Abstract: In an embodiment, bitstream elements of sub-frames are encoded differentially to a global gain value so that a change of the global gain value results in an adjustment of an output level of the decoded representation of the audio content. Concurrently, the differential coding saves bits. Even further, the differential coding enables the lowering of the burden of globally adjusting the gain of an encoded bitstream. In another embodiment, a global gain control across CELP coded frames and transform coded frames is achieved by co-controlling the gain of the codebook excitation of the CELP codec, along with a level of the transform or inverse transform of the transform coded frames. In another embodiment, the gain value determination in CELP coding is performed in the weighted domain of the excitation signal.

Abstract: An adaptive time/frequency-based encoding mode determination apparatus including a time domain feature extraction unit to generate a time domain feature by analysis of a time domain signal of an input audio signal, a frequency domain feature extraction unit to generate a frequency domain feature corresponding to each frequency band generated by division of a frequency domain corresponding to a frame of the input audio signal into a plurality of frequency domains, by analysis of a frequency domain signal of the input audio signal, and a mode determination unit to determine any one of a time-based encoding mode and a frequency-based encoding mode, with respect to the each frequency band, by use of the time domain feature and the frequency domain feature.

Abstract: Provided are systems, methods and techniques for processing frame-based data. A frame of data, an indication that a transient occurs within the frame, and a location of the transient within the frame are obtained. Based on the indication of the transient, a block size is set for the frame, thereby effectively defining a plurality of equal-sized blocks within the frame. In addition, different window functions are selected for different ones of the plurality of equal-sized blocks based on the location of the transient, and the frame of data is processed by applying the selected window functions.

Abstract: A speech signal processing device is equipped with a power acquisition unit, a probability distribution acquisition unit, and a correspondence degree determination unit. The power acquisition unit accepts an inputted speech signal and, based on the accepted speech signal, acquires power representing the intensity of a speech sound represented by the speech signal. The probability distribution acquisition unit acquires a probability distribution using the intensity of the power acquired by the power acquisition unit as a random variable. The correspondence degree determination unit determines whether a correspondence degree representing a degree that power acquired by the power acquisition unit in a case that a predetermined reference speech signal is inputted into the power acquisition unit corresponds with predetermined reference power is higher than a predetermined reference correspondence degree, based on the probability distribution acquired by the probability distribution acquisition unit.

Abstract: A vocal effect processing system may include an effect modification module configured to selectively and dynamically apply effects to an input audio signal in accordance with a degree of likelihood that the input audio signal includes a vocal signal and/or based on a proximate location of a source of vocal audio with respect to a vocal microphone. Determination of the degree of likelihood that the input audio signal includes a vocal signal and/or the proximate location may be based on processing of the input audio signal or a plurality of input audio signals. Determination of the proximate location may alternatively, or in addition, be estimated based on a proximity sensor. The effect modification module may dynamically and selectively adjust the effects in response to changes in the degree of likelihood that the vocal signal is included in the input audio signal and/or changes in the estimated proximate location.

Abstract: A vocal effect processing system may include an effect modification module configured to selectively and dynamically apply effects to an input audio signal in accordance with a degree of likelihood that the input audio signal includes a vocal signal and/or based on a proximate location of a source of vocal audio with respect to a vocal microphone. Determination of the degree of likelihood that the input audio signal includes a vocal signal and/or the proximate location may be based on processing of the input audio signal or a plurality of input audio signals. Determination of the proximate location may alternatively, or in addition, be estimated based on a proximity sensor. The effect modification module may dynamically and selectively adjust the effects in response to changes in the degree of likelihood that the vocal signal is included in the input audio signal and/or changes in the estimated proximate location.

Abstract: The invention concerns a method and apparatus for performing packet loss or Frame Erasure Concealment (FEC) for a speech coder that does not have a built-in or standard FEC process. A receiver with a decoder receives encoded frames of compressed speech information transmitted from an encoder. A lost frame detector at the receiver determines if an encoded frame has been lost or corrupted in transmission, or erased. If the encoded frame is not erased, the encoded frame is decoded by a decoder and a temporary memory is updated with the decoder's output. A predetermined delay period is applied and the audio frame is then output. If the lost frame detector determines that the encoded frame is erased, a FEC module applies a frame concealment process to the signal. The FEC processing produces natural sounding synthetic speech for the erased frames.

