LPC Vocoder Research Articles

Matrix Quantization and LPC Vocoder Based Linear Predictive for Low-Resource Speech Recognition system

Over the last ten years, there has been significant progress in the use of low-rate speech coders in voice applications for computers, military communications, and civil communications. This advancement has been made possible by the development of new speech coders that can generate high-quality speech at low data rates. The majority of existing coders include spectral representation of speech, speech waveform matching, and ”optimization” of the coder’s performance for human hearing. The goal of this paper is to provide a thorough evaluation of voice coding methods for educational purposes, with a particular emphasis on the algorithms used in low-rate cellular communication standards. The algorithm we developed using a voice-excited LPC vocoder produces clear, low-distortion results. Ordinary LPCs, on the other hand, fall short of vocoders because they can handle signals other than speech, such as music. To improve quality, additional bandwidth is used to reduce the bit rate. To improve the quality, we tried two approaches. The first was to increase the number of bits required to quantize the DCT coefficients. This coefficient would outperform the inverse DCT in closer error rearrangements. The second possibility is to increase the total number of quantized coefficients. As a result, error array rearrangements would be more accurate. The goal is to identify the point at which a method improvement outperforms the previous, better result. Other coding methods become more complex, but this vocoder suffices.

Effective Spectral and Excitation Modeling Techniques for LSTM-RNN-Based Speech Synthesis Systems

In this paper, we report research results on modeling the parameters of an improved time-frequency trajectory excitation (ITFTE) and spectral envelopes of an LPC vocoder with a long short-term memory (LSTM)-based recurrent neural network (RNN) for high-quality text-to-speech (TTS) systems. The ITFTE vocoder has been shown to significantly improve the perceptual quality of statistical parameter-based TTS systems in our prior works. However, a simple feed-forward deep neural network (DNN) with a finite window length is inadequate to capture the time evolution of the ITFTE parameters. We propose to use the LSTM to exploit the time-varying nature of both trajectories of the excitation and filter parameters, where the LSTM is implemented to use the linguistic text input and to predict both ITFTE and LPC parameters holistically. In the case of LPC parameters, we further enhance the generated spectrum by applying LP bandwidth expansion and line spectral frequency-sharpening filters. These filters are not only beneficial for reducing unstable synthesis filter conditions but also advantageous toward minimizing the muffling problem in the generated spectrum. Experimental results have shown that the proposed LSTM-RNN system with the ITFTE vocoder significantly outperforms both similarly configured band aperiodicity-based systems and our best prior DNN-trainecounterpart, both objectively and subjectively.

Experimental Analysis of Voice Ooze Algorithm using Voice-excited LPC Vocoder

This research paper examines the recent development in speech coding. Speech coding is a basic element of digital communications and constantly attracting attention due to the increase of power in telecommunication services and capabilities. This paper focuses on the design and implementation of a Voice-excited Linear Predictive coder at low bit-rate. And at last analyze result by using different voice examples. This research paper is divided into six divisions. General Terms Voice-excited LPC Vocoder et. al.

Adaptive Feedback Cancellation With Band-Limited LPC Vocoder in Digital Hearing Aids

Feedback oscillation is one of the major issues with hearing aids. An effective way of feedback suppression is adaptive feedback cancellation, which uses an adaptive filter to estimate the feedback path. However, when the external input signal is correlated with the receiver input signal, the estimate of the feedback path is biased. This so-called “bias problem” results in a large modeling error and a cancellation of the desired signal. This paper proposes a band-limited linear predictive coding based approach to reduce the bias. The idea is to replace the hearing-aid output with a synthesized signal, which sounds perceptually the same as or similar to the original signal but is statistically uncorrelated with the external input signal at high frequencies where feedback oscillation usually occurs. Simulation results show that the proposed algorithm can effectively reduce the bias and the misalignment between the real and the estimated feedback path. When combined with filtered-X adaptation in the feedback canceller, this approach reduces the misalignment even further.

