BIP Post Audio Quantize - Sample Variance Issues

Dear Forum,
I have Audio Quantized a bass part and looking to bounce it out. Everything I tried introduces small timing and waveform variances. BIP, Convert New Region, Bounce. I am guessing this is due to the Flex feature which is not being incorporated into the bounce?
Would be grateful for a bit of help. PS. Logic 9

Does your tv have an optical audio output?  Hook both audio and video to your tv, then run the optical cable out of your tv to your receiver.  the problem is that the tv takes longer to decode the hdmi video that you receiver does the audio, so then they don't sync up.

Similar Messages

  • MiniDV audio sample rate issue

    I cannot get the audio sample rate to match. I have captured miniDV footage (32mhz/12bit). In Sequence settings the audio options for bit are 8, 16, 24. My audio is out of sync!!!! What can I do? I have 2 hours of raw footage and can't proceed! Any help is VERY appreciated. thanks in advance!

    Hi again!
    Hum ... that´s not as simple as it could be!
    That way i´ll have to capture twice?!
    There must be another way of doing it , i don´t think Avid can do it and FCP don´t .
    As i told you i use both systems and i know they´re limitions (or i think i know!), but it´s strange because Avid Xpress DV 3.5 (from the age of stone ) do it in a blink ... ok ... found an FCP limitation!
    Thank you!
    If there´s another way ... please feel free to post it!
    (and is was supposed to "reply" a "miniDV audio sample rate issue.
    HBars

  • External sound card sample rate issue

    After installing Yosemite I am having problem with Sample Rate settings on my external audio interface. It was working perfectly before the upgrade.
    The issue is that the sample rate on my system is set to 44.1 Khz. When a new application attempts to access the audio card it plays but breaks up. This is indicative of a sample rate issue so I tried to set the rate to 48 Khz and then back to 44.1 Khz. Once I do this the system works properly until another application tried to use the card and I have to repeat the same process over again. While it DOES work it is not correct functionality and there is obviously something wrong.
    At first I thought is was just my system until my friend also commented on it happening to him too.
    I am using an RME Fireface UFX with the latest drivers issued on October 8th of this year.
    My friend is using an M-Audio unit and he is finding the same issue.
    I have already checked everything in Audio / Midi Set-up control and reinstalled the drivers for my audio interface. I also looked for any new settings to Yosemite to see if there was something new that might affect this. No luck and no leads on where to look for the next option.
    Anyone have any ideas?

    Hi,
         The  sound card do not support all sample rates. Which sample rates your sound card support you will find in the the manual.goto help in the toolbar select find examples and search for sound Open the Sound Input to File.vi file.This will give you a template for recording sound.You have to set the sample rate then open a sound file for write.
    Please post a sample VI that you have created.
    also,please go through this:-
    http://www.zeitnitz.de/Christian/waveio
    Thanks as kudos only

  • OS 10.4.10 Update Causing Audio Popping (AND HUM) Issues:

    OS 10.4.10 Update Causing Audio Popping (AND HUM) Issues:
    The problem on my Intel Mac mini with external Klipsch speakers (two mid-range/high and a subwoofer) is REAL obvious as the system emits a hum like a Spinal Tap amplifier turned up to #11 when 10.4.10 powers down the audio circuitry
    Then, if I play a song in iTunes, it first pops and the hum disappears right before playing.
    Quite annoying. I restored the mini from a recent backup to 10.4.9. Problem gone.
    I'm glad I test updates on the mini before applying them to my Intel iMac 24. I guess I'll just hide & watch what happens before doing any more updates.
    Here's what I sent to Apple tech support:
    Guys/Gals (OS X Engineers):
    Concerning the Snap, Crackle Pop Audio problem in the 10.4.10 update:
    Please. Just do the right thing and fix the audio.
    There is no need to power cycle the audio subsystem to save .0005 amps on the laptop line.
    And then to make your Mac mini/iMac/other desktop users have to put up with it when using AC power is ridiculous.
    Put the audio subsystem back like it was in 10.4.9 and let the laptop battery freaks looking for mini-amp savings get a "work-a-round".
    Crikey!
    iMac 2.16 Ghz Intel Core 2 Duo 2GB RAM Mac OS X (10.4.9) 20 Years of Microsoft-Induced Grey Hair
    Mac Mini 1.83 Ghz Intel Core Duo 1GB RAM   Mac OS X (10.4.9)   20 Years of Microsoft-Induced Grey Hair

