ADPCM and GSM to wav file with RAWCONVERTER

Hi, is it possible to codify ADPCM and GSM in a .wav file with the RawAudioDataConverter(jsresources) or do they need a targetEncoding?. I have recorded GSM data in a wav file but I can't decode them.
Example of my GSM coder and decoder
AudioInputStream ais = null;
          AudioFormat.Encoding targetEncoding = null;
          AudioInputStream audioInputStream;
          File encodedFile = null;
encodedFile = new File("gsm" + audioFile.getName());
               try
                    ais = AudioSystem.getAudioInputStream(audioFile);
               } catch (Exception e)
                    e.printStackTrace();
               if (ais == null)
                    System.out.println("No se puede abrir el archivo");
               AudioFormat sourceFormat = ais.getFormat();
               if (!sourceFormat.getEncoding().equals(
                         AudioFormat.Encoding.PCM_SIGNED)
                         || sourceFormat.getSampleRate() != 8000.0F
                         || sourceFormat.getSampleSizeInBits() != 16
                         || sourceFormat.getChannels() != 1)
                    System.out
                              .println("The format of the input data has to be PCM 8 kHz 16 bit mono");
                    System.exit(1);
               AudioFormat.Encoding targetEncod = new AudioFormat.Encoding(
                         "GSM0610");
               AudioInputStream gsmAIS = AudioSystem.getAudioInputStream(
                         targetEncod, ais);
               AudioFileFormat.Type fileType = new AudioFileFormat.Type("GSM",
                         ".gsm");
               int nWrittenFrames = 0;
               try
                    nWrittenFrames = AudioSystem.write(gsmAIS, fileType,
                              encodedFile);
               } catch (IOException e)
                    e.printStackTrace();
               }Example of my GSM decoder:
private File decodificar(File FileIn)
          AudioInputStream ais = null;
          File in = new File(archivoCodificado.getName());
          File FileOut = new File("decod" + in.getName());
          try
               ais = AudioSystem.getAudioInputStream(in);
          } catch (Exception e)
               e.printStackTrace();
          if (ais == null)
               System.out.println("cannot open input file");
          AudioFormat.Encoding targetEncoding = AudioFormat.Encoding.PCM_SIGNED;
          AudioInputStream pcmAIS = AudioSystem.getAudioInputStream(
                    targetEncoding, ais);
          AudioFileFormat.Type fileType = AudioFileFormat.Type.WAVE;
          int nWrittenBytes = 0;
          try
               nWrittenBytes = AudioSystem.write(pcmAIS, fileType, FileOut);
               System.out.println("AudioDecoder: written (bytes): "
                         + nWrittenBytes);
          } catch (IOException e)
               e.printStackTrace();
          return FileOut;
     }I think that it might be well recorded but it doesn't works, giving and exception. I think that its all right because the number of byts in my GSM recorded file is about the tenth part of a PCM lineal file, and when I try to reproduce it with winamp for example it detects a sound track but it can't calculate the length of the GSM file(wav extension I mean)

QuickTime Player X does not export WAV. You could use QuickTime Player 7, which if it's not in your Utilities folder already can be installed from the Optional Installs on the Snow Leopard install disk, but you'll need to purchase a QuickTime Pro key.
Regards.

Similar Messages

  • Analyzing and recognizing a sound file with a squeak

    I am running a brake test on a piece of equipment. I would like to be able to use Labview and a microphone, record every stop cycle and analyze the wav file for a squeak. I would just like to be able to keep track of how many squeaks occured during testing. I am using the professional development system of Labview 7.0. Any suggestions would be greatly appreciated. I have included two wav files with this posting, one squeak, one no squeak. Thanks!
    Attachments:
    no_squeak.wav ‏97 KB
    squeak4.wav ‏62 KB

    Hi,
    A response to your posting can be found at:
    Analyzing a wav file recorded with a microphone for a high pitch squeak
    I hope this helps
    Sincerely,
    Feroz

  • Importing wav files with dead space?

