Audition sample rate mismatch

I have been using Audobe Audition 2.0 for a number of years, and now have upgraded hardware and software (Windows 7 running on a new good spec 64bit Dell), with the intention of running CS6.
Unfortunately when I go to record, using the same linein as previously, I get the following message:
"Sample rate of the audio input and output devices do not match. Audio cannot be recorded until this is corrected."
I think I know what it means, and have ensured (as far as I am aware) that the sound card is set to 44100, 16bit, the same as the settings on Audition, or appear to. So what have I missed. (I do know XP better than 7).
Many thanks
Peter G

You also have to set the sample rate in the Windows Sound control panel. Select your audio source in the Windows Sound Recording panel and click on Properties. GO to the Advanced tab to make sure that the sample rate is set correctly there as well as everywhere else. Hopefully that should sort the problem. This is an extra vaguery of Windos 7 audio drivers.

Similar Messages

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

Audio problem ( sample rate/ mismatch samplerate )

Hi everyone,
I'm using a Macbook white Mid 2010, everything is fine but recenly i bought a USB interface to record music from my instruments ( Mic and guitar ) through garageband..
I have watched steps on how to connect guitar to garageband, and i did those steps by selecting USB channel as input..
The problem is every time i test my guitar or microphone, it sounds terrible just like little chipmunk or something. Some ppl told me my sample rate is mismatch. I also was checking around on Audio Midi Setup, i can't change my sample rate for my USB audio interface. The input for it, is 28000KHz, and i was told garageband only accepts 44000KHz.. So i thought my USB interface was having a problem..
However today i have tested my microphone on my old iMac white, i did plug in my USB interface and microphone, it worked so fine. The sound was not like on my macbook. it just fine and very clear. So my USB interface is not the problem..
BTW audacity also works fine on my macbook, only garageband..
Could someone help me? I can give you more details about it..

Canon cameras are notorious for recording with nonstandard sampling. Usually, though, material will capture without synch problems.
Make sure you have "abort capture on dropped frames" and "create new clip on timecode break" enabled in fcp: user preferences: general and do not capture over a control track break (where you see just "noise" in the video

Sample rate mismatch - FCP7-3/4" Video-Canon ZR40 digitizer.

Using FCP7, I am capturing 3/4" Video w/ Audio using a Canon ZR40 camera as digitizer. Capture setting of DV NTSC 48khz. This works fine for VHS capture, but when inputting 3/4" I get the dreaded "Audio sample rate...does not match sample rate on your source tape....." This setting worked fine for VHS source tapes. Audio is out of sync with video in captured sequence. What am I doing wrong?

Canon cameras are notorious for recording with nonstandard sampling. Usually, though, material will capture without synch problems.
Make sure you have "abort capture on dropped frames" and "create new clip on timecode break" enabled in fcp: user preferences: general and do not capture over a control track break (where you see just "noise" in the video

Slow Playback Adobe Audition CS5.5 - not sample rate setting error!

I'm currently using Windows 7 and Adobe Audition CS5.5 build 1815. Soundcard is RME Digi96 Pad. Ok its an old card and the drivers date from 2005. In Adobe 3.1 and CS5.5 I get slow, around half speed, playback at 44k and 96k sample rates. YES the project sample rates and sound card sample rates are set correctly, clock is set to master on the soundcard.
However, in using Cockos Reaper 3 or 4 on exactly the same computer, same soundcard, same OS, I can play 44k and 96k wav files perfectly.
This suggests to me there is an error in the way Adobe Audition 3 / CS5.5 is handling the ASIO drivers in Windows 7 with some soundcards.
If it was a driver problem, or indeed a soundcard hardware issue, then Reaper would not perform properly.
So what's going on with the ASIO drivers in Adobe Audition with Windows 7?

