SdpOfferAnswerHandler: Significantly reduce audio impairment.

It was found from Chrome tracing that worker packet progression in
https://webrtc.github.io/samples/src/content/peerconnection/negotiate-timing/
during renegotiation of 100 transceivers is hindered by a multi-hundred
millisecond Invoke from the signaling to the worker thread. This
causes audio impairment.

Fix this by splitting the single Invoke into a series of Invokes,
allowing packets received during the renegotiation to be processed
between the worker invocations.

Experimental data of negotiation from 1 to 100 video transceivers

WebRtcDistinctWorkerThread OFF, before change:
4415.60 milliseconds, audio impairment 29760
4216.00 milliseconds, audio impairment 25560
4298.40 milliseconds, audio impairment 25440

WebRtcDistinctWorkerThread OFF, after change:
4258.70 milliseconds, audio impairment 26280
4255.50 milliseconds, audio impairment 25920
4363.10 milliseconds, audio impairment 25200

WebRtcDistinctWorkerThread ON, before change:
4407.80 milliseconds, audio impairment 24840
4541.00 milliseconds, audio impairment 26160
4377.80 milliseconds, audio impairment 17040

WebRtcDistinctWorkerThread ON, after change:
4364.80 milliseconds, audio impairment 0
4174.30 milliseconds, audio impairment 0
4309.00 milliseconds, audio impairment 0

We should reconsider this split after lazy decoders and decoder stream
projects have shipped, see
- bugs.webrtc.org/12462
- crbug.com/1157227
- crbug.com/1187289

Bug: webrtc:12840, webrtc:12462, chromium:1157227, chromium:1187289
Change-Id: I8e3b3943bd76f09da74b457690799415335b51f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221103
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34202}
1 file changed
tree: 7e959ab691b621987256e8eb96b9736c472ab859
  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. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. DEPS
  36. DIR_METADATA
  37. ENG_REVIEW_OWNERS
  38. g3doc.lua
  39. LICENSE
  40. license_template.txt
  41. native-api.md
  42. OWNERS
  43. PATENTS
  44. PRESUBMIT.py
  45. presubmit_test.py
  46. presubmit_test_mocks.py
  47. pylintrc
  48. README.chromium
  49. README.md
  50. WATCHLISTS
  51. webrtc.gni
  52. webrtc_lib_link_test.cc
  53. 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