Record GSM and ADPCM in wav files

Hello,
is there any way to record .wav files using GSM and ADPCM format(tritonus implementation for example). I need to record .wav files in this formats and send then through a network to another computer.
Is somebody has a code example please, post it or send me it to my e-mail:
[email protected]

why these files don't work in iTunes 5. Perhaps there is a bug in iTunes?
A bug or a new "feature". ;-)>
The problem is unique to iTunes 5, as I said - I had no problems with any iTunes version through 4.9.
The important thing is I will still have access to the files for classroom use (and I also just plain listened to them as well).

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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

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

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

Manually importing greetings and name id wav files to Unity Connection7.1(3)

We are going to be doing a fresh install of Unity connection in one of our sites. The current Unity is version 4.0(4) running on a windows 2000 server. Because the hardware, software and the OS are so out of date and buggy we don't want to attempt any upgrades on the server, we want to rebuild the system on new hardware. My question is, can we copy and paste wav files from the current unity to the new unity connection 7.1(3) server? We want to be able to copy the end users greetings and name id's for them in advance.
Please let me know if you do.
Thank you.
Rgds,
Vicky

Hi
I normally use COBRAS for this type of upgrade - basically you can run the export tool on your Unity server to export subscriber data, greetings, and even messages if you want. You then run the import tool for Unity Connection to push the users into the new server and map them to new CoS, schedules etc.
Unfortunately you do need to be running Unity 4.0.5 in order to get the export to work, but for the sake of a smoother migration it is worth the effort.
Search for COBRAS on this forum to see mroe info on it, or get the software from this link:
http://www.ciscounitytools.com/SiteList/AllDownloads/alldownloads.html
It's also worth reading through the help files to see what it can do, or maybe looking at the vids.
Regards
Aaron
Please rate helpful posts...

When I drag and drop a wav file into garage band it is rejected. How to imposrt a wav file?

Hi Everyone,
I want to edit a voice recording, done in WMA format, in Garage Band.
I saw a video in which the fie is simply draged and dropped in the timeline area, and the time line appears. Then youcan edit it.
But when I drag and drop this WMA file into the editing area, it jumps back out to where I dragged it from.
Is there any way to import WMA files intoGarage Band?
Thank you, Giorafromny

Giorafromny wrote:
Is there any way to import WMA files intoGarage Band?
not directly, you'll need to use some software to convert it to a format supported by GB:
http://www.bulletsandbones.com/GB/GBFAQ.html#importformats
(Let the page FULLY load. The link to your answer is at the top of your screen)

External Table - possible bug related to record size and total bytes in fil

I have an External Table defined with fixed record size, using Oracle 10.2.0.2.0 on HP/UX. At 279 byte records (1 or more fields, doesn't seem to matter), it can read almost 5M bytes in the file (17,421 records to be exact). At 280 byte records, it can not, but blows up with "partial record at end of file" - which is nonsense. It can read up to 3744 records, just below 1,048,320 bytes (1M bytes). 1 record over that, it blows up.
Now, If I add READSIZE and set it to 1.5M, then it works. I found this extends further, for instance 280 recsize with READSIZE 1.5M will work for a while but blows up on 39M bytes in the file (I didn't bother figuring exactly where it stops working in this case). Increasing READSIZE to 5M works again, for 78M bytes in file. But change the definition to have 560 byte records and it blows up. Decrease the file size to 39M bytes and it still won't work with 560 byte records.
Anyone have any explanation for this behavior? The docs say READSIZE is the read buffer, but only mentions that it is important to the largest record that can be processed - mine are only 280/560 bytes. My table definition is practically taken right out of the example in the docs for fixed length records (change the fields, sizes, names and it is identical - all clauses the same).
We are going to be using these external tables a lot, and need them to be reliable, so increasing READSIZE to the largest value I can doesn't make me comfortable, since I can't be sure in production how large an input file may become.
Should I report this as a bug to Oracle, or am I missing something?
Thanks,
Bob

I have an External Table defined with fixed record size, using Oracle 10.2.0.2.0 on HP/UX. At 279 byte records (1 or more fields, doesn't seem to matter), it can read almost 5M bytes in the file (17,421 records to be exact). At 280 byte records, it can not, but blows up with "partial record at end of file" - which is nonsense. It can read up to 3744 records, just below 1,048,320 bytes (1M bytes). 1 record over that, it blows up.
Now, If I add READSIZE and set it to 1.5M, then it works. I found this extends further, for instance 280 recsize with READSIZE 1.5M will work for a while but blows up on 39M bytes in the file (I didn't bother figuring exactly where it stops working in this case). Increasing READSIZE to 5M works again, for 78M bytes in file. But change the definition to have 560 byte records and it blows up. Decrease the file size to 39M bytes and it still won't work with 560 byte records.
Anyone have any explanation for this behavior? The docs say READSIZE is the read buffer, but only mentions that it is important to the largest record that can be processed - mine are only 280/560 bytes. My table definition is practically taken right out of the example in the docs for fixed length records (change the fields, sizes, names and it is identical - all clauses the same).
We are going to be using these external tables a lot, and need them to be reliable, so increasing READSIZE to the largest value I can doesn't make me comfortable, since I can't be sure in production how large an input file may become.
Should I report this as a bug to Oracle, or am I missing something?
Thanks,
Bob

HT4229 Does Logic Pro or logic express allow me to plug in a Halion VST player or to import and play VST wave files on any other player plugin?

Hi
Can I plug Halion 2 into Logic Pro or Logic Express? Or another VST player?
I cant find out how....

