Abstract: According to various embodiments, an electronic device may include: a transceiver; and at least one processor, wherein the at least one processor is configured to: control the electronic device to establish a wireless communication connection between the electronic device and an external electronic device, generate, by encoding a first frame included in an audio stream, multiple pieces of compressed data corresponding to the first frame, transmit a first packet, including main compressed data corresponding to the first frame from among the multiple pieces of compressed data, to the external electronic device via the transceiver based on the wireless communication connection, and based on a first signal being received from the external electronic device via the transceiver based on the wireless communication connection within a specified time after transmission of the first packet, transmit a second packet, including first sub-compressed data corresponding to the first frame from among the multiple pieces of c

Abstract: An apparatus, method and program for dividing a conversational dialog into utterance. The apparatus includes: a computer processor; a word database for storing spellings and pronunciations of words; a grammar database for storing syntactic rules on words; a pause detecting section which detects a pause location in a channel making a main speech among conversational dialogs inputted in at least two channels; an acknowledgement detecting section which detects an acknowledgement location in a channel not making the main speech; a boundary-candidate extracting section which extracts boundary candidates in the main speech, by extracting pauses existing within a predetermined range before and after a base point that is the acknowledgement location; and a recognizing unit which outputs a word string of the main speech segmented by one of the extracted boundary candidates after dividing the segmented speech into optimal utterance in reference to the word database and grammar database.

Abstract: An error concealment method and apparatus for an audio signal and a decoding method and apparatus for an audio signal using the error concealment method and apparatus. The error concealment method includes selecting one of an error concealment in a frequency domain and an error concealment in a time domain as an error concealment scheme for a current frame based on a predetermined criteria when an error occurs in the current frame, selecting one of a repetition scheme and an interpolation scheme in the frequency domain as the error concealment scheme for the current frame based on a predetermined criteria when the error concealment in the frequency domain is selected, and concealing the error of the current frame using the selected scheme.

Abstract: A high-quality decoded signal is synthesized. A coding method of the present invention includes a local decoding coefficient searching step. The local decoding coefficient searching step includes a replication determining sub-step, a candidate replication shift signal sequence generating sub-step, a distance calculating sub-step, and a minimum distance shift amount finding sub-step. The replication determining sub-step determines, for each source signal sequence to be coded, whether or not a candidate replication shift signal sequence is to be generated from a decoded signal sequence and outputs a replication determination flag. If the replication determination flag indicates that a candidate replication shift signal sequence is to be generated, the candidate replication shift signal sequence generating sub-step generates a candidate replication shift signal sequence for each predetermined candidate signal shift amounts.

Abstract: A multi-object audio encoding and decoding apparatus supporting a post downmix signal may be provided. The multi-object audio encoding apparatus may include: an object information extraction and downmix generation unit to generate object information and a downmix signal from input object signals; a parameter determination unit to determine a downmix information parameter using the extracted downmix signal and the post downmix signal; and a bitstream generation unit to combine the object information and the downmix information parameter, and to generate an object bitstream.

Abstract: Systems and methods are described for managing bit errors present in a series of encoded bits representative of a portion of an audio signal, wherein the series of encoded bits is received over a communication link in an audio communications system. At least one characteristic of a portion of a received modulated carrier signal that is demodulated to produce the series of encoded bits is determined. A number of bit errors present in the series of encoded bits is then determined based on the at least one characteristic. Based on the estimated number of bit errors, one of a plurality of methods for producing a series of digital audio samples representative of the portion of the audio signal is selectively performed. The series of digital audio samples produced by the selected method is then converted into a form suitable for playback to a user.

Abstract: A method for performing packet loss or Frame Erasure Concealment (FEC) for a speech coder receives encoded frames of compressed speech information transmitted from an encoder. The method determines whether an encoded frame has been lost, corrupted in transmission, or erased, synthesizes properly received frames, and decides on an overlap-add window to use in combining a portion of the synthesized speech signal with a subsequent speech signal resulting from a received and decoded packet, where the size of the overlap-add window is based on the unavailability of packets. If it is determined that an encoded frame has been lost, corrupted in transmission, or erased, the method performed an overlap-add operation on the portion of the synthesized speech signal and the subsequent speech signal, using the decided-on overlap-add window.

