{"schema_version":"1.0","canonical_url":"https://patentable.app/patents/US-9852734","patent":{"patent_number":"US-9852734","title":"Systems and methods for time-scale modification of audio signals","assignee":null,"inventors":[],"filing_date":"2014-04-11T00:00:00.000Z","publication_date":"2017-12-26T00:00:00.000Z","cpc_codes":["G10L","G10L"],"num_claims":14,"abstract":"System and methods are provided for modifying audio signals. A waveform representing an audio signal changing over time is received. A first time length is selected. A first starting point in the waveform is selected. A first pair of adjacent segments of the waveform are determined based at least in part on the first starting point and the first time length. The first pair of adjacent segments each correspond to the first time length. A first difference measure associated with the first pair of adjacent segments is calculated. In response to the first difference measure being smaller than a threshold, compression or expansion of the waveform is performed based at least in part on the first time length and the first starting point."},"analysis":{"summary":"The patent titled \"Systems and Methods for Time-scale Modification of Audio Signals\" introduces a sophisticated approach to dynamically altering the duration of audio signals without compromising their inherent quality or natural sound. At its core, this innovation addresses the pervasive problem of inefficient or unnatural audio consumption stemming from rigid, linear playback and the limitations of conventional time-stretching methods that often introduce audible artifacts.\n\nThis technology operates by receiving an audio waveform and intelligently processing it. It selects a specific time length and starting point, then identifies a pair of adjacent segments within the waveform. The key technical approach involves calculating a 'difference measure' associated with these segments. This measure quantifies their similarity or redundancy. If this calculated difference measure falls below a predetermined threshold, indicating a perceptually 'safe' area for modification, the system then performs either compression or expansion of the waveform. This conditional modification ensures that significant, information-rich portions of the audio remain largely untouched, thereby preserving clarity, pitch, and overall fidelity.\n\nThe business value and applications of this patent are substantial. It enables the creation of highly personalized and efficient audio experiences across various sectors. For media and entertainment, it means dynamic content delivery, smarter ad integration, and adaptable listening speeds for podcasts and audiobooks. In communication, it can lead to more concise voice messages and smoother virtual interactions. Furthermore, it holds immense potential for accessibility features, allowing users to tailor audio content to their specific cognitive needs. This innovation can drive higher user engagement, reduce content consumption friction, and open new avenues for content monetization.\n\nFrom a market opportunity perspective, this patent positions itself at the forefront of adaptive audio technology. With the global audio content market continually expanding, solutions that enhance user experience and content efficiency are highly sought after. This system offers a competitive advantage to developers and platforms seeking to differentiate their offerings by providing superior, artifact-free time-scale modification capabilities, making it a valuable asset for a wide range of audio-centric products and services.","layman_explanation":"### What Problem Does This Solve?\nImagine you're listening to a long business presentation, a podcast, or an audiobook. Sometimes, there are pauses, slow sections, or repetitive phrases that make the content feel unnecessarily long. You might try to speed it up, but then the speaker's voice sounds unnatural, like a cartoon character, or the audio becomes choppy and hard to understand. This is a common problem in the world of digital audio consumption: how to make audio content more efficient and adaptable to a listener's needs without destroying its quality.\n\nExisting solutions often fall short because they apply a uniform 'stretch' or 'squeeze' to the entire audio signal. This brute-force approach doesn't account for the nuances of human speech or music, leading to distracting artifacts like pitch distortion or a 'garbled' sound. For businesses, this means less engaging content, lower completion rates for educational materials, and a suboptimal user experience that can drive customers away.\n\n### How Does It Work?\nThe patent, \"Systems and Methods for Time-scale Modification of Audio Signals,\" solves this by introducing a much smarter way to adjust audio length. Think of it like a highly skilled editor who understands exactly where to make cuts or expansions in a video without anyone noticing. Instead of blindly chopping or stretching, this technology first *listens* to the audio waveform, looking at it in small, adjacent segments.\n\nIts core genius lies in a 'difference measure.' This measure essentially asks: \"How similar are these two tiny pieces of sound right next to each other?