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Search found 10 results on 1 pages for 'nyquist'.

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• Nyquist won't play audio

  - by erjiang

  I downloaded Nyquist, and am having trouble playing sounds from it. If I run it normally, I get: Nyquist -- A Language for Sound Synthesis and Composition Copyright (c) 1991,1992,1995 by Roger B. Dannenberg Version 2.29 > (play (osc 60)) Saving sound file to ./eric-temp.wav error: snd_save -- could not open audio output > If I wrap it by running padsp ny, the sound plays fine for about half a second, and then I get garbage fed to my speakers. Any solutions?

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• Figuring out the Nyquist performance limitation of an ADC on an example PIC microcontroller

  - by AKE

  I'm spec-ing the suitability of a dsPIC microcontroller for an analog-to-digital application. This would be preferable to using dedicated A/D chips and a separate dedicated DSP chip. To do that, I've had to run through some computations, pulling the relevant parameters from the datasheets. I'm not sure I've got it right -- would appreciate a check! (EDITED NOTE: The PIC10F220 in the example below was selected ONLY to walk through a simple example to check that I'm interpreting Tacq, Fosc, TAD, and divisor correctly in working through this sort of Nyquist analysis. The actual chips I'm considering for the design are the dsPIC33FJ128MC804 (with 16b A/D) or dsPIC30F3014 (with 12b A/D).) A simple example: PIC10F220 is the simplest possible PIC with an ADC Runs at clock speed of 8MHz. Has an instruction cycle of 0.5us (4 clock steps per instruction) So: Taking Tacq = 6.06 us (acquisition time for ADC, assume chip temp. = 50*C) [datasheet p34] Taking Fosc = 8MHz (? clock speed) Taking divisor = 4 (4 clock steps per CPU instruction) This gives TAD = 0.5us (TAD = 1/(Fosc/divisor) ) Conversion time is 13*TAD [datasheet p31] This gives conversion time 6.5us ADC duration is then 12.56 us [? Tacq + 13*TAD] Assuming at least 2 instructions for load/store: This is another 1 us [0.5 us per instruction] Which would give max sampling rate of 73.7 ksps (1/13.56) Supposing 8 more instructions for real-time processing: This is another 4 us Thus, total ADC/handling time = 17.56us (12.56us + 1us + 4us) So expected upper sampling rate is 56.9 ksps. Nyquist frequency for this sampling rate is therefore 28 kHz. If this is right, it suggests the (theoretical) performance suitability of this chip's A/D is for signals that are bandlimited to 28 kHz. Is this a correct interpretation of the information given in the data sheet in obtaining the Nyquist performance limit? Any opinions on the noise susceptibility of ADCs in PIC / dsPIC chips would be much appreciated! AKE

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• How would i down-sample a .wav file then reconstruct it using nyquist? - in matlab [closed]

  - by martin

  This is all done in MatLab 2010 My objective is to show the results of: undersampling, nyquist rate/ oversampling First i need to downsample the .wav file to get an incomplete/ or impartial data stream that i can then reconstuct. Heres the flow chart of what im going to be doing So the flow is analog signal - sampling analog filter - ADC - resample down - resample up - DAC - reconstruction analog filter what needs to be achieved: F= Frequency F(Hz=1/s) E.x. 100Hz = 1000 (Cyc/sec) F(s)= 1/(2f) Example problem: 1000 hz = Highest frequency 1/2(1000hz) = 1/2000 = 5x10(-3) sec/cyc or a sampling rate of 5ms This is my first signal processing project using matlab. what i have so far. % Fs = frequency sampled (44100hz or the sampling frequency of a cd) [test,fs]=wavread('test.wav'); % loads the .wav file left=test(:,1); % Plot of the .wav signal time vs. strength time=(1/44100)*length(left); t=linspace(0,time,length(left)); plot(t,left) xlabel('time (sec)'); ylabel('relative signal strength') **%this is were i would need to sample it at the different frequecys (both above and below and at) nyquist frequency.*I think.*** soundsc(left,fs) % shows the resaultant audio file , which is the same as original ( only at or above nyquist frequency however) Can anyone tell me how to make it better, and how to do the various sampling at different frequencies?

