Abstract: Methods and apparatus to extract data encoded in media content are disclosed. An example method includes receiving a media content signal, sampling the media content signal to generate digital samples, determining a frequency domain representation of the digital samples, determining a first rank of a first frequency in the frequency domain representation, determining a second rank of a second frequency in the frequency domain representation, combining the first rank and the second rank with a set of ranks to create a combined set of ranks, comparing the combined set of ranks to a set of reference sequences, determining a data represented by the combined set of ranks based on the comparison, and storing the data in a tangible memory.

Abstract: An audio synthesizing apparatus receives an encoded signal comprising a downmix signal and parametric extension data for expanding the downmix signal to a multi-sound source signal. A decomposition processor (205) performs a signal decomposition of the downmix signal to generate at least a first signal component and a second signal component, where the second signal component is at least partially decorrelated with the first signal component. A position processor (207) determines a first spatial position indication for the first signal component in response to the parametric extension data and a binaural processor (211) synthesizes the first signal component based on the first spatial position indication and the second signal component to originate from a different direction. The invention may provide improved spatial experience from e.g. headphones by using a direct synthesis of a main directional signal from the appropriate position rather than as a combination of signals from virtual loudspeaker positions.

Abstract: The present invention is based on the finding that a reconstructed output channel, reconstructed with a multi-channel reconstructor using at least one downmix channel derived by downmixing a plurality of original channels and using a parameter representation including additional information on a temporal fine structure of an original channel can be reconstructed efficiently with high quality, when a generator for generating a direct signal component and a diffuse signal component based on the downmix channel is used. The quality can be essentially enhanced, if only the direct signal component is modified such that the temporal fine structure of the reconstructed output channel is fitting a desired temporal fine structure, indicated by the additional information on the temporal fine structure transmitted.

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: Provided are a signal processing method capable of inserting additional information into a bitstream without departing from the AC-3 standard or the Enhanced AC-3 standard. The signal processing method includes: extracting additional information which is for performing at least one of a channel number expansion of an audio signal in an encoding apparatus and a three-dimensional (3D) reproduction of the audio signal in the encoding apparatus; encoding the audio signal according to a predetermined standard; formatting the encoded audio signal into a bitstream including bitstream information, at least one audio block and an auxiliary data field; and inserting the additional information into at least one of additional bitstream information included in the bitstream information, a skip field included in the audio block and auxiliary data bits included in the auxiliary data field.

Abstract: Provided are a method and apparatus for encoding and decoding a stereo signal or a multi-channel signal. According to the method and apparatus, a stereo signal or a multi-channel signal can be encoded and/or decoded by generating parameters based on a mono signal.

Abstract: An active audio matrix decoding method and apparatus to generate multi-channel audio signals from a stereo channel audio signal.

Abstract: A multi-channel output device for a multimedia device includes a plurality of reception ends, an amplifier having an input end and an output end, a plurality of sound output devices for outputting sound, a first switching unit coupled between the plurality of reception ends and the input end of the amplifier for coupling a reception end of the plurality of reception ends to the input end of the amplifier according to a control signal, a second switching unit coupled between the output end of the amplifier and the plurality of sound output devices for coupling the output end of the amplifier to a sound output device of the plurality of sound output devices according to the control signal, and a control unit coupled to the first switching unit and the second switching unit for outputting the control signal.

Abstract: The application relates to audio encoder and decoder systems. An embodiment of the encoder system comprises a downmix stage for generating a downmix signal and a residual signal based on a stereo signal. In addition, the encoder system comprises a parameter determining stage for determining parametric stereo parameters such as an inter-channel intensity difference and an inter-channel cross-correlation. Preferably, the parametric stereo parameters are time- and frequency-variant. Moreover, the encoder system comprises a transform stage. The transform stage generates a pseudo left/right stereo signal by performing a transform based on the downmix signal and the residual signal. The pseudo stereo signal is processed by a perceptual stereo encoder. For stereo encoding, left/right encoding or mid/side encoding is selectable. Preferably, the selection between left/right stereo encoding and mid/side stereo encoding is time- and frequency-variant.

Abstract: Methods of encoding and decoding a multi-channel audio signal and apparatuses for encoding and decoding a multi-channel audio signal are provided. The method of decoding a multi-channel audio signal includes an unpacking unit which extracts a quantized CLD between a pair of channels of a plurality of channels from a bitstream, and an inverse quantization unit which inverse-quantizes the quantized CLD using a quantization table that considers the location properties of the pair of channels. The methods of encoding and decoding a multi-channel audio signal and the apparatuses for encoding and decoding a multi-channel audio signal can enable an efficient encoding/decoding by reducing the number of quantization bits required.

