I am new to audio processing. What I wanted to do is amplify certain frequencies in my data. I have written an application in c++ with a...

I am new to audio processing. What I wanted to do is amplify certain frequencies in my data. I have written an application in c++ with a callback function that gets the audio data to a buffer. Now I need to apply some function to this data to amplify my frequencies. Can I do this using a filter? Is there an existing code I can use? Can someone point me in the right direction ? ...

Hi, I have created a tutorial that helps you to understand transmission lines, the reflection coefficient and VSWR. It is an...

Hi, I have created a tutorial that helps you to understand transmission lines, the reflection coefficient and VSWR. It is an interactive flash program with embedded audio lectures. It is located here: http://www.fourier-series.com/rf-concepts/reflection.html It is about a 10M download, due to a lot of audio embedded into the program.

What is the theoretically highest possible: 1. Frequency Response (range of frequencies) 2. SNR 3. Dynamic Range 4. SPL 5. Musical...

What is the theoretically highest possible: 1. Frequency Response (range of frequencies) 2. SNR 3. Dynamic Range 4. SPL 5. Musical Pitch (highest acceptable frequency) of PCM audio?

Hi there, Okay so I had already put up a post how in order to find the spectrogram of an audio file, I was going to perform a series of FFT's...

Hi there, Okay so I had already put up a post how in order to find the spectrogram of an audio file, I was going to perform a series of FFT's and from these compute the spectrogram.. I was confused about whether to use real fft or complex etc.. so I used the real fft function from kiss_fft, i.e. kiss_fftr( ) The spectrogram I get has the right shape overall, however it looks terribly nois...

I didn't cross-post it, and I won't multiple-post it, but some of you may have heard of Tom Stockham (pioneering applier of...

I didn't cross-post it, and I won't multiple-post it, but some of you may have heard of Tom Stockham (pioneering applier of blind-deconvolution techniques and Other Matters), and might like to see memorial remarks that I and others posted recently on rec.audio.pro, where I first saw this news. (I cited certain other highly-technical innovators also, and a little DSP history, in my own message...

Hi, I am implementing an audio pattern recognition application, the pattern recognition is done using persons correlation coefficient on...

Hi, I am implementing an audio pattern recognition application, the pattern recognition is done using persons correlation coefficient on a set of N dimensional feature vectors. I have considered usage of PCA (Principal components analysis) for dimensionality reduction ( as done in some speech 2 text applications ). PCA take the most dominant dimensions of a collection of N dimensional...

Hi, Many audio codec uses sigma-delta technology. I do not understand why there are so many bits, which are far more than the SNR spec. For...

Hi, Many audio codec uses sigma-delta technology. I do not understand why there are so many bits, which are far more than the SNR spec. For example, ADI has ADAU1761 which is 24-bit, 98 dB SNR. Could you tell me that? Thanks.

We have several needs for DSP engineers in Portland, Oregon. The positions range from mid-level to senior engineers with a focus on audio...

We have several needs for DSP engineers in Portland, Oregon. The positions range from mid-level to senior engineers with a focus on audio related product development. Must have a range of audio compression standards experience (AC3, MP3, AAC, etc.) Some relocation assistance within the USA or Canada is available for the right candidate. US Citizen, EAD or Permanent Resident status is preferred...

Hi Everyone, I'm a newcomer to DSP and am trying to write a Fixed Point implementation of the bilinear transform lowpass filter as outlined...

Hi Everyone, I'm a newcomer to DSP and am trying to write a Fixed Point implementation of the bilinear transform lowpass filter as outlined in the Audio EQ Cookbook (http://www.harmony-central.com/Computer/Programming/Audio-EQ-Cookbook.txt): y[n] = (b0/a0)*x[n] + (b1/a0)*x[n-1] + (b2/a0)*x[n-2] - (a1/a0)*y[n-1] - (a2/a0)*y[n-2] where for a lowpass filter: b0 = (1 - ...

Is there a reason (founded in DSP theory) to resample in stages instead of going directly from one sample rate to another? If so what is the...

Is there a reason (founded in DSP theory) to resample in stages instead of going directly from one sample rate to another? If so what is the reason? And where can I read about it? For example: When I resample from 8kHz to 48kHz the resulting audio sounds worse compared to the audio I get if I resample in stages from 8kHz to 24kHz to 48kHz. Thanks

On Sep 11, 8:37 pm, Jerry Avins wrote http://groups.google.com/group/comp.dsp/msg/b1ef0c8f50008e77?hl=en& : > He's > contemplating...

