Sound cards (Was: Re: no sound
On Fri, 2011-05-20 at 12:48 +0200, Robert Klaar wrote: Any Rme card is good, although they're a bit expensive. Didn't have to do a thing to get my hdsp 9632 working. Hi Robert :) the main reason to switch the sound cards is the audio sound quality. I'm able to pay around 700,- EUR / 800,- EUR (right now, regarding to my profession, I usually don't have any money). Btw. I noticed that not only my sound cards do cause loss. When recording soft synth just by JACK, without the sound cards being involved, there's a loss for the sound quality too! And soft synth already do sound less good than real old synth. Unfortunately those real old synth can break and there're no microchips available to repair those synth, resp. they are hard to get, very expensive and without warranty. My two TerraTec EWX 24/96 needs to be replaced, before I don't have got the money anymore. FWIW S/PDIF doesn't work for my Ubuntu Studio with the TerraTecs, hence I can't use good analog IOs that would be available via S/PDIF. Btw. good analog IOs in this context does mean consumer DAT Sony DTC-670 and Aiwa HD-S1, both are without any loss of sound quality, when listening by my consumer equipment. At least good consumer sound quality is what I expect of a professional sound card, even if internal Linux there still would be loss caused by JACK or caused by what issue ever. Once Brauner borrowed me a Mac with a Motu firewire device. This devices sound quality and the Mac's sound quality accomplished this requirement! No, I'm not using Brauner microphones for my home studio ;), all I need is good consumer sound quality. Two question about the RME card, I'll read more about sound cards later and during the weekend and maybe I'll order a card next week. http://www.thomann.de/gb/search_dir.html?xsid=294c1645e0e16b761c35fa8f9ebcec51sw=HDSP+9632x=0y=0 At a max of 4 analog IOs? The unbalanced breakout cables are part of the product content? Cheers! Ralf -- Ubuntu-Studio-users mailing list Ubuntu-Studio-users@lists.ubuntu.com Modify settings or unsubscribe at: https://lists.ubuntu.com/mailman/listinfo/ubuntu-studio-users
Re: Sound cards (Was: Re: no sound
Am Fri, 20 May 2011 13:54:57 +0200 schrieb Ralf Mardorf ralf.mard...@alice-dsl.net: When recording soft synth just by JACK, without the sound cards being involved, there's a loss for the sound quality too! Wait a minute... could you explain that? You have a loss of quality compared to live playback of the soft synths (using JACK?) when playing back a recording taken from JACK? A recording that preserves 32 bit floating point sample format (heck, or 24 bit integer) and the sample rate, of course? I have to wonder what you did there to alter the data from the soft synth. I mean ... we're talking bit-exact copy here, aren't we? Can you present a test setup to observe that issue? Alrighty then, Thomas. signature.asc Description: PGP signature -- Ubuntu-Studio-users mailing list Ubuntu-Studio-users@lists.ubuntu.com Modify settings or unsubscribe at: https://lists.ubuntu.com/mailman/listinfo/ubuntu-studio-users
Re: Sound cards (Was: Re: no sound
Am 20.05.2011 13:54, schrieb Ralf Mardorf: On Fri, 2011-05-20 at 12:48 +0200, Robert Klaar wrote: Any Rme card is good, although they're a bit expensive. Didn't have to do a thing to get my hdsp 9632 working. Hi Robert :) the main reason to switch the sound cards is the audio sound quality. I'm able to pay around 700,- EUR / 800,- EUR (right now, regarding to my profession, I usually don't have any money). Btw. I noticed that not only my sound cards do cause loss. When recording soft synth just by JACK, without the sound cards being involved, there's a loss for the sound quality too! No, there is none. Any software, that generates sounds from scratch like a softsynth wil produce exactly the same stream with any soun card. In fact such software will even generate the very same stream if no soundcard exists. And if you record such a stream with Jack you simply store that very stream bit by bit. I assure you, you get the very same data in a recording via Jack if you play the same patch of the same synth on a work station with a RME Hammerfall or on a Laptop with a built-in HDA. The only level on wich a soundcard is related to a softsynth is the one on wich you actually hear the stream. And some synths can render differntly, if Jack is running at 96KHz instead of 48 or 44.1. But this has only remotely to do with the sound card let alone its quality. And soft synth already do sound less good than real old synth. Unfortunately those real old synth can break and there're no microchips available to repair those synth, resp. they are hard to get, very expensive and without warranty. My two TerraTec EWX 24/96 needs to be replaced, Tell me where you dump them, these 2 more stereo-dacs would be most welcome in my box ;-) All the trouble with the envy24-cards is related to (mis)configuration and to stupidities like automatically zeroing all channels caused by PA in most cases. As of now these problems can be solved by the user. They are *not* acceptable, they are bugs that need to be solved. But these bugs are not show-stoppers. before I don't have got the money anymore. FWIW S/PDIF doesn't work for my Ubuntu Studio with the TerraTecs, hence I can't use good analog IOs that would be available via S/PDIF. Btw. good analog IOs in this context does mean consumer DAT Sony DTC-670 and Aiwa HD-S1, both are without any loss of sound quality, when listening by my consumer equipment. At least good consumer sound quality is what I expect of a professional sound card, even if internal Linux there still would be loss caused by JACK or caused by what issue