Changing sampling rate and bitrate

If I drop in an .aiff audio file into Garage Band 2 (or iTunes), is there a function that allows me to adjust the audio sampling rate and bitrate?
When I created this podcast in Adobe Auditions, the were about 10MB for a 22 min show.
When I do it through GB, it's about 26MB for the same time;
I just want to tune it down slightly.
Thanks!

GB exports uncompressed 44.1K 16-Bit AIFF files. You need to convert them to Mp3s or AAC files
http://thehangtime.com/gb/gbfaq2.html#converttomp3

Similar Messages

Sample Rate And Bitrate Not Allowed ... ?

I have almost 70GB of my music in my iTunes, ALL in AAC 320kbps 48Khz. Now that I just updated to iTunes 9, I just tried to add more of my new CD's to my library and they are being imported at 44Khz...?? I went to change it back to 320/48 and I get ...
"The selected combination of bitrate and sample rate is not allowed"
Bug? Anyone else have this problem? HELP!
btw ..I tired to use my laptop and my work PC to see if it's just this machine that is causing the problem....and those too (now updated to iTunes 9) are doing the same thing.

I don't know why a previous version might have allowed a 48khz sample rate, but audio CDs are 44.1kHz (that sample rate is locked into the Red Book CD standard) so there would be no good reason I can think of to attempt to import at a 48KHz sample rate. It won't increase quality and could cause artifacting.
Regards.

Multi Track settings (Sample Rate and Bit Rate)

I'm setting up my multi-track in 5.5 as 44100 16 bit, but when I create it...it's saying its 44100 32 bit (at the bottom of the multi-track). Is there a way I can change this setting? I've explored the preferences all day long and still can't find any answers.
Thanks

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

Multitrack mode: change sample rate

Is there a way to change the sample rate of a multitrack project that has already been created? The only point at which I have seen one can set this is when creating a new project.
I have a couple of empty projects (meaning no audio data) with the correct mixer settings, buses, etc., into which I want to pop various audio files with various sampling rates, one at a time. Therefore I would just as soon adjust the rate of the project according to the audio files being inserted rather than perform conversions of the audio files to fit the project.

TheodoreG - if you save your session file as an XML and email it to me I will change the sample rate and send it back to you.
While I haven't tried this before Ozpeter is perfectly correct it is technically possible.
After I send the XML session file back to you let us know if it was successful - if so - I will build in a facility to change it in one of my utilities.
I'm not sure if some other maths may be required but I'm digressing
Contact me through http://www.mediasweeper.com.au

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

DASYLAB QUERIES on Sampling Rate and Block Size

HELP!!!! I have been dwelling on DASYLAB for a few weeks regarding certain problems faced, yet hasn't come to any conclusion. Hope that someone would be able to help.Lots of thanks!
1. I need to have more data points, thus I increase the sampling rate(SR). When sampling rate is increased, Block size(BS) will increase correspondingly.
For low sampling rate (SR<100Hz) and Block size of 1, the recorded time in dasy and the real experimental time is the same. But problem starts when SR>100Hz for BS=1. I realized that the recorded time in dasylab differs from the real time. To solve the time difference problem, I've decided to use "AUTO" block size.
Qn1: Is there any way to solve the time difference problem for high SR?
Qn2: For Auto Block Size, Is the recorded result in dasylab at one time moment the actual value or has it been overwritten by the value from the previous block when AUTO BS is chosen.
2. I've tried getting the result for both BS=1 and when BS is auto. Regardless of the sampling rate, the values gotten when BS=1 is always larger than that of Auto Block size. Qn1: Which is the actual result of the test?
Qn2: Is there any best combination of the block size and sampling rate that can be used?
Hope someone is able to help me with the above problem.
Thanks-a-million!!!!!
Message Edited by JasTan on 03-24-2008 05:37 AM

