Heiko Purnhagen

HILN-the MPEG-4 parametric audio coding tools

The MPEG-4 Audio Standard combines tools for efficient and flexible coding of audio. For very low bitrate applications, tools based on a parametric signal representation are utilised. The parametric speech coding tools (HVXC) are already available in Version 1 of MPEG-4. The main focus of this paper is on the parametric audio coding tools Harmonic and Individual Lines plus Noise (KILN) which are included in Version 2 of MPEG-4. As already indicated by their name, the HILN tools are based on the decomposition of the audio signal into components which are described by appropriate source models and represented by model parameters. This paper gives an overview of the HILN tools, presents the recent advances in signal modelling and parameter coding, and concludes with an evaluation of the subjective audio quality.

Advances in parametric audio coding

Parametric modelling provides an efficient representation of general audio signals and is utilised in very low bit rate audio coding. It is based on the decomposition of an audio signal into components which are described by appropriate source models and represented by model parameters. Perception models are utilised in signal decomposition and model parameter coding. This paper gives a brief tutorial overview of parametric audio coding and describes the parametric coder currently developed in the MPEG-4 audio standardisation. Recent advances as well as novel approaches in this field are presented.

Sinusoidal coding using loudness-based component selection

Sinusoidal modelling forms the base of parametric audio coding systems, like MPEG-4 HILN, where it is combined with noise and transient models. A parametric encoder decomposes the audio signal into components that are described by appropriate models and represented by model parameters. To achieve efficient coding at very low bitrates, selection of the perceptually most relevant signal components (e.g. sinusoids) is essential, as only a limited number of component parameters can be conveyed in the bitstream. Various strategies for sinusoidal component selection have been proposed in the literature. This paper introduces a new, loudness-based strategy and tries to compare the different strategies using objective and subjective criteria.

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.

An Overview of the Coding Standard MPEG-4 Audio Amendments 1 and 2: HE-AAC, SSC, and HE-AAC v2

In 2003 and 2004, the ISO/IEC MPEG standardization committee added two amendments to their MPEG-4 audio coding standard. These amendments concern parametric coding techniques and encompass Spectral Band Replication (SBR), Sinusoidal Coding (SSC), and Parametric Stereo (PS). In this paper, we will give an overview of the basic ideas behind these techniques and references to more detailed information. Furthermore, the results of listening tests as performed during the final stages of the MPEG-4 standardization process are presented in order to illustrate the performance of these techniques.

Parametric audio coding

For very low bit rate audio coding applications in mobile communications or on the Internet, parametric audio coding has evolved as a technique complementing the more traditional approaches. These are transform codecs originally designed for achieving CD-like quality on one hand, and specialized speech codecs on the other hand. Both of these techniques usually represent the audio signal waveform in a way such that the decoder output signal gives an approximation of the encoder input signal, while taking into account perceptual criteria. Compared to this approach, in parametric audio coding the models of the signal source and of human perception are extended. The source model is now based on the assumption that the audio signal is the sum of components, each of which can be approximated by a relatively simple signal model with a small number of parameters. The perception model is based on the assumption that the sound of the decoder output signal should be as similar as possible to that of the encoder input signal. Therefore, the approximation of waveforms is no longer necessary. This approach can lead to a very efficient representation. However, a suitable set of models for signal components, a good decomposition, and a good parameter estimation are all vital for achieving maximum audio quality. We give an overview on the current status of parametric audio coding developments and demonstrate advantages and challenges of this approach. Finally, we indicate possible directions of further improvements.

Decorrelation for audio object coding

Object-based representations of audio content are increasingly used in entertainment systems to deliver immersive and personalized experiences. Efficient storage and transmission of such content can be achieved by joint object coding algorithms that convey a reduced number of downmix signals together with parametric side information that enables object reconstruction in the decoder. This paper presents an approach to improve the performance of joint object coding by adding one or more decorrelators to the decoding process. Listening test results illustrate the performance as a function of the number of decorrelators. The method is adopted as part of the Dolby AC-4 system standardized by ETSI.

Delivering Scalable Audio Experiences using AC-4

AC-4 is a state-of-the-art audio codec standardized in ETSI (TS 103 190 and TS 103 190-2) and included in the DVB toolbox (TS 101 154 V2.2.1 and DVB BlueBook A157) and, at the time of writing, is a candidate standard for ATSC 3.0 as per A/342 part 2. AC-4 is an audio codec designed to address the current and future needs of video and audio entertainment services, including broadcast and Internet streaming. As such, it incorporates a number of features beyond the traditional audio coding algorithms, such as capabilities to support immersive and personalized audio, support for advanced loudness management, video-frame synchronous coding, dialog enhancement, etc. This paper will outline the thinking behind the design of the AC-4 codec, explain the different coding tools used, the systemic features included, and give an overview of performance and applications. It further outlines metadata aspects (immersive and personalized, essential for broadcast), metadata carriage, aspects of interchange of immersive programing, as well as immersive playback and rendering.