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#1
Posted to rec.audio.tech
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need help to set up a M-Audio Audiophile 2496
Hi to Everyone,
Could someone help me managing my new audio card ? It operates under Debian Linux, but I also tested it under Windows XP, with same results. The input volume for digitilizing vinyl records is too low, and I can't enhance it. On the contrary, with my old audio card that was part of my ASUS motherboard, I was getting to high input volumes and I had to lower the input to 7 to 13% of the maximum so as to avoid saturating, such low settings were not in the green area for recording. Here is what I get with Alsamixer : Alsamixer v1.0.13 Card: M Audio Audiophile 24/96 Chip: ICE 1712 - multitrack View: Playback Capture [All] Item : list follows: Item: IEC958 set on H/W In 0 Item: IEC958 Multi. Was in 'L CAPTUR R' mode, but I de-activated it, to no avail. I am not in multichannel mode, only in stereo. Item: IEC958 Multi 1: same thing Item: IEC958 1 set to 'H/W In 1' other options for IEC958 et IEC958 1 were H/W In 2 ... H/W In 7, IEC958 In L, IEC958 In R, PCM Out, Digital Mixer. Item: ADC (Analog to Digital Controller ??) set to 55%, dB gain = -18.5 (in the green area) Item: ADC1: idem : dB gain=-18.5 Item: ADC Capture. dB gain=-23 (in the green) Item: ADC1 Capture. dB gain=-23 (in the green) Item: DAC (Digital to Analog Controller ??) dB gain=-23 idem Item: DAC Capture -23 dB Item: DAC1 -23 dB Item DAC1 Capture -23 dB Item: Deemphasis. Set to 48 kHz. Other options : 32, 44.1, OFF Item: H/W set to H/W In 0 Item: H/W1 set to H/W In 1 other options : H/W In 2, H/W In 3, .... H/W In 7, IEC958 In L, IEC958 In R Item: H/W Multi L CAPTUR R CAPTUR de-activated Item: H/W Multi1 L CAPTUR R de-activated Items: Multi, Multi1, Multi2... Multi9 set to Mute (MM) Item: Multi Track Internal Clock : 44100 Item: Multi Track Internal Clock Default : 44100 Item: Multi Track Peak 5656 Item: Multi Track Peak 6969 (two homonymous items, same for the next two) Item: Multi Track Peak. Here a single figure : 69 Item: Multi Track Peak. Same thing, 69 Item: Multi Track Capture. Set to 86, against my will. The green area starts at 39, but if I rise the setting, it comes back to low values after restarting 'alsamixer'. True enough, I am not in multichannel mode, so chances are that this setting is obsolete here. Items: Third and fourth 'multi track peak' Item: Multi Track Rate Locking. 2 options : Mute ou 00. Item: Multi Track Rate Reset: idem Item: Multi Track Volume Rate : 67 Thanks in advance for comments and advises. |
#2
Posted to rec.audio.tech
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need help to set up a M-Audio Audiophile 2496
Bernard wrote:
Could someone help me managing my new audio card ? It operates under Debian Linux, but I also tested it under Windows XP, with same results. The input volume for digitilizing vinyl records is too low, and I can't enhance it. You must use an RIAA phono preamp between the pickup cartridge and the line input to the sound card. AFAIK, there are NO computer audio cards which have inputs appropriate for phono cartrdges (47K input impedance, high gain, and the RIAA equalization curve). |
#3
Posted to rec.audio.tech
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need help to set up a M-Audio Audiophile 2496
Le Mon, 13 Jul 2009 13:13:14 -0700, Richard Crowley a écrit*:
Bernard wrote: Could someone help me managing my new audio card ? It operates under Debian Linux, but I also tested it under Windows XP, with same results. The input volume for digitilizing vinyl records is too low, and I can't enhance it. You must use an RIAA phono preamp between the pickup cartridge and the line input to the sound card. AFAIK, there are NO computer audio cards which have inputs appropriate for phono cartrdges (47K input impedance, high gain, and the RIAA equalization curve). Yes, I do have a RIAA phono preamp between the two. |
#4
Posted to rec.audio.tech
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need help to set up a M-Audio Audiophile 2496
"Bernard" wrote ...
Richard Crowley a écrit : Bernard wrote: It operates under Debian Linux, but I also tested it under Windows XP, with same results. The input volume for digitilizing vinyl records is too low, and I can't enhance it. You must use an RIAA phono preamp between the pickup cartridge and the line input to the sound card. AFAIK, there are NO computer audio cards which have inputs appropriate for phono cartrdges (47K input impedance, high gain, and the RIAA equalization curve). Yes, I do have a RIAA phono preamp between the two. Then either the phono preamp output is too low or the 2496 card input gain is too low. The common differential diagnosis techniques would quickly identify which was the case. |
#5
Posted to rec.audio.tech
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need help to set up a M-Audio Audiophile 2496
In article , Richard Crowley wrote: It operates under Debian Linux, but I also tested it under Windows XP, with same results. The input volume for digitilizing vinyl records is too low, and I can't enhance it. You must use an RIAA phono preamp between the pickup cartridge and the line input to the sound card. AFAIK, there are NO computer audio cards which have inputs appropriate for phono cartrdges (47K input impedance, high gain, and the RIAA equalization curve). Yes, I do have a RIAA phono preamp between the two. Then either the phono preamp output is too low or the 2496 card input gain is too low. The common differential diagnosis techniques would quickly identify which was the case. One possibility, I suppose, is that the OP is using a low-output moving-coil cartridge. These require some form of additional step-up device (either a transformer, or a "head amp") prior to the normal RIAA preamplifier. It might also just be a relatively low-level moving-iron or moving-magnet cartridge. Another possibility is that the RIAA preamp was designed for the older consumer standard ("full output" being 1 volt peak-to-peak) but that the Audiophine 2496 is designed for the newer (CD-era) consumer standard ("full scale input" being 2 volts peak-to-peak). This would result in a 50% reduction in apparent input level. An additional question for the OP: just how much "too low" is the signal level once you've digitized it? If the highest peak volume level you're seeing on your LPs is, say, up to 25% of full-scale, then I'd actually recommend that you not fiddle at all with the input gain. If you need to normalize, do so in the digital domain after capturing. Here's why. There can be a fairly wide variation in the peak signal level of LPs. Different manufacturers "cut" the albums differently. There's a complex