Generally, the DASYLab sampling rate to block size ratio should be between 2:1 and 10:1.
If your sample rate is 1000, the block size should be 500 to no smaller than 100.
Very large block sizes that encompass more than 1 second worth of data often cause display delays that frustrate users.
Very small block sizes that have less than 10 ms of data cause DASYLab to bog down.
Sample rate of 100 samples / second and a block size of 1 is going to cause DASYLab to bog down.
There are many factors that contribute to performance, or lack there of - the speed and on-board buffers of the data acquisition device, the speed, memory, and video capabilities of the computer, and the complexity of the worksheet. As a result, we cannot be more specific, other than to provide you with the rule of thumb above, and suggest that you experiment with various settings, as you have done.
Usually the only reason that you want a small block size is for closed loop control applications. My usual advice is that DASYLab control is around 1 to 10 samples/second. Much faster, and delays start to set in. If you need fast, tight control loops, there are better solutions that don't involve Microsoft Windows and DASYLab.
Q1 - without knowing more about your hardware, I cannot answer the question, but, see above. Keep the block size ratio between 2:1 and 10:1.
Q2 - without knowing more about your hardware, and the driver, I'm not sure that I can fully answer the question. In general, the DASYLab driver instructs the DAQ device driver to program the DAQ device to a certain sampling rate and buffer size. The DASYLab driver then retrieves the data from the intermediate buffers, and feeds it to the DASYLab A/D Input module. If the intermediate buffers are too small, or the sample rate exceeds the capability of the built-in buffers on the hardwar, then data might be overwritten. You should have receive warning or error messages from the driver.
Q3 - See above.
It may be that your hardware driver is not configured correctly. What DAQ device, driver, DASYLab version, and operating system are you using? How much memory do you have? How complex is your worksheet? Are you doing control?
Have you contacted your DASYLab reseller for more help? They should know your hardware better than I do.
- cj
Measurement Computing (MCC) has free technical support. Visit www.mccdaq.com and click on the "Support" tab for all support options, including DASYLab.

HT5848 What is the sampling rate and codec for iTunes Radio. Is it lossless encoded?

What is the sample rate and codec for iTunes Radio? Is it lossless encoded?

I have to agree with you. There are several forum discussions on bit rate being as high as 256 kbps but I don't see how it could be more than 96 kbps based on the poor sound quality I'm hearing. I'm comparing it to an internet radio station that is 128 kbps and sounds much better.
Am I missing something?

Conflict between the saved data and the sampling rate and samples to read using PXI 6070e

Hello, I am using PXI 6070e to read an analog voltage. I was sampling at 6.6 MHz and the samples to read were 10. So, that means it should sample 10 points every 1.5 um. The x-axis of the graph on the control panel was showing ns and us scale, which I think because of the fast sampling and acquiring data. I use "write to measurement file" block to save the data. However, the data was saved every 0.4 second and as 35 points data at the beginning of each cycle (e.g. 35 points at 0.4 sec and 35 at 0.8 sec, and so on) and there was no data in between. Can anyone help me how there are 35 reading points every cycle? I could not find the relation between the sampling rate and samples to read, to 35 points every 0.4 second!
Another thing, do I need to add a filter after acquiring the data (after the DAQ assistant block)? Is there anti-aliasing filter is built in PXI 6070e?
Thanks for the help in advance,
Alaeddin

I'm not seeing anything that points to this issue. Your DAQ is set to continuous acquire. I'm not sure if this is really what you want because your DAQ buffer will keep overwriting. You probably just want to set to Read N Samples.
I'm not a fan of using the express VIs. And since you are writing to a TDMS file, I would use the Stream to TDMS option in DAQmx. If you use the LabVIEW Example Finder, search for "TDMS Log" for a list of some good examples.
There are only two ways to tell somebody thanks: Kudos and Marked Solutions
Unofficial Forum Rules and Guidelines

Sample rate and / or Bit depth probl

I am in the middle of mastering a tune, but when I come to play the tune in Soundforge 8 I get the error message: One or more playback devices do not support the current Sample rate and / or Bit depth. I am using a Audigy2 Platinum with the ASIO A400 driver and I'm sure it should be A9000, I can't find the driver update for this, does anyone have a link to this, or is it something else I should be looking at?

