Abstract: Audio objects are associated with positional metadata. A received downmix signal comprises downmix channels that are linear combinations of one or more audio objects and are associated with respective positional locators. In a first aspect, the downmix signal, the positional metadata and frequency-dependent object gains are received. An audio object is reconstructed by applying the object gain to an upmix of the downmix signal in accordance with coefficients based on the positional metadata and the positional locators. In a second aspect, audio objects have been encoded together with at least one bed channel positioned at a positional locator of a corresponding downmix channel. The decoding system receives the downmix signal and the positional metadata of the audio objects. A bed channel is reconstructed by suppressing the content representing audio objects from the corresponding downmix channel on the basis of the positional locator of the corresponding downmix channel.

Abstract: A system may receive a communication from a user, which may include a vocal sample. The system may transform the vocal sample from a wavelength domain into a frequency domain. The system may determine a divergence of one or more amplitude values of the transformed frequency domain from a predetermined frequency distribution. According to some embodiments, the predetermined frequency distribution may be a Benford's distribution. When the divergence exceeds a predetermined threshold, the system may execute one or more security measures. The one or more security measures may include (i) transferring the user from an automated operator to a human operator, (ii) requiring second factor authentication from the user, and/or (iii) denying a user-initiated request.

Abstract: An apparatus for processing an audio signal and method thereof are disclosed. The present invention includes receiving a downmix signal and side information; extracting control restriction information from the side information; receiving control information for controlling gain or panning at least one object signal; generating at least one of first multi-channel information and first downmix processing information based on the control information and object information, without using the control restriction information; and, generating an output signal by applying the at least one of the first multichannel information and the first downmix processing information to the downmix signal, wherein the control restriction information relates to a parameter indicating limiting degree of the control information.

Abstract: The audio coding method and system of lattice vector quantization is provided in the invention. The method comprises: dividing frequency domain coefficients of an audio signal for which a modified discrete cosine transform (MDCT) has been performed into a plurality of coding sub-bands, and quantizing and coding an amplitude envelope value of each coding sub-band to obtain coded bits of amplitude envelopes; performing bit allocation on each coding sub-band, and performing normalization, quantization and coding respectively on vectors in a low bit coding sub-band with pyramid lattice vector quantization and on vectors in a high bit coding sub-band with sphere lattice vector quantization to obtain coded bits of the frequency domain coefficients; multiplexing and packing the coded bits of the amplitude envelope and the coded bits of the frequency domain coefficients of each coding sub-band, then sending them to a decoding side.

Abstract: Downmixing multi-channel audio signals to target channels by pre-downmixing frequency coefficients that are encoded using a most frequently used block type in stereo channels in the frequency domain, thereby reducing an amount of calculations and an amount of power required to downmix the multi-channel audio signals.

Abstract: A method is provided for coding a multi-channel audio signal representing a sound scene comprising a plurality of sound sources. The method comprises decomposing the multi-channel signal into frequency bands and the following performed per frequency band: obtaining data representative of the direction of the sound sources of the sound scene, selecting a set of sound sources constituting principal sources, adapting the data representative of the direction of the selected principal sources, as a function of restitution characteristics of the multi-channel signal, determining a matrix for mixing the principal sources as a function of the adapted data, matrixing the principal sources by the matrix determined so as to obtain a sum signal with a reduced number of channels and coding the data representative of the direction of the sound sources and forming a binary stream comprising the coded data, the binary stream being transmittable in parallel with the sum signal.

Abstract: A method of encoding stereo audio that minimizes a number of pieces of side information required for parametric-encoding and parametric-decoding of the stereo audio. The side information may include parameters about interchannel intensity difference (IID), interchannel correlation (IC), overall phase difference (OPD), and interchannel phase difference (IPD), which are required to restore the mono audio to the stereo audio.

Abstract: A particular method includes loading one or more memory images into a multi-way cache. The memory images are associated with an audio decoder, and the multi-way cache is accessible to a processor. Each of the memory images is sized not to exceed a page size of the multi-way cache.

Abstract: A method of encoding stereo audio that minimizes a number of pieces of side information required for parametric-encoding and parametric-decoding of the stereo audio. The side information may include parameters about interchannel intensity difference (IID), interchannel correlation (IC), overall phase difference (OPD), and interchannel phase difference (IPD), which are required to restore the mono audio to the stereo audio.

