An audio system has an amplifier for driving an audio actuator and includes a switched-mode power supply that draws power from a power source (e.g., battery) to supply power to the amplifier, a capacitor charged by the switched-mode power supply to supply power to the amplifier, and a feed-forward compressor that performs dynamic range compression of an audio input to provide an audio output for amplification by the amplifier. The compressor includes a sidechain frequency-biasing filter that generates a frequency-biased version of the audio input that is attenuated as frequency increases which causes the compressor to decrease the compression as frequency increases. A control block limits current drawn from the battery by the switched-mode power supply independent of audio input frequency, but the frequency-biasing filter enables the amplifier to service audio power transients greater than the current-limited power supply can supply by advantageously concurrently sourcing extra power from the capacitor.

Volume leveler controller and controlling method are disclosed. In one embodiment, A volume leveler controller includes an audio content classifier for identifying the content type of an audio signal in real time; and an adjusting unit for adjusting a volume leveler in a continuous manner based on the content type as identified. The adjusting unit may configured to positively correlate the dynamic gain of the volume leveler with informative content types of the audio signal, and negatively correlate the dynamic gain of the volume leveler with interfering content types of the audio signal.

In some embodiments, a method for processing an audio signal in an audio processing apparatus is disclosed. The method includes receiving an audio signal and a parameter, the parameter indicating a location of an auditory event boundary. An audio portion between consecutive auditory event boundaries constitutes an auditory event. The method further includes applying a modification to the audio signal based in part on an occurrence of the auditory event. The parameter may be generated by monitoring a characteristic of the audio signal and identifying a change in the characteristic.

Described herein is a method and apparatus for enhancing the dynamic range of a digital-to-analog conversion circuit. Dynamic range enhancement (DRE) is accomplished by modifying the gain of components of the circuit so that the gain of components generating noise is effectively reduced. In a circuit utilizing a plurality of 1-bit DACs, analog signal gain is decreased when the full nominal gain of the analog portion of the circuit is not needed to obtain a desired peak output amplitude. The reduction is accomplished by effectively “disconnecting” some of the plurality of 1-bit DACs. Some or all of the 1-bit DACs are configured to have a third or “tri-state” in which there is no connection to the normal two reference levels thus providing no output. If some portion of the 1-bit DACs is placed in the tri-state, both the signal and noise gain will be reduced.

A computer-implemented method of audio processing, the method comprising: receiving audio object data and audio description data, wherein the audio object data includes a first plurality of audio objects; calculating a long-term loudness of the audio object data and a long- term loudness of the audio description data; calculating a plurality of short-term loudnesses of the audio object data and a plurality of short-term loudnesses of the audio description data; reading a first plurality of mixing parameters that correspond to the audio object data; generating a second plurality of mixing parameters based on the first plurality of mixing parameters, the long-term loudness of the audio object data, the long-term loudness of the audio description data, the plurality of short-term loudnesses of the audio object data, and the plurality of short-term loudnesses of the audio description data; generating a gain adjustment visualization corresponding to the second plurality of mixing parameters, the audio object data and the audio description data; and generating mixed audio object data by mixing the audio object data and the audio description data according to the second plurality of mixing parameters, wherein the mixed audio object data includes a second plurality of audio objects, wherein the second plurality of audio objects correspond to the first plurality of audio objects mixed with the audio description data according to the second plurality of mixing parameters.

An audio normalization gain value is applied to an audio signal to produce a normalized signal. The normalized signal is processed to compute dynamic range control (DRC) gain values in accordance with a selected one of several pre-defined DRC characteristics. The audio signal is encoded, and the DRC gain values are provided as metadata associated with the encoded audio signal. Several other embodiments are also described and claimed.

The present technology relates to a decoding apparatus, a decoding method and a program which make it possible to obtain sound with higher quality. A demultiplexing circuit demultiplexes an input code string into a gain code string and a signal code string. A signal decoding circuit decodes the signal code string to output a time series signal. A gain decoding circuit decodes the gain code string. That is, the gain decoding circuit reads out gain values and gain inclination values at predetermined gain sample positions of the time series signal and interpolation mode information. An interpolation processing unit obtains a gain value at each sample position between two gain sample positions through linear interpolation or non-linear interpolation according to the interpolation mode based on the gain values and the gain inclination values. A gain applying circuit adjusts a gain of the time series signal based on the gain values. The present technology can be applied to a decoding apparatus.

Methods, systems, and apparatuses are described herein for improved processing audio in a video stream. A system may split audio in a frame of video content into multiple bands based on their audio levels. The system may then dynamically compress and dynamically normalize the audio level in each band. When dynamically compressing the bands, the system may determine, based on stored information, what audio level range is acceptable for an end user and may smooth and maintain the ranges of the audio to be within the acceptable range. The system may include the dynamically normalized and dynamically compressed frames as a second audio track in the video content. A computing device receiving the video content may select the second audio track during playback. If an end user selects the second audio track, the video is delivered with the modified sound of the second audio track.

Example embodiments provide a process that includes one or more of receiving an audio signal at a feedback compressor circuit, multiplying the received audio signal with a power feedback signal to create a product audio signal, wherein the feedback signal comprises a low-pass filtered signal, applying a power amplifier to the product audio signal, and providing the amplified product audio signal as an output signal to a speaker.

Apparatus, systems, articles of manufacture, and methods for volume adjustment are disclosed herein. An example method includes collecting data corresponding to a volume of an audio signal as the audio signal is output through a device, when an average volume of the audio signal does not satisfy a volume threshold for a specified timespan, determining a difference between the average volume and a desired volume, and applying a gain to the audio signal to adjust the volume of the audio signal to the desired volume, the gain determined based on the difference between the average volume and the desired volume.