Re: [Alsa-user] best card for bitperfect SPDIF I/O with external clock sync ?
Paolo Saggese wrote: On Thursday 22 November 2007 15:25, you wrote: This is a valid wish, isn't it? And at least the M-Audio Audiphile cards can sync themselves to the S/PDIF In clock. They can, but the word clock is used only as a sample clock. It will not reduce the amount of jitter of the card's S/PDIF output. that's not a concern! Let' try to explain it better: it's all about avoiding the use of the SPDIF reconstructed clock altogether (on the DAC side). The S/PDIF standard specifies that the clock must be reconstructed from the signal itself. If I can slave the SPDIF clock coming into the external DAC to the same DAC local clock, I can simply ignore the incoming SPDIF clock and just use the local one without any need for reclocking, PLLs, etc! Yes, _if_ you can. http://www.audiocraftersguild.com/AandE/npt.on.jitter2.htm says: | A few companies which make both transports and external DACs have | implemented schemes in which the S/PDIF signal is supplemented with a | second line carrying the master clock back from the external DAC to | the transport. In this way the DAC's crystal becomes the master rather | than the transport and the problems of recovering a spectrally pure | clock are eliminated. No standards for this type of implementation | exist. In reference #4 Dr. Hawksford calls for the clock signal to be | transmitted on a second S/PDIF line, I know of no actual product which | implements this scheme. Sony (in one product) and Arcam send the | actual clock, Linn argues this leads to RFI problems and so they send | a DC servo voltage which controls a VCOX in the transport. So it seems you wouldn't be able to find a sound card that does what you want. If all your audio data comes from the PC anyway, you might want to drop S/PDIF and use a sound card with a break-out box. The I2S bus used to control the DACs in those devices has a separate master clock line. Regards, Clemens - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] best card for bitperfect SPDIF I/O with external clock sync ?
Hi Darrell Bellerive! On 2007.11.22 at 17:54:19 -0800, Darrell Bellerive wrote next: I have never been happy with this card. While it works okay for playing basic sound, getting it to do anything more sophisticated is pure black magic. For example, I have never gotten full duplex to work. Well now, when pulseaudio era came, hopefully it's not true anymore. Also while the Audiophile 24/96 does sample at 96 KHz, the audio bandwidth of the card is limited to 22 Hz to 22 KHz +/- 0.4 dB. How does that matter when all is needed is digital output? Anyway, in analog mode, are you sure there is no option to switch off bandwidth filter? -- Vladimir - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] best card for bitperfect S PDIF I/O with external clock sync ?
On Friday 23 November 2007, Bill Unruh wrote: On Fri, 23 Nov 2007, Gene Heskett wrote: On Friday 23 November 2007, Vladimir Mosgalin wrote: Hi Darrell Bellerive! On 2007.11.22 at 17:54:19 -0800, Darrell Bellerive wrote next: I have never been happy with this card. While it works okay for playing basic sound, getting it to do anything more sophisticated is pure black magic. For example, I have never gotten full duplex to work. Well now, when pulseaudio era came, hopefully it's not true anymore. Also while the Audiophile 24/96 does sample at 96 KHz, the audio bandwidth of the card is limited to 22 Hz to 22 KHz +/- 0.4 dB. How does that matter when all is needed is digital output? Anyway, in analog mode, are you sure there is no option to switch off bandwidth filter? No one in their right mind would want to do that as the aliasing would drive you up a wall. The other delay distortions the filter might give are 100's of times more tolerable to listen to. Of course there is no aliasing problem at sampling at 96K and having the frequency go up to 40K. There is an aliasing problem if you then downconvert that to 48K or 44.1K but surely the downconverter should handle that not the soundcard. Mind you why you want more than 22K I have no idea. For sound sources, that is all that you can hear. (NOt me, my ears are old and have trouble with 10K, but if you have little children they might appreciate the extra few KHz, but probably not since they already have to tune out that annoying 15.7KHz scream from the TV. I'm 73, and my ears aren't in that good a shape. The last time I was tested I had a Carhart notch over 120 db deep at 4 kilohertz, rising back up to about 15 db down at around 7khz, and another notch at 15.735 khz about 40 db deep, but then I was the CE at a tv station for around 30 years, with 10 to 25 monitors in the control room, all screaming at or about that frequency, you can hear the doppler shift as the satellite carrying the signal wanders around in it box in the sky, and it was back up to about -20 db at 22 khz. I can still hear that stuff very plainly as I also have many years of using SSB on chicken band radio where cheap filters let a lot of the opposite sideband thru. So if those badly worn ears can hear and identify it, pity the child who hasn't yet worn out 3 centerfire rifle barrels before he figured he'd better be wearing some earmuffs. That BTW is the cause of the Carhart Notch. - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user -- Cheers, Gene There are four boxes to be used in defense of liberty: soap, ballot, jury, and ammo. Please use in that order. -Ed Howdershelt (Author) Mix's Law: There is nothing more permanent than a temporary building. There is nothing more permanent than a temporary tax. - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] best card for bitperfect S PDIF I/O with external clock sync ?
