This is not normal sleep it's microsecond sleep. Its done because we are doing nonblocking read on the ringbuffer that's tied to the hardware. Since this is voip, we must drop audio frames when they are late and in order to do that we must have a high resolution loop. This is only a problem when the audio device is not sending audio at the interval we asked it to. Some cheap hardware cannot reliably deliver audio at 20ms intervals which is why i suggested higher value intervals in the config. The request to remove the sleep is to confirm the proposition that sleep 1ms was really taking 15ms.<br>
<br>I appreciate the suggestion and I understand you are not trying to be a smart ass.<br><br><br><br><div class="gmail_quote">On Mon, May 5, 2008 at 4:35 PM, Michael Jerris <<a href="mailto:mike@jerris.com">mike@jerris.com</a>> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">The place this is trickiest is when you are in a loop where you want<br>
to wait for audio, AND do something every x ms or so. You can't do a<br>
blocking read, and a read with timeout tends to be quite expensive.<br>
There are some ways around this, but sometimes its just the most<br>
efficient, even if not preferred method.<br>
<br>
Mike<br>
<div><div></div><div class="Wj3C7c"><br>
<br>
On May 5, 2008, at 5:26 PM, ?ukasz Zwierko wrote:<br>
<br>
> Hi,<br>
><br>
> Just a thought here: using calls like delay(), sleep() etc. for very<br>
> short amount of time (like in this case a couple of miliseconds)<br>
> should in my opinion be really discouraged. It is often a symptom of<br>
> bad programming even in an embedded enviroment, not mentioning<br>
> platforms like Windows or Linux where you can't really tell what<br>
> priority does your task have, and will it not be starved for a long<br>
> time by other tasks. I should be avoided whenever possible.<br>
> Don't want to sound like a smart ass here, but isn't there any other<br>
> way? From what I understood you wait until some amount of voice<br>
> samples is collected? If that's the case than perhaps you can measure<br>
> an amount of data collected not the time... If these are PCM samples<br>
> than the correlation is straightforward.<br>
> Again, sorry if I'm being a smart ass here but I've seen some really<br>
> bad code with sleep() calls and such like, and I can tell you that it<br>
> only worked fine in specific conditions, and had a tendency to work<br>
> very poorly when for example CPU was under heavy load.<br>
><br>
> Luaksz<br>
><br>
> 2008/5/5 Anthony Minessale <<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>>:<br>
>> did you try setting the ptime on the rtp to 30 or 60ms<br>
>><br>
>> when you choose a codec in your sip settings on FS in vars.xml,<br>
>> instead of<br>
>> PCMU try PCMU@30i or PCMU@60i<br>
>> it may be that the other side is doing 30 or 60 ms and not telling<br>
>> us.<br>
>><br>
>> also in pablio.c in the portaudio_mod directory in ReadAudioStream<br>
>> func,<br>
>> there is a sleep 1 ms too<br>
>> if the windows is really sleeping a lot longer than that, try<br>
>> omitting line<br>
>> 158.<br>
>> This probably will consume the whole cpu but if it fixes your<br>
>> problem it<br>
>> will support the theory that the sleep on windows in inaccurate.<br>
>><br>
>><br>
>><br>
>><br>
>><br>
>> On Mon, May 5, 2008 at 11:38 AM, Csaba Zelei<br>
>> <<a href="mailto:csaba.zelei@gmail.com">csaba.zelei@gmail.com</a>> wrote:<br>
>>><br>
>>><br>
>>><br>
>>><br>
>>> With a little hack I can make mod_portaudio to send rtp packets with<br>
>> ~24ms, ~16ms delay alternately on linux. This result in a constant<br>
>> 4ms<br>
>> jitter but its better than the original. (diff attached)<br>
>>> However on windows the delta between rtp packets is 15-32 ms<br>
>>> randomly,<br>
>> with occasionally high 70-100ms delta.<br>
>>> I also tried to tweak the windows timer without success.<br>
>>> Does anybody has any idea how to make windows xp more accurate?<br>
>>><br>
>>> Sluschny, Thomas wrote:<br>
>>><br>
>>><br>
>>><br>
>>><br>
>>> as you can see here:<br>
>>><br>
>>><br>
>>> <a href="http://jira.freeswitch.org/browse/MODENDP-40" target="_blank">http://jira.freeswitch.org/browse/MODENDP-40</a><br>
>>><br>
>>> i have this problem all the time (the error mentioned in this<br>
>>> issue was<br>
>> only related with this).<br>
>>><br>
>>> It has to do with windows handle sleep() method, you has say<br>
>>> sleep(1) for<br>
>> 1ms but on my<br>
>>><br>
>>> machine it waits 15ms (it depends on your hardware, other PCs behave<br>
>> different!). So i tested around with high performance counters.<br>
>>><br>
>>> For now i ignore that problem an set jitterbuffers on other device<br>
>>> big<br>
>> enough.<br>
>>><br>
>>> Thomas<br>
>>><br>
>>> ________________________________<br>
>> Von: <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a><br>
>> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a>] Im Auftrag<br>
>> von Zelei<br>
>> Csaba<br>
>>> Gesendet: Donnerstag, 24. April 2008 19:05<br>
>>> An: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
>>> Betreff: [Freeswitch-users] mod_portaudio send 3 rtp packet/60ms<br>
>>> instead<br>
>> of1 packet/20ms<br>
