Abstract: Visual information is used to alter or set an operating parameter of an audio signal processor, other than a beamformer. A digital camera captures visual information about a scene that includes a human speaker and/or a listener. The visual information is analyzed to ascertain information about acoustics of a room. A distance between the speaker and a microphone may be estimated, and this distance estimate may be used to adjust an overall gain of the system. Distances among, and locations of, the speaker, the listener, the microphone, a loudspeaker and/or a sound-reflecting surface may be estimated. These estimates may be used to estimate reverberations within the room and adjust aggressiveness of an anti-reverberation filter, based on an estimated ratio of direct to indirect (reverberated) sound energy expected to reach the microphone.

Abstract: Described herein are systems, methods, and apparatus for determining audio context between an audio source and an audio sink and selecting signal profiles based at least in part on that audio context. The signal profiles may include noise cancellation which is configured to facilitate operation within the audio context. Audio context may include user-to-user and user-to-device communications.

Abstract: Exemplary embodiments provide systems, devices and methods that allow creation and management of lists of items in an integrated manner on an interactive graphical user interface. A user may speak a plurality of list items in a natural unbroken manner to provide an audio input stream into an audio input device. Exemplary embodiments may automatically process the audio input stream to convert the stream into a text output, and may process the text output into one or more n-grams that may be used as list items to populate a list on a user interface.

Abstract: A method for encoding and decoding a digital audio signal is provided, said method comprising the steps of: encoding a first sequence of samples of the digital signal according to a transform encoding; encoding a second sequence of samples of the digital signal according to a predictive encoding; wherein the second sequence starts before the end of the first sequence, a subsequence common to the first and second sequences being thus encoded both by predictive encoding and by transform encoding.

Abstract: The present invention relates to a method of evaluating intelligibility of a degraded speech signal received from an audio transmission system conveying a reference signal. The method comprises sampling said reference and degraded signal into frames, and forming frame pairs. For each pair one or more difference functions representing a difference between the degraded and reference signal are provided. A difference function is selected and compensated for different disturbance types, such as to provide a disturbance density function adapted to human auditory perception. An overall quality parameter is determined indicative of the intelligibility of the degraded signal. The method comprises determining a switching parameter indicative of audio power level of said degraded signal, for performing said selecting.

Abstract: Facts are extracted from speech and recorded in a document using codings. Each coding represents an extracted fact and includes a code and a datum. The code may represent a type of the extracted fact and the datum may represent a value of the extracted fact. The datum in a coding is rendered based on a specified feature of the coding. For example, the datum may be rendered as boldface text to indicate that the coding has been designated as an “allergy.” In this way, the specified feature of the coding (e.g., “allergy”-ness) is used to modify the manner in which the datum is rendered. A user inspects the rendering and provides, based on the rendering, an indication of whether the coding was accurately designated as having the specified feature. A record of the user's indication may be stored, such as within the coding itself.

Abstract: A method includes receiving an input signal comprising an original domain signal and creating a first window data set and a second window data set from the signal, wherein an initiation of the second window data set is offset from an initiation of the first window data set, converting the first window data set and the second window data set to a frequency domain and storing the resulting data as data in a second domain different from the original domain, performing complex spectral phase evolution (CSPE) on the second domain data to estimate component frequencies of the first and second window data sets, using the component frequencies estimated in the CSPE, sampling a set of second-domain high resolution windows to select a mathematical representation comprising a second-domain high resolution window that fits at least one of the amplitude, phase, amplitude modulation and frequency modulation of a component of an underlying signal wherein the component comprises at least one oscillator peak, generating an ou

Abstract: Systems and methods for updating electronic calendar information. Speech is received from a user at a vehicle telematics unit (VTU), wherein the speech is representative of information related to a particular vehicle trip. The received speech is recorded in the VTU as a voice memo, and data associated with the voice memo is communicated from the VTU to a computer running a calendaring application. The data is associated with a field of the calendaring application, and stored in association with the calendaring application field.

