<?xml version="1.0" encoding="utf-8" standalone="yes"?><rss version="2.0" xmlns:atom="http://www.w3.org/2005/Atom"><channel><title>Audio &amp; Acoustic Sensors on Embedded Systems Development</title><link>https://applied-ee.github.io/embedded/docs/sensor-integration/audio-and-acoustic/</link><description>Recent content in Audio &amp; Acoustic Sensors on Embedded Systems Development</description><generator>Hugo</generator><language>en-us</language><atom:link href="https://applied-ee.github.io/embedded/docs/sensor-integration/audio-and-acoustic/index.xml" rel="self" type="application/rss+xml"/><item><title>MEMS Microphone Selection &amp; Interfacing</title><link>https://applied-ee.github.io/embedded/docs/sensor-integration/audio-and-acoustic/mems-microphone-interfacing/</link><pubDate>Mon, 01 Jan 0001 00:00:00 +0000</pubDate><guid>https://applied-ee.github.io/embedded/docs/sensor-integration/audio-and-acoustic/mems-microphone-interfacing/</guid><description>&lt;h1 id="mems-microphone-selection--interfacing"&gt;MEMS Microphone Selection &amp;amp; Interfacing&lt;a class="anchor" href="#mems-microphone-selection--interfacing"&gt;#&lt;/a&gt;&lt;/h1&gt;
&lt;p&gt;MEMS microphones have largely replaced electret condenser microphones (ECMs) in embedded systems. The key advantage is integration: a MEMS microphone contains the acoustic transducer, a charge amplifier, and — in digital variants — a sigma-delta ADC and digital interface, all in a single surface-mount package typically 3.5 x 2.65 x 1 mm. The result is a microphone that solders directly to a PCB with no external biasing components, no analog signal routing concerns, and consistent part-to-part sensitivity. Understanding the output format (analog, PDM, or I2S), the critical specifications, and the power supply requirements determines whether a given MEMS microphone will deliver clean audio or an unusable noise floor.&lt;/p&gt;</description></item><item><title>PDM &amp; I2S Audio Interfaces</title><link>https://applied-ee.github.io/embedded/docs/sensor-integration/audio-and-acoustic/pdm-and-i2s-audio-interfaces/</link><pubDate>Mon, 01 Jan 0001 00:00:00 +0000</pubDate><guid>https://applied-ee.github.io/embedded/docs/sensor-integration/audio-and-acoustic/pdm-and-i2s-audio-interfaces/</guid><description>&lt;h1 id="pdm--i2s-audio-interfaces"&gt;PDM &amp;amp; I2S Audio Interfaces&lt;a class="anchor" href="#pdm--i2s-audio-interfaces"&gt;#&lt;/a&gt;&lt;/h1&gt;
&lt;p&gt;Streaming audio data from a digital microphone to an MCU requires a well-defined serial protocol. Two dominate embedded audio: PDM (Pulse Density Modulation), which carries a 1-bit sigma-delta bitstream at megahertz clock rates, and I2S (Inter-IC Sound), which carries multi-bit PCM samples in a synchronous frame format. The choice between them affects peripheral selection, CPU load, memory bandwidth, and firmware complexity. PDM is simpler in hardware (two wires) but requires decimation filtering that consumes either a dedicated hardware peripheral or significant CPU cycles. I2S delivers ready-to-use PCM samples but requires three signal lines and a more complex peripheral configuration.&lt;/p&gt;</description></item><item><title>Analog Microphone Front-Ends</title><link>https://applied-ee.github.io/embedded/docs/sensor-integration/audio-and-acoustic/analog-microphone-front-ends/</link><pubDate>Mon, 01 Jan 0001 00:00:00 +0000</pubDate><guid>https://applied-ee.github.io/embedded/docs/sensor-integration/audio-and-acoustic/analog-microphone-front-ends/</guid><description>&lt;h1 id="analog-microphone-front-ends"&gt;Analog Microphone Front-Ends&lt;a class="anchor" href="#analog-microphone-front-ends"&gt;#&lt;/a&gt;&lt;/h1&gt;
&lt;p&gt;Electret condenser microphones (ECMs) remain relevant in embedded systems despite the rise of MEMS alternatives. ECMs are inexpensive ($0.10-0.50), available in a wide range of form factors (6 mm capsules through large-diaphragm studio types), and produce an analog signal that can be read by any MCU with an ADC channel. The challenge is that the raw output from an ECM is tiny — typically 5-50 mV peak for normal speech — and rides on a DC bias that must be stripped before amplification. A proper analog front-end includes biasing, coupling, preamplification, DC offset for single-supply operation, and anti-alias filtering before the ADC. Each stage has concrete design choices that determine whether the captured audio is clean or buried in noise.&lt;/p&gt;</description></item><item><title>Ultrasonic Transducers &amp; Ranging</title><link>https://applied-ee.github.io/embedded/docs/sensor-integration/audio-and-acoustic/ultrasonic-transducers/</link><pubDate>Mon, 01 Jan 0001 00:00:00 +0000</pubDate><guid>https://applied-ee.github.io/embedded/docs/sensor-integration/audio-and-acoustic/ultrasonic-transducers/</guid><description>&lt;h1 id="ultrasonic-transducers--ranging"&gt;Ultrasonic Transducers &amp;amp; Ranging&lt;a class="anchor" href="#ultrasonic-transducers--ranging"&gt;#&lt;/a&gt;&lt;/h1&gt;
&lt;p&gt;Ultrasonic ranging measures distance by timing the round-trip of a sound pulse — the same principle as sonar and radar, scaled down to centimeter-resolution distances using 40 kHz piezoelectric transducers. The technique is conceptually simple: transmit a burst, start a timer, listen for the echo, and compute distance from the elapsed time. The implementation details, however, involve transmit driver circuits, receive amplifier chains, envelope detection, blanking intervals, and careful timing — all within firmware running on a general-purpose MCU. The HC-SR04 module hides most of this complexity behind a trigger/echo interface, but understanding the underlying signal chain is essential for building custom ranging systems with better performance, multiple channels, or integration into space-constrained designs.&lt;/p&gt;</description></item></channel></rss>