Abstract: A method performed by an audio decoder for reconstructing N audio channels from an audio signal containing M audio channels is disclosed. The method includes receiving a bitstream containing an encoded audio signal having M audio channels and a set of spatial parameters, the set of spatial parameters including an inter-channel intensity difference parameter and an inter-channel coherence parameter. The encoded audio bitstream is then decoded to obtain a decoded frequency domain representation of the M audio channels, and at least a portion of the frequency domain representation is decorrelated with an all-pass filter having a fractional delay. The all-pass filter is attenuated at locations of a transient. A matrixed version of the decorrelated signals are summed with a matrixed version of the decoded frequency domain representation to obtain N audio signals that collectively having N audio channels where M is less than N.

Abstract: Embodiments relate to an audio processing unit that includes a buffer, bitstream payload deformatter, and a decoding subsystem. The buffer stores at least one block of an encoded audio bitstream. The block includes a fill element that begins with an identifier followed by fill data. The fill data includes at least one flag identifying whether enhanced spectral band replication (eSBR) processing is to be performed on audio content of the block. A corresponding method for decoding an encoded audio bitstream is also provided.

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: The present document relates an audio encoding and decoding system (referred to as an audio codec system). In particular, the present document relates to a audio codec system which is particularly well suited for voice encoding/decoding. A transform-based speech encoder is configured to encode a speech signal into a bitstream is described. A speech decoder configured to decode audio signals from a bitstream is further described.

Abstract: Pixel data of a video sequence with enhanced dynamic range (EDR) are predicted based on pixel data of a corresponding video sequence with standard dynamic range (SDR) and a piecewise inter-layer predictor. The output parameters of the piecewise predictor are computed based atleast on two sets of pre-computed values and a prediction cost criterion. The first set of pre-computed values applies to all input SDR frames and comprises a set of SDR pixel values raised to one or more integer power terms. The second set of pre-computed values is frame specific and is computed based on a histogram of an input SDR frame and pixel values of the corresponding EDR frame. The pre-computed values allow for a fast iterative algorithm to identify the best pivot points for the piecewise polynomials according to a prediction cost and to solve for the coefficients of the piecewise predictor.

Abstract: There is provided methods and apparatuses for decoding and encoding of audio signals. In particular, a method for decoding includes receiving a waveform-coded signal having a spectral content corresponding to a subset of the frequency range above a cross-over frequency. The waveform-coded signal is interleaved with a parametric high frequency reconstruction of the audio signal above the cross-over frequency. In this way an improved reconstruction of the high frequency bands of the audio signal is achieved.

Abstract: A multi-channel decoder for generating a binaural signal from a downmix signal using upmix rule information on an energy-error introducing upmix rule for calculating a gain factor based on the upmix rule information and characteristics of head related transfer function based filters corresponding to upmix channels. The one or more gain factors are used by a filter processor for filtering the downmix signal so that an energy corrected binaural signal having a left binaural channel and a right binaural channel is obtained.

Abstract: The present invention relates to transposing signals in time and/or frequency and in particular to coding of audio signals. More particular, the present invention relates to high frequency reconstruction (HFR) methods including a frequency domain harmonic transposer. A method and system for generating a transposed output signal from an input signal using a transposition factor T is described. The system comprises an analysis window of length La, extracting a frame of the input signal, and an analysis transformation unit of order M transforming the samples into M complex coefficients. M is a function of the transposition factor T. The system further comprises a nonlinear processing unit altering the phase of the complex coefficients by using the transposition factor T, a synthesis transformation unit of order M transforming the altered coefficients into M altered samples, and a synthesis window of length Ls, generating a frame of the output signal.

Abstract: For converting a channel-based 3D audio signal to a higher-order Ambisonics HOA audio signal, the channel-based 3D audio signal is transformed (21) from time domain to frequency domain. A primary ambient decomposition (22) is carried out for three-channel triplets of blocks of the domain channel-based 3D audio signal, wherein directional signals and ambient signals are provided (37) for each triplet. From the directional signals directional information of a total directional signal for each triple is derived (23). That total directional signal is HOA encoded (25) according to the derived directions, and ambient signals are HOA encoded (24) according to channel positions. The HOA coefficients of the HOA encoded directional signal and the HOA coefficients of the HOA encoded ambient signal are superimposed (27) in order to obtain a HOA coefficients signal for the channel-based 3D audio signal, followed by a transformation (26) into time domain.

Abstract: The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.

Abstract: The present document relates to a method of layered encoding of a compressed sound representation of a sound or sound field. The compressed sound representation comprises a basic compressed sound representation comprising a plurality of components, basic side information for decoding the basic compressed sound representation to a basic reconstructed sound representation of the sound or sound field, and enhancement side information including parameters for improving the basic reconstructed sound representation.

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: Exemplary embodiments provide encoding and decoding methods, and associated encoders and decoders, for encoding and decoding of an audio scene which at least comprises one or more audio objects (106a). The encoder (108, 110) generates a bit stream (116) which comprises downmix signals (112) and side information which includes individual matrix elements (114) of a reconstruction matrix which enables reconstruction of the one or more audio objects (106a) in the decoder (120).

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: The present document relates to audio source coding systems. In particular, the present document relates to audio source coding systems which make use of linear prediction in combination with a filterbank. A method for estimating a first sample (615) of a first subband signal in a first subband of an audio signal is described. The first subband signal of the audio signal is determined using an analysis filterbank (612) comprising a plurality of analysis filters which provide a plurality of subband signals in a plurality of subbands from the audio signal, respectively.

Abstract: Methods and systems for improving coding efficiency of video. In one aspect of the disclosure, efficiency may be improved by trimming candidate motion vectors used for spatial and/or temporal prediction. In another aspect of the disclosure, efficiency may be improved by choosing selective combinations of syntax elements from video data.

Abstract: Diffuse or spatially large audio objects may be identified for special processing. A decorrelation process may be performed on audio signals corresponding to the large audio objects to produce decorrelated large audio object audio signals. These decorrelated large audio object audio signals may be associated with object locations, which may be stationary or time-varying locations. For example, the decorrelated large audio object audio signals may be rendered to virtual or actual speaker locations. The output of such a rendering process may be input to a scene simplification process. The decorrelation, associating and/or scene simplification processes may be performed prior to a process of encoding the audio data.

Abstract: A multi-channel decoder for generating a binaural signal from a downmix signal using upmix rule information on an energy-error introducing upmix rule for calculating a gain factor based on the upmix rule information and characteristics of head related transfer function based filters corresponding to upmix channels. The one or more gain factors are used by a filter processor for filtering the downmix signal so that an energy corrected binaural signal having a left binaural channel and a right binaural channel is obtained.

Abstract: A system for decoding a video bitstream includes receiving a bitstream and a plurality of enhancement bitstreams together with receiving a video parameter set and a video parameter set extension. The system also receives an output layer set change message including information indicating a change in at least one output layer set.

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.