| Entries |
| Document | Title | Date |
|---|
| 20080249767 | METHOD AND SYSTEM FOR REDUCING FRAME ERASURE RELATED ERROR PROPAGATION IN PREDICTIVE SPEECH PARAMETER CODING - Predictive encoding methods, predictive encoders, and digital systems are provided that encode input frames by computing quantized predictive frame parameters for an input frame, recomputing the quantized predictive frame parameters wherein a previous frame is assumed to be erased and frame erasure concealment is used, and encoding the input frame based on the results of the computing and the recomputing. In embodiments of these methods, encoders, and digital systems, two phase codebook search techniques used in the encoding process are provided that compute the predictive parameters in the first phase, and the predictive parameters assuming the prior frame is erased in the second phase. In the second phase, a frame erasure concealment technique is used in the computation of the predictive parameters. | 10-09-2008 |
| 20080249768 | METHOD AND SYSTEM FOR SPEECH COMPRESSION - 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. | 10-09-2008 |
| 20080255832 | Scalable Encoding Apparatus and Scalable Encoding Method - A scalable encoding apparatus wherein stereo audio signals can be scalable encoded by use of a CELP encoding to improve the encoding efficiency. In the apparatus, an adder and a multiplier obtain an average of first and second channel signals as a monophonic signal. A CELP encoding part performs a CELP encoding of the monophonic signal. A first channel difference information encoding part performs an encoding of the first channel signal in conformance with the CELP encoding and obtains a difference between a resulting encoded parameter and an encoded parameter outputted from the CELP encoding part. The first channel difference information encoding part then encodes this difference and outputs the resulting encoded parameter. | 10-16-2008 |
| 20080255833 | Scalable Encoding Device, Scalable Decoding Device, and Method Thereof - A scalable encoding device for realizing scalable encoding by CELP encoding of a stereo sound signal and improving the encoding efficiency. In this device, an adder and a multiplier obtain an average of a first channel signal CH | 10-16-2008 |
| 20080294428 | Packet loss concealment - A method for using a waveform segment in place of a missing portion of an audio waveform generated in response to a packet stream encoding portions of the audio waveform, the method comprising: phase matching a trailing portion of the waveform segment with a trailing portion of the audio waveform that follows the missing portion; and adding the phase matched waveform segment to the audio waveform. | 11-27-2008 |
| 20080306732 | Method and Device for Carrying Out Optimal Coding Between Two Long-Term Prediction Models - Disclosed is a system and method for implementing compression coding of audio signals, such as speech signals, using two long-term prediction (LTP) models. The method determines the parameters of a second long-term prediction model on the basis of the parameters of at least one first LTP model. The present invention is aimed at switching from an LTP model with a single coefficient (monotap) to an LTP model with several coefficients, (multitap) and vice versa, as well as at switching between two multitap LTP models. The complexity of the method may be adjusted, especially as a function of a desired compromise between a target complexity and a desired quality. A device for implementing the method according to the invention is, moreover, very useful for multiple codings in cascade (transcodings) or in parallel (multi-codings and multi-mode codings). | 12-11-2008 |
| 20080312915 | Audio Encoding - 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. | 12-18-2008 |
| 20090012781 | SPEECH CODER AND SPEECH DECODER - A speech coder includes a seed storage that stores a plurality of seeds used as an initial state of oscillation. An oscillator generates different vector sequences in accordance with values of the seeds stored in the seed storage and outputs the vector sequences as excitation vectors. A linear predictive coding synthesis filter receives, as input, the excitation vectors synthesizes the excitation vectors, and outputs a synthesized speech. The seed storage stores the plurality of seeds prepared in advance as the initial state of oscillation such that the vector sequences generated in the oscillator serve as effective excitation vectors from which the synthesized speech can be generated when the vector sequences are input to the linear predictive coding synthesis filter the oscillator receives, the seeds from the seed storage, generates, using the input seeds, vector sequences that serve as the effective excitation vectors from which the synthesized speech can be generated in the linear predictive coding synthesis filter, and outputs the vector sequences. | 01-08-2009 |
| 20090030677 | SCALABLE ENCODING APPARATUS, SCALABLE DECODING APPARATUS, AND METHODS OF THEM - 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 ( | 01-29-2009 |
| 20090037168 | Apparatus for Improving Packet Loss, Frame Erasure, or Jitter Concealment - The invention presents a method to improve the recovering from packet loss, frame erasure or jitter concealment during signal communication, especially for VoIP (Voice Over Internet Protocol) applications. A variable delay concept (instead of constant delay) is introduced to guarantee the continuity and periodicity of signal after recovering lost frames, adding frames or removing frames. During the recovering of lost frames or the adding of extra frames, the copy of previous signal from history buffer into missing frame(s) is based on the frame length, onset, and offset information. | 02-05-2009 |
| 20090076809 | AUDIO ENCODING DEVICE AND AUDIO ENCODING METHOD - There is provided an audio encoding device capable of effectively encoding stereo audio in audio encoding having monaural-stereo scalable configuration. In this device, a correlation degree comparison unit ( | 03-19-2009 |
| 20090083031 | CLIPPED-WAVEFORM REPAIR IN ACOUSTIC SIGNALS USING GENERALIZED LINEAR PREDICTION - A method and system for optimally repairing a clipped audio signal. Clipping occurs when a waveform exceeds a dynamic range of a recording device. Portions of an audio signal exceeding the dynamic range or saturation level of the recording device are clipped, causing distortion when the clipped recorded signal is played. To address this problem, successive frames of the clipped audio data are repaired to fill in gaps where the data were clipped. For each frame, an iterative process repetitively estimates an auto-covariance and detects clipped samples in the frame or a sub-frame in order to compute a least-squares solution for the frame that interpolates the clipped data. The process can cause inverted peaks in the repaired data, which must then be rectified to produced corrected repaired data. The corrected repaired data for the successive frames are recombined using interpolation, to produce a complete repaired audio data set. | 03-26-2009 |
| 20090094023 | Method, medium, and apparatus encoding scalable wideband audio signal - Provided is a method and apparatus for encoding a scalable wideband audio signal, the method including: filtering a voiced signal by performing linear prediction on the voiced signal and modulating the filtered signal; encoding the modulated signal in the time domain, and outputting a core layer encoding result of the voiced signal; subtracting a signal obtained by decoding the core layer encoding result from the modulated signal and outputting an error signal; and encoding the error signal and outputting an enhancement layer encoding result of the voiced signal. | 04-09-2009 |
| 20090094024 | CODING DEVICE AND CODING METHOD - 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. | 04-09-2009 |
| 20090094025 | METHOD FOR SPEECH CODING, METHOD FOR SPEECH DECODING AND THEIR APPARATUSES - A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal. In speech coding method according to a code-excited linear prediction (CELP)speech coding, a noise level of a speech in a concerning coding period is evaluated by using a code or coding result of at least one of spectrum information, power information, and pitch information, and various excitation codebooks are used based on an evaluation result | 04-09-2009 |
| 20090112581 | METHOD AND APPARATUS FOR TRANSMITTING AN ENCODED SPEECH SIGNAL - A method and apparatus for processing speech in a wireless communication system uses CELP speech encoded signals. A speech input receives samples of a speech signal and a codebook analysis block for selects an index of a code from at least one of a plurality of codebooks. A weighted synthesis filter is used in the generation of a prediction error between a predicted current sample and a current sample of the speech samples. The index is transmitted to the receiver to enable reconstruction of the speech signal at the receiver. | 04-30-2009 |
| 20090119098 | SIGNAL PROCESSING METHOD, PROCESSING APPARATUS AND VOICE DECODER - The present invention discloses a signal processing method adapted to process a synthesized signal in packet loss concealment. The method includes the following steps: receiving a good frame following a lost frame, obtaining an energy ratio of energy of a signal in the signal of the good frame signal to energy of a synthesized signal corresponding to the same time of the good frame, and adjusting the synthesized signal in accordance with the energy ratio. The present invention also discloses a signal processing apparatus and a voice decoder. Through using the method provided by the present invention, the synthesized signal is adjusted in accordance with the energy ratio of the energy of the first good frame following the lost frame to the energy of the synthesized signal to ensure that there be not a waveform sudden change or an energy sudden change at the place where the lost frame and the first good frame following the lost frame are jointed in the synthesized signal, to realize the waveform's smooth transition and to avoid music noises. | 05-07-2009 |
| 20090138260 | VOICE JUDGING SYSTEM, VOICE JUDGING METHOD AND PROGRAM FOR VOICE JUDGMENT - A voice judging system including feature value extraction means that analyzes a sound signal input from a sound signal input device, and extracts a time series of the feature values, sub-word boundary score calculating means that calculates a time series of sub-word boundary scores, by having reference to sound models of voice stored in a voice model storage unit, temporal regularity analyzing means that analyzes temporal regularity of the sub-word boundary scores, and voice judgment means judges whether the input sound signal is voice or non-voice using of the temporal regularity of the sub-word boundary scores. | 05-28-2009 |
| 20090157397 | Voice Rule-Synthesizer and Compressed Voice-Element Data Generator for the same - A voice rule-synthesizer synthesizes a voice waveform based on the voice data stored in a database, which stores a large number of compressed voice data sections in a data stream. Each voice data section is stored as a plurality of frames compressed in a fixed-length frame format. The storage capacity of the database is reduced because the compressed voice data sections are stored as the data stream. | 06-18-2009 |
| 20090164210 | Codebook sharing for LSF quantization - In accordance with one aspect of the invention, a selector supports the selection of a first encoding scheme or the second encoding scheme based upon the detection or absence of the triggering characteristic in the interval of the input speech signal. The first encoding scheme has a pitch pre-processing procedure for processing the input speech signal to form a revised speech signal biased toward an ideal voiced and stationary characteristic. The pre-processing procedure allows the encoder to fully capture the benefits of a bandwidth-efficient, long-term predictive procedure for a greater amount of speech components of an input speech signal than would otherwise be possible. In accordance with another aspect of the invention, the second encoding scheme entails a long-term prediction mode for encoding the pitch on a sub-frame by sub-frame basis. The long-term prediction mode is tailored to where the generally periodic component of the speech is generally not stationary or less than completely periodic and requires greater frequency of updates from the adaptive codebook to achieve a desired perceptual quality of the reproduced speech under a long-term predictive procedure. | 06-25-2009 |
| 20090171657 | Hybrid Approach in Voice Conversion - A hybrid approach is described for combining frequency warping and Gaussian Mixture Modeling (GMM) to achieve better speaker identity and speech quality. To train the voice conversion GMM model, line spectral frequency and other features are extracted from a set of source sounds to generate a source feature vector and from a set of target sounds to generate a target feature vector. The GMM model is estimated based on the aligned source feature vector and the target feature vector. A mixture specific warping function is generated each set of mixture mean pairs of the GMM model, and a warping function is generated based on a weighting of each of the mixture specific warping functions. The warping function can be used to convert sounds received from a source speaker to approximate speech of a target speaker. | 07-02-2009 |
| 20090177465 | METHOD AND ARRANGEMENT FOR SPEECH CODING IN WIRELESS COMMUNICATION SYSTEMS - The present invention relates to speech coding in wireless and wireline communication systems. The present invention provides a method of saving bandwidth by a controlled dropping of speech frames at an encoder in a sending communication device. The dropping is controlled in a manner to minimize the effects on the speech quality after the decoding in the receiving communication device, by assuring that the state mismatch between the encoder and the decoder is removed or at least significantly reduced. This is achieved by letting the encoder run an ECU algorithm with a similar behavior as the one running in the decoder in the receiving communication device. | 07-09-2009 |