Abstract: An apparatus for processing an audio signal and method thereof applied to an audio playback system are disclosed. The apparatus comprises a decoder, an error-correcting circuit and an audio correcting module. The method for processing audio signals in accordance with the present invention decodes the audio signal to generate a decoded signal by the decoder. Then, the error-correcting circuit performs an error-correcting algorithm in the decoded signal to generate an error indication signal and an output audio signal. And the audio correcting module corrects the output audio signal to generate a corrected audio signal when the error indication signal indicates that the output audio signal has error.

Abstract: An apparatus for upmixing a downmix audio signal describing one or more downmix audio channels into an upmixed audio signal describing a plurality of upmixed audio channels includes an upmixer configured to apply temporally variable upmixing parameters to upmix the downmix audio signal in order to obtain the upmixed audio signal. The apparatus also includes a parameter interpolator, wherein the parameter interpolator is configured to obtain one or more temporally interpolated upmix parameters to be used by the upmixer on the basis of a first complex-valued upmix parameter and a subsequent second complex-valued upmix parameter.

Abstract: Processing an audio signal associated with a sound recording made available to be rendered to an end user is disclosed. The audio signal is received. A high-order perceptual attribute of the audio signal as rendered is changed by modifying the audio signal. The modification may be based on real-time analysis of the audio signal.

Abstract: The invention proposes a method for embedding an ancillary data into a compressed audio signal. This is achieved by replacing Least Significant Bits (LSBs) in at least one frequency subband of the compressed audio signal by the ancillary data. When replacing LSB bits of compressed subband signals with the ancillary data, the subband signal is effectively modified, resulting in a different decoded output. The replaced LSB bits corresponding to the ancillary data are conveyed as part of the bitstream and can be easily extracted at the decoder. In such a way the decoder obtains the ancillary data that can be used for more advanced audio reproduction at the decoder. The compressed audio itself maintains a good audio quality despite the replacement of the LSB bits of the frequency subband, because the LSB bits do not contribute to the audible artefacts.

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: A method for generating a parametrically encoded audio signal, the method comprising: inputting a multi-channel audio signal comprising a plurality of audio channels; generating at least one combined signal of the plurality of audio channels; and generating one or more corresponding sets of side information including channel configuration information for controlling audio source locations in a synthesis of a binaural audio signal.

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 method and apparatus to encode audio with spatial information in a manner that does not depend on the exhibition setup, and to decode and play out optimally for any given exhibition setup, maximizing the sweet-spot area, and including setups with loudspeakers at different heights, and headphones. The part of the audio that requires very precise localization is encoded into a set of mono tracks with associated directional parameters, whereas the remaining audio is encoded into a set of Ambisonics tracks of a chosen order and mixture. Upon specification of a given exhibition system, the exhibition-independent format is decoded adapting to the specified system, by using different decoding methods for each assigned group.

Abstract: Provided is an audio data decoding system and method that may selectively transfer compressed audio data from a top system to at least one audio input buffer of a separately provided and/or separately power management controlled subsystem, restore the compressed audio data to digital pulse code modulation (PCM) data using an audio decoding unit of the subsystem, convert the digital PCM data to analog PCM data or an audio output signal, and output the converted audio output signal.

Abstract: A variable equalizer apparatus for forward and/or reverse equalizers in an amplifier. The system can include a structure to allow continuous contact of the signal flow as an equalizer is removed; jumpers with fixed resistors and associated capacitors and inductors to produce a variable range over many different values; and/or variable resistance potentiometers with fixed resistors and associated capacitors and inductors to produce a variable range over separate value equalizers.

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 system, method, and method of encoding/decoding a multi-channel audio signal, including a decoding level generation unit producing decoding-level information that helps a bitstream including a number of audio channel signals and space information to be decoded into a number of audio channel signals, wherein the space information includes information about magnitude differences and/or similarities between channels, and an audio decoder decoding the bitstream according to the decoding-level information. Accordingly, even a single input bitstream can be decoded into a suitable number of channels depending on the type of a speaker configuration used. Scalable channel decoding can be achieved by partially decoding an input bitstream. In the scalable channel decoding, a decoder may set decoding levels and outputs audio channel signals according to the decoding levels, thereby reducing decoding complexity.

