Sample rate control not supported

Hi, I am trying to slow down the tempo of a Clip. I loaded the clip like this:
DataLine.Info info = new DataLine.Info(Clip.class, format);
try{
     clip = (Clip) AudioSystem.getLine(info);
     clip.open(mixedInputStream);
catch(Exception e){
        e.printStackTrace();
}When I try to obtain a sample rate control like this:
(FloatControl) clip.getControl(FloatControl.Type.SAMPLE_RATE);I get the message: java.lang.IllegalArgumentException: Unsupported control type: Sample Rate
Does this mean I can''t change the sample rate of my clip? Is there a way to fix this?
Thanks!

jamama wrote:
I get the message: java.lang.IllegalArgumentException: Unsupported control type: Sample Rate
Does this mean I can''t change the sample rate of my clip? Is there a way to fix this?I believe that the SampleRate control is only supported on data lines you're capturing from...to control the sample rate of the capture.

Similar Messages

Podcast not syncing because bit rate is not supported

For some reason 3 podcasts will not sync because the bit rate is not supported. How do I fix this so I can play them on my iPod. They are Video Podcasts.
PC Windows XP

Loaded HD is the High Definition version, intended for the Apple TV device. You cannot play this version on an iPod.
Look in the iTunes Store for the non-HD version - that should play. If you have trouble finding it, use the *Search iTunes Store* facility and it will appear in the Podcasts section. You may need to "see all".
Phil

TS1717 when trying to import cd the following message appears "the current encoder settings for bit rate and sample rate are not valid for this file"?

Trying to import cd when this message appears "the current encoder settings for bit rate and sample rate are not valid for this file". Any suggestions?

Thanks so very much. I chose MP3 and Download and it is working beautifully. Have a wonderful day and it is so nice of you to get right back to me. I'm trying to cut a DVD for my grandson's grad party and I got most of the photos and didn't have any music since I lost everything in a clean install. Could hug you!

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.

The selected combination of bit rate and sample rate is not allowed.

I've always burned my purchased CDs into iTunes using a AAC 320 kbps and 48.000kHz. Now when I select that option in iTunes 9 I'm getting a "The selected combination of bit rate and sample rate is not allowed." pop up screen. So is it impossible to burn my albums in a 320kbps/48.000 AAC format?. I had no problems in previous versions of iTunes using this setting. I guess I might have to switch over using a MP3 import setting instead.

Audio CDs are encoded with a sample rate of 44.1 kHz. Ripping them at 48 kHz requires resampling and does not improve the quality. Unless you have a special need for 48, use 44.1.

Audio sample rate does not match (HDcam to dvcam)

I'm trying to import clips and keep getting this 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."
Footage was originally shot on HDCAM and transferred to DVCAM elsewhere. Using FCP 5, am importing via firewire from a Sony DSR-11 deck. Using DV NTSC 48kHz Anamorphic as capture settings (though I've tried everything that I thought might possibly work with no success). The audio does not seem to drift over the course of several 5 minute or so clips. Clip settings show audio at 48 kHz (don't know if that's from capture settings or from actual data). Seems to me all audio should be 48 kHz 16 bits, so can't figure out what's going on. I have to export an EDL for the project to be finished in HD. Read some similar threads that ended in December, seemingly without much resolution. My broader concern is why this is happening; my immediate concern is do I need to worry about this right now since the media files will need to be recaptured in HD anyway. Any thoughts?
Thanks

A little more info. I'm having this problem on 4 tapes (from different cameras) that were transferred to DVCAM in a squished format to appear full screen on a 4x3 monitor. Video that was letterboxed and I can bring in with the standard DV NTSC capture settings does not have this problem. Still have the problem if I try to import the clips from the squished video with standard settings. Any thoughts?

