Abstract: A scheme for stereo and multi-channel synthesis of inter-channel correlation (ICC) (normalized cross-correlation) cues for parametric stereo and multi-channel coding. The scheme synthesizes ICC cues such that they approximate those of the original. For that purpose, diffuse audio channels are generated and mixed with the transmitted combined (e.g., sum) signal(s). The diffuse audio channels are preferably generated using relatively long filters with exponentially decaying Gaussian impulse responses. Such impulse responses generate diffuse sound similar to late reverberation. An alternative implementation for reduced computational complexity is proposed, where inter-channel level difference (ICLD), inter-channel time difference (ICTD), and ICC synthesis are all carried out in the domain of a single short-time Fourier transform (STFT), including the filtering for diffuse sound generation.

Abstract: 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.

Abstract: A new audio playback architecture may be used, which allows the use of much larger buffering than that used by a typical audio subsystem in a computing system to improve power efficiency of the system and at the same time allows to maintain the quality (e.g., fidelity and responsiveness) of the audio playback. The audio controller in the new architecture may be made to report back to the host system a more accurate indication of which audio frame is being set to the audio codec than a currently available audio controller does. Additionally, the controller is capable of re-fetching previously buffered (but not yet transmitted) data. Furthermore, the controller may dynamically adjust the size of its local buffer.

Abstract: A multi-channel decoder is provided for decoding a mono signal and an associated inter-channel coherence measure, the inter-channel coherence measure representing a coherence between a plurality of original channels, the mono signal being derived from the plurality of original channels.

Abstract: A television audio signal encoder includes a matrix that sums a left channel audio signal and a right channel audio signal to produce a sum signal. The matrix also subtracts one of the left and right audio signals from the other to produce a difference signal. The encoder also includes a configurable infinite impulse response digital filter that selectively uses one or more sets of filter coefficients to filter the difference signal. Each selectable set of filter coefficients is associated with a unique filtering application to prepare the difference signal for transmission.

Abstract: A system and method of remotely enabling sound enhancement techniques is disclosed. In an embodiment, a watermark is embedded in an encoded multi-channel audio stream to remotely enable an enhancement decoder portion of a multi-channel audio decoder.

Abstract: An BTSC encoder with an improved digital filter structure substantially implemented on a single CMOS integrated circuit is described. By cascading first and second order allpass filter structures to form a higher order digital filter, such as a Cauer low pass filter, limit cycle oscillations are reduced or eliminated, word-length growth from one stage to the next is contained, and a more efficient overall filter structure and performance is obtained.

Abstract: A process for combining audio channels combines the audio channels to produce a combined audio channel and dynamically applies one or more of time, phase, and amplitude or power adjustments to the channels, to the combined channel, or to both the channels and the combined channel. One or more of the adjustments are controlled at least in part by a measure of auditory events in one or more of the channels and/or the combined channel. Applications include the presentation of multichannel audio in cinemas and vehicles. Not only methods, but also corresponding computer program implementations and apparatus implementations are included.

Abstract: A distributing circuit distributes audio signals of a predetermined number of channels to audio signals of a predetermined number of channels. A first signal processing circuit processes audio signals that are output from the distributing circuit in parallel, reproduces the resultant signals with a plurality of speakers, and localizes sound images of the individual audio signals at predetermined positions. A second signal processing circuit inputs audio signals that are output to the plurality of speakers and performs signal processes equivalent to the transfer functions from the individual speakers to both the ears of the listener. The output signals of the second signal processing circuit are reproduced with the headsets.

Abstract: In one embodiment, the method includes receiving audio frame data having at least first and second channel data. The first and second channel data includes a plurality of blocks, where the blocks are classified by a block type. Block information indicating the block type is obtained. The block information corresponds to the first and second channel data being common when the first and second channel data are paired. The first and second channel data is lossless decoded based on the block information.

