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 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 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: Many portable playback devices cannot decode and playback encoded audio content having wide bandwidth and wide dynamic range with consistent loudness and intelligibility unless the encoded audio content has been prepared specially for these devices. This problem can be overcome by including with the encoded content some metadata that specifies a suitable dynamic range compression profile by either absolute values or differential values relative to another known compression profile. A playback device may also adaptively apply gain and limiting to the playback audio. Implementations in encoders, in transcoders and in decoders are disclosed.

Abstract: Individual loudspeaker dynamics processing configuration data, for each of a plurality of loudspeakers of a listening environment, may be obtained. Listening environment dynamics processing configuration data may be determined, based on the individual loudspeaker dynamics processing configuration data. Dynamics processing may be performed on received audio data based on the listening environment dynamics processing configuration data, to generate processed audio data. The processed audio data may be rendered for reproduction via a set of loudspeakers that includes at least some of the plurality of loudspeakers, to produce rendered audio signals. The rendered audio signals may be provided to, and reproduced by, the set of loudspeakers.

Abstract: A system utilizing a high throughput coding mode for CABAC in HEVC is described. The system may include an electronic device configured to obtain a block of data to be encoded using an arithmetic based encoder; to generate a sequence of syntax elements using the obtained block; to compare an Absolute-3 value of the sequence or a parameter associated with the Absolute-3 value to a preset value; and to convert the Absolute-3 value to a codeword using a first code or a second code that is different than the first code, according to a result of the comparison.

Abstract: An apparatus for processing an input audio signal relies on a cascade of filterbanks, the cascade having a synthesis filterbank for synthesizing an audio intermediate signal from the input audio signal, the input audio signal being represented by a plurality of first subband signals generated by an analysis filterbank, wherein a number of filterbank channels of the synthesis filterbank is smaller than a number of channels of the analysis filterbank. The apparatus furthermore has a further analysis filterbank for generating a plurality of second subband signals from the audio intermediate signal, wherein the further analysis filterbank has a number of channels being different from the number of channels of the synthesis filterbank, so that a sampling rate of a subband signal of the plurality of second subband signals is different from a sampling rate of a first subband signal of the plurality of first subband signals.

Abstract: The present document relates to audio coding systems. In particular, the present document relates to efficient methods and systems for parametric multi-channel audio coding. An audio encoding system configured to generate a bitstream indicative of a downmix signal and spatial metadata for generating a multi-channel upmix signal from the downmix signal is described. The system comprises a downmix processing unit configured to generate the downmix signal from a multi-channel input signal; wherein the downmix signal comprises m channels and wherein the multi-channel input signal comprises n channels; n, m being integers with m<n. Furthermore, the system comprises a parameter processing unit configured to determine the spatial metadata from the multi-channel input signal.

Abstract: A method for providing media content within a media distribution network. The method comprises transforming source media content into an interim format, thereby providing transformed content. Furthermore, the method comprises storing the transformed content on at least one core storage unit. In addition, the method comprises receiving a request for the source media content from a client. The method further comprises encoding the transformed content or intermediate coded content derived therefrom into encoded content suitable for transmission over a core network and/or an edge network, as well as sending the encoded content via the core network and/or the edge network to the client.

Abstract: An audio session management method for an audio environment having multiple audio devices may involve receiving, from a first device implementing a first application and by a device implementing an audio session manager, a first route initiation request to initiate a first route for a first audio session. The first route initiation request may indicate a first audio source and a first audio environment destination. The first audio environment destination may correspond with at least a first person in the audio environment, but in some instances will not indicate an audio device. The method may involve establishing a first route corresponding to the first route initiation request. Establishing the first route may involve determining a first location of at least the first person in the audio environment, determining at least one audio device for a first stage of the first audio session and initiating or scheduling the first audio session.

Abstract: The present disclosure relates to an apparatus for decoding an encoded Unified Audio and Speech stream. The apparatus comprises a core decoder for decoding the encoded Unified Audio and Speech stream. The core decoder includes an upmixing unit adapted to perform mono to stereo upmixing. The upmixing unit includes a decorrelator unit D adapted to apply a decorrelation filter to an input signal. The decorrelator unit is adapted to determine filter coefficients for the decorrelation filter by referring to pre-computed values. The present disclosure further relates to a an apparatus for encoding a Unified Audio and Speech stream, as well as to corresponding methods and storage media.

Abstract: A rendering mode may be determined for received audio data, including audio signals and associated spatial data. The audio data may be rendered for reproduction via a set of loudspeakers of an environment according to the rendering mode, to produce rendered audio signals. Rendering the audio data may involve determining relative activation of a set of loudspeakers in an environment. The rendering mode may be variable between a reference spatial mode and one or more distributed spatial modes. The reference spatial mode may have an assumed listening position and orientation. In the distributed spatial mode(s), one or more elements of the audio data may each be rendered in a more spatially distributed manner than in the reference spatial mode and spatial locations of remaining elements of the audio data may be warped such that they span a rendering space of the environment more completely than in the reference spatial mode.

