Audio processor, audio processing system, audio decoder, and method for providing a representation of a processed audio signal.

Description

1 / 121 “AUDIO PROCESSOR, AUDIO PROCESSING SYSTEM, AUDIO DECODER AND METHOD FOR PROVIDING A REPRESENTATION OF A PROCESSED AUDIO SIGNAL” FIELD OF TECHNOLOGY

[001] The embodiments according to the invention relate to an audio processor for providing a processed audio signal representation based on an input audio signal representation.

[002] Other embodiments according to the invention relate to an audio processing system for providing a processed audio signal representation based on an input audio signal representation.

[003] Other embodiments according to the invention relate to an audio decoder for providing a decoded audio representation based on an encoded audio representation.

[004] Other embodiments according to the invention relate to a method for providing a processed audio signal representation based on an input audio signal representation.

[005] Other embodiments of the invention relate to a computer program.

[006] The embodiment according to the invention relates to an audio processor with multiple stages and (one or more) immediate audio signals and time scale modification on an intermediate audio signal. BACKGROUND OF THE INVENTION

[007] Some conventional devices, systems, and methods will be described below. However, it should be noted that features, functionalities, and details of these conventional devices, systems, and methods may optionally be introduced into any embodiment of the present invention, either individually or in combination. In particular, features, functionalities, and details disclosed in the cited / referenced documents may optionally be introduced into the embodiments according to the present invention.

[008] It has been found that in a communication system, the uninterrupted reproduction of the decoded audio signal is important for good communication (or auditory impression), preferably with the least possible delay. Typically, a communication codec creates information for consecutive frames of 20 Petition 870250092380, dated 09 / 10 / 2025, page 9 / 153 2 / 121 milliseconds. If the (audio) information is sent through a packet-based network, there is no guarantee that the individual packets containing one or more frames of audio information will arrive at the receiver at constant time intervals, but may arrive at different times than expected within the framing grid of the codec itself. This deviation is called network variation.

[009] To ensure smooth playback despite this, a mechanism can be introduced to manage the possible variation, called temporary variation storage management (JBM). In the simplest case, a temporary storage is used, large enough to compensate for the variation, keeping the playback delay fixed.Since this approach is very simplistic (for example, for many applications) and would require large temporary storage (and consequently create large delays), so-called adaptive JBMs are used that can adapt to varying network conditions, keeping the playback delay low to cover only the variation in network variation, for example, by discarding silent frames or adding artificial silent frames [1], or in a more sophisticated way, in addition to discarding frames or adding silent frames or frames marked as corrupted, time scale modification (TSM) is applied to the decoded audio signal for finer adjustment of the playback delay [2][3]. For example, Figure 5a shows the JBM modules described in [2].

[010] It should be noted that modules of the EVS variation temporary storage management solution can optionally be used in embodiments according to the invention. Furthermore, embodiments of the invention can optionally be used in connection with EVS frames.

[011] Furthermore, the complete decoding stream, including the acoustic front-end of a device where the decoding process is performed, is shown in Figure 5b. The acoustic front-end synchronously requests a constant frame (or a constant stream) of decoded PCM samples from the receiver's output temporary storage, i.e., it pulls the PCM temporary storage. Typically, the size of the extracted frame is equal to the frame size of a decoded audio data frame, for example, 20 milliseconds in EVS.

[012] The time scaling procedure in the EVS pattern is a waveform similarity overlap addition (WSOLA). Figures 5c and 5d Petition 870250092380, dated 09 / 10 / 2025, page 10 / 153 Figures 3 / 121 show how the shortening and lengthening of the input frames are performed.

[013] Within the scope of a communication codec such as the upcoming 3GPP Immersive Audio and Voice Services (IVAS) [4], it is now possible to send complex audio scenes comprising more than one audio channel. To achieve low bitrates for such complex scenes, usually based on multi-channel configurations, object-based audio or, for example, scene-based audio in an Ambisonics representation, typically fewer audio channels than those placed in the encoder are encoded as transport channels (TCs) along with parametric information to reconstruct the complete audio scene in the decoder, i.e., the number of channels reconstructed in the decoder is normally greater than the number of audio channels carried in the encoded frames [5][6]. Figure 6 shows a generic block diagram for such modes of operation.The requested output format may differ from the encoded format, and additional rendering may be required to convert from the encoded format to the output format.

[014] The renderer can work directly on the time-domain samples of the decoded transport channels or it can employ a time-frequency transformation before processing, i.e., transforming the time-domain samples to the frequency domain or, in a more general sense, to the domain of the filter bank used for processing (FD), typically dividing a time-domain sample frame into a smaller number of time intervals. Furthermore, the metadata associated with the decoded transport channels can also be divided into subframes, where one of several sets of metadata for a frame is associated with a specific subframe. Each subframe comprises multiple samples of the decoded transport channels or, in the case of frequency-domain processing in the renderer, one or more time intervals.

[015] Figure 5e shows this processing in which the decoded transport channel frame (TC frame) is processed by a time-frequency transformation, resulting in a number of time intervals (TS) of frequency domain samples. The renderer reconstructs and renders the output format using parameters associated with the TC frame. In this example, several Petition 870250092380, dated 09 / 10 / 2025, page 11 / 153 4 / 121 parameter sets are used, each associated with a subframe (SF), where each comprises numerous time intervals. The renderer produces the desired output channels in the frequency domain, and the time samples of the desired output channels are generated by inverse time transformation in the frequency domain of the rendered frequency domain time intervals.

[016] Furthermore, rendering can include binaural rendering with additional information about the listener's position (part of the rendering parameters in Figure 6), here the so-called motion delay for sound is of great importance for the auditory impression and should be as low as possible, especially if one wants to obtain a fine-grained JBM as in [2][3], the TSM should be performed in the same way for all output channels.

[017] The simplest way to have a JBM for such a decoder would be a TSM as in [2][3] applied to the output channels and synchronized across all output channels. Figure 8 shows a block diagram of this solution. The only difference from the EVS JBM solution is that the TSM module is now a multi-channel (MC) TSM synchronized across all output channels. This naive approach has been found to have several disadvantages. It increases complexity as the TSM needs to be applied across all output channels. It is more difficult to obtain good quality across all frames due to differences in the audio signal and the need for the TSM to be synchronized across all channels. The motion-to-sound delay in the case of a binaural renderer with listener position information is increased. Furthermore, the TSM, along with the receiver's temporary output storage, causes the delay, and therefore the motion-to-sound delay, to be non-constant.

[018] Generally, this decoder can be seen as belonging to a class of audio processors in which the processor has several stages with intermediate audio signals and each processing stage can optionally obtain parameters to direct the processing (Figure 7).

[019] It was recognized that the IVAS parametric reconstruction of Figure 6 can be seen as a 2-stage audio processor, where the representation of the input audio signal are the encoded frames, stage 1 is the transport channel and metadata decoder (and does not need processing parameters), the Petition 870250092380, dated 09 / 10 / 2025, page 12 / 153 5 / 121 decoded transport channels are the intermediate audio signal 1, stage 2 is reconstruction and rendering, processing parameters 2 are the decoded metadata and additional rendering parameters, and the processed audio signal are the final output channels.

[020] It should be noted that the general concepts mentioned above may optionally be employed in embodiments according to the present invention.

[021] In view of this situation, there is a desire to have a concept that allows a modification of the time scale with a better balance between complexity, processing latency and achievable auditory impression. SUMMARY OF THE INVENTION

[022] One embodiment of the invention creates an audio processor (e.g., an audio decoder) to provide a processed (e.g., decoded and rendered) audio signal representation (e.g., in the form of a plurality of rendered time-domain audio signals) based on an input audio signal representation (e.g., an encoded representation of a multi-channel audio content). The audio processor is configured to perform a plurality of processing steps (e.g., separate processing functionalities; e.g., a first processing and a second processing) in order to provide the processed audio signal representation based on the input audio signal representation.The audio processor is configured to perform a timescale modification (e.g., a modification of the length of a portion of an audio signal, e.g., an audio frame, provided by the first processing) on ​​one or more intermediate audio signals, which are provided by a first processing (e.g., a first processing step), in order to obtain one or more timescale-modified intermediate audio signals. The audio processor is configured to perform a second processing [e.g., a second processing step; e.g., an upmix and / or a rendering], which follows the first processing [e.g., the first processing step], based on the one or more timescale-modified intermediate audio signals.

[023] This approach is based on the finding that it is advantageous, in many Petition 870250092380, dated 09 / 10 / 2025, page 13 / 153 6 / 121 applications, perform a time-scaling modification on one or more intermediate signals in an audio signal processor, wherein a first processing provides one or more intermediate signals, wherein the time-scaling modification is applied to these one or more intermediate signals, and wherein a second subsequent processing, which follows the first processing, is performed on (i.e., applied to) the one or more time-scaling modified intermediate audio signals.

[024] It has been recognized that one or more intermediate signals are often more suitable for time-scaling modification with high temporal resolution (e.g., with fine temporal granularity), whereas one or more input audio signals from the first processing are generally not suitable for such time-scaling modification. Furthermore, it has been recognized that it is relatively easy to adapt the processing parameters appropriately, which can, for example, adjust the second processing to the time-scaling modification. It has been recognized that the processing parameters can be adapted more easily and accurately to the time-scaling modification if the time-scaling modification is applied to one or more intermediate audio signals between the first processing and the second processing.For example, it was recognized that applying timescale modification before the first processing would make it significantly more difficult to adapt the processing parameters of the second processing to said timescale modification. Furthermore, it was recognized that applying timescale modification after the second processing would, in some cases, result in an inappropriately long latency until the adaptations of the second processing parameters exhibited an effect on the representation of the processed audio signal (output). Additionally, applying timescale modification after the second processing would result in a variable delay between movement and sound, which would degrade the auditory impression.

[025] Conversely, by applying time scale modification before the second processing, which can, for example, use one or more motion parameters to obtain the representation of the processed audio signal and adapt the representation of the processed audio signal to a motion defined by one or more motion parameters, a motion delay for sound can Petition 870250092380, dated 09 / 10 / 2025, page 14 / 153 7 / 121 can be kept constant, since there is no need for any modification of the time scale after the second processing. In this way, it is possible to obtain a particularly good auditory impression.

[026] In other words, it was recognized that processing after time scale modification (TSM) allows achieving a constant delay between the processing parameters and the processed audio signal.

[027] Furthermore, it was also recognized that additional temporary receiver output storage is no longer necessary, as we can now always render exactly the amount of audio signal requested, for example, by an acoustic front-end. In this way, the implementation effort is reduced.

[028] Furthermore, it was also recognized that, in many applications, the number of intermediate audio signals is less than the number of audio signals representing the processed audio signal, so applying time scale modification to one or more intermediate audio signals can also lead to a reduction in complexity.

[029] In conclusion, it was recognized that applying time scale modification to one or more intermediate audio signals, i.e., between a first processing and a second processing, constitutes a particularly good solution that brings with it a good balance between achievable audio quality, implementation complexity and responsiveness.

[030] In one embodiment, the audio processor is configured to perform the second processing in dependence on one or more parameters (for example, in dependence on decoded metadata or rendering parameters), where, for example, one or more parameters that are used in the second processing may be temporally associated with frames or subframes of one or more intermediate audio signals (for example, before timescale modification).

[031] It has been recognized that efficient audio processing can be achieved by performing secondary processing depending on one or more parameters. Consequently, secondary processing can be adapted according to the processing parameters that are entered into the audio processor, wherein the processing parameters may, for example, originate from the input audio representation (which may, for example, comprise the Petition 870250092380, dated 09 / 10 / 2025, page 15 / 153 8 / 121 processing parameters as secondary information) and / or where the processing parameters may be based on user interaction. Thus, the second processing can be used to adapt the representation of the intermediate audio signal to have characteristics defined in the representation of the input audio signal and / or to have characteristics that are defined by user interaction. However, it has been found that applying the processing parameters in the second processing, i.e., after modifying the time scale, often results in the processed audio signal representation providing a good auditory impression.Furthermore, applying processing parameters that depend on user interaction during the second processing stage, i.e., after modifying the time scale, typically results in a fast response of the processed audio signal representation to user interactions, which usually meets user expectations well.

[032] In one embodiment, the audio processor is configured to adapt the second processing to the timescale change (for example, by adapting metadata used in the second processing, which may, for example, be derived from an encoded audio representation, to the timescale change, and / or by adapting one or more (internal) processing parameters of the second processing, for example, interpolation steps, to the timescale change).

[033] It has been recognized that such adaptation of the second processing to the timescale modification usually results in a particularly good auditory impression. For example, a shortening or lengthening of one or more timescale-modified intermediate audio signals (when compared to one or more (original) audio signals) can be taken into account in the second processing. Just as an example, the speed at which the parameters of the second processing change can be adapted to the timescale modification (so that, for example, the parameters applied in the second processing change (relatively) slowly if the timescale modification involves a lengthening, and so that the parameters used in the second processing change comparatively faster if the timescale modification involves a shortening). Consequently, having the possibility of adapting characteristics of internal processing functions of Petition 870250092380, dated 09 / 10 / 2025, page 16 / 153 9 / 121 By modifying the timescale and / or adapting one or more parameters used in the second processing to the timescale modification, it is possible to achieve a second processing that fits well with the characteristics of the timescale-modified intermediate audio signals. In this way, it is possible to obtain a good auditory impression.

[034] It has been revealed that an adaptation of one or more processing parameters, which are considered in the second processing (i.e., which adapt the characteristics of the second processing) results in good audio processing that provides a good auditory impression. For example, it has been revealed that timescale modification, which results in a lengthening or shortening of the timescale-modified intermediate audio signal, can have the effect that the original processing parameters intended for use by the second processing may no longer fit the one or more timescale-modified intermediate audio signals that are fed into the second processing.Thus, by adapting the processing parameters to the timescale modification (for example, using interpolation, extrapolation, or repetition), or by generating an additional parameter or set of parameters (for example, in the case of lengthening the timescale-modified intermediate audio signals) or by canceling a parameter or set of parameters (for example, in the case of shortening the timescale-modified intermediate audio signals), it is typically possible to reduce or even eliminate audible artifacts in the representation of the processed audio signal that might otherwise originate from the timescale modification.Thus, the processing parameters, which are typically temporally related to one or more intermediate audio signals, can be adapted to a new timescale-modified intermediate audio signals, so that the second processing handles portions of one or more timescale-modified intermediate audio signals with the associated “appropriate” parameters. In this way, good audio quality can be achieved with moderate effort.

[035] In one embodiment, the audio processor (for example, a parameter adapter that is part of the audio processor) is configured to adapt one or more processing parameters (for example, decoded metadata). Petition 870250092380, dated 09 / 10 / 2025, page 17 / 153 10 / 121 and / or rendering parameters; for example, metadata that defines how the second processing should perform a rendering and / or an upmix; for example, spatial metadata that defines how the second processing should perform a rendering and / or an upmix; for example, metadata that defines a relationship (e.g., a level difference and / or a phase difference and / or a correlation and / or a covariance) between two or more upmixed and / or rendered audio signals that are obtained by the second processing; for example, metadata that defines a direction;For example, metadata associated with one or more transport channel signals, which are intermediate audio signals) to modify the time scale (for example, to a time scaling result, where, for example, a target value for the time scaling is determined by a temporary variation storage control or by a user setting, and where, for example, an actual value of the time scaling is determined depending on the characteristics of one or more intermediate signals), to obtain adapted processing parameters. The audio processor can be configured to perform the second processing depending on one or more adapted parameters.

[036] Adapting one or more processing parameters to the time scale change allows the second processing to be performed in sync with the time-scale-modified intermediate audio signals and therefore gives a good auditory impression.

[037] In one embodiment of the audio processor, one or more processing parameters are temporally associated with one or more intermediate audio signals (for example, temporally associated with respective frames or subframes of one or more intermediate audio systems), wherein, for example, a processing parameter, or a set of processing parameters, may be associated with each subframe of one or more intermediate audio signals (wherein, for example, the one or more processing parameters are spatial metadata with subframes associated with the intermediate audio signal in the time domain). The audio processor may be configured to obtain the adapted processing parameters so that the adapted processing parameters are adapted to a timing of the intermediate audio signals modified on the time scale. Petition 870250092380, dated 09 / 10 / 2025, page 18 / 153 11 / 121

[038] It has been recognized that obtaining adapted parameters in such a way that the adapted processing parameters are adapted to the timing of the timescale-modified intermediate audio signals can ensure proper association between one or more timescale-modified intermediate audio signals and the processing parameters. For example, shortening or lengthening one or more intermediate audio signals, caused by timescale modification, can invalidate an original temporal association between portions (subframes) of one or more intermediate audio signals and processing parameters. However, adapting the processing parameters can re-establish proper temporal alignment between the adapted processing parameters and the associated portions (e.g., subframes) of one or more timescale-modified intermediate audio signals.However, adapting the processing parameters can involve establishing an appropriate temporal association, but it can also involve a quantitative modification of the processing parameters, for example, by interpolation or extrapolation or other processing functionalities. Thus, it is possible to achieve that the appropriate processing parameters, which can be modified when compared to the original processing parameters, are associated with portions (e.g., subframes) of one or more intermediate audio signals modified in time scaling. Consequently, the degradation of the auditory impression due to time scaling modification can be avoided or at least reduced to an acceptable level.

[039] In one embodiment, the audio processor is configured to obtain the adapted processing parameters using temporal interpolation (e.g., between two or more (original) processing parameters), wherein, for example, the temporal interpolation is determined by a time associated with the adapted processing parameters and times associated with the original processing parameters (e.g., so that the adapted processing parameters are adapted according to the time scale modification). Alternatively, however, temporal extrapolation could be used.

[040] By obtaining the adapted processing parameters using temporal interpolation, it is possible to resolve the effect that the parameters of Petition 870250092380, dated 09 / 10 / 2025, page 19 / 153 12 / 121 Original processing parameters do not properly fit the timescale of one or more timescale-modified intermediate audio signals. For example, as a result of timescale modification, the original processing parameters, which are originally associated with portions (e.g., subframes) of the (original) intermediate audio signals, may no longer fit a timescale grid of processing parameters of one or more timescale-modified intermediate audio signals. Just as an example, an original processing parameter (or an original set of processing parameters) might be associated with a subframe of a given length of one or more (original) intermediate audio signals. However, this subframe of a given length might be mapped onto a portion of one or more timescale-modified intermediate audio signals with a different length.If the second processing requires that there be a processing parameter, or a set of processing parameters, for a given length of one or more timescale-modified intermediate audio signals, it becomes evident that the original parameter or the original set of parameters does not fit the complete temporal portion of one or more timescale-modified intermediate audio signals in which the corresponding portion of the (original) intermediate audio signals are timescale-mapped.In other words, since one or more timescaled-modified intermediate audio signals have a different temporal duration from one or more (original) intermediate audio signals on which the timescaled-modified intermediate audio signals are based, there is no one-to-one correspondence between the parameter time portions of the original intermediate audio signals (to which the original processing parameters are associated) and the parameter time portions of the timescaled-modified intermediate audio signals.

[041] However, it was recognized that an adaptation of processing parameters associated with parameter time portions of one or more intermediate audio signals modified in the time scale can be obtained by interpolation, whereby, for example, one or more processing parameters or sets of processing parameters can be associated with respective time instances of one or more intermediate (original) audio signals and also with respective time instances of one or more intermediate audio signals. Petition 870250092380, dated 09 / 10 / 2025, page 20 / 153 13 / 121 modified in the time scale, and can therefore be considered as support points for an interpolation-based determination of one or more processing parameters associated with necessary parameter time instances of the time-scaled modified intermediate audio signals.In other words, using knowledge of timescale modification, the time instances of the processing parameters can be mapped from the timescale of one or more (original) intermediate audio signals to the timescale of the (one or more) timescale-modified intermediate audio signals, where the timescale instances of the timescale-modified intermediate audio signals for which the processing parameters are needed may be different from those parameter time instances obtained by mapping the parameter time instances based on the original processing parameter time instances. Thus, processing parameters adapted to the necessary parameter time instances can be obtained by interpolation, for example, using the mapped parameter time instances.Consequently, interpolated processing parameters can be obtained that are very useful for further processing of one or more time-scaled modified intermediate audio signals.

[042] In particular, it was found that interpolation techniques are suitable for obtaining a representation of the processed and undistorted output audio signal by the second processing.

[043] In one embodiment, the audio processor is configured to apply a time-scaling modification to one or more decoded transport channels (where the one or more decoded transport channels are the one or more intermediate audio signals), for example, to counteract a transport packet oscillation, which comprise an encoded representation of the one or more transport channels (for example, to shorten or lengthen frames of the decoded transport channels, for example, with a subframe resolution; for example, controlled by a temporary variation storage control).

[044] It has been found that applying a timescale modification to one or more decoded transport signals generally results in a particularly good compromise between computational complexity and quality. Petition 870250092380, dated 09 / 10 / 2025, page 21 / 153 14 / 121 achievable. For example, it was recognized that applying a timescale modification to one or more decoded transport channels is typically possible with finer temporal granularity (resolution) compared to the timescale of an encoded representation (e.g., an encoded representation of transport channels). Furthermore, it was also recognized that it is typically advantageous to apply a timescale modification to decoded transport channels, since applying the timescale modification to decoded transport channels usually requires less processing effort compared to applying a timescale modification to the output audio signals (e.g., individual channel signals) provided, for example, by subsequent processing of the decoded transport channels.For example, it was recognized that the output audio signals obtained by further processing of the decoded transport channels typically comprise a greater amount of audio information (e.g., more channels and / or a larger bandwidth) compared to the transport channels. Furthermore, it was recognized that applying timescale modification to the decoded transport channels is also particularly advantageous, since this typically does not affect the “real-time” characteristics of the output audio signals derived from the decoded transport channels, as the relevant “real-time” parameters are typically applied in subsequent processing after the timescale modification.In other words, it was recognized that applying time-scaling modification to the decoded transport channels allows for efficient time scaling, which will provide the opportunity to introduce real-time characteristics into the representation of the output audio signal. Furthermore, it should be noted that such time-scaling modification of one or more decoded transport channels allows for partial compensation of variations in transport packets, the content of which may represent one or more transport channels in an encoded form.

[045] In one embodiment, the audio processor is configured to obtain (for example, using a rendering, where, for example, the number of output channels may be different from the number of input channels in an audio encoder) or reconstruct (where, for example, the number of output channels may be equal to the number of input channels in an audio encoder) Petition 870250092380, dated 09 / 10 / 2025, p. 22 / 153 15 / 121 a plurality of output audio channels (which may form the output audio representation) based on one or more decoded transport channels (which may be one or more intermediate audio signals) and metadata (wherein the metadata may, for example, be obtained based on the encoded representation, for example, using a rendering and / or using an upmix, wherein, for example, the rendering may comprise an upmix of the transport channels for speaker signals associated with a plurality of speakers, and / or wherein the rendering may comprise binauralization, in which two headphone signals are generated using information about a listener's position (wherein the information about the listener's position may, for example,to be used to determine a head-related transfer function that is applied to derive the two headphone signals based on one or more decoded transport channels.

[046] It has been recognized that obtaining or reconstructing a plurality of output audio channels based on one or more decoded transport channels after time scaling modification (e.g., after time scaling modification is applied to one or more decoded transport channels) and using metadata provides a particularly good compromise between an achievable auditory impression and complexity. As described earlier, for example, real-time processing can be applied in “second processing,” i.e., when obtaining or reconstructing a plurality of output audio channels based on one or more decoded transport channels after time scaling. Consequently, the output audio representation can be obtained such that the output audio representation can be adapted to the user's requirements in real-time or near real-time, without having a delay impact from time scaling modification.Thus, dividing the process into a decoding phase, which is performed before the timescale modification, and a rendering or bottom-up blending phase, which is performed after the timescale modification, proves to yield very good results with low complexity.

[047] In one embodiment, the audio processor is configured to apply the same time-scaling modification to a plurality of decoded transport channels, or even to all decoded transport channels, for example, in a synchronized manner (for example, to ensure that the Petition 870250092380, dated 09 / 10 / 2025, page 23 / 153 16 / 121 lengths of a plurality of decoded transport channels on the resulting time scale, which are, for example, inserted into a rendering and / or a mix, are equal; for example, to thus obtain a plurality of decoded transport channels on the modified time scale).

[048] It has been found that applying a timescale modification to a plurality of decoded transport channels (or even to all decoded transport channels) in a synchronized manner results in timescale-modified intermediate audio signals, i.e., timescale-modified decoded transport channels, which are well usable in an upmix or subsequent rendering, or in any other multichannel audio channel generation. Furthermore, synchronizing the timescale modification of the different decoded transport channels can help avoid audible distortion or at least significant audible distortion in all transport channels, since the specific characteristics of all decoded transport channels can, for example, be taken into account.

[049] In one embodiment, the audio processor is configured to adapt metadata for obtaining or reconstructing the output audio channels (wherein the metadata that is adapted may, for example, be associated with the transport channels, and wherein the metadata that is adapted may, for example, be included in the encoded audio representation) depending on the timescale modification (for example, depending on the time by which one or more decoded transport channels are shortened or lengthened; for example, so that the adapted metadata is adapted to an altered duration of the decoded transport channels modified in the timescale).

[050] By using this adaptation of metadata to obtain the reconstruction of the output audio signals, a good auditory impression can be achieved even if the time scale modification occurs between the first processing (e.g., the decoding of the transport channels) and the second processing. Thus, appropriate metadata can be provided even if a time relationship between one or more output signals from the first processing and one or more input signals from the second processing is varied (e.g., in a way that varies over time) by the time scale modification.

[051] In one mode, the audio processor is configured to Petition 870250092380, dated 09 / 10 / 2025, page 24 / 153 17 / 121 perform the first processing (e.g., decoding of encoded transport channels) for (e.g., by) time segments (e.g., of an audio content) of a first length (e.g., frame by frame; e.g., with a first temporal granularity, where, e.g., the first temporal granularity is a frame, which comprises a plurality of subframes). In addition, the audio processor is configured to perform the second processing for (e.g., by) time segments (e.g., of the audio content) of a second length that is shorter than the first length (e.g., in terms of subframes;For example, with a second temporal granularity that is finer than the first temporal granularity, where, for example, the second temporal granularity is a subframe, for example, comprising a plurality of time intervals], or the audio processor is configured to execute the second processing with temporal increments (e.g., in steps) of a second length, which is shorter than the first length (where, for example, portions of one or more intermediate output signals of the second length are inserted into the second processing for each execution of the second processing, so that, for example, each execution of the second processing advances one processing of the timescale-modified intermediate audio signals by one step of the second length (measured on the timescale of one or more timescale-modified intermediate audio signals)).

[052] It has been revealed that using different temporal granularity (e.g., time segments of different durations) for the first processing and for the second processing allows for high processing efficiency and also brings good real-time characteristics of the decoding (e.g., with the use of a relatively short temporal granularity in the second processing). For example, by decoding frame-encoded transport channels in the first processing, a good compromise is achieved between bitrate requirements and achievable audio quality. On the other hand, it has been found that “post-processing” (e.g., mixing or rendering performed by the second processing) is more efficient and provides better audio quality by operating on shorter time portions (of the timescale-modified intermediate audio signals) than the first processing. For example, Petition 870250092380, dated 09 / 10 / 2025, page 25 / 153 18 / 121 By using relatively small portions in the second processing, or by making comparatively small (gradual) temporal increments in the second processing, signals comprising multiple sources can be efficiently manipulated while still providing a good impression. Thus, it was recognized that using different temporal granularities in different processing stages is a good choice in terms of efficiency and achievable audio quality, where applying time scaling between these processing stages using different temporal granularities was considered an advantageous concept that does not significantly compromise the processing results.

[053] In one embodiment, a granularity of the timescale modification (e.g., a sample) is finer than the second length (e.g., a subframe, which comprises a plurality of time intervals).

[054] It has been recognized that, in many applications, it is desirable to have a timescale modification granularity that is finer than the second length. In particular, it has been recognized that a relatively fine timescale modification granularity helps to reduce artifacts caused by timescale modification, even though such fine timescale granularity may result in timescale-modified intermediate audio signal frame lengths that deviate by integer multiples of the second length. However, it has been recognized that it is possible to handle such timescale-modified intermediate information block or frame lengths without introducing significant artifacts.

[055] In one embodiment of the audio processor, the first processing (e.g., the decoding of one or more transport channels) is configured to provide block-by-block time-domain samples (e.g., using a block-by-block frequency-domain to time-domain transformation; e.g., using a modified discrete cosine transform), where a block of time-domain samples provided by the first processing comprises a first number of time-domain samples. Furthermore, the second processing (e.g., the upmixing and / or rendering of one or more transport channels into output audio signals that comprise the representation of the processed audio signal) is configured to operate in blocks. Petition 870250092380, dated 09 / 10 / 2025, page 26 / 153 19 / 121 of time-domain samples that are based on time-scaled modified time-domain samples that are derived from the time-domain samples provided by the first processing using time-scale modification (e.g., transforming time-scaled modified time-domain sample blocks into a spectral domain), wherein the time-domain sample blocks processed by the second processing comprise a shorter length than the time-domain sample blocks provided by the first processing.Alternatively, the second processing (e.g., mixing and / or rendering one or more transport channels into output audio signals that make up the representation of the processed audio signal) is configured to insert, per processing step, a block of a plurality of time-domain samples that is based on time-scale modified time-domain samples that are derived from the time-domain samples provided by the first processing using time-scale modification (e.g., advancing one processing by the block of time-scale modified time-domain samples), wherein the block of a plurality of time-domain samples inserted in the second processing per processing step comprises a shorter length than the blocks of time-domain samples provided by the first processing.

