GSM Network and Services

Similar documents
Digital Speech Coding

Speech-Coding Techniques. Chapter 3

EUROPEAN ETS TELECOMMUNICATION January 1999 STANDARD

Open AMR Initiative. Technical Documentation. Version 1.0 Revision

Source Coding Basics and Speech Coding. Yao Wang Polytechnic University, Brooklyn, NY11201

Voice Quality Assessment for Mobile to SIP Call over Live 3G Network

ETSI EN V8.0.1 ( )

RTP implemented in Abacus

AUDIO. Henning Schulzrinne Dept. of Computer Science Columbia University Spring 2015

ETSI TS V ( )

EUROPEAN ETS TELECOMMUNICATION February 1998 STANDARD

A MULTI-RATE SPEECH AND CHANNEL CODEC: A GSM AMR HALF-RATE CANDIDATE

2.4 Audio Compression

CT516 Advanced Digital Communications Lecture 7: Speech Encoder

The Steganography In Inactive Frames Of Voip

Alcatel OmniPCX Enterprise

Synopsis of Basic VoIP Concepts

EN V6.0.1 ( )

ETSI TS V5.2.0 ( )

ABSTRACT. that it avoids the tolls charged by ordinary telephone service

Voice Over LTE (VoLTE) Technology. July 23, 2018 Tim Burke

Audio Fundamentals, Compression Techniques & Standards. Hamid R. Rabiee Mostafa Salehi, Fatemeh Dabiran, Hoda Ayatollahi Spring 2011

Principles of Audio Coding

GSM System Overview. Ph.D. Phone Lin.

ETSI TS V ( )

ETSI TS V8.2.1 ( )

ETSI TS V ( )

Mahdi Amiri. February Sharif University of Technology

TELE4652 Mobile and Satellite Communication Systems

Data Compression. Audio compression

Multimedia Systems Speech II Mahdi Amiri February 2012 Sharif University of Technology

Multimedia Systems Speech II Hmid R. Rabiee Mahdi Amiri February 2015 Sharif University of Technology

ETSI TS V3.1.0 ( )

Program Title: Telecommunication Engineering Technology Postsecondary Number: (AS) (AAS

Audio Coding and MP3

ETSI TS V7.3.0 ( )

Overcoming Barriers to High-Quality Voice over IP Deployments

Network Extension Unit (NXU)

GSM. Course requirements: Understanding Telecommunications book by Ericsson (Part D PLMN) + supporting material (= these slides) GPRS

Perceptual coding. A psychoacoustic model is used to identify those signals that are influenced by both these effects.

ETSI TS V4.2.0 ( )

Bluray (

Nokia Q. Xie Motorola April 2007

White Paper Voice Quality Sound design is an art form at Snom and is at the core of our development utilising some of the world's most advance voice

EN V6.0.1 ( )

Both LPC and CELP are used primarily for telephony applications and hence the compression of a speech signal.

Lecture overview. Modifications and derivatives of GSM Data transmission in GSM: HSCSD GPRS part one EDGE

3G TS V3.0.0 ( )

Application of wavelet filtering to image compression

ETSI TR V7.2.0 ( )

Application Report SPRA582

Pertemuan 7 GSM Network. DAHLAN ABDULLAH

Real-time Audio Quality Evaluation for Adaptive Multimedia Protocols

ETSI TS V ( )

ITNP80: Multimedia! Sound-II!

Assessing Call Quality of VoIP and Data Traffic over Wireless LAN

Introduction to Quality of Service

CONCEPTS AND SOLUTIONS FOR LINK ADAPTATION AND INBAND SIGNALING FOR THE GSM AMR SPEECH CODING STANDARD

UNIT-5. GSM System Operations (Traffic Cases) Registration, call setup, and location updating. Call setup. Interrogation phase

CAPTURING AUDIO DATA FAQS

For Mac and iphone. James McCartney Core Audio Engineer. Eric Allamanche Core Audio Engineer

Automated Testbench Generation for Communication Systems

CDMA. Fundamentals of Wireless Communications & CDMA. Student Guide CDMA X6 January 24, 2000

ETSI TS V1.1.1 ( )

Publication of specifications for the mobile network interfaces offered by Wind

2 Framework of The Proposed Voice Quality Assessment System

Investigation of Algorithms for VoIP Signaling

ENSC 427 COMMUNICATION NETWORKS

Internal. GSM Fundamentals.

