Transferring music recorded at 32kHz sampling rate

In my project transferring music from DAT tapes to HD, I've run into a problem. A few tapes were recorded at 32kHz sampling rate for longer recording time.
STP can record at 32kHz so in theory I can transfer music digitally. But there is no sound when I monitor the tape and the Audio MIDI setup doesn't have a 32kHz option to choose for recording.
Can someone steer me in the right direction to get this music off the tapes? Is recording to the line analogue input the only way?

The problem is not with STP, the problem is with the hardware. if you want to record at 32Khz, you'll need an audio interface that is 32Khz compatible and I don't think the Mac's built-in audio is. Otherwise, we could select that in the AudioMidi Setup.

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Changing audio recording depth and sample rate in QuickTime Pro?

Last week, I set up QuickTime Pro 7.4.5 to record audio in 24-bit, 48 KHz format -- only now, I can't remember how I did that. I think that I may have used some other software to change the settings of my Mac Pro's built-in sound card. Does anyone know how to do this?
I made several such recordings, just last week, as a test, simply using a microphone plugged directly into my Mac Pro, and using QuickTime Pro as the recording application. During playback of such QuickTime audio files, I can hit <command-i> to see that the recordings are indeed 24-bit, 48 KHz. But now, when I repeat the same process, my QuickTime audio recordings are made with 16-bit resolution, at 44.1 KHz. I remember putting the settings back to those values after my recording session of last week. But I can't remember how I adjusted those settings. Does anyone know how to do that?
Thanks.

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.
As for bit depth, I want to sample at 24-bit, since I'll be processing the sound afterwards and don't want audible rounding errors during the number "crunching". Afterwards, I'll just drop the eight least significant bits, or perhaps round them up or down (to zero or 256).

My usb turntable records at 48000hz sample rate is 44100hz?

do i haveto change sample rates to match recording device?

iTunes can play songs with 48kHz sample rate, if that is what you are asking.
However, I have run into issues with the iPod on certain files at 48kHz. In general, it is more trouble-free to have everything in your library at 44.1kHz if possible.

Recording LP records as source material- Sample Rate

Using recorded tracks from LP records to make DVDs, Blu-Ray DVDs or simple CD's. Am not sure what maximum sample rate to use. I understand the end product limits of the various digital media, but LPs are analog. Do I gain any sound quality by recording the original LP at a sample rate higher than 48000/32bit, say 96000 sample rate) and then resampling (downsizing) the audio file if the end product cannot produce the higher sample rate?

Conversion de LP -Archivos Digitales
Se recomienda Grabalos con estas velocidad de Muestreo
Blue-Ray 96000Hrz. / 32 bits
DVD & 48000 Hrz. /32 Bits
CD DE AUDIO A 44100Hrz /32 Bits ó 24Bits
Te recomendaria Cambiarte a Adobe Audition
Saludos
http://soundcloud.com/creativoxpro/restaurando-audio-de-un-vinil
Para audio
8000 muestras/s
Teléfonos, adecuado para la voz humana pero no para la reproducción musical. En la práctica permite reproducir señales con componentes de hasta 3,5 kHz.
22050 muestras/s
Radio En la práctica permite reproducir señales con componentes de hasta 10 kHz.
32000 muestras/s
Vídeo digital en formato miniDV.
44100 muestras/s
CD, En la práctica permite reproducir señales con componentes de hasta 20 kHz. También común en audio en formatos MPEG-1 (VCD,SVCD, MP3).
47250 muestras/s
Formato PCM de Nippon Columbia (Denon). En la práctica permite reproducir señales con componentes de hasta 22 kHz.
48000 muestras/s
Sonido digital utilizado en la televisión digital, DVD, formato de películas, audio profesional y sistemas DAT.
50000 muestras/s
Primeros sistemas de grabación de audio digital de finales de los 70de las empresas 3M y Soundstream.
96000 ó 192400 muestras/s
HD DVD, audio de alta definición para DVD y BD-ROM (Blu-ray Disc).
2 822 400 muestras/s
SACD, Direct Stream Digital, desarrollado por Sony y Philips.
Para vídeo
50 Hz
Vídeo PAL.
60 Hz
Vídeo NTSC.
*informacion extraida para apoyo de la pregunta en el foro // http://es.wikipedia.org/wiki/Frecuencia_de_muestreo
> http://en.wikipedia.org/wiki/Sampling_rate

E-MU 0204 for Vinyl Recording: Missing Sampling Rates Using Windows 7 Drivers?

