Abstract: A method and apparatus for implementing fixed codebooks as a common module are provided. In the method of implementing fixed codebooks of a plurality of speech codecs as a common module, it is possible to include only a part excluding fixed codebooks in a communication terminal or communication system, support various speech codecs without using a chip with high price and high performance, and reduce a memory space that is occupied by the speech codecs by generating a track of a fixed codebook corresponding to a speech codec based on information on the speech codec among the plurality of speech codecs and selecting a codebook vector corresponding to a target signal among codebook vectors constructed with combinations of pulses represented by the generated track. In addition, it is possible to reduce processing complexity as compared with a case of embodying the common fixed codebook module in software by embodying the common fixed codebook module in hardware.

Abstract: A method of mitigating errors in a distributed speech recognition process. The method comprises the steps of identifying a group comprising one or more vectors which have undergone a transmission error, and replacing one or more speech recognition parameters in the identified group of vectors. In one embodiment all the speech recognition parameters of each vector of the group are replaced by replacing the whole vectors, and each respective replaced whole vector is replaced by a copy of whichever of the preceding or following vector without error is closest in receipt order to the vector being replaced.

Abstract: A system enhances speech by detecting a speaker's utterance through a first microphone positioned a first distance from a source of interference. A second microphone may detect the speaker's utterance at a different position. A monitoring device may estimate the power level of a first microphone signal. A synthesizer may synthesize part of the first microphone signal by processing the second microphone signal. The synthesis may occur when power level is below a predetermined level.

Abstract: The present invention provides a method, a computer-software-product and an apparatus for enabling a determination of speech related audio data within a record of digital audio data. The method comprises steps for extracting audio features from the record of digital audio data, for classifying one or more subsections of the record of digital audio data, and for marking at least a part of the record of digital audio data classified as speech. The classification of the digital audio data record is performed on the basis of the extracted audio features and with respect to at least one predetermined audio class.

Abstract: A CELP speech decoder includes an adaptive codebook that generates an adaptive code vector and a random codebook that generates a random code vector. The random codebook includes an input vector provider that provides an input vector including at least one pulse, each pulse having a position and a polarity, a fixed waveform storage that stores at least one fixed waveform, and a selector that selects at least one of a first process and a second process based on a value of an adaptive codebook gain. The random codebook further includes a convolution section that generates the random code vector by convoluting the at least one fixed waveform with the input vector when the first process is selected. A synthesis filter outputs synthesized speech by performing linear prediction coefficient synthesis on a signal based on the adaptive code vector and the random code vector.

Abstract: An optional short SACCH is used when the network detects high error rate in the normal SACCH channel. The communication returns to normal mode when the network detects that the link quality has improved. In the short SACCH message only the most relevant information fields are sent. The extra bits are used for channel coding so that in the channel conditions where the most robust AMR codings still work, the BLER of short SACCH is still tolerable.

Abstract: 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.

Abstract: A method and apparatus include a voice activity detection module configured to detect silent frames, and a codec mode selection module configured to determine a codec mode. The voice activity detection module includes a receiver configured to receive a frame, a first determiner configured to determine a first set of parameters from the frame, and a providing unit configured to provide the first set of parameters to the codec mode selection module. The codec mode selection module includes a second determiner configured to determine a second set of parameters in dependence on the first set of parameters, and a selector configured to select a codec mode in dependence on the second set of parameters.

Abstract: An audio decoding system performs frame loss concealment (FLC) when portions of a bit stream representing an audio signal are lost within the context of a digital communication system. The audio decoding system employs two different FLC methods: one designed to perform well for music, and the other designed to perform well for speech. When a frame is deemed lost, the audio decoding system analyzes a previously-decoded audio signal corresponding to previously-decoded frames of an audio bit-stream. Based on the results of the analysis, the lost frame is classified as either speech or music. Using this classification, other signal analysis, and knowledge of the employed FLC methods, the audio decoding system selects the appropriate FLC method which then performs FLC on the lost frame.

Abstract: A normalization factor for a current frame of a signal may be determined. The normalization factor may depend on an amplitude of the current frame of the signal. The normalization factor may also depend on values of states after one or more operations were performed on a previous frame of a normalized signal. The current frame of the signal may be normalized based on the normalization factor that is determined. The states' normalization factor may be adjusted based on the normalization factor that is determined.

