Abstract: Disclosed is a stereo audio encoding device capable of reducing a bit rate. In this device, a stereo audio encoding unit (103) performs LPC analysis on an L channel signal and an R channel signal so as to obtain an L channel LPC coefficient and an R channel LPC coefficient. An LPC coefficient adaptive filter (105) obtains an LPC coefficient adaptive filter parameter to minimize the mean square error between the L channel LPC coefficient and the R channel LPC coefficient. An LPC coefficient reconfiguration unit (106) reconfigures the R channel LPC coefficient by using the L channel LPC coefficient and the LPC coefficient adaptive filter parameter. A route calculation unit (107) calculates a polynomial route indicating the safety of the R channel reconfigured LPC coefficient. A selection unit (108) selects and outputs the LPC coefficient adaptive filter parameter or the R channel LPC coefficient according to the safety of the R channel reconfigured LPC coefficient.

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: A fixed code book (FCB) search device simplifies an error minimizing process and reduces a calculation amount so as to prevent deterioration of a coding performance. The FCB search device (100) includes: a pulse shape convolution inverse filter (104) having an inverse feature of a pulse diffusion filter and supplied with an ideal residual signal; a pulse candidate preparatory selection unit (105) for pre-selecting a plurality of pulse candidates from the ideal residual signal to which the inverse filter is applied; and a pulse candidate final selection unit (109) for finally selecting one pulse from the selected candidates. By using this configuration, search is made for an algebra code book to which the pulse diffusion is applied.

Abstract: A frame loss compensating method wherein even when audio codec, which utilizes past sound source information of adaptive codebook or the like, is used as a main layer, the degradation in quality of the decoded audio of a lost frame and following frames is small. In this method, it is assumed that a pitch period ‘T’ and a pitch gain ‘g’ have been obtained as encoded information of a current frame. The sound source information of a preceding frame is expressed by use of a single pulse, and a pulse position ‘b’ and a pulse amplitude ‘a’ are used as encoded information for compensation. Then, an encoded sound source signal is a vector that builds up a pulse having an amplitude ‘a’ at a position that precedes by ‘b’ from the front position of the current frame. This vector is used as the content of the adaptive codebook, so that a vector, which builds up a pulse having an amplitude (g×a) at the position of the current frame (T?b), can be used as an adaptive codebook vector at the current frame.

Abstract: A technique is described for concealing the effect of a lost frame in a series of frames representing an encoded audio signal in a sub-band predictive coding system. In accordance with the technique, a first synthesized sub-band audio signal is synthesized, wherein synthesizing the first synthesized sub-band audio signal comprises performing waveform extrapolation based on a stored first sub-band decoded audio signal. A second synthesized sub-band audio signal is also synthesized, wherein synthesizing the second synthesized sub-band audio signal comprises performing waveform extrapolation based on the stored second sub-band decoded audio signal. The first synthesized sub-band audio signal and the second synthesized sub-band audio signal are combined to generate a synthesized full-band output audio signal corresponding to a lost frame.

Abstract: A method and apparatus for searching fixed codebook are provided. The method includes: obtaining a basic codebook which comprises position information of N pulses on M tracks, wherein N and M are positive integers; choosing n pulses as search pulses, wherein the n pulses are parts of the N pulses and n is a positive integer smaller than N; and replacing position information of the n search pulses respectively with other position information on the tracks to obtain a searched codebook; executing the search process for K times, wherein K is a positive integer larger than or equal to 2, at least two or more search pulses are chosen in one of the K search processes , and the chosen search pulses vary in each of the K search processes; and obtaining an optimal codebook from the basic codebook and the searched codebook according to a preset criterion.

