Reland "Pass true stream resolutions to GetSimulcastConfig()"

This is a reland of commit 09f03be54804e81f626c26e8fde8c86cc952545f

Use max_num_layers instead of encoder_config.number_of_streams when calculation stream resolutions in EncoderStreamFactory::GetStreamResolutions().

Original change's description:
> Pass true stream resolutions to GetSimulcastConfig()
>
> Before this change GetSimulcastConfig() received only maximum resolution as an input parameter and derived resolutions for low quality simulcast streams assuming 1/2 scaling factor [1]. These days applications can configure resolution scaling factors via RtpEncodingParameters. If the configured resolution scaling factors were different from 1/2 then we got wrong bitrate limits from GetSimulcastConfig(). Now resolutions are calculated using scaling factor from RtpEncodingParameters (or default 1/2) for all streams in EncoderStreamFactory::CreateEncoderStreams() and then passed to GetSimulcastConfig().
>
> Moved tests from simulcast_unittest.cc to encoder_stream_factory_unittest.cc. Mapping of old to new tests:
> * GetConfigWithLimitedMaxLayersForResolution -> ReducesStreamCountWhenResolutionIsLow
> * GetConfigWithLowResolutionScreenshare -> ReducesLegacyScreencastStreamCountWhenResolutionIsLow
> * GetConfigWithNotLimitedMaxLayersForResolution -> KeepsStreamCountUnchangedWhenLegacyLimitIsDisabled
> * GetConfigWithNormalizedResolution -> AdjustsResolutionWhenUnaligned
> * GetConfigWithNormalizedResolutionDivisibleBy4 -> MakesResolutionDivisibleBy4
> * GetConfigWithNormalizedResolutionDivisibleBy8 -> not needed (MakesResolutionDivisibleBy4 should be enough).
> * GetConfigForLegacyLayerLimit -> KeepsStreamCountUnchangedWhenResolutionIsHigh and ReducesStreamCountWhenResolutionIsLow
> * GetConfigForLegacyLayerLimitWithRequiredHD -> KeepsStreamCountUnchangedWhenLegacyLimitIsDisabled
>
> [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/video/config/simulcast.cc;l=297-298;drc=1b78a7eb3f418460da03672b1d1af1d9488bb544
>
> Bug: webrtc:351644568, b/352504711
> Change-Id: I0028904ab0bb1e27b9c1b7cd3fb9a8ccf447fa35
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357280
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42651}

Bug: webrtc:351644568, b/352504711
Change-Id: Ib3fd859257b61c2a5d695b8b8f45c95495117c0e
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357520
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42654}
8 files changed
tree: 68532d5e56cd953834292e242f62a1d236782463
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. experiments/
  11. g3doc/
  12. infra/
  13. logging/
  14. media/
  15. modules/
  16. net/
  17. p2p/
  18. pc/
  19. resources/
  20. rtc_base/
  21. rtc_tools/
  22. sdk/
  23. stats/
  24. system_wrappers/
  25. test/
  26. tools_webrtc/
  27. video/
  28. .clang-format
  29. .git-blame-ignore-revs
  30. .gitignore
  31. .gn
  32. .mailmap
  33. .style.yapf
  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. LICENSE
  43. license_template.txt
  44. native-api.md
  45. OWNERS
  46. OWNERS_INFRA
  47. PATENTS
  48. PRESUBMIT.py
  49. presubmit_test.py
  50. presubmit_test_mocks.py
  51. pylintrc
  52. pylintrc_old_style
  53. README.chromium
  54. README.md
  55. WATCHLISTS
  56. webrtc.gni
  57. webrtc_lib_link_test.cc
  58. 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