1907 results about "Vocal tract" patented technology

In an encoder, multiple channels of audio information representing multidimensional sound fields are split into subband signals and the subband signals in one or more subbands are combined to form composite signals. The composite signals, the subband signals not combined into a composite signal and information describing the spectral levels of subband signals combined into composite signals are assembled into an encoded output signal. The spectral level information conveys either the amplitude or power of the combined subband signals or the apparent direction of the sound field represented by the combined subband signals. In digital implementations, adaptive bit allocation may be used to reduce the informational requirements of the encoded signal.

Disclosed is a wireless surround sound speaker system wherein a transmitter broadcasts a variety of FM signals that correspond to the individual speaker channels commonly found in a surround sound system. Receivers, individually equipped with signal receiving, conditioning and amplification components, are configured to receive any one of the broadcast signals in a remote location and are used to drive a conventional loudspeaker in that location. Powered by wall socket or via DC battery packs, the receivers, used in conjunction with the transmitter, provide surround sound capabilities without the need for complex and difficult wiring.

An audio signal decorrelator for deriving an output audio signal from an input audio signal has a frequency analyzer for extracting from the input audio signal a first partial signal descriptive of an audio content in a first audio frequency range and a second partial signal descriptive of an audio content in a second audio frequency range having higher frequencies compared to the second audio frequency range. A partial signal modifier for modifies the first and second partial signals, to obtain first and second processed partial signals, so that a modulation amplitude of a time variant phase shift or time variant delay applied to the first partial signal is higher than that applied to the second partial signal, or for modifying only the first partial signal. A signal combiner combines the first and second processed partial signals, or combines the first processed partial signal and the second partial signal, to obtain an output audio signal.

An audio privacy system reduces the intelligibility of speech in an audio signal while preserving prosodic information, such as pitch, relative energy and intonation so that a listener has the ability to recognize environmental sounds but not the speech itself. An audio signal is processed to separate non-vocalic information, such as pitch and relative energy of speech, from vocalic regions, after which syllables are identified within the vocalic regions. Representations of the vocalic regions are computed to produce a vocal tract transfer function and an excitation. The vocal tract transfer function for each syllable is then replaced with the vocal tract transfer function from another prerecorded vocalic sound. In one aspect, the identity of the replacement vocalic sound is independent of the identity of the syllable being replaced. A modified audio signal is then synthesized with the original prosodic information and the modified vocal tract transfer function to produce unintelligible speech that preserves the pitch and energy of the speech as well as environmental sounds.

A two-channel phase-amplitude stereo encoding and decoding scheme enabling flexible and spatially accurate interactive 3-D audio reproduction via standard audio-only two-channel transmission. The encoding scheme allows associating a 2-D or 3-D positional localization to each of a plurality of sound sources by use of frequency independent inter-channel phase and amplitude differences. The decoder is based on frequency-domain spatial analysis of 2-D or 3-D directional cues in a two-channel stereo signal and re-synthesis of these cues using any preferred spatialization technique, thereby allowing faithful reproduction of positional audio cues and reverberation or ambient cues over arbitrary multi-channel loudspeaker reproduction formats or over headphones, while preserving source separation despite the intermediate encoding over only two audio channels.

The present invention describes techniques that can be used to provide novel methods of spatial audio rendering using adapted M-S matrix shuffler topologies. Such techniques include headphone and loudspeaker-based binaural signal simulation and rendering, stereo expansion, multichannel upmix and pseudo multichannel surround rendering.

An acoustic channel is used for fault diagnosis, repair, and upgrades. Remote diagnosis uses self-test data encoded into sound waves. Repair data and upgrades are also encoded and transmitted as sound waves.

