Abstract: A method in an apparatus, the method comprising receiving a set of learning targets based on a user selection; determining a user skill level by analyzing an input of a user performance of at least one musical piece; determining a difficulty level for each learning target of the set of learning targets; determining an effort parameter for each learning target of the set of learning targets based on the user skill level and/or the difficulty level of each learning target; arranging the set of learning targets based on the effort parameters; and providing a personalized curriculum based on the arranged set of learning targets.

Abstract: A method in an apparatus, the method comprising maintaining a repertoire of musical pieces at a user profile; receiving a user performance of a musical piece; analyzing the user performance of the musical piece; determining based on analyzing the performance whether the user performance of the musical piece fulfills at least one predetermined criterion; and based on the user performance of the musical piece fulfilling the at least one predetermined criterion, storing the musical piece into the repertoire.

Abstract: A method and apparatus for presenting at least a portion of a musical notation of a starting point of a song to a user, including a first note or chord to be played; indicating a tempo of the song to the user; receiving from a musical instrument played a signal indicating how the musical instrument is being played; monitoring triggers and responsively to detecting a starting trigger beginning to playback one or more backing tracks; detecting the starting trigger by determining that the user has provided a starting sign independent of the apparatus; and synchronizing the playback of the one or more backing tracks using a time when the user has provided the starting sign.

Abstract: A computer-implemented method includes: providing backing track audio data, wherein each backing track includes information of at least: song tempo, tonal content that is synchronized with the backing track audio, selecting a song, receiving a real-time audio signal of the user's performance, estimating parameters, based on the audio signal, including at least: playing activity of the user, wherein detecting whether the user is producing any sounding notes with a musical instrument, tempo of the user's playing, and playing position of the user within the selected song, estimating the reliability of the estimated tempo and play position of the user, wherein a value of the reliability represents the probability that the amount of error in the estimated user tempo and play position is sufficiently small, and when the estimated reliability of the user position and tempo is sufficiently high, start playing the backing track at the user position and tempo.

Abstract: A user interface presents structural musical information in a score in a way where both the start and the end points of each jump in the score are visible simultaneously. Each jump is presented in a manner that allows the user to select during performance, which one of different alternatives to choose, when approaching a decision point like a repeat in the song.

Abstract: A method, apparatus and computer program product are provided to cause AN audio source to be modified in a manner consistent with the corresponding video images once the user and/or a display upon which the images are rendered has been tilted. In regards to a method, an initial virtual position of an audio source is determined. The method also includes determining a tilt angle that defines an angle that an apparatus embodying a display for rendering images has been tilted relative to a reference orientation of the apparatus with respect to a user of the apparatus. The method may also include modifying a virtual position of an audio source based upon the tilt angle and an initial virtual position of the audio source. The method may also include causing the audio source to be rendered in accordance with the virtual position as modified.

Abstract: A method, apparatus and computer program product are provided to cause AN audio source to be modified in a manner consistent with the corresponding video images once the user and/or a display upon which the images are rendered has been tilted. In regards to a method, an initial virtual position of an audio source is determined. The method also includes determining a tilt angle that defines an angle that an apparatus embodying a display for rendering images has been tilted relative to a reference orientation of the apparatus with respect to a user of the apparatus. The method may also include modifying a virtual position of an audio source based upon the tilt angle and an initial virtual position of the audio source. The method may also include causing the audio source to be rendered in accordance with the virtual position as modified.

Abstract: A method, apparatus and computer program product are provided to cause AN audio source to be modified in a manner consistent with the corresponding video images once the user and/or a display upon which the images are rendered has been tilted. In regards to a method, an initial virtual position of an audio source is determined. The method also includes determining a tilt angle that defines an angle that an apparatus embodying a display for rendering images has been tilted relative to a reference orientation of the apparatus with respect to a user of the apparatus. The method may also include modifying a virtual position of an audio source based upon the tilt angle and an initial virtual position of the audio source. The method may also include causing the audio source to be rendered in accordance with the virtual position as modified.

Abstract: This invention describes a method for adjusting the loudness and the spectral content of digital audio signals in a real-time using warped spectral filtering. A warped processing module modifies a spectral content of a digital audio signal with a set of gains for a plurality of non-linearly-scaled frequency bands determined by a warping factor ? of a warped delay line. Warped delay line signals, generated by the warped delay line, are processed by a warped filter block containing multiple warped finite impulse response filters, e.g., Mth band filters, using individual warped spectral filtering in said plurality of the non-linearly-scaled frequency bands, which is followed by a conventional processing by a dynamic range control/equalization block. The present invention describes another innovation, that is embedding the warped processing module in a two-channel quadrature mirror filter (QMF) bank for improving processing efficiency at high sample rates.

Abstract: An apparatus for utilizing spatial information for audio signal enhancement in a multiple distributed network may include a processor. The processor may be configured to receive representations of a plurality of audio signals including at least one audio signal received at a first device and at least a second audio signal received at a second device. The first and second devices may be part of a common acoustic space network and may be arbitrarily positioned with respect to each other. The processor may be further configured to combine the first and second audio signals to form a composite audio signal, and provide for communication of the composite audio signal along with spatial information relating to a sound source of at least one of the plurality of audio signals to another device.

