Abstract: In a method for embedding steganographic information into the signal information of a signal encoder, a solution is to be created, which enables steganographic information being embedded into the signal information of a signal encoder such that a reduction of the voice quality is largely avoided.

Abstract: Implementations related to echo cancellation are depicted and described herein.

Abstract: There is provided a transition mode device and method for use in a predictive-type sound signal codec for producing a transition mode excitation replacing an adaptive codebook excitation in a transition frame and/or a frame following the transition in the sound signal, comprising an input for receiving a codebook index and a transition mode codebook for generating a set of codevectors independent from past excitation. The transition mode codebook is responsive to the index for generating, in the transition frame and/or frame following the transition, one of the codevectors of the set corresponding to the transition mode excitation. There is also provided an encoding device and method and a decoding device and method using the above described transition mode device and method.

Abstract: A method is provided for concealing a transmission error in a digital signal chopped into a plurality of successive frames associated with different time intervals in which, on reception, the signal may comprise erased frames and valid frames, the valid frames comprising information relating to the concealment of frame loss. The method is implemented during a hierarchical decoding using a core decoding and a transform-based decoding using windows introducing a time delay of less than a frame with respect to the core decoding. The method includes concealing a first set of missing samples for the erased frame, implemented in a first time interval; a step of concealing a second set of missing samples utilizing information of said valid frame and implemented in a second time interval; and a step of transition between the first and the second set of missing samples to obtain at least part of the missing frame.

Abstract: A method, device and system for signal encoding and decoding are disclosed. The method includes: encoding a core layer signal to obtain a core layer signal code; selecting an enhancement sample point that requires enhancement layer signal encoding according to the core layer signal code and the number of bits that can be used by an enhancement layer; obtaining an enhancement layer signal code of the enhancement sample point; and outputting a bit stream, where the bit stream includes the core layer signal code and the enhancement layer signal code. In embodiments of the present invention, according to the number of bits that can be used by the enhancement layer, the enhancement sample point that requires enhancement layer signal encoding is selected; the enhancement layer signal of the selected enhancement sample point is encoded and decoded; when no sufficient bits are available for the enhancement layer, the enhancement quality of the core layer can be improved.

Abstract: Improved indexing of telephony sessions is achieved by: (a) receiving, during the recording of the telephony session or during a playback of the recording, an indication including parameters which identify a discrete segment of the recording as being of interest; and (b) storing, in an index associated with the recording of the session, an identifier which identifies that discrete segment of the recording.

Abstract: Disclosed is a code conversion method to convert a first code sequence conforming to a first speech coding scheme into a second code sequence conforming to a second speech coding scheme. The method includes the following steps. The first step discriminates whether the first code sequence corresponds to a speech part or to a non-speech part, and generates a numerical value that indicates the discrimination result as a control flag. The second step converts the first code sequence into the second code sequence and outputs said second code sequence, when the value of the control flag corresponds to the speech part. The third step outputs the second code sequence that corresponds to the value of the control flag, when the value of the control flag corresponds to the non-speech part.

Abstract: Apparatus and methods are provided for measuring perceptual quality of a signal transmitted over a communication network, such as a circuit-switching network, packet-switching network, or a combination thereof. In accordance with one embodiment, a distributed apparatus is provided for measuring perceptual quality of a signal transmitted over a communication network. The distributed apparatus includes communication ports located at various locations in the network. The distributed apparatus may also include a signal processor including a processor for providing non-intrusive measurement of the perceptual quality of the signal. The distributed apparatus may further include recorders operatively connected to the communication ports and to the signal processor, wherein at least one of the recorders processes the signal at one of the communication ports and the recorder sends the signal to the signal processor to measure the perceptual quality of the signal.

Abstract: A hearing aid includes a microphone to convert sounds into electrical signals and a memory to store a plurality of voice prints and a plurality of sound-shaping instructions. Each of the plurality of sound-shaping instructions is associated with one of the plurality of voice prints. The hearing aid further includes a processor coupled to the microphone and the memory. The processor is configured to compare at least one sample from the electrical signals to the plurality of voice prints to identify a voice print. The processor selects sound-shaping instructions associated with the voice print and applies the sound-shaping instructions to selectively shape a portion of the electrical signals corresponding to the voice print to produce a shaped signal. The hearing aid further includes a speaker coupled to the processor and configured to reproduce the shaped signal as an audible output.

