摘要:
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.
摘要:
A speech parameter encoder capable of encoding spectrum parameters at a bit rate of 1 kb/s or less with comparatively small amount of operations and memory capacity. A spectrum parameter calculation unit 130 derives a spectrum parameter representing the spectrum envelope of a discrete input speech signal through division thereof into frames each having a predetermined time length. A weighted coefficient calculation unit 150 derives a weighted coefficient corresponding to an auditory masking threshold value through derivation thereof from the speech signal. A spectrum parameter quantization unit 160 receives the weighted coefficient and the spectrum parameter and quantizes the spectrum parameter through search of a codebook such as to minimize the weighting distortion based on the weighted coefficient.
摘要:
A speech coding method in which spectrum parameter representing a spectrum envelope and a pitch parameter representing a pitch are obtained from an input discrete speech signal. A frame interval is divided into subintervals in accordance with the pitch parameter. A sound source signal in one of the subintervals is obtained by obtaining a multipulse with respect to a difference signal obtained by performing prediction on the basis of a past sound source signal. Correction information for correcting at least one of the amplitude and the phase of the sound source signal are obtained and output in other pitch intervals in the frame.
摘要:
A voice encoding system is constituted by a short time voice signal series producing circuit inputted with a discrete voice signal series for dividing the same at each short time; a parameter extracting circuit for extracting a parameter representative of a spectrum envelope from the short time voice signal series and encoding the parameter; an impulse response series calculating circuit for calculating the impulse response series based on the parameter representative of the spectrum envelope; an autocorrelation function sequence calculating circuit utilizing the impulse response series; a cross-correlation function sequence calculating circuit utilizing the impulse response series and the short time voice signal series; a circuit for calculating and encoding an excitation signal series of the short time voice signal series by utilizing the autocorrelation function sequence; and a circuit for combining and outputting a code of the parameter representative of the spectrum envelope and a code representative of the excitation signal series. With the system, high quality voice encoding can be made at a transmission rate of less than 10k bits/second with a relatively small amount of calculation.
摘要:
A gateway apparatus receives a call control signal and/or a packet with voice data stored therein in a predetermined protocol from a packet transfer apparatus on a mobile high-speed network and converts the received protocol into a circuit-switched protocol used when an RNC connects to a circuit switching equipment on a mobile circuit-switched network, for output to the circuit switching equipment The gateway apparatus, on receipt of a call process signal and/or a voice signal, from the circuit switching equipment, converts the received protocol for output to the packet transfer apparatus.
摘要:
A server device includes: a storage unit that, upon receiving content storage instruction information from a mobile terminal via a packet forwarding device in a mobile network and a content stream or content file, stores the content stream or content file; a virtual client unit that, upon receiving content reproduction instruction information from the mobile terminal, reads, decodes and reproduces a content stream or content file stored in the storage unit to generate display screen data; and an encoder unit that transmits an encoding result obtained by compression-encoding part or all of the display screen data as a packet to the packet forwarding device.
摘要:
During audio communication between terminals, an audio quality analyzing device picks up from a network a packet containing a bit stream obtained by compression encoding of audio sent by at least one of the terminals. The device has an audio quality analyzing unit that, in addition to analyzing a header of the packet picked up, performs at least one of: an analysis of a payload header and an analysis of the bit stream contained in the payload, detects deterioration in the quality of the audio communication service, and notifies a result of the detection to an upper level device.
摘要:
A gateway apparatus arranged between the mobile circuit switched network and the mobile IMS network, includes a call control unit which converts a control signal output from the mobile circuit switched network into at least one of SIP and SDP to output a signal converted to the mobile IMS network and a conversion unit which receives a line switching protocol for an audio signal output from the mobile circuit switched network. The conversion unit decides, using at least one of SIP and SDP, output from the mobile IMS network and received by the call control unit, on whether the partner terminal is to be connected via the mobile IMS network to the mobile circuit switched network, via the mobile IMS network to a fixed network or via the mobile IMS network to a mobile broadband network, then converts the line switching protocol to one of different protocol, depending on a result of the decision, and outputs the protocol converted signal to the mobile IMS network (FIG. 2).
摘要:
In a delivery network system, deliver a server that holds requested content is determined, using information included in a content request message from a communication terminal. A file or stream of the requested content is read from the determined server. When the content is moving image content, a bandwidth of a mobile network is estimated, based on a response signal or a congestion state notification signal from the communication terminal. The moving content is transmitted by controlling a timing of transmitting the read file or the read stream so as not exceed the estimated bandwidth.
摘要:
When a location of a portable terminal connected to another network beyond a packet communication network side changes, the gateway apparatus that is arranged between a mobile circuit switching network and a packet communication network (IMS network) and that performs mutual conversion between a circuit switching protocol and a packet communication protocol to execute voice communication, exercises control such that at least one of a sequence number and a timestamp in a packet destined to a portable terminal and output to the packet communication network, is made to vary continuously before and after movement of the portable terminal, in case the portable terminal connected to a network beyond the packet communication network side, moves its location.