This paper presents an approach, referred to as frequency domain interpolation (FDI), for achieving high-quality speech at low bit-rates (4 kb/s and below) within reasonable complexity and delay. FDI methods, like the prototype waveform interpolation (PWI) methods, derive a prototype waveform (PW) at regular intervals of time. But, unlike PWI, there is no separation into a slowly evolving waveform (SEW) and a rapidly evolving waveform (REW) component. Instead, the PW is encoded after gain normalization in magnitude-phase form. The magnitude is modeled as a sum of mean and deviation values in multiple frequency bands and this model is quantized using switched backward adaptive VQ techniques. The phase information is represented as a composite vector of PW correlations in multiple frequency bands and an overall voicing measure. This information is quantized using a VQ at the encoder. At the decoder, a phase model is employed that uses the received phase (and magnitude) information to reproduce PWs with the correct periodicity and evolutionary characteristics. Speech is synthesized by interpolating the reconstructed PWs after gain adjustment and filtering it using the short-term predictor and a postfilter. The design of a 4-kb/s and a 2.4-kb/s FDI codec are presented in this paper and their performance is characterized in terms of delay, complexity, and subjective voice quality. The results confirm that FDI techniques have the potential for delivering high-quality speech at low bit-rates in a cost-effective manner.

We present an enhancement front-end for speech codecs, which consists of the integrated elements of noise reduction and echo cancellation. By including these elements, the front-end performs the task of mitigating the objectionable effects of the two major factors, i.e., noise and echo, which adversely affect the quality of most transmission systems, especially when low bit rate codecs are used. The use of this front-end is demonstrated with the 7.4 kbps IS-641 codec (enhanced full-rate standard for IS-136 systems). The integrated speech-processing unit has the advantage of utilizing the synergy among its components: the voice activity detector in the speech codec, the noise reduction, and the echo canceller. This synergy manifests itself both in the form of a reduction of the overall computational complexity by the use of a number of shared elements among the unit's various components, as well as an improved performance resulting from these components working together. The system displays high performance in both clean and noisy environments and it works well with low bit rate codecs.

In this paper, we present a combined speech quality enhancement solution for IS-136 systems. Since echo and background noise are the two major factors that adversely affect the speech quality in most transmission systems, our solution consists of echo cancellation and noise reduction elements. These are used in conjunction with the IS-641 (enhanced fullrate standard for IS-136 systems) coder to form an integrated speech-processing unit. The echo canceller uses the normalized least mean square (NLMS) method. Because of the existence of high levels of background noise, variable step-size techniques are employed. The noise reduction consists of a single microphone method and it uses a spectral amplitude enhancement gain function with minimal spectral distortion. The noise reduction is utilized in the pre-compression configuration, and it comes after the echo canceller on the send path reducing the residual echo as well as noise.

Robust vector quantization of LSF parameters at a low bit rate is essential for voice coders operating below 5 Kbps. A novel aspect of the proposed technique is the use of decorrelated residual LSF vectors from speech mode based backward prediction along with a multi-stage VQ design. Rates as low as 12 bits per 20 ms speech frame for the stationary voiced speech mode and 22 bits/frame for unvoiced and non-stationary voiced frames are shown to result in efficient quantization. In our classification scheme, spectrally stationary voiced frames constitute around 30% of active speech frames resulting in a minimum average bit rate of 19 bits/frame. The objective VQ performance is compared with cellular standard coders such as the IS-641 and IS-127. The proposed VQ has been integrated into a speech mode based 4.8 Kbps coder resulting in subjective performance close to that of the 7.4 Kbps IS-641 coder.

The recently adopted ITU-T G.729 8 kbps codec standard's performance is evaluated for digital cellular channels that are characterized by Rayleigh fading. Two forward error correction schemes (FEC) are studied. They are the convolutional code based FEC and the Nordstrom Robinson based FEC. The effects of interleaving depth are investigated for both FEC schemes. Both flat fading as well as co-channel interference limited models are used in our study.

The design of a speech and channel codec for the North American TDMA digital cellular half rate channel which meets the eligibility requirements of the Telecommunications Industry Association (TIA) contest is described. Its design objectives are largely shaped by the selection criteria. The codec has been implemented in real time on two Tiger C40 boards. Informal listening tests using this real-time hardware seem to indicate that this candidate is in close proximity to the full rate standard.>

The design and implementation of a real-time CELP coder for mobile communication applications are discussed. To realize a single-chip implementation, several tradeoffs were made without compromising speech quality. In addition, techniques that make the coder more robust under a variety of channel conditions are discussed. The real-time coder can be operated at different bit rates (8, 6.8, 4.6 kb/s) by simply changing the frame update rates. The speech quality was evaluated through a formal listening test, and it was found that this coder compares favorably with other (standardized) coders operating at similar or higher rates.>

A general framework is developed for uniform DFT, parallel quadrature mirror filter banks, which enables us to derive a number of conclusions in an elegant manner. Closed form expressions for the synthesis filters are obtained both for FIR and IIR case, and a number of properties of importance are placed in evidence.