Abstract: A voice processing apparatus, which processes a first voice signal, includes: an acoustic analysis part which analyzes a feature quantity of an input second voice signal; a reference range calculation part which calculates a reference range based on the feature quantity; a comparing part which compares the feature quantity and the reference range and outputs a comparison result; and a voice processing part which processes and outputs the input first voice signal based on the comparison result.

Abstract: The invention pertains to a method and apparatus of efficient encoding and decoding of vector quantized data. The method and system explores and implements sub-division of a quantization vector space comprising class-leader vectors and representation of the class-leader vectors by a set of class-leader root-vectors facilitating faster encoding and decoding, and reduced storage requirements.

Abstract: Wideband speech signals must be converted to narrowband speech signals if the transmission medium or the destination terminal is constructed with narrowband constraints. A typical wideband-to-narrowband conversion method is the elimination of frequencies above 3400 Hz using a low pass filter and a down sampler. However, this method produces a muffled speech sound since the resulting narrowband signal has a flat frequency response. Methods and apparatus are presented herein to enhance the acoustic quality of a wideband-to-narrowband converted signal. A bandwidth switching filter is used to emphasize a mid-range frequency portion of the wideband signal so that the resulting narrowband signal has a non-flat frequency spectrum.

Abstract: In one embodiment, the present invention comprises a vocoder having at least one input and at least one output, an encoder comprising a filter having at least one input operably connected to the input of the vocoder and at least one output, a decoder comprising a synthesizer having at least one input operably connected to the at least one output of the encoder, and at least one output operably connected to the at least one output of the vocoder, wherein the decoder comprises a memory and the decoder is adapted to execute instructions stored in the memory comprising phase matching and time-warping a speech frame.

Abstract: A fixed code book (FCB) search device simplifies an error minimizing process and reduces a calculation amount so as to prevent deterioration of a coding performance. The FCB search device includes a pulse shape convolution inverse filter having an inverse feature of a pulse diffusion filter and supplied with an ideal residual signal; a pulse candidate preparatory selector that pre-selects a plurality of pulse candidates from the ideal residual signal to which the inverse filter is applied; and a pulse candidate final selector that finally selects one pulse from the selected candidates. Using this configuration, a search is made for an algebra code book to which the pulse diffusion is applied.

Abstract: Document data corresponding to each page included in a document is stored, and furthermore, feature data indicative of a feature of the document data and a document index indicating the document are associated with the document data. A document extracting apparatus obtains input document data, calculates feature data from the input document data, judges similarity between the input document data and the document data based on the feature data, obtains a document index associated with document data similar to the input document data, and extracts a plurality of pieces of document data associated with the document index. Thus, document data concerning the document including a page corresponding to the document data similar to the input document data is extracted for a plurality of pages.

Abstract: In a device configurable to encode speech performing an closed loop re-decision may comprise representing a speech signal by amplitude components and phase components for a current frame and a past frame. In a first closed loop stage, a first set of compressed components and a first set of uncompressed components for a current frame may be generated. A first set of features may be generated by comparing current and past frame amplitude and/or phase components. In a second closed loop stage, a second set of compressed components for the current frame may be generated by compressing the first set of compressed components and compressing the first set of uncompressed components. Generation of a second set of features may be based on the second set of compressed components from the current frame and a combination of amplitude and/or phase components from the past frame.

Abstract: Techniques for implementing vocoders in parallel digital signal processors are described. A preferred approach is implemented in conjunction with the BOPS® Manifold Array (ManArray™) processing architecture so that in an array of N parallel processing elements, N channels of voice communication are processed in parallel. Techniques for forcing vocoder processing of one data-frame to take the same number of cycles are described. Improved throughput and lower clock rates can be achieved.

Abstract: Differential dynamic content delivery including providing a session document for a presentation, wherein the session document includes a session grammar and a session structured document; selecting from the session structured document a classified structural element in dependence upon user classifications of a user participant in the presentation; presenting the selected structural element to the user; streaming presentation speech to the user including individual speech from at least one user participating in the presentation; converting the presentation speech to text; detecting whether the presentation speech contains simultaneous individual speech from two or more users; and displaying the text if the presentation speech contains simultaneous individual speech from two or more users.

