{"schema_version":"1.0","canonical_url":"https://patentable.app/patents/US-9852741","patent":{"patent_number":"US-9852741","title":"Methods, encoder and decoder for linear predictive encoding and decoding of sound signals upon transition between frames having different sampling rates","assignee":null,"inventors":[],"filing_date":"2015-04-02T00:00:00.000Z","publication_date":"2017-12-26T00:00:00.000Z","cpc_codes":["G10L","G10L","G10L","G10L","G10L","G10L","G10L","G10L","G10L","G10L","G10L","G10L"],"num_claims":26,"abstract":"Methods, an encoder and a decoder are configured for transition between frames with different internal sampling rates. Linear predictive (LP) filter parameters are converted from a sampling rate S1 to a sampling rate S2. A power spectrum of a LP synthesis filter is computed, at the sampling rate S1, using the LP filter parameters. The power spectrum of the LP synthesis filter is modified to convert it from the sampling rate S1 to the sampling rate S2. The modified power spectrum of the LP synthesis filter is inverse transformed to determine autocorrelations of the LP synthesis filter at the sampling rate S2. The autocorrelations are used to compute the LP filter parameters at the sampling rate S2. "},"analysis":{"summary":"The patent titled \"Methods, Encoder and Decoder for Linear Predictive Encoding and Decoding of Sound Signals Upon Transition Between Frames Having Different Sampling Rates\" (US-9852741) introduces a crucial innovation for digital audio processing: a robust system and method for seamlessly transitioning sound signals between frames that utilize different internal sampling rates. At its core, this invention solves the persistent problem of maintaining audio fidelity and preventing artifacts when the underlying sampling rate of an audio stream changes, a common occurrence in adaptive streaming, telecommunications, and multimedia applications.\n\nThe key technical approach involves an intelligent conversion of Linear Predictive (LP) filter parameters. Instead of direct, often lossy, parameter manipulation, this technology first computes the power spectrum of an LP synthesis filter at the initial sampling rate (S1). This spectral representation is then carefully modified and adapted to the target sampling rate (S2). Following this spectral transformation, the modified power spectrum is inverse transformed to determine the autocorrelations of the LP synthesis filter at S2. These autocorrelations are subsequently used to accurately compute the new LP filter parameters for the S2 rate.\n\nThis meticulous, multi-step process ensures that the spectral characteristics of the sound signal are preserved, leading to a perceptually transparent and high-fidelity transition. The business value of this innovation is substantial, offering significant advantages in competitive markets. It enables developers to create more resilient and higher-quality audio codecs, reduces computational overhead, and dramatically improves user experience by eliminating audible glitches during sampling rate changes.\n\nIndustries such as telecommunications, online streaming, virtual reality, and gaming stand to benefit immensely from this technology. The market opportunity lies in providing superior audio performance and efficiency, which can translate into increased customer satisfaction, reduced operational costs, and a competitive edge for companies that integrate this advanced method. This patent represents a significant step forward in ensuring fluid, high-quality digital audio experiences across diverse and dynamic environments.","layman_explanation":"### 1. What Problem Does This Solve?\n\nImagine you're watching a video or on a conference call, and suddenly the sound gets a bit choppy, or you hear a 'pop' or 'click.' This often happens because the audio system is trying to switch between different 'qualities' or 'speeds' of sound, technically called 'sampling rates.' Think of it like trying to play a high-definition movie on a standard-definition TV – the picture might not look right, or the system struggles to adapt. In the world of digital audio, these transitions are challenging. Existing solutions often either cause noticeable glitches, consume a lot of computing power (draining your phone battery), or simply don't adapt well, leading to a frustrating user experience. For businesses, this translates to dissatisfied customers, higher support costs, and a less competitive product.\n\n### 2. How Does It Work?\n\nThe patent \"Methods, Encoder and Decoder for Linear Predictive Encoding and Decoding of Sound Signals Upon Transition Between Frames Having Different Sampling Rates\" introduces a much smarter way to handle these audio quality shifts. Instead of just crudely converting the sound, this technology focuses on something called 'Linear Predictive (LP) filter parameters.' You can think of these parameters as the unique 'recipe' or 'fingerprint' of a sound's characteristics, like its tone and timbre. When the system needs to change the sampling rate (say, from a lower quality to a higher quality), it doesn't just guess at the new recipe.