Abstract: A method and apparatus to quantize/dequantize frequency amplitude data and a method and apparatus to audio encode/decode using the method and apparatus to quantize/dequantize the frequency amplitude data. The method includes calculating and quantizing power of frequency amplitudes for each of a plurality of bands constituting an audio frame, normalizing frequency amplitude data for each of the bands using the quantized power, and quantizing a first one of even-numbered or odd-numbered data among the normalized frequency amplitude data.

Abstract: There is provided an audio encoding device capable of generating an appropriate monaural signal from a stereo signal while suppressing the lowering of encoding efficiency of the monaural signal. In a monaural signal generation unit (101) of this device, an inter-channel prediction/analysis unit (201) obtains a prediction parameter based on a delay difference and an amplitude ratio between a first channel audio signal and a second channel audio signal; an intermediate prediction parameter generation unit (202) obtains an intermediate parameter of the prediction parameter (called intermediate prediction parameter) so that the monaural signal generated finally is an intermediate signal of the first channel audio signal and the second channel audio signal; and a monaural signal calculation unit (203) calculates a monaural signal by using the intermediate prediction parameter.

Abstract: Spatial information associated with an audio signal is encoded into a bitstream, which can be transmitted to a decoder or recorded to a storage media. The bitstream can include different syntax related to time, frequency and spatial domains. In some embodiments, the bitstream includes one or more data structures (e.g., frames) that contain ordered sets of slots for which parameters can be applied. The data structures can be fixed or variable. The data structure can include position information that can be used by a decoder to identify the correct slot for which a given parameter set is applied. The slot position information can be encoded with either a fixed number of bits or a variable number of bits based on the data structure type as indicated by the data structure type indicator.

Abstract: The invention relates to the compression coding of digital signals such as multimedia signals (audio or video), and more particularly a method for multiple coding, wherein several encoders each comprising a series of functional blocks receive an input signal in parallel. Accordingly, a method is provided in which, a) the functional blocks forming each encoder are identified, along with one or several functions carried out of each block, b) functions which are common to various encoders are itemized and c) said common functions are carried out definitively for a part of at least all of the encoders within at least one same calculation module.

Abstract: The present invention is based on the finding that an efficient code for encoding information values can be derived, when two or more information values are grouped in a tuple in a tuple order and when an encoding rule is used, that assigns the same code word to tuples having identical information values in different orders and that does derive an order information, indicating the tuple order, and when the code word is output in association with the order information.

Abstract: Spatial information associated with an audio signal is encoded into a bitstream, which can be transmitted to a decoder or recorded to a storage media. The bitstream can include different syntax related to time, frequency and spatial domains. In some embodiments, the bitstream includes one or more data structures (e.g., frames) that contain ordered sets of slots for which parameters can be applied. The data structures can be fixed or variable. The data structure can include position information that can be used by a decoder to identify the correct slot for which a given parameter set is applied. The slot position information can be encoded with either a fixed number of bits or a variable number of bits based on the data structure type.

Abstract: Spatial information associated with an audio signal is encoded into a bitstream, which can be transmitted to a decoder or recorded to a storage media. The bitstream can include different syntax related to time, frequency and spatial domains. In some embodiments, the bitstream includes one or more data structures (e.g., frames) that contain ordered sets of slots for which parameters can be applied. The data structures can be fixed or variable. The data structure can include position information that can be used by a decoder to identify the correct slot for which a given parameter set is applied. The slot position information can be encoded with either a fixed number of bits or a variable number of bits based on the data structure type.

Abstract: An encoding device (200) includes an MDCT unit (202) that transforms an input signal in a time domain into a frequency spectrum including a lower frequency spectrum, a BWE encoding unit (204) that generates extension data which specifies a higher frequency spectrum at a higher frequency than the lower frequency spectrum, and an encoded data stream generating unit (205) that encodes to output the lower frequency spectrum obtained by the MDCT unit (202) and the extension data obtained by the BWE encoding unit (204). The BWE encoding unit (204) generates as the extension data (i) a first parameter which specifies a lower subband which is to be copied as the higher frequency spectrum from among a plurality of the lower subbands which form the lower frequency spectrum obtained by the MDCT unit (202) and (ii) a second parameter which specifies a gain of the lower subband after being copied.

