Reland "Reland "ObjC ADM: record/play implementation via RTCAudioDevice [3/3]""

This is a reland of commit 9a0a6a198e8e247884fe01d7e0aa6bd425721c14

Original change's description:
> Reland "ObjC ADM: record/play implementation via RTCAudioDevice [3/3]"
>
> This is a reland of commit 2b9aaad58f56744f5c573c3b918fe072566598a5
>
> Original change's description:
> > ObjC ADM: record/play implementation via RTCAudioDevice [3/3]
> >
> > # Overview
> > This CL chain exposes new API from ObjC WebRTC SDK to inject custom
> > means to play and record audio. The goal of CLs is achieved by having
> > additional implementation of `webrtc::AudioDeviceModule`
> > called `ObjCAudioDeviceModule`. The feature
> > of `ObjCAudioDeviceModule` is that it does not directly use any
> > of OS-provided audio APIs like AudioUnit, AVAudioEngine, AudioQueue,
> > AVCaptureSession etc. Instead it delegates communication with specific
> > system audio API to user-injectable audio device instance which
> > implements `RTCAudioDevice` protocol.
> > `RTCAudioDevice` is new API added to ObC WebRTC SDK in the CL chain.
> >
> > # AudioDeviceBuffer
> > `ObjCAudioDeviceModule` does conform to heavy `AudioDeviceModule`
> > interface providing stubs for unrelated methods. It also implements
> > common low-level management of audio device buffer, which glues audio
> > PCM flow to/from WebRTC.
> > `ObjCAudioDeviceModule` owns single `webrtc::AudioDeviceBuffer` which
> > with the help of two `FineAudioBuffer` (one for recording and one for
> > playout) is exchanged audio PCMs with user-provided `RTCAudioDevice`
> > instance.
> > `webrtc::AudioDeviceBuffer` is configured to work with specific audio:
> > it has to know sample rate and channels count of audio being played and
> > recorded. These formats could be different between playout and
> > recording. `ObjCAudioDeviceModule` stores current audio  parameters
> > applied  to `webrtc::AudioDeviceBuffer` as fields of
> > type `webrtc::AudioParameters`. `RTCAudioDevice` has it's own variable
> > audio parameters like sample rate, channels  count and IO buffer
> > duration. The audio parameters of `RTCAudioDevice` must be kept in sync
> > with audio parameters applied to `webrtc::AudioDeviceBuffer`, otherwise
> > audio playout and recording will be corrupted: audio is sent only
> > partially over the wire and/or audio is played with artifacts.
> > `ObjCAudioDeviceModule` reads current `RTCAudioDevice` audio parameters
> > when playout or recording is initialized. Whenever `RTCAudioDevice`
> > audio parameters parameters are changed, there must be a notification to
> > `ObjCAudioDeviceModule` to allow it to reconfigure
> > it's `webrtc::AudioDeviceBuffer`. The notification is performed
> > via `RTCAudioDeviceDelegate` object, which is provided
> > by `ObjCAudioDeviceModule` during initialization of `RTCAudioDevice`.
> >
> > # Threading
> > `ObjCAudioDeviceModule` is stick to same thread between initialization
> > and termination. The only exception is two IO functions invoked by SDK
> > user code presumably from real-time audio IO thread.
> > Implementation of `RTCAudioDevice` may rely on the fact that all the
> > methods of `RTCAudioDevice` are called on the same thread between
> > initialization and termination. `ObjCAudioDeviceModule` is also expect
> > that the implementation of `RTCAudioDevice` will call methods related
> > to notification of audio parameters changes and audio interruption are
> > invoked on `ObjCAudioDeviceModule` thread. To facilitate this
> > requirement `RTCAudioDeviceDelegate` provides two functions to execute
> > sync and async block on `ObjCAudioDeviceModule` thread.
> > Async block could be useful when handling audio session notifications to
> > dispatch whole block re-configuring audio objects used
> > by `RTCAudioDevice` implementation.
> > Sync block could be used to make sure changes to audio parameters
> > of ADB owned by `ObjCAudioDeviceModule` are notified, before interrupted
> > playout/recording restarted.
> >
> > Bug: webrtc:14193
> > Change-Id: I5587ec6bbee3cf02bad70dd59b822feb0ada7f86
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269006
> > Reviewed-by: Henrik Andreasson <henrika@google.com>
> > Commit-Queue: Yury Yarashevich <yura.yaroshevich@gmail.com>
> > Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
> > Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> > Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#37928}
>
> Bug: webrtc:14193
> Change-Id: Iaf950d24bb2394a20e50421d5122f72ce46ae840
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273380
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37946}

Bug: webrtc:14193
Change-Id: I84a6462c233daae7f662224513809b13e7218029
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273662
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37977}
5 files changed
tree: 15a463b6b3270021c4a826b72e0c20784e747bcf
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. g3doc/
  11. infra/
  12. logging/
  13. media/
  14. modules/
  15. net/
  16. p2p/
  17. pc/
  18. resources/
  19. rtc_base/
  20. rtc_tools/
  21. sdk/
  22. stats/
  23. system_wrappers/
  24. test/
  25. tools_webrtc/
  26. video/
  27. .clang-format
  28. .git-blame-ignore-revs
  29. .gitignore
  30. .gn
  31. .mailmap
  32. .style.yapf
  33. .vpython
  34. .vpython3
  35. AUTHORS
  36. BUILD.gn
  37. CODE_OF_CONDUCT.md
  38. codereview.settings
  39. DEPS
  40. DIR_METADATA
  41. ENG_REVIEW_OWNERS
  42. g3doc.lua
  43. LICENSE
  44. license_template.txt
  45. native-api.md
  46. OWNERS
  47. PATENTS
  48. PRESUBMIT.py
  49. presubmit_test.py
  50. presubmit_test_mocks.py
  51. pylintrc
  52. README.chromium
  53. README.md
  54. WATCHLISTS
  55. webrtc.gni
  56. webrtc_lib_link_test.cc
  57. 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