Revert "Reland "Add ability to set RTCP sender ssrc at construction time""

This reverts commit 6f420e424885dab1d9f885365ea9abea5cc4a901.

Reason for revert: Speculative revert (some perf test are failing)

Original change's description:
> Reland "Add ability to set RTCP sender ssrc at construction time"
>
> This is a reland of 94c58fd815f0c7c6429aa53a79621ea9ef39c770
>
> Patch set 1 is the original CL.
> Patch set 2 introduced a trivial fix. In RtcpSender::SetSSRC(), check
> if either current SSRC is 0 or if the SSRC is identical to the current
> one. If so, don't schedule an early report.
> This prevents a regression in which audio jitter became too low(?)
>
> Original change's description:
> > Add ability to set RTCP sender ssrc at construction time
> >
> > Bug: webrtc:10774
> > Change-Id: Iaf5857e24359e9795434bcd0cdbe1658a2f9f5d3
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144632
> > Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28506}
>
> Bug: webrtc:10774
> Change-Id: I103dfa48719aa41d6ab633cdac8b3a5c46b54843
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144565
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28520}

TBR=asapersson@webrtc.org,sprang@webrtc.org,ilnik@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:10774
Change-Id: I39238d942b2bbe0a9c8ca752387a35ed9dd70650
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145327
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28555}
6 files changed
tree: bb443b76eb565fb5b32e7a3f8d391b78e693144a
  1. .clang-format
  2. .git-blame-ignore-revs
  3. .gitignore
  4. .gn
  5. .vpython
  6. AUTHORS
  7. BUILD.gn
  8. CODE_OF_CONDUCT.md
  9. DEPS
  10. ENG_REVIEW_OWNERS
  11. LICENSE
  12. OWNERS
  13. PATENTS
  14. PRESUBMIT.py
  15. README.chromium
  16. README.md
  17. WATCHLISTS
  18. abseil-in-webrtc.md
  19. api/
  20. audio/
  21. build_overrides/
  22. call/
  23. codereview.settings
  24. common_audio/
  25. common_types.h
  26. common_video/
  27. crypto/
  28. data/
  29. examples/
  30. license_template.txt
  31. logging/
  32. media/
  33. modules/
  34. native-api.md
  35. p2p/
  36. pc/
  37. presubmit_test.py
  38. presubmit_test_mocks.py
  39. pylintrc
  40. resources/
  41. rtc_base/
  42. rtc_tools/
  43. sdk/
  44. stats/
  45. style-guide.md
  46. style-guide/
  47. system_wrappers/
  48. test/
  49. tools_webrtc/
  50. video/
  51. webrtc.gni
  52. 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 http://www.webrtc.org/native-code/development 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