Abstract: Provided is a method and apparatus for encoding/decoding a multi-channel audio signal. The apparatus for encoding a multi-channel audio signal includes a frame converter for converting the multi-channel audio signal into a framed audio signal; means for downmixing the framed audio signal; means for encoding the downmixed audio signal; a source location information estimator for estimating source location information from the framed multi-channel audio signal; means for quantizing the estimated source location information; and means for multiplexing the encoded audio signal and the quantized source location information, to generate an encoded multi-channel audio signal.

Abstract: Spatial information associated with an audio signal is encoded into a bitstream, which can be transmitted to a decoder or recorded to a storage media. The bitstream can include different syntax related to time, frequency and spatial domains. In some embodiments, the bitstream includes one or more data structures (e.g., frames) that contain ordered sets of slots for which parameters can be applied. The data structures can be fixed or variable. The data structure can include position information that can be used by a decoder to identify the correct slot for which a given parameter set is applied. The slot position information can be encoded with either a fixed number of bits or a variable number of bits based on the data structure type.

Abstract: Spatial information associated with an audio signal is encoded into a bitstream, which can be transmitted to a decoder or recorded to a storage media. The bitstream can include different syntax related to time, frequency and spatial domains. In some embodiments, the bitstream includes one or more data structures (e.g., frames) that contain ordered sets of slots for which parameters can be applied. The data structures can be fixed or variable. The data structure can include position information that can be used by a decoder to identify the correct slot for which a given parameter set is applied. The slot position information can be encoded with a fixed number of bits or a variable number of bits based on the data structure type.

Abstract: An apparatus for processing a signal and method thereof are disclosed. Data coding and entropy coding are performed with interconnection, and grouping is used to enhance coding efficiency. The present invention includes the steps of obtaining a pilot reference value corresponding to a plurality of data and a pilot difference value corresponding to the pilot reference value and obtaining the data using the pilot reference value and the pilot difference value.

Abstract: Apparatus, method, and medium for processing an audio signal using a correlation between bands are provided. The apparatus includes an encoding unit encoding an input audio signal and a decoding unit decoding the encoded input audio signal.

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

Abstract: The method compresses image data into a fixed sized memory. The image is encoded in scans, where the scans are ordered from a perceptually most significant scan to a perceptually least significant scan. The scans also have an attribute 210 determining whether a scan is active or inactive. The method comprises an encoding, transferring and setting step. The encoding step 326, 348 encodes active scans of image data into scan bit-stream data. The transferring step 330, 352 transfers the encoded scan bit-stream data to the fixed size memory. The setting step 452 sets, if the fixed size memory becomes full, the attribute of a perceptually least significant scan to inactive.

Abstract: A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal. In speech coding method according to a code-excited linear prediction (CELP) speech coding, a noise level of a speech in a concerning coding period is evaluated by using a code or coding result of at least one of spectrum information, power information, and pitch information, and various excitation codebooks are used based on an evaluation result.

Abstract: A parametric model is used for error concealment. The model filter allows for recovering signal components of original audio channel signals that now are lost or erroneous from signal components of at least one other audio channel. During error-free reception of valid frames, the parameters of that model will be derived and stored. In case of frame loss or frame error affecting the multi-channel information, a conjecture of the missing information is recovered by applying the model, using the stored parameters. In case of several subsequent lost or erroneous frames, it is possible either to use the parameters derived during the last valid frame or to use parameters derived from the recovered multi-channel information of the respective previous invalid frame. Furthermore, if there are long sequences of lost frames, it can be beneficial to apply some gradual muting of the model parameters, which essentially results in a gradual attenuation of the recovered multi-channel information.

Abstract: A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal. In speech coding method according to a code-excited linear prediction (CELP) speech coding, a noise level of a speech in a concerning coding period is evaluated by using a code or coding result of at least one of spectrum information, power information, and pitch information, and various excitation codebooks are used based on an evaluation result.

Abstract: A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal. In speech coding method according to a code-excited linear prediction (CELP) speech coding, a noise level of a speech in a concerning coding period is evaluated by using a code or coding result of at least one of spectrum information, power information, and pitch information, and various excitation codebooks are used based on an evaluation result.

