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webrtc: stats: Fill data-channel transport stats

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Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8698>
This commit is contained in:
Philippe Normand 2025-03-26 10:38:06 +00:00 committed by GStreamer Marge Bot
parent b088d53970
commit ac465ebfe1
2 changed files with 119 additions and 0 deletions
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24
subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcstats.c
View File

@ -1015,6 +1015,28 @@ out:
return TRUE;
}
static void
_get_data_channel_transport_stats (GstWebRTCBin * webrtc, GstStructure * s)
{
struct transport_stream_stats ts_stats = {
NULL,
};
GObject *gst_rtp_session;
if (!webrtc->priv->data_channel_transport)
return;
ts_stats.stream = webrtc->priv->data_channel_transport;
g_signal_emit_by_name (webrtc->rtpbin, "get-session",
ts_stats.stream->session_id, &gst_rtp_session);
ts_stats.transport_id =
_get_stats_from_dtls_transport (webrtc, ts_stats.stream->transport,
GST_WEBRTC_ICE_STREAM (ts_stats.stream->stream), NULL, s);
g_clear_pointer (&ts_stats.transport_id, g_free);
}
GstStructure *
gst_webrtc_bin_create_stats (GstWebRTCBin * webrtc, GstPad * pad)
{
@ -1039,6 +1061,8 @@ gst_webrtc_bin_create_stats (GstWebRTCBin * webrtc, GstPad * pad)
gst_structure_free (pc_stats);
}
_get_data_channel_transport_stats (webrtc, s);
if (pad)
_get_stats_from_pad (webrtc, pad, s);
else

95
subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c
View File

@ -2924,6 +2924,100 @@ GST_START_TEST (test_data_channel_pre_negotiated)
GST_END_TEST;
typedef struct _stat_type_find_data
{
GstWebRTCStatsType t;
GstStructure *res;
} stat_type_find_data;
static gboolean
find_typed_stat (const GstIdStr * id, const GValue * value, gpointer user_data)
{
stat_type_find_data *data = (stat_type_find_data *) user_data;
if (!GST_VALUE_HOLDS_STRUCTURE (value))
return TRUE;
const GstStructure *structure = gst_value_get_structure (value);
GstWebRTCStatsType statsType;
if (!gst_structure_get (structure, "type", GST_TYPE_WEBRTC_STATS_TYPE,
&statsType, NULL))
return TRUE;
if (statsType == data->t) {
data->res = gst_structure_copy (structure);
return FALSE;
}
return TRUE;
}
static GstStructure *
get_typed_stats (GstElement * webrtcbin, GstWebRTCStatsType t)
{
GstPromise *p;
GstPromiseResult res;
stat_type_find_data data;
data.t = t;
data.res = NULL;
p = gst_promise_new ();
g_signal_emit_by_name (webrtcbin, "get-stats", NULL, p);
res = gst_promise_wait (p);
fail_unless_equals_int (res, GST_PROMISE_RESULT_REPLIED);
gst_structure_foreach_id_str (gst_promise_get_reply (p), find_typed_stat,
&data);
gst_promise_unref (p);
return data.res;
}
GST_START_TEST (test_data_channel_ice_stats)
{
struct test_webrtc *t = test_webrtc_new ();
GObject *channel = NULL;
VAL_SDP_INIT (media_count, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL);
VAL_SDP_INIT (offer, on_sdp_has_datachannel, NULL, &media_count);
GstStructure *local_ice_cand_stats, *remote_ice_cand_stats;
t->on_negotiation_needed = NULL;
t->on_ice_candidate = NULL;
t->on_prepare_data_channel = have_prepare_data_channel;
t->on_data_channel = signal_data_channel;
fail_if (gst_element_set_state (t->webrtc1, GST_STATE_READY) ==
GST_STATE_CHANGE_FAILURE);
fail_if (gst_element_set_state (t->webrtc2, GST_STATE_READY) ==
GST_STATE_CHANGE_FAILURE);
g_signal_emit_by_name (t->webrtc1, "create-data-channel", "label", NULL,
&channel);
g_assert_nonnull (channel);
fail_if (gst_element_set_state (t->webrtc1, GST_STATE_PLAYING) ==
GST_STATE_CHANGE_FAILURE);
fail_if (gst_element_set_state (t->webrtc2, GST_STATE_PLAYING) ==
GST_STATE_CHANGE_FAILURE);
/* Wait SCTP transport creation */
test_validate_sdp_full (t, &offer, &offer, 1 << STATE_CUSTOM, FALSE);
local_ice_cand_stats =
get_typed_stats (t->webrtc1, GST_WEBRTC_STATS_LOCAL_CANDIDATE);
remote_ice_cand_stats =
get_typed_stats (t->webrtc1, GST_WEBRTC_STATS_REMOTE_CANDIDATE);
g_assert_nonnull (local_ice_cand_stats);
g_assert_nonnull (remote_ice_cand_stats);
gst_structure_free (local_ice_cand_stats);
gst_structure_free (remote_ice_cand_stats);
g_object_unref (channel);
test_webrtc_free (t);
}
GST_END_TEST;
static void
_count_non_rejected_media (struct test_webrtc *t, GstElement * element,
GstWebRTCSessionDescription * sd, gpointer user_data)
@ -6729,6 +6823,7 @@ webrtcbin_suite (void)
tcase_add_test (tc, test_sdp_no_media);
tcase_add_test (tc, test_session_stats);
tcase_add_test (tc, test_stats_with_stream);
tcase_add_test (tc, test_data_channel_ice_stats);
if (vp8enc) {
tcase_add_test (tc, test_stats_with_two_streams);
} else {