Method for generating a replacement frame for a lost audio frame from an audio signal

Description

1 / 28 “METHOD FOR GENERATING A REPLACEMENT FRAME FOR A LOST AUDIO FRAME FROM AN AUDIO SIGNAL” Separated from BR112022025235-8, filed on June 10, 2021. CROSS-REFERENCE TO RELATED ORDERS

[001] This application claims priority with respect to the following Priority Applications: U.S. Provisional Application 63 / 037,673 (reference: D20058USP1), filed on June 11, 2020, and U.S. Provisional Application 63 / 193,974 (reference: D20058USP2), filed on May 27, 2021, which are incorporated herein by reference. TECHNOLOGY

[002] The present disclosure relates generally to a method and apparatus for frame loss concealment for a low-frequency effects (LFE) channel. More specifically, the present disclosure relates to frame loss concealment based on linear predictive coding (LPC) for an LFE channel of a multichannel audio signal. The techniques presented can be applied, for example, to 3GPP IVAS coding.

[003] Although some modalities are described in this document with particular reference to this disclosure, it will be appreciated that this disclosure is not limited to such field of use and is applicable in broader contexts. BACKGROUND

[004] Any discussion of the preceding technique throughout the disclosure should in no way be considered as an admission that such art is widely known or forms part of the common general knowledge in the field.

[005] The LFE (low-frequency effects) channel is the low-frequency effects channel in multichannel audio, such as in 5.1 or 7.1 audio. This channel is intended to drive the subwoofer of loudspeaker playback systems for such multichannel audio. As the term LFE indicates, this channel should only provide... Petition 870260006730, dated 01 / 23 / 2026, page 9 / 78 2 / 28 bass information, a typical upper frequency limit is 120 Hz.

[006] However, this frequency limit may not always be very sharp, which means that in practice the LFE channel may contain even some higher frequency component, for example, up to 400 or 700 Hz. Whether such components will have a perceptible effect when rendered to the speaker system may depend on the actual frequency characteristics of the subwoofer.

[007] In some cases, multichannel audio can also be reproduced through stereo headphones. Specific rendering techniques are employed to generate an equivalent sound experience, in this case, as if the multichannel audio were heard on a multi-speaker system. This is even the case for the LFE channel, where appropriate rendering techniques ensure that the sound experience of the LFE channel is as close as possible to the experience if a subwoofer system had been used for playback.

[008] Given that the LFE channel typically has only a very limited frequency content, it can be encoded and transmitted at a relatively low bit rate. A suitable encoding technique for LFE is transform-based encoding using a modified discrete cosine transform (MDCT). With this technique, for example, it is possible to represent LFE at bit rates of around 2,000 to 4,000 bits per second.

[009] A particular situation in multichannel audio transmissions, especially in wireless channels, is that the transmission may be subject to errors. Transmission is typically based on packets, and a transmission error can result in the loss of one or more complete encoded frames of the multichannel audio. There are so-called packet or frame loss concealment techniques employed by a multichannel audio decoding system that has the Petition 870260006730, dated 01 / 23 / 2026, page 10 / 78 3 / 28 objective to make the effects of lost audio frames less noticeable.

[010] For regular multichannel audio signal channels, there are well-established frame loss concealment techniques. A variety of suitable techniques are, for example, part of the 3GPP EVS codec [3 GPP TS 26.447].

[011] For the MDCT-encoded LFE channel, in principle, the same techniques can be applied. For example, it would be possible to reuse the MDCT coefficients from the most recent valid audio frame and use these coefficients after gain scaling (attenuation) and signal prediction or randomization. The EVS standard also offers other techniques, such as a technique that reconstructs the missing audio frame in the time domain, according to a sinusoidal approach.

[012] A major problem with applying these cutting-edge techniques to the LFE channel is that they are not designed or optimized for very low frequency content. While they are very powerful for audio channels with regular frequency content, applying them to the LFE channel results in irritating low-frequency noise.

[013] Therefore, it is an objective of this disclosure to describe a new technique that overcomes the problems and limitations of the frame loss concealment techniques of the preceding technique applied to the LFE channel. The range of application of the new method may, however, not be limited to LFE channels. SUMMARY

[014] According to a first aspect of the present disclosure, a method is presented for generating a replacement frame for a missing audio frame from an audio signal. The method may comprise determining an audio filter based on samples from a valid audio frame preceding the missing audio frame. The method may comprise generating the replacement frame based on the audio filter and samples from the valid audio frame. Petition 870260006730, dated 01 / 23 / 2026, page 11 / 78 4 / 28 preceding the lost audio frame. The step of generating the replacement frame based on the audio filter and samples from the valid audio frame may include initializing a filter memory of the audio filter with samples from the valid audio frame. The method may comprise determining a modified audio filter based on the audio filter. The modified audio filter may replace the audio filter, and the step of generating the replacement frame based on the audio filter may include generating the replacement frame based on the modified audio filter and samples from the valid audio frame.

[015] The audio filter can be a multipolar filter. The audio filter can be a linear predictive coding (LPC) synthesis filter. The audio filter can be derived from an all-pass filter operated on at least one sample of a valid frame. The method can comprise determining the audio filter based on a denominator polynomial of a transfer function of the all-pass filter.

