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webrtc-version-updater 8de2cf7c93 Update WebRTC code version (2024-05-08T04:01:40). 
Bug: None
Change-Id: I4635d184e8a3ebdf83b956076c17232e761115b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350163
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#42254}
2024-05-08 05:48:56 +00:00
..
adaptation
Delete rtc::TaskQueue
2024-02-28 10:22:49 +00:00
test
Add missing header
2024-04-19 06:13:16 +00:00
audio_receive_stream.cc
Rename AudioReceiveStream to AudioReceiveStreamInterface
2022-05-23 08:44:26 +00:00
audio_receive_stream.h
Propagate time of the last received packet with Timestamp type
2023-06-02 14:29:19 +00:00
audio_send_stream.cc
Reland "Wire up non-sender RTT for audio, and implement related standardized stats."
2021-09-06 14:26:55 +00:00
audio_send_stream.h
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
audio_sender.h
Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api
2020-01-13 18:31:30 +00:00
audio_state.cc
…
audio_state.h
Move webrtc::AudioDeviceModule include to api/ folder
2024-04-22 08:56:31 +00:00
bitrate_allocator_unittest.cc
Adopt absl::string_view in call/
2022-05-17 12:00:45 +00:00
bitrate_allocator.cc
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 1
2022-03-09 13:23:21 +00:00
bitrate_allocator.h
Use backticks not vertical bars to denote variables in comments for /call
2021-07-27 18:29:33 +00:00
bitrate_estimator_tests.cc
Move call/simulated_network to test/network
2024-04-29 09:55:06 +00:00
BUILD.gn
Move call/simulated_network to test/network
2024-04-29 09:55:06 +00:00
call_config.cc
Use Environment in RtpTransportyControllerSend
2023-12-20 14:47:51 +00:00
call_config.h
Reland "FrameCadenceAdapter: align video encoding to metronome"
2024-01-08 13:54:56 +00:00
call_perf_tests.cc
Move call/simulated_network to test/network
2024-04-29 09:55:06 +00:00
call_unittest.cc
Pass Environment instead of clock to Fake video encoders at construction
2024-04-12 07:42:48 +00:00
call.cc
Provide Environment for ReceiveSideConfestionController construction
2024-05-07 08:02:36 +00:00
call.h
Pass Clock through Environment when constructing Call
2023-12-06 19:13:39 +00:00
create_call.cc
Pass Clock through Environment when constructing Call
2023-12-06 19:13:39 +00:00
create_call.h
Delete CallFactoryInterface as no longer needed
2023-12-05 15:44:43 +00:00
degraded_call.cc
Remove internal overrides using old SendRtp and SendRtcp interfaces.
2023-08-15 13:20:21 +00:00
degraded_call.h
Move call/simulated_network to test/network
2024-04-29 09:55:06 +00:00
DEPS
Move call/simulated_network to test/network
2024-04-29 09:55:06 +00:00
fake_network_pipe_unittest.cc
Move call/simulated_network to test/network
2024-04-29 09:55:06 +00:00
fake_network_pipe.cc
Delete unused constructor of FakeNetworkPipe
2023-08-18 13:07:10 +00:00
fake_network_pipe.h
Delete unused constructor of FakeNetworkPipe
2023-08-18 13:07:10 +00:00
flexfec_receive_stream_impl.cc
stats: implement flexfec fecBytesReceived stats for FlexFEC
2023-06-21 13:04:31 +00:00
flexfec_receive_stream_impl.h
stats: implement flexfec fecBytesReceived stats for FlexFEC
2023-06-21 13:04:31 +00:00
flexfec_receive_stream_unittest.cc
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
flexfec_receive_stream.cc
…
flexfec_receive_stream.h
stats: implement flexfec fecBytesReceived stats for FlexFEC
2023-06-21 13:04:31 +00:00
OWNERS
Make perkj owner of call/
2024-05-07 15:13:29 +00:00
packet_receiver.h
Dont create RTX receive stream before media SSRC is known
2024-02-22 14:40:43 +00:00
rampup_tests.cc
In EncoderStreamFactory pass field trials as required parameter
2024-04-17 12:53:30 +00:00
rampup_tests.h
Move call/simulated_network to test/network
2024-04-29 09:55:06 +00:00
receive_stream.h
Remove rtp header extension from config of Call audio and video receivers
2023-01-31 11:58:43 +00:00
receive_time_calculator_unittest.cc
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 2
2022-03-09 22:17:52 +00:00
receive_time_calculator.cc
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +00:00
receive_time_calculator.h
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +00:00
rtp_bitrate_configurator_unittest.cc
Revert "In RtpBitrateConfigurator ignore new parameters when set to default values."
2020-01-10 16:39:51 +00:00
rtp_bitrate_configurator.cc
Allow setting a bandwidth cap for relayed connections.
