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

A method for decoding an encoded audio signal is described. The encoded audio signal comprises a sequence of frames, and is indicative of a plurality of different dynamic range control (DRC) profiles for a corresponding plurality of different rendering modes. The method comprises determining a first rendering mode from the plurality of different rendering modes; determining one or more DRC profiles from a subset of DRC profiles comprised within a current frame of the sequence of frames; determining whether at least one of the one or more DRC profiles is applicable to the first rendering mode; selecting a default DRC profile as a current DRC profile, if none of the one or more DRC profiles is applicable to the first rendering mode; wherein definition data of the default DRC profile is known at a decoder; and decoding the current frame using the current DRC profile.

The present document relates to processing of multimedia data, notably the encoding, the transmission, the decoding and the rendering of multimedia data, e.g. audio files or bitstreams. In particular, the present document relates to the implementation of loudness control in multimedia players. A method for providing loudness related data to a media player is described. The method comprises the steps of providing a first loudness related value associated with an audio signal; wherein the first loudness related value has been determined according to a first procedure; of converting the first loudness related value into a second loudness related value using a reversible relation; wherein the second loudness related value is associated with a second procedure for determining loudness related values; of storing the second loudness related value in metadata associated with the audio signal; and of providing the metadata to the media player.

In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

A method for decoding an encoded audio signal is described. The encoded audio signal comprises a sequence of frames, and is indicative of a plurality of different dynamic range control (DRC) profiles for a corresponding plurality of different rendering modes. The method comprises determining a first rendering mode from the plurality of different rendering modes; determining one or more DRC profiles from a subset of DRC profiles comprised within a current frame of the sequence of frames; determining whether at least one of the one or more DRC profiles is applicable to the first rendering mode; selecting a default DRC profile as a current DRC profile, if none of the one or more DRC profiles is applicable to the first rendering mode; wherein definition data of the default DRC profile is known at a decoder; and decoding the current frame using the current DRC profile.

The present application provides a method for controlling illumination effect, an electronic device and a storage medium. The method includes: obtaining sound signals, and constructing an impulse response filter according to a sound field type of the sound signals; filtering the sound signals by using the impulse response filter and obtaining analog signals; transforming the analog signals and obtaining a plurality of spectrum signals; calculating a target excitation value for each of the plurality of spectrum signals, and calculating human-perceived loudness of each of the plurality of spectrum signals according to the target excitation value; controlling a lighting effect of at least one lighting device according to the human-perceived loudness of the sound signals. The above method is able to improve an accuracy for controlling illumination effect.

A signal processing method for processing audio signals and in particular for adjusting the tone of music signals is disclosed. The method provides adaptive tonality control. Here, a shape dependent filter magnitude response is calculated, a level dependent filter magnitude response is calculated, a result response is calculated based on the shape and level dependent filter magnitude responses, and an output signal is calculated by convolving an input signal with the result response. In particular, the calculation of the shape dependent filter magnitude response comprises at least the steps of calculating a magnitude spectrum of a signal, filtering the resulting magnitude spectrum using a first smoothing filter, applying a predetermined weighting curve to the spectrum, detrending the weighted spectrum by fitting a curve to said weighted spectrum and subtracting said fitted curve from said weighted spectrum, and inverting the resulting spectrum to form the shape dependent filter magnitude response.

The present document relates to processing of multimedia data, notably the encoding, the transmission, the decoding and the rendering of multimedia data, e.g. audio files or bitstreams. In particular, the present document relates to the implementation of loudness control in multimedia players. A method for providing loudness related data to a media player is described. The method comprises the steps of providing a first loudness related value associated with an audio signal; wherein the first loudness related value has been determined according to a first procedure; of converting the first loudness related value into a second loudness related value using a reversible relation; wherein the second loudness related value is associated with a second procedure for determining loudness related values; of storing the second loudness related value in metadata associated with the audio signal; and of providing the metadata to the media player.

Some noise compensation methods involve receiving microphone signals corresponding to ambient noise from a noise source location in or near an audio environment, determining or estimating a listener position in the audio environment and estimating at least one critical distance, which is a distance from the noise source location at which directly propagated sound pressure is equal to diffuse field sound pressure. Some examples involve estimating whether the listener position is within the at least one critical distance and implementing a noise compensation method for the ambient noise based, at least in part, on an estimate of whether the listener position is within the critical distance.

