Abstract: A system and method are described for selecting a target sound object from a sound mixture. In embodiments, a sound mixture comprises a plurality of sound objects superimposed in time. A user can select one of these sound objects by providing reference audio data corresponding to a reference sound object. The system analyzes the audio data and the reference audio data to identify a portion of the audio data corresponding to a target sound object in the mixture that is most similar to the reference sound object. The analysis may include decomposing the reference audio data into a plurality of reference components and the sound mixture into a plurality of components guided by the reference components. The target sound object can be re-synthesized from the target components.

Abstract: The present invention provides an audio codec for converting digital audio signals to analogue audio signals, the audio codec comprising: two digital audio bus interfaces for coupling to respective digital audio buses; a digital-only signal path between the two digital audio bus interfaces, such that no analogue processing of the audio signals occurs in the digital-only signal path.

Abstract: An embodiment of the present disclosure provides an audio mixing processing method for audio signals. The method includes: determining a terminal that needs to adjust a location of an audio signal, and adjusting location information about the audio signal for the terminal; and performing audio mixing processing for the audio signal whose location is adjusted and other signals to be audio mixed. An embodiment of the present disclosure further provides an audio mixing processing apparatus for audio signals. According to the solutions provided in the embodiments, the audio location information about terminals whose audio signal locations overlap are adjusted, so that the location of each sending terminal is separated from each other as far as possible, and the sound of each sending terminal is more clear, thereby improving the field experience and feeling of a user.

Abstract: A method, an apparatus, and a system for cross-platform conference convergence are disclosed in embodiments of the present invention, where implementation of the method includes: receiving, after attending a first conference, an INVITE request sent by a first application server, where the INVITE request is a request for inviting a second terminal served by a second application server to attend a first conference; sending an invitation call to the second application server to request creating a second conference; and sending, after receiving conference creation success information sent by the second application server, an invitation to the second application server to invite the second terminal to attend the second conference. Usage of a private line between an enterprise and an operator can be eliminated in the entire implementation process.

Abstract: A distributed audio signal processing system having a plurality of linked audio signal processing units is disclosed. Each audio signal processing unit has physical channels for receiving and sending local audio signals and a high bandwidth interface for exchanging audio signals with other linked audio signal processing units. Each of the physical channels of each of the audio signal processing units are mapped to a corresponding global channel. The audio signal processing system provides a tightly integrated, distributed signal processing matrix where any input signal or physical channel on any one of the signal processing units can be exchanged with any other of the linked audio signal processing units, thereby providing maximum flexibility for selecting signals to mix and determining how to process the signals.

Abstract: The invention relates to a hearing aid with at least one sound receiver and a sound generator. The hearing aid has an audio signal unit operatively connected to the sound generator, the audio signal unit being designed to generate an audio signal perceptible to a human ear. The audio signal has a plurality of mutually different frequencies of a frequency range. The audio signal unit has at least one tone generator. The at least one tone generator is designed to generate the audio signal with at least one fundamental frequency representing a tone and with harmonics of the fundamental frequency.

Abstract: Systems and methods for time-aligning a first and second media signals. The first and second media signals may be audio signals recorded for combining as one audio performance or as an audio portion of an audio-video recording. The system retrieves the audio signals as digital samples generated using a common sampling rate. The system includes at least one coarse alignment function for reducing an initial misalignment between the first and second signal. The coarse alignment function may be configured to generate a coarse-shifted second signal. A fine alignment function determines an aligning time shift between the coarse-shifted second signal and the first signal by performing a cross-correlation between the first signal and the coarse-shifted second signal.

Abstract: Described herein are one or more implementations offering a centralized (e.g., single-point) user-interface, where a user may manipulate audio properties (e.g., volume level) for one or more audio submixes produced by each of multiple active operating-system processes.

