re Zoom R16

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Re: re Zoom R16

Set Hallstrom
Hi!

>> Plus, I don't know how to change the sample rate on the device.
>> There is no control exposed for alsamixer, and I cannot change it
>> from the device itself. When I tested it, as soon I started jackd
>> the device changed the sample rate to 96kHz (when I was trying 48kHz
>> from the jackd command, and the device was at 44.1kHz before), and
>> (q)jack(ctl)d stopped like waiting for the device.

I'm 99% sure you can only switch the bitrate on the r16 from 16-bits to
24-bits. The samplerate is always set to 44.1kHz.

Perhaps configuring jack to use 44.1 will solve the problem?

Just a thaught, i still havn't got my thumb out to try Clemens Patch :/

Have a good awoken time y'all!

*set


On 2013-10-21 22:57, Natanael Olaiz wrote:

> El 10/18/2013 10:04 PM, Natanael Olaiz escribió:
>> El 10/18/2013 08:27 AM, Clemens Ladisch escribió:
>>> Natanael Olaiz wrote:
>>>
>>>> [...]
>>>> On the other hand, audio doesn't seem to works straightforward. The
>>>> device is seen, but I was not able to make it work with jack nor aplay.
>>>
>>> What happens?
>>>
>> I think it had some trouble with the sample rate or some of the other
>> defined arguments, but I still didn't had many time to try different
>> combinations...
>>
>> Plus, I don't know how to change the sample rate on the device. There is
>> no control exposed for alsamixer, and I cannot change it from the device
>> itself. When I tested it, as soon I started jackd the device changed the
>> sample rate to 96kHz (when I was trying 48kHz from the jackd command,
>> and the device was at 44.1kHz before), and (q)jack(ctl)d stopped like
>> waiting for the device.
>>
>> ...
>>
>> But after my previous mail I tried again with alsa_in on a running 48kHz
>> jackd server (Focusrite Saffire), and it worked!!!! I was able to use
>> all the 8 input channels!!! (my impression was an incredibly -for the
>> price...- clean sound!)
>>
>> I tried then adding an alsa_out (at the same time than alsa_in), but it
>> said the device was already in use (I have a H2n and as far I can
>> remember I can execute both together...)
> I'm using the R16 as MIDI controller [R16 -> a2jmidid (alsa midi)->
> (jack midi) ardour3] and audio capture [R16 -> alsa_in (alsa device with
> 8 input channels) -> ffado jack host].
>
> It seems to work quite well when is running, but there is something not
> fine:
>
>  - it takes a long time to the device to appears as an alsa device since
> it is detected from the kernel module
>
>  - at least when it is not used (but I think it can occurs even when
> streaming) it seems to block ALSA for moments. Meanwhile alsa seems is
> unresponsive. Trying to start qjackctl (just open the frontend, not even
> starting jack) opened an unresponsive widget. As soon I disconnected the
> USB from the R16, the buttons and the control of the widget appeared.
>
> - I was with the above described chain (jack + R16 alsa_in) , and all
> the audio devices suddenly hanged. I saw in the R16 the sample rate at
> 96kHz (when it was -as JACK- at 48kHz)
>
> A dmesg shows this :
>
>> [  268.877864] 3:1:1: cannot get freq at ep 0x3
>> [  272.489903] firewire_core 0000:07:00.0: created device fw1: GUID
>> 00130e0100060dce, S400
>> [  273.879240] 3:1:1: cannot set freq 44100 to ep 0x3
>> [  278.903419] 3:1:1: cannot get freq at ep 0x3
>> [  283.904662] 3:1:1: cannot set freq 44100 to ep 0x3
>> [  286.633916] 3:1:1: cannot get freq at ep 0x3
>> [  286.638154] 3:1:1: cannot set freq 44100 to ep 0x3
>> [  286.671291] 3:1:1: usb_set_interface failed (-71)
>> [  286.675542] 3:1:1: usb_set_interface failed (-71)
>> [  286.679792] 3:1:1: usb_set_interface failed (-71)
>>  "" * 20
>> [  286.799821] 3:1:1: usb_set_interface failed (-71)
>> [  286.830497] usb 1-1.4: USB disconnect, device number 3
>> [  304.980599] hda-intel: IRQ timing workaround is activated for card
>> #0. Suggest a bigger bdl_pos_adj.
>> [  305.930727] usb 1-1.4: new high-speed USB device number 4 using
>> ehci-pci
>> [  306.022714] usb 1-1.4: config 1 interface 3 altsetting 0 bulk
>> endpoint 0x1 has invalid maxpacket 64
>> [  306.022717] usb 1-1.4: config 1 interface 3 altsetting 0 bulk
>> endpoint 0x82 has invalid maxpacket 64
>> [  306.023508] usb 1-1.4: New USB device found, idVendor=1686,
>> idProduct=00dd
>> [  306.023513] usb 1-1.4: New USB device strings: Mfr=1, Product=2,
>> SerialNumber=3
>> [  306.023515] usb 1-1.4: Product: R16
>> [  306.023517] usb 1-1.4: Manufacturer: ZOOM Corporation
>> [  306.023519] usb 1-1.4: SerialNumber: 0
>> [  311.025093] 4:1:1: cannot get freq at ep 0x3
>> [  321.027467] 4:2:1: cannot get freq at ep 0x84
>> [  331.090900] 4:1:1: cannot get freq at ep 0x3
>> [  336.092152] 4:1:1: cannot set freq 44100 to ep 0x3
>> [  341.115367] 4:1:1: cannot get freq at ep 0x3
>> [  346.116557] 4:1:1: cannot get freq at ep 0x3
>> [  356.273029] 4:1:1: cannot get freq at ep 0x3
>> [  358.352531] 4:2:1: usb_set_interface failed (-110)
>> [  820.268089] 4:1:1: cannot get freq at ep 0x3
>> [  825.269267] 4:1:1: cannot set freq 44100 to ep 0x3
>
> Any hint?
>
>
> Thanks in advance,
> Natanael.
>
>
>>
>> Soon I'll do more controlled tests and I'll update you. But it look so
>> good!! :-)
>>
>> Thanks a lot!!!
>>
>> Natanael.
>>> Regards,
>>> Clemens
>>>
>

