Abstract: An apparatus for decoding a signal and method thereof are disclosed, by which the audio signal can be controlled in a manner of changing/giving spatial characteristics (e.g., listener's virtual position, virtual position of a specific source) of the audio signal. The present invention includes receiving an object parameter; extracting object information by parsing the received object parameter; generating a control parameter using the extracted object information and control information including at least one of user control information, default control information, device control information, and device information; and, generating a rendering parameter determining a position and level of an object in an output signal using the object parameter and the control parameter.

Abstract: An audio waveform processing not imparting any feeling of strangeness and high in definition, in which time stretch and pitch shift are performed by a vocoder method, and the variation of phase over the whole waveform caused by the vocoder method at all times is reduced. An audio input waveform is handled as one band as it is or subjected to frequency band division into bands. While performing time stretch and pitch shift of each band waveform like conventional vocoder methods, the waveforms are combined. The combined waveform of the band is phase-synchronized at regular intervals to reduce the variation of phase. The phase-synchronized waveforms of the band are added, thus obtaining the final output waveform.

Abstract: An exemplary embodiment of the invention can generate multiple output audio signals from multiple input audio signals, in which the number of output signals is equal to or higher than the number of input signals. The embodiment includes computing one or more independent sound subbands representing signal components which are independent between the input subbands; computing one or more localized direct sound subbands representing signal components which are contained in more than one of the input subbands and direction factors representing the ratios with which these signal components are contained in two or more input subbands; generating the output subband signals, where each output subband signal is a linear combination of the independent sound subbands and the localized direct sound subbands; and converting the output subband signals to time domain audio signals.

Abstract: An apparatus calculates a number of spectral envelopes to be derived by a spectral band replication (SBR) encoder, wherein the SBR encoder is adapted to encode an audio signal using a plurality of sample values within a predetermined number of subsequent time portions in an SBR frame extending from an initial time to a final time, the predetermined number of subsequent time portions being arranged in a time sequence given by the audio signal. The apparatus has a decision value calculator for determining a decision value, the decision value measuring a deviation in spectral energy distributions of a pair of neighboring time portions. The apparatus further has a detector for detecting a violation of a threshold by the decision value and a processor for determining a first envelope border between the pair of neighboring time portions when the violation of the threshold is detected.

Abstract: One or more attributes (e.g., pan, gain, etc.) associated with one or more objects (e.g., an instrument) of a stereo or multi-channel audio signal can be modified to provide remix capability. An audio decoding apparatus obtains an audio signal having a set of objects and side information. The apparatus obtains a set of mix parameters from a user input and an attenuation factor from the set of mix parameters. The apparatus then generates a plural-channel audio signal using at least one of the side information, the attenuation factor or the set of mix parameters.

Abstract: An audio information processing apparatus and method include dividing an audio signal, determining a time period having a power change ratio of an audio signal larger than a first threshold value as an attack candidate, searching the time period of the attack candidate and a time period immediately before the time period of the attack candidate for an attack starting point, correcting a power of an audio signal included in the time period, and determining whether a power change ratio of the audio signal included in the time period is larger than a second threshold value for attack detection which is larger than the first threshold value.

Abstract: An audio coding terminal and method is provided. The terminal includes a coding mode setting unit to set an operation mode, from plural operation modes, for input audio coding by a codec, configured to code the input audio based on the set operation mode such that when the set operation mode is a high frame erasure rate (FER) mode the codec codes a current frame of the input audio according to a select frame erasure concealment (FEC) mode of one or more FEC modes. Upon the setting of the operation mode to be the High FER mode, the one FEC mode is selected, from the one or more FEC modes predetermined for the High FER mode, to control the codec by incorporating of redundancy within a coding of the input audio or as separate redundancy information separate from the coded input audio according to the selected one FEC mode.

Abstract: The present invention provides a signal processing apparatus, a signal processing method and a program for outputting a high-quality coded string. A signal processing apparatus according to an embodiment of the present invention includes a normalization coefficient information increasing/decreasing circuit 12 for modifying normalization coefficient information of a signal component of a frame and normalization coefficient information of a primary additional signal component according to a normalization coefficient information primary increase/decrease amount, and an additional signal component normalization coefficient information increasing/decreasing circuit 14 for modifying normalization coefficient information of a secondary additional signal component, which is a copy of the primary additional signal component, according to a normalization coefficient information secondary increase/decrease amount.