[056] It has been found that processing comparatively short blocks of time-domain samples in the second processing, or inputting a comparatively short block of time-domain samples as new time-domain samples in the second processing, brings with it comparatively good temporal resolution in the second processing, which allows the processed audio representation (output) to have good audio quality and allows for real-time or near-real-time adjustment in the second processing. On the other hand, the high efficiency, for example, a high bit rate efficiency, of processing comparatively large blocks in the first processing can still be exploited.Furthermore, it was recognized that the use of different temporal granularities in the first processing and in the second processing can still be used advantageously, even if the time scale modification occurs between the first and second processing. Petition 870250092380, dated 09 / 10 / 2025, page 27 / 153 20 / 121 processing. Thus, it is possible to obtain a good compromise between efficiency and achievable audio quality, where the processors also include good real-time or near-real-time capabilities.

[057] In one embodiment of the audio processor, blocks of time-domain samples processed by the second processing comprise a plurality of time-domain samples, wherein, for example, a time resolution of the second processing, for example, of a time-frequency transformation that is performed by the second processing, is less than a time resolution of a time-domain signal that makes up the intermediate signal (for example, a time resolution of the decoded transport channel time signal).

[058] However, it was found that processing, in the second processing, time-domain sample blocks comprising a plurality of time-domain samples brings good efficiency, and that it is not necessary to use a second temporally very fine processing after the time scale modification (even if the time scale modification is adjustable with a single-sample granularity). Thus, good computational efficiency can be achieved with such a concept, since a block processing is typically more efficient than a second processing of single-sample granularity.

[059] In one embodiment of the audio processor, metadata (e.g., metadata sets), which are used in the second processing, are associated with subframes of the input audio signal representation and / or are associated with subframes of one or more intermediate audio signals, wherein a subframe comprises one or more time intervals (e.g., 4 time intervals). The audio processor is configured to provide the one or more time-scaled modified intermediate audio signals, such that a respective frame of the one or more time-scaled modified intermediate audio signals comprises a different length (e.g., a different number of audio samples) when compared to a corresponding frame of the input audio signal representation and / or when compared to a corresponding frame of one or more intermediate audio signals.

[060] It has been recognized that, in some cases, the metadata that is Petition 870250092380, dated 09 / 10 / 2025, p. 28 / 153 21 / 121 frames associated with subframes of the input audio signal representation and / or that are associated with subframes of one or more intermediate audio signals can still be used with good results, even if the timescale-modified frames of one or more intermediate audio signals comprise a different length when compared to the corresponding frames of the input audio signal representation and / or when compared to the corresponding frames of one or more intermediate signals. This has been found to be particularly true if the temporal granularity of the metadata is smaller, or even significantly smaller, than the temporal variations caused by timescale modification or temporal misalignments caused by timescale modification.Just as an example, in some cases, modifying the timescale causes a temporal misalignment between the subframes processed in the second processing and the metadata associated with those subframes processed in the second processing, which is smaller or significantly smaller than the temporal granularity of the metadata; typically, there are no significant audible artifacts caused by this. Thus, a good impression can be achieved with high efficiency by exploiting the different temporal granularities of the processing steps and the metadata, and by exploiting the listener's tolerance for small temporal variations in the second processing.

[061] In one embodiment, the audio signal processor is configured to provide one or more time-scaled modified intermediate audio signals such that the length of a respective frame of the one or more time-scaled modified intermediate audio signals (e.g., between 10 ms and 30 ms, for example, with a granularity of less than 1 ms; for example, with a single-sample temporal granularity) is different from a multiple of a block size (e.g., 1.25 ms or 5 ms) used in the second processing (at least in the case where time-scale modification is applied).Alternatively or additionally, the audio processor is configured to perform secondary processing with temporal increments (e.g., in steps) (e.g., with increments in 1.25 ms steps, which may correspond to 60 samples at a sampling rate of 48 kHz), where the audio signal processor is configured to provide one or more modified intermediate audio signals. Petition 870250092380, dated 09 / 10 / 2025, page 29 / 153 22 / 121 time scale such that the length of a respective frame of one or more intermediate audio signals modified in the time scale (e.g., between 10 ms and 30 ms; e.g., between 480 samples and 1920 samples at a sampling rate of 48 kHz; e.g., with a temporal granularity of less than 1 ms; e.g., with a single-sample temporal granularity) is different from a multiple of the temporal increments (e.g., 1.25 ms or 60 samples at a sampling rate of 48 kHz).

[062] It was recognized that such a choice of time scale modification, which allows fine-tuning of the time scale without being tied to a granularity of the second processing, provides a high-quality time scale while also allowing good quality output signals from the second processing.

[063] In one embodiment, the audio signal processor is configured to form a sample block for second processing (e.g., a sample block associated with a current frame; e.g., a sample block that is processed in second processing or a sample block that is inserted into second processing as a time increment in a processing step of second processing) based on one or more time-domain samples (e.g., between 1 and 59 samples, if second processing requires a 60-sample block for processing or as a time increment) associated with a previous frame of one or more time-scaled modified intermediate audio signals and based on one or more time-domain samples (e.g., between 1 and 59 samples,If the second processing requires a block of 60 samples for processing or as a time increment) associated with a current frame of one or more time-scaled modified intermediate audio signals. The audio signal processor is configured to form one or more sample blocks (e.g., of 60 samples) for the second processing (e.g., a sample block associated with a current frame) based on a plurality of time-domain samples associated with a current frame of one or more time-scaled modified intermediate audio signals (e.g., until there are not enough time-domain samples associated with a current frame of the one or more time-scaled modified intermediate audio signals remaining to form a block of, Petition 870250092380, dated 09 / 10 / 2025, page 30 / 153 23 / 121 samples for the second processing). In addition, the audio signal processor is configured to store in temporary storage one or more residual (e.g., excess) time-domain samples (which are not sufficient for the formation of a sample block for the second processing) (e.g., between 1 and 59 samples if the second processing requires a block of 60 samples for processing or as a time increment) associated with a current frame of one or more intermediate audio signals modified in time scale for (subsequent) sample block formation (e.g.,of 60 samples) for the second processing that includes one or more time-domain samples associated with a subsequent frame of one or more time-scaled modified intermediate audio signals (where one or more temporarily stored residual time-domain samples can be attached to a subsequent frame of one or more time-scaled modified intermediate audio signals).

[064] By using such sample block formation for the second processing, which considers, for the formation of a single sample block, samples stored in temporary storage associated with a previous frame and samples associated with a current frame, and which considers, for the formation of one or more additional sample blocks for the second processing, only time-domain samples associated with the current frame, it is possible to form sample blocks for the second processing with uniform length, even if the time-scale modification changes the length of a frame to be different from an integer multiple of the length of the frame blocks used for the second processing.This is also supported by the possibility of temporarily storing one or more time-domain samples associated with the current frame, which makes sense mainly if the number of frames temporarily stored is less than the number of samples needed to form a sample block for the second processing. Consequently, the temporarily stored samples from the current frame can be used later, for example, when a sufficient number of time-domain samples (or frequency-domain samples) associated with the subsequent frame are available. In this way, the concept allows for efficient handling of the second processing, where sample blocks of equal length can be used. Petition 870250092380, dated 09 / 10 / 2025, page 31 / 153 24 / 121 easily provided for the second processing. Furthermore, it was recognized that the concept does not severely degrade the second processing, even if processing parameters are used in the second processing, since the granularity of the processing parameters is often smaller (or even significantly smaller) than the length of the sample blocks used for the second processing. In addition, difficulties in the second processing, which would be caused by using sample blocks of different lengths for the second processing, can be avoided. In this way, the concept brings with it high processing efficiency and good results.

[065] In one embodiment, the audio processor is configured to use one or more residual (e.g., oversized) time-domain samples stored in temporary storage associated with a previous frame (e.g., all residual (e.g., oversized) time-domain samples stored in temporary storage associated with a previous frame) and a maximum possible number of time-domain samples associated with the current frame of one or more time-scaled modified intermediate audio signals (such that a number of residual time-domain samples associated with the current frame of one or more time-scaled modified intermediate audio signals is less than a block size used in second-way block processing) for a block formation of (e.g.,60) samples for the second processing associated with the current frame (where the maximum possible number of time-domain samples can be chosen so that the remaining number of residual time-domain samples is less than the size of a block of samples processed by the second processing or inserted into the second processing by temporal increment).

[066] Using this approach, the number of samples from the current frame, which are stored in temporary storage for later formation of sample blocks for the second processing, can be kept reasonably small. Furthermore, using this approach, latency can be kept as low as possible while using blocks of equal length in the second processing, even in the presence of fine-grained timescale modification, thus reducing processing effort. Additionally, an overlap of timedomain sample blocks used in the second processing can Petition 870250092380, dated 09 / 10 / 2025, page 32 / 153 25 / 121 can be avoided in this way.

[067] In one embodiment of the audio processor, a metadata temporal resolution, which is used in the second processing, is lower (i.e., coarser) than a second processing temporal resolution.

[068] In this case, variations in the actual edges of the time-domain sample blocks used in the second processing (which are, for example, caused by the use of constant-length blocks, even in the case of a timescale modification) do not actually degrade the audio quality of an output audio representation provided by the second processing. Furthermore, using such a concept, the effort for metadata encoding can be kept reasonably small while the temporal resolution of the second processing can still be comparatively high, which in turn allows for the consideration of one or more real-time or near-real-time processing parameters.

[069] In one embodiment of the audio processor, one or more intermediate audio signals are one or more transport channel signals (decoded) (e.g., TC signals), wherein, for example, the one or more transport channel signals may comprise respective frames, wherein the respective frames of the one or more transport channel signals are, for example, shortened or lengthened by time scale modification, so that the time scale-modified intermediate signals comprise respective shortened or lengthened frames (e.g., comprising fewer or more time domain samples per frame than the original frames).

[070] It has been recognized that applying time-scaling modification to one or more decoded transport channel signals is particularly efficient, since decoding typically operates on relatively long blocks of samples (e.g., audio frames), while time-scaling modification can achieve a relatively fine timescale. Furthermore, it has been recognized that decoded transport channels typically comprise time-domain samples, which are often subjected to further processing that is substantially separate from the decoding of the transport channels. In other words, it has been recognized that decoded transport channels constitute intermediate signals, where further processing Petition 870250092380, dated 09 / 10 / 2025, page 33 / 153 26 / 121 (for example, the second processing) is performed in a different domain (e.g., in a different transformation domain and / or using a different spectral domain representation) when compared to the decoding of the transport channel that is performed by the first processing. Consequently, it was recognized that timescale modification can be efficiently interposed between the (core) decoding of the transport channels (which is performed in the first processing or as the first processing) and a subsequent post-processing that is performed in the second processing or as the second processing.It was also recognized that the "post-processing" of transport channels (original or timescale-modified) is often sufficiently independent of the decoding of the transport channels, so that variations in frame size due to timescale modification do not significantly impair said post-processing, which is performed in the second processing or as the second processing. In this way, good audio quality can be achieved by the described solution.

[071] In one embodiment, the audio processor is configured to perform time scale modification using a quality control mechanism (so that, for example, a real time scale value is adjusted depending on a desired time scale value and taking into account the characteristics of one or more intermediate audio signals; wherein, for example, the correlation between different portions of an intermediate audio signal, or of a processed version thereof, is taken into account to adjust a time scale).

[072] By using this timescale modification with a quality control mechanism, timescale artifacts can be significantly reduced, since the actual timescale is normally adjusted (or adapted) to the characteristics (e.g., the length of a tone cycle) of the intermediate (original) audio signals. Thus, by using the quality control mechanism, the resulting length of the timescale-modified intermediate audio signals is not determined solely by a desired timescale, but rather the exact timescale is chosen to achieve reasonably small (or even minimal) distortions, for example, by performing a timescale overlay and addition operation. Thus, the Petition 870250092380, dated 09 / 10 / 2025, page 34 / 153 27 / 121 artifacts can be kept low, typically at the cost that a length (e.g., in terms of time-domain samples) of the time-scaled modified intermediate audio signals (or frames of the time-scaled modified intermediate audio signals) cannot be predetermined independently of the signal. This, in turn, has the consequence that the duration of the time-scaled modified intermediate audio signals may not exactly match the “original” requirements of the second processing, but it has been recognized that this does not constitute a serious problem. Thus, the concept creates a good compromise between implementation efficiency and audio quality.

[073] In one embodiment, the audio processor is configured to switch between a time-scale modification mode with quality control and a time-scale modification mode without quality control, depending on one or more characteristics of one or more intermediate audio signals and / or depending on side information.

[074] By using this approach, high efficiency can be achieved in a wide range of circumstances.

[075] In one embodiment, the audio processor is configured to operate in (e.g., to selectively switch to) a time-scale modification mode without quality control (e.g., to selectively switch to time-scale modification mode without quality control).

[076] When operating in timescale modification mode without quality control, the duration of the timescale-modified intermediate audio signals can be (precisely) predetermined by a controller, allowing for frame alignment or predefining how many subframes (or time-domain sample blocks) should be provided to the second processing for a frame of the (original) intermediate audio signals. Consequently, precise control can be exercised independently of the audio signal details, which can, for example, be useful for frame boundary synchronization or for synchronization between an audio encoder and an audio decoder. Furthermore, this mechanism can reduce the effort of adapting processing parameters and thus reduce processing complexity in some cases.

[077] In one mode, the audio processor is configured to Petition 870250092380, dated 09 / 10 / 2025, p. 35 / 153 28 / 121 selectively switch to a special timescale modification mode (silent frame) (e.g., to a timescale modification mode without quality control) for frames classified as silent (e.g., for frames with power below a threshold level, or for frames for which there is a silent frame signal). Furthermore, the audio processor is configured to perform timescale modification so that no temporary storage of residual frames is required (e.g., so that a number of timedomain samples from a modified frame at the current timescale plus numerous timedomain samples stored in temporary storage from a modified frame at the previous timescale is an integer multiple of a number of timedomain samples from a block of timedomain samples processed by the second processing).or such that a number of time-domain samples from a modified frame in the current timescale plus numerous time-domain samples stored in temporary storage from a modified frame in the previous timescale is an integer multiple of a number of time-domain samples inserted in the second processing to perform a temporal increment), by operating in the special timescale modification mode (silent frame) (in which the timescale modification is adjusted to provide such a number of modified time-domain samples in the timescale that the boundaries of the subframes used in the second processing are aligned with the boundaries of the frames of the subframes used in the second processing).

[078] After switching to a special time-scaling modification mode for frames classified as silent, time alignment can be achieved efficiently without significantly compromising audio quality. In other words, by using a special time-scaling modification mode, it is possible to have a frame of the time-scaling modified intermediate audio signals fully processed by the second processing (taking into account a processing time grid applied by the second processing) without leaving any time-domain or frequency-domain values ​​of this (current) frame unprocessed by the second processing. Thus, the detection of a frame classified as silent can be used to efficiently arrive at a Petition 870250092380, dated 09 / 10 / 2025, page 36 / 153 29 / 121 situation where it is not necessary to store in temporary storage any time-domain or frequency-domain samples of a current frame of the time-scaled modified intermediate audio signals for later processing (along with samples from a subsequent frame). In this way, a low-latency state can be achieved without audio quality issues. For example, the time-scale modification can be adjusted in such a way that it provides a desired number of time-scaled modified time-domain samples without being tied to a quality control mechanism, since the audio quality is not significantly degraded in a silent frame condition.

[079] In conclusion, the audio processor exploits the detection of the silent frame condition to efficiently achieve time synchronization between the timescale-modified audio signals and the second processing.

[080] In one embodiment, the audio processor is configured to adapt a subframe size, which are used (e.g., processed) in the second processing or which are inserted into the second processing (e.g., as a block of new time-domain samples) to make a temporal increment, for the modification of the time scale.

[081] After adapting the size of the subframes that are processed in the second processing or that are inserted into the second processing to make a temporal increment in the timescale modification, a synchronization between the timescale-modified intermediate audio signals and the second processing can be achieved. For example, adapting a subframe size (e.g., in terms of time intervals or in terms of time-domain samples or in terms of frequency-domain samples) to the timescale modification can, for example, avoid the need to store in temporary storage any values ​​of the timescale-modified intermediate audio signals (for processing along with a subsequent frame).For example, the size of the subframes that are used in a second processing, or that are inserted into the second processing to perform a temporal increment, can be adjusted in such a way that all time intervals, or all time-domain samples, or all frequency-domain samples of a frame (shortened or lengthened) of the audio signals are included. Petition 870250092380, dated 09 / 10 / 2025, page 37 / 153 30 / 121 timescale-modified intermediates are associated with the subframes of the second processing, where, for example, a final subframe of the second processing ends at the end of the respective frame of the timescale-modified intermediate audio signals. Consequently, temporary storage or joint processing of time-domain samples or frequency-domain samples or slots from a current frame and a subsequent frame is not necessary, making the processing resource-efficient and also keeping the processing effort low.

[082] In one embodiment, the audio processor is configured to adapt a number of time intervals (for example, a number of frequency domain intervals or frequency domain samples that form a subframe of one or more time-scaled modified intermediate audio signals, or a number of time domain intervals or time domain samples that form a subframe of one or more time-scaled modified intermediate audio signals; for example, a subframe size) that are processed in the second processing, or that are inserted in the second processing (for example, as a block of new frequency domain samples) to make a temporal increment in the time scale modification.

[083] After adapting the number of time intervals or the size of subframes that are processed in the second processing, or that are inserted into the second processing to make a temporal increment, to the timescale modification, it is possible to achieve that a current frame is fully processed by the second processing before a subsequent frame is available (or before a timescale modification has been applied to the subsequent frame), even if the timescale modification results in a change in the number of time intervals in a frame of the timescale-modified intermediate audio signal (for example, when compared to the number of time intervals in a frame of the original intermediate audio signals). Consequently, the advantages mentioned above, such as low latency and efficient processing, can be achieved.

[084] In one mode, the audio processor is configured to perform timescale modification using one or more overlay and addition operations by overlapping different segments (e.g., frames, Petition 870250092380, dated 09 / 10 / 2025, page 38 / 153 31 / 121 subframes or intervals) of one or more intermediate (original) audio signals. Furthermore, the audio processor is configured to adapt metadata (e.g., metadata associated with frames, subframes, or intervals of the intermediate (original) audio signals) where overlay and addition operations are applied to different segments (e.g., frames, subframes, or intervals) of one or more intermediate audio signals, in order to obtain adapted metadata associated with frames or subframes of the one or more intermediate audio signals modified in timescale.

[085] It has been found that the overlapping and addition of different segments of one or more intermediate audio signals results in a timescale-modified intermediate audio signal comprising different characteristics when compared to the original intermediate audio signal. However, it has also been recognized that an adaptation of the metadata, which is associated with the (original) intermediate audio signals, particularly for time portions that are subject to an overlapping and addition operation, can be used to provide metadata that is well adapted to the signal characteristics of the timescale-modified portion of the intermediate audio signals obtained by the overlapping and addition operation. Consequently, metadata adaptation can help avoid audio quality degradation for time portions obtained using an overlapping and addition operation.For example, adapting metadata can correspond to the overlay and add-on operation, where metadata from different segments that are overlaid and added are combined linearly (e.g., in a manner similar to overlay and add-on combination).

[086] In one embodiment, the audio processor is configured to linearly combine metadata associated with frames or subframes of one or more intermediate (original) audio signals (e.g., unmodified decoded transport channels) that are involved in an overlay and add-on operation, to (thus) obtain metadata associated with frames or subframes resulting from one or more timescale-modified intermediate audio signals (e.g., frames resulting from an overlay and add-on operation).

[087] It was found that the linear combination of metadata associated with Petition 870250092380, dated 09 / 10 / 2025, page 39 / 153 32 / 121 frames or subframes of one or more intermediate audio signals that are involved in an overlay and addition operation is suitable for obtaining metadata that fits parts of the timescale-modified intermediate audio signals obtained using the overlay and addition operation. In particular, it was recognized that the overlay and addition operation and the linear metadata combination are similar operations, so that the result is a good fit between adapted (processed) metadata and parts of the timescale-modified intermediate audio signals obtained by the overlay and addition operation.

[088] In one embodiment, the audio processor (e.g., timescale modification; e.g., WSOLA-based timescale modification) is configured to identify (e.g., using cross-correlation) a segment (also designated as “identified segment” or “best match”), from a plurality of candidate segments that are portions of a given audio frame of an intermediate audio signal, for overlay and addition with a predetermined portion (e.g., an initial portion; e.g., “seg. a”) of the given audio frame, using a waveform similarity determination, in order to obtain a shortened version of the given audio frame (which is, e.g., a frame of the timescale-modified intermediate audio signal).

[089] After identifying a segment, from among a plurality of candidate segments that are portions of a given audio frame of an intermediate audio signal, for superimposition and addition with the predetermined portion using a waveform similarity determination, it helps to keep audible distortions reasonably small. For example, it is possible to identify a segment very similar to the predetermined part of the given audio frame and, consequently, a superimposition and addition operation generates very small audible distortions. Thus, a real time scale can be determined by the characteristics of the audio signal, since a time alignment between the portions that are superimposed and added is obtained by determining the waveform similarity, i.e., by the characteristics of the actual audio signals.Thus, modifying the timescale is well suited to the characteristics of audio signals, which provides good audio quality (but brings a challenge). Petition 870250092380, dated 09 / 10 / 2025, page 40 / 153 33 / 121 that the actual amount of timescale modification is affected by the signal characteristics and cannot be arbitrarily determined by a control). However, it was recognized that the concept of timescale modification is well suited for current applications.

[090] In one embodiment, the audio processor is configured to include, at the end of the shortened version of the provided audio frame, an unmodified portion of the provided audio frame (e.g., rem) that follows the identified segment in an original version of the provided audio frame (e.g., so that the shortened version of the provided audio frame is a concatenation of an overlay and add portion, in which the identified segment is combined overlay and add (e.g., overlay and add) with the predetermined portion, and a remaining unmodified portion (e.g., “rem”) of the provided audio frame).

[091] By including, at the end of the shortened version of the provided audio frame, an unmodified portion of the provided audio frame, it is possible to ensure that the timescale modification affects only a portion of one or more intermediate audio signals that are suitable for a timescale modification. Thus, the overlay and add-on operation, which constitutes the core of the timescale modification, can be applied to a portion that is suitable for the overlay and add-on operation, while other portions of one or more intermediate audio signals can be left unchanged and can, for example, be appended to the end of a frame (e.g., to the end of a shortened version of a given audio frame). In this way, the timescale modification can be performed with good audio quality and with reasonable computational effort.

[092] In one embodiment, the audio processor (e.g., timescale modification; e.g., WSOLA-based timescale modification) is configured to identify (e.g., using cross-correlation) a segment (also designated as the “identified segment” or “best match”) from a plurality of candidate segments that are portions (e.g., designated as “candidates”) comprising values ​​from a previous audio frame of an intermediate audio signal preceding a given audio frame (and possibly values ​​from the current audio frame of the signal). Petition 870250092380, dated 09 / 10 / 2025, page 41 / 153 34 / 121 intermediate audio), for an overlay and addition with a predetermined portion (e.g., an initial portion; e.g., “sec. a”) of the given audio frame, using a waveform similarity determination, in order to obtain an elongated version of the given audio frame.

[093] After identifying a segment comprising values ​​from a previous audio frame of an intermediate audio signal preceding a given audio frame for overlay and addition with a predetermined portion of the given audio frame, a lengthening of the given audio frame can be achieved. For example, a well-suited portion of the audio frame preceding the given audio frame can be “reused” to obtain a lengthened audio frame, where a good auditory impression can be obtained by appropriately choosing the “identified segment”, taking into account the similarity between the predetermined portion of the given audio frame and the candidate segments (from which the “identified segment” is selected).Thus, it is possible to obtain a smooth transition between the predetermined portion of the provided audio frame and an audio portion that begins in the previous audio frame (which precedes the provided audio frame) and typically ends at the end of the provided audio frame. Consequently, the total length of the stretched version of the provided audio frame is greater than the length of the provided audio frame, since time-domain samples or frequency-domain samples from the previous audio frame are also included (through the overlay and addition operation) in the provided audio frame. In conclusion, a modification of the stretching timescale can be efficiently achieved using this concept, where good audio quality can be obtained with reasonable computational complexity.

[094] In one embodiment, the audio processor is configured to include, at the end of the extended version of the provided audio frame, an unmodified portion comprising values ​​from the provided audio frame (and optionally also values ​​from the preceding audio frame) (e.g., “rem”) that follows the identified segment (e.g., designated as “best match”), for example, so that the extended version of the provided audio frame is a concatenation of an overlay and add portion, in which the identified segment is combined (e.g., overlaid and added) with the Petition 870250092380, dated 09 / 10 / 2025, page 42 / 153 35 / 121 predetermined portion (e.g., with an original segment of the provided frame) and a remaining unmodified portion that is based on a portion that follows the segment identified in an original version of the provided audio frame and possibly the previous audio frame.

[095] By including, at the end of the extended version of the audio frame, an unmodified portion comprising values ​​from the provided audio frame that follows the identified segment (to which the overlay and addition is applied), a good auditory impression can be achieved. In particular, it can be achieved that the length of the final portion of an audio frame remains substantially unchanged, where the timescale modification is effectively performed at the beginning of the provided frame (e.g., using the overlay and addition operation). However, it has been recognized that such a concept is computationally efficient and provides a good auditory impression, since the overlay and addition operation is not applied to the entire frame, but only to a portion of the frame that is recognizably suitable for the overlay and addition operation.Consequently, the operations of overlapping and adding, as well as the modification of the stretching timescale, can be efficiently applied even if one or more intermediate audio signals are not stationary throughout the entire length of an audio frame.

[096] In one embodiment, the audio processor is configured to interpolate in time between one or more processing parameters (e.g., inputs from a mixing matrix Mprev or Mk,prev; e.g., metadata values ​​or processing values ​​derived from metadata values;for example, inputs of a mixing matrix) associated with a previous frame of one or more intermediate audio signals and one or more processing parameters (for example, inputs of a mixing matrix M or Mk) associated with a current frame of one or more intermediate audio signals, in order to (for example, thereby) obtain processing parameters associated with different time instances (for example, designated with time indices n) of a current frame of one or more intermediate audio signals modified in time scale using an interpolation function (for example, g[n] or ga[n]) that determines the contributions of one or more processing parameters associated with the previous frame and the contributions of one or more associated processing parameters; Petition 870250092380, dated 09 / 10 / 2025, page 43 / 153 36 / 121 to the current frame in a linear combination of processing parameters.

[097] It has been found that, by means of temporal interpolation between one or more processing parameters associated with a previous frame and one or more processing parameters associated with a current frame, the processing parameters can be adapted to the time scale, thus avoiding audible distortions. It has also been recognized that the use of an interpolation function (which can, for example, define an interpolation that deviates from a purely linear interpolation) allows the interpolation to be adapted to the real time scale, which yields particularly good results. In particular, the use of such an interpolation function is advantageous in the case where the time scale modification is not performed linearly in a complete frame (which is an advantageous situation, as described above).

[098] In conclusion, it was recognized that temporal interpolation between one or more processing parameters or between one or more sets of processing parameters using an interpolation function yields good results in terms of audio quality and efficiency.

[099] In one embodiment of the audio processor, the interpolation function is a linear or piecewise linear function that maps a time index (e.g., a sample index; e.g., n) to a weighting value (e.g., g[n] or ga[n]) that determines the contributions of a processing parameter associated with a previous frame and a processing parameter associated with a current frame to a time-interpolated processing parameter.

[0100] By choosing an interpolation function, which is preferably a linear or piecewise linear function, the interpolation of one or more processing parameters can be performed efficiently, where the described form of the interpolation is in accordance with the functionality of the timescale modification.

[0101] In one embodiment of the audio processor, for a temporally shortened audio frame (e.g., current), the interpolation function comprises a greater slope (e.g., variation of the interpolation function value with the time index) for a portion (e.g., Lsec) of the modified timescale of the current frame that was obtained using a combination of overlapping and addition than for a portion (e.g., Lrem) of the current frame modified in the timescale that is not modified (e.g., relative to the Petition 870250092380, dated 09 / 10 / 2025, page 44 / 153 37 / 121 intermediate audio signal (e.g., original); for example, a “remaining part”). Alternatively or additionally, for a temporally elongated audio frame (e.g., current), the interpolation function comprises a smaller slope (e.g., variation of the interpolation function value with the time index) for a portion (e.g., Lseg) of the modified timescale of the current frame that was obtained using a combination of overlay and addition than for a portion (e.g., Lrem) of the current frame modified on the timescale that is not modified (e.g., relative to the intermediate audio signal (e.g., original); for example, a “remaining part”).

[0102] It has been found that such a choice of interpolation function is, for example, well suited for use with timescale modification concepts in which a timescale occurs primarily in an initial portion of a given frame, while a final portion of the given frame remains substantially unchanged.

[0103] Thus, in the case of a shortening timescale modification, the choice of the interpolation function is, for example, made so that the interpolation function comprises a (comparatively) larger slope in a part of an audio frame where real time compression (shortening) occurs, and comprises a (comparatively) smaller slope (e.g., “normal”) in a part of an audio frame where no time compression occurs. Consequently, the interpolation occurs comparatively fast (e.g., faster than “normal”) in a part of the frame where real shortening occurs, and the interpolation of the parameters occurs comparatively slowly in a portion of the frame where no time compression (shortening) occurs.

