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Abstract: A speech/audio codec based on FrequencyDomainLinearPrediction(FDLP) exploits auto-regressive modeling to approximate instantaneous energy in critical frequency sub-bands of relatively long input segments. The current version of the FDLP codec operating at 66 kbps has been shown to provide comparable subjective listening quality results to state-of-the-art codecs on similar bit-rates even without employing standard blocks such as entropy coding or simultaneous masking. This paper describes an experimental work to increase compression efficiency of the FDLP codec by employing entropy coding. Unlike conventional Huffman coding employed in current speech/audio coding systems, we describe an efficient way to exploit arithmetic coding to entropy compress quantized spectral magnitudes of the sub-band FDLP residuals. Such an approach provides 11\% ( 3 kbps) bit-rate reduction compared to the Huffman coding algorithm ( 1 kbps).
Abstract: FrequencyDomainLinearPrediction(FDLP) represents the technique for approximating temporal envelopes of a signal using autoregressive models. In this paper, we propose a wide-band audio coding system exploiting FDLP. Specifically, FDLP is applied on critically sampled sub-bands to model the Hilbert envelopes. The residual of the linearprediction forms the Hilbert carrier, which is transmitted along with the envelope parameters. This process is reversed at the decoder to reconstruct the signal. In the objective and subjective quality evaluations, the FDLP based audio codec at $66$ kbps provides competitive results compared to the state-of-art codecs at similar bit-rates.
Abstract: Audio codec based on FrequencyDomainLinearPrediction(FDLP) exploits auto-regressive modeling to approximate instantaneous energy in critical frequency sub-bands of relatively long input segments. Current version of the FDLP codec operating at 66 kbps has shown to provide comparable subjective listening quality results to the state-of-the-art codecs on similar bit-rates even without employing strategic blocks, such as entropy coding or simultaneous masking. This paper describes an experimental work to increase compression efficiency of the FDLP codec provided by employing entropy coding. Unlike traditionally used Huffman coding in current audio coding systems, we describe an efficient way to exploit Arithmetic coding to entropy compress quantized magnitude spectral components of the sub-band FDLP residuals. Such approach outperforms Huffman coding algorithm and provides more than 3 kbps bit-rate reduction.
Abstract: FrequencyDomainLinearPrediction(FDLP) represents a technique for auto-regressive modelling of Hilbert envelopes of a signal. In this paper, we propose a speech coding technique that uses FDLP in Quadrature Mirror Filter (QMF) sub-bands of short segments of the speech signal (25 ms). Line Spectral Frequency parameters related to au-toregressive models and the spectral components of the residual signals are transmitted. For simulating the effects of lossy transmission channels, bit-packets are dropped randomly. In the objective and subjective quality evaluations, the proposed FDLP speech codec is judged to be more resilient to bit-packet losses compared to the state-of-the-art Adaptive Multi-Rate Wide-Band (AMR-WB) codec at 12 kbps.
Abstract: FrequencyDomainLinearPrediction(FDLP) represents a technique for auto-regressive modelling of Hilbert envelopes of a signal. In this paper, we propose a speech coding technique that uses FDLP in Quadrature Mirror Filter (QMF) sub-bands of short segments of the speech signal (25 ms). Line Spectral Frequency parameters related to au-toregressive models and the spectral components of the residual signals are transmitted. For simulating the effects of lossy transmission channels, bit-packets are dropped randomly. In the objective and subjective quality evaluations, the proposed FDLP speech codec is judged to be more resilient to bit-packet losses compared to the state-of-the-art Adaptive Multi-Rate Wide-Band (AMR-WB) codec at 12 kbps.
Abstract: Automatic Speech Recognition (ASR) systems usually fail when they encounter speech from far-field microphone in reverberant environments. This is due to the application of short-term feature extraction techniques which do not compensate for the artifacts introduced by long room impulse responses. In this paper, we propose a front-end, based on FrequencyDomainLinearPrediction(FDLP), that tries to remove reverberation artifacts present in far-field speech. Long temporal segments of far-field speech are analyzed in narrow frequency sub-bands to extract FDLP envelopes and residual signals. Filtering the residual signals with gain normalized inverse FDLP filters result in a set of sub-band signals which are synthesized to reconstruct the signal back. ASR experiments on far-field speech data processed by the proposed front-end show significant improvements (relative reduction of $30 \%$ in word error rate) compared to other robust feature extraction techniques.
Abstract: Automatic speech recognition (ASR) systems, trained on speech signals from close-talking microphones, generally fail in recognizing far-field speech. In this paper, we present a Hilbert Envelope based feature extraction technique to alleviate the artifacts introduced by room reverberations. The proposed technique is based on modeling temporal envelopes of the speech signal in narrow sub-bands using FrequencyDomainLinearPrediction(FDLP). ASR experiments on far-field speech using the proposed FDLP features show significant performance improvements when compared to other robust feature extraction techniques (average relative improvement of $43 \%$ in word error rate).
Abstract: In this paper, we present a spectro-temporal feature extraction technique using sub-band Hilbert envelopes of relatively long segments of speech signal. Hilbert envelopes of the sub-bands are estimated using FrequencyDomainLinearPrediction(FDLP). Spectral features are derived by integrating the sub-band Hilbert envelopes in short-term frames and the temporal features are formed by converting the FDLP envelopes into modulation frequency components. These are then combined at the phoneme posterior level and are used as the input features for a phoneme recognition system. In order to improve the robustness of the proposed features to telephone speech, the sub-band temporal envelopes are gain normalized prior to feature extraction. Phoneme recognition experiments on telephone speech in the HTIMIT database show significant performance improvements for the proposed features when compared to other robust feature techniques (average relative reduction of $11\%$ in phoneme error rate).
