Revert "Improves native Android audio implementations." This reverts commit 92b1ffd0f655e88532cb7313707f300fec911b46. Reason for revert: <INSERT REASONING HERE> Original change's description: > Improves native Android audio implementations. > > Summary: > > Adds AudioTrackStartErrorCode to separate different types of error > codes in combination with StartPlayout. > > Harmonizes WebRtcAudioRecord and WebRtcAudioTrack implementations > to ensure that init/start/stop is performed identically. > > Adds thread checking in WebRtcAudio track. > > Bug: webrtc:8453 > Change-Id: Ic913e888ff9493c9cc748a7b4dae43eb6b37fa85 > Reviewed-on: https://webrtc-review.googlesource.com/15481 > Commit-Queue: Henrik Andreassson <henrika@webrtc.org> > Reviewed-by: Alex Glaznev <glaznev@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20448} TBR=henrika@webrtc.org,glaznev@webrtc.org Change-Id: If1d1d9717387a4a8f6d9d6acf7e86ded4c655b5e No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8453 Reviewed-on: https://webrtc-review.googlesource.com/16321 Reviewed-by: Zhi Huang <zhihuang@webrtc.org> Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20452}
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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