Abstract: A system for synthesising expressive speech includes: an interface configured to receive an input text for conversion to speech; a memory; and at least one processor coupled to the memory. The processor is configured to generate, using an expressivity characterisation module, a plurality of expression vectors, wherein each expression vector is a representation of prosodic information in a reference audio style file, and synthesise expressive speech from the input text, using an expressive acoustic model comprising a deep convolutional neural network that is conditioned by at least one of the plurality of expression vectors.

Abstract: Provided are a signal processing method and apparatus for enhancing sound quality. The signal processing method performed by a signal transmitting apparatus includes determining, based on a plurality of parameters, a valid bandwidth so as to encode an input signal; performing pre-processing on the input signal, based on the valid bandwidth; and encoding the pre-processed input signal, based on the valid bandwidth, and the signal processing method performed by a signal receiving apparatus includes decoding a bitstream or a packet received via a transmission channel; determining a valid bandwidth, based on a plurality of parameters used in the decoding; and performing post-processing on a decoded signal, based on the valid bandwidth.

Abstract: A system for synthesising expressive speech includes: an interface configured to receive an input text for conversion to speech; a memory; and at least one processor coupled to the memory. The processor is configured to generate, using an expressivity characterisation module, a plurality of expression vectors, wherein each expression vector is a representation of prosodic information in a reference audio style file, and synthesise expressive speech from the input text, using an expressive acoustic model comprising a deep convolutional neural network that is conditioned by at least one of the plurality of expression vectors.

Abstract: Provided is a method for processing a signal adaptively to a noisy environment, the method including: determining a class related to enhancement of intelligibility, based on a noise spectrum of a reception side or a near-end device and a received voice spectrum; generating guide information for the enhancement of intelligibility based on the noise spectrum, the voice spectrum, and the determined class; and generating a changed voice spectrum by applying the guide information to the voice spectrum.

Abstract: A voice signal processing method according to an embodiment of the present disclosure for overcoming the problem includes: acquiring a real-time near-end noise signal; acquiring a far-end voice signal according to an incoming call; measuring subjective speech quality and perceptual-objective speech quality of test signals generated based on a reference signal and the real-time near-end noise signal; selecting at least one speech quality enhancement method based on the subjective speech quality and the perceptual-objective speech quality, and determining parameters that are to be applied to the selected at least one speech quality enhancement method; and enhancing speech quality of the far-end voice signal by using the selected at least one speech quality enhancement method, based on the determined parameters, wherein the test signals are generated by mixing the acquired real-time near-end noise signal with the reference signal whose speech quality is enhanced by applying a combination of parameter values to s

Abstract: Provided are a signal processing method and apparatus for enhancing sound quality. The signal processing method performed by a signal transmitting apparatus includes determining, based on a plurality of parameters, a valid bandwidth so as to encode an input signal; performing pre-processing on the input signal, based on the valid bandwidth; and encoding the pre-processed input signal, based on the valid bandwidth, and the signal processing method performed by a signal receiving apparatus includes decoding a bitstream or a packet received via a transmission channel; determining a valid bandwidth, based on a plurality of parameters used in the decoding; and performing post-processing on a decoded signal, based on the valid bandwidth.

Abstract: Provided are a signal processing method and apparatus for enhancing sound quality. The signal processing method performed by a signal transmitting apparatus includes determining, based on a plurality of parameters, a valid bandwidth so as to encode an input signal; performing pre-processing on the input signal, based on the valid bandwidth; and encoding the pre-processed input signal, based on the valid bandwidth, and the signal processing method performed by a signal receiving apparatus includes decoding a bitstream or a packet received via a transmission channel; determining a valid bandwidth, based on a plurality of parameters used in the decoding; and performing post-processing on a decoded signal, based on the valid bandwidth.

Abstract: A voice signal processing method according to an embodiment of the present disclosure for overcoming the problem includes: acquiring a real-time near-end noise signal; acquiring a far-end voice signal according to an incoming call; measuring subjective speech quality and perceptual-objective speech quality of test signals generated based on a reference signal and the real-time near-end noise signal; selecting at least one speech quality enhancement method based on the subjective speech quality and the perceptual-objective speech quality, and determining parameters that are to be applied to the selected at least one speech quality enhancement method; and enhancing speech quality of the far-end voice signal by using the selected at least one speech quality enhancement method, based on the determined parameters, :herein the test signals are generated by mixing the acquired real-time near-end noise signal with the reference signal whose speech quality is enhanced by applying a combination of parameter values to s

Abstract: Provided is a method for processing a signal adaptively to a noisy environment, the method including: determining a class related to enhancement of intelligibility, based on a noise spectrum of a reception side or a near-end device and a received voice spectrum; generating guide information for the enhancement of intelligibility based on the noise spectrum, the voice spectrum, and the determined class; and generating a changed voice spectrum by applying the guide information to the voice spectrum.

Abstract: There is provided a noise suppressor 2 comprising a receiver 4 operable to receive an input audio signal 14 and to produce from the input audio signal 14 a first signal 16 and a second signal 18, the input audio signal 14 comprising desired audio and transmission end noise. The noise suppressor 2 further comprises a first processor 6 operable to perform a first process on the first signal 16, the first process comprising noise suppression to remove at least a portion of the transmission end noise from the first signal 16 before outputting the first signal 16 to a first audio channel 10. The noise suppressor further comprises a second processor 8 operable to perform a second process on the second signal 18, the second process comprising outputting the second signal 18 to a second audio channel 12. The first process comprises more aggressive noise suppression than the second process.

Abstract: Provided are a signal processing method and apparatus for enhancing sound quality. The signal processing method performed by a signal transmitting apparatus includes determining, based on a plurality of parameters, a valid bandwidth so as to encode an input signal; performing pre-processing on the input signal, based on the valid bandwidth; and encoding the pre-processed input signal, based on the valid bandwidth, and the signal processing method performed by a signal receiving apparatus includes decoding a bitstream or a packet received via a transmission channel; determining a valid bandwidth, based on a plurality of parameters used in the decoding; and performing post-processing on a decoded signal, based on the valid bandwidth.

Abstract: An apparatus receives an input encoded audio data signal comprising a base layer and at least one enhancement layer. A reference unit (103) generates reference audio data corresponding to audio data of a reference set of layers. A layer unit (105) divides the layers of the input signal into a first subset and a second subset. A sample unit (107) generates sample audio data corresponding to the audio data of the first subset. A comparison unit (109) generates a difference measure by comparing the sample audio data to the reference audio data based on a perceptual model. An output unit (111) then determines if the difference measure meets a similarity criterion and generates an output signal without audio data from a layer of the second subset if the similarity criterion is met and including the audio data of the layer otherwise. The invention may provide reduced data rates without an unacceptable degradation of quality.

Abstract: An apparatus receives an input encoded audio data signal comprising a base layer and at least one enhancement layer. A reference unit (103) generates reference audio data corresponding to audio data of a reference set of layers. A layer unit (105) divides the layers of the input signal into a first subset and a second subset. A sample unit (107) generates sample audio data corresponding to the audio data of the first subset. A comparison unit (109) generates a difference measure by comparing the sample audio data to the reference audio data based on a perceptual model. An output unit (111) then determines if the difference measure meets a similarity criterion and generates an output signal without audio data from a layer of the second subset if the similarity criterion is met and including the audio data of the layer otherwise. The invention may provide reduced data rates without an unacceptable degradation of quality.