Remove deprecated SetRates/SetRateAllocation from VideoEncoder.

This CL removes two deprecated methods from the VideoEncoder interface:
* int32_t SetRates(uint32_t, uint32_t);
* int32_t SetRateAllocation(const VideoBitrateAllocation&, uint32_t);

These are no longer used, instead the new version must be implemented:
  void SetRates(const RateControlParameters&) = 0;

Migrating is straight forward. For the old SetRates, simple replace:
  int32_t MyEncoder::SetRates(uint32_t bitrate, uint32_t framerate) {
with
  void MyEncoder::SetRates(const RateControlParameters& parameters) {
    uint32_t bitrate = parameters.bitrate.get_sum_kbps();
    uint32_t framerate =
      static_cast<uint32_t>(parameters.framerate_fps + 0.5);

For SetRateAllocation, replace:
  int32_t MyEncoder::SetRateAllocation(
      const VideoBitrateAllocation& allocation,
      uint32_t framerate) {
with
  void MyEncoder::SetRates(const RateControlParameters& parameters) {
    const VideoBitrateAllocation& allocation = parameters.bitrate;
    uint32_t framerate =
      static_cast<uint32_t>(parameters.framerate_fps + 0.5);

Two more things to note:
1. The new method is void. Previously the only use of the return value
   in production code was to log a more or less generic error message.
   Instead, log the actual error from the encoder when it happens,
   then just return.

2. The new method is pure virtual; it must be overriden even in test.

This CL is intended to be landed two weeks after creation, on Thursday
May 9th 2019.

Bug: webrtc:10481
Change-Id: I61349571a280bd40cd100ca9f93c4aa7748ed30d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134214
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27926}
3 files changed
tree: 10d172806c3f1a0c2905daf75f34d36dbf978077
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. crypto/
  8. data/
  9. examples/
  10. logging/
  11. media/
  12. modules/
  13. p2p/
  14. pc/
  15. resources/
  16. rtc_base/
  17. rtc_tools/
  18. sdk/
  19. stats/
  20. style-guide/
  21. system_wrappers/
  22. test/
  23. tools_webrtc/
  24. video/
  25. .clang-format
  26. .git-blame-ignore-revs
  27. .gitignore
  28. .gn
  29. .vpython
  30. abseil-in-webrtc.md
  31. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. common_types.h
  36. DEPS
  37. ENG_REVIEW_OWNERS
  38. LICENSE
  39. license_template.txt
  40. native-api.md
  41. OWNERS
  42. PATENTS
  43. PRESUBMIT.py
  44. presubmit_test.py
  45. presubmit_test_mocks.py
  46. pylintrc
  47. README.chromium
  48. README.md
  49. style-guide.md
  50. WATCHLISTS
  51. webrtc.gni
  52. 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 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