nice_agent_set_stream_tos (ice->priv->nice_agent, item->nice_stream_id, tos);
}
+static const gchar *
+_relay_type_to_string (GstUri * turn_server)
+{
+ const gchar *scheme;
+ const gchar *transport;
+
+ if (!turn_server)
+ return "none";
+
+ scheme = gst_uri_get_scheme (turn_server);
+ transport = gst_uri_get_query_value (turn_server, "transport");
+
+ if (g_strcmp0 (scheme, "turns") == 0) {
+ return "tls";
+ } else if (g_strcmp0 (scheme, "turn") == 0) {
+ if (!transport || g_strcmp0 (transport, "udp") == 0)
+ return "udp";
+ if (!transport || g_strcmp0 (transport, "tcp") == 0)
+ return "tcp";
+ }
+
+ return "none";
+}
+
+static gchar *
+_get_server_url (GstWebRTCICE * ice, NiceCandidate * cand)
+{
+ switch (cand->type) {
+ case NICE_CANDIDATE_TYPE_RELAYED:
+ return g_strdup (gst_uri_get_host (ice->turn_server));
+ case NICE_CANDIDATE_TYPE_SERVER_REFLEXIVE:
+ return g_strdup (gst_uri_get_host (ice->stun_server));
+ default:
+ return g_strdup ("");
+ }
+}
+
+/* TODO: replace it with nice_candidate_type_to_string()
+ * when it's ready for use
+ * https://libnice.freedesktop.org/libnice/NiceCandidate.html#nice-candidate-type-to-string
+ */
+static const gchar *
+_candidate_type_to_string (NiceCandidateType type)
+{
+ switch (type) {
+ case NICE_CANDIDATE_TYPE_HOST:
+ return "host";
+ case NICE_CANDIDATE_TYPE_SERVER_REFLEXIVE:
+ return "srflx";
+ case NICE_CANDIDATE_TYPE_PEER_REFLEXIVE:
+ return "prflx";
+ case NICE_CANDIDATE_TYPE_RELAYED:
+ return "relay";
+ default:
+ g_assert_not_reached ();
+ return NULL;
+ }
+}
+
+static void
+_populate_candidate_stats (GstWebRTCICE * ice, NiceCandidate * cand,
+ GstWebRTCICEStream * stream, GstWebRTCICECandidateStats * stats,
+ gboolean is_local)
+{
+ gchar ipaddr[INET6_ADDRSTRLEN];
+
+ g_assert (cand != NULL);
+
+ nice_address_to_string (&cand->addr, ipaddr);
+ stats->port = nice_address_get_port (&cand->addr);
+ stats->ipaddr = g_strdup (ipaddr);
+ stats->stream_id = stream->stream_id;
+ stats->type = _candidate_type_to_string (cand->type);
+ stats->prio = cand->priority;
+ stats->proto =
+ cand->transport == NICE_CANDIDATE_TRANSPORT_UDP ? "udp" : "tcp";
+ if (is_local) {
+ if (cand->type == NICE_CANDIDATE_TYPE_RELAYED)
+ stats->relay_proto = _relay_type_to_string (ice->turn_server);
+ stats->url = _get_server_url (ice, cand);
+ }
+}
+
+static void
+_populate_candidate_list_stats (GstWebRTCICE * ice, GSList * cands,
+ GstWebRTCICEStream * stream, GArray * result, gboolean is_local)
+{
+ GSList *item;
+
+ for (item = cands; item != NULL; item = item->next) {
+ GstWebRTCICECandidateStats stats;
+ NiceCandidate *c = item->data;
+ _populate_candidate_stats (ice, c, stream, &stats, is_local);
+ g_array_append_val (result, stats);
+ }
+}
+
+GArray *
+gst_webrtc_ice_get_local_candidates (GstWebRTCICE * ice,
+ GstWebRTCICEStream * stream)
+{
+ GSList *cands = NULL;
+
+ GArray *result =
+ g_array_new (FALSE, TRUE, sizeof (GstWebRTCICECandidateStats));
+
+ cands = nice_agent_get_local_candidates (ice->priv->nice_agent,
+ stream->stream_id, NICE_COMPONENT_TYPE_RTP);
+
+ _populate_candidate_list_stats (ice, cands, stream, result, TRUE);
+ g_slist_free_full (cands, (GDestroyNotify) nice_candidate_free);
+
+ return result;
+}
+
+GArray *
+gst_webrtc_ice_get_remote_candidates (GstWebRTCICE * ice,
+ GstWebRTCICEStream * stream)
+{
+ GSList *cands = NULL;
+
+ GArray *result =
+ g_array_new (FALSE, TRUE, sizeof (GstWebRTCICECandidateStats));
+
+ cands = nice_agent_get_remote_candidates (ice->priv->nice_agent,
+ stream->stream_id, NICE_COMPONENT_TYPE_RTP);
+
+ _populate_candidate_list_stats (ice, cands, stream, result, FALSE);
+ g_slist_free_full (cands, (GDestroyNotify) nice_candidate_free);
+
+ return result;
+}
+
+gboolean
+gst_webrtc_ice_get_selected_pair (GstWebRTCICE * ice,
+ GstWebRTCICEStream * stream, GstWebRTCICECandidateStats ** local_stats,
+ GstWebRTCICECandidateStats ** remote_stats)
+{
+ NiceCandidate *local_cand = NULL;
+ NiceCandidate *remote_cand = NULL;
+
+ if (stream) {
+ if (nice_agent_get_selected_pair (ice->priv->nice_agent, stream->stream_id,
