Fixed but still says Inconsistent Audio Sample Rates

I imported some purchased music .aif files; completed my piece then tried to export a .wmv file through Quicktime. It told me I had "Inconsistent Audio Sample Rates" so I went back and re-exported the file through Quicktime with a sample rate matching the rest of my project.
I'm still getting the "Inconsistent Audio Sample Rates".
What do I need to do to get the project exported?
Cheers,
D

I have also been getting this error - just starting today. I tried exporting a mixdown in aiff, then converting that to MP3, bringing it back into a new sequence -- all with no luck.
However, I then exported this mixed-down version as a self-contained movie. Then, I did the wmv conversion with Cleaner 6 and it works. Slight glitch in frame 1 for some reason (motion file within the FCP sequence) -- but it seems like it's on the right track.
I also get the feeling this is due to the audio update. These are 8-track sequences I've been working with on and off for two months -- and the first time I try to render out after the audio, osx, and QT update, and it's crapping out on me. Could also be the changes to the h.264?
I've been spending all day on this, and I'm at my wits end!

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Inconsistent Audio Sample Rate - Export issue

I am attempting to export my project into a Windows Media file and I'm getting an error that says "inconsistent audio sample rate" Does anyone know how to fix this problem?

That error was introduced with the release of Quicktime 7.2. Flip4Mac released a beta with the fix for that particular audio error thing HERE. I don't think the final is ready yet, just the beta, but it's working for me. Don't be "put off" by it, but, before you can download the beta, they ask you to fill out some registration info so you can report back any problems you might encounter.

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

Message: inconsistent audio sample rate

when trying to Share my movie to create a .wmv file, I get the following message: "inconsistent audio sample rate -- the media you are exporting contains audio with multiple sample rates."
The audio to this clip is not the native audio. I replaced it ( on the second track - the first track audio I muted by turning off the check mark next to that track) with audio imported from Itunes. That audio clip is an aac audio file, 128 kbps, 44.1 khz. In expert settings I matched this, and indicated coding method: One pass constant bit rate (CBR).
Anyone know what it is I am doing wrong? Many thanks

No. That is not how it works.
44100Hz is the frequency and the audio sample rate could be 8, 12, 16 or even 24. Same with 48, 96 and higher frequencies. Open the Audio MIDI Setup app to see what your machine and sound card can offer.
QuickTime files can combine up to 99 tracks so a mix of sample rates wouldn't be much of an issue. WMP exports (I've never made one) seem to require only one audio source or constant sample sizes.
iMovie uses 48KHz because that is how it comes off DV tape. AIFF files are usually the same as audio CD (44.1KHz and 16 bit sample size) and I don't know if iMovie upsamples or otherwise changes them at import but I doubt it.
But 44100 does not equal 12 bit sample size.
Hope this helps clear that issue up.

Error message stops .wmv creation: "Inconsistent Audio Sample Rate..."

Just recently when attempting to export a video into the .wmv format, QuickTime Pro 7.2 gives me this alert message, "Inconsistent Audio Sample Rate. The media you are exporting contains audio with multiple sample rates." QuickTime will not compress (export) to .wmv. It will export to .mov. and .m4v.
This problem is new. For more than a year my weekly video exported to the .wmv format without fail.
I am working on a Mac G4 Dual Processor and iMovieHD.
Any help you can offer would be greatly appreciated.
Mac G4 Dual Processor Mac OS X (10.4.10)

Just in from flip4mac support:
The recent release of QuickTime 7.2 created an audio compression error for Flip4Mac WMV Studio, Studio Pro and Studio Pro HD customers. Specifically, when encoding WMV files with Flip4Mac WMV v.2.1.2 using uncompressed audio sources on QuickTime 7.2 systems, customers were receiving "Inconsistent Audio Sample Rate" error and the encode will fail. This problem has been resolved with the beta release of Flip4Mac WMV v2.1.33. If you are using QuickTime 7.2, please upgrade to this release at http://www.flip4mac.com/wmv_beta.htm

Error Msg.: "Inconsistent Audio Sample Rate" with Flip4Mac

If you're using Flip4Mac to produce WMV files, the latest QT updates will create a conflict with Flip4Mac, which produces the error msg. referenced in the subj. heading above.
If you're using Flip4Mac, you may want to wait until Telestream resolves the issue before upgrading QT.
More info is here: http://www.flip4mac.com/knowledge/kb_0070.htm
and here: http://forum.flip4mac.com/forum/messageview.aspx?catid=10&threadid=270&enterthre ad=y
MacPro, MBP 17in (2 GB RAM), PB G4 15in, PB G4 12in, iMac G3 Mac OS X (10.4.9) Loyal to Apple since the Apple ][+