--
Set Hallstrom
AKA Sakrecoer
http://sakrecoer.com


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Re: re Zoom R16

Jan Depner
On Wed, 2013-10-23 at 10:02 +0200, Set Hallstrom wrote:

> Hi!
>
> >> Plus, I don't know how to change the sample rate on the device.
> >> There is no control exposed for alsamixer, and I cannot change it
> >> from the device itself. When I tested it, as soon I started jackd
> >> the device changed the sample rate to 96kHz (when I was trying 48kHz
> >> from the jackd command, and the device was at 44.1kHz before), and
> >> (q)jack(ctl)d stopped like waiting for the device.
>
> I'm 99% sure you can only switch the bitrate on the r16 from 16-bits to
> 24-bits. The samplerate is always set to 44.1kHz.
>

The bitrate is either 16 or 24.  The samplerate is only restricted to
44.1KHz as a standalone unit.  When used as a computer sound interface
you are supposed to be able to set it at the usual places between
44.1KHz and 96KHz.

On a related yet unrelated note, I have two of these and use them in a
master/slave configuration to record gigs using direct outs from our
board.  I did have to make a couple of 8 port, 50dB L-pad boxes though
because the inputs on the R-16 are expecting line input at consumer
level (-10dBV) while the board puts out pro level (+4dBu).  If anyone is
interested in the L-pad boxes I've posted a tutorial on my web page:

https://googledrive.com/host/0Bw2lTTCOj0kBeGdrbGVqNXJaODA/ZR16.html

I've since found that I should have made them 40dB so now I have to swap
out the 33K ohm resistors for 10K ohm.  Oh well ;-)


