Hi Pau! I'll excerpt a bit out of order here: On Wed, 2007-02-14 at 15:56 +0100, Pau Arumi wrote: GStreamer has two basic modes, push and pull. Push is optimized for the case when you are dealing with a container format, and don't know exactly what will come out of the format. The source starts a thread, pushing buffers, and the rest of the pipeline deals with buffers as they come. An audio sink in push mode spawns an additional thread to write to the audio device. It communicates with the rest of the pipeline via a ring buffer. Multiple sinks within the same pipeline are typically driven from separate threads. Synchronization is easy with multiple threads: a clock is selected somehow and the sinks perform blocking waits on the clock until it is time to render their data. In the audio sink case it is the thread feeding the ring buffer that performs the blocking wait. Typically, the audio sink also provides the pipeline clock based on how many samples have been played. Pull mode, more appropriate to this list ;), has to be explicitly supported by elements. It's a pain to implement a decoder such that you can ask for 64 frames of audio data, which is why they normally run in push mode. Pull mode in GStreamer is more appropriate to when dealing with a format with a constant number of bytes per unit time. In pull mode, the sink is the one that spawns the thread, driving the pipeline by pulling N bytes as it can. The process of thread spawning is generic enough so that the jack process() callback can run the gstreamer pipeline directly, pulling out only the number of samples that it wants. In the past there have been other scheduling mechanisms in GStreamer (cothreads, notably), but hopefully we have seen the last of those. gstreamer-devel(a)lists.sourceforge.net would be the list to continue a discussion, as it's probably more specific than this list is interested in. The best written documents we have, which are only slightly out of date, are these: http://webcvs.freedesktop.org/gstreamer/gstreamer/docs/design/part-overview… in the section "Dataflow and buffers", as an overview; http://webcvs.freedesktop.org/gstreamer/gstreamer/docs/design/part-activati… as how the threads get spawned or not, and how the scheduling method is chosen; and http://webcvs.freedesktop.org/gstreamer/gstreamer/docs/design/part-negotiat… for some trickiness regarding format ("caps") negotiation. On Wed, 2007-02-14 at 15:56 +0100, Pau Arumi wrote: Interesting! Most of the review that it needs is GStreamer-specific: how to ensure optimal behavior appropriate for audio synthesis in a very general framework. It's not an area that has been well-explored from the GStreamer side, although I've tried to make sure that the current stable series can support efficient pull-mode in a compatible manner. So getting use cases, ideas, analysis, and example apps would be useful indeed. I am indeed, although my Catalan is not yet up to standard ;-) In fact I think we're only a few blocks away. I'll follow with a private mail :) Cheers, Andy. -- http://wingolog.org/