Audio recording setting

I do not hear my voice when I talk after the recoding It says I have to check my audio recording setting. Where do I go?

seraj.ali6 wrote:
I do not hear my voice when I talk after the recoding It says I have to check my audio recording setting. Where do I go?

Similar Messages

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    Are you are using an audio interface?
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    I don't like that approach for adult learners, cannot refrain from mentioning that.
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    For generic performance and trouble-shooting advice I’d like to recommend reading these:
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    Hi again
    Here is a link to an article about uninstalling.
    Click
    here to view
    And here is a link for obtaining an older version. Perhaps 9.
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    here to view
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  • Spinning wheel when audio recording

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  • Lapel Microphone with a Digital Audio Recorder?

    I am looking for a device that would look a lot like the microphone and transmitter portion of a wireless microphone system.
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    For wedding videography I put a traditional wireless lapel microphone on the groom and connect the receiver to my camcorder.
    Sometime there is noise and/or cutouts in the audio from this system.
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    All this has to be stuffed into the groom's pocket.
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  • Audio recording redirection in Windows 2008 R2

    Hi, i try to setting up audio recording redirection with no success. I use Windows 2008 R2 with Remote Desktop Services role installed and Windows Seven as client computer. I edit local group policy on Windows 2008 R2. Computer settings - Administrative Templates - Windows Components - Remote Desktop Services - Remote Desktop Session Host - Device and resource redirection:
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    With this setting i can hear audio on my local computer, but no recording devices in sound properties. Anybody know how to troubleshooting this behavior?
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    Hello Pavel,
    Thanks for your feedback.
    Regarding the sound latency issue, I agree with TP on the suggestions to troubleshoot it via hardware/driver level. Based on your feedback, this issue doesn’t appear after rebooting the terminal server. Please let us know if it happens again.
    Regarding your new question, I’d like to share the some assessment based on my experience.
    After the Remote Desktop Services role is removed from the server, the remote desktop connection will start administrative sessions to access the server. In such a case, redirections audio recording redirection is not supported any more, but Audio Playback redirection is still supported. Therefore, you can still get the sound redirection provided it is configured so. therefore, it is a normal behavior.
    Please feel free to let us know if you need any further assistance. Thanks.
    Lionel Chen
    TechNet Subscriber Support in forum
    If you have any feedback on our support, please contact [email protected]

  • Audio recorded from lavalier mic is not audible when printing to tape

    hello!
    i am very confused : ( i have an edited work with some audio recorded directly from the built-in camera mic, and most of the audio from a lavalier (lapel) mic. all the audio plays fine from the timeline, and from the "print to video" play back option, when the video playback is set to "none" or "preview"; however, when it is set to ntsc (which i believe i need in order to print it back onto a mini DV tape to submit it) only the audio recorded with the built-in camera mic is clear and audible. the other recorded sounds from the lav mic are very quiet (or silent) and have static noises.
    i have exported the timeline and burnt it to a dvd, and all the audio works fine. but i can not seem to print it to a tape without losing most of the audio.
    i have tried all different combinations of settings (audio to built in audio, or to firewire), etc etc and nothing seems to fix this problem.
    any suggestions would be GREATLY appreciated! thanks : )

    You might try highlighting all the audio tracks
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    Also, I'm not in front of the interface right now,
    but if you go to FINAL CUT PRO drop down - AUDIO / VIDEO
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    Message was edited by: brvideo

  • Audio, recorded separately, strays from sync

    Hello,
    I have three sources of media that I'm using in my project, which is a recording of a gig that my band played:
    - Sony AVCHD video camera (front view), includes one video and a stereo pair
    - Sony AVCHD video camera (side view), includes one video and a stereo pair
    - three tracks of audio that were recorded using a separate multitrack audio recorder (Zoom R16).
    I've figured out how to sync the sources together - I set an in point on a distinct piece of audio that exists in all three sources (actually it's someone coughing, I didn't think to have the drummer clap his sticks together before we started). Then I set an in point in the sequence, edited in the three sources, and adjusted the in points of the clips forward to the beginning of the set. At that point where the clips were matched up, the sources are perfectly in sync.
    However, within 30 seconds before or after the sync point, some drift is evident in the audio that was captured on the R16. And a few minutes out, it's seconds off. I capture these gigs one set at a time, so we're talking about clips that are an hour long (we usually do two or three sets).
    I didn't pay much attention to the detailed settings of the audio recorder before I captured the gig. Actually, I recall that the basic marketing/quickstart tutorials on the Apple website say you can pretty much use any video or audio in a single project in FCE and it would convert/transcode for you as necessary.
    Had a look at the clip properties, and I see that the audio captured in the cameras is 48.0 KHz, while the R16 captured the audio at 44.1 KHz. If I'd noticed the difference beforehand I would have changed the R16's settings to match, just to be thorough - but I am a little disappointed - I'd thought that there would be compensation for differences like these... this is digital!
    Could this difference in the sample rates be the problem? Regardless, can anyone suggest how I might be able to resample or convert the audio so that it doesn't drift like this?
    Thanks much in advance, and cheers.