Abstract: The present invention relates to a spectral enhancement method and to an apparatus carrying out this method. The method of the invention enhanced the spectral content of a signal having an incomplete spectrum including a first spectral frequency band, the method comprising the following stages: at least one spectral content transposition of said first frequency band into a second spectral frequency band not included in said spectrum for the purpose of generating a transposed spectrum signal having a spectrum limited to said second spectral frequency band, shaping the spectrum of the transposed spectrum signal for the purpose of producing an enhanced signal, combining an incomplete spectrum signal and the enhanced signal for the purpose of producing an enhanced spectrum signal, characterized in that said spectral content is subject to a stage of whitening.

Abstract: A multi-stage recursive sample rate converter (“SRC”) typically embodied as digital signal processor provides for an efficient structure for converting digital audio samples at one frequency, such as 48 kHz, to another frequency, such as 44.1 kHz. A parameter codebook comprising memory stores parameters used at a plurality of stages by the SRC. For each stage, a controller coordinates the SRC to use the appropriate set of parameters from the codebook, process an input audio sample stream, and store the intermediate results in a buffer. The controller then causes the intermediate results to be processed again as input to the SRC in a subsequent stage of processing using a different set of parameters. The process is repeated until all stages are completed, and the final results are the output digital audio data stream at the desired sampling rate.

Abstract: The invention relates to a method of selecting a respective coding model for encoding consecutive sections of an audio signal, wherein at least one coding model optimized for a first type of audio content and at least one coding model optimized for a second type of audio content are available for selection. In general, the coding model is selected for each section based on signal characteristics indicating the type of audio content in the respective section. For some remaining sections, such a selection is not viable, though. For these sections, the selection carried out for respectively neighboring sections is evaluated statistically. The coding model for the remaining sections is then selected based on these statistical evaluations.

Abstract: A decoder receives (501) a bitstream comprising an encoded mono signal and stereo data. A time scale processor (503) generates a time scaled mono signal. A time-to frequency processor generates frequency sample blocks of the time scaled signal, the block length being fixed and independent of the time scaling. A parametric stereo decoder (509) generates a stereo signal for the frequency sample blocks and these are converted to the time domain by a frequency-to-time processor (511). A synchronization processor (515) synchronizes the stereo data with the time scaled signal by determining a time association between a parameter value and a frequency sample block. The parameter value and time association is used to determine synchronized stereo parameter values for that and other frequency sample blocks. The invention is particularly suitable for low complexity generation of time scaled stereo signals from MPEG-4 encoded signals.

Abstract: A method for suppressing data is provided that includes receiving a first packet communicated by an end user and identifying a difference in a comfort noise level associated with a second packet received as compared to the first packet. The second packet may be communicated without a data payload in cases where the difference in comfort noise level associated with the second packet as compared to the first packet is below a predetermined threshold.

Abstract: The central idea of the present invention is that the prior procedure, namely interpolation relative to the filter coefficients and the amplification value, for obtaining interpolated values for the intermediate audio values starting from the nodes has to be dismissed. Coding containing less audible artifacts can be obtained by not interpolating the amplification value, but rather taking the power limit derived from the masking threshold, for each node, i.e. for each parameterization to be transferred, and then performing the interpolation between these power limits of neighboring nodes, such as, for example, a linear interpolation. On both the coder and the decoder side, an amplification value can then be calculated from the intermediate power limit determined such that the quantizing noise caused by quantization, which has a constant frequency before post-filtering on the decoder side, is below the power limit or corresponds thereto after post-filtering.

Abstract: Signals of different channels are combined into one mono signal. A set of adaptive filters, preferably one for each channel, is derived in a respective filter adaptation unit. When an adaptive filter is applied to the mono signal it reconstructs the signal of the respective channel under a perceptual constraint. The perceptual constraint is a gain and/or shape constraint. The gain constraint allows the preservation of the relative energy between the channels while the shape constraint allows more stability by avoiding unnecessary filtering of spectral nulls. The transmitted parameters are the mono signal, in encoded form, and the parameters of the adaptive filters, preferably also encoded. The receiver reconstructs the signal of the different channels by applying the adaptive filters and possibly some additional post-processing.