Early linear predictive speech coding at Bell Telephone Laboratories and Electrical Communication Laboratory

At the 6th ICA in Tokyo (1968), two papers on the linear predictive coding of speech were presented by Manfred Schroeder and by me in the same session. The first paper was on the adaptive linear predictive waveform coding for medium band coding (time domain modeling) and the latter was on the LPC vocoder based on the autoregressive spectral estimation for narrow‐band coding (frequency domain modeling). Although their approaches were very different in both concept and scope, there were dramatic common points: the covariance and autocorrelation methods were used to solve normal equation to determine the LPC parameters and also the pitch period was extracted to reproduce speech periodicity in both systems. Subsequently the multipulse (1982), stochastic (1984), and code (1985) excitation LP were proposed at AT&T BTL, whereas the transformation of LPC parameters to PARCOR (1969) and LSP (1975) and their efficient quantization were intensively studied at ECL of NTT. These efforts were eventually combined for practical implementations in the secure, cellular, and IP telephone in the 1980s. The brilliant ideas by Manfred Schroeder and his associates at BTL, in particular Bishnu Atal, have always been a helpful lighthouse to navigate the world‐wide speech coding research in the right direction since 1980s.

Fifty years of progress in speech analysis (LPC)

Although the history of statistical linear prediction is very long, it was first applied to speech analysis in 1966 at NTT to estimate the all-pole speech spectrum envelope in order to implement ASR and vocoder. All-pole spectral parameters are investigated in detail to find a better representation with respect to quantization and interpolation characteristics, partly at NTT and Bell Labs, leading to PARCOR and LSP. These parameters were applied to narrow band speech coders or LPC vocoders and speech synthesizer chips in the late 1970s and early 1980s, but the speech quality was insufficient for digital mobile telephone application. The problem was later solved by using hybrid CELP and MLP coding mainly at BTL. LPC analysis was also applied to acoustic front-end for ASR. Again it was found that LPC suffers with additive noise and linear/nonlinear distortions. Whereas computational efficiency of LPC was used to be the most prominent advantage, today we have gained thousands of times the processing power at a ten thousandth of the cost in 40 years. It is hoped a novel series of speech analysis methods, whose competence is comparable to human auditory system, should be developed at any expense of computational complexity.

An LPC vocoder based on phase-equalized pitch waveform

This paper presents a speech coder operating at a very low bit-rate using a model called “phase-equalized pitch waveform”. The basic idea of the coder is to employ pitchwise extraction of the linear predictive residual signal, a pitch waveform, in voiced speech. The residual signal is processed with a phase-equalization filter to increase the efficiency of both the pitch waveform quantization and interpolation. Listening tests showed that efficient and high-quality coding is achieved at 2.0 kbits/s. The quality of the coder is equal to that of the DoD FS1016 standard CELP at 4.8 or 2.4 kbits/s MELP.

A new 1.2 kbit/s speech coding method based on a sinusoidal harmonic vocoder

A very low bit rate speech coder at 1.2 kbit/s is proposed in which a speech signal is synthesized by using sinusoids whose frequencies are multiples of the fundamental frequency, and whose amplitudes are adaptively modulated in accordance with auditory perceptual characteristics, in order to improve subjective quality. The auditory perceptual characteristics are simulated by using Gammatone filters. The phases of the sinusoids are also controlled with reference to the subjective quality of the synthesized speech. The quality of the synthesized speech was improved by 0.45 in the Mean Opinion Score compared with that of a simple LPC vocoder operating at the same rate, and was comparable to that of a 2.4 kbit/s MELP coder. © 2000 Scripta Technica, Syst Comp Jpn, 31(14): 64–73, 2000

Model of the human speech production based on vocoder techniques

A tutorial is presented with which the student first gets basic knowledge about the human speech production. It is explained that speech is generated by an air flow coming from the lungs which is, in the case of voiced sounds, periodically interrupted by the vocal cords and then modulated with the resonance rooms of the mouth and nasal tract. For voiceless sounds (like ‘‘f,’’ ‘‘sch’’) the vocal cords are open and the air flow excites the mouth and nasal tract with a noise-like pressure signal. It is therefore adequate to distinguish two components in the speech production: the excitation and the articulation component. In our tutorial, the two components are realized (simulated) by a technical system, known as linear predictive vocoder (LPC vocoder). Its principle is explained by a short text and some graphics and the analogy to the human speech organs is clearly shown. The main emphasis is put on the control parameters of the system—in the main the pitch frequency and the vocal tract parameters (prediction coefficients)—and the effect when.

