Tommi bc37504297 Make sure to apply media options in RtpTransceiver::CreateChannel ... Ensures audio/video options are applied when RtpTransceiver::CreateChannel is called and the transceiver has pre-constructed channel objects. This involves some additional plumbing, setters for NetEq and AudioReceiveStream to allow updating jitter buffer settings. Added test for the regression detected in issue webrtc:470860829. Bug: webrtc:470860829 Change-Id: I30f6e0c4811c3d7f55114cceaeb6e4ab452a32fd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/440200 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#46585}		2026-01-08 10:28:53 -08:00
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mock_audio_receive_stream.h	Make sure to apply media options in RtpTransceiver::CreateChannel	2026-01-08 10:28:53 -08:00
mock_audio_send_stream.h	Apply include-cleaner to call/	2025-03-13 01:00:52 -07:00
mock_bitrate_allocator.h	Apply include-cleaner to call/	2025-03-13 01:00:52 -07:00
mock_rtp_packet_sink_interface.h	In call/ replace mock macros with unified MOCK_METHOD macro	2020-05-15 13:36:00 +00:00
mock_rtp_transport_controller_send.h	Reland "Ensure Answer after PrAnswer can change usage from CCFB to TWCC feedback"	2025-11-05 05:45:35 -08:00