Abstract: A frame type for a current SBR frame is determined according to a type of end border of a previous frame, as well as presence of a transient in the current SBR frame. A start border is determined according to the end border of the previous SBR frame. For a FIXFIX frame, a low time-resolution setting is used. For a FIXVAR or a VARVAR frame, a search for intermediate borders is conducted in the region between the transient and maximum allowed end border location. The end border is also determined at this stage. If there is excess capacity for more borders, another search is conducted in the region between the transient and the start border. For a VARFIX frame, only one search needs to be conducted, in the whole region partitioned by a variable start border and a fixed end border. All of the above are accomplished with two Forward Search operations and one Backward Search operation.

Abstract: A system and method of remotely enabling sound enhancement techniques is disclosed. In an embodiment, a watermark is embedded in an encoded multi-channel audio stream to remotely enable an enhancement decoder portion of a multi-channel audio decoder.

Abstract: 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.

Abstract: According to one embodiment, an audio outputting apparatus comprises: an audio outputting apparatus comprising: a first input unit to which a first compressed audio stream is input with respect to each frame, the first compressed audio stream including a compressed audio data and an extended data; a second input unit to which a second compressed audio stream is input with respect to each frame, the second compressed audio stream including a compressed audio data; a multiplex unit that inserts the second compressed audio stream into an extended data region of a corresponding frame of the first compressed audio stream to generate a multiplexed compressed audio stream; and an output unit that externally outputs the multiplexed compressed audio stream in a predetermined transmission mode.

Abstract: A method of encoding a multichannel signal, such as a stereophonic audio signal, including at least first and second signal components includes transforming at least the first and second signal components by a predetermined transformation into a principal signal including most of the signal energy and at least one residual signal including less energy than the principal signal. The predetermined transformation is parameterized by at least one transformation parameter. The method further includes representing the multichannel signal at least by the principal signal and the transformation parameter.

Abstract: Apparatus and method for transmitting and reproducing stereophonic audio signals are disclosed. The method comprises splitting first and second channels into two frequency bands, combining the lower frequency band signals of the two channels, and transmitting the combined signals of the two channels or signals representative thereof. The apparatus comprises: a splitter for splitting each of the two channels into two frequency bands, a combiner, for combining the lower frequency band signals of the two channels, and a transmitter for transmitting the combined signals, or signals representative thereof, and the higher frequency band signals of the two channels, or signals representative thereof.

Abstract: Provided is a method and apparatus for encoding/decoding an audio signal. Sections which are not used to output noise components near important spectral components and sub-bands which are not used to output noise components, are determined to be encoded or decoded, so that the efficiency of encoding and decoding an audio signal increases, and sound quality can be improved using less bits.

Abstract: 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.

Abstract: An apparatus for generating a multi-channel output signal performs a center channel cancellation to obtain improved base channels for reconstructing left-side output channels or right-side output channels. In particular, the apparatus includes a cancellation channel calculator for calculating a cancellation channel using information related to the original center channel available at the decoder. The device furthermore includes a combiner for combining a transmission channel with the cancellation channel. Finally, the apparatus includes a reconstructor for generating the multi-channel output signal. Due to the center channel cancellation, the channel reconstructor not only uses a different base channel for reconstructing the center channel but also uses base channels different from the transmission channels for reconstructing left and right output channels which have a reduced or even completely cancelled influence of the original center channel.

Abstract: A multi-channel audio reproduction apparatus and method for loudspeaker reproduction using virtual sound images whose positions can be adjusted is provided. The multi-channel audio reproduction apparatus includes a virtual sound image forming unit for compensating for the occurrence of cross-talk in at least one input audio signal according to the arrangement of loudspeakers, obtaining transfer functions occurring when sound from a position in a three dimensional space is transmitted to both ears of a listener, and forming a plurality of first virtual sound images in a three dimensional space using the transfer functions. A controller generates adjusting factors for adjusting the position of at least one second virtual sound image.

