Converting a WAV file to a Raw Audio File

Is there an easy way to convert a WAV file to a raw audio file? Maybe some way to just strip the file header?
Would if work if I just loaded a WAV file into an AudioInputStream and read all the bytes out and dumped them directly back onto disk via a DataOutputStream? Even if so, there must be a quicker solution, right?
Thanks!

philgmo3 wrote:
Is there an easy way to convert a WAV file to a raw audio file? Maybe some way to just strip the file header?
Would if work if I just loaded a WAV file into an AudioInputStream and read all the bytes out and dumped them directly back onto disk via a DataOutputStream? Even if so, there must be a quicker solution, right?That would work, sure.
And I'm not entirely sure what you mean by a "quicker solution". That's literally about 10 lines of code...
JavaSound wasn't really meant to do audio processing, it was designed to give you access to sound data so you can do audio processing if you want it. It's not particularly useful, but it is extremly powerful...

Similar Messages

Can i convert a wav file to a digital signal without using DAQ?

i tried the coversion of a sine wave to digital signal, ofcourse it doesnt work on converting wav file to a digital signal. i am also new in labview. i tried to explore it but there are many functions that i can't use.
so my these are my questions.
1) is a .wav file is an analog signal?
2) can i convert it without using any DAQ device?
tnx guys! i'll post the programs that uses quantization of a simple sine wave. i tried it to .wav file but it didnt work.
Attachments:
act04-02.vi ‏41 KB

.WAV is a digital "signal". It uses Pulse Code Modulation to encode analog signal. You don't need a DAQ device to "convert" it to analog. Any computer with a sound card, and media player can "convert" a .wav file to the analog audio signal it represents
Wan L
Applications Engineer
National Instruments
http://www.ni.com/support

When I try to convert a wav file to an mp3 it only gives me a 6 or 7 second mp3....what am i doing wrong?

When I try to convert a wav file to an mp3 it only gives me a 6 or 7 second mp3....what am i doing wrong?

Sorry you are having problems. Can you give an example preferably of some small publicly available file and the site you are trying to download it from ?
As a simple example of a download (ok it is a .png image file not not a .exe file) but can you try to download a copy of your avtar/icon, as used on these forums. In Windows one method is that you right click & should get an option save. <br/> Mine alongside uses the link
* https://support.mozilla.com/media/uploads/avatars/avatar-257447.png
Are you able to download such an image, then see it in your download manager, and find it on your computer.
Once you have got Firefox to save your binary file you can use it. In the case of the .exe file you are trying to download that should be usable as intended with your Windows Operating System, the avatar also should be easily opened and viewed.
P.S.
You could try installing the add-on https://addons.mozilla.org/en-US/firefox/addon/opendownload-10902/ if you wished to adapt Firefox to run .exe files instead of just save them.

How can I convert a wave file to a jpeg file (picture) is it possible???

Hello:
How can I convert a wave file to a jpeg file (picture) is it possible???

zuckini;
I am pretty sure you know, but just to be completely sure: a wave is (usually) sound signal and a jpeg file is an image.
If you mean how to display the wave signal in a graph and then save the graph as a jpeg file, then just drop a graph indicator in the control panel, wire the signal. Check any of the example shipped with LabVIEW on how to do this. To get the image, drop a graph indicator's invoke node in the block and select "Get Image". Check this page for detailed information.
If you have something else in mind, please reply to this message with additional information about the wave signal, the image, etc.
Regards;
Enrique
www.vartortech.com

Converting recorded WAV files in Advanced

Hi
When I record my vinyl onto my laptop I record them as WAV files. When I want to convert them I simply go to 'Advanced' & convert them to MP3's...Very easy.
How ever, when I use my brothers laptop to do exactly the same process it only gives the option to convert them to AAC versions! They take much much longer & it simply does not give me the option the covert to MP3!
We both have exactly the same operating system (10.4.11) & iTunes (8.0.2)
Any ideas?

When you convert a song in iTunes it will be converted to the format that you have set to import your CDs in. If you are using iTunes 8 go to Edit>Preferences>General and click on Import Settings, change your option to MP3. Go back to your library and highlight the songs you want to convert and go to the Advanced menu at the top of your iTunes window and choose "Create MP3 version:
iTunes: How to convert a song to a different file format

HT3823 may I known, my itunes song is ACC file how to convert to WAV file

may I known, my itunes song is ACC file how to convert to WAV file

Here's a free online file converter I use all the time: http://www.convertfiles.com/
And don't you mean AAC, not ACC? lol

How do I convert a wave file of 50-60 meg into an MP3 to send to a client?

