How to render audio with sample rate 48000hz using jmf

hi,
In my application i need to play the audio with jmf player with sampling rate 48000hz. but i found that jmf player plays the audio with sampling rate of 44100hz only.but my application needs to play the audio with sampling rate of 48000hz .please help me how to do this using jmf .
thanks in advance,

hi,
In my application i need to play the audio with jmf player with sampling rate 48000hz. but i found that jmf player plays the audio with sampling rate of 44100hz only.but my application needs to play the audio with sampling rate of 48000hz .please help me how to do this using jmf .
thanks in advance,

Similar Messages

  • 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?

  • DIgital audio in sample rates, 441 48 96 but no 882... is there a way to ad

    When feeding my macbookpro a digital audio signal via optical cable while viewing the audio midi setup utility, it shows that the mbp's internal clock automatically switches to the incoming sample rate. I got it to work with 441,48 and 96k. When I fead it the 882 sample rate it automatically switches back to the mbp's internal clock and reverts to line input instead of digital. So the audio midi setup utility shows when it is in digital input mode the three sample rates 441,48 and 96k in the format pop up window and not 882 so it must be just a software issue and I am wondering if there is a way to add the sample rate to the list. If it can take in 96k it should also be ale to take in 882. So does anyone know how you could ad a sample rate of 882 to the list in the audio midi utility format list?

    One added note. The reason that I want to be able to do this is I like to record my audio projects at 882 and if the mac book pro could except the sample rate I could then use it as a 882 digital 2 track recorder. Enabling me to skip hitting the bounce to disk function in pro tools and actually be able to ride a fader as the mix is printing to the macbookpro, where as when you bounce to disk in pro tools you cannot access the mix as it bounces to disk.

  • Problems with sample rate

    hello:
    I have a sound blaster audigy platinum ex and I'm trying to make some recordings with adobe audition, but this program works at 44khz and the sound card seems to work only at 48khz, so it doesn't let me recording. how can I change the sampleing rate of the card?
    thanks.

    By setting it in Audition audio settings. W/ ASIO device drivers you can have 16-bit/48kHz (and IIRC, 24-bit/96kHz) mode(s) only). For other modes you need to use other device driver mode. Also, Asio4All should handle all modes (16/24-bit and 44.1, 48 and 96kHz).
    - http://www.asio4all.com
    The quality in 44.1kHz mode is POOR for your card. If you switch to 24-bit/44.1kHz quality is otherwise better but the frequency gets cutted already around 15kHz.
    You get the best quality by setting the sample rate to 96kHz (16 and 24 bit resolution are ~same in these cards) for recordings (for project?) and then when export the final mix just use SRC to CD quality (Audition do have quite good SRC quality but, if you use additional software for SRC (like Voxengo r8brain (free)), export the final mix using 96kHz and then execute SRC).
    Something more on Creative ASIO -
    http://forums.creative.com/creativel...essage.id=1726
    jutapa
    Message Edited by jutapa on 11-09-200602:43 PM

  • Core audio and sample rate conversion

    I would like to know how to take manual control over core audio regarding sample rate destruct... er conversion. First - I know the workarounds - simply closing the audio player of choice, resetting the external hardware and relaunching the audio player of choice.
    setup:
    I run external converters with an external sample rate clock source. No problems or issues here. I keep my music collection segregated by SR (96k, 88.2k, 48k, 44.1k) as I ALWAYS listen through external converters.
    The annoyance is when one forgets to keep track of core audio and inadvertently ends up listen to a piece of music sample rate destructed. You know - walk up to music server computer, forget that you had DAC set to 44.1k for last music played, put on 96k source, SRC takes over and you don't notice the artifacts and distortion for a few songs. No one wants that! Don't get me wrong - it's a convenient feature and the amateur user would be sunk without it.
    What I want is an indicator that will tell when SRC is turned on and further, what the input and output sample rates are (as far as core audio is able to determine from the hardware anyway). In my world this would have been a check box in a preference setting. Perhaps someone has written a script or app for this? Command line instruction?
    Thanks

    Start with http://developer.apple.com/documentation/MusicAudio/Conceptual/CoreAudioOverview /Introduction/Introduction.html and direct further queries to the developer forums under OS X Technologies.

  • How i can deal with oracle file by using php api

    how I can deal with oracle file by using php api ?

    What has this to do with Reflections and Reference Objects?

  • How do I text with the iPad 2 using cellular device?

