Pulse-code Modulated Signals Research Articles

Design of Intelligent Speech Translation System Based on Deep Learning

In order to solve the problem of low translation accuracy caused by complex sentence parameters in traditional machine translation systems, a method based on deep learning was proposed. First, MCU SPCE061A is used to study the problem of complex digital signal. The training data in the synchronous translation server support the translation services of a large number of users, and the translation results were displayed through the session interface of the user terminal. The PMDL model is used to detect the keyword signal, record the PCM audio data, and slice the collected pulse code modulation signal, so as to wake up the artificial intelligence voice service. Then, this study establishes a speech recognition process that accurately outputs the speech-related semantics. In this paper, a manual interactive synchronous translation program is designed with the input text as the search criterion, and the set is trimmed to obtain the best translation effect. The experimental results show that the sentence translation accuracy of the system is 0.9 ∼ 1.0. It is proved that the method based on deep learning solves the problem of low accuracy of the traditional translation system.

Digital Sine Oscillators Based on Higher Order Sigma–Delta Modulation

In this paper, a new class of digital sine oscillators based on higher-order multibit sigma-delta modulation is presented. The samples of a given pure sine signal are quantized, resulting in a pulse-code modulated (PCM) sequence. A numeric condition for oscillation (NCO) based on the PCM signal is disclosed. If the NCO is fulfilled, a system constant <italic xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">F</i> with a finite bit width can be derived and a hardware-efficient practical realization of a digital oscillator based on <italic xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">R</i> <sup xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">th</sup> -order sigma-delta modulation is possible. The oscillator output sequence is an <italic xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">R</i> -times differentiated version of the PCM signal and thus the PCM signal can be reconstructed by <italic xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">R</i> -subsequent accumulations. All signals are absolutely periodic and are represented by binary numbers with well-defined finite bit widths. Initial conditions for the registers of the oscillator are determined. An example for the generation of a high-purity PCM signal with 2048 samples per period is demonstrated. The noise spectrum of this signal is white and the ratio between the mean noise amplitude and the signal amplitude is about one over one billion. The resulting oscillator requires a third order sigma-delta modulator. The chip size is estimated for the oscillator being integrated in a commonly used CMOS technology.

Music Rhythm Detection Algorithm Based on Multipath Search and Cluster Analysis

Music rhythm detection and tracking is an important part of the music comprehension system and visualization system. The music signal is subjected to a short-time Fourier transform to obtain the frequency spectrum. According to the perception characteristics of the human auditory system, the spectrum amplitude is logarithmically processed, and the endpoint intensity curve and the phase information of the peak value are output through half-wave rectification. The Pulse Code Modulation (PCM) characteristic value is extracted according to the autocorrelation characteristic of the endpoint intensity curve. This article proposes a rhythm detection algorithm based on multipath search and cluster analysis; that is, based on the clustering algorithm, it absorbs the idea of multipath tracking and proposes its own detection and tracking algorithm. It overcomes the weakness of the clustering algorithm that needs to use Musical Instrument Digital Interface (MIDI) auxiliary input to achieve the desired effect. This algorithm completely uses the PCM signal as the input, which is more robust than the clustering algorithm. The whole process is carried out in the time domain, and the amount of calculation is much smaller than the frequency domain calculation of multipath tracking, and the linear relationship with the rhythm of the music is much better than the filter bank algorithm. This algorithm can successfully detect the rhythm of the music with a strong sense of rhythm and can track the specific position of the rhythm point.

Improvement in Speed of Trains in Poor Visibility &amp; Advancement of Train Transport Signaling Mechanism

The Railway system is heavily hampered by poor weather conditions like fog, smog and heavy rainfall especially in Northern India. These factors lead to poor visibility and reduces the average speed of trains which is the main reason for their delay, particularly in winter season. The average speed of the train is kept low so that the locomotive pilot can easily watch out signals and control the movement of the train. This paper focuses on techniques to boost the average speed of trains during low visibility by improving the transport signaling mechanism using wireless communication mechanisms using pulse code modulated signals.