Abstract: An encoding method and apparatus and a decoding method and apparatus are provided. The decoding method includes extracting a three-dimensional (3D) down-mix signal and spatial information from an 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 the spatial information and a down-mix signal obtained by the removal. 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 a reproduction environment.

Abstract: A voice band expansion device includes a time-frequency converter that calculates a frequency spectrum of a voice signal having a first frequency band; a separator that extracts, from the frequency spectrum, an envelope amplitude spectrum, a periodic amplitude spectrum, and a random amplitude spectrum; an envelope amplitude spectrum band expander that expands a frequency band to a second frequency band that is different from the first frequency band; a periodic amplitude spectrum band expander that expands a frequency band to the second frequency band; a random amplitude spectrum band expander that expands a frequency band of the random amplitude spectrum to the second frequency band; a broadband spectrum calculator that calculates a broadband frequency spectrum having the first frequency band and the second frequency band; and a frequency-time converter generates a voice signal having the first frequency band and the second frequency band.

Abstract: An audio decoder for decoding a multi-audio-object signal having an audio signal of a first type and an audio signal of a second type encoded therein is described, the multi-audio-object signal having a downmix signal and side information, the side information having level information of the audio signals of the first and second types in a first predetermined time/frequency resolution, and a residual signal specifying residual level values in a second predetermined time/frequency resolution, the audio decoder having a processor for computing prediction coefficients based on the level information; and an up-mixer for up-mixing the downmix signal based on the prediction coefficients and the residual signal to obtain a first up-mix audio signal approximating the audio signal of the first type and/or a second up-mix audio signal approximating the audio signal of the second type.

Abstract: An apparatus for decoding an encoded audio signal having first and second portions encoded in accordance with first and second encoding algorithms, respectively, BWE parameters for the first and second portions and a coding mode information indicating a first or a second decoding algorithm, includes first and second decoders, a BWE module and a controller. The decoders decode portions in accordance with decoding algorithms for time portions of the encoded signal to obtain decoded signals. The BWE module has a controllable crossover frequency and is configured for performing a bandwidth extension algorithm using the first decoded signal and the BWE parameters for the first portion, and for performing a bandwidth extension algorithm using the second decoded signal and the bandwidth extension parameter for the second portion. The controller controls the crossover frequency for the BWE module in accordance with the coding mode information.

Abstract: In order to achieve the best improvement of ITU G.711 related codec perceptual quality, perceptual weighting controlling parameter(s) should be at least adaptive to relative quantization error statistics or adaptive to signal level. When the relative quantization error statistics are larger or the signal level is lower, the perceptual weighting should be “stronger”, which means ? in (5) is smaller; when the relative quantization error statistics are smaller or the signal level is larger, the perceptual weighting should be “weaker”, which means ? in (5) is larger.

Abstract: A method, medium, and apparatus encoding and/or decoding an audio signal to surround data. While encoding spatial information, which can up-mix an audio signal to a surround signal, to extension data, a length of a payload corresponding to the spatial information is encoded and a payload of the spatial information is decoded using the length of the payload. Accordingly, compatibility of the spatial information can be provided, and the spatial information can be transmitted by effectively embedding the spatial information.

Abstract: In processing a multi-channel audio signal having at least three original channels, a first downmix channel and a second downmix channel are provided, which are derived from the original channels. For a selected original channel of the original channels, channel side information are calculated such that a downmix channel or a combined downmix channel including the first and the second downmix channels, when weighted using the channel side information, results in an approximation of the selected original channel. The channel side information and the first and second downmix channels form output data to be transmitted to a decoder, which, in case of a low level decoder only decodes the first and second downmix channels or, in case of a high level decoder provides a full multi-channel audio signal based on the downmix channels and the channel side information.

Abstract: An audio system comprises an encoder (209) which encodes audio objects in an encoding unit (403) that generates a down-mix audio signal and parametric data representing the plurality of audio objects. The down-mix audio signal and parametric data is transmitted to a decoder (215) which comprises a decoding unit (301) which generates approximate replicas of the audio objects and a rendering unit (303) which generates an output signal from the audio objects. The decoder (215) furthermore contains a processor (501) for generating encoding modification data which is sent to the encoder (209). The encoder (209) then modifies the encoding of the audio objects, and in particular modifies the parametric data, in response to the encoding modification data. The approach allows manipulation of the audio objects to be controlled by the decoder (215) but performed fully or partly by the encoder (209).

