9results about "Volume compression/expansion in digital/coded amplifiers" patented technology

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.

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.

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.

It provides a delayed loudness adjustment technique for dynamic range control. [Solution] A bitstream containing an encoded version of the audio signal and an instantaneous loudness sequence of the audio signal is acquired by the decoder. The instantaneous loudness sequence is not loudness normalized. A Dynamic Range Control (DRC) gain sequence is generated by applying the instantaneous loudness sequence with loudness normalization to the DRC characteristics. This DRC gain sequence is applied to the decoded audio signal. Other embodiments are also described and claimed.

A method for dynamically controlling enhancement of an audio stream is provided, where the audio stream defines a sequence of audio segments over time. Each audio segment defines a waveform having a plurality of waveform attributes. For each audio segment of the sequence of audio segments, the method includes: (i) determining a set of waveform-attribute values of the audio segment's waveform attributes, (ii) computing a first distance between the determined set of waveform-attribute values and a first predefined set of waveform-attribute values representative of speech, and computing a second distance between the determined set of waveform-attribute values and a second predefined set of waveform-attribute values representative of music, (iii) using the computed first and second distances as a basis to classify the audio segment as primarily speech or rather primarily music, and (iv) controlling, based on the classifying, whether or not to enhance the audio segment for output.

The present disclosure relates to loudness adjustment for downmixing audio content. Audio content encoded for a reference speaker configuration is downmixed to downmixed audio content encoded for a particular speaker configuration. One or more gain adjustments are performed on individual portions of the downmixed audio content encoded for the particular speaker configuration. Loudness measurements are then performed on the individual portions of the downmixed audio content. Audio content is produced that includes the audio content encoded for the reference speaker configuration and downmix loudness metadata. The downmix loudness metadata is created based at least in part on the loudness measurements on the individual portions of the downmixed audio content.

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