43results about "Digital/coded signal control" patented technology

Described herein is a method of metadata-based dynamic processing of audio data for playback, the method including: receiving, by a decoder, a bitstream including audio data and metadata for dynamic loudness adjustment; decoding, by the decoder, the audio data and the metadata to obtain decoded audio data and the metadata; determining, by the decoder, from the metadata, one or more processing parameters for dynamic loudness adjustment based on a playback condition; applying the determined one or more processing parameters to the decoded audio data to obtain processed audio data; and outputting the processed audio data for playback. Described is further a method of encoding audio data and metadata for dynamic loudness adjustment into a bitstream. Moreover, described are a respective decoder and encoder, a respective system and computer program products.

Certain aspects of the present disclosure provide techniques for configuring sidelink slots with multiple automatic gain control symbols. One aspect provides a method for wireless communication by a user equipment, including enabling a slot configuration comprising at least two symbols configured for use by a receiver for automatic gain control in preconfigured symbol locations within a slot, and transmitting the at least two symbols while transmitting symbols within the slot.

The technical scheme of the present application provides a dynamic range control circuit, an audio processing chip and an audio processing method thereof, wherein the dynamic range control circuit has a plurality of parallel gain generation modules and an output module, and each gain generation module has a separate gain generator and a gain smoothing processing module. In the same gain generation module, the gain generator is used for gain processing of an input signal, and the gain smoothing processing module is used for setting the compression time and release time of the gain generation module based on the output gain of the gain generator. It can be seen that in each gain generation module, the compression time and release time of each gain generation module can be set by the respective smoothing processing module, so that the compression time and release time of different gain generation modules are separated, which can greatly improve the tuning of the audio power amplifier and the user's auditory perception.

The present application describes a method for metadata-based dynamic processing of audio data for playback, the method comprising: receiving, by a decoder, a bitstream including audio data and metadata for dynamic loudness adjustment; decoding the audio data and metadata to obtain decoded audio data and metadata; determining from the metadata one or more processing parameters for dynamic loudness adjustment based on playback conditions; applying the determined one or more processing parameters to the decoded audio data to obtain processed audio data; and outputting the processed audio data for playback. Further, a method is described for encoding the audio data and metadata for dynamic loudness adjustment into a bitstream. Further, respective data and encoders, respective systems and computer program products are described.

This application provides a volume adjustment method, an electronic device, and a system. The method includes: In response to a user operation of starting a hearing test, an electronic device separately plays preset audio at a plurality of volumes, where the plurality of volumes include a first volume; determines, based on a confirmation operation for the first volume, that a hearing level of a user is a first hearing level corresponding to the first volume; when playing a downlink signal by using a sound-emitting apparatus, collects the downlink signal and an ambient noise signal by using at least one microphone; and adjusts a play volume of the sound-emitting apparatus from a first sound level to a second sound level based on the first hearing level when at least one of the collected downlink signal and the collected ambient noise signal changes. Embodiments of this application are implemented, so that the electronic device can adjust the play volume of the sound-emitting apparatus based on the ambient noise signal, the downlink signal, and the hearing level of the user, to improve user experience.

A method of avoiding clipping in audio signals is disclosed (figure 2). Sound signals are amplified in a pre-amplifier that successively uses a plurality of analogue gain values to generate a pluralit

There is provided a method of amplifying a signal. The method comprises: applying analogue gain to a signal, an analogue gain amount being applied at an offset amount below a set gain amount; applying digital gain to the signal, a digital gain amount applied being an amount up to the offset amount; and converting the signal between digital and analogue between the steps of applying gain. The digital gain amount is adjustable based on an amplitude of the signal. An amplifier is also provided with an analogue gain module, a digital gain module and a converter connected between the modules.

The present invention discloses a dynamic range control circuit, an audio processing chip and an audio processing method thereof, which have a first gain branch, a second gain branch and an output module. The first gain branch includes a plurality of first gain generation modules connected in series. The second gain branch includes a plurality of second gain generation modules connected in parallel. The output module is used to output target data based on the output gain of the first gain branch, the output gain of the second gain branch and the initial input data. The first gain generation module and the second gain generation module each have a separate gain generator and a gain smoothing processing module. It can be seen that in each gain generation module, different compression times and release times can be set through their respective smoothing processing modules, so that the compression times and release times of different gain generation modules are separated, which can greatly improve the tuning of the audio amplifier and the user's auditory experience.

