87bbb91469
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}
22 lines
506 B
Python
22 lines
506 B
Python
include_rules = [
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"+call",
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"+common_audio",
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"+common_video",
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"+logging/rtc_event_log",
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"+media",
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"+modules/audio_device",
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"+modules/audio_coding/audio_network_adaptor",
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"+modules/audio_coding/neteq/include",
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"+modules/audio_coding/neteq/tools",
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"+modules/audio_processing",
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"+modules/bitrate_controller",
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"+modules/remote_bitrate_estimator",
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"+modules/congestion_controller",
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"+modules/pacing",
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"+modules/rtp_rtcp",
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"+system_wrappers",
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"+p2p",
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"+third_party/libyuv",
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]
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