Abstract: Techniques are described herein that suppress noise in a Mel-filtered spectral domain. For example, a window may be applied to a representation of a speech signal in a time domain. The windowed representation in the time domain may be converted to a subsequent representation of the speech signal in the Mel-filtered spectral domain. A noise suppression operation may be performed with respect to the subsequent representation to provide noise-suppressed Mel coefficients.

Abstract: A method comprising receiving at a user equipment encrypted content. The content is stored in said user equipment in an encrypted form. At least one key for decryption of said stored encrypted content is stored in the user equipment.

Abstract: A lossless coding technique for near-logarithmic companded PCM that achieves high compression performance is provided. In coding, the coding method that produces the smaller code amount is selected between the prediction coding method, which performs linear prediction of samples in a frame and codes the amplitude of the prediction error, and the normalization coding method, which normalizes the amplitude of the samples in the frame and codes the normalized amplitude, and a selection code that indicates the selection result is output. The samples in the frame are coded according to the selected coding method to produce a compression code. In decoding, the compression code is decoded according to a decoding process corresponding to the coding method specified by the selection code.

Abstract: Method and arrangement in a network node for adapting a property of source coding to the quality of a communication link in packet switched conversational services in a communication system. The method comprises obtaining (404) information related to the quality of a communication link. The method further comprises selecting (406) a source coding mode with an associated source coding delay, based on the obtained information and the associated source coding delay. The selected source coding mode is selected from a set of at least two source coding modes associated with different source coding delays, and is to be used when source coding voice data to be transmitted over the communication link.

Abstract: Systems, methods, apparatus, and machine-readable media for voice activity detection in a single-channel or multichannel audio signal are disclosed.

Abstract: The present invention relates to an encoding device and an encoding method, a decoding device and a decoding method, and a program that reduce deterioration of sound quality due to encoding of audio signals. An envelope emphasis part (51) emphasizes an envelope (ENV). A noise shaping part (52) divides an emphasized envelope (D) formed by emphasis of the envelope (ENV) by a value larger than 1, and subtracts noise shaping (G) specified by information (NS) from a result of the division. A quantization part (14) sets a result of the subtraction as a quantization bit count (WL), and quantizes a normalized spectrum (S1) formed by normalization of a spectrum (S0) based on the quantization bit count (WL). A multiplexing part (53) multiplexes the information (NS), a quantized spectrum (QS) formed by quantization of the normalized spectrum (S1), and the envelope (ENV). The present invention can be applied to an encoding device encoding audio signals, for example.

Abstract: Low complexity detection of a time-wise position of a representative segment in media data is described. A subset of offset values is located in a set of offset values in media data using a first type of one or more types of features, which are extractable from (e.g., derivable from components of) the media data. The subset of offset values comprise values that are selected from the set of offset values based on one or more selection criteria. A set of candidate seed time points is identified based on the subset of offset values using a second type of the one or more types of features.

Abstract: 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.

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: Provided are methods and systems for improving quality of speech communications. The method may be for improving quality of speech communications in a system having a speech encoder configured to encode a first audio signal using a first set of encoding parameters associated with a first noise suppressor. A method may involve receiving a second audio signal at a second noise suppressor which provides much higher quality noise suppression than the first noise suppressor. The second audio signal may be generated by a single microphone or a combination of multiple microphones. The second noise suppressor may suppress the noise in the second audio signal to generate a processed signal which may be sent to a speech encoder. A second set of encoding parameters may be provided by the second noise suppressor for use by the speech encoder when encoding the processed signal into corresponding data.

Abstract: Methods and apparatus for audio and speech processing including generating a plurality of frames, each of the frames comprising a plurality of transform coefficients, and allocating bits to the transform coefficients in each of the frames such that at least two of the transform coefficients in the same frame have different bit allocations and the total number of the bits allocated to the transform coefficients in at least two of the frames is equal.

Abstract: A method is disclosed for controlling a voice-activated device by interpreting a spoken command as a series of voiced and non-voiced intervals. A responsive action is then performed according to the number of voiced intervals in the command. The method is well-suited to applications having a small number of specific voice-activated response functions. Applications using the inventive method offer numerous advantages over traditional speech recognition systems including speaker universality, language independence, no training or calibration needed, implementation with simple microcontrollers, and extremely low cost. For time-critical applications such as pulsers and measurement devices, where fast reaction is crucial to catch a transient event, the method provides near-instantaneous command response, yet versatile voice control.

