Thanks for the replies.
So I guess that's as solid a confirmed no as I am going to get.
On 12/21/2011 7:00 PM, Gerry Beauregard wrote:
Hi Anthony,
On every platform I've tried, when using the microphone input each
SampleDataEvent gives 2048 samples.
Up until now, I really didn't care about things like latency, as I was merely
attempting to learn algorithms for post processing related tasks; yet, now that
I have become interested in realtime tasks, and would love to be able to
perform them in the browser, it's becoming an issue.
I hear ya! The high latency for Flash audio both on input and output makes it
basically unusable for any application where near real-time response is
required. The lack of support for stereo audio input and MIDI, the
single-threaded execution model, and the generally poor performance of AS3
(compared to C#, for example) also don't help matters! That said, there are
still plenty of interesting audio applications that are possible even within
the above limitations.
Incidentally, the latency for the Flash audio output (and perhaps the audio
input) varies a lot depending on the operating system. In my experience,
Android is the worst, Windows is better, and Mac is quite good. But the best
I've seen is on the BlackBerry Playbook.
Got a Playbook for free at the Singapore Blackberry DevCon, which I went to
mainly because I could get a free copy of FlashBuilder Premium (which normally
costs more than the conference fees). I figured I'd immediately put the free
Playbook up for sale on eBay, but decided to keep it after being pleasantly
surprised at how well it worked, especially for Flash.
Cheers,
-Gerry
On 2011-12-22 , at 07:15 , Anthony Pace wrote:
Hey Karl,
Not sure if you read my question completely, or if you did and what I wrote was
too confusing; yet, I do already know how to set the sampling rate (e.g.
mic.rate = 44, or 11, or w/e ).
What I was asking, is if there is anyway to access the sample buffer before the
bytes get reported by the SampleDataEvent; for, at least on my machine, it only
reports a minimum of 2048 samples about every 47 ms, but I really want to have
access to 512 samples in about 12 ms, or even 256 in sample in about 6 ms.
When you adjust the sample rate, you can get lower sample sets returned; yet,
as the rate has been decreased you receive them with the same amount of
latency. (e.g at a rate of 11 you get 512 samples when the SampleDataEvent
fires, but you still get them in the3 same amount of time, which is about 47 ms
on average).
Now do you see what see what I am referring to?
On 12/21/2011 5:29 PM, Karl DeSaulniers wrote:
Did a google for "controlling mic sample rates in as3"
FYI
Best,
Karl
On Dec 21, 2011, at 2:08 PM, Anthony Pace wrote:
Wow. Just read through my post, and saw the errors. Guess I shouldn't get 3
hours sleep and then try to type.
Second try:
No replies? Okay then. As the player doesn't seem to make the buffered data
available unless it has a full 2^n chunk within a given timeframe, and also
considering that there doesn't seem to be a way to just grab the currently
buffered bytes, then I guess doing what I want just isn't possible, at least in
this version of the player (hope for the future).
This is really sad; for, imho, the developer should be able to at least state
the minimum sample size he wants reported per SampleDataEvent, and not just the
rate per second.
In combination with setting the desired sampling rate (e.g mic.rate = 44),
being able to set the minimum buffer amount to be reported, and to set no
delays would be perfect .
e.g.
mic.sampleBufferMin = 8; // 2^8 = 256 must be value between 8 and etc...
mic.sampleBufferMin = 9;// 2^9 = 512 is what would work in my case
mic.reportImmediately = true; //to avoid having to wait until the next frame is
collected if it couldn't collect enough to report in time
or even being able to access the bytes already in the buffer without having to
wait until SampleDataEvent is allowed to fire
e.g.
myByteArray = mic.pollBuffer(); //or bytesLoaded or w/e you would prefer to
call it
Up until now, I really didn't care about things like latency, as I was merely
attempting to learn algorithms for post processing related tasks; yet, now that
I have become interested in realtime tasks, and would love to be able to
perform them in the browser, it's becoming an issue.
On 12/20/2011 4:32 PM, Anthony Pace wrote:
Just in case you are wondering why...
although I know I could just split up the sample amount into smaller 512 sample
chunks, I am interested in doing some lower latency effects, and the delay is
just too much to bare right now.
On 12/20/2011 3:37 PM, Anthony Pace wrote:
Is this possible?
When recording from the Mic, in this case at 44100hz I can't get the
SampleDataEvent to report sample chunks of less than 2048 samples. The
problem really is that it flash waits about 47 ms on average to report a sample
chunk (although my counter shows it updates between 33ms and 60ms), and in some
cases even though there is no load, it waits until it can report 4098 samples
(even though there is no load); yet, I only need/want 512 samples in
approximately 12ms on average.
If, while keeping the same sample rate, there is some official way to decrease
the chunk size being reported and the time it takes to report it, I can't see
it; therefore, I am hoping there may be some way of just getting to the Sample
Data before it's reported?
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Karl DeSaulniers
Design Drumm
http://designdrumm.com
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