The invention provides an efficient implementation of cross-product enhanced high-frequency reconstruction (HFR), wherein a new component at frequency QΩ+rΩ.sub.0 is generated on the basis of existing components at Ω and Ω+Ω.sub.0. The invention provides a block-based harmonic transposition, wherein a time block of complex subband samples is processed with a common phase modification. Superposition of several modified samples has the net effect of limiting undesirable intermodulation products, thereby enabling a coarser frequency resolution and/or lower degree of oversampling to be used. In one embodiment, the invention further includes a window function suitable for use with block-based cross-product enhanced HFR. A hardware embodiment of the invention may include an analysis filter bank, a subband processing unit configurable by control data and a synthesis filter bank.

A method includes generating a first current through a first node based on a differential pair of signals received by a differential pair of input nodes of a differential circuit of a first integrated circuit die of an isolator product. The method includes generating a second current through a second node. The second current matches the first current through the first node and is based on an attenuated version of an output measurement signal. The method includes generating the output measurement signal having a level corresponding to an average amplitude of the differential pair of signals based on the first current and the second current.

In one aspect, an example method includes (i) presenting first media content from a first source; (ii) encountering a trigger to switch from presenting the first media content from the first source to presenting second media content from a second source; (iii) determining a first loudness level of the first media content; (iv) determining a second loudness level of the second media content; (v) based on a difference between the first loudness level and the second loudness level, adjusting a loudness level of the second media content so as to generate modified media content having a third loudness level that is different from the second loudness level; and (vi) responsive to encountering the trigger, presenting the modified media content having the third loudness level.

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.

In one aspect, an example method includes (i) presenting first media content from a first source; (ii) encountering a trigger to switch from presenting the first media content from the first source to presenting second media content from a second source; (iii) determining a first loudness level of the first media content; (iv) determining a second loudness level of the second media content; (v) based on a difference between the first loudness level and the second loudness level, adjusting a loudness level of the second media content so as to generate modified media content having a third loudness level that is different from the second loudness level; and (vi) responsive to encountering the trigger, presenting the modified media content having the third loudness level.

A system comprising audio processing circuitry is provided. The audio processing circuitry is operable to receive audio signals. The audio processing circuitry is operable to process the audio signals to detect strength of a chat component of the audio signals and strength of a game component of the audio signals. The audio processing circuitry is operable to automatically control a volume setting based on one or both of: the detected strength of the chat component, and the detected strength of the game component. The combined-game-and-chat audio signals may comprise a left channel signal and a right channel signal. The processing of the combined-game-and-chat audio signals may comprise measuring strength of a vocal-band signal component that is common to the left channel signal and the right channel signal.

Audio data in a first format may be processed to produce audio data in a second format, which may be a reduced or simplified version of the first format. A loudness correction process may produce loudness-corrected audio data in the second format. A first power of the audio data in the second format and a second power of the loudness-corrected audio data in the second format may be determined. A second-format loudness correction factor for the audio data in the second format may be based, at least in part, on a power ratio between the first power and the second power. A first-format loudness correction factor for the audio data in the first format may be based, at least in part, on the power ratio and a power relationship between the audio data in the first format and the audio data in the second format.

A system and method provide audio processing for on-line communications, including the elimination of unwanted and disruptive noises, enhancing the clarity of the participants voices, and further processing to establish an immersive 3D spatial audio experience. The combination of the three main processing components which make up the Core and the processes of how audio streams and related data are manipulated leveraging machine learning algorithms and finely tuned component configurations to establish a clear, immersive on-line audio communication listening experience for each participant is a primarily unique feature of the present invention.

Methods, systems, and apparatuses are described for optimizing user content consuming experience by recognizing and classifying different sounds while a user views a program. The system may have or may access information related to the program audio being presented, enabling it to distinguish between conversations occurring in the program audio and conversations between users in the viewing environment. The system may turn the program volume down on one or more sound producing devices if it detects a conversation. The system may turn the program volume up if it detects an interrupting noise. The system may also adjust the program content based on locations of various objects within the listening or viewing environment, and types of users in the environment.

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.