Revert "red: remove special-casing of no-redundancy"

This reverts commit 320c57b7c6bf68d8612a4f135f44b3d29e802113.

Reason for revert:
Breaks CI tests: https://ci.chromium.org/ui/p/webrtc/builders/ci/Linux32%20Release/27236/overview
All CI tests: https://ci.chromium.org/p/webrtc/g/ci/console
Error is of the following type:
```
../../modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc:195: Failure
Expected equality of these values:
  1u
    Which is: 1
  encoded_info_.encoded_bytes
    Which is: 2
Stack trace:
  0x56b298d9: webrtc::AudioEncoderCopyRedTest_CheckPayloadSizesSingle_Test::TestBody()
  0x572fe317: testing::internal::HandleExceptionsInMethodIfSupported<>()
  0x572fe1d4: testing::Test::Run()
  0x572ff2ee: testing::TestInfo::Run()
```

Original change's description:
> red: remove special-casing of no-redundancy
>
> removes the special-casing of not sending a RED header when there is no redundant payload.
> This avoids switching back and forth between the primary and the red payload format (primarily at the start of the connection).
>
> BUG=webrtc:11640
>
> Change-Id: I8e0044bef1ed7c4168d9527645522392db2ed068
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220932
> Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34703}

TBR=henrik.lundin@webrtc.org,devicentepena@webrtc.org,minyue@webrtc.org,philipp.hancke@googlemail.com,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com

Change-Id: Ide409232720df32b24022f99228f3b6ae81f06fe
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11640
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228421
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34707}
2 files changed
tree: 40ac3cd6db644feb182e18e921851361871d2514
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. docs/
  9. examples/
  10. g3doc/
  11. logging/
  12. media/
  13. modules/
  14. net/
  15. p2p/
  16. pc/
  17. resources/
  18. rtc_base/
  19. rtc_tools/
  20. sdk/
  21. stats/
  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. DEPS
  36. DIR_METADATA
  37. ENG_REVIEW_OWNERS
  38. g3doc.lua
  39. LICENSE
  40. license_template.txt
  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. 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 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