[0104] In the case of a time-stretching modification, the choice of the interpolation function can, for example, be made so that the interpolation function comprises a (comparatively) smaller slope in a part of an audio frame where a real-time extension (stretching) occurs, and comprises a (comparatively) larger slope (e.g., “normal”) in a part of an audio frame where no time extension occurs. Consequently, the interpolation proceeds comparatively more slowly (e.g., slower than “normal”; e.g., slower than if the temporal interpolation were performed on the entire frame) in a part of the frame. Petition 870250092380, dated 09 / 10 / 2025, page 45 / 153 38 / 121 where the actual stretching occurs, and the interpolation of parameters proceeds comparatively faster in a part of the frame where no time extension (stretching) occurs.

[0105] It was found that such interpolation of processing parameters is well adapted to the time scale modification and results in interpolated processing parameters that are well adapted to the time scale-modified intermediate audio signals.

[0106] In one embodiment, the audio processor is configured to obtain interpolated processing parameters Pinter[n] associated with time instances with time index n according to Pinter[n]=ga,f[n] P + (1 -ga,f)Pprev, where P is a processing parameter or a set of processing parameters (for example, a scalar parameter, or a vector of parameters, or a matrix of parameters) associated with a current frame of one or more intermediate audio signals, where Pprev is a processing parameter or a set of processing parameters (for example, a scalar parameter, or a vector of parameters, or a matrix of parameters) associated with a previous frame of one or more intermediate audio signals, where Pinter is a processing parameter or a set of processing parameters (for example, a scalar parameter, or a vector of parameters, or a matrix of parameters) associated with a time instance n in a current frame of one or more intermediate audio signals modified in the time scale, and where ga,f is an interpolation function.

[0107] It was found that such determination of interpolated processing parameters, which effectively constitutes a linear combination controlled by a single scaling interpolation function, and which can also be considered as a panning (controlled by the interpolation function) between different processing parameters or sets of processing parameters, is computationally efficient and yields good results.

[0108] In one embodiment of the audio processor, the interpolation function ga,f is defined as íiLfW = max (O,£Π[?Φ θ where Petition 870250092380, dated 09 / 10 / 2025, p. 46 / 153 39 / 121 f η + 1 + ~ LreJ H = -I (Crt + l) 2LSeg+ 1 — £ re?Ji 21·„βηι·„β+ 1) ,0 <n<L„sem que n é um índice de tempo, em que Lrem=Lf-Lseg, em que Lf é urn comprimento de urn quadro a ser processado (por exemplo, um comprimento de um quadro atual de um ou mais sinais de áudio intermediários modificados na escala de tempo; por exemplo, um comprimento que é diferente de um comprimento de um quadro original de um ou mais sinais de áudio intermediários originais) e em que Lseg é o comprimento de uma porção na qual uma sobreposição e adição é aplicada pela modificação da escala de tempo.

[0109] It was recognized that such a choice of interpolation function, which brings different slopes of the interpolation functions in an initial portion of an audio frame and in a final portion of the audio frame, is well suited to a time scaling modification in which a final portion of a frame is left substantially unchanged, while an overlay and addition operation of time compression or time extension is performed on an initial portion of a given audio frame. It was recognized that the piecewise linear interpolation function, having two parts of different interpolation slopes, is well suited to an advantageous implementation of time scaling modification and can be implemented with reasonable effort.Furthermore, the interpolation function can be adapted to the actual frame length of the timescale-modified intermediate audio signals, although it should be noted that the frame length of the timescale-modified intermediate audio signals is typically variable and signal-dependent. Thus, such a choice of interpolation function has been found to be particularly advantageous.

[0110] In one embodiment, the audio processor is configured to perform processing (e.g., time-domain processing) of values ​​of one or more intermediate audio signals modified in the time scale (e.g., x[n] or xk[n]) according to y[«] = (ga,f [ti]M + (1 - gaif[n])MJB.e„)x[nL0 < τι < Lz or according to ytM = (j« / [n]Mk+ (1 < n < 1 / , Petition 870250092380, dated 09 / 10 / 2025, page 47 / 153 40 / 121 where x[n] and xk[n] are values ​​or sets of values ​​of one or more time-scaled modified intermediate audio signals (e.g., time-domain values ​​or spectral-domain values), where M is a processing parameter or a set of processing parameters (e.g., a scalar parameter, a parameter vector, or a parameter matrix) associated with a current frame of one or more intermediate audio signals, where Mprev is a processing parameter or a set of processing parameters (e.g., a scalar parameter, a parameter vector, or a parameter matrix) associated with a previous frame of one or more intermediate audio signals, where n is a time index, where Lf is a length of a frame to be processed (e.g., a length of a current frame of one or more time-scaled modified intermediate audio signals;for example, a length that is different from the length of an original frame of one or more original intermediate audio signals) and where ga,f is an interpolation function.

[0111] It has been recognized that such processing of values ​​of one or more timescale-modified audio signals, to derive one or more output signals y[n] or yk[n], yields particularly good processing results. In particular, the effects of timescale modification can be efficiently taken into account by such processing, where, for example, the timescale function mentioned above can be efficiently applied.For example, interpolation between processing parameters (or sets of processing parameters, or matrices of processing parameters) associated with the previous frame and the current frame can be performed with very high temporal resolution, i.e., by processing step, where a processing step can be a matrix-vector modification between a vector of intermediate audio signals modified on the x or xk time scale and a processing matrix (e.g., a mixing matrix or a rendering matrix) obtained by parameter interpolation.

[0112] It has been revealed that such a concept can be used efficiently for different processing operations, such as an upmix, a rendering or similar that uses some processing parameters that are, for example, associated with frames or subframes of the original intermediate audio signals. Petition 870250092380, dated 09 / 10 / 2025, page 48 / 153 41 / 121

[0113] In conclusion, it was recognized that this concept is efficient and allows the provision of high-quality audio output signals, even in the presence of a timescale modification.

[0114] In one embodiment, the audio processor is configured to perform secondary processing (e.g., a rendering or a mixing of the time-scaled modified intermediate audio signals) using processing parameters (e.g., metadata sets) associated (e.g., individually associated) with subframes of an audio frame (e.g., so that there are multiple metadata sets per frame; e.g., so that there is one metadata set per subframe), wherein, for example, a rendering rule or a rendering matrix or a mixing rule or a mixing matrix is ​​derived from the processing parameters, or wherein, for example, the processing parameters define a rendering rule or a rendering matrix or a mixing rule or a mixing matrix.

[0115] By using processing parameters associated with subframes of an audio frame in the second processing, high temporal resolution can be achieved. For example, there may be one processing parameter per subframe or there may be a set of processing parameters per subframe. Thus, the second processing can react quickly to changes in audio content, at a subframe temporal resolution.

[0116] Thus, good reproduction or rendering of audio content can be achieved through second processing.

[0117] In one embodiment, the audio processor is configured to obtain a processing rule for the second processing using processing parameters associated with a given subframe (e.g., from one or more timescale-modified intermediate audio signals) and, optionally, using signal characteristics of one or more timescale-modified intermediate audio signals in the given subframe (e.g., using a combined energy of one or more timescale-modified intermediate audio signals).

[0118] By obtaining a processing rule for the second processing using processing parameters associated with a Petition 870250092380, dated 09 / 10 / 2025, page 49 / 153 42 / 121 given subframe, fine temporal granularity can be achieved, which provides good quality reproduction or rendering of audio content. Furthermore, by taking into account the signal characteristics of one or more timescale-modified intermediate audio signals in the given subframe when determining the processing rule, the signal characteristics can be considered in determining the processing rule, so that a reasonably small amount of metadata (or secondary information) can be kept. For example, the combined energy of one or more timescale-modified intermediate audio signals can be used to adjust the processing rule for the second processing, which typically results in processing well-adapted to the actual signal characteristics without requiring a high bit rate.

[0119] In one embodiment, the audio processor is configured to determine a mapping vector that defines an association between time intervals (e.g., of one or more timescale-modified intermediate audio signals) and subframe indices (e.g., of subframes of one or more original intermediate audio signals) depending on the timescale modification, for example, such that the mapping describes which processing parameters are allocated with (or best fit) which time interval of the timescale-modified intermediate audio signals.

[0120] It has been found that such a concept, in which an association between time intervals and subframe indices is determined depending on a timescale modification, is useful for determining processing parameters with good temporal resolution.For example, it can be determined in which subframe of the original intermediate audio signals a given timescale interval on which the modified intermediate audio signals are based (or primarily based) belongs, and this association can be reflected in the mapping vector. Thus, each timescale interval of the modified intermediate audio signals can be associated with a subframe of the original intermediate audio signals in the mapping vector, and this association can be used to determine processing parameters (or metadata) for frames or subframes of the modified intermediate audio signals. In this regard, it should be noted that there is an association of timescale intervals to subframes. Petition 870250092380, dated 09 / 10 / 2025, page 50 / 153 43 / 121 [n] — max í 0. The timescaled modified intermediate audio signals may differ from the original intermediate audio signals. However, by having an (original) subframe (and consequently certain metadata) associated with each time interval of the scaled modified intermediate audio signal, determining processing parameters (or metadata) associated with a subframe of the timescaled modified intermediate audio signals is easily possible.

[0121] In one embodiment, the audio processor is configured to determine the mapping vector according to Γ Ί — jir I íπ ιιρ[Π]]\ jv [n] - modulo(nDffset. + max I 0,----------------------- , Lm(i) \ / or according to Lsf where mSf[n] is an element of the mapping vector; where n is a time interval index, and noffset is an offset value; where Lseg is half the length of a frame of one or more original intermediate audio signals at time intervals; where 2Lseg is the length of a frame of one or more original intermediate audio signals at time intervals; where Lsf is the length of a subframe in time intervals; where Lmd is the length of a temporary metadata storage; where noffset is ; and where ga,map[n] is an interpolation function; where L 1 denotes rounding to the nearest whole number; where modulo is a modulo operation, and where division is integer division.

[0122] It was found that such determination of the mapping vector is particularly efficient. In particular, the modification of the time scale can be taken into account by considering the interpolation function ga. However, it was Petition 870250092380, dated 09 / 10 / 2025, page 51 / 153 44 / 121 recognized that such a mapping vector is suitable for deriving processing parameters (or metadata) associated with time intervals or subframes of the timescale-modified intermediate audio signal.

[0123] In one modality, ' ^Seg —n< a,map ,0 <n<Lsegem que Lf é um comprimento de um quadro a ser processado em intervalos de tempo; e em que Lrem=Lf-Lseg.

[0124] In one embodiment, the audio processor is configured to determine a mapping vector that defines an association between time intervals of one or more timescale-modified intermediate audio signals and time intervals of one or more original intermediate audio signals depending on the timescale modification (where, for example, a temporary storage stores metadata, for example, transmitted metadata, in an interval-by-interval manner (even if the metadata is provided by subframe, where a subframe comprises a plurality of intervals)).

[0125] In one embodiment, the audio processor is configured to determine the mapping vector according to = max (O,[(2Δ30ίT- 1)#π,ίηπρ[n]j) where n is a time interval index, where 2Lseg is the length of a frame of one or more original intermediate audio signals at time intervals; and where ga,map[n] is an interpolation function (e.g., as described above).

[0126] In one embodiment, the audio processor is configured to determine a mapping vector that defines an association between time intervals of one or more timescale-modified intermediate audio signals and time intervals of one or more original intermediate audio signals depending on the timescale modification, for example, so that the mapping vector describes which processing parameters are allocated with (or Petition 870250092380, dated 09 / 10 / 2025, page 52 / 153 45 / 121 best fits a) which time interval of the intermediate audio signals modified on the time scale (e.g., if the processing parameters are available by time interval, for example, in the input audio representation (e.g., in an encoded form) or as a result of an analysis performed in the first processing).

[0127] After defining such an association between time intervals of one or more timescale-modified intermediate audio signals and time intervals of one or more original intermediate audio signals, a particularly high temporal resolution can be achieved. For example, the temporal resolution can be significantly finer than a subframe (where, for example, a subframe can comprise a plurality of time intervals). Thus, after using the mapping vector that indicates which processing parameters (such as metadata) are available (or should be used) with a very high temporal resolution, for example, by time interval, an efficient and precise determination of the timescale-modified intermediate audio signals can be achieved.For example, the mapping vector can define, for instance, for each time interval of the timescale-modified intermediate audio signals, which time interval of the original intermediate audio signals the time interval of the timescale-modified intermediate audio signals is most closely associated with. Thus, the mapping vector can be efficiently derived based on knowledge of the timescale modification. Consequently, the mapping vector can determine which processing parameters (e.g., metadata) (which are typically defined in association with time intervals of the original intermediate audio signals) best fit the different time intervals of the timescale-modified intermediate audio signals.Thus, the mapping vector that defines the association between time intervals of one or more timescaled intermediate audio signals and time intervals of one or more original intermediate audio signals is very useful in a derivation of processing parameters for the second processing.

[0128] In one mode, the audio processor is configured to determine processing parameters associated with a given subframe. Petition 870250092380, dated 09 / 10 / 2025, page 53 / 153 46 / 121 of one or more timescaled intermediate audio signals using associated processing parameters (e.g., metadata) (e.g., according to the mapping vector, which may be a metadata mapping vector) with time intervals that make up the given subframe of one or more timescaled intermediate audio signals (e.g., using an average of processing parameters associated with time intervals that make up the given subframe). The audio processor may, for example, be configured to obtain metadata associated with the given subframe using an average of metadata associated with the time intervals that make up the given subframe, or the audio processor may be configured to obtain rendering rules or upmixing rules or intermediate parameters by intervals using processing parameters associated with the respective intervals (e.g.,using metadata for the interval) (and optionally also using signal characteristics of one or more original intermediate audio signals or one or more timescaled modified intermediate audio signals; for example, using signal characteristics of the transport channel) and to obtain a rendering rule or an upmix rule or intermediate parameters associated with a subframe using an average of the rendering rules or upmix rules or intermediate parameters associated with the intervals that make up the subframe.

[0129] It has been revealed that, in the presence of high temporal resolution metadata or high temporal resolution processing parameters, the determination of processing parameters (or metadata) associated with a given subframe of one or more timescale-modified intermediate audio signals can be reliably performed by taking into account processing parameters (or metadata) associated with time intervals that make up the given subframe. It has been revealed that, in some situations, it is possible to associate the time intervals of the timescale-modified intermediate audio signals with the corresponding time intervals of the original intermediate audio signals. Consequently, it is often possible to determine processing parameters (e.g., metadata) associated with the time intervals of the timescale-modified intermediate audio signals. Consequently, Petition 870250092380, dated 09 / 10 / 2025, p. 54 / 153 47 / 121 it is also efficiently possible to determine processing parameters (e.g., metadata) associated with a subframe of the timescale-modified intermediate audio signals based on processing parameters (e.g., metadata) associated with those time intervals (of the timescale-modified intermediate audio signals) that make up the subframe (of the timescale-modified intermediate audio signals). Thus, while it is not normally possible to have a clear association between a subframe of the timescale-modified intermediate audio signals and a subframe of the original intermediate audio signals (due to the timescale modification), a temporal granularity of the time intervals can be fine enough to make such an allocation meaningful.Consequently, using processing parameters associated with time intervals as an intermediate quantity to determine processing parameters associated with a subframe yields reliable results efficiently.

[0130] In one embodiment, the audio processor is configured to adapt an interpolation (for example, of processing parameters, such as parameters that define a mixing rule) between intervals of a subframe to a subframe size obtained by modifying the time scale (for example, if a size in time intervals of a subframe is changed as a result of modifying the time scale).

[0131] After adapting the interpolation between time intervals of a subframe to the subframe size obtained by modifying the time scale, it is possible to efficiently handle situations in which the size of the subframes varies depending on the modification of the time scale. Thus, the processing parameters (e.g., metadata) associated with a subframe can be reliably and efficiently obtained depending on the actual size of the subframe (e.g., measured in terms of time intervals).

[0132] In one embodiment, the audio processor is configured to modify (e.g., shift) a grid of subframes of one or more intermediate audio signals modified in timescale, depending on a timescale modification of a current frame (and optionally also depending on timescale modifications of one or more previous frames), wherein, for example, the lengths of the subframes of one or more audio signals Petition 870250092380, dated 09 / 10 / 2025, page 55 / 153 48 / 121 intermediates modified in the timescale can be left unchanged.

[0133] After shifting a subframe grid of one or more timescale-modified intermediate audio signals, depending on the timescale modification, it is possible to have subframes of equal lengths, for example, in terms of a number of time intervals, even if there is a timescale modification. For example, the subframe grid can be varied so that the allocation of time intervals to the subframes is altered. However, the subframe grid can even change with finer resolution than the duration of a time interval. Consequently, the time-domain values ​​or the frequency-domain values ​​of the timescale-modified intermediate audio signals can be used efficiently, and a subframe-based operation of the second processing can remain unchanged even in the presence of a timescale modification.For example, an overlap of subframes can be avoided and the subframe grid can be adjusted so that the temporary storage effort is kept small or that temporary storage can even be avoided. Thus, modifying the subframe grid has proven to be an efficient solution for the subsequent processing of timescale-modified intermediate audio signals.

[0134] In one embodiment of the audio processor, the first processing is an audio decoding (where, for example, the audio processor comprises an audio decoder to perform the first processing).

[0135] It has been recognized that it is advantageous to perform timescale modification only after audio decoding, since it is often not possible to perform timescale modification with a reasonably fine temporal granularity before audio decoding, as audio decoding typically operates on comparatively large data blocks to achieve reasonable encoding efficiency. However, it has been recognized that implementing timescale modification between audio decoding and subsequent processing (second processing) is usually a very efficient solution, since subsequent processing typically increases the amount of audio information and therefore makes timescale modification after the second processing more challenging. Petition 870250092380, dated 09 / 10 / 2025, page 56 / 153 49 / 121

[0136] In conclusion, it was found that the use of timescale modification after a first processing, which is a decoding, constitutes an efficient solution, in particular if there is an additional processing (second processing) that follows the decoding and that is carried out based on the intermediate audio signals modified in the timescale.

[0137] In one embodiment of the audio processor, the first processing comprises at least one of the following options: decoding, rendering, reconstruction, and encoding, and / or the second processing comprises at least one of the following options: decoding, rendering, reconstruction, and encoding.

[0138] It has been recognized that the introduction of timescale modification between such processing functionalities is efficient and can result in good audio quality, where it should be noted that an adaptation of the processing parameters, which may be necessary in the second processing after the timescale modification, can usually be performed with moderate effort. Furthermore, it should be noted that the “second” processing may, for example, comprise a plurality of combined or separate processing steps.In other words, it was recognized that, in many cases, it is sufficient to have the timescale modification in the middle of a multi-step processing chain that may comprise two or more processing functionalities.

[0139] In one embodiment, the audio processor is configured to receive the representation of the input audio signal from an input temporary storage (where the input temporary storage may or may not be part of the audio processor).

[0140] It has been recognized that the use of a temporary input store, which can be organized in a processing order before the first processing, is useful for handling variation (e.g., packet variation including a representation of the input audio signals). For example, this temporary variation store can be useful for compensating for large packet delays relatively quickly. However, the presence of timescale modification in the audio processor can allow the size of the temporary input store to remain reasonably small, since the timescale modification can adapt to a relatively large time variation, provided that the large time variation occurs on a timescale. Petition 870250092380, dated 09 / 10 / 2025, page 57 / 153 50 / 121 is reasonably long. Thus, it should be noted that the combination of an input temporary storage (which may be a variation temporary storage) and which may, for example, store a plurality of packets comprising an encoded representation of the input audio signals, and a time-scaling modification allows handling both short-term and long-term variation. In conclusion, it was found that the structure of having an input temporary storage, a first processing, and a time-scaling modification after the first processing, as well as a second processing that is executed after the time-scaling modification, is advantageous in many situations.

[0141] In one embodiment, the audio processor is configured to receive the input audio signal representation from a temporary variation store (wherein the temporary variation store may or may not be part of the audio processor), and the audio processor comprises a time scale control to adjust the time scale modification. The time scale control is a temporary variation store control (which may, for example, adjust the time scale modification depending on the completeness of the temporary variation store and / or depending on the characteristics of a packet variation of packets that make up the input audio representation and / or depending on a user setting).

[0142] As described above, it has been found that the use of a temporary variation store is advantageous as it can be useful to compensate for packet variation. Furthermore, it has been recognized that the use of a time scale control, which is a temporary variation store control, is particularly advantageous as it allows adapting the time scale modification to a fullness of the temporary variation store or to a change in the fullness of the temporary variation store. Consequently, the time scale modification can be used, under the control of the time scale control, to prevent an overflow or overflow of the temporary variation store.It was also recognized that placing the timescale modification after the first processing (which could, for example, be a decoding of an encoded audio representation that is stored, for example, in a packaged format, in the temporary variation storage) constitutes a solution. Petition 870250092380, dated 09 / 10 / 2025, page 58 / 153 51 / 121 computationally efficient, providing good audio quality.

[0143] In one embodiment of the audio processor, the first processing comprises a decoding of encoded transport channels (wherein, for example, the audio processor comprises a decoder, and wherein, for example, an encoded representation of one or more transport channels may compose the input audio representation, and wherein, for example, a decoded representation of one or more transport channels may compose the intermediate (original) audio signals). Furthermore, the second processing comprises a rendering (e.g., a renderer) configured to reconstruct an encoded or intermediate audio format based on timescale-modified transport channels and associated metadata and to render the encoded or intermediate audio format into an output format (wherein the timescale-modified transport channels may be the timescale-modified intermediate audio signals).

[0144] This concept is based on the observation that decoded transport channels are normally suitable for applying time-scaling modification, since decoded transport channels are normally represented with sufficiently fine temporal granularity, for example, in the form of time-domain samples or in the form of frequency-domain samples, so that time-scaling modification is efficiently possible. Furthermore, decoded transport channels generally comprise a comparatively small volume of data, since decoded transport channels often form the basis for an upmix / rendering in which a number of audio signals are increased, and since decoded transport channels also sometimes form the basis for a bandwidth extension that increases the amount of data.Thus, it was recognized that it is more advantageous to apply the time scale modification to the decoded transport channels than to the output signals derived from the decoded transport channels in the second processing.

[0145] In one embodiment of the audio processor, the rendering (e.g., a renderer) is configured to reconstruct the encoded audio format or the intermediate audio format and render the encoded audio format or the intermediate audio format to the output format in two Petition 870250092380, dated 09 / 10 / 2025, page 59 / 153 52 / 121 consecutive processing steps (e.g., separate).

[0146] It has been revealed that such a multi-step procedure is computationally efficient, since the reconstruction of the encoded audio format or the intermediate audio format and the rendering of the encoded audio format or the intermediate audio format to the output format can be performed as separate processing steps, for example, using separate processing parameters. Such processing can, for example, allow for easy implementation of systems in which different types of output formats can be handled.

[0147] In one embodiment of the audio processor, the audio processor functionalities are distributed to two or more devices, with an interface between them.

[0148] Such an implementation may, for example, be advantageous if a main processing functionality is performed on a device with greater computing power, while some processing functionality is performed by a device with less computing power. For example, the interface may be inserted at a point in a processing chain where the amount of data to be transmitted is transferable through the interface, while processing effort on a device with less processing power is kept reasonably small.For example, in some applications, it may be advantageous to route the timescale-modified intermediate audio signals to another device via an interface, since the timescale-modified intermediate audio signals typically have a comparatively low data rate (e.g., when compared to the output audio signals), while computationally challenging decoding and timescale modification can be performed on a powerful device with high processing capacity.

[0149] Thus, the concept of distributing the audio processor functionalities to two or more devices can be advantageous in many situations.

[0150] In one embodiment, the audio processor comprises an interface configured to provide the timescale-modified intermediate audio signals (e.g., timescale-modified decoded transport channels) and associated metadata (e.g., metadata associated with the channels of Petition 870250092380, dated 09 / 10 / 2025, page 60 / 153 53 / 121 decoded transport modified in timescale, which can be adapted to timescale modification or dependent on timescale modification) for subsequent rendering.

[0151] It has been recognized that timescale-modified intermediate audio signals and associated metadata can be efficiently transmitted via an interface (e.g., between different and possibly spatially separated devices). In this way, it is possible to distribute functionalities advantageously, making the most of the processing power and interface capabilities of the respective equipment.

[0152] In one embodiment of the audio processor, the second processing is a rendering, where the rendering (e.g., a renderer) is configured to process a reconstruction of an encoded audio format and / or an intermediate audio format and a rendering of the reconstructed encoded audio format and / or the intermediate audio format in a combined processing step.

[0153] It has been revealed that, in some cases, it may be advantageous to process a reconstruction of the encoded audio format and / or an intermediate audio format and a rendering of the reconstructed encoded format and / or the intermediate audio format in a combined processing step. For example, it has been recognized that a single matrix mixing operation can be used to perform the processing, which can, for example, reduce complexity and eliminate the need to store (or temporarily store) intermediate signals. Thus, the concept described in this document is efficient in some processing scenarios.

[0154] In one embodiment of the audio processor, one or more intermediate audio signals are time-domain signals, and the audio processor is configured to perform time-scaling modification on the intermediate time-domain audio signals (e.g., in a time domain).

[0155] It has been recognized that time-domain signals are efficient in many cases. For example, it has been recognized that timescale modification can be efficiently performed based on time-domain signals, since a correlation, which is useful for high-quality timescale modification, can be easily performed in the time domain. Furthermore, also Petition 870250092380, dated 09 / 10 / 2025, page 61 / 153 54 / 121 recognized that time-domain signals are suitable for transitioning between different transformation domains (e.g., different spectral domains or frequency domains). Thus, it was found that using one or more time-domain signals as one or more intermediate audio signals is an advantageous solution.

[0156] In a preferred embodiment, the audio processor (e.g., an audio processor time scaler) is configured to synchronize a time scaling operation over a plurality of intermediate audio signals (e.g., using a multi-channel time scaler that provides the same number of time domain samples for each intermediate audio signal modified in the time scale).

[0157] It has been recognized that synchronizing the time scaling operation of a plurality of intermediate audio signals helps to reduce audible artifacts in the case of multiple intermediate audio signals. For example, synchronizing the time scaling operation over a plurality of intermediate audio signals can perform multiple correlations to determine an appropriate common time offset that allows for good quality overlay and addition operation for all intermediate audio signals. In this way, audible artifacts can be reduced to a tolerable level.

[0158] In one embodiment, the audio processor is configured to modify transient position information depending on the timescale change.

[0159] In one embodiment, the audio processor is configured to obtain an adapted transient position according to nT,a = niax(0,íiT + (Lf - 2^„)) where ητ is an original transient position value, where Lf is a frame length of one or more time-scaled modified intermediate audio signals; and where 2Lsg is the frame length of one or more original intermediate audio signals. It should be noted that ητ, Lf, and Lsg should preferably be expressed with the same time resolution, for example, with the time resolution of the transient position value (e.g., with a time resolution of one sample, or with a time resolution of a given number of Petition 870250092380, dated 09 / 10 / 2025, page 62 / 153 55 / 121 samples, or with a time resolution of one time interval, or with a time resolution of one subframe, or with a time resolution of two time intervals).

[0160] One embodiment of the invention creates an audio processing system (e.g., an audio decoder) to provide a processed (e.g., decoded and rendered) audio signal representation (e.g., in the form of a plurality of rendered time-domain audio signals) based on an input audio signal representation (e.g., an encoded representation of a multi-channel audio content). The audio processing system is configured to perform a plurality of processing steps (e.g., separate processing functionalities; e.g., a first processing and a second processing) in order to provide the processed audio signal representation based on the input audio signal representation.The audio processing system is configured to perform a timescale modification (e.g., a modification of the length of a portion of an audio signal, e.g., an audio frame, provided by the first processing) on ​​one or more intermediate audio signals, which are provided by a first processing (e.g., a first processing step), in order to obtain one or more timescale-modified intermediate audio signals. Furthermore, the audio processing system is configured to perform a second processing (e.g., a second processing step; e.g., an upmix and / or a rendering), which follows the first processing (e.g., the first processing step), based on the one or more timescale-modified intermediate audio signals or based on signals derived from the one or more timescale-modified intermediate audio signals.The functionalities of the audio processing system are distributed across two or more devices.

[0161] It is worth noting that this audio processing system is based on the same configurations as the audio processor mentioned above. However, it should be noted that the different functionalities of the audio processing system can be distributed to different physical entities, for example, to physically separate devices. However, it is recognized that the distribution of Petition 870250092380, dated 09 / 10 / 2025, page 63 / 153 56 / 121 functionalities for two or more devices brings advantages in several situations, for example, in situations where processing must be distributed among devices with different processing power, when interface capabilities also need to be considered. In conclusion, the audio processing system allows an adaptation of the concept presented in this document for distributed applications.

[0162] Furthermore, it should be noted that the audio processing system may optionally be supplemented by any of the features, functionalities and details disclosed in this document, for example, with regard to the audio processor, either individually or in combination.

[0163] In one embodiment, the audio processing system comprises an interface for providing timescale-modified intermediate audio signals (e.g., timescale-modified decoded transport channels) and adapted metadata associated with a second processing (e.g., a rendering) that takes place on a device different from the first processing.

[0164] This implementation is based on the finding that timescale-modified intermediate audio signals and associated metadata can typically be transferred efficiently from one device to another, since this information typically comprises a sufficiently small bitrate to be suitable for efficient exchange via an interface. Furthermore, reference is made to the above discussion of distributed audio processor implementations.

