Revert "Move CodecParameterMap functions to rtp_parameters"

This reverts commit 882b287979da2d1a78e50849a3883bb1aad64a5c.

Reason for revert: Breaks downstream build during linking: backward reference detected: _ZN6webrtc27SessionDescriptionInterface6CreateENS_7SdpTypeENSt3__u10unique_ptrINS_18SessionDescriptionENS2_14default_deleteIS4_EEEENS2_17basic_string_viewIcNS2_11char_traitsIcEEEESB_NS2_6vectorINS_22IceCandidateCollectionENS2_9allocatorISD_EEEE in blaze-out/k8-opt/bin/third_party/webrtc/files/stable/webrtc/pc/_objs/webrtc_session_description_factory/webrtc_session_description_factory.pic.o refers to blaze-out/k8-opt/bin/third_party/webrtc/files/stable/webrtc/pc/_objs/webrtc_sdp/jsep_session_description.pic.o

I assume this is related to the circular dependency referenced in the comment of pc/BUILD.gn

Original change's description:
> Move CodecParameterMap functions to rtp_parameters
>
> This leaves webrtc_sdp.* with only the code related to parsing for
> SessionDescriptionInterface, unlocking the next step of co-locating that
> code with the jsep code which (circularly) depends on it.
>
> Bug: webrtc:42222470
> Change-Id: I460ea4f5fc47b5dafcefea2658a465f8ff97e84c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/414840
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#45864}

Bug: webrtc:42222470
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I2dc766968073958cc2f30307e848d56a2bf97414
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/415341
Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45869}
9 files changed
tree: 328a093a19e9d611bcb250d919124e6e8f2f9021
  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. .clang-tidy
  30. .git-blame-ignore-revs
  31. .gitignore
  32. .gn
  33. .mailmap
  34. .rustfmt.toml
  35. .style.yapf
  36. .vpython3
  37. AUTHORS
  38. BUILD.gn
  39. CODE_OF_CONDUCT.md
  40. codereview.settings
  41. DEPS
  42. DIR_METADATA
  43. ENG_REVIEW_OWNERS
  44. LICENSE
  45. license_template.txt
  46. native-api.md
  47. OWNERS
  48. OWNERS_INFRA
  49. PATENTS
  50. PRESUBMIT.py
  51. presubmit_test.py
  52. presubmit_test_mocks.py
  53. pylintrc
  54. pylintrc_old_style
  55. README.chromium
  56. README.md
  57. WATCHLISTS
  58. webrtc.gni
  59. webrtc_lib_link_test.cc
  60. 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