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webrtc / src / webrtc / a99933612c91b87df9f397ce70d1192d3069b628
commita99933612c91b87df9f397ce70d1192d3069b628[log] [tgz]
authorkwiberg@webrtc.org <kwiberg@webrtc.org>Wed Oct 29 08:38:50 2014
committerkwiberg@webrtc.org <kwiberg@webrtc.org>Wed Oct 29 08:38:50 2014
tree76c5ffb7997fdb47da111555b576334b0c7e5b2e
parentff8f833892a08f89314cc886c9a3190f6baa3212 [diff]
AudioEncoder: num_10ms_frames_per_packet -> Num10MsFramesInNextPacket

Rename this accessor function to reflect its new, slightly changed
meaning. The reason for the change is that some codecs (iSAC) vary the
number of 10 ms frames from packet to packet, and so can't return a
truly constant value.

BUG=3926
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7556 4adac7df-926f-26a2-2b94-8c16560cd09d
6 files changed
tree: 76c5ffb7997fdb47da111555b576334b0c7e5b2e
  1. base/
  2. build/
  3. common_audio/
  4. common_video/
  5. examples/
  6. libjingle/
  7. modules/
  8. overrides/
  9. p2p/
  10. sound/
  11. system_wrappers/
  12. test/
  13. tools/
  14. video/
  15. video_engine/
  16. voice_engine/
  17. .gitignore
  18. BUILD.gn
  19. call.h
  20. common.gyp
  21. common.h
  22. common_types.h
  23. config.cc
  24. config.h
  25. engine_configurations.h
  26. experiments.h
  27. frame_callback.h
  28. LICENSE
  29. LICENSE_THIRD_PARTY
  30. OWNERS
  31. PATENTS
  32. PRESUBMIT.py
  33. README.chromium
  34. rtc_unittests.isolate
  35. supplement.gypi
  36. transport.h
  37. typedefs.h
  38. video_encoder.h
  39. video_engine_tests.isolate
  40. video_frame.h
  41. video_receive_stream.h
  42. video_renderer.h
  43. video_send_stream.h
  44. webrtc.gyp
  45. webrtc_examples.gyp
  46. webrtc_perf_tests.isolate
  47. webrtc_tests.gypi