Example techniques involve calibration of one or more playback devices. An example implementation involves a playback device playing back audio content using a first calibration via the one or more audio transducers and the one or more amplifiers. The first calibration is based on a response of a listening environment to audio content playback by the playback device. The playback device records, via one or more microphones, at least a portion of the played back audio content. Based on the recorded audio content, the playback device detects a change in the response of the listening environment to audio content playback by the playback device. Responsive to detecting the change in the response, the playback device causes output of a prompt to initiate a calibration procedure for the playback device. The calibration procedure involves a mobile device recording playback by the playback device.

A signal processing apparatus includes a control section, a signal processing section connected with a plurality of signal processing elements and performs signal processing for enhancing or attenuating an input signal in a specific frequency band, and a crossfade signal section including a crossfade signal processing element capable of replacing at least one of the signal processing elements, wherein the control section controls any one of the signal processing elements among the plurality of signal processing elements, and the crossfade signal processing element, to crossfade to the crossfade signal processing element having the signal processing element as a new characteristic, to perform processing for replacing any one of the signal processing elements by the crossfade signal processing element, and to perform the processing on remaining signal processing elements of the plurality of signal processing elements in the signal processing section.

A method of acoustically controlling an equalizer in a natural language includes receiving a speech; performing automatic speech recognition; outputting a text message corresponding to the speech; obtaining a digital signal processing parameter of the corresponding text message from a database according to the text message; and adjusting the equalizer according to the digital signal processing parameter. The method may be implemented on a computer readable storage medium storing executable software.

A playback device is configured to: based on a room setting of the playback device, determine that the playback device is associated with a first room of a media playback system; operate in a first mode that is associated with a first user type and a first set of playback control options; determine that the room setting of the playback device has changed; determine that the playback device is no longer associated with the first room and is associated with a second room of the media playback system; transition to operating in a second mode; determine that the playback device is no longer associated with the second room and is associated with the first room; transition to operating in the first mode; receive a playback command; determine that the playback command corresponds to a playback control option within the first set; and execute the playback command.

A ribbon microphone assembly may include a ribbon capsule assembly, an amplifier connected to the capsule assembly, an analog-to-digital converter connected to the amplifier, a digital signal processor connected to the analog-to-digital converter to compensate for an on-axis frequency response caused by a directional polar response of the ribbon microphone assembly, and a USB output port.

An integrated system for hearing protection and enhancement having earpieces including microphones and receivers to provide for user control of ambience level (mic gain), monitor input, control over limiting of excessive levels, and frequency equalization, all with very wide input dynamic range capability and controlled output dynamic range. The user control comprises a wired or wireless smartphone app. Also, the wide input dynamic range is achieved by placement of limiter circuitry capable of handling the entire input dynamic range prior to the most dynamic range limited circuit blocks: digital EQ block, output amplifier, earpiece receiver, and the users' ears. Switches allow combining left and right signals to be directed to a user's favored ear, if necessary.

Disclosed are a sound effect configuration method and system and a related device. The method comprises: determining, when a starting operation on a target application of a mobile terminal is detected, a sound effect setting time period to which the system time belongs, the target application being used for controlling an audio output device; determining a reference sound effect parameter corresponding to the sound effect setting time period to which the system time belongs according to mapping relationships between pre-stored sound effect setting time periods of the audio output device and sound effect parameters; sending a pre-configuration instruction comprising the reference sound effect parameter to the audio output device, the pre-configuration instruction being used for instructing the audio output device to create, according to the reference sound effect parameter, a sound effect instance for an audio stream to be played, and the sound effect instance being used for configuring the sound effect of the audio stream to be played. Embodiments of the present invention can improve convenience and intelligence of sound effect setting and user experience.

A digital signal processor for audio includes: an interface circuit communicating with an microcomputer; a multi-band equalizer including digital filters corresponding respectively to bands, wherein the frequency characteristic of each digital filter is changed according to coefficient group; a memory; and a coefficient setting circuit setting the coefficient group of each digital filter, wherein the frequency characteristic of the multi-band equalizer is selected from presets, wherein, when the digital signal processor start, the interface circuit receives setting data corresponding to the presets from the microcomputer and stores the received setting data in the memory, and wherein, at the time of audio reproduction, the interface circuit receives selection data to designate one of the plurality of presets and the coefficient setting circuit sets the coefficient group of each digital filter according to one of the setting data stored in the memory, the one setting data corresponding to the selection data.

Examples involve calibration of a playback device including multiple audio transducers. An example implementation involves forming a particular playback configuration that configures the multiple audio transducers to form two or more channels and receiving a command to initiate calibration of the particular playback configuration. The calibration includes emitting a calibration sound according to a sequence that involves the calibration sound cycling through frequencies of a calibration frequency range. Based on the command, the playback device emits the calibration sound contemporaneously via the two or more channels. Emission of the calibration sound by each channel is staggered such that each emitted calibration sound is delayed relative to a preceding emitted calibration sound. The playback device receives audio processing coefficients that are based on one or more recordings of the emitted calibration sound; and applies the audio processing coefficients when playing back the audio content as part of the particular playback configuration.

A method of processing an audio signal is disclosed. According to embodiments of the method, magnitude response information of a prototype filter is determined. The magnitude response information includes a plurality of gain values, at least one of which includes a first gain corresponding to a first frequency. The magnitude response information of the prototype filter is stored. The magnitude response information of the prototype filter at the first frequency is retrieved. Gains are computed for a plurality of control frequencies based on the retrieved magnitude response information of the prototype filter at the first frequency, and the computed gains are applied to the audio signal.