Setting default sample rate to 48KHz???

Can someone tell me how to set my Logic Pro to default to 48 KHz instead of 44.1 KHz. Also can you tell me how to convert the sample rate of audio files within Logic (the command appears moved since Logic 7). Thanks. Dave

to convert the sample rate of audio files you have to use the Copy/Convert command in the Audio Bins' local menu, under *Audio File*.

Similar Messages

Changing default sample rate breaks Pulseaudio

When I change default sample rate from s16le to s24le which my card supports (and it always worked under Ubuntu) Pulseaudio goes bananas and doesn't play any sound in Firefox and Spotify, but works in Audacious or for system sounds (like speaker test from Gnome settings).
Output of pulseaudio -vvvv in default settings:
http://pastebin.com/zJiAi0T2
Output with default sample rate set to s24le:
http://pastebin.com/SKWjcg19
Last edited by gutigen (2014-11-15 22:32:45)

Thanks. That will also be nice in that I won't have to re-do all of my saved screensets every time.
Is there an easy way to open templates? I ask because Logic requires them to be saved to the "Project Template" folder, whereas I have all of my music stuff saved on an external firewire drive. Also, templates don't appear to show up under the "open recent" menu so I'm wondering if I'm going to have to go into that folder every time I need to use that template.

Setting the sampling rate in SignalExpress

I am using a cDAQ-9172 with a strain gauge module and a thermocouple module and using SignalExpress. I want to acquire data at a pretty low frequency rate (2Hz), but I am unable to use 1 sample on demand in the acquisition setup. I get the following error.
Error -201087 occurred at DAQ Assistant
Possible Reason(s):
Measurements: Task contains physical channels on one or more devices that require you to specify the Sample Clock Rate.
Specify a Sample Clock Rate.
Device: cDAQ1
I am unable to setup a sample clock rate as it is 'blacked' out by the software in the 1 sample on demand mode.
When I try to use continuous or 'N' sample mode, the cDaq is sampling at a rate of around 1600 Hz, even though I am putting in a value of 1Hz on the seti[ screen. The large amount of data will prevent me from downloading the data to Excel for further reporting. I obviouisly don't need to sample my TC module at that rate either.
Is there anything I can 'easily' do to decrease the sampling rate down to a low level? Everything I have tried doesn't seem to work and I don't really want to go away from SignalExpress.
Jay

Hi Jay
I am assuming that you are using the NI-9237. This strain module has only a specific set of sampling rates. If you specify a rate that is not supported it will corers the rate to the next higher sampling rate. Also since you are using the cDAQ chassis, there is only sample clock so it must choose the highest sampling rate. So it is not possible to sample at a lower rate. Please see the following link.
If you have access to LabVIEW then I would suggest taking your readings at the fast rate and then average the samples before you write them to your file. The following link has some more information on this.
If you are limited to Signal Express, this task is tricky. If you select a N-Sample acquisition, set the samples to read to greater than two, and set the post acquisition delay under the execution control tab to 1000, you acquisition will acquire the two samples at the very fast rate, but it will wait 1000mS between each acquisition. This will slow the overall sample rate.
Chris_K
National Instruments
Applications Engineer

[Q] How to set the sampling rate separately?

Hello,
I'm using LV5.1 in Windows98 with SCXI-1200.
I want to set different sampling rate in each input channel.
I've ever used "AI acquire waveforms.vi" only.
Anybody can help me?
Example codes are highly appreciated.
Regards,
Hyun-ho Lee
[email protected]

The inputs of the SCXI-1200 are multiplexed to a single Analog-to-Digital converter. Because of this, the only difference in sampling rates that are achievable would be integer divisions of a common high frequency. This is functionally identical to acquiring all channels at the highest rate, and decimating (throwing away) data from channels that need lower data rates.

FMLE won't let me set the sample rate with Blackmagic Decklink Studio 2 (Windows)

Hi,
I am trying to set the audio device on FMLE to the Decklink Studio 2's audio inputs. However,
the sample rate sticks to 44.1k and I can't change it to the 48kHz sample rate that
the Decklink defaults to (standard for broadcast is 48k).
Try as I may, it isn't changeable. I seem to remember a similar problem where the
sample rate defaults to the system audio card (on the motherboard) which is at
44.1.
Any ideas on a work-around? The audio from the Decklink is *much* better than
the built in card.....
thanks
jeff

One thing you might try doing is trashing the preference file in your user directory:
~/Library/Preferences/com.apple.compressor.Compressor.plist
Like repairing permissions, this is a bit of a "catch-all" fix, not something specific to this particular problem.
Also, have you tried different file types like: TIFF or JPEG, there could be an issue with output from Fireworks. You could use QT Pro to transcode the file. Sometimes that can work to fix funky problems.
Does the Text Overlay filter work?
BG

Changing Default Sample Rate Settings?

Hello,
When I open up a new song in Logic it always defaults to 44.1. Sometimes I forget to switch it to 96 before recording only to discover later that I am going to have to convert the files or re-record. Is there a way to set a "default" so the sample rate is always at 96 when I start a new song? Thanks.

Thanks. That will also be nice in that I won't have to re-do all of my saved screensets every time.
Is there an easy way to open templates? I ask because Logic requires them to be saved to the "Project Template" folder, whereas I have all of my music stuff saved on an external firewire drive. Also, templates don't appear to show up under the "open recent" menu so I'm wondering if I'm going to have to go into that folder every time I need to use that template.

