We review the variable frame rate (VFR) transmission methodology that we developed, implemented, and tested during the period 1973-1978 for efficiently transmitting LPC vocoder parameters extracted from the input speech at a fixed frame rate. In the VFR method, parameters are transmitted only when their values have changed sufficiently over the interval since their preceding transmission. We explored two distinct approaches to automatic implementation of the VFR method. The first approach bases the transmission decisions on comparisons of the parameter values of the present frame and the last transmitted frame. The second approach, which is based on a functional perceptual model of speech, compares the parameter values of all the frames that lie in the interval between the present frame and the last transmitted frame against a linear model of parameter variation over that interval. The application of VFR transmission to the design of narrow-band LPC speech coders with average bit rates of 2000-2400 bits/s is also considered. The transmission decisions are made separately for the three sets of LPC parameters, pitch, gain, and spectral parameters, using separate VFR schemes. A formal subjective spccch quality test of six selected LPC coders is described, and the results are presented and analyzed in detail. It is shown that a 2075 bit/s VFR coder produces speech quality equal to or better than that of a 5700 bit/s fixed frame rate coder.

Variable Frame Rate Transmission: A Review of Methodology and Application to Narrow-Band LPC Speech Coding

Speech synthesis system implementable in an integrated circuit device capable of converting frames of speech data at a variable frame rate into analog signals representative of human speech. The frames of speech data comprise digital representations of values of pitch, energy, filter coefficients and coded frame rate data. The speech synthesis system includes a linear predictive coding filter as a speech synthesizer which utilizes the speech data at a varying frame rate to produce digital speech signals representative of human speech. Frames of digital speech data including coded frame rate data are received by an input, with the frame rate data being decoded to control both the rate at which the incoming variable-length frames of speech data are accepted by the speech synthesizer and the number of interpolation calculations required to define interpolated speech values between adjacent incoming frames of speech data. A frame control circuit accomplishes the foregoing utilization of speech data at a variable frame rate by the speech synthesizer by providing for a variable number of interpolation calculations between adjacent speech frames from last implemented speech data in which the number of interpolation calculations in a given instance is determined by the frame rate data. A microprocessor controls the access of selected speech data which is stored in a memory. The system also includes a digital-to-analog converter for converting the digital speech signals produced by the filter into analog signals and a speaker for generating audible sounds in the form of synthesized human speech from the analog signals provided by the digital-to-analog converter.

Speech synthesis system utilizing variable frame rate

Method and system for encoding digital speech information to characterize spoken human speech with an optimally reduced speech data rate while retaining speech quality in the audible reproduction of the encoded digital speech information. Markov modeling is applied to quantized speech parameters to represent their time behavior in a probabilistic manner. This is accomplished by representing the quantized speech parameters as finite state machines having predetermined matrices of transitional probabilities from which the conditional probabilities as to the quantized speech parameter values of successive speech data frames are established. The probabilistic description as so obtained is then used to represent the respective quantized values of the speech parameters by a digital code through Huffman coding in which digital codewords of variable length represent the quantized speech parameter values in accordance with their probability of occurrence such that more probable quantized values are assigned digital codewords of a shorter bit length while less probable quantized values are assigned digital codewords of a longer bit length.

Method and system for encoding digital speech information

A variable frame length vocoder extracts a feature vector for each given frame, a predetermined number of frames being defined as a section. The feature vectors in each section are stored, changes in feature vectors within a section being approximated by a given number of variable time length flat sections with a constant time length portion between adjacent flat sections, adjacent flat sections being interconnected by an inclined section of the constant time length duration. A feature vector of each flat section is outputted as a representative vector of the flat section, and the number of frames comprising the flat section is outputted as a repeat signal. This information is processed at the synthesis side of the vocoder to produce the feature vector in each inclined section by interpolating the representative vectors of the flat sections on both sides of the inclined section.

Variable frame length vocoder

Methods and apparatus are presented for reducing the number of bits needed to represent an excitation waveform. An acoustic signal in an analysis frame is analyzed to determine whether it is a band-limited signal. (305) A sub-sampled sparse codebook is used to generate the excitation waveform if the acoustic signal is a band-limited signal. (320) The sub-sampled sparse codebook is generated by decimating permissible pulse locations from the codebook track in accordance with the frequency characteristic of the acoustic signal.

Sub-sampled excitation waveform codebooks

Variable data rate LPC speech compression schemes are employed to transmit LPC parameters only when speech characteristics have changed sufficiently since the last transmission, yielding improved speech quality relative to fixed-rate schemes for a given average transmission rate. Transmission of variable-rate LPC speech over fixed-rate channels is accomplished using transmit and receive buffers, with resulting transmission delays. Development of proper buffer control strategy is essential to minimize losses caused by exhausting either buffer, or by corrective actions, namely, forced or suppressed transmission. Certain aspects of such strategy and their impact on speech quality and data rate are discussed for a narrowband (2400 bps) speech transmission system.

http://ieeexplore.ieee.org/iel6/8365/26347/01170225.pdf

Variable-to-fixed rate conversion of narrowband LPC speech

Recently we described a variable-frame-rate LPC vocoder designed to transmit good quality speech over 2400 bps fixed-rate noisy channels with bit-error probabilities ranging up to 5% [3]. The basic idea was to lower the data rate by transmitting LPC parameters only when speech characteristics have changed sufficiently since the last transmission, and to employ the resulting bit-rate savings for protecting important transmission data against channel noise. This paper describes our continuing efforts which have concentrated on minimizing loss of synchronization between the receiver and the transmitter. In one approach, we emphasize heavy protection of header, and rapid resynchronization. Alternatively, we apply constraints which guarantee synchronization at a cost of some freedom in the selection of data for transmission. Results from the first approach are presented; results from both methods will be compared at the conference.

E. Blackman

Papers

Variable-to-fixed rate conversion of narrowband LPC speech

Narrowband LPC speech transmission over noisy channels