Abstract: Disclosed is a code conversion method to convert a first code sequence conforming to a first speech coding scheme into a second code sequence conforming to a second speech coding scheme. The method includes the following steps. The first step discriminates whether the first code sequence corresponds to a speech part or to a non-speech part, and generates a numerical value that indicates the discrimination result as a control flag. The second step converts the first code sequence into the second code sequence and outputs said second code sequence, when the value of the control flag corresponds to the speech part. The third step outputs the second code sequence that corresponds to the value of the control flag, when the value of the control flag corresponds to the non-speech part.

Abstract: An encoder improves inter-channel prediction (ICP) performance in scalable stereo sound encoding using an ICP. In the encoder, ICP analysis units use, as reference signal candidates, a frequency coefficient in the low-band portion of a side residual signal, a frequency coefficient in each sub-band portion of a monaural residual signal, and a frequency coefficient in the low-band portion of the monaural residual signal, respectively, and perform an ICP analysis between the these respective candidates and a frequency coefficient in each sub-band portion of the side residual signal to generate first, second, and third ICP coefficients.

Abstract: Perceptual audio coder refers to audio compression schemes that exploit the properties of human auditory perception. The coder allocates the quantization noise below the masking threshold such that even with the bit rate limitation, the noise is imperceptible to the ear. These distortion and bit rate requirement makes the bit allocation-quantization process a considerable computational effort. One method includes incrementally adjusting a global gain according to a gradient. The gradient could be adjusted each time the number of bits used to represent a quantized value is counted. Another method includes limiting a rate controlling parameter to a predetermined number of loops. The method could also include deriving a global gain to ensure exit from the loop. Accordingly, embodiments of the present disclosure provide a fast and efficient method to derive the rate controlling parameter and can be applied to generic perceptual audio encoders where low computational complexity is required.

Abstract: Audio encoding method and device comprising the transmission, in addition to the data representing a frequency-limited signal, of information relating to a temporal filter that is to be applied to the entire enhanced signal, both in its transmitted low-frequency part and in its reconstituted high-frequency part. The application of this filter for reshaping the reconstituted high-frequency part and the correction of compression artefacts present in the transmitted low-frequency part. In this way, the application of the temporal filter, simple and inexpensive, to all or part of the reconstituted signal, makes it possible to obtain a signal of good perceived quality.

Abstract: Various techniques are disclosed for reducing artifacts generated by time compression. by adapting the time compression based on the state of the received audio. The amount of time compression may be bounded based on audio characteristics. Another feature provides a way of determining the most correlated portions of segments of audio. Voiced speech may be distinguished from unvoiced speech. Another feature provides a way of distinguishing between silence, voiced speech, and unvoiced speech. Time compression may be adapted during periods of lengthy silence. Another feature allows for reducing time compression during sensitive portions of the received audio. One or more of these features may be present in different embodiments.

Abstract: An apparatus and method for encoding and decoding using mutual information between a high band signal and a low band signal to increase a coding efficiency in a portable terminal are provided. The apparatus includes a bandwidth extender for extracting auxiliary information relating to a characteristic of a high band signal using the high band signal and a low band signal and an encoder for encoding residual high band signal obtained by subtracting auxiliary information acquired from the low band signal from auxiliary information acquired from the high band signal.

Abstract: A coding device is provided with features in which optimum coding in a higher layer is flexibly carried out based on a coding result of a lower layer and a quality audio signal in limited circumstances is served to users. In this coding device, a basic layer coding unit codes an input signal to generate a basic layer information source code and outputs a linear prediction coefficient (LPC) and a quantum LPC, which are parameters calculated at coding, to an expanded layer control unit. A basic layer decoding unit decodes the basic layer information source code. An adding unit reverses a polarity of a basic layer decoded signal, adds the same to the input signal, and calculates a difference signal. The expanded layer control unit generates expanded layer mode information indicative of a coding mode in an expanded layer based on the LPC and the quantum LPC. An expanded layer coding unit codes the difference signal obtained from the adding unit under control of the expanded layer control unit.

