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.

Similar Messages

Core Audio - Sample rate

I've just got myself a copy of Logic Pro 8 as a complete newbie and have hit a bit of a hurdle within 24 hours of opening the software.
I was using the software without a problem just tinkering around using a Carillion MIDI interface to play some software instruments. I don't have a mic yet so thought I'd try my bluetooth headset just so I could play with recorded vocals. When I started speaking while Logoc was recording it came up with an error message
"Core Audio
Sample Rate 8000 not allowed."
I then turned off my bluetooth headset and turned bluetooth off on my Mac so I could go back to using the MIDI keyboard but got no sound. I closed Logic and re-opened, selected "Empty Project" withing the "Explore" tab and was again presented with a series of the messages, all reading as per the above error messages but I get 3 with "8000" and one with "73536".
When I'm playing the keyboard I can see the note I'm playing or the chord in the dialog box at the bottom of the Logic interface screen but again, no sound.
Any help would be hugely appreciated. Again, I am a COMPLETE Logic novice so please be gentle with me.
Regards,
Mark

Hi Jounik,
I went into Audio MIDI Setup and the headset didn’t appear. The keyboard appears as “Midilink” but I couldn’t see where to check the sample rate. I did however use the Test Setup facility and the keyboard was only producing a sound when the key was released.
I also had a look at: Logic Pro > Preferences > Audio (Devices) but wasn’t really sure what to look for. This is what I was confronted with http://adoseof.co.uk/resources/Picture1.png
The project sample rate was set to 44.100 KHz
I like the idea of using the internal mic. I’d tried that before but was getting feedback. Plugging the headphones in seems like a good solution.
To try and get the keyboard working again I opened a new Empty Project and left the dialog with the default settings as indicated here: http://adoseof.co.uk/resources/Picture2.png
I’m not sure what you mean by “In the I/O slot of the channel strip open a synth, e.g. ES1”
I then used the on-screen keyboard and still no sound was coming out.
Any other ideas?
I tried the keyboard in garage band and it worked fine so I can only assume it must be something to do with Logic.
Mark
Message was edited by: hotsawz

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 do you set sequence audio sample rate?

I tried posting this to another, but it was already answered, and noone will see it.
I am getting the capture error "audio sample rate doesn't match" and yes, I can see in my browser that the clip is 48khz/16bit, but the sequence is 48khz/32 bit. Howver, wherever I look to change the sequence setting, it is 48/16 already. I've gone to FCP on the menu dropdown to audio/video settings - it's correct all through there. I've gone to the menu dropdown Sequence settings, and it's correct there. I've closed down, opened a new sequence, restarted, everything I can think of. Is there a secret to getting them to match? And, can I fix a project already edited with this discrepancy? Its export to QT is WAY out of synch.

Annoying - I can't see your post when I am in reply mode.
Yes, I get this error when i am capturing. From reading a previous post about the error, I thought the solution was to check the audio rate of the captured clip in the browser, and then make sure it matched the audio rate of the sequence. Like I said, everywhere you get to change the sequence setting, it SAYS it is 48/16, but yet, whe I scroll over in the browser, it says the audio rate is 48 KHz and the audio format is 32-bit floating point. Am I looking at the right places?
I can't check the settings in the camera until this evening...don't ask.
I'm not so sure this is not a QT issue instead of anything to do with capturing, etc. It plays back fine in the timeline.

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

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 do I determine that my audio sample rate matches my tape?

How do I determine that my audio sample rate of my capture preset matches the sample rate of my tape?

You've really got two issues.
DV records audio and video differently and if the camera is substandard, you will have drift over long captures. Solutions: new, better camera or capture in small (max 10 minutes) chunks.
The other issue is the 32k sample rate, which will exacerbate issue 1. You can duplicate the 48k preset and modify the copy for 32k and work in that until you are ready to deliver. Then copy it into a 48k sequence, render and output.
It's funny, sometimes I get that pop up but everything is fine.

