Abstract: Methods are described to communicate source color volume information in a coded bitstream using SEI messaging. Such data include at least the minimum, maximum, and average luminance values in the source data plus optional data that may include the color volume x and y chromaticity coordinates for the input color primaries (e.g., red, green, and blue) of the source data, and the color x and y chromaticity coordinates for the color primaries corresponding to the minimum, average, and maximum luminance values in the source data. Messaging data signaling an active region in each picture may also be included.

Abstract: An apparatus and method of pre-conditioning audio for machine perception. Machine perception differs from human perception, and different processing parameters are used for machine perception applications (e.g., speech to text processing) as compared to those used for human perception applications (e.g., voice communications). These different parameters may result in pre-conditioned audio that is worsened for human perception yet improved for machine perception.

Abstract: Methods, systems, and bitstream syntax are described for metadata signaling and film-grain parameter adaptation based on a viewing environment which may differ from a reference environment. Example adaptation models are provided for viewing parameters that include: ambient room illumination, viewing distance, and pixels per inch in a target display. Example systems include a single reference viewing environment model and a multi-reference viewing environment model supporting adaptation of film-grain model parameters via adaptation functions or interpolation.

Abstract: A quantization parameter signaling mechanism for both SDR and HDR content in video coding is described using two approaches. The first approach is to send the user-defined QpC table directly in high level syntax. This leads to more flexible and efficient QP control for future codec development and video content coding. The second approach is to signal luma and chroma QPs independently. This approach eliminates the need for QpC tables and removes the dependency of chroma quantization parameter on luma QP.

Abstract: Methods are described to communicate source color volume information in a coded bitstream using SEI messaging. Such data include at least the minimum, maximum, and average luminance values in the source data plus optional data that may include the color volume x and y chromaticity coordinates for the input color primaries (e.g., red, green, and blue) of the source data, and the color x and y chromaticity coordinates for the color primaries corresponding to the minimum, average, and maximum luminance values in the source data. Messaging data signaling an active region in each picture may also be included.

Abstract: Volume leveler controller and controlling method are disclosed. In one embodiment, A volume leveler controller includes an audio content classifier for identifying the content type of an audio signal in real time; and an adjusting unit for adjusting a volume leveler in a continuous manner based on the content type as identified. The adjusting unit may configured to positively correlate the dynamic gain of the volume leveler with informative content types of the audio signal, and negatively correlate the dynamic gain of the volume leveler with interfering content types of the audio signal.

Abstract: Systems and methods for encoding and decoding multiple-intent images and video using metadata. When encoding an image as a multiple-intent image, at least one appearance adjustment may be made to the image. Metadata characterizing the at least one appearance adjustment may be included in, or transmitted along with, the encoded multiple-intent image. When decoding a multiple-intent image, a system may obtain a selection of a desired rendering intent and, based on that selection, either render the multiple-intent image with the applied appearance adjustments or may use the metadata to invert the appearance adjustments and recover the image pre-appearance adjustments.

Abstract: Methods and apparatus for improving noise compensation in mask-based speech enhancement are described. A method of processing an audio signal, which includes one or more speech segments, includes obtaining a mask for mask-based speech enhancement of the audio signal and obtaining a magnitude of the audio signal. An estimate of residual noise is determined in the audio signal after mask-based speech enhancement, based on the mask and the magnitude of the audio signal. A modified mask is determined based on the estimate of the residual noise. Further described are corresponding programs and computer-readable storage media.

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: 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.

Abstract: The present disclosure provides methods, devices and computer program products for encoding and decoding a stereo audio signal based on an input signal. According to the disclosure, a hybrid approach of using both parametric stereo coding and a discrete representation of the stereo audio signal is used which may improve the quality of the encoded and decoded audio for certain bitrates.

