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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 01:13:27 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 01:13:27 +0000 |
commit | 40a355a42d4a9444dc753c04c6608dade2f06a23 (patch) | |
tree | 871fc667d2de662f171103ce5ec067014ef85e61 /third_party/libwebrtc/pc/rtp_sender_receiver_unittest.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/rtp_sender_receiver_unittest.cc')
-rw-r--r-- | third_party/libwebrtc/pc/rtp_sender_receiver_unittest.cc | 80 |
1 files changed, 38 insertions, 42 deletions
diff --git a/third_party/libwebrtc/pc/rtp_sender_receiver_unittest.cc b/third_party/libwebrtc/pc/rtp_sender_receiver_unittest.cc index 3092e53c2d..4387aedf53 100644 --- a/third_party/libwebrtc/pc/rtp_sender_receiver_unittest.cc +++ b/third_party/libwebrtc/pc/rtp_sender_receiver_unittest.cc @@ -105,7 +105,7 @@ class RtpSenderReceiverTest : network_thread_(rtc::Thread::Current()), worker_thread_(rtc::Thread::Current()), video_bitrate_allocator_factory_( - webrtc::CreateBuiltinVideoBitrateAllocatorFactory()), + CreateBuiltinVideoBitrateAllocatorFactory()), // Create fake media engine/etc. so we can create channels to use to // test RtpSenders/RtpReceivers. media_engine_(std::make_unique<cricket::FakeMediaEngine>()), @@ -119,16 +119,16 @@ class RtpSenderReceiverTest // Fake media channels are owned by the media engine. voice_media_send_channel_ = media_engine_->voice().CreateSendChannel( &fake_call_, cricket::MediaConfig(), cricket::AudioOptions(), - webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create()); + CryptoOptions(), AudioCodecPairId::Create()); video_media_send_channel_ = media_engine_->video().CreateSendChannel( &fake_call_, cricket::MediaConfig(), cricket::VideoOptions(), - webrtc::CryptoOptions(), video_bitrate_allocator_factory_.get()); + CryptoOptions(), video_bitrate_allocator_factory_.get()); voice_media_receive_channel_ = media_engine_->voice().CreateReceiveChannel( &fake_call_, cricket::MediaConfig(), cricket::AudioOptions(), - webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create()); + CryptoOptions(), AudioCodecPairId::Create()); video_media_receive_channel_ = media_engine_->video().CreateReceiveChannel( &fake_call_, cricket::MediaConfig(), cricket::VideoOptions(), - webrtc::CryptoOptions()); + CryptoOptions()); // Create streams for predefined SSRCs. Streams need to exist in order // for the senders and receievers to apply parameters to them. @@ -162,8 +162,8 @@ class RtpSenderReceiverTest audio_track_ = nullptr; } - std::unique_ptr<webrtc::RtpTransportInternal> CreateDtlsSrtpTransport() { - auto dtls_srtp_transport = std::make_unique<webrtc::DtlsSrtpTransport>( + std::unique_ptr<RtpTransportInternal> CreateDtlsSrtpTransport() { + auto dtls_srtp_transport = std::make_unique<DtlsSrtpTransport>( /*rtcp_mux_required=*/true, field_trials_); dtls_srtp_transport->SetDtlsTransports(rtp_dtls_transport_.get(), /*rtcp_dtls_transport=*/nullptr); @@ -515,12 +515,12 @@ class RtpSenderReceiverTest test::RunLoop run_loop_; rtc::Thread* const network_thread_; rtc::Thread* const worker_thread_; - webrtc::RtcEventLogNull event_log_; + RtcEventLogNull event_log_; // The `rtp_dtls_transport_` and `rtp_transport_` should be destroyed after // the `channel_manager`. std::unique_ptr<cricket::DtlsTransportInternal> rtp_dtls_transport_; - std::unique_ptr<webrtc::RtpTransportInternal> rtp_transport_; - std::unique_ptr<webrtc::VideoBitrateAllocatorFactory> + std::unique_ptr<RtpTransportInternal> rtp_transport_; + std::unique_ptr<VideoBitrateAllocatorFactory> video_bitrate_allocator_factory_; std::unique_ptr<cricket::FakeMediaEngine> media_engine_; rtc::UniqueRandomIdGenerator ssrc_generator_; @@ -540,7 +540,7 @@ class RtpSenderReceiverTest rtc::scoped_refptr<MediaStreamInterface> local_stream_; rtc::scoped_refptr<VideoTrackInterface> video_track_; rtc::scoped_refptr<AudioTrackInterface> audio_track_; - webrtc::test::ScopedKeyValueConfig