Abstract: A wideband speech encoder according to one embodiment includes a lowband encoder and a highband encoder. The lowband encoder is configured to encode a lowband portion of a wideband speech signal as a set of filter parameters and an encoded excitation signal. The highband encoder is configured to calculate values for coding parameters that specify a spectral envelope and a temporal envelope of a highband portion of the wideband speech signal. The temporal envelope is based on a highband excitation signal that is derived from the encoded excitation signal. In one such example, the temporal envelope is based on a difference in levels between the highband portion and a synthesized highband signal, wherein the synthesized highband signal is generated according to the highband excitation signal and a set of highband filter parameters.

Abstract: A method, system and computer program product for assigning one or more conditions to a telephonic communication system to enable recording, replaying, and pausing of a telephone conversation. Recording of a telephone conversation may be manually or dynamically initiated during the telephone conversation. Dynamic replay of a telephone conversation is automatically initiated when a conversation experiences insufficient call quality, an interruption to the telephone service is detected, the user is distracted, a displacement of the user's phone is detected, or the user toggles between listening modes. A pause mode is automatically entered into during the telephone conversation when a section of the telephone conversation is replayed. An automated pause alert may be played during repeat of the telephone conversation to inform one or more parties that the user is temporarily unavailable. Real-time access to the telephone conversation is resumed when the replaying of the recorded telephone conversation has ended.

Abstract: A method to transmit a broadband multimedia resource locator using a narrowband communication system embeds the broadband multimedia resource locator into a narrowband audio stream and transmits the narrowband audio stream to one or more receiving communication devices over the narrowband communication system. The receiving communication device(s) subsequently extract the broadband multimedia resource locator from the narrowband audio stream and use the broadband multimedia resource locator to access a broadband communication system to retrieve multimedia content.

Abstract: A system capable of separating sound source signals with high precision while improving a convergence rate and convergence precision. A process of updating a current separation matrix Wk to a next separation matrix Wk+1 such that a next value J(Wk+1) of a cost function is closer to a minimum value J(W0) than a current value J(Wk) is iteratively performed. An update amount ?Wk of the separation matrix is increased as the current value J(Wk) of the cost function is increased and is decreased as a current gradient ?J(Wk)/?W of the cost function is rapid. On the basis of input signals x from a plurality of microphones Mi and an optimal separation matrix W0, it is possible to separate sound source signals y(=W0·x) with high precision while improving a convergence rate and convergence precision.

Abstract: A method for automatic transmission of status information from a first communications terminal set up for speech communication to a second communications terminal set up for text communication is provided. The speech communication between communications terminals is processed over a speech communications server and the text communication between communications terminals over a text communications server. The speech communications server and the text communications server exchange messages over at least one converter device. The status information will be transmitted from the first communications terminal over the speech communications server, the converter device, and the text communications server to the second communications terminal.

Abstract: A system and method for automatically adjusting floor controls for a conversation is provided. Audio streams are received, which each originate from an audio source. Floor controls for a current configuration including at least a portion of the audio streams are maintained. Conversational characteristics shared by two or more of the audio sources are determined. Possible configurations for the audio streams are identified based on the conversational characteristics. An analysis of the current configuration and the possible configurations is performed. A change threshold is applied to the analysis. When the analysis satisfies the change threshold, the floor controls are automatically adjusted. The audio streams are mixed into one or more outputs based on the adjusted floor controls.

Abstract: Methods and systems for handling speech recognition processing in effectively real-time, via the Internet, in order that users do not experience noticeable delays from the start until they receive responsive feedback. A user uses a client to access the Internet and a server supporting speech recognition processing. The user inputs speech to the client, which transmits the user speech to the server in approximate real-time. The server evaluates the user speech, and provides responsive feedback to the client, again, in approximate real-time, with minimum latency delays. The client upon receiving responsive feedback from the server, displays, or otherwise provides, the feedback to the user.

