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Robust Audio Tool (RAT) Speaker : Wei-Shin Speaker : Wei-Shin Pan Pan DATE : 09/07/02 DATE : 09/07/02

Robust Audio Tool (RAT) Speaker : Wei-Shin Pan DATE : 09/07/02

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Page 1: Robust Audio Tool (RAT) Speaker : Wei-Shin Pan DATE : 09/07/02

Robust Audio Tool (RAT)

Speaker : Wei-Shin Speaker : Wei-Shin PanPanDATE : 09/07/02DATE : 09/07/02

Page 2: Robust Audio Tool (RAT) Speaker : Wei-Shin Pan DATE : 09/07/02

2009 Graduate Institute of Communication Engineeri2009 Graduate Institute of Communication Engineeringng2

OutLine

Introduction

RAT

DEMO

Keep goingKeep goingKeep goingKeep going

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Introduction

• An open-source audio conferencing and streaming application

• RAT transmission– Unicast (Point-to-point streaming)– Multicast (conference)

• 224.2.0.0 – 224.2.255.255

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Architecture(1)

• Three components

RATRAT

RatmediaRatmedia

RatuiRatui

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Architecture(2)

• RAT – arguments analyzing and controls the other c

omponents.

• RATmedia– voice processing and transmisstion

• RATui– Graphical user Interface

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RAT

• How to start– RAT Get !

• https://frostie.cs.ucl.ac.uk/mbone/mmedia/wiki/RatDownload• Choose the build platform

• The stable version : RAT4 (4.4.01)

– Installation– Start it !

• RAT launched from the command line• Syntax :

– > rat address/port

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User interface

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Configuration

• Options– Personal– Transmission– Reception– Audio– Codecs– Security– Interface

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Personal

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Transmission

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Codecs

• G.711 u-law & a-law 64kbit/s , MOS 4.3

• Linear-16 128kbit/s (RAW)

• GSM 13.2kbit/s , MOS 3.5

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Loss protection(1)

• Loss protection

1 2 3 4 5 6 7 8 9FRAMES

1 2 3Sequential Frames

(A)4 5 6 7 8 9

1 4 7 2 5 8 3 6 9Distribution Frames(B)

1 2 3 7 8 9Packet loss

(A)

1 4 73 6 9Packet loss

(B)

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Loss protection(2)

• Redundancy

1 2 3 4 5FRAMES

1 2 3 41 2 3

High quality Low quality

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Loss protection(3)

• Layering

Time

Frames

(Low Priority )

(High priority) Base

Normal

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Repair scheme

• Pattern-Match– 將遺失的部份聲音 , 以它前後的聲音透過 interpolate 來

回復 , 需較大的計算量• Repeat

– 基於人聲沒有快速變化的特性 , 將遺失掉的封包用前一個封包來代替 , 但當封包長度和封包遺失率太大時 , 就沒辦法利用人聲的特性了

• Noise– 透過隨機分佈雜訊聲音來取代遺失的聲音區塊 , 會被封

包長度跟封包遺失率大小所影響

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Sample rate conversion

• If local’s soundcard can’t support sample rate of received packets , and converting the sample rate by this option.

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Limit Playout delay

• Jitter– jitter buffer

• Adjustable range : 0-2000ms

• IF value is high, the voice packets will play smoothly and poor quality (high delay)

• IF value is low, some voice packets can’t play because the delay too short.

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Jitter Buffer (With Playout buffer )is designed to rem

ove the effects of jitter from a decoded voice stream and buffer each arriving packet for a short interval before playing out. This compensates an additional delay and packet loss for jitter.

Jitter Jitter bufferbuffer

INTERNET||||||||||||||||||||||||||||||||

J I TTER buffer

||||||||||||||||||||||||||||||||||||||||

VOIP system with J itter buffer

I nside the PC

Voice pacKets

SenderReceiver

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Silence suppression

• Much silence gaps (No talking) in a conversation

• Background Noise– No packet sending will reduce 50%

bandwidth

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DES (data encryption standard)

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DEMO

Multicast

Unicast

Run -> cmd> rat destination _address/port

Run -> cmd> rat 224.224.1.2/port

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Reference [1] The Study and Analysis of the Robust-Audio Tool for Real-Time Audio Conference, Ting-Yu Chiu

[2] Robust Audio Tool official website