An audio processing device includes a signal processing circuit, a power switch circuit, a power amplifier and a detection circuit. The signal processing circuit is configured to process an audio input to generate an audio signal. The power switch circuit is configured to generate an operating voltage. The power amplifier is coupled with the signal processing circuit and the power switch circuit, and configured to drive a load circuit by the operating voltage according to the audio signal. The detection circuit is coupled with the signal processing circuit and the power switch circuit, configured to obtain a volume value from the audio signal and compare the volume value with a first volume threshold to generate a comparison result, and configured to control, according to the comparison result, the power switch circuit to set the operating voltage to have a corresponding one of multiple voltage levels.

An example first playback device includes programming to perform functions including: (1) storing an active volume state variable in memory, wherein the active volume state variable corresponds to a current playback volume; (2) storing a volume limit state variable in memory, wherein the volume limit state variable corresponds to a playback volume limit of the first playback device; (3) detecting a command to begin playback of media at a proposed playback volume different from the current playback volume; (4) based on comparing (i) the playback volume limit and (ii) the proposed playback volume, selecting a startup playback volume; (5) playing back media at the startup playback volume; and (6) causing at least a second playback device of the media playback system to play back media at the startup playback volume.

Systems and methods for limiting volume in an audio playback device using a feedback controller are disclosed herein. In one example, a gain stage modulates gain of an audio signal based in part on feedback from a downstream limiter. The gain stage receives a first audio signal as well as a feedback signal from the feedback controller. Based at least in part on the feedback signal from the feedback controller, the gain stage modulates a gain of the first audio signal to provide a second audio signal. The second audio signal is delivered to the limiter, which limits the second audio signal to produce an output signal. The output signal is played back via a transducer. The feedback controller receives a gain reduction value from the limiter and determines a feedback signal to provide to the gain stage upstream of the limiter.

Provided is a method for audio peak reduction using an all-pass filter, including: determining a delay parameter m and a gain parameter g based on a formula (1):

[00001]$\begin{array}{cc}\underset{m,g}{\arg \min }\underset{n}{\max }.Math.y_{m,g}\left(n\right).Math.,& \left(1\right)\end{array}$

absolute peak map

[00002]$Y\left(m,g\right)=\underset{n}{\max }.Math.y_{m,g}\left(n\right).Math.,$

y.sub.m,g(n) represents a processed signal with a time-domain response function and is calculated based on a formula (2): y.sub.m,g(n)=(h.sub.s*x)(n)(2), where h.sub.s represents an impulse response function, x (n) represents an input signal, and h.sub.s is calculated based on formula (3):

[00003]$\begin{array}{cc}{H}_S\left(z\right)=\frac{g+z^{-m}}{1+{\mathrm{gz}}^{-m}}.& \left(3\right)\end{array}$

This method is widely used in the reproduction, storage and broadcasting of sound, and the computational complexity is small, which i

Many portable playback devices cannot decode and playback encoded audio content having wide bandwidth and wide dynamic range with consistent loudness and intelligibility unless the encoded audio content has been prepared specially for these devices. This problem can be overcome by including with the encoded content some metadata that specifies a suitable dynamic range compression profile by either absolute values or differential values relative to another known compression profile. A playback device may also adaptively apply gain and limiting to the playback audio. Implementations in encoders, in transcoders and in decoders are disclosed.

This application relates to Class D amplifier circuits. A modulator controls a Class D output stage based on a modulator input signal (Dm) to generate an output signal (Vout) which is representative of an input signal (Din). An error block, which may comprise an ADC, generates an error signal (ε) from the output signal and the input signal. In various embodiments the extent to which the error signal (ε) contributes to the modulator input signal (Dm) is variable based on an indication of the amplitude of the input signal (Din). The error signal may be received at a first input of a signal selector block. The input signal may be received at a second input of the signal selector block. The signal selector block may be operable in first and second modes of operation, wherein in the first mode the modulator input signal is based at least in part on the error signal; and in the second mode the modulator input signal is based on the digital input signal and is independent of the error signal. The error signal can be used to reduce distortion at high signal levels but is not used at low signal levels and so the noise floor at low signal levels does not depend on the component of the error block.

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.

Systems and methods for limiting volume in an audio playback device using a feedback controller are disclosed herein. In one example, a gain stage modulates gain of an audio signal based in part on feedback from a downstream limiter. The gain stage receives a first audio signal as well as a feedback signal from the feedback controller. Based at least in part on the feedback signal from the feedback controller, the gain stage modulates a gain of the first audio signal to provide a second audio signal. The second audio signal is delivered to the limiter, which limits the second audio signal to produce an output signal. The output signal is played back via a transducer. The feedback controller receives a gain reduction value from the limiter and determines a feedback signal to provide to the gain stage upstream of the limiter.

The present invention provides methods and systems for digital processing of an input audio signal. Specifically, the present invention includes a high pass filter configured to filter the input audio signal to create a high pass signal. A first filter module then filters the high pass signal to create a first filtered signal. A first compressor modulates the first filtered signal to create a modulated signal. A second filter module then filters the modulated signal to create a second filtered signal. The second filtered signal is processed by a first processing module. A band splitter splits the processed signal into low band, mid band, and high band signals. The low band and high band signals are modulated by respective compressors. A second processing module further processes the modulated low band, mid band, and modulated high band signals to create an output signal.

An audio encoder device includes an audio encoder configured for producing an encoded audio bitstream from an audio signal having consecutive audio frames; a dynamic range control encoder configured for producing an encoded dynamic range control bitstream from an dynamic range control sequence corresponding to the audio signal and having consecutive dynamic range control frames, wherein each dynamic range control frame of the dynamic range control frames has one or more nodes, wherein each node of the one or more nodes has gain information for the audio signal and time information indicating to which point in time the gain information corresponds.