Benefits of 96khz vs 44.1khz Sample Rate

Who knows if there are benefits to using a sample rate of 96khz instead of 44.1khz?
If so, what are they?
If not, why aren't there?

The main advantage is it lets you record higher frequencies. Whether anyone can hear those frequencies is debatable.
Other advantages are lower latency at the same buffer size (although you need more CPU power), and it's useful if you plan to do extreme pitch shifting, including using a recording as a sample in EXS. It also can make it easier for an A/D converter to do filtering properly but most modern converters can do it just fine at 44.1. For instruments that don't have higher frequencies, you won't hear a difference - it's not really more "precise". You are capturing more data, but if there are no frequencies over 20k present, that extra data is redundant and doesn't capture the waveform any more accurately than 44.1.
There's certainly benefit to going from 16 to 24 bit, but in most cases I wouldn't recommend bothering with 96 or 192.

Similar Messages

AppleTV Dac really 16bit@44.1khz sampling rate?

I was thinking about getting an AppleTV, but now have concerns with the 16bit@44khz dac. Most HD content needs 24bit@96khz, so this would suggest everything you buy on the Apple Store is down converted. What do you audio experts think about the sound quality of these things?
Also noticed that audio from a flash based ipod sounds better than a hard drive based ipod, so that leads to questions about noise and jitter on hard drive based AppleTV's. Any noise and jitter comments?

These formats produce massive bitrates, they are what are known as optional formats so would need another audio track sat alongside them, then there's licensing and the problem of delivering such large files to the AppleTV
The bitrate for 96/24 is 4mbps. Not really massive. I routinely "deliver" 96/24 files from my Macmini to the AppleTV without any problem whatsover. I have no idea of what you speak when you say "the problem of delivering such large files to the AppleTV". In fact there are over 20 such files sitting on my AppleTV as I type this, and that number increases nearly daily.
Oh and let's not forget there is no audio available in these formats.
The Beatles DVD "Love" allows my DVD player to output the music at 96/24. I captured that data stream to a disk file and now that data file resides on my AppleTV, just like I rip CDs.
Chesky download site adds 96/24 music: http://mobile.twice.com/index.asp?layout=article&articleid=CA6603951
Nine Inch Nails album The Slip available free of charge in 96/24 format: http://dl.nin.com/theslip/signup btw, this 96/24 set of files copied over to my AppleTV without any problem.

Keeping Sample Rates in Sync: 44.1kHz or 48kHz?

After reading about, and working with, sample rates in FCE and Soundtrack, I'm a bit confused about which sample rate(s) to use to keep audio and video in sync.
For example:
In Soundtrack Express, I'm working with a music track -- 44.1kHz/16bit/stereo; and a voice track -- 44.1kHz/16bit/mono.
When I export the mix to FCE I get a default option of 48kHz/16bit, which resulted in a 48kHz .aif file.
And since my FCE sequence defaults to an Aud Rate of 48kHz, it seems like the audio and video sample rates in this example would be in synch. Is this correct?
On the other hand ...
if I imported the same 44.1kHz music file directly into FCE, and created the same 44.1kHz voice over file, I'd have two files with a 44.1kHz sample rate as part of the FCE sequence, which has an Aud Rate of 48kHz. Would this configuration be considered out of synch? And if so, what problems would it cause?
Finally:
1. Do the sample rates of all media in a FCE sequence need to have the same sample rate as the sequence?
2. If so, does this sample rate need to be 48kHz?
3. Or, am I off in how I'm thinking about this?
Any feedback would be appreciated.
Thanks.

Tom Wolsky wrote:
2. It should all be the same. If you captured video that was 32K then everything should be 32K.
In my example I didn't capture any video: I imported a music file and did a voice over, both of which were 44.1K. So are you saying, that by rendering the media sample rates get synched up with the sequence aud rate?

