Sample rate problem
I have an odd sample rate problem: I have put in some music at 44.1 (while all other audio is at 48kHz.)
At most points this is, surprisingly, not an issue. But if I haved 4 channels of 48kHz sound then EITHER the 44.1 sound OR the 48kHz sound will not play.
Anyone know why this would be? I obviously need to redo the 44.1kHz sound - is there a quick and easy way to do this, or do I need to reimport everything seperately?
Export the 44.1khz audio as an AIF, 48khz, 16 bit file themn import the resulting file. Normally you'd want to do this with QT Pro before importing it into FCP.
-DH
Similar Messages
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Sample Rate problem as well as CoreAudio overload and anyone using DFH?
I've only switched from Pro Tools to Logic Pro recently and am confused about a lot of things. I am running Toontrack DFH Superior with 7 audio tracks and one more software instrument track. I keep getting message about sample rate problem and something about conflict betwwen logic and external device. The only device I use is a M-Audio 49e MIDI controller and the M-Box, and I am running these on a Quad PowerMac G5 wih 4 GB RAM... I've tried to adjust the setting here and there, but none of them seem to work... Please help!
Often MIDI driver conflicts and processor overloads, or bottlenecks too. Freezing, removing old drivers, if you really are running Jaguar and if it is actually installed from those days then it is time to do a fresh install and maybe use Panther or Tiger. Panther can be found really cheap these days. As far as old drivers you don't use for any audio or MIDI interfaces you once used with your sytem, are there any? If so which ones? You may be able to direct yourself to your old driver uninstall by looking at the receipt of whichever driver you no longer use. Also, Jaguar was just really good at getting this message to come out, in my experience. Some plugins require alot of MIDI I/O or CPU power, some of NI's stuff really (momentarily) spikes on MIDI data input.
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Sample Rate Problems going from Logic to Garage Band
I've downloaded the Trial Version of Logic Express. When it launched it changes the i/o sample Rate that is required by Garage Band (I'm using a digi002 i/o). The result is that garage band plays back at a fraction of it normal rate (sounds like crap).
It wasn't a problem until the trial ran out on Logic Express. I could change the sample rate back inside Logic. Hummm ... what to do? I don't see any place to change the rate in the garage band preferences or in the core audio interface. Any suggestions would be appreciated.
G4 1.25 dual, mirror Mac OS X (10.4.6)
G4 1.25 dual, mirror Mac OS X (10.3.9)updated
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Sample Rate Problems going from Garage Band to Logic Express
I've downloaded the Trial Version of Logic Express. When it launched it changes the i/o sample Rate that is required by Garage Band (I'm using a digi002 i/o). The result is that garage band plays back at a fraction of it normal rate (sounds like crap).
It wasn't a problem until the trial ran out on Logic Express. I could change the sample rate back inside Logic. Hummm ... what to do? I don't see any place to change the rate in the garage band preferences or in the core audio interface. Any suggestions would be appreciated.
G4 1.25 dual, mirror Mac OS X (10.4.6)
G4 1.25 dual, mirror Mac OS X (10.3.9)
G4 1.25 dual, mirror Mac OS X (10.4.6)I've never had any such problem with Logic Express. GB files just load up, and work fine. Did you change anything in Logic Express? There are setting for that there, but even songs I open after I changed LE to 24 bit, work fine.
Wait, are you saying LE is changing the sample rate, and then it stays that way when you open GB? Hmmm, What do you have LE set for? I think mine is set at 24/96, but I'd have to look to be sure. But I am not using the same interface as you are. It might be something you just have to live with. Have you tried posting this in the LE7 forum? Maybe one the gurus there knows what you can do.
Oh, and I am using GB 2, not GB 1, but I am not sure that is relevant. -
DV 16:9 but Only Exports 4:3 WHY? Also Audio Sample Rate Problem
I'm quite new to Final Cut and have FC 6.01. I use PAL 25fps and a 16:9 SD DV Camcorder with FireWire SONY DV VTR Deck. I have two problems:
1. When i capture my 16:9 DV footage the Logging Window shows only 4:3 with a distorted image in it. Though I have chosen the easy set up and told Final Cut that im editing Anamorphic 16:9 I also found that FC will only export 4:3. However, during the editing process i see 16:9 in the Browser Preview and Canvas Window.