The present document relates to processing of multimedia data, notably the encoding, the transmission, the decoding and the rendering of multimedia data, e.g. audio files or bitstreams. In particular, the present document relates to the implementation of loudness control in multimedia players. A method for providing loudness related data to a media player is described. The method comprises the steps of providing a first loudness related value associated with an audio signal; wherein the first loudness related value has been determined according to a first procedure; of converting the first loudness related value into a second loudness related value using a reversible relation; wherein the second loudness related value is associated with a second procedure for determining loudness related values; of storing the second loudness related value in metadata associated with the audio signal; and of providing the metadata to the media player.

The present application is applicable to the technical field of audio processing, and provides an audio compression method and an audio compression apparatus. The method includes: obtaining a first audio signal to be processed; performing, by using a spectrum compensation filter, filtering processing on the first audio signal based on a preset target spectrum to obtain a second audio signal, where the target spectrum is correlated with a distorted frequency band of the audio playing device; using a preset dynamic range compressor (DRC) to process the second audio signal to obtain a compression coefficient; and outputting a compressed signal according to the compression coefficient. The method may expand a compression range, reduce an occurrence rate of distortion when playing the audio signal and improve a playing quality of the audio signal, without significantly reducing the gain of the loud sound frequency band.

To provide an audio processing unit, an audio processing method and a non-temporary medium, which decode a bitstream.SOLUTION: A method by a decoder for decoding an encoded audio bitstream including program loudness metadata and audio data includes adaptive loudness processing of audio data of an audio program indicated by the bitstream or processing for performing authentication and / or validation confirmation of metadata and / or audio data of such audio program. The decoder includes a buffer memory for storing at least one frame of the audio bitstream encoded based on any embodiment of the method.SELECTED DRAWING: Figure 3

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.

Provided are an information processing device and an information processing method capable of correcting the frequency characteristics of sound output from a speaker and making it easier to hear the sound clearly. The information processing apparatus includes: a volume level information acquisition unit that acquires volume level information relating to a setting of a volume level by a user; and a signal processing unit that corrects the frequency characteristic of the sound output from the speaker, on the basis of the volume level, such that the attenuation amount from the maximum volume of the sound output from the speaker at a prescribed frequency falls within a volume lower limit value range with respect to the maximum volume at the prescribed frequency, and corrects the frequency characteristic of the sound output from the speaker, on the basis of the volume level. The volume lower limit value is based on the difference between the maximum volume of the speaker at the reference frequency and the minimum volume that can be set by the sound output from the speaker. The sound pressure level difference is determined according to a first sound pressure level difference between the maximum volume under the reference frequency and the minimum audible volume which can be heard by people and a second sound pressure level difference between the maximum volume and the minimum audible volume under other frequencies different from the reference frequency.

The present invention provides an information processing device and an information processing method that correct the frequency characteristics of sound output from a speaker to make it easier to hear. [Solution] The information processing device includes a volume level information acquisition unit that acquires volume level information related to the volume level setting by the user, and a signal processing unit that corrects the frequency characteristics of the sound output from the speaker so that the amount of attenuation from the maximum volume of the sound output from the speaker at a predetermined frequency is within a lower volume limit relative to the maximum volume at the predetermined frequency, according to the volume level. The lower volume limit is determined based on the difference between the maximum volume of the speaker at a reference frequency and the minimum volume that can be set as the sound output from the speaker, according to a first sound pressure level difference between the maximum volume at a reference frequency and the minimum audible volume that a person can hear, and a second sound pressure level difference between the maximum volume and the minimum audible volume at a frequency other than the reference frequency.

To provide optimization of the loudness and the dynamic range across different playback devices.SOLUTION: Embodiments are directed to a method and system for receiving, in a bitstream, metadata associated with the audio data, and analyzing the metadata to determine whether a loudness parameter for a first group of audio playback devices are available in the bitstream. In response to determining that the parameters are present for the first group, the system uses the parameters and audio data to render audio. In response to determining that the loudness parameters are not present for the first group, the system analyzes one or more characteristics of the first group, and determines the parameter on the basis of the one or more characteristics.SELECTED DRAWING: Figure 16