field_trials_; + test::ScopedKeyValueConfig field_trials_; }; // Test that `voice_channel_` is updated when an audio track is associated @@ -651,15 +651,13 @@ TEST_F(RtpSenderReceiverTest, LocalVideoTrackDisable) { TEST_F(RtpSenderReceiverTest, RemoteVideoTrackState) { CreateVideoRtpReceiver(); - EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, video_track_->state()); - EXPECT_EQ(webrtc::MediaSourceInterface::kLive, - video_track_->GetSource()->state()); + EXPECT_EQ(MediaStreamTrackInterface::kLive, video_track_->state()); + EXPECT_EQ(MediaSourceInterface::kLive, video_track_->GetSource()->state()); DestroyVideoRtpReceiver(); - EXPECT_EQ(webrtc::MediaStreamTrackInterface::kEnded, video_track_->state()); - EXPECT_EQ(webrtc::MediaSourceInterface::kEnded, - video_track_->GetSource()->state()); + EXPECT_EQ(MediaStreamTrackInterface::kEnded, video_track_->state()); + EXPECT_EQ(MediaSourceInterface::kEnded, video_track_->GetSource()->state()); DestroyVideoRtpReceiver(); } @@ -888,9 +886,9 @@ TEST_F(RtpSenderReceiverTest, AudioSenderCanSetParametersAsync) { RtpParameters params = audio_rtp_sender_->GetParameters(); EXPECT_EQ(1u, params.encodings.size()); - absl::optional<webrtc::RTCError> result; + absl::optional<RTCError> result; audio_rtp_sender_->SetParametersAsync( - params, [&result](webrtc::RTCError error) { result = error; }); + params, [&result](RTCError error) { result = error; }); run_loop_.Flush(); EXPECT_TRUE(result->ok()); @@ -918,13 +916,13 @@ TEST_F(RtpSenderReceiverTest, audio_rtp_sender_ = AudioRtpSender::Create(worker_thread_, /*id=*/"", nullptr, nullptr); - absl::optional<webrtc::RTCError> result; + absl::optional<RTCError> result; RtpParameters params = audio_rtp_sender_->GetParameters(); ASSERT_EQ(1u, params.encodings.size()); params.encodings[0].max_bitrate_bps = 90000; audio_rtp_sender_->SetParametersAsync( - params, [&result](webrtc::RTCError error) { result = error; }); + params, [&result](RTCError error) { result = error; }); run_loop_.Flush(); EXPECT_TRUE(result->ok()); @@ -932,7 +930,7 @@ TEST_F(RtpSenderReceiverTest, EXPECT_EQ(params.encodings[0].max_bitrate_bps, 90000); audio_rtp_sender_->SetParametersAsync( - params, [&result](webrtc::RTCError error) { result = error; }); + params, [&result](RTCError error) { result = error; }); run_loop_.Flush(); EXPECT_TRUE(result->ok()); @@ -1016,13 +1014,13 @@ TEST_F(RtpSenderReceiverTest, RtpParameters params = audio_rtp_sender_->GetParameters(); EXPECT_EQ(1u, params.encodings.size()); - absl::optional<webrtc::RTCError> result; + absl::optional<RTCError> result; audio_rtp_sender_->SetParametersAsync( - params, [&result](webrtc::RTCError error) { result = error; }); + params, [&result](RTCError error) { result = error; }); run_loop_.Flush(); EXPECT_TRUE(result->ok()); audio_rtp_sender_->SetParametersAsync( - params, [&result](webrtc::RTCError error) { result = error; }); + params, [&result](RTCError error) { result = error; }); run_loop_.Flush(); EXPECT_EQ(RTCErrorType::INVALID_STATE, result->type()); @@ -1081,7 +1079,7 @@ TEST_F(RtpSenderReceiverTest, SetAudioMaxSendBitrate) { CreateAudioRtpSender(); EXPECT_EQ(-1, voice_media_send_channel()->max_bps()); - webrtc::RtpParameters params = audio_rtp_sender_->GetParameters(); + RtpParameters params = audio_rtp_sender_->GetParameters(); EXPECT_EQ(1U, params.encodings.size()); EXPECT_FALSE(params.encodings[0].max_bitrate_bps); params.encodings[0].max_bitrate_bps = 1000; @@ -1106,10 +1104,9 @@ TEST_F(RtpSenderReceiverTest, SetAudioMaxSendBitrate) { TEST_F(RtpSenderReceiverTest, SetAudioBitratePriority) { CreateAudioRtpSender(); - webrtc::RtpParameters params = audio_rtp_sender_->GetParameters(); + RtpParameters params = audio_rtp_sender_->GetParameters(); EXPECT_EQ(1U, params.encodings.size()); - EXPECT_EQ(webrtc::kDefaultBitratePriority, - params.encodings[0].bitrate_priority); + EXPECT_EQ(kDefaultBitratePriority, params.encodings[0].bitrate_priority); double new_bitrate_priority = 2.0; params.encodings[0].bitrate_priority = new_bitrate_priority; EXPECT_TRUE(audio_rtp_sender_->SetParameters(params).ok()); @@ -1140,9 +1137,9 @@ TEST_F(RtpSenderReceiverTest, VideoSenderCanSetParametersAsync) { RtpParameters params = video_rtp_sender_->GetParameters(); EXPECT_EQ(1u, params.encodings.size()); - absl::optional<webrtc::RTCError> result; + absl::optional<RTCError> result; video_rtp_sender_->SetParametersAsync( - params, [&result](webrtc::RTCError error) { result = error; }); + params, [&result](RTCError error) { result = error; }); run_loop_.Flush(); EXPECT_TRUE(result->ok()); @@ -1170,19 +1167,19 @@ TEST_F(RtpSenderReceiverTest, video_rtp_sender_ = VideoRtpSender::Create(worker_thread_, /*id=*/"", nullptr); - absl::optional<webrtc::RTCError> result; + absl::optional<RTCError> result; RtpParameters params = video_rtp_sender_->GetParameters(); ASSERT_EQ(1u, params.encodings.size()); params.encodings[0].max_bitrate_bps = 90000; video_rtp_sender_->SetParametersAsync( - params, [&result](webrtc::RTCError error) { result = error; }); + params, [&result](RTCError error) { result = error; }); run_loop_.Flush(); EXPECT_TRUE(result->ok()); params = video_rtp_sender_->GetParameters(); EXPECT_EQ(params.encodings[0].max_bitrate_bps, 90000); video_rtp_sender_->SetParametersAsync( - params, [&result](webrtc::RTCError error) { result = error; }); + params, [&result](RTCError error) { result = error; }); run_loop_.Flush(); EXPECT_TRUE(result->ok()); @@ -1350,13 +1347,13 @@ TEST_F(RtpSenderReceiverTest, RtpParameters params = video_rtp_sender_->GetParameters(); EXPECT_EQ(1u, params.encodings.size()); - absl::optional<webrtc::RTCError> result; + absl::optional<RTCError> result; video_rtp_sender_->SetParametersAsync( - params, [&result](webrtc::RTCError error) { result = error; }); + params, [&result](RTCError error) { result = error; }); run_loop_.Flush(); EXPECT_TRUE(result->ok()); video_rtp_sender_->SetParametersAsync( - params, [&result](webrtc::RTCError error) { result = error; }); + params, [&result](RTCError error) { result = error; }); run_loop_.Flush(); EXPECT_EQ(RTCErrorType::INVALID_STATE, result->type()); @@ -1453,7 +1450,7 @@ TEST_F(RtpSenderReceiverTest, VideoSenderDetectInvalidNumTemporalLayers) { CreateVideoRtpSender(); RtpParameters params = video_rtp_sender_->GetParameters(); - params.encodings[0].num_temporal_layers = webrtc::kMaxTemporalStreams + 1; + params.encodings[0].num_temporal_layers = kMaxTemporalStreams + 1; RTCError result = video_rtp_sender_->SetParameters(params); EXPECT_EQ(RTCErrorType::INVALID_RANGE, result.type()); @@ -1536,7 +1533,7 @@ TEST_F(RtpSenderReceiverTest, SetVideoMinMaxSendBitrate) { CreateVideoRtpSender(); EXPECT_EQ(-1, video_media_send_channel()->max_bps()); - webrtc::RtpParameters params = video_rtp_sender_->GetParameters(); + RtpParameters params = video_rtp_sender_->GetParameters(); EXPECT_EQ(1U, params.encodings.size()); EXPECT_FALSE(params.encodings[0].min_bitrate_bps); EXPECT_FALSE(params.encodings[0].max_bitrate_bps); @@ -1589,10 +1586,9 @@ TEST_F(RtpSenderReceiverTest, SetVideoMinMaxSendBitrateSimulcast) { TEST_F(RtpSenderReceiverTest, SetVideoBitratePriority) { CreateVideoRtpSender(); - webrtc::RtpParameters params = video_rtp_sender_->GetParameters(); + RtpParameters params = video_rtp_sender_->GetParameters(); EXPECT_EQ(1U, params.encodings.size()); - EXPECT_EQ(webrtc::kDefaultBitratePriority, - params.encodings[0].bitrate_priority); + EXPECT_EQ(kDefaultBitratePriority, params.encodings[0].bitrate_priority); double new_bitrate_priority = 2.0; params.encodings[0].bitrate_priority = new_bitrate_priority; EXPECT_TRUE(video_rtp_sender_->SetParameters(params).ok()); |