Just in from flip4mac support:
The recent release of QuickTime 7.2 created an audio compression error for Flip4Mac WMV Studio, Studio Pro and Studio Pro HD customers. Specifically, when encoding WMV files with Flip4Mac WMV v.2.1.2 using uncompressed audio sources on QuickTime 7.2 systems, customers were receiving "Inconsistent Audio Sample Rate" error and the encode will fail. This problem has been resolved with the beta release of Flip4Mac WMV v2.1.33. If you are using QuickTime 7.2, please upgrade to this release at http://www.flip4mac.com/wmv_beta.htm

Export/share wmv "inconsistent audio sample rate"

I've never used iMovie before today. I successfully imported a mini DV clip and need to "share" it as an wmv. I keep getting an error stating "inconsistent audion sample rate". What does this mean and how do I fix it so I can save my clip as a wmv file?
Thanks much.

That error was introduced with the release of Quicktime 7.2. Flip4Mac released a beta with the fix for that particular audio error thing HERE. I don't think the final is ready yet, just the beta, but it's working for me. Don't be "put off" by it, but, before you can download the beta, they ask you to fill out some registration info so you can report back any problems you might encounter.

Inconsistent audio sample rates....? pt. 2

I'm running Compressor3. The last three projects I've done have all been put together on fcp6, and the audio has been sampled at 48khz. When I go to compress, I get an 'inconsistent/multiple audio sample rates' warning. However, all my audio is set to the same rates. I've tried cutting down to only one audio source, and tried also resampling through Soundtrack to 44.1, adjusting my settings in Compressor as well. Any suggestions?
I should add that I can compress from .mov to .mp4, etc.....but the one format I need is .wmv, and that's the one that is giving me trouble.

It is a well-known issue but it's not highly publicized. Flip4Mac has a problem with the last round of Quciktime. You willneed to download a new rev of Flip from Telestream's site. I think it's still being called a beta version, even after several months.
bogiesan

Inconsistent audio sample rates....?

I'm running Compressor3. The last three projects I've done have all been put together on fcp6, and the audio has been sampled at 48khz. When I go to compress, I get an 'inconsistent/multiple audio sample rates' warning. However, all my audio is set to the same rates. I've tried cutting down to only one audio source, and tried also resampling through Soundtrack to 44.1, adjusting my settings in Compressor as well. Any suggestions?

Which codec are you using in Compressor? If you are using Flip4Mac, have a look at this knowledge base article: http://www.flip4mac.com/knowledge/kb_0070.htm

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.

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.

Audio Sample Rates Don't Match

I am capturing a MiniDV tape via Firewire to a Final Cut project
w/ a Capture Preset of DV/DVCPRO-NTSC w/ 2 channel 48KHz 16-bit
audio sample rate
and when I check the Item Props on said clip in Browser it says the same info
I've toggled both Mono and Stereo on Capture Settings, but after capturing,
in all cases, I get the same warning message:
Audio Sample rate one your captured media files doesn't match the sample rate on source tape. This may cause it to go out of sync. Make sure the capture preset matches sample rate of tape.
Please advise, thanks

I am capturing with a Sony MiniDV camera w/ a Firewire400 cable,
and the tape is a Sony MiniDV (DVM60) Premium tape
I am unsure of the camera that shot the footage, which is from 1998
and was one of the earliest MiniDV cameras (a large one)
I tried both Firewire Basic and Log and Capture
but the problem still persists
any other options?
thank you

What is the best way to deal with different audio sample rates on the same timeline ?

what is the best way to deal with different audio sample rates on the same timeline ?

You don't have to do anything special. If possible, start your project with a clip that has the desired target frame rate and audio sample rate, and your project parameters will be set automatically. Other sample rates will be converted under the covers.
For example, if your video is shot at 48khz, you can add music files at 44.1khz with no problem.
If you are recording audio that you want to synch with video (multicam), you will get best results if everything is 48khz, but you can use 44.1 if that is all you have. Once I forgot to reset my Zoom to 48,000 and it still worked.

