Reland "Remove legacy RtpVideoStreamReceiver."

This reverts commit 768de78d10506ae192e5f12af48fce0c9b9e31ef.

Reason for revert: downstream cleaned up

Original change's description:
> Revert "Remove legacy RtpVideoStreamReceiver."
>
> This reverts commit da12e10aba4d12e7a6fb3882dc667901c9e17aa2.
>
> Reason for revert: Speculative revert due to failing downstream tests
>
> Original change's description:
> > Remove legacy RtpVideoStreamReceiver.
> >
> > Bug: none
> > Change-Id: I434a56980f4d6c68381abae973cd846c71441b08
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236846
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#37177}
>
> Bug: none
> Change-Id: I960c7693955c061ad95f921905d35c87733a3283
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265397
> Commit-Queue: Björn Terelius <terelius@google.com>
> Owners-Override: Björn Terelius <terelius@google.com>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Auto-Submit: Björn Terelius <terelius@google.com>
> Cr-Commit-Position: refs/heads/main@{#37183}

Bug: none
Change-Id: Ia4a84c6287d5a6a24bb98174617bec2e469e194b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265840
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Björn Terelius <terelius@google.com>
Cr-Commit-Position: refs/heads/main@{#37221}
4 files changed
tree: cdf9fd7e390a467ef2e150ccba51aead62b112f0
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. g3doc/
  11. infra/
  12. logging/
  13. media/
  14. modules/
  15. net/
  16. p2p/
  17. pc/
  18. resources/
  19. rtc_base/
  20. rtc_tools/
  21. sdk/
  22. stats/
  23. system_wrappers/
  24. test/
  25. tools_webrtc/
  26. video/
  27. .clang-format
  28. .git-blame-ignore-revs
  29. .gitignore
  30. .gn
  31. .mailmap
  32. .style.yapf
  33. .vpython
  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. g3doc.lua
  43. LICENSE
  44. license_template.txt
  45. native-api.md
  46. OWNERS
  47. PATENTS
  48. PRESUBMIT.py
  49. presubmit_test.py
  50. presubmit_test_mocks.py
  51. pylintrc
  52. README.chromium
  53. README.md
  54. WATCHLISTS
  55. webrtc.gni
  56. webrtc_lib_link_test.cc
  57. 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