Audition 3 seeing a different sample rate setting than what the device shows

Hi,
I have just installed Adobe Audition 3, along with the 3.01 patch, on a brand new system running Windows 7 64 bit. The mother board is an Asus Sabertooth X58 using Realtek High Definition Audio. The device drivers show that the audio sampling rate for line input is set to 24 bit 192K. I wanted to set it to the maximum that the sound card would allow to test performance and audio quality.
The problem is when I bring up Audition 3 and hit record, I get the message "We do not support recording when your file does not match your hardware sample rate. Your current hardware sample rate is 44100Hz". Clearly this is not the case since the Line In Properties - Advanced tab is displaying "2 channel, 24 bit, 192000 Hz (Studio Quality).
Under Audition's Audio Hardware Setup it shows only one choice for Audio Driver: Audition 3.0 Windows Sound. It also displays Sample Rate: 44100Hz, Clock Source: Internal, Buffer Size: 2048 samples with no way to change these values.
If I click on the Control Panel button I get:
DirectSound Input Ports:
Device Name: Line In (High Definition Audio Device
Audio Channels: 2
Bits per Sample: 16
Anyone know of how I can change these settings to get Audition to agree with the device settings?
Thanks
Dale

DaleChamberlain wrote: Anyone know of how I can change these settings to get Audition to agree with the device settings?
I'm afraid that life is nowhere near that simple. The main issue here is that Audition, in common with most audio software, uses a driver system called ASIO to talk to the sound device - this cuts out a lot of the OS and reduces the latency of the system considerably. There are several problems with ASIO though - the first being that it only supports a single device per system (or sometimes multiple identical devices if the manufacturer can make them look like a single device), and with software designed to use this driver, then to use any other driver (like a native Windows one) you have to use a converter stage like ASIO4ALL. This will convert the ASIO streams to WDM, and let you use multiple sound devices - but with increased latency.
It's the second problem that's really going to stuff you though - and that is that quite reasonably, ASIO is limited by its inventors to run only at three sample rates; 44.1k, 48k and 96k. So there's no way you can run at what you think might be a higher quality setting. All settings above even 48k are making your sound device work much harder, and for what? All that happens is that you increase the potential frequency response to way beyond the human hearing range - to no purpose at all. You don't have sources that can produce useful output at these frequencies, and you certainly don't have the means to reproduce them. This has all been well documented and explained before, so I'm not going over all that again. In a nutshell, Nyquist points out that any digital sampling device has a frequency response limited to a maximum of half of the sample rate, so for 48k that gives us a frequency response up to 24kHz - comfortably higher than any adult can hear by quite a long way. Anything you sample and record beyond this by using even 96k is nothing but noise as far as humans are concerned, and unpercievable noise at that.
So what the line input properties tab is saying is, if you have a non-ASIO driver designed to support all potential rates, possible. You don't have an ASIO driver available, because it's a built-in sound device, and anyway you've already pointed out that it's using the Audition Windows driver (a cut-down version of ASIO4ALL, effectively), so a conversion is already taking place. What Realtek refer to as 'High Definition Audio' is no such thing - all on-board sound devices of this nature are of universally low quality, and to improve this you'd need an external device - of which there are many available, usually with dedicated ASIO drivers. But none of them will work with ASIO beyond 96k, simply because the standard doesn't support any higher rates.
If you download and use ASIO4ALL (it's free), then you will get an additional control panel which will show you exactly what your sound device is capable of doing as far as Audition or any other ASIO software is concerned, and this is a useful diagnostic tool anyway, so it's worth doing. You just select this option when installed, instead of the Audition Windows Driver.
I'm sorry to be the bearer of what seems like bad news, but actually, it isn't. You will percieve no quality difference at all running at anything beyond 48k sample rates; all you will be doing is wasting your computer's resources unnecessarily. You waste both processing resources and hard drive space by processing at ridiculously high sample rates, and there are zero returns.

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.

Capturing issues-audio sample rate & locating timecode break

I am a first time FCP user.
I am working on an educational video with hours of DV source tapes.
The source tapes are from a 2R50MC Cannon mini DV camcorder also used as the deck I am capturing from via firewire. My scratch disk is Maxtor III 500GB external drive hooked up via a firewire (I'm realizing the firewire is years old could that be a problem?)
In my first capture test I captured 1 minute of tape by setting in and out points and after completeing the capture I received an error message—"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 capture preset matches the sample rate of your tape."
For the past day I have been scouring the user manual, this discussion group and training tapes from Lynda.com and can not resolve this issue. I have determined that the source tape is 12 bit or 32 kHz and yet can not find a way to set up capture preferences to 32 kHz. Is this the problem/resolution? Page 320 of the user manual shows the QuickTime Audio Settings dialog box but I can't find it in FCP or Quicktime. Is this where I make the change to 32kHz?
After doing the first test, I tried again and ran into a new problem—it stopped capturing with the message: Locating timecode break [press esc to abort]. It never seemed to resume capturing and I pressed escape to abort. If I'm reading FCP time code window correctly I received this time code break message within a section I had already captured previously (with the audio sync warning).
I have done several capture tests at different points in the tape and on different tapes all with the same results. I've used the capture clip and capture now buttons. I've tested with drop frame turned on and off. Confirmed my setting of: At timecode break "Make New Clip." Confirmed my Easy Setup as DV-NTSC. My capture preset is DV NTSC 48 kHz. I've turned off and on FCP. Restarted my computer. Restarted my computer with the shift key down and ran permissions.
Any ideas are GREATLY appreciated, T.

