Combine WAV files?

Kind of a weird idea, but I'd like to see if it would be
possible to "assemble" a WAV file (using CF) from smaller WAV
files.
My idea: We report of the heights of rivers across the U.S.
(as in, at this moment, the Potomac River is 5.45 feet). I am able
to have people send in an email asking for a certain river, and CF
produces an email back to them with the river height (process runs
every 5 minutes).
But I thought of sending them back a voice (WAV) file to
their cell phones, so they could hear the river height instead of
using a text email. I thought if I could have a library of small
WAV files for each number, a "point", and some words, maybe I could
combine the needed pieces from multiple WAV files:
"Potomac" + "Height is" + "5" + "point" + "4" + "5" + "feet"
Somehow join those little WAV files into a single WAV file:
"Potomac height is 5 point 4 5 feet"
I can't believe CF could join these files, but if anyone has
any ideas, please let me know!
Thanks....

Is there any shareware online that lets you merge wav files?
I would imagine you could string the phrases and send them to
the program
with <cfexec> or something, haven't looked into it
myself, but it sounds
like a interesting idea...
"Howard Perlman" <[email protected]> wrote
in message
news:f3h85b$jba$[email protected]..
> Kind of a weird idea, but I'd like to see if it would be
possible to
> "assemble"
> a WAV file (using CF) from smaller WAV files.
>
> My idea: We report of the heights of rivers across the
U.S. (as in, at
> this
> moment, the Potomac River is 5.45 feet). I am able to
have people send in
> an
> email asking for a certain river, and CF produces an
email back to them
> with
> the river height (process runs every 5 minutes).
>
> But I thought of sending them back a voice (WAV) file to
their cell
> phones, so
> they could hear the river height instead of using a text
email. I thought
> if I
> could have a library of small WAV files for each number,
a "point", and
> some
> words, maybe I could combine the needed pieces from
multiple WAV files:
> "Potomac" + "Height is" + "5" + "point" + "4" + "5" +
"feet"
>
> Somehow join those little WAV files into a single WAV
file:
> "Potomac height is 5 point 4 5 feet"
>
> I can't believe CF could join these files, but if anyone
has any ideas,
> please
> let me know!
> Thanks....
>

Similar Messages

Combining 2+ WAV Files into a single WAV file

Hello there,
I am trying to combine at least 2 WAV files (my example uses 3) into a single file. I have tried several examples (http://www.jsresources.org/examples/AudioConcat.html and http://stackoverflow.com/questions/653861/join-two-wav-files-from-java) but both have the same problem: the output file is an exact copy of the last file to have been joined.
Here is the code for the simpler of the 2 examples which I have altered to meet my needs.
        File outputFile = new File("concat.wav");
        for (int i = 0; i < files.length - 1; i++) {
            try {
                AudioInputStream clip1 = AudioSystem.getAudioInputStream(files);
AudioInputStream clip2 = AudioSystem.getAudioInputStream(files[i + 1]);
AudioInputStream appendedFiles =
new AudioInputStream(
new SequenceInputStream(clip1, clip2),
clip1.getFormat(),
clip1.getFrameLength() + clip2.getFrameLength());
AudioSystem.write(appendedFiles,
AudioFileFormat.Type.WAVE,
outputFile);
files[i + 1] = outputFile;//first file read in next loop, will contain joined file so far
} catch (Exception e) {
e.printStackTrace();
return outputFile;
I am unsure as where the problem lies, so would be grateful for anyone to point out the (no doubt obvious) mistake i've made :p

I don't see any problem with your actual code in terms of JavaSound, but...
The first run through, you'd be concating "file1.wav" and "file2.wav" to a file called "concat.wav". The second time through, you'd be concating "concat.wav" and "file3.wav" to a file called "concat.wav"...
On the second iteration, "concat.wav" will have a read-lock put on it until such time as the sequence input stream releases it... which will occur as soon as it's done reading it. At that point, the read lock is released and you can obtain a write-lock on it, but, you've already dumped all of the data from concat.wav into the bit bucket...so the only thing that gets to go in there now is "file3.wav".
So, don't try to read from and write to the same file at the same time...

