PT redesign: handle raw, change allocation and update golden tests

Refactored `MergeCodecsFromConfigurations` to use a multi-pass approach
(Primary+RTX, then RED, then FEC) to restore the expected payload type
allocation order for video RED and ULPFEC codecs.

Modified `BasicOfferAnswerPayloadTypesStable` to use a different golden
set when the `WebRTC-PayloadTypesInTransport` field trial is enabled,
accounting for the different payload type allocation order in the
redesign path. Added the new golden set for
`kWebRtcTipOfTreeWithPayloadTypeRedesign`.

Also cleaned up `auto` usage in several files to comply with the style
guide.

Added a specific `SetRawPacketization` mechanism to `TypedCodecVendor`
and `CodecVendor` to support the `packetization=raw` hack applied via
SDP munging. This ensures that subsequent offers generated in the
redesign path also include the `packetization=raw` parameter for the
affected codec.

This avoids adding generic backwards propagation of munged codec changes
while keeping this specific hack working.

Verified with a new unit test in `codec_vendor_redesign_unittest.cc`
and by verifying that `MungeRawPacketizationChangesSubsequentSections`
now passes.

Bug: webrtc:360058654, webrtc:412904801
Change-Id: Ic6e5a270304ecbd69cfeb805b209336081c33c62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/475003
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47825}
10 files changed
tree: ae9cb9ca6703fcaab32dc0f528071932dcdb314d
  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