Abstract: A processor-based method of reducing loudspeaker-induced harmonic distortion in sound generated by a loudspeaker responsive to a first audio signal, the method including: attenuating a first band in the first audio signal by an attenuation gain, the first band including a fundamental frequency, wherein the attenuating dynamically adjusts the attenuation gain in accordance with, at least, a ratio between: i) an energy of the first band in the first audio signal; and ii) an energy of a second band in the first audio signal, the second band including one or more frequencies that is an integer multiple of the fundamental frequency.

Abstract: An audio system including a first microphone-equipped playback device and a second microphone-equipped playback device. The audio system is configured to synchronize playing of audio to a listener position by receiving by the first microphone-equipped playback device an audio stream, and playing the audio stream on a speaker of the first microphone-equipped playback device, in accordance with a playback delay ?t. The playback delay ?t is in accordance with a first calibration sound originating at the listening position, a second calibration sound originating at the second microphone-equipped playback device, and a third calibration sound originating at the first microphone-equipped playback device.

Abstract: Presentation of a recommendation to a user for individual processing of audio tracks in a digital audio workstation. Training audio tracks are provided to a human sound mixer and responsive to the training audio tracks individually processed training audio tracks are received from the human sound mixer. The training audio tracks and the individually processed training audio tracks are input to a machine to train the machine. Audio processing operations are output from the trained machine and stored in a record of a database.

Abstract: Adjusting gains of an audio mix. Audio tracks using respective gains are individually analyzed to compute a first metric of frequency content. A user input specifies a desired second metric of the frequency content. Responsive to the user input, respective gains of the audio tracks are collectively and simultaneously adjusted to produce respective adjusted gains of the audio tracks. A second audio mix when played of the audio tracks with the respective adjusted gains has a third metric of frequency content different from the first metric of frequency content. The third metric is closer to the second metric than said first metric.

Abstract: An audio system including a first microphone-equipped playback device and a second microphone-equipped playback device. The audio system is configured to synchronize playing of audio to a listener position by receiving by the first microphone-equipped playback device an audio stream, and playing the audio stream on a speaker of the first microphone-equipped playback device, in accordance with a playback delay ?t. The playback delay ?t is in accordance with a first calibration sound originating at the listening position, a second calibration sound originating at the second microphone-equipped playback device, and a third calibration sound originating at the first microphone-equipped playback device.

Abstract: A trained machine configured to input a stereo sound track and separate the stereo sound track into multiple N separated stereo audio signals respectively characterized by multiple N audio content classes. All stereo audio as input in the stereo sound track is included in the N separated stereo audio signals. A mixing module is configured to spatially localize symmetrically and without cross-talk, between left and right, the N separated stereo audio signals into multiple output channels. The output channels include respective mixtures of one or more of the N separated stereo audio signals. Gain is adjusted of the output channels into left and right binaural outputs to conserve summed levels of the N separated stereo audio signals distributed over the output channels.

Abstract: A computerized microphone-equipped audio playback device comprising a speaker and microphone and being configured to receive digital audio; and play the digital audio on a speaker, in accordance with a playback delay that is derivative of, at least: an arrival time of a first calibration sound at the processor and an arrival time of the first calibration sound at a second microphone-equipped audio playback device, wherein the first calibration sound originated at the listener position; a generation time of a second calibration sound at the second microphone-equipped audio playback device, and an arrival time of the second calibration sound at the processor; and a generation time of a third calibration sound at the processor, and an arrival time of the third calibration sound at the second microphone-equipped audio playback device; thereby synchronizing arrival of sound of the two playback devices at the listener position.

Abstract: A processor-based method of reducing loudspeaker-induced harmonic distortion in sound generated by a loudspeaker responsive to a first audio signal, the method including: attenuating a first band in the first audio signal by an attenuation gain, the first band including a fundamental frequency, wherein the attenuating dynamically adjusts the attenuation gain in accordance with, at least, a ratio between: i) an energy of the first band in the first audio signal; and ii) an energy of a second band in the first audio signal, the second band including one or more frequencies that is an integer multiple of the fundamental frequency.

Abstract: Adjusting gains of an audio mix. Audio tracks using respective gains are individually analyzed to compute a first metric of frequency content. A user input specifies a desired second metric of the frequency content. Responsive to the user input, respective gains of the audio tracks are collectively and simultaneously adjusted to produce respective adjusted gains of the audio tracks. A second audio mix when played of the audio tracks with the respective adjusted gains has a third metric of frequency content different from the first metric of frequency content. The third metric is closer to the second metric than said first metric.

Abstract: Presentation of a recommendation to a user for individual processing of audio tracks in a digital audio workstation. Training audio tracks are provided to a human sound mixer and responsive to the training audio tracks individually processed training audio tracks are received from the human sound mixer. The training audio tracks and the individually processed training audio tracks are input to a machine to train the machine. Audio processing operations are output from the trained machine and stored in a record of a database.

Abstract: A method for conveying to a listener a directionality-preserving pseudo low frequency psycho-acoustic sensation of a multichannel sound signal, including: deriving from the sound signal, by a processing unit, a high frequency multichannel signal and a low frequency multichannel signal, generating a multichannel harmonic signal, the loudness of at least one channel signal of the multichannel harmonic signal substantially matching the loudness of a corresponding channel in the low frequency multichannel signal; and at least one interaural level difference (ILD) of at least one frequency of the at least one channel pair of the multichannel harmonic signal substantially matching an ILD of a corresponding fundamental frequency in a corresponding channel pair in the low frequency multichannel signal; and summing the harmonic multichannel signal and the high frequency multichannel signal thereby giving rise to a psychoacoustic alternative signal.

