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webrtc / src / / 9f68535e68bbb4505fe2897952202ff20f682b9d
commit9f68535e68bbb4505fe2897952202ff20f682b9d[log] [tgz]
authorHenrik Boström <hbos@webrtc.org>Fri Jan 17 11:06:53 2025
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Fri Jan 17 17:17:11 2025
tree1823df138e3de6e528236e6172126c3940ee7b26
parentb0038dd14a1cc637ffbc48e36e55bd96452eef5d [diff]
Fix setParameters() throwing when level-id does not match.

In order to align with this PR[1], setParameters() should not throw if
the H265 level ID we're trying to send does not match what was
negotiated. This was believed to be fixed by [2] but we were still
throwing due to a check on a different layer (media_engine.cc).

In order to reproduce the issue despite WebRTC lacking SW
encoder/decoder for H265, peer_connection_encodings_integrationtest.cc
gets a new test with real stack but fake encoder/decoder factory. This
allows negotiating H265 and doing SetParameters() even though the codec
is not processing any frames.
- Basic test coverage is added for singlecast and simulcast H265.
- Test coverage for the bug being fixed added.
- In Chrome the equivalent WPTs exists for when real HW is available
  here[3]. Those tests PASS with this CL (currently FAIL).

[1] https://github.com/w3c/webrtc-pc/pull/3023
[2] https://webrtc-review.googlesource.com/c/src/+/368781
[3] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/web_tests/external/wpt/webrtc/protocol/h265-level-id.https.html

Bug: chromium:381407888
Change-Id: I3619a124586b8b26d3695cfad8890cf40bd475db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374164
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#43759}
5 files changed
tree: 1823df138e3de6e528236e6172126c3940ee7b26
  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