Abstract: Systems, methods, and devices are described for implementing learning algorithms on data sets. A data set may be partitioned into a plurality of data partitions that may be distributed to two or more processors, such as a graphics processing unit. The data partitions may be processed in parallel by each of the processors to determine local counts associated with the data partitions. The local counts may then be aggregated to form a global count that reflects the local counts for the data set. The partitioning may be performed by a data partition algorithm and the processing and the aggregating may be performed by a parallel collapsed Gibbs sampling (CGS) algorithm and/or a parallel collapsed variational Bayesian (CVB) algorithm. In addition, the CGS and/or the CVB algorithms may be associated with the data partition algorithm and may be parallelized to train a latent Dirichlet allocation model.

Abstract: A method for decoding an audio signal having a bandwidth that extends beyond a bandwidth of a CELP excitation signal in an audio decoder including a CELP-based decoder element. The method includes obtaining a second excitation signal having an audio bandwidth extending beyond the audio bandwidth of the CELP excitation signal, obtaining a set of signals by filtering the second excitation signal with a set of bandpass filters, scaling the set of signals using a set of energy-based parameters, and obtaining a composite output signal by combining the scaled set of signals with a signal based on the audio signal decoded by the CELP-based decoder element.

Abstract: A speech masking apparatus includes a microphone and a speaker. The microphone can detect a human voice. The speaker can output a masking language which can include phonemes resembling human speech. At least one component of the masking language can have a pitch, a volume, a theme, and/or a phonetic content substantially matching a pitch, a volume, a theme, and/or a phonetic content of the voice.

Abstract: An electronic device for determining a set of pitch cycle energy parameters is described. The electronic device includes a processor and executable instructions stored in memory. The electronic device obtains a frame, a set of filter coefficients and a residual signal based on the frame and the set of filter coefficients. The electronic device determines a set of peak locations based on the residual signal and segments the residual signal such that each segment includes one peak. The electronic device determines a first set of pitch cycle energy parameters based on a frame region between two consecutive peak locations and maps regions between peaks in the residual signal to regions between peaks in a synthesized excitation signal to produce a mapping. The electronic device determines a second set of pitch cycle energy parameters based on the first set of pitch cycle energy parameters and the mapping.

Abstract: The present invention relates to a method for encoding a voice signal, a method for decoding a voice signal, and an apparatus using the same. The method for encoding the voice signal according to the present invention, includes the steps of: determining an eco-zone in a present frame; allocating bits for the present frame on the basis of the location of the eco-zone; and encoding the present frame using the allocated bits, wherein the step of allocating the bits allocates more bits in the section in which the eco-zone is located than in the section in which the eco-zone is not located.

Abstract: A communication system according to the present invention includes a plurality of terminal devices that are able to communicate mutually. Each of the terminal devices includes a voice input conversion device, a voice transmitting device, a voice receiving device, and a voice reproducing device. When there is a plurality of voice signals which has not been completed reproduction, the voice reproducing device reproduces after arranging the voice signals so that respective voices corresponding to the respective voice signals do not overlap.

Abstract: An original loudness level of an audio signal is maintained for a mobile device while maintaining sound quality as good as possible and protecting the loudspeaker used in the mobile device. The loudness of an audio (e.g., speech) signal may be maximized while controlling the excursion of the diaphragm of the loudspeaker (in a mobile device) to stay within the allowed range. In an implementation, the peak excursion is predicted (e.g., estimated) using the input signal and an excursion transfer function. The signal may then be modified to limit the excursion and to maximize loudness.

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: A multi-channel synthesizer includes a post processor for determining post processed reconstruction parameters or quantities derived from the reconstruction parameter for an actual time portion of the input signal so that the post processed reconstruction parameter or the post processed quantity is different from the corresponding quantized and inversely quantized reconstruction parameter in that the value of the post processed reconstruction parameter or the derived quantity is not bound by the quantization step size. A multi-channel reconstructor uses the post-processed reconstruction parameter for reconstructing the multi-channel output signal.

