commit | 311428fecbc8b7e717391c60a71f43fe80645c47 | [log] [tgz] |
---|---|---|
author | Piasy Xu <xz4215@gmail.com> | Thu May 31 11:47:33 2018 |
committer | Commit Bot <commit-bot@chromium.org> | Thu May 31 18:44:28 2018 |
tree | e9ff9a25f0978eada25208e6ec9479d6d45c1c51 | |
parent | cfecd9e8d32360c5e2a0edb61f80b2011743d27a [diff] |
Remove unnecessary set_stream_ids call Both AudioRtpSender and VideoRtpSender receive stream_ids in their constructor, no need to call set_stream_ids again. Bug: None Change-Id: I6238a6d6e31076a0b3245c89e2825d8dee5166c0 Reviewed-on: https://webrtc-review.googlesource.com/80220 Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23476}
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.