Reland "rtpsender interface: make pure virtual again"

This reverts commit fbb7ce8a935db1988b3571639cab1eaed88980d1.

Reason for revert: Relanding because the upstream project should be compatible with the changes now.

Original change's description:
> Revert "rtpsender interface: make pure virtual again"
>
> This reverts commit 021512b76a872b04e803d61f46c740ed363d641b.
>
> Reason for revert: Breaks upstream project. It relies on the default implementation. The CL will be relanded after the migration is done. We will make sure to do it shortly.
>
> Original change's description:
> > rtpsender interface: make pure virtual again
> >
> > after providing default implementations in Chromium tests
> >
> > BUG=None
> >
> > Change-Id: I53bf26b3a99416f4005e7df75b9b86dfbf2489cb
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273100
> > Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#37941}
>
> Bug: None
> Change-Id: I40f27c36819365fadae32032521f7e11184bee62
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273484
> Owners-Override: Andrey Logvin <landrey@google.com>
> Commit-Queue: Andrey Logvin <landrey@google.com>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Auto-Submit: Andrey Logvin <landrey@google.com>
> Cr-Commit-Position: refs/heads/main@{#37947}

Bug: None
Change-Id: I531e17d5252d4bd5450d5ac5c64fc8f51b4a1d1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273701
Commit-Queue: Andrey Logvin <landrey@google.com>
Reviewed-by: Philipp Hancke <phancke@microsoft.com>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Andrey Logvin <landrey@google.com>
Cr-Commit-Position: refs/heads/main@{#37969}
5 files changed
tree: 39d05b34cfb5eaab7dee1742e233c47c327836a5
  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