2.(Not sure if its related) I get the following message everytime after I capture individual clips or if i press the Escape key during a capture:
"The audio sample rate of one or more of your captured media files does not match the sample rate on your source tape. This may cause the video and audio of these media files to be out of sync. Make sure the audio sample rate of your captured preset matches the sample rate of your tape"
Does any one else have this problem?
Apple, as yet have not given me any answers.Danny Boy.. Thanks for your reply and I'll be happy if it is my fault and not FCP's. Actually i made an error in my post. It does indeed display correctly in DVDSP, it's iDVD it does not even when told to display it in 16:9. However, when i used the PAL DVD Anamorphic file preset, iDVD still couldn't display it. To get it working I had to tell Compressor specifically (in the additional settings) to encode 16:9 despite what the presets stated! No matter how one looks at this, this is confusing to say the least! If a preset says 16:9 then one should expect 16:9! Remember, Im using the display window of iDVD to show me the output.
To detail my steps as hanumang has said, im doing the following:
1. in FCP I encode to QuickTime Move
2. Open iDVD & Create a Project
3. 'Drag' The QT file into the menu
4. Using iDVD Preview function, Preview the QT file
5. ITS STILL 4:3
now, the above was done with QT Conversion which as also set to encode 16:9 and still had the same result.
Thnaks to you both. -
Audigy 2 platinum sample rate problem
People i need help with my audigy... Its connect to my Ht sony ddw880 via coax cable... In audio console, when selected spdif digital out (PCM) sample rate for 48 khz... everything goes fine..But when i select sample rate for 96khz, my receiver detects the signal as 96 pcm but theres no LFE signal... The sound plays normal but without subwoofer... Ive contacted sony and the tell me thats a problem with the signal that the pc is outputting in 96khz mode... cause in 48 khz its ok... I need help with this problem... please anyone.....!!!Thanks a lot.. PS: Sorry about my english (Brazilian)
People i need help with my audigy... Its connect to my Ht sony ddw880 via coax cable... In audio console, when selected spdif digital out (PCM) sample rate for 48 khz... everything goes fine..But when i select sample rate for 96khz, my receiver detects the signal as 96 pcm but theres no LFE signal... The sound plays normal but without subwoofer... Ive contacted sony and the tell me thats a problem with the signal that the pc is outputting in 96khz mode... cause in 48 khz its ok... I need help with this problem... please anyone.....!!!Thanks a lot.. PS: Sorry about my english (Brazilian)
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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 problems while re-opening .PCM file I recorded
I recorded an audio using adobe audition and saved as a .PCM format but am having a problem with with file, it keeps giving me a heavy noise. Please any help?
What is the sample rate and bit depth of the file?
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Non-Uniformed Sampling Rate Problem
Hi All,
I have 2 analog inputs being sampled at 500 Hz. But after looking at the plot of my samples, it doesn't seemed like the intervals are of the same size? Can any one tell me what might cause that and how I can make them all the same?
Thanks much.
hellolv
Attachments:
FILE_4.vi 51 KBhellolv,
I have taken a look at your program, and I have made a few modifications.
I have found a couple problems with the code.
First, you are performing your DAQmx read in two positions inside your
while loop. This means that for every loop iteration you read in two
samples (not one). This makes the logic you use to stop the loop
incorrect because it assumes that one sample per loop iteration is read.
I have modified the VI so that only 1 DAQmx read is used. I also changed
how many samples you pass to the DAQmx timing. In this application it only
takes 120 seconds to acquire 60 000 samples (at 500 Hz).
Next, the interval between samples should be constant. The DAQmx device
will acquire data at the specified sampling rate and store the data in the PC
buffer. The graph will display data based upon waveform data type.
This data type only has a t0 (start time), dt (period between each sample), and
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Regards,
Jesse O.
Applications Engineering
National Instruments
Jesse O. | National Instruments R&D
Attachments:
FILE_4_new.vi 49 KB -
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. -
I recorded my band at a studio at 48k in protools, then I took the ausio files and loaded them into logic pro, it worked great, I mixed a few songs, spend a huge amount of time mixing and always thought to myself I cant believe we played the songs so slow. at any rate I realized now that I brought all the 48k ausio files into Logic and logic was set at 44k, so it converted them. I tried changing the sample rate in the perferences to 48k and now everything does not line up like it should etc...
