286results about "Automatic tone/bandwidth control" patented technology

Technology for grouping, consolidating, and pairing individual playback devices with network capability (players) to stimulate a multi-channel listening environment is disclosed. An example method includes receiving an audio signal containing a range of audio frequencies; amplifying, according to a gain parameter, the audio signal to be reproduced by at least one speaker. The example method includes automatically increasing the gain parameter to a higher gain parameter responsive to a determination that no more than a subset of the range of audio frequencies is to be reproduced by the at least one speaker. The example method includes amplifying, according to the higher gain parameter, the audio signal containing no more than the subset of the range of audio frequencies to be reproduced by the at least one speaker.

Example methods and systems provide automatic audio equalization for online conferences. A target, ideal, audio frequency response for use in online conferencing audio can be designed and preset for reference, and the equalizer can operate in real time to reach the target frequency response for any input. The equalizer can selectively apply per-band equalization to the audio signal as needed to adjust an energy value for each of multiple frequency bands to produce an output audio signal that that can be, as an example, routed through meeting servers or other infrastructure to other conference participants. The equalization can compensate for conditions local to a speaker that would otherwise adversely affect audio quality.

Embodiments are provided for receiving a request to output audio at a first speaker and a second speaker of an electronic device, determining that the electronic device is oriented in a portrait orientation or a landscape orientation, identifying, based on the determined orientation, a first equalization setting for the first speaker and a second equalization setting for the second speaker, providing, for output at the first speaker, a first audio signal with the first equalization setting, and providing, for output at the second speaker, a second audio signal with the second equalization setting.

Embodiments described herein provide for detecting presence of an object in proximity to a playback device and responsively performing one or more operations. In an example implementation, a playback device detects, via a proximity detector of the playback device, presence of an object in proximity to the playback device. In response to detecting the presence of the object, the playback device performs one or more operations.

A method for processing an audio signal, an electronic apparatus, and a computer-readable recording medium that records a program, the method, the electronic apparatus, and the program enabling appropriately-engineered sound to be listened to on various devices. After model information about a model of a device to which an audio signal is supplied has been acquired, from among a plurality of sound-setting information items associated with this model information, one of the sound-setting information items, the one corresponding to a type of sound, the type having been specified in accordance with an instruction to be input, is selected. Then, an audio signal is processed in accordance with this selected one of the sound-setting information items, and the processed audio signal is supplied to the device.

Embodiments are provided for receiving a request to output audio at a first speaker and a second speaker of an electronic device, determining that the electronic device is oriented in a portrait orientation or a landscape orientation, identifying, based on the determined orientation, a first equalization setting for the first speaker and a second equalization setting for the second speaker, providing, for output at the first speaker, a first audio signal with the first equalization setting, and providing, for output at the second speaker, a second audio signal with the second equalization setting.

Techniques are described for wirelessly receiving, at first and second speaker assemblies, signals representing left and right audio channels from each of first and second sources, and using the signals, simultaneously playing, on both speaker assemblies, music, computer game audio, computer chat audio, and speech picked up by speaker microphones. User-adjusted and non-user-adjusted EQ may be implemented on various portions of the signal processing.

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.

Amplitude and phase error correction in a transceiver circuit is provided. In embodiments disclosed herein, a transceiver circuit is configured to equalize an input vector to thereby correct amplitude-amplitude (AM-AM) and amplitude-phase (AM-PM) errors across a modulation bandwidth of the wireless communication circuit. Unlike conventional methods where complicated memory digital predistortion (mDPD) coefficients must be defined and calibrated for each modulation frequency within the modulation bandwidth, the transceiver circuit is configured herein to eliminate modulation frequency dependency of the AM-AM and AM-PM errors. As a result, it is possible to correct the AM-AM and AM-PM errors across the modulation bandwidth with reduced complexity to thereby improve efficiency and linearity of the wireless communication circuit.

The invention discloses an audio equalization method and device, equipment and a storage medium, and relates to the technical field of audio. The method comprises the following steps: acquiring an input audio signal and an equalization adjustment parameter, wherein the equalization adjustment parameter comprises at least one frequency point needing to be adjusted and a gain corresponding to the at least one frequency point; filtering the input audio signal through a filter bank to obtain a plurality of audio sub-bands; according to the at least one frequency point and the gain corresponding to the at least one frequency point, obtaining delay time corresponding to the plurality of audio sub-bands; performing delay processing on the plurality of audio sub-bands according to the delay time corresponding to the plurality of audio sub-bands to obtain a plurality of delayed audio sub-bands; and synthesizing the plurality of delayed audio sub-bands to obtain an output audio signal. According to the invention, the influence on the energy of the audio signal is avoided, the normal output of the audio signal is ensured, the equalization adjustment of the audio signal is realized, and the audio equalization quality is improved.

An example method includes receiving data indicating a configuration of one or more playback devices. The one or more playback devices may include one or more transducers. The method further includes, based on the received data, associating each of one or more audio streams respectively with at least one transducer of the one or more transducers. The method further includes generating the one or more audio streams and sending at least one of the generated one or more audio streams to each of the one or more playback devices. An example non-transitory computer readable medium and an example computing device related to the example method are also disclosed herein.

The invention relates to a method for calibrating a loudspeaker system in an acoustic environment. The method comprises steps of providing a local frequency response based on a recorded test signal and providing a target frequency response. A difference frequency response is established based on a difference between the local frequency response and the target frequency response. A list of exclusion frequency ranges associated with minima of the local frequency response is generated and one or more filter frequency ranges which are non-overlapping with the exclusion frequency ranges are identified. A target filter frequency is selected from the identified filter frequency ranges and a calibration filter related to the target filter frequency is implemented in an equalizer of the loudspeaker system. The invention further relates to a loudspeaker system arranged to carry out the above method for calibrating a loudspeaker system.

