Abstract: An apparatus for generating a synthesis audio signal using a patching control signal has a first converter, a spectral domain patch generator, a high frequency reconstruction manipulator and a combiner. The first converter is configured for converting a time portion of an audio signal into a spectral representation. The spectral domain patch generator is configured for performing a plurality of different spectral domain patching algorithms, wherein each patching algorithm generates a modified spectral representation having spectral components in an upper frequency band derived from corresponding spectral components in a core frequency band of the audio signal.

Abstract: An apparatus for generating a high band extension of a low band excitation signal (eLB) defined by parameters representing a CELP encoded audio signal includes the following elements: upsamplers (20) configured to upsample a low band fixed codebook vector (uFCB) and a low band adaptive codebook vector (uACB) to a predetermined sampling frequency. A frequency shift estimator (22) configured to determine a modulation frequency (?) from an estimated measure representing a fundamental frequency (F0) of the audio signal. A modulator (24) configured to modulate the upsampled low band adaptive codebook vector (uACB?) with the determined modulation frequency to form a frequency shifted adaptive codebook vector. A compression factor estimator (28) configured to estimate a compression factor. A compressor (34) configured to attenuate the frequency shifted adaptive codebook vector and the upsampled fixed codebook vector (uFCB?.) based on the estimated compression factor.

Abstract: A method, medium, and apparatus hierarchically coding/decoding audio data, such as bit sliced arithmetic coding (BSAC), such that payloads of audio data and extension data can be grouped and interleaved according to priority so that some groups of the payloads are dropped, and the remainder of groups are transmitted. Therefore, extension data that is more important than a top layer of audio data, in terms of reproducing of original sounds, can be transmitted with priority.

Abstract: Methods and systems for communicating data on a cellular telephone voice channel are disclosed. The method includes segmenting a data stream into one or more n-bit symbols; identifying a human vocal sound corresponding to each n-bit symbol according to a predetermined assignment of each n-bit symbol to a human vocal sound; and retrieving data representing the human vocal sound, wherein data representing the human vocal sound is configured to be passed through a vocoder.

Abstract: Provided are a method and apparatus for measuring sound quality in a variable band multi-codec.

Abstract: In one example, a mobile device encodes a digital bitstream using a particular set of modulation parameters to generate an audio signal that has different audio tones selected to pass through a vocoder of the mobile device. The particular set of modulation parameters is optimized for a subset of a plurality of vocoding modes without a priori knowledge of which one of the vocoding modes is currently operated by the vocoder. The mobile device conducts transmissions over the wireless telecommunications network through the vocoder using the particular set of modulation parameters, and monitors these transmissions for errors. If the errors reach a threshold, then the vocoder may be using one of the vocoding modes that are not included in the subset for which the particular set of modulation parameters is optimized, and accordingly, the modulation device switches from the particular set of modulation parameters to a different set of modulation parameters.

Abstract: Provided is a technology of controlling an electronic device using a broadcasting system. A control signal may be modulated to an audible frequency band and thereby be transmitted from a transmission apparatus to a reception apparatus using the broadcasting system. The reception apparatus may reproduce the control signal of the audible frequency band. A controller may perform a control operation based on the control signal.

Abstract: A scalable encoding apparatus, a scalable decoding apparatus and the like are disclosed which can achieve a band scalable LSP encoding that exhibits both a high quantization efficiency and a high performance. In these apparatuses, a narrow band-to-wide band converter receives and converts a quantized narrow band LSP to a wide band, and then outputs the quantized narrow band LSP as converted (i.e., a converted wide band LSP parameter) to an LSP-to-LPC converter. The LSP-to-LPC converter converts the quantized narrow band LSP as converted to a linear prediction coefficient and then outputs it to a pre-emphasizer. The pre-emphasizer calculates and outputs the pre-emphasized linear prediction coefficient to an LPC-to-LSP converter. The LPC-to-LSP converter converts the pre-emphasized linear prediction coefficient to a pre-emphasized quantized narrow band LSP as wide band converted, and then outputs it to a prediction quantizer.