Abstract: The present invention provides improvements to prior art audio codecs that generate a stereo-illusion through post-processing of a received mono signal. These improvements are accomplished by extraction of stereo-image describing parameters at the encoder side, which are transmitted and subsequently used for control of a stereo generator at the decoder side. Furthermore, the invention bridges the gap between simple pseudo-stereo methods, and current methods of true stereo-coding, by using a new form of parametric stereo coding. A stereo-balance parameter is introduced, which enables more advanced stereo modes, and in addition forms the basis of a new method of stereo-coding of spectral envelopes, of particular use in systems where guided HFR (High Frequency Reconstruction) is employed. As a special case, the application of this stereo-coding scheme in scalable HFR-based codecs is described.

Abstract: An L-channel signal and an R-channel signal are input into a subtractor in which the difference between the L-channel signal and the R-channel signal is determined. The resulting signal is input into first and second multipliers. In the first multiplier, a mixed output signal is generated by using the channel mixture ratio calculated in a previous frame. In the second multiplier, a mixed output signal is generated by using the channel mixture ratio calculated in a current frame. The mixed output signals from the first and second multipliers respectively undergo spectrum transform in first and second spectrum transformers, and quantized in first and second quantizers. A comparator compares a quantization error in the first quantizer with that in the second quantizer. If the quantization error in the second quantizer is greater than that in the first quantizer by a predetermined factor, the comparator controls a switch to select the output of the first quantizer.

Abstract: A method and system accept left and right channel inputs of audio data and provide output audio data for at least three channels in a fashion that may effect a facsimile of the original soundstage on which the left and right inputs were recorded.

Abstract: A sensor includes one or more sensor transducers coupled with a signal conditioning IC incorporating signal conditioning circuitry and memory devoted to storing end-user downloadable coefficients. In a preferred embodiment, the IC is an ASIC and the end-user downloadable coefficients are pre-selected by the end-user based on its needs, and the coefficients are pre-stored in the ASIC when the sensor is calibrated. This results in a more cost-effective and space-efficient sensor device with improved functionality over that available in the prior art.

Abstract: A signal conditioning integrated circuit includes both signal conditioning circuitry and memory devoted to storing end-user downloadable coefficients. In a preferred embodiment, the integrated circuit is an Application Specific Integrated Circuit (ASIC), and the end-user downloadable coefficients, based upon a mathematical equation pre-selected by the end-user, are pre-stored in the ASIC when a sensor device with which the ASIC is associated is calibrated. This results in a customized and more cost-effective and space-efficient signal-conditioning apparatus with improved functionality over that available in the prior art.

Abstract: Part of the spectrum of two or more input signals is encoded using conventional coding techniques, while encoding the rest of the spectrum using binaural cue coding (BCC). In BCC coding, spectral components of the input signals are downmixed and BCC parameters (e.g., inter-channel level and/or time differences) are generated. In a stereo implementation, after converting the left and right channels to the frequency domain, pairs of left- and right-channel spectral components are downmixed to mono. The mono components are then converted back to the time domain, along with those left- and right-channel spectral components that were not downmixed, to form hybrid stereo signals, which can then be encoded using conventional coding techniques. For playback, the encoded bitstream is decoded using conventional decoding techniques. BCC synthesis techniques may then apply the BCC parameters to synthesize an auditory scene based on the mono components as well as the unmixed stereo components.

Abstract: A method derives at least three audio signals, each associated with a direction, from two input audio signals. In response to the two input signals, a passive matrix generates a plurality of passive matrix audio signals, including two pairs of passive matrix audio signals, a first pair of passive amtrix audio signals represent directions lying on a first axis and a second pair of passive matrix audio signals represent direction lying on a second axis, the first and second aces being substantially at ninety degrees to ach other. The pairs of passive matrix audio signals are processed to derive a plurality of matrix coefficients therefrom, The processing includes deriving a pair of intermediate signals and urging each pair of intermediate signals toward equality in response to a respective error signal. At least three output signals are produced by matrix multiplying the two input signals by the matrix coefficients.

