Abstract: A method, computer program, and computer system is provided for synthesizing speech at one or more speeds. A context associated with one or more phonemes corresponding to a speaking voice is encoded, and the one or more phonemes are aligned to one or more target acoustic frames based on the encoded context. One or more mel-spectrogram features are recursively generated from the aligned phonemes and target acoustic frames, and a voice sample corresponding to the speaking voice is synthesized using the generated mel-spectrogram features.

Abstract: A method for decoding an encoded audio bitstream is disclosed. The method includes receiving the encoded audio bitstream and decoding the audio data to generate a decoded lowband audio signal. The method further includes extracting high frequency reconstruction metadata and filtering the decoded lowband audio signal with an analysis filterbank to generate a filtered lowband audio signal. The method also includes extracting a flag indicating whether either spectral translation or harmonic transposition is to be performed on the audio data and regenerating a highband portion of the audio signal using the filtered lowband audio signal and the high frequency reconstruction metadata in accordance with the flag.

Abstract: Some implementations involve analyzing audio packets received during a time interval that corresponds with a conversation analysis segment to determine network jitter dynamics data and conversational interactivity data. The network jitter dynamics data may provide an indication of jitter in a network that relays the audio data packets. The conversational interactivity data may provide an indication of interactivity between participants of a conversation represented by the audio data. A jitter buffer size may be controlled according to the network jitter dynamics data and the conversational interactivity data. The time interval may include a plurality of talkspurts.

Abstract: A communication system detects a state of a connected television. The communication system receives a request to establish a communication session with a local communication device via a network. The communication system outputs a reference audio signal to a locally connected audio output device. The communication system captures an ambient audio signal from a local microphone and compares the reference audio signal and the ambient audio signal to determine if the reference audio signal and the ambient audio signal match by meeting a similarity threshold. initiates an action with respect to the communication session based on the comparison.

Abstract: Efficient assignment of bit numbers is performed even under a low bit rate condition. A quantizer 12 obtains a quantized spectral sequence from a frequency spectral sequence. An integer transformer 13 obtains a unified quantized spectral sequence by obtaining, by a bijective transformation, a transformed integer for each of the sets, each being made up of integer values, obtained from the quantized spectral sequence. An integer encoder 15 obtains an integer code by encoding the unified quantized spectral sequence using a bit assignment sequence. An object-to-be-encoded estimator 18 obtains an estimated unified spectral sequence from the frequency spectral sequence by a transformation which is performed by the integer transformer 13 or a transformation that approximates the magnitude relationship between values before and after the above transformation. A bit assigner 14 obtains a bit assignment sequence and a bit assignment code from the estimated unified spectral sequence.

Abstract: A system is provided for electronic data compression by automated time-dependent compression algorithm. In particular, the system may track instances in which a particular dataset is used, copied, or accessed over time. For certain datasets (e.g., datasets that have not been accessed for a threshold amount of time), the system may use a time-based compression algorithm that progressively removes the least significant bits of such datasets as time passes. The compression of the datasets may continue until the system detects that further compression would cause the dataset to be unreadable or unrecoverable. In this way, the system may minimize the computing resources allocated to storing datasets that are not frequently accessed.

Abstract: A method for encoding a high frequency signal includes determining a signal type of a high frequency signal of a current frame, smoothing and scaling time envelopes of the high frequency signal of the current frame and obtaining time envelopes of the high frequency signal of the current frame that require to be encoded when the high frequency signal of the current frame is a non-transient signal and a high frequency signal of the previous frame is a transient signal, and quantizing and encoding the time envelopes of the high frequency signal of the current frame that require to be encoded, and frequency information and signal type information of the high frequency signal of the current frame.

Abstract: A digital audio quality monitoring device uses a deep neural network (DNN) to provide accurate estimates of signal-to-noise ratio (SNR) from a limited set of features extracted from incoming audio. Some embodiments improve the SNR estimate accuracy by selecting a DNN model from a plurality of available models based on a codec used to compress/decompress the incoming audio. Each model has been trained on audio compressed/decompressed by a codec associated with the model, and the monitoring device selects the model associated with the codec used to compress/decompress the incoming audio. Other embodiments are also provided.

