Abstract: A voice quality measurement device that measures voice quality of a decoded voice signal outputted from a voice decoder unit. The voice quality measurement device includes a packet buffer unit and a voice information monitoring unit. The packet buffer unit accumulates voice packets that arrive non-periodically as voice information, and outputs the voice information to the voice decoder unit periodically. The voice information monitoring unit monitors continuity of the voice information inputted to the voice decoder unit, and calculates an index of voice quality of the decoded voice signal that reflects acceptability of this continuity.

Abstract: There are provided with an IP telephone system having both convenience of the softphone and durability of the hardphone, and a speech processing peripheral device (10) for the IP telephone system. The speech processing peripheral device as a peripheral device of an information processing device (20) is connected to the information processing device (20) for executing a call control program and functioning as a call controller. This speech processing peripheral device (10) includes a host connection means for sending/receiving an audio signal to/from the information processing device (20) and a speech processing means for executing speech processing in IP telephone communication. The IP telephone system according to the present invention includes the information processing device (20) for executing the call control program and functioning as the call controller and the speech processing peripheral device (10) according to the present invention, connected to the information processing device (20).

Abstract: Data received by a receiver is processed at a sampling rate indicated manually or automatically in the receiver. The sampling rate of the received data is controlled in accordance with the processing rate. The sampling rate controlled data is then processed so as to convert its frequency distribution to that the received data originally had.

Abstract: An apparatus estimates the direction of a sound source from signals plural microphones capture sound to produce. Data are stored on reverse characteristics of spatial transfer functions defined on sound transmitted from sound source positions to the respective microphones. To the signal produced by each microphone, applied are the reverse characteristics of the spatial transfer functions thus stored in connection with that microphone with respect to the sound source positions to thereby estimate a sound source signal on a sound source position associated with the sound captured. Between the sound source signals estimated on the sound source positions associated with the sounds captured by the microphones, coincidence or higher correlation is found on a sound source position to thereby produce information on at least the direction of the sound source thus found.

Abstract: A voice mixing apparatus decodes input encoded narrowband voice data and encoded voice data for narrowband region of input encoded wideband voice data, and detects a speaker in accordance with the decoded voice signals of the entire narrowband. When encoded voice data from a speaker is included in the narrowband, a signal in a region outside the narrowband of the expanded data is encoded. When the data is included in the wideband, encoded voice data of the region outside the narrowband is extracted for output. When the destination terminal is compatible with the encoded narrowband voice data, the narrowband voice signal mixed is encoded and output. When the destination terminal is compatible with wideband, the narrowband voice signal mixed is encoded for the narrowband region, and the voice data of the speaker is used as the encoded voice data for the region outside the narrowband.

Abstract: A bandwidth extension apparatus can generate from an inputted speech signal a bandwidth-extended signal whose bandwidth is automatically adjusted according to the surrounding hearing situation by means of a parameter. The apparatus has a surrounding characteristic predictor for predicting the characteristic amount of the surrounding sound of the phone terminal on which the apparatus is installed; an adjustment amount predictor for comparing the characteristic amount with the parameter to predict the adjustment amount for the parameter; a speech characteristic predictor for predicting the characteristic amount of the inputted speech signal; and an adjustment amount determiner for determining an adjustment amount for updating the parameter based on the characteristic amount of the surrounding sound and the inputted speech signal.

Abstract: Voice communications apparatus is connected to a general subscriber telephone set or a broadband telephone set, to communicate over the public switched telephone network. The apparatus includes a filter converting signals such as to satisfy signal conditions prescribed for the telephone network. The apparatus also includes a terminal class determiner for determining the class of the telephone set connected, and a circuit for changing at least the sampling frequency at which the analog signals from the telephone set are sampled. This establishes high quality in broadband voice communications. The terminal class determiner may automatically determine the class of the telephone set at any timing of a call sequence by detecting a band component of the signals from the telephone set, a predetermined frequency of a signal intermittently transmitted from the telephone set or the characteristics of the telephone set.

Abstract: A band-limited voice signal is processed to reduce its spectral envelope or harmonic structure, or both. The resulting reduced signal is moved into a frequency band above the upper limit frequency of the band-limited voice signal, and then combined with the band-limited voice signal to form a band expanded signal with improved quality and comprehensibility, free of unnatural high-frequency resonances and unnaturally strong high-frequency harmonics.

Abstract: A voice quality measurement device that measures voice quality of a decoded voice signal outputted from a voice decoder unit. The voice quality measurement device includes a packet buffer unit and a voice information monitoring unit. The packet buffer unit accumulates voice packets that arrive non-periodically as voice information, and outputs the voice information to the voice decoder unit periodically. The voice information monitoring unit monitors continuity of the voice information inputted to the voice decoder unit, and calculates an index of voice quality of the decoded voice signal that reflects acceptability of this continuity.

Abstract: There are provided: a means (101a) for storing standard head related transfer functions for reference positions from a virtual listener; a means (101) for, when given information (DIR, DIST) about a virtual sound source position, forming head related transfer functions (hL(k), hR(k)) as left ear and right ear head related transfer functions by selecting one of the stored standard head related transfer functions or by selecting two or more of them and interpolating; means (102, 103) for imprinting a sense of direction and distance on the audio listening signal by using the head related transfer functions thus obtained; and means (104, 105) for correcting a distance related to the obtained head related transfer functions and the sense of distance to the virtual sound source position, in the audio listening signals (sL(n), sR(n)) given the sense of direction and distance or the source audio listening signal (s(n)). A highly precise sense of distance can be provided in a small structure.

