[muckshot]

I was hoping someone might be able to give me some advice. I am using an M-Audio 2496 and I was trying to do some inbound analog recording, but have been running into glitches; small repetitions (sound like high-speed skips) of varying length are introduced into the recording. They are the kind of artifact that might be introduced if the computer was working hard at some other task (like playing an intensive game or video encoding etc) while simultaneously making a recording. Even after turning off or disabling almost everything running, I am still gettng the same glitchy recordings...

I don't think there have been any new drivers released for the 2496 in ages, but I could be wrong. Could it be a problem with the way Sound Forge is configured? Some kind of buffer issue or sound card misconfiguration? I am running a record player into an NAD phono amp and then directly into the jacks at the back of the M-Audio card, using Sound Forge 8.0 under Windows 2000 to record. Any ideas? Thanks

 

[jiiteepee]

Quote:

Originally Posted by muckshot I was hoping someone might be able to give me some advice. I am using an M-Audio 2496 and I was trying to do some inbound analog recording, but have been running into glitches; small repetitions (sound like high-speed skips) of varying length are introduced into the recording. They are the kind of artifact that might be introduced if the computer was working hard at some other task (like playing an intensive game or video encoding etc) while simultaneously making a recording. Even after turning off or disabling almost everything running, I am still gettng the same glitchy recordings... I don't think there have been any new drivers released for the 2496 in ages, but I could be wrong. Could it be a problem with the way Sound Forge is configured? Some kind of buffer issue or sound card misconfiguration? I am running a record player into an NAD phono amp and then directly into the jacks at the back of the M-Audio card, using Sound Forge 8.0 under Windows 2000 to record. Any ideas? Thanks

Use ASIO or WDM/KS device drivers, and when/if you're not multitracking w/ monitoring, then just set the I/O latency to 20 ms or more.

jiitee

 

[muckshot]

Thanks for the reply! Ok, you'll have to bear with me cause I'm sort of dumb

In Sound Forge, I've selected 'Analog In Delta-AP' as my default audio recording device. Where do I set the latency? When I bring up the Delta mixer, I can set the 'DMA buffer size' in terms of samples-- right now it's set to 'Latency 512 samples', the range is from 64 to 2048.

Also, in the Delta Mixer's options, I have an a box checked that is listed as 'Disable audio app use of Monitor Mixer and Patchbay/Router' - should I leave this checked?

 

[muckshot]

btw, I use the Delta Mixer to attenuate the sound level while recording to keep it out of the red, so in the Delta Mixer's 'Patchbay / Router' menu, I have the 'Monitor Mixer' selected under the 'H/W out 1/2' section. Not sure if that's at all relevant...

 

[jiiteepee]

Quote:

Originally Posted by muckshot Thanks for the reply! Ok, you'll have to bear with me cause I'm sort of dumb In Sound Forge, I've selected 'Analog In Delta-AP' as my default audio recording device. Where do I set the latency? When I bring up the Delta mixer, I can set the 'DMA buffer size' in terms of samples-- right now it's set to 'Latency 512 samples', the range is from 64 to 2048. Also, in the Delta Mixer's options, I have an a box checked that is listed as 'Disable audio app use of Monitor Mixer and Patchbay/Router' - should I leave this checked?

From SF Manual:

[size=xx-small] Quote:

Item - Description --------------------------------------------------------------------------------------------------------------------------- Audio device type - Choose a driver type from the drop-down list. - Microsoft Sound Mapper - The default setting. Allows the Sound Mapper to choose appropriate playback and recording devices. - Windows Classic Wave Driver - Allows you to choose a specific audio device using a classic Wave driver. - ASIO - Allows you to choose a specific audio device using a low-latency ASIO driver. --------------------------------------------------------------------------------------------------------------------------- Default playback device - When Windows Classic Wave Driver or ASIO is selected in the Audio device type drop-down list, you can choose the audio device that you want to use for playing back sound data. - Playback Buffering (seconds) - Specifies the total amount of buffering that is used during playback. - The larger the number, the more buffering is performed during playback. This value must be as low as possible without gapping. To set it, start at .25 and play back a typical song. Move some of the track faders. If the playback gaps, try increasing this slider in small increments until the gapping goes away. As you increase this slider, the RAM meter at the bottom of the ACID window will indicate more RAM usage. You need to strike a balance between RAM usage and playback buffering. - If you simply cannot get playback to be free of gapping, you need to either decrease the number of tracks you are trying to play simultaneously, install more RAM in your computer so you can increase buffering, buy a faster access hard drive, or minimize the number of audio plug-ins you are trying to use simultaneously. --------------------------------------------------------------------------------------------------------------------------- Default recording device - When Windows Classic Wave Driver or ASIO is selected in the Audio device type drop-down list, you can choose the device that you want to use for recording sound data. This device will be used by default in the Device drop-down list in the Record dialog. - Selecting a device such as the Wave Mapper or Microsoft Sound Mapper allows Windows to select an appropriate device to use for the current sound data. - Record buffering - Specifies the total amount of buffering that is used during recording. - If you use your computer for other tasks while recording, increasing this setting can reduce the likelihood of those tasks interrupting recording. --------------------------------------------------------------------------------------------------------------------------- Advanced - Click this button to open the Advanced Audio Configuration dialog. --------------------------------------------------------------------------------------------------------------------------- Default All - Click to restore the Audio tab to the default settings.

