PT: Fix test failures and PT mapping conflicts under WebRTC-PayloadTypesInTransport

This CL addresses multiple test failures in rtc_pc_unittests when the
WebRTC-PayloadTypesInTransport field trial is enabled:

1. Reverts a too-permissive directionality check bypass in
pc/codec_vendor.cc to restore correct negotiation of kInactive via
reversed offer direction, fixing legacy AudioCodecsAnswerTest
failures.

2. Adds BUNDLE support to FakePayloadTypeSuggester by mapping bundled
MIDs to share the same PayloadTypeRecorder instance.

3. Fixes TestBundleOfferWithSameCodecPlType by configuring bundle
groups on the fake suggester.

4. Avoids force-registering conflicting preferred payload types in
RegisterExpectations inside CodecLookupHelperForTesting if they are
already mapped. This allows fake suggester conflict resolution to
work correctly in tests.

5. Updates 15 asymmetric H265 level negotiation tests to use
CodecListsMatch instead of strict EXPECT_EQ, as TypedCodecVendor
does not pre-assign payload types in video_sendrecv_codecs().

Bug: webrtc:360058654
Change-Id: I9431d77b2e3eceba25c2df71f5fef69f70480495
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/477860
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47887}
3 files changed
tree: b993c1ae21a22b6e7f15592cd3aa25fb400b297a
  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.yapf
  39. .vpython3
  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. WATCHLISTS
  64. webrtc.gni
  65. webrtc_lib_link_test.cc
  66. 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