    I'm curious if anyone else has experienced slow loading times/overview calculations when importing wav files with lots of dead space into Logic?
    Working on vocal comps from a Pro Tools session. Logic has no problem importing full takes (24-bit, 44kHz wav files that basically start at the beginning of the song). But alternate takes that might have complete silence until the 3rd chorus, for example, take forever to load. 10+ minutes in some cases.
    If anyone can shed some light on this or share a similar experience, I'd appreciate it.
    Tech details, if it helps:  Macbook Pro 2.53 GHz Intel Core Duo, 4GB RAM, Lion 10.7.2, Logic Pro 9.1.6

    Well, iTunes has no ability to undo edits, so the problem may lie elsewhere. Please try the following extra-extra-careful procedure:
    - Edit the file and save it.
    - Exit the editor.
    - Make sure you know filename/location of the edited file, and that you are not confusing it with some unedited copy of the same song.
    - Play the edited file in some other player than iTunes and make sure the edits are as you wished.
    - Now, add the file to your iTunes library.
    Pls try this and let us know.
      Windows XP  

  • Export wav file with matching dropframe timecode

    I'm sure I'm just missing something simple, but I need to export a small corrected part of a finished as a WAV file with timecode that matches the project. Being for broadcast in the US, the project is in dropframe timecode. While I'm working in STP, I see drop frame time code. But when I export, I get non-drop timecode that doesn't match the orginal code. This is a five second fix for a one hour DVD and I hate to have to send the authorers an entire new sound track when they only need the five seconds with matching timecode.
    How can I export and make this work. Without having to get ProTools, that is!

    If you're talking about timecode stamping in a WAV file exported from STP . . . well it doesn't seem to work for me no matter what timecode rate I use. 29DF, 29, 24, 23. I suspect you need to either give them the timecode location manually or export the whole thing, and let them just punch in the section they need.

  • FLASH CS3[AS3] – MICROPHONE – RECORD AND SAVE AS WAVE FILE

    HI,
    I need solution for this one.Can we record,play and save an audio using flash CS3 astion script.
    It should be something like this:http://justflash.byethost12.com/microphone/microphone_recandsave1.html
    I have searched for the soluiton but i found it in flash cs5.I need it in flash cs3.
    Looking for the solution.
    Thank You

    Hello Jeff,
    From the LabVIEW Help description:
    Sound File Write Simple (Windows, Linux)
    Writes data from an array of waveforms to a .wav file. This VI automatically opens, writes to, and closes the .wav file. Windows You must have DirectX 8.0 or later to use this VI. Linux You must have the Open Sound System (OSS) driver to use this VI.
    So, the data input to Sound File Write Simple.vi is an array of waveforms. However, if you only have one waveform, you can build an array that contains only your single waveform using the Build Array VI. As described in the LabVIEW Help description:
    Build Array
    Concatenates multiple arrays or appends elements to an n-dimensional array. You also can use the Replace Array Subset function to modify an existing array.
    So, for example, you could run your single waveform into the Build Array function and get an array of waveforms containing only your waveform. You could then wire this array of waveforms to the Sound File Write Simple.vi. This is shown in the image below.
    Message Edited by Matt A on 06-28-2007 09:58 PM
    Matt Anderson
    Hardware Services Marketing Manager
    National Instruments
    Attachments:
    Build Array.JPG ‏11 KB

  • How to get string (specified by line and column) from txt file with labview

    Hi everyone
    How to get string (specified by line and column) from txt file with labview
    thx 
    Solved!
    Go to Solution.

    As far as I know, a text file has no columns.  Please be more specific.  Do you mean something like the 5th word on line 4, where words are separated by a space, and lines are separated by a newline character?  You could the Read from Spreadsheet String function and set the delimiter to a space.  This will produce a 2D array of strings.  Then use index array and give the line number and column number.
    - tbob
    Inventor of the WORM Global

  • Compressing the wav file  with JMF

    Hi Guys,
    Any one of you know about how to compress a wav file with jmf.Please its very urgent.
    Thnaks
    Jast

    Specify a compressed encoding for the audio output format. You'll have to enumerated the supported formats on the output track to finds out what's available to you. In Windows, JMF should be able to use the available system codecs. First try transcoding a wave file in JMStudio to see what works.

