183results about "Digital/coded signal combination control" patented technology

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 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.

An audio device is provided. The audio device comprises a controller and an acoustic transducer. The controller is configured to receive an input audio signal. The controller is further configured to generate, via a loudness detector, a loudness level of a first portion of the input audio signal. The controller is further configured to identify a compression curve of a plurality of compression curves corresponding to a user volume setting. The controller is further configured to adjust, via an audio compressor, a second portion of the input audio signal based on the loudness level and the compression curve to generate an output audio signal. The loudness level is determined according to LUFS model.

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.

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.

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.

Systems and methods for digital audio measurement such as perceived loudness. An example of the method includes dividing, by a processor, a digital audio file into a plurality of blocks in a time sequence at a first time length; determining, by the processor, a respective Loudness Units relative to Full Scale (LUFS) for each block; determining, by the processor, a difference between the LUFS of each block with a reference value; and generating, by the processor, an indicator from one or more of the differences of the plurality of blocks within a second time length.

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.

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.

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 speech equalization, comprising the steps of receiving an input audio signal, processing said input audio signal in dependence on frequency and to providing an equalized electric audio signal according to an equalization function, wherein said equalization function comprises at least an actuator part configured to dynamically applying a compensation filter to the received input signal and dynamically applying a transparent filter to the received input signal, and further transmitting an output signal perceivable by a user as sound representative of said electric acoustic input signal or a processed version thereof.

The invention relates to a method for operating a hearing system (2), in which a dynamic scheme for an automatic gain control is modified depending on the situation. Furthermore, a corresponding hearing system (2) is disclosed.

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.

An audio system for reproducing amplified output audio signals by at least one loudspeaker of the audio system. The audio system has a signal processing unit for actuating the loudspeaker, including a digital signal processor and at least one digital / analog converter and at least one amplifier. The signal processing unit is connected / connectable to a data source for providing input audio signals to be amplified, or the data source is integrated into the signal processing unit. The audio system has a reproduction mode and a hearing test mode and additionally a data storage device for storing a result of the at least one, frequency-dependent listening test carried out with the audio system. The loudspeaker is a multi-path loudspeaker having at least two mutually different frequency band speakers for reproducing the amplified output audio signal, with the at least two mutually different frequency band speakers each having distinct useful frequency bands.

Apparatus and methods for generating an encoded audio bitstream, including by including program loudness metadata and audio data in the bitstream, and optionally also program boundary metadata in at least one segment (e.g., frame) of the bitstream. Other aspects are apparatus and methods for decoding such a bitstream, e.g., including by performing adaptive loudness processing of the audio data of an audio program indicated by the bitstream, or authentication and / or validation of metadata and / or audio data of such an audio program. Another aspect is an audio processing unit (e.g., an encoder, decoder, or post-processor) configured (e.g., programmed) to perform any embodiment of the method or which includes a buffer memory which stores at least one frame of an audio bitstream generated in accordance with any embodiment of the method.

A dynamic range control unit (210) configured to apply 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), to determine a DRC gain (329) based on the downsampled subband signal (321), and to apply the DRC gain (329) to the subband signal (212), to provide a compressed subband signal (213) of a compressed audio signal (214).

A method for decoding an encoded audio signal is described. The encoded audio signal may comprise a sequence of frames and may be indicative of a plurality of different dynamic range control (DRC) profiles for a corresponding plurality of different rendering modes. Different subsets of DRC profiles may be comprised within different frames. The method may comprise 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; determining whether at least one of the DRC profiles is applicable to the first rendering mode; selecting a default DRC profile as a current DRC profile, if none of the 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.

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.

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.