Abstract: A method of processing a representation of a multichannel audio signal is provided. The representation includes a first channel and metadata relating to a second channel. The metadata includes, for each of a plurality of first bands of a first filter bank, a respective prediction parameter. The method includes: applying a second filterbank with a plurality of second bands to the first channel to obtain, for each second band, a banded version of the first channel; for each second band, generating a respective time-domain filter based on the prediction parameters and first filters corresponding to the first bands; and for each second band, generating a prediction for the second channel based on a filtered version of the first channel, the filtered version being obtained by applying the respective time-domain filter in that second band to the banded version of the first channel. Also provided are corresponding apparatus, programs, and computer-readable storage media.

Abstract: Methods, systems, and bitstream syntax are described for a file container that supports the storage and transmission of multi-plane images. Examples are provided for coding texture and opacity information using HEVC or VVC coding and the HEIF container. Examples of carrying coded MPI images according to V3C and an example HEIF-based player are also presented.

Abstract: Methods, systems, and media for determining user movement direction are provided. In some embodiments, a method involves obtaining, using a control system, user acceleration data associated with a user. The method involves determining, using the control system, a movement period associated with a movement activity of the user using the user acceleration data, wherein the movement period indicates a duration between two sequential movements by the user. The method involves determining, using the control system, a movement direction corresponding to the movement activity using the user acceleration data based on a direction of acceleration orthogonal to the movement direction in which at least a portion of the user acceleration data is anti-periodic over a period of time corresponding to the movement period.

Abstract: The present disclosure relates to a method and audio processing system (1) for performing source separation. The method comprises obtaining (S1) an audio signal (Sin) including a mixture of speech content and noise content, determining (S2a, S2b, S2c), from the audio signal, speech content (formula A), stationary noise content (formula C) and non-speech content (formula B). The stationary noise content (formula C) is a true subset of the non-speech content (formula B) and the method further comprises determining (S3), based on a difference between the stationary noise content (formula C) and the non-speech content (formula B) a non-stationary noise content formula D), obtaining (S5) a set of weighting factors and forming (S6) a processed audio signal based on a combination of the speech content (formula A), the stationary noise content (formula C), and the non-stationary noise content (formula D) weighted with their respective weighting factor.

Abstract: The present disclosure relates to a method for designing a processor (20) and a computer implemented neural network. The method comprises obtaining input data and corresponding ground truth target data and providing the input data to a processor (20) for outputting a first prediction of target data given the input data. The method further comprises providing the latent variables output by a processor module (21: 1, 21: 2, . . . 21: n?1) to a supervisor module (22: 1, 22: 2, 22: 3, . . . 22: n?1) which outputs a second prediction of target data based on latent variables and determining a first and second loss measure by comparing the predictions of target data with the ground truth target data. The method further comprises training the processor (20) and the supervisor module (22: 1, 22: 2, 22: 3, . . . 22: n?1) based on the first and second loss measure and adjusting the processor by at least one of removing, replacing and adding a processor module.

Abstract: Embodiments are described for a method of rendering audio for playback through headphones comprising receiving digital audio content, receiving binaural rendering metadata generated by an authoring tool processing the received digital audio content, receiving playback metadata generated by a playback device, and combining the binaural rendering metadata and playback metadata to optimize playback of the digital audio content through the headphones.

Abstract: A method (1000) of generating a plurality of audio channels from audio containing height and non-height audio channels for playing back them with an immersive loudspeaker system of with at least one height loudspeaker (5) inside a listening environment, comprising: applying (1500) a virtual height filter (1300) to a portion of each height channel (1010) for, when playing back the height channel by one of the loudspeakers, attenuating spectral components of the height channel directly emanating from said loudspeaker (1;2;3;4) and for amplifying spectral components of the height channel reflected from a roof or an area close to the roof inside the listening environment, to generate a plurality of virtual height filtered audio signals which are added to the corresponding non-height audio channels for playback by corresponding loudspeakers; and playing back the remaining portions of each height audio channel with the at least one height loudspeaker (5).

