Abstract: A coding apparatus reduces a circuit scale and the amount of coding processing calculation. A frequency domain conversion section performs a frequency analysis of the signal sampled at a sampling rate Fx with an analysis length of 2·Na and calculates first spectrum S1(k)(0?k<Na). A band extension section extends the effective frequency band of first spectrum S1(k) to 0?k<Nb so that a new spectrum can be assigned to the extended area following to the frequency k=Na of first spectrum S1(k). An extended spectrum assignment section assigns extended spectrum S1?(k)(Na?k<Nb) input to the extended frequency band from the outside. A spectral information specification section outputs information necessary to specify extended spectrum S1?(k) out of the spectrum given from the extended spectrum assignment section as a code.

Abstract: An audio signal band expanding apparatus (100a) includes a harmonic generator (3) that receives an input audio signal having a predetermined band and generates, based on the input audio signal, harmonic signals, and an adder (2) that adds the harmonic signals generated by the harmonic generator (3) to the input audio signal. The harmonic generator (3) simulates the input-output characteristics of a predetermined amplifier or that of a device to generate the harmonic signals from the input audio signal.

Abstract: A coding method, a decoding method, a coder, and a decoder are disclosed herein. A coding method includes: obtaining the pulse distribution, on a track, of the pulses to be encoded on the track; determining a distribution identifier for identifying the pulse distribution according to the pulse distribution; and generating a coding index that includes the distribution identifier. A decoding method includes: receiving a coding index; obtaining a distribution identifier from the coding index, wherein the distribution identifier is configured to identify the pulse distribution, on a track, of the pulses to be encoded on the track; determining the pulse distribution, on a track, of all the pulses to be encoded on the track according to the distribution identifier; and reconstructing the pulse order on the track according to the pulse distribution.

Abstract: Provided are, among other things, systems, methods and techniques for decoding an audio signal from a frame-based bit stream. Each frame includes processing information pertaining to the frame and entropy-encoded quantization indexes representing audio data within the frame. The processing information includes: (i) code book indexes, (ii) code book application information specifying ranges of entropy-encoded quantization indexes to which the code books are to be applied, and (iii) window information. The entropy-encoded quantization indexes are decoded by applying the identified code books to the corresponding ranges of entropy-encoded quantization indexes. Subband samples are then generated by dequantizing the decoded quantization indexes, and a sequence of different window functions that were applied within a single frame of the audio data is identified based on the window information.

Abstract: A stream of raw acoustic data can be received at a client device. The client device can frame the stream of raw acoustic data at particular intervals with alignment information to create framed acoustic data, and buffer the framed acoustic data while waiting for a data request from a host device. In response to receiving the data request, the client device can provide the framed acoustic data to the host device.

Abstract: Methods and apparatus are disclosed for approximating an MDCT coefficient of a block of windowed sinusoid having a defined frequency, the block being multiplied by a window sequence and having a block length and a block index. A finite trigonometric series is employed to approximate the window sequence. A window summation table is pre-computed using the finite trigonometric series and the defined frequency of the sinusoid. A block phase is computed for each block with the defined frequency, the block length and the block index. An MDCT coefficient is approximated by the dot product of a phase vector computed using the block phase with a corresponding row of the window summation table.

Abstract: A coding apparatus reduces a circuit scale and the amount of coding processing calculation. A frequency domain conversion section performs a frequency analysis of the signal sampled at a sampling rate Fx with an analysis length of 2·Na and calculates first spectrum S1(k)(0?k<Na). A band extension section extends the effective frequency band of first spectrum S1(k) to 0?k<Nb so that a new spectrum can be assigned to the extended area following to the frequency k=Na of first spectrum S1(k). An extended spectrum assignment section assigns extended spectrum S1?(k)(Na?k<Nb) input to the extended frequency band from the outside. A spectral information specification section outputs information necessary to specify extended spectrum S1?(k) out of the spectrum given from the extended spectrum assignment section as a code.

