tag | fe62945a4b42048efa0709178055bcc96b767bcd | |
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tagger | The Android Open Source Project <initial-contribution@android.com> | Wed Feb 13 10:36:30 2019 -0800 |
object | b3cb8ab4ede8bb77f0bdef2715efc2c1e6267072 |
Android CTS 7.1 Release 25 (5240545)
commit | b3cb8ab4ede8bb77f0bdef2715efc2c1e6267072 | [log] [tgz] |
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author | Chih-hung Hsieh <chh@google.com> | Wed Jan 20 17:50:13 2016 +0000 |
committer | android-build-merger <android-build-merger@google.com> | Wed Jan 20 17:50:13 2016 +0000 |
tree | 28c4cf735dd5bd9cc8f1ccd06fff8a173b20d1cb | |
parent | a4acd9d6bc9b3b033d7d274316e75ee067df8d20 [diff] | |
parent | 9a337512d97e37afc142dee4fd50a41b741a87d2 [diff] |
Merge "Merge upstream SHA 04cb763" am: 9a337512d9 * commit '9a337512d97e37afc142dee4fd50a41b741a87d2': (797 commits) Add tests for verifying transport feedback for audio and video. Eliminate defines in talk/ Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ ) Remove assert which was incorrectly added to TcpPort::OnSentPacket. Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. Update with new default boringssl no-aes cipher suites. Re-enable tests. Revert of Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ ) Re-land: "Use an explicit identifier in Config" Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. Revert of Delete remnants of non-square pixel support from cricket::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/1586613002/ ) Remove libfuzzer trybot from default trybot set. Add ramp-up tests for transport sequence number with and w/o audio. Delete remnants of non-square pixel support from cricket::VideoFrame. Fix IPAddress::ToSensitiveString() to avoid dependency on inet_ntop(). Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ ) Re-enable tests that failed under Linux_Msan. Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ ) Roll chromium_revision 346fea9..099be58 (369082:369139) Disable WebRtcVideoChannel2BaseTest.SendManyResizeOnce for TSan Add build_protobuf variable. ...
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. This page is maintained by the Google Chrome team.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.