[Jack-Devel] Cannot combine audio devices with more than 64 channels

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[Jack-Devel] Cannot combine audio devices with more than 64 channels

Jörg Müller
I need to combine two HDSPe MADI FX cards to one virtual device. I have a working driver, which is alsa-compatible. I can select each single card in any alsa-compatible application and all channels work flawless.
For combining the MADI FX cards to one virtual device, I created an .asoundrc with 194 inputs for each card. When I start that virtual device via
jackd -R -d alsa -C madifx_record_all -P madifx_playback_all

I get this error:
> creating alsa driver ...
> madifx_playback_all|madifx_record_all|1024|2|48000|0|0|nomon|swmeter|-|32bit
> jackd: pcm_multi.c:1060: snd_pcm_multi_open: Assertion `!slave_map[sidxs[i]][schannels[i]]' failed.

However, it works when I reduce the amount from 194 to 64 channels per card. I tried to use 128 channels per card, but that fails the same way. See my alsa-info attached, which also includes the .asoundrc content.

I also found this, which might be related: https://ccrma.stanford.edu/mirrors/lalists/lad/2005/06/0202.html

To me, this looks like a bug. What do you think?

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Re: Cannot combine audio devices with more than 64 channels

Adrian Knoth
On Mon, Jan 23, 2017 at 01:57:09PM +0100, Jörg Müller wrote:

> I need to combine two HDSPe MADI FX cards to one virtual device. I have a

Fancy!

> For combining the MADI FX cards to one virtual device, I created an
> .asoundrc with 194 inputs for each card. When I start that virtual device

You can (and probably should) ask this again on the ALSA ML if you want
to do it via ALSA virtual devices. The other option is to have jack open
both cards:

   https://github.com/jackaudio/jack1/commit/40b01cc3691f25840b40432ec16fd312dbfbd365

I have never tried it, but in theory, you just add -A 2ndALSAname to the
command line and be done.

Not sure how much documentation is available for this on the internet,
but might be worth trying.


Cheers

--
mail: [hidden email]   http://adi.thur.de        PGP/GPG: key via keyserver

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Re: Cannot combine audio devices with more than 64 channels

Chris Caudle
On Mon, January 23, 2017 7:31 am, Adrian Knoth wrote:
> ...The other option is to have jack open
> both cards:
> I have never tried it, but in theory, you just add -A 2ndALSAname to the
> command line and be done.

Note that using jack -A uses the zita resampler, which means the audio
data will not be bit-exact.  That often does not matter, but could be
important to know if some channels are going to a Dolby Digital decoder,
for example.

--
Chris Caudle


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Re: Cannot combine audio devices with more than 64 channels

Joakim Hernberg-2
On Mon, 23 Jan 2017 09:14:16 -0600
"Chris Caudle" <[hidden email]> wrote:

> Note that using jack -A uses the zita resampler, which means the audio
> data will not be bit-exact.  That often does not matter, but could be
> important to know if some channels are going to a Dolby Digital
> decoder, for example.

Is there really SRC going on if the cards are synced externally?

--

   Joakim
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Re: Cannot combine audio devices with more than 64 channels

Fons Adriaensen-3
On Mon, Jan 23, 2017 at 04:22:04PM +0100, Joakim Hernberg wrote:
> On Mon, 23 Jan 2017 09:14:16 -0600
> "Chris Caudle" <[hidden email]> wrote:
>
> > Note that using jack -A uses the zita resampler, which means the audio
> > data will not be bit-exact.  That often does not matter, but could be
> > important to know if some channels are going to a Dolby Digital
> > decoder, for example.
>
> Is there really SRC going on if the cards are synced externally?

With the official zita-ajbridge you can disable resampling. Don't
know about the Jack built-ins.

Ciao,

--
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)

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Re: Cannot combine audio devices with more than 64 channels

John Rigg-16
On Mon, Jan 23, 2017 at 05:35:57PM +0000, Fons Adriaensen wrote:
> With the official zita-ajbridge you can disable resampling.

Good to know.