Abstract: Aspects of the invention provide methods, computer-readable media, and apparatuses for digital processing of acoustic signals to create a reproduction of a natural or an artificial spatial sound environment. An aspect of the invention supports spatial audio processing such as extracting a center channel in up-mixing stereo sound for multi-channel loudspeaker setup or headphone virtualization. An aspect of the invention also supports directional listening in which sound sources in a desired direction may be amplified or attenuated. Direction and diffuseness parameters for regions of input channels are determined and an extracted channel is extracted from the input channels according to the direction and diffuseness parameters. A gain estimate is estimated for each signal component being fed into the extracted channel and an extracted channel may be synthesized from a base signal and the gain estimate. The input channels may be partitioned into a plurality of time-frequency regions.

Abstract: This invention relates to a device, a method, a software application program, a software application program product and an audio device for processing a digital signal, wherein the digital signal is separated and downsampled into at least two downsampled subband signals, wherein at least one of the at least two downsampled subband signals is equalized, and wherein the at least two downsampled subband signals are upsampled and combined into a digital output signal.

Abstract: This invention describes a method for adjusting the loudness and the spectral content of digital audio signals in a real-time using warped spectral filtering. A warped processing module modifies a spectral content of a digital audio signal with a set of gains for a plurality of non-linearly-scaled frequency bands determined by a warping factor ? of a warped delay line. Warped delay line signals, generated by the warped delay line, are processed by a warped filter block containing multiple warped finite impulse response filters, e.g., Mth band filters, using individual warped spectral filtering in said plurality of the non-linearly-scaled frequency bands, which is followed by a conventional processing by a dynamic range control/equalization block. The present invention describes another innovation, that is embedding the warped processing module in a two-channel quadrature mirror filter (QMF) bank for improving processing efficiency at high sample rates.

Abstract: An apparatus for providing a beat and tatum tracker includes an accent filter bank, a periodicity estimator, a period estimator and a phase estimator. The accent filter bank is configured to downsample an input audio signal. The periodicity estimator is configured to determine a periodicity based on the downsampled signal. The period estimator is configured to determine a period based on the periodicity. The phase estimator is configured to estimate a phase based on the period for determining beat and tatum times of the input audio signal.

Abstract: An apparatus for utilizing spatial information for audio signal enhancement in a multiple distributed network may include a processor. The processor may be configured to receive representations of a plurality of audio signals including at least one audio signal received at a first device and at least a second audio signal received at a second device. The first and second devices may be part of a common acoustic space network and may be arbitrarily positioned with respect to each other. The processor may be further configured to combine the first and second audio signals to form a composite audio signal, and provide for communication of the composite audio signal along with spatial information relating to a sound source of at least one of the plurality of audio signals to another device.

Abstract: This invention describes a method for adjusting the loudness and the spectral content of digital audio signals in a real-time using warped spectral filtering. A warped processing module modifies a spectral content of a digital audio signal with a set of gains for a plurality of non-linearly-scaled frequency bands determined by a warping factor ? of a warped delay line. Warped delay line signals, generated by the warped delay line, are processed by a warped filter block containing multiple warped finite impulse response filters, e.g., Mth band filters, using individual warped spectral filtering in said plurality of the non-linearly-scaled frequency bands, which is followed by a conventional processing by a dynamic range control/equalization block. The present invention describes another innovation, that is embedding the warped processing module in a two-channel quadrature mirror filter (QMF) bank for improving processing efficiency at high sample rates.

Abstract: Aspects of the invention provide methods, computer-readable media, and apparatuses for digital processing of acoustic signals to create a reproduction of a natural or an artificial spatial sound environment. An aspect of the invention supports spatial audio processing such as extracting a center channel in up-mixing stereo sound for multi-channel loudspeaker setup or headphone virtualization. An aspect of the invention also supports directional listening in which sound sources in a desired direction may be amplified or attenuated. Direction and diffuseness parameters for regions of input channels are determined and an extracted channel is extracted from the input channels according to the direction and diffuseness parameters. A gain estimate is estimated for each signal component being fed into the extracted channel and an extracted channel may be synthesized from a base signal and the gain estimate. The input channels may be partitioned into a plurality of time-frequency regions.

Abstract: Aspects of the invention provide methods, computer-readable media, and apparatuses for re-panning multiple audio signals by applying spatial cue coding. Sound sources in each of the signals may be re-panned before the signals are mixed to a combined signal. Processing may be applied in a conference bridge that receives two omni-directionally recorded audio signals. The conference bridge subsequently re-pans one of the signals to the listeners left side and the signal to the right side. The source image mapping and panning may further be adaptively based on the content and use case. Mapping may be done by manipulating the directional parameters prior to directional decoding or before directional mixing. Directional information that is associated with an audio input signal is remapped order to compress input source positions into virtual source positions. The virtual sources may be placed with respect to actual loudspeakers using binaural cue panning.

Abstract: The invention relates to an audio processing system 1. In order to improve the audio processing, the system comprises at least one audio processing component 11, 12, 13 with a group of real-time functions 14 for processing audio data and a group of control functions 15 for processing control signals. The system further comprises at least one processor 16 providing a first process 20 for executing real-time functions 14 of the at least one audio processing component 11, 12, 13 using a basically constant processing power and at least one further process 30 for executing control functions 15 of the at least one audio processing component 11, 12, 13 when needed without affecting the processing power employed for the first process 20. The invention relates equally to a corresponding method and to a corresponding software program product.

Abstract: Provided are improved systems, methods, and computer program products for direct encoding of spatial sound into a directional audio coding format. The direct encoding may also include providing spatial information for a monophonic sound source. The direct encoding of spatial information may be used, for example, in interactive audio applications such as gaming environments and in teleconferencing applications such as multi-party teleconferencing.