Abstract: An input signal, in the form of a sequence of feature vectors, is transformed to an output signal by first storing parameters of a model of the input signal in a memory. Using the vectors and the parameters, a sequence of vectors of hidden variables is inferred. There is at least one vector hn of hidden variables hi,n for each feature vector xn, and each hidden variable is nonnegative. The output signal is generated using the feature vectors, the vectors of hidden variables, and the parameters. Each feature vector xn is dependent on at least one of the hidden variables hi,n for the same n. The hidden variables are related according to h i , n = ? j , l ? ? c i , j , l ? ? l , n ? h j , n - 1 , where j and l are summation indices. The parameters include non-negative weights ci,j,l, and ?l,n are independent non-negative random variables.

Abstract: A method and system for peak limiting of speech signals for delay sensitive voice communication is disclosed. In an embodiment, a position of a sample with highest magnitude within a current block of samples is determined. Further, a peak gain to be applied for the current block of samples to bring down the highest magnitude to a predetermined threshold value is determined. Furthermore, a gain delta by which an old gain is updated to the peak gain is computed. Then, a gain factor is computed for the current block of samples based on the position of the sample with highest magnitude and the gain delta. Subsequently, the gain factor is set to a predetermined minimum gain factor when the computed gain factor is less than the predetermined minimum gain factor. In addition, gain is applied to the current block of samples using the gain factor.

Abstract: A hybrid speech encoder detects changes from music-like sounds to speech-like sounds. When the encoder detects music-like sounds (e.g., music), it operates in a first mode, in which it employs a frequency domain coder. When the encoder detects speech-like sounds (e.g., human speech), it operates in a second mode, and employs a time domain or waveform coder. When a switch occurs, the encoder backfills a gap in the signal with a portion of the signal occurring after the gap.

Abstract: A method for optimizing multiple psychoacoustic effects in a sound system includes synthesizing a high-frequency restored version of a input signal; adding the high-frequency restored version of the input signal to the input signal to create a second signal; synthesizing a third signal having enhanced spatialization from the second signal; synthesizing a fourth signal having virtual bass from the second signal; and, adding the third and fourth signals, or second, third and fourth signals, together to create an output signal.

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

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

Abstract: A system for classification of the emotional content of music is provided. An encoder receives a digital audio recording of a piece of music, and encodes it using musical notes and associated amplitudes. The artificial neural network is configured to take a plurality of encoded time slices and provide output indicative of the emotional content of the music.

Abstract: A data compression/decompression apparatus, for example, acquires sampling data obtained by sampling an audio signal with a predetermined period, and converts the sampling data into frequency domain data. The data compression/decompression apparatus divides a data sequence of the converted frequency domain data into a plurality of blocks such that the number of pieces of data included in each block is variable, and compresses each block.

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: A voice-coded in-band communication device monitors a voice-coded channel to detect data to present to a user. During operation, the communication device can detect a data-encoding signal from the voice-coded channel, such that the voice-coded channel can carry an audio signal that includes a voice signal and the data-encoding signal. The device decodes the data-encoding signal to detect a data element. The data element can include information that is to be presented to a local user, a request from a remote device for information about the local user, or information that the system can use to establish a peer-to-peer connection with the remote device over a separate data channel. The device can also generate a filtered audio signal to present to the user by removing the detected data-encoding signal from the voice-coded channel, and then reproduces the filtered audio signal for the user.

Abstract: A method and apparatus provide for audio frame recovery by identifying a sequence of lost frames of coded audio data as being lost or corrupted; identifying a first frame of coded audio data which immediately preceded the sequence of lost frames, as having been encoded using a time domain coding method; identifying a second frame of coded audio data, which immediately followed the sequence of lost frames of coded audio data, as having been encoded using a transform domain coding method; obtaining a pitch delay; generating a second decoded audio portion of the second frame based on the second frame; generating a first decoded audio portion of the second frame based on the pitch delay and decoded audio samples; and generating a decoded audio output of the second frame based on a sequential combination of the first and second decoded audio portions.

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

Abstract: An embodiment of the present invention discloses a system and method for decoding multiple independent encoded audio streams using a single decoder. The system includes one or more parsers, a preprocessor, an audio decoder, and a renderer. The parser extracts individual audio frames from each input audio stream. The preprocessor combines the outputs of all parsers into a single audio frame stream and enables sharing of the audio decoder among multiple independent encoded audio streams. The audio decoder decodes the single audio frame stream and provides a single decoded audio stream. And the renderer renders the individual reconstructed audio streams from the single decoded audio stream.