In one aspect, a network may perform a method that includes determining one or more slots among a plurality of slots that allow an analog gain change of an automatic gain control (AGC) process, and signaling one or more of the plurality of slots that allow the analog gain change to a user equipment (UE) coupled to or in communication with the network. The network may schedule the downlink based on one or more of the plurality of slots that allow analog gain variations.

Described herein is a method of metadata-based dynamic processing of audio data for playback, the method including: receiving, by a decoder, a bitstream including audio data and metadata for dynamic loudness adjustment; decoding, by the decoder, the audio data and the metadata to obtain decoded audio data and the metadata; determining, by the decoder, from the metadata, one or more processing parameters for dynamic loudness adjustment based on a playback condition; applying the determined one or more processing parameters to the decoded audio data to obtain processed audio data; and outputting the processed audio data for playback. Described is further a method of encoding audio data and metadata for dynamic loudness adjustment into a bitstream. Moreover, described are a respective decoder and encoder, a respective system and computer program products.

A method of amplifying a signal is provided. The method includes: applying an analog gain to a signal, wherein an analog gain amount is applied with an offset lower than a set gain amount; applying a digital gain to the signal, wherein the amount of the applied digital gain is up to the offset; and converting the signal between digital and analog between the steps of applying the gain. The digital gain amount is adjustable based on the amplitude of the signal. The amplifier is further provided with an analog gain module, a digital gain module and a converter connected between the modules.

The application provides a dynamic range control method and device and electronic equipment, comprising: performing real-time amplitude detection on an input signal to obtain amplitude data; determining to-be-adjusted amplitude in the amplitude data; performing gain processing on the to-be-adjusted amplitude to obtain a current adjustment gain; and before obtaining a next adjustment gain, adjusting and outputting the input signal by using the current adjustment gain. According to the above content, the technical scheme provided by the application only needs to perform gain processing on part of the to-be-adjusted amplitude in the amplitude data, and before obtaining the next adjustment gain, the input signal is adjusted and output by using the current adjustment gain, so that gain processing on all amplitudes in the amplitude data is not needed, and the operation amount in the dynamic range control process is reduced.

The invention relates to an acoustic signal amplifier adapted for connection to a loudspeaker and for communication with a digital computer via at least one first data bus. The amplifier is characterized in that it comprises: a sampling register including several current samples captured at a sampling frequency; a first threshold volume register associated with a first threshold value; a first comparator adapted for comparing current samples from the sampling register to the first threshold value of the first volume register; a first time counter adapted for being triggered when a compared current sample is greater than the first threshold value and adapted for being reset when another compared current sample is less than the first threshold value; and an alert unit.adapted to send an alert on the first data bus to the digital computer when the first time counter reaches a first threshold time value. Figure 2a,

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 sound signal processing includes obtaining sound signals; amplifying the sound signals based on a plurality of analog gain values to generate a plurality of amplified sound signals, where each amplified sound signal corresponds to a respective analog gain value; digitizing the plurality of amplified sound signals to produce plurality of digital sound signals; determining whether the plurality of digital sound signals are clipped or non-clipping, and selectively outputting the non-clipping sound signals; and applying a dynamic range control program to the non-clipping sound signals to adjust them.

A method and system for processing an input audio mix, by extracting at least two source audio signals, each audio source signal representing a separate audio source, extracting audio mix information from the input audio mix, the audio mix information including at least one of audio mix semantic properties and audio mix signal properties, determining audio processing parameters based on the audio mix information, and processing the source audio signals based on the audio processing parameters to generate a processed audio mix. The processing of the sources can thus be based on properties of the input audio mix before source separation, thereby allowing a more automated source processing.

There is provided a method of separating at least one candidate audio from a sound source comprising the at least one candidate audio and a background sound using an audio separation system, the method comprising the steps of: extracting a first audio feature from the sound source; extracting a background sound feature from the sound source, wherein the background sound feature is used for identifying a correlation degree between the first audio feature and the background sound; generating a second audio feature based on the first audio feature, the background sound feature and a background control parameter, wherein the background control parameter is configured to control the background sound; and generating at least one separated audio based on the target information corresponding to the one candidate audio, the first audio feature, and the second audio feature in which the background sound is adjusted.

