摘要:
Aspects of the invention provide methods, computer-readable media, and apparatuses for digital processing of acoustic signals to create a reproduction of a natural or an artificial spatial sound environment. An aspect of the invention supports spatial audio processing such as extracting a center channel in up-mixing stereo sound for multi-channel loudspeaker setup or headphone virtualization. An aspect of the invention also supports directional listening in which sound sources in a desired direction may be amplified or attenuated. Direction and diffuseness parameters for regions of input channels are determined and an extracted channel is extracted from the input channels according to the direction and diffuseness parameters. A gain estimate is estimated for each signal component being fed into the extracted channel and an extracted channel may be synthesized from a base signal and the gain estimate. The input channels may be partitioned into a plurality of time-frequency regions.
摘要:
Aspects of the invention provide methods, computer-readable media, and apparatuses for digital processing of acoustic signals to create a reproduction of a natural or an artificial spatial sound environment. An aspect of the invention supports spatial audio processing such as extracting a center channel in up-mixing stereo sound for multi-channel loudspeaker setup or headphone virtualization. An aspect of the invention also supports directional listening in which sound sources in a desired direction may be amplified or attenuated. Direction and diffuseness parameters for regions of input channels are determined and an extracted channel is extracted from the input channels according to the direction and diffuseness parameters. A gain estimate is estimated for each signal component being fed into the extracted channel and an extracted channel may be synthesized from a base signal and the gain estimate. The input channels may be partitioned into a plurality of time-frequency regions.
摘要:
This invention describes a method for adjusting the loudness and the spectral content of digital audio signals in a real-time using warped spectral filtering. A warped processing module modifies a spectral content of a digital audio signal with a set of gains for a plurality of non-linearly-scaled frequency bands determined by a warping factor λ of a warped delay line. Warped delay line signals, generated by the warped delay line, are processed by a warped filter block containing multiple warped finite impulse response filters, e.g., Mth band filters, using individual warped spectral filtering in said plurality of the non-linearly-scaled frequency bands, which is followed by a conventional processing by a dynamic range control/equalization block. The present invention describes another innovation, that is embedding the warped processing module in a two-channel quadrature mirror filter (QMF) bank for improving processing efficiency at high sample rates.
摘要:
This invention describes a method for adjusting the loudness and the spectral content of digital audio signals in a real-time using warped spectral filtering. A warped processing module modifies a spectral content of a digital audio signal with a set of gains for a plurality of non-linearly-scaled frequency bands determined by a warping factor λ of a warped delay line. Warped delay line signals, generated by the warped delay line, are processed by a warped filter block containing multiple warped finite impulse response filters, e.g., Mth band filters, using individual warped spectral filtering in said plurality of the non-linearly-scaled frequency bands, which is followed by a conventional processing by a dynamic range control/equalization block. The present invention describes another innovation, that is embedding the warped processing module in a two-channel quadrature mirror filter (QMF) bank for improving processing efficiency at high sample rates.
摘要:
This invention describes a method for adjusting the loudness and the spectral content of digital audio signals in a real-time using warped spectral filtering. A warped processing module modifies a spectral content of a digital audio signal with a set of gains for a plurality of non-linearly-scaled frequency bands determined by a warping factor λ of a warped delay line. Warped delay line signals, generated by the warped delay line, are processed by a warped filter block containing multiple warped finite impulse response filters, e.g., Mth band filters, using individual warped spectral filtering in said plurality of the non-linearly-scaled frequency bands, which is followed by a conventional processing by a dynamic range control/equalization block. The present invention describes another innovation, that is embedding the warped processing module in a two-channel quadrature mirror filter (QMF) bank for improving processing efficiency at high sample rates.
摘要:
This invention describes a method for adjusting the loudness and the spectral content of digital audio signals in a real-time using warped spectral filtering. A warped processing module modifies a spectral content of a digital audio signal with a set of gains for a plurality of non-linearly-scaled frequency bands determined by a warping factor λ of a warped delay line. Warped delay line signals, generated by the warped delay line, are processed by a warped filter block containing multiple warped finite impulse response filters, e.g., Mth band filters, using individual warped spectral filtering in said plurality of the non-linearly-scaled frequency bands, which is followed by a conventional processing by a dynamic range control/equalization block. The present invention describes another innovation, that is embedding the warped processing module in a two-channel quadrature mirror filter (QMF) bank for improving processing efficiency at high sample rates.
摘要:
This invention relates to a device, a method, a software application program, a software application program product and an audio device for processing a digital signal, wherein the digital signal is separated and downsampled into at least two downsampled subband signals, wherein at least one of the at least two downsampled subband signals is equalized, and wherein the at least two downsampled subband signals are upsampled and combined into a digital output signal.
摘要:
An apparatus for providing a beat and tatum tracker includes an accent filter bank, a periodicity estimator, a period estimator and a phase estimator. The accent filter bank is configured to downsample an input audio signal. The periodicity estimator is configured to determine a periodicity based on the downsampled signal. The period estimator is configured to determine a period based on the periodicity. The phase estimator is configured to estimate a phase based on the period for determining beat and tatum times of the input audio signal.
摘要:
An apparatus for utilizing spatial information for audio signal enhancement in a multiple distributed network may include a processor. The processor may be configured to receive representations of a plurality of audio signals including at least one audio signal received at a first device and at least a second audio signal received at a second device. The first and second devices may be part of a common acoustic space network and may be arbitrarily positioned with respect to each other. The processor may be further configured to combine the first and second audio signals to form a composite audio signal, and provide for communication of the composite audio signal along with spatial information relating to a sound source of at least one of the plurality of audio signals to another device.
摘要:
An apparatus for utilizing spatial information for audio signal enhancement in a multiple distributed network may include a processor. The processor may be configured to receive representations of a plurality of audio signals including at least one audio signal received at a first device and at least a second audio signal received at a second device. The first and second devices may be part of a common acoustic space network and may be arbitrarily positioned with respect to each other. The processor may be further configured to combine the first and second audio signals to form a composite audio signal, and provide for communication of the composite audio signal along with spatial information relating to a sound source of at least one of the plurality of audio signals to another device.