Described herein is a method of metadata-based dynamic processing of audio data for playback, the method including: receiving, by a decoder, a bitstream including audio data and metadata for dynamic loudness adjustment; decoding, by the decoder, the audio data and the metadata to obtain decoded audio data and the metadata; determining, by the decoder, from the metadata, one or more processing parameters for dynamic loudness adjustment based on a playback condition; applying the determined one or more processing parameters to the decoded audio data to obtain processed audio data; and outputting the processed audio data for playback. Described is further a method of encoding audio data and metadata for dynamic loudness adjustment into a bitstream. Moreover, described are a respective decoder and encoder, a respective system and computer program products.
A method for performing gain control on audio signals is provided. In some implementations, the method involves determining downmixed signals associated with one or more downmix channels associated with a current frame of an audio signal to be encoded. In some implementations, the method involves determining whether an overload condition exists for an encoder. In some implementation, the method involves determining a gain parameter. In some implementations, the method involves determining at least one gain transition function based on the gain parameter and a gain parameter associated with a preceding frame of the audio signal. In some implementations, the method involves applying the at least one gain transition function to one or more of the downmixed signals. In some implementations, the method involves encoding the downmixed signals in connection with information indicative of gain control applied to the current frame.
G10L 19/008 - Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing
Disclosed is an audio signal encoding/decoding method that uses an encoding downmix strategy applied at an encoder that is different than a decoding re-mix/upmix strategy applied at a decoder. Based on the type of downmix coding scheme, the method comprises: computing input downmixing gains to be applied to the input audio signal to construct a primary downmix channel; determining downmix scaling gains to scale the primary downmix channel; generating prediction gains based on the input audio signal, the input downmixing gains and the downmix scaling gains; determining residual channel(s) from the side channels by using the primary downmix channel and the prediction gains to generate side channel predictions and subtracting the side channel predictions from the side channels; determining decorrelation gains based on energy in the residual channels; encoding the primary downmix channel, the residual channel(s), the prediction gains and the decorrelation gains; and sending the bitstream to a decoder.
G10L 19/008 - Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing
H04S 5/00 - Pseudo-stereo systems, e.g. in which additional channel signals are derived from monophonic signals by means of phase shifting, time delay or reverberation
G10L 19/24 - Variable rate codecs, e.g. for generating different qualities using a scalable representation such as hierarchical encoding or layered encoding
4.
ACOUSTIC ECHO CANCELLATION CONTROL FOR DISTRIBUTED AUDIO DEVICES
An audio processing method may involve receiving output signals from each microphone of a plurality of microphones in an audio environment, the output signals corresponding to a current utterance of a person and determining, based on the output signals, one or more aspects of context information relating to the person, including an estimated current proximity of the person to one or more microphone locations. The method may involve selecting two or more loudspeaker-equipped audio devices based, at least in part, on the one or more aspects of the context information, determining one or more types of audio processing changes to apply to audio data being rendered to loudspeaker feed signals for the audio devices and causing one or more types of audio processing changes to be applied. In some examples, the audio processing changes have the effect of increasing a speech to echo ratio at one or more microphones.
The present disclosure relates to a method of processing audio content including directivity information for at least one sound source, the directivity information comprising a first set of first directivity unit vectors representing directivity directions and associated first directivity gains. The disclosure further relates to corresponding methods of encoding and decoding audio content including directivity information for at least one sound source.
G10L 19/00 - Speech or audio signal analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
6.
LAYERED CODING FOR COMPRESSED SOUND OR SOUND FIELD REPRESENTATIONS
A method of layered encoding of a compressed sound representation of a sound or sound field is disclosed. 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. A method of decoding a compressed sound representation of a sound or sound field is also disclosed, wherein the compressed sound representation is encoded in a plurality of hierarchical layers that include a base layer and one or more hierarchical enhancement layers, as well as to an encoder and a decoder for layered coding of a compressed sound representation.
7.
LAYERED CODING FOR COMPRESSED SOUND OR SOUND FIELD REPRESENTATIONS
A method of layered encoding of a compressed sound representation of a sound or sound field is disclosed. 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. A method of decoding a compressed sound representation of a sound or sound field is also disclosed, wherein the compressed sound representation is encoded in a plurality of hierarchical layers that include a base layer and one or more hierarchical enhancement layers, as well as to an encoder and a decoder for layered coding of a compressed sound representation.
