search for: dacs

from the zaptel sample config: # "dacs" : The zaptel driver cross connects the channels starting at # the channel number listed at the end, after a colon # "dacsrbs" : The zaptel driver cross connects the channels starting at # the channel number listed at the end, after a colon and #...

The following is mostly for Nick, but perhaps there are others who might be interested in one part or another of the following... Nick, I think we're basically on the same page here. I'll first try to cover some of my basic ideas in general, then respond to your specific comments inline. For one thing, I have long believed that it is smart to keep analog audio circuits

I'm after the cheapest way to decode a flac stream with following criteria; - Transport from UPnP (DNLA) NAS using either WiFi or Cat5 networking - Toslink digital out for using existing HiFi DAC. - Low power consuption (eg just few Watts - it's just a decoder anyway). There seems a real lack of Flac players that are cheap and HiFi separates integratable like current-day CD players. All

From: Steve Underwood <steveu at coppice.org> > It seems some cards use a PLL for their ADC, so they can lock to an > incoming SPDIF signal, but always use a local crystal clock source for > their DAC. These cards do not have their ADC and DAC synchronised. Do common on-board or PCI sound card lock to some incoming signal? Yes, there is a crystal oscillator and a PLL or divider to

http://bugs.freedesktop.org/show_bug.cgi?id=25823 Summary: Not possible to create second X session Product: xorg Version: unspecified Platform: x86 (IA32) OS/Version: Linux (All) Status: NEW Severity: normal Priority: medium Component: Driver/nouveau AssignedTo: nouveau at lists.freedesktop.org

Hi all, The conclusion of the discussion is that most sound cards indeed have different capture and playing frequencies for the unknown reasons. But we all know the adaptive filter of the AEC relies on the synchronization of the far-end and near-end sampling rates. Then Has anybody tried to use speex AEC in Windows system? How do you solve this problem? (I have tested speex AEC. In most

The values are already stored on chipset specific basis in the ctor. Make the most of them and simplify the code further by using a temporary variable to avoid code duplication. Signed-off-by: Emil Velikov <emil.l.velikov at gmail.com> --- Found this patch laying around in a local branch. While the it does not reduce the number of lines, I believe that it makes the code is alot more

I am using ST micro's STM32F103 chip with their AN2812 development code. It uses a PWM output, resulting in low-quality sound. Can I just use a scaled version of the PWM to drive a DAC? (i.e. is the output linear, or is there shaping to make the PWM output work?) Any feedback, links or pointers would be greatly appreciated. David. -------------- next part -------------- An HTML attachment

...t; Asterisk -> Adtran Channel Bank -> (analog) -> Nortel Don't complain that it's ugly. I've already done plenty of that. The CLEC manages their Adtran remotely and needs to be able to continue to do so. I assume they use FDL to do the management. We are using the Zaptel DACS/DACSRBS to cross connect some of the channels directly between the CLEC and the Adtran. These are channels we don't really care about (data, other voice, etc). We are not cross connecting all the channels, just some of them. I'm wondering if/how I can make sure the remote management v...

Heya everybody. I have been peering into the code for zaptel for a while now, I am keenly interested in the dacs support, being able to apparently redirect certain spans to other spans. Not sure if this has to be on the same T1 interface or can be used between T1 interfaces on the same board or possible two different cards. Any information on the functionality of this would be greatly appreciated! I don...

On 8/19/19 9:53 PM, Nir Soffer wrote: > Hi, > > I upgraded to a Fedora 29 host using virt-preview repo to > libvirt-daemon-5.6.0-1.fc29.x86_64 > The host was using plain Fedora 29 without virt-preview before that. > > After the upgrade, starting some vms that were running fine fail now with > this error: > > Error starting domain: internal error: child reported

...rnal DAC as a way to bypass the DAC's internal oversampling and found that wavpak is the only compressed container format that supports the 705/768 sample rates but it is very poorly supported and extremely slow during decompression, and fully uncompressed WAV data is extremely cumbersome. Most DACs do appear to turn off their anti-aliasing filters at sample rates above 192kHz so 384 may well be the limit of any theoretical benefit, but many DACs now support input up to 768kHz (a very small number support even higher.) Studio ADCs often record in the Mhz range, and although they apparently alw...

----- Original Message ----- > From: Eric Blake <eblake@redhat.com> > To: Cristian Ciupitu <cristian.ciupitu@yahoo.com> > Cc: libvirt-users <libvirt-users@redhat.com> > Sent: Monday, August 19, 2013 11:24 PM > Subject: Re: [libvirt-users] Stop the relabeling of CD images > So maybe this would do it: > > <source file=...> >   <seclabel

Then why ONE sound card have different capture and playback rate? It must be ONE single physical clock generator which is used by both ADC and DAC in the sound card, isn't it? If you are a hardware engineer. Will you design two different physical clock for ADC and DAC seperately? What on earth causes this problem? Who knows its intrinsic real reason? Isn't there any other solutions? For

----- Original Message ----- > From: Martin Kletzander <mkletzan@redhat.com> > To: Cristian Ciupitu <cristian.ciupitu@yahoo.com> > Cc: Eric Blake <eblake@redhat.com>; libvirt-users <libvirt-users@redhat.com> > Sent: Tuesday, August 20, 2013 6:05 PM > Subject: Re: [libvirt-users] Stop the relabeling of CD images > > On 08/20/2013 04:19 AM, Cristian

> Hi glad you're interested - comments below. > > On Fri, Jan 15, 2010 at 5:05 PM, Brian Willoughby <brianw at sounds.wa.com > > wrote: > > On Jan 14, 2010, at 20:59, Nicholas Bower wrote: > I'm after the cheapest way to decode a flac stream with following > criteria; > - Transport from UPnP (DNLA) NAS using either WiFi or Cat5 networking > - Toslink

Use the new non-coherent DMA API including proper ownership transfers. This also means we can allocate the buffer memory with the proper direction instead of bidirectional. Signed-off-by: Christoph Hellwig <hch at lst.de> --- sound/mips/hal2.c | 58 ++++++++++++++++++++++------------------------- 1 file changed, 27 insertions(+), 31 deletions(-) diff --git a/sound/mips/hal2.c

hmmm. my associate needs a USB sound thing in his stack... its always worked before, but when I brought him up on a new system with 6.7, the device is recognized.... usb 1-1: New USB device found, idVendor=08bb, idProduct=2704 usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=0 usb 1-1: Product: USB Audio DAC usb 1-1: Manufacturer: Burr-Brown from TI usb 1-1: configuration

Hi all, I use libvirt/kvm for my virtualization needs and have a little scerio in mind that I want to implement: a small linux box boots up, launches an X session and after that launches a virtual machine with a fullscreen sdl window, With this I hope to achive several goals: user gets a familiar experience with guest os ( I had windows in mind), near native graphics responsiveness ( we deploy

Hi Underwood, Thank you for your help. I agree with your opinion. But it is almost impossible to further reduce the frequent difference between play and capture. 1. I used a 2^18 step FFT to analyse the echo frequency. So the freq resolution is 8000HZ/(2^17)=0.0625Hz. The analyser need at least 2^18/8000=32 seconds acoustic echo record signal from the microphone. Better freq resolution relies