Now, in order for this to work the latency has to be repeatable and I think this is the bigger issue here. This is more reliable as it captures the latency of the entire signal chain. My ASBD uses kAudioFormatFlagsCanonical and kAudioFormatLinearPCM as format. The typical way Ive seen DAWs find the latency of an audio interface is to have the user loopback their channels and play a test signal.
![audio loopback latency test audio loopback latency test](https://dt7v1i9vyp3mf.cloudfront.net/styles/news_large/s3/imagelibrary/p/pcmusician1-0105-WfSjh.CHli7G9Vf4fjYdMqnhW9GVUbxk.jpg)
I was expecting 20ms or lower latency, but clearly the result gave me 50~60ms. This test is not very useful for detailed analysis by itself because of the nature of the test, but it can be useful for calibrating other tests, and for establishing an upper bound. This provides a crude measure of combined output and input latency by timing an impulse response loop. So that the error should mainly come from the peak distance measurement, which I assume should be much smaller than the audio pipeline latency and can be ignored. One of the easiest latency tests is an audio feedback (Larsen effect) test.
#Audio loopback latency test mac#
This was by no means a super accurate measurement, but at least the innate latency of the Mac recording pipeline should have been cancelled out this way.
![audio loopback latency test audio loopback latency test](https://d154hy2b5aa41t.cloudfront.net/course_covers/7d3a37ce-ecef-4edb-b7cb-c340361f07f7/512x512/f9565bec1c57ddaf2f6c247a4d3fd338.512.512.jpg)
Finally I simply visually observe the waveform in Audacity's recorded track and measure the time interval between the peaks of the two recorded snaps. Then, with Audacity in recording mode, the Mac would pick up both the sound from my fingers and its "clone" from the iOS speaker in close range. The only listener to the iOS output (through a speaker connected via
#Audio loopback latency test android#
In order to compute latency for your own audio device, you need to connect the audio out and audio in ports using a loopback cable. Audio loopback dongle to test audio latency on android phones Gallery. The best way to get low latency is to 1) set your audio interface buffers as low as will work - 48 samples or lower if you can, and 2) set your Audio FrameSize to 2.5ms - do this in JK Manage/Audio Settings/Audio Booster. Measuring Latency with audioLatencyMeasurementExampleApp.m That is, the latency incurred when playing audio through a device, looping back the audio with a physical loopback cable, and recording the loopback audio with the same audio device. In addition to potentially increasing latency, the amount of processing involved in the audio algorithm can also cause dropouts. A better test signal would be a short audio sample of maybe a cowbell or other click sound. However, the tradeoff is a higher chance of dropouts occurring (overruns/underruns). I think it used to happen when there was a big difference in ASIO input and output latency figures The loopback test could be of interest when testing the ping value for external FX. Smaller frame sizes and higher sampling rates reduce the roundtrip latency.
![audio loopback latency test audio loopback latency test](https://i.ytimg.com/vi/6sCLzzZhdfQ/maxresdefault.jpg)
![audio loopback latency test audio loopback latency test](https://source.android.com/devices/audio/images/round_trip.png)
However, the latency involved should be the same either way provided the other factors (frame size, sampling rate, algorithm latency) don't change. Typically the processing chain consists of recording audio, processing it, and playing the processed audio. Also, most practical applications will not use a loopback setup. It measures only the combined effect of the two. Roundtrip latency does not break down the measurement between output latency and input latency.