Re: [pjsip] Audio latency in Windows Mobile

NI
Nanang Izzuddin
Fri, Dec 11, 2009 5:54 AM

How could you be sure that it's not audio device performance? Audio
device burst level could be checked using 'device test' of
pjsystest[1]. Also, please try making call using lightweight codec
such as G711, as CPU load spikes may disturb audio clock. The audio
latency section in FAQ, you've mentioned before, may describe more.


[1] http://trac.pjsip.org/repos/wiki/Testing_Audio_Device_with_pjsystest

BR,
nanang

On Fri, Dec 11, 2009 at 1:05 AM, Nuno Costa nuno.costa@wit-software.com wrote:

Hi,

I've been taking a deeper look into the media flow implementation and I
think the problem for the high latency might be in the network management
(maybe at the device level).
Beyond the JB summary from the previous message, check the following logs:

17:49:30.000   strm01D5B674  Jitter buffer empty (prefetch=40)
17:49:30.000     ec01CF73F0  Underflow, buf_cnt=0, will generate 1 frame
17:49:30.000   Master/sound  140 samples reduced, buf_cnt=815
17:49:30.000   Master/sound  Buffer size adjusted from 955 to 815
(eff_cnt=689)
17:49:31.000     ec01CF73F0  Underflow, buf_cnt=0, will generate 1 frame
17:49:31.000   strm01D5B674  jb updated(2), prefetch=40, size=49
17:49:31.000   strm01D5B674  jb updated(2), prefetch=40, size=41
17:49:31.000   Master/sound  143 samples reduced, buf_cnt=832
17:49:31.000   Master/sound  Buffer size adjusted from 975 to 832
(eff_cnt=689)
17:49:32.000     ec01CF73F0  Underflow, buf_cnt=0, will generate 1 frame
17:49:32.000   strm01D5B674  Jitter buffer empty (prefetch=40)
17:49:32.000   strm01D5B674  Jitter buffer empty (prefetch=40)
17:49:32.000   strm01D5B674  jb updated(2), prefetch=40, size=49
17:49:33.000   Master/sound  141 samples reduced, buf_cnt=851
17:49:33.000   Master/sound  Buffer size adjusted from 992 to 851
(eff_cnt=676)
17:49:33.000   Master/sound  140 samples reduced, buf_cnt=711
17:49:33.000   Master/sound  Buffer size adjusted from 851 to 711
(eff_cnt=676)
17:49:33.000   strm01D5B674  Jitter buffer empty (prefetch=40)
17:49:33.000     ec01CF73F0  Underflow, buf_cnt=0, will generate 1 frame
17:49:34.000   Master/sound  136 samples reduced, buf_cnt=735
17:49:34.000   Master/sound  Buffer size adjusted from 871 to 735
(eff_cnt=667)
17:49:34.000   strm01D5B674  jb updated(2), prefetch=40, size=41

The jitter buffer is becoming many times empty which means that the audio
latency experienced is not related with the audio device performance.
I've rechecked the network latency and it is very low. Could it be a network
device issue?

Have anyone experienced this before?
Any ideas on a possible solution?

Nanang: do you have any suggestion, looking to the JB summary?

Thank you very much for your support.

Cheers,
Nuno Costa

On 09-12-2009 19:31, Nuno Costa wrote:

Nanang,

Thank you very much for your reply.
The JB summary below is from a test call:

  JB summary:
    size=44 prefetch=40
    delay (min/max/avg/dev)=20/500/362/128 ms
    burst (min/max/avg/dev)=2/98/33/29 frames
    lost=86 discard=781 empty=427

The average delay (362ms) is very high and has impact on the overall
latency.
Any clue on how to optimize this value?

Best regards,
Nuno Costa

On 09-12-2009 17:41, Nanang Izzuddin wrote:

Hi Nuno,

Problem behind ticket #783 is high latency issue on WM :)

Agree, 300ms different is quite strange/high. It is usually useful to
examine the JB statistics at the end of call (printed by
pjmedia_jbuf_destroy()), e.g: if JB was used to get full, clock-drift
or high burst of audio device might take place.

BR,
nanang

On Thu, Dec 3, 2009 at 5:30 AM,  ncosta@wit-software.com wrote:

Hi all,

I've developed an application using PJSIP for Microsoft Windows Mobile 6.x
devices and I've been testing it on some devices.

Unfortunately the audio latency is very high on both devices (Samsung
Omnia II and LG LM730). Values are typically as high as 700ms, measured
from end-to-end.

I've checked your wiki:
http://trac.pjsip.org/repos/wiki/FAQ#audio-latency

And also this ticket:
http://trac.pjsip.org/repos/ticket/783

I've done several tests adjusting the parameters but results haven't
changed. Running 'pjsystest', the average value is 400ms for both devices.

Has anyone faced this problem already?

I've done these tests on a local network with a low network latency so I
do not quite understand the 300ms difference between the end-to-end test
and pjsystest.

Any clue on the problem behind ticket #783?

Thanks!

