commit 9da1af43f53511632550d076969823d498d9dab3 parent 6ab7a68262240625f16ced2522206376712d9428 Author: Dan Baker <dbaker@mozilla.com> Date: Mon, 27 Oct 2025 15:31:54 -0600 Bug 1995393 - Vendor libwebrtc from 3c9a550086 Upstream commit: https://webrtc.googlesource.com/src/+/3c9a5500863b3b4a82610264602c53f6bf49026b clang-tidy: apply bugprone-argument-comment https://clang.llvm.org/extra/clang-tidy/checks/bugprone/argument-comment.html and add it to .clang-tidy Bug: webrtc:424706384 Change-Id: Icc2664b4a324a92cde52f5f4c00743cbf2ab1d0d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/406900 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#45490} Diffstat:
26 files changed, 74 insertions(+), 71 deletions(-)
diff --git a/third_party/libwebrtc/.clang-tidy b/third_party/libwebrtc/.clang-tidy @@ -1,6 +1,7 @@ --- --- Checks: '-*, + bugprone-argument-comment llvm-namespace-comment, modernize-use-designated-initializers, modernize-use-override, diff --git a/third_party/libwebrtc/README.mozilla.last-vendor b/third_party/libwebrtc/README.mozilla.last-vendor @@ -1,4 +1,4 @@ # ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc -libwebrtc updated from /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-10-27T21:29:10.797855+00:00. +libwebrtc updated from /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-10-27T21:31:41.709632+00:00. # base of lastest vendoring -73aba0d728 +3c9a550086 diff --git a/third_party/libwebrtc/audio/utility/audio_frame_operations_unittest.cc b/third_party/libwebrtc/audio/utility/audio_frame_operations_unittest.cc @@ -26,7 +26,7 @@ class AudioFrameOperationsTest : public ::testing::Test { AudioFrameOperationsTest() = default; // Set typical values. - AudioFrame frame_{/*sample_rate=*/32000, /*num_channels*/ 2}; + AudioFrame frame_{/*sample_rate_hz=*/32000, /*num_channels=*/2}; }; class AudioFrameOperationsDeathTest : public AudioFrameOperationsTest {}; diff --git a/third_party/libwebrtc/call/rtp_video_sender_unittest.cc b/third_party/libwebrtc/call/rtp_video_sender_unittest.cc @@ -1402,7 +1402,7 @@ TEST(RtpVideoSenderTest, ClearsPendingPacketsOnInactivation) { encoded_image.SetEncodedData( EncodedImageBuffer::Create(kPayload, sizeof(kPayload))); EXPECT_EQ(test.router() - ->OnEncodedImage(encoded_image, /*codec_specific=*/nullptr) + ->OnEncodedImage(encoded_image, /*codec_specific_info=*/nullptr) .error, EncodedImageCallback::Result::OK); @@ -1438,7 +1438,7 @@ TEST(RtpVideoSenderTest, ClearsPendingPacketsOnInactivation) { encoded_image.SetRtpTimestamp(3); encoded_image.capture_time_ms_ = 4; EXPECT_EQ(test.router() - ->OnEncodedImage(encoded_image, /*codec_specific=*/nullptr) + ->OnEncodedImage(encoded_image, /*codec_specific_info=*/nullptr) .error, EncodedImageCallback::Result::OK); test.AdvanceTime(TimeDelta::Millis(33)); @@ -1480,7 +1480,7 @@ TEST(RtpVideoSenderTest, encoded_image.SetEncodedData( EncodedImageBuffer::Create(kImage, std::size(kImage))); EXPECT_EQ(test.router() - ->OnEncodedImage(encoded_image, /*codec_specific=*/nullptr) + ->OnEncodedImage(encoded_image, /*codec_specific_info=*/nullptr) .error, EncodedImageCallback::Result::OK); @@ -1519,7 +1519,7 @@ TEST(RtpVideoSenderTest, encoded_image.SetRtpTimestamp(3); encoded_image.capture_time_ms_ = 4; EXPECT_EQ(test.router() - ->OnEncodedImage(encoded_image, /*codec_specific=*/nullptr) + ->OnEncodedImage(encoded_image, /*codec_specific_info=*/nullptr) .error, EncodedImageCallback::Result::OK); test.AdvanceTime(TimeDelta::Millis(33)); diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/alignment_mixer_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/alignment_mixer_unittest.cc @@ -160,7 +160,7 @@ TEST(AlignmentMixer, FixedMode) { /*adaptive_selection*/ false, /*excitation_limit*/ 1.f, /*prefer_first_two_channels*/ false); - Block x(/*num_band=*/1, num_channels); + Block x(/*num_bands=*/1, num_channels); const auto channel_value = [](int frame_index, int channel_index) { return static_cast<float>(frame_index + channel_index); }; diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/echo_canceller3_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec3/echo_canceller3_unittest.cc @@ -974,7 +974,7 @@ TEST(EchoCanceller3, DetectionOfProperStereo) { EchoCanceller3 aec3(CreateEnvironment(), mono_config, multichannel_config, /*sample_rate_hz=*/kSampleRateHz, /*num_render_channels=*/kNumChannels, - /*num_capture_input_channels=*/kNumChannels); + /*num_capture_channels=*/kNumChannels); EXPECT_FALSE(aec3.StereoRenderProcessingActiveForTesting()); EXPECT_EQ( @@ -1022,7 +1022,7 @@ TEST(EchoCanceller3, DetectionOfProperStereoUsingThreshold) { EchoCanceller3 aec3(CreateEnvironment(), mono_config, multichannel_config, /*sample_rate_hz=*/kSampleRateHz, /*num_render_channels=*/kNumChannels, - /*num_capture_input_channels=*/kNumChannels); + /*num_capture_channels=*/kNumChannels); EXPECT_FALSE(aec3.StereoRenderProcessingActiveForTesting()); EXPECT_EQ( @@ -1069,7 +1069,7 @@ TEST(EchoCanceller3, DetectionOfProperStereoUsingHysteresis) { EchoCanceller3 aec3(CreateEnvironment(), mono_config, surround_config, /*sample_rate_hz=*/kSampleRateHz, /*num_render_channels=*/kNumChannels, - /*num_capture_input_channels=*/kNumChannels); + /*num_capture_channels=*/kNumChannels); EXPECT_FALSE(aec3.StereoRenderProcessingActiveForTesting()); EXPECT_EQ( @@ -1135,7 +1135,7 @@ TEST(EchoCanceller3, StereoContentDetectionForMonoSignals) { EchoCanceller3 aec3(env, mono_config, multichannel_config, /*sample_rate_hz=*/kSampleRateHz, /*num_render_channels=*/1, - /*num_capture_input_channels=*/1); + /*num_capture_channels=*/1); EXPECT_FALSE(aec3.StereoRenderProcessingActiveForTesting()); EXPECT_EQ( diff --git a/third_party/libwebrtc/modules/audio_processing/agc/agc_manager_direct_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc/agc_manager_direct_unittest.cc @@ -509,7 +509,7 @@ TEST_P(AgcManagerDirectParametrizedTest, // controller read the input volume. That is needed because clipping input // causes the controller to stay in idle state for // `AnalogAgcConfig::clipped_wait_frames` frames. - WriteAudioBufferSamples(/*samples_value=*/0.0f, /*clipping_ratio=*/0.0f, + WriteAudioBufferSamples(/*samples_value=*/0.0f, /*clipped_ratio=*/0.0f, audio_buffer); manager_no_analog_agc.AnalyzePreProcess(audio_buffer); manager_with_analog_agc.AnalyzePreProcess(audio_buffer); @@ -521,7 +521,7 @@ TEST_P(AgcManagerDirectParametrizedTest, GetOverrideOrEmpty(-18.0f)); // Feed clipping input to trigger a downward adapation of the analog level. - WriteAudioBufferSamples(/*samples_value=*/0.0f, /*clipping_ratio=*/0.2f, + WriteAudioBufferSamples(/*samples_value=*/0.0f, /*clipped_ratio=*/0.2f, audio_buffer); manager_no_analog_agc.AnalyzePreProcess(audio_buffer); manager_with_analog_agc.AnalyzePreProcess(audio_buffer); diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal_unittest.cc b/third_party/libwebrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal_unittest.cc @@ -99,14 +99,14 @@ TEST(RnnVadTest, ComputePitchPeriod48kHzBitExactness) { y_energy_view, cpu_features); // TODO(bugs.webrtc.org/8948): Add when the issue is fixed. // FloatingPointExceptionObserver fpe_observer; - EXPECT_EQ( - ComputePitchPeriod48kHz(test_data.PitchBuffer24kHzView(), y_energy_view, - /*pitch_candidates=*/{280, 284}, cpu_features), - 560); - EXPECT_EQ( - ComputePitchPeriod48kHz(test_data.PitchBuffer24kHzView(), y_energy_view, - /*pitch_candidates=*/{260, 284}, cpu_features), - 568); + EXPECT_EQ(ComputePitchPeriod48kHz( + test_data.PitchBuffer24kHzView(), y_energy_view, + /*pitch_candidates_24kHz=*/{280, 284}, cpu_features), + 560); + EXPECT_EQ(ComputePitchPeriod48kHz( + test_data.PitchBuffer24kHzView(), y_energy_view, + /*pitch_candidates_24kHz=*/{260, 284}, cpu_features), + 568); } struct PitchCandidatesParameters { diff --git a/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_adapter.cc b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_adapter.cc @@ -253,7 +253,7 @@ TransportFeedbackAdapter::ProcessTransportFeedback( << " packets because they were sent on a different route."; } return ToTransportFeedback(std::move(packet_result_vector), - feedback_receive_time, /*suports_ecn=*/false); + feedback_receive_time, /*supports_ecn=*/false); } std::optional<TransportPacketsFeedback> diff --git a/third_party/libwebrtc/modules/pacing/pacing_controller_unittest.cc b/third_party/libwebrtc/modules/pacing/pacing_controller_unittest.cc @@ -2320,7 +2320,7 @@ TEST_F(PacingControllerTest, BudgetDoesNotAffectRetransmissionInsTrial) { // Send a video packet so that we have a bit debt. pacer.EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo, kVideoSsrc, /*sequence_number=*/1, - /*capture_time=*/1, kPacketSize.bytes())); + /*capture_time_ms=*/1, kPacketSize.bytes())); EXPECT_CALL(callback_, SendPacket); pacer.ProcessPackets(); EXPECT_GT(pacer.NextSendTime(), clock_.CurrentTime()); @@ -2330,7 +2330,7 @@ TEST_F(PacingControllerTest, BudgetDoesNotAffectRetransmissionInsTrial) { pacer.EnqueuePacket(BuildPacket(RtpPacketMediaType::kRetransmission, kVideoSsrc, /*sequence_number=*/1, - /*capture_time=*/1, kPacketSize.bytes())); + /*capture_time_ms=*/1, kPacketSize.bytes())); pacer.ProcessPackets(); } @@ -2349,10 +2349,10 @@ TEST_F(PacingControllerTest, AbortsAfterReachingCircuitBreakLimit) { // Send