Revert "Detect codec collisions between audio and video sections"

This reverts commit a8aae89f379e36cd418bf4ef194115d2ca231a1c.

Reason for revert: The test is failing on downstream bots:

pc/test/integration_test_helpers.h:900
Value of: SetRemoteDescription(std::move(desc))
  Actual: false
Expected: true

Original change's description:
> Detect codec collisions between audio and video sections
>
> This bug was reproduced as a side effect of fixing
> issues.chromium.org/395077842
>
> Bug: webrtc:42224689
> Change-Id: I41c2bb02a6ec9fb9e9c057d64255dd7896da4f4d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/377460
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Auto-Submit: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#47664}

Bug: webrtc:42224689
Change-Id: Iefc1872c330a03ea2b6e9ef12aee1fcdfeceba9e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/471260
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47670}
4 files changed
tree: 04a3e1eb42d9a2596a6ba3eb2a632a2aef10a31a
  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