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Pulse density modulation (PDM) represents a signal with a rapid stream of binary bits: the proportion of high pulses over time corresponds to the signal’s amplitude. In a PDM audio microphone, a receiving peripheral or codec filters and decimates that high-rate stream into pulse-code modulation (PCM) samples for conventional audio processing.
What pulse density modulation means
PDM is a way to encode a changing signal as a sequence of two binary levels. An individual bit does not specify an amplitude value. Instead, the relative density of high pulses across time carries that information: a higher signal level corresponds to a greater proportion of high pulses, and a lower level to a smaller proportion.
Infineon’s PDM_PCM_v2 technical reference explains that PDM uses relative pulse density to represent analog amplitude, rather than encoding amplitude values as weighted pulse codewords as PCM does.
How PDM becomes audio samples
A PDM MEMS microphone converts sound into a clocked, high-rate one-bit stream. The host processor’s audio peripheral or a codec receives that stream, applies a digital filter, and decimates it—reducing the sample rate—to produce PCM audio samples. Those multibit samples can then be handled by ordinary digital-audio software and interfaces.
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For example, Nordic Semiconductor documents an nRF5340 PDM peripheral with hardware decimation filters and PCM output. That product family supports up to two microphones as a left/right pair; this is a feature of the documented implementation, not a guarantee about every PDM microphone or host interface. See the nRF5340 product specification for its specific behavior.
PDM and PCM compared
| Aspect | PDM | PCM |
|---|---|---|
| How amplitude is represented | Density of one-bit pulses over time | Quantized multibit sample values |
| Typical audio data pattern | High-rate bitstream, commonly oversampled | Sample values at the selected audio sample rate |
| What conversion is needed for ordinary PCM workflows | Filtering and decimation | Already represented as sample values |
| Key design considerations | PDM clock, channel timing, filter and decimator | Sample rate, sample width and audio data format |
These formats describe information differently; neither is universally better. PDM is useful at the microphone interface, while PCM is the multibit sample representation used by many audio-processing workflows.
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Why PDM uses a fast clock
Audio PDM is oversampled: its bit clock runs much faster than the resulting audio sample rate so that filtering and decimation can recover the signal. The required clock depends on the microphone, the target sample rate, and the receiving peripheral’s clock-divider and decimation options.
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What to check when connecting a PDM microphone
A PDM microphone is not interchangeable with a PCM-output microphone merely because both are digital. Before selecting a part or wiring a prototype, check the interface requirements at both ends:
- Clock: Confirm the microphone’s supported PDM clock range and that the host can generate a suitable clock.
- Data and voltage: Verify pinout, logic voltage, and electrical requirements in the microphone and development-board documentation.
- Audio settings: Match the desired output sample rate with the peripheral’s clock divider, filter, and decimation settings.
- Channels: For two microphones, check how the devices distinguish left and right data, including any clock-edge or channel timing requirements. Do not assume all PDM interfaces use the same method.
NXP’s i.MXRT600 PDM MEMS Microphone Audio Path Optimal Settings identifies the Knowles SPH0641LM4H-1 as an example of a PDM MEMS microphone. Treat that as an example part, not a statement about current availability or compatibility with a particular host board.
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PDM can also produce an analog output
PDM is not limited to microphone input. In a documented Microchip DAC implementation, a module generates a PDM pulse stream whose density corresponds to a data-register value. A multistage resistor-capacitor (RC) filter smooths the pulses into a voltage. This illustrates the same principle in the opposite direction: filtering the binary stream yields an analog level.
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