Add AudioMultiVector::ReadInterleavedFromIndex()

The new method differs from the existing method with the same name in
two respects:

* It reads into an InterleavedView<> instead of pointer.
* It either reads exactly the requested number of samples or fails

After examining the behavior in NetEq, where this method is called from,
it seems that what we need is a method that reads exactly the required
number of samples, not partially.

A follow-up change to NetEqImpl and SyncBuffer will change
SyncBuffer::GetNextAudioInterleaved() to read the requested buffer size
or fail, as well as update NetEqImpl::GetAudioInternal accordingly.

Bug: chromium:335805780
Change-Id: Ia78f888d52c9acb4b53b7042b31b8b0c53199e7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392143
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#44787}
6 files changed
tree: 4a876d28679b3b8c92e783ad0e2bf4d7b70d918c
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. experiments/
  11. g3doc/
  12. infra/
  13. logging/
  14. media/
  15. modules/
  16. net/
  17. p2p/
  18. pc/
  19. resources/
  20. rtc_base/
  21. rtc_tools/
  22. sdk/
  23. stats/
  24. system_wrappers/
  25. test/
  26. tools_webrtc/
  27. video/
  28. .clang-format
  29. .git-blame-ignore-revs
  30. .gitignore
  31. .gn
  32. .mailmap
  33. .rustfmt.toml
  34. .style.yapf
  35. .vpython3
  36. AUTHORS
  37. BUILD.gn
  38. CODE_OF_CONDUCT.md
  39. codereview.settings
  40. DEPS
  41. DIR_METADATA
  42. ENG_REVIEW_OWNERS
  43. LICENSE
  44. license_template.txt
  45. native-api.md
  46. OWNERS
  47. OWNERS_INFRA
  48. PATENTS
  49. PRESUBMIT.py
  50. presubmit_test.py
  51. presubmit_test_mocks.py
  52. pylintrc
  53. pylintrc_old_style
  54. README.chromium
  55. README.md
  56. WATCHLISTS
  57. webrtc.gni
  58. webrtc_lib_link_test.cc
  59. 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