John
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Re: Cannot combine audio devices with more than 64 channels

Ralf Mardorf
On Tue, 24 Jan 2017 07:41:50 +0000, [hidden email] wrote:
>On Mon, Jan 23, 2017 at 05:35:57PM +0000, Fons Adriaensen wrote:
>> With the official zita-ajbridge you can disable resampling.  
>
>Good to know.

Could this be used without word clock, by using a digital interface to
sync, e.g. S/PDIF? If so, assuming two identical cards would be used,
would sync be hat good, that each combination of IOs, e.g. a channel
from one card for the left and of the other card for the right channel
of a stereo signal, wouldn't cause phasing issues?

Regards,
Ralf
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Re: Cannot combine audio devices with more than 64 channels

Thomas Brand
On Tue, January 24, 2017 09:22, Ralf Mardorf wrote:

> On Tue, 24 Jan 2017 07:41:50 +0000, [hidden email] wrote:
>
>> On Mon, Jan 23, 2017 at 05:35:57PM +0000, Fons Adriaensen wrote:
>>
>>> With the official zita-ajbridge you can disable resampling.
>>>
>>
>> Good to know.
>>
>
> Could this be used without word clock, by using a digital interface to
> sync, e.g. S/PDIF? If so, assuming two identical cards would be used, would
> sync be hat good, that each combination of IOs, e.g. a channel from one
> card for the left and of the other card for the right channel of a stereo
> signal, wouldn't cause phasing issues?
>
yes i think that works pretty good (drawing from earlier experiments).
Using S/PDIF just for sync is a cheap alternative to a "real" wordclock.
One of both interfaces will be slaved to the other and then both should
run the same cycle at the same time (so theoretically no phasing issues).
You just have to manage to shove forth and back the sample data from/to
interface at good enough rate which isn't a particular problem on
localhost using UDP like zita does. If there is no resampling involved,
this can work for many many channels without hitting the CPU too much.

Greetings
Thomas



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Re: Cannot combine audio devices with more than 64 channels

Thomas Brand
On Tue, January 24, 2017 16:27, Thomas Brand wrote:

> On Tue, January 24, 2017 09:22, Ralf Mardorf wrote:
>
>> On Tue, 24 Jan 2017 07:41:50 +0000, [hidden email] wrote:
>>
>>
>>> On Mon, Jan 23, 2017 at 05:35:57PM +0000, Fons Adriaensen wrote:
>>>
>>>
>>>> With the official zita-ajbridge you can disable resampling.
>>>>
>>>>
>>>
>>> Good to know.
>>>
>>>
>>
>> Could this be used without word clock, by using a digital interface to
>> sync, e.g. S/PDIF? If so, assuming two identical cards would be used,
>> would sync be hat good, that each combination of IOs, e.g. a channel
>> from one card for the left and of the other card for the right channel
>> of a stereo signal, wouldn't cause phasing issues?
>>
> yes i think that works pretty good (drawing from earlier experiments).
> Using S/PDIF just for sync is a cheap alternative to a "real" wordclock.
> One of both interfaces will be slaved to the other and then both should
> run the same cycle at the same time (so theoretically no phasing issues).
> You just have to manage to shove forth and back the sample data from/to
> interface at good enough rate which isn't a particular problem on localhost
> using UDP like zita does. If there is no resampling involved, this can
> work for many many channels without hitting the CPU too much.
>
> Greetings
> Thomas
>
Even if the interfaces are synchronized with S/PDIF, there will be some
cycles (hopefully a static number) needed to shove data forth and back.
This needs to be considered to effectively get synchronous behaviour for
input or output to both interfaces.



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Re: Cannot combine audio devices with more than 64 channels

Jörg Müller
Thanks for the information.

The two MADI FX cards I'm using can be synced via word clock.

However, the information needs to be unaltered, no resampling or any modification is allowed for our processing. What command line settings do I have to use to make sure of that? How can I use the official zita-ajbridge or is there some documentation about it?

