Anthony seams to be right.
I tested around a little bit and i see:
- the main problem is the soundcard (-driver), it gives 480 frames all 60 ms
- if i chose 160 (256) frames ringbuffer size i get only 160 frames all 60 ms, and it sounds really croppy (of course, there are not enough samples ...)
- i get really good results if i store the last timestamp when ReadAudioStream() returns, and if current timestamp is less than 20ms from last i wait 20 ms,
so the packet difference went from 60-0-0-60-0-0 ms to 20-20-20-20-20-20
- i use HighPerformanceCounter to measure the time in WinXP
 
I thought we could use switch_core_timer_check() to do that timestamp work, but it doesn't work as expected,
seams we have to synchronize to the 60ms clock from soundcard, at least at the beginning, which means the delay increase.
 
Thomas
 
PS: i attach a patch for better understanding 

Von: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Anthony Minessale
Gesendet: Dienstag, 6. Mai 2008 00:38
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] mod_portaudio send 3 rtp packet/60ms insteadof1 packet/20ms

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.

I appreciate the suggestion and I understand you are not trying to be a smart ass.



On Mon, May 5, 2008 at 4:35 PM, Michael Jerris <mike@jerris.com> wrote:
The place this is trickiest is when you are in a loop where you want
to wait for audio, AND do something every x ms or so.  You can't do a
blocking read, and a read with timeout tends to be quite expensive.
There are some ways around this, but sometimes its just the most
efficient, even if not preferred method.

Mike


On May 5, 2008, at 5:26 PM, ?ukasz Zwierko wrote:

> Hi,
>
> Just a thought here: using calls like delay(), sleep() etc. for very
> short amount of time (like in this case a couple of miliseconds)
> should in my opinion be really discouraged. It is often a symptom of
> bad programming even in an embedded enviroment, not mentioning
> platforms like Windows or Linux where you can't really tell what
> priority does your task have, and will it not be starved for a long
> time by other tasks. I should be avoided whenever possible.
> Don't want to sound like a smart ass here, but isn't there any other
> way? From what I understood you wait until some  amount of voice
> samples is collected? If that's the case than perhaps you can measure
> an amount of data collected not the time... If these are PCM samples
> than the correlation is straightforward.
> Again, sorry if I'm being a smart ass here but I've seen some really
> bad code with sleep() calls and such like, and I can tell you that it
> only worked fine in specific conditions, and had a tendency to work
> very poorly when for example CPU was under heavy load.
>
> Luaksz
>
> 2008/5/5 Anthony Minessale <anthony.minessale@gmail.com>:
>> did you try setting the ptime on the rtp to 30 or 60ms
>>
>> when you choose a codec in your sip settings on FS in vars.xml,
>> instead of
>> PCMU try PCMU@30i or PCMU@60i
>> it may be that the other side is doing 30 or 60 ms and not telling
>> us.
>>
>> also in pablio.c in the portaudio_mod directory in ReadAudioStream
>> func,
>> there is a sleep 1 ms too
>> if the windows is really sleeping a lot longer than that, try
>> omitting line
>> 158.
>> This probably will consume the whole cpu but if it fixes your
>> problem it
>> will support the theory that the sleep on windows in inaccurate.
>>
>>
>>
>>
>>
>> On Mon, May 5, 2008 at 11:38 AM, Csaba Zelei
>> <csaba.zelei@gmail.com> wrote:
>>>
>>>
>>>
>>>
>>> With a little hack I can make mod_portaudio to send rtp packets with
>> ~24ms, ~16ms delay alternately on linux. This result in a constant
>> 4ms
>> jitter but its better than the original. (diff attached)
>>> However on windows the delta between rtp packets is 15-32 ms
>>> randomly,
>> with occasionally high 70-100ms delta.
>>> I also tried to tweak the windows timer without success.
>>> Does anybody has any idea how to make windows xp more accurate?
>>>
>>> Sluschny, Thomas wrote:
>>>
>>>
>>>
>>>
>>> as you can see here:
>>>
>>>
>>> http://jira.freeswitch.org/browse/MODENDP-40
>>>
>>> i have this problem all the time (the error mentioned in this
>>> issue was
>> only related with this).
>>>
>>> It has to do with windows handle sleep() method, you has say
>>> sleep(1) for
>> 1ms but on my
>>>
>>> machine it waits 15ms (it depends on your hardware, other PCs behave
>> different!). So i tested around with high performance counters.
>>>
>>> For now i ignore that problem an set jitterbuffers on other device
>>> big
>> enough.
>>>
>>> Thomas
>>>
>>> ________________________________
>> Von: freeswitch-users-bounces@lists.freeswitch.org
>> [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag
>> von Zelei
>> Csaba
>>> Gesendet: Donnerstag, 24. April 2008 19:05
>>> An: freeswitch-users@lists.freeswitch.org
>>> Betreff: [Freeswitch-users] mod_portaudio send 3 rtp packet/60ms
>>> instead
>> of1 packet/20ms
>>>
>>> Dear all,
>>>
>>> I tried to use FS in client mode, starting calls with
>>> mod_portaudio to our
>> providers gateway ( a Cirpack softswitch )
>>> I experienced that there is 2-3 sec delay in the call, its choppy
>>> and
>> robot like.
>>> I tested it with a softphone, and an ip phone and everything was
>>> fine. I
>> traced back the problem to mod_portaudio sending 3 rtp packet in 60ms
>> instead of 1 packet/20ms.
>>>
>>> Here is an rtp statistic from a call: (see
>> http://pastebin.freeswitch.org/4307 for the complete list and sip
>> trace)
>>>
>>> Packet    Sequence    Delta (ms)
>>> 42  26138    0.00
>>> 43    26139    0.02
>>> 46    26140    45.69
>>> 47    26141    0.02
>>> 48    26142    2.96
>>> 52    26143    56.31
>>> 53    26144    5.75
>>> 54    26145    0.02
>>> 58    26146    51.99
>>> 59    26147    0.03
>>> 60    26148    2.96
>>> 63    26149    42.95
>>> 65    26150    17.06
>>> 66    26151    0.02
>>> 67    26152    2.90
>>> 71    26153    56.99
>>> 72    26154    0.03
>>> 73    26155    0.02
>>>
>>> Did anyone else experience similar problems?
>>> Is this the desired behaviour, because portaudio get data in 60ms
>>> interval
>> or can I set it to 20ms somehow?
>>>
>>> Thanks,
>>>
>>> Csaba Zelei
>>>
>>>
>>> ________________________________
>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users@lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> Index: mod_portaudio.c
>>> ===================================================================
>>> --- mod_portaudio.c     (revision 8260)
>>> +++ mod_portaudio.c     (working copy)
>>> @@ -121,6 +121,7 @@
>>>       int ring_interval;
>>>       GFLAGS flags;
>>>       switch_timer_t timer;
>>> +       switch_timer_t sync_timer;
>>> } globals;
>>>
>>>
>>> @@ -282,7 +283,15 @@
>>>       }
>>>
>>>       switch_set_flag_locked(tech_pvt, TFLAG_IO);
>>> +
>>> +       /* Start Synchronization Timer */
>>> +       //Is it ok to always use 20ms? What about the 160 sample????
>>> +       if (
>> switch_core_timer_init(&globals.sync_timer,"soft",
>> 20,160,switch_core_session_get_pool(session))
>> != SWITCH_STATUS_SUCCESS)
>>> +       {
>>> +               switch_log_printf(SWITCH_CHANNEL_LOG,
>>> SWITCH_LOG_DEBUG,
>> "Sync Timer failed!!\n");
>>> +       }
>>>
>>> +
>>>       /* Move Channel's State Machine to RING */
>>>       switch_channel_set_state(channel, CS_RING);
>>>
>>> @@ -412,6 +421,8 @@
>>>       }
>>>
>>>       switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "%s
>>> CHANNEL
>> HANGUP\n",
>> switch_channel_get_name(switch_core_session_get_channel(session)));
>>> +       /* Destroy timer */
>>> +       switch_core_timer_destroy(&globals.sync_timer);
>>>
>>>       return SWITCH_STATUS_SUCCESS;
>>> }
>>> @@ -542,12 +553,17 @@
>>>       switch_mutex_lock(globals.device_lock);
>>>
>>> get_samples:
>>> -
>>> +
>>>       if ((samples = ReadAudioStream(globals.audio_stream,
>> globals.read_frame.data,
>>>
>> globals.read_codec.implementation->samples_per_frame,
>>>
>> &globals.timer)) == 0) {
>>> +
>>> +       //switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG,
>>> "No data
>> reset timer\n");
>>> +       switch_core_timer_sync(&globals.sync_timer);
>>>        switch_yield(1000);
>>> +
>>>        goto get_samples;
>>> +
>>>       } else {
>>>               globals.read_frame.datalen = samples * 2;
>>>               globals.read_frame.samples = samples;
>>> @@ -562,7 +578,9 @@
>>>               status = SWITCH_STATUS_SUCCESS;
>>>       }
>>>       switch_mutex_unlock(globals.device_lock);
>>> -
>>> +
>>> +       switch_core_timer_next(&globals.sync_timer);
>>> +
>>>       return status;
>>>
>>> }
>>>
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users@lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>
>> AIM: anthm
>> MSN:anthony_minessale@hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
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>> iax:guest@conference.freeswitch.org/888
>> googletalk:conf+888@conference.freeswitch.org
>> pstn:213-799-1400
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>>
>>
>
> _______________________________________________
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--
Anthony Minessale II

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