Abstract: Methods and apparatuses for encoding and decoding a multi-channel audio signal are provided. In the encoding method, spatial information is calculated based on a multi-channel audio signal and a down-mix signal, and a compensation parameter that compensates for the down-mix signal is calculated based on the multi-channel audio signal and the down-mix signal. Thereafter, a bitstream is generated by encoding the spatial information, the compensation parameter, and the down-mix signal and combining the results of the encoding. Therefore, it is possible to prevent deterioration of the quality of sound regarding a multi-channel audio signal by compensating for the multi-channel audio signal using a compensation parameter that compensates for a down-mix signal.

Abstract: An audio encoder has a common preprocessing stage, an information sink based encoding branch such as spectral domain encoding branch, a information source based encoding branch such as an LPC-domain encoding branch and a switch for switching between these branches at inputs into these branches or outputs of these branches controlled by a decision stage. An audio decoder has a spectral domain decoding branch, an LPC-domain decoding branch, one or more switches for switching between the branches and a common post-processing stage for post-processing a time-domain audio signal for obtaining a post-processed audio signal.

Abstract: A method and apparatus which encode multi-channel audio signals and a method and apparatus which decode multi-channel audio signals. When encoding, a downmixed audio signal, first additional information for restoring multi-channel audio signals from the downmixed audio signal, and second additional information representing characteristics of a residual signal are multiplexed. When decoding, restored multi-channel audio signals having a predetermined phase difference are combined using the second additional information, and an audio signal of each channel is corrected, in order to improve quality of the restored audio signals.

Abstract: A method of synthesizing a first (L) and a second (R) output signal from an input signal (x). The method comprises: filtering (201) the input signal to generate a filtered signal (Hx); obtaining a correlation parameter (?) indicative of a desired correlation between the first and second output signals; obtaining a level parameter (c) indicative of a desired level difference between the first and second input signals; and transforming the input signal and the filtered signal by a matrixing operation (203) into the first and second output signals, where the matrixing operation depends on the correlation parameter and the level parameter.

Abstract: An embodiment of the present invention provides a method for generating a downmixed signal, including: performing a time-frequency transform on a received left sound channel signal and a received right sound channel signal to obtain a frequency domain signal, and dividing the frequency domain signal into several frequency bands; calculating a sound channel energy ratio and a sound channel phase difference of each frequency band; calculating a phase difference between the downmixed signal and a first sound channel signal in each frequency band according to the sound channel energy ratio and the sound channel phase difference; and calculating a frequency domain downmixed signal according to the left sound channel signal, the right sound channel signal, and the phase difference between the downmixed signal and the first sound channel signal in each frequency band. This method effectively improves quality of stereo encoding and decoding.

Abstract: The present technique relates to a decoding device, a decoding method, an encoding device, an encoding method, and a program which can obtain a high-quality realistic sound. The encoding device stores speaker arrangement information in a comment region in a PCE of an encoded bit stream and stores a synchronous word and identification information in the comment region such that other public comments and the speaker arrangement information stored in the comment region can be distinguished from each other. When an encoded bit stream is decoded, it is determined whether the speaker arrangement information is stored on the basis of the synchronous word and the identification information stored in the comment region. Audio data included in the encoded bit stream is output according to the arrangement of the speakers corresponding to the determination result. The present technique can be applied to an encoding device.

Abstract: The present technology relates to an encoding device and method, a decoding device and method, and a program that enable audio of a high audio quality to be obtained with a smaller code amount. The encoding device multiplexes low frequency encoding data obtained by encoding a low frequency component of an input signal and high frequency encoding data obtained by encoding data including an estimation coefficient to acquire a high frequency component of the input signal by estimation and outputs multiplexed data. When the input signal is encoded, a calculation unit calculates pseudo high frequency subband power to be an estimation value of power of the high frequency component from an estimation coefficient selected in a frame immediately before a frame of a processing target and the high frequency component of the input signal.

Abstract: A perceptually motivated, frequency domain active matrix decoder and decoding method which decodes N audio input signals to generate M audio output signals, where M is greater than N, including by generating M streams of output frequency components which determine the audio output signals, in response to N streams of input frequency components indicative of the audio input signals, determining power ratios from the input frequency components without use of feedback, including at least one power ratio for each critical frequency band in a set of critical frequency bands, and determining gain control values for each of the critical frequency bands from the power ratios including by shaping the power ratios in nonlinear fashion without use of feedback. An active matrix element is steered using the gain control values.

