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webrtc / src / cd83ae2d6555ab19458820c04c5ce3a3714b1233
commitcd83ae2d6555ab19458820c04c5ce3a3714b1233[log] [tgz]
authorSteve Anton <steveanton@webrtc.org>Thu Apr 08 18:26:04 2021
committerCommit Bot <commit-bot@chromium.org>Tue Apr 13 17:18:47 2021
tree0e4a4858d843e1a6c89c6406cf718e9118cadf8c
parent32347b50bade365c42c676925374a3ee8d26f6ac [diff]
Speed up FrameCombiner::Combine by 3x

There were a couple operations in the mixer which touched
AudioFrame data() and mutable_data() getters in a hot loop. These
getters have a if (muted) conditional in them which led to
inefficient code generation and execution.

Profiled using Google Meet with 6 audio-only speaking participants.
Meet uses 3 audio receive streams.

Before: https://pprof.corp.google.com/user-profile?id=02526c98ca1f60ba7b340b2f5dabb72a&tab=flame&path=18l9q740udb80g1iq9r1c1gv6b9k1cuuq200eztpq0054kuq0
After: https://pprof.corp.google.com/user-profile?id=32a33e5c90c650e013bdf5008d9b5fd3&tab=flame&path=18l9q740udb80g1iq9r1c1gv6b9k1cuuq200eztpq0054kuq0

(Zoomed in on the audio render thread.)

Bug: webrtc:12662
Change-Id: If6ecb5de02095b8b0e4938f1a1817b55d388e01a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214560
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33712}
2 files changed
tree: 0e4a4858d843e1a6c89c6406cf718e9118cadf8c
  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. style-guide/
  23. system_wrappers/
  24. test/
  25. tools_webrtc/
  26. video/
  27. .clang-format
  28. .git-blame-ignore-revs
  29. .gitignore
  30. .gn
  31. .vpython
  32. abseil-in-webrtc.md
  33. AUTHORS
  34. BUILD.gn
  35. CODE_OF_CONDUCT.md
  36. codereview.settings
  37. DEPS
  38. DIR_METADATA
  39. ENG_REVIEW_OWNERS
  40. g3doc.lua
  41. LICENSE
  42. license_template.txt
  43. native-api.md
  44. OWNERS
  45. PATENTS
  46. PRESUBMIT.py
  47. presubmit_test.py
  48. presubmit_test_mocks.py
  49. pylintrc
  50. README.chromium
  51. README.md
  52. style-guide.md
  53. WATCHLISTS
  54. webrtc.gni
  55. webrtc_lib_link_test.cc
  56. 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