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webrtc / src / 271195f33686d6fc1aba51c0d43d76dc6839c4dd
commit271195f33686d6fc1aba51c0d43d76dc6839c4dd[log] [tgz]
authorDanil Chapovalov <danilchap@webrtc.org>Mon Feb 11 10:30:03 2019
committerCommit Bot <commit-bot@chromium.org>Mon Feb 11 13:21:55 2019
treef36be800e091a3ae1277180a62d36a48dfffe8f8
parent501bfba0cb5f9649956451dd0d14f56b62af9af4 [diff]
Fix potential crash when building rtx packet

rtx packet may have addition extension (mid) and may use different
header size for extension (e.g. if repaired rtp stream id registered
to larger id than rtp stream id)

As a result rtx packet size calculation as orginial size + 2 bytes in
some scenarious may be incorrect. This chenage avoids crash in that cases.

Bug: None
Change-Id: I620d95e0592d6bdac0d3623b2675a49fc2177580
Reviewed-on: https://webrtc-review.googlesource.com/c/122180
Reviewed-by: Erik Varga <erikvarga@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26635}
1 file changed
tree: f36be800e091a3ae1277180a62d36a48dfffe8f8
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. data/
  8. examples/
  9. logging/
  10. media/
  11. modules/
  12. p2p/
  13. pc/
  14. resources/
  15. rtc_base/
  16. rtc_tools/
  17. sdk/
  18. stats/
  19. style-guide/
  20. system_wrappers/
  21. test/
  22. tools_webrtc/
  23. video/
  24. .clang-format
  25. .git-blame-ignore-revs
  26. .gitignore
  27. .gn
  28. .vpython
  29. abseil-in-webrtc.md
  30. AUTHORS
  31. BUILD.gn
  32. CODE_OF_CONDUCT.md
  33. codereview.settings
  34. common_types.h
  35. DEPS
  36. ENG_REVIEW_OWNERS
  37. LICENSE
  38. license_template.txt
  39. native-api.md
  40. OWNERS
  41. PATENTS
  42. PRESUBMIT.py
  43. presubmit_test.py
  44. presubmit_test_mocks.py
  45. pylintrc
  46. README.chromium
  47. README.md
  48. style-guide.md
  49. WATCHLISTS
  50. webrtc.gni
  51. 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 http://www.webrtc.org/native-code/development 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