Truncate CSRC list in RtpPacket::SetCsrcs if too large

Truncates the CSRC list to a maximum of 15 entries as per the 4-bit RTP
header field limitation, and logs a warning when truncation occurs. Also
updates the buffer capacity check and allocation to happen prior to
writing data to avoid heap buffer overflow.

Additionally, ensure that the underlying buffer size is updated before
writing the CSRC data. By calling SetSize prior to WriteAt, we prevent
potential heap buffer overflows that could occur if the buffer was not
sufficiently allocated before the write operation.

Bug: chromium:486317116
Change-Id: I2821867f94c0a8d174fb25047ac1f4b94ee5b867
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/459943
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47266}
3 files changed
tree: 736244883b96bb19b976b0e9633c35243c799bc7
  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.yapf
  38. .vpython3
  39. AUTHORS
  40. BUILD.gn
  41. CODE_OF_CONDUCT.md
  42. codereview.settings
  43. DEPS
  44. DIR_METADATA
  45. ENG_REVIEW_OWNERS
  46. GEMINI.md
  47. LICENSE
  48. license_template.txt
  49. native-api.md
  50. OWNERS
  51. OWNERS_INFRA
  52. PATENTS
  53. PRESUBMIT.py
  54. presubmit_test.py
  55. presubmit_test_mocks.py
  56. pylintrc
  57. pylintrc_old_style
  58. README.chromium
  59. README.md
  60. unsafe_buffers_paths.txt
  61. WATCHLISTS
  62. webrtc.gni
  63. webrtc_lib_link_test.cc
  64. 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