Embodiments described herein provide for smart configuration of audio settings for a playback device. According to an embodiment, while a playback device is a part of a first zone group that includes the playback device and at least one first playback device, the playback device applies a first audio setting. The embodiment also includes the playback device joining a second zone group that includes the playback device and at least one second playback device. The embodiment further includes the playback device applying a second audio setting based on an audio content profile corresponding to the second zone group.

The present technology relates to a sound processing apparatus and a sound processing system for enabling more stable localization of a sound image.

A virtual speaker is assumed to exist on the lower side among the sides of a tetragon having its corners formed with four speakers surrounding a target sound image position on a spherical plane. Three-dimensional VBAP is performed with respect to the virtual speaker and the two speakers located at the upper right and the upper left, to calculate gains of the two speakers at the upper right and the upper left and the virtual speaker, the gains being to be used for fixing a sound image at the target sound image position. Further, two-dimensional VBAP is performed with respect to the lower right and lower left speakers, to calculate gains of the lower right and lower left speakers, the gains being to be used for fixing a sound image at the position of the virtual speaker. The values obtained by multiplying these gains by the gain of the virtual speaker are set as the gains of the lower right and lower left speakers for fixing a sound image at the target sound image position. The present technology can be applied to sound processing apparatuses.

A method and an apparatus for sound effect compensation, a non-transitory computer-readable storage medium, and a terminal device are provided. The method includes the following. An acoustic transmission function of an audio processing device in an ear canal of a user is determined according to an audio signal currently output from the audio processing device. Compensate for the audio signal according to the acoustic transmission function.

The disclosed technologies calibrate audio output based on a computer-based hearing evaluation. In an embodiment, the disclosed technologies perform operations that include playing an electronic file that contains segments of digital music that have different playback attributes, wherein playback attribute include at least one of volume data, frequency data, left/right data; during the playing of the electronic file, recording timestamp data and user account data that are associated with a user input; mapping the timestamp data to a set of playback attributes of a segment of the digital music; determining calibration data based on the timestamp data and the set of playback attributes; storing the calibration data in a user profile that is associated with the user account data; when different digital music is played, automatically calibrating the playing of the different digital music based on the user profile that is associated with the user account data.

A method of analyzing and displaying spectral-dynamic properties of an audio signal, including measuring a level over time in multiple neighboring frequency-bands based on the audio signal; calculating spectral-dynamic characterizing values based on two or more statistics of a distribution of the measured levels within each of the frequency-bands; and displaying a graphical representation of the spectral-dynamic characterizing values as a function of frequency.

Example embodiments disclosed herein relate to separated audio analysis and processing. A system for processing an audio signal is disclosed. The system includes an audio analysis module configured to analyze an input audio signal to determine a processing parameter for the input audio signal, the input audio signal being represented in time domain. The system also includes an audio processing module configured to process the input audio signal in parallel with the audio analysis module. The audio processing module includes a time domain filter configured to filter the input audio signal to obtain an output audio signal in the time domain, and a filter controller configured to control a filter coefficient of the time domain filter based on the processing parameter determined by the audio analysis module. Corresponding method and computer program product of processing an audio signal are also disclosed.

An audio processor (1) includes a first filter coefficient calculator (31) that calculates a first filter coefficient so as to correspond to first gains for respective bands set by a user, a second filter coefficient calculator (32) that if values of third gains for respective bands of the first filter coefficient are greater than an absolute value of a second gain set by the user, calculates a second filter coefficient by limiting the values of the third gains for the respective bands to the amplitude value of the second gain, and a filtering unit (35) that filters an audio signal that has been transformed into a frequency-domain signal, using the second filter coefficient.

A speaker device with equalization (EQ) control is provided, wherein the speaker device may be a pair of earbuds or earphones or a headset, comprising: an audio receiving port for receiving audio signals from an audio source, a circuitry comprising at least a processor and a memory, the processor being configured to execute one or more firmware or software programs to perform tasks for processing the audio signals, including equalization of the audio signals according to one of multiple EQ settings stored in the memory, a user input terminal for detecting a user input to select one of the EQ settings; and a speaker driver for emitting sound corresponding to the processed audio signals.

A device is configured to (i) receive media content from a media source, (ii) generate a first series of frames including first portions of the media content and first playback timing information, (iii) generate a second series of frames including second portions of the media content and second playback timing information, (iv) transmit the first series of frames to a first playback device in a synchrony group for playback of the first portions of the media content in accordance with the first playback timing information, and (v) transmit the second series of frames to a second playback device in the synchrony group for playback of the second portions of the media content in accordance with the second playback timing information such that the media content is played back in synchrony by the synchrony group.

Traditional audio feedback elimination systems may attempt to reduce the effect of the audio feedback by simply scaling down the audio volume of the signal frequencies that are prone to howling. Other traditional feedback elimination systems may also employ adaptive notch filtering to detect and notch the so-called singing or howling frequencies as they occur in real-time. Such devices may typically have several knobs and buttons needing tuning, for example: the number of adaptive parametric equalizers (PEQs) versus fixed PEQs; attack and decay timers; and/or PEQ bandwidth. Rather than removing the singing frequencies with PEQs, the devices described herein attempt to holistically model the feedback audio and then remove the entire feedback signal. Two advantages of the devices described herein are: 1.) the system can operate at a much larger loop-gain (and hence with a much higher loudspeaker volume); and 2) setup is greatly simplified (i.e., no tuning knobs or buttons).