tradeoff between maximum cutting level, play time per side, background noise level, and distortion. I've had some particularly aggressive albums whose peak levels seemed to be almost 10 dB above that of albums with a quieter cutting (although this is a fairly extreme case). In my opinion, when digitizing LPs, the most important thing to do is make sure that the highest peak level, on the "hottest" LP, doesn't clip. Clipping sounds really nasty, and cannot be corrected properly. The dynamic range of even a 16-bit audio capture is significantly greater than the dynamic range of an LP... even that of a mint-condition, never-before-played LP pressed on really good virgin vinyl. You can afford to throw away a couple of bits of dynamic range at the top end, by having your average input level down at half- or quarter-scale, in order to avoid the risk of clippping; the quietest passages on the LP, and the residual surface noise will still be comfortably far above the 16-bit quantization threshold. This is even more true if you have a "24-bit" converter, and can capture audio in a 24-bit format. I say "24-bit" because such converters usually don't have a true 24-bit dynamic range... last time I looked the better ones were in the 21- to 22-bit range. Nevertheless, there's plenty of range here - quite a bit more than an LP requires. Best recipe: capture in a 24-bit format (with the input gain adjusted to keep the worst-case peak on a really loud record some distance below full-scale), edit (clean up pops and ticks), then convert to 16-bit format (gain-boosting to normalize while still avoiding clipping, if you wish) using a decent dithering algorithm. -- Dave Platt AE6EO Friends of Jade Warrior home page: http://www.radagast.org/jade-warrior I do _not_ wish to receive unsolicited commercial email, and I will boycott any company which has the gall to send me such ads! |
#6
Posted to rec.audio.tech
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need help to set up a M-Audio Audiophile 2496
Bernard wrote:
Could someone help me managing my new audio card ? It operates under Debian Linux, but I also tested it under Windows XP, with same results. The input volume for digitilizing vinyl records is too low, and I can't enhance it. On the contrary, with my old audio card that was part of my ASUS motherboard, I was getting to high input volumes and I had to lower the input to 7 to 13% of the maximum so as to avoid saturating, such low settings were not in the green area for recording. Here is what I get with Alsamixer : Use m-audio's driver software. You may prefer not to use the newest version, the newest version does not let you record the output of the monitor mixer as I recall this. It is generally preferable to use a complete hifi-preamp (or "pre-out" from an integrated with a patch point beteeen preamp and poweramp) and use the output to power-amplifier, that gives you a record volume control. The next best thing may be to use the headphone output. Tape out may however be ok and it is likely to be the cleanest sounding choice. It doesn't really matter with a 24 bit card if the recorded audio generally is at -10 to -6 dB, something that is quite likely with this card and a ""standard riaa amp and cartridge"", it is still cleaner to adjust level afterwards AND you DO need to record all scratches cleanly. The concept of an AD conversion card as this is is that there is that it is a line in card with no controls, the only settings are for matching the output level. Alsamixer v1.0.13 Unless this software is broken it should work to select the cards inputs in the recording software and leave everything in their default settings. My recollection is - it may be wrong - that there is linux driver and software available on m-audio's site, perhaps you should use that. Thanks in advance for comments and advises. Kind regards Peter Larsen |
#7
Posted to rec.audio.tech
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need help to set up a M-Audio Audiophile 2496
Bernard wrote:
Hi to Everyone, Could someone help me managing my new audio card ? It operates under Debian Linux, but I also tested it under Windows XP, with same results. The input volume for digitilizing vinyl records is too low, and I can't enhance it. On the contrary, with my old audio card that was part of my ASUS motherboard, I was getting to high input volumes and I had to lower the input to 7 to 13% of the maximum so as to avoid saturating, such low settings were not in the green area for recording. Here is what I get with Alsamixer : Alsamixer v1.0.13 Card: M Audio Audiophile 24/96 Chip: ICE 1712 - multitrack View: Playback Capture [All] Item : list follows: Item: IEC958 set on H/W In 0 Item: IEC958 Multi. Was in 'L CAPTUR R' mode, but I de-activated it, to no avail. I am not in multichannel mode, only in stereo. Item: IEC958 Multi 1: same thing Item: IEC958 1 set to 'H/W In 1' other options for IEC958 et IEC958 1 were H/W In 2 ... H/W In 7, IEC958 In L, IEC958 In R, PCM Out, Digital Mixer. Item: ADC (Analog to Digital Controller ??) set to 55%, dB gain = -18.5 (in the green area) Item: ADC1: idem : dB gain=-18.5 Item: ADC Capture. dB gain=-23 (in the green) Item: ADC1 Capture. dB gain=-23 (in the green) Item: DAC (Digital to Analog Controller ??) dB gain=-23 idem Item: DAC Capture -23 dB Item: DAC1 -23 dB Item DAC1 Capture -23 dB Item: Deemphasis. Set to 48 kHz. Other options : 32, 44.1, OFF Item: H/W set to H/W In 0 Item: H/W1 set to H/W In 1 other options : H/W In 2, H/W In 3, ... H/W In 7, IEC958 In L, IEC958 In R Item: H/W Multi L CAPTUR R CAPTUR de-activated Item: H/W Multi1 L CAPTUR R de-activated Items: Multi, Multi1, Multi2... Multi9 set to Mute (MM) Item: Multi Track Internal Clock : 44100 Item: Multi Track Internal Clock Default : 44100 Item: Multi Track Peak 5656 Item: Multi Track Peak 6969 (two homonymous items, same for the next two) Item: Multi Track Peak. Here a single figure : 69 Item: Multi Track Peak. Same thing, 69 Item: Multi Track Capture. Set to 86, against my will. The green area starts at 39, but if I rise the setting, it comes back to low values after restarting 'alsamixer'. True enough, I am not in multichannel mode, so chances are that this setting is obsolete here. Items: Third and fourth 'multi track peak' Item: Multi Track Rate Locking. 2 options : Mute ou 00. Item: Multi Track Rate Reset: idem Item: Multi Track Volume Rate : 67 Thanks in advance for comments and advises. Bernard, you are treading a well worn path. I have been doing what you describe for a number of years with a moving coil cartridge, NAD PP2 pre amp and the software levels set at max. Regardless of operating system, I use my analog mixer between the pre amp and M-Audio card to raise the level and often to raise the treble slightly. Best wishes, Alan. |