Abstract: Codebook indices for a scalable speech and audio codec may be efficiently encoded based on anticipated probability distributions for such codebook indices. A residual signal from a Code Excited Linear Prediction (CELP)-based encoding layer may be obtained, where the residual signal is a difference between an original audio signal and a reconstructed version of the original audio signal. The residual signal may be transformed at a Discrete Cosine Transform (DCT)-type transform layer to obtain a corresponding transform spectrum. The transform spectrum is divided into a plurality of spectral bands, where each spectral band having a plurality of spectral lines. A plurality of different codebooks are then selected for encoding the spectral bands, where each codebook is associated with a codebook index. A plurality of codebook indices associated with the selected codebooks are then encoded together to obtain a descriptor code that more compactly represents the codebook indices.

Abstract: A scalable speech and audio codec is provided that implements combinatorial spectrum encoding. A residual signal is obtained from a Code Excited Linear Prediction (CELP)-based encoding layer, where the residual signal is a difference between an original audio signal and a reconstructed version of the original audio signal. The residual signal is transformed at a Discrete Cosine Transform (DCT)-type transform layer to obtain a corresponding transform spectrum having a plurality of spectral lines. The transform spectrum spectral lines are transformed using a combinatorial position coding technique. The combinatorial position coding technique includes generating a lexicographical index for a selected subset of spectral lines, where each lexicographic index represents one of a plurality of possible binary strings representing the positions of the selected subset of spectral lines. The lexicographical index represents non-zero spectral lines in a binary string in fewer bits than the length of the binary string.

Abstract: Encoding and decoding apparatuses and encoding and decoding methods are provided. The decoding method includes extracting a plurality of encoded signals from an input bitstream, determining which of a plurality of decoding methods is to be used to decode each of the encoded signals, decoding the encoded signals using the determined decoding methods, synthesizing the decoded signals, and restoring an original signal by performing a post-processing operation on the single signal. Accordingly, it is possible to encode signals having different characteristics at an optimum bitrate by classifying the signals into one or more classes according to the characteristics of the signals and encoding each of the signals using an encoding unit that can best serve the class where a corresponding signal belongs. In addition, it is possible to efficiently encode various signals including audio and speech signals.

Abstract: Provided are methods and apparatuses for more efficiently encoding and decoding a high frequency band signal which is from an audio signal and which is greater than a predetermined threshold frequency. The method and apparatus for encoding the audio signal encodes a linear prediction coding (LPC) coefficient and gain information of a residual signal, which are generated by performing LPC analysis, thereby encoding a high frequency signal so as to have enhanced sound quality, while using less bits.

Abstract: Provided are a method and apparatus for effectively encoding/decoding remaining difference signals excluding sinusoidal components, from input audio signals. In the method and apparatus for encoding audio signals, sinusoidal analysis is performed on low frequency signals of less than a predetermined critical frequency in order to extract sinusoidal signals and then, an encoding operation is performed on the remaining difference signals excluding the sinusoidal signals, from input audio signals, by using linear prediction coding (LPC) analysis.

Abstract: Embodiments related to recording and retrieving of voice data signals are described and depicted.

Abstract: A method and apparatus for processing a signal compressed in accordance with a specific alternative coding scheme are disclosed. In detail, a coding method for signal compression and signal restoration using a specific alternative coding scheme, and an apparatus therefor are disclosed. Data coding and entropy coding according to the present invention are executed under the condition in which they have a co-relation with each other. The method for signal processing includes 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: Disclosed is a wideband audio signal coding/decoding device and method that may code a wideband audio signal while maintaining a low bit rate. The wideband audio signal coding device includes an enhancement layer that extracts a first spectrum parameter from an inputted wideband signal having a first bandwidth, quantizes the extracted first spectrum parameter, and converts the extracted first spectrum parameter into a second spectrum parameter; and a coding unit that extracts a narrowband signal from the inputted wideband signal and codes the narrowband signal based on the second spectrum parameter provided from the enhancement layer, wherein the narrowband signal has a second bandwidth smaller than the first bandwidth. The wideband audio signal coding/decoding device and method may code a wideband audio signal while maintaining a low bit rate.