Abstract: A computerized microphone-equipped audio playback device comprising a speaker and microphone and being configured to receive digital audio; and play the digital audio on a speaker, in accordance with a playback delay that is derivative of, at least: an arrival time of a first calibration sound at the processor and an arrival time of the first calibration sound at a second microphone-equipped audio playback device, wherein the first calibration sound originated at the listener position; a generation time of a second calibration sound at the second microphone-equipped audio playback device, and an arrival time of the second calibration sound at the processor; and a generation time of a third calibration sound at the processor, and an arrival time of the third calibration sound at the second microphone-equipped audio playback device; thereby synchronizing arrival of sound of the two playback devices at the listener position.

Abstract: A method for conveying to a listener a directionality-preserving pseudo low frequency psycho-acoustic sensation of a multichannel sound signal, including: deriving from the sound signal, by a processing unit, a high frequency multichannel signal and a low frequency multichannel signal, generating a multichannel harmonic signal, the loudness of at least one channel signal of the multichannel harmonic signal substantially matching the loudness of a corresponding channel in the low frequency multichannel signal; and at least one interaural level difference (ILD) of at least one frequency of the at least one channel pair of the multichannel harmonic signal substantially matching an ILD of a corresponding fundamental frequency in a corresponding channel pair in the low frequency multichannel signal; and summing the harmonic multichannel signal and the high frequency multichannel signal thereby giving rise to a psychoacoustic alternative signal.

Abstract: A method, and one or more non-transitory computer-readable media storing instructions, and a device for removal of unwanted components in an audio signal, the device comprising a processor, coupled to memory, configured to receive reference and processed inputs into memory where the processed input is a result of a reduction process of unwanted components of the audio signal, estimate envelope values for processed and reference inputs at a plurality of time and frequency instances, for each time and frequency instance: compute a first gain in relation to a ratio of the estimated envelope value of the processed input to the estimated envelope value of the reference input, apply a nonlinear process to said first gain to produce a second gain, compute an output gain as the ratio between second gain and first gain and, apply the output gain to processed input, thereby producing a filtered output with unwanted components suppressed.

Abstract: A device for removal of unwanted components in an audio signal, the device comprising a processor, coupled to memory, configured to receive reference and processed inputs into memory where the processed input is a result of a reduction process of unwanted components of the audio signal, estimate envelope values for processed and reference inputs at a plurality of time and frequency instances, for each time and frequency instance: compute a first gain in relation to a ratio of the estimated envelope value of the processed input to the estimated envelope value of the reference input, apply a nonlinear process to said first gain to produce a second gain, compute an output gain as the ratio between second gain and first gain and, apply the output gain to processed input, thereby producing a filtered output with unwanted components suppressed.

Abstract: A circuit, method, and system for producing artificial ambience effect for an input audio signal, mono, stereo, or surround. The ambience effect enhances artificial reverberation, replaces artificial reverberation, or synthesizes extra audio channels, such as surround channels. The circuit may include a transient reduction module and a reverberation filter. The transient reduction module may be adapted to reduce transients in an input audio signal of one or more channels. The reverberation filter maybe adapted to receive a transient-reduced signal of one or more channels corresponding to the transient-reduced signal.

Abstract: In accordance with some embodiments of the present invention there is provided a system, a method and a circuit for processing a stereo audio signal. According to some embodiments, the system for processing a stereo audio signal may include an audio processing circuit. The audio processing module or circuit may be operatively connected to an audio input interface and to an output audio interface. Through the audio input interface, the audio processing circuit may be adapted for receiving a 2-channels stereo audio signal. The audio processing module may be adapted for determining an output mono audio signal representing the center sound and a stereo audio signal representing the stereo sound without the center. Through the output interface the audio processing circuit may provide each of an output mono audio signal representing the center sound and a stereo audio signal representing the stereo sound without the center.

Abstract: An apparatus for classifying an input audio signal into audio contents of a first and second class, comprising an audio segmentation module adapted to segment said input audio signal into segments of a predetermined length; a feature computation module adapted to calculate for the segments features characterizing said audio input signal; a threshold comparison module adapted to generate a feature vector for each of said one or more segments based on a plurality of predetermined thresholds, the thresholds including for each of the audio contents of the first class and of the second class a substantially near certainty threshold, a substantially high certainty threshold, and a substantially low certainty threshold; and a classification module adapted to analyze the feature vector and classify each one of said one or more segments as audio contents of the first class, of the second class, or as non-decisive audio contents.

Abstract: A circuit, method, and system for producing artificial ambience effect for an input audio signal, mono, stereo, or surround. The ambience effect enhances artificial reverberation, replaces artificial reverberation, or synthesizes extra audio channels, such as surround channels. The circuit may include a transient reduction module and a reverberation filter. The transient reduction module may be adapted to reduce transients in an input audio signal of one or more channels. The reverberation filter maybe adapted to receive a transient-reduced signal of one or more channels corresponding to the transient-reduced signal.

Abstract: In accordance with some embodiments of the present invention there is provided a system, a method and a circuit for processing a stereo audio signal. According to some embodiments, the system for processing a stereo audio signal may include an audio processing circuit. The audio processing module or circuit may be operatively connected to an audio input interface and to an output audio interface. Through the audio input interface, the audio processing circuit may be adapted for receiving a 2-channels stereo audio signal. The audio processing module may be adapted for determining an output mono audio signal representing the center sound and a stereo audio signal representing the stereo sound without the center. Through the output interface the audio processing circuit may provide each of an output mono audio signal representing the center sound and a stereo audio signal representing the stereo sound without the center.