Abstract: In the field of communications, a method and a device for determining a decoding mode of in-band signaling are provided, which improve accuracy of in-band signaling decoding. The method includes: calculating a probability of each decoding mode of in-band signaling of a received signal at a predetermined moment by using a posterior probability algorithm; and from the calculated probabilities of the decoding modes, selecting a decoding mode having a maximum probability value as a decoding mode of the in-band signaling of the received signal at the predetermined moment. The method and the device are mainly used in a process for determining a decoding mode of in-band signaling in a speech frame transmission process.

Abstract: The described method and system provide for HMI steering for a telematics-equipped vehicle based on likelihood to exceed eye glance guidelines. By determining whether a task is likely to cause the user to exceed eye glance guidelines, alternative HMI processes may be presented to a user to reduce ASGT and EORT and increase compliance with eye glance guidelines. By allowing a user to navigate through long lists of items through vocal input, T9 text input, or heuristic processing rather than through conventional presentation of the full list, a user is much more likely to comply with the eye glance guidelines. This invention is particularly useful in contexts where users may be searching for one item out of a plurality of potential items, for example, within the context of hands-free calling contacts, playing back audio files, or finding points of interest during GPS navigation.

Abstract: According to one embodiment, an electronic apparatus includes: a microphone; a storage unit which stores at least one of record start instruction keyword; a voice recognition section which recognizes a voice content that is input through the microphone; and a record start execution section which, in a case where the voice content recognized by the voice recognition section is coincident with the record start instruction keyword, executes a record start.

Abstract: A method of analyzing an audio signal is disclosed. A digital representation of an audio signal is received and a first output function is generated based on a response of a physiological model to the digital representation. At least one property of the first output function may be determined. One or more values are determined for use in analyzing the audio signal, based on the determined property of the first output function.

Abstract: Provided are an apparatus and a method for integrally encoding and decoding a speech signal and a audio signal. The encoding apparatus may include: an input signal analyzer to analyze a characteristic of an input signal; a first conversion encoder to convert the input signal to a frequency domain signal, and to encode the input signal when the input signal is a audio characteristic signal; a Linear Predictive Coding (LPC) encoder to perform LPC encoding of the input signal when the input signal is a speech characteristic signal; a frequency band expander for expanding a frequency band of the input signal whose output is transmitted to either the first conversion encoder or the LPC encoder based on the input characteristic; and a bitstream generator to generate a bitstream using an output signal of the first conversion encoder and an output signal of the LPC encoder.

Abstract: A method for managing an interaction of a calling party to a communication partner is provided. The method includes automatically determining if the communication partner expects DTMF input. The method also includes translating speech input to one or more DTMF tones and communicating the one or more DTMF tones to the communication partner, if the communication partner expects DTMF input.

Abstract: A method for providing text to speech from digital content in an electronic device is described. Digital content including a plurality of words and a pronunciation database is received. Pronunciation instructions are determined for the word using the digital content. Audio or speech is played for the word using the pronunciation instructions. As a result, the method provides text to speech on the electronic device based on the digital content.

Abstract: Embodiments of the present invention use one or more audible tones to communicate metadata during a transfer of an audio file. Embodiments of the present invention communicate an audio file from a speaker in a recording device (e.g., a recordable book, toy, computing device) to a microphone in a receiving device. The audio file is transferred by audibly broadcasting the audio file content. The audio file may be a recording made by the user (e.g., the user singing a song, a child responding to a storybook prompt intended to elicit a response). The file transfer process uses one or more audible tones, such as dual-tone multi-frequency signaling (“DTMF”) tones to communicate metadata associated with the audio file. Audible tones may also be used to communicate commands that delineate the beginning and/or end of a file broadcast.

Abstract: A network communication node includes an audio outputter that outputs an audible representation of data to be provided to a requester. The network communication node also includes a processor that determines a categorization of the data to be provided to the requester and that varies a pause between segments of the audible representation of the data in accordance with the categorization of the data to be provided to the requester.