Abstract: A method of encoding stereo audio that minimizes a number of pieces of side information required for parametric-encoding and parametric-decoding of the stereo audio. The side information may include parameters about interchannel intensity difference (IID), interchannel correlation (IC), overall phase difference (OPD), and interchannel phase difference (IPD), which are required to restore the mono audio to the stereo audio.

Abstract: A digital audio interface may include two signal inputs to transmit audio data. A first signal line may carry digital serial audio data. The second signal line may carry a word clock signal to differentiate the serial audio data transmitted over the first signal line. In the case of stereo audio data, the word clock signal may correspond to a left-right clock signal and may differentiate audio data intended for a right channel from that intended for a left channel. The audio data may also be differentiated differently depending on the configuration, such as in the case that the transmitted audio data include audio for more than two channels. The word clock signal may be scaled to regenerate a bit clock signal used to encode the serial audio data over the first signal line. The encoding bit clock signal need not be transmitted.

Abstract: A system (200) for processing a sound signal (212) that allows dynamic customization of perceived spatial positions and sound qualities of sound components associated with the sound signal (212). The system provides apparatus for processing a sound signal (212) that includes an input to receive the sound signal (212), a sound unmixer (204) coupled to the input to receive the sound signal (212) and unmix at least one sound stream (216) from the sound signal (212) based on at least one unmixing instruction (214), and an output coupled to the sound unmixer (214) to output the at least one sound stream (216).

Abstract: A sound encoder for efficiently encoding stereophonic sound. A prediction parameter analyzer determines a delay difference D and an amplitude ratio g of a first-channel sound signal with respect to a second-channel sound signal as channel-to-channel prediction parameters from a first-channel decoded signal and a second-channel sound signal. A prediction parameter quantizer quantizes the prediction parameters, and a signal predictor predicts a second-channel signal using the first decoded signal and the quantization prediction parameters. The prediction parameter quantizer encodes and quantizes the prediction parameters (the delay difference D and the amplitude ratio g) using a relationship (correlation) between the delay difference D and the amplitude ratio g attributed to a spatial characteristic (e.g., distance) from a sound source of the signal to a receiving point.

Abstract: A panner is provided that incorporates a surround sound decoder. The panner takes as input the desired panning effect that a user requests, separates sounds using surround sound decoding, and places the separated sounds in the desired places in an output sound field.

Abstract: Provided are an encoding apparatus and a decoding apparatus of a multi-channel signal. The encoding apparatus of the multi-channel signal may process a phase parameter associated with phase information between a plurality of channels constituting the multi-channel signal, based on a characteristic of the multi-channel signal. The encoding apparatus may generate an encoded bitstream with respect to the multi-channel signal using the processed phase parameter and a mono signal extracted from the multi-channel signal.

Abstract: A method for processing an audio signal at an audio decoder, the method including receiving a downmix signal, a residual signal, and object information; extracting a background-object signal and a foreground-object signal from the downmix signal using the residual signal and object information; receiving mix information including gain information for the background-object signal; generating a downmix processing information based on the object information and the mix information; and generating an output signal including a modified background-object signal and a modified foreground-object signal. The modified background-object signal is obtained by modifying a gain of the background-object signal using the mix information. The modified foreground-object signal is obtained by modifying a gain of the foreground-object signal using the downmix processing information.

Abstract: A method for multi-channel processing in a multi-channel sound system, in which a channel or a channel mixture is first split into individual channels, the individual channels are limited by setting the values of the parameters channel fader, threshold, release, and output level and then encoding the individual channels. At least two channels are compressed and/or limited with a uniform output level value in method step, one channel is provided with a deviating output level value, which is set, depending on the audio material to be processed, and every further channel is compressed and/or limited in such a manner that it has an output level value that is at least one decibel less than the uniform output level value. The individual channels are combined into an encoded (coded) channel by setting a value of at least one of the parameters channel fader, threshold, release, and output level.