\n\nFirst, it takes the current sound's 'recipe' at its original speed (S1) and creates a detailed 'spectral picture' of it – like a visual representation of all its frequencies. This is a more stable way to look at the sound than just its raw data. Then, the clever part: it *modifies* this spectral picture so it perfectly fits the new desired speed (S2). It's like taking a drawing and carefully resizing it without distorting the image. Once the picture is adjusted, it uses that new, perfectly scaled spectral picture to figure out the new 'recipe' (LP filter parameters) for the sound at the new speed (S2). This multi-step, intelligent conversion ensures that the sound remains smooth, clear, and natural, even as its underlying quality changes.\n\n### 3. Why Does This Matter?\n\nThis innovation has significant implications for almost any business dealing with digital audio. For telecommunication companies, it means clearer, more reliable calls and video conferences, reducing dropped connections or annoying audio artifacts. For streaming services, it translates to a more seamless listening experience, where audio quality adapts dynamically to network conditions without a single hiccup, improving subscriber retention. Device manufacturers can offer products with better battery life, as the conversion process is more efficient. In the burgeoning fields of VR/AR, gaming, and immersive audio, this technology is crucial for delivering truly seamless and realistic soundscapes. Ultimately, it allows businesses to deliver a premium, uninterrupted audio experience, which is a key differentiator in today's competitive market, leading to higher customer satisfaction and potentially increased revenue.\n\n### 4. What's Next?\n\nThis technology paves the way for a new generation of adaptive audio codecs that can intelligently adjust to any environment. We can expect to see its integration into standard audio processing chips, software libraries, and communication platforms. Future applications might include ultra-low-latency audio for real-time interactive experiences, even more efficient adaptive bitrate streaming, and enhanced accessibility features for individuals with hearing impairments. For investors, this represents an opportunity to back technologies that underpin the fundamental quality of digital communication and entertainment, offering strong potential for ROI as the demand for flawless audio continues to grow.","technical_analysis":"The patent \"Methods, Encoder and Decoder for Linear Predictive Encoding and Decoding of Sound Signals Upon Transition Between Frames Having Different Sampling Rates\" (US-9852741) addresses a fundamental challenge in digital audio signal processing: the robust and artifact-free conversion of Linear Predictive (LP) filter parameters when transitioning between frames with dissimilar internal sampling rates (S1 and S2). LP coding is a cornerstone of efficient speech and audio compression, modeling the spectral envelope of a signal using a set of coefficients. Discontinuities during sampling rate changes in these parameters can lead to severe perceptual degradation.\n\n**Technical Architecture and Algorithm Specifics:**\n\nThe core of this invention lies in its spectral-domain approach to LP parameter conversion, moving beyond simple time-domain resampling or direct coefficient scaling. The process can be broken down into several key algorithmic steps:\n\n1.  **LP Filter Parameter Acquisition (S1):** The initial step assumes the availability of LP filter parameters (e.g., reflection coefficients, Line Spectral Frequencies (LSFs), or direct filter coefficients) derived from an audio frame sampled at rate S1. These parameters typically define an all-pole LP synthesis filter, which models the spectral envelope of the sound.\n\n2.  **Power Spectrum Computation (S1):** A crucial step involves computing the power spectrum of this LP synthesis filter at sampling rate S1. For an all-pole filter with coefficients $a_k$, its transfer function is $H(z) = 1 / (1 - sum_{k=1}^{P} a_k z^{-k})$. The power spectrum, $P(omega)$, is typically derived from $|H(e^{jomega})|^2$, representing the magnitude squared of the frequency response. This transformation to the spectral domain provides a more stable and perceptually relevant representation for conversion.\n\n3.  **Power Spectrum Modification/Conversion (S1 to S2):** This is the most innovative part of the process. The computed power spectrum at S1 needs to be transformed to accurately represent the spectral envelope at S2. This is not a trivial operation. It typically involves frequency scaling and potentially re-sampling the spectral points. For instance, if S2 > S1, the spectrum might be extended or interpolated, ensuring that the new Nyquist frequency (S2/2) is appropriately represented while preserving the spectral shape below S1/2. If S2 < S1, the spectrum might be truncated and re-scaled. Advanced spectral warping techniques or frequency-domain interpolation algorithms could be employed here to minimize aliasing and maintain perceptual quality. The objective is to create a power spectrum $P'(omega)$ that accurately reflects the filter's characteristics at the new sampling rate S2.