I was wondering whether anyone here had any recommendations for methods of improving speaker audio performance with the use of a DSP...

I was wondering whether anyone here had any recommendations for methods of improving speaker audio performance with the use of a DSP (digital signal processor). I have heard of different methods of improving the system. I have heard that you can make measurements of the loudspeaker excursion limits and use these parameters to control a dsp compressor to get a lot of performance and volume...

Hi Friends, I am a new bee in embedded systems. My task is to test a filter(low pass) with audio in and out connections of hardware board. I...

Hi Friends, I am a new bee in embedded systems. My task is to test a filter(low pass) with audio in and out connections of hardware board. I have developed a low pass filter in C and tested it using goldwave(signal processing tool) and need guidance on how to test the same filter with audio in and out connections of hardware. Thanks in advance. Aamer

Hello, I would like to apply an FIR filter (at least 512 points) to a 1D array of audio data (512 points). This convolution would take place...

Hello, I would like to apply an FIR filter (at least 512 points) to a 1D array of audio data (512 points). This convolution would take place within a audio callback function so the input and output data arrays both need to be the same length. My problem is (if I understand correctly) convolving the filter kernel and input data would result in an output signal N + M -1 long, i.e. different i...

I want to extract audio from a computer, pc or Mac to start with (maybe later on Smartphones, tablets). I may want to have start/stop recording...

I want to extract audio from a computer, pc or Mac to start with (maybe later on Smartphones, tablets). I may want to have start/stop recording button as well. There seems a lot of APIs - libspeex API, PortAudio API ... Xuggle API...? I am wondering what might be an optimal choice - better performance - trouble free implementation? Currently the preference is to extract in wav format....

I've been extremely interested in Spectral Analysis & Resynthesis of Audio Signals for as long as I've known of it. The idea of understanding...

I've been extremely interested in Spectral Analysis & Resynthesis of Audio Signals for as long as I've known of it. The idea of understanding and manipulating a timbre in terms of basic, intuitive building blocks seems like the most useful tool imaginable for sound design. There are a number of computer applications which combine spectral analysis, resynthesis, and spectral manipulation. My f...

I have given a audio file (sine wave)800 Hz as an input to my FFT algorithm.I have got 8192 power spectrum samples in a array .What is the best...

I have given a audio file (sine wave)800 Hz as an input to my FFT algorithm.I have got 8192 power spectrum samples in a array .What is the best and easy way to check whether my output is right or wrong.If i give a silent audio file. The out out is zero for all samples.In a sine wav the o/p increases from 20 (0th sample )to 26059811(743 sample) and decreases gradually to 40.Any ideas will be more h...

Hi I want to use microphone array for DoA estimation.For DoA estimation,I require the data (which will be processed to find angle of arrival).I...

Hi I want to use microphone array for DoA estimation.For DoA estimation,I require the data (which will be processed to find angle of arrival).I want to know how to take input audio data on PC .On my LAPTOP,i have only one microphone input so how can I get multiple channels on single microphone input. Thanks Salrome _____________________________________ Do you know a company who emplo...

Hello Everyone, I am developing an audio application using Blackfin BF537 EzKit lite. I have some problems with DAC 1854 Sport interface. I...

Hello Everyone, I am developing an audio application using Blackfin BF537 EzKit lite. I have some problems with DAC 1854 Sport interface. I want to pass the audio data samples via SPORT interface to DAC 1854 without using DMA. I am writing to Sport 0 Tx register on receiving the SPORT0_TX interrup as follows *pSPORT_TX=*(audio_data); Should I need wait to check any status register of SPO...

Hi! Please help to choose a dsp chip for a audio processing project. I want to make a kind of effect box with reverberation or surround...

Hi! Please help to choose a dsp chip for a audio processing project. I want to make a kind of effect box with reverberation or surround algorithm inside. Since have experience mostly in dsp programming area, I need an all-in-one solution including the dsp, memory, adc/dac etc, to make board design as simple as possible. The requirements are: 2-3ch ADC, 2-8ch DAC digital in/out (optional) ...

How can I converto a 4 bit/sample raw audio file to an 8 bit/sample file?

How can I converto a 4 bit/sample raw audio file to an 8 bit/sample file?

Hi everybody I am trying to enhance the low freq (

Hi everybody I am trying to enhance the low freq (

Hi all, I'm having a few problems with the basics of IFFT. I have created a frequency repsonse (which is an acoustical model of the response of...