[016] The step in determining the modified audio filter may include bandwidth sharpening. Bandwidth sharpening may be applied so that the duration of an impulse response of the modified audio filter is extended, relative to the duration of an impulse response of the audio filter. Bandwidth sharpening may be applied so that a distance between a pole of the modified audio filter and the unit circle is reduced compared to a distance between a corresponding pole of the audio filter and the unit circle. Bandwidth sharpening may be applied so that a pole of the modified audio filter with the highest magnitude is equal to 1 or at least close to 1. Bandwidth sharpening may be applied so that a frequency of a pole of the modified audio filter with the highest magnitude is equal to a frequency of a pole of the audio filter with the highest magnitude. Petition 870260006730, dated 01 / 23 / 2026, page 12 / 78 5 / 28

[017] The method may comprise determining the magnitudes and frequencies of the audio filter poles using a root-finding method. Bandwidth sharpness may be applied such that the magnitudes of the modified audio filter poles are equal to or at least close to 1, wherein the frequencies of the modified audio filter poles are identical to the frequencies of the audio filter poles. A magnitude of a modified audio filter pole may be set equal to or at least close to 1 only if a magnitude of the corresponding audio filter pole has a magnitude that exceeds a certain threshold value.

[018] The method may comprise determining audio filter coefficients. The method may comprise applying bandwidth sharpening using a bandwidth sharpening factor such that Sy(z) = S(z / y), where Sy indicates a transfer function of the modified audio filter, S indicates a transfer function of the audio filter, and y indicates the bandwidth sharpening factor. The method may comprise generating the replacement frame based on the audio filter coefficients, the samples of the valid audio frame preceding the lost audio frame, and the bandwidth sharpening factor y. The bandwidth sharpening factor may be determined in an iterative procedure by step-by-step increasing and / or decreasing the bandwidth sharpening factor.The method can involve checking if a pole of the modified audio filter is inside the unit circle by converting polynomial coefficients of the modified audio filter into reflection coefficients. This conversion of the polynomial coefficients of the modified audio filter into reflection coefficients can be based on Levinson's inverse recursion. The bandwidth sharpness factor r can be determined so that a pole of the modified audio filter with the highest magnitude is moved as close as possible to the unit circle while simultaneously locating all poles of the modified audio filter. Petition 870260006730, dated 01 / 23 / 2026, page 13 / 78 6 / 28 within the unit circle. The replacement frame can be generated using the equation X(n) = ΣΓ=ι a1· Y1X(n - i), n > 0, where ai indicates the audio filter coefficients, P indicates the audio filter order, y indicates the bandwidth sharpness factor, X(-1... - P) indicates the audio filter memory, and X(n), n > 0 indicates replacement samples of the replacement frame.

[019] The method may comprise determining the audio filter coefficients by applying bandwidth sharpening by reducing the distance of a pair of line spectral frequencies representing the audio filter coefficients, thereby generating modified line spectral frequencies. The method may comprise deriving the modified audio filter coefficients from the modified line spectral frequencies. The method may comprise generating the replacement frame based on the modified audio filter coefficients and samples of the valid audio frame preceding the lost audio frame.

[020] The lost audio packet may be associated with a low-frequency effect LFE channel of a multichannel audio signal. In particular, the lost audio packet may have been transmitted over a wireless channel from a transmitter to a receiver. The method may be performed at the receiver.

[021] The method may comprise reducing the sampling of the valid audio frame samples before generating replacement samples of the replacement frame. The method may comprise increasing the sampling of the replacement frame samples after generating the replacement frame.

[022] A plurality of audio frames may be lost and the method may comprise determining a first modified audio filter by scaling the audio filter coefficients of the audio filter using a first bandwidth sharpness factor. The method may comprise determining a second modified audio filter by scaling said audio filter coefficients using a second bandwidth sharpness factor. The method Petition 870260006730, dated 01 / 23 / 2026, page 14 / 78 7 / 28 may involve generating replacement frames based on the first modified audio filter for the first M lost audio frames. The method may involve generating replacement frames based on the second modified audio filter for the (M+1)th lost audio frame and all subsequent lost audio frames, so that the audio signal is damped for the last frames.

[023] The method may comprise splitting the audio signal into a first sub-band signal and a second sub-band signal. The method may comprise generating a first sub-band audio filter for the first sub-band signal. The method may comprise generating first sub-band replacement frames based on the first sub-band audio filter. The method may comprise generating a second audio filter for the second sub-band signal. The method may comprise generating second sub-band replacement frames based on the second sub-band audio filter. The method may comprise generating the replacement frame by combining the first and second sub-band replacement frames.

[024] The audio filter can be configured to operate as a resonator. The resonator can be tuned to samples of the valid audio frame preceding the missing audio frame. The resonator can be initially excited with at least one sample among the samples of the valid audio frame preceding the missing audio frame. The replacement frame can be generated by using the resonator's tuning to extend at least one sample into the missing audio frame.

