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DCT in audio compression

Started by eblade in comp.dsp16 years ago 2 replies

After reading, what seems like a lot of material, I'm still in doubt why DCT isnt used in audio-compression. Seeing as DCT assumes mirror...

After reading, what seems like a lot of material, I'm still in doubt why DCT isnt used in audio-compression. Seeing as DCT assumes mirror symmetry, my only idea for not validating DCT over MDCT is that with real-life signals you can rarely (if ever) expect a signal to have that property? Is that correct or is there a better explanation.


Best choice for EMG signal processing

Started by maestro in comp.dsp14 years ago 4 replies

Searching for a DSK at the Texas site, I've noticed that some kits are available with audio ports and audio codecs, while some DSPs are...

Searching for a DSK at the Texas site, I've noticed that some kits are available with audio ports and audio codecs, while some DSPs are used mainly for video applications. I'd like to use a DSK that is less "guided". Any suggestions? My application processing requirements are not defined yet, but the signal must be sampled at least at 10kHz. Thank you,


low-cost audio waveform display doubles as space heater

Started by Robert Adams in comp.dsp14 years ago 3 replies

For those of you who on a budget who need to visualize audio waveforms, check out the following...

For those of you who on a budget who need to visualize audio waveforms, check out the following link. http://www.youtube.com/watch?v=wBydVCF4DrY All we need to do is figure out how to make the equivalent of a horizontal sync circuit. Might want to check with your local fire department first ... Happy New Year! Bob Adams


Audio synthesis problem

Started by Brent in comp.dsp17 years ago 2 replies

Hi, I would like to generate an audio clip with a varying frequency. A few attempts using either matlab/octave or Python code have made...

Hi, I would like to generate an audio clip with a varying frequency. A few attempts using either matlab/octave or Python code have made "interesting" sounds but not what I'm looking for yet. The sound I want will have an initial frequency F0 which will shift (interpolate) to a frequency F1 midway through the clip, and remain at F1 until the end. For example, in a two-second clip the fr...


noise removal from audio signal using cepstral analysis

Started by enricophpdsp in comp.dsp13 years ago 6 replies

hello, i have an audio signal mixed with white noise. How do i use the cepstrum technique to remove the noise from it. please tell me from...

hello, i have an audio signal mixed with white noise. How do i use the cepstrum technique to remove the noise from it. please tell me from where should i start. i want to know the basics of cepstrum and how can i use it to fullfill my purpose. please guide me about the literature which i should read. thanks


Audio phase lock loop design

Started by Jorgito in comp.dsp16 years ago 6 replies

Hi All, Looking for basic design reference info for audio phase lock loops. For example, given a specified analog input S/N, what...

Hi All, Looking for basic design reference info for audio phase lock loops. For example, given a specified analog input S/N, what acquisition time is needed to achieve 1 degree phase resolution? Thanks. Jorgito


PC audio crossover

Started by Martin in comp.dsp19 years ago 4 replies

Does anyone know whether a PCI card exists that will act as an audio active crossover network? Ie it should accept a digital input from a PC...

Does anyone know whether a PCI card exists that will act as an audio active crossover network? Ie it should accept a digital input from a PC soundcard then be capable of splitting it digitally into a number of frequency bands (at least 3), with user definable cut off frequencies and attenuation rates. Obviously seperate outputs should be available for each frequency band per channel. Or is i...


Frames solution ?? and complete audio silence file

Started by ranjeet in comp.dsp20 years ago 1 reply

Hi all !! ( FIRST OF ALL I AM THANKFULL TO JERRY, TIM, JOHAN, RANDY for clearing my doubts in the Floatiing and fixed point. Well i will...

Hi all !! ( FIRST OF ALL I AM THANKFULL TO JERRY, TIM, JOHAN, RANDY for clearing my doubts in the Floatiing and fixed point. Well i will like to know the algo which jerry talked about mutilpication (how it is done)). Now I want to share my understanding as I am Engineering student so please let me know about my blunder. I have a audio file, (8Khz, 16bit data) Now the thin...


Problems with Sonogram

Started by Rock Lobster in comp.dsp16 years ago 11 replies

Hello, I'm programming an audio application that should display a sonagram. I convert audio samples by FFT into the frequency data, so far so...

Hello, I'm programming an audio application that should display a sonagram. I convert audio samples by FFT into the frequency data, so far so good. The problem is that the values now are in a logarithmic scale of course and I want to transform them to a linear scale. But I don't know how. I've got 256 colors, so I'd like to have 256 linear steps from 0 to 255, so maybe it would be best to re...


Regarding PSNR for audio compression

Started by vasindagi in comp.dsp16 years ago

Its known that PSNR (peak signal to noise ratio) is used to compare image quality after compression. Can it also be used for audio quality?

Its known that PSNR (peak signal to noise ratio) is used to compare image quality after compression. Can it also be used for audio quality?


audio input extension for ADSP-21369 EZ-Kit Lite

Started by Arne Borsum in comp.dsp18 years ago 2 replies

Hi, I am developing audio applications on the ADSP-21369 EZ-Kit Lite board. The board has 8 channels audio output but only 2 channels input,...

