Abstract: A method and device for audio steering from a loudspeaker line array of a display device toward a user direction is disclosed. Data corresponding to a viewer gesture is obtained from at least one sensor of a display device. A distance and an angle between the viewer and a plurality of loudspeakers coupled to the display is determined based on the obtained data. Phase shifting is applied to an audio signal powering the plurality of loudspeakers based on the determined distance and angle to audio steer toward the user direction.

Abstract: Method for modifying a style of an audio object, and corresponding electronic device, computer readable program products and computer readable storage medium The disclosure relates to a method for processing an input audio signal. According to an embodiment, the method includes obtaining a base audio signal being a copy of the input audio signal and generating an output audio signal from the base signal, the output audio signal having style features obtained by modifying the base signal so that a distance between base style features representative of a style of the base signal and a reference style feature decreases. The disclosure also relates to corresponding electronic device, computer readable program product and computer readable storage medium.

Abstract: Method for modifying a style of an audio object, and corresponding electronic device, computer readable program products and computer readable storage medium The disclosure relates to a method for processing an input audio signal. According to an embodiment, the method includes obtaining a base audio signal being a copy of the input audio signal and generating an output audio signal from the base signal, the output audio signal having style features obtained by modifying the base signal so that a distance between base style features representative of a style of the base signal and a reference style feature decreases. The disclosure also relates to corresponding electronic device, computer readable program product and computer readable storage medium.

Abstract: A method and a system (20) of audio source separation are described. The method comprises: receiving (10) an audio mixture and at least one text query associated to the audio mixture; retrieving (11) at least one audio sample from an auxiliary audio database; evaluating (12) the retrieved audio samples; and separating (13) the audio mixture into a plurality of audio sources using the audio samples. The corresponding system (20) comprises a receiving (21) and a processor (22) configured to implement the method.

Abstract: To represent and recover the constituent sources present in an audio mixture, informed source separation techniques are used. In particular, a universal spectral model (USM) is used to obtain a sparse time activation matrix for an individual audio source in the audio mixture. The indices of non-zero groups in the time activation matrix are encoded as the side information into a bitstream. The non-zero coefficients of the time activation matrix may also be encoded into the bitstream. At the decoder side, when the coefficients of the time activation matrix are included in the bitstream, the matrix can be decoded from the bitstream. Otherwise, the time activation matrix can be estimated from the audio mixture, the non-zero indices included in the bitstream, and the USM model. Given the time activation matrix, the constituent audio sources can be recovered based on the audio mixture and the USM model.

Abstract: A method and a system of audio retrieval and source separation are described. The method comprises the steps of: receiving a textual query; retrieving a preliminary audio sample from an auxiliary audio database; retrieving a target audio sample from a target audio database; and separating the retrieved target audio sample into a plurality of audio source signals. The corresponding system comprises an input unit, a storing unit and a processing unit to implement the method.

Abstract: A method for processing an input signal having an audio component is described. The method includes obtaining a set of time parameters from a time frequency transformation of the audio component of the input signal, the audio component being a mixture of audio signals comprising at least one first audio signal of a first audio source; determining at least one motion feature of the first audio source from a visual sequence corresponding to the first audio signal; obtaining a weight vector of the set of time parameters based on the motion feature; and determining a time frequency transformation of the first audio signal based on the weight vector.

Abstract: A solution for delivery of audiovisual content to a receiver device is provided. At the transmitter side, a transmission buffer is constituted, while offering fast channel change and fast trick modes at the receiver side. At least one GoP, starting with a first I-frame is sought in the content that is to be transmitted. The timing references of the data in the at least one GoP that is prepared for delivery to a receiver device are modified so that the data is decoded by the receiver at a slowed-down rate for a given duration. This creates a lag between reading of data in by the transmitter and decoding of data by the receiver. The lag is used by the transmitter to fill the transmission buffer, while the receiver does not have to wait for the transmission buffer to be filled to start decoding.

Abstract: A method for performing audio inpainting, wherein missing portions in an input audio signal are recovered and a recovered audio signal is obtained, comprises computing a Short-Time Fourier Transform (STFT) on portions of the input audio signal, computing conditional expectations of the source power spectra of the input audio signal, wherein estimated source power spectra P(f, n, j) are obtained and wherein the variance tensor V and complex Short-Time Fourier Transform (STFT) coefficients of the input audio signals are used, iteratively re-calculating the variance tensor V from the estimated power spectra P(f, n, j) and re-calculating updated estimated power spectra P(f, n, j), computing an array of STFT coefficients ? from the resulting variance tensor V according to ?(f, n, j)=E{S(f, n, j)|x, Is, IL, V}, and converting the array of STFT coefficients ? to the time domain, wherein coefficients {tilde over (s)}1, {tilde over (s)}2, . . . , {tilde over (s)}j of the recovered audio signal are obtained.

