search for: dac

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 #...

...d seem to point to SPDIF and AES3 as the ideal interconnect, but the unfortunate truth is that those two specific digital audio standards create more problems than they solve. I'll try to explain: Every digital audio playback system combines some sort of storage media with some sort of DAC. Between these basic elements is a flow of data at a particular sampling rate. Without getting into multichannel systems that combine multiple DAC boxes into one system, there are basically only two choices for clocking. Either the DAC is the master or the media device is the master....

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 solutions I've seen are either expensive boutique heavy-weights or portable devices. Squeezebo...

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 generate assigned clock signal. But if playing and capture are assigned the same sampling rate, why clock of ADC an...

....0: nouveau_channel_alloc: initialised FIFO 1 [ 228.587351] [drm] nouveau 0000:01:00.0: Setting dpms mode 3 on vga encoder (output 0) [ 228.597304] [drm] DAC-7: set mode 1280x1024_75.00 3e [ 228.607513] [drm] nouveau 0000:01:00.0: Setting dpms mode 0 on vga encoder (output 0) [ 228.607519] [drm] nouveau 0000:01:00.0: Output VGA-1 is running on CRTC 0 using o...

...;> >> From: Steve Underwood <steveu at coppice.org <mailto: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? >> >I already said they do, and what the lock to - SPDIF ports. Most modern >machines have one, and if they don't its generally still there...

...< priv->pior.nr; i++) { tmp = nv_rd32(priv, 0x61e000 + (i * 0x800)); nv_wr32(priv, 0x6101f0 + (i * 0x04), tmp); } @@ -834,10 +834,11 @@ exec_script(struct nv50_disp_priv *priv, int head, int id) u8 ver, hdr, cnt, len; u16 data; u32 ctrl = 0x00000000; + u32 reg; int i; /* DAC */ - for (i = 0; !(ctrl & (1 << head)) && i < 3; i++) + for (i = 0; !(ctrl & (1 << head)) && i < priv->dac.nr; i++) ctrl = nv_rd32(priv, 0x610b5c + (i * 8)); /* SOR */ @@ -845,19 +846,18 @@ exec_script(struct nv50_disp_priv *priv, int head, int id...

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 was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20111025/a032fb...

...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...

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...

...t 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 (status=125): > Requested operation is not valid: Setting different DAC user or group on > /home/libvirt/images/voodoo4-os.img which is already in use > > Traceback (most recent call last): > File "/usr/share/virt-manager/virtManager/asyncjob.py", line 75, in > cb_wrapper > callback(asyncjob, *args, **kwargs) > File "/usr...

...ontainer format to support sample rates above the 655k current limit. Please note this is not for using ridiculously high sample rate sourced audio for some imaginary audible benefit. I've been involved in some experimentation with offline upsampling in software prior to delivery to an external 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...

...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 model='selinux' relabel='no'/> >   <seclabel model='dac' relabel='no'/> > </source> I've just tried it and the SELinux label is not changed anymore, but the ownership is still changed to qemu:qemu. > I'm also not sure why you think to resort to chattr +i, but if using > that causes libvirt heartburn, maybe we have...

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 example, I have also tested Microsoft MSN and S...

...4 PM >>> Subject: Re: [libvirt-users] Stop the relabeling of CD images >> >>> So maybe this would do it: >>> >>> <source file=...> >>>     <seclabel model='selinux' relabel='no'/> >>>     <seclabel model='dac' relabel='no'/> >>> </source> >> >> I've just tried it and the SELinux label is not changed anymore, but >> the ownership is still changed to qemu:qemu. >> >>> I'm also not sure why you think to resort to chattr +i, but if using...

....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 digital out for using existing HiFi DAC. > - Low power consuption (eg just few Watts - it's just a decoder > anyway). > Your criteria fall short of the full potential for quality when you > separate the transport from the DAC via TOSLINK. I suppose that if > you mention "cheapest" then you're perha...

...09,21 +512,16 @@ static int hal2_playback_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); - int err; runtime->hw = hal2_pcm_hw; - - err = hal2_alloc_dmabuf(hal2, &hal2->dac); - if (err) - return err; - return 0; + return hal2_alloc_dmabuf(hal2, &hal2->dac, DMA_TO_DEVICE); } static int hal2_playback_close(struct snd_pcm_substream *substream) { struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream); - hal2_free_dmabuf(hal2, &hal2->dac); + hal2...

...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 #1 chosen from 1 choice usb 1-1: ctrl urb status -75 received input: Burr-Brown from TI USB Audio DAC as /devices/pci0000:00/0000:00:01.2/usb1/1-1/1-1:1.2/input/input3 generic-usb 0003:08BB:2704.0001: input,hidraw0:...

...xcb_connection_has_error() returned true pulseaudio: pa_context_connect() failed pulseaudio: Reason: Connection refused pulseaudio: Failed to initialize PA contextaudio: Could not init `pa' audio driver ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave alsa: Could not initialize DAC alsa: Failed to open `default': alsa: Reason: Device or resource busy ALSA lib pcm_dmix.c:1022:(snd_pcm_dmix_open) unable to open slave alsa: Could not initialize DAC alsa: Failed to open `default': alsa: Reason: Device or resource busy audio: Failed to create voice `dac' No protocol sp...

...n longer FFT and also longer record echo signal. It is almost not possible for a practical system. 2. Better resampler with smaller step size is also a challenge in its performance. So it is difficult to reduce the frequent difference further more. About the change of sample rates of the ADC and DAC. I don't think it is a problem if the ADC and DAC share the same clock. The real problem is why they have different freq clock even if they if they are assigned with the same sample rate. For example: 1. for a pure hardware sound card. There is a a crystal quartz oscillator and two PLL circu...