Abstract: Methods and arrangements for effecting a cloud representation of audio content. An audio cloud is created and rendered, and user interaction with at least a clip portion of the audio cloud is afforded.

Abstract: An audio encoder and decoder use architectures and techniques that improve the efficiency of multi-channel 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 performs a pre-processing multi-channel transform on multi-channel audio data, varying the transform so as to control quality. The encoder groups multiple windows from different channels into one or more tiles and outputs tile configuration information, which allows the encoder to isolate transients that appear in a particular channel with small windows, but use large windows in other channels. Using a variety of techniques, the encoder performs flexible multi-channel transforms that effectively take advantage of inter-channel correlation. An audio decoder performs corresponding processing and decoding. In addition, the decoder performs a post-processing multi-channel transform for any of multiple different purposes.

Abstract: Embodiments of the present invention disclose an audio decoding method, including: determining that bitstreams to be decoded are monophony coding layer and first stereo enhancement layer bitstreams; decoding the monophony coding layer to obtain a monophony decoded frequency-domain signal; reconstructing left and right channel frequency-domain signals in a first sub-band region by utilizing the monophony decoded frequency-domain signal after an energy adjustment; and reconstructing left and right channel frequency-domain signals in a second sub-band region by utilizing the monophony decoded frequency-domain signal without the energy adjustment.

Abstract: Encoder-assisted frame loss concealment (FLC) techniques for decoding audio signals are described. A decoder may discard an erroneous frame of an audio signal and may implement the encoder-assisted FLC techniques in order to accurately conceal the discarded frame based on neighboring frames and side-information transmitted from the encoder. The encoder-assisted FLC techniques include estimating magnitudes of frequency-domain data for the frame based on frequency-domain data of neighboring frames, and estimating signs of the frequency-domain data based on a subset of signs transmitted from the encoder as side-information. Frequency-domain data for a frame of an audio signal includes tonal components and noise components. Signs estimated from a random signal may be substantially accurate for the noise components of the frequency-domain data.

Abstract: A system and method may be configured to analyze audio information derived from an audio signal. The system and method may track sound pitch across the audio signal. The tracking of pitch across the audio signal may take into account change in pitch by determining at individual time sample windows in the signal duration an estimated pitch and a representation of harmonic envelope at the estimated pitch. The estimated pitch and the representation of harmonic envelope may then be implemented to determine an estimated pitch for another time sample window in the signal duration with an enhanced accuracy and/or precision.

Abstract: A decoded sound analysis unit (104) calculates, regarding the frequency-region stereo signals L(b) and R(b) decoded by the PS decoding unit (103), a second degree of similarity (109) and a second intensity difference (110) from the decoded sound signals. A spectrum correction unit (105) detects a distortion added by the parametric-stereo conversion by comparing the second degree of similarity (109) and the second intensity difference (110) calculated at the decoding side with the first degree of similarity (107) and the first intensity difference (108) calculated and transmitted from the encoding side, and corrects the spectrum of the frequency-region stereo decoded signals L(b) and R(b).

Abstract: A device and a method for maintaining voice communication security in a terminal are provided, which enable the terminal to maintain security for a conversation during voice communication. The device includes a microphone for receiving voice data through a microphone in a voice communication mode; and a controller for making a control to decode encoded characters included in voice data received through a microphone in a voice communication mode, and then transmitting the decoded characters to a counterpart terminal communicating with the device.

Abstract: Provided are a coding device, a communication processing device, and a coding method, whereby processing operation load (computational load) is significantly reduced for a configuration which computes either frame energy or sub-frame energy of an input signal, using auto-correlation operations, without causing a decline in the precision of either the frame energy or the sub-frame energy. In a coding device (101), a sub-frame energy computation unit (201) computes the sub-frame energy by substituting the sum of input signal auto-correlation operations in a first range with the sum of auto-correlation operations in a second range which differs at least partially from the first range.

Abstract: Embodiments herein include receiving a request to modify an audio characteristic associated with a first user for a voice communication system. One or more suggested modified audio characteristics may be provided for the first user, based on, at least in part, one or more audio preferences established by another user. An input of one or more modified audio characteristics may be received for the first user for the voice communication system. A user-specific audio preference may be associated with the first user for voice communications on the voice communication system, the user-specific audio preference including the one or more modified audio characteristics.