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?

How to set accurate sampling rate in FPGA programming.

Hi all,
I'm using cRIO 9004 and NI 9233 for acquiring high speed analog inputs. Module 9233 specify some data rate values which can be programmed. I want to acquire data without missing a single sample. please comment on this.

Couple things to add to what Mike said:
For C Series modules like the NI 9233 when used on the CompactRIO platform, all configuration and programming is done from within LabVIEW.
To configure the data rate at which the 9233 will acquire data, you need to either get into the module's properties page (in LabVIEW) and set it there, or you programmatically do it by using a LV FPGA property node on the module. I suggest you look at the examples that ship with CompactRIO for the 9233.
JMota

How does Core Audio set sample rate?

When I play a particular movie in QuickTime, the audio and video are out of sync. The movie plays fine on other computers.
This Mac Pro (OS 10.6.8) is used exclusively for Pro Tools and Final Cut Pro. The audio hardware is a Pro Tools HD Native card:
Native card --> Pro Tools Digital I/O boxes --> Lavry D/A converters --> monitors
The sample rate of the Digital I/O boxes is set by an external master clock, a Lavry Gold A/D converter.
I opened the DigiDesign Core Audio Manager and it says:
Connected @ 44.1K, 32 In/32 Out, Buffer Size 512
Yet the movie is at 48K (I know because I created it in Final Cut Pro), and the external clock is set at 48K. So, I don't know where the 41K in the Digidesign Core Audio Manager came from. This is why I am guessing the problem is sample rate.
Please help me understand what sets the sample rate.
Does the application using Core Audio set it?
If so, how is this made consistent with the sample rate set for the hardware by the external master clock?
Do I have to be sure I always change the external clock setting to be consistent with the movie being played?
BTW, I have not had a sync problem when I play video in Final Cut Pro..........it is always in sync. So, I have never worried about how Core Audio works. The sync problem is only with QuickTime.

Well, here is more information:
- I reinstalled QuickTime 7 and did many other things (trashed preferences,etc)
- I set the audio hardware (Pro Tools/HD Native card) to 48k
- I checked the Digi CoreAudio Manager and it said "Connected at 44.1K", even though the hardware was set to 48K
- I played the problem videos, and all are now in sync
- I check the Digi CoreAudio Manager and it still says 44.1K
- I checked the hardware and it is now set to 44.1K
I know that the videos are 48K, because I created them in Final Cut Pro X.
Does this mean that QuickTime 7 always converts videos to 44.1K, regardless of their sample rate?

Set sample rate to 96 kHz in iMac

If I go to Applications>Utilities>Audio Midi Support, I can set the Sample Rate to 96 kHz, but in fact the Sample Rate is back at 48 kHz next time I look.
Question: How can I get the machine to accept and stay at the new sample rate of 96 kHz?
Robert

Hi Robert, try this...
Safe Boot , (holding Shift key down at bootup), use Disk Utility from there to Repair Permissions, test if things work OK in Safe Mode.
Then move these files to the Desktop for now...
/Users/YourUserName/Library/Preferences/com.apple.finder.plist
/Users/YourUserName/Library/Preferences/com.apple.systempreferences.plist
/Users/YourUserName/Library/Preferences/com.apple.desktop.plist
/Users/YourUserName/Library/Preferences/com.apple.recentitems.plist
/Users/YourUserName/Library/Preferences/com.apple.audio.AudioMIDISetup.plist
Reboot & test.
PS. Safe boot may stay on the gray radian for a long time, let it go, it's trying to repair the Hard Drive.

Can I set an arbritrary sampling rate in Biobench (need 60/sec to sync to video)?

Is there a way to set a sampling rate that is not in the drop-down menu in Biobench? Thanks!

Hi George,
BioBench comes with a set of predefined sampling rates. I encourage you to try using LabVIEW to trigger/sync your DAQ with video. The sample freq. in BB are std. for low freq. biomedical signals.
Please let me know if you have any further questions.
Regards,
Morten Jensen
National Instruments

How is Core Audio sample rate set?

When I play a particular movie in QuickTime, the audio and video are out of sync. The movie plays fine on other computers.
This Mac Pro (OS 10.6.8) is used exclusively for Pro Tools and Final Cut Pro. The audio hardware is a Pro Tools HD Native card:
Native card --> Pro Tools Digital I/O boxes --> Lavry D/A converters --> monitors
The sample rate of the Digital I/O boxes is set by an external master clock, a Lavry Gold A/D converter.
I opened the DigiDesign Core Audio Manager and it says:
Connected @ 44.1K, 32 In/32 Out, Buffer Size 512
Yet the movie is at 48K (I know because I created it in Final Cut Pro), and the external clock is set at 48K. So, I don't know where the 41K in the Digidesign Core Audio Manager came from. This is why I am guessing the problem is sample rate.
Please help me understand what sets the sample rate.
Does the application using Core Audio set it?
If so, how is this made consistent with the sample rate set for the hardware by the external master clock?
Do I have to be sure I always change the external clock setting to be consistent with the movie being played?
BTW, I have not had a sync problem when I play video in Final Cut Pro..........it is always in sync. So, I have never worried about how Core Audio works. The sync problem is only with QuickTime.

You might ask on the GarageBand, Logic or Final Cut forums. There is not much audio traffic here.

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.

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

Setting default sample rate to 48KHz???

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