Sound & Music Forum RSS Feed

mp3 audio xing header TOC specification/information

Sun, 28 Jun 2009 05:03:25 -0700

Can anyone point me at the definition/specification of the contents of a Xing header TOC please.
I've spent hours googling for this and can only find
"Every TOC entry contains the size of the n-th frame. Calculating the position of the 3rd frame should look as following: header_size + toc[0] + toc[1] + toc[2]"
which obviously isn't correct as there is always 100 TOC entries (and the calculation example would give the start position of the 4th frame).
Ive investigated existing Xing files and the TOC contains 100 bytes, the first being zero and each incremented by 2, 3, 4 etc.
Example 0, 2, 4, 7, 9, 11 ........ 248, 251, 253.
I am writing a C program which creates a .mp3 audio file from extracts from various other .mp3 files and so the result will be a Variable Bit Rate file.
I've everything working now and the result plays fine. However, I need to allow positioning within the file when being played (e.g. by Windows Media Player) which I think creating a correct TOC will do.
[If anyone is interested the program produces music to run to: selected tracks with 'timers' inside telling you your remaining duration etc. Once working I will be giving it away for free, including the C source if required.]

blending audio data

Fri, 05 Jun 2009 02:45:40 -0700

hello,

can someone tell me how to blend audio data (its voice data actually) from a byte array point of view ? so lets say ive got 2 arrays of audio data, and each array is 10 bytes long:

array1 -> 0x01, 0x02, 0x03, 0x04, 0x05, 0x01, 0x02, 0x03, 0x04, 0x05
array2 -> 0x06, 0x05, 0x09, 0x05, 0x04, 0x10, 0x0A, 0x06, 0x0B, 0x03

what i want to do is to blend/combine/merge these into one array. note i dont want to concatenate one onto the other, to give me a 20 byte array, i want to play both arrays at the same time. for example, if array one was someone saying 'hello', and array 2 was someone saying 'goodbye', then in the merged array you would hear both people saying 'hello' and 'goodbye' at the same time. is this even possible ?

a bit of background on what im doing - im making a VOIP system using the WAV api in a C# app. im using the GSM610 codec, and the result is that the average bytes per second recorded from the microphone is 6500 bytes. because the buffer size is 650 bytes, so each client sends 650 bytes of voice data to the server every 10th of a second, thus resulting in 6500 bytes per sec.

the server must then forward these 650 bytes onto all other clients. so if only 2 clients are connected to the server, this is fine, because client A sends 650 bytes to the server, and the server forwards these onto client B. in this case client B is playing client A's voice data at the same rate as client A recorded it. however when a third client C is introduced, now both client A and client C are sending 650 bytes to client B = 1300 bytes. if another client connected, this would now be 650 * 3 = 1950 bytes and so on....but each client only processes 6500 bytes per second, hence things start to sound wrong.

so i need a way of merging each set of 650 bytes into one set of 650 bytes, and then send this merged set to each client. is this performed using a bitwise operation on each byte for example ?

How to decode a sound track

Sun, 31 May 2009 07:10:12 -0700

Hi
I really want to know when we record a sound in the computer how does it change into binary codes after that I want to know how can I decode mp3 format for making changes in the sound(In c++ preferred).

Thanks a lot

MP3 Classes for C#

Sat, 21 Feb 2009 19:59:04 -0700

Hello. Newbie here, so I apolozie for the simplistic question. I'm trying to write what I thought would be a very simple C# application to replace an MP3's title based on it's file name. I don't have time to write my own classes to access the ID3 tags, so I was hoping to find something on line to allow me to do this. I've tried several that I've found, but none of them seem to work correctly. The closest I've gotten is with taglib-sharp, but I'm not sure that's working for me.....it seems to process the files the first time I run it, but when I try to run it on some of the same files a 2nd time I get an error. That indicates to me that maybe something is getting hosed up when I save them. Can anyone reccomend some pre-built classes I can use?

Coding a Sampler

Wed, 11 Feb 2009 14:08:10 -0700

Hello,
I'd like to create a software sampler. I am a vb.net programmer with a little c# experience as well. I've been doing allot of researching and I'm a little confused on which 3rd party plugin to use. DirectSound, DirectMusic, ChucK, the list goes on and on...
I was excited when I came across DirectMusic, but it appears thats no longer used??? I'm not sure. What's the top of the line technology for developing a music player with all the basic music player functions today?

I have some specific needs, but the basics are:
- Play various audio file formats (mainly wav & mp3).
- volume control, pan, output meters
- ability to change the lows, mids & hi freq.
- ability to view the wave form.
- ability to slow down / speed up the music.
- midi functionality... meaning the ability to trigger a sample using a midi interface & controller.

Any guidance/direction would be greatly appreciated... Thanks!

Problem with soundcard - Desktop Konnekt 6

Wed, 07 Jan 2009 02:29:22 -0700

Hi there!
I bought my T.C electronics Desktop Konnekt 6 last week and have this far spent at least 40 hours trying to get the sodding thing to work.

I have OS: Windows Vista 32-bits on a Dell Studio 1737.