[0165] One embodiment of the invention creates an audio decoder to provide a decoded audio representation based on an encoded audio representation. The audio decoder comprises an audio processor as disclosed in this document. For example, the first processing comprises a decoding of one or more encoded transport channels, wherein, for example, one or more intermediate audio signals are one or more decoded transport channels. For example, the second processing comprises a rendering or mixing of one or more time-scale modified intermediate audio signals, wherein, for example, rendered or mixed audio channels compose the audio representation. Petition 870250092380, dated 09 / 10 / 2025, page 64 / 153 57 / 121 processed.

[0166] It has been recognized that the audio processor disclosed in this document is suitable for use in an audio decoder. In particular, the audio processor disclosed in this document is well usable in an audio decoder comprising multiple processing stages or processing steps, wherein one or more of these steps may be performed as the first processing and one or more other processing steps may be performed as the second processing. In other words, it has been recognized that the audio processor (and the concept of performing a time-scaling modification on an intermediate audio signal) is well applicable in an audio decoder.

[0167] An embodiment of the invention creates a method for providing a processed (e.g., decoded and rendered) audio signal representation (e.g., in the form of a plurality of rendered time-domain audio signals) based on an input audio signal representation (e.g., an encoded representation of a multi-channel audio content). The method comprises performing a plurality of processing steps (e.g., separate processing functionalities; e.g., a first processing and a second processing) in order to provide the processed audio signal representation based on the input audio signal representation.The method comprises performing a timescale modification (e.g., a modification of the length of a portion of an audio signal, e.g., an audio frame, provided by the first processing) on ​​one or more intermediate audio signals, which are provided by a first processing (e.g., a first processing step), in order to obtain one or more timescale-modified intermediate audio signals, and the method comprises performing a second processing (e.g., a second processing step; e.g., an upmix and / or a rendering), which follows the first processing (e.g., the first processing step), based on the one or more timescale-modified intermediate audio signals.

[0168] The method is based on the same considerations as the audio processor disclosed in this document. Consequently, the method may optionally be Petition 870250092380, dated 09 / 10 / 2025, page 65 / 153 58 / 121 supplemented by any of the features, functionalities and details disclosed here regarding the audio processor, both individually and in combination.

[0169] One embodiment of the invention creates a computer program to perform the method as disclosed herein when the computer program is run on a computer. BRIEF DESCRIPTION OF THE FIGURES

[0170] The embodiments according to the invention will be described with reference to the accompanying figures, in which: Figure 1 shows a schematic block diagram of an audio processor, according to one embodiment of the present invention; Figure 2 shows a schematic block diagram of an audio processing system, according to an embodiment of the present invention; Figure 3 shows a schematic block diagram of an audio decoder, according to one embodiment of the present invention; Figure 4 shows a flowchart of a method, according to an embodiment of the invention; Figure 5a shows a schematic block diagram of the modules of the EVS variable temporary storage management solution (Figure 2 of [2]); Figure 5b shows a schematic block diagram of a receiver output temporary storage (Figure 7 of [2]); Figure 5c shows a schematic representation of a shortening of an input frame (Figure 5 of [2]), which can optionally be used in modifying the time scale of modes according to the invention; Figure 5d shows a schematic representation of an extension of an input frame (Figure 6 of [2]), which can optionally be used in modifying the time scale in embodiments according to the invention; Figure 5e shows a schematic representation of a frequency domain rendering with subframes; Figure 6 shows a generic block diagram of IVAS with parametric reconstruction; Figure 7 shows a schematic block diagram of a processor. Petition 870250092380, dated 09 / 10 / 2025, page 66 / 153 59 / 121 of generic multistage audio, according to an embodiment of the invention; Figure 8 shows a schematic block diagram of IVAS JBM (IVAS Variation Temporary Storage Management) with TSM (Time Scale Modification) on the output channels, according to one embodiment of the invention; Figure 9 shows a schematic block diagram of a TSM (time scale modification) audio processor on an intermediate audio signal, according to one embodiment of the invention; Figure 10 shows a schematic block diagram of an audio processor with parameters and TSM (time scale modification) on an intermediate audio signal, according to one embodiment of the invention; Figure 11 shows a schematic block diagram of an audio processor with TSM in an intermediate and time-scaling audio signal processor, according to one embodiment of the invention; Figure 12 shows a schematic block diagram of an audio processor with TSM on an intermediate audio signal, processing parameter adapters and a time scale controller, according to one embodiment of the invention; Figure 13 shows a schematic block diagram of a speech playback control, according to one embodiment of the invention; Figure 14 shows a schematic block diagram of an IVAS JBM (variable temporary storage management), according to one embodiment of the invention; Figure 15a shows a schematic representation of a temporary receiver storage for the IVAS JBM (variation temporary storage management), according to one embodiment of the invention; Figure 15b shows a schematic representation of a temporary receiver storage for the IVAS JBM (variation temporary storage management), according to another embodiment of the invention; Figure 16 shows a schematic representation of a temporary TC (transport channel) frame storage for FD rendering. Petition 870250092380, dated 09 / 10 / 2025, page 67 / 153 60 / 121 according to an embodiment of the invention; Figure 17 shows a schematic representation of a frequency domain (FD) rendering with adapted subframe size, according to an embodiment of the invention; Figure 18 shows a schematic representation of a rendering with constant subframe size and metadata adaptation, according to an embodiment of the invention; and Figure 19 shows a graphical representation of a concept for adapting interpolation functions, according to one embodiment of the invention. DETAILED DESCRIPTION OF THE MODALITIES 1. Audio processor as shown in Figure 1

[0171] Figure 1 shows a schematic block diagram of an audio processor 100, according to one embodiment of the invention. The audio processor 100 is configured to receive an input audio signal representation 110 and provide, based on it, a processed audio signal representation 112, which can also be considered as an output audio signal representation. The audio processor 100 is configured to perform a plurality of processing steps in order to provide the processed audio signal representation 112 based on the input audio signal representation. In particular, the audio processor 100 comprises a first processing 120, which receives the input audio signal representation 110 and provides, based on it, one or more intermediate audio signals 122.For example, the first processing 120 may comprise a first processing step, but optionally, the first processing 120 may comprise a plurality of processing steps.

[0172] The audio processor 100 is additionally configured to perform a time-scale modification on one or more intermediate (original) audio signals 122, which are provided by the first processing 120, in order to obtain one or more time-scale modified intermediate audio signals 132. In other words, the time-scale modification 130 receives one or more intermediate audio signals 122, performs a time-scale modification, and provides, as a result of the time-scale modification, one or more time-scale modified intermediate audio signals 132. The audio processor 100 is Petition 870250092380, dated 09 / 10 / 2025, page 68 / 153 61 / 121 is additionally configured to perform a second processing, which follows the first processing, based on one or more timescale-modified intermediate audio signals. For example, audio processor 100 may comprise a second processing 140 that receives one or more timescale-modified intermediate audio signals 132, processes one or more timescale-modified intermediate audio signals, and provides, as a result of the processing, the representation of the processed audio signal 112. For example, the second processing 142 may comprise a second processing step. However, the second processing may optionally comprise a plurality of processing steps to derive the representation of the processed audio signal 112 based on one or more timescale-modified intermediate audio signals 132.

[0173] The audio processor 100 is based on the finding that it is advantageous to perform timescale modification based on one or more intermediate signals, that is, between a first processing, which is performed before the timescale modification, and a second processing, which is performed after the timescale modification and using one or more intermediate audio signals modified in timescale provided by the timescale modification.In particular, it has been recognized that a modification in the time scale between different processing steps, which provide a processed audio signal representation 112 (for example, a decoded and rendered audio signal representation or a decoded and reconstructed audio signal representation) based on an input audio signal representation, is often more efficient than applying a time scale modification to the final output signals provided at the end of a processing chain.For example, the number of channels in the output of the first processing 120 may be less than the number of channels in the representation of the processed audio signal 112, so that applying the timescale modification 130 to one or more intermediate audio signals 122 requires less effort (e.g., computational resources) than a possible application of a timescale modification to the final representation of the processed audio signal 112. For example, it has been found that it is advantageous to apply the timescale modification 130 before a second processing step, which increases the number of channels and / or. Petition 870250092380, dated 09 / 10 / 2025, page 69 / 153 62 / 121 which increases the bandwidth of audio channels, since applying a timescale modification to a smaller number of audio channels is easier than applying the timescale modification to a larger number of channels (e.g., rendered, mixed, or with extended bandwidth) of the representation of the processed audio signal 112, which is derived by additional processing (e.g., the second processing 140) based on one or more intermediate audio signals 122. However, it was also recognized that, in many situations, it is easier to apply the timescale modification 130 to one or more intermediate signals when compared to a timescale modification that would be applied to the representation of the input audio signal. For example, if the representation of the input audio signal 110 is in an encoded form, a timescale modification would be very complicated.Conversely, applying timescale modification after decoding (which can be done as part of the first processing or as the first processing step) was found to be efficient.

[0174] In conclusion, the 100 audio processor brings with it a good balance between computational complexity and achievable audio quality, where the application of time scale modification to one or more intermediate signals, i.e., between two processing stages, was considered very advantageous.

[0175] Furthermore, it should be noted that the audio processor 100 may optionally be supplemented by any of the features, functionalities and details disclosed in this document, either individually or in combination.

[0176] For example, the second processing may optionally comprise two or more processing functions (or processing steps) that may be substantially separate or that may be performed in combination. 2. Audio Processing System as per Figure 2

[0177] Figure 2 shows a schematic block diagram of an audio processing system according to one embodiment of the present invention. The audio processing system 200 according to Figure 2 is configured to receive an input audio signal representation 210 and provide, based on the same, a processed audio signal representation 212. The system of Petition 870250092380, dated 09 / 10 / 2025, p. 70 / 153 Audio processing system 200 comprises a first processing unit or a first processor 220, which receives the representation of the input audio signal 210 and provides, based on it, one or more intermediate audio signals 222. The audio processing system 200 further comprises a time-scaling modification or a time-scaling modifier 230, which receives one or more intermediate audio signals 222 and applies the time-scaling modification. Consequently, the time-scaling modification / time-scaling modifier 230 provides, as a result, one or more intermediate audio signals modified in time scale 232 for a second processing unit or for a second processor 240.The second processing / second processor 240 applies additional processing to one or more timescale-modified intermediate audio signals, or to signals derived from timescale-modified intermediate audio signals, and provides, as a result of the processing, the representation of the processed audio signal 212. For example, there may be an interface between the first processing 220 and the timescale modification 230. Alternatively or additionally, there may be an interface between the timescale modification / timescale modifier 230 and the second processing / second processor 240. For example, the first processing / first processor 220 and the second processing / second processor 240 may be distributed to two or more devices, for example, to two or more separate hardware devices that may, for example, be coupled using a wired link or using a wireless link.For example, the first processing / first processor 220 and the second processing / second processor 240 may be part of physically separate devices that together form the audio processing system 200. The time scale modification 230 may, for example, be included in the device comprising the first processing / first processor 220 or may be included in the device comprising the second processing / second processor 240.

[0178] In conclusion, the audio processing system can be adapted to provide a representation of the processed audio signal 212 based on the representation of the input audio signal 210. The audio processing system is configured to perform a plurality of processing steps, Petition 870250092380, dated 09 / 10 / 2025, page 71 / 153 64 / 121 in order to provide the representation of the processed audio signal 212 based on the representation of the input audio signal 210. The separate processing steps can, for example, be performed by the first processing / first processor 220 and the second processing / second processor 240. The audio processing system is configured to perform a timescale modification 230 on one or more intermediate audio signals 222 that are provided by the first processing 230, in order to obtain one or more intermediate audio signals modified in timescale 232. Furthermore, the audio processing system is configured to perform a second processing 240, which follows the first processing 220, based on the one or more intermediate audio signals modified in timescale 232.In particular, the functionalities of the audio processing system are distributed to two or more devices, where, for example, the first processing 220 and the second processing 240 can be included in separate devices.

[0179] However, it should be noted that the audio processing system 200 may optionally comprise a third processing / third processor 250, which may follow the second processing / second processor 240. For example, the third processing / third processor 250 may perform one or more additional processing functions that result in a processed (final) audio signal (e.g., an audio signal that can be sent to a user).

[0180] It should be noted that, optionally, the third processing can be performed on a different (physical) device than the first processing. For example, the third processing can be performed on the same device as the second processing. Alternatively, the third processing can be performed on a different (physical) device than the second processing (for example, with a wired or wireless interface between the second processing and the third processing).

[0181] For example, a split rendering can be implemented (i.e., transport channel (TC) decoding, binaural rendering for three different head positions, sending to an edge device with metadata, rendering completion on the edge device according to the actual head position). For example, transport channel decoding, a scaling modification of Petition 870250092380, dated 09 / 10 / 2025, page 72 / 153 65 / 121 time and binaural rendering (or pre-rendering) for three different head positions can be performed on a first device or even on two devices; the pre-rendering result and metadata (e.g., adapted metadata) can be transmitted to an edge device (e.g., via a wired or wireless interface), and the completion of rendering according to the actual head position can be performed on the edge device. However, a different distribution of functionalities across two or more hardware devices or hardware appliances is also possible.

[0182] However, it should be noted that the audio processing system 200 includes similar advantages to those of the audio processor 100.

[0183] Furthermore, it should be noted that the audio processing system 200 may optionally be supplemented by any of the features, functionalities and details disclosed in this document. In particular, the audio processing system 200 may optionally be supplemented by any of the features, functionalities and details disclosed with respect to the audio processor 100, either individually or in combination. 3. Audio decoder according to Figure 3

[0184] Figure 3 shows a schematic block diagram of an audio decoder 300, according to an embodiment of the present invention. The audio decoder 300 is configured to receive an encoded audio representation 310 and provide, based on the same, a decoded audio representation 312. In particular, the audio decoder 300 comprises an audio processor 340, wherein the audio processor 340 may, for example, correspond to the audio processor 100 according to Figure 1. For example, the first processing performed by the audio processor 340 may comprise a decoding of one or more encoded transport channels, which may, for example, be represented by the encoded audio representation. Consequently, one or more intermediate audio signals may be one or more decoded transport channels.For example, the second processing involves a rendering or mixing of one or more time-scaled modified intermediate audio signals, where, for example, the rendered or mixed audio channels in an ascending order compose the processed audio representation (and may form the decoded audio representation 312). Petition 870250092380, dated 09 / 10 / 2025, page 73 / 153 66 / 121 In other words, different functionalities of the audio decoder 300, such as, for example, transport channel decoding and up-rendering / mixing, can be performed by the audio processor 340, for example, as the first processing 120 and as the second processing 140. Thus, time scale modification can be performed between different functionalities that are executed by the audio decoder 300. Insofar as the audio decoder 300 comprises the same advantages as the audio processor 100.

[0185] In addition, it should be noted that the 300 audio decoder may optionally be supplemented by any of the features, functionalities and details disclosed in this document, for example, with regard to the audio processor. 4. METHOD ACCORDING TO FIGURE 4

[0186] Figure 4 shows a flowchart of a method 400 for providing a processed audio signal representation based on an input audio signal representation. The method 400 comprises performing 410 a first processing, to provide one or more intermediate audio signals. The method 400 further comprises performing 420 a time-scaling modification on one or more intermediate audio signals. The method 400 further comprises performing 430 a second processing based on the one or more time-scaling modified intermediate audio signals. Thus, the method can, for example, comprise (or perform) the functionality of audio processor 100 or audio processing system 200 or audio processor 340. Thus, the method 400 can bring the same advantages as audio processor 100 discussed above.

[0187] Furthermore, it should be noted that method 400 may optionally be supplemented by any of the features, functionalities and details disclosed in this document, for example, with regard to the audio processor, with regard to the audio processing system and with regard to the audio decoder, either individually or in combination. 5. Audio Processing According to Figure 9

[0188] Figure 9 shows a schematic block diagram of an audio processor / audio processing unit 900, according to an embodiment of the present invention. Petition 870250092380, dated 09 / 10 / 2025, page 74 / 153 67 / 121 invention. The audio processor according to Figure 9 is configured to receive an input audio signal representation 910 and provide, based on it, a processed audio signal 912 (or a plurality of processed audio signals). The audio processor 900 comprises a first processing stage (or a first processing) 920, which receives the input audio signal representation 910 and provides, based on it, an intermediate audio signal 1, which is designated 922. The audio processor 900 further comprises a time-scale modification (TSM) 930, which receives the intermediate audio signal 1 (designated 922) and provides, based on it, a time-scale modified intermediate audio signal 1, which is designated 932.However, it should be noted that the intermediate audio signal 1 can optionally be replaced by a plurality of intermediate audio signals, and that the timescale-modified intermediate audio signal 1 (932) can optionally be replaced by a plurality of timescale-modified intermediate audio signals (where it should be noted that one or more intermediate audio signals can be used in either mode). The timescale-modified intermediate audio signal 1 (932) is fed into a second processing stage 940, which provides an intermediate audio signal 2 (designated 942) based on the timescale-modified intermediate audio signal 1.Optionally, audio processor 100 may comprise one or more additional processing stages, for example, up to a processing stage n (or processing n, or processor n), which is designated 950, which further processes intermediate audio signal 2 (designated 942) or yet another intermediate audio signal derived from intermediate audio signal 2. For example, each processing stage (or each processing, or each processor) may provide a new intermediate audio signal, which is then fed into the subsequent processing stage (or processing or processor). Consequently, processing stage n 950 may receive an intermediate audio signal n-1. However, the last processing stage in the chain (in the present example, processing stage n, designated 950) provides the processed audio signal 912.

[0189] In this regard, it should be noted that in total there may be two processing steps (for example, processing step 1, designated with 920, and step Petition 870250092380, dated 09 / 10 / 2025, page 75 / 153 68 / 121 of processing 2, designated with 940). However, it should be noted that, optionally, there may be two or more processing steps before the timescale modification 930 and that, optionally, there may be two or more processing steps after the timescale modification 930.

[0190] In conclusion, Figure 9 shows an audio processor with a time-scaling modification on an intermediate audio signal. Consequently, the time-scaling modification is applied to an intermediate audio signal, which is provided by a processing step that precedes the time-scaling modification, and the time-scaling modification provides a time-scaled modified intermediate audio signal for a subsequent processing step.

[0191] However, it should be noted that the 900 audio processor may offer the same advantages that have already been discussed in relation to the 100 audio processor. In addition, the 900 audio processor may optionally be supplemented by any of the features, functionalities and details described in this document, either individually or in combination. 6. Audio processor according to Figure 10

[0192] Figure 10 shows a schematic block diagram of an audio processor with parameters and time scale modification (TSM) on an intermediate audio signal.

[0193] The audio processor 1000 according to Figure 10 is similar to the audio processor 900 according to Figure 9, so identical characteristics, functionalities and details will not be discussed again in this document. Instead, reference is made to the discussion of the audio processor 900 according to Figure 9.

[0194] However, audio processor 1000 is configured to receive an input audio signal representation 1010 that corresponds to input audio signal representation 910. Audio processor 1000 is configured to provide a processed audio signal 1012 based on the input audio signal representation 1010. Audio processor 1000 comprises a first processing stage 1020 which, in principle, corresponds to the first processing stage 920. However, the first processing stage 1020 of audio processor 1000 is configured to use a Petition 870250092380, dated 09 / 10 / 2025, page 76 / 153 69 / 121 or more processing parameters 1021 to adjust the processing performed by the first processing stage 1020. The first processing stage 1020 is configured to provide an intermediate audio signal 1, which is designated with 1022, based on the representation of the input audio signal 1010, where the first processing stage 1020 is controlled or adjusted using one or more processing parameters 1021. Regarding this matter, it should be noted that Figure 10 refers to a “processing parameter 1”. However, alternatively, a plurality of first processing parameters can be used by the first processing stage 1020. Furthermore, it should be noted that Figure 10 shows that the first processing stage provides an “intermediate audio signal 1”. However, alternatively, the first processing stage 1020 can provide a plurality of intermediate audio signals.

[0195] The audio processor 1000 comprises a timescale modification 1030, which corresponds to the timescale modification 930. The timescale modification 1030 provides an intermediate audio signal 1 modified in timescale, which is designated with 1032. Alternatively, however, the timescale modification 1030 may provide a plurality of intermediate audio signals modified in timescale.

[0196] The audio processor 1000 also comprises a second processing stage 1040, which receives the timescale-modified intermediate audio signal 1 (or a plurality of timescale-modified first intermediate audio signals). The second processing stage 1040 may, in principle, correspond to the second processing stage 940. However, the second processing stage 1040 uses a processing parameter 2, also designated 1041, to control or adjust the second processing stage 1040. Alternatively, however, the second processing stage 1040 may receive and use a plurality of processing parameters. Thus, the second processing stage 1040 provides an intermediate audio signal 2, which is designated 1042. However, alternatively, the second processing stage 1040 may provide a plurality of intermediate audio signals.

[0197] It should be noted that the first 1020 processing can provide any number of intermediate audio signals for scaling modification of Petition 870250092380, dated 09 / 10 / 2025, page 77 / 153 70 / 121 timescale 1030, where the 1030 timescale modification typically provides the same number of timescale-modified intermediate audio signals (1032). In other words, the 1030 timescale modification typically does not alter the number of intermediate audio signals. However, the second 1040 processing stage may, for example, receive a first number of timescale-modified intermediate audio signals from the 1030 timescale modification and provide a second number of intermediate audio signals or processed audio signals as an output. In other words, the second 1040 processing stage may, for example, perform an upmix or a downmix, where an upmix would be preferable.

[0198] Furthermore, the audio processor 1000 may optionally comprise one or more additional processing steps, for example, an nth processing step 1050. One or more optional additional processing steps (for example, the nth processing step 1050) may or may not use one or more processing parameters. Consequently, the audio processor 1000 obtains the processed audio signal 1012 as an output of a final processing step (wherein, in some embodiments, the second processing step 1040 may be the final processing step). Furthermore, it should be noted that in some cases it may be sufficient for only a single processing step (for example, of processing steps 1020, 1040, 1050) to use one or more processing parameters.

[0199] However, it is preferable that the second 1040 processing stage use one or more processing parameters to adjust the processing performed by the second processing stage. The processing parameters may, for example, be parameters that define a rendering operation performed by the second 1040 processing stage or an upmixing operation performed by the second 1040 processing stage or parameters that control a bandwidth extension functionality performed by the second processing stage, or any other processing parameters that adjust the processing performed by the second 1040 processing stage.

[0200] In conclusion, the functionality of the 1000 audio processor is similar to the functionality of the 900 audio processor, in that, however, one or more processing steps use one or more processing parameters, and in that Petition 870250092380, dated 09 / 10 / 2025, page 78 / 153 71 / 121 it should be noted that different processing steps may use different processing parameters, and that the number of processing parameters used by the different processing steps may vary, and in that, for example, each of the processing steps may use 0, 1 or more processing parameters. Furthermore, it should be noted that the temporal granularity of the processing parameters may also vary over a wide range. For example, a processing parameter, or a set of processing parameters, may be provided per time domain sample, or per frequency domain sample, or per time interval, or per subframe, or per frame, or even per block of frames.

[0201] However, it should be noted that the audio processor 1000 according to Figure 10 comprises similar advantages to those of the audio processor 100 according to Figure 1. In particular, by using the time scale modification 1030 between different processing stages (e.g., separate ones), good processing efficiency can be achieved and good audio quality can also be achieved.

[0202] Furthermore, it should be noted that the 1000 audio processor according to Figure 10 can optionally be supplemented by any of the features, functionalities and details disclosed in this document, either individually or in combination. 7. Audio processor according to Figure 11

[0203] Figure 11 shows a schematic block diagram of an 1100 audio processor with time scaling modification (TSM) on an intermediate audio signal and time scaling controller, according to embodiments of the present invention.

[0204] The 1100 audio processor is similar to the 100 audio processor and similar to the 1000 audio processor.

[0205] In particular, the audio processor 1100 is configured to receive an input audio signal representation 1110 and provide, based on the same, a processed audio signal representation 1112. The audio processor 1100 comprises a first processing stage 1120 which may correspond to the first processing stage 920 or the first processing stage 1020. The first processing stage 1120 provides a signal of Petition 870250092380, dated 09 / 10 / 2025, page 79 / 153 72 / 121 intermediate audio signal 1 (designated 1122) and may alternatively provide a plurality of first intermediate audio signals. The audio processor 1100 comprises a timescale modification 1130 which may, for example, correspond to timescale modification 930 or timescale modification 1030. However, timescale modification 1130 may receive a control signal 1134 from a timescale controller 1160 and may provide timescale information 1136 to the timescale controller 1160. Furthermore, timescale modification 1130 may be configured to provide a modified intermediate audio signal at timescale 1, which is designated 1132, for a second processing stage 1140.The timescale-modified intermediate audio signal 1, designated 1132, may be based on the timescale-modified intermediate audio signal 1, designated 1122, and may be derived from the timescale-modified intermediate audio signal 1 (1122) using a timescale modification. However, it should be noted that both the timescale-modified intermediate audio signal 1 and the timescale-modified intermediate audio signal 1 may optionally comprise a plurality of channels. The second processing stage 1140 may, for example, correspond to the second processing stage 940 and the second processing stage 1040, and may provide an intermediate audio signal 2, designated 1142, based on the timescale-modified intermediate audio signal 1132.However, the “intermediate audio signal 2” may, for example, comprise a larger number of channel signals, where the number of channel signals of the intermediate audio signal 2 may be equal to or different from the number of channels of the timescale-modified intermediate audio signal 1, which is inserted into the second processing 1140. However, in addition to the functionalities of the second processing 940, 1040, the second processing 1140 may receive a control signal 1144 from the timescale controller 1160.

[0206] Furthermore, the audio processor 1100 comprises one or more additional processing stages, for example, a processing stage n which is also designated with 1150. Thus, the processed audio signal 1012 can be provided as an output of processing stage n, or alternatively it can be provided as an output signal of processing stage 2 (1140) if there are no additional processing stages beyond the first processing stage. Petition 870250092380, dated 09 / 10 / 2025, page 80 / 153 73 / 121 1120 and the second processing stage 1140.

[0207] However, in general, any of the explanations applied above with respect to audio processor 900 and with respect to audio processor 1000 are applicable, except for the fact that time scale modification 1130 receives control signal 1134 from time scale controller 1160, that time scale modification 1130 provides time scale information 1136 to time scale controller 1160, and that one or more of the processing steps after time scale modification (for example, the second processing step 1140 and / or the nth processing step 1150) receive respective control signals 1144, 1154 from time scale controller 1160.

[0208] The timescale controller 1160 can, for example, control the timescale modification 1130 using the control signal 1134. Thus, the timescale controller 1160 can, for example, receive information about a desired quantity of a timescale (e.g., from a temporary storage management of variation) and can provide the control information 1134 to the timescale modification 1130 in order to adjust the timescale modification. Furthermore, the timescale controller 1160 can receive the timescale information 1136 from the timescale modification 1130. Regarding this matter, it should be noted that the timescale modification 1130 can determine an actual quantity of the timescale modification based on the control signal 1134, but also independently of the characteristics of the intermediate audio signal 1 (1122).In other words, the amount of timescale modification can be adapted according to the characteristics of the intermediate audio signal 1, in order to reduce artifacts. Consequently, the actual amount of timescale modification indicated by the timescale information 1136 may deviate from a desired timescale modification indicated by the control signal 1134.

[0209] However, the timescale controller 1160 can, for example, adapt the desired timescale modification depending on the timescale information 1136, for example, in an iterative manner (for example, for a subsequent frame). Furthermore, the timescale controller 1160 can provide control information 1144, 1154 for one or more steps of Petition 870250092380, dated 09 / 10 / 2025, page 81 / 153 74 / 121 processing 1140, 1150 after time scale modification. For example, the time scale controller 1160 can provide control information 1144, 1154 for the processing steps depending on the time scale information 1136 (and possibly depending on additional information received by the time scale controller 1160). Consequently, the time scale controller 1160 can provide the control signals 1144, 1154 for the processing steps 1140, 115 after time scale modification in order to adapt said processing steps to the time scale modification (e.g., to a desired time scale modification and / or to an actual time scale modification). For example, control signals 1144, 1154 can be used to adapt a frame grid or a subframe grid for a determination of processing blocks by processing steps 1140, 1150.However, control signals 1144, 1154 can also adjust any other processing characteristics of processing steps 1140, 1150.

[0210] In conclusion, the audio processor 1100 comprises a time-scaling modification between a plurality of processing steps, such that the time-scaling modification operates on an intermediate audio signal 1122. Furthermore, the time-scaling modification 1130 is controlled by the time-scaling controller 1160, and the processing steps 1140, 1150 after the time-scaling modification are also adapted to the actual time-scaling modification. Thus, the audio processor 1100 brings with it the advantages described above in relation to the audio processor 100 and, additionally, provides improved audio quality due to the adaptation of the processing steps 1140 and 1150 to the time-scaling modification.

[0211] In addition, it should be noted that the 1100 audio processor can optionally be supplemented by any of the features, functionalities and details described in this document. 8. Audio processor according to Figure 12

[0212] Figure 12 shows a schematic block diagram of a 1200 audio processor with time-scaling modification (TSM) on an intermediate audio signal, processing parameter adapters and a time-scaling controller, according to one embodiment of the invention. Petition 870250092380, dated 09 / 10 / 2025, page 82 / 153 75 / 121

[0213] It should be noted that the 1200 audio processor according to Figure 12 is very similar to the 1100 audio processor according to Figure 11. Consequently, those features, functionalities and details that have already been described with regard to the 1100 audio processor will not be explained in detail again. Instead, reference is made to the above discussion of the 1100 audio processor.