    How do I text with the iPad 2 using cellular device? I also have iPhone 4, can I use them both for texting?

    The iPad doesn't have any text over cellular capabilities. You can use apps like Text Free to send texts to non-iOS devices, though.

  • Problems with audio Parallels sample rate

    Hi!
    I recently I installed Finale 2014 in my iMac (Os X 10.9.2) and, when I try to run the application, I receive an error message telling that I must adjust my audio sample rate to 44.1 kHz and Finale crashes. So, by going to Applications --> Utillities --> Audio and MIDI Setup, I adjusted the sample rate to this value and the issue continues to appear. By contacting Finale support, they told me that I must remove my audio device from Parallels Desktop 9 (I was not allowed to adjust this device). But, the problem is that I simply can not remove the device - when I select the device, the " - " signal is in gray. How can I remove the device, in this case?
    Thank you!

    Hello!, first of all thanks for your reply. Yes, the camera is the same, the problem is on my computer, I have discovered a solution: first I capture the audio, only the audio, and then the video, only the video. After this I soncronize the two separate files on Final Cut timeline. It´s a slow process, but I am really desesperated. I don´t know what happens, I few minutes ago I have captured a 30 mins. video and the sample rate of the audio was 42.145k (???). It´s a crazy thing...

  • How to sync. 50Hz sample rate with 5Hz external digital trigger signal?

    Hello there,
    I want to sample an analog input at 50Hz, where I further want to use
    the 5Hz PPS signal of a GPS receiver as an external trigger/clock
    signal for the NI DAQ USB-6008 device. However, I do not want to read
    10 samples every 200ms (= 50Hz avg. sample rate), but I want to read 1
    sample every 20ms, i.e. with "real" 50Hz.
    The "natural" way, it seems would be to use a task that waits for a
    rising edge on the ext. trigger input, then reads 10 samples, then
    waits for the next trigger edge, and so on.
    However, it seems that with this approach I cannot read the data from
    the DAQ with real 50Hz, further it seems that every 10 samples the task
    needs to be restarted, which takes longer than the 20ms (corresponding
    to the 50Hz target sample rate).
    I hope the problem description is clear enough....
    Thank you in advance for any advice.
    Regards,
    Oliver

    You have to use the counter together with the analog input to achive this. You could configure the counter to generate 10 pulses with 50 Hz whenever the counter sees a rising edge on the inpput of the counter. The output of the counter would be used as the external clock for the analog input.

  • How to force digital input sample rate?

    Does anyone know how to force a Digital Input sample rate and/or tell Core Audio which conversion clock to use... internal or external? Apparently both things can be done from Pro Tools, but I don't have it.  My problem is locking onto an A/D converter's 48K Toslink output.  I can set 48K in 'Audio - Midi' setup and the audio is perfect, but a few seconds later  Core Audio returns to the default 44.1K.  I need to force it to stay at 48K, or clock on the signal rather than internal clock.

    Braver,
    Let me clarify a few things. First, the 6733 supports correlated digital I/O. This means that the digital operation does not have its own clock. You can correlate the digital operation with the AO clock, which will allow a digital operation up to the max AO rate. You can generate a pulse train with an onboard counter and use that as your digital clock source. Also, you can use a completely external clock source to achieve the max digital rate of 10MHz. The knowledge base, linked below, points you to a couple of examples that ship with LabVIEW. Be sure and read the documentation associated with each example.
    Knowlege Base
    As for
    the example CDI Single Point clocked by Counter (E).vi, notice how the output of...

  • 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

  • Q: Convert audio bit & sample rate during import

    I have a project that's in 24bit/48KHz, and want to import an audio file (mp3) that's in 16bit/44.1KHz.
    When I import the mp3 file into the project either through drag & drop or the Add Audio File dialog box, it doesn't convert the file into the project's native bit & sample rate. Consequently, when I play it back in the Arrange window it plays in chipmunk voice.
    I thought Logic used to automatically convert audio files into native resolution. How do I do this in Logic 8?
    Thanks,
    Steven

    Thanks for the response. My project currently does have those options checked, and it's still not converting.
    To double check, the mp3 file does get copied to the project's Audio Files directory. It also creates a file with the same name and the file extension .ovw - but doesn't seem to be converted.
    Any ideas? Thanks again -