Abstract: An apparatus for generating bandwidth extension output data for an audio signal has a noise floor measurer, a signal energy characterizer and a processor. The audio signal has components in a first frequency band and components in a second frequency band, the bandwidth extension output data are adapted to control a synthesis of the components in the second frequency band. The noise floor measurer measures noise floor data of the second frequency band for a time portion of the audio signal. The signal energy characterizer derives energy distribution data, the energy distribution data characterizing an energy distribution in a spectrum of the time portion of the audio signal. The processor combines the noise floor data and the energy distribution data to obtain the bandwidth extension output data.

Abstract: Methods and apparatus for coordinating audio data processing and network communication processing in a communication device are disclosed. An exemplary method begins with demodulating a series of received communication frames, using a network communication processing circuit, to produce received encoded audio frames. An event report for each of one or more of the received encoded audio frames is generated, the event report indicating a network communication circuit processing time associated with the corresponding received encoded audio frames. The received encoded audio frames are decoded, using an audio data processing circuit, and the decoded audio is output to an audio circuit. The timing of the outputting of the decoded audio is adjusted, based on the generated event reports.

Abstract: An SBR encoder includes a filter bank that receives an input signal, a time/frequency grid generator that controls a number of bits of various parameters, a parameter calculator that calculates various parameters, a parameter coding unit that encodes the parameters, an upper-limit number-of-bit storage unit that stores an upper limit of the number of bit of encoded data of high-frequency component finally generated in a high-pass encoding process, and a number-of-bit controller. The number-of-bit controller controls the high-pass encoding process by preferentially encoding a parameter having a large influence to sound quality and not encoding a parameter having a small influence to the sound quality relative to a plurality of parameters, so that the number of bits of the encoded data of high-frequency component finally generated in the high-pass encoding process becomes equal to or less than the upper limit to be stored in the upper-limit number-of-bit storage unit.

Abstract: The technology of the present application provides a method and apparatus to allow for dynamically updating a language model across a large number of similarly situated users. The system identifies individual changes to user profiles and evaluates the change for a broader application, such as, a dialect correction for a speech recognition engine, as administrator for the system identifies similarly situated user profiles and downloads the profile change to effect a dynamic change to the language model of similarly situated users.

Abstract: The detection of double-talk in audio communication is provided. A communication device may receive an echo signal mixed with a speech signal at a near end location. The echo signal may be generated by speech transmitted by a remote party at a far end location to a local party at the near end location. The speech signal may be received from the local party for transmission to the remote party. The communication device may then filter the echo signal and the speech signal. The communication device may then analyze the speech signal to identify speech characteristics which indicate the presence of double-talk. The communication device may then set a flag upon identifying the speech characteristics which indicate the presence of the double-talk. The communication device may then process the filtered signals to further suppress remaining echo prior to transmission of the speech signal to the remote party.

Abstract: There is provided an information processing device including a storage unit that stores music data for playing music and lyrics data indicating lyrics of the music, a display control unit that displays the lyrics of the music on a screen, a playback unit that plays the music and a user interface unit that detects a user input. The lyrics data includes a plurality of blocks each having lyrics of at least one character. The display control unit displays the lyrics of the music on the screen in such a way that each block included in the lyrics data is identifiable to a user while the music is played by the playback unit. The user interface unit detects timing corresponding to a boundary of each section of the music corresponding to each displayed block in response to a first user input.

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: 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: Computationally implemented methods and systems include receiving speech data correlated to one or more words spoken by a particular party, receiving adaptation data that is at least partly based on at least one speech interaction of a particular party that is discrete from the received speech data, wherein at least a portion of the adaptation data has been stored on a particular device associated with the particular party, obtaining target data regarding a target configured to process at least a portion of the received speech data, and determining whether to apply the adaptation data for processing at least a portion of the received speech data, at least partly based on the acquired target data. In addition to the foregoing, other aspects are described in the claims, drawings, and text.

