What Is Audio Latency? Meaning, Causes, Measurement & How to Reduce It

Audio latency is the time shift that occurs as audio signals pass through an audio processing system: during transmission, processing, and playback. Audio latency inevitably degrades the user experience, especially in remote meetings. In this article, we'll explain what audio latency is, its causes, measurement methods, and ways to reduce it.

Contents

1. What is audio latency?

2. How does audio latency occur?

3. Comparing audio latency in different systems

4. How to measure audio latency?

5. How to reduce audio latency?

6. Conclusion

1. What is audio latency?

Audio latency is the time shift between a sound source (such as a microphone input or a piano key press) and the final audio output (the sound played through a speaker), typically measured in milliseconds (ms).

Sound appears to travel instantaneously. However, in any modern system, audio goes through several stages: a microphone captures the sound; the analog signal is converted to digital data; the digital audio is processed for data transmission (local or network); the audio is decoded to convert the digital signal back to analog; and finally, the speaker plays the sound. Each step adds a small delay. These delays accumulate to form the overall audio delay.

A common example of audio delay occurs when watching video: the speaker's lips move before the sound is heard. This misalignment is the audio delay.

2. How does audio latency occur?

Audio latency is the cumulative time shift caused by many small delays along the entire audio signal path—from the moment the sound is generated to the moment it is perceived by the listener. These small delays accumulate at various stages of processing.

1. Input Latency

This is the first stage where sound enters a digital system.

Microphone: Sound waves (air pressure fluctuations) are converted by the microphone into analog electrical signals. This step is extremely fast—typically taking only microseconds.

ADC (Analog-to-Digital Converter): The analog signal is converted into digital data. This takes some time depending on the sampling rate, bit depth, and internal buffering. Latency typically ranges from less than 1 ms to several milliseconds. It is the unavoidable initial delay in any digital audio system.

2. Processing Latency

This stage often contributes the largest share of overall latency and occurs in the digital domain.

Digital Signal Processing (DSP): As audio passes through the DSP for noise reduction, echo cancellation, equalization, mixing, and other functions, each algorithm requires computational time and buffering. The more complex the processing chain, the higher the latency—typically from a few milliseconds to tens of milliseconds.

Encoding/Decoding: For network transmission or storage, audio is encoded and compressed (for example, using OPUS or AAC). Encoders must accumulate a certain number of audio samples to form a "frame" before processing. This frame buffering adds latency. Low-latency codecs use shorter frames to reduce latency.

3. Transmission Latency

This occurs when sending audio from one location to another. Network Transmission: In IP audio systems, audio packets are transmitted over networks such as Ethernet or the internet. Latency depends on network speed, the number of transit nodes, switching, and congestion. A local network can add only a few milliseconds, while wide area networks or cloud routing can add tens to hundreds of milliseconds.

Jitter Buffer: Since network packets don't always arrive at regular intervals (jitter), the receiver uses a jitter buffer to temporarily store the data and play it back evenly. This improves stability but directly introduces additional latency.

4. Output Latency

This is the final stage where the audio is converted back into sound.

Decoding: If the audio has been compressed, it must be decoded before playback. This introduces a processing delay similar to encoding (usually smaller).

DAC (Digital-to-Analog Converter): The digital signal is converted back to analog. Like an ADC, this adds a small, fixed latency.

Loudspeaker and Sound Propagation: An analog signal drives a loudspeaker to produce sound. The mechanical movement of the cone adds a slight delay. The sound then travels through the air to the listener. Each additional meter of distance adds about 3 ms of delay—this becomes noticeable in large halls.

5. Other Sources of Latency

In addition to the main stages, small delays are also introduced by:

• Physical interfaces (e.g., USB or Bluetooth);

• Audio scheduling and buffering in the operating system;

• Audio pipelines within specific applications.

Audio delay does not have a single source. It is the sum of many small delays along the entire signal chain—from input, processing, and transmission to the final output. Each stage adds a small amount, and together they form the overall audio delay.

3. Comparing Audio Delay in Different Systems

There is no single, fixed standard for acceptable audio latency. It depends on both the human hearing threshold and the functional requirements of a specific application.

According to industry consensus, the human ear has the following distinct perception thresholds:

• < 20 ms: practically unnoticeable

• 20–50 ms: slightly noticeable

• > 100 ms: clearly noticeable

• > 150 ms: disrupts the natural flow of conversation

Conference systems: Natural dialogue is critical. The recommended latency is below 50 ms, with the optimal target being below 30 ms. Higher latency causes interruptions, awkward pauses, and echo.

Intercom systems: Designed for real-time communication, latency should be as low as possible—typically below 100 ms, ideally below 50 ms. High latency makes conversations slow and unreliable.

Public address (PA) systems: Generally tolerate higher latency. Latencies above 100 ms are often acceptable. However, a key requirement is a consistent delay between zones to ensure synchronicity of announcements.