[025] According to a second aspect of the present disclosure, a system is presented. The system may comprise one or more processors and a non-transient, computer-readable medium that stores instructions which, when executed by one or more processors, cause one or more Petition 870260006730, dated 01 / 23 / 2026, page 15 / 78 8 / 28 processors execute operations of the method described above.

[026] According to a third aspect of the present disclosure, a non-transient, computer-readable medium is presented. Said non-transient, computer-readable medium can store instructions that, when executed by one or more processors, cause one or more processors to perform operations of the method described above. BRIEF DESCRIPTION OF THE DRAWINGS

[027] Exemplary forms of revelation will now be described, by way of example only, with reference to the attached drawings in which: Figure 1 illustrates a flowchart of an example frame loss hiding process, and Figure 2 illustrates an exemplary mobile device architecture for implementing the features and processes described in this document. DESCRIPTION OF EXEMPLARY MODALITIES

[028] A key idea of ​​this revelation is to extrapolate the missing audio frame samples from the most recent valid audio samples by operating a resonator. The resonator is tuned to the most recent valid audio samples and is then operated to extend the audio samples into the missing audio frame. For example, if the most recent valid audio samples are a sine wave of frequency f° and phase φ, then a suitable resonator would be an oscillator tuned to extend that sine wave into the missing audio frame.

[029] In this example, the most recent valid signal can be expressed as: fx(n) = a sin(2rc — η + φ~), n < 0. fs

[030] The extrapolated samples generated by the resonator would then be: fx(n) = a sin(2rc — η + φ~), n > 0. fs

[031] In these equations, α is the sinusoidal amplitude, fs is the sampling frequency. Petition 870260006730, dated 01 / 23 / 2026, page 16 / 78 9 / 28

[032] One possible embodiment of this resonator is the following pass filter: 1-ζ-1·2 εοδΣπ^+ζ-2 H (z) =----------£---. 1-ζ-1·2 cos2n^-+z~2j s

[033] Since the numerator and denominator of this filter are identical, the resulting transfer function would be one, and therefore the filter would pass through the most recent valid audio samples without modification. However, to generate the extrapolated samples, only the denominator of the filter would be used, transforming it into an oscillator. The extrapolated samples would then be generated as follows: x(n) = —2 cos 2π— x(n — 1) + x(n — 2), n > 0. fs

[034] The initial values ​​for x(-i) ex(-2) would be the two most recent valid samples x(— 1) ex(— 2).

[035] In other words, the extrapolated samples can be constructed as the touch of the resonator filter that was originally excited with the most recent audio samples, which thus determine the state memories of the initial filter, and then letting the filter touch (or oscillate) by itself, i.e., without other (non-zero) input samples.

[036] The sample extrapolation approach described would be possible if the signal could be sufficiently well approximated with a sinusoid. However, this would still require identifying the sinusoidal frequency fo and the resonance frequency of the resonator.

[037] A more generic approach that overcomes the limitation of a single sinusoid and also solves the problem of determining the resonator's resonance frequencies is to apply a linear predictive filter (LPC) approach. The linear predictive synthesis filter has traditionally been employed in frame-based Analysis by Synthesis speech coding systems. In the present document, the LPC filter excitation of a current frame is calculated by taking Petition 870260006730, dated 01 / 23 / 2026, page 17 / 78 10 / 28 considering the synthesis filter touch of the preceding frame. The LPC synthesis filter touch was also used to extrapolate some samples in the case of switching the ACELP codec mode, where some future samples are not available [3 GPP TS 26.445],

[038] Just as the filter passes everything above, an H(z) filter is constructed as: W = Λ(ζ)

[039] In this document, A(z) is the LPC analysis filter that generates the linear predictive error signal. In this exemplary formulation of H(z), A(z) is a transversal filter. — is the LPC synthesis filter reconstructing the speech signal A(z) from the predictive error signal 4(z) or some other suitable excitation signal. is a recursive filter (multipolar filter), is a scaling factor of the excitation signal to be chosen so that the power of the synthesized signal matches the power of the original signal, can be optional and / or set to 1 in some implementations.

[040] The approach to extrapolate signal samples is similar to the oscillator case described above: x(n) = Zf=iai %(η— i),n> 0.

[041] The initial values ​​for *(-i) up to x(-P) are the most recent valid samples x(— 1) up to x(— P). P is the order of the LPC synthesis filter.

[042] Notably, the analysis filter A(z) can be generated / determined with conventional approaches, such as the Levinson-Durbin approach. The all-pass filter H(z) can be constructed from A(z) as described above. In case of frame loss, the synthesis filter part of H(z), i.e., the LPC synthesis filter S(z) = can be used to construct the replacement frame for ϊ4(ζ) the lost frame.

[043] It is also noteworthy that the LPC approach solves the problem of determining the resonator's resonance frequencies, as explained in Petition 870260006730, dated 23 / 01 / 2026, p. 18 / 78 11 / 28 following: A property of LPC analysis, well known from speech coding, is that the frequency response of the corresponding LPC synthesis filter is consistent with the speech structure. In general, this means that the synthesis filter matches the resonance frequencies of the dominant spectral components (dominant frequencies) of the analyzed input signal. Therefore, the LPC approach is suitable for determining a resonator with corresponding resonance frequencies.