[0214] The audio processor 1200 is configured to receive an input audio signal representation 1210, which corresponds to the input audio signal representation 1110, and provides, based on this, a processed audio signal 1212 that corresponds to the processed audio signal 1112.

[0215] The audio processor 1200 comprises a first processing stage 1220, which corresponds to the first processing stage 1120, where it should be noted that the first processing stage 1220 may, for example, use a processing parameter 1, which is designated with 1221 (or, alternatively, a plurality of first processing parameters). The first processing stage 1220 provides an intermediate audio signal 1, which is designated with 1222, based on the representation of the input audio signal 1210, where the first processing stage 1220 is controlled by the processing parameter 1 (1221). The audio processor 1200 comprises a time-scaling modification 1230, which receives the intermediate audio signal 1 (1222) and provides, based on the same, a time-scaled modified intermediate audio signal 1, which is designated with 1232.Time scale modification 1230 receives a control signal (or control information) 1234, which corresponds to the control signal 1134, and provides a time scale information 1236, which corresponds to the time scale information 1136.

[0216] The audio processor 1200 comprises a second processing stage 1240, which receives the timescale-modified intermediate audio signal 1 (1332) and provides, based on the same, an intermediate audio signal 2, which is designated with 1242. The second processing stage 1240 can, for example, combine the functionalities of the second processing stage 1040 (for example, in the sense that the second processing stage 1240 uses a processing parameter) and of the second processing stage 1140 (for example, in the sense that the second processing stage 1240 receives a signal Petition 870250092380, dated 09 / 10 / 2025, page 83 / 153 76 / 121 of control). For example, the second processing stage 1240 receives a control signal 1244 (for example, from a time-scaling controller 1260) and also receives an adapted processing parameter 2, which is designated with 1245 from a first parameter adapter 1246. For example, the first parameter adapter 1246 receives a processing parameter 2, which is designated with 1241 and provides, based on the same, the adapted processing parameter 2 (designated with 1245). The adaptation of the processing parameter 2 (1241) by the parameter adapter 1246 is, for example, adjusted depending on a control signal 1247 that is provided by the time-scaling controller 1260.

[0217] Thus, the time-scaling controller 1260 can, for example, control the adaptation of one or more processing parameters 1241 used by the second processing 1240 (which follows the time-scaling modification 1230) and can also control a further adaptation of the processing 1240 using the control signal 1244. Consequently, the time-scaling controller 1260 can, for example, have a double impact on the processing performed by the second processing (or second processing stage) 1240.

[0218] Similarly, optional additional processing steps (e.g., an nth optional additional processing step 1250) may be similar to the second processing step 1240. For example, the nth processing step 1250 may receive a control signal 1251a from the timescale controller 1260, wherein an effect of the control signal 1251a may be similar to an effect of the control signal 1154. Furthermore, the nth processing step 1250 may receive one or more adapted processing parameters 1255 from a parameter adapter 1256. For example, the parameter adapter 1256 (e.g., an nth parameter adapter) may receive one or more nth processing parameters that are also designated with 1251. Furthermore, the parameter adapter 1256 receives a control signal 1257 from the timescale controller 1260.Consequently, the parameter adapter 1256 can adapt one or more processing parameters 1251 depending on the control signal 1257 provided by the time-scaling controller 1260, in order to obtain the one or more adapted processed parameters 1255 for the nth processing step 1250. Furthermore, it should be noted that the nth processing step 1250 can also receive a control signal 1254 from the controller. Petition 870250092380, dated 09 / 10 / 2025, page 84 / 153 77 / 121 of time scale 1260, where the control signal 1254 can, for example, adapt a processing performed by the nth processing step 1250 to the modification of the time scale.

[0219] The timescale controller 1260 can, for example, receive information indicating a desired timescale modification and provide the control signal 1234 to the timescale modification 1230. Thus, the timescale controller 1260 can signal to the timescale modification 1230 which timescale modification is desired. However, the timescale modification 1230 can provide the timescale information 1236 to the timescale controller 1260, wherein the timescale information 1236 can, for example, indicate a real timescale modification, which may be different from the desired timescale modification due to the signal characteristics of one or more intermediate audio signals 1222.The time-scaling controller can, for example, provide control signals 1244, 1254 for processing steps 1240, 1250 after time-scaling modification 1230 depending on time-scaling information 1236. Furthermore, the time-scaling controller 1260 can, for example, provide control signals 1247, 1257 controlling the adaptation of the respective processing parameters 1241, 1251 depending on time-scaling information 1236. Consequently, the time-scaling controller can control the adaptation of the respective processing parameters (or sets of processing parameters) to time-scaling modification and can also control an adaptation of a processing functionality of processing steps 1240, 1250 to time-scaling modification.

[0220] However, it is important to emphasize that it is not necessary to have all the functionalities presented in Figure 12. For example, it should be noted that any of the processing steps subsequent to the second processing step 1240 can be considered optional. In addition, both the parameter adapters 1246, 1256 and the use of the control signals 1244, 1254 can be considered optional. For example, it is not necessary for all these functionalities to be present in some modalities.

[0221] In conclusion, the audio processor 1200 comprises a timescale modification 1230 that is between a previous processing step 1220 Petition 870250092380, dated 09 / 10 / 2025, page 85 / 153 78 / 121 and a subsequent processing stage 1240. The audio processor 1200 also comprises a time scale controller 1260 which is configured to control an adaptation of one or more processing parameters for processing 1240 depending on the time scale modification and / or to adopt processing 1240.

[0222] However, it should be noted that the 1200 audio processor comprises the advantage described here in relation to the 1000 audio processor and in relation to the 1100 audio processor.

[0223] In addition, it should be noted that the 1200 audio processor may optionally be supplemented by any of the features, functionalities and details disclosed in this document, either individually or in combination. 9. Audio processor according to Figure 13

[0224] Figure 13 shows a schematic block diagram of an audio processor 1300 comprising a speech playback control, according to one embodiment of the invention. The audio processor 1300 is configured to receive an input audio representation 1310 and provide, based thereon, a plurality of reconstructed audio channels that are designated with 1312 and which can, for example, be considered as a processed audio signal representation. The audio processor 1300 (which can also be considered as an audio decoder) comprises a decoding (or decoder) 1320, which can, for example, perform an IVAS decoding of transport channels (and optionally, metadata). Thus, the decoding 1320 can provide one or more decoded transport channels 1322 and can also provide decoded metadata 1324.The audio processor 1300 also comprises a multi-channel timescale modification 1330 which can, for example, receive the decoded transport channels 1322 and provide one or more modified (decoded) transport channels on a timescale 1332. Furthermore, the audio processor 1300 comprises a reconstruction and rendering 1340 which receives the modified transport channels on a timescale 1332 and provides, based on them, the reconstructed audio channels 1312. Additionally, the audio decoder comprises a metadata adaptation 1370 which receives the decoded metadata 1324 and provides, based on them, adapted metadata 1372 for reconstruction and... Petition 870250092380, dated 09 / 10 / 2025, page 86 / 153 79 / 121 rendering 1340. In addition, the audio processor 1300 comprises a playback speed control 1380, which receives playback speed information 1382 (e.g., a playback speed factor) and provides, based on the same, a plurality of control signals 1334, 1344, 1347. For example, the playback speed control may also receive time scale information 1336, which may, for example, describe a real time scale achieved by modifying the time scale of multiple channels 1330. The playback speed control 1380 may comprise adaptation control logic 1381, which may process the playback speed information 1382 and the time scale information 1336 and provide, based on the same, the control signals 1334, 1344, 1347.Thus, playback speed control 1380 can control time scaling, which is performed by modifying the time scale of multiple channels 1330, depending on playback speed information 1382, for example, to modify a playback speed (which may, for example, be useful for facilitating the understanding of audio content or which may, for example, be useful for achieving synchronization between an audio encoder and an audio decoder running at different rates or on different clocks).

[0225] It should be noted that the multi-channel timescale modification 1330 can be configured to apply the timescale modification to a plurality of decoded transport channels (e.g., in parallel), for example, in a synchronized manner. Furthermore, it should be noted that rendering and reconstruction 1340 can, for example, be controlled by the control signal 1344 which is provided by the playback speed control 1380. However, reconstruction and rendering 1340 can, for example, also receive rendering parameters 1349 which can, for example, describe a (desired) rendering of audio content represented by the timescale-modified transport channels 1332, to reconstructed output channels 1312.It should be noted that reconstruction rendering 1340 may, for example, include a mix, so that, for example, the number of reconstructed output channels 1312 is greater than the number of modified transport channels in the timescale 1332. A rendering parameter 1349 may, for example, be provided by a user interaction (e.g., using an interface). Petition 870250092380, dated 09 / 10 / 2025, page 87 / 153 80 / 121 user) or it can, for example, be derived from an encoded audio representation.

[0226] Furthermore, metadata adaptation 1370 can adapt metadata 1324 to the timescale modification, thus obtaining adapted metadata 1372. Adapted metadata 1372 can, for example, be used in reconstruction and rendering 1340. Just as an example, metadata 1324 or adapted metadata 1372 can, for example, describe a blend that must be performed by reconstruction and / or rendering 1340. For example, adapted metadata 1372 can be inputs to a blend matrix, or it can be parameters that describe inputs to a blend matrix that is used in reconstruction and rendering 1340 to derive the reconstructed output channels 1312 based on the timescale-modified transport channels 1332. However, different types of metadata can also be used.

[0227] In conclusion, in the audio processor 1300, multi-channel timescale modification 1330 can be performed between a decoding 1320 and a reconstruction and rendering 1340, wherein the timescale modification can be a multi-channel timescale modification that operates (e.g., in a synchronized manner) on a plurality of decoded transport channels. The playback speed control 1380 controls the multi-channel timescale modification 1330 using a control signal 1334 and also controls the reconstruction rendering 1340 using a control signal 1344 and also controls the metadata adaptation 1370 using a control signal 1347. Thus, the adapted metadata 1372 can be provided in such a way that it is adapted to the timescale modification, which brings a good result from the reconstruction and rendering 1340.

[0228] However, it should be noted that the 1300 audio processor may include any of the advantages of the other audio processors disclosed in this document.

[0229] In addition, it should be noted that the 1300 audio processor may optionally be supplemented by any of the features, functionalities and details disclosed in this document, either individually or in combination. 10. Audio processor according to Figure 14

[0230] Figure 14 shows a schematic block diagram of a Petition 870250092380, dated 09 / 10 / 2025, page 88 / 153 81 / 121 audio processor 1400, according to one embodiment of the invention. In particular, the audio processor 1400 can be combined with an IVAS Variable Temporary Storage Management (IVAS JBM). However, it should be noted that the main functionality of the audio processor 1400 can be similar to the main functionality of the audio processor 1300. In particular, the audio signal processor 1400 comprises a decoding 1420, which can be configured to perform an IVAS decoding of transport channels (and optionally also of metadata). The decoding 1420 can correspond to the decoding 1320. Thus, the decoding 1420 provides one or more decoded transport channels 1422 (preferably a plurality of decoded transport channels 1422).Furthermore, the audio processor comprises a multi-channel timescale modification 1430 that may substantially correspond to the multi-channel timescale modification 1330. The multi-channel timescale modification 1430 receives one or more decoded transport channels 1422 and also receives a control signal 1434. The multi-channel timescale modification 1430 provides one or more timescale-modified transport channels 1432, which may substantially correspond to the timescale-modified transport channels 1332. Additionally, the multi-channel timescale modification 1430 provides timescale information 1436 that may substantially correspond to timescale information 1336. Furthermore, the audio processor 1400 also comprises a reconstruction and rendering 1440 that may substantially correspond to reconstruction and rendering 1340.Reconstruction and rendering 1440 receives the modified transport channels at timescale 1432 and also receives a control signal 1444. Reconstruction and rendering 1440 also receives one or more rendering parameters 1449 and also receives adapted metadata 1472. For example, the control signal 1444 may correspond to the control signal 1341a, and one or more rendering parameters 1449 may correspond to one or more rendering parameters 1349. Adapted metadata 1472 may correspond to adapted metadata 1372. Reconstruction and rendering 1340 provides, based on the modified transport channels at timescale 1432, the reconstructed output channels 1412. Reconstruction and rendering 1440 may use the control signal 1444, one or more. Petition 870250092380, dated 09 / 10 / 2025, page 89 / 153 82 / 121 rendering parameters 1449 and adapted metadata 1472 to provide the reconstructed output channels 1412 based on the timescale-modified transport channels 1432. For example, reconstruction and rendering 1440 may perform an upmix and / or a rendering, where inputs of an upmix matrix or a rendering matrix may, for example, be determined by the adapted metadata 1472. Furthermore, the control signal 1444 and / or the rendering parameters 1449 may have an additional impact on reconstruction and rendering 1440 and may, for example, vary matrix inputs used by reconstruction and rendering.

[0231] The audio processor 1400 also comprises a metadata adaptation 1470, which may, for example, correspond to the metadata adaptation 1370. Thus, the metadata adaptation 1470 may, for example, adapt the metadata 1424 provided by the decoding 1420 depending on the timescale modification (e.g., depending on a control signal 1447), in order to obtain the adapted metadata 1472. Consequently, the metadata may adjust to the timescale-modified transport channels 1432, so that the reconstruction and rendering 1440 provide reconstructed (and / or rendered) output channels 1412 with good audio quality.

[0232] However, unlike the audio processor 1300, the audio processor 1300 does not include a playback speed control, but rather a temporary variation storage control 1480. In addition, the audio processor 1400 may also include packet handling 1490. The functionality of packet handling 1490 and temporary variation storage control 1480 will be described below. It should be noted that the audio processor 1400 can receive real-time transport protocol packets (RTP packets) 1492. The incoming real-time transport protocol packets 1492 can, for example, be obtained from a storage medium or can, for example, be supplied to the audio processor 1400 via a wired or wireless data connection, such as a TCP / IP network (or any other data communication network).For example, the received RTP packets may be subject to a timing variation, which may be caused, for example, by variations in the delay of a communication network through which the RTP packets are transported to the audio processor 1400. The audio processor. Petition 870250092380, dated 09 / 10 / 2025, page 90 / 153 83 / 121 1400 comprises a real-time transport protocol unpacker 1493, which receives real-time transport protocol packets and extracts a packet payload from the real-time payload packets. The real-time transport unpacker 1493 therefore receives the incoming RTP packets 1492 and provides, based on them, a packet payload 1494. The packet payload 1494 may, for example, comprise IVAS frames, for example, an encoded representation of frames according to the IVAS standard. The audio processor 1400 also comprises a variance-compensation temporary storage 1495 which receives the payload information and has the capacity to store the (unpacked) payload information (e.g., IVAS frames) in temporary storage. Thus, the variance-compensation temporary storage 1495 provides payload information in temporary storage 1496 for decoding 1420.For example, the 1495 variation compensation temporary storage can be configured to receive 1494 payload information at a rate at which the payload is unpacked and can provide payload information in temporary storage 1496 for decoding 1420 at a rate that is required (or can be processed) by decoding 1420. Thus, the 1495 variation compensation temporary storage can receive an IVAS frame when the IVAS frame is unpacked from the RTP packet and can provide that IVAS frame for decoding 1420 when said IVAS frame can be processed (or should be processed) by decoding 1420.

[0233] Variance temporary storage control 1480 can control variance compensation temporary storage 1495, multi-channel time scale modification 1430, and can also control reconstruction and rendering 1440, as well as parameter adaptation 1470. For example, variation temporary storage control 1480 can comprise a network variation analysis and a destination delay estimate 1481, which can, for example, receive timing information 1493a from the RTP unpacker. For example, timing information 1493a can describe when RTP packets arrive at the unpacker 1493 and / or when different payload information (e.g., different IVAS frames) arrive at the RTP unpacker (or when said payload information can be sent to the Petition 870250092380, dated 09 / 10 / 2025, page 91 / 153 84 / 121 temporary variation compensation storage 1495). In other words, information 1493a can be any information that is usable for obtaining information about a network variation and that may be useful for estimating a target delay. Thus, analysis of temporary variation storage 1481 can, for example, determine a network variation and can, for example, obtain information about a (desired) average fill of the temporary variation compensation storage (i.e., information indicating how many payload information units should be stored in the temporary variation compensation storage 1495 on average).

[0234] For example, the variation temporary storage control 1480 may comprise an adaptation control logic 1483, which receives a network variation analysis result 1482 from the network variation analysis 1481. Furthermore, the variation temporary storage control 1480 may comprise a playback delay estimator 1484 that may receive temporary storage filling information 1485 from the variation compensation temporary storage 1495 and that may also receive time scale information 1436 from the multi-channel time scale modification 1430.

[0235] Consequently, the playback delay estimator 1484 can, for example, determine or estimate a playback delay and provide playback delay information 1486 to the adaptation control logic 1483. The adaptation control logic 1483 can, for example, receive the network variation analysis result information 1482 from the network variation analysis 1481 and the playback delay information 1486 from the playback delay estimator 1484 and provide the control signals 1434, 1444 and 1447 for multi-channel timescale modification 1430, for reconstruction and rendering 1440 and for metadata adaptation 1470. Furthermore, the adaptation control logic 1483 can provide a control signal 1487 for the temporary storage of variation compensation 1495.

[0236] Consequently, the control of the temporary variation storage 1480 can receive information about network variation, information about the filling of the temporary variation reduction storage, information about an actual time scale modification and possibly information Petition 870250092380, dated 09 / 10 / 2025, page 92 / 153 85 / 121 additional. Based on this information, the variation temporary storage control 1480 may have the capability to determine a playback delay and may also have the capability to determine how many payload units (e.g., IVAS frames) should preferentially be stored in the variation compensation temporary storage 1495 to allow an uninterrupted supply of the reconstructed output channels 1412, even in view of an expected network variation. For example, the variation temporary storage control 1480 may be able to effect short-duration variation compensation by appropriately controlling the variation compensation temporary storage 1495. However, the variation temporary storage control 1480 may also be configured to adapt to timescale changes, for example, to handle variation on a longer timescale.Furthermore, controlling the temporary storage of 1480 variation can manage metadata adaptation, reconstruction, and rendering to adapt well to potential timescale changes.

[0237] However, it should be noted that the details regarding the functionality of the temporary variation storage control may differ from the functionalities described above.

[0238] In conclusion, the audio processor 1400 allows for network variation compensation using a variation compensation temporary storage 1495 and also using a multi-channel time scale modification 1430. The variation temporary storage control 1480 can be configured to control both the variation compensation temporary storage 1495 and the multi-channel time scale modification 1430 to adapt a speed at which the reconstructed output channels 1412 are provided for network variation.Temporary variation compensation storage 1495 can be used to compensate for short-term network variation, and multi-channel time scaling modification 1430 can be used to compensate for network delay variations on a larger time scale and may possibly also be used in the case of an encoder providing incoming RTP packets 1492 at a rate different from that required by the audio processor 1400. In particular, the fact that the time scaling modification of... Petition 870250092380, dated 09 / 10 / 2025, page 93 / 153 86 / 121 multiple channels 1430, positioned between 1420 decoding and 1440 reconstruction rendering, allows for efficient timescale modification without introducing significant audible artifacts.

[0239] Furthermore, it should be noted that any of the advantages discussed in this document with respect to other audio processors also apply to the 1400 audio processor.

[0240] It should also be noted that the 1400 audio processor can optionally be supplemented by any of the features, functionalities and details disclosed in this document, either individually or technically. 11. TEMPORARY STORAGE OF RECEIVER ACCORDING TO FIGURES 15A AND 15B

[0241] Figure 15a shows a schematic representation of an operation of a temporary storage receiver for the IVAS JBM (IVAS Variation Temporary Storage Management). It should be noted that the variation temporary storage management according to Figure 15a can optionally be used in any of the embodiments according to the invention disclosed in this document.

[0242] It should be noted that the management of the temporary variation storage can be configured to control a temporary output storage of the receiver 1510, as well as decoding, time scale modification, and data exchange between the temporary output storage of the receiver 1510 and the acoustic front-end 1520. For example, the control may comprise checking 1530 if there are sufficient samples available (e.g., if there are sufficient samples included in the receiver's temporary output storage to transfer a block of samples (e.g., sample values ​​associated with an audio frame) to the acoustic front-end 1520). If it is found that there are sufficient samples available, the control comprises retrieving the samples from the queue to a PCM / FD temporary data storage and returning them (e.g., the samples) to the acoustic front-end.For example, if it is found that there are sufficient samples available in the output temporary storage of receiver 1510, these samples can be transferred to a transmission data temporary storage (e.g., a PCM / FD temporary data storage) for transmission to the acoustic front-end 1520. For example, the acoustic front-end 1520 can then... Petition 870250092380, dated 09 / 10 / 2025, page 94 / 153 87 / 121 extract PCM / FD data from temporary PCM / FD data storage. For example, the 1520 acoustic front-end can extract time-domain or frequency-domain values ​​from a (processed) frame or subframe from temporary PCM / FD data storage for further processing (and preferably for outputting audio content to a user).

[0243] However, if it is found, in check 1530, that there are not enough samples available (e.g., in the receiver's output temporary storage), the control initiates a decoding of a next frame (e.g., a next frame from the variation temporary storage). For example, the decoding may comprise a decoding of audio samples (with time scale modification and adapted metadata and reconstruction / rendering). Consequently, variable-size PCM / FD data (or, alternatively, fixed-size PCM / FD data) can be queued in a data temporary storage (e.g., in the receiver's output temporary storage). Thus, for example, if the queued PCM / FD data (e.g., variable-size) includes enough samples, the samples can be transferred to the acoustic front-end 1520 (e.g., via the PCM / FD data temporary storage).

[0244] In conclusion, the provision of samples in the receiver's output temporary storage and / or in the PCM / FD temporary data storage may involve decoding and may also involve timescale modification. Optionally, processing may also involve metadata adaptation and reconstruction / rendering.

[0245] Furthermore, it should be noted that the mechanism according to Figure 15a may optionally be introduced into any of the embodiments disclosed in this document.

[0246] Figure 15b shows a schematic representation of an operation of a temporary variation storage management or control mechanism, according to another embodiment of the present invention. The temporary variation storage management (or control mechanism) described in this document can, for example, be used in combination with temporary storage for the IVAS JBM (IVAS temporary variation storage management). Petition 870250092380, dated 09 / 10 / 2025, page 95 / 153 88 / 121

[0247] It should be noted that the operation of the temporary variation storage management (or control mechanism) is based on the assumption that there is a decoding of the transport channels and a timescale modification of the decoded transport channels. It is further assumed that a 1560 reconstructor / renderer follows the timescale modification and that the reconstructor / renderer comprises an input temporary storage that stores in temporary storage, for example, timescale-modified transport channels, or that there is a temporary storage between the timescale modification and the reconstructor / renderer to store in temporary storage timescale-modified intermediate audio signals.

[0248] For example, the management of temporary variation storage (or the control mechanism) comprises checking 1570 whether there are sufficient samples available to perform a reconstruction / rendering (where the reconstruction / rendering may, for example, operate in a block-by-block manner and therefore may require blocks (e.g., frames or subframes) of transport channels modified in timescale). Checking 1570 may, for example, be performed at a time when it is desirable to provide new reconstructed / rendered values, for example, for an output (or transmission) to an acoustic front-end and / or an edge device.If it is found that sufficient samples are available (for example, in a temporary input storage of the reconstructor / renderer, or in a temporary storage between the time scale modification and the reconstructor / renderer), a reconstruction / rendering is performed (for example, triggered by the control mechanism), which provides samples that can be inserted into a temporary output storage of the optional receiver 1590, or that can be provided directly (for example, without temporary storage) to an acoustic front-end or to an edge device (for example, via an interface) 1595.

[0249] If, however, it is found in check 1570 that there are not enough samples available to perform a reconstruction and / or rendering (e.g., in blocks) (e.g., by the reconstructor / renderer 1560), the temporary variation storage management (or the mechanism of Petition 870250092380, dated 09 / 10 / 2025, p. 96 / 153 (89 / 121 control) can, for example, trigger a decoding of a subsequent frame, for example, a 1580 decoding of audio samples from the transport channel (for example, along with or followed by a timescale modification, for example, of the decoded transport channels, and a determination of adapted metadata). As a result of the 1580 decoding and the subsequent timescale modification, samples from one or more timescale-modified transport channels become available and are inserted into a temporary input storage of the reconstructor / renderer or into a temporary storage that is organized between the timescale modification and the reconstructor / renderer. Furthermore, adapted metadata can also be provided if such adapted metadata is required.Thus, the 1560 reconstructor / renderer can perform a reconstruction and / or rendering based on samples stored in temporary storage and taking into account adapted metadata (if necessary), in order to provide one or more reconstructed and / or rendered audio signals that can be inserted into the receiver output temporary storage (optional) 1590, or that can be provided (directly, without temporary storage) to the acoustic front-end or edge device 1595. For example, in the presence of the receiver output temporary storage 1590, the acoustic front-end 1595 can then extract PCM / FD data from said receiver output temporary storage.For example, the 1595 acoustic front-end can extract time-domain or frequency-domain values ​​from a (processed) frame or subframe from receiver output temporary storage for further processing (and preferably for outputting audio content to a user). However, in the absence of receiver output temporary storage, the reconstructor / renderer can provide (e.g., stream or send) the processed frame or subframe to the 1595 acoustic edge / front-end device.

[0250] In conclusion, the reconstructor / renderer can provide the reconstructed and / or rendered audio signals when needed for the acoustic front-end or edge device. Because the timescale modification is performed before reconstruction / rendering, the timescale modification does not cause a variable delay between reconstruction / rendering and playback of the reconstructed and / or rendered audio signals. Thus, in the case of motion parameters Petition 870250092380, dated 09 / 10 / 2025, page 97 / 153 Since 90 / 121 are considered by reconstruction / rendering, a variable motion delay for audio can be avoided. Furthermore, temporary storage of the receiver output can be optionally omitted, since (in the presence of sufficient temporary frame storage, for example, before decoding), it is possible to request the provision of decoded transport channels (with modified timescale) whenever necessary.

[0251] In conclusion, the control mechanism in Figure 15b is an advantageous solution.

[0252] Furthermore, it should be noted that the mechanism according to Figure 15b may optionally be introduced into any of the embodiments disclosed in this document. 12. TEMPORARY STORAGE OF TRANSPORT CHANNEL FRAME FOR FD RENDERING ACCORDING TO FIGURE 16

[0253] Figure 16 shows a schematic representation of a temporary storage of a transport channel frame for FD rendering, according to an embodiment of the invention.

[0254] An original transport channel frame, which can be provided by a decoding (e.g., by decoding 1320 or decoding 1420) is shown in reference numeral 1610. A timescale modification is applied to the original transport channel frame (which can be considered as an intermediate audio signal). For example, timescale modification 1330 or timescale modification 1430 can be applied. Consequently, a shortened transport channel frame 1620 is obtained. For example, the shortened transport channel frame 1620 can be considered as a timescale-modified intermediate audio signal.

[0255] Assuming that one or more time-domain samples from the previous frame (associated with the previous frame) were not processed beforehand (for example, because they did not fit into the frame grid or sub-frame grid of reconstruction and rendering 1340, 1440), these one or more samples from the previous frame, which are designated with 1640, are appended at the beginning of the shortened transport channel frame 1620, thus resulting in a sample portion 1632 (which may be time-domain samples or frequency-domain samples). In other words, portion 1632 comprises one or more samples Petition 870250092380, dated 09 / 10 / 2025, page 98 / 153 91 / 121 stored in temporary storage 1630 from the previous frame and the samples from the current frame shortened (timescale modified) from the transport channel 1620. Now, a sample frame 1640 is selected from the sample portion 1632. For example, the length of frame 1640 is determined by the size of a processing frame from the second processing after the timescale modification or as an integer multiple of the size of a subframe from the second processing. For example, the length of frame 1640 (also designated as “frame to reconstruct / render”) can be determined by the number of samples needed (or processed) in a processing step of the reconstruction and rendering 1340, 1440, and can, for example, be an integer multiple of that number of samples.

[0256] However, if sample portion 1632 comprises more samples than frame 1640, which is inserted into reconstruction and rendering 1340, 1440, one or more samples 1642 will be stored in temporary storage for later processing along with samples from the subsequent frame (and, insofar as they correspond to the samples stored in temporary storage 1630). Thus, by using one or more samples stored in temporary storage 1630 and defining the size of frame 1640 according to the requirements of reconstruction and rendering 1340, 1440, a reconstruction and rendering can be performed efficiently even in the presence of a change in the timescale.For example, a time-frequency transformation (or time-domain-to-frequency-domain transformation) (shown in reference numeral 1650) can be applied to frame 1640 and can provide a frequency-domain representation of frame 1640, for example, for a number of time intervals that are shown in reference numeral 1660. For example, the frequency-domain representation shown in reference numeral 1660 can be fed into a renderer 1670, which can also receive one or more rendering parameters or other parameters 1620. The renderer can, for example, provide a frequency-domain representation of reconstructed output channels. An inverse time-frequency transformation (e.g., frequency-domain-to-time-domain transformation) 1680 can be applied to the frequency-domain representation (e.g., of reconstructed output channels or rendered output channels). Petition 870250092380, dated 09 / 10 / 2025, page 99 / 153 92 / 121 provided by renderer 1670 to obtain a time-domain representation of a "shortened" output frame, which is shown in reference number 1690.

