How ReSpeaker Enables Clear Voice Pickup Even in Noise: A Technical Explainer
In the world of Voice AI, there’s a crucial difference between simply “hearing” sound and “hearing clearly.” Simply detecting speech is easy, the real challenge lies in capturing clean, intelligible audio. Raw audio is messy, filled with echoes, noise, and reverberation. For AI applications to accurately process and interpret those sounds, microphone arrays must first receive high-quality input. Without clarity, there can be no “understanding“.

In this blog, we’ll break down five technical insights of Seeed’s latest AI-powered 4-Mic Array. We’ll be covering:
- Introduing AEC and Dereverberation
- From NS to Dynamic NS: Smarter Noise Suppression
- PDM MEMS Microphones vs. Analog Microphones
- How to Place Microphones?
- Bottom-Firing Microphone Design Update
With these insights, this article will provide you with a better understanding of how the ReSpeaker XMOS XVF3800 combines upgraded algorithms, powerful hardware, and efficient design to systematically deliver exceptional voice pickup.
Introduing AEC and Dereverberation
In a conference room, microphones often pick up not only the speaker’s voice but also annoying echoes and reverberations. To tackle these issues, ReSpeaker integrates two essential acoustic algorithms: AEC and dereverberation. While both aim to deliver cleaner and clearer audio, they address fundamentally different challenges.
Acoustic Echo Cancellation (AEC)
Definition: AEC is the process of removing the echo generated when sound from a speaker re-enters the microphone.
Why it matters: Without AEC, users in a call or conference would hear their own voices fed back, making communication distracting and unpleasant. Effective AEC ensures that only the intended voice is transmitted, which is critical in scenarios like online meetings, call centers, and voice AI applications.
How it works: AEC systems continuously analyze the reference signal sent to the loudspeaker and compare it with the signal captured by the microphone. Using adaptive filters, the algorithm predicts and subtracts the loudspeaker’s contribution from the microphone input in real time. This way, only the near-end user’s clean speech is preserved and transmitted.
Dereverberation
Definition: Dereverberation is the process of reducing or removing reverberation—the prolonged “tail”of sound caused by reflections in a room. Unlike AEC, dereverberation targets the acoustic environment itself, not the loudspeaker’s playback.
Why it matters: Reverberation blurs speech and makes words less distinct, especially in large rooms or spaces with hard surfaces. For far-field voice capture, reverberation drastically lowers automatic speech recognition (ASR) accuracy. Dereverberation directly improves clarity and makes AI-driven voice systems more reliable in real-world environments.
How it works: Modern dereverberation algorithms analyze the microphone signal in the time-frequency domain, estimate the late reverberation components, and suppress them while preserving the direct speech signal. This enhances speech intelligibility and ensures that the input to speech recognition engines or conferencing systems is clean and natural.
From NS to Dynamic NS: Smarter Noise Suppression for ReSpeaker
One key improvement of the new ReSpeaker XMOS XVF3800 is the shift from traditional noise suppression to dynamic noise suppression.
What Traditional Noise Suppression Does
- Works with fixed noise models, such as steady background hums or white noise.
- Struggles with non-stationary or sudden noises like keyboard clicks, door knocks, or side conversations.
- Often results in distortion or artifacts, making voices sound unnatural.
What Dynamic Noise Suppression Brings
- Real-time adaptation: Continuously analyzes the environment, detects both steady and transient noise, and adjusts suppression strength dynamically.
- Smarter classification: Recognizes different noise types and applies the right strategy for each.
Why It Matters
- More natural conversations: Suppresses disruptive noises while preserving the clarity and richness of the human voice. No more “breathing effects” or clipped speech.
- Stronger adaptability: Performs consistently across dynamic environments such as offices, coffee shops, or cars.
PDM MEMS Microphones vs. Analog Microphones
Even the most advanced algorithms depend on high-quality “ears” to capture sound. This leads us to the next topic: microphones.
In Seeed Studio’s ReSpeaker product family, most newer models have transitioned to using PDM MEMS microphones, while only the Raspberry Pi-compatible ReSpeaker Pi HAT still utilize Analog microphones. So, what are the differences between these two types of microphones, and where is each most suitable?
| Analog Microphones | PDM MEMS Microphones | |
| Definition | Convert sound waves into continuous electrical voltage signals | Use MEMS diaphragm + circuitry to output sound as digital PDM signal |
| Output Signal Type | Analog voltage | Digital (PDM format) |
| Advantages | Simple structure Low cost | Strong EMI immunity Direct connection to digital processors Compact size High consistency Suitable for arrays |
| Disadvantages | Susceptible to EMI Limited transmission distance Needs external ADC | Higher cost than analog Dependent on digital processing compatibility |
| Ideal Use Cases | Cost-sensitive, short-distance audio with minimal interference | Noise-resistant, long-distance, high-integration, and multi-microphone array applications |
How to Place Microphones on ReSpeaker?
When it comes to microphone arrays, choosing the right component is only half way to success. Another key factor that often gets overlooked is how the microphones are actually placed.
Microphone arrays are the backbone of advanced sound algorithms. To run these algorithms effectively, the spacing between microphones must stay within a carefully defined range. If the distance is too wide or too narrow, the algorithms lose accuracy and stability, resulting in poor performance.
The Science Behind the Design
- Algorithm-driven requirements: Each algorithm has its own tolerance for microphone spacing. Beamforming, for example, depends on precise distance to steer audio “beams” accurately.
- Voice frequency range: Human speech typically falls between 300 Hz and 3.4 kHz. The wavelength of these frequencies sets physical limits on how microphones can be placed without causing aliasing or signal degradation.
Seeed ReSpeaker‘s Design
The ReSpeaker series has been engineered with these constraints in mind. For example, the minimum spacing between microphones on the ReSpeaker XMOS XVF3000 is 44 mm; with more powerful voice processing, the ReSpeaker XMOS XVF3800 increases the minimum spacing to 66 mm.

