Pulse-density modulation (PDM) audio explained in a quick primer

Pulse-density modulation (PDM) is a compact digital audio format used in devices like MEMS microphones and embedded systems. This compact primer eases you into the essentials of PDM audio.

Let’s begin by revisiting a ubiquitous PDM MEMS microphone module based on MP34DT01-M—an omnidirectional digital MEMS audio sensor that continues to serve as a reliable benchmark in embedded audio design.

Figure 1 A MEMS microphone mounted on a minuscule module detects sound and produces a 1-bit PDM signal. Source: Author

When properly implemented, PDM can digitally encode high-quality audio while remaining cost-effective and easy to integrate. As a result, PDM streams are now widely adopted as the standard data output format for MEMS microphones.

On paper, the anatomy of a PDM microphone boils down to a few essential building blocks like:

  • MEMS microphone element, typically a capacitive MEMS structure, unlike the electret capsules found in analog microphones.
  • Analog preamplifier boosts the low-level signal from the MEMS element for further processing.
  • PDM modulator converts the analog signal into a high-frequency, 1-bit pulse-density modulated stream, effectively acting as an integrated ADC.
  • Digital interface logic handles timing, clock synchronization, and data output to the host system.

Next is the function block diagram of T3902, a digital MEMS microphone that integrates a microphone element, impedance converter amplifier, and fourth-order sigma-delta (Σ-Δ) modulator. Its PDM interface enables time multiplexing of two microphones on a single data line, synchronized by a shared clock.

Figure 2 Functional block diagram outlines the internal segments of the T3902 digital MEMS microphone. Source: TDK

The analog signal generated by the MEMS sensing element in a PDM microphone—sometimes referred to as a digital microphone—is first amplified by an internal analog preamplifier. This amplified signal is then sampled at a high rate and quantized by the PDM modulator, which combines the processes of quantization and noise shaping. The result is a single-bit output stream at the system’s sampling rate.

Noise shaping plays a critical role by pushing quantization noise out of the audible frequency range, concentrating it at higher frequencies where it can be more easily filtered out. This ensures relatively low noise within the audio band and higher noise outside it.

The microphone’s interface logic accepts a master clock signal from the host device—typically a microcontroller (MCU) or a digital signal processor (DSP)—and uses it to drive the sampling and bitstream transmission. The master clock determines both the sampling rate and the bit transmission rate on the data line.

Each 1-bit sample is asserted on the data line at either the rising or falling edge of the master clock. Most PDM microphones support stereo operation by using edge-based multiplexing: one microphone transmits data on the rising edge, while the other transmits on the falling edge.

During the opposite edge, the data output enters a high-impedance state, allowing both microphones to share a single data line. The PDM receiver is then responsible for demultiplexing the combined stream and separating the two channels.

As a side note, the aforesaid microphone module is hardwired to treat data as valid when the clock signal is low.

The magic behind 1-bit audio streams

Now, back in the driveway. PDM is a clever way to represent a sampled signal using just a stream of single bits. It relies on delta-sigma conversion, also known as sigma-delta, and it’s the core technology behind many oversampling ADCs and DACs.

At first glance, a one-bit stream seems hopelessly noisy. But here is the trick: by sampling at very high rates and applying noise-shaping techniques, most of that noise is pushed out of the audible range—above 20 kHz—where it no longer interferes with the listening experience. That is how PDM preserves audio fidelity despite its minimalist encoding.

There is a catch, though. You cannot properly dither a 1-bit stream, which means a small amount of distortion from quantization error is always present. Still, for many applications, the trade-off is worth it.

Diving into PDM conversion and reconstruction, we begin with the direct sampling of an analog signal at a high rate—typically several megahertz or more. This produces a pulse-density modulation stream, where the density of 1s and 0s reflects the amplitude of the original signal.

Figure 3 An example that renders a single cycle of a sine wave as a digital signal using pulse density modulation. Source: Author

Naturally, the encoding relies on 1-bit delta-sigma modulation: a process that uses a one-bit quantizer to output either a 1 or a 0 depending on the instantaneous amplitude. A 1 represents a signal driven fully high, while a 0 corresponds to fully low.

And, because the audio frequencies of interest are much lower than the PDM’s sampling rate, reconstruction is straightforward. Passing the PDM stream through a low-pass filter (LPF) effectively restores the analog waveform. This works because the delta-sigma modulator shapes quantization noise into higher frequencies, which the low-pass filter attenuates, preserving the desired low-frequency content.

Inside digital audio: Formats at a glance

It goes without saying that in digital audio systems, PCM, I²S, PWM, and PDM each serve distinct roles tailored to specific needs. Pulse code modulation (PCM) remains the most widely used format for representing audio signals as discrete amplitude samples. Inter-IC Sound (I²S) excels in precise, low-latency audio data transmission and supports flexible stereo and multichannel configurations, making it a popular choice for inter-device communication.

Though not typically used for audio signal representation, pulse width modulation (PWM) plays a vital role in audio amplification—especially in Class D amplifiers—by encoding amplitude through duty cycle variation, enabling efficient speaker control with minimal power loss.

On a side note, you can convert a PCM signal to PDM by first increasing its sample rate (interpolation), then reducing its resolution to a single bit. Conversely, a PDM signal can be converted back to PCM by reducing its sampling rate (decimation) and increasing its word length. In both cases, the ratio of the PDM bit rate to the PCM sample rate is known as the oversampling ratio (OSR).

Crisp audio for makers: PDM to power simplified

Cheerfully compact and maker-friendly PDM input Class D audio power amplifier ICs simplify the path from microphone to speaker. By accepting digital PDM signals directly—often from MEMS mics—they scale down both complexity and component count. Their efficient Class D architecture keeps the power draw low and heat minimal, which is ideal for battery-powered builds.

That is to say, audio ICs like MAX98358 require minimal external components, making prototyping a pleasure. With filterless Class D output and built-in features, they simplify audio design, freeing makers to focus on creativity rather than complexity.

Sidewalk: For those eager to experiment, ample example code is available online for SoCs like the ESP32-S3, which use a sigma-delta driver to produce modulated output on a GPIO pin. Then with a passive or active low-pass filter, this output can be shaped into clean, sensible analog signal.

Well, the blueprint below shows an active low-pass filter using the Sallen & Key topology, arguably the simplest active two-pole filter configuration you will find.

Figure 4 Circuit blueprint outlines a simple active low-pass filter. Source: Author

Echoes and endings

As usual, I feel there is so much more to cover, but let’s jump to a quick wrap-up.

Whether you are decoding microphone specs or sketching out a signal chain, understanding PDM is a quiet superpower. It is not just about 1-bit streams; it’s about how digital sound travels, transforms, and finds its voice in your design. If this primer helped demystify the basics, you are already one step closer to building smarter, cleaner audio systems.

Let’s keep listening, learning, and simplifying.

T. K. Hareendran is a self-taught electronics enthusiast with a strong passion for innovative circuit design and hands-on technology. He develops both experimental and practical electronic projects, documenting and sharing his work to support fellow tinkerers and learners. Beyond the workbench, he dedicates time to technical writing and hardware evaluations to contribute meaningfully to the maker community.

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