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2025​​Activity reportProject-TeamEMERAUDE​​​‌

RNSR: 202224250T
  • Research center‌ Inria Lyon Centre
  • In‌​‌ partnership with:Institut national​​ des sciences appliquées de​​​‌ Lyon, Générateur de Ressources‌ et d’Activités Musicales Exploratoires‌​‌
  • Team name: EMbEdded pRogrammable​​ AUDio systEms
  • In collaboration​​​‌ with:Centre d'innovation en‌ télécommunications et intégration de‌​‌ services

Creation of the​​​‌ Project-Team: 2022 March 01​

Each year, Inria research​‌ teams publish an Activity​​ Report presenting their work​​​‌ and results over the​ reporting period. These reports​‌ follow a common structure,​​ with some optional sections​​​‌ depending on the specific​ team. They typically begin​‌ by outlining the overall​​ objectives and research programme,​​​‌ including the main research​ themes, goals, and methodological​‌ approaches. They also describe​​ the application domains targeted​​​‌ by the team, highlighting​ the scientific or societal​‌ contexts in which their​​ work is situated.

The​​​‌ reports then present the​ highlights of the year,​‌ covering major scientific achievements,​​ software developments, or teaching​​​‌ contributions. When relevant, they​ include sections on software,​‌ platforms, and open data,​​ detailing the tools developed​​​‌ and how they are​ shared. A substantial part​‌ is dedicated to new​​ results, where scientific contributions​​​‌ are described in detail,​ often with subsections specifying​‌ participants and associated keywords.​​

Finally, the Activity Report​​​‌ addresses funding, contracts, partnerships,​ and collaborations at various​‌ levels, from industrial agreements​​ to international cooperations. It​​​‌ also covers dissemination and​ teaching activities, such as​‌ participation in scientific events,​​ outreach, and supervision. The​​​‌ document concludes with a​ presentation of scientific production,​‌ including major publications and​​ those produced during the​​​‌ year.

Keywords

Computer Science​ and Digital Science

  • A1.1.2.​‌ Hardware accelerators (GPGPU, FPGA,​​ etc.)
  • A2.2. Compilation
  • A5.7.1.​​​‌ Sound
  • A5.7.2. Music
  • A5.7.5.​ Synthesis
  • A5.9. Signal processing​‌
  • A8.10. Computer arithmetic

Other​​ Research Topics and Application​​​‌ Domains

  • B6.6. Embedded systems​
  • B9.2.1. Music, sound

1​‌ Team members, visitors, external​​ collaborators

Research Scientists

  • Stéphane​​​‌ Letz [GRAME]​
  • Romain Michon [INRIA​‌, ISFP, HDR​​]
  • Anastasia Volkova [​​​‌INRIA, Researcher]​

Faculty Members

  • Tanguy Risset​‌ [Team leader,​​ INSA LYON, Professor​​​‌ Delegation, HDR]​
  • Christine Solnon [INSA​‌ LYON, Professor,​​ HDR]
  • Florent de​​​‌ Dinechin [INSA LYON​, Professor, HDR​‌]

Post-Doctoral Fellows

  • Aurélien​​ Delage [INSA LYON​​​‌, Post-Doctoral Fellow,​ until Aug 2025]​‌
  • Romain Fontaine [INSA​​ LYON, from Sep​​​‌ 2025]
  • Romain Fontaine​ [INSA LYON,​‌ Post-Doctoral Fellow, until​​ Aug 2025]
  • Louis​​​‌ Ledoux [INSA LYON​, Post-Doctoral Fellow]​‌
  • Xiao Peng [INSAVALOR​​, Post-Doctoral Fellow,​​​‌ until Mar 2025]​

PhD Students

  • Bastien Barbe​‌ [INSA LYON]​​
  • Romain Bouarah [INSA​​​‌ LYON]
  • Theo Cantaloube​ [INSA LYON,​‌ from Oct 2025]​​
  • Eric Chen [THALES​​​‌, CIFRE]
  • Oregane​ Desrentes [KALRAY,​‌ CIFRE, until Sep​​ 2025]
  • Adrien Pichon​​​‌ [UBS]
  • Benjamin​ Quiedeville [GRAME,​‌ CIFRE]
  • Florian Rascoussier​​ [IMT ATLANTIQUE]​​​‌
  • Thomas Rushton [INRIA​]
  • Clemens Wegener [​‌INRIA, from Dec​​ 2025]

Technical Staff​​​‌

  • Pierre Cochard [INSA​ LYON]
  • Yann Orlarey​‌ [INRIA, Engineer​​, until Sep 2025​​​‌]

Interns and Apprentices​

  • Mohammad Ali [GRAME​‌, from Sep 2025​​]
  • Elise Bachet [​​​‌INSA LYON, from​ Jun 2025 until Aug​‌ 2025]
  • William Barran​​ [INSA LYON,​​ from Jun 2025 until​​​‌ Aug 2025]
  • Bastien‌ Candela Marty [INSA‌​‌ LYON, Intern,​​ from Jun 2025 until​​​‌ Sep 2025]
  • Theo‌ Cantaloube [INRIA,‌​‌ Intern, from Mar​​ 2025 until Jul 2025​​​‌]
  • Adam Guglielmino [‌INRIA, Intern,‌​‌ from Sep 2025]​​
  • Remi Guillotte [INRIA​​​‌, Intern, from‌ Mar 2025 until Sep‌​‌ 2025]
  • Simon Jacquin​​ [INRIA, Intern​​​‌, from Mar 2025‌ until Sep 2025]‌​‌
  • Leslie Mendoza [INSA​​ LYON, from Apr​​​‌ 2025 until Jul 2025‌]

Administrative Assistants

  • Sylvie‌​‌ Boyer [INRIA]​​
  • Anouchka Ronceray [INRIA​​​‌]
  • Linda Soumari [‌INSA LYON]

External‌​‌ Collaborator

  • Yann Orlarey [​​POLE EMPLOI, from​​​‌ Oct 2025]

2‌ Overall objectives

The goal‌​‌ of the Emeraude project-team​​1 is to combine​​​‌ the multidisciplinary skills of‌ CITI laboratory and Grame-CNCM‌​‌ to foster the development​​ of new programming tools​​​‌ and signal processing techniques‌ for embedded audio systems.‌​‌

Grame-CNCM 2 is a​​ National Center for Musical​​​‌ Creation (CNCM3)‌ hosting a research team‌​‌ specialized in music technology.​​ Grame is also the​​​‌ inventor of the Faust‌ programming language,4 which‌​‌ has met great success​​ in the audio processing​​​‌ community. The skills in‌ compilation, embedded systems, and‌​‌ FPGAs of former Inria​​ Socrate team members, as​​​‌ well as the experience‌ acquired in signal processing‌​‌ is also useful for​​ research in audio and​​​‌ acoustic signal processing.

Embedded‌ programmable audio systems are‌​‌ ubiquitously used in our​​ day-to-day life. Whether it's​​​‌ in our headphones or‌ our car to carry‌​‌ out active noise cancellation,​​ in virtual home assistants​​​‌ (e.g., Alexa, Google Home,‌ etc.), or in modern‌​‌ musical instruments and sound​​ systems, they are everywhere.​​​‌ Real-time audio processing is‌ known to be relatively‌​‌ computationally expensive, but progress​​ in processor architectures in​​​‌ recent years – including‌ microcontrollers, microprocessors, Digital Signal‌​‌ Processors (DSPs), Graphics Processing​​ Unit (GPUs), etc. –​​​‌ have made computing power‌ much more affordable. The‌​‌ generalization of hardware floating​​ point support, and the​​​‌ availablilty of high-level IDEs‌ (Integrated Development Environments) for‌​‌ these new architectures has​​ made them accessible to​​​‌ audio programmers.

Programming embedded‌ audio systems requires specific‌​‌ skills: Digital Signal Processing​​ (DSP), low-level C/C++ programming,​​​‌ and a deep understanding‌ of system architecture. Few‌​‌ engineers (whether they are​​ on the DSP or​​​‌ the programming side) fully‌ master all these domains,‌​‌ and even fewer people​​ in the maker community.​​​‌ The scientific credo of‌ the Emeraude Inria-Insa joint‌​‌ project-team is that Domain​​ Specific Languages (DSLs) are​​​‌ a major technical evolution‌ to enable audio programming‌​‌ on emerging embedded systems​​. There currently exists​​​‌ a few software solutions‌ addressing audio programming such‌​‌ as libpd46 or​​ the SOUL programming language,​​​‌5 but none of‌ them is as generic‌​‌ and as universal as​​ Faust  88, which​​​‌ has been developed at‌ Grame for more than‌​‌ 15 years.

Emeraude uses​​ the Faust programming language​​​‌ as the main platform‌ for experimenting fundamental research.‌​‌ Faust88 is a​​​‌ DSL for real-time audio​ signal processing. A screenshot​‌ of the Faust IDE​​ is shown in Fig.​​​‌ 1. Faust is​ widely used for audio​‌ plugin design (i.e., effects​​ and synthesizers), DSP research,​​​‌ mobile and web app​ design, etc. The success​‌ of Faust is due​​ to its natural data-flow​​​‌ paradigm and on a​ compiler “translating” DSP specifications​‌ written in Faust into​​ a wide range of​​​‌ lower-level languages (e.g., C,​ C++, Rust, Java, LLVM​‌ bitcode, WebAssembly, etc.). Thanks​​ to its highly re-targetable​​​‌ compilation flow, generated DSP​ objects can be embedded​‌ into template programs (wrappers)​​ used to turn a​​​‌ Faust program into a​ specific ready-to-use object (e.g.,​‌ standalone, plug-in, smartphone app,​​ webpage, etc.).

Figure 1

The faust​​​‌ IDE

Figure 1:​ The Faust Web IDE​‌ allowing for the compilation​​ of Faust programs on​​​‌ any machines without having​ to install any particular​‌ tool.

While Faust was​​ not originally designed with​​​‌ embedded audio systems in​ mind, its development took​‌ a significant turn in​​ that direction by targeting​​​‌ an increasingly large number​ of hardware platforms such​‌ as the Teensy6​​ 81 and the ESP-32​​​‌ microcontrollers7 82,​ the SHARC Audio Module​‌ DSP,8 the BELA,​​9 the ELK,10​​​‌ etc. Since Faust can​ generate various types of​‌ standalone programs for Linux,​​ it can also target​​​‌ most embedded Linux systems​ such as the Raspberry​‌ Pi or the BeagleBone​​ for real-time audio signal​​​‌ processing applications. This recent​ availability of Faust compilation​‌ on tiny embedded systems​​ and micro-controllers in particular​​​‌ opens the door to​ the creation of innovative​‌ audio objects. Fig. 2​​ shows the Gramophone, a​​​‌ device designed by the​ Grame team and that​‌ is used in schools​​ to teach basic science​​​‌ concepts to children.

Faust​ is now a well-established​‌ language in the audio​​ DSP community. It is​​​‌ used both in academia​ for teaching in prestigious​‌ institutions such as Stanford​​ University,11 Aalborg University,​​​‌ the University of Michigan,​ and in the industry​‌ (e.g., moForte Inc.,12​​ ExpressiveE). Faust is also​​​‌ used as a prototyping​ tool at Korg, Apple,​‌ Google, Tesla, etc.

Figure 2.a
     
Figure 2.b

The​​ Gramophone instrument

The Gramophone​​​‌ instrument

Figure 2:​ The Gramophone is a​‌ speaker/musical instrument programmable with​​ Faust designed to facilitate​​​‌ the teaching of programming,​ maths, and physics in​‌ middle and high schools.​​ A picture of the​​​‌ board used inside it​ (an ESP-32 microcontroller programmed​‌ directly with a Faust​​ program) can be seen​​​‌ on the right-hand-side of​ the figure.

While embedded​‌ audio systems are already​​ widespread, many limitations remain,​​​‌ especially for real-time applications​ where latency plays a​‌ crucial role. For instance,​​ efficient active control of​​​‌ sound where audio processing​ should be faster than​‌ the propagation of acoustical​​ waves 60, digital​​​‌ musical instruments playability 73​, digital audio effects,​‌ etc. cannot be deployed​​ on lightweight systems. While​​​‌ latency can be potentially​ reduced on “standard” computing​‌ platforms such as personal​​ computers, going under the​​​‌ “one millisecond threshold” is​ usually impossible because of​‌ audio samples buffering induced​​ by software audio drivers.​​

Up to now, most​​​‌ of the research efforts‌ on audio signal processing‌​‌ have been focused on​​ throughput and computing power,​​​‌ leaving aside ultra-low latency‌ as it seemed inaccessible‌​‌ on software platforms. We​​ believe that enabling ultra-low​​​‌ latency for audio application‌ will open a wide‌​‌ range of new domains​​ of application from active​​​‌ acoustic control to new‌ musical instruments (see Fig.‌​‌ 3, “stolen” from​​ the ANR FAST project​​​‌ which started in 2021).‌

Figure 3

overview of applications of‌​‌ low latency audio

Figure​​ 3: Example of​​​‌ target applications for ultra-low‌ latency audio processing on‌​‌ FPGA: module A and​​ module B are two​​​‌ possible “products” based on‌ the same faust2FPGA compilation‌​‌ flow.

FPGAs (Field Programmable​​ Gate Arrays) can help​​​‌ solve current limitations of‌ traditional computing platforms used‌​‌ for musical and artistic​​ applications, especially in terms​​​‌ of audio latency. FPGAs‌ are known for their‌​‌ high computational capabilities 50​​, 89 and their​​​‌ very low-latency performances 106‌. They also provide‌​‌ a large number of​​ GPIOs (General Purpose Inputs​​​‌ and Outputs) which can‌ be exploited to implement‌​‌ modern real-time multi-channel processing​​ algorithms (e.g., sound field​​​‌ capture using a very‌ large number of digital‌​‌ microphones 95, active​​ sound control over a​​​‌ large spatial region 110‌, etc.).

But FPGAs‌​‌ remain extremely complex to​​ program, even with state-of-the-art​​​‌ high-level tools,13 making‌ them largely inaccessible to‌​‌ DSP researchers, musicians, digital​​ artists, and maker communities.​​​‌ There are currently only‌ a few examples of‌​‌ professional FPGA-based real-time audio​​ DSP systems (i.e., Antelope​​​‌ Audio,14 Korora Audio‌15) and in‌​‌ these applications, FPGAs are​​ dedicated to a specific​​​‌ task and not exploited‌ as user-programmable devices.

Emeraude‌​‌ provides a combination of​​ skills that is unique​​​‌ in the world: audio‌ signal processing, compilation, high-level‌​‌ synthesis, computer arithmetic, FPGA​​ programming, acoustics, and embedded​​​‌ system design. This gives‌ a hint on what‌​‌ initially motivated the creation​​ of Emeraude: a compiler​​​‌ from Faust to FPGA‌ as considered in the‌​‌ SyFaLa project16 enabling​​ very low latency processing​​​‌ (less than 100μ‌s, or equivalently‌​‌ between 1 and 5​​ sample latency).

The objective​​​‌ of the research axes‌ described in the next‌​‌ section is to deeply​​ understand and enable new​​​‌ compilation flows for audio‌ signal processing.