Abstract: Encoding a sequence of digital speech samples into a bit stream includes dividing the digital speech samples into one or more frames and computing a set of model parameters for the frames. The set of model parameters includes at least a first parameter conveying pitch information. The voicing state of a frame is determined and the first parameter conveying pitch information is modified to designate the determined voicing state of the frame, if the determined voicing state of the frame is equal to one of a set of reserved voicing states. The model parameters are quantized to generate quantizer bits which are used to produce the bit stream.

Abstract: An encoding device reduces the encoding distortion as compared to the conventional technique and obtains a preferable sound quality for auditory sense. In the encoding device, a shape quantization unit quantizes the shape of an input spectrum with a small number of pulse positions and polarities. The shape quantization unit sets a pulse amplitude width to be searched later upon search of the pulse position to a value not greater than the pulse amplitude width which has been searched previously. A gain quantization unit calculates a gain of a pulse searched by the shape quantization unit for each of bands.

Abstract: A spectrum modifying method and the like wherein the efficiencies of the signal estimation and prediction can be improved and the spectrum can be more efficiently encoded. According to this method, the pitch period is calculated from an original signal, which serves as a reference signal, and then a basic pitch frequency (f0) is calculated. Thereafter, the spectrum of a target signal, which is a target of spectrum modification, is divided into a plurality of partitions. It is specified here that the width of each partition be the basic pitch frequency. Then, the spectra of bands are interleaved such that a plurality of peaks having similar amplitudes are unified into a group. The basic pitch frequency is used as an interleave pitch.

Abstract: Systems and programs for improving the efficiency of a sorting process in a computer system are disclosed. Data is provided in an input file external to the central processing unit of the computer system. In one embodiment, the implemented process involves investigating the contents of the input file in order to identify presorted portions thereof; incorporating the identified presorted portions of the input file into a second file external to the central processing unit, performing this step by rearranging directory information, without physically transferring the presorted portions from the input file. In sort processes involving both a string generation phase and a merge phase, the techniques described may be used in either or both phases, as well as in any output phase. Rearrange directory information rather than physically transferring data provides for greater efficiency in disk I/O.

Abstract: A communication device includes memory, an input interface, a processing module, and a transmitter. The processing module receives a digital signal from the input interface, wherein the digital signal includes a desired digital signal component and an undesired digital signal component. The processing module identifies one of a plurality of codebooks based on the undesired digital signal component. The processing module then identifies a codebook entry from the one of the plurality of codebooks based on the desired digital signal component to produce a selected codebook entry. The processing module then generates a coded signal based on the selected codebook entry, wherein the coded signal includes a substantially unattenuated representation of the desired digital signal component and an attenuated representation of the undesired digital signal component. The transmitter converts the coded signal into an outbound signal in accordance with a signaling protocol and transmits it.

Abstract: Encoding an audio signal is provided wherein the audio signal includes a first audio channel and a second audio channel, the encoding comprising subband filtering each of the first audio channel and the second audio channel in a complex modulated filterbank to provide a first plurality of subband signals for the first audio channel and a second plurality of subband signals for the second audio channel, downsampling each of the subband signals to provide a first plurality of downsampled subband signals and a second plurality of downsampled subband signals, further subband filtering at least one of the downsampled subband signals in a further filterbank in order to provide a plurality of sub-subband signals, deriving spatial parameters from the sub-subband signals and from those downsampled subband signals that are not further subband filtered, and deriving a single channel audio signal comprising derived subband signals derived from the first plurality of downsampled subband signals and the second plurality of

Abstract: Embodiments of the invention include apparatuses, systems, computer readable media, and methods for processing speech signals in a manner that enhances capacity, efficiency and hardware utilization of a communications network. A method, according to one embodiment, includes receiving speech signals, determining a subchannel power imbalance ratio of at least two subchannels, and selecting a receiver architecture for processing the speech signals in accordance with the determined subchannel power imbalance ratio.

Abstract: Communication apparatus having interfaces for exchanging data with first and second neighbors, a memory for storing codec information regarding the communication apparatus and a control entity operative to detect a message from the first neighbor, the first message being indicative of codec information regarding an originating entity. In response, the control entity assesses compatibility between the codec information regarding the originating entity and the codec information regarding the communication apparatus. If the assessment is positive, the control entity self-identifies the communication apparatus as a candidate for terminally supporting a subsequent codec-bypass negotiation with the originating entity. If the assessment is negative, the control entity self-identifies the communication apparatus as a candidate for non-terminally supporting such negotiation.