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: 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 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: An audio encoder and decoder use architectures and techniques that improve the efficiency of quantization (e.g., weighting) and inverse quantization (e.g., inverse weighting) in 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 quantizes audio data in multiple channels, applying multiple channel-specific quantizer step modifiers, which give the encoder more control over balancing reconstruction quality between channels. The encoder also applies multiple quantization matrices and varies the resolution of the quantization matrices, which allows the encoder to use more resolution if overall quality is good and use less resolution if overall quality is poor. Finally, the encoder compresses one or more quantization matrices using temporal prediction to reduce the bitrate associated with the quantization matrices. An audio decoder performs corresponding inverse processing and decoding.

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: 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: A lossless audio codec segments audio data within each frame to improve compression performance subject to a constraint that each segment must be fully decodable and less than a maximum size. For each frame, the codec selects the segment duration and coding parameters, e.g., a particular entropy coder and its parameters for each segment, that minimizes the encoded payload for the entire frame subject to the constraints. Distinct sets of coding parameters may be selected for each channel or a global set of coding parameters may be selected for all channels. Compression performance may be further enhanced by forming M/2 decorrelation channels for M-channel audio. The triplet of channels (basis, correlated, decorrelated) provides two possible pair combinations (basis, correlated) and (basis, decorrelated) that can be considered during the segmentation and entropy coding optimization to further improve compression performance.

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 including level information corresponding to at least one object signal, converting the level information corresponding to the object signal to the level information corresponding to an output channel by applying a control parameter to the object parameter, and generating a rendering parameter including the level information corresponding to the output channel to control an object downmix signal resulting from downmixing the object signal.

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: 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: 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 electronic device includes a connector, at least two kinds of signal transmitting circuits, and a selecting system. The connector is capable of transmitting audio and video signals when connected to a peripheral device. The selecting system includes a detecting module, a memory module, a comparing module, and a connecting module. The detecting module is capable of detecting and measuring the voltage of a signal line of the connector. The memory module is capable of storing predetermined voltage ranges corresponding to different kinds of peripheral devices. The comparing module is capable of comparing the voltage of the signal line measured by the detecting module to the predetermined voltage ranges and determining what kind of peripheral device is connected to the connector. The connecting module is capable of connecting the connector to one of the signal transmitting circuits according to the comparing module.

Abstract: A method for the selection of synthesis units of a piece of information that can be decomposed into synthesis units, comprises at least the following steps for a considered information segment: determining the mean fundamental frequency value F0 for the information segment considered; selecting a sub-set of synthesis units defined as being the sub-set whose mean pitch values are the closest to the pitch value F0; applying one or more proximity criteria to the selected synthesis units to determine a synthesis unit representing the information segment.

Abstract: In one alternative, an audio signal is analyzed using multiple psychoacoustic criteria to identify a region of the signal in which time scaling and/or pitch shifting processing would be inaudible or minimally audible, and the signal is time scaled and/or pitch shifted within that region. In another alternative, the signal is divided into auditory events, and the signal is time scaled and/or pitch shifted within an auditory event. In a further alternative, the signal is divided into auditory events, and the auditory events are analyzed using a psychoacoustic criterion to identify those auditory events in which the time scaling and/or pitch shifting processing of the signal would be inaudible or minimally audible. Further alternatives provide for multiple channels of audio.

Abstract: A speech decoding device of the invention smoothes, in decoding speech signal in a voice-less period, RMS and filter coefficients which is discontinuously transmitted, and provides them to a synthesis filter. Thereby, it is capable of preventing discontinuous changing of the filter coefficient caused by the intermittent transmission of the filter coefficient. As a result, a quality of decoding can be improved. Also, to remove an effect, caused by the smoothing process, from the filter coefficients or the RMS which are transmitted in the past frames, a smoothing factor is adjusted not to perform smoothing while a certain time period (or a certain number of frames) from when a transition is made from a voice period from a voice-less period, or when a decoded feature parameter satisfies a predetermined condition.

