Audio Player With EQ OpenAL (Audio, C++ / C, Digital, Other (Programming))

Audio Player With EQ OpenAL

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Job Summary:

Job Type:	Project
Budget:	$ 20-100
Required Skills:	Audio, C++ / C, Digital, Other (Programming)
Attached Files:	(None)

Status:	Closed for bidding

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Pullman, WA, United States

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November 14, 2008

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Description

I need a C++ class written. The project should be a very simple one for someone with the right knowledge. I'm guessing it could be as simple as a single day project. There is a lot of example code out there that does what I want (see links below), but I just don't know how to put it together.

I need a fairly simple OpenAL 1.1 based audio player class with a real-time EQ filter built in. It should take streaming samples of data from a stereo OpenAL source (i.e. a wav file) and apply up to 100 bands of active "bandpass" equalization in real-time. Then it should feed the new "filtered" samples into alSourceQueueBuffer().

The EQ is essentially a 100 band parametric EQ, yet there are no graphics to it.

The player should play a stereo wave file, thus disabling the 3D spatialization effects of OpenAL.

SoundSource.cpp

I want to be able to make a call to play the sound and just pass the following parameters:

1. wave file path

2. volume level

example call:

SoundSource1.playSound("c:\test.wav", 0.95);

I also need a function within the class that I can call to continuously update the frequency values. I will pass this function a 3D array of values representing the required EQ parameters for playback

- 100 frequency values in Hz- i.e. 100, 300, 500, 700 ...

- 100 db Gain setpoints that correspond to the desired volume level of each
frequency band - i.e.: -3.00, 4.50, 0.76, 12.81 ...

- 100 "Q" values for the width of each individual filter

example call:

freqValues[100][100][100];

SoundSource1.EqUpdate(freqValues);

_________________________

Internally the class should do the following:

- instantiate an openAL sound source and start streaming playback immediately

- Apply real-time EQ to the buffered audio samples just before they are being queued for playback (i.e. just prior to the alSourceQueueBuffer() call)

- The function that applies the EQ will take the freqValues[100][100][100] array and manipulate the buffer of sampled audio data (I'm guessing through convolution) such that up to 100 narrow bands of EQ are applied.

- handle memory management properly

__________________________

The winning bidder will be someone who has a good understanding of the following:

- C and C++
- OpenAL
- digital signal processing
- digital filtering

I need extensive code commenting that explains exactly what the code is doing, in case I need to modify anything.

Misc. notes:
Due to the number of filters applied, I'm concerned about the processor burden, so some of the specs might have to be adjusted to balance functionality with the processor (i.e. less frequency values). I don't think it's possible or efficient to do real-time FFTs to the signal, so the winning bidder should know how to avoid that approach. I'm assuming that convolution will be the solution.

Here are some links to example code:

http://www.devmaster.net/articles/openal-tutorials/lesson5.php

http://www.musicdsp.org/files/biquad.c

http://musicdsp.org/archive.php?classid=3

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$150.00

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1 Day(s)

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Dear Mr.Freeballs , We are ready to start up with your Project with our excellent team of programmers, Designers & Project Managers. We are 100% confident for a successful delivery of your Project. We are providing our Services to many Big Client Across Globe & are looking for an Excellent Long term Business Relationship with your Esteemed Organization. Please Check your PMB for our company Profile & Our Portfolio. Warm regards
Bid Time: 11-18-2008 12:13

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