hi friends can any one suggest me a good material to understand and implement "Vector Quantization".. waiting for u r...

hi friends can any one suggest me a good material to understand and implement "Vector Quantization".. waiting for u r suggestion.... thanx in advance ramu

In order to do this efficiently you need to perform the conversion in multiuple stages See here : ...

In order to do this efficiently you need to perform the conversion in multiuple stages See here : http://www.sound slogical.com/support/resample/documentation/english/docu mentparts/re

The answer from harishankar is not quite correct. Theory tells the following: Re-sampling from 44100 Hz to 48000 Hz: 1)...

The answer from harishankar is not quite correct. Theory tells the following: Re-sampling from 44100 Hz to 48000 Hz: 1) Up-sample by expansion factor of 160 via zero insertion (sampling frequency is now 7.056 MHz!!) 2) Run up-sampled signal through a

hi all in the AMR standard, a signal called y[n] is used. it is the "filtered adaptive codebook vector" and is defined as: ...

hi all in the AMR standard, a signal called y[n] is used. it is the "filtered adaptive codebook vector" and is defined as: y[n] = v[n]*h[n] (read convolution of v[n] and h[n]), where v[n] => adaptive codebook vector and h[n] => impulse response of teh wei

dear all: In the Find_best() function of G.723.1 reference code, there is a sentence (line 527 of EXC2.C or Line 655 of...

dear all: In the Find_best() function of G.723.1 reference code, there is a sentence (line 527 of EXC2.C or Line 655 of EXC_LBC.C) I don't understand: /* Search for the best sequence */ for ( k = 0 ; k < Sgrid ; k ++ ) { /* Find maximum amplitude *

Hello All: I am converting floating point code to fixed point of Qcelp13 coder. I need to compute the autocorrelation of a...

Hello All: I am converting floating point code to fixed point of Qcelp13 coder. I need to compute the autocorrelation of a temporary vector and then multiply their sum with the autocoeffs of the input signal. The Q format of autocoeffs of the tempVector is Q16 in 32bit

Hi guys; What is a speech data file, .spd format? What are the resources for getting more information on this? How can it be...

Hi guys; What is a speech data file, .spd format? What are the resources for getting more information on this? How can it be listened to? Looking keenly for the response, Sincerely, Prateek Shrivastava

The smooth is for LPC coefficients calculation. When G.723.1 calculates the pitch by open-loop or close-loop pitch predictor, it...

The smooth is for LPC coefficients calculation. When G.723.1 calculates the pitch by open-loop or close-loop pitch predictor, it uses the the original signal source not the smoothed one. //Yidan -----Original Message----- From: gray_1024 [mailto:] Sent: Friday, Mar

Hi guys, I am doing a project, comparing ADPCM & CELP. For this i have chosen G.721 for ADPCM & G.729 standard for CELP. In...

Hi guys, I am doing a project, comparing ADPCM & CELP. For this i have chosen G.721 for ADPCM & G.729 standard for CELP. In these, 1)What are the parameters that can be compared in these 2? 2)How can the Mean squared error be found out? 3)Are the MOS scores standa

Hi list,   I'm involved in integrating G.729 algorithm.   I know that G.729 encodes with 16 bits samples but our...

Hi list,   I'm involved in integrating G.729 algorithm.   I know that G.729 encodes with 16 bits samples but our system provides 8 bits input, so I've allocated these 8 bits samples in