Is there anyway to now change my mixes to 48k without losing any infomation including automation?
ThanksTry making a copy of the song, Erasing your bad tracks and removing them from the audio window. Then reimport at the correct rate. I think you will have to shift the automation.
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i! I have two audio PC's with 1212M's. One PCI-express, and one PCI, recently bought.
I have some hi-res audio files that I'd like to play at 96 khz, but the PCI version gives no sound if I make a new session at a higher sample rate.
Any ideas?
THanks, DaanTry to fresh the driver or Profile session. How do you feel about the PCI-Express and Audio Response compared to PCI?
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Hi,
I am using the CANOpen toolkit to communicate with a sensor which can be set to a sample rate of 1kHz. Before I've used the standard Frame API and there I've received my PDO-objects every ms. Because I want to make the programming more easy I would like to use the CANOpen. But I can't set the sample rate to 1ms. When using 2ms the data isn't transmitted with the correct timestamp intervall.
I've attached the errorcode message when using 1ms as sample rate and the part of the block diagram with the PDO initialization.
Maybe somebody can give me a hint.
Best Regards,
Joachim
Solved!
Go to Solution.
Attachments:
canopen_block-diagram.JPG 26 KB
canopen_error_1ms.JPG 16 KBHello DirkW,
thanks for your reply. I've attached 2 example VIs. One with the communication via Frame API (Interface NET) and one via CanOPEN (Frame API (Interface Object)). When running the example with Interface NET I can poll my data every ms. I only have to start the communication via ncWriteNET.vi. After that I am receiving my PDO objects. That means my hardware (slave) is able to deliver data every ms. I've seen that the CanOpen library uses the method 'coPdoCreate' from the ni_cano.* - library. How can I adapt this CanOpen object to communicate via the Interface NET method. I need for starting the PDO-transfer the PDO-object reference for the PDO Create Object.
My aim is to communicate with the maximum sample rate via CanOpen.
Joachim
Attachments:
MAIN_MP55_RECEIVE_CANFrame.vi 52 KB
MAIN_MP55_RECEIVE_CANOpen.vi 49 KB -
Hi,
I have a quicktime movie I'm trying to edit in FCP but the audio is 22050Hz and I don't see that option in my sequence settings. Now I'm getting little audio pops & I can't figure out how to get rid of them.
Thanks,
GabriealI'n not understanding what you mean by originate, I guess. I downloaded it from youtube to a mov on my computer.
I wanted to edit the clip in FCP and when I check 'item properties' under 'format' it says:
Type: Clip
Creator: Quick Time Player
Source: (Path on my computer)
Offline: (nothing here)
Size: 21.3 MB
Last Modified: (date)
Tracks: 1V, 1A
Vid Rate: 30fps
Frame size: 320 x 240
Compressor: MPEG-4 Video
Data Rate: 37.5 k/sec
Pixel Aspect Ratio: Square
Anamorphic: (nothing here)
Field Dominance: None
Alpha: None/Ignore
Reverse Alpha: (nothing here)
Composite: Normal
Audio: 1 Mono
Aud Rate: 22050Hz
Aud Format: 32-bit Floating Point
Angle: (nothing here) -
Audio Sample Rate Problem:
I am getting an error message when digitizing footage
I'm having the audio out of synch error as well, and was ignoring it, but after exporting one project, found it was terribly out of synch. So I've gone back and figured out how to inspect settings such as you all are suggesting, and have found out that yes,in the browser, my SEQUENCE says its audio format is 48 khz 32-bit floating point, whereas all my CLIPS are 16-bit. However, when I go to look at the audio presets under the menu FCP audio/vdeio settings it IS set to 48.0 Khz, 16 bit - all the capture settings, the sequence setting ...everything matches. From the menu, Sequence, the properties says it is the same. Another odd thing is that - as I mentioned in another post - the easy setup that my husband created - the one that works best for capturing - does not come up in audio/video settings so I can even look at it anymore. While I'm here, I might as well add that last week I was surprised to find that I COULD use device control, but no matter what I try now, no device control...he giveth and he taketh away, apparently. I was liking it....
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