT_THAT(parameters.encodings, SizeIs(3)); |
| parameters.encodings[0].scalability_mode = "L3T3_KEY"; |
| parameters.encodings[0].scale_resolution_down_by = 1; |
| parameters.encodings[1].active = false; |
| parameters.encodings[2].active = false; |
| EXPECT_TRUE(sender->SetParameters(parameters).ok()); |
| |
| NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); |
| local_pc_wrapper->WaitForConnection(); |
| remote_pc_wrapper->WaitForConnection(); |
| |
| // Since the standard API is configuring simulcast we get three outbound-rtps, |
| // but only one is active. |
| ASSERT_TRUE_WAIT(HasOutboundRtpBytesSent(local_pc_wrapper, 3u, 1u), |
| kDefaultTimeout.ms()); |
| // Wait until scalability mode is reported and expected resolution reached. |
| // Ramp up time is significant. |
| ASSERT_TRUE_WAIT(HasOutboundRtpWithRidAndScalabilityMode( |
| local_pc_wrapper, "f", "L3T3_KEY", 720), |
| kLongTimeoutForRampingUp.ms()); |
| |
| // GetParameters() is consistent with what we asked for and got. |
| parameters = sender->GetParameters(); |
| ASSERT_THAT(parameters.encodings, SizeIs(3)); |
| EXPECT_THAT(parameters.encodings[0].scalability_mode, |
| Optional(std::string("L3T3_KEY"))); |
| EXPECT_FALSE(parameters.encodings[1].scalability_mode.has_value()); |
| EXPECT_FALSE(parameters.encodings[2].scalability_mode.has_value()); |
| } |
| |
| // Exercise common path where `scalability_mode` is not specified until after |
| // negotiation, requring us to recreate the stream when the number of streams |
| // changes from 1 (legacy SVC) to 3 (standard simulcast). |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| VP9_SwitchFromLegacySvcToStandardSingleActiveEncodingSvc) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); |
| ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); |
| |
| std::vector<cricket::SimulcastLayer> layers = |
| CreateLayers({"f", "h", "q"}, /*active=*/true); |
| rtc::scoped_refptr<RtpTransceiverInterface> transceiver = |
| AddTransceiverWithSimulcastLayers(local_pc_wrapper, remote_pc_wrapper, |
| layers); |
| std::vector<RtpCodecCapability> codecs = |
| GetCapabilitiesAndRestrictToCodec(local_pc_wrapper, "VP9"); |
| transceiver->SetCodecPreferences(codecs); |
| |
| // The original negotiation triggers legacy SVC because we didn't specify |
| // any scalability mode. |
| NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); |
| local_pc_wrapper->WaitForConnection(); |
| remote_pc_wrapper->WaitForConnection(); |
| |
| // Switch to the standard mode. Despite only having a single active stream in |
| // both cases, this internally reconfigures from 1 stream to 3 streams. |
| // Test coverage for https://crbug.com/webrtc/15016. |
| rtc::scoped_refptr<RtpSenderInterface> sender = transceiver->sender(); |
| RtpParameters parameters = sender->GetParameters(); |
| ASSERT_THAT(parameters.encodings, SizeIs(3)); |
| parameters.encodings[0].active = true; |
| parameters.encodings[0].scalability_mode = "L2T2_KEY"; |
| parameters.encodings[0].scale_resolution_down_by = 2.0; |
| parameters.encodings[1].active = false; |
| parameters.encodings[1].scalability_mode = absl::nullopt; |
| parameters.encodings[2].active = false; |
| parameters.encodings[2].scalability_mode = absl::nullopt; |
| sender->SetParameters(parameters); |
| |
| // Since the standard API is configuring simulcast we get three outbound-rtps, |
| // but only one is active. |
| ASSERT_TRUE_WAIT(HasOutboundRtpBytesSent(local_pc_wrapper, 3u, 1u), |
| kDefaultTimeout.ms()); |
| // Wait until scalability mode is reported and expected resolution reached. |
| // Ramp up time may be significant. |
| ASSERT_TRUE_WAIT(HasOutboundRtpWithRidAndScalabilityMode( |
| local_pc_wrapper, "f", "L2T2_KEY", 720 / 2), |
| kLongTimeoutForRampingUp.ms()); |
| |
| // GetParameters() does not report any fallback. |
| parameters = sender->GetParameters(); |
| ASSERT_THAT(parameters.encodings, SizeIs(3)); |
| EXPECT_THAT(parameters.encodings[0].scalability_mode, |
| Optional(std::string("L2T2_KEY"))); |
| EXPECT_FALSE(parameters.encodings[1].scalability_mode.has_value()); |
| EXPECT_FALSE(parameters.encodings[2].scalability_mode.has_value()); |
| } |
| |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| VP9_AllLayersInactive_LegacySvc) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); |
| ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); |
| |
| std::vector<cricket::SimulcastLayer> layers = |
| CreateLayers({"f", "h", "q"}, /*active=*/true); |
| rtc::scoped_refptr<RtpTransceiverInterface> transceiver = |
| AddTransceiverWithSimulcastLayers(local_pc_wrapper, remote_pc_wrapper, |
| layers); |
| std::vector<RtpCodecCapability> codecs = |
| GetCapabilitiesAndRestrictToCodec(local_pc_wrapper, "VP9"); |
| transceiver->SetCodecPreferences(codecs); |
| |
| // Legacy SVC mode and all layers inactive. |
| rtc::scoped_refptr<RtpSenderInterface> sender = transceiver->sender(); |
| RtpParameters parameters = sender->GetParameters(); |
| ASSERT_THAT(parameters.encodings, SizeIs(3)); |
| parameters.encodings[0].active = false; |
| parameters.encodings[1].active = false; |
| parameters.encodings[2].active = false; |
| sender->SetParameters(parameters); |
| |
| NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); |
| local_pc_wrapper->WaitForConnection(); |
| remote_pc_wrapper->WaitForConnection(); |
| |
| // Ensure no media is flowing (1 second should be enough). |
| rtc::Thread::Current()->SleepMs(1000); |
| rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper); |
| std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps = |
| report->GetStatsOfType<RTCOutboundRtpStreamStats>(); |
| ASSERT_THAT(outbound_rtps, SizeIs(1u)); |
| EXPECT_EQ(*outbound_rtps[0]->bytes_sent, 0u); |
| } |
| |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| VP9_AllLayersInactive_StandardSvc) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); |
| ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); |
| |
| std::vector<cricket::SimulcastLayer> layers = |
| CreateLayers({"f", "h", "q"}, /*active=*/true); |
| rtc::scoped_refptr<RtpTransceiverInterface> transceiver = |
| AddTransceiverWithSimulcastLayers(local_pc_wrapper, remote_pc_wrapper, |
| layers); |
| std::vector<RtpCodecCapability> codecs = |
| GetCapabilitiesAndRestrictToCodec(local_pc_wrapper, "VP9"); |
| transceiver->SetCodecPreferences(codecs); |
| |
| // Standard mode and all layers inactive. |
| rtc::scoped_refptr<RtpSenderInterface> sender = transceiver->sender(); |
| RtpParameters parameters = sender->GetParameters(); |
| ASSERT_THAT(parameters.encodings, SizeIs(3)); |
| parameters.encodings[0].scalability_mode = "L3T3_KEY"; |
| parameters.encodings[0].scale_resolution_down_by = 1; |
| parameters.encodings[0].active = false; |
| parameters.encodings[1].active = false; |
| parameters.encodings[2].active = false; |
| sender->SetParameters(parameters); |
| |
| NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); |
| local_pc_wrapper->WaitForConnection(); |
| remote_pc_wrapper->WaitForConnection(); |
| |