Hi, I am developing audio applications on the ADSP-21369 EZ-Kit Lite board. The board has 8 channels audio output but only 2 channels input, although the DSP itself supports more channels. Now I am searching for a extension board for the EZ-Kit which provides 6 or more (analog) input channels. Is there anything like this availble? Regards, Arne Borsum


[O.T.] Audio DAC as AWG (test source)?

Started by Pete Fraser in comp.dsp13 years ago 7 replies

I'm just finishing up design of an analog data acquisition system (16 bits, 100 kHz sampling), and need to come up with a test / verification...

I'm just finishing up design of an analog data acquisition system (16 bits, 100 kHz sampling), and need to come up with a test / verification scheme. I've looked at some of the arbitrary waveform generators available, and they seem to be expensive, and not very accurate (13 or 14 bits). There seems to be a wide variety of inexpensive USB and firewire audio DACs available, and some of them ...


Speech/Speaker recognition

Started by Anonymous in comp.dsp19 years ago 5 replies

Hi, where is a good place to learn about speaker recognition? Specifically, I would like to use/develop an application that can take an audio...

Hi, where is a good place to learn about speaker recognition? Specifically, I would like to use/develop an application that can take an audio clip of human speech and determine who the speaker is (I assume there would be a database of audio samples that the app would compare to). Can anyone point me to some resources or apps that may help? Thank you!


Audio Equalization Using MDCT

Started by foxcob in comp.dsp15 years ago 1 reply

I have done real-time audio equalization using fast-convolution (overlap-add) where the frequency domain values were specified by the end user...

I have done real-time audio equalization using fast-convolution (overlap-add) where the frequency domain values were specified by the end user through a graphical interface. If the phase response did not need to be specified, could a MDCT give better frequency resolution for around the same number of computations? Would this work or am I completely off here. Thanks, Jacob


Spectral Shaping in freq. domain for audio input

Started by aaab in comp.dsp11 years ago 5 replies

Hi all, I have a digital audio signal sampled at 20000 Hz. I would like to increase the amplitude by two of certain freq. components, which...

Hi all, I have a digital audio signal sampled at 20000 Hz. I would like to increase the amplitude by two of certain freq. components, which are from 1100 Hz to 1540 Hz. This is to be done in the Frequency domain using Matlab code. However I am getting artefacts especially at the window edges. Please advice as to what am I doing wrong in the following steps: 1. Identify the frequency bins th...


Instrument for multi-channel audio recording

Started by reju_vg in comp.dsp19 years ago 2 replies

I want to record more than two channels audio signal at the same time. Can anybody suggest one instrument, which is readily available in the...

I want to record more than two channels audio signal at the same time. Can anybody suggest one instrument, which is readily available in the market? For example say I have 8 microphones and I want to record all the outputs of the microphone at the same time at 8KHz sampling rate. Thanks Reju This message was sent using the Comp.DSP web interface on www.DSPRelated.com


Multistage Interpolation - Need help understanding each stage

Started by Rick S. in comp.dsp20 years ago 16 replies

From scouring this group, I see that someone else is attempting the same thing that I'm trying to accomplish - multistage interpolation of an...

From scouring this group, I see that someone else is attempting the same thing that I'm trying to accomplish - multistage interpolation of an audio signal. Just getting my feet wet with this topic, I'd like to ask you guys a few general qustions. Taking a bandlimited audio signal sampled at 44.1kHz and interpolating by 160 broken into multiple stages (4x, 5x, and 8x respectively), my qu...


Allpass filter

Started by Kuba in comp.dsp19 years ago 4 replies

Hallo, I'm a student and I'm makins an audio device. I have to make a allpass filter or lowpass and highpass, one after another. The point...

Hallo, I'm a student and I'm makins an audio device. I have to make a allpass filter or lowpass and highpass, one after another. The point is that i have to cutt the audio band at about 20Hz and at about 20kHz to get good signal parametres before parametric equalizer. Do You have some sugestions what kind of filters would be best in this case(IIR or FIR and what kind)? It will be im...


Looking for audio effect library

Started by Karmas Lin in comp.dsp20 years ago 4 replies

Hi all, I'm looking for the audio effect library (commercial or public domain), I need the compressor, EQ, reverb, chorus, flanger... to...

Hi all, I'm looking for the audio effect library (commercial or public domain), I need the compressor, EQ, reverb, chorus, flanger... to process wave files on PC. It may be static or dynamic library and can be called by my Windows application. Can any one tell me where to get/buy? I've search for the FAQ of comp.dsp and can not find suitable library... Thanks in advance.


Audio morphing?

Started by Verified by Kerberos in comp.dsp20 years ago 3 replies

Hi, I'm reading this list for quite some time trying to learn something about DSP. Most of it is too complicated for me. I'm a guitarist...

Hi, I'm reading this list for quite some time trying to learn something about DSP. Most of it is too complicated for me. I'm a guitarist and musician from Switzerland and not a DSP guy so please apologize if I'm asking stupid or obvious questions. I recently discovered a product by Prosoniq (http://www.prosoniq.com) that does morphing with audio signals. I've listened to the demo files ...