Abstract: An MPEG-1 layer 3 audio encoder, including a scalefactor generator for determining first scalefactors for encoding a block of audio data if a temporal masking transient is not detected in said block of audio data; and for selecting the maximum of said scalefactors for encoding said block of audio data if a temporal masking transient is detected in said block of audio data to enable greater compression of said audio data. Increases in quantization error, due to use of the maximum scalefactor are pre-masked or post-masked by the temporal masking transient. In cases where the last portion of a block includes a temporal masking transient that masks the preceding portions of the block, the maximum scalefactor is only used to encode the block if the resulting increase in quantization error is less than 30% of the quantization error for the block.

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

Abstract: Coding an audio signal of a sequence of audio values into a coded signal includes determining first and second listening thresholds for first and second blocks of audio values of the sequence of audio values; calculating versions of first second parameterizations of the parameterizable filter such that the transfer function thereof roughly corresponds to the inverse of the magnitude of the first and second listening thresholds, respectively; filtering a predetermined block of audio values of the sequence of audio values with the parameterizable filter using a predetermined parameterization which depends on the version of the second parameterization to obtain a block of filtered audio values corresponding to the predetermined block which is quantized; forming a difference between the versions of the first and second parameterizations; integrating information on, inter alias, the difference into the coded signal.

Abstract: A method of decoding audio data, encoded in multiple DIF blocks in a Digital Video (DV) data stream, and outputting said audio data as a PCM frame, includes fetching a single Digital Interface Frame (DIF) block from the DV data stream. A first byte in the single DIF block is de-shuffled to determine its index (n) in the PCM frame. Each byte in the in the single DIF block is de-shuffled to determine its respective index (n) in the PCM frame. The de-shuffled data is written into the PCM frame for output if the present DIF block is the last in the present DV frame. Subsequent DIF blocks in the DV frame are processed in the manner described above.

Abstract: There is disclosed an image processing apparatus which implements an image process suitable for an effect for compositing or switching images, and an image process that allows smooth high-speed playback of even an image in motion. The apparatus has band segmentation means for segmenting an image signal into a plurality of frequency band components, and image composition means for, after the band segmentation means segments input first and second image signals, outputting a third image signal by replacing image data for respective segmented band components. When image data, which are recorded while being segmented into a plurality of frequency band components, are composited for respective band, and the composite image data is output, image data obtained by compositing a plurality of image frequency components is decoded and played back in a high-speed playback mode.

Abstract: A coding apparatus including a base layer, a speech quality enhancement layer, and a multiplexer. The base layer filters an input speech signal using linear prediction coding and generates an excitation signal corresponding to the filtered speech signal through a fixed codebook search and an adaptive codebook search. The speech quality enhancement layer searches a fixed codebook using parameters obtained through the fixed codebook search in the base layer, or searches the fixed codebook using a target signal, which is obtained by removing a contribution of a fixed codebook of the base layer and a signal which is obtained by synthesizing and filtering a previous fixed codebook of the speech quality enhancement layer from a target signal for the fixed codebook search of the base layer. The multiplexer multiplexes signals generated by the base layer and the at least one speech quality enhancement layer.

Abstract: Automated testing of audio performance of applications across platforms is provided for via capture of audio data. The audio data can include, inter alia, output sounds from a sound card or pre-rendered buffer data. The audio data is processed to produce descriptive data including data describing the audio data at least a first resolution and a second resolution. This descriptive data is used to compare data samples and describe the degree of similarity of two or more data samples. This comparison enables a determination as to whether the audio performance is satisfactory.

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

Abstract: A method for generating an encoded audio signal, and a method for processing the same during the multi-channel audio coding are disclosed. The present invention provides the method for generating an encoded audio signal comprising: generating basic spatial information including basic configuration information requisite for a multi-channel audio coding process and basic data corresponding to the basic configuration information; and generating extension spatial information including extension configuration information selectively required for the multi-channel audio coding process and extension data corresponding to the extension configuration information.

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

Abstract: To provide an enhanced true-to-life atmosphere enjoyed in multipoint connecting, and reduce a calculation load at a multipoint connection unit, as well.

Abstract: A method, system and computer program product for computationally efficient estimation of the scale factors of one or more frequency bands in an encoder. These scale factors are dependant on a plurality of variables. One of the variables is approximated according to embodiments of the invention. This reduces the complexity of the estimation of scale factors, especially in digital signal processors.