2020-03-26 20:41:46 +00:00
rtp_bitrate_configurator.h
Remove RTC_DISALLOW_COPY_AND_ASSIGN more.
2022-01-20 11:00:18 +00:00
rtp_config.cc
Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED
2021-11-15 21:44:59 +00:00
rtp_config.h
Use independet frame IDs between simulcast streams when WebRTC-GenericDescriptorAuth is disabled.
2024-03-19 10:03:36 +00:00
rtp_demuxer_unittest.cc
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
rtp_demuxer.cc
Remove SSRCs from libSRTP when removing them from the rtp_demuxer
2023-11-08 10:24:10 +00:00
rtp_demuxer.h
Remove SSRCs from libSRTP when removing them from the rtp_demuxer
2023-11-08 10:24:10 +00:00
rtp_packet_sink_interface.h
…
rtp_payload_params_unittest.cc
Use independet frame IDs between simulcast streams when WebRTC-GenericDescriptorAuth is disabled.
2024-03-19 10:03:36 +00:00
rtp_payload_params.cc
Use independet frame IDs between simulcast streams when WebRTC-GenericDescriptorAuth is disabled.
2024-03-19 10:03:36 +00:00
rtp_payload_params.h
Use independet frame IDs between simulcast streams when WebRTC-GenericDescriptorAuth is disabled.
2024-03-19 10:03:36 +00:00
rtp_stream_receiver_controller_interface.h
Demote RtpStreamReceiverController AddSink/RemoveSink to private
2022-07-06 09:31:54 +00:00
rtp_stream_receiver_controller.cc
Change RecoveredPacket::OnRecoveredPacket to produce webrtc::RtpPacketReceived
2022-12-22 14:04:21 +00:00
rtp_stream_receiver_controller.h
Change RecoveredPacket::OnRecoveredPacket to produce webrtc::RtpPacketReceived
2022-12-22 14:04:21 +00:00
rtp_transport_config.h
Add PeerConnectionInterface::ReconfigureBandwidthEstimation
2024-02-07 14:10:02 +00:00
rtp_transport_controller_send_factory_interface.h
Use Environment in RtpTransportyControllerSend
2023-12-20 14:47:51 +00:00
rtp_transport_controller_send_factory.h
Use Environment in RtpTransportyControllerSend
2023-12-20 14:47:51 +00:00
rtp_transport_controller_send_interface.h
PacketRouter directly notify RtpTransportControllerSender when sending
2024-03-28 09:27:43 +00:00
rtp_transport_controller_send.cc
Add ProbeController::EnableRepeatedInitialProbing
2024-05-07 19:37:23 +00:00
rtp_transport_controller_send.h
PacketRouter directly notify RtpTransportControllerSender when sending
2024-03-28 09:27:43 +00:00
rtp_video_sender_interface.h
Refactor RtpVideoSender::SetActiveModules.
2024-01-26 10:34:46 +00:00
rtp_video_sender_unittest.cc
Use independet frame IDs between simulcast streams when WebRTC-GenericDescriptorAuth is disabled.
2024-03-19 10:03:36 +00:00
rtp_video_sender.cc
PacketRouter directly notify RtpTransportControllerSender when sending
2024-03-28 09:27:43 +00:00
rtp_video_sender.h
Use independet frame IDs between simulcast streams when WebRTC-GenericDescriptorAuth is disabled.
2024-03-19 10:03:36 +00:00
rtx_receive_stream_unittest.cc
Store RtpPacketReceived::arrival_time as Timestamp.
2021-05-05 16:22:33 +00:00
rtx_receive_stream.cc
Updated associated payload types without recreating receive streams.
2022-08-16 13:38:24 +00:00
rtx_receive_stream.h
Updated associated payload types without recreating receive streams.
2022-08-16 13:38:24 +00:00
simulated_network.h
Move call/simulated_network to test/network
2024-04-29 09:55:06 +00:00
simulated_packet_receiver.h
…
syncable.cc
…
syncable.h
Rename AudioReceiveStream to AudioReceiveStreamInterface
2022-05-23 08:44:26 +00:00
version.cc
Update WebRTC code version (2024-05-08T04:01:40).
2024-05-08 05:48:56 +00:00
version.h
Add WebRTC code freshness version string.
2020-12-14 16:22:35 +00:00
video_receive_stream.cc
Add missing comma in VideoReceiveStreamInterface::Stats::ToString
2023-10-17 10:42:06 +00:00
video_receive_stream.h
stats: implement remote-outbound-rtp for video
2024-04-15 15:10:54 +00:00
video_send_stream.cc
Cleanup usasge of ReportBlockData::report_block accessor
2023-05-05 09:56:30 +00:00
video_send_stream.h
Remove VideoSendStream::StartPerRtpStream
2024-01-26 09:19:50 +00:00