Variable Frame Rate Speech Coding Using Optimal Interpolation

A VFR LPC vocoder using optimal interpolation is presented in this paper. In the encoder, some representative frames of an utterance are selected for transmission. In the decoder, LPC parameters of all untransmitted frames are restored by optimal interpolation. Simulation results show that this coding scheme outperforms the conventional VFR vocoder using linear interpolation. By incorporating contour quantization of gain and pitch information, a low variable-rate LPC vocoder is realized. An informal listening test shows that very high intelligible reconstructed speech was obtained at an average data rate of 300 bps

Use of a single chip fixed-point DSP for multiple speech channel vocoders

In this paper, the use of an advanced digital signal processor (DSP) makes possible the parallel real-time implementation of more than one speech coder on a single-chip DSP. Thus, many speech channels can share a single digital communications channel at low cost. Besides the hardware requirements, the algorithms' implementation is considered for multi-channel single-pitch excited LPC vocoders. The Motorola DSP56000/DSP56001 is selected and the execution times are presented. A variable length coding scheme for reduced bit-rate is also described. Up to 10 vocoders are simultaneously implemented on a single DSP and the algorithms can be used as the basis for the implementation of more sophisticated speech coders.

High quality multiband LPC coding of speech at 2.4 kbit/s

Until recently good quality digital speech transmission was not possible below about 9.6 kbit/s. With the development of code excited linear prediction (CELP) and multiband excitation (MBE) vocoders,2 high quality digital speech transmission became possible at bit rates as low as 4.8 kbit/s. Below 4.8 kbit/s however, vocoders such as LPC-103 and the channel vocoder4 which can only produce synthetic and unnatural speech are still in use. A new coding scheme termed the multiband LPC (MB-LPC) vocoder is proposed, which produces good and natural quality speech at 2.4 kbit/s. Subjective performance of speech at 2.4kbit/s produced by the CELP is very close to MBE coded speech at 4.8 kbit/s. Informal listening tests have shown that in most cases people could not tell the difference between the new 2.4kbit/s MB-LPC coder and the 4.8 kbit/s MBE vocoder.

Autocorrelation-based pitch determination algorithms for realtime vocoders with the TMS32020/C25

In the paper, an autocorrelation-based pitch determination algorithm (PDA) is described. Realtime implementation on TMS32020/C25 digital signal processors is considered. Several optimizations and variations of autocorrelation PDAs have been implemented on a single TMS32020-based system. The algorithms are primarily designed with TMS32020/C25 LPC vocoders in mind and combine high speed, good performance and flexibility.

Very low bit rate speech coding based on joint segmentation and variable length segment quantizer

This paper presents a very low bit rate LPC vocoder based on a joint segmentation and quantization method using spectral segments having variable time length. The method exploits the nonuniform distribution of speech characteristics in the time and spectral domains. A measure of the spectral distance between a variable‐length input speech segment and a fixed‐length segment template is introduced based on linear time warping. The optimum segment boundaries and templates for a spectral sequence are efficiently determined using a dynamic programming technique so that the total spectral distortion in a voice interval is minimized. The segment templates are obtained by a sub‐optimum pattern learning method, which guarantees a monotonic decrease in distortion, using a combined segmentation and clustering technique. Experimental results for a single male speaker show that this method reduces the initial distortion by 20% and yields a sound articulation score of 78%.