G10L 19/24 - Variable rate codecs, e.g. for generating different qualities using a scalable representation such as hierarchical encoding or layered encoding
8.
LAYERED CODING AND DATA STRUCTURE FOR COMPRESSED HIGHER-ORDER AMBISONICS SOUND OR SOUND FIELD REPRESENTATIONS
The present document relates to a method of layered encoding of a frame of a compressed higher-order Ambisonics, HOA, representation of a sound or sound field. The compressed HOA representation comprises a plurality of transport signals. The method comprises assigning the plurality of transport signals to a plurality of hierarchical layers, the plurality of layers including a base layer and one or more hierarchical enhancement layers, generating, for each layer, a respective HOA extension payload including side information for parametrically enhancing a reconstructed HOA representation obtainable from the transport signals assigned to the respective layer and any layers lower than the respective layer, assigning the generated HOA extension payloads to their respective layers, and signaling the generated HOA extension payloads in an output bitstream. The present document further relates to a method of decoding a frame of a compressed HOA representation of a sound or sound field, an encoder and a decoder for layered coding of a compressed HOA representation, and a data structure representing a frame of a compressed HOA representation of a sound or sound field.
9.
LAYERED CODING AND DATA STRUCTURE FOR COMPRESSED HIGHER-ORDER AMBISONICS SOUND OR SOUND FIELD REPRESENTATIONS
The present document relates to a method of layered encoding of a frame of a compressed higher-order Ambisonics, HOA, representation of a sound or sound field. The compressed HOA representation comprises a plurality of transport signals. The method comprises assigning the plurality of transport signals to a plurality of hierarchical layers, the plurality of layers including a base layer and one or more hierarchical enhancement layers, generating, for each layer, a respective HOA extension payload including side information for parametrically enhancing a reconstructed HOA representation obtainable from the transport signals assigned to the respective layer and any layers lower than the respective layer, assigning the generated HOA extension payloads to their respective layers, and signaling the generated HOA extension payloads in an output bitstream. The present document further relates to a method of decoding a frame of a compressed HOA representation of a sound or sound field, an encoder and a decoder for layered coding of a compressed HOA representation, and a data structure representing a frame of a compressed HOA representation of a sound or sound field.
10.
AUDIO ENCODING AND DECODING USING PRESENTATION TRANSFORM PARAMETERS
A method for encoding an input audio stream including the steps of obtaining a first playback stream presentation of the input audio stream intended for reproduction on a first audio reproduction system, obtaining a second playback stream intended for reproduction on a second audio reproduction system, determining a set of transform parameters suitable for transforming an intermediate playback stream presentation to an approximation of the second playback stream presentation, wherein the transform parameters are determined by minimization of a measure of a difference between the approximation of the second playback stream presentation and the second playback stream presentation, and encoding the first playback stream presentation and the set of transform parameters for transmission to a decoder.
11.
DECODING AUDIO BITSTREAMS WITH ENHANCED SPECTRAL BAND REPLICATION METADATA IN AT LEAST ONE FILL ELEMENT
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.
G10L 19/02 - Speech or audio signal analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
12.
METHOD FOR AND APPARATUS FOR DECODING AN AMBISONICS AUDIO SOUNDFIELD REPRESENTATION FOR AUDIO PLAYBACK USING 2D SETUPS
Sound scenes in 3D can be synthesized or captured as a natural sound field. For decoding, a decode matrix is required that is specific for a given loudspeaker setup and is generated using the known loudspeaker positions. However, some source directions are attenuated for 2D loudspeaker setups like e.g. 5.1 surround. An improved method for decoding an encoded audio signal in soundfield format for L loudspeakers at known positions comprises steps of adding (10) a position of at least one virtual loudspeaker to the positions of the L loudspeakers, generating (11) a 3D decode matrix (D'), wherein the positions (Formula l) of the L loudspeakers and the at least one virtual position (Formula II) are used, downmixing (12) the 3D decode matrix (D'), and decoding (14) the encoded audio signal (i14) using the downscaled 3D decode matrix (Formula III). As a result, a plurality of decoded loudspeaker signals (q14) is obtained.
G10L 19/008 - Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing
G10L 19/02 - Speech or audio signal analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
13.
