801500cf99
Specifically, don't expect the ACM to be able to create encoders; we have to give it an encoder that we make ourselves. The new way of creating encoders used a 32 kbit/s bitrate unconditionally for iSAC; I had to change it to 32 kbit/s for 16 kHz and 56 kbit/s for 32 kHz, which is what the old way of creating encoders has used since forever. I also had to change some test expectations on Opus, because the new way defaults to 32 kbit/s for mono and 64 kbit/s for stereo (which I believe to be correct), while the old way defaults to 64 kbit/s in both cases. Bug: webrtc:8396 Change-Id: I3aab944175a8e27f4c63380e822b27e839bba7f2 Reviewed-on: https://webrtc-review.googlesource.com/94540 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24375}
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.
Authoritative list of directories that contain the native API header files.
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