This is an integration test using peerconnectiontestwrapper.h to set up and end to end test using a real PeerConnection implementation. These tests will complement rtcstatscollector_unittest.cc which collects all stats using mocks. The integration test is set up so that all stats types are returned by GetStats and verifies that expected dictionary members are defined. The test could in the future be updated to include sanity checks for the values of members. There is a sanity check that references to other stats dictionaries yield existing stats of the appropriate type, but other than that members are only tested for if they are defined not. StatsCallback of rtcstatscollector_unittest.cc is moved so that it can be reused and renamed to RTCStatsObtainer. TODO: Audio stream track stats members are missing in the test. Find out if this is because of a real problem or because of testing without real devices. Do this before closing crbug.com/627816. BUG=chromium:627816 Review-Url: https://codereview.webrtc.org/2521663002 Cr-Commit-Position: refs/heads/master@{#15287}
Revert of CQ: Disable android_more_configs trybot (patchset #1 id:1 of https://codereview.webrtc.org/2522953003/ )
Revert of Bug in ExtractFrame API (extracts frames incorrectly) (patchset #9 id:130001 of https://codereview.webrtc.org/2529923002/ )
…
…
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.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Languages
C++
90.3%
Java
2.9%
C
2.2%
Objective-C++
2%
Python
1.3%
Other
1%