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A fixed rate speech coder based on the filter bank and the non-uniform sampling technique is proposed. The non-uniform sampling is achieved by the detection of inflection points (IPs). A speech block is band passed by the filter bank, and the subband signals are processed by the IP detector, and the detected IP patterns are compared with entries of the IP database. For each subband signal, the address of the closest member of the database and the energy of the IP pattern are transmitted through channel. In the receiver, the decoder recovers the subband signals using the received addresses and the energy information, and reconstructs the speech via the filter bank summation. As results, the coder shows fixed data rate contrary to the existing speech coders based on the non-uniform sampling. Through computer simulation, the usefulness of the proposed technique is confirmed. The signal-to-noise ratio (SNR) performance of the proposed method is comparable to that of the uniform sampled pulse code modulation (PCM) below 20 kbps data rate.

A fixed rate speech coder having improved database search performance is proposed. In the speech coder, the 10 ms block of speech is band-passed using the filter bank, and the inflection points (IP) for each band are detected and analyzed to get the short-time statistics like the energy and IP rate. The obtained IP statistics are used to decide if the speech block is voiced or unvoiced. Then the encoder searches for the closest IP pattern either at the voiced part or at the unvoiced part of the IP database. For further reduction of the search time, the entries of the database parts corresponding to the neighboring subbands are checked for the obtained IP pattern. As results, about 94.5% decrease in search time is achieved compared to the exhaustive search. With 10 subbands and 240 entries for each band, the reconstructed signal shows about 5.2 dB SNR at 32 kbps data rate.

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A low bit rate speech coder based on the non-uniform sampling technique is proposed. The non-uniform sampling technique is based on the detection of inflection points (IP). A speech block is processed by the IP detector, and the detected IP pattern is compared with entries of the IP database. The address of the closest member of the database is transmitted with the energy of the speech block. In the receiver, the decoder reconstructs the speech block using the received address and the energy information of the block. As results, the coder shows fixed data rate contrary to the existing speech coders based on the non-uniform sampling. Through computer simulation, the usefulness of the proposed technique is shown. The SNR performance of the proposed method is approximately 5.27 dB with the data rate of 1.5 kbps.

In order to reduce the data amount, the nonuniform sampling (NUS) method detects samples of a signal, such as local maxima and minima. To overcome the sparseness problem of the NUS method, an inflection point detection (IPD) method is proposed to sample a signal nonuniformly. The IPD samples a signal not only at the local maxima and minima, but also at the inflection points where the slope of the signal changes. To show its usefulness, the IPD is applied to speech coding. The encoder transmits the time instants and sample amplitude values of the inflection points. At the receiver, the decoder estimates the sample amplitude values at the noninflection points by interpolating the received information. Simulation results show that the IPD method produces 7% mean square error improvement over the NUS method. With a small threshold to detect inflection points, the proposed coding method shows 0.38-8.72 dB signal-to-noise ratio (SNR) and 0.5-1.3 mean opinion score improvement, compared to the continuously variable slope delta modulation algorithm (CVSDM). The IPD method produces up to 8.5 dB improvement in SNR over the CVSDM at bit error rates (BER) below 5 × 10(-5), while the IPD method becomes worse than the CVSDM at BER above 5 × 10(-5).

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