AAC sample rates for N95

Which AAC sample rates are supported by the N95 other than 44.1khz; is 48KHz supported for example?
If so is it possible to change the sample rate used within Nokia Music Manager?
Thanks,
Zack

YES both 44.1 KHz and 48 KHz sample rates are supported for AAC.
You should use something "other" than Nokia Music Manager if you can. Like WINAMP or other audio encoder (SUPER) to convert files. Lot's of problems trying to use Nokia MM.
Message Edited by el_loco on 29-Sep-2008 06:49 PM
640K Should be enough for everybody
El_Loco Nokia Video Blog

Similar Messages

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!

Audio Sample Rate for Video Clips?

How do you check the audio sample rate for a clip of video?
Thanks.

scroll over in the browser until you get to the column for audio. That's one way.

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.

Separate sampling rate for two different channels for a USB-6009 daq

Hi,
I am using a USB-6009 and incorporating the 'daq assistant' to change the sample rate. I am trying to find a way to set the sampling rate to two unique values for two separate channels. I've tried setting up two daq assistants and adjusting the sample rate different for each channel, though this does not work. Is there any way to set the sample rate high for all channels then reduce the rate for a different channel - or an alternative? I would appreciate any input on this, thank you!
- Anthony
Solved!
Go to Solution.

All tasks on a DAQ board that use a sampling clock must use the same clock. Therefore, you cannot have two tasks on the same DAQ board sample at different rates.
Alternatives would be:
1. to combine all of the channels into a single task and just accept the extra data
2. get an extra DAQ board to use in parallel
There are only two ways to tell somebody thanks: Kudos and Marked Solutions
Unofficial Forum Rules and Guidelines

Why does the sampling rate for my DAQCard-700 max out at 6578Hz regardless of the number of channels I'm sampling?

My DAQCard-700 is acting funny. I am using LABView 6.0.2 on a Dell Inspiron 8100 laptop for continuous analog input. For some reason my DAQCard-700 won't let me acquire multiple channels unless I create virtual channels in max and then run AI Config with a list of the virtual channel names. That's fine (but weird), but I max out my sampling rate at 6578 Hz, which is the max sampling rate if I would have sampled all 16 (Referenced, single-ended) AI channels. But I'm not -- I'm sampling 6 channels, as proven by the "number of channels" indicator in AI config and just looking at my data. If I sample just one channel, I can get up to 105248Hz. Is my card f
ried, or am I missing a hidden setting somewhere? I have the latest NIDAQ.

Hello;
By the description of your problem, everything is pointing us to a hardware failure.
There aren't any tricks to acquire multiple channels with that card.
The best way to go about this is to have NI to fix the board for you.
Regards
Filipe

Different Sampling rates for different channels in Analog Input

Hi,
I would like to acquire data at different sampling rates on different channels say ACH 0, ACH 1 and so on. I have a PCI 6052E board and NI DAQ 6.9.2. Also is it possible to simultaneously perform Analog output on two different channels along with the Analog input? What will be the problems/consequences as far as the system resources are concerned. I am a beginner in this area and would greatly appreciate any help/pointers for my queries.
Thanking you in advance
Deepak

Search the eaxamples that ship with LV.
Theer is one called simultaneous input and output or something like that.
It will get you started.
re: multiple scan rates. This is acoomplished by sampling all channels at the highest rate and throw away the expttra samples you do not need.
Ben
Ben Rayner
I am currently active on.. MainStream Preppers
Rayner's Ridge is under construction

Sample rate for digital sampling (cDAQ-9172 & NI 9401)

Hi!
I have a cDAQ-9172 with a NI 9401 C-series module (digital). I would like to sample the digital inputs with a sample rate of e.g. 400 kHz or 200 kHz. My problem is that I can only select a the 100kHzTimebase clock, and therefore only get a 100 kHz sample rate. The 20MHzTimebase clock is too fast, since it gives me a sample rate of 20 MHz). Is it possible to get a user defined sample rate of e.g. 200 kHz, by e.g. dividing down the 20MHzTimebase clock?
Solved!
Go to Solution.

The cDAQ-9172 chassis does not have an internal timing engine for digital input however you can use one of the onboard counters to generate your clock. Set your pulse train generation counter to be one of the internal counters, such as "cDAQ1/_ctr0" and your digital input sample clock source to be /cDAQ1/Ctr0InternalOutput".

How to know the sampling rate for NI6624?

