Have you tried setting the codec-ms in the portaudio.conf.xml to 60 or 120 ms?
Maybe the soundcard is not able to do 20ms intervals and portaudio is doing the least common multiple and chopping it up for us.
I think what's happening is the timer in the module is set to the interval from the config file (20ms) and during every 60ms period there is no audio until the last ms.  so in each 60 ms:

20ms (timeout..... flush buffer)
20ms (timeout..... flush buffer)
20ms (get 60ms worth of audio at once [3 20ms packets] but we have already read 2 filler frames from the timeouts)

So now we have read 5 packets instead of 3 and erased some of our buffer because of perceived timeouts.
The code is using the assumption that if the device will obey the chosen frame size and sample rate requests down to the interval.

If you find and edit conf/autoload_configs/portaudio.conf.xml

look for this:

 <param name="codec-ms" value="20"/>

and change 20 to 60

Setting this to 60 will change the frame size of all the packets from 320 to 960 and set the timer to clock at an interval of 60ms
Since the card seems to be able to reliably produce 3 20ms packets every 60ms it should also be able to produce 1 60ms packet.

FreeSWITCH should then buffer the audio and still deliver it over SIP at 20ms if you want but you can opt to set the codec PCMU@60i to disable buffering if you are in a reliable network.

The same should be true for setting the codec-ms to 120


On Wed, May 7, 2008 at 3:27 AM, Sluschny, Thomas <Thomas.Sluschny@siemens.com> wrote:
Hi Anthony,
 
i also tested your patch with no success.
As i already described below, the problem with all 60ms 3 packets comes from the soundcard.
The hardware delivers its samples all 60 ms.
Our problem is (like Csaba said) that we read out the buffer after 60ms, 3 times, each with samples for 20ms, AND WITH NO DELAY!
So we get: 60ms wait and 3 RTP packets within <1ms to send, and after that we already wait 60 ms for the next samples.
 
In my patch i wait appr.20 ms if last method call was no longer than 4ms ago,
but i think we can do better with switch_core_timer_check() method, but i don't know exactly how.
 
You are absolutly right with your demand for a better timing resolution under Windows,
but this 60ms mystery is caused by the soundcard.
 
Thomas

Von: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Zelei Csaba
Gesendet: Dienstag, 6. Mai 2008 20:22

An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] mod_portaudio send 3 rtppacket/60msinsteadof1 packet/20ms

Yes, but it didnt help with portaudio (just together with the patch)
What else do you need to test it with? Please tell me, and i will do it for you tomorrow.
I also found a test code for timeBeginPeriod on this site: http://www.geisswerks.com/ryan/FAQS/timing.html and experienced that sleep(1) is usually 2 ms long (sometimes 4)


2008/5/6 Anthony Minessale <anthony.minessale@gmail.com>:
Did you try the trunk with no pathc since my latest email r8269?  I added code that makes the windows timer accurate to 1ms precision.  If the windows timer must be made accurate or many other things in FS will not work right.



On Tue, May 6, 2008 at 10:59 AM, Csaba Zelei <csaba.zelei@gmail.com> wrote:
Hi Anthony,

I tried your suggestions (the PCMU@30i  60i and omitting line 158 in pablio.c) but none solve the problem.
I also tried the latest trunk but its the same.
I reached the best performance with the latest trunk and the patch Thomas wrote.(~20ms rtp packet delta, rarely 100+ms)
The only problem with it thats FS ate up 95% cpu but I am not sure if it has to do anything with mod_portaudio or the windows timer.

If I am correct we have 2 problem here.
The first is windows timer isnt very accurate.
The second is that mod_portaudio dont have any delay when it can return audio data immadiately .
Let me explain:

1. call for channel_read_frame
2. waiting for audio data (~60ms)
3. got 60ms audio data
4. get 20ms (remaining 40ms) audio data, return (and send rtp frame)
5. call for channel_read_frame
6. get 20ms, return(and send rtp frame) within ~1ms
7. call for channel_read_frame
8. get 20ms, return(and send rtp frame) within ~1ms
go back to 1 and start again

That's what I tried to solve with my patch, but because of the windows timer it failed.
I hope I could help, and dont misunderstand something.

Csaba

Anthony Minessale írta:
I added a small patch to turn up the resolution of Sleep to 1ms.
Can you see if that helps?

On Tue, May 6, 2008 at 8:38 AM, Sluschny, Thomas <Thomas.Sluschny@siemens.com> wrote:
i already tried to implement such a routine, but as i remember i had problems to compile,
anyway this may be help you:
 
// sleep with smaller tick time, wait in µs
void betterSleep(long wait)
{
 LARGE_INTEGER lElapse;
 BOOL succ;
 HANDLE timerHandle = 0;
 DWORD dwWaitResult;
 
 __try {
 
  // return on 0
  if (wait < 1)
   return;
 
  // create the timer
  timerHandle = (HANDLE) CreateWaitableTimerA(NULL, FALSE, NULL);
 
  // set the timer
  lElapse.QuadPart = 2500LL - (wait * 10000LL);
  succ = SetWaitableTimer(timerHandle, &lElapse, 0, NULL, NULL, FALSE);
 
  if (!succ) {
   printf( "set timer not successful!\n" );
  }
 
  dwWaitResult = WaitForSingleObject(
   timerHandle,  
   wait   // fall back timeout with 15ms granularity
  );
 
  switch (dwWaitResult)
  {
   // The thread got mutex ownership.
  case WAIT_OBJECT_0:
   //printf( "WAIT_OBJECT_0!\n" );
   break;
 
   // Cannot get mutex ownership due to time-out.
  case WAIT_TIMEOUT:
   printf( "WAIT_TIMEOUT!\n" );
   break;
 
   // Got ownership of the abandoned mutex object.
  case WAIT_ABANDONED:
   printf( "WAIT_ABANDONED!\n" );
   break;
  }
 }
 __finally {
  if (timerHandle) CloseHandle(timerHandle);
 }
}
 
It may be hard work but: Its cool to be cross platform - you reach all people around
we count on you ;)
 
Thomas

Gesendet: Dienstag, 6. Mai 2008 15:08

An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] mod_portaudio send 3 rtp packet/60msinsteadof1 packet/20ms

if switch_yield for windows is not working properly maybe if we fix that the code will work as planned.

switch_timer_check tells you if the timer has ticked or not and it's based on a single timer thread that also has the necessity to do 1ms sleeps to broadcast the time changes to the rest of the system.  so let's try using performance counters to implement proper switch_yield and see if the code begins to work as planned.  I'll try to come up with a patch because there are several places including the RTP code where semi-accurate 1ms sleeps are absolutely necessary.

This is one of the many joys of being cross platform. =D




On Tue, May 6, 2008 at 7:37 AM, Sluschny, Thomas <Thomas.Sluschny@siemens.com> wrote:
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 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
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>
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