29 Jul
2009
29 Jul
'09
11:51 a.m.
Hi guys,
I read Goller's documentation "About the GSM-Dm-Channels", but I'm a bit
confused about something.
I understand that one TDMA frame has a period of 4.615 ms and consists
of 8 Time slots, each of a period of 576.9 us. When a MS registers to
the BTS, it gets one Time-slot, but how is speech transfered? Normally
the Fs (samplefrequency) is 8 KHz, this means every 128 us one byte
sample. If I have one timeslot of let's say 577 us , than there is a gap
of 7 other timeslots, which makes a 4 ms gap. Also the speech data is
compressed. How is that damn speech data multiplexed with other MSs and
finally received on the other side as 8 bit samplevalue at a rate of 128
us (8 KHz).
It gets fuzzy to me, cause I wrote a microcontroller project which,
among other things, samples audio and than transfered to a GSM module.
This is done at 8 KHz samplerate, so how is that done when I have only
one TimeSlot of 577 us every 4 ms period?
I can't find a good link about how several MSs on one ARFCN can send and
receive speech data.
Thank you.
29 Jul
29 Jul
1 p.m.
I understand that one TDMA frame has a period of 4.615
ms and consists
of 8 Time slots, each of a period of 576.9 us. When a MS registers to
the BTS, it gets one Time-slot, but how is speech transfered? Normally
the Fs (samplefrequency) is 8 KHz, this means every 128 us one byte
sample. If I have one timeslot of let's say 577 us , than there is a gap
of 7 other timeslots, which makes a 4 ms gap. Also the speech data is
compressed. How is that damn speech data multiplexed with other MSs and
finally received on the other side as 8 bit samplevalue at a rate of 128
us (8 KHz).
I am not sure if this is what you want to know but I suggest you have to
take a look at GSM fullrate and halfrate speech codecs. I am currently
doing the speech processing lecture in here in Aachen so you might have
a look at the following books: "Digital Speech Transmisson" by Peter
Vary and Rainer Martin and "Wireless Communications Second Edition" by
Theodore S. Rappaport. Or you read it from the standard...
You will find that speech is not transmitted by sending just the 8kHz
samples. There is a complex prediction filter structure. In principle
the filter coefficients and the error are send and the receiver does a
reconstruction of the signal.
Additionally everything is quantized heavily to achieve the low
datarates.
If you use fullrate, every 8th timeslot is given to you, if you use
halfrate you get every 16th timeslot. You also need to now about
multiplexing of the logical GSM channels. Sometimes a timeslot is used
to send signaling or control information instead of speech data. The
Speechcodec is able to hide this missing data so you will not notice
that while talking.
regards, Johannes
5392
days inactive
5392
days old
1 comments
2 participants
participants (2)
-
Johannes Schmitz -
Nordin