Abstract: A method and system for compressing an audio signal. The method includes receiving a segment of an audio signal and selectively disabling noise suppression for the received segment. The segment is filtered in a noise-suppression module if noise suppression is not disabled. The method also includes calculating an autocorrelation coefficient and an LSP coefficient, predicting a short-term coefficient and long-term coefficients according to the LSP coefficient and calculating one or more bandwidth-expanded correlation coefficients. Further, the method includes determining the type of packet in which to encode the segment. An encoding rate is selected from among a full rate encode, a half-rate encode, and an eight-rate encode if noise suppression is not disabled. An encoding rate is selected from among a full rate encode and a half-rate encode if noise suppression is disabled. Furthermore, the segment is formed into a packet of the determined type and selected rate.

Abstract: A method for decoding an audio signal comprises receiving a combined downmix, a combined object information, and a mix information, the combined downmix being generating using at least two downmix signals, the combined object information being made by combination of at least two sets of object information, generating a downmix processing information using the combined object information and the mix information, and processing the combined downmix using the downmix processing information. The method and an apparatus for decoding an audio signal comprising the combined downmix and the combined object information can control object gain and output in a remote conference and so on. The method and the apparatus for decoding audio signal that contains multi-object signals are fast and efficiently by reducing process time, computer resource, thereby relieving the resource requirement like the wide bandwidth by using the combined object information.

Abstract: A method and apparatus are disclosed for generating a coded audio signal based on a multiple channel audio input signal. A balance factor having balance factor components each associated with an audio signal of the multiple channel audio signal is generated. A gain value to be applied to the coded audio signal to generate an estimate of the multiple channel audio signal based on the balance factor and the multiple channel audio signal is determined, with the gain value configured to minimize a distortion value between the multiple channel audio signal and the estimate of the multiple channel audio signal.

Abstract: A hierarchical audio coder for use in a frequency band divided into adjacent first and second sub-bands, the coder including: a core coder (305) for coding an original signal in the first sub-band of the frequency band; a stage (306) for calculating a residual signal (e) from the original signal and the signal from the core coder; a device (307) for perceptually weighting the residual signal (e). The perceptual weighting device includes a perceptually weighted filter (307) with gain compensation adapted to realize spectral continuity between the output signal of the perceptually weighted filter with gain compensation and the signal in the second sub-band. Application to transmitting and storing digital signals, such as audio-frequency speech, music, etc. signals.

Abstract: An efficient encoded representation of a first and a second input audio signal can be derived using correlation information indicating a correlation between the first and the second input audio signals, when a signal characterization information, indicating at least a first or a second, different characteristic of the input audio signal is additionally considered. Phase information indicating a phase relation between the first and the second input audio signals is derived, when the input audio signals have the first characteristic. The phase information and a correlation measure are included into the encoded representation when the input audio signals have the first characteristic, and only the correlation information is included into the encoded representation when the input audio signals have the second characteristic.

Abstract: An audio encoding method previously estimates better initial iterative values of global-gain and scalefactor for avoiding heavy calculation. The estimating process of the encoding method includes calculating the bit allocation of one frequency sample based on a sampling rate, a bit rate, and the number of audio channels according to an input frame, and the psychoacoustic model, searching one frequency sample having the greatest sample energy in each of a plurality of scalefactor bands, quantizing the frequency sample to comply with the bit allocation and to generate a corresponding scalefactor, searching a maximum scalefactor of all scalefactor bands corresponding to the input frame, and setting initial values of scalefactors and an initial value of global-gain for the quantization iterative loop process according to the corresponding scalefactor and the maximum scalefactor.