Audiovisual security systems: The primary requirement is synchronicity of audio and video. With a latency above 100 ms, users clearly notice a mismatch between audio and video.

The acceptability of latency depends on the project requirements and the application scenario. As a general rule, the lower the latency, the better—especially for real-time communications and security systems.

4. How to measure audio latency?

In public address, sound reinforcement, and intercom systems, latency affects more than just audio quality. It directly impacts speech intelligibility, emergency response times, and multi-zone synchronization. Accurate measurement is critical for system design, commissioning, and acceptance testing.

Objective: To measure the internal processing latency of a single device, such as a DSP processor, power amplifier, or codec.

Method:

1. Using a signal generator, apply a short pulse (e.g., a positive 1 kHz pulse) to the device under test.

2. Split the input signal and connect it to channel 1 of the oscilloscope.

3. Connect the device output to channel 2 of the oscilloscope.

4. Measure the time difference (ΔT) between the rising edges of the two pulses. This value is the device's accurate latency.

Industry applications: Comparing latency characteristics of different brands of DSPs or audio processors; generating reference latency data for each component in large systems for subsequent global timing calibration.

Method 2: Measuring Digital Audio Loop (Round-Trip) Latency

Objective: To measure the total latency of the entire digital audio chain, including the computer, software, and audio interface. Method:

1. Use a professional USB or Thunderbolt audio interface with known specifications.

2. Connect one output (e.g., Line Out) directly to one input (Line In) with a physical cable.

3. Run specialized tools (e.g., RTL Utility or command-line tools for loopback). The software automatically sends pulses and measures the round-trip time.

4. The resulting output includes ADC latency, DAC latency, and computer buffer latency.

Industry applications: Evaluating the latency of media servers used for background music or emergency audio; testing the recording output latency of digital mixers for synchronization with live sound.

Method 3: Acoustic End-to-End Measurement

Goal: Measure the complete acoustic chain—from microphone input to speaker output—under real-world conditions.

Method:

1. At the front end of the system (in the control room), use measurement equipment (e.g., a computer running a signal generator application) to inject a short pulse into the mixer or DSP.

2. Place a measurement microphone near the final loudspeaker and connect it to another recording device (or a separate input channel outside the system under test).

3. Compare the original signal with the waveform recorded by the microphone and measure the time difference.

Note: The result includes the air-propagation time of the sound (approximately 3 ms per meter).

Industry Applications: Measuring the total intercom delay from "press PTT and speak" to "perception by the remote party"; testing the delay of a PA system from the signal input to the farthest loudspeaker in large venues.

Method 4: Multi-Zone Timing Measurement

Goal: Ensure that announcements are played simultaneously in all zones (on different floors or in different buildings) of large PA systems, eliminating echo and confusion. Method:

1. Send a global broadcast test signal.

2. Place network recorders in each key zone (or use multiple synchronized measurement microphones connected to a central analyzer).

3. Record all points simultaneously and align them on the software timeline to check for waveform offsets.

4. Adjust DSP delay settings in each zone until all outputs are synchronized.

Industry Application: A key step in commissioning large distributed PA systems in stadiums, airports, and educational institutions.

5. How to reduce audio latency?

Audio latency can occur during encoding, DSP processing, network transmission, and depending on the hardware architecture.

Codec Optimization

Key point: for low-latency IP audio, always prioritize PCM or OPUS and keep the codec frame length as short as possible. Consumer codecs like MP3 and AAC are optimized for quality and storage, not real-time performance.

Optimizing DSP Processing

Each DSP block adds buffering and processing time. Simplifying signal chains is one of the most effective ways to reduce DSP latency.

Key point: shorter DSP chains mean faster audio. For real-time communications, avoid excessive filtering, multi-layer mixing, and overly complex AEC settings.

Network Optimization

In IP audio systems, network design directly impacts transmission latency and jitter buffering.

Key point: proper network design is essential for reducing audio latency. VLAN + QoS + fewer transit nodes often yield a greater effect than changing codecs.

Hardware Architecture Optimization

Hardware architecture matters. Professional IP audio devices are designed specifically for real-time operation.

Key point: Professional low-latency IP audio systems use SIP architecture, local processing, edge DSP, and hardware acceleration to achieve predictable performance in real time.

6. Conclusion

Audio latency not only degrades listening quality—noticeable delays in critical meetings can disrupt communication, reducing decision-making and team collaboration.

To address this challenge, SPON introduces the LCN-8000 digital conference system, designed for large conference rooms and multi-microphone configurations. The main unit supports up to 128 simultaneous conference microphones using a robust architecture focused on real-time transmission. Wired microphones provide an end-to-end latency of just 7 ms, ensuring natural and synchronized speech transmission for smoother and more reliable communication.