| 20090192790 | SYSTEMS, METHODS, AND APPARATUS FOR CONTEXT SUPPRESSION USING RECEIVERS - Configurations disclosed herein include systems, methods, and apparatus that may be applied in a voice communications and/or storage application to remove, enhance, and/or replace the existing context. | 07-30-2009 |
| 20090192791 | SYSTEMS, METHODS AND APPARATUS FOR CONTEXT DESCRIPTOR TRANSMISSION - Configurations disclosed herein include systems, methods, and apparatus that may be applied in a voice communications and/or storage application to remove, enhance, and/or replace the existing context. | 07-30-2009 |
| 20090192792 | METHODS AND APPARATUSES FOR ENCODING AND DECODING AUDIO SIGNAL - 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. | 07-30-2009 |
| 20090198491 | LSP VECTOR QUANTIZATION APPARATUS, LSP VECTOR INVERSE-QUANTIZATION APPARATUS, AND THEIR METHODS - Disclosed is an LPC vector quantization device capable of quantizing an LSP vector by using correlation between divided vectors. The device includes: a vector dividing unit ( | 08-06-2009 |
| 20090222261 | Apparatus and Method for Encoding and Decoding Signal - 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. | 09-03-2009 |
| 20090240491 | TECHNIQUE FOR ENCODING/DECODING OF CODEBOOK INDICES FOR QUANTIZED MDCT SPECTRUM IN SCALABLE SPEECH AND AUDIO CODECS - 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. | 09-24-2009 |
| 20090240492 | PACKET LOSS CONCEALMENT FOR SUB-BAND PREDICTIVE CODING BASED ON EXTRAPOLATION OF SUB-BAND AUDIO WAVEFORMS - 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. | 09-24-2009 |
| 20090240493 | METHOD AND APPARATUS FOR SEARCHING FIXED CODEBOOK - 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. | 09-24-2009 |
| 20090248403 | DEREVERBERATION APPARATUS, DEREVERBERATION METHOD, DEREVERBERATION PROGRAM, AND RECORDING MEDIUM - A model application unit calculates linear prediction coefficients of a multi-step linear prediction model by using discrete acoustic signals. Then, a late reverberation predictor calculates linear prediction values obtained by substituting the linear prediction coefficients and the discrete acoustic signals into linear prediction term of the multi-step linear prediction model, as predicted late reverberations. Next, a frequency domain converter converts the discrete acoustic signals to discrete acoustic signals in the frequency domain and also converts the predicted late reverberations to predicted late reverberations in the frequency domain. A late reverberation eliminator calculates relative values between the amplitude spectra of the discrete acoustic signals expressed in the frequency domain and the amplitude spectra of the predicted late reverberations expressed in the frequency domain, and provides the relative values as predicted amplitude spectra of a dereverberation signal. | 10-01-2009 |
| 20090248404 | LOST FRAME COMPENSATING METHOD, AUDIO ENCODING APPARATUS AND AUDIO DECODING APPARATUS - 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. This vector is used to synthesize a decoded signal. The pulse position ‘b’ and pulse amplitude ‘a’ are then decided such that a difference between the synthesized signal and an input signal becomes minimum. | 10-01-2009 |
| 20090248405 | PACKET LOSS CONCEALMENT FOR A SUB-BAND PREDICTIVE CODER BASED ON EXTRAPOLATION OF EXCITATION WAVEFORM - Systems and methods are described for performing packet loss concealment using an extrapolation of an excitation waveform in a sub-band predictive speech coder, such as an ITU-T Recommendation G.722 wideband speech coder. The systems and methods are useful for concealing the quality-degrading effects of packet loss in a sub-band predictive coder and address some sub-band architectural issues when applying excitation extrapolation techniques to such sub-band predictive coders. | 10-01-2009 |
| 20090265167 | SPEECH ENCODING APPARATUS AND SPEECH ENCODING METHOD - Disclosed is an audio encoding device capable of adjusting a spectrum inclination of a quantized noise without changing the Formant weight. The device includes: an HPF ( | 10-22-2009 |
| 20090281798 | PREDICTIVE ENCODING OF A MULTI CHANNEL SIGNAL - A multi channel encoder ( | 11-12-2009 |
| 20090292534 | FIXED CODE BOOK SEARCH DEVICE AND FIXED CODE BOOK SEARCH METHOD - 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 ( | 11-26-2009 |
| 20090299737 | Method for adapting for an interoperability between short-term correlation models of digital signals - The invention relates to the code conversion of digital signals, particularly voice signals, and in particular coding according to a second format from information obtained by carrying out a coding according to a first format. These first and second formats use LPC (linear predictive coding) short-term prediction models on digital signal sample blocks while using filters represented by respective LPC coefficients. The LPC coefficients of the second format are determined from an interpolation on the representative values of the LPC coefficients of at least the first format, between at least one given block and a preceding block. According to the invention, the interpolation ( | 12-03-2009 |
| 20100010811 | STEREO AUDIO ENCODING DEVICE, STEREO AUDIO DECODING DEVICE, AND METHOD THEREOF - Disclosed is a stereo audio encoding device capable of reducing a bit rate. In this device, a stereo audio encoding unit ( | 01-14-2010 |
| 20100023323 | Multi-Reference LPC Filter Quantization and Inverse Quantization Device and Method - A multi-reference quantization device and method for quantizing an input LPC filter, comprises a plurality of differential quantizers using respective, different references, and a selector of a reference amongst the different references of the differential quantizers using a reference selection criterion. The input LPC filter is differentially quantized by the differential quantizer using the selected reference. A device and method for inverse quantizing a multi-reference differentially quantized LPC filter extracted from a bitstream, comprises an extractor from the bitstream of information about a reference amongst a plurality of possible references used for quantizing the multi-reference differentially quantized LPC filter, and a differential inverse quantizer using the reference corresponding to the extracted reference information to inverse quantize the multi-reference differentially quantized LPC filter. | 01-28-2010 |
| 20100023324 | Device and Method for Quanitizing and Inverse Quanitizing LPC Filters in a Super-Frame - A device and a method for quantizing, in a super-frame including a sequence of frames, LPC filters calculated during the frames of the sequence. The LPC filter quantizing device and method comprises: an absolute quantizer for first quantizing one of the LPC filters using absolute quantization; and at least one quantizer of the other LPC filters using a quantization mode selected from the group consisting of absolute quantization and differential quantization relative to at least one previously quantized filter amongst the LPC filters. For inverse quantizing, at least the first quantized LPC filter is received and an inverse quantizer inverse quantizes the first quantized LPC filter using absolute inverse quantization. If any quantized LPC filter other than the first quantized LPC filter is received, an inverse quantizer inverse quantizes this quantized LPC filter using one of absolute inverse quantization and differential inverse quantization relative to at least one previously received quantized LPC filter. | 01-28-2010 |
| 20100023325 | Variable Bit Rate LPC Filter Quantizing and Inverse Quantizing Device and Method - A device and a method for quantizing a LPC filter in the form of an input vector in a quantization domain, comprises a calculator of a first-stage approximation of the input vector, a subtractor of the first-stage approximation from the input vector to produce a residual vector, a calculator of a weighting function from the first-stage approximation, a warper of the residual vector with the weighting function, and a quantizer of the weighted residual vector to supply a quantized weighted residual vector. A device and a method for inverse quantizing of a LPC filter, comprises means for receiving coded indices representative of a first-stage approximation of a vector representative of the LPC filter in a quantization domain and of a quantized weighted residual version of the vector, a calculator of an inverse weighting function from the first-stage approximation, an inverse quantizer of the quantized weighted residual version of the vector to produce a weighted residual vector, a multiplier of the weighted residual vector by the inverse weighting function to produce a residual vector, and an adder of the first-stage approximation with the residual vector to produce the vector representative of the LPC filter in the quantization domain. | 01-28-2010 |
| 20100023326 | SPEECH ENDODING DEVICE - The present invention is a synthetic speech encoding device that produces a synthetic speech signal which closely matches an actual speech signal. The actual speech signal is digitized, and excitation pulses are selected by minimizing the error between the actual and synthetic speech signals. The preferred pattern of excitation pulses needed to produce the synthetic speech signal is obtained by using an excitation pattern containing a multiplicity of weighted pulses at timed positions. The selection of the location and amplitude of each excitation pulse is obtained by minimizing an error criterion between the synthetic speech signal and the actual speech signal. The error criterion function incorporates a perceptual weighting filter which shapes the error spectrum. | 01-28-2010 |
| 20100049507 | APPARATUS FOR NOISE SUPPRESSION IN AN AUDIO SIGNAL - An apparatus for noise suppression having a linear prediction analysis circuit having an LP error filter (LFF), which takes a first, noisy voice signal y(n)=x(n)+ε(n) as a basis for producing an LP-error-filter output signal e(n), having a coefficient calculation unit (KBE), which updates the coefficients of the LP error filter on the basis of the internal signals (including the input and out signals y(n) and e(n)) in the LP error filter, and having a subtraction unit, which subtracts the LP error filter output signal e(n) from the first voice signal y(n) in a subtractor and, following the subtraction, outputs the remainder as a second voice signal x(n)=y(n)−e(n) in which the noise is suppressed. A noise estimation unit (GSE) is provided which takes the internal signals of the LP error filter as a basis for producing a noise power signal σ | 02-25-2010 |
| 20100049508 | AUDIO ENCODING DEVICE AND AUDIO ENCODING METHOD - Provided is an audio encoding device which performs a closed loop search of a gain and a sound source vector without significantly increasing the calculation amount as compared to an open loop search. In the audio encoding device, firstly, a first parameter decision unit ( | 02-25-2010 |
| 20100049509 | AUDIO ENCODING DEVICE AND AUDIO DECODING DEVICE - Disclosed are an audio encoding device and an audio decoding device which reduce degradation of subjective quality of a decoding signal caused by power mismatch of a decoding signal which is generated by a concealing process upon disappearance of a frame. When a frame is lost, a past encoding parameter is used to obtain a concealed LPC of the current frame and a concealed sound source parameter. A normal CELP decoding is performed from the obtained concealed sound source parameter. Correction is performed by using a conceal parameter on the obtained concealed LPC and the concealed sound source signal. The power of the corrected concealed sound source signal is adjusted to match a reference sound source power. A filter gain of the synthesis filter is adjusted so as to adjust the power of a decoded sound signal to the power of a decoded sound signal during an error-free state. Moreover, a synthesis filter gain adjusting coefficient is calculated by using an estimated normalized residual power so that a filter gain of a synthesis filter formed by using a concealed LPC is a filter gain during an error-free state. | 02-25-2010 |
| 20100049510 | METHOD AND DEVICE FOR PERFORMING PACKET LOSS CONCEALMENT - A method, device and system to implement hiding the loss packet are provided. The provided method, device and system recover the lost frame according to the data before and after the lost frame and enhances the correlation of the recovered lost frame data and the data after the lost frame. A method and device for estimating pitch period are also provided which select a pitch period from the initial pitch period and the pitch periods corresponding to the frequencies which are one or more times higher than the frequencies corresponding to the initial pitch period as the final estimated pitch period, may improve frequency multiplication when estimating the pitch period; in addition, by tuning of the pitch period by matching the waves, the error of estimating pitch period may be reduced and the quality of the audio data may be improved. | 02-25-2010 |
| 20100057447 | PARAMETER DECODING DEVICE, PARAMETER ENCODING DEVICE, AND PARAMETER DECODING METHOD - Provided is a parameter decoding device which performs parameter compensation process so as to suppress degradation of a main observation quality in a prediction quantization. The parameter decoding device includes amplifiers ( | 03-04-2010 |
| 20100070271 | Transmission error concealment in audio signal - A method of concealing transmission error in a digital audio signal, wherein a signal that has been decoded after transmission is received, the samples decoded while the transmitted data is valid are stored, at least one short-term prediction operator and one long-term prediction operator are estimated as a function of stored valid samples, and any missing or erroneous samples in the decoder signal are generated using the estimated operators. The energy of the synthesized signal that is thus generated is controlled by means of a gain that is computed and adapted sample by sample. | 03-18-2010 |