A method for processing an audio signal to be supplied to a device (3) that converts the audio signal into sound, wherein an electronic device (1) that processes the stored audio signal stores: a plurality of sound setting information (53), each specifying the content of processes on the audio signal, and a plurality of model information elements (51), each containing information elements about models of the device, wherein the plurality of sound setting information (53) each correspond to different types of sounds, the plurality of model information elements (51) each are associated with one or more sound setting information elements (53) from the plurality of sound setting information elements (53), wherein the one or more sound setting information elements (53) correspond to one or more types of sound from the various types of sound, wherein the method comprises: capturing,by the electronic device (1), the plurality of model information (51) about the models of the device to which the audio signal, which has undergone the processes, is supplied; selecting, by the electronic device (1), a sound setting information element (53) from the plurality of sound setting information elements associated with the acquired plurality of model information elements (51), wherein the one sound setting information element (53) corresponds to one type among the various types of sound, the one type being specified in accordance with an instruction to be entered; processing the audio signal by the electronic device in accordance with the selected sound setting information (53); acquiring sound type information (52) by the electronic device (1), which indicates the various types of sound,which is expressed by the audio signal to be subjected to the processes; and selecting, by the electronic device (1), a sound setting information element (53) other than a preset sound setting information element (53) from the plurality of sound setting information elements (53) associated with the detected plurality of model information elements (51), wherein the other sound setting information element (53) corresponds to a different type among the various types of sound, wherein the other type is indicated by a sound type information element (52) among the detected sound type information elements (52), wherein the selection of the one sound setting information (53) by the electronic device (3) includes selecting, in accordance with the instruction to be entered, the one sound setting information (53) that is the same as or different from the preset sound setting information (53),characterized by the acquisition of sound type information (52) by the electronic device (1), including the analysis of features of the sound expressed by the audio signal to be subjected to the process, and the differentiation of the different types of sound expressed by the audio signal to be subjected to the process by applying the analyzed features of the sound to a learned model that was generated to differentiate the different types of sound on the basis of the analyzed features of the sound.

The present document relates to processing of multimedia data, notably the encoding, the transmission, the decoding and the rendering of multimedia data, e.g. audio files or bitstreams. In particular, the present document relates to the implementation of loudness control in multimedia players. A method for providing loudness related data to a media player is described. The method comprises the steps of providing a first loudness related value associated with an audio signal; wherein the first loudness related value has been determined according to a first procedure; of converting the first loudness related value into a second loudness related value using a reversible relation; wherein the second loudness related value is associated with a second procedure for determining loudness related values; of storing the second loudness related value in metadata associated with the audio signal; and of providing the metadata to the media player.

Noise compensation method comprising: (a) receiving a content stream including content audio data; (b) receiving first microphone signals from a first device; (c) detecting ambient noise from a noise source location in or near the audio environment; (d) causing a first wireless signal to be transmitted from the first device to a second device, the first wireless signal including instructions for the second device to record an audio segment; (e) receiving a second wireless signal from the second device; (f) determining a content stream audio segment time interval for a content stream audio segment; (g) receiving a third wireless signal from the second device, including a recorded audio segment captured via a second device microphone; (h) determining a second device ambient noise signal at the second device location; and (i) implementing a noise compensation method for the content audio data based, at least in part, on the second device ambient noise signal.

Embodiments are directed to a method and system for receiving, in a bitstream, metadata associated with the audio data, and analyzing the metadata to determine whether a loudness parameter for a first group of audio playback devices are available in the bitstream. Responsive to determining that the parameters are present for the first group, the system uses the parameters and audio data to render audio. Responsive to determining that the loudness parameters are not present for the first group, the system analyzes one or more characteristics of the first group, and determines the parameter based on the one or more characteristics.

In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

The present disclosure relates to a method and audio processing system for performing dynamic range adjustment of spatial audio objects. The method comprises obtaining (step S1) a plurality of spatial audio objects (10), obtaining (step S2) at least one rendered audio presentation of the spatial audio objects (10) and determining (step S3) signal level data associated with each presentation audio channel in said set of presentation audio channels. The method further comprises obtaining (step S31) a threshold value and, for each time segment, selecting (step S4) a selected presentation audio channel which is associated with a highest or a lowest signal level, determining (step S5) a gain based on the threshold value and the representation of the signal level of the selected audio channel, and applying (step S6) the gain of each time segment to corresponding time segments of the spatial audio objects.

