Systems and methods are provided for automatically adjusting a reference gain of an audio mixer having a reference channel for receiving a far end audio signal from a remote location as a reference signal and a plurality of audio input channels for receiving audio signals captured by a plurality of microphone element. An exemplary method includes determining an echo level in an input audio signal received at a given audio input channel, and automatically determining a gain amount for the reference channel based on the echo level. An exemplary system includes a reference gain adjuster configured to automatically determine a gain amount for the reference channel based on an echo level detected in an input audio signal received at a given audio input channel.

Techniques described herein may involve audio settings based on an environment. An example implementation may involve a playback device playing back first audio content and during playback of at least a portion of the first audio content, detecting, via the at least one microphone, an audio signal. At least a portion of the detected audio signal may include a reflection of the first audio content played back by the playback device. The playback device may determine an equalization setting based on at least the determined one or more reflection characteristics and apply the determined equalization setting during playback of second audio content.

A content output device includes at least one processor and at least one communicator. The at least one communicator is configured to output a plurality of channels from the at least one processor to a plurality of audio devices signals. When the number of audio devices increases and at least one additional channel is available to be reproduced, the assigning task assigns the at least one additional channel to the increased number of audio devices and the outputting task outputs the at least one additional channel in accordance with the changed assignment of the plurality of channels.

According to certain embodiments, a microphone array having a plurality of microphone elements is calibrated by ensonifying the microphone array at a first direction relative to the microphone array with a first acoustic signal to concurrently generate a first set of audio signals from two or more of the microphone elements and processing the first set of audio signals to calibrate the two or more microphone elements. One or more other sets of audio signals can be generated by ensonifying the microphone array with one or more other acoustic signals at one or more other directions relative to the microphone array, where the two or more microphone elements are calibrated using the first set and the one or more other sets of audio signals. The calibration process can be performed outside of an anechoic chamber using one or more acoustic sources located outside or inside the microphone array.

The present application relates to the technical field of loudspeakers. Disclosed are a loudspeaker device, a method, apparatus and device for adjusting the sound effect thereof, and a medium. The method is used for increasing the diversity of the sound effect of the loudspeaker device, and comprises: determining the spatial distribution state of multiple loudspeaker units in the loudspeaker device; determining a sound effect mode corresponding to the spatial distribution state; and adjusting the sound effect of the loudspeaker device according to the sound effect mode. According to the present application, the sound effect mode is accordingly adjusted according to the spatial distribution state of multiple loudspeaker units in the loudspeaker device. In this way, when the spatial distribution state changes, the sound effect mode of the loudspeaker device also changes, thereby overcoming the defect of the single sound effect of the loudspeaker device, i.e., according to the present application, the diversity of the sound effect of the loudspeaker device can be increased efficiently.

A signal processing device according to aspects of the present disclosures comprises a measuring section configured to measure an impulse response between each of a plurality of speakers and a predetermined listening position, a Fourier transformer configured to obtain a frequency spectrum corresponding to each of the plurality of speakers by applying a Fourier transform, a phase adjustment amount calculator configured to calculate a phase adjustment amount for each frequency, a band detector configured to detect a leading phase band, a phase converter configured to convert a phase of the leading phase band to a lagging phase, and a filter coefficient generator configured to generate a filter coefficient based on the phase adjustment amount after conversion by the phase converter.

A computer system for processing audio content may receive content that includes metadata on spatial features about a plurality of objects, convert a format set according to a production environment of the content to a format according to a playback environment in an electronic apparatus, and transmit the content in the converted format to the electronic apparatus. The computer system may support content produced in various production environments and various playback environments.

Some implementations involve receiving a content stream that includes audio data, determining a content type corresponding to the content stream and determining, based at least in part on the Receiving, by a control system and via an interface system, a content stream that includes audio data content type, a noise compensation method. Some examples involve performing the noise compensation method on the audio data to produce noise-compensated audio data, rendering the noise-compensated audio data for reproduction via a set of audio reproduction transducers of the audio environment, to produce rendered audio signals, and providing the rendered audio signals to at least some audio reproduction transducers of the audio environment.

The disclosure relates to a soundbar and a method for automatic surround pairing and calibration of a surround sound system. The soundbar includes two built-in microphones on the left and right respectively, which can be used for determining relative positions of left and right surround speakers. When the relative positions of the left and right speakers are not correct, configurations of left and right surround channels can be automatically swapped with each other without manually swapping physical positions of the surround speakers by a user. In addition, latencies including a latency of each channel of a main system may also be calibrated, and magnitude compensation may be achieved by calculating a filter compensation coefficient of each line and merging it into an original filter. The automatic surround pairing and calibration of the surround sound system may be one-click completed automatically by the user by pressing a start button.

Technologies are disclosed for improving the efficiency of real-time audio processing, and specifically for improving the efficiency of continuously modifying a real-time audio signal. Efficiency is improved by reducing memory bandwidth requirements and by reducing the amount of processing used to modify the real-time audio signal. In some configurations, memory bandwidth requirements are reduced by selectively transferring active samples in the frequency domain—e.g. avoiding the transfer samples with amplitudes of zero or near-zero. This has particular importance when the specialized hardware retrieves samples from main memory in real-time. In some configurations, the amount of processing needed to modify the audio signal is reduced by omitting operations that do not meaningfully affect the output audio signal. For example, a multiplication of samples may be avoided when at least one of the samples has an amplitude of zero or near-zero.