Highest audio sampling rate in CS4?

Hello,
I apologize if this has already been asked, but I have been searching everywhere and I simply cannot find the answer to this.
What is the highest audio sampling rate that can be utilized in Premiere Pro CS4? Can it import and export 192kHz 24-bit audio?
Thanks in advance

Hey Hacienda,
I might not have the experience in audio work you have since I've only been doing this for the past 6 years or so. But I've been a musician for far longer than that, and I've learned A LOT mostly from really smart people in the industry. So, I'm not gonna lie to you and say that I've done extensive testing in this area because I simply do not have the equipment, nor the money to buy it (WAY too expensive). But we do share the neophyte status when it comes to video editing :-P
Anyways, the Nyquist Theorem is not a theory, which is what people are led to believe. It is a theorem, meaning it's already mathematically proven. It is proven that, as long as you follow the premise of capturing twice the highest frequency of the sound source, you'll get a perfect reproduction of it. To capture more than that is a waste of bandwith specially because most people won't even hear above 18KHz, nor do they have the equipment to reproduce such frequencies. Most consumer systems and audio gear, including those found in professional studios, go up to about 22KHz. You need to spend BIG dolars for anything that goes beyond that. So, who are we really making music for here? The super rich? Dolphins?
Now, I know you're not just talking about higher frequencies, but the amount of samples needed to recronstruct a perfect copy of the original waveform. OK, well, this is the kind of snake oil marketing BS I was talking about. The biggest one being that 1bit DSD crap that Sony/Phillips is pushing. Adding more samples to the recording will not make any difference on how faithfully you can reproduce a sound. It'll just make the files bigger for no reason. Again, the Nyquist Theorem already proves this. This is FACT! Here's a link I found interesting regarding these audio industry lies, maybe you will too: http://theaudiocritic.com/back_issues/The_Audio_Critic_26_r.pdf It starts on page 5, but the one pertaining this discussion is lie #3 on page 6. :-D
Don't forget that modern converters already sample at much higher frequencies than the target sampling rate. I believe my RME Fireface 400 samples at 5.6MHz, which is twice the amount of samples compared to DSD technology, before going back down to the target rate. But, like I said, it does so for other reasons and NOT because it needs that many samples in order to faithfully reproduce a waveform. Of more importance are the quality of the FIR (Finite Impulse Response) filter and the clock inside the converters. These components are what make a converter high grade, among others. The converter chips themselves are very inexpensive (in the tens of dolars) which why you hear some companies advertizing having the same converter chip as a ProTools HD rig (not the best example I know).
By the way, I didn't say humans can only hear up to 20KHz. I'm sure there are people who can hear above that. My point was that the 20Hz - 20KHz range is what's generally accepted as an average for humans (which implies that there are people who can hear avobe/below that). Also, the reason why modern-day pop records causes headaches and sound horrible is because of a totally different issue known as "The Loudness War" (I'm sure you know about it so I won't go into details). However, I do agree with you as far as compressed audio goes. Unfortunately, there's a reason for that and there's nothing we can do about it until the day Internet bandwith becomes more accessible and cheaper. Eventually it'll get to the point where uncompressed audio can be streamed reliably through the net. But, until then, we're stuck with MP3, AAC, DTS and other audio compression formats. As far as digital media distribution goes, it's the future and companies are seeing that. More and more people download music rather than buying CDs, so I do believe those numbers are accurate. Just look at sales from iTunes and even games like Guitar Hero and Rock Band. It's just a matter of time.
Take care!

Change capture settings to 32k audio sample rate???

I have a DVCAM tape that my client recorded and the audio sample rate is 32k. I have noticed that the audio seems to get out of sync when I capture it using the 48k NTSC. I also get a warning saying that my source tape's audio sample rate doesn't match the capture settings and that audio could be out of sync. I looked at past posts and it does seem to be the 32k verses 48k.
My question is how do I change my capture settings to 32k? I see how you do it on the sequence settings, but I can't figure it out on the capture settings.
Thanks,
John Schenkel

John,
I recently read a post where there were four likely answers to the problem, yet I was not sure which one was the best solution.
Why? Because even though the question was marked as answered the person posting the original question didn't mark which individual's answer was the solution.
Please help future users to find answers more quickly, who knows, you may find yourself in the same situation someday.

Fixed but still says Inconsistent Audio Sample Rates

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