Abstract: An encoder (109) comprises a receiver (201) which receives a time domain audio signal. A filter bank (203) generates a first subband signal from the time domain audio signal where the first subband signal corresponds to a non-critically sampled complex subband domain representation of the time domain signal. A conversion processor (205) generates a second subband signal from the first subband signal by subband processing. The second subband signal corresponds to a critically sampled complex subband domain representation of the time domain audio signals. An encode processor (207) then generates a waveform encoded data stream by encoding data values of the second subband signal. The conversion processor (205) generates the second subband signal by direct subband conversion without converting back to the time domain. The invention allows an oversampled subband signal typically generated in parametric encoding to be waveform encoded with reduced complexity. A decoder performs the inverse operation.

Abstract: An audio encoder and decoder use architectures and techniques that improve the efficiency of multi-channel audio coding and decoding. The described strategies include various techniques and tools, which can be used in combination or independently. For example, an audio encoder performs a pre-processing multi-channel transform on multi-channel audio data, varying the transform so as to control quality. The encoder groups multiple windows from different channels into one or more tiles and outputs tile configuration information, which allows the encoder to isolate transients that appear in a particular channel with small windows, but use large windows in other channels. Using a variety of techniques, the encoder performs flexible multi-channel transforms that effectively take advantage of inter-channel correlation. An audio decoder performs corresponding processing and decoding. In addition, the decoder performs a post-processing multi-channel transform for any of multiple different purposes.

Abstract: Methods and an apparatus for enhancement of source coding systems utilizing high frequency reconstruction (HFR) are introduced. The problem of insufficient noise contents is addressed in a reconstructed highband, by using Adaptive Noise-floor Addition. New methods are also introduced for enhanced performance by means of limiting unwanted noise, interpolation and smoothing of envelope adjustment amplification factors. The methods and apparatus used are applicable to both speech coding and natural audio coding systems.

Abstract: A multi-channel audio decoder reconstructs multi-channel audio of more than two physical channels from a reduced set of coded channels based on correlation parameters that specify a full power cross-correlation matrix of the physical channels, or merely preserve a partial correlation matrix (such as power of the physical channels, and some subset of cross-correlations between the physical channels, or cross-correlations of the physical channels with coded or virtual channels).

Abstract: A decoding apparatus that decodes a first encoded data that is encoded into a first time range from a low-frequency component of an audio signal, and a second encoded data that is used when creating a high-frequency component of the audio signal from the low-frequency component and encoded into a second time range, into the audio signal. In the decoding apparatus, a high-frequency component compensating unit that compensates the high-frequency component created from the second encoded data based on the first time range. A decoding unit that decodes into the audio signal by synthesizing the high-frequency component compensated by the high-frequency component compensating unit, and the low-frequency component decoded from the first encoded data.

Abstract: Disclosed is an adaptive sound source vector quantization device capable of reducing deviation of the quantization accuracy of the adaptive sound source vector quantization of each sub-frame when performing an adaptive sound source vector quantization in a sub-frame unit by using a greater information amount in a first sub-frame than in a second sub-frame.

Abstract: Embodiments of the invention provide a method and encoder for encoding a frame in of a communication system. The method includes calculating a first set of parameters associated with the frame, wherein said first set of parameters comprises filter bank parameters. The method further includes selecting, in a first stage, one of a plurality of encoding methods based on the first set of parameters one of modes for encoding, calculating a second set of parameters associated with the frame, selecting, in a second stage, one of the plurality of encoding methods based on the result of the first stage selection and the second set of parameters one of modes for encoding, and encoding the frame using the selected encoding excitation method from the second stage.

Abstract: In one embodiment, a highband burst suppressor includes a first burst detector configured to detect bursts in a lowband speech signal, and a second burst detector configured to detect bursts in a corresponding highband speech signal. The lowband and highband speech signals may be different (possibly overlapping) frequency regions of a wideband speech signal. The highband burst suppressor also includes an attenuation control signal calculator configured to calculate an attenuation control signal according to a difference between outputs of the first and second burst detectors. A gain control element is configured to apply the attenuation control signal to the highband speech signal. In one example, the attenuation control signal indicates an attenuation when a burst is found in the highband speech signal but is absent from a corresponding region in time of the lowband speech signal.

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: In one embodiment, C input audio channels are encoded to generate E transmitted audio channel(s), where one or more cue codes are generated for two or more of the C input channels, and the C input channels are downmixed to generate the E transmitted channel(s), where C>E?1. One or more of the C input channels and the E transmitted channel(s) are analyzed to generate a flag indicating whether or not a decoder of the E transmitted channel(s) should perform envelope shaping during decoding of the E transmitted channel(s). In one implementation, envelope shaping adjusts a temporal envelope of a decoded channel generated by the decoder to substantially match a temporal envelope of a corresponding transmitted channel.