Abstract: Described herein is a method of processing audio data for playback, the method including: receiving, by a decoder, a bitstream including encoded audio data and metadata, wherein the metadata includes one or more dynamic range control (DRC) sets, and for each DRC set, an indication of whether the DRC set is configured for providing a dynamic loudness compensation effect; parsing the metadata, by the decoder, to identify DRC sets that are configured for providing the dynamic range compensation effect; decoding, by the decoder, the encoded audio data to obtain decoded audio data; selecting, by the decoder, one of the identified DRC sets configured for providing the dynamic loudness compensation effect: extracting from the bitstream, by the decoder, one or more DRC gains corresponding to the selected DRC set; applying to the decoded audio data, by the decoder, the one or more DRC gains corresponding to the selected DRC set to obtain dynamic loudness compensated audio data; and outputting the dynamic loudness comp

Abstract: The present document relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), and to digital effect processors, e.g. so-called exciters, where generation of harmonic distortion adds brightness to the processed signal. In particular, a system configured to generate a high frequency component of a signal from a low frequency component of the signal is described. The system may comprise an analysis filter bank (501) configured to provide a set of analysis subband signals from the low frequency component of the signal; wherein the set of analysis subband signals comprises at least two analysis subband signals; wherein the analysis filter bank (501) has a frequency resolution of ?f.

Abstract: Described herein is a method of performing content-aware audio processing for an audio signal comprising a plurality of audio components of different types. The method includes source separating the audio signal into at least a voice-related audio component and a residual audio component. The method further includes determining a dynamic audio gain based on the voice-related audio component and the residual audio component. The method also includes performing audio level adjustment for the audio signal based on the determined audio gain. Further described are corresponding apparatus, programs, and computer-readable storage media.

Abstract: A method comprising acquiring a set of voltage signals from a set of electrodes arranged in proximity to the ears of a user, based on the set of voltage signals, determining an EOG gaze vector in ego-centric coordinates, determining a head pose of the user in display coordinates, using a sensor device worn by the user, combining the EOG gaze vector and head pose to obtain a gaze vector in display coordinates, and determining a gaze point by calculating an intersection of the gaze vector and an imaging surface having a known position in display coordinates.

Abstract: A method for performing denoising on audio signals is provided. In some implementations, the method involves determining an aggressiveness control parameter value that modulates a degree of speech preservation to be applied. In some implementations, the method involves obtaining a training set of training samples, a training sample having a noisy audio signal and a target denoising mask. In some implementations, the method involves training a machine learning model, wherein the trained machine learning model is usable to take, as an input, a noisy test audio signal and generate a corresponding denoised test audio signal, and wherein the aggressiveness control parameter value is used for: 1) generating a frequency domain representation of the noisy audio signals included in the training set: 2) modifying the target denoising masks: 3) determining an architecture of the machine learning model: or determining a loss during training of the machine learning model.

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: Improved tools for authoring and rendering audio reproduction data are provided. Some such authoring tools allow audio reproduction data to be generalized for a wide variety of reproduction environments. Audio reproduction data may be authored by creating metadata for audio objects. The metadata may be created with reference to speaker zones. During the rendering process, the audio reproduction data may be reproduced according to the reproduction speaker layout of a particular reproduction environment.

Abstract: A method for estimating a user's location in an environment may involve receiving output signals from each microphone of a plurality of microphones in the environment. At least two microphones of the plurality of microphones may be included in separate devices at separate locations in the environment and the output signals may correspond to a current utterance of a user. The method may involve determining multiple current acoustic features from the output signals of each microphone and applying a classifier to the multiple current acoustic features. Applying the classifier may involve applying a model trained on previously-determined acoustic features derived from a plurality of previous utterances made by the user in a plurality of user zones in the environment. The method may involve determining, based at least in part on output from the classifier, an estimate of the user zone in which the user is currently located.

Abstract: 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 presentation of the input audio 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.