3 Research‌​‌ program

The Emeraude project​​ team was officially created​​​‌ in March 2022, though‌ we had been working‌​‌ together for two years​​ prior. At that time,​​​‌ we had decided to‌ organize the team around‌​‌ four research axes:

  1. Ultra-Low​​ Audio Latency on FPGA​​​‌
  2. Advanced Arithmetics for Digital‌ Audio
  3. Digital Audio Signal‌​‌ Processing
  4. Language, Compilation, Deployment​​ and Interfaces for Audio​​​‌ Signal Processing

However, it‌ soon became clear that‌​‌ it was challenging to​​ separate axis 3 (Digital​​​‌ Audio Signal Processing) from‌ axes 1 and 4.‌​‌ Specifically, we have a​​ very active research group​​​‌ in embedded audio systems‌ and FPGAs (with four‌​‌ permanent members) and a​​ strong group in FPGA-based​​​‌ arithmetic (with two permanent‌ members). Within the embedded‌​‌ audio systems group, certain​​​‌ topics are directly concerned​ with Faust , the​‌ language design, the compiler,​​ and the ecosystem. Therefore,​​​‌ the three research focuses​ presented in this document​‌ represent the most effective​​ way to organize the​​​‌ team going forward in​ a balanced manner:

  1. Embedded​‌ Audio Systems and FPGAs​​
  2. Optimization of Arithmetic for​​​‌ FPGAs
  3. The Faust Programming​ Language and its Ecosystem​‌

The recent arrival of​​ Christine Solnon and her​​​‌ students (four people as​ of September 2024) has​‌ been in planning for​​ six months and impacts​​​‌ the team’s structure. Christine​ Solnon is a renowned​‌ researcher in graph algorithms​​ and constraint programming and​​​‌ also has a solid​ background in optimization more​‌ broadly. Many of the​​ problems we study give​​​‌ rise to unique optimization​ challenges. For example, optimizing​‌ the bit width in​​ FIR or IIR filters​​​‌ can be framed as​ a complex integer linear​‌ programming problem, while certain​​ compilation problems addressed in​​​‌ the Faust compiler can​ be expressed as graph​‌ algorithms. More generally, optimization​​ has applications in numerous​​​‌ scientific fields, and we​ intend to establish it​‌ as an important axis​​ for the Emeraude team​​​‌ in the next years.​

3.1 Embedded Audio Systems​‌ and FPGAs

Participants: Florent​​ de Dinechin, Stéphane​​​‌ Letz, Romain Michon​, Yann Orlarey,​‌ Tanguy Risset.

The​​ main objective of this​​​‌ research axis is to​ enable the construction of​‌ audio systems reacting with​​ a latency smaller than​​​‌ (or at least comparable​ to) the duration of​‌ a single audio sample.​​

Low-latency digital audio processing​​​‌ might seem easy: computer​ systems operate at GHz​‌ frequencies whereas audible sound​​ stops at about 20​​​‌ kHz (high-resolution sound processing​ means 192 kHz sample​‌ frequency; CD-quality is 44.1​​ kHz). Concerning sound propagation,​​​‌ electronic data may be​ transmitted at speeds close​‌ to the speed of​​ light while sound travels​​​‌ one million times slower.​ Still, achieving ultra-low latency​‌ remains a huge technical​​ challenge. For the main​​​‌ applications of mass-produced audio​ devices (mostly sound playback​‌ and telephony), a latency​​ of a thousand audio​​​‌ cycles translates to an​ audible delay that is​‌ barely noticeable. However, for​​ the applications envisioned in​​​‌ Emeraude, sound must be​ captured, processed, and emitted​‌ with sub-millisecond latencies.

For​​ that, we need to​​​‌ provide a real compilation​ flow from high-level audio​‌ DSP programs to FPGA​​ IPs. Our proposal is​​​‌ to target a new​ Faust architecture backend for​‌ FPGA-based platforms as depicted​​ in Fig. 4.​​​‌ One of the challenges​ here is the optimization​‌ of the module generated​​ by Faust . The​​​‌ real breakthrough will be​ obtained with the use​‌ of two recent technologies​​ in the Faust compilation​​​‌ workflow: (i)​ High Level Synthesis (HLS)​‌ for compiling Faust programs​​ to VHDL and (​​​‌ii) fixed-point​ support in the code​‌ generated by the Faust​​ compiler, building on the​​​‌ expertise developed at CITI​ around the FloPoCo project​‌ (and studied in next​​ research axis: §3.2​​​‌).

In Audio, sampling​ rate is between 20kHz​‌ and 200kHz. The sampling​​ rate has of course​​ an impact on achievable​​​‌ latency: at 48kHz, one‌ sample arrives every 20‌​‌μs and the​​ achievable latency is limited​​​‌ to one sample because‌ of the audio codec‌​‌ (ADC/DAC) serial protocol. However,​​ what is called “low​​​‌ latency” in current systems‌ is usually close to‌​‌ 1ms (50 samples at​​ 48kHz). Various systems, both​​​‌ in the industry and‌ in academia, have been‌​‌ targeting low audio latency​​ through the use of​​​‌ different hardware solutions. The‌ most affordable ones are‌​‌ embedded Linux systems enhanced​​ with dedicated audio hardware.​​​‌ They run audio signal‌ processing tasks outside of‌​‌ the operating system. The​​ BELA 79 and the​​​‌ Elk,17 which belong‌ to this category, can‌​‌ achieve relatively low latency​​ with buffer sizes as​​​‌ low as 8 samples.‌

Microcontrollers have been used‌​‌ more and more in​​ recent years for sound​​​‌ synthesis and processing because‌ of their increasing power.‌​‌ The Teensy 81 and​​ the ESP32 82 are​​​‌ good examples of such‌ systems. When programmed “bare-metal”‌​‌ (i.e., without an OS),​​ their latency can be​​​‌ similar to that of‌ dedicated/specialized embedded Linux systems‌​‌ (buffer size of 8​​ samples as well).

Digital​​​‌ Signal Processors (DSPs) can‌ target even lower latency‌​‌ with buffer sizes as​​ low as 4 samples​​​‌ and provide tremendous amounts‌ of computational power for‌​‌ signal processing applications. Their​​ programming needs specific developer​​​‌ tools, making them less‌ accessible than the other‌​‌ types of systems mentioned​​ in this section. Additionally,​​​‌ many of them do‌ not provide native support‌​‌ for floating-points computations, further​​ increasing the complexity of​​​‌ their programming. The Analog‌ Devices SHARC Processor18‌​‌ is a leader on​​ the market which can​​​‌ be used as a‌ prototyping system through the‌​‌ SHARC Audio Module. It​​ also provides an official​​​‌ Faust support.

The only‌ way to take audio‌​‌ latency one step further​​ down is to use​​​‌ FPGAs, which is what‌ we plan to do‌​‌ in this research axis.​​

Programming FPGAs is usually​​​‌ done with a hardware‌ description language (VHDL or‌​‌ Verilog). Developing a VHDL​​ IP19 is extremely​​​‌ time consuming. Hence, FPGA‌ programmers have two possibilities:‌​‌ re-using existing IPs and​​ assembling them to compose​​​‌ a circuit solving their‌ problem (as proposed by‌​‌ LABVIEW20), or​​ using High-Level Synthesis to​​​‌ compile a VHDL specification‌ from a higher-level description.‌​‌

High Level Synthesis (HLS)​​ 86 has been referred​​​‌ to for decades as‌ the mean to enable‌​‌ fast and safe circuit​​ design for programmers. However,​​​‌ the design space offered‌ to a hardware designer‌​‌ is so huge that​​ no automatic tool is​​​‌ able to capture all‌ the constraints and come‌​‌ up with the optimal​​ solution (which does not​​​‌ exists anyway since multiple‌ objectives are to be‌​‌ optimized). Many HLS tools​​ have been proposed (i.e.,​​​‌ Pico 96, CatapultC‌ 43, Gaut 99‌​‌, to cite a​​ few) dedicated to specific​​​‌ target application domains. Most‌ of the existing tools‌​‌ start from a high-level​​ representation that is based​​​‌ on a programming language‌ (i.e., C, C++, or‌​‌ Python) which is instrumented​​​‌ using pragmas to guide​ the HLS process.

Using​‌ HLS today still requires​​ very specific skills 68​​​‌ to write a source​ description that is correctly​‌ processed by the tools,​​ but we believe that​​​‌ this technology has reached​ a certain level of​‌ maturity and can now​​ be foreseen as a​​​‌ valuable tool for audio​ designers.

Figure 4

syfala toolflow

Figure​‌ 4: The complete​​ faust2FPGA flow targeted by​​​‌ this research axis. Different​ possible compilation flows for​‌ generating VHDL from a​​ Faust program will be​​​‌ studied.

Another goal is​ to adapt the different​‌ design flows to target​​ high-performance FPGA boards, such​​​‌ as the Genesys ZU​ based on a Zynq​‌ Ultrascale FPGA for instance.​​ These new targets are​​​‌ used for the compute-bound​ studied algorithms. High computing​‌ power implies the introduction​​ of parallelization techniques in​​​‌ the compilation flow (either​ using the HLS process​‌ or by direct VHDL​​ generation from Faust ).​​​‌ This research direction might​ require the parallelization techniques​‌ (Polyhedral tools in particular)​​ developed within Inria in​​​‌ particular (e.g., CASH, Taran,​ CAMUS, CORSE, etc.).

The​‌ main outcome of this​​ research axis, namely the​​​‌ new open-source compilation flow​ from Faust to FPGA​‌ is useful in many​​ contexts: for musicians, acoustic​​​‌ engineers or mechanical vibration​ engineers. In order to​‌ convince these people to​​ use it, we are​​​‌ prototyping a large number​ of audio treatments (e.g.,​‌ filters, reverb effects, etc.)​​ and study the resulting​​​‌ performances – in terms​ of latency and computing​‌ power – depending of​​ the configuration chosen for​​​‌ the flow.

3.2 Optimization​ of Arithmetic for FPGAs​‌

Participants: Florent de Dinechin​​, Anastasia Volkova,​​​‌ Christine Solnon, Yann​ Orlarey.

In this​‌ research axis, Emeraude builds​​ upon the expertise developed​​​‌ in Socrate in application-specific​ arithmetic. Florent de Dinechin​‌ is an expert of​​ computer arithmetics in general​​​‌ (including floating-point 85 and​ alternatives 54, 103​‌) but also in​​ arithmetics for FPGAs, in​​​‌ particular with the FloPoCo​ project 59. Anastasia​‌ Volkova specializes in error​​ analysis and optimization of​​​‌ computer arithmetic, with a​ focus on fixed-point operator​‌ design for digital signal​​ processing and machine learning.​​​‌ Christine Solnon, as an​ expert in graph algorithms​‌ and constraint programming, brings​​ a unique insight into​​​‌ efficient design-space exploration and​ optimization of mathematical models.​‌ Their expertise is helping​​ us addressing challenges related​​​‌ to low-latency digital audio​ by combining complementary approaches:​‌ compilation of digital audio​​ to fixed-point arithmetic, an​​​‌ arithmetic-centered approach to digital​ filter design, and the​‌ scheduling and tiling problems.​​ In these three directions,​​​‌ audio applications fuel research​ that has an impact​‌ well beyond audio.

Audio-to-fixed​​ easier than float-to-fixed

In​​​‌ audio processing, we know​ that the inputs and​‌ outputs are fixed-point data,​​ and we also have​​​‌ a lot of domain​ knowledge about audio physics.​‌ This gives serious hope​​ that Faust audio can​​​‌ be compiled directly to​ fixed-point. This is a​‌ requirement for FPGAs, but​​ it will also reduce​​​‌ the latency and power​ consumption on software targets​‌ if we can use​​ their integer units. It​​ will also enable the​​​‌ compilation of Faust to‌ ultra-low-power microcontrollers without floating-point‌​‌ hardware.

“Domain-specific” is the​​ key word here making​​​‌ us confident that a‌ problem that is generally‌​‌ intractable (float-to-fixed conversion) can​​ be addressed with little​​​‌ or no modification to‌ a Faust program. The‌​‌ challenge here is to​​ keep this additional work​​​‌ so high-level and sound-related‌ that it is not‌​‌ a burden for a​​ musician or a sound​​​‌ engineer. A central objective‌ is that Faust programmers‌​‌ should not need to​​ become fixed-point experts. They​​​‌ should actually not be‌ anymore aware of the‌​‌ underlying arithmetic than they​​ currently are with floating-point.​​​‌ Being high-level is a‌ key reason for the‌​‌ success of Faust .​​

Automated error analysis for​​​‌ hardware computing just right‌

The main issue is‌​‌ to understand how arithmetic​​ errors propagate, are amplified,​​​‌ are accumulated, etc. in‌ a computation and in‌​‌ a circuit. This is​​ called error analysis.​​​‌ Then a general technique‌ 57 is to add‌​‌ enough bits to the​​ right of internal fixed-point​​​‌ formats so that errors‌ accumulate in these bits‌​‌ and the overall error​​ accumulation does not hinder​​​‌ the final quality of‌ the result. Error analysis‌​‌ is also managed by​​ a worst-case combination, but​​​‌ here there is nothing‌ implicit or hidden. This‌​‌ is therefore a comparatively​​ well understood problem, and​​​‌ there is no reason‌ to believe it cannot‌​‌ be fully automated in​​ a compiler that is​​​‌ already able to derive‌ the format information, building‌​‌ on the experience accumulated​​ when designing complex FloPoCo​​​‌ operators 56, 55‌, 47, 102‌​‌, 109.

Digital​​ filters as arithmetic objects​​​‌

Digital filters are essential‌ components of everyday electronics‌​‌ like radios, mobile phones,​​ etc., but also in​​​‌ audio systems of course.‌ Their design is a‌​‌ core topic in digital​​ signal processing and control​​​‌ theory, one that has‌ received significant research interest‌​‌ for the better part​​ of the last half​​​‌ century. A lot of‌ effort has gone into‌​‌ constructing flexible filter design​​ methods. For designing software-based​​​‌ digital filters with floating-point‌ coefficients, there are many‌​‌ powerful approaches that are​​ relatively easy to use​​​‌ by the filter designer‌ (all the more as‌​‌ they rely on over-dimensioned​​ floating-point operators). When designing​​​‌ hardware, things are not‌ that simple for several‌​‌ reasons:

  • algorithms developed for​​ software-implemented filters cannot be​​​‌ transferred directly to hardware:‌ what is a constraint‌​‌ in software (e.g., “use​​ a 32-bit fixed-point format”)​​​‌ becomes a degree of‌ freedom in hardware design‌​‌ (“What is the smallest​​ fixed-point format that can​​​‌ be used?”);
  • another degree‌ of freedom comes from‌​‌ different available realization techniques​​ to implement the arithmetic​​​‌ itself, for instance the‌ construction of multipliers by‌​‌ constants.

A consequence is​​ that popular tools, such​​​‌ as the popular fdatool‌ (filter design and analysis‌​‌ tool) from Matlab's Signal​​ Processing toolbox, offer a​​​‌ complex interface, requiring a‌ tedious hand-tuning process, and‌​‌ expect some domain expertise.​​ Such tools input a​​​‌ frequency response, and decompose‌ the filter implementation problem‌​‌ in three steps: 1/​​​‌ the filter design (FD)​ step consists in finding​‌ a filter with ideal​​ (high precision) coefficients that​​​‌ adheres to the frequency​ response; 2/ the quantization​‌ (Q) step converts the​​ obtained coefficients to hardware-friendly​​​‌ fixed-point values ; 3/​ the implementation (I) step​‌ generates a valid hardware​​ description (e.g., a VHDL​​​‌ or Verilog description) using​ the quantized coefficients.

