396results about "Two-channel systems" patented technology

A device and method for providing spatialized audio with dynamic head tracking that includes, in a static phase, providing a spatialized acoustic signal to a user that is perceived as originating from a virtual soundstage at a first location, and, upon determining one or more predetermined conditions are satisfied, which can include whether the users head has exceeded an angular bound, rotating the virtual soundstage to track the movement of the user's head.

Methods, apparatus, systems, and articles of manufacture are disclosed to generate spatial audio based on computer vision. An example apparatus includes at least one memory, instructions in the apparatus, and processor circuitry to execute the instructions to determine a position of an audio source based on an image generated via a camera, and apply an audio spatialization filter to an audio signal generated by a microphone based on the position of the audio source.

Disclosed are systems, apparatuses, processes, and computer-readable media to. According to some aspects, a method of processing audio data may include obtaining, at a computing device, sensing information from an audio device outputting a spatial audio stream for a user, wherein the audio device includes a first audio output device and a second audio output device; determining, based on the sensing information, that the second audio output device is not in use; modifying the spatial audio stream based on determining that the second audio output device is not in use and a head pose of the user to create a modified spatial audio stream; and providing the modified spatial audio stream to the first audio output device.

The invention relates to an audio processing method and device, a storage medium, a controller and a vehicle, and the method comprises the steps: receiving an audio signal inputted by a sound source, determining a processing mode of the audio signal according to the type of the sound source, processing the audio signal based on the processing mode, and obtaining a processed audio signal, and finally, the processed audio signal is transmitted to an audio playing assembly of the target sound area for playing, and the audio playing assembly comprises loudspeakers installed at at least two different target positions, so that a three-dimensional sound field space is constructed in the target sound area. Therefore, the audio processing effect can be improved on the basis of saving the cost, and an independent sound area and three-dimensional audio experience are provided for a user.

Techniques are described for wirelessly receiving, at first and second speaker assemblies, signals representing left and right audio channels from each of first and second sources, and using the signals, simultaneously playing, on both speaker assemblies, music, computer game audio, computer chat audio, and speech picked up by speaker microphones.

Surround sound systems can dramatically expand the size of a user's sound field. Much surround sound content is mixed in a simplistic way where the front audio is copied to the rear, at a lower volume. It can be difficult for users to appreciate the value proposition of a surround sound system without more compelling spatially complex content. Quantifying surround sound complexity of various content items based on an audio spatial complexity scoring system can address this issue. Algorithms can be implemented to determine an audio spatial complexity score based on audio channels of a content item. Large catalog of content items can be analyzed, and audio spatial complexity scores can be associated with various content items. If a user has a surround sound system, content items with a high audio spatial complexity can be retrieved or recommended to the user to demonstrate the surround sound system's value better.

In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

This application provides a display device and its audio output method. The display device includes a display screen, a first speaker, and a second speaker. The first speaker is located behind the display screen, and its sound emission direction is towards the upper rear of the display device. The second speaker's sound emission direction is towards the front of the display device or towards the lower part of the display device. The first speaker and the second speaker emit sound asynchronously. The display device of this application has good stereo sound effects.

A lavalier microphone has improved dynamic range. The lavalier microphone incorporates a pair of microphone transducer elements into the same housing or attached housings, with one of the microphone transducer elements being more sensitive to acoustic input than the other. The voltage signals from the respective transducer elements are processed and combined, resulting in a digital output audio signal with improved signal to noise both at lower and higher input acoustic levels.

A delay estimation method includes determining a cross-correlation coefficient of a multi-channel signal of a current frame, determining a delay track estimation value of the current frame based on buffered inter-channel time difference information of at least one past frame, determining an adaptive window function of the current frame, performing weighting on the cross-correlation coefficient based on the delay track estimation value of the current frame and the adaptive window function of the current frame, to obtain a weighted cross-correlation coefficient, and determining an inter-channel time difference of the current frame based on the weighted cross-correlation coefficient.

This invention is entitled "Arrangement for Generating a Head-Related Transfer Function Filter". The invention discloses an arrangement for acquiring images used to generate a head-related transfer function filter. In this arrangement, the camera of a mobile phone or similar portable device is adjusted to take an image. All acquired images are analyzed, and only suitable images are further sent for generating the head-related transfer function filter. The arrangement is further configured to provide instructions to the user to adequately cover the entire head and other relevant body parts.

The present invention provides a method for processing music audio data, comprising the steps of providing input audio data representing a first piece of music containing a mixture of predetermined musical timbres, decomposing the input audio data to generate at least a first audio track representing a first musical timbre selected from the predetermined musical timbres, and a second audio track representing a second musical timbre selected from the predetermined musical timbres, applying a predetermined first audio effect to the first audio track, applying no audio effect or a predetermined second audio effect, which is different from the first audio effect, to the second audio track, and obtaining recombined audio data by recombining the first audio track with the second audio track.