    Libby,
    Backup is unable to make a restorable by Disk Utility backup. You can restore pieces of the software as outlined in my FAQ about backing up user data:
    http://www.macmaps.com/backup.html
    But to be able to restore a backup with Disk Utility, you need a cloning utility to have made the backup.
    I suggest starting a new thread here, where someone else could help you out better:
    http://discussions.apple.com/forum.jspa?forumID=752&start=0
    First off, you'll get a wider audience who may be able to solve your problem. Secondly, responses to you won't confuse the original poster with solutions that don't apply to them. And thirdly, you'll know for certain whether or not a response applies to you.

  • How does the decoder retrieve the quantized samples in this loop?

    please check the block diagram loop. 
    Attachments:
    Decoder.vi ‏23 KB

    Hi Mahady,
    Looking at the encoder VI you posted I can see that there is a separate input called quantized samples. I am pretty sure the q[k] is an array of possible values that the input to the encoder may hold after being quantized. For each value in to the program that array is searched and the encoded value corresponds to the index position in the array. The output encoded array is optimized to take up as little memory as possible. The information is then sent as a digital waveform and as you say the decoder is the inverse of the encoder.
    I have worked all this out by using the Context Help tool in LabVIEW and Highlight Execution. Both can really help to understand a VI that is not your own.
    Can I ask where you got this VI from?
    Jack. W
    Applications Engineer
    National Instruments

  • Audio capture sample rate

    I have read other posts that deal with the issue audio rate that cause the synch with the picture to be off. I recorded this Dv at sp speed and the camera is a pd 150 that had the audio rate set to 32khz. I haved looked in the settings and there is no 32 khz option and you can't make one by copying the generic template. I remember in version four and prior versions that there was the template for 32khz and now its gone, why? How do you get Final Cut Pro 5 to allow 32khz 2 channel sound?

    First, it is better to record at 48k for a lot of reasons. However, go to your capture presets, duplicate the appropriate preset and look at the quicktime audio format options. You should be able to change it to 32k. I can.

  • 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.

  • PI Interface Posting Files - Trigger Process Chain Issue

    Dear Reader,
    Situation -
    We get multiple flat files from source system via PI interface. To process this in BW 7.0 side we have created the web service interface. In the function module we have written a code to trigger process chain, once a data is posted.
    Issue -
    As there are multiple files being posted, the PC runs on completion of the every single post, which is not desired. We need to run the process chain only once at the end of all the files being posted.
    Notes -
    1. Number of files keep varying.
    2. Clubbing all the files in a single file and then posting it would cause performance issues.
    Request your help in find a way like -
    1. A file being posted of name, say 'START PC', which can be trapped in the funciton module and controll the PC call.
    2. <any Other idea>
    regards,
    vinay gupta

    Hi Dhanya,
    This is the code i have in the ABAP program in the process chain. I just included the API_SEMBPS_POST part, but still it doesn't work. Please give me your email address so that i can send some screenshots.
    REPORT  ZHTEST.
    DATA: l_subrc TYPE sy-subrc.
    DATA: ls_return TYPE bapiret2.
    CALL FUNCTION 'API_SEMBPS_POST'
    IMPORTING
       E_SUBRC         = l_subrc
       ES_RETURN       = ls_return.
    CALL FUNCTION 'RSAPO_CLOSE_TRANS_REQUEST'
      EXPORTING
        I_INFOCUBE               = 'ZMAP_TAB'
    EXCEPTIONS
      ILLEGAL_INPUT            = 1
      REQUEST_NOT_CLOSED       = 2
      INHERITED_ERROR          = 3
      OTHERS                   = 4
    IF SY-SUBRC <> 0.
    MESSAGE ID SY-MSGID TYPE SY-MSGTY NUMBER SY-MSGNO
            WITH SY-MSGV1 SY-MSGV2 SY-MSGV3 SY-MSGV4.
    ENDIF.