Abstract: A quantizing method is provided that includes quantizing an input signal by selecting one of a first quantization scheme not using an inter-frame prediction and a second quantization scheme using the inter-frame prediction, in consideration of one or more of a prediction mode, a predictive error and a transmission channel state.

Abstract: Systems and methods that can be utilized to convert a voice communication received over a telecommunication network to text are described. In an illustrative embodiment, a call processing system coupled to a telecommunications network receives a call from a caller intended for a first party, wherein the call is associated with call signaling information. At least a portion of the call signaling information is stored in a computer readable medium. A greeting is played the caller, and a voice communication from the caller is recorded. At least a portion of the voice communication is converted to text, which is analyzed to identify portions that are inferred to be relatively more important to communicate to the first party. A text communication is generated including at least some of the identified portions and including fewer words than the recorded voice communication. At least a portion of the text communication is made available to the first party over a data network.

Abstract: A method, medium, and apparatus encoding and/or decoding a multichannel audio signal. The method includes detecting the type of spatial extension data included in an encoding result of an audio signal, if the spatial extension data is data indicating a core audio object type related to a technique of encoding core audio data, detecting the core audio object type; decoding core audio data by using a decoding technique according to the detected core audio object type, if the spatial extension data is residual coding data, decoding the residual coding data by using the decoding technique according to the core audio object type, and up-mixing the decoded core audio data by using the decoded residual coding data. According to the method, the core audio data and residual coding data may be decoded by using an identical decoding technique, thereby reducing complexity at the decoding end.

Abstract: An audio buffer is used to capture audio in anticipation of a user command to do so. Sensors and processor activity may be monitored, looking for indicia suggesting that the user command may be forthcoming. Upon detecting such indicia, a circular buffer is activated. Audio correction may be applied to the audio stored in the circular buffer. After receiving the user command instructing the device to process or record audio, at least a portion of the audio that was stored in the buffer before the command is combined with audio received after the command. The combined audio may then be processed, transmitted or stored.

Abstract: A method for communication includes receiving modulated signals, which convey encoded speech. A measure of information entropy associated with the received signals is estimated. A speech encoding scheme is selected responsively to the estimated measure of the information entropy. A request to encode subsequent speech using the selected speech encoding scheme is sent to a transmitter.

Abstract: A system and method for performing dual mode speech recognition, employing a local recognition module on a mobile device and a remote recognition engine on a server device. The system accepts a spoken query from a user, and both the local recognition module and the remote recognition engine perform speech recognition operations on the query, returning a transcription and confidence score, subject to a latency cutoff time. If both sources successfully transcribe the query, then the system accepts the result having the higher confidence score. If only one source succeeds, then that result is accepted. In either case, if the remote recognition engine does succeed in transcribing the query, then a client vocabulary is updated if the remote system result includes information not present in the client vocabulary.

Abstract: An audio data processing system is a client-server system including an audio data communication device and an audio data processing device which are linked together via a communication network. The audio data communication device includes an acoustic generator, a control device, a transmitter and a receiver in connection with first and second storage areas. The transmitter sequentially transmits a time series of unprocessed data DA[n] stored in the first storage area, while the receiver sequentially receives a time series of processing-completed data DB[n] from the acoustic data processing device so that processing-completed data are stored in the second storage area and sequentially reproduced. When specific processing-completed data is not stored in the second storage area, the control device designates and reproduces specific unprocessed data, which is unprocessed acoustic data corresponding to specific processing-completed data.

Abstract: A method for setting a voice tag is provided, which comprises the following steps. First, counting a number of phone calls performed between a user and a contact person. If the number of phone calls exceeds a predetermined times or a voice dialing performed by the user is failed before calling to the contact person within a predetermined duration, the user is inquired whether or not to set a voice tag corresponding to the contact person after the phone call is complete. If the user decides to set the voice tag, a voice training procedure is executed for setting the voice tag corresponding to the contact person.