Abstract: A prediction performance between the individual channels of a stereo signal is improved to improve the sound quality of a decoded signal. An LPF (101-1) interrupts the high-range component of an S1, and outputs an S1? (a low-range component). An LPF (101-2) interrupts the high-range component of an S2, and outputs an S2? (a low-range component). A prediction unit (102) predicts the S2? from the S1?, and outputs a prediction parameter composed of a delay time difference (t) and an amplitude ratio (g). A first channel encoding unit (103) encodes the S1. A prediction parameter encoding unit (104) encodes the prediction parameter. The encoded parameters of the encoded parameter of the S1 and the prediction parameter are finally outputted.

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: A sound encoding device for efficiently encoding stereophonic sound. In this sound encoding device, a prediction parameter analyzing section (21) determines the delay difference D and the amplitude ratio g of a first-channel sound signal with respect to a second-channel sound signal as channel-to-channel prediction parameters from a first-channel decoded signal and a second-channel sound signal, a prediction parameter quantizing section (22) quantizes the prediction parameters, a signal predicting section (23) predicts a second-channel signal by using the first decoded signal and the quantization prediction parameters. The prediction parameter quantizing section (22) encodes and quantizes the prediction parameters (the delay difference D and the amplitude ratio g) by using the relationship (correlation) between the delay difference D and the amplitude ratio g attributed to the spatial characteristic (e.g., distance) from the sound source of the signal to the receiving point.

Abstract: In one embodiment, the method includes receiving audio frame data having at least first and second channel data. The first and second channel data include a plurality of blocks, where the blocks are classified by a block type. The first and second channel data are provided jointly if the first and second channel data are paired with each other. The embodiment further includes obtaining block information indicating the block type, and lossless decoding the first and second channel data based on the block information.

Abstract: In one embodiment, the method includes receiving the audio signal including at least one block of audio data and configuration information, and reading coding type information and partitioning information from the configuration information. The coding type information indicates an entropy coding scheme used in encoding the audio signal, and the partitioning information indicates a sub-block partition scheme by which the block is divided into sub-blocks. Sub-block information is read from the block of audio data, and the sub-block information indicates a number of the sub-blocks into which the block is partitioned given the sub-block partitioning scheme. The number of the sub-blocks is determined based on the entropy coding scheme and the sub-block partition scheme. The partitioned sub-blocks are decoded based on the entropy coding scheme.

Abstract: In one embodiment, the method includes receiving audio frame data having at least first and second channel data. The first and second channel data includes a plurality of blocks, where the blocks are classified by a block type. The embodiment further includes obtaining frame length information indicating a length of the audio frame data, and obtaining block information indicating the block type. The block information corresponds to the first and second channel data being common when the first and second channel data are paired. The first and second channel data are lossless decoded based on the frame length information and the block information.

Abstract: A scalable encoding apparatus capable of suppressing the quality degradation of a decoded signal without increasing the bit rate. In this apparatus, a core layer encoding part (101) and an extended layer encoding part (102) encode an input signal for each of audio frames. When a replacement determining part (103) determines that a degree to which the input signal changes between a preceding frame and a current frame is equal to or greater than a predetermined value or that a degree, to which the quality of the decoded signal is improved by an extended layer encoding process in the preceding frame, is equal to less than a predetermined level, a replacing part (105) replaces a part of an extended layer encoded data of the preceding frame by a core layer encoded data of the current frame. That is, a transmitting part (108) transmits, as a backup, the core layer encoded data of the current frame to a decoding end in advance.

Abstract: Provided is an audio encoding device for modeling a spectrum waveform and accurately restoring the spectrum waveform. The audio encoding device includes: an FFT unit (104) for subjecting a spectrum amplitude of a drive sound source signal to an FFT process to obtain an FFT transform coefficient; a second spectrum amplitude calculation unit (105) for calculating a second spectrum amplitude of the FFT transform coefficient; a peak point position identification unit (106) for identifying the positions of the most significant N peaks of the second spectrum amplitude; a coefficient selection unit (107) for selecting FFT transform coefficients corresponding to the identified positions; and a quantization unit (108) for quantizing the selected FFT transform coefficients.

