98results about "Volume compression/expansion in digital/coded amplifiers" patented technology

On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters (α) from the bitstream, where 1 ≤ m < n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC2) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.

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 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.

Various aspects include active noise reduction (ANR) headsets and methods of controlling such headsets. In some implementations, a headset includes: at least one electro-acoustic transducer; a power source for powering the at least one electro-acoustic transducer; and a control circuit configured to apply active noise reduction (ANR) to environmental sound using the at least one electro-acoustic transducer, sample voltage drops across the power source, and adjust a compressor threshold for the ANR based on the sampled voltage drops across the power source.

A device for adjusting the volume of an audio signal in an audio system is proposed, wherein the audio system has an input for receiving the audio signal from an input source and an output for outputting the audio signal. The device comprises a determining unit configured to determine the loudness of the audio signal and an adjustment unit configured to regulate the level of the audio signal to a target value. The device further comprises an attenuation unit downstream of the adjustment unit, which is configured to attenuate the audio signal regulated by the adjustment unit by a predetermined attenuation value based on the determined loudness.

Methods for performing dynamic range compression (DRC) on audio in a manner intended to produce output audio for playback by systems or devices with limited power handling capabilities and preferably also to reduce or prevent undesirable artifacts (e.g., pumping and / or breathing) in the output audio. Some embodiments perform the DRC so as to maximize average loudness (while preventing loss of quieter elements) during playback, and also to reduce or prevent distortion. Other aspects are systems or devices configured to perform embodiments of the method. In some embodiments, reduced DRC is applied when average loudness of the input audio approaches (or matches or exceeds) a target (e.g., a knee point for DRC, or a signal level near to a maximum playback level of the intended playback system), since such input audio is assumed to have already been compressed, and otherwise applying full DRC to the input audio.

Aspects of the present disclosure relate to a method and system for enhancing conversation intelligibility in a raw audio signal including a conversation component and a non-conversation component. The method comprises providing a dialogue component of the original audio signal in a first individual audio signal, providing a non-dialogue component of the original audio signal in a second individual audio signal, separately processing the first individual audio signal and the second individual audio signal, wherein processing the first and second individual audio signals comprises processing the loudness of the first individual audio signal and / or the second individual audio signal, and combining the processed first and second individual audio signals to provide a processed audio signal.

A device for adjusting the volume of an audio signal in an audio system is provided, wherein the audio system has an input for receiving the audio signal from an input source and an output for outputting the audio signal. The device includes a determination unit configured to determine a loudness of the audio signal, and an adjustment unit configured to regulate a level of the audio signal to a target value. The device further includes a damping unit, downstream of the adjustment unit, which is configured to attenuate the audio signal regulated by the adjustment unit by a predetermined damping value based on the determined loudness.

An audio signal processing method, and an audio signal processing device for executing the audio signal processing method includes, obtaining an audio signal and compressing the audio signal to generate a first compressed signal; calculating a duration for which a power value of the first compressed signal equals a first power threshold; determining whether the duration exceeds a first time threshold; when the duration exceeds the first time threshold, compressing the first compressed signal to generate a second compressed signal, and amplifying and outputting the second compressed signal.

Aspects include active noise reduction (ANR) headphones and methods of controlling such headphones. In some implementations, a headphone comprises: at least one electro-acoustic transducer; the power supply is used for supplying power to the at least one electroacoustic transducer; and a control circuit configured to: apply active noise reduction (ANR) to ambient sound using the at least one electro-acoustic transducer; sampling the voltage drop at the two ends of the power supply; and adjusting a compressor threshold for the ANR based on the sampled voltage drop across the power supply.

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.

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.

An audio system with a base station connected to multiple mobile devices. The base station and the mobile devices are configured to exchange data. The base station is connected to an audio mixer configured to receive multiple input signals from the mobile devices, mix them into a single signal, and provide it as an in-ear monitoring signal. This signal is then transmitted to a mobile device that includes an output amplifier and connected transducers. The mobile device receiving the in-ear monitoring signal has a volume control that allows the user to adjust the playback volume of the in-ear monitoring signal. The mobile device also has an audio compressor that compresses the input signal level of the in-ear monitoring signal as a function of the volume setting, ensuring that the output level of the output amplifier does not exceed a predefined limit.

This invention is related to signal processing of music. The invention specifically concerns removal of unwanted artifacts from a music signal. The invention provides a method for attenuating resonances in a sound signal, which method uses subsampling of at least one sub-band of the original signal to save computing resources. As a coarse overview description, the method comprises at least steps in which an incoming signal block is filtered (120) into at least two different frequency sub-band signals, at least one of the sub-band signals is subsampled (121), a resonance analysis step is performed (125) for each sub-band signal, said step resulting in a filter response, a sub-band signal from each sub-band is convolved (130) with the corresponding filter response to produce a filtered sub-band signal, for each subsampled sub-band, the filtered sub-band signal is upsampled (140), and finally the upsampled sub-band signals of subsampled sub-bands and filtered sub-band signals of any non-subsampled sub-bands are combined (150) to form a result signal block. This method saves computing resources by performing analysis based on subsampled sub-band blocks.