Abstract: Computationally implemented methods and systems include detecting speech data related to a speech-facilitated transaction, acquiring adaptation data that is at least partly based on at least one speech interaction of a particular party that is discrete from the detected speech data, wherein at least a portion of the adaptation data has been stored on a particular device associated with the particular party, obtaining a destination of one or more of the adaptation data and the speech data, and transmitting one or more of the speech data and the adaptation data to the acquired destination. In addition to the foregoing, other aspects are described in the claims, drawings, and text.

Abstract: Computationally implemented methods and systems include detecting speech data related to a speech-facilitated transaction, acquiring adaptation data that is at least partly based on at least one speech interaction of a particular party that is discrete from the detected speech data, wherein at least a portion of the adaptation data has been stored on a particular device associated with the particular party, obtaining a destination of one or more of the adaptation data and the speech data, and transmitting one or more of the speech data and the adaptation data to the acquired destination. 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: The instant application includes computationally-implemented systems and methods that include acquiring indication of a speech-facilitated transaction between a particular party and a target device, receiving adaptation data correlated to the particular party, the receiving facilitated by a particular device associated with the particular party, processing audio data from the particular party at least partly using the received adaptation data correlated to the particular party, and updating the adaptation data based at least in part on a result of the processed audio data, such that the updated adaptation data is configured to be transmitted to the particular device. In addition to the foregoing, other aspects are described in the claims, drawings, and text.

Abstract: Computationally implemented methods and systems 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, in response to an indicator related to a speech-facilitated transaction of a particular party, wherein the adaptation data is correlated to at least one aspect of speech of the particular party, and determining whether to update the adaptation data, said determination at least partly based on a result of at least a portion of the speech-facilitated transaction In addition to the foregoing, other aspects are described in the claims, drawings, and text.

Abstract: Computationally implemented methods and systems include receiving speech data correlated to one or more words spoken by a particular party, receiving adaptation data that is at least partly based on at least one speech interaction of a particular party that is discrete from the received speech data, wherein at least a portion of the adaptation data has been stored on a particular device associated with the particular party, obtaining target data regarding a target configured to process at least a portion of the received speech data, and determining whether to apply the adaptation data for processing at least a portion of the received speech data, at least partly based on the acquired target data. 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 acquiring indication of a speech-facilitated transaction between a particular party and a target device, receiving adaptation data correlated to the particular party, the receiving facilitated by a particular device associated with the particular party, processing audio data from the particular party at least partly using the received adaptation data correlated to the particular party, and updating the adaptation data based at least in part on a result of the processed audio data, such that the updated adaptation data is configured to be transmitted to the particular device. In addition to the foregoing, other aspects are described in the claims, drawings, and text.

Abstract: Configurations disclosed herein include systems, methods and apparatus that may be applied in a voice communications and/or storage application to remove, enhance, and/or replace the existing context. Example embodiments may first remove any existing context from a digital audio signal to obtain a context suppressed signal. The context suppressed signal may then be encoded. An audio context may be selected from among a plurality of audio contexts, with the selected audio context inserted into a signal based on the encoded context suppressed signal.

Abstract: A non-acoustic sensor is used to measure a user's speech and then broadcasts an obscuring acoustic signal diminishing the user's vocal acoustic output intensity and/or distorting the voice sounds making them unintelligible to persons nearby.

Abstract: A vector joint encoding/decoding method and a vector joint encoder/decoder are provided, more than two vectors are jointly encoded, and an encoding index of at least one vector is split and then combined between different vectors, so that encoding idle spaces of different vectors can be recombined, thereby facilitating saving of encoding bits, and because an encoding index of a vector is split and then shorter split indexes are recombined, thereby facilitating reduction of requirements for the bit width of operating parts in encoding/decoding calculation.

Abstract: An advanced audio coding (AAC) encoder quantization architecture is described. The architecture includes an efficient, low computation complexity approach for estimating scalefactors in which a base scalefactor estimate is adjusted by a delta scalefactor estimate that is based, in part, on global scalefactor adjustments applied to the previously quantized/encoded frame. Using such feedback, the AAC encoder quantization architecture is able to produce scalefactor estimates that are very close to the actual scalefactor applied by the subsequent quantization and encoding process. The architecture further includes a frequency hole avoidance approach that reduces a magnitude of an estimated scalefactor to avoid generating frequency holes in quantized SFBs.

