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}
2 files changed
tree: 76fd5e06b64e78a5d31893adb7726d2426d79160
  1. build_overrides/
  2. chromium/
  3. data/
  4. infra/
  5. resources/
  6. third_party/
  7. tools/
  8. webrtc/
  9. .clang-format
  10. .gitignore
  11. .gn
  12. all.gyp
  13. AUTHORS
  14. BUILD.gn
  15. check_root_dir.py
  16. codereview.settings
  17. DEPS
  18. LICENSE
  19. license_template.txt
  20. LICENSE_THIRD_PARTY
  21. OWNERS
  22. PATENTS
  23. PRESUBMIT.py
  24. pylintrc
  25. README.md
  26. setup_links.py
  27. sync_chromium.py
  28. WATCHLISTS
README.md

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