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lpc-10 coded speech

Started by Anonymous in Speech Coding20 years ago

Hi all, I remember that Texas Instruments have had a big library with pre-coded words, spoken by a TV-moderator from Dallas...

Hi all, I remember that Texas Instruments have had a big library with pre-coded words, spoken by a TV-moderator from Dallas Radio located in Dallas/Tx. This was a library of single words and phrases coded n lpc-10 with 8KHz-sample-frequency for the TMS52xx speech-synthesizer f


AMR codec (ETSI version v.s. 3GPP version)

Started by Murphy Chen in Speech Coding20 years ago 3 replies

ETSI has a AMR reference code. 3GPP also has AMR reference code for both fix-point and floating point. What's the difference...

ETSI has a AMR reference code. 3GPP also has AMR reference code for both fix-point and floating point. What's the difference between them? Which version is preferred?


FS-1016 CELP

Started by Ray Mc Carthy in Speech Coding20 years ago 4 replies

Hi All, I am working with the FS-1016 4.8kb/s CELP coder but I cant understand why the LSP's are interpolated for every...

Hi All, I am working with the FS-1016 4.8kb/s CELP coder but I cant understand why the LSP's are interpolated for every subframe.Can someone tell me is there a specific advantage for doing this? Regards, Ray. -------------------Legal Disclaimer--------------


G729 quantization of the gains

Started by shabnamso in Speech Coding20 years ago 1 reply

Hello I would be thankful if anybody who has tested the G729A source code by ITU can tell me if the QUA_GAIN function has been...

Hello I would be thankful if anybody who has tested the G729A source code by ITU can tell me if the QUA_GAIN function has been implemented exactly as the algorithm. I try to run it on Code Composer Studio C6000 simulator with various ITU test vectors and everytime I


About EVRC

Started by Wei Chen in Speech Coding20 years ago 1 reply

Hi, buddies, Do u have the following paper? M. C. Recchione, "The enhanced variable rate coder: Toll quality speech for...

Hi, buddies, Do u have the following paper? M. C. Recchione, "The enhanced variable rate coder: Toll quality speech for CDMA", International Journal of Speech Technology, pp. 305- 315, 1999 I could not get it. If you have, could you pls send it to me at


C- code for G.723.1 and LSP coefficients, gain, ...

Started by setumi71 in Speech Coding20 years ago 1 reply

Hi, I have downloaded c code for the codec (it is attached with G.723.1 annex A, the version with fixed point). How to find the...

Hi, I have downloaded c code for the codec (it is attached with G.723.1 annex A, the version with fixed point). How to find the coefficients LSP, the values of gain, amplitude of the frames? I find attached the code a vector sequence test, but there is no a help on using


G.729 bitstream file format

Started by rishi_ahuja in Speech Coding20 years ago 3 replies

Hi! All, I have captured a G.729 call using Ethereal (in rtpdump payload format) and would like to use ITU G.729 decoder to...

Hi! All, I have captured a G.729 call using Ethereal (in rtpdump payload format) and would like to use ITU G.729 decoder to decode the conversation. Is there a utility available which would help me convert me the G.729 payload from rtpdump payload format to the bi


MP-MLQ

Started by prajesh_etc in Speech Coding21 years ago 3 replies

Hi everyone, I am working on ITU Standard G.723.1, ie. speech coder(Hybrid coder)i have started the project one month back. I have...

Hi everyone, I am working on ITU Standard G.723.1, ie. speech coder(Hybrid coder)i have started the project one month back. I have almost designed the h/w for it. And in s/w part i have gone through th b/d of G.723.1, I am clear with most of the thing but i am unable to underst


FS-1016 Adaptive Codebook

Started by Ray McCarthy in Speech Coding21 years ago

Hi All, I'm having a bit of trouble with the adaptive codebook.From theory it seems that the codebook is a 256 x 60 array,so to...

Hi All, I'm having a bit of trouble with the adaptive codebook.From theory it seems that the codebook is a 256 x 60 array,so to speak, but from the standard code it seems that it is a 256 x 1 array and to get an adptive excitation,you select different elements of the codebook.I dont


waveform interpolation

Started by dr_simic in Speech Coding21 years ago 2 replies

I'm trying to implement WI codec,for now just layer I(without quantisation),as it is described in Eddie Choy's thesis,but I don't...

I'm trying to implement WI codec,for now just layer I(without quantisation),as it is described in Eddie Choy's thesis,but I don't get perfectly sounding decoded speech,as they stated,there are audible errors,and these errors usually occur when pitch doubles. Can anybody help me,or send me


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