Abstract: An audio codec losslessly encodes audio data into a sequence of analysis windows in a scalable bitstream. This is suitably done by separating the audio data into MSB and LSB portions and encoding each with a different lossless algorithm. An authoring tool compares the buffered payload to an allowed payload for each window and selectively scales the losslessly encoded audio data, suitably the LSB portion, in the non-conforming windows to reduce the encoded payload, hence buffered payload. This approach satisfies the media bit rate and buffer capacity constraints without having to filter the original audio data, reencode or otherwise disrupt the lossless bitstream.

Abstract: A multi-channel synthesizer for generating at least three output channels using an input signal having at least one base channel, the base channel being derived from the original multi-channel signal, the input signal further including at least two different up-mixing parameters, and an up-mixer mode indication indicating, in a first state that a first up-mixing rule is to be performed, and, indicating, in a second state, that a different second up-mixing rule is to be performed, uses an up-mixer for up-mixing the at least one base channel using the at least two different up-mixing parameters based on the first or the second up-mixing rule in response to the up-mixer mode indication so that the at least three output channels are obtained.

Abstract: A lossless audio coding and/or decoding method and apparatus are provided. The coding method includes: mapping the audio signal in the frequency domain having an integer value into a bit-plane signal with respect to the frequency; obtaining a most significant bit and a Golomb parameter for each bit-plane; selecting a binary sample on a bit-plane to be coded in the order from the most significant bit to the least significant bit and from a lower frequency component to a higher frequency component; calculating the context of the selected binary sample by using significances of already coded bit-planes for each of a plurality of frequency lines existing in the vicinity of a frequency line to which the selected binary sample belongs; selecting a probability model by using the obtained Golomb parameter and the calculated contexts; and lossless-coding the binary sample by using the selected probability model.

Abstract: There is provided a communication device for effectively encoding an audio/music signal while maintaining a predetermined quality by controlling the transmission bit rate of the transmission side considering the use environment of the reception side. In this device, a transmission mode decision unit (101) detects an environment noise contained in the background of the audio/music signal in the input signal and decides the transmission mode controlling the transmission bit rate of the signal transmitted from a communication terminal device (150), which is a communication terminal of the partner side, according to the environment noise level. A signal decoding unit (103) decodes encoded information transmitted from the communication terminal device (150) via a transmission path (110) and outputs the obtained signal as an output signal.

Abstract: A technique to encode an audio signal based on a perceptual model. In one example embodiment, this is accomplished by shaping quantization noise in the spectral lines on a band-by-band basis using their local gains. The noise shaped spectral lines are then fitted within a predetermined bit rate to form an encoded bit stream.

Abstract: The system for transmitting and receiving a wideband speech signal includes an A/D converter for receiving an analog speech signal to convert it into a digital speech signal, a transmitter analysis filter for receiving the digital speech signal and dividing it into a baseband signal and an enhancement residual band signal, a standard baseband encoder for accepting the baseband signal and coding it using an ITU-T encoder, an additional baseband encoder for reducing standard coding distortion in the baseband signal, an enhancement residual band encoder for coding a signal obtained by removing the coded baseband signal from the original digital speech signal, and an IP network interface for multiplexing the coded standard and additional baseband signals and enhancement residual band signal.

Abstract: A method and an apparatus for encoding digital audio data with reduced bit rates, the apparatus having a provider of psycho-acoustically quantized digital audio data with a bit rate being higher than the reduced bit rate. The apparatus further has an identifier for identifying a frequency band according to a selection criterion, the selection criterion being such that an impact on the quality of the digital audio data when the data in the identified frequency band is replaced by generated noise is smaller than the impact on the quality of the digital audio data, which would arise when the data in a different frequency band is replaced by generated noise. The apparatus further has a replacer for replacing data in the identified frequency band of the digital audio data by a noise synthesis parameter.

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: 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: 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: A subband synthesis filtering apparatus for M sets of signals is provided. Each set of signals includes N subband sample signals. The apparatus includes a processor for processing the ith set of signals among the M sets of signals, wherein i is an integer index ranging from 0 to (M?1). The processor includes a DCT converting module and a generating module. The DCT converting module converts the N subband sample signals of the ith set of signals into N converted vectors. If i is an odd number, the (2j?1)th subband sample signal among the N subband sample signals is multiplied by negative one in the converting module, wherein j is an integer index ranging from 1 to (N/2). The generating module generates N pulse code modulation signals based on the N converted vectors.