| 20100070272 | METHOD AND AN APPARATUS FOR PROCESSING A SIGNAL - An apparatus for processing an encoded signal and method thereof are disclosed, by which an audio signal can be compressed and reconstructed in higher efficiency. An audio signal processing method includes the steps of identifying whether a coding type of the audio signal is a music signal coding type using first type information, if the coding type of the audio signal is not the music signal coding type, identifying whether the coding type of the audio signal is a speech signal coding type or a mixed signal coding type using second type information, if the coding type of the audio signal is the mixed signal coding type, extracting spectral data and a linear predictive coefficient from the audio signal, generating a residual signal for linear prediction by performing inverse frequency conversion on the spectral data, reconstructing the audio signal by performing linear prediction coding on the linear predictive coefficient and the residual signal, and reconstructing a high frequency region signal using an extension base signal corresponding to a partial region of the reconstructed audio signal and band extension information. Accordingly, various kinds of audio signals can be encoded/decoded in higher efficiency. | 03-18-2010 |
| 20100082337 | ADAPTIVE SOUND SOURCE VECTOR QUANTIZATION DEVICE, ADAPTIVE SOUND SOURCE VECTOR INVERSE QUANTIZATION DEVICE, AND METHOD THEREOF - Disclosed is an adaptive sound source vector quantization device capable of improving quantization accuracy of adaptive sound source vector quantization while suppressing increase of the calculation amount in CELP sound encoding which performs encoding in sub-frame unit. In the device, a search adaptive sound source vector generation unit ( | 04-01-2010 |
| 20100088090 | ARITHMETIC ENCODING FOR CELP SPEECH ENCODERS - A communication system ( | 04-08-2010 |
| 20100088091 | FIXED CODEBOOK SEARCH METHOD THROUGH ITERATION-FREE GLOBAL PULSE REPLACEMENT AND SPEECH CODER USING THE SAME METHOD - Provided are a fixed codebook search method based on iteration-free global pulse replacement in a speech codec, and a Code-Excited Linear-Prediction (CELP)-based speech codec using the method. The fixed codebook search method based on iteration-free global pulse replacement in a speech codec includes the steps of: (a) determining an initial codevector using a pulse-position likelihood vector or a correlation vector; (b) calculating a fixed-codebook search criterion value for the initial codevector; (c) calculating fixed-codebook search criterion values for respective codevectors obtained by replacing a pulse of the initial codevector each time for respective tracks, and determining a pulse position generating the largest fixed-codebook search criterion value as a candidate pulse position for the respective tracks, respectively; (d) calculating fixed-codebook search criterion values for respective codevectors of all combinations obtained by replacing at least one pulse position of the initial codevector with the candidate pulse positions of the respective tracks, and determining the largest value of the fixed-codebook search criterion values; and (e) comparing the fixed-codebook search criterion value for the initial codevector obtained in step (b) with the largest value determined in step (d) to determine an optimum fixed codevector. | 04-08-2010 |
| 20100100373 | AUDIO DECODING DEVICE AND AUDIO DECODING METHOD - Provided is an audio decoding device which can adjust the high-range emphasis degree in accordance with a background noise level. The audio decoding device includes: a sound source signal decoding unit ( | 04-22-2010 |
| 20100106492 | ADAPTIVE SOUND SOURCE VECTOR QUANTIZATION UNIT AND ADAPTIVE SOUND SOURCE VECTOR QUANTIZATION METHOD - Disclosed is an adaptive sound source vector quantization device capable of reducing deviation of the quantization accuracy of the adaptive sound source vector quantization of each sub-frame when performing an adaptive sound source vector quantization in a sub-frame unit by using a greater information amount in a first sub-frame than in a second sub-frame. In this device: when the device performs the adaptive sound source vector quantization of the first sub-frame, an adaptive sound source vector generation unit ( | 04-29-2010 |
| 20100106493 | ENCODING DEVICE AND ENCODING METHOD - Provided is an encoding device which can achieve both of highly effective encoding/decoding and high-quality decoding audio when executing a scalable stereo audio encoding by using MDCT and ICP. In the encoding device, an MDCT conversion unit ( | 04-29-2010 |
| 20100114566 | Method and apparatus for encoding/decoding speech signal - An apparatus and method for encoding/decoding a speech signal which determines a variable bit rate based on reserved bits obtained from a target bit rate, is provided. The variable bit rate is determined based on a source feature of the speech signal and the reserved bits is obtained based on the target bit rate. The apparatus for encoding the speech signal may include a linear predictive (LP) analysis unit/quantization unit to determine an immittance spectral frequencies (ISF) index, a closed loop pitch search unit, a fixed codebook search unit, a gain vector quantization (VQ) unit to determine a gain vector quantization (VQ) index, and a bit rate control unit to control at least two indexes of the ISF index, the pitch index, the code index, and the gain VQ index to be encoded to be variable bit rates based on a source feature of a speech signal and the reserved bits. | 05-06-2010 |
| 20100114567 | Method And Arrangement For Smoothing Of Stationary Background Noise - In a method of smoothing background noise in a telecommunication speech session; receiving and decoding S | 05-06-2010 |
| 20100125453 | APPARATUS AND METHOD FOR ENCODING AT LEAST ONE PARAMETER ASSOCIATED WITH A SIGNAL SOURCE - Apparatus ( | 05-20-2010 |
| 20100125454 | PACKET LOSS CONCEALMENT FOR SUB-BAND CODECS - Packet loss concealment systems and methods are described that may be used in conjunction with a Bluetooth® Low-Complexity Sub-band Coding (LC-SBC) codec or other sub-band codecs, including but not limited to an MPEG-1 Audio Layer 3 (MP3) codec, an Advanced Audio Coding (AAC) codec, and a Dolby AC-3 codec. | 05-20-2010 |
| 20100125455 | AUDIO ENCODING AND DECODING WITH INTRA FRAMES AND ADAPTIVE FORWARD ERROR CORRECTION - Various strategies for rate/quality control and loss resiliency in an audio codec are described. The various strategies can be used in combination or independently. For example, a real-time speech codec uses intra frame coding/decoding, adaptive multi-mode forward error correction [“FEC”], and rate/quality control techniques. Intra frames help a decoder recover quickly from packet losses, while compression efficiency is still emphasized with predicted frames. Various strategies for inserting intra frames and signaling intra/predicted frames are described. With the adaptive multi-mode FEC, an encoder adaptively selects between multiple modes to efficiently and quickly provide a level of FEC that takes into account the bandwidth currently available for FEC. The FEC information itself may be predictively encoded and decoded relative to primary encoded information. Various rate/quality and FEC control strategies allow additional adaptation to available bandwidth and network conditions. | 05-20-2010 |
| 20100138219 | Coding Apparatus and Decoding Apparatus - A coding apparatus capable of coding a spectrum at a low bit rate and with high quality without producing any disturbance in a harmonic structure of the spectrum. In this apparatus, internal state setting section sets an internal state of a filtering section using a first spectrum S | 06-03-2010 |
| 20100153099 | SPEECH ENCODING APPARATUS AND SPEECH ENCODING METHOD - A speech coder and so forth for preventing deterioration of the quality of a reproduced speech signal while reducing the coding rate. In a speech signal modifying section ( | 06-17-2010 |
| 20100153100 | ADDRESS GENERATOR FOR SEARCHING ALGEBRAIC CODEBOOK - An address generator for searching an algebraic codebook is disclosed. The address generator includes: a multiplier multiplying the dimension and a width value of a correlation matrix; a first adder adding a length value and an offset address of the correlation matrix; and a second adder adding the results of the multiplier and the first adder to generate an address for algebraic codebook searching. The amount of calculation required for an address calculation to search an algebraic codebook can be reduced. | 06-17-2010 |
| 20100169086 | SIGNAL COMPRESSION METHOD AND APPARATUS - A signal compression method includes: multiplying an input signal by a window function, calculating original autocorrelation coefficients of a windowed input signal. The method also includes calculating a white-noise correction factor or a lag-window according to the original autocorrelation coefficients, and calculating modified autocorrelation coefficients according to the original autocorrelation coefficients, the white-noise correction factor and the lag-window. The method further includes calculating linear prediction coefficients according to the modified autocorrelation coefficients, and outputting a coded bit stream according to the linear prediction coefficients. | 07-01-2010 |
| 20100174537 | Speech coding - A method, system and computer program for encoding speech according to a source-filter model. The method comprises deriving a spectral envelope signal representative of a modelled filter and a first remaining signal representative of a modelled source signal, and deriving a second remaining signal from the first remaining signal by, at intervals during the encoding: exploiting a correlation between approximately periodic portions in the first remaining signal to generate a predicted version of a later portion from a stored version of an earlier portion, and using the predicted-version of the later portion to remove an effect of said periodicity from the first remaining signal. The method further comprises, once every number of intervals, transforming the stored version of the earlier portion of the first remaining signal prior to generating the predicted version of the respective later portion. | 07-08-2010 |
| 20100174538 | Speech encoding - A method, system and program for encoding and decoding speech according to a source-filter model whereby speech is modelled to comprise a source signal filtered by a time-varying filter. The method comprises: receiving a speech signal comprising successive frames. For each of a plurality of frames of the speech signal: adding a predetermined noise signal to the speech signal to generate a simulated signal, determining linear predictive coding coefficients based on the simulated signal frame, and determining a linear predictive coding residual signal based on the linear predictive coding coefficients and one of the speech signal and the simulated signal. Then forming an encoded signal representing said speech signal, based on the linear predictive coding coefficients and the linear predictive coding residual signal. | 07-08-2010 |
| 20100191526 | AUDIO ENCODING DEVICE AND AUDIO ENCODING METHOD - An audio encoding device which can improve encoding performance while performing division search on an algebraic codebook in an audio encoding. In a distortion minimizing unit ( | 07-29-2010 |
| 20100211386 | METHOD FOR MANUFACTURING A SEMICONDUCTOR PACKAGE - The present research can decrease the amount of computation and enhance speech quality by using a global pulse replacement method in a fixed codebook search. The fixed codebook search method in a speech encoder based upon global pulse replacement, includes the steps of: (a) computing absolute values of the pulse-position likelihood-estimator vectors; (b) temporarily obtaining a codebook vector; (c) computing a mathematical equation by replacing a pulse; (d) determining whether a value computed based upon the mathematical equation is increased after pulse replacement; (e) obtaining a new codebook vector by replacing the pulse; and (f) maintaining a previous codebook vector. | 08-19-2010 |
| 20100217585 | Method and Arrangement for Enhancing Spatial Audio Signals - In a method of enhancing spatial audio signals, receiving (S | 08-26-2010 |
| 20100223053 | EFFICIENT SPEECH STREAM CONVERSION - Speech frames of a first speech coding scheme are utilized as speech frames of a second speech coding scheme, where the speech coding schemes use similar core compression schemes for the speech frames, preferably bit stream compatible. An occurrence of a state mismatch in an energy parameter between the first speech coding scheme and the second speech coding scheme is identified, preferably either by determining an occurrence of a predetermined speech evolution, such as a speech type transition, e.g. an onset of speech following a period of speech inactivity, or by tentative decoding of the energy parameter in the two encoding schemes followed by a comparison. Subsequently, the energy parameter in at least one frame of the second speech coding scheme following the occurrence of the state mismatch is adjusted. The present invention also presents transcoders and communications systems providing such transcoding functionality. | 09-02-2010 |
| 20100223054 | SINGLE-MICROPHONE WIND NOISE SUPPRESSION - A technique for suppressing non-stationary noise, such as wind noise, in an audio signal is described. In accordance with the technique, a series of frames of the audio signal is analyzed to detect whether the audio signal comprises non-stationary noise. If it is detected that the audio signal comprises non-stationary noise, a number of steps are performed. In accordance with these steps, a determination is made as to whether a frame of the audio signal comprises non-stationary noise or speech and non-stationary noise. If it is determined that the frame comprises non-stationary noise, a first filter is applied to the frame and if it is determined that the frame comprises speech and non-stationary noise, a second filter is applied to the frame. | 09-02-2010 |