Abstract: A sound decoding device is capable of improving the lost frame compensation performance and improving quality of the decoded sound. A rise frame sound source compensation unit generates a compensation sound source signal when the current frame is a lost frame and a rise frame. An average sound source pattern update unit updates the average sound source pattern held in an average sound source pattern holding unit over a plurality of frames. When a frame is lost, an LPC synthesis unit performs LPC synthesis on a decoded sound source signal by using the compensation sound source signal inputted via a switching unit and a decoded LPC parameter from an LPC decoding unit and outputs the compensation decoded sound signal.

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

Abstract: Disclosed are systems, methods, and computer readable media for performing speech recognition. The method embodiment comprises selecting a codebook from a plurality of codebooks with a minimal acoustic distance to a received speech sample, the plurality of codebooks generated by a process of (a) computing a vocal tract length for a each of a plurality of speakers, (b) for each of the plurality of speakers, clustering speech vectors, and (c) creating a codebook for each speaker, the codebook containing entries for the respective speaker's vocal tract length, speech vectors, and an optional vector weight for each speech vector, (2) applying the respective vocal tract length associated with the selected codebook to normalize the received speech sample for use in speech recognition, and (3) recognizing the received speech sample based on the respective vocal tract length associated with the selected codebook.

Abstract: Described are the architecture of such a system, algorithms for time synchronization during a multiway conferencing session, methods to fight with network imperfections such as jitter to improve synchronization, methods of introducing buffering delays to create handicaps for players with faster connections, methods which help players with synchronization (such as a synchronized metronome during a music conferencing session), methods for synchronized recording and live delivery of synchronized data to the audience watching the distributed interaction live over the Internet.

Abstract: Packets of real-time information are sent with a source rate greater than zero kilobits per second, and a time or path or combined time/path diversity rate initially being zero kilobits per second. This results in a quality of service QoS, optionally measured at the sender or the receiver. When the QoS is on an unacceptable side of a threshold of acceptability, the sender sends diversity packets at an increased rate. Increasing the diversity rate while either reducing or maintaining the overall transmission rate is new. CELP-based multiple-description data partitioning sends the base or important information plus a subset of fixed excitation in one packet and sends the base or important information plus the complementary subset of fixed excitation in another packet. Reconstruction produces acceptable quality when only one of the two packets is received and better quality when both packets are received. Reconstruction provides for single and multiple lost packets.

Abstract: Prefixes are registered on a first list as index elements for respective registration patterns. Each prefix is selected as the longest of different-length prefixes that are extractable from a registration pattern in accordance with an extraction rule. Suffixes, which are the remaining parts of the registration patterns excluding the respective prefixes, are registered on a second list. Using different-length prefixes that are extracted from a retrieval key in accordance with the extraction rule, a prefix retriever searches the first list to retrieve a registration pattern whose prefix matches any of the prefixes of the retrieval key. A suffix checker carries out a check on the suffix of the registration pattern retrieved by the prefix retriever, among the suffixes on the second list, as to whether the suffix of the registration pattern matches the suffix of the retrieval key.

Abstract: Methods and devices provide improved perceived quality of an audio (or other) coded signal at a low bit-rate. An input signal may be split into an outlier portion and a stationary portion. The outlier portion of the input signal may be encoded. The stationary portion may be divided into subvectors. Each subvector may be classified as trivial or non-trivial. Each trivial subvector may be encoded using a pre-defined pattern. Each non-trivial subvector may be encoded with at least one location of at least one significant sample and a sign of the significant sample.

Abstract: A method and apparatus for a voice transcoder that converts a bitstream representing frames of data encoded according to a first voice compression standard to a bitstream representing frames of data according to a second voice compression standard using perceptual weighting that uses tuned weighting factors, such that the bitstream of a second voice compression standard to produce a higher quality decoded voice signal than a comparable tandem transcoding solution.

Abstract: One embodiment of the disclosures made herein is a method for facilitating mediated communication. In such an embodiment of the disclosures made herein, a voice-based communication request transmitted from a first communication device is received by a mediation system. The voice based communication request is requesting voice-based communication between the first communication device and a second communication device. In response to receiving the request for voice-based communication, a capability of the first communication device for communicating via a prescribed text messaging protocol is determined. Facilitating presentation of a text messaging follow-through action at the second communication device is performed in response to determining that the first communication device is capable of communicating via the prescribed text messaging protocol.

