Abstract: In the acoustic object extraction device, beam forming processing units generate a first acoustic signal by beam forming in an arrival direction of a signal from an acoustic object with respect to a microphone array and generate a second acoustic signal by beam forming in an arrival direction of a signal from the acoustic object with respect to a microphone array, and a common component extraction unit extracts, on the basis of a similarity between the spectrum of the first acoustic signal and the spectrum of the second acoustic signal and from the first acoustic signal and the second acoustic signal, a signal containing a common component corresponding to the acoustic object. The common component extraction unit divides the spectrums of the first acoustic signal and the second acoustic signal into a plurality of frequency sections and calculates a similarity for each of the frequency sections.

Abstract: A method generates binaural headphone playback signals given multiple audio source signals with associated metadata and a binaural room impulse response (BRIR) database, where the audio source signals can be channel-based, object-based, or a mixture of both signals. The method groups the audio source signals according to positions of the audio sources, divides BRIR into blocks and frames, where the BRIR is divided into a direct block and diffuse blocks, and divides each audio source signal into blocks and frames, wherein the source signal is divided into a current block and previous blocks, and the current block is further divided into the frames. The method further averages, for each of previous frames of the source signals, the divided BRIR identified with the grouping result by downmixing the previous frames of the source signals according to the grouping result, and performs a convolution with the downmixed previous frame.

Abstract: A communication node determines a codec and a codec mode to be used by two terminals that perform communication in a first network, when one of the two terminals performs handover to a second network that is different from the first network. The communication node includes: a determiner that sets common part for a codec and a codec mode to be used by the two terminals, the common part being common among information indicating codecs and codec modes used for communication in the first network, information indicating codecs and codec modes supported by the one terminal, and information indicating codecs and codec modes supported by the second network; and a generator that generates signaling for requesting the two terminals to perform changing to the set codec and codec mode to be used by the two terminals.

Abstract: A communication method supports Enhanced Voice Services (EVS) codec performed by a communication terminal apparatus. The method includes performing negotiation to use an EVS codec for communication between a communication terminal apparatus and a counterpart terminal, using an IP multimedia subsystem (IMS) signaling including one of a session description protocol (SDP) offer and an SDP answer, and performing negotiation to specify one or more audio-bandwidths of input signals in Hertz (Hz) or kilohertz (kHz) for the EVS codec. In a communication session after the codec negotiation session, the processor causes the receiver to receive a signaling for changing an audio-bandwidth of an input signal to the EVS codec from a network node, changes the audio-bandwidth of the input signal to the EVS codec to another audio-bandwidth without changing the EVS codec based on the signaling, and causes the transmitter to transmit encoded data in the changed audio-bandwidth.

Abstract: A method of generating binaural headphone playback signals given multiple audio source signals with an associated metadata and binaural room impulse response (BRIR) database is provided, wherein the audio source signals can be channel-based, object-based, or a mixture of both signals. The method includes grouping the audio source signals according to positions of the audio sources, parameterizing BRIR to be used for rendering, and dividing each audio source signal to be rendered into a number of blocks and frames. The method also includes averaging the parameterized BRIR sequences, downmixing the divided audio source signals using the diffuse blocks of BRIRs, and performing late reverberation processing on the downmixed version of the previous blocks of the audio source signals.

Abstract: A communication terminal apparatus is provided that includes a negotiator that negotiates a codec used for communication between the communication terminal apparatus and a counterpart terminal at a start of the communication. The negotiator uses an IP multimedia subsystem (IMS) signaling including one of a session description protocol (SDP) offer and an SDP answer. The negotiated codec supports bandwidths of input signals of a plurality of codecs, and changes a bandwidth of an input signal during the communication with the counterpart terminal by changing a frequency range of the input signal. The apparatus also includes a bandwidth determiner that limits the bandwidth and an encoding bit rate of the input signal of the codec during the communication with the counterpart terminal, according to signaling for limiting the bandwidth and the encoding bit rate of the input signal of the codec, the signaling being notified from a network node.

