Linear predictive coding

About: Linear predictive coding is a research topic. Over the lifetime, 6565 publications have been published within this topic receiving 142991 citations. The topic is also known as: Linear predictive coding, LPC.

Speech analysis and synthesis system

Abstract: Method and apparatus for speech analysis and synthesis adapted for analyzing and multiplexing speech signals from a plurality of voice grade telephone lines for further transmission through a single voice grade telephone line. A plurality of specialized digital signal processing techniques are employed to analyze in real time four speech channels in parallel and multiplex speech frame parameters of the channels into a single data output channel for transmission through a suitable media. The received data channel is demultiplexed and the speech frame parameters for the individual channels are utilized to synthesize, in parallel, the four speech signals. Certain of the digital processing techniques utilize the characteristics of speech signals to truncate conventional signal processing time while other processing techniques are substantially statistical analyses of speech to resolve ambiguities, particularly in making the voiced/unvoiced decision for a frame of analyzed speech data.

Core estimator and adaptive gains from signal to noise ratio in a hybrid speech enhancement system

Abstract: A speech enhancement system receives noisy speech and produces enhanced speech. The noisy speech is characterized by a spectral amplitude spanning a plurality of frequency bins. The speech enhancement system modifies the spectral amplitude of the noisy speech without affecting the phase of the noisy speech. The speech enhancement system includes a core estimator that applies to the noisy speech one of a first set of gains for each frequency bin. A noise adaptation module segments the noisy speech into noise-only and signal-containing frames, maintains a current estimate of the noise spectrum and an estimate of the probability of signal absence in each frequency bin. A signal-to-noise ratio estimator measures an a-posteriori signal-to-noise ratio and estimates an a-priori signal-to-noise ratio based on the noise estimate. Each one of the first set of gains is based on the a-priori signal-to-noise ratio, as well as the probability of signal absence in each bin and a level of aggression of the speech enhancement. A soft decision module computes a second set of gains that is based on the a-posteriori signal-to-noise ratio and the a-priori signal-to-noise ratio, and the probability of signal absence in each frequency bin.

Improving Lpc Spectral Envelope Extraction Of Voiced Speech By True-Envelope Estimation

Abstract: In this work we address the problem of all pole spectral envelope estimation for speech signals. The currently widely used all pole spectral envelope model suffers from well-known systematic errors and more severely from model order mismatch. We will propose a procedure to first establish a band limited interpolation of the observed spectrum using a recently rediscovered true envelope estimator and then using the band limited envelope to derive an all pole envelope model named TE-LPC. The band-limited envelope that is used to derive the all pole envelope model reduces the problem of the unknown all pole model order. For the experimental investigation we propose a new perceptually motivated residual spectral peak flatness measure. The experimental results demonstrate that the proposed method significantly increases the spectral flatness for the perceptually especially important low order harmonics of voiced utterances.

A robust low-delay CELP speech coder at 16 kbits/s

Abstract: A candidate algorithm for the new CCITT 16-kb/s speech coding standard is presented. This algorithm is based on backward-adaptive CELP (code-excited linear prediction) where the predictor and the excitation gain are updated by analyzing previously quantized signals. The only information transmitted is the excitation vector with a size as small as five samples so as to achieve a one-way coding delay of less than 2 ms. With a clear channel, this 16-kb/s coder slightly outperformed the CCITT standard 32 kb/s ADPCM (adaptive differential pulse code modulation) (G.721) in speech quality as measured by the mean opinion score (MOS). With noisy channels, the coder scored slightly higher than G.721 fora bit-error rate of 10/sup -2/ and significantly higher (by a margin of 0.5) for bit-error rate of 10/sup -3/. >