Abstract: The spectral or pitch parameters of a speech signal are quantized, and impulse responses thereof are predicted by using a filter. An orthogonal transform is made of the speech signal, or a signal derived therefrom, or of the impulse responses or signals derived therefrom. The result of the orthogonal transform is entirely or partly quantized to obtain a plurality of pulses. More preferably, these pulses are retrieved recurrently by also using codevectors retrieved from a codebook or collectively quantizing their senses or amplitudes. This method optimizes speech signal coding.

Abstract: For voice and musical signal, a signal coding system which can obtain good sound quality even at a low bit rate is provided. The signal coding system predicts an input signal in a predicting circuit and performs orthogonal transformation in a first orthogonal transformation circuit of a predicted residual error signal. In a coefficient calculating circuit, a coefficient of a smaller degree is calculated for expressing an envelop of the orthogonal transformation coefficient in the orthogonal transformation circuit. In a quantization circuit, quantization is performed by expressing the orthogonal transformation coefficient with a plurality of pulse trains with determining positions for generating a pulse using the output of the coefficient calculating circuit. The envelop of the orthogonal transformation coefficient is expressed by the coefficient with smaller degree.

Abstract: A telephone apparatus of a telephone system, including at least one server for connecting a plurality of telephone apparatuses connected to a computer network, comprises: a communication circuit means for processing data to and from said computer network; an audio input means for generating a digital audio signal from input voice; an audio compression means for compressing the audio signal from the audio input means and for supplying the compressed audio signal to the communication circuit means; and a control means for determining a compression rate of the audio compression means in response to information on the ratio of use of the computer network obtained from the server.

Abstract: When a secondary battery such as a lithium ion battery, etc. requiring constant-voltage charging is charged, the battery voltage and charging current of the secondary battery are detected under a specified state, and in accord with the detected state, a voltage at which charging is carried out, the charging time or charging stop period thereof, or connecting condition of the charging circuit etc. are controlled in order to favorably charge the secondary battery to a full-charge or to a nearly full-charged state, thereby effectively preventing deterioration of characteristics of the secondary battery.

Abstract: A telephone system including telephone terminals in computer networks connected via servers to transmit data containing at least audio data through the Internet. Upon receipt of information on a source terminal sent via a server, a destination terminal performs operations based on the information. For example, the destination terminal judges whether the source terminal is a previously specified terminal, and connects or does not connect itself to the source terminal in accordance with the judgement.

Abstract: Backward movement of a heavy motorcycle is assisted by a starter motor for cranking an engine of the motorcycle. A backward drive control system includes a pulse width modulation circuit for supplying operating current to the starter motor in a controlled manner and a device for estimating operating temperature of the starter motor based on the current supplied to the starter motor. When the estimated temperature reaches a predetermined maximum temperature, the operation of the starter motor is compulsorily stopped and prohibited for a certain period or until the starter motor is cooled down to a certain temperature level. Overheating of the starter motor is effectively prevented without using a temperature sensor for measuring the operating temperature.Moreover, chattering of the starter motor does not occur because its operation is prohibited for a certain period after the starter motor is once compulsorily stopped.

Abstract: An excitation quantizer 60 in a speech encoder includes a divider, which divides M pulses representing in combination a speech signal into groups each of L pulses, L being smaller than M. The amplitude of pulses, i.e., L pulses as each unit, is quantized, using spectral parameter. The quantization is executed on at least one quantization candidate, which is selected through distortion evaluation made through addition of the evaluation value based on an adjacent group quantization candidate output value and the evaluation value based on the pertinent group quantization value.

Abstract: A telephone system using terminals in computer network connected via servers to transmit data containing at least audio data through the computer networks. A server is provided with a data base storing call-out information used to connect to a destination terminal. When a source terminal makes a call to a destination terminal, the server of the computer network searches out the connection address of the destination terminal from the data base. Then, the server calls up the destination terminal and establishes connection between the server of the computer network and the destination terminal.

Abstract: A first coefficient generating unit derives, from a past speech reproduction signal, a first coefficient signal representing a spectral characteristic of the past speech reproduction signal. A residual signal generating unit derives, from a speech signal for each frame, a predicted residue signal by using the first coefficients. A second coefficient generator derives second coefficients representing a spectral characteristic of the predicted residue signal. A second coefficient quantizing unit quantizes the second coefficients and provides a quantized coefficient signal. An excitation quantizing unit derives an excitation signal concerning the speech signal by using the speech signal, the first coefficient signal, the second coefficient signal and the quantized coefficient signal, quantizes the excitation signal thus derived and provides a quantized excitation signal.

Abstract: In a speech encoder, a gain codebook switching circuit is supplied with short-term prediction gains from a short-term prediction gain calculator circuit and with mode information through an input terminal and compares the short-term prediction gains with a predetermined threshold value when the mode information indicates a predetermined mode. As a result of comparison, the gain codebook switching circuit produces gain codebook switching information which is delivered to a gain quantizer circuit. The gain codebook quantizer circuit is supplied with adaptive code vectors, excitation code vectors, impulse response information, and the gain codebook switching information, and gain code vectors from a particular gain codebook connected to one of a plurality of input terminals that is selected by the gain codebook switching information.

Abstract: In a speech coder, an excitation quantizer 360 retrieves the positions of M non-zero amplitude pulses, which together constitute an excitation, by using spectral parameters and with a different gain for each group of the pulses less in number than M.