  • Audio pitch / playback speed issue, MacBook Pro 10.4.10

    I experienced a very strange audio pitch and speed issue that affected a recording I was making in Ableton Live 6.0.3.
    Setup is MacBook Pro 2.16GHz, Mackie Onyx Satellite FW interface for audio, recording straight to internal HD.
    I made two multitrack recordings (3-4 tracks each) which played back fine initially within Live. I then rendered the audio to AIFF in order to burn data and audio CDs.
    Testing the playback in iTunes, the audio was strangely pitched lower than what we originally recorded. Checking other tracks from my iTunes library, they were playing back FASTER or higher-pitched than normal, and so did audio streamed from MySpace that I was familiar with.
    CDs burnt from these AIFFs exhibited the same "lower pitch" / slower playback.
    Reopening the Live Set, they were now also playing back lower pitched than what we had heard just minutes beforehand during the recording / mixing.
    After I let the computer sleep for a few hours, I came back and checked -- system and internet audio was now playing back at normal speed, but the Live Set was still lower-pitched - the individual WAV files in the set were all pitched lower.
    The Live recording prefs were stock standard - set to record 24 bit WAV at 44.1Khz. Since the issue seemed to affect non-Live audio as well, I'm guessing it was something more at the CoreAudio level?
    I managed to salvage the session by taking the exported AIFFs into Tracktion 3 and using the pitch change function in the Clip Properties pane. One track was 1.5x off, the other was 1.125x off.
    I'm at a complete loss as to why this would happen. It's a first (and hopefully only) occurrence. Has anyone had this happen to them? Could this be some sort of intermittent hardware (clock) glitch? Something to do with 6.0.3 and OS 10.4.10?
    I do need to take this MacBook Pro in for servicing - it experienced the "bulging battery" issue and has a partially dead keyboard (the keys near the upper right) - but even with those issues I've not had audio issues like this.

    In any case, if my Mac had problems, I'd send it in for service....!!!
    About your problem, I don't think that it depends on hardware.
    I'd check Live configuration and the SR of the audio files: you know, Live has so many features to modify speed and a lot of other parameters that bacomes easy to forget something.
    Try recording something with Garage Band so you will be sure if there is a real problem.
    cheers
    rob

  • Problem Posting Receipt From Production and Issue for Production

    Hi Everyone,
    I am posting Receipt from Production and Issue for Production through DIAPI in SAP Business One 2007A SP00 PL03.
    I am using following lines of code
    oDocument.Lines.SetCurrentLine(0);
    oDocument.Lines.BaseEntry = int.Parse(BaseEntry);
    oDocument.Lines.Quantity = double.Parse(Quantity);
    lRetCode = oDocument.Add();
    The Document adds perfectly but when I open the document in SAP Business One i am not able to see the Order Number at Line level on which this document was Based.
    I identified the problem being Order Number Column being binded to BaseRef field hence form is not able to Show the Order Number and we cannot set BaseRef via DIAPI since the property to set that field is not exposed.
    I donot want to update the table through a Recordset Update Query. I would appreciate if the property to set that field is exposed by SAP or any other workaround can be put forward by anyone who has faced the same problem.
    Thanks
    Kapil

    Do you need to set
    oDocument.Lines.BaseType = 17 (if this is sales order)
    and then baseentry = baseentry....etc
    If you bind the base entry correctly it should alwasy shows the base document on both database as well as front end. I have never had any problem with this
    Sincerely

  • Problem recording audio from sampler...

    Problem recording audio from sampler (MPC 1000) to Logic using midi trigger thru Lexicon Omega box.
    When I hit the record button in Logic, it triggers my mpc to start playing. Logic then picks up the incoming audio signal on the appropriate track and starts recording the audio... PROBLEM- Logic appears to keep restarting the incoming audio, repeating the first 1/4 of the first 1Bar. So every time I hear the metronome tempo counter as it records, the incoming audio replays from the beginning and this is what gets recorded. On the contrary when I'm controlling the sampler via midi and just hit the play button in Logic, the audio signal comes threw and plays the full 4 bars of the audio that I want recorded.
    Any idea on how I can fix this? It seems like it has to do with my metronome counter or midi time code?
    Thanks to all that help!

