Recording wav file

HI is there any way to record wav file using Code composer studio to TI 5510 DSK and play them back. I am having problem writing a code to record in wav format. Plz help if there is any simple code to understand. Thank you

akki20,
I think you are looking for Texas Instruments. That is, unless you are somehow interfacing the 5510 DSK with LabVIEW or are using Measurement Studio for display / control. Let us know if that is the case.
Regards,
Ryan Verret
Product Marketing Engineer
Signal Generators
National Instruments

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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

Cisco AQM 8.5 not recording : wav files are deleted from client side

Daer Networkers,
We do have Cisco AQM 8.5 SR2 ES1 installed with UCCX 8.5
The issue is that when accessing the Web interface of AQM and try to look for calls, I can't find any one.
When checking the client side, I can see that the call are being recorded : The FROM and TO files are there. But once the call is terminated, the wav file appear for some seconds and then it dissappears asi fi it's deleted.
The wav files are not in the server too;
I don't know what is the issue. Can you please help ?
Thanks in advance.

Hi,
This issue was resolved by doing a repair to the QM Base services, then run postinstall as if it was run for the first once.
The Proxy Gateway program was missing for some reason. After doing the repair the issue was resovled and calls are uploaded to the server.
Now I am facing another issue ! : I can hear the client voice only. Agent's voice couln't be heard in the recorded files. As if the agent's voice is not recorded. Can you please advise ?

Problem in playing back mp3 which was recorded as wav file

Hi all,
I have recorded sound using my program. It uses classes like AudioSystem. And it's
able to record and play also. but in my site I have an mp3 player. So i thought of converting the recorded wav file into mp3. For this i used LAME utility in Linux. And it was converted. Also it is being played in normal players like Winamp. But when i try to play this converted mp3 file using my player in my site it's having problem. Why is it like that?
Should i do something while capturing the sound from microphone so that it can be converted into mp3 format??
Or
Can I directly capture voice to an mp3 file??
In my program,
targetDataLine.open(audioFormat) where audioFormat is like
float sampleRate = 8000.0F;
int sampleSizeInBits = 16;
int channels = 1;
boolean signed = true;
boolean bigEndian = false;
Should I do something more or change something??
Please help me...
Thanx and Regards,
sand...

Thanks Walter.
Unfortunately, the same issue happens... In whichever window I select "Bounce" from -- Environment, Audio Mixer, or even FILE > BOUNCE -- a window titled "Bounce 'Output 1-2' " pops up. I'm guessing this is what you mean by the "Bounce Output Window." But, there I'm only able to name the file in the "Save As" field, choose where to bounce the file to on my harddrive, and then click on "Bounce" in the bottom-right of the window. I have no tabs or anything else to choose from. So I'm still a little stumped. Any additional thoughts?
I really appreciate the help.

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).

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.

Wave Files? Where?

Where does GB put recorded wave files? I've been looking for my recorded raw materials and can't seem to locate them. I would like to know this so that I can, among other things, be able to delete unused takes.
Thank you very much in advance to everyone,
Javier Calderon

Okay. Thank you for the spotlight info . . . I was wondering why my test wasn't showing up there . . .
Okay . . . So you say I've got it now/am getting closer . . . But I'm still a bit confused.
I opened up a NEW file, and saved it in a folder named "Test" under the filename "Test" on my desktop. Then I made a quick, 5 second recording on a track I titled "Test" . . . I'm assuming that recording should have showin up in the folder I named "Test" on my desktop, right?
I see nothing in there, however, but the GB file named "Test".
- hence my present confusion . . .
Of course I'm missing something . . . I'm just not sure what it is at this point . . .
Javier