Abstract: A method and apparatus for estimating speech intelligibility in a mobile communications network component handling two-way communication between two ends of a signal path. Test signals adapted for speech intelligibility measurements are inserted into the signal path to simulate two-way communication. Double-talk is detected during the communication, and speech intelligibility measurements are performed only during periods of double-talk. This enables the effect of echo to be taken into account while avoiding undesirable effects from non-linear processing, and comfort noise if present, in the signal path. Voice enhancement devices may then be adjusted in response to the estimated speech intelligibility.

Abstract: The invention provides a compensation method for audio frame loss in a MDCT domain, the method comprising: when a frame currently lost is a Pth frame, obtaining a set of frequencies to be predicted, and for each frequency in the set, using phases and amplitudes of a plurality of frames before a (P?1)th frame in a MDCT-MDST domain to predict a phase and an amplitude of the Pth frame, and using the predicted phase and amplitude to obtain a MDCT coefficient of the Pth frame at each corresponding frequency; for a frequency outside the set, using MDCT coefficients of a plurality of frames before the Pth frame to calculate a MDCT coefficient value of the Pth frame at the frequency; performing an IMDCT for the MDCT coefficients of the Pth frame to obtain a time domain signal of the Pth frame.

Abstract: A speech synthesizing apparatus includes a selector configured to select a plurality of speech units for synthesizing a speech of a phoneme sequence by referring to speech unit information stored in an information memory. Speech unit waveforms corresponding to the speech units are acquired from a plurality of speech unit waveforms stored in a waveform memory, and the speech is synthesized by utilizing the speech unit waveforms acquired. When acquiring the speech unit waveforms, at least two speech unit waveforms from a continuous region of the waveform memory are copied onto a buffer by one access, wherein a data quantity of the at least two speech unit waveforms is less than or equal to a size of the buffer.

Abstract: The present invention relates to methods and devices for forward time-domain aliasing cancellation in a coded signal transmitted from a coder to a decoder. Information related to correction of the time-domain aliasing in the coded signal is calculated at the coder and added in a bitstream sent from the coder to the decoder. The decoder receives the bitstream and cancels the time-domain aliasing in the coded signal in response to the information comprised in the bitstream. The information may be representative of a difference between a frame of audio signal to be encoded in a first coding mode and a decoded signal from the frame including time-domain aliasing effects.

Abstract: A method and system for monitoring and analyzing at least one signal are disclosed. An abstract of at least one reference signal is generated and stored in a reference database. An abstract of a query signal to be analyzed is then generated so that the abstract of the query signal can be compared to the abstracts stored in the reference database for a match. The method and system may optionally be used to record information about the query signals, the number of matches recorded, and other useful information about the query signals. Moreover, the method by which abstracts are generated can be programmable based upon selectable criteria. The system can also be programmed with error control software so as to avoid the re-occurrence of a query signal that matches more than one signal stored in the reference database.

Abstract: A method, device, system, and computer program product expand narrowband speech signals to wideband speech signals. The method includes determining signal type information from a signal, obtaining characteristics for forming an upper band signal using the determined signal type information, determining signal noise information, using the determined signal noise information to modify the obtained characteristics for forming the upper band signal, and forming the upper band signal using the modified characteristics.

Abstract: A parameter decoding apparatus includes a prediction residue decoder that finds a quantized prediction residue based on encoded information included in a current frame subject to decoding and a moving-average predictor produces a predicted parameter by multiplying a predictive coefficient with a past quantized prediction residue. An adder decodes a parameter by adding the quantized prediction residue and the predicted parameter, wherein the prediction residue decoder, when the current frame is erased, finds a current-frame quantized prediction residue from a weighted linear sum of a parameter decoded in the past and a future-frame quantized prediction residue.

Abstract: A method and system for analysis-by-synthesis encoding of an information signal is provide. The encoder (400) can include the steps of generating a first synthetic signal based on a first pitch-related codebook (402), generating a second synthetic signal based on a second pitch-related codebook (404), selecting a codebook configuration parameter based on the reference signal and the first and second synthetic signals, and conveying the codebook configuration for use in reconstructing an estimate of the input signal. The encoder can include an error expression having an error bias (506) and a prediction gain having a prediction gain bias (508) for determining the codebook configuration. The encoder can employ variable length coding and combinatorial subframe coding (600) for efficiently compressing the codebook configuration parameter and codebook related parameters for one or more subframes.

Abstract: The embodiments of the present invention improves conventional attenuation schemes by replacing constant attenuation with an adaptive attenuation scheme that allows more aggressive attenuation, without introducing audible change of signal frequency characteristics.