An audio post-processor (100) for post-processing an audio signal (102) having a time-variable high frequency gain information (104) as side information comprises: a band extractor (110) for extracting a high frequency band (112) of the audio signal (102) and a low frequency band (114) of the audio signal (102); a high band processor (120) for performing a time-variable modification of the high frequency band (112) in accordance with the time-variable high frequency gain information (104) to obtain a processed high frequency band (122); and a combiner (130) for combining the processed high frequency band (122) and the low frequency band (114). Furthermore, a pre-processor is illustrated.

An example method of operation may include applying a set of initial power and gain parameters for a speaker, playing a stimulus signal via the speaker, determining a sound level at a microphone location and a sound level at a predefined distance from the speakers, determining a gain at the microphone location based on a difference of the sound level at the microphone location and the sound level at the predefined distance from the speaker, and applying the gain to the speaker output.

A computer system for manipulating, mixing, or synthesizing dynamic sounds accesses a package of one or more music tracks. The computer system then receives input variables from the one or more vehicle sensors. One or more vehicle sensors measure a condition of a driving parameter of a vehicle. In response to the input variable, the computer system generates a particular audio effect using the one or more music tracks.

Certain embodiments provide methods and systems for managing a sound profile. An example playback device includes a network interface and a non-transitory computer readable storage medium having stored therein instructions executable by the processor. When executed by the processor, the instructions are to configure the playback device to receive, via the network interface over a local area network (LAN) from a controller device, an instruction. The example playback device is to obtain, based on the instruction, via the network interface from a location outside of the LAN, data comprising a sound profile. The example playback device is to update one or more parameters at the playback device based on the sound profile. The example playback device is to play back an audio signal according to the sound profile.

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 computer system for manipulating, combining, or composing dynamic sounds accesses a package of one or more music stems. The computer system then receive an input variable from one or more vehicle sensors. The one or more vehicle sensors measure an aspect of driving parameters of a vehicle. In response to the input variable, the computer system generate a particular audio effect with the one or more music stems.

To provide an audio generation method and a system.SOLUTION: A method for generating audio assets includes a step of receiving a plurality of input audio assets, a step of converting each of the input audio assets into an input graphic representation, a step of generating an input image from each of the input graphic representations, a step of feeding the input image to a generative model and extracting an output graphic representation from each of the output images in order to learn the generative model and generate one or more output images including the output graphic representations, and a step of converting the output graphic representation into an output audio asset.SELECTED DRAWING: Figure 2

An audio system, including at least one approximating circuit, the at least one approximating circuit configured to apply an approximated gain to the input signal, the least one approximating circuit outputting an approximated signal; a limiter configured to reduce a magnitude of the input signal by a gain determined by an amount the magnitude of the approximated filtered signal exceeds a predetermined threshold, the limiter outputting a limited input signal; a filter being respectively configured to receive the limited input signal and to respectively apply a filter gain, the filters outputting a plurality of filtered signals, wherein the approximated gain is substantially similar to the filter gain of one of the plurality of equalization filters; and a speaker array comprising a plurality of speakers, each speaker configured to receive a respective filtered signal of the plurality of filtered signals and to transduce the filtered signal to an audio signal.

In various embodiments, the disclosed techniques automatically change the audio equalization levels of audio content. When an audio content is requested for playback, the equalization manager of an audio system extracts the frequency components of a segment of the audio content. The equalization manager analyzes the frequency components of the segment and separates the frequency components into predefined frequency bands based on the analysis. The equalization manager determines sound pressure level (SPL) metrics of each frequency component. The equalization manager generates a custom equalization setting using gains calculated for each frequency band based on the SPL metrics. The equalization manager then adjusts the audio segment by applying the custom equalization setting to generate a modified segment (e.g., applying the calculated gains to each of the frequency bands). The modified segment is then output as audio to a user. The process then repeats for additional segments of the audio.

In various embodiments, the disclosed technology automatically alters audio equalization levels of audio content. When a request to play audio content is made, an equalization manager of an audio system extracts frequency components of a segment of the audio content. The equalization manager analyzes the frequency components of the segment and separates the frequency components into predefined frequency bands based on the analysis. The equalization manager determines a sound pressure level (SPL) indicator for each frequency component. The equalization manager generates a customized equalization setting based on the SPL indicators using gains computed for each frequency band. The equalization manager then adjusts the audio segment by applying the customized equalization setting to generate a modified segment (e.g., applying the computed gains to each of the frequency bands). The modified audio segment is then output as audio to a user. The process is then repeated for additional segments of the audio.

A signal processing system for adjusting an audio signal, the signal processing system configured to: receive an audio signal for adjustment at a signal input; at a function generator, receive a selection of a musical key and scale, the musical key and scale including musical notes, and generate an audio signal processing function, configured to adjust the amplitude of frequencies of an audio signal, the audio signal processing function being dependent on the selected musical key and scale; apply the generated audio signal processing function to the received audio signal to generate an adjusted audio signal; and, output the adjusted audio signal at a signal output.

A method for generating an analog signal for signal analysis, which includes the steps of receiving analog spectrogram data that characterizes one or more bands, constructing an analog signal from the analog spectrogram data, discretely sampling the analog signal for a time period comprised of a sequence of equal time intervals to create a digital signal with values that characterize the analog signal for each of the time periods, constructing a digital spectrogram from the digital signal, wherein the digital spectrogram represents the rate of change of amplitude with respect to time and frequency; and computing an analog spectrogram from the digital spectrogram, wherein an analog spectrogram represents the one or more bands in units of time, frequency and amplitude for the time period.