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

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Audigy 4 pro vs. cubase sx sample rate - fight to the de

helloI recently bought an audigy 4 pro soundcard, heard it was probably the best for home recording etc.
I have encountered some problems... I have a load of backing tracks stored in cubase sx 2.2, but now when i open them up it shows the message "sample rate could not be set. This may be due to the sample clock being set to external sync."The files now run at 48kHz instead of 44.kHz, making them jittery, out of time, or chipmonk like. I've read through various messages around similar problems, i've tried everything & nothing has worked...in cubase - project/project set up/sample rate = 48kHz and can not be changed.
in cubase - device set up/vst multitrack/asio driver = creative asio + clock source = internalit only offers 5 asio drivers, asio direct x full duplex, asio mulitmedia, creative asio, SB audigy4 asio 24/96 [a400] and SB audigy4asio [a400] i do not get the offer of asio4all.While i have also tried going through the control panel...controlpanel/audio control panel/device settings/digital out samplerate and setting it to 44.kHz this doesn't seem to change anything. I have no problem with recording new songs at a higher sample rate, but has audigy 4 pro rendered all my old songs useless?please help!!!!!!!!

Doogs wrote:
helloI recently bought an audigy 4 pro soundcard, heard it was probably the best for home recording etc. I have encountered some problems... I have a load of backing tracks stored in cubase sx 2.2, but now when i open them up it shows the message "sample rate could not be set. This may be due to the sample clock being set to external sync."The files now run at 48kHz instead of 44.kHz, making them jittery, out of time, or chipmonk like. I've read through various messages around similar problems, i've tried everything & nothing has worked...in cubase - project/project set up/sample rate = 48kHz and can not be changed.in cubase - device set up/vst multitrack/asio driver = creative asio + clock source = internalit only offers 5 asio drivers, asio direct x full duplex, asio mulitmedia, creative asio, SB audigy4 asio 24/96 [a400] and SB audigy4asio [a400] i do not get the offer of asio4all.While i have also tried going through the control panel...controlpanel/audio control panel/device settings/digital out samplerate and setting it to 44.kHz this doesn't seem to change anything. I have no problem with recording new songs at a higher sample rate, but has audigy 4 pro rendered all my old songs useless?please help!!!!!!!!
If you RTFM, you'll find out, Audigy 4 is locked into 6-bit/48kHz and 24-bit/96kHz resolutions when ASIO driver is in use.
I suppose, you still can (if not saved @ 48kHz) load your projects into SX @ 44. kHz, by selecting MME drivers instead of any ASIO.
Perhaps installing Asio4All gets it popped into that list you have there.
There are also tools to convert from 48-->44., but the source has to be as wave format.
Here is one freeware SRC tool @ http://www.voxengo.com/product/r8brain/.
.jtp

Problems with sample rate

hello:
I have a sound blaster audigy platinum ex and I'm trying to make some recordings with adobe audition, but this program works at 44khz and the sound card seems to work only at 48khz, so it doesn't let me recording. how can I change the sampleing rate of the card?
thanks.

By setting it in Audition audio settings. W/ ASIO device drivers you can have 16-bit/48kHz (and IIRC, 24-bit/96kHz) mode(s) only). For other modes you need to use other device driver mode. Also, Asio4All should handle all modes (16/24-bit and 44.1, 48 and 96kHz).
- http://www.asio4all.com
The quality in 44.1kHz mode is POOR for your card. If you switch to 24-bit/44.1kHz quality is otherwise better but the frequency gets cutted already around 15kHz.
You get the best quality by setting the sample rate to 96kHz (16 and 24 bit resolution are ~same in these cards) for recordings (for project?) and then when export the final mix just use SRC to CD quality (Audition do have quite good SRC quality but, if you use additional software for SRC (like Voxengo r8brain (free)), export the final mix using 96kHz and then execute SRC).
Something more on Creative ASIO -
http://forums.creative.com/creativel...essage.id=1726
jutapa
Message Edited by jutapa on 11-09-200602:43 PM

USB Mics, Sampling Rates and "dumb" apps--help please

I hope someone here can explain this better than the support people at Blue did.
I have a Blue Yeti USB mic. It has a fixed output of 16 bits/ 48kHz. In most recording apps, I can set the sample rate in the app to match at 48kHz. What happens when using this mic with a "dumb" app like Camfrog or Skype, where the user cannot set a matching sample rate in the app? Will the pitch be affected due to the 48 kHz fixed rate of the mic, as opposed to a more common 44.1 kHz?
Any insight into this would be great. I could ask people on Camfrog if my voice sounds right, but they don't know what my normal voice sounds like!
Thank you.

Darn. Not being able to type at login sounds like an issue mine has every so often, although if you don't log in RIGHT AWAY on mine, you have to force it down and restart because when you type, it pinwheels and never stops. Mine also hasn't been able to restart or shutdown so I haven't been able to apply updates.

How do I capture at an audio sample rate of 44.1 KHz?

Hey,
I'm wanting to capture some footage that's been recorded at a 44.1 KHz sample rate. Presets on FCP 5 only have 48 KHz.
How do I change this?

I just checked the import settings on two other editing systems I use... Canopus DV Storm II and the Matrox RTX 100 Extreme. Only the Canopus has a preset for 44.1k import. Even Adobe Premiere, using the native IEEE 1394 connection, doesn't have it. I am not sure if FCP has a "CUSTOM" import dialog that might allow you to manualy select?
If you have found this information helpful, please select the icon for this. If it helps you resolve this issue, please select "answered".

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.

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.

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.

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:
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-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

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.

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.

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).

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

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.

Recording LP records as source material- Sample Rate

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