Abstract: A speech signal encoder includes a speech analyzer for determining short-term prediction codes at a predetermined time interval. The prediction codes indicate frequency characteristics of a speech signal. A reverse filter is provided for calculating residual signals of first synthesis filter. The residual signals are defined by the short-term prediction codes. A residual code book stores past residual signals. Further, a plurality of delay codes, each of which represents pitch correlation of the speech signal, are predetermined. A vector generator issues, using the residual code book, delay residual vectors each of which corresponds to the delay code. A filter is provided for generating a synthesis signal using second synthesis filter which receives the delay residual vectors and which is defined by the short-term prediction codes. A distance between the speech signal and the synthesis signal is calculated. Subsequently, a pitch path estimator estimates a pitch path which varies smoothly.

Abstract: In a speech coding apparatus, a spectral parameter calculator determines spectral parameters from an inputted speech signal, quantizes the spectral parameters, and outputs a plurality of quantization candidates. An adaptive code book determines delays with respect to each of the quantization candidates outputted from the spectral parameter calculator, generates a pitch predictive signal based on a past excitation signal for each of the delays and associating quantization candidates, and outputs a quantization candidate and a delay that provide a minimum distortion between the speech signal and the pitch predictive signal. An excitation quantizer quantizes and outputs the excitation signal of the speech signal. A gain quantizer quantizes and outputs a gain of at least one of the adaptive code book and the quantized excitation signal.

Abstract: A vertical drive circuit comprises a shift circuit composed of a plurality of cascaded half-bit scan circuits, a plurality of NAND gate circuits controlled by output signals of the scan circuits and control signals, and a plurality of output buffer circuits connected to the NAND gate circuits. A horizontal drive circuit comprises a shift circuit composed of a plurality of cascaded half-bit scan circuits, a plurality of first NAND gate circuits controlled by output signals of the scan circuits and control signals, a plurality of second NAND gate circuits controlled by output signals of the first NAND gate circuits and enable signals, and a plurality of data sampling and holding circuits controlled by output signals of the second NAND gate circuits.

Abstract: Wide-band speech signals and also music signals are coded with relatively less computational efforts and less sound quality deterioration even at low bit rates. A spectral parameter calculator obtains a spectral parameter from sub-frames of an input signal from a sub-frame divider, and quantizes the obtained spectral parameter. A divider divides the difference result from a subcontractor into a plurality of sub-bands. Adaptive codebook circuits obtain a pitch prediction signal by obtaining pitch data in at least one of the sub-bands. Judging circuits execute pitch prediction judgment by using the pitch data in at least one of the sub-bands. A synthesizer synthesizes a pitch prediction signal. A subtractor subtracts the pitch prediction signal from the difference result obtained from a subtractor and thus obtains an excitation signal. An excitation quantizer quantizes the excitation signal with reference to an excitation codebook.

Abstract: An audio recording and reproducing apparatus includes a controller for controlling the entire behaviors, hard disc for write and read of audio data, audio compression/expansion circuit for expanding compressed audio data, and external I/O port. The audio recording and reproducing apparatus is connected to a network service center to obtain desired music data from storage of the network service center and to store it in the hard disc.

Abstract: A receiving unit receives input speech data on a frame-by-frame basis. An error detection unit checks whether errors exist in each frame, and outputs a signal indicative thereof to a first switch circuit. The first switch circuit outputs the input speech data to a second switch circuit if an error is detected, while it outputs the input speech data to a speech decoder unit if no error is detected. A data memory stores the input speech data after delaying the data by one frame, and outputs the delayed data to a bad frame masking unit for voiced frame, and a bad frame masking unit for unvoiced frame. The speech decoder unit decodes the input speech data by using spectral parameter data, delay of an adaptive codebook, an index of an excitation codebook, gains of the adaptive and excitation codebooks, and the amplitude of the input speech signal. The speech decoder unit outputs a decoding result to a voiced/unvoiced frame judging unit, as well as to an output terminal.

Abstract: A signal coding apparatus includes a signal dividing section for dividing an input signal in units of frames and in units of bands, a pulse allocating section for determining a performance request value for each of the bands, for determining a number of pulses for each of the bands from the performance request value for the band, and for adaptively allocating the determined numbers of pulses to the bands for every frame, and a plurality of coding circuits respectively provided for the bands. Each of said plurality of coding circuits generates a transmission signal for a corresponding band for every frame based on the number of pulses allocated to the corresponding band.

Abstract: The invention provides a speech coding apparatus by which a good sound quality can be obtained even when the bit rate is low. The speech coding apparatus includes an excitation quantization circuit which quantizes an excitation signal using a plurality of pulses. The position of at least one of the pulses is represented by a number of bits determined in advance, and the amplitude of the pulse is determined in advance depending upon the position of the pulse.

Abstract: A block length judging circuit 120 switches block lengths based on a feature quantity obtained from an input signal. A transform circuit 200 executes transform of the signal into frequency components according to the block length. A masking threshold calculating circuit 250 calculates a masking threshold simulating the masking characteristic of psychoacoustical property for each predetermined intra-block section. An inter-block/intra-block bit assignment circuit 300 executes inter-block bit number assignment and/or intra-block bit number assignment to each predetermined intra-block section. A vector quantization circuit 350 vector quantizes transform signal by switching codebooks 360.sub.1 to 360.sub.N according to the assignment bit number, and also quantizes gain by using a gain codebook 370.