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• How would i down-sample a .wav file then reconstruct it using nyquist? - in matlab [closed]

  - by martin

  Possible Duplicate: How would i down-sample a .wav file then reconstruct it using nyquist? - in matlab This is all done in MatLab 2010 My objective is to show the results of: undersampling, nyquist rate/ oversampling First i need to downsample the .wav file to get an incomplete/ or impartial data stream that i can then reconstuct. Heres the flow chart of what im going to be doing So the flow is analog signal - sampling analog filter - ADC - resample down - resample up - DAC - reconstruction analog filter what needs to be achieved: F= Frequency F(Hz=1/s) E.x. 100Hz = 1000 (Cyc/sec) F(s)= 1/(2f) Example problem: 1000 hz = Highest frequency 1/2(1000hz) = 1/2000 = 5x10(-3) sec/cyc or a sampling rate of 5ms This is my first signal processing project using matlab. what i have so far. % Fs = frequency sampled (44100hz or the sampling frequency of a cd) [test,fs]=wavread('test.wav'); % loads the .wav file left=test(:,1); % Plot of the .wav signal time vs. strength time=(1/44100)*length(left); t=linspace(0,time,length(left)); plot(t,left) xlabel('time (sec)'); ylabel('relative signal strength') **%this is were i would need to sample it at the different frequecys (both above and below and at) nyquist frequency.*I think.*** soundsc(left,fs) % shows the resaultant audio file , which is the same as original ( only at or above nyquist frequency however) Can anyone tell me how to make it better, and how to do the various sampling at different frequencies?

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• How would i down-sample a .wav file then reconstruct it using nyquist? - in MATLAB

  - by Andrew

  This is all done in MATLAB 2010 My objective is to show the results of: undersampling, nyquist rate/ oversampling First i need to downsample the .wav file to get an incomplete/ or impartial data stream that i can then reconstuct. Heres the flow chart of what im going to be doing So the flow is analog signal - sampling analog filter - ADC - resample down - resample up - DAC - reconstruction analog filter what needs to be achieved: F= Frequency F(Hz=1/s) E.x. 100Hz = 1000 (Cyc/sec) F(s)= 1/(2f) Example problem: 1000 hz = Highest frequency 1/2(1000hz) = 1/2000 = 5x10(-3) sec/cyc or a sampling rate of 5ms This is my first signal processing project using matlab. what i have so far. % Fs = frequency sampled (44100hz or the sampling frequency of a cd) [test,fs]=wavread('test.wav'); % loads the .wav file left=test(:,1); % Plot of the .wav signal time vs. strength time=(1/44100)*length(left); t=linspace(0,time,length(left)); plot(t,left) xlabel('time (sec)'); ylabel('relative signal strength') **%this is were i would need to sample it at the different frequecys (both above and below and at) nyquist frequency.*I think.*** soundsc(left,fs) % shows the resaultant audio file , which is the same as original ( only at or above nyquist frequency however) Can anyone tell me how to make it better, and how to do the sampling at verious frequencies?

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• SQL Server 2008 Express - "Best" backup solution?

  - by Alexander Nyquist

  Hi! What backup solutions would you recommend when using SQL Server 2008 express? I'm pretty new to SQL Server, but as I'm coming from an MySql background i thought of setting up replication on another computer and just take x-copy backups of that server. But unfortanetly replication is not available in the express edition. The site is heavily accessed, so there has to be no delays och downtime. I'm also thinking of doing a backup twice a day or something. What would you recommend? I have multiple computers I can use, but don't know if that helps me since i'm using the express version. Thanks

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• Can anyone give me a sample DSP script in C/C++

  - by Andrew

  Im working on a (Audio) DSP project and just wondering if there are any sample (Open source) DSP example that are written in c or c++, for my MSP430 Chip. I just want something as a guideline so i can program my own script using the ACD and DCA on my board for sampling. http://focus.ti.com/docs/toolsw/folders/print/msp-exp430f5438.html Thats my board, MSP430F5438 Experimenter Board, from what i herd it can run dsp script via the USB connection with the computer. Im using CCS ( From TI, code composer studio) and Octave/Matlab. Just any DSP example scripts or sites that will help me create my own would be appreciated. What im tying to do, Partial audio (sampled) track -- Nyquist rate sampling -- over- and undersampling -- reconstruction of the audio track.

