Distorted wave file (region)

I was wondering if someone could help me with this:
I have a great vocal track that I transferred analog from my 1680 to Logic express. The only problem is that only one syllable throughout the whole region is distorted. I tried using the automated volume rides in this section but it is too noticeable. Are there any tricks in the editing functions of Logic Express that could resolve this issue? I would like to keep this stellar vocal track and not have to punch in for one word...any help is much appreciated.
thanks
G5 Mac OS X (10.4.4)

Is the word repeated somewhere else in the song with roughly the same phrasing? If so, some judicious editing should fix it, even if it's a different note - you should be able to pitch correct it. Maybe even the syllable…
Hard to make realistic suggestions without hearing it, but distortion/clipping is really hard as you cant really correct the source material. Editing is probably your best bet.

Similar Messages

Wav file distortion problem

When putting radio programs together, I'm adding .wav files to .mp3 files in the multitrack view in Audition 3.0 and they sound fine to start with. When I export the file to mp3 and listen to it in the edit view the .wav spots that I added sound distorted, as if there are 2 or 3 spots playing on top of each other. I go back to Multitrack view and they also now sound distorted. The other thing that happens is that sometimes once I export something of say 10mins, it only saves the first 20 seconds or so! It doesn't happen every time but it is quite often. If I go to another computer it seems to work fine and nobody else I know seems to get this problem which makes me think it must be my laptop. Any idea what is causing these issues? Maybe my Audio hardware or sample rate? Please help! Thanks.

jonnykibbs wrote:
To be honest I'm not sure what the levels are,
Surely you can see on Audition's meters what the signal is peaking at? If levels are peaking very near to '0' then when you encode to a lossy format like .mp3 you can generate levels above '0' and which lead to clipping and distortion.

Distortion after converting wav file to mp3

Hello,
I am working with a .wav file of 8k sample rate, 16-bit, mono of a Skype conference call that was recorded through the computer. The wav is file is Microsoft ACM: Microsoft ADPCM. The audio quality of the original is not great, as you can imagine, but when I make my edits and convert to mp3 it sounds horrible. I don't know the technical term for it, but it sounds muffled and distorted--the effect you get when you cup your hand in front of your mouth and move it around while talking. I'm at a loss for what to do. Is there some setting I need to change before converting? I know it's a low sample rate (I usually work with 44.1K), would that contribute to the problem?
Any help is greatly appreciated!
Thank you.

In a quick test, I was able to get significantly better results with an 8K, 16-bit source converting to MP3 when changing the MP3 setting to 16Khz. In the File > Save As dialog, after selecting MP3, click the "Change" button next to "Sample Type" and change the sample rate there. The file size was practically unchanged, but the audio quality was increased and looking at both files in spectral view showed a definite improvement.
Edit: Oops, I responded via e-mail and didn't realize this was posted to the Audition 3 forum. I suspect the results will still be better with the MP3 encoder in Au3 - both are by Fraunhofer. Here were the differences I saw when writing to 8K vs. 16K. The resulting file was 29KB in both cases.
MP3 saved at 8K Sample Rate setting
MP3 saved at 16K Sample Rate setting.

Since 10.0.1 dropping wave files in the arranger leads to a empty region.

Logic X 10.0.0. had no problems with the directly dropping .wav files from the finder in the arrangement area.
since upgrade to 10.0.1 this leads to a event with just a bit noise in it.
When I import it via the menu function, the same process leads to a functional event with the wave file.
Am I the only one?

Hmmm... I can't reproduce the issue you have... Works normal here with LPX 10.0.1

Cannot hear imported wav files in the output

Hi, I am using Macromedia Captivate. Version 1.01.1447
I have imported a wav file and when I am playing the slide containing the audio file everything is fine, in particular I hear my audio file. However, when going to the preview of the entire slideshow (or just 5 slides), or if I preview the show in the browser, I do not hear anything. Also, when exporting the movie as a standalone file, I don't hear anything.
Does anyone know a solution for this issue?
Thanks!

If you can get to the instrument editor, drag and drop your regions or files onto the keyboard and answer the following questions. To preview it have an EXS24 loaded up with EXS Instrument #5678 (or whichever number it happens to be.

ITunes update and WAV files

I recently updated my iTunes and since then my WAV files playback very distorted. Not being a mac boffin I tried to retrieve my old iTunes through my timemachine. This told me that I didn't have enough privileges with iTunes to carry out the function.....now I can't access my iTunes at all. How can I sort this all out?

VictorSMiller wrote:
I have a number of WAV files on my PC (from an Olympus digital recorder). Before iTunes version 8, the played just fine. Starting with version 8, they all play extremely noisy and choppy. I've tried playing selected files on Quicktime and they all play just fine. Clearly some sort of bug was introduced in version 8. Has anyone else encountered this?
Victor
Yes, that problem with WAVs from voice recorders has been reported here several times. Haven't seen anything about a fix. Converting the files to MP3 will probably help.