Abstract: A method and apparatus are provided for visualizing the latency in a conversation between a local speaker and at least one remote speaker separated from the local speaker by a communication medium. A latency estimate is obtained. A timing indication of at least the end of a conversational turn by the local speaker is obtained, and an outbound graphic is displayed, indicating the progress of at least the end-of-turn across the communication medium toward the remote speaker. The outbound graphical indication is displayed with a transit time across the medium that is derived from the latency estimate. An inbound graphic is displayed, indicating the progress across the communication medium toward the local speaker, of a start of a conversational turn by the remote speaker, which is imputed to begin when the remote speaker receives the local speaker's end-of-turn. The inbound graphical indication is displayed with a transit time across the medium that is derived from the latency estimate.

Abstract: A method of communicating digitized speech from a transmitting forum participant comprises the step of receiving a data structure that includes said digitized speech. The data structure is analyzed to determine whether the digitized speech is redundantly represented in a plurality of forms in the data structure. A portion of the data structure is forwarded to a receiving forum participant, thereby communicating the digitized speech from the transmitting forum participant. In this method, when the digitized speech is redundantly represented in the data structure in a plurality of forms, the forwarding step includes a step of selecting one or more forms, based on a function, from the plurality of forms in the data structure. Furthermore, the portion of the data structure that is forwarded to the receiving forum participant includes data in the data structure that corresponds to each of the selected one or more forms.

Abstract: The present invention relates to a postfilter and a postfilter control to be associated with a postfilter for improving perceived quality of speech reconstructed at a speech decoder. The postfilter control comprises means for measuring stationarity of a speech signal reconstructed at a decoder, means for determining a coefficient to a postfilter control parameter based on the measured stationarity, and means for transmitting the determined coefficient to a postfilter, such that the postfilter can process the reconstructed speech signal by applying the determined coefficient to the postfilter control parameter to obtain an enhanced speech signal.

Abstract: In one configuration, erasure of a significant frame of a sustained voiced segment is detected. An adaptive codebook gain value for the erased frame is calculated based on the preceding frame. If the calculated value is less than (alternatively, not greater than) a threshold value, a higher adaptive codebook gain value is used for the erased frame. The higher value may be derived from the calculated value or selected from among one or more predefined values.

Abstract: An method and apparatus that modifies static media, such as music files being played to a user of the device, upon the generation or receipt of an alert, notification or message, so that information in the alert, notification or message can be incorporated into the media files then communicated to the user. In a further embodiment, a user's response to the communicated information can be sensed using one or more sensors and transducers so as to provide feedback to the device, and then optionally to a node in a system.

Abstract: Aspects of a method and system for an asynchronous pipeline architecture for multiple independent dual/stereo channel PCM processing are provided. Asynchronously pipeline processing of audio information comprised within a decoded PCM frame may be based on metadata information generated from the decoded PCM frame and an output decoding rate. The asynchronously pipeline processing may comprise mixing a primary audio information portion and a secondary audio information, portion, sample rate converting the audio information, and buffering the audio information. The asynchronously pipeline processing may comprise multiple pipeline stages. Feeding back an output of one of the pipeline stages to an input of a previous one of the pipeline stages may be enabled. The metadata information may comprise a frame start indicator associated with the decoded PCM frame and/or a plurality of mixing coefficients.

Abstract: An audio encoder implements multi-channel coding decision, band truncation, multi-channel rematrixing, and header reduction techniques to improve quality and coding efficiency. In the multi-channel coding decision technique, the audio encoder dynamically selects between joint and independent coding of a multi-channel audio signal via an open-loop decision based upon (a) energy separation between the coding channels, and (b) the disparity between excitation patterns of the separate input channels. In the band truncation technique, the audio encoder performs open-loop band truncation at a cut-off frequency based on a target perceptual quality measure. In multi-channel rematrixing technique, the audio encoder suppresses certain coefficients of a difference channel by scaling according to a scale factor, which is based on current average levels of perceptual quality, current rate control buffer fullness, coding mode, and the amount of channel separation in the source.

Abstract: An audio encoder has a common preprocessing stage, an information sink based encoding branch such as spectral domain encoding branch, a information source based encoding branch such as an LPC-domain encoding branch and a switch for switching between these branches at inputs into these branches or outputs of these branches controlled by a decision stage. An audio decoder has a spectral domain decoding branch, an LPC-domain decoding branch, one or more switches for switching between the branches and a common post-processing stage for post-processing a time-domain audio signal for obtaining a post-processed audio signal.

