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rtphdrext-rfc6464: Add test for inserting in payloader using the API

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This makes it clearer how to use the plugin in an API driven application.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1058>
This commit is contained in:
Olivier Crête 2021-08-19 10:32:27 -04:00 committed by GStreamer Marge Bot
parent f70ccd6d86
commit aa3d2c3369
1 changed files with 47 additions and 0 deletions
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47
tests/check/elements/rtphdrextrfc6464.c
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@ -303,6 +303,52 @@ GST_START_TEST (rtprfc6464_payloader_depayloader)
GST_END_TEST;
GST_START_TEST (rtprfc6464_payloader_api)
{
GstHarness *h;
GstRTPHeaderExtension *ext;
GstBuffer *b;
GstFlowReturn fret;
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
guint8 *data;
guint size;
guint8 level;
gboolean voice_activity;
h = gst_harness_new ("rtpL16pay");
gst_harness_set_src_caps_str (h, "audio/x-raw, rate=44100, channels=1,"
" layout=interleaved, format=S16BE");
ext = gst_rtp_header_extension_create_from_uri (URN);
gst_rtp_header_extension_set_id (ext, 2);
fail_unless (ext);
g_signal_emit_by_name (h->element, "add-extension", ext);
b = gst_buffer_new_allocate (NULL, 100, NULL);
gst_buffer_add_audio_level_meta (b, 12, TRUE);
fret = gst_harness_push (h, b);
fail_unless (fret == GST_FLOW_OK);
b = gst_harness_pull (h);
fail_unless (gst_rtp_buffer_map (b, GST_MAP_READ, &rtp));
fail_unless (gst_rtp_buffer_get_extension_onebyte_header (&rtp, 2, 0,
(gpointer *) & data, &size));
fail_unless (size == 1);
level = data[0] & 0x7F;
voice_activity = (data[0] & 0x80) >> 7;
fail_unless (level == 12);
fail_unless (voice_activity == TRUE);
gst_rtp_buffer_unmap (&rtp);
gst_buffer_unref (b);
gst_object_unref (ext);
gst_harness_teardown (h);
}
GST_END_TEST;
static Suite *
rtprfc6464_suite (void)
{
@ -316,6 +362,7 @@ rtprfc6464_suite (void)
tcase_add_test (tc_chain, rtprfc6464_two_bytes);
tcase_add_test (tc_chain, rtprfc6464_no_meta);
tcase_add_test (tc_chain, rtprfc6464_payloader_depayloader);
tcase_add_test (tc_chain, rtprfc6464_payloader_api);
return s;
}