| // Ensure no media is flowing (1 second should be enough). |
| rtc::Thread::Current()->SleepMs(1000); |
| rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper); |
| std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps = |
| report->GetStatsOfType<RTCOutboundRtpStreamStats>(); |
| ASSERT_THAT(outbound_rtps, SizeIs(3u)); |
| EXPECT_EQ(*outbound_rtps[0]->bytes_sent, 0u); |
| EXPECT_EQ(*outbound_rtps[1]->bytes_sent, 0u); |
| EXPECT_EQ(*outbound_rtps[2]->bytes_sent, 0u); |
| } |
| |
| TEST_F(PeerConnectionEncodingsIntegrationTest, VP9_TargetBitrate_LegacyL1T3) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); |
| ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); |
| |
| std::vector<cricket::SimulcastLayer> layers = |
| CreateLayers({"f", "h", "q"}, /*active=*/true); |
| rtc::scoped_refptr<RtpTransceiverInterface> transceiver = |
| AddTransceiverWithSimulcastLayers(local_pc_wrapper, remote_pc_wrapper, |
| layers); |
| std::vector<RtpCodecCapability> codecs = |
| GetCapabilitiesAndRestrictToCodec(local_pc_wrapper, "VP9"); |
| transceiver->SetCodecPreferences(codecs); |
| |
| // In legacy SVC, disabling the bottom two layers encodings is interpreted as |
| // disabling the bottom two spatial layers resulting in L1T3. |
| rtc::scoped_refptr<RtpSenderInterface> sender = transceiver->sender(); |
| RtpParameters parameters = sender->GetParameters(); |
| parameters.encodings[0].active = false; |
| parameters.encodings[1].active = false; |
| parameters.encodings[2].active = true; |
| sender->SetParameters(parameters); |
| |
| NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); |
| local_pc_wrapper->WaitForConnection(); |
| remote_pc_wrapper->WaitForConnection(); |
| |
| // Wait until 720p L1T3 has ramped up to 720p. It may take additional time |
| // for the target bitrate to reach its maximum. |
| ASSERT_TRUE_WAIT(HasOutboundRtpWithRidAndScalabilityMode(local_pc_wrapper, |
| "f", "L1T3", 720), |
| kLongTimeoutForRampingUp.ms()); |
| |
| // The target bitrate typically reaches `kVp9ExpectedMaxBitrateForL1T3` |
| // in a short period of time. However to reduce risk of flakiness in bot |
| // environments, this test only fails if we we exceed the expected target. |
| rtc::Thread::Current()->SleepMs(1000); |
| rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper); |
| std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps = |
| report->GetStatsOfType<RTCOutboundRtpStreamStats>(); |
| ASSERT_THAT(outbound_rtps, SizeIs(1)); |
| DataRate target_bitrate = |
| DataRate::BitsPerSec(*outbound_rtps[0]->target_bitrate); |
| EXPECT_LE(target_bitrate.kbps(), kVp9ExpectedMaxBitrateForL1T3.kbps()); |
| } |
| |
| // Test coverage for https://crbug.com/1455039. |
| TEST_F(PeerConnectionEncodingsIntegrationTest, VP9_TargetBitrate_StandardL1T3) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); |
| ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); |
| |
| std::vector<cricket::SimulcastLayer> layers = |
| CreateLayers({"f", "h", "q"}, /*active=*/true); |
| rtc::scoped_refptr<RtpTransceiverInterface> transceiver = |
| AddTransceiverWithSimulcastLayers(local_pc_wrapper, remote_pc_wrapper, |
| layers); |
| std::vector<RtpCodecCapability> codecs = |
| GetCapabilitiesAndRestrictToCodec(local_pc_wrapper, "VP9"); |
| transceiver->SetCodecPreferences(codecs); |
| |
| // With standard APIs, L1T3 is explicitly specified and the encodings refers |
| // to the RTP streams, not the spatial layers. The end result should be |
| // equivalent to the legacy L1T3 case. |
| rtc::scoped_refptr<RtpSenderInterface> sender = transceiver->sender(); |
| RtpParameters parameters = sender->GetParameters(); |
| parameters.encodings[0].active = true; |
| parameters.encodings[0].scale_resolution_down_by = 1.0; |
| parameters.encodings[0].scalability_mode = "L1T3"; |
| parameters.encodings[1].active = false; |
| parameters.encodings[2].active = false; |
| sender->SetParameters(parameters); |
| |
| NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); |
| local_pc_wrapper->WaitForConnection(); |
| remote_pc_wrapper->WaitForConnection(); |
| |
| // Wait until 720p L1T3 has ramped up to 720p. It may take additional time |
| // for the target bitrate to reach its maximum. |
| ASSERT_TRUE_WAIT(HasOutboundRtpWithRidAndScalabilityMode(local_pc_wrapper, |
| "f", "L1T3", 720), |
| kLongTimeoutForRampingUp.ms()); |
| |
| // The target bitrate typically reaches `kVp9ExpectedMaxBitrateForL1T3` |
| // in a short period of time. However to reduce risk of flakiness in bot |
| // environments, this test only fails if we we exceed the expected target. |
| rtc::Thread::Current()->SleepMs(1000); |
| rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper); |
| std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps = |
| report->GetStatsOfType<RTCOutboundRtpStreamStats>(); |
| ASSERT_THAT(outbound_rtps, SizeIs(3)); |
| auto* outbound_rtp = FindOutboundRtpByRid(outbound_rtps, "f"); |
| ASSERT_TRUE(outbound_rtp); |
| DataRate target_bitrate = DataRate::BitsPerSec(*outbound_rtp->target_bitrate); |
| EXPECT_LE(target_bitrate.kbps(), kVp9ExpectedMaxBitrateForL1T3.kbps()); |
| } |
| |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| SimulcastProducesUniqueSsrcAndRtxSsrcs) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); |
| ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); |
| |
| std::vector<cricket::SimulcastLayer> layers = |
| CreateLayers({"f", "h", "q"}, /*active=*/true); |
| rtc::scoped_refptr<RtpTransceiverInterface> transceiver = |
| AddTransceiverWithSimulcastLayers(local_pc_wrapper, remote_pc_wrapper, |
| layers); |
| std::vector<RtpCodecCapability> codecs = |
| GetCapabilitiesAndRestrictToCodec(local_pc_wrapper, "VP8"); |
| transceiver->SetCodecPreferences(codecs); |
| |
| NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); |
| local_pc_wrapper->WaitForConnection(); |
| remote_pc_wrapper->WaitForConnection(); |
| |
| // Wait until media is flowing on all three layers. |
| // Ramp up time is needed before all three layers are sending. |
| ASSERT_TRUE_WAIT(HasOutboundRtpBytesSent(local_pc_wrapper, 3u), |
| kLongTimeoutForRampingUp.ms()); |
| // Verify SSRCs and RTX SSRCs. |
| rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper); |
| std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps = |
| report->GetStatsOfType<RTCOutboundRtpStreamStats>(); |
| ASSERT_THAT(outbound_rtps, SizeIs(3u)); |
| |
| std::set<uint32_t> ssrcs; |
| std::set<uint32_t> rtx_ssrcs; |
| for (const auto& outbound_rtp : outbound_rtps) { |
| ASSERT_TRUE(outbound_rtp->ssrc.has_value()); |
| ASSERT_TRUE(outbound_rtp->rtx_ssrc.has_value()); |
| ssrcs.insert(*outbound_rtp->ssrc); |
| rtx_ssrcs.insert(*outbound_rtp->rtx_ssrc); |
| } |
| EXPECT_EQ(ssrcs.size(), 3u); |
| EXPECT_EQ(rtx_ssrcs.size(), 3u); |
| } |
| |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| EncodingParameterCodecIsEmptyWhenCreatedAudio) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| |
| auto transceiver_or_error = |
| local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO); |
| rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver = |
| transceiver_or_error.MoveValue(); |
| webrtc::RtpParameters parameters = |
| audio_transceiver->sender()->GetParameters(); |
| EXPECT_FALSE(parameters.encodings[0].codec.has_value()); |
| } |
| |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| EncodingParameterCodecIsEmptyWhenCreatedVideo) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| |
| auto transceiver_or_error = |
| local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO); |
| rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver = |
| transceiver_or_error.MoveValue(); |
| webrtc::RtpParameters parameters = |
| video_transceiver->sender()->GetParameters(); |
| EXPECT_FALSE(parameters.encodings[0].codec.has_value()); |
| } |
| |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| EncodingParameterCodecIsSetByAddTransceiverAudio) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); |
| ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); |
| |
| rtc::scoped_refptr<webrtc::MediaStreamInterface> stream = |
| local_pc_wrapper->GetUserMedia( |
| /*audio=*/true, {}, /*video=*/false, {}); |
| rtc::scoped_refptr<AudioTrackInterface> track = stream->GetAudioTracks()[0]; |
| |
| absl::optional<webrtc::RtpCodecCapability> pcmu = |
| local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO, |
| "pcmu"); |
| ASSERT_TRUE(pcmu); |
| |
| webrtc::RtpTransceiverInit init; |
| init.direction = webrtc::RtpTransceiverDirection::kSendOnly; |
| webrtc::RtpEncodingParameters encoding_parameters; |
| encoding_parameters.codec = pcmu; |
| init.send_encodings.push_back(encoding_parameters); |
| |
| auto transceiver_or_error = |
| local_pc_wrapper->pc()->AddTransceiver(track, init); |
| rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver = |
| transceiver_or_error.MoveValue(); |
| webrtc::RtpParameters parameters = |
| audio_transceiver->sender()->GetParameters(); |
| EXPECT_EQ(*parameters.encodings[0].codec, *pcmu); |
| |
| NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); |
| local_pc_wrapper->WaitForConnection(); |
| remote_pc_wrapper->WaitForConnection(); |
| |
| rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper); |
| std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps = |
| report->GetStatsOfType<RTCOutboundRtpStreamStats>(); |
| ASSERT_EQ(outbound_rtps.size(), 1u); |
| std::string codec_name = GetCurrentCodecMimeType(report, *outbound_rtps[0]); |
| EXPECT_STRCASEEQ(("audio/" + pcmu->name).c_str(), codec_name.c_str()); |
| } |
| |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| EncodingParameterCodecIsSetByAddTransceiverVideo) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); |
| ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); |
| |
| rtc::scoped_refptr<webrtc::MediaStreamInterface> stream = |
| local_pc_wrapper->GetUserMedia( |
| /*audio=*/false, {}, /*video=*/true, {.width = 1280, .height = 720}); |
| rtc::scoped_refptr<VideoTrackInterface> track = stream->GetVideoTracks()[0]; |
| |
| absl::optional<webrtc::RtpCodecCapability> vp9 = |
| local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO, |
| "vp9"); |
| ASSERT_TRUE(vp9); |
| |
| webrtc::RtpTransceiverInit init; |
| init.direction = webrtc::RtpTransceiverDirection::kSendOnly; |
| webrtc::RtpEncodingParameters encoding_parameters; |
| encoding_parameters.codec = vp9; |
| encoding_parameters.scalability_mode = "L3T3"; |
| init.send_encodings.push_back(encoding_parameters); |
| |
| auto transceiver_or_error = |
| local_pc_wrapper->pc()->AddTransceiver(track, init); |
| rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver = |
| transceiver_or_error.MoveValue(); |
| webrtc::RtpParameters parameters = |
| audio_transceiver->sender()->GetParameters(); |
| EXPECT_EQ(*parameters.encodings[0].codec, *vp9); |
| |
| NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); |
| local_pc_wrapper->WaitForConnection(); |
| remote_pc_wrapper->WaitForConnection(); |
| |
| EXPECT_TRUE_WAIT( |
| IsCodecIdDifferentWithScalabilityMode(local_pc_wrapper, 0, "", "L3T3"), |
| kDefaultTimeout.ms()); |
| |
| rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper); |
| std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps = |
| report->GetStatsOfType<RTCOutboundRtpStreamStats>(); |
| ASSERT_EQ(outbound_rtps.size(), 1u); |
| std::string codec_name = GetCurrentCodecMimeType(report, *outbound_rtps[0]); |
| EXPECT_STRCASEEQ(("video/" + vp9->name).c_str(), codec_name.c_str()); |
| EXPECT_EQ(outbound_rtps[0]->scalability_mode.value(), "L3T3"); |
| } |
| |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| EncodingParameterCodecIsSetBySetParametersBeforeNegotiationAudio) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); |
| ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); |
| |
| rtc::scoped_refptr<webrtc::MediaStreamInterface> stream = |
| local_pc_wrapper->GetUserMedia( |
| /*audio=*/true, {}, /*video=*/false, {}); |
| rtc::scoped_refptr<AudioTrackInterface> track = stream->GetAudioTracks()[0]; |
| |
| absl::optional<webrtc::RtpCodecCapability> pcmu = |
| local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO, |
| "pcmu"); |
| |
| auto transceiver_or_error = local_pc_wrapper->pc()->AddTransceiver(track); |
| rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver = |
| transceiver_or_error.MoveValue(); |
| webrtc::RtpParameters parameters = |
| audio_transceiver->sender()->GetParameters(); |
| parameters.encodings[0].codec = pcmu; |
| EXPECT_TRUE(audio_transceiver->sender()->SetParameters(parameters).ok()); |
| |
| parameters = audio_transceiver->sender()->GetParameters(); |
| EXPECT_EQ(parameters.encodings[0].codec, pcmu); |
| |
| NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); |
| local_pc_wrapper->WaitForConnection(); |
| remote_pc_wrapper->WaitForConnection(); |
| |
| rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper); |
| std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps = |
| report->GetStatsOfType<RTCOutboundRtpStreamStats>(); |
| ASSERT_EQ(outbound_rtps.size(), 1u); |
| std::string codec_name = GetCurrentCodecMimeType(report, *outbound_rtps[0]); |
| EXPECT_STRCASEEQ(("audio/" + pcmu->name).c_str(), codec_name.c_str()); |
| } |
| |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| EncodingParameterCodecIsSetBySetParametersAfterNegotiationAudio) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); |
| ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); |
| |
| rtc::scoped_refptr<webrtc::MediaStreamInterface> stream = |
| local_pc_wrapper->GetUserMedia( |
| /*audio=*/true, {}, /*video=*/false, {}); |
| rtc::scoped_refptr<AudioTrackInterface> track = stream->GetAudioTracks()[0]; |
| |
| absl::optional<webrtc::RtpCodecCapability> pcmu = |
| local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO, |
| "pcmu"); |
| |
| auto transceiver_or_error = local_pc_wrapper->pc()->AddTransceiver(track); |
| rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver = |
| transceiver_or_error.MoveValue(); |
| |
| NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); |
| local_pc_wrapper->WaitForConnection(); |