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: An audio decoding method is provided. In the audio decoding method, a synchronization word and a corresponding packet header are inserted at the beginning of each packet data. A position of the packet data is confirmed according to the synchronization word, and the packet data is then decoded according to information in the packet header. Accordingly, when an error occurs during the decoding process, the decoding process skips to a next packet data for decoding to avoid noise. In addition, a packet header can be directly accessed in the situation of a fast-forward operation to obtain decoding information of the packet data to perform audio decoding.

Abstract: The invention can be a simple method for data transfer from one electronic device to another. In this embodiment, a sender can upload data to a server using an out-of-band connection while broadcasting an identification signal over one or several mediums, such as acoustic and/or radio (Ultrasound, Bluetooth, infrared, etc. . . . ). In the case that a connection to the server can be established, the receiver will detect the identification signal, decode it, and request the information from the server. The receiver can then send an authorization for a transaction through the server via an out-of-band connection or directly to the sender via one of the primary communication mediums, at which point the transaction is complete.

Abstract: A stereo audio decoder generates a set of stereo output channels in response to a parametric audio input including signal parameters and stereo related parameters. A parameter processor generates two different set of parameters based on the input signal parameters thus up-mixing the signal parameters by altering or manipulating the signal parameters corresponding to the stereo related parameters. The two different parameters are synthesized by separate signal synthesizers to form respective stereo output channels. The signal synthesizers may be sinusoidal synthesizers, and the decoder also includes transient and noise synthesizers to generate transient and noise signal portions to be applied to the stereo output channels. Further, different transient and noise signal portions to the output channels may be provided by applying different gains based on the stereo related parameter. The two different parameters may be determined from current and previous signal parameter inputs using an input delay line.

Abstract: System and method to improve intelligibility of coded speech, the method including: receiving an encoded speech signal from a network; extracting an encoded media data stream and one or more control data packets from the encoded speech signal; decoding the encoded media data stream to produce a decoded speech signal; boosting an upper spectral portion of the decoded speech signal to produce a boosted speech signal; and outputting the boosted speech signal. In another embodiment, the method may include: receiving an uncoded speech signal; processing the uncoded speech signal, wherein the processing comprises generating an unencoded data stream from the uncoded speech signal; boosting an upper spectral portion of the unencoded data stream to produce a boosted speech signal; encoding the boosted speech signal to produce an encoded speech signal; and outputting the boosted speech signal.

Abstract: A composite memory card has 4-bit and 1-bit data transfer modes; in which, when the composite memory card plugs in a general reader, it operates in the 4-bit data transfer mode to form 4-bit format information signal linking with the external transfer through four data pins; and when it plugs in a dedicated reader, it operates in the 1-bit data transfer mode to form 1-bit format data signal linking with the external transfer through one data pin, while it starts the internal audio processing module to read the internal audio file streaming and to transform into voice output signal output to the dedicated reader through the other data pins for broadcast, and to receive the voice input signal of the dedicated reader through data pins to transform into audio file streaming to import for storage; thus, the composite memory card combines the standard memory card and the audio processing function.

Abstract: Methods and apparatus for voice and data interlacing in a system having a shared antenna. In one embodiment, a voice and data communication system has a shared antenna for transmitting and receiving information in time slots, wherein the antenna can only be used for transmit or receive at a given time. The system determines timing requirements for data transmission and reception and interrupts data transmission for transmission of speech in selected intervals while meeting the data transmission timing and throughput requirements. The speech can be manipulated to fit with the selected intervals, to preserve the intelligibility of the manipulated speech.

Abstract: An audio signal processing system includes parallel speech and generic audio signal processing paths. One path includes a linear predictive coder and a resampling filter having a non-linear phase characteristic. A phase compensation filter is disposed along the one of the processing paths to compensate for the non-linearity of the resampling filter thereby enabling relatively seamless switching between the coders resulting in a reduction of audio artifacts that would otherwise result from the non-linear phase characteristic of the resampling filter during playback.

Abstract: Transferring data via audio link is described. In an example a short sequence of data can be transferred between two devices by encoding the sequence of data as an audio sequence. For example, the audio sequence may be a sequence of tones which vary in dependence on the encoded data. The sequence of data may be encoded by a first device and transmitted using a loudspeaker associated with the first device. At least one mobile communications device can be used to capture the audio sequence, for example using a microphone, and to decode the sequence, retrieving the data encoded therein. In some examples the encoded data may comprise a shortened URL or other information which can be used to control one or more aspects of the capture device.