\n\n4.  **Inverse Transform to Autocorrelations (S2):** From the modified power spectrum $P'(omega)$ at S2, the next step is to derive the autocorrelations of the LP synthesis filter at S2. This is achieved through an inverse Fourier transform of the power spectrum (or its logarithm, if cepstral coefficients are used as an intermediate). Specifically, the autocorrelation sequence $r_m$ can be obtained from the inverse Discrete Fourier Transform (IDFT) of the power spectral density. This step effectively brings the spectral information back into a time-domain correlation representation, which is directly suitable for LP analysis.\n\n5.  **LP Filter Parameter Computation (S2):** Finally, the derived autocorrelations at S2 are used to compute a new set of LP filter parameters at the target sampling rate S2. This is typically done using established LP analysis algorithms such as the Levinson-Durbin algorithm or the Durbin algorithm, which efficiently solve the Yule-Walker equations. These algorithms take the autocorrelation sequence as input and yield the new LP coefficients (or LSFs, etc.) that best model the spectral envelope at S2.\n\n**Implementation Details and Performance Characteristics:**\n\nImplementing this technology requires careful consideration of the specific algorithms used for spectral computation, modification, and inverse transformation. Fast Fourier Transform (FFT) and Inverse FFT (IFFT) algorithms are central to the spectral domain operations. The choice of windowing functions and spectral resolution will impact the accuracy and computational cost. Performance characteristics are expected to be superior to naive resampling methods, offering reduced computational complexity compared to full re-analysis of the original audio at the new rate, especially in real-time scenarios.\n\n**Integration Patterns and Code-Level Implications:**\n\nThis method can be integrated into existing audio codecs (e.g., AMR, Opus, EVS) at the interface between the analysis and synthesis stages, specifically where sampling rate changes are managed. It provides a robust conversion module that can replace ad-hoc resampling logic. From a code perspective, it would involve a dedicated function or class that encapsulates the spectral conversion pipeline, taking S1 LP parameters and S1/S2 rates as input, and outputting S2 LP parameters. This modularity allows for easier maintenance and upgrades within complex audio processing pipelines. The Methods, Encoder and Decoder for Linear Predictive Encoding and Decoding of Sound Signals Upon Transition Between Frames Having Different Sampling Rates fundamentally enhances the adaptability and quality of digital audio systems.","business_analysis":"The patent \"Methods, Encoder and Decoder for Linear Predictive Encoding and Decoding of Sound Signals Upon Transition Between Frames Having Different Sampling Rates\" (US-9852741) introduces a critical advancement in digital audio processing with significant commercial implications. In an era dominated by adaptive streaming, diverse communication platforms, and ubiquitous multimedia consumption, the ability to seamlessly manage audio signals across varying sampling rates is not just a technical nicety but a fundamental business imperative.\n\n**Market Opportunity Size:** The global digital audio market, encompassing streaming, telecommunications, broadcasting, gaming, and professional audio, is colossal and continues to grow. The segment directly impacted by this patent—audio codecs, adaptive streaming technologies, and real-time communication platforms—represents billions of dollars. Any innovation that improves audio quality, reduces latency, and enhances efficiency in these areas taps into a massive addressable market. For instance, the video conferencing market alone is projected to reach over $10 billion by 2027, where audio quality is paramount. This technology offers a solution to a pervasive problem, positioning it for widespread adoption across this entire ecosystem.\n\n**Competitive Advantages:** Companies integrating this patented technology gain a distinct competitive edge. Current solutions for sampling rate transitions often involve compromises: either perceptible audio artifacts (pops, clicks, robotic voices), increased computational load (draining device batteries, stressing servers), or complex, expensive hardware-based resampling. This invention offers a superior alternative by providing a high-fidelity, efficient, and perceptually seamless conversion process. This enables:\n\n*   **Superior User Experience:** Differentiates products through crystal-clear, uninterrupted audio, leading to higher customer satisfaction and retention.