two packets. pacer.EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo, kVideoSsrc, /*sequence_number=*/1, - /*capture_time=*/1, kPacketSize.bytes())); + /*capture_time_ms=*/1, kPacketSize.bytes())); pacer.EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo, kVideoSsrc, /*sequence_number=*/2, - /*capture_time=*/2, kPacketSize.bytes())); + /*capture_time_ms=*/2, kPacketSize.bytes())); // Advance time to way past where both should be eligible for sending. clock_.AdvanceTime(TimeDelta::Seconds(1)); @@ -2370,7 +2370,7 @@ TEST_F(PacingControllerTest, DoesNotPadIfProcessThreadIsBorked) { // Add one packet to the queue, but do not send it yet. pacer.EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo, kVideoSsrc, /*sequence_number=*/1, - /*capture_time=*/1, + /*capture_time_ms=*/1, /*size=*/1000)); // Advance time to waaay after the packet should have been sent. @@ -2402,17 +2402,17 @@ TEST_F(PacingControllerTest, FlushesPacketsOnKeyFrames) { // Enqueue a video packet and a retransmission of that video stream. pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kVideo, kSsrc, - /*sequence_number=*/1, /*capture_time=*/1, - /*size_bytes=*/100)); - pacer->EnqueuePacket(BuildPacket(RtpPacketMediaType::kRetransmission, - kRtxSsrc, - /*sequence_number=*/10, /*capture_time=*/1, + /*sequence_number=*/1, /*capture_time_ms=*/1, /*size_bytes=*/100)); + pacer->EnqueuePacket( + BuildPacket(RtpPacketMediaType::kRetransmission, kRtxSsrc, + /*sequence_number=*/10, /*capture_time_ms=*/1, + /*size_bytes=*/100)); EXPECT_EQ(pacer->QueueSizePackets(), 2u); // Enqueue the first packet of a keyframe for said stream. auto packet = BuildPacket(RtpPacketMediaType::kVideo, kSsrc, - /*sequence_number=*/2, /*capture_time=*/2, + /*sequence_number=*/2, /*capture_time_ms=*/2, /*size_bytes=*/1000); packet->set_is_key_frame(true); packet->set_first_packet_of_frame(true); diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc @@ -280,7 +280,7 @@ TEST(RtcpPacketTest, TransportFeedbackWithLargeBaseTimeIsConsistent) { constexpr Timestamp kTimestamp = Timestamp::Zero() + int64_t{0x7fff'ffff} * TransportFeedback::kDeltaTick; tb.SetBase(/*base_sequence=*/0, /*ref_timestamp=*/kTimestamp); - tb.AddReceivedPacket(/*base_sequence=*/0, /*ref_timestamp=*/kTimestamp); + tb.AddReceivedPacket(/*base_sequence=*/0, /*timestamp=*/kTimestamp); EXPECT_TRUE(tb.IsConsistent()); } diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc @@ -855,7 +855,8 @@ TEST_F(RtpSenderEgressTest, SendPacketUpdatesStats) { const ArrayView<const RtpExtensionSize> kNoRtpHeaderExtensionSizes; FlexfecSender flexfec(env_, kFlexfectPayloadType, kFlexFecSsrc, kSsrc, /*mid=*/"", - /*header_extensions=*/{}, kNoRtpHeaderExtensionSizes, + /*rtp_header_extensions=*/{}, + kNoRtpHeaderExtensionSizes, /*rtp_state=*/nullptr); RtpRtcpInterface::Configuration config = DefaultConfig(); config.fec_generator = &flexfec; diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_video_stream_receiver_frame_transformer_delegate_unittest.cc @@ -59,7 +59,7 @@ constexpr int kLastSeqNum = 2; std::unique_ptr<RtpFrameObject> CreateRtpFrameObject( const RTPVideoHeader& video_header, std::vector<uint32_t> csrcs) { - RtpPacketInfo packet_info(/*ssrc=*/123, csrcs, /*rtc_timestamp=*/0, + RtpPacketInfo packet_info(/*ssrc=*/123, csrcs, /*rtp_timestamp=*/0, /*receive_time=*/Timestamp::Seconds(123456)); return std::make_unique<RtpFrameObject>( kFirstSeqNum, kLastSeqNum, /*markerBit=*/true, @@ -324,7 +324,7 @@ TEST(RtpVideoStreamReceiverFrameTransformerDelegateTest, TestRtpVideoFrameReceiver receiver; auto mock_frame_transformer = make_ref_counted<NiceMock<MockFrameTransformer>>(); - SimulatedClock clock(/*initial_timestamp_us=*/12345000); + SimulatedClock clock(/*initial_time_us=*/12345000); auto delegate = make_ref_counted<RtpVideoStreamReceiverFrameTransformerDelegate>( &receiver, &clock, mock_frame_transformer, Thread::Current(), diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_receiver_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/ulpfec_receiver_unittest.cc @@ -525,7 +525,7 @@ TEST_F(UlpfecReceiverTest, MediaWithPadding) { ForwardErrorCorrection::PacketList media_packets; PacketizeFrame(2, 0, &augmented_media_packets, &media_packets); - augmented_media_packets.front().SetPadding(/*padding_bytes=*/4); + augmented_media_packets.front().SetPadding(/*padding_size=*/4); media_packets.front()->data = augmented_media_packets.front().Buffer(); std::list<ForwardErrorCorrection::Packet*> fec_packets; diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_vp8_unittest.cc @@ -197,7 +197,7 @@ TEST(VideoRtpDepacketizerVp8Test, TooShortHeader) { TEST(VideoRtpDepacketizerVp8Test, WithPacketizer) { uint8_t data[10] = {0}; - RtpPacketToSend packet(/*extenions=*/nullptr); + RtpPacketToSend packet(/*extensions=*/nullptr); RTPVideoHeaderVP8 input_header; input_header.nonReference = true; input_header.pictureId = 300; diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_test.cc b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_test.cc @@ -328,7 +328,7 @@ TEST_P(SpatialQualityTest, SpatialQuality) { VideoSourceSettings source_settings = ToSourceSettings(video_info); EncodingSettings encoding_settings = VideoCodecTester::CreateEncodingSettings( - env, codec_type, /*scalability_mode=*/"L1T1", width, height, + env, codec_type, /*scalability_name=*/"L1T1", width, height, {DataRate::KilobitsPerSec(bitrate_kbps)}, Frequency::Hertz(framerate_fps)); @@ -413,14 +413,14 @@ TEST_P(BitrateAdaptationTest, BitrateAdaptation) { VideoSourceSettings source_settings = ToSourceSettings(video_info); EncodingSettings encoding_settings = VideoCodecTester::CreateEncodingSettings( - env, codec_type, /*scalability_mode=*/"L1T1", + env, codec_type, /*scalability_name=*/"L1T1", /*width=*/640, /*height=*/360, {DataRate::KilobitsPerSec(bitrate_kbps.first)}, /*framerate=*/Frequency::Hertz(30)); EncodingSettings encoding_settings2 = VideoCodecTester::CreateEncodingSettings( - env, codec_type, /*scalability_mode=*/"L1T1", + env, codec_type, /*scalability_name=*/"L1T1", /*width=*/640, /*height=*/360, {DataRate::KilobitsPerSec(bitrate_kbps.second)}, /*framerate=*/Frequency::Hertz(30)); @@ -505,14 +505,14 @@ TEST_P(FramerateAdaptationTest, FramerateAdaptation) { VideoSourceSettings source_settings = ToSourceSettings(video_info); EncodingSettings encoding_settings = VideoCodecTester::CreateEncodingSettings( - env, codec_type, /*scalability_mode=*/"L1T1", + env, codec_type, /*scalability_name=*/"L1T1", /*width=*/640, /*height=*/360, /*bitrate=*/{DataRate::KilobitsPerSec(512)}, Frequency::Hertz(framerate_fps.first)); EncodingSettings encoding_settings2 = VideoCodecTester::CreateEncodingSettings( - env, codec_type, /*scalability_mode=*/"L1T1", + env, codec_type, /*scalability_name=*/"L1T1", /*width=*/640, /*height=*/360, /*bitrate=*/{DataRate::KilobitsPerSec(512)}, Frequency::Hertz(framerate_fps.second)); diff --git a/third_party/libwebrtc/net/dcsctp/timer/timer_test.cc b/third_party/libwebrtc/net/dcsctp/timer/timer_test.cc @@ -119,7 +119,7 @@ TEST_F(TimerTest, TimerWithNoRestarts) { std::unique_ptr<Timer> t1 = manager_.CreateTimer( "t1", on_expired_.AsStdFunction(), TimerOptions(TimeDelta::Seconds(5), TimerBackoffAlgorithm::kFixed, - /*max_restart=*/0)); + /*max_restarts=*/0)); EXPECT_CALL(on_expired_, Call).Times(0); t1->Start(); @@ -141,7 +141,7 @@ TEST_F(TimerTest, TimerWithOneRestart) { std::unique_ptr<Timer> t1 = manager_.CreateTimer( "t1", on_expired_.AsStdFunction(), TimerOptions(TimeDelta::Seconds(5), TimerBackoffAlgorithm::kFixed, - /*max_restart=*/1)); + /*max_restarts=*/1)); EXPECT_CALL(on_expired_, Call).Times(0); t1->Start(); @@ -170,7 +170,7 @@ TEST_F(TimerTest, TimerWithTwoRestart) { std::unique_ptr<Timer> t1 = manager_.CreateTimer( "t1", on_expired_.AsStdFunction(), TimerOptions(TimeDelta::Seconds(5), TimerBackoffAlgorithm::kFixed, - /*max_restart=*/2)); + /*max_restarts=*/2)); EXPECT_CALL(on_expired_, Call).Times(0); t1->Start(); diff --git a/third_party/libwebrtc/net/dcsctp/tx/retransmission_queue_test.cc b/third_party/libwebrtc/net/dcsctp/tx/retransmission_queue_test.cc @@ -1577,8 +1577,8 @@ TEST_F(RetransmissionQueueTest, CanAlwaysSendOnePacket) { // Ack 12, and report an empty receiver window (the peer obviously has a // tiny receive window). - queue.HandleSack( - now_, SackChunk(TSN(9), /*rwnd=*/0, {SackChunk::GapAckBlock(3, 3)}, {})); + queue.HandleSack(now_, SackChunk(TSN(9), /*a_rwnd=*/0, + {SackChunk::GapAckBlock(3, 3)}, {})); // Force TSN 10 to be retransmitted. queue.HandleT3RtxTimerExpiry(); @@ -1590,8 +1590,8 @@ TEST_F(RetransmissionQueueTest, CanAlwaysSendOnePacket) { EXPECT_THAT(queue.GetChunksToSend(now_, mtu), IsEmpty()); // Don't ack any new data, and still have receiver window zero. - queue.HandleSack( - now_, SackChunk(TSN(9), /*rwnd=*/0, {SackChunk::GapAckBlock(3, 3)}, {})); + queue.HandleSack(now_, SackChunk(TSN(9), /*a_rwnd=*/0, + {SackChunk::GapAckBlock(3, 3)}, {})); // There is in-flight data, so new data should not be allowed to be send since // the receiver window is full. @@ -1599,15 +1599,15 @@ TEST_F(RetransmissionQueueTest, CanAlwaysSendOnePacket) { // Ack that packet (no more in-flight data), but still report an empty // receiver