| 20100228544 | SPEECH CODER AND SPEECH DECODER - A vector quantization apparatus performs coding of a linear predictive coding coefficient by multi-stage vector quantization. A first codebook and a second codebook store code vectors, and a storing section stores scalars. A first quantizing section extracts a first code vector stored in the first codebook and performs first stage quantization for quantizing a target vector using the first code vector. A second quantizing section extracts a second code vector stored in the second codebook, calculates a third code vector by multiplying the second code vector and one of the scalars stored in the storing section, performs distance calculation using the target vector, the first code vector and the third code vector, and performs second stage quantization for quantizing the target vector using a result of the distance calculation. Each scalar stored in the storing section is associated with at least one of the vectors stored in the first codebook. | 09-09-2010 |
| 20100274557 | METHOD AND AN APPARATUS FOR PROCESSING A SIGNAL - A method of processing a signal is disclosed. The present invention includes receiving extension information and at least one downmix signal of a first downmix signal decoded by a audio coding scheme and a second downmix signal decoded by a speech coding scheme; determining an extension base signal corresponding to a partial region of the downmix signal based on the extension information; and generating an extended downmix signal having a bandwidth extended by reconstructing a high frequency region signal using the extension base signal and the extension information. According to a signal processing method and apparatus of the present invention, signal corresponding to a partial frequency region of the downmix signal is used as the extension base signal. Therefore, the high frequency region of the downmix signal is reconstructed by using the extension base signal having variable bandwidth. | 10-28-2010 |
| 20100274558 | ENCODER, DECODER, AND ENCODING METHOD - An encoder capable of reducing the degradation of the quality of the decoded signal in the case of band expansion in which the high band of the spectrum of an input signal is estimated from the low band. In this encoder, a first layer encoding section ( | 10-28-2010 |
| 20100292986 | ENCODER - An encoder is configured to receive an audio signal and output a scaled encoded signal. The encoder is further configured to generate a synthesized audio signal and an encoded signal. The encoder is further configured to scale the encoded signal dependent on the synthesized audio signal. | 11-18-2010 |
| 20100312552 | SYSTEMS AND METHODS FOR PREVENTING THE LOSS OF INFORMATION WITHIN A SPEECH FRAME - A method for preventing the loss of information within a speech frame is described. A first speech frame to be encoded is selected. A determination is made as to whether or not a second speech frame is a critical speech frame based on the information within the second speech frame and one or more adjacent speech frames. At least a part of an encoded version of the second speech frame is created according to a selected forward error correction (FEC) mode if the second speech frame is a critical speech frame. The first speech frame and the at least a part of the encoded version of the second speech frame are transmitted. | 12-09-2010 |
| 20100324892 | EXCITATION VECTOR GENERATOR, SPEECH CODER AND SPEECH DECODER - A code excited linear prediction type speech coder, which includes a seed storage that stores seeds used as an initial state of oscillation, and an oscillator that generates different vector sequences in accordance with values of the seeds stored in the seed storage and outputs the vector sequences as excitation vectors. The speech coder also includes a linear predictive coding synthesis filter that receives, as input, the excitation vectors, which are the vector sequences generated in accordance with the values of the seeds, that synthesizes the excitation vectors, and that outputs a synthesized speech. | 12-23-2010 |
| 20110004469 | VECTOR QUANTIZATION DEVICE, VECTOR INVERSE QUANTIZATION DEVICE, AND METHOD THEREOF - Disclosed are a vector quantization device and others capable of adaptively adjusting a vector space of a code vector for quantization of a second stage by using a quantization result of a first stage and improving the quantization accuracy. In the device, the first quantization unit ( | 01-06-2011 |
| 20110010168 | MULTIMODE CODING OF SPEECH-LIKE AND NON-SPEECH-LIKE SIGNALS - The invention relates to the coding of audio signals that may include both speech-like and non-speech-like signal components. It describes methods and apparatus for code excited linear prediction (CELP) audio encoding and decoding that employ linear predictive coding (LPC) synthesis filters controlled by LPC parameters, a plurality of codebooks each having codevectors, at least one codebook providing an excitation more appropriate for non-speech-like signals and at least one codebook providing an excitation more appropriate for speech-like signals, and a plurality of gain factors, each associated with a codebook. The encoding methods and apparatus select from the codebooks codevectors and/or associated gain factors by minimizing a measure of the difference between the audio signal and a reconstruction of the audio signal derived from the codebook excitations. The decoding methods and apparatus generate a reconstructed output signal from the LPC parameters, codevectors, and gain factors. | 01-13-2011 |
| 20110022382 | Adaptive Reduction of Noise Signals and Background Signals in a Speech-Processing System - An audio input signal is filtered using an adaptive filter to generate a prediction output signal with reduced noise, wherein the filter is implemented using a plurality of coefficients to generate a plurality of prediction errors and to generate an error from the plurality of prediction errors, wherein the absolute values of the coefficients are continuously reduced by a plurality of reduction parameters. | 01-27-2011 |
| 20110066427 | Receiver Intelligibility Enhancement System - Embodiments of the invention provide a communication device and methods for enhancing audio signals. A first audio signal buffer and a second audio signal buffer are acquired. Thereafter, the magnitude spectrum calculated from the Fast Fourier Transform (FFT) of the second audio signal is processed based on the Linear Predictive Coding (LPC) spectrum of the first audio signal to generate an enhanced second audio signal. | 03-17-2011 |
| 20110071821 | RECEIVER INTELLIGIBILITY ENHANCEMENT SYSTEM - Embodiments of the invention provide a communication device and methods for enhancing audio signals. A first audio signal buffer and a second audio signal buffer are acquired. Thereafter, the second audio signal is processed based on the linear predictive coding coefficients and gains based on noise power of the first audio signal to generate an enhanced second audio signal. | 03-24-2011 |
| 20110099008 | BIT ERROR MANAGEMENT AND MITIGATION FOR SUB-BAND CODING - Systems and method for managing and/or mitigating the impact of bit errors on encoded frames received by an LC-SBC (Low Complexity Sub-band Coding) decoder are described herein. For example, in one embodiment, the impact of bit errors on an LC-SBC frame received by an LC-SBC decoder is estimated and one of a plurality of bit error management techniques is applied to the LC-SBC frame based on the estimated impact, wherein the bit error management techniques may include performing PLC, performing normal SBC decoding, or performing some other technique for managing and/or mitigating the impact of the bit errors. Techniques for concealing bit errors in LC-SBC frames are also described. | 04-28-2011 |
| 20110166854 | METHOD, APPARATUS, PROGRAM AND RECORDING MEDIUM FOR LONG-TERM PREDICTION CODING AND LONG-TERM PREDICTION DECODING - A method and apparatus multiplies a past sample a time lag τ older than a current sample by a quantized multiplier ρ′ 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 ρ′ 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 ρ′ is not smaller than 0.2. A multiplier ρ is coded by a multiplier coder and the multiplier ρ′ obtained by decoding the multiplier ρ is outputted. The process is performed for each frame. | 07-07-2011 |
| 20110172995 | METHOD FOR SPEECH CODING, METHOD FOR SPEECH DECODING AND THEIR APPARATUSES - A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal. In speech coding method according to a code-excited linear prediction (CELP) speech coding, a noise level of a speech in a concerning coding period is evaluated by using a code or coding result of at least one of spectrum information, power information, and pitch information, and various excitation codebooks are used based on an evaluation result | 07-14-2011 |
| 20110184733 | SYSTEM AND METHOD FOR ENCODING AND DECODING PULSE INDICES - Methods, and corresponding codec-containing devices are provided that have source coding schemes for encoding a component of an excitation. In some cases, the source coding scheme is an enumerative source coding scheme, while in other cases the source coding scheme is an arithmetic source coding scheme. In some cases, the source coding schemes are applied to encode a fixed codebook component of the excitation for a codec employing codebook excited linear prediction, for example an AMR-WB (Adaptive Multi-Rate-Wideband) speech codec. | 07-28-2011 |
| 20110202336 | FIXED CODEBOOK SEARCHING APPARATUS AND FIXED CODEBOOK SEARCHING METHOD - A fixed codebook searching apparatus, includes a convolution operator, implemented by at least one processor, that convolves an impulse response of a perceptually weighted synthesis filter with an impulse response vector that has values at negative times, to generate a second impulse response vector that has values at negative times. A matrix generator, implemented by at least one processor, generates a Toeplitz-type convolution matrix using the second impulse response vector generated by the convolution operator. A searcher, implemented by at least one processor, performs a codebook search by maximizing a term using the Toeplitz-type convolution matrix. | 08-18-2011 |
| 20110264448 | METHOD FOR HIGH QUALITY AUDIO TRANSCODING - A method and apparatus for a voice transcoder that converts a bitstream representing frames of data encoded according to a first voice compression standard to a bitstream representing frames of data according to a second voice compression standard using perceptual weighting that uses tuned weighting factors, such that the bitstream of a second voice compression standard to produce a higher quality decoded voice signal than a comparable tandem transcoding solution. The method includes pre-computing weighting factors for a perceptual weighting filter optimized to a specific source and destination codec pair, pre-configuring the transcoding strategies, mapping CELP parameters in the CELP parameter space according to the selected coding strategy, performing Linear Prediction analysis if specified by the transcoding strategy, perceptually weighting the speech using with tuned weighting factors, and searching for adaptive codebook and fixed-codebook parameters to obtain a quantized set of destination codec parameters. | 10-27-2011 |
| 20110270608 | METHOD AND APPARATUS FOR RECEIVING AN ENCODED SPEECH SIGNAL - A method and apparatus for processing speech in a wireless communication system uses CELP speech encoded signals. A decoder receives encoded speech including a code index, a code index gain, a pitch lag, a pitch gain, and a line spectral pair (LSP) index. An innovation codevector and an adaptive codevector are determined and scaled. An excitation sequence is generated. Reconstructed speech is then output based on the excitation sequence and LSP index. | 11-03-2011 |
| 20110295600 | Apparatus and method determining weighting function for linear prediction coding coefficients quantization - An apparatus determining a weighting function for line prediction coding coefficients quantization converts a linear prediction coding (LPC) coefficient of an input signal into one of a line spectral frequency (LSF) coefficient and an immitance spectral frequency (ISF) coefficient and determines a weighting function associated with one of an importance of the ISF coefficient and importance of the LSF coefficient using one of the converted ISF coefficient and the converted LSF coefficient. | 12-01-2011 |
| 20110301946 | TONE DETERMINATION DEVICE AND TONE DETERMINATION METHOD - Disclosed is a tone determination device that determines the tonality of an input signal using correlations between the frequency components of a current frame with the frequency components of the preceding frame, such that the tone determination device is able to decrease the calculation complexity. In the device, a vector coupling unit ( | 12-08-2011 |
| 20110301947 | SYSTEMS, PROCESSES AND INTEGRATED CIRCUITS FOR RATE AND/OR DIVERSITY ADAPTATION FOR PACKET COMMUNICATIONS - Packets of real-time information are sent with a source rate greater than zero kilobits per second, and a time or path or combined time/path diversity rate initially being zero kilobits per second. This results in a quality of service QoS, optionally measured at the sender or the receiver. When the QoS is on an unacceptable side of a threshold of acceptability, the sender sends diversity packets at an increased rate. Increasing the diversity rate while either reducing or maintaining the overall transmission rate is new. CELP-based multiple-description data partitioning sends the base or important information plus a subset of fixed excitation in one packet and sends the base or important information plus the complementary subset of fixed excitation in another packet. Reconstruction produces acceptable quality when only one of the two packets is received and better quality when both packets are received. Reconstruction provides for single and multiple lost packets. | 12-08-2011 |
