Revert "Call: Deduplicate SentPacket notifications"

This reverts commit 61a8d9caaa31ab4ef953415882f97be5a4248774.

Reason for revert: We have identified some downstream regressions caused by this change (https://crbug.com/webrtc/13437).

Original change's description:
> Call: Deduplicate SentPacket notifications
>
> When bundling is in effect, multiple senders may be sharing the same
> transport. It means every |sent_packet| will be multiply notified from
> different channels, WebRtcVoiceMediaChannel or WebRtcVideoChannel.
> Record |last_sent_packet_| to deduplicate redundant notifications to
> downstream objects.
>
> This CL reduces 50% PostTask/Wakeup of Dynamic Mode Pacer.
>
> [1] https://datatracker.ietf.org/doc/html/rfc8829#section-4.1.1
> [2] https://datatracker.ietf.org/doc/html/rfc8843
>
> Bug: webrtc:13417
> Change-Id: Ib121d5af07abe208bd7d36715a234f48cdabb032
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238720
> Reviewed-by: Markus Handell <handellm@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Tommi <tommi@webrtc.org>
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35417}

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

Bug: webrtc:13417
Change-Id: Ib1230fa07db56c33941a5b529a28f83d6d08d74d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239441
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Owners-Override: Jakob Ivarsson <jakobi@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35442}
1 file changed
tree: d3a6689b96a2ea91c3b0b9bf07247d4e5d1bdc7e
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. g3doc/
  11. logging/
  12. media/
  13. modules/
  14. net/
  15. p2p/
  16. pc/
  17. resources/
  18. rtc_base/
  19. rtc_tools/
  20. sdk/
  21. stats/
  22. system_wrappers/
  23. test/
  24. tools_webrtc/
  25. video/
  26. .clang-format
  27. .git-blame-ignore-revs
  28. .gitignore
  29. .gn
  30. .vpython
  31. .vpython3
  32. AUTHORS
  33. BUILD.gn
  34. CODE_OF_CONDUCT.md
  35. codereview.settings
  36. DEPS
  37. DIR_METADATA
  38. ENG_REVIEW_OWNERS
  39. g3doc.lua
  40. LICENSE
  41. license_template.txt
  42. native-api.md
  43. OWNERS
  44. PATENTS
  45. PRESUBMIT.py
  46. presubmit_test.py
  47. presubmit_test_mocks.py
  48. pylintrc
  49. README.chromium
  50. README.md
  51. WATCHLISTS
  52. webrtc.gni
  53. webrtc_lib_link_test.cc
  54. 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