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 Ã…hgren <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. .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. DIR_METADATA
  11. ENG_REVIEW_OWNERS
  12. LICENSE
  13. OWNERS
  14. PATENTS
  15. PRESUBMIT.py
  16. README.chromium
  17. README.md
  18. WATCHLISTS
  19. abseil-in-webrtc.md
  20. api/
  21. audio/
  22. build_overrides/
  23. call/
  24. codereview.settings
  25. common_audio/
  26. common_video/
  27. data/
  28. docs/
  29. examples/
  30. g3doc.lua
  31. g3doc/
  32. license_template.txt
  33. logging/
  34. media/
  35. modules/
  36. native-api.md
  37. net/
  38. p2p/
  39. pc/
  40. presubmit_test.py
  41. presubmit_test_mocks.py
  42. pylintrc
  43. resources/
  44. rtc_base/
  45. rtc_tools/
  46. sdk/
  47. stats/
  48. style-guide.md
  49. style-guide/
  50. system_wrappers/
  51. test/
  52. tools_webrtc/
  53. video/
  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