Disable AudioOptions in Scream and L4S BWE integration tests

WebRtcVoiceEngine::ApplyOptions disables software noise suppression,
echo cancellation, and auto gain control on iOS (WEBRTC_IOS) while
leaving them enabled by default on other platforms. In PeerScenario BWE
integration tests using TestAudioDeviceModule::CreatePulsedNoiseCapturer,
this platform difference caused un-attenuated pulsed noise on iOS to
drive Opus and Audio Network Adaptor to higher bitrates, increasing
bottleneck queue delay in
ScreamTest.AudioOnlyBothDirectionsLinkCapacity40To90KbpsRtt200ms
(causing ICE STUN ping RTTs around 653 ms vs. the 600 ms limit) and
previously requiring CcFbSendRateAdaptation to be disabled on iOS.

Explicitly disable echo_cancellation, auto_gain_control,
noise_suppression, and highpass_filter in SendMedia and
CcFbSendRateAdaptation so all platforms run with identical audio
processing settings, relax the upper RTT bound in
AudioOnlyBothDirectionsLinkCapacity40To90KbpsRtt200ms from 600 ms to
700 ms, and re-enable CcFbSendRateAdaptation on iOS.

Bug: webrtc:42225697, webrtc:447037083
Change-Id: I36660d614cc67b17c9cc56ac610fb332f243c8d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/516140
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#49263}
2 files changed
tree: 0a65658c1bc0d8b0765d54632b2d4f1d7b067551
  1. .agents/
  2. agents/
  3. api/
  4. audio/
  5. build_overrides/
  6. call/
  7. common_audio/
  8. common_video/
  9. data/
  10. docs/
  11. examples/
  12. experiments/
  13. g3doc/
  14. infra/
  15. knowledge_base/
  16. logging/
  17. media/
  18. modules/
  19. net/
  20. p2p/
  21. pc/
  22. resources/
  23. rtc_base/
  24. rtc_tools/
  25. rust/
  26. sdk/
  27. stats/
  28. system_wrappers/
  29. test/
  30. tools_webrtc/
  31. video/
  32. .clang-format
  33. .clang-tidy
  34. .git-blame-ignore-revs
  35. .gitignore
  36. .gn
  37. .mailmap
  38. .rustfmt.toml
  39. .style.mdformat
  40. .style.yapf
  41. .yapfignore
  42. AUTHORS
  43. BUILD.gn
  44. CODE_OF_CONDUCT.md
  45. codereview.settings
  46. DEPS
  47. DIR_METADATA
  48. ENG_REVIEW_OWNERS
  49. GEMINI.md
  50. LICENSE
  51. license_template.txt
  52. native-api.md
  53. OWNERS
  54. OWNERS_INFRA
  55. PATENTS
  56. PRESUBMIT.py
  57. presubmit_test.py
  58. presubmit_test_mocks.py
  59. pylintrc
  60. pylintrc_old_style
  61. README.chromium
  62. README.md
  63. unsafe_buffers_paths.txt
  64. vpython.toml
  65. vpython.toml.uv.lock
  66. WATCHLISTS
  67. webrtc.gni
  68. webrtc_lib_link_test.cc
  69. 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