Abstract: Methods and systems for extracting speech from such packet streams. The methods and systems analyze the encoded speech in a given packet stream, and automatically identify the actual speech coding scheme that was used to produce it. These techniques may be used, for example, in interception systems where the identity of the actual speech coding scheme is sometimes unavailable or inaccessible. For instance, the identity of the actual speech coding scheme may be sent in a separate signaling stream that is not intercepted. As another example, the identity of the actual speech coding scheme may be sent in the same packet stream as the encoded speech, but in encrypted form.

Abstract: An apparatus for encoding includes a first domain converter, a switchable bypass, a second domain converter, a first processor and a second processor to obtain an encoded audio signal having different signal portions represented by coded data in different domains, which have been coded by different coding algorithms. Corresponding decoding stages in the decoder together with a bypass for bypassing a domain converter allow the generation of a decoded audio signal with high quality and low bit rate.

Abstract: A method is presented that uses steganographic codeword(s) carried in a speech payload in such a way that (i) the steganographic codeword(s) survive compression and/or transcoding as the payload travels from a transmitter to a receiver across at least one diverse network, and (ii) the embedded steganographic codeword(s) do not degrade the perceived voice quality of the received signal below an acceptable level. The steganographic codewords are combined with a speech payload by summing the amplitude of a steganographic codeword to the amplitude of the speech payload at a relatively low steganographic-to-speech bit rate. Advantageously, the illustrative embodiment of the present invention enables (i) steganographic codewords to be decoded by a compliant receiver and applied accordingly, and (ii) legacy or non-compliant receivers to play the received speech payload with resultant voice quality that is acceptable to listeners even though the steganographic codeword(s) remain in the received speech payload.

Abstract: The present disclosure discloses a speech recognition method and a terminal, which belong to the field of communications. The method comprises: receiving speech information inputted by a user; acquiring the current environment information, and judging whether the speech information needs to be played according to the current environment information; and recognizing the speech information as text information, when it is judged that the speech information needs not to be played. The terminal comprises an acquisition module, a judgment module and a recognition module. The present disclosure provides the speech receiver with a speech recognition function, when the speech information of the instant messaging is received by the terminal, it can help the receiver to normally acquire the content to be expressed by the speech sender under an inconvenient situation.

Abstract: A method of operating an audio processing device to improve a user's perception of an input sound includes defining a critical frequency fcrit between a low frequency range and a high frequency range, receiving an input sound by the audio processing device, and analyzing the input sound in a number of frequency bands below and above the critical frequency. The method also includes defining a cut-off frequency fcut below the critical frequency fcrit, identifying a source frequency band above the cut-off frequency fcut, and extracting an envelope of the source band. Further, the method identifying a corresponding target band below the critical frequency fcrit, extracting a phase of the target band, and combining the envelope of the source band with the phase of the target band.

Abstract: In an audio output terminal device, a buffer control unit adjusts the buffer size of a jitter buffer in accordance with the setting of a sound output mode instructed in an instruction receiving unit. If the instruction receiving unit acknowledges an instruction for setting an audio output mode that requires low delay in outputting sound, the buffer control unit reduces the buffer size of the jitter buffer. Further, the buffer control unit controls, in accordance with the instructed setting of the sound output mode, timing for allowing a media buffer to transmit one or more voice packets to the jitter buffer.

Abstract: Disclosed is a frame comparison apparatus and method for comparing frames included in an audio signal by using spectrum information. The frame comparison apparatus includes a spectrum information estimation apparatus for receiving an audio signal and estimating and outputting spectrum information for the respective frames included in the audio signal, an estimation operation option determiner for determining an estimation order of the spectrum information estimated from the spectrum information estimation apparatus, a frame comparison option determiner for determining a comparison order for the frames output from the spectrum information estimation apparatus, and a frame comparator for determining a comparison target frame which is a comparison target for a current frame included in the audio signal, comparing the spectrum information for the current frame with the spectrum information for the comparison target frame, and outputting a comparison result value.