ever. Jack does not cause any loss in sound-quality because Jack does not have any influence on the way, the pcm-stream is produced by the driver/sound-card. Once Brauner borrowed me a Mac with a Motu firewire device. The MOTUs are quite okayish and they sound exactly the same on any system that supports them. the Mac's sound quality There is no such thing. No Mac-expert would endorse something like a special Mac-related sound quality. Do not mix that up with sound performance that has to do with latencies and stability but *not* with how good it sounds in the end. You know why? Because no professional would want to buy/use any computer/OS, that attempts to manipulate the sound produced from a pro-interface, be it for better or worse. OS/Driver etc *has* to be absolutely neutral in that, everything else is super-bass-enancer nonsense that one my expect in a cheap MP3-player but certainly *not* in a computer-system built for pros. accomplished this requirement! No, I'm not using Brauner microphones for my home studio ;), all I need is good consumer sound quality. All your experience is fired by the real quality of the DAC/ADC-hardware on the cards you have used and to some extend may be influenced by mixer-settings. no offence ment but RTFM please. best regs HZN Two question about the RME card, I'll read more about sound cards later and during the weekend and maybe I'll order a card next week. http://www.thomann.de/gb/search_dir.html?xsid=294c1645e0e16b761c35fa8f9ebcec51sw=HDSP+9632x=0y=0 At a max of 4 analog IOs? The unbalanced breakout cables are part of the product content? Cheers! Ralf -- Ubuntu-Studio-users mailing list Ubuntu-Studio-users@lists.ubuntu.com Modify settings or unsubscribe at: https://lists.ubuntu.com/mailman/listinfo/ubuntu-studio-users
Re: Sound cards (Was: Re: no sound
On Fri, 2011-05-20 at 14:36 +0200, Hartmut Noack wrote: Am 20.05.2011 13:54, schrieb Ralf Mardorf: On Fri, 2011-05-20 at 12:48 +0200, Robert Klaar wrote: Any Rme card is good, although they're a bit expensive. Didn't have to do a thing to get my hdsp 9632 working. Hi Robert :) the main reason to switch the sound cards is the audio sound quality. I'm able to pay around 700,- EUR / 800,- EUR (right now, regarding to my profession, I usually don't have any money). Btw. I noticed that not only my sound cards do cause loss. When recording soft synth just by JACK, without the sound cards being involved, there's a loss for the sound quality too! No, there is none. Any software, that generates sounds from scratch like a softsynth wil produce exactly the same stream with any soun card. In fact such software will even generate the very same stream if no soundcard exists. And if you record such a stream with Jack you simply store that very stream bit by bit. Wrong! There still could be rounding errors and dithering involved, a digital copy very often isn't a digital copy ;). But ok, here I didn't use dithering and I do use 32-bit float, but the result has a loss. I dunno what do cause this loss. For Jack there also is a zero-copy issue! You can't do wild connections using Jack, but you need to take care about the order, when e.g. connecting a client to itself. I assure you, you get the very same data in a recording via Jack if you play the same patch of the same synth on a work station with a RME Hammerfall or on a Laptop with a built-in HDA. The only level on wich a soundcard is related to a softsynth is the one on wich you actually hear the stream. And some synths can render differntly, if Jack is running at 96KHz instead of 48 or 44.1. But this has only remotely to do with the sound card let alone its quality. And soft synth already do sound less good than real old synth. Unfortunately those real old synth can break and there're no microchips available to repair those synth, resp. they are hard to get, very expensive and without warranty. My two TerraTec EWX 24/96 needs to be replaced, Tell me where you dump them, these 2 more stereo-dacs would be most welcome in my box ;-) All the trouble with the envy24-cards is related to (mis)configuration and to stupidities like automatically zeroing all channels caused by PA in most cases. As of now these problems can be solved by the user. They are *not* acceptable, they are bugs that need to be solved. But these bugs are not show-stoppers. before I don't have got the money anymore. FWIW S/PDIF doesn't work for my Ubuntu Studio with the TerraTecs, hence I can't use good analog IOs that would be available via S/PDIF. Btw. good analog IOs in this context does mean consumer DAT Sony DTC-670 and Aiwa HD-S1, both are without any loss of sound quality, when listening by my consumer equipment. At least good consumer sound quality is what I expect of a professional sound card, even if internal Linux there still would be loss caused by JACK or caused by what issue ever. Jack does not cause any loss in sound-quality because Jack does not have any influence on the way, the pcm-stream is produced by the driver/sound-card. Once Brauner borrowed me a Mac with a Motu firewire device. The MOTUs are quite okayish and they sound exactly the same on any system that supports them. the Mac's sound quality There is no such thing. No Mac-expert would endorse something like a special Mac-related sound quality. Do not mix that up with sound performance that has to do with latencies and stability but *not* with how good it sounds in the end. You know why? Because no professional would want to buy/use any computer/OS, that attempts to manipulate the sound produced from a pro-interface, be it for better or worse. OS/Driver etc *has* to be absolutely neutral in that, everything else is super-bass-enancer nonsense that one