#8
Posted to rec.audio.tech
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need help to set up a M-Audio Audiophile 2496
On Mon, 13 Jul 2009 13:13:14 -0700, "Richard Crowley"
wrote: Bernard wrote: Could someone help me managing my new audio card ? It operates under Debian Linux, but I also tested it under Windows XP, with same results. The input volume for digitilizing vinyl records is too low, and I can't enhance it. You must use an RIAA phono preamp between the pickup cartridge and the line input to the sound card. AFAIK, there are NO computer audio cards which have inputs appropriate for phono cartrdges (47K input impedance, high gain, and the RIAA equalization curve). RIAA computer audio cards http://www.terratec.net/en/products/...inyl_2064.html http://www.hama.de/portal/searchSele...bySearch*40515 |
#9
Posted to rec.audio.tech
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need help to set up a M-Audio Audiophile 2496
Le Tue, 14 Jul 2009 07:07:00 +0100, Peter Larsen a écrit*:
Bernard wrote: Could someone help me managing my new audio card ? It operates under Debian Linux, but I also tested it under Windows XP, with same results. The input volume for digitilizing vinyl records is too low, and I can't enhance it. On the contrary, with my old audio card that was part of my ASUS motherboard, I was getting to high input volumes and I had to lower the input to 7 to 13% of the maximum so as to avoid saturating, such low settings were not in the green area for recording. Here is what I get with Alsamixer : Use m-audio's driver software. You may prefer not to use the newest version, the newest version does not let you record the output of the monitor mixer as I recall this. Ever since I wrote my first post, I have found this link, where my audio card is being discussed by users. All comments are positive, except for the first one on the list, and it just happens that the guy purchased that card for the same purpose as mine, that is, digitilizing vinyl records. He also found input levels too low, and digital 'burps' in his audio samples (I don't know what that is, and I doubt if I have these). He also says that that card did a poor sampling job, compared to his former audio card. All other comments are positive, but no one else seems to have digitilized vinyls with that card... http://www.newegg.com/Product/Produc...e=1 00&Page=1 However, it is not sure whether this guy's problems would reproduce on my system. At the end of this post, I will propose links for a few 30 seconds samples that I obtained : maybe someone could then tell me if he or she notices such digital burps or other defects besides scratches, and what to think of the quality of sampling and of background noise. It is generally preferable to use a complete hifi-preamp (or "pre-out" from an integrated with a patch point beteeen preamp and poweramp) and use the output to power-amplifier, that gives you a record volume control. The next best thing may be to use the headphone output. Great idea : I just tested it ! Whatever comes from this output has gone through the amplifier, while what comes from Tape Out has only gone through the RIAA adjustment. So, I can enhance the output level so that it gets close to 0 dB for treatment and record. As you point out, processing that way may not be the best strategy for quality, and the "phones" output is obviously not meant for that purpose. I propose samples below : http://www.teaser.fr/~bdebreil/musiq...Mozart.wav.bz2 is a 30 sec sample of a vinyl record that has been digitilized using my M-Audio 2496 audio card. No treatment was applied to the files, except an export to 32bits float wav file (and then losslessly compressed using bzip2) http://www.teaser.fr/~bdebreil/musiq...t_ampl.wav.bz2 is the same music sample, but I have amplified it of 7.3 dB using audacity. http://www.teaser.fr/~bdebreil/musiq...phones.wav.bz2 Is another digitilizing of the same music sample, processed from the headphones output, amplification adjusted on the hardware amplifier so as to approach 0 dB. No treatment was done afterwards. Thanks in advance for your comments. |
#10
Posted to rec.audio.tech
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need help to set up a M-Audio Audiophile 2496
Bernard wrote:
Ever since I wrote my first post, I have found this link, where my audio card is being discussed by users. All comments are positive, except for the first one on the list, and it just happens that the guy purchased that card for the same purpose as mine, that is, digitilizing vinyl records. He also found input levels too low, Give it more input then. Seriously: it is already too sensitive to reall give you 24 bit AD conversion because of the analog noise floor. and digital 'burps' in his audio samples (I don't know what that is, and I doubt if I have these). That guy has something else broke in his setup, perhaps an on board graphics card, perhaps something networking that needs to be disabled in that specific box. Or his actual card is defective out of the box. He also says that that card did a poor sampling job, compared to his former audio card. All other comments are positive, but no one else seems to have digitilized vinyls with that card... It is the card I use and I use it as I suggested that you should, ie. with a japanese preamp from the age o the quality wars and via tape out, levels match perfectly, ie. I can digitize the loudest scratch without clipping and it is REQUIRED to be able to do just that if you want optimum results. Thanks in advance for your comments. I didn't read the rest ... busy morning, bear with me. Kind regards Peter Larsen |
#11
Posted to rec.audio.tech
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need help to set up a M-Audio Audiophile 2496
"Ken" wrote ...
"Richard Crowley" wrote: Bernard wrote: Could someone help me managing my new audio card ? It operates under Debian Linux, but I also tested it under Windows XP, with same results. The input volume for digitilizing vinyl records is too low, and I can't enhance it. You must use an RIAA phono preamp between the pickup cartridge and the line input to the sound card. AFAIK, there are NO computer audio cards which have inputs appropriate for phono cartrdges (47K input impedance, high gain, and the RIAA equalization curve). RIAA computer audio cards http://www.terratec.net/en/products/...inyl_2064.html http://www.hama.de/portal/searchSele...bySearch*40515 Not exactly plug-in I/O cards, but thanks. OTOH, the sound card cited does NOT have RIAA inputs and no mention of a phono preamp was made in the OP's initial post. |
#12
Posted to rec.audio.tech
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need help to set up a M-Audio Audiophile 2496
"Bernard" wrote ...
