Patent classifications
H04S3/002
AUDIO SIGNAL PROCESSING APPARATUS AND METHOD FOR CROSSTALK REDUCTION OF AN AUDIO SIGNAL
The disclosure relates to an audio signal processing apparatus for filtering a left channel input audio signal (L) and a right channel input audio signal (R), a left channel output audio signal (X.sub.1) and a right channel output audio signal (X.sub.2) to be transmitted over acoustic propagation paths to a listener, wherein transfer functions of the acoustic propagation paths are defined by an acoustic transfer function matrix. The audio signal processing apparatus comprises a decomposer, a first cross-talk reducer, a second cross-talk reducer, and a combiner. The first cross-talk reducer is configured to reduce a cross-talk within a first predetermined frequency band upon the basis of the acoustic transfer function matrix. The second cross-talk reducer is configured to reduce a cross-talk within a second predetermined frequency band upon the basis of the acoustic transfer function matrix.
IMMERSIVE AUDIO REPRODUCTION SYSTEMS
Systems and methods can provide an elevated, virtual loudspeaker source in a three-dimensional soundfield using loudspeakers in a horizontal plane. In an example, a processor circuit can receive at least one height audio signal that includes information intended for reproduction using a loudspeaker that is elevated relative to a listener, and optionally offset from the listener's facing direction by a specified azimuth angle. A first virtual height filter can be selected for use based on the specified azimuth angle virtualized audio signal can be generated by applying the first virtual height filter to the at least one height audio signal. When the virtualized audio signal is reproduced using one or more loudspeakers in the horizontal plane, the virtualized audio signal can be perceived by the listener as originating from an elevated loudspeaker source that corresponds to the azimuth angle.
Method, Apparatus or Systems for Processing Audio Objects
Diffuse or spatially large audio objects may be identified for special processing. A decorrelation process may be performed on audio signals corresponding to the large audio objects to produce decorrelated large audio object audio signals. These decorrelated large audio object audio signals may be associated with object locations, which may be stationary or time-varying locations. For example, the decorrelated large audio object audio signals may be rendered to virtual or actual speaker locations. The output of such a rendering process may be input to a scene simplification process. The decorrelation, associating and/or scene simplification processes may be performed prior to a process of encoding the audio data.
APPARATUS AND METHOD FOR DETERMINING DELAY AND GAIN PARAMETERS FOR CALIBRATING A MULTI CHANNEL AUDIO SYSTEM
A method and an apparatus for adjusting delay and gain parameters for calibrating a multichannel audio system to which a plurality of loudspeakers is connected. A calibration process includes emitting a plurality of test tones by an audio processing device on a plurality of loudspeakers with predetermined timings and amplitude levels, according to a calibration signal. A calibration device having a microphone captures the audio signal corresponding to the test tones from the listener's position. The captured audio signal is analyzed, either by the calibration device or the audio processing device, to determine the delays between loudspeakers and difference of amplitude levels between loudspeakers. Corresponding delay and gain parameters are determined and used by the audio processing device to correct the sound to be played back. A calibration device and an audio processing device implementing the method are disclosed as well as a calibration signal utilized in the calibration process.
IN-VEHICLE ACOUSTIC SYSTEM AND VEHICLE PROVIDED WITH THIS IN-VEHICLE ACOUSTIC SYSTEM
An in-vehicle acoustic system includes a first amplifier that outputs a first audio signal to a first mid-range speaker; a second amplifier that outputs a second audio signal to a second mid-range speaker; a third amplifier that outputs a third audio signal to a first high-range speaker and a first low-range speaker; and a fourth amplifier that outputs a fourth audio signal to a second high-range speaker and a second low-range speaker, in which the third audio signal is inputted to the first high-range speaker, the fourth audio signal is inputted to the second high-range speaker, the first audio signal is inputted to a deep-bass speaker, the third audio signal is inputted to the first low-range speaker, the fourth audio signal is inputted to the second low-range speaker.
Button sound-emitting apparatus and electronic device
Disclosed are a button sound-emitting apparatus and an electronic device. The button sound-emitting apparatus comprises a cap part, a exciter, and an elastic component; said elastic component is arranged at an edge of the cap part; said exciter is provided in the middle of the cap part; the elastic component is connected to a housing of the electronic device; the exciter is suspended in the housing; the elastic component is configured for providing an elastic restoring force; the exciter is configured to be used for providing a driving force. The button sound-emitting apparatus is simple in structure and eliminates the need to arrange a front acoustic cavity and a sound outlet hole, and ensures good sound performance.
Method and apparatus for binaural rendering audio signal using variable order filtering in frequency domain
The present invention relates to a method and an apparatus for binaural rendering an audio signal using variable order filtering in frequency domain. To this end, provided are a method for processing an audio signal including: receiving an input audio signal; receiving a set of truncated subband filter coefficients for filtering each subband signal of the input audio signal, the set of truncated subband filter coefficients being constituted by one or more FFT filter coefficients generated by performing FFT by a predetermined block size; generating at least one subframe for each subband; generating at least one filtered subframe for each subband; performing inverse FFT on the filtered subframe for each subband; and generating a filtered subband signal by overlap-adding the transformed subframe for each subband and an apparatus for processing an audio signal using the same.
Stereo unfold with psychoacoustic grouping phenomenon
The Stereo Unfold Technology solves the inherent problems in the stereo reproduction by utilizing modern DSP technology to extract information from the Left (L) and Right (R) stereo channels to create a number of new channels that feeds into processing algorithms. The Stereo Unfold Technology operates by sending the ordinary stereo information in the customary way towards the listener to establish the perceived location of performers in the sound field with great accuracy and then projects delayed and frequency shaped extracted signals forward as well as in other directions to provide additional psychoacoustically based clues to the ear and brain. The additional clues generate the sensation of increased detail and transparency as well as establishing the three dimensional properties of the sound sources and the acoustic environment in which they are performing. The Stereo Unfold Technology manages to create a real believable three-dimensional soundstage populated with three-dimensional sound sources generating sound in a continuous real sounding acoustic environment.
In-vehicle independent sound field system and control system based on mini-speakers
An in-vehicle independent sound field system and control system based on mini-speakers are provided. The system includes a mini-speaker array installed in front of a user and a mini-speaker installed near the user's ear. Each mini-speaker is provided with a coupling rear cavity, and the mini-speaker near the user's ear is provided with a sound guide front cavity. A controller is provided to control the mini-speaker array to perform an intermediate-and-high frequency directional replay, and control the mini-speakers near the user's ear to perform low-frequency replay and autonomous counterbalance, to solve the problem that a speaker array cannot be installed with a traditional in-vehicle speaker size and that a low-frequency effect is bad when merely using a speaker array to control sound direction, Besides, when different sources are being played, they rarely interfere with each other, thereby improving personalized requirements for sound and user experience of users in a vehicle.
ANTI-CAUSAL FILTER FOR AUDIO SIGNAL PROCESSING
An audio signal processor includes a digital filter block configured to receive an audio signal and output a first filtered audio signal, and a phase linearization block configured to receive the first filtered audio signal and output a second filtered audio signal with a more linear phase.