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: Techniques for processing input signals in accordance with an output configuration are disclosed. According to one aspect of the techniques, a module implemented in a sound reproducing device is configured to determine whether a sound in the input audio source can be reproduced properly through speakers of an output configuration of the device by examining the sound effects in the input audio sound and the output channel in the output configuration, producing synthetically a sound effect if the number of sound channels is greater than the number of sound effects, or reducing synthetically a sound effect if the number of sound channels is less than the number of sound effects.

Abstract: A binaural cue coding scheme in which cue codes are derived from the transmitted audio signal. In one embodiment, an encoder downmixes C input channels to generate E transmitted channels, where C>E>1. A decoder derives cue codes from the transmitted channels and uses those cue codes to synthesize playback channels. For example, in one 5-to-2 BCC embodiment, the encoder downmixes a 5-channel surround signal to generate left and right channels of a stereo signal. The decoder derives stereo cues from the transmitted stereo signal, maps those stereo cues to surround cues, and applies the surround cues to the transmitted stereo channels to generate playback channels of a 5-channel synthesized surround signal.

Abstract: A novel system and method for correcting the residual phase offset between a recovered pilot signal and the received stereo signal. The invention uses a Costas loop as an auxiliary loop in addition to the pilot recovery phase locked loop (PLL) to lock onto the stereo component itself. This auxiliary loop functions to generate a pilot to stereo component phase correction signal that is added to the stereo carrier phase The resultant phase is used to generate the recovered pilot carrier used to demodulate the stereo MPX signal. The Costas loop is activated together with the main pilot recovery PLL that locks onto the pilot tone in the demodulated MPX signal. The auxiliary Costas loop is operative to track and determine a residual phase error of up to several degrees.

Abstract: The invention relates to methods and units supporting a multichannel audio extension. In order to allow an efficient extension requiring a low computational complexity, it is proposed that at an encoding end, at least state information is provided as side information for a provided mono audio signal (M) generated out of a multichannel audio signal. The state information indicates for each of a plurality of frequency bands how a predetermined or equally provided gain value is to be applied in the frequency domain to the mono audio signal (M) for obtaining first and a second channel signals (L,R) of a reconstructed multichannel audio signal.

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 method for processing an audio signal, comprising: receiving a downmix signal, an object information, and a mix information; generating a downmix processing information using the object information and the mix information; processing the downmix signal using the downmix processing information; and, generating a multi-channel information using the object information and the mix information, wherein the number of channel of the downmix signal is equal to the number of channel of the processed downmix signal is disclosed.

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 method for processing an audio signal, comprising: receiving a downmix signal in time domain; if the downmix signal corresponds to a mono signal, bypassing the downmix signal; if the number of channel of the downmix signal corresponds to at least two, decomposing the downmix signal into a subband signal, and processing the subband signal using a downmix processing information, wherein the downmix processing information is estimated based on an object information and a mix information is disclosed.

Abstract: A method for processing an audio signal, comprising: receiving a downmix signal and a downmix processing information; and, processing the downmix signal using a downmix processing information, comprising: de-correlating the downmix signal; and, mixing the downmix signal and the de-correlated signal in order to output the processed downmix signal, wherein the downmix processing information is estimated based on an object information and a mix information is disclosed.

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 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 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 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, buffers in both the audio controller and the main memory may be dynamically adjusted during playback of audio data and/or for different applications.

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 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 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 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: An audio signal having at least two channels can be efficiently down-mixed into a downmixe signal and a residual signal, when the down-mixing rule used depends on a spatial parameter that is derived from the audio signal and that is post-processed by a limiter to apply a certain limit to the derived spatial parameter with the aim of avoiding instabilities during the up-mixing or down-mixing process. By having a down-mixing rule that dynamically depends on parameters describing an interrelation between the audio channels, one can assure that the energy within the down-mixed residual signal is as minimal as possible, which is advantageous in the view of coding efficiency. By post processing the spatial parameter with a limiter prior to using it in the down-mixing, one can avoid instabilities in the down- or up-mixing, which otherwise could result in a disturbance of the spatial perception of the encoded or decoded audio signal.