>>><br>
>>> Dear all,<br>
>>><br>
>>> I tried to use FS in client mode, starting calls with<br>
>>> mod_portaudio to our<br>
>> providers gateway ( a Cirpack softswitch )<br>
>>> I experienced that there is 2-3 sec delay in the call, its choppy<br>
>>> and<br>
>> robot like.<br>
>>> I tested it with a softphone, and an ip phone and everything was<br>
>>> fine. I<br>
>> traced back the problem to mod_portaudio sending 3 rtp packet in 60ms<br>
>> instead of 1 packet/20ms.<br>
>>><br>
>>> Here is an rtp statistic from a call: (see<br>
>> <a href="http://pastebin.freeswitch.org/4307" target="_blank">http://pastebin.freeswitch.org/4307</a> for the complete list and sip<br>
>> trace)<br>
>>><br>
>>> Packet Sequence Delta (ms)<br>
>>> 42 26138 0.00<br>
>>> 43 26139 0.02<br>
>>> 46 26140 45.69<br>
>>> 47 26141 0.02<br>
>>> 48 26142 2.96<br>
>>> 52 26143 56.31<br>
>>> 53 26144 5.75<br>
>>> 54 26145 0.02<br>
>>> 58 26146 51.99<br>
>>> 59 26147 0.03<br>
>>> 60 26148 2.96<br>
>>> 63 26149 42.95<br>
>>> 65 26150 17.06<br>
>>> 66 26151 0.02<br>
>>> 67 26152 2.90<br>
>>> 71 26153 56.99<br>
>>> 72 26154 0.03<br>
>>> 73 26155 0.02<br>
>>><br>
>>> Did anyone else experience similar problems?<br>
>>> Is this the desired behaviour, because portaudio get data in 60ms<br>
>>> interval<br>
>> or can I set it to 20ms somehow?<br>
>>><br>
>>> Thanks,<br>
>>><br>
>>> Csaba Zelei<br>
>>><br>
>>><br>
>>> ________________________________<br>
>><br>
>>> _______________________________________________<br>
>>> Freeswitch-users mailing list<br>
>>> <a href="mailto:Freeswitch-users@lists.freeswitch.org">Freeswitch-users@lists.freeswitch.org</a><br>
>>> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
>>> UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
>>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>>><br>
>>><br>
>>><br>
>>> Index: mod_portaudio.c<br>
>>> ===================================================================<br>
>>> --- mod_portaudio.c (revision 8260)<br>
>>> +++ mod_portaudio.c (working copy)<br>
>>> @@ -121,6 +121,7 @@<br>
>>> int ring_interval;<br>
>>> GFLAGS flags;<br>
>>> switch_timer_t timer;<br>
>>> + switch_timer_t sync_timer;<br>
>>> } globals;<br>
>>><br>
>>><br>
>>> @@ -282,7 +283,15 @@<br>
>>> }<br>
>>><br>
>>> switch_set_flag_locked(tech_pvt, TFLAG_IO);<br>
>>> +<br>
>>> + /* Start Synchronization Timer */<br>
>>> + //Is it ok to always use 20ms? What about the 160 sample????<br>
>>> + if (<br>
>> switch_core_timer_init(&globals.sync_timer,"soft",<br>
>> 20,160,switch_core_session_get_pool(session))<br>
>> != SWITCH_STATUS_SUCCESS)<br>
>>> + {<br>
>>> + switch_log_printf(SWITCH_CHANNEL_LOG,<br>
>>> SWITCH_LOG_DEBUG,<br>
>> "Sync Timer failed!!\n");<br>
>>> + }<br>
>>><br>
>>> +<br>
>>> /* Move Channel's State Machine to RING */<br>
>>> switch_channel_set_state(channel, CS_RING);<br>
>>><br>
>>> @@ -412,6 +421,8 @@<br>
>>> }<br>
>>><br>
>>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "%s<br>
>>> CHANNEL<br>
>> HANGUP\n",<br>
>> switch_channel_get_name(switch_core_session_get_channel(session)));<br>
>>> + /* Destroy timer */<br>
>>> + switch_core_timer_destroy(&globals.sync_timer);<br>
>>><br>
>>> return SWITCH_STATUS_SUCCESS;<br>
>>> }<br>
>>> @@ -542,12 +553,17 @@<br>
>>> switch_mutex_lock(globals.device_lock);<br>
>>><br>
>>> get_samples:<br>
>>> -<br>
>>> +<br>
>>> if ((samples = ReadAudioStream(globals.audio_stream,<br>
>> globals.read_frame.data,<br>
>>><br>
>> globals.read_codec.implementation->samples_per_frame,<br>
>>><br>
>> &globals.timer)) == 0) {<br>
>>> +<br>
>>> + //switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG,<br>
>>> "No data<br>
>> reset timer\n");<br>
>>> + switch_core_timer_sync(&globals.sync_timer);<br>
>>> switch_yield(1000);<br>
>>> +<br>
>>> goto get_samples;<br>
>>> +<br>
>>> } else {<br>
>>> globals.read_frame.datalen = samples * 2;<br>
>>> globals.read_frame.samples = samples;<br>
>>> @@ -562,7 +578,9 @@<br>
>>> status = SWITCH_STATUS_SUCCESS;<br>
>>> }<br>
>>> switch_mutex_unlock(globals.device_lock);<br>
>>> -<br>
>>> +<br>
>>> + switch_core_timer_next(&globals.sync_timer);<br>
>>> +<br>
>>> return status;<br>
>>><br>
>>> }<br>
>>><br>
>>><br>
>>> _______________________________________________<br>
>>> Freeswitch-users mailing list<br>
>>> <a href="mailto:Freeswitch-users@lists.freeswitch.org">Freeswitch-users@lists.freeswitch.org</a><br>
>>> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
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>>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>>><br>
>>><br>
>><br>
>><br>
>><br>
>> --<br>
>> Anthony Minessale II<br>
>><br>
>> FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>
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>><br>
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>> GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</a><br>
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>><br>
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>><br>
>><br>
><br>
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</div></div></blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
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