The​‌ objective of this research​​ axis is to offer​​​‌ an optimal solution to​ the global FD +​‌ Q + I problem.​​ Optimal techniques exist for​​​‌ each of the FD,​ Q and I steps​‌ in isolation. The combination​​ of the FD &​​​‌ Q steps have been​ studied since the 1960's​‌  69, and can​​ even be regarded as​​​‌ solved for certain practical​ instances of fixed-point Finite​‌ Impulse Response (FIR) design​​ 70. A large​​​‌ body of work also​ exists for the I​‌ step, with recent optimal​​ approaches 40, 71​​​‌, 72. However,​ these approaches are only​‌ optimal for a given​​ set of coefficients, and​​​‌ therefore strongly depend on​ the FD and Q​‌ steps.

Arithmetic-oriented scheduling and​​ tiling for low-latency audio​​​‌

Finally, we also want​ to formally insert arithmetic​‌ considerations in the global​​ problem of distributing a​​​‌ very heavy computation between​ space (we have up​‌ to several thousands multipliers​​ in an FPGA, and​​​‌ many more if we​ multiply by a constant)​‌ and time (we have​​ thousands of FPGA cycles​​​‌ within one audio cycle).​ These are well-researched compilation​‌ issues, called the scheduling​​ and tiling problems. There​​​‌ is local expertise in​ Lyon (in particular in​‌ the CASH team and​​ its spin-off XtremLogic21​​​‌) who have worked​ on these problems for​‌ FPGAs. However, scheduling and​​ tiling techniques so far​​​‌ consider each operation as​ having a standard, constant​‌ cost (e.g., multiplication costs​​ cm and has​​​‌ latency tm,​ addition costs ca​‌ in space and t​​a in time). This​​​‌ is a very crude​ simplification if one attempts​‌ to optimize each operator,​​ think for multiplications by​​​‌ constant for instance. The​ availability of many audio-related​‌ case studies in Emeraude​​ will allow us (hopefully​​​‌ in collaboration with CASH)​ to develop arithmetic-aware scheduling​‌ and tiling techniques that​​ will eventually prove useful​​​‌ well beyond the world​ of digital audio.

Towards​‌ provably optimal solutions

Recent​​ arrivals of Christine Solnon​​​‌ and Anastasia Volkova into​ the team bring more​‌ focus into the optimization.​​ Given a mathematical object​​​‌ to implement in hardware​ (for instance the 2D​‌ norm x2+​​y2), the​​​‌ standard practice so far​ has been to 1/​‌ define a family of​​ hardware algorithms parameterized by​​​‌ architectural parameters (typically the​ number of bits for​‌ the intermediate results), 2/​​ express the constraints for​​​‌ a given solution to​ be acceptable (typically that​‌ the hardware should provide​​ a faithful or correct​​​‌ rounding of the exact​ result), and 3/ finally​‌ define a heuristic that​​ provides the parameters for​​​‌ an acceptable solution with​ good performance (performance being,​‌ for instance, area or​​ delay of the hardware).​​ The shift is to​​​‌ replace the heuristic in‌ the third step with‌​‌ a mathematical model that​​ can be solved with​​​‌ standard solvers (ILP, SAT‌ or CP) to provide‌​‌ hardware operators with provably​​ optimal performance. This approach​​​‌ was pioneered by Martin‌ Kumm about ten years‌​‌ ago, and used in​​ Emeraude recently for squarers​​​‌ 44, elementary function‌ evaluators 53, and‌​‌ digital filters 108.​​ In the coming years​​​‌ we will apply this‌ approach more broadly to‌​‌ operators for digital signal​​ processing (in collaboration with​​​‌ axes 1 and 3)‌ and artificial intelligence accelerators‌​‌ (in collaboration with other​​ members of the PEPR​​​‌ IA). We will also‌ use optimization for core‌​‌ classical arithmetic problems such​​ as function approximation and​​​‌ evaluation, compressor tree synthesis,‌ or word-length optimization, again‌​‌ in collaboration with Axis​​ 3.

3.3 The Faust​​​‌ Programming Language and its‌ Ecosystem

Participants: Florent de‌​‌ Dinechin, Stéphane Letz​​, Romain Michon,​​​‌ Yann Orlarey, Tanguy‌ Risset, Christine Solnon‌​‌.

Audio signal processing​​ is an applied field​​​‌ where each result, algorithm,‌ method, or tool ends‌​‌ up being validated by​​ the human ear. This​​​‌ validation requires efficient tools‌ to rapidly prototype audio‌​‌ signal processing algorithms. For​​ many years, languages and​​​‌ tools for audio DSP‌ have been developed by‌​‌ researchers to ease the​​ implementation and the deployment​​​‌ of new audio processing‌ algorithms. The Faust programming‌​‌ language and environment were​​ invented in that context​​​‌ at Grame-CNCM. Emeraude continues‌ to bring new developments‌​‌ around these tools.

The​​ Faust language and its​​​‌ compiler

A large part‌ of Emeraude's research results‌​‌ is visible thanks to​​ the Faust ecosystem development.​​​‌ Faust has gained an‌ international recognition, especially since‌​‌ it is used for​​ teaching at Stanford University​​​‌ (in the context of‌ courses on signal processing,‌​‌ physical interaction design, etc.)​​ and for developing new​​​‌ audio plugins 83.‌ The efforts needed to‌​‌ keep Faust as the​​ most efficient language for​​​‌ real-time audio processing involve‌ research in: language design,‌​‌ compiler design, and development​​ of DSP libraries.

One​​​‌ of the reason of‌ Faust 's success is‌​‌ that it is both​​ a language and an​​​‌ environment for audio signal‌ processing. The Faust compiler‌​‌ typically generates high-level codes​​ (in languages such as​​​‌ C, C++, etc.), following‌ every compiler's goal: providing‌​‌ better code than manually​​ written code. For that,​​​‌ it has to stick‌ to the most recent‌​‌ compiler technologies and processors​​ evolutions 77. For​​​‌ instance, a back-end for‌ WebAssembly was recently added‌​‌ to the Faust compiler​​ 76. An important​​​‌ deployment step was the‌ embedding of the Faust‌​‌ compiler in a web​​ browser 75 which makes​​​‌ it easily accessible on‌ all computers.

Faust language‌​‌ design research in Emeraude​​

The current design of​​​‌ Faust , inspired by‌ lambda-calculus, combinatory logic and‌​‌ John Backus’ functional programming,​​ has to be extended​​​‌ to face new challenges,‌ in particular multi-dimensional and‌​‌ multi-rate signals and linear​​ algebra.

Faust allows for​​​‌ the description of synchronous‌ mono-rate scalar DSP computations.‌​‌ This is sufficient to​​​‌ implement most time-domain algorithms​ such as filters, oscillators,​‌ waveguides, etc. However, this​​ makes the implementation of​​​‌ frequency-domain algorithms (i.e. based​ on FFT) very inefficient,​‌ not to say impossible.​​ One of our goals​​​‌ is to extend the​ language to enable multi-rate​‌ as well as vector​​ computations. While we already​​​‌ have a working prototype​ for this, some challenges​‌ have yet to be​​ overcome.

Along the lines​​​‌ of the previous point,​ Faust currently doesn't provide​‌ any support for efficient​​ matrix operations and more​​​‌ generally linear algebra. This​ prevents the implementation of​‌ some classes of DSP​​ algorithms such as Finite-Difference​​​‌ Time-Domain (FDTD) method for​ physical modeling. The skills​‌ of former Socrate members​​ on seminal Alpha language​​​‌ 58 and polyhedral optimization​ are very useful here.​‌

Support for the main​​ target programming languages in​​​‌ Faust is essential. Recently​ added languages (WebAssembly, Rust,​‌ and CMajor) have opened​​ many new opportunities. The​​​‌ FPGA target, studied in​ §3.1, introduces​‌ new challenges such as​​ the ability to use​​​‌ fixed-point arithmetic or the​ use of HLS for​‌ targeting hardware platforms (e.g.,​​ VHDL, Verilog, etc.). Other​​​‌ “exotic” architectures such as​ GPUs or MPSoCs should​‌ be studied for compute-bound​​ algorithms.

Musicians have to​​​‌ deal with a large​ variety of operating systems,​‌ software environments and hardware​​ architectures. Faust is designed​​​‌ to favor an easy​ deployment of DSP programs​‌ on all these targets​​ by making a clear​​​‌ separation between computation itself,​ as described by the​‌ program code, and how​​ this computation should be​​​‌ related to the external​ world. This relation (with​‌ audio drivers, GUIs, sensors,​​ etc.) is described in​​​‌ specific architecture files62​. Architecture files concern​‌ both hardware (i.e., audio​​ interfaces/sound cards) as well​​​‌ as software control interfaces​ (e.g., GUI, OSC,22​‌ MIDI), new luthieries (e.g.,​​ SmartFaust, Gramophone), Web platforms​​​‌ (Web audio Plugin), etc.​ One of the goal​‌ of the work of​​ Emeraude on Faust is​​​‌ to ease the programming​ of these audio systems.​‌

Faust ecosystem and DSP​​ libraries

Faust users are​​​‌ very attached to its​ ecosystem, including native applications,​‌ online and “embedded” audio​​ applications, Just In Time​​​‌ (JIT) compiler, etc. Recent​ developments include a JIT​‌ Faust compiler on the​​ Web, a JIT compiler​​​‌ in the Max/MSP environment,​ tools to find the​‌ best compilation parameters and​​ ease compilation for multiple​​​‌ CPUs. This is constantly​ evolving to answer to​‌ users' demand.

The Faust​​ DSP libraries currently implement​​​‌ hundreds of functions/objects ranging​ from simple oscillators and​‌ filters to advanced filter​​ architectures, physical models, and​​​‌ complete ready-to-use audio plugins.​ These libraries are at​‌ the heart of Faust​​ 's success and international​​​‌ position. Julius Smith23​ (Stanford professor) is one​‌ of the most respected​​ figures in the field​​​‌ of audio DSP and​ one of the main​‌ contributors to the Faust​​ libraries. One of the​​​‌ ambitions of the Emeraude​ team is to maintain​‌ and extend this tool​​ to make it as​​​‌ exhaustive and as universal​ as possible. Along these​‌ lines, new developments made​​ to the language presented​​ above (e.g., multi-rate, linear​​​‌ algebra, etc.) should be‌ ported to the libraries.‌​‌ Finally, dedicated libraries targeting​​ specific hardware platforms (e.g.,​​​‌ microcontrollers, FPGAs) should be‌ made available too.

Machine‌​‌ learning for digital signal​​ processing

Machine learning and​​​‌ deep learning in particular,‌ are playing an increasingly‌​‌ important role in the​​ field of audio DSP.​​​‌ Researchers are revisiting the‌ algorithmic techniques of signal‌​‌ synthesis and processing in​​ the light of machine​​​‌ learning, for instance for‌ speech processing 64.‌​‌ Recent breakthroughs such as​​ the use of machine​​​‌ learning use in the‌ context of Differentiable Digital‌​‌ Signal Processing (DDSP) 61​​ demonstrate its power. The​​​‌ extension of Faust applications‌ to artificial intelligence began‌​‌ with the PhD work​​ of Benjamin Quiédeville, expanding​​​‌ the scope of Faust‌ in the field of‌​‌ AI. The objective is​​ the introduction of an​​​‌ autodifferentiation primitive for Faust‌ programs, aimed at machine‌​‌ learning applications. The development​​ of new backends is​​​‌ planned, particularly for MLIR,‌ MOJO, and GPUs, as‌​‌ well as support for​​ emerging architectures, especially in​​​‌ the context of game‌ engines and VR/AR applications.‌​‌

Embedded systems for audio​​ processing

As Emeraude's name​​​‌ suggests it, the implementation‌ of audio Digital Signal‌​‌ Processing on embedded hardware​​ is at the heart​​​‌ of the project. We‌ naturally rely on the‌​‌ Faust language for these​​ implementations. The skills of​​​‌ Emeraude members in compilation‌ and embedded systems are‌​‌ used to add new​​ embedded target for audio​​​‌ processing, in particular FPGAs,‌ as explained previously. This‌​‌ action is a mix​​ of research and engineering​​​‌ work, it should be‌ very useful for the‌​‌ dissemination of audio processing​​ programming.

Haptics is a​​​‌ huge topic, especially in‌ the field of New‌​‌ Interfaces for Musical Expression​​ (NIME), which has been​​​‌ studied for many years‌ 51, 105.‌​‌ It has always been​​ tightly coupled to physical​​​‌ modeling because this sound‌ synthesis technique provides natural‌​‌ connections to the physical​​ world. A big part​​​‌ of the challenge is‌ technological because haptics require‌​‌ ultra low-latency and high​​ sampling resolution in order​​​‌ to be accurate. This‌ is at the heart‌​‌ of Emeraude 's goals.​​

Virtual and Augmented Reality​​​‌ (VR/AR) is not limited‌ to immersive 3D graphics,‌​‌ sound also has an​​ important role to play​​​‌ in that emerging field.‌ Lots of work have‌​‌ been done around using​​ VR environments as a​​​‌ creative tool for audio‌ 74, 49,‌​‌ 48. While many​​ VR-based musical instruments have​​​‌ been created in the‌ past 97, little‌​‌ work has been done​​ around implementing interfaces specifically​​​‌ targeting VR/AR audio environments,‌ especially in the context‌​‌ of 3D sound. This​​ is something that we​​​‌ plan to explore as‌ part of Emeraude.

Finally,‌​‌ beside ergonomic and HCI​​ aspects, the design of​​​‌ musical interfaces is impacted‌ by various kinds of‌​‌ technical limitations that we​​ plan to address as​​​‌ part of Emeraude. First,‌ just like for real-time‌​‌ audio processing, latency plays​​ a crucial role in​​​‌ this context. Similarly, the‌ “time resolution” (e.g., the‌​‌ sampling rate of the​​​‌ interfaces) can have a​ huge impact, especially when​‌ targeting specific kinds of​​ instruments such as drums.​​​‌ Finally, the “spatial resolution”​ (e.g., the number of​‌ sensor points per squared​​ centimeters on a tabletop​​​‌ interface) also impacts its​ quality. In this context,​‌ we would like to​​ develop an embedded, high-resolution,​​​‌ high-sampling-rate, multi-touch multi-dimensional (X​ and Y + pressure)​‌ interface/instrument leveraging the development​​ carried out in the​​​‌ previous axes. This work​ would be followed by​‌ a user study to​​ measure the impact of​​​‌ this type of advanced​ system on perception.

4​‌ Application domains

Emeraude aims​​ at being a world​​​‌ leading research team on​ audio systems, carrying out​‌ fundamental research. However, Emeraude's​​ research topics do belong​​​‌ to the spectrum of​ applied research. Hence, discoveries​‌ made in the context​​ of Emeraude should be​​​‌ illustrated with experimental prototypes​ and demonstrations. Here is​‌ a brief overview of​​ various application fields where​​​‌ research developed in Emeraude​ could be applied.