| remote_pc_wrapper->WaitForConnection(); |
| |
| rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper); |
| std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps = |
| report->GetStatsOfType<RTCOutboundRtpStreamStats>(); |
| ASSERT_EQ(outbound_rtps.size(), 1u); |
| std::string codec_name = GetCurrentCodecMimeType(report, *outbound_rtps[0]); |
| EXPECT_STRCASENE(("audio/" + pcmu->name).c_str(), codec_name.c_str()); |
| std::string last_codec_id = outbound_rtps[0]->codec_id.value(); |
| |
| webrtc::RtpParameters parameters = |
| audio_transceiver->sender()->GetParameters(); |
| parameters.encodings[0].codec = pcmu; |
| EXPECT_TRUE(audio_transceiver->sender()->SetParameters(parameters).ok()); |
| |
| parameters = audio_transceiver->sender()->GetParameters(); |
| EXPECT_EQ(parameters.encodings[0].codec, pcmu); |
| |
| EXPECT_TRUE_WAIT(IsCodecIdDifferent(local_pc_wrapper, 0, last_codec_id), |
| kDefaultTimeout.ms()); |
| |
| report = GetStats(local_pc_wrapper); |
| outbound_rtps = report->GetStatsOfType<RTCOutboundRtpStreamStats>(); |
| ASSERT_EQ(outbound_rtps.size(), 1u); |
| codec_name = GetCurrentCodecMimeType(report, *outbound_rtps[0]); |
| EXPECT_STRCASEEQ(("audio/" + pcmu->name).c_str(), codec_name.c_str()); |
| } |
| |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| EncodingParameterCodecIsSetBySetParametersBeforeNegotiationVideo) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); |
| ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); |
| |
| rtc::scoped_refptr<webrtc::MediaStreamInterface> stream = |
| local_pc_wrapper->GetUserMedia( |
| /*audio=*/false, {}, /*video=*/true, {.width = 1280, .height = 720}); |
| rtc::scoped_refptr<VideoTrackInterface> track = stream->GetVideoTracks()[0]; |
| |
| absl::optional<webrtc::RtpCodecCapability> vp9 = |
| local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO, |
| "vp9"); |
| |
| auto transceiver_or_error = local_pc_wrapper->pc()->AddTransceiver(track); |
| rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver = |
| transceiver_or_error.MoveValue(); |
| webrtc::RtpParameters parameters = |
| video_transceiver->sender()->GetParameters(); |
| parameters.encodings[0].codec = vp9; |
| parameters.encodings[0].scalability_mode = "L3T3"; |
| EXPECT_TRUE(video_transceiver->sender()->SetParameters(parameters).ok()); |
| |
| parameters = video_transceiver->sender()->GetParameters(); |
| EXPECT_EQ(parameters.encodings[0].codec, vp9); |
| EXPECT_EQ(parameters.encodings[0].scalability_mode, "L3T3"); |
| |
| NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); |
| local_pc_wrapper->WaitForConnection(); |
| remote_pc_wrapper->WaitForConnection(); |
| |
| EXPECT_TRUE_WAIT( |
| IsCodecIdDifferentWithScalabilityMode(local_pc_wrapper, 0, "", "L3T3"), |
| kDefaultTimeout.ms()); |
| rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper); |
| std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps = |
| report->GetStatsOfType<RTCOutboundRtpStreamStats>(); |
| ASSERT_EQ(outbound_rtps.size(), 1u); |
| std::string codec_name = GetCurrentCodecMimeType(report, *outbound_rtps[0]); |
| EXPECT_STRCASEEQ(("video/" + vp9->name).c_str(), codec_name.c_str()); |
| EXPECT_EQ(outbound_rtps[0]->scalability_mode.ValueOrDefault(""), "L3T3"); |
| } |
| |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| EncodingParameterCodecIsSetBySetParametersAfterNegotiationVideo) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); |
| ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); |
| |
| rtc::scoped_refptr<webrtc::MediaStreamInterface> stream = |
| local_pc_wrapper->GetUserMedia( |
| /*audio=*/false, {}, /*video=*/true, {.width = 1280, .height = 720}); |
| rtc::scoped_refptr<VideoTrackInterface> track = stream->GetVideoTracks()[0]; |
| |
| absl::optional<webrtc::RtpCodecCapability> vp9 = |
| local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO, |
| "vp9"); |
| |
| auto transceiver_or_error = local_pc_wrapper->pc()->AddTransceiver(track); |
| rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver = |
| transceiver_or_error.MoveValue(); |
| |
| NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); |
| local_pc_wrapper->WaitForConnection(); |
| remote_pc_wrapper->WaitForConnection(); |
| |
| rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper); |
| std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps = |
| report->GetStatsOfType<RTCOutboundRtpStreamStats>(); |
| ASSERT_EQ(outbound_rtps.size(), 1u); |
| std::string codec_name = GetCurrentCodecMimeType(report, *outbound_rtps[0]); |
| EXPECT_STRCASENE(("audio/" + vp9->name).c_str(), codec_name.c_str()); |
| std::string last_codec_id = outbound_rtps[0]->codec_id.value(); |
| |
| webrtc::RtpParameters parameters = |
| video_transceiver->sender()->GetParameters(); |
| parameters.encodings[0].codec = vp9; |
| parameters.encodings[0].scalability_mode = "L3T3"; |
| EXPECT_TRUE(video_transceiver->sender()->SetParameters(parameters).ok()); |
| |
| parameters = video_transceiver->sender()->GetParameters(); |
| EXPECT_EQ(parameters.encodings[0].codec, vp9); |
| EXPECT_EQ(parameters.encodings[0].scalability_mode, "L3T3"); |
| |
| EXPECT_TRUE_WAIT(IsCodecIdDifferentWithScalabilityMode(local_pc_wrapper, 0, |
| last_codec_id, "L3T3"), |
| kDefaultTimeout.ms()); |
| |
| report = GetStats(local_pc_wrapper); |
| outbound_rtps = report->GetStatsOfType<RTCOutboundRtpStreamStats>(); |
| ASSERT_EQ(outbound_rtps.size(), 1u); |
| codec_name = GetCurrentCodecMimeType(report, *outbound_rtps[0]); |
| EXPECT_STRCASEEQ(("video/" + vp9->name).c_str(), codec_name.c_str()); |
| EXPECT_EQ(outbound_rtps[0]->scalability_mode.value(), "L3T3"); |
| } |
| |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| AddTransceiverRejectsUnknownCodecParameterAudio) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| |
| webrtc::RtpCodec dummy_codec; |
| dummy_codec.kind = cricket::MEDIA_TYPE_AUDIO; |
| dummy_codec.name = "FOOBAR"; |
| dummy_codec.clock_rate = 90000; |
| dummy_codec.num_channels = 2; |
| |
| webrtc::RtpTransceiverInit init; |
| init.direction = webrtc::RtpTransceiverDirection::kSendOnly; |
| webrtc::RtpEncodingParameters encoding_parameters; |
| encoding_parameters.codec = dummy_codec; |
| init.send_encodings.push_back(encoding_parameters); |
| |
| auto transceiver_or_error = |
| local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init); |
| EXPECT_FALSE(transceiver_or_error.ok()); |
| EXPECT_EQ(transceiver_or_error.error().type(), |
| RTCErrorType::UNSUPPORTED_OPERATION); |
| } |
| |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| AddTransceiverRejectsUnknownCodecParameterVideo) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| |
| webrtc::RtpCodec dummy_codec; |
| dummy_codec.kind = cricket::MEDIA_TYPE_VIDEO; |
| dummy_codec.name = "FOOBAR"; |
| dummy_codec.clock_rate = 90000; |
| |
| webrtc::RtpTransceiverInit init; |
| init.direction = webrtc::RtpTransceiverDirection::kSendOnly; |
| webrtc::RtpEncodingParameters encoding_parameters; |
| encoding_parameters.codec = dummy_codec; |
| init.send_encodings.push_back(encoding_parameters); |
