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PCM

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3.9 | 9 ratings Rate this file 51 Downloads (last 30 days) File Size: 2.78 KB File ID: #6585

PCM

by Jawad Khalid

 

21 Dec 2004 (Updated 23 Dec 2004)

This file does the PCM thing for any voice signal.

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File Information
Description

This file converts the voice signal into PCM mode for transmission.

Acknowledgements
This submission has inspired the following:
PCM coding
MATLAB release MATLAB 6.5.1 (R13SP1)
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Comments and Ratings (14)
17 Feb 2005 Ali Hassan

a very efficient code and totaly correct
Thank you for your participating

15 Jun 2005 Stefan Bunov  
26 Oct 2005 vasfi tuna

my project was about sound zipping and I found this excellent code on this page ....it works and thanks to mathworks...

27 Nov 2005 JInbo li

The sampling frequency is useless..

31 Jan 2006 nicolas chavarriaga

i have a question, i have to coder a .WAV archive, but when i change the MP3 for .WAV, the matlab command window shows a mistake. why? , how can i run the M-FILE with a .wav archive.
  Index exceeds matrix dimensions.
  Error in ==> pcm at 72
     samp_in(k) = m(k*Ts);

tank you

10 May 2006 qiguang miao  
01 Aug 2006 ganesh ramasamy

fair

06 Feb 2007 Abel Valero

I have the same problem. when I change the .mp3 file for a .wav file it produces a mistake, related with the index of the matrix in line 72

29 Nov 2007 yu sun  
20 Apr 2008 abas turki hamza

good

24 Aug 2008 mrudula athi

i haev this same problem of index exceeding matrix dimension..plz suggest..

09 Dec 2009 Eng FA

hi

I wana help that how can i make a program for PCM on the matlab??

I did not know who can i start, i tried yours but nothing done
So can you please help me on that
Thank you :)

11 Apr 2010 Doddes Douddoussa

hi
I have the same problem. I want to apply this code on a wav file but it produces a mistake, related with the index of the matrix in line 72.
Please help me to get a PCM code applied on the wav, I need this code my project at the end of study
thank you:)

22 Aug 2010 Peter Robert

The index of the matrix problem occurs because your file doesn't have 2*Fs = 16000 samples in it (<2 seconds). I noticed other problems as well, like the error is divided with the number of samples, and the read wav file isn't necessarily sampled with 8kHz.
Uploaded a fixed implementation based on this one. After it has been reviewed you should be able to find it on this site.

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Tag Activity for this File
Tag Applied By Date/Time
pcm Jawad Khalid 22 Oct 2008 07:38:32
converts Jawad Khalid 22 Oct 2008 07:38:32
voice signal Jawad Khalid 22 Oct 2008 07:38:32
transmission Jawad Khalid 22 Oct 2008 07:38:32
pcm Zis 19 Jan 2009 07:09:15
voice signal Zis 19 Jan 2009 07:10:03
pcm Saeed Bamashmos 01 Feb 2009 02:34:44
converts Luiz Bonani 11 Jun 2010 14:46:02
pcm ++ teixeira 17 Oct 2010 15:21:16
pcm Manish Agrawal 22 Apr 2011 01:42:16
converts Manish Agrawal 22 Apr 2011 01:42:16
converts Pabblin P 13 Nov 2011 19:03:40
converts alfa perez 05 Jan 2012 04:03:56
converts Manuel Antunes 02 Apr 2012 09:20:26
pcm Manuel Antunes 02 Apr 2012 09:20:33

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