Ever since I wrote my first post, I have found this link, where my audio card is being discussed by users. All comments are positive, except for the first one on the list, and it just happens that the guy purchased that card for the same purpose as mine, that is, digitilizing vinyl records. He also found input levels too low, Unless your card is broken I can't see how "digitizing vinyl records" has anything to do with "levels too low". If you are using an RIAA phono preamp which is putting out the proper line level, it should drive the M-Audio sound card perfectly. I am quite certain that hundreds (or even thousands) of people are digitizing vinyl records with this M-Audio card with no problems at all. Repeating: Test both your RIAA phono preamp AND your M-Audio card with other equipment to determine if the preamp is outputing the nominal consumer line level signal. And try some other source (such as a CD player, etc.) as the source into the M-Audio card to establish whether that is the problem. Nothing you have written causes me to believe that there is anything intrinsicly problematic with the M-Audio card that causes it to be unsuitable for digitizing vinyl records. You appear to have a level- setting problem and so far you have not reported doing any of the recommended tests to establish where the problem lies. Only you can do this experiment, we cannot guess how your hardware is working, especially at this distance. and digital 'burps' in his audio samples (I don't know what that is, and I doubt if I have these). He also says that that card did a poor sampling job, compared to his former audio card. All other comments are positive, but no one else seems to have digitilized vinyls with that card... http://www.newegg.com/Product/Produc...e=1 00&Page=1 Nothing about that comment makes me believe there is anything wrong with the M-Audio card. It seems quite obvious that the person has not set up his hardware and/or software properly. ...I propose samples below : http://www.teaser.fr/~bdebreil/musiq...Mozart.wav.bz2 If you post files that only Linux users can easily decrypt, then you can expect answers from only Linux users and there are very very few (if any?) of them here. You might have better success taking your question over to a Linux newsgroup. But PLEASE do not cross-post anything from Linux-land to this newsgroup. The past results are consistently disasterous. |
#13
Posted to rec.audio.tech
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need help to set up a M-Audio Audiophile 2496
Richard Crowley wrote:
Nothing you have written causes me to believe that there is anything intrinsicly problematic with the M-Audio card that causes it to be unsuitable for digitizing vinyl records. it is excellent for the purpose and fits the tape-out level on my preamp perfectly when using a high output MC. The record level may not be louder than what allows a clean recording of the loudest possible click on a record. Kind regards Peter Larsen |
#14
Posted to rec.audio.tech
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need help to set up a M-Audio Audiophile 2496
Le Wed, 15 Jul 2009 10:34:01 -0700, Richard Crowley a écrit*:
"Bernard" wrote ... Ever since I wrote my first post, I have found this link, where my audio card is being discussed by users. All comments are positive, except for the first one on the list, and it just happens that the guy purchased that card for the same purpose as mine, that is, digitilizing vinyl records. He also found input levels too low, Unless your card is broken I can't see how "digitizing vinyl records" has anything to do with "levels too low". If you are using an RIAA phono preamp which is putting out the proper line level, it should drive the M-Audio sound card perfectly. I am quite certain that hundreds (or even thousands) of people are digitizing vinyl records with this M-Audio card with no problems at all. Repeating: Test both your RIAA phono preamp AND your M-Audio card with other equipment to determine if the preamp is outputing the nominal consumer line level signal. And try some other source (such as a CD player, etc.) as the source into the M-Audio card to establish whether that is the problem. Nothing you have written causes me to believe that there is anything intrinsicly problematic with the M-Audio card that causes it to be unsuitable for digitizing vinyl records. You appear to have a level- setting problem and so far you have not reported doing any of the recommended tests to establish where the problem lies. Only you can do this experiment, we cannot guess how your hardware is working, especially at this distance. and digital 'burps' in his audio samples (I don't know what that is, and I doubt if I have these). He also says that that card did a poor sampling job, compared to his former audio card. All other comments are positive, but no one else seems to have digitilized vinyls with that card... http://www.newegg.com/Product/Produc...e=1 00&Page=1 Nothing about that comment makes me believe there is anything wrong with the M-Audio card. It seems quite obvious that the person has not set up his hardware and/or software properly. ...I propose samples below : http://www.teaser.fr/~bdebreil/musiq...Mozart.wav.bz2 If you post files that only Linux users can easily decrypt, then you can expect answers from only Linux users and there are very very few (if any?) of them here. You might have better success taking your question over to a Linux newsgroup. But PLEASE do not cross-post anything from Linux-land to this newsgroup. The past results are consistently disasterous. I don't know anything about 'Linux-land', assuming that this is not a generic word. The files which I have proposed links for downloading, are fully workable under MSWIN systems, as I have tested myself. 'Audacity', which is the software I am using for my purpose, is available for MSWIN operating system ; I even tested it under Windows XP. The only thing I have mentioned is the uncompressing of bz2 files. I made sure that 'bunzip' existed as MSWIN version, but I am quite aware that not everyone may have this software on his/her system. For your information and that of everyone else in this group, I have rightfully purchased the Windows XP version that I have installed in a partition of my HDs, even though I only boot on this system maybe once a month or so : this is my choice, and I have no complain about this. I am in favour of GNU software, still being fully aware that this would not exist if such commercial work as that of Microsoft had not existed. If you or anyone else are willing to test those links, I am quite prepared to propose other files without compression, so as to avoid installing weird compressing software. |
#15
Posted to rec.audio.tech
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need help to set up a M-Audio Audiophile 2496
"Bernard" wrote ...