Abstract: Provided are a method and an apparatus for encoding and decoding an audio signal. A method for encoding an audio signal includes receiving a transformed audio signal, dividing the transformed audio signal into a plurality of subbands, performing a first sinusoidal pulse coding operation on the subbands, determining a performance region of a second sinusoidal pulse coding operation among the subbands on the basis of coding information of the first sinusoidal pulse coding operation, and performing the second sinusoidal pulse coding operation on the determined performance region, wherein the first sinusoidal pulse coding operation is performed variably according to the coding information. Accordingly, it is possible to further improve the quality of a synthesized signal by considering the sinusoidal pulse coding of a lower layer when encoding or decoding an audio signal in an upper layer by a layered sinusoidal pulse coding scheme.

Abstract: A speech-synthesizing device includes a hierarchical prosodic module, a prosody-analyzing device, and a prosody-synthesizing unit. The hierarchical prosodic module generates at least a first hierarchical prosodic model. The prosody-analyzing device receives a low-level linguistic feature, a high-level linguistic feature and a first prosodic feature, and generates at least a prosodic tag based on the low-level linguistic feature, the high-level linguistic feature, the first prosodic feature and the first hierarchical prosodic model. The prosody-synthesizing unit synthesizes a second prosodic feature based on the hierarchical prosodic module, the low-level linguistic feature and the prosodic tag.

Abstract: A data processing method for performing data processing on wireless received data and an associated data processing apparatus are provided, where the data processing method is applied to an electronic device. The data processing method includes the steps of: wirelessly receiving a plurality of packets corresponding to a same set of speech data from another electronic device; and selectively performing error correction operation on at least one of the plurality of packets to obtain the set of speech data, wherein whether to perform the error correction operation is determined according to at least one characteristic of the plurality of packets. More particularly, the error correction operation is selectively performed for at least one scenario of a timing critical scenario and a re-transmission limited scenario.

Abstract: A dialogue system and a method for the same are disclosed. The dialogue system includes a multimodal input unit receiving speech and non-speech information of a user, a domain reasoner, which stores a plurality of pre-stored situations, each of which is formed by a combination one or more speech and non-speech information, calculating each adaptability of the pre-stored situations on the basis of a generated situation based on the speech and the non-speech information received from the multimodal input unit, and determining a current domain according to the calculated adaptability, a dialogue manager to select a response corresponding to the current domain, and a multimodal output unit to output the response. The dialogue system performs domain reasoning using a situation including information combinations reflected in the domain reasoning process, current information, and a speech recognition result, and reduces the size of a dialogue search space while increasing domain reasoning accuracy.

Abstract: A mobile device is adapted for protecting private information on the mobile device in a hybrid automatic speech recognition arrangement. The mobile device includes a speech input component for receiving a speech input signal from a user. Additionally, the mobile device includes a local ASR arrangement for performing local ASR processing of the speech input signal and determining if private information is included within the speech input signal. A control unit on the mobile device obscures private information in the speech input signal if the local ASR arrangement identifies information within a speech recognition result as private information. The control unit releases the speech input signal with the obscured private information for transmission to a remote server for further ASR processing. Results from the remote server's ASR processing are integrated and combined with results from local ASR processing to display information on the mobile device.

Abstract: A method, device, system, and computer program product calculate a gradient index as a sum of magnitudes of gradients of speech signals from a received frame at each change of direction; and provide an indication that the frame contains babble noise if the gradient index, energy information, and background noise level exceed pre-determined thresholds or a voice activity detector algorithm and sound level indicate babble noise.

Abstract: A method of processing a signal, which includes receiving at least one of a first signal and a second signal, receiving mode information, and decoding the at least one of the first signal and the second signal using at least one of a first coding scheme and a second coding scheme according to the mode information. Further, the mode information is information for indicating that a prescribed mode corresponds to which one of at least three modes.

Abstract: This document discusses, among other things, apparatus and methods including an analog-to-digital controller (ADC) configured to receive an enable signal and to provide an ADC output signal to control logic, wherein the control logic is configured to provide a control voltage to a control input of a switch. In an example, the control voltage includes the ADC output signal when the ADC output signal is below a first threshold or above a second threshold. In certain examples, the control logic is configured to transition the control voltage from the first threshold to the second threshold when the ADC output signal is between the first and second thresholds.

