An Explanation of Audio Latency

In this post, I will talk about audio latency which is a fairly technical term that is often applied to audio transmission products. First of all, the term latency itself is usually fairly well understood by everybody. It means that there’s a certain delay. If something happens, something else will happen a certain amount of time after the first event occurred. The time difference is called latency.

Let’s talk about audio transmission. Audio transmission systems are fairly widespread and are sometimes built by using wires and sometimes are wireless. A wireless audio receiver that is used for these setups such as a Bluetooth receiver will output the audio is certain amount of time after it is fed into the transmitting unit. In case of Bluetooth, however, it is not easy to measure this latency because the audio is oftentimes not directly apply to a transmitter but instead being stream from an integrated device such as a cell phone.

However, for the sake of measurement, instead of using an integrated device, you can pick up a Bluetooth audio transmitter. This transmitter allows to connect an analog audio signal to its input and measure the difference in phase between the signal going into the transmitter and appearing at the receiver. This latency is typical even for integrated Bluetooth streaming devices from The question is, what is causing this latency. More interestingly, latency does also occur in transmission systems which use wires.

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First of all, the signal itself has to propagate which is dictated by the speed of light. Therefore, depending on the distance the signal has to travel, it will require a certain amount of time. Second of all, the signal will usually not travel directly but be converted into data inside the transmitter. This is done by a conversion process which uses filtering that also causes a certain delay. This process is then reversed inside the receiver unit. Another delay occurs there. In addition, most systems are using some sort of buffering. This buffering ensures that the receiver can request data packets which have not been properly received. However, for this purpose, both the transmitter and receiver will have to buffer data packets. This buffering will by definition incurred for the delay. As a result, the audio as it appears at the receiver will be delayed time efficiently cases. For practical applications, you should try to minimize the latency below 40 ms. However, in some applications such as when streaming music from a cell phone, latency doesn’t matter much because there’s no sync issue.

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