Sample rate and audio-MIDI sync issues

Disclaimer: I did read other posts similar to this but couldn't find an answer to my specific situation. So here it is:
Logic was perfectly fine when everything was running at 44.1kHz sample rate. Then I got vocals at 48 kHz so I had to convert the sample rate in Logic Pro to match it.
Suddenly I get a slew of "Error trying to sync MIDI and audio" messages. After crying and changing the sample rate back to 44.1, then to 48, and over and over again, it finally works again.
So Logic is fine at 48kHz. But when I go to a track that's at 44.1kHz, I get the sync messages again and have to play "toggle the sample rate" for about 5-15 minutes before Logic decides whose master again.
Why is it doing this? Do I need to change Logic Pro to some kind of default settings every time I go from one song to another with a different sample rate? Or will this not be an issue if I upgrade to 7.2? (I have 7.1.0)
PowerPC G5 Mac OS X (10.4.6)

This is not a bug, but a nuisance.
You should upgrade to LP 7.1.1, which is way more stable than 7.1.0. No need for you to go to LP 7.2.
"Then I got vocals at 48 kHz so I had to convert the sample rate in Logic Pro to match it."
Are you certain, that in your song, in your regions, you used the newly converted 44.1kHz files, chosen from the Audio Window, and not (still) the old 48kHz files?
"So Logic is fine at 48kHz. But when I go to a track that's at 44.1kHz, I get the sync messages again"
LP doesn't do this well. And for a reason, but we'll not get into this now.
See: "Audio > Sample Rate > ..." and select one.
Perform proper conversion and make sure ALL of your audio files running in your song are congruent. Check your Audio Window and the files associated to the regions.
Set up your autoload to contain these settings. From then on, whenever you know that you will be importing other sample rates, change the settings in "Audio > Sample Rate > ..." before loading the sounds/files, if indeed you are starting from scratch. This will save you significant time.
Been there, done that.
sonther

Bit Depth & Sample Rate: 24 bit 96kHz? 192kHz?

I am using the Apogee Duet for Mac and iOS on my Mac and I love it - I'm thinking about getting an iPad for mobile recording (voice overs, mostly) and I wonder if Garage Band can manage 24 bit audio at 96 kHz or 192 kHz? I know that the Auria app can, so if nothing else I can just buy that, but since all I would use the iPad for is Voice Overs to edit later in a computer, a $50 app feels like overkill. Comments? Thoughts? Specs?

Well, I am new to high-end sound cards, and I may be misinterpreting the terminology, but the sound card is supposed to be a 24bit/96kHz card.
I am under the impression that one should be able to set the output quality of the card to 24bits of depth and a 96kHz sample rate, despite the speaker setting that one may be using, to decode good quality audio streams (say an audio cd or the dolby digital audio of a dvd movie.) I can currently achieve this only on 2. speaker systems (or when i set the speaker setting of the card to 2.) Otherwise the maximum bit depth/sample rate I can set the card output to is a sample rate of 48kHz and a bit depth of 6bits.
Am I mistaken in thinking that if I am playing a good quality audio stream I should be able to raise the output quality of the card to that which it is advertised and claims to have?
Thnx

How to create a wav file from 24bit 96Khz sampling rate data

Hi
I am trying to make an VI which will play sound while acquiring data from PXI 4472 DAQ card.
My sampling rate is 96Khz and PXI 4472 card is a 24bit card.
Wave files are in 8 or 16 bit and the sampling rate is 8000, 11025, 22050 and 44100. How will I be able to play the data which I am acquiring.
How would i normalize the data into the required format needed for most of the sound cards to play.
Or are there any codec available in Windows XP which i call to play a 96KHz 24 bit sample
Does anybody ever encountered this type of problem.
Thanks in advance
Nitin

Whilst the 'standard' RIFF format specification usually accomodates 16 bit data, there is of course no reason that you can not create your own extension. It just won't be playable by Media player using the 'standard' installed drivers or codec. This may not be a problem....
WAV files can and do support other formats, you just need to know how to handle them......
There is howerver a 4GB limit (related to the pointer size in the WAV specification) which with higher bit depths on the sampling does start to become a bit of a problem.
To give you a few samples of other types of wav files check out the following site here
http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples.html
The following definitions for WAV audio formats may also be of interest here
http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/WAVE.html
Good luck with 24 bits.