[044] One disadvantage with the touch approach of the LPC synthesis filter is that the impulse response of the LPC synthesis filter is typically quite fast (approximately and exponentially) decaying. The approach would therefore not be sufficient to generate a replacement frame for a lost 20 ms audio frame. In the case of several successive lost frames, correspondingly, multiples of 20 ms of replacement signal would need to be generated. A typical LPC synthesis filter would have already faded and would not be able to produce a useful replacement signal.

[045] To overcome this limitation, the LPC synthesis filter cannot be employed as such and calculated using standard techniques, such as the Levinson / Durbin approach. Instead, through bandwidth sharpening, the filter is modified so that its poles are moved as close as possible to the unit circle, while maintaining stability. According to one of these approaches, the poles of the LPC synthesis filter are calculated using a standard root-finding method. Then, given an original pole location z, = η · βιω<, the magnitude of the pole η is replaced by a magnitude of 1, or at least close to 1. The effect of this operation is that the frequency of the pole is maintained as long as the filter response for the frequency of that pole fi = fs·^ does not gradually decrease. A slight modification of the method is that only the poles whose magnitude exceeds a certain value are moved towards the unit circle. Petition 870260006730, dated 01 / 23 / 2026, page 19 / 78 12 / 28 limit of, for example, 0.75.

[046] A practical disadvantage of the described method may, in some implementations, be the numerical complexity required for root discovery. A method that avoids this processing step is to take the provided LPC synthesis filter and modify it by a bandwidth sharpness factor γ, as follows: SY(z) = S(Z / y).

[047] This operation has the effect that the posterior poles are all moved by the factor γ towards the unit circle. However, since the locations of the poles are unknown, a given factor γ can be very large, so that at least the pole with the greatest magnitude is moved outside the unit circle, resulting in a still unstable filter. Thus, it is possible, after applying a given factor γ, to check whether the filter has become unstable or is still stable. If the filter is unstable, a smaller γ is chosen; otherwise, a larger γ. This procedure can then be repeated iteratively (using nested interval techniques) until a bandwidth-boosting factor γ is found for which the filter is very close to instability, but still stable.

[048] Notably, other bandwidth filter sharpening techniques can also be employed, such as sharpening based on line spectral frequency. In this technique, the LPC filter coefficients are represented as line spectral frequencies (pairs). The sharpening effect is obtained by reducing the distance between line spectral frequency pairs. If the distance is reduced to zero, this is identical to moving the filter poles to the unit circle or pushing the filter to the stability limit. The correspondingly modified filter, represented by the modified line spectral frequencies, can then be represented again by LPC coefficients which are obtained by a conversion. Petition 870260006730, dated 01 / 23 / 2026, page 20 / 78 13 / 28 inverse of the modified line spectral frequencies for modified LPC coefficients.

[049] The LPC-based approach above can be summarized as follows: In a first step, an audio filter (which can be viewed as a resonator) can be tuned to a previously received and / or reconstructed audio signal (such as, for example, an LFE audio signal). For example, the LPC coefficients at a,, i = 1 ... / ', can be calculated. The tuning to the previously received and / or reconstructed signal can be performed in such a way that the audio filter obtained in this step has characteristics (e.g., resonance frequencies) based on (e.g., derivatives) those of the previously received and / or reconstructed signal.

[050] The sharpness of the bandwidth of the corresponding LPC synthesis filter can be achieved using a modified synthesis filter Scrit(z) = sÇz / Ycrit) where ycrit is chosen so that the LPC filter is at the stability limit. Alternatively, sharpness based on the line spectral frequency can be employed. The memories of the LPC synthesis filter can be initialized with the most recent samples of the previously received and / or reconstructed audio signal: X(-1...-P) = x(-1...-P). The replacement signal for a lost frame can then be determined based on the following formula: X(n) = EiLia, •YlcritX(n - i),n > 0. In other words, the resonator's touch can be used to reconstruct or estimate the replacement signal.

[051] The stability check of the filter in the above procedure can be done by converting the polynomial coefficients of the modified LPC synthesis filter into reflection coefficients. This can be done using an inverse Levinson feature. The reflection coefficients allow a direct stability test: if any of the absolute values ​​of the reflection coefficients is greater than or equal to 1, the filter is unstable, otherwise it is guaranteed to be stable. Petition 870260006730, dated 01 / 23 / 2026, page 21 / 78 14 / 28

[052] For implementation reasons, it may be advantageous to perform the operations described above in the subsampled domain. Under the assumption that the LFE signal does not have significant frequency content above 800 Hz, it is possible to perform the frame loss hiding operations described in the subsampled domain, for example, using a sampling frequency of fs = 1600 Hz instead of an original sampling frequency of 48,000 Hz. This allows, for example, reducing the memory required to store the preceding valid samples by a corresponding factor of 1 / 30 = 1600^z / 4800^z. The complexity of certain numerical operations is reduced by the same factor. Under the assumption that the LFE signal is sufficiently band-limited, no additional filtering before subsampling is required.However, during sampling to the original sampling frequency, after calculating the replacement samples, corresponding interpolation filtering is necessary, typically applying a linear phase low-pass filter. The delay induced by the filter can be considered, and a corresponding additional number of replacement samples must be calculated.