my expect in a cheap MP3-player but certainly *not* in a computer-system built for pros. accomplished this requirement! No, I'm not using Brauner microphones for my home studio ;), all I need is good consumer sound quality. All your experience is fired by the real quality of the DAC/ADC-hardware on the cards you have used and to some extend may be influenced by mixer-settings. no offence ment but RTFM please. best regs HZN Two question about the RME card, I'll read more about sound cards later and during the weekend and maybe I'll order a card next week. http://www.thomann.de/gb/search_dir.html?xsid=294c1645e0e16b761c35fa8f9ebcec51sw=HDSP+9632x=0y=0 At a max of 4 analog IOs? The unbalanced breakout cables are part of the product content? Cheers! Ralf -- Ubuntu-Studio-users mailing list Ubuntu-Studio-users@lists.ubuntu.com Modify settings or unsubscribe at:
Re: Sound cards (Was: Re: no sound
Am 20.05.2011 14:37, schrieb Ralf Mardorf: On Fri, 2011-05-20 at 14:04 +0200, Thomas Orgis wrote: Am Fri, 20 May 2011 13:54:57 +0200 schrieb Ralf Mardorfralf.mard...@alice-dsl.net: When recording soft synth just by JACK, without the sound cards being involved, there's a loss for the sound quality too! Wait a minute... could you explain that? You have a loss of quality compared to live playback of the soft synths (using JACK?) when playing back a recording taken from JACK? A recording that preserves 32 bit floating point sample format (heck, or 24 bit integer) and the sample rate, of course? Yes and other people who can't hear it, do have it too. I do not. You can see it by watching the waves spectral by Audacity. I did this regarding to a zero-copy issue, that appears if a Jack client is connected directly to itself, e.g. to do the mastering. 48 and 96 KHz, 32-bit wav 32-bit float. If a synth has dynamic filters it will never produce the exactly same stream twice. But if you think about yourself you will find out, that given you use the same settings for Jack on a HDA or a HDSP you will get exactly the same quality. Simply because a synth-software only delivers, what it renders to Jack and Jack does *not* change anything in that rendered data. There is simply not soundcard and not even a driver involved in the rendering itself. DSPs only do the very same thing faster as cheap chips. All difference in sound quality is related to DAC/ADC period I have to wonder what you did there to alter the data from the soft synth. I mean ... we're talking bit-exact copy here, aren't we? Can you present a test setup to observe that issue? Any Linux install I know, e.g. 64 Studio 64-bit 3.0, 3.3, Suse 64-bit 11.2 and Edubuntu 32-bit Maverick + Ubuntu Studio meta packages and others! If you can't here it, try to see it. If you don't have this issue too, some people claim that they get 100% correct digital copies, then something on my machine might cause a software issue, but I don't think so. Ralf Alrighty then, Thomas. -- Ubuntu-Studio-users mailing list Ubuntu-Studio-users@lists.ubuntu.com Modify settings or unsubscribe at: https://lists.ubuntu.com/mailman/listinfo/ubuntu-studio-users
Re: Sound cards (Was: Re: no sound
Am 20.05.2011 14:58, schrieb Ralf Mardorf: On Fri, 2011-05-20 at 14:36 +0200, Hartmut Noack wrote: Am 20.05.2011 13:54, schrieb Ralf Mardorf: On Fri, 2011-05-20 at 12:48 +0200, Robert Klaar wrote: Any Rme card is good, although they're a bit expensive. Didn't have to do a thing to get my hdsp 9632 working. Hi Robert :) the main reason to switch the sound cards is the audio sound quality. I'm able to pay around 700,- EUR / 800,- EUR (right now, regarding to my profession, I usually don't have any money). Btw. I noticed that not only my sound cards do cause loss. When recording soft synth just by JACK, without the sound cards being involved, there's a loss for the sound quality too! No, there is none. Any software, that generates sounds from scratch like a softsynth wil produce exactly the same stream with any soun card. In fact such software will even generate the very same stream if no soundcard exists. And if you record such a stream with Jack you simply store that very stream bit by bit. Wrong! There still could be rounding errors and dithering involved Yes, dithering makes a differnce -- if you use it. And do you think, sound cards or drivers provoke specific rounding errors in streams that are not even deliverd to them by Jack? a digital copy very often isn't a digital copy ;). But ok, here I didn't use dithering and I do use 32-bit float, but the result has a loss. I dunno what do cause this loss. For Jack there also is a zero-copy issue! You can't do wild connections using Jack, but you need to take care about the order, when e.g. connecting a client to itself. I assure you, you get the very same data in a recording via Jack if you play the same patch of the same synth on a work station with a RME Hammerfall or on a Laptop with a built-in HDA. The only level on wich a soundcard is related to a softsynth is the one on wich you actually hear the stream. And some synths can render differntly, if Jack is running at 96KHz instead of 48 or 44.1. But this has only remotely to do with the sound card let alone its quality. And soft synth already do sound less good than real old synth. Unfortunately those real old synth can break and there're no microchips available to repair those synth, resp. they are hard to get, very expensive and without warranty. My two TerraTec EWX 24/96 needs to be replaced, Tell me where you dump them, these 2 more stereo-dacs would be most welcome in my box ;-) All the trouble with the envy24-cards is related to (mis)configuration and to stupidities like automatically zeroing all channels caused by PA