Disclosed are systems and methods for automatic control of gain in audio and video conferencing applications to maintain a predetermined and stable audio level. In one embodiment, a first stage applies a first stage gain based on a long-term estimate of signal level, while a second stage gain, based on a short-term estimate of signal level assists the first stage gain to achieve a target level. Some embodiments of long-term level estimation utilize statistical analysis of a buffer to validate or arrive at a more accurate long-term signal level estimate.

An audio signal output device (1) includes an audio signal output portion (15), a volume setting information acquisition portion (15), and a control portion (10). The control portion is configured to calculate the output volume of an external device on the basis of volume information of content and volume setting information of the external device and perform control of lowering the output volume of the external device in the case where the output volume is higher than a prescribed volume threshold.

Techniques are disclosed for synchronizing gain adjustments across a cascaded network of audio gain stages having variant operating delays. In particular, a delay-synchronized volume adjustment system configured in accordance with an embodiment of the present disclosure includes a controller operatively coupled to the cascaded network of audio and configured to apply gain adjustments in a synchronized manner that accounts for operating delays that are inherent to each gain stage. In an embodiment, the controller synchronously adjusts each gain stage relative to a corresponding operating delay such that gain adjustments fully propagate at substantially a same point in time within a given acceptable tolerance, and thus, eliminates or otherwise mitigates perceivable volume shifts when mixing audio from two or more audio sources.

An audio reproduction apparatus is shown and includes an amplifier with a power amplification stage having transistors in a push-pull arrangement. A bias generator biases the transistors with a standing current. A processor receives a data stream comprising digital samples of an analog audio signal and analyzes the peak level of each group. It then determines the appropriate standing currents to maintain Class A operation of the power amplification stage given the peak levels of each of the groups. A digital to analog converter produces an analog input signal for the input stage of the amplifier from the data stream. A feedforward path between the processor and the bias generator allows the standing current to be adjusted prior to the arrival of the analog input signal in the power amplification stage.

A gain control system for controlling gain applied to an audio signal includes a power estimator configured to estimate the power of a digital signal derived from the audio signal, a digital gain estimator configured to determine, in dependence on the estimated power, a digital gain which would modify the power of the digital signal so as to reach a target power level, and a gain controller configured to adjust an analogue gain applied to the audio signal in dependence on the determined digital gain.

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.

Disclosed are systems and methods for processing an audio signal. In particular, there is provided a method for determining dynamic gain values to be applied on a digital input signal. The digital signal may be arranged in blocks. The dynamic gain values may be used for attenuating input signal values exceeding a clipping threshold. More particularly, the method comprising, for each signal block, passing backwards over the next signal block and the current signal block to produce a preliminary gain contour from the input signal; and passing forwards over the current signal block to produce a final gain contour for the current signal block based on the preliminary gain contour, wherein the gain contours are produced by applying an instant gain ascent and a smooth gain decay to the gain contours.

An audio mixer includes a user interface, panners, a first adder, a localization device, a second adder, and an output circuit. The user interface supplies a first parameter and a second parameter for each channel based on a user operation. The first parameter indicates a position in a right-left direction. The second parameter specifies internalization or externalization. The panners respectively correspond to channels and, based on the first parameter, pan a sound signal corresponding to the each channel to generate first stereo signals. The first adder generates a second stereo signal by mixing first stereo signals respectively corresponding to externalization channels. The localization device generates two third stereo signals. The second adder generates a fourth stereo signal by mixing the two third stereo signals and first stereo signals respectively corresponding to internalization channels. The output circuit outputs the fourth stereo signal.

Example techniques may provide volume enhancements to a media playback system. An example implementation involves a first playback device in a first playback configuration playing audio at a first volume level and storing the first volume level in association with the first playback configuration. The first playback device receives instructions to change to a second playback configuration. Based on receiving the instructions, the first playback device joins a synchrony group with a second zone and adjusts the first playback device to play audio at a second volume level which is based on a volume level of a second playback device in the second zone. While in the second playback configuration, the first playback device receives instructions to change to the first playback configuration and, based on the instructions, leaves the synchrony group and adjusts the first playback device to play audio content at the stored first volume level.

An audio loudness data collection module collects and stores audio data. The data can be stored for a first time period. A monitoring system can access and store the audio data collected and stored by the loudness data collection module according to a second time period. The stored audio data can be processed and displayed to a user in a number of ways. The data can also be used to provide loudness data control. For example, a loudness data control module can perform an audio AGC function to control audio loudness.