| commit | 3cc47ebd2d0efc6a48ddb5b142fa9ea9c1ae4435 | [log] [tgz] |
|---|---|---|
| author | asapersson <asapersson@webrtc.org> | Fri Sep 30 10:16:19 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Fri Sep 30 10:16:26 2016 |
| tree | 76fd5e06b64e78a5d31893adb7726d2426d79160 | |
| parent | f5297a019ea6de709a2bb94793d637bb44538931 [diff] |
Add sanity check for decreasing RTP timestamp in RtpToNtpMs. The capture time for a frame (capture_ms) is set later (in ViEEncoder::IncomingCapturedFrame) than the timestamp. Could potentially cause the RTP timestamp in consecutive RTCP SR to decrease. Example: // Frame1 46371: timestamp:2732, capture_ms:46373, rtcp SR ms: 46423 -> estimated current RTP timestamp:2732+(46423-46373)*90 = 7232 // Frame2 46404: timestamp:5702, capture_ms:46412, rtcp SR ms: 46428 -> estimated current RTP timestamp:5702+(46428-46412)*90 = 7142 // Diff: 33 ms: 33 ms, 39 ms, 5 ms BUG=b/31154867 Review-Url: https://codereview.webrtc.org/2354843003 Cr-Commit-Position: refs/heads/master@{#14454}
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.