.wav file not recognized in Premiere 8

I am trying to edit a video for which I am using a .wav audio file for the audio. I have done this before with no problem. Now Premiere tells me it does not recognize the .wav media file, and may be missing a codec.
When I go back to other projects I have done this way before, I find that I can't open the audio portion of ANY of them. On the project, the audio line shows, but is flat-lined.
Have I lost something. FYI, I am combining audio from concerts (recorded on Pro-tools and bounced to a .wav file) with video of the same concerts recorded on two video cameras. Then, I combine and synchronize to create a finished video with good sound and picture.
Thanks for any suggestions!
Thad Reynolds

Thad,
Welcome to the forum.
With PCM/WAV, the Audio is not compressed, so this should not be a CODEC issue. One thing to note is that there is also LPCM/WAV, and for some reason, a few of the Adobe products have issues with the LPCM, where the PCM version causes no issues.
I'd just run one of the WAV files through G-Spot (PC-only), or MediaInfo (PC, or Mac), and see what it says about the files.
I also assume that you are on the Mac, considering that you mention ProTools. Is that correct?
Were the older Projects, also with WAV files, created on the Mac, or perhaps another computer?
In very general terms, PrE should work perfectly with PCM/WAV files at 48KHz 16-bit. These are uncompressed files, and are the native format for Audio in PrE.
Good luck,
Hunt

Combining MP4 Files

I want to programatically combine MP4 files. I have seen many options for combining files using a gui but that will not work for my application. Does anyone know of an SDK or something that i can tie into (preferably from within .NET) and access some functionality that will allow me to combine multiple MP4 files into one file?
Thanks!

Steve Grisetti covers this well in his Basic Training Series on Muvipix.com.
First, you will want to start with a New Project, that matches this source footage 100%.
Import the MP4's (for VHS tapes, a better format would have been DV-AVI Type II)
Drag the two files to the Timeline, in the order that you want them
As you play the Clips in the Timeline View Mode, you can use either the Scissors/Razor Tool, or Ctrl+K to Cut the areas, that you wish to remove
Rt-click on the areas to be deleted, and choose "Delete and Close Gap"
If desired, you can add Transitions
Add any music, etc., using PCM/WAV 48KHz 16-bit (not MP3)
Export/Share to the format desired for delivery
Good luck, and happy editing,
Hunt

How to edit wav files

I've been using SoundTrack Pro 3 to edit WAV files, but I've now bought a sound recorder that creates 24bit 192kHz files. These are too big for SoundTrack, so I thought that Logic Pro would do the trick. So far I've had no success in trying to combine two files that are part of the same recording; the first segment is 3.9Gb and the second 4.27Gb.
Can Logic Pro do this this, or will I need to ask for a refund and find some alternative software?
Thanks
Nick

Why 192 KHz? Do you have a compelling reason to use this absurdly high sampling frequency? That is, are you recording bats, or dogwhistles? If not, I'd advise you to convert the files to 24/48 - that'll do for almost all situations. If you really, rea-heally want a high sampling frequency for (very subtle) quality difference (imperceptable but for (maybe) some of the best trained sensitive ears), you could go with 24/96. But again, 24/48 (or 24/44.1) will do. The 24 is much more important (significant diffence with 16)...

How do I edit wav files on my mac air?

Hi Im new to the site and to mac. Can anyone help me edit a simple wav file on my mac air? I've found plenty of blogs which say I can use quick time app but when I open mine, there are no options to edit or do anything other than play and record. Confused!

Hi lyndafromkingston,
I see that you would like more information about editing files in QuickTime. Here is an article for you that can give you some steps to help you accomplish this:
Mac Basics: QuickTime Player creates, plays, edits, and shares video and audio files - Apple Support
http://support.apple.com/en-us/HT201066
Editing content
QuickTime Player lets you easily edit content you've created or downloaded. You can trim, combine, and rearrange clips. When you're done, just save or export your finished video.
Trimming
To remove unwanted footage from your media:
Open the file in QuickTime Player.
Choose Edit > Trim. The trimming bar appears.
To select the portion of the recording you want to keep, drag the yellow trimming bar handles to the left and right as desired.
When you are satisfied with your selection area, click Trim. The unwanted footage outside the the yellow trimming bar is removed.
Thanks for coming to the Apple Support Communities!
Cheers,
Braden

Streaming wav files once edited won't work in flash

I need some help with streaming wav files in flash. I can load a wav file and stream it ok using actionscript 3 into my fla file. However if I go to edit the wav and make it shorter the saved wav file will not play. I have used a trial version of soundbooth to see if I could somehow change the settings for the wav but I still cannot make it work. It will stream as an mp3 but I want the file to loop and need the file to be wav. I have attached the file. I have no problem when I upload the file into the fla file but that is not what I want to do.
Can anyone help me.
Cheers.