Exporting WAV Files with looping

Hi
I have a collection of sound effects that I've prepared in Logic. I've been exporting them to WAV files for use by another application. I do this with File->Export->Region as Audio File. This is all good.
The prolem I have is that I'd like some of them to loop. So I've been using a PC and Sound Forge to add the WAV sustain information to the WAV file. I'd really like to keep my workflow on the Mac and in Logic.
So, my question: is there a Mac app that can add the WAV sustain or can Logic export it? (I've tried Sound Studio and Audacity already with no success).
cheers
peter

Even though Logic has a 'write loop to file' function, I often find that the loop info either doesn't stick, or is invisible to other apps.
Loops added in DSPQuattro seem to always work, and probably with Peak too (I don't have Peak so I can't check.)
But you can get SoundHack for free and that will also write readable loop points to files.
No Logic user should be without it.
It does so many useful things you can't usually do in DAW apps like Logic.
Get it here:
http://www.soundhack.com/freeware.php

Editing 1.3 Gigabyte Wave File in GarageBand

I have a 1.32 GB wave file that I want to edit in GarageBand. But it seems to be to big for Garageband. It is a speach that is recorded with 44.1 kHZ sample rate and 24 bits. It is about 70 minutes of recording.
I imported the file and it plays, but when trying to insert a split, the part that should become region 2, disappears from the track, and leaves only region 1 in the track.
I would like to edit the file and use parts of it in a CD project and other parts in a DVD project as movie scores.
Anyone who has experience with such big files and how to split up one big region into smaller parts, of which some will could deleted?
Secondly, what is the best way to produce CD/DVD after editing this file?
I will be most grateful for help here.
Macbook core 2 duo Mac OS X (10.4.9)

I would suggest that editing a file is best done with, well, an Audio Editor:
http://www.bulletsandbones.com/GB/GBFAQ.html#audioeditors
what is the best way to produce CD/DVD after editing this file?
No question, Toast 8
<hr width="250" align="left">
HangTime [Will Compute for Food] B-)>
♪♫ The Bullets and Bones Band ♫♪
Disclaimer<hr width="250" align="left">

In garage band, i want to import a 20 minute wav file and cut it into individual songs, and export them to itunes or dropbox or ...

in garage band, i want to import a 20 minute wav file and cut it into individual songs, and export them to itunes or dropbox or ...
i've imorted the file to audio for imovie, but can't see how to chop it up, save the pieces as individual units, and export them.

Create a new GarageBand project using the "Voice" template.
Drag you wav file into the Track area of the GarageBand window to create an audio track for it.
Use the playback button to find the gaps between the songs.
Mark each song in turn with the cycle region. Press the button with the two blue chasing arrows to enable the cycle region. This will create a yellow bar at the top of the track area. Drag the ends of this bar to mark one of your songs.
Now use the "Share" menu to send your song to iTunes.
When done, adjust the cycle region to mark the second song.

Corrupt wav files in Adobe but play fine in other apps

I have some old projects that have been backed up several times. The problem is some of the sound files appear corrupt when I load them into Adobe Audition. The sound plays all distorted with VERY LOUD clicks/pops in it. However, I can play the same files just fine in any standard media player (Winamp, WMP, VLC player)
I tried opening the project in the new Audition but same thing happens. I've also tried the "Open As" option and specifying what windows reports as it's sample rate etc but it comes out the same. Most of the other wav files in the projects work just fine, but one here and there is like this and is preventing me from remixing anything. Worst case scenario I suppose I could play the file in winamp and record it but then I'd have to match the timing up and everything and I just don't have the motivation to do that on all these projects.
http://home.comcast.net/~chicken3/Track17.zip
This is one of the files if somebody wants to take a moment and look at it. I just don't understand why winamp and all of them can play it fine if it was corrupt in any way, other than VLC which I know doesn't need file headers to play properly most of the time.
Below is a pic of what the wave looks like in Audition. Oh, and Reaper reads the files just fine as well, if I hadn't trimmed all my wav files down i'd just paste the whole project into Reaper and remix but unfortunately I trimmed all the files already.

Well as I said it plays in standard media players such as Winamp (which generally will not play a file with a corrupt header or any sort of problem like that) However I checked it out in that app you mentioned and the results seem normal to me. I've tried Open As in Audition and I can choose the 44.1KHz and 32 bits but it opens the same as it did before. So far I've only noticed 32 bit files with this problem but it's possible it has nothing to do with that. Again, the file plays fine in all standard players, and Reaper and other audio apps open it just fine. Audition will not.
Format                           : Wave
File size                        : 36.8 MiB
Duration                         : 1mn 49s
Overall bit rate                 : 2 822 Kbps
Audio
ID                               : 0
Format                           : PCM
Codec ID                         : 1
Codec ID/Hint                    : Microsoft
Duration                         : 1mn 49s
Bit rate                         : 2 822 Kbps
Channel(s)                       : 2 channels
Sampling rate                    : 44.1 KHz
Resolution                       : 32 bits
Stream size                      : 36.8 MiB (100%)
Edit: Downloaded goldwave trial just to see, it opens flawlessly in Goldwave. I suppose I can try resaving the files from within goldwave and see what happens...Audition is really pissing me off.

