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webrtc-version-updater d1fd9b5e73 Update WebRTC code version (2022-04-19T04:02:48). 
Bug: None
Change-Id: I7f1b7a2dc0e79b9cedf158e928d714eaccfd6ada
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259188
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@{#36567}
2022-04-19 05:03:52 +00:00
..
adaptation Remove //rtc_base:stringutils from public deps 2022-04-05 22:42:19 +00:00
test Remove unnecessary copy of suspended_ssrcs. 2021-11-29 10:54:30 +00:00
audio_receive_stream.cc Remove chromium clang style errors affecting sdk/android/media_jni 2018-04-09 13:55:49 +00:00
audio_receive_stream.h Add ability to set rtp header extensions without recreating streams. 2021-09-08 13:39:36 +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 Remove typing detection 2022-03-23 10:23:54 +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 Async audio processing API 2020-10-02 12:33:34 +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_allocator_unittest.cc Use backticks not vertical bars to denote variables in comments for /call 2021-07-27 18:29:33 +00:00
bitrate_estimator_tests.cc Adopt absl::string_view in rtc_base/log_sinks 2022-03-14 12:47:55 +00:00
BUILD.gn Remove platform_thread from //rtc_base:rtc_base_approved public_deps 2022-04-18 23:12:52 +00:00
call.cc WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf 2022-03-29 10:14:00 +00:00
call.h WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf 2022-03-29 10:14:00 +00:00
call_config.cc Add ability to pass factory for RtpTransportControllerSend to PeerConnectionFactoryDependencies. 2021-06-01 06:57:31 +00:00
call_config.h WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf 2022-03-29 10:14:00 +00:00
call_factory.cc WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf 2022-03-29 10:14:00 +00:00
call_factory.h Use SequenceChecker from public API 2021-02-10 15:04:55 +00:00
call_perf_tests.cc WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 4/inf 2022-03-14 14:36:35 +00:00
call_unittest.cc Use backticks not vertical bars to denote variables in comments for /call 2021-07-27 18:29:33 +00:00
degraded_call.cc WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf 2022-03-29 10:14:00 +00:00
degraded_call.h WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf 2022-03-29 10:14:00 +00:00
DEPS SimulcastEncoderAdapter: Use FramerateController instead of FramerateControllerDeprecated. 2021-08-30 10:20:55 +00:00
fake_network_pipe.cc Delete some unneeded references to ProcessThread. 2022-01-03 15:36:02 +00:00
fake_network_pipe.h Remove RTC_DISALLOW_COPY_AND_ASSIGN more. 2022-01-20 11:00:18 +00:00
fake_network_pipe_unittest.cc Use backticks not vertical bars to denote variables in comments for /call 2021-07-27 18:29:33 +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 Add rtp_config() accessor to ReceiveStream. 2021-06-14 17:57:57 +00:00
flexfec_receive_stream_impl.cc Delete some unneeded references to ProcessThread. 2022-01-03 15:36:02 +00:00
flexfec_receive_stream_impl.h Add ability to set rtp header extensions without recreating streams. 2021-09-08 13:39:36 +00:00
flexfec_receive_stream_unittest.cc Deduplicate set of the rtp header extension uri constants 2021-09-14 13:38:44 +00:00
OWNERS Fix WebRTC version updater bot. 2021-11-22 08:44:15 +00:00
packet_receiver.h Remove DeliverPacketAsync. 2021-05-29 07:37:33 +00:00
rampup_tests.cc WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 2 2022-03-09 22:17:52 +00:00
rampup_tests.h Add ImproveDirection to RampupTests 2022-01-10 12:34:23 +00:00
receive_stream.h Add ability to set rtp header extensions without recreating streams. 2021-09-08 13:39:36 +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