A sound processing method obtains first audio data representing first sound, obtains second audio data representing second sound created in advance, analyzes the first audio data, compares the second audio data with an analysis result of the analyzing, reproduces third audio data from the second audio data by omitting a type of sound from the second sound that matches a type of sound in the first sound, based on a comparison result of the comparing, and outputs an audio signal representing the reproduced third audio data.

A computer system and a processing method thereof of sound signal are provided. The computer system includes a platform path controller (PCH), a high-definition audio (HDA) codec, and a digital microphone. The HDA codec is coupled to the PCH. The digital microphone is coupled to the PCH and the HDA codec. The digital microphone is used to generate sound signal. The HDA codec processes the sound signal from the digital microphone. Accordingly, the audio recording with high quality could be provided.

Example techniques relate to audio generation in a media playback system. Based on one or more first functions and first characteristics of an area, the system may generate first audio that includes a first audio signal and a second audio signal. The system provides the first audio signal to at least one first audio driver and the second audio signal to at least one second audio driver, thereby causing a first playback device and a second playback device to play back the first audio synchronously. The system receives second characteristics of the area and based on one or more second functions and the second characteristics, generates second audio comprising a third audio signal and a fourth audio signal. The system provides the third audio signal to the at least one first audio driver and the fourth audio signal to the at least one second audio driver.

The present invention improves the quality of signals of a virtual stereo microphone. This communication device comprises: a reception circuit that receives encoded information obtained by encoding signals of a stereo microphone; and a generation circuit that generates signals of a virtual stereo microphone configured from virtual microphones arranged on both sides of the stereo microphone through an extrapolation process performed on the phase spectrum of a decoded stereo signal obtained by decoding the encoded information in a frequency band in which the phase difference of signals between microphones constituting the stereo microphone is 2π or less.

Audio signal processor for generating a two-channel audio signal, comprising: an input interface for providing single-channel acoustic data describing an acoustic environment; a two-channel synthesizer for synthesizing two-channel acoustic data from the single-channel acoustic data using a listener position or rotation; and a sound generator for generating the two-channel audio signal from an audio signal and the two-channel acoustic data, wherein the two-channel synthesizer is configured to separate the single-channel acoustic data into at least two parts consisting of a direct sound part and at least one of an early reflection part and a late reverberation part, and to individually process the at least two parts for generating two-channel acoustic data for each part, to determine a separation time instant in the single-channel acoustic data between the direct sound part and the early reflection part or between the early reflection part and the late reverberation part, to extend at least one of the two parts of the separation time instant by a certain number of samples, so that an overlap at the separation time instant is achieved, and to window the at least one extended part using a certain window function accounting for the sample extension.

An apparatus comprises receivers (101, 102) receiving microphone signals and a video signal capturing a scene. An image generator (113) generates an image signal comprising images generated from the video signal and a zoom determiner (115) determines a zoom property, such as a zoom factor, for the images. An audio component generator (107) generates a stereo signal component for a given mono audio signal to comprise the mono audio signal positioned at a position in a stereo image of the stereo signal component. A combiner (109) generates the stereo audio signal to include at least one stereo signal component and an output (111) generates the audiovisual signal to comprise the stereo audio signal and the image signal. In the approach the position is determined in dependence on the zoom property, and typically also dependent on a direction of arrival for the given mono audio signal. The approach may allow closer consistency between visual and auditory spatial cues.

A computing apparatus with the proposed Method HRTF- for soundstage extension, upon which some inherent issues of the conventional Method HRTF such as the need for pre-measuring or simulating, highly individual dependence, small sweet spot, etc., have been effectively resolved. With the approaches that decoupling the individual-dependent segment from the entire HRTF chain, moving the intersection-point forward outside the ear, and simplifying the frequency response with relative-quantity approximations, the goal- "listening with one's own ear" is achieved with robustness improved and sweet spot enlarged. The present invention can be deployed in diverse scenarios: the stereo speakers; a foundational algorithmic module for advanced applications such as home theater systems, etc.; and an audio file in a particular encoding format can be also generated upon the invention that the soundstage can be extended readily through replaying with conventional stereo speakers.

The invention relates to a method for downmixing a stereo signal to obtain a monaural signal, the method comprising selecting (305), for a stereo signal frame, between: a first downmixing mode (T1), comprising filtering with phase correction in each channel of the stereo signal at an angle defined by half of a phase difference determined (303, 304) between two channels of the stereo signal, and-a second downmix mode (T2), comprising filtering with phase correction in one channel of the stereo signal according to an angle defined by the phase difference determined (303, 304) between two channels of the stereo signal, the selection of the first downmix mode or the second downmix mode depending on the value of the phase indicator determined (302) per frame. The invention also relates to a downmixing device for implementing the method.

