Re: Web Audio API sequencer capabilities

A gain node's gain parameter effectively serves as an envelope node if you feed a unity signal to the gain node. This has gotten really expressive particularly after connect() began supporting AudioParams as targets. Do you have a use case in mind that cannot be covered by such a gain node that would be covered by an envelope node?

-Kumar

On 4 Oct, 2012, at 1:58 AM, Jussi Kalliokoski <jussi.kalliokoski@gmail.com> wrote:

> Let me be more specific, do you think the envelope functionality being in the AudioParam is more powerful than if it were in a separate node? If you do, why? What is the advantage it offers?
> 
> Cheers,
> Jussi
> 
> On Wed, Oct 3, 2012 at 8:40 PM, Chris Rogers <crogers@google.com> wrote:
> 
> 
> On Wed, Oct 3, 2012 at 12:59 AM, Jussi Kalliokoski <jussi.kalliokoski@gmail.com> wrote:
> Chris (Rogers), could I get your opinion regarding the introducing an envelope node and simplifying the AudioParam?
> 
> AudioParam has been designed with lots of care and thought for implementing envelopes, so I believe it's in a very good spot right now.  As an example of how people are using these envelope capabilities in sequencer applications, here's a good example from Patrick Borgeat:
> https://dl.dropbox.com/u/15744891/www1002/macro_seq_test1002.html
> 
> Chris
> 
> 
>  
> 
> Cheers,
> Jussi
> 
> 
> On Sat, Aug 25, 2012 at 8:19 AM, Srikumar Karaikudi Subramanian <srikumarks@gmail.com> wrote:
> > This would be a very basic setup, but with the current API design there are some hard problems to solve here. The audio is relatively easy, regardless of whether it's coming from an external source or not. It's just a source node of some sort. The sequencing part is where stuff gets tricky.
> 
> Yes it does appear tricky, but given that scheduling with native nodes suffices mostly, it seems to me that the ability to schedule JS audio nodes using noteOn/noteOff (renamed now as start/stop), together with dynamic lifetime support solves the scheduling problems completely. Such scheduling facility need only be present for JS nodes that have no inputs - i.e. are source nodes.
> 
> We (at anclab) were thinking about similar scheduling issues within the context of building compose-able "sound models" using the Web Audio API. A prototype framework for this purpose that we built (http://github.com/srikumarks/steller) will generalize if JS nodes can be scheduled similar to buffer source nodes and oscillators. A bare bones example of using the framework is available here - http://srikumarks.github.com/steller .
> 
> "Steller" is intended for interactive high level sound/music models (think foot steps, ambient music generators and the like) and so doesn't have time structures that are editable or even a "play position" as a DAW would require, but it may be possible to build them atop/beside Steller. At the least, it suggests the sufficiency of the current scheduling API for native nodes.
> 
> Best,
> -Kumar
> 
> On 21 Aug, 2012, at 11:28 PM, Jussi Kalliokoski <jussi.kalliokoski@gmail.com> wrote:
> 
> > Hello group,
> >
> > I've been thinking about how to use the Web Audio API to write a full-fledged DAW with sequencing capabilities (e.g. MIDI), and I thought I'd share some thoughts and questions with you.
> >
> > Currently, it's pretty straight-forward to use the Web Audio API to schedule events in real time, which means it would play quite well together with other real time APIs, such as the Web MIDI API. For example, you can just schedule an audiobuffer to play whenever a noteon event is received from a MIDI source.
> >
> > However, here's something of a simple idea of how to build a DAW with a plugin architecture using the Web Audio API:
> >
> >  * You have tracks, which may contain audio and sequencing data (e.g. MIDI, OSC and/or user-defined envelopes). All of these inputs can be either being recorded from an external source, or be static pieces.
> >
> >  * You have an effects list for each track, effects being available to pick from plugins.
> >
> >  * You have plugins. The plugins are given references to two gain nodes, one for input and one for output, as well as a reference to the AudioContext. In response, they will give AudioParam references back to the host, as well as some information of what the AudioParams stand for, min/max values and so on. The plugin will set up a sub-graph between the given gain nodes.
> >
> > This would be a very basic setup, but with the current API design there are some hard problems to solve here. The audio is relatively easy, regardless of whether it's coming from an external source or not. It's just a source node of some sort. The sequencing part is where stuff gets tricky.
> >
> > In the plugin models I've used, the sequencing data is paired with the audio data in processing events, i.e. you're told to fill some buffers, given a few k-rate params, a few a-rate params and some sequencing events as well as the input audio data. This makes it very simple to synchronize the sequencing events with the audio. But with the Web Audio API, the only place where you get a processing event like this is the JS node, and even there you currently only get the input audio.
> >
> > What would be the proposed solution for handling this case? And please, no setTimeout(). A system is as weak as its weakest link and building a DAW/Sequencer that relies on setTimeout is going to be utterly unreliable, which a DAW can't afford to be.
> >
> > Cheers,
> > Jussi
> 
> 
> 
>