Hi all,
Just so you know, the primary use for the E-MU 0204 is to convert the mounds of LPs I have into digital format (either 88.2 or 96 kHz sampling at 24 bit). Having said that, I'm going through a lot of growing pains with the E-MU 0204 unit I just purchased. The specs of my setup is as follows:
-Windows 7 Pro 64Bit 8GB ram
-Stanton T.92 USB (with internal preamp removed)
-Shure M97xE cartridge
-ART DJ PreII Phono Preamp.
As far as I can tell, the E-MU drivers are properly loaded onto the OS and in the Device Manager>Sound, and video game controllers, windows correctly identifies the unit as a "E-MU 0204 USB".
From the specifications of the unit here:
"Sample Rates: 44.1, 48, 88.2, 96, 176.4, 192kHz from internal crystal (no sample rate conversion)
Bit Depth: 24-bit I/O, 32-bit processing
USB Hi-Speed
- Full 24-bit resolution at all sample rates
- Stereo in, stereo out at all sample rates"
Yet, in the "Recording Devices" of my OS, I get only the following choices:
-2 channel, 16 bit, 88200 Hz (Studio Quality)
-2 channel, 16 bit, 96000 Hz (Studio Quality)
-2 channel, 16 bit, 176400 Hz (Studio Quality)
-2 channel, 16 bit, 192000 Hz (Studio Quality)
-2 channel, 24 bit, 44100 Hz (Studio Quality)
-2 channel, 24 bit, 48000 Hz (Studio Quality)
-2 channel, 24 bit, 96000 Hz (Studio Quality)
-2 channel, 24 bit, 192000 Hz (Studio Quality).
The 44.1 kHz/16 bit, 48 kHz/24 bit, 88.2 kHz/24 bit, and 176.4 kHz/24 bit modes seem to be missing.
My preferred choice for recording LPs is to use 96/24 for LP music that I know will not be burned onto CD, and 88.2/24 for LP music I know may be burned onto CD in the future at some point.
Am I missing something here? Please help!
I'm pretty new to this whole E-MU interface stuff, so I appreciate your patience in advance.
Thanks kindly,
Andrew

dgoh88 wrote:
okay just got home, turned the PC back on and now all the sounds are distorted again. Tried completely disabling wireless ports, still the same. About to throw in the towl already
There can be other high priority prcosses as well.
Keep trying with DPC Latency Tool ...
Also, you could try by
- turning off all visual effects of O/S (incl. Aero)
- turn off indexing service
- launch perfmon.msc / system monitoring tool to see which processes uses CPU/etc.
If you already uninstalled XP then why not install XP as well (dual boot system). Most audio interfaces works better on 32-bit XP compared to Vista/W7 & 64-bit systems.
jutapa

On the fence for which sample rate to record at (44.1 vs 96)

Been reading tons of posts on the sampe rate debate. My friends (across the country) and I are about to start to collaborate on the great American rock album that we didn't quite get right back in the day in college. I'll be running the show sending them scratch tracks with clicks so they can lay down individual tracks and I'll import them.
I'm torn on which sample rate(s) to use -- and want the best quality possible, of course. I've boiled it down to the following pros per sample rate. Any advice/comments much appreciated. thanks
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
- Smaller file size to post/share (Dropbox cost per/storage size)
- 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?
96 Pros
- 96 sounds noticeably better than 44.1?
- While 44.1 standard for CDs, that’s no longer how music is largely distributed
- Should always record to max resolution you can be “future proofed”?
- Current lossy formats support up to 48kHz so 96kHz good as will cut cleanly in half when bouncing
88.2 Pros
- 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)
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
- Smaller file size to post/share (Dropbox cost per/storage size)
- 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?
96 Pros
- 96 sounds noticeably better than 44.1?
- While 44.1 standard for CDs, that’s no longer how music is largely distributed
- Should always record to max resolution you can be “future proofed”?
- Current lossy formats support up to 48kHz so 96kHz good as will cut cleanly in half when bouncing
88.2 Pros
- 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)
thanks!
44.1 kHz still is pretty much standard for MP3's.
Your friends/collaborators can pretty much use any application that can record PCM (or even MP3) audio; even if they're not playing to a steady tempo, you can line everything up in Logic, with flex.
Using Garageband and one set tempo should also work. Just remember that you cannot open Logic files in Garageband, only Garageband files in Logic. The Audio Files recorded by either, can be used (imported) by either.
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.
Bitdepth on the other hand does make a significant difference. There is no reason not to record everything at 24 bits. Shorter: always record at 24 bits.
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.