On Thursday 22 November 2007 21:01, Vladimir Mosgalin wrote: I meant M-Audio Revolution 7.1, and quite possibly M-Audio Revolution 5.1 allow you to use external clock source. In theory, yes. In practice, I wasn't able to make my M-Audio Audiophile USB get clock from external source. Well it kinda works, but at some point distortions appear, and one must force clock resync or something like that by turning card off and on. mmh... I guess the USB interface may be the problem source here. Though USB could be operated in asynchronous mode, AFAIK most (all?) of the current USB audio devices operate it in synchronous mode, with the clock provided from the PC. Of course, if the USB interface (from which the digital audio stream is coming) is operated synchronously with source-based clock, then the sound card MUST be somehow in sync with that stream clock... The only way to loosen the sync with that clock and try to sync to another one is to do some sort of reclocking, but unfortunately that is not gonna work quite well. In particular when the upstream clock is that of an USB connection which - apart from being usually dirt and not so stable - is at odd rates with respect to audio clocks. The only real solution would be a sound card which connect to the PC through an _asynchronous_ interface (be it USB, firewire, Ethernet, HDMI or whatever else, as long as it's fast enough and asynchronous). BTW, do you know if any such device exists? I wanted to create setup similar to this, and one of the things I learned that in order to reduce jitter, you'd want to have power as clean as possible. indeed, absolutely. On-board soundchips produce lowest quality signal, pci/pcie boards have much better filtering and produce better signal, but if you want something better, you have to use card which isn't powered by PSU of your PC, and doesn't suffer from problems of its signal. So if you want best digital audio, you probably should look among external cards (usb/firewire) which aren't bus powered, and use external AC adapter. Interface doesn't matter as long as card doesn't get power from it, so choose most compatible card. well, I have to disagree here. Interface would/should not matter ONLY if it is asynchronous. If the interface is synchronous, then the card has to lock itself to the interface clock and, unfortunately, IMHO/IME no reclocking and/or resampling will ever be able to really clean up the mess. For the USB, the standard bus clock frequency makes things even worse... don't know about firewire. Ciao, Paolo. -- Skype: Paolo.Saggese http://borex.lngs.infn.it/saggese You can still escape from the GATES of hell: Use Linux! - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] M-Audio Audiophile 2496 was: Re: best card for bitperfect SPDIF I/O with?external clock sync ?
Hi Darrell Bellerive! On 2007.11.23 at 04:25:56 -0800, Darrell Bellerive wrote next: Well now, when pulseaudio era came, hopefully it's not true anymore. Pulseaudio shows some promise, but not all apps support it yet. Wasn't Most of them do - some through wrappers or alsa-pulse routing, for example SDL, openal and wine, but unlike jack, it works much better. The most problematic thing is gstreamer, it works with pulse, but audio quality suffers for unknow reason (not much, but quite noticeable on some material, and very annoying if you have good ears). Besides, with stuff like per-application mixers which stores its state (a feature you'll fall in love with instantly), on the fly audio card detection and stream switching, network transparency with transparent stream migration (just a few clicks to move sounds from your notebook to home audio system connected to main pc or home media server when you're home - I find it really useful), high quality resampling and so on really makes it the most desired think to have in the core of your linux audio system. JACK, and then GStreamer, and soon Phonon, supposed to solve all our Jack has completely different purpose. It is good (bwt you can run pulse on top of jack, and some day probably will be able to run jack on top of pulse - though it wouldn't have much sense). gstreamer and photon are systems working at different level. They don't conflict with any sound server and support most of them. audio problems? Yet another sound server. You might think of it like this, however after discovering pulseaudio I think that it's holy grail of linux audio we've been searching for years. Still a bit edgy, though. Also while the Audiophile 24/96 does sample at 96 KHz, the audio bandwidth of the card is limited to 22 Hz to 22 KHz +/- 0.4 dB. Anyway, in analog mode, are you sure there is no option to switch off bandwidth filter? If there is, it is not documented. http://alsa.opensrc.org/index.php/Ice1712 http://alsa.opensrc.org/index.php/Envy24control Well there are Audiophile 192, Audiophile USB and other cards in Audiophile series - most are even more interesting than older Audiophile 24/96. So I don't really see any problem. -- Vladimir - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] M-Audio Audiophile 2496 was: Re: best card for bitperfect SPDIF I/O with external clock sync ?