[0257] In conclusion, in the processing shown in Figure 16, a 1640 frame (e.g., of time-domain samples) can be obtained, which can be fed into a 1650 time-frequency transformation. The length of the 1640 frame can, for example, fit a processing grid of the 1650 time-frequency transformation, so that the 1650 time-frequency transformation can provide an appropriate 1660 frequency-domain representation of the 1640 frame. This 1660 frequency-domain representation can be fed into a renderer, which performs a rendering in the frequency domain. A result of the 1670 rendering, which is performed in the frequency domain, will then be transformed back to the time domain, to obtain a time-domain representation of the rendered (and / or reconstructed) output channels.

[0258] However, it should be noted that temporary storage and time-domain sample selection for the 1650 time-frequency transformation can, for example, be performed by a stage that can be arranged between the time-scaling modification and the actual reconstruction / rendering. In other words, the concept as explained in relation to Figure 16 can optionally be performed by any of the audio processors disclosed in this document.

[0259] It should be noted that, in the processing of Figure 16, a frame length of 1640 is preferably chosen so that the frame length fits an original processing grid (e.g., subframe grid) of the subsequent processing (e.g., time-frequency transformation 1650 and / or renderer 1670). Thus, the frame length of 1640 can be chosen so that the number of time intervals that are provided to the time-frequency transformation 1650 is an integer multiple of the size of a subframe in terms of time intervals. Consequently, the time-frequency transformation 1650 can, for example, be performed with a predetermined (e.g., fixed) transformation length. However, by adapting the frame length of 1640 so that the time-frequency transformation 1650 results in a number of time intervals that are a multiple Petition 870250092380, dated 09 / 10 / 2025, pp. 100 / 153 93 / 121 integer of a predetermined subframe length (time interval), it may happen that a relatively large number of samples 1642 must be stored in temporary storage for later processing. However, the possibility of performing time-frequency transformation and subsequent rendering with predetermined (e.g., fixed) framing (e.g., in terms of subframe size) can compensate for this problem.

[0260] Furthermore, the processing shown in Figure 16 can optionally be supplemented by any of the features, functionalities, and details disclosed in this document, either individually or technically. 13. FD RENDERING WITH SUBFRAME SIZE Adapted according to Figure 17

[0261] Figure 17 shows a schematic representation of a frequency domain rendering with adapted subframe size, according to an embodiment of the invention. It should be noted that the processing shown in Figure 17 can optionally be used in any of the audio processors disclosed in this document.

[0262] The processing receives the transport channel frame 1710, which may, for example, be provided by a decoding. A timescale modification (e.g., as disclosed in this document) provides a shortened transport channel frame 1720 based on the (original) transport channel frame 1710, as described in this document. The sample values ​​portion, which is designated with 1732, may comprise one or more samples stored in temporary storage 1730 of a previous frame (e.g., associated with a previous frame) and the samples of the shortened transport channel frame 1720. A frame 1740 may be chosen from the portion 1732, wherein a frame length 1740 may, for example, be chosen such that a time-frequency transformation 1750 can provide an integer number of time intervals based on the selected frame 1740.However, it should be noted that, in 1700 processing, the selection of the 1740 frame may not be linked to a predetermined frame or subframe grid for subsequent processing (e.g., time-domain to frequency-domain transformation). Thus, the number of samples in the 1740 frame may differ from the number of samples required to operate using a frame grid or grid. Petition 870250092380, dated 09 / 10 / 2025, pp. 101 / 153 94 / 121 predetermined subframe “fixed”. Consequently, the number of time intervals for which the 1750 time frequency transformation provides samples (or sets of samples) based on the 1740 frame may be different from an integer multiple of a preferred subframe length (in terms of time intervals). However, processing can adjust or subdivide the time portions for which the 1750 time frequency transformation provides transformed samples (or transformed sets of samples) to / in subframes of different lengths. For example, a first subframe SF1 may comprise four time intervals, a second subframe SF2 may also comprise four time intervals, a third subframe SF3 may comprise three time intervals, and a fourth subframe SF4 may comprise two time intervals.Thus, the 1750 time-frequency transformation and / or the subsequent allocation of time intervals to subframes can operate in such a way that subframes of varying lengths are provided for further processing. Subframes of varying lengths (e.g., SF1, SF2, SF3, SF4) can, for example, be fed into a 1770 renderer, which can, for example, receive one or more 1772 parameters and provide, based on the subframes and one or more 1772 parameters, a frequency domain representation of rendered and / or reconstructed output channels.An inverse time-frequency transformation (e.g., a frequency-domain to time-domain transformation) 1780 can be applied to the frequency-domain representation of the rendered and / or reconstructed output channels provided by the renderer 1770, in order to obtain time-domain representations of the rendered or reconstructed output channels, which are shown in the reference numeral 1790. For example, the time-domain representation of the rendered and / or reconstructed output channels can be provided in the form of a shortened output frame.

[0263] In conclusion, modifying the timescale can result in a shortened transport channel frame that, for example, can be combined with one or more samples stored in temporary storage from a previous frame. This portion of samples comprising the one or more samples stored in temporary storage associated with a previous frame and the samples from the shortened (current) transport channel frame can then be used Petition 870250092380, dated 09 / 10 / 2025, pp. 102 / 153 95 / 121 to provide subframes of varying lengths, where the use of varying subframe lengths can allow the processing of a large number of samples from the 1740 portion (while only a small number of 1742 samples need to be stored in temporary storage for later processing). The subframes of varying lengths (e.g., SF1, SF2, SF3, and SF4 subframes obtained by time and frequency transformation 1750) can then be processed by the renderer 1770, and the rendering result, which is performed in the frequency domain, can then be transformed back to the time domain to produce rendered or reconstructed output channels (e.g., in the form of a shortened output frame 1790). Thus, the use of SF1, SF2, SF3, and SF4 subframes of varying lengths allows for particularly good processing efficiency and also helps to keep latencies low.However, the processing is naturally a bit more complicated when compared to using fixed subframe lengths.

[0264] It should be noted that the processing as described in relation to Figure 17 can optionally be used in any of the audio processors disclosed in this document. It can also be noted that the processing of Figure 17 can optionally be supplemented by any of the features, functionalities and details disclosed in this document. 14. RENDERING WITH CONSTANT SUBFRAME SIZE AND METADATA ADAPTATION, ACCORDING TO FIGURE 18

[0265] Figure 18 shows a schematic representation of a rendering with constant subframe size and metadata adaptation. The processing is based on a transport channel frame 1810 and applies a time-scaling modification to the transport channel frame 1810 to obtain a shortened transport channel frame 1820. A sample portion is obtained by combining one or more samples stored in temporary storage 1830 associated with a previous frame with the samples from the shortened transport channel frame 1820. The portion obtained by this combination is designated 1832. A frame 1840 is selected based on this portion 1832, where the size of the frame 1840 can be chosen so that a time-frequency transformation 1850 provides an integer number of time intervals based on the frame 1840. Samples from the portion 1830, which are not included in the frame 1840 Petition 870250092380, dated 09 / 10 / 2025, pp. 103 / 153 Samples 96 / 121 and designated 1842 may be stored in temporary storage for subsequent processing, for example, in combination with samples associated with the subsequent framework.

[0266] The 1850 time-frequency transformation provides a set of time intervals based on the selected frame 1840, for example, time intervals TS1 to TSm. However, if the number of time intervals is not equal to an integer multiple of the size of a subframe (in terms of time intervals), one or more time intervals from a previous frame (e.g., TS-1,m) and / or one or more time intervals from a subsequent frame (e.g., time intervals TS+11 and TS+12) are used to obtain subframes of equal length. For example, in the modality of Figure 18, all subframes SF1, SF2, SF3 and SF4 can each comprise four time intervals (where, for example, the first subframe comprises time intervals TS-1,m, TS1, TS2 and TS3, and where, for example, the last subframe can comprise samples TSm-1, TSm, TS+11 and TS+i2).Thus, a corresponding (equal) subframe size can be obtained by using one or more time intervals from a previous frame and / or one or more time intervals from a subsequent frame to obtain the subframes.

[0267] The subframes can then be processed by an 1870 renderer, wherein the 1870 renderer can, for example, use one or more 1872 parameters. Furthermore, an inverse time-frequency transformation (e.g., a frequency domain to time domain transformation) 1880 can be applied to the frequency domain representation of reconstructed and / or rendered audio channels provided by the 1870 renderer. Consequently, a time domain representation of the reconstructed and / or rendered output channels can be obtained, which is designated with 1890.

[0268] Thus, a shortened output frame of reconstructed or rendered output channels can be obtained.

[0269] However, it should be noted that the metadata, which is, for example, used by the 1870 renderer, can be adapted to the timescale modification, as disclosed in this document. Consequently, a good auditory impression can be achieved, since the adapted metadata can fit well to the frequency domain representation that is inserted into the 1870 renderer (when, for example, the metadata can fit to Petition 870250092380, dated 09 / 10 / 2025, pp. 104 / 153 97 / 121 subframes that are processed by the 1870 renderer).

[0270] In conclusion, the processing as shown in Figure 18 can provide efficiently reconstructed and / or rendered output channels, where rendering is performed with a constant subframe size due to the aforementioned subframe composition. A good auditory impression can be obtained by adapting the metadata to the subframes.

[0271] However, it should be noted that the processing disclosed in relation to Figure 18 may optionally be used in any of the audio processors or audio processing systems disclosed in this document.

[0272] It should also be noted that the processing of Figure 18 may optionally be supplemented by any of the resources, functionalities and details disclosed in this document. 15. ADAPTATION OF INTERPOLATION FUNCTIONS

[0273] An adaptation of the interpolation functions is illustrated in Figure 19. In relation to this issue, it should be noted that the adaptation of the interpolation functions, shown in Figure 19, and which can be used in any of the modalities disclosed in this document, must be understood in combination with the concept of shortening a frame shown in Figure 5c and also in combination with the concept of lengthening an input frame shown in Figure 5d (where these concepts for shortening an input frame and for lengthening an input frame can optionally be used in modifying the time scale described in this document).

[0274] Taking Figure 5c as a reference, it can be observed that an input frame can have a length of 2L sec. The input frame (which can, for example, be a frame of the intermediate (original) audio signal) can be subdivided (at least logically) into a first portion (SEG A) and a second portion (SEG B). However, starting from a positional pmin within the input frame, a search can be performed to identify a portion of the input frame that can be best superimposed and added to the SEG A portion. In this search, a plurality of candidate portions can be considered, and a cross-correlation between a respective candidate portability and the SEG A portion can be used as a criterion to identify a portion that can be best superimposed and added. A portion that has been identified as having a Petition 870250092380, dated 09 / 10 / 2025, pages 105 / 153 98 / 121 overlap and sufficiently good (or better) addition with the SEG A portion, which is designated 521c, and the remainder of the input frame 521a, which is designated “REM”, are used to obtain the output frame 521d. For example, the identified portion 521c (which is also designated “best match”) will be overlapped and added to the “SEG A” portion of the input frame 521a. Thus, an initial portion 521e of the shortened output frame 521d will be formed by an overlap and addition of the “SEGA” portion and the “best match” portion identified 521c. A final portion 521f of the shortened output frame 521d can be formed by the remaining “REM” portion that follows the identified portion 521c. Thus, the shortened output frame 521d is a concatenation of an initial portion 521e, which is obtained by an overlapping and addition operation (of the SEGA and 521c portions), and an end portion 521f (REM), which is an unmodified portion (end) of the input frame 521a.For example, it can be assumed that the portions “SEG A” and “SEG B” can both have the same length of Lseg. It is further assumed that the remaining portion “REM” has a length of Lrem. Therefore, it can be assumed that the shortened output frame 521d has a length of Lseg Lrem, where the initial portion of length Lseg is formed using an overlay and addition operation that effectively causes a shortening, while the final portion of length Lrem remains unchanged.

[0275] Taking Figure 5d as a reference, an extension of an input frame will be described. It should be noted that an input frame 531a is considered to have a length of 2 Lseg. The input frame 531a is, for example, (at least logically) divided into a first segment “SEG A” and a second segment “SEG B”. Now, a plurality of candidate portions are selected, which comprise one or more samples from a previous input frame that precedes the current input frame 531a and which are consequently located partially or entirely within the previous input frame. Consequently, a “best match” portion 531c is selected, which is judged to provide the best possible overlap and addition (or at least sufficiently good) with the first portion “SEG A” of the input frame 531a.Consequently, a remaining portion “REM” extends from one end of the identified portion 531c to the end of the current input frame 531a. Thus, the remaining portion may, in some cases, be even larger than the frame. Petition 870250092380, dated 09 / 10 / 2025, pp. 106 / 153 99 / 121 of input 531a, but it can also be smaller than the input frame 531a. However, the remaining portion will normally be larger than Lsec.

[0276] Consequently, an initial portion of the elongated output frame 531d is obtained by using an overlap and addition of the “SEG A” portion of the input frame 531a and the identified “best match” portion 531c. Furthermore, the remaining portion “REM” is appended after the overlap and addition region, thus obtaining the elongated output frame 531d. Consequently, the elongated output frame 531d is a concatenation of an overlap region, in which the “SEG A” portion and the identified portion 531c are overlapped and added, and the remaining portion “REM”. Thus, the length of the shortened output frame 531d is equal to the sum of the length Lseg (which is the length of the overlap and addition portion) and a length Lrem, which is the length of the remaining portion. Since the length of the remaining portion is typically greater than Lsec in the case of elongating an input frame, the total length is typically greater than 2 Lsec.Consequently, the elongated output frame 531d is longer than the input frame 531a, where a good quality of the elongated output frame 531d is achieved by selecting the identified portion 531c or, for example, depending on a cross-correlation between a plurality of candidate portions and the segment “SEG A”.

[0277] In conclusion, an initial portion 531e of length Lseg is obtained using an overlay and addition operation, and a final portion 531f of the elongated output frame 531d remains unchanged.

[0278] With regard to the concept of Figures 5c and 5d, it should be noted that the length of the remaining portion, Lrem, is determined by the characteristics of the audio signal and, therefore, is not a quality that can be exactly predetermined by a control that provides information, but a desired time scale modification.

[0279] Taking Figure 19 as a reference, the interpolation function that can be used to determine the adapted parameters will be described. In the case of an unchanged frame size, the frame can have a length of 2 Lsec, where the temporal extent of the frame is shown on an abscissa 1910. An ordinate 1912 describes a value of the interpolation function. It can be observed that, in the case of an unchanged frame size, shown in the reference number Petition 870250092380, dated 09 / 10 / 2025, pp. 107 / 153 100 / 121 In 1902, the interpolation function extends linearly from a value of 0 (shown as reference number 1914) to 1 (as shown in reference number 1916). It should be noted that the interpolation function is a direct linear function, which is shown in reference number 1915.

[0280] However, in the case of a shortened frame size, which can be obtained using the concept of Figure 5c, a frame length is equal to Lseg + Lrem, and is less than 2 Lseg. This situation is shown in reference number 1922. An abscissa of 1930 describes the temporal extent of the shortened frame. An ordinate of 1932 describes a value of the interpolation function. However, it can be observed that the interpolation function increases from 0, which is shown as reference number 1934, to a first value less than 1, which is shown as reference number 1935, during the initial portion of the Lseg length of the shortened frame. Furthermore, the interpolation function increases from the first value (shown in reference number 1935) to a final value of 1, which is shown in reference number 1936, during the final portion of the Lrem length of the shortened frame.It becomes evident that the slope of the interpolation function is greater in the initial portion of the shortened frame (for example, in the initial portion 521e of the shortened frame of length Lseg) when compared to the slope in the final portion (for example, in the final portion 521f) of the shortened frame of length Lrem. It becomes evident that the slope (of the interpolation function) is greater in the initial portion 521e in which the overlay and addition operation is performed when compared to the final portion 521f of the shortened frame.

[0281] A different situation arises in the case of an elongated frame size. The case of the elongated frame size is shown in reference number 1942. The elongated frame comprises a length of Lseg + Lrem, where Lrem is greater than Lseg, and where the length of the elongated frame is greater than 2 Lseg. A temporal extension of the elongated frame is shown on an abscissa of 1950. A value of the interpolation function is shown on an ordinate of 1952. It can be observed that in the initial portion (e.g., in the initial portion 531e) of the elongated output frame, in which the overlapping and addition is performed, the slope of the interpolation function is smaller than in the final portion of the elongated output frame (e.g., in the final portion 531f). For example, the interpolation function may start at a value of zero, which is shown in reference number 1954, and increase up to a Petition 870250092380, dated 09 / 10 / 2025, pp. 108 / 153 The first value, 101 / 121, shown in reference number 1955, occurs during the initial part of the Lseg length. The interpolation function can then increase to a final value of one, shown in reference number 1956, during the final portion (of length Lren) of the stretched output frame. In other words, the slope of the interpolation function is small in the part where the overlapping and addition, which performs the stretching, is carried out, and the slope of the interpolation function is larger in the remaining part, which is not modified. However, it has been found that such adaptation of the interpolation function provides good interpolation of the processing parameters, resulting in good audio quality. 16. OTHER ASPECTS AND MODALITIES

[0282] Other aspects and embodiments of the invention will be described below, which can be used individually and in combination with the other features, functionalities and details described in this document.

[0283] In other words, it should be noted that the modalities described below can be used individually or can be supplemented by any of the resources, functionalities and details disclosed in this document.

[0284] On the other hand, any of the modalities disclosed in this document may optionally be supplemented by any of the resources, functionalities and details disclosed in this section. HIGHER LEVEL

[0285] According to one aspect of the invention, the main idea of ​​the invention is shown in Figure 9, wherein an audio processor with multiple processing stages has a time-scaling modification performed on one of the intermediate audio signals.

[0286] In one embodiment (Figure 10), processing parameters are used at each processing step.

[0287] In another embodiment (Figure 11), a common time scale controller is present, which controls the time scale modification itself and, based on the time scale result, adapts the processing steps to the time scale modification.

[0288] In another embodiment (Figure 12), the time scale controller also controls the adaptation of the processing parameters of each processing step. Petition 870250092380, dated 09 / 10 / 2025, pp. 109 / 153 102 / 121 CONTROL OF THE SPEED OF ENCODED AUDIO PLAYBACK

[0289] Figure 13 shows a block diagram of a playback speed control based on the invention. JBM Communication Codec

[0290] Figure 14 shows a block diagram of a JBM for a communication codec based on the invention. For example, among other features, the difference and inventive step compared to the methods described in Figure 2 and Figure 8 are in the Decoded Audio Signal block and in the way the adaptation control logic directs the processing blocks. The adaptation control logic is, for example, an embodiment of the time scale controller mentioned earlier. As mentioned in the prior art overview, in a communication codec such as IVAS, the decoded audio signal is reconstructed (or rebuilt) by decoding transport channels and associated metadata and by reconstructing the output channels in a reconstruction / rendering stage that uses the transport channels, the associated metadata and, optionally, additional rendering parameters from outside.

[0291] According to one aspect, the invention, instead of applying the TSM controlled by the Temporary Variation Storage control to the reconstructed output channels, applies the same to the decoded transport channels. For example, depending on the Temporary Variation Storage control, the decoded frames of the transport channels can be shortened or lengthened or kept as they are. For example, if the number of transport channels is greater than one, the TSM is configured to apply the same timescale modification to all transport channels in a synchronized manner, for example, to ensure that the duration of the resulting timescale-modified transport channels placed in the renderer has the same duration over time and the same modification over time.

[0292] In addition, for example, Temporary Variation Storage Control also controls the adaptation of metadata so that optional metadata associated with transport channels is adapted to the altered duration of the modified transport channels in the timescale.

[0293] According to one aspect, for some possible transported formats Petition 870250092380, dated 09 / 10 / 2025, pp. 110 / 153 103 / 121 there is no metadata required to reconstruct the encoded format; in this case, the transport channels already represent the encoded format. If, in this case, the desired output format is, for example, also equivalent to the encoded format, the renderer will not be necessary and the transport channels already represent the output channels, and the renderer block is ignored or can be seen as a simple pass-through block.

[0294] Figure 15a shows a possible (optional) treatment of the Receiver's temporary storage for the invention. The receiver's output temporary storage, compared to Figure 2, must have the capacity (or, in some cases, must even have the capacity) to hold a plurality of output channels. One use case for such a codec is, for example, split processing of the decoded signal, i.e., part of the decoding and rendering is done on one device and the remaining part on another linked device, for example, a portable device as the first device and a so-called edge device as the second device, where the final rendering and the acoustic front-end are located (e.g., glasses for augmented / virtual reality).In this case, the receiver's temporary output storage can, for example, be configured to provide PCM samples or samples of the frequency domain representation of the requested output format (for example, as samples in the CLDFB domain). Furthermore, the duration of the requested segments from the receiver's temporary output storage can, for example, not be the duration of the decoded frames, but an integer fraction of that duration.

[0295] Figure 15b shows an alternative implementation of the temporary storage handling. For more details, see the description above. Temporary storage of TSM transport channels for FD reconstruction / rendering.

[0296] For example, the renderer can be a time-domain-based renderer or a renderer applied to a time-frequency transformed representation of the transport channel, for example, employing a Low Latency Complex Filter Bank (CLDFB). Time-frequency transforms typically (but not necessarily) have a lower time resolution than the time resolution of the time-domain signal, i.e., the time signal is, for example, divided into finite-duration intervals where, for example, each Petition 870250092380, dated 09 / 10 / 2025, pp. 111 / 153 The 104 / 121 interval is transformed into the frequency representation of that interval using a time-frequency transformation.

[0297] The TSM module can, for example, produce a modified frame duration (e.g., modified time scale) that is not an integer multiple of the TF Transform interval size (e.g., time-frequency transform). Therefore, it is advantageous to only pass the number of samples to the FD renderer (e.g., frequency domain renderer) that represents the largest possible number of intervals. For example, any excess samples are stored in temporary storage and attached to the modified TSM transport channels of the next frame.In other words, the time-domain samples placed in the FD renderer comprise the samples stored in temporary storage from the transport channels of the previous frame, the samples from the TSM transport channels of the current frame, shortened to the largest possible number of FD slots based on the sum of the length of the samples stored in temporary storage and the length of the samples from the TSM transport channel of the current frame. Overall, this results in a possible minor incompatibility with the associated metadata, but since the temporal resolution of the metadata is typically lower than the temporal resolution of the TF transformation—that is, for example, a set of metadata parameters is associated with multiple slots of the FD representation of the transport channels—the impact on the quality of the reconstructed and rendered output channels is minimal.

[0298] Figure 16 shows, for example, the process flow for this modality; the example is for a shortened frame; for an elongated TC frame, the processing is performed analogously. The TC frame is, for example, processed by the TSM, resulting in a shortened TC frame. The complete time-domain samples available for the current frame are now, for example, constructed by taking the TC samples stored in temporary storage from the previous frame (1) and appending the shortened TC frame. The frame to be constructed is now, for example, taken as the portion of samples that fits the largest possible number of TS (e.g., time intervals) for the time-frequency transformation, and the possible remaining samples from the available samples (2) are, for example, stored in temporary storage for Petition 870250092380, dated 09 / 10 / 2025, pp. 112 / 153 105 / 121 usage in the next frame. The remaining processing is then, for example, performed in the manner already described in the time-frequency domain transformation frame to reconstruct / render.

[0299] Note that, for example, for a renderer processing the decoded transport channel in the time domain, such temporary storage is not necessary, and the complete frame of the decoded transport channels modified on the time scale can, for example, be rendered in the current frame.

[0300] The TSM operation in [2] contains a special mode (section 5.4.3.4) where for very quiet frames the timescale modification does not operate in a quality-controlled manner, but a maximum stretching or shrinking is performed. In one embodiment of the invention, in frames classified as quiet, the signal is not, for example, stretched or shortened to the maximum allowed values, but, for example, in such a way that no sample of the transport channels needs to be stored in temporary storage in the case of an FD renderer, i.e., the TSM transport channels (e.g., timescale-modified) together with the samples stored in temporary storage form a frame to be rendered that has a length that is an integer multiple of the time interval size of the time-frequency transformation without excessive samples needing to be stored in temporary storage.This results in advantageous synchronization of the TSM transport channels with the frame edge. ADAPTATION OF THE RECONSTRUCTION / RENDERING

[0301] As stated in the reconstruction, the metadata associated with the decoded transport channels may, for example, comprise several sets of metadata, for example, each associated with a subframe of the decoded transport channel frame. In this case, it will, for example, be advantageous to adapt the rendering and / or metadata according to the timescale modification of the decoded transport channels. For example, it is also advantageous to adapt certain internal parameters of a renderer to the applied TSM, for example, interpolation steps. RENDERING: SUBFRAME SIZE ADJUSTMENT

[0302] In one embodiment, the size of the subframes, that is, for example, the number of FD slots in the case of FD domain processing or the number of Petition 870250092380, dated 09 / 10 / 2025, pp. 113 / 153 106 / 121 time-domain samples in the case of time-domain processing in the renderer, which includes each of the subframes within a frame, is adapted based on the modification of the transport channel timescale.

[0303] Figure 17 shows an example of such a modality (in this case with FD / slot processing and TSM leading to a modified transport channel frame on the shortened timescale). The parameters in the simplest case can be the unmodified parameters associated with the subframes of the decoded transport channel frame before TSM. Since TSM typically (but not necessarily) involves overlaying and adding different segments of the unmodified decoded transport channels in another modality, it is, for example, also advantageous to adapt the metadata for subframes where the overlay and addition operations have been performed, for example, by a linear combination of the metadata associated with the unmodified decoded transport channel subframes that are involved in the overlay and addition process. RENDERING: KEEP SUBFRAME SIZE CONSTANT, ADAPT METADATA

[0304] The proposed system may, for example, be required to provide portions of the signal that are shorter in duration than the duration of a decoded frame (Figure 15) and / or provide external parameters for rendering (Figure 14). The temporal resolution of dynamic external parameters, for example, the orientation of a listener, would typically have, for example, a temporal resolution equal to the subframes of the rendering. For these cases, also considering a motion delay for sound as small as possible, rendering with a constant subframe size is advantageous.

[0305] Figure 10 (or Figure 16 or 18) shows an example of a mode that achieves this (the example is again for FD processing in the renderer with subframes). Unlike the previous modes with adapted subframe size, the subframe size, for example, is not changed in relation to the non-adapted case, but, for example, the subframe grid is shifted, for example, according to previous timescale modifications and the timescale modification in the current frame.

[0306] In the Figure, during the rendering stage, some slots of the first subframe Petition 870250092380, dated 09 / 10 / 2025, pp. 114 / 153 107 / 121, for example, were already processed in the previous frame (indicated by the hash padding and time intervals indicated by TS-i), and the first subframe is, for example, shortened by that number, or in other words, it is the number of intervals that still need to be processed to complete that subframe. For example, subsequent subframes have a constant size. The last subframe in the frame may, for example, exceed the size of the frame to be processed, so, for example, only the number of intervals that fit within the frames are processed, and the number of intervals from that subframe that still need to be processed is stored in temporary storage for the next frame.

[0307] For example, in general, the number of subframes may now be different from the number of subframes in a frame where no TSM was applied. Therefore, it is, for example, advantageous to adapt the metadata to the different number of subframes and / or the modification of the applied timescale, for example, by a linear combination for subframes where overlay and addition processing was applied in the TSM and / or a linear interpolation of the parameter set over the new number of subframes. An additional advantage in this mode is that, for example, optional external parameters for the renderer that have the same time resolution as the subframe processing do not need to be adapted and can be used directly.

[0308] Note that for all modality examples showing FD reconstruction / rendering and TC frame shortening with TSM, examples with frame stretching and / or time-domain rendering follow directly. DETAILS OF IMPLEMENTATION OF PROCESSING PARAMETERS AND METADATA ADAPTATION FOR IVAS JBM Adapt interpolation functions for complete frames.

[0309] Some IVAS technologies (e.g., Multi-Channel Object-Based Parametric Audio Coding) have only one set of metadata per frame, so in these cases, adaptation of the metadata itself to timescale changes is not necessary. In reconstruction and rendering, these formats typically (but not necessarily) generate a mixing rule (e.g., a reconstruction / rendering rule or matrix) (e.g., a Petition 870250092380, dated 09 / 10 / 2025, pages 115 / 153 108 / 121 mixing matrix)M for the current frame and, together with the mixing matrix, generate the rendered output signal using linear interpolation between the current and previous mixing rule. For example, mixing matrices can be a single wideband mixing rule for time-domain processing or a band-dependent rule for frequency-domain processing. Therefore, for a frame with unchanged length, the time-domain output signal, in the case of time-domain processing, is: y [n] = (í[n]M + (1 - g [n])Mplw)a:[n],0 < n < (2Lsec- 1)

[0310] where are the transport channel samples in the sample instance71 (and possibly uncorrelated versions of the transport channels), 5Í71] is the interpolator function and yM are the resulting output channel samples in the sample instance71, and L™ is half the frame length in samples.

[0311] Similar to frequency domain processing of an unaltered frame: yJ71] = GffalMfc + (1 -5[n])Mk,pl191,)xk[n],0 < n < (2L„g- 1)

[0312] where * denotes the band index of the frequency representation used and is half the number of time intervals of the frequency domain in an unaltered frame and 71 is the time interval index to be processed.