Abstract: In a method to improve backwards compatibility when decoding high-dynamic range images coded in a wide color gamut (WCG) space which may not be compatible with legacy color spaces, hue and/or saturation values of images in an image database are computed for both a legacy color space (say, YCbCr-gamma) and a preferred WCG color space (say, IPT-PQ). Based on a cost function, a reshaped color space is computed so that the distance between the hue values in the legacy color space and rotated hue values in the preferred color space is minimized. HDR images are coded in the reshaped color space. Legacy devices can still decode standard dynamic range images assuming they are coded in the legacy color space, while updated devices can use color reshaping information to decode HDR images in the preferred color space at full dynamic range.

Abstract: The present document describes a method (200) for determining N coefficients of an asymmetric prototype filter p0 for use in a low delay M-channel analysis and/or synthesis filter bank (101, 102) comprising M analysis filters hk (103) and/or M synthesis filters fk(106), k=0, . . . , M?1, wherein M is greater than 1, and wherein subband signals which are processed by the analysis and/or synthesis filter bank (101, 102) are decimated by a decimation factor S, with S<M. The method (200) comprises determining (201) a target transfer function of the analysis and/or synthesis filler bank (101, 102) comprising a target delay D; wherein D is smaller or equal to N. Furthermore.

Abstract: Some disclosed methods involve receiving audio data of at least a first audio data type and a second audio data type, including audio signals and associated spatial data indicating intended perceived spatial positions for the audio signals, determining at least a first feature type from the audio data and applying a positional encoding process to the audio data, to produce encoded audio data. The encoded audio data may include representations of at least the spatial data and the first feature type in first embedding vectors of an embedding dimension. Some methods may involve training a neural network, based on the encoded audio data, to transform audio data from an input audio data type having an input spatial data type to a transformed audio data type having a transformed spatial data type. Some methods may involve training a neural network to identify an input audio data type.

Abstract: Some disclosed methods involve receiving, by a control system, location control data from a sound source as the sound source emits sound in a plurality of sound source locations within an audio environment. Some such methods involve receiving, by the control system, direction of arrival data from each audio device of a plurality of audio devices in the audio environment. In some examples, each audio device of the plurality of audio devices includes a microphone array and the direction of arrival data corresponding to microphone signals from microphone arrays responsive to sound emitted by the sound source in the plurality of sound source locations. Some such methods involve estimating, by the control system, sound source locations and audio device locations based, at least in part, on the location control data and the direction of arrival data.

Abstract: A projection system and method therefore related to a first projection device; a second projection device; at least one spatial modulator; and an electronic processor configured to: receive a two-dimensional video data, generate a first plurality of intensity values of a first color gamut and a second plurality of intensity values of a second color gamut, subtract a luminance threshold from a plurality of pixel values of the second color gamut to yield a plurality of positive pixel values and a plurality of negative pixel values, set respective ones of the plurality of pixel values to predetermined values to obtain a residual of the second color gamut, add the residual to the first color gamut, add maximized pixel values to the first color gamut and the second color gamut to create an output gamut, and blend the output gamut with a native color gamut to generate a blended gamut.

Abstract: Systems and methods to transmit data over multiple communication channels in parallel with forward error correction. Original packets are evenly distributed to the channels as the initial systematically channel-encoded packets. Subsequent channel-encoded packets are configured to be linearly independent of their base sets of channel-encoded packets, where a base set for a subsequent channel-encoded packet includes those scheduled to be transmitted before the subsequent packet in the same channel as the subsequent packet, and optionally one or more initial packets from other channels. The compositions of the sequences of the encoded packets can be predetermined without the content of the packets; and the channel-encoded packets can be generated from the original packets on-the-fly by the transmitters of the channels during transmission. When a sufficient number of packets have been received via the channels, a recipient may terminate their transmissions.

