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.

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Having problem storing incoming MU-LAW to PMC WAV file

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I am attaching the code below:
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outraspace wrote:
Hi,
I am able to receive ULAW/rtp, 8000.0 Hz, 8-bit, Mono stream and play it using the Player class. What I need to do is to store the stream data into WAV file. I have followed the JMF tutorial to set the output file format. I can also create the WAV file no problem. But when I try to play the gnerated WAV file using Windows Media Player, it says : Windows Media Player cannot play the file. The Player might not support the fle type or might not support the codec that was used to compress the file.
Does anyone know what is wrong with my code? Lots of things. As tbm pointed out, don't use a processor model. Use create a regular processor, and then create a ContentDescriptor object with all of the formatting you want, and set that on the Processor.
In your code "Here we choose the ULAW format", it doesn't appear that you're chosing ULAW. It appears you're chosing linear encoding. ULaw isn't linear, it's logarithmic, so certainly that's not the right format.
In the examples I've seen, the processor is started before the DataSink is started. You're doing this in reverse.
If you plan to configure a processor, do not get a realized one. That's like say "I'd like some ice cream", going out and buying ice-cream sandwiches and scraping out the ice cream...
You should disable all of the tracks except the ones you want data from. That means if it's not an audio format, you'll want to go ahead and disable it.
Without nitpicking anymore, I have no idea where your sample code came from, but it seems pretty useless. The best idea would be to take your code, delete it, and start over with the following code:
[http://java.sun.com/javase/technologies/desktop/media/jmf/2.1.1/solutions/RTPExport.java]

Distortion after converting wav file to mp3

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MP3 saved at 8K Sample Rate setting
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Tempo problems with imported wav files

Hey everyone, sorry if there's a quick fix for this in the forums that I couldn't find, but I've been having some tempo problems with imported .wav files.
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Hi,
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[phil@discovery hack]$ cdrdao simulate --device /dev/cdrom1 toc
Cdrdao version 1.2.3 - (C) Andreas Mueller <[email protected]>
ERROR: ./01_achilles_last_stand.wav: not in PCM format.
ERROR: Header of audio file "./01_achilles_last_stand.wav" is corrupted.
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CD_DA
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COPY
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TWO_CHANNEL_AUDIO
FILE "./01_achilles_last_stand.wav" 0
START 00:02:00
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01_achilles_last_stand.wav: RIFF (little-endian) data, WAVE audio, stereo 44100 Hz
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test2.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz
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I am not sure I understood what you actually did or intended.
Did you plug in the camera cards into a card reader and import with Lightroom, copy to destination on one portable HD?
That should work.
As soon as you unplug the portable HD, the storage location will be labeled as offline and all thumbnails have the "?". As soon as you replug the portable HD and ensure that it has the same drive letter as before, the "?" will vanish.
The "?" indicates that the file is not accessible at the storage location which you told LR during import.
It has nothing to do with being able to save.
Note: only in importing with "copy" chosen will LR actually touch a file (and copy it to the indicated storage location). Inside LR there are only records pointing to the image files and holding all sorts of descriptions, like metadata and develop instructions.
If you use NX2 for the task to copy images from camera card to portable HD, and import them afterwards to LR (add in import settings), that makes no difference.
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Although WAV files have the same audio coding as an audio CD, a disc full of WAV files does not act like an audio CD; it lacks the indexing and ToC needed in order for a "regular" CD player to play it. As Diane notes, put your WAV files in an iTunes playlist and burn it as an *audio CD.*

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Thanks for your answers!
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It's a stereo 48KHz 24bit file. I did a try with a 48KHz 16bit version: same problem...
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Problems converting WAV file

I'm trying to convert a ~250MB wav file to an MP3 using iTunes and I keep getting an error message that says:
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Anyone have any suggestions? I'm gonna try using GarageBand to convert the file perhaps...

If Garageband doesn't work, try Audacity.

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?
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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.

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
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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)
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else
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}

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How can i create a .wav file from an array in v5.0?

I am trying to output an aquired vibration trace as a .wav file.

Hi Brian,
no problem, see attached file.
The file called "To Wav.vi"
Hope it help you.
Mike
Brian wrote:
> I am trying to output an aquired vibration trace as a .wav file.
Attachments:
To_Wav_Folder.zip ‏53 KB

Wav file distortion problem

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