  • Reading, filtering and playing large WAV files

    Hi!
    I want to build an application that reads LARGE wavefiles, filters them and plays them via the soundcard.
    I tried to build something with the standard blocks available in Labview 7.0, but I keep running out of memory. The files I want to filter and play are simply too large The filesize is typ. 650Mb = 1 hour of stereo CD quality audio. The files I am using just won't play even on a WIN XP machine with 1Gb of memory.
    Next step was to build a function that loads the WAV files in blocks of ~16kB. This works fine, but now the filtering does not work anymore; after each block, the 'FIR Filter' block seems to reset its states. The result is a very bad repeating artifact in the sound. The repetition-rate is related to the buffer size.
    Does anyone have experience with this type of application? Help and examples are -very- welcome (as I am new to Labview... .
    Many thanks,
    Dr.dB

    I have created a VI that will do the continuous FIR filtering you need. I assume you want to filter a 'stereo' signal so the VI takes an array of waveforms as input. In this version the FIR filter coefficients (kernel) is common to all channels. If this is not what you need you can correct the VI accordingly.
    The control 'algorithm' specifies whether the FIR convolution is internally performed in the time or frequency domain. Changing the 'algorithm' does not affect your result but affects your performances. Select time domain (direct) if your kernel size (number of FIR coefficients) is, say less than 100, otherwise select 'frequency domain'.
    Try the attached example and let me know if you have any problems.
    Attachments:
    continuous FIR filtering on multiple channels.llb ‏79 KB

  • 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

  • How to read .wav files with DIADEM

    I have done a acoustic analysis with a microphone, and I have save the different Files with .wav format.
    I need to read these waveform files with DIADEM.
    Diadem recognize the different files however it returns the message "Diadem d'ont load the files ---.wav"

    Hello Jean-Julien!
    Get and install the Audio Data PlugIn. You can get ist here: http://zone.ni.com/devzone/cda/epd/p/id/44
    Matthias
    Matthias Alleweldt
    Project Engineer / Projektingenieur
    Twigeater?  

  • How to play wav files with Garageband/Midi keyboard

    Hello,
    I'm using Garagband (on my MAC, not iPad) and a midi keyboard for making songs. I just got a couple of cd's from Miroslav Philharmonik, but the sounds are all in standalone WAV files. There is no installation required. How can i use these sounds with Garageband? Must i copy/paste them in a specified directory on the HD? Or is a sample library necessary? I tried drag and drop, but that dind't work out...
    Thanx in advance for your help.

    this should help:
    import com.ms.win32.*;
    public void play()
              try
              URL url = new URL("file","C:\\","Hallo.wav");
              String wav= new String(url.getHost()+url.getFile());
              Winmm.PlaySound(wav,1,1);
              catch(MalformedURLException e)
    or something like this ;))
    Quaxi

  • Create wav file with freetts

    i cannot create an wav file using the code given in freetts readme.
    please help me to get out of this problem.
    if possible give me the code to do this with the Player demo given with freetts.
    my email id is [email protected]

    Hi,
    If you need to create a new command, you can use SAP Developer Studio. After deploy you will see it in
    System administration->system configuration->cm->user interface->Commands.
    There are examples about how to create commands.
    You need to create your own layout set. This one use a resource renderer, here you must setup the commands group that you created.
    After that, you create a navigation iview and layout set field you setup your layout set.
    Patricio.

  • Can iTunes convert a file and replace the old file with the new one?