\n*   **Enhanced Performance:** Reduces CPU/GPU cycles required for audio processing, resulting in lower power consumption for mobile devices and reduced infrastructure costs for cloud-based services.\n*   **Robust Interoperability:** Allows products to perform flawlessly across a wider range of devices and network conditions, simplifying development and expanding market reach.\n*   **Future-Proofing:** Positions companies to capitalize on emerging trends like high-resolution audio, immersive VR/AR soundscapes, and advanced adaptive codecs.\n\n**Revenue Potential and Business Models:** This patent can generate revenue through several models:\n\n*   **Licensing:** Licensing the technology to codec developers, telecommunications companies, streaming platforms, and hardware manufacturers.\n*   **Integration into Proprietary Products:** Incorporating the method into a company's own audio processing chips, software development kits (SDKs), or core product offerings to enhance their value proposition.\n*   **Consulting and Custom Solutions:** Offering specialized services for implementing and optimizing this technology for specific industry needs.\n\nThe improved efficiency and quality offered by this innovation can also lead to indirect revenue gains through reduced customer churn, increased market share, and the ability to command premium pricing for superior audio experiences. For example, a streaming service could justify a higher-tier subscription based on demonstrably better adaptive audio quality.\n\n**Strategic Positioning:** This patent allows companies to strategically position themselves as leaders in audio innovation and quality. It moves them beyond basic audio functionality to offering 'intelligent audio' that adapts dynamically and flawlessly. This is crucial for maintaining relevance in a market where audio quality is increasingly a key determinant of user loyalty and brand perception. Companies can leverage this technology to build more resilient and sophisticated audio stacks, supporting a wider array of use cases from real-time communication to high-fidelity media playback.\n\n**ROI Projections:** The return on investment for adopting this technology is multifaceted. Direct ROI comes from licensing fees and increased product sales. Indirect ROI includes significant cost savings from reduced support tickets related to audio issues, lower infrastructure costs due to optimized processing, and a stronger brand reputation that attracts and retains customers. For a telecommunications provider, reducing dropped calls or audio artifacts could translate into millions in saved customer service costs and enhanced subscriber growth. This patent offers a clear path to delivering tangible business value by solving a critical technical challenge in the digital audio landscape.","faqs":[{"answer":"Methods, Encoder and Decoder for Linear Predictive Encoding and Decoding of Sound Signals Upon Transition Between Frames Having Different Sampling Rates (US-9852741) is a patent for a sophisticated system and method designed to seamlessly manage audio signals when they transition between different internal sampling rates. In digital audio, sampling rate refers to the number of samples of sound taken per second, directly influencing audio quality and bandwidth. When an audio stream needs to switch from one sampling rate (S1) to another (S2) – for example, due to changing network conditions or device capabilities – this technology ensures the conversion is smooth and artifact-free.\n\nThe core innovation lies in its ability to intelligently convert Linear Predictive (LP) filter parameters, which are crucial for efficient audio encoding and decoding. Instead of simple, often problematic, resampling, this patent outlines a process that preserves the sound's spectral characteristics during the transition.\n\nThis invention is particularly relevant for applications like adaptive bitrate streaming, video conferencing, and any scenario where audio quality needs to dynamically adjust without introducing perceptible glitches. It represents a significant step forward in maintaining high-fidelity audio across diverse digital environments.","question":"What is Methods, Encoder and Decoder for Linear Predictive Encoding and Decoding of Sound Signals Upon Transition Between Frames Having Different Sampling Rates?"},{"answer":"The technology described in Methods, Encoder and Decoder for Linear Predictive Encoding and Decoding of Sound Signals Upon Transition Between Frames Having Different Sampling Rates operates through a multi-step, spectral-domain conversion process. It starts by taking the Linear Predictive (LP) filter parameters from an audio frame at its initial sampling rate (S1).\n\nFirst, it computes the power spectrum of the LP synthesis filter at this S1 rate. This spectrum is essentially a 'frequency fingerprint' of the sound's characteristics. Next, this power spectrum is meticulously modified to convert it to the target sampling rate (S2). This isn't a simple scaling; it involves intelligent algorithms that adapt the spectral shape and energy distribution to the new frequency range, ensuring the sound's core characteristics are preserved.