Abstract: A system and method for collecting audio data for use in establishing an audio recording such as a surround sound recording are described. One exemplary method includes recording audio data at a plurality of mobile terminals located in a plurality of location zones in a given location, sending the recorded audio data from the plurality of mobile terminals to a central network server, and the central network server recording the received audio data into a plurality of audio tracks that may be used in establishing an audio recording.

Abstract: Surround audio decoding for selectively generating an audio signal from a multi-channel signal. In the surround audio decoding, a down-mixed signal, e.g., as down-mixed by an encoding terminal, is selectively up-mixed to a stereo signal or a multi-channel signal, by generating spatial information for generating the stereo signal, using spatial information for up-mixing the down-mixed signal to the multi-channel signal.

Abstract: An audio processing unit that provides a high-sound-quality reproduction even when a band property of an input encoded signal drops at or below a Nyquist frequency. The audio processing unit has a band-variable bandpass filter 1500; a band determining section 1700 for determining a passband of the band-variable bandpass filter with respect to a band extension component from an extended band generating section 1300 by using decoding information from a decode-processing section 1100 as a band determination information; and a bandpass filter controller 1600 for controlling the passband of the band-variable bandpass filter in accordance with an indication from the band determining section.

Abstract: This invention addresses sound recording and mixing methods for 3-D audio rendering of multiple sound sources over headphones or loudspeaker playback systems. Economical techniques are provided, whereby directional panning and mixing of sounds are performed in a multi-channel encoding format which preserves interaural time difference information and does not contain head-related spectral information. Decoders are provided for converting the multi-channel encoded signal into signals for playback over headphones or various loudspeaker arrangements. These decoders ensure faithful reproduction of directional auditory information at the eardrums of the listener and can be adapted to the number and geometrical layout of the loudspeakers and the individual characteristics of the listener. A particular multi-channel encoding format is disclosed, which, in addition to the above advantages, is associated with a practical microphone technique for producing 3-D audio recordings compliant with the decoders described.

Abstract: A video camera capable of recording and/or reproducing multiple channel audio signals and video signals simultaneously in a single recording medium in such a manner as to reproduce omnidirectional ambient sounds as surround sound. The video camera includes a number of microphones and records first multiple audio signals obtained by processing the audio signals outputted from the microphones to generate the multiple channel audio signals and also records a second audio signal obtained by processing all of the audio signals outputted from the microphones in the recording medium simultaneously with the video signals, so that the first multiple audio signals and the second audio signal can be reproduced independently of each other.

Abstract: Parametric stereo coders use perceptually relevant parameters of the input signal to describe spatial properties. One of these parameters is the phase difference between the input signals (ITD or IPD). This time difference only determines the relative time difference between the input signals, without any information about how these time differences should be divided over the output signals in the decoder. An additional parameter is included in the encoded signal that describes how the ITD or IPD should be distributed between the output channels. To this goal the delay between a computed monaural signal and one of the input signals is used.

Abstract: A method and circuit for deriving a set of multichannel audio signals from a conventional monaural or stereo audio signal uses an auxiliary multichannel spectral mapping data stream. Audio can be played back in stereo and multichannel formats from a conventional stereo signal on compact discs, FM radio, or other stereo or monaural delivery systems. The invention reduces the data rate needed for the transmission of multichannel digital audio.

Abstract: A record carrier carries a stereo signal and a data signal which includes a first data portion and a remaining data portion. A first channel on the record carrier includes a composite signal which is obtained by combining the stereo signal and the first data portion. A second channel on the record carrier includes the remaining data portion of the data signal. The stereo signal and the data signal are recorded to and reproduced from the record carrier using a recording apparatus and a reproducing apparatus.

