Multiresolution Source/Filter Model for Low Bitrate Coding of Spot Microphone Signals

Athanasios Mouchtaris,Kiki Karadimou,Panagiotis Tsakalides

doi:10.1155/2008/624321

Athanasios Mouchtaris, Kiki Karadimou + Show 1 more

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https://doi.org/10.1155/2008/624321

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Abstract

A multiresolution source/filter model for coding of audio source signals (spot recordings) is proposed. Spot recordings are a subset of the multimicrophone recordings of a music performance, before the mixing process is applied for producing the final multichannel audio mix. The technique enables low bitrate coding of spot signals with good audio quality (above 3.0 perceptual grade compared to the original). It is demonstrated that this particular model separates the various microphone recordings of a multimicrophone recording into a part that mainly characterizes a specific microphone signal and a part that is common to all signals of the same recording (and can thus be omitted during transmission). Our interest in low bitrate coding of spot recordings is related to applications such as remote mixing and real-time collaboration of musicians who are geographically distributed. Using the proposed approach, it is shown that it is possible to encode a multimicrophone audio recording using a single audio channel only, with additional information for each spot microphone signal in the order of 5 kbps, for good-quality resynthesis. This is verified by employing both objective and subjective measures of performance.

Highlights

Multichannel audio offers significant advantages regarding music reproduction when compared to two-channel stereo audio. (In the following, when we are referring to stereo audio, we always mean two-channel stereo.) The use of a large number of channels around the listener results in a more realistic acoustic space, adding more sound directions, and immersing the listener into the acoustic scene
We use microphone signals obtained from a US orchestra hall by placing 16 microphones at various locations throughout the hall. (Provided by Professor Kyriakakis of the University of Southern California.) Our objective is to indicate that the model and the coding method we propose result in a good-quality recording with low datarate requirements
This normalization was achieved by dividing each signal with its maximum absolute value, and in practice this procedure proved to be sufficient for all signals to sound as having the same sound level; the reader is referred to the authors’ website mentioned in Section 5.2 for listening to the audio waveforms that were used in the listening tests

Summary

Introduction

Multichannel audio offers significant advantages regarding music reproduction when compared to two-channel stereo audio. (In the following, when we are referring to stereo audio, we always mean two-channel stereo.) The use of a large number of channels around the listener results in a more realistic acoustic space, adding more sound directions, and immersing the listener into the acoustic scene. MPEG-1 audio coding [1, 2] (including the popular Layer III known as MP3 audio coding), MPEG-2 AAC (advanced audio coding) [3,4,5], and Dolby AC-3 [6, 7] are some well-known audio coding methods for stereo and multichannel audio content. These methods mainly exploit the masking property of the human auditory system for shaping the quantization noise so that it will be inaudible.

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Conclusion