[eside-ghost] musica en telefonia - moviles vs fijos tradicionales

Jon Urionaguena juriona en nesys-st.com
Vie Nov 10 15:18:47 CET 2006


Por desbarrar...

GSM usa algún algoritmo de detección de silencios para conmutar...

Mirando por ahí...

http://kbs.cs.tu-berlin.de/~jutta/gsm/js-intro.html

  "Speech coding
/References:/ [NHdB89
<http://www.cs.tu-berlin.de/%7Ejutta/gsm/js-bib.html#natv>, V+89
<http://www.cs.tu-berlin.de/%7Ejutta/gsm/js-bib.html#vary>, S+89
<http://www.cs.tu-berlin.de/%7Ejutta/gsm/js-bib.html#sout>]

GSM is a digital system, so speech signals, inherently analog, have to
be digitized.  The method employed by ISDN, and by current telephone
systems for multiplexing voice lines over high speed trunks and optical
fiber lines, is Pulse Coded Modulation (PCM).  The output stream from
PCM is 64 kbps, too high a rate to be feasible over a radio link.  The
64 kbps signal contains much redundancy, although it is simple to
implement.  The GSM group studied several voice coding algorithms on the
basis of subjective speech quality and complexity (which is related to
cost, processing delay, and power consumption once implemented) before
arriving at the choice of a Regular Pulse Excited - Linear Predictive
Coder (RPE­LPC) with a Long Term Predictor loop.  Basically, information
from previous samples, which does not change very quickly, is used to
predict the current sample.  The coefficients of the linear combination
of the previous samples, plus an encoded form of the residual, the
difference between the predicted and actual sample, represent the
signal.  Speech is divided into 20 millisecond samples, each of which is
encoded as 260 bits, giving a total bit rate of 13 kbps.

.....

  Discontinuous transmission
/References:/ [S+89
<http://www.cs.tu-berlin.de/%7Ejutta/gsm/js-bib.html#sout>, Che91
<http://www.cs.tu-berlin.de/%7Ejutta/gsm/js-bib.html#chee>]

Minimizing co­channel interference is a goal of any cellular system,
since it allows better service for a given cell size, or the use of
smaller cells, thus increasing the overall capacity of the system. 
Discontinuous transmission (DTX) is a method that takes advantage of the
fact that a person speaks less that 40 percent of the time in normal
conversation [S+89
<http://www.cs.tu-berlin.de/%7Ejutta/gsm/js-bib.html#sout>], by turning
the transmitter off during silence periods.  An added benefit of DTX is
that power is conserved at the mobile unit.

The most important component of DTX is, of course, Voice Activity
Detection.  It must distinguish between voice and noise inputs, a task
that is not as trivial as it appears, considering background noise.  If
a voice signal is misinterpreted as noise, the transmitter is turned off
and a very annoying effect called clipping is heard at the receiving
end.  If, on the other hand, noise is misinterpreted as a voice signal
too often, the efficiency of DTX is dramatically decreased.  Another
factor to consider is that when the transmitter is turned off, there is
a very silent silence heard at the receiving end, due to the digital
nature of GSM.  To assure the receiver that the connection is not dead,
/comfort noise/ is created at the receiving end by trying to match the
characteristics of the transmitting end's background noise.

  Discontinuous reception
Another method used to conserve power at the mobile station is
discontinuous reception.  The paging channel, used by the base station
to signal an incoming call, is structured so that the mobile station
knows when it needs to check for a paging signal.  In the time between
paging signals, the mobile can go into sleep mode, when almost no power
is used. 
"

Entiendo que estás cargándote ese algoritmo transmitiendo contínuamente
en uno de los sentidos, no permitiendo que ese algoritmo VAD [1] permita
conmutar al receptor de modo "recepción" a modo "transmisión"...

¿Alguno ve lo mismo que yo? No estoy nada seguro de nada...

Esta droga es wuapaaaa!!!

[1] http://en.wikipedia.org/wiki/Voice_activity_detection

-- 

Jon



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