Audio content coded for a reference speaker configuration is downmixed to downmix audio content coded for a specific speaker configuration. One or more gain adjustments are performed on individual portions of the downmix audio content coded for the specific speaker configuration. Loudness measurements are then performed on the individual portions of the downmix audio content. An audio signal that comprises the audio content coded for the reference speaker configuration and downmix loudness metadata is generated. The downmix loudness metadata is created based at least in part on the loudness measurements on the individual portions of the downmix audio content.

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

To provide loudness adjustment for downmixed audio content.SOLUTION: Audio content coded for a reference speaker configuration is downmixed to downmix audio content coded for a specific speaker configuration. One or more gain adjustments are performed on individual portions of the downmix audio content coded for the specific speaker configuration. Loudness measurements are then performed on the individual portions of the downmix audio content. An audio signal that comprises the audio content coded for the reference speaker configuration and downmix loudness metadata is generated. The downmix loudness metadata is created on the basis of at least in part the loudness measurements on the individual portions of the downmix audio content.SELECTED DRAWING: Figure 3

A rendering method of an object-based audio signal and an apparatus for performing the same are provided. The rendering method of an object-based audio signal includes obtaining a rendered audio signal, performing clipping prevention on the rendered audio signal using a first limiter, mixing a signal output by the first limiter using a mixer, and performing clipping prevention on the mixed signal using a second limiter.

The invention relates to a system for maintaining reversible dynamic range control information associated with a parametric audio encoder. On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and a multi-channel coding parameter (alpha) from the bitstream, where 1 < = m < n. A pre-processing dynamic range control parameter, i.e., a pre-processing DRC parameter (DRC2), that quantizes the encoder-side dynamic range limitation of the core signal is also derived from the bitstream. The n-channel audio signal is obtained by parameter synthesis according to the multi-channel encoding parameters while eliminating any encoder-side dynamic range limitations based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes the use of compensated post-processing DRC parameters that quantify possible decoder-side dynamic range compression. Preferably, encoder-side range limit cancellation and range compression are performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC preprocessor.

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 processing system includes an audio source device configured to provide an analog based audio signal; an audio converting device including an analog gain amplifier configured to amplify the analog based audio signal into an amplified analog signal, a first analog-to-digital converter (ADC) configured to convert the amplified analog signal into an amplified digital signal, a digital gain reducer configured to reduce the amplified digital signal into a first digital signal, an analog gain reducer configured to reduce the analog based audio signal into a reduced analog signal, a second ADC configured to convert the reduced analog signal into a reduced digital signal, and a digital gain amplifier configured to amplify the reduced digital signal into a second digital signal.

A class-D amplifier circuit for amplifying an input signal, the class-D amplifier circuit including a controller configured to control the class-D amplifier circuit to selectively transition between an open-loop operation mode and a closed-loop operation mode based on an indication of the amplitude of the input signal.

A method (800) comprising: - receiving (850-1) input samples of a signal to be processed by a subsequent device; - performing (850-2) a first limiting step including for at least one sample, replacing the concerned input sample with a reduced sample by applying an offset to the concerned input sample, wherein the reduced sample has the same sign, but a lower magnitude than the concerned input sample, wherein the offset is computed based on parameters of a filter modelling the behavior of the subsequent device and an overshoot difference between a threshold and a predicted output value generated by this filter.

The method (800) includes receiving (850-1) input samples of a signal to be processed by a subsequent device, clipping, and performing a limiting process on at least one of the input samples (850-2), the limiting process including generating an inter-sample prediction value by applying a sub-filter to a corresponding plurality of input samples including the relevant input sample, the sub-filter modeling the inter-sample behavior of the subsequent device, generating an inter-sample prediction value by applying a sub-filter to a corresponding plurality of input samples including the relevant input sample, and replacing the relevant input sample with a reduced sample by applying an offset to the relevant input sample, the reduced sample having the same sign as the relevant input sample but having an amplitude smaller than that of the relevant input sample, and the offset being calculated based on a parameter of the sub-filter and an overshoot difference between a threshold and one of the inter-sample prediction values.

Described herein is a method of processing audio data for playback, the method including: receiving, by a decoder, a bitstream including encoded audio data and metadata, wherein the metadata includes one or more dynamic range control (DRC) sets, and for each DRC set, an indication of whether the DRC set is configured for providing a dynamic loudness compensation effect; parsing the metadata, by the decoder, to identify DRC sets that are configured for providing the dynamic range compensation effect; decoding, by the decoder, the encoded audio data to obtain decoded audio data; selecting, by the decoder, one of the identified DRC sets configured for providing the dynamic loudness compensation effect: extracting from the bitstream, by the decoder, one or more DRC gains corresponding to the selected DRC set; applying to the decoded audio data, by the decoder, the one or more DRC gains corresponding to the selected DRC set to obtain dynamic loudness compensated audio data; and outputting the dynamic loudness compensated audio data for playback. Moreover, described are respective decoder and computer program products.

On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters (α) from the bitstream, where 1 ≤ m < n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC2) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.