On Sat, Dec 12, 2015 at 9:07 AM, Len Ovens <len(a)ovenwerks.net> wrote: This is incorrect. There is no API to change the SR of a running JACK instance. You can, of course, stop and restart JACK with a new SR, but that's not really what is being requested.

There *is* a way to change the sample rate of a running jack instance, but it's jack2 + dbus only. This is by using switch-master. You can change your entire JACK setup on the fly, including the selected soundcard. If you keep the same jack settings but only change the sample rate you'll effectly change the jack sample-rate on the fly. Note that JACK2 will call the buffersize and samplerate callbacks to inform clients of the change. But yeah, since this was something that JACK was not designed for, *all* major clients that exist right now do not support this.

On 12.12.2015 16:07, Paul Davis wrote: The system ports do go away and appear again, but all existing clients stay unchanged. As far as clients are concerned, jack never stopped.

On Sat, 12 Dec 2015 16:12:02 +0100 Filipe Coelho <falktx(a)gmail.com> wrote: So then all the clients have to stop, re-calculate timings buffers etc. then restart. How do you syncronise that? This seems like a sledgehammer to crack a nut, and apart from the OP's specific situation I can't think of a use-case that's critical enough to justify it. Also, as the OP is wanting this for DJ work wouldn't it be simpler to accept some additional latency and pass the offending tracks through a 'live' samplerate converter? -- Will J Godfrey http://www.musically.me.uk Say you have a poem and I have a tune. Exchange them and we can both have a poem, a tune, and a song.

Am 12.12.2015 um 16:54 schrieb Len Ovens: Thanks to the group for the answers. I think y have to explain my intentions: I am a Tango-DJ in germany. Now there is a strong trend to play tracks in the highest possible quality. There are sellers of new masterd peaces from shellacs in flac 24/96k sampled. http://www.tangotunes.com/orquestas/juan-d-arienzo.html?tt_filtertype=509 The DJ's like to play it bit-precise. They invest a lot of money to do so. They use Windos or OSX apple. As a linux enthusiast I would like to say to them, it is possible in Linux also. They do not want to discuss if sombody can hear it, they only will do it with the highest quality as possible. Most of them have tracks from CD rippt with 44,1k and Flacs 24/96k. At the moment I think its not possible with Jack and Jack-Clients like Calf-studio-gear. I did it with master-switch, it works but calf-studio-gear-euqualizer will be stoppt and I have to start it new. It takes to long in Tango Event to switch from 44,1k to 96k and 15min. later back to 44.1k. It is sad, but what can I do. PeterW

On Sun, 13 Dec 2015, Wenger Peter wrote: Highest quality for a 24/96k file is resample to 48k and play. Pretty much you are less likely to have bad artifacts from an audio system trying to reproduce higher than 20k audio. 24/96k is just PR no real increase in quality, just bigger numbers to impress the easily impressed... and more money in the pockets of those who sell equipment. Bit precise? what is that? It sounds impressive, but meaningless. in fact at 96k it is probably false because most audio cards no matter what SR they use still low pass the i/o to 20k. So a 96k file is not bit precise in the way that a 8bit/32k file can be. So your DJs are in effect using false advertising to increase profits. No they only want to make money. It is important to be seen as using the latest, greatest, stuff. If it is or not. The best way for you to join in (with this maddness :) ) is to upsample your 44.1k stuff (on the fly) to 96k and be happy. If win/OSx allow you to mix from one tune to the next (fade from song to song) then they are internally already resampling to allow this to happen. The only other way to fade from one song to the next with no resample is to use more than one audio interface and mix externally. This would allow mixing one track at 96k with another at 44.1k. This is true no matter what OS you use. As I say, I think you will find that many of these DJs are upsampling the 44.1k material if they are aware of it or not. (they may even be downsampling the 96k material unaware) Thats life... -- Len Ovens www.ovenwerks.net

On Sun, 13 Dec 2015 15:53:25 +0100, Wenger Peter wrote: Hi, as others already pointed out, you could do the same as the Mac and Windows DJs do, you could resample. Do you play the music in a dance hall, cafe, bar ...? If the music is played in a dance hall, cafe, bar ..., I wouldn't care about any scientific and/or psychical arguments about differences that exist or don't exists regarding (re)sampling quality. The sound is much more biased by the PA, room acoustics and ambient noise. If there should be audible differences between the sound sources in some good circumstances, they unlikely will be audible when played in a dance hall, cafe, bar ... . Regards, Ralf

On Sun, 13 Dec 2015 20:39:18 +0100 Robin Gareus <robin(a)gareus.org> wrote: Actually you can go one better... "Hey. I've got this gizmo that turns 44.1k into 96K. It's the mutt's nuts!" Let them fill in what gaps they like. You've just told the unvarnished truth :) -- Will J Godfrey http://www.musically.me.uk Say you have a poem and I have a tune. Exchange them and we can both have a poem, a tune, and a song.

On 13/12/2015 20:39, Robin Gareus wrote: And if you want to be a bit 'artistic' - apply a some hi frequency EQ boosting (pick above 5 kHz): in a smoky room with PA it will add a bit of 'brilliance' and you can say "can't you hear how crisp the bit-accurate sound is?" :-)

Len Ovens &lt;len(a)ovenwerks.net&gt; writes: Not for audio, but having JACK run at 96000 Hz is quite useful for low-cost software-defined radio... -- Adam Sampson &lt;ats(a)offog.org&gt; <http://offog.org/>

Will Godfrey &lt;willgodfrey(a)musically.me.uk&gt; writes: A common architecture for an SDR receiver uses a quadrature sampling detector: you mix the incoming radio signal with a pair of clocks 90 degrees apart, and sample the two resulting signals (I/Q). Some DSP then allows you to "tune in" any signal within the bandwidth that you've sampled -- if my detector's clock is running at 7 MHz and I'm sampling I/Q at 48 kHz, my SDR receiver can demodulate any signals between 6.976 MHz and 7.024 MHz (and can receive several signals simultaneously provided they're within that range of frequencies). So the advantage of sampling at a higher rate is that you can receive a wider chunk of the radio spectrum without having to retune. High-end SDR receivers have a dedicated ADC that can run at a high sampling rate, but cheap radios like the SoftRock kits (http://wb5rvz.com/sdr/) use a regular soundcard. I'm using a SoftRock RXTX with dttsp (and sdr-shell/softrig/fldigi), which uses JACK for audio IO. Running JACK at 96 kHz for this means that I can see most of the 30m amateur band at the same time... Cheers, -- Adam Sampson &lt;ats(a)offog.org&gt; <http://offog.org/>

On Mon, Dec 14, 2015 at 09:09:16AM -0800, Len Ovens wrote: Depends on how it works. Just like DA converters can be upsampling, an AD converter could actually sample at a much higher rate than the nominal one and then downsample digitally - for the same reason as the DA: it allows to use a simpler analog antialising filter. With such an architecture, switching to 96 kHz would provide the full, near FS/2, bandwidth even if that isn't very useful for audio. No, but you can have a very low level signal which you want to demodulate very near (in frequency) to a high level unwanted signal. In other words, you need high dynamic range *before* the demodulator, even if the demodulated signal can have a low S/N ratio. High linear dynamic range is one of the things you pay for with the more expensive receivers. 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)

On Tue, 15 Dec 2015, Adam Sampson wrote: Thankyou for that page. I would imagine some of that applies to my delta 66 as well. -- Len Ovens www.ovenwerks.net

On Mon, Dec 14, 2015 at 04:46:47PM +0000, Adam Sampson wrote: So your soundcard actually provides ~48 Khz of BW when running at 96 kHz sample rate ? 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)