Revert "Allow injecting media channel into VideoRtpReceiver"

This reverts commit f0edfc399074da2c0b6a3fe866462fa464addf5c.

Reason for revert: Breaks downstream project

Original change's description:
> Allow injecting media channel into VideoRtpReceiver
>
> This aligns VideoRtpReceiver and AudioRtpReceiver by allowing the
> media channel to be passed directly during construction. This
> dependency injection is necessary preparation for creating senders
> and receivers from within the RtpTransceiver class.
>
> Also create parity between the audio and video receiver
> constructors by removing the explicit `is_unified_plan` boolean from
> `AudioRtpReceiver`.
>
> Key changes:
> * Update `VideoRtpReceiver` to accept an optional
>   VideoMediaReceiveChannelInterface` in the constructor.
> * Refactor `AudioRtpReceiver` constructors to not have the
>   `is_unified_plan` argument in constructors that are only used with
>   unified plan. Leave one dedicated constructor for planb.
> * Update call sites in `RtpTransmissionManager` and unit tests.
>
> Bug: webrtc:42222804
> Change-Id: Ifa82f5ac4087b048f5e6d78284a4e79678c8e3b6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/433841
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#46499}

Bug: webrtc:42222804
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I1c52e87c05fec0e4cbab002d2d381d584e8349cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/434720
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#46505}
9 files changed
tree: 782fa84e9125d36128eb8df918bb2c070151a435
  1. agents/
  2. api/
  3. audio/
  4. build_overrides/
  5. call/
  6. common_audio/
  7. common_video/
  8. data/
  9. docs/
  10. examples/
  11. experiments/
  12. g3doc/
  13. infra/
  14. logging/
  15. media/
  16. modules/
  17. net/
  18. p2p/
  19. pc/
  20. resources/
  21. rtc_base/
  22. rtc_tools/
  23. sdk/
  24. stats/
  25. system_wrappers/
  26. test/
  27. tools_webrtc/
  28. video/
  29. .clang-format
  30. .clang-tidy
  31. .git-blame-ignore-revs
  32. .gitignore
  33. .gn
  34. .mailmap
  35. .rustfmt.toml
  36. .style.yapf
  37. .vpython3
  38. AUTHORS
  39. BUILD.gn
  40. CODE_OF_CONDUCT.md
  41. codereview.settings
  42. DEPS
  43. DIR_METADATA
  44. ENG_REVIEW_OWNERS
  45. GEMINI.md
  46. LICENSE
  47. license_template.txt
  48. native-api.md
  49. OWNERS
  50. OWNERS_INFRA
  51. PATENTS
  52. PRESUBMIT.py
  53. presubmit_test.py
  54. presubmit_test_mocks.py
  55. pylintrc
  56. pylintrc_old_style
  57. README.chromium
  58. README.md
  59. WATCHLISTS
  60. webrtc.gni
  61. webrtc_lib_link_test.cc
  62. 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