A400 doesn't have anything to do with a version number. It's related to the ressource allocated to your card.
What kind of source are you trying to play ?

PXI-4462 actual sample rate and delay...

Hi All!
Is the DAQmx properties, which give me actual sample rate and filter delay? Or I must manualy calculate this values, as described in DSA manual?
Jury

Hi Jury,
I am not sure if I fully understand your question, but there are DAQmx properties that do return the actual sample rate, but there is not one for filter delay. If you place a DAQmx Timing property node down, the actual sample rate will be returned by selecting Sample Clock » Rate.
Regards,
Jason D
Applications Engineer
National Instruments

Changing sample rate during acquisition

I need to programmatically change sample rate during acquisition: My application involves sampling at slow rate most of the time but during "region of interest" I need to change sampling rate to fast (close to max rate of board).

I think it's also possible to use the same (high) sample rate during your
complete acquisition, but in the case you need the 'slow' rate only take
from the array sample 0, 10, 20 etc. or any number you prefer.
In the end I think you want to make sure your computer doesn't have to work
that hard all the way, or if you want to save the data, it's not that big...
If I'm wrong, please tell me.
Ron@ld
Lancer schreef in berichtnieuws
506500000008000000DF570000-1027480788000@exchange.ni.com...
> I need to programmatically change sample rate during acquisition: My
> application involves sampling at slow rate most of the time but during
> "region of interest" I need to change sampling rate to fast (close to
> max rate of board).

Restore Old LP'S AT 96000 Hz Sample Rate and 24 bit Resolution?

I restore LP vinyl albums using the above Sample Rate and Resolution.
Do I have to convert to 44100 Hz, and 14-bit before adding to my iTunes Library?

Ringmaster wrote:
Thanks for your help!
I guess I had better keep all of them in this format. It might make a difference if I decide to burn CD's; also, all of my back ups will be of the same format.
Do what works for you, but neither of those two things is an advantage. If you burn audio CDs, they will be downsampled to 44.1 kHz anyway. And as far as backups, iTunes does not care if different songs are in different formats, nor does any other music player that I know of.
Unless you plan on further mastering or remixing, or you really just like the higher fidelity, there is no obvious reason not to use 44/16.

I have one application that has requirement to do low and high speed acquisition. I want to change sample rate while running. BUT... I have E series Device

I am writing control software for a process that is usually dull and
requires only 10 Hz acquisition rate. At particular times during
the sequence, however, we are interested in looking at a couple of
channels at 1000 Hz. My approach so far is to configure my
Buffered DAQ to run at the higher rate at all times. When we are
in the 'high-speed DAQ' mode, the program logs every point to
disk. In the 'low-speed' mode, I am picking off every nth (in
this case, 10th) point to log to disk. At all times, I update my
GUI indicators on the front panel at a maximum of 4 times per second (I
find that anything faster results in an uncomfortable display), so I
fill up a FIFO with data in my acquisition / logging loop, and read the
FIFO in the display loop. The data in my GUI display can be up to
250 milliseconds off, but I find this acceptable . As a side note, I
need buffered Daq with hardware timing, as software timing results in
lost data at 1000 Hz.
This all works fine and dandy, but I am convinced that it is not the
most elegant solution in the world. Has anyone developed a
buffered DAQ loop where the scan rate can be adjusted during
operation? I would like to change the rate of the E-Series card
rather than relying on down-sampling as I am now doing.
The reason I have concern is that at the moment I am simulating my AI
using MAX and when running the down-sampling routine, I consistently
miss a particular event on the simulated data becuase the event in
question on the simulated data always occurs at the same 'time', and I
always miss it. Granted, while it is unlikely that my measured
signal and my acquisition are perfectly synchronized in the real world,
this particular situation points out the weakness in my approach.
More than anything, I am looking for ideas from the community to see
how other people have solved similar problems, and to have you guys
either tear apart my approach or tell me it is 'ok'. What do you
think?
Wes Ramm, Cyth UK
CLD, CPLI

