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f766b85b96
Add a source-info property that will read/write meta to the buffers about RTP source information. The GstRTPSourceMeta can be used to transport information about the origin of a buffer, e.g. the sources that is included in a mixed audio buffer. A new function gst_rtp_base_payload_allocate_output_buffer() is added for payloaders to use to allocate the output RTP buffer with the correct number of CSRCs according to the meta and fill it. RTPSourceMeta does not make sense on RTP buffers since the information is in the RTP header. So the payloader will strip the meta from the output buffer. https://bugzilla.gnome.org/show_bug.cgi?id=761947
36 lines
1.3 KiB
C
36 lines
1.3 KiB
C
/* GStreamer
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* Copyright (C) 2012 GStreamer developers
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*
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* gstrtp.h: single include header for gst-rtp library
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifndef __GST_RTP_H__
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#define __GST_RTP_H__
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#include <gst/rtp/gstrtpdefs.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include <gst/rtp/gstrtcpbuffer.h>
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#include <gst/rtp/gstrtppayloads.h>
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#include <gst/rtp/gstrtphdrext.h>
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#include <gst/rtp/gstrtpbaseaudiopayload.h>
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#include <gst/rtp/gstrtpbasepayload.h>
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#include <gst/rtp/gstrtpbasedepayload.h>
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#include <gst/rtp/gstrtpmeta.h>
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#include <gst/rtp/gstrtp-enumtypes.h>
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#endif /* __GST_RTP_H__ */
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