in most cases. As of now these problems can be solved by the user. They are *not* acceptable, they are bugs that need to be solved. But these bugs are not show-stoppers. before I don't have got the money anymore. FWIW S/PDIF doesn't work for my Ubuntu Studio with the TerraTecs, hence I can't use good analog IOs that would be available via S/PDIF. Btw. good analog IOs in this context does mean consumer DAT Sony DTC-670 and Aiwa HD-S1, both are without any loss of sound quality, when listening by my consumer equipment. At least good consumer sound quality is what I expect of a professional sound card, even if internal Linux there still would be loss caused by JACK or caused by what issue ever. Jack does not cause any loss in sound-quality because Jack does not have any influence on the way, the pcm-stream is produced by the driver/sound-card. Once Brauner borrowed me a Mac with a Motu firewire device. The MOTUs are quite okayish and they sound exactly the same on any system that supports them. the Mac's sound quality There is no such thing. No Mac-expert would endorse something like a special Mac-related sound quality. Do not mix that up with sound performance that has to do with latencies and stability but *not* with how good it sounds in the end. You know why? Because no professional would want to buy/use any computer/OS, that attempts to manipulate the sound produced from a pro-interface, be it for better or worse. OS/Driver etc *has* to be absolutely neutral in that, everything else is super-bass-enancer nonsense that one my expect in a cheap MP3-player but certainly *not* in a computer-system built for pros. accomplished this requirement! No, I'm not using Brauner microphones for my home studio ;), all I need is good consumer sound quality. All your experience is fired by the real quality of the DAC/ADC-hardware on the cards you have used and to some extend may be influenced by mixer-settings. no offence ment but RTFM please. best regs HZN Two question about the RME card, I'll read more about sound cards later and during the weekend and maybe I'll order a card next week. http://www.thomann.de/gb/search_dir.html?xsid=294c1645e0e16b761c35fa8f9ebcec51sw=HDSP+9632x=0y=0 At a max of 4 analog IOs? The unbalanced breakout cables are part of the product content? Cheers! Ralf -- Ubuntu-Studio-users mailing list
Re: Sound cards (Was: Re: no sound
On Fri, 2011-05-20 at 15:00 +0200, Hartmut Noack wrote: Am 20.05.2011 14:37, schrieb Ralf Mardorf: On Fri, 2011-05-20 at 14:04 +0200, Thomas Orgis wrote: Am Fri, 20 May 2011 13:54:57 +0200 schrieb Ralf Mardorfralf.mard...@alice-dsl.net: When recording soft synth just by JACK, without the sound cards being involved, there's a loss for the sound quality too! Wait a minute... could you explain that? You have a loss of quality compared to live playback of the soft synths (using JACK?) when playing back a recording taken from JACK? A recording that preserves 32 bit floating point sample format (heck, or 24 bit integer) and the sample rate, of course? Yes and other people who can't hear it, do have it too. I do not. You can see it by watching the waves spectral by Audacity. I did this regarding to a zero-copy issue, that appears if a Jack client is connected directly to itself, e.g. to do the mastering. 48 and 96 KHz, 32-bit wav 32-bit float. If a synth has dynamic filters it will never produce the exactly same stream twice. But if you think about yourself you will find out, that given you use the same settings for Jack on a HDA or a HDSP you will get exactly the same quality. I'm an expert for audio engineering. I did work for Brauner microphones development and others, hence I know a little bit about how to do tests ;). No dynamic filters are involved! It's very simple, there's a natural sounding drum set as example drum kit for Hydrogen. Play a rhythm, record this Rhythm and then record this recording and compare both recordings. They should be equal, but they aren't equal. I can here a !clear! loss and it's visible by spectral waves. Simply because a synth-software only delivers, what it renders to Jack and Jack does *not* change anything in that rendered data. There is simply not soundcard and not even a driver involved in the rendering itself. DSPs only do the very same thing faster as cheap chips. All difference in sound quality is related to DAC/ADC period No! Before any converter is involved, there at least could be rounding errors, if you don't use 32-bit float all the times. And by the way, the sound card will effect the original and the digital copy in the same way, even with a bad sound card both recordings shouldn't differ. Hey, do a recording of a recording and then run the diff command to compare them ;)! I have to wonder what you did there to alter the data from the soft synth. I mean ... we're talking bit-exact copy here, aren't we? Can you present a test setup to observe that issue? Any Linux install I know, e.g. 64 Studio 64-bit 3.0, 3.3, Suse 64-bit 11.2 and Edubuntu 32-bit Maverick + Ubuntu Studio meta packages and others! If you can't here it, try to see it. If you don't have this issue too, some people claim that they get 100% correct digital copies, then something on my machine might cause a software issue, but I don't think so. Ralf Alrighty then, Thomas. -- Ubuntu-Studio-users mailing list Ubuntu-Studio-users@lists.ubuntu.com Modify settings or unsubscribe at: https://lists.ubuntu.com/mailman/listinfo/ubuntu-studio-users
Re: Sound cards (Was: Re: no sound