This is ridiculous...I'm reading the help page and it's been absolutely no help whatsoever.

From the iTunes help page searching for "Convert".
" Saving a copy of a song in a new file format
You can convert a song to a different file format (and keep a copy of the original). For example, you can save a copy of a compressed song file such as MP3 or AAC in an uncompressed song format (AIFF or WAV).
When converting from a compressed to uncompressed file format (for example, from MP3 to AIFF), you shouldn't notice any reduction in sound quality. However, when converting between compressed formats (for example, MP3 and AAC), you may notice a reduction in the sound quality. For the best results, if you want your music encoded in a different file format, import the music again from the original source using the new encoding format.
IMPORTANT: Except for iTunes Plus songs, you can't convert iTunes Store purchases.
To convert a song's file format:
Choose iTunes > Preferences, click Advanced, and then click Importing.
In the Import Using pop-up menu, choose the format you want to convert songs to, and then click OK to save the settings.
Select one or more songs in your library (Command-click to choose multiple songs), and then choose Advanced > "Convert Selection to...."
To convert all the songs in a folder or on a disk, hold down the Option key and choose Advanced > "Convert Selection to," and then choose the folder or disk containing the songs you want to convert.
The song in its original format and the newly converted song appear in your library."

Converting WAV files to MP3

When trying to convert my wav files to mp3 in itunes, I get an error pop-up that says "The disk could not be read from or written to." These files worked fine before I updated to the new itunes.

PaulSimpson wrote:
The files are on my hard drive Steve. They were audio files that I downloaded from my personal camera. I didn't even think these kinds of files had permissions.
OK just wanted to make sure. Sometimes, read/write errors occur when write protection is enabled on, say, an SD card. (that little tab on the top)
All files actually have permissions under Mac OS X, one of the aspects of the underlying UNIX file system. Every file is capable of being readable, writable, and/or executable by either the owner, the group to which he/she belongs to, and/or everybody. Sometimes accessing files becomes erratic when the file's permissions unknowingly change, thus the good idea to "repair permissions" from time to time with Disk Utility.
However, you mentioned that the original WAV files came from your digital camera. What happens if you try to open these WAVs with the QuickTime Player? Do they play as per usual?
PaulSimpson wrote:
If you still think repairing my permissions with my install disc is the way to go, I'm game, but I'm going to need a bit more step by step.
We'll save that for a little later. Originally, I was leaning toward the feeling that the hard disk may have needed verifying/repairing (through Disk Utility) - because of the read/write errors. But it is possible that it may be an issue with the WAV files themselves.

Converting WAV files to MP3 files

I have a USB turntable for converting my vinyl records to digital. When the files are imported to itunes they are large WAV files. I there a way to set up itunes to automatically convert the WAV files to MP3 files?

Old thread, but still very useful. I too looked for a converter for Mac
for ages and stumbled across your forum. Now you can use an app called audio converter for mac, It costs a bit, but u can try it out free first to see if it works for what ,you need it for. Plz click the link below to get more info: http://www.mp3converter-mac.com/wav-to-mp3-converter-for-mac/

How do I convert soundbooth .wav to .cda files?

I have tried to convert soundbooths .wav file to .cda file format using Nero software, but it says I am missing a required plugin to do this.
Does anyone know what plugin this may be?
I am using soundbooth 1.0 and Nero 8.
Does soundbooth add something to there .wav format?

.wav must be 44.1K sample rate 16 bit stereo in order to burn to CD.

Converting .wav files to mp3 format.

Hi,
I am trying to convert a wav file into mp3 format using the below code. I am not seeing any errors in the console but at the same time the mp3 file is created without any content. Any ideas on how to address this issue ?
Format[] FORMATS = new Format[] { new AudioFormat(
AudioFormat.LINEAR) };
String srcFilePath = "C:" + File.separator + "Users"
+ File.separator + "vkiran" + File.separator + "Desktop"
+ File.separator + "test" + File.separator + "4a714ea2.wav";
String mp3FilePath = "C:" + File.separator + "Users"
+ File.separator + "vkiran" + File.separator + "Desktop"
+ File.separator + "test" + File.separator + "4a714ea2.mp3";
File srcFile = new File(srcFilePath);
File destFile = new File(mp3FilePath);
MediaLocator source = new MediaLocator(srcFile.toURL());
DataSource d = Manager.createDataSource(source);
ContentDescriptor cd = new ContentDescriptor(
FileTypeDescriptor.MPEG_AUDIO);
Processor p = Manager.createRealizedProcessor(new ProcessorModel(d,
FORMATS, cd));
p.configure();
System.out.println((p.getDuration()).getSeconds());
MediaLocator destination = new MediaLocator(destFile.toURL());
DataSource ds = p.getDataOutput();
DataSink sink = Manager.createDataSink(p.getDataOutput(),
destination);
sink.open();
sink.start();
sink.stop();
sink.close();
p.stop();
p.close();