Abstract: A method of encoding samples in a digital signal is provided that includes receiving a frame of N samples of the digital signal, determining L possible distinct data values in the N samples, determining a reference data value in the L possible distinct data values and a coding order of L?1 remaining possible distinct data values, wherein each of the L?1 remaining possible distinct data values is mapped to a position in the coding order, decomposing the N samples into L?1 coding vectors based on the coding order, wherein each coding vector identifies the locations of one of the L?1 remaining possible distinct data values in the N samples, and encoding the L?1 coding vectors.

Abstract: Methods and apparatus to provide speech privacy are disclosed. An example method includes forming a sampling block based on a first received audio sample, the sampling block representing speech of a user, creating, with a processor, a mask based on the sampling block, the mask to reduce the intelligibility of the speech of the user, wherein the mask is created by converting the sampling block from a time domain to a frequency domain to form a frequency domain sampling block, identifying a first peak within the frequency domain sampling block, demodulating the frequency domain sampling block at the first peak to form a first envelope of the sampling block, distorting the first envelope to form a first distorted envelope, and emitting an acoustic representation of the mask via a speaker.

Abstract: A speech coding method of significantly reducing error propagation due to voice packet loss, while still greatly profiting from a pitch prediction or Long-Term Prediction (LTP), is achieved by limiting or reducing a pitch gain only for the first subframe or the first two subframes within a speech frame. The method is used for a voiced speech class; a pitch cycle length is compared to a subframe size to decide to reduce the pitch gain for the first subframe or the first two subframes within the frame. Speech coding quality loss due to the pitch gain reduction is compensated by increasing a bit rate of a second excitation component or adding one more stage of excitation component only for the first subframe or the first two subframes within the speech frame.

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 system and method of updating automatic speech recognition parameters on a mobile device are disclosed. The method comprises storing user account-specific adaptation data associated with ASR on a computing device associated with a wireless network, generating new ASR adaptation parameters based on transmitted information from the mobile device when a communication channel between the computing device and the mobile device becomes available and transmitting the new ASR adaptation data to the mobile device when a communication channel between the computing device and the mobile device becomes available. The new ASR adaptation data on the mobile device more accurately recognizes user utterances.

Abstract: A parameter transformer generates level parameters, indicating an energy relation between a first and a second audio channel of a multi-channel audio signal associated to a multi-channel loudspeaker configuration. The level parameter are generated based on object parameters for a plurality of audio objects associated to a down-mix channel, which is generated using object audio signals associated to the audio objects. The object parameters have an energy parameter indicating an energy of the object audio signal. To derive the coherence and the level parameters, a parameter generator is used, which combines the energy parameter and object rendering parameters, which depend on a desired rendering configuration.

Abstract: In an embodiment, a system maintains a database of a plurality of persons. The database includes an audio clip of a pronunciation of a name of a first person in the database. The system determines from a calendar database that a second person has an event in common with the first person, and transmits to a device associated with the second person an indication that the database includes the pronunciation of the name of the first person.

Abstract: Microphone arrays (MAs) are described that position and vent microphones so that performance of a noise suppression system coupled to the microphone array is enhanced. The MA includes at least two physical microphones to receive acoustic signals. The physical microphones make use of a common rear vent (actual or virtual) that samples a common pressure source. The MA includes a physical directional microphone configuration and a virtual directional microphone configuration. By making the input to the rear vents of the microphones (actual or virtual) as similar as possible, the real-world filter to be modeled becomes much simpler to model using an adaptive filter.

Abstract: A computer-implemented system and method for enhancing audio to individuals participating in a conversation is provided. Audio data for individuals participating in one or more conversations is analyzed. Possible conversational configurations of the individuals are generated based on the audio data, and each possible conversational configuration includes one or more subconfigurations of at least two of the individuals. A probability weight is assigned to each of the subconfigurations and includes a likelihood that the individuals of that subconfiguration are participating in one of the conversations. A probability of each possible conversational configuration is determined by combining the probability weights for the subconfigurations of that possible conversational configuration. The possible conversational configuration with the highest probability is selected as a most probable configuration. The individuals participating in the conversations are determined based on the most probable configuration.