Abstract: A display device includes a display panel, a first storage device, a second storage device and a control device. The first storage device is disposed on the display panel and stores a first firmware. The second storage device stores a second firmware. The control device is coupled to the first storage device and the second storage device. The control device reads the first firmware and uses the first firmware to update the second firmware, or receives a third firmware transmitted by a host terminal device and uses the third firmware to update the second firmware. The control device executes the updated second firmware to generate an image signal corresponding to the updated second firmware to the display panel. Therefore, the efficiency of updating the firmware of the display device may be effectively increased, and the convenience of use is increased.

Abstract: A method for extracting big beat information from music beat points includes: acquiring at least one candidate beat cycle according to beat points of music; generating at least one beat point sequence corresponding to the candidate beat cycle and calculating a weight of the beat point sequence, wherein two adjacent beat points in the beat point sequence take the candidate beat cycle as an interval; selecting a beat point sequence with a maximum weight from the at least one beat point sequence of the candidate beat cycle as a maximum weight sequence of the candidate beat cycle; and acquiring the big beat information according to the maximum weight sequence of the at least one candidate beat cycle and the corresponding weight.

Abstract: An apparatus for generating one or more audio output channels is provided. The apparatus includes a parameter processor for calculating output channel mixing information and a downmix processor for generating the one or more audio output channels. The downmix processor is configured to receive an audio transport signal including one or more audio transport channels, wherein two or more audio object signals are mixed within the audio transport signal, and wherein the number of the one or more audio transport channels is smaller than the number of the two or more audio object signals. The audio transport signal depends on a first mixing rule and on a second mixing rule. The first mixing rule indicates how to mix the two or more audio object signals to obtain a plurality of premixed channels. Moreover, the second mixing rule indicates how to mix the plurality of premixed channels.

Abstract: A high-band encoding/decoding method and device for bandwidth extension are provided. A high-band encoding method comprising the steps of: generating sub band-specific bit allocation information on the basis of a low-band envelope; determining, on the basis of the sub band-specific bit allocation information, the sub band requiring an envelope update in a high band; and generating, for the determined sub band, refinement data relating to the envelope update. A high-band decoding method comprising the steps of: generating sub band-specific bit allocation information on the basis of a low-band envelope; determining, on the basis of the sub band-specific bit allocation information, the sub band requiring an envelope update in a high band; and decoding, for the determined sub band, refinement data relating to the envelope update, thereby updating the envelope.

Abstract: A sound source estimation unit (101) estimates, in a space as a target of sparse sound field decomposition, an area where a sound source is present at second granularity that is coarser than first granularity of a position where a sound source is assumed to be present in the sparse sound field decomposition. A sparse sound field decomposition unit (102) decomposes an acoustic signal observed by a microphone array into a sound source signal and an ambient noise signal by performing a sparse sound field decomposition process at the first granularity for the acoustic signal in the area at the second granularity where the sound source is estimated to be present in the space.

Abstract: A method comprising: receiving at least two audio channel signals; determining, for a first frame, at least two parameters representing a difference between the at least two channel audio signals; scalar quantising the at least two parameters to generate at least two index values; adaptively encoding an initial scalar quantized parameter of the at least two parameters; determining whether the initial scalar quantized parameter has a value different from a predetermined value; adaptively encoding any unencoded scalar quantized parameters where the initial scalar quantized parameter has a value different from the predetermined value; determining whether all of the at least two scalar quantized parameters have values equal to the predetermined value where the initial scalar quantized parameter has a value equal to the predetermined value; adaptively encoding any unencoded scalar quantized parameters and generating an indicator that an output is one of fixed or variable rate coding where the initial scalar quanti

Abstract: A method and apparatus for signaling and receiving a video signal for processing is disclosed. Methods for determining a most probable mode for predicting a current prediction block are provided as well as new methods for grouping intra prediction modes into prediction groups. Methods for predicting a prediction block of video data as well as signaling intra prediction modes are also provided.

Abstract: An apparatus for generating one or more audio output channels is provided. The apparatus includes a parameter processor for calculating output channel mixing information and a downmix processor for generating the one or more audio output channels. The downmix processor is configured to receive an audio transport signal including one or more audio transport channels, wherein two or more audio object signals are mixed within the audio transport signal, and wherein the number of the one or more audio transport channels is smaller than the number of the two or more audio object signals. The audio transport signal depends on a first mixing rule and on a second mixing rule. The first mixing rule indicates how to mix the two or more audio object signals to obtain a plurality of premixed channels. Moreover, the second mixing rule indicates how to mix the plurality of premixed channels.

