GStreamer

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  GStreamer | gst-plugins-good | Ver: HEAD CVS

Wim Taymans changed:

           What    |Removed                     |Added
----------------------------------------------------------------------------
                 CC|                            |wim.taymans@...




------- Comment #1 from Wim Taymans  2008-10-28 10:02 UTC -------
First problem was basetransform not calling the fixate caps function in all
cases so that sometimes the channel positions were not set resulting in the
warnings above.

        * libs/gst/base/gstbasetransform.c:
        (gst_base_transform_find_transform), (gst_base_transform_getrange):
        If we have a fixate function, call it even if we already have fixed
caps
        because the subclass might add some caps. Makes audioconvert add a
        default channel layout.


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  GStreamer | gst-plugins-good | Ver: HEAD CVS




------- Comment #2 from Wim Taymans  2008-10-28 10:04 UTC -------
Second patch, add channel mappings and checks.

        * gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_setcaps):
        * gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_setcaps),
        (gst_rtp_L16_pay_getcaps):
        * gst/rtp/gstrtpchannels.c: (check_channels),
        (gst_rtp_channels_get_by_pos), (gst_rtp_channels_get_by_order),
        (gst_rtp_channels_create_default):
        * gst/rtp/gstrtpchannels.h:
        Add mappings for multichannel support. Does not completely just work
        because the getcaps function does not yet return the allowed channel
        mappings. See #556641.


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  GStreamer | gst-plugins-good | Ver: HEAD CVS




------- Comment #3 from tmatth  2008-10-28 18:59 UTC -------
Created an attachment (id=121532)
 --> (http://bugzilla.gnome.org/attachment.cgi?id=121532&action=view)
Test case that fails

This program creates 8 audiotestsrcs, interleaves them, then sends them through
rtpL16pay, rtpL16depay, audioconvert, jackaudiosink. Requires a jack server to
be running. 

Compile with:
gcc -Wall -Werror -g -pthread -I/usr/local/include/gstreamer-0.10
-I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -I/usr/include/libxml2
-L/usr/local/lib -lgstreamer-0.10 -lgstaudio-0.10 -lgobject-2.0 -lgmodule-2.0
-ldl -lgthread-2.0 -lrt -lxml2 -lglib-2.0 multiChannel.c -o multiChannel 



Currently fails with the message:

** (multiChannel:13409): WARNING **: audioconvert10: size 1388 is not a
multiple of unit size 16
0:00:00.320827207 13409 0x80dbaa0 ERROR basetransform
gstbasetransform.c:1304:gst_base_transform_prepare_output_buffer:<audioconvert10>
unknown output size


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  GStreamer | gst-plugins-good | Ver: HEAD CVS

Wim Taymans changed:

           What    |Removed                     |Added
----------------------------------------------------------------------------
             Status|UNCONFIRMED                 |RESOLVED
         Resolution|                            |FIXED
   Target Milestone|HEAD                        |0.10.12




------- Comment #4 from Wim Taymans  2008-10-29 18:28 UTC -------
        * gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_flush),
        (gst_rtp_L16_pay_getcaps):
        Only put an integral amount of samples in the RTP packet.
        Fixes #556641.


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  GStreamer | gst-plugins-good | Ver: git

Sebastian Dröge changed:

           What    |Removed                     |Added
----------------------------------------------------------------------------
                 CC|                            |slomo@...
   Target Milestone|0.10.12                     |0.10.14




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