4.1​‌ Spatial active noise control​​

Noise control is a​​​‌ major issue in many​ industries: transport, construction, multimedia,​‌ etc. Active noise control​​ techniques can help to​​​‌ partially remedy this problem.​

However, the implementation of​‌ such approaches requires several​​ microphones and loudspeakers, whose​​​‌ signal processing must be​ done in real-time and​‌ faster than the propagation​​ time of the acoustical​​​‌ waves. In these applications,​ FPGA solutions are therefore​‌ the most suitable way​​ to program such devices,​​​‌ and the flow proposed​ in §3.1 is​‌ of great interest in​​ this context.

For instance,​​​‌ it could be used​ for single-channel controllers: a​‌ theme already developed, for​​ example for active headsets​​​‌ 41. In that​ case, low latency allows​‌ for fully digital feedback​​ control to be implemented.​​​‌ More generally, the feedback​ control previously limited to​‌ small, non-modular spaces, can​​ be extended to a​​​‌ variety of situations, given​ the flexibility and adaptability​‌ of digital filters. Another​​ extension would be the​​​‌ implementation of multichannel controllers:​ experiments have already been​‌ performed for the implementation​​ of multichannel feedforward FPGA​​​‌ controllers with the development​ of architectures adapted from​‌ the FXLMS reference algorithm​​ 98. This allows​​​‌ developments to be considered​ in a real-world context.​‌

4.2 Virtual acoustics/spatial audio​​

Controlling noise is only​​​‌ one of the applications​ of the aforementioned system.​‌ There is a rather​​ strong interest at the​​​‌ moment for the replication​ of virtual acoustic spaces​‌ for “immersive experiences.” Stanford​​ is currently discussing the​​​‌ possibility of integrating a​ virtual acoustics component to​‌ the replica of the​​ Chauvet cave in Ardèche​​​‌ with the scientific director​ of the Chauvet cave​‌ program. The idea would​​ be to make acoustic​​​‌ measurements of the real​ cave and to set​‌ up a system which,​​ by capturing the position​​​‌ of the visitor's head,​ would allow him to​‌ hear the guide's voice​​ as if he were​​​‌ in the real cave​ (in 3D). Emeraude (Romain​‌ Michon) is part of​​ the think-tank on this​​​‌ topic.

Research around Virtual​ Reality (VR) and Augmented​‌ Reality (AR) systems is​​ very active today: immersive/augmented​​ experience: audio guides, AR​​​‌ headsets implementing binaural rendering,‌ augmented acoustics experience, with‌​‌ a strong focus on​​ the development of systems​​​‌ supporting binaural rendering. Emeraude‌ will be active in‌​‌ this domain too.

4.3​​ Industrial acoustics

Industrial developments​​​‌ of active noise control‌ systems have so far‌​‌ been limited either to​​ small spaces (e.g., active​​​‌ headsets, low-frequency ducts for‌ aeraulic systems, etc.) or‌​‌ to noises of a​​ particular nature (e.g., periodic​​​‌ noise from propeller aircraft,‌ land vehicle engines, etc.).‌​‌ Our FPGA-based solution, which​​ offers low latency and​​​‌ high computational capabilities, would‌ enable the extension of‌​‌ controlled volumes, and the​​ possibility of active noise​​​‌ control over any kind‌ of noise. This includes‌​‌ for instance the automotive​​ sectors where the reduction​​​‌ of road noise inside‌ the passenger compartment is‌​‌ a big concern 67​​.

Another application would​​​‌ be the active treatment‌ of boundary conditions with‌​‌ the realisation of “smart​​ surfaces” for absorption 78​​​‌, 42, or‌ vibro-acoustic isolation 80,‌​‌ 65, 111.​​ The development of active​​​‌ material is based on‌ multi-channel control systems combining‌​‌ global control and decentralized​​ feedback systems. The use​​​‌ of FPGAs would enable‌ them to be applied‌​‌ on a large scale,​​ in buildings and also​​​‌ in transport systems (e.g.,‌ aircraft, turbojet nacelles, etc.).‌​‌ The LMFA is developing​​ both the experimental means​​​‌ (i.e., MATISSE and CAIMAN‌ test benches, ECL-B3 test‌​‌ bench from Equipex PHARE,​​ etc.), and the numerical​​​‌ codes of acoustic propagation‌ 52, 101,‌​‌ within the framework of​​ a strong partnership with​​​‌ Safran Aircraft Engines (ANR‌ ADOPSYS and ARENA industrial‌​‌ chairs). The development of​​ a high-level compiler dedicated​​​‌ to Acoustic Digital Signal‌ processing on FPGAs is‌​‌ therefore of high interest​​ for many researchers in​​​‌ acoustic for numerous industrial‌ applications.

4.4 Medicine/sonification

There‌​‌ is a trend in​​ the medical world towards​​​‌ the “sonification” of medical‌ data such as EEGs,‌​‌ etc. The idea behind​​ this concept is that​​​‌ our brain can process‌ time series much faster‌​‌ and with much more​​ precision if they are​​​‌ “encoded” as sound than‌ if they are plotted‌​‌ on a graph. For​​ instance, trained doctors can​​​‌ spot patterns which are‌ characteristics of seizures in‌​‌ EEGs just by listening​​ to their sonified version,​​​‌ which would not be‌ possible just by looking‌​‌ at the corresponding plot.​​ In that context, a​​​‌ “brain stethoscope” which basically‌ sonifies the output signal‌​‌ of an EEG cap​​ in real-time is currently​​​‌ being developed and will‌ be released soon.24‌​‌ This type of development​​ will be greatly simplified​​​‌ by the tools developed‌ by Emeraude.

4.5 Low-latency‌​‌ audio effect processors and​​ synthesizers

Custom low-latency synthesizers​​​‌ and sound processors (i.e.,‌ audio effects) are currently‌​‌ mostly out of reach​​ to people in the​​​‌ audio and music technology‌ communities. Indeed, the high-level‌​‌ programming environments used by​​ these groups (e.g., Max/MSP,​​​‌ SuperCollider, etc.) cannot be‌ used to program embedded‌​‌ audio platforms targeting low-latency​​ applications. Instead, they were​​​‌ meant to be executed‌ on personal computers which‌​‌ have potentially way more​​​‌ audio latency than embedded​ systems. Providing people in​‌ these communities with a​​ tool (from §3.1​​​‌) solving this problem​ would completely revolutionize the​‌ way they approach their​​ tool chain.

4.6 Digital​​​‌ luthiery

Since the 1980s,​ digital equipment has become​‌ deeply embedded in all​​ parts of the popular​​​‌ music production, distribution and​ consumption chain. In a​‌ market whose worldwide sales​​ exceed 15 billion euros,​​​‌ digital instruments (also known​ as “Digital Luthiery”) are​‌ only the latest chapter​​ in the long history​​​‌ of music technology. Digital​ instruments have sped up​‌ the evolution process by​​ increasing accessibility of musical​​​‌ equipment to practitioners, especially​ young people, who can​‌ now achieve at home​​ with inexpensive devices the​​​‌ kind of professional-calibre sounds​ that previously would have​‌ needed a large recording​​ studio. Modern musical instruments​​​‌ are all in need​ of some form of​‌ embedded audio processing in​​ which Emeraude could play​​​‌ a central role.

Grame​ is actively contributing to​‌ this effort by creating​​ tools easily accessible to​​​‌ the maker community: open​ platform to design musical​‌ instruments, educational tools, etc.​​

5 Highlights of the​​​‌ year

5.1 JIMLAC-25

Figure 5

Participants​ of the 2025 Linux​‌ Audio Conference.

Figure 5​​: 2025 Linux Audio​​​‌ Conference participants.

In 2025,​ Emeraude organized JIMLAC-2525​‌ in collaboration with GRAME-CNCM.​​ This major scientific and​​​‌ artistic event combined the​ 2025 Journées d'Informatique Musicale​‌ (JIM) and the 2025​​ Linux Audio Conference (LAC)​​​‌ and it took place​ at INSA Lyon on​‌ June 23-28, 2025. With​​ a total of six​​​‌ days of scientific conference​ and six concerts, it​‌ hosted around 200 participants.​​ Romain Michon was the​​​‌ general chair of JIMLAC-25​ and Stéphane Letz the​‌ technical chair.

5.2 Awards​​

Orégane Desrentes and Florent​​​‌ de Dinechin received a​ Best Paper Award for​‌ their article 12 at​​ the the ARITH 2025​​​‌ conference. Florent de Dinechin​ received the Best Presentation​‌ Award at the 2nd​​ FPGA Developers' Forum meeting​​​‌ organized at CERN.

6​ Latest software developments, platforms,​‌ open data

6.1 Latest​​ software developments

6.1.1 FloPoCo​​​‌

  • Name:
    Floating-Point Cores, but​ not only
  • Keyword:
    Synthesizable​‌ VHDL generator
  • Functional Description:​​
    The purpose of the​​​‌ open-source FloPoCo project is​ to explore the many​‌ ways in which the​​ flexibility of the FPGA​​​‌ target can be exploited​ in the arithmetic realm.​‌
  • URL:
  • Contact:
    Florent​​ De Dinechin
  • Participant:
    2​​​‌ anonymous participants
  • Partners:
    ENS​ Lyon, Insa de Lyon,​‌ Inria, Fulda University of​​ Applied Science

6.1.2 Syfala​​​‌

  • Name:
    Low-Latency Synthesizer on​ FPGA
  • Keywords:
    FPGA, Compilers,​‌ High-level synthesis, Audio signal​​ processing
  • Functional Description:

    The​​​‌ goal of Syfala is​ to design an FPGA-based​‌ platform for multichannel ultra-low-latency​​ audio Digital Signal Processing​​​‌ programmable at a high-level​ with Faust and C++​‌ and usable for various​​ applications ranging from sound​​​‌ synthesis and processing to​ active sound control and​‌ artificial sound field/room acoustics.​​

    A series of tools​​​‌ are currently being developed​ around SyFaLa. SyFaLa is​‌ freely accessible on GitHub:​​ https://github.com/inria-emeraude/syfala.

  • URL:
  • Contact:​​​‌
    Tanguy Risset

6.1.3 FAUST​

  • Name:
    Functional Audio Stream​‌
  • Keywords:
    Audio, Functional programming​​
  • Functional Description:

    The core​​ component of Faust is​​​‌ its compiler. It allows‌ to "translate" any Faust‌​‌ digital signal processing (DSP)​​ specification to a wide​​​‌ range of non-domain specific‌ languages such as C++,‌​‌ C, LLVM bit code,​​ WebAssembly, Rust, etc. In​​​‌ this regard, Faust can‌ be seen as an‌​‌ alternative to C++ but​​ is much simpler and​​​‌ intuitive to learn.

    Thanks‌ to a wrapping system‌​‌ called "architectures," codes generated​​ by Faust can be​​​‌ easily compiled into a‌ wide variety of objects‌​‌ ranging from audio plug-ins​​ to standalone applications or​​​‌ smartphone and web apps,‌ etc.

  • URL:
  • Contact:‌​‌
    Yann Orlarey
  • Partners:
    GRAME,​​ Insa de Lyon, Inria​​​‌

7 New results

7.1‌ Interaction within the team‌​‌

During 2025, Emeraude team​​ members strengthened collaborations between​​​‌ the team’s research axes‌ (embedded audio and FPGA,‌​‌ computer arithmetic and optimisation),​​ resulting in several concrete​​​‌ outcomes. First, the computer‌ arithmetic and optimisation groups‌​‌ initiated joint work on​​ new constraint programming models​​​‌ for designing efficient hardware‌ architectures for multiplication circuits,‌​‌ which led to the​​ publication of two papers​​​‌11, 9.‌ This collaboration notably included‌​‌ the supervision by Christine​​ Solnon and Anastasia Volkova​​​‌ of Théo Cantaloube (a‌ master internship that evolved‌​‌ into a PhD thesis),​​ focusing on the intersection​​​‌ of the two research‌ axes. Second, Romain Michon‌​‌ and Anastasia Volkova collaborated​​ through the co-supervision of​​​‌ a master student on‌ the optimisation of sigma-delta‌​‌ DAC circuits, with results​​ published at the DASIP​​​‌ 2026 conference. Third, Pierre‌ Cochard and Louis Ledoux‌​‌ made significant progress 34​​, 34, 2​​​‌ toward proposing a complete‌ MLIR flow for simple‌​‌ Faust programs leveraging the​​ FLoPoCo tool, thereby further​​​‌ reinforcing the links between‌ arithmetic tools developed by‌​‌ the team and signal​​ processing applications. Finally, an​​​‌ article summarizing coupling research‌ result on audio on‌​‌ FPGA and hardware design​​ 38 untitled "Frugalité et​​​‌ conception de circuits pour‌ le traitement du signal‌​‌ audio numérique" was published​​ in French in "Revue​​​‌ Francophone d'Informatique et Musique".‌

7.2 Immersive Audio

7.2.1‌​‌ Distributing spatial audio

Participants:​​ Thomas Rushton, Romain​​​‌ Michon, Tanguy Risset‌.

In 21,‌​‌ we introduced a low-cost,​​ open, and scalable platform​​​‌ for distributed spatial audio‌ rendering. It seeked to‌​‌ reduce the financial and​​ technical barriers imposed by​​​‌ conventional immersive audio systems.‌ For that we proposed‌​‌ a distributed architecture in​​ which sound field synthesis​​​‌ algorithms, such as Wave‌ Field Synthesis (WFS) or‌​‌ ambisonics, are parallelized across​​ many inexpensive, networked embedded​​​‌ audio processors (see Fig.‌ 6). We focused‌​‌ on the central technical​​ challenge of achieving sufficiently​​​‌ precise time synchronization between‌ physically separate audio devices‌​‌ to preserve wavefront integrity.​​ To address this, we​​​‌ demonstrated how the IEEE‌ 1588 Precision Time Protocol‌​‌ (PTP), implemented using open-source​​ software on low-cost microcontroller​​​‌ platforms (specifically the Teensy‌ 4.1), can be used‌​‌ both to align audio​​ start times and to​​​‌ continuously correct sampling frequency‌ drift by conditioning each‌​‌ device's audio phase-locked loop.​​ Using an experimental setup​​​‌ with multiple microcontroller-based audio‌ nodes synchronized via a‌​‌ PTP-capable Ethernet switch, we​​​‌ show that without correction,​ clock drift accumulates to​‌ the millisecond range, whereas​​ with PTP-derived sampling frequency​​​‌ conditioning, relative timing errors​ are reduced by three​‌ orders of magnitude to​​ the microsecond level, corresponding​​​‌ to sub-millimeter acoustic discrepancies.​ These results demonstrate that​‌ accurate, synchronous audio reproduction​​ is feasible using inexpensive​​​‌ hardware and standard networking​ infrastructure, validating our approach​‌ as a practical foundation​​ for accessible distributed spatial​​​‌ audio systems and motivating​ future work on scalability,​‌ integration with conventional audio​​ workflows, and more computationally​​​‌ demanding applications such as​ real-time virtual acoustics and​‌ auralization.

Figure 6

Microcontroller-based WFS.

Figure​​ 6: The holophonic​​​‌ effect of primary-source WFS​ is created by applying​‌ appropriate per-loudspeaker delays to​​ a virtual sound source;​​​‌ delays are independent and​ can be computed in​‌ distributed fashion. A server​​ delivers audio and control​​​‌ data to a collection​ of Microcontroller-based signal processors​‌ via an ethernet switch;​​ each microcontroller is informed​​​‌ of its position, and​ the position of the​‌ virtual sound source; microcontroller​​ audio clocks are conditioned​​​‌ via PTP to match​ that of the server.​‌ Synchronicity amongst the group​​ of microcontrollers is necessary​​​‌ to ensure the integrity​ of the synthesised wavefront.​‌

7.2.2 Real-Time Auralization on​​ FPGA

Participants: Rémi Jeunehomme​​​‌, Romain Michon,​ Tanguy Risset, Pierre​‌ Cochard, Stéphane Letz​​.

