Topic
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.
Papers published on a yearly basis
Papers
More filters
TL;DR: The design of an underwater acoustic diver communication system controlled by a digital signal processor and the extracted parameters are transmitted through the water to a synchronized receiver by employing digital pulse position modulation.
Abstract: This paper describes the design of an underwater acoustic diver communication system controlled by a digital signal processor. The speech signal transmission rate is compressed by using linear predictive coding (LPC) and the extracted parameters are transmitted through the water to a synchronized receiver by employing digital pulse position modulation (DPPM). The pulse position in each time frame is estimated by an energy detection and decision algorithm which enables the received LPC parameters to be recovered and used to synthesize the speech signal.
45 citations
19 Apr 1994
TL;DR: A toll quality speech codec at 8 kbit/s with a 10 ms speech-frame currently under standardization by the CCITT is presented and initial subjective tests showed that the codec quality is equivalent to that of G.726 ADPCM in error-free conditions and it performs adequately under tandeming conditions.
Abstract: A toll quality speech codec at 8 kbit/s with a 10 ms speech-frame currently under standardization by the CCITT is presented. The encoding algorithm is based on algebraic code-excited linear prediction (ACELP). Efficient pitch and codebook search strategies, along with efficient quantization procedures, have been developed to achieve toll quality encoded speech with a complexity implementable on current fixed-point DSP chips. Initial subjective tests showed that the codec quality is equivalent to that of G.726 ADPCM at 32 kbit/s in error-free conditions and it outperforms G.726 under error conditions. The codec can support a frame erasure rate up to 3% with slight degradation and performs adequately under tandeming conditions. The algorithm has been implemented on a single fixed-point DSP for the CCITT qualification test. It requires about 24 MIPS. >
45 citations
TL;DR: It is shown in the paper how the coefficients of a modified model can be obtained and how the inverse and synthesis filters can be implemented.
Abstract: In conventional one-step forward linear prediction, an estimate for the current sample value is formed as a linear combination of previous sample values. In this paper, a generalized form of this scheme is studied. Here, the prediction is not based simply on the previous sample values but on the signal history as seen through an arbitrary filterbank. It is shown in the paper how the coefficients of a modified model can be obtained and how the inverse and synthesis filters can be implemented. Various properties of such systems are derived in this article. As an example, a novel linear predictive system using inherently logarithmic frequency representation is introduced.
45 citations
05 Jun 2000
TL;DR: A 1.2 kbps speech coder based on the mixed excitation linear prediction (MELP) analysis algorithm that achieves approximately the same quality as the proposed federal standard 2.4 kbps MELP coder.
Abstract: This paper presents a 1.2 kbps speech coder based on the mixed excitation linear prediction (MELP) analysis algorithm. In the proposed coder, the MELP parameters of three consecutive frames are grouped into a superframe and jointly quantized to obtain a high coding efficiency. The interframe redundancy is exploited with distinct quantization schemes for different unvoiced/voiced (U/V) frame combinations in the superframe. Novel techniques for improving performance make use of the superframe structure. These include pitch vector quantization using pitch differentials, joint quantization of pitch and U/V decisions and LSF quantization with a forward-backward interpolation method. Subjective test results indicate that the 1.2 kbps speech coder achieves approximately the same quality as the proposed federal standard 2.4 kbps MELP coder.
45 citations
TL;DR: In this paper, a speech signal is received into a bank of bandpass filters and the instantaneous amplitude modulation and frequency modulation of each harmonic in the speech waveform is determined, for example, by computing a weighted average of the frequency modulations of the harmonics.
Abstract: A method and apparatus for extracting information from human speech are disclosed. A speech signal is received into a bank of bandpass filters and the instantaneous amplitude modulation and frequency modulation of each harmonic in the speech waveform is determined. A logarithm of the instantaneous frequency of the speech fundamental frequency is determined, for example, by computing a weighted average of the frequency modulations of the harmonics. An output signal is formed having the logarithm of the frequency of the thus determined speech fundamental and the logarithms of the amplitude modulation for the ten lowest frequency speech harmonics and/or the speech envelope.
44 citations