Abstract: An audio decoding method includes: acquiring, from encoded audio data, a reception audio signal and first auxiliary decoded audio information; calculating coefficient information from the first auxiliary decoded audio information; generating a decoded output audio signal based on the coefficient information and the reception audio signal; decoding to result in a decoded audio signal based on the first auxiliary decoded audio signal and the reception audio signal; calculating, from the decoded audio signal, second auxiliary decoded audio information corresponding to the first auxiliary decoded audio information; detecting a distortion caused in a decoding operation of the decoded audio signal by comparing the second auxiliary decoded audio information with the first auxiliary decoded audio information; correcting the coefficient information in response to the detected distortion; and supplying the corrected coefficient information as the coefficient information when generating the decoded output audio signal.

Abstract: At an audio encoder, cue codes are generated for one or more audio channels, wherein an envelope cue code is generated by characterizing a temporal envelope in an audio channel. At an audio decoder, E transmitted audio channel(s) are decoded to generate C playback audio channels, where C>E?1. Received cue codes include an envelope cue code corresponding to a characterized temporal envelope of an audio channel corresponding to the transmitted channel(s). One or more transmitted channel(s) are upmixed to generate one or more upmixed channels. One or more playback channels are synthesized by applying the cue codes to the one or more upmixed channels, wherein the envelope cue code is applied to an upmixed channel or a synthesized signal to adjust a temporal envelope of the synthesized signal based on the characterized temporal envelope such that the adjusted temporal envelope substantially matches the characterized temporal envelope.

Abstract: A method for processing an audio signal, comprising: receiving a downmix signal, a first multi-channel information, and an object information; processing the downmix signal using the object information and a mix information; and, transmitting one of the first multi-channel information and a second multi-channel information according to the mix information, wherein the second channel information is generated using the object information and the mix information is disclosed.

Abstract: A decoded sound analysis unit (104) calculates, regarding the frequency-region stereo signals L(b) and R(b) decoded by the PS decoding unit (103), a second degree of similarity (109) and a second intensity difference (110) from the decoded sound signals. A spectrum correction unit (105) detects a distortion added by the parametric-stereo conversion by comparing the second degree of similarity (109) and the second intensity difference (110) calculated at the decoding side with the first degree of similarity (107) and the first intensity difference (108) calculated and transmitted from the encoding side, and corrects the spectrum of the frequency-region stereo decoded signals L(b) and R(b).

Abstract: An acoustic signal encoding device for down-mixing at different ratios to encode a multichannel signal with a small number of channels, and an acoustic signal decoding device for decoding the signal encoded by the acoustic signal encoding device. In these devices, weighting means (103) in the acoustic signal encoding device (100) weights input signals of two channels individually according to a down-mixing coefficient thereby to calculate the level difference of the signals of two channels weighted by a level difference calculation unit (104). A separating unit (202) in the acoustic signal decoding device (200) separates the down-mixed signals into signals of two channels with the level difference information weighted.

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: A method for processing an audio signal, comprising the steps of extracting an ancillary signal for generating the audio signal and an extension signal included in the ancillary signal from a received bit stream, checking a level of the bit stream, selectively decoding the extension signal according to the level of the bit stream, and generating the audio signal using the ancillary signal. Accordingly, in case of processing the audio signal by the present invention, it is able to reduce a corresponding load of operation to enable efficient processing and enhance a sound quality.

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: The method and system of present invention sequences audio post-processing algorithms to simulate live or theater sound. An audio signal is selectively post-processed according to equipment availability and listener preferences. Downmixing or Prologic algorithms are applied to a signal arriving at sound system. A listener inputs their speaker configuration to a player console. Desired post-processing effects are likewise indicated to the console. For instance, if surround sound equipment is both available and selected, then surround portions of the audio signal are parsed to surround speakers. Bass management techniques then transfer low frequency channels of the signal to compatible speakers. VES or DCS algorithms further manipulate the surround portion of the signal to create an illusion of immersion, and a center channel equalizer balances the signal playback. Alternatively, the post-processed signal is transmitted to a headphone set.

