Abstract: Systems and methods augment a target media with a plurality of source media. The target media and source media are processed to form time frequency distributions (TFDs). Target features are extracted from the associated TFD and source features are extracted from each of the associated source TFDs. The target features are segmented into temporal portions that are compared with each of the plurality of source features to determine one or more matched source features having nearest matches to the target feature segments. Portions of the source media associated with the matched source features are mixed with the target media to form an augmented target media, wherein the mixing is based upon a probabilistic mixing algorithm that uses a distance between the matched target feature and source features to define an amplitude of each portion of the source media.

Abstract: According to the invention, a device for post-processing at least one channel signal of a plurality of channel signals of a multi-channel signal is described, the at least one channel signal being generated from a decoded downmix signal by a low-bit-rate audio coding/decoding system, the device comprising: a receiver for receiving the at least one channel signal generated from the decoded downmix signal, a time envelope of the decoded downmix signal, an interchannel time difference between the channel signal and the downmix signal, and a classification indication indicating a transient type of the downmix signal; and a post-processor for post-processing the at least one channel signal based on the time envelope of the decoded downmix signal weighted by a respective weighting factor and in dependence on the classification indication and the interchannel time difference.

Abstract: Audio characteristics of audio data corresponding to a plurality of audio channels may be determined. The audio characteristics may include spatial parameter data. Decorrelation filtering processes for the audio data may be based, at least in part, on the audio characteristics. The decorrelation filtering processes may cause a specific inter-decorrelation signal coherence (“IDC”) between channel-specific decorrelation signals for at least one pair of channels. The channel-specific decorrelation signals may be received and/or determined. Inter-channel coherence (“ICC”) between a plurality of audio channel pairs may be controlled. Controlling ICC may involve at receiving an ICC value and/or determining an ICC value based, at least partially, on the spatial parameter data. A set of IDC values may be based, at least partially, on the set of ICC values. A set of channel-specific decorrelation signals, corresponding with the set of IDC values, may be synthesized by performing operations on the filtered audio data.

Abstract: Dynamically adjusting stream quality level is disclosed. In some embodiments, playing media content includes playing a first stream having a first quality level of the media content, determining that a different available quality level of the media content would result in improved playback performance, and switching to playing a second stream having a second quality level of the media content.

Abstract: A method of encoding a multi-channel 3-dimensional (3D) audio signal mixed with a multi-channel 3D object signal is provided. The method includes: obtaining a location parameter indicating a virtual location of the multi-channel 3D object signal on a multi-channel speaker layout based on a gain value of the multi-channel 3D object signal for each channel; and encoding the multi-channel 3D audio signal and the location parameter.

Abstract: An apparatus for decoding an audio signal and method thereof are disclosed. The present invention includes receiving the audio signal and spatial information, identifying a type of modified spatial information, generating the modified spatial information using the spatial information, and decoding the audio signal using the modified spatial information, wherein the type of the modified spatial information includes at least one of partial spatial information, combined spatial information and expanded spatial information. Accordingly, an audio signal can be decoded into a configuration different from a configuration decided by an encoding apparatus. Even if the number of speakers is smaller or greater than that of multi-channels before execution of downmixing, it is able to generate output channels having the number equal to that of the speakers from a downmix audio signal.

Abstract: An image decoding method which can improve both image quality and coding efficiency is an image decoding method for decoding a coded stream which includes a plurality of processing units and a header for the processing units, the coded stream being generated by coding a moving picture, the processing units including at least one processing unit layered to be split into a plurality of smaller processing units, the image decoding method including specifying a hierarchical layer having a processing unit in which a parameter necessary for decoding is stored, by parsing hierarchy depth information stored in the header, and decoding the processing unit using the parameter stored in the processing unit located at the specified hierarchical layer.