Recording audio files in PCM_FLOAT format

Hello!
I'm trying to record voice sample and I've got a problem with my .wav format. With:
AudioFormat format = new AudioFormat(AudioFormat.Encoding.PCM_SIGNED, 44100.0F, 16, 1, 2, 44100.0F, false);everything works fine. But what I need (because of further analysis of this audio file in Matlab) is PCM_FLOAT format:
AudioFormat format = new AudioFormat(AudioFormat.Encoding.PCM_FLOAT, 8000.0F, 32, 1, 4, 8000.0F, true);which unfortunately returns an error:
Exception in thread "AWT-EventQueue-0" java.lang.IllegalArgumentException:
No line matching interface TargetDataLine supporting format PCM_FLOAT 8000.0 Hz, 32 bit, mono, 4 bytes/frame, is supported.I've tried a lot of different combinations with PCM_FLOAT but every one of them returns same exception. I'm probably missing something but the question is: what? How can I find what audio formats could work with my sound card? If I can only use PCM_SIGNED or UNSIGNED how can I convert recorded .wav file to PCMFLOAT?

I've finally been able to try what you suggested. As server received from the client an array of bytes I tried to convert it to an array of floats:
IntBuffer sbuf = ByteBuffer.wrap(myAudioBytes).order(ByteOrder.LITTLE_ENDIAN).asIntBuffer();
int[] audioInts = new int[sbuf.capacity()];
sbuf.get(audioInts);
float[][] audioFloats = new float[audioInts.length][1];
for (int k = 0; k < audioInts.length; k++)
    audioFloats[k][0] = ((float)audioInts[k])/0x8000; // Two dimensions in audioFloats because Utils.floatsToWav needs to know whether it's mono or not.
}and then save it to file using http://code.google.com/p/j-simple-wav-io/ :
Utils.floatsToWAV(audioFloats,new File("output.wav"),8000);Unfortunately it didn't work. I checked the .wav files with AudioFileInfo from jsresources.org and it recognizes the format of output.wav as:
Type: WAVE
AudioFormat: PCM_SIGNED, 8000.0 Hz, 16 bit, mono, 2 bits/frame, little-endian
Lenght of audio data: 22050 frames, 44100 bytes, 2,75625 seconds
Total lenght of file (including headers): 44144 byteswhile valid audio file (I mean the one that actually works with my Matlab script) is:
Type: WAVE
AudioFormat: PCM_FLOAT, 8000.0 Hz, 32 bit, mono, 4 bits/frame
Lenght of audio data: unknown
Total lenght of file (including headers): unknownBefore that I tried to do it another way:
- getTargetEncodings(originalFormat) --> returned PCM_FLOAT (amongst others);
- isConversionSupported(targetFormat, originalFormat) --> returned true;
(where originalFormat is "PCM_SIGNED, 8000.0F, 16, 1, 2, 8000.0F, false" and targetFormat is "PCM_FLOAT, 8000.0F, 32, 1, 4, 8000.0F, false" - as you can see above I don't really know if Matlab wants little or big endian).
so I thought sth like this will do the trick:
ByteArrayInputStream bais = new ByteArrayInputStream(myAudioBytes);
AudioInputStream AIS = new AudioInputStream(bais, originalFormat, myAudioBytes.length / originalFormat.getFrameSize());
if(AudioSystem.isConversionSupported(targetFormat, originalFormat))
    AIS = AudioSystem.getAudioInputStream(targetFormat, AIS);
File file = new File(fileName);
int nWrittenBytes = 0;
try
    nWrittenBytes = AudioSystem.write(AIS, AudioFileFormat.Type.WAVE, file);
}It didn't work either (but it seems that this bit can help switching from 16 to 32 bits and from 2 to 4 bits/frame).
So - clearly I've done something wrong but again have no idea what. I'm sort of new to Java and an absolute novice to processing audio files so thanks for all the advices :-).

Assigning app to .WAV file

I've got a dandy new audio recorder (Edirol R-09) that records WAV files onto SD cards, connects via USB. When I connect the recorder to my OS9 G3, the WAV file shows up with a RealPlayer icon. When I double click the file, my RealPlayer crashes with a Type 1 error.
I've gone into the File Exchange control panel, made sure that WAV files are assigned to QuickTime Player. Other WAV files in my computer do show up as QuickTime icons and open properly in QT. When I hook the recorder up to an OSX iMac, the WAV file does show up as a QuickTime icon.
How can I get my system to properly label and open these files as QuickTime files?
Thanks,
UW

