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webrtc / src.git / 36928454fab7078ae1d867b5abda1bb1e2e7685b
commit36928454fab7078ae1d867b5abda1bb1e2e7685b[log] [tgz]
authorphilipel <philipel@webrtc.org>Mon Nov 07 09:42:36 2016
committerphilipel <philipel@webrtc.org>Mon Nov 07 09:42:43 2016
tree5559ced94c3b7bfa942a77a4ada9d5a7e54e7730
parent18ee17d1e7876b95fa48a82360b20236624350b6 [diff]
Allocate extra buffer space in FrameObject in case of H264.

Since ffmpeg use an optimized bitstream reader that reads in chunks of 32/64
bits the bitstream buffer has to be increased in order for the reader to not
read out of bounds.

BUG=webrtc:5514
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2476513004 .

Cr-Commit-Position: refs/heads/master@{#14941}
2 files changed
tree: 5559ced94c3b7bfa942a77a4ada9d5a7e54e7730
  1. build_overrides/
  2. chromium/
  3. data/
  4. infra/
  5. resources/
  6. third_party/
  7. tools/
  8. webrtc/
  9. .clang-format
  10. .git-blame-ignore-revs
  11. .gitignore
  12. .gn
  13. all.gyp
  14. AUTHORS
  15. BUILD.gn
  16. check_root_dir.py
  17. codereview.settings
  18. DEPS
  19. LICENSE
  20. license_template.txt
  21. LICENSE_THIRD_PARTY
  22. OWNERS
  23. PATENTS
  24. PRESUBMIT.py
  25. pylintrc
  26. README.md
  27. setup_links.py
  28. sync_chromium.py
  29. WATCHLISTS
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. This page is maintained by the Google Chrome team.

Development

See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

More info