This is the problem I've been experiencing:
At first, I couldn't even make the computer detect the soundcard. Even though i had installed the CD correctly and plugged in the firewire cable. But then, dont ask me how, I managed to get the computer to detect the external soundcard, maybe because of the many uppdates for drivers and software I had downloaded.
So at this point I have made sure I have all the latest uppdatetes for the soundcard and for the sounddriver in the computer, but as fast as I try to play something in the speakers connected to the soundcard, I just get an errormessage.
And yes i have been in the controlpanel and made sure everything is correctly set. I even tried to turn of all other soundsources in the Device Manager but nothing works.

It seems as Cubase (a soundediting program) isn't having any problems connecting with the soundcard. I've succeded to record music from instruments that I've plugged inte to soundcard and have even sometimes managed to get the sound from the playback in Cubase to stream out of the speakers connected to the soundcard. But this doesn't either work all the time, it's like a broken watch: sometimes going, sometimes not.

Is there anyone else that's experiencing the same problem, or having any solution to offer?

I started thinking that it might have something to do with the Firewire Chipset in the computer, because it's a firewire soundcard.
I checked wich chipset I had and the result was:
RICOH OHCI Compliant IEEE 1394 Host Controller
And as i checked on some other forums I saw that this chipset, Ricoh, is known to cause alot of trouble, is this something i should be worried about?

Im very greatfull for all responses!

Recording Sound and encoding to g711

Wed, 17 Sep 2008 06:49:31 -0700

Hi@all,

I recording sound with Windows Media Library in Waveform. Now I want to encode the recorded data into g711. Anyone here who can tell me what I have to do or knows a good tutorial or manual to this?

Greetings
CrazyPlaya

hi im new!! media player design? help!!!

Sun, 14 Sep 2008 11:07:50 -0700

im new to this whole thing so i want to give it a try...

basicly im wanting to design a media player

any one have any ideas where i can start what programs i need to use..

the media player will need to work on xp operating system..

i have uploaded a pic of what i want it to be like..

any help would be greatly appreciated

Attachment: 23773_large.jpg (71661 bytes | downloaded 119 times)

You tube Video Downloads

Tue, 09 Sep 2008 04:05:58 -0700

hallo i am looking for a software to download youtube videos in
my pc and convert them to mp3 to listen them in my ipod too, can
you help me pls?

Convert raw drive data to PCM wav

Tue, 29 Apr 2008 10:21:32 -0700

I noticed a thread about converting wav to binary...

Im wondering if anyone can help me as I need to go the other way:

I have a Hard drive from a digital recording desk containing deleted audio.
The machine records in 16bit/44.1KHz PCM but works on its own OS and has no such features or interfaces to restore deleted content back to its previous TOC state... so the audio data is there, but inaccessable because the machine has allocated that space as empty due to (wrongly) deleting those tracks...

The drive is IDE so I can plug it into the PC and read the raw binary data with a number of generic hex/disk editors but what I need is a method or program that can find the PCM strings and let me make wavs...

can anyone help me here?

Win32 API waveIn recording issue

Wed, 02 Apr 2008 09:46:48 -0700

Hi,

I am having some difficulty comprehending the result of the Win32API waveIn.

As you know, to receive the input from the microphone , you have to call waveInOpen, waveInStart and to add some buffers in the queue.
When the buffer is filled , the appropriate callback function is called.

I tried in my program to see how often the function is called compared to the number of seconds passed.

The WaveFormatEx I specified was 8192 samples , 16 bits and 1 channel . The buffer's length was 8192 bytes. Therefore , I expected that the callback function be called twice / second , bringing 4096 samples each time.
However , I discovered that the function was called about 3.1 times / second, which is weird.

Any idea why this might happen?
What does the other data mean? Must it be taken in account when doing the Fourier analysis?

Thanks a lot,
George

Cross Platform MP3 Manipulation

Sun, 09 Mar 2008 10:49:09 -0700

Hello,

I am planning to write a cross platform mp3 player for musicians. It will need to be able to slow down an mp3 without changing it's pitch and loop a section. I am considering using JMF(Java media platform). It appears to have the methods i will need to accomplish this, but i was wondering if there is anything else available. And also if JMF has a reputation of good performance. Any information or thoughts on thesubject would be great. Thanks alot.

"The world is a tragedy when you feel, but a comedy when you think."

Measuring Line In or Mic input voltage

Mon, 07 Jan 2008 09:01:30 -0700

Is there a way to measure the voltage from the line in or mic inputs in a sound card?

Win32 sound program written for command line?

Fri, 28 Dec 2007 14:40:31 -0700

Need help getting started with a small development project for writing to a sound device in windows:

I've got a need to write a stream of bytes (8-bit samples) to windows audio output port. These bytes will arrive in UDP datagrams (rather than a file), as this is part of an audio-over-LAN experiment. I expect to develop the UDP datagram receiver as part of this, but I want to start with the sound device interface because finding a development environment (compiler/linker) with sound device library API seems like the critical path.