Abstract: Disclosed is an audio data packet format for transmitting an MPEG-4 HE-AAC frame via a voice channel of a mobile communication network, a method for decoding the audio data packet format, a method for correcting a codec setup error by identifying a codec used to encode sound source data inserted into a data field of voice slot data, based on the sequence number of the voice slot data, and correcting the codec setup error when a codec set up in a mobile communication terminal is different from the codec used to encode the sound source data, and a mobile communication terminal adapted to correct a codec setup error.

Abstract: Playback by a decoder of a lossy compressed digital media file without quantization gaps is disclosed. The digital media file can be formed of a number of audio samples grouped into a corresponding number of audio frames. As a method, one embodiment can be carried out by identifying an encoder used to compress the digital media file; obtaining an encoder delay value for the identified encoder; obtaining a decoder delay value for the decoder; determining a audio sample count corresponding to a last valid audio sample; setting a re-synchronization after seek option marker N audio frames from the last valid audio sample; and decoding valid audio samples using the encoder delay value, the decoder delay value, and the sample count corresponding to the last valid audio sample.

Abstract: A system and method to change codec information to provide a coloring service in a Voice over Internet Protocol (VoIP) terminal uses different compression methods depending on a calling state and a call connecting state between the VoIP terminals so that a more efficient coloring service is provided. The system for changing codec information includes a gateway adapted to compress ring back tone data in a calling state and to compress voice signal data in a call connecting state between communication terminals according to preset different compression information and to transmit both data to a receiving terminal.

Abstract: Provided are an encoding method and apparatus for efficiently encoding a sinusoidal signal whose magnitude is less than a masking value according to a psychoacoustic model, a decoding method and apparatus for decoding an encoded sinusoidal signal, and a computer-readable recording medium having recorded thereon a program for executing the encoding method/the decoding method. By using a particular code indicating that the magnitude of a first sinusoidal signal is less than a masking value according to a psychoacoustic model to encode the first sinusoidal signal, difference coding for a third sinusoidal signal of a next frame, which is connected to the first sinusoidal signal, is performed using a sinusoidal signal or sinusoidal signals selected according to a method to use the particular code, and a decoding apparatus obtains a sum with a transmitted difference using the selected sinusoidal signal(s).

Abstract: An encoding method and apparatus and a decoding method and apparatus are provided. The decoding method includes extracting a down-mix signal and down-mix identification information from an input bitstream, determining, based on the down-mix identification information, whether the down-mix signal is a 3D down-mix signal obtained by performing a three-dimensional (3D) rendering operation, and if the down-mix signal is not 3D down-mix signal, generating a 3D down-mix signal by performing a 3D rendering operation. 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: In a multi-channel encoder generating several different parameter sets for reconstructing a multi-channel output signal using at least one transmission channel, the data stream is written such that the two parameter sets are decodable independently of each other. Thus, a multi-channel decoder is enabled to skip a parameter set which is marked as optional and/or has a higher version number when reading the data stream and still to perform a valid multi-channel reconstruction using a data set marked as mandatory or a data set having a sufficiently low version number. This achieves a flexible encoder/decoder concept suitable for future updates characterized by backward compatibility and reliability.

Abstract: A method, device and system for signal encoding and decoding, the method comprising: encoding a core layer signal to obtain a core layer signal code; selecting an enhancement sample point that requires enhancement layer signal encoding according to the core layer signal code and the number of bits that can be used by an enhancement layer; obtaining an enhancement layer signal code of the enhancement sample point; and outputting a bit stream, where the bit stream includes the core layer signal code and the enhancement layer signal code. In embodiments of the present invention, according to the number of bits that can be used by the enhancement layer, the enhancement sample point that requires enhancement layer signal encoding is selected; the enhancement layer signal of the selected enhancement sample point is encoded and decoded; when no sufficient bits are available for the enhancement layer, the enhancement quality of the core layer can be improved.