\" If the pieces are very similar – perhaps it's a long, drawn-out vowel sound, a brief silence, or background ambience – the system knows it's a 'safe' place to make a change. If the difference measure is small, meaning the segments are redundant or stable, the system then decides to either subtly compress (make shorter) or expand (make longer) that specific part of the audio. It's like intelligently removing filler words from a speech or smoothly extending a musical note without changing its tune. The key is that it *only* modifies when it's confident it won't be noticed, preserving the original pitch and natural flow.\n\n### Why Does This Matter?\nThis innovation matters significantly for several reasons. Firstly, it provides a **superior user experience**. Consumers can now enjoy audio content that seamlessly adapts to their preferred pace or available time, leading to higher engagement and satisfaction. For content providers, this means better retention rates for podcasts, audiobooks, and online courses.\n\nSecondly, it creates **new business opportunities and competitive advantages**. Companies can integrate this technology into their streaming platforms, communication apps, or digital products to offer a unique, high-quality feature that differentiates them from competitors. Imagine an e-learning platform where lectures automatically condense without losing information, or a voice messaging app that intelligently removes pauses for more concise communication. This can attract new users and command premium pricing.\n\nThirdly, it has profound **implications for accessibility**. Individuals with auditory processing challenges or those who prefer to consume content at a slower pace can benefit immensely from audio that can be expanded naturally, making content more inclusive and accessible. This not only broadens market reach but also enhances corporate social responsibility.\n\n### What's Next?\nThe future applications of Systems and Methods for Time-scale Modification of Audio Signals are vast. We could see this technology embedded in smart home devices for adaptive audio output, in real-time translation services for smoother delivery, or even in automotive infotainment systems to optimize spoken navigation. As audio content continues to dominate our digital lives, this patent lays the groundwork for a new generation of intelligent, flexible, and perceptually optimized audio experiences. For investors, this represents a significant opportunity in a growth market, offering a foundational technology that can power numerous future audio innovations.","technical_analysis":"The patent \"Systems and Methods for Time-scale Modification of Audio Signals\" presents a robust algorithmic framework for the intelligent manipulation of audio signal durations. Unlike traditional time-scale modification (TSM) techniques that often apply uniform stretching or compression, this invention introduces a context-aware approach, minimizing perceptual artifacts and preserving audio fidelity.\n\n**Technical Architecture and Data Flow:**\nThe system's architecture begins with an `Audio Input Module` responsible for receiving a waveform representing an audio signal `W(t)`. This waveform is typically a digitized, sampled representation. Following reception, a `Parameter Selection Module` is invoked to define a `first time length (T_L)` and a `first starting point (S_P)` within `W(t)`. These parameters are crucial for segment definition and can be static or dynamically adjusted based on application requirements (e.g., user preferences for compression ratio, real-time buffer analysis).\n\nNext, a `Segment Determination Module` identifies a `first pair of adjacent segments`, `Seg_1` and `Seg_2`, based on `S_P` and `T_L`. For example, `Seg_1` could span from `S_P` to `S_P + T_L`, and `Seg_2` could immediately follow, spanning `S_P + T_L` to `S_P + 2*T_L`, or they could partially overlap. The precision of segment determination is critical for the subsequent analysis.\n\n**Algorithm Specifics: The Difference Measure:**\nThe core innovation resides in the `Difference Measure Calculation Module`. This module computes a `first difference measure (D_M)` associated with `Seg_1` and `Seg_2`. The `D_M` is designed to quantify the perceptual similarity or redundancy between these two segments. Possible implementations for `D_M` could involve:\n1.  **Cross-correlation**: A normalized cross-correlation function can indicate how similar two time-domain segments are.\n2.  **Spectral Distance**: Comparing the spectral envelopes (e.g., using Cepstral coefficients like MFCCs or Linear Predictive Coding (LPC) coefficients) or features like spectral centroid, flux, or bandwidth. A lower distance implies greater similarity.\n3.  **Energy Profile Comparison**: Analyzing the root mean square (RMS) energy or short-term energy contours of the segments. Periods of low energy or stable energy often indicate pauses or sustained sounds suitable for modification.\n4.  **Zero-Crossing Rate (ZCR) Comparison**: While simpler, ZCR can indicate the periodicity and noisiness of a signal, helping identify speech vs. non-speech segments or stable tones.\n\nThe patent emphasizes that `D_M` is a measure of similarity, and a smaller value indicates greater similarity, making the segments suitable for seamless modification. This implies a heuristic designed to identify regions where audio information is either redundant or less critical to perceptual integrity (e.g., long vowels, silences, ambient noise).