Abstract: Example methods, apparatus and articles of manufacture to perform audio watermark decoding are disclosed. A disclosed example method includes receiving an audio signal including an audience measurement code embedded therein using a first plurality of frequency components, sampling the audio signal, transforming the sampled audio signal into a first frequency domain representation, determining whether the code is detectable in the first plurality of frequency components of the first frequency domain representation, and when the code is not detected in the first plurality of frequency components, examining a second plurality of frequency components of a second frequency domain representation to determine whether the code is detected, the second plurality of frequency components being offset from the first plurality of frequency components by a first offset, the first offset corresponding to a sampling frequency mismatch.

Abstract: A system may time-shift the distribution high-definition (HD) audio. The system can obtain an audio stream from a specified audio source, transcode the audio stream into an HD audio stream, and store the HD audio stream in a memory. The system may later forward the stored HD audio stream to a destination device, which can be a communication device linked to the system through a local telephone network or a remote communication device. The system can also store HD audio when a local communication device receives an incoming call request that interrupts a current HD audio distribution process. The system may resume distribution of the HD audio after processing the incoming call request from a point when the distribution was interrupted.

Abstract: Systems, methods, apparatuses, and computer programs for transfer of recorded digital voice memos to a computing system and processing of the transferred digital voice memos by the computing system or another computing system are disclosed. A recording device is configured to record a voice memo from a user and store the voice memo. The recording device is also configured to transfer the recorded voice memo to a computing system. The computing system is configured to translate the transferred voice memo into a computer-readable format and parse the translated voice memo. The computing system is also configured to determine a type of software application to which the voice memo pertains via a preamble, a keyword, or a keyphrase in the translated voice memo. The computing system is further configured to create an item in the determined software application based on the translated voice memo.

Abstract: When encoding an audio signal, it is possible to efficiently encode the audio signal while maintaining high register signal components, and prevent deterioration of sound quality of decoded signal. A digital audio signal is divided into a plurality of frequency bands. The digital audio signal having been divided into each band is function-approximated for each divided band. Further, parameters of function having been function-approximated are encoded. When performing decoding process, parameters of the function of each band are used to perform function interpolation, synthesize the function-interpolated signal of each band interpolated, and decode the signal. Thus, when function-approximating each band, by suitably setting the function equation, it is possible to perform an encoding process while maintaining the high register components and perform a compression-coding process which enables reproduction with very good sound quality.

Abstract: Provided are an encoding apparatus and a decoding apparatus of a multi-channel signal. The encoding apparatus of the multi-channel signal may process a phase parameter associated with phase information between a plurality of channels constituting the multi-channel signal, based on a characteristic of the multi-channel signal. The encoding apparatus may generate an encoded bitstream with respect to the multi-channel signal using the processed phase parameter and a mono signal extracted from the multi-channel signal.

Abstract: A system for leaving and transmitting speech messages automatically analyzes input speech of at least a reminder, fetches a plurality of tag informations, and transmits speech message to at least a message receiver, according to the transmit criterions of the reminder. A command or message parser parses the tag informations at least including at least a reminder ID, at least a transmitted command and at least a speech message. The tag informations are sent to a message composer for being synthesized into a transmitted message. A transmitting controller controls a device switch according to the reminder ID and the transmitted command, to allow the transmitted message send to the message receiver via a transmitting device.

Abstract: In a computer system that permits multiple audio capture applications to get an audio capture feed concurrently, an audio manager manages audio capture and/or audio playback in reaction to trigger events. For example, a trigger event indicates an application has started, stopped or otherwise changed a communication stream, or indicates an application has gained, lost or otherwise changed focus or visibility in a user interface, or indicates a user change. In response to a trigger event, the audio manager applies a set of rules to determine which audio capture application is allowed to get an audio capture feed. Based on the decisions, the audio manager manages the audio capture feed for the applications. The audio manager also sends a notification to each of the audio capture applications that has registered for notifications, so as to indicate whether the application is allowed to get the audio capture feed.

Abstract: A method is provided for decoding data streams in a voice communication system. The method includes: receiving two or more data streams having voice data encoded therein; decoding each data stream into a set of speech coding parameters; forming a set of combined speech coding parameters by combining the sets of decoded speech coding parameters, where speech coding parameters of a given type are combined with speech coding parameters of the same type; and inputting the set of combined speech coding parameters into a speech synthesizer.