Abstract: The invention concerns a method for spectral enhancement and a device therefor. The inventive method is a method for enhancing spectral content of a signal having an incomplete spectrum including a first spectral band, the method including the following steps: at least transposing the spectral content of the first band into a second spectral band not included in the spectrum to generate a transposed spectrum signal, with spectrum limited to the second spectral band; transforming the spectrum of the transposed spectrum signal to obtain an enhancing signal; combining the incomplete spectrum signal and the enhancing signal to produce a spectrum enhanced signal. The invention is characterized in that the spectral content is subjected to a whitening step.

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, which is obtained by downmixing a plurality of object signals, and object side information, extracting metadata from the object-side information and displaying an information regarding the object signals based on the metadata.

Abstract: There is provided a wide-band LSP prediction device and others capable of predicting a wide-band LSP from a narrow-band LSP with a high quantization efficiency and a high accuracy while suppressing the size of a conversion table correlating the narrow-band LSP to the wide-band LSP. In this device, a non-linear prediction unit (102) performs non-linear prediction by using a converted wide-band LSP inputted from a narrow-band/wide-band conversion unit (101) and inputs the non-linear prediction result to an amplifier (103). The converted wide-band LSP is inputted to an amplifier (104). An adder (122) adds multiplication results (vectors) inputted from the amplifiers (103, 104).

Abstract: In the method and device for interoperating a first station using a first communication scheme and comprising a first coder and a first decoder with a second station using a second communication scheme and comprising a second coder and a second decoder, communication between the first and second stations is conducted by transmitting signal-coding parameters related to a sound signal from the coder of one of the first and second stations to the decoder of the other station. The sound signal is classified to determine whether the signal-coding parameters should be transmitted from the coder of one station to the decoder of the other station using a first communication mode in which full bit rate is used for transmission of the signal-coding parameters.

Abstract: According to one embodiment, an improved audio coding technique encodes audio having a low frequency transient signal, using a long block, but with a set of adapted masking thresholds. Upon identifying an audio window that contains a low frequency transient signal, masking thresholds for the long block may be calculated as usual. A set of masking thresholds calculated for the 8 short blocks corresponding to the long block are calculated. The masking thresholds for low frequency critical bands are adapted based on the thresholds calculated for the short blocks, and the resulting adapted masking thresholds are used to encode the long block of audio data. The result is encoded audio with rich harmonic content and negligible coder noise resulting from the low frequency transient signal.

Abstract: A method is provided for coding a source audio signal. The method includes the following steps: coding a quantization profile of coefficients representative of at least one transform of the source audio signal, according to at least to distinct coding techniques, delivering at least two sets of data representative of a quantization profile; selecting one of the sets of data representative of a quantization profile, as a function of a predetermined selection criterion; transmitting and/or storing the set of data representative of a selected quantization profile and an indicator representative of the corresponding coding technique.

Abstract: Presented herein are systems and methods for processing sound signals for use with electronic speech systems. Sound signals are temporally parsed into frames, and the speech system includes a speech codebook having entries corresponding to frame sequences. The system identifies speech sounds in an audio signal using the speech codebook.

Abstract: An encoder/decoder for multi-channel audio data, and in particular for audio reproduction through wave field synthesis. The encoder comprises a two-dimensional filter-bank to the multi-channel signal, in which the channel index is treated as an independent variable as well as time, and and the resulting spectral coefficient are quantized according to a two-dimensional psychoacoustic model, including masking effect in the spatial frequency as well as in the temporal frequency. The coded spectral data are organized in a bitstream together with side information containing scale factors and Huffman codebook identifiers.

Abstract: An overall encoding procedure and associated decoding procedure are presented. The encoding procedure involves at least two signal encoding processes operating on signal representations of a set of audio input channels. Local synthesis is used in connection with a first encoding process to generate a locally decoded signal, including a representation of the encoding error of the first encoding process. This locally decoded signal is applied as input to a second encoding process. The overall encoding procedure generates at least two residual encoding error signals from at least one of said encoding processes, including at least said second encoding process. The residual error signals are then subjected to compound residual encoding in a further encoding process, preferably based on correlation between the residual error signals.

