Reason for revert: Reverting because of the reasons given in comment #16: "This change breaks a scenario that is unfortunately not covered by unit tests, but can easily happen in a real call. The scenario that is broken by the change is this: 1. A sends an offer to B, with a set of codecs C_a (which is a subset of C_b, the codecs supported by B) 2. B responds with an answer, and sets the receive codecs to C_a. 3. At a later time, B generates a new offer which by default includes all codecs in C_b. 4. B calls SetLocalDescription() with this offer, that adds new receive codecs. 5. Adding the new codecs fails, because of the check at https://cs.chromium.org/chromium/src/third_party/webrtc/voice_engine/channel..... This causes SetLocalDescription() itself to fail. The net effect is that B cannot set a local description it just generated. Before the CL mentioned above, we'd stop playout before changing the codecs, and the operation would succeed." Original issue's description: > Removed the legacy behavior of stopping playout when setting new receive codecs. > > BUG=webrtc:4690 > > Committed: https://crrev.com/917d4e1e7131f35764cff932a8793151585e8179 > Cr-Commit-Position: refs/heads/master@{#14610} TBR=solenberg@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:4690 Review-Url: https://codereview.webrtc.org/2478433003 Cr-Commit-Position: refs/heads/master@{#14905}
Revert of Removed the legacy behavior of stopping playout when setting new receive codecs. (patchset #1 id:1 of https://codereview.webrtc.org/2409483003/ )
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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. This page is maintained by the Google Chrome team.
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://chromium.googlesource.com/external/webrtc
- 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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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