> Perhaps configuring jack to use 44.1 will solve the problem?
>
> Just a thaught, i still havn't got my thumb out to try Clemens Patch :/
>
> Have a good awoken time y'all!
>
> *set
>
>
> On 2013-10-21 22:57, Natanael Olaiz wrote:
> > El 10/18/2013 10:04 PM, Natanael Olaiz escribió:
> >> El 10/18/2013 08:27 AM, Clemens Ladisch escribió:
> >>> Natanael Olaiz wrote:
> >>>
> >>>> [...]
> >>>> On the other hand, audio doesn't seem to works straightforward. The
> >>>> device is seen, but I was not able to make it work with jack nor aplay.
> >>>
> >>> What happens?
> >>>
> >> I think it had some trouble with the sample rate or some of the other
> >> defined arguments, but I still didn't had many time to try different
> >> combinations...
> >>
> >> Plus, I don't know how to change the sample rate on the device. There is
> >> no control exposed for alsamixer, and I cannot change it from the device
> >> itself. When I tested it, as soon I started jackd the device changed the
> >> sample rate to 96kHz (when I was trying 48kHz from the jackd command,
> >> and the device was at 44.1kHz before), and (q)jack(ctl)d stopped like
> >> waiting for the device.
> >>
> >> ...
> >>
> >> But after my previous mail I tried again with alsa_in on a running 48kHz
> >> jackd server (Focusrite Saffire), and it worked!!!! I was able to use
> >> all the 8 input channels!!! (my impression was an incredibly -for the
> >> price...- clean sound!)
> >>
> >> I tried then adding an alsa_out (at the same time than alsa_in), but it
> >> said the device was already in use (I have a H2n and as far I can
> >> remember I can execute both together...)
> > I'm using the R16 as MIDI controller [R16 -> a2jmidid (alsa midi)->
> > (jack midi) ardour3] and audio capture [R16 -> alsa_in (alsa device with
> > 8 input channels) -> ffado jack host].
> >
> > It seems to work quite well when is running, but there is something not
> > fine:
> >
> >  - it takes a long time to the device to appears as an alsa device since
> > it is detected from the kernel module
> >
> >  - at least when it is not used (but I think it can occurs even when
> > streaming) it seems to block ALSA for moments. Meanwhile alsa seems is
> > unresponsive. Trying to start qjackctl (just open the frontend, not even
> > starting jack) opened an unresponsive widget. As soon I disconnected the
> > USB from the R16, the buttons and the control of the widget appeared.
> >
> > - I was with the above described chain (jack + R16 alsa_in) , and all
> > the audio devices suddenly hanged. I saw in the R16 the sample rate at
> > 96kHz (when it was -as JACK- at 48kHz)
> >
> > A dmesg shows this :
> >
> >> [  268.877864] 3:1:1: cannot get freq at ep 0x3
> >> [  272.489903] firewire_core 0000:07:00.0: created device fw1: GUID
> >> 00130e0100060dce, S400
> >> [  273.879240] 3:1:1: cannot set freq 44100 to ep 0x3
> >> [  278.903419] 3:1:1: cannot get freq at ep 0x3
> >> [  283.904662] 3:1:1: cannot set freq 44100 to ep 0x3
> >> [  286.633916] 3:1:1: cannot get freq at ep 0x3
> >> [  286.638154] 3:1:1: cannot set freq 44100 to ep 0x3
> >> [  286.671291] 3:1:1: usb_set_interface failed (-71)
> >> [  286.675542] 3:1:1: usb_set_interface failed (-71)
> >> [  286.679792] 3:1:1: usb_set_interface failed (-71)
> >>  "" * 20
> >> [  286.799821] 3:1:1: usb_set_interface failed (-71)
> >> [  286.830497] usb 1-1.4: USB disconnect, device number 3
> >> [  304.980599] hda-intel: IRQ timing workaround is activated for card
> >> #0. Suggest a bigger bdl_pos_adj.
> >> [  305.930727] usb 1-1.4: new high-speed USB device number 4 using
> >> ehci-pci
> >> [  306.022714] usb 1-1.4: config 1 interface 3 altsetting 0 bulk
> >> endpoint 0x1 has invalid maxpacket 64
> >> [  306.022717] usb 1-1.4: config 1 interface 3 altsetting 0 bulk
> >> endpoint 0x82 has invalid maxpacket 64
> >> [  306.023508] usb 1-1.4: New USB device found, idVendor=1686,
> >> idProduct=00dd
> >> [  306.023513] usb 1-1.4: New USB device strings: Mfr=1, Product=2,
> >> SerialNumber=3
> >> [  306.023515] usb 1-1.4: Product: R16
> >> [  306.023517] usb 1-1.4: Manufacturer: ZOOM Corporation
> >> [  306.023519] usb 1-1.4: SerialNumber: 0
> >> [  311.025093] 4:1:1: cannot get freq at ep 0x3
> >> [  321.027467] 4:2:1: cannot get freq at ep 0x84
> >> [  331.090900] 4:1:1: cannot get freq at ep 0x3
> >> [  336.092152] 4:1:1: cannot set freq 44100 to ep 0x3
> >> [  341.115367] 4:1:1: cannot get freq at ep 0x3
> >> [  346.116557] 4:1:1: cannot get freq at ep 0x3
> >> [  356.273029] 4:1:1: cannot get freq at ep 0x3
> >> [  358.352531] 4:2:1: usb_set_interface failed (-110)
> >> [  820.268089] 4:1:1: cannot get freq at ep 0x3
> >> [  825.269267] 4:1:1: cannot set freq 44100 to ep 0x3
> >
> > Any hint?
> >
> >
> > Thanks in advance,
> > Natanael.
> >
> >
> >>
> >> Soon I'll do more controlled tests and I'll update you. But it look so
> >> good!! :-)
> >>
> >> Thanks a lot!!!
> >>
> >> Natanael.
> >>> Regards,
> >>> Clemens
> >>>
> >
>