Abstract: A method for processing sound signal including an L side enhanced difference signal comprising a sum of a signal being removed low frequency component from L?R signal and an L+R signal, an R side enhanced difference signal comprising a sum of a signal being removed low frequency component from R?L signal and an R+L signal, an L side output signal comprising a subtraction of signals, in which said R side enhanced difference signal being delayed to remove low and high ranges and amplified by a predetermined amplifying rate for a purpose of cross cancellation talk is subtracted from said L side enhanced difference signal and an R side output signal comprising a subtraction of signals, in which said L side enhanced difference signal being delayed to remove low and high ranges and amplified by a predetermined amplifying rate for a purpose of cross cancellation talk is subtracted from said R side enhanced difference signal.

Abstract: Using an M:N variable matrix, M audio input signals, each associated with a direction, are translated to N audio output signals, each associated with a direction, wherein N is larger than M, M is two or more and N is a positive integer equal to three or more. The variable matrix is controlled in response to measures of: (1) the relative levels of the input signals, and (2) the cross-correlation of the input signals so that a soundfield generated by the output signals has a compact sound image in the nominal ongoing primary direction of the input signals when the input signals are highly correlated, the image spreading from compact to broad as the correlation decreases and progressively splitting into multiple compact sound images, each in a direction associated with an input signal, as the correlation continues to decrease to highly uncorrelated.

Abstract: A method of encoding input signals (l, r) to generate encoded data (100) is provided. The method involves processing the input signals (l, r) to determine first parameters (?1, ?2) describing relative phase difference and temporal difference between the signals (l, r), and applying these first parameters (?1, ?2) to process the input signals to generate intermediate signals. The method involves processing the intermediate signals to determine second parameters (?; IID, ?) describing angular rotation of the first intermediate signals to generate a dominant signal (m) and a residual signal (s), the dominant signal (m) having a magnitude or energy greater than that of the residual signal (s). These second parameters are applicable to process the intermediate signals to generate the dominant (m) and residual (s) signals.

Abstract: A method is shown for supporting a multichannel audio extension at an encoding end of a multichannel audio coding system. In order to improve the audio quality over a large frequency range, the method comprises transforming each channel of a multichannel audio signal into the frequency domain and dividing a bandwidth of the frequency domain signals into a first region of lower frequencies and at least one further region of higher frequencies. Then, the frequency domain signals are encoded in each of the frequency regions with another type of coding to obtain parametric multichannel extension information for the respective frequency region. The invention relates equally to a method for supporting in a corresponding manner a multichannel audio extension at a decoding end. Also shown are a corresponding encoder, a corresponding decoder, and corresponding devices, systems and software program products.

Abstract: A coded signal conveys encoded audio information and metadata that may be used to control the loudness and dynamic range of the audio information during its playback. If the values for these metadata parameters are set incorrectly, annoying fluctuations in loudness during playback can result. The present invention overcomes this problem by detecting incorrect metadata parameter values in the signal and replacing the incorrect values with corrected values.

Abstract: An audio encoder, generating a stereo signal based on a multi-channel signal, includes a downmix unit for downmixing a multi-channel signal exceeding two channels to a two-channel stereo signal, a first coding unit for generating a first coded signal by coding the downmixed stereo signal, a second coding unit for generating a second coded signal by coding information to restore the downmixed stereo signal to a multi-channel signal, a code size calculating unit for calculating a code size of the second coded signal, and a first multiplexing unit for multiplexing the calculated code size in either the first coded signal or the second coded signal. Accordingly a decoder is able to easily extract a coded signal of the multi-channel signal based on the code size, and the decoder reproducing only the downmixed signal can be configured inexpensively.

Abstract: There is described a multi-channel encoder (10; 600) for processing input signals conveyed in N input channels to generate corresponding output signals conveyed in M output channels together with complementary parametric data; M and N are integers wherein N>M. The encoder (10; 600) includes a down-mixer for down-mixing the input signals to generate the corresponding output signals, the encoder also comprising an analyser for processing the input signals to generate the parameter data, said parametric data describing mutual differences between the N channels of input signal to allow for regenerating during decoding one or more of the N channels of input signals from the M channels of output signal. Such an encoder (10; 600) is capable of providing highly efficient data encoding and also of being backwards compatibility with relatively simpler decoders having fewer than N decoding output channels. The invention also concerns decoders (800) compatible with such a multi-channel encoder (10; 600).

