Refactor: Move ssrc, media_channel, SetFrameDecryptor, and SetFrameTransformer to RtpReceiverBase

- Add media_channel() pure virtual getter to RtpReceiverInternal
- Move ssrc() implementation and signaled_ssrc_ member to RtpReceiverBase
- Move SetFrameTransformer/frame_transformer_ to RtpReceiverBase
- Move SetFrameDecryptor/GetFrameDecryptor/frame_decryptor_ to RtpReceiverBase
- Add media_channel() override to AudioRtpReceiver and VideoRtpReceiver

Both concrete receivers had identical implementations of these methods.
Consolidating them in the base class reduces duplication and provides a
foundation for wiring SFrame decryption at the base class level.

Bug: webrtc:479862368
Change-Id: I8c2769006820e683d06491522b31ba321e975d16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/466421
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47550}
8 files changed
tree: 62057f45b32f52572b59877245aa5d62ac668795
  1. .agents/
  2. agents/
  3. api/
  4. audio/
  5. build_overrides/
  6. call/
  7. common_audio/
  8. common_video/
  9. data/
  10. docs/
  11. examples/
  12. experiments/
  13. g3doc/
  14. infra/
  15. logging/
  16. media/
  17. modules/
  18. net/
  19. p2p/
  20. pc/
  21. resources/
  22. rtc_base/
  23. rtc_tools/
  24. sdk/
  25. stats/
  26. system_wrappers/
  27. test/
  28. tools_webrtc/
  29. video/
  30. .clang-format
  31. .clang-tidy
  32. .git-blame-ignore-revs
  33. .gitignore
  34. .gn
  35. .mailmap
  36. .rustfmt.toml
  37. .style.mdformat
  38. .style.yapf
  39. .vpython3
  40. .yapfignore
  41. AUTHORS
  42. BUILD.gn
  43. CODE_OF_CONDUCT.md
  44. codereview.settings
  45. DEPS
  46. DIR_METADATA
  47. ENG_REVIEW_OWNERS
  48. GEMINI.md
  49. LICENSE
  50. license_template.txt
  51. native-api.md
  52. OWNERS
  53. OWNERS_INFRA
  54. PATENTS
  55. PRESUBMIT.py
  56. presubmit_test.py
  57. presubmit_test_mocks.py
  58. pylintrc
  59. pylintrc_old_style
  60. README.chromium
  61. README.md
  62. unsafe_buffers_paths.txt
  63. WATCHLISTS
  64. webrtc.gni
  65. webrtc_lib_link_test.cc
  66. 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