You are converting the WAV files to mp3 format before transferring them to the iPod?
You're using iTunes to do this?
Just for the record, the iPod will not play mp3 files below a bit rate of 32kbps nor will it play compressed WAV files.

Trying to convert wav file to mp3 using itunes?

Is there not an easier way to convert a wav file to an mp3. I import my file or folder but it does not convert. I have had to manually ask it to create an aac or aca out of each import. shouldn't it convert audimatically when I import file or folder. It did on the last version before this new update. Please help.
Flustered

If Garageband doesn't work, try Audacity.

Can no longer listen to low-quality WAV files

I have used iTunes to organize and listen to voice files recorded on portable digital recorders and cameras, but today discovered that they no longer work. Furthermore I can no longer import these files, with no error message. Should I look to replace iTunes with other software? Install Windows?
Kind: WAV audio file
Size: 4MB
Bit Rate: 32 kbps
Sample Rate: 8.000 kHz
Date Modified: 22/05/08 7:59 AM
Play Count: 2
Last Played: 21/08/08 10:22 AM
Channels: Mono
Sample Size: 4 bit
Compression: DVI IMA

I am having the same trouble. It started with the iTunes 8.1 update. What I've found is that you can solve this with a work around, it is not pretty, but it does work. The very strange thing is that Quicktime plays them fine and Quickview plays them fine(my guess is it is using Quicktime to play them).
Basically use another program to convert your WAV files to MP3 format. Then you can import them into iTunes.
This is a good one(open source and allows for editing the sound files)
http://audacity.sourceforge.net/

Problem with cdrdao reading .wav files

Hi,
I have being having problems with using cdrdao to read .wav files that I created from other sources (.flac, .ogg and .mp3) using flac123/ogg123/mpg123.
Basically, this is the error
[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.
The 'toc' file contains....
[phil@discovery hack]$ cat toc
CD_DA
TRACK AUDIO
COPY
NO PRE_EMPHASIS
TWO_CHANNEL_AUDIO
FILE "./01_achilles_last_stand.wav" 0
START 00:02:00
Having used the 'cdrdao' program previously, I am pretty sure the 'toc' file is sane.
I then noticed a difference between .wav files using the 'file' program:
[phil@discovery hack]$ file *.wav
01_achilles_last_stand.wav: RIFF (little-endian) data, WAVE audio, stereo 44100 Hz
test.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz
I found that the 'test.wav' file worked when substituted as the FILE in the 'toc' file......
With a bit of research I found a work around, namely to convert the .wav file to another .wav file using 'ffmpeg'.
[phil@discovery hack]$ ffmpeg -loglevel quiet -i 01_achilles_last_stand.wav -f wav -ar 44100 -ac 2 test2.wav
The resulting files...
01_achilles_last_stand.wav: RIFF (little-endian) data, WAVE audio, stereo 44100 Hz
test.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz
test2.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz
Changing the 'toc' file to pick up the two test* files worked fine.
Does anybody know either a better fix or the reasons why this is happening?
Regards
Phil Q

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.
LR will show the "?" whenever the portable HD is unavailable or the path invalid because with a different drive letter.
Cornelia

Audition 5.5 - Help with .wav files for phone system - making a hold message PC & Mcc