[053] It is noteworthy that an LPC filter order of P=20 was considered adequate in a practical implementation, operated in an undersampled domain with a sampling frequency fs = 1600 Hz.

[054] Another factor to be taken into consideration in frame loss concealment of MDCT-based encoding is that the frame to be recovered may need to be prepared to match the particular realization of that MDCT transform (overlay). This means that replacement samples, after applying the frame loss concealment technique described above, may be windowed and then converted to time-folded domain. The time-folded domain conversion may then be inverted, the resulting signal frame then subjected to the inverted time window. Note that time folding and unfolding Petition 870260006730, dated 01 / 23 / 2026, page 22 / 78 15 / 28 can be combined in one step. After these operations, the recovered frame can be combined with the remainder of the preceding (valid) frame to produce replacement samples for the deleted frame. Depending on the MDCT frame size and window format and the interpolation filter mentioned, this may require the reconstruction of more samples with the described method than would be expected by the nominal step or frame size of the encoding system, which could, for example, be 20 ms.

[055] A specific case is when several consecutive frames are lost in sequence. In principle, the processing described above remains unchanged if the frame loss is the second, third, etc., consecutive loss in the sequence. The preceding frame recovered by the described technique can be considered as if it were a valid frame received without errors. Or, the ringing can be extended to the next lost frame, whereby the resonator or synthesis filter parameters (modified) are maintained from the initial calculation for the first frame loss. However, after very long bursts of frame losses (e.g., more than 10 consecutive frames corresponding to 200 ms), it is advantageous for a listener to start muting the replacement signal. Otherwise, the listener may become confused by a seemingly endless replacement signal despite the interrupted connection.

[056] A particularly inventive method suitable for silencing is to modify the bandwidth sharpness factor γ found according to the steps described above. Although the factor γ found ensures that the modified synthesis filter S(z / y) produces a sustained replacement signal, for silencing, γ is further modified (scaled) to ensure adequate attenuation. This has the effect that the poles of the modified synthesis filter are moved by the scaling factor into the circled unity and, consequently, the response of the synthesis filter decays exponentially. Petition 870260006730, dated 01 / 23 / 2026, page 23 / 78 16 / 28

[057] For example, if an attenuation (att_per_frame) of 3dB per 20ms frame is desired (flen = 0.02s), and assuming that the synthesis filter operates at a sampling frequency of fs = 1600Hz, the following scaling factor would be applied: amute=flen•fs^10-att_Per-frame / 20= °'02^10-3^ / 20.

[058] The resulting yimuth factor is the original γ scaled with an amute, as follows: Ymute = Y · ^mute

[059] It should be noted that, generally, muting should only be initiated after a very long burst of frame drops, for example, after 10 consecutive frame drops. That is, only then would γ be replaced by Ymute·

[060] The preceding embodiments of the invention are based on the assumption that the signal for which frame loss concealment is to be performed is the LFE channel of a multichannel audio signal. However, analogous principles can be applied to any audio signal without bandwidth limitations. An obvious possibility is to perform the operations in full band, at the nominal sampling frequency of the signal. However, this can result in practical difficulties, especially using the LPC approach. If the sampling frequency is 48 kHz, it may be difficult to find a sufficiently high-order LPC filter that can adequately represent the spectral properties of the signal to be extended. The challenges can be numerical (to calculate a sufficiently high-order LPC filter) and conceptual. The conceptual difficulty may be that low frequencies may require a longer LPC analysis window than higher frequencies.

[061] An effective way to address these challenges is to perform the operations described in a sub-band / split-band approach. To this end, the initial full-band signal is split by a bank of analysis filters. Petition 870260006730, dated 01 / 23 / 2026, page 24 / 78 17 / 28 in several sub-band signals, each representing a partial frequency band. The split-band approach can be combined with the use of quadrature mirror filtering and subsampling (QMF approach), which offers advantages in terms of complexity and memory economy (due to critical sampling). After analyzing the operation of the filter that produces the sub-band signals, the frame loss concealment techniques described above can be applied to all sub-band signals in parallel. With this approach, it is especially possible to use a wider LPC analysis window for low-frequency bands than for high-frequency bands, thus making the frequency selective of the LPC approach.

[062] After frame loss concealment operations for the initial sub-bands, the sub-bands can be combined again for a full-band replacement signal. In the case of QMF, QMF synthesis also involves upsampling and QMF interpolation filtering. Interpretation

[063] Unless specifically stated otherwise, as will appear in the discussions that follow, it is understood that, throughout the disclosure discussions, terms such as processing, computation, calculation, determination, analysis or the like are used to refer to the action and / or processes of a computer or computing system, or similar electronic computing devices, that manipulate and / or transform data represented as physical quantities, such as electronic ones, into other data similarly represented as physical quantities.