Abstract: An audio/speech sender and an audio/speech receiver and methods thereof. The audio/speech sender comprising a core encoder adapted to encode a core frequency band of an input audio/speech signal having a first sampling frequency, wherein the core frequency band comprises frequencies up to a cut-off frequency. The audio/speech sender further comprises a segmentation device adapted to perform a segmentation of the input audio/speech signal into a plurality of segments, a cut-off frequency estimator adapted to estimate a cut-off frequency for each segment and adapted to transmit information about the estimated cut-off frequency to a decoder, a low-pass filter adapted to filter each segment at said estimated cut-off frequency, and a re-sampler adapted to resample the filtered segments with a second sampling frequency that is related to said cut-off frequency in order to generate an audio/speech frame to be encoded by said core encoder.

Abstract: During operation an input signal to be coded is received and coded to produce a coded audio signal. The coded audio signal is then scaled with a plurality of gain values to produce a plurality of scaled coded audio signals, each having an associated gain value and a plurality of error values are determined existing between the input signal and each of the plurality of scaled coded audio signals. A gain value is then chosen that is associated with a scaled coded audio signal resulting in a low error value existing between the input signal and the scaled coded audio signal. Finally, the low error value is transmitted along with the gain value as part of an enhancement layer to the coded audio signal.

Abstract: An apparatus for processing a media signal and method thereof are disclosed, by which the media signal can be converted to a surround signal by using spatial information of the media signal. The present invention provides a method of processing a signal, the method comprising of extracting spatial information and a downmix signal from a bitstream; and generating rendering information by using the spatial information and filter information having a surround effect, wherein the rendering information comprises first rendering information applied to one channel of the downmix signal extracted from the bitstream and then transmitted on the same channel and second rendering information applied to the channel and then transmitted on another channel.

Abstract: Disclosed is a method in a network element of a communication network, which communication network is capable of transparently transferring coded data at least in some part of the communication network. The method includes detecting a need to change codec rate to a second codec rate in a downlink connection from the communication network to an end user device; receiving coded data destined to said end user device, which data is coded with a first codec rate, and starting, in response to said detecting, rate transformation for transforming codec rate of said data destined to the end user device into said second codec rate. Also disclosed are an apparatus, a system and a computer program.

Abstract: A down-sampler 101 down-samples the sampling rate of an input signal from sampling rate FH to sampling rate FL. A base layer coder 102 encodes the sampling rate FL acoustic signal. A local decoder 103 decodes coding information output from base layer coder 102. An up-sampler 104 raises the sampling rate of the decoded signal to FH. A subtracter 106 subtracts the decoded signal from the sampling rate FH acoustic signal. An enhancement layer coder 107 encodes the signal output from subtracter 106 using a decoding result parameter output from local decoder 103.

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, which is obtained by downmixing a plurality of object signals, and object side information, extracting metadata from the object-side information and displaying an information regarding the object signals based on the metadata.

Abstract: An iterative rate-distortion optimization algorithm for MPEG I/II Layer-3 (MP3) encoding based on the method of Lagrangian multipliers. Generally, an iterative method is performed such that a global quantization step size is determined while scale factors are fixed, and thereafter the scale factors are determined while the global quantization step size is fixed. This is repeated until a calculated rate-distortion cost is within a predetermined threshold. The methods are demonstrated to be computationally efficient and the resulting bit stream is fully standard compatible.

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 “0.1” indicates a single low-frequency effects (LFE) channel and “0.2” indicates two LFE channels.

Abstract: Synthesizing a set of digital speech samples corresponding to a selected voicing state includes dividing speech model parameters into frames, with a frame of speech model parameters including pitch information, voicing information determining the voicing state in one or more frequency regions, and spectral information. First and second digital filters are computed using, respectively, first and second frames of speech model parameters, with the frequency responses of the digital filters corresponding to the spectral information in frequency regions for which the voicing state equals the selected voicing state. A set of pulse locations are determined, and sets of first and second signal samples are produced using the pulse locations and, respectively, the first and second digital filters. Finally, the sets of first and second signal samples are combined to produce a set of digital speech samples corresponding to the selected voicing state.