The document discusses speech compression using GSM RPE-LTP encoding. It begins with an introduction to GSM standards and architecture. It then describes how speech is generated and modeled in the GSM 6.10 vocoder using the RPE-LTP algorithm. The algorithm compresses speech by analyzing the signal to determine if it is voiced or unvoiced, encoding the periodicity of voiced sounds, and transmitting the filter parameters. At the receiver, the decoder uses these parameters to reconstruct the speech signal through linear predictive coding, long term prediction synthesis filtering, and residual pulse decoding.

1. Speech Compression
using
GSM RPE-LTP
Faiza Nawaz
Bisma Hashmi
Mehrin Kiani

2. 2
Introduction to GSM
 The Global System for Mobile Communications is the most
popular standard for mobile phones in the world.
 GSM service is used by over 2 billion people across more than
212 countries and territories.
 The ubiquity of the GSM standard makes international roaming
very common between mobile phone operators.
 GSM differs significantly from its predecessors in that both
signaling and speech channels are Digital call quality.
(so it is considered a second generation (2G) mobile phone
system.)

3. 3
Architecture Of GSM

4. 4
What is Speech?
 Speech Generation:

5. 5
GSM 6.10 Vocoder
 Key principle: mathematical modeling of the human vocal tract,
leading to an efficient compression method for transmitting
speech.
 A vocoder (combination of voice and coder) is used to describe
GSM systems tailored for the compression of speech.
 The sampling rate is 8000 sample/s leading to an average bit
rate for the encoded bit stream of 13 K bit/s

6. 6
GSM 6.10 Vocoder
 Coding scheme used by GSM 6.10 Vocoder is the Regular Pulse
Excitation - Long Term prediction - Linear Predictive Coder
(RPE-LTP)
 Vocoder sends three kinds of information to the receiver:
 Voiced or unvoiced signal
 (If it is voiced) The period of the excitation signal
 The parameters of the prediction filter.

7. 7
Linear Predictive Coder (LPC)
 LPC algorithm assumes that each speech sample is a linear
combination of previous samples.
 Speech is sampled, stored and analyzed.
 Coefficients calculated from the sample are transmitted and
processed in the receiver.
 Receiver accurately processes and categorizes voiced and
unvoiced sounds.

8. 8
Residual Pulse Excited (RPE) Coder
 Determines if the signal is voiced or unvoiced
 Determines the period for voiced sounds, encodes periodicity and
transmits the coefficient
 When the signal changes from voiced to unvoiced, RPE transmits a
code that stops the receiver from generating periodic pulses
 Starts generating random pulses to correspond to the noise like
nature of unvoiced

9. 9
GSM Compression Technologies
 Four compression technologies are:
 Full Rate
 Enhanced Full Rate (EFR)
 Adaptive Multi-Rate (AMR)
 Half Rate

10. 10
GSM Full Rate Vocoder Using RPE-LTP
 Described as an RPE-LTP linear predictive coder.
 Models the human vocal tract as a series of cylinders of
different widths.
 By forcing air through these cylinders, speech sounds
can be generated— the LPC coder models this with a
set of simultaneous equations.

11. 11
GSM Full Rate Vocoder Using RPE-LTP
(…contd)
 The input data to the RPE-LTP coder is 20ms of speech
composed of 160 samples, each with 13bit resolution.
 The data is first passed through a pre-emphasis filter:
 Enhances high-frequency components of the signal. (better
transmission efficiency.)
 Also removes any offset on the signal. (Simplifies computation.)

12. 12
LPC Speech Generation
 The model of speech generation can be thought of as air passing
through a set of different size cylinders.

13. 13
Short Term Analysis Stage
 Uses autocorrelation to calculate a set of eight reflection
coefficients.
 Schur recursion is used to efficiently solve the set of
equations resulting from it.
 The parameters are then converted into log-area ratios
(LARs) -- that allow better quantizing in a smaller
number of bits — the first eight parameters of the
transmission stream.

14. 14
 The coded LARs is then decoded back to coefficients
and used to filter the input samples.
 The reason for decoding the LARs is to ensure that the
encoder uses the same information available at the
decoder to perform the filtering.
 An array of weights lpc[P] is computed such that
s[n] ~ lpc[0]*s[n--1]+lpc[1]*s[n--2]+_+lpc[P--1]*s[n--P]
(P is usually between 8 and 14, GSM uses 8.)
Short Term Analysis Stage (…contd)

15. 15
Long Term Prediction Stage
 The 160 samples are split into 4 sub-windows of 40
samples each.

16. 16
 The long-term predictor produces two parameters for
each sub window: the lag and the gain.
 The LTP lag describes the source of the copy in time.

The LTP gain describes the scaling factor.
Long Term Prediction Stage (…contd)

17. 17
Calculating Lag and Gain
 LAG:
Compute resemblance by correlation.
correlation of x[n] and y[n] =
Sum of products x[n]*y[n-lag]
 GAIN:
Maximum correlation divided by the energy of the
reconstructed short-term residual signal.

18. 18
Residual Pulse Encoding
 To remove the long-term predictable signal from
its input, the algorithm then subtracts the scaled
40 samples.
 The residual signal is either weak or random and
consequently cheaper to encode and transmit.

19. 19
Residual Signal(…contd)
 The algorithm down-samples by a factor of three,
discarding two out of three sample values.
 Results in four evenly spaced 13-value subsequences to
choose from, starting with samples 1, 2, 3, and 4.
 The algorithm picks the sequence with the most energy.
 That leaves us with 13 3-bit sample values and a 6-bit
scaling factor that turns the PCM encoding into an
APCM

20. 20
Speech Decoder
 Decoder consists of three parts
 RPE Decoding
 LTP synthesis filter
 LPC short term synthesis filter

21. 21
Speech Decoder(…contd)

22. 22
Speech Decoder (…contd)
 Algorithm multiplies the 13 3-bit samples by the scaling factor and
expands them back into 40 samples, zero-padding the gaps
 Resulting residual pulse is fed to the long-term synthesis filter
 40-sample segment is cut from the old estimated short-term residual
signal, scaled by the LTP gain and added to the incoming pulse
 Estimated short-term residual signal passes through the short-term
synthesis filter whose reflection coefficients are calculated by the
LPC module
 Noise from the excited long-term synthesis filter passes through the
tubes of the simulated vocal tract--and emerges as speech

23. 23
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