Abstract: A voice communication apparatus includes a communication portion that receives a plurality of frames including at least a first frame having first voice data and a second frame having second voice data subsequent to the first frame, the first voice data and the second voice data being encoded by a predetermined encoding system, a decoding portion that decodes the first voice data and the second voice data received by the communication portion, a buffer that retains the first voice data and the second voice data decoded by the decoding portion, a calculation portion that calculates an amplitude envelope based on the first voice data decoded by the decoding portion, and a controlling portion that judges whether or not the second voice data decoded by the decoding portion exceeds the amplitude envelope and corrects the second voice data that exceeds the amplitude envelope.

Abstract: An audio signal band expanding apparatus (100a) includes a harmonic generator (3) that receives an input audio signal having a predetermined band and generates, based on the input audio signal, harmonic signals, and an adder (2) that adds the harmonic signals generated by the harmonic generator (3) to the input audio signal. The harmonic generator (3) simulates the input-output characteristics of a predetermined amplifier or that of a device to generate the harmonic signals from the input audio signal.

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 technique is described herein for reducing audible artifacts in an audio output signal generated by decoding a received frame in a series of frames representing an encoded audio signal in a predictive coding system. In accordance with the technique, it is determined if the received frame is one of a predefined number of received frames that follow a lost frame in the series of the frames. Responsive to determining that the received frame is one of the predefined number of received frames, at least one parameter or signal associated with the decoding of the received frame is altered from a state associated with normal decoding. The received frame is then decoded in accordance with the at least one parameter or signal to generate a decoded audio signal. The audio output signal is then generated based on the decoded audio signal.

Abstract: In one embodiment, the method includes receiving an audio data frame having at least first and second channels. The first and second channels are synchronously subdivided into blocks such that the lengths of the blocks into which the second channel is subdivided correspond to the lengths of the blocks into which the first channel is subdivided if the first and second channels are correlated with each other and difference coding is used. The first and second channels are decoded and the subdivided blocks of the first and second channels are interleaved if the first and second channels are synchronously subdivided.

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: In one embodiment, at least first and second channels in a frame of the audio signal are independently subdivided into blocks if the first and second channels are not correlated with each other. At least two of the blocks have different block lengths. Furthermore, the first and second channels are correspondingly subdivided into blocks such that the lengths of the blocks into which the second channel is subdivided correspond to the lengths of the blocks into which the first channel is subdivided if the first and second channels are correlated with each other. At least two of the blocks have different block lengths.

Abstract: The invention is based on the idea of providing a method for high-resolution, waveform-preserving digitization of analog signals, wherein conventional scalar logarithmic quantization is transferred to multi-dimensional spherical coordinates, and the advantages resulting from this, e.g., a constant signal/noise ratio over an extremely high dynamic range with very low loss with respect to the rate-distortion theory. In order to make use of the statistical dependencies present in the source signal for an additional gain in the signal/noise ratio, the differential pulse code modulation (DPCM) is combined with spherical logarithmic quantization. The resulting method achieves an effective data reduction with a high long-term and short-term signal/noise ratio with an extremely small signal delay.

Abstract: In one embodiment, the method includes receiving an audio data frame having at least first and second channels. The first and second channels are independently subdivided into blocks if the first and second channels are not correlated with each other. The first and second channels are decoded, and the subdivided blocks of the first and second channels are not interleaved if the first and second channels are independently subdivided.

Abstract: Provided are, among other things, systems, methods and techniques for decoding an audio signal from a frame-based bit stream. Each frame includes processing information pertaining to the frame and entropy-encoded quantization indexes representing audio data within the frame. The processing information includes: (i) code book indexes, (ii) code book application information specifying ranges of entropy-encoded quantization indexes to which the code books are to be applied, and (iii) window information. The entropy-encoded quantization indexes are decoded by applying the identified code books to the corresponding ranges of entropy-encoded quantization indexes. Subband samples are then generated by dequantizing the decoded quantization indexes, and a sequence of different window functions that were applied within a single frame of the audio data is identified based on the window information.

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: An interrupt sensitive extract byte instruction scheme is presented herein. The interrupt sensitive extract by instruction extracts bytes from data, depending on the presence of an interrupt. The extract byte instruction extracts bytes from data in the absence of the interrupt and does not extract bytes in the presence of the interrupt. The interrupt can be triggered by a set of counters that count the number of extracted bytes. By loading the counters with a particular number, the interrupt can be generated when the particular number of data bytes is extracted.

