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Re: Event dispatch relative to AudioContext time

From: Joseph Berkovitz <joe@noteflight.com>
Date: Mon, 2 Feb 2015 11:57:53 -0500
Cc: public-audio@w3.org
Message-Id: <C5B62F29-5456-4FAF-9DC1-9F5D9B4F26AD@noteflight.com>
To: Steven Yi <stevenyi@gmail.com>
Thanks Steven.

Could you please post your comment about jitter and the Brandt/Dannenberg reference to the issue thread? Those are valuable points. 


On Feb 2, 2015, at 11:49 AM, Steven Yi <stevenyi@gmail.com> wrote:

> Hi Joe,
> Thank you for filing that issue, I have subscribed to it and will try
> to add some notes to it.  Regarding #5, I should have noted that I
> think the "Two Clocks" design with ahead of time scheduling of a
> partial set of events is completely valid for a number of realtime
> uses cases. (I believe it's the same as Brandt and Dannenberg's
> "Forward Synchronous Model", as discussed in [1]). However, I think
> the design also best works when some expectations can be made about
> how bounded the jitter can be, which with the JS Main thread seems
> very difficult.
> Thanks!
> steven
> [1] - "Time in Distributed Real-Time Systems", Eli Brandt and Roger B.
> Dannenberg. 1999. Available at:
> http://www.cs.cmu.edu/~rbd/papers/synchronous99/synchronous99.pdf
> On Mon, Feb 2, 2015 at 11:09 AM, Joseph Berkovitz <joe@noteflight.com> wrote:
>> Hi Steven,
>> Many points here worth responding to, but I will just reference your #5
>> since it was also raised by some other people at the WAC and I think it is
>> an important issue for both realtime and offline audio rendering.
>> Please see the newly filed
>> https://github.com/WebAudio/web-audio-api/issues/473
>> …Joe
>> On Jan 30, 2015, at 7:57 PM, Steven Yi <stevenyi@gmail.com> wrote:
>> Hello All,
>> First, it was a great pleasure to be at the Web Audio conference. I
>> enjoyed the sessions and gigs and getting to the meet the other
>> members of community that I did.  Cheers to IRCAM and Mozilla for the
>> lovely conference!
>> That said, I have some comments and questions about the Web Audio API
>> and specification. (Note: these comments are in reference to the 06
>> January 2015 draft, found at
>> http://webaudio.github.io/web-audio-api/.)
>> #1 - The specification is not clear to me when a node become live. I
>> assume it is when a node is connected to the active part of the audio
>> graph that is "live" and processing. Since node creation and graph
>> assembly is done in the JS Main thread, it seems that the following
>> from "3.3 Example: Mixer with Send Busses", it possible that nodes
>> might get attached across buffers in the audio thread:
>>  compressor = context.createDynamicsCompressor();
>>   // Send1 effect
>>   reverb = context.createConvolver();
>>   // Convolver impulse response may be set here or later
>>   // Send2 effect
>>   delay = context.createDelay();
>>   // Connect final compressor to final destination
>>   compressor.connect(context.destination);
>>   // Connect sends 1 & 2 through effects to main mixer
>>   s1 = context.createGain();
>>   reverb.connect(s1);
>>   s1.connect(compressor);
>>   s2 = context.createGain();
>>   delay.connect(s2);
>>   s2.connect(compressor);
>> For example, could it be the case that "s1.connect(compresor)" above
>> happens just before buffer n starts to generate, and
>> "s2.connect(compressor)" happens such that it starts in when buffer n
>> + 1 is generating?
>> If this is the case, would connecting the compressor to the
>> context.destination at the end of the example, rather than the
>> beginning, guarantee that the graph of nodes connected to the
>> compressor are started at the same time?  If so, then maybe this
>> aspect of node graph creation could be clarified and the example in
>> 3.3 updated so that the sub-graph of nodes is clearly formed before
>> attaching to the active audio-graph.
>> #2 - Following from #1, what would happen if one is dynamically
>> altering a graph to remove an intermediary node?  For example, lets
>> say one has a graph like:
>>  gain = contxt.createGainNode();
>>  compressor = context.createDynamicsCompressor();
>>  reverb = context.createConvolver();
>>  gain.connect(reverb);
>>  reverb.connect(compressor);
>>  compressor.connect(context.destination);
>> and later the user decides to remove the reverb with something like:
>>  reverb.disconnect();
>>  // gain.disconnect();
>>  gain.connect(compressor);
>> (Assuming the above uses a gain node as a stable node for other nodes
>> to attach to.) My question is: when does connect and disconnect
>> happen?  Does it happen at block boundaries?  I assume it must or a
>> graph can get in a bad state if the graph changes while a block is
