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webrtc / src / a60b9a4162bfb8f4f7134e096a5dccfdcc99dc48
commita60b9a4162bfb8f4f7134e096a5dccfdcc99dc48[log] [tgz]
authorHenrik Boström <hbos@webrtc.org>Wed Aug 27 09:33:01 2025
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Wed Aug 27 10:05:37 2025
tree5747a08408f0b87e5d1ebb83353a5ba23de36ebc
parent1c1c160ff95083a934792eb78e9cec5a0e659ea6 [diff]
Test inbound-rtp is created for the early media use case.

Once DTLS has been established, you can offer to receive media and have
packets flowing before returning to "stable". Because we can receive
packets early like this, we should also have "early" inbound-rtp stats
object creation.

Bug: chromium:406585888
Change-Id: I3bf7eff5099f4c070b77487a301a9ee6f003aadc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/406521
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45451}
1 file changed
tree: 5747a08408f0b87e5d1ebb83353a5ba23de36ebc
  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. .clang-tidy
  30. .git-blame-ignore-revs
  31. .gitignore
  32. .gn
  33. .mailmap
  34. .rustfmt.toml
  35. .style.yapf
  36. .vpython3
  37. AUTHORS
  38. BUILD.gn
  39. CODE_OF_CONDUCT.md
  40. codereview.settings
  41. DEPS
  42. DIR_METADATA
  43. ENG_REVIEW_OWNERS
  44. LICENSE
  45. license_template.txt
  46. native-api.md
  47. OWNERS
  48. OWNERS_INFRA
  49. PATENTS
  50. PRESUBMIT.py
  51. presubmit_test.py
  52. presubmit_test_mocks.py
  53. pylintrc
  54. pylintrc_old_style
  55. README.chromium
  56. README.md
  57. WATCHLISTS
  58. webrtc.gni
  59. webrtc_lib_link_test.cc
  60. 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