Johannes Hilpert

Unified speech and audio coding scheme for high quality at low bitrates

Traditionally, speech coding and audio coding were separate worlds. Based on different technical approaches and different assumptions about the source signal, neither of the two coding schemes could efficiently represent both speech and music at low bitrates. This paper presents a unified speech and audio codec, which efficiently combines techniques from both worlds. This results in a codec that exhibits consistently high quality for speech, music and mixed audio content. The paper gives an overview of the codec architecture and presents results of formal listening tests comparing this new codec with HE-AAC(v2) and AMR-WB+. This new codec forms the basis of the reference model in the ongoing MPEG standardization activity for Unified Speech and Audio Coding.

The MPEG Surround Audio Coding Standard [Standards in a Nutshell]

This article provides a compact overview of the history, technology, and performance of MPEG Surround. The technology of MPEG Surround is based on the spatial audio coding (SAC) principle: In the encoder, a mono- or stereophonic down- mix is generated from the multichannel input signal, and additional parametric side information is extracted to guide the subsequent up-mix procedure in the decoder.With Moving Pictures Expert Group (MPEG) Surround, a data-rate efficient coding scheme for high-quality multichannel sound using novel parametric coding techniques has been standardized.

Efficient transform coding of two-channel audio signals by means of complex-valued stereo prediction

Traditional MDCT-based perceptual audio coding schemes employ mid/side and intensity stereo techniques to allow efficient joint coding of the two channels of a stereophonic signal. These techniques, however, provide only little coding gain for critical stereo signals characterized by spectral components with a distinct level or phase difference between the channels. To overcome this deficiency, we propose an extension to the mid/side coding paradigm that utilizes complex-valued inter-channel linear prediction in the MDCT spectral domain. The required imaginary spectrum (MDST) is calculated in a computationally efficient manner without additional algorithmic delay. A formal listening test conducted in the course of the ISO/MPEG standardization of the unified speech and audio codec USAC illustrates that the proposed stereo prediction approach provides significant improvements in coding efficiency and shows that at 96 kb/s, excellent quality can be obtained even for critical signals.

MPEG-4 AAC-ELD v2 – The New State of the Art in High Quality Communication Audio Coding

Recently MPEG finished the standardization of a Low Delay MPEG Surround tool that is tailored for enhancing the widely adopted AAC-ELD low delay codec for high-quality audio communication into AAC-ELD v2. In combination with the Low Delay MPEG Surround tool, the coding efficiency for stereo content outperforms competing low delay audio codecs. This paper describes the technical challenges and solutions for designing a low delay codec that delivers a performance which is comparable to that of existing state of the art compression schemes. It provides a comparison to competing proprietary and ITU-T codecs, as well as a guideline for how to select the best possible points of operation. Applications facilitated by AAC-ELD v2 in the area of broadcasting and mobile video conferencing are discussed.

Development of the MPEG-H TV Audio System for ATSC 3.0

A new TV audio system based on the MPEG-H 3D audio standard has been designed, tested, and implemented for ATSC 3.0 broadcasting. The system offers immersive sound to increase the realism and immersion of programming, and offers audio objects that enable interactivity or personalization by viewers. Immersive sound may be broadcast using loudspeaker channel-based signals or scene-based components in combination with static or dynamic audio objects. Interactivity can be enabled through broadcaster-authored preset mixes or through user control of object gains and positions. Improved loudness and dynamic range control allows tailoring the sound for best reproduction on a variety of consumer devices and listening environments. The system includes features to allow operation in HD-SDI broadcast plants, storage, and editing of complex audio programs on existing video editor software or digital audio workstations, frame-accurate switching of programs, and new technologies to adapt current mixing consoles for live broadcast production of immersive and interactive sound. Field tests at live broadcast events were conducted during system design and a live demonstration test bed was constructed to prove the viability of the system design. The system also includes receiver-side components to enable interactivity, binaural rendering for headphone, or tablet computer listening, a “3D soundbar” for immersive playback without overhead speakers, and transport over HDMI 1.4 connections in consumer equipment. The system has been selected as a proposed standard of ATSC 3.0 and is the sole audio system of the UHD ATSC 3.0 broadcasting service currently being deployed in South Korea.

MPEG Unified Speech and Audio Coding – Bridging the Gap

Speech and audio coding schemes originate from different worlds. Speech coding schemes typically assume a source model i.e. the human vocal tract. General audio coding schemes primarily rely on a sinkmodel i.e. the human auditory system. While speech coding schemes work well for the signal class they were designed for at very low rates, they are known to fail for general audio signals even at higher rates. In contrast, general audio coders work well for any content at higher rates, but typically have limited performance especially for speech signals at very low rates. Recently the ISO/MPEG group started a standardization activity to develop a new Unified Speech and Audio Coding scheme. A state of the art AAC based general audio coder, featuring transform coding, parametric bandwidth extension and parametric stereo coding,was extended by source model coding tools. All codec modules were further improved and revised for enhanced performance in particular at very low bitrates. The new unified coding scheme outperforms dedicated speech and general audio coding schemes and bridges the gap between both worlds. This paper describes the new codec in detail and shows how the goal of consistent high quality for all signal types is reached.