I am concerned about available memory when capturing to my hard drive and the babysitting and extra steps involved considering the amount of tape I want to capture...but it does work that's GREAT!
This is always a concern, but in your case I think having the camera and external on the same bus was causing your problem. You may have to capture a little and then transfer, rinse and repeat. Just don't try to do too much at once and let your System Drive get too full, you'll run into other problems there. Slow and steady is the pace!
I shouldn't have stacked my questions since you answered one and Chris answered the other. Don't know how to apply the answered question and who gets the points.
No problem, just mark it answered and divide up the solved and helpful points as you wish. All in all we really don't care too much about the points, but they do make us feel good! Thanks for your desire to use the forums properly.
K

Capturing-audio sample rate

I have fce hd v 3.5. My camera is a sony dcr-trv11. I am trying to "capture now" but it looks like I am dropping frames. When i hit "esc" it stops capture of course, but I get a message that reads "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 ans audio of these media files to be out of sync. Make sure the audio sample rate of your capture preset matches the sample rate of your tape." Okay, so how do I do that? When I originally captured my footage with imovie I had no problems. I had trouble importing them to FCE so I scrapped them all and decided to start from scratch with FCE. I have 199.5 GB of free space.

Most miniDV cameras can be set to record audio as 12-bit or 16-bit; and most of them come set to 12-bit as the default.
Your preset (easy setup) in FCE has to match both the video & audio setup of your camera. Normally, you would select the DV-NTSC easy setup in FCE, which would give you a sequence that expects DV-NTSC video and 16-bit (aka 48KHz) audio. If your video was recorded as 12-bit (aka 32KHz) audio, but you captured into a 16-bit (48KHz) sequence in FCE, that would give you the mismatch.
Check your camera - if it was set to 12 bit audio for the tape you are trying to capture, then in FCE you should select the DV-NTSC 32KHz easy setup for your sequence before you capture your tape.
There are many different reasons you might be getting dropped frames - can you tell us more about your exact setup, esp. if you have an external HD connected to your system. Oh, and by the way, 512MB is the bare minimum to run FCE, you will find things much better overall if you upgrade to at least 1GB.

Geting audio sample rate error, help

Hey all, been doing a massive project where I ma bring in tons of old 8 mm tapes, hi8 and digital 8. This one tape I brought on though however is giving me grief. I keep getting this error:
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 files to be out of sync. Make sure the audio sample rate of your capture preset matches the sample rate of your tape.
If find this odd though cause the tape should be standard, NTSC dv 48 K. Any suggestions. Also how do I reset my final cut so that when I plug in a camera it always reads it as what it recognizes. I ask this because I had to used the setting uncontrollable device because on the original old 8mm tapes there is no time code so I had to capture that way.
Anyhow, any suggestions on this would be great for if I recapture. Cause I could line it up by eye but want to find the problem so I know for the future. Thank you.
Nathan

This is a recent problem that seems related to a recent upgrade of QuickTime. Here's why.
In the last month, a rash of these posts have begun to appear:
"DV Capture Audio problem"
http://discussions.apple.com/thread.jspa?messageID=6708693&#6708693
"audio/video"
http://discussions.apple.com/thread.jspa?messageID=6591262&#6591262
Plus this thread, plus my own.
In my case, nothing changed in my operating system or Final Cut Pro version. I upgraded to QuickTime 7.1.6, and the problem began. I have upgraded all the way to 7.4 to no avail. When I attempt to import a DV clip using the same Sony DVCAM deck that imported the same clip in December, I get the error every time. Nothing has changed in the tape, the deck, the Project or Final Cut. I am simply unable to import DV video. I can import other kinds (Panasonic P2, for example), but DV is a no go. I cannot get rid of this error.

