ARTICLE Ganapathy_IEEETASLP_2010/IDIAP Autoregressive Models of Amplitude Modulations in Audio Compression Ganapathy, Sriram Motlicek, Petr Hermansky, Hynek https://publications.idiap.ch/index.php/publications/showcite/Ganapathy_Idiap-RR-33-2009 Related documents IEEE TRANSACTIONS ON AUDIO, SPEECH, AND LANGUAGE PROCESSING 2010 http://www.signalprocessingsociety.org/publications/periodicals/taslp/ URL We present a scalable medium bit-rate wide-band audio coding technique based on frequency domain linear prediction (FDLP). FDLP is an efficient method for representing the long-term amplitude modulations of speech/audio signals using autoregressive models. For the proposed audio codec, relatively long temporal segments (1000 ms) of the input audio signal are decomposed into a set of critically sampled sub-bands using a quadrature mirror filter (QMF) bank. The technique of FDLP is applied on each sub-band to model the sub-band temporal envelopes. The residual of the linear prediction, which represents the frequency modulations in the sub-band signal [1], are encoded and transmitted along with the envelope parameters. These steps are reversed at the decoder to reconstruct the signal. The proposed codec utilizes a simple signal independent non-adaptive compression mechanism for a wide class of speech and audio signals. The subjective and objective quality evaluations show that the reconstruction signal quality for the proposed FDLP codec compares well with the state-of-the-art audio codecs in the 32-64 kbps range. REPORT Ganapathy_Idiap-RR-33-2009/IDIAP Autoregressive Models of Amplitude Modulations in Audio Compression Ganapathy, Sriram Motlicek, Petr Hermansky, Hynek Idiap-RR-33-2009 2009 Idiap Rue Marconi 19, Martigny December 2009 We present a scalable medium bit-rate wide-band audio coding technique based on frequency domain linear prediction (FDLP). FDLP is an efficient method for representing the long-term amplitude modulations of speech/audio signals using autoregressive models. For the proposed audio codec, relatively long temporal segments (1000 ms) of the input audio signal are decomposed into a set of critically sampled sub-bands using a quadrature mirror filter (QMF) bank. The technique of FDLP is applied on each sub-band to model the sub-band temporal envelopes. The residual of the linear prediction, which represents the frequency modulations in the sub-band signal [1], are encoded and transmitted along with the envelope parameters. These steps are reversed at the decoder to reconstruct the signal. The proposed codec utilizes a simple signal independent non-adaptive compression mechanism for a wide class of speech and audio signals. The subjective and objective quality evaluations show that the reconstruction signal quality for the proposed FDLP codec compares well with the state-of-the-art audio codecs in the 32-64 kbps range.