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webrtc / src / f237dc146debcfde3d70038c2b66f71bfea8d24b
commitf237dc146debcfde3d70038c2b66f71bfea8d24b[log] [tgz]
authorJohannes Kron <kron@webrtc.org>Thu Aug 15 09:49:54 2024
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Thu Aug 22 14:15:40 2024
tree629318e24640291581faa0f2fae3d4447b08dfd4
parente7686023a186ac233ed1284da45cc166c0df4e1a [diff]
[M128] Ensure calls to QP convergence controller are on the same sequence

The original CL overlooked the possibility that the encoder may be
reconfigured in the middle of a stream.

Restructure the code so that all calls to QP convergence controller
happen on the encoder queue.

A side effect of this CL is that `EncodedImage::SetAtTargetQuality()`
is never called. The information is supplied to the frame cadence
adapter directly without this intermediate step.

`EncodedImage::SetAtTargetQuality()` and
`EncodedImage::IsAtTargetQuality()` are being marked as deprecated
in https://webrtc-review.googlesource.com/c/src/+/359660.

(cherry picked from commit b47cd6fbe315690756f2f03e7658d4e26fe27b1e)

Bug: chromium:359410061
Change-Id: I941b5f60b1a9fd7694dbedf2f3e4ff5253ccf357
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359640
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Original-Commit-Position: refs/heads/main@{#42788}
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360240
Cr-Commit-Position: refs/branch-heads/6613@{#3}
Cr-Branched-From: 1ac162ee20a214bf97f6594a7effcbbc21f1effb-refs/heads/main@{#42664}
4 files changed
tree: 629318e24640291581faa0f2fae3d4447b08dfd4
  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