This document describes a dynamic range control unit (210) configured to apply a dynamic range control, referred to as DRC, to an audio signal (211). The DRC unit (210) is configured to: downsample a subband signal (212) derived from the audio signal (211) to provide a downsampled subband signal (321); determine a DRC gain (329) based on the downsampled subband signal (321); and apply the DRC gain (329) to the subband signal (212) to provide a compressed subband signal (213) of a compressed audio signal (214).

An information processing apparatus comprises a volume level information acquirer that acquires volume level information regarding a setting of a volume level by a user; and a signal processor that corrects a frequency characteristic of an output sound from a speaker and thus causes an attenuation amount from a maximum volume of the output sound from the speaker at a predetermined frequency to be within a range indicated by a volume lower limit value relative to the maximum volume at the predetermined frequency, according to the volume level indicated by the volume level information, wherein the volume lower limit value is determined according to a first sound pressure level difference at a reference frequency between the maximum volume of the speaker and a minimum audible volume that a human is capable of hearing and a second sound pressure level difference at different frequency different from the reference frequency between the maximum volume and the minimum audible volume, with a difference at the reference frequency between the maximum volume and a minimum volume that is settable for the output sound from the speaker as a reference.

An adaptive control system may identify, from a user's natural language instructions, an audio parameter of an audio rendering system that the user wishes to adjust. The adaptive control system may identify the context in which the user is providing the instructions, where the context can include sensor information characterizing an environment around the user, device information characterizing the device through which the user is consuming audio, or states of audio parameters as tracked on a parametric space. The adaptive control system may input data characterizing the context into one or more machine learning models to determine a likely audio parameter and a corresponding degree of change the user is requesting through their instruction. The adaptive control system may generate recommended adjustments to audio parameters using machine learning based on the context in which the user is utilizing a controllable system.

A method for performing loudness normalization, the method being performed by an audio signal processing device, includes the steps of receiving an audio signal; receiving information about an integrated loudness of the audio signal; receiving information about a target loudness of the audio signal; correcting the integrated loudness based on one or more processing steps to obtain a corrected integrated loudness; and normalizing the audio signal based on the corrected integrated loudness and the target loudness to obtain a normalized audio signal.

A method for decoding an encoded audio signal is described. The encoded audio signal comprises a sequence of frames, and is indicative of a plurality of different dynamic range control (DRC) profiles for a corresponding plurality of different rendering modes. The method comprises determining a first rendering mode from the plurality of different rendering modes; determining one or more DRC profiles from a subset of DRC profiles comprised within a current frame of the sequence of frames; determining whether at least one of the one or more DRC profiles is applicable to the first rendering mode; selecting a default DRC profile as a current DRC profile, if none of the one or more DRC profiles is applicable to the first rendering mode; wherein definition data of the default DRC profile is known at a decoder; and decoding the current frame using the current DRC profile.

An exemplary audio enhancement system substantially eliminates latency by returning audio input signals in amplified form directly in the analog domain to the source, thereby reducing signal degradation and removing redundancy from digital and analog audio transmission and processing architectures.

It provides efficient dynamic range control (DRC) profile transmission. [Solution] The encoded audio signal 102 has a sequence of frames and represents multiple different DRC profiles for a plurality of different rendering modes. The method includes the steps of: determining a first rendering mode from a plurality of different rendering modes; determining one or more DRC profiles from a subset of DRC profiles contained in the current frame of the sequence of frames; determining whether at least one of the one or more DRC profiles is applicable to the first rendering mode; selecting a default DRC profile if none of the one or more DRC profiles are applicable to the first rendering mode; and decoding the current frame using the current DRC profile.

The present application is applicable to the technical field of audio processing. Provided are an audio compression method and apparatus. The method comprises: acquiring a first audio signal to be processed; on the basis of a preset target spectrum, using a spectrum compensation filter to perform filtering processing on the first audio signal, so as to obtain a second audio signal, wherein the target spectrum is related to a distortion frequency band of an audio playing device; processing the second audio signal by means of a preset dynamic range compressor (DRC), so as to obtain a compression coefficient; and on the basis of the compression coefficient, outputting a compressed signal. The method can expand the compression range without significantly reducing the gain of high-volume frequency bands, thereby reducing the occurrences of distortion during audio signal playing, and improving playing quality.

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.