Logic unfortunately does not support any VST's by themselves. Rather they support AU's (Audio Units). So you cannot plug Halion 2 into Logic easily.
However, I have heard that you can install an AU called Plogue Bidule that allows you to run VST's. It acts as a sort of middle man so that you open the AU that opens the VST. I don't know too much about it, but it appears that they have some content on their website to look at if you are interested.
Hope this information helps!
Ryan

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.

Need advice on a strong laptop for playing/ recording music / .wav files

I primarily use my laptop for music and enjoy making .wav files and hooking up to main stereo of house. I've been experimenting with my daughter's laptop but I had to buy a sound card that attaches to the laptop. Are there any good laptops out there with a good enough sound card available internally? On my desktop I had a similar problem and needed to upgrade sound card to get the "what U hear" option.

TheLexMachine wrote:
SlimJim77 wrote:
TheLexMachine wrote:
Entropy wrote:
Most newer laptops have pretty good playback quality, and also most have optical digital audio outputs if you have a surround receiver with digital in.
Recording is a different story. In general, almost no system will have good internal recording capability, even with an internal addon card. Recording is best done with an external device.
You are confusing TOSLINK optical ouputs with digital coax outputs which are what they normally put on multi-media desktops to provide digital audio output though some older desktops did have TOSLINK ports. There was a period of time several years ago when you might find TOSLINK on a high-end multi-media notebook but digital coax was the standard in those days and that has obviously been replaced by HDMI which carries both digital audio and video signals.
In my experience, when it comes to laptops, HDMI will produce no sound. I too have seen a handful of laptops that support TOSLINK in the form of a 3.5mm connector, and even the standard sqare connector(not common, however).
If you aren't getting sound from your HDMI port that means your digital audio output is incorrectly configured, the digital audio signal is not in a format that your HDTV or display can recognize (simply uncheck the Dolby Digital, DTS, or WMA Pro boxes in the properties panel for your digital audio output), or the port on your display doesn't accept digital audio or is set to use analog audio in the settings. If you look in your TV's manual or dig up the manual online, the specs page will tell you what digital audio formats it accepts if any. A lot of older HDTVs with HDMI, as well as a lot of the first monitors with it, didn't accept digital audio over HDMI because they were using repurposed DVI chipsets which could be modified for use with HDMI by swapping out the DVI connector with an HDMI jack due to the fact that HDMI shares it's base video protocols with DVI. All it takes is fifteen seconds to figure it out if you know what you're doing.
Ya, a mere few hours after I made that post, a friend hooked up his newer laptop to my TV via HDMI, and sure enough...there was sound. Me stands corrected.
Disclosure: Former BBY employee.

Record waveform wav file

Hello,
I am trying to record a waveform in a wav file. If i play the waveform with the VI Labview the quality of the sound is ok but when i try to make the wav file the quality is not ok. I attached the file i want to save as a wav file and the VI i programmed. Thanks a lot!
Another question: There is any way to stop playing the waveform (or stop the program) when it enters into the "Play Waveform.vi"?
Thank you very much
Javier
Attachments:
MakeWavFile.vi ‏43 KB
file1.txt ‏1984 KB

I forgot before... i am using labview 8.0 so if you send me a VI, could you please send it to me saved in this version? Thanks

Cmus doesn't play .wav files and pulseaudio isn't working properly

The first problem I noticed was that Chromium was using about 90% of my CPU while I wasn't connected to the internet. So I deleted Chromium because I couldn't fix the problem any other way. After deleting Chromium the CPU problem was fixed but after checking htop again I noticed this message with -11 being in red:
(http://pastie.org/4234521)
I started using cmus and noticed that .wav files aren't playing anymore but I am not sure if that has anything to do with this issue. Cmus played .wav files just fine when I had Chromium however. Also, RAM is steadily increasing at about 700 MB which never happened before. I think maybe the problem with cmus and the RAM increase could be caused by pulseaudio not working properly?
Also, when I run "pulseaudio" in the terminal I get this message in red:
$ pulseaudio
E: [pulseaudio] pid.c: Daemon already running.
E: [pulseaudio] main.c: pa_pid_file_create() failed.
Last edited by rg_arc (2012-07-10 23:56:41)

I have stopped using cmus and I am currently using moc(Music On Console). I found out that for some reason pulseaudio is taking up alot of RAM.
Here is a picture of what is going on in htop:
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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

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.

WAV file won't replay in LabView or load in WMP, but works in VLC

I'm creating an audio file for an interview with the code snippet attached (VI is very large and does a lot of unrelated processing). This audio is recorded in the "interview" subvi. At completion, this subvi is closed and the "process" subvi is opened. If this process is followed, labview will open and play the WAV file with no problem. Alternatively, "process" can be run as a stand-alone. If this is done, I get the error 4810 at Sound File Read Open (possible reason: cannot recognize sound format). If I try to open the file in Windows Media Player, the wav will open, but will have a length of 0. If I open the wav in VLC, it plays with no problem. I haven't tried saving as a new file, but this is a sortof untenable solution anyway. For what it's worth, this problem arose when the audio recording was changed to a queue-based process. I've attached the old (working) code for reference.
Thanks,
Brendan
Solved!
Go to Solution.
Attachments:
interview sound recording.JPG ‏65 KB
playback audio.JPG ‏21 KB
old audio storage.JPG ‏94 KB

I'm also adding a screen cap of the working (right) and non-working (left) headers created by the snippets attached above. I don't have a lot of experience with this, but it strikes me as curious that the non-working (left) header has no value for what I think should be the File Size (data) section (as referenced here http://www.topherlee.com/software/pcm-tut-wavformat.html). I note also that the header seems to be 12 bytes longer that most references suggest, but this is the case in both the working and non-working headers (I'm looking specifically at offset 36-47 in both).
Attachments:
WAV headers.JPG ‏61 KB

Record GSM and ADPCM in wav files

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