Abstract: According to one embodiment, a communication network includes an analog voice bridge coupled to two secure network domains that each have a differing security level. The analog voice bridge includes two codecs that are coupled together through an analog voice line that transfers analog voice signals. The analog voice bridge is coupled to each secure network domain through a network switch that transfers data packet streams from their respective networks to the codecs while restricting data packets that are not associated with the data packet stream.

Abstract: In one aspect of the present invention, a network gateway is configured to facilitate on line and off line bi-directional communication between a number of near end data and telephony devices with far end data termination devices via a hybrid fiber coaxial network and a cable modem termination system. The described network gateway combines a QAM receiver, a transmitter, a DOCSIS MAC, a CPU, a voice and audio processor, an Ethernet MAC, and a USB controller to provide high performance and robust operation.

Abstract: 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 S1(k). A pitch coefficient setting section outputs a pitch coefficient T by gradually changing it. The filtering section calculates an estimated value S?2(k) of a second spectrum S2(k) based on a pitch coefficient T. A search section calculates the degree of similarity between S2(k) and S?2(k). At this time, pitch coefficient T? corresponding to the maximum calculated degree of similarity is given to a filter coefficient calculation section. The filter coefficient calculation section determines a filter coefficient ?i using this pitch coefficient T?.

Abstract: A device for lost frame concealment comprises: a lost frame detector for detecting whether a voice frame is lost, a decoding module for decoding the current voice frame, a low band delay module for delaying the low band signal, a low band signal recovering module for recovering the lost low band signal, a high band lost frame concealment module for processing the lost frame concealment for the high band signal, and a QMF synthesis filter for synthetically filtering the low band signal and the high band signal. The invention makes full use of the delay of the coding/decoding device itself, enhances the effect of lost frame concealment for the low band signal and the high band signal, and introduces no nearby delay during the process of lost frame concealment.

Abstract: 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.

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: Provided is an encoding device which can obtain a sound quality preferable for auditory sense even if the number of information bits is small. The encoding device includes a shape quantization unit (111) having: a section search unit (121) which searches for a pulse for each of bands into which a predetermined search section is divided; and a whole search unit (122) which performs search for a pulse over the entire search section. The shape of an input spectrum is quantized by a small number of pulse positions and polarities. A gain quantization unit (112) calculates a gain of the pulse searched by the shape quantization unit (111) and quantizes the gain for each of the bands.

Abstract: 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.

Abstract: Provided is a decoding device and others which can mitigate the spectrum energy discontinuity and improves the decoded signal quality even when a sub-band is subjected to a spectrum attenuation process in the band extension method. The device includes: a substitution unit (181) which substitutes a second layer decoding spectrum of the sub-band indicated by the sub-band information with a third layer decoding error spectrum of the sub-band indicated by the sub-band information; and an adjusting unit (185) which makes an adjustment so that the energy of the second layer decoding spectrum after the substitution approaches the energy of the spectrum before the replacement.

Abstract: Disclosed is a decoding device and others 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 decoding unit (202) decodes first layer encoded information to generate a first layer decoded signal; a second layer decoding unit (204) decodes second layer encoded information to generate a second layer decoded signal; a spectrum decoding unit (205) performs a band extension process by using the second layer decoded signal and the first layer decoded signal up-sampled in an up-sampling unit (203) so as to generate a all-band decoded signal; and a switch (206) outputs the first layer decoded signal or the all-band decoded signal according to the control information generated in a control unit (201).

Abstract: A codebook generation system and associated methods are generally described herein.