How do I change the audio sample rate from 48kHz to 44.1kHz for Mpeg2

Hey all,
I've been searching for a while but haven't found any direct answers in the forums or the user manual so here goes.
I have to dispatch a file to Bloomberg TV and the file specs they have given me are as follows:
Video Standard: MPEG-2, MP:ML, 4:2:0
Frame Rate: PAL 25fps
Video Size 720 x 405
Aspect Ratio 16:9
Audio Standard MPEG-1 Layer 2
MPEG-2 Program Stream Mux rate 6mbs per second
Bit Rate Type: CBR
Video Bit Rate 5.7mbs
Audio Bit Rate 192kbs
Audio Sample Rate 44.1Khz
Interlacing: Upper Field first (why they want interlaced for web streaming is beyond me)
GOP Structure: IBBP
I-Frame distance 12
Now everything above is fine except the audio encoding because even though I have set up a new setting from scratch I cant find anywhere to adjust the audio sample rate. The Inspector tells me the Audio encoder is set to:
Format: MPEG
Sample Rate: 48.000kHz
Channels: 2
Bits Per Sample: 16
Anyone Know how or even if I can change these audio settings? The only adjustments I can find are the filters or the transport/program stream option. I have it set to program as specified by Bloomberg.
Thanks in advance
J

The only setting that I could find in compressor that lets your change the bitrate to 44.1 is when you create a new dolby digital setting and then under the inspectors audio tab/Target System button, change the button to Generic AC-3. When done, you can change the Sample Rate to 44.1.
Hope this helps?

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

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.

Converting a whole project from 96khz to 44.1khz

I've recently switched from recording at 24bit / 44.1khz to 24/96. Unfortunately this has brought me swiftly to the limits of my G5's CPU capabilities. What is the best way to convert an entire arrangement from 96khz to 44.1khz? I am not looking forward to going through each audio file one by one in Sample editor. There surely is an easy way to convert them all at once leaving the arranged regions uneffected.
My arrangements are typically quite big and I use a few power hungry plug-ins such as BFD, Stylus RMX and Atmosphere. I have easily enough RAM and I have 4 fast hard disks with critical streaming things like BFD on separate disks. The main audio files are on the really fast RAID 0 equipped Miglia DualDisk 320. The only thing holding me back is the actual CPU. Things have got a bit unusable lately with Core Audio error messages stopping playback etc.
What sort of plug in / track counts are people getting who have faster processors and who record at 24/96?

Go to the Audio Window. Select all of the files. Pull down "Audio File" menu to "Copy/Convert File(s)". Choose what rates, etc. you wish.
All done at once.

Creative Audigy 2 NX Bit Depth / Sample Rate Prob

This is my first post to this form
Down to business: I recently purchased a Creative Audigy 2 NX sound card. I am using it on my laptop (an HP Pavilion zd 7000, which has plenty of power to support the card.) I installed it according to the instructions on the manual, but I have been having some problems with it. I can't seem to set the bit depth and sample rate settings to their proper values.
The maximum bit depth available from the drop down menu in "Device Control" -> "PCI/USB" tab is 6 bits and the maximum sample rate is 48kHz. I have tried repairing and reinstalling the drivers several times, but it still wont work. The card is connected to my laptop via USB 2.0.
I looked around in the forms and found out that at least one other person has had the same problem but no solution was posted. If anyone knows of a way to resolve this issue I would appreciate the input!
Here are my system specs:
HP Pavilion zd 7000
Intel Pentium 4 3.06 GHz
GB Ram
Windows XP Prof. SP 2
Thnx.
-cmsleimanMessage Edited by cmsleiman on -27-2004 09:38 PM

Well, I am new to high-end sound cards, and I may be misinterpreting the terminology, but the sound card is supposed to be a 24bit/96kHz card.
I am under the impression that one should be able to set the output quality of the card to 24bits of depth and a 96kHz sample rate, despite the speaker setting that one may be using, to decode good quality audio streams (say an audio cd or the dolby digital audio of a dvd movie.) I can currently achieve this only on 2. speaker systems (or when i set the speaker setting of the card to 2.) Otherwise the maximum bit depth/sample rate I can set the card output to is a sample rate of 48kHz and a bit depth of 6bits.
Am I mistaken in thinking that if I am playing a good quality audio stream I should be able to raise the output quality of the card to that which it is advertised and claims to have?
Thnx