Audio Sample Rate Query

Hi FCE users,
I am about to start capturing 4 hrs of wedding video. I have a fair bit of experience on FCP but am new to FCE and its variations in functionality.
When I ran a quick capture test, it warned me on completion that the audio sample rates were different. The video was shot in Hungary and I have no way of contacting the videographer, however I assume that it was recorded in 32khz as when I changed the capture preset to this, no warning message occurred. The DV cam I'm using most helpfully doesn't display a bit rate but given that no warning occurs after capture, I'm happy to go with the 32khz preset.
Most of the articles posted on the net however state that the 'SEQUENCE PRESETS' as well as the 'Capture Presets' need to be changed. Almost all of it relates to FCP, as opposed to FC Express.
The equivalent on FCE - 'Sequence Settings' doesn't seem to offer a 32khz option, and the Audio Rate shown in the Browser seems to be fixed to 48khz.
My question is:
Does the sequence setting variation matter? - Do I assume that given the scaled down functionality of FCE that it takes care of this automatically OR that 48khz is a fixed audio rate for sequences in FCE and all non 48khz audio clips needs to be rendered prior to output.
No doubt I will be bringing music from CD's in to the project and sequence also, so your help would be greatly appreciated.
I don't want to bring in 4 hours worth of footage and then have to recapture!
Thanks for your help!
Rich

Thanks for your swift reply Tom.
I've just re-reviewed the menus. I can display the 12/16 info on record, but it doesn't offer it as a playback display option. However... I've just recorded more footage into FCE and received no error message with the capture preset on 32 so can I be fairly confident that its 32?
I've just figured out how to change the sequence format to 32 (it seems to be created by default with capture preset set to 32, wheras I was trying to alter an existing sequence).
Rich

Cannot export as WMV - Inconsistent Audio Sample Rate error

Hello,
I am getting an error message when I try to export a sequence using quicktime conversion to a WMV. "Inconsistent Audio Sample Rate - The media you are exporting contains audio with multiple sample rates".
I used Soundtrack Pro for the first time- to analysis and fix an audio file with pops. It worked wonderfully. I don't have much experience with this program and the audio is set to 32-bit stereo with a file ext stp. Not sure how to change this or if I need to.
I have done some tests and I have no problem exporting as a quicktime. I am using the latest version of flip4mac software. Your help is appreciated.
Cheers,
Chris

Hi Susan,
Thanks for your quick reply. The link helped alot.
For others facing this problem, I found this solution.
1. Export sequence as a Quicktime movie (not conversion) using current settings and make movie self-contained.
2. Create a new sequence and change the sequence audio setting (under sequence-general) to 44.1 kHz - Dept 16-bit config default
3. Drag your Quicktime movie into this new sequence.
4. Export using Quicktime conversion to WMV with these settings for audio.
5. WMA9 Standard - One pass, CBR at 64kbps, 44.1 khz, Stereo.
Voila.
I have struggled with flip4mac from the start. Never had any luck using the 2 pass method which 1'm told is better quality. Tech support at the flip4mac discussion group generated much discussion but no solution. I have settled with single pass quality in order to get the job done.
Does any remember the days of linear editing when you delivered a master, protection copy and some VHS dubs and you were done? Oh, I'm showing my age. Ha.Ha.
Cheers,
Chris

Matching Audio Sample Rates In FCP

When the message comes up, "Audio sample rate must match sample rate of source tape (Log and Capture window) what exactly does one have to do to ensure a change to the correct .. ?

What I would really like first is for someone to point out the steps on the DSR11 remote for selection of both video and audio. I'm having to remember the three different selection places as I'm doing it "blind" and I can't see the results of my inputs whilst enabling it. The only confirmation I get is the Capture screen showing the SVHS tape playing if I get it right but that doesn't answer the question of how come I don't get a TV monitor screen showing up when then playing the captured material. That, as well as no audio which could possibly be due to a wrong audio preset in the DSR 11 due to my lack of visual. The video setting has to be right for me to get a Capture screen on my desktop. Normally, when starting FCP, the TV Monitor screen changes to a colour indicating connection.
With frustration at a peak, and just in case I'm in some way unhappily missing the right Capture preset, I'll bow to anyones step by step workthrough on the procedure.
Firewire between Macpro and DSR. SVHS in and out connections between DSR and SVHS VCR. Ongoing scart connection from VCR to TV monitor.

How is Core Audio sample rate set?

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