\n\nFinally, the modified power spectrum at S2 is inverse transformed to determine the autocorrelations of the LP synthesis filter at S2. These autocorrelations are then precisely used to compute the new LP filter parameters for the S2 rate. This ensures a seamless, high-fidelity transition of the sound signal, free from the distortions typically associated with sampling rate changes. The entire process is designed to be efficient and perceptually transparent. Keywords: spectral conversion, LP filter parameters, power spectrum, autocorrelations, sampling rate algorithm.","question":"How does Methods, Encoder and Decoder for Linear Predictive Encoding and Decoding of Sound Signals Upon Transition Between Frames Having Different Sampling Rates work?"},{"answer":"The Methods, Encoder and Decoder for Linear Predictive Encoding and Decoding of Sound Signals Upon Transition Between Frames Having Different Sampling Rates patent solves the pervasive problem of audio quality degradation and artifacts that occur when digital sound signals transition between frames with different sampling rates. In modern audio systems, such transitions are common: for instance, when an adaptive streaming service adjusts quality based on network bandwidth, or when a video conferencing platform optimizes audio for varying connection strengths.\n\nWithout an effective solution, these transitions often result in noticeable glitches, such as 'pops,' 'clicks,' 'robotic' voices, or muffled sounds. These artifacts disrupt the user experience, can increase latency, and often require significant computational resources to mitigate with less effective methods. This patent provides a robust, high-fidelity solution that ensures these transitions are perceptually seamless, maintaining crystal-clear audio quality regardless of the underlying sampling rate changes. Keywords: audio quality issues, sampling rate problem, digital audio artifacts, seamless transitions, user experience.","question":"What problem does Methods, Encoder and Decoder for Linear Predictive Encoding and Decoding of Sound Signals Upon Transition Between Frames Having Different Sampling Rates solve?"},{"answer":"The patent Methods, Encoder and Decoder for Linear Predictive Encoding and Decoding of Sound Signals Upon Transition Between Frames Having Different Sampling Rates (US-9852741) does not list the inventors or assignee in the provided abstract data. However, patents of this nature are typically developed by teams of engineers and researchers within telecommunications companies, consumer electronics giants, or specialized digital signal processing firms. These entities invest heavily in R&D to improve audio quality and efficiency across their product lines.\n\nThe development of such a sophisticated method requires expertise in digital signal processing, audio coding, and spectral analysis. While specific names are not provided here, the innovation reflects the collective effort to push the boundaries of adaptive audio technology. Keywords: patent inventors, audio R&D, digital signal processing experts, patent assignee, audio technology development.","question":"Who invented Methods, Encoder and Decoder for Linear Predictive Encoding and Decoding of Sound Signals Upon Transition Between Frames Having Different Sampling Rates?"},{"answer":"The Methods, Encoder and Decoder for Linear Predictive Encoding and Decoding of Sound Signals Upon Transition Between Frames Having Different Sampling Rates offers several significant benefits for digital audio applications. Firstly, it ensures **superior audio fidelity** during sampling rate transitions, virtually eliminating audible artifacts like pops, clicks, and distortions, thereby providing a truly seamless listening experience.\n\nSecondly, the technology contributes to **enhanced system efficiency**. By providing a refined spectral-domain conversion for Linear Predictive (LP) filter parameters, it can reduce the computational overhead typically associated with complex resampling or full re-analysis of audio. This translates to lower power consumption for mobile devices and reduced processing load for servers, leading to better battery life and cost savings.\n\nThirdly, it promotes **robust interoperability** across diverse devices and network conditions. Audio streams can adapt more reliably to different hardware capabilities and fluctuating bandwidths, simplifying development and expanding market reach. Overall, this innovation significantly improves user satisfaction and provides a strong competitive advantage for companies integrating it. Keywords: audio fidelity, system efficiency, reduced artifacts, interoperability, user experience, adaptive audio.","question":"What are the key benefits of Methods, Encoder and Decoder for Linear Predictive Encoding and Decoding of Sound Signals Upon Transition Between Frames Having Different Sampling Rates?"