| |
| auto transceiver_or_error = |
| local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init); |
| EXPECT_FALSE(transceiver_or_error.ok()); |
| EXPECT_EQ(transceiver_or_error.error().type(), |
| RTCErrorType::UNSUPPORTED_OPERATION); |
| } |
| |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| SetParametersRejectsUnknownCodecParameterAudio) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| |
| webrtc::RtpCodec dummy_codec; |
| dummy_codec.kind = cricket::MEDIA_TYPE_AUDIO; |
| dummy_codec.name = "FOOBAR"; |
| dummy_codec.clock_rate = 90000; |
| dummy_codec.num_channels = 2; |
| |
| auto transceiver_or_error = |
| local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO); |
| ASSERT_TRUE(transceiver_or_error.ok()); |
| rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver = |
| transceiver_or_error.MoveValue(); |
| |
| webrtc::RtpParameters parameters = |
| audio_transceiver->sender()->GetParameters(); |
| parameters.encodings[0].codec = dummy_codec; |
| RTCError error = audio_transceiver->sender()->SetParameters(parameters); |
| EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION); |
| } |
| |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| SetParametersRejectsUnknownCodecParameterVideo) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| |
| webrtc::RtpCodec dummy_codec; |
| dummy_codec.kind = cricket::MEDIA_TYPE_VIDEO; |
| dummy_codec.name = "FOOBAR"; |
| dummy_codec.clock_rate = 90000; |
| |
| auto transceiver_or_error = |
| local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO); |
| ASSERT_TRUE(transceiver_or_error.ok()); |
| rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver = |
| transceiver_or_error.MoveValue(); |
| |
| webrtc::RtpParameters parameters = |
| video_transceiver->sender()->GetParameters(); |
| parameters.encodings[0].codec = dummy_codec; |
| RTCError error = video_transceiver->sender()->SetParameters(parameters); |
| EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION); |
| } |
| |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| SetParametersRejectsNonPreferredCodecParameterAudio) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| |
| absl::optional<webrtc::RtpCodecCapability> opus = |
| local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO, |
| "opus"); |
| ASSERT_TRUE(opus); |
| |
| std::vector<webrtc::RtpCodecCapability> not_opus_codecs = |
| local_pc_wrapper->pc_factory() |
| ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO) |
| .codecs; |
| not_opus_codecs.erase( |
| std::remove_if(not_opus_codecs.begin(), not_opus_codecs.end(), |
| [&](const auto& codec) { |
| return absl::EqualsIgnoreCase(codec.name, opus->name); |
| }), |
| not_opus_codecs.end()); |
| |
| auto transceiver_or_error = |
| local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO); |
| ASSERT_TRUE(transceiver_or_error.ok()); |
| rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver = |
| transceiver_or_error.MoveValue(); |
| ASSERT_TRUE(audio_transceiver->SetCodecPreferences(not_opus_codecs).ok()); |
| |
| webrtc::RtpParameters parameters = |
| audio_transceiver->sender()->GetParameters(); |
| parameters.encodings[0].codec = opus; |
| RTCError error = audio_transceiver->sender()->SetParameters(parameters); |
| EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION); |
| } |
| |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| SetParametersRejectsNonPreferredCodecParameterVideo) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| |
| absl::optional<webrtc::RtpCodecCapability> vp8 = |
| local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO, |
| "vp8"); |
| ASSERT_TRUE(vp8); |
| |
| std::vector<webrtc::RtpCodecCapability> not_vp8_codecs = |
| local_pc_wrapper->pc_factory() |
| ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO) |
| .codecs; |
| not_vp8_codecs.erase( |
| std::remove_if(not_vp8_codecs.begin(), not_vp8_codecs.end(), |
| [&](const auto& codec) { |
| return absl::EqualsIgnoreCase(codec.name, vp8->name); |
| }), |
| not_vp8_codecs.end()); |
| |
| auto transceiver_or_error = |
| local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO); |
| ASSERT_TRUE(transceiver_or_error.ok()); |
| rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver = |
| transceiver_or_error.MoveValue(); |
| ASSERT_TRUE(video_transceiver->SetCodecPreferences(not_vp8_codecs).ok()); |
| |
| webrtc::RtpParameters parameters = |
| video_transceiver->sender()->GetParameters(); |
| parameters.encodings[0].codec = vp8; |
| RTCError error = video_transceiver->sender()->SetParameters(parameters); |
| EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION); |
| } |
| |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| SetParametersRejectsNonNegotiatedCodecParameterAudio) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); |
| ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); |
| |
| absl::optional<webrtc::RtpCodecCapability> opus = |
| local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO, |
| "opus"); |
| ASSERT_TRUE(opus); |
| |
| std::vector<webrtc::RtpCodecCapability> not_opus_codecs = |
| local_pc_wrapper->pc_factory() |
| ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO) |
| .codecs; |
| not_opus_codecs.erase( |
| std::remove_if(not_opus_codecs.begin(), not_opus_codecs.end(), |
| [&](const auto& codec) { |
| return absl::EqualsIgnoreCase(codec.name, opus->name); |
| }), |
| not_opus_codecs.end()); |
| |
| auto transceiver_or_error = |
| local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO); |
| ASSERT_TRUE(transceiver_or_error.ok()); |
| rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver = |
| transceiver_or_error.MoveValue(); |
| ASSERT_TRUE(audio_transceiver->SetCodecPreferences(not_opus_codecs).ok()); |
| |
| NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); |
| local_pc_wrapper->WaitForConnection(); |
| remote_pc_wrapper->WaitForConnection(); |
| |
| webrtc::RtpParameters parameters = |
| audio_transceiver->sender()->GetParameters(); |
| parameters.encodings[0].codec = opus; |
| RTCError error = audio_transceiver->sender()->SetParameters(parameters); |
| EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION); |
| } |
| |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| SetParametersRejectsNonRemotelyNegotiatedCodecParameterAudio) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); |
| ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); |
| |
| absl::optional<webrtc::RtpCodecCapability> opus = |
| local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO, |
| "opus"); |
| ASSERT_TRUE(opus); |
| |
| std::vector<webrtc::RtpCodecCapability> not_opus_codecs = |
| local_pc_wrapper->pc_factory() |
| ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO) |
| .codecs; |
| not_opus_codecs.erase( |
| std::remove_if(not_opus_codecs.begin(), not_opus_codecs.end(), |
| [&](const auto& codec) { |
| return absl::EqualsIgnoreCase(codec.name, opus->name); |
| }), |
| not_opus_codecs.end()); |
| |
| auto transceiver_or_error = |
| local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO); |
