G10L19/06

Methods, encoder and decoder for linear predictive encoding and decoding of sound signals upon transition between frames having different sampling rates
09852741 · 2017-12-26 · ·

Methods, an encoder and a decoder are configured for transition between frames with different internal sampling rates. Linear predictive (LP) filter parameters are converted from a sampling rate S1 to a sampling rate S2. A power spectrum of a LP synthesis filter is computed, at the sampling rate S1, using the LP filter parameters. The power spectrum of the LP synthesis filter is modified to convert it from the sampling rate S1 to the sampling rate S2. The modified power spectrum of the LP synthesis filter is inverse transformed to determine autocorrelations of the LP synthesis filter at the sampling rate S2. The autocorrelations are used to compute the LP filter parameters at the sampling rate S2.

Method and apparatus for processing speech signal

An apparatus for processing a speech signal is provided. The apparatus includes a communicator comprising communication circuitry configured to transmit and receive data, an actuator comprising actuation circuitry configured to generate vibration and to output a signal, a formant enhancement filter configured to increase a formant of the speech signal, and a controller comprising processing circuitry configured to control the speech signal to be received through the communicator, to estimate at least one formant frequency from the speech signal based on linear predictive coding (LPC), to estimate a bandwidth of the at least one formant frequency, to determine whether the speech signal is a voiced sound or a voiceless sound, to configure the formant enhancement filter based on the at least one formant frequency, the bandwidth of the at least one formant frequency, characteristics of the determined voiced sound or voiceless sound, and signal delivery characteristics of a human body, to apply the formant enhancement filter to the speech signal, and to control the speech signal to which the formant enhancement filter is applied to be output using the actuator through the human body.

Method and apparatus for processing speech signal

An apparatus for processing a speech signal is provided. The apparatus includes a communicator comprising communication circuitry configured to transmit and receive data, an actuator comprising actuation circuitry configured to generate vibration and to output a signal, a formant enhancement filter configured to increase a formant of the speech signal, and a controller comprising processing circuitry configured to control the speech signal to be received through the communicator, to estimate at least one formant frequency from the speech signal based on linear predictive coding (LPC), to estimate a bandwidth of the at least one formant frequency, to determine whether the speech signal is a voiced sound or a voiceless sound, to configure the formant enhancement filter based on the at least one formant frequency, the bandwidth of the at least one formant frequency, characteristics of the determined voiced sound or voiceless sound, and signal delivery characteristics of a human body, to apply the formant enhancement filter to the speech signal, and to control the speech signal to which the formant enhancement filter is applied to be output using the actuator through the human body.

Systems and methods for mitigating potential frame instability

A method for mitigating potential frame instability by an electronic device is described. The method includes obtaining a frame subsequent in time to an erased frame. The method also includes determining whether the frame is potentially unstable. The method further includes applying a substitute weighting value to generate a stable frame parameter if the frame is potentially unstable.

Systems and methods for mitigating potential frame instability

A method for mitigating potential frame instability by an electronic device is described. The method includes obtaining a frame subsequent in time to an erased frame. The method also includes determining whether the frame is potentially unstable. The method further includes applying a substitute weighting value to generate a stable frame parameter if the frame is potentially unstable.

Method for siren detection based on audio samples
09842602 · 2017-12-12 · ·

The present disclosure provides methods and apparatuses that enable an apparatus to identify sounds from short samples of audio. The apparatus may capture an audio sample and create several audio signals of different lengths, each containing audio from the captured audio sample. The apparatus my process the several audio signals in an attempt to identify features of the audio signal that indicate an identification of the captured sound. Because shorter audio samples can be analyzed more quickly, the system may first process the shortest audio samples in order to quickly identify features of the audio signal. Because longer audio samples contain more information, the system may be able to more accurately identify features in the audio signal in longer audio samples. However, analyzing longer audio signals takes more buffered audio than identifying features in shorter signals. Therefore, the present system attempts to identify features in the shortest audio signals first.

Method for siren detection based on audio samples
09842602 · 2017-12-12 · ·

The present disclosure provides methods and apparatuses that enable an apparatus to identify sounds from short samples of audio. The apparatus may capture an audio sample and create several audio signals of different lengths, each containing audio from the captured audio sample. The apparatus my process the several audio signals in an attempt to identify features of the audio signal that indicate an identification of the captured sound. Because shorter audio samples can be analyzed more quickly, the system may first process the shortest audio samples in order to quickly identify features of the audio signal. Because longer audio samples contain more information, the system may be able to more accurately identify features in the audio signal in longer audio samples. However, analyzing longer audio signals takes more buffered audio than identifying features in shorter signals. Therefore, the present system attempts to identify features in the shortest audio signals first.

Audio-visual dialogue system and method

The present invention provides an audio-visual dialogue system that allows a user to create an ‘avatar’ which may be customised to look and sound a particular way. The avatar may be created to resemble, for example, a person, animal or mythical creature, and generated to have a variable voice which may be female or male. The system then employs a real-time voice conversion in order to transform any audio input, for example, spoken word, into a target voice that is selected and customised by the user. The system is arranged to facially animate the avatar using a real-time lip-synching algorithm such that the generated avatar and the target voice are synchronised.

Audio-visual dialogue system and method

The present invention provides an audio-visual dialogue system that allows a user to create an ‘avatar’ which may be customised to look and sound a particular way. The avatar may be created to resemble, for example, a person, animal or mythical creature, and generated to have a variable voice which may be female or male. The system then employs a real-time voice conversion in order to transform any audio input, for example, spoken word, into a target voice that is selected and customised by the user. The system is arranged to facially animate the avatar using a real-time lip-synching algorithm such that the generated avatar and the target voice are synchronised.

Signal decorrelation in an audio processing system

Audio processing methods may involve receiving audio data corresponding to a plurality of audio channels. The audio data may include a frequency domain representation corresponding to filterbank coefficients of an audio encoding or processing system. A decorrelation process may be performed with the same filterbank coefficients used by the audio encoding or processing system. The decorrelation process may be performed without converting coefficients of the frequency domain representation to another frequency domain or time domain representation. The decorrelation process may involve selective or signal-adaptive decorrelation of specific channels and/or specific frequency bands. The decorrelation process may involve applying a decorrelation filter to a portion of the received audio data to produce filtered audio data. The decorrelation process may involve using a non-hierarchal mixer to combine a direct portion of the received audio data with the filtered audio data according to spatial parameters.