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webrtc / src / / d8c4de71722c9de38f942932be21d4015f32a3bc
commitd8c4de71722c9de38f942932be21d4015f32a3bc[log] [tgz]
authorPer Kjellander <perkj@webrtc.org>Thu Nov 17 12:23:04 2022
committerWebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com>Mon Nov 21 12:41:39 2022
treef5e6622e7794fe19d2c75280bb4486e76e56717c
parentd742382eb0c576efd7ac2fac0e239177b31cc52b [diff]
Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream

VideoSendStreamImpl::Start and VideoSendStream::Start are not used by PeerConnections, only StartPerRtpStream.
Therefore this cl:
- Change implementation of VideoSendStream::Start to use VideoSendStream::StartPerRtpStream. VideoSendstream::Start is kept for convenience.
- Remove VideoSendStreamImpl::Start() since it was only used by tests that use call and is confusing.
- RtpVideoSender::SetActive is removed/changed to RtpVideoSender::Stop(). For normal operations RtpVideoSender::SetActiveModules is used.

Bug: none
Change-Id: I43b153250b07c02fe63c84e3c4cec18d4ec0d47a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283660
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38698}
15 files changed
tree: f5e6622e7794fe19d2c75280bb4486e76e56717c
  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. .vpython
  35. .vpython3
  36. AUTHORS
  37. BUILD.gn
  38. CODE_OF_CONDUCT.md
  39. codereview.settings
  40. DEPS
  41. DIR_METADATA
  42. ENG_REVIEW_OWNERS
  43. g3doc.lua
  44. LICENSE
  45. license_template.txt
  46. native-api.md
  47. OWNERS
  48. PATENTS
  49. PRESUBMIT.py
  50. presubmit_test.py
  51. presubmit_test_mocks.py
  52. pylintrc
  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