On Fri, 2011-05-20 at 15:04 +0200, Hartmut Noack wrote: Am 20.05.2011 14:58, schrieb Ralf Mardorf: On Fri, 2011-05-20 at 14:36 +0200, Hartmut Noack wrote: Am 20.05.2011 13:54, schrieb Ralf Mardorf: On Fri, 2011-05-20 at 12:48 +0200, Robert Klaar wrote: Any Rme card is good, although they're a bit expensive. Didn't have to do a thing to get my hdsp 9632 working. Hi Robert :) the main reason to switch the sound cards is the audio sound quality. I'm able to pay around 700,- EUR / 800,- EUR (right now, regarding to my profession, I usually don't have any money). Btw. I noticed that not only my sound cards do cause loss. When recording soft synth just by JACK, without the sound cards being involved, there's a loss for the sound quality too! No, there is none. Any software, that generates sounds from scratch like a softsynth wil produce exactly the same stream with any soun card. In fact such software will even generate the very same stream if no soundcard exists. And if you record such a stream with Jack you simply store that very stream bit by bit. Wrong! There still could be rounding errors and dithering involved Yes, dithering makes a differnce -- if you use it. And do you think, sound cards or drivers provoke specific rounding errors in streams that are not even deliverd to them by Jack? No! But I suspect a bug for Jack and btw. there already was a rounding bug that was fixed, perhaps years ago. The you can't connect every client with every client, regarding to a zero-copy issue still was a topic some days ago on Jack devel mailing list, but I didn't follow the thread, just noticed it randomly. I won't lead on this discussion about a loss or no loss. When I noticed sync issues, MIDI jitter and loss for audio before, most from the community vetoed, then after a while the bugs I heard became a topic and bug-fixes too. I've got a good reputation regarding to professional audio engineering and a bad name by the Linux community. I don't care about this. A lot of people claimed that they don't have sync issues, but the coders said that e.g. there was a sync issue for Hydrogen, when using Jack transport, btw. I never used Hydrogen's step sequencer myself, but noticed similar issues, for other usages. Very often people simply aren't able to hear KNOWN issues. Talking, resp. writing about some bugs is useless. I need to live with those issues. Ralf a digital copy very often isn't a digital copy ;). But ok, here I didn't use dithering and I do use 32-bit float, but the result has a loss. I dunno what do cause this loss. For Jack there also is a zero-copy issue! You can't do wild connections using Jack, but you need to take care about the order, when e.g. connecting a client to itself. I assure you, you get the very same data in a recording via Jack if you play the same patch of the same synth on a work station with a RME Hammerfall or on a Laptop with a built-in HDA. The only level on wich a soundcard is related to a softsynth is the one on wich you actually hear the stream. And some synths can render differntly, if Jack is running at 96KHz instead of 48 or 44.1. But this has only remotely to do with the sound card let alone its quality. And soft synth already do sound less good than real old synth. Unfortunately those real old synth can break and there're no microchips available to repair those synth, resp. they are hard to get, very expensive and without warranty. My two TerraTec EWX 24/96 needs to be replaced, Tell me where you dump them, these 2 more stereo-dacs would be most welcome in my box ;-) All the trouble with the envy24-cards is related to (mis)configuration and to stupidities like automatically zeroing all channels caused by PA in most cases. As of now these problems can be solved by the user. They are *not* acceptable, they are bugs that need to be solved. But these bugs are not show-stoppers. before I don't have got the money anymore. FWIW S/PDIF doesn't work for my Ubuntu Studio with the TerraTecs, hence I can't use good analog IOs that would be available via S/PDIF. Btw. good analog IOs in this context does mean consumer DAT Sony DTC-670 and Aiwa HD-S1, both are without any loss of sound quality, when listening by my consumer equipment. At least good consumer sound quality is what I expect of a professional sound card, even if internal Linux there still would be loss caused by JACK or caused by what issue ever. Jack does not cause any loss in sound-quality because Jack does not have any influence on the way, the pcm-stream is produced by the driver/sound-card. Once Brauner borrowed me a Mac with a Motu firewire device. The MOTUs are quite okayish and they sound exactly the same on any system that supports them. the Mac's sound quality There is no such thing. No Mac-expert would endorse something like a special
Re: Sound cards (Was: Re: no sound
Am 20.05.2011 15:15, schrieb Ralf Mardorf: On Fri, 2011-05-20 at 15:00 +0200, Hartmut Noack wrote: Am 20.05.2011 14:37, schrieb Ralf Mardorf: On Fri, 2011-05-20 at 14:04 +0200, Thomas Orgis wrote: Am Fri, 20 May 2011 13:54:57 +0200 schrieb Ralf Mardorfralf.mard...@alice-dsl.net: When recording soft synth just by JACK, without the sound cards being involved, there's a loss for the sound quality too! Wait a minute... could you explain that? You have a loss of quality compared to live playback of the soft synths (using JACK?) when playing back a recording taken from JACK? A recording that preserves 32 bit floating point sample format (heck, or 24 bit integer) and the sample rate, of course? Yes and other people who can't hear it, do have it too. I do not. You can see it by watching the waves spectral by Audacity. I did this regarding to a zero-copy issue, that appears if a Jack client is connected directly to itself, e.g. to do the mastering. 