Abstract: Differential dynamic content delivery including providing a session document for a presentation, wherein the session document includes a session grammar and a session structured document; selecting from the session structured document a classified structural element in dependence upon user classifications of a user participant in the presentation; presenting the selected structural element to the user; streaming presentation speech to the user including individual speech from at least one user participating in the presentation; converting the presentation speech to text; detecting whether the presentation speech contains simultaneous individual speech from two or more users; and displaying the text if the presentation speech contains simultaneous individual speech from two or more users.

Abstract: The present invention provides a method and a device for identifying and obtaining AMBE encoding and decoding rate information in an SDP. The method for identifying the AMBE encoding and decoding rate information in the SDP includes: setting an SDP, which specifically includes: in an m attribute line of the SDP, using a payload type PT value to describe AMBE encoding and decoding, and in an fmtp attribute of an a attribute line of the SDP, identifying, in a format field, a PT value same as that of the m attribute line, and identifying AMBE encoding and decoding rate information in a format specific parameter field; and sending the set SDP to a peer end. The method and the device for identifying and obtaining the AMBE encoding and decoding rate information in the SDP provided by the present invention can implement automatic rate switching in a bearer plane.

Abstract: A system and method of generating at least two distinctive auditory alerts upon receiving a transmission or telephone call at a device is described. Data indicative of a first plurality of sounds corresponding to a user of a device configured to receive the transmission or telephone call is accessed, such as from a memory. The first plurality of sounds is played at the device so as to identify a received transmission or telephone call being directed to the user. A telephone number, subscriber name or identifier associated with a transmitting or calling party of the transmission or telephone call is accessed. Data indicative of a second plurality of sounds designating the transmitting or calling party based on the subscriber name, telephone number or identifier is retrieved, such as from a data structure, and the second plurality of sounds is played at the device so as to identify the transmitting or calling party.

Abstract: 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.

Abstract: A speech signal processing device is equipped with a power acquisition unit, a probability distribution acquisition unit, and a correspondence degree determination unit. The power acquisition unit accepts an inputted speech signal and, based on the accepted speech signal, acquires power representing the intensity of a speech sound represented by the speech signal. The probability distribution acquisition unit acquires a probability distribution using the intensity of the power acquired by the power acquisition unit as a random variable. The correspondence degree determination unit determines whether a correspondence degree representing a degree that power acquired by the power acquisition unit in a case that a predetermined reference speech signal is inputted into the power acquisition unit corresponds with predetermined reference power is higher than a predetermined reference correspondence degree, based on the probability distribution acquired by the probability distribution acquisition unit.

Abstract: Apparatus (119) for encoding at least one parameter associated with a signal source for transmission over k frames to a decoder comprises a processor (119) which is configured in operation to assign a predetermined bit pattern to n bits associated with the at least one parameter of a first frame of k frames and set the n bits associated with the at least one parameter of each of k?1 subsequent frames to values, such that the values of the n bits of the k?1 subsequent frames represent the at least one parameter. The predetermined bit pattern indicates a start of the at least one parameter.

Abstract: An audio encoder for encoding an audio signal includes an impulse extractor for extracting an impulse-like portion from the audio signal. This impulse-like portion is encoded and forwarded to an output interface. Furthermore, the audio encoder includes a signal encoder which encodes a residual signal derived from the original audio signal so that the impulse-like portion is reduced or eliminated in the residual audio signal. The output interface forwards both, the encoded signals, i.e., the encoded impulse signal and the encoded residual signal for transmission or storage. On the decoder-side, both signal portions are separately decoded and then combined to obtain a decoded audio signal.

Abstract: In one aspect, a method of processing a voice signal to extract information to facilitate training a speech synthesis model is provided. The method comprises acts of detecting a plurality of candidate features in the voice signal, performing at least one comparison between one or more combinations of the plurality of candidate features and the voice signal, and selecting a set of features from the plurality of candidate features based, at least in part, on the at least one comparison. In another aspect, the method is performed by executing a program encoded on a computer readable medium. In another aspect, a speech synthesis model is provided by, at least in part, performing the method.

Abstract: A system and method for providing voice communications with desired characteristics based upon the intended recipient of a voice communication. An apparatus includes a list of dial strings associated with parties having desired voice communication characteristics. A dial string entered by a user and associated with an intended recipient is compared to a list of preferred dial strings to determine the characteristics of an encoded voice signal to be sent to the recipient. The apparatus can include a vocoder having different bit rate modes and a bit rate mode is selected based upon the dial string entered by a user. Dial strings can be stored at the device or on a network. The apparatus can include a mode selector to select a desired vocoder mode to generate an encoded voice signal.