Adding to Alan's answer:
One of the problems that comes with these tricks for variable-rate acquisition is being able to match up sample data with the time that it was sampled.
If you weren't using either of E-series board's counters, there is a nifty solution to this! You'll be using 1 of the counters to generate the variable-rate sampling clock. You can then use the 2nd counter to perform a buffered period measurement on the output of the 1st counter. This gives you a hw-timed measurement of every sampling interval. You would need to keep track of a cumulative sum of these periods to generate a hw-accurate timestamp value for each sample.
Note: the very first buffered period measurement is the time from starting the 2nd counter until the first active edge from the 1st. For your app, you should ignore it.
-Kevin P.

Can't change sample rate for digital input on Mac Pro

Hello all,
on my Mac Pro with 10.4.10 I can't change the sample rate for the digital input. Whenever I choose 48000 or 96000 Hz it returns to 44100 Hz after a few seconds. Feeding a 24 bit/96000 Hz signal from an external ADC into the optical input doesn't help. No input signal is available for digital recording software such as Sound Studio or Cubase. After changing the sample rate in audio midi configuration to 96000 Hz the sound can be heard for a few seconds but when the setup returns to 44100 automatically the signal is, of course, lost.
I've deleted all relevant preferences and restarted with resetting paramter RAM. Still the same. With external hardware such as M-Audio Firewire equipment setting the sample rates works properly.
Thanks to all for helpful clues.

Hi,
when E&M signaling is configured on digital interface like the VWIC is, 2 or 4 wires operation is not applicable because there are no wires at all, and reported only for compatibility with the analog E&M card.
Consequently, you cannot configure that and it will not make any difference to effect of the connection.
Please rate post if it helps!

How to use on board counter to change sample rate dynamically on pci-6134

Hi,
I am relatively new in LabView.
I am making power quality measurement system and I need to vary the sampling rate of my pci-6134 dynamically (all channels simultaneously). What I need is to have a constant amount of samples in each period of measured signal (grid voltage), which changes slightly all the time. Therefore I will have to measure the voltage, find its exact frequency and then adjust the sampling rate of daq accordingly. I know that there will always be some delay, but I would rather like not to go into any predictive algorithms...
I have found an information in the Forum that one of possible solutions is to use an onboard counter to change the sampling frequency but I have no idea how to make that. Can someone help me or possibly show an example?
Is there a simple way to solve that problem?
Thanks in advance
Andrzej

At least at a glance, the code generally looks like it ought to work. Two thoughts:
1. Instead of getting into PFI3 vs PFI8 routing stuff, can't you just specify "Dev1/Ctr0InternalOutput" as the AI Sample Clock source? (You may need to right-click the terminal to get at the menu that exposes the so-called "advanced terminals").
2. Try writing both the freq AND duty cycle properties when you want to update the freq. Or try using the DAQmx Write vi instead of a property node. My past experience suggests that writing only the freq property *should* still work, but writing both isn't hard to try and may turn out to help if the behavior of your version of DAQmx differs somehow.
-Kevin P.
P.S. Bonus 3rd thought. I just went back to reread the thread more carefully, including your first screenshot. I'm now thinking that maybe the hardware actually WAS behaving properly, and that you just weren't aware of it. When you query the AI task for it's sampling rate, all the task can know is whatever rate you told it when you configured it outside the loop. So even as you change the counter freq to change the actual hardware sampling rate on the fly, the AI task will continue to report its orig freq. After all, how is *it* supposed to know?
Try an experiment: Set your original freq very low so that the AI task produces a timeout error without getting all the requested samples within the 10 sec timeout window. Run and verify the timeout. Then run again, but after 3-5 seconds set a new frequency that will produce all those samples in another 1 sec or less. Verify that you get the samples rather than timing out. That should demonstrate taht the counter freq change really *does* produce a change to the hardware sample rate, even though the task property node remains unaware.

Changing sampling rate and bitrate

Similar Messages

Maybe you are looking for