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• GUI Control For Audio Presentation

  - by Boris

  I need GUI control for audio file presentation. The language is not very important but it should run on windows platform. I should be able to :- load the file play the sound put and move markers across the audio bar. it would be nice if it can load itself from RTP wireshark captures (and not wav files). An example may be seen in audacity (may be someone even had an experience extracting it from there). Writing nyquist scripts in audacity is not a good option because I have to operate on RTP captures and not on raw sound samples. Another example of such control is wireshark RTP analyzer. Any advise?

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• ADSL throughput loss from Reed-Solomon encoding

  - by javano

  I'm reading about ADSL starting here and I am confused by how the Reed-Solomon encoding for ECC is limiting the available transfer rate, as much as it does (nearly half). This pdf on the same subject contains the following; A maximum of 255 sub-carriers can be used to modulate data in the downstream direction. Sub-carrier 256, the downstream Nyquist frequency, and sub-carrier 64, the downstream pilot frequency, are not available for user data, thus limiting the total number of available downstream sub-carriers to 254. Each of these 254 sub-carriers can support the modulation of 0 to 15 bits. Since the ADSL DMT data frame rate is 4000 frames per second, the maximum theoretical downstream data rate of an ADSL system is 15.24Mbps. Due to limitations in system architecture, specifically the maximum allowable Reed-Solomon codeword size (255 bytes), the maximum achievable downstream data rate is 8.16Mbps. How is this nearly halving the throughput? Is all that extra bandwidth overhead of the RS encoding? 15240000 bps (15.24Mbps) - 8160000 bps (8.12Mbps) = 7080000 bps (7.08Mbps). Where has that 7Mbps of throughput gone? EDIT: I tried to read the wiki page on Reed-Soloman but it's all crazy maths and algerbra, which I don't understand. I can understand that data is split into 255 byte codewords, because that maybe the max codeword size whilst still maintaining accuracy during transmission; But I don't understand why that means less data is sent?

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• Figuring out the performance limitation of an ADC on a PIC microcontroller

  - by AKE

  I'm spec-ing the suitability of a microcontroller like PIC for an analog-to-digital application. This would be preferable to using external A/D chips. To do that, I've had to run through some computations, pulling the relevant parameters from the datasheets. I'm not sure I've got it right -- would appreciate a check! Here's the simplest example: PIC10F220 is the simplest possible PIC with an ADC. Runs at clock speed of 8MHz. Has an instruction cycle of 0.5us (4 clock steps per instruction) So: Taking Tacq = 6.06 us (acquisition time for ADC, assume chip temp. = 50*C) [datasheet p34] Taking Fosc = 8MHz (? clock speed) Taking divisor = 4 (4 clock steps per CPU instruction) This gives TAD = 0.5us (TAD = 1/(Fosc/divisor) ) Conversion time is 13*TAD [datasheet p31] This gives conversion time 6.5us ADC duration is then 12.56 us [? Tacq + 13*TAD] Assuming at least 2 instructions for load/store: This is another 1 us [0.5 us per instruction] Which would give max sampling rate of 73.7 ksps (1/13.56) Supposing 8 more instructions for real-time processing: This is another 4 us Thus, total ADC/handling time = 17.56us (12.56us + 1us + 4us) So expected upper sampling rate is 56.9 ksps. Nyquist frequency for this sampling rate is therefore 28 kHz. If this is right, it suggests the (theoretical) performance suitability of this chip's A/D is for signals that are bandlimited to 28 kHz. Is this a correct interpretation of the information given in the data sheet? Any pointers would be much appreciated! AKE

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