Abstract: A method of processing an audio signal, the method including receiving a downmix signal including at least one object signal, and object information extracted when the downmix signal is generated, the at least one object signal including a background object and at least one independent object; receiving mix information including mode selection information, the mix information for controlling the at least object signal; receiving enhanced object information corresponding to a residual signal generated when the at least one object signal is downmixed to the downmix signal; generating downmix processing information by using the object information and the mix information; generating an output signal by applying the downmix processed information to the downmix signal; and extracting the background object and the at least one independent object from the downmix signal, by using the enhanced object information.

Abstract: A method of encoding stereo audio that minimizes a number of pieces of side information required for parametric-encoding and parametric-decoding of the stereo audio. The side information may include parameters about interchannel intensity difference (IID), interchannel correlation (IC), overall phase difference (OPD), and interchannel phase difference (IPD), which are required to restore the mono audio to the stereo audio.

Abstract: A decoding method, medium, and device decoding an input signal, including compressed multi-channel signals as a mono or stereo signal, into 2-channel binaural signals. A full band channel level of each channel in the multi-channel system is calculated from channel level differences between the channels, and data of each channel included in the input signal is localized in directions corresponding to the channels based on the calculated full band channel levels of the channels. Accordingly, the input signal can be output as the 2-channel binaural signals by using simple operations without having to reconstruct multi-channel signals from the input signal in a quadrature mirror filter (QMF) domain.

Abstract: A decoding method of audio data is applied to an electronic device. The method includes: calculating difference values of the left and right channel audio signal values; determining time slots, wherein each of the first difference values exceeds a threshold value and a time length of the time slots exceeds a preset time; respectively multiplying the time slots with the left channel audio signal values and the right channel audio signal values to obtains and then making subtraction to obtain DM1˜n; finding a U shaped pattern or an inverse U shaped pattern which meets proportionality from the waveform of DM1˜n; and decoding the written symbol “0” or “1” according to the found U shaped pattern or inverse U shaped pattern.

Abstract: For flexibly signaling a synchronous mode or an asynchronous mode in the multi-channel parameter reconstruction, a parameter configuration cue is inserted in the data stream, which is used by a configurator on the side of a multi-channel decoder to configure a multi-channel reconstructor. If the parameter configuration cue has a first meaning, the configurator will look for further configuration information in its input data, while, when the parameter configuration cue has another meaning, the configurator performs a configuration setting of the multi-channel reconstructor based on information on a coding algorithm with which transmission channel data have been coded, so that it is ensured efficiently on the one hand and flexibly on the other hand that there will always be obtained a correct association between parameter data and decoded transmission channel data.

Abstract: An audio signal processor receives a plurality of encoded multi-channel audio signals. A multi-channel decoder (105) decodes a first encoded multi-channel signal to generate a first decoded multi-channel signal. A generator (109) generates an encoded further audio signal by selecting audio encoding data from at least a second encoded multi-channel audio signal such that a number of channels of the encoded further audio signal comprising audio encoding data from the second encoded multi-channel audio signal is less than a number of channels in the second encoded multi-channel signal. Thus, a channel reduction is performed in the encoded data domain. A further decoder (111) generates a further decoded signal by decoding the further encoded audio signal. A combiner (107) combines the first decoded multi-channel signal and the further decoded signal to generate a multi-channel output signal. An exciting user experience can be provided while maintaining low complexity and resource usage.

Abstract: A selected channel of a multi-channel signal which is represented by frames composed from sampling values having a high time resolution can be encoded with higher quality when a wave form parameter representation representing a wave form of an intermediate resolution representation of the selected channel is derived, the wave form parameter representation including a sequence of intermediate wave form parameters having a time resolution lower than the high time resolution of the sampling values and higher than a time resolution defined by a frame repetition rate. The wave form parameter representation with the intermediate resolution can be used to shape a reconstructed channel to retrieve a channel having a signal envelope close to that one of the selected original channel. The time scale on which the shaping is performed is shorter than the time scale of a framewise processing, thus enhancing the quality of the reconstructed channel.

Abstract: An object decoding unit receives a plurality of pieces of coded acoustic information coded on an object basis and decodes the received coded acoustic information on an object basis to generate object signals. A rendering unit positions, for each acoustic signal resulting from synthesizing the object signals, a virtual sound source of each of the object signals resulting from the object-based decoding, in a listening space, and priority information indicating a priority of the acoustic signal is multiplexed with the coded acoustic information. The rendering unit determines, with reference to the priority indicated in the priority information or the number of object signals included in the acoustic signal, a size of an area of the listening space which is allocated to the acoustic signal, and positions, within the area, the virtual sound source of each of the object signals included in the acoustic signal.