48 and 96 KHz, 32-bit wav 32-bit float. If a synth has dynamic filters it will never produce the exactly same stream twice. But if you think about yourself you will find out, that given you use the same settings for Jack on a HDA or a HDSP you will get exactly the same quality. I'm an expert for audio engineering. I did work for Brauner microphones development and others, hence I know a little bit about how to do tests ;). No dynamic filters are involved! It's very simple, there's a natural sounding drum set as example drum kit for Hydrogen. Play a rhythm, record this Rhythm and then record this recording and compare both recordings. They should be equal, but they aren't equal. I can here a !clear! loss and it's visible by spectral waves. Simply because a synth-software only delivers, what it renders to Jack and Jack does *not* change anything in that rendered data. There is simply not soundcard and not even a driver involved in the rendering itself. DSPs only do the very same thing faster as cheap chips. All difference in sound quality is related to DAC/ADC period No! Before any converter is involved, there at least could be rounding errors, if you don't use 32-bit float all the times. And why should I not use 32bit float all the time? Of course there are differences, if format-conversion is involved. But you did not mention such conversions, you only talked about sound cards causing mysterious differences when Jack delivers a stream from a synth-application directly to a recorder. The normal, sane setup fpr recording a synth directly with Jack is, that the synth, Jack and the recorder all run with the same samplerate and 32bit float or at least all 3 with 16bit Int. And if that is set up like this, there is zero influence of the soundcard on the recording. And by the way, the sound card will effect the original and the digital copy in the same way, even with a bad sound card both recordings shouldn't differ. Hey, do a recording of a recording You mean, like recording something from ams via jack then play the wav-file with mhw and record this with ardour. Then compare the two recorded streams? If in such a process recording A would differ from recording B then MHW or Ardour *could* cause such a difference. Jack itself could only be charged, if resampling and/or dithering would be involved. That is: if the synth-engine would work with 44.1Hz while Jack runs with 48Hz. And you will not want to set up a synth like this. But you said, that there would be a difference between a 1st-level recording from a synth via Jack with sound card A and another recording of the same synth with soundcard B. And that is not the case. and then run the diff command to compare them ;)! I have to wonder what you did there to alter the data from the soft synth. Format conversion? Some synths (like ZynaddSubFX) can be configured to produce streams in a certain format (such as 44.1 Hz / 16bit Int) and still run with Jack that runs with different settings. Every body sets Zynadd to run with the same SR as Jack and Zynadd recommends that if started from the command line. I mean ... we're talking bit-exact copy here, aren't we? Can you present a test setup to observe that issue? Any Linux install I know, e.g. 64 Studio 64-bit 3.0, 3.3, Suse 64-bit 11.2 and Edubuntu 32-bit Maverick + Ubuntu Studio meta packages and others! If you can't here it, try to see it. If you don't have this issue too, some people claim that they get 100% correct digital copies, then something on my machine might cause a software issue, but I don't think so. Ralf Alrighty then, Thomas. -- Ubuntu-Studio-users mailing list Ubuntu-Studio-users@lists.ubuntu.com Modify settings or unsubscribe at: https://lists.ubuntu.com/mailman/listinfo/ubuntu-studio-users
Re: Sound cards (Was: Re: no sound
Ralph, the sound card *cant* have anything to do with the audio that is generated by software synths. this is not a grey area. and expert such as yourself might want to look at the source code for the software in question, and see what about the hardware is being utilized for rendering the audio. heres a test scenario: take all the sound devices out of the machine (or disable them). take a MIDI file and render it using JACK utilizing the 'dummy' driver. you can render the same file with a sound card in use, and share both of those here if you would like. *i am not talking about monitoring those sounds, OR recording them analog from the main outs of the sound card. by render, im thinking recording in ardour and exporting or exporting from something else. -- MH http://opensourcemusician.libsyn.com/ http://wnclug.ourproject.org/ -- Ubuntu-Studio-users mailing list Ubuntu-Studio-users@lists.ubuntu.com Modify settings or unsubscribe at: https://lists.ubuntu.com/mailman/listinfo/ubuntu-studio-users
Re: Sound cards (Was: Re: no sound
Am Fri, 20 May 2011 15:33:39 +0200 schrieb Ralf Mardorf ralf.mard...@alice-dsl.net: No! But I suspect a bug for Jack and btw. there already was a rounding bug that was fixed, perhaps years ago. Ah, OK. I'll ignore the discussion about different sound cards influencing this as a heap of confusion and settle for this: If JACK would not have bugs, by design it would do bit-exact copies, or at least copies with floating point computation errors in non-audible amplitude. I don't claim that this ideal case is indeed the reality. When you find those bugs, I hope they'll get reported and fixed, for the benefit to us all. In the meantime, those of us with some time on their hands can just try to verify the issue of bit-exact copying of data through the pipeline ... I just don't want to go down that path right now, since I'm already not able to do my actual work because of debugging some gcc 4.6 breakage on code of mine. Alrighty then, Thomas. signature.asc Description: PGP signature -- Ubuntu-Studio-users mailing list Ubuntu-Studio-users@lists.ubuntu.com Modify settings or unsubscribe at: https://lists.ubuntu.com/mailman/listinfo/ubuntu-studio-users