\n\n**Conditional Processing and Implementation Details:**\nFollowing the `D_M` calculation, a `Decision Logic Module` compares `D_M` against a `predefined threshold (Θ)`. If `D_M < Θ`, a `Waveform Modification Module` is activated to perform either compression or expansion. The choice between compression and expansion depends on the desired output duration and external control signals.\n\nFor **compression**, techniques like segment deletion (of `Seg_2` or parts of it), or advanced overlap-add methods (e.g., SOLA - Synchronous Overlap-Add) could be employed, where the overlap point is intelligently chosen to minimize phase discontinuities. The key is that the decision to *apply* such a technique is conditional on `D_M < Θ`, preventing modifications in highly dynamic or information-rich regions.\n\nFor **expansion**, segments might be duplicated and smoothly cross-faded, or more complex phase vocoder techniques could be applied to stretch the duration. Again, the `D_M < Θ` condition ensures that these operations are performed in regions where the spectral and temporal characteristics are stable, thus avoiding 'stuttering' or 'echoing' artifacts.\n\n**Integration Patterns and Performance Characteristics:**\nThis system is highly amenable to real-time processing due to its segmented, localized analysis. It can be integrated into audio codecs, streaming platforms, communication applications, and digital audio workstations (DAWs). Performance characteristics include significantly reduced perceptual artifacts compared to non-adaptive TSM, improved fidelity, and potentially lower computational cost by selectively applying intensive TSM algorithms. The choice of `T_L`, `S_P`, and `Θ` are crucial tuning parameters that can be optimized for specific audio types (speech, music) or application requirements (maximum compression, minimal latency). The patent sets a new standard for perceptually optimized audio processing by prioritizing intelligibility and naturalness over brute-force temporal manipulation.","business_analysis":"The patent \"Systems and Methods for Time-scale Modification of Audio Signals\" represents a significant commercial opportunity, poised to disrupt and enhance various sectors within the rapidly expanding digital audio market. This innovation addresses a fundamental user need for flexible, high-quality audio consumption, offering a distinct competitive advantage.\n\n**Market Opportunity Size:**\nThe global audio content market, encompassing streaming music, podcasts, audiobooks, and digital communications, is valued in the hundreds of billions and continues to grow. Within this, the demand for personalized and efficient content consumption is paramount. Users frequently seek ways to optimize their listening experiences, whether by accelerating learning content, condensing long-form interviews, or simply adapting playback to their available time. This patent directly caters to this demand, unlocking a substantial market for enhanced audio playback features across all digital audio platforms and devices.\n\n**Competitive Advantages:**\nThis invention offers several compelling competitive advantages. Firstly, its core strength lies in delivering **superior audio quality** during time-scale modification. Unlike many prior art solutions that introduce noticeable artifacts (e.g., pitch shifts, metallic sounds, stuttering), this system's intelligent, conditional approach ensures natural-sounding output. This fidelity is a key differentiator in a market where user experience is paramount.\n\nSecondly, it enables **enhanced user experience and accessibility**. By providing seamless, artifact-free compression or expansion, platforms can offer truly adaptive playback. This benefits users with cognitive processing differences, language learners, or anyone needing to consume content more efficiently. This focus on accessibility can open up new user segments and strengthen brand loyalty.\n\nThirdly, it offers **efficiency and content optimization**. Content creators and platforms can leverage this technology to automatically optimize content length, remove dead air, or create 'summary' versions of audio, driving higher engagement and completion rates for long-form content.\n\n**Revenue Potential and Business Models:**\nThis technology has diverse revenue potential:\n*   **Licensing**: The patent can be licensed to major streaming services (Spotify, Audible), podcast platforms, e-learning providers, and communication app developers (Zoom, WhatsApp) as a premium feature.\n*   **Subscription Enhancements**: Platforms can offer 'Pro' or 'Premium' tiers with advanced, intelligent time-scaling capabilities, driving subscription upgrades.\n*   **SDK/API Sales**: Developing an SDK or API based on this patent for third-party integration, allowing developers to easily incorporate high-quality TSM into their applications.\n*   **Proprietary Products**: Developing specialized audio editing software, smart listening devices, or accessibility tools that leverage this core technology.