2017-01-24 16:36 GMT+01:00 Thomas Brand <[hidden email]>:
On Tue, January 24, 2017 16:27, Thomas Brand wrote:
> On Tue, January 24, 2017 09:22, Ralf Mardorf wrote:
>
>> On Tue, 24 Jan 2017 07:41:50 +0000, [hidden email] wrote:
>>
>>
>>> On Mon, Jan 23, 2017 at 05:35:57PM +0000, Fons Adriaensen wrote:
>>>
>>>
>>>> With the official zita-ajbridge you can disable resampling.
>>>>
>>>>
>>>
>>> Good to know.
>>>
>>>
>>
>> Could this be used without word clock, by using a digital interface to
>> sync, e.g. S/PDIF? If so, assuming two identical cards would be used,
>> would sync be hat good, that each combination of IOs, e.g. a channel
>> from one card for the left and of the other card for the right channel
>> of a stereo signal, wouldn't cause phasing issues?
>>
> yes i think that works pretty good (drawing from earlier experiments).
> Using S/PDIF just for sync is a cheap alternative to a "real" wordclock.
> One of both interfaces will be slaved to the other and then both should
> run the same cycle at the same time (so theoretically no phasing issues).
> You just have to manage to shove forth and back the sample data from/to
> interface at good enough rate which isn't a particular problem on localhost
> using UDP like zita does. If there is no resampling involved, this can
> work for many many channels without hitting the CPU too much.
>
> Greetings
> Thomas
>
Even if the interfaces are synchronized with S/PDIF, there will be some
cycles (hopefully a static number) needed to shove data forth and back.
This needs to be considered to effectively get synchronous behaviour for
input or output to both interfaces.



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Re: Cannot combine audio devices with more than 64 channels

Thomas Brand
On Tue, January 24, 2017 17:13, Jörg Müller wrote:

> Thanks for the information.
>
>
> The two MADI FX cards I'm using can be synced via word clock.
>
>
> However, the information needs to be unaltered, no resampling or any
> modification is allowed for our processing. What command line settings do
> I
> have to use to make sure of that? How can I use the official zita-ajbridge
>  or is there some documentation about it?
>
yes; man zita-n2j

in a nutshell, for two local JACK servers:
main server (default):
jackd -R -d...
zita-j2n --float 10.10.10.19 4445
zita-n2j --buf 0 10.10.10.19 6667

second server (foo):
jackd --name foo ...
JACK_DEFAULT_SERVER=foo zita-n2j --buff 0 10.10.10.19 4445
JACK_DEFAULT_SERVER=foo zita-j2n --float 10.10.10.19 6667

This creates 4 clients in 2 JACK servers ready to be connected in both
directions, for 2 channels (default setting) using MTU of 1500 (default
setting). The delay is constant from what i see in a scope.
10.10.10.19 is the IP address of the local network interface (127.0.0.1 or
localhost should work too).
You still have to find in man page of a recent zita-n2j how to turn off
resampling and set channel counts.

Good luck,
Thomas

> 2017-01-24 16:36 GMT+01:00 Thomas Brand <[hidden email]>:
>
>
>> On Tue, January 24, 2017 16:27, Thomas Brand wrote:
>>
>>> On Tue, January 24, 2017 09:22, Ralf Mardorf wrote:
>>>
>>>
>>>> On Tue, 24 Jan 2017 07:41:50 +0000, [hidden email] wrote:
>>>>
>>>>
>>>>
>>>>> On Mon, Jan 23, 2017 at 05:35:57PM +0000, Fons Adriaensen wrote:
>>>>>
>>>>>
>>>>>
>>>>>> With the official zita-ajbridge you can disable resampling.
>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>> Good to know.
>>>>>
>>>>>
>>>>>
>>>>
>>>> Could this be used without word clock, by using a digital interface
>>>> to sync, e.g. S/PDIF? If so, assuming two identical cards would be
>>>> used, would sync be hat good, that each combination of IOs, e.g. a
>>>> channel from one card for the left and of the other card for the
>>>> right channel of a stereo signal, wouldn't cause phasing issues?
>>>>
>>> yes i think that works pretty good (drawing from earlier
>>> experiments). Using S/PDIF just for sync is a cheap alternative to a
>>> "real" wordclock.
>>> One of both interfaces will be slaved to the other and then both
>>> should run the same cycle at the same time (so theoretically no
>>> phasing issues). You just have to manage to shove forth and back the
>>> sample data from/to interface at good enough rate which isn't a
>>> particular problem on
>> localhost
>>> using UDP like zita does. If there is no resampling involved, this
>>> can work for many many channels without hitting the CPU too much.
>>>
>>> Greetings
>>> Thomas
>>>
>>>
>> Even if the interfaces are synchronized with S/PDIF, there will be some
>>  cycles (hopefully a static number) needed to shove data forth and
>> back. This needs to be considered to effectively get synchronous
>> behaviour for input or output to both interfaces.
>>
>>
>>
>> _______________________________________________
>> Jack-Devel mailing list
>> [hidden email]
>> http://lists.jackaudio.org/listinfo.cgi/jack-devel-jackaudio.org
>>
>>
>