Best regards,
Nuno Costa


Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org


Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

--

---========
Nuno Costa
Senior Engineer
WIT Software S.A.
Coimbra, Lisboa, Leiria, Porto (Portugal)
San Jose (California, USA)
Phone : +351 239 801030
Mobile: +351 91 9821825
Email : nuno.costa@wit-software.com
Web: http://www.wit-software.com

---========

--

---========
Nuno Costa
Senior Engineer
WIT Software S.A.
Coimbra, Lisboa, Leiria, Porto (Portugal)
San Jose (California, USA)
Phone : +351 239 801030
Mobile: +351 91 9821825
Email : nuno.costa@wit-software.com
Web: http://www.wit-software.com

---========

How could you be sure that it's not audio device performance? Audio device burst level could be checked using 'device test' of pjsystest[1]. Also, please try making call using lightweight codec such as G711, as CPU load spikes may disturb audio clock. The audio latency section in FAQ, you've mentioned before, may describe more. --- [1] http://trac.pjsip.org/repos/wiki/Testing_Audio_Device_with_pjsystest --- BR, nanang On Fri, Dec 11, 2009 at 1:05 AM, Nuno Costa <nuno.costa@wit-software.com> wrote: > Hi, > > I've been taking a deeper look into the media flow implementation and I > think the problem for the high latency might be in the network management > (maybe at the device level). > Beyond the JB summary from the previous message, check the following logs: > > 17:49:30.000   strm01D5B674  Jitter buffer empty (prefetch=40) > 17:49:30.000     ec01CF73F0  Underflow, buf_cnt=0, will generate 1 frame > 17:49:30.000   Master/sound  140 samples reduced, buf_cnt=815 > 17:49:30.000   Master/sound  Buffer size adjusted from 955 to 815 > (eff_cnt=689) > 17:49:31.000     ec01CF73F0  Underflow, buf_cnt=0, will generate 1 frame > 17:49:31.000   strm01D5B674  jb updated(2), prefetch=40, size=49 > 17:49:31.000   strm01D5B674  jb updated(2), prefetch=40, size=41 > 17:49:31.000   Master/sound  143 samples reduced, buf_cnt=832 > 17:49:31.000   Master/sound  Buffer size adjusted from 975 to 832 > (eff_cnt=689) > 17:49:32.000     ec01CF73F0  Underflow, buf_cnt=0, will generate 1 frame > 17:49:32.000   strm01D5B674  Jitter buffer empty (prefetch=40) > 17:49:32.000   strm01D5B674  Jitter buffer empty (prefetch=40) > 17:49:32.000   strm01D5B674  jb updated(2), prefetch=40, size=49 > 17:49:33.000   Master/sound  141 samples reduced, buf_cnt=851 > 17:49:33.000   Master/sound  Buffer size adjusted from 992 to 851 > (eff_cnt=676) > 17:49:33.000   Master/sound  140 samples reduced, buf_cnt=711 > 17:49:33.000   Master/sound  Buffer size adjusted from 851 to 711 > (eff_cnt=676) > 17:49:33.000   strm01D5B674  Jitter buffer empty (prefetch=40) > 17:49:33.000     ec01CF73F0  Underflow, buf_cnt=0, will generate 1 frame > 17:49:34.000   Master/sound  136 samples reduced, buf_cnt=735 > 17:49:34.000   Master/sound  Buffer size adjusted from 871 to 735 > (eff_cnt=667) > 17:49:34.000   strm01D5B674  jb updated(2), prefetch=40, size=41 > > The jitter buffer is becoming many times empty which means that the audio > latency experienced is not related with the audio device performance. > I've rechecked the network latency and it is very low. Could it be a network > device issue? > > Have anyone experienced this before? > Any ideas on a possible solution? > > Nanang: do you have any suggestion, looking to the JB summary? > > Thank you very much for your support. > > Cheers, > Nuno Costa > > > On 09-12-2009 19:31, Nuno Costa wrote: > > Nanang, > > Thank you very much for your reply. > The JB summary below is from a test call: > >   JB summary: >     size=44 prefetch=40 >     delay (min/max/avg/dev)=20/500/362/128 ms >     burst (min/max/avg/dev)=2/98/33/29 frames >     lost=86 discard=781 empty=427 > > The average delay (362ms) is very high and has impact on the overall > latency. > Any clue on how to optimize this value? > > Best regards, > Nuno Costa > > > On 09-12-2009 17:41, Nanang Izzuddin wrote: > > Hi Nuno, > > Problem behind ticket #783 is high latency issue on WM :) > > Agree, 300ms different is quite strange/high. It is usually useful to > examine the JB statistics at the end of call (printed by > pjmedia_jbuf_destroy()), e.g: if JB was used to get full, clock-drift > or high burst of audio device might take place. > > BR, > nanang > > > On Thu, Dec 3, 2009 at 5:30 AM, <ncosta@wit-software.com> wrote: > > > Hi all, > > I've developed an application using PJSIP for Microsoft Windows Mobile 6.x > devices and I've been testing it on some devices. > > Unfortunately the audio latency is very high on both devices (Samsung > Omnia II and LG LM730). Values are typically as high as 700ms, measured > from end-to-end. > > I've checked your wiki: > http://trac.pjsip.org/repos/wiki/FAQ#audio-latency > > And also this ticket: > http://trac.pjsip.org/repos/ticket/783 > > I've done several tests adjusting the parameters but results haven't > changed. Running 'pjsystest', the average value is 400ms for both devices. > > Has anyone faced this problem already? > > I've done these tests on a local network with a low network latency so I > do not quite understand the 300ms difference between the end-to-end test > and pjsystest. > > Any clue on the problem behind ticket #783? > > Thanks! > > Best regards, > Nuno Costa > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > -- > ========================================= > Nuno Costa > Senior Engineer > WIT Software S.A. > Coimbra, Lisboa, Leiria, Porto (Portugal) > San Jose (California, USA) > Phone : +351 239 801030 > Mobile: +351 91 9821825 > Email : nuno.costa@wit-software.com > Web: http://www.wit-software.com > ========================================= > > -- > ========================================= > Nuno Costa > Senior Engineer > WIT Software S.A. > Coimbra, Lisboa, Leiria, Porto (Portugal) > San Jose (California, USA) > Phone : +351 239 801030 > Mobile: +351 91 9821825 > Email : nuno.costa@wit-software.com > Web: http://www.wit-software.com > =========================================