Abstract: User is allowed to designate a desired mode defining the respective numbers of channels and mixing buses, and processing for mixing input signals of the number of channels corresponding to the designated mode is performed repetitively to generate signals for the individual buses. The time of arrival of the last step in the mixing processing for the number of channels, corresponding to the designated mode, is detected to output an accumulation result obtained at the last step, and new accumulation is started with a digital audio signal inputted at a step following the last step. Digital audio signals processed by a first signal processing circuit are stored into a memory and transmitted to a second signal processing circuit via a cascade-connection. The second signal processing circuit adds the audio signal, processed for each of the steps, to audio signals input via the cascade-connection and writes added signal into the memory.

Abstract: In an apparatus having a display for supplying parameters to a mixing system, a first display control section controls the display to present a scene list containing identifications of each scene and configuration data of the mixing system specified by first specification data contained in each scene. A scene selection acceptance section accepts selection of a scene from the scene list presented on the display. A second display control section controls the display to present operation data of the mixing system specified by second specification data contained in the selected scene, such that the display presents each portion of the operation data. Each portion contains one option of the operation data specified by the second specification data in manner distinguishable from another option not specified by the second specification data.

Abstract: To provide an enhanced true-to-life atmosphere enjoyed in multipoint connecting, and reduce a calculation load at a multipoint connection unit, as well.

Abstract: An audio processor (202) receives a non-priority audio signal (302) and a priority audio signal (304). The priority audio signal occupies a frequency band (408). The audio processor filters (320) the non-priority audio signal by suppressing frequency content in the same frequency region occupied by the priority signal, creating a filtered non-priority signal (412). The filtered non-priority signal and the priority signal are combined (328) and played over an audio transducer (110).

Abstract: Briefly, in accordance with one or more embodiments, a decentralized mixer includes an audio input circuit to receive one or more local audio input signals, a network adapter to couple to a network and to receive one or more mix busses from one or more other mixers coupled to the network, a digital signal processor to create a local mix of the local audio input signals in combination with the mix busses from the network, and a recording circuit to record the local mix, and/or to record the raw signals from the mix busses and the local audio input signals.

Abstract: In each of a plurality of input channels of an audio mixer, there are provided a fader operator for adjusting a tone volume level, and a send level adjustment section for adjusting a send level of an audio signal to be sent from the channel to individual mixing buses. On an operation panel of the mixer, there are provided a plurality of bus selection switches in corresponding relation to the mixing buses. Once any one of the bus selection switches is depressed once, the mixing bus corresponding to the operated bus selection switch is allocated to a selected send level operator. Activation/deactivation of a Sends On Fader (SOF) function may be instructed by a user depressing any one of the bus selection switches twice in succession.

Abstract: A system for substantially simultaneous data presentation is provided having a processor and memory with instructions for execution by the processor for: receiving attended data, including data for a user to attends to by paying attention to the data; receiving unattended data, which includes data that the user does not have to attend to in order to sense and process it; and generating presentation data. The presentation is suitable for presentation by a presentation device for substantially simultaneous presentation of the attended data and the unattended data in a manner in which the unattended data at least partially overlaps the attended data and the unattended data does not interfere with the attended data by replacing, erasing, or suppressing the attended data, and vice versa. The presented unattended data gives unconscious cues that the user can sense and provide information to the user while attending to the presented attended data, without attending to the presented unattended data.

Abstract: A method, a device and a system for mixing processing of an audio signal are provided in the embodiments of the present invention. The method includes: judging a channel type of a receiving terminal; for a single-channel receiving terminal, sending a mixed audio signal and meanwhile sending location information of a sending terminal that has maximum audio signal energy on each sub-band of the mixed audio signal to the single-channel receiving terminal; for a double-channel receiving terminal or a multi-channel receiving terminal, performing up-mixing to obtain double-channel or multi-channel audio data according to location information that is allocated to a single-channel sending terminal, performing mixing processing on audio data that participates in mixing to obtain double-channel or multi-channel mixed audio data, and sending the double-channel or multi-channel mixed audio data.

Abstract: A distributed audio signal processing system having a plurality of linked audio signal processing units is disclosed. Each audio signal processing unit has physical channels for receiving and sending local audio signals and a high bandwidth interface for exchanging audio signals with other linked audio signal processing units. Each of the physical channels of each of the audio signal processing units is mapped to a corresponding global channel. Global channels can be combined to form virtual channels that can be processed as a signal channel.