Hello everyone,
I am really frustrated trying to create a .wav file for our phone system and I was hoping someone might have some suggestions as I am new to using Adobe Audition. Our phone system will only accept .wav files for our hold messages but the tricky part is that is can only be a maxiimum streaming bit rate of 64 kbps. Everything we've created under 256 kbps becomes in-audible and I'm really not sure what to do as the current messages sound fantastic. Our current hold message is 35 seconds long and we have another one we are about to do which will be 2 minutes long (uuuugh).
- I've tried first converting the .wav file to an .mp3, lowering the bit rate to 64 kbps, then converting back to .wav but the bit rate goes up to almost 1500 kbps after re-converting.
- I've tried making the track mono, reducing the sample rate to 22k but the size ends up being somewhere between 512 and 768. If I lower the sample rate much further and convert to 8bit, the audio sounds terrible.
- I've also tried using Adobe Media Encoder, Audacity, Hand Brake, OS X Automator and a few other programs all with the same results.
- I've tried re-recording the audio on a handheld .mp3 recorder hoping the initial quality would be less (originally recorded on a PC through Audition with a external mic).
Unfortuantely, the person who used to do these conversions is no longer with the company, never documented the process and all I know is that they used to do this on a Mac. Also, the software that uplaods the files is a Linux based which is out of my area of expertise so I'm not able to tinker with it to make any headway.
** Any sort of suggestion would be greatly aprpeciated as I really don't know much about audio. I've been working on this for 2 weeks now and this is really driving me crazy. **

9ov7 wrote:
Hello everyone,
I am really frustrated trying to create a .wav file for our phone system and I was hoping someone might have some suggestions as I am new to using Adobe Audition. Our phone system will only accept .wav files for our hold messages but the tricky part is that is can only be a maxiimum streaming bit rate of 64 kbps. Everything we've created under 256 kbps becomes in-audible and I'm really not sure what to do as the current messages sound fantastic. Our current hold message is 35 seconds long and we have another one we are about to do which will be 2 minutes long (uuuugh).
- I've tried first converting the .wav file to an .mp3, lowering the bit rate to 64 kbps, then converting back to .wav but the bit rate goes up to almost 1500 kbps after re-converting.
Eugh! Converting to MP3 is absolutely not the thing to do!
- I've tried making the track mono, reducing the sample rate to 22k but the size ends up being somewhere between 512 and 768. If I lower the sample rate much further and convert to 8bit, the audio sounds terrible.
- I've tried re-recording the audio on a handheld .mp3 recorder hoping the initial quality would be less (originally recorded on a PC through Audition with a external mic).
Okay, in reverse order: I know it sounds daft, but actually you want to start with the best recording you can possibly have. You are going to have to reduce its bit rate, but if you reduce a crap signal's bit rate, it inevitably ends up worse than reducing a good signal's one; when you think about that for a moment, it's going to make sense (I hope...).
As for what you need to do - well it's going to have to be a mono file anyway (you haven't got a stereo telephone system), and that will immediately halve the streaming rate, but the other thing you need to bear in mind is that telephones have a limitited bandwidth, and that your sample rate, even at 22k, is going to give you more bandwidth than the system has. If you can reduce it to 8k, it will work fine. This will still give you an upper frequency limit of 4kHz, and that's more than adequate for telephone speech, believe it or not...
The other thing, and this is where I think you may be running into trouble, is that 8 bit streaming is pretty much the norm here. All it does is to increase the noise floor, so that the signal to noise ratio is about 48dB instead of the 96dB it would be with a 16-bit signal. But, to get this to work to your advantage, you have to optimise the audio you are using in the first place. Chances are that you need to compress your recording so that it has a relatively limited dynamic range before reducing it to 8-bit, and then you'll be fine. This is better anyway for telephone systems - there's far less chance of bits of the message being missed because they're too quiet. Also, you don't want massive great peaks in the recording - these just reduce the maximum overall level for the majority of the message sound. So, you need to apply a limiter to get rid of them, leaving the majority of the speech sounding just as good, only louder. All of this you can easily do in Audition - don't need any other software at all.
So, my guess is that the previous maker of your messages knew exactly what he was doing. Yes, he could have documented the basic steps, but he couldn't have been absolutely specific about it because no two recordings are the same. I'll attempt to explain the basic steps in English, but how well you will be able to follow this, I'm not sure...
The basic process I'd go through with the recordings is firstly to normalize them to 0dB, then look at how far down most of the 'green' parts are - this is the bulk of the signal, and where all of the meaning is. Then, you apply a limiter to this so that those green parts don't have peaks sticking out of them. Then, if you need to, normalize again so that the green part moves to the upper part of the waveform space (towards where 0dB is). Then, if you need to, apply compression to the signal so that the level of it sounds about the same all through the message.
Now, when you reduce your signal to mono, 8k, and 8-bit, pretty much everything about the original signal will still be there. That should stream 8-bit words at 8k, making 64k streaming for a single mono channel. This will work fine, but only if you optimise the signal first - otherwise no chance!

Converting a WAV file to a Raw Audio File

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