Microphone record sample rate problem

Good day. Our company is developing a Flash-application for audio processing
purporses. We encountered the following problem: when we use flash.media.Microphone
object (ActionScript 3) with "rate" property set to 22kHz or 44kHz then
using a spectrum analysis tool it is clear that
actual sampling frequency of recording is 16kHz (instead of 22kHz or 44kHz) which is
too low for high quality voice recording.
Is there any way to solve this problem?
PS. We trying to compile our flash movie under version of player 10.3,11.1,11.2 in CS5 and in CS6 and we have the same result.
PPS. More tested getMicrophone and getEnhancedMicrophone. When microphone initialized with getMicrophone, quality of recorded data is same as rate property of microphone object. When initialized with getEnhancedMicrophone the sample rate of recorded data is 16kHz (view using a spectrum analysis tool. Adobe audition for example). looks like a BUG in flash player.

TheJackAttack wrote:
Some of this is a bit disinenuous. I don't know any "professional" audio engineer that is using on-chip audio. All pro engineers I know-on PC based machines-are using ASIO and professional level interfaces whether PCIe or 1394. The OS is a non starter in that regard. For all the crying about Win7, I just haven't had the issues or problems that seem to make it grossly inadequate for some of my colleagues on this forum.
Unfortunately (perhaps for them?) I know many many professional audio users who mix and match external asio, external wdm, internal on-board wdm drivers. They may not be engineers, although some will be, but they are professionals. An external audio interface running wdm drivers may well be affected by Microsoft settings.
The people I know use laptops, and include professional music writers and critics, broadcast and print journalists, composers, forensic experts and so on. They need to be able to play back occasionally over the laptop speakers and, if they have to listen or work with material from on-line sources, they will often have to use the wdm drivers with their professional audio interface.
Many of them have moved successfully from XP to Windows 7, but they do have to be aware of the possible problems. I find that it takes time to get them set up and running comfortably in Windows 7, but it can be done.
The trouble is that this software is used by so many people in so many different circumstances.

If your Lightroom 6 is crashing in the Slideshow module, check the SAMPLE RATE of your music track

After spending over 2 hours with an Adobe technician who had control over my Windows 7 machine, I've learned something that I don't think Adobe knows yet... I upgraded from LR 5.7 to LR 6 (not CC) and began having issues when I was working in the Slideshow module. When trying to add a music track, Lightroom would crash ("unexpectedly quit"). I couldn't for the life of me figure out why. I turned off the GPU since a fair amount of chatter on the forum related to that, but it didn't change anything.
So I finally waited patiently for the Adobe Chat line to finally get a tech to respond...and he was exceptionally detailed and methodical in his approach to diagnosing my issues. I could follow on my screen as he did a variety of things that indicated he understood what he was doing. Unfortunately he got to a point where he thought (as did I) that he had found the problem/solution...which he thought was related to the bit rate. He concluded that music with a Variable Bit Rate (VBR) as opposed to a Constant Bit Rate (CBR) was causing the problem. Also unfortunate was that this "discovery" was near the 2-1/2 hour mark and we did not apply the scientific method to property determine if we had narrowed down to cause/effect.
Based on my further analysis, he was close, but not right on. It isn't the BIT rate mode that is causing the crash, but rather the SAMPLE rate. Turns out that the vast majority of songs in my iTunes library have a sample rate of 44.100 kHz--and those work just fine. However, I have about 15 songs, mostly downloaded as free background music from the YouTube creation library, that have a sample rate of 48.000 kHz--every single one of those cause my LR6 to crash.
If you're having a problem with Lightroom crashing in the Slideshow module, check the SAMPLE RATE of your music track.

If you're on Windows, check out a free program called Audacity--it's fantastic. With Audacity the process of converting a 48.000 kHz to 44.100 kHz is as simply as opening, changing the sample rate, and then exporting again, as an MP3.

Sample rate and data recording rate on NI Elvis

I am currently working on a project that requires me to record my data at 1ms intervals or less. Currently the lowest timing interval I can record is at 10ms. If I change my wait timer to anything below 10 the recorded data in excel will skip time. For example instead of it starting at 1 ms and counting 2,3,4,5,6...,etc. It is skipping from 2,5,12,19,....,etc. So my question is if it is a limitation that I have reached on the NI Elvis or if it is possibly a problem with how I've created my LabVIEW code. My program from an operational stand point is working great, but it is my data recording that is causing me to not be able to move to my testing phase. Any help on this matter would be greatly appreciated.
Other information that might be relevant:
Operating System: Windows 7
Processor: Intel(R) Xeon(R) CPU E31245 @ 3.00 GHz
Memory: 12GB
DirectX Version: 11
Attachments:
Count Digital(mod12).vi ‏76 KB

