Hi, On Tue, Jul 29, 2008 at 5:43 PM, Dennis Fleming <arsantiqua at sbcglobal.net> wrote: > The interesting thing is that uncompressed WAV files are causing the problem > while MP3s were fixed by setting the buffer-time and latency-time to values > smaller than found on a desktop. What would adding a queue do to latency > through the system? There is no latency in this case because there are no live-sources. [1] > Also, I suppose, that I will need to break up the > playbin and create a pipeline myself, yes? playbin has the queue elements on the correct location, no changes needed. You where already using a custom pipeline, no? Gr, [1] http://webcvs.freedesktop.org/gstreamer/gstreamer/docs/design/draft-latency.txt?view=markup > > Dennis > > ----- Original Message ---- > From: Thijs Vermeir <thijsvermeir at gmail.com> > To: Zhao Liang-E3423C <E3423C at motorola.com> > Cc: Dennis Fleming <arsantiqua at sbcglobal.net>; > gstreamer-embedded at lists.sourceforge.net > Sent: Tuesday, July 29, 2008 2:46:42 AM > Subject: Re: noise and stuttering > > Hi, > > On Tue, Jul 29, 2008 at 11:15 AM, Zhao Liang-E3423C <E3423C at motorola.com> > wrote: >> What's the rootcause of noise and stuttering ? > > Now you are using only 1 thread for all the elements and if the > filesrc or the decoder is too slow sometimes > you don't have time to catch up. By adding the queue you put the sink > in another thread and now the filesrc+decoder can > do some decoding in advance. > > Gr, > Thijs > >> >> For normal playback, it should not have issues. If decoder didn't drop >> data, I think alsasink did it. >> By gstaudiosink mechanism, it will drop data replaced with blank data when >> data is late. I guess the rootcause is that. >> >> If that, I have no ideas except adding a queue before alsasink, and when >> queue is empty, pause the pipeline, it will not cause dropout, but still >> discontinous. >> >> Zhao liang >> ________________________________ >> From: gstreamer-embedded-bounces at lists.sourceforge.net >> [mailto:gstreamer-embedded-bounces at lists.sourceforge.net] On Behalf Of >> Dennis Fleming >> Sent: Tuesday, July 29, 2008 4:37 AM >> To: gstreamer-embedded at lists.sourceforge.net >> Subject: noise and stuttering >> >> I'm trying to create an audio player on an IMX31 target and I've found a >> discrepancy in the output of various formats. If I send MP3 data I have >> to >> set the buffer-time and latency-time to 10000 and 100 respectively to play >> without severe dropouts. However WAV files still have drop-out at a >> consistent rate (about 1 per 10 sec). Are there some general features I'm >> missing or is there some guidance on the buffer-time/latency time that >> would >> account for this difference? >> >> Linux 2.6.22.19 >> gstreamer 0.10.17 (open-embedded) >> gst-launch filesrc location=<file> ! decodebin ! alsasink >> buffer-time=10000 >> latency-time=100 >> >> Dennis >> >> ------------------------------------------------------------------------- >> This SF.Net email is sponsored by the Moblin Your Move Developer's >> challenge >> Build the coolest Linux based applications with Moblin SDK & win great >> prizes >> Grand prize is a trip for two to an Open Source event anywhere in the >> world >> http://moblin-contest.org/redirect.php?banner_id=100&url=/ >> _______________________________________________ >> Gstreamer-embedded mailing list >> Gstreamer-embedded at lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/gstreamer-embedded >> >> >
