commit | 8b3db59b6ec51c73a86ba991e8a2cb3334ec69b8 | [log] [tgz] |
---|---|---|
author | Alex Loiko <aleloi@webrtc.org> | Wed Feb 20 15:17:39 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Feb 20 15:17:49 2019 |
tree | 6db95d6d632fabcca8f4955e667e8d3a087de7bc | |
parent | 01fe3098030d87342cd9428cbdda7157d100dbe6 [diff] |
Revert "Reland of https://webrtc-review.googlesource.com/c/src/+/114883" This reverts commit 5341aaccdb64e3336abf5875e8828222446adffa. Reason for revert: Order of initialization of global static strings. Original change's description: > Reland of https://webrtc-review.googlesource.com/c/src/+/114883 > > The difference to the original is new bitexactness strings AND > global static file string constants. The reason for reland is breaking > downstream projects. > > Original CL description: > > Tests for multi-stream Opus. > > This CL (mainly) adds bit-exactness tests for multi-stream Opus. The > tests are in audio_coding_unittest.cc. Some refactoring of > AcmSendTestOldApi, AcmSenderBitExactnessOldApi is done to make it > possible. A few checks for "channels \in {1, 2}" are replaced with > "channels \in {1, 2, 4, 6, 8}" in the WebRTC Opus codec wrapper. A few > other changes are made to be able to write and read multi-channel WAV > files. > > The SDP changes are NOT included; as of this CL there is no way to set > up a multi-stream opus en/de-coder from SDP strings. > > Bug: webrtc:8649 > Change-Id: I9fd47c790c241c1876c4a731b0840bec30b4f1b2 > Reviewed-on: https://webrtc-review.googlesource.com/c/123387 > Reviewed-by: Oskar Sundbom <ossu@webrtc.org> > Commit-Queue: Alex Loiko <aleloi@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#26774} TBR=aleloi@webrtc.org,ossu@webrtc.org Change-Id: I88060f2050ccee83d6091b042a10f79b3c4edc47 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8649 Reviewed-on: https://webrtc-review.googlesource.com/c/123580 Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26777}
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.