Abstract: A method and apparatus for processing an audio signal is disclosed. Herein, the method includes receiving a downmix information having at least two independent objects and a background object downmixed therein; separating the downmix information into a first independent object and a temporary background object using a first enhanced object information; and extracting a second independent object from the temporary background object using a second enhanced object information.

Abstract: A two-channel phase-amplitude stereo encoding and decoding scheme enabling flexible and spatially accurate interactive 3-D audio reproduction via standard audio-only two-channel transmission. The encoding scheme allows associating a 2-D or 3-D positional localization to each of a plurality of sound sources by use of frequency independent inter-channel phase and amplitude differences. The decoder is based on frequency-domain spatial analysis of 2-D or 3-D directional cues in a two-channel stereo signal and re-synthesis of these cues using any preferred spatialization technique, thereby allowing faithful reproduction of positional audio cues and reverberation or ambient cues over arbitrary multi-channel loudspeaker reproduction formats or over headphones, while preserving source separation despite the intermediate encoding over only two audio channels.

Abstract: The present invention relates to means and methods of automated difference recognition between speech and music signals in voice communication systems, devices, telephones, and methods, and more specifically, to systems, devices, and methods that automate control when either speech or music is detected over communication links. The present invention provides a novel system and method for monitoring the audio signal, analyze selected audio signal components, compare the results of analysis with a pre-determined threshold value, and classify the audio signal either as speech or music.

Abstract: An in-vehicle audio device and an in-vehicle audio system are disclosed. The in-vehicle audio device acquires and registers metadata of music data from multiple music possession devices, at least one of which is a portable music carrier device capable of performing wireless communication. Regarding the metadata acquired from the portable music carrier device, the in-vehicle audio device maintains the registration of the metadata after a communication connection between the portable music carrier device and the in-vehicle audio device is cut off. The in-vehicle audio device creates a playlist based on the registered metadata.

Abstract: A surround sound virtualization apparatus and method. The surround sound virtualization apparatus may include an audio decoder to perform head-related transfer function (HRTF) filtering, and a time delay unit to provide a time delay to a plurality of output signals of the audio decoder.

Abstract: An audio processor for converting a multi-channel audio input signal, such as a B-format sound field signal, into a set of audio output signals, such as a set of two or more audio output signals arranged for headphone reproduction or for playback over an array of loudspeakers. A filter bank splits each of the input channels into frequency bands. The input signal is decomposed into plane waves to determine one or two dominant sound source directions. The(se) are used to determine a set of virtual loudspeaker positions selected such that the dominant direction(s) coincide(s) with virtual loudspeaker positions. The input signal is decoded into virtual loudspeaker signals corresponding to each of the virtual loudspeaker positions, and the virtual loudspeaker signals are processed with transfer functions suitable to create the illusion of sound emanating from the directions of the virtual loudspeakers.

Abstract: An encoder and encoding method for use in a surround sound system wherein at least four audio input signals representing an original sound field are encoded into two channel signals and the encoded two channel signals are decoded into at least four audio output signals corresponding to the four audio input signals. The encoder includes matrix structure connected to receive the four audio input signals for encoding the four input signals into two channel output signals. The matrix structure is responsive to the four input signals for producing L and R output signals as follows: L=FL+kFR+jRL+jkRR R=FR+kFL?jRR?jkRL wherein k denotes a transformation or matrix constant having a value approximately 0.207 and j denotes a 90 degree phase shift.

Abstract: Embodiments of the present invention disclose an audio encoding method. The method includes: obtaining audio data of N channels; and performing channel interleaving and packetization on the obtained audio data of the N channels to obtain data packets, where each data packet includes X*N segments of audio data, where X is a ratio of an amount of audio data included in one data packet to an amount of audio data included in one audio frame, X is an integer greater than or equal to 1, and in the X*N segments of audio data, at least X+1 segments of audio data belong to different audio frames.