[0313] Generally, the linear interpolator in these cases takes the form: r i71 + 1

[0314] By using the time scale modification of [2], it is possible to see that, in some cases, there is always a remaining part (rem in Figures 3 and 4) that remains unchanged and the first part of lengthL^gé is an added overlay version that is generated with parts of the previous frame or a shortened version of the current frame.

[0315] According to one aspect, it is therefore a favorable adaptation of the interpolation function to maintain the slope of the unadapted interpolator in the remaining part and adapt the section with the addition of overlap (the first long part). Again here, if we employ, for example, temporary storage of Petition 870250092380, dated 09 / 10 / 2025, pp. 116 / 153 109 / 121 transport channels to maintain a certain time resolution, it is assumed that the influence on quality is small if Lrsm is assumed to be:

[0316] where Lf is the length of the frame to be processed (and Lrem is no longer, for example, the actual length of the remaining part of the transport channels after the timescale modification)

[0317] This interpolator adaptation principle is illustrated in Figure 19.

[0318] So, in general, for example, the adapted interpolation function now has the form: M = TI + 1 + (Lseg Usjji) . ---------------,LSgg <n<Lfr । 2Lseg + 1—Lrgm(1)' ”'

[0319] It is generally, for example, favorable to restrict 8 to M to the values ​​°: 8a.f M = max (0,£π[ti])

[0320] So now the processing for the frame adapted for time-domain processing is, for example: y[n] = (sa,f [Φ1+ (1 - 8a,f [n])Mp.ei,)a:[n],0 < n < Lf

[0321] And, for frequency domain processing, for example: yk M = (8a,f [n]Mk+ <n<Lf

[0322] For example, a loop-based pseudocode implementation of the adapted interpolator function is (with C-style array indexing, i.e., the first entry in the array has index zero (0)): Algorithm 1: Pseudocode to generate the adapted interpolator = get-interpolatoríUsg- Lf) begin float dec = 1 / (2*^^)32?π[^-1]=1 7 7 for (n=L / -2 ; n >=Ls<>g ; n = n-1) start 5n[n] = max(0, £n[n+1]-dec) Petition 870250092380, dated 09 / 10 / 2025, pp. 117 / 153 110 / 121 end8se > o ) start ofC = 3a[LSes]lLSeg for ( n = -1 ; n >= 0 ; n=n-1 ) 5n[n] = max(0, £n[n+1]-dec) end or j for ( n =Lseg-1 ; n >= 0 ; n=n-1 )162?n[n] = 0 end end 19 return 3a 20 end SUBFRAME METADATA ADAPTATION BY MAPPING

[0323] Other IVAS technologies (DirAC, MASA, McMASA), mainly those based on scene-based audio (SBA), where the audio to be encoded is normally presented in Ambisonics, employ, for example, several sets of metadata per frame, i.e., they have a set of metadata per subframe. In this case, processing is performed, for example, per subframe, based on the metadata of that specific subframe. Typically (but not necessarily), a mixing rule is generated from the metadata and, optionally, based on some characteristic of the signal, for example, the combined energy of the transport channels, the so-called reference energy. Furthermore, the rendering itself is performed, for example, by linear interpolation between the mixing rule of the previous subframe with an interpolator such as , where now is, for example, the length (in time intervals) of the subframe.

[0324] For example, here too we generally like to follow the idea presented in the full-frame adaptation of the interpolators, i.e., for the remaining part which represents unaltered portions of the transport channels, the metadata was also used as unchanged as possible and was adapted only in the parts where signals were superimposed / added in the scale modification. Petition 870250092380, dated 09 / 10 / 2025, pp. 118 / 153 111 / 121 of time.

[0325] Metadata is typically (but not necessarily) stored in temporary storage where, for example, each subframe is stored in a temporary storage range, where the index of that range is the subframe index.

[0326] For example, for certain combinations of technologies that share common metadata, a delay compensation is required for the metadata (e.g., by using DirAC metadata for SPAR processing). In this case, the metadata may, for example, be stored in a temporary ring storage (module) that contains more metadata ranges than are needed for an unaltered frame, generally the number of ranges needed for an unaltered frame and the number of metadata ranges needed for delay compensation, where the length of the temporary metadata storage is Lmd. For example, to access correct metadata for processing in a given technology, an offset cf / ^t- in the metadata is determined, for example, for each frame to be processed.

[0327] For example, for metadata adaptation, a mapping vector 1- / M is now created, in which, for example, for each interval, the most suitable subframe index is stored. This can be accomplished, for example, using the rule 9a,maPΙλΐ5η,ϊηίΙρ M being an interpolator with the same characteristic as 9aM, but suitable for generating mapping indices for a system where the indices in matrices or vectors start with zero; for systems with indices starting with 1, a similar interpolator can be derived directly from the following, adding one to the resulting map: + (^seg UsjíO 2Lseg- 1 'Ee3^•^“Seg Ue;ji . 71-----7-----------Γ, 0 < [τι] = moduloÇncf fsgt+ max 'seg ' jb^md) where L 1 denotes rounding to the nearest integer, U / is the length of a subframe, and the division in the formula for is a division Petition 870250092380, dated 09 / 10 / 2025, pp. 119 / 153 112 / 121 is an integer division (i.e., a Euclidean division that produces an integer and a remainder, which is not used in this case), and modulo is the modulo operation with the divisor End (i.e., a function that retrieves the remainder of an integer division).

[0328] The following pseudocode algorithm is one possible way to implement this mapping: Algorithm 2: generating a metadata mapm3f = get_map(£,Sfl.íj' start int src_idx, src_idx_map float src_idx_f t int src_idx = 2*Lse. <j-1 float dec para (n="Lr-1" ; n>=Lsea; n = n-1) begin ® src_idx_map = max(0,src_idx / £'sf)msf [n] = modulo(src_idx_map + ) src_idx = src_idx - 1 end if ( src_idx > 0 ) begin i dec = (src_idx + 1) / src_idx_f = (src_idx + 1) - dec for ( n =Lse.<} -1 ; n >= 0 ; n=n-1 ) ® src_idx = max(0, [src_idx_f] / íJf)m5f[n] = modulo(src_idx +nofp0r),Un[i ) src_idx_f = src_idx_f - dec end or for ( n =Lsea -1 ; n >= 0 ; n=n-1 ) = nDff3Btend end # integer division # float division # integer division Petition 870250092380, dated 09 / 10 / 2025, pages 120 / 153 113 / 121 returnmsf 28 end

[0329] In the processing itself, for the generation of the mixing rule, if it is not dependent on one or more properties of the transport channel signal, it can be generated, for example, using the average of the metadata over the subframe, which can be seen as a linear interpolation of the metadata based on the characteristic of the interpolator (frame). Another possibility (example) is to generate a mixing rule or an intermediate set of parameters per interval based on the metadata of the slot indicated by the metadata mapping vector and calculate the average of the mixing rules or the intermediate parameter. The choice of which domain the average will be calculated depends, for example, on which is the most suitable for the average.In cases where the mixing rule also takes into account the properties of the transport channel signal, it is often advantageous to average over the domain of the intermediate or mixing rule; that is, if, for example, the reference power is the signal characteristic, this can be viewed as a signal (transport channel) energy-weighted interpolation of the metadata.

[0330] Some modes (SBA, DirAC) [7] not only understand encoded and transmitted metadata for multiple subframes, but, for example, also apply metadata analysis to the transport channels at the decoder reconstruction / rendering stage with a time resolution higher than the transported metadata, for example, one per time interval for bands where no metadata was transmitted.

[0331] In this case, the internal temporary storage also stores the transmitted metadata intermittently, so the mapping vector in this case is, for example: ?JifJ[?i] = max (0, [(2L30Í- 1 )^njín[tí3[?i]])

[0332] This can, for example, be implemented using Algorithm 2 and defining Gy=\

[0333] Typically (but not necessarily), the decoder metadata analysis also stores the estimated metadata in the same Petition 870250092380, dated 09 / 10 / 2025, pp. 121 / 153 114 / 121 Internal temporary storage. For adapted subframe processing, as the map can, for a number of consecutive intervals, process points for the same interval from the internal temporary metadata storage, it is preferable to copy the metadata of a subframe to a local temporary storage before processing the subframe to avoid overwriting the metadata estimated in the decoder for the same interval (source) during processing.

[0334] The mapping vector can, for example, also be employed for IVAS modes when the mixing rules do not depend on the signal properties and are calculated and stored based on the assumption of a frame to be processed with unchanged length. The final mixing rule for an interval is then determined based on the (unchanged) interval index. In the case of an adapted length frame, it can be used to map the interval index of the interval to be processed in the adapted length frame to the unchanged interval index to determine the mixing rule for that interval. ADAPTING A TRANSITIONAL POSITION INDICATOR

[0335] Some IVAS technologies (e.g., ParamMC[6]) have metadata that indicates a transient position ητ. The TSM can, for example, shift the transient position in the frame to be processed, so that the transient position can, for example, be adapted as follows: = max(0,nr+ - 2L^g}')

[0336] where ητ,α is the adapted transient position and Lf and L^a are the length of the frame to be processed and the length of the TSM segment expressed in the time resolution of the transient position metadata, for example, the time resolution of the transient position is 2 CLDFB time intervals (ParamMC). ADAPTING SUBFRAME PROCESSING INTERPOLATION

[0337] In the proposed system, the time interval size of a subframe to be processed may, for example, be smaller than the standard processing subframe size. When an interpolator (e.g., interpolating blending rules of the current subframe and the previous subframe) is used in processing, the interpolation may, for example, simply be adapted to the smaller size of the Petition 870250092380, dated 09 / 10 / 2025, pp. 122 / 153 115 / 121 subframe, scaling it linearly to the size of the subframe to be processed. So, for example, if a linear interpolator is used, the length of the interpolator will simply be adapted to the size of the subframe to be processed. 17. IMPLEMENTATION ALTERNATIVES

[0338] Although some aspects have been described in the context of an apparatus, it is clear that these aspects also represent a description of the corresponding method, where a block or device corresponds to a step of the method or to a feature of a step of the method. Analogously, the aspects described in the context of a step of the method also represent a description of a corresponding block, item, or feature of a corresponding apparatus. Some or all of the steps of the method may be performed by (or using) a hardware apparatus, such as a microprocessor, a programmable computer, or an electronic circuit. In some embodiments, one or more of the most important steps of the method may be performed by such an apparatus.

[0339] Depending on certain implementation requirements, embodiments of the invention may be implemented in hardware or in software. Implementation may be carried out using a digital storage medium, for example, a floppy disk, a DVD, a Blu-ray, a CD, a ROM, a PROM, an EPROM, an EEPROM or a FLASH memory, having electronically readable control signals stored therein, which cooperate (or have the capacity to cooperate) with a programmable computer system so that the respective method is executed. Therefore, the digital storage medium may be computer-readable.

[0340] Some embodiments of the invention comprise a data carrier with electronically readable control signals, which have the capability to cooperate with a programmable computer system so that one of the methods described in this document is executed.

[0341] In general, embodiments of the present invention can be implemented as a computer program product with program code, wherein the program code is operable to execute one of the methods when the computer program product is run on a computer. The program code can, for example, be stored on a reader-readable medium. Petition 870250092380, dated 09 / 10 / 2025, pp. 123 / 153 116 / 121 machine.

[0342] Other embodiments include the computer program for executing one of the methods described in this document, stored on a machine-readable medium.

[0343] In other words, an embodiment of the inventive method is therefore a computer program that has program code to execute one of the methods described in this document, when the computer program is run on a computer.

[0344] Another embodiment of inventive methods is, therefore, a data carrier (or a digital storage medium, or a computer-readable medium) comprising, recorded thereon, the computer program for executing one of the methods described in this document. The data carrier, the digital storage medium or the recorded medium is typically tangible and / or non-transitory.

[0345] Another embodiment of the inventive method is, therefore, a data stream or a sequence of signals that represents the computer program for executing one of the methods described in this document. The data stream or sequence of signals may, for example, be configured to be transferred via a data communication connection, for example, via the Internet.

[0346] Another embodiment comprises a means of processing, for example, a computer or a programmable logic device, configured or adapted to perform one of the methods described in this document.

[0347] Another embodiment comprises a computer that has the computer program installed on it to execute one of the methods described in this document.

[0348] Another embodiment of the invention comprises an apparatus or system configured to transfer (e.g., electronically or optically) a computer program to execute one of the methods described herein to a receiver. The receiver may be, for example, a computer, a mobile device, a memory device, or the like. The apparatus or system may, for example, comprise a file server for transferring the computer program to the receiver.

[0349] In some embodiments, a programmable logic device (e.g., Petition 870250092380, dated 09 / 10 / 2025, pp. 124 / 153 117 / 121 a field-programmable gate array) can be used to perform some or all of the functionalities of the methods described in this document. In some embodiments, a set of field-programmable gates can cooperate with a microprocessor to perform one of the methods described in this document. Generally, the methods are preferably performed by any hardware device.

[0350] The device described in this document can be implemented using a hardware device, or using a computer, or using a combination of a hardware device and a computer.

[0351] The device described in this document, or any components of the device described in this document, may be implemented at least partially in hardware and / or software.

[0352] The methods described in this document can be performed using a hardware device, or using a computer, or using a combination of a hardware device and a computer.

[0353] The methods described in this document, or any components of the apparatus described in this document, may be implemented at least partially by hardware and / or by software.

[0354] The embodiments described above are merely illustrative of the principles of the present invention. It is understood that modifications and variations of the arrangements and details described herein will be apparent to those skilled in the art. The intention, therefore, is to be limited only by the scope of the imminent patent claims and not by the specific details presented through the description and explanation of the embodiments described herein. 19. ADDITIONAL MODALITIES

[0355] Other aspects and embodiments of the invention will be described below.

[0356] It is worth noting that the aspects described below can be used individually or in combination.

[0357] It should also be noted that the aspects described below may optionally be supplemented by any of the resources, functionalities and details disclosed in this document.

[0358] In addition, any of the aspects disclosed below may optionally be introduced into any of the other disclosed modalities. Petition 870250092380, dated 09 / 10 / 2025, pages 125 / 153 118 / 121 in this document, both individually and in combination. 1. An audio processor for delivering time-scaled audio content, where the audio processor has multiple processing stages and intermediate audio signals when time scaling is performed on an intermediate audio signal. (Figure 9) 2. The audio processor with parameters for the processing steps (Figure 10, the parameters are, for example, the decoded metadata or rendering parameters according to Figure 6 that are applied sequentially) 3. The audio processor in which the timescale is controlled by a timescale controller (Figure 11) 4. The audio processor with parameter adapters (Figure 12) 5. The audio processor according to aspect 2, where the parameter for a processing step is metadata (e.g., spatial metadata describing how to perform bottom-up mixing). 6. The audio processor of aspect 5 where metadata is associated with the intermediate audio signal (e.g., spatial metadata with subframes synchronized with the TD signal) 7. The audio processor, according to aspects 4 to 6, where the metadata adapter is configured to adapt metadata associated with audio samples from the transport channel (i.e., the metadata adapter follows the timescale ratio of the TSM of the TD signal). 8. The audio processor, according to the previous aspects, uses a time-domain signal as the intermediate audio signal, and time scaling is performed on the time-domain signal. 9. The audio processor, according to the previous aspects, where the time scaler is configured to synchronize the time scaling operation across multiple audio channels (e.g., multi-channel TSM that provides the same number of samples for each channel). 10. (Cancelled) 11. Aspect dependent on temporary storage of TSM transport channel samples to maintain a time resolution FD (Figure 16, see description) Petition 870250092380, dated 09 / 10 / 2025, pp. 126 / 153 119 / 121 12. Dependent aspect to adapt the subframe size and / or metadata for the renderer for each associated subframe based on the TSM applied in the TCs (Figure 17, see description) 13. Aspect-dependent for metadata adaptation and initial subframe shift for rendering with constant subframe size (Figure 18, see description) 14. Aspect dependent on the adapted TSM in silent frames (no figure, see description) 15. Dependent aspect describing a decoder that contains the audio processor, and also another view that the decoder is one of the processing steps. 16. Dependent aspect: the processing steps can be decoding, rendering, reconstruction, encoding (in the case of another interface, for example, at the edge). 17. An audio processor, in accordance with the previous aspects, where the representation of the audio input signal comes from a temporary input storage. 18. An audio processor in accordance with the preceding aspects, wherein the input temporary storage for the decoder is a variation temporary storage and the time scale control is a variation temporary storage control (see, for example, [2] or [3]) 19. An audio processor according to the preceding aspects comprising a decoder for generating transport channels, a renderer for reconstructing an encoded or intermediate audio format from transport channels and associated metadata and rendering it into an output format; (The renderer is a single box in the figures, but in fact has two stages) 20. (The audio processor, according to the previous aspects, where the transport channels already represent the output channels and the renderer is just a pass-through block) 21. The audio processor according to aspect 19, where the renderer is configured to reconstruct the encoded audio format or an intermediate format and render the encoded audio format or the intermediate audio format into an output format in two processing steps. Petition 870250092380, dated 09 / 10 / 2025, pp. 127 / 153 120 / 121 consecutive. 22. An audio processor according to the preceding aspects comprising an interface for providing timescaled transport channels and associated adapted metadata for subsequent rendering; (see description, for example, renderer splitting on the end device, decoder on the edge device, connected in the FD domain, plus parameters) 23. The audio processor according to one of the previous aspects, where the renderer is configured to process the reconstruction of the encoded and intermediate output format and the rendering to the output format in a combined processing step (see most Figures where rendering and reconstruction are a single block) 20. CONCLUSIONS

[0359] In conclusion, the embodiments according to the invention relate to a novel multi-stage audio processing system with intermediate audio signals, in which a timescale modification is applied to one of the intermediate audio signals. Furthermore, the embodiments according to the invention show ways of adapting processing steps and processing parameters based on the timescale modification of the intermediate audio signal, controlled by a common timescale modification controller. As a detailed embodiment, a temporary variation storage management solution is disclosed for a communication codec capable of encoding and / or decoding complex audio scenes based on transport channels and associated metadata to reconstruct encoded audio formats.

[0360] The modalities are related to multi-stage audio processing, time scale modification, parameter adaptation, playback speed, JBM and / or IVAS.

[0361] The embodiments according to the invention are usable in the context of 3GPP IVAS. REFERENCES [1] Kramer, Kris W.; Robert, Chris C. Forrester; Joly, Nepean: JITTER BUFFER MANAGEMENT; Patent No. US 6,658,027 B1; December 2003 [2 ] ETSI TS 126 448: Universal Mobile Telecommunications System Petition 870250092380, dated 09 / 10 / 2025, pp. 128 / 153 121 / 121 (UMTS); LTE; EVS Codec Jitter Buffer Management (3GPP TS 26.448 version 12.0.0 Release 12) [3] Reuschl, Stefan; Dohla, Stefan; Lecomte, Jeremy; Jander, Manuel: Jitter Buffer Control, Audio Decoder, Method and Computer Program, Patent No. US 9,997,162 B2; June 2018 [4] IVAS Design Constraints (IVAS-4), Version 1.0.0, https: / / www.3gpp.org / ftp / tsg_sa / WG4_CODEC / IVAS_Permanent_Documents / IVAS4_S4-2211619.zip [5 ] Fuchs, Guillaume, et al: Apparatus and Method for Encoding or Decoding Directional Audio Coding Parameters using Quantization and Entropy Coding; Patent No. US 11,367,454 B2, June 2022 [6] Bayer, Stefan; Boutheon, Alexandre; Fuchs, Guillaume; Multrus, Markus; Küch, Fabian; Thiergart, Oliver; Disch, Sascha; Herre, Jürgen: Parameter Encoding and Decoding, Patent Application No. EP3984028A2, June 2020 [7] Bayer, Stefan, et al: Audio Scene Encoder, Audio Scene Decoder and Related Methods Using Hybrid Encoder / Decoder Spatial Analysis, Patent Descriptive Report EP 3 724 876 B1, May 2022 Petition 870250092380, dated 09 / 10 / 2025, pp. 129 / 153

Claims

1 / 32 CLAIMS 1.An audio processor is designed to provide a representation of a processed audio signal based on an input audio signal representation. The audio processor is characterized by being configured to perform a plurality of processing steps in order to provide the representation of the processed audio signal based on the input audio signal representation. In this case, the audio processor is configured to perform a time-scaling modification on one or more intermediate audio signals, provided by a first processing based on the input audio representation, in order to obtain one or more time-scaling modified intermediate audio signals. Furthermore, the audio processor is configured to perform a second processing, following the first processing, which provides the representation of the processed audio signal based on the one or more time-scaling modified intermediate audio signals.

2. Audio processor, according to claim 1, the audio processor being characterized in that it is configured to perform secondary processing depending on one or more parameters.

3. Audio processor, according to claim 1 or 2, the audio processor being characterized by being configured to adapt the second processing to the modification of the time scale.

4. Audio processor, according to any one of claims 1 to 3, the audio processor being characterized by being configured to adapt one or more processing parameters to the modification of the time scale, in order to obtain adapted processing parameters, wherein the audio processor is configured to perform the second processing depending on the one or more adapted parameters.

5. Audio processor, according to claim 4, characterized in that one or more processing parameters are temporally associated with one or more intermediate audio signals, wherein the audio processor is configured to obtain the adapted processing parameters such that the adapted processing parameters are adapted to a timescale-modified timing of the intermediate audio signals.

6. Audio processor, according to claim 4 or 5, the audio processor being characterized by being configured to obtain the adapted processing parameters using temporal interpolation or using temporal extrapolation.

7. Audio processor, according to any one of claims 1 to 6, the audio processor being characterized in that it is configured to apply a time-scaling modification to one or more decoded transport channels.

8. Audio processor, according to any one of claims 1 to 7, the audio processor being characterized by being configured to obtain or reconstruct a plurality of output audio channels based on one or more decoded transport channels and metadata.

9. Audio processor, according to any one of claims 1 to 8, the audio processor being characterized in that it is configured to apply the same time-scaling modification to a plurality of decoded transport channels.

10. Audio processor, according to any one of claims 1 to 9, the audio processor being characterized by being configured to adapt metadata for obtaining or reconstructing the output audio channels depending on the timescale modification.

11. Audio processor, according to any one of claims 1 to 10, the audio processor being characterized in that it is configured to perform the first processing for time segments of a first length; and wherein the audio processor is configured to perform the second processing for time segments of a second length that is shorter than the first length, or wherein the audio processor is configured to perform the second processing with temporal increments in steps of a second length that is shorter than the first length, wherein one or more intermediate audio signals of the second length are inserted into the second processing for execution of the second processing.

12. Audio processor, according to claim 11, characterized in that the granularity of the timescale modification is finer than the second length, such that the timescale modification can result in frame lengths of one or more timescale-modified intermediate audio signals that deviate by integer multiples of the second length.

13. Audio processor, according to any one of claims 1 to 12, characterized in that the first processing is configured to provide time-domain samples in blocks, wherein a block of time-domain samples provided by the first processing comprises a first number of time-domain samples; and wherein the second processing is configured to operate on blocks of time-domain samples that are based on time-scale modified time-domain samples that are derived from the time-domain samples provided by the first processing using the time-scale modification, wherein the blocks of time-domain samples processed by the second processing comprise a shorter length than the blocks of time-domain samples provided by the first processing;or wherein the second processing is configured to insert, per processing step, a block of a plurality of time-domain samples that is based on time-scale modified time-domain samples that are derived from the time-domain samples provided by the first processing using the time-scale modification, wherein the block of a plurality of time-domain samples Petition 870250092384, dated 09 / 10 / 2025, page 43 / 72 4 / 32 inserted in the second processing per processing step comprises a shorter length than the blocks of time-domain samples provided by the first processing.

14. Audio processor, according to any one of claims 1 to 13, characterized in that blocks of time-domain samples processed by the second processing comprise a plurality of time-domain samples.

15. Audio processor, according to any one of claims 1 to 14, characterized in that metadata, which are used in secondary processing, are associated with subframes of the input audio signal representation and / or are associated with subframes of one or more intermediate audio signals, wherein a subframe comprises one or more time intervals, in which the audio processor is configured to provide the one or more time-scaled modified intermediate audio signals, such that a respective frame of the one or more time-scaled modified intermediate audio signals comprises a different length when compared to a corresponding frame of the input audio signal representation and / or when compared to a corresponding frame of one or more intermediate audio signals.

16. Audio processor, according to any one of claims 1 to 15, the audio signal processor being characterized in that it is configured to provide one or more time-scaled modified intermediate audio signals such that the length of a respective frame of the one or more time-scaled modified intermediate audio signals is different from a multiple of a block size used in the second processing; and / or wherein the audio processor is configured to perform the second processing with temporal increments in steps, wherein the audio signal processor is configured to provide one or more time-scaled modified intermediate audio signals such that Petition 870250092384, dated 09 / 10 / 2025, page 44 / 72 5 / 32 the length of a respective frame of the one or more time-scaled modified intermediate audio signals is different from a multiple of the temporal increments.

17. Audio processor, according to any one of claims 1 to 16, the audio signal processor being characterized in that it is configured to form a sample block for second processing based on one or more time-domain samples associated with a previous frame of one or more time-scaled modified intermediate audio signals and based on one or more time-domain samples associated with a current frame of one or more time-scaled modified intermediate audio signals, and wherein the audio signal processor is configured to form one or more sample blocks for second processing based on a plurality of time-domain samples associated with a current frame of one or more time-scaled modified intermediate audio signals, and wherein the audio signal processor is configured to temporarily store one or more residual time-domain samples, i.e.Excessive time-domain samples that are insufficient to form a sample block for the second processing, associated with a current frame of one or more time-scaled modified intermediate audio signals to form a sample block for the second processing that includes one or more time-domain samples associated with a subsequent frame of one or more time-scaled modified intermediate audio signals.

18. Audio processor, according to any one of claims 1 to 17, the audio processor being characterized by being configured to use one or more temporarily stored residual time-domain samples, i.e., excess time-domain samples that are not sufficient to form a sample block for the second processing, associated with a previous frame and the maximum possible number of time-domain samples associated with the current frame of one or more intermediate audio signals modified in time scale to form sample blocks for the second processing associated with the current frame. Petition 870250092384, dated 09 / 10 / 2025, p. 45 / 72 6 / 32 19. Audio processor, according to any one of claims 1 to 18, characterized in that the temporal resolution of metadata, which is used in the second processing, is lower than the temporal resolution of the second processing.

20. Audio processor, according to any one of claims 1 to 19, characterized in that one or more intermediate audio signals are one or more transport channel signals.

21. Audio processor, according to any one of claims 1 to 20, the audio processor being characterized in that it is configured to perform time scale modification using a quality control mechanism.

22. Audio processor, according to any one of claims 1 to 21, the audio processor being characterized in that it is configured to switch between a time-scaling modification mode with quality control and a time-scaling modification mode without quality control, depending on one or more characteristics of one or more intermediate audio signals and / or depending on side information.

23. Audio processor, according to any one of claims 1 to 22, the audio processor being characterized in that it is configured to operate in a time-scale modification mode without quality control.

24. Audio processor, according to any one of claims 1 to 23, the audio processor being characterized in that it is configured to selectively switch to a special time-scaling modification mode for frames classified as silent; and wherein the audio processor is configured to perform the time-scaling modification in such a way that no temporary storage of residual frames is required during operation in the special time-scaling modification mode.wherein the number of time-domain samples from a modified frame at the current timescale plus the number of time-domain samples stored in temporary storage from a modified frame at a previous timescale is an integer multiple of the number of time-domain samples from a block of time-domain samples processed by the second processing, or wherein the number of time-domain samples from a modified frame at the current timescale plus the number of time-domain samples stored in temporary storage from a modified frame at a previous timescale is an integer multiple of the number of time-domain samples inserted in the second processing to perform a temporal increment during the operation in special timescale modification mode.

25. Audio processor, according to any one of claims 1 to 24, the audio processor being characterized by being configured to adapt a subframe size, which are used in the second processing or which are inserted in the second processing as a block of new samples to make a temporal increment, to the modification of the time scale.

26. Audio processor, according to claim 25, the audio processor being characterized by being configured to adapt a number of time intervals that are processed in the second processing, or that are inserted in the second processing as a block of new samples to make a temporal increment, to the modification of the time scale.

27. Audio processor, according to any one of claims 1 to 26, the audio processor being characterized by being configured to perform timescale modification using one or more overlay and add-on operations by overlaying different segments of one or more intermediate audio signals, wherein the audio processor is configured to adapt metadata whereby overlay and add-on operations are applied to different segments of one or more intermediate audio signals, in order to obtain adapted metadata associated with frames or subframes of the timescale-modified intermediate audio signals.

28. Audio processor, according to claim 27, the audio processor being characterized by being configured to linearly combine metadata associated with frames or subframes of one or more intermediate audio signals that are involved in an overlay and addition operation, to obtain metadata associated with frames or subframes resulting from one or more time-scale modified intermediate audio signals.

29. Audio processor, according to any one of claims 1 to 28, the audio processor being characterized in that it is configured to identify a segment, from among a plurality of candidate segments that are portions of a given audio frame of an intermediate audio signal, for superimposition and addition with a predetermined portion of the given audio frame, using a waveform similarity determination, in order to obtain a shortened version of the given audio frame.

30. Audio processor, according to claim 29, the audio processor being characterized in that it is configured to include, at the end of the shortened version of the given audio frame, an unmodified portion of the given audio frame that follows the segment identified in an original version of the given audio frame.