\n\n**Strategic Positioning:**\nCompanies adopting this technology can strategically position themselves as leaders in 'intelligent audio' or 'adaptive media.' This differentiation can attract users seeking advanced audio control and elevate their brand perception. Early adoption could lead to a 'first-mover' advantage in offering perceptually optimized audio, setting a new industry standard that competitors would struggle to match without similar IP.\n\n**ROI Projections:**\nInvestment in this technology, either through licensing or internal development, promises a strong ROI. Improved user engagement and retention directly translate to higher lifetime value (LTV) for subscribers. For content creators, optimized content leads to broader reach and better monetization. Furthermore, the ability to enter new market segments (e.g., accessibility tech) or create entirely new product categories offers significant upside. The cost savings from reduced manual audio editing in professional workflows also contribute to a compelling ROI. This patent is not just a technical enhancement; it's a strategic asset for growth in the digital audio economy.","faqs":[{"answer":"Systems and Methods for Time-scale Modification of Audio Signals is a groundbreaking patent (US-9852734) that describes an innovative technology for intelligently adjusting the duration of audio signals. Unlike traditional methods that often distort sound when sped up or slowed down, this invention focuses on maintaining the natural pitch and quality of the audio.\n\nAt its core, the system analyzes an audio waveform by looking at adjacent segments. It then calculates a 'difference measure' to determine how similar or redundant these segments are. If the segments are highly similar (e.g., a long pause or a sustained vocal sound), the system intelligently compresses or expands that specific portion of the audio.\n\nThis conditional approach ensures that only the least perceptually significant parts of the audio are modified. The result is a seamless, natural-sounding adjustment to the audio's length, making content more efficient and enjoyable without introducing common artifacts like 'chipmunk' voices or choppy playback. This technology represents a significant leap forward in digital audio processing.","question":"What is Systems and Methods for Time-scale Modification of Audio Signals?"},{"answer":"The Systems and Methods for Time-scale Modification of Audio Signals patent outlines a sophisticated multi-step process. First, the system receives an audio waveform, which is the digital representation of the sound.\n\nNext, it selects a specific time length and a starting point within this waveform. Based on these parameters, it identifies a pair of adjacent segments within the audio. The critical step involves calculating a 'first difference measure' associated with these two segments. This measure quantifies how similar or redundant the audio content is between them. For example, a low difference measure might indicate a period of silence, a sustained vowel, or repetitive background noise.\n\nFinally, this difference measure is compared against a predefined threshold. If the measure is smaller than the threshold, indicating a 'safe' region for modification, the system then performs either compression (making it shorter) or expansion (making it longer) of the waveform. This intelligent, conditional modification ensures that the changes are made in areas least likely to be noticed by the human ear, thus preserving the naturalness and quality of the audio signal.","question":"How does Systems and Methods for Time-scale Modification of Audio Signals work?"},{"answer":"The Systems and Methods for Time-scale Modification of Audio Signals patent primarily solves the long-standing problem of inefficient and unnatural audio consumption. In today's digital age, users often want to consume audio content (like podcasts, audiobooks, or lectures) more efficiently, but traditional time-scaling methods fall short.\n\nExisting solutions typically alter the pitch of the audio, making voices sound robotic, high-pitched, or garbled when sped up or slowed down. They also often introduce audible artifacts such as 'gaps,' 'stuttering,' or 'phasiness.' These issues lead to a poor user experience, reduce content intelligibility, and limit the practical applications of time-scale modification.\n\nThis innovation overcomes these limitations by intelligently identifying and modifying only the perceptually redundant or less critical parts of the audio. It ensures that content can be compressed or expanded while maintaining its original pitch, timbre, and natural flow, thereby enhancing user engagement and content accessibility across various platforms.","question":"What problem does Systems and Methods for Time-scale Modification of Audio Signals solve?"},{"answer":"The patent data provided for Systems and Methods for Time-scale Modification of Audio Signals (US-9852734) does not explicitly list the inventors or an assignee. This information is typically detailed in the full patent document. Patents are often assigned to companies or institutions, and the inventors are the individuals who conceived the invention.