On Sep 11, 8:37 pm, Jerry Avins wrote http://groups.google.com/group/comp.dsp/msg/b1ef0c8f50008e77?hl=en& : > He's > contemplating pitch shifting, not frequency shifting. In the audio domain, if I am contemplating pitch shifting, then I am also contemplating frequency shifting.

Hello, I am playing a little bit with audio signals and Fourier analysis. Since audio is a real 2-channel thing, I used the twofft algorithm...

Hello, I am playing a little bit with audio signals and Fourier analysis. Since audio is a real 2-channel thing, I used the twofft algorithm to simultaneously obtain both left and right channel Fourier transforms, see: http://www.library.cornell.edu/nr/bookcpdf/c12-3.pdf However, I cannot manage to simultaneously perform the inverse Fourier. It should be very easy, according to the ...

Hi, I'm working on a Nicam modulator in fpga,and such a system is required to perform pre-emphasis on the audio channels.Pre-emphasis...

Hi, I'm working on a Nicam modulator in fpga,and such a system is required to perform pre-emphasis on the audio channels.Pre-emphasis filter specified in the CCITT J.17 recommendation. Has anybody implemented this pre-emphasis filter and could you give me some tips on how you generated the coefficients? The pre-emphasis attenuation curve is given by the followingformula: Insertion los...

Hi, I'm working on a Nicam modulator in fpga,and such a system is required to perform pre-emphasis on the audio channels.Pre-emphasis...

Hi, I'm working on a Nicam modulator in fpga,and such a system is required to perform pre-emphasis on the audio channels.Pre-emphasis filter specified in the CCITT J.17 recommendation. Has anybody implemented this pre-emphasis filter and could you give me some tips on how you generated the coefficients? The pre-emphasis attenuation curve is given by the followingformula: Insertion los...

Hi, I'm working on a Nicam modulator in fpga,and such a system is required to perform pre-emphasis on the audio channels.Pre-emphasis...

Hi, I'm working on a Nicam modulator in fpga,and such a system is required to perform pre-emphasis on the audio channels.Pre-emphasis filter specified in the CCITT J.17 recommendation. Has anybody implemented this pre-emphasis filter and could you give me some tips on how you generated the coefficients? The pre-emphasis attenuation curve is given by the followingformula: Insertion los...

Hi, I'm working on a Nicam modulator in fpga,and such a system is required to perform pre-emphasis on the audio channels.Pre-emphasis...

Hi, I'm working on a Nicam modulator in fpga,and such a system is required to perform pre-emphasis on the audio channels.Pre-emphasis filter specified in the CCITT J.17 recommendation. Has anybody implemented this pre-emphasis filter and could you give me some tips on how you generated the coefficients? The pre-emphasis attenuation curve is given by the followingformula: Insertion los...

Hi to everyone, I've to calculate the SPL value of the waveform from the digital values of audio samples incoming from the audiocard (Echo...

Hi to everyone, I've to calculate the SPL value of the waveform from the digital values of audio samples incoming from the audiocard (Echo Layla24). I get the audio sample buffer using Asio protocol. I can calculate the RMS value of buffer, but how can i calculate the SPL value?? Is there a way to obtain the SPL from RMS?? Thanks in advance!

Hello I had just finished making an audio spectrum analyzer, when someone told me that since voice is a random signal you need to...

Hello I had just finished making an audio spectrum analyzer, when someone told me that since voice is a random signal you need to first correlate it before finding the fft of its samples. I had read somewhere that correlation is generally used for removing noise, and I tols him that I am assuming noise free environment. But he said that random signals are infinite energy signals and taking...

Hello everyone. I'm looking for the DSP Texas Instruments and Analog Devices libraries for audio. Can someone help me to find the source files or...

Hello everyone. I'm looking for the DSP Texas Instruments and Analog Devices libraries for audio. Can someone help me to find the source files or the list of the procedures available? Thank you! --------------------------------------- Posted through http://www.DSPRelated.com

Hi, I am implementing an application for matching/earching audio pattern in audio files ( using MFCC ), I am looking for ways to increase...

Hi, I am implementing an application for matching/earching audio pattern in audio files ( using MFCC ), I am looking for ways to increase the efficiency of my detection algorithm. Recently, I have been investigating PCA (Principal Components Analysis) as advised by Ikaro of this group, PCA is used to express the data by it's most significant dimensions discarding less significant dimensio...