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webrtc_m130/call
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Tommi 16a8b25d80 Update local_ssrc without needing to recreate video streams. 
This is comparable to this change done previously for for audio streams:
https://webrtc-review.googlesource.com/c/src/+/222042

Bug: webrtc:11993
Change-Id: Ic953f816a8f7c56d1c3dc9a16d85bef3696a663d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261960
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36876}
2022-05-13 10:08:54 +00:00
..
adaptation
Avoid more usage of implicit conversion from scoped_refptr<T> to T*
2022-04-27 13:18:30 +00:00
test
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audio_receive_stream.cc
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audio_receive_stream.h
Add GetRtpExtensionMap to ReceiveStream and remove GetRtpExtensions.
2022-05-10 13:50:31 +00:00
audio_send_stream.cc
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audio_send_stream.h
Remove typing detection
2022-03-23 10:23:54 +00:00
audio_sender.h
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audio_state.cc
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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
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BUILD.gn
Add GetRtpExtensionMap to ReceiveStream and remove GetRtpExtensions.
2022-05-10 13:50:31 +00:00
call_config.cc
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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
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call_perf_tests.cc
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call_unittest.cc
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call.cc
Update local_ssrc without needing to recreate video streams.
2022-05-13 10:08:54 +00:00
call.h
Update local_ssrc without needing to recreate video streams.
2022-05-13 10:08:54 +00:00
degraded_call.cc
Update local_ssrc without needing to recreate video streams.
2022-05-13 10:08:54 +00:00
degraded_call.h
Update local_ssrc without needing to recreate video streams.
2022-05-13 10:08:54 +00:00
DEPS
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fake_network_pipe_unittest.cc
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fake_network_pipe.cc
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fake_network_pipe.h
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flexfec_receive_stream_impl.cc
Update local_ssrc without needing to recreate video streams.
2022-05-13 10:08:54 +00:00
flexfec_receive_stream_impl.h
Update local_ssrc without needing to recreate video streams.
2022-05-13 10:08:54 +00:00
flexfec_receive_stream_unittest.cc
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flexfec_receive_stream.cc
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flexfec_receive_stream.h
Add a transport_cc() getter and remove rtp_config().
2022-05-09 20:21:14 +00:00
OWNERS
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packet_receiver.h
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rampup_tests.cc
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rampup_tests.h
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receive_stream.h
Add GetRtpExtensionMap to ReceiveStream and remove GetRtpExtensions.
2022-05-10 13:50:31 +00:00
receive_time_calculator_unittest.cc
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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
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rtp_bitrate_configurator.cc
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rtp_bitrate_configurator.h
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rtp_config.cc
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rtp_config.h
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rtp_demuxer_unittest.cc
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rtp_demuxer.cc
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rtp_demuxer.h
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rtp_packet_sink_interface.h
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rtp_payload_params_unittest.cc
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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_stream_receiver_controller_interface.h
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rtp_stream_receiver_controller.cc
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rtp_stream_receiver_controller.h
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rtp_transport_config.h
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +00:00
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
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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_video_sender_interface.h
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rtp_video_sender_unittest.cc
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
2022-03-29 10:14:00 +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
rtx_receive_stream_unittest.cc
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rtx_receive_stream.cc
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rtx_receive_stream.h
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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 (2022-05-13T04:03:44).
2022-05-13 04:53:33 +00:00
version.h
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video_receive_stream.cc
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video_receive_stream.h
Add a transport_cc() getter and remove rtp_config().
2022-05-09 20:21:14 +00:00
video_send_stream.cc
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video_send_stream.h
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