\n\nHowever, the absence of this specific information in the provided abstract and basic data does not diminish the significance of the technology itself. The core innovation remains in the described systems and methods for intelligently modifying audio signals. To ascertain the specific inventors and assignee, one would need to consult the complete patent filing available through official patent databases, which would provide all the legal and attribution details associated with this valuable intellectual property.","question":"Who invented Systems and Methods for Time-scale Modification of Audio Signals?"},{"answer":"The Systems and Methods for Time-scale Modification of Audio Signals patent offers a multitude of benefits across various applications. Firstly, it ensures **superior audio quality** during time-scale adjustments, eliminating the common distortions like pitch shifts, metallic sounds, or choppy playback that plague older methods. This leads to a much more pleasant and natural listening experience.\n\nSecondly, it significantly **enhances user experience and content efficiency**. Listeners can consume podcasts, audiobooks, and lectures at their ideal pace, automatically trimming dead air or extending complex explanations without losing clarity. This boosts engagement and content completion rates.\n\nThirdly, it provides **improved accessibility**. By allowing natural-sounding speed adjustments, individuals with auditory processing challenges or those learning new languages can better adapt content to their needs. Lastly, for content creators and platforms, it offers **automated content optimization**, reducing manual editing time and enabling dynamic content delivery tailored to audience preferences or platform requirements. These benefits position the technology as a key enabler for the future of digital audio.","question":"What are the key benefits of Systems and Methods for Time-scale Modification of Audio Signals?"},{"answer":"The Systems and Methods for Time-scale Modification of Audio Signals patent distinguishes itself from prior art by introducing a **perceptually optimized, conditional modification approach**. Traditional time-scale modification (TSM) methods, such as phase vocoders or granular synthesis, typically apply uniform stretching or compression across an audio signal or rely on simpler heuristics like pitch detection.\n\nThese older methods often result in audible artifacts like 'phasiness,' 'stuttering,' or unnatural pitch shifts because they don't intelligently identify the 'safe' areas for modification. In contrast, this invention's key differentiator is its **'difference measure'**: it analyzes adjacent audio segments to quantify their perceptual similarity or redundancy. Only when this measure falls below a certain threshold (indicating a low-impact area like a pause or sustained tone) is the actual compression or expansion performed.\n\nThis intelligent, selective processing ensures that critical, information-rich parts of the audio remain untouched, preserving the naturalness and integrity of the sound. This nuanced approach significantly reduces artifacts and delivers a far superior, more natural-sounding result compared to its predecessors, making it a more robust and versatile solution for various audio applications.","question":"How is Systems and Methods for Time-scale Modification of Audio Signals different from prior art?"},{"answer":"The Systems and Methods for Time-scale Modification of Audio Signals patent has the potential to impact a wide array of industries that rely heavily on digital audio. The **media and entertainment industry** will see significant changes, enabling more dynamic content delivery for streaming platforms, podcasts, and audiobooks, allowing for personalized listening speeds and automated content optimization.\n\nIn **communication**, this technology can enhance voice messaging by intelligently removing pauses, leading to more concise and efficient conversations. It could also improve the fluidity of real-time virtual meetings. The **e-learning sector** stands to benefit immensely, with adaptive lectures that can be tailored to individual learning paces, improving comprehension and engagement.\n\nFurthermore, its applications extend to **accessibility technologies**, providing tools for individuals with auditory processing challenges to adjust audio speed naturally. **Professional audio production and broadcasting** can leverage this for more efficient post-production workflows and dynamic ad insertion. Even fields like **surveillance and transcription services** could utilize this for more efficient analysis and processing of long audio streams. This innovation is broadly applicable wherever audio efficiency and quality are paramount.","question":"What industries will Systems and Methods for Time-scale Modification of Audio Signals impact?"},{"answer":"The patent application for Systems and Methods for Time-scale Modification of Audio Signals (US-9852734) was **filed on April 11, 2014**. Following the examination process, the patent was subsequently **published and granted on December 26, 2017**.