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webrtc / src.git / e5447fb6d1a1dafe5f0ece08bfc431ca4755139a
commite5447fb6d1a1dafe5f0ece08bfc431ca4755139a[log] [tgz]
authorSebastian Jansson <srte@webrtc.org>Tue Feb 20 18:38:47 2018
committerCommit Bot <commit-bot@chromium.org>Wed Feb 21 13:39:16 2018
treebbc958d06a8bc86895406b05c53ec17186372deb
parentdf023aa6b434e6845fa06a0dc3be0d6268658063 [diff]
Removed fake rtp transport controller send.

The fake rtp transport controller is only used by CallBitrateTest, but
the functionality tested in CallBitrateTest is now tested in
RtpBitrateConfiguratorTest. Removing the fake rtp transport controller
send reduces the complexity of refactoring the rtp transport controller
send interface.

Bug: webrtc:8415
Change-Id: I4673daea4e68521e7e14293514830d6e704219bc
Reviewed-on: https://webrtc-review.googlesource.com/54480
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22125}
5 files changed
tree: bbc958d06a8bc86895406b05c53ec17186372deb
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. examples/
  9. infra/
  10. logging/
  11. media/
  12. modules/
  13. ortc/
  14. p2p/
  15. pc/
  16. resources/
  17. rtc_base/
  18. rtc_tools/
  19. sdk/
  20. stats/
  21. system_wrappers/
  22. test/
  23. tools_webrtc/
  24. video/
  25. .clang-format
  26. .git-blame-ignore-revs
  27. .gitignore
  28. .gn
  29. .vpython
  30. AUTHORS
  31. BUILD.gn
  32. CODE_OF_CONDUCT.md
  33. codereview.settings
  34. common_types.cc
  35. common_types.h
  36. DEPS
  37. LICENSE
  38. license_template.txt
  39. LICENSE_THIRD_PARTY
  40. native-api.md
  41. OWNERS
  42. PATENTS
  43. PRESUBMIT.py
  44. presubmit_test.py
  45. presubmit_test_mocks.py
  46. pylintrc
  47. README.chromium
  48. README.md
  49. style-guide.md
  50. typedefs.h
  51. WATCHLISTS
  52. webrtc.gni
  53. whitespace.txt
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.

Development

See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

Authoritative list of directories that contain the native API header files.

More info