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Introduce MediaTransportConfig

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Currently we pass media_transport from PeerConnection to media layers. The goal of this change is to replace media_transport with struct MediaTransportCondif, which will enable adding different transports (i.e. we plan to add DatagramTransport) as well as other media-transport related settings without changing 100s of files.

TODO: In the future we should consider also adding rtp_transport in the same config, but it will require a bit more work, so I did not include it in the same change.


Bug: webrtc:9719
Change-Id: Ie31e1faa3ed9e6beefe30a3da208130509ce00cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137181
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28016}
This commit is contained in:
Anton Sukhanov
2019-05-21 11:12:57 -07:00
committed by Commit Bot
parent 4880e15707
commit 4f08faae82
42 changed files with 329 additions and 206 deletions
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3
call/call.cc
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@ -708,7 +708,8 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream(
TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
RTC_DCHECK_RUN_ON(&configuration_sequence_checker_);
RTC_DCHECK(media_transport() == config.media_transport);
RTC_DCHECK_EQ(media_transport(),
config.media_transport_config.media_transport);
RegisterRateObserver();