Files
platform-external-webrtc/webrtc/tools/e2e_quality/audio
kjellander@webrtc.org 3c0aae17f0 Change gflags and gmock includes to be full paths.
This will fix PRESUBMIT warnings developers will get due to
r7014 and r7020.

Also some minor style cleanup in:
webrtc/modules/audio_coding/main/test/RTPFile.cc
webrtc/modules/audio_coding/neteq/test/RTPjitter.cc

BUG=
R=henrik.lundin@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7058 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 09:55:40 +00:00
..

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/

webrtc/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})"