Abstract: A DTMF signal processing apparatus of the present invention comprises a data divider unit, a DTMF signal component analyzer unit, a weighting processing unit, a buffer, a DTMF signal erasure determination unit, and a DTMF signal erasure processing unit. The data divider unit divides speech data into a plurality of divided speech data, and the DTMF signal component analyzer unit analyzes whether or not the divided speech data has a DTMF signal component. The weighting processing unit applies a weighting value to divided speech data analyzed at this time and stores the resultant speech data in the buffer, and also applies a weighting value to past divided speech data previously stored in the buffer when the result of the analysis indicates that the analyzed divided speech data has the DTMF signal component. The DTMF signal erasure determination unit determines based on the weighting value whether or not to erase the divided speech data stored in the buffer.

Abstract: A scalable audio data arithmetic decoding method, medium, and apparatus, and a method, medium, and apparatus truncating an audio data bitstream. The arithmetic decoding method of decoding a scalable arithmetic coded symbol may include arithmetic decoding of a symbol by using the symbol and a probability value for the symbol desired to be decoded, and determining whether or not to continue decoding by checking an ambiguity indicating whether or not decoding of the symbol to be decoded is completed. According to a method, medium, and apparatus of the present invention, data to which scalability is applied when arithmetic coding is performed in MPEG-4 scalable lossless audio coding can be efficiently decoded. Even when a bitstream is truncated, a decoding termination point can be known such that additional decoding of the truncated part can be performed.

Abstract: The invention provides an apparatus and a method for performing a calculation operation with at least one input signal consisting of signal sections, wherein each signal section of said input signal has a constant amplitude. The apparatus comprises a signal transformation unit for transforming at least one input signal into a first intermediary signal having a virtual amplitude with respect to at least one carrier signal. The calculation unit is provided for performing the calculation operation on said first intermediary signal to generate a second intermediary signal. A signal re-transformation unit re-transforms the second intermediary signal into an output signal consisting of signal sections, wherein each signal section of said output signal has a constant amplitude.

Abstract: A coding apparatus capable of reducing a circuit scale and also reducing the amount of coding processing calculation is disclosed. In this apparatus, frequency domain conversion section (103) performs a frequency analysis of the signal sampled at a sampling rate Fx with an analysis length of 2·Na and calculates first spectrum S1(k) (0?k<Na). Band extension section (104) extends the effective frequency band of first spectrum S1(k) to 0?k<Nb so that a new spectrum can be assigned to the extended area following to the frequency k=Na of first spectrum S1(k). Extended spectrum assignment section (105) assigns extended spectrum S1?(k) (Na?k<Nb) input to the extended frequency band from outside. Spectral information specification section (106) outputs information necessary to specify extended spectrum S1?(k) out of the spectrum given from extended spectrum assignment section (105) as a code.

Abstract: A system extends a bandwidth of bandlimited audio signals by analyzing bandlimited audio signals at a transmission cycle rate. The analyzer may obtain a bandlimited parameter at a transmission cycle rate. A mapping device or logic in the system obtains a wideband parameter based on the bandlimited parameter. An audio signal generator generates a highband and/or lowband audio signal based on the wideband parameter at the transmission cycle rate. In some systems, the bandlimited audio signal is analyzed at the transmission cycle rate. The highband and/or lowband audio signals and the combined wideband audio signal are generated at the transmission cycle rate.

Abstract: A low bit rate digital audio coding system includes an encoder which assigns codebooks to groups of quantization indexes based on their local properties resulting in codebook application ranges that are independent of block quantization boundaries. The invention also incorporates a resolution filter bank, or a tri-mode resolution filter bank, which is selectively switchable between high and low frequency resolution modes or high, low and intermediate modes such as when detecting transient in a frame. The result is a multichannel audio signal having a significantly lower bit rate for efficient transmission or storage. The decoder is essentially an inverse of the structure and methods of the encoder, and results in a reproduced audio signal that cannot be audibly distinguished from the original signal.