Abstract: In an audio signal encoding apparatus, a first audio signal and a second audio signal are added into an addition-result signal. The first audio signal is subtracted from the second audio signal to generate a subtraction-result signal. A first difference signal is generated which represents a difference in the addition-result signal. A second difference signal is generated which represents a difference in the subtraction-result signal. A plurality of first predictors have different prediction characteristics respectively, and are responsive to the first difference signal for generating first different prediction signals for the first difference signal, respectively. A plurality of first subtracters operate for generating first prediction-error signals representing differences between the first difference signal and the first different prediction signals, respectively. A first minimum prediction-error signal representative of a smallest difference is selected from among the first prediction-error signals.

Abstract: Essentially all of the processing parameters which control processing of a source audio signal to produce an encoded audio signal are stored in an audio processing profile. Multiple audio processing profiles are stored in a processing profile database such that specific combinations of processing parameters can be retrieved and used at a later time. Audio processing profiles are organized according to specific delivery bandwidths such that a sound engineer can quickly and efficiently encode audio signals for each of a number of distinct delivery media. Synchronized A/B switching during playback of various encoded audio signals allows the sound engineer to detect nuances in the sound characteristics of the various encoded audio signals.

Abstract: The present invention relates to an encoding device for saving the number of bits of codes. In step S11, the differential value between a normalization coefficient Bi to be encoded and a normalization coefficient Bi-1 for an encoding unit Ai-1 in a band adjacent to the lower side of an encoding unit Ai corresponding to the normalization coefficient Bi is computed. In step S12, reference is made to a table in which a differential value having a high frequency of occurrence is associated with a code having a small number of bits, and a code corresponding to the computed differential value is read. In step S13, it is determined whether or not all normalization coefficients B have been encoded. If it is determined that all normalization coefficients B have been encoded, in step S14, the code read in step S12 is output. The present invention is applicable to an audio recorder.

Abstract: The degree of correlation between two audio signals is determined and the channels are normalized according to first and second normalization modes responsive to correlation and uncorrelation respectively.

Abstract: A sound signal encoding apparatus for encoding two different sound signals, comprising: compression level calculating means for calculating a compression level for each of the sound signal sections; compression level judging means for judging whether or not the calculated compression level for each of the sound signal sections exceeds a predetermined threshold compression value; frequency components encoding means for encoding the quantized frequency components for each of the sound signal sections to a multiplexed bit stream with a predetermined bit rate under two different states consisting of the first state in which the frequency components for each of the sound signal sections are compressed by the first frequency components compressing means when the compression level judging means is operative to judge that the compression level for each of the sound signal sections exceeds the predetermined threshold compression value and the second state in which the frequency components for each of the sound signal

Abstract: 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.

Abstract: A surround sound field reproduction system and method is disclosed by which, in such a case that a listener changes its posture so that a portion of a video image to which attention should be paid may be observed in front of the listener, a sound image can be localized accordingly in a direction in which the video image is observed in front of the listener. Specifically, the surround sound field reproduction apparatus includes a plurality of horizontal panners for receiving a plurality of sound signals from microphones disposed horizontally and cooperating with each other to control the horizontal localization, and a vertical panner for receiving sound signals from microphones disposed vertically to control the vertical localization. The horizontal panners level-distribute the received sound signals to a plurality of output channels provided corresponding to surround reproduction output channels.

Abstract: A system and method of producing an output sound field that is representative of an input sound field compatible with both existing prior art sound reproduction systems, for example ITU 5.1/6.1, and with a three-dimensional reproduction system unique to this disclosure. One embodiment of the disclosed system is comprised of a microphone array, an encoder, a decoder, and a plurality of speakers, some of which may not be located in the plane of the listener. A further embodiment discloses matrices to encode and decode the signals representative of the input and output sound fields respectively.

Abstract: An apparatus and method for assigning each conferee to a conference a three-dimensional position with respect to a central listening position and the other conferees. Each conferees audio stream is encoded with the assigned three-dimensional position to produce an encoded audio stream corresponding to each conferee. For each conferee, the encoded audio streams of the other conferees are mixed to produce a mixed audio stream wherein the conferee listens to the conference from the central listening position.

