Revert "frame transformer: make GetPayloadType pure virtual again"

This reverts commit 209ac5fd95594ab3834dad3e3dbd14c8196637bc.

Reason for revert: Breaks WebRTC autoroll presubmit:
https://chromium-review.googlesource.com/c/chromium/src/+/3134502
Example failure https://ci.chromium.org/ui/p/chromium/builders/try/mac-rel/775468/overview

../../buildtools/third_party/libc++/trunk/include/__memory/unique_ptr.h:725:32: error: allocating an object of abstract class type 'testing::NiceMock<blink::(anonymous namespace)::MockTransformableVideoFrame>'
    return unique_ptr<_Tp>(new _Tp(_VSTD::forward<_Args>(__args)...));
                               ^
../../third_party/blink/renderer/platform/peerconnection/rtc_encoded_video_stream_transformer_test.cc:69:26: note: in instantiation of function template specialization 'std::make_unique<testing::NiceMock<blink::(anonymous namespace)::MockTransformableVideoFrame>>' requested here
  auto mock_frame = std::make_unique<NiceMock<MockTransformableVideoFrame>>();
                         ^
../../third_party/webrtc/api/frame_transformer_interface.h:36:19: note: unimplemented pure virtual method 'GetPayloadType' in 'NiceMock'
  virtual uint8_t GetPayloadType() const = 0;
                  ^


Original change's description:
> frame transformer: make GetPayloadType pure virtual again
>
> after chrome was updated in
>   https://chromium-review.googlesource.com/c/chromium/src/+/3103323
>
> BUG=webrtc:13077
>
> Change-Id: I7e5ff6aaae81c5dcfbaa41b09ef01bc95bb7251a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230143
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> Cr-Commit-Position: refs/heads/main@{#34877}

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:13077
Change-Id: I6b2e4e2804890c857f1f832a6a4faa614ec026c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230920
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34891}
1 file changed
tree: 66ba286f43611ed6a6ff093f73bfb7c864917b4c
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. g3doc/
  11. logging/
  12. media/
  13. modules/
  14. net/
  15. p2p/
  16. pc/
  17. resources/
  18. rtc_base/
  19. rtc_tools/
  20. sdk/
  21. stats/
  22. system_wrappers/
  23. test/
  24. tools_webrtc/
  25. video/
  26. .clang-format
  27. .git-blame-ignore-revs
  28. .gitignore
  29. .gn
  30. .vpython
  31. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. DEPS
  36. DIR_METADATA
  37. ENG_REVIEW_OWNERS
  38. g3doc.lua
  39. LICENSE
  40. license_template.txt
  41. native-api.md
  42. OWNERS
  43. PATENTS
  44. PRESUBMIT.py
  45. presubmit_test.py
  46. presubmit_test_mocks.py
  47. pylintrc
  48. README.chromium
  49. README.md
  50. WATCHLISTS
  51. webrtc.gni
  52. webrtc_lib_link_test.cc
  53. 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