Toward improving speech intelligibility

One difference observed between conversational and clear speech appears to be the presence of greater amounts of vowel reduction and target formant frequency “undershoot” in conversational speech. Thus vowel formant frequencies in clear speech more closely reach targetlike values, and transitions out of semivowels are steeper in slope [Picheny and Durlach (1979)]. To begin examining the contribution of such effects to the overall improvement in intelligibility of clear speech, an algorithm has been developed which, approximately, inverts the effects of reduction and undershoot in conversational speech. The algorithm is applied to the parameters used by an LPC vocoder program so that, in three steps, synthesis parameters are obtained via LPC analysis, changes in the time course of each synthesis parameter are magnified, and a synthetic utterance is generated which contains enhanced time‐changing information. Thus the algorithm operates on spectral change in the original version of an utterance, and magnifi...

Low bit rate real‐time half duplex LPC vocoder utilizing minimal random access memory

A low bit rate real‐time half duplex LPC vocoder was constructed based upon the Texas Instruments TMS32010 digital signal processor IC. Approximately 2700 words of external ROM were used for program and tables. The 144 words of onboard RAM were the only words of RAM used. The analysis was performed on preemphasized windowed speech using autocorrelation accumulation and the LeRoux‐Gueguen method to generate reflection coefficients and gain. A modified Rabiner‐Gold pitch algorithm was used for pitch and voicing. The software was split into foreground computations done on each speech sample and background computations done on each frame. There were 54 LPC information bits generated per frame.

An enhanced LPC vocoder with no voiced/Unvoiced switch

A modified LPC system (LPC plus) which requires no voiced/unvoiced switch at the synthesizer is presented in this paper. The excitation functions of the synthesizer filter are modeled to be the sum of the conventional pulse and noise sources. The mixture ratio is estimated from the LPC residual error signal, and this parameter controls the amplitudes of the pulse and noise sources. Since the V/UV switch is eliminated, this system produces robust speech in highly noisy environments, while a conventional LPC system produces degraded speech due to voicing errors. In addition, this technique has been applied to the speech of two simultaneous speakers and produces synthesized speech which is more natural and intelligible than that produced by conventional LPC.

Novel programmable wave-SC filter structure for pole-zero modelling of vocal tract

In the letter a novel programmable wave-SC filter structure is presented which is suitable for implementation of a LPC vocoder including transmission zeros. Owing to distributed arithmetic five basic cells and two new slightly modified cells, which correspond to the nasal branch, are required for the entire SC vocoder. Each cell contains one programmable capacitor array and the capacitive loading of all op-amps is constant. The latter essentially simplifies the op-amp design and allows the use of low-power dynamic amplifiers. The presented structure employs only fully stray-insensitive SC-amphfier and integrator circuits and offers a low sensitivity to coefficient quantisation.

Implementation of a Two-Channel LPC Vocoder

In this paper we describe hardware implementation of a 2400 bit/s two-channel linear predictive vocoder. The vocoder hardware uses the latest 2900 series "bit slice" microprocessor chips, and 2K RAM's and 1K ROM's of data memory. The system design is a two bus structure with a 208 ns cycle time. A significant feature of the vocoder is that it is capable of processing two voice channels simultaneously. Throughout the paper, emphasis is placed on details of firmware development of the vocoder system. Efficient design of the vocoder hardware is also discussed.

Variable rate speech compression by encoding subsets of the PARCOR coefficients

In LPC analysis, the speech signal is divided into frames each of which is represented by a vector of estimated vocal tract parameters, assumed to be constant throughout the frame. For many sounds, these parameters do not change significantly from one frame to the next, and some of them can often be adequately represented by previously transmitted values. In the LPC coding systems described in this paper, a number of alternative representations are considered for each frame. These representations (vectors) are combinations of PARCOR coefficients from the current frame and from previous frames. Several consecutive frames are analyzed at once, and all the possible sequences of PARCOR coefficient vectors are examined. The sequence which minimizes a preselected cost function is chosen for transmission, resulting in a reduced overall data rate. The examination of all the decision sequences is equivalent to a decision tree search, which is most efficiently accomplished through dynamic programming. Using these techniques, LPC encoded speech at 1200 bits/s is demonstrated to be of quality comparable to a constant rate LPC vocoder at 2400 bits/s.