Abstract: To address the above technical problem, a method of processing an audio signal includes: receiving an audio bitstream that is encoded by using MPEG Surround 212 (MPS212); generating an internal channel signal with respect to one channel pair element (CPE) based on the received audio bitstream and rendering parameters with respect to MPS212 output channels defined in a format converter; allocating a group of internal channels based on an output channel location of a core codec; and generating stereo channel output signals based on the generated internal channel signal and the allocated group of the internal channels.

Abstract: A method of neighbor cell detection performed in a communication device is disclosed. The method comprises selecting at least one subband-time interval resource for processing, wherein each subband is part of a system frequency bandwidth and each time interval is a part of a synchronization signal repetition time period, recording one or more radio samples associated with the at least one subband-time interval resource, executing cell detection on the recorded radio samples, and repeating the selecting, recording and executing until a stop criterion is fulfilled. A corresponding communication device computer program and computer program product are also provided.

Abstract: An audio decoder for providing a decoded audio information on the basis of an encoded audio information includes an error concealment configured to provide an error concealment audio information for concealing a loss of an audio frame following an audio frame encoded in a frequency domain representation using a time domain excitation signal.

Abstract: The invention provides methods and devices for stereo encoding and decoding using complex prediction in the frequency domain. In one embodiment, a decoding method, for obtaining an output stereo signal from an input stereo signal encoded by complex prediction coding and comprising first frequency-domain representations of two input channels, comprises the upmixing steps of: (i) computing a second frequency-domain representation of a first input channel; and (ii) computing an output channel on the basis of the first and second frequency-domain representations of the first input channel, the first frequency-domain representation of the second input channel and a complex prediction coefficient. The upmixing can be suspended responsive to control data.

Abstract: An audio encoder for encoding segments of coefficients, the segments of coefficients representing different time or frequency resolutions of a sampled audio signal, the audio encoder including a processor for deriving a coding context for a currently encoded coefficient of a current segment based on a previously encoded coefficient of a previous segment, the previously encoded coefficient representing a different time or frequency resolution than the currently encoded coefficient. The audio encoder further includes an entropy encoder for entropy encoding the current coefficient based on the coding context to obtain an encoded audio stream.

Abstract: An apparatus for processing an audio signal including a sequence of blocks of spectral values, includes: a processor for calculating an aliasing-affected signal using at least one first modification value for a first block of the sequence of blocks and using at least one different second modification value for a second block of the sequence of blocks and for estimating an aliasing-error signal representing an aliasing-error in the aliasing-affected signal; and a combiner for combining the aliasing-affected signal and the aliasing-error signal such that a processed signal obtained by the combining is an aliasing-reduced or aliasing-free signal.

Abstract: A method for controlling a three-dimensional multi-layer speaker arrangement having a plurality of speakers arranged in spaced layers. The method includes: providing information for a sound to be played back from a 3D source position assigned to the sound, wherein the source position is defined with respect to a reference point (RP) within the multi-layer speaker arrangement, extracting a 3D source position (SPXY) from the source position and calculating layer specific speaker coefficients using a 2D calculator to position the sound two dimensional source position, and feeding a vertical pan or 3D source position into a multilayer calculator for obtaining a layer gain factor for each layer for obtaining speaker coefficients used as individual gains enabling the speakers to play back the sound.

Abstract: Audio processing methods may involve receiving audio data corresponding to a plurality of audio channels. The audio data may include a frequency domain representation corresponding to filterbank coefficients of an audio encoding or processing system. A decorrelation process may be performed with the same filterbank coefficients used by the audio encoding or processing system. The decorrelation process may be performed without converting coefficients of the frequency domain representation to another frequency domain or time domain representation. The decorrelation process may involve selective or signal-adaptive decorrelation of specific channels and/or specific frequency bands. The decorrelation process may involve applying a decorrelation filter to a portion of the received audio data to produce filtered audio data. The decorrelation process may involve using a non-hierarchal mixer to combine a direct portion of the received audio data with the filtered audio data according to spatial parameters.

Abstract: A plurality of candidates of prediction coefficients of a channel signal out of a plurality of channel signals are extracted from a code book storing a plurality of prediction coefficients, each candidate having a predictive error falling within a specific range of predictive error of predictive encoding of two other channel signals, and a prediction coefficient is selected from the extracted candidates as a result of the predictive encoding in accordance with a specific data embedding rule and embedding embed target data into the selected prediction coefficient.

