Revert "Store a raw_packetization bool in RtpSenderVideo rather than inferring"

This reverts commit fc1cbcf05221bf26b6fa94ecb018a5225060ddb4.

Reason for revert: Reverting while we figure out why the check is triggering downstream

Original change's description:
> Store a raw_packetization bool in RtpSenderVideo rather than inferring
>
> Ensure all calls to SendVideo() use a raw packetizer when they should
> rather than inferring it from the absence of |codec_type| - an issue
> when the caller doesn't know the correct packetization (eg
> RTPSenderVideoFrameTransformerDelegate).
>
> Bug: b/446768451
> Change-Id: Ib628d69ac1697de63cc293c5f4c681d6450f72d9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/411560
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Auto-Submit: Tony Herre <herre@google.com>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#45724}

Bug: b/446768451
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I456daecded27f9e52e126b8e5760238704f67300
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/411940
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45731}
4 files changed
tree: 91a2addcd84508e519734c9f57f366d9f132733d
  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. .clang-tidy
  30. .git-blame-ignore-revs
  31. .gitignore
  32. .gn
  33. .mailmap
  34. .rustfmt.toml
  35. .style.yapf
  36. .vpython3
  37. AUTHORS
  38. BUILD.gn
  39. CODE_OF_CONDUCT.md
  40. codereview.settings
  41. DEPS
  42. DIR_METADATA
  43. ENG_REVIEW_OWNERS
  44. LICENSE
  45. license_template.txt
  46. native-api.md
  47. OWNERS
  48. OWNERS_INFRA
  49. PATENTS
  50. PRESUBMIT.py
  51. presubmit_test.py
  52. presubmit_test_mocks.py
  53. pylintrc
  54. pylintrc_old_style
  55. README.chromium
  56. README.md
  57. WATCHLISTS
  58. webrtc.gni
  59. webrtc_lib_link_test.cc
  60. 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