40results about "Volume compression/expansion" patented technology

Presented is a method (100) for real-time processing an audio stream in an audio speaker arrangement comprising at least one audio speaker. The method (100) comprises receiving (110) an audio-stream and a playback volume associated with the audio-stream and determining (120) a playback sound pressure level of the audio speaker based on the playback volume and pre-stored acoustic transfer data. The pre- stored acoustic transfer data is associating the playback volume with the playback sound pressure level of the audio speaker. The method further comprises generating (130) a compensation filter based on amplitude compensation data associated with the determined playback sound pressure level and filtering (140) the audio-stream using the compensation filter and applying a gain factor to the audio stream generating a compensated audio-stream. The gain factor is configured such that a total audio power of the audio stream is preserved regardless of the compensation filter. The compensated audio stream is provided (150) to the audio speaker (21). Additionally, an audio speaker arrangement and a computer program product are presented.

To provide an audio compression method and device for suppressing distortion of an audio signal during reproduction and improving reproduction quality. [Solution] The audio compression method includes the steps of obtaining a first audio signal to be processed (an original audio signal to be reproduced), filtering the first audio signal with a frequency spectrum compensation filter based on a predetermined target frequency spectrum related to the distortion band of the audio reproduction device to obtain a second audio signal, processing the second audio signal with a predetermined dynamic range compressor (DRC) to obtain a compression coefficient, and outputting a compressed signal based on the compression coefficient.

Examples of the disclosure relate to apparatus, methods and computer programs for enabling audio zooming. The apparatus can comprise means for determining, for an audio signal, if sound energy in at least one first direction is different to sound energy in at least one second direction by at least a threshold amount. The means may also be for controlling an amount of headroom provided based on whether or not sound energy in at least one first direction is different to the sound energy in at least one second direction by at least a threshold amount.

Examples of the disclosure relate to apparatus, methods and computer programs for enabling audio zooming. The apparatus can include circuitry configured for determining, for an audio signal, if sound energy in at least one first direction is different to sound energy in at least one second direction by at least a threshold amount. The circuitry may also be configured for controlling an amount of headroom provided based on whether or not sound energy in at least one first direction is different to the sound energy in at least one second direction by at least a threshold amount.

The invention relates to the technical field of audio signal processing, in particular to an audio signal processing method and device, electronic equipment and a storage medium. The audio signal processing method comprises the following steps: firstly, receiving a currently input first audio signal, and when the amplitude of the first audio signal is determined to be greater than or equal to a first signal threshold value, firstly turning down a digital gain in the first audio signal to obtain a second audio signal, and then increasing an analog gain in the second audio signal to obtain a third audio signal; or, when it is determined that the amplitude of the first audio signal is smaller than the first signal threshold value, firstly increasing the digital gain in the first audio signal to obtain a fourth audio signal, and then reducing the analog gain in the fourth audio signal; in this way, the digital gain and the analog gain in the first audio signal can be dynamically adjusted according to the amplitude change condition of the input first audio signal, so that it is guaranteed that the quality of the processed audio signal is high, and the processed audio signal has a high signal-to-noise ratio and a wide signal dynamic range.

The invention discloses an audio signal processing method and device, equipment, a storage medium and a product, and belongs to the technical field of signal processing. According to the method, the input first signal can be acquired, and the audio signal type of the first signal is identified, so that the audio signal type of the audio signal included in the first signal is analyzed. The dynamic range control curves of the audio signals under different audio signal types can be the same or different, and are customized by fully considering the characteristics of the audio signal types in energy distribution and playing performance, so that a better dynamic range control effect can be achieved. For each related audio signal type included in the first signal, extracting a corresponding characteristic dynamic range control curve to obtain a target dynamic range control curve corresponding to the first signal, therefore, a good control effect can be achieved by performing dynamic range control synthesis on the first signal based on the target dynamic range control curve.