Abstract: A parameter transformer generates level parameters, indicating an energy relation between a first and a second audio channel of a multi-channel audio signal associated to a multi-channel loudspeaker configuration. The level parameter are generated based on object parameters for a plurality of audio objects associated to a down-mix channel, which is generated using object audio signals associated to the audio objects. The object parameters have an energy parameter indicating an energy of the object audio signal. To derive the coherence and the level parameters, a parameter generator is used, which combines the energy parameter and object rendering parameters, which depend on a desired rendering configuration.

Abstract: A method of processing an audio signal, comprising: receiving a downmix signal, a residual signal and object information; extracting at least one of a background-object signal and a foreground-object signal from the downmix signal using the residual signal; receiving mix information comprising gain control information for the background-object signal; generating a downmix processing information based on the object information and the mix information; and, generating a processed downmix signal comprising a modified background-object signal to which an adjusted gain corresponding to the gain control information is applied, by applying the downmix processing information to the at least one of the background-object signal and the foreground-object signal is disclosed.

Abstract: Wearers of hearing apparatuses and in particular of hearing device systems having two speakers are to be able to enjoy the experience of spatial multi-channel reproduction. Provision is accordingly made to generate a dual-channel audio signal for a binaural hearing apparatus comprising a multi-channel audio signal having at least three individual channels. Accordingly at least one spatial impression-influencing signal level in at least one of the individual channels is changed, and a signal of at least one of the individual channels is connected with signals of the remaining individual channels to the dual-channel audio signal. A corresponding hearing apparatus and in particular a corresponding hearing device have a transformation system that takes over this preprocessing from the multi-channel audio signal to the dual-channel audio signal.

Abstract: Method, apparatus and computer program product for processing an input stereophonic audio signal to thereby generate a converted stereophonic audio signal representing the input stereophonic audio signal, the input stereophonic audio signal comprising a left input audio signal and a right input audio signal, and the converted stereophonic audio signal comprising a first converted audio signal and a second converted audio signal. The first converted audio signal is generated based on the sum of the left input audio signal and the right input audio signal. The second converted audio signal is generated based on the difference between a first function of the left input audio signal and a second function of the right input audio signal. The first and second functions are adjustable to thereby adjust at least one characteristic of the converted stereophonic audio signal.

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 audio signal interpolation device is presented, including an input unit for receiving an input audio signal, a phase splitting unit for splitting the input audio signal, a high range interpolation unit for interpolating a high range component into the signal, a phase combining unit for combining an in-phase component signal with a differential phase component, a high-pass filter for high-pass filtering the audio signal from by the phase combining unit, a delay unit for producing a delayed audio signal, and an addition processing unit for adding the delayed audio signal to the audio signal output from the high-pass filter.

Abstract: Methods, systems and computer-readable medium reduce and/or eliminate errors in coding/decoding of streamed audio due to resetting of decoder state values based on playback buffer access. An inaudible compensation signal is included with the audio signal. The compensation signal is generated having a characteristic selected so that the encoded streamed audio signal substantially matches the reset state values at block boundaries. In an ADPCM example, the compensation signal is chosen such that the sum of the compensation signal and the original audio signal (=the compensated audio signal) has the characteristic that, at the block boundaries, the compensated audio signal matches the initial predictor value.

Abstract: Fluctuation in decoded signal localization is suppressed to maintain the feel of stereo. A selection unit selects balance parameters when the balance parameters are input from a gain coefficient decoding unit, or selects balance parameters input from a gain coefficient calculation unit when there is no balance parameter input from the gain coefficient decoding unit, and outputs the selected balance parameters to a multiplication unit. The multiplication unit multiplies a gain coefficient input from the selection unit with a decoded monaural signal input from a monaural decoding unit to perform balance adjustment processing.

Abstract: An audio decoding device includes a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute, decoding, using a first channel signal and a second channel signal included in a plurality of channels of an audio signal having a first frequency range and a second frequency range, a first prediction coefficient of the first frequency range and a second prediction coefficient of the second frequency range, both selected from a code book when prediction-encoding a third channel signal that is not subjected to prediction encoding and that is included in the plurality of channels; decoding a residual signal included in the first frequency range, the residual signal representing an error occurring in prediction encoding; and prediction-decoding the third channel signal subjected to prediction-encoding in the second frequency range from the first channel signal, the second channel signal.

