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webrtc / src / 860a13c6fd6569144d3cb8e99bb9924a4417a003
commit860a13c6fd6569144d3cb8e99bb9924a4417a003[log] [tgz]
authorHenrik Boström <hbos@webrtc.org>Mon Jan 20 13:31:12 2025
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Tue Jan 21 07:55:17 2025
tree902dbcbd2415dd68fa11db0fadd3fa14f3d4293d
parentee7371f1f863bd7d81f58c4ea43f4a73676ec1aa [diff]
Misc improvements to RtpTransceiver unit tests and test utils.

In order to reduce the size and scope of a follow-up CL, this CL makes
some cleaning up and improvements to existing tests and adds some minor
test utility methods that will be used in the follow-up.

No change in behavior, this CL...
- Makes use of NiceMock in RtpTransceiver tests to avoid wall of text
  spam for various "uninteresting" method calls in all tests in this
  file.
- Refactors creating senders, receivers and transceivers to allow the
  follow-up CL to create such objects for kind "video" as well.
- Exposes cricket::FakeVideoEngine* to RtpTranscieverTest and allows
  adding unidirectional video codecs in the fake engine, to be used by
  the follow-up CL's tests.
- Allows creating fake video engine codecs from SdpVideoFormat in the
  fake decoder factory (already possible in the fake encoder factory).

Bug: chromium:381407888
Change-Id: Ie07eff79d832dd21800b95fd584891ebf4520798
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374900
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43776}
5 files changed
tree: 902dbcbd2415dd68fa11db0fadd3fa14f3d4293d
  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