| ASSERT_TRUE(transceiver_or_error.ok()); |
| rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver = |
| transceiver_or_error.MoveValue(); |
| |
| // Negotiation, create offer and apply it |
| std::unique_ptr<SessionDescriptionInterface> offer = |
| CreateOffer(local_pc_wrapper); |
| rtc::scoped_refptr<MockSetSessionDescriptionObserver> p1 = |
| SetLocalDescription(local_pc_wrapper, offer.get()); |
| rtc::scoped_refptr<MockSetSessionDescriptionObserver> p2 = |
| SetRemoteDescription(remote_pc_wrapper, offer.get()); |
| EXPECT_TRUE(Await({p1, p2})); |
| |
| // Update the remote transceiver to reject Opus |
| std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> remote_transceivers = |
| remote_pc_wrapper->pc()->GetTransceivers(); |
| ASSERT_TRUE(!remote_transceivers.empty()); |
| rtc::scoped_refptr<RtpTransceiverInterface> remote_audio_transceiver = |
| remote_transceivers[0]; |
| ASSERT_TRUE( |
| remote_audio_transceiver->SetCodecPreferences(not_opus_codecs).ok()); |
| |
| // Create answer and apply it |
| std::unique_ptr<SessionDescriptionInterface> answer = |
| CreateAnswer(remote_pc_wrapper); |
| p1 = SetLocalDescription(remote_pc_wrapper, answer.get()); |
| p2 = SetRemoteDescription(local_pc_wrapper, answer.get()); |
| EXPECT_TRUE(Await({p1, p2})); |
| |
| local_pc_wrapper->WaitForConnection(); |
| remote_pc_wrapper->WaitForConnection(); |
| |
| webrtc::RtpParameters parameters = |
| audio_transceiver->sender()->GetParameters(); |
| parameters.encodings[0].codec = opus; |
| RTCError error = audio_transceiver->sender()->SetParameters(parameters); |
| EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION); |
| } |
| |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| SetParametersRejectsNonNegotiatedCodecParameterVideo) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); |
| ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); |
| |
| absl::optional<webrtc::RtpCodecCapability> vp8 = |
| local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO, |
| "vp8"); |
| ASSERT_TRUE(vp8); |
| |
| std::vector<webrtc::RtpCodecCapability> not_vp8_codecs = |
| local_pc_wrapper->pc_factory() |
| ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO) |
| .codecs; |
| not_vp8_codecs.erase( |
| std::remove_if(not_vp8_codecs.begin(), not_vp8_codecs.end(), |
| [&](const auto& codec) { |
| return absl::EqualsIgnoreCase(codec.name, vp8->name); |
| }), |
| not_vp8_codecs.end()); |
| |
| auto transceiver_or_error = |
| local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO); |
| ASSERT_TRUE(transceiver_or_error.ok()); |
| rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver = |
| transceiver_or_error.MoveValue(); |
| ASSERT_TRUE(video_transceiver->SetCodecPreferences(not_vp8_codecs).ok()); |
| |
| NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); |
| local_pc_wrapper->WaitForConnection(); |
| remote_pc_wrapper->WaitForConnection(); |
| |
| webrtc::RtpParameters parameters = |
| video_transceiver->sender()->GetParameters(); |
| parameters.encodings[0].codec = vp8; |
| RTCError error = video_transceiver->sender()->SetParameters(parameters); |
| EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION); |
| } |
| |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| SetParametersRejectsNonRemotelyNegotiatedCodecParameterVideo) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); |
| ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); |
| |
| absl::optional<webrtc::RtpCodecCapability> vp8 = |
| local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO, |
| "vp8"); |
| ASSERT_TRUE(vp8); |
| |
| std::vector<webrtc::RtpCodecCapability> not_vp8_codecs = |
| local_pc_wrapper->pc_factory() |
| ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO) |
| .codecs; |
| not_vp8_codecs.erase( |
| std::remove_if(not_vp8_codecs.begin(), not_vp8_codecs.end(), |
| [&](const auto& codec) { |
| return absl::EqualsIgnoreCase(codec.name, vp8->name); |
| }), |
| not_vp8_codecs.end()); |
| |
| auto transceiver_or_error = |
| local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO); |
| ASSERT_TRUE(transceiver_or_error.ok()); |
| rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver = |
| transceiver_or_error.MoveValue(); |
| |
| // Negotiation, create offer and apply it |
| std::unique_ptr<SessionDescriptionInterface> offer = |
| CreateOffer(local_pc_wrapper); |
| rtc::scoped_refptr<MockSetSessionDescriptionObserver> p1 = |
| SetLocalDescription(local_pc_wrapper, offer.get()); |
| rtc::scoped_refptr<MockSetSessionDescriptionObserver> p2 = |
| SetRemoteDescription(remote_pc_wrapper, offer.get()); |
| EXPECT_TRUE(Await({p1, p2})); |
| |
| // Update the remote transceiver to reject VP8 |
| std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> remote_transceivers = |
| remote_pc_wrapper->pc()->GetTransceivers(); |
| ASSERT_TRUE(!remote_transceivers.empty()); |
| rtc::scoped_refptr<RtpTransceiverInterface> remote_video_transceiver = |
| remote_transceivers[0]; |
| ASSERT_TRUE( |
| remote_video_transceiver->SetCodecPreferences(not_vp8_codecs).ok()); |
| |
| // Create answer and apply it |
| std::unique_ptr<SessionDescriptionInterface> answer = |
| CreateAnswer(remote_pc_wrapper); |
| p1 = SetLocalDescription(remote_pc_wrapper, answer.get()); |
| p2 = SetRemoteDescription(local_pc_wrapper, answer.get()); |
| EXPECT_TRUE(Await({p1, p2})); |
| |
| local_pc_wrapper->WaitForConnection(); |
| remote_pc_wrapper->WaitForConnection(); |
| |
| webrtc::RtpParameters parameters = |
| video_transceiver->sender()->GetParameters(); |
| parameters.encodings[0].codec = vp8; |
| RTCError error = video_transceiver->sender()->SetParameters(parameters); |
| EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION); |
| } |
| |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| EncodingParametersCodecRemovedAfterNegotiationAudio) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); |
| ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); |
| |
| absl::optional<webrtc::RtpCodecCapability> opus = |
| local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO, |
| "opus"); |
| ASSERT_TRUE(opus); |
| |
| std::vector<webrtc::RtpCodecCapability> not_opus_codecs = |
| local_pc_wrapper->pc_factory() |
| ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO) |
| .codecs; |
| not_opus_codecs.erase( |
| std::remove_if(not_opus_codecs.begin(), not_opus_codecs.end(), |
| [&](const auto& codec) { |
| return absl::EqualsIgnoreCase(codec.name, opus->name); |
| }), |
| not_opus_codecs.end()); |
| |
| webrtc::RtpTransceiverInit init; |
| init.direction = webrtc::RtpTransceiverDirection::kSendOnly; |
| webrtc::RtpEncodingParameters encoding_parameters; |
| encoding_parameters.codec = opus; |
| init.send_encodings.push_back(encoding_parameters); |
| |
| auto transceiver_or_error = |
| local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init); |
| ASSERT_TRUE(transceiver_or_error.ok()); |
| rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver = |
| transceiver_or_error.MoveValue(); |
| |
| NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); |
