[Overuse] Encoding pipeline as input signals in the abstract interface.

This defines the following methods:
- OnFrame(), replaces SetLastFramePixelCount().
- OnFrameDroppedDueToSize(), a rename of FrameDroppedDueToSize() to
  match the other methods.
- OnEncodeStarted(), a rename of the incorrectly named FrameCaptured().
- OnEncodeCompleted(), a rename of the poorly named FrameSent().

In order to get rid of SetLastFramePixelCount(), the "we don't know the
frame size" use case - which was previously implicitly avoided by
invoking SetLastFramePixelCount() with a made-up value for
last_frame_info_ - is now avoided using ".value_or()" in
LastInputFrameSizeOrDefault(). This does mean that a constant 144p
resolution value is referenced in two places, but the fact that this is
a magic value is at least made explicit. This may help future

Bug: webrtc:11222
Change-Id: I3b28daa8c5ecf57c6537957d4759f15e24bb2234
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166961
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30352}
6 files changed
tree: a3c0c98a1ba6ad719f6c592a359e75b636249f2e
  1. .clang-format
  2. .git-blame-ignore-revs
  3. .gitignore
  4. .gn
  5. .vpython
  7. BUILD.gn
  9. DEPS
  12. OWNERS
  14. PRESUBMIT.py
  15. README.chromium
  16. README.md
  18. abseil-in-webrtc.md
  19. api/
  20. audio/
  21. build_overrides/
  22. call/
  23. codereview.settings
  24. common_audio/
  25. common_types.h
  26. common_video/
  27. data/
  28. docs/
  29. examples/
  30. license_template.txt
  31. logging/
  32. media/
  33. modules/
  34. native-api.md
  35. p2p/
  36. pc/
  37. presubmit_test.py
  38. presubmit_test_mocks.py
  39. pylintrc
  40. resources/
  41. rtc_base/
  42. rtc_tools/
  43. sdk/
  44. stats/
  45. style-guide.md
  46. style-guide/
  47. system_wrappers/
  48. test/
  49. tools_webrtc/
  50. video/
  51. webrtc.gni
  52. webrtc_lib_link_test.cc
  53. whitespace.txt

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.


See http://www.webrtc.org/native-code/development 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