commit | bd7392829a81d0720887dae9ac0bf83da8d9e80d | [log] [tgz] |
---|---|---|
author | Taylor Brandstetter <deadbeef@webrtc.org> | Fri Apr 20 15:58:11 2018 |
committer | Commit Bot <commit-bot@chromium.org> | Fri Apr 20 15:58:25 2018 |
tree | f514e73f8ca9837e978cad3b76902fe92e8a14d5 | |
parent | 4049a25afd47ae0299c075ea22e2de8fbd205e16 [diff] |
Revert "Reland "Remove our stream << overloads from non-test build targets."" This reverts commit d7ee72041f882c023c73e27a7436c626c4e43604. Reason for revert: Broke downstream build which was using SdpAudioFormat operator<< Original change's description: > Reland "Remove our stream << overloads from non-test build targets." > > This is a reland of c841d18d257ba8e4ed7d77d105e3c46006bb1e7e > > Original change's description: > > Remove our stream << overloads from non-test build targets. > > > > Most are removed entirely, but RtcErrorType, RtpTransceiverDirection, IPAddress and > > SocketAddress are kept behind gtest's #ifdef UNIT_TEST. > > > > Bug: webrtc:8982 > > Change-Id: I36db19891e7d25aeacb08b9a08aa2b4004765e70 > > Reviewed-on: https://webrtc-review.googlesource.com/64143 > > Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> > > Reviewed-by: Benjamin Wright <benwright@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Reviewed-by: Åsa Persson <asapersson@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22916} > > TBR=deadbeef@webrtc.org,kwiberg@webrtc.org,asapersson@webrtc.org,jonasolsson@webrtc.org,benwright@webrtc.org > > Bug: webrtc:8982 > Change-Id: Ibe08c6270e5e693eb661a6ce9e8f074b34ef8123 > Reviewed-on: https://webrtc-review.googlesource.com/71161 > Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> > Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22949} TBR=deadbeef@webrtc.org,kwiberg@webrtc.org,asapersson@webrtc.org,jonasolsson@webrtc.org,benwright@webrtc.org Change-Id: I3c2b18ec2877d68a522ecbae7a2955c4eecf36df No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8982 Reviewed-on: https://webrtc-review.googlesource.com/71446 Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22963}
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.