Abstract: An uplink or downlink audio processor contains a multi band compressor that receives an input, uplink or downlink, audio signal. The multi-band compressor has a band splitter that splits the input audio signal into a number of different band signals. Each band signal is input to a respective compressor block, which is independently programmable so that its audio frequency response (a) differs from a linear response in at least two non-overlapping windows of its input signal, and (b) differs from the frequency response of another one of the compressor blocks. Other embodiments are also described and claimed.

Abstract: A stochastic codebook associates a pulse position of a predetermined channel with a pulse position of another channel, searches for a pulse position by means of a predetermined algorithm, and outputs a code combining a found pulse position with a polarity code to an excitation vector creation section as a stochastic excitation vector code. By this means, it is possible to secure variations so that there are no positions where there is no pulse at all while achieving a reduction of the number of bits used when coding stochastic codebook pulses in order to attain a lower bit rate.

Abstract: One or more methods and systems of detecting or identifying one or more types of algorithms used in the encoding of a voice or speech waveform is presented. The system and method may be used as a testing tool to identify whether a voice data stream is encoded using a linear G.711, ?-law G.711, or A-law G.711 algorithm. The system and method are applied to a voice data stream to ensure that a codec with the appropriate algorithm is used to reproduce an audio waveform.

Abstract: Described is a system and method for communicating using a telecommunications device which includes an automatic speech recognition (ASR) system. The ASR system which is capable of recognizing at least one prestored command from a voice input to the telecommunication device is activated. The ASR system compares the voice input to the at least one prestored command. When at least a first portion of the voice input matches the prestored command, the ASR system reviews a second portion of the voice input to determine a parameter value. A tone system generates a tone frequency corresponding to the parameter value.

Abstract: A digital recording apparatus for recording 1-bit digital audio data of a first sampling frequency on a recording medium in accordance with the recording format of multi-bit PCM data of a second sampling frequency includes a storage section to which input 1-bit digital audio data of the first sampling frequency is written; an encoder configured to read, from the storage section, the 1-bit digital audio data at a clock synchronized with the second sampling frequency and configured to convert the 1-bit digital audio data in such a manner that bits of the 1-bit digital audio data are arrayed in a 1-bit data area provided in the multi-bit PCM data that is in accord with the recording format; and a recorder configured to record data output from the encoder on the recording medium in accordance with the recording format.

Abstract: A method, apparatus and computer memory are provided for generating an audio fingerprint of an audio recording. A memory stores stable frequency family data corresponding to a plurality of stable frequency families. A processor curve fits audio recording data to at least one of the stable frequency families, extracts at least one variation from the curve fitted audio recording data, and creates the audio fingerprint of the audio recording from the at least one variation.

Abstract: A digital wireless communications device enabling call distance to be increased while avoiding mute processing as far as possible, even if code error occurs in an ADPCM code. The digital wireless communications device may be configured to include: an error detector for detecting code error in an ADPCM code received via a wireless circuit; an ADPCM decoder for generating a PCM signal by decoding the ADPCM code; and a substitution unit for determining that a click noise is generated if the high-speed scale factor and the low-speed scale factor determined by the ADPCM decoder within a predetermined time period after the error detector has judged that there is a code error in the ADPCM code, and the received ADPCM code itself, respectively exceed prescribed threshold values, and for substituting the ADPCM code with a predetermined prescribed code.

Abstract: The present invention discloses a method and a system for converting vocal sounds into digital data format. This technique will significantly decrease the amount of memory needed to store the digital data of the recorded voice. The system is comprised of a microphone converting the vocal sound signals into electrical signal, amplifying and filtering module for analyzing the electrical signals, a comparator module for comparing the analog signal to pre-defined value and sampling by clock edge module for representing the output signal of the comparator as a digital data format, a memory module for storing said digital data, a filtering module for reducing the alternating the analog signal higher harmonics, an amplifying module increasing the filtered signals amplitude and transducer module for converting the electrical amplifying signals into vocal sound signal.

Abstract: The invention utilizes low complexity estimates of complex functions to perform combinatorial coding of signal vectors. The invention disregards the accuracy of such functions as long as certain sufficient properties are maintained. The invention in turn may reduce computational complexity of certain coding and decoding operations by two orders of magnitude or more for a given signal vector input.