I have two XLR wireless mics, which are great for my video work, but I need to do audio recordings for podcasts and wanted a mic that would be able to do just audio podcasts, as well as video podcasts, live via my laptop. So while I do agree that the audio recording from my XLR mics is great, I am not happy with the audio quality of standard computer mics I've tried and the only way I can get my XLR mics into the computer directly is through a USB connector (which I also have, I'm getting the Yeti set-up for my business partner to use).
Maybe Audition just isn't the best software option for this, but since we have CS6 Premium I had hoped I could utilize software I already had. (And for the live video casts I'm looking at starting out with Google hangouts...)
With that being said, I appreciate your reply and thought I had already set-up the hardware correctly in Windows & Audition. Here is a screenshot that shows the settings I have - my question at this point is whether the master clock needs to be changed (the audio channel mapping tab shows the left and right speaker of the default Realtek speaker):
I just double checked the properties of both the speakers and mic and they are both set to 48,000/16/stereo, as is the audio file in Audition.
I'm a journalist, not an audio pro, and I fully accept that the sound device must not be properly connected. The problem is that I cannot figure out why that is the case.
Since I need to use the USB mic and I want to be able to use the power of Audition to put together podcasts, using other software for the livecasts, I would really like to sort through this. Any and all help is greatly appreciated!

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.

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.

I'm trying to record in Adobe Audition and it keeps saying my the sample rates for my input and output devices do not match. How do I correct this?

I'm trying to record in Audition and it keeps saying my sample rates for the input and output devices don't match. How do I fix this?

I finally have communication between the ISA One and Audition. I moved the optic cable to another SPDIF on the computer and from the SPDIF to ADAT on the ISA One and it finally agreed that the communication was at the same clock speed. The audio was not intelligible so I moved the optic cable back to the original configuration and I can record voice that is clean.

Setting Sample Rates in Audition

Hi I am trying to record in Audition and it says my sample rates don't match. I have tried and failed at this so I keep going back to sound booth. Any ideas please?

The problem may be either the settings for your soundcard or more likely the operating systems sound settings if you are using Windows. What soundcard or audio interface are you using?
See this Audition FAQ:
http://forums.adobe.com/thread/973133?tstart=0

Mismatching Audio Sample Rates Upon Capture

I recently bought a cheap MiniDv JVC video camera, and I am having trouble capturing video and audio together in FCP 6.0. The capture seems fine, but when I finish an error pops up which reads: "The Audio Sample rate in your presets does not match the one on your source tape." As a result there is a delay between the video and audio tracks that creates huge amounts of extra work.
My video camera captures sound in 12-bit format, and from what I understand that usually corresponds to a 32MHz sample rate. I can't find any 32 MHz presets in the Easy Set Ups or other menus, the system seems geared entirely towards 48 MHz sound. I've messed around with some different combinations of settings, but so far to no avail. Any help someone can give me will be much appreciated.

You might want to duplicate the capture preset for DV/NTSC 48k and change the audio to 32k.
Then try that.
But I'd also change the JVC's audio settings to record in 48 while you're waiting. Should be able to do it... cheap or not.
CaptM

Sample rate issues in Audition CC and CS6

I'm using a Blue Spark mic with the usb adapter and I keep getting an error message saying the sample rates don't match the device. I've tried several different sample rates and none of them work. I can monitor the audio from the mic but when I hit record the error message pops up.
Here is the message I keep getting and the settings I'm using: http://imgur.com/a/WpJST. (For some reason the insert image button isn't working so I just linked the screenshots through an image host.) For some reason the icicle doesn't show up as an output device so I just chose a default.

Try the Bob Howe's suggestion in this thread:
http://forums.adobe.com/thread/940613

Can I mix down to 32 bit at a higher sample rate than 44.1 kHz ?

When I use Ableton Live, it lets me choose 16, 24, or 32 bit, and then I can choose a sample rate all the way up to 192000. Is this possible in Audition ? I have been going through all the preferences and all the tabs and I can't find this option. All I find is a convert option, or the adjust option. But that's not what I want. I want to mix down this way.
The closest thing I found is when I go to "Export Audio Mix Down", I can select 32 bit. Then there is a box for sample rate, with all the different values. But it won't allow me to change it from 44100.