Abstract: A hybrid sinusoidal/pulse excitation encoder has been recently proposed for constructing a scalable audio encoder The base layer consisting of data supplied by the sinusoidal encoder retains the main features of the input signal achieving medium to high quality audio at a very low bit rate. Quality can be further enhanced by adding excitation signal layers associated with a decreasing decimation that increasingly model more subtle aspects of the original signal. The invention provides a method of mixing the different excitation signal layers so that the full concept of scalability is realised without compromising the quality of the encoded signals. The mixing is controlled via a quality parameter that weights the significance of previous layers when constructing a new higher layer.

Abstract: Methods, encoders, and digital systems are provided for predictive encoding of speech parameters in which an input frame is encoded by quantizing a parameter vector of the input frame with a strongly-predictive codebook and a weakly-predictive codebook to obtain a strongly-predictive distortion and a weakly-predictive distortion, adjusting a correlation indicator based on a relative correlation of the input frame to a previous frame, wherein the correlation indicator is indicative of the strength of the correlation of previously encoded frames, and encoding the input frame with the weakly-predictive codebook unless the correlation indicator has reached a correlation threshold.

Abstract: A method and apparatus to convert a linear predictive coding (LPC) coefficient into a coefficient having order characteristics, such as a line spectrum frequency (LSF), and to vector quantize the coefficient having the order characteristics when a speech signal is encoded. The method and apparatus split the vector of the coefficient having the order characteristics into a plurality of subvectors, select a codebook in which an available bit is variably allocated to each subvector according to distribution of elements of each subvector, and quantize each subvector according to the selected codebook. The method and apparatus use normalized codebooks.

Abstract: A speaker of encoded speech data recorded in a semiconductor storage device in an IC recorder is to be retrieved easily. An information receiving unit 10 in a speaker retrieval apparatus 1 reads out the encoded speech data recorded in a semiconductor storage device 107 in an IC recorder 100. A speech decoding unit 12 decodes the encoded speech data. A speaker frequency detection unit 13 discriminates the speaker based on a feature of the speech waveform decoded to find the frequency of conversation (frequency of occurrence) of the speaker in a preset time interval. A speaker frequency graph displaying unit 14 displays the speaker frequency on a picture as a two-dimensional graph having time and the frequency as two axes.

Abstract: A method and apparatus multiplies a past sample a time lag ? older than a current sample by a quantized multiplier ?? on a frame by frame basis, subtracts the multiplication result from the current sample, codes the subtraction result, and codes the time lag using a fixed-length coder if the multiplier ?? is smaller than 0.2 or if information about the previous frame is unavailable, or codes the time lag using a variable-length coder if ?? is not smaller than 0.2. A multiplier ? is coded by a multiplier coder and the multiplier ?? obtained by decoding the multiplier ? is outputted. The process is performed for each frame.

Abstract: Lost frame reconstruction is described. A previous good or reconstructed frame may be analyzed to determine a category for the lost frame. A percentage Pi may be associated with the determined category of the lost frame. A top Pi percent magnitude samples may be zeroed out in an excitation of the previous good or reconstructed frame to produce a reconstruction excitation. The reconstruction excitation may be applied to one or more linear prediction coefficients for the previous good or reconstructed frame to generate a reconstructed frame.

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: A signal processing apparatus includes a decoding unit, an analyzing unit, a synthesizing unit, and a selecting unit. The decoding unit decodes an input encoded audio signal and outputs a playback audio signal. When loss of the encoded audio signal occurs, the analyzing unit analyzes the playback audio signal output before the loss occurs and generates a linear predictive residual signal. The synthesizing unit synthesizes a synthesized audio signal on the basis of the linear predictive residual signal. The selecting unit selects one of the synthesized audio signal and the playback audio signal and outputs the selected audio signal as a continuous output audio signal.