Hi all, I'm having a few problems with the basics of IFFT. I have created a frequency repsonse (which is an acoustical model of the response of a room). I want to convolve this with an audio file - thereby making the audio sound like it was played in the room. However, I'm having problems with the IFFT and time domain issues. My frequency response is originally 600 points long, representing ...

Hi, I have an hardly used Lyrtech PADK for sale. Also have a XDS510 USB JTAG interface that goes with it. For more info on the PADK...

Hi, I have an hardly used Lyrtech PADK for sale. Also have a XDS510 USB JTAG interface that goes with it. For more info on the PADK see: http://focus.ti.com/docs/toolsw/folders/print/tmdspdk6727.html and http://www.lyrtech.com/Products/PADK.php It's a great Development kit with all possible audio and digital interfaces included (24-bit four-channel ADCs (2×) and DACs (2×) up to 192 kHz, Co...

Hi, I'm using the C6416 DSK with the DSignT DSK-91C111 Ethernet daughter card. In fact this card just connects an SMSC 91C111 MAC+PHY to...

Hi, I'm using the C6416 DSK with the DSignT DSK-91C111 Ethernet daughter card. In fact this card just connects an SMSC 91C111 MAC+PHY to the DSP's EMIFA and an interrupt line to one of the DSP's external interrupt pins. What I want to do is to receive audio data via ethernet and send it to the board's audio output. So at the moment, whenever I get an HWI from the MAC I start a service routi...

* Does anyone know of source code for decoding Biphase/Manchester encoding from analog signal? I am looking for source code for a biphase...

* Does anyone know of source code for decoding Biphase/Manchester encoding from analog signal? I am looking for source code for a biphase decoder. The application is extracting SMPTE/EBU time code data from analog audio signals (recorded by professional audio/video equipment). I found source code that looks at signal crossing an adjustable threshold (by Maarten de Boer) ftp://www.iua...

I want to convert some .WAV files to a delta-sigma bitstream. The .WAV files are typical 2 channel 16 bits per sample 44kHz audio files. I'd...

I want to convert some .WAV files to a delta-sigma bitstream. The .WAV files are typical 2 channel 16 bits per sample 44kHz audio files. I'd like to use some kind of tool (linux preferably, windows if necessary) to transform them into a single channel delta-sigma 8x oversampled bit stream, and preserve as much of the original audio spectrum below 6kHz as possible. This is for output from a mi...

I am having some trouble receiving an audio signal through the MIC input of the AIC23 part of the EVMDM642 board? I am trying to use the...

I am having some trouble receiving an audio signal through the MIC input of the AIC23 part of the EVMDM642 board? I am trying to use the "audio echo" example and have changed the INSEL register in evmdm642_aic23.h to select the MIC input instead of Line in, but still I get no signal from the MIC (and the LINE-IN still works as an input!). Does this indicate that I need to do something else ...

I have to design a simple low pass filter (for audio) to reject frequencies beyond 3.6kHz. The design of the filter is quite simple. However, I...

I have to design a simple low pass filter (for audio) to reject frequencies beyond 3.6kHz. The design of the filter is quite simple. However, I want to know what affect the sampling frequency of the audio will have on such a filter design. In other words, I can sample at 16kHz, filter the data and downsample to 8kHz (which is finally the sampling rate I want) OR I can sample at 8kHz and fi...

I have to design a simple low pass filter (for audio) to reject frequencies beyond 3.6kHz. The design of the filter is quite simple. However, I...

I have to design a simple low pass filter (for audio) to reject frequencies beyond 3.6kHz. The design of the filter is quite simple. However, I want to know what affect the sampling frequency of the audio will have on such a filter design. In other words, I can sample at 16kHz, filter the data and downsample to 8kHz (which is finally the sampling rate I want) OR I can sample at 8kHz and fi...

Hi, I am currently investigating the use of fractional delay filters for the asynchronous sample rate conversion of audio signals. I'm getting...

Hi, I am currently investigating the use of fractional delay filters for the asynchronous sample rate conversion of audio signals. I'm getting quite confused with the various methods and hopefully some of you guys could help. First of all, let's make it clear that this is for 'asynchronous' conversion. Nothing is known about the input to output ratio, except that it's within a 'normal aud...