[064] Similarly, the term “processor” can refer to any device or part of a device that processes electronic data, for example, from registers and / or memory to transform that electronic data into other electronic data that can, for example, be stored in Petition 870260006730, dated 01 / 23 / 2026, page 25 / 78 18 / 28 registers and / or memory. A "computer" or a "computing machine" or a "computing platform" may include one or more processors.

[065] The methodologies described in this document are, in an exemplary embodiment, executable by one or more processors that accept computer-readable (also called machine-readable) code containing a set of instructions that, when executed by one or more of the processors, perform at least one of the methods described in this document. Any processor capable of executing a set of instructions (sequential or not) that specify the actions to be performed is included. Thus, an example is a typical processing system that includes one or more processors. Each processor may include one or more CPUs, a graphics processing unit, and a programmable DSP unit. The processing system may also include a memory subsystem including main RAM and / or static RAM and / or ROM. A bus subsystem may be included for communication between components.The processing system may also be a distributed processing system with processors coupled over a network. If the processing system requires a display, such a display may include, for example, a liquid crystal display (LCD) or a cathode ray tube (CRT) display. If manual data input is required, the processing system also includes an input device, such as one or more alphanumeric input units, such as a keyboard, a pointing control device, such as a mouse, and so on. The processing system may also encompass a storage system, such as a hard disk drive. The processing system in some configurations may include a sound output device and a network interface device. The memory subsystem therefore includes a computer-readable transport medium that carries computer-readable code (e.g., software) including an instruction set. Petition 870260006730, dated 01 / 23 / 2026, page 26 / 78 19 / 28 to cause the execution, when executed by one or more processors, of one or more of the methods described in this document. Note that when the method includes multiple elements, for example, multiple steps, no order of such elements is implied unless specifically indicated. Software may reside on the hard disk, or it may also reside, wholly or at least partially, within RAM and / or within the processor during its execution by the computer system. Thus, memory and the processor also constitute a computer-readable transport medium containing computer-readable code. Furthermore, a computer-readable transport medium may form or be included in a computer program product.

[066] In exemplary alternative embodiments, one or more processors operate as a standalone device or may be connected, for example, in a network to another processor(s), in a network deployment, one or more processors may operate in the capacity of a server or a user machine in a server-user network environment or as a peer machine in a peer-to-peer or distributed network environment. The one or more processors may form a personal computer (PC), a tablet, a Personal Digital Assistant (PDA), a mobile phone, a web device, a network router, switch or bridge, or any machine capable of executing a set of instructions (sequential or not) that specify the actions to be performed by that machine.

[067] Note that the term machine should also be considered as including any collection of machines that individually or collectively execute a set (or multiple sets) of instructions to perform any one or more of the methodologies discussed in this document.

[068] Thus, an example of an embodiment of each of the methods described in this document is in the form of a readable transport medium. Petition 870260006730, dated 23 / 01 / 2026, p. 27 / 78 20 / 28 A computer that carries a set of instructions, for example, a computer program intended for execution on one or more processors, for example, one or more processors that are part of a web server arrangement. Thus, as will be appreciated by those skilled in the art, exemplary embodiments of the present disclosure may be embodied as a method, an apparatus such as a special-purpose apparatus, an apparatus such as a data processing system, or a computer-readable transport medium, for example, a computer program product. The computer-readable transport medium carries computer-readable code, including a set of instructions that, when executed on one or more processors, cause the processor or processors to implement a method.Consequently, aspects of the present disclosure may take the form of a method, an exemplary embodiment consisting entirely of hardware, an exemplary embodiment consisting entirely of software, or an exemplary embodiment combining aspects of software and hardware. Furthermore, the present disclosure may take the form of a carrier medium (for example, a computer program product on a computer-readable storage medium) carrying computer-readable program code embedded in the medium.

[069] Software can also be transmitted or received over a network via a network interface device. Although the carrier medium is, in an exemplary embodiment, a single medium, the term carrier medium should be understood as including a single medium or multiple media (e.g., a centralized or distributed database and / or associated caches and servers) that store one or more instruction sets. The term carrier medium should also be understood as including any medium capable of storing, encoding, or transporting an instruction set for execution by one or more processors and that causes one or more Petition 870260006730, dated 01 / 23 / 2026, page 28 / 78 21 / 28 processors execute any one or more of the methodologies of the present disclosure. A carrier medium can take various forms, including but not limited to non-volatile media, volatile media, and transmission media. Non-volatile media includes, for example, optical, magnetic, and magneto-optical disks. Volatile media includes dynamic memory, such as main memory. Transmission media include coaxial cables, copper wires, and optical fiber, including the wires that make up a bus subsystem. Transmission media can also take the form of acoustic or light waves, such as those generated during data communications by radio waves and infrared.For example, the term "carrier medium" should be understood as including, but not limited to, solid-state memories; a computer product embedded in optical and magnetic media; a medium containing a propagated signal detectable by at least one processor or one or more processors and representing a set of instructions that, when executed, implement a method; and a transmission medium in a network containing a propagated signal detectable by at least one processor or one or more processors and representing the instruction set.

