ITU

Committed to connecting the world

ITU-T work programme

[2005-2008] : [SG16] : [Q9/16]

[Declared patent(s)]  - [Publication]

Work item: G.718 (ex G.VBR-EV)
Status: Approved on 2008-06-13 
Approval process: AAP
Type of work item: Recommendation
Version: New
Provisional name: G.VBR-EV
Equivalent number: -
Timing: -
Liaison: 3GPP, 3GPP2, ISO/IEC JTC1/SC29
Subject/title: Frame error robust narrowband and wideband embedded variable bit-rate coding of speech and audio from 8-32 kbit/s
Summary: This Recommendation describes a narrowband (NB) and wideband (WB) embedded variable bit-rate coding algorithm for speech and audio operating in the range from 8 to 32 kbit/s which is designed to be robust to frame erasures. This codec provides state-of-the-art NB speech quality over the lower bit rates and state-of-the-art WB speech quality over the complete range of bit rates. In addition, G.718 is designed to be highly robust to frame erasures, thereby enhancing the speech quality when used in IP transport applications on fixed, wireless and mobile networks. Despite its embedded nature, the codec also perform with both NB and WB generic audio signals. This codec has an embedded scalable structure, enabling maximum flexibility in the transport of voice packets through IP networks of today and in future media-aware networks. In addition, the embedded structure of G.718 will easily allow the codec to be extended to provide a superwideband and stereo capability through additional layers which are currently under development. The bitstream may be truncated at the decoder side or by any component of the communication system to instantaneously adjust the bit rate to the desired value without the need for out-of-band signalling. The encoder produces an embedded bitstream structured in five layers corresponding to the five available bit rates: 8, 12, 16, 24 & 32 kbit/s. The G.718 encoder can accept WB sampled signals at 16 kHz, or NB signals sampled at either 16 or 8 kHz. Similarly, the decoder output can be 16 kHz WB, in addition to 16 or 8 kHz NB. Input signals sampled at 16 kHz, but with bandwidth limited to NB, are detected by the encoder. The output of the G.718 codec is capable of operating with a bandwidth of 50-4000 Hz at 8 and 12 kbit/s and 50-7000 Hz from 8 to 32 kbit/s. The high quality codec core represents a significant performance improvement, providing 8 kbit/s wideband clean speech quality equivalent to G.722.2 at 12.65 kbit/s whilst the 8 kbit/s narrowband codec operating mode provides clean speech quality equivalent to G.729E at 11.8 kbit/s. The codec operates on 20 ms frames and has a maximum algorithmic delay of 42.875 ms for wideband input and wideband output signals. The maximum algorithmic delay for narrowband input and narrowband output signals is 43.875 ms. The codec may also be employed in a low-delay mode when the encoder and decoder maximum bit rates are set to 12 kbit/s. In this case the maximum algorithmic delay is reduced by 10 ms. The codec also incorporates an alternate coding mode, with a minimum bit rate of 12.65 kbit/s, which is bitstream interoperable with ITU-T Recommendation G.722.2, 3GPP AMR-WB and 3GPP2 VMR-WB mobile WB speech coding standards. This option replaces Layer 1 and Layer 2, and the layers 3-5 are similar to the default option with the exception that in Layer 3 few bits are used to compensate for the extra bits of the 12.65 kbit/s core. The decoder is further able to decode all other G.722.2 operating modes. G.718 also includes discontinuous transmission mode (DTX) and comfort noise generation (CNG) algorithms that enable bandwidth savings during inactive periods. An integrated noise reduction algorithm can be used provided that the communication session is limited to 12 kbit/s. The underlying algorithm is based on a two-stage coding structure: the lower two layers are based on Code-Excited Linear Prediction (CELP) coding of the band (50-6400 Hz) where the core layer takes advantage of signal-classification to use optimized coding modes for each frame. The higher layers encode the weighted error signal from the lower layers using overlap-add MDCT transform coding. Several technologies are used to encode the MDCT coefficients to maximize performance for both speech and music. This text contains an electronic attachment with the ANSI C source code, which is an integral part of this Recommendation.
Comment: Former title: Variable bit rate embedded coding of speech signals
Base text(s):
[ToR & schedule: TD 295R1/WP3 Annexes Q08.B & Q08.C]
Contact(s):
Jon Gibbs, Rapporteur
Milan Jelinek, Editor (C Code)
ITU-T A.5 reference(s):
Generate A.5 drat TD
-
[Submit new A.5 reference ] 
First registration in the WP: 2008-02-15 09:45:05
Last update: 2008-05-13 12:25:41