Abstract: An apparatus includes a plurality of applications and an integrator having a voice recognition module configured to identify at least one voice command from a user. The integrator is configured to integrate information from a remote source into at least one of the plurality of applications based on the identified voice command. A method includes analyzing speech from a first user of a first mobile device having a plurality of applications, identifying a voice command based on the analyzed speech using a voice recognition module, and incorporating information from the remote source into at least one of a plurality of applications based on the identified voice command.

Abstract: Provided is a communication apparatus for direct communication between networks of different types. The communication apparatus includes a transmission data selector determining whether or not data input from a first communication network is speech data, a data processor digitizing and packetizing the data transferred from the transmission data selector, and a modem for converting the digitized and packetized data into analog data and then directly transmitting the analog data to a second communication network different from the first communication network through a speech channel.

Abstract: In a method of improving perceived loudness and sharpness of a reconstructed speech signal delimited by a predetermined bandwidth, performing the steps of providing (S10) the speech signal, and separating (S20) the provided signal into at least a first and a second signal portion. Subsequently, adapting (S30) the first signal portion to emphasize at least a predetermined frequency or frequency interval within the first bandwidth portion. Finally, reconstructing (S40) the second signal portion based on at least the first signal portion, and combining (S50) the adapted first signal portion and the reconstructed second signal portion to provide a reconstructed speech signal with an overall improved perceived loudness and sharpness.

Abstract: The present invention relates to means and methods of automated difference recognition between speech and music signals in voice communication systems, devices, telephones, and methods, and more specifically, to systems, devices, and methods that automate control when either speech or music is detected over communication links. The present invention provides a novel system and method for monitoring the audio signal, analyze selected audio signal components, compare the results of analysis with a pre-determined threshold value, and classify the audio signal either as speech or music.

Abstract: Methods and apparatus for voice and data interlacing in a system having a shared antenna. In one embodiment, a voice and data communication system has a shared antenna for transmitting and receiving information in time slots, wherein the antenna can only be used for transmit or receive at a given time. The system determines timing requirements for data transmission and reception and interrupts data transmission for transmission of speech in selected intervals while meeting the data transmission timing and throughput requirements. The speech can be manipulated to fit with the selected intervals, to preserve the intelligibility of the manipulated speech.

Abstract: A voice quality measurement device that measures voice quality of a decoded voice signal outputted from a voice decoder unit. The voice quality measurement device includes a packet buffer unit and a voice information monitoring unit. The packet buffer unit accumulates voice packets that arrive non-periodically as voice information, and outputs the voice information to the voice decoder unit periodically. The voice information monitoring unit monitors continuity of the voice information inputted to the voice decoder unit, and calculates an index of voice quality of the decoded voice signal that reflects acceptability of this continuity.

Abstract: An audio coding terminal and method is provided. The terminal includes a coding mode setting unit to set an operation mode, from plural operation modes, for input audio coding by a codec configured to code the input audio based on the set operation mode such that when the set operation mode is a high frame erasure rate (FER) mode the codec codes a current frame of the input audio according to a select frame erasure concealment (FEC) mode of one or more FEC modes. Upon the setting of the operation mode to be the High FER mode the one FEC mode is selected, from the one or more FEC modes predetermined for the High FER mode, to control the codec by incorporating of redundancy within a coding of the input audio or as separate redundancy information separate from the coded input audio according to the selected one FEC mode.

Abstract: An apparatus includes a user input unit, a display unit, a control unit, and a buffer unit. The display unit includes a speed setting menu. The control unit selects a mode from the speed setting menu in response to the selection signal of the user, and controls a compression ratio of a voice codec and a transfer rate of a modem corresponding to a transmission-side radio, and a reception rate of a modem and a restoration rate of a voice codec corresponding to a reception-side radio, based on the selected mode. The buffer unit performs a storage function if there is a difference between the compression ratio of the voice codec and the transfer rate of the modem or if there is a difference between the reception rate of the modem and the restoration rate of the voice codec.

