Fuzzing support for RTPDump VP8 and VP9 Streams.

This change integrates fuzzing support for RtpDumps in WebRTC. This allows
LibFuzzer to directly fuzz the RTP code path from packet arrival all the way
to actual decoding and rendering. It does this by replaying each RTP packet
in the RTPDump which can be mutated directly by the fuzzer.

For fuzzing support the RtpFileReader needs to support reading from a
buffer instead of an file. The test class requires FILE* for all its
parsing operations and is deeply coupled this way. I chose to solve this
problem at an OS level by using the tmpfile() option and copying the buffer
to the tmpfile(). fmemopen() is no available on most platforms so couldn't
be used as a generic solution. The additional copy isn't ideal but won't
be a bottleneck for the fuzzing.

In the future I plan for the fuzzers to read from a configuration file. But
given the current packaging strategy for fuzzers in WebRTC this isn't easy.

Bug: webrtc:9860
Change-Id: I2560120e82663f9e9fb5b9640e6a6d16f9c1a360
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126682
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27151}
10 files changed
tree: 115c09d51c2464ec0b2dfc383780ab5c901f7db6
  1. api/
  2. audio/
  3. build_overrides/
  4. call/
  5. common_audio/
  6. common_video/
  7. crypto/
  8. data/
  9. examples/
  10. logging/
  11. media/
  12. modules/
  13. p2p/
  14. pc/
  15. resources/
  16. rtc_base/
  17. rtc_tools/
  18. sdk/
  19. stats/
  20. style-guide/
  21. system_wrappers/
  22. test/
  23. tools_webrtc/
  24. video/
  25. .clang-format
  26. .git-blame-ignore-revs
  27. .gitignore
  28. .gn
  29. .vpython
  30. abseil-in-webrtc.md
  31. AUTHORS
  32. BUILD.gn
  33. CODE_OF_CONDUCT.md
  34. codereview.settings
  35. common_types.h
  36. DEPS
  37. ENG_REVIEW_OWNERS
  38. LICENSE
  39. license_template.txt
  40. native-api.md
  41. OWNERS
  42. PATENTS
  43. PRESUBMIT.py
  44. presubmit_test.py
  45. presubmit_test_mocks.py
  46. pylintrc
  47. README.chromium
  48. README.md
  49. style-guide.md
  50. WATCHLISTS
  51. webrtc.gni
  52. 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