Abstract: The present document relates to audio source coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), as well as to digital effect processors, e.g. exciters, where generation of harmonic distortion add brightness to the processed signal, and to time stretchers where a signal duration is prolonged with maintained spectral content. A system and method configured to generate a time stretched and/or frequency transposed signal from an input signal is described. The system comprises an analysis filterbank (101) configured to provide an analysis subband signal from the input signal; wherein the analysis subband signal comprises a plurality of complex valued analysis samples, each having a phase and a magnitude. Furthermore, the system comprises a subband processing unit (102) configured to determine a synthesis subband signal from the analysis subband signal using a subband transposition factor Q and a subband stretch factor S.

Abstract: An audio decoder provides a combination of decoding components including components implementing base band decoding, spectral peak decoding, frequency extension decoding and channel extension decoding techniques. The audio decoder decodes a compressed bitstream structured by a bitstream syntax scheme to permit the various decoding components to extract the appropriate parameters for their respective decoding technique.

Abstract: On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters (a) from the bitstream, where 1?m<n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC2) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.

Abstract: Wind noise reduction is described for audio signals received in a device. In one embodiment, an audio signal is decomposed into a plurality of sub-bands, the audio signal including wind noise, a first sub-band of the plurality of sub-bands low-pass filtered, wind noise is removed from the first sub band and the first sub-band is combined with the other sub-bands after removing wind noise.

Abstract: Embodiments of packet loss concealment for phone-to-hearing-aid streaming are generally described herein. A method for packet loss concealment can include receiving a first frame at a hearing assistance device, determining, at the hearing assistance device, that a second frame was not received within a predetermined time, and determining a first set of sequential samples that match the first frame. The method can include cross-fading the first frame and the first set of sequential samples to create a first cross-faded frame and extrapolating a third frame to replace the second frame using the first set of sequential samples and an autoregressive model.

Abstract: The present invention discloses a speech/audio signal processing method and apparatus.

Abstract: The embodiments of the present invention improves conventional attenuation schemes by replacing constant attenuation with an adaptive attenuation scheme that allows more aggressive attenuation, without introducing audible change of signal frequency characteristics.

Abstract: The invention introduces a new concept for hierarchical coding of HOA content. A method for encoding a hierarchical audio bitstream comprises rendering a HOA input signal to surround sound, encoding the surround sound for a base layer output signal, decoding the encoded surround sound to obtain a reconstructed surround sound signal, performing dimensionality reduction on the received HOA input signal, calculating a residual between the dimensionality-reduced HOA signal and the reconstructed surround sound signal, encoding the residual signal, and multiplexing structural information about the HOA input signal, the encoded residuals and the encoded surround sound into a bitstream to obtain a hierarchical audio bitstream.

Abstract: An encoder and an encoding method for a multi-channel signal, and a decoder and a decoding method for a multi-channel signal are disclosed. A multi-channel signal may be efficiently processed by consecutive downmixing or upmixing.

Abstract: The invention relates to a codec and a signal classifier and methods therein for signal classification and selection of a coding mode based on audio signal characteristics. A method embodiment to be performed by a decoder comprises, for a frame m: determining a stability value D(m) based on a difference, in a transform domain, between a range of a spectral envelope of frame m and a corresponding range of a spectral envelope of an adjacent frame m?1. Each such range comprises a set of quantized spectral envelope values related to the energy in spectral bands of a segment of the audio signal. The method further comprises selecting a decoding mode, out of a plurality of decoding modes, based on the stability value D(m); and applying the selected decoding mode.

Abstract: Sound-producing devices such as headphone sets or loudspeakers are disclosed that can removably mate with an external interface. The external interface may plug into a port on an exterior of the sound-producing device. The external interface includes conversion circuitry that is powered by a power source in the sound-producing device. The external interface receives a first audio-encoded signal from a source device such as a mobile phone, tablet computer, or the like, and converts such signal to a conditioned signal which is compatible with the sound-producing device. Connection between the source device and the external interface may be a first type of wired or wireless connection. The sound-producing device may operate with other source devices that produce different audio-encoded signals by replacing the external interface with a different external interface whose conversion circuitry converts the different audio-encoded signal to the same conditioned signal.