Abstract: A method and apparatus that manages speech decoders in a communication device is disclosed. The method may include detecting a change in transmission rate from a higher rate to a lower rate, decoding and shifting a first, second and third received first decoder set of frame parameters, generating a first decoder output audio frame from the previously shifted frame parameters, generating a first, second and third second decoder audio fill frame, the second decoder being a higher rate decoder than first decoder, outputting a first and second second decoder audio fill frame, combining the first decoder audio frame and the third second decoder audio fill frame with overlapping triangular windows, and outputting combined first decoder and second decoder frames to an audio buffer for subsequent transmission to a user of the communication device. In an alternative embodiment, another method may include detecting and processing a change in transmission rate from a lower rate to a higher rate.

Abstract: The invention proposes several methods for codec handling. In specific, methods involving providing a supported codec list of a Call Control Server are described. A node receives information, whether a terminal supports a wideband codec, wherein the information is received in call set up signaling from the terminal of the subscriber. Furthermore, configuration information is retrieved, whether a Radio Access Node supports the wideband codec. Additionally, information is retrieved, whether a media gateway supports the wideband codec, wherein the information is either provided by the operator or retrieved from the media gateway (MGW1, MGW2, MGWx). The information is analyzed and in response to the analysis a supported codec list is provided. Furthermore, alternative embodiments and devices adapted for the methods are disclosed.

Abstract: A system and method for data communication over a cellular communications network that allows the transmission of digital data over a voice channel using a vocoder that operates in different modes depending upon characteristics of the inputted signal it receives. To prepare the digital data for transmission, one or more carrier signals are encoded with the digital data using one of a number of modulation schemes that utilize differential phase shift keying to give the modulated carrier signal certain periodicity and energy characteristics that allow it to be transmitted by the vocoder at full rate. The modulation schemes include DPSK using either a single or multiple frequency carriers, combined FSK-DPSK modulation, combined ASK-DPSK, PSK with a phase tracker in the demodulator, as well as continuous signal modulation (ASK or FSK) with inserted discontinuities that can be independent of the digital data.

Abstract: A mass-scale, user-independent, device-independent, voice messaging system that converts unstructured voice messages into text for display on a screen is disclosed. The system comprises (i) computer implemented sub-systems and also (ii) a network connection to human operators providing transcription and quality control; the system being adapted to optimise the effectiveness of the human operators by further comprising 3 core sub-systems, namely (i) a pre-processing front end that determines an appropriate conversion strategy; (ii) one or more conversion resources; and (iii) a quality control sub-system.

Abstract: Provided are a method and apparatus for measuring sound quality in a variable band multi-codec.

Abstract: A sound encoding device enabling the amount of delay to be kept small and the distortion between frames to be mitigated. In the sound encoding device, a window multiplication part (211) of a long analysis section (21) multiplies a long analysis frame signal of analysis length M1 by an analysis window, the resultant signal multiplied by the analysis window is outputted to an MDCT section (212), and the MDCT section (212) performs MDCT of the input signal to obtain the transform coefficients of the long analysis frame and outputs it to a transform coefficient encoding section (30). The window multiplication part (221) of a short analysis section (22) multiplies a short analysis frame signal of analysis length M2 (M2<M1) by an analysis window and the resultant signal multiplied by the analysis window is outputted to the MDCT section (222).

Abstract: The present invention includes a time-division-multiple-access (TDMA) communication system having a base station and at least one mobile station, each transmitting and receiving an analog radio-frequency signal carrying digitally coded speech. The speech is encoded using a vocoder which samples a voice signal at variable encoding rates. During periods when the radio-frequency channel is experiencing high levels of channel interference, the encoded voice channel having a lower encoding rate is chosen. This low-rate encoded voice is combined with the high degree of channel coding necessary to ensure reliable transmission. When the radio-frequency channel is experiencing low levels of channel interference, less channel coding is necessary and the vocoder having a higher encoding rate is used. The high-rate encoded voice is combined with the lower degree of channel coding necessary to ensure reliable transmission.

Abstract: A conferencing system is provided that utilizes both time domain signal mixing and direct signal fast transcoding. An exemplary embodiment of the present invention utilizes both time domain signal mixing and direct signal fast transcoding to process a bit-stream from a same channel during a conference.