Abstract: Embodiments related to utilizing substantially optimal entries for a relatively low complexity dictionary for matching pursuits coding is disclosed. In various embodiments, methods are invoked for determining a substantially optimal entry from a bases dictionary comprising a plurality of entries; and utilizing the substantially optimal entry in a relatively low complexity matching pursuits data coding. In various embodiments, a system is provided comprising a bases dictionary comprising a plurality of entries each with a width of 15 or less; a signal to be coded; and a selection module configured to receive at least one of the plurality of entries from the bases dictionary, to calculate an inner product between the at least one of the plurality of entries and the signal to be coded, and to select the entry from the at least one of the plurality of entries that produces a maximum inner product for use in at least partially coding the signal to be coded.

Abstract: An encoding device and an encoding method are provided for encoding by reducing the number of samples to be processed when encoding higher-band spectrum data according to lower-band spectrum data in a wide-band signal. The device and the method can obtain a high-quality decoded signal even if a large quantization distortion is caused in the lower-band spectrum data. When encoding higher-band spectrum data in a signal to be encoded, according to lower-band spectrum data in the signal, only for a part (a head portion) of the higher-band spectrum data, the lower-band spectrum data after being quantized is subjected to approximate partial search and higher-band spectrum data is generated according to the search result.

Abstract: The disclosure relates to a communication terminal having a bandwidth expansion device for expanding the bandwidth of a narrowband voice signal, on a low-frequency and/or high-frequency side, by synthesizing at least one frequency band on the basis of the narrowband voice signal. A qualitatively satisfactory bandwidth expansion is thus performed using a plurality of net bit rates. The bandwidth expansion device is further connected to a memory containing a lookup table comprising at least one parameter value for the bandwidth expansion, for at least two net bit rates of the narrowband voice signal. A method for expanding a bandwidth of a narrowband voice signal having at least two net bit rates in a communication terminal is also disclosed herein.

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

Abstract: Storing audio data encoded in any of a plurality of different audio encoding formats is enabled by parametrically defining the underlying format in which the audio data is encoded, in audio format and packet table chunks. A flag can be used to manage storage of the size of the audio data portion of the file, such that premature termination of an audio recording session does not result in an unreadable corrupted file. This capability can be enabled by initially setting the flag to a value that does not correspond to a valid audio data size and that indicates that the last chunk in the file contains the audio data. State information for the audio data, to effectively denote a version of the file, and a dependency indicator for dependent metadata, may be maintained, where the dependency indicator indicates the state of the audio data on which the metadata is dependent.

Abstract: The present invention proposes a new method and a new apparatus for enhancement of audio source coding systems utilizing high frequency reconstruction (HFR). It utilizes a detection mechanism on the encoder side to assess what parts of the spectrum will not be correctly reproduced by the HFR method in the decoder. Information on this is efficiently coded and sent to the decoder, where it is combined with the output of the HFR unit.

Abstract: An audio decoder for providing a decoded audio information on the basis of an encoded audio information includes a window-based signal transformer configured to map a time-frequency representation, which is described by the encoded audio information, to a time-domain representation. The window-based signal transformer is configured to select a window, out of a plurality of windows including windows of different transition slopes and windows of different transform length, on the basis of a window information. The audio decoder includes a window selector configured to evaluate a variable-codeword-length window information in order to select a window for a processing of a given portion of the time-frequency representation associated with a given frame of the audio information.

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

Abstract: 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: Storing audio data encoded in any of a plurality of different audio encoding formats is enabled by parametrically defining the underlying format in which the audio data is encoded, in audio format and packet table chunks. A flag can be used to manage storage of the size of the audio data portion of the file, such that premature termination of an audio recording session does not result in an unreadable corrupted file. This capability can be enabled by initially setting the flag to a value that does not correspond to a valid audio data size and that indicates that the last chunk in the file contains the audio data. State information for the audio data, to effectively denote a version of the file, and a dependency indicator for dependent metadata, may be maintained, where the dependency indicator indicates the state of the audio data on which the metadata is dependent.

