Udar Mittal

A joint source-channel speech coder using hybrid digital-analog (HDA) modulation

A joint source-channel coding system for transmitting speech on a bandlimited additive white Gaussian noise (AWGN) channel is presented. The proposed method uses a hybrid of digital and analog modulation techniques. The digital part of the system consists of a Federal Standard 1016 code-excited linear predictive (FS 1016 CELP) speech coder followed by a rate-3/5 parallel concatenated (turbo) error correcting code. The analog part, which transmits the quantization error due to the FS 1016 CELP coder, consists of a linear encoder and decoder. The advantage of the proposed system is that it achieves excellent rate-distortion/capacity performance that is common in digital systems while maintaining a graceful degradation characteristic that is common in analog systems. Comparisons are made with three purely digital systems and an analog system-with all systems operating at the same overall rate. A formal listening test shows that, at high channel SNR, the improved performance of the proposed hybrid system (versus the purely digital systems) is noticeable to the average listener. Finally, an informal listening test indicates that at low channel SNR (where the error correcting code breaks down) the decoded speech of the hybrid system is still intelligible.

Duality theorems for joint source-channel coding

We consider joint source-channel coding for a memoryless Gaussian source and an additive white Gaussian noise (AWGN) channel. For a given code defined by an encoder-decoder pair (/spl alpha/, /spl beta/), its dual code is obtained by interchanging the encoder and decoder: (/spl beta/, /spl alpha/). It is shown that if a code (/spl alpha/, /spl beta/) is optimal at rate p channel uses per source sample and if it satisfies a certain uniform continuity condition, then its dual code (/spl beta/, /spl alpha/) is optimal for rate 1//spl rho/ channel uses per source sample. Further, it is demonstrated that there is a code which is optimal but its dual code is not optimal. Finally, using random coding, we show that there is an optimal code which has an optimal dual. The duality concept is also presented for the cases of (i) binary memoryless equiprobable source and binary-symmetric channel (BSC), and (ii) colored Gaussian source and additive colored Gaussian noise (ACGN) channel.

Signal/noise KLT based approach for enhancing speech degraded by colored noise

A signal/noise KLT based approach for enhancing speech degraded by colored noise is proposed. The noisy speech frames are classified into speech-dominated frames and noise-dominated frames. In the speech-dominated frames, the signal KLT matrix is used and in the noise-dominated frames, the noise KLT matrix is used. The approach does not require noise whitening and hence works well even with narrowband noise. A two-dimensional objective measure which captures both the speech distortion and the noise shaping characteristics of the algorithm is proposed. This measure indicates that the proposed method performs better noise shaping than a modified version of the signal subspace approach proposed by Ephraim and Van Trees and the standard spectral subtraction method. Informal listening tests show that the proposed algorithm does not suffer from the problem of residual musical noise and performs better noise masking than the signal subspace approach.

Robust joint source-channel coding theory

We consider joint source-channel coding for a memoryless Gaussian source and an additive white Gaussian noise (AWGN) channel. A code is said to be robust if its performance is arbitrarily close to the theoretical limit for a wide range of channel noise conditions. The existence of robust codes depends on the ratio of the channel bandwidth to the source bandwidth-a quantity we called /spl rho/. We first show that when /spl rho/=1, robust codes do indeed exist. Secondly, no robust code exists for /spl rho//spl ne/1.

A technique of multi-tap long term predictor (LTP) filter using sub-sample resolution delay [speech coding applications]

The method of a 1/sup st/ order long-term predictor (LTP) filter, using a sub-sample resolution delay, is extended to a multi-tap LTP filter, or, equivalently, the conventional integer-sample resolution multi-tap LTP filter is extended to use sub-sample resolution delay. Defining the delay with sub-sample resolution enables this novel multi-tap LTP filter to explicitly model delay values that have a fractional component. The filter coefficients, largely freed from implicitly modeling the effect of delays that have a fractional component, seek to maximize the prediction gain of the LTP filter by modeling the frequency dependent gain. This is in contrast to a conventional multitap LTP filter, which applies a single model to tackle the dual tasks of representing the non-integer valued delays and the frequency dependent gain. Experimental results are presented for narrowband and wideband speech. This technique is part of the 3GPP2 source-controlled variable-rate multimode wideband speech codec (VMR-WB) rate set 1 standard.

A duality theorem for joint source-channel coding

We consider joint source-channel coding for a memoryless Gaussian source and an additive white Gaussian noise (AWGN) channel. For a given code defined by an encoder-decoder pair (/spl alpha/,/spl beta/), its dual code is obtained by interchanging the encoder and decoder: (/spl beta/,/spl alpha/). It is shown that if a code (/spl alpha/,/spl beta/) is optimal at rate /spl rho/ channel uses per source sample and if it satisfies a certain uniform continuity condition, then its dual code (/spl beta/,/spl alpha/) is optimal for rate 1//spl rho/ channel uses per source sample. It is demonstrated that there is a code which is optimal but its dual code is not optimal. Finally, using random coding, we show that there is an optimal code which has an optimal dual.