Abstract: A decoder for generating an audio output signal having one or more audio output channels from a downmix signal having a plurality of time-domain downmix samples is provided. The downmix signal encodes two or more audio object signals. The decoder has a window-sequence generator for determining a plurality of analysis windows, each having a plurality of time-domain downmix samples of the downmix signal and a window length indicating the number of the time-domain downmix samples. Moreover, the decoder has a t/f-analysis module for transforming the plurality of time-domain downmix samples of each analysis window from a time-domain to a time-frequency domain depending on the window length of said analysis window, to obtain a transformed downmix. Furthermore, the decoder has an un-mixing unit for un-mixing the transformed downmix based on parametric side information on the two or more audio object signals to obtain the audio output signal. Moreover, an encoder is provided.

Abstract: An apparatus for generating a representation of a bandwidth-extended signal on the basis of an input signal representation includes a phase vocoder configured to obtain values of a spectral domain representation of a first patch of the bandwidth-extended signal on the basis of the input signal representation. The apparatus also includes a value copier configured to copy a set of values of the spectral domain representation of the first patch, which values are provided by the phase vocoder, to obtain a set of values of a spectral domain representation of a second patch, wherein the second patch is associated with higher frequencies than the first patch. The apparatus is configured to obtain the representation of the bandwidth-extended signal using the values of the spectral domain representation of the first patch and the values of the spectral domain representation of the second patch.

Abstract: An apparatus for generating one or more audio output channels is provided. The apparatus includes a parameter processor for calculating output channel mixing information and a downmix processor for generating the one or more audio output channels. The downmix processor is configured to receive an audio transport signal including one or more audio transport channels, wherein two or more audio object signals are mixed within the audio transport signal, and wherein the number of the one or more audio transport channels is smaller than the number of the two or more audio object signals. The audio transport signal depends on a first mixing rule and on a second mixing rule. The first mixing rule indicates how to mix the two or more audio object signals to obtain a plurality of premixed channels. Moreover, the second mixing rule indicates how to mix the plurality of premixed channels.

Abstract: A method for controlling a three-dimensional multi-layer speaker arrangement having a plurality of speakers arranged in spaced layers. The method includes: providing information for a sound to be played back from a 3D source position assigned to the sound, wherein the source position is defined with respect to a reference point (RP) within the multi-layer speaker arrangement, extracting a 2D source position (SPXY) from the source position and calculating layer specific speaker coefficients using a 2D calculator to position the sound two dimensional source position, and feeding a vertical pan or 3D source position into a multilayer calculator for obtaining a layer gain factor for each layer for obtaining speaker coefficients used as individual gains enabling the speakers to play back the sound.

Abstract: A browser efficiently extracts media from a video presented through a web application having a different domain from a media server providing the web application while complying with cross-origin resource restrictions. A video portion is selected, and a header portion of the video is identified. A truncated video portion, which includes the selected video portion and a video portion contiguous to the selected video portion, is loaded into memory. A truncated video file is generated including the truncated video portion and a truncated video header. The truncated video file is encoded as a video tag included in the web application, and the media corresponding to the selected video portion is extracted by transcoding the truncated video file in the video tag. A server may also extract the media by loading the truncated video portion, generating the truncated video file, and transcoding the truncated video file to obtain the media.

Abstract: A stereo decoding method and apparatus are disclosed. The method includes: restoring a monophonic signal from a received code stream through decoding; restoring an interchannel level difference, a group delay, and a group phase from the received code stream through decoding; and processing the monophonic signal according to the interchannel level difference, group delay, and group phase to obtain a first channel signal and a second channel signal. According to the stereo decoding method and apparatus provided in embodiments of the present invention, the first and second channel signals are obtained according to the monophonic signal, ILD, group delay, and group phase by referring to not only the ILD but also the group delay and group phase, thereby yielding favorable stereo sound field effect for the obtained first and second channel signals.

Abstract: The present invention includes various embodiments of a mobile app remote virtualization system and process that enables users of remote client devices to control mobile apps running in a host server environment. The resulting user experience is practically equivalent to running native mobile apps, even when such mobile apps require access to local client device resources, as well as when native versions of such mobile apps do not exist for the user's client device. The functionality afforded by the mobile app remote virtualization system and process of the present invention enables a variety of novel scenarios and “use cases” that have not previously been available to mobile device users.