This application provides a volume adjustment method, an electronic device, and a system. The method includes: In response to a user operation of starting a hearing test, an electronic device separately plays preset audio at a plurality of volumes, where the plurality of volumes include a first volume; determines, based on a confirmation operation for the first volume, that a hearing level of a user is a first hearing level corresponding to the first volume; when playing a downlink signal by using a sound-emitting apparatus, collects the downlink signal and an ambient noise signal by using at least one microphone; and adjusts a play volume of the sound-emitting apparatus from a first sound level to a second sound level based on the first hearing level when at least one of the collected downlink signal and the collected ambient noise signal changes. Embodiments of this application are implemented, so that the electronic device can adjust the play volume of the sound-emitting apparatus based on the ambient noise signal, the downlink signal, and the hearing level of the user, to improve user experience.

The invention discloses a wide gain range VGA circuit with SAR-AGC, and relates to the technical field of integrated circuits, and the VGA circuit comprises an SAR-AGC module and a VGA system. The VGA system has differential input and differential output or has single-ended input and single-ended output; the output end of the VGA system is connected with the SAR-AGC module, and the output end of the SAR-AGC module is connected with the VGA system; the output of the VGA system is the input of the SAR-AGC module, and the output of the SAR-AGC module is a gain control signal of the VGA system; if the VGA system adopts a DVGA system, the gain control signal output by the SAR-AGC module is a digital code; and if the VGA system adopts an AVGA system, the gain control signal output by the SAR-AGC module is an analog signal. The SAR-AGC module provided by the invention has an extremely wide automatic gain control range and occupies a small area.

A method for use in an audio system including a plurality of amplifier channels of a corresponding audio channel. Each of the amplifier channels receives a respective input audio signal and is powered by the same power supply. The method includes obtaining (810) a value of an indicator, the indicator representing an energy reserve of the power supply, and for each of the amplifier channels, determining (830) a gain to be applied to the respective input audio signal by an attenuation function and a clipping level to be applied to the output of the attenuation function by a clipping function, wherein when the value of the indicator is below a threshold, the gain decreases over time, when the value of the indicator is above the threshold, the gain increases over time to a maximum value, the threshold represents a nominal energy level at which the power supply is in a nominal operating state, and the clipping level is determined as a monotonic function of the value of the indicator.

A modulator circuit may include an integrator circuit and a gain compensation circuit. The integrator circuit may include a first amplifier including an input to receive a signal and an output to provide an inverted signal and a capacitor including a first terminal coupled to the input and a second terminal coupled to the output. The gain compensation circuit may include a second inverter amplifier including a gain input coupled to the output of the first amplifier and including a gain output; a first resistor including a first terminal coupled to the gain output and including a second terminal; and a second capacitor including a first terminal coupled to the second terminal of the first resistor and including a second terminal coupled to the gain input. The integrator circuit has a finite DC gain and the gain compensation circuit is configured to compensate for the finite DC gain.

Described herein is a method of metadata-based dynamic processing of audio data for playback, the method including: receiving, by a decoder, a bitstream including audio data and metadata for dynamic loudness adjustment; decoding, by the decoder, the audio data and the metadata to obtain decoded audio data and the metadata; determining, by the decoder, from the metadata, one or more processing parameters for dynamic loudness adjustment based on a playback condition; applying the determined one or more processing parameters to the decoded audio data to obtain processed audio data; and outputting the processed audio data for playback. Described is further a method of encoding audio data and metadata for dynamic loudness adjustment into a bitstream. Moreover, described are a respective decoder and encoder, a respective system and computer program products.

Provided is an audio system in which an audio signal to be heard can be automatically set to an appropriate volume or sound quality. The audio system is provided with: a mobile terminal device provided with a microphone unit that collects surrounding sound waves, and a transmission unit that, on the basis of the sound waves collected by the microphone unit, sets the volume or sound quality of an audio signal to be amplified by an audio device, and transmits the set audio signal volume or sound quality information to the audio device; and an audio device provided with an amplification unit that amplifies the audio signal, a control unit that adjusts the audio signal volume or sound quality by means of the amplifier, and a receiving unit that receives the audio signal volume or sound quality setting information from the mobile terminal device. The control unit controls the amplification unit so as to adjust the audio signal volume or sound quality on the basis of the audio signal volume or sound quality setting information received by the receiving unit.

