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A Tribute to NY Giants by Tanya T6

Sat, 04 Feb 2012 15:07:28 -0800

New York Singer/Songwriter Tanya T6 releases tribute anthem in support of the NFC Champions, New York Giants.
New York Singer/Songwriter Tanya “T6” Dormevil has just released a song called "NY (Hands High)", a tribute song that demonstrates her love and support for her hometown.
Following the New York Giants 20-17 victory in overtime over the San Francisco 49ers for the NFC title, R&B Singer/Songwriter Tanya “T6” Dormevil released “NY (HANDS HIGH)” a tribute song that demonstrates her love and support for her hometown New York and a Giants team that no one saw coming but has found its way to the Super Bowl.
The song “NY (Hands High)” was written by Tanya T6 and produced by J-rum Beats. Tanya has been working diligently behind the scenes with several artists such as Jae Millz, Mims, NEYO, Royce Da 59, Max B, French Montana, and DJ Webstar. Tanya is currently recording a new EP series; a follow-up to her debut EP entitled "Pieces". Fans can expect this exciting new EP to be released spring 2012.
View NY (Hands High) - http://bit.ly/AaQ8eO
Visit:
http:/www.associates-firm.net
http://www.thebestoftanya.com

Effects of Audio Mastering

Thu, 24 Mar 2011 07:05:29 -0800

Music drives a nation. When the music reaches hearts, it binds. To make the effect of music last forever, it is essential to add finishing to the final track. Mastering is the process of preparing and transferring the source code having the recorded audio file to the master, which is a data storage device. The master is then used for developing copies via replication or duplication and pressing. The format of choice is analog or digital. Mastering is necessary for the perfection that is attained when the audio files are mastered. When the audio file is mastered and played on radio, the effect of mastering becomes obvious. The mastering services are especially important when there is more than one track in the album. With mastering, the gaps between the sings are adjusted while the intros and the end are fixed. The musical instruments used in the track are made clearer and louder. When the mastered music is played on radio, it is more audible and professional. The popularity of the mastered music has made the services of online mastering, highly useful. This is evident in the amount of convenience and affordability it offers. Being economical and quick does not mean that the quality of the mastered audio files is any less. The mastering is done with the same set of equipments that are used at attended mastering. The quality is good enough for the mastered audio files or tracks to be played on any medium and not just the one on which it is formed.

Read the Apple Final Cut Studio Soundtrack Pro to Win the Free Gift!

Mon, 21 Mar 2011 23:29:23 -0800

Soundtrack Pro 3 helps you edit and mix faster than ever, with new multitrack editing tools and easy ways to fix common audio problems. Whether you’re enhancing a single clip or crafting a feature-length soundtrack, you’ll find all the tools you need to create perfect sound for picture.

Powerful audio editing

A full range of professional features is at your fingertips in Soundtrack Pro, including tools designed specifically for audio post. Work fast with new navigation shortcuts in the multitrack interface, and edit files with precision in the File Editor.

Soundtrack Pro 3 makes it easy to craft perfect dialogue tracks. Use the Multitake Editor to record and find the best takes, and tap into a variety of innovative tools to match tone and level across your entire project in just a few clicks.

Special-purpose tools in Soundtrack Pro 3 let you repair common audio problems in seconds. Remove unwanted background noise in a few simple steps. You can also quickly identify, preview, and fix audio that contains pops, clicks, hum, DC offset, and more.

With more than 50 professional effect plug-ins and over 6000 royalty-free Foley effects, sound effects, and music beds, Soundtrack Pro 3 makes it easy to design the perfect sound to enhance any project.

Easily synchronize your soundtrack to changes made in your video edit. If you change your picture edit in Final Cut Pro, you can use the Conform feature in Soundtrack Pro to automatically compare versions and update your multitrack audio project.

Soundtrack Pro 3 gives you everything you need to create immersive 5.1 surround for your video. Whether you’re mixing an independent film or a multimillion-dollar blockbuster, creating an enveloping cinematic soundtrack has never been easier.

Tackle the most complex audio post-production tasks, with the flexibility to work the way you want. Soundtrack Pro 3 combines powerful integration with Final Cut Studio applications and seamless interchange with other audio software.

Plus: If you like this post in programmersheaven.com, you can get a free license code of HD Video Converter Factory Pro(the original price is 29.95$) by following some easy and interesting rules:

1. Firstly, Leave a comment here
2. Secondly, Subscribe to programmersheaven.com via email or via Rss.
3. Thirdly, Share this post in your facebook account or tweet this post on twitter (optional)
4. Lastly, after you have done the job that are requred above, you just need to contact me via this email:

cklarry6@gmail.com

. Then I will give your the free license code via your email!

I hope all of you will get the free code, this activity will end till March 21, 2011

I hope and I belive I will be happy all the time!

A Collection of Useful C++ Classes for Digital Signal Processing

Thu, 03 Mar 2011 08:47:44 -0800

Greetings. I would like to offer you my latest creation, an open source MIT-licensed C++ library for implementing IIR (Infinite Impulse Response) digital filters. It is fully documented, and even comes with a graphical/audio demo to showcase it's features. It uses templates. Here it is;

http://code.google.com/p/dspfilterscpp/

"A Collection of Useful C++ Classes for Digital Signal Processing"

By Vincent Falco

"Techniques for digital signal processing are well guarded and held close to the chest, as they have valuable applications for multimedia content. The black art of Infinite Impulse Response ("IIR") filtering has remained shrouded in secrecy with little publicly available source code....until now."