Abstract: An apparatus comprising: a channel analyzer configured to determine for a first frame of at least one audio signal a set of first frame audio signal multi-channel parameters; a multichannel difference selector configured to select for the first frame groups of elements of the set of first frame audio signal multi-channel parameters based on a value associated with the first frame; and a multichannel parameter encoder configured to generate an encoded first frame audio signal multi-channel parameter based on the selected groups of elements of the set of first frame audio signal multi-channel parameters.

Abstract: The present disclosure discloses a method, a device, and a mobile terminal for performing automatic recording. The method comprises: obtaining content of an ongoing call; performing a recording operation on the content of the ongoing call, and saving a recording of the content of the ongoing call in a predefined region of a system ring buffer. The technical solutions provided in the present disclosure solve the problems of troublesome manual recording adjustments and excessive waste of system resources arising from recording valueless call content, thereby improving user experience.

Abstract: A network device is configured to receive information regarding a group of content streams and determine a buffer size for each of the content streams. The network device is further configured to receive the content streams from one or more encoding devices. The network device is further configured to buffer an amount of each of the content streams based on the respective buffer size. The network device is further configured to send a first content stream to a user device. The network device is further configured to determine that the first content stream has a quality of experience issue and send the second content stream to the user device.

Abstract: In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

Abstract: In certain embodiments, a web services system receives a request to provision a device, such as a telephone, as an authentication device. The web services system initiates display of an image communicating a key to allow the telephone to capture the image and to send key information associated with the key. The web services system receives the key and determines that the key information is valid. In response to the determination, the web services system sends a seed to the telephone to provision the telephone to be an authentication device. The telephone can use the seed to generate one-time passcodes to access a service of the web services system.

Abstract: In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

Abstract: The present invention extends to methods, systems, and computer program products for using cinematic techniques to present data. Embodiments of the invention can be used to infer and generate cinematic techniques or combinations thereof based on a model and user action. Cinematic techniques can be used to meet the data exploration and analysis requirements of a user. As such, embodiments of the invention permit users (including non-programmers) to employ cinematic techniques (possibly in combination with other techniques) to gain insights into their data and also convey appropriate emotional messages.

Abstract: A decoding system (100) reconstructs an n-channel audio signal on the basis of an input signal (A) representing the audio signal either by parametric coding or as n discretely coded channels. Parametric decoding proceeds on the basis of a core signal and mixing parameters (a) controlling a spatial synthesis stage (150), which is supplied with a downmix signal from a downmix stage (140). A selector (170) controls the components of the decoding system, in steady-state parametric and discrete decoding mode and transitions between these. The downmix stage realizes a projection on the downmix signal based on an n-channel signal, either an n-channel input signal or a core signal padded with neutral values. The downmix stage is active in each time frame in which the input signal represents the audio signal by parametric coding and in at least the first time frame after the last time frame in each episode of parametrically coded time frames.

Abstract: Media content presentation systems and methods are operable to present a media content event comprising a video portion and an associated audio portion to a user. An exemplary embodiment compares a detected level of background noise with a first background noise level threshold; increases output volume of the presented audio portion of the media content event when the level of background noise exceeds the first background noise level threshold; determines text when the level of background noise exceeds a second background noise level threshold; determines sentences from the determined text; and presents on a display the beginning portion of a current sentence in the dialog of the text when the increasing level of background noise exceeds a maximum background noise level threshold.

Abstract: The embodiments of the present invention improves conventional attenuation schemes by replacing constant attenuation with an adaptive attenuation scheme that allows more aggressive attenuation, without introducing audible change of signal frequency characteristics.