Chapter 9 Xoring Stage

6MPEG-4 audio coding tools

ARIB STD-T53-C.S Circuit-Switched Video Conferencing Services

Meeting #29 Agenda items: rd 25 th June, 1999, Miami. Adaptive Multi-Rate Wideband (AMR-WB) Feasibility study report. Version 1.0.

Extraction and Representation of Features, Spring Lecture 4: Speech and Audio: Basics and Resources. Zheng-Hua Tan

Overview. A Survey of Packet-Loss Recovery Techniques. Outline. Overview. Mbone Loss Characteristics. IP Multicast Characteristics

3GPP TS V ( )

PASS4TEST. IT Certification Guaranteed, The Easy Way! We offer free update service for one year

Speech and audio coding

ETSI TS V5.0.0 ( )

ETSI TS V ( )

Compressed Audio Demystified by Hendrik Gideonse and Connor Smith. All Rights Reserved.

Networking Infrastructure Markets

ELL 788 Computational Perception & Cognition July November 2015

ETSI TS V (201

A Synchronization Scheme for Hiding Information in Encoded Bitstream of Inactive Speech Signal

SAOC and USAC. Spatial Audio Object Coding / Unified Speech and Audio Coding. Lecture Audio Coding WS 2013/14. Dr.-Ing.

Audio and video compression

A Review on various Voices over IP codecs for WMN

ON-LINE SIMULATION MODULES FOR TEACHING SPEECH AND AUDIO COMPRESSION TECHNIQUES

3GPP TS V ( )

Transporting audio-video. over the Internet

MPEG-4 General Audio Coding

ETSI TS V (201

Discontinuous Transmission (DTX) of Speech in cdma2000 Systems

White Paper. Optimal Codec Selection in International IP based Voice Networks. (Release 2.0) May 2010

CS 335 Graphics and Multimedia. Image Compression

MOHAMMAD ZAKI BIN NORANI THESIS SUBMITTED IN FULFILMENT OF THE DEGREE OF COMPUTER SCIENCE (COMPUTER SYSTEM AND NETWORKING)

ITU-T G.113. Transmission impairments due to speech processing

Improving QoS of VoIP over Wireless Networks (IQ-VW)

GSM System Protocol Architecture

Transcription:

GSM Network and Services Voice coding 1

From voice to radio waves voice/source coding channel coding block coding convolutional coding interleaving encryption burst building modulation diff encoding symbol coding carrier modulation 2

Why digital? bit rate digital analog speech quality compressed link quality 3

Analog vs Digital If radio conditions are very good, an analog connection could do the job and in many cases a better job. Digital coding allows: compression error detection and error correction half duplex implementation adaption to radio conditions Digital drawback: processing, delay, voice only codecs. 4

GSM voice codec Sampled at 8 khz, 13 bits per sample. Compare this to regular A-law coding that samples at 8kHz using 8 bits per sample. Divides the sampled voice into 20 ms blocks for the codec to work on. Each block is coded in 260 bits resulting in 13 kb/s. The GSM codec is based on modeling of human speech organs. Tuned for males! Voice activation to allow discontinuous transmission. 5

Remember The 260 bits from the voice codec are encoded using 456 bits that are interleaved into eight bursts (half of each burst). A 26-multiframe last for 120 ms and holds 24 TCH burst, 1 SACCH burst and one idle frame. 24/4 = 6 voice samples every 26-multiframe. 26-multiframe takes 120 ms that is equal to 6 voice samples of 20 ms. One frame takes 120/26 = 4.6 ms. 6