JimMcMahon85 wrote:
Can someone explain this process in laymens terms:
http://www.izotope.com/products/audio/ozone/OzoneDitheringGuide.pdf ---> specifically Section: VIII "Don't believe the hype"
I don't read graphs well, can someone put in laymens terms how to do this test, step by step, and where do i get a pure sinewave to import into audition in the first place??
UNbelievable: So I have to first run visual tests using a sinewave to make sure dither is working properly, then do listening tests with different types of dither to hear which I like best on my source material, and then for different source material it's best to use different types of dither techniques???... Am I getting this right???...
Hmm... you only need to run tests and do all that crap if you are completely paranoid. Visual tests prove nothing in terms of what you want to put on a CD - unless it's test tones, of course. For the vast majority of use, any form of dither at all is so much better than no dither that it simply doesn't matter. At the extreme risk of upsetting the vast majority of users, I'd say that dither is more critical if you are reproducing wide dynamic range acoustic material than anything produced synthetically in a studio - simply because the extremely compressed nature of most commercial music means that even the reverb tails drop off into noise before you get to the dither level. And that's one of the main points really - if the noise floor of your recording is at, say, -80dB then you simply won't be hearing the effects of dither, whatever form it takes - because that noise is doing the dithering for you. So you'd only ever hear the effect of LSB dither (what MBIT+, etc. does) when you do a fade to the 16-bit absolute zero at the end of your track.
Second point: you cannot dither a 32-bit Floating Point recording, under any circumstances at all. You can only dither a recording if it's stored in an integer-based format - like the 16-bit files that go on a CD. Technically then, you can dither a 24-bit recording - although there wouldn't be any point, because the dither would be at a level which was impossible to reproduce on real-world electronics - which would promptly swamp the entire effect you were listening for with its own noise anyway. Bottom line - the only signals you need to dither are the 16-bit ones on the file you use for creating CD copies. And you should only dither once - hence the seemingly strange instructions in the Ozone guide about turning off the Audition dither when you save the converted copy. The basic idea is that you apply the dither to the 32-bit file during the truncation process - and that's dithered the file (albeit 'virtually') just the once. Now if you do the final file conversion in Audition, you need to make sure the dithering is turned off during the process, otherwise all the good work that Ozone did is undone. What you need to do is to transfer the Ozone dithering at the 16-bit level directly to a 16-bit file proper, without anything else interfering with it at all. So what you do with Ozone is to do the dithering to your master file, and save that as something else - don't leave the master file like that at all. After saving it, undo the changes to the original, in fact - otherwise it's effectively not a file you could use to generate a master with a greater-than-16-bit depth from any more, because it will all have been truncated. Small point, but easy to overlook!
just what's the easiest way to test if a simple dithering setting is
working for 32-bit down to 16-bit in Audition?... Why is there no info
about dithering from 32 bit to 16 bit (which is better then dithering
from 24-bit isn't it)?
I hope that the answers to at least some of this are clearer now, but just to reiterate: The easiest way to test if its working is to burn a CD with your material on it, and at the end of a track, turn the volume right up. If it fades away smoothly to absolute zero on a system with lower noise than the CD produces then the dither has worked. If you hear a strange sort-of 'crunchy' noise at the final point, then it hasn't. There is info about the 32 to 16-bit dithering process in the Ozone manual, but you probably didn't understand it, and the reason that there's nothing worth talking about in the Audition manual is because it's pretty useless. Earlier versions of it were better, but Adobe didn't seem to like that too much, so it's been systematically denuded of useful information over the releases. Don't ask me why; I don't know what the official answer to the manual situation is at all, except that manuals are expensive to print, and have also to be compatible with the file format for the help files - which are essentially identical to it.
Part of the answer will undoubtedly be that Audition is a 'professional' product, and that 'professionals' should know all this stuff already, therefore the manual only really has to be a list of available functions, and not how to use them. I don't like that approach very much - there's no baseline definition of what a 'professional' should know (or even how they should behave...), and it's an unrealistic view of the people that use Audition anyway. Many of them would regard themselves as professional journalists, or whatever, but they still have to use the software, despite knowing very little about it technically. For these people, and probably a lot of others, the manual sucks big time.
It's all about educating people in the end - and as you are in the process of discovering, all education causes brain damage - otherwise it hasn't worked.

Recording LP records as source material- Sample Rate

Using recorded tracks from LP records to make DVDs, Blu-Ray DVDs or simple CD's. Am not sure what maximum sample rate to use. I understand the end product limits of the various digital media, but LPs are analog. Do I gain any sound quality by recording the original LP at a sample rate higher than 48000/32bit, say 96000 sample rate) and then resampling (downsizing) the audio file if the end product cannot produce the higher sample rate?