Thanks for the input however I had already tried a variety of combinations. But I did go through it again.
The Flash Player supports 8-bit or 16-bit audio at sample rates of 5.512, 11.025, 22.05, or 44.1 kHz. I have tried all permutations of wav from 8 bit to 16bit and the all sampling rates from 11.025. The original file is attached at it was at 32000hz. Nothing I can do will make these play in flash once edited. I am using soundbooth howeve I have used audacity and wavepad. I have removed the metadata tags as that was suggested in one of the forums. I have attached the original files and the shortened file that I cannot get to work. I went into Soundbooth and went through all the preferences and checked the box where it asked for FLash compaticble at 16bit 44 hz etc. This is really frustrating. I know wav can be streamed.

How do I change the key of a wav file?

I'm using Apple Loops Utility and the wav file plays the correct key & tempo there. But when I save it, and open it up in LE or add it into the arrangement, it's playing the original key? The only thing that is correct is the tempo. Can you change the key and save the change using Apple Loops Utility?

I'm confused. Are you saying that in Apple Loops Utility the wav plays in the key and tempo of that wave, but when you add the same wav to LE it plays in a different key? You said in LE it plays in the "original" key. Seems like that is what you would expect. I would think it would play in the "original" key in both ALU and LE.

Placing 24/192 WAV files on iTunes for playback through Wadia 170 iTranspor

I record live music on a Sound Devices 722 digital audio recorder at 24/192 as WAV files. I can import the 24/192 WAV file into iTunes on a Mac and the bit rate is displayed as 9216 kps. This makes sense since 24x192=9216. When I click on the song in iTunes it plays back at the correct tempo.
I bought a Wadia 170 iTransport since it says it will output bit perfect digital files. I'd love to hear my recordings this way over my sound system. I understand that this would chew up drive space real fast on an iPod. However . . .
I can't get my iPod to accept the 9216 kps song files. I went to set iTunes / Preferences / Advanced / Importing to WAV but the options for sample rate only go up to 48.0 kps and the bit rate up to 16-bit.
Any thoughts or suggestions on how to overcome this will be appreciated.

Hi,
I'm from germany so no 'best buy' but I think I will lock around sleeve/bag for a different product with same measures. Any tip?
The clear plastic case is not what I want. I would like to use the player without such protections. I just want a sleeve to protect the player while its in my jacket/pants.
Btw, i have bought this version: Creative Zen (maybe european version?). I also doesn't seem to have your mentioned sleeve. I also couldn't find it on the website.
Anyway, thank you,
Chris

Broadcast Wav files from PT to Logic not moving to original record position

Okay, my system needs upgrading but I'm stable (at least I think I am til now!) I have a TV network sending me broadcast wav files from PT. They shoot their video on a 24 hour clock. I'm to be editing and mixing (in Logic) their audio tracks from PT and send them back as broadcast wav files for them to sync back to their video. After years and years of recording and producing music, this TV world is a first for me. 2 issues at hand that are causing trouble.
NUMBER 1: The wav files will not snap to original record position. I keep getting an error message about "out of range" and of all craziness with a negative SMPTE number. ??? The prompt suggests an offset error. But there isn't I can find. Start time is set to 0:0:0.0.0 here and at the TV studio. They film at 29.97d. That matches as well. Sample Rate is correct at 48k 24bit. I've uncovered every thing I can think of. I did have to set the tempo down to 5 BPM in order to see a full 24 hours in SMPTE.
NUMBER 2: I can't figure out how to see the timestamp start time of the audio files in the audio window. I have every option turned on in the VIEW option, I can see length, but I see nothing about start time of original record time. Am I missing something simple here? Seems like a very standard option that should be easily accessible. It would allow me to prepare the location on the time line.
Overall, I'm hoping I can do all the above with my current rig but realize that the clock is ticking toward a whole new system with Logic 9.
Help please. Don't want to lose this client.

