The recently standardized 3GPP codec for Enhanced Voice Services (EVS) offers new features and improvements for low-delay real-time communication systems. Based on a novel, switched low-delay speech/audio codec, the EVS codec contains various tools for better compression efficiency and higher quality for clean/noisy speech, mixed content and music, including support for wideband, super-wideband and full-band content. The EVS codec operates in a broad range of bitrates, is highly robust against packet loss and provides an AMR-WB interoperable mode for compatibility with existing systems. This paper gives an overview of the underlying architecture as well as the novel technologies in the EVS codec and presents listening test results showing the performance of the new codec in terms of compression and speech/audio quality.

Overview of the EVS codec architecture

EVS, the newly standardized 3GPP Codec for Enhanced Voice Services (EVS) was developed for mobile services such as VoLTE, where error resilience is highly essential. The presented paper outlines all aspects of the advances brought during the EVS development on packet loss concealment, by presenting a high level description of all technical features present in the final standardized codec. Coupled with jitter buffer management, the EVS codec provides robustness against late or lost packets. The advantages of the new EVS codec over reference codecs are further discussed based on listening test results.

Packet-loss concealment technology advances in EVS

A highly error resilient mode of the newly standardized 3GPP EVS speech codec is described. Compared to the AMR-WB codec and other conversational codecs, the EVS channel aware mode offers significantly improved error resilience in voice communication over packet-switched networks such as Voice-over-IP (VoIP) and Voice-over-LTE (VoLTE). The error resilience is achieved using a form of in-band forward error correction. Source-controlled coding techniques are used to identify candidate speech frames for bitrate reduction, leaving spare bits for transmission of partial copies of prior frames such that a constant bit rate is maintained. The self-contained partial copies are used to improve the error robustness in case the original primary frame is lost or discarded due to late arrival. Subjective evaluation results from ITU-T P.800 Mean Opinion Score (MOS) tests are provided, showing improved quality under channel impairments as well as negligible impact to clean channel performance.

Improved error resilience for volte and VoIP with 3GPP EVS channel aware coding

Audio signal processing is the most challenging field in the current era for an analysis of an audio signal. Audio signal classification (ASC) comprises of generating appropriate features from a sound and utilizing these features to distinguish the class the sound is most likely to fit. Based on the application’s classification domain, the characteristics extraction and classification/clustering algorithms used may be quite diverse. The paper provides the survey of the state-of art for understanding ASC’s general research scope, including different types of audio; representation of audio like acoustic, spectrogram; audio feature extraction techniques like physical, perceptual, static, dynamic; audio pattern matching approaches like pattern matching, acoustic phonetic, artificial intelligence; classification, and clustering techniques. The aim of this state-of-art paper is to produce a summary and guidelines for using the broadly used methods, to identify the challenges as well as future research directions of acoustic signal processing.

Pattern analysis based acoustic signal processing: a survey of the state-of-art

This paper describes the time-domain bandwidth extension (TBE) framework employed to code wideband and super-wideband speech in the newly standardized 3GPP EVS codec. The TBE algorithm uses a nonlinear harmonic modeling technique that incorporates principles of time-domain envelope-modulated noise mixing. At 13.2 kbps, the super-wideband coding of speech uses as low as 1.55 kbps for encoding the spectral content from 6.4–14.4 kHz. Subjective evaluation results from ITU-T P.800 Mean Opinion Score (MOS) tests are provided, showing significantly improved quality compared to the other standardized SWB codecs under both clean speech and speech with background noise.

Super-wideband bandwidth extension for speech in the 3GPP EVS codec

An audio decoder (100; 300) for providing a decoded audio information (112; 312) on the basis of an encoded audio information (110; 310) comprises an error concealment (130; 380; 500) configured to provide an error concealment audio information (132; 382; 512) for concealing a loss of an audio frame following an audio frame encoded in a frequency domain representation (322) using a time domain excitation signal (532).

Audio decoder and method for providing a decoded audio information using an error concealment based on a time domain excitation signal

This paper describes new time domain techniques for concealing packet loss in the new 3GPP Enhanced Voice Services codec. Enhancements to the existing ACELP concealment methods include guided, improved pitch prediction, increased flexibility and accuracy of pulse resynchronization. Furthermore, the new method of separate linear predictive (LP) filter synthesis aims for sound quality improvement in case of multiple packet loss, especially for noisy signals. Another enhancement consists of a guided LP concealment approach to limit the risk of creating artifacts during recovery. These enhancements are also used in the presented advanced TCX concealment method. Subjective listening tests show that quality is significantly increased with these methods.

Enhanced time domain packet loss concealment in switched speech/audio codec

An encoding concept which is linear prediction based and uses spectral domain noise shaping is rendered less complex at a comparable coding efficiency in terms of, for example, rate/distortion ratio, by using the spectral decomposition of the audio input signal into a spectrogram comprising a sequence of spectra for both linear prediction coefficient computation as well as spectral domain shaping based on the linear prediction coefficients. The coding efficiency may remain even if such a lapped transform is used for the spectral decomposition which causes aliasing and necessitates time aliasing cancellation such as critically sampled lapped transforms such as an MDCT.

Linear prediction based coding scheme using spectral domain noise shaping

A multisignal encoder for encoding at least three audio signals, comprises: a signal preprocessor (100) for individually preprocessing each audio signal to obtain at least three preprocessed audio signals, wherein the preprocessing is performed so that a preprocessed audio signal is whitened with respect to the signal before preprocessing; an adaptive joint signal processor (200) for performing a processing of the at least three preprocessed audio signals to obtain at least three jointly processed signals or at least two jointly processed signals and an unprocessed signal; a signal encoder (300) for encoding each signal to obtain one or more encoded signals; and an output interface (400) for transmitting or storing an encoded multisignal audio signal comprising the one or more encoded signals, side information relating to the preprocessing and side information relating to the processing.

Multisignal audio coding using signal whitening as preprocessing

FIELD: audio signal processing means.SUBSTANCE: method comprises filtering temporal noise generation (TNS) of an information signal which includes a plurality of frames. For each frame, selecting (S34) between filtering by first filter (14a) and filtering by second filter (15a), impulse response of which has lower energy, based on frame metric, wherein second filter (15a) is not an identity filter. Frame is filtered using filtration selected between filtration by first filter (14a) and filtration by second filter (15a). First filter (14a) is modified to produce second filter (15a) with reduced pulse response energy.EFFECT: technical result is higher efficiency of audio encoding.22 cl, 14 dwg

Temporal formation of noise

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