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Re: re Zoom R16

jmancine
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This post was updated on .
In reply to this post by Set Hallstrom
Well, I guess my first attempt was pure luck because I can not make it work again.  Grrrr

Every time I try to start LADI or Jack, I get this error:

Could not connect to JACK server as client.
- Overall operation failed.
Unable to connect to server
Please check the messages window for more information.


And the Jack messages shows this:

09:52:59.093 JACK is starting...
09:52:59.095 /usr/bin/jackd -dalsa -dhw:R16,0 -r96000 -p512 -n3 -C -i8
Cannot connect to server socket err = No such file or directory
Cannot connect to server request channel
jack server is not running or cannot be started
no message buffer overruns
no message buffer overruns
09:52:59.146 JACK was started with PID=4138.
no message buffer overruns
jackdmp 1.9.10
Copyright 2001-2005 Paul Davis and others.
Copyright 2004-2013 Grame.
jackdmp comes with ABSOLUTELY NO WARRANTY
This is free software, and you are welcome to redistribute it
under certain conditions; see the file COPYING for details
JACK server starting in realtime mode with priority 10
audio_reservation_init
Acquire audio card Audio0
creating alsa driver ... -|hw:R16,0|512|3|96000|8|0|nomon|swmeter|-|32bit
configuring for 96000Hz, period = 512 frames (5.3 ms), buffer = 3 periods
ALSA: final selected sample format for capture: 32bit integer little-endian
ALSA: use 3 periods for capture
09:53:01.164 Could not connect to JACK server as client. - Overall operation failed. - Unable to connect to server. Please check the messages window for more info.
Cannot connect to server socket err = No such file or directory
Cannot connect to server request channel
jack server is not running or cannot be started
ALSA: prepare error for capture on "hw:R16,0" (Connection timed out)
Cannot start driver
JackServer::Start() failed with -1
Failed to start server
Released audio card Audio0
audio_reservation_finish
09:53:09.436 JACK was stopped with exit status=255.


I have tried every combination of settings possible, and it is always the same result.   If I change the sample rate, when I start Jack it will change the sample rate on the R16 and all 8 channel meters flash for a second (showing that they are being recognized) before the error terminates Jack.

I even tried a fresh install of Ubuntu 13.10, modified and installed the kernel, and have the same result.

Here is what I think the problem is:

creating alsa driver ... -|hw:R16,0|512|3|96000|8|0|nomon|swmeter|-|32bit
configuring for 96000Hz, period = 512 frames (5.3 ms), buffer = 3 periods
ALSA: final selected sample format for capture: 32bit integer little-endian

Jack/Alsa are trying to intitialize the R16 at 32-bits, when we know that it is 24. Other devices I have initialize properly with their given bit-rates.  So, my question is:  How do I tell Jack to initialize the R16 at 24 bit instead of 32??

Also, I did try the "Force 16 bit" option in Jack just to see what happens, and it STILL tries to set it at 32 bit and results in the same error.

creating alsa driver ... hw:R16|hw:R16|512|3|96000|0|0|nomon|swmeter|-|16bit
configuring for 96000Hz, period = 512 frames (5.3 ms), buffer = 3 periods
ALSA: final selected sample format for capture: 32bit integer little-endian

Again, when I plug in other devices, the "final selected sample format for capture" is properly set.   If there were a "Force 24-bit" option, I have a feeling we would be in business!

