Re: [SailfishDevel] gstreamer rtspsrc issue
On Mon, Sep 14, 2015 at 11:40:53AM +0300, Kaj-Michael Lang wrote: > On Sun, 2015-09-13 at 23:49 +0300, Mohammed Hassan wrote: > > That won't work then. I thought -based on our previous discussions- > > that you are > > using GStreamer directly. > > I'm not as it is not allowed in harbour apps, and I like to have it > available there. Strictly Qt Multimedia only. But I've been hoping for > direct usage as then I could add things like equalizers, visualization, > etc. Direct usage will come. I just need to prepare the needed patches for the validator. You could help by listing the GStreamer libraries you need. > > BTW what's your app? Is it in harbour? > > It is, Y-Radio. 100% unusable on 1.1.9.28 right now. I will do my best then to push it to 1.1.9 but I cannot promise anything yet as it's really late. Cheers, ___ SailfishOS.org Devel mailing list To unsubscribe, please send a mail to devel-unsubscr...@lists.sailfishos.org
Re: [SailfishDevel] gstreamer rtspsrc issue
On pe, 2015-09-11 at 17:24 +0300, Mohammed Hassan wrote: > If you want to be updated on the issue then please file a t.j.c > post ;-) here https://together.jolla.com/question/109672/bug-gstreamer-1x-rtspsrc-does-not-work-properly/ -- Kaj-Michael Lang___ SailfishOS.org Devel mailing list To unsubscribe, please send a mail to devel-unsubscr...@lists.sailfishos.org
Re: [SailfishDevel] gstreamer rtspsrc issue
On Mon, Sep 14, 2015 at 08:27:26PM +0300, Kaj-Michael Lang wrote: > On pe, 2015-09-11 at 17:24 +0300, Mohammed Hassan wrote: > > If you want to be updated on the issue then please file a t.j.c > > post ;-) > > here > https://together.jolla.com/question/109672/bug-gstreamer-1x-rtspsrc-does-not-work-properly/ Thank you. Cheers, ___ SailfishOS.org Devel mailing list To unsubscribe, please send a mail to devel-unsubscr...@lists.sailfishos.org
Re: [SailfishDevel] gstreamer rtspsrc issue
On Sun, 2015-09-13 at 23:49 +0300, Mohammed Hassan wrote: > That won't work then. I thought -based on our previous discussions- > that you are > using GStreamer directly. I'm not as it is not allowed in harbour apps, and I like to have it available there. Strictly Qt Multimedia only. But I've been hoping for direct usage as then I could add things like equalizers, visualization, etc. > BTW what's your app? Is it in harbour? It is, Y-Radio. 100% unusable on 1.1.9.28 right now. -- Kaj-Michael Lang___ SailfishOS.org Devel mailing list To unsubscribe, please send a mail to devel-unsubscr...@lists.sailfishos.org
Re: [SailfishDevel] gstreamer rtspsrc issue
On su, 2015-09-13 at 20:39 +0300, Mohammed Hassan wrote: > I cannot promise. GStreamer 0.10 is still there in 1.1.9 so you can > continue using it until the glib bug gets fixed. How would that work now that Qt Multimedia is built against 1.x ? -- Kaj-Michael Lang___ SailfishOS.org Devel mailing list To unsubscribe, please send a mail to devel-unsubscr...@lists.sailfishos.org
Re: [SailfishDevel] gstreamer rtspsrc issue
On Sat, Sep 12, 2015 at 05:22:15PM +0300, Kaj-Michael Lang wrote: > On pe, 2015-09-11 at 17:24 +0300, Mohammed Hassan wrote: > > I did some more digging and it is a bug in glib. > > _g_uri_parse_authority() fails to parse the URI but this seems to has > > been fixed already. We are using an old glib and thus we are missing > > those fixes. I will see what can be done regarding that. > > Thanks for digging deeper on this. I hope a fix will be included in the > public final release. Unfortunately it's too late to include the fix in the final 1.1.9 release. I am trying to -however- include the fix in the update after 1.1.9 but I cannot promise. GStreamer 0.10 is still there in 1.1.9 so you can continue using it until the glib bug gets fixed. > > If you want to be updated on the issue then please file a t.j.c > > post ;-) > > Sure, I can do that. Thanks. Please share the link when you do so. Cheers, ___ SailfishOS.org Devel mailing list To unsubscribe, please send a mail to devel-unsubscr...@lists.sailfishos.org
Re: [SailfishDevel] gstreamer rtspsrc issue
On Sun, Sep 13, 2015 at 10:51:17PM +0300, Kaj-Michael Lang wrote: > On su, 2015-09-13 at 20:39 +0300, Mohammed Hassan wrote: > > I cannot promise. GStreamer 0.10 is still there in 1.1.9 so you can > > continue using it until the glib bug gets fixed. > > How would that work now that Qt Multimedia is built against 1.x ? That won't work then. I thought -based on our previous discussions- that you are using GStreamer directly. BTW what's your app? Is it in harbour? Cheers, ___ SailfishOS.org Devel mailing list To unsubscribe, please send a mail to devel-unsubscr...@lists.sailfishos.org