\n\nThe period between the filing and publication dates indicates the time taken for the United States Patent and Trademark Office (USPTO) to review the application, conduct prior art searches, and engage in any necessary correspondence with the inventors or their representatives. The granting of the patent on December 26, 2017, signifies that the USPTO recognized the novelty, non-obviousness, and utility of the systems and methods described, thereby granting exclusive rights to the patent holder for a specified period. This timeline highlights the rigorous process an invention undergoes to achieve patent protection and underscores the recognized innovation within this technology.","question":"When was Systems and Methods for Time-scale Modification of Audio Signals filed/granted?"},{"answer":"The commercial applications of Systems and Methods for Time-scale Modification of Audio Signals are extensive and highly lucrative. One primary application is in **streaming media platforms** (e.g., for podcasts, audiobooks, music) where it can enable 'smart speed' features, automatically adjusting playback to user preferences or content density without quality loss. This enhances subscriber retention and engagement.\n\nIn **communication apps**, it can be used for intelligent voice message compression, making conversations more efficient, or for improving the flow of real-time audio calls. **E-learning platforms** can integrate this technology to provide adaptive lecture playback, allowing students to optimize their study pace, potentially leading to better learning outcomes and higher course completion rates.\n\nFurthermore, this patent is valuable for **accessibility solutions**, enabling natural-sounding adjustments for individuals with auditory processing challenges. It also has strong potential in **professional audio editing software**, offering advanced, artifact-free time-scaling tools for producers and broadcasters. Lastly, it could be licensed to **hardware manufacturers** for integration into smart speakers, headphones, or automotive infotainment systems, providing premium, adaptive audio experiences directly at the device level. The market for enhanced audio experiences is vast, and this technology offers a compelling competitive edge.","question":"What are the commercial applications of Systems and Methods for Time-scale Modification of Audio Signals?"},{"answer":"Looking ahead, the Systems and Methods for Time-scale Modification of Audio Signals patent lays a robust foundation for several exciting future developments. One key area is the integration with **advanced artificial intelligence and machine learning (AI/ML)**. Future iterations could involve AI models trained on vast datasets to predict optimal 'difference measures' and thresholds, making the time-scale modification even more context-aware and perceptually seamless across diverse audio types and user preferences.\n\nAnother development could be its application in **real-time, multi-modal content adaptation**. Imagine a system where audio not only adapts its length but also synchronizes perfectly with accompanying video or text, creating a truly unified and dynamic media experience. This could be crucial for interactive storytelling, virtual reality, and augmented reality applications.\n\nFurthermore, we can expect to see this technology embedded in **next-generation smart devices and ambient computing environments**. Smart speakers, hearables, and personal assistants could proactively adapt audio content based on environmental noise, user activity, or even biometric feedback, delivering information at the most opportune and comfortable pace. This moves beyond simple playback to a truly intelligent and responsive audio ecosystem, where sound is not just heard but experienced in a deeply personalized and adaptive manner. The principles of this patent will continue to drive innovation in making digital audio more flexible, efficient, and user-centric.","question":"What are the future developments expected for Systems and Methods for Time-scale Modification of Audio Signals?"}],"topics":["Systems and Methods for Time-scale Modification of Audio Signals","time-scale modification","audio processing patent","audio compression","audio expansion","challenge","scale","modification"],"tech_cluster":null},"seo":{"title":"Systems and Methods for Time-scale Modification of Audio Signals - Patent US-9852734","description":"Discover this groundbreaking patent: Systems and Methods for Time-scale Modification of Audio Signals. Intelligently compress or expand audio without quality loss. Full analysis, applications, and technical details.","keywords":["Systems and Methods for Time-scale Modification of Audio Signals","time-scale modification","audio processing patent","audio compression","audio expansion","digital signal processing","audio waveform analysis","perceptual audio","audio technology","patent US-9852734","G10L"]},"attribution":{"source":"Patentable","source_url":"https://patentable.app","canonical_url":"https://patentable.app/patents/US-9852734","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-9852734","citation_suggestion":"Patentable. \"Systems and methods for time-scale modification of audio signals\" (US-9852734). https://patentable.app/patents/US-9852734","copyright_holder":"Nomic Interactive Technology LLC"},"links":{"html":"https://patentable.app/patents/US-9852734","json":"https://patentable.app/api/llm-context/US-9852734","site":"https://patentable.app","llms_txt":"https://patentable.app/llms.txt"},"generated_at":"2026-06-06T04:56:03.877Z"}