Abstract: The invention relates to a method, computer, computer program and computer program product for speech quality estimation. The method comprises the steps of: determining a coding distortion parameter (QCOD), a bandwidth related distortion parameter (BW) and a presentation level distortion parameter (PL) of a speech signal; extracting a first coefficient (?l) and a second coefficient (?2), the first coefficient and the second coefficient being dependent on the coding distortion parameter; and calculating a signal quality measure (Q), where the signal quality measure is QCOD+?1BW+?2PL using the signal quality measure in a quality estimation of the speech signal.

Abstract: In a semiconductor device, a vocoder processing unit requests, after executing a first vocoder process being one of an encoding process and a decoding process and before executing a following second vocoder process being other one of the encoding process and the decoding process, a cache memory to prefetch first program data to be used for the second vocoder process from an external memory.

Abstract: Systems and methods for automatic user specific, condition specific communication system intelligibility testing and optimization are provided. The intelligibility of speech for a particular user is determined using a test of intelligibility administered by an interactive voice response (IVR) application running on a communication server. The intelligibility test can be run for a particular user under different conditions. For each user and/or set of conditions, a set of speech signal adjustment parameters can be determined. A set of speech signal adjustment parameters that will enhance the intelligibility of a speech signal for a user are applied when that user is involved in a communication session. The particular set of speech signal adjustment parameters selected can depend on the communication equipment and/or environment associated with the communication session.

Abstract: A voice activity detection (VAD) module analyzes a media file, such as an audio file or a video file, to determine whether one or more frames of the media file include speech. A speech recognizer generates feedback relating to an accuracy of the VAD determination. The VAD module leverages the feedback to improve subsequent VAD determinations. The VAD module also utilizes a look-ahead window associated with the media file to adjust estimated probabilities or VAD decisions for previously processed frames.

Abstract: The instant application includes computationally-implemented systems and methods that include managing adaptation data, wherein the adaptation data is correlated to at least one aspect of speech of a particular party, facilitating transmission of the adaptation data to a target device, wherein the adaptation data is configured to be applied to the target device to assist in execution of a speech-facilitated transaction, facilitating reception of adaptation result data that is based on at least one aspect of the speech-facilitated transaction between the particular party and the target device, determining whether to modify the adaptation data at least partly based on the adaptation result data, and facilitating transmission of at least a portion of modified adaptation data to a receiving device. In addition to the foregoing, other aspects are described in the claims, drawings, and text.

Abstract: The instant application includes computationally-implemented systems and methods that include managing adaptation data, the adaptation data is at least partly based on at least one speech interaction of a particular party, facilitating transmission of the adaptation data to a target device when there is an indication of a speech-facilitated transaction between the target device and the particular party, such that the adaptation data is to be applied to the target device to assist in execution of the speech-facilitated transaction, and facilitating acquisition of adaptation result data that is based on at least one aspect of the speech-facilitated transaction and to be used in determining whether to modify the adaptation data. In addition to the foregoing, other aspects are described in the claims, drawings, and text.

Abstract: Traditional audio encoders may conserve coding bit-rate by encoding fewer than all spectral coefficients, which can produce a blurry low-pass sound in the reconstruction. An audio encoder using wide-sense perceptual similarity improves the quality by encoding a perceptually similar version of the omitted spectral coefficients, represented as a scaled version of already coded spectrum. The omitted spectral coefficients are divided into a number of sub-bands. The sub-bands are encoded as two parameters: a scale factor, which may represent the energy in the band; and a shape parameter, which may represent a shape of the band. The shape parameter may be in the form of a motion vector pointing to a portion of the already coded spectrum, an index to a spectral shape in a fixed code-book, or a random noise vector. The encoding thus efficiently represents a scaled version of a similarly shaped portion of spectrum to be copied at decoding.

Abstract: A dialog server which provides dialogs made by at least one user through their respective avatars in a virtual space. A method and a computer readable article of manufacture tangibly embodying computer readable instructions for executing the steps of the method are also provided. The dialog server includes: a position storage unit which stores positional information on the avatars; an utterance receiver which receives at least one utterance of avatars and utterance strength representing an importance or attention level of the utterance; an interest level calculator which calculates interest levels between avatars based on their positional information; a message processor which generates a message based on the utterance in accordance with a value calculated from the interest levels and the utterance strength; and a message transmitter which transmits the message to the avatars.

