Abstract: A voice presence/absence discriminator can accurately determine the presence or absence of a voice in a frame that includes an uttered syllable head portion of an input voice and avoids performing erroneous determinations in bad environments such as those where background noise is of a high magnitude. In a sub-frame power calculation section, a sub-frame power Pm is calculated in units of sub-frames prepared by dividing a frame into four sub-frame portions. Based on this sub-frame power Pm, in a frame maximum power production section, a moving average (short-period average value) of the power of a sub-frame and the power of a sub-frame that precedes this sub-frame by one unit are calculated in units of a sub-frame and the short-period average values are compared with each other among the sub-frames that constitute the same frame to thereby select a maximum one of them as the frame maximum power Pf of this frame.

Abstract: A silence-suppression scheme for a packet voice transmission system (FIG. 1) accurately regenerates the silence signals at the receiving end. The following discipline is implemented at the transmitting end (100): if speech is detected (200), transmit (204) a packet carrying the speech; if silence is detected (200) and the last-transmitted packet carried speech (202), transmit (204) a packet carrying the silence; if silence is detected (200) and the last-transmitted packet carried silence (202), transmit nothing. The following discipline is implemented at the receiving end (101): if a packet is received during a packet time interval (300), output (302) the contents of the packet; if a packet is not received during a packet time interval (300), output (306) again the contents of the last-received packet. Thus, while only one silence-carrying packet is transmitted, the receiver regenerates silence signals for the entire silence interval.

Abstract: One or more dynamically updated measures are generated for an audio stream. Processing is performed using the measures to distinguish silent periods from non-silent periods in the audio stream and the audio stream is encoded, wherein the silent periods are encoded differently from the non-silent periods. The processing is re-initialized during the encoding of the audio stream, if certain conditions are met. In a preferred embodiment, the processing is re-initialized if either of the following two conditions is met: (1) one of the non-silent periods is longer than a duration threshold or (2) an energy measure for the silent periods of the audio stream exceeds an energy threshold level.

Abstract: A voice recognition apparatus is provided which includes a first detection circuit for receiving an electric signal corresponding to voice. The first detection circuit detects a voice termination point representing a time at which the input of the electric signal corresponding to the voice is terminated based on the electric signal. The apparatus further includes a second detection circuit for determining a speech period, the speech period being a period in which the voice is uttered within a whole period in which the voice is input, based on the electric signal. In addition, the apparatus includes a feature amount extracting circuit for producing a feature amount vector, on the basis of a part of the electric signal corresponding to the speech period. A memory is provided for storing feature amount vectors for a plurality of voice candidates which are previously generated.

Abstract: A signal detector for detecting the presence of a intermittent signal component in a signal. The signal detector receives each of the signal strength samples during a corresponding iteration, and compares a threshold value with the received signal sample. The signal detector sets a counter to a pre-determined number if the sample compared is greater than the threshold value. The signal detector decrements the persistence counter if a corresponding sample is not greater than the threshold value. If the persistence counter is greater than a trigger value, the detector indicates the presence of a intermittent signal component or otherwise declares the absence of a intermittent signal component. The detector may indicate the presence of a intermittent signal component by a logical value of 1 and the absence by a logical value of 0. The threshold value is composed of two components; the intermittent signal component and the background signal component.

Abstract: An input speech signal is compressed and encoded by a speech encoding unit 3 and sent to an RF transmission processing unit 4 where it is channel-encoded, modulated and transmission-processed so as to be transmitted over an antenna 11. A signal received over the antenna 11 is reception-processed, demodulated and channel-decoded so as to be expanded and decoded a speech decoding unit 6. A recording/playback control unit 8 controls writing of a signal from the speech encoding unit 3 to a semiconductor memory 7 or readout of a signal from the semiconductor memory 7 to the speech encoding unit 3. This enables the semiconductor memory 7 to be efficiently utilized in single/dual communication without increasing the circuit construction.