Hi crashdx,
So my immediate thought on this issue is that the code inside your primary while loop might be taking too long to process to achieve such a high sample rate. Especially when making calls into external applications (such as Excel) which can take a large amount of itme.
There is a very useful debugging tool called the Performance and Memory tool. If you aren't familiar with this tool, it will allow you to see how much memory the various chunks of your code are using and, more importantly here, how much time each subVI is taking to execute. Does the code inside your while loop take longer than 1ms to run? If so, then you will definitely see unwanted logging behavior and will need to change your approach. Would it be possible to collect more than a single sample at a time and perform calculations on a large number of samples at once before writing them to Excel in bigger chunks?
I've included a link to the LabVIEW helpful detailing the Profile Performance and Memory tool.
http://zone.ni.com/reference/en-XX/help/371361H-01/lvdialog/profile/
I would first try and figure out how long it's taking your loop code to execute and go from there.
I hope this helps!
Andy C.
Applications Engineer
National Instruments

Record data at one sample rate and calculate velocity

I need to record position data at a sample rate of 10 hz but I would like to calculate velocity every half a second. What is the best way to do this so the velocity display does not jump? I want to capture quick responses but I would also like to calculate the velocity over a longer period of time like 1 minute.

I am trying to calculate vertical speed based on pressure altitude data. I am using serial communication to measure pressure at 10 Hz. I am using Labview 8.2 for all of the calculations and measurements.
I am looking for some way to average the data over some time interval that I have yet to determine. I want to be able to see any major pressure changes whithin about .5 seconds but I also need to see trends over a longer time period say 2 to 10 seconds.
I have thought about using an array and then averaging the array or implimenting a rolling digital filter.

Audigy 2 NX recording sample rate question

Hi, I have two questions about the sample rate used when recording in the Audigy 2 NX -
When recording an analog signal (Line In), is the actual?sampling rate at which the analog signal is being sampled different from the requested sample rate ? For example, when recording at XX Khz, does the actual sampling of the analog signal really?happen at XX Khz or does it get sampled at a different fixed rate(48khz?) and then gets re-sampled to XX Khz ?
When recording a digital signal (spdif in), if the recording sampling rate specified is the same as the sampling rate of the input signal then does the card record this digital input signal in a bit-perfect fashion ? For example, if the digital input signal is 6bit and 44.Khz, and the recording is also being done at 6bit and 44.Khz then does the card actually record the signal at exactly that rate without modifying the data in any way ? or does it record it at a different rate and then re-sample internally to arri've at the requested recording rate ?
Now I know this card works a little different in terms of re-sampling when the 'Digital Input only' checkbox is checked. So I am looking for the answers to these questions keeping in my mind this option as well.
Thanks!
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
ST LD33 - 3.3v regulator (?)
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

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.

I'm trying to record in Adobe Audition and it keeps saying my the sample rates for my input and output devices do not match. How do I correct this?

I'm trying to record in Audition and it keeps saying my sample rates for the input and output devices don't match. How do I fix this?

I finally have communication between the ISA One and Audition. I moved the optic cable to another SPDIF on the computer and from the SPDIF to ADAT on the ISA One and it finally agreed that the communication was at the same clock speed. The audio was not intelligible so I moved the optic cable back to the original configuration and I can record voice that is clean.

Changing sample rates for recorded audio?

I've been trying to record VO through premiere. I'm using an m-audio mobile pre which has never given me trouble in any other application. When i turn on Meter Input(s) Only from the panel menu of the track mixer i can clearly see the input move in time and responsively. When I record however the resulting clip is much shorter than the ammount of time recorded and the audio is all garbled and chopped. It clearly jumps out to me as a sample rate problem. I looked at the clips and they are 44.1 but the sequence i'm recording to is 48. I made a 44.1 sequence and recorded to that which worked fine but there must be a way to change the recording sample rate right? I can't find a solution anywhere. Anybody know?

i'm just following up here. Thanks for the replies. I understood i would lose data as one answer suggests, but i wanted to convert it so i could work in a new song environment in which all the NEW tracks were recorded at 192. So the second answer was very helpful. incidentally, with further research we are recording at 96.

Why can't I record in 96.000 khz sample rate ?

I am running Logic Pro 9.1.7 (1700.57) (32-bit). My computer is Mac OS 10.6.8. I am using a Duet Apogee
I have always recorded in 44.100 khz, but now someone has asked me to record in 96.000 khz. Everything is fine until I change the sample rate in Logic. Then I get a tinny, distorted sound, and this is for any audio I play through my computer, not just the track that's in Logoc. Is there a way out of this ?
Thanks

You own the hardware and don't know how to use it?
Didn't you install some software for the Duet? Somewhere on your system will be a software applet that let's you change settings on the Duet.
I just took a quick look online.. the "Maestro software Mixer Applet". Try looking in Applications or Utilities.
Need to read up about what you're using. Do not change Duet settings with Logic running.
I don't own a Duet but all of this stuff works basically the same.

Transferring music recorded at 32kHz sampling rate

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