Abstract: Information about excitation signals of a first signal encoded by CELP is used to derive a limited set of candidate excitation signals for a second correlated second signal. Preferably, pulse locations of the excitation signals of the first encoded signal are used for determining the set of candidate excitation signals. More preferably, the pulse locations of the set of candidate excitation signals are positioned in the vicinity of the pulse locations of the excitation signals of the first encoded signal. The first and second signals may be multi-channel signals of a common speech or audio signal. However, the first and second signals may also be identical, whereby the coding of the second signal can be utilized for re-encoding at a lower bit rate.

Abstract: A hardware device for analyzing an audio signal comprises a calculator for calculating a neural activity pattern over time resulting at nerve fibers of an ear model based on the audio signal and a processor for processing the neural activity pattern to obtain a sequence of time information as an analysis representation describing a temporal position of consecutive trajectories, wherein a trajectory includes activity impulses on different nerve fibers based on the same event in the audio signal. A two-dimensional representation of the neural activity pattern is gradually distorted over time, and it is recognized when an approximately straight line is contained in the distorted two-dimensional representation of the neural activity pattern over time. Accordingly, a time information belonging to the trajectory is provided.

Abstract: An apparatus for processing adaptive codebook pitch lag from one CELP based standard to another CELP based standard. The apparatus has various modules that perform at least the functionality described herein. The apparatus includes a time-base subframe checker inspection module, which is adapted to associate one or more incoming subframes with an outgoing subframes of a destination codec. The apparatus also has a decision module coupled to the time-base subframe inspection module. The decision module is adapted to determine a desired pitch lag parameter from a plurality of pitch lag parameters among respective two or more incoming subframes. The apparatus has a pitch lag selection module coupled to the decision module. The pitch lag selection module is adapted to select the desired pitch lag parameter.

Abstract: A sampling-rate-independent method of automated speech recognition (ASR). Speech energies of a plurality of codebooks generated from training data created at an ASR sampling rate are compared to speech energies in a current frame of acoustic data generated from received audio created at an audio sampling rate below the ASR sampling rate. A codebook is selected from the plurality of codebooks, and has speech energies that correspond to speech energies in the current frame over a spectral range corresponding to the audio sampling rate. Speech energies above the spectral range are copied from the selected codebook and appended to the current frame.

Abstract: There is provided a voice activity detection method for indicating an active voice mode and an inactive voice mode. The method comprises receiving a first portion of an input signal; determining that the first portion of the input signal includes an active voice signal; indicating the active voice mode in response to the determining that the first portion of the input signal includes the active voice signal; receiving a second portion of the input signal immediately following the first portion of the input signal; determining that the second portion of the input signal includes an inactive voice signal; extending the indicating the active voice mode for a period of time after determining that the second portion of the input signal includes the inactive voice signal, wherein the period of time varies based on one or more conditions; and indicating the inactive voice mode after expiration of the period of time.

Abstract: Encoding a sequence of digital speech samples into a bit stream includes dividing the digital speech samples into one or more frames and computing a set of model parameters for the frames. The set of model parameters includes at least a first parameter conveying pitch information. The voicing state of a frame is determined and the first parameter conveying pitch information is modified to designate the determined voicing state of the frame, if the determined voicing state of the frame is equal to one of a set of reserved voicing states. The model parameters are quantized to generate quantizer bits which are used to produce the bit stream.

Abstract: An implementation of the present invention comprises a voice encoder and decoder method and system that uses voice excitation, eliminating the voice/unvoiced pitch tracking, and the first formant up to 2400 Hertz for synchronous and up to 1600 Hertz for asynchronous, does not use pulse code modulation encoding, but uses the zero crossings only of the first formant, frequency dividing by two and sampling at the formant frequency. The resulting combination uses half or less of the bit rate for excitation and the remainder for short-term spectrum analysis. The spectrum could be updated each 20 milliseconds using 49 bits for the spectrum frame and 49 bits for excitation and one frame bit for synchronous Asynchronous operation could be update at 21.25 milliseconds using 49 bits for the spectrum information and 34 bits for excitation with one bit for frame synchronization.

Abstract: 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.

Abstract: The present invention discloses a method for obtaining an attenuation factor. The method is adapted to process the synthesized signal in packet loss concealment, and includes: obtaining a change trend of a pitch of a signal; obtaining an attenuation factor, according to the change trend of the pitch of the signal. The present invention also discloses an apparatus for obtaining an attenuation factor. A self-adaptive attenuation factor is adjusted dynamically by using the latest change trend of a history signal by using the present invention. The smooth transition from the history data to the data last received is realized so that the attenuation speed is kept consistent between the compensated signal and the original signal as much as possible for adapting to the characteristic of various human voices.