Abstract: An echo canceller which can respond to a sudden change in echo characteristics in real time and does not require an alteration outside the echo canceller, includes a smoothed sending-speech signal calculation means for calculating a smoothed sending-speech signal from the sending-speech signal, the smoothed sending-speech signal being obtained by smoothing the sending-speech signal; a smoothed receiving-speech signal calculation means for calculating a smoothed receiving-speech signal from the receiving-speech signal, the smoothed receiving-speech signal being obtained by smoothing the receiving-speech signal; a delay time information generation means for obtaining delay time information reflecting delay characteristics of an echo path in accordance with a correlation between the smoothed sending-speech signal and the smoothed receiving-speech signal; and an update information generation means for obtaining update information indicating execution or suspension of updating of the tap coefficients of the adapti

Abstract: A speech band extension device (100), which generates an audio signal capable of realizing natural audibility after speech band extension, includes a band-extended audio generator which generates a band-extended audio signal from an original audio signal, the band-extended audio signal including components lying within a frequency band that is not included in a frequency band of the original audio signal, and an adjustment adder (20) which detects a timing shift between the original audio signal and the band-extended audio signal, adjusts timing of the original audio signal and timing of the band-extended audio signal in accordance with the detected timing shift, and combines the both signals after the adjusting of the timing, wherein the detection of the timing shift is performed, for example, using zero-crossing and cross-correlation.

Abstract: Voice communications apparatus intervenes between a telephone set and a broad-band transmission path that has a communication band broader than a voice band limited by the transmitter and receiver characteristics of the telephone set. The voice communications apparatus includes a transmitter frequency characteristic corrector circuit for correcting the frequency characteristic of a signal input from the telephone set, and a receiver frequency characteristic corrector circuit for correcting the frequency characteristic of a signal to be input to the telephone set.

Abstract: A voice packet communications system which objectively evaluates the voice quality in real time. Voice packets received from a network are input into an adjustment circuit. The adjustment circuit evens out the periods of the voice packets, detects the drop-out of voice packets, and inserts substitute packets in the voice packet sequence. Furthermore, the adjustment circuit outputs the number of voice packets that have dropped out, the number of substitute packets that have been inserted, and the number of voice packets that have accumulated inside the circuit, as quality information in each monitoring period. A decoding circuit converts the voice packets adjusted by the adjustment circuit into voice information. A judgement circuit calculates an index that indicates the communications quality of the voice packets from the quality information using a specified calculation formula.

Abstract: An apparatus estimates the direction of a sound source from signals plural microphones capture sound to produce. Data are stored on reverse characteristics of spatial transfer functions defined on sound transmitted from sound source positions to the respective microphones. To the signal produced by each microphone, applied are the reverse characteristics of the spatial transfer functions thus stored in connection with that microphone with respect to the sound source positions to thereby estimate a sound source signal on a sound source position associated with the sound captured. Between the sound source signals estimated on the sound source positions associated with the sounds captured by the microphones, coincidence or higher correlation is found on a sound source position to thereby produce information on at least the direction of the sound source thus found.

Abstract: A voice communication apparatus is interposed between a telephone set and a public switched telephone network (PSTN) or a broadband network which can provide a frequency bandwidth broader than a frequency bandwidth of the public switched telephone network. The apparatus selects one of the PSTN and the broadband network, from which a voice signal is input. The apparatus takes account of the selected network, frequency characteristics of the selected network and of the telephone set, and appropriately adjusts frequency characteristics of a voice signal transmitted between the telephone set and the selected network by means of transmitter and receiver band-pass filter units. As a result, both the telephone set and the counterpart communication terminal can receive a voice signal of better transmission quality.

Abstract: Voice communications apparatus intervenes between a telephone set and a broad-band transmission path that has a communication band broader than a voice band limited by the transmitter and receiver characteristics of the telephone set. The voice communications apparatus includes a transmitter frequency characteristic corrector circuit for correcting the frequency characteristic of a signal input from the telephone set, and a receiver frequency characteristic corrector circuit for correcting the frequency characteristic of a signal to be input to the telephone set.

Abstract: A bandwidth extension apparatus can generate from an inputted speech signal a bandwidth-extended signal whose bandwidth is automatically adjusted according to the surrounding hearing situation by means of a parameter. The apparatus has a surrounding characteristic predictor for predicting the characteristic amount of the surrounding sound of the phone terminal on which the apparatus is installed; an adjustment amount predictor for comparing the characteristic amount with the parameter to predict the adjustment amount for the parameter; a speech characteristic predictor for predicting the characteristic amount of the inputted speech signal; and an adjustment amount determiner for determining an adjustment amount for updating the parameter based on the characteristic amount of the surrounding sound and the inputted speech signal.

Abstract: Data received by a receiver is processed at a sampling rate indicated manually or automatically in the receiver. The sampling rate of the received data is controlled in accordance with the processing rate. The sampling rate controlled data is then processed so as to convert its frequency distribution to that the received data originally had.

Abstract: A voice mixing apparatus decodes input encoded narrowband voice data and encoded voice data for narrowband region of input encoded wideband voice data, and detects a speaker in accordance with the decoded voice signals of the entire narrowband. When encoded voice data from a speaker is included in the narrowband, a signal in a region outside the narrowband of the expanded data is encoded. When the data is included in the wideband, encoded voice data of the region outside the narrowband is extracted for output. When the destination terminal is compatible with the encoded narrowband voice data, the narrowband voice signal mixed is encoded and output. When the destination terminal is compatible with wideband, the narrowband voice signal mixed is encoded for the narrowband region, and the voice data of the speaker is used as the encoded voice data for the region outside the narrowband.