In 15,​​​‌ we investigated the feasibility​ of real-time auralization on​‌ FPGAs by implementing and​​ evaluating convolution-based artificial reverberation​​​‌ algorithms tailored to FPGA​ architectures. We focused on​‌ the computational challenges posed​​ by long room impulse​​​‌ responses and explored FPGA-based​ solutions as an alternative​‌ to traditional CPU- and​​ GPU-based approaches, motivated by​​​‌ the low latency, parallelism,​ and scalability offered by​‌ reconfigurable hardware. Using Syfala​​ 91 and High-Level Synthesis​​​‌ (HLS), we implemented two​ uniformly partitioned overlap-save convolution​‌ algorithms in C: a​​ time-domain approach (TUPOLS) based​​​‌ on direct convolution, and​ a frequency-domain approach (FUPOLS)​‌ relying on FFT-based circular​​ convolution. We analyzed their​​​‌ architectural suitability, detailing memory​ organization, dataflow pipelining, numerical​‌ format choices, and hardware-specific​​ optimizations required to meet​​​‌ real-time constraints on a​ Xilinx Zynq-based FPGA platform.​‌ We showed that while​​ the time-domain approach is​​​‌ severely limited by latency​ and resource consumption for​‌ impulse responses longer than​​ about one second, the​​​‌ frequency-domain approach achieved real-time​ performance for impulse responses​‌ exceeding ten seconds with​​ acceptable FPGA resource usage.​​​‌ This work opens the​ way to potential large-scale​‌ multichannel auralization applications on​​ FPGA.

7.2.3 Auralization of​​​‌ Paleoacoustics landscapes in Chauvet​ Cave

Participants: Romain Michon​‌.

In 22,​​ we reported on an​​​‌ ongoing interdisciplinary study carried​ out in the context​‌ of the MIRAGES associate​​ team (see §9.1.1​​​‌) aimed at measuring,​ analyzing, and auralizing the​‌ paleoacoustic landscapes of Chauvet​​ Cave (south of France),​​​‌ in order to better​ understand how sound may​‌ have shaped human experiences​​ of the cave during​​​‌ the Upper Paleolithic. We​ described the methodological and​‌ conservation-driven constraints that governed​​ our work, including restricted​​​‌ access to the cave,​ limitations on equipment, and​‌ the fact that measurements​​ were confined to modern​​ walkways, leaving large portions​​​‌ of the cave acoustically‌ undocumented. To address these‌​‌ challenges, we carried out​​ several field campaigns between​​​‌ 2022 and 2024, during‌ which we collected thousands‌​‌ of impulse responses using​​ low-impact measurement protocols based​​​‌ on exponential sine sweeps,‌ omnidirectional and Ambisonic microphones,‌​‌ and portable loudspeakers. We​​ analyzed the resulting data​​​‌ to assess signal quality‌ and extract room-acoustic parameters‌​‌ such as reverberation time,​​ clarity, and definition, showing​​​‌ that despite logistical constraints,‌ most measurements achieved sufficiently‌​‌ high signal-to-noise ratios for​​ reliable analysis. We then​​​‌ discussed how these measurements‌ informed multiple auralization frameworks,‌​‌ including offline convolution, web-based​​ interactive tools, real-time multichannel​​​‌ installations, virtual reality environments,‌ and museum exhibits, each‌​‌ offering different balances between​​ realism, interactivity, and accessibility.​​​‌ This work opens the‌ way to future research‌​‌ directions, including predictive acoustic​​ modeling based on reconstructed​​​‌ Paleolithic geometries, improved material‌ characterization, and expanded interactive‌​‌ auralization platforms, positioning this​​ work as a foundational​​​‌ step toward scientifically grounded‌ yet explicitly speculative reconstructions‌​‌ of prehistoric soundscapes for​​ both research and public​​​‌ engagement.

Figure 7

Measurements in the‌ Chauvet cave.

Figure 7‌​‌: Acoustical measurements in​​ the Salle Hillaire of​​​‌ the Chauvet cave.

7.2.4‌ The Space Bar, an‌​‌ Embedded WFS Sound System​​

Participants: Benjamin Quiedeville,​​​‌ Romain Michon, Pierre‌ Cochard, Stéphane Letz‌​‌.

From November to​​ March, an embedded WFS​​​‌ system was designed and‌ built at the laboratory‌​‌ in the context of​​ an exhibition at the​​​‌ Grenoble INRIA centre, based‌ on an earlier prototype‌​‌ created by the Emeraude​​ team. It leverages the​​​‌ Syfala toolchain 91 developed‌ in the Emeraude team‌​‌ to create an FPGA​​ based, frugal and autonomous​​​‌ device capable of spatialising‌ a high number of‌​‌ audio sources on 32​​ speakers using a Wave​​​‌ Field Synthesis (WFS) 100‌ algorithm and controlled via‌​‌ a standard game controller.​​

The construction involved the​​​‌ building of the casing‌ and the electronic wiring‌​‌ of the FPGA board​​ on special multi channel​​​‌ audio interfaces 90 and‌ the programming of the‌​‌ integrated ARM CPU using​​ a custom Alpine Linux​​​‌ distribution provided by Syfala.‌ The role of this‌​‌ program is to open​​ audio files and send​​​‌ the audio data alongside‌ position control signals to‌​‌ the FPGA. The project​​ involved the publication of​​​‌ an article in the‌ proceeding of the Journées‌​‌ de l'Informatique Musicale conference​​ in Lyon in June​​​‌ 2025.

The composer Frédéric‌ Khan has worked in‌​‌ collaboration with GRAME on​​ the production of a​​​‌ musical piece to be‌ integrated in the device‌​‌ in march 2026, a​​ special version of the​​​‌ control program was implemented‌ to accomodate for composers‌​‌ for enhanced creativity and​​ ease of writting. Additional​​​‌ resources used during the‌ work were 92,‌​‌ 107, 104.​​

7.3 Acceleration of AI​​​‌ inference

7.3.1 HAtorch: Hardware-aware‌ quantization-aware training for CNNs‌​‌

Participants: Bastien Barbe,​​ Romain Bouarah, Anastasia​​​‌ Volkova, Florent de‌ Dinechin.

Popular machine‌​‌ learning frameworks like PyTorch​​ and TensorFlow provide complete​​​‌ toolchains to design, train,‌ quantize and deploy convolutional‌​‌ neural networks (CNNs) on​​​‌ standard hardware (CPUs, GPUs,​ TPUs, NPUs, microcontrollers). However,​‌ custom hardware targets like​​ FPGAs, ASICs and research​​​‌ accelerators typically use non-standard​ number representations as well​‌ as limited operator sets,​​ and are poorly served​​​‌ by current lowering backends.​ Existing backend-driven approaches rigidly​‌ force training to their​​ internal formats and often​​​‌ hide lowering choices behind​ opaque layers and closed​‌ software development kits. Conversely,​​ unconstrained quantization that permits​​​‌ arbitrary numeric and operator​ freedom often produces models​‌ that are difficult to​​ implement efficiently, leading to​​​‌ an hardware gap between​ the trained and deployed​‌ model. This work introduces​​ HATorch, a PyTorch-based hardware-aware​​​‌ training framework that reduces​ this gap by enhancing​‌ quantization with hardware architecture​​ model on the arithmetic​​​‌ operator/format level. HATorch supports​ custom hardware-friendly quantization flows,​‌ exposing lowering decisions for​​ transparent model-hardware co-design. We​​​‌ demonstrate the toolflow on​ the new hardware-friendly multipliers​‌ based on 9.​​

This QAT framework is​​​‌ a shared contribution of​ two PhD students from​‌ the team, Bastien Barbe​​ and Romain Bouarah .​​​‌ The conference paper was​ accepted in 2025 and​‌ will be presented in​​ early 2026 at the​​​‌ 8th AccML Accelerated Machine​ Learning Workshop at the​‌ HiPEAC conference in Krakow.​​

HATorch is also being​​​‌ used for the exploration​ of other type of​‌ neural network architectures, based​​ on Kolmogorov-Arnlodi Networks (KANs)​​​‌ in the preliminary work​ 33.

Figure 8

From training​‌ to deplyment

Figure 8​​: Classical flows for​​​‌ lowering of trained models​ to HW exhibit a​‌ a substantial performance gap.​​ HATorch incorporates HW model​​​‌ into the quantization-aware training​ process in a clear​‌ and transparent way, permitting​​ utilization of unconventional arithmetics​​​‌ and reducing the final​ performance mismatch between software​‌ and hardware

7.3.2 Towards​​ frugality in Natural Language​​​‌ Processing classification models

Participants:​ Anastasia Volkova.

This​‌ line of work is​​ part of a broader​​​‌ collaboration on mixed-precision quantization-aware​ training (QAT) for NLP​‌ models, involving Anastasia Volkova,​​ Cédric Gernigon, Richard Dufour,​​​‌ and Xavier Pillet from​ Nantes University.

The first​‌ contribution, based on 14​​ and to be published​​​‌ at DASIP 2026, investigates​ mixed-precision quantization for BERT​‌ inference, with the goal​​ of reducing memory and​​​‌ computational costs while preserving​ accuracy. Unlike most prior​‌ work, the study jointly​​ considers mixed-precision quantization of​​​‌ both weights and activations​ (see Fig. 9),​‌ integrates knowledge distillation into​​ the quantization pipeline, and​​​‌ evaluates the impact of​ quantizing the embedding layer​‌ beyond token weights. Experiments​​ on the SQuAD and​​​‌ GLUE benchmarks show that​ mixed-precision configurations achieve substantial​‌ reductions in resource usage​​ without degrading performance.

Figure 9

Example​​​‌ of a mixed-precision configuration​ learned by AdaQAT

Figure​‌ 9: Example of​​ learned mixed-precision formats for​​​‌ BERT layers on the​ CoLa dataset. Weights (blue,​‌ on top) and activations​​ (red, in the bottom)​​​‌ have variable precisions learned​ during trianing.

The second​‌ contribution (to be presented​​ at HiPEAC AccML workshop)​​​‌ extends this work to​ the question of whether​‌ low-bitwidth representations can be​​ learned jointly with new​​​‌ domain-specific knowledge during fine-tuning.​ Focusing on biomedical NLP​‌ under strong hardware constraints,​​ it studies the interaction​​ between domain-adaptive pre-training, fine-tuning,​​​‌ and extreme quantization. Experiments‌ on French biomedical data‌​‌ show that domain-specialized models​​ are more robust to​​​‌ aggressive quantization than generalist‌ ones, and that structural‌​‌ ternary quantization enables stable​​ training in very low-precision​​​‌ regimes. Even limited domain‌ adaptation significantly improves convergence,‌​‌ indicating that domain learning​​ and adaptation to low​​​‌ numerical precision can be‌ achieved simultaneously.

7.4 Optimization‌​‌ of arithmetic cores

7.4.1​​ Double-Word Decomposition in a​​​‌ Combined FP16, BF16 and‌ FP32 Dot Product Add‌​‌ Operator

Participants: Oregane Desrentes​​, Florent de Dinechin​​​‌.

This work 12‌ introduces a fused Dot‌​‌ Product Add (DPA) operator​​ that supports mixed-precision operations​​​‌ for both machine learning‌ and numerical computing by‌​‌ combining FP16, BF16, and​​ FP32 multiplicands within a​​​‌ single hardware unit. The‌ key technical innovation is‌​‌ the use of an​​ intermediate floating-point format (E9S12)—with​​​‌ a 9-bit exponent and‌ 12-bit significand—that enables an‌​‌ exact double-word decomposition of​​ FP32 multiplicands and efficient​​​‌ internal representation of lower-precision‌ inputs. The resulting operator‌​‌ is correctly rounded for​​ FP16 products accumulated into​​​‌ FP32 and also emulates‌ the IEEE-compliant FP32 fused‌​‌ multiply-add when operating on​​ decomposed FP32 inputs, simplifying​​​‌ hardware compared to traditional‌ double-word software techniques. Synthesis‌​‌ results for a 4​​ nm technology node show​​​‌ that this combined operator‌ improves both performance and‌​‌ accuracy over software decomposition​​ approaches.

This contribution was​​​‌ recognised with the Best‌ Paper Award at the‌​‌ ARITH 2025 conference, highlighting​​ its significance in understanding​​​‌ and optimising the trade-offs‌ inherent in mixed-precision hardware‌​‌ design.

7.4.2 Reconfigurable constant​​ multiplication

Participants: Bastien Barbe​​​‌, Louis Ledoux,‌ Anastasia Volkova, Florent‌​‌ de Dinechin.

This​​ work 9 has been​​​‌ done in the context‌ of the PhD thesis‌​‌ of Bastien Barbe in​​ collaboration with the team's​​​‌ postdoc Xiao Peng .‌ It addresses the design‌​‌ of efficient hardware multipliers​​ in the context of​​​‌ extremely quantized deep neural‌ networks, where multiplications are‌​‌ performed with constants drawn​​ from a small, fixed​​​‌ set. Rather than relying‌ on standard numerical formats‌​‌ (e.g. int4), which constrain​​ both representation and hardware​​​‌ design, the approach assumes‌ that only a carefully‌​‌ chosen set of integer​​ constants needs to be​​​‌ supported. It proposes dedicated‌ reconfigurable shift-and-add architectures capable‌​‌ of implementing these constants​​ efficiently by switching between​​​‌ pre-computed computation graphs. In‌ this setting, constants are‌​‌ represented implicitly through configuration​​ bits that select the​​​‌ appropriate circuit, allowing values‌ of larger magnitude to‌​‌ be handled with fewer​​ bits than standard encodings.​​​‌ Experimental results show that‌ the proposed algorithms outperform‌​‌ existing heuristics, support a​​ larger number of constants​​​‌ simultaneously, and lead to‌ more efficient multiplier designs‌​‌ in terms of hardware​​ cost.

Figure 10

Example of a​​​‌ two-adder RSCM.

Figure 10‌: Example 2-adder RSCM‌​‌ (reconfigurable single constant multiplier)​​

7.4.3 Constraint programming models​​​‌ for multiple constant multiplication‌

Participants: Anastasia Volkova,‌​‌ Christine Solnon, Theo​​ Cantaloube.

The Multiple​​​‌ Constant Multiplication (MCM) problem‌ arises in many applications‌​‌ such as, for example,​​ digital signal processing. Given​​​‌ a set T of‌ target constants, the goal‌​‌ of MCM is to​​​‌ find the most efficient​ way for multiplying an​‌ input number with each​​ constant in T,​​​‌ where multiplications are realized​ through bit-shifts and additions,​‌ and where intermediate results​​ may be shared to​​​‌ produce different target constants.​ State-of-the-art methods are based​‌ on Integer Linear Programming​​ (ILP), and suffer from​​​‌ numerous performance and scalability​ bottlenecks.

In 11 we​‌ have proposed for the​​ first time a Constraint​​​‌ Programming (CP) model for​ minimizing the number of​‌ adders for the MCM.​​ Compared to the state-of-the-art​​​‌ ILP approach, CP does​ not suffer from the​‌ curse of linearization, hence​​ permits significantly simpler formulations​​​‌ of the mathematical model.​ It is also more​‌ efficient, especially for the​​ hardest instances. We have​​​‌ also introduced a pseudo-polynomial​ time algorithm which is​‌ able to efficiently solve​​ some instances.