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Re: Cannot combine audio devices with more than 64 channels

Thomas Brand
In reply to this post by Jörg Müller

Hi again,

I'd like to throw in a possible alternative for non-resampled audio
transfer via UDP.

jack_audio_send and jack_audio_receive [1]

These programs have several limitations however for a case with two
externally synced interfaces both connected to the same host running JACK
with the same settings it could be worth a try. The number of channels
that can be transimitted depends on the period size.

For two JACK running at -p64, this *should* work:
jack_audio_send --in 196 localhost 1234
jack_audio_receive --out 196 1234
...
196 channels are quite heavy and the send/receive aren't possibly robust
enough so you'd need to see if it works four you or where it breaks.

===
period size: 64 samples (1.333 ms, 256 bytes)
channels (capture): 196
bytes per sample: 4 (32 bit float)
immediate send, no pause or shutdown: no
multi-channel period size: 50176 bytes
message rate: 750.0 messages/s
message length: 51200 bytes
transfer length: 52398 bytes (4.2 % overhead)
/!\ receiver(s) must support message size > 32768
expected network data rate: 314388.0 kbit/s (39.30 MB/s)
/!\ high data rate (OK on localhost), use GigE
===
-> almost 40 MB/s of data moving on localhost, just in one direction.
That's a lot for "just" creating a common audio data domain for both devices.

Good Luck
Thomas

[1] https://github.com/7890/jack_tools/tree/master/audio_rxtx

On Tue, January 24, 2017 17:13, Jörg Müller wrote:

> Thanks for the information.
>
>
> The two MADI FX cards I'm using can be synced via word clock.
>
>
> However, the information needs to be unaltered, no resampling or any
> modification is allowed for our processing. What command line settings do
> I
> have to use to make sure of that? How can I use the official zita-ajbridge
>  or is there some documentation about it?
>
> 2017-01-24 16:36 GMT+01:00 Thomas Brand <[hidden email]>:
>
>
>> On Tue, January 24, 2017 16:27, Thomas Brand wrote:
>>
>>> On Tue, January 24, 2017 09:22, Ralf Mardorf wrote:
>>>
>>>
>>>> On Tue, 24 Jan 2017 07:41:50 +0000, [hidden email] wrote:
>>>>
>>>>
>>>>
>>>>> On Mon, Jan 23, 2017 at 05:35:57PM +0000, Fons Adriaensen wrote:
>>>>>
>>>>>
>>>>>
>>>>>> With the official zita-ajbridge you can disable resampling.
>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>> Good to know.
>>>>>
>>>>>
>>>>>
>>>>
>>>> Could this be used without word clock, by using a digital interface
>>>> to sync, e.g. S/PDIF? If so, assuming two identical cards would be
>>>> used, would sync be hat good, that each combination of IOs, e.g. a
>>>> channel from one card for the left and of the other card for the
>>>> right channel of a stereo signal, wouldn't cause phasing issues?
>>>>
>>> yes i think that works pretty good (drawing from earlier
>>> experiments). Using S/PDIF just for sync is a cheap alternative to a
>>> "real" wordclock.
>>> One of both interfaces will be slaved to the other and then both
>>> should run the same cycle at the same time (so theoretically no
>>> phasing issues). You just have to manage to shove forth and back the
>>> sample data from/to interface at good enough rate which isn't a
>>> particular problem on
>> localhost
>>> using UDP like zita does. If there is no resampling involved, this
>>> can work for many many channels without hitting the CPU too much.
>>>
>>> Greetings
>>> Thomas
>>>
>>>
>> Even if the interfaces are synchronized with S/PDIF, there will be some
>>  cycles (hopefully a static number) needed to shove data forth and
>> back. This needs to be considered to effectively get synchronous
>> behaviour for input or output to both interfaces.
>>
>>
>>
>> _______________________________________________
>> Jack-Devel mailing list
>> [hidden email]
>> http://lists.jackaudio.org/listinfo.cgi/jack-devel-jackaudio.org
>>
>>
>