Abstract: A method for improving packetized speech transmitted over a wireless LAN is disclosed. Speech packets transmitted over the wireless LAN are monitored for errors. Any of the speech packets found to have errors are replaced with synthesized speech packets. The synthesized speech packets may be created from a vocal tract model generated from the received speech packets during periods of time when there are no errors.

Abstract: 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.

Abstract: A system and method for detection of rate determination algorithm errors in variable rate communications system receivers. The disclosed embodiments prevent rate determination algorithm errors from causing audible artifacts such as screeches or beeps. The disclosed system and method detects frames with incorrectly determined data rates and performs frame erasure processing and/or memory state clean up to prevent propagation of distortion across multiple frames. Frames with incorrectly determined data rates are detected by checking illegal rate transitions, reserved bits, validating unused filter type bit combinations and analyzing relationships between fixed code-book gains and linear prediction coefficient gains.

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: A method conceals dropouts in one or more audio channels of a multi-channel arrangement. The method maps transmitted signals into a frequency domain during an error-free signal transmission of two or more channels. A magnitude spectra and spectral filter coefficients are derived. The spectral filter coefficients relate the magnitude spectrum of the audio channel to the magnitude spectrum of at least one other channel. When a dropout occurs, a replacement signal is generated through the filter coefficients and a substitution signal. The filter coefficients may be generated prior to the detection of the dropout.

Abstract: Techniques and tools related to delayed or lost coded audio information are described. For example, a concealment technique for one or more missing frames is selected based on one or more factors that include a classification of each of one or more available frames near the one or more missing frames. As another example, information from a concealment signal is used to produce substitute information that is relied on in decoding a subsequent frame. As yet another example, a data structure having nodes corresponding to received packet delays is used to determine a desired decoder packet delay value.

Abstract: 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 (131) which extracts a high-frequency component of the frequency region from an input audio signal; a high-frequency energy level calculation unit (132) which calculates an energy level of the high-frequency component in a frame unit; an LPF (133) which extracts a low-frequency component of the frequency region from the input audio signal; a low-energy level calculation unit (134) which calculates an energy level of a low-frequency component in a frame unit; an inclination correction coefficient calculation unit (141) multiplies the difference between SNR of the high-frequency component and SNR of the low-frequency component inputted from an adder (140) by a constant and adds a bias component to the product so as to calculate an inclination correction coefficient ?3.

Abstract: The present invention discloses a solution for a speech processing system to determine end-of-utterance (EOU) events. The solution is a modified dual factor technique, where one factor is based upon a number of silence frames received and a second factor is based upon an end-of-path occurrence. The solution permits a set of configurable timeout delay values to be established, which can be configured on an application specific basis by application developers. The solution can speed up EOU determinations made through a dual factor technique, by situationally making finalization determination before a silence frame window is full.

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

Abstract: Disclosed is an audio data packet format for transmitting an IYIPEG-4 HE-AAC frame via a voice channel of a mobile communication network, a method for decoding the audio data packet format, a method for correcting a codec setup error by identifying a codec used to encode sound source data inserted into a data field of voice slot data, based on the sequence number of the voice slot data, and correcting the codec setup error when a codec set up in a mobile communication terminal is different from the codec used to encode the sound source data, and a mobile communication terminal adapted to correct a codec setup error.

Abstract: The present invention provides, methods, computer-readable media, and apparatuses for tuning and adjusting the computational complexity of algorithm that is executed by a signal encoder. The signal encoder may comprise a speech encoder. When a resource shortage on a computer platform is detected, a degree of the resource shortage and a corresponding complexity adjustment for a speech encoder are determined. The speech encoder is then tuned to adjust the computational complexity of an executed speech processing algorithm. The resource shortage may correspond to a computational capability, audio buffer memory, or battery of a mobile device. A speech process being executed by the mobile device is tuned to adjust the computational demands in accordance with a complexity adjustment. A number of iteration rounds may be adjusted while the speech encoder is executing a speech processing algorithm. The iterations may correspond to an algebraic codebook search.