Microphone spacing and layout are optimized to maximize algorithm performance while keeping the array compact. This balance means users don’t have to sacrifice either audio quality or product flexibility. ReSpeaker are deliberately sized to fit into a wide variety of projects and enclosures. Whether you’re building a smart speaker, a conference device, or an embedded system, ReSpeaker are ready to integrate without compromise.
Bottom-Firing Microphone Design Update
What Is Bottom-firing
Seeed Studio’s new ReSpeaker XMOS XVF3800 adopts a bottom-firing microphone placement, which means the sound port is on the underside of the PCB, rather than on the top surface. This subtle change in layout brings meaningful benefits for performance, reliability, and product design.

Why It Matters
- Enhanced Protection
The downward-facing port is less exposed to dust, moisture, and accidental contact. This design improves durability and long-term reliability in real-world applications.
- Consistent Acoustic Paths
In microphone arrays, having all microphones share a uniform plane ensures consistent acoustic performance. Bottom-firing placement avoids interference from tall components on the PCB, enabling cleaner beamforming and more accurate algorithm processing.
- Built-in Noise Shielding
The PCB itself acts as a physical barrier against electrical and thermal noise generated by CPUs, power circuits, and other components. This natural shielding improves the signal-to-noise ratio at the hardware level.
- Simplified Product Design
Developers no longer need to design complex acoustic isolation structures or multiple housing openings to reduce interference. This cuts down on both cost and design complexity, while allowing for sleeker, cleaner product aesthetics. Sound can still be guided through bottom or side channels without compromising performance.
The latest ReSpeaker XMOS XVF3800 delivers superior voice capture, higher audio quality, and natural speech interaction—made possible through advanced algorithms, optimized hardware, and smart design.
The ReSpeaker family spans a range of options, from compact 2-mic arrays to versatile 4-mic arrays, giving users the flexibility to choose the right fit for their application.
Explore the lineup to find the solution that works best for you.