[070] It will be understood that the steps of the methods discussed are performed in a modality example by an appropriate processor (or processors) of a processing system (e.g., computer) executing instructions (computer-readable code) stored in storage. It will also be understood that the disclosure is not limited to any particular implementation or programming technique and that the disclosure can be implemented using any techniques appropriate to implement the functionality described herein. The disclosure is not limited to any specific programming language or operating system.

[071] The reference throughout this revelation to an exemplary modality, some exemplary modalities” means that a particular resource, framework Petition 870260006730, dated 01 / 23 / 2026, page 29 / 78 22 / 28 or a feature described in connection with the example of a modality is included in at least one example of a modality of the present disclosure. Thus, when the phrases “in an exemplary modality,” “in some exemplary modalities,” or “in an exemplary modality” are used in several places throughout this disclosure, they do not necessarily refer to the same exemplary modality. Furthermore, particular features, structures, or characteristics may be combined in any suitable manner, as would be evident to one skilled in the art from this disclosure, in one or more exemplary modalities.

[072] As employed in this document, unless otherwise specified, the use of the ordinal adjectives “first”, “second”, “third”, etc. to describe a common object merely indicates that different forms of similar objects are being referred to and does not mean that the objects thus described must be in a given sequence, whether temporal, spatial, classification or otherwise.

[073] In the claims below and in the description in this document, any of the terms comprising, included or comprising is an open term meaning to include at least the following elements / characteristics, but not excluding others. Thus, the term comprising, when used in the claims, should not be interpreted as limiting the means or elements or steps listed below. For example, the scope of the expression a device comprising A and B should not be limited to devices consisting only of elements A and B. Any of the terms including or comprising as used in this document is also an open term meaning to include at least the following elements / characteristics, but not excluding others. Thus, include is synonymous with and means to comprise.

[074] It should be appreciated that in the above description of modalities Petition 870260006730, dated 01 / 23 / 2026, page 30 / 78 23 / 28 exemplary embodiments of the disclosure, several features of the disclosure are sometimes grouped into a single exemplary embodiment, Figure, or description thereof for the purpose of simplifying the disclosure and aiding in the understanding of one or more of the various inventive aspects. This method of disclosure, however, should not be interpreted as reflecting an intention that the claims require more features than are expressly cited in each claim. Instead, as the following claims reflect, the inventive aspects reside in fewer than all the features of a single exemplary embodiment disclosed previously. Thus, the claims following the Description are expressly incorporated into this Description, with each independent claim as a separate exemplary embodiment of this disclosure.

[075] Furthermore, although some exemplary embodiments described herein include some, but not others, features included in other exemplary embodiments, combinations of features from different exemplary embodiments must be within the scope of disclosure and form different exemplary embodiments, as would be understood by those skilled in the art. For example, in the following claims, any of the claimed exemplary embodiments may be employed in any combination.

[076] In the description provided in this document, several specific details are presented. However, it is understood that exemplary modalities of revelation can be practiced without these specific details. In other cases, well-known methods, structures, and techniques have not been shown in detail so as not to obscure the understanding of this description.

[077] Thus, although what are believed to be the best modes of revelation have been described, those versed in the technique will recognize that other and further modifications may be made without departing from the spirit of revelation, and it is intended to claim all changes and modifications that fall within the Petition 870260006730, dated 01 / 23 / 2026, p. 31 / 78 24 / 28 scope of disclosure. For example, any formulas given above are merely representative of procedures that may be employed. Functionality may be added to or removed from the block diagrams, and operations may be swapped between functional blocks. Steps may be added to or removed from the methods described within the scope of this disclosure.

[078] Finally, Figure 1 illustrates a flowchart of an example of a frame loss hiding process. This example process can be performed, for example, by an 800 mobile device architecture represented in Figure 2. The 800 architecture can be implemented in any electronic device, including, but not limited to: a desktop computer, client audiovisual (AV) equipment, radio transmission equipment, mobile devices (e.g., smartphone, tablet, laptop, wearable device).In the exemplary embodiment shown, the 800 architecture is intended for a smartphone and includes processor(s) 801, peripheral interface 802, audio subsystem 803, speakers 804, microphone 805, sensors 806 (e.g., accelerometers, gyroscopes, barometer, magnetometer, camera), location processor 807 (e.g., GNSS receiver), wireless communication subsystems 808 (e.g., Wi-Fi, Bluetooth, cellular), and I / O subsystem(s) 809, which includes touch controller 810 and other input controllers 811, touch surface 812, and other input / control devices 813. Other architectures with more or fewer components may also be employed to implement the embodiments disclosed.

[079] The memory interface 814 is coupled to the processors 801, peripheral interface 802 and memory 815 (e.g., flash, RAM, ROM). Memory 815 stores instructions and computer program data, including, but not limited to: operating system instructions 816, communication instructions 817, GUI instructions 818, sensor processing instructions 819, Petition 870260006730, dated 01 / 23 / 2026, page 32 / 78 25 / 28 telephone instructions 820, electronic messaging instructions 821, web browsing instructions 822, audio processing instructions 823, GNS S / navigation instructions 824 and applications / data 825. The audio processing instructions 823 include instructions for performing the audio processing described in reference to Figure 1.