Abstract: Method and apparatus for processing audio signals are provided. The method for decoding an audio signal includes extracting a downmix signal and spatial information from a received audio signal, generating surround converting information using the spatial information and rendering the downmix signal to generate a pseudo-surround signal in a previously set rendering domain, using the surround converting information. The apparatus for decoding an audio signal includes a demultiplexing part extracting a downmix signal and spatial information from a received audio signal, an information converting part generating surround converting information using the spatial information and a pseudo-surround generating part rendering the downmix signal to generate a pseudo-surround signal in a previous set rendering domain, using the surround converting information.

Abstract: Conventional audio compression technologies perform a standardized signal transformation, independent of the type of the content. Multi-channel signals are decomposed into their signal components, subsequently quantized and encoded. This is disadvantageous due to lack of knowledge on the characteristics of scene composition, especially for e.g. multi-channel audio or Higher-Order Ambisonics (HOA) content. An improved method for encoding pre-processed audio data comprises encoding the pre-processed audio data, and encoding auxiliary data that indicate the particular audio pre-processing. An improved method for decoding encoded audio data comprises determining that the encoded audio data had been pre-processed before encoding, decoding the audio data, extracting from received data information about the pre-processing, and post-processing the decoded audio data according to the extracted pre-processing information.

Abstract: Provided is an audio encoding method. The audio encoding method includes: acquiring envelopes based on a predetermined sub-band for an audio spectrum; quantizing the envelopes based on the predetermined sub-band; and obtaining a difference value between quantized envelopes for adjacent sub-bands and lossless encoding a difference value of a current sub-band by using a difference value of a previous sub-band as a context. Accordingly, the number of bits required to encode envelope information of an audio spectrum may be reduced in a limited bit range, thereby increasing the number of bits required to encode an actual spectral component.

Abstract: A method for packet loss concealment, that includes: continuously receiving a digital audio stream; extracting audio features from the digital audio stream while the digital audio stream is unharmed; and upon detecting a gap in the digital audio stream, filling the gap with one or more previous segments of the digital audio stream, wherein the filling is based on a matching of the one or more of the extracted audio features with one or more audio features adjacent to the gap.

Abstract: An apparatus for generating one or more audio output channels is provided. The apparatus includes a parameter processor for calculating 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. One or more audio channel signals are mixed within the audio transport signal, and one 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 one or more audio channel signals plus the number of the one or more audio object signals. The parameter processor is configured to receive downmix information indicating information on how the one or more audio channel signals and the one or more audio object signals are mixed.

Abstract: An encoding system (100) encodes a first (E1) and further (E2, E3) audio signals as a layered bitstream (B), wherein a quantizer for each frequency band of each signal is selected using a rate allocation rule based on signal-specific rate allocation data, a spectral envelope of the signal and a reference level (EnvE1Max), which is determined based on the spectral envelope of the first signal and is not necessarily included in the bitstream. Further disclosed is a decoding system for reconstructing the audio signals based on the bitstream. In embodiments, the bitstream has a basic layer (BE1), which contains data that enable decoding of the first audio signal, and a spatial layer (Bspatial) facilitating decoding of the further audio signal(s). In embodiments, the encoding system prepares the bitstream subject to a basic-layer bitrate constraint and a total bitrate constraint.

Abstract: A mobile apparatus and control method thereof are provided, which includes an audio data signal input unit arranged to receive an audio data signal. An audio output unit is arranged to output an audio signal according to the received audio data signal within a first reproduction frequency. A balanced armature is arranged to output an audio signal according to the received audio data signal within a second reproduction frequency band. The second reproduction frequency band is different from the first reproduction frequency band. An audio signal processor is arranged to adjust the first reproduction frequency band and the second reproduction frequency band such that a combination of the first reproduction frequency band and the second reproduction frequency band is wider than the first reproduction frequency band and the second reproduction frequency band individually.

Abstract: Systems and methods of performing blind bandwidth extension are disclosed. In an embodiment, a method includes determining, based on a set of low-band parameters of an audio signal, a first set of high-band parameters and a second set of high-band parameters. The method further includes generating a predicted set of high-band parameters based on a weighted combination of the first set of high-band parameters and the second set of high-band parameters.

