Disclosed is an audio signal processing method. The audio signal processing method according to the present invention comprises the steps of: receiving a bit-stream including at least one of a channel signal and an object signal; receiving a user's environment information; decoding at least one of the channel signal and the object signal on the basis of the received bit-stream; generating the user's reproducing channel information on the basis of the user's received environment information; and generating a reproducing signal through a flexible renderer on the basis of at least one of the channel signal and the object signal and the user's reproducing channel information.

An audio system is proposed, dynamically playing optimized audio signals based on user position. A sensor circuits dynamically senses a target space to generate field context information. First speaker and second speaker are arranged for audio playback. A host device recognizes a user from the field context information, determines the user position corresponding to the target space, and adaptively assigns the user position as a target listening spot. A sensor circuit contains a camera capturing an ambient image out of the target space. A control circuit utilizes a user interface circuit to perform a configuration procedure which determines location, size and acoustic attribute information of an ambient object, allowing the control circuit to accordingly perform an object-based compensation operation on the target listening spot to generate optimized first channel audio signal and second channel audio signal.

A method is disclosed for audio object extraction from an audio content which includes identifying a first set of projection spaces including a first subset for a first channel and a second subset for a second channel of the plurality of channels. The method may further include determining a first set of correlations between the first and second channels, each of the first set of correlations corresponding to one of the first subset of projection spaces and one of the second subset of projection spaces. Still further, the method may include extracting an audio object from an audio signal of the first channel at least in part based on a first correlation among the first set of correlations and the projection space from the first subset corresponding to the first correlation, the first correlation being greater than a first predefined threshold. Corresponding system and computer program products are also disclosed.

Some audio processing methods may involve receiving audio data corresponding to a plurality of audio channels and determining audio characteristics of the audio data, which may include transient information. An amount of decorrelation for the audio data may be based, at least in part, on the audio characteristics. If a definite transient event is determined, a decorrelation process may be temporarily halted or slowed. Determining transient information may involve evaluating the likelihood and/or the severity of a transient event. In some implementations, determining transient information may involve evaluating a temporal power variation in the audio data. Explicit transient information may or may not be received with the audio data, depending on the implementation. Explicit transient information may include a transient control value corresponding to a definite transient event, a definite non-transient event or an intermediate transient control value.

The present invention relates to a method for generating a filter for an audio signal and a parameterization device for the same, and more particularly, to a method for generating a filter for an audio signal, to implement filtering of an input audio signal with a low computational complexity, and a parameterization device therefor. To this end, provided are a method for generating a filter for an audio signal, including: receiving at least one binaural room impulse response (BRIR) filter coefficients for binaural filtering of an input audio signal; converting the BRIR filter coefficients into a plurality of subband filter coefficients; obtaining average reverberation time information of a corresponding subband by using reverberation time information extracted from the subband filter coefficients; obtaining at least one coefficient for curve fitting of the obtained average reverberation time information; obtaining flag information indicating whether the length of the BRIR filter coefficients in a time domain is more than a predetermined value; obtaining filter order information for determining a truncation length of the subband filter coefficients, the filter order information being obtained by using the average reverberation time information or the at least one coefficient according to the obtained flag information and the filter order information of at least one subband being different from filter order information of another subband; and truncating the subband filter coefficient by using the obtained filter order information and a parameterization device therefor.

A control section (1009) receives signaling (S) indicating one of at least two coding formats (F.sub.1, F.sub.2, F.sub.3) of an M-channel audio signal (L, LS, LB, TFL, TBL), the coding formats corresponding to different partitions of the channels of the audio signal into respective first and second groups (601, 602), wherein, in the indicated coding format, first and second channels (L.sub.1, L.sub.2) of a downmix signal correspond to linear combinations of the first and second groups, respectively; and a decoding section (900) reconstructs the audio signal based on the downmix signal and associated upmix parameters (α.sub.L). In the decoding section: a decorrelation input signal (D.sub.1, D.sub.2, D.sub.3) is determined based on the downmix signal and the indicated coding format; and wet and dry upmix coefficients, controlling linear mappings of the downmix signal and a decorrelated signal, generated based on the decorrelation input signal, are determined based on the upmix parameters and the indicated coding format.

Embodiments are described for a system of rendering object-based audio content through a system that includes individually addressable drivers, including at least one driver that is configured to project sound waves toward one or more surfaces within a listening environment for reflection to a listening area within the listening environment; a renderer configured to receive and process audio streams and one or more metadata sets associated with each of the audio streams and specifying a playback location of a respective audio stream; and a playback system coupled to the renderer and configured to render the audio streams to a plurality of audio feeds corresponding to the array of audio drivers in accordance with the one or more metadata sets.

A method for processing an audio signal in accordance with a room impulse response is described. The audio signal is separately processed with an early part and a late reverberation of the room impulse response, and the processed early part of the audio signal and the reverberated signal are combined. A transition from the early part to the late reverberation in the room impulse response is reached when a correlation measure reaches a threshold, the threshold being set dependent on the correlation measure for a selected one of the early reflections in the early part of the room impulse response.

A method of processing an audio signal includes receiving an audio bitstream encoded via MPEG Surround 212 (MPS212); generating an internal channel (IC) signal for a single channel pair element (CPE), based on the received audio bitstream, equalization (EQ) values for MPS212 output channels defined in a format converter, and gain values for the MPS212 output channels; and generating stereo output channels, based on the generated IC signal.

A method for rendering an audio output based on an audio data stream including M audio signals, side information including a series of reconstruction instances of a reconstruction matrix C and first timing data, the side information allowing reconstruction of N audio objects from the M audio signals, and object metadata defining spatial relationships between the N audio objects. The method includes generating a synchronized rendering matrix based on the object metadata, the first timing data, and information relating to a current playback system configuration, the synchronized rendering matrix having a rendering instance for each reconstruction instance, multiplying each reconstruction instance with a corresponding rendering instance to form a corresponding instance of an integrated rendering matrix, and applying the integrated rendering matrix to the audio signals in order to render an audio output.