PJSIP Audio fades, distorts and inaudible on ARM-i.MX53

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Hi,

 

I ran the sndtest and capture the call dump. Does this say anything about
the audio going bad.

 

# ./sndtest

08:13:14.760      sndtest.c !Found 2 devices:

08:13:14.761      sndtest.c   0: default:CARD=imx3stack (capture=1,
playback=1)

08:13:14.761      sndtest.c   1: sysdefault:CARD=imx3stack (capture=1,
playback=1)

08:13:15.220      sndtest.c  Testing playback device default:CARD=imx3stack

08:13:15.220      sndtest.c  Testing capture device default:CARD=imx3stack

08:13:15.424      sndtest.c   Please wait while test is in progress (~11
secs)..

08:13:26.581      sndtest.c   Dumping results:

08:13:26.582      sndtest.c    Parameters: clock rate=8000Hz, 80
samples/frame

08:13:26.582      sndtest.c    Playback stream report:

08:13:26.582      sndtest.c     Duration: 9s.990

08:13:26.582      sndtest.c     Frame interval: min=0.029ms, max=67.945ms

08:13:26.582      sndtest.c     Jitter: min=9.948ms, avg=26.685ms,
max=67.913ms

08:13:26.583      sndtest.c    Capture stream report:

08:13:26.583      sndtest.c     Duration: 9s.980

08:13:26.583      sndtest.c     Frame interval: min=0.092ms, max=63.500ms

08:13:26.583      sndtest.c     Jitter: min=9.893ms, avg=26.536ms,
max=63.406ms

08:13:26.583      sndtest.c    Checking for clock drifts:

08:13:26.583      sndtest.c     Sound capture is 80 samples faster than
playback at the end of the test (average is 8 samples per second)

08:13:26.583      sndtest.c   Test completed with some warnings

=======================================================

Call statistics:

 

[DISCONNCTD] To: <sip:192.168.1.156>;tag=3d928773698d4280957744e929d8aa73

    Call time: 00h:04m:42s, 1st res in 4587 ms, conn in 4642ms

    #0 audio PCMA @8kHz, sendrecv, peer=192.168.1.156:4004

       SRTP status: Not active Crypto-suite:

       RX pt=8, last update:00h:00m:01.465s ago

          total 12.6Kpkt 2.03MB (2.53MB +IP hdr) @avg=57.5Kbps/71.9Kbps

          pkt loss=249 (1.9%), discrd=0 (0.0%), dup=0 (0.0%), reord=3 (0.0%)

                (msec)    min     avg     max     last    dev

          loss period:  20.000  21.652 100.000  20.000   6.131

          jitter     :   0.250   8.685  29.000   7.000   3.389

       TX pt=8, ptime=20, last update:00h:00m:01.318s ago

          total 11.8Kpkt 1.89MB (2.37MB +IP hdr) @avg=53.7Kbps/67.2Kbps

          pkt loss=435 (3.5%), dup=0 (0.0%), reorder=0 (0.0%)

                (msec)    min     avg     max     last    dev

          loss period:  20.000 164.151 1540.000  80.000  41.859

          jitter     :   0.000  14.582  21.875  12.125   4.515

       RTT msec      :   3.082  21.143  73.908  14.831  18.172

00:34:55.055  pjsua_media.c  ......Call 1: deinitializing media..

00:34:55.056  pjsua_media.c  ........Media stream call01:0 is destroyed

00:34:56.055    pjsua_aud.c  Closing sound device after idle for 1 second(s)

00:34:56.055    pjsua_app.c  .Turning sound device OFF

00:34:56.055    pjsua_aud.c  .Closing default:CARD=imx3stack sound playback
device and default:CARD=imx3stack sound capture device

 

 

Thanks & Regards

 

Varma SVRP

From: pjsip-bounces@xxxxxxxxxxxxxxx [mailto:pjsip-bounces at lists.pjsip.org]
On Behalf Of Varma
Sent: Monday, April 07, 2014 9:04 AM
To: pjsip at lists.pjsip.org
Subject: PJSIP Audio fades, distorts and inaudible on ARM-i.MX53

 

Hi,

 

I built PJSIP and run on Freescale ARM i.MX53. The video call works well.
The issue is with the audio. 

The call gets initiated from the i.MX base board to a mobile running the
csipsimple android app. The audio is heard properly on the mobile but not on
the i.MX board. The audio is heard clearly on for 1-2 secs and it starts to
fade out or distorted. Sometimes audio is both distorted and faded. I tried
all the steps in the audio troubleshooting as mentioned in the PJSIP
website. 

 

I checked my sound output by playing an audio file. It plays well.

 

Following steps I tried.

1.       Tried using PCMU and PCMA as the board supports only PCM. SPEEX is
disabled on the board.

2.       Disabled the echo canceller. Did not have any effect on the result.

3.       In-call volume to increased 5.0x, 10.0x and went up to 100.0x.
Every time I increased this the audio gets worser.

4.       Tried forcing 8KHz sample. Still no improvement.

 

Can you suggest what could fix my issue?

 

I could not get the packet statistics. Somehow the TX and RX packet count
always shows zero when I use the 'dq' while in call.

 

Thanks & Regards

 

Varma SVRP

Technical Lead | Orvito Technologies India Pvt Ltd.

M: +91-9032867017

Description: Description: Description: Description:
cid:image002.png at 01CDD3CC.69D07270

8-2-269/S/41 | Plot No.: 41 | Sagar Society| Banjara Hills | Road No.: 2 |
Hyderabad - 500038

www.orvito.com <http://www.orvito.com/> 

 

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