76results about "Digital/coded signal combination control" patented technology

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.

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 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.

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.

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 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.

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.

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.

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).

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.

In various applications, the system provides a method for processing audio signals, including: receiving, by a processor, a digital audio signal from a recorded audio file; analyzing, by the processor, the digital audio signal to identify pitch distortion caused by changes in momentary sound level; determining, by the processor, an amount of compensation of the audio signal to correct the identified pitch distortion; dynamically adjusting, by the processor, the digital audio signal by the compensation amount to correct the identified pitch distortion; and outputting, by the processor, the digital audio signal to an audio transducer device of a listener to improve a listening experience for the listener of the recorded audio file.

The invention provides a concept for combined dynamic range compression and guided clipping prevention for audio devices. An audio decoder for decoding an audio bitstream and a metadata bitstream related to the audio bitstream according to the concept comprises an audio processing chain configured to receive a decoded audio signal derived from the audio bitstream and to adjust characteristics of the audio signal in order to produce an audio output signal, the audio adjustment chain comprising a plurality of adjustment stages including a dynamic range control stage for adjusting a dynamic range of the audio output signal and a guided clipping prevention stage for preventing clipping of the audio output signal; and a metadata decoder configured to receive the metadata bitstream and to extract dynamic range control gain sequences and guided clipping prevention gain sequences from the metadata bitstream, at least a part of the dynamic range control gain sequences being supplied to the dynamic range control stage, and at least a part of the guided clipping prevention gain sequences being supplied to the guided clipping prevention stage.

The present disclosure describes techniques of generating voices for virtual characters. A plurality of source sounds may be received. The plurality of source sounds may correspond to a plurality of frames of a video. The video may comprise a virtual character. The plurality of source sounds may be converted into a plurality of representations in a latent space using a first model. Each representation among the plurality of representations may comprise a plurality of parameters. The plurality of parameters may correspond to a plurality of sound features. A plurality of sounds may be generated in real time for the virtual character in the video based at least in part on modifying at least one of the plurality of parameters of each representation.

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.

Distortion sensing, distortion prevention, and distortion aware bass enhancement are disclosed. Systems and methods that provide distortion sensing, prevention, and / or distortion aware bass enhancement in an audio system can be implemented in a variety of applications. Sensing circuitry can generate statistics based on a received input signal for which an acoustic output is generated. In various embodiments, the statistics can be used such that a multi-notch filter can be used to provide an input to a speaker to generate the acoustic output. In various embodiments, the statistics from the sensing circuitry can be provided to a bass parameter controller coupled to a bass enhancement circuit to operatively provide a parameter to the bass enhancement circuit. The bass enhancement circuit can provide a bass enhanced signal for generation of the acoustic output based on the parameter. Various combinations of bass enhancement circuits and multi-notch filters using statistics from the sensing circuitry can be implemented to provide an enhanced acoustic output.

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 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 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.

Disclosed are systems and methods for a frontend capture module of a video conferencing application, which can modify an input signal, received from a microphone device to match predetermined signal characteristics, such as voice signal level and expected noise floor. An Input stage, a suppression module and an output stage amplify the voice signal portion of the input signal and suppress the noise signal of input signal to predetermined ranges. The input stage selectively applies gains defined by a gain table, based on signal level of the input signal. The suppression module selectively applies a suppression gain to the input signal based on presence or absence of voice signal in the input signal. The output stage further amplifies the input signal in portions having a voice signal and applies a gain table to maintain a consistent noise floor.

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.

The invention provides a concept for combined dynamic range compression and guided clipping prevention for audio devices. An audio decoder for decoding an audio bitstream and a metadata bitstream related to the audio bitstream according to the concept comprises an audio processing chain configured to receive a decoded audio signal derived from the audio bitstream and to adjust characteristics of the audio signal in order to produce an audio output signal, the audio adjustment chain comprising a plurality of adjustment stages including a dynamic range control stage for adjusting a dynamic range of the audio output signal and a guided clipping prevention stage for preventing clipping of the audio output signal; and a metadata decoder configured to receive the metadata bitstream and to extract dynamic range control gain sequences and guided clipping prevention gain sequences from the metadata bitstream, at least a part of the dynamic range control gain sequences being supplied to the dynamic range control stage, and at least a part of the guided clipping prevention gain sequences being supplied to the guided clipping prevention stage.

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.