Add rtp_generator utility to rtc_tools.

This CL introduces a new rtp_generator tool that can be utilized to generate
.rtpdump files that can be replayed by the video_replayer. This allows
automated generation of corpus material for the new WebRTC RTP fuzzers in
addition to allowing anyone who is experimenting with a new RTP feature to
quickly debug issues.

It can be used as follows:
./rtp_generator --input_config=./rtc_tools/rtp_generator/configs/vp8.json --output_rtpdump=/tmp/vp8.rtpdump
./video_replay --config_file test/fuzzers/configs/replay_packet_fuzzer/vp8_config.json --input_file /tmp/vp8.rtpdump

It works by generating squares randomly on the screen for a given duration. This
initial version is very limited and doesn't support FEC, RED and other
configurations. I plan to extend it to support these in future CLs.

Bug: webrtc:10117
Change-Id: I31d3dbb6fad73c727145ead4e7d085113d11fc51
Reviewed-on: https://webrtc-review.googlesource.com/c/119964
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26517}
9 files changed
tree: 8e57d8b453274802bebd2deeb968faa9900a9d7b
  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