If you or anyone else are willing to test those links, I am quite prepared to propose other files without compression, so as to avoid installing weird compressing software. I think it would be a very good idea for you to use a more familiar file type if you want help from people (most of whom are NOT using Linux). I was unable to find anything that would decode BZ2 files under MS Windows. |
#16
Posted to rec.audio.tech
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need help to set up a M-Audio Audiophile 2496
Le Wed, 15 Jul 2009 15:09:41 -0700, Richard Crowley a écrit*:
"Bernard" wrote ... If you or anyone else are willing to test those links, I am quite prepared to propose other files without compression, so as to avoid installing weird compressing software. I think it would be a very good idea for you to use a more familiar file type if you want help from people (most of whom are NOT using Linux). I was unable to find anything that would decode BZ2 files under MS Windows. I am very sorry for this. I had just checked that bzip2 and bunzip2 did exist for Windows platforms, but I should have thought that it may have not been so easy and quick to get it and install. I have prepared new files : 2 are uncompressed ; you will find them adequate if you have a fast DSL connexion. Two more are zip compressed: 'Winzip' will handle them as did 'pkzip' and 'pkunzip' in the old days of MSDOS 6.2. The original uncompressed files are about 10 mB, the zip files about 6 mB (the bz2 were about 4.4 mB). Other compressing lossless utilities that I know don't handle that sort of files (FLAC only compress 16 bit audio files, with a ratio that is about 2:1) Here are the new links below : http://www/teaser.fr/~bdebreil/musiq...ozart_ampl.wav http://www/teaser.fr/~bdebreil/musiq...art_phones.wav http://www/teaser.fr/~bdebreil/musiq...t_ampl.wav.zip http://www/teaser.fr/~bdebreil/musiq...phones.wav.zip |
#17
Posted to rec.audio.tech
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need help to set up a M-Audio Audiophile 2496
Le Thu, 16 Jul 2009 14:33:16 +0200, Bernard a écrit*:
Le Wed, 15 Jul 2009 15:09:41 -0700, Richard Crowley a écrit*: "Bernard" wrote ... If you or anyone else are willing to test those links, I am quite prepared to propose other files without compression, so as to avoid installing weird compressing software. I think it would be a very good idea for you to use a more familiar file type if you want help from people (most of whom are NOT using Linux). I was unable to find anything that would decode BZ2 files under MS Windows. I am very sorry for this. I had just checked that bzip2 and bunzip2 did exist for Windows platforms, but I should have thought that it may have not been so easy and quick to get it and install. I have prepared new files : 2 are uncompressed ; you will find them adequate if you have a fast DSL connexion. Two more are zip compressed: 'Winzip' will handle them as did 'pkzip' and 'pkunzip' in the old days of MSDOS 6.2. The original uncompressed files are about 10 mB, the zip files about 6 mB (the bz2 were about 4.4 mB). Other compressing lossless utilities that I know don't handle that sort of files (FLAC only compress 16 bit audio files, with a ratio that is about 2:1) Here are the new links below : http://www/teaser.fr/~bdebreil/musiq...ozart_ampl.wav http://www/teaser.fr/~bdebreil/musiq...art_phones.wav http://www/teaser.fr/~bdebreil/musiq...t_ampl.wav.zip http://www/teaser.fr/~bdebreil/musiq...phones.wav.zip I am even more sorry for these links , especially for the typing mistake. Here are the valid links below : http://www.teaser.fr/~bdebreil/musiq...ozart_ampl.wav http://www.teaser.fr/~bdebreil/musiq...art_phones.wav http://www.teaser.fr/~bdebreil/musiq...t_ampl.wav.zip http://www.teaser.fr/~bdebreil/musiq...phones.wav.zip |
#18
Posted to rec.audio.tech
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need help to set up a M-Audio Audiophile 2496
Bernard wrote:
I am even more sorry for these links , especially for the typing mistake. Here are the valid links below : http://www.teaser.fr/~bdebreil/musiq...ozart_ampl.wav http://www.teaser.fr/~bdebreil/musiq...art_phones.wav Thank you, it appears to me that you do not have a problem, but if you get the forte at offset 5 seconds at -6 or even -10 dB via tape out, then that is how you should record a record. As I have said previously you need to allow for making a clean recording of the loudest scratch or click because fixing it requires it to be properly recorded. You will also very soon start wanting to fix the channel imbalance, there is no such thing as a "no processing" transfer .... What I do with vinyl is to equalize the actual cartridge + riaa combination to linear based on playback of a bruel and kjaer test-record I got at some equipment demonstration way long time ago and then I do a manual "fix single click" of the large clicks and nothing else. I have decided on this strategy after detecting just how many clicks automated click fixing routines will find and fix in pristine recordings causing them to sound plain boring. As in the old days: if clicks and "needle noise" is a problem in playback, then you should fix the playback system. Note: you should NOT try to compensate for the treble roll-off innermost on the record, leave it be. With some records it seems to me that the "innermost bands" are cut at lower level so as to make room for more music, that I compensate some of of the time, but not always fully. You need to respect the artistic choices, the Droit Morale. I always digitize to 16 bit 96 kHz sample rate and downsample, my preferred audio software has very good downsampling. It would have merit to digitize to cd-format with the target being CD's, but generally no grammophone playback is acceptable without fixing frequency response errors and gross channel imbalance, I usually normalize to -2.5 dB after fixing large clicks manually with "fix single click functionality". Software that doesn't offer that functionality is toy software, but - that said - it does happen that I use some kind of automated repair, generally I prefer the clean sound of doing less. Kind regards Peter Larsen |
#19
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need help to set up a M-Audio Audiophile 2496
"Bernard" wrote ...
