Sample rate is resetted by recording program

E-MU locked.jpg
Hi,
I try to record some of my old DAT (digital audio tape) recordings to PC (Vista) using the E-MU 0404. Sample rate 48 kHZ, sync source external, locked; I can hear the sound coming from earphones connected tot the E-MU device.
Then when I try to record, the sample rate is reset automatically to 44.1 kHz. No sound is coming from the earphones anymore; the blue 'locked' signal is out and there is no signal shown in the meters of the recording program. This happens with Ableton Live (Free download) as well as Adobe Soundbooth CS4 and Audacity 2.0. What can I do to prevent the E-MU device being reset?
greetings from The Hague, Holland

Resolved.
The trick is in the Windows Sound Settings, Digital Input:
WindowsDigitalInput.jpg
Signal at the microphone input of the E-MU can be recorded.

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dgoh88 wrote:
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xfggfx wrote:
I am exporting from Logic at48khz and wondered what is the highest quality I can export at...what is the sample rate at which Logic records?
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http://www.motu.com/techsupport/manuals

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Conversion de LP -Archivos Digitales
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thanks!

rcook349 wrote:
44.1 Pros
- Friends across the country can use GarageBand (44.1kHz is max) to lay down a single track and send to me to import into Logic Pro and mix (and will match up)
- GarageBand is free, Logic Pro is $199; Apogee JAM is $99, Jam96kHz is $129
- GarageBand requires fewer resources (1GB RAM vs 4GB for Logic Pro) so they don’t have to have beefy Macs
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- Picture mixing a bunch of 44.1 tracks vs a bunch of 96 tracks even if on my beefy Mac running Logic Pro; fan running, gasping for air etc?
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- 96 sounds noticeably better than 44.1?
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- Maybe choose this so friends using GarageBand can record at 44.1 so upscales more cleanly? But then bouncing down to 48kHz not as clean? (I don’t recall Logic Pro allowing me to choose 44.1 for bounce rate.. always says 48 for MP3/M4A)
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44.1 kHz still is pretty much standard for MP3's.
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Higher sampling rates will not "future proof" anything. In fact, that whole concept is flawed. Your best bet for now is simply 44.1 kHz 24 bit uncompressed PCM files in their most widely used form: AIFF or WAV.
96 does not noticeably sound better than 44.1, unless you have a top end interface and a very delicate and very complicated mix, and admirably acute hearing. In some interfaces 96 or 88.2 have been found to sound worse than 44.1, because of clocking inaccuracies getting progressively worse at higher sampling frequencies. I would stick to 44.1, it has lots of practical advantages (as you pointed out), and the sonic difference with 96 kHz is marginal at best, and certainly not worth the price: "double" rates need double the CPU power for any plugin processing. That's the biggest loss. Half a Mac.
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O, also just spotted your remark about Logic not "letting you" bounce MP3/M4a to 44.1 kHz. You must remember incorrectly, because I never bounce MP3 or AAC to any other frequency than 44.1 kHz. However, it may be that this rate is tied to the projects' sampling frequency as set in the project settings, and the last time I used 48 kHz was in LP 8. I'll check that now.

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Thanks for your response.
I will use Apple MIDI Setup, as you suggest.
I want to change the sample rate to 48 KHz, as the final destination is DVD and I don't want to convert from 44.1 to 48 and thus suffer a possible loss in quality.
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Message Edited by whatcreative on 02-8-2009 08: PMMessage Edited by whatcreative on 02-8-2009 08: PM

thanks for your replies.
katman,
I checked that thread. Although I wonder if the same thing applies to recording at rates higher than 48khz as well ?as I am more interested in finding out if when recording 96khz does the signal really get sampled at true 96khz or something else ? But, Upsampling a recording (not playback)?to a higher frequency doesn't make sense?because you can't create what you lost by recording at a?lower sampling rate. A test for checking this would be to try and record a signal that has frequency content well above 24khz but less than 48khz. If that content gets captured that means it is indeed recording at 96khz, if it gets lost then its recording at 48khz.
flipflop,
I have opened up the soundcard and messed around with a lot. Here's a list of the chips alongwith their descriptions based on whatever I could find?about them -
Sigmatel STAC9460S - Codec
JRC 4556A - Headphone output opamp
ST MC33078 - Line level output/input dual opamp. (Total 4).
Philips UDA334 - Low Power DAC
Philips ISP58BD - USB device controller
Creative CA086 - DSP
Cirrus Logic EP7309 - High-Performance, Low-Power System-on-Chip Enhanced Digital Audio Interface
ST M29W400BB - 4Mbit Flash Memory
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ST LD8 - .8v regulator (?)
Although the datasheet of the codec suggests that it supports all sampling rates, I'd have to dig deeper to really find out if it is handicapped to only one standard rate for recording - http://www.alsa-project.org/~james/d...tac9460-ds.pdf
Message Edited by whatcreative on 02-2-2009 :35 AM

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Sample rate is resetted by recording program

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