Abstract: An audio decoding system including a decoder decoding a first part of audio data, and an audio buffer compressor compressing and storing the decoded first part of audio data in a first time interval and decompressing the stored first part of audio data in a second time interval.

Abstract: Some embodiments disclosed herein store a target application and a dictation application. The target application may be configured to receive input from a user. The dictation application interface may include a full overlay mode option, where in response to selection of the full overlay mode option, the dictation application interface is automatically sized and positioned over the target application interface to fully cover a text area of the target application interface to appear as if the dictation application interface is part of the target application interface. The dictation application may be further configured to receive an audio dictation from the user, convert the audio dictation into text, provide the text in the dictation application interface and in response to receiving a first user command to complete the dictation, automatically copy the text from the dictation application interface and inserting the text into the target application interface.

Abstract: A method of converting speech from the characteristics of a first voice to the characteristics of a second voice, the method comprising: receiving a speech input from a first voice, dividing said speech input into a plurality of frames; mapping the speech from the first voice to a second voice; and outputting the speech in the second voice, wherein mapping the speech from the first voice to the second voice comprises, deriving kernels demonstrating the similarity between speech features derived from the frames of the speech input from the first voice and stored frames of training data for said first voice, the training data corresponding to different text to that of the speech input and wherein the mapping step uses a plurality of kernels derived for each frame of input speech with a plurality of stored frames of training data of the first voice.

Abstract: A speech enhancement system enhances transitions between speech and non-speech segments. The system includes a background noise estimator that approximates the magnitude of a background noise of an input signal that includes a speech and a non-speech segment. A slave processor is programmed to perform the specialized task of modifying a spectral tilt of the input signal to match a plurality of expected spectral shapes selected by a Codec.

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: Method, system, and computer program product for voice transformation are provided. The method includes transforming a source speech using transformation parameters, and encoding information on the transformation parameters in an output speech using steganography, wherein the source speech can be reconstructed using the output speech and the information on the transformation parameters. A method for reconstructing voice transformation is also provided including: receiving an output speech of a voice transformation system wherein the output speech is transformed speech which has encoded information on the transformation parameters using steganography; extracting the information on the transformation parameters; and carrying out an inverse transformation of the output speech to obtain an approximation of an original source speech.

Abstract: An information processing method and an electronic device are disclosed. The information processing method is applied to a first electronic device. When the device orientation of the first electronic device is a first device orientation at a first time instant, the method includes: obtaining, by a first sensor of the first electronic device, a first sensing parameter indicating that the device orientation is a second device orientation at a second time instant after the first time instant; determining, based on the first sensing parameter, whether the second device orientation differs from the first device orientation, and obtaining a first determination; and generating a first instruction for entering into a voice record state when the second device orientation differs from the first device orientation and the second device orientation meets a predetermined condition.

Abstract: A method for measuring speech signal quality by an electronic device is described. The method includes obtaining a modified single-channel speech signal. The method also includes estimating multiple objective distortions based on the modified single-channel speech signal. The multiple objective distortions include at least one foreground distortion and at least one background distortion. The method further includes estimating a foreground quality and a background quality based on the multiple objective distortions. The method additionally includes estimating an overall quality based on the foreground quality and the background quality.

Abstract: A voice correction device includes a detector that detects a response from a user, a calculator that calculates an acoustic characteristic amount of an input voice signal, an analyzer that outputs an acoustic characteristic amount of a predetermined amount when having acquired a response signal due to the response from the detector, a storage unit that stores the acoustic characteristic amount output by the analyzer, a controller that calculates an correction amount of the voice signal on the basis of a result of a comparison between the acoustic characteristic amount calculated by the calculator and the acoustic characteristic amount stored in the storage unit, and a correction unit that corrects the voice signal on the basis of the correction amount calculated by the controller.