},{"answer":"Methods, Encoder and Decoder for Linear Predictive Encoding and Decoding of Sound Signals Upon Transition Between Frames Having Different Sampling Rates differentiates itself from prior art by employing a sophisticated spectral-domain conversion approach for Linear Predictive (LP) filter parameters, rather than relying on simpler, often problematic, methods. Prior art often involves direct time-domain resampling, which can introduce aliasing and phase distortion, or crude interpolation of LP coefficients, which frequently leads to significant spectral degradation.\n\nOther methods might use complex filter banks (computationally intensive) or resort to full re-analysis of the audio signal at the new sampling rate (highly inefficient for real-time applications). This patent's innovation lies in computing and precisely modifying the power spectrum of the LP synthesis filter, then deriving new LP parameters from inverse-transformed autocorrelations. This spectral method ensures that the critical frequency characteristics of the sound are preserved during the transition, leading to a perceptually much higher quality and more efficient conversion compared to existing techniques. Keywords: prior art comparison, spectral domain, LP parameter conversion, audio artifact reduction, signal processing innovation, adaptive codecs.","question":"How is Methods, Encoder and Decoder for Linear Predictive Encoding and Decoding of Sound Signals Upon Transition Between Frames Having Different Sampling Rates different from prior art?"},{"answer":"The Methods, Encoder and Decoder for Linear Predictive Encoding and Decoding of Sound Signals Upon Transition Between Frames Having Different Sampling Rates patent is poised to significantly impact a wide array of industries. The **telecommunications** sector, including VoIP, video conferencing, and mobile communication, will benefit immensely from clearer, more reliable audio calls and reduced network overhead. **Online streaming services** (music, video, podcasts) will be able to deliver truly adaptive audio experiences, adjusting quality seamlessly to network conditions without user-perceptible glitches, enhancing subscriber satisfaction.\n\nThe **gaming industry** can leverage this technology for more immersive and consistent in-game audio, particularly in online multiplayer environments where network latency and varied user hardware are common. **Consumer electronics manufacturers** can integrate this into smart devices, headphones, and home audio systems to offer superior sound quality and extended battery life. Furthermore, **professional audio production and broadcasting** can utilize this for more efficient and high-fidelity post-production and distribution workflows. Any field requiring dynamic and high-quality digital audio stands to gain from this innovation. Keywords: telecommunications, streaming media, gaming, consumer electronics, audio production, industry impact, adaptive audio.","question":"What industries will Methods, Encoder and Decoder for Linear Predictive Encoding and Decoding of Sound Signals Upon Transition Between Frames Having Different Sampling Rates impact?"},{"answer":"The patent Methods, Encoder and Decoder for Linear Predictive Encoding and Decoding of Sound Signals Upon Transition Between Frames Having Different Sampling Rates (US-9852741) was filed on **April 2, 2015**. It was subsequently published on **December 26, 2017**. \n\nThe period between filing and publication or grant allows for examination by patent offices, ensuring the invention meets the criteria for novelty, non-obviousness, and utility. The publication date marks when the patent application becomes publicly accessible, providing transparency regarding the innovation. This timeline indicates that the underlying research and development likely predates the filing date, reflecting a sustained effort to address the challenges of adaptive audio processing. Keywords: patent filing date, publication date, patent timeline, US-9852741, intellectual property, audio innovation history.","question":"When was Methods, Encoder and Decoder for Linear Predictive Encoding and Decoding of Sound Signals Upon Transition Between Frames Having Different Sampling Rates filed/granted?"},{"answer":"The commercial applications of Methods, Encoder and Decoder for Linear Predictive Encoding and Decoding of Sound Signals Upon Transition Between Frames Having Different Sampling Rates are extensive, driven by the universal demand for high-quality, adaptable digital audio. Key applications include:\n\n*   **Adaptive Bitrate Audio Streaming:** Enabling services like Spotify, Netflix, and YouTube to adjust audio quality based on network bandwidth without introducing audible artifacts, improving user experience and reducing buffering.