Abstract: A method for generating a bitstream indicative of an object based audio program is described. The bitstream comprises a sequence of containers. A first container of the sequence of containers comprises a plurality of substream entities for a plurality of substreams of the object based audio program and a presentation section. The method comprises determining a set of object channels. The method further comprises providing a set of object related metadata for the set of object channels. In addition, the method comprises inserting a first set of object channel frames and a first set of object related metadata frames into a respective set of substream entities of the first container. Furthermore, the method comprises inserting presentation data into the presentation section.

Abstract: A method for audio rendering by an apparatus comprising at least one audio rendering device, the method comprising: extracting (S10) a plurality of frequency band components from an input audio signal; determining (S15) from the device frequency response and the plurality of extracted frequency band components at least one indicator representative of masking frequencies energy, masking frequencies corresponding to frequency bands that are above a frequency threshold; determining at least one correction factor (S16) from the indicator representative of masking frequencies energy; e) for each frequency band, determining a second acoustic level threshold (S20) by modifying (S17) with the correction factor a predetermined first acoustic level threshold associated with said frequency band, and determining a reduction gain from a comparison (S30) between an acoustic level of the extracted frequency band and the second acoustic level threshold associated to said extracted frequency band, and applying (S40) the reduc

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 a multichannel audio signal has a plurality of channel signals. The apparatus performs a time scale modulation of the multichannel audio signal and has a phase adaptor and a separator. The phase adaptor provides a processed signal by modifying a phase of a signal based on a combination of the channel signals. The separator provides separated signals based on the processed signal. A corresponding method is provided.

Abstract: Embodiments are provided for updating a playlist that has been added to a playback queue in response to changes to the playback queue. The playback queue may be associated with a zone of a network media system such that items in the playback queue are to be rendered by the zone. The playlist may include one or more items playable by the zone, and may be stored separately from where the playback queue is maintained. Embodiments are also provided for updating a playback queue in response to modifications to a playlist included in the playback queue.

Abstract: Provided are an audio data processing method and apparatus, a device and a storage medium. The method includes: acquiring audio data to be processed and a variable-speed rate of at least one audio frame in the audio data; sequentially using the at least one audio frame as a current audio frame to be processed, and converting the current audio frame to a frequency domain; determining a target phase signal of the current audio frame according to a variable-speed rate of the current audio frame and a variable-speed rate of a previous audio frame; and performing, according to the target phase signal, time domain conversion on the current audio frame converted to the frequency domain to obtain a processed current audio frame.

Abstract: Systems and methods for generating a graphical representation of audio signal data during time compression or expansion are provided. The system may include a processor that performs a method including displaying a waveform during audio-signal playback at a first speed by scrolling the waveform from aright portion of a display to a left portion of the display. The method includes receiving a command to increase or decrease the audio-signal playback speed and horizontally expanding or horizontally contracting the waveform in response to receiving the command to increase or decrease the audio-signal playback speed.

Abstract: Utilizing different data compression algorithms based on characteristics of a storage system, including: selecting, in dependence upon a priority for conserving processing resources or storage resources in a storage system, a data compression algorithm to utilize to compress data; detecting that at least one of an amount of processing resources available in the storage system or the amount of space available to store additional data in the storage system has changed; and responsive to detecting that at least one of the amount of processing resources available in the storage system or the amount of space available to store additional data in the storage system has changed, selecting a different data compression algorithm to utilize to compress data.

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: 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 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: 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 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: 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: 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.