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webrtc / src.git / add7ef974ee2642a3b55a36ec80be50a615bc60a
commitadd7ef974ee2642a3b55a36ec80be50a615bc60a[log] [tgz]
authorSteve Anton <steveanton@webrtc.org>Wed Jul 31 01:04:40 2019
committerCommit Bot <commit-bot@chromium.org>Wed Jul 31 19:19:56 2019
treef247154c01dd6a6c6d5fa332565ee138e571fab2
parentf6f3ecf4bbb72a586f74489d63bc72f184d33b3d [diff]
Sanitize the codec list before sending it to the media engine

The SDP can assign the same codec to two different payload types
which gets represented as two separate codecs in the SDP structure.
The media engine assumes that the client does not pass down
duplicate codecs. This change adds logic to BaseChannel to filter
out codecs of the same name with different payload types, picking
the one which is listed first in the m= line.

Bug: chromium:987598
Change-Id: I6fa813db1769e572ff7c3f322dc9b1de39817ea2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147602
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28726}
2 files changed
tree: f247154c01dd6a6c6d5fa332565ee138e571fab2
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. crypto/
  8. data/
  9. examples/
  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