Here are the new links below : http://www/teaser.fr/~bdebreil/musiq...ozart_ampl.wav http://www/teaser.fr/~bdebreil/musiq...art_phones.wav I am not sure what these files are an example of, but they both appear to be completely normal to me. There is some very slight clipping on some peaks around 3.4 seconds into the "phones" file. But if these are the levels you are getting for the original recording, you are better at setting levels than I am. I would have been more conservative unless I had identified the peaks on the vinyl and set the levels accordingly. (which may be what you did???) DO NOT compare levels from a vinyl disc with anything coming from a modern CD, even the same performance of the same music. Note that the dynamics on modern CDs are MUCH more processed than vinyl discs ever were. |
#20
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need help to set up a M-Audio Audiophile 2496
In article ,
Richard Crowley wrote: DO NOT compare levels from a vinyl disc with anything coming from a modern CD, even the same performance of the same music. Note that the dynamics on modern CDs are MUCH more processed than vinyl discs ever were. Often true, and it's a dreadful shame. The CD medium has quite a bit more dynamic range capability than vinyl... and most CDs are mastered so much level compression that most of the dynamic range is squandered. -- Dave Platt AE6EO Friends of Jade Warrior home page: http://www.radagast.org/jade-warrior I do _not_ wish to receive unsolicited commercial email, and I will boycott any company which has the gall to send me such ads! |
#21
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need help to set up a M-Audio Audiophile 2496
Le Thu, 16 Jul 2009 16:36:28 +0100, Peter Larsen a écrit*:
Bernard wrote: I am even more sorry for these links , especially for the typing mistake. Here are the valid links below : http://www.teaser.fr/~bdebreil/musiq...ozart_ampl.wav http://www.teaser.fr/~bdebreil/musiq...art_phones.wav Thank you, it appears to me that you do not have a problem, but if you get the forte at offset 5 seconds at -6 or even -10 dB via tape out, This is what I got at first,but the file I sent has been amplified after recording, using the audacity tool for that purpose, so it was closer to 0 dB. then that is how you should record a record. As I have said previously you need to allow for making a clean recording of the loudest scratch or click because fixing it requires it to be properly recorded. You will also very soon start wanting to fix the channel imbalance, there is no such thing as a "no processing" transfer .... What I do with vinyl is to equalize the actual cartridge + riaa combination to linear based on playback of a bruel and kjaer test-record I got at some equipment demonstration way long time ago and then I do a manual "fix single click" of the large clicks and nothing else. I have decided on this strategy after detecting just how many clicks automated click fixing routines will find and fix in pristine recordings causing them to sound plain boring. As in the old days: if clicks and "needle noise" is a problem in playback, then you should fix the playback system. Note: you should NOT try to compensate for the treble roll-off innermost on the record, leave it be. With some records it seems to me that the "innermost bands" are cut at lower level so as to make room for more music, that I compensate some of of the time, but not always fully. You need to respect the artistic choices, the Droit Morale. I always digitize to 16 bit 96 kHz sample rate and downsample, So far, I have digitized at 32 bit float and 44.1 kHz, which were default values on my software (audacity), (other possible values are 24 bits and 16 bits). Then, at 'export', I do a downgrading to 16 bits so as to comply with CD format. The sample I sent was not downgraded, so it was 32 bit float and 44.1 kHz. Do you think that I would be better digitizing to 16 bits at start, so as to avoid a resampling process ? I thought that digitizing at 32 bits would allow a better sampling afterward, the job being carried with a greater number of data... also, the help file of audacity says : "it is recommended that you use 32-bit float unless you have a slow computer or are running out of disk space". In any case, my audio card is 24 bit. As for the sampling rate, this was not until a few minutes ago that I tried changing the default value (44.1 kHz) to 96 kHz, which is the maximum value allowed with my audio card. I felt that I also had to change the master clock using the audio card monitor, and set it from 44100 int to 96000 int. The audacity help file says that the sampling rate can be set to anything, and that tracks are automatically resampled to the project sample rate. my preferred audio software has very good downsampling. How do I know if 'audacity' does a good downsampling job ? If I uploaded a 96000 Hz file, and a downsampled version of it at 44100 Hz, would you have a mean to say if it went through a good or average or poor downsampling job ? In 'audacity', you have a choice of two processes for downsampling : Real-time sample rate converter : set by default to 'fast interpolation' High-quality sample rate converter : set by default to 'high quality interpolation'. Besides this, it has dithering facilities, which, on my audacity, are set by default to : Real-time dithering is set to 'none'. Other options are 'triangle', 'rectangular' and 'wave' High quality dithering is set to 'Triangle'. Other options are 'rectangular', 'wave' and 'none' It would have merit to digitize to cd-format with the target being CD's, but generally no grammophone playback is acceptable without fixing frequency response errors and gross channel imbalance, my "normalize" function "allows you to correct for DC offset (a vertical displacement of the track) and/or to amplify such that...". Is this what you call 'fixing the channel imbalance' ? I usually normalize to -2.5 dB after fixing large clicks manually with "fix single click functionality". Software that doesn't offer that functionality is toy software, but - that said - it does happen that I use some kind of automated repair, generally I prefer the clean sound of doing less. Clicks don't bother me that much either. I don't know whether audacity has a 'fix single click' fonction. I think so, even though no function bears that name. In order to manually remove a click that I can see on the graph, I operate the 'zoom in' tool till I get, say, 0.05 sec per inch on the screen. Then I set 'snap off', which de-activate the 'snap on' mode which constrain the selection to the nearest interval, by default the second. In Snap off mode, I can select very narrow ranges such as one thousandth of a second or so, and cut that much, no more. Besides, there is another facility that I have not tested yet. It is called 'find zero crossings': 'modify the selection slightly so that both left and right edge of the selection appear on a positive slope zero crossing. This make it easier to cut and paste audio without resulting in a large audible clicking sound'. Thanks again for your very relevant help which I appreciate much Kind regards Bernard Debreil |
#22
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need help to set up a M-Audio Audiophile 2496
Le Thu, 16 Jul 2009 09:58:50 -0700, Richard Crowley a écrit*:
"Bernard" wrote ... Here are the new links below : http://www/teaser.fr/~bdebreil/musiq...ozart_ampl.wav http://www/teaser.fr/~bdebreil/musiq...art_phones.wav I am not sure what these files are an example of, but they both appear to be completely normal to me. There is some very slight clipping on some peaks around 3.4 seconds into the "phones" file. But if these are the levels you are getting for the original recording, But no... the original levels were much lower (about -8 dB at the peaks). That file that I sent as example of has been processed through the 'audacity' amplifier (+7 dB if I recall well), this is why I used the word 'ampl' in the filename. Do you wish to see the original unprocessed file ? http://www/teaser.fr/~bdebreil/musiq...art_noampl.wav As for the other file (test_Mozart_phones.wav), the audio card was fed from what came out of the "phones" (headphones out), therefore it went through the amplifier... but this is supposedly not the right way to do if you expect a clean record |
#23
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need help to set up a M-Audio Audiophile 2496
Bernard wrote:
But no... the original levels were much lower (about -8 dB at the peaks). That file that I sent as example of has been processed through the 'audacity' amplifier (+7 dB if I recall well), this is why I used the word 'ampl' in the filename. Do you wish to see the original unprocessed file ? http://www/teaser.fr/~bdebreil/musiq...art_noampl.wav That looks exactly right. I don't see anything wrong with that recording. I sometimes visually scan a disc to see where the highest levels are likely recorded, and then I set the recording level given that likely peak. That sometimes lets me get a big higher level recording without fear of clipping. But your example seems perfectly acceptable. I don't know what you are complaining about? Again, DO NOT compare the levels you get from a vinyl disc from anything recorded on a modern CD. |
#24
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need help to set up a M-Audio Audiophile 2496
Bernard wrote:
But no... the original levels were much lower (about -8 dB at the peaks). That file that I sent as example of has been processed through the 'audacity' amplifier (+7 dB if I recall well), this is why I used the word 'ampl' in the filename. Do you wish to see the original unprocessed file ? http://www/teaser.fr/~bdebreil/musiq...art_noampl.wav An excellent example of the preferred record level for set and forget transfer of grammophone records. It is in my opinion malpractice to adjust record level on a pr. record basis, there is so much that needs fixing even on a high quality record that it is better to have a standard start level for processing that allows you ample headroom. As for the other file (test_Mozart_phones.wav), the audio card was fed from what came out of the "phones" (headphones out), therefore it went through the amplifier... but this is supposedly not the right way to do if you expect a clean record I assume the example is from tape out, and it is recorded perfectly for processing, and there is headroom enough to allow a clean recording of the loudest possible transient. Again, you DO need to record the clicks and scratches cleanly, otherwise they can get more difficult to fix. Some people go to agonizing lenghts to get perfection, in my opinion that is misunderstood, vinyl never was silent anyway and I much prefer good violins and small record scratches to plastic violins and fake digital silence. Kind regards Peter Larsen |
#25
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need help to set up a M-Audio Audiophile 2496
Bernard wrote:
Le Tue, 14 Jul 2009 07:07:00 +0100, Peter Larsen a ?crit?: the first one on the list, and it just happens that the guy purchased that card for the same purpose as mine, that is, digitilizing vinyl records. He also found input levels too low, and digital 'burps' in his audio samples (I don't know what that is, and I doubt if I have these). He also says that that card did a poor sampling job, compared to his former audio card. All other comments are positive, but no one else seems to have digitilized vinyls with that card... I have. You need a preamp with adjustable gain inline, if you want to manually raise input level. If you record from a D/A source, like a CDP or DVDP, you may actually find you have to use a preamp to LOWER the input gain, to avoid clipping. Digital 'burps' only occurred for me when there was an IRC conflict with my PC hardware. In my case it turned out to be a conflict with my aftermarket wireless network card, which I now disable whenever I need to record with the 2496. Other causes of digital pops and clicks are noted in the Troubleshooting faq of the user;s manual. Inadequate DMA buffer size is one. The claim that the sampling (I assume he meant A/D conversion) is inadequate is nonsense. The M-audio 2496 is an excellent card when it and your system are configured correctly, as Rightmark testing can demonstrate. http://ixbtlabs.com/articles/maudioaudiophile/ It is not the most user-friendly card by any means, though. You do need to read the manual. Such as this: Variable Signal Levels. Here you can choose two volume levels of the output signal: +2 dBV (Consumer) and -4 dBV (-10 dBV). The level of the input signal is +2 dBV and can't be changed. It is obvious that for recording from the linear-in it's necessary to use audio equipment with a standard linear-out or a recording level controller of the external device (for example, faders of the audio mixer). H/W 1/2 In in the Monitor Mixer means that we can monitor a signal and adjust volume of the playback level, but it doesn't affect a recording level. -- -S We have it in our power to begin the world over again - Thomas Paine |
#26
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was : need help to set up a M-Audio Audiophile 2496
My older posts dealt with too low levels for inputs in audacity... I now
got a better RIAA preamp with higher output, and all seemed OK... except that I now have a vinyl record that gives too high output, and I have problem recording/digitizing it. I can't monitor the analog signal input using Audacity. The input comes from a Cambridge Audio 640 P RIAA preamp, which is fed from a vinyl turntable Thorens TD 90. The preamp does not have an output setting knob. In audacity, there is a cursor that allows to decrease the input... but it does not really work like I would expect, since the decrease shows only on the spectrogram level, not on the meter toolbar, which keeps showing same high ratings with some over zero, even when output setting is set to 0 and the spectrogram is flat. Am I right to think that, even if the recording level is lowered, the input is still distorted because of too high levels ? How am I to lower the volume of the signal fed into Audacity ? I have achieved nothing trying to play with the sound card control (envy24control under Linux), and I don't know really how to use it. Alsamixer has done no better, and, most times what you lower in alsamixer also gets lowered in envy24control. Here are the settings that I have in envy24control : Monitor Inputs panel : H/W In 1 Left = 58 Right = Mute H/W In 2 Left = Mute Right = 58 SPDIF In L Mute SPDIF In R Mute The H/W In 1 and 2 boxes show green bars that go up and down while the music plays, sometimes in the red. Moving the cursors up and down to change the figures does not change anything in the level of green bars in each box, but it shows changes in the left panel called "digital mixer". The lower the figure is for each channel, the higher the same channel goes in the "digital mixer" box. At first I had thought that I should set each channel to a figure that would allow the green bar in the digital mixer to go as high as possible without entering the red zone... but I soon figured out that this had no effect on the input level in audacity nor in the output of my earphones. I can mute both H/W In 1 and H/W In 2 monitors, and the recording input in Audacity is still at maximum, as well as output in earphones. Monitor PCMs panel : PCM Out 1 is "2" on Left channel, Mute on right channel, PCM Out 2 is Mute on Left channel, 2 on right PCM Out 3 to 8 are Mute Patchbay Router panel : H/W Out 1 (L) : the check is on H/W In 1 H/W Out 2 (R) : the check is on H/W In 2 SPDIF Out L : the check is on H/W In 1 SPDIF Out R : the check is on H/W In 2 Hardware settings panel : Master Clock : Int 96000 Rate State : Locked and Reset both checked Volume Change : Rate 170 Analog Volume panel : DAC 0 = 87 DAC 1 = 90 ADC 0 = 96 ADC 1 = 96 Thanks in advance for helping me with envy24control settings. |