Abstract: This disclosure describes decoding techniques for audio information encoded according to mid-side (MS) stereo encoding. The techniques involve a number of audio decoding tasks that are performed in a non-conventional order. By rearranging the order of the decoding tasks, various efficiencies can be achieved. In one embodiment, a method of decoding MS stereo audio information includes decoding a first channel of the audio information, computing an inverse modified discrete cosine transform (IMDCT) for the first channel prior to generating MS stereo information, decoding the second channel of the audio information, computing an IMDCT for the second channel prior to generating the MS stereo information, and generating the MS stereo information using the IMDCTs for the first and second channels.

Abstract: Parameters being a measure for a characteristic of a channel or of a pair of channels, wherein the parameter is a measure for a characteristic of the channel or of the pair of channels with respect to another channel of a multi-channel signal can be quantized more efficiently using a quantization rule that is generated based on a relation of an energy measure of the channel or the pair of channels and an energy measure of the multi-channel signal. With generation of the quantization rule taking into account a psycho acoustic approach, the size of an encoded representation of the multi-channel signal can be decreased by coarser quantization without significantly disturbing the perceptual quality of the multi-channel signal when reconstructed from the encoded representation.

Abstract: Provided is an encoding device (1) including: a pitch contour analysis unit (101) which detects information, a dynamic time-warping unit (102) which generates, based on the information, pitch change ratios (Tw_ratio in FIG. 18) within a range (86) including a range (86a) of the pitch change ratios corresponding to absolute pitch differences of 42 cents or larger; a first lossless coding unit (103) which codes the generated pitch parameters (102x); a time-warping unit (104) which shifts a pitch of a signal according to the information; and a second encoding unit which codes a signal (104x) obtained by the shifting.

Abstract: Disclosed is an audio signal decoding device and a method of balance adjustment that reduces a fluctuation of a decoded signal orientation and maintains a stereo perception. An interchannel correlation computation unit (224) computes a correlation between a left channel decoded stereo signal and a right channel decoded stereo signal, and if the interchannel correlation is low, a peak detection unit (225) uses a peak component of a decoded monaural signal of the current frame and a peak component of either a left or a right channel of the preceding frame to detect a peak component with a high temporal correlation. The peak detection unit (225) combines and outputs, from among the frequencies of the detected peak components, a peak frequency of a frame n?1 and a peak frequency of a frame n. A peak balance coefficient computation unit (226) computes, from the peak frequency of the frame n?1, a balance parameter that is used in converting a peak frequency component of the monaural signal to stereo.

Abstract: Biphase mark codes (BMC) are used in digital communications. Most BMC formats use preambles for rate determination and synchronization. A decoder compares the intervals of continuous high or continuous low voltages in a BMC stream to predetermined minimum and maximum values of half cell, full cell and one-and-a-half cell intervals for all supported sampling rates. If a pattern matching a preamble is found, the sampling rate is locked in and the decoder is synchronized to the BMC stream. Once locked, the decoder uses the predetermined minimum and maximum values at the locked rate to generate half cell, full cell and one-and-a-half cell indicators for a decoding state machine which decodes data in the BMC stream or decodes expected preambles.

Abstract: A system for generating stereo-channel audio signals with surround information is disclosed. The system includes a surround mapping unit configured to receive signals from a number of audio channels and generate a pair of stereo-channel audio signals based on the audio channels. The pair of stereo-channel audio signals includes binaural and spatial information. The system also includes a stereo-channel encoder configured to receive and encode the pair of stereo-channel audio signals from the surround mapping unit thereby generating a pair of encoded stereo-channel audio signals. The system further includes a stereo-channel decoder configured to receive and decode the pair of encoded stereo-channel audio signals thereby obtaining the pair of stereo-channel audio signals. The pair of stereo-channel audio signals are capable of being used to generate surround effect.

Abstract: A system and method is described herein in which an audio source wirelessly transmits audio content to a first audio sink over one wireless link and to a second audio sink over another wireless link. The two audio sinks also exchange forward error correction (FEC) streams over a wireless link between the two audio sinks, wherein the FEC streams are generated by FEC encoding the audio content received from the audio source. The audio sinks advantageously use the exchanged FEC information to synchronize the playback of the audio content as well as to improve the robustness of the wireless links with the audio source in a manner that does not consume additional bandwidth on those links.

