On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters (?) from the bitstream, where 1?m<n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC2) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.

A calibration method and a computer readable recording medium are provided. The calibration method is applied to an electronic device of an electronic system. The electronic system includes the electronic device and a broadcasting device. The calibration method comprises: obtaining device information of the broadcasting device to determine a type of the broadcasting device; providing a user interaction interface, and obtaining an interactive operation via the user interaction interface; generating a calibration command corresponding to the broadcasting device according to the interactive operation; and calibrating a broadcasting setting of the broadcasting device according to the calibration command.

On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters () from the bitstream, where 1m<n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC2) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.

An input ambisonic audio signal includes multiple channels, each of which is made up of audio data representing sound captured by an ambisonic microphone. A remote audio signal made up of audio data representing sound captured by remote meeting equipment is output by a local loudspeaker. Acoustic echo cancellation is performed on the input ambisonic audio signal by removing the remote audio signal from the input ambisonic audio signal. The acoustic echo cancellation may be performed on ambisonic A-format or B-format encoded audio data, or on an output encoding generated from the B-format encoded audio data. Comfort noise may be generated based on spectral and spatial characteristics of noise in the input audio data, for insertion into the input signal during acoustic echo cancellation. Automatic gain control may be performed across the multiple channels of the input audio signal.

A method for processing a digital audio broadcast signal includes: separating an analog audio portion and a digital audio portion of the digital audio broadcast signal; determining the loudness of the analog audio portion and the digital audio portion over a first short time interval; using the loudness of the analog and digital audio portions to calculate a short term average gain; determining a long term average gain; converting one of the long term average gain or the short term average gain to dB; if an output has been blended to digital, adjusting a digital gain parameter by a preselected increment to produce a digital gain parameter; if an output has not been blended to digital, setting the digital gain parameter to the short term average gain; providing the digital gain parameter to an audio processor; and repeating the above steps using a second short time interval.

A method for processing a digital audio broadcast signal includes: separating an analog audio portion and a digital audio portion of the digital audio broadcast signal; determining the loudness of the analog audio portion and the digital audio portion over a first short time interval; using the loudness of the analog and digital audio portions to calculate a short term average gain; determining a long term average gain; converting one of the long term average gain or the short term average gain to dB; if an output has been blended to digital, adjusting a digital gain parameter by a preselected increment to produce a digital gain parameter; if an output has not been blended to digital, setting the digital gain parameter to the short term average gain; providing the digital gain parameter to an audio processor; and repeating the above steps using a second short time interval.

A method for processing a digital audio broadcast signal includes: separating an analog audio portion and a digital audio portion of the digital audio broadcast signal; determining the loudness of the analog audio portion and the digital audio portion over a first short time interval; using the loudness of the analog and digital audio portions to calculate a short term average gain; determining a long term average gain; converting one of the long term average gain or the short term average gain to dB; if an output has been blended to digital, adjusting a digital gain parameter by a preselected increment to produce a digital gain parameter; if an output has not been blended to digital, setting the digital gain parameter to the short term average gain; providing the digital gain parameter to an audio processor; and repeating the above steps using a second short time interval.

A process and system for generating three dimensional sound conferencing includes generating a virtual map with a plurality of positions, each participant selecting one of the positions, determining a direction from each position to each other position on the map, determining a distance from each position to each other position on the map, receiving sound from each participant, mixing the received sound, transforming the mixed sound into binaural audio, and directing the binaural audio sound to each participant via a speaker associated with the virtual position of the speaking participant. The result is a clarified sound that gives to the listening participant a sense of where the speaking participant is positioned relative to the listening participant.

A method including: receiving at least a first input audio channel and a second input audio channel; and using an inter-channel prediction model to form at least an inter-channel direction of reception parameter.

A method including: receiving at least a first input audio channel and a second input audio channel; and using an inter-channel prediction model to form at least an inter-channel direction of reception parameter.