Abstract: A decoder arrangement comprising a receiver input for parameters of frame-based coded signals and a decoder arranged to provide frames of decoded audio signals based on the parameters. The receiver input and/or the decoder is arranged to establish a time difference between the occasion when parameters of a first frame is available at the receiver input and the occasion when a decoded audio signal of the first frame is available at an output of the decoder, which time difference corresponds to at least one frame. A postfilter is connected to the output of the decoder and to the receiver input. The postfilter is arranged to provide a filtering of the frames of decoded audio signals into an output signal in response to parameters of a respective subsequent frame.

Abstract: A system and method may be configured to analyze audio information derived from an audio signal. The system and method may track sound pitch across the audio signal. The tracking of pitch across the audio signal may take into account change in pitch by determining at individual time sample windows in the signal duration an estimated pitch and a representation of harmonic envelope at the estimated pitch. The estimated pitch and the representation of harmonic envelope may then be implemented to determine an estimated pitch for another time sample window in the signal duration with an enhanced accuracy and/or precision.

Abstract: An audio encoder adapted for encoding frames of a sampled audio signal to obtain encoded frames, wherein a frame includes a number of time domain audio samples. The audio encoder includes a predictive coding analysis stage for determining information on coefficients of a synthesis filter and a prediction domain frame based on a frame of audio samples. The audio encoder further includes a time-aliasing introducing transformer for transforming overlapping prediction domain frames to the frequency domain to obtain prediction domain frame spectra, wherein the time-aliasing introducing transformer is adapted for transforming the overlapping prediction domain frames in a critically-sampled way. Moreover, the audio encoder includes a redundancy reducing encoder for encoding the prediction domain frame spectra to obtain the encoded frames based on the coefficients and the encoded prediction domain frame spectra.

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

Abstract: A sound reproducing device has a loudspeaker arranged to produce sound from an audio signal provided by an audio signal source. A microphone is positioned to pick up ambient noise and generate a microphone signal which comprises the noise. An ambient noise cancellation (ANC) system receives the microphone signal from the microphone and generates anti-noise corresponding to the ambient noise in the microphone signal. An automatic polarity adaptation (AAP) system monitors the ANC system and, when a decision criterion is fulfilled, causes a switch in polarity for the generated anti-noise.

Abstract: A scalable decoder device (50) for signals representing audio comprises a primary decoder (21) connected to an input (40). The primary decoder (21) is arranged to provide a primary decoded signal (23) based on received parameters (4). A primary postfilter (31) is connected to the primary decoder (23) to provide a primary postfiltered signal (32). A secondary enhancement decoder (45) is connected to the input (40) and arranged to provide a secondary decoded enhancement signal (44). The device further comprises a combiner arrangement (55), arranged for combining the primary postfiltered signal (32) and a signal (53) based on the secondary decoded enhancement signal (44) into an output signal (6) to be provided at an output (6). The combining is made with an adaptable strength relation between contributions from the two signals. A method for decoding coded signals representing audio operates in analogy with the scalable decoder device (50).

Abstract: Methods and apparatus of managing a communication system, wherein a decision regarding a level of activity at a first end is made based at least in part on the level of activity at the second end. In one embodiment, the energy level of a first-end audio signal is measured. The first end is declared voice-active if the first-end energy level is greater than or equal to a first threshold value. The first end is declared voice-inactive if the first-end energy level is less than the first threshold value. To determine the value of the first threshold value, the energy level of a second-end audio signal is measured. If the second-end energy level is greater than or equal to a second threshold value, the second end is declared voice-active, in which case the first threshold is maintained at a relatively high level. If the second-end energy level is less than the second threshold value, the second end is declared voice-inactive, in which case the first threshold is maintained at a relatively lower level.

Abstract: A decoding device is capable of flexibly calculating high-band spectrum data with a high accuracy in accordance with an encoding band selected by an upper-node layer of the encoding side. In this device: a first layer decoder decodes first layer encoded information to generate a first layer decoded signal; a second layer decoder decodes second layer encoded information to generate a second layer decoded signal; a spectrum decoder performs a band extension process by using the second layer decoded signal and the first layer decoded signal up-sampled in an up-sampler so as to generate an all-band decoded signal; and a switch outputs the first layer decoded signal or the all-band decoded signal according to the control information generated in a controller.

Abstract: A method and an apparatus for inserting an included message into an e-mail message, wherein the e-mail message is transferred through a unified messaging solution have been provided. In one embodiment, the unified messaging solution detects transmission of a voice mail message as the e-mail attachment. The voice mail message is received by a system that facilitates the transfer of the e-mail message. The system associates the included message with the voice mail message. The included message is inserted into the e-mail message. The system sends the e-mail message along with the included message and the attached voice mail message to an intended user. In a preferred embodiment, the included message is an advertising message.