METHOD FOR AND APPARATUS FOR DECODING AN AMBISONICS AUDIO SOUNDFIELD REPRESENTATION FOR AUDIO PLAYBACK USING 2D SETUPS
Sound scenes in 3D can be synthesized or captured as a natural sound field. For decoding, a decode matrix is required that is specific for a given loudspeaker setup and is generated using the known loudspeaker positions. However, some source directions are attenuated for 2D loudspeaker setups like e.g. 5.1 surround. An improved method for decoding an encoded audio signal in soundfield format for L loudspeakers at known positions comprises steps of adding (10) a position of at least one virtual loudspeaker to the positions of the L loudspeakers, generating (11) a 3D decode matrix (D'), wherein the positions (Formula l) of the L loudspeakers and the at least one virtual position (Formula II) are used, downmixing (12) the 3D decode matrix (D'), and decoding (14) the encoded audio signal (i14) using the downscaled 3D decode matrix (Formula III). As a result, a plurality of decoded loudspeaker signals (q14) is obtained.
G10L 19/008 - Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing
14.
METHOD AND APPARATUS FOR GENERATING FROM A COEFFICIENT DOMAIN REPRESENTATION OF HOA SIGNALS A MIXED SPATIAL/COEFFICIENT DOMAIN REPRESENTATION OF SAID HOA SIGNALS
There are two representations for Higher Order Ambisonics denoted HOA: spatial domain and coefficient domain. The invention generates from a coefficient domain representation a mixed spatial/coefficient domain representation, wherein the number of said HOA signals can be variable. A vector of coefficient domain signals is separated into a vector of coefficient domain signals having a constant number of HOA coefficients and a vector of coefficient domain signals having a variable number of HOA coefficients. The constant-number HOA coefficients vector is transformed to a corresponding spatial domain signal vector. In order to facilitate high-quality coding, without creating signal discontinuities the variable-number HOA coefficients vector of coefficient domain signals is adaptively normalised and multiplexed with the vector of spatial domain signals.
G10L 19/00 - Speech or audio signal analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
Audio encoding and decoding systems and methods. An example method for encoding a vector of parameters in an audio encoding system, each parameter corresponding to a non¬ periodic quantity, the vector having a first element and at least one second element, may comprise: representing each parameter in the vector by an index value which may take N values; calculating one or more symbols; associating each of the at least one second element with a respective symbol of the one or more symbols; and encoding each of the at least one second element by entropy coding of the symbol associated with the at least one second element based on a probability table comprising probabilities of the symbols.
G10L 19/00 - Speech or audio signal analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
G10L 19/02 - Speech or audio signal analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
16.
METHOD AND APPARATUS FOR COMPRESSING AND DECOMPRESSING A HIGHER ORDER AMBISONICS REPRESENTATION
Higher Order Ambisonics represents three-dimensional sound independent of a specific loudspeaker set-up. However, transmission of an HOA representation results in a very high bit rate. Therefore compression with a fixed number of channels is used, in which directional and ambient signal components are processed differently. The ambient HOA component is represented by a minimum number of HOA coefficient sequences. The remaining channels contain either directional signals or additional coefficient sequences of the ambient HOA component, depending on what will result in optimum perceptual quality. This processing can change on a frame-by-frame basis.
G10L 19/00 - Speech or audio signal analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
17.
METHOD AND APPARATUS FOR COMPRESSING AND DECOMPRESSING A HIGHER ORDER AMBISONICS REPRESENTATION
Higher Order Ambisonics represents three-dimensional sound independent of a specific loudspeaker set-up. However, transmission of an HOA representation results in a very high bit rate. Therefore compression with a fixed number of channels is used, in which directional and ambient signal components are processed differently. The ambient HOA component is represented by a minimum number of HOA coefficient sequences. The remaining channels contain either directional signals or additional coefficient sequences of the ambient HOA component, depending on what will result in optimum perceptual quality. This processing can change on a frame-by-frame basis.
G10L 19/00 - Speech or audio signal analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
18.
METHOD AND APPARATUS FOR COMPRESSING AND DECOMPRESSING A HIGHER ORDER AMBISONICS REPRESENTATION
Higher Order Ambisonics represents three-dimensional sound independent of a specific loudspeaker set-up. However, transmission of an HOA representation results in a very high bit rate. Therefore compression with a fixed number of channels is used, in which directional and ambient signal components are processed differently. The ambient HOA component is represented by a minimum number of HOA coefficient sequences. The remaining channels contain either directional signals or additional coefficient sequences of the ambient HOA component, depending on what will result in optimum perceptual quality. This processing can change on a frame-by-frame basis.