Thanks for the reply. On Friday 23 November 2007 00:42, Vladimir Mosgalin wrote: On 2007.11.22 at 17:54:19 -0800, Darrell Bellerive wrote next: I have never been happy with this card. While it works okay for playing basic sound, getting it to do anything more sophisticated is pure black magic. For example, I have never gotten full duplex to work. Well now, when pulseaudio era came, hopefully it's not true anymore. Pulseaudio shows some promise, but not all apps support it yet. Wasn't JACK, and then GStreamer, and soon Phonon, supposed to solve all our audio problems? Yet another sound server. Also while the Audiophile 24/96 does sample at 96 KHz, the audio bandwidth of the card is limited to 22 Hz to 22 KHz +/- 0.4 dB. Anyway, in analog mode, are you sure there is no option to switch off bandwidth filter? If there is, it is not documented. http://alsa.opensrc.org/index.php/Ice1712 http://alsa.opensrc.org/index.php/Envy24control -- Darrell Bellerive - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] best card for bitperfect S PDIF I/O with external clock sync ?
On Friday 23 November 2007, Vladimir Mosgalin wrote: Hi Darrell Bellerive! On 2007.11.22 at 17:54:19 -0800, Darrell Bellerive wrote next: I have never been happy with this card. While it works okay for playing basic sound, getting it to do anything more sophisticated is pure black magic. For example, I have never gotten full duplex to work. Well now, when pulseaudio era came, hopefully it's not true anymore. Also while the Audiophile 24/96 does sample at 96 KHz, the audio bandwidth of the card is limited to 22 Hz to 22 KHz +/- 0.4 dB. How does that matter when all is needed is digital output? Anyway, in analog mode, are you sure there is no option to switch off bandwidth filter? No one in their right mind would want to do that as the aliasing would drive you up a wall. The other delay distortions the filter might give are 100's of times more tolerable to listen to. -- Cheers, Gene There are four boxes to be used in defense of liberty: soap, ballot, jury, and ammo. Please use in that order. -Ed Howdershelt (Author) Pie are not square. Pie are round. Cornbread are square. - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] Re : Re : Problem with emu 0404 P CI, and alsa 1.0.15rc*
Hi all Finaly, after some weeks of error (lot of unknow symbol errors, when inserting modules), I finaly decided to reinstall my gusty system. Then I tried the latest alsa daily build, and that almost work ! I now have a problem loading firmware, at boot time it stops and my systeme dont boot, I must uninstall the module to be able to boot. when inserting the modules, I look in /var/log/messages, and found this : []ACPI: PCI Interrupt :01:08.0[A] - Link [APC1] - GSI 16 (level, low) - []IRQ 22 []emu1010: Special config. []emu1010: EMU_HANA_ID=0x7f []emu1010: filename emu/emu0404.fw testing []firmware size=0xd67c []emu1010: Loading Hana Firmware file failed, reg=0x7f []c013b9ee I compiled the latest firmware succesfully, they are in /lib/firmware/emu/ Any idea ? Thanks :) -fx On Monday 05 November 2007 19:43:56 James Courtier-Dutton wrote: fx wrote: Hi Still the same problem, same error, nothing changed :( Thank for the try ! -fx Try the latest alsa-kernel and alsa-driver from the hg repository. I don't think you are installing the new drivers correctly. James - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] best card for bitperfect S PDIF I/O with external clock sync ?
On Fri, 23 Nov 2007, Gene Heskett wrote: On Friday 23 November 2007, Vladimir Mosgalin wrote: Hi Darrell Bellerive! On 2007.11.22 at 17:54:19 -0800, Darrell Bellerive wrote next: I have never been happy with this card. While it works okay for playing basic sound, getting it to do anything more sophisticated is pure black magic. For example, I have never gotten full duplex to work. Well now, when pulseaudio era came, hopefully it's not true anymore. Also while the Audiophile 24/96 does sample at 96 KHz, the audio bandwidth of the card is limited to 22 Hz to 22 KHz +/- 0.4 dB. How does that matter when all is needed is digital output? Anyway, in analog mode, are you sure there is no option to switch off bandwidth filter? No one in their right mind would want to do that as the aliasing would drive you up a wall. The other delay distortions the filter might give are 100's of times more tolerable to listen to. Of course there is no aliasing problem at sampling at 96K and having the frequency go up to 40K. There is an aliasing problem if you then downconvert that to 48K or 44.1K but surely the downconverter should handle that not the soundcard. Mind you why you want more than 22K I have no idea. For sound sources, that is all that you can hear. (NOt me, my ears are old and have trouble with 10K, but if you have little children they might appreciate the extra few KHz, but probably not since they already have to tune out that annoying 15.7KHz scream from the TV. - This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse012070mrt/direct/01/ ___ Alsa-user mailing list Alsa-user@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user
