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Re: Round Trip Latency test

From: Norbert Schnell <Norbert.Schnell@ircam.fr>
Date: Tue, 2 Sep 2014 23:29:42 +0200
Cc: Chris Wilson <cwilso@google.com>, Alex Russell <slightlyoff@google.com>, public-audio@w3.org
Message-Id: <44FEE812-F31A-4693-954E-A1CC52E0ED6A@ircam.fr>
To: Stephen Band <stephband@cruncher.ch>
Hi Stephen,

These two lines seem to assume that the latency through the ScriptProcessorNode is twice its buffer size:
  - 89: outputTimes.push(bufferLength * frame + bufferLength);
  - 104: inputTimes.push(bufferLength * frame + n - bufferLength);

Is that really always the case?

The only idea I had so far to reliably measure the latency through the ScriptProcessorNode was to send a sort of time code (e.g. a ramp counting the samples of a second generated via an AudioBufferSourceNode) through a ScriptProcessorNode (that just copies the input into the output) and to compare (e.g. subtract) the signal coming out of the node with the signal going into the node. 
I guess, one would need an additional ScriptProcessorNode to look at the resultů

(BTW: The time code can also be used to see if frames get lost on the way into the audio process code.)

Norbert

On 02 Sep 2014, at 22:59, Stephen Band <stephband@cruncher.ch> wrote:
> Hey Stephen,
>  
> do you have an un-minimized version of your code?
>  
> Did you see this? The meat and potatoes of the code is in this gist:
> 
> https://gist.github.com/stephband/f032a69c54f3a5d0ebf9
> 
> 
> Bjorn Melinder also had a look at it. I didn't get chance to look at what he said in his latest two posts, yet.
> 
> 
> 
> If you're hopping process boundaries, and you usually are, you'll need to double-buffer.  That's 6ms.  The input has the same buffering - so you're up to 12ms.  And that's an idealized path...)  This is why even pro audio hardware frequently has a "direct pass-through"... :)
> For sound.io I'm working on a looper that compensates for latency, so that you can use the direct channel of a pro audio interface and process loops through sound.io, yet still expect the loops to stay in time with you. 
> 
> I just tested my Metric Halo and I'm getting 38ms at 48000. That's not as good as I thought it was, there may well be some mistakes in the assumptions I've made in the code.
> 
> Stephen.
> 
> 
> 
> On 2 Sep 2014 22:29, "Chris Wilson" <cwilso@google.com> wrote:
> Hey Stephen,
> do you have an un-minimized version of your code?  I can't understand how you're accounting for the inherent ScriptProcessor latency.  I also didn't see a clear 2x drop when I doubled my sample rate, which I wanted to investigate.
> 
> The design of the Web Audio API was intended to provide low-latency in audio; realistically, <10ms is hard to do without an optimized audio path *and* a high sample rate.  (A single 128-sample block at 44.1kHz is just under 3ms.  If you're hopping process boundaries, and you usually are, you'll need to double-buffer.  That's 6ms.  The input has the same buffering - so you're up to 12ms.  And that's an idealized path...)  This is why even pro audio hardware frequently has a "direct pass-through"... :)
> 
> 
> 
> 
> On Sat, Aug 30, 2014 at 2:21 PM, Alex Russell <slightlyoff@google.com> wrote:
> On Sat, Aug 30, 2014 at 12:46 PM, Stephen Band <stephband@cruncher.ch> wrote:
> It's nothing to do with the UI really.
> 
> 
> I understand that this wasn't in any way a test of UI, but in terms of the goal of reducing latency, I'd have assumed that being able to match UI closely (in response to input, e.g.) would be a goal and impls are some distance of that (although we also have bad delay in touch inputs for various reasons that are boring).
> You're doing well if you get less than 40ms out of a standard sound card, but if you use a good external audio interface you could see as low as 5ms.
> 
> Above 15-20ms is when the ear starts to hear two distinct sounds, although it can be uncomfortable to sing and monitor with a latency of >10ms.
> 
> Thanks for the context. 
> So I would say a good latency would be <10ms. But good luck getting there :)
> 
> Looks like we're gonna need it = )
>  
> On 30 Aug 2014 21:21, "Alex Russell" <slightlyoff@google.com> wrote:
> What's a "good" number for this? I'm assuming less than a UI frame (16ms) is preferred? I'm seeing ~50ms on Chrome Dev/OS X/MBP and FF doesn't seem to detect all of the signals in my view.
> 
> 
> On Sat, Aug 30, 2014 at 11:24 AM, Stephen Band <stephband@cruncher.ch> wrote:
> In case someone should find it useful, here's a round-trip latency tester:
> 
> https://sound.io/latency/
> 
> 
> 
> 
> 
Received on Tuesday, 2 September 2014 21:30:13 UTC

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