Sure do appreciate everyone's willingness to help. Setting SMPTE start to 24:00:00.0.0 didn't work either. I confirmed with the studio manager at the TV station also that they run a 24hour clock with 0 being midnight.
I'm really hoping that the error is on their end.
*It would help me if anyone knew how to VIEW the Timestamp info attached to the wav files. That would at least give me an approximate idea of where they should be.*
I'm supposed to be editing and mixing the voice dialog from numerous different scene takes that were shot throughout a day of filming. It's critical that I can see all these different takes that were captured over hours of open recording. Some of the files are almost 4 hours long. It's a big project and I'm sure hoping I can stay inside Logic to get it all done with excellence.

Wav files on external drive will not work in Apple Loops

When I import wav files (from Nu Jazz Funk library) by dropping on the AL browser from an external drive, Apple Loops insists on copying them to my system drive (no option to make use current location). If I manually make an alias, the files show up, but play at the wrong tempo. I even tried letting AL make the copy, then moving that to an external drive & making an alias of that. No go. BTW, aif files work fine.

Yeah, that's because Apple Loops are .AIF files, not .WAV.

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.
Long story short, my system couldn't handle playing all the tracks for a song while recording drums, so I bounced out an mp3 of the song and put it in a new Logic file so my drummer could just play along to that as I recorded him. Unfortunately, the original song is at 167 bpm, but I forgot to change the bpm in the new Logic file with the .mp3 file of the song to 167 bpm, so it was left at the default 120 bpm.
So, the drums were recorded at the correct 167 bpm, but Logic thinks that those new drum .wav files should be played at 120 bpm, so when I import my drum tracks back into the original file, they do not play correctly at all.
I could get record it all again, but I wanted to check if there was a way I could salvage what I already have, since my drummer lives about an hour away right now and can't just come over whenever he wants.
Thanks for the help! I really appreciate it.

Hi,
First, do not use MP3 in Logic, the sound quality is less than AIFF, WAV or CAF, and Logic has to decode it for playback, making it a heavier burden on the CPU than an uncoded audiofile, such as AIFF, WAV or CAF.
Secondly, audio files are independent of Logic's tempo. If you bounce down an audio file in any format (other than Apple Loop), it will play back at the same speed, +regardless of Logics' tempo setting+, either at recording or playback. Logic doesn't 'think' anything. The BPM is only important to MIDI tracks, or to the spacing between audio files. The audio files themselves *are not affected* by the tempo setting. If you import an audio file of tempo 167 into a 120 BPM project, the file will still play at 167, only Logic will indicate the wrong bar positions.
regards, Erik.

Wav file from remix stem pack has no volume - Logic Pro 9

I recently downloaded a remix stem pack and can't get the vocal wav file to play. It show's a wave form, but no volume. If I load a sample wav file, a drum hit for example, it will play no problem. Anyone have any idea's? It's not the wav file as it plays in itunes, and also plays in my friends logic pro.
Thanks

The hierarchy is [LOGIC FILE NAME]>LgDoc>(files ... Displaystate, DocumentData, Windowimage.jpg)
Like this:
You need to recreate the folder structure in order to be able to open the file. (look at the path bar at the bottom of the picture.)
Do not disassociate the displaystate from the documentData files. They should be together. The windowimage file is optional and will be recreated when the file is saved again.
Hope this helps!

How can I open a WAV file in LabVIEW 7?

I need to make a program that opens a WAV file and converts it into amplitudes values and saves the in a .txt file. But how can I make LabVIEW 7 open up a WAV file? It's probably easy, but I didn't come up with anything up to now!
Thanks in advance!

try this
regards
timo
Attachments:
wav2txt.vi ‏17 KB

Mixing external wave file into video

I've recorded a video including sound of a mandolin being played. At the same time I recorded the sound with my external recorder which resulted in a wave file of better quality that the sound from the camera. I would like to import the video and the externally recorded wave file into a project, line up the external wave file with the sound recorded by the video camera and then delete the sound recorded with the video camera so I end up with a video that has the sound from the external recording system. Any recommendations on a good approach to accomplishing this task?

There's information here about how to use the new merged clips features in Premiere Pro CS5.5 to merge the audio from an external source with the video from the camera.

Combine WAV files?

Similar Messages

Maybe you are looking for