| 20110313761 | METHOD FOR ENCODING SIGNAL, AND METHOD FOR DECODING SIGNAL - The present disclosure relates to a method, apparatus, and system for encoding and decoding signals. The encoding method includes: converting a first-domain signal into a second-domain signal; performing Linear Prediction (LP) processing and Long-Term Prediction (LTP) processing for the second-domain signal; obtaining a long-term flag according to decision criteria; obtaining a second-domain contribution signal according to the LP processing result and the LTP processing result when the long-term flag is a first flag; obtaining a second-domain contribution signal according to the LP processing result when the long-term flag is a second flag; converting the second-domain contribution signal into a first-domain contribution signal, and calculating a first-domain predictive residual signal; and outputting a bit stream that includes the first-domain predictive residual signal. With the present disclosure, a subsequent encoding or decoding process is performed adaptively according to the long-term flag; when the long-term flag is the second flag, it is not necessary to consider the LTP processing result, thus improving the compression performance of a codec. | 12-22-2011 |
| 20110320195 | METHOD, APPARATUS AND SYSTEM FOR LINEAR PREDICTION CODING ANALYSIS - The present invention relates to communication technologies and discloses a method, an apparatus and a system for Linear Prediction Coding (LPC) analysis to improve LPC prediction performance and simplify analysis operation. The method includes: obtaining signal feature information of at least one sample point of input signals; comparing and analyzing the signal feature information to obtain an analysis result; selecting a window function according to the analysis result to perform adaptive windowing for the input signals and obtain windowed signals; and processing the windowed signals to obtain an LPC coefficient for linear prediction. The embodiments of the present invention are applicable to LPC. | 12-29-2011 |
| 20120095756 | Apparatus and method for determining weighting function having low complexity for linear predictive coding (LPC) coefficients quantization - Proposed is a method and apparatus for determining a weighting function for quantizing a linear predictive coding (LPC) coefficient and having a low complexity. The weighting function determination apparatus may convert an LPC coefficient of a mid-subframe of an input signal to one of a immitance spectral frequency (ISF) coefficient and a line spectral frequency (LSF) coefficient, and may determine a weighting function associated with an importance of the ISF coefficient or the LSF coefficient based on the converted ISF coefficient or LSF coefficient. | 04-19-2012 |
| 20120095757 | AUDIO SIGNAL BANDWIDTH EXTENSION IN CELP-BASED SPEECH CODER - A method for decoding an audio signal having a bandwidth that extends beyond a bandwidth of a CELP excitation signal in an audio decoder including a CELP-based decoder element. The method includes obtaining a second excitation signal having an audio bandwidth extending beyond the audio bandwidth of the CELP excitation signal, obtaining a set of signals by filtering the second excitation signal with a set of bandpass filters, scaling the set of signals using a set of energy-based parameters, and obtaining a composite output signal by combining the scaled set of signals with a signal based on the audio signal decoded by the CELP-based decoder element. | 04-19-2012 |
| 20120095758 | AUDIO SIGNAL BANDWIDTH EXTENSION IN CELP-BASED SPEECH CODER - A method for decoding an audio signal in a decoder having a CELP-based decoder element including a fixed codebook component, at least one pitch period value, and a first decoder output, wherein a bandwidth of the audio signal extends beyond a bandwidth of the CELP-based decoder element. The method includes obtaining an up-sampled fixed codebook signal by up-sampling the fixed codebook component to a higher sample rate, obtaining an up-sampled excitation signal based on the up-sampled fixed codebook signal and an up-sampled pitch period value, and obtaining a composite output signal based on the up-sampled excitation signal and an output signal of the CELP-based decoder element, wherein the composite output signal includes a bandwidth portion that extends beyond a bandwidth of the CELP-based decoder element. | 04-19-2012 |
| 20120116757 | METHOD AND APPARATUS FOR ENCODING AND DECODING HIGH FREQUENCY SIGNAL - Provided are a method and apparatus for encoding and decoding a high frequency signal by using a low frequency signal. The high frequency signal can be encoded by extracting a coefficient by linear predicting a high frequency signal, and encoding the coefficient, generating a signal by using the extracted coefficient and a low frequency signal, and encoding the high frequency signal by calculating a ratio between the high frequency signal and an energy value of the generated signal. Also, the high frequency signal can be decoded by decoding a coefficient, which is extracted by linear predicting a high frequency signal, and a low frequency signal, and generating a signal by using the decoded coefficient and the decoded low frequency signal, and adjusting the generated signal by decoding a ratio between the generated signal and an energy value of the high frequency signal. | 05-10-2012 |
| 20120143602 | SPEECH DECODER AND METHOD FOR DECODING SEGMENTED SPEECH FRAMES - A method for decoding segmented speech frames includes: generating parameters of a segmented current speech frame by using parameters of a segmented previous speech frame; and decoding a speech frame by using the parameters of the current speech frame, which are generated in the generating of the parameters of the segmented current speech frame. | 06-07-2012 |
| 20120150535 | METHOD FOR SPEECH CODING, METHOD FOR SPEECH DECODING AND THEIR APPARATUSES - A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal. In speech coding method according to a code-excited linear prediction (CELP) speech coding, a noise level of a speech in a concerning coding period is evaluated by using a code or coding result of at least one of spectrum information, power information, and pitch information, and various excitation codebooks are used based on an evaluation result | 06-14-2012 |
| 20120209599 | DEVICE AND METHOD FOR QUANTIZING THE GAINS OF THE ADAPTIVE AND FIXED CONTRIBUTIONS OF THE EXCITATION IN A CELP CODEC - A device and method for quantizing a gain of a fixed contribution of an excitation in a frame, including sub-frames, of a coded sound signal. The gain is estimated in a sub-frame using a frame classification parameter, and is then quantized in the sub-frame using the estimated gain. The device and method can be used in jointly quantizing gains of adaptive and fixed contributions of an excitation. For retrieving a quantized gain of a fixed contribution of an excitation in a sub-frame, the gain of the fixed excitation contribution is estimated using a frame classification parameter, a gain codebook supplies a correction factor in response to a received, gain codebook index, and a multiplier multiplies the estimated gain by the correction factor to provide the quantized gain. | 08-16-2012 |
| 20120209600 | INTEGRATED VOICE/AUDIO ENCODING/DECODING DEVICE AND METHOD WHEREBY THE OVERLAP REGION OF A WINDOW IS ADJUSTED BASED ON THE TRANSITION INTERVAL - A Unified Speech and Audio Codec (USAC) for adjusting an overlap area of a window based on a transition is provided. To increase an encoding efficiency, encoding may be performed by overlapping relatively long windows. Additionally, when a transition is generated between frames, an overlap area of a window may be reduced based on the transition, thereby preventing a noise from occurring due to the transition. | 08-16-2012 |
| 20120215529 | Speech Enhancement - A method for processing and iteratively enhancing and estimating a source audio signal received at two audio receivers is provided. In one embodiment, the method involves the use of codebook constrained iterative binaural Wiener filter (CCIBWF). The provided CCIBWF embodiment can improve the quality of speech received at two audio receivers both in terms of noise reduction and speech intelligibility. In one embodiment, optimum speech enhancement performance was achieved within two iterations of the CCIBWF scheme. Further, the embodiment of the CCIBWF scheme introduces minimal distortion to the binaural cues, such as the interaural time delay cues, thereby preserving localization information of the audio source. The embodiment of the CCIBWF is also able to relatively accurately track the Time Delay of Arrival (TDOA) when the audio source is moving. This ensures that the performance of the CCIBWF scheme is not significantly degraded due to the selection of wrong codebooks. | 08-23-2012 |
| 20120226496 | APPARATUS FOR PROCESSING A SIGNAL AND METHOD THEREOF - An apparatus for processing a signal and method thereof are disclosed. The present invention includes receiving coding mode information indicating a speech coding scheme or an audio coding scheme, linear prediction coding degree information indicating a linear prediction coding degree, and the signal including at least one of a speech signal and an audio signal; decoding the signal according to the speech coding scheme or the audio coding scheme based on the coding mode information; decoding linear prediction coding coefficients of the signal based on the linear prediction coding degree information; and generating an output signal by applying the decoded linear prediction coding coefficients to the decoded signal. In this case, the linear prediction coding degree information is determined based on a variation of a value of an LPC residual generated from performing the linear prediction coding on the signal. | 09-06-2012 |
| 20120232888 | APPARATUS AND METHOD FOR CONCEALING FRAME ERASURE AND VOICE DECODING APPARATUS AND METHOD USING THE SAME - An apparatus and method for concealing frame erasure and a voice decoding apparatus and method using the same. The frame erasure concealment apparatus includes: a parameter extraction unit determining whether there is an erased frame in a voice packet, and extracting an excitement signal parameter and a line spectrum pair parameter of a previous good frame; and an erasure frame concealment unit, if there is an erased frame, restoring the excitement signal and line spectrum pair parameter of the erased frame by using a regression analysis from the excitement signal and line spectrum pair parameter of the previous good frame. According to the method and apparatus, by predicting and restoring the parameter of the erased frame through the regression analysis, the quality of the restored voice signal can be enhanced and the algorithm can be simplified. | 09-13-2012 |
| 20120245930 | METHOD AND APPARATUS FOR ENCODING A SPEECH SIGNAL - According to the present invention, a linear prediction filter coefficient of a current frame is acquired from an input signal using linear prediction, a quantized spectrum candidate vector of the current frame, corresponding to the linear prediction filter coefficient of the current frame, is acquired on the basis of first best information, and the quantized spectrum candidate vector of the current frame and the quantized spectrum vector of the previous frame are interpolated. Accordingly, in contrast to conventional phased optimization techniques, optimum parameters which minimize quantization errors, can be obtained. | 09-27-2012 |
| 20120253797 | MULTI-MODE AUDIO CODEC AND CELP CODING ADAPTED THEREFORE - In an embodiment, bitstream elements of sub-frames are encoded differentially to a global gain value so that a change of the global gain value results in an adjustment of an output level of the decoded representation of the audio content. Concurrently, the differential coding saves bits. Even further, the differential coding enables the lowering of the burden of globally adjusting the gain of an encoded bitstream. In another embodiment, a global gain control across CELP coded frames and transform coded frames is achieved by co-controlling the gain of the codebook excitation of the CELP codec, along with a level of the transform or inverse transform of the transform coded frames. In another embodiment, the gain value determination in CELP coding is performed in the weighted domain of the excitation signal. | 10-04-2012 |
| 20120259624 | SYSTEMS, PROCESSES AND INTEGRATED CIRCUITS FOR RATE AND/OR DIVERSITY ADAPTATION FOR PACKET COMMUNICATIONS - Packets of real-time information are sent with a source rate greater than zero kilobits per second, and a time or path or combined time/path diversity rate initially being zero kilobits per second. This results in a quality of service QoS, optionally measured at the sender or the receiver. When the QoS is on an unacceptable side of a threshold of acceptability, the sender sends diversity packets at an increased rate. Increasing the diversity rate while either reducing or maintaining the overall transmission rate is new. CELP-based multiple-description data partitioning sends the base or important information plus a subset of fixed excitation in one packet and sends the base or important information plus the complementary subset of fixed excitation in another packet. Reconstruction produces acceptable quality when only one of the two packets is received and better quality when both packets are received. Reconstruction provides for single and multiple lost packets. | 10-11-2012 |
| 20120271629 | APPARATUS FOR QUANTIZING LINEAR PREDICTIVE CODING COEFFICIENTS, SOUND ENCODING APPARATUS, APPARATUS FOR DE-QUANTIZING LINEAR PREDICTIVE CODING COEFFICIENTS, SOUND DECODING APPARATUS, AND ELECTRONIC DEVICE THEREFORE - A quantizing apparatus is provided that includes a quantization path determiner that determines a path from a first path not using inter-frame prediction and a second path using the inter-frame prediction, as a quantization path of an input signal, based on a criterion before quantization of the input signal; a first quantizer that quantizes the input signal, if the first path is determined as the quantization path of the input signal; and a second quantizer that quantizes the input signal, if the second path is determined as the quantization path of the input signal. | 10-25-2012 |
| 20120278069 | METHOD OF QUANTIZING LINEAR PREDICTIVE CODING COEFFICIENTS, SOUND ENCODING METHOD, METHOD OF DE-QUANTIZING LINEAR PREDICTIVE CODING COEFFICIENTS, SOUND DECODING METHOD, AND RECORDING MEDIUM AND ELECTRONIC DEVICE THEREFOR - A quantizing method is provided that includes quantizing an input signal by selecting one of a first quantization scheme not using an inter-frame prediction and a second quantization scheme using the inter-frame prediction, in consideration of one or more of a prediction mode, a predictive error and a transmission channel state. | 11-01-2012 |