G10L 19/00 - Speech or audio signal analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
19.
METHOD AND APPARATUS FOR COMPRESSING AND DECOMPRESSING A HIGHER ORDER AMBISONICS REPRESENTATION
Higher Order Ambisonics represents three-dimensional sound independent of a specific loudspeaker set-up. However, transmission of an HOA representation results in a very high bit rate. Therefore compression with a fixed number of channels is used, in which directional and ambient signal components are processed differently. The ambient HOA component is represented by a minimum number of HOA coefficient sequences. The remaining channels contain either directional signals or additional coefficient sequences of the ambient HOA component, depending on what will result in optimum perceptual quality. This processing can change on a frame-by-frame basis.
G10L 19/00 - Speech or audio signal analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
20.
METHOD AND APPARATUS FOR COMPRESSING AND DECOMPRESSING A HIGHER ORDER AMBISONICS REPRESENTATION
Higher Order Ambisonics represents three-dimensional sound independent of a specific loudspeaker set-up. However, transmission of an HOA representation results in a very high bit rate. Therefore compression with a fixed number of channels is used, in which directional and ambient signal components are processed differently. The ambient HOA component is represented by a minimum number of HOA coefficient sequences. The remaining channels contain either directional signals or additional coefficient sequences of the ambient HOA component, depending on what will result in optimum perceptual quality. This processing can change on a frame-by-frame basis.
G10L 19/00 - Speech or audio signal analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
21.
METHOD AND APPARATUS FOR COMPRESSING AND DECOMPRESSING A HIGHER ORDER AMBISONICS REPRESENTATION
Higher Order Ambisonics represents three-dimensional sound independent of a specific loudspeaker set-up. However, transmission of an HOA representation results in a very high bit rate. Therefore compression with a fixed number of channels is used, in which directional and ambient signal components are processed differently. The ambient HOA component is represented by a minimum number of HOA coefficient sequences. The remaining channels contain either directional signals or additional coefficient sequences of the ambient HOA component, depending on what will result in optimum perceptual quality. This processing can change on a frame-by-frame basis.
G10L 19/00 - Speech or audio signal analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
22.
MODEL BASED PREDICTION IN A CRITICALLY SAMPLED FILTERBANK
The present document describes methods and systems which improve the quality of audio source coding employing prediction in the subband domain of a critically sampled filterbank. The methods and systems may make use of a compact description of subband predictors, wherein the description is based on signal models. Alternatively or in addition, the methods and systems may make use of an efficient implementation of predictors directly in the subband domain. Alternatively or in addition, the methods and systems may make use of cross subband predictor terms, as described in the present document, to allow for a reduction of alias artifacts.
G10L 19/04 - Speech or audio signal analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
23.
VIDEO DECODER WITH REDUCED DYNAMIC RANGE TRANSFORM WITH INVERSE TRANSFORM SHIFTING MEMORY
A method for decoding video includes receiving quantized coefficients representative of a block of video representative of a plurality of pixels. The quantized coefficients are dequantized based upon a function of a remainder. The dequantized coefficients are inverse transformed to determine a decoded residue.
24.
CROSS PRODUCT ENHANCED SUBBAND BLOCK BASED HARMONIC TRANSPOSITION
A system for generating a time stretched and/or frequency transposed signal from an audio input signal includes an analysis filter bank configured to derive a number of analysis subband signals from the audio input signal. The analysis subband signals comprise complex- valued analysis samples. A subband processing unit generates a synthesis subband signal from the analysis subband signals. The subband processing unit comprises a block extractor and a nonlinear frame processing unit. For each of the analysis subband signals the block extractor i) forms a frame of L input samples extracted from the complex-valued analysis samples; and ii) applies a block hop size of h samples to the complex-valued analysis samples, prior to forming a subsequent frame of input samples. The nonlinear frame processing unit generates a frame of processed samples by determining a phase and magnitude for each processed sample of the frame. For at least one processed sample: i) the phase of the processed sample is based on the respective phases of the corresponding input sample in the frames of input samples; and ii) the magnitude of the processed sample is based on the magnitude of the corresponding input sample in the frames of input samples. A windowing unit applies window function to the frame of processed samples. Overlapped and added window samples of a plurality of window functions, when weighted by complex weights and shifted with a hop size, form a substantially constant sequence. An overlap and add unit determines the synthesis subband signal by overlapping and adding the samples of a sequence of frames of windowed processed samples. A synthesis filter bank generates the time stretched and/or frequency transposed signal from at least the synthesis subband signal output by the subband processing unit.