Conversion de LP -Archivos Digitales
Se recomienda Grabalos con estas velocidad de Muestreo
Blue-Ray 96000Hrz. / 32 bits
DVD & 48000 Hrz. /32 Bits
CD DE AUDIO A 44100Hrz /32 Bits ó 24Bits
Te recomendaria Cambiarte a Adobe Audition
Saludos
http://soundcloud.com/creativoxpro/restaurando-audio-de-un-vinil
Para audio
8000 muestras/s
Teléfonos, adecuado para la voz humana pero no para la reproducción musical. En la práctica permite reproducir señales con componentes de hasta 3,5 kHz.
22050 muestras/s
Radio En la práctica permite reproducir señales con componentes de hasta 10 kHz.
32000 muestras/s
Vídeo digital en formato miniDV.
44100 muestras/s
CD, En la práctica permite reproducir señales con componentes de hasta 20 kHz. También común en audio en formatos MPEG-1 (VCD,SVCD, MP3).
47250 muestras/s
Formato PCM de Nippon Columbia (Denon). En la práctica permite reproducir señales con componentes de hasta 22 kHz.
48000 muestras/s
Sonido digital utilizado en la televisión digital, DVD, formato de películas, audio profesional y sistemas DAT.
50000 muestras/s
Primeros sistemas de grabación de audio digital de finales de los 70de las empresas 3M y Soundstream.
96000 ó 192400 muestras/s
HD DVD, audio de alta definición para DVD y BD-ROM (Blu-ray Disc).
2 822 400 muestras/s
SACD, Direct Stream Digital, desarrollado por Sony y Philips.
Para vídeo
50 Hz
Vídeo PAL.
60 Hz
Vídeo NTSC.
*informacion extraida para apoyo de la pregunta en el foro // http://es.wikipedia.org/wiki/Frecuencia_de_muestreo
> http://en.wikipedia.org/wiki/Sampling_rate

Converting sample rate when exporting

CS5.5
I'm using Audition to do some final mastering to 16-bit stereo 44100 wav files, in 32 float multi-track sessions.
All I'm doing is increasing the clip gain, then exporting to 16-bit stereo 44100 wave and 320 mp3.
I'm not changing the sample rate, channels or bit depth, but I am applying triangular dither when exporting.
The thing I've noticed is even though I'm not changing any of the properties of the files, when exporting Audition still tells me its "converting sample rate", despite the sample rate being the same as the source (44100). The quality advanced tab on the sample rate is greyed out.
Also, when I check the history box for the newly exported files it tells me "convert sample type on save as"
What is Audition doing at this stage?
How can it be converting the sample rate, when the sample rate is the exact same as the source? And the quality tab for sample rate conversion is greyed out?
All I'm doing is upping the clip gain slightly and then applying dither to these files. No sample rates are changed.
Any help greatly appreciated.

brazil101 wrote:
I'm using Audition to do some final mastering to 16-bit stereo 44100 wav files, in 32 float multi-track sessions.
I'm not changing the sample rate, channels or bit depth,
Except that you are...
If you open a 16-bit file as a 32-bit float, then you've changed the bit depth. It's the right thing to do if you are doing amplitude changes, as it makes any change effectively lossless, but it will still have to be converted back to 16-bit if you elect to store it like that.

Sample rate

I recorded some files with a sample rate of 44100.
When I went back to cut the files up, I would select the section I wanted to save and "saved selection as" to save the file. I choose the MP3 format.
Now on the menu I can change the Bit rate and select a mode but I can not alter the original Sample rate. My problem is that Sound booth saved some files at a sample rate of 44100 and some at 48000.
When imported into Captivate the 48000 files run slow.
Most likely I will need to install an old version of Audition to batch fix my files. I guess from here on I will use Audition so I don't run into this problem again. Or, since I'm new with Soundbooth one of you will tell me what I screwed up. :|
Thanks

The sample rate is generally based on the source media and the mp3 bit rate selection. You could convert the 48K WAV source files to 44K by choosing File, Save As, and selecting the sample rate you wish to use. Then export the converted file to mp3 with your desired bit rate selection.
I'll need to check and see if this has changed in recent releases, but Flash Player has, in the past, only supported 44K and 22K sample rate mp3 files.

Audition sample rate mismatch

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