A universal four-channel multimedia computer speaker system is connectable to audio sub-system control circuits (e.g., “sound cards”) of both the four-channel type and the conventional two-channel (stereo) type. With the audio sub-system control circuit being of a four-channel type, the universal four-channel audio system functions as a conventional multimedia computer four-channel surround sound audio system. With a four audio channel multimedia computer work and the audio sub-system control circuit being of a conventional two-channel (stereo) type, the universal four-channel audio system is configured to provide one pair of wide-band speakers (e.g., the front) with distinct audio playback according to respective right-front and left-front audio channels in a four or two audio channel multimedia computer work such as a game, music, etc. The universal four-channel audio system is configured to provide at the rear speakers distinct audio playback that are generated from the right-front and left-front audio channels in the four or two audio channel multimedia computer work. With respect to the rear speakers, the universal four-channel audio system includes a proxy audio signal component that provides respective right and left rear proxy audio signals to proximate the typical sound of actual right and left rear audio signals carried on a four audio channel multimedia computer work or simulated surround signals generated from conventional two channel multimedia work.

A panner is provided that incorporates a surround sound decoder. The panner takes as input the desired panning effect that a user requests, separates sounds using surround sound decoding, and places the separated sounds in the desired places in an output sound field.

A method and apparatus processes multi-channel audio by encoding, transmitting or recording “dry” audio tracks or “stems” in synchronous relationship with time-variable metadata controlled by a content producer and representing a desired degree and quality of diffusion. Audio tracks are compressed and transmitted in connection with synchronized metadata representing diffusion and preferably also mix and delay parameters. The separation of audio stems from diffusion metadata facilitates the customization of playback at the receiver, taking into account the characteristics of local playback environment.

Systems, methods, and apparatus for spectral contrast enhancement of speech signals, based on information from a noise reference that is derived by a spatially selective processing filter from a multichannel sensed audio signal, are disclosed.

An apparatus for and a method of audio recording in packet based telephony systems. Using the present invention, the equivalent of four audio channels are recorded utilizing only two recording channels. Each channel recorded comprises the stream of packets generated and transmitted by each endpoint to the other side. The RTP packets include the samples generated by the particular endpoint in addition to the timestamp of the samples received from the other side actually played by the endpoint. The recording device has knowledge of what was played at the other endpoint in order to accurately playback the audio samples generated by and received from the other endpoint. The recording device receives a packet stream containing the audio generated on each endpoint and the timestamp of the packet from the other side that was played on the endpoint. The recording device can reconstruct from this data the audio signal that was actually played on each endpoint.

An information handling system includes a signal processor such as an audio accelerator based sub-system with the capabilities to define output connectors such that the same set of connectors which provide, for example, home theatre AC3 (Dolby Digital) decoded signals (front, rear, cent/LFE) can provide, for example, 3 discrete stereo channels meant to be routed to different zones in, for example, a living or business space. The information handling system can provide a home theatre experience with one application, taking advantage of audio accelerator capabilities for AC3 signal decode, and provide multi-zone audio distribution, taking advantage of audio accelerator capabilities for decoding multiple streams of compressed digital audio. Home computing architectures that utilize a consumer personal computer (PC) are, thus, now suitable as the audio storage and manipulation center. In one embodiment, the PC provides DSP/decode pre-amplification functions and routes audio output to a generic 6-channel amplifier. In one embodiment, the system also includes a simple to use user interface (UI) to (i) select and initiate processing of multiple streams of audio for playback and (ii) map routing of the audio signals to stereo signal compatible jacks. Each pair of stereo output signals can be associated with a zone. Each zone can be represented in the UI such that the user can direct the desired audio stream to the desired zone. In another embodiment such as a broadcast situation, one stream can be sent to multiple zones, accomplishing distributed coverage with one audio stream.

A multi-channel audio system is constituted using at least one line array speaker unit, in which plural speakers are arrayed in line, wherein the same audio signal is supplied with a prescribed delay time to each of the speakers, thus forming plural sound beams. The plural sound beams are reflected on a wall surface and a ceiling of a room so as to form plural virtual sound sources surrounding a listening position, and emission directions and intensities of the sound beams are controlled so as to localize a phantom at a prescribed position based on the plural virtual sound sources. By appropriately arranging plural line array speaker units horizontally, vertically, and slantingly in such a way that each line array speaker unit forms sound beams distributed and spread in a sectorial form, it is possible to realize a surround audio system having a high degree of freedom with regard to setup positions for forming virtual sound sources.

