Revert "Fix bug where we assume new m= sections will always be bundled."

This reverts commit d2b885fd91909f1b17fb11292a8c989d5d883b22.

Reason for revert: Speculative revert for Chromium importer

Original change's description:
> Fix bug where we assume new m= sections will always be bundled.
> A recent change [1] assumes that all new m= sections will share the
> first BUNDLE group (if one already exists), which avoids generating
> ICE candidates that are ultimately unnecessary. This is fine for JSEP
> endpoints, but it breaks the following scenarios for non-JSEP endpoints:
> * Remote offer adding a new m= section that's not part of any BUNDLE
>   group.
> * Remote offer adding an m= section to the second BUNDLE group.
> The latter is specifically problematic for any application that wants
> to bundle all audio streams in one group and all video streams in
> another group when using Unified Plan SDP, to replicate the behavior of
> using Plan B without bundling. It may try to add a video stream only
> for WebRTC to bundle it with audio.
> This is fixed by doing some minor re-factoring, having BundleManager
> update the bundle groups at offer time.
> Also:
> * Added some additional validation for multiple bundle groups in a
>   subsequent offer, since that now becomes relevant.
> * Improved rollback support, because now rolling back an offer may need
>   to not only remove mid->transport mappings but alter them.
> [1]:
> Bug: webrtc:12906, webrtc:12999
> Change-Id: I4c6e7020c0be33a782d3608dee88e4e2fceb1be1
> Reviewed-on:
> Reviewed-by: Harald Alvestrand <>
> Reviewed-by: Henrik Boström <>
> Commit-Queue: Taylor Brandstetter <>
> Cr-Commit-Position: refs/heads/master@{#34544}

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:12906, webrtc:12999
Change-Id: I00179d7573f322ad539ff16cad1f85320cfb2270
Reviewed-by: Björn Terelius <>
Reviewed-by: Taylor Brandstetter <>
Commit-Queue: Taylor Brandstetter <>
Cr-Commit-Position: refs/heads/master@{#34578}
7 files changed
tree: 8578b27289d5d42d45b4b624653b1fb52abb4ab0
  1. .clang-format
  2. .git-blame-ignore-revs
  3. .gitignore
  4. .gn
  5. .vpython
  9. DEPS
  13. OWNERS
  16. README.chromium
  19. api/
  20. audio/
  21. build_overrides/
  22. call/
  23. codereview.settings
  24. common_audio/
  25. common_video/
  26. data/
  27. docs/
  28. examples/
  29. g3doc.lua
  30. g3doc/
  31. license_template.txt
  32. logging/
  33. media/
  34. modules/
  36. net/
  37. p2p/
  38. pc/
  41. pylintrc
  42. resources/
  43. rtc_base/
  44. rtc_tools/
  45. sdk/
  46. stats/
  47. system_wrappers/
  48. test/
  49. tools_webrtc/
  50. video/
  51. webrtc.gni
  53. whitespace.txt

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.


See here 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