A system and method for transmitting at least one multichannel audio matrix across a global network is shown and described. The method for transmitting at least one multichannel audio matrix across a global network begins by capturing audio from a production source. The captured audio is then converted from an analog format to a digital format. The digital audio may then be encoded using an audio codec. The audio is sent to a network and received at a second location. The second location uses a specialized computing device to ensure the audio is properly received. If the audio was encoded, it is now decoded. Once decoded if needed the audio is converted back to analog format. This will allow for the audio to be mixed on a mixing device.

A method including: obtaining at least two audio signals for reproduction, each of the at least two audio signals associated with a respective one of at least two reproduction locations within an audio reproduction space; obtaining within the audio reproduction space at least two zones; obtaining at least one location for a user's position within the audio reproduction space, the at least one location being relative to at least one of the at least two zones and the at least two reproduction locations; and processing the at least two audio signals based on the obtained at least one location for the user's position within the audio reproduction space to generate at least one output audio signal, the at least one output audio signal is reproduced from at least one of the at least two reproduction locations.

In some embodiments, a method comprises receiving audio content comprising left input channel signals and right input channel signals, and generating first and second input signals from the left and right input channel signals. The first input signal is based on a sum of the left and right input channel signals, and the second input signal is based on a difference of the left and right input channel signals. An array transfer function is applied to the first and second input signals to produced audio output signals, which can be provided to a plurality of audio transducers on one or more playback devices.

A device includes a memory configured to store untransformed ambisonic coefficients at different time segments. The device also includes one or more processors configured to obtain the untransformed ambisonic coefficients at the different time segments, where the untransformed ambisonic coefficients at the different time segments represent a soundfield at the different time segments. The one or more processors are also configured to apply one adaptive network, based on a constraint, to the untransformed ambisonic coefficients at the different time segments to generate transformed ambisonic coefficients at the different time segments, wherein the transformed ambisonic coefficients at the different time segments represent a modified soundfield at the different time segments, that was modified based on the constraint.

A method and a device encode N audio signals, from N microphones where N≥3. For each pair of the N audio signals an angle of incidence of direct sound is estimated. A-format direct sound signals are derived from the estimated angles of incidence by deriving from each estimated angle an A-format direct sound signal. Each A-format direct sound signal is a first-order virtual microphone signal, for example, a cardioids signal.

A method and a device encode N audio signals, from N microphones where N≥3. For each pair of the N audio signals an angle of incidence of direct sound is estimated. A-format direct sound signals are derived from the estimated angles of incidence by deriving from each estimated angle an A-format direct sound signal. Each A-format direct sound signal is a first-order virtual microphone signal, for example, a cardioids signal.

An apparatus including at least one processor and at least one memory including a computer program code, the at least one memory and the computer program code configured to, with the at least one processor, cause the apparatus at least to: determine, for two or more microphone audio signals, at least one spatial audio parameter for providing spatial audio reproduction; determine at least one coherence parameter associated with a sound field based on the two or more microphone audio signals, such that another sound field is configured to be reproduced based on the at least one spatial audio parameter and the at least one coherence parameter.

An apparatus including at least one processor and at least one memory including a computer program code, the at least one memory and the computer program code configured to, with the at least one processor, cause the apparatus at least to: determine, for two or more microphone audio signals, at least one spatial audio parameter for providing spatial audio reproduction; determine at least one coherence parameter associated with a sound field based on the two or more microphone audio signals, such that another sound field is configured to be reproduced based on the at least one spatial audio parameter and the at least one coherence parameter.

The present invention relates to a method and an apparatus for processing a signal, which are used for effectively reproducing a multimedia signal, and more particularly, to a method and an apparatus for processing a signal, which are used for implementing filtering for multimedia signal having a plurality of subbands with a low calculation amount. To this end, provided are a method for processing a multimedia signal including: receiving a multimedia signal having a plurality of subbands; receiving at least one proto-type filter coefficients for filtering each subband signal of the multimedia signal; converting the proto-type filter coefficients into a plurality of subband filter coefficients; truncating each subband filter coefficients based on filter order information obtained by at least partially using characteristic information extracted from the corresponding subband filter coefficients, the length of at least one truncated subband filter coefficients being different from the length of truncated subband filter coefficients of another subband; and filtering the multimedia signal by using the truncated subband filter coefficients corresponding to each subband signal and an apparatus for processing a multimedia signal using the same.

The present invention relates to a method and an apparatus for processing a signal, which are used for effectively reproducing a multimedia signal, and more particularly, to a method and an apparatus for processing a signal, which are used for implementing filtering for multimedia signal having a plurality of subbands with a low calculation amount. To this end, provided are a method for processing a multimedia signal including: receiving a multimedia signal having a plurality of subbands; receiving at least one proto-type filter coefficients for filtering each subband signal of the multimedia signal; converting the proto-type filter coefficients into a plurality of subband filter coefficients; truncating each subband filter coefficients based on filter order information obtained by at least partially using characteristic information extracted from the corresponding subband filter coefficients, the length of at least one truncated subband filter coefficients being different from the length of truncated subband filter coefficients of another subband; and filtering the multimedia signal by using the truncated subband filter coefficients corresponding to each subband signal and an apparatus for processing a multimedia signal using the same.