| local_pc_wrapper->WaitForConnection(); |
| remote_pc_wrapper->WaitForConnection(); |
| |
| webrtc::RtpParameters parameters = |
| audio_transceiver->sender()->GetParameters(); |
| EXPECT_EQ(parameters.encodings[0].codec, opus); |
| |
| ASSERT_TRUE(audio_transceiver->SetCodecPreferences(not_opus_codecs).ok()); |
| NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); |
| |
| parameters = audio_transceiver->sender()->GetParameters(); |
| EXPECT_FALSE(parameters.encodings[0].codec); |
| } |
| |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| EncodingParametersRedEnabledBeforeNegotiationAudio) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); |
| ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); |
| |
| std::vector<webrtc::RtpCodecCapability> send_codecs = |
| local_pc_wrapper->pc_factory() |
| ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO) |
| .codecs; |
| |
| absl::optional<webrtc::RtpCodecCapability> opus = |
| local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO, |
| "opus"); |
| ASSERT_TRUE(opus); |
| |
| absl::optional<webrtc::RtpCodecCapability> red = |
| local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO, |
| "red"); |
| ASSERT_TRUE(red); |
| |
| webrtc::RtpTransceiverInit init; |
| init.direction = webrtc::RtpTransceiverDirection::kSendOnly; |
| webrtc::RtpEncodingParameters encoding_parameters; |
| encoding_parameters.codec = opus; |
| init.send_encodings.push_back(encoding_parameters); |
| |
| auto transceiver_or_error = |
| local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init); |
| ASSERT_TRUE(transceiver_or_error.ok()); |
| rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver = |
| transceiver_or_error.MoveValue(); |
| |
| // Preferring RED over Opus should enable RED with Opus encoding. |
| send_codecs[0] = red.value(); |
| send_codecs[1] = opus.value(); |
| |
| ASSERT_TRUE(audio_transceiver->SetCodecPreferences(send_codecs).ok()); |
| NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); |
| local_pc_wrapper->WaitForConnection(); |
| remote_pc_wrapper->WaitForConnection(); |
| |
| webrtc::RtpParameters parameters = |
| audio_transceiver->sender()->GetParameters(); |
| EXPECT_EQ(parameters.encodings[0].codec, opus); |
| EXPECT_EQ(parameters.codecs[0].payload_type, red->preferred_payload_type); |
| EXPECT_EQ(parameters.codecs[0].name, red->name); |
| |
| // Check that it's possible to switch back to Opus without RED. |
| send_codecs[0] = opus.value(); |
| send_codecs[1] = red.value(); |
| |
| ASSERT_TRUE(audio_transceiver->SetCodecPreferences(send_codecs).ok()); |
| NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); |
| |
| parameters = audio_transceiver->sender()->GetParameters(); |
| EXPECT_EQ(parameters.encodings[0].codec, opus); |
| EXPECT_EQ(parameters.codecs[0].payload_type, opus->preferred_payload_type); |
| EXPECT_EQ(parameters.codecs[0].name, opus->name); |
| } |
| |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| SetParametersRejectsScalabilityModeForSelectedCodec) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| |
| absl::optional<webrtc::RtpCodecCapability> vp8 = |
| local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO, |
| "vp8"); |
| ASSERT_TRUE(vp8); |
| |
| webrtc::RtpTransceiverInit init; |
| init.direction = webrtc::RtpTransceiverDirection::kSendOnly; |
| webrtc::RtpEncodingParameters encoding_parameters; |
| encoding_parameters.codec = vp8; |
| encoding_parameters.scalability_mode = "L1T3"; |
| init.send_encodings.push_back(encoding_parameters); |
| |
| auto transceiver_or_error = |
| local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init); |
| ASSERT_TRUE(transceiver_or_error.ok()); |
| rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver = |
| transceiver_or_error.MoveValue(); |
| |
| webrtc::RtpParameters parameters = |
| video_transceiver->sender()->GetParameters(); |
| parameters.encodings[0].scalability_mode = "L3T3"; |
| RTCError error = video_transceiver->sender()->SetParameters(parameters); |
| EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION); |
| } |
| |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| EncodingParametersCodecRemovedByNegotiationVideo) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); |
| ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); |
| |
| absl::optional<webrtc::RtpCodecCapability> vp8 = |
| local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO, |
| "vp8"); |
| ASSERT_TRUE(vp8); |
| |
| std::vector<webrtc::RtpCodecCapability> not_vp8_codecs = |
| local_pc_wrapper->pc_factory() |
| ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO) |
| .codecs; |
| not_vp8_codecs.erase( |
| std::remove_if(not_vp8_codecs.begin(), not_vp8_codecs.end(), |
| [&](const auto& codec) { |
| return absl::EqualsIgnoreCase(codec.name, vp8->name); |
| }), |
| not_vp8_codecs.end()); |
| |
| webrtc::RtpTransceiverInit init; |
| init.direction = webrtc::RtpTransceiverDirection::kSendOnly; |
| webrtc::RtpEncodingParameters encoding_parameters; |
| encoding_parameters.rid = "h"; |
| encoding_parameters.codec = vp8; |
| encoding_parameters.scale_resolution_down_by = 2; |
| init.send_encodings.push_back(encoding_parameters); |
| encoding_parameters.rid = "f"; |
| encoding_parameters.scale_resolution_down_by = 1; |
| init.send_encodings.push_back(encoding_parameters); |
| |
| auto transceiver_or_error = |
| local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init); |
| ASSERT_TRUE(transceiver_or_error.ok()); |
| rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver = |
| transceiver_or_error.MoveValue(); |
| |
| NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); |
| local_pc_wrapper->WaitForConnection(); |
| remote_pc_wrapper->WaitForConnection(); |
| |
| webrtc::RtpParameters parameters = |
| video_transceiver->sender()->GetParameters(); |
| ASSERT_EQ(parameters.encodings.size(), 2u); |
| EXPECT_EQ(parameters.encodings[0].codec, vp8); |
| EXPECT_EQ(parameters.encodings[1].codec, vp8); |
| |
| ASSERT_TRUE(video_transceiver->SetCodecPreferences(not_vp8_codecs).ok()); |
| NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); |
| |
| parameters = video_transceiver->sender()->GetParameters(); |
| EXPECT_FALSE(parameters.encodings[0].codec); |
| EXPECT_FALSE(parameters.encodings[1].codec); |
| } |
| |
| TEST_F(PeerConnectionEncodingsIntegrationTest, |
| AddTransceiverRejectsMixedCodecSimulcast) { |
| // Mixed Codec Simulcast is not yet supported, so we ensure that we reject |
| // such parameters. |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); |
| ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); |
| |
| absl::optional<webrtc::RtpCodecCapability> vp8 = |
| local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO, |
| "vp8"); |
| ASSERT_TRUE(vp8); |
| absl::optional<webrtc::RtpCodecCapability> vp9 = |
| local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO, |
| "vp9"); |
| |
| webrtc::RtpTransceiverInit init; |
| init.direction = webrtc::RtpTransceiverDirection::kSendOnly; |
| webrtc::RtpEncodingParameters encoding_parameters; |
