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