commit | e21a4fec9f7f0b47c0467e191a8027961c3e8624 | [log] [tgz] |
---|---|---|
author | Lambros Lambrou <lambroslambrou@chromium.org> | Wed Jul 10 01:18:30 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Jul 10 01:18:30 2019 |
tree | e87df0b5bd89a70142ce99aef756e9b40a6d14f2 | |
parent | 6d599092b746d1dc34c31b293149aa302aa2753f [diff] |
[remoting host] AddTrack() -> AddTransceiver() This replaces the calls to PeerConnection::AddTrack() with AddTransceiver(), and updates the plumbing to pass RtpTransceivers instead of RtpSenders. Some of the newer WebRTC APIs operate on Transceivers, and it is easier to get a Sender from a Transceiver than it is to do the reverse. This will facilitate using the newer APIs to migrate away from SDP-munging, and perhaps some other future uses. There is no corresponding RemoveTransceiver() API, so the previous cleanup code is no longer needed. This also fixes a TODO to store the video sender (transceiver) instead of looping over all senders (transceivers). Tested by manually verifying video and audio are working, over multiple connect/disconnect cycles. Also verified audio is still received at roughly 160kbps (20k bytes/s). Change-Id: I071b16fa57dcee57faf4ab22e895185cf105aa9c Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/1694151 Commit-Queue: Lambros Lambrou <lambroslambrou@chromium.org> Commit-Queue: Jamie Walch <jamiewalch@chromium.org> Auto-Submit: Lambros Lambrou <lambroslambrou@chromium.org> Reviewed-by: Jamie Walch <jamiewalch@chromium.org> Cr-Commit-Position: refs/heads/master@{#675914}
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