Sample Rate Madness

I have a Project shoot with a Canon 7D 1080p 23,98. The sound was recorded with a MiniDisc at 44.1kHz. I converted the clips to 48 kHz but FinalCut still shows a greenbar in the timeline and the sound is going out of sync. I tried to convert it with soundtrack, compressor and even Logic 8. None of this works. Do I miss the point somewhere?
Thanks to everybody!
P.S.
Im running FCP Studio 3 on an iMac 20" Mac OSX 10.6.3

I always use Quicktime for the conversion, using these settings:
Linear PCM, 48 kHz, 16 bit, Stereo (or 2 discrete channels, depends on what you need), BEST settings (slider). NOT Big Endian

Similar Messages

Sample Rate Madness : What Did I do This Time?

I typically record at 24/48, but last week I had to mix an old project that was originally recorded at 24/44.1. I had some problems switching to and from 44.1 and 48k, probably due to my M-Audio Lightbridge, which seems a little buggy at time. It's clocked over lightpipe by the master, hd24XR, which is also set to match.
Problem:
I changed the autload to 44.1 while mixing the project, it made the devices more stable not having to change back and forth. I forgot to change the autoload back, and recorded a new project today. Both the XR and Lightbridge were set properly to 48k.
I didn't discover my error until the session was 2/3 done, but I had gone back and forth about whether the audio didn't sound quite the same on playback. It seemed a little clearer when I was tracking, as I was probably listening to D/A at 48k. When I checked the files, they were at 44.1. Its nice that I noticed the difference, but I messed up. oops...
Question:
Where did the conversion from 48 to 44.1 take place? The XR said 48, and the Lightbridge said 48 on both the hardware and software control panel. Did Logic automatically down-convert? Or, please tell me no, the M-Audio Lightbridge?
Thanks guys!

Hi Ryan,
Do you think it would sound any different if I changed the headers?
Changing the header will not alter the samples/audio data of the files. What you do is change it to the actual sample rate (WC) used when the recording took place, when it presently states the old session sample rate - you then change your session sample rate.
Whats the quickest way to change them all?
As always make backups. I don't know what you have for software (beyond Logic, WaveBurner, iTunes...) there is a free utility called soundhack which should take care of this in minutes - ( I don't know how many you have ).
So, you change the SR for all files then reopen the session.
J

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?

How can I find out the audio sampling rate of BetacamSP tape?

Hi guys
I'm trying to digitize BetacamSP tape. But I'm afraid if I might choose wrong setting...
This tape is from very long time ago so we don't know which audio sampling rate we recorded with..
How can I find out the audio sampling rate of this BetacamSP tape?
Thanks:)

The sampling rate is set by the Sony DVMCDA2 you are using, when the conversion is made from the analog input to the digital (DV) output. You should be outputting standard DV which is 16bit 48khz audio.
Assuming you are in the US, your Easy Setup for FCP should be DV-NTSC, and then open the Log and Capture Pane and set the Capture Settings Device Control to Non-Controlable Device and you should be good to go.
You will have to roll the deck manually and start and end your capture manually.
You can download a user manual for the DVMCDA2 by clicking here.
MtD

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.

Having trouble with wav files and sample rates

Hi ,I am having trouble with wav files and sample rates .I have been sent multiple projects on wav as the main instrumental ; I wish to record in 48.000kHz .Now comes the problem.When I try to change the project to 48k It seems to pitch up the track.I can't have them send the logic/project file as most have outboard synths,different plug ins etc.This particular case the producer has recorded the synth task in 41.000 kHz .My successful outcome would be to be able t create a project file in 48 kHz .And NOT pitch up whne I add the instrumenta wav file .Any help would be gratefully recieved,this is my first post so any mistakes I may have made go easy

You'll have to convert the actual synth audio file file that the producer gave you to 48kHz. You can do this in the audio Bin in Logic.