Abstract: A method for encoding an audio signal including: processing a selected subset of a lower series of samples forming a lower frequency spectral band of the audio signal and a higher series of samples forming a higher frequency spectral band of the audio signal to parametrically encode the higher series of samples forming the higher frequency spectral band by identifying a sub-series of the lower series of samples.

Abstract: A method and a system for transmitting an audio signal are provided. The system includes a transmission device and a receiving device communicating with the transmission device via a network. The method includes receiving and sampling the audio signal, recording values of points of the sampled audio signal using the transmission device, segmenting the sampled audio signal into a plurality of frames, extracting and encoding characteristic information from each frame, to obtain a group of generated codes, transmitting each group of generated codes to a receiving device sequentially using the transmission device, and decoding each group of generated codes using the receiving device, to obtain a decoded audio signal.

Abstract: Various embodiments provide an interactive, shared, story-reading experience in which stories can be experienced from remote locations. Various embodiments enable augmentation or modification of audio and/or video associated with the story-reading experience. This can include augmentation and modification of a reader's voice, face, and/or other content associated with the story as the story is read.

Abstract: An encoding method includes: extracting core layer characteristic parameters and enhancement layer characteristic parameters of a background noise signal, encoding the core layer characteristic parameters and enhancement layer characteristic parameters to obtain a core layer codestream and an enhancement layer codestream. The disclosure also provides an encoding device, a decoding device and method, an encapsulating method, a reconstructing method, an encoding-decoding system and an encoding-decoding method. By describing the background noise signal with the enhancement layer characteristic parameters, the background noise signal can be processed by using more accurate encoding and decoding method, so as to improve the quality of encoding and decoding the background noise signal.

Abstract: Provided are a method for emotion communication to share a user's emotions between an emotion signal sensing device and an emotion service providing device. The method for emotion communication includes: the emotion signal sensing device's sensing biological and environmental information of the user and generating an emotion signal and emotion information of the user based on the biological and environmental information; establishing an emotion communication connection with the emotion service providing device; transmitting the emotion signal and the emotion information to the emotion service providing device by the emotion communication connection establishment; and breaking the connection with the emotion service providing device.

Abstract: A Yule-Walker based, low-complexity voice activity detector (VAD) is disclosed. An input signal is typically noisy speech (i.e., corrupted with, for example, babble noise). In one embodiment, a first initialization stage of the VAD computes an occurrence of a silent period within the input signal and the AR parameters. The VAD could accordingly compute a tentative adaptive threshold and output hypothesis H1 (which means speech is present) during this stage. During the second initialization stage, the VAD generally builds a database of associated values and computes the adaptive threshold accordingly. The second initialization stage could also output tentative VAD decisions based on the tentative threshold computed in the first initialization stage. Finally, the VAD periodically retrains or updates AR parameters, threshold values and/or the database and outputs VAD decisions accordingly.

Abstract: A home appliance and an operation method thereof are disclosed. The operation method of the home appliance includes entering a voice recognition mode, receiving a voice data through a microphone, recognizing the received voice date, and, in a case in which the recognized voice data contains information related to another home appliance, transmitting the recognized voice data to the corresponding home appliance. Consequently, sharing of voice data between home appliances is achieved.

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: A sound encoder for efficiently encoding stereophonic sound. A prediction parameter analyzer determines a delay difference D and an amplitude ratio g of a first-channel sound signal with respect to a second-channel sound signal as channel-to-channel prediction parameters from a first-channel decoded signal and a second-channel sound signal. A prediction parameter quantizer quantizes the prediction parameters, and a signal predictor predicts a second-channel signal using the first decoded signal and the quantization prediction parameters. The prediction parameter quantizer encodes and quantizes the prediction parameters (the delay difference D and the amplitude ratio g) using a relationship (correlation) between the delay difference D and the amplitude ratio g attributed to a spatial characteristic (e.g., distance) from a sound source of the signal to a receiving point.

