Don't crash when SDP describes fewer send layers

RtpSenderBase::SetSsrcTask() applies the encodings the application
configured via addTransceiver() to the media channel once a local
description has been applied. The media channel derives its encodings
from the negotiated StreamParams, so the number of layers it knows
about is whatever the local description describes.

When the description describes fewer layers than the application asked
for, for instance because simulcast was removed by SDP munging, or
because the sender is attached to an ssrc the media channel has no send
stream for, the two views disagree and the RTC_CHECK_GE in the worker
thread lambda terminates the process. The mismatch is a negotiation
outcome, not memory corruption, so crashing is too strong a response
and the SDP remains authoritative either way.

Apply the layers that the media channel knows about, log a warning for
the ones that are dropped, and leave the channel untouched when it has
no send stream for the ssrc, which is the same treatment ssrc 0 gets.

Bug: webrtc:558756689, webrtc:562462476
Change-Id: I1775221261eae116f55d5fc45e3e5a7417db88b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/503820
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#48615}
3 files changed
tree: 7ff2099cc8d1de75f0c9f7b0f64def56c6c35e3e
  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