Figure 11

Performance comparison​​​‌ of MCM models.

Figure​ 11: Performance comparison​‌ of state-of-the-art approaches to​​ our CP model on​​​‌ a comprehensive benchmark.

This​ work has been done​‌ in the context of​​ a 6-month internship of​​​‌ Theo Cantaloube , co-supervised​ by Anastasia Volkova and​‌ Christine Solnon . Theo​​ Cantaloube has started a​​​‌ PhD thesis in October​ 2025. The goal of​‌ this thesis is to​​ investigate the capabilities of​​​‌ CP for solving this​ kind of problems.

7.4.4​‌ Optimization for other application​​ domains

Participants: Romain Fontaine​​​‌, Xiao Peng,​ Christine Solnon.

Beyond​‌ the application to arithmetic,​​ which is at the​​​‌ core of the research​ topics of Emeraude, we​‌ have also applied optimization​​ and constraint programming techniques​​​‌ to other problems: surface​ coverage by tethered robots​‌ 4, the travelling​​ salesman problem with time​​​‌ windows 5, 7​, and graph generation​‌ 19.

7.5 From​​ FLoPoCo to MLIR dialects​​​‌ and flows

Emeraude-MLIR is​ a multi-level arithmetic optimization​‌ compilation framework. It can​​ be viewed as an​​​‌ end-to-end compilation flow that​ targets a wide range​‌ of inputs, including DSP​​ applications such as Faust,​​​‌ machine learning workloads expressed​ in PyTorch and LLaMA,​‌ and HPC kernels such​​ as BLAS routines and​​​‌ polyhedral benchmarks from PolyBench.​ These inputs are mapped​‌ to multiple backends, including​​ FPGA, ASIC, and software​​​‌ targets through LLVM.

These​ compilation paths naturally expose​‌ dozens of lowering transformations,​​ with a strong emphasis​​​‌ on arithmetic specialization. Many​ of these transformations are​‌ context-aware, for example selecting​​ an accumulator bitwidth for​​​‌ a given sub-matrix block​ or adapting internal ROM​‌ sizes to FPGA resource​​ constraints. The internal transformations​​​‌ and intermediate representations, expressed​ through a hierarchy of​‌ dialects, encompass polynomial approximations​​ of real-valued computations, IEEE​​​‌ 754–compliant floating-point combinational IR​ construction, and graph rewriting​‌ over structured control flow​​ to capture exact accumulation​​​‌ semantics.

Overall, the flow​ has demonstrated its capabilities​‌ by lowering a LLaMA​​ attention block to silicon​​​‌ and by producing a​ silicon-proven Faust soft-clipping function​‌ fabricated through a recent​​ TinyTapeout shuttle, occupying wafer​​​‌ area in GlobalFoundries 180​ nm technology.2627​‌

Figure 12
Figure 12: Emeraude​​ MLIR flow
Figure 13
Figure 13​​​‌: TinyTapeout GF180 silicon​ for the Faust MLIR​‌ soft-clipping design.

7.6 The​​ Faust Programming Language and​​ its Ecosystem

Participants: Florent​​​‌ de Dinechin, Stéphane‌ Letz, Romain Michon‌​‌, Yann Orlarey,​​ Tanguy Risset, Christine​​​‌ Solnon.

Audio signal‌ processing is an applied‌​‌ field where success is​​ ultimately determined by the​​​‌ human ear, requiring advanced‌ tools to prototype and‌​‌ implement algorithms rapidly and​​ efficiently. The Faust programming​​​‌ language and environment, developed‌ at Grame-CNCM in 2002‌​‌ 87, represented a​​ significant development by enabling​​​‌ researchers and developers to‌ prototype and deploy audio‌​‌ processing algorithms more efficiently.​​ At the time, Faust​​​‌ was the first fully‌ compiled audio programming language,‌​‌ which played an important​​ role in making the​​​‌ field of real-time embedded‌ audio systems more accessible.‌​‌

Since its inception, Faust​​ has gained international recognition​​​‌ and is widely used‌ in both academic and‌​‌ professional contexts. It has​​ been adopted for teaching​​​‌ advanced topics such as‌ signal processing and physical‌​‌ interaction design at Stanford​​ University and for developing​​​‌ audio plugins 83.‌

Today, the Faust research‌​‌ project, conducted by Emeraude,​​ focuses on three interrelated​​​‌ areas:

  • The Faust Programming‌ Language: Developing a‌​‌ high-level language for sound​​ synthesis and signal processing​​​‌ that is accessible to‌ non-computer scientists.
  • The Compiler‌​‌ and Compilation Techniques:​​ Producing tools to automatically​​​‌ generate highly optimized code,‌ comparable in efficiency to‌​‌ code written by experienced​​ C programmers.
  • The Ecosystem​​​‌: Expanding and maintaining‌ architecture files (which allow‌​‌ the same Faust code​​ to run on over​​​‌ twenty platforms), development tools,‌ libraries, and documentation.

The‌​‌ following sections present the​​ work carried out during​​​‌ the period in each‌ of these areas.

7.6.1‌​‌ The Faust programming language​​

Faust is a synchronous​​​‌ functional programming language inspired‌ by lambda calculus, combinatory‌​‌ logic, and John Backus'​​ FP. Its semantics is​​​‌ centered around the concept‌ of audio circuits. Programming‌​‌ in Faust primarily involves​​ constructing new audio circuits​​​‌ by assembling primitive ones‌ using an algebra of‌​‌ five composition operations. During​​ the period, however, Faust​​​‌ has evolved beyond simple‌ circuit assembly.

7.6.2 The‌​‌ Faust compiler

To support​​ Faust 's evolving functionality​​​‌ and optimize performance across‌ a wider range of‌​‌ platforms, recent advancements in​​ the Faust compiler—driven in​​​‌ part by the Syfala‌ project—have introduced innovative compilation‌​‌ techniques that now target​​ not only CPUs but​​​‌ also FPGAs, enabling efficient‌ code generation and dynamic‌​‌ processing capabilities on diverse​​ hardware architectures.

FIR/IIR reconstruction​​​‌

Research work on FIR‌ and IIR reconstruction in‌​‌ the Faust compiler continued​​ in 2025 with the​​​‌ implementation of new techniques‌ that accelerate and simplify‌​‌ the process.

Let's recall​​ the project's starting point.​​​‌ The minimalist design philosophy‌ of the Faust language‌​‌ deliberately excludes high-level signal​​ processing functions that could​​​‌ be expressed using lower-level‌ primitives. This is why‌​‌ the Faust language does​​ not have any filter-type​​​‌ primitives (FIR or IIR)‌ since they can easily‌​‌ be expressed using delay​​ lines and recursions. Thus,​​​‌ all common signal processing‌ filters are defined in‌​‌ the standard library `filters.lib`,​​ which contains over 150​​​‌ of them.

Thanks to‌ various optimization techniques, such‌​‌ as delay line sharing,​​​‌ there is no penalty​ for not having these​‌ filters as language primitives.​​ However, this "unrolled" approach​​​‌ is not the most​ efficient with high-level synthesis​‌ tools like Vivado HLS.​​

This is why we​​​‌ introduced new compilation options​ that automatically reconstruct FIR​‌ and IIR filters in​​ the compiler's internal representations.​​​‌ While these options do​ not necessarily improve CPU​‌ performance, they are clearly​​ very efficient on FPGAs​​​‌ in terms of latency​ and resource trade-offs.

The​‌ new approach consists of​​ assembling binary additions and​​​‌ subtractions into n-ary weighted​ sums. Signals that appear​‌ with a fixed delay​​ in these sums are​​​‌ grouped into `FIR[x,a,b,c,..]` and​ recursive definitions of the​‌ type `x = y​​ + FIR[x,a,b,c,...]` where `y`​​​‌ does not depend on​ `x`, are transformed into​‌ `IIR[y,a,b,c,...]`.

Automatic differentiation

The​​ main subject is the​​​‌ implementation of automatic differentiation​ tools for the Faust​‌ programming language in order​​ to extend its capabilities​​​‌ to solving machine learning​ problems.

Automatic differentiation is​‌ a mechanism in which​​ a program can compute​​​‌ the derivative of an​ arbitrary function at runtime,​‌ avoiding the need for​​ the programmer to differentiate​​​‌ it beforehand or relying​ on other methods like​‌ symbolic calculus engines (WX​​ Maxima 28, Sympy​​​‌ 29) or discrete​ calculus tools. Automatic differentiation​‌ is necessary to perform​​ gradient descents and thus​​​‌ necessary for many optimisation​ problems and machine learning.​‌

Previous work has been​​ done on this topic.​​​‌ David Braun created a​ Jax 45 backend for​‌ Faust30 allowing the​​ use of its gradient​​​‌ computation tools and the​ integration in its deep​‌ learning ecosystem. But implementing​​ the mechanism directly inside​​​‌ Faust would then propagate​ to all the existing​‌ and future backends and​​ make use of the​​​‌ special aspects of the​ language to compute gradients​‌ in a new way.​​

Thomas Rushton of the​​​‌ Emeraude team has worked​ in the context of​‌ two Google Summer of​​ Code on implementing automatic​​​‌ differentiation in the compiler​ and in the syntax​‌ of the language. His​​ work was published in​​​‌ the proceedings of the​ 2024 edition of the​‌ Faust International Conference94​​ and is the main​​​‌ tool used in the​ beginning of this PHD​‌ to experiment with simple​​ examples.

Automatic differentiation is​​​‌ split into two modes:​ forward mode, and reverse​‌ mode, that present different​​ performance characteristics depending on​​​‌ the use case. For​ each mode, several implementation​‌ methods have been explored​​ depending on the programming​​​‌ environment. Many projects exist​ for the implementation of​‌ the main methods: source​​ transformation 66 (generating the​​​‌ derivative code given the​ primal function), gradient tape​‌ 63 (recording all the​​ operation in a first​​​‌ pass and unrolling the​ derivative in a second,​‌ backward pass), operator overloading​​ 93, LLVM IR​​​‌ transformation 84 (implemented as​ a plugin in the​‌ clang compiler).

Here, Rushton's​​ library falls in between​​​‌ source transformation and operator​ overloading as it is​‌ leveraging the pattern matching​​ capabilities of the Faust​​​‌ language to do operator​ overloading and the compiler​‌ is generating new code​​ for that derivative to​​ the chosen backend. This​​​‌ shows that the specificities‌ of the Faust language,‌​‌ as a Domain Specific​​ Language (DSL) can open​​​‌ new ways of computing‌ automatic differentiation.

A part‌​‌ of the work was​​ to implement several basic​​​‌ methods in the Julia‌31 programming language to‌​‌ understand a maximum of​​ details on Forward and​​​‌ Reverse mode in a‌ procedural environment rather than‌​‌ a functional one. Now​​ the main task is​​​‌ to continue the work‌ done by Thomas Rushton‌​‌ and create, measure and​​ test specific examples to​​​‌ compare against the state‌ of the art.

A‌​‌ differentiated version of the​​ NLMS algorithm (consisting in​​​‌ optimising an FIR filter‌ to match an unknown‌​‌ LTI system) was tried​​ and the current results​​​‌ show that the Faust‌ compiler is struggling too‌​‌ much to generate the​​ C++ code for an​​​‌ interesting amount of filter‌ coefficients (> 100). It‌​‌ showed some of the​​ limits of the compiler​​​‌ that will need to‌ be worked on for‌​‌ a working implementation of​​ differentiation algorithms. These results​​​‌ were presented in the‌ context of the annual‌​‌ Emeraude team seminary in​​ May 2025.

It was​​​‌ later found that delay‌ lines of variable lengths‌​‌ could not be differentiated​​ in the time domain​​​‌ and thus could not‌ be integrated in our‌​‌ automatic differenciation schemes. This​​ was a blockage as​​​‌ a significant part of‌ the Faust ecosystem as‌​‌ well as many audio​​ algorithms rely on that​​​‌ mecanism.

7.6.3 The Faust‌ ecosystem

Faust and Evolutionary‌​‌ / Genetic algorithms

Starting​​ in september 2025, exploration​​​‌ of genetic algorithms and‌ their potential integration with‌​‌ the Faust programming language​​ started. Genetic algorithms are​​​‌ an approach to automatic‌ optimisation inspired by biology.‌​‌ Similar to classical gradient​​ descent-based machine learning, a​​​‌ genetic algorithm is an‌ iterative method that converges‌​‌ toward an optimal solution​​ to a quantifiable problem.​​​‌ But unlike machine learning,‌ genetic algorithms converge to‌​‌ the optimal solution with​​ this scheme:

  • generating a​​​‌ generation of chromosomes (a‌ set of solutions);
  • evaluating‌​‌ every chromosome and computing​​ their respective loss function​​​‌ (called the fitness function)‌ value.
  • randomly selecting chromosomes‌​‌ for the next generation​​ with a bias toward​​​‌ ones with the best‌ fitness values.
  • randomly modifying‌​‌ the selected chromosomes.
  • repeating​​ the process until convergence​​​‌ or for a given‌ number of iterations.

For‌​‌ our use case, genetic​​ algorithms have the advantages​​​‌ to garantee the convergence‌ of the optimisation given‌​‌ enough iterations, and to​​ not require the differentiability​​​‌ of the system.

The‌ work has been targeted‌​‌ toward implementing a genetic​​ algorithm based optimisation tool​​​‌ that can optimize the‌ parameters of a Faust‌​‌ program given a target​​ behaviour. For the experiments,​​​‌ the target systems were‌ digital filters and the‌​‌ Faust programs were designed​​ to match the frequency​​​‌ response. A FIR (Finite‌ Impulse Response) filter was‌​‌ chosen for the higher​​ number of parameters (one​​​‌ per filter coefficient), better‌ suited for genetic algorithms.‌​‌

Faust MCP server architecture​​

In the Faust ecosystem,​​​‌ an architecture is a‌ software layer that connects‌​‌ compiled DSP code to​​​‌ the external world (audio​ drivers, user interfaces, communication​‌ protocols, etc.). Faust architectures​​ implement the glue code​​​‌ that bridges signal processing​ algorithms with various platforms​‌ and interaction modalities.

In​​ early 2025, we developed​​​‌ a new component that​ allows Faust audio applications​‌ to expose their sound​​ parameters for control by​​​‌ AI assistants such as​ ChatGPT or Claude. This​‌ component relies on the​​ Model Context Protocol (MCP)​​​‌ 32, an open​ standard introduced by Anthropic​‌ in November 2024 that​​ enables AI assistants to​​​‌ connect with external data​ sources and tools. The​‌ protocol defines a client-server​​ architecture using JSON-RPC 2.0​​​‌ communication over stdio or​ HTTP.

This Faust architecture​‌ uses MCP's tool system​​ to expose DSP parameters​​​‌ as callable functions. The​ MCP protocol distinguishes between​‌ tools (functions that can​​ be invoked by AI​​​‌ clients), resources (data sources​ such as files, APIs,​‌ and databases), prompts (reusable​​ templates for AI interactions),​​​‌ and servers (applications that​ expose tools and resources​‌ via the protocol). In​​ our implementation, each Faust​​​‌ widget becomes an MCP​ tool with a defined​‌ input schema. Tool names​​ follow hierarchical naming based​​​‌ on Faust UI structure.​ Parameter validation and range​‌ clamping are handled automatically,​​ and communication occurs via​​​‌ JSON-RPC over stdin/stdout.

