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:
Aspects of the invention provide methods, computer-readable media, and apparatuses for re-panning multiple audio signals by applying spatial cue coding. Sound sources in each of the signals may be re-panned before the signals are mixed to a combined signal. Processing may be applied in a conference bridge that receives two omni-directionally recorded audio signals. The conference bridge subsequently re-pans one of the signals to the listeners left side and the signal to the right side. The source image mapping and panning may further be adaptively based on the content and use case. Mapping may be done by manipulating the directional parameters prior to directional decoding or before directional mixing. Directional information that is associated with an audio input signal is remapped order to compress input source positions into virtual source positions. The virtual sources may be placed with respect to actual loudspeakers using binaural cue panning.
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:
The invention relates to an audio processing system 1. order to improve the audio processing, the system comprises at least one audio processing component 11, 12, 13 with a group of real-time functions 14 for processing audio data and a group of control functions 15 for processing control signals. The system further comprises at least one processor 16 providing a first process 20 for executing real-time functions 14 of the at least one audio processing component 11, 12, 13 using a basically constant processing power and at least one further process 30 for executing control functions 15 of the at least one audio processing component 11, 12, 13 when needed without affecting the processing power employed for the first process 20. The invention relates equally to a corresponding method and to a corresponding software program product.
摘要:
The invention relates to an audio processing system 1. In order to improve the audio processing, the system comprises at least one audio processing component 11, 12, 13 with a group of real-time functions 14 for processing audio data and a group of control functions 15 for processing control signals. The system further comprises at least one processor 16 providing a first process 20 for executing real-time functions 14 of the at least one audio processing component 11, 12, 13 using a basically constant processing power and at least one further process 30 for executing control functions 15 of the at least one audio processing component 11, 12, 13 when needed without affecting the processing power employed for the first process 20. The invention relates equally to a corresponding method and to a corresponding software program product.
摘要:
Provided are improved systems, methods, and computer program products for direct encoding of spatial sound into a directional audio coding format. The direct encoding may also include providing spatial information for a monophonic sound source. The direct encoding of spatial information may be used, for example, in interactive audio applications such as gaming environments and in teleconferencing applications such as multi-party teleconferencing.
摘要:
An apparatus for providing low frequency expansion of speech includes a nonlinear function element, a band-pass filter element and a level control element. The non-linear function element is configured to receive a signal including at least two harmonic components and to produce a signal including at least one lower frequency harmonic component having a lower frequency than a highest frequency component of the at least two harmonic components responsive to the signal including at least two harmonic components. The band-pass filter element is in communication with the non-linear function element and configured to filter the signal including the at least one lower frequency harmonic component. The level control element is configured to apply a level control to alter the filtered signal based on a feature vector associated with an input speech signal.
摘要:
A media subsystem of a processing element includes a plurality of elements and a latency manager. The plurality of elements are capable of processing media data including a plurality of instances wherein a first element inserts a length of media data into buffer(s) from which a second element thereafter reads the length of media data for subsequent output from the media subsystem. The latency manager is capable of determining a latency requirement of the media subsystem, and then dynamically tuning the length of media data inserted into the buffer(s) based upon the latency requirement, including increasing or decreasing the length of media data inserted into the buffer(s) during one or more instances(s).