Abstract: A method for coding a multi-channel audio signal representing a sound scene comprising a plurality of sound sources is provided. This method comprises decomposing the multi-channel signal into frequency bands and, per frequency band, obtaining directivity information per sound source of the sound scene, the information being representative of the spatial distribution of the sound source in the sound scene, of selecting a set of sound sources of the sound scene constituting principal sources, of matrixing the selected principal sources to obtain a sum signal with a reduced number of channels and, of coding the directivity information and of forming a binary stream comprising the coded directivity information, the binary stream being transmittable in parallel with the sum signal. A decoding method is also provided that is able to decode the sum signal and the directivity information to obtain a multi-channel signal, to an adapted coder and an adapted decoder.

Abstract: A method is provided for coding a multi-channel audio signal representing a sound scene comprising a plurality of sound sources. The method comprises decomposing the multi-channel signal into frequency bands and the following performed per frequency band: obtaining data representative of the direction of the sound sources of the sound scene, selecting a set of sound sources constituting principal sources, adapting the data representative of the direction of the selected principal sources, as a function of restitution characteristics of the multi-channel signal, determining a matrix for mixing the principal sources as a function of the adapted data, matrixing the principal sources by the matrix determined so as to obtain a sum signal with a reduced number of channels and coding the data representative of the direction of the sound sources and forming a binary stream comprising the coded data, the binary stream being transmittable in parallel with the sum signal.

Abstract: A coded signal conveys encoded audio information and metadata that may be used to control the loudness 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: 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 multi-channel decoder for generating a binaural signal from a downmix signal using upmix rule information on an energy-error introducing upmix rule for calculating a gain factor based on the upmix rule information and characteristics of head related transfer function based filters corresponding to upmix channels. The one or more gain factors are used by a filter processor for filtering the downmix signal so that an energy corrected binaural signal having a left binaural channel and a right binaural channel is obtained.

Abstract: A temporal processing apparatus includes: a splitter splitting an audio signal, included in the sub-band domain, into diffuse signals indicating reverberating components and direct signals indicating non-reverberating components; a downmix unit generating a downmix signal by downmixing the direct signals; BPFs respectively generating a bandpass downmix signal and bandpass diffuse signals; normalization processing units respectively generating a normalized downmix signal and normalized diffuse signals; a scale computation processing unit computing, on a predetermined time slot basis, a scale factor indicating the magnitude of energy of the normalized downmix signal with respect to energy of the normalized diffuse signals; a calculating unit generating scale diffuse signals; a HPF generating high-pass diffuse signals; an adding unit generating addition signals; and a synthesis filter bank performing synthesis filter processing on the addition signals and transforming the addition signals into the time domains.

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: 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: A parametric representation of a multi-channel audio signal having parameters suited to be used together with a monophonic downmix signal to calculate a reconstruction of the multi-channel audio signal can efficiently be derived in a stereo-backwards compatible way when a parameter combiner is used to generate the parametric representation by combining a one or more spatial parameters and a stereo parameter resulting in a parametric representation having a decoder usable stereo parameter and an information on the one or more spatial parameters that represents, together with the decoder usable stereo parameter, the one or more spatial parameters.

Abstract: An encoder for mixing a plurality of independent mono audio channels into a stereo recording and generating a restricted set of additional parameters used to master an audio track of a storage device is described. The plurality of independent mono audio channels are constructed such that the storage device can be played using an optical disk player so that in a first mode all of the plurality of independent mono audio channels are played as the stereo recording and in a second mode at least one of the plurality of independent mono audio channels can be unmixed and the stereo recording played with at least one mono audio channel removed. A corresponding decoder and an audio system comprising such encoder and decoder are also described.

Abstract: An encoding device (1) and method convert a set of signals (l, r) into a dominant signal (m) containing most signal energy, a residual signal (s) containing a remainder of the signal energy, and signal parameters (IID, ICC) associated with the conversion. The dominant signal (m) and selected parts of the residual signal (s) are encoded. Selecting parts of the residual signal involves a residual signal (s?) passing perceptually relevant parts of the residual signal (s), attenuating perceptually less relevant parts of the residual signal and suppressing least relevant parts of the residual signal. An associated decoding device (2) and method decode the encoded dominant signal and the encoded residual signal so as to produce a decoded dominant signal (m?u) and a decoded residual signal (s?mod) respectively. A synthetic residual signal (s?syn) is derived from the decoded dominant signal (m?u) and is attenuated so as to produce an attenuated synthetic residual signal (s?syn,mod).

