[Freeswitch-users] mod_portaudio send 3 rtp packet/60ms insteadof1 packet/20ms
Sluschny, Thomas
Thomas.Sluschny at siemens.com
Tue May 6 08:37:19 EDT 2008
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 at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
Anthony Minessale
Gesendet: Dienstag, 6. Mai 2008 00:38
An: freeswitch-users at 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 at 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 at 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 at 30i or PCMU at 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 at 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 at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im
Auftrag
>> von Zelei
>> Csaba
>>> Gesendet: Donnerstag, 24. April 2008 19:05
>>> An: freeswitch-users at 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 at 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 at lists.freeswitch.org
>>>
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>>>
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com>
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>>
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>
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--
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