Add support for ICE/STUN/TURN in res_rtp_asterisk and chan_sip.
[asterisk/asterisk.git] / res / pjproject / tests / pjsua / scripts-pesq / 200_codec_speex_16000.py
1 # $Id$
2 #
3 from inc_cfg import *
4
5 ADD_PARAM = ""
6
7 if (HAS_SND_DEV == 0):
8         ADD_PARAM += "--null-audio"
9
10 # Call with Speex/16000 codec
11 test_param = TestParam(
12                 "PESQ codec Speex WB",
13                 [
14                         InstanceParam("UA1", ADD_PARAM + " --max-calls=1 --clock-rate 16000 --add-codec speex/16000 --play-file wavs/input.16.wav --no-vad"),
15                         InstanceParam("UA2", "--null-audio --max-calls=1 --clock-rate 16000 --add-codec speex/16000 --rec-file  wavs/tmp.16.wav --auto-answer 200")
16                 ]
17                 )
18
19 pesq_threshold = 3.8