    I am trying to convert my wav files to Apple Lossless to save some space. How can I convert the files and make them replace the old files? Right now it just converts new files so that I end up with two files. I have thousands I want to convert and just replace the old ones.
    I have an external program that will do it for me, however if I go that route then when my ITunes files are converted to lossless files I need to re-link the files as they show up with the ! showing they are missing. That would mean manually re-linking every song, which would take forever unless there is a way to mass re-link all the files.

    iTunes does not see the newly created files as the same file because they are not the same file.  They are new, different files.  If you have a track that is 1234.wav and you create 1234.mp3 it simply isn't the same file and you can't trick iTunes into thinking it is.  iTunes will handle it as a brand new file.
    Check Dougscripts web site for scripts which help modifying tag data.  However, I suspect you won't find any doing exactly what you want doing.  Some tag data are only accessible by iTunes itself.
    As for playlists, there's things you can try though they may be more effort than they are worth.  For example, you could add a special code for each playlist to some element of a tag (e.g., grouping).  Convert the file, then make a smart playlist selecting for the code you added to the group of tracks in the old playlist.  If you have hundreds of playlists with a dozen tracks each this will be a grand nuisance.  If it is a handful of playlists with hundreds of tracks each this might be worth it.
    Another way to preserve playlists would be to edit the .xml copy of the iTunes library file, replacing ".wav" with ".mp3" or whatever format you are using.  However, this won't help with play counts because that is only stored in the .itl copy of the file which you cannot edit.

  • Importing WAV file with Flash Cue Points Premiere Pro CS6

    I want to import a Japanese voice-over . WAV file which has Flash cue points marked in it so that I can clearly identify edit points on the video timeline when I import it into Premiere pro CS6.
    Or is there another way of getting a voice-over artist to mark up audio so that I can clearly identify when certain words or phrases have been spoken? Trouble is, Japanese characters mean nothing to me! Usually this is not a problem with European languages like French and German, as I can easily mark up the script with a highlighter as I can follow the voice over on the supplied script, but Japanese characters mean nothing to me.
    Please help.
    Many thanks.

    Premiere has a function now where you can import a clip and drag onto a timeline and it will prompt you with a message about changing the timeline settings to match your clip, (or if there is no timeline it will automatically create one that matches your clip.) This is in CC of course so you might consider upgrading. They have fixed a lot of compatibility issues from CS6 to now. If you don't want to get all of CC for 60/month then you can at least shell out 20/month on premiere. It's worth it. They've made it where you don't have to worry about 99% of all video formats. They just work.

  • Logic and BWF/Interlaced .WAV files

    Hi
    I am new to Logic and indeed D.A.W's in general but not to sound recording.
    Introductions aside
    I wish to import polyphonic .wavs (files created on a sound devices 744t) into Logic in order to mix and separate the tracks.
    It all appears to be straight forward however when I import the audio the individual tacks do not separate out on to different tracks, as I would expect them to do (It does it in audacity!!!). Instead I end up with a ''mix'' of all the tracks on one audio track.
    I have looked in the manual and googled the problem but don't seem to coming up with a solution can anybody help.
    Alan

    It's very certainly something your colleague did that triggered the creation of new audio files. Check his audio file folder of the project, I'm pretty sure it will contain the new files along with the old ones.
    Cheers,
    Jazz

Maybe you are looking for

  • How to manage playlists with multiple devices?

    We have multiple devices in our household that use the same Macbook Pro.  We all like different kinds of music and we are now having a difficult time trying to manage the playlists on the different devices.  How do we manually manage playlists for ea

  • Possible to fill a list using a custom Workflow in Sharepoint 2013

    Is it possible to develop a custom workflow that compares a Contact List and the Active Directory and add any missing people to the list? Or is this a functionality not available with workflows and sharepoint 2013?

  • Ipod not to be a detachable hard drive

    now i know what my problem is... how do i get my computer to not recognize my ipod as a detachable hard drive? i want it to recognize it as an ipod again.

  • Can any one give pros and cons of upgradation for Essbase 6.5 to 9.3.1

    Hi, plz let know, major differences between these two(Essbase 6.5 and 9.3.1) which one is stable, advantages and disadvanges, which one you suggest Plz let me know. Thanks, sudhakar

  • Observer Pattern

    Hi. I'm coding the observer pattern with generics. All is well up to a point. I have an ObserverI interface public interface ObserverI <     S extends Subject<S,D>,      D >      void update( S subject, D details ) ; }and a Subject base class public