Revert "Moving iOS Audio Device to sdk."

This reverts commit a167212657f8450296ac518162ef0b28ba4214c5.

Reason for revert: Breaks Chromium build.
Log:
https://ci.chromium.org/buildbot/chromium.webrtc.fyi/ios-device/
Writing """\
additional_target_cpus = [ "arm64" ]
goma_dir = "/b/c/goma_client"
ios_enable_code_signing = false
is_component_build = false
is_debug = false
target_cpu = "arm"
target_os = "ios"
use_goma = true
""" to /b/c/b/ios_device/src/out/Release-iphoneos/args.gn.
/b/c/b/ios_device/src/buildtools/mac/gn gen //out/Release-iphoneos --check
  -> returned 1
ERROR at //third_party/webrtc/sdk/BUILD.gn:108:9: Can't load input file.
        "../../rtc_base:checks",
        ^----------------------
Unable to load:
  /b/c/b/ios_device/src/third_party/rtc_base/BUILD.gn
I also checked in the secondary tree for:
  /b/c/b/ios_device/src/build/secondary/third_party/rtc_base/BUILD.gn

Original change's description:
> Moving iOS Audio Device to sdk.
> 
> This change forks the existing iOS audio device module and audio device
> from modules/audio_device/ into sdk/objc/Framework. It also updates
> RTCPeerConnectionFactory to use the forked implementation.
> 
> The unit tests are re-implemented as XCTests.
> 
> Bug: webrtc:9120
> Change-Id: Ie60cafae796efbd7966d21ff6877c92cbe850fb7
> Reviewed-on: https://webrtc-review.googlesource.com/67300
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23163}

TBR=andersc@webrtc.org,kthelgason@webrtc.org,peterhanspers@webrtc.org

Change-Id: Iebe52e9775409a3bdd6d5e44f4f985d56b859cbe
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9120
Reviewed-on: https://webrtc-review.googlesource.com/75220
Reviewed-by: Max Morin <maxmorin@webrtc.org>
Commit-Queue: Max Morin <maxmorin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23166}
19 files changed
tree: 8856c12528750b8828562d667e13eaf5538aa0c9
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. examples/
  9. infra/
  10. logging/
  11. media/
  12. modules/
  13. ortc/
  14. p2p/
  15. pc/
  16. resources/
  17. rtc_base/
  18. rtc_tools/
  19. sdk/
  20. stats/
  21. style-guide/
  22. system_wrappers/
  23. test/
  24. tools_webrtc/
  25. video/
  26. .clang-format
  27. .git-blame-ignore-revs
  28. .gitignore
  29. .gn
  30. .vpython
  31. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. common_types.cc
  36. common_types.h
  37. DEPS
  38. LICENSE
  39. license_template.txt
  40. LICENSE_THIRD_PARTY
  41. native-api.md
  42. OWNERS
  43. PATENTS
  44. PRESUBMIT.py
  45. presubmit_test.py
  46. presubmit_test_mocks.py
  47. pylintrc
  48. README.chromium
  49. README.md
  50. style-guide.md
  51. typedefs.h
  52. WATCHLISTS
  53. webrtc.gni
  54. 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 http://www.webrtc.org/native-code/development 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