High sampling rate is not accurate

Dear alls,
Any one know why the time loop is not accurate at defined period at the RealTime Trace Viewer ? In figure of VI, I set the period for the time loop is 250 us, but when I analyze in Real time Trace Viewer, it shows the real period time about 676 us. That means my data acquisition will be sampled in wrong sampling rate.
Could you give me some explane, please? And how can I solve this problem? I tested this code with MyRIO board.
Best Regards,
Kien

crossrulz wrote:
Why do you have a wait in your timed loop? The point of the Timed Loop is to state the loop rate.
??? Bigger problems than that! Why do you have any timing other that the TASK timing? And really, The task timing is most likely the loop rate.
Can you post your actual VI? I suspect something else is happening here that we cannot see (configurations, channels used, etc).
And a snip of the Task needs posting

Which sample rate can be supported by flash player?

Hi support:
I am a flex developer, i encounted a problem when i develop a flash program for my ipcamera,
i add a adpcm format audio stream to the rtmp stream, but the specification says that only 5.5k,11k,22k and 44k is supported,
but my device can only generate an audio stream which has a 8k sample rate, can you help me to do some thing.

This ISN'T support. It's a user to user forum. And unfortunately it ISN'T the one where the answer would be. You'd be better asking in the Flex Forum, as there are most likely people there who have a lot more knowledge about frame rates in Flex development than we do here.

Sample rate will not change

Logic refuses to change its sample rate from 44100. This happened all of a sudden last night when I opened a session I was mixing at 48000 and noticed it was playing back at 44100 (as it sounded flat and slow, obviously).
When I select a new sample rate from the Audio menu there is a click, Logic thinks for a moment, then there is another click, and Logic thinks a little longer then a brief spinning rainbow. When it has finished the sample rate is still 44100.
I am using the built-in audio driver on my MacBook. I have imported the audio from a 48000/24bit pro-tools session. I'm used to working in 44100 so I may have missed something obvious, but it seemed to be working for me yesterday.
Urgently need this working. Any help much appreciated thanks.

I've just noticed this today, and found the solution...
I was convinced that you could set the sound card's sample rate independently of Logic and I believed I had been running it at 192 KHz before, during and after a Logic session. Whether or not Logic would temporarily change this to 44.1 I can't say for sure, but I'm pretty sure if it didn't keep it at 192 during, it certainly reverted to it on closing Logic. But now, this isn't the case, and I'm finding Logic sets it and then keeps it at 44.1
The solution...
Set your project's sample rate to the desired rate in File > Project Settings > Audio
My gripe about this is that it's not trivial to change this once you've done a lot of audio work because Logic doesn't seem to understand that an audio sample at 44.1 is the same temporal length as one that's at 192, so it time compresses them (+and the pitch goes up+) if you up the sample rate. I would imagine the opposite would be true, although I haven't tried it.
So you really need to decide on the project's sample rate from the start, and stick with that (+unless you are just working with Apple Loops and Midi, in which case everything seems OK+).

Flash AX control not supported on firefox 4

Hi, when i went to play a game on jackpotcity.net i had a pop up message that told me that flash AX control was not supported & not compatible with firefox 4.
I dont have a problem with loading the game on IE.
also i had no problem with it until i installed firefox4.
Is there something i can do to rectify this as i cannot go back to the older 3.6 version of firefox.

Firefox 4 beta on the Android is not compatible with Flash. Mozilla are working with Adobe to add Flash support, but no time scale for when it will work.

"Variable Bit Rate Audio Not Supported"

Hello -
I have several AC-3 files - sourced from satellite broadcast downloads - surround 5.1, 448kbps/48kHz - that I cannot add to DVD SP. When I attempt to import them into the assests pane, DVD SP reports they are incompatible giving the error message in the subject.
Any ideas?

I too have noticed this today when trying to import a new CD. Every track has a variable bit rate despite not requesting it.
The older version of iTunes did not do this, just this new 7.5. I cannot seem to resolve. I tried to create a custom import without VBR but it ignored this too and did it anyway.
Apple, please help! This must be fixable somehow!