Abstract: A sound signal encoder for high efficiency encoding of sound signals from a plurality of channels is provided which includes a to-be-correlated object setter (52), to-be-correlated object selector (56) and a variable-length encoder (58). The to-be-correlated object setter (52) sets, on the basis of left-channel frequency information held in a left-channel frequency information holder (50) and right-channel frequency information held in a right-channel frequency information holder (51), index [i] indicating which ones of sine waves on the left channel are to be correlated with, namely, are to be subtracted from, sine waves on the right channel. The to-be-correlated object selector (56) selects a default value read from a storage unit (55) or index [i]-th amplitude information read from a left-channel amplitude information holder (53) as an object to be subtracted from the i-th amplitude information on the right channel according to the index [i].

Abstract: Three surround sound channels are provided within the current formats of the Dolby Digital, Sony SDDS and DTS digital soundtrack systems in a manner that provides compatibility with conventional two surround channel playback in standard 5.1 channel and 7.1 channel systems while allowing the soundtrack preparer to send the same signal to all surround sound channels and preserving the ability to pan among the three matrix decoded surround sound channels in an arrangement that employs an active matrix decoder to provide the three surround sound channels.

Abstract: In an audio signal encoding apparatus, a first audio signal and a second audio signal are added into an addition-result signal. The first audio signal is subtracted from the second audio signal to generate a subtraction-result signal. A first difference signal is generated which represents a difference in the addition-result signal. A second difference signal is generated which represents a difference in the subtraction-result signal. A plurality of first predictors have different prediction characteristics respectively, and are responsive to the first difference signal for generating first different prediction signals for the first difference signal, respectively. A plurality of first subtracters operate for generating first prediction-error signals representing differences between the first difference signal and the first different prediction signals, respectively. A first minimum prediction-error signal representative of a smallest difference is selected from among the first prediction-error signals.

Abstract: In an audio signal encoding apparatus, a first audio signal and a second audio signal are added into an addition-result signal. The first audio signal is subtracted from the second audio signal to generate a subtraction-result signal. A first difference signal is generated which represents a difference in the addition-result signal. A second difference signal is generated which represents a difference in the subtraction-result signal. A plurality of first predictors have different prediction characteristics respectively, and are responsive to the first difference signal for generating first different prediction signals for the first difference signal, respectively. A plurality of first subtracters operate for generating first prediction-error signals representing differences between the first difference signal and the first different prediction signals, respectively. A first minimum prediction-error signal representative of a smallest difference is selected from among the first prediction-error signals.

Abstract: In an audio signal encoding apparatus, a first audio signal and a second audio signal are added into an addition-result signal. The first audio signal is subtracted from the second audio signal to generate a subtraction-result signal. A first difference signal is generated which represents a difference in the addition-result signal. A second difference signal is generated which represents a difference in the subtraction-result signal. A plurality of first predictors have different prediction characteristics respectively, and are responsive to the first difference signal for generating first different prediction signals for the first difference signal, respectively. A plurality of first subtracters operate for generating first prediction-error signals representing differences between the first difference signal and the first different prediction signals, respectively. A first minimum prediction-error signal representative of a smallest difference is selected from among the first prediction-error signals.

Abstract: In an audio signal encoding apparatus, a first audio signal and a second audio signal are added into an addition-result signal. The first audio signal is subtracted from the second audio signal to generate a subtraction-result signal. A first difference signal is generated which represents a difference in the addition-result signal. A second difference signal is generated which represents a difference in the subtraction-result signal. A plurality of first predictors have different prediction characteristics respectively, and are responsive to the first difference signal for generating first different prediction signals for the first difference signal, respectively. A plurality of first subtracters operate for generating first prediction-error signals representing differences between the first difference signal and the first different prediction signals, respectively. A first minimum prediction-error signal representative of a smallest difference is selected from among the first prediction-error signals.