Re: Round Trip Latency test

Ah yes, you're right. I was assuming that the output buffer starts to play
at the same time as the input buffer gets filled with new data onaudioprocess.
But you're right, perhaps that's a totally erroneous assumption.

I suppose another way to do it would be to record a bunch of echos after a
single impulse and measure the time between them. You could do that with a
single script processor.



On 2 September 2014 23:29, Norbert Schnell <Norbert.Schnell@ircam.fr> wrote:

> 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
> result…
>
> (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 <https://gist.github.com/stephband/f032a69c54f3a5d0ebf9>://
> <https://gist.github.com/stephband/f032a69c54f3a5d0ebf9>gist.github.com
> <https://gist.github.com/stephband/f032a69c54f3a5d0ebf9>
> /stephband/f032a69c54f3a5d0ebf9
> <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/
>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>
>>
>


-- 

Cheers,
Stephen


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