Abstract: A fixed codebook search method includes: initializing a counter; searching for pulses and calculating the value of a cost function Qk; initializing the counter if the Qk value increases; increasing the value of the counter if the Qk value does not increase; judging whether the value of the counter is greater than the threshold value; continuing the search process if the value of the counter is not greater than the threshold value; and ending the whole search process if the value of the counter is greater than the threshold value. The present invention reduces the search count and improves the search efficiency.

Abstract: A fixed codebook search method includes initializing a counter, searching for pulses and calculating the value of a cost function Qk, initializing the counter if the Qk value increases, increasing the value of the counter if the Qk value does not increase, judging whether the value of the counter is greater than the threshold value, continuing the search process if the value of the counter is not greater than the threshold value, and ending the whole search process if the value of the counter is greater than the threshold value.

Abstract: The invention concerns a method and apparatus for performing packet loss or Frame Erasure Concealment (FEC) for a speech coder that does not have a built-in or standard FEC process. A receiver with a decoder receives encoded frames of compressed speech information transmitted from an encoder. A lost frame detector at the receiver determines if an encoded frame has been lost or corrupted in transmission, or erased. If the encoded frame is not erased, the encoded frame is decoded by a decoder and a temporary memory is updated with the decoder's output. A predetermined delay period is applied and the audio frame is then output. If the lost frame detector determines that the encoded frame is erased, a FEC module applies a frame concealment process to the signal. The FEC processing produces natural sounding synthetic speech for the erased frames.

Abstract: For a bandwidth extension of an audio signal, in a signal spreader the audio signal is temporally spread by a spread factor greater than 1. The temporally spread audio signal is then supplied to a demicator to decimate the temporally spread version by a decimation factor matched to the spread factor. The band generated by this decimation operation is extracted and distorted, and finally combined with the audio signal to obtain a bandwidth extended audio signal. A phase vocoder in the filterbank implementation or transformation implementation may be used for signal spreading.

Abstract: There is provided a method of transcoding an Enhance Full Rate (EFR) 12.2 Kbps encoded frame into an Adaptive Multi-Rate (AMR) 12.2 Kbps encoded frame, where the method comprises receiving the EFR 12.2 Kbps encoded frame from a first codec; determining if the EFR 12.2 Kbps encoded frame is a Silence Insertion Descriptor (SID) frame; if the EFR 12.2 Kbps encoded frame is determined to be the SID frame, the method further comprises transcoding the EFR SID frame. There is also provided a method of transcoding an EFR 12.2 Kbps encoded frame into an AMR 12.2 Kbps encoded frame, where the method comprises receiving the AMR 12.2 Kbps encoded frame from a first codec; determining if the AMR 12.2 Kbps encoded frame is an SID frame; if the AMR 12.2 Kbps encoded frame is determined to be the SID frame, the method further comprises transcoding the AMR SID frame.

Abstract: An electronic circuit (1100) including a processor circuit (1110) and a storage circuit establishing a speech coder (1170) for execution by said processor (1110), the speech coder (1170) for approximating speech by pulses having pulse positions selectable from a codebook (550), the speech coder (1170) operable to obtain (1310) a set of estimated pulse positions having a first number of pulse tracks of the estimated pulse positions, use (1320) a cost function (epsilon tilde {tilde over (?)}) relating to approximation to speech to find a first subset including a second number of one or more pulse tracks fewer in number than the first number wherein the first subset of pulse tracks contributed a lower contribution to the cost function relative to a second subset of pulse tracks, and control (1330) a subsequent pulse position search beginning with the lower-contributing subset of pulse tracks to yield pulse positions to provide a value of the cost function representing a better approximation to speech.

Abstract: A method and arrangement for dynamically adapting thresholds used for selecting a codec mode to be used is presented. Thresholds are adapted in response to the current received signal quality. An estimate of actual prevailing received signal quality is obtained on which the adaptation is based. The present invention can be applied either on the mobile terminal side or on the network side, working on the uplink and/or the downlink. The thresholds can be modified on the receiving side, or, when operating in the network and working on the downlink, the threshold adaptation can be initiated in the terminal.