From the UDP datagrams, I receive a stream of raw audio samples (each 8 bits unsigned) that I want to send to a windows sound device, so I need to develop a program that opens the windows default sound device, sets it to 8 bit samples at a given sample rate (say 8000 Hz), and let me write sample bytes to the device.

IMPORTANT - I don't need a GUI at all - in fact the ideal solution would be a program written in C or C++ that can be compiled and run from the windows command line. I am thinking this would make the overall application simpler, as I don't want to delve into Windows GUI application development for this experiment.

I also do not have a development system (compiler/linker) in mind yet, but would need one for this purpose. I've got my trivial application working in linux (using one of the alsa utilities, 'aplay', as my starting point.) If there was something like 'aplay' for windows (I realize alsa itself is not part of windows, so the similarity would end there), that would be ideal.

Very tks,

Dave

Playsound replays old sounds after new sounds

Thu, 23 Aug 2007 13:46:42 -0700

In a c++ program, I play a series of wave files taken from a list of filenames using playsound. Each wave file plays for less than 1 second. After playing a series of up to 5 files, there is a 10 to 30 second delay in which no sound is played, and then another series of files is played. My playsound function call looks like this:

PlaySound(FilePathName, NULL, SND_SYNC | SND_FILENAME | SND_NODEFAULT)

My problem is that trailing sounds from a previous longer series play after a subsequent shorter series completes. Apparently what isn't overwritten in the sound buffer gets played.

Is there a way to clear the sound buffer of older wave format data or otherwise stop play at the end of a new list? (Obviously, the SND_PURGE flag does not and is not meant to accomplish this.)

Using DirectSound on a WebPage

Mon, 30 Jul 2007 04:11:00 -0700

Hi all, this looks like a great forum to solve my dilema. Looking to utilize DirectSound for audio playback/capture but would like to shy away from WinForms and make it a web app. Looks like DirectSound needs a form object, am I completely oblivious, is there a workaround? WOuld I need to create an ActiveX Object? Not sure if there are any examples that would help. Any help would be greatly appreciated.

Does anybody know ho w to program CASIO WK-3200 KEYBOARDS

Wed, 25 Jul 2007 04:38:56 -0700

I have software downloaded from the CASIO website and I want to upgrade it to improve the quality of the WK-3200...I have WInDOWS VISTA...I'm new at this...Where do I start...PEACE and BLESSING...

Free Multimedia Ambient Loops

Sat, 30 Jun 2007 10:58:29 -0700

Hi everyone,

I have uploaded a free sound pack containing ambient audio loops for use in your projects. You can find them on the Free Multimedia Sounds section , here is the link:

http://www.equinoxsounds.com/products.htm

Sound programming problem

Mon, 04 Jun 2007 10:06:30 -0700

Hi friends
I am trying to make sound recorder in C at first I only want to sample bytes but the the problem is that when I give command 20h to dsp the bit 7 of port Eh does not set please tell me how to solve this problem?

amplitude of wav

Sun, 27 May 2007 04:06:39 -0700

Hello all. How i can change amplitude of .wav file in C++?

alert: need help....

Wed, 14 Feb 2007 17:19:00 -0700

first and foremost, i would like to say hello to all programmers!

i would like to present to you myself, a first year programming student. just call me "RED3WARLORD"...

i would like to ask for favor if you don't mind.....

could anyone introduce me this programming language or should i say, this type of programming? i just want to extend my knowledge about programming....

respectfully yours,

red3warlord with gratitude

sinustone

Tue, 30 Jan 2007 15:49:56 -0700

Hi!
Can someone show me how to program a tone, say a sinustone, through my PC? Either in C, C++, or Pascal. And what parameter to change the frequenzy?
Thanks!

Sound card programming

Wed, 29 Nov 2006 23:01:01 -0700

Hello Everybody,

I'm new to programming. I have knowledge of C and C++.Now, I need to generate a 10KHz audio signal from a sound card and thereafter perform signal processing at the receiving end.

I'm not able to understand how to start.Can anybody give similiar examples which could help me out?

Thanking you,
vasu.

How to do Fast Fourier Transform for .wav files?

Mon, 23 Oct 2006 11:06:19 -0700

My objective is to make a speaker recognition program using Delphi.

So far, I have managed to record my voice through a microphone into a .wav file. However, I get confused about how to process the .wav file using FFT to get the frequency domain. I have read several papers about FFT, yet they are too theoritical and when I find it difficult to put it into Delphi codes.

Any help will be greatly appreciated.

Windows MIDI instrument definitions?

Mon, 16 Oct 2006 01:52:28 -0700

Hi all!

I'm programming a DOS game in Borland Pascal 7.0 and have found a pretty good unit for playing MIDI files:

http://www.shdon.com/view?cat=dos&doc=sound

However, the MIDIs don't sound exactly like they do in Windows. The author of the unit is aware of this and therefor also supplies an instrument editor for the FM.DAT file.

Does any of you know how I should edit the instruments to make the MIDIs played with this unit sound like they do when played in Windows? Where can I find the Windows instrument definitions so I can edit the instruments in the FM.DAT file to match the Windows instruments?