Abstract: Disclosed are a frame error concealment method and apparatus and an audio decoding method and apparatus. The frame error concealment (FEC) method includes: selecting an FEC mode based on at least one of a state of at least one frame and a phase matching flag, with regard to a time domain signal generated after time-frequency inverse transform processing; and performing corresponding time domain error concealment processing on the current frame based on the selected FEC mode, wherein the current frame is an error frame or the current frame is a normal frame when the previous frame is an error frame.

Abstract: An embodiment of the present invention provides a method for generating a downmixed signal, including: performing a time-frequency transform on a received left sound channel signal and a received right sound channel signal to obtain a frequency domain signal, and dividing the frequency domain signal into several frequency bands; calculating a sound channel energy ratio and a sound channel phase difference of each frequency band; calculating a phase difference between the downmixed signal and a first sound channel signal in each frequency band according to the sound channel energy ratio and the sound channel phase difference; and calculating a frequency domain downmixed signal according to the left sound channel signal, the right sound channel signal, and the phase difference between the downmixed signal and the first sound channel signal in each frequency band. This method effectively improves quality of stereo encoding and decoding.

Abstract: A commercial detection system generates a probe audio fingerprint of an audio signal associated with a media stream on a broadcast channel and determines whether the media stream has commercial content based on analysis of the probe audio fingerprint. The commercial detection system determines whether a same match between the probe audio fingerprint and a reference audio fingerprint is observed across multiple broadcast channels. Responsive to the number of same matches exceeding a predetermined threshold, the commercial detection system determines that the media stream has commercial content. The commercial detection system may also apply a trained feature analysis model to extracted acoustic features of the audio signal. The commercial detection system determines whether the media stream has commercial content based on a confidence score assigned to the probe audio fingerprint. The commercial detection system reduces false positive detection using program guide information of the media stream.

Abstract: Systems, methods, and computer program products are provided for editing digital audio data. In some implementations a method is provided that includes receiving digital audio data, identifying a modification to a portion of the digital audio data, and automatically correcting audio data surrounding one or more edit boundaries resulting from the identified modification including interpolating audio data from a region associated with the one or more edit boundaries.

Abstract: A method and apparatus are provided for generating a personalized radio channel playlist by simultaneously buffering tracks from multiple received channels from one or more source streams and selectively playing back tracks from the buffered channels. Navigation tools permit users to skip buffered songs in their playlist (e.g., skip forward and backward). Users can specify favorite channels for building personal playlists, or multiple default playlist channels can be provided (e.g., by genre). Thumbs up/down buttons on the radio receiver permit entering a song or artist being played back into a favorites list that is used to search all channels for matches or a banned list used to block songs from future playlists. A matched channel carrying the favorite can be added to a playlist. Segments on the playlist can be played back in full or truncated to facilitate preview of playlist segments.

Abstract: In general, techniques are described for grouping audio objects into clusters. In some examples, a device for audio signal processing comprises a cluster analysis module configured to group, based on spatial information for each of N audio objects, a plurality of audio objects that includes the N audio objects into L clusters, where L is less than N, wherein the cluster analysis module is configured to receive information from at least one of a transmission channel, a decoder, and a renderer, and wherein a maximum value for L is based on the information received. The device also comprises a downmix module configured to mix the plurality of audio objects into L audio streams, and a metadata downmix module configured to produce, based on the spatial information and the grouping, metadata that indicates spatial information for each of the L audio streams.

Abstract: A method comprising determining a reverberation time estimate for an audio signal from a first part of an encoded audio signal representing the audio signal.

