Apparatuses, systems, and techniques to facilitate understanding of media content using neural networks to adjust playback speed and volume based on environmental and other factors. In at least one embodiment, playback of media content is slowed down or sped up if audio associated with said media content is difficult to understand based on background noise, accent, difficulty of material, as well as other factors that decrease understandability of media content.

According to one embodiment, in a semiconductor integrated circuit, the second circuit samples an amplitude of the output second signal at a plurality of timings every given cycle in a period corresponding to a second period of the pattern. The second circuit controls a parameter relating to the frequency characteristic for the first circuit according to a first magnitude relation and a second magnitude relation. The first magnitude relation is a relation between an absolute value of a first amplitude and a first threshold. The first amplitude is an amplitude sampled at a first timing among the plurality of timings. The second magnitude relation is a relation between an absolute value of a second amplitude and the first threshold. The second amplitude is an amplitude sampled at a second timing. The second timing is a timing after the first timing among the plurality of timings.

A system comprising audio processing circuitry is provided. The audio processing circuitry is operable to receive audio signals. The audio processing circuitry is operable to process the audio signals to detect strength of a chat component of the audio signals and strength of a game component of the audio signals. The audio processing circuitry is operable to automatically control a volume setting based on one or both of: the detected strength of the chat component, and the detected strength of the game component. The combined-game-and-chat audio signals may comprise a left channel signal and a right channel signal. The processing of the combined-game-and-chat audio signals may comprise measuring strength of a vocal-band signal component that is common to the left channel signal and the right channel signal.

The response of a Rogowski coil based current measuring circuit is often proportional to frequency. To correct for this a low pass or integrating function is applied to the response to linearize it. The low pass filter is made from real resistors and capacitors, and tolerances in their values significantly affect the estimate of current. This disclosure relates to a way of addressing such problems. This allows consumers of electricity to have confidence in the accuracy of, for example, their electricity meter.

In one aspect, an example method includes (i) determining, by a playback device, a first loudness level of a first portion of first media content from a first source while the playback device presents the first media content, with the first portion having a first length; (ii) switching, by the playback device, from presenting the first media content from the first source to presenting second media content from a second source; (iii) based on the switching, determining, by the playback device, second loudness levels of second portions of the first media content while the playback device presents the second media content, with the second portions having a second length that is shorter than the first length; and (iv) while the playback device presents the second media content, adjusting, by the playback device, a volume of the playback device based on one or more of the second loudness levels.

In one aspect, an example method includes (i) determining, by a playback device, a first loudness level of a first portion of first media content from a first source while the playback device presents the first media content, with the first portion having a first length; (ii) switching, by the playback device, from presenting the first media content from the first source to presenting second media content from a second source; (iii) based on the switching, determining, by the playback device, second loudness levels of second portions of the first media content while the playback device presents the second media content, with the second portions having a second length that is shorter than the first length; and (iv) while the playback device presents the second media content, adjusting, by the playback device, a volume of the playback device based on one or more of the second loudness levels.

Methods and apparatus to perform an automated gain control protocol with an amplifier based on historical data corresponding to contextual data are disclosed. Example apparatus disclosed herein are to select an automatic gain control (AGC) parameter for an AGC protocol based on historical data corresponding to contextual data, the contextual data including at least one of a time during which the AGC protocol is performed, a panelist identified by a meter, demographics of an audience identified by the meter, a location of the meter, a station identified by the meter, a media type identified by the meter, or a sound pressure level identified by the meter. The disclosed example apparatus are also to perform the AGC protocol based on the selected AGC parameter.

In some embodiments, a method for processing an audio signal in an audio processing apparatus is disclosed. The method includes receiving an audio signal and a parameter, the parameter indicating a location of an auditory event boundary. An audio portion between consecutive auditory event boundaries constitutes an auditory event. The method further includes applying a modification to the audio signal based in part on an occurrence of the auditory event. The parameter may be generated by monitoring a characteristic of the audio signal and identifying a change in the characteristic.

Methods and systems for modification of electronic system operation based on acoustic ambience classification are presented. In an example method, at least one audio signal present in a physical environment of a user is detected. The at least one audio signal is analyzed to extract at least one audio feature from the audio signal. The audio signal is classified based on the audio feature to produce at least one classification of the audio signal. Operation of an electronic system interacting with the user in the physical environment is modified based on the classification of the audio signal.

A microphone may comprise a microphone element for detecting sound, and a digital signal processor configured to process a first audio signal that is based on the sound in accordance with a selected one of a plurality of digital signal processing (DSP) modes. Each of the DSP modes may be for processing the first audio signal in a different way. For example, the DSP modes may account for distance of the person speaking (e.g., near versus far) and/or desired tone (e.g., darker, neutral, or bright tone). At least some of the modes may have, for example, an automatic level control setting to provide a more consistent volume as the user changes their distance from the microphone or changes their speaking level, and that may be associated with particular default (and/or adjustable) values of the parameters attack, hold, decay, maximum gain, and/or target gain, each depending upon which DSP is being applied.