Abstract: An encoding method and apparatus and a decoding method and apparatus are provided. The decoding method includes skipping extension information included in an input bitstream, extracting a three-dimensional (3D) down-mix signal and spatial information from the input bitstream, removing 3D effects from the 3D down-mix signal by performing a 3D rendering operation on the 3D down-mix signal, and generating a multi-channel signal using a down-mix signal obtained by the removal and the spatial information. Accordingly, it is possible to efficiently encode multi-channel signals with 3D effects and to adaptively restore and reproduce audio signals with optimum sound quality according to the characteristics of an audio reproduction environment.

Abstract: An audio decoder (100) comprising: effect means, decoding means, and rendering means. The effect means (500) generate modified down-mix audio signals from received down-mix audio signals. Said received down-mix audio signals comprise a down-mix of a plurality of audio objects. Said modified down-mix audio signals are obtained by applying effects to estimated audio signals corresponding to audio objects comprised in said received down-mix audio signals. Said estimated audio signals are derived from the received down-mix audio signals based on received parametric data. Said received parametric data comprise a plurality of object parameters for each of the plurality of audio objects. Said modified down-mix audio signals based on a type of the applied effect are decoded by decoding means or rendered by rendering means or combined with the output of rendering means.

Abstract: An audio decoder for decoding an encoded audio signal, the encoded audio signal including a first channel element and a second channel element in a payload section of a data stream and first decoder configuration data for the first channel element and second decoder configuration data for the second channel element in a configuration section of the data stream, includes: a data stream reader for reading the configuration data for each channel element in the configuration section and for reading the payload data for each channel element in the payload section; a configurable decoder for decoding the plurality of channel elements; and a configuration controller for configuring the configurable decoder so that the configurable decoder is configured in accordance with the first decoder configuration data when decoding the first channel element and in accordance with the second decoder configuration data when decoding the second channel element.

Abstract: A method for reducing howling in a communication system containing collocated mobile devices is presented. In a transmitter, an audio signal is received at a microphone. Acoustic feedback is removed from the audio signal and the resulting signal is encoded and transmitted either using direct or trunked mode operation to a receiver. The encoded signal is decoded at the transmitter, in addition to at the receiver, and fed back to an echo canceller with sufficient delay to account for substantially the entirety of a loop delay from encoding of the audio signal to reception of the acoustic feedback at the microphone to enable removal of the acoustic feedback. An estimate of the acoustic feedback is used to initially remove the acoustic feedback, the error being fed back to the processor to adaptively change the signal being subtracted from the audio signal to better reduce the acoustic feedback.

Abstract: An encoding method and apparatus and a decoding method and apparatus are provided. The decoding method includes extracting a down-mix signal and spatial information regarding a plurality of channels from an input bitstream, and generating a three-dimensional (3D) down-mix signal by performing a 3D rendering operation on the down-mix signal using the spatial information and a filter, wherein the sum of the number of valid signals of the down-mix signal, the number of valid signals of the spatial information, and the number of valid signals of co-efficients of the filter is less than the number of valid signals of the 3D down-mix signal. Accordingly, it is possible to efficiently encode multi-channel signals with 3D effects and to adaptively restore and reproduce audio signals with optimum sound quality according to the characteristics of an audio reproduction environment.

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: Provided are an audio encoding method and apparatus and an audio decoding method and apparatus in which audio signals can be encoded or decoded so that sound images can be localized at any desired position for each object audio signal. The audio decoding method includes extracting a downmix signal and object-based side information from an audio signal; generating channel-based side information based on object-based side information and control information for rendering the downmix signal; processing the downmix signal using a decorrelated channel signal; and generating a multi-channel audio signal using the processed downmix signal and the channel-based side information.

Abstract: Two digital data sets are combined by equating a first subset of samples to neighboring samples from a second subset which is interleaved with the first subset of samples where the equated samples of the two digital data sets do not correspond in time, and by subsequently adding corresponding samples from both digital data sets. This results in a third digital data set that allows the unraveling of the two digital data sets. The third digital data set, when combining two digital audio streams into a single digital audio stream, is still a good mono representation of the two combined digital audio streams and can thus be reproduced on regular reproduction equipment, yet the use of a decoder according to the invention allows the unraveling of the two digital data sets from the third digital data set.

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.