Re: [SailfishDevel] gstreamer rtspsrc issue
On pe, 2015-09-11 at 17:24 +0300, Mohammed Hassan wrote: > I did some more digging and it is a bug in glib. > _g_uri_parse_authority() fails to parse the URI but this seems to has > been fixed already. We are using an old glib and thus we are missing > those fixes. I will see what can be done regarding that. Thanks for digging deeper on this. I hope a fix will be included in the public final release. > If you want to be updated on the issue then please file a t.j.c > post ;-) Sure, I can do that. -- Kaj-Michael Lang___ SailfishOS.org Devel mailing list To unsubscribe, please send a mail to devel-unsubscr...@lists.sailfishos.org
Re: [SailfishDevel] gstreamer rtspsrc issue
On Fri, 11 Sep 2015 14:23:07 +0300 Kaj-Michael Langwrote: > Hi > > Upgraded my phone to 1.1.9.28 (upgrade was smoonth) and tried my > softwares, unfortunately something is broken with rtspsrc in > gstreamer. > > On my desktop with gst 1.4.5 it works ok, but on the device I get a > connection error, and it is not my network, users have reported the > same thing. > > Commandline used: > gst-launch-1.0 -t -c -v playbin > uri=rtsp://rtspstreamer.yle.fi/radio/yleliveradiohd_6_64@113883 > > On desktop, connects and plays ok: > Setting pipeline to PAUSED ... > /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: > ring-buffer-max-size = 0 > /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: buffer-size = -1 > /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: buffer-duration = > -1 /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: use-buffering = > false > /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: download = false > /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: uri = > rtsp://rtspstreamer.yle.fi/radio/yleliveradiohd_6_64@113883 > /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: connection-speed > = 0 /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: source = > "\(GstRTSPSrc\)\ source" > Pipeline is live and does not need PREROLL ... > Progress: (open) Opening Stream > Progress: (connect) Connecting to > rtsp://rtspstreamer.yle.fi/radio/yleliveradiohd_6_64@113883 > Progress: (open) Retrieving server options > Progress: (open) Retrieving media info > Progress: (request) SETUP stream 0 > . > etc and the stream plays ok > > On device, I get a connection error: > Setting pipeline to PAUSED ... > /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: > ring-buffer-max-size = 0 > /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: buffer-size = -1 > /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: buffer-duration = > -1 /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: use-buffering = > false > /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: download = false > /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: uri = > rtsp://rtspstreamer.yle.fi/radio/yleliveradiohd_6_64@113883 > /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: connection-speed > = 0 /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: source = > "\(GstRTSPSrc\)\ source" > Pipeline is live and does not need PREROLL ... > Progress: (open) Opening Stream > Progress: (connect) Connecting to > rtsp://rtspstreamer.yle.fi/radio/yleliveradiohd_6_64@113883 > ERROR: from > element /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0/GstRTSPSrc:source: > Could not open resource for reading and writing. Additional debug > info: gstrtspsrc.c(6868): gst_rtspsrc_retrieve_sdp > (): /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0/GstRTSPSrc:source: > Failed to connect. (Generic error) > ERROR: pipeline doesn't want to preroll. > Setting pipeline to PAUSED ... > Setting pipeline to READY ... > Setting pipeline to NULL ... > Freeing pipeline ... > > any ideas ? I am surprised it complains about an invalid URI: Progress: (connect) Connecting to rtsp://rtspstreamer.yle.fi/radio/yleliveradiohd_6_64@113883 0:00:00.074042423 14227 0x1ebf460 ERRORdefault gstrtspconnection.c:877:gst_rtsp_connection_connect: failed to connect: Invalid URI 'rtsp://rtspstreamer.yle.fi/radio/yleliveradiohd_6_64@113883' 0:00:00.074591790 14227 0x1ebf460 ERRORrtspsrc gstrtspsrc.c:4198:gst_rtsp_conninfo_connect: Could not connect to server. (Generic error) 0:00:00.074988555 14227 0x1ebf460 WARN rtspsrc gstrtspsrc.c:6868:gst_rtspsrc_retrieve_sdp: error: Failed to connect. (Generic error) You would be better submitting a t.j.c issue so I can at least try to update you as I make progress debugging it. Cheers, ___ SailfishOS.org Devel mailing list To unsubscribe, please send a mail to devel-unsubscr...@lists.sailfishos.org
