Digital Rise Technology Co., Ltd.

Guoguang Electric Company Limited

Yuli You

Jimeng Zheng

Provided are systems, methods and techniques for compressing the dynamic range of an audio signal. In one implementation: an input audio signal is obtained, and a time-varying gain signal is provided based on it and a desired output range. The time-varying preliminary gain signal is then applied to the input audio signal to provide an output audio signal. Provision of the time-varying gain signal involves lowpass filtering a signal that is based on the input audio signal, using an attack gain-response time and a release gain-response time as filtering parameters, with the attack gain-response time decreased and the release gain-response time increased in response to a determination that a transient is occurring in the input audio signal.

Audio signal dynamic range compression

Provided are, among other things, systems, methods and techniques for processing an audio signal to add virtual bass. In one representative embodiment, an apparatus includes: an input line that inputs an original audio signal; an estimator, coupled to the input line, that estimates a fundamental frequency of a bass sound within the original audio signal; a bass extraction filter, coupled to the input line, that extracts a bass portion of the original audio signal that is at least 1 octave wide and includes the fundamental frequency; a frequency translator, coupled to the bass extraction filter, that shifts the bass portion, in its entirety, by a positive frequency increment that is an integer multiple of the fundamental frequency estimated by the estimator, thereby providing a virtual bass signal; and an adder having 1) inputs coupled to the original audio signal and to the virtual bass signal and 2) an output.

Addition of virtual bass

Provided are, among other things, systems, methods and techniques for reducing echo in an audio signal. One representative embodiment involves obtaining an input signal, an estimate of a system-characterizing function, and a reference signal, each at a corresponding sample rate and each divided into a plurality of sub-bands; separately processing such sub-bands, where for a given sub-band the estimate of the system-characterizing function and the reference signal are processed to generate an echo-estimation signal and then the echo-estimation signal is subtracted from the input signal to provide an echo-corrected signal for such given sub-band; and combining the echo-corrected signal from each of different ones of the plurality of the sub-bands to provide a final output signal, with the echo-estimation signal generated using a processing sample rate that is lower than the sample rate for the input signal.

Audio signal echo reduction

Provided is an audio-signal dynamic-range-compression system. An input line accepts an input signal. A filter set coupled to the input line divides the input signal into a set of frequency bands, including a first frequency band and a second frequency band. A compressor has a first input that receives the first frequency band, a second input and an output. An adder has a first input coupled to the output of the compressor, a second input coupled to the second frequency band, and an output. An overshoot estimation module has an input coupled to the output of the adder and an output coupled to the second input of the compressor. The output of the overshoot estimation module provides an overshoot signal that indicates a result of a comparison between a signal at the input of the overshoot estimation module and a specified threshold.

Multiband audio signal dynamic range compression with overshoot suppression

Provided are, among other things, systems, methods and techniques for loudness-based audio-signal compensation. According to one such system, an input line accepts an input audio signal; a loudness estimator, coupled to the input line, processes the input audio signal to obtain an audio playback loudness level and also provides a production loudness level for the input audio signal; and a filter generator/selector coupled to an output of the loudness estimator provides an audio compensation filter based on the production loudness level and the audio playback loudness level. The input signal is processed by the audio compensation filter to provide an output signal.

Loudness-based audio-signal compensation

Provided are, among other things, systems, methods and techniques for processing an audio signal to add virtual bass. In one representative embodiment, an apparatus includes: (a) an input line that inputs an original audio signal; (b) a bass extraction filter that extracts a bass portion of such original audio signal; (c) an estimator that estimates a fundamental frequency of a bass sound within such bass portion; (d) a frequency translator that shifts the bass portion by a positive frequency increment that is an integer multiple of the fundamental frequency estimated by the estimator, thereby providing a virtual bass signal; (f) an adder having (i) inputs coupled to the original audio signal and to the virtual bass signal and (ii) an output; and (g) an audio output device coupled to the output of the adder.

Provided are, among other things, systems, methods and techniques for audio-signal processing. One representative embodiment includes HT sub-band analysis/decomposition modules, e.g., one for each audio channel and one for an echo reference signal. Each HT sub-band analysis/decomposition module includes a Hilbert Transformation module and an analysis/decomposition filter bank and provides sub-band outputs. Echo-cancellation modules, e.g., one for each audio channel, perform echo-cancellation processing on such sub-bands. Beamforming modules, e.g., one for each sub-band, then perform beamforming, e.g., across all audio channels. Finally, a resynthesis stage combines the different sub-band outputs in order to provide a system output signal.

Multichannel sub-band audio-signal processing using beamforming and echo cancellation

Provided are, among other things, systems, methods and techniques for compressing an audio signal. According to one representative embodiment, an audio signal that includes quantization indexes, identification of segments of such quantization indexes, and indexes of entropy codebooks that have been assigned to such segments is obtained, with a single entropy codebook index having been assigned to each such segment. Potential merging operations in which adjacent ones of the segments potentially would be merged with each are identified, and bit penalties for the potential merging operations are estimated. At least one of the potential merging operations is performed based on the estimated bit penalties, thereby obtaining a smaller updated set of segments of quantization indexes and corresponding assigned codebooks. The quantization indexes in each of the segments in the smaller updated set are then entropy encoded by using the corresponding assigned entropy codebooks, thereby compressing the audio signal.

Codebook segment merging

Provided are, among other things, systems, methods and techniques for processing an audio signal to add virtual bass. In one representative embodiment, an apparatus includes: (a) an input line that inputs an original audio signal; (b) a transform module that transforms the original audio signal into a set of frequency components; (c) a bass extraction filter that extracts a bass portion of such frequency components; (d) an estimator that estimates a fundamental frequency of a bass sound within such bass portion; (e) a frequency translator that shifts the bass portion by a frequency that is an integer multiple of the fundamental frequency estimated by the estimator, thereby providing a virtual bass signal; (f) an adder having (i) inputs coupled to the original audio signal and to the virtual bass signal and (ii) an output; and (g) an audio output device coupled to the output of the adder.

Addition of virtual bass in the frequency domain

Provided are, among other things, systems, methods and techniques for processing an audio signal to add virtual bass. In one representative embodiment, an apparatus includes: (a) an input line that inputs an original audio signal in the time domain; (b) a bass extraction filter that extracts a bass portion of the original audio signal, which also is in the time domain; (c) an estimator that estimates a fundamental frequency of a bass sound within the bass portion; (d) a frequency translator that shifts the bass portion by a positive frequency increment that is an integer multiple of the fundamental frequency estimated by the estimator, thereby providing a virtual bass signal; (e) an adder having (i) inputs coupled to the original audio signal and to the virtual bass signal and (ii) an output; and (f) an audio output device coupled to the output of the adder.

Addition of virtual bass in the time domain

Provided are, among other things, systems, methods and techniques for detecting whether a transient exists within an audio signal. According to one representative embodiment, a segment of a digital audio signal is divided into blocks, and a norm value is calculated for each of a number of the blocks, resulting in a set of norm values for such blocks, each such norm value representing a measure of signal strength within a corresponding block. A maximum norm value is then identified across such blocks, and a test criterion is applied to the norm values. If the test criterion is not satisfied, a first signal indicating that the segment does not include any transient is output, and if the test criterion is satisfied, a second signal indicating that the segment includes a transient is output. According to this embodiment, the test criterion involves a comparison of the maximum norm value to a different second maximum norm value, subject to a specified constraint, within the segment.

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Laguna Beach, CA, United States of America

Yuli You