An audio mixing device includes a gain setting circuit configured to set first to n-th gains based on a command input from the outside, n being an integer of two or more, and a mixing circuit configured to output a mixing signal obtained by mixing two or more of first to n-th multiplication data obtained by respectively multiplying first to n-th audio data by the first to n-th gains. In a case where the j-th gain of the first to n-th gains is set to a first value, when the j-th gain is to be set to a second value different from the first value, the gain setting circuit transitions the j-th gain from the first value to the second value in a stepwise manner in a j-th gain change section of first to n-th gain change sections.

The present application discloses an audio recording method, device, storage medium and computer equipment, the method is applied to a terminal device, the terminal device includes at least one microphone, at least one amplifier and at least one analog-to-digital converter, each microphone corresponds to an amplifier, and each amplifier corresponds to an analog-to-digital converter; the method includes: obtaining an original audio signal collected by the microphone; obtaining a first audio signal and a second audio signal output by the analog-to-digital converter corresponding to the microphone, wherein the analog-to-digital converter outputs a first audio signal corresponding to the original audio signal based on a first gain parameter, the first gain parameter of the amplifier is amplified to a second gain parameter, and the analog-to-digital converter outputs a second audio signal corresponding to the original audio signal based on the second gain parameter; determining a target audio signal in the first audio signal and the second audio signal, and saving the target audio signal. The present application reduces the cost of the audio recording device.

A method and system for processing input audio mixing by extracting at least two source audio signals, each audio source signal representing an independent audio source; extracting audio mixing information from the input audio mixing, wherein the audio mixing information comprises at least one of an audio mixing semantic attribute and an audio mixing signal attribute; determining audio processing parameters based on the audio mixing information; and processing the source audio signal based on the audio processing parameters to generate a processed audio mix. Thus, processing of the source may be based on attributes of the input audio mix prior to source separation, thereby enabling more automated source processing.

There is provided a method of amplifying a signal, which may be an audio signal. The method comprises: applying analogue gain 10 to a signal, an analogue gain amount being applied at an offset amount

A method for use in an audio system comprising a plurality of amplifier channels corresponding to the audio channels. Each of the amplifier channels receives a respective input audio signal and is powered by the same power source. The method comprises: obtaining (810) a value of an indicator, where the indicator represents an energy reserve of the power source; determining (830), for each of the amplifier channels, a gain to be applied to the respective input audio signal by an attenuation function and a clipping level to be applied to an output of the attenuation function by a clipping function; wherein, when the value of the indicator is below a threshold value, the gain decreases over time, and when the value of the indicator is above the threshold value, the gain increases over time to a maximum value, where the threshold value represents a nominal energy level at which the power supply is in nominal operation; wherein the clipping level is determined as a monotonic function of the value of the indicator.

A method includes, in response to a user operation of starting a hearing test, an electronic device separately plays preset audio at a plurality of volumes, where the volumes include a first volume; determines, based on a confirmation operation for the first volume, that a hearing level of a user is a first hearing level corresponding to the first volume; when playing a downlink signal by using a sound-emitting apparatus, collects the downlink signal and an ambient noise signal by using at least one microphone; and adjusts a play volume of the sound-emitting apparatus from a first sound level to a second sound level based on the first hearing level when at least one of the collected downlink signal and the collected ambient noise signal changes.

Amplifier circuit with programmable gain, featuring the following: at least one device pin (208A) designed to be coupled to an external gain adjustment resistor component (RG); a resistor component chain (202) comprising several resistor components (206A-206H); a comparator chain (204) with several comparator circuits (U1-U8), wherein the several comparator circuits (U1-U8) have the following: first inputs, which are coupled in such a way that they receive voltages generated by corresponding components of the multiple resistors; and second inputs that are coupled in such a way that they receive a voltage applied to at least one device pin (208A); a logic circuit (210) with multiple inputs coupled to corresponding outputs of the multiple comparator circuits (U1-U8); and an amplifier component that is adjustable by the logic circuit (210) based on outputs of the multiple comparator circuits (U1-U8), wherein the logic circuit (210) is configured to adjust the gain of the amplifier component based on a comparison between the voltages from the resistor component chain (202) and the voltage applied to the at least one device pin (208A).