Split RTPSender packet building into helpers

The code that attaches the MID and RID header extensions was duplicated
in AllocatePacket(), BuildRtxPacket() and both padding paths of
GeneratePadding(), and GeneratePadding() had grown long and hard to
read.

- Move the MID and RID logic into AttachMidAndRid().
- Build media and padding packets in BuildMediaPacket() and
  BuildPaddingPacket(), and track the maximum CSRC count in
  UpdateMaxNumCsrcs().
- Split GeneratePadding() with AppendPayloadPadding() and
  PaddingBytesInPacket(). Checks that cannot change inside the padding
  loop now return before the loop instead of breaking out of it.

The helpers still run under send_mutex_. This prepares
https://webrtc-review.googlesource.com/c/src/+/507280, which builds
packets after releasing the lock and then only needs to change which
state the helpers read. BuildMediaPacket() and UpdateMaxNumCsrcs() will
also be used by a per-frame variant of AllocatePacket().

No change in behavior.

Bug: webrtc:42223727
Change-Id: I22b7682a65fb7c56d1c92b8cd07af3ef998b0b44
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/507642
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#48809}
2 files changed
tree: 07ecef2c3ad5e789c9801c60631d260119faed8c
  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. rust/
  25. sdk/
  26. stats/
  27. system_wrappers/
  28. test/
  29. tools_webrtc/
  30. video/
  31. .clang-format
  32. .clang-tidy
  33. .git-blame-ignore-revs
  34. .gitignore
  35. .gn
  36. .mailmap
  37. .rustfmt.toml
  38. .style.mdformat
  39. .style.yapf
  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. vpython.toml
  64. vpython.toml.uv.lock
  65. WATCHLISTS
  66. webrtc.gni
  67. webrtc_lib_link_test.cc
  68. 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