The utility model discloses a wide dynamic range power detection circuit, which relates to the field of radio communication detection circuits, aims to obtain a wide range and high precision detection circuit, and adopts the technical scheme that the wide dynamic range power detection circuit comprises a coupling circuit, the coupling circuit is connected with a detected signal, the output end of the coupling circuit is connected with an amplifying circuit and then connected with a filter circuit, and the amplifying circuit is connected with the filter circuit. A main path of the coupling circuit is in direct connection with a signal output end; the rear end of the filter circuit is connected with a level detection circuit, the level detection circuit adopts a logarithmic amplification level conversion circuit, the rear end of the level detection circuit is connected with a microprocessor with an ADC (Analog to Digital Converter) port, and the microprocessor is connected with a display screen. After the detected signal is coupled, amplified and filtered, the precision and accuracy of the detected signal can be better ensured, so that the precision meeting the requirement can be obtained while a wide dynamic range is obtained.

A computing device capable of controlling a networked media playback system receives data identifying a first zone player, a second zone player, and a zone scene including a predefined grouping of zone players including at least the first zone player and the second zone player. The computing device displays a representation of available zone players in the networked media playback system including a first selectable indication of the first zone player, a second selectable indication of the second zone player, and a third selectable indication of the zone scene that, when selected, causes the zone scene to be invoked such that the predefined grouping of zone players becomes configured for synchronous playback of media. While displaying the representation, the computing device receives a user input indicating that the zone scene has been selected to be invoked. Based on the user input, the computing device causes the zone scene to be invoked.

To provide a method and system for adaptive equalization for an acoustic system.SOLUTION: A method comprises: equalizing an input signal that a signal input module 102 receives through a standard fixed equalization module 104; processing equalization parameters by an adaptive equalization module 110 capable of dynamic adjustment; and outputting by an output module 114 via DRC 112. A signal detection and energy calculation module 106, on the other hand, detects the input signal that the input module 102 receives, and calculates signal energy. A dynamic gain calculation and leveling module 108 calculates a dynamic gain on the basis of the calculated signal energy, and dynamically adjusts parameters of the adaptive equalization module 110 on the basis of the calculated dynamic gain.SELECTED DRAWING: Figure 1

A computing device capable of controlling a networked media playback system receives data identifying a first zone player, a second zone player, and a zone scene including a predefined grouping of zone players including at least the first and second zone players. The computing device displays a representation of available zone players in the networked media playback system including a first selectable indication of the first zone player, a second selectable indication of the second zone player, and a third selectable indication of the zone scene that, when selected, causes the zone scene to be invoked such that the predefined grouping of zone players becomes configured for synchronous playback of media. While displaying the representation, the computing device receives a user input indicating that the zone scene has been selected to be invoked. Based on the user input, the computing device causes the zone scene to be invoked.

A method (100) for real-time processing of an audio stream in an audio speaker arrangement comprising at least one audio speaker is proposed. The method (100) comprises receiving (110) an audio stream and a playback volume associated with the audio stream, and determining (120) a playback sound pressure level of the audio speaker based on the playback volume and pre-stored acoustic transmission data. The pre-stored acoustic transmission data associates the playback volume with the playback sound pressure level of the audio speaker. The method further comprises generating (130) a compensation filter based on amplitude compensation data associated with the determined playback sound pressure level, and filtering the audio stream using the compensation filter (140) and applying a gain factor to the audio stream, thereby generating a compensated audio stream. The gain factor is configured such that a total audio power of the audio stream is maintained regardless of the compensation filter. The compensated audio stream is provided (150) to the audio speaker (21). In addition, an audio speaker arrangement and a computer program product are proposed.

A signal processing method comprises comparing a first frequency domain representation of a sequence of power values for respective windows of source input samples of a source input signal with a second frequency domain representation of a sequence of power values for respective windows of target input samples of a target input signal so as to generate a frequency domain difference representation; inverse-frequency-transforming the frequency domain difference representation to generate a modification indication; and applying the modification indication to the source input samples to generate respective output samples of an output signal.