commit | 41478c7c1b2f9aeb5ba201d6d0c4bf91a2fdd4f2 | [log] [tgz] |
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author | Sam Zackrisson <saza@webrtc.org> | Tue Oct 15 08:10:26 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Tue Oct 15 09:23:16 2019 |
tree | 02a87f97797f6c2bef6e3b3a9ce6880194f2a34e | |
parent | eb90e6ffe39510d270095f7739beb097f49fd6dc [diff] |
Remove AudioProcessing::gain_control() getter This change also resolves a bug in audioproc_f: The implicit ApplyConfig calls to enable gain control settings in aec_dump_simulator.cc:377-406 [1] are overwritten by the ApplyConfig call on line 500 using a config from line 292. Compared to a ToT build including a fix for that bug, these changes are bitexact on a large number of aecdumps. [1] https://cs.chromium.org/chromium/src/third_party/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc?l=377&rcl=8bbf9e2c6e40feb8efcbf276b43945a14d651e9b Bug: webrtc:9878 Change-Id: Id427d34e838c999d996d58193977ac2a9198edd6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156463 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29481}
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.
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.