This​ Faust MCP architecture is​‌ the first implementation enabling​​ AI-driven control of audio​​​‌ DSP parameters through natural​ language. This approach opens​‌ up possibilities that go​​ far beyond simple parameter​​​‌ adjustment. Users can now​ express complex temporal behaviors​‌ and relationships between parameters​​ in natural language—for example,​​​‌ "gradually increase the reverb​ while slowly decreasing the​‌ delay feedback" or "oscillate​​ the filter cutoff between​​​‌ 500Hz and 2kHz every​ three seconds while keeping​‌ the resonance stable."

This​​ capability effectively transforms natural​​​‌ language into a form​ of performance notation or​‌ score, where intricate parameter​​ evolutions can be described​​​‌ conversationally rather than programmed​ explicitly. Such an interface​‌ is particularly valuable for​​ interactive sound installations, where​​​‌ visitors with no technical​ knowledge can engage with​‌ and perform the sonic​​ environment using everyday language.​​​‌ Instead of confronting control​ panels or learning specific​‌ commands, they can simply​​ describe what they want​​​‌ to hear, making sound​ art and experimental audio​‌ systems accessible to broader​​ audiences while preserving the​​​‌ depth and complexity of​ the underlying synthesis.

Wasmtime​‌ architecture

Wasmtime 33 is​​ a high-performance WebAssembly runtime​​​‌ designed to execute WebAssembly​ modules outside the browser​‌ in an efficient and​​ secure way. In particular​​​‌ it allows native applications​ to do JIT compilation​‌ starting from a pre-compiled​​ wasm module. This makes​​​‌ it particularly interesting for​ real-time domains such as​‌ audio where performance and​​ safety are critical. The​​​‌ wasmtime code is written​ in Rust, but offers​‌ C and C++ exports,​​ so that pure C/C++​​​‌ projects can easily link​ to wasmtime as a​‌ library.

The Faust /wasmtime​​ project explores how Faust​​​‌ DSP code compiled to​ WebAssembly can be executed​‌ natively using Wasmtime. Its​​ main goal is to​​​‌ make WebAssembly an alternative​ standalone deployment format for​‌ Faust DSPs, without relying​​ on JavaScript (used in​​ a Node.js environment) or​​​‌ a web browser.

The‌ libfaust library is used‌​‌ with its WebAssembly backend​​ to produce a wasm​​​‌ module to be JIT‌ compiled using wasmtime and‌​‌ executed on the fly.​​ The wasm module is​​​‌ then wrapped with a‌ specific architecture which connects‌​‌ it with the wasmtime​​ machinery, exposing the internal​​​‌ API (initialization, compute, parameters‌ access), accessing some needed‌​‌ functions from the wastime​​ time (like math functions),​​​‌ and defining the memory‌ block to be shared‌​‌ between the wasm module​​ and the execution runtime.​​​‌

The final API is‌ exposed in a factory/instance‌​‌ model, where the DSP​​ instance pointer can be​​​‌ used with already developed‌ audio and controller managers,‌​‌ port of the Faust​​ general architecture machinery.

Two​​​‌ concrete use-cases have been‌ developed:

  • faustwasmtime: using‌​‌ JACK as the audio​​ layer and accessing the​​​‌ control parameters with a‌ GTK based GUI or‌​‌ a HTTPD controller.
  • faustbench-wasmtime​​: to benchmark the​​​‌ DSP CPU consumption.
faust2wwise‌ tool

faust2wwise aimed to‌​‌ integrate the Faust programming​​ language with Audiokinetic's Wwise​​​‌ 34, the multi-platform‌ industry standard audio middleware‌​‌ in game development. This​​ framework allows sound designers​​​‌ to set up audio‌ environments, explore spatial audio,‌​‌ interactive music and real-time​​ synthesis. The project implemented​​​‌ 2 different and complementary‌ use-cases:

  • a faust2wwise tool‌​‌ for statically compiling Faust​​ DSP code into Wwise​​​‌ modules, to be loaded‌ in the Wwise environment.‌​‌
  • a plugin that embeds​​ the libfaust and LLVM​​​‌ JIT compiler for dynamic‌ Faust DSP live-coding within‌​‌ Wwise, allows a explore​​ different Faust DSP programs​​​‌ at runtime.

Drawing on‌ experience in C++ audio‌​‌ programming, DSP techniques, and​​ plugin development, a practical​​​‌ connection has been built‌ that allows audio designers‌​‌ and programmers to use​​ Faust 's DSP capabilities​​​‌ directly within their Wwise‌ workflows.

faust2clap tool

The‌​‌ faust2clap tool generates a​​ fully working CLAP35​​​‌ plugin starting from a‌ Faust DSP program. The‌​‌ tool supports both statically​​ compiled and dynamically written​​​‌ and compiled (hot reload)‌ DSP code:

  • in the‌​‌ static mode model, a​​ given DSP program is​​​‌ wrapped with additional C++‌ code then compiled into‌​‌ a CLAP plugin binary.​​ Control parameters are exposed​​​‌ to be usable by‌ the host application (typically‌​‌ a DAW that would​​ load the plugin and​​​‌ instantiate it to generate‌ or transform an audio‌​‌ track. Synthesizer and effect​​ are supported, as well​​​‌ as MIDI controllable polyphonic‌ instruments.
  • a dynamic plugin‌​‌ has also been developed.​​ It compiles and reloads​​​‌ any .dsp program while‌ it is running, so‌​‌ that the user does​​ not need to build​​​‌ a new binary or‌ close and reopen the‌​‌ plugin in the host.​​ This approach makes testing​​​‌ and development faster, and‌ allows experiments to be‌​‌ carried out in the​​ same environment in which​​​‌ the final plugin will‌ be used, for example‌​‌ in a DAW such​​ as Reaper.
Faust PWA​​​‌ project

The project 18‌ aimed at moving Faust‌​‌ instruments deployed on iOS​​ and Android platforms, coded​​​‌ with native frameworks and‌ languages, to web-based audio‌​‌ applications, focusing on the​​​‌ use of WebAssembly, the​ Web Audio API, and​‌ Progressive Web Applications (PWA).​​ Its main goal is​​​‌ to simplify the deployment​ and maintenance of real-time​‌ musical applications while preserving​​ performance and interactivity comparable​​​‌ to native solutions.

Initially,​ Faust applications such as​‌ SmartFaust and GameLan were​​ developed as native apps​​​‌ for iOS and Android,​ relying on platform-specific audio​‌ backends and programming languages,​​ like Objective C on​​​‌ iOS and Java on​ Android for GUI implementations.​‌ Although this approach ensured​​ low-latency audio and effective​​​‌ use of device sensors,​ it required maintaining multiple​‌ codebases and complying with​​ restrictive app store distribution​​​‌ and validation processes.

The​ emergence of WebAssembly and​‌ modern web standards enabled​​ a move toward a​​​‌ single and cross-platform web​ architecture. By compiling Faust​‌ DSP code to WebAssembly,​​ applications can run directly​​​‌ in browsers with near-native​ performance, while benefiting from​‌ browser security, sandboxing, and​​ permissions management.

To support​​​‌ this transition, the faustwasm​ package was introduced as​‌ a JavaScript/TypeScript library. It​​ automates the generation of​​​‌ self-contained web applications, supports​ MIDI by analyzing DSP​‌ metadata, efficiently integrates sensor​​ data usage, and allows​​​‌ the use of soundfiles,​ making it suitable for​‌ interactive musical systems.

Finally,​​ the adoption of the​​​‌ Progressive Web Application model​ allows Faust applications to​‌ be installed easily on​​ mobile devices via URLs​​​‌ or QR codes, with​ offline support and secure​‌ access to sensors. This​​ approach greatly improves accessibility​​​‌ and usability, opening new​ opportunities for artistic creation​‌ and music education using​​ web-based Faust tools.

8​​​‌ Bilateral contracts and grants​ with industry

Participants: Florent​‌ de Dinechin, Stephane​​ Letz, Romain Michon​​​‌, Yann Orlarey,​ Tanguy Risset, Anastasia​‌ Volkova, Christine Solnon​​.

8.1 Bilateral contracts​​​‌ with industry

The PhD​ thesis of Benjamin Quiédeville,​‌ in collaboration with GRAME-CNCM,​​ includes a support contract​​​‌ of 30,000€ for the​ duration of the thesis.​‌

The PhD thesis of​​ Orégane Desrentes, in collaboration​​​‌ with Kalray, includes a​ support contract of 47,500€​‌ for the duration of​​ the thesis.

Anastasia Volkova,​​​‌ co-advises an industrial PhD​ thesis with Valeuriad company​‌ and Nantes University on​​ the subject of frugal​​​‌ Natural Language Processing systems.​ The collaboration includes a​‌ 25,000€ support transferred to​​ Emeraude for 2023-2027.

Florent​​​‌ de Dinechin co-advises an​ industrial PhD thesis with​‌ Thales on the subject​​ of Vision transformers and​​​‌ FPGAs. The collaboration includes​ a 45,000€ support for​‌ 2025-2028.

9 Partnerships and​​ cooperations

9.1 International initiatives​​​‌

9.1.1 Associate Teams in​ the framework of an​‌ Inria International Lab or​​ in the framework of​​​‌ an Inria International Program​

MIRAGES
  • Title:
    Modeling and​‌ Immersive Rendering of Archaeoacoustics​​ in Grotto Environments
  • Duration:​​​‌
    2025 -> 2027
  • Coordinator:​
    Romain Michon and Chris​‌ Chafe (cc@ccrma.stanford.edu)
  • Partner Institution(s):​​
    • Center for Computer Research​​​‌ in Music and Acoustics​ (CCRMA), Stanford University, USA​‌
  • Inria contact:
    Romain Michon​​
  • Summary:
    As computing power​​​‌ has increased since the​ 2000s, virtual acoustics based​‌ on impulse response measurement​​ and convolution have made​​​‌ it possible to recreate​ the sound of real​‌ spaces with high fidelity,​​ driving growing interest from​​ virtual reality, cultural institutions,​​​‌ and immersive media. While‌ stereo convolution reverberation is‌​‌ now commonplace, delivering truly​​ immersive, real-time, multi-speaker virtual​​​‌ acoustics remains challenging due‌ to cost, system complexity,‌​‌ and computational demands. The​​ MIRAGES associate team addresses​​​‌ these challenges through the‌ Chauvet Cave as a‌​‌ unique case study, aiming​​ to restore the missing​​​‌ acoustical dimension of its‌ public replica by developing‌​‌ more immersive rendering techniques,​​ leveraging FPGA platforms for​​​‌ high-performance real-time processing, and‌ implementing an accurate, interactive‌​‌ simulation of the cave.​​ Building on advances from​​​‌ the FAST ANR project‌ and the PLASMA associate‌​‌ team that preceded it,​​ MIRAGES seeks to create​​​‌ scalable, low-cost systems capable‌ of driving hundreds or‌​‌ thousands of speakers and​​ supporting advanced spatial audio​​​‌ methods such as Wave-Field‌ Synthesis, with broad implications‌​‌ for research, industry, and​​ public-facing audio experiences.

9.2​​​‌ National initiatives

9.2.1 ANR‌ DIStrib

Interest around spatial‌​‌ audio has been booming​​ in recent years. An​​​‌ increasingly high number of‌ movie theaters, concert halls,‌​‌ Virtual Reality (VR) platforms​​ in museums, attractions in​​​‌ amusement parks, etc. are‌ equipped with advanced spatial‌​‌ audio systems involving a​​ large number of speakers.​​​‌ The automotive industry has‌ also recently shown interest‌​‌ in spatial audio for​​ its applications in the​​​‌ context of active noise‌ cancellation. When considering interactive‌​‌ applications (i.e., virtual acoustics,​​ soundscape rendering in VR,​​​‌ noise canceling, speaker correction,‌ speech intelligibility enhancement, etc.)‌​‌ involving real-time operations and​​ the ability the reprogram/customize​​​‌ the system, managing a‌ large number of individual‌​‌ audio channels requires a​​ tremendous amount of computational​​​‌ power and incredibly high‌ bandwidths, which current systems‌​‌ fail to provide. The​​ norm to implement such​​​‌ systems is to rely‌ on a centralized software-based‌​‌ approach: a powerful computer​​ connected to one or​​​‌ multiple audio interfaces providing‌ a limited number of‌​‌ audio outputs. In that​​ case, the bottleneck is​​​‌ the computer's throughput and‌ hence its ability to‌​‌ manage a large number​​ of audio streams in​​​‌ parallel with potential computations‌ applied to each of‌​‌ them.

The goal of​​ DIStrib is to rethink​​​‌ the way we approach‌ spatial audio systems by‌​‌ relying on a distributed​​ computing approach leveraging the​​​‌ computational power of large‌ Field-Programmable Gate Arrays (FPGA).‌​‌ In this system, each​​ FPGA is in charge​​​‌ of computing the sound‌ of a limited set‌​‌ of speakers. Conversely, the​​ distributed approach allows for​​​‌ a very large number‌ of speakers to be‌​‌ targeted. In order to​​ reach this goal, multiple​​​‌ challenges ranging from transmitting‌ audio streams to a‌​‌ large number of audio​​ devices with perfect synchronicity​​​‌ to running complex audio‌ Digital Signal Processing (DSP)‌​‌ algorithms on FPGAs must​​ be tackled. We believe​​​‌ that such a system‌ has the potential to‌​‌ be highly disruptive in​​ the field of spatial​​​‌ audio. DIStrib is also‌ the occasion to explore‌​‌ the potential of artificial​​ intelligence in the context​​​‌ of immersive sound and‌ virtual acoustics rendering by‌​‌ relying on emerging platforms​​ such as embedded NPUs​​​‌ (Neural Processing Unit).

DIStrib‌ is a 42 months‌​‌ JCJC ANR project that​​​‌ started in October 2025.​ Romain Michon is the​‌ Principal Investigator (PI) of​​ DIStrib.

9.2.2 PEPR HoliGrail​​​‌

A key challenge to​ reach the maximal efficiency​‌ is to match algorithms​​ with the underlying hardware.​​​‌ As the end of​ Moore’s law steadily approaches,​‌ hardware becomes increasingly heterogeneous,​​ and the interplay between​​​‌ algorithms and hardware even​ more important. From this​‌ point of view, artificial​​ intelligence algorithms are not​​​‌ efficient when run on​ commodity hardware such as​‌ microprocessors and GPUs. The​​ last decade has seen​​​‌ a boom of accelerators​ for common inference tasks​‌ (first in line was​​ Google’s TPU, followed by​​​‌ many others), now embedded​ in many consumer products.​‌ These accelerators offer hardware​​ that better matches the​​​‌ algorithms, but are still​ far from achieving the​‌ “inference per joule” performance​​ of the Human brain.​​​‌ Training deep neural networks​ (DNNs) is even less​‌ efficient, requiring orders of​​ magnitude more computation operations​​​‌ than inference.