Re: Sound cards (Was: Re: no sound
On Fri, 2011-05-20 at 16:53 +0200, Hartmut Noack wrote: Am 20.05.2011 15:15, schrieb Ralf Mardorf: On Fri, 2011-05-20 at 15:00 +0200, Hartmut Noack wrote: Am 20.05.2011 14:37, schrieb Ralf Mardorf: On Fri, 2011-05-20 at 14:04 +0200, Thomas Orgis wrote: Am Fri, 20 May 2011 13:54:57 +0200 schrieb Ralf Mardorfralf.mard...@alice-dsl.net: When recording soft synth just by JACK, without the sound cards being involved, there's a loss for the sound quality too! Wait a minute... could you explain that? You have a loss of quality compared to live playback of the soft synths (using JACK?) when playing back a recording taken from JACK? A recording that preserves 32 bit floating point sample format (heck, or 24 bit integer) and the sample rate, of course? Yes and other people who can't hear it, do have it too. I do not. You can see it by watching the waves spectral by Audacity. I did this regarding to a zero-copy issue, that appears if a Jack client is connected directly to itself, e.g. to do the mastering. 48 and 96 KHz, 32-bit wav 32-bit float. If a synth has dynamic filters it will never produce the exactly same stream twice. But if you think about yourself you will find out, that given you use the same settings for Jack on a HDA or a HDSP you will get exactly the same quality. I'm an expert for audio engineering. I did work for Brauner microphones development and others, hence I know a little bit about how to do tests ;). No dynamic filters are involved! It's very simple, there's a natural sounding drum set as example drum kit for Hydrogen. Play a rhythm, record this Rhythm and then record this recording and compare both recordings. They should be equal, but they aren't equal. I can here a !clear! loss and it's visible by spectral waves. Simply because a synth-software only delivers, what it renders to Jack and Jack does *not* change anything in that rendered data. There is simply not soundcard and not even a driver involved in the rendering itself. DSPs only do the very same thing faster as cheap chips. All difference in sound quality is related to DAC/ADC period No! Before any converter is involved, there at least could be rounding errors, if you don't use 32-bit float all the times. And why should I not use 32bit float all the time? Of course there are differences, if format-conversion is involved. But you did not mention such conversions, you only talked about sound cards causing mysterious differences when Jack delivers a stream from a synth-application directly to a recorder. No, a misunderstanding. 1. Yes my sound card is bad, but ... 2. I was writing about using 32-bit float only and use audio streams internal Jack only, without the sound cards being involved. The normal, sane setup fpr recording a synth directly with Jack is, that the synth, Jack and the recorder all run with the same samplerate and 32bit float or at least all 3 with 16bit Int. And if that is set up like this, there is zero influence of the soundcard on the recording. Correct, there's no influence of the sound cards, but on my machine there's loss, even when the sound cards aren't involved, just by recording a soft synth. Everything is set to the same sample rate, 96 KHz or 48 KHz and 32-bit float ... as far as I know ... I don't know if e.g. Yoshimi does use 32-bit float. And by the way, the sound card will effect the original and the digital copy in the same way, even with a bad sound card both recordings shouldn't differ. Hey, do a recording of a recording You mean, like recording something from ams via jack then play the wav-file with mhw and record this with ardour. Then compare the two recorded streams? If in such a process recording A would differ from recording B then MHW or Ardour *could* cause such a difference. Jack itself could only be charged, if resampling and/or dithering would be involved. That is: if the synth-engine would work with 44.1Hz while Jack runs with 48Hz. And you will not want to set up a synth like this. But you said, that there would be a difference between a 1st-level recording from a synth via Jack with sound card A and another recording of the same synth with soundcard B. And that is not the case. and then run the diff command to compare them ;)! I have to wonder what you did there to alter the data from the soft synth. Format conversion? Some synths (like ZynaddSubFX) can be configured to produce streams in a certain format (such as 44.1 Hz / 16bit Int) and still run with Jack that runs with different settings. Every body sets Zynadd to run with the same SR as Jack and Zynadd recommends that if started from the command line. I mean ... we're talking bit-exact copy here, aren't we? Can you present a test setup to observe that issue? Any Linux install I know, e.g. 64 Studio 64-bit 3.0, 3.3,
Re: Sound cards (Was: Re: no sound
