5611results about "Stereophonic systems" patented technology

Techniques for controlling zone group and zone group characteristics such as audio volume in a multi-zone system are disclosed. The multi-zone system includes a number of multimedia players, each preferably located in a zone. A controller may control the operations of all of the zone players remotely from any one of the zones. Two or more zone players may be dynamically grouped as a zone group for synchronized operations. According to one aspect of the techniques, a zone group configuration can be managed, updated, modified via an interactive user interface provided in a controlling device. The zone group configuration may be saved in one of zone players. According to another aspect of the techniques, the audio volume control of a zone group can be performed individually or synchronously as a group.

Described herein are various methods and systems relating to the unification of media devices, and more specifically to the provision of wireless audio systems. In overview, two or more wireless speaker subsystem units substantially autonomously form a single wireless audio system having its own control interface. This control interface is used to apply operational changes across the wireless audio system, such as volume adjustment. That is, an operational change may be applied to the system as a whole, and this change is subsequently implemented by each of the individual wireless speaker subsystem units.

The present invention is directed to a method and system for automatic calibration of an acoustic system. The acoustic system may include a source A / V device, calibration computing device, and multiple rendering devices. The calibration system may include a calibration component attached to each rendering device and a source calibration module. The calibration component on each rendering device includes a microphone. The source calibration module includes distance and optional angle calculation tools for automatically determining a distance between the rendering device and a specified reference point upon return of the test signal from the calibration component.

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.

The apparatus for constructing a multi-channel output signal using an input signal and parametric side information, the input signal including the first input channel and the second input channel derived from an original multi-channel signal, and the parametric side information describing interrelations between channels of the multi-channel original signal uses base channels for synthesizing first and second output channels on one side of an assumed listener position, which are different from each other. The base channels are different from each other because of a coherence measure. Coherence between the base channels (for example the left and the left surround reconstructed channel) is reduced by calculating a base channel for one of those channels by a combination of the input channels, the combination being determined by the coherence measure. Thus, a high subjective quality of the reconstruction can be obtained because of an approximated original front / back coherence.

Techniques for loudspeaker localization are provided. Sound is received from a loudspeaker at a plurality of microphone locations. A plurality of audio signals is generated based on the sound received at the plurality of microphone locations. Location information is generated that indicates a loudspeaker location for the loudspeaker based on the plurality of audio signals. Whether the generated location information matches a predetermined desired loudspeaker location for the loudspeaker is determined. A corrective action with regard to the loudspeaker is enabled to be performed if the generated location information is determined to not match the predetermined desired loudspeaker location for the loudspeaker.

In processing a multi-channel audio signal having at least three original channels, a first downmix channel and a second downmix channel are provided, which are derived from the original channels. For a selected original channel of the original channels, channel side information are calculated such that a downmix channel or a combined downmix channel including the first and the second downmix channels, when weighted using the channel side information, results in an approximation of the selected original channel. The channel side information and the first and second downmix channels form output data to be transmitted to a decoder, which, in case of a low level decoder only decodes the first and second downmix channels or, in case of a high level decoder provides a full multi-channel audio signal based on the downmix channels and the channel side information. Since the channel side information only occupy a low number of bits, and since the decoder does not use dematrixing, an efficient and high quality multi-channel extension for stereo players and enhanced multi-channel players is obtained.

The invention relates to mobile audio playback devices and methods of its operation for enhanced music and sound experience for mobile devices. To provide stereo and surround sound, the present invention provides an audio playback device with a receiving means for receiving multi-channel audio data, obtaining means connected to the receiving means, for obtaining first channel audio data from the multi-channel audio data for playback on the audio playback device, and playback means having at least one loudspeaker, and being connected to the obtaining means for outputting the first channel audio data. The other not selected audio channels or all received audio data may be transferred to other terminals for playback or may be discarded.