Abstract: A behavior control system of a robot for learning a phoneme sequence includes a sound inputting device inputting a phoneme sequence, a sound signal learning unit operable to convert the phoneme sequence into a sound synthesis parameter and to learn or evaluate a relationship between a sound synthesis parameter of a phoneme sequence that is generated by the robot and a sound synthesis parameter used for sound imitation, and a sound synthesizer operable to generate a phoneme sequence based on the sound synthesis parameter obtained by the sound signal learning unit.

Abstract: A device relating to information processing technologies and including an encoding and decoding method configured to solve the poor decoding quality problem. The method includes: encoding each sample of an input signal to generate an encoded signal of a core layer; comparing residuals of all or a part of the samples of the input signal with encoding thresholds, where the residuals are generated by core layer encoding, and performing encoding according to comparison results to generate an encoded signal of an enhancement layer; and writing the encoded signal of the core layer and the encoded signal of the enhancement layer into a bitstream to generate an encoded signal of the input signal.

Abstract: A system distinguishes a primary audio source and background noise to improve the quality of an audio signal. A speech signal from a microphone may be improved by identifying and dampening background noise to enhance speech. Stochastic models may be used to model speech and to model background noise. The models may determine which portions of the signal are speech and which portions are noise. The distinction may be used to improve the signal's quality, and for speaker identification or verification.

Abstract: Automatic level control of speech portions of an audio signal is provided. An audio signal is received in the form of a sequence of samples and may contain speech portion and non-speech portions. The sequence of samples is divided into a sequence of sub-frames. Multiple sub-frames adjacent to a present sub-frame are examined to determine a peak value of samples in the sub-frames. A gain factor is computed for the present sub-frame based on the peak value and a desired maximum value for said speech portion, and each sample in the present sub-frame is amplified by the gain factor. In an embodiment, variations in filtered energy values of multiple sub-frames enable determination of whether a sub-frame corresponds to a speech or non-speech/noise portion.

Abstract: Provided is a transmission apparatus for matching sound quality measurement sections of a variable bandwidth multi-codec. The apparatus includes a measurement section setting unit setting a measurement section, which is to be measured for sound quality, in units of time; a first conversion unit converting the measurement section into a measurement section in units of samples; and an information synthesis unit synthesizing information regarding the measurement section in units of samples with a digital original sound and outputting the synthesis result. In addition, provided is a method of matching a measurement section of a reference sound, based on which the end-to-end sound quality measurement of the variable bandwidth multi-codec is performed, and a measurement section of a sound produced by the variable bandwidth multi-codec in a real-time Internet multimedia service. Therefore, distortion of measurement results due to un-matching measurement sections can be reduced.

Abstract: A voice recording method is applied in a recording device that includes a voice receiving unit and a storage unit. The voice receiving unit receives voice signals. The storage unit stores voice models and personal information associated with each voice model. The recording method includes: recording voice signals received by the voice receiving unit and storing the recorded voice signals to the storage unit. Extracting speaker voice features from the recorded speaker's voice. Comparing the extracted features with the voice models to find a match. Obtaining the speaker personal information associated with the voice model when a match is found. Obtaining the storage path of the voice signals stored in the storage unit, then generating an index document according to the obtained voice model and the obtained storage path of the voice signals.

Abstract: Provided is an audio encoding device capable of improving performance of an adaptive codebook and improving quality of a decoded audio. In this audio encoding device, an adaptive codebook cuts out a vector specified by a comparator from adaptive code vectors stored in an internal buffer and outputs it to a filter and a switch. The filter performs a predetermined filtering process on the adaptive sound source signal and outputs the obtained adaptive code vector to the switch. According to an instruction from the comparator, the switch outputs the adaptive code vector directly output from the adaptive codebook to a adjuster when the adaptive codebook is searched and outputs the adaptive code vector output from the filter after being subjected to the filtering process to the gain adjuster when a fixed sound source is searched after the adaptive sound source search.