Abstract: Some embodiments disclosed herein store a target application and a dictation application. The target application may be configured to receive input from a user. The dictation application interface may include a full overlay mode option, where in response to selection of the full overlay mode option, the dictation application interface is automatically sized and positioned over the target application interface to fully cover a text area of the target application interface to appear as if the dictation application interface is part of the target application interface. The dictation application may be further configured to receive an audio dictation from the user, convert the audio dictation into text, provide the text in the dictation application interface and in response to receiving a first user command to complete the dictation, automatically copy the text from the dictation application interface and inserting the text into the target application interface.

Abstract: Methods of encoding a signal using a perceptual model are described in which a signal to mask ratio parameter within the perceptual model is tuned. The signal to mask ratio parameter is tuned based on a function of the bitrate of the part of the signal which has already been encoded and the target bitrate for the encoding process. The tuned signal to mask ratio parameter is used to compute a masking threshold for the signal which is then used to quantize the signal.

Abstract: A message device records and plays messages. In a first aspect, actuation is effected by way of a single control. The control generally corresponds to the whole cover. Relatedly, in a second aspect the device is extremely simple and promotes openness communication. The device can be placed or hung almost anywhere to facilitate message leaving and receiving in almost any group or organization. In one embodiment actuation is achieved by pressing on substantially any portion of the cover. In a second embodiment, a rocker configuration is implemented to actuate a record mode by pressing on one end of the cover and a play/listen mode by pressing on an opposite end. The device comprises a base, electrical components, a cover, and in some embodiments, a side wall.

Abstract: Systems and methods are described for performing packet loss concealment (PLC) to mitigate the effect of one or more lost frames within a series of frames that represent a speech signal. In accordance with the exemplary systems and methods, PLC is performed by searching a codebook of speech-related parameter profiles to identify content that is being spoken and by selecting a profile associated with the identified content for use in predicting or estimating speech-related parameter information associated with one or more lost frames of a speech signal. The predicted/estimated speech-related parameter information is then used to synthesize one or more frames to replace the lost frame(s) of the speech signal.

Abstract: An operation method capable of adaptively operating at least one of a Speech To Text (STT) service and a Text To Speech (TTS) service according to setting or user operation and a system thereof are provided. The method includes requesting a specific type of a communication service connection to a reception side terminal by a transmission side terminal, and performing an operation of at least one of a speech to text service providing speech recognition based text and a text to speech service converting the text into speech data between the reception side terminal and the transmission side terminal, and includes one of recognizing speech data provided from the transmission side terminal and converting the speech data into a text based on a first speech process supporting device connected to the transmission side terminal.

Abstract: A voice quality enhancement (VQE) detector for a network element receiving an audio signal from a previous network element of a network, wherein said voice quality enhancement detector is adapted: to perform a voice quality enhancement detection based on the received audio signal, wherein said voice quality enhancement detection comprises detecting that at least one voice quality enhancement function, VQEF, was applied to the received audio signal by at least one previous network element of the network; and to control a voice quality enhancement processing of the received audio signal depending on the detection result.

Abstract: A method and associated device are provided for spatial synthesis of a sum signal to obtain at least two output signals, the sum signal as well as the spatialization parameters being output from a parametric coding by matrixing of an original multi-channel signal. The method comprises: decorrelation of the sum signal to obtain a decorrelated signal; applying a synthesis matrix, whose coefficients depend on the spatialization parameters, to the decorrelated signal and to the sum signal to obtain said output signals, wherein for at least one range of value of at least one spatialization parameter, the coefficients of the synthesis matrix are determined according to a criterion of minimizing a quantitative function, relating to the quantity of decorrelated signal in each of the output signals obtained by applying the synthesis matrix.