Abstract: The invention relates to a method for transcoding audio signals in a communications system. In order to improve the inter-operability between units (2,40) capable of handling wideband audio signals and units (3,46) or network components (50) capable of handling narrowband audio signals, it is proposed that first, an audio signal is received in a network element (42) of a communications network via which said audio signal is transmitted. Next, it is determined in said network element (42) whether a transcoding of the received audio signal is required. In case a narrowband-to-wideband transcoding of the received signal is required, the received narrowband audio signal is transcoded into a wideband audio signal in the network element (1,42). The generated wideband audio signal is then forwarded to the receiving terminal (2,40). The invention equally relates to a corresponding communications system and its components.

Abstract: A DPCM operation which can reduce the size of differential data and a method and an apparatus for encoding data using the DPCM operation are provided. A method for generating differential data includes generating differential data by performing a DPCM operation on quantized data and generating predicted differential data by performing a predicted DPCM operation on the quantized data, generating circular-quantized differential data and circular-quantized predicted differential data by performing a circular quantization operation on the differential data and the predicted differential data so as to reduce their ranges, and selecting one of the circular-quantized differential data and the circular-quantized predicted differential data depending on their magnitudes.

Abstract: A prediction signal calculator with a limit function is provided with a multiplier calculating a partial prediction signal composed of the product of a polar prediction coefficient for generating a regenerative signal and a quantized regenerative signal, a display conversion section for converting the partial prediction signal from floating point representation to an absolute value display, and a limiter executing processing for substituting limit values in the partial prediction signal satisfying overflow conditions during conversion of the partial prediction signal from floating point representation to an absolute value display in the event that the error detector determines that there are code errors in the audio data for a predetermined number of frames of the audio data.

Abstract: Embodiments of the invention achieve increased compression of audio data in comparison to prior art ADPCM compression schemes using modest processing power and resources. For one embodiment, an asymmetric ADPCM encoding scheme is implemented to increase apparent encoding resolution for a specified number of encoding bits. Additionally, or alternatively, such techniques as pattern recognition and encoding, as well as calculation simplification, are employed to increase data compression of audio data.

Abstract: A method of compressing sounds in mobile terminals according to the present invention transforms pulse code modulations (PCM) codes, which are source data of bell sounds using recorded sounds or voice memos and are generated by sampling the sounds, through applying a differential method and, then, compresses the PCM codes using Lempel Ziv Welch (LZW) compresses technique, thus reducing a storage space required for storing bell sounds using sounds or voice memos in mobile terminals. According to the present invention, compression efficiency is maximized upon using LZW algorithm by transforming PCM code through applying differential method, thereby increasing restoration efficiency of original sounds and heightening compression efficiency by about 50%, compared with the existing compression storage method using ADPCM.

Abstract: A call setup procedure is presented to permit vocoder bypass, which will allow the transmission of wideband speech packets between wideband terminals over narrowband transmission constraints. In addition, methods and apparatus are presented that allow the conversion between a wideband tandem-free operation, a narrowband tandem-free operation, and a standard tandem operation.

Abstract: A digital microphone system and a method thereof are described. The digital microphone comprises a receiving unit, an amplifier, a modulator, an audio processing unit and a high definition audio link. After the receiving unit receives an analog audio signal, the analog audio signal is amplified by the amplifier. The analog audio signal is then converted into a first digital data by the modulator. The high definition audio link is set between the modulator and the audio processing unit for transmitting the first digital data. The audio processing unit comprises a digital filter for converting the first digital data into a second digital data. The audio processing unit and the digital filter are integrated and implemented by software.

Abstract: An apparatus for decoding encoded voice data comprises a demodulator (101) which demodulates the encoded voice data (RF) and provides a demodulated encoded voice data (APO, RD), an adaptive differential pulse code modulation decoder (102) which decodes the demodulated encoded voice data and provides a pulse code modulation data (PO), an error detector (103) which detects whether error is present in the encoded voice data based on the demodulated encoded voice data and outputs a detection result (CRCERR) and a limiter (104) which outputs either the pulse code modulation data (POL) or a limit data (POL) in accordance with the detection result (CRCERR).

