ADAPTIVE

MULTIRATE
SPEECH CODING

Presented By:
Irfan Yaqoob

RADIO PARAMETERS

FRAME STRUCTURE

SOURCE CODING

Compresses the amount of information over

the given channel.
Improves the spectral efficiency of the radio
interface.
Related to the network capacity.
Allows the system to introduce more powerful
encoding techniques to counter the
propagation & interference effects.

CHANNEL CODING

Conditions the output of the source encoder

for transmission over the channel.
Transcoded speech is error protected by
passing it through the channel encoder.
Related to the network quality.
Includes :

Coding for forward error correction & detection

Bit interleaving
Modulation

SPEECH ENCODING &

MODULATION

Digitized Speech is passed at 64kbps through

speech coder, compresses the 64kbps PCM
speech to 13kbps data rate.
The transcoded speech is error protected by
passing it through channel encoder, that
utilizes parity & convolutional code thus
increasing the bite rate to 22.8kbps (for
GSM-FR) and 11.4kbps (for GSM-HR).
The 456 bits are interleaved to combat burst
errors. Channel encoded bits over two
adjacent 20ms interval are split into eight
blocks (114 bits) and transmitted over eight
frames.

Interleaved data is then modulated at

270.8kbps using GMSK.
Then it is passed through Duplexer to isolate,
transmit and receive signals.
The reverse process is applicable for the
downlink of signals.

CODEC

Digital algorithm used to encode speech

signals.
Codecs used in GSM

Half Rate (HR)

Full Rate (FR)
Enhanced Full Rate (EFR)
Adaptive Multirate (AMR)

FULL RATE (GSM - FR)

Specified in ETSI 06.10

Based on RPE-LTP (Regular Pulse ExcitationLong Term Prediction)
Uses linear prediction in as many others.
Provides a bit rate of 13kbps. (260bits/20ms)
Gradually replaced by EFR and AMR due its poor
quality.

REGULAR PULSE EXCITATION LONG

TERM PREDICTION (RPE-LTP)

Used in order to reduce the data sent b/w MS

and the BTS.
When a voltage level of a particular speech
sample is quantified, the Mobile Station's
internal logic predicts the voltage level for the
next sample.
When the next sample is quantified, the packet
sent by the MS to the BTS contains only the
error.

HALF RATE (GSM - HR)

Specified in ETSI 06.20

Based on VSELP (Vector Sum Excited Linear
Prediction) Algorithm.
Provides a bit rate 6.5kbps. (130bits/20ms)
Requires half the bandwidth, so network capacity
is doubled at the expense of audio quality.
Consumes 30% less energy.

CODE EXCITED LINEAR PREDICTOR

(CELP)

Coder & decoder have predetermined book of

stochastic excitation signals.
For each speech signal, the transmitter searches
its code book of stochastic signals and the
index of one that gives the best match is
transmitted.
The receiver uses this index of code book to pick
the correct excitation signal for its synthesizer
filter.
CELP are extremely complex, but can achieve
bit rates of as low as 4.8kbps.

VECTOR SUM EXCITED LINEAR

PREDICTOR (VSELP)

Utilizes three excitation sources or codebooks.

Each of these contain the equivalent of 128
vectors.
The three excitation sequences are multiplied by
corresponding gain terms and summed up.
The combined excitation sequence is used for
synthesizer filter.
Provides highest speech quality, low
computational complexity & robustness to
channel errors.

ALGERBRAIC CODE EXCITED LINEAR

PREDICTOR (ACELP)

ACELP codebooks have a specific algebraic

structure.
A 16-bit algebraic codebook shall be used in the
innovative codebook search, the aim of which
is to find the best innovation and gain parameters.
The innovation vector contains, at most, four
non-zero pulses.

ENHANCED FULL RATE (GSM EFR)

Developed to improve the poor quality of FR.

Provides a bit rate of 12.2kbps. (244bits/20ms)
Compatible with the highest AMR mode.
Consumes 5% more energy.
Recommended to use only in poor reception
areas.

ADAPTIVE MULTIRATE (AMR)

Audio data compression technique for speech

encoding.
The AMR codec uses eight source codecs with
bit rates of 12.2, 10.2, 7.95, 7.40, 6.70,
5.90, 5.15 and 4.75 kbps.
It uses link adaptation to select from one of
eight different bit rates based on link
conditions.
Link adaptation is the selection of the best
codec mode to meet the local radio channel
and capacity requirements.

ADAPTIVE MULTIRATE (AMR)

If the radio conditions are bad, source coding

is reduced and channel coding is increased.
This improves the quality and robustness of
the network connection while sacrificing
some network capacity.
AMR utilizes Discontinuous Transmission
(DTX), with Voice Activity Detection
(VAD) and Comfort Noise Generation (CNG) to
reduce bandwidth usage during silence
periods.

ADAPTIVE MULTIRATE (AMR)

Mode

Bit Rate (kbps)

Channel

AMR_12.20

12.2 (244 bits) FR

AMR_10.20

10.2 (204 bits) FR

AMR_7.95

7.95 (159 bits) FR/HR

AMR_7.40

7.4

(148 bits) FR/HR

AMR_6.70

6.7

(134 bits) FR/HR

AMR_5.90

5.9

(118 bits) FR/HR

AMR_5.15

5.15 (103 bits) FR/HR

AMR_4.75

4.75 (95 bits)

FR/HR

VOICE ACTIVITY DETECTION

Technique in speech processing where the

presence of human speech is detected in the
regions of audio.
Its main uses are in speech coding and
speech recognition.
Deactivates some processes during nonspeech segments to avoid unnecessary
coding/transmission of silence packets.
Done at the transmitters (MS) end.

COMFORT NOISE

Artificial background noise used to fill the

silence in a transmission resulting from voice
activity detection.
The result of receiving total silence, especially
for a prolonged period, has a number of
unwanted effects on the listener.
To counteract these effects, comfort noise is
added.
Done at the receivers (BTS) end.

DISCONTINOUS TRANSMISSION
(DTX)

Method of momentarily powering-down, or

muting, a mobile set when there is no voice
input to the set.
This conserves battery power, eases the
workload of the components in the
transmitter amplifiers, and reduces
interference.
Resources freed up when one user is in
silence can be used to serve another user, thus
increasing capacity of the network.
Operates using VAD and CNG.

The End