Abstract: In an encoder (100), a signal analysis unit (101) performs signal analysis on an L channel signal and an R channel signal that constitute a stereo signal and generates a parameter used to determine a coding mode for each of an L channel and an R channel. A DMA stereo encoding unit (104) encodes the L channel signal and the R channel signal by using a coding mode common to the L channel signal and the R channel signal. At this time, the DMA stereo encoding unit (104) determines the common coding mode by selecting, out of the L channel and the R channel, the one that has a lower ratio of energy of an environmental sound component to the entire energy of the channel and using the parameter of the selected channel.

Abstract: A voice signal decoding device includes a first decoder, a second decoder, a signal switch, and a noise adder. The first decoder decodes first encoded data encoded by a first encoding method. The second decoder decodes second encoded data encoded by a second encoding method. The second encoded data has a narrower band than a band of the first encoded data. The signal switch switches an output signal of the first decoder and an output signal of the second decoder. The noise adder adds a noise signal to a high-frequency band in the output signal of the second decoder when the signal switch switches an output signal from the output signal of the first decoder to the output signal of the second decoder. The high-frequency band is a band where a signal component is lacking as compared with the output signal of the first decoder.

Abstract: A method of generating binaural headphone playback signals given multiple audio source signals with an associated metadata and binaural room impulse response (BRIR) database is provided, wherein the audio source signals can be channel-based, object-based, or a mixture of both signals. The method includes grouping the audio source signals according to positions of the audio sources, parameterizing BRIR to be used for rendering, and dividing each audio source signal to be rendered into a number of blocks and frames. The method also includes averaging the parameterized BRIR sequences, downmixing the divided audio source signals using the diffuse blocks of BRIRs, and performing late reverberation processing on the downmixed version of the previous blocks of the audio source signals.

Abstract: A sound source estimation unit (101) estimates, in a space as a target of sparse sound field decomposition, an area where a sound source is present at second granularity that is coarser than first granularity of a position where a sound source is assumed to be present in the sparse sound field decomposition. A sparse sound field decomposition unit (102) decomposes an acoustic signal observed by a microphone array into a sound source signal and an ambient noise signal by performing a sparse sound field decomposition process at the first granularity for the acoustic signal in the area at the second granularity where the sound source is estimated to be present in the space.

Abstract: A sound source estimation unit (101) estimates, in a space as a target of sparse sound field decomposition, an area where a sound source is present at second granularity that is coarser than first granularity of a position where a sound source is assumed to be present in the sparse sound field decomposition. A sparse sound field decomposition unit (102) decomposes an acoustic signal observed by a microphone array into a sound source signal and an ambient noise signal by performing a sparse sound field decomposition process at the first granularity for the acoustic signal in the area at the second granularity where the sound source is estimated to be present in the space.

Abstract: A method of generating binaural headphone playback signals given multiple audio source signals with an associated metadata and binaural room impulse response (BRIR) database, wherein the multiple audio source signals can be channel-based, object-based, or a mixture of both signals. The method includes grouping the multiple audio source signals according to positions of the audio sources in a hierarchical manner, and parameterizing BRIR to be used for rendering. The method also includes dividing each audio source signal to be rendered into a number of blocks and frames, averaging the parameterized BRIR sequences identified with a hierarchically grouping result, and downmixing the divided audio source signals identified with the hierarchically grouping result.

Abstract: An audio signal coding apparatus includes a time-frequency transformer that outputs sub-band spectra from an input signal; a sub-band energy quantizer; a tonality calculator that analyzes tonality of the sub-band spectra; a bit allocator that selects a second sub-band on which quantization is performed by a second quantizer on the basis of the analysis result of the tonality and quantized sub-band energy, and determines a first number of bits to be allocated to a first sub-band on which quantization is performed by a first quantizer; the first quantizer that performs first coding using the first number of bits; the second quantizer that performs coding using a second coding method; and a multiplexer.