Sample rate of cio-das08jr

hello;
after an internet search. and reading the help, I still post on this forum, hoping that I find an answer;
At first, I realized the aquisiion using cio-das08 Jr. with the help of someone on this forum (cj), I thought I solved the problem by that when I want to change the sample rate I can not find my device inTime base, and the acquisition is made with a sample rate that Ido not know, but when I check hardware Clocked; of hardwaresetup, I find my device in time bases, but when execution, I found a message that says invalid option
my question is how can I change the sample rate of my devicecio-das08jr
Attachments:
hardware Setup.GIF ‏42 KB

hello
I have not found the solution to control the sample rate of cio-das08jr, please help me it urgent
Attachments:
sample rate.GIF ‏40 KB

8kHz sample rate option broken for stream sounds in CS5

I've made several test files where all I do is add a sound to a keyframe, set it to Stream sound, add a sufficient number of frames to hear the sound (about 300), then change publish settings to override as MP3 with "use 8kHz sample rate" option.
Testing movie yields a SWF that has NO SOUND at all.
Changing the sound to Event sound and republishing PLAYS THE SOUND AS EXPECTED.
Changing it back to Stream sound and turning off the "use 8kHz sample rate" option again yields a SWF that has NO SOUND at all.
It seems to make no difference what sound I import and use.
I'm on a Mac Octocore.
This Publish Setting appears to be broken in Flash CS5.

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 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.

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.

Why Audition CC don't show sample rate correctly?

Hi
Using Audition CC on Mac OS X with Mackie 1220, 12-channel premium analog mixer with integrated 24-bit/96kHz Onyx FireWire I/O (Quality is amazing!)
Need to record on mono a horrible old tape recording with voice only to "hero"'s repairing audio and configure the best sample rate for recording.
My preferences Audio Hardware on Audition are:
Device Class: CoreAudio
Default Input: Onyx Firewire (0025)
Default Output: Onyx Firewire (0025)
Clock Source: Device
I/O Buffer size 512 Samples
Sample Rate: 96000 Hz
Question 1: Expert said do not use more than 48000 Hz for recording voice only and use mono. Humans can't listen more than 20000 Hz... well... My recording on 96000 Hz is better than 48000 Hz and I'm not Superman... and can hear better difference with 96000 Hz recording. Don't know why.
Question 2: Why after selecting Audio Hardware preferences to 96000 Hz after recording the File Panel said my track is 48000 Hz Mono 32 (Float)? My hardware can record at 96000 Hz and preferences is set correcty at that Bit Depth then WHY don't appears 96000Hz on File Panel after recording and instead appears 48000Hz?
Question 3: Don't understand the differences of my 24-bit/96kHz Onyx FireWire I/O and 32 bit (Float) of Audition. If I record using 24 Bits (Maximum bit depth of my hardware) why appears 32 bit (Float)? My recording are 24 Bits or 32 Bits?
Thanks :-)
Tom

CS6 has a serious issue with saving files correctly. The program is asuming that 48kHz is the maximum you will be using and in my case it saved a 96kHz recording with a 48kHz internal header. The file size is consistent with all my previous 24/96 recordings and it sounds just fine interpreted correctly - but played an octave low in frequency and tempo it really sucks unless you are a Blue Whale..
I can play it "interpreted" as a 192kHz file just fine, and it now sounds 100% right, but I cannot save it correctly. I have yet to find out how to recover because when I use "convert Sample Type" it saves it with the same mistake - the wrong sample rate off by the same ratio again.
It is a program flaw - so at this point you cannot record with sample rates higher than 48kHz in CS6 and depend on your file being OK.
Tom the reason you can tell the difference between sample rates is that your hearing has two dimensions, frequency and timing, I sure hope you can't hear the bats at night but I expect that you can tell a good drummer from a bad drummer. In addition there is also the issue (dimension) of bit-depth - instrument decay and acoustic space occupies the time space between notes and if it is not sampled at the right time and place or rounded off to the nearest digital digit you have a problem.
You all know that some humans have perfect pitch and others dont, this gives you some indication how much we each differ. Some people have even learned to use echolocation; the best know cases being blind people because they are not supposed to be able to find their way and know where they are. You can learn underlying concepts of this little discussed aspect of human hearing here and hit the university libraries for the rest. http://en.wikipedia.org/wiki/Human_echolocationhttp://
SteveG's comment are only true in one of the three dimensions - frequency range. We all hear about the same for starters anyway.
The second dimension, timing, is what provides spatial information. Coarse sampling effects this as well. My most recent experience was when transcribing some old casette tapes - when I experimented with FLAC and WMA lossless I found that they din NOT downsample well. Spatial information was significantly deminished be that aucoustic or studio work. Needless to say this surprised me because I was auming that I could drop the samplerate to 48kHz as soon I was done editing and save a lot of drive space. For now I made some excellent 48kHz/24 bit mp3s (320) because they altered the sound the least.
Now you know some additional reasons why older guys like me who have lost the high end and a lot of decibel as well can still tell the difference:we have good sense of timing.
Anyway - I need to learn how to edit my files to reset the sample frequency header - real fast. I just recorder a fabulous Madrigal group for 2 hours and my files are lost in the vortex.