A prefilter (5) for an audio system comprising a loudspeaker (1) in a room (2), which corrects both amplitude and phase errors due to the loudspeaker (1) by a linear phase correction filter response and corrects the amplitude response of the room (2) whilst introducing the minimum possible amount of extra phase distortion by employing a minimum phase correction filter stage. A test signal generator (8) generates a signal comprising a periodic frequency sweep with a greater phase repetition period than the frequency repetition period. A microphone (7) positioned at various points in the room (2) measures the audio signal processed by the room (2) and loudspeaker (1), and a coefficient calculator (6) (e.g. a digital signal processor device) derives the signal response of the room and thereby a requisite minimum phase correction to be cascaded with the linear phase correction already calculated for the loudspeaker (1). Filter (5) may comprise the same digital signal processor as the coefficient calculator (6). Applications in high fidelity audio reproduction, and in car stereo reproduction.

A multi-channel encoder / decoder scheme additionally preferably generates a waveform-type residual signal. This residual signal is transmitted together with one or more multi-channel parameters to a decoder. In contrast to a purely parametric multi-channel decoder, the enhanced decoder generates a multi-channel output signal having an improved output quality because of the additional residual signal.

The apparatus for constructing a multi-channel output signal using an input signal and parametric side information, the input signal including the first input channel and the second input channel derived from an original multi-channel signal, and the parametric side information describing interrelations between channels of the multi-channel original signal uses base channels for synthesizing first and second output channels on one side of an assumed listener position, which are different from each other. The base channels are different from each other because of a coherence measure. Coherence between the base channels (for example the left and the left surround reconstructed channel) is reduced by calculating a base channel for one of those channels by a combination of the input channels, the combination being determined by the coherence measure. Thus, a high subjective quality of the reconstruction can be obtained because of an approximated original front / back coherence.

An illustrative embodiment includes a method for use in performing acoustic calibration of at least one audio output device for a plurality of listening locations. An audio input device generates a data signal based on a series of one or more tones output by the at least one audio output device. The audio input device wirelessly transmits the data signal to a calibration device. The audio input device is one of a plurality of audio input devices deployed at respective ones of the plurality of listening locations. The data signal is one of a plurality of data signals generated by respective ones of the plurality of audio input devices based on the series of one or more tones output by the at least one audio output device. The plurality of data signals are wirelessly transmitted by the respective ones of the plurality of audio input devices to the calibration device.

A method of disassembling a pair of input signals L(t) and R(t) to form subband representations of N output channel signals o1(t), o2(t), . . . , oN(t), wherein t is time. The method includes the steps of generating a subband representation of the signal L(t) containing a plurality of subband components Lk(t) where k is an integer ranging from 1 to M; generating a subband representation of the signal R(t) containing a plurality of subband components Rk(t); and constructing the subband representation for each of the plurality of output channel signals, each of those subband representations containing a plurality of subband components oj,k(t), wherein oj,k(t) represents the kth subband of the jth output channel signal and is constructed by combining components of the input signals L(t) and R(t) according to an output construction rule: oj,k(t)=f(Lk(t),Rk(t)) for k=1, 2, . . . , M and j=1, 2, . . . , N.

Embodiments are described for a system of rendering object-based audio content through a system that includes individually addressable drivers, including at least one driver that is configured to project sound waves toward one or more surfaces within a listening environment for reflection to a listening area within the listening environment; a renderer configured to receive and process audio streams and one or more metadata sets associated with each of the audio streams and specifying a playback location of a respective audio stream; and a playback system coupled to the renderer and configured to render the audio streams to a plurality of audio feeds corresponding to the array of audio drivers in accordance with the one or more metadata sets.

The invention relates to an automated estimation of the position (co-ordinates) of a set of loudspeakers in a ioom Based on measured impulse responses the distances between each pair of loudspeakers are estimated, thereby forming a distance matrix, and the resultant distance matrix is used by a multidimensional scaling (MDS) algorithm to estimate the co-ordinates of each individual loudspeaker An improved co-ordinate estimation can, if desired, be derived by utilizing the stress values provided by the MDS algorithm.

The invention discloses a system applied to optimize the perceived sound quality in virtual sound zones. The system includes a method to establish a threshold of acceptability for an interfering audio programme on a target audio programme. The method includes physical parameters like target programme and interferer programme which combined with the scenarios: information gathering, entertainment and reading and / or working, constitutes modes of operations that may be processed and controlled by a system controller.

In a multi-speaker audio system for, e.g., a home entertainment system or other entertainment system, each networked-speaker (wired or wireless) is activated or deactivated as appropriate such that audio play follows a user as the user moves around the home.