Re: [SailfishDevel] gstreamer rtspsrc issue
On Fri, 11 Sep 2015 16:10:09 +0300 Mohammed Hassanwrote: > On Fri, 11 Sep 2015 14:23:07 +0300 > Kaj-Michael Lang wrote: > > > Hi > > > > Upgraded my phone to 1.1.9.28 (upgrade was smoonth) and tried my > > softwares, unfortunately something is broken with rtspsrc in > > gstreamer. > > > > On my desktop with gst 1.4.5 it works ok, but on the device I get a > > connection error, and it is not my network, users have reported the > > same thing. > > > > Commandline used: > > gst-launch-1.0 -t -c -v playbin > > uri=rtsp://rtspstreamer.yle.fi/radio/yleliveradiohd_6_64@113883 > > > > On desktop, connects and plays ok: > > Setting pipeline to PAUSED ... > > /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: > > ring-buffer-max-size = 0 > > /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: buffer-size = -1 > > /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: buffer-duration > > = -1 /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: > > use-buffering = false > > /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: download = false > > /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: uri = > > rtsp://rtspstreamer.yle.fi/radio/yleliveradiohd_6_64@113883 > > /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: connection-speed > > = 0 /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: source = > > "\(GstRTSPSrc\)\ source" > > Pipeline is live and does not need PREROLL ... > > Progress: (open) Opening Stream > > Progress: (connect) Connecting to > > rtsp://rtspstreamer.yle.fi/radio/yleliveradiohd_6_64@113883 > > Progress: (open) Retrieving server options > > Progress: (open) Retrieving media info > > Progress: (request) SETUP stream 0 > > . > > etc and the stream plays ok > > > > On device, I get a connection error: > > Setting pipeline to PAUSED ... > > /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: > > ring-buffer-max-size = 0 > > /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: buffer-size = -1 > > /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: buffer-duration > > = -1 /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: > > use-buffering = false > > /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: download = false > > /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: uri = > > rtsp://rtspstreamer.yle.fi/radio/yleliveradiohd_6_64@113883 > > /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: connection-speed > > = 0 /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0: source = > > "\(GstRTSPSrc\)\ source" > > Pipeline is live and does not need PREROLL ... > > Progress: (open) Opening Stream > > Progress: (connect) Connecting to > > rtsp://rtspstreamer.yle.fi/radio/yleliveradiohd_6_64@113883 > > ERROR: from > > element > > /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0/GstRTSPSrc:source: > > Could not open resource for reading and writing. Additional debug > > info: gstrtspsrc.c(6868): gst_rtspsrc_retrieve_sdp > > (): /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0/GstRTSPSrc:source: > > Failed to connect. (Generic error) > > ERROR: pipeline doesn't want to preroll. > > Setting pipeline to PAUSED ... > > Setting pipeline to READY ... > > Setting pipeline to NULL ... > > Freeing pipeline ... > > > > any ideas ? > > I am surprised it complains about an invalid URI: > Progress: (connect) Connecting to > rtsp://rtspstreamer.yle.fi/radio/yleliveradiohd_6_64@113883 > 0:00:00.074042423 14227 0x1ebf460 ERRORdefault > gstrtspconnection.c:877:gst_rtsp_connection_connect: failed to > connect: Invalid URI > 'rtsp://rtspstreamer.yle.fi/radio/yleliveradiohd_6_64@113883' > 0:00:00.074591790 14227 0x1ebf460 ERRORrtspsrc > gstrtspsrc.c:4198:gst_rtsp_conninfo_connect: Could not connect > to server. (Generic error) 0:00:00.074988555 14227 0x1ebf460 > WARN rtspsrc > gstrtspsrc.c:6868:gst_rtspsrc_retrieve_sdp: error: Failed to > connect. (Generic error) > > You would be better submitting a t.j.c issue so I can at least try to > update you as I make progress debugging it. I did some more digging and it is a bug in glib. _g_uri_parse_authority() fails to parse the URI but this seems to has been fixed already. We are using an old glib and thus we are missing those fixes. I will see what can be done regarding that. If you want to be updated on the issue then please file a t.j.c post ;-) Cheers, ___ SailfishOS.org Devel mailing list To unsubscribe, please send a mail to devel-unsubscr...@lists.sailfishos.org