>> being processed.
>> Also, without the gain.disconnect(), will there be a hidden reference
>> to the reverb from gain? (I guess a "connection" reference according
>> to 2.3.3). If so, this seems like it could be a source of a memory
>> leak (assuming that the above object references to reverb are all
>> cleared from the JS main thread side).
>> #3 -  In "2.3.2 Methods", for an AudioNode to connect to another audio
>> node, it is not clear whether fan-out/fan-in is supported.  The
>> documentation for connecting to AudioParams explicitly states that
>> this is supported.  Should the first connect() method documentation be
>> clarified for this when connecting to nodes?
>> #4 - Also in regards to 2.3.2, the API of disconnect() seems odd as it
>> does not mirror connect(). connect() is given an argument of what node
>> or audioParam to connect to.  disconnect() however does not have a
>> target argument. It's not clear then what this disconnects from. For
>> example, if I connect a node to two different nodes and also to
>> another node's parameter, then call disconnect, what happens?  As it
>> is now, it doesn't seem possible then to create a GUI editor where one
>> could connect the output of a node to multiple other nodes/params,
>> then click and disconnect a single connection.
>> #5 - In the music systems I've seen, event processing is done within
>> the audio-thread.  This generally happens for each buffer, something
>> like:
>> 1. Process incoming messages
>> 2. Process a priority queue of pending events
>> 3. Handle audio input
>> 4. Run processing graph for one block
>> 5. Handle audio output
>> I'm familiar with this from Csound and SuperCollider's engines, as
>> well as the design in my own software synthesizer Pink. (Chuck's
>> design follow the same basic pattern above, but on a sample-by-sample
>> basis.)
>> As it is today, the Web Audio API does not have any kind of reified
>> event object.  One can schedule some things like automations via
>> param's setXXXatTime() methods and have that run within the time of
>> the audio engine, but there is nothing built-in for events in the Web
>> Audio API.
>> Now, I have no issues with the Web Audio API not having a concrete
>> event system, and think it should not have one, as people have
>> different notions and needs out of what is encoded in an event.
>> However, I think that there should be a way to create one's own event
>> system, one that is clocked to the same audio system clock (i.e. run
>> within the audio thread).
>> I was a bit concerned when at the conference there was mention of "A
>> Tale of Two Clocks".  The design of trying to reference two clocks can
>> not, by definition, allow for a queue of events to be processed
>> synchronously with audio. If one formalizes events processing
>> functions and audio processing functions as functions of time, by
>> having two clocks you get two different variables, ta and tb, which
>> are not equivalent unless the clocks are proven to advance at the same
>> exact rate (i.e. ta0 == tb0, ta1 == tb1, ... tan == tbn).  However,
>> the JS Main thread and audio thread are not run at the same rate, so
>> we can at best implement some kind of approximation, but it can not be
>> a formally correct solution.
>> Event processing in a thread other than the audio thread has problems.
>> One mentioned at the conference was what to do with offline rendering,
>> where the clock of an audio engine runs faster than realtime, and may
>> advance faster or slower in terms of wall-clock time while rendering,
>> depending on how heavy the processing needs of the graph is.  Second,
>> I seemed to remember hearing a problem during one of the concerts when
>> I turned off my phone's screen and I continued to hear audio but all
>> events stopped, then a number of events fired all at once when I
>> turned my screen back on. The piece used an event scheduling system
>> that ran in the JS Main thread. I assume this situation is similar to
>> what could happen with backgrounded tabs, but I'm not quite sure about
>> all this. Either way, I think there are real problems here that need
>> to be addressed.
>> This also leads to a bigger question: with Web Audio, if I run the
>> same project twice that uses an event system to reify graph
>> modifications in time (as events in audio engines are mostly used for,
>> i.e. alloc this graph of nodes and add to the live audio graph), will
>> I get the same result?  Assuming to use only referentially transparent
>> nodes (i.e. no random calculations), I believe the only way to
>> guarantee this is if the event system is processed as part of the
>> audio thread.