To perform appropriate audio reproduction depending on the receiver's capabilities.SOLUTION: A receiver that receives broadcasting that includes MPEG-H audio in an audio component obtains identification information indicating whether the MPEG-H audio is present from broadcast airwaves at a multiplexing layer, and selects the audio component according to the receiver's capabilities on the basis of the identification information, and decodes the audio data of the selected audio component.SELECTED DRAWING: Figure 15

An audio signal processing method, and an audio signal processing device for executing the audio signal processing method includes, obtaining an audio signal and compressing the audio signal to generate a first compressed signal; calculating a duration for which a power value of the first compressed signal equals a first power threshold; determining whether the duration exceeds a first time threshold; when the duration exceeds the first time threshold, compressing the first compressed signal to generate a second compressed signal, and amplifying and outputting the second compressed signal.

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.

This application provides a stereo signal encoding method and apparatus. The encoding method includes: determining a residual signal encoding parameter of a current frame of a stereo signal based on downmixed signal energy and residual signal energy of each of M sub-bands of the current frame, where the residual signal encoding parameter of the current frame is used to indicate whether to encode residual signals of the M sub-bands, the M sub-bands are at least some of N sub-bands, N is a positive integer greater than 1, M ≤ N, and M is a positive integer (301); and determining, based on the residual signal encoding parameter of the current frame, whether to encode the residual signals of the M sub-bands of the current frame (302). According to the encoding method, high-frequency distortion of a decoded stereo signal can be reduced as much as possible while a spatial sense and sound-image stability of the decoded stereo signal are improved, thereby improving encoding quality

A method of using an earphone set to deliver binaural beats to a user (e.g., for altering a state of consciousness) and an earphone set 100 therefore are provided. The method involves simultaneously p

Multichannel audio data processing device for motor vehicles. A multichannel audio data processing device for motor vehicles includes a data storage unit (110) that receives multichannel audio data comprising rank audio data including a first audio channel and a second audio channel, a first rank digital filter, and a second rank digital filter for each rank. The first rank digital filter and a second rank digital filter for a given rank are determined from a digital filter transformation of rank correction data for each row of speakers. A computer (120) applies each first rank digital filter to all or part of the first audio channel of the rank audio data and each second rank digital filter to all or part of the second audio channel of the rank audio data. Fig. 1

This disclosure describes the classification (hereinafter referred to as "UNCLR classification") and crosstalk detection (hereinafter referred to as "XTALK detection") of irrelevant stereo content in an input stereo audio signal. This disclosure also describes stereo mode selection, such as automatic LRTD / DFT stereo mode selection. Furthermore, this disclosure uses the classification to select one of a first stereo mode and a second stereo mode for encoding a stereo audio signal including the left and right channels; to detect crosstalk in the stereo audio signal including the left and right channels in response to features extracted from the stereo audio signal including the left and right channels; or to classify irrelevant stereo content in the stereo audio signal including the left and right channels in response to features extracted from the stereo audio signal including the left and right channels.

The apparatus and methods of the present invention provide improved accuracy of response for a tactile transducer included in a body-mounted device such as a headphone, VR / AR headset or similar device. Accuracy is increased through the application of digital signal processing, such as with Infinite Impulse Response filters or Finite Impulse Response filters.

The present disclosure relates to a stereo sound decoding method and a stereo sound decoding system. The stereo sound decoding method includes receiving encoded parameters including encoded parameters of a primary channel and encoded parameters of a secondary channel, wherein the primary channel encoded parameters include LP filter coefficients of the primary channel; decoding the primary channel in response to the primary channel encoded parameters; and decoding the secondary channel using one of a plurality of encoding models, wherein (a) at least one of the encoding models uses the primary channel LP filter coefficients to decode the secondary channel, and (b) at least one of the encoding models uses primary channel encoded parameters other than the LP filter coefficients to decode the secondary channel.

An audio signal processor includes a harmonic signal generator to generate a harmonic signal that includes a signal generated by frequency scaling components of a target audio signal by a factor of k, a left audio signal generator to generate a left audio signal by convolving a head-related transfer function from a target localization position of the target audio signal to a left ear of a listener with the target audio signal, a right audio signal generator to generate a right audio signal by convolving a head-related transfer function from the target localization position to a right ear of the listener with the target audio signal, a left synthesizer to generate a left channel audio signal by synthesizing the left audio signal and the harmonic signal, and a right synthesizer to generate a right channel audio signal by synthesizing the right audio signal and the harmonic signal.