commit | 5ad789ceffd41f5b8dab3bf45f487b9c394bca16 | [log] [tgz] |
---|---|---|
author | Alessio Bazzica <alessiob@webrtc.org> | Wed Mar 13 10:51:44 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Mar 13 15:33:29 2019 |
tree | a630e039e67cebf4a4ed87991c5fdd6323e80a60 | |
parent | 123f3453e2df51e797f75652f6ed4c3c052704c2 [diff] |
Reland "NetEQ RTP Play: Optionally write output audio file" This reverts commit c4b391a257ebf85448e58e73a96eb267635b6d6a. Reason for revert: issue fixed Original change's description: > Revert "NetEQ RTP Play: Optionally write output audio file" > > This reverts commit 6330818ec8159ee476481ba4a89f884fb3653f3f. > > Reason for revert: This breaks api/test/neteq_simulator_factory.cc, which unfortunately was not caught by our bots. > > Original change's description: > > NetEQ RTP Play: Optionally write output audio file > > > > This CL makes the output audio file optional to more > > quickly run neteq_rtpplay when no audio output is needed. > > The CL also includes necessary adaptations because of pre-existing > > dependencies (e.g., the output audio file name is used to create > > the plotting script file names). > > > > The command line arguments are retro-compatible - i.e., same behavior > > when specifying the output audio file and the new flag > > --output_files_base_name is not used. > > > > This CL also includes a test script with which the retro-compatibility > > has been verified. > > > > Bug: webrtc:10337 > > Change-Id: Ie3f301b3b2ed0682fb74426d9cf452396f2b112b > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126224 > > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > > Reviewed-by: Ivo Creusen <ivoc@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#27067} > > TBR=henrik.lundin@webrtc.org,alessiob@webrtc.org,ivoc@webrtc.org > > Change-Id: I0c63a8ba9566ef567ee398f571f2a511916fa742 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:10337 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127293 > Reviewed-by: Ivo Creusen <ivoc@webrtc.org> > Commit-Queue: Ivo Creusen <ivoc@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27078} TBR=henrik.lundin@webrtc.org,alessiob@webrtc.org,ivoc@webrtc.org Change-Id: Ia7061f7c2d69db61638ad612e82cd429eb49d539 Bug: webrtc:10337 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127540 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27106}
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.