[/size]

As it says in manual too ... (underlined/bolded text above).

Samples set to 512 --> means latency in ms w/
44.1 kHz = ~11.6 ms
48 kHz = ~10.7 ms
96 kHz = ~5.3 ms

So you can set the buffer in samples to MAX.

If you need to set SR/bit-depth on M-Audio software too, just check everything equals w/ settings made on SF.

BTW: With which resolution are you recording? I did record using 24-bit/ 96 kHz (converted to WMA 24/96) and the results are excellent.

jiitee

 

[muckshot]

I actually had my inbound recording set to the Mixer/Monitor rather that the straight 'Audio In' as I remarked earlier. I stupidly never thought to look in the SF manual, and the 2496's manual was less than exhastive and clear AFAIK, but I'm still pretty green here so maybe it's just me...

Ok, I will play around and see if I can't get things worked out, I basically only used my 2496 for MIDI and didn't have any problems so I never needed to learn much about the card or it's settings.

Just to be clear, I've set my 2496 to '2048 samples' under the DMA buffer configuration menu, which popped up to tell me my latency would be 21 milliseconds for 96kHz / 46 milliseconds for 44kHz. This ought to be OK?

To be honest, I've never recorded anything beyond the standard 16bit/44kHz setting, although I've just now configured it to 24/96 (as per the card's name) and will report back with my impressions. Thanks again!

 

[darky_mtp]

Quote:

Originally Posted by muckshot I don't think there have been any new drivers released for the 2496 in ages, but I could be wrong.

You are wrong.

 

[muckshot]

Quote:

Originally Posted by darky_mtp You are wrong.

ha! Thanks for the link, but I'm still running 2000 and the last update for my platform is from early 2005, still newer than the one I've got installed though

 

[Steve999]

I used to use a super-lean freeware called CD Wave Editor to get analog sources digitized on my stone-age computer. CD Wave Editor is also a marvelously easy wave track splitter after a little practice.

Nowadays I have been using the LP and Tape Assistant that comes with Easy Media Creator 8. The rest of the utility package I don't have much use for, but the LP and Tape assistant, with really top-notch track splitting and tagging features and MP3 encoder, is pretty nice and very feature-rich.

But anyway, I'm just a novice, but I'd suggesting trying CD Wave Editor for starters to keep things as lean on the software side as possible. I've had a lot of success with this.

Sound-Forge uses a LOT of resources, does it not? And my experience has been it's much much slower than CD Wave Editor and LP and Tape Assistant. You can import things into Sound Forge later if you are using its best-of-class hiss, pop and click removal features, etc.

Also, I'd keep things at 44.1 khz sampling rate if possible. I think it would tax your resources less than 24/96 and it's more than capable for a transparent upload of LP or tape analog sources, IMHO. If you were doing studio work with lots of panning, overdubbing, editing, EQ, etc., 24/96 might give you some margin for error, but otherwise I think 44.1 khz sampling rate is more than needed. I use a cheap external USB sound card. The A/B comparisons I've tried after a 44.1 khz sampling rate import have been great.

Hope this helps.

Quote:

Originally Posted by muckshot I was hoping someone might be able to give me some advice. I am using an M-Audio 2496 and I was trying to do some inbound analog recording, but have been running into glitches; small repetitions (sound like high-speed skips) of varying length are introduced into the recording. They are the kind of artifact that might be introduced if the computer was working hard at some other task (like playing an intensive game or video encoding etc) while simultaneously making a recording. Even after turning off or disabling almost everything running, I am still gettng the same glitchy recordings... I don't think there have been any new drivers released for the 2496 in ages, but I could be wrong. Could it be a problem with the way Sound Forge is configured? Some kind of buffer issue or sound card misconfiguration? I am running a record player into an NAD phono amp and then directly into the jacks at the back of the M-Audio card, using Sound Forge 8.0 under Windows 2000 to record. Any ideas? Thanks

 

[muckshot]

Quote:

Originally Posted by Steve999 I used to use a super-lean freeware called CD Wave Editor to get analog sources digitized on my stone-age computer. CD Wave Editor is also a marvelously easy wave track splitter after a little practice. Nowadays I have been using the LP and Tape Assistant that comes with Easy Media Creator 8. The rest of the utility package I don't have much use for, but the LP and Tape assistant, with really top-notch track splitting and tagging features and MP3 encoder, is pretty nice and very feature-rich. But anyway, I'm just a novice, but I'd suggesting trying CD Wave Editor for starters to keep things as lean on the software side as possible. I've had a lot of success with this. Sound-Forge uses a LOT of resources, does it not? And my experience has been it's much much slower than CD Wave Editor and LP and Tape Assistant. You can import things into Sound Forge later if you are using its best-of-class hiss, pop and click removal features, etc. Also, I'd keep things at 44.1 khz sampling rate if possible. I think it would tax your resources less than 24/96 and it's more than capable for a transparent upload of LP or tape analog sources, IMHO. If you were doing studio work with lots of panning, overdubbing, editing, EQ, etc., 24/96 might give you some margin for error, but otherwise I think 44.1 khz sampling rate is more than needed. I use a cheap external USB sound card. The A/B comparisons I've tried after a 44.1 khz sampling rate import have been great. Hope this helps.

Hey man, thanks for sharing your thoughts/setup. Regarding software, I have been using Sound Forge for years and strangely never had problems before this recent glitch business, I've done plenty of analog transfers in the last few years, but I think with the config changes and a driver update, things have worked themselves out

I will try to find a copy of the prog you're using and see how I find it, but I'm a creature of habit and know SF pretty well by now. I actually never use any of the vinyl restoration plug-ins, I don't mind it sounding like vinyl and most of the LPs I have are in decent condition. I've messed around with a bunch of the noise reduction & click/pop removers, but I find it softens the overall presentation and personally I prefer a bit of noise to a damp sounding recording.

Did a quick recording session with 24/96 and a standard 16/44.1 and did a quick comparo. I personally thought the 24/96 was much more dynamic sounding, I was impressed. Not sure if I'm gonna use it as a standard as my turntable/needle could be much better and I'm not sure it's totally worth it, but disk space is really not an issue so I'm debating it.

Anyhow, thanks again for the help from all those who contributed!

 

[Steve999]

You're welcome. It's fun to see how other people are doing this.

I never use the noise reduction or click/pop removers either. I prefer the original LP sound warts and all. I've heard that Sound Forge is supposed to be the very best for noise reduction and pop and click removal though.

Glad to hear everything has worked itself out for you. This stuff can be a little tricky. Like you, I tend to stay with what works for me once I find it. The biggest trip is hearing my old LPs, divided into tracks and titled, with album art, on my computer and on my ipod. I love it.

Quote:

Originally Posted by muckshot Hey man, thanks for sharing your thoughts/setup. Regarding software, I have been using Sound Forge for years and strangely never had problems before this recent glitch business, I've done plenty of analog transfers in the last few years, but I think with the config changes and a driver update, things have worked themselves out

 

[muckshot]

Quote:

Originally Posted by Steve999 The biggest trip is hearing my old LPs, divided into tracks and titled, with album art, on my computer and on my ipod. I love it.

No doubt, reminds me of the feeling I got after I burned my first audio CDR mix, back when nobody I knew had one yet

 

[jiiteepee]

Quote:

Originally Posted by muckshot ... Did a quick recording session with 24/96 and a standard 16/44.1 and did a quick comparo. I personally thought the 24/96 was much more dynamic sounding, I was impressed. Not sure if I'm gonna use it as a standard as my turntable/needle could be much better and I'm not sure it's totally worth it, but disk space is really not an issue so I'm debating it. Anyhow, thanks again for the help from all those who contributed!

So, you got the same result I did (even I'm near '50). Though, this difference can be clearly seen on FFT Spectrum Analyzer too (I've checked this).

What becomes to disk space, there are good compression formats for 24-bit/96 kHz. I did use WMA's :
- (Pro) 24-bit/96kHz/256-440kbps (CBR) [size 171 MB wav --> < 10MB - < 17MB]
- (Pro) Quality 98 24-bit/96kHz (VBR) [size 171 MB wav --> < 11MB]

and to my ears there are not much difference from original 24/96 wav.

WMA lossless Quality 100 96 kHz Stereo gave 'bout 15% smaller file size compared to 16-bit/44.1 kHz (WAV --> Q100/96 - 54MB vs std WAV - 65MB).

[size=xx-small]If you're too lazy to type, the file naming 'process' is easiest to do using simple copy/paste operations. There are sites like Artist Direct you'll find track indexes (and much more data) for most of your albums (quite accurate information too) --> just search the artist --> select the album (best done in listen/watc -tab -->view all songs) --> select the album you're after --> copy all necessary data and paste to notepad --> modify the track naming --> ...[/size]


jiitee