sdp: backfill default codec parameters for AV1

as required by
  https://aomediacodec.github.io/av1-rtp-spec/#72-sdp-parameters
Also unify usage of profile fmtp parameter. Most notably this causes
SDP answers to include the default values.

These default values correspond to libaom's default values for AV1E_SET_TARGET_SEQ_LEVEL_IDX, AV1E_SET_TIER_MASK as used in
https://source.chromium.org/chromium/chromium/src/+/main:third_party/libaom/source/libaom/aom/aomcx.h
and g_profile in aom_codec_enc_cfg
https://source.chromium.org/chromium/chromium/src/+/main:third_party/libaom/source/libaom/aom/aom_encoder.h;l=415;drc=b58207f5aecc39db7d3da766e7d171e5d2c3598e

Note: AV1 is inconsistently cased in variable/struct/method/class names. The canonical casing should probably be "Av1" since it is an acronym standing for "AOMedia Video 1".

BUG=webrtc:15703

Change-Id: I11864b7666fea906cd1a0759c7ad45997beab90e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331360
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41654}
12 files changed
tree: 0bd0566a0254fba5b195556e95d67273443d8359
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. experiments/
  11. g3doc/
  12. infra/
  13. logging/
  14. media/
  15. modules/
  16. net/
  17. p2p/
  18. pc/
  19. resources/
  20. rtc_base/
  21. rtc_tools/
  22. sdk/
  23. stats/
  24. system_wrappers/
  25. test/
  26. tools_webrtc/
  27. video/
  28. .clang-format
  29. .git-blame-ignore-revs
  30. .gitignore
  31. .gn
  32. .mailmap
  33. .style.yapf
  34. .vpython3
  35. AUTHORS
  36. BUILD.gn
  37. CODE_OF_CONDUCT.md
  38. codereview.settings
  39. DEPS
  40. DIR_METADATA
  41. ENG_REVIEW_OWNERS
  42. LICENSE
  43. license_template.txt
  44. native-api.md
  45. OWNERS
  46. OWNERS_INFRA
  47. PATENTS
  48. PRESUBMIT.py
  49. presubmit_test.py
  50. presubmit_test_mocks.py
  51. pylintrc
  52. pylintrc_old_style
  53. README.chromium
  54. README.md
  55. WATCHLISTS
  56. webrtc.gni
  57. webrtc_lib_link_test.cc
  58. 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