#27
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need help to set up a M-Audio Audiophile 2496
Bernard wrote:
My older posts dealt with too low levels for inputs in audacity... Again: your levels then were perfect. I now got a better RIAA preamp with higher output, and all seemed OK... except that I now have a vinyl record that gives too high output, and I have problem recording/digitizing it. You then have to reduce the signal level before the signal enters the soundcard. Otherwise the sound card will clip. A simple 10 dB attenuator will do, say 10 kOhm in series and 5 kOhm to ground in a small plastic box. I can't monitor the analog signal input using Audacity. The input comes from a Cambridge Audio 640 P RIAA preamp, which is fed from a vinyl turntable Thorens TD 90. The preamp does not have an output setting knob. With grammophone cartridges having their output signals within a 10 dB window it seems like folly to me not to have an output level adjustment option, a -5 and -10 dB switch would have been wise design from the manufacturer ... Thanks in advance for helping me with envy24control settings. Forget everything about adjusting the record level after digitizing, it needs to be adjusted before enters the sound card or the sound card will clip. Welcome to the simplicity of line level contraptions. Your problem is partly that the AP2496 is adapted to home audio line levels and partly that Cambrigde appear to have put extra gain in that RIAA to match the +8 dB (ref. 0.775 volt) output from a CD-player. If you need to vent your frustration, then write to Cambridge Audio and ask them how come they forgot a -5 and -10 dB option on the output of the RIAA box. It is well warranted to have that since cartridges for 47 kOhm loading have their out spread within approximately 12 dB. Kind regards Peter Larsen |
#28
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need help to set up a M-Audio Audiophile 2496
On 7/22/2009 12:13 PM Peter Larsen spake thus:
Bernard wrote: I now got a better RIAA preamp with higher output, and all seemed OK... except that I now have a vinyl record that gives too high output, and I have problem recording/digitizing it. You then have to reduce the signal level before the signal enters the soundcard. Otherwise the sound card will clip. A simple 10 dB attenuator will do, say 10 kOhm in series and 5 kOhm to ground in a small plastic box. I'd say better yet would be an adjustable attenuator, which is just a potentiometer across the output of the preamp with the wiper connected to the input of the sound card. This is what I use between my sound card and my amplifier (Technics) which produces a slightly too-high signal. I use a 50K pot. Mine fits in a small metal box (like an Altoids tin). -- Found--the gene that causes belief in genetic determinism |
#29
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need help to set up a M-Audio Audiophile 2496
David Nebenzahl wrote:
On 7/22/2009 12:13 PM Peter Larsen spake thus: Bernard wrote: I now got a better RIAA preamp with higher output, and all seemed OK... except that I now have a vinyl record that gives too high output, and I have problem recording/digitizing it. You then have to reduce the signal level before the signal enters the soundcard. Otherwise the sound card will clip. A simple 10 dB attenuator will do, say 10 kOhm in series and 5 kOhm to ground in a small plastic box. I'd say better yet would be an adjustable attenuator, which is just a potentiometer across the output of the preamp with the wiper connected to the input of the sound card. This is what I use between my sound card and my amplifier (Technics) which produces a slightly too-high signal. I use a 50K pot. There are many ways to do things, but you really should not go higher than 10 kOhm ... otherwise the cable capacity between box and sound card begins to matter too much. The disadvantages of an adjustable record level in this context are - with my work process - a lot larger than the advantages. Kind regards Peter Larsen |
#30
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need help to set up a M-Audio Audiophile 2496
"Peter Larsen" wrote in message
k David Nebenzahl wrote: On 7/22/2009 12:13 PM Peter Larsen spake thus: Bernard wrote: I now got a better RIAA preamp with higher output, and all seemed OK... except that I now have a vinyl record that gives too high output, and I have problem recording/digitizing it. You then have to reduce the signal level before the signal enters the soundcard. Otherwise the sound card will clip. A simple 10 dB attenuator will do, say 10 kOhm in series and 5 kOhm to ground in a small plastic box. I'd say better yet would be an adjustable attenuator, which is just a potentiometer across the output of the preamp with the wiper connected to the input of the sound card. This is what I use between my sound card and my amplifier (Technics) which produces a slightly too-high signal. I use a 50K pot. There are many ways to do things, but you really should not go higher than 10 kOhm ... otherwise the cable capacity between box and sound card begins to matter too much. The disadvantages of an adjustable record level in this context are - with my work process - a lot larger than the advantages. Yes indeed. My standard line-level attenuation pot is a 5K 2 watt RN50 type. |
#31
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need help to set up a M-Audio Audiophile 2496
Peter Larsen wrote:
There are many ways to do things, but you really should not go higher than 10 kOhm ... otherwise the cable capacity between box and sound card begins to matter too much. Not to quibble, but "capacitance" ;-) geoff |
#32
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need help to set up a M-Audio Audiophile 2496
geoff wrote:
Peter Larsen wrote: There are many ways to do things, but you really should not go higher than 10 kOhm ... otherwise the cable capacity between box and sound card begins to matter too much. Not to quibble, but "capacitance" Thanks, will try to remember. ;-) geoff Kind regards Peter Larsen |
#33
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need help to set up a M-Audio Audiophile 2496
Peter Larsen wrote:
geoff wrote: Peter Larsen wrote: There are many ways to do things, but you really should not go higher than 10 kOhm ... otherwise the cable capacity between box and sound card begins to matter too much. Not to quibble, but "capacitance" Thanks, will try to remember. Tha8t's fine - even some cable manufacturers don't seem to get the subtle difference, and print it on cable jackets ! ;-) geoff |
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