Fix receiver SSRC tracking for unsignaled streams

Since https://webrtc-review.googlesource.com/c/src/+/499601,
RTCRtpReceiver.getStats() filters the report by the SSRC that the
receiver tracks on the signaling thread, ssrc_s(). For unsignaled
streams, that SSRC could get out of sync with the worker thread's view:

- Renegotiating without signaled SSRCs reset it to nullopt. If the media
  channel kept the unsignaled stream, no new first packet notification
  followed and the receiver stayed unbound, so getStats() returned no
  inbound-rtp and the receiver's sources stopped updating.
- A first packet notification for a new unsignaled SSRC, e.g. when the
  remote endpoint changed the SSRC, was ignored since the receiver was
  already bound to the previous SSRC.

Track the signaled SSRC and the most recent unsignaled SSRC separately
on the signaling thread, mirroring RtpReceiverBase::ssrc() on the worker
thread. Setting up the receiver only updates the signaled SSRC, and a
first packet for a new unsignaled stream rebinds the first receiver that
doesn't have a signaled SSRC.

Bug: b:575939917
Change-Id: I1dff1ad7ce686322bd0dd60c1fee3557e7a7907e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/510881
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#49023}
12 files changed
tree: a180010ece9aa06fca087ca71c9467a678488130
  1. .agents/
  2. agents/
  3. api/
  4. audio/
  5. build_overrides/
  6. call/
  7. common_audio/
  8. common_video/
  9. data/
  10. docs/
  11. examples/
  12. experiments/
  13. g3doc/
  14. infra/
  15. logging/
  16. media/
  17. modules/
  18. net/
  19. p2p/
  20. pc/
  21. resources/
  22. rtc_base/
  23. rtc_tools/
  24. rust/
  25. sdk/
  26. stats/
  27. system_wrappers/
  28. test/
  29. tools_webrtc/
  30. video/
  31. .clang-format
  32. .clang-tidy
  33. .git-blame-ignore-revs
  34. .gitignore
  35. .gn
  36. .mailmap
  37. .rustfmt.toml
  38. .style.mdformat
  39. .style.yapf
  40. .yapfignore
  41. AUTHORS
  42. BUILD.gn
  43. CODE_OF_CONDUCT.md
  44. codereview.settings
  45. DEPS
  46. DIR_METADATA
  47. ENG_REVIEW_OWNERS
  48. GEMINI.md
  49. LICENSE
  50. license_template.txt
  51. native-api.md
  52. OWNERS
  53. OWNERS_INFRA
  54. PATENTS
  55. PRESUBMIT.py
  56. presubmit_test.py
  57. presubmit_test_mocks.py
  58. pylintrc
  59. pylintrc_old_style
  60. README.chromium
  61. README.md
  62. unsafe_buffers_paths.txt
  63. vpython.toml
  64. vpython.toml.uv.lock
  65. WATCHLISTS
  66. webrtc.gni
  67. webrtc_lib_link_test.cc
  68. 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