Abstract: An apparatus, a server, a method, and a tangible machine-readable medium thereof for processing and recognizing a sound signal are provided. The apparatus is configured to sense the sound signal of the environment and to dynamically derive and to transmit a feature signal and a sound feature message of the sound signal to the server. The server is configured to retrieve the stored sound models according to the sound feature message and to compare each of the sound models with the feature signal to determine whether the sound signal is abnormal after receiving the feature signal and the sound feature message.

Abstract: An apparatus and a method for voice communication of a mobile terminal are provided. More particularly, an apparatus and a method for clearly receiving a counterpart user's voice signal in a mobile terminal positioned at a place where a noise occurs are provided. The apparatus includes an input unit, an extension signal generator, and an adder. The input unit receives a voice signal. The extension signal generator generates, based on a voice signal received via the input unit, a harmonics signal corresponding to a frequency band that represents a reaction sensitive to a sense of hearing. The adder merges the generated harmonics signal with the received voice signal.

Abstract: A digital audio communication control apparatus includes a first mixing unit that mixes a voice input from a voice input unit and uttered by a specific speaker with a voice input from a digital audio packet receiving unit and uttered by at least one speaker except for the specific speaker, and a second mixing unit that mixes the voices mixed by the first mixing unit with the voice of the specific speaker. The voices mixed by the second mixing unit are fed back to the specific speaker.

Abstract: Apparatus and method to allow retrieval of voice messages deleted from the voice message memory of a voice messaging system. A voice messaging system such as a telephone answering device includes a deleted voice message memory for storing voice messages deleted from the voice message memory. The deleted voice messages stored in the deleted voice message memory are retrievable by the user for review subject to rules for permanent deletion of the deleted voice messages (e.g., after a period of time, when the deleted voice message memory approaches capacity, periodically, etc.).

Abstract: In one embodiment, at least one channel in a frame of the audio signal is subdivided into a plurality of blocks such that at least two of the blocks having different lengths. A length of the frame is a user defined value and is determined within a predetermined value. Furthermore, information indicating the subdivision of the channel into the blocks is generated.

Abstract: Methods and apparatus to extract data encoded in media content are disclosed. An example method includes receiving a media content signal, sampling the media content signal to generate digital samples, determining a frequency domain representation of the digital samples, determining a first rank of a first frequency in the frequency domain representation, determining a second rank of a second frequency in the frequency domain representation, combining the first rank and the second rank with a set of ranks to create a combined set of ranks, comparing the combined set of ranks to a set of reference sequences, determining a data represented by the combined set of ranks based on the comparison, and storing the data in a tangible memory.

Abstract: In a voice recording equipment and method, voice data from a speaker is received using a microphone. Threshold values T1 and T2 of surrounding environment of the voice recording equipment are determined. If an intensity of the voice data is less than or equal to the threshold value T2, the voice recording is stopped and the speaker is informed that the voice data is not useful. If the intensity of the voice data is greater than the threshold values, the voice data is stored into a storage unit.

Abstract: Disclosed is a voice decoding apparatus wherein the processor may be continuously employed for other applications for a prescribed time but, in response to an urgent interrupt, the processor can generate synthesised sound even when being used for other applications, without interruption. In this apparatus, a packet receiving section (101) receives packets of the layers of a plurality of frames and extracts code from the received packets. A state/code storage section (103) stores the code and decoding state of the code. A layer selection section (104) selects a layer number and a frame number corresponding to the code to be initially decoded, based on the decoding state. A decoding section (105) decodes the code of the selected frame number and layer number.

Abstract: A mobile replacement-dialogue recording system enables the creation of replacement-dialogue items by mobile users not at a media recording studio. Studio-users prepare guide media video, audio and text data which are made available to mobile users through a media server. A mobile user's mobile replacement-dialogue recording device obtains guide media and allows the user to view the guide media in rehearsal mode. The mobile replacement-dialogue recording device then records the mobile user's dialogue performance while presenting the mobile user with synchronized guide media. The mobile user can review, delete, and rerecord the resulting potential replacement dialogue, as well as create feedback media characterizing the replacement dialogue. Selected replacement dialogue items can be transmitted to the media server. A studio-module can then obtain the selected replacement dialogue items and feedback media from the media server so that they may be used in media-replacement.