Methods and systems are disclosed for generating a plurality of energy- adjusted high frequency subband signals covering a high frequency interval from a plurality of low frequency subband signals. The method involves: receiving the plurality of low frequency subband signals; receiving a set of target energies; generating a plurality of high frequency subband signals from the plurality of low frequency subband signals and from a plurality of spectral gain coefficients associated with the plurality of low frequency subband signals, respectively, by applying the plurality of spectral gain coefficients to the plurality of low frequency subband signals; and adjusting the energy of the plurality of high frequency subband signals using the set of target energies to generate the plurality of energy-adjusted high frequency subband signals. Adjusting the energy of the plurality of high frequency subband signals comprises deteimining, for each target interval, a common envelope adjustment value for each of the high frequency subband signals within the target interval. The methods and sytems may be applied for providing audio signals having increased perceived audio quality.
G10L 19/02 - Speech or audio signal analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
26.
IMPROVED SUBBAND BLOCK BASED HARMONIC TRANSPOSITION
A subband processing unit is configured to determine a synthesis subband signal from an analysis subband signal comprising plural complex-valued analysis samples. The subband processing unit comprises: a block extractor configured to generate a suite of frames of input samples; a nonlinear frame processing unit configured to determine phase and magnitude for each processed sample of a frame; and an overlap and add unit configured to determine the synthesis subband signal by overlapping and adding the samples of a suite of frames of processed samples. The overlap and add unit may be used to control the degree of time stretching and/or harmonic transposition of the system.
Signal processing devices and methods filter and process an audio signal. A signal processing device includes a high frequency reconstructor or a parametric stereo processor that generates modified complex-valued subband sample; and a synthesis filter bank that receives the modified complex-valued subband samples and generates time domain output audio samples. The analysis filter bank comprises analysis filters (h k(n)) and the synthesis filter bank comprises synthesis filters (f k(n)) that are complex exponential modulated versions of a prototype filter (p0(n)) that has a length N. The analysis filter bank and synthesis filter bank have a system delay of D samples, wherein D is smaller than N. This structure can provide a reduced overall delay.
G10L 19/02 - Speech or audio signal analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
The present invention relates to transposing signals in time and/or frequency and in particular to coding of audio signals. More particularly, the present invention relates to a system for transposing an input audio signal by a transposition factor T that includes: means for extracting a frame of L time-domain samples of the input audio signal using an analysis window; means for transforming the time- domain samples into M complex frequency-domain coefficients; a nonlinear processing unit for altering a phase of one or more of the M complex frequency- domain coefficients using the transposition factor T to obtain an altered set of M frequency-domain coefficients; a means for transforming the altered set of M frequency-domain coefficients into M altered time-domain samples; and a means for generating a frame of L time-domain output samples from the M altered time- domain samples using a synthesis window. The invention has example applications in high frequency reconstruction (HFR) and coding of audio signals.
G10L 19/02 - Speech or audio signal analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
The present invention relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR). A system and a method for generating a high frequency component of a signal from a low frequency component of the signal is described. The system comprises an analysis filter bank providing a plurality of analysis subband signals of the low frequency component of the signal. It also comprises a non-linear processing unit to generate a synthesis subband signal with a synthesis frequency by modifying the phase of a first and a second of the plurality of analysis subband signals and by combining the phase-modified analysis subband signals. Finally, it comprises a synthesis filter bank for generating the high frequency component of the signal from the synthesis subband signal.
G10L 19/02 - Speech or audio signal analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
30.
METHOD AND APPARATUS FOR REPRESENTING IMAGE GRANULARITY BY ONE OR MORE PARAMETERS
To characterize film grain, a Film Grain Characterizer (23) receives an incoming information stream (12) and a filtered information stream (24) from which the film grain has been removed. From these streams, the Film Grain Characterizer (23) outputs a message that contains an identity of a model for simulating grain, as well at least one of a set of several parameters, including correlation parameters, intensity-independent parameters and intensity-dependent parameters used by the identified model. An encoder (26, 13) encodes the film grain information for subsequent transmission.
patents
