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android / platform / external / chromium_org / third_party / webrtc / 24065dc
commit24065dc5d44fb5eb37a0c6050cb1f85bddf972c9[log] [tgz]
authorstefan@webrtc.org <stefan@webrtc.org>Mon Sep 08 08:45:25 2014 +0000
committerstefan@webrtc.org <stefan@webrtc.org>Mon Sep 08 08:45:25 2014 +0000
treed8095498bf29734069652f2776cea0f7a0bc7b65
parent0b52f751e37055f5b91f4fd6a662b9daedb2635b [diff]
Fix RTT calculations for send-only channels.

As we don't know the SSRC of the other end in a send-only channel since we haven't received packets from that end, we are required to assume that the SSRC of the first report block is the correct SSRC to use for RTT calculations.

BUG=3781
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7097 4adac7df-926f-26a2-2b94-8c16560cd09d
1 file changed
tree: d8095498bf29734069652f2776cea0f7a0bc7b65
  1. base/
  2. build/
  3. common_audio/
  4. common_video/
  5. examples/
  6. libjingle/
  7. modules/
  8. overrides/
  9. sound/
  10. system_wrappers/
  11. test/
  12. tools/
  13. video/
  14. video_engine/
  15. voice_engine/
  16. .gitignore
  17. BUILD.gn
  18. call.h
  19. common.gyp
  20. common.h
  21. common_types.h
  22. config.cc
  23. config.h
  24. engine_configurations.h
  25. experiments.h
  26. frame_callback.h
  27. LICENSE
  28. LICENSE_THIRD_PARTY
  29. OWNERS
  30. PATENTS
  31. PRESUBMIT.py
  32. README.chromium
  33. supplement.gypi
  34. transport.h
  35. typedefs.h
  36. video_engine_tests.isolate
  37. video_receive_stream.h
  38. video_renderer.h
  39. video_send_stream.h
  40. webrtc.gyp
  41. webrtc_examples.gyp
  42. webrtc_perf_tests.isolate
  43. webrtc_tests.gypi