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: Methods and apparatuses for encoding and decoding a multi-channel audio signal are provided. In the encoding method, spatial information is calculated based on a multi-channel audio signal and a down-mix signal, and a compensation parameter that compensates for the down-mix signal is calculated based on the multi-channel audio signal and the down-mix signal. Thereafter, a bitstream is generated by encoding the spatial information, the compensation parameter, and the down-mix signal and combining the results of the encoding. Therefore, it is possible to prevent deterioration of the quality of sound regarding a multi-channel audio signal by compensating for the multi-channel audio signal using a compensation parameter that compensates for a down-mix signal.

Abstract: A method for determining mantissa bit allocation of frequency domain audio data to be encoded, including by performing adaptive low frequency compensation on each frequency band of a set of low frequency bands of the data. The low frequency compensation includes steps of: performing tonality detection on the audio data to generate compensation control data indicative of whether each frequency band in the set has prominent tonal content; and performing low frequency compensation on each frequency band in the set having prominent tonal content, including by correcting a preliminary masking value for each frequency band having prominent tonal content, but not performing low frequency compensation on the audio data in any other frequency band in the set. Other aspects are audio encoding methods including such tonality detection and low frequency compensation steps, and a system configured to perform any embodiment of the inventive method.

Abstract: A method (700, 800) and apparatus (100, 200) processes audio frames to transition between different codecs. The method can include producing (720), using a first coding method, a first frame of coded output audio samples by coding a first audio frame in a sequence of frames. The method can include forming (730) an overlap-add portion of the first frame using the first coding method. The method can include generating (740) a combination first frame of coded audio samples based on combining the first frame of coded output audio samples with the overlap-add portion of the first frame. The method can include initializing (760) a state of a second coding method based on the combination first frame of coded audio samples. The method can include constructing (770) an output signal based on the initialized state of the second coding method.

Abstract: A digital audio signal can be processed using continuously variable time-frequency resolution by selecting a portion of an input digital audio signal, resampling the selected portion of the input digital audio signal, generating a plurality of spectral characteristics associated with the resampled portion of the input digital audio signal, generating a portion of an output digital audio signal from the plurality of spectral characteristics, and resampling the portion of the output digital audio signal. Further, resampling the selected portion of the input digital audio signal can comprise determining a sampling ratio and resampling the selected portion of the input digital audio signal in accordance with the determined sampling ratio. Additionally, the portion of the output digital audio signal can be resampled in accordance with the inverse of the determined sampling ratio. The sampling ratio can be determined based on a time-frequency resolution requirement associated with an audio processing algorithm.

Abstract: Systems and methods for determining mean opinion scores (MOS) for variable bit rate (VBR) audio streams transmitted over VoIP networks are described. In an embodiment, a method may include monitoring a communication over a network and detecting portions of the communication including packets having a different packet payload sizes. The method may also include deriving bit rates corresponding to those portions. The method may then include calculating MOS values for each portion based on the derived bit rates, and calculating an overall MOS value for the communication based upon each individual MOS value averaged according to a distribution of packets having the different packet payload sizes.

Abstract: An apparatus for distinguishing a voice is described. In one embodiment, the apparatus includes a server with a communication interface, a frame generator, and a sound analyzer. The communication interface processes an incoming communication stream with an echo canceller to cancel echo in the communication stream. The frame generator operates on a processor and generates a plurality of frames from the communication stream. Each of the plurality of frames contains data for a period of time from the communication stream. The frame generator also assigns a frame value to each of the plurality of frames. The sound analyzer determines a status of the communication stream by analyzing the frame values of the plurality of frames.

Abstract: A method and apparatus provides for encoding an audio signal. A bit rate value is received. A set of energy thresholds based on the bit rate value is selected. The set of energy thresholds is one of a plurality of sets of energy thresholds. The energy thresholds of each set of energy thresholds correspond on a one-to-one basis with a set of sub-bands of the audio signal. The audio signal is received. The energy of each sub-band of the set of sub-bands is determined. A highest frequency sub-band that has an energy exceeding the corresponding threshold is determined. A selected bandwidth of the audio signal is encoded. The selected bandwidth includes only those frequencies of the audio signal that are in the highest frequency sub-band that has an energy exceeding the corresponding threshold, as well as the lower frequencies of the audio signal that are above a high-pass cut-off frequency.