Abstract: Provided is an apparatus and method for coding and decoding multi-object audio signals with various channels and providing backward compatibility with a conventional spatial audio coding (SAC) bitstream. The apparatus includes: an audio object coding unit for coding audio-object signals inputted to the coding apparatus based on a spatial cue and creating rendering information for the coded audio-object signals, where the rendering information provides a coding apparatus including spatial cue information for audio-object signals; channel information of the audio-object signals; and identification information of the audio-object signals, and used in coding and decoding of the audio signals.

Abstract: A downmixer provides a listener of an output signal with a substantially accurate rendition of the apparent direction and relative loudness of an input signal. Downmixing certain channels of the input signal independently may substantially preserve the energy and intended direction of the input signal. The downmixer may include a test downmixer that operates over a limited frequency range to more accurately reflect the loudness of the input signal at the output, as perceived by a listener. The downmixer may demand fewer resources, freeing up resources for use in other operations.

Abstract: A distributed audio signal processing system having a plurality of linked audio signal processing units is disclosed. Each audio signal processing unit has physical channels for receiving and sending local audio signals and a high bandwidth interface for exchanging audio signals with other linked audio signal processing units. Each of the physical channels of each of the audio signal processing units is mapped to a corresponding global channel. Global channels can be combined to form virtual channels that can be processed as a signal channel. Combinations of virtual channels can be grouped into logical channel groups that can be controlled as a group.

Abstract: A method of processing an audio signal is disclosed. The present invention comprises receiving downmix signal including object signals, transforming the downmix signal per frequency band, determining a direction of an object signal from the transformed downmix signal, and determining blind information by estimating a level of the object signal corresponding to the direction. Accordingly, the present invention generates blind information in case of using an encoder incapable of generating object information, thereby enabling a gain and panning of object to be controlled using the blind information.

Abstract: A control device and an associated method for driving a multi-function speaker supporting a plurality of predetermined functions including at least an audio function and a non-audio function includes a digital signal mixing block and a digital-to-analog block. The digital signal mixing block is arranged for receiving a plurality of digital input signals corresponding to the predetermined functions, respectively, and generating a digital mixed signal according to the digital input signals. The digital-to-analog block is coupled to the digital signal mixing block, for generating an analog driving signal to the multi-function speaker according to the digital mixed signal.

Abstract: The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

Abstract: A signal manipulator for manipulating an audio signal having a transient event may have a transient remover, a signal processor and a signal inserter for inserting a time portion in a processed audio signal at a signal location where the transient event was removed before processing by the transient remover, so that a manipulated audio signal has a transient event not influenced by the processing, whereby the vertical coherence of the transient event is maintained instead of any processing performed in the signal processor, which would destroy the vertical coherence of a transient.

Abstract: An apparatus for processing a media signal and method thereof are disclosed, by which the media signal can be converted to a surround signal by using spatial information of the media signal. The present invention provides a method of processing a signal, the method comprising of extracting a downmix signal from a bitstream; generating a decorrelated downmix signal by applying a decorrelator to the downmix signal; and generating a surround signal by applying rendering information for generating a surround signal to the downmix signal and the decorrelated downmix signal.

Abstract: An audio mixing device that can get simpler processing done safely without depending on the property of an input signal is provided.

Abstract: Each of level meter block images 21 to 2m arranged on a touch panel display 1 displays, as a bar graph, respective signal levels of channels of a corresponding one of m blocks into which a plurality of channels are divided. The level meter block images 21 to 2m also serve as assignment selection buttons of an assignment selection portion 2. For each level meter block image, an image generation portion 8 generates not only an image of a bar graph but also an image which indicates respective color indicators of the respective channels such that, when a signal type of a channel indicates that there is a signal correlation between the channel and its neighboring channel, the channel has a color indicator of a connected shape shared with the neighboring channel, and synthesizes these images to output the synthesized image to an image display control circuit 3.