Abstract: 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.

Abstract: Provided are methods and systems of managing handoffs in a wireless communication system having different types of vocoders. Some embodiments include translating state memory of a first vocoder to a second vocoder using a state memory transcoder. The state memory may be delayed to align differences in algorithmic delays between the first vocoder and the second vocoder. In one embodiment, a speech signal may be decoded from the first vocoder, delayed, and encoded to the second vocoder. Furthermore, for a period of time during and/or adjacent to the handoff, the first vocoder may output with decreasing amplitude while the second vocoder outputs with increasing amplitude. Such techniques may be used alone or in combination.

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

Abstract: Information that includes first information identifying integer quotients obtained by divisions using prediction residuals or integers not smaller than 0 that increase monotonically with increases in the amplitude of the prediction residuals, as dividends, and a separation parameter decided for a time segment corresponding to the prediction residuals or a mapped integer value of the separation parameter, as a modulus, and second information identifying the remainders obtained when the dividends are divided by the modulus is generated as a code corresponding to the prediction residuals, and each piece of side information that includes the separation parameter is subjected to variable length coding.

Abstract: Differential dynamic content delivery including providing a session document for a presentation, wherein the session document includes a session grammar and a session structured document; selecting from the session structured document a classified structural element in dependence upon user classifications of a user participant in the presentation; presenting the selected structural element to the user; streaming presentation speech to the user including individual speech from at least one user participating in the presentation; converting the presentation speech to text; detecting whether the presentation speech contains simultaneous individual speech from two or more users; and displaying the text if the presentation speech contains simultaneous individual speech from two or more users.

Abstract: A method of transmitting speech data to a remote device in a distributed speech recognition system, includes the steps of: dividing an input speech signal into frames; calculating, for each frame, a voice activity value representative of the presence of speech activity in the frame; grouping the frames into multiframes, each multiframe including a predetermined number of frames; calculating, for each multiframe, a voice activity marker representative of the number of frames in the multiframe representing speech activity; and selectively transmitting, on the basis of the voice activity marker associated with each multiframe, the multiframes to the remote device.

Abstract: A system and method for modeling speech in such a way that both voiced and unvoiced contributions can co-exist at certain frequencies. In various embodiments, three spectral bands (or bands of up to three different types) are used. In one embodiment, the lowest band or group of bands is completely voiced, the middle band or group of bands contains both voiced and unvoiced contributions, and the highest band or group of bands is completely unvoiced. The embodiments of the present invention may be used for speech coding and other speech processing applications.

Abstract: A method for determining an upperband speech signal from a narrowband speech signal is disclosed. A list of narrowband line spectral frequencies (LSFs) is determined from the narrowband speech signal. A first pair of adjacent narrowband LSFs that have a lower difference between them than every other pair of adjacent narrowband LSFs in the list is determined. A first feature that is a mean of the first pair of adjacent narrowband LSFs is determined. Upperband LSFs are determined based on at least the first feature using codebook mapping.

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: This invention provides remote audio surveillance by recording audio data via three microphones and storage on a removable digital mass storage device, operating on battery power. The housing is of a weather resistant design to withstand outdoor conditions. Recording can be done in person or recording times can be defined so that the unit will only ‘listen’ during the desired times of the day, on a day to day basis. The user does not have to be in the vicinity but simply programs the record time(s) and leaves the device in the woods. The device also has play back capabilities for any recorded audio data and can interface with personal computers via the removable digital mass storage device. In addition to the audio collection and playback capabilities, PC software will be provided with the device which will analyze the data and provide direction of sound (based upon relative amplitude of the 3 microphones) and distance of sound (based on absolute and relative recorded amplitudes).

Abstract: It is possible to improve quality of a decoding signal in a band spread for estimating a high band from a low band of a decoding signal. A first layer encoder encodes a lower band portion below a predetermined frequency of an input signal so as to generate first layer encoded information. A first layer decoder decodes the first layer encoded information so as to generate a first layer demodulated signal. A second layer encoder divides a high band portion higher, than a predetermined frequency, of an input signal into a plurality of sub-bands and estimates each of the sub-bands from the input signal or the first layer decoded signal by using the estimation result of the sub-band adjacent to the lower band side so as to generate second encoded information including the estimation results of the sub-bands.