The invention relates to a method and a system for adapting a loudspeaker to a specific listening position relative to the loudspeaker according to which method and system the acoustic power radiated by the loudspeaker is corrected by means of a correction filter inserted in the signal path through the loudspeaker, the response of said correction filter being determined by comparison between the a quantity characterising the radiated acoustic power measured at an actual listening position and a similar quantity measured at a reference listening position. According to a specific embodiment of the invention said characterising quantities are the radiation resistances measured at the actual listening position and the reference listening position respectively.

A system and method are provided for smoothing the in-room acoustic magnitude response of an audio reproduction system. An in-room acoustic magnitude response analysis is performed to determine a room resonance induced peak associated with an audio signal. A replica of the audio signal is filtered at the room resonance induced peak. The filtered replica signal is added with the audio signal. Through this, smoothing of the room resonance induced peak may be achieved, such that a subjective impression of transient response and dynamics of the audio signal are preserved.

An audio system to calibrate an audio signal based on a wirelessly received signal, and a signal calibration method are provided. The audio system includes a sound output unit to output a sound corresponding to a received audio signal. A transceiver is connected to an external device to enable wireless communication between a main unit and the external device. The external device converts the sound output from the sound output unit into an electric signal to generate a calibration audio signal. The main unit performs calibration on an audio signal to be played back through the sound output unit using the calibration audio signal.

An audio content playback method for a portable terminal The audio content playback method includes checking a channel that is supportable by audio content that is currently engaged in group's simultaneous playback, in group's simultaneous playback of the audio content. The method includes allocating a channel to each of devices included in a group based on position information of each device included in the group or based on an input state in a user interface environment that is preset for channel allocation for each device included in the group, and transmitting the allocated channel information to each device included in the group to allow the device to select its allocated channel and play the audio content.

Disclosed is subject matter that proposes a system and method for a media device to automatically detect the characteristics of an attached speaker. Speakers have many different characteristics, for example, power, impedance, frequency response, etc. With knowledge of the speaker characteristics, audio output can be equalized appropriately, and an amplifier of the media device, for example, can prevent exceeding the maximum power handling capability of the speaker. Described is a device and method for retrieving information about the speaker from a memory that is coupled to the speaker. A media device can read the data from the memory over existing speaker wires. Software and / or hardware in the media device can optimize the output to the attached speaker. Accordingly, the media device can interrogate the speaker directly over speaker wire to obtain the characteristics of the speaker.

A method of compensating for spatial audio frequency characteristics that varies in accordance with a mounting condition of a down firing speaker of an audio / video (AV) apparatus includes calculating a listening distance between the AV apparatus and a listener, calculating a distance between a speaker mounted on the AV apparatus and a neighboring reflective surface, setting a spatial frequency compensation filter value and a speaker frequency characteristic compensation filter value based on the calculated distances, and compensating for frequency characteristics of an audio signal by combining the spatial frequency compensation filter value and the speaker frequency characteristic compensation filter value.

A sound reproducing apparatus and a sound reproducing method. The sound reproducing apparatus includes an actual listening environment feature function database where an actual listening space feature function is stored for correcting the virtual source in response to a feature of an actual listening space provided at the time of listening; and an actual listening space feature correcting unit of reading out the actual listening space feature function stored in the actual listening environment feature function database, and correcting the virtual source based on the reading result. Accordingly, causes of each distortion may be removed to provide sounds having the best quality.

A method for equalizing audio and a video apparatus using the audio equalizing method are provided. The method for equalizing audio includes detecting the distance between a speaker mounted in a video apparatus and a reflective surface, and equalizing an audio signal to be output from the speaker based on the detected distance. Accordingly, attenuation of audio output is reduced, so audio output is optimized.

An audio system installed in a listening space may include a signal processor and a plurality of loudspeakers. The audio system may be tuned with an automated audio tuning system to optimize the sound output of the loudspeakers within the listening space. The automated audio tuning system may provide automated processing to determine at least one of a plurality of settings, such as channel equalization settings, delay settings, gain settings, crossover settings, bass optimization settings and group equalization settings. The settings may be generated by the automated audio tuning system based on an audio response produced by the loudspeakers in the audio system. The automated tuning system may generate simulations of the application of settings to the audio response to optimize tuning.