Abstract: A signal manipulator for manipulating an audio signal having a transient event may have a transient remover, a signal processor and a signal inserter for inserting a time portion in a processed audio signal at a signal location where the transient event was removed before processing by the transient remover, so that a manipulated audio signal has a transient event not influenced by the processing, whereby the vertical coherence of the transient event is maintained instead of any processing performed in the signal processor, which would destroy the vertical coherence of a transient.

Abstract: A method and apparatus are provided for generating a personalized radio channel playlist by simultaneously buffering multiple received channels from one or more source streams, and then selecting songs or tracks to playback from the buffered channels. Users can specify favorite channels for building their personal playlists, or multiple default playlist channels can be provided by genre or channels related in some other way. Navigation tools permit users to skip ahead and backward in the playback stream. A personalized radio channel playlist can be implemented as (1) content selected from buffered channels based on user preferences for artists, songs and the like, or (2) as a Mix Channel in which content from selected buffered channels is automatically mixed for playback in response to selection of a preset button assigned to the Mix Channel.

Abstract: In one aspect, audio streams are added to a mix until the mix is either complete (i.e., all audio streams have been added) or the mix is closed early (i.e., before the mix is complete). In another aspect, audio and video streams are synchronized by playing back the audio stream and then synchronizing display of the video frames to the playback of the audio stream.

Abstract: Audio encoding method and device comprising the transmission, in addition to the data representing a frequency-limited signal, of information relating to a temporal filter that is to be applied to the entire enhanced signal, both in its transmitted low-frequency part and in its reconstituted high-frequency part. The application of this filter for reshaping the reconstituted high-frequency part and the correction of compression artefacts present in the transmitted low-frequency part. In this way, the application of the temporal filter, simple and inexpensive, to all or part of the reconstituted signal, makes it possible to obtain a signal of good perceived quality.

Abstract: When a copy mode is set, all channel selecting switches on a control panel are set to valid for a parameter copy operation, in which, when an operation of the channel selecting switch is detected while the copy mode is set, (a) in a case where the operation of the channel selecting switch is determined to be a setting operation of a copy-source channel, the channel corresponding to the operated channel selecting switch is set as a copy-source channel and channel selecting switches corresponding to the channel of a different type from a type of the copy-source channel is set invalid so that only the channel selecting switch corresponding to the channel of the same type as of the copy-source channel is placed into a valid state, and (b) in a case where the operation of the channel selecting switch is determined to be the setting operation of a copy-destination channel, the channel corresponding to the channel selecting switch operated is set as a copy-destination channel.

Abstract: A method for processing an audio signal, comprising: receiving a downmix signal, an object information, and a mix information; generating a multi-channel information including at least one gain modification factor using the object information and the mix information, wherein the gain modification factor corresponds to a time-subband-variant factor for controlling gain of the downmix signal is disclosed.

Abstract: A system of linear equations is used to upmix a number N of audio signals to generate a larger number M of audio signals that are psychoacoustically decorrelated with respect to one another and that can be used to improve the representation of a diffuse sound field. The linear equations are defined by a matrix that specifies a set of vectors in an M dimensional space that are substantially orthogonal to each other. Methods for deriving the system of linear equations are disclosed.

Abstract: Methods are provided for solid-state implementation of a vacuum tube replacement device (VTRD) that includes a communication interface. The communication interface enables the VTRD to command or retrieve information regarding VTRD performance, function, and status. A VTRD may communicate with other VTRDs in a target amplifier retrofitted with compatible VTRDs and to an external user interface for programming or query/response. The methods presented for the vacuum tube replacement device system(s) are equally useful for non-vacuum tube systems such as audio amplifier circuits.

