This article is an overview of the standardization, architecture, and performance of the new ITU-T Recommendation G.718. G.718 is an embedded variable bit rate codec providing a scalable solution for compression of 8 and 16 kHz sampled speech and audio signals at rates between 8 kb/s and 32 kb/s. It comprises five layers where higher-layer bitstreams can be discarded without affecting the lower layersiquest decoding. The codec also has an optional core layer interoperable with ITU-T G.722.2 (3GPP AMR-WB) at 12.65 kb/s. G.718 was designed to provide high speech quality at low bit rates and to be robust to significant rates of frame erasures or packet losses. It is also targeting good quality for generic audio at higher rates.

G.718: A new embedded speech and audio coding standard with high resilience to error-prone transmission channels

Recent trends in speech and audio codec standardization include scalability and extending the signal bandwidth beyond wideband (WB) to superwideband (SWB). In this paper we introduce a SWB extension for the ITU-T G.718 WB codec. In the SWB extension the high frequency content is generated utilizing the quantized MDCT domain coefficients of the WB core, which enables low additional delay. The proposed implementation is scalable with 4 kbps layers. In the first layer two different coding modes are used depending on the input signal type. The proposed SWB extension is evaluated with listening tests and complexity analysis.

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Scalable superwideband extension for wideband coding

A method for encoding audio frames by producing a first frame of coded audio samples by coding a first audio frame in a sequence of frames, producing at least a portion of a second frame of coded audio samples by coding at least a portion of a second audio frame in the sequence of frames, and producing parameters for generating audio gap filler samples, wherein the parameters are representative of either a weighted segment of the first frame of coded audio samples or a weighted segment of the portion of the second frame of coded audio samples.

Encoder for audio signal including generic audio and speech frames

The invention utilizes low complexity estimates of complex functions to perform combinatorial coding of signal vectors. The invention disregards the accuracy of such functions as long as certain sufficient properties are maintained. The invention in turn may reduce computational complexity of certain coding and decoding operations by two orders of magnitude or more for a given signal vector input.

Apparatus and method for low complexity combinatorial coding of signals

During operation an input signal to be coded is received and coded to produce a coded audio signal. The coded audio signal is then scaled with a plurality of gain values to produce a plurality of scaled coded audio signals, each having an associated gain value and a plurality of error values are determined existing between the input signal and each of the plurality of scaled coded audio signals. A gain value is then chosen that is associated with a scaled coded audio signal resulting in a low error value existing between the input signal and the scaled coded audio signal. Finally, the low error value is transmitted along with the gain value as part of an enhancement layer to the coded audio signal.

Method and apparatus for generating an enhancement layer within an audio coding system

This paper presents ITU-T Embedded Variable Bit-Rate (EV-VBR) codec being standardized by Question 9 of Study Group 16 (Q9/16) as recommendation G.718. The codec provides a scalable solution for compression of 16 kHz sampled speech and audio signals at rates between 8 kbit/s and 32 kbit/s, robust to significant rates of frame erasures or packet losses. It comprises 5 layers where higher layer bitstreams can be discarded without affecting the lower layer decoding. The core layer takes advantage of signal-classification based CELP encoding. The second layer reduces the coding error from the first layer by means of additional pitch contribution and another algebraic codebook. The higher layers encode the weighted error signal from lower layers using MDCT transform coding. Sev-eral technologies are used to encode the MDCT coefficients for best performance both for speech and music. The codec performance is demonstrated with selected results from ITU-T Characterization test.

ITU-T EV-VBR: A robust 8-32 kbit/s scalable coder for error prone telecommunications channels

ITU-T has selected the candidate submitted by Ericsson, Nokia, Motorola, VoiceAge, and Texas Instruments as the baseline for the G.EV-VBR coding standard. G.EV-VBR is an embedded scalable speech codec that uses state-of-the-art technology to provide the most efficient encoded speech available for various real-time applications. EV-VBR encodes both narrowband (NB) and wideband (WB) speech signals starting at 8 kbps. Near perfect wideband representation is achieved at 32 kbps for all signal types. The bit stream is divided into five robust layers, providing sufficient granularity, in particular for VoIP applications. In addition, an extension to the codec will provide super- wideband and stereo capability by adding layers to the codec. Extensive listening tests were conducted during the ITU-T selection phase to support selection of the best- performing candidate. The selected EV-VBR candidate passed 69 of 70 required and 25 of 28 objective terms of reference.

A. Erdem Ertan

Papers

ITU-T EV-VBR: A robust 8-32 kbit/s scalable coder for error prone telecommunications channels

Quality evaluation of the G.EV-VBR speech codec