| 20120284020 | SYSTEM AND METHOD OF SPEECH COMPRESSION USING AN INTER FRAME PARAMETER CORRELATION - The disclosure provides a speech encoder, decoder, speech processor and methods of encoding and decoding speech. In one embodiment, the speech encoder includes: (1) a speech frame generator configured to form a speech frame from an input speech signal, the speech frame having a length of multiple samples, (2) a speech frame processor configured to determine if the speech frame is a subsequent voiced frame of a group of consecutive voiced frames and, based thereon, perform speech analysis of the subsequent voiced frame; and (3) a speech frame coder configured to perform, if the speech frame is a subsequent voiced frame, differential coding of speech parameters of the subsequent voiced frame with respect to previous speech parameters of the previous voiced frame of the consecutive voiced frames. | 11-08-2012 |
| 20120290295 | Transform-Domain Codebook In A Celp Coder And Decoder - Codebook Arrangement for use in coding an input sound signal includes First and Second Codebook Stages. First Codebook Stage includes one of a time-domain CELP codebook and a transform-domain codebook. Second Codebook Stage follows the first codebook stage and includes the other of the time-domain CELP codebook and the transform-domain codebook. Codebook Stage includes an adaptive codebook may be provided before First Codebook Stage. A selector may be provided to select an order of the time-domain CELP codebook and the transform-domain codebook in First and Second Codebook Stages, respectively, as a function of characteristics of the input sound signal. The selector may also be responsive to both the characteristics of the input sound signal and a bit rate of the codec using Codebook Arrangement to bypass Second Codebook Stage. Codebook Arrangement can be used in a coder of an input sound signal. | 11-15-2012 |
| 20120303362 | NOISE-ROBUST SPEECH CODING MODE CLASSIFICATION - A method of noise-robust speech classification is disclosed. Classification parameters are input to a speech classifier from external components. Internal classification parameters are generated in the speech classifier from at least one of the input parameters. A Normalized Auto-correlation Coefficient Function threshold is set. A parameter analyzer is selected according to a signal environment. A speech mode classification is determined based on a noise estimate of multiple frames of input speech. | 11-29-2012 |
| 20130030798 | METHOD AND APPARATUS FOR AUDIO CODING AND DECODING - An encoder and decoder for processing an audio signal including generic audio and speech frames are provided herein. During operation, two encoders are utilized by the speech coder, and two decoders are utilized by the speech decoder. The two encoders and decoders are utilized to process speech and non-speech (generic audio) respectively. During a transition between generic audio and speech, parameters that are needed by the speech decoder for decoding frame of speech are generated by processing the preceding generic audio (non-speech) frame for the necessary parameters. Because necessary parameters are obtained by the speech coder/decoder, the discontinuities associated with prior-art techniques are reduced when transitioning between generic audio frames and speech frames. | 01-31-2013 |
| 20130030799 | ACOUSTIC SHOCK PROTECTION DEVICE AND METHOD THEREOF - An acoustic shock protection device includes a prediction gain estimator and an audio compressor. The prediction gain estimator is configured to analyze a plurality of linear prediction coefficients of an audio signal and determine a category of the audio signal. The audio compressor is coupled to the prediction gain estimator, and the audio compressor is configured to adjust a signal level of the audio signal according to the category of the audio signal. | 01-31-2013 |
| 20130030800 | ADAPTIVE VOICE INTELLIGIBILITY PROCESSOR - Systems and methods for adaptively processing speech to improve voice intelligibility are described. These systems and methods can adaptively identify and track formant locations, thereby enabling formants to be emphasized as they change. As a result, these systems and methods can improve near-end intelligibility, even in noisy environments. The systems and methods can be implemented in Voice-over IP (VoIP) applications, telephone and/or video conference applications (including on cellular phones, smart phones, and the like), laptop and tablet communications, and the like. The systems and methods can also enhance non-voiced speech, which can include speech generated without the vocal track, such as transient speech. | 01-31-2013 |
| 20130046534 | MECHANISM FOR DYNAMIC SIGNALING OF ENCODER CAPABILITIES - The present disclosure provides systems and methods for dynamically signaling encoder capabilities of vocoders of corresponding communication nodes. In one embodiment, during a call between a first communication node and a second communication node, a control node (e.g., base station controller or mobile switching center) for the first communication node sends capability information for a voice encoder of a vocoder of the first communication node to a control node for the second communication node. As a result, the second communication node is enabled to select and request a preferred encoder mode for the voice encoder of the vocoder of the first communication node based on the capabilities of the voice encoder of the vocoder of the first communication node. | 02-21-2013 |
| 20130054230 | ESTIMATION OF SPEECH ENERGY BASED ON CODE EXCITED LINEAR PREDICTION (CELP) PARAMETERS EXTRACTED FROM A PARTIALLY-DECODED CELP-ENCODED BIT STREAM AND APPLICATIONS OF SAME - Methods, systems, and non-transitory computer readable media for estimating speech energy of an encoded bit stream based on coding parameters extracted from the partially-decoded bit stream are disclosed. According to one aspect, a method for estimating speech energy of an encoded bit stream based on coding parameters extracted from the partially-decoded bit stream includes receiving a CELP-encoded bit stream, partially decoding the bit stream, and estimating the speech energy of the bit stream based a set of four or fewer CELP parameters extracted from the partially decoded bit stream. According to another aspect, a method for estimating speech energy of an encoded bit stream based on coding parameters extracted from the partially-decoded bit stream includes receiving a CELP-encoded bit stream, partially decoding the bit stream, extracting at least one CELP parameter from the partially-decoded bit stream, and estimating the speech energy of the bit stream based on the extracted at least one CELP parameter without calculating a linear prediction coding (LPC) filter response energy. | 02-28-2013 |
| 20130096913 | METHOD AND APPARATUS FOR ADAPTIVE MULTI RATE CODEC - There is provided an apparatus and method for encoding a speech signal. The encoding comprises: receiving a plurality of current samples of the speech signals; extrapolating a plurality of look-ahead samples from the current samples; and performing linear prediction analysis using the current samples and the extrapolated look-ahead samples. | 04-18-2013 |
| 20130110507 | Adding Second Enhancement Layer to CELP Based Core Layer | 05-02-2013 |
| 20130132076 | SMART REJECTER FOR KEYBOARD CLICK NOISE - According to various embodiments of the invention, a new and effective keyboard click noise reduction scheme is presented. The keyboard click noise reduction scheme may have various processing units including: Dynamic Signal Modeler, Smart Model Selector, Adaptive Filtering Module, Keyboard/Impulse Noise and Voice Activity Detectors, and a Post-Processing Unit. By adaptively changing the coefficients of the proposed adaptive filter through minimizing the output energy, the scheme can provide the target signal/voice with nearly zero keyboard click noise. The scheme could be used in real-time to minimize keyboard click noise or any kind of unwanted noise, especially noise having transient impulse characteristics. | 05-23-2013 |
| 20130179159 | SYSTEMS AND METHODS FOR DETECTING OVERFLOW - A method for detecting overflow on an electronic device is described. The method includes determining a linear predictive coding synthesis filter gain. The method further includes determining whether overflow is detected based on the linear predictive coding synthesis filter gain and a fixed codebook gain. The method further includes determining a scaling factor if overflow is detected. | 07-11-2013 |
| 20130185062 | SYSTEMS, METHODS, APPARATUS, AND COMPUTER-READABLE MEDIA FOR CRITICALITY THRESHOLD CONTROL - Systems, methods, and apparatus as disclosed herein may be implemented to adjust criticality thresholds for speech frames, based on channel conditions. Such a threshold may be used to control retransmission frequency in response to changes in channel state. | 07-18-2013 |
| 20130185063 | MULTIPLE CODING MODE SIGNAL CLASSIFICATION - Improved audio classification is provided for encoding applications. An initial classification is performed, followed by a finer classification, to produce speech classifications and music classifications with higher accuracy and less complexity than previously available. Audio is classified as speech or music on a frame by frame basis. If the frame is classified as music by the initial classification, that frame undergoes a second, finer classification to confirm that the frame is music and not speech (e.g., speech that is tonal and/or structured that may not have been classified as speech by the initial classification). Depending on the implementation, one or more parameters may be used in the finer classification. Example parameters include voicing, modified correlation, signal activity, and long term pitch gain. | 07-18-2013 |
| 20130204615 | METHOD FOR SPEECH CODING, METHOD FOR SPEECH DECODING AND THEIR APPARATUSES - A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal. In speech coding method according to a code-excited linear prediction (CELP) speech coding, a noise level of a speech in a concerning coding period is evaluated by using a code or coding result of at least one of spectrum information, power information, and pitch information, and various excitation codebooks are used based on an evaluation result | 08-08-2013 |
| 20130226569 | METHODS EMPLOYING PHASE STATE ANALYSIS FOR USE IN SPEECH SYNTHESIS AND RECOGNITION - A computer-implemented method for automatically analyzing, predicting, and/or modifying acoustic units of prosodic human speech utterances for use in speech synthesis or speech recognition. Possible steps include: initiating analysis of acoustic wave data representing the human speech utterances, via the phase state of the acoustic wave data; using one or more phase state defined acoustic wave metrics as common elements for analyzing, and optionally modifying, pitch, amplitude, duration, and other measurable acoustic parameters of the acoustic wave data, at predetermined time intervals; analyzing acoustic wave data representing a selected acoustic unit to determine the phase state of the acoustic unit; and analyzing the acoustic wave data representing the selected acoustic unit to determine at least one acoustic parameter of the acoustic unit with reference to the determined phase state of the selected acoustic unit. Also included are systems for implementing the described and related methods. | 08-29-2013 |
| 20130226570 | APPARATUS AND METHOD FOR PROCESSING AN AUDIO SIGNAL AND FOR PROVIDING A HIGHER TEMPORAL GRANULARITY FOR A COMBINED UNIFIED SPEECH AND AUDIO CODEC (USAC) - An apparatus for processing an audio signal is provided. The apparatus has a signal processor and a configurator. The configurator is adapted to configure the signal processor based on configuration information such that a configurable upsampling factor is equal to a first upsampling value when a first ratio of the second configurable number of samples to a first configurable number of samples has a first ratio value. Moreover, the configurator is adapted to configure the signal processor such that the configurable upsampling factor is equal to a different second upsampling value, when a different second ratio of the second configurable number of samples to the first configurable number of samples has a different second ratio value. The first or the second ratio value is not an integer value. | 08-29-2013 |
| 20130238323 | Method of Accessing a Dial-Up Service - A method of accessing a dial-up service is disclosed. An example method of providing access to a service includes receiving a first speech signal from a user to form a first utterance; recognizing the first utterance using speaker independent speaker recognition; requesting the user to enter a personal identification number; and when the personal identification number is valid, receiving a second speech signal to form a second utterance and providing access to the service. | 09-12-2013 |
| 20130246057 | SYSTEMS, PROCESSES AND INTEGRATED CIRCUITS FOR RATE AND/OR DIVERSITY ADAPTATION FOR PACKET COMMUNICATIONS - Packets of real-time information are sent with a source rate greater than zero kilobits per second, and a time or path or combined time/path diversity rate initially being zero kilobits per second. This results in a quality of service QoS, optionally measured at the sender or the receiver. When the QoS is on an unacceptable side of a threshold of acceptability, the sender sends diversity packets at an increased rate. Increasing the diversity rate while either reducing or maintaining the overall transmission rate is new. CELP-based multiple-description data partitioning sends the base or important information plus a subset of fixed excitation in one packet and sends the base or important information plus the complementary subset of fixed excitation in another packet. Reconstruction produces acceptable quality when only one of the two packets is received and better quality when both packets are received. Reconstruction provides for single and multiple lost packets. | 09-19-2013 |