Abstract: An audio signal processing apparatus comprises input channels, a mix bus, a volume level controller, a path disconnector that connects or disconnects a path from each input channel to the mix bus, a signal selector that select either one of a post-signal and a pre-signal, a volume adjusting operator that groups at least a part of the plurality of input channels and collectively adjusts volume levels of the grouped input channels, a mute setting device that sets muting of the grouped input channels, a mute target selector that selects either one of a first mode in which only the post-signal is muted and a second mode in which both pre-signal and post-signal are muted, and a mute controller that controls, when the mute setting device has set muting of the grouped input channels, the volume level controller to execute the muting of the grouped input channels on the first mode, and controls the path disconnector to execute the muting of the grouped input channels on the second mode.

Abstract: A configurable amplifier circuit for a karaoke player is described. In one embodiment, a karaoke player includes: a DVD/CD reader port for coupling to a DVD/CD reader, a microphone input to receive audio signals from an external microphone, an MPEG decoder coupled to the DVD/CD reader port and the microphone input, the MPEG decoder having first and second output terminals, a processor coupled to the MPEG decoder, the processor to control the MPEG decoder, and a configurable amplifier coupled to the first and second output terminals, wherein a first audio signal originating from the first output terminal is selectively amplified by the configurable amplifier circuit in accordance with a second signal originating from the second output terminal.

Abstract: The quality of music output from audio systems is improved by simulating the effect of low frequency signals in the human ear. This thus allows listeners to perceive the lower frequency signals, even though the speakers may be incapable of providing such low frequency outputs. Method and systems provided for processing enhancing bass effect in audio signals. Said method and systems result in the bass enhancement being computationally less intensive. The bass effect enhancement techniques described are based on the response of sine and cosine transfer functions and on the directional independence of low frequency components. The human ear is unable to resolve directions from low frequency components. The bass effect enhancement technique alternatively is based on response of an exponential transfer function.

Abstract: The invention relates to a method and an apparatus for mixing audio signals. The invention is based on the idea to base selection of what audio channels to mix on the dynamic behavior of the audio channels in terms of whether they are active or not, rather than basing the selection on quantative measures of the audio channels. According to the invention, a set of audio channels are arranged in a mixing stack and a mixed audio signal is produced from the topmost channels above a predetermined threshold level in the stack. Whenever a channel becomes active, it is placed at the top of the stack.

Abstract: Calculation is performed for sound paths 112-1, 114-1 along which sounds emitted from a sound emitting point 104 in an acoustic space 102 are reflected and delivered to a sound receiving point 106. By the calculation, entering angles eR1, eR2 by which the sound paths enter the front side 106a of the sound receiving point 106 are obtained. Calculation is then performed to obtain angles by which respective speakers 52C, 52L, 52R, 52SR, 52SL of a 5.1 surround system are arranged in a listening room, with the front side 106a of the sound receiving point 106 centered thereon. Audio signals on the respective sound paths are distributed among channels for any two speakers. Consequently, sharp localization of sound images is achieved, requiring less calculation in simulating acoustic characteristics of the acoustic space 102 in which the sound emitting point 104 for emitting sounds and the sound receiving point 106 for receiving the sounds are placed.

Abstract: An apparatus for generating at least one audio output signal representing a superposition of at least two different audio objects comprises a processor for processing an audio input signal to provide an object representation of the audio input signal, where this object representation can be generated by a parametrically guided approximation of original objects using an object downmix signal. An object manipulator individually manipulates objects using audio object based metadata referring to the individual audio objects to obtain manipulated audio objects. The manipulated audio objects are mixed using an object mixer for finally obtaining an audio output signal having one or several channel signals depending on a specific rendering setup.

Abstract: A mixing control device has a plurality of input channels for processing and mixing audio signals according to parameters specified in the respective input channels. In the mixing control device, a plurality of channel strips are grouped into a first portion and a second portion, each channel strip having a fader operable for controlling a parameter. An allocating part allocates the plurality of the input channels to the plurality of the channel strips so that the fader of the channel strip is operable to control the parameter of the input channel allocated to the channel strip. An instructing part provides an instruction to switch the parameter to be controlled by the fader.