Abstract: Some disclosed methods involve receiving microphone signals from a microphone system, including signals corresponding to one or more sounds detected by the microphone system. Some methods may involve determining, via a trained neural network, a filtering scheme for the microphone signals, the filtering scheme including one or more filtering processes. The trained neural network may be configured to implement one or more subband-domain adaptive filter management modules. Some methods may involve applying the filtering scheme to the microphone signals, to produce enhanced microphone signals.

Abstract: A system for real-time monitoring of user-generated audio content for audio anomaly and a related method are disclosed. In some embodiments, the system is programmed to receive, in real time, audio data generated by a first mobile device, such as a smartphone. The system is programed to determine, in real time, whether an audio anomaly has occurred from the audio data. The system is programmed to cause, in real time, a presentation of an alert to a second mobile device, which could be the same smartphone, in response to detecting an occurrence of audio anomaly.

Abstract: Light projection systems using white light illumination. One embodiment provides a projection system using white light illumination. The projection system includes an illumination assembly configured to receive a white light input. A prism is configured to separate the white light input into color light inputs, redirect the color light inputs to respective modulators, and combine modulated color light inputs from the respective modulators into a white light output. An optical filter is configured to spatially Fourier transform the white light output to generate a filtered white light output. A projection lens assembly is configured to project the filtered white light output.

Abstract: Embodiments described herein provide a unified container format for delivering different multimedia applications. One embodiment provides a data structure utilized for implementing a plurality of multimedia applications. The data structure includes a first metadata level including low-level metadata used to perform operations associated with media data in a bitstream. The data structure includes a second metadata level including mid-level metadata used to apply operation metadata to render the media data. The data structure includes a third metadata level including upper-level metadata used to utilize the low-level metadata and the mid-level metadata to deliver the plurality of multimedia applications.

Abstract: Described is a method of modelling a Doppler effect when rendering audio content for a 6 degrees of freedom (6DoF) environment on a user side. In particular, the method may comprise obtaining first parameter values of one or more first parameters indicative of an allowable range of pitch factor modification values. The method may further comprise obtaining a second parameter value of a second parameter indicative of a desired strength of the to-be-modelled Doppler effect. The method may yet further comprise determining a pitch factor modification value based on a relative velocity between a listener and an audio source in the audio content, and the first and second parameter values, using a predefined pitch factor modification function. Particularly, the predefined pitch factor modification function may have the first and second parameters and may be a function for mapping relative velocities to pitch factor modification values.

Abstract: Sample data and metadata related to spatial regions in images may be received from a coded video signal. It is determined whether specific spatial regions in the images correspond to a specific region of luminance levels. In response to determining the specific spatial regions correspond to the specific region of luminance levels, signal processing and video compression operations are performed on sets of samples in the specific spatial regions. The signal processing and video compression operations are at least partially dependent on the specific region of luminance levels.

Abstract: Methods and systems for designing binaural room impulse responses (BRIRs) for use in headphone virtualizers, and methods and systems for generating a binaural signal in response to a set of channels of a multi-channel audio signal, including by applying a BRIR to each channel of the set, thereby generating filtered signals, and combining the filtered signals to generate the binaural signal, where each BRIR has been designed in accordance with an embodiment of the design method. Other aspects are audio processing units configured to perform any embodiment of the inventive method. In accordance with some embodiments, BRIR design is formulated as a numerical optimization problem based on a simulation model (which generates candidate BRIRs) and at least one objective function (which evaluates each candidate BRIR), and includes identification of a best one of the candidate BRIRs as indicated by performance metrics determined for the candidate BRIRs by each objective function.

Abstract: A video encoding method according to an embodiment of the present invention includes generating header information that includes information about resolutions of motion vectors of respective blocks, determined based on motion prediction for a unit image. Here, the header information includes flag information indicating whether resolutions of all motion vectors included in the unit image are integer-pixel resolutions. Further, a video decoding method according to another embodiment of the present invention includes extracting information about resolutions of motion vectors of each unit image from header information included in a target bitstream to be decoded; and a decoding unit for decoding the unit image based on the resolution information. Here, the header information includes flag information indicating whether resolutions of all motion vectors included in the unit image are integer-pixel resolutions.