Abstract: A method for decoding an audio signal in a decoder having a CELP-based decoder element including a fixed codebook component, at least one pitch period value, and a first decoder output, wherein a bandwidth of the audio signal extends beyond a bandwidth of the CELP-based decoder element. The method includes obtaining an up-sampled fixed codebook signal by up-sampling the fixed codebook component to a higher sample rate, obtaining an up-sampled excitation signal based on the up-sampled fixed codebook signal and an up-sampled pitch period value, and obtaining a composite output signal based on the up-sampled excitation signal and an output signal of the CELP-based decoder element, wherein the composite output signal includes a bandwidth portion that extends beyond a bandwidth of the CELP-based decoder element.

Abstract: Method for speaker identification includes detecting a target speaker's utterance locally; extracting features from the detected utterance locally, analyzing the extracted features in the local device to obtain information on the speaker identification and/or encoding the extracted features locally, transmitting the encoded extracted features to a remote server, decoding and analyzing the received extracted features by the server to obtain information on the speaker identification, and transmitting the information on the speaker identification from the server to the location where the speaker's utterance was detected. The method further includes detecting speech activity locally. Extracting features, encoding the extracted features, and/or transmitting the encoded extracted features to the server, are only performed if speech activity above some predetermined threshold is detected.

Abstract: A method for coding and decoding an audio signal or speech signal and an apparatus adopting the method are provided.

Abstract: A system and method for providing an audio representation of a name includes providing a list of a plurality of users of a network and respective presence information regarding each of the plurality of users; receiving a request from an endpoint to receive an audio representation of a name of a particular user of the plurality of users, and providing the audio representation to the endpoint. Moreover, the audio representation of the name at least generally approximates a pronunciation of the name as pronounced by the particular user.

Abstract: Method and apparatus that dynamically adjusts operational parameters of a text-to-speech engine in a speech-based system. A voice engine or other application of a device provides a mechanism to alter the adjustable operational parameters of the text-to-speech engine. In response to one or more environmental conditions, the adjustable operational parameters of the text-to-speech engine are modified to increase the intelligibility of synthesized speech.

Abstract: A computer-implemented method and apparatus are disclosed for decoding an encoded data signal. In one embodiment, the method includes accessing, in a memory, a set of signal elements. The encoded data signal is received at a computing device. The signal includes signal fragments each having a projection value and an index value. The projection value has been calculated as a function of at least one signal element of the set of signal elements and at least a portion of the data signal. The index value associates its respective signal fragment with the at least one signal element used to calculate the projection value. The computing device determines amplitude values based on the projection values in the signal fragments. The decoded signal is determined using the amplitude values and the signal elements associated with the at least some of the signal fragments.

Abstract: This document describes various techniques for dual-band speech encoding. In some embodiments, a first type of speech feature is received from a remote entity, an estimate of a second type of speech feature is determined based on the first type of speech feature, the estimate of the second type of speech feature is provided to a speech recognizer, speech-recognition results based on the estimate of the second type of speech feature are received from the speech recognizer, and the speech-recognition results are transmitted to the remote entity.

Abstract: There is a need to enable decompression of a speech signal even if no network synchronizing signal is output from a baseband processing portion. For this purpose, an information processing device includes a first serial interface. The first serial interface includes a notification signal generation circuit that generates a notification signal each time compressed data incorporated from the baseband processing portion reaches a predetermined data quantity, and notifies a speech processing portion of this state using the notification signal. The speech processing portion includes a synchronizing signal generation circuit that generates a network synchronizing signal based on the notification signal. A clock signal for PCM communication is generated based on the network synchronizing signal. A speech signal can be decompressed even if no network synchronizing signal is output from the baseband processing portion.

Abstract: Aspects relate to machine recognition of human voices in live or recorded audio content, and delivering text derived from such live or recorded content as real time text, with contextual information derived from characteristics of the audio. For example, volume information can be encoded as larger and smaller font sizes. Speaker changes can be detected and indicated through text additions, or color changes to the font. A variety of other context information can be detected and encoded in graphical rendition commands available through RTT, or by extending the information provided with RTT packets, and processing that extended information accordingly for modifying the display of the RTT text content.

