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webrtc-version-updater e9d066d3b7 Update WebRTC code version (2024-07-26T04:03:34). 
Bug: None
Change-Id: Ic9535697379751a522a125a131340e97c90fdbfb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357841
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@{#42673}
2024-07-26 06:05:27 +00:00
..
adaptation
Deprecate rtc::RefCountInterface
2024-06-07 09:47:26 +00:00
test
Expose a PeerConnection's NetworkControllerInterface instance
2024-06-18 08:04:03 +00:00
audio_receive_stream.cc
Rename AudioReceiveStream to AudioReceiveStreamInterface
2022-05-23 08:44:26 +00:00
audio_receive_stream.h
Computing and propagating the audio stats totalprocessingdelay.
2024-06-18 08:05:28 +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
RTCP: implement reduced size RTCP for audio
2024-05-16 18:24:10 +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
Remove chromium clang style errors affecting sdk/android/media_jni
2018-04-09 13:55:49 +00:00
audio_state.h
Deprecate rtc::RefCountInterface
2024-06-07 09:47:26 +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
Mass removal of absl_deps in all BUILD.gn files
2024-05-23 15:09:46 +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
Ensure test capturer does not set ntp time
2024-05-29 10:26:54 +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
Expose network to deliverrtp histogram metric
2024-05-29 08:50:22 +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
Reland "Replace usage of link_capacity_kbps with DataRate link_capacity"
2024-05-16 10:39:10 +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
[Cleanup] Add missing #include. Remove useless ones.
2018-10-23 11:32:56 +00:00
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
Reland "Replace usage of link_capacity_kbps with DataRate link_capacity"
2024-05-16 10:39:10 +00:00
rampup_tests.h
Move call/simulated_network to test/network
2024-04-29 09:55:06 +00:00
receive_stream.h
RTCP: implement reduced size RTCP for audio
2024-05-16 18:24:10 +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
Expose a PeerConnection's NetworkControllerInterface instance
2024-06-18 08:04:03 +00:00
rtp_transport_controller_send.cc
Fix GoogCcNetworkController::OnNetworkStateEstimate behaviour
2024-06-10 10:43:57 +00:00
rtp_transport_controller_send.h
Expose a PeerConnection's NetworkControllerInterface instance
2024-06-18 08:04:03 +00:00
rtp_video_sender_interface.h
Refactor RtpVideoSender::SetActiveModules.
2024-01-26 10:34:46 +00:00
rtp_video_sender_unittest.cc
Change WebRTC-Video-SimulcastIndependentFrameIds to a kill-switch.
2024-05-23 09:42:23 +00:00
rtp_video_sender.cc
Change WebRTC-Video-SimulcastIndependentFrameIds to a kill-switch.
2024-05-23 09:42:23 +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
Calculate next process time in simulated network.
2019-02-08 19:33:17 +00:00
syncable.cc
…
syncable.h
Rename AudioReceiveStream to AudioReceiveStreamInterface
2022-05-23 08:44:26 +00:00
version.cc
Update WebRTC code version (2024-07-26T04:03:34).
2024-07-26 06:05:27 +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
RTCP: implement reduced size RTCP for audio
2024-05-16 18:24:10 +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