Abstract: An apparatus comprising an ingress port configured to receive a signal comprising a plurality of encoded audio signals corresponding to a plurality of sources; and a processor coupled to the ingress port and configured to calculate a parameter for each of the plurality of encoded audio signals, wherein each parameter is calculated without decoding any of the encoded audio signals, select some, but not all, of the plurality of encoded audio signals according to the parameter for each of the encoded audio signals, decode the selected signals to generate a plurality of decoded audio signals, and combine the plurality of decoded audio signals into a first audio signal.

Abstract: Noise robust template matching may be performed. First features of a first signal may be computed. Based at least on a portion of the first features, second features of a second signal may be computed. A new signal may be generated based on at least another portion of the first features and on at least a portion of the second features.

Abstract: A particular method includes transitioning out of a low-power state at a processor. The method also includes retrieving audio feature data from a buffer after transitioning out of the low-power state. The audio feature data indicates features of audio data received during the low-power state of the processor.

Abstract: A method and apparatus for reproducing audio data using low power are provided. The apparatus may reproduce the audio data by determining a power mode based on a memory resource of an internal memory, and an amount of a memory required for reproducing the audio data, controlling a power based on the determined power mode, and decoding the audio data.

Abstract: The embodiments described herein are directed to systems and methods for transmitting audio data and control segment in a single bitstream and reducing audio disturbance associated with the control segment when the bitstream is processed by an audio digital-to-analog converter. The system, according to one aspect, comprises a first audio unit, a transmitter coupled to the first audio unit, a receiver coupled to the transmitter, a second audio unit coupled to the receiver, a first processor coupled to at least one of the first audio unit and the transmitter, a second processor coupled to the second audio unit and the receiver, and an audio digital-to-analog converter connected to the second processor.

Abstract: A portable electronic device includes an audio coder-decoder ‘CODEC’ capable of generating analog audio signals from digital audio representations, and includes a transmit coil capable of producing an alternating magnetic field upon passage of the analog audio signals through the transmit coil. Automatically determining the presence of conditions for magnetic coupling between the portable electronic device and an audio reproduction accessory results in the portable electronic device causing the analog audio signals to be routed from the CODEC to the transmit coil. When an audio reproduction accessory for a portable electronic device is magnetically coupled to the portable electronic device, the sole source of energy for audible sound generated by the audio reproduction accessory may be energy contained in a magnetic field that acts on the audio reproduction accessory, the magnetic field produced by a transmit coil of the portable electronic device.

Abstract: An audio encoding device capable of efficient encoding processing includes: a storage unit which stores audio data; a data acquisition controller which acquires the audio data from the storage unit; a transformation unit which processes an audio data signal outputted from the data acquisition unit for frequency transformation; a harmonic overtone generation/synthesizing unit which generates a harmonic based on a first output wave out of an output wave of the transformation unit and synthesizes the harmonic and a second output wave out of the output wave of the transformation unit, the second output wave being higher in frequency than the first output wave; and an encoder which subjects an output from the harmonic overtone generation/synthesizing unit to encoding processing.

Abstract: A sound process apparatus includes a processor. The processor may execute instructions, which are stored on a memory, and when executed cause the sound process apparatus to perform operations. An obtaining operation may obtain sound data in a remote site. A first determining operation may determine volume levels of voice and noise in the remote site based on the sound data. A second determining operation may determine a volume level of noise in a local site based on the sound in the local site. A third determining operation may determine a target volume level based on the volume level of the voice in the remote site, the volume level of the noise in the remote site, and the volume level of the noise in the local site. A notifying operation may notify a user of the target volume level.

Abstract: A method and device are provided for modifying a compounded voice message having at least one first voice component. The method includes a step of obtaining at least one second voice component, a step of updating at least one item of information belonging to a group of items of information associated with the compounded voice message as a function of the at least one second voice component and a step of making available the compounded voice message comprising the at least one first and second voice components, and the group of items of information associated with the compounded voice message. The compounded voice message is intended to be consulted by at least one recipient user.

Abstract: The present application discloses coding and decoding (CODEC) devices and operating and driving methods thereof. The CODEC device includes a first interface compatible with High Definition Audio (HDA) specification, a second interface compatible with Musical Instrument Digital Interface (MIDI) specification, and a converter. The converter is configured to convert a first MIDI command received from the first interface and output a corresponding first converted MIDI command via the second interface, and to convert a second MIDI command received from the second interface and output a corresponding second converted MIDI command via the first interface.