A lighting effect control method, an electronic device (10), and a storage medium. The method comprises: acquiring a sound signal (S11), and on the basis of a sound field type of the sound signal, constructing a corresponding pulse response filter (S12); on the basis of the pulse response filter, filtering the sound signal to obtain an analog signal (S13); transforming the analog signal to obtain a plurality of spectrum signals (S14); and calculating a target excitation value of each spectrum signal (S15), and on the basis of the target excitation value, calculating the human-perceived loudness of each spectrum signal (S16); and on the basis of the human-perceived loudness of the sound signal, controlling the lighting effect of at least one lighting device (S17). Thus, the control accuracy of the lighting effect can be improved.

Some noise compensation methods involve: receiving a microphone signal 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, the critical distance being a distance from the noise source location at which the directly propagated sound pressure equals the diffuse-field sound pressure. Some examples involve estimating whether the listener position is within the at least one critical distance, and performing the noise compensation method for the ambient noise based at least in part on the estimation of whether the listener position is within the critical distance.

To enable a patron of a venue attending a performance to feel as if he or she is ‘front and center’ even if located much farther away or angled from a stage of the venue, system and methods that enhance audio quality of audio signals especially over the voice frequency spectrum to generate a processed audio signal including an increased voice frequency spectrum are provided. The processed audio signal may be communicated to a receiving system with a headset or other transducer (e.g., loudspeakers) that allows for ambient sounds to be heard by a patron and for the patron to control volume of the processed audio signal, thereby mixing the processed audio signal with the ambient sounds. The receiving system may be rented and an operator may control usage for both performance or long-term rentals using various control features of the processed audio signals and / or receiving system.

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

The invention relates to a system and method for providing personalized audio streaming and rendering. Systems and methods for providing personalized audio streaming and rendering include receiving a movie audio track of selected content and a maximum accessible audio track of the content, and adjustably combining the movie audio track and the accessible audio track to provide an improved dialog audio track, the dialog audio track allows a user / listener to hear a dialog on ambient background noise, the combination being provided by a cross-tapered renderer that is manually adjusted by the user / listener or automatically adjusted based on measured background noise, and optionally providing a personalized equalizer for hearing impairment. It also allows the content provider to create a maximum accessible audio track for a given content independently of the user / listener, which may also use an X-Fade renderer to verify the quality of the accessible audio track.

The invention is related to signal processing methods for processing audio signals. Especially the invention concerns signal processing methods for adjusting the tone of music signals. The invention provides a signal processing method that provides an adaptive tonality control. In the method 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 (110) a magnitude spectrum of a signal, - filtering (120) the resulting magnitude spectrum using a first smoothing filter, - applying (130) a predetermined weighting curve to the spectrum, - detrending (140) said weighted spectrum by fitting a curve to said weighted spectrum and subtracting said fitted curve from said weighted spectrum, and - inverting (150) the resulting spectrum to form the shape dependent filter magnitude response.

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.

Techniques are described herein that are capable of adaptively modulating audio content based on background noise. For instance, a loudness difference between a loudness of the audio content and a loudness of the background noise is determined. The loudness of the audio content is compared to an upper loudness threshold. A modulation is selected from multiple modulations based on the loudness difference and / or the comparison. Subsequent outputted audio content is modulated using the selected modulation.

An audio system (1) for reproducing an amplified output audio signal by means of at least one loudspeaker, the audio system (1) having a signal processing unit (24) having a digital signal processor (2), at least one digital / analog converter (3) and at least one amplifier (4) for driving and controlling the loudspeaker, the signal processing unit (24) being connected or connectable to a data source (5) providing an input audio signal to be amplified or the data source (5) being integrated within the signal processing unit (24), the audio system (1) being connected or connectable to a data source (5) providing an input audio signal to be amplified, The sound system (1) has a playback mode of operation and a hearing test mode of operation, and further has an additional data memory (10) for storing the results of at least one frequency-related hearing test performed by the sound system (1), and the speaker is a multi-way speaker (11) including at least two mutually distinct frequency band toners (12, 13, 14) for reproducing an amplified output sound signal, the at least two mutually distinct frequency band toners (12, 13, 14) having mutually distinct effective frequency bands (15, 16, 17).