Hi, Uri -
If you made the link-WAV-to-QuickTime Player association in File Exchange in the File Translation section, try also making it in the PC Exchange section.
Nonetheless, Finder may try to open those files with RealPlayer. In OS 9, if a file has File Type and Creator codes Finder will search for an application which matches those; if Finder locates one such it will attempt to use it, regardless of the settings in File Exchange. The settings in File Exchange are used only when Finder can not locate an application that matches the codes. If those files were given codes that match RealPlayer, then that is what Finder will try to use.
The solution when that happens is to do one of two things -
• Hide the appplication you don't want used, so that Finder can't try to use it. One way to do that is to delete the 'wrong' program; however, if there is a recurring though occasional need for it, doing that can be inconvenient. A less inconvenient way is to compress it using Stuffit (or DropStuff), keeping the Stuffit archive but tossing away the original. A fresh copy of the program can be extracted from the archive when needed.
• Change the File Type and Creator codes of the files to those that match the program you want to use. If the files are on the hard drive, that is relatively easy to do using either FileBuddy or FileTyper. If they remain on the attached device's media, that may not be easy to do - and may be impossible.

Playback of Wav file

Hi there,
I have had a look through the back posts to try and sort my problem out, but have been unable to succeed in understanding. Problem is that I'm not very good with Labview, and the people who normally are have left my department. So I'm trying to work things out in ignorance. Basically I want to change my current Labview VI so that instead of continuously playing back a 'Simulate Signal' VI (a sine wave as it happens) I want to continuously playback a pre-recorded wave file. I just cannot work out how to incorporate this task into the program. We're using an SCB-68 interface board. I've attatched the program we're using below, in case any kind soul can offer advice.
Many thanks,
mike
Attachments:
Wav_playback.vi ‏102 KB

Hi Mike,
You have not stated what version of LabVIEW you have so I have modified your VI in LabVIEW 7.1 you will need LabVIEW 7.1 or later to open the attachments.
I have modified you original VI and kept the name the same "Wav_playback-1.vi" in THis I have removed the Tradiotional DAQ call, as you also had a DAQmx call, I have wired everything together so it should run. However I don't think this will work as you are using Tasks (which you have not provided) and then you setup timing and triggering again. All this is setup in the Task.
For the reason above, I have created "Wav_playback-_mod.vi" which has my original modifications as well as removing timing from you AO section, as this is in your task. In the AI line I have removed the Task and added the create Virtual channel call to setup the input as a Voltage input without using tasks as you need control of the sample timing VI, so seting this up as each step is better than using tasks.
Regards
JamesC
NIUK and Ireland
Attachments:
Wav_playback-1.vi ‏116 KB
Wav_playback-_mod.vi ‏120 KB

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.

How do i convert a garage band recording into a wav file?

I made a recording in Garage Band that I need upload to a particular website, but it only accepts wav files.
Supported WAV formats:
- Mono is preferable to stereo, but stereo is accepted.
- 8000Hz, 16-bit, PCM, mono is the recommended format.
- 44,100Hz, 16-bit, PCM is also supported
- mp3 files are NOT supported
Does anyone know how to do this? I went on You Tube which says to send to iTunes selecting WAV, but it does not seem to be working. Any suggestions?
Andrea

GB doesn't export .wav files. Sharet he file to disk as an uncompressed AIFF file, then use iTunes to convert it to .wav. (Go to iTunes' import settings and select .wav.)

Record a speech signal and save as a wav file?

Dear All:
I'd like to record a speech signal using one microphone and write/save as a .wav file. Can anyone help me on this issue.
I'm using 6024E PCI NI-DAQ 7.0.
Thank you
regards,
Jeffrey

Jeffrey,
Below is a link to a knowledge base that discusses the specifics in creating a .wav file and even includes an example of how to accomplish this with your daq board. Also, if you have a sound card, you can use the LabVIEW sound VI's to acquire a microphone signal through your sound card. Attached is the installer for the sound VI's for LabVIEW.
Wave File Creation
Attachments:
Sound_VIs_Installer_-_Win32.zip ‏2130 KB

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

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.

Recording wav file

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