Patent classifications
H04S2400/03
Time-domain stereo encoding and decoding method and related product
An audio encoding and decoding method and a related apparatus are provided. The audio encoding method may include: determining a coding mode of a current frame; when determining that the coding mode of the current frame is an anticorrelated signal coding mode, performing time-domain downmix processing on left and right channel signals in the current frame by using a time-domain downmix processing manner corresponding to the anticorrelated signal coding mode, to obtain a primary channel signal and a secondary channel signal, where the time-domain downmix processing manner corresponding to the anticorrelated signal coding mode is a time-domain downmix processing manner corresponding to an anticorrelated signal channel combination scheme, and the anticorrelated signal channel combination scheme is a channel combination scheme corresponding to a near out of phase signal; and encoding the obtained primary channel signal and secondary channel signal in the current frame.
NOISE FILLING IN MULTICHANNEL AUDIO CODING
In multichannel audio coding, an improved coding efficiency is achieved by the following measure: the noise filling of zero-quantized scale factor bands is performed using noise filling sources other than artificially generated noise or spectral replica. In particular, the coding efficiency in multichannel audio coding may be rendered more efficient by performing the noise filling based on noise generated using spectral lines from a previous frame of, or a different channel of the current frame of, the multichannel audio signal.
BINAURAL MULTI-CHANNEL DECODER IN THE CONTEXT OF NON-ENERGY-CONSERVING UPMIX RULES
A multi-channel decoder for generating a binaural signal from a downmix signal using upmix rule information on an energy-error introducing upmix rule for calculating a gain factor based on the upmix rule information and characteristics of head related transfer function based filters corresponding to upmix channels. The one or more gain factors are used by a filter processor for filtering the downmix signal so that an energy corrected binaural signal having a left binaural channel and a right binaural channel is obtained.
APPARATUS AND METHOD FOR GENERATING A DIFFUSE REVERBERATION SIGNAL
An audio apparatus for generating a diffuse reverberation signal comprises a receiver (501) receiving audio signals representing sound sources and metadata comprising a diffuse reverberation signal to total source relationship indicative of a level of diffuse reverberation sound relative to total emitted sound in the environment. The metadata also for each audio signal comprises a signal level indication and a directivity data indicative of directivity of sound radiation from the sound source represented by the audio signal. A circuit (505, 507) determines a total emitted energy indication based on the signal level indication and the directivity data, and a downmix coefficient based on the total emitted energy and the diffuse reverberation signal to total signal relationship. A downmixer (509) generates a downmix signal by combining signal components for each audio signal generated by applying the downmix coefficient for each audio signal to the audio signal. A reverberator (407) generates the diffuse reverberation signal for the environment from thedownmix signal component.
Efficient coding of audio scenes comprising audio objects
There is provided encoding and decoding methods for encoding and decoding of object based audio. An exemplary encoding method includes inter alia calculating M downmix signals by forming combinations of N audio objects, wherein M≦N, and calculating parameters which allow reconstruction of a set of audio objects formed on basis of the N audio objects from the M downmix signals. The calculation of the M downmix signals is made according to a criterion which is independent of any loudspeaker configuration.
Method for generating filter for audio signal, and parameterization device for same
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.
Networked speaker system with LED-based wireless communication and object detection
A networked speaker system communicates using Li-Fi. The LEDs implementing the Li-Fi may also have modes in which they are used to map the walls of a room in which the speakers are located, detect the locations of speakers in the room, and detect and classify listeners in the room. Based on this, waveform analysis may be applied to input audio to establish equalization and delays that are optimal for the room geometry, speaker locations, and listener locations.
MATRIX DECODER WITH CONSTANT-POWER PAIRWISE PANNING
A constant-power pairwise panning upmixing system and method for upmixing from a two-channel stereo signal to a multi-channel surround sound (having more than two channels). Each output channel is some combination of the two input channels. Closed-form solutions are used to calculate dematrixing coefficients that are used to weight each input channel. The dematrixing coefficients are computed based on an inter-channel level difference and an inter-channel phase difference between the two input signals. The weighted input channels then are mixed uniquely for each output channel to generate a surround sound output from the stereo input signal. Each dematrixing coefficient has an in-phase component and an out-of-phase component. The phase coefficients for each component vary in time and are based on the phase difference between the input signals. The resultant surround sound output faithfully simulates the audio content as originally mixed.
AUDIO RENDERING USING 6-DOF TRACKING
The methods and apparatus described herein optimally represent full 3D audio mixes (e.g., azimuth, elevation, and depth) as “sound scenes” in which the decoding process facilitates head tracking. Sound scene rendering can be performed for the listener's orientation (e.g., yaw, pitch, roll) and 3D position (e.g., x, y, z), and can be modified for a change in the listener's orientation or 3D position. As described below, the ability to render an audio object in both the near-field and far-field enables the ability to fully render depth of not just objects, but any spatial audio mix decoded with active steering/panning, such as Ambisonics, matrix encoding, etc., thereby enabling full translational head tracking (e.g., user movement) beyond simple rotation in the horizontal plane, or 6-degrees-of-freedom (6-DOF) tracking and rendering.
Generating Binaural Audio in Response to Multi-Channel Audio Using at Least One Feedback Delay Network
In some embodiments, virtualization methods for generating a binaural signal in response to channels of a multi-channel audio signal, which apply a binaural room impulse response (BRIR) to each channel including by using at least one feedback delay network (FDN) to apply a common late reverberation to a downmix of the channels. In some embodiments, input signal channels are processed in a first processing path to apply to each channel a direct response and early reflection portion of a single-channel BRIR for the channel, and the downmix of the channels is processed in a second processing path including at least one FDN which applies the common late reverberation. Typically, the common late reverberation emulates collective macro attributes of late reverberation portions of at least some of the single-channel BRIRs. Other aspects are headphone virtualizers configured to perform any embodiment of the method.