Abstract: The APPARATUSES, METHODS AND SYSTEMS FOR A DIGITAL CONVERSATION MANAGEMENT PLATFORM (“DCM-Platform”) transforms digital dialogue from consumers, client demands and, Internet search inputs via DCM-Platform components into tradable digital assets, and client needs based artificial intelligence campaign plan outputs. In one implementation, The DCM-Platform may capture and examine conversations between individuals and artificial intelligence conversation agents. These agents may be viewed as assets. One can measure the value and performance of these agents by assessing their performance and ability to generate revenue from prolonging conversations and/or ability to effect sales through conversations with individuals.

Abstract: Disclosed is a method for controlling a cordless telephone device for use in a system that allows remote control of a home electric appliance. The method includes a first generation step of causing a first generation unit in a handset to encode audio input via a sound receiving unit in the handset to generate a first stream, and a first transmission step of transmitting the first stream to a base unit. The first generation step includes causing the first generation unit to generate instruction bit information and a first instruction stream when a first trigger indicating a request to start the remote control is given to the first generation unit. The first transmission step includes transmitting the instruction bit information and the first instruction stream to the base unit through a multiplexing scheme that is common to transmission of a first stream generated when the first trigger is not given.

Abstract: Methods, apparatus, and systems for voice processing are provided herein. An exemplary method can be implemented by a terminal. A voice bit stream to be sent can be obtained. Voice control information corresponding to the voice bit stream to be sent can be obtained. The voice control information can be used for a voice server to determine a voice-mixing strategy. The voice bit stream and the voice control information can be sent to the voice server. At least one voice bit stream, returned by the voice server based on the voice-mixing strategy, can be received. The at least one voice bit stream can be outputted.

Abstract: An audio encoder has a window function controller, a windower, a time warper with a final quality check functionality, a time/frequency converter, a TNS stage or a quantizer encoder, the window function controller, the time warper, the TNS stage or an additional noise filling analyzer are controlled by signal analysis results obtained by a time warp analyzer or a signal classifier. Furthermore, a decoder applies a noise filling operation using a manipulated noise filling estimate depending on a harmonic or speech characteristic of the audio signal.

Abstract: A method and system for performing sample rate conversion is provided. The method may include configuring a system to convert a sample rate of a first audio channel of a plurality of audio channels to produce a first audio stream of samples. The system may be dynamically reconfigured to convert a sample rate of a second of the plurality of audio channels to produce a second audio stream of samples, wherein the first and second audio streams are output from the system at the same time. The method may further include arbitrating between request for additional data from the first and second audio stream of samples, where processing of the first channel is suspended when the request corresponds to a second channel that is of higher priority.

Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for generating direct speech messages based on voice commands that include indirect speech messages. In one aspect, a method includes receiving a voice input corresponding to an utterance. A determination is made whether a transcription of the utterance includes a command to initiate a communication to a user and a segment that is classified as indirect speech. In response to determining that the transcription of the utterance includes the command and the segment that is classified as indirect speech, the segment that is classified as indirect speech is provided as input to a machine translator. In response to providing the segment that is classified as indirect speech to the machine translator, a direct speech segment is received from the machine translator. A communication is initiated that includes the direct speech segment.

Abstract: According to certain embodiments, training a transcription system includes accessing recorded voice data of a user from one or more sources. The recorded voice data comprises voice samples. A transcript of the recorded voice data is accessed. The transcript comprises text representing one or more words of each voice sample. The transcript and the recorded voice data are provided to a transcription system to generate a voice profile for the user. The voice profile comprises information used to convert a voice sample to corresponding text.

Abstract: A system that incorporates teachings of the present disclosure may include, for example, a controller configured to obtain information associated with media content, to generate a first group of tones representative of the information associated with the media content, and to generate a media stream comprising the media content and the first group of tones; and a communication interface configured to transmit the media stream to a media device whereby the media device presents the media content and a sequence of tones, where the sequence of tones is generated based at least in part on the first group of tones, where the first group of tones comprises high frequency tones and low frequency tones, and where one of the high and low frequency tones represents a binary one and the other of the high and low frequency tones represents a binary zero. Other embodiments are disclosed.

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: 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 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: 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: 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: 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: 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: 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: 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: 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: 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: 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 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: 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: 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: 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 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: 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.