Abstract: An object-based 3-D audio system. An audio input unit receives object-based sound sources. An audio editing/producing unit converts the sound sources into 3-D audio scene information. An audio encoding unit encodes 3-D information and object signals of the 3-D audio scene to transmit them through a medium. An audio decoding unit receives the encoded data through the medium, and decodes the same. An audio scene-synthesizing unit selectively synthesizes the object signals and 3-D information into a 3-D audio scene. A user control unit outputs a control signal according to the user's selection so as to selectively synthesize the audio scene by the audio scene synthesizing unit. An audio reproducing unit reproduces the audio scene synthesized by the audio scene-synthesizing unit.

Abstract: In an audio-information encoding apparatus, in order to encode an audio signal containing a white-noise component, an index iL indicating the energy level of the white-noise component and an index iR designating the start index of a random-number table are introduced into a code train. In an audio-information decoding apparatus (20), a white-noise generating unit (25) uses the indices iL and iR contained in the code train, thereby generating a white-noise signal Sw(t) on the time axis, which has the same level as the white noise, and an adder (26) adds the white-noise signal to an audio signal Sf(t) decoded on the time axis, outputting as an output audio signal So(t).

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 method and circuit for performing CRC calculations permits variable width data input. Preferably, multiple CRC calculations are performed in parallel, each CRC calculation involving a different number of data bits from the data word and terminating within one clock cycle. The CRC polynomial is preferably incorporated into the hardware for each CRC calculation.

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 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 method of audio matrix decoding in which a moving sound image is restored includes decoding multichannel signals from stereo signals, extracting strengths and positions of virtual sound sources existing between channels based on power vectors of the decoded multichannel signals, comparing the strengths and positions of the extracted previous and current virtual sound sources to predict position movement and the strengths of the virtual sound sources, and redistributing powers to positions of channel speakers in a multichannel arrangement based on the predicted position of a sound image.

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: The beginning detection, accommodation and frequency bias properties of the human hearing mechanism have been modeled to create systems that can detect directional transients (“sound events”) in a sound field and localize them. These systems break down a sound field into sound events and non-sound events and separately localize the sound events and non-sound events. Sound events are generally identified according to the frequency bias and beginning detection properties. Once detected, the sound events are generally localized according to differential steering angles (steering angles to which the steady-state signals have been accommodated) or ordinary steering angles, both of which reflect the direction of a sound event indicated during the rise-time of the sound event. When no sound events are detected, non-sound events are localized according to a steering angle that does not reflect rapid motion.

Abstract: An audio system for processing two channels of audio input to provide more than two output channels. The input may be conventional stereo material or compressed audio signal data. The audio processing includes separating the input signals into frequency bands and processing the frequency bands according to processes which may differ from band to band. The audio processing includes no processing of L?R signals.

Abstract: For achieving an audio reproduction with high sound quality, in a multi-channel A/V amplifier, front speakers are driven with a parallel-drive bi-amplifier arrangement upon stereo reproduction. In the case of the multi-channel reproduction mode, the switching circuit allows an output signal of each channel of the decoder to be sent to speakers via amplifiers for each channel in one-to-one correspondence. On the other hand, in the case of the 2-channel stereo reproduction mode, the switching circuit allows at least two amplifiers among the plurality of amplifiers to be connected in parallel between the output signal for each of the channels L and R of the decoder and the speakers for each of the channels L and R, and also allows the timing of the output signals of the respective amplifiers to be varied.

Abstract: A Japanese BTSC audio broadcast signal has three modes of transmission: mono, stereo, or dual mono. The control channel centering at 3.5 fH is one of the three channels in the J-BTSC signal, and contains information to indicate to the decoder which one of the three modes the audio transmission is in. The present invention uses a bandpass filter directly in the AM band, followed by envelope filtering and a decision circuit. Therefore, the need for AM demodulation and AM carrier detection is eliminated.

Abstract: A sound processing system reduces speaker distortion at high volume levels by attenuating the filter gain and/or tone of audio and mixed output signals. The sound processing system has one or more filters to attenuate the filter gain and tone in response to the high volume levels. The sound processing system also can attenuate the filter gain and tone in response to a sound pressure level, which may be provided by a microphone.