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Re: Cannot combine audio devices with more than 64 channels

John Rigg-16
In reply to this post by Jörg Müller
On Tue, Jan 24, 2017 at 05:13:53PM +0100, Jörg Müller wrote:
> How can I use the official zita-ajbridge or is there some
> documentation about it?

man zita-ajbridge

Use zita-a2j for capture and zita-j2a for playback.
The -S option disables resampling.

Make sure you get latest version as earlier ones couldn't disable SRC:

http://kokkinizita.linuxaudio.org/downloads/zita-ajbridge-0.7.0.tar.bz2

John
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Re: Cannot combine audio devices with more than 64 channels

Fons Adriaensen-3
In reply to this post by Thomas Brand
On Tue, Jan 24, 2017 at 04:27:00PM +0100, Thomas Brand wrote:

> One of both interfaces will be slaved to the other and then both should
> run the same cycle at the same time (so theoretically no phasing issues).
> You just have to manage to shove forth and back the sample data from/to
> interface at good enough rate which isn't a particular problem on
> localhost using UDP like zita does. If there is no resampling involved,
> this can work for many many channels without hitting the CPU too much.

This is all as wrong as it can possibly be.

First, even when resampling is disabled there is no defined relation
between the period timing of Jack's main soundcard and the one added
with zita-ajbridge. Zita-ajbridge will detect the relative period
phases, and insert buffering so as to ensure that the difference in
latency is always the same.

Second, there is no UDP or anything network involved at all.

Ciao,

--
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)

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Re: Cannot combine audio devices with more than 64 channels

Fons Adriaensen-3
In reply to this post by Jörg Müller
On Tue, Jan 24, 2017 at 05:13:53PM +0100, Jörg Müller wrote:
 
> However, the information needs to be unaltered, no resampling or any
> modification is allowed for our processing. What command line settings do I
> have to use to make sure of that? How can I use the official zita-ajbridge
> or is there some documentation about it?

Use the -h option or man zita-a2j or man zita-j2a.

Without resampling the samples will not be modified, but the latency
of the two cards will not be the same (but the difference will be
repeatable).

Ciao,

--
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)

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Re: Cannot combine audio devices with more than 64 channels

Thomas Brand
In reply to this post by Fons Adriaensen-3
On Tue, January 24, 2017 20:41, Fons Adriaensen wrote:

> On Tue, Jan 24, 2017 at 04:27:00PM +0100, Thomas Brand wrote:
>
>
>> One of both interfaces will be slaved to the other and then both should
>>  run the same cycle at the same time (so theoretically no phasing
>> issues). You just have to manage to shove forth and back the sample data
>> from/to interface at good enough rate which isn't a particular problem
>> on localhost using UDP like zita does. If there is no resampling
>> involved, this can work for many many channels without hitting the CPU
>> too much.
>
> This is all as wrong as it can possibly be.
>
>
> First, even when resampling is disabled there is no defined relation
> between the period timing of Jack's main soundcard and the one added with
> zita-ajbridge. Zita-ajbridge will detect the relative period phases, and
> insert buffering so as to ensure that the difference in latency is always
> the same.
>
> Second, there is no UDP or anything network involved at all.
>

I wrongly mixed this up with zita-n2j/j2n while its about zita-a2j/j2a,
sorry for that.