Abstract: When performing audio communication by using different encoding/decoding methods, a code obtained by encoding audio by a certain method is converted into a code decodable by another method with a high audio quality and a small calculation amount. In a code conversion device for converting a first code string into a second code string, an audio decoding circuit acquires a first linear prediction coefficient and excitation signal information from the first code string and drives the filter having the first linear prediction coefficient by the excitation signal obtained from the excitation signal information, thereby creating a first audio signal. A fixed codebook code generation circuit uses the fixed codebook information and minimizes the distance between the second audio signal generated from the information obtained from the second code string and the first audio signal, thereby obtaining the fixed codebook information in the second code string.

Abstract: In a method of providing a backward and forward compatible speech codec payload format, the following steps are included: providing S10 an RTP package; including S20 payload according to a first codec into the provided RTP package, and appending S50 payload according to a second codec into the provided RTP package. In addition, at least one unused bit is located S30 in the included first codec payload, and the located at least one unused bit is designated S40 as a codec compatibility bit. Finally, the designated at least one codec compatibility bit is utilized S60 to provide an indication of the presence of the appended second codec payload.

Abstract: A recording and playback system is provided. The system includes an audio capturing device configured to receive an analog input and an encoder coupled to the audio capturing device configured to generate a digital signal based on the analog input. The system further includes a recognition engine coupled to the audio capturing device and configured to generate text data based on the analog input, wherein the encoder and the recognition engine simultaneously generate the digital signal and the text data such that the digital signal and the text data can be provided in a synchronized manner.

Abstract: The present invention is a system and method that improves upon voice activity detection by packetizing actual noise signals, typically background noise. In accordance with the present invention an access network receives an input voice signal (including noise) and converts the input voice signal into a packetized voice signal. The packetized voice signal is transmitted via a network to an egress network. The egress network receives the packetized voice signal, converts the packetized voice signal into an output voice signal, and outputs the output voice signal. The egress network also extracts and stores noise packets from the received packetized voice signal and converts the packetized noise signal into an output noise signal. When the access network ceases to receive the input voice signal while the call is still ongoing, the access network instructs the egress network to continually output the output noise signal.

Abstract: In one embodiment, a method, system and apparatus for recording audio is provided so that the recording can be authenticated. The system may be implemented as a central server that is accessed via one or more lines for audio communication, or as a stand-alone unit. The system operates by encrypting communicated data (e.g., audio signals), storing the encrypted information, and providing at least one user with a key that can be used to decrypt the stored information.

Abstract: A method for streaming audio data in a network, the audio data having a sequence of samples, includes encoding the sequence of samples into a plurality of coded base bitstreams, generating a plurality of enhancement streams, and transmitting the coded base bitstreams and the enhancement bitstreams to a receiver for decoding. Each of the enhancement bitstreams is generated from one of a plurality of non-overlapping portions of the sequence of samples.

Abstract: Speech signals to be sent between a first node and a second node via a wireless communication system are Adaptive Multi-Rate (AMR) encoded. A need to change the first node's first data transmission rate over a radio interface to a second different data transmission rate is determined. A new AMR source bit rate is then determined for both nodes. Information is sent to the second node, in advance of changing the data transmission rate over the radio interface, requesting the second node to change towards the new AMR source bit rate. After a predetermined time period sufficient for the second node to change from the current AMR source bit rate to the new AMR source bit rate expires or after the second node indicates a change to the new AMR source bit rate, the first node starts transmitting at the second data transmission rate over the radio interface.

Abstract: In one embodiment, at least one channel in a frame of the audio signal is subdivided into a plurality of blocks such that at least two of the blocks having different lengths, and information indicating the subdivision of the channel into the blocks is generated.