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: An audio decoding method and apparatus and an audio encoding method and apparatus which can efficiently process object-based audio signals are provided. The audio decoding method includes receiving a downmix signal and object-based side information, the downmix signal comprising at least two downmix channel signals; extracting gain information from the object-based side information and generating modification information for modifying the downmix channel signals on a channel-by-channel basis based on the gain information; and modifying the downmix channel signals by applying the modification information to the downmix channel signals.

Abstract: An apparatus for determining a spatial output multi-channel audio signal based on an input audio signal and an input parameter. The apparatus includes a decomposer for decomposing the input audio signal based on the input parameter to obtain a first decomposed signal and a second decomposed signal different from each other. Furthermore, the apparatus includes a renderer for rendering the first decomposed signal to obtain a first rendered signal having a first semantic property and for rendering the second decomposed signal to obtain a second rendered signal having a second semantic property being different from the first semantic property. The apparatus comprises a processor for processing the first rendered signal and the second rendered signal to obtain the spatial output multi-channel audio signal.

Abstract: A coding apparatus which suppresses an extreme increase in a bit rate, includes: a downmixing and coding unit (301) that downmixes audio signals that have been provided, to reduce the number of channels to be fewer than the number of the provided audio signals, and to code the downmix signals; an object parameter extracting unit (304) that extracts parameters indicating correlation between the audio signals; and a multiplexing circuit (309) that multiplexes the extracted parameters with the generated downmix coded signals. The object parameter extracting unit (304) includes: an object classifying unit (305) that classifies each of the provided audio signals into a predetermined one of types based on audio characteristics; and an object parameter extracting circuit (308) that extracts parameters using a temporal granularity and a frequency granularity each of which is determined for a corresponding one of the types.

Abstract: A decoding apparatus (10) is disclosed which includes: a storing means (11) for storing encoded audio signals including multi-channel audio signals; a transforming means (40) for transforming the encoded audio signals to generate transform block-based audio signals in a time domain; a window processing means (41) for multiplying the transform block-based audio signals by a product of a mixture ratio of the audio signals and a first window function, the product being a second window function; a synthesizing means (43) for overlapping the multiplied transform block-based audio signals to synthesize audio signals of respective channels; and a mixing means (14) for mixing audio signals of the respective channels between the channels to generate a downmixed audio signal. Furthermore, an encoding apparatus is also disclosed which downmixes the multi-channel audio signals, encodes the downmixed audio signals, and generates the encoded, downmixed audio signals.

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: The envelope of a decorrelated signal derived from an original signal can be shaped without introducing additional distortion, when a spectral flattener is used to spectrally flatten the spectrum of the decorrelated signal and the original signal prior to using the flattened spectra for deriving a gain factor describing the energy distribution between the flattened spectra, and when the so derived gain factor is used by an envelope shaper to timely shape the envelope of the decorrelated signal.

Abstract: On an encoder-side, a multi-channel input signal is analyzed for obtaining smoothing control information, which is to be used by a decoder-side multi-channel synthesis for smoothing quantized transmitted parameters or values derived from the quantized transmitted parameters for providing an improved subjective audio quality in particular for slowly moving point sources and rapidly moving point sources having tonal material such as fast moving sinusoids.

Abstract: Generic and specific C-to-E binaural cue coding (BCC) schemes are described, including those in which one or more of the input channels are transmitted as unmodified channels that are not downmixed at the BCC encoder and not upmixed at the BCC decoder. The specific BCC schemes described include 5-to-2, 6-to-5, 7-to-5, 6.1-to-5.1, 7.1-to-5.1, and 6.2-to-5.1, where “.1” indicates a single low-frequency effects (LFE) channel and “.2” indicates two LFE channels.

Abstract: A motion picture soundtrack reproduction system has a center front soundtrack channel and a plurality of other soundtrack channels. A volume control adjusts the gain of all the channels. The volume control has a range of settings from a minimum to a maximum, the gain of the center front channel having substantially a first relationship to the volume control settings and the gain of the other channels having substantially a second relationship to the volume control settings, the relationships being such that for a range of volume control settings less than a first setting the gain of the center front channel remains substantially constant while the gain of the other channels decreases as the setting decreases or decreases more gradually than the gain of the other channels as the setting decreases.