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webrtc-version-updater 683db76c0f Update WebRTC code version (2023-10-23T04:12:13). 
Bug: None
Change-Id: Ib3e4ddcef7425761811aab4c27e18b4d64161846
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324581
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@{#40985}
2023-10-23 05:29:57 +00:00
..
adaptation
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
2023-05-03 11:09:26 +00:00
test
Revert "Clean up last_packet_received_time_ as it's no longer used."
2023-09-25 08:49:53 +00:00
audio_receive_stream.cc
…
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
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audio_send_stream.h
Replace "rcvd" with "received" for readability
2023-04-24 15:30:07 +00:00
audio_sender.h
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audio_state.cc
…
audio_state.h
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bitrate_allocator_unittest.cc
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bitrate_allocator.cc
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bitrate_allocator.h
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bitrate_estimator_tests.cc
Reland: FrameGeneratorCapturer: don't generate video before Start is called
2023-06-29 14:47:05 +00:00
BUILD.gn
Cleanup Call construction
2023-10-16 06:34:26 +00:00
call_config.cc
…
call_config.h
Introduce support for video packet batching.
2023-05-08 16:24:03 +00:00
call_factory.cc
Cleanup Call construction
2023-10-16 06:34:26 +00:00
call_factory.h
Cleanup Call construction
2023-10-16 06:34:26 +00:00
call_perf_tests.cc
Cleanup Call construction
2023-10-16 06:34:26 +00:00
call_unittest.cc
Cleanup Call construction
2023-10-16 06:34:26 +00:00
call.cc
Delete unnesseccary Call::RegisterReceiveStream and Call::DeregisterReceiveStream methods.
2023-10-18 10:17:52 +00:00
call.h
Cleanup Call construction
2023-10-16 06:34:26 +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
Remove internal overrides using old SendRtp and SendRtcp interfaces.
2023-08-15 13:20:21 +00:00
DEPS
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fake_network_pipe_unittest.cc
Reland "Delete PacketReceiver::DeliverPacket from all implementations"
2023-01-25 18:18:29 +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
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packet_receiver.h
Allow injecting packets of type Any to Call::DeliverRtpPacket
2023-03-29 06:36:17 +00:00
rampup_tests.cc
Replace WebRTC-QuickPerfTest field trial with a flag
2023-10-10 08:59:10 +00:00
rampup_tests.h
Stop overriding extensions in rampup tests
2023-01-25 13:18:49 +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
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receive_time_calculator.cc
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receive_time_calculator.h
…
rtp_bitrate_configurator_unittest.cc
…
rtp_bitrate_configurator.cc
…
rtp_bitrate_configurator.h
…
rtp_config.cc
…
rtp_config.h
Introduce support for video packet batching.
2023-05-08 16:24:03 +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
…
rtp_demuxer.h
…
rtp_packet_sink_interface.h
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rtp_payload_params_unittest.cc
When simulating chains from VP9 codec specific info support first_active_layer > 0
2023-08-03 13:19:00 +00:00
rtp_payload_params.cc
Add codec name H265 to support H265 in WebRTC
2023-09-20 09:25:32 +00:00
rtp_payload_params.h
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rtp_stream_receiver_controller_interface.h
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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
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rtp_transport_controller_send_factory_interface.h
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rtp_transport_controller_send_factory.h
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rtp_transport_controller_send_interface.h
Revert "Clean up last_packet_received_time_ as it's no longer used."
2023-09-25 08:49:53 +00:00
rtp_transport_controller_send.cc
Change expectation of GoogCCNetworkController::OnNetworkAvailability
2023-10-19 17:34:42 +00:00
rtp_transport_controller_send.h
Revert "Clean up last_packet_received_time_ as it's no longer used."
2023-09-25 08:49:53 +00:00
rtp_video_sender_interface.h
Reland "Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream"
2022-12-02 12:03:25 +00:00
rtp_video_sender_unittest.cc
Rename EncodedImage property Timetamp to RtpTimestamp
2023-09-24 20:06:48 +00:00
rtp_video_sender.cc
Move field trial check WebRTC-DisableRtxRateLimiter
2023-10-11 10:34:17 +00:00
rtp_video_sender.h
Add SetRetransmissionMode() to FecController, this will be used to control RTX settings in FecController.
2023-06-01 07:51:56 +00:00
rtx_receive_stream_unittest.cc
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rtx_receive_stream.cc
…
rtx_receive_stream.h
…
simulated_network_unittest.cc
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simulated_network.cc
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simulated_network.h
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simulated_packet_receiver.h
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syncable.cc
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syncable.h
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version.cc
Update WebRTC code version (2023-10-23T04:12:13).
2023-10-23 05:29:57 +00:00
version.h
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video_receive_stream.cc
Add missing comma in VideoReceiveStreamInterface::Stats::ToString
2023-10-17 10:42:06 +00:00
video_receive_stream.h
Remove default "unknown" encoderImplementation/decoderImplementation
2023-06-22 11:49:58 +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 default "unknown" encoderImplementation/decoderImplementation
2023-06-22 11:49:58 +00:00