Improved performance of analog and digital acousto-optic modulation with feedback under profiled beam propagation for secure communication using chaos

Using intensity feedback, the closed-loop behavior of an acousto-optic hybrid device under profiled beam propagation has been recently shown to exhibit wider chaotic bands potentially leading to an increase in both the dynamic range and sensitivity to key parameters that characterize the encryption. In this work, a detailed examination is carried out vis-a-vis the robustness of the encryption/decryption process relative to parameter mismatch for both analog and pulse code modulation signals, and bit error rate (BER) curves are used to examine the impact of additive white noise. The simulations with profiled input beams are shown to produce a stronger encryption key (i.e., much lower parametric tolerance thresholds) relative to simulations with uniform plane wave input beams. In each case, it is shown that the tolerance for key parameters drops by factors ranging from 10 to 20 times below those for uniform plane wave propagation. Results are shown to be at consistently lower tolerances for secure transmission of analog and digital signals using parameter tolerance measures, as well as BER performance measures for digital signals. These results hold out the promise for considerably greater information transmission security for such a system.

Apparatus for converting PCM signal

An apparatus for converting pulse code modulation (PCM) signals in a telecommunication system between two different modulation standards. In the apparatus, a channel selector generates a channel select signal for identifying at least one channel of the multiple channels containing input digital signals modulated by said one of said two different modulation standards. A first codec demodulates a signal on a channel designated by the channel select signal which is modulated by the first modulation standard, into an analog signal to be supplied to a second codec, which modulates the converted analog signal input into a digital signal using the second modulation standard. The reverse process is executed in a similar manner. A second codec using the second modulation standard demodulates a digital signal on the channel designated by the channel select signal into an analog signal to be supplied to the first codec, and the first codec in turn modulates the converted analog signal into a digital signal using the first modulation standard. As a result, the present invention is able to convert pulse code modulation signals from either one of two different modulation standards to the other one of said two different modulation standards in a telecommunication system.

Acoustic characteristics of an electrodynamic planar digital loudspeaker.

In this paper, an electrodynamic planar loudspeaker driven by a digital signal is experimentally discussed. The digital loudspeaker consists of 22 voice coils, 11 permanent magnets, a diaphragm with streamlined sections molded in plastic, and a suspension made of handmade Japanese paper between the diaphragm and the frame. First, the acoustic responses are affected by the arrangement of the voice coils, so an asymmetric arrangement is studied. This asymmetric arrangement is designed to obtain as flat a frequency response to an analog signal as possible. This arrangement is compared with a symmetric one and results show that the flatness of the frequency response around 1 kHz and 4 kHz is improved and that the sound reproduction band is from 40 Hz to 10 kHz. Second, to evaluate the acoustic responses to a digital signal, the digital loudspeaker is driven with a pulse code modulation signal. Results show that the digital loudspeaker can reproduce pure sound with a total harmonic distortion of less than 5% from 40 Hz to 10 kHz, exceeding this value only in a narrow frequency band near 4 kHz. This digital loudspeaker was demonstrated to have good linearity over its dynamic range of 84 dB.

A study on the digitization of audio signal for S-VHS VCR

The recording and reproducing characteristics of the PCM (pulse code modulation) audio signal, reduction of interference to the video signal by the PCM audio signal, and performance against tracking error have been studied on a digital audio system for S-VHS VCR (video cassette recorder). A block error rate of 5*10/sup -4/ is attained without degradation of both video and FM audio signals. Amorphous heads have been used for both video and audio heads and the PCM signal carrier-to-noise ratio of 19 dB in the EP mode has been obtained without degradation of the video signal. It is confirmed that the uncorrectable error rate is under 1 symbol/second even in the case of a tracking error of -20 mu m to 9 mu m in the EP mode.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

Unequal error protection of PCM signals by self-orthogonal convolutional codes

In the PCM (pulse-code modulation) transmission system discussed, the influence of bit errors on the demodulated signal-to-noise ratio differs in the erroneous bit position. The demodulated signal-to-noise ratio can be improved when the PCM signals are protected so that the decoded bit error rate in more important bit position may be lower. Assigning a different number of orthogonal check sums to each information bit of self-orthogonal convolutional code provides a different error protection capability to each information bit. The author presents an unequal error protection self-orthogonal convolutional code with different error protection capability for each information bit, and discusses the condition that makes the demodulated signal-to-noise ratio of PCM signals maximum. >