Abstract: A decoding device includes: a separating unit separating first encoded data, a spectrum including a low-band spectrum of audio signals having been encoded, and second encoded data, a high-band spectrum of a higher band having been encoded, based on the first encoded data; a first decoding unit decoding the first encoded data and generating a first decoded spectrum; a first amplitude normalizer dividing amplitude of the first decoded spectrum into sub-bands, normalizing the spectrum of each sub-band by the largest amplitude of the first decoded spectrum within each sub-band, and generating a normalized spectrum; an addition unit adding noise spectrum to the normalized spectrum and generating a noise-added normalized spectrum; a second decoding unit decoding the second encoded data using the noise-added normalized spectrum, and generating a second noise-added spectrum; and a converter performing time-frequency conversion regarding a spectrum coupled based on the first decoded spectrum and second noise-added spe

Abstract: An inter-channel correlation calculation unit (102) calculates an inter-channel correlation between a left channel and a right channel by using a left channel signal and a right channel signal that constitute a stereo signal. A DMA stereo encoding unit (104) and a DM stereo encoding unit (105) encode the left channel signal and the right channel signal by using a common coding mode if the inter-channel correlation is greater than a threshold value and individually encode the left channel signal and the right channel signal by using a coding mode determined for each of the left channel signal and the right channel signal if the inter-channel correlation is less than or equal to the threshold value.

Abstract: A decoding device includes: a separating unit separating first encoded data, a spectrum including a low-band spectrum of audio signals having been encoded, and second encoded data, a high-band spectrum of a higher band having been encoded, based on the first encoded data; a first decoding unit decoding the first encoded data and generating a first decoded spectrum; a first amplitude normalizer dividing amplitude of the first decoded spectrum into sub-bands, normalizing the spectrum of each sub-band by the largest amplitude of the first decoded spectrum within each sub-band, and generating a normalized spectrum; an addition unit adding noise spectrum to the normalized spectrum and generating a noise-added normalized spectrum; a second decoding unit decoding the second encoded data using the noise-added normalized spectrum, and generating a second noise-added spectrum; and a converter performing time-frequency conversion regarding a spectrum coupled based on the first decoded spectrum and second noise-added spe

Abstract: A voice signal decoding device includes a first decoder, a second decoder, a signal switch, and a noise adder. The first decoder decodes first encoded data encoded by a first encoding method. The second decoder decodes second encoded data encoded by a second encoding method. The second encoded data has a narrower band than a band of the first encoded data. The signal switch switches an output signal of t first decoder and an output signal of the second decoder. The noise adder adds a noise signal to a high-frequency band in the output signal of t second decoder when the signal switch switches an output signal from the output signal of the first decoder to the output signal of the second decoder. The high-frequency band is a band where a signal component is lacking as compared with the output signal of the first decoder.

Abstract: A communication terminal and method are provided, wherein a processor executes instructions to configure a codec mode or a codec mode set for a communication terminal. The configuration is based on Session Description Protocol (SDP) offer/answer messages between the communication terminal and a partner terminal in a codec negotiation session when communication starts between the communication terminal and the partner terminal, when the SDP offer/answer messages include a codec, and when a codec mode or a codec mode set is included. The processor also configures a Real-time Transport Protocol (RTP) payload format header based on the SDP offer/answer messages in the codec negotiation session, a CMR being a request for the partner terminal to change the codec mode. The processor further causes the transmitter to transmit, after the codec negotiation session, a codec mode of the communication terminal currently in use, using an RTP payload.

Abstract: An audio signal coding apparatus includes a time-frequency transformer that outputs sub-band spectra from an input signal; a sub-band energy quantizer; a tonality calculator that analyzes tonality of the sub-band spectra; a bit allocator that selects a second sub-band on which quantization is performed by a second quantizer on the basis of the analysis result of the tonality and quantized sub-band energy, and determines a first number of bits to be allocated to a first sub-band on which quantization is performed by a first quantizer; the first quantizer that performs first coding using the first number of bits; the second quantizer that performs coding using a second coding method; and a multiplexer.

Abstract: A method of generating binaural headphone playback signals given multiple audio source signals with an associated metadata and binaural room impulse response (BRIR) database, wherein the multiple audio source signals can be channel-based, object-based, or a mixture of both signals. The method includes grouping the multiple audio source signals according to positions of the audio sources in a hierarchical manner, and parameterizing BRIR to be used for rendering. The method also includes dividing each audio source signal to be rendered into a number of blocks and frames, averaging the parameterized BRIR sequences identified with a hierarchically grouping result, and downmixing the divided audio source signals identified with the hierarchically grouping result.