Provide RtpState at construction when restoring an RTP module

When a suspended stream was recreated, its previous state was applied
with SetRtpState() and SetRtxState() right after construction. This
required state to be mutable that otherwise could be const, which in
turn required a lock for every read on the encoder queue.

* Add rtp_state and rtx_rtp_state to RtpRtcpInterface::Configuration.
* Apply it when constructing RTPSender and ModuleRtpRtcpImpl2.
* RtpVideoSender and AudioSendStream, via ChannelSend, now pass the
  suspended states at creation.
* Remove SetStartTimestamp(), SetRtpState() and SetRtxState() from
  RtpRtcpInterface, ModuleRtpRtcpImpl2, as well as
  RTPSender::SetRtxRtpState().
* Deprecate RTPSender::SetTimestampOffset() and SetRtpState(). Once
  direct users of RTPSender have migrated, they can be removed and the
  timestamp offset made const.

Bug: webrtc:42223727
Change-Id: Idad26fe1205c1288a3455865dfb79027ed7f83da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/507320
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#48831}
17 files changed
tree: ff8eb710fdd54219d3b5f2bd224a9de8c36c1a65
  1. .agents/
  2. agents/
  3. api/
  4. audio/
  5. build_overrides/
  6. call/
  7. common_audio/
  8. common_video/
  9. data/
  10. docs/
  11. examples/
  12. experiments/
  13. g3doc/
  14. infra/
  15. logging/
  16. media/
  17. modules/
  18. net/
  19. p2p/
  20. pc/
  21. resources/
  22. rtc_base/
  23. rtc_tools/
  24. rust/
  25. sdk/
  26. stats/
  27. system_wrappers/
  28. test/
  29. tools_webrtc/
  30. video/
  31. .clang-format
  32. .clang-tidy
  33. .git-blame-ignore-revs
  34. .gitignore
  35. .gn
  36. .mailmap
  37. .rustfmt.toml
  38. .style.mdformat
  39. .style.yapf
  40. .yapfignore
  41. AUTHORS
  42. BUILD.gn
  43. CODE_OF_CONDUCT.md
  44. codereview.settings
  45. DEPS
  46. DIR_METADATA
  47. ENG_REVIEW_OWNERS
  48. GEMINI.md
  49. LICENSE
  50. license_template.txt
  51. native-api.md
  52. OWNERS
  53. OWNERS_INFRA
  54. PATENTS
  55. PRESUBMIT.py
  56. presubmit_test.py
  57. presubmit_test_mocks.py
  58. pylintrc
  59. pylintrc_old_style
  60. README.chromium
  61. README.md
  62. unsafe_buffers_paths.txt
  63. vpython.toml
  64. vpython.toml.uv.lock
  65. WATCHLISTS
  66. webrtc.gni
  67. webrtc_lib_link_test.cc
  68. 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