A microphone equalization system determines an equalization filter which is used to spectrally shape a signal from a microphone. A loudspeaker has an enclosure, a back volume, a driver diaphragm, and a sensor located inside the back volume. An external microphone is located outside the loudspeaker. A processor determines the equalization filter based on comparison of a first signal from an output of the sensor and a second signal from an output of the external microphone both of which are produced while the driver diaphragm produces acoustic pressure waves produced by the driver diaphragm. The processor is to then spectrally shape a third signal from the output of the external microphone that is responsive to further acoustic pressure waves from a source other than the loudspeaker. Other aspects are also described and claimed.

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.

A preamplifier for musical instruments includes: an operational amplifier 40 to amplify an inputted analog audio signal; a dual-unit variable resistor 30 to change an amplification factor of the operational amplifier 40 by manually operating an operation unit; an A/D converter 51 to convert the amplified analog audio signal to a digital audio signal; and a digital signal processor 60 to digital-signal process the digital audio signal, wherein the dual-unit variable resistor 30 includes a second variable resistor 32 to output a detection signal in accordance with an amount of operation of the operation unit, and the digital signal processor 60 is capable of implementing, based on a value of the detection signal, a first digital gain process to amplify the digital audio signal and/or a second digital gain process to attenuate the digital audio signal.

A preamplifier for musical instruments includes: an operational amplifier 40 to amplify an inputted analog audio signal; a dual-unit variable resistor 30 to change an amplification factor of the operational amplifier 40 by manually operating an operation unit; an A/D converter 51 to convert the amplified analog audio signal to a digital audio signal; and a digital signal processor 60 to digital-signal process the digital audio signal, wherein the dual-unit variable resistor 30 includes a second variable resistor 32 to output a detection signal in accordance with an amount of operation of the operation unit, and the digital signal processor 60 is capable of implementing, based on a value of the detection signal, a first digital gain process to amplify the digital audio signal and/or a second digital gain process to attenuate the digital audio signal.

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 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).

A network device communicates with one or another set of antennas depending on an orientation of the network device. The network device includes a first set of one or more antennas, a second set of one or more antennas, a processor, and memory having stored thereon instructions executable by the processor to cause the device to perform functions. The functions include (1) determining that an orientation of the network device is one of a first orientation and a second orientation; (2) if the determined orientation is the first orientation, then causing the network device to communicate using the first set of one or more antennas; and (3) if the determined the orientation is the second orientation, then causing the network device to communicate using the second set of one or more antennas.

Techniques for altering audio being output by a voice-controlled device, or another device, to enable more accurate automatic speech recognition (ASR) by the voice-controlled device. For instance, a voice-controlled device may output audio within an environment using a speaker of the device. While outputting the audio, a microphone of the device may capture sound within the environment and may generate an audio signal based on the captured sound. The device may then analyze the audio signal to identify speech of a user within the signal, with the speech indicating that the user is going to provide a subsequent command to the device. Thereafter, the device may alter the output of the audio (e.g., attenuate the audio, pause the audio, switch from stereo to mono, etc.) to facilitate speech recognition of the user's subsequent command.

Techniques for altering audio being output by a voice-controlled device, or another device, to enable more accurate automatic speech recognition (ASR) by the voice-controlled device. For instance, a voice-controlled device may output audio within an environment using a speaker of the device. While outputting the audio, a microphone of the device may capture sound within the environment and may generate an audio signal based on the captured sound. The device may then analyze the audio signal to identify speech of a user within the signal, with the speech indicating that the user is going to provide a subsequent command to the device. Thereafter, the device may alter the output of the audio (e.g., attenuate the audio, pause the audio, switch from stereo to mono, etc.) to facilitate speech recognition of the user's subsequent command.