Let WebRtcVideoChannel::ResetUnsignaledRecvStream delete all default streams.

This CL changes WebRtcVideoChannel::ResetUnsignaledRecvStream so
that it deletes all default streams created by
WebRtcVideoChannel::AddRecvStream. This is needed for the case that
there are lingering default streams, whose SSRCs are different
from the SSRCs that were subsequently signaled. This can happen
when there are multiple "m= sections" and the early media is
sent to an "m= section" that is later not supposed to be the
sink for that particular SSRC.

Default streams whose SSRC match the subsequently signaled
SSRC is already handled here: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/media/engine/webrtc_video_engine.cc;l=1386;drc=22387b44ff173d263b434889d394cea90368ab06?originalUrl=https:%2F%2Fcs.chromium.org%2F

Bug: webrtc:11477
Change-Id: I96ed7e35b4904fb0757fe5824f8afa6f1b9a565e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172622
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30971}
2 files changed
tree: a9006fa0e0545430b2de38d6b463ecc2a879e4ca
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  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. 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 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