Speech/Audio Coding for IP networks
Transcript of Speech/Audio Coding for IP networks
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ETSI STQ Workshop “Compensating for Packet Loss in Real-Time Applications”, Feb 2003
Speech/Audio Coding for IP networks
Alan DuricSen. Systems ArchitectSIP/email: [email protected]
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Agenda
• Introduction• Traditional approach to speech coding for VoIP
applications• New paradigm• iLBC• Questions• Demo
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QoS – (endpoints) perspective
• Year after year the same story
• More then 3000 papers since 1984
• Limited ToS support at the end points
• Introduction of new technologies and applications is making situation even more difficult
QoSQoS is already 19 years old is already 19 years old ––is it time to get a real job !? is it time to get a real job !? ☺☺[ ]
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Traditional approach to speech coding for VoIP
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CELP SPECIFICS
• Current low bit rate codecs: ITU G.729, G.723.1, GSM-EFR, and 3GPP-AMR were developed for circuit switched & wireless telephony and are all based on the CELP (Code Excited Linear Prediction) paradigm.
• CELP coders are stateful, they have memory, error propagation results from lost or delayed packets.
• Long time is needed to resynchronize coder and decoder (often 70-100 ms)
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original
iLBC
g729
g723
PLC Staterecovery
iLBC Advantage over CELP
iLBC, like other GIPS codecs treats every packet individually, making it suitable for packet communications.[ ]
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New Paradigm
• Approach & diagnose• What can be improved?• How?• Proof of (concept and design)
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Approach
We need holistic view/approach for both
• Horizontal (end-to-end) QoS perspective
• Vertical (top-down) QoS perspective
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Vertical (Top Down) Perspective
PhysicalPhysical
LinkLink MLPPP/FR/ATM AAL1MLPPP/FR/ATM AAL1
NetworkNetwork IP/WFQ/IPIP/WFQ/IP--precprec
TransportTransport RTP/UDP/RSVPRTP/UDP/RSVP
SessionSession SIP/H.323SIP/H.323
PresentationPresentation Speech Codecs/…Speech Codecs/…
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EC
What impacts perceived quality?VoIP End Point
Lost Packets
CODEC
Delay & Jitter
Packet discardsby jitter buffer
VoIP End Point
Needed to mitigate impact of Needed to mitigate impact of delay/jitter and packet lossdelay/jitter and packet loss[ ]
Echo
Playout Controller
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What can be improved?
• One side solutions:– Advanced Playout Controller– AEC, NEC with right design
• Both end solution:– Codec
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Design principles
• High basic quality• Robustness (e.g. for codec no inter-frame dependency,
MDC)• Low complexity• …• Realistic test methodology and tools during design
phases
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MDC
Input SpeechFrame
Packets sent onIP Network
N + 1 N N - 1N + 2 NN + 1D2D1D1D1 D2D2
20 msN+2
20 msN+1
20 msN
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Improvements for “one-side” solutions
Source: Lockheed Martin Global Telecommunications (COMSAT)
Adaptive jitter bufferJitter Fixed jitter buffer NetEQ™Adaptive jitter bufferAdaptive jitter bufferJitterJitter Fixed jitter bufferFixed jitter buffer NetEQ™NetEQ™140
120
20
80
100
40
60
0
140
120
20
80
100
40
60
00 200 400 600 800 1000 1200 1400 1600 1800 2000
Packet number
Dela
y (m
s)
0 200 400 600 800 1000 1200 1400 1600 1800 20000 200 400 600 800 1000 1200 1400 1600 1800 2000
Packet number
Dela
y (m
s)Saved approximately 30—80 ms
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iLBC Performance
Source: Dynastat
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GIPS Ehanced G.711+GIPS NetEQ™
G.711+GIPS NetEQ™
G.711+ITU PLC
G.729A
G.711+No PLC
Matching PSTN QualityTelephony bandwidth speech test result
SOURCE LOCKHEED MARTIN GLOBAL TELECOMMUNICATION (COMSAT)
5
4.5
4.0
3.5
3.0
2.5
2.0
1.5
1.0
NETWORK CONDITION (% PACKET LOSS)
0% 5% 30%10% 15% 25%20%
MO
S
GIPS iPCM™-wb+GIPS NetEQ™-wb
G.722+GIPS NetEQ™-wbG.722.1
Source +no PLC
Better Than PSTN QualityWideband speech
0% 5% 10% 15% 20% 25%
5
4.5
4.0
3.5
3.0
2.5
2.0
1.5
1.0
MO
S
NETWORK CONDITION (% PACKET LOSS)
wide band sound quality
Proof of concept and design (part 3)
telephony band sound quality
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iLBC (Internet Low Bitrate Codec)
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iLBC (internet Low Bitrate Codec)
• Speech sampled at 8 kHZ,• using a block-independent linear-predictive coding (LPC)
algorithm.• Bandwidth 13.33 kbps (50 bytes per 30 ms)• Frame size 30 ms (support for 20 ms in the next revision)• Complexity and memory requirements are similar to ITU
G.729A• Basic Quality is equal to or better than G.729. Packet loss
robustness is significantly better than G.729. • Packet loss concealment - Integrated example solution
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The Core iLBC method
• Gain-shape waveform matching forward in time• Gain-shape waveform matching backward in time
• Start state encoding
• Pitch enhancementOriginal speech segment
Decoded segment representation
• Packet loss concealment
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iLBC - IETF work
• IETF deliverables, submitted during February ‘02:– iLBC codec specification draft - experimental
standards track– iLBC RTP Payload Profile - regular standards
track (AVT)– Statement about IPRs in ILBC and its “freeware nature”
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Summary
• Accelerate deployment of VoIP technology by using realistic QoS enhancements and solutions that are already available
• VoIP endpoints, focus on both: one side improving solutions and both end improving solution
• Move quality exprience to the next level with wideband coders
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Questions ???
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Demo
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More information
• Web site www.ilbcfreeware.org with:– Info about initiative– Info about codec– Latest iLBC IETF drafts (spec and payload format)– Latest iLBC float point Source code– FAQ list
• IETF drafts:– draft-ietf-avt-ilbc-00.txt - codec spec (exper. stds track)– draft-ietf-avt-rtp-ilbc-00.txt - RTP payload profile (AVT group)
• Web site www.globalipsound.com
• Free demo SIP client available, please request at:SIP/email: [email protected]