Dears,
I am trying to measure a transient signal that is a time-vary counter train. The target frequency is increased from 0 Hz to 50 Hz when the measurement time rises to 1 s from 0 s. The NI 6624 card and the LabVIEW DAQmx have been adopted. In the block diagram, the terminal of measurement method “Low Frequency with 1 Counter” is set in “DAQmx Create Channel (CI-Frequency).vi”, and the “Finite Samples” mode is chosen in “DAQmx Timing (Implicit).vi”. Then the transient signal points (increasing-frequency points) will be got successfully within 1 s. Now I have a question: how do I estimate the time step “dt” between these data points? Knowing the default sampling rate of the card seems a better way to help me to define the "dt", and calculates the time stamp at each data point. If the foregoing concept is true, how the internal sampling rate in NI6624 obtains? Beside, for the transient counter signal, any way to get the time stamp of data points is also welcome.
Thanks for anyone comment,
Adan

Adan,
When selecting "Implicit" as the DAQmx Timing type, you are indicating
that a data point will be taken for every measurement the counter
performs. When you create a task of type "Low Frequency with 1
Counter," the counter simply uses the card's internal timebase to
measure the period between edges of your signal. It then takes this
period measurement and converts it to a frequency. Therefore, the
spacing between the samples you read out is simply the inverse of the
subsequent frequency measurement sample.
Hope this helps,
Ryan Verret
Product Marketing Engineer
Signal Generators
National Instruments

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.

Samples rates for sounds

I am thinking of purchasing some sounds from sounddogs.com, they offer their sounds in sample rates of 44.1 kHz and 48 kHz. Obviously 48 kHz is better, but would there be much of difference between 48 kHz and 44.1 kHz?

As David says, 48kHz is better but you probably wouldn't notice the difference. Almost everything I have ever used has been 44.1kHz and I've never had a problem.
If you are buying them it makes sense to opt for 48.

Setting sample rate for sinus analog output

Hello,
I've been trying to do something very simple : using an analog output of the card PCI 6221 to produce a sinus curve of frequency 50 Hz. For this I used a Vi to create a sinus curve and the different DAQmx VIs. But I have difficulties understanding the principle of virtual channel and I think I'm doing a mistake setting the sample rate and samples number : one time for the sinus vi, second time for "DAQmx - Timing". Should I use the same values for both of these VIs ?
On my oscilloscope, with frequency=50Hz and sample rate=1kHz, I get a null signal. Then depending on both values, I get differently rated signals. For example with f=1Hz and sr=10kHz, a sinus of frequency 0,7 Hz.
Solved!
Go to Solution.
Attachments:
Sinus analog output.vi ‏32 KB

Yes, thanks for your advice. I used the structure given in the example and now it's working fine. I'm still not sure what I did wrong though.
I would have a second question now (should I create a new topic?):
I put a continuous sine wave on the analog output. As soon as this is running, (or maybe after a short delay) I want to measure a limited amount of samples on my analog input. How can I be sure, it's not going to start measuring before the output is properly set ?
I don't think a trigger would solve the problem since I'm going to vary the output Amplitude.

Minimum sample rate for 10KHz square wave with MIO

Need to display a 10KHz square wave input to a 16E MIO. Presently sampling 1 channel at 400KS/s. Gives me 2.5uS/sample. Even at 800KS/s-same thing. How come I can't get a decent picture of the waveform. Presently I see a ramp up-down akin to an RC time constant. At 1KHz I get a decent waveform. I'm wondering if I have something set wrong in my program. Did I miss something in determining the proper sample rate?

Hi Steve,
Are you sure you are sampling at 400 and 800 KS/s?
I thought those boards only go up to about 100KS/s.
(show your point on your plot. Are you getting 40 samples per period?)
If you really are sampling that fast, then you have partially answered your own question. There is a small capacitance at teh input to the board. A high impeadance source will combine the input capacitance of the board to give you the charge ramp you mentioned. Only solution to this situation is ussing a low impeadance output buffer between your signal source and the AI.
Ben
Ben Rayner
I am currently active on.. MainStream Preppers
Rayner's Ridge is under construction

What is the current sampling rate for iTunes store purchases

As a dinosaur, I still like getting my music on CD's (and, in fact, I still have quite an LP collection). But even CD's are becoming harder to come by these days and I have found that with the right sampling and codec (256KB/vbr AAC), my lossy compressed iPod music is audibly no different than the original CDs ...at least as far as my critical ears can tell. Which brings me to my question:
What is the sampling and codec used for downloads from the iTunes Store? I have read in older posts ('05/'06) it is 128KB/AAC but then I recently read iTunes downloads will be non-DRM encoded and come in at 256KB/AAC.
Appreciate any insght into this as I am really enjoying the "Record Store" on my desktop.
Thanks,
Stegasaurus

All itunes music has been 256/ non drm for years now.

Converting Units in LabView and specifying sampling rate with Universal Library functions

Hi,
I am having trouble converting units in LabView 7.0 and having it write to a file / output on a chart in Nm instead of in volts. I can't seem to find any straightforward instructions, eventhough it seems like a simple task.
Another task that would seem rather simple is changing the sampling rate with the XAIn program from the Universal Library for LabView (this is version 5.40).
If anyone could help me out, I would greatly appreciate it - I have tried other sources without much luck!
Thanks
Jenna

Are you really needing to change units or do you need to scale the voltages read from your sensors? If it's scaling, it would be trivial to set this up in mAX or with the DAQ Assistant if you were using an NI board. Since you are not, then you could use the Scaling and Mapping Express VI. The y axis of a chart is just a label. You can use the text tool to change it to anything you want.
I have no idea how to change sample rate for your board. Have you asked the vendor?

AAC sample rates for N95

Similar Messages

Maybe you are looking for