[Rollback] Don't end tracks when transceiver is still in use.

Prior to this CL, calling RtpTransceiver::SetChannel() with null
arguments would cause the receiver's track to end. This is wrong,
because the channel can be nulled for other reasons than the transceiver
being stopped/removed - such as when the transceiver is rolled back but
still in use. Also, stopping a transceiver will end the track, so we
should simply ensure to always stop the transceiver when that is needed.

This CL makes sure that the transceiver is stopped or stopping in all
appropriate places, allowing us to remove the ability to end the source
for any other reason. A side-effect of this is that:
- The track never ends prematurely, fixing https://crbug.com/1315611.
- Removed transceivers are always stopped, fixing
  https://crbug.com/webrtc/14005.

This CL fixes the issue of track being ended in the ontrack event when
running https://jsfiddle.net/henbos/nxebusjm/.
- We don't have WPT test coverage for this, so I'll add that separately.

With SetSourceEnded() removed, some stopping/stop in response to
rejecting locally SDP munged content had to be added in order not to
regress the existing test coverage for this:
*PeerConnectionInterfaceTest.RejectMediaContent/1

Bug: chromium:1315611, webrtc:14005.
Change-Id: I21f30a1259e51324066dc84f72a72485b9e0fadc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260180
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36669}
9 files changed
tree: 5199ba149965b704a08bf71dacff1ccf30bc7fcd
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. g3doc/
  11. infra/
  12. logging/
  13. media/
  14. modules/
  15. net/
  16. p2p/
  17. pc/
  18. resources/
  19. rtc_base/
  20. rtc_tools/
  21. sdk/
  22. stats/
  23. system_wrappers/
  24. test/
  25. tools_webrtc/
  26. video/
  27. .clang-format
  28. .git-blame-ignore-revs
  29. .gitignore
  30. .gn
  31. .mailmap
  32. .style.yapf
  33. .vpython
  34. .vpython3
  35. AUTHORS
  36. BUILD.gn
  37. CODE_OF_CONDUCT.md
  38. codereview.settings
  39. DEPS
  40. DIR_METADATA
  41. ENG_REVIEW_OWNERS
  42. g3doc.lua
  43. LICENSE
  44. license_template.txt
  45. native-api.md
  46. OWNERS
  47. PATENTS
  48. PRESUBMIT.py
  49. presubmit_test.py
  50. presubmit_test_mocks.py
  51. pylintrc
  52. README.chromium
  53. README.md
  54. WATCHLISTS
  55. webrtc.gni
  56. webrtc_lib_link_test.cc
  57. 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