Abstract: In one embodiment, a method of receiving a decoded audio signal that has a transmitted pitch lag is disclosed. The method includes estimating pitch correlations of possible short pitch lags that are smaller than a minimum pitch limitation and have an approximated multiple relationship with the transmitted pitch lag, checking if one of the pitch correlations of the possible short pitch lags is large enough compared to a pitch correlation estimated with the transmitted pitch lag, and selecting a short pitch lag as a corrected pitch lag if a corresponding pitch correlation is large enough. The postprocessing is performed using the corrected pitch lag. In another embodiment, when the existence of irregular harmonics or wrong pitch lag is detected, a coded-excited linear prediction (CELP) postfilter is made more aggressive.

Abstract: An encoding apparatus comprises a frame processor (105) which receives a multi channel audio signal comprising at least a first audio signal from a first microphone (101) and a second audio signal from a second microphone (103). An ITD processor 107 then determines an inter time difference between the first audio signal and the second audio signal and a set of delays (109, 111) generates a compensated multi channel audio signal from the multi channel audio signal by delaying at least one of the first and second audio signals in response to the inter time difference signal. A combiner (113) then generates a mono signal by combining channels of the compensated multi channel audio signal and a mono signal encoder (115) encodes the mono signal. The inter time difference may specifically be determined by an algorithm based on determining cross correlations between the first and second audio signals.

Abstract: A method and apparatus of providing an audio output to a user in a communications system in which the audio to be output to a user, preferably an audio frame, is assessed before it is broadcast to the user, and then selectively changed on the basis of the assessment. The assessment may be carried out in the audio encoding process, in the audio decoding process and/or after the audio decoding process. The selective changing of the audio output may comprise selectively replacing the audio output and/or re-encoding of the audio output.

Abstract: Method and apparatus for processing audio signals are provided. The method for decoding an audio signal includes extracting a downmix signal and spatial information from a received audio signal and generating a pseudo-surround signal using the downmix signal and the spatial information. The apparatus for decoding an audio signal includes a demultiplexing part extracting a downmix signal and spatial information from a received audio signal and a pseudo-surround decoding part generating a pseudo-surround signal from the downmix signal, using the spatial information.

Abstract: Provided are a method and apparatus for generating a pseudo-random number which is unpredictable and which has a small memory work area, and also a method and apparatus for encrypting data, for each predetermined amount, based on the generated pseudo-random number. A seed is divided into a predetermined number of blocks, new blocks are created by calculating an exclusive-OR of the blocks being different from each other, and the new blocks are merged to generate a new pseudo-random number. The data is encrypted for each determined amount based on the generated pseudo-random number. At this time, a pseudo-random number to be used for the succeeding encryption is generated by using as a seed a predetermined amount of random number of the pseudo-random number used for the preceding encryption of the predetermined amount of data.

Abstract: A coding method, a decoding method, a coding apparatus, and a decoding apparatus are disclosed herein. A coding method includes: obtaining a value of each sample of an input data frame; determining pulse samples and non-pulse samples in the input data frame according to the distribution of values of samples of the input data frame; encoding the determined pulse samples in the input data frame in a first coding mode to obtain a first data stream; encoding the determined non-pulse samples in the input data frame in a second coding mode to obtain a second data stream; and multiplexing the first data stream and the second data stream to obtain an output coded data stream of the input data frame. Compared with the prior art, the technical solution under the present disclosure reduces the number of bits required for encoding the entire data frame is reduced, and improves the compression efficiency of the data frame with a wide dynamic range.

Abstract: A method and apparatus for transmitting an audio signal over a communication channel comprising encoding the audio signal with an encoder 204 using a first sampling rate, filtering the audio signal using a first cut off frequency, the first cut off frequency being chosen in dependence upon the first sampling rate, and transmitting the encoded and filtered audio signal over the communication channel. The presence of a condition in which the sampling rate of the encoder 204 is to be switched to a second sampling rate at a switching time is determined and if the condition has been determined to be present, the cut off frequency used in the filtering step is gradually changed from the first cut off frequency to a second cut off frequency, the second cut off frequency being chosen in dependence upon the second sampling rate, such that the audio bandwidth of the transmitted signal changes gradually when the sampling rate is switched to the second sampling rate.