Abstract: Prefixes are registered on a first list as index elements for respective registration patterns. Each prefix is selected as the longest of different-length prefixes that are extractable from a registration pattern in accordance with an extraction rule. Suffixes, which are the remaining parts of the registration patterns excluding the respective prefixes, are registered on a second list. Using different-length prefixes that are extracted from a retrieval key in accordance with the extraction rule, a prefix retriever searches the first list to retrieve a registration pattern whose prefix matches any of the prefixes of the retrieval key. A suffix checker carries out a check on the suffix of the registration pattern retrieved by the prefix retriever, among the suffixes on the second list, as to whether the suffix of the registration pattern matches the suffix of the retrieval key.

Abstract: Methods and apparatus are presented for determining the type of acoustic signal and the type of frequency spectrum exhibited by the acoustic signal in order to selectively delete parameter information before vector quantization. The bits that would otherwise be allocated to the deleted parameters can then be re-allocated to the quantization of the remaining parameters, which results in an improvement of the perceptual quality of the synthesized acoustic signal. Alternatively, the bits that would have been allocated to the deleted parameters are dropped, resulting in an overall bit-rate reduction.

Abstract: In a device configurable to encode speech performing an open loop re-decision may comprise representing a speech signal by amplitude components and phase components for a current frame and a past frame. During the current frame, there may be an extraction of uncompressed amplitude components and uncompressed phase components. The amplitude components and the phase components from the past frame may then be retrieved. A set of features may be generated based on the uncompressed amplitude components from the current frame, the uncompressed phase components from the current frame, the amplitude components from the past frame, and the phase components from the past frame. The set of features may be checked as part of the open loop re-decision, and determining a final encoding decision based on the checking may be performed. The final encoding decision may be an encoding mode and/or encoding rate.

Abstract: A method and an apparatus for inserting an included message into an e-mail message, wherein the e-mail message is transferred through a unified messaging solution have been provided. In one embodiment, the unified messaging solution detects transmission of a voice mail message as the e-mail attachment. The voice mail message is received by a system that facilitates the transfer of the e-mail message. The system associates the included message with the voice mail message. The included message is inserted into the e-mail message. The system sends the e-mail message along with the included message and the attached voice mail message to an intended user. In a preferred embodiment, the included message is an advertising message.

Abstract: Encoding an audio signal is provided wherein the audio signal includes a first audio channel and a second audio channel, the encoding comprising subband filtering each of the first audio channel and the second audio channel in a complex modulated filterbank to provide a first plurality of subband signals for the first audio channel and a second plurality of subband signals for the second audio channel, downsampling each of the subband signals to provide a first plurality of downsampled subband signals and a second plurality of downsampled subband signals, further subband filtering at least one of the downsampled subband signals in a further filterbank in order to provide a plurality of sub-subband signals, deriving spatial parameters from the sub-subband signals and from those downsampled subband signals that are not further subband filtered, and deriving a single channel audio signal comprising derived subband signals derived from the first plurality of downsampled subband signals and the second plurality of

Abstract: A signal processing apparatus includes a decoding unit, an analyzing unit, a synthesizing unit, and a selecting unit. The decoding unit decodes an input encoded audio signal and outputs a playback audio signal. When loss of the encoded audio signal occurs, the analyzing unit analyzes the playback audio signal output before the loss occurs and generates a linear predictive residual signal. The synthesizing unit synthesizes a synthesized audio signal on the basis of the linear predictive residual signal. The selecting unit selects one of the synthesized audio signal and the playback audio signal and outputs the selected audio signal as a continuous output audio signal.

Abstract: The present invention relates to a transmitter and a receiver for speech coding and decoding by using an additional bit allocation method. The transmitter and the receiver according to the present invention realize a voice communication service of high quality by using additional bits permitted in system requirements while using a conventional speech coder as it is. In addition, the transmitter and the receiver according to the present invention have an advantage in that they enable insertion of additional quantization blocks while not changing the structure of the conventional standard speech coder, since they allocate additional bits by applying a multi-stage quantization procedure not in a speech signal domain but in a parameter domain.

Abstract: An apparatus and an associated method for facilitating selection of CODEC availability from amongst a set of CODECs at a communication device. A battery power measurer measures the stored energy level of a battery power supply that powers a communication device of which the CODEC forms a portion. A selector selects the available CODECs responsive to the measured power level. If the measured level is less than a threshold, then high-sampling-rate CODECs are at least selectably made unavailable for use. If the battery level is higher than the threshold, then the high-sampling-rate CODECs are made available for use. If the level is greater than a threshold then both a high sampling-rate and the low sampling-rate CODEC are available. An indication generator generates an indication of selection made by the selector.