Abstract: Provided is a frequency band extension apparatus and method, an encoding apparatus and method, a decoding apparatus and method, and a program. Band-pass filters obtain a plurality of lowband subband signals from an input signal. A frequency envelope extracting circuit extracts a frequency envelope from the plurality of lowband subband signals obtained by the plurality of band-pass filters. A highband signal generating circuit generates highband signal components on the basis of the frequency envelope obtained by the frequency envelope extracting circuit, and the plurality of subband signals obtained by the band-pass filters. A frequency band extension apparatus extends the frequency band of the input signal on the basis of the highband signal components generated by the highband signal generating circuit.

Abstract: A method and apparatus for generating synthesis audio signals are provided. The method includes decoding a bitstream; splitting the decoded bitstream into n sub-band signals; generating n transformed sub-band signals by transforming the n sub-band signals in a frequency domain; and generating synthesis audio signals by respectively multiplying the n transformed sub-band signals by values corresponding to synthesis filter bank coefficients.

Abstract: Systems, methods, and computer-readable storage media for generating personalized tag recommendations using speech analytics. The system first analyzes an audio stream to identify topics in the audio stream. Next, the system identifies tags related to the topics to yield identified tags. Based on the identified tags, the system then generates a tag recommendation for tagging the audio stream. The system can also send the tag recommendation to a device associated with a user for presentation to the user.

Abstract: Embodiments of the present invention provide a method, device, and system for classifying voice conference minutes. The method is: performing voice source locating according to audio data of the conference site so as to acquire a location of a voice source corresponding to the audio data, writing the location of the voice source into additional field information of the audio data, writing a voice activation flag into the additional field information, packaging the audio data as an audio code stream, and sending the audio code stream and the additional field information of the audio code stream to a recording server, so that the recording server classifies the audio data according to the additional field information and writes a participant identity that corresponds to the location of the voice source corresponding to the audio data into the additional field information of the audio code stream.

Abstract: The present disclosure discloses a method of using a mobile terminal as a microphone, an audio playback method, and related device and system. The method of using a mobile terminal as a microphone comprises: receiving identification information from a media device; establishing a data connection with the media device based on the identification information; converting a voice signal into audio data and sending the audio data to the media device, enabling the media device to output the audio data. According to the present disclosure, a mobile device and a media device can coordinate with each other. By connecting a mobile device, such as a mobile phone etc., with a media device, the mobile device can be used as a microphone. This makes it convenient for a user to use a microphone whenever and wherever, and meets the user's need.

Abstract: An audio recording and reproducing apparatus includes a controller for controlling the entire behaviors, hard disc for write and read of audio data, audio compression/expansion circuit for expanding compressed audio data, and external I/O port. The audio recording and reproducing apparatus is connected to a network service center to obtain desired music data from storage of the network service center and to store it in the hard disc.

Abstract: A wireless comprises at least one subscriber unit in wireless communication with an infrastructure. Each subscriber unit implements a speech recognition client, and the infrastructure comprises a speech recognition server. A given subscriber unit takes as input an unencoded speech signal that is subsequently parameterized by the speech recognition client. The parameterized speech is then provided to the speech recognition server that, in turn, performs speech recognition analysis on the parameterized speech. Information signals, based in part upon any recognized utterances identified by the speech recognition analysis, are subsequently provided to the subscriber unit. The information signals may be used to control the subscriber unit itself; to control one or more devices coupled to the subscriber unit, or may be operated upon by the subscriber unit or devices coupled thereto.

Abstract: A wireless system comprises at least one subscriber unit in wireless communication with an infrastructure. Each subscriber unit implements a speech recognition client, and the infrastructure comprises a speech recognition server. A given subscriber unit takes as input an unencoded speech signal that is subsequently parameterized by the speech recognition client. The parameterized speech is then provided to the speech recognition server that, in turn, performs speech recognition analysis on the parameterized speech. Information signals, based in part upon any recognized utterances identified by the speech recognition analysis, are subsequently provided to the subscriber unit. The information signals may be used to control the subscriber unit itself; to control one or more devices coupled to the subscriber unit, or may be operated upon by the subscriber unit or devices coupled thereto.