[gst-devel] OpenAL sink
tristan at sat.qc.ca
Mon Feb 1 18:15:46 CET 2010
2010/2/1 Babineau, Denis <Denis.Babineau at gtech.com>
> All 4 values are under [in/out] so I don’t know if it’s inputting
> recommended values with the option of tweaking them?
Have you looked at code in jackaudiosink? It has reasonable defaults. Also,
if you need more fine grained control over scheduling and synchronization,
your plugin should be derived from GstBaseAudioSink rather than GstAudioSink
(see section 20.1 of the plugin writer's guide:
I'm not familiar enough with OpenAL to know if this is really necessary.
Good luck, this would be a welcome addition to gstreamer.
> Regarding latency_time/buffer_time, I’m leaving them as defaults (they can
> be modified thru the properties) and by default I get values of 10,000/0
> respectively with my test. Segsize/segtotal I take as “buffer size” and
> “number of buffers”, with OpenAL I have to pre-allocate buffers and while
> running I write to the buffer and queue them (and unqueue processed
> buffers). And so in my test I get 882 segsize and 20 segtotal so I
> pre-allocate 20 OpenAL buffers and fill/queue them during the write() call
> but what appears to be happening is that all the buffers gets full before
> even one buffer gets processed by OpenAL so I end up overrunning my buffers.
> Should I be stalling the write() call? Also, do those values make sense?
> When I normally stream a sound in OpenAL (not synced to video) I would
> allocate fairly large buffers, but not so many (2-3 buffers large enough to
> keep about 2-3 seconds buffered). But here from what I understand it’s
> suggesting I allocate buffers of only a few samples each (i.e. 882 / 4 = 220
> At this point I’m not too worried about latency/delays; I’m just trying to
> get something playing sound so that I get a better feel as to how this
> interface works.
> Any input appreciated!
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email: tristan at sat.qc.ca
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