window. - queue.HandleSack( - now_, SackChunk(TSN(10), /*rwnd=*/0, {SackChunk::GapAckBlock(2, 2)}, {})); + queue.HandleSack(now_, SackChunk(TSN(10), /*a_rwnd=*/0, + {SackChunk::GapAckBlock(2, 2)}, {})); // Then TSN 11 can be sent, as there is no in-flight data. EXPECT_THAT(queue.GetChunksToSend(now_, mtu), ElementsAre(Pair(TSN(11), _))); EXPECT_THAT(queue.GetChunksToSend(now_, mtu), IsEmpty()); // Ack and recover the receiver window - queue.HandleSack(now_, SackChunk(TSN(12), /*rwnd=*/5 * mtu, {}, {})); + queue.HandleSack(now_, SackChunk(TSN(12), /*a_rwnd=*/5 * mtu, {}, {})); // That will unblock sending remaining chunks. EXPECT_THAT(queue.GetChunksToSend(now_, mtu), ElementsAre(Pair(TSN(13), _))); diff --git a/third_party/libwebrtc/p2p/base/transport_description.cc b/third_party/libwebrtc/p2p/base/transport_description.cc @@ -87,7 +87,7 @@ RTCError ValidateIcePwd(absl::string_view raw_pwd) { RTCErrorOr<IceParameters> IceParameters::Parse(absl::string_view raw_ufrag, absl::string_view raw_pwd) { IceParameters parameters(std::string(raw_ufrag), std::string(raw_pwd), - /* renomination= */ false); + /* ice_renomination= */ false); auto result = parameters.Validate(); if (!result.ok()) { return result; diff --git a/third_party/libwebrtc/pc/peer_connection_callsetup_perf_tests.cc b/third_party/libwebrtc/pc/peer_connection_callsetup_perf_tests.cc @@ -63,7 +63,7 @@ class PeerConnectionDataChannelOpenTest : background_thread_(std::make_unique<Thread>(&vss_)) { RTC_CHECK(background_thread_->Start()); // Delay is set to 50ms so we get a 100ms RTT. - vss_.set_delay_mean(/*delay_ms=*/50); + vss_.set_delay_mean(/*delay_mean=*/50); vss_.UpdateDelayDistribution(); } diff --git a/third_party/libwebrtc/rtc_base/rate_statistics_unittest.cc b/third_party/libwebrtc/rtc_base/rate_statistics_unittest.cc @@ -319,7 +319,7 @@ TEST_F(RateStatisticsTest, HandlesSomewhatLargeNumbers) { } TEST_F(RateStatisticsTest, HandlesLowFps) { - RateStatistics fps_stats(/*window_size_ms=*/1000, /*scale=*/1000); + RateStatistics fps_stats(/*max_window_size_ms=*/1000, /*scale=*/1000); const int64_t kExpectedFps = 1; constexpr int64_t kTimeDelta = 1000 / kExpectedFps; @@ -341,7 +341,7 @@ TEST_F(RateStatisticsTest, HandlesLowFps) { } TEST_F(RateStatisticsTest, Handles25Fps) { - RateStatistics fps_stats(/*window_size_ms=*/1000, /*scale=*/1000); + RateStatistics fps_stats(/*max_window_size_ms=*/1000, /*scale=*/1000); constexpr int64_t kExpectedFps = 25; constexpr int64_t kTimeDelta = 1000 / kExpectedFps; diff --git a/third_party/libwebrtc/rtc_tools/rtc_event_log_to_text/converter.cc b/third_party/libwebrtc/rtc_tools/rtc_event_log_to_text/converter.cc @@ -85,7 +85,7 @@ bool Convert(std::string inputfile, ParsedRtcEventLog::UnconfiguredHeaderExtensions handle_unconfigured_extensions) { ParsedRtcEventLog parsed_log(handle_unconfigured_extensions, - /*allow_incomplete_logs=*/true); + /*allow_incomplete_log=*/true); auto status = parsed_log.ParseFile(inputfile); if (!status.ok()) { diff --git a/third_party/libwebrtc/rtc_tools/video_replay.cc b/third_party/libwebrtc/rtc_tools/video_replay.cc @@ -598,7 +598,7 @@ class RtpReplayer final { int64_t replay_start_ms = -1; int num_packets = 0; std::map<uint32_t, int> unknown_packets; - Event event(/*manual_reset=*/false, /*initially_signalled=*/false); + Event event(/*manual_reset=*/false, /*initially_signaled=*/false); uint32_t start_timestamp = absl::GetFlag(FLAGS_start_timestamp); uint32_t stop_timestamp = absl::GetFlag(FLAGS_stop_timestamp); diff --git a/third_party/libwebrtc/test/pc/e2e/peer_connection_quality_test.cc b/third_party/libwebrtc/test/pc/e2e/peer_connection_quality_test.cc @@ -468,7 +468,7 @@ void PeerConnectionE2EQualityTest::OnTrackCallback( auto* video_track = static_cast<VideoTrackInterface*>(track.get()); std::unique_ptr<VideoSinkInterface<VideoFrame>> video_sink = video_quality_analyzer_injection_helper_->CreateVideoSink( - peer_name, peer_subscription, /*report_infra_stats=*/false); + peer_name, peer_subscription, /*report_infra_metrics=*/false); video_track->AddOrUpdateSink(video_sink.get(), VideoSinkWants()); output_video_sinks_.push_back(std::move(video_sink)); } diff --git a/third_party/libwebrtc/video/end_to_end_tests/corruption_detection_tests.cc b/third_party/libwebrtc/video/end_to_end_tests/corruption_detection_tests.cc @@ -34,7 +34,7 @@ namespace { RtpExtension GetCorruptionExtension() { return RtpExtension(RtpExtension::kCorruptionDetectionUri, /*extension_id=*/1, - /*encrypted=*/true); + /*encrypt=*/true); } } // namespace diff --git a/third_party/libwebrtc/video/video_stream_encoder_unittest.cc b/third_party/libwebrtc/video/video_stream_encoder_unittest.cc @@ -9335,12 +9335,13 @@ TEST_F(VideoStreamEncoderTest, RecreatesEncoderWhenEnableVp9SpatialLayer) { video_encoder_config.encoder_specific_settings = make_ref_counted<VideoEncoderConfig::Vp9EncoderSpecificSettings>( vp9_settings); - video_encoder_config.spatial_layers = GetSvcConfig(1280, 720, - /*fps=*/30.0, - /*first_active_layer=*/0, - /*num_spatial_layers=*/2, - /*num_temporal_layers=*/3, - /*is_screenshare=*/false); + video_encoder_config.spatial_layers = + GetSvcConfig(1280, 720, + /*fps=*/30.0, + /*first_active_layer=*/0, + /*num_spatial_layers=*/2, + /*num_temporal_layers=*/3, + /*is_screen_sharing=*/false); ConfigureEncoder(video_encoder_config.Copy(), VideoStreamEncoder::BitrateAllocationCallbackType:: kVideoLayersAllocation); @@ -9511,7 +9512,7 @@ TEST_P(VideoStreamEncoderWithRealEncoderTest, HandlesLayerToggling) { /*first_active_layer=*/0, /*num_spatial_layers=*/3, /*num_temporal_layers=*/3, - /*is_screenshare=*/false); + /*is_screen_sharing=*/false); } else if (codec_type_ == VideoCodecType::kVideoCodecAV1) { test::FillEncoderConfiguration(codec_type_, 1, &config); config.max_bitrate_bps = kSimulcastTargetBitrate.bps(); @@ -9520,7 +9521,7 @@ TEST_P(VideoStreamEncoderWithRealEncoderTest, HandlesLayerToggling) { /*first_active_layer=*/0, /*num_spatial_layers=*/3, /*num_temporal_layers=*/3, - /*is_screenshare=*/false); + /*is_screen_sharing=*/false); config.simulcast_layers[0].scalability_mode = ScalabilityMode::kL3T3_KEY; } else { // Simulcast for VP8/H264.