Abstract: A method to detect a presence of a hangover period in a speech decoder in a communication system that utilizes discontinuous transmission between a transmitter and a receiver. The discontinuous transmission is defined by temporally sequential frame periods of which some periods will contain a transmitted frame and some periods will contain no transmission. The discontinuous transmission is constituted by information transmission periods of at least one frame that contain information given by a user to the transmitter, and silence periods having a length of at least one frame period and containing other information than that given by the user. An irregularly occurring period (T) of at least one frame exists between an information transmission period and the following silence period, the irregularly occurring period forming a hangover period for determining the information relating to the silence period.

Abstract: What is disclosed is a signal processing system, wherein a method and apparatus identify background noise in a signal containing speech and noise by separating the signal into frames, evaluating energy levels of selected frames, and identifying the frames as noise if certain consistency tests are met, and speech if certain pulsing, monotone, and speech level tests are met, and transition between speech and noise if transition deviation and consistency levels are met.

Abstract: A voice encoder which pauses outputting codewords in accordance with the absence of voice activity. An input aural signal is divided into frames and inputted to the voice encoder. The voice encoder has a voice activity detection circuit for determining at each frame whether voice activity is absent or present, a voice encoding circuit, a background noise update judging circuit for detecting a change in the characteristics of the input aural signal, and a control circuit. If the absence of voice activity is detected, the control circuit causes the frame at that time to be encoded as a background noise frame, and then pauses the operation of the voice encoding circuit. If the presence of voice activity is detected, the operation of the voice encoding circuit is resumed.

Abstract: A audio reproducing apparatus includes an audio decoder and a voice speed converting unit. The audio decoder decodes an audio data stream to produce an audio signal. The voice speed converting unit converts the audio signal in such a manner that when a bit rate is higher than a normal bit rate, a pitch of a reproduced sound interval is the same as the pitch of the sound interval in a normal playback mode and a voice speed in the reproduced sound interval approaches a voice speed in a sound interval in the normal playback mode. The voice speed converting unit further performs voice speed conversion on the audio signal in such a manner that when the bit rate is lower than the normal bit rate, the sound interval is not noticeably interrupted.

Abstract: Components include: an audio input, an audio output, a signal processor, a buffer memory, an interface, a command input, a display, and a controller. When a record command is input, the controller detects a non-sound section of the audio signal input, generates time data corresponding to the non-sound section, forms a data transmission block using the time data and sound data, records the result into an IC memory card, and registers a list of the newly recorded content at a content table. When a reproduction command is input, the controller reads the start/end address of a selected musical piece with reference to the TOC which is loaded into the buffer memory from the IC memory card, accesses the corresponding audio data from the IC memory card, and controls the system according to a reproduction operation. Silence sections are only stored at the beginning of a musical piece.

Abstract: In a speech decoding apparatus, a conversion unit converts a received encoded signal into a parameter in units of frames. A memory repeatedly updates and stores the parameter representing a pause state and output from the conversion unit for the pause interval of the speech signal. A synthesis filter coefficient generation unit generates a synthesis filter coefficient on the basis of the parameter read out from the memory. A smoothed filter coefficient generation unit generates a smoothed filter coefficient on the basis of the synthesis filter coefficient output from the synthesis filter coefficient generation unit. The smoothed filter coefficient generation unit generates the smoothed filter coefficient which is smoothed such that the synthesis filter coefficient changes in accordance with a count value of the frames during the predetermined period. A background noise generation unit generates background noise on the basis of the parameter read out from the memory for the pause interval of the speech signal.

Abstract: In this invention, in quantizing time region samples or frequency region samples of an input signal, an audio signal, etc., an approach is employed to allow a MDCT coefficient as the frequency region sample to undergo block floating processing. Thereafter, the processed MDCT coefficient is quantized to generate a first quantized value which is utilized calculate a quantization error between the first quantized value and the MDCT coefficient each quantization error is individually generated using one sample. The quantization error is then utilized to generate a second quantized value to thereby decompose the MDCT coefficient into at least two words. Thus, in accordance with this invention, degradation of sound quality can be minimized, and backward compatibility can be ensured.