Abstract: The invention enables the inclusion of voice and remaining audio information at different parts of the audio production process. In particular, the invention embodies special techniques for VRA-capable digital mastering, accommodation of PCPV/PCA and/or SCRA signals in audio CODECs, VRA-capable encoders and decoders, and VRA in DVD and other digital audio file formats. The invention facilitates an end-listener's voice-to-remaining audio (VRA) adjustment upon the playback of digital audio media formats by focusing on new configurations of multiple parts of the entire digital audio system, thereby enabling a new technique intended to benefit audio end-users (end-listeners) who wish to control the ratio of the primary vocal/dialog content of an audio program relative to the remaining portion of the audio content in that program.

Abstract: The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.

Abstract: Disclosed is a method capable of adaptively aligning windows to extract features according to the types and characteristics of voice signals. To this end, window lengths based on the window update points in a corresponding order are determined by employing the concept of a higher order peak, and windows are aligned according to window lengths. When the windows are aligned according to such a manner, the start and end points of each window is known, so that it becomes possible to easily extract and analyze peak feature information.

Abstract: An audio file generation method and system. A computing system receives a first audio file comprising first speech data associated with a first party. The computing system receives a second audio file comprising second speech data associated with a second party. The first audio file differs from the second audio file. The computing system generates a third audio file from the second audio file. The third audio file differs from the second audio file. The process to generate the third audio file includes identifying a first set of attributes missing from the second audio file and adding the first set of attributes to the second audio file. The process to generate the third audio file additionally includes removing a second set of attributes from the second audio file. The third audio file includes third speech data associated with the second party. The computing system broadcasts the third audio file.

Abstract: Spatial information associated with an audio signal is encoded into a bitstream, which can be transmitted to a decoder or recorded to a storage media. The bitstream can include different syntax related to time, frequency and spatial domains. In some embodiments, the bitstream includes one or more data structures (e.g., frames) that contain ordered sets of slots for which parameters can be applied. The data structures can be fixed or variable. The data structure can include position information that can be used by a decoder to identify the correct slot for which a given parameter set is applied. The slot position information can be encoded with either a fixed number of bits or a variable number of bits based on the data structure type.

Abstract: In accordance with an embodiment, A method of encoding an audio bitstream at an encoder includes encoding an original low band signal at the encoder by using a closed loop analysis-by-synthesis approach to obtain a coded low band signal, encoding an original high band signal at the encoder by using an open loop energy matching approach to obtain coded high band energy envelopes, comparing an energy of the coded low band signal with an energy of a corresponding original low band signal for a subframe, and generating an indication flag that indicates whether an energy envelope perceptual correction is needed for the subframe based on comparing the energy.

Abstract: According to one embodiment, an AV device comprises a receiving section, a processing section, a storage section and a control section. The receiving section receives a digital voice signal. The processing section applies a predetermined signal processing operation to the digital voice signal received by the receiving section. The storage section stores information indicating time required for the signal processing operation at the processing section, and when a voice has been set in a mute state, stores the information indicating the time required for the signal processing operation by the processing section which is rewritten into a value that cannot be taken in general. The control section outputs information stored in the storage section upon an external request. Other embodiments are also described.

Abstract: Disclosed are systems, methods, and computer readable media for performing speech synthesis. The method embodiment comprises applying a first part of a speech synthesizer to a text corpus to obtain a plurality of phoneme sequences, the first part of the speech synthesizer only identifying possible phoneme sequences, for each of the obtained plurality of phoneme sequences, identifying joins that would be calculated to synthesize each of the plurality of respective phoneme sequences, and adding the identified joins to a cache for use in speech synthesis.

Abstract: The disclosed embodiments include systems, methods, apparatuses, and computer-readable mediums for compensating one or more signals and/or one or more parameters for time delays in one or more signal processing paths.

