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android / platform / external / chromium_org / third_party / webrtc / 9b707ca
commit9b707ca664bd22f7a8bfad14a06f9ffdc6129d3d[log] [tgz]
authorpbos@webrtc.org <pbos@webrtc.org>Wed Sep 03 16:17:12 2014 +0000
committerpbos@webrtc.org <pbos@webrtc.org>Wed Sep 03 16:17:12 2014 +0000
treee09d6211956e473e48f241439232d246f1f233ff
parentfc9d7791c3e19a69b1d7151d4011285792e42609 [diff]
Network up/down signaling in Call.

BUG=2429
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13109005

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7044 4adac7df-926f-26a2-2b94-8c16560cd09d
7 files changed
tree: e09d6211956e473e48f241439232d246f1f233ff
  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