Reland "red: generate and parse the red fmtp format"

This is a reland of 9d0730942677a520ce7e184d081b4c5a2469fc48

Original change's description:
> red: generate and parse the red fmtp format
>
> generates a fmtp line like
>   a=fmtp:<red payloadtype> <opus payloadtype>/<opus payloadtype>
> and matches the incoming redundant payload types against the
> send codec one. Offers without an FMTP line will not use RED.
> Redundancy levels of 1 (plus main packet ) to 32 are accepted but
> this is not wired up to the encoder since the O/A semantic of
> RFC 2198 is not clear.
>
> This decreases the chance of a collision with the SATIN codec
> which also runs on 48khz (but so far does not specify a channelCount of 2)
>
> BUG=webrtc:11640
>
> Change-Id: I8755e5b1e944d105212f1bbe4f330cf4e0753e67
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229583
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#34848}

Bug: webrtc:11640
Change-Id: I9465e489897a8ded9845592477fe14678af7ab61
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230545
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34965}
2 files changed
tree: 372973381f70a9dbed01c44ee81bdc3344516621
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. g3doc/
  11. logging/
  12. media/
  13. modules/
  14. net/
  15. p2p/
  16. pc/
  17. resources/
  18. rtc_base/
  19. rtc_tools/
  20. sdk/
  21. stats/
  22. system_wrappers/
  23. test/
  24. tools_webrtc/
  25. video/
  26. .clang-format
  27. .git-blame-ignore-revs
  28. .gitignore
  29. .gn
  30. .vpython
  31. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. DEPS
  36. DIR_METADATA
  37. ENG_REVIEW_OWNERS
  38. g3doc.lua
  39. LICENSE
  40. license_template.txt
  41. native-api.md
  42. OWNERS
  43. PATENTS
  44. PRESUBMIT.py
  45. presubmit_test.py
  46. presubmit_test_mocks.py
  47. pylintrc
  48. README.chromium
  49. README.md
  50. WATCHLISTS
  51. webrtc.gni
  52. webrtc_lib_link_test.cc
  53. 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