Abstract: This specification describes technologies relating to generating audio mixes for listening environments. A method is provided that includes receiving digital audio data; receiving an environmental input, the environmental input being associated with the listening environment; calculating one or more audio parameters for the digital audio data based on the received environmental input, the calculating including: calculating a particular intensity level for the digital audio data, and processing the digital audio data according to specified reference levels; and generating an audio mix for the digital audio data according to the calculated audio parameters.

Abstract: An integrated circuit for digital signal routing. Signal routing is achieved by means of a multiply-accumulate block, which takes data from one or more data source and, after any required scaling, generates output data for a data destination. The multiply-accumulate block operates on a time division multiplexed basis, so that multiple signal paths can be processed within one period of the sample clock. Each signal path has a respective sample clock rate, and paths with different sample clock rates can be routed through the multiply-accumulate block on a time division multiplexed basis independently of each other. Thus, speech signals at 8 kHz or 16 kHz can be processed concurrently with audio data at 44 kHz or 48 kHz.

Abstract: A method and system for performing audio signal processing is provided. A phase of a signal may be shifted between 0° and 360° for all frequencies of the signal, without any significant delay distortion. The phase shift can be accomplished through the use of two signal outputs (which are each shifted in phase by 90° relative to the other), inverters and an analog summing circuit. Selective use of the inverter on one or both of the 90° phase shifted outputs, and their linear combinations, allows the phase of the audio signal outputs to be shifted between about 0° and about 360° by a recording engineer, for example.

Abstract: A vehicle engine harmonic enhancement system. A method for operating the system includes providing a first engine harmonic enhancement audio signal and equalizing the first engine harmonic enhancement audio signal separately for each of the plurality of loudspeakers to provide individually equalized loudspeaker engine harmonic enhancement audio signals. A vehicle audio system includes a plurality of loudspeakers, an entertainment audio system, and an engine harmonic enhancement system. A method includes equalizing an entertainment audio signal to provide an equalized entertainment audio signal; providing an engine harmonic enhancement audio signal; equalizing the engine harmonic enhancement audio signal separately from the entertainment audio signal to provide an equalized engine harmonic enhancement audio signal; and combining the equalized entertainment audio signal with the equalized engine harmonic enhancement audio signal.

Abstract: An apparatus and method for autonomously mixing multiple devices capable of storing and playing audio signals is provided. Multiple devices can be mixed into one standard stereo signal that can then be played on any sort of powered speakers or amplifier. The apparatus is capable of receiving multiple audio inputs and can combine multiple iPods®, iPhones®, MP3 devices, or other devices capable of storing and playing audio signals, such that more than one device can be played at one time. No human intervention is required to control the device when the device is in autonomous mode. The autonomous mode can include random song playback using multiple devices.

Abstract: A semiconductor IC includes an analog audio input circuit, a selecting unit, and an audio processing circuit, which maybe a car navigation chip. The analog audio input circuit includes an RF module and at least one analog audio input module, respectively for providing a first analog audio input signal and at least one second analog audio input signal. The RF module includes an FM/AM broadcast receiving function. The selecting unit is coupled to the RF module and analog audio input module, and outputs a target analog audio input signal according to the first and second analog audio input signals. The audio processing circuit is coupled to the selecting unit, and performs an audio signal process upon the target analog audio signal to generate an audio output signal.

Abstract: An apparatus for generating at least one audio output signal representing a superposition of at least two different audio objects comprises a processor for processing an audio input signal to provide an object representation of the audio input signal, where this object representation can be generated by a parametrically guided approximation of original objects using an object downmix signal. An object manipulator individually manipulates objects using audio object based metadata referring to the individual audio objects to obtain manipulated audio objects. The manipulated audio objects are mixed using an object mixer for finally obtaining an audio output signal having one or several channel signals depending on a specific rendering setup.

Abstract: In a channel patching apparatus, an output patch setting part assigns a name to an audio signal of an internal channel of an audio device, which outputs the audio signal to a transmission channel of an audio network. A transmission channel allocation setting part allocates a transmission channel of the audio network to the audio signal assigned the name, and performs setting of the audio device such that the audio signal assigned the name is output from an internal channel associated with the audio signal assigned the name to the allocated transmission channel.