[080] Aspects of the systems described in this document can be implemented in a computer-based sound processing network environment, suitable for processing digital or digitized audio files. Parts of the adaptive audio system may include one or more networks comprising any desired number of individual machines, including one or more routers (not shown) that serve to buffer and route the data transmitted between the computers. This network may be built on several different network protocols and may be the Internet, a Wide Area Network (WAN), a Local Area Network (LAN), or any combination thereof.

[081] One or more of the components, blocks, processes, or other functional components may be implemented by means of a computer program that controls the execution of a processor-based computing device of the system. It should also be noted that the various functions disclosed herein may be described using any number of combinations of hardware, firmware, and / or as data and / or instructions embedded in various machine-readable or computer-readable media, in terms of their behavior, register transfer, logical component, and / or other characteristics. Computer-readable media in which such formatted data and / or instructions may be embedded include, but are not limited to, non-volatile physical (non-transient) storage media in various forms, such as optical, magnetic, or semiconductor storage media. Petition 870260006730, dated 01 / 23 / 2026, page 33 / 78 26 / 28

[082] Although one or more implementations have been described as examples and in terms of specific embodiments, it should be understood that one or more implementations are not limited to the embodiments disclosed. On the contrary, it is intended to encompass various similar modifications and arrangements, as would be evident to those skilled in the art. Therefore, the scope of the appended claims should be given the broadest interpretation so as to encompass all such similar modifications and arrangements. Listed Exemplary Modalities

[083] Various aspects and implementations of the present invention can also be appreciated from the following enumerated exemplary embodiments (EEEs), which are not claims.

[084] EEE1. A method for recovering a lost audio frame, comprising: to tune a resonator to samples of a valid audio frame preceding the lost audio frame; Adapt the resonator to operate as an oscillator according to the samples of the valid audio frame; and extend an audio signal generated by the oscillator to the lost audio frame. The resonator may correspond to the audio filter H(z) described above, while the oscillator may correspond to the term described above S(z) = -^ .

[085] EEE2. The EEE 1 method, where the resonator / oscillator combination is constructed using linear predictive techniques (LPC) and where the oscillator is implemented as an LPC synthesis filter.

[086] EEE3. The EEE 2 method, in which the LPC synthesis filter is modified using bandwidth sharpness.

[087] EEE4. The EEE 3 method, in which the LPC synthesis filter is modified using a bandwidth sharpening factor γ, resulting in the following Petition 870260006730, dated 01 / 23 / 2026, page 34 / 78 27 / 28 modified filter: SY(z) = 5{Z / γ}.

[088] EEE5. The EEE 4 method, in which the bandwidth sharpness factor γ is selected, so that the modified LPC synthesis filter is close to instability, but still stable.

[089] EEE6. The method of any of the EEE 1-5, wherein the method is operated in the subsampled domain.

[090] EEE7. A method for recovering a frame from a sequence of consecutive audio frame losses, comprising: Apply a first modified LPC synthesis filter using a sharpness factor γ to an nth consecutive frame loss, where n is below a threshold M; and gradually silence further frame losses in the sequence using a second modified LPC synthesis filter using an additional modified sharpness factor Ymute to a kth consecutive frame loss, where k is above or equal to the threshold M, and where Ymute is the sharpness factor γ scaled by an amute factor.

[091] EEE8. The EEE 7 method, in which the M limit and the amute scaling factor are chosen so that a muting behavior is achieved with an attenuation of 3dB per 20 ms audio frame, starting on the tenth consecutive frame loss.

[092] EEE9. The method of any of the EEE 1-8, wherein the method is applied to the low frequency effect (LFE) channel of a multichannel audio signal.

[093] EEE10. A system comprising: one or more processors; and a non-transient, computer-readable medium that stores instructions which, when executed by one or more processors, cause one or more processors to perform a specific action. Petition 870260006730, dated 01 / 23 / 2026, page 35 / 78 28 / 28 more processors perform operations of any EEE from EEE 1-9.

[094] EEE 11. A non-transient, computer-readable medium that stores instructions which, when executed by one or more processors, cause one or more processors to perform operations of any EEE from EEE 1-9. Petition 870260006730, dated 01 / 23 / 2026, page 36 / 78

Claims

1 / 1 CLAIMS 1. A method for generating a replacement frame for a lost audio frame from an audio signal, the method characterized by the fact that it comprises: Determine an audio filter based on samples from a valid audio frame preceding the missing audio frame, where the audio filter is tuned to samples from the valid audio frame preceding the missing audio frame; Determine a modified audio filter, based on the audio filter, the modified audio filter including bandwidth sharpness; to operate the audio filter as a resonator, wherein the resonator's touch is initially excited with at least one sample of the valid audio frame preceding the missing audio frame; and to generate the replacement frame based on the modified audio filter and samples of the valid audio frame preceding the missing audio frame, wherein the replacement frame is generated using the resonator's touch to extend to at least one sample in the missing audio frame, and wherein the missing audio frame is associated with a low-frequency effect LFE channel of a multichannel audio signal. Petition 870260006730, dated 01 / 23 / 2026, page 37 / 78

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