Abstract: Even when a combination of the stegonography technique and prediction encoding is applied to sound encoding, a sound encoder does not cause deterioration in quality of decoded signals. In the device, an encoding section (102) outputs an encoding code (I) to a bit embedding section (104). A function extension encoding section (103) generates an encoding code (J) for information required for extending functions of the sound encoder (100) and outputs it to the bit embedding section (104). The bit embedding section (104) embeds information on the encoding code (J) into a part of bits of the encoding code (I) and outputs the resultant encoding code (I?). A synchronization information generating section (106) generates synchronization information according to the encoding code (I?) after the bit embedding and outputs the synchronization information to the encoding section (102).

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.

Abstract: The invention enables the inclusion of voice and remaining audio information at different parts of the audio production process. In particular, the invention embodies special techniques for VRA-capable digital mastering, accommodation of PCPV/PCA and/or SCRA signals in audio CODECs, VRA-capable encoders and decoders, and VRA in DVD and other digital audio file formats. The invention facilitates an end-listener's voice-to-remaining audio (VRA) adjustment upon the playback of digital audio media formats by focusing on new configurations of multiple parts of the entire digital audio system, thereby enabling a new technique intended to benefit audio end-users (end-listeners) who wish to control the ratio of the primary vocal/dialog content of an audio program relative to the remaining portion of the audio content in that program.

Abstract: Herein disclosed is an acoustic signal encoding device comprising: a coefficient table (17) having described therein coefficients representable in the form of a matrix with 2 rows by N columns simulating head-related transfer characteristics to be applied when a multi-channel signal is reproduced, a first signal outputting unit (12) for downmixing a N-channel frequency domain signal to have a 2-channel downmixed signal outputted therethrough in accordance with the coefficient table (17), and a second signal outputting unit (14) for generating subsidiary information to be used to reconstruct a multi-channel signal based on the 2-channel downmixed signal, thereby making it possible for the downmixed signal to be filtered in accordance with a desired transfer function, and thus enabling the acoustic signal decoding device to reproduce the original multi-channel spatial information simply by reproducing the first coded signal, and the original multi-channel signal by reproducing the first coded signal with the ai

Abstract: An audio reproduction circuit has an input buffer for holding MP3 data. The audio reproduction circuit also has an MP3 decoder for reading and decoding, in certain processing units, the MP3 data held in the input buffer, and generating voice data. The audio reproduction circuit also has an output buffer for holding the voice data supplied from the MP3 decoder. The audio reproduction circuit also has a digital-analog converter for reading the voice data from the output buffer in synchronization with a clock signal, and converting the voice data to an audio signal. A cut-out detector is provided for generating a cut-out detection (prediction) signal on the basis of the quantity of MP3 data remaining in the input buffer and the quantity of voice data remaining in the output buffer. The cut-out signal indicates a timing at which MP3 data is to be introduced to the input buffer.

Abstract: A speech coding apparatus includes a base layer coder that codes an input signal and generates first coded information. A base layer decoder decodes the first coded information and generates a first decoded signal. The base layer decoder also generates long term prediction information comprising information representing long term correlation of speech or sound. An adder obtains a residual signal representing a difference between the input signal and the first decoded signal. An enhancement layer coder calculates a long term prediction coefficient using the residual signal obtained in the adder and a long term prediction signal fetched from a previous long term prediction signal sequence based on the long term prediction information. The enhancement layer coder further codes the long term prediction coefficient and generates second coded information.

Abstract: Applications of dim-and-burst techniques to coding of wideband speech signals are described. Reconstruction of a highband portion of a frame of a wideband speech signal using information from a previous frame is also described.