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TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/915006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d |
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audio_e2e_harness.cc | ||
daemon.conf | ||
default.pa | ||
README | ||
run_audio_test.py |
The tools here run an end-to-end audio quality test on Linux using PulseAudio. INSTALLATION The test depends on PulseAudio virtual devices (null sinks). Without additional arguments, run_audio_test.py expects a pair of sinks named "capture" and "render". To create these devices at machine startup, place the provided default.pa file in ~/.pulse. Alternately, the "pacmd" commands therein can be run on the command-line to create the devices. Similarly, place the provided daemon.conf file in ~/.pulse to use high quality resampling in PulseAudio. This will reduce the resampling impact on the outcome of the test. Build all WebRTC targets as usual (or just the audio_e2e_harness target) to generate the VoiceEngine harness. USAGE Run run_audio_test.py to start. The script has reasonable defaults and will use the expected location of audio_e2e_harness. Some settings will usually be provided by the user, particularly the comparison tool command-line and regular expression to extract the quality metric. An example command-line, run from trunk/ tools/e2e_quality/audio/run_audio_test.py \ --input=data/voice_engine/audio_short16.pcm --output=e2e_audio_out.pcm \ --codec=L16 --compare="comparison-tool" --regexp="(\d\.\d{3})"