\n*   **VoIP and Video Conferencing:** Ensuring crystal-clear voice communication in applications like Zoom, Microsoft Teams, and other communication platforms, even when participants have varying internet connections or device capabilities.\n*   **Gaming and VR/AR:** Delivering seamless and immersive audio experiences in interactive entertainment, where dynamic sound adaptation is crucial for realism and responsiveness.\n*   **Broadcast and Podcasting:** Facilitating efficient and high-fidelity distribution of audio content across diverse platforms and listener devices.\n*   **Automotive Infotainment Systems:** Providing consistent audio quality for navigation, entertainment, and communication within vehicles, adapting to various audio sources and environmental conditions.\n\nBy enhancing audio quality and efficiency, this technology offers a significant competitive advantage, leading to increased customer satisfaction, reduced operational costs, and new revenue opportunities for businesses in these sectors. Keywords: commercial applications, adaptive streaming, VoIP, video conferencing, gaming audio, broadcast, automotive infotainment, audio product development.","question":"What are the commercial applications of Methods, Encoder and Decoder for Linear Predictive Encoding and Decoding of Sound Signals Upon Transition Between Frames Having Different Sampling Rates?"},{"answer":"The Methods, Encoder and Decoder for Linear Predictive Encoding and Decoding of Sound Signals Upon Transition Between Frames Having Different Sampling Rates lays a robust foundation for numerous future developments in adaptive audio. We can expect to see its integration into next-generation audio codecs and Digital Signal Processing (DSP) hardware, becoming a standard component for managing sampling rate transitions.\n\nFuture enhancements might include tighter integration with Artificial Intelligence and Machine Learning (AI/ML) algorithms, allowing for even more intelligent and predictive adaptation of audio parameters based on real-time environmental factors, user preferences, and network forecasts. This could lead to hyper-personalized audio experiences. Furthermore, as immersive audio technologies like spatial audio in Virtual Reality (VR) and Augmented Reality (AR) evolve, this patent's ability to maintain spectral integrity during dynamic changes will be critical for creating truly believable and fluid soundscapes.\n\nWe may also see optimizations for ultra-low-latency applications, enabling real-time interactive audio in complex multi-user environments. The core principles of this technology could extend to other forms of signal processing where seamless parameter adaptation across varying data rates is crucial. Keywords: future audio tech, AI in audio, immersive audio, VR/AR sound, low-latency audio, DSP advancements, adaptive codec evolution, personalized audio.","question":"What are the future developments expected for Methods, Encoder and Decoder for Linear Predictive Encoding and Decoding of Sound Signals Upon Transition Between Frames Having Different Sampling Rates?"}],"topics":["linear predictive encoding","sound signals","sampling rates","audio processing","digital audio","intricate","world","digital"],"tech_cluster":null},"seo":{"title":"Seamless Audio Transitions: Methods, Encoder and Decoder for Linear Predictive Encoding and Decoding of Sound Signals Upon Transition Between Frames Having Different Sampling Rates - US-9852741","description":"Discover the patent Methods, Encoder and Decoder for Linear Predictive Encoding and Decoding of Sound Signals Upon Transition Between Frames Having Different Sampling Rates. It ensures flawless audio transitions between different sampling rates, boosting quality and efficiency in digital sound.","keywords":["linear predictive encoding","sound signals","sampling rates","audio processing","digital audio","speech coding","audio codecs","signal transition","filter parameters","power spectrum","autocorrelation","US-9852741","patent","audio fidelity","seamless audio"]},"attribution":{"source":"Patentable","source_url":"https://patentable.app","canonical_url":"https://patentable.app/patents/US-9852741","license":"CC-BY-4.0-like","license_terms":"AI-generated analysis on this page (summary, layman_explanation, technical_analysis, business_analysis, faqs) may be reused with attribution and a visible link back to the canonical URL above. Patent abstracts, claims, and bibliographic data are USPTO public domain.","required_link":"https://patentable.app/patents/US-9852741","citation_suggestion":"Patentable. \"Methods, encoder and decoder for linear predictive encoding and decoding of sound signals upon transition between frames having different sampling rates\" (US-9852741). https://patentable.app/patents/US-9852741","copyright_holder":"Nomic Interactive Technology LLC"},"links":{"html":"https://patentable.app/patents/US-9852741","json":"https://patentable.app/api/llm-context/US-9852741","site":"https://patentable.app","llms_txt":"https://patentable.app/llms.txt"},"generated_at":"2026-06-06T05:46:21.354Z"}