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]: https://webrtc-review.googlesource.com/c/src/+/221601

Bug: webrtc:12906, webrtc:12999
Change-Id: I4c6e7020c0be33a782d3608dee88e4e2fceb1be1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225642
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34544}
7 files changed
tree: a73b904cc39e4688a3f236a169fc0766bc31f10b
  1. .clang-format
  2. .git-blame-ignore-revs
  3. .gitignore
  4. .gn
  5. .vpython
  6. AUTHORS
  7. BUILD.gn
  8. CODE_OF_CONDUCT.md
  9. DEPS
  10. DIR_METADATA
  11. ENG_REVIEW_OWNERS
  12. LICENSE
  13. OWNERS
  14. PATENTS
  15. PRESUBMIT.py
  16. README.chromium
  17. README.md
  18. WATCHLISTS
  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/
  35. native-api.md
  36. net/
  37. p2p/
  38. pc/
  39. presubmit_test.py
  40. presubmit_test_mocks.py
  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
  52. webrtc_lib_link_test.cc
  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 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