Revert "Add AV1 RtpDepacketizer class"

This reverts commit 49470c2ac460ed8cce250942e8525c5f14e32778.

Reason for revert: Seems to trigger linker error on iOS64. See:
https://ci.chromium.org/p/webrtc/builders/ci/iOS64%20Debug/17733

Original change's description:
> Add AV1 RtpDepacketizer class
> 
> Implement Parse function that extracts is_first_packet_in_frame,
> is_last_packet_in_frame, and frame_type fields.
> 
> Bug: webrtc:11042
> Change-Id: I9360ea52ef274281b5c5e4c31955100b92155bfe
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159180
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29814}

TBR=danilchap@webrtc.org,saza@webrtc.org,philipel@webrtc.org

Change-Id: I2eb5994d8e31e12d6cb6e9f792b691ed10d9df81
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11042
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159940
Reviewed-by: Yves Gerey <yvesg@google.com>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#29815}
6 files changed
tree: e5ce09139118b94efb68d33d3595e45dc732920e
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. logging/
  11. media/
  12. modules/
  13. p2p/
  14. pc/
  15. resources/
  16. rtc_base/
  17. rtc_tools/
  18. sdk/
  19. stats/
  20. style-guide/
  21. system_wrappers/
  22. test/
  23. tools_webrtc/
  24. video/
  25. .clang-format
  26. .git-blame-ignore-revs
  27. .gitignore
  28. .gn
  29. .vpython
  30. abseil-in-webrtc.md
  31. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. common_types.h
  36. DEPS
  37. ENG_REVIEW_OWNERS
  38. LICENSE
  39. license_template.txt
  40. native-api.md
  41. OWNERS
  42. PATENTS
  43. PRESUBMIT.py
  44. presubmit_test.py
  45. presubmit_test_mocks.py
  46. pylintrc
  47. README.chromium
  48. README.md
  49. style-guide.md
  50. WATCHLISTS
  51. webrtc.gni
  52. webrtc_lib_link_test.cc
  53. 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