On Fri, 2011-05-20 at 10:54 -0400, Mike Holstein wrote: Ralph, the sound card *cant* have anything to do with the audio that is generated by software synths. Again, a misunderstanding, perhaps regarding to my broken English. There're two issues, one is caused by my sound cards, they do produce a muddy bass, the other issue has nothing to do with my sound cards. A recording internal Linux, without the sound cards involved does cause loss here. Audible loss, you don't need trained or good ears to hear it. this is not a grey area. and expert such as yourself might want to look at the source code for the software in question I can't read C and much more worse, never coded for modern PCs. , and see what about the hardware is being utilized for rendering the audio. heres a test scenario: take all the sound devices out of the machine (or disable them). take a MIDI file and render it using JACK utilizing the 'dummy' driver. you can render the same file with a sound card in use, and share both of those here if you would like. *i am not talking about monitoring those sounds, OR recording them analog from the main outs of the sound card. by render, im thinking recording in ardour and exporting or exporting from something else. No need to do this, because ... again ... the sound cards already aren't involved. This was a misunderstanding. Btw. doing the mastering for a real song might not work by an exporting option. At least Qtractor can't do this for faster than real-time included apps, in other words, DSSI plugins and especially not for inserts, e.g. jconvolver. That I wish to buy a good sound card has nothing to do with the loss internal Linux on my machine (and I suspect on many other machines too, at least I know some people who do have the same issue). -- MH http://opensourcemusician.libsyn.com/ http://wnclug.ourproject.org/ -- Ubuntu-Studio-users mailing list Ubuntu-Studio-users@lists.ubuntu.com Modify settings or unsubscribe at: https://lists.ubuntu.com/mailman/listinfo/ubuntu-studio-users
Re: Sound cards (Was: Re: no sound
On Sat, 2011-05-21 at 00:30 +0200, Thomas Orgis wrote: Am Fri, 20 May 2011 15:33:39 +0200 schrieb Ralf Mardorf ralf.mard...@alice-dsl.net: No! But I suspect a bug for Jack and btw. there already was a rounding bug that was fixed, perhaps years ago. Ah, OK. I'll ignore the discussion about different sound cards influencing this as a heap of confusion and settle for this: If JACK would not have bugs, by design it would do bit-exact copies, or at least copies with floating point computation errors in non-audible amplitude. I don't claim that this ideal case is indeed the reality. When you find those bugs, I hope they'll get reported and fixed, for the benefit to us all. In the meantime, those of us with some time on their hands can just try to verify the issue of bit-exact copying of data through the pipeline ... I just don't want to go down that path right now, since I'm already not able to do my actual work because of debugging some gcc 4.6 breakage on code of mine. Alrighty then, Thomas. Is there a way of monitoring the data that is captured and read by Jack and Jack clients in a human readable way, e.g. to see if there would be rounding errors? I always used Audacity's spectral view (to show recordings of square and sine waves), when writing with Rui. For Qtractor there really was and is an issue when connecting it's outputs to the inputs, that is caused by Jack. Phew, a lot of OT blah blah, you don't need to read does follow, pardon: (That's why I wish to stop the discussion about the audio quality loss issue.) I guess it's because a vector points to the original buffer, instead of coping it, but I might be completely wrong. Anyway, if a client's input is connected to it's output this only should work correctly, if there's a special order. Here I did avoid to do this and there still is loss, similar to cheap 4-Track cassette recorders. If I directly connect Hydrogen or Yoshimi to Qtractor or Ardour2, that does mean one client to _another_ client, there's this kind of audible loss and this shouldn't happen :(. I can't help with reading the source code. When I programmed in C 20 years ago, I just did it for one program and then switched back to Assembler. On Linux I tried to learn C again, but I wasn't able to write even simplest make files, IIRC 20 years ago it was the work, the compiler had to do ;), I could take a look on my Atari ST's 80286 hardware emulation, there still should be the old editor and compiler. So, I'm not a coder ;). I'm an audio/video engineer and the computers I privately programmed have less in common with current PCs, e.g. the Atari's TOS and the emulation running DR DOS, are sharing a 42MB hard disk. Especially regarding to music software, there was the advantage that those machines don't do real multitasking and the hardware, e.g. for the C64 MIDI interface, was directly accessible, that's why there's no MIDI jitter for old hardware. Turn of the interrupt, check the ACIA/UART and send in real-time, that's how it did work years ago, today we do have USB protocols etc. that at all events do cause MIDI jitter. Some smart guys, I guess Stéphane is one of them, do work on this issue and they/he already had success, at least for my machine. If I use PCI MIDI and edubuntu@edubuntu:~$ jackd -V jackdmp 1.9.8 using the -Xalsarawmidi switch + a2jmidi_bridge I'm able to use external MIDI equipment without audible jitter, unfortunately I need to disable -Xalsarawmidi, if I wish to record MIDI events with a Linux sequencer, because there's no bridge vice versa and as far as I know no good sequencer using Jack MIDI, resp. I guess Ardour3 might use Jack MIDI. Btw. if I run latency tests I also get best results for Alsa, but at least I'm able to hear that there's jitter, a lot of people with less good results aren't able to hear jitter. There're a lot of issues for Linux audio, some are already solved by svn versions, other issues aren't solved. It makes me wonder that there are so less people in the Linux community who notice those issues. I can't program myself, but I could run tests and report issues. Linux audio and MIDI IMO do need a lot of upgrades and some isolation from the non-audio community. -- Ubuntu-Studio-users mailing list Ubuntu-Studio-users@lists.ubuntu.com Modify settings or unsubscribe at: https://lists.ubuntu.com/mailman/listinfo/ubuntu-studio-users