Abstract: In one aspect, a method of processing a voice signal to extract information to facilitate training a speech synthesis model is provided. The method comprises acts of detecting a plurality of candidate features in the voice signal, performing at least one comparison between one or more combinations of the plurality of candidate features and the voice signal, and selecting a set of features from the plurality of candidate features based, at least in part, on the at least one comparison. In another aspect, the method is performed by executing a program encoded on a computer readable medium. In another aspect, a speech synthesis model is provided by, at least in part, performing the method.

Abstract: Methods and devices are described for entropy coding data using an entropy coder to encode quantized transform domain coefficient data. Last significant coefficient information is signaled in the bitstream using two-dimensional coordinates for the last significant coefficient. The context for bins of one of the coordinates is based, in part, upon the value of the other of the coordinates. In one case, instead of signaling last significant coefficient information, the number of non-zero coefficients is binarized and entropy encoded.

Abstract: In an encoding process, a CPU transforms an audio signal from the real-time domain to the frequency domain, and transforms the signal into spectra consisting of MDCT coefficients. The CPU separates the audio signal into several frequency bands, and performs bit shifting in each band such that the MDCT coefficients can be expressed with pre-configured numbers of bits. The CPU re-quantizes the MDCT coefficients at a precision differing for each band, and transmits the values acquired thereby and shift bit numbers as encoded data. Meanwhile, in a decoding process, a CPU receives encoded data and inverse re-quantizes and inverse bit shifts the data, thereby restoring the MDCT coefficients. Furthermore, the CPU transforms the data from frequency domain to the real-time domain by using the inverse MDCT, and restores and outputs the audio signal.

Abstract: An audio encoder capable of implementing a plurality of encoding functions, wherein an adaptation controller adjusts the implementation of the encoding functions in response to feedback received by the adaptation controller during use. The adjustment may involve adapting encoding algorithms or selecting alternative encoding algorithms. The encoder may also include an operations scheduler to adjust the order in which the encoding functions are applied. The feedback may be received from internally of the encoder, for example from the currently implemented encoding functions, or from externally of the encoder. A corresponding decoder is also provided.

Abstract: A vector quantizer which improves the accuracy of vector quantization in switching over a vector quantization codebook on a first stage depending on the type of feature having the correlation with a quantization target vector. In the vector quantizer, a classifier generates classification information representing a type of narrowband LSP vector having the correlation with wideband LSP (Line Spectral Pairs) of the plural types. A first codebook selects one sub-codebook corresponding to the classification information as a codebook used for the quantization of the first stage from plural sub-codebooks corresponding to each of the types of narrowband LSP vectors. A multiplier multiplies the quantization residual vector of the first stage inputted from an adder by a scaling factor corresponding to the classification information of plural scaling factors stored in a scaling factor determiner and outputs it to an adder as the quantization target of a second stage.

Abstract: The present invention relates to information processing technologies and discloses an encoding and decoding method and device to solve the poor decoding quality problem. The technical solution of the present invention includes: encoding each sample of an input signal to generate an encoded signal of a core layer; comparing residuals of all or a part of the samples of the input signal with encoding thresholds, where the residuals are generated by core layer encoding, and performing encoding according to comparison results to generate an encoded signal of an enhancement layer; and writing the encoded signal of the core layer and the encoded signal of the enhancement layer into a bitstream to generate an encoded signal of the input signal.

Abstract: An audio decoding device of the present invention includes: a decoding unit decoding a stream to a spectrum coefficient, and outputting stream information when a frame included in the stream cannot be decoded; an orthogonal transformation unit transforming the spectrum coefficient to a time signal; a correction unit generating a correction time signal based on an output waveform within a reference section that is in a section that overlaps between an error frame section to which the stream information is outputted and an adjacent frame section and that is a section in the middle of the adjacent frame section, when the decoding unit outputs the stream information: and an output unit generating the output waveform by synthesizing the correction time signal and the time signal.

Abstract: Devices, systems, and methods for recording call sessions over a VoIP network using a distributed record server architecture are disclosed. An example recording device for recording segments of a call session includes a record server configured to receive an agent voice data stream and an external caller voice data stream from an agent telephone station, and a file repository configured to store voice data and call data associated with each recorded segment of the call session. The recording device is configured to tag recorded segments of each call session, which can be later used by a third-party application or database to check the status and/or integrity of the recorded call session.

Abstract: A voice recognition system comprises: a voice input unit that receives an input signal from a voice input element and output it; a voice detection unit that detects an utterance segment in the input signal; a voice recognition unit that performs voice recognition for the utterance segment; and a control unit that outputs a control signal to at least one of the voice input unit and the voice detection unit and suppresses a detection frequency if the detection frequency satisfies a predetermined condition.