    fudster wrote:
    RyanManUtd wrote:
    Hmmm ... honestly .... I am running out of ideas ...
    Can I ask:
    1. did you get both their starting point correct ?
    2. Are the duration of both the video clips & audio clips you are trying to sync the same length ? That is, if the audio is 10 secs, then is the video also 10 secs ?
    Thanks
    Hey -
    1. Yes, I'm certain that I matched up the in points correctly while making the edits. In the audio tracks, I zoomed in all the way so I could get sub-frame position - as described starting on pg 474 of the manual. At that point, they are in perfect sync. And as you get further away from that matched point in the sequence, the drift gets worse.
    2. The length of the audio that was recorded on the cameras is exactly the same length as the video (of course). The audio clips recorded on the R16 are all exactly the same length as each other as well.
    However the R16 clips are not the same length as the video clips, and the two videos are not the same length as each other. They are a few seconds different from each other, because of the way I start the recording - I step onto the stage and before we start playing, I use the infrared remote to start the recording on each of the cameras one at a time, and then I start the recording on the R16.
    I don't actually know how exact clip lengths from separate recording sources could be achieved, and since I'm editing them in separately, I think it shouldn't make a different. The keyword being "shouldn't", of course. In any case, if you extend the clips all the way out to play all the available media in the track (using up all the handles), the extents of the clips in the tracks appear a few seconds different from each other, as they should.
    Thanks for your suggestions, I'll keep trying to come up with something. Feel free to suggest other things you think of, I do appreciate your time and help.
    I have run out of ideas ... Anyone here can help ?

  • Audio recording - prompts after every slide?

    This question concerns the prompt box that appears after
    recording audio for a slide. Basically, if the audio track is
    longer than the slide length, you are prompted on how to handle
    this after each slide. Is there a universal setting for this so
    that we are not prompted after each slide? Or do you have to
    increase the slide length before each audio recording to avoid this
    issue? Either way, this is a time constraint. Thanks in advance for
    any insight you may offer on these topics.

    Hi stunamistyle
    There is no setting I'm aware of for this. What you see is
    what you get.
    Cheers... Rick

  • Audio recording issues

    Hi,
    For language teaching purposes, we've developed a flash media
    server based app which allows users to record audio. It works well,
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    itself so I'm thinking there might be a setting to change or
    something we might not be doing well.
    thanks in advance for any help!!
    Emmanuel

    Sounds like you're listening to the same source (the X6) being played twice.
    Could be you are hearing  MIDI thru playback as well as Logic's "Software Monitoring" you must also be enabling a MIDI track for recording. If you unhook the MIDI connection from the X6 does the hollow sound go away.
    What Audio interface are you using? The other possibility is your Audio Interface has "Direct Monitoring" this allows you to monitor the audio inputs before it goes into the computer. Having Logic's Software Monitoring enable along with Direct Monitoring would also produce the hollow sound.
    I suggest the thorough reading of the manual so you understand what all the preferences do. Most of the time it is better to read and understand/experiment on your own. It's one thing to be shown something another to figure it out for yourself.
    Here's a Link to all of the Preferences/Audio settings.
    http://help.apple.com/logicpro/mac/10/#lgcpbb81aca5

  • Audio recording (wav) isn't streamable?

    Hi dear all,
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    Am I doing something wrong or that's the fact?
    I read from Sun (See below) wav audio format is not streamable since some fields in the header can only be filled when finish recording.
    If I want to have the realtime (nearly) audio, what should I do?
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    An example of a non-streamable format would be a WAVE audio format or a
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        rc = (RecordControl)p.getControl("RecordControl");
         rc.setRecordStream(outputStream);
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         try {
              Thread.currentThread().sleep(5000);
         } catch (InterruptedException ie) {
    } catch (IOException ioe) {
         ioe.printStackTrace();
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    //stopRecorder();Any suggestions, hint will be appreciated.

    unfortunly
    I decomplied the file RecordCtrl (the implementation of the RecordControl control ) to check this case too
    i found that the class buffers all the data before it begins to send any single byte.
    the only solution i find are :
    1) to do a series of startRecord() / commit() calls on a small time base
    2) to find some way to override the method
          public void record(byte buf[], int offset, int length)
    to have only this line
    {stream.write(buff, offset, length);}
    instead of all the lines in it
    beside making your own OutputStream class that accepts the coming data....
    well
    the problem is
    how to override that method !!!!!!!!!!!!!!!!!!
    i am in this problem nowadays
    can any body help me regarding this ?!!!
    thank you in advance

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