UPDATE:  Check this link out:  http://linux-audio.com/jack/.  Under the JACK Specific Options section, it says that Jack only  supports 16-bit and 24-packed-in-32 formats...NOT true 24 bit.   Hmmmm.  This would seem to prohibit the device from running, but I had it working yesterday!  There has to be a way.

Thanks,
Jason
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Re: re Zoom R16

jmancine
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This post was updated on .
In reply to this post by Clemens Ladisch
Clemens, I have been tinkering with adding an .data/audioformat function to this quirk...  so far no luck with copying some of the others already in the quirk table.  Any tips?  Assuming that the device operates at 24 bit integer, would the correct .formats setting be "SNDRV_PCM_FMTBIT_S24_LE" ?

Thanks,
Jason


P.S.  I have some development background, but am new to kernel compiling.  Is there a way to compile only the sound section of the kernel, or do I need to recompile the whole thing each time?  If the latter, can anyone recommend a lightweight system to test on (the 83MB debian kernels are taking 3 hours to compile).

Thanks

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Re: re Zoom R16

jmancine
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GOT IT!   THANK YOU everyone for the work.

For the record, this is running on AVLinux 6.0.1 on the plain 3.6.11 kernel straight from www.kernel.org

As I suspected, ALSA was not able to read the correct bitrate.  With an addition to Clemens' code, we can tell ALSA that the R16 is running at 24 bits, and also explicitly tell it what sampling rates are supported.

PLEASE NOTE:  You must set it for "capture only" in JACK...if you try to run duplex or playback, it will produce runaway xruns.   (Duplex WILL work if you have a second device for playback, just not with the R16 as the playback device).

Here is my entry into quirks-table.h that has made the R16 work (FINALLY!!!) in Linux!!  

{
        /* ZOOM R16 in USB 2.0 mode */
        USB_DEVICE(0x1686, 0x00dd),
        .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
                .ifnum = QUIRK_ANY_INTERFACE,
                .type = QUIRK_COMPOSITE,
                .data = (const struct snd_usb_audio_quirk[]) {
                        {
                                .ifnum = 0,
                                .type = QUIRK_IGNORE_INTERFACE
                        },
                        {
                                .ifnum = 1,
                                .type = QUIRK_AUDIO_STANDARD_INTERFACE
                        },
                        {
                                .ifnum = 2,
                                .type = QUIRK_AUDIO_STANDARD_INTERFACE
                        },
                        {
                                .ifnum = 3,
                                .type = QUIRK_MIDI_STANDARD_INTERFACE
                        },

{
                                .ifnum = 4,
                                .type = QUIRK_AUDIO_FIXED_ENDPOINT,
                                .data = & (const struct audioformat) {
                                        .formats = SNDRV_PCM_FMTBIT_S24_LE,
                                        .channels = 8,
                                        .iface = 1,
                                        .altsetting = 1,
                                        .altset_idx = 1,
                                        .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE,
                                        .rates = SNDRV_PCM_RATE_44100 |
                                                 SNDRV_PCM_RATE_48000 |
                                                 SNDRV_PCM_RATE_88200 |
                                                 SNDRV_PCM_RATE_96000,
                                        .rate_min = 44100,
                                        .rate_max = 96000,
                                        .nr_rates = 4,
                                        .rate_table = (unsigned int[]) {
                                                        44100, 48000, 88200, 96000
                                        }
                                }
                        },



                        {
                                .ifnum = .1
                        },
             

                        }

        }

},


Notes:

1. Sample rate can not be changed without resetting the device.  For example, if you start jack at 48000, stop jack, change settings to 96000 and restart jack, the R16 will not change.  

2. Jack will start for the R16 with completely non-working configurations like mismatched sample rates, ridiculously low latencies, etc.  Doesn't do this for any other devices I have.

3. No control surface functions seem to work with this configuration.


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Re: re Zoom R16

jmancine
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In reply to this post by Set Hallstrom
Has anyone had success getting the control surface function to work?  I have tried it in Ardour 2 and Ardour 3 and  I can connect the midi ports using a2j (or ALSA).  I have configured Ardour to use Mackie Control...also tried Generic Midi.   Nothing seems to work.  