| encoding_parameters.rid = "h"; |
| encoding_parameters.codec = vp8; |
| encoding_parameters.scale_resolution_down_by = 2; |
| init.send_encodings.push_back(encoding_parameters); |
| encoding_parameters.rid = "f"; |
| encoding_parameters.codec = vp9; |
| encoding_parameters.scale_resolution_down_by = 1; |
| init.send_encodings.push_back(encoding_parameters); |
| |
| auto transceiver_or_error = |
| local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init); |
| ASSERT_FALSE(transceiver_or_error.ok()); |
| EXPECT_EQ(transceiver_or_error.error().type(), |
| RTCErrorType::UNSUPPORTED_OPERATION); |
| } |
| |
| // Tests that use the standard path (specifying both `scalability_mode` and |
| // `scale_resolution_down_by`) should pass for all codecs. |
| class PeerConnectionEncodingsIntegrationParameterizedTest |
| : public PeerConnectionEncodingsIntegrationTest, |
| public ::testing::WithParamInterface<std::string> { |
| public: |
| PeerConnectionEncodingsIntegrationParameterizedTest() |
| : codec_name_(GetParam()), mime_type_("video/" + codec_name_) {} |
| |
| // Work-around for the fact that whether or not AV1 is supported is not known |
| // at compile-time so we have to skip tests early if missing. |
| // TODO(https://crbug.com/webrtc/15011): Increase availability of AV1 or make |
| // it possible to check support at compile-time. |
| bool SkipTestDueToAv1Missing( |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper) { |
| if (codec_name_ == "AV1" && |
| !HasSenderVideoCodecCapability(local_pc_wrapper, "AV1")) { |
| RTC_LOG(LS_WARNING) << "\n***\nAV1 is not available, skipping test.\n***"; |
| return true; |
| } |
| return false; |
| } |
| |
| protected: |
| const std::string codec_name_; // E.g. "VP9" |
| const std::string mime_type_; // E.g. "video/VP9" |
| }; |
| |
| TEST_P(PeerConnectionEncodingsIntegrationParameterizedTest, AllLayersInactive) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| if (SkipTestDueToAv1Missing(local_pc_wrapper)) { |
| return; |
| } |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); |
| ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); |
| |
| std::vector<cricket::SimulcastLayer> layers = |
| CreateLayers({"f", "h", "q"}, /*active=*/true); |
| rtc::scoped_refptr<RtpTransceiverInterface> transceiver = |
| AddTransceiverWithSimulcastLayers(local_pc_wrapper, remote_pc_wrapper, |
| layers); |
| std::vector<RtpCodecCapability> codecs = |
| GetCapabilitiesAndRestrictToCodec(local_pc_wrapper, codec_name_); |
| transceiver->SetCodecPreferences(codecs); |
| |
| // Standard mode and all layers inactive. |
| rtc::scoped_refptr<RtpSenderInterface> sender = transceiver->sender(); |
| RtpParameters parameters = sender->GetParameters(); |
| ASSERT_THAT(parameters.encodings, SizeIs(3)); |
| parameters.encodings[0].scalability_mode = "L1T3"; |
| parameters.encodings[0].scale_resolution_down_by = 1; |
| parameters.encodings[0].active = false; |
| parameters.encodings[1].active = false; |
| parameters.encodings[2].active = false; |
| sender->SetParameters(parameters); |
| |
| NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); |
| local_pc_wrapper->WaitForConnection(); |
| remote_pc_wrapper->WaitForConnection(); |
| |
| // Ensure no media is flowing (1 second should be enough). |
| rtc::Thread::Current()->SleepMs(1000); |
| rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper); |
| std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps = |
| report->GetStatsOfType<RTCOutboundRtpStreamStats>(); |
| ASSERT_THAT(outbound_rtps, SizeIs(3u)); |
| EXPECT_EQ(*outbound_rtps[0]->bytes_sent, 0u); |
| EXPECT_EQ(*outbound_rtps[1]->bytes_sent, 0u); |
| EXPECT_EQ(*outbound_rtps[2]->bytes_sent, 0u); |
| } |
| |
| TEST_P(PeerConnectionEncodingsIntegrationParameterizedTest, Simulcast) { |
| rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); |
| if (SkipTestDueToAv1Missing(local_pc_wrapper)) { |
| return; |
| } |
| rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); |
| ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); |
| |
| std::vector<cricket::SimulcastLayer> layers = |
| CreateLayers({"f", "h", "q"}, /*active=*/true); |
| rtc::scoped_refptr<RtpTransceiverInterface> transceiver = |
| AddTransceiverWithSimulcastLayers(local_pc_wrapper, remote_pc_wrapper, |
| layers); |
| std::vector<RtpCodecCapability> codecs = |
| GetCapabilitiesAndRestrictToCodec(local_pc_wrapper, codec_name_); |
| transceiver->SetCodecPreferences(codecs); |
| |
| rtc::scoped_refptr<RtpSenderInterface> sender = transceiver->sender(); |
| RtpParameters parameters = sender->GetParameters(); |
| ASSERT_THAT(parameters.encodings, SizeIs(3)); |
| parameters.encodings[0].scalability_mode = "L1T3"; |
| parameters.encodings[0].scale_resolution_down_by = 4; |
| parameters.encodings[1].scalability_mode = "L1T3"; |
| parameters.encodings[1].scale_resolution_down_by = 2; |
| parameters.encodings[2].scalability_mode = "L1T3"; |
| parameters.encodings[2].scale_resolution_down_by = 1; |
| sender->SetParameters(parameters); |
| |
| NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); |
| local_pc_wrapper->WaitForConnection(); |
| remote_pc_wrapper->WaitForConnection(); |
| |
| // GetParameters() does not report any fallback. |
| parameters = sender->GetParameters(); |
| ASSERT_THAT(parameters.encodings, SizeIs(3)); |
| EXPECT_THAT(parameters.encodings[0].scalability_mode, |
| Optional(std::string("L1T3"))); |
| EXPECT_THAT(parameters.encodings[1].scalability_mode, |
| Optional(std::string("L1T3"))); |
| EXPECT_THAT(parameters.encodings[2].scalability_mode, |
| Optional(std::string("L1T3"))); |
| |
| // Wait until media is flowing on all three layers. |
| // Ramp up time is needed before all three layers are sending. |
| ASSERT_TRUE_WAIT(HasOutboundRtpBytesSent(local_pc_wrapper, 3u), |
| kLongTimeoutForRampingUp.ms()); |
| EXPECT_TRUE(OutboundRtpResolutionsAreLessThanOrEqualToExpectations( |
| local_pc_wrapper, {{"f", 320, 180}, {"h", 640, 360}, {"q", 1280, 720}})); |
| // Verify codec and scalability mode. |
| rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper); |
| std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps = |
| report->GetStatsOfType<RTCOutboundRtpStreamStats>(); |
| ASSERT_THAT(outbound_rtps, SizeIs(3u)); |
| EXPECT_THAT(GetCurrentCodecMimeType(report, *outbound_rtps[0]), |
| StrCaseEq(mime_type_)); |
| EXPECT_THAT(GetCurrentCodecMimeType(report, *outbound_rtps[1]), |
| StrCaseEq(mime_type_)); |
| EXPECT_THAT(GetCurrentCodecMimeType(report, *outbound_rtps[2]), |
| StrCaseEq(mime_type_)); |
| EXPECT_THAT(*outbound_rtps[0]->scalability_mode, StrEq("L1T3")); |
| EXPECT_THAT(*outbound_rtps[1]->scalability_mode, StrEq("L1T3")); |
| EXPECT_THAT(*outbound_rtps[2]->scalability_mode, StrEq("L1T3")); |
| } |
| |
| INSTANTIATE_TEST_SUITE_P(StandardPath, |
| PeerConnectionEncodingsIntegrationParameterizedTest, |
| ::testing::Values("VP8", |
| "VP9", |
| #if defined(WEBRTC_USE_H264) |
| "H264", |
| #endif // defined(WEBRTC_USE_H264) |
| "AV1"), |
| StringParamToString()); |
| |
| } // namespace webrtc |