Changing audio recording depth and sample rate in QuickTime Pro?

Last week, I set up QuickTime Pro 7.4.5 to record audio in 24-bit, 48 KHz format -- only now, I can't remember how I did that. I think that I may have used some other software to change the settings of my Mac Pro's built-in sound card. Does anyone know how to do this?
I made several such recordings, just last week, as a test, simply using a microphone plugged directly into my Mac Pro, and using QuickTime Pro as the recording application. During playback of such QuickTime audio files, I can hit <command-i> to see that the recordings are indeed 24-bit, 48 KHz. But now, when I repeat the same process, my QuickTime audio recordings are made with 16-bit resolution, at 44.1 KHz. I remember putting the settings back to those values after my recording session of last week. But I can't remember how I adjusted those settings. Does anyone know how to do that?
Thanks.

Thanks for your response.
I will use Apple MIDI Setup, as you suggest.
I want to change the sample rate to 48 KHz, as the final destination is DVD and I don't want to convert from 44.1 to 48 and thus suffer a possible loss in quality.
As for bit depth, I want to sample at 24-bit, since I'll be processing the sound afterwards and don't want audible rounding errors during the number "crunching". Afterwards, I'll just drop the eight least significant bits, or perhaps round them up or down (to zero or 256).

Sample Rates not Matching

I'm using Firewire from my Sony HVR M15U to FCP 6. After ever capture I'm getting a message "that 1 or more of the media files does not match the sample rate of your source tape". It also warns that audio may be out of sync with video. However, that does not seem to be the case. I have check the sample rates and they seem to match. Any thoughts?

I saw this warning several times at a station I was freelancing at several months ago. Never could figure out the difference it was seeing. Never made a difference with sync or anything else.
I might have noticed a longer render time with those jobs that had the warning, but that is just something I thought I noticed and I never had time to set up a test of that.
All jobs output to tape just fine in the end. Happy me, happy client.
They were using a Blackmagic card for input if that could be a control variable. Sorry, I don't recall the model number.

Audible glitching - audio sample rate mismatch?

I'm capturing some old DV tapes for a friend who no longer has a DV camera to capture from. I've captured into FCE4 and the video looks fine, but all the audio sounds like a sample rate mismatch, that static-y glitch sound that I've heard before in audio programs when there's a sample rate problem. The audio on the captured clip is 16-bit, 48KHz, big endian.
I'm using a sequence with the DV-NTSC preset with 48KHz audio. I made sure to set it up in Easy Setup beforehand so it would be the same as the sequence. The camcorder I'm using for capture is my old Sony TRV-70 (have also tried my TRV-50). I can't get either camera to tell me what sample rate the tape was shot in; all i can get it to tell me is 16 bit and SP mode. I'm assuming this will be 48KHz?
When i try capturing with the DV 32KHz easy setup (and sequence) it gives me the error about the sequence rate not matching what's on the tape, and it sounds worse.
I don't know what camera originally shot the tapes. But the audio sounds perfectly normal coming out of the speaker in my sony camcorder.
Am I missing something?

16 bit and SP mode. I'm assuming this will be 48KHz?
That's correct.
Make sure the Sequence you select is a 48 kHz choice abd double check that you do not have a PAL/NTSC mix up.
Al

The dreaded "sample rates do not match" error: imac

Greetings,
I am getting the "samples rates do not match" error when I try to record in an audio file. I see a lot of stuff online about fixing this on windows, but haven't found the solution for my late 2012 iMac.
I've made the audio file stereo, 24 bit, 48kHz.
I've checked the audio midi setup and both the mic and the speakers are set to 2ch 24 bit 48kHz.
Still get the error....
What am I missing?
Thanks!

What audio interface are you using? You need to make that that is set to 48k as well.

Sample Rate Madness

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