Abstract: The present document relates to methods and systems for estimating the tempo of a media signal, such as audio or combined video/audio signal. In particular, the document relates to the estimation of tempo perceived by human listeners, as well as to methods and systems for tempo estimation at scalable computational complexity. A method and system for extracting tempo information of an audio signal from an encoded bit-stream of the audio signal comprising spectral band replication data is described. The method comprises the steps of determining a payload quantity associated with the amount of spectral band replication data comprised in the encoded bit-stream for a time interval of the audio signal; repeating the determining step for successive time intervals of the encoded bit-stream of the audio signal, thereby determining a sequence of payload quantities; identifying a periodicity in the sequence of payload quantities; and extracting tempo information of the audio signal from the identified periodicity.

Abstract: In general, techniques are described for performing a positional analysis to code audio data. Typically, this audio data comprises a hierarchical representation of a soundfield and may include, as one example, spherical harmonic coefficients (which may also be referred to as higher-order ambisonic coefficients). An audio compression device that includes one or more processors may perform the techniques. The processors may be configured to allocate bits to one or more portions of the audio data, at least in part by performing positional analysis on the audio data.

Abstract: A linear prediction coefficient of a signal represented in a frequency domain is obtained by performing linear prediction analysis in a frequency direction by using a covariance method or an autocorrelation method. After the filter strength of the obtained linear prediction coefficient is adjusted, filtering may be performed in the frequency direction on the signal by using the adjusted coefficient, whereby the temporal envelope of the signal is shaped. This reduces the occurrence of pre-echo and post-echo and improves the subjective quality of the decoded signal, without significantly increasing the bit rate in a bandwidth extension technique in the frequency domain represented by SBR.

Abstract: In order to enable audio applications for a mobile device that utilize more than one audio input, a peripheral audio device is provided for encoding multichannel audio signals into a reduced number of channels. The peripheral audio device receives audio signals from an audio input/output device, generates at least one output audio signal by combining the received audio signals, and transmits the at least one generated output audio signal to the mobile device. The number of received audio signals is greater than the number of generated output audio signals.

Abstract: This invention introduces audio/speech encoding apparatus audio/speech decoding apparatus, audio/speech encoding method and audio/speech decoding method to efficiently encode the quantization parameters of split multi-rate lattice vector quantization. In this invention, the position of the sub-vector whose codebook indication consumes the most bits is firstly located, and then the value of the codebook is estimated based on the total number of bits available and the bits usage information for other sub-vectors. The difference value is calculated between the actual value and estimated value. Finally, instead of transmitting the codebook indication which consumes the most bits, the position of the sub-vector whose codebook indication consumes the most bits and the difference value between the actual value and the estimated value are transmitted. By applying of the invented method, bits can be saved by the codebook indications.

Abstract: The invention relates to a method and the associated device 1 for separating one or more particular digital audio source signals (si) contained in a mixed multichannel digital audio signal (smix) obtained by mixing a plurality of digital audio source signals (s1, . . . , sp). According to the invention: the modulus of the amplitude or the normalized power of the particular source signal(s) (si) is determined from representative values of said particular source signal(s) contained in the mixed signal; and then linearly constrained minimum variance spatial filtering is performed on the mixed signal in order to obtain each particular source signal (s?i), said filtering being based on the distribution of said particular source signal between at least two channels of the mixed signal, and the modulus of the amplitude or the normalized power of said particular source signal is used as a linear constraint of the filter.

Abstract: The present document relates to audio source coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), as well as to digital effect processors, e.g. exciters, where generation of harmonic distortion add brightness to the processed signal, and to time stretchers where a signal duration is prolonged with maintained spectral content. A system and method configured to generate a time stretched and/or frequency transposed signal from an input signal is described. The system comprises an analysis filterbank (101) configured to provide an analysis subband signal from the input signal; wherein the analysis subband signal comprises a plurality of complex valued analysis samples, each having a phase and a magnitude. Furthermore, the system comprises a subband processing unit (102) configured to determine a synthesis subband signal from the analysis subband signal using a subband transposition factor Q and a subband stretch factor S.