Abstract: A stereo decoding method and apparatus are disclosed. The method includes: restoring a monophonic signal from a received code stream through decoding; restoring an interchannel level difference, a group delay, and a group phase from the received code stream through decoding; and processing the monophonic signal according to the interchannel level difference, group delay, and group phase to obtain a first channel signal and a second channel signal. According to the stereo decoding method and apparatus provided in embodiments of the present invention, the first and second channel signals are obtained according to the monophonic signal, ILD, group delay, and group phase by referring to not only the ILD but also the group delay and group phase, thereby yielding favorable stereo sound field effect for the obtained first and second channel signals.

Abstract: Various embodiments relate to a system and method for decorrelating an audio signal with a hybrid filter. The hybrid filter is generated by first generating a decorrelation filter. A frequency-dependent warping is applied to the decorrelation filter. The warped decorrelation filter is then mixed with a carrier filter to generate the hybrid filter. The carrier filter may include filters for spatial processing of an audio signal, filters for upmixing an audio signal, and/or filters for downmixing an audio signal.

Abstract: A stereo encoding method, a stereo encoding device, and an encoder are provided. The stereo encoding method includes: obtaining a left channel energy relation coefficient and a right channel energy relation coefficient; obtaining a left energy sum and a right energy sum respectively; performing cross correlation between sub-bands of a first monophonic signal at a wave trough and sub-bands of the left channel signal according to the left channel energy relation coefficient, and performing cross correlation between sub-bands of the first monophonic signal at the wave trough and sub-bands of the right channel signal according to the right channel energy relation coefficient; obtaining a scaling factor by using the left energy sum, the right energy sum, and cross correlation results; and encoding the stereo left and right channel signals according to the scaling factor.

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 decoding apparatus (10) is disclosed which includes: a storing means (11) for storing encoded audio signals including multi-channel audio signals; a transforming means (40) for transforming the encoded audio signals to generate transform block-based audio signals in a time domain; a window processing means (41) for multiplying the transform block-based audio signals by a product of a mixture ratio of the audio signals and a first window function, the product being a second window function; a synthesizing means (43) for overlapping the multiplied transform block-based audio signals to synthesize audio signals of respective channels; and a mixing means (14) for mixing audio signals of the respective channels between the channels to generate a downmixed audio signal. Furthermore, an encoding apparatus is also disclosed which downmixes the multi-channel audio signals, encodes the downmixed audio signals, and generates the encoded, downmixed audio signals.

Abstract: An audio signal decoder has a time warp contour calculator, a time warp contour data rescaler and a warp decoder. The time warp contour calculator is configured to generate time warp contour data repeatedly restarting from a predetermined time warp contour start value, based on time warp contour evolution information describing a temporal evolution of the time warp contour. The time warp contour data rescaler is configured to rescale at least a portion of the time warp contour data such that a discontinuity at a restart is avoided, reduced or eliminated in a rescaled version of the time warp contour. The warp decoder is configured to provide the decoded audio signal representation, based on an encoded audio signal representation and using the rescaled version of the time warp contour.

Abstract: Systems and methods are presented for efficient cross-fading (or other multiple clip processing) of compressed domain information streams on a user or client device, such as a telephone, tablet, computer or MP3 player, or any consumer device with audio playback. Exemplary implementation systems may provide cross-fade between AAC/Enhanced AAC Plus (EAACPIus) information streams or between MP3 information streams or even between information streams of unmatched formats (e.g. AAC to MP3 or MP3 to AAC). Furthermore, these systems are distinguished by the fact that cross-fade is directly applied to the compressed bitstreams so that a single decode operation may be performed on the resulting bitstream. Moreover, using the described methods, similar cross fade in the compressed domain between information streams utilizing other formats of compression, such as, for example, MP2, AC-3, PAC, etc. can also be advantageously implemented.

Abstract: An audio signal decoder for providing a decoded multi-channel audio signal representation on the basis of an encoded multi-channel audio signal representation has a time warp decoder configured to selectively use individual audio channel specific time warp contours or a joint multi-channel time warp contour for a reconstruction of a plurality of audio channels represented by the encoded multi-channel audio signal representation.

Abstract: A method for detecting barge-in in a speech dialog system comprising determining whether a speech prompt is output by the speech dialog system, and detecting whether speech activity is present in an input signal based on a time-varying sensitivity threshold of a speech activity detector and/or based on speaker information, where the sensitivity threshold is increased if output of a speech prompt is determined and decreased if no output of a speech prompt is determined. If speech activity is detected in the input signal, the speech prompt may be interrupted or faded out. A speech dialog system configured to detect barge-in is also disclosed.