Abstract: Methods and corresponding systems to process face regions are disclosed. The described methods include providing face bounding boxes and confidence levels for the faces, generating a histogram of the pixels and the faces, generating a probability of face, and generating a face probability map. A face contrast adjustment and a face saturation adjustment can be applied to the face probability map.

Abstract: This disclosure falls into the field of audio coding, in particular it is related to the field of providing a framework for providing loudness consistency among differing audio output signals. In particular, the disclosure relates to methods, computer program products and apparatus for encoding and decoding of audio data bitstreams in order to attain a desired loudness level of an output audio signal.

Abstract: An input image represented in an input domain is received from an input video signal. Forward reshaping is performed on the input image to generate a forward reshaped image represented in a reshaped image domain. Non-reshaping encoding operations are performed to encode the reshaped image into an encoded video signal. At least one of the non-reshaping encoding operations is implemented with an ML model that has been previously trained with training images in one or more training datasets in a preceding training stage. A recipient device of the encoded video signal is caused to generate a reconstructed image from the forward reshaped image.

Abstract: An audio processing method may involve receiving audio signals and associated spatial data, listener position data, loudspeaker position data and loudspeaker orientation data, and rendering the audio data for reproduction, based, at least in part, on the spatial data, the listener position data, the loudspeaker position data and the loudspeaker orientation data, to produce rendered audio signals. The rendering may involve applying a loudspeaker orientation factor that tends to reduce a relative activation of a loudspeaker based, at least in part, on an increased loudspeaker orientation angle. In some examples, the rendering may involve modifying an effect of the loudspeaker orientation factor based, at least in part, on a loudspeaker importance metric. The loudspeaker importance metric may correspond to a loudspeaker's importance for rendering an audio signal at the audio signal's intended perceived spatial position.

Abstract: This disclosure falls into the field of audio coding, in particular it is related to the field of providing a framework for providing loudness consistency among differing audio output signals. In particular, the disclosure relates to methods, computer program products and apparatus for encoding and decoding of audio data bitstreams in order to attain a desired loudness level of an output audio signal.

Abstract: Input images are received as input to a multi-node system. The input images are divided into segments assigned to respective nodes of the multi-node system. Primary and secondary scenes are identified in the segments to ensure compliance with minimum and average distance constraints. Scene-level forward reshaping mappings are generated for the scenes by a respective node for an assigned segment. Forward reshaped images in the segment are generated by the node using the forward reshaping mappings and encoded into an output video signal, which enables a recipient device to generate reconstructed images and to render display images derived from the reconstructed images on an image display.

Abstract: Methods for dialogue enhancing audio content, comprising providing a first audio signal presentation of the audio components, providing a second audio signal presentation, receiving a set of dialogue estimation parameters configured to enable estimation of dialogue components from the first audio signal presentation, applying said set of dialogue estimation parameters to said first audio signal presentation, to form a dialogue presentation of the dialogue components; and combining the dialogue presentation with said second audio signal presentation to form a dialogue enhanced audio signal presentation for reproduction on the second audio reproduction system, wherein at least one of said first and second audio signal presentation is a binaural audio signal presentation.

Abstract: 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.

Abstract: The present disclosure relates to a method and variable quality playback system for selecting a quality of media content. The method comprising receiving (S4001) media content of a data segment (1010) over at least one network path (1031a, 1031b, 1031c), the media content being encoded with network or application-layer code and storing (S4002) the media content in a network or application-layer decoder (1050). The network or application-layer decoder (1050) is configured to decode the media content and provide decoded media content to a buffer (5061) associated with a media renderer (1060). The method further comprises obtaining a decoding metric of the network or application-layer decoder (1050), the decoding metric indicating a property of the decoding process and selecting the quality of the media content of subsequent data segments based on the decoding metric.