This reverts commit a8264dbdd97f5e125d45fd0e84356f2e1f747df1. Reason for revert: Reverting to unblock rolls into Chromium. See failure here: https://build.chromium.org/p/tryserver.chromium.linux/builders/linux_chromium_rel_ng/builds/565449 Fails: external/wpt/webrtc/RTCPeerConnection-setRemoteDescription-offer.html I'm guessing these lines from the output are relevant: 12:15:32.525 11839 [1:19:1015/121532.495175:16438293900:ERROR:webrtcsession.cc(350)] Failed to set remote offer sdp: The order of m-lines in subsequent offer doesn't match order from previous offer/answer. 12:15:32.525 11839 [1:20:1015/121532.497199:16438296127:WARNING:delay_based_bwe.cc(326)] BWE Setting start bitrate to: 300000 12:15:32.525 11839 [1:1:1015/121532.498272:16438296963:ERROR:webrtcsdp.cc(359)] Failed to parse: "Invalid SDP". Reason: Expect line: v= 12:15:32.525 11839 [1:1:1015/121532.498364:16438297040:ERROR:rtc_peer_connection_handler.cc(2183)] Failed to create native session description. Type: offer SDP: Invalid SDP 12:15:32.525 11839 [1:1:1015/121532.498432:16438297104:ERROR:rtc_peer_connection_handler.cc(1458)] Failed to parse SessionDescription. Invalid SDP Expect line: v= Original change's description: > Reject the subsequent offer with fewer m= sections. > > If the subsequent offer contains fewer m= sections than the existing > description, it would be rejected. > > The helper method MediaSectionsInSameOrder is modified and it will > compare the number of m= sections before matching the media type. > > Bug: chromium:773620 > Change-Id: Ic8999445f4bc023da1d85a65659583db1687ec37 > Reviewed-on: https://webrtc-review.googlesource.com/9621 > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Commit-Queue: Zhi Huang <zhihuang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20298} TBR=deadbeef@webrtc.org,zhihuang@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: chromium:773620 Change-Id: I4a3ff7a42abb95144615b1dd37fb21585ee07b5d Reviewed-on: https://webrtc-review.googlesource.com/10920 Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20300}
Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ )
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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