Abstract: An audio encoder includes a time-frequency mapping block, a psychoacoustic model block, a middle/side (M/S) encoding block, a parameter calculation block, a bit allocation and quantization block and a bitstream formatting block. The encoder is forced to operate in M/S mode for reducing the calculation time of the parameter used for bit allocation, quantization and encoding. In addition, the calculation of the parameter only needs to consider the middle and side channels but not the left and right channels, thus the complexity of the psychoacoustic model for analyzing the input audio signal can be reduced.

Abstract: The present invention relates to a new method for enhancement of source coding systems using high-frequency reconstruction. The invention teaches that tonal signals can be classified as either pulse-train-like or non-pulse-train-like. Relying on this classification, significant improvements on the perceived audio quality can be obtained by adaptive switching of transposers. The invention shows that the so-switched transposers must have fundamental differences in their characteristics.

Abstract: There is disclosed a method of controlling a speech code of speech communications between mobile terminals via an IP network, between mobile switching centers which are interconnected through the IP network. Two mobile switching centers communicate with each other via the IP network using a field in an IP header of a packet, and determines whether coding processes used by two mobile terminals are the same as each other. If the coding processes are the same as each other, then the two mobile switching centers do not convert the coding process for a speech signal, and transmit speech signals from the mobile terminals directly carried on packets through the IP network. If the coding processes are not the same as each other, then the two mobile switching centers convert the coding process for the speech signal into a general-purpose coding process for the speech signal to be transmitted through the IP network.

Abstract: A method for searching an excitation (or fixed) codebook in a speech coding system. In a speech coding system including a synthesis filter for synthesizing a speech signal, a fixed codebook searcher according to the present invention segments a speech signal frame into a plurality of subframes to generate an excitation signal to be used in a synthesis filter, segments again each of the subframes into a plurality of subgroups, and searches the respective subframes each comprised of a plurality of pulse position/amplitude combinations for pulses. The fixed codebook searcher searches the respective subgroups for a predetermine number of pulses having non-zero amplitude, and generates the searched pulses as an initial vector. Next, the fixed codebook searcher selects a pulse combination including at least one pulse among the pulses of the initial vector, and then substitutes pulses of the selected pulse combination for pulses in other positions in the subgroups.

Abstract: The present invention relates to packet-distributed data transmission of compressed data. According to the invention, parity bits are supplied to the compressed data. The parity bits are used in the entire transmission chain between an encoder having compressed the data, and a decoder which decompresses it. According to one embodiment, the data is speech and the packet-distributed network is a mobile radio network with packet-distribution in included links. However, sending in the radio link of the compressed speech is circuit switched.

Abstract: A digital signal of which input data has been segmented as block each having a predetermined data amount and highly efficiently encoded along with an adjacent block is decoded, edited, and then highly efficiently encoded. A delay that takes place in such signal processes is compensated. Thus, part of a digital signal that has been highly efficiently encoded digital signal can be edited.

Abstract: An audio signal coding device, an audio signal decoding device and a method to improve audio quality. The audio signal coding device and method include a quantizer that quantizes a given signal according to a number of assigned bits in order to generate a codeword. The coding device includes an extractor that extracts core bits from the generated codeword. The coding device also includes a determiner that determines an optimal value of the number of assigned bits based on an energy level corresponding to the extracted core bits. The audio signal decoding device and method include a dequantizer that dequantizes a given codeword according to the number of assigned bits to generate a decoded signal. The decoding device includes an extractor that extracts core bits from the given codeword. The decoding device also includes a determiner that determines an optimal value of the number of assigned bits used in the dequantizer.

Abstract: Digital signal samples X in a floating-point format, each of which is composed of 1 bit of sign, 8 bits of exponent E and 23 bits of mantissa M, are converted through rounding by an integer formatting part 12 into digital signal samples Y in an integer format, the sequence of the digital signal samples Y is losslessly compression-coded by a compressing part 13 into a code sequence Ca, and the code sequence Ca is output. The digital signal samples Y are converted by a floating point formatting part 15 into digital signal samples X? in the floating-point format, a difference signal ?X indicating the difference between the digital signal sample X? and the digital signal sample X is determined by a subtraction part 16, the difference signal ?X is losslessly coded, and the resulting code sequence Cb is output.