| 20130253922 | PARAMETER DECODING APPARATUS AND PARAMETER DECODING METHOD - A parameter decoding apparatus includes a prediction residue decoder that finds a quantized prediction residue based on encoded information included in a current frame subject to decoding and a moving-average predictor produces a predicted parameter by multiplying a predictive coefficient with a past quantized prediction residue. An adder decodes a parameter by adding the quantized prediction residue and the predicted parameter, wherein the prediction residue decoder, when the current frame is erased, finds a current-frame quantized prediction residue from a weighted linear sum of a parameter decoded in the past and a future-frame quantized prediction residue. | 09-26-2013 |
| 20130268266 | Method and Apparatus for Generating a Candidate Code-Vector to Code an Informational Signal | 10-10-2013 |
| 20130282368 | APPARATUS AND METHOD FOR ENCODING/DECODING FOR HIGH FREQUENCY BANDWIDTH EXTENSION - A method and apparatus for performing coding and decoding for high-frequency bandwidth extension. The coding apparatus may down-sample an input signal, perform core coding on the down-sampled input signal, perform frequency transformation on the input signal, and perform bandwidth extension coding by using a base signal of the input signal in a frequency domain. | 10-24-2013 |
| 20130289981 | LOW-DELAY SOUND-ENCODING ALTERNATING BETWEEN PREDICTIVE ENCODING AND TRANSFORM ENCODING - An encoder and a method for encoding a digital signal are provided. The method includes encoding a preceding frame of samples of the digital signal according to a predictive encoding process, and encoding a current frame of samples of the digital signal according to a transform encoding process. The method is implemented such that a first portion of the current frame is also encoded by predictive encoding that is limited relative to the predictive encoding of the preceding frame by reusing at least one parameter of the predictive encoding of the preceding frame and only encoding the parameters of said first portion of the current frame that are not reused. A decoder and a decoding method are also provided, which correspond to the described encoding method. | 10-31-2013 |
| 20130317813 | SPECTRAL ENVELOPE CODING OF ENERGY ATTACK SIGNAL - MDCT or FFT-based audio coding algorithms often have the problem named here spectral pre-echoes when coding an energy attack signal. This invention presents several possibilities to avoid the spectral pre-echoes existing in decoded signal segment before the energy attack point. The spectral envelope before the attack point can be improved by performing spectrum smoothing, replacing the segment of having spectral pre-echoes or filtering the segment with a combined filter obtained by doing LPC analysis. | 11-28-2013 |
| 20130317814 | ENCODING METHOD, DECODING METHOD, ENCODER, DECODER, PROGRAM, AND RECORDING MEDIUM - In encoding, the number of bits to be assigned to codes corresponding to noise or a pulse sequence obtained according to prediction analysis applied to time series signals included in a predetermined time interval is switched according to whether an index that indicates a level of periodicity and/or stationarity of input time series signals satisfies a condition that indicates high periodicity and/or high stationarity or a condition that indicates low periodicity and/or low stationarity, to acquire the codes corresponding to the noise and the pulse sequence. In decoding, a decoding mode for codes corresponding to noise or a pulse sequence included in codes corresponding to a predetermined time interval is switched according to the same criterion as that described above to decode the codes corresponding to the noise or the pulse sequence to acquire noise or a pulse sequence corresponding to the predetermined time interval. | 11-28-2013 |
| 20130332152 | APPARATUS AND METHOD FOR ERROR CONCEALMENT IN LOW-DELAY UNIFIED SPEECH AND AUDIO CODING - An apparatus for generating spectral replacement values for an audio signal has a buffer unit for storing previous spectral values relating to a previously received error-free audio frame. Moreover, the apparatus includes a concealment frame generator for generating the spectral replacement values, when a current audio frame has not been received or is erroneous. The previously received error-free audio frame has filter information, the filter information having associated a filter stability value indicating a stability of a prediction filter. The concealment frame generator is adapted to generate the spectral replacement values based on the previous spectral values and based on the filter stability value. | 12-12-2013 |
| 20130332153 | LINEAR PREDICTION BASED CODING SCHEME USING SPECTRAL DOMAIN NOISE SHAPING - An encoding concept which is linear prediction based and uses spectral domain noise shaping is rendered less complex at a comparable coding efficiency in terms of, for example, rate/distortion ratio, by using the spectral decomposition of the audio input signal into a spectrogram having a sequence of spectra for both linear prediction coefficient computation as well as spectral domain shaping based on the linear prediction coefficients. The coding efficiency may remain even if such a lapped transform is used for the spectral decomposition which causes aliasing and necessitates time aliasing cancellation such as critically sampled lapped transforms such as an MDCT. | 12-12-2013 |
| 20130339012 | SPEECH/AUDIO ENCODING APPARATUS, SPEECH/AUDIO DECODING APPARATUS, AND METHODS THEREOF - Provided is a speech/audio encoding apparatus with which it is possible to code a significant frequency domain region with high precision, and to enable high audio quality. A speech/audio encoding apparatus codes a linear prediction coefficient. A significant frequency domain region detection unit identifies a frequency domain region which is aurally significant from the linear prediction coefficient. A frequency domain region repositioning unit repositions the significant frequency domain region which is identified by the significant frequency domain region detection unit. A bit allocation computation unit determines a coding bit allocation on the basis of the significant frequency domain region which is repositioned by the frequency domain region repositioning unit. | 12-19-2013 |
| 20140012571 | METHOD AND APPARATUS FOR PROVIDING SIGNAL PROCESSING COEFFICIENTS - A method and apparatus for providing signal processing coefficients for processing an input signal at a predetermined signal processing sampling rate, wherein the input signal is received at an input signal sampling rate, the method comprising the steps of computing a correlation or covariance function based on the received input signal at the input signal sampling rate to provide correlation or covariance coefficients at the input signal sampling rate, re-sampling the computed correlation or covariance coefficients having the input signal sampling rate to provide correlation or covariance coefficients at the predetermined signal processing sampling rate, and calculating the signal processing coefficients based on the correlation or covariance coefficients at the predetermined signal processing sampling rate. | 01-09-2014 |
| 20140032213 | ADAPTIVE TIME/FREQUENCY-BASED AUDIO ENCODING AND DECODING APPARATUSES AND METHODS - Adaptive time/frequency-based audio encoding and decoding apparatuses and methods. The encoding apparatus includes a transformation & mode determination unit to divide an input audio signal into a plurality of frequency-domain signals and to select a time-based encoding mode or a frequency-based encoding mode for each respective frequency-domain signal, an encoding unit to encode each frequency-domain signal in the respective encoding mode, and a bitstream output unit to output encoded data, division information, and encoding mode information for each respective frequency-domain signal. In the apparatuses and methods, acoustic characteristics and a voicing model are simultaneously applied to a frame, which is an audio compression processing unit. As a result, a compression method effective for both music and voice can be produced, and the compression method can be used for mobile terminals that require audio compression at a low bit rate. | 01-30-2014 |
| 20140052439 | METHOD AND APPARATUS FOR POLYPHONIC AUDIO SIGNAL PREDICTION IN CODING AND NETWORKING SYSTEMS - A method, device, and apparatus provide the ability to predict a portion of a polyphonic audio signal for compression and networking applications. The solution involves a framework of a cascade of long term prediction filters, which by design is tailored to account for all periodic components present in a polyphonic signal. This framework is complemented with a design method to optimize the system parameters. Specialization may include specific techniques for coding and networking scenarios, where the potential of each enhanced prediction is realized to considerably improve the overall system performance for that application. One specific technique provides enhanced inter-frame prediction for the compression of polyphonic audio signals, particularly at low delay. Another specific technique provides improved frame loss concealment capabilities to combat packet loss in audio communications. | 02-20-2014 |
| 20140074460 | SCALABLE CODE EXCITED LINEAR PREDICTION BITSTREAM - The present invention provides for methods and apparatuses for processing audio data. In one embodiment, there is a provided a method for achieving bitstream scalability in a multi-channel audio encoder, said method comprising receiving audio input data; organizing said input data by a Code Excited Linear Predictor (CELP) processing module for further encoding by arranging said data according to significance of data, where more significant data is placed ahead of less significant data; and providing a scalable output bitstream. The organized CELP data comprises of a first part and a second part. The first part comprises a frame header, sub frame parameters and innovation vector quantization data from the first frame from all channels. The innovation vector quantization data from the first frames from all channels is arranged according to channel number. | 03-13-2014 |
| 20140074461 | METHOD AND APPARATUS FOR ENCODING/DECODING SPEECH SIGNAL USING CODING MODE - An apparatus and a method to encode and decode a speech signal using an encoding mode are provided. An encoding apparatus may select an encoding mode of a frame included in an input speech signal, and encode a frame having an unvoiced mode for an unvoiced speech as the selected encoding mode. | 03-13-2014 |
| 20140129214 | Method and Apparatus for Generating a Candidate Code-Vector to Code an Informational Signal | 05-08-2014 |
| 20140188465 | CODING MODE DETERMINATION METHOD AND APPARATUS, AUDIO ENCODING METHOD AND APPARATUS, AND AUDIO DECODING METHOD AND APPARATUS - Provided are a method and an apparatus for determining an encoding mode for improving the quality of a reconstructed audio signal. A method of determining an encoding mode includes determining one from among a plurality of encoding modes including a first encoding mode and a second encoding mode as an initial encoding mode in correspondence to characteristics of an audio signal, and if there is an error in the determination of the initial encoding mode, generating a modified encoding mode by modifying the initial encoding mode to a third encoding mode. | 07-03-2014 |
| 20140214413 | SYSTEMS, METHODS, APPARATUS, AND COMPUTER-READABLE MEDIA FOR ADAPTIVE FORMANT SHARPENING IN LINEAR PREDICTION CODING - A method of processing an audio signal includes determining an average signal-to-noise ratio for the audio signal over time. The method includes, based on the determined average signal-to-noise ratio, a formant-sharpening factor is determined. The method also includes applying a filter that is based on the determined formant-sharpening factor to a codebook vector that is based on information from the audio signal. | 07-31-2014 |
| 20140236587 | SYSTEMS AND METHODS FOR CONTROLLING AN AVERAGE ENCODING RATE - A method for controlling an average encoding rate by an electronic device is described. The method includes obtaining a speech signal. The method also includes determining a first average rate. The method further includes determining a first threshold based on the first average rate. The method additionally includes controlling the average encoding rate by determining at least one other threshold based on the first threshold. The method also includes sending an encoded speech signal. | 08-21-2014 |
| 20140236588 | SYSTEMS AND METHODS FOR MITIGATING POTENTIAL FRAME INSTABILITY - A method for mitigating potential frame instability by an electronic device is described. The method includes obtaining a frame subsequent in time to an erased frame. The method also includes determining whether the frame is potentially unstable. The method further includes applying a substitute weighting value to generate a stable frame parameter if the frame is potentially unstable. | 08-21-2014 |
| 20140358531 | Speech Encoding Utilizing Independent Manipulation of Signal and Noise Spectrum - Some embodiments describe methods, programs, and systems for speech encoding. Among other things, a received input signal representing a property of speech is quantized to generate a quantized output signal. Prior to the quantization, a version of the input signal is supplied to a first noise shaping filter having a first set of filter coefficients effective to generate a first filtered signal. Following the quantization, the quantized output signal is supplied to a second noise shaping filter having a second set of filter coefficients, thus generating a second filtered signal. A noise shaping operation is performed to control a frequency spectrum of a noise effect in the quantized output signal caused by the quantization, wherein the noise shaping operation is based on both the first and second filtered signals. Finally, the quantised output signal is transmitted in an encoded signal. | 12-04-2014 |
| 20140365212 | Receiver Intelligibility Enhancement System - Embodiments of the invention provide a system and a method for enhancing audio signals. A first audio signal buffer and a second audio signal buffer are acquired. Thereafter, the magnitude spectrum calculated from the Fast Fourier Transform (FFT) of the second audio signal is processed based on the Linear Predictive Coding (LPC) spectrum of the first audio signal to generate an enhanced second audio signal. | 12-11-2014 |