Abstract: An electronic device includes a camera and two microphones. The space in front of the camera is divided into a plurality of imaginary cubic areas. Each imaginary cubic area is associated with a delay parameter. The camera locates a face of a user and determines an imaginary cubic area in which the face is located from the plurality of imaginary cubic areas. A wave beam pointing to the imaginary cubic area is calculated according to the delay parameter associated with the imaginary cubic area. The two microphone record voices within a range of the wave beam. A voice recording method is also provided.

Abstract: Systems and methods are provided for voice data processing. For example, a first data packet included in voice data transmitted by a client is received; the first data packet is stored in a storage area; whether to process one or more second data packets stored in the storage area is determined based on at least information associated with a type of the first data packet and a current storage state of the storage area; in response to a determination to process the second data packets, voice resources are applied for; and the second data packets stored in the storage area are processed using the voice resources.

Abstract: A multi-layered speech recognition apparatus and method, the apparatus includes a client checking whether the client recognizes the speech using a characteristic of speech to be recognized and recognizing the speech or transmitting the characteristic of the speech according to a checked result; and first through N-th servers, wherein the first server checks whether the first server recognizes the speech using the characteristic of the speech transmitted from the client, and recognizes the speech or transmits the characteristic according to a checked result, and wherein an n-th (2?n?N) server checks whether the n-th server recognizes the speech using the characteristic of the speech transmitted from an (n?1)-th server, and recognizes the speech or transmits the characteristic according to a checked result.

Abstract: An apparatus for processing an audio signal and method thereof are disclosed. The present invention includes receiving, by an audio processing apparatus, an audio signal including a first data of a first block encoded with rectangular coding scheme and a second data of a second block encoded with non-rectangular coding scheme; receiving a compensation signal corresponding to the second block; estimating a prediction of an aliasing part using the first data; and, obtaining a reconstructed signal for the second block based on the second data, the compensation signal and the prediction of aliasing part.

Abstract: An audio analysis system includes a terminal apparatus and a host system. The terminal apparatus acquires an audio signal of a sound containing utterances of a user and another person, discriminates between portions of the audio signal corresponding to the utterances of the user and the other person, detects an utterance feature based on the portion corresponding to the utterance of the user or the other person, and transmits utterance information including the discrimination and detection results to the host system. The host system detects a part corresponding to a conversation from the received utterance information, detects portions of the part of the utterance information corresponding to the user and the other person, compares a combination of plural utterance features corresponding to the portions of the part of the utterance information of the user and the other person with relation information to estimate an emotion, and outputs estimation information.

Abstract: A difference signal calculating unit of a noise detecting device calculates a difference between the amplitudes of a residual signal at each sample timing and a residual signal at the preceding sample timing. A difference signal comparing unit determines whether or not an impulsive noise is present on the basis of the difference signal at the current sample timing, and the difference signal at each sample timing within a predetermined duration from the current sample timing.

Abstract: A voice recording and playback device of the present invention is capable of storage management of generated voice data, to which date information has been attached, as voice files, and comprises a display section capable of calendar display, and a control section for, at the time of retrieving voice files from a storage section, performing movable identification on a calendar display, as well as retrieving voice files that have been stored in the storage section based on date information attached to the files, and performing display of results of this retrieval indicating the existence of voice files close to day display on the calendar display, wherein the control section moves the identification position based on an instruction operation by the retrieval instructions section, and generates a notification in accordance with voice files that exist on the date of the identification position that has been moved.

Abstract: A system that incorporates teachings of the present disclosure may initiate a communication session with a member device of a social network and may activate a speech capture element based on a pattern of prior speech messages. A speech message may be detected at the speech capture element and, in turn, transmitted to the member device.