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

Maximum audio sample rate and bit depth question

Anyone worked out what the maximum sample rates and bit depths AppleTV can output are?
I'm digitising some old LPs and while I suspect I can get away with 48kHz sample rate and 16 bit depth, I'm not sure about 96kHz sample rate or 24bit resolution.
If I import recordings as AIFFs or WAVs to iTunes it shows the recording parameters in iTunes, but my old Yamaha processor which accepts PCM doesn't show the source data values, though I know it can handle 96kHz 24bit from DVD audio.
It takes no more time recording at any available sample rates or bit depths, so I might as well maximise an album's recording quality for archiving to DVD/posterity as I only want to do each LP once!
If AppleTV downsamples however there wouldn't be much point streaming higher rates.
I wonder how many people out there stream uncompressed audio to AppleTV? With external drives which will hold several hundred uncompressed CD albums is there any good reason not to these days when you are playing back via your hi-fi? (I confess most of my music is in MP3 format just because i haven't got round to ripping again uncompressed for AppleTV).
No doubt there'll be a deluge of comments saying that recording LPs at high quality settings is a waste of time, but some of us still prefer the sound of vinyl over CD...
AC

I guess the answer to this question relies on someone having an external digital amp/decoder/processor that can display the source sample rate and bit depth during playback, together with some suitable 'demo' files.
AC

Tuner / sample-rate problem

I'm using MainStage 2.0.1 on OS X 10.6.2 with an Echo AudioFire12 interface.
I have the interface set to 44.1kHz and the sample rate in MainStage Audio Preferences set to the same sample rate, however the tuner readout is sharp by almost a semitone when I try to use it.
Also the audio output is sharp by the same amount when I use the built-in ouput of the mac with interface input.
Seems to be a sample rate issue but I don't know how to fix it.
Any suggestions?

Turned out this was a hardware issue with the Echo AudioFire12.
It was repaired under warranty with excellent customer service from Echo Audio.

NI-DAQmx frequency sampling rate

Hi there!
I'm working on setting up a data acquisition Labview VI, to measure different signals on a test rig.
I'm using the NI-DAQmx assistance (the Express VI?) to continously measure analog signals (Variable current, voltage and temperatures). This is working just fine, and i can change the sampling rate by writing to the express VI. The idea is, that the user can change the sampling rate from around 1 to 500 Hz.
We do however have a sensor that transmittes digital signals (a frequency), and are using a NI-9423 module to "read" it. As this is a digital signal, another NI-DAQmx express VI is needed to handle it (that's ok), but so far we can't figure out how to alter the sampling rate - it's apperently locked at 1kHz.
Being that we want to merge the analog and digital signals to one array, we are recieving overflow errors from the "analog" DAQ, if it's not set at exactly 1kHz.
So, in short - is it possible to change the sampling rate of a DAQmx recieving frequencies? So that we to DAQ assistences have the same sampling rate?
Help would be greatly appreciated!
- Nicklas
Attachments:
DAQissue.PNG ‏64 KB

Unlike voltage measurements, which tend to be (more or less) instantaneous, frequency measurements take a finite (and often variable) amount of time.
If it is a slow signal then you measure the number of counts of your reference clock that occur in one period of your input signal. As your input signal varies in frequency, so does the measurement rate. If it is a fast signal, you can either measure how long it takes to get n cycles or your input (again variable) or you could count how many cycles of your input occur in a fixed time period.
The NI help on frequency measurements describes three different ways you can configure a counter to measure frequency.
The long and short of this is that generally counter measurements come at variable measurement rates which can be problematic to fit in with a fixed rate loggin system. If the measurement period is much smaller than your desired rate then you can wait and trigger a measurement at regular intervals. If not, you can let the counter run at its own rate, placing the latest result on a notifier, and in another loop just read the latest measurement from the notifier each time you want to record a result. Depending if you counter is running faster or slower than your desired logging rate you will end up with either missed samples or repeated samples. There are inherant timing inaccuracies in both approaches because, unlike analog measurements, the counter measurement is not made at 'that exact time, now!' but over a period of time which may be long or short compared to your logging rate.

Benefits of 96khz vs 44.1khz Sample Rate

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