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Re: Cannot combine audio devices with more than 64 channels

Chris Caudle
In reply to this post by Ralf Mardorf
On Tue, January 24, 2017 1:45 pm, Fons Adriaensen wrote:
> Without resampling the samples will not be modified, but the latency of
the two cards will not be the same (but the difference will be
repeatable).

Meaning some output ports will have more latency than others?  I don't
think jack has a way to notify the server that some backends have
different latency so that it can be reported to applications like Ardour
that adjust for latency.

--
Chris Caudle




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Re: Cannot combine audio devices with more than 64 channels

Jörg Müller
That would be a problem. We want to use these cards for wave field synthesis. They can have WordClock connections, so they are precisely synced. If jack would somehow add latency to the second card, the results won't work.

2017-01-25 2:59 GMT+01:00 Chris Caudle <[hidden email]>:
On Tue, January 24, 2017 1:45 pm, Fons Adriaensen wrote:
> Without resampling the samples will not be modified, but the latency of
the two cards will not be the same (but the difference will be
repeatable).

Meaning some output ports will have more latency than others?  I don't
think jack has a way to notify the server that some backends have
different latency so that it can be reported to applications like Ardour
that adjust for latency.

--
Chris Caudle




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Re: Cannot combine audio devices with more than 64 channels

Jörg Müller
In reply to this post by Fons Adriaensen-3
Thanks for your answer!

In your first mail, you write
> Zita-ajbridge will detect the relative period phases, and insert buffering so as to ensure that the difference in latency is always the same.
That sounds to me like the two cards get additional latency, but it will be exactly the same so their outputs remain in correct phase.

However, in your second mail, you write:
> Without resampling the samples will not be modified, but the latency of the two cards will not be the same (but the difference will be repeatable).
That sounds like the very opposite.

So will there be any additional latency when using zita? Can it be configured to be the same for all cards? Could the latency be configured to something minimal like 64 samples?


2017-01-24 20:45 GMT+01:00 Fons Adriaensen <[hidden email]>:
On Tue, Jan 24, 2017 at 05:13:53PM +0100, Jörg Müller wrote:

> However, the information needs to be unaltered, no resampling or any
> modification is allowed for our processing. What command line settings do I
> have to use to make sure of that? How can I use the official zita-ajbridge
> or is there some documentation about it?

Use the -h option or man zita-a2j or man zita-j2a.

Without resampling the samples will not be modified, but the latency
of the two cards will not be the same (but the difference will be
repeatable).

Ciao,

--
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)

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Re: Cannot combine audio devices with more than 64 channels

Jörg Müller
In reply to this post by Adrian Knoth
So I've been trying to send the problem to the alsa mailing list. I successfully registered there and get their mails delivered. However, the mails I sent to their mailing list remain unanswered and they cannot be seen in the archive of January 2017. Something seems to block my mails. What could it be? Could someone of you maybe forward the initial mail to their list?

2017-01-23 14:31 GMT+01:00 Adrian Knoth <[hidden email]>:
On Mon, Jan 23, 2017 at 01:57:09PM +0100, Jörg Müller wrote:

> I need to combine two HDSPe MADI FX cards to one virtual device. I have a

Fancy!

> For combining the MADI FX cards to one virtual device, I created an
> .asoundrc with 194 inputs for each card. When I start that virtual device

You can (and probably should) ask this again on the ALSA ML if you want
to do it via ALSA virtual devices. The other option is to have jack open
both cards:

   https://github.com/jackaudio/jack1/commit/40b01cc3691f25840b40432ec16fd312dbfbd365

I have never tried it, but in theory, you just add -A 2ndALSAname to the
command line and be done.

Not sure how much documentation is available for this on the internet,
but might be worth trying.


Cheers

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