Sample rates do not match

Hi.
I am trying to generate a sweep waveform with 48KHz sampling frequency. However, I see that the lower frequencies have a 48KHz sampling frequencies but the higher frequencies have a bad sampling frequencies.
Can someone tell me how to generate tones with the same sampling frequencies?
It is the same issue with higher frequencies even if the waveform is single frequency (example 9khz waveform cannot be generated at 48khz freq). I am using a trial version Adobe Audition CC Build 732 (64 bit)

Generate Tones uses an entirely internal timing system, and effectively isn't influenced by any clocks at all. But even if you assumed that an internal clock drifted, it wouldn't account for anything like these apparent discrepancies. As an example, consider a 48k clock with a 2% error - which is way worse than you'd ever find these days. If you counted the number of samples over a millisecond, that would, unless you were extremely unlucky, still give you 48, because a 2% error is only 0.96 of a sample over that period.
But, if you opened up this recording in Audition as 48k, it would display it just the same as internally generated tone - so you'd get 48 samples anyway. What would be wrong when you played back the file (using an accurate sample reference for the playback) would be the pitch of your recording, and if it was a good drift, you could even end up with wow and flutter!
The answer to the question about recording external sources though is that it depends... If you have a S/PDIF input, it will use the clock from that (or you get a huge glitch when the sample rates drift to a one sample error). Same thing happens with a USB mic - the sync has to be established from the A-D clock source in the mic, and with a cheap example, that could be even worse than a computer-generated clock. So, it's only non-digital devices connected to the on-board sound device that are subjected to the vagiaries of the internal clock.
And yes, it's because I've considered this in some depth that I'm just a little sceptical about what's happening with the OP, if you see what I mean...

Extra colume controls not supported by KT4V

Hi
I have a KT4V M/B.I noticed that after installing the sound driver, the Windows Volume control shows many extra controls like-
Modem Speaker
Aux (Auxillary)
etc.
But the M/B does not have connectors for these in the PCB.Why ar they shown?

Yeah, that should work, as the CD-analogue and TV tuner outputs will be of similar level - you just have to remember that the fader labelled "CD Player" (or whatever) is the TV Tuner level.
As Piioo said, the Volume Control covers all of the audio sources that the audio chip can handle, even though some might not be connected on a particular mobo. You can turn off the unavailable channels in Properties - note that there are separate sets for Recording and Playback.
The Recording mixer is a bit strange, in that it will only allow one source to be active at a time. The benefit of this arrangement is that if you are only recording from one source, you won't get added noise from other sources that are connected but "silent". On the other hand, if you are doing a set of recordings from different devices (e.g. Line In, Midi, CD), you have to keep going back to the Recording Mixer to change the source. If you want to record from a mix of sources, you need to select the Playback mixer as the source - this might be labelled "Stereo Mixer" or "What U Hear" etc.
Cheers

Itunes to Ipod "Sample rate" Transfer Problem

In trying to transfer music files from Itunes 10.1.1 to a Classic Ipod, I get the following message.
"(song name) was not copied because the sample rate is not supported by the iPod (ipod name)
ipod".
How do I fix this. I have 55 music files that were transferred to Itunes from large wave files recorded on a Sony digital recorder.
They transferred fine to Itunes and play great on the Mac pro but can't be transferred to the ipod because
of the sample rate.. I need them on the ipod.
How do I fix this?
Thanks for any advice.

Set your import settings (Preferences, General tab) to a suitable alternate format, e.g. 320kbps mp3 or Apple Lossless. Select the files in question, option-click on them and click *Create <format> version*.
tt2

Why voice control/ Siri is not supported in the middile east?

I am trying to use voice control / Siri to get near restaurants, shops. It keep gives me message says can't find restaurants or any place i asked for in Kuwait.
I hope I can find solution for this. It's disappointed If voice control not supported in Kuwait :(
Anyone had this kind of problems with Siri?

I had problems on one of my pages (and only one) under chrome. First the buttons stopped working...they were supposed to branch to another page in the application but instead the url they went to was 'localhost'/the page so of ourse this didn't work. I flixed this by manually adding the whole url for the button. Also strangely enough from this page/tab only clicking on the other tabs does not work. I have yet to work out how to fix this...luckily i have few chrome users. I am considering posting a message warning people not to use chrome...
maybe this will go away when apex 4.0 comes out.... hopefully before many people use chrome...when is 4.0 coming out? Yes I am holding my breath....
(btw you can see that page if you to to www.schedulemail.net and go to the "options" tab.)
Edited by: Emu on Jun 10, 2009 9:01 AM

Sample rate control not supported

Similar Messages

Maybe you are looking for