Abstract: Encoding and decoding devices for encoding the channels of an audio system having at least four channels are disclosed. The decoding device has a first stereo decoding component which subjects a first pair of input channels to a first stereo decoding, and a second stereo decoding component which subjects a second pair of input channels to a second stereo decoding. The results of the first and second stereo decoding components are crosswise coupled to a third and a fourth stereo decoding component which each performs stereo decoding on one channel resulting from the first stereo decoding component, and one channel resulting from the second stereo decoding component.

Abstract: Some disclosed methods involve encoding or decoding directional audio data. Some encoding methods may involve receiving a mono audio signal corresponding to an audio object and a representation of a radiation pattern corresponding to the audio object. The radiation pattern may include sound levels corresponding to plurality of sample times, a plurality of frequency bands and a plurality of directions. The methods may involve encoding the mono audio signal and encoding the source radiation pattern to determine radiation pattern metadata. Encoding the radiation pattern may involve determining a spherical harmonic transform of the representation of the radiation pattern and compressing the spherical harmonic transform to obtain encoded radiation pattern metadata.

Abstract: Methods for rendering audio for playback by two or more speakers are disclosed. The audio includes one or more audio signals, each with an associated intended perceived spatial position. Relative activation of the speakers may be a cost function of a model of perceived spatial position of the audio signals when played back over the speakers, a measure of proximity of the intended perceived spatial position of the audio signals to positions of the speakers, and one or more additional dynamically configurable functions. The dynamically configurable functions may be based on at least one or more properties of the audio signals, one or more properties of the set of speakers and/or one or more external inputs.

Abstract: Exemplary embodiments provide encoding and decoding methods, and associated encoders and decoders, for encoding and decoding of an audio scene which is represented by one or more audio signals. The encoder generates a bit stream which comprises downmix signals and side information which includes individual matrix elements of a reconstruction matrix which enables reconstruction of the one or more audio signals in the decoder.

Abstract: Methods and systems for improving coding decoding efficiency of video by providing a syntax modeler, a buffer, and a decoder. The syntax modeler may associate a first sequence of symbols with syntax elements. The buffer may store tables, each represented by a symbol in the first sequence, and each used to associate a respective symbol in a second sequence of symbols with encoded data. The decoder decodes the data into a bitstream using the second sequence retrieved from a table.

Abstract: Methods, systems, and computer program products for network-based processing and distribution of multimedia content of a live performance are disclosed. In some implementations, recording devices can be configured to record a multimedia event (e.g., a musical performance). The recording devices can provide the recordings to a server while the event is ongoing. The server automatically synchronizes, mixes and masters the recordings. The server performs the automatic mixing and mastering using reference audio data previously captured during a rehearsal. The server streams the mastered recording to multiple end users through the Internet or other public or private network. The streaming can be live streaming.

Abstract: An encoding system encodes an N-channel audio signal (X), wherein N?3, as a single-channel downmix signal (Y) together with dry and wet upmix parameters ({tilde over (C)}, {tilde over (P)}). In a decoding system, a decorrelating section outputs, based on the downmix signal, an (N?1)-channel decorrelated signal (Z); a dry upmix section maps the downmix signal linearly in accordance with dry upmix coefficients (C) determined based on the dry upmix parameters; a wet upmix section populates an intermediate matrix based on the wet upmix parameters and knowing that the intermediate matrix belongs to a predefined matrix class, obtains wet upmix coefficients (P) by multiplying the intermediate matrix by a predefined matrix, and maps the decorrelated signal linearly in accordance with the wet upmix coefficients; and a combining section combines outputs from the upmix sections to obtain a reconstructed signal ({circumflex over (X)}) corresponding to the signal to be reconstructed.

Abstract: The present disclosure provides methods, devices and computer program products for non-uniform quantization of parameters. The disclosure further relates to a method and apparatus for reconstructing an audio object in an audio decoding system taking the non-uniformly quantized parameters into account. According to the disclosure, such an approach renders it possible to reduce bit consumption without substantially reducing the quality of the reconstructed audio object.

Abstract: Embodiments relate to an audio processing unit that includes a bitstream payload deformatter and a decoding subsystem. The decoding subsystem is coupled to the bitstream payload deformatter and configured to decode at least a portion of a block of an encoded audio bitstream. The block includes a fill element with an identifier indicating a start of the fill element and fill data after the identifier. The fill data includes at least one flag identifying whether a base form of spectral band replication or an enhanced form of spectral band replication is to be performed on audio content of the block. The identifier is a three bit unsigned integer transmitted most significant bit first and having a value of 0x6.

Abstract: The invention provides an efficient implementation of cross-product enhanced high-frequency reconstruction (HFR), wherein a new component at frequency Q?+r?0 is generated on the basis of existing components at ? and ?+?0. The invention provides a block-based harmonic transposition, wherein a time block of complex subband samples is processed with a common phase modification. Superposition of several modified samples has the net effect of limiting undesirable intermodulation products, thereby enabling a coarser frequency resolution and/or lower degree of oversampling to be used. In one embodiment, the invention further includes a window function suitable for use with block-based cross-product enhanced HFR. A hardware embodiment of the invention may include an analysis filter bank, a subband processing unit configurable by control data and a synthesis filter bank.