receive_time_calculator_unittest.cc WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 2 2022-03-09 22:17:52 +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_bitrate_configurator_unittest.cc Revert "In RtpBitrateConfigurator ignore new parameters when set to default values." 2020-01-10 16:39:51 +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 backticks not vertical bars to denote variables in comments 2021-08-10 10:40:03 +00:00
rtp_demuxer.cc Add ctor to RtpDemuxerCriteria to allow supplying 'mid' and 'rsid'. 2022-01-03 20:00:52 +00:00
rtp_demuxer.h Make RtpDemuxerCriteria's mid_ and rsid_ const. 2022-01-05 08:51:44 +00:00
rtp_demuxer_unittest.cc Add ctor to RtpDemuxerCriteria to allow supplying 'mid' and 'rsid'. 2022-01-03 20:00:52 +00:00
rtp_packet_sink_interface.h
rtp_payload_params.cc WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf 2022-03-29 10:14:00 +00:00
rtp_payload_params.h WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf 2022-03-29 10:14:00 +00:00
rtp_payload_params_unittest.cc WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 2 2022-03-09 22:17:52 +00:00
rtp_stream_receiver_controller.cc Remove lock from RtpStreamReceiverController. 2021-01-18 09:10:14 +00:00
rtp_stream_receiver_controller.h Use SequenceChecker from public API 2021-02-10 15:04:55 +00:00
rtp_stream_receiver_controller_interface.h
rtp_transport_config.h WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf 2022-03-29 10:14:00 +00:00
rtp_transport_controller_send.cc Update TQ pacer holdback window defaults. 2022-03-31 13:54:46 +00:00
rtp_transport_controller_send.h WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf 2022-03-29 10:14:00 +00:00
rtp_transport_controller_send_factory.h Make KeyValueConfig mandatory in the pacer. 2022-03-15 15:07:46 +00:00
rtp_transport_controller_send_factory_interface.h Add ability to pass factory for RtpTransportControllerSend to PeerConnectionFactoryDependencies. 2021-06-01 06:57:31 +00:00
rtp_transport_controller_send_interface.h Remove unnecessary copy of suspended_ssrcs. 2021-11-29 10:54:30 +00:00
rtp_video_sender.cc WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf 2022-03-29 10:14:00 +00:00
rtp_video_sender.h WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf 2022-03-29 10:14:00 +00:00
rtp_video_sender_interface.h Remove top-level const from parameters in function declarations. 2022-01-26 11:05:25 +00:00
rtp_video_sender_unittest.cc WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf 2022-03-29 10:14:00 +00:00
rtx_receive_stream.cc Store RtpPacketReceived::arrival_time as Timestamp. 2021-05-05 16:22:33 +00:00
rtx_receive_stream.h IWYU: uint32_t is defined in cstdint 2020-05-07 17:04:15 +00:00
rtx_receive_stream_unittest.cc Store RtpPacketReceived::arrival_time as Timestamp. 2021-05-05 16:22:33 +00:00
simulated_network.cc Use backticks not vertical bars to denote variables in comments 2021-08-10 10:40:03 +00:00
simulated_network.h Use backticks not vertical bars to denote variables in comments for /call 2021-07-27 18:29:33 +00:00
simulated_network_unittest.cc Replace DataSize and DataRate factories with newer versions 2020-02-18 16:09:50 +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 Make AV sync robust to failures to set a desired minimum delay 2020-09-09 15:44:47 +00:00
version.cc Update WebRTC code version (2022-04-19T04:02:48). 2022-04-19 05:03:52 +00:00
version.h Add WebRTC code freshness version string. 2020-12-14 16:22:35 +00:00
video_receive_stream.cc Prepare WebRtcVideoReceiveStream for configuration changes. 2021-07-01 11:23:51 +00:00
video_receive_stream.h Add a way to set keyframe request method on VideoReceiveStream 2022-01-26 08:09:40 +00:00
video_send_stream.cc Change the type of RTCVideoSourceStats.framesPerSecond 2021-11-16 11:21:41 +00:00
video_send_stream.h Remove more top-level const from parameters in function declarations 2022-02-01 09:15:50 +00:00