Abstract: A method is provided for determining an indicator evaluating the voice quality of a coded speech signal. The method includes the following steps: calculation per signal frame, of a predetermined number of coefficients of a linear prediction filter for the coded speech signal; determination per frame, of a speech signal reconstructed on the basis of the filter coefficients thus calculated; obtaining per sample, of the residual between the coded speech signal and the reconstructed speech signal; calculation of an evaluation indicator on the basis of the mean or the absolute value of the residuals obtained for all the samples. Also provided are a device for determining an indicator implementing the above method, a method of evaluating the quality or of identifying the class of coding of the coded signal using the indicator determined, as well as a measurement terminal implementing these methods.

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: An encoding method and apparatus and a decoding method and apparatus are provided. The decoding method includes skipping extension information included in an input bitstream, extracting a three-dimensional (3D) down-mix signal and spatial information from the input bitstream, removing 3D effects from the 3D down-mix signal by performing a 3D rendering operation on the 3D down-mix signal, and generating a multi-channel signal using a down-mix signal obtained by the removal and the spatial information. Accordingly, it is possible to efficiently encode multi-channel signals with 3D effects and to adaptively restore and reproduce audio signals with optimum sound quality according to the characteristics of an audio reproduction environment.

Abstract: An audio decoder (100) comprising: effect means, decoding means, and rendering means. The effect means (500) generate modified down-mix audio signals from received down-mix audio signals. Said received down-mix audio signals comprise a down-mix of a plurality of audio objects. Said modified down-mix audio signals are obtained by applying effects to estimated audio signals corresponding to audio objects comprised in said received down-mix audio signals. Said estimated audio signals are derived from the received down-mix audio signals based on received parametric data. Said received parametric data comprise a plurality of object parameters for each of the plurality of audio objects. Said modified down-mix audio signals based on a type of the applied effect are decoded by decoding means or rendered by rendering means or combined with the output of rendering means.

Abstract: The present invention discloses an apparatus for automatic extraction of important events in audio signals comprising: signal input means for supplying audio signals; audio signal fragmenting means for partitioning audio signals supplied by the signal input means into audio fragments of a predetermined length and for allocating a sequence of one or more audio fragments to a respective audio window; feature extracting means for analyzing acoustic characteristics of the audio signals comprised in the audio fragments and for analyzing acoustic characteristics of the audio signals comprised in the audio windows; and important event extraction means for extracting important events in audio signals supplied by the audio signal fragmenting means based on predetermined important event classifying rules depending on acoustic characteristics of the audio signals comprised in the audio fragments and on acoustic characteristics of the audio signals comprised in the audio windows, wherein each important event extracted

Abstract: Described herein are systems, methods and apparatus for decoding in-band on-channel signals and extracting audio and data signals. Memory requirements are reduced by selectively filtering a bit stream of data in the signal so that services of interest which are encoded therein are processed. A single pool of memory may be shared between physical layer and data link layer processing. Memory in this pool may be allocated dynamically between processing of data at the physical and data link layers. When the available memory is not sufficient to support the required services, the dynamic allocation allows for graceful degradation.

Abstract: In one embodiment, a method includes monitoring activity in an environment, and storing a snippet of the monitored activity. Monitoring the activity in the environment includes operating a device arranged to capture the activity between approximately a first time and approximately a second time. The snippet has a particular duration that is arranged to end at approximately the second time. The method also includes storing the snippet in a storage module and determining when a request to provide the snippet is obtained from a party. If it is determined that the request to play the snippet is obtained, the method includes accessing the storage module to obtain the snippet and providing the snippet to the party if it is determined that the request to provide the snippet is obtained.

Abstract: A voice signal encoding and decoding method, device, and codec system are provided. The coding method includes: encoding an input voice signal to obtain a broadband code stream, where the broadband code stream includes a core layer bit stream and an extension enhancement layer bit stream (101); compressing the core layer bit stream to obtain a compressed code stream (102); and packing the compressed code stream and the extension enhancement layer bit stream to obtain a packed code stream (103). The core layer bit stream compressed, and the compressed code stream and the extension enhancement layer bit stream are packed, thereby reducing transmission bandwidth occupied by the input voice signal. Since the broadband voice encoding is performed on the input voice signal, a broadband voice code stream is transmitted by using narrowband transmission bandwidth, thereby improving the cost performance of voice signal transmission.

Abstract: A narration session between a plurality of participants can be set up to allow participants to collaboratively narrate an electronic book. Information can be transmitted to each participant so that the views of the participants remain in sync. Visual cues can also be transmitted to notify a participant of text that is to read aloud and audio snippets of read text are collected to form a narration file. Participants without access rights to the electronic book can be granted temporary rights.