Method of detecting the presence of speech in a telephone signal and speech detector implementing said method

In a method of detecting the presence of speech in a telephone signal, a signal sequence is interpreted as constituting speech if its mean power level exceeds a threshold, the value of which is decreased as the number of passages through zero level in the sequence approaches 1.25 per millisecond (0.25 Hz). A speech detector implementing this method receives pulse code modulated signal samples on its input. It comprises a squaring circuit driving a digital accumulator whose output signal is proportional to the mean power level of each of consecutive 4 millisecond signal sequences. A change of sign detector and a second digital accumulator indicate the number of passages through zero level in each of these sequences. Banks of threshold comparators and a selector circuit are used to choose between the various threshold values. The selected value is lowest when the number of passages through zero level is between 5 and 6 per millisecond.

Embedded delta modulation

In an embedded delta modulation (EDM) encoder, a supplemental pulse-code-modulation (PCM) encoder processes the error signal of a delta modulator. The PCM and DM bit streams are multiplexed and then transmitted. The decoder adds and then filters the analog versions of the separately decoded PCM and DM signals. To enhance the efficiency of EDM, it is desirable to filter and decimate the DM error signal prior to PCM encoding. Corresponding filtering and decimation of the decoded DM signal is then required. The theory of both full-rate and reduced-rate EDM, which include filtering and decimation, is presented. The performance of these codes are those of compared to PCM, DM, and two embedded codes, embedded-differential PCM and multistage DM. It is found that reduced-rate EDM has a higher signal-to-noise ratio than DM and PCM. It is comparable in performance to multistage DM but simpler to implement. The signal-to-noise ratio of EDM is lower than that of embedded DPCM. However, EDM is simpler to implement, and the signal-to-noise ratio penalty is less than 2.6 dB over a range of transmission rates of practical interest. >

Sampled speech compression system

A sampled speech compression and expansion system, for two-dimensional prssing of speech or other type of audio signal, comprises transmit/encode apparatus and receive/decode apparatus. The transmit/encode apparatus comprises a low-pass filter, adapted to receive an input signal, for passing through low-frequency analog signals. A converter is connected to the low-pass filter for converting the analog signal into a digital signal. A buffer memory, whose input is connected to the converting means, stores the digitized signals. A correlator, having inputs from the A/D converter and the buffer memory, correlates the digital signal received directly from the converter with a delayed signal from the buffer memory. An "interval-select" circuit, whose input is connected to the output of the correlator, uses the autocorrelation value as a basis for comparison with subsequent peaks in the correlation value which are greater than a specified fraction of the autocorrelation value. The interval-select circuit has an output which is connected to the buffer memory, the value of the fractional peaks and their timing being stored in the buffer memory. A transform circuit, whose input is connected to the buffer memory, performs an even discrete cosine transform (EDCT) of the stored signal. A first modulator, whose input is connected to the output of the EDCT means, differentially pulse code modulates (DPCM) its input signal. A second modulator, whose input is connected to the output of the interval select circuit, differentially pulse code modulates its input signal. A multiplexer, having an input connected to the output of the first and second modulating means, combines the two differentially pulse code modulated signals. A receiver/decoder has circuits which perform an inverse function to those of the transmitter/coder and are arranged in inverse order, from input to output, to those of the transmitter/coder.

A Circuit That Changes the Word Rate of Pulse Code Modulated Signals

In this paper we describe a circuit that accepts pulse code modulated signals sampled at about 8 kHz and resamples them at any desired rate up to 512 kHz. When the sampling satisfies Nyquist's criterion, the distortion introduced is at least 35 dB below the signal level. The circuit uses a digital low-pass filter to interpolate sample values, and it may be integrated as about 2500 gates on a 5 mm2 chip.