    Thanks, good suggestion but these are music instruction dvds that aren't copyrighted.
    cheers
    Eric

  • 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,

  • TIMECODES are important for audio and video sync issues

    Hello, just wanted to pass on what I learned so that others can avoid the trouble that I've had to go through. Perhaps this may help someone who is stuck on the launch pad. :)
    BOTTOM LINE: Info for the beginner. Audio and video not in sync in Premiere Pro CS3 V3.2.0
    PROBLEM: Capture works great it seems. When I go to the folder that contains the captured file and view in Windows Media Player audio and video are in sync. BUT when viewing the video asset in the source and program monitors, the audio and video are not in sync.
    SOLUTION:
    Before capturing a tape make certain the following is checked:
    (1) Edit->Preferences->Capture->Use device control timecode
    (2) Edit->Preferences->Device Control->Options->Timecode Format
    (3) Project->Project Settings->General->Video->Display format
    As for the devices timecode choose something other than Auto Detect. Then match the project timecode with what was chosen during capture. The projects display format could of course be set to frames.
    I searched everywhere for audio and video sync issues in google, adobe forums, F1 help, and hv20.com and everyone was talking about:
    (1) Presets: 1080p30 vs. 1080i30 (60i).
    (2) brakes in the tape where timecode for the audio and video get misaligned during capture.
    But choosing the correct hardware settings and timecodes to solve audio and video sync issues never popped up.
    MY HARDWARE: Canon HV30, HDV

    >Audio and video not in sync in Premiere Pro CS3 V3.2.0
    Must be an HDV only issue because my synch is always perfect.

  • Getting best video for my Post Audio

    Another amateur project question. I'm just doing post audio for the movie. The director/editor has DVCPRO HD footage at 29.97. I think I should use DV for my Pro Tools session.
    How do you recommend the editor get the video from FCP to a DV file? Duplicate the FCP sequence and change settings then export? or Export a full-res DVCPRO HD file and use Compressor to change? Export from FCP using Compressor or QT Conversion?
    Thanks.

    I'd make a DV copy of the exported master, that is, there should be no question that you got a copy of the exported video. Depending on your team's penchant for blaming each other for screwing up, you should have timecode and a supered version identifier burned into the video you are using.
    If you export from the timeline, changes made to the timeline will not be reflected in anyone's copy. That is good. Everyone at every stage gets a coy , in whatever format they want, of the exported movie, not their own exports made from the timeline.
    Your production team must come up with a version numbering system if you have not already. It is critical that everyone have copies of the same exported version and immediately be certain they know what they're looking at.
    bogiesan

  • LogicX - Maverick - Sample Rate issue with Audio Interface

    As a test prior to updating my Mac Pro and iMac, I loaded ML so I could test out LogicX.  I've been running Logic 9 either using the built in audio or a Line6 guitar Port.  The MBP is a late 2009, non retina, 13 in with 4 GB Ram.  As compared to my iMac, the laptop can barely play back my tracks which are mostly software instruments (I don't record audio until the midi arrangements are finalized). 
    So, I updated to ML to try out X and of course, had to deal with the 32 bit plugin issue.  Luckily, 32 Lives and Vienna helped out with 90% of my plugins.
    Now, I updated to MAV and had to go through an arduous process of re validating the AU plugins for both Logic 9 and X.
    After doing so, I loaded up a song that did have a single bass audio track.  I updated the Guitar Port driver from Line 6 to the latest driver.  The song will start, but then I get a stuttering and an error message about running the audio at a weird sample rate, eg 28654.  I tried using the built in audio and same issue.
    I opened the song in Logic 9 and it works.
    Any suggestions?  Do you think one of the resurrected plugins is causing this problem?  Everything worked OK in ML though.
    Thanks and appreciate any help.
    Todd
    trademarkauthority

    Let me start over ... the typical audio setting that FCP works with is 48khz, 16 bit stereo. If you check the sequence settings, it will probably show that to be the case unless you've created a sequence with other settings. The audio of any clip you bring into FCP really needs to match the sequence settings. To check the sequence settings go to Sequence->Settings.
    So far, you've confirmed the clip is 48khz sampling rate but you didn't say what it's bit depth was. In FCP, right click on the clip and select "Item Properties." That will tell you the information I've been asking about.
    -DH

Maybe you are looking for