mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-07-03 13:16:04 +00:00
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
111 lines
3.6 KiB
C
111 lines
3.6 KiB
C
/* GStreamer RTP meta unit tests
|
|
* Copyright (C) 2016 Stian Selnes <stian@pexip.com>
|
|
*
|
|
* This library is free software; you can redistribute it and/or
|
|
* modify it under the terms of the GNU Library General Public
|
|
* License as published by the Free Software Foundation; either
|
|
* version 2 of the License, or (at your option) any later version.
|
|
*
|
|
* This library is distributed in the hope that it will be useful,
|
|
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
* Library General Public License for more details.
|
|
*
|
|
* You should have received a copy of the GNU Library General
|
|
* Public
|
|
* License along with this library; if not, write to the
|
|
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
|
|
* Boston, MA 02110-1301, USA.
|
|
*/
|
|
|
|
#include <gst/gst.h>
|
|
#include <gst/check/gstcheck.h>
|
|
#include <gst/rtp/rtp.h>
|
|
|
|
GST_START_TEST (test_rtp_source_meta_set_get_sources)
|
|
{
|
|
GstBuffer *buffer;
|
|
GstRTPSourceMeta *meta;
|
|
guint32 ssrc = 1000, ssrc2 = 2000;
|
|
const guint32 csrc[] = {
|
|
0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14
|
|
};
|
|
|
|
buffer = gst_buffer_new ();
|
|
meta = gst_buffer_add_rtp_source_meta (buffer, &ssrc, csrc, 12);
|
|
|
|
fail_unless_equals_int (gst_rtp_source_meta_get_source_count (meta), 12 + 1);
|
|
fail_unless (meta->ssrc_valid);
|
|
fail_unless_equals_int (meta->ssrc, ssrc);
|
|
for (gint i = 0; i < 12; i++)
|
|
fail_unless_equals_int (meta->csrc[i], csrc[i]);
|
|
|
|
/* Unset the ssrc */
|
|
fail_unless (gst_rtp_source_meta_set_ssrc (meta, NULL));
|
|
fail_unless_equals_int (gst_rtp_source_meta_get_source_count (meta), 12);
|
|
fail_if (meta->ssrc_valid);
|
|
|
|
/* Set the ssrc again */
|
|
fail_unless (gst_rtp_source_meta_set_ssrc (meta, &ssrc2));
|
|
fail_unless_equals_int (gst_rtp_source_meta_get_source_count (meta), 12 + 1);
|
|
fail_unless (meta->ssrc_valid);
|
|
fail_unless_equals_int (meta->ssrc, ssrc2);
|
|
|
|
/* Append multiple csrcs */
|
|
fail_unless (gst_rtp_source_meta_append_csrc (meta, &csrc[12], 2));
|
|
fail_unless_equals_int (gst_rtp_source_meta_get_source_count (meta), 14 + 1);
|
|
for (gint i = 0; i < 14; i++)
|
|
fail_unless_equals_int (meta->csrc[i], csrc[i]);
|
|
|
|
gst_buffer_unref (buffer);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_rtp_source_meta_set_get_max_sources)
|
|
{
|
|
GstBuffer *buffer;
|
|
GstRTPSourceMeta *meta;
|
|
guint32 ssrc = 1000;
|
|
const guint32 csrc[16] = { 0, };
|
|
|
|
buffer = gst_buffer_new ();
|
|
meta = gst_buffer_add_rtp_source_meta (buffer, &ssrc, csrc, 14);
|
|
|
|
fail_unless_equals_int (gst_rtp_source_meta_get_source_count (meta), 14 + 1);
|
|
fail_unless_equals_int (meta->csrc_count, 14);
|
|
fail_unless (meta->ssrc_valid);
|
|
fail_unless_equals_int (meta->ssrc, ssrc);
|
|
|
|
/* Append one more csrc */
|
|
/* The source count should cap at 15 for convenient use with
|
|
* gst_rtp_buffer-functions! */
|
|
fail_unless (gst_rtp_source_meta_append_csrc (meta, &csrc[14], 1));
|
|
fail_unless_equals_int (gst_rtp_source_meta_get_source_count (meta), 15);
|
|
fail_unless_equals_int (meta->csrc_count, 15);
|
|
|
|
/* Try to append one more csrc, but we've reached max */
|
|
fail_if (gst_rtp_source_meta_append_csrc (meta, &csrc[15], 1));
|
|
fail_unless_equals_int (gst_rtp_source_meta_get_source_count (meta), 15);
|
|
fail_unless_equals_int (meta->csrc_count, 15);
|
|
|
|
gst_buffer_unref (buffer);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
static Suite *
|
|
rtp_meta_suite (void)
|
|
{
|
|
Suite *s = suite_create ("rtp_meta_tests");
|
|
TCase *tc_chain;
|
|
|
|
suite_add_tcase (s, (tc_chain = tcase_create ("GstRTPSourceMeta")));
|
|
tcase_add_test (tc_chain, test_rtp_source_meta_set_get_sources);
|
|
tcase_add_test (tc_chain, test_rtp_source_meta_set_get_max_sources);
|
|
|
|
return s;
|
|
}
|
|
|
|
GST_CHECK_MAIN (rtp_meta)
|