HI, I am having trouble when importing .wav files into logic.

Everytime I import this .wav file of a DJ set i recorded, it shows me the waveform but no sound comes out when its played. I have tried converting the bitrate to 16bit and the filetype to aiff as I saw it had helped people with similar problems but when i do this its boosts the gain massivly and the recrding becomes very distored? Really dont know what to try??
Also the reason I am trying to get the Wav in was bevause i need to normalise it as it had to be recorded very quitely to avoid distortion on the recording. I recorded it by using the imput on the back of my monitors as they are usb, and th programme i used to record it is "audio recorder tool" from the app store
Thanks for your help, Joe

I think the problem is you're recorded 32-bit wav files which Logic doesn't support.
(don't convert to aiff, unless you have the right program, that understands converting little endian 32 bit wavs to big endian aiff format, you will get distortion)
I think your best bet is to download "Audacity" the free audio editor and convert to 24-bit WAV files. Re-save the audio file under a different name after converting. In Audacity this is done using the Export function, selecting other non-compressed formats.
http://audacity.sourceforge.net/download/mac
The reason you want 24-bit WAVs to work with is you're going to be raising the volume of the file, 24-bit files will suffer less damage when editing with plugins, normalizing...etc.

Assign wav file in EXS

I'm looking for a simple, quick way to import sound effects I have created into Mainstage and simply assign them to keys on my MalletKAT. I know there must be a simple, direct method which I cannot seem to discover.
I have wav files and want to assign each file to one pad (or key) on my instrument. I'm trying to use the EXS engine.
Any help?

I think there is a better solution: drag all your files on a track in the arrange window in Logic, select all the files on track by clicking the track header, then choose the local menu Audio>Convert Regions to New Sampler Track. In the window that opens, select Regions as the creation option, and then the note you want the first audio file to be placed.
Then click OK and you have created a EXS track with a midi region (useless for you) AND the sampler instrument in one single click.
Save your instrument and you can use it from the Mainstage Library. TIP: put an asterix at the beginning so it appears on top of your list in MS.
Cheers,
JL
Message was edited by: Sarl Apaxx Designs
Message was edited by: Sarl Apaxx Designs

Need help with audio wav file in multicam clip

I created a multicam clip with 3 cameras with crappy, far away, room audio and 1 stereo audio wav file with 2 lavaliere mics of "good Audio" 1 lav left & 1 lav right. on one channel there is a lot of distortion from shirt ruffle etc.... but only at times. I would like to ride the mix and use sections of "good" audio and get rid of sections of "Bad" audio from the one channel. but, I cant figure it out. anyone??? I'm hoping Apple updates this issue soon with great audio support with "simultaneos multicam audio switching!!!" Please, Please Apple...

You can do what you need right now, no need for Apple to do anything, they've already don it. Check this out.
http://www.macprovideo.com/hub/final-cut/final-cut-pro-multiclip-audio-channel-e diting-1006

How do you load wav files in the EXS24?

Hey all - I'm new w/the EXS24 and am trying to import wav files off a sampler CD I bought. I can't figure out an easy way to do it. When I imported some Akai CD's before, it was as simple as opening AKAI convert and inserting the CD. Now, when I go to the "load mult-sample" for these wav's - it wants me to do everything one at a time. Surely, there's got to be an easier way than loading hundreds of samples one at a time. Any suggestions from all you EXS24 pros out there?

If you can get to the instrument editor, drag and drop your regions or files onto the keyboard and answer the following questions. To preview it have an EXS24 loaded up with EXS Instrument #5678 (or whichever number it happens to be.

Merge/append/concatenate wav files

hi,
I would like to ask.. does anyone knows how i can go about appending 2 PCM wav files to a new wav file with different format without any sound distortions?
Lets say i have 2 PCM wav files..
Eg:
1.wav
numChannel = 1(mono)
sampleRate= 16khz
bitsPerSample = 16bits
2.wav
numChannel = 2(stereo)
sampleRate= 44khz
bitsPerSample = 16bits
I am able to append 2.wav to 1.wav but because the new wav file's header takes the format of 1.wav, the sound for 2.wav gets distorted.
What should i do/change so that both files when appended, play exactly as they originally sound without distortion to either file?
Pls help. Any suggestions are appreciated. thanks.

I doubt that the problem is coming from the joining. Try putting just one file through the process, without adding anything alse to it. Try to simulate it as closely as possible. If that eliminates the hissing, then check to make sure you're not coming in on the wrong offset. If that doesn't work, no clue.
If it doesn't eliminate the hissing, there's probably a problem with data conversion, i.e. casting or chopping or mangling in general.
if(simulate with one eliminates hissing)
      check for offfset errors;
      if(still hisses)
          no clue;
else
      check for data mangling;
}

Distorted wave file (region)

Similar Messages

Maybe you are looking for