  • Blue lines through audio, NOT sampling rate

    Hello,
    A friend is editing some audio, using a lot of tracks (around twenty) and shifting the speed. For some reason audio clips that were slowed to 80 percent are appearing with a blue line through them and playing at a very low volume. I checked the sampling rate and it's 48kHz. Also frustrating is that these playback issues are intermittent.
    Any suggestions would be great.
    thanks,
    jesse

    He probably doesn't have the CPU horsepower to render twenty audio tracks on the fly. In User Preferences, you can do a few things to reduce the need to render audio:
    a: increase the number of tracks for 'Realtime Playback' - the default is 8 I think, and it can usually handle more;
    b. reduce the audio quality to low - it really doesn't sound bad at all, and it only affects your preview audio;

  • How do i ensure my sampling rate is a constant and correct (@250Hz) when using AI Sample Channel vi

    I am running a VI that samples 2 channels using the AI Sample Channel vi within a for loop (executing equal to the number of samples i require), i then calculate the difference between the channels. Within the for loop i use a chart to displays this difference in realtime.
    On exiting the loop the data is converted to an array, which in turn i then convert to a waveform (using dT as 0.01 as an arbitary solution at the moment). The waveform is then compared to limits using the limit specs and limit testing vi's.
    There is also other code displaying graphs from the previous iterations of the VI.
    My question is how do i control the aquisition rat
    e so that i know that the AI Sample Channel vi is sampling my data at a set rate (250Hz)?
    I have tried to use some of the hardware timed exaples supplied by NI to no avail. They can't give me the single point output required within the initial for loop for the real-time display.
    Any solutions welcome!!

    How about buffered acquisition? You can let your DAQCard acquire the number of samples you require instead of doing it with a FOR loop. Using a FOR loop means that you are software timed, which may as well be untimed since it's about as deterministic as the weather in Florida.
    Look for the "Acquire N Scans.vi" example.
    If you want to do this the RIGHT way, use continuous acquisition. Start your acquisition and keep doing AI Read in a loop. Also include your porcessing and display functions in the loop. Just make sure your loop runs fast enough to keep up with your acquisition. At 250 Hz, you should have no problem.
    Dan Press
    www.primetest.com

  • Question on supported audio input sampling rate

    Hi,
    I have USB capture device which supports HDMI input and I wish to use it to connect to my digicam and stream it out with FMLE. While I can see that the video input is working properly, the audio is not working. The error message "Problem with audio capture device - Please verify that the audio device is working correctly and is not already in use." always appear when I try to select the device's audio device. Using the graphedit, I can see that the capture device's audio output pin only supports stereo 16-bit 48 kHz PCM output. As such, can I confirm that that FMLE only support 44.1kHz (as well as other 44.1kHz divisible) sampling rate?
    Thanks
    John

    After researching and playing with the system,  I found a solution.  Uninstalling the "Realtek High Definition Audio Driver" under the Control Panel -> Hardware and Sound enables sampling rate up to 384KHz  instead of 48KHz.
    If there are other solutions without uninstalling the HD audio driver,  please post.  Thanks
    Eddie 

Maybe you are looking for

  • Windows reboots immediately when I start iTunes

    The Problem: When I click on the iTunes icon to start up the software, my computer restarts IMMEDIATELY. There is no error message, the computer does not freeze, and there is no indication that an iTunes window opened at all, even briefly. The screen

  • How to trun off batch determination for certain shipping points ?

    During Delivery Notes creation we would like to turn off the batch determination on basis of certail shipping points. can some please guide what is the best way to do it. We do not want it turned off for all the shipping points but for a special ones

  • ITunes 12.1.0.7 64-bit not syncing with Outlook 2013 Calendar

    After updating to iTunes 12.1.0.7, Outlook 2013 (on Windows 7) is no longer syncing (via USB).  It was working fine before.  I'm using an iPhone 6  on iOS 8.1.3.  iTunes doesn't indicate any errors, it just plain doesn't work.  Thoughts?

  • [SOLVED] systemd/cron as user: timer doesn't start after boot

    Hello, I've tried to follow the wiki's systemd/cron functionality article to have a systemd service to run hourly. Everything seems to be working ok, except that the hourly timer doesn't start automatically : I have to issue a systemctl --user timer-

  • Why can't I downloaded or update apps anymore?

    Ever since March 17th, 2012, I have not been able to update or download apps. I've already lost 4 apps now! Pandora, Billminder, Movies by Flixster and Dental Expert refused to update. I couldn't use them because they were "Waiting". I deleted them t