Abstract: A scalable decoder device (50) for signals representing audio comprises a primary decoder (21) connected to an input (40). The primary decoder (21) is arranged to provide a primary decoded signal (23) based on received parameters (4). A primary postfilter (31) is connected to the primary decoder (23) to provide a primary postfiltered signal (32). A secondary enhancement decoder (45) is connected to the input (40) and arranged to provide a secondary decoded enhancement signal (44). The device further comprises a combiner arrangement (55), arranged for combining the primary postfiltered signal (32) and a signal (53) based on the secondary decoded enhancement signal (44) into an output signal (6) to be provided at an output (6). The combining is made with an adaptable strength relation between contributions from the two signals. A method for decoding coded signals representing audio operates in analogy with the scalable decoder device (50).

Abstract: A method, medium, and apparatus encoding and/or decoding a multichannel audio signal. The method includes detecting the type of spatial extension data included in an encoding result of an audio signal, if the spatial extension data is data indicating a core audio object type related to a technique of encoding core audio data, detecting the core audio object type; decoding core audio data by using a decoding technique according to the detected core audio object type, if the spatial extension data is residual coding data, decoding the residual coding data by using the decoding technique according to the core audio object type, and up-mixing the decoded core audio data by using the decoded residual coding data. According to the method, the core audio data and residual coding data may be decoded by using an identical decoding technique, thereby reducing complexity at the decoding end.

Abstract: A method and apparatus for laughter detection. Laughter is detected through the presence of a sequence of at least a predetermined number such as three consecutive bursts, each burst comprising a voiced portion and an unvoiced portion. After detecting bursts, n-tuples such as triplets are detected, and a likelihood of each burst N-tuple to represent laughter is provided by comparison to predetermined thresholds. Finally, a total score is assigned to the signal based on the grades associated with the triplets and parameters such as the distance between the N-tuples, the total score representing the probability that the audio signal comprises a laughter episode. The method and apparatus preferably comprise a training step and module for determining the thresholds according to manually marked audio signals.

Abstract: Methods, systems, and apparatuses are provided for performing jitter buffer enhanced joint source channel decoding. Jitter buffer enhanced joint source channel decoding may be performed in a manner that exploits parameter domain correlation. A jitter buffer stores hard bits of properly channel decoded packets, and a secondary jitter buffer is implemented to store soft bits associated with packets that are improperly channel decoded. Joint source channel decoding may be delayed to perform channel decoding of a frame in the penultimate position of the jitter buffer. The soft bits stored in the secondary jitter buffer as well as hard bits stored in the jitter buffer, which may include future frames, are utilized to perform channel decoding. The delayed jitter buffer enhanced joint source channel decoding may also be extended to iteratively perform channel decoding for giving frames at each position in the jitter buffer as they traverse the jitter buffer.

Abstract: Disclosed is a method of processing a signal, which includes receiving at least one of a first signal and a second signal, receiving mode information, and decoding the at least one of the first signal and the second signal using at least one of a first coding scheme and a second coding scheme according to the mode information. The mode information is information for indicating that a prescribed mode corresponds to one of at least three modes. The method includes detecting when a restricted mode change occurs and changing at least one mode when detecting a restricted mode change.

Abstract: Configurations disclosed herein include systems, methods and apparatus that may be applied in a voice communications and/or storage application to remove, enhance, and/or replace the existing context. Example embodiments may decode two sets of encoded frames from an encoded audio signal. The two frame sets may be encoded using different encoding schemes. For example, the bit rate or coding mode may differ between the two encoded frame sets. Based on information from one of the decoded sets of frames, a context component included in a signal represented by the other frame set may be suppressed. Other embodiments may generate an audio context signal within the mobile user terminal, and mix the generated audio signal with another decoded audio signal.

Abstract: Provided are an apparatus and method for visualizing multichannel audio signals. The apparatus includes a spatial audio decoding unit for receiving a downmix signal of a time domain, converting the downmix signal into a signal of a frequency domain to output a frequency domain downmix signal, and synthesizing a multichannel audio signal based on the spatial parameter and the downmix signal; and a multichannel visualizing unit for creating visualization information of the multichannel audio signal based on the frequency domain downmix signal and the spatial parameter.