A Per-Channel A/D Converter Having 15-Segment µ-255 Companding

This paper describes a companded analog-to-digital (A/D) converter for voiceband signals that is simple and potentially inexpensive. The converter uses only 18 coarsely spaced analog levels. Fine resolution is obtained by oscillating between these levels at an increased speed and averaging the result over a Nyquist interval. The companding used in the converter is effectively the same as that of μ-255 pulse-code modulation (PCM). In the encoding process a one-bit code is generated at 256 000 samples/s. This 1-bit per sample signal can be transmitted and decoded directly, or a simple digital circuit will produce a 13-bit, 8-kHz linear PCM signal that can be compressed to 8-bit companded PCM format. In this paper the basic operation of the 1-bit coder is described and its performance when connected to a 1-bit decoder is illustrated. Methods for obtaining both linear and compressed PCM are then presented, and the properties of these PCM signals with respect to noise, gain tracking, and harmonic content are described. Relative insensitivity to circuit component variations, absence of analog gates, along with the need to generate only a few analog levels, make the coder especially well suited to integrated circuit realization.

PCM Encoded NTSC Color Television Subjective Tests

Digital television may effectively serve as the means to store, process and control color television signals within a broadcast plant. Pulse code modulation (PCM) involves the periodic amplitude sampling of a video signal and the conversion of each sample into a binary number of n bits. The PCM signal bit stream can be in n parallel channels or be time multiplexed into a single channel. In either case the total bit rate equals n times the encoding rate. We desire to minimize the bit rate in order to save circuit elements and bandwidth, consistent of course with acceptable picture quality. — A series of subjective tests was conducted to determine the importance of various parameters when PCM encoding an NTSC, 525-line, 60-field, composite television signal. Among these parameters are: (1) the number of amplitude levels being sampled that, in turn, define the n-bit digital word; (2) the encoding rate, both locked and unlocked with respect to the 3.58-MHz color subcarrier; and (3) the role of dither consisting of coherent square waves and weighted noise. The test setup consisted of a color film camera, encoding rate generators, dither generators, A/D and D/A converters, color and monochrome monitors. A test panel of experienced viewers examined the pictures in a controlled environment and rated each picture on a stepped acceptance scale. We concluded that NTSC television pictures that had been encoded with 6 or more bits per codeword suffered negligible impairment. The encode rate should be phase coherent with the color subcarrier in order to avoid beats in the picture. Dither signals did not make any improvement in picture quality except to mask contours during 4- or 5-bit encoding.

Multiplexing Systems for Pulse-Code-Modulated Hierarchial Networks

Various synchronization systems which function to multiplex lower speed pulse-code-modulated (PCM) signals into a high-speed signal and to drop each lower signal from the highspeed PCM signal, are considered from a practical viewpoint for a useful PCM hierarchial network, and a suitable evaluation basis for such systems is introduced. Three new stuff-synchronization methods are proposed and designed. A severly reliable bit synchronization function necessary for bit multiplexing can be provided by using a quartz crystal resonant circuit. Relations between the system parameters and transmission performance, such as framing time and stuffing error rate, especially stuff-jitter response for coded frequency-division-multiplex (FDM) signal, are analyzed, and the design technique has been established. The test equipments operated perfectly to verify the expected performances. The results suggest these systems are two or four times as advantageous as usual systems on the evaluation basis.

A 100-megabit nine-digit electron tube encoder

A beam deflection tube has been developed which will convert an amplitude modulated signal into a nine-digit binary pulse code modulated signal at sampling speeds in excess of 12 mcs/sec. The tube employs a ribbon beam which is electrostatically deflected over nine columns of apertures in a code plate. This paper describes the tube and the critical tolerances required to achieve a signal-to-noise ratio of 54 db, which is within 3 db of the theoretical limit for a nine-digit encoder. The fabrication techniques which were required to achieve these tolerances are radically different from those used in the encoding tube described at the 1960 Electron Device Meeting. Factors that degrade encoding accuracy include thin oxide films on electrodes and residual gas pressure; techniques for minimizing these effects will be discussed. A secondary emitting target comprised of gold sputtered on a molybdenum substrate, found to be very stable in accelerated aging tests, has been evolved. An output current of 5/µamps per digit is achieved with a peak cathode loading of approximately 100 ma/cm2.