CODEWITHSUNDEEP
androidaudioaudiotrack
tmm1
tmm1

Reputation: 2115

How to use AudioTrack.getTimestamp() on Android to calculate latency?

I see many resources recommending that AudioTrack.getTimestamp() be used on modern Android versions to calculate audio latency for audio/video sync.

For instance:

https://stackoverflow.com/a/37625791/332798

https://developer.amazon.com/docs/fire-tv/audio-video-synchronization.html#section1-1

https://groups.google.com/forum/#!topic/android-platform/PoHfyNK54ps

However, none of these explain how to use the timestamp to calculate the latency? I'm struggling to figure what to do with the timestamp's framePosition/nanoTime to come up with a latency number.

Upvotes: 3

Views: 2489

Answers (1)

tmm1
tmm1

Reputation: 2115

So prior to this API, you would use AudioTrack.getPlaybackHeadPosition() which was just an approximation. Thus, to account for latency you had to offset that value with a latency value from one of two hidden methods: AudioManager.getOutputLatency() or AudioTrack.getLatency().

With the new AudioTrack.getTimestamp() API, you get a snapshot of the playhead position at a given time, taken directly at the output. As such, it is fully accurate and already accounts for device latency. Thus there's no need to call any other APIs now to add/remove latency.

The caveat is that this timestamp is only a snapshot, and the docs recommend you don't call this new method very often. So the trick to getting the "current" position is to use your last snapshot and linearly interpolate what the current value should be:

playheadPos = timestamp.framePosition + 
              (System.nanoTime() - timestamp.nanoTime) * samplerate / 1e9;

This position can then be compared against how many frames you've written into the AudioTrack, by maintaining another counter which increments every time AudioTrack.write() completes:

int bytesWritten = track.write(...);
writtenPos += bytesWritten / pcmFrameSize;

If you're working with ENCODING_AC3, the playhead position reported by AudioTrack is still in terms of samples. You will either need to convert it to bytes, or convert the number of bytes you've written in back into samples. Either way, you will need to know the bitrate of your AC3 stream (i.e. 384000bps)

int bytesWritten = track.write(...);
writtenPos += bytesWritten * samplerate / (bitrate / 8);

Upvotes: 5

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