We believe​ that there is a​‌ significant room for improvements​​ in both “inference per​​​‌ joule" and "training per​ joule”. The vision of​‌ this action is to​​ create a synergy with​​​‌ the research on the​ foundations of AI frugality​‌ (as proposed in SHARP​​ action) to propose cutting-edge​​​‌ methods that significantly improve​ the energy efficiency of​‌ both inference and training​​ of a model. We​​​‌ will propose (i) more​ compact and efficient number​‌ representations that still maintain​​ a quality of inference​​​‌ or training close to​ the reference, (ii) hardware-aware​‌ training algorithms that enhance​​ certain types of sparsity​​​‌ (e.g., more structured), coding​ compactness (aggressive quantization, maximum​‌ entropy) and tensor transformations.​​ Most state-of-the-art solutions are​​​‌ agnostic of the hardware​ they run on. By​‌ taking advantage of this​​ interplay between the hardware​​​‌ and the algorithms, we​ can achieve breakthroughs beyond​‌ current solutions, in particular​​ by developing (iii) efficient​​​‌ hardware mechanisms, especially optimized​ to take advantage of​‌ sparsity, extreme quantization and​​ ad-hoc number representations, together​​​‌ with (iv) compiler optimizations,​ to demonstrate the effectiveness​‌ of the proposed methods.​​ Our approaches are holistic​​​‌ in the sense that​ they will jointly optimize​‌ the whole computing stack​​ of AI, i.e., at​​​‌ the algorithm, arithmetic, compiler​ and hardware levels.

PEPR​‌ HOLIGRAIL kicked off in​​ March 2024 and will​​​‌ end in 2029 (extension​ has been anticipated). The​‌ project combines following partners:​​ Inria/IRISA Taran (Univ. Rennes,​​​‌ Inria, CNRS), List /LIAE​ (Université Paris Saclay, CEA),​‌ Inria Corse (Université Grenoble​​ Alpes), TIMA SLS (CNRS,​​​‌ Université Grenoble Alpes), CITI​ lab (INSA Lyon /​‌ Inria), List / LVML​​ (Université Paris Saclay, CEA).​​​‌ The scientific leaders are​ Olivier Bichler (CEA List)​‌ and Olivier Sentieys (Inria​​ Taran).

10 Dissemination

Participants:​​​‌ Tanguy Risset, Romain​ Michon, Pierre Cochard​‌, Florent de Dinechin​​, Anastasia Volkova,​​​‌ Stéphane Letz, Christine​ Solnon.

10.1 Promoting​‌ scientific activities

10.1.1 Scientific​​ events: organisation

General chair,​​​‌ scientific chair
  • Romain Michon​ served as General Chair​‌ and Stephane Letz as​​ Technical Chair for the​​​‌ JIMLAC-25 conference which was​ mainly organized by Emeraude​‌ (see §5.1).​​

10.1.2 Scientific events: selection​​

Member of the conference​​​‌ program committees
  • Romain Michon‌ served in the program‌​‌ committees of SMC (Sound​​ and Music Computing conference),​​​‌ DAFx (Digital Audio Effects‌ conference), NIME (New Interfaces‌​‌ for Musical Expression conference),​​ ICMC (International Computer Music​​​‌ Conference).
  • Tanguy Risset served‌ in the program committees‌​‌ of ASAP 2025 and​​ DATE 2025 (Design Automation​​​‌ and Test in Europe),‌ on track " Architectural‌​‌ and Microarchitectural Design".
  • Christine​​ Solnon served in the​​​‌ programme committee of CPAIOR‌ 2025, and as‌​‌ senior PC member for​​ CP 2025.
  • Florent​​​‌ de Dinechin served in‌ the programme committees of‌​‌ ARITH 2025, ASAP​​ 2025, FCCM 2025​​​‌, FPL 2025.‌
  • Anastasia Volkova served in‌​‌ the programme committees of​​ ARITH 2025 and ASAP​​​‌ 2025

10.1.3 Journal

Member‌ of the editorial boards‌​‌
  • Christine Solnon is associate​​ editor of JAIR and​​​‌ Constraints.

10.1.4 Invited‌ talks

10.1.5 Leadership‌​‌ within the scientific community​​

  • Christine Solnon is member​​​‌ of the executive committee‌ of GDR RADIA.‌​‌
  • Anastasia Volkova is a​​ part of the organizing​​​‌ comittee for the "Embedded‌ HPC" axis of GDR‌​‌ SoC2.
  • Romain Michon is​​ the president of the​​​‌ Sound and Music Computing‌ (SMC) Network that steers‌​‌ the planning of the​​ SMC conference, one of​​​‌ the most prestigious scientific‌ event in the sound‌​‌ and music technology community.​​
  • Romain Michon is a​​​‌ board member the Association‌ Française d'Informatique Musicale (AFIM)‌​‌ and of GRAME-CNCM, both​​ as secretary.

10.2 Teaching​​​‌ - Supervision - Juries‌ - Educational and pedagogical‌​‌ outreach

  • Tanguy Risset is​​ professor at the Telecommunications​​​‌ Department of Insa Lyon.‌
  • Florent de Dinechin is‌​‌ a professor at the​​ Computer Science Department of​​​‌ Insa Lyon and head‌ of the 4th year.‌​‌ He also teaches computer​​ architecture at ENS-Lyon.
  • Christine​​​‌ Solnon is a professor‌ at the Computer Science‌​‌ Department of Insa Lyon.​​
  • Romain Michon is a​​​‌ part-time associate professor at‌ the Telecommunications Department of‌​‌ Insa Lyon.
  • Romain Michon​​ teaches 2 courses as​​​‌ part of the RIM/RAN‌ Masters Program at the‌​‌ université of Saint-Étienne.
  • Romain​​ Michon teaches 2 one​​​‌ week workshops at Aalborg‌ University in Copenhagen every‌​‌ year.
  • Stephane Letz teaches​​ 1 course as part​​​‌ of the RIM/RAN Masters‌ Program at the université‌​‌ of Saint-Étienne.
  • Anastasia Volkova​​ teaches 1 course at​​​‌ the Telecommunications department of‌ INSA and offers multiple‌​‌ semester projects.

10.2.1 Juries​​​‌

  • Anastasia Volkova was in​ the PhD jury of:​‌
    • Sami Ben Ali (Univ.​​ de Rennes), as examiner.​​​‌
  • Romain Michon was in​ the PhD jury of:​‌
    • David Fiero (U. Paris​​ 8), as reviewer,
    • Alexandre​​​‌ d'Hooge (U. of Lille),​ as reviewer,
    • Daniel Picciola​‌ (U. Paris 8), as​​ reviewer.
  • Christine Solnon was​​​‌ in the PhD jury​ of:
    • Jean-Baptiste Sciau (IMT​‌ Albi) as president,
    • Jorge​​ Mortes Alcaraz (IMT Atlantique)​​​‌ as president,
    • Stevan Stanovic​ (ENSICAEN) as president,
    • Guillaume​‌ Ghienne (IMT Atlantique) as​​ president,
    • Léon Fauste (U.​​​‌ Grenoble Alpes) as co-director,​
    • Fei Ge (ENTPE) as​‌ president,
    • Bachtiar Herdianto (IMT​​ Atlantique) as examiner,
    • Julien​​​‌ Rouzot (U. Toulouse) as​ examiner,
    • Djawad Bekkoucha (U.​‌ Caen) as examiner,
    • Alexandre​​ Ronsain (ENAC) as reviewer​​​‌ and president,
    • Fayad Ali​ Banna (U. Saint Etienne)​‌ as president,
    • Antoine Lhomme​​ (U. Grenoble Alpes) as​​​‌ president,
    • Arsène Marzorati (INSA​ Lyon) as president.
  • Florent​‌ de Dinechin was in​​ the PhD jury of:​​​‌
    • Cheolyong Bae (Linköping University,​ Suède),
    • Jonas Bertels (KULeuven,​‌ Belgique),
    • Quentin Milot (Université​​ de Rennes).

10.3 Popularization​​​‌

10.3.1 Productions (articles, videos,​ podcasts, serious games, ...)​‌

Romain Michon served in​​ the scientific committee of​​​‌ the “Ça résonne37​” exhibit at the​‌ Maison des Mathématiques et​​ de l'Informatique (MMI) of​​​‌ Lyon University.

Christine Solnon​ wrote a paper on​‌ women in computation 28​​, and a column​​​‌ for La Recherche 39​.

10.3.2 Participation in​‌ Live events

Several concerts​​ where organized during the​​​‌ JIMLAC-25 conference which was​ mainly organized by Emeraude,​‌ the concert list can​​ be seen on JIMLAC​​​‌ program 38

10.3.3 Others​ science outreach relevant activities​‌

  • Benjamin Quiedeville proposed a​​ demonstration of the spacebar​​​‌ 20 at the "fête​ de la science" in​‌ October 2025.
  • Romain Michon​​ actively took part in​​​‌ the Chiche! program in​ 2025.
  • Christine Solnon actively​‌ took part in the​​ Chiche! program in 2025,​​​‌ and she gave a​ conference on AI for​‌ the staff of ENSSIB​​
  • Anastasia Volkova actively participated​​​‌ in Comptoire des Sciences​ and Déclic programs in​‌ 2025.
    Figure 14

    The image is​​ a promotional poster with​​​‌ a green, high-tech aesthetic​ featuring a complex network​‌ of wires and electronic​​ components. The text reads​​​‌ "Exposant: Grame INRIA/INSA Lyon"​ at the top, indicating​‌ the exhibitors. It mentions​​ the event date as​​​‌ 15.11 (November 15) in​ Lyon, with the location​‌ specified as "@Périscope, Lyon."​​ At the bottom right,​​​‌ the phrase "Media Land"​ is partially visible, suggesting​‌ the event's theme or​​ name. (Description generated at​​​‌ February 2nd, 2026 by​ Albert AI with the​‌ model Mistral-Small-3.2-24B)

    Figure 14​​: Logo of the​​​‌ Grame/Emeraude demonstration at Moduland​
  • Louis Ledoux participated in​‌ the first edition of​​ the Moduland electronic arts​​​‌ festival, organized by Le​ Séquenceur, by representing the​‌ Émeraude research team (INSA–Inria–GRAME).​​ He presented an experimental​​​‌ electronic instrument during a​ long public demonstration, leading​‌ to numerous exchanges with​​ festival attendees and practitioners.​​​‌ This highly successful inaugural​ edition offered strong visibility​‌ and contributed to the​​ promotion of the team’s​​​‌ work, emphasizing its concrete​ artistic realizations and the​‌ close articulation between research,​​ digital instrument design, and​​ contemporary electronic art practices.​​​‌ 3940

11 Scientific‌ production

11.1 Major publications‌​‌

  • 1 bookF.Florent​​ de Dinechin and M.​​​‌Martin Kumm. Application-Specific‌ Arithmetic.Springer International‌​‌ Publishing2024HALDOI​​

11.2 Publications of the​​​‌ year

International journals

Invited conferences​​

  • 7 inproceedings C.Christine​​​‌ Solnon. Anytime and‌ exact search for planning‌​‌ problems: How to explore​​ a DP-based state transition​​​‌ graph with A*, CP‌ and LS? 31st International‌​‌ Conference on Principles and​​ Practice of Constraint Programming​​​‌ (CP 2025) 41 2‌ Glasgow, United Kingdom 2025‌​‌ HAL DOI back to​​ text

International peer-reviewed conferences​​​‌

National peer-reviewed Conferences‌​‌

Conferences‌ without proceedings

  • 24 inproceedings‌​‌A.Aurélien Delage and​​ C.Christine Solnon.​​​‌ Un modèle de théorie‌ des jeux pour l’étude‌​‌ de stratégies de réduction​​ de l’impact environnemental humain​​​‌.26ème congrès annuel‌ de la société française‌​‌ de recherche opérationnelle et​​ d’aide à la décision​​​‌ (ROADEF)Marne-la-Vallée, FranceFebruary‌ 2025, 226-227HAL‌​‌DOI
  • 25 inproceedingsO.​​Orégane Desrentes, F.​​​‌Florent de Dinechin and‌ B.Benoît Dupont de‌​‌ Dinechin. Reciprocal Square​​ Root Accelerated Using Hardware​​​‌ and Software Techniques.‌RAIM Meeting 2025: 16th‌​‌ Rencontres de l'Arithmétique en​​ Informatique Mathématique – A​​​‌ Tribute to Jean-Michel Muller‌Lyon, FranceNovember 2025‌​‌HAL
  • 26 inproceedingsR.​​Romain Fontaine and C.​​​‌Christine Solnon. Large-scale‌ experiments on the Traveling‌​‌ Salesman Problem with Time​​ Windows.ROADEF 2025​​​‌ - 26ème congrès annuel‌ de la Société Française‌​‌ de Recherche Opérationnelle et​​ d’Aide à la Décision​​​‌Champs-sur-Marne, FranceFebruary 2025‌HAL
  • 27 inproceedingsX.‌​‌Xavier Pillet, C.​​Cédric Gernigon, A.​​​‌Anastasia Volkova, R.‌Richard Dufour, A.‌​‌Adeline Granet and N.​​Nicolas Greffard. Quantization-aware​​​‌ training: a tradeoff between‌ training and fine-tuning for‌​‌ domain-specific language models.​​AccML 2026 - 8th​​​‌ Workshop on Accelerated Machine‌ LearningKrakow (Cracovie), Poland‌​‌January 2026HAL

Scientific​​ book chapters

Edition (books,‌​‌ proceedings, special issue of​​ a journal)

  • 29 proceedings​​​‌Proceedings of the 32nd‌ Journées d'Informatique Musicale.‌​‌Journées d'Informatique MusicaleLyon,​​ FranceJuly 2025HAL​​​‌
  • 30 proceedingsR.Romain‌ Michon and P.Pierre‌​‌ Lecomte, eds. Proceedings​​ of the 19th Linux​​​‌ Audio Conference.Proceedings‌ of the 19th Linux‌​‌ Audio ConferenceLyon, France​​June 2025HAL

Reports​​​‌ & preprints

Other scientific publications​‌

Scientific​​ popularization

11.3 Cited publications​‌

  • 40 articleL.Levent​​ Aksoy, E.Eduardo​​​‌ da Costa, P.​Paulo Flores and J.​‌José Monteiro. Exact​​ and Approximate Algorithms for​​​‌ the Optimization of Area​ and Delay in Multiple​‌ Constant Multiplications.IEEE​​ Transactions on Computer-Aided Design​​​‌ of Integrated Circuits and​ Systems2762008​‌, 1013--1026back to​​ text
  • 41 inproceedingsP.​​​‌ R.P. Rivera Benois​, P.Patrick Nowak​‌ and U.Udo Zölzer​​. Fully Digital Implementation​​​‌ of a Hybrid Feedback​ Structure for Broadband Active​‌ Noise Control in Headphones​​.2017 Proceedings of​​​‌ the 24th International Congress​ on Sound and Vibration​‌2017back to text​​
  • 42 articleB.Benjamin​​​‌ Betgen and M.-A.Marie-Annick​ Galland. A New​‌ Hybrid Active/Passive Sound Absorber​​ with Variable Surface Impedance​​​‌.Mechanical systems and​ signal processing255​‌2011, 1715--1726back​​ to text
  • 43 inbook​​​‌T.Thomas Bollaert.​ Catapult Synthesis: A Practical​‌ Introduction to Interactive C​​ Synthesis.High-Level Synthesis:​​ From Algorithm to Digital​​​‌ CircuitP.Philippe Coussy‌ and A.Adam Morawiec‌​‌, eds. DordrechtSpringer​​ Netherlands2008, 29--52​​​‌back to text
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