358c94f5dd
This patch modifies AudioState to always call InitRecording before StartRecording(). This makes it possible to do SetRecording(false) + SetRecording(true), which before this patch would not actually work if there was sending streams. The only way was to add/remove streams...via SDP operations, puh :(. Bonus: We also needed to modifu AndroidAudioDeviceModule (which is a thin wrapper) so that StopRecording() will call AudioInput->StopRecording() even when recording is not enabled. BUG=b/397376626 Change-Id: I954b5caab11225b544c3e6a78c5dde357d4eedb5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/378140 Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Auto-Submit: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43946}
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 here 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: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs
- Documentation