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.

The present application relates in one aspect to a method of controlling diaphragm excursion of an electrodynamic loudspeaker. The method comprises dividing the audio input signal into at least a low-frequency band signal and a high-frequency band signal by a band-splitting network and applying the low-frequency band signal to a diaphragm excursion estimator. The instantaneous diaphragm excursion is determined based on the low-frequency band signal. The determined instantaneous diaphragm excursion is compared with an excursion limit criterion. The low-frequency band signal is limited based on a result of the comparison between the instantaneous diaphragm excursion and the excursion limit criterion to produce a limited low-frequency band signal which is combined with the high-frequency band signal to produce an excursion limited audio signal.

To control loudness during a junction between different types of broadcast content, such as a junction between programme and commercial or promotional content, representative loudness values for content respectively before (P) and after (C) the junction are received from a playout automation system. A time-varying gain control is applied before and after the junction in order to smooth loudness around the junction. The audio gain is smoothly increased prior to the junction to a gain (P+C)/2P times higher than the original gain value. Then, the gain is reduced shortly before the junction to a value (P+C)/2C times lower than the original gain value. After the junction, the gain is returned smoothly to the original value.

A wireless communication device is disclosed. The wireless communication device includes a processor, a memory, a transceiver configured to receive an audio signal, a codec to decode the audio signal, a dynamic range controller and a phoneme processor. The phoneme processor is configured to extract acoustic cues from each frame of the decoded audio signal and to identify a phoneme class in the each frame. The dynamic range controller is configured to apply dynamic range compression on the each frame based on the identified phoneme class.

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.

A method for decoding an encoded audio signal is described. The encoded audio signal comprises a sequence of frames, and is indicative of a plurality of different dynamic range control (DRC) profiles for a corresponding plurality of different rendering modes. The method comprises determining a first rendering mode from the plurality of different rendering modes; determining one or more DRC profiles from a subset of DRC profiles comprised within a 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 as a current DRC profile, if none of the one or more DRC profiles is applicable to the first rendering mode; wherein definition data of the default DRC profile is known at a decoder; and decoding the current frame using the current DRC profile.

A method or apparatus for delivering audio programming such as music to listeners may include identifying, capturing and applying a listener's audio perception characteristics (sometimes referred to as the listener's Personal Waveform Profile) to transform audio content so that the listener perceives the content similarly to how the content would be perceived by a different listener. An audio testing tool may be implemented as software application to identify and capture respective listeners' Personal Waveform Profiles. A signal processor may operate an algorithm processing source audio content using the respective listeners' Personal Waveform Profiles to provide audio output based on a difference between different profiles.

Systems are disclosed including a television (TV) set having included audio system. The TV set permits control over various functions, at least including audio volume, via a remote control. When the viewer activates the remote volume control, a graphic appears indicating the state of the volume control and, optionally the mute status. The graphic can be presented on the TV screen. In alternate embodiments, the graphic may be presented on a display of the remote, or even some other location, e.g., a different remote control. If a mute button is provided on the remote control, when the viewer activates the mute button, a graphic appears indicating the state of muting and, optionally, the volume control status. The TV set also offers control over various aspects of the audio system, including settings which go beyond volume up/down, generally through some sort of menu system.

The present invention provides methods and systems for narrow bandwidth digital processing of an input audio signal. Particularly, the present invention includes a high pass filter configured to filter the input audio signal. A first compressor then modulates the filtered signal in order to create a partially processed signal. In some embodiments, a clipping module further limits the gain of the partially processed signal. A splitter is configured to split the partially processed signal into a first signal and a second signal. A low pass filter is configured to filter the first signal. A pass through module is configured to adjust the gain of the second signal. A mixer then combines the filtered first signal and the gain-adjusted second signal in order to output a combined signal. In some embodiments, a tone control module further processes the combined signal, and a second compressor further modulates the processed signal.

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.