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Re: re Zoom R16

Atte-4
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Re: re Zoom R16

Jan Depner
On Fri, 2013-11-08 at 14:36 +0100, Atte wrote:

> On 10/23/2013 02:18 PM, Jan Depner wrote:
>
> > On a related yet unrelated note, I have two of these
>
> I might be interested in getting one or two of these, but from reading
> this thread, it's not clear to me exactly what works and how robust it
> is under linux.
>
> Could someone please fill me in ?
>

I have no idea if the R16 works as an interface or as a controller under
Linux.  What I use mine for is to record my band live and then dump the
24 bit/44.1KHz audio to my hard drive.  I then import the WAV files to
Ardour and mix them.  With the exception of the line level problem that
I addressed here

https://googledrive.com/host/0Bw2lTTCOj0kBeGdrbGVqNXJaODA/ZR16.html

I have had no problem with them.  I did move up to 32GB SDHC cards so I
could record a 4 hour gig.


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Re: re Zoom R16

jmancine
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In reply to this post by Atte-4
With the previously mentioned procedures, it is working 100% for me as an audio interface.  I have not yet made it work as a control surface, but I believe it shouldn't be that difficult since it is recognized properly as a MIDI device.

Basically, it is a sub $400 device with 8 XLR inputs that also doubles as a field recorder that runs on AA batteries...pretty solid deal.  The preamps and trim controls are not *great*, but they are certainly adequate for most needs.

As for performance, I did a test in Ardour 2 the other day and ran 8 simultaneous tracks of recording at 48KHz for 2 hours, and then recorded another 8 tracks over those (while playing back through another device)...no xruns, and no problems whatsoever.   This is on an old AMD Athlon X2 system running 32-bit AVLinux 6.0.1 -- certainly no powerhouse!




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Re: re Zoom R16

jmancine
In reply to this post by Set Hallstrom
Control surface is working in Ardour 3 using A2JMIDID to connect the ports.  For Ardour 2, you also need to connect the MCU and SEQ ports to the R16 MIDI out.  

R16 is now fully functional in linux as a control surface and recording interface!  
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Re: re Zoom R16

Atte-4
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Re: re Zoom R16

Atte-4
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Re: re Zoom R16

jmancine

I guess most of my posts never hit the mailing list!  Wondered why no one was interested.  Goto the nabble link at the bottom of the email for extensive info.

On Nov 16, 2013 3:23 AM, "Atte-3 [via Linux Audio]" <[hidden email]> wrote:
On 11/15/2013 08:25 PM, jmancine wrote:
> Control surface is working in Ardour 3 using A2JMIDID to connect the ports.
> For Ardour 2, you also need to connect the MCU and SEQ ports to the R16 MIDI
> out.
>
> R16 is now fully functional in linux as a control surface and recording
> interface!
>

Could you list what you had to do? Patching which kernel, etc...

--
Atte

http://atte.dk   http://modlys.dk
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Re: re Zoom R16

Atte-4
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Re: re Zoom R16

jmancine
I can't imagine many people wanting to use an 8 channel input device for playback, so I think it is safe to consider it "working".  :)

It may very well work for playback... I haven't tested it other than the first time I started JACK with default settings after recompiling the kernel.  I am not sure why you would want to be playing back through the R16 anyway... was it was even designed to do this?   Even if it is capable of duplex playback, it would seem counterproductive to try and force it do so while it is recording 8 channels of audio.

That said, if there is interest in the R16 as a playback device I can try to get it running...



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Re: re Zoom R16

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Re: re Zoom R16

jmancine
This post was updated on .
I guess I was under the assumption that the standard setup for most linux users includes a playback device... not playing back audio through the recording interface.   I don't think the R16 was intended to decode/playback a stereo 24/96 stream while recording 8 individual 24/96 streams... especially through a USB 2.0 connection.

So, I may be able to verify if it works, but you would probably find it better to use a separate playback device for monitoring when it comes to the R16, and DAW recording in general.

UPDATE:  So, JACK seems to accept the R16 as a playback device... but I can't figure out a way to actually hear the audio.  It really isn't set up to monitor audio from the computer (i.e. like a "sound card").  The headphone jack on the R16 only monitors the input channels and I don't know a way to mix in the signal from the playback as there is no way to control the fader functions when it is in recording interface mode.

I guess I need to research how it is *intended* to work in Windows and OSX... i.e., does it allow full duplex, and if so how does the user control the output?

