70results 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.

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.

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.

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 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.

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.

The invention relates to a device for adjusting the volume of an audio signal in an audio system, the audio system having an input for receiving the audio signal from an input source and an output for outputting the audio signal, the device having a determination unit and an adjustment unit, and a determination unit configured to determine a loudness of the audio signal, and an adjustment unit configured to adjust a level of the audio signal to a target value. The apparatus also has an attenuation unit connected downstream of the adjustment unit and configured to attenuate the audio signal adjusted by the adjustment unit by a predetermined attenuation value based on the determined loudness.

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

To provide a companding method and system to reduce encoding noise in an audio codec.SOLUTION: A compression process includes: reducing a dynamic range of an audio signal; dividing the audio signal into a plurality of time segments using a defined window shape; calculating a wideband gain for each time segment in a frequency domain using a non-energy based average of a frequency domain representation of the audio signal; and applying individual gain values so as to amplify segments of relatively low intensity and attenuate segments of relatively high intensity. Next, the compressed audio signal is expanded to return to a substantially original dynamic range. Inverse gain values are applied to amplify the segments of relatively high intensity and attenuate the segments of relatively low intensity. A QMF filter bank is used to obtain a frequency domain representation.SELECTED DRAWING: Figure 1

An electronic device outputting a sound according to embodiments of the disclosure may comprise memory storing at least one instruction, a processor executing the at least one instruction, at least one speaker, and an audio processing module processing the input audio signal and outputting the processed input audio signal to the at least one speaker. The audio processing module may include a digital amplifier amplifying a digital audio signal, a digital-to-analog converter (DAC) converting the digital audio signal into an analog audio signal, an analog amplifier amplifying the analog audio signal, a boost amplifier boosting the analog audio signal to a reference high voltage or more, and at least one noise gate removing noise included in the analog audio signal. The digital amplifier may apply a positive (+) digital gain to the digital audio signal. The analog amplifier may apply a negative (−) analog gain to the analog audio signal.

Various embodiments include active noise reduction (ANR) headsets and methods for controlling such headsets. In some implementations, the headset includes at least one electroacoustic transducer, a power supply for powering at least one electroacoustic transducer, and a control circuit configured to apply active noise reduction (ANR) to ambient noise using at least one electroacoustic transducer, sample a voltage drop from the power supply, and adjust a compressor threshold for the ANR based on the sampled voltage drop from the power supply.

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.

Aspects of the present invention relate to a method and system for improving the dialogue intelligibility of an original audio signal including dialogue and non-dialogue components. The method includes providing the dialogue components of the original audio signal in a first separated audio signal, providing the non-dialogue components of the original audio signal in a second separated audio signal, and processing the first separated audio signal and the second separated audio signal separately, wherein processing the first and second separated audio signals includes processing the loudness of the first separated audio signal and / or the second separated audio signal, and combining the processed first and second separated audio signals to provide a processed audio signal.