| 20140379332 | IDENTIFICATION OF A LOCAL SPEAKER - Method for speaker identification includes detecting a target speaker's utterance locally; extracting features from the detected utterance locally, analyzing the extracted features in the local device to obtain information on the speaker identification and/or encoding the extracted features locally, transmitting the encoded extracted features to a remote server, decoding and analyzing the received extracted features by the server to obtain information on the speaker identification, and transmitting the information on the speaker identification from the server to the location where the speaker's utterance was detected. The method further includes detecting speech activity locally. Extracting features, encoding the extracted features, and/or transmitting the encoded extracted features to the server, are only performed if speech activity above some predetermined threshold is detected. | 12-25-2014 |
| 20150081285 | SPEECH SIGNAL PROCESSING APPARATUS AND METHOD FOR ENHANCING SPEECH INTELLIGIBILITY - A speech signal processing apparatus and a speech signal processing method for enhancing speech intelligibility are provided. The speech signal processing apparatus includes an input signal gain determiner to determine a gain of an input signal based on a harmonic characteristic of a voiced speech, a voiced speech output unit to output a voiced speech in which a harmonic component is preserved by applying the gain to the input signal, a linear predictive coefficient determiner to determine a linear predictive coefficient based on the voiced speech, and an unvoiced speech preserver to preserve an unvoiced speech of the input signal based on the linear predictive coefficient. | 03-19-2015 |
| 20150081286 | LPC RESIDUAL SIGNAL ENCODING/DECODING APPARATUS OF MODIFIED DISCRETE COSINE TRANSFORM (MDCT)-BASED UNIFIED VOICE/AUDIO ENCODING DEVICE - Disclosed is an LPC residual signal encoding/decoding apparatus of an MDCT based unified voice and audio encoding device. The LPC residual signal encoding apparatus analyzes a property of an input signal, selects an encoding method of an LPC filtered signal, and encode the LPC residual signal based on one of a real filterbank, a complex filterbank, and an algebraic code excited linear prediction (ACELP). | 03-19-2015 |
| 20150127328 | Vector Joint Encoding/Decoding Method and Vector Joint Encoder/Decoder - A vector joint encoding/decoding method and a vector joint encoder/decoder are provided, more than two vectors are jointly encoded, and an encoding index of at least one vector is split and then combined between different vectors, so that encoding idle spaces of different vectors can be recombined, thereby facilitating saving of encoding bits, and because an encoding index of a vector is split and then shorter split indexes are recombined, thereby facilitating reduction of requirements for the bit width of operating parts in encoding/decoding calculation. | 05-07-2015 |
| 20150134328 | UNIFIED SPEECH/AUDIO CODEC (USAC) PROCESSING WINDOWS SEQUENCE BASED MODE SWITCHING - A Unified Speech and Audio Codec (USAC) that may process a window sequence based on mode switching is provided. The USAC may perform encoding or decoding by overlapping between frames based on a folding point when mode switching occurs. The USAC may process different window sequences for each situation to perform encoding or decoding, and thereby may improve a coding efficiency. | 05-14-2015 |
| 20150302861 | Methods, Encoder And Decoder For Linear Predictive Encoding And Decoding Of Sound Signals Upon Transition Between Frames Having Different Sampling Rates - Methods, an encoder and a decoder are configured for transition between frames with different internal sampling rates. Linear predictive (LP) filter parameters are converted from a sampling rate S | 10-22-2015 |
| 20150332694 | APPARATUS AND METHOD FOR SYNTHESIZING AN AUDIO SIGNAL, DECODER, ENCODER, SYSTEM AND COMPUTER PROGRAM - A method and an apparatus for synthesizing an audio signal are described. A spectral tilt is applied to the code of a codebook used for synthesizing a current frame of the audio signal. The spectral tilt is based on the spectral tilt of the current frame of the audio signal. Further, an audio decoder operating in accordance with the inventive approach is described. | 11-19-2015 |
| 20150332695 | LOW-FREQUENCY EMPHASIS FOR LPC-BASED CODING IN FREQUENCY DOMAIN - The invention provides an audio encoder including a combination of a linear predictive coding filter having a plurality of linear predictive coding coefficients and a time-frequency converter, wherein the combination is configured to filter and to convert a frame of the audio signal into a frequency domain in order to output a spectrum based on the frame and on the linear predictive coding coefficients; a low frequency emphasizer configured to calculate a processed spectrum based on the spectrum, wherein spectral lines of the processed spectrum representing a lower frequency than a reference spectral line are emphasized; and a control device configured to control the calculation of the processed spectrum by the low frequency emphasizer depending on the linear predictive coding coefficients of the linear predictive coding filter. | 11-19-2015 |
| 20150348563 | ENCODER AND DECODER TO ENCODE SIGNAL INTO A SCALABLE CODEC AND TO DECODE SCALABLE CODEC, AND ENCODING AND DECODING METHODS OF ENCODING SIGNAL INTO SCALABLE CODEC AND DECODING THE SCALABLE CODEC - An encoder and decoder to encode one or more input signals into a scalable codec and to decode the scalable codec, and encoding and decoding methods using a bitstream with a layered structure in the scalable codec change a top coding bit rate to encode the input signals according to a network status, and the bitstream is decoded by analyzing the top coding bit rate included in the bitstream. | 12-03-2015 |
| 20150348572 | DETECTING A USER'S VOICE ACTIVITY USING DYNAMIC PROBABILISTIC MODELS OF SPEECH FEATURES - Method of detecting voice activity starts with by generating probabilistic models that respectively model features of speech dynamically over time. Probabilistic models may model each feature dependent on a past feature and a current state. Features of speech may include a nonstationary signal presence feature, a periodicity feature, and a sparsity feature. Noise suppressor may then perform noise suppression on an acoustic signal to generate a nonstationary signal presence signal and a noise suppressed acoustic signal. An LPC module may then perform residual analysis on the noise suppressed data signal to generate a periodicity signal and a sparsity signal. Inference generator receives the probabilistic models and receives, in real-time, nonstationary signal presence signal, periodicity signal, and sparsity signal. Inference generator may then generate in real time an estimate of voice activity based on the probabilistic models, nonstationary signal presence signal, periodicity signal, and sparsity signal. Other embodiments are also described. | 12-03-2015 |
| 20150380007 | TEMPORAL GAIN ADJUSTMENT BASED ON HIGH-BAND SIGNAL CHARACTERISTIC - The present disclosure provides techniques for adjusting a temporal gain parameter and for adjusting linear prediction coefficients. A value of the temporal gain parameter may be based on a comparison of a synthesized high-band portion of an audio signal to a high-band portion of the audio signal. If a signal characteristic of an upper frequency range of the high-band portion satisfies a first threshold, the temporal gain parameter may be adjusted. A linear prediction (LP) gain may be determined based on an LP gain operation that uses a first value for an LP order. The LP gain may be associated with an energy level of an LP synthesis filter. The LP order may be reduced if the LP gain satisfies a second threshold. | 12-31-2015 |
| 20160042745 | Device And Method For Quantizing The Gains Of The Adaptive And Fixed Contributions Of The Excitation In A Celp Codec - A device is for retrieving a quantized gain of a fixed contribution of an excitation in a sub-frame of a frame. The device includes a receiver of a gain codebook index; an estimator of the gain of the fixed contribution of the excitation in the sub-frame, wherein the estimator is supplied with a parameter representative of a classification of the frame; a gain codebook for supplying a correction factor in response to the gain codebook index; and a multiplier of the estimated gain by the correction factor to provide a quantized gain of the fixed contribution of the excitation in said sub-frame. | 02-11-2016 |
| 20160055852 | FRAME LOSS CORRECTION BY WEIGHTED NOISE INJECTION - A method for processing a digital signal, implemented during decoding of the signal, in order to replace a succession of samples lost during decoding, the method comprising steps of: generating a structure of a signal for replacing the lost succession, this structure comprising spectral components determined from valid samples received during decoding before the succession of lost samples; generating a residue between a digital signal available to the decoder, comprising received valid samples, and a signal generated from the spectral components; and extracting blocks from the residue, method in which window weighted blocks are injected into the structure using an overlap-add approach, the injected blocks partially overlapping in time. | 02-25-2016 |
| 20160104487 | APPARATUS AND METHOD REALIZING A FADING OF AN MDCT SPECTRUM TO WHITE NOISE PRIOR TO FDNS APPLICATION - An apparatus for decoding an encoded audio signal to obtain a reconstructed audio signal includes a receiving interface for receiving one or more frames comprising information on a plurality of audio signal samples of an audio signal spectrum of the encoded audio signal, and a processor for generating the reconstructed audio signal. The processor is configured to generate the reconstructed audio signal by fading a modified spectrum to a target spectrum, if a current frame is not received by the receiving interface or if the current frame is received by the receiving interface but is corrupted, wherein the modified spectrum includes a plurality of modified signal samples, wherein, for each of the modified signal samples of the modified spectrum, an absolute value of the modified signal sample is equal to an absolute value of one of the audio signal samples of the audio signal spectrum. | 04-14-2016 |
| 20160104489 | APPARATUS AND METHOD REALIZING IMPROVED CONCEPTS FOR TCX LTP - An apparatus for decoding an encoded audio signal to obtain a reconstructed audio signal is provided. The apparatus includes a receiving interface, a delay buffer and a sample processor for processing the selected audio signal samples to obtain reconstructed audio signal samples of the reconstructed audio signal. The sample selector is configured to select, if a current frame is received by the receiving interface and if the current frame being received by the receiving interface is not corrupted, the plurality of selected audio signal samples from the audio signal samples being stored in the delay buffer depending on a pitch lag information being included by the current frame. | 04-14-2016 |
| 20160111101 | APPARATUS AND METHOD FOR CONCEALING FRAME ERASURE AND VOICE DECODING APPARATUS AND METHOD USING THE SAME - An apparatus and method for concealing frame erasure and a voice decoding apparatus and method using the same. The frame erasure concealment apparatus includes: a parameter extraction unit determining whether there is an erased frame in a voice packet, and extracting an excitement signal parameter and a line spectrum pair parameter of a previous good frame; and an erasure frame concealment unit, if there is an erased frame, restoring the excitement signal and line spectrum pair parameter of the erased frame by using a regression analysis from the excitement signal and line spectrum pair parameter of the previous good frame. According to the method and apparatus, by predicting and restoring the parameter of the erased frame through the regression analysis, the quality of the restored voice signal can be enhanced and the algorithm can be simplified. | 04-21-2016 |
| 20160140976 | SPEECH CODING APPARATUS AND METHOD THEREFOR - In a CELP-type speech coding apparatus, switching between an orthogonal search of a fixed codebook and a non-orthogonal search is performed in a practical and effective manner. The CELP-type speech coding apparatus includes a parameter quantizer that selects an adaptive codebook vector and a fixed codebook vector so as to minimize an error between a synthesized speech signal and an input speech signal. The parameter quantizer includes a fixed codebook searcher that switches between the orthogonal fixed codebook search and the non-orthogonal fixed codebook search based on a correlation value between a target vector for the fixed codebook search and the adaptive codebook vector obtained as a result of a synthesis filtering process. | 05-19-2016 |
| 20160155457 | METHOD AND ARRANGEMENT FOR CONTROLLING SMOOTHING OF STATIONARY BACKGROUND NOISE | 06-02-2016 |
| 20160163325 | METHOD FOR SPEECH CODING, METHOD FOR SPEECH DECODING AND THEIR APPARATUSES - A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal. In speech coding method according to a code-excited linear prediction (CELP) speech coding, a noise level of a speech in a concerning coding period is evaluated by using a code or coding result of at least one of spectrum information, power information, and pitch information, and various excitation codebooks are used based on an evaluation result | 06-09-2016 |
| 20160171986 | METHOD OF DETECTING A PREDETERMINED FREQUENCY BAND IN AN AUDIO DATA SIGNAL, DETECTION DEVICE AND COMPUTER PROGRAM CORRESPONDING THERETO | 06-16-2016 |
| 20160379656 | CONCEPT FOR ENCODING OF INFORMATION - An information encoder for encoding an information signal includes: a converter for converting the linear prediction coefficients of the predictive polynomial A(z) to frequency values f | 12-29-2016 |