UPDATE 2:  The specs say that it IS capable of duplex playback, and that the master fader's balance control will fade between the recording signal and the playback signal.  I will fiddle with it and see if I can make it work.  The button above the Master fader toggles between REC/SOLO/MUTE... not sure how this factors in.  

Seems much more complicated than the setup where your speakers/headphones are plugged into a playback device and you can swap out recording devices on-the-fly... but I will see if I can make it work, nonetheless.



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Re: re Zoom R16

michael noble-2

On Mon, Nov 18, 2013 at 12:58 AM, jmancine <[hidden email]> wrote:
but you would probably find it
better to use a separate playback device for monitoring when it comes to the
R16, and DAW recording in general.

That seems to contradict even my admittedly basic understanding of how digital audio devices work. Mixing unsynced devices for simultaneous I/O is asking for trouble. JACK has no inbuilt capacity for synchronizing devices, so you will get clock drift and as far as I know potential sample misalignment between what you are listening to and what you are recording. Using a single device with the same clock for both input and output is always preferable, followed by two devices with some kind of hard sync if possible, followed by an aggregate driver level (ALSA in Linux) device only if the other options are unavailable. I've probably made a mess of explaining this, but there are multiple articles on the web about the importance of clock sync if you want to read up on it.

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Re: re Zoom R16

Ralf Mardorf
On Mon, 2013-11-18 at 14:53 +0800, michael noble wrote:

>
> On Mon, Nov 18, 2013 at 12:58 AM, jmancine <[hidden email]> wrote:
>         but you would probably find it
>         better to use a separate playback device for monitoring when
>         it comes to the
>         R16, and DAW recording in general.
>
>
> That seems to contradict even my admittedly basic understanding of how
> digital audio devices work. Mixing unsynced devices for simultaneous
> I/O is asking for trouble. JACK has no inbuilt capacity for
> synchronizing devices, so you will get clock drift and as far as I
> know potential sample misalignment between what you are listening to
> and what you are recording. Using a single device with the same clock
> for both input and output is always preferable, followed by two
> devices with some kind of hard sync if possible, followed by an
> aggregate driver level (ALSA in Linux) device only if the other
> options are unavailable. I've probably made a mess of explaining this,
> but there are multiple articles on the web about the importance of
> clock sync if you want to read up on it.

Professional gear can be synced by hardware and devices that don't have
a sync option can be synced by software. I don't remember the name, at
least this seems to do the job using libsamplerate:
http://jackaudio.org/multiple_devices

Wasn't there an app written by Fons too?

Regards,
Ralf


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Re: re Zoom R16

jmancine

Dont want to stray too far off here, but if that were "asking for trouble" I'd have been out of business many years ago.  The sync is fine, one clock becomes master.

I have been recording and playing back on separate devices for a long time... As long as various softwares/platforms have supported it, but I am talking back in the days of cramming 3 or 4 soundblasters into PCI slots. 

I record generally 16 to 32 tracks, mostly overdubs.  This year I did a 90 minute film score in two single Ardour sessions with literally thousands of takes and +/- 1 frame accuracy

In other words, it is a non-issue.




On Nov 18, 2013 7:13 AM, "Ralf Mardorf [via Linux Audio]" <[hidden email]> wrote:
On Mon, 2013-11-18 at 14:53 +0800, michael noble wrote:

>
> On Mon, Nov 18, 2013 at 12:58 AM, jmancine <[hidden email]> wrote:
>         but you would probably find it
>         better to use a separate playback device for monitoring when
>         it comes to the
>         R16, and DAW recording in general.
>
>
> That seems to contradict even my admittedly basic understanding of how
> digital audio devices work. Mixing unsynced devices for simultaneous
> I/O is asking for trouble. JACK has no inbuilt capacity for
> synchronizing devices, so you will get clock drift and as far as I
> know potential sample misalignment between what you are listening to
> and what you are recording. Using a single device with the same clock
> for both input and output is always preferable, followed by two
> devices with some kind of hard sync if possible, followed by an
> aggregate driver level (ALSA in Linux) device only if the other
> options are unavailable. I've probably made a mess of explaining this,
> but there are multiple articles on the web about the importance of
> clock sync if you want to read up on it.
Professional gear can be synced by hardware and devices that don't have
a sync option can be synced by software. I don't remember the name, at
least this seems to do the job using libsamplerate:
http://jackaudio.org/multiple_devices

Wasn't there an app written by Fons too?

Regards,
Ralf


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