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webrtc / src.git / 63a176b34f5fced24b7f8c80b4221436c2a3fce3
commit63a176b34f5fced24b7f8c80b4221436c2a3fce3[log] [tgz]
authorPiotr (Peter) Slatala <psla@webrtc.org>Fri Jan 25 16:25:33 2019
committerCommit Bot <commit-bot@chromium.org>Fri Jan 25 18:19:17 2019
treee307b7cc66cca22ca35d8dabbdb533de452722c9
parent18f65dc20a84cab43303b091b1bdde594c39fe73 [diff]
Do not modify media transport config when falling back to RTP

It is possible that media transport is re-set by the caller, but once
disabled it should stay disabled.

it's possible to fail this check the check in JsepTransportController::SetMediaTransportFactory in such case.

We should also change the caller to not invoke SetMediaTransportFactory
multiple times (with the same value), but I'll leave it as an excercise
to someone else :)

Bug: webrtc:9719
Change-Id: Ideea8a50d863edf4ef59e594a78c74bb9aba5aa7
Reviewed-on: https://webrtc-review.googlesource.com/c/119911
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26411}
2 files changed
tree: e307b7cc66cca22ca35d8dabbdb533de452722c9
  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. p2p/
  14. pc/
  15. resources/
  16. rtc_base/
  17. rtc_tools/
  18. sdk/
  19. stats/
  20. style-guide/
  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. abseil-in-webrtc.md
  31. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. common_types.h
  36. DEPS
  37. ENG_REVIEW_OWNERS
  38. LICENSE
  39. license_template.txt
  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. WATCHLISTS
  51. webrtc.gni
  52. 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