diff options
author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 01:13:27 +0000 |
---|---|---|
committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 01:13:27 +0000 |
commit | 40a355a42d4a9444dc753c04c6608dade2f06a23 (patch) | |
tree | 871fc667d2de662f171103ce5ec067014ef85e61 /third_party/libwebrtc/pc/peer_connection_encodings_integrationtest.cc | |
parent | Adding upstream version 124.0.1. (diff) | |
download | firefox-40a355a42d4a9444dc753c04c6608dade2f06a23.tar.xz firefox-40a355a42d4a9444dc753c04c6608dade2f06a23.zip |
Adding upstream version 125.0.1.upstream/125.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/pc/peer_connection_encodings_integrationtest.cc')
-rw-r--r-- | third_party/libwebrtc/pc/peer_connection_encodings_integrationtest.cc | 214 |
1 files changed, 96 insertions, 118 deletions
diff --git a/third_party/libwebrtc/pc/peer_connection_encodings_integrationtest.cc b/third_party/libwebrtc/pc/peer_connection_encodings_integrationtest.cc index c7181c53ae..ae238671c2 100644 --- a/third_party/libwebrtc/pc/peer_connection_encodings_integrationtest.cc +++ b/third_party/libwebrtc/pc/peer_connection_encodings_integrationtest.cc @@ -77,18 +77,17 @@ struct StringParamToString { // 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) { +bool IsReliabilityMechanism(const 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) { + rtc::scoped_refptr<const RTCStatsReport> report, + const RTCOutboundRtpStreamStats& outbound_rtp) { return outbound_rtp.codec_id.is_defined() - ? *report->GetAs<webrtc::RTCCodecStats>(*outbound_rtp.codec_id) - ->mime_type + ? *report->GetAs<RTCCodecStats>(*outbound_rtp.codec_id)->mime_type : ""; } @@ -98,8 +97,8 @@ struct RidAndResolution { uint32_t height; }; -const webrtc::RTCOutboundRtpStreamStats* FindOutboundRtpByRid( - const std::vector<const webrtc::RTCOutboundRtpStreamStats*>& outbound_rtps, +const RTCOutboundRtpStreamStats* FindOutboundRtpByRid( + const std::vector<const 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) { @@ -121,8 +120,8 @@ class PeerConnectionEncodingsIntegrationTest : public ::testing::Test { 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()); + pc_wrapper->CreatePc({}, CreateBuiltinAudioEncoderFactory(), + CreateBuiltinAudioDecoderFactory()); return pc_wrapper; } @@ -130,10 +129,9 @@ class PeerConnectionEncodingsIntegrationTest : public ::testing::Test { 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<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>> @@ -973,8 +971,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, 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(); + RtpParameters parameters = audio_transceiver->sender()->GetParameters(); EXPECT_FALSE(parameters.encodings[0].codec.has_value()); } @@ -986,8 +983,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, 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(); + RtpParameters parameters = video_transceiver->sender()->GetParameters(); EXPECT_FALSE(parameters.encodings[0].codec.has_value()); } @@ -997,19 +993,19 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); - rtc::scoped_refptr<webrtc::MediaStreamInterface> stream = + rtc::scoped_refptr<MediaStreamInterface> stream = local_pc_wrapper->GetUserMedia( /*audio=*/true, {}, /*video=*/false, {}); rtc::scoped_refptr<AudioTrackInterface> track = stream->GetAudioTracks()[0]; - absl::optional<webrtc::RtpCodecCapability> pcmu = + absl::optional<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; + RtpTransceiverInit init; + init.direction = RtpTransceiverDirection::kSendOnly; + RtpEncodingParameters encoding_parameters; encoding_parameters.codec = pcmu; init.send_encodings.push_back(encoding_parameters); @@ -1017,8 +1013,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, local_pc_wrapper->pc()->AddTransceiver(track, init); rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver = transceiver_or_error.MoveValue(); - webrtc::RtpParameters parameters = - audio_transceiver->sender()->GetParameters(); + RtpParameters parameters = audio_transceiver->sender()->GetParameters(); EXPECT_EQ(*parameters.encodings[0].codec, *pcmu); NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); @@ -1039,19 +1034,19 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); - rtc::scoped_refptr<webrtc::MediaStreamInterface> stream = + rtc::scoped_refptr<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 = + absl::optional<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; + RtpTransceiverInit init; + init.direction = RtpTransceiverDirection::kSendOnly; + RtpEncodingParameters encoding_parameters; encoding_parameters.codec = vp9; encoding_parameters.scalability_mode = "L3T3"; init.send_encodings.push_back(encoding_parameters); @@ -1060,8 +1055,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, local_pc_wrapper->pc()->AddTransceiver(track, init); rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver = transceiver_or_error.MoveValue(); - webrtc::RtpParameters parameters = - audio_transceiver->sender()->GetParameters(); + RtpParameters parameters = audio_transceiver->sender()->GetParameters(); EXPECT_EQ(*parameters.encodings[0].codec, *vp9); NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper); @@ -1087,20 +1081,19 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); - rtc::scoped_refptr<webrtc::MediaStreamInterface> stream = + rtc::scoped_refptr<MediaStreamInterface> stream = local_pc_wrapper->GetUserMedia( /*audio=*/true, {}, /*video=*/false, {}); rtc::scoped_refptr<AudioTrackInterface> track = stream->GetAudioTracks()[0]; - absl::optional<webrtc::RtpCodecCapability> pcmu = + absl::optional<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(); + RtpParameters parameters = audio_transceiver->sender()->GetParameters(); parameters.encodings[0].codec = pcmu; EXPECT_TRUE(audio_transceiver->sender()->SetParameters(parameters).ok()); @@ -1125,12 +1118,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); - rtc::scoped_refptr<webrtc::MediaStreamInterface> stream = + rtc::scoped_refptr<MediaStreamInterface> stream = local_pc_wrapper->GetUserMedia( /*audio=*/true, {}, /*video=*/false, {}); rtc::scoped_refptr<AudioTrackInterface> track = stream->GetAudioTracks()[0]; - absl::optional<webrtc::RtpCodecCapability> pcmu = + absl::optional<RtpCodecCapability> pcmu = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO, "pcmu"); @@ -1150,8 +1143,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, 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(); + RtpParameters parameters = audio_transceiver->sender()->GetParameters(); parameters.encodings[0].codec = pcmu; EXPECT_TRUE(audio_transceiver->sender()->SetParameters(parameters).ok()); @@ -1174,20 +1166,19 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); - rtc::scoped_refptr<webrtc::MediaStreamInterface> stream = + rtc::scoped_refptr<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 = + absl::optional<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(); + 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()); @@ -1218,12 +1209,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); - rtc::scoped_refptr<webrtc::MediaStreamInterface> stream = + rtc::scoped_refptr<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 = + absl::optional<RtpCodecCapability> vp9 = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO, "vp9"); @@ -1243,8 +1234,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, 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(); + 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()); @@ -1269,15 +1259,15 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, AddTransceiverRejectsUnknownCodecParameterAudio) { rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); - webrtc::RtpCodec dummy_codec; + 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; + RtpTransceiverInit init; + init.direction = RtpTransceiverDirection::kSendOnly; + RtpEncodingParameters encoding_parameters; encoding_parameters.codec = dummy_codec; init.send_encodings.push_back(encoding_parameters); @@ -1292,14 +1282,14 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, AddTransceiverRejectsUnknownCodecParameterVideo) { rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); - webrtc::RtpCodec dummy_codec; + 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; + RtpTransceiverInit init; + init.direction = RtpTransceiverDirection::kSendOnly; + RtpEncodingParameters encoding_parameters; encoding_parameters.codec = dummy_codec; init.send_encodings.push_back(encoding_parameters); @@ -1314,7 +1304,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, SetParametersRejectsUnknownCodecParameterAudio) { rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); - webrtc::RtpCodec dummy_codec; + RtpCodec dummy_codec; dummy_codec.kind = cricket::MEDIA_TYPE_AUDIO; dummy_codec.name = "FOOBAR"; dummy_codec.clock_rate = 90000; @@ -1326,8 +1316,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver = transceiver_or_error.MoveValue(); - webrtc::RtpParameters parameters = - audio_transceiver->sender()->GetParameters(); + 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); @@ -1337,7 +1326,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, SetParametersRejectsUnknownCodecParameterVideo) { rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); - webrtc::RtpCodec dummy_codec; + RtpCodec dummy_codec; dummy_codec.kind = cricket::MEDIA_TYPE_VIDEO; dummy_codec.name = "FOOBAR"; dummy_codec.clock_rate = 90000; @@ -1348,8 +1337,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver = transceiver_or_error.MoveValue(); - webrtc::RtpParameters parameters = - video_transceiver->sender()->GetParameters(); + 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); @@ -1359,12 +1347,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, SetParametersRejectsNonPreferredCodecParameterAudio) { rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); - absl::optional<webrtc::RtpCodecCapability> opus = + absl::optional<RtpCodecCapability> opus = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO, "opus"); ASSERT_TRUE(opus); - std::vector<webrtc::RtpCodecCapability> not_opus_codecs = + std::vector<RtpCodecCapability> not_opus_codecs = local_pc_wrapper->pc_factory() ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO) .codecs; @@ -1382,8 +1370,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, transceiver_or_error.MoveValue(); ASSERT_TRUE(audio_transceiver->SetCodecPreferences(not_opus_codecs).ok()); - webrtc::RtpParameters parameters = - audio_transceiver->sender()->GetParameters(); + 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); @@ -1393,12 +1380,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, SetParametersRejectsNonPreferredCodecParameterVideo) { rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); - absl::optional<webrtc::RtpCodecCapability> vp8 = + absl::optional<RtpCodecCapability> vp8 = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO, "vp8"); ASSERT_TRUE(vp8); - std::vector<webrtc::RtpCodecCapability> not_vp8_codecs = + std::vector<RtpCodecCapability> not_vp8_codecs = local_pc_wrapper->pc_factory() ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO) .codecs; @@ -1416,8 +1403,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, transceiver_or_error.MoveValue(); ASSERT_TRUE(video_transceiver->SetCodecPreferences(not_vp8_codecs).ok()); - webrtc::RtpParameters parameters = - video_transceiver->sender()->GetParameters(); + 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); @@ -1429,12 +1415,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); - absl::optional<webrtc::RtpCodecCapability> opus = + absl::optional<RtpCodecCapability> opus = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO, "opus"); ASSERT_TRUE(opus); - std::vector<webrtc::RtpCodecCapability> not_opus_codecs = + std::vector<RtpCodecCapability> not_opus_codecs = local_pc_wrapper->pc_factory() ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO) .codecs; @@ -1456,8 +1442,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, local_pc_wrapper->WaitForConnection(); remote_pc_wrapper->WaitForConnection(); - webrtc::RtpParameters parameters = - audio_transceiver->sender()->GetParameters(); + 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); @@ -1469,12 +1454,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); - absl::optional<webrtc::RtpCodecCapability> opus = + absl::optional<RtpCodecCapability> opus = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO, "opus"); ASSERT_TRUE(opus); - std::vector<webrtc::RtpCodecCapability> not_opus_codecs = + std::vector<RtpCodecCapability> not_opus_codecs = local_pc_wrapper->pc_factory() ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO) .codecs; @@ -1519,8 +1504,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, local_pc_wrapper->WaitForConnection(); remote_pc_wrapper->WaitForConnection(); - webrtc::RtpParameters parameters = - audio_transceiver->sender()->GetParameters(); + 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); @@ -1532,12 +1516,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); - absl::optional<webrtc::RtpCodecCapability> vp8 = + absl::optional<RtpCodecCapability> vp8 = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO, "vp8"); ASSERT_TRUE(vp8); - std::vector<webrtc::RtpCodecCapability> not_vp8_codecs = + std::vector<RtpCodecCapability> not_vp8_codecs = local_pc_wrapper->pc_factory() ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO) .codecs; @@ -1559,8 +1543,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, local_pc_wrapper->WaitForConnection(); remote_pc_wrapper->WaitForConnection(); - webrtc::RtpParameters parameters = - video_transceiver->sender()->GetParameters(); + 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); @@ -1572,12 +1555,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); - absl::optional<webrtc::RtpCodecCapability> vp8 = + absl::optional<RtpCodecCapability> vp8 = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO, "vp8"); ASSERT_TRUE(vp8); - std::vector<webrtc::RtpCodecCapability> not_vp8_codecs = + std::vector<RtpCodecCapability> not_vp8_codecs = local_pc_wrapper->pc_factory() ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO) .codecs; @@ -1622,8 +1605,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, local_pc_wrapper->WaitForConnection(); remote_pc_wrapper->WaitForConnection(); - webrtc::RtpParameters parameters = - video_transceiver->sender()->GetParameters(); + 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); @@ -1635,12 +1617,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); - absl::optional<webrtc::RtpCodecCapability> opus = + absl::optional<RtpCodecCapability> opus = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO, "opus"); ASSERT_TRUE(opus); - std::vector<webrtc::RtpCodecCapability> not_opus_codecs = + std::vector<RtpCodecCapability> not_opus_codecs = local_pc_wrapper->pc_factory() ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO) .codecs; @@ -1651,9 +1633,9 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, }), not_opus_codecs.end()); - webrtc::RtpTransceiverInit init; - init.direction = webrtc::RtpTransceiverDirection::kSendOnly; - webrtc::RtpEncodingParameters encoding_parameters; + RtpTransceiverInit init; + init.direction = RtpTransceiverDirection::kSendOnly; + RtpEncodingParameters encoding_parameters; encoding_parameters.codec = opus; init.send_encodings.push_back(encoding_parameters); @@ -1667,8 +1649,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, local_pc_wrapper->WaitForConnection(); remote_pc_wrapper->WaitForConnection(); - webrtc::RtpParameters parameters = - audio_transceiver->sender()->GetParameters(); + RtpParameters parameters = audio_transceiver->sender()->GetParameters(); EXPECT_EQ(parameters.encodings[0].codec, opus); ASSERT_TRUE(audio_transceiver->SetCodecPreferences(not_opus_codecs).ok()); @@ -1684,24 +1665,24 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); - std::vector<webrtc::RtpCodecCapability> send_codecs = + std::vector<RtpCodecCapability> send_codecs = local_pc_wrapper->pc_factory() ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO) .codecs; - absl::optional<webrtc::RtpCodecCapability> opus = + absl::optional<RtpCodecCapability> opus = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO, "opus"); ASSERT_TRUE(opus); - absl::optional<webrtc::RtpCodecCapability> red = + absl::optional<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; + RtpTransceiverInit init; + init.direction = RtpTransceiverDirection::kSendOnly; + RtpEncodingParameters encoding_parameters; encoding_parameters.codec = opus; init.send_encodings.push_back(encoding_parameters); @@ -1720,8 +1701,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, local_pc_wrapper->WaitForConnection(); remote_pc_wrapper->WaitForConnection(); - webrtc::RtpParameters parameters = - audio_transceiver->sender()->GetParameters(); + 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); @@ -1743,14 +1723,14 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, SetParametersRejectsScalabilityModeForSelectedCodec) { rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc(); - absl::optional<webrtc::RtpCodecCapability> vp8 = + absl::optional<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; + RtpTransceiverInit init; + init.direction = RtpTransceiverDirection::kSendOnly; + RtpEncodingParameters encoding_parameters; encoding_parameters.codec = vp8; encoding_parameters.scalability_mode = "L1T3"; init.send_encodings.push_back(encoding_parameters); @@ -1761,8 +1741,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver = transceiver_or_error.MoveValue(); - webrtc::RtpParameters parameters = - video_transceiver->sender()->GetParameters(); + 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); @@ -1774,12 +1753,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); - absl::optional<webrtc::RtpCodecCapability> vp8 = + absl::optional<RtpCodecCapability> vp8 = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO, "vp8"); ASSERT_TRUE(vp8); - std::vector<webrtc::RtpCodecCapability> not_vp8_codecs = + std::vector<RtpCodecCapability> not_vp8_codecs = local_pc_wrapper->pc_factory() ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO) .codecs; @@ -1790,9 +1769,9 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, }), not_vp8_codecs.end()); - webrtc::RtpTransceiverInit init; - init.direction = webrtc::RtpTransceiverDirection::kSendOnly; - webrtc::RtpEncodingParameters encoding_parameters; + RtpTransceiverInit init; + init.direction = RtpTransceiverDirection::kSendOnly; + RtpEncodingParameters encoding_parameters; encoding_parameters.rid = "h"; encoding_parameters.codec = vp8; encoding_parameters.scale_resolution_down_by = 2; @@ -1811,8 +1790,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, local_pc_wrapper->WaitForConnection(); remote_pc_wrapper->WaitForConnection(); - webrtc::RtpParameters parameters = - video_transceiver->sender()->GetParameters(); + 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); @@ -1833,17 +1811,17 @@ TEST_F(PeerConnectionEncodingsIntegrationTest, rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc(); ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper); - absl::optional<webrtc::RtpCodecCapability> vp8 = + absl::optional<RtpCodecCapability> vp8 = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO, "vp8"); ASSERT_TRUE(vp8); - absl::optional<webrtc::RtpCodecCapability> vp9 = + absl::optional<RtpCodecCapability> vp9 = local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO, "vp9"); - webrtc::RtpTransceiverInit init; - init.direction = webrtc::RtpTransceiverDirection::kSendOnly; - webrtc::RtpEncodingParameters encoding_parameters; + RtpTransceiverInit init; + init.direction = RtpTransceiverDirection::kSendOnly; + RtpEncodingParameters encoding_parameters; encoding_parameters.rid = "h"; encoding_parameters.codec = vp8; encoding_parameters.scale_resolution_down_by = 2; |