Abstract: A method and system for performing sample rate conversion is provided. The method may include configuring a system to convert a sample rate of a first audio channel of a plurality of audio channels to produce a first audio stream of samples. The system may be dynamically reconfigured to convert a sample rate of a second of the plurality of audio channels to produce a second audio stream of samples, wherein the first and second audio streams are output from the system at the same time. The method may further include arbitrating between request for additional data from the first and second audio stream of samples, where processing of the first channel is suspended when the request corresponds to a second channel that is of higher priority.

Abstract: Various techniques are disclosed for reducing artifacts generated by time compression. by adapting the time compression based on the state of the received audio. The amount of time compression may be bounded based on audio characteristics. Another feature provides a way of determining the most correlated portions of segments of audio. Voiced speech may be distinguished from unvoiced speech. Another feature provides a way of distinguishing between silence, voiced speech, and unvoiced speech. Time compression may be adapted during periods of lengthy silence. Another feature allows for reducing time compression during sensitive portions of the received audio. One or more of these features may be present in different embodiments.

Abstract: For a bandwidth extension of an audio signal, in a signal spreader the audio signal is temporally spread by a spread factor greater than 1. The temporally spread audio signal is then supplied to a demicator to decimate the temporally spread version by a decimation factor matched to the spread factor. The band generated by this decimation operation is extracted and distorted, and finally combined with the audio signal to obtain a bandwidth extended audio signal. A phase vocoder in the filterbank implementation or transformation implementation may be used for signal spreading.

Abstract: Example systems and methods concern a sub-block parser that is configured with a variable sized window whose size varies as a function of the actual or expected entropy of data to be parsed by the sub-block parser. Example systems and methods also concern a sub-block parser configured to compress a data sequence to be parsed before parsing the data sequence. One example method facilitates either actually changing the window size or effectively changing the window size by manipulating the data before it is parsed. The example method includes selectively reconfiguring a data set to be parsed by a data-dependent parser based, at least in part, on the entropy level of the data set, selectively reconfiguring the data-dependent parser, based, at least in part, on the entropy level of the data set, and parsing the data set.

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: The present disclosure is directed towards methods and systems for virtualizing audio hardware for one or more virtual machines. A control virtual machine (VM) may translate a first stream of audio functions calls from a first VM hosted by a hypervisor. The translated first stream of audio function calls may be destined for a sound card of the computing device executing the hypervisor. The control VM may detect a second stream of audio functions calls from a second VM hosted by the hypervisor. The control VM may translate the second stream of audio functions calls from the second VM. The control VM may further merge the translated first stream of audio function calls and the translated second stream of the audio function calls in response to the detected second stream. The control VM may transmit the merged stream of audio function calls to the sound card.

Abstract: A method is provided for hierarchical coding of a digital audio signal comprising, for a current frame of the input signal: a core coding, delivering a scalar quantization index for each sample of the current frame and at least one enhancement coding delivering indices of scalar quantization for each coded sample of an enhancement signal. The enhancement coding comprises a step of obtaining a filter for shaping the coding noise used to determine a target signal and in that the indices of scalar quantization of said enhancement signal are determined by minimizing the error between a set of possible values of scalar quantization and said target signal. The coding method can also comprise a shaping of the coding noise for the core bitrate coding. A coder implementing the coding method is also provided.

Abstract: Systems (1600) and methods (1500) for frame synchronization. The methods involve: extracting bit sequences S0 and S1 from a Bit Stream (“BS”) of a Data Burst (“DB”); decoding S0 and S1 to obtain decoded bit sequences S?0 and S?1; using S?0 and S?1 to determine Bit Error Rate (“BER”) estimates (516, 518); combining the BER estimates to obtain a combined BER estimate; modifying S0 and S1 so that each includes at least one bit of BS which is not included in its current set of bits and so that it is absent of at least one of the bits in the current set of bits; iteratively repeating the decoding, using, combining and modifying steps to obtain more combined BER estimates; analyzing the combined BER estimates to identify a minimum combined BER estimate; and using the minimum combined BER estimate to determine a location of a vocoder voice frame within DB.

Abstract: A method comprising receiving at a user equipment encrypted content. The content is stored in said user equipment in an encrypted form. At least one key for decryption of said stored encrypted content is stored in the user equipment.