Abstract: This paper describes employment of non-uniform QMF decomposition to increase the efficiency of a generic wide-band audio coding system based on FrequencyDomainLinearPrediction(FDLP). The base line FDLP codec, operating at high bit-rates ( 136 kbps), exploits a uniform QMF decomposition into 64 sub-bands followed by sub-band processing based on FDLP. Here, we propose a non-uniform QMF decomposition into 32 frequency sub-bands obtained by merging 64 uni- form QMF bands. The merging operation is performed in such a way that bandwidths of the resulting critically sampled sub-bands emulate the characteristics of the critical band filters in the human auditory system. Such frequency decomposition, when employed in the FDLP audio codec, results in a bit-rate reduction of 40\% over the base line. We also describe the complete audio codec, which provides high-fidelity audio compression at 66 kbps. In subjective listening tests, the FDLP codec outperforms MPEG-1 Layer 3 (MP3) and achieves similar qualities as MPEG-4 HE-AAC codec.
Abstract: We present a new feature extraction technique for phoneme recognition that uses short-term spectral envelope and modulation frequency features. These features are derived from sub-band temporal envelopes of speech estimated using FrequencyDomainLinearPrediction(FDLP). While spectral envelope features are obtained by the short-term integration of the sub-band envelopes, the modulation frequency components are derived from the long-term evolution of the sub-band envelopes. These features are combined at the phoneme posterior level and used as features for a hybrid HMM-ANN phoneme recognizer. For the phoneme recognition task on the TIMIT database, the proposed features show an improvement of 4.7\% over the other feature extraction techniques.
Abstract: Performance of a typical automatic speech recognition (ASR) system severely degrades when it encounters speech from reverberant environments. Part of the reason for this degradation is the feature extraction techniques that use analysis windows which are much shorter than typical room impulse responses. We present a feature extraction technique based on modeling temporal envelopes of the speech signal in narrow sub-bands using FrequencyDomainLinearPrediction(FDLP). FDLP provides an all-pole approximation of the Hilbert envelope of the signal obtained by linearprediction on cosine transform of the signal. ASR experiments on speech data degraded with a number of room impulse responses (with varying degrees of distortion) show significant performance improvements for the proposed FDLP features when compared to other robust feature extraction techniques (average relative reduction of $24 \%$ in word error rate). Similar improvements are also obtained for far-field data which contain natural reverberation in background noise. These results are achieved without any noticeable degradation in performance for clean speech.
Abstract: Audio coding based on FrequencyDomainLinearPrediction(FDLP) uses auto-regressive models to approximate Hilbert envelopes in frequency sub-bands. Although the basic technique achieves good coding efficiency, there is a need to improve the reconstructed signal quality for tonal signals with impulsive spectral content. For such signals, the quantization noise in the FDLP codec appears as frequency components not present in the input signal. In this paper, we propose a technique of Spectral Noise Shaping (SNS) for improving the quality of tonal signals by applying a Time DomainLinearPrediction (TDLP) filter prior to the FDLP processing. The inverse TDLP filter at the decoder shapes the quantization noise to reduce the artifacts. Application of the SNS technique to the FDLP codec improves the quality of the tonal signals without affecting the bit-rate. Performance evaluation is done with Perceptual Evaluation of Audio Quality (PEAQ) scores and with subjective listening tests.
Abstract: FrequencyDomainLinearPrediction(FDLP) provides an efficient way to represent temporal envelopes of a signal using auto-regressive models. For the input speech signal, we use FDLP to estimate temporal trajectories of sub-band energy by applying linearprediction on the cosine transform of sub-band signals. The sub-band FDLP envelopes are used to extract spectral and temporal features for speech recognition. The spectral features are derived by integrating the temporal envelopes in short-term frames and the temporal features are formed by converting these envelopes into modulation frequency components. These features are then combined in the phoneme posterior level and used as the input features for a hybrid HMM-ANN based phoneme recognizer. The proposed spectro-temporal features provide a phoneme recognition accuracy of $69.1 \%$ (an improvement of $4.8 \%$ over the Perceptual LinearPrediction (PLP) base-line) for the TIMIT database.
Abstract: Audio coding based on FrequencyDomainLinearPrediction(FDLP) uses auto-regressive model to approximate Hilbert envelopes in frequency sub-bands for relatively long temporal segments. Although the basic technique achieves good quality of the reconstructed signal, there is a need for improving the coding efficiency. In this paper, we present a novel method for the application of temporal masking to reduce the bit-rate in a FDLP based codec. Temporal masking refers to the hearing phenomenon, where the exposure to a sound reduces response to following sounds for a certain period of time (up to $200$ ms). In the proposed version of the codec, a first order forward masking model of the human ear is implemented and informal listening experiments using additive white noise are performed to obtain the exact noise masking thresholds. Subsequently, this masking model is employed in encoding the sub-band FDLP carrier signal. Application of the temporal masking in the FDLP codec results in a bit-rate reduction of about $10$\% without degrading the quality. Performance evaluation is done with Perceptual Evaluation of Audio Quality (PEAQ) scores and with subjective listening tests.