Abstract: A “STAC Codec” provides audio transcoding and decoding by processing an encoded audio signal using a backward-adaptive run-length Golomb-Rice (RLGR) decoder to recover transform coefficients of the encoded audio signal. The transform coefficients are then either transcoded in the transform domain to lossy or other formats, or decoded to the time domain by applying an inverse integer-reversible modulated lapped transform (MLT) to the recovered transform coefficients to recover an uncompressed time domain representation compressed audio signal. In additional embodiments, an inter-block spectral estimation and inverse data sorting strategy is used in recovering the transform coefficients from the encoded audio signal.

Abstract: To provide an acoustic signal processing apparatus which can reduce the amount of calculation in matrix arithmetic. An acoustic signal processing apparatus converts down-mixed acoustic signals of NI channels to acoustic signals of NO channels, where NO>NI.

Abstract: An audio encoder and decoder use architectures and techniques that improve the efficiency of quantization (e.g., weighting) and inverse quantization (e.g., inverse weighting) in 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 quantizes audio data in multiple channels, applying multiple channel-specific quantizer step modifiers, which give the encoder more control over balancing reconstruction quality between channels. The encoder also applies multiple quantization matrices and varies the resolution of the quantization matrices, which allows the encoder to use more resolution if overall quality is good and use less resolution if overall quality is poor. Finally, the encoder compresses one or more quantization matrices using temporal prediction to reduce the bitrate associated with the quantization matrices. An audio decoder performs corresponding inverse processing and decoding.

Abstract: In one embodiment, the method includes receiving the audio signal including a block of audio data partitioned into N sub-blocks, and restoring a plurality of code parameters s(0), s(1), . . . , s(N?1), respectively. The restoring step includes detecting s(0) from the audio signal, where s(0) represents the code parameter of the first sub-block; detecting a difference s(i)?s(i?1) from the audio signal for i=1, . . . N?1, where s(i) representing the code parameter of each sub-block following the first sub-block. The difference s(i)?s(i?1) is encoded by using first entropy code. The restoring step further includes calculating s(i) for i=1, . . . , N?1 using s(0) and the detected differences, and the method further includes decoding the N sub-blocks using the restored code parameters.

Abstract: There is described a method of encoding an input signal (20) to generate a corresponding encoded output signal (30), and also encoders (10) arranged to implement the method. The method comprises steps of: (a) distributing the input signal to sub-encoders (300, 310, 320) of the encoder (10); (b) processing the distributed input signal (20) at the sub-encoders (300, 310, 320) to generate corresponding representative parameter outputs (200, 210, 220) from the sub-encoders (300, 310, 320); and (c) combining the parameter outputs (200, 210, 220) of the sub-encoders (300, 310, 320) to generate the encoded output signal (30). Processing of the input signal (20) in the sub-encoders (300, 310, 320) involves segmenting the input signal (20) for analysis, such segments having associated temporal durations which are dynamically variable at least partially in response to information content present in the input signal (20).

Abstract: An entropy encoder includes an apparatus for producing a data stream which comprises two reference points, of code words of variable lengths, the apparatus comprising a first device for writing at least a part of a code word into the data stream in a first direction of writing, starting from a first reference point, and a second device for writing at least a part of a code word into the data stream in a second direction of writing, which is opposite to the first direction of writing, starting from the other reference point. In particular, when a raster having a plurality of segments is used to write the code words of variable lengths into the data stream, the number of the code words which can be written starting at raster points is doubled, in the best case, such that the data stream of code words of variable lengths is robust toward a propagation of sequence errors.

Abstract: A method of processing a decoded speech (DS) signal including successive DS frames, each DS frame including DS samples. The method comprises: adaptively filtering the DS signal to produce a filtered signal; gain-scaling the filtered signal with an adaptive gain updated once a DS frame, thereby producing a gain-scaled signal; and performing a smoothing operation to smooth possible waveform discontinuities in the gain-scaled signal.

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

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

Abstract: The present invention proposes a new method and a new apparatus for enhancement of audio source coding systems utilizing high frequency reconstruction (HFR). It utilizes a detection mechanism on the encoder side to assess what parts of the spectrum will not be correctly reproduced by the HFR method in the decoder. Information on this is efficiently coded and sent to the decoder, where it is combined with the output of the HFR unit.