GSM discontinuous transmission voice activation det. VAD frame sample voice coder voice DTX coder SP comfort noise synth SID Silence Descriptor 7

DTX coder The coder will detect if the sample is voice or silence. When detected the Voice activity detection will set a VAD bit to 1. The voice coder will output a voice frame of 260 bits or a SID frame of 35 bits. Depending on the VAD bit the DTX will output voice frames or SID frames. The DTX will pass the voice frames or in band encoded SID frames to the channel coder. 8

DTX encoder The channel coder will, based on the SP flag, control when data is actually sent over the air: any frame that contains speech information the first frame after a speech frame since it holds a SID frame frames to keep the radio measurement active 9

Comfort noise If there is no noise in the background when you speak you will think that the connection is lost. This is extremely annoying Since phone conversations are almost always half duplex it is wasteful to keep a full duplex connection open. Why? Wasting what? 10

DTX decoder BFI bad frame DTX voice voice coder sample SID comfort noise synth 11

DTX decoder Correct voice frames are passed directly to the voice decoder. Correct SID frames are passed to the comfort noise synthesizer. If the Bad Frame Indicator (BFI) is set then an incorrect voice frame is replaced by the previous an incorrect SID frame is replaced by the last valid SID frame or last valid speech frame (that probably contains noise) 12

Waveform vs Vocoder Waveform codecs encode audio with no knowledge of the signal. Pulse Code Modulation (PCM) is one example that is used in fixed line telephony. Vocoder, or source coders, try to represent the audio generator, in this case the vocal tract, and represent a sample as a the parameters that define the generator. Vocoders can encode speech at a very low bit rate. Even if the result does not sound like the original audio signal. 13

quality 10 9 8 Voice codec Codecs waveform 7 6 hybrid 5 4 vocoder 3 2 1 0 0 1 2 4 8 16 32 64 bit rate kb/s 14

Hybrid coder In a hybrid coder a vocoder is used to encode the voice signal. The decoded sample of the result is then compared to the original sample. The difference between the original and the decoded result is encoded using a waveform coder. The result is a high quality low bit rate codec. 15

GSM codec 2080 bits 104kb/s Linear Predictive Coder short-term LPC short-term filter long-term filter 36 bits (1.8 kb/s) Regular Pulse Excitation RPE encoding 188 bits 9.4 kb/s RPE-LTP LPC LTP analysis Long Term Prediction 36 bits (1.8 kb/s) 16

GSM codec The resulting 260 bits are divided into: Class 1A: 50 most significant bits Class 1B: 132 significant bits Class 2: 78 less significant bits Classing is done by subjective evaluation of the perceived speech quality (Mean Opinion Score, MOS). 17

Channel coding IA 50 b IB 132 b II 78 b 25 25 3 25 66 3 66 25 4 378 b after ½ convolutional coding II 78 b 18

Other codecs Half rate (HR): 6.5 kb/s used to increase capacity since it only needs a half rate TCH. Speech quality is considerably less. Enhanced Full Rate (EFR): 12.2 kb/s but further protected by CRC in a resulting 13 kb/s codec. Better speech quality. Adaptive Multi Rate (AMR): the codec adapts to the radio conditions; high BER will use a low rate codec that is better protected. 19

AMR kb/s 12.2 10.2 6.7 4.75 Block 244 204 134 95 Class 1A 81 65 55 39 Rate 1/2 1/3 1/4 1/5 Raw 508 642 576 535 Puncturing 60 194 128 87 Result 448 448 448 448 In band 8 8 8 8 Frame 456 456 456 456 20

BSS side coder MS GSM voice signaling BSC TRAU A-law ISDN MSC The TRAU needs not only the 13kb/s voice stream but also BFI information etc. 3 kb/s signaling and error detection is added resulting in a 16 kb/s stream. 4 streams will share a 64 kb/s ISDN channel 21