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: An audio decoding device comprises an audio decoding section for decoding first stream data and second stream data to generate two pieces of audio data and a data processing parameter, an external setting section in which a parameter corresponding to the data processing parameter is set, and an audio data processing section for processing the two pieces of audio data. When the data processing parameter contained in the second stream data is inappropriate, the audio data processing section performs data processing using the parameter set in the external setting section. When the data processing parameter contained in the second stream data is appropriate, the audio data processing section performs data processing using the data processing parameter generated by the audio decoding section.

Abstract: A gateway apparatus receives a call control signal and/or a packet with voice data stored therein in a predetermined protocol from a packet transfer apparatus on a mobile high-speed network and converts the received protocol into a circuit-switched protocol used when an RNC connects to a circuit switching equipment on a mobile circuit-switched network, for output to the circuit switching equipment The gateway apparatus, on receipt of a call process signal and/or a voice signal, from the circuit switching equipment, converts the received protocol for output to the packet transfer apparatus.

Abstract: Provided is a speech encoding device that is capable of performing encoding in an extension encoder even when the core encoder and core decoder of each layer have been interchanged, and that is also capable of performing high precision encoding by using the appropriate codec for each situation. The speech encoding device (100) performs hierarchical encoding of a speech signal by using the information of a lower layer in a higher layer. A core encoder (102) in the speech encoding device (100) generates a code by encoding the speech signal. A core decoder (104) generates a decoded signal by decoding the code generated by the core encoder (102). An adding unit (106) detects the encoding residual between the speech signal and the decoded signal generated by the core decoder (104). An auxiliary analyzing unit (107) inputs the decoded signal and generates lower layer information by conducting analysis processing and adjustment processing.

Abstract: A speech enhancement system that improves the intelligibility and the perceived quality of processed speech includes a frequency transformer and a spectral compressor. The frequency transformer converts speech signals from the time domain to the frequency domain. The spectral compressor compresses a pre-selected portion of the high frequency band and maps the compressed high frequency band to a lower band limited frequency range.

Abstract: A “STAC Codec” provides audio transcoding and decoding by processing an encoded audio signal using a backward-adaptive run-length Golomb-Rice (RLGR) decoder to recover transform coefficients of the encoded audio signal. The transform coefficients are then either transcoded in the transform domain to lossy or other formats, or decoded to the time domain by applying an inverse integer-reversible modulated lapped transform (MLT) to the recovered transform coefficients to recover an uncompressed time domain representation compressed audio signal. In additional embodiments, an inter-block spectral estimation and inverse data sorting strategy is used in recovering the transform coefficients from the encoded audio signal.

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 voice transmission apparatus includes a jitter absorbing buffer unit provided in the receiving unit and which absorbs a transmission delay in the packets, and a transmission wait control unit provided in a transmitting unit and which links a packet to be transmitted when transmission temporarily breaks down to a queue. A transmission wait control unit 15 includes an accumulated time computing unit 18 that computes the accumulated time of voice data in packets linked to a queue 19. The transmission wait control unit 15 extracts an oldest linked packet from the queue 19 and discards the same so that the accumulated time computed by the accumulated time computing unit 18 becomes equal to or less than a threshold in order to continuously link a new packet to the queue 19, thereby equalizing the threshold to an accumulation time of a jitter absorbing buffer unit 21.

Abstract: A method and device for updating statuses of synthesis filters are provided. The method includes: exciting a synthesis filter corresponding to a first encoding rate by using an excitation signal of the first encoding rate, outputting reconstructed signal information, and updating status information of the synthesis filter and a synthesis filter corresponding to a second encoding rate. In the present disclosure, the status of the synthesis filter corresponding to the current rate and the statuses of the synthesis filters at other rates are updated. Thus, synchronization between the statuses of the synthesis filters corresponding to different rates at the encoding terminal may be realized, thereby facilitating the consistency of the reconstructed signals of the encoding and decoding terminals when the encoding rate is switched, and improving the quality of the reconstructed signal of the decoding terminal.