Building on the work of cherished luminaries such as Sophocles Orfanidis, Andreas Antoniou, Martin Holters, and Udo Zolzer, this library harnesses the power of C++ templates to solve a useful problem in digital signal processing: the realization of multichannel IIR filters of arbitrary order and prescribed specifications with various properties such as Butterworth, Chebyshev, Elliptic, and Optimum-L (Legendre) responses. The library is provided under the MIT license and is therefore fully compatible with proprietary usage.

Classes are designed as independent re-usable building blocks. Use some or all of the provided features, or extend the functionality by writing your own objects that plug into the robust framework. Only the code that you need will get linked into your application. Here's a list of features:

* Exclusive focus on IIR filters instead of boring FIR filters
* Complete implementation of all "RBJ Biquad" Cookbook filter formulas
* Butterworth, Chebyshev, Elliptic, Bessel, Legendre designs
* Low Pass, High Pass, Band Pass, Band Stop transformations
* Low, High, and Band Shelf filter implementations for most types
* Smooth interpolation of filter settings, pole/zeros, and biquad coefficients to achieve seamless parameter changes
* Representation of digital filters using poles and zeros
* Realization using Direct Form I, Direct Form II, or user provided class
* Fully factored to minimize template instantiations
* "Design" layer provides runtime introspection into a filter
* Utility template functions for manipulating buffers of sample data
* No calls to malloc or new, great for embedded systems
* No external dependencies, just the standard C++ library!
* Documentation!

Using these filters is easy:

    // Create a Chebyshev type I Band Stop filter of order 3
    // with state for processing 2 channels of audio.
    Dsp::SimpleFilter <Dsp::ChebyshevI::BandStop <3>, 2> f;
    f.setup (3,    // order
             44100,// sample rate
             4000, // center frequency
             880,  // band width
             1);   // ripple dB
    f.process (numSamples, arrayOfChannels);

An accompanying demonstration program that works on most popular platforms by using the separately licensed Juce application framework (included), exercises all the functionality of the library, including these features:

* Dynamic interface creates itself using filter introspection capabilities
* Audio playback with real time application of a selected filter
* Live time stretching and amplitude modulation without clicks or popping
* Charts to show magnitude, phase response and pole/zero placement
* Thread safety "best practices" for audio applications

Here's a screenshot of the DspFilters Demo

http://dspfilterscpp.googlecode.com/files/dspfiltersdemo.png

If you've been searching in futility on the Internet for some source code for implementing high order filters, then look no further because this is it! Whether you are a student of C++ or digital signal processing, a writer of audio plugins, or even a VST synthesizer coder, "A Collection of Useful C++ Classes for Digital Signal Processing" might have something for you!

features of an audio

Wed, 29 Dec 2010 01:55:20 -0800

hai all,
I am in need of extracting audiofingerprint which is features of an audio.
can any body provide source code or frame work of procedure, since its very urgent.

Thanking you

Audio broadcast crossfire

Wed, 12 May 2010 18:51:47 -0800

Hi, i'm encountering an error where audio streaming crossfire will occur randomly when there are 2 sessions taking place. my system is created by c# codes, and uses udp to transmit data to multicast ip. both sessions points to same ip but different ports.

CD player with JUST S/PDIF output?

Wed, 03 Feb 2010 18:34:06 -0800

Hey Everyone,

I'm looking to buy a dedicated CD player for use with my new receiver, I only search it in www.ebay.com and www.sourcinggate.com. .I only need it to have digital output, not any inbuilt DAC/RCA-out functionality. I'd prefer the CD player to be quite small (at least in height), so that it perches on top of the receiver.

Does such a CD player exist? I've looked about, but most of the main companies seem to sell high-end CD players with digital output as an extra to high quality RCA output, which I don't need, and which I suspect add a great deal of bulk to the device.

Any recommendations would be appeciated, and I'm happy to look at older hardware, including stuff available only second-hand.

Thanks!

Encryption of wav files by converting into text format and playing

Sun, 29 Nov 2009 10:58:05 -0800

Hey all ,

I have to convert wav files into text format to encrypt using Identity based encryption (using AES algorithm) and then decrypt that file.The resulted text should be converted so that it is playeable as a wav file again.

Could you please help me on how to perform these conversions viz wav to text and text to wav .

Thanks
regards
bk

fourier transform

Tue, 03 Nov 2009 13:33:24 -0800

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.

Simply play audio?

Tue, 14 Jul 2009 20:48:45 -0800

Is there a simple/efficient way to play a frequency at a volume using c++? I need a way to play sound through the speakers so I don't think Beep() will do the job, but I want to have access to the audio in the memory so I don't thing PlaySound() will work either. I do not want to use any libraries either. I just want to play a frequency, I can load the input file myself. Is there some way I can do this by just using the libraries that come standard on windows XP OS?

mp3 audio xing header TOC specification/information

Sun, 28 Jun 2009 05:03:25 -0800

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 -0800

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 -0800

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 -0800

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 -0800

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 -0800

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 -0800

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 -0800

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 275 times)

Convert raw drive data to PCM wav

Tue, 29 Apr 2008 10:21:32 -0800

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 -0800

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 -0800

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 -0800

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 -0800

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 -0800

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 -0800

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.