On 6/26/07, Justin Tulloss <[EMAIL PROTECTED]> wrote:
Hello, I'm just getting into PulseAudio and I'm looking forward to using it for some challenging projects. I do have a few questions however: 1. Is good floating point performance necessary? I saw that some of your algorithms do rely on floating point, but I didn't get far enough into it to see whether it was necessary for basic PCM routing.
No, it isn't neccesarily. I been working with pulseaudio in an embedded enviroment for a while. There are two current floating point weaknesses. These are software volume and resampling. However I had made a patch to use a newer fixed point speex resampler. It got to the list for a little while. I'll resubmit once the speex stuff is done and stable. 2. How do direct plugin sources/sinks work? For example, the gstreamer
plugin communicates directly with pulse. Therefore it doesn't show up as a source in the manager. Is there a way to get a handle to that source? For instance, if I had 2 sinks (A and B) and I wanted to say that all alsa sound (from the alsa plugin) should go to sink A where all gstreamer audio should go to sink B, how would I do that? That's a contrived example, but I am looking for that kind of flexibility.
This is very easy. There is a API to get current sources and sinks. Each source can tell what sink it wants to go to. (ie. gst-launch audiotestsrc | pulsesink device=secondaudiodevice) 3. Is anybody actively working on a D-Bus control for this? Seems like it
would be easy enough.
I believe there has already been work to integrate dbus with pulse's mainloop and there seems to be some code that does this. I haven't found it necessary to use because the native protocol works well. (although I use dbus for everything else.) 4. How do you define policy for HAL events? I don't see an obvious way of
saying "now that this sink is connected, all audio goes to it and the old sink will be turned off"
Sorry, This is the only question I don't know.... Keith
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