Abstract: A portable player or a multi-channel home player includes: a mixed signal decoding unit that extracts, from a first inputted coded stream, a second coded stream representing a downmix signal into which multi-channel audio signals are mixed and supplementary information for reverting the downmix signal back to the multi-channel audio signals before being downmixed, and that decodes the second coded stream representing the downmix signal; a signal separation processing unit that separates the downmix signal obtained by decoding based on the extracted supplementary information and that generates audio signals which are acoustically approximate to the multi-channel audio signals before being downmixed; and headphones or speakers that reproduce the decoded downmix signal or speakers that reproduce the multi-channel audio signals separated from the downmix signal.

Abstract: In a noise suppression apparatus for suppressing noise contained in a speech signal, the speech signal is converted to a first vector of spectral speech components and a second vector of spectral speech components identical to the first vector. A vector of noise suppression coefficients is determined based on the first vector spectral speech components. A vector of estimated noise components is determined based on the first vector spectral speech components, and a speech section correction factor and a nonspeech section correction factor are calculated from the estimated noise components and the first-vector spectral speech components to produce a combined correction factor. The noise suppression coefficients are weighted by the combined correction factor to produce a vector of post-suppression coefficients. The second vector spectral speech components are weighted by the post-suppression coefficients to produce a vector of enhanced speech components.

Abstract: A mobile or in-vehicle communication system and method facilitate communication among groups. The system and method also facilitate the creation of such groups. The system and method may convert speech from one member of the group to text for distribution to other members of the group, for whom the text is converted to audible speech.

Abstract: An underwater communications system is provided that transmits electromagnetic and/or magnetic signals to a remote receiver. The transmitter includes a data input. A digital data compressor compresses data to be transmitted. A modulator modulates compressed data onto a carrier signal. An electrically insulated, magnetic coupled antenna transmits the compressed, modulated signals. The receiver that has an electrically insulated, magnetic coupled antenna for receiving a compressed, modulated signal. A demodulator is provided for demodulating the signal to reveal compressed data. A de-compressor de-compresses the data. An appropriate human interface is provided to present transmitted data into text/audio/visible form. Similarly, the transmit system comprises appropriate audio/visual/text entry mechanisms.

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 method and apparatus for automated analysis of emotional content of speech is presented. Telephony calls are routed via a network such as public service telephone network (PSTN) and delivered to an interactive voice response system (IVR) where prerecorded or synthesized prompts guide a caller to speech responses. Speech responses are analyzed for emotional content in real time or collected via recording and analyzed in batch. If performed in real time, results of emotional content analysis (ECA) may be used as input to IVR call processing and call routing. In some applications this might involve ECA input to expert system process whose results interact with an IVR for prompt creation and call processing. In any case, ECA data is valuable on its own and may be culled and restated in the form of reports for business application.

Abstract: A digital voice recording device includes a storage unit, a display unit, and a processing unit. The processing unit includes a recording module, a storing module, a marking module, and a playing module. The recording module converts audio into digital signals, and records the digital signals into an audio file. Each audio file is associated with a document including textual content of the audio file. The storing module stores the audio file and the document. The display module displays the document. The marking module creates a plurality of flags for the audio file. Each flag is associated with a time point in the audio file, and is assigned an identifier. The playing module identifies an identifier of a flag to acquire a time point in response to a user input, and begin playing the audio file from the acquired time point.

Abstract: An audio encoder and decoder use architectures and techniques that improve the efficiency of quantization (e.g., weighting) and inverse quantization (e.g., inverse weighting) in audio coding and decoding. The described strategies include various techniques and tools, which can be used in combination or independently. For example, an audio encoder quantizes audio data in multiple channels, applying multiple channel-specific quantizer step modifiers, which give the encoder more control over balancing reconstruction quality between channels. The encoder also applies multiple quantization matrices and varies the resolution of the quantization matrices, which allows the encoder to use more resolution if overall quality is good and use less resolution if overall quality is poor. Finally, the encoder compresses one or more quantization matrices using temporal prediction to reduce the bitrate associated with the quantization matrices. An audio decoder performs corresponding inverse processing and decoding.