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In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
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32 lines
1 KiB
Text
This file describes how to set up and use the RTP log analyzer.
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First build the tool with
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ninja -C out/my_build webrtc:rtp_analyzer
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The tool is built by default, so
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ninja -C out/my_build
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is enough.
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After building, run the tool as follows:
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./out/my_build/rtp_analyzer.sh [options] <rtc event log>
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where <rtc event log> is a recorded RTC event log, which is stored in
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protobuf format. Such logs are generated in multiple ways, e.g. by
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Chrome through the chrome://webrtc-internals page.
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Options:
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-h, --help show this help message and exit
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--dump_header_to_stdout
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print header info to stdout; similar to rtp_analyze
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--query_sample_rate always query user for real sample rate
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The script has been tested to work in python versions 3.4.1 and 2.7.6,
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but should work in all python versions.
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Working versions of NumPy (http://www.numpy.org/) and matplotlib
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(http://matplotlib.org/) are needed to run this tool. See this link
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with installation instructions (http://www.scipy.org/install.html).
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