>> Now, what can a user do with Web Audio to create their own Event
>> system that is in sync with the audio thread?  Currently, there is the
>> ScriptProcessorNode.  Of course, the design of ScriptProcessorNode is
>> deeply flawed for all the reasons discussed at the conference
>> (Security, Inefficient due to context switching, potential for
>> breakups, etc.).  However, what it does do is allow for one to process
>> events in sync with the audio thread, allowing to build formally
>> correct audio systems where one processes event time according to the
>> same time as is used by the audio nodes. Additionally, according to
>> those events, one can dynamically modify the graph (i.e. add new
>> instances of a sub-graph of nodes to the live graph, representing a
>> "note"), via reference to other nodes and the audio context. So while
>> flawed in terms of performance and security, it does allow one to
>> build correct systems that generate consistent output.
>> My concern is that there was discussion of not only deprecating
>> ScriptProcessorNode, but removing it altogether.  I would have no
>> problems with this, except that from reading the current specification
>> for AudioWorker, I do not see how it would be possible to create an
>> event system with it.  While one can pass messages to and from an
>> AudioWorker, one has no access to the AudioContext. In that regards,
>> one can not say, within an AudioWorker, create new nodes and attach to
>> the context.destination. I am not very familiar with transferables and
>> what can be passed between the AudioWork and the JS Main thread via
>> postMessage, but I assume AudioNodes can not be made transferable.
>> At this point, I'm questioning what can be done. It seems
>> AudioWorker's design is not meant for event processing (fair enough),
>> and ScriptProcessor can only do this by accident and not design. Is
>> there any solution to this problem with the Web Audio API moving
>> forward?  For example, would this group be willing to consider
>> extending the API for non-audio nodes?  (Processing nodes?). If
>> processing nodes could be added that has a larger context than what is
>> proposed for AudioWorkGlobalContext--say, has access to the
>> AudioContext, and can modify the audio node graph dynamically--I could
>> see it as a solution to allow building higher level constructs like an
>> event system.
>> #6 - For the AudioWorker specification, I think it would be useful to
>> have clarification on when postMessage is processed.  In, it
>> has a link to "the algorithm defined by the Worker Specification".
>> That in turn mentions:
>> "The postMessage() method on DedicatedWorkerGlobalScope objects must
>> act as if, when invoked, it immediately invoked the method of the same
>> name on the port, with the same arguments, and returned the same
>> return value."
>> If it meant to be processed immediately, then this can cause problems
>> if the AudioWorker is already part of a live graph and values mutate
>> while an audio worker is processing a block. I think it would be good
>> to have clarification on this, perhaps with a recommendation that in
>> onaudioprocess functions, one should make a local copy of a value of a
>> mutable value and use that for the duration of onaudioprocess to get a
>> consistent result for the block.
>> #7 - Related to #6, I noticed in " A Bitcrusher Node", the
>> example uses a phaser variable that is scoped to the AudioWorker.  I
>> assume this would then be on the heap. This is perhaps more of general
>> JS question, but I normally see in block-based audio programming that
>> for a process() function, one generally copies any state variables of
>> a node/ugen/etc. to local variables, runs the audio for-loop with
>> local variable, then saves the state for the next run.  This is done
>> for performance (better locality, stack vs. heap access, better
>> compiler optimizations, etc.). I don't know much about JavaScript
>> implementations; can anyone comment if these kinds of optimizations
>> are effective in JS?  If so, the example might benefit from rewriting
>> and give some guidance. (i.e. phase and lastDataValue are copied to a
>> local var before the for-loop, and saved again after the for-loop, in
>> onaudioprocess).
>> Thanks!
>> steven
>> .            .       .    .  . ...Joe
>> Joe Berkovitz
>> President
>> Noteflight LLC
>> Boston, Mass.
>> phone: +1 978 314 6271
>> www.noteflight.com
>> "Your music, everywhere"

.            .       .    .  . ...Joe

Joe Berkovitz

Noteflight LLC
Boston, Mass.
phone: +1 978 314 6271
"Your music, everywhere"
Received on Monday, 2 February 2015 16:58:30 UTC

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