Abstract: Methods and apparatus are provided for sending data communications over wireless digital voice communications networks which transmit voice communications in voice frames, each of which contains a digitized segment of a voice communication in a voice frame format. The method including the steps of: encoding the data communication into a plurality of data frames, each of the data frames having the same format as the voice frame format; transmitting the data frames over the wireless digital voice communications network; and decoding the data frames to reconstruct the data communication. The apparatus includes: a processor for encoding the data communication into a plurality of data frames, each of the data frames having the same format as the voice frame format; and a transmitter for transmitting the data frames over the wireless digital voice communications network.

Abstract: There is disclosed an audio encoding device capable of realizing effective encoding while using audio encoding of the CELP method in an extended layer when hierarchically encoding an audio signal. In this device, a first encoding section (115) subjects an input signal (S11) to audio encoding processing of the CELP method and outputs the obtained first encoded information (S12) to a parameter decoding section (120). The parameter decoding section (120) acquires a first quantization LSP code (L1), a first adaptive excitation lag code (A1), and the like from the first encoded information (S12), obtains a first parameter group (S13) from these codes, and outputs it to a second encoding section (130). The second encoding section (130) subjects the input signal (S11) to a second encoding processing by using the first parameter group (S13) and obtains second encoded information (S14).

Abstract: The present invention relates to encoding technology. The encoding method includes selecting a second encoding mode for encoding an input frame signal according to an analysis on signal characteristic of the input frame signal; obtaining coding demand values for a preset first encoding mode and the second encoding mode which are used to encode the input frame signal; determining, from the above encoding modes based on the coding demand values, an encoding mode for encoding the input frame signal; and multiplexing information of the determined encoding mode and encoded data which are encoded according to the determined encoding mode. Hence, the compatibility and the prioritization in terms of the encoding modes can be achieved.

Abstract: A fixed codebook search method includes initializing a counter, searching for pulses and calculating the value of a cost function Qk, initializing the counter if the Qk value increases, increasing the value of the counter if the Qk value does not increase, judging whether the value of the counter is greater than the threshold value, continuing the search process if the value of the counter is not greater than the threshold value, and ending the whole search process if the value of the counter is greater than the threshold value.

Abstract: The conventional error conceal processing generates a greatly fluctuating irregular sound which is unpleasant to ears and causes a remarkable echo effect and click noise. A notification signal detection unit (301) judges processing for an input frame. In case of an error frame, a sound detection unit (303) makes judgment whether a preceding non-error data frame is a sound signal. If it is a sound frame, a sound copying unit (304) generates a replacing frame. If it is a non-sound frame, a transient signal detection unit (305) judges whether it is an attack signal by the transient signal detection and selects an appropriate area from the preceding non-error frame.

Abstract: An audio codec system and an encoding method using the same are provided. According to the method, encoding and decoding processes are repeatedly performed so as to determine optimized coding parameters when analog audio signals being inputted are encoded. The processes of encoding and decoding inputted analog audio signals using initial coding parameters, and computing new parameters using a differential computed during the encoding process are repeatedly performed.

Abstract: A system and method of remotely enabling sound enhancement techniques is disclosed. In an embodiment, a watermark is embedded in an encoded multi-channel audio stream to remotely enable an enhancement decoder portion of a multi-channel audio decoder.

Abstract: A method and apparatus for prediction in a speech-coding system is provided herein. The method of a 1st order long-term predictor (LTP) filter, using a sub-sample resolution delay, is extended to a multi-tap LTP filter, or, viewed from another vantage point, the conventional integer-sample resolution multi-tap LTP filter is extended to use sub-sample resolution delay. This novel formulation of a multi-tap LTP filter offers a number of advantages over the prior-art LTP filter configurations. Particularly, defining the lag with sub-sample resolution makes it possible to explicitly model the delay values that have a fractional component, within the limits of resolution of the over-sampling factor used by the interpolation filter. The coefficients of such a multi-tap LTP filter are thus largely freed from modeling the effect of delays that have a fractional component.

Abstract: An electronic circuit includes a storage circuit and a microprocessor operable together with the storage circuit as a speech coder. The speech coder has a backward pitch enhancement in frames or subframes having a length and at least one main pulse and at least one backward pitch enhancement pulse preceding the main pulse by a portion of the length called a pitch lag, and is operable to limit in number any such backward pitch enhancement pulse or pulses to a predetermined maximum number more than none upon an occurrence when the length divided by the pitch lag is at least one more than that maximum number. Other forms of the invention involve systems, circuits, devices, processes and methods of operation.