Abstract: A particular implementation determines parameters of a generative probabilistic model from visual descriptors extracted from at least one image. The extracted visual descriptors are quantized and encoded using the model-based arithmetic encoding to be stored or for transmission to a decoder. The model parameters are also stored to be available to a decoder, or transmitted directly to a decoder. A decoder uses the stored, or received, model parameters to reconstruct the generative probabilistic model and then to decode the visual descriptors. The visual descriptors are used for image analysis tasks, such as image retrieval or object detection. A particular implementation uses a Gaussian mixture model as a generative probabilistic model.

Abstract: A method and an apparatus for separating speech data from background data in an audio communication are suggested. The method comprises: applying a speech model to the audio communication for separating the speech data from the background data of the audio communication; and updating the speech model as a function of the speech data and the background data during the audio communication.

Abstract: A method and apparatus for processing audio content is described. The method and apparatus include receiving (510) audio content, the audio content including an input audio signal, a first reference audio signal, and a second reference audio signal, determining (550) a processing function for the input audio signal, the processing function determined based on a cost function between the input audio signal, the first reference audio signal and a second reference audio signal, and processing (560) the input audio signal using the determined processing function in order to produce an output audio signal.

Abstract: A method for encoding multiple audio signals comprises random sampling and quantizing each of the multiple audio signals, and encoding the sampled and quantized multiple audio signals as side information that can be used for decoding and separating the multiple audio signals from a mixture of said multiple audio signals. A method for decoding a mixture of multiple audio signals comprises decoding and demultiplexing side information, the side information comprising quantized samples of each of the multiple audio signals, receiving or retrieving from any data source a mixture of said multiple audio signals, and generating multiple estimated audio signals that approximate said multiple audio signals, wherein said quantized samples of each of the multiple audio signals are used.

Abstract: A method is proposed for encoding at least two signals. The method includes mixing the at least two signals in a mixture; sampling a map Z representative of locations of the at least two signals in a time-frequency plane at sampling locations, the sampling delivering a first list of values Z?; and transmitting the mixture of the at least two signals and information representative of the first list of values Z?. The disclosure also relates to the corresponding method for separating signals in a mixture, and corresponding computer program products, devices and bitstream.

Abstract: An apparatus and method for generating visual content from an audio signal are described. The method includes receiving (310) audio content, processing (320) the audio content to separate into a first and second portion of the audio content, converting (330) the second portion into visual content, delaying (340) the first portion based on a time relationship between the audio content and the visual content, the delaying accounting for time to process the first portion and convert the second portion, and providing (350) the visual content and audio content for reproduction. The apparatus includes a source separation module (210) processing the received audio content to separate into a first and second portion of the audio content, a converter module (220) converting the second portion into visual content, and a synchronization module (230) delaying the first portion based on a time relationship between the audio content and the visual content.

Abstract: A method and an apparatus for separating speech data from background data in an audio communication are suggested. The method comprises: applying a speech model to the audio communication for separating the speech data from the background data of the audio communication; and updating the speech model as a function of the speech data and the background data during the audio communication.

Abstract: Separation of speech and background from an audio mixture by using a speech example, generated from a source associated with a speech component in the audio mixture, to guide the separation process.

Abstract: A method and apparatus for processing audio content is described. The method and apparatus include receiving (510) audio content, the audio content including an input audio signal, a first reference audio signal, and a second reference audio signal, determining (550) a processing function for the input audio signal, the processing function determined based on a cost function between the input audio signal, the first reference audio signal and a second reference audio signal, and processing (560) the input audio signal using the determined processing function in order to produce an output audio signal.

Abstract: A method and a system (20) of audio source separation are described. The method comprises: receiving (10) an audio mixture and at least one text query associated to the audio mixture; retrieving (11) at least one audio sample from an auxiliary audio database; evaluating (12) the retrieved audio samples; and separating (13) the audio mixture into a plurality of audio sources using the audio samples. The corresponding system (20) comprises a receiving (21) and a processor (22) configured to implement the method.

Abstract: Isolation of an active participant in a group of participants commences by first capturing images and audio of participants. Thereafter, an active one of the participants in the group of participants (e.g., a participant that is currently speaking) is identified. After identification of the active participant, at least one of participants' images and participants' audio are rendered to isolate the active participant.