A method and an apparatus to reproduce a multi-channel audio signal, in which mixing of a center channel signal is performed with a center channel signal in a home theater system. The method of reproducing a multi-channel audio signal includes calculating a delay value of a center channel signal according to location relationships of a listener, a center channel speaker and other channel speakers, regulating a time delay of the center channel signal according to the calculated delay value, and mixing the time-delay regulated center channel signal with other channel signals.

Binaural rendering a multi-channel audio signal into a binaural output signal is described. The multi-channel audio signal has a stereo downmix signal into which a plurality of audio signals are downmixed, and side information having a downmix information, as well as object level information of the plurality of audio signals and inter-object cross correlation information. Based on a first rendering prescription, a preliminary binaural signal is computed from the first and second channels of the stereo downmix signal. A decorrelated signal is generated as an perceptual equivalent to a mono downmix of the first and second channels of the stereo downmix signal being, however, decorrelated to the mono downmix. Depending on a second rendering prescription, a corrective binaural signal is computed from the decorrelated signal and the preliminary binaural signal is mixed with the corrective binaural signal to obtain the binaural output signal.

A desired acoustic signal is extracted from a noisy environment by generating a signal representative of the desired signal with processor (30). Processor (30) receives aural signals from two sensors (22, 24) each at a different location. The two inputs to processor (30) are converted from analog to digital format and then submitted to a discrete Fourier transform process to generate discrete spectral signal representations. The spectral signals are delayed to provide a number of intermediate signals, each corresponding to a different spatial location relative to the two sensors. Locations of the noise source and the desired source, and the spectral content of the desired signal are determined from the intermediate signal corresponding to the noise source locations. Inverse transformation of the selected intermediate signal followed by digital to analog conversion provides an output signal representative of the desired signal with output device (90). Techniques to localize multiple acoustic sources are also disclosed. Further, a technique to enhance noise reduction from multiple sources based on two-sensor reception is described.

A method for compressing audio input signals to form a master bit stream that can be scaled to form a scaled bit stream having an arbitrarily prescribed data rate. A hierarchical filterbank decomposes the input signal into a multi-resolution time/frequency representation from which the encoder can efficiently extract both tonal and residual components. The components are ranked and then quantized with reference to the same masking function or different psychoacoustic criteria. The selected tonal components are suitably encoded using differential coding extended to multichannel audio. The time-sample and scale factor components that make up the residual components are encoded using joint channel coding (JCC) extended to multichannel audio. A decoder uses an inverse hierarchical filterbank to reconstruct the audio signals from the tonal and residual components in the scaled bit stream.

A frequency-domain method for format conversion or reproduction of 2-channel or multi-channel audio signals such as recordings is described. The reproduction is based on spatial analysis of directional cues in the input audio signal and conversion of these cues into audio output signal cues for two or more channels in the frequency domain.

The present invention includes apparatuses and methods comprising a means for detecting the presence of speakers and microphones coupled to a portable multi-function device (such as Apple's iPhone™). In response, a portable multi-function device can adapt its output depending on the nature of the coupled headset device. In particular, a portable multi-function device containing the present invention can, upon detecting only one speaker in a coupled headset accessory device, combine the multiple channels of a stereo audio signal into a single mono audio signal. Likewise, a portable multi-function device containing the present invention can alert users to the absence of a coupled microphone.

Systems and methods for providing object-oriented audio are described. Audio objects can be created by associating sound sources with attributes of those sound sources, such as location, velocity, directivity, and the like. Audio objects can be used in place of or in addition to channels to distribute sound, for example, by streaming the audio objects over a network to a client device. The objects can define their locations in space with associated two or three dimensional coordinates. The objects can be adaptively streamed to the client device based on available network or client device resources. A renderer on the client device can use the attributes of the objects to determine how to render the objects. The renderer can further adapt the playback of the objects based on information about a rendering environment of the client device. Various examples of audio object creation techniques are also described.