TELE4652 Mobile and Satellite Communications Lecture 9 – Speech Coding Overview • Important in the development of Cellular Networks • Speech compression advances were significant in the growth in network capacity from 1G networks to 2G networks (and beyond) • Techniques can reduce the amount of digital data required to represent speech by factors of ten or more at no loss in perceptual quality • This lecture will provide a brief overview of the important ideas of speech compression TELE4652
33
Embed
TELE4652 Mobile and Satellite Communications · •Basic PCM system •Quantisation is the loss of information –signal distortion TELE4652 Quantisation of Speech Samples •Perceptual
This document is posted to help you gain knowledge. Please leave a comment to let me know what you think about it! Share it to your friends and learn new things together.
Transcript
TELE4652 Mobile and Satellite
Communications
Lecture 9 – Speech Coding
Overview
• Important in the development of Cellular
Networks
• Speech compression advances were significant in
the growth in network capacity from 1G networks
to 2G networks (and beyond)
• Techniques can reduce the amount of digital data
required to represent speech by factors of ten or
more at no loss in perceptual quality
• This lecture will provide a brief overview of the
important ideas of speech compression
TELE4652
Speech Processing
• Speech Compression is one aspect of the overall
speech processing
• For example, GSM speech processing is shown
below:
TELE4652
GSM Speech Processing
Aspects of a speech processing system:
1.A/D conversion. GSM samples at 8kHz with 13
bits/sample. Raw data rate is 104kbps.
2.Framing – groups together a sequence of speech
samples to form a frame. GSM takes frames of 160
samples every 20ms.
3.Frame classification – a Voice Activity Detection
(VAD) algorithm is used to identify whether frame
contains speech or not
TELE4652
GSM Speech Processing
4. Discontinuous Transmission System (DTX) – don’t transmit if there is no speech within the frame. This
• Prolongs battery life
• Reduces the level of interference across the network
There should be speech data to transmit less than half the time on average.
5. Comfort Noise Generation – If there is no speech data, the receiver will generate some comfort noise (background noise)
6. Silence Descriptor (SID) – Tx sends an estimate of the background noise. In GSM this is done once every 480ms
TELE4652
GSM Speech Processing
7. Error Concealment – bit errors on channel can cause speech frames to be corrupted. If detected -> Bad Frame Indicator (BFI). Lost speech frame is replaced by prediction from previous frames. 16 consecutive lost frames results in failure of acoustic channel.
8. Speech Codec (Compression) – GSM uses a technique called Regular Pulse Excitation – Long Term Prediction (RPE-LTP). This is a type of Linear Predictive Coding (LPC). Output is 260 bits for every frame (data rate out is thus 13kbps)!!!
TELE4652
GSM Speech Processing
Overview:
TELE4652
Digitising Speech
• Telephone quality speech is
generally taken as band-limited to
[300Hz, 3.4kHz)
• Usual sampling frequency is 8kHz
• High quality audio (for music)
requires a greater bandwidth
• Speech digitisation systems can
be understood as Pulse Code
Modulation (PCM)
• Sample, Quantise, Encode
TELE4652
Pulse Code Modulation
• Basic PCM system
• Quantisation is the loss of information – signal
distortion
TELE4652
Quantisation of Speech Samples
• Perceptual quality requires at least 13 bits/sample
• From Signal to Quantisation Noise Ratio (SQNR)
• Resulting data rate, 8kHz x 13 bits/sample =
104kbps is much too high for cellular application
• Non-uniform quantisation – companding
• Companding = Compressing + Expanding. Non-
linear amplification of speech
• Common schemes – A-law and µ-law
• Used in the PSTN
TELE4652
Companding
• Non-uniform amplification in the time domain
• Idea is that there is more information contained in
the low amplitude parts of a speech waveform
• There should be more quantisation levels at low
amplitudes
TELE4652
Companding
• Reduces the number of bits per sample to 8
• Resultant bit rate is only 64kbps
• Not enough for cellular systems, but used in
Cordless Phones
• Practically either performed by non-linear
amplification prior to quantisation, or by direct non-
uniform quantiser
TELE4652
Adaptive Differential PCM
• Use an adaptive algorithm to predict the next speech
sample
• Quantise and encode the difference between the
prediction and the actual sample
• Can adapt step-size based on the received signal
characteristic
• This reduces the dynamic range of the quantiser, and hence
the number of bits required in representation
• The receiver can use the same predictive algorithm, and
then receives the difference to add to its prediction
TELE4652
Adaptive DPCM
• Adaptive DPCM encoder
TELE4652
Adaptive DPCM
• Predictive – estimate the next sample from the
previous sample(s)
• Then quantise and encode these prediction errors