31. Audio processor, according to any one of claims 1 to 30, the audio processor being characterized in that it is configured to identify a segment, from among a plurality of candidate segments that are portions comprising values ​​of a previous audio frame of an intermediate audio signal preceding a given audio frame, for superimposition and addition with a predetermined portion of the given audio frame, using a waveform similarity determination, in order to obtain an elongated version of the given audio frame.

32. Audio processor, according to claim 31, being the audio processor characterized by being configured to include, at one end of the elongated version of the given audio frame, an unmodified portion comprising values ​​of the given audio frame that follows the identified segment.

33. Audio processor, according to any one of claims 1 to 32, the audio processor being characterized by being configured to interpolate in time between one or more processing parameters associated with a previous frame of one or more intermediate audio signals and one or more processing parameters associated with a current frame of one or more intermediate audio signals, in order to obtain processing parameters associated with different time instances of a current frame of one or more intermediate audio signals modified in time scale using an interpolation function that determines the contributions of the one or more processing parameters associated with the previous frame and the contributions of one or more processing parameters associated with the current frame in a linear combination of processing parameters.

34. Audio processor, according to claim 33, characterized in that the interpolation function is a linear or piecewise linear function that maps a time index to a weighting value that determines the contributions of a processing parameter associated with a previous frame and a processing parameter associated with a current frame to a time-interpolated processing parameter.

35. Audio processor, according to claim 33 or 34, characterized in that, for a temporally shortened audio frame, the interpolation function comprises a greater slope for a portion of the current frame modified in the timescale that was obtained using an overlay-addition combination than for a portion of the current frame modified in the timescale that was not modified; and / or in that, for a temporally lengthened audio frame, the interpolation function comprises a lesser slope for a portion of the current frame modified in the timescale that was obtained using an overlay-addition combination than for a portion of the current frame modified in the timescale that was not modified.

36. Audio processor, according to any one of claims 33 to 35, the audio processor being characterized in that it is configured to obtain interpolated processing parameters Pinter[n] associated with time instances with time index n according to Pinter[n]=ga,f[n] P + (1-ga,f)Pprev, where P is a processing parameter or a set of processing parameters associated with a current frame of one or more intermediate audio signals, where Pprev is a processing parameter or a set of processing parameters associated with a previous frame of one or more intermediate audio signals, wherein Pinter is a processing parameter or a set of processing parameters associated with a time instance n in a current frame of one or more time-scaled modified intermediate audio signals, and wherein ga,f is an interpolation function.

37. Audio processor, according to claim 36, characterized in that the interpolation function ga,f is defined as 9a.f M = max (0, ga [n]j θ where TI + 1 + ---------------,LSgg <n<L r 1 Lre,„ (Ü em que n é um índice de tempo, em que Lrem=Lf-Lseg, em que Lf é o comprimento de um quadro a ser processado; em que Lseg é o comprimento de uma porção na qual uma sobreposição e adição é aplicada pela modificação da escala de tempo. Petição 870250092384, de 09 / 10 / 2025, pág. 50 / 72 11 / 32 38. Audio processor, according to any one of claims 1 to 37, the audio processor being characterized in that it is configured to perform, as a second processing step, the processing of values ​​of one or more intermediate audio signals modified in time scale according to y[íi] = (gBtf [?i]M + (1 - ga>f [ΐΐ])Μρϊ19I,)χ[η], 0 <n<Lf ou de acordo com Vk M = (õa,f [n]Mk + (1 - Sa,f ΗprtfB )xk [n], 0 < n < Lf _ em que x[n] e xk[n] são valores ou conjuntos de valores do um ou mais sinais de áudio intermediários modificados em escala de tempo, em que M é um parâmetro de processamento ou um conjunto de parâmetros de processamento associados a um quadro atual do um ou mais sinais de áudio intermediários;where Mprev is a processing parameter or a set of processing parameters associated with a previous frame of one or more intermediate audio signals, where n is a time index, where Lf is the length of a frame to be processed, and where ga,f is an interpolation function.

39. Audio processor, according to any one of claims 1 to 38, the audio processor being characterized in that it is configured to perform secondary processing using processing parameters associated with subframes of an audio frame.

40. Audio processor, according to any one of claims 1 to 39, the audio processor being characterized by being configured to obtain a processing rule for the second processing using processing parameters associated with a given subframe.

41. Audio processor, according to any one of claims 1 to 40, the audio processor being characterized by being configured to determine a mapping vector that defines an association between time intervals of one or more time-scaled modified intermediate audio signals and subframe indices of one or more original intermediate audio signals depending on the timescale modification, such that the mapping describes which process parameters are allocated to which time interval of the timescaled modified intermediate audio signals.

42. Audio processor, according to claim 41, wherein the audio processor is characterized by being configured to determine the mapping vector according to wiíZ[n] = moàulo(nDffset. + max [(2L 1)5 M]\ T / or according to m3f [n] = max 1)^ rt.innp Ml where mSf[n] is an element of the mapping vector; where n is a time interval index; where noffset is an offset value; where Lseg is half the length of a frame of one or more original intermediate audio signals at time intervals; where 2Lseg is the length of a frame of one or more original intermediate audio signals at time intervals; where Lsf is the length of a subframe at time intervals; where Lmd is the length of a temporary metadata storage; where noffset is ; and where ga,map[n] is an interpolation function; where L 1 denotes rounding to the nearest integer; where modulo is a modulo operation, and where division is integer division.

43. Audio processor, according to claim 42, Petition 870250092384, dated 09 / 10 / 2025, page 52 / 72 13 / 32 characterized by dseg — 71 < i / , o < 71 < Lsgg 71 Ί (dseg Uem ^seg 1 2 Ue g Uerti where Lf is the length of a frame to be processed in time intervals; where Lrem =Lf-Lseg.

44. Audio processor, according to any one of claims 1 to 43, the audio processor being characterized in that it is configured to determine a mapping vector that defines an association between time intervals of one or more timescale-modified intermediate audio signals and time intervals of one or more original intermediate audio signals depending on the timescale modification, such that the mapping vector describes which processing parameters are allocated to which time interval of the timescale-modified intermediate audio signals.

45. Audio processor, according to claim 44, the audio processor being characterized in that it is configured to determine the mapping vector according to 7ii^ M = max (0, [(2Lje5 - 1)£α,ίηπί3 M]) where n is a time interval index, where 2Lseg is the length of a frame of one or more original intermediate audio signals at time intervals; and where ga,map[n] is an interpolation function.

46. ​​Audio processor, according to any one of claims 1 to 45, the audio processor being characterized in that it is configured to determine a mapping vector that defines an association between time intervals of one or more timescale-modified intermediate audio signals and time intervals of one or more original intermediate audio signals depending on the timescale modification. Petition 870250092384, 09 / 10 / 2025, pp. 53 / 72 14 / 32 such that the mapping vector describes which processing parameters are allocated with which time interval of the timescale-modified intermediate audio signals.

47. Audio processor, according to any one of claims 41 to 46, the audio processor being characterized in that it is configured to determine processing parameters associated with a given subframe of one or more time-scaled modified intermediate audio signals using processing parameters associated with time intervals that make up the given subframe of one or more time-scaled modified intermediate audio signals, or wherein the audio processor can be configured to obtain rendering rules or upmixing rules or interval-based intermediate parameters using processing parameters associated with the respective intervals,and to obtain a rendering rule or an up-blending rule or intermediate parameters associated with a subframe using an average of the rendering rules or up-blending rules or intermediate parameters associated with the slots that make up the subframe.

48. Audio processor, according to any one of claims 1 to 47, the audio processor being characterized in that it is configured to adapt an interpolation between intervals of a subframe to a subframe size obtained by modifying the time scale.

49. Audio processor, according to any one of claims 1 to 48, the audio processor being characterized in that it is configured to modify a grid of subframes of one or more timescaled modified intermediate audio signals, depending on a timescaled modification of a current frame, wherein the lengths of the subframes of the one or more timescaled modified intermediate audio signals are left unchanged, and wherein the second processing is configured to perform a subframe-based operation.

50. Audio processor, according to any one of claims 1 to 49, characterized in that the first processing step is audio decoding.

51. Audio processor, according to any one of claims 1 to 50, characterized in that the first processing comprises at least one of: decoding, rendering, reconstruction and encoding; and / or in that the second processing comprises at least one of: decoding, rendering, reconstruction and encoding.

52. Audio processor, according to any one of claims 1 to 51, the audio processor being characterized in that it is configured to receive the representation of the input audio signal from a temporary input storage.

53. Audio processor, according to any one of claims 1 to 52, the audio processor being characterized in that it is configured to receive the representation of the input audio signal from a temporary variation storage; and wherein the audio processor comprises a time scale control for adjusting the time scale modification, wherein the time scale control is a temporary variation storage control.

54. Audio processor, according to any one of claims 1 to 53, characterized in that the first processing comprises a decoding of encoded transport channels; and in that the second processing comprises a rendering configured to reconstruct an encoded or intermediate audio format based on timescale-modified transport channels and associated metadata and to render the encoded or intermediate audio format into an output format.

55. Audio processor, according to claim 54, characterized in that the rendering is configured to reconstruct the encoded audio format or the intermediate audio format and render the encoded audio format or the intermediate audio format to the output format in two consecutive processing steps.

56. Audio processor, according to any one of claims 1 to 55, characterized in that the audio processor functionalities are distributed to two or more devices, with an interface between them.

57. Audio processor, according to any one of claims 1 to 56, the audio processor being characterized by comprising an interface configured to provide time-scale modified intermediate audio signals and associated metadata for subsequent rendering.

58. Audio processor, according to any one of claims 1 to 57, characterized in that the second processing is a rendering, wherein the rendering is configured to process a reconstruction of an encoded audio format and / or an intermediate audio format and a rendering of the reconstructed encoded audio format and / or the intermediate audio format in a combined processing step.

59. Audio processor, according to any one of claims 1 to 58, characterized in that one or more intermediate audio signals are time-domain signals, and wherein the audio processor is configured to perform time-scaling modification on the intermediate time-domain audio signals.

60. Audio processor, according to any one of claims 1 to 59, the audio processor being characterized by being configured to synchronize a time-scaling operation on a plurality of intermediate audio signals.

61. Audio processor, according to any one of claims 1 to 60, the audio processor being characterized in that it is configured to modify transient position information depending on the modification of the time scale.

62. Audio processor, according to claim 61, the audio processor being characterized in that it is configured to obtain an adapted transient position according to 7irj„ = max(0,nT+ (Lf - 2Lse3)) wherein ητ is an original transient position value, wherein Lf is the frame length of one or more time-scaled modified intermediate audio signals; and wherein 2Lseg is the frame length of one or more original intermediate audio signals.

63. Audio processing system to provide a representation of a processed audio signal based on a representation of an input audio signal, the audio processing system being characterized by being configured to perform a plurality of processing steps in order to provide the representation of the processed audio signal based on the representation of the input audio signal;wherein the system is configured to perform a time-scaling modification on one or more intermediate audio signals, which are provided by a first processing based on the input audio representation, in order to obtain one or more time-scaling modified intermediate audio signals, and wherein the audio processing system is configured to perform a second processing, which follows the first processing and which provides the processed audio signal representation, based on the one or more time-scaling modified intermediate audio signals, or based on signals derived from one or more time-scaling modified intermediate audio signals; wherein the functionalities of the audio processing system are distributed to two or more devices.

64. Audio processing system according to claim 63, the audio processing system being characterized by comprising an interface for providing time-scale modified intermediate audio signals and adapted metadata associated with a second processing that takes place on a device different from the first processing.

65. Audio decoder for providing a decoded audio representation based on an encoded audio representation, the audio decoder being characterized by comprising an audio processor, as defined in any one of claims 1 to 62.

66. A method for providing a processed audio signal representation based on an input audio signal representation, the method being characterized by comprising performing a plurality of processing steps in order to provide the processed audio signal representation based on the input audio signal representation; wherein the method comprises performing a time-scaling modification on one or more intermediate audio signals, which are provided by a first processing based on the input audio representation, in order to obtain one or more time-scaling modified intermediate audio signals, and wherein the method comprises performing a second processing, which follows the first processing and which provides the processed audio signal representation, based on the one or more time-scaling modified intermediate audio signals.

67. Computer program characterized by being able to execute the method, as defined in claim 66, when the computer program is run on a computer.

68. Audio processor to provide a processed audio signal representation based on an input audio signal representation, the audio processor being characterized by being configured to perform a plurality of processing steps in order to provide the processed audio signal representation based on the input audio signal representation;wherein the audio processor is configured to perform a time-scaling modification on one or more intermediate audio signals, which are provided by a first processing, which is performed by the audio processor, based on the input audio representation, in order to obtain one or more time-scaling modified intermediate audio signals, and wherein the audio processor is configured to perform a second processing, which follows the first processing and which provides the processed audio signal representation, based on the one or more time-scaling modified intermediate audio signals; wherein the audio processor is configured to perform the first processing for time segments of a first length;and wherein the audio processor is configured to perform the second processing for time segments of one second length that is shorter than the first length, or wherein the audio processor is configured to perform the second processing with temporal increments of one second length that is shorter than the first length, wherein one or more intermediate audio signals of the second length are inserted into the second processing for each execution of the second processing; wherein a granularity of the timescale modification is finer than the second length, such that the timescale modification may result in frame lengths of one or more timescale-modified intermediate audio signals that deviate by integer multiples of the second length.

69. Audio processor to provide a processed audio signal representation based on an input audio signal representation, the audio processor being characterized by being configured to perform a plurality of processing steps in order to provide the processed audio signal representation based on the input audio signal representation;Petition 870250092384, dated 09 / 10 / 2025, page 59 / 72 20 / 32, in which the audio processor is configured to perform a time-scaling modification on one or more intermediate audio signals, which are provided by a first processing based on the input audio representation, in order to obtain one or more time-scaling modified intermediate audio signals, and in which the audio processor is configured to perform a second processing, which follows the first processing and which provides the representation of the processed audio signal, based on the one or more time-scaling modified intermediate audio signals;wherein the audio signal processor is configured to provide one or more time-scaled modified intermediate audio signals such that the length of a respective frame of one or more time-scaled modified intermediate audio signals is different from a multiple of a block size used in the second processing; and / or wherein the audio processor is configured to perform the second processing with gradual temporal increments, wherein the audio signal processor is configured to provide one or more time-scaled modified intermediate audio signals such that the length of a respective frame of one or more time-scaled modified intermediate audio signals is different from a multiple of the temporal increments.

70. Audio processor for providing a representation of a processed audio signal based on an input audio signal representation, the audio processor being characterized by being configured to perform a plurality of processing steps in order to provide the representation of the processed audio signal based on the input audio signal representation; wherein the audio processor is configured to perform a time-scaling modification on one or more intermediate audio signals, which are provided by a first processing based on the input audio representation, in order to obtain one or more time-scaled modified intermediate audio signals, and wherein the audio processor is configured to perform a second processing, which follows the first processing and which provides the processed audio signal representation.based on one or more time-scaled modified intermediate audio signals; wherein the audio signal processor is configured to form a sample block for second processing based on one or more time-domain samples associated with a previous frame of the one or more time-scaled modified intermediate audio signals and based on one or more time-domain samples associated with a current frame of the one or more time-scaled modified intermediate audio signals, and wherein the audio signal processor is configured to form one or more sample blocks for second processing based on a plurality of time-domain samples associated with a current frame of the one or more time-scaled modified intermediate audio signals,until there are not enough time-domain samples associated with a current frame of one or more remaining time-scaled modified intermediate audio signals to form a sample block for the second processing, such that sample blocks for the second processing of uniform length are formed even if the time-scale modification alters the length of a frame to be different from an integer multiple of the length of the sample blocks used for the second processing, and wherein the audio signal processor is configured to store in temporary storage one or more residual time-domain samples, i.e., excess time-domain samples that are not sufficient for the formation of a sample block for the second processing,associated with a current frame of one or more time-scaled modified intermediate audio signals to form a sample block for the second processing, which includes one or more time-domain samples associated with a subsequent frame of one or more time-scaled modified intermediate audio signals.

71. Audio processor to provide a representation of a processed audio signal based on a representation of an input audio signal, the audio processor being characterized by being configured to perform a plurality of processing steps in order to provide the Petition 870250092384, dated 09 / 10 / 2025, page 61 / 72 22 / 32 representation of the processed audio signal based on the representation of the input audio signal;wherein the audio processor is configured to perform a time-scaling modification on one or more intermediate audio signals, which are provided by a first processing based on the input audio representation, in order to obtain one or more time-scaling modified intermediate audio signals, and wherein the audio processor is configured to perform a second processing, which follows the first processing and which provides the processed audio signal representation, based on the one or more time-scaling modified intermediate audio signals;where the audio processor is configured to use one or more residual time-domain samples stored in temporary storage, i.e., excess time-domain samples that are not sufficient for the formation of a sample block for the second processing, associated with a previous frame and the maximum possible number of time-domain samples associated with the current frame of one or more intermediate audio signals modified in time scale for the formation of sample blocks for the second processing associated with the current frame, so that sample blocks for the second processing with uniform length are formed even if the time-scaling modification alters the length of a frame to be different from an integer multiple of the length of the sample blocks used for the second processing.

72. Audio processor to provide a representation of a processed audio signal based on a representation of an input audio signal, the audio processor being characterized by being configured to perform a plurality of processing steps in order to provide the representation of the processed audio signal based on the representation of the input audio signal;wherein the audio processor is configured to perform a time-scaling modification on one or more intermediate audio signals, which are provided by a first processing based on the input audio representation, in order to obtain one or more time-scaling modified intermediate audio signals, and wherein the audio processor is configured to perform a second processing, which follows the first processing and which provides the processed audio signal representation, based on the one or more time-scaling modified intermediate audio signals; wherein the audio processor is configured to selectively switch to a special time-scaling modification mode for frames classified as silent;and in which the audio processor is configured to perform timescale modification so that no temporary storage of residual frames is required during operation in special timescale modification mode;wherein the number of time-domain samples from a modified frame at the current timescale plus the number of time-domain samples stored in temporary storage from a modified frame at a previous timescale is an integer multiple of the number of time-domain samples from a block of time-domain samples processed by the second processing, or wherein the number of time-domain samples from a modified frame at the current timescale plus the number of time-domain samples stored in temporary storage from a modified frame at a previous timescale is an integer multiple of the number of time-domain samples inserted in the second processing to perform a temporal increment during operation in the special timescale modification mode.

73. Audio processor to provide a representation of a processed audio signal based on a representation of an input audio signal, the audio processor being characterized by being configured to perform a plurality of processing steps in order to provide the representation of the processed audio signal based on the representation of the input audio signal;wherein the audio processor is configured to perform a time-scaling modification on one or more intermediate audio signals, which are provided by a first processing based on the input audio representation, in order to obtain one or more time-scaling modified intermediate audio signals, and wherein the audio processor is configured to perform a second processing, which follows the first processing and which provides the processed audio signal representation, based on the one or more time-scaling modified intermediate audio signals;where the audio processor is configured to interpolate in time between one or more processing parameters associated with a previous frame of one or more intermediate audio signals and one or more processing parameters associated with a current frame of one or more intermediate audio signals, in order to obtain processing parameters associated with different time instances of a current frame of one or more intermediate audio signals modified in time scale using an interpolation function that determines the contributions of the one or more processing parameters associated with the previous frame and the contributions of one or more processing parameters associated with the current frame in a linear combination of processing parameters;wherein, for a temporally shortened audio frame, the interpolation function comprises a greater slope for a portion of the current frame modified in the timescale that was obtained using a combination of overlapping and adding than for a portion of the current frame modified in the timescale that was not modified; and / or wherein, for a temporally lengthened audio frame, the interpolation function comprises a smaller slope for a portion of the current frame modified in the timescale that was obtained using a combination of overlapping and adding than for a portion of the current frame modified in the timescale that was not modified.

74. Audio processor for providing a representation of a processed audio signal based on an input audio signal representation, the audio processor being characterized by being configured to perform a plurality of processing steps in order to provide the representation of the processed audio signal based on the input audio signal representation; Petition 870250092384, dated 09 / 10 / 2025, pp. 64 / 72 25 / 32 wherein the audio processor is configured to perform a time-scaling modification on one or more intermediate audio signals, which are provided by a first processing based on the input audio representation, in order to obtain one or more time-scaling modified intermediate audio signals, and wherein the audio processor is configured to perform a second processing, which follows the first processing and which provides the processed audio signal representation.based on one or more time-scaled modified intermediate audio signals; wherein the audio processor is configured to interpolate in time between one or more processing parameters associated with a previous frame of the one or more intermediate audio signals and one or more processing parameters associated with a current frame of the one or more intermediate audio signals.in order to obtain processing parameters associated with different time instances of a current frame of one or more time-scaled intermediate audio signals using an interpolation function that determines the contributions of one or more processing parameters associated with the previous frame and the contributions of one or more processing parameters associated with the current frame in a linear combination of processing parameters; wherein the audio processor is configured to obtain interpolated processing parameters Pinter[n] associated with time instances with time index n according to Pinter[n]=ga,f[n] P + (1-ga,f)Pprev, where P is a processing parameter or a set of processing parameters associated with a current frame of one or more intermediate audio signals,where Pprev is a processing parameter or a set of processing parameters associated with a previous frame of one or more intermediate audio signals, where Pinter is a processing parameter or a set of processing parameters associated with a time instance n in a current frame of one or more time-scaled modified intermediate audio signals, Petition 870250092384, dated 09 / 10 / 2025, p. 65 / 72 26 / 32 where ga,f is an interpolation function. where the interpolation function ga,f is defined as 9a.f [n] = max (0, ga [n]), and where h· + 1 + (Lseg Lrem. da [n] = 2L seg , Lseg <n < Lf 2 , .·>2Lseg + 1 Lrem (n + 1)------7--------r-, 0 < n < Lseq ' 2Lses(_LSeS + 1) S where n is a time index, where Lrem=Lf-Lseg, where Lf is the length of a frame to be processed; where Lseg is the length of a portion in which an overlay and addition is applied by modifying the time scale., 75. Audio processor for providing a representation of a processed audio signal based on an input audio signal representation, the audio processor being characterized by being configured to perform a plurality of processing steps in order to provide the representation of the processed audio signal based on the input audio signal representation; wherein the audio processor is configured to perform a time-scaling modification on one or more intermediate audio signals, which are provided by a first processing based on the input audio representation, in order to obtain one or more time-scaling modified intermediate audio signals, and wherein the audio processor is configured to perform a second processing, which follows the first processing and which provides the representation of the processed audio signal, based on the one or more time-scaling modified intermediate audio signals;where the audio processor is configured to perform, as a second processing step, the processing of values ​​of one or more intermediate audio signals modified in time scale in accordance with Petition 870250092384, dated 09 / 10 / 2025, p. 66 / 72 27 / 32 y[η] = (gaj [n]M + (1 - ga,f [n])Mprev)x[n], 0 ; <n < lf ou de acordo com}'i<h="(ga,f" hmk + (1 - ga,f [n])mk,prev )xfc[n], 0 n lf, em que x[n] e xk[n] são valores conjuntos do um mais sinais áudio intermediários modificados escala tempo, m é parâmetro processamento conjunto parâmetros associados a quadro atual intermediários; mprev anterior intermediários, índice o comprimento ser processado, uma função interpolação.

76. Audio processor for providing a representation of a processed audio signal based on an input audio signal representation, the audio processor being characterized by being configured to perform a plurality of processing steps in order to provide the representation of the processed audio signal based on the input audio signal representation; wherein the audio processor is configured to perform a time-scaling modification on one or more intermediate audio signals, which are provided by a first processing based on the input audio representation, in order to obtain one or more time-scaling modified intermediate audio signals, and wherein the audio processor is configured to perform a second processing, which follows the first processing and which provides the representation of the processed audio signal, based on the one or more time-scaling modified intermediate audio signals;where the audio processor is configured to determine a mapping vector that defines an association between time intervals of one or more timescale-modified intermediate audio signals and subframe indices of one or more original intermediate audio signals in Petition 870250092384, dated 10 / 09 / 2025, pp. 67 / 72 28 / 32, depending on the timescale modification, such that the mapping describes which processing parameters are allocated with which time interval of the timescale-modified intermediate audio signals.

77. Audio processor for providing a representation of a processed audio signal based on an input audio signal representation, the audio processor being characterized by being configured to perform a plurality of processing steps in order to provide the representation of the processed audio signal based on the input audio signal representation; wherein the audio processor is configured to perform a time-scaling modification on one or more intermediate audio signals, which are provided by a first processing based on the input audio representation, in order to obtain one or more time-scaling modified intermediate audio signals, and wherein the audio processor is configured to perform a second processing, which follows the first processing and which provides the representation of the processed audio signal, based on the one or more time-scaling modified intermediate audio signals;where the audio processor is configured to determine a mapping vector that defines an association between time intervals of one or more timescale-modified intermediate audio signals and time intervals of one or more original intermediate audio signals depending on the timescale modification, such that the mapping vector describes which processing parameters are allocated to which time interval of the timescale-modified intermediate audio signals.

78. Audio processor to provide a representation of a processed audio signal based on a representation of an input audio signal, the audio processor being characterized by being configured to perform a plurality of processing steps in order to provide the representation of the processed audio signal based on the representation of the input audio signal;wherein the audio processor is configured to perform a time-scale modification on one or more intermediate audio signals, which are provided by a first processing based on the input audio representation, in order to obtain one or more time-scale modified intermediate audio signals, and wherein the audio processor is configured to perform a second processing, which follows the first processing and which provides the processed audio signal representation, based on the one or more time-scale modified intermediate audio signals;wherein the audio processor is configured to modify a grid of subframes of one or more timescaled modified intermediate audio signals, depending on a timescaled modification of a current frame, wherein the subframe lengths of one or more timescaled modified intermediate audio signals are left unchanged, and wherein the second processing is configured to perform a subframe-based operation.

79. Audio processor for providing a representation of a processed audio signal based on an input audio signal representation, the audio processor being characterized by being configured to perform a plurality of processing steps in order to provide the representation of the processed audio signal based on the input audio signal representation; wherein the audio processor is configured to perform a time-scaling modification on one or more intermediate audio signals, which are provided by a first processing based on the input audio representation, in order to obtain one or more time-scaling modified intermediate audio signals, and wherein the audio processor is configured to perform a second processing, which follows the first processing and which provides the representation of the processed audio signal, based on the one or more time-scaling modified intermediate audio signals;where the audio processor is configured to modify transient position information depending on the change in the time scale. Petition 870250092384, dated 09 / 10 / 2025, pages 69 / 72 30 / 32; 80. Audio processor for providing a representation of a processed audio signal based on an input audio signal representation, the audio processor being characterized by being configured to perform a plurality of processing steps in order to provide the representation of the processed audio signal based on the input audio signal representation; wherein the audio processor is configured to perform a time-scaling modification on one or more intermediate audio signals, which are provided by a first processing based on the input audio representation, in order to obtain one or more time-scaling modified intermediate audio signals, and wherein the audio processor is configured to perform a second processing, which follows the first processing and which provides the representation of the processed audio signal, based on the one or more time-scaling modified intermediate audio signals;where the audio processor is configured to determine a mapping vector that defines an association between time intervals of one or more time-scaled modified intermediate audio signals and subframe indices of one or more original intermediate audio signals depending on the time-scale modification, such that the mapping describes which processing parameters are allocated to which time interval of the time-scaled modified intermediate audio signals; where the audio processor is configured to determine the mapping vector according to r -i , , , (^ [(2^seg msf ín] = modulo(nOffSet + maxl 0,----1)ff a,map [n]h 7 Λ £ J , ^md ) or according to Γ 1 (n K2^seg - 1)9a,map msf [n] = max I 0,-----------------------I \ Lsf / where msf[n] is an element of the mapping vector; where n is a time interval index; where noffset is an offset value;Petition 870250092384, dated 09 / 10 / 2025, pp. 70 / 72 31 / 32 where Lseg is half the length of a frame of one or more original intermediate audio signals at time intervals; where 2Lseg is the length of a frame of one or more original intermediate audio signals at time intervals; where Lsf is the length of a subframe at time intervals; where Lmd is the length of a temporary metadata storage; where noffset is ; and where ga,map[n] is an interpolation function; where [ ] denotes rounding to the nearest integer; where modulo is a modulo operation, and where division is integer division.; 81. Audio processor for providing a representation of a processed audio signal based on a representation of an input audio signal, the audio processor being characterized by being configured to perform a plurality of processing steps in order to provide the representation of the processed audio signal based on the representation of the input audio signal; wherein the audio processor is configured to perform a time-scaling modification on one or more intermediate audio signals, which are provided by a first processing based on the input audio representation, in order to obtain one or more time-scaling modified intermediate audio signals, and wherein the audio processor is configured to perform a second processing, which follows the first processing and which provides the representation of the processed audio signal, based on the one or more time-scaling modified intermediate audio signals;wherein the audio processor is configured to determine a mapping vector that defines an association between time intervals of one or more timescale-modified intermediate audio signals and time intervals of one or more original intermediate audio signals depending on the timescale modification, Petition 870250092384, 09 / 10 / 2025, p. 71 / 72 32 / 32 such that the mapping vector describes which processing parameters are allocated with which timescale-modified intermediate audio signals. wherein the audio processor is configured to determine the mapping vector according to ™ts [A] max (0, [(2Lsep 1-)9a,map IA]1) where n is a time interval index, where 2Lseg is the length of a frame of one or more original intermediate audio signals in time intervals; and where ga,map[n] is an interpolation function. Petition 870250092384, dated 09 / 10 / 2025, page 72 / 72;< / n>

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