Abstract: A wideband speech coding method comprising identifying whether an input speech signal is a narrowband signal or a wideband signal, and coding the input speech signal by controlling a predetermined parameter of a wideband speech coding process based on the identification result.

Abstract: An audio encoding apparatus and the like are disclosed which can improve the sound quality of encoded audio signals even in a case of scalable CELP encoding the audio signals in sections that vary with time. In this apparatus, an enhancement layer extended adaptive codebook generating part (102) generates an extended adaptive codebook (d_enh_ext[i]) from both one frame of core layer drive sound source signals (exc_core[n]) received from a core layer CELP encoding part (101) and past enhancement layer drive sound source signals (exc_enh[n]) received from an adder (106), and further inputs the generated extended adaptive codebook (d_enh_ext[i]) to an enhancement layer extended adaptive codebook (103) for each of sub-frames. That is, the enhancement layer extended adaptive codebook generating part (102) updates the extended adaptive codebook (d_enh_ext[i]) for each of the sub-frames.

Abstract: 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.

Abstract: The present invention relates to a method of coding a signal, in particular an audio or speech signal, wherein a codebook comprising k code vectors is provided for vector quantization of a signal vector representing a set of signal values of said signal(s), and wherein an optimal code vector of said codebook is determined by performing a codebook search. Parallelism is employed to accelerate the coding procedure. In particular, the codebook search is highly parallelised.

Abstract: A method of converting a voice signal spoken by a source speaker into a converted voice signal having acoustic characteristics that resemble those of a target speaker. The method includes the following steps: determining (1) at least one function for the transformation of the acoustic characteristics of the source speaker into acoustic characteristics similar to those of the target speaker; and transforming the acoustic characteristics of the voice signal to be converted using the at least one transformation function. The method is characterized in that: (i) the aforementioned transformation function-determining step (1) consists in determining (1) a function for the joint transformation of characteristics relating to the spectral envelope and characteristics relating to the fundamental frequency of the source speaker; and (ii) the transformation includes the application of the joint transformation function.

Abstract: A sound encoder multiplexes a plurality of codes into a sound code in an order determined by a multiplexing order determination unit (12), and a sound decoder demultiplexes the sound code into a plurality of codes one by one in an order determined by a demultiplexing order determination unit (14).

Abstract: A sound encoder multiplexes a plurality of codes into a sound code in an order determined by a multiplexing order determination unit (12), and a sound decoder demultiplexes the sound code into a plurality of codes one by one in an order determined by a demultiplexing order determination unit (14).

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

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

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

Abstract: An audio decoding device (1) generates a first decoded audio from a first code string by a first decoding method, AN audio encoding device (2) judges whether the first decoded audio is an audio signal or a non-audio signal by using the information contained in the first code string and generates a second code string by encoding the first decoded audio by the second encoding method according to the judgment. Thus, there are provided a device and a method for converting a code obtained y encoding audio by a certain method into a code decodable by the other method with a low calculation amount.

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

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

Abstract: A method for transcoding a CELP based compressed voice bitstream from source codec to destination codec. The method includes processing a source codec input CELP bitstream to unpack at least one or more CELP parameters from the input CELP bitstream and interpolating one or more of the plurality of unpacked CELP parameters from a source codec format to a destination codec format if a difference of one or more of a plurality of destination codec parameters including a frame size, a subframe size, and/or sampling rate of the destination codec format and one or more of a plurality of source codec parameters including a frame size, a subframe size, or sampling rate of the source codec format exist. The method includes encoding the one or more CELP parameters for the destination codec and processing a destination CELP bitstream by at least packing the one or more CELP parameters for the destination codec.

Abstract: A conversion entity and method for converting higher-rate speech parameters into lower-rate parameters including dimmed excitation parameters. The conversion entity comprises a first decoder configured to produce a target excitation from the higher-rate parameters, based on a first fixed contribution and a first adaptive contribution. The conversion entity also comprises a second decoder configured to produce a second adaptive contribution, and configured to selectably operate in a first or a second mode. In the first mode, the second adaptive component is generated based on the first fixed contribution for a previous frame, while in the second mode, the second adaptive component is generated based on a second fixed contribution for the previous frame. The second decoder operates in the second mode in response to a rate reduction request.