+ NICE_COMPONENT_TYPE_RTP, &local_cand, &remote_cand)) {
+ *local_stats = g_new0 (GstWebRTCICECandidateStats, 1);
+ _populate_candidate_stats (ice, local_cand, stream, *local_stats, TRUE);
+
+ *remote_stats = g_new0 (GstWebRTCICECandidateStats, 1);
+ _populate_candidate_stats (ice, remote_cand, stream, *remote_stats,
+ FALSE);
+
+ return TRUE;
+ }
+ }
+
+ return FALSE;
+}
+
static void
_clear_ice_stream (struct NiceStreamItem *item)
{
#include "gstwebrtcstats.h"
#include "gstwebrtcbin.h"
+#include "icestream.h"
#include "transportstream.h"
#include "transportreceivebin.h"
#include "utils.h"
}
}
+/* https://www.w3.org/TR/webrtc-stats/#icecandidate-dict* */
+static gchar *
+_get_stats_from_ice_candidates (GstWebRTCBin * webrtc,
+ GstWebRTCICECandidateStats * can, const gchar * transport_id,
+ const gchar * candidate_tag, GstStructure * s)
+{
+ GstStructure *stats;
+ GstWebRTCStatsType type;
+ gchar *id;
+ double ts;
+
+ gst_structure_get_double (s, "timestamp", &ts);
+
+ id = g_strdup_printf ("ice-candidate-%s_%u_%s_%u", candidate_tag,
+ can->stream_id, can->ipaddr, can->port);
+ stats = gst_structure_new_empty (id);
+
+ if (strcmp (candidate_tag, "local")) {
+ type = GST_WEBRTC_STATS_LOCAL_CANDIDATE;
+ } else if (strcmp (candidate_tag, "remote")) {
+ type = GST_WEBRTC_STATS_REMOTE_CANDIDATE;
+ } else {
+ GST_WARNING_OBJECT (webrtc, "Invalid ice candidate tag: %s", candidate_tag);
+ return NULL;
+ }
+ _set_base_stats (stats, type, ts, id);
+
+ /* RTCIceCandidateStats
+ DOMString transportId;
+ DOMString address;
+ long port;
+ DOMString protocol;
+ RTCIceCandidateType candidateType;
+ long priority;
+ DOMString url;
+ DOMString relayProtocol;
+ */
+
+ if (transport_id)
+ gst_structure_set (stats, "transport-id", G_TYPE_STRING, transport_id,
+ NULL);
+ gst_structure_set (stats, "address", G_TYPE_STRING, can->ipaddr, NULL);
+ gst_structure_set (stats, "port", G_TYPE_UINT, can->port, NULL);
+ gst_structure_set (stats, "candidate-type", G_TYPE_STRING, can->type, NULL);
+ gst_structure_set (stats, "priority", G_TYPE_UINT, can->prio, NULL);
+ gst_structure_set (stats, "protocol", G_TYPE_STRING, can->proto, NULL);
+ if (can->relay_proto)
+ gst_structure_set (stats, "relay-protocol", G_TYPE_STRING, can->relay_proto,
+ NULL);
+ if (can->url)
+ gst_structure_set (stats, "url", G_TYPE_STRING, can->url, NULL);
+
+ gst_structure_set (s, id, GST_TYPE_STRUCTURE, stats, NULL);
+ gst_structure_free (stats);
+
+ return id;
+}
+
/* https://www.w3.org/TR/webrtc-stats/#candidatepair-dict* */
static gchar *
_get_stats_from_ice_transport (GstWebRTCBin * webrtc,
- GstWebRTCICETransport * transport, const GstStructure * twcc_stats,
+ GstWebRTCICETransport * transport, GstWebRTCICEStream * stream,
+ const GstStructure * twcc_stats, const gchar * transport_id,
GstStructure * s)
{
GstStructure *stats;
gchar *id;
+ gchar *local_cand_id = NULL, *remote_cand_id = NULL;
double ts;
+ GstWebRTCICECandidateStats *local_cand = NULL, *remote_cand = NULL;
gst_structure_get_double (s, "timestamp", &ts);
id = g_strdup_printf ("ice-candidate-pair_%s", GST_OBJECT_NAME (transport));
stats = gst_structure_new_empty (id);
- _set_base_stats (stats, GST_WEBRTC_STATS_TRANSPORT, ts, id);
-
-/* XXX: RTCIceCandidatePairStats
- DOMString transportId;
- DOMString localCandidateId;
- DOMString remoteCandidateId;
- RTCStatsIceCandidatePairState state;
- unsigned long long priority;
- boolean nominated;
- unsigned long packetsSent;
- unsigned long packetsReceived;
- unsigned long long bytesSent;
- unsigned long long bytesReceived;
- DOMHighResTimeStamp lastPacketSentTimestamp;
- DOMHighResTimeStamp lastPacketReceivedTimestamp;
- DOMHighResTimeStamp firstRequestTimestamp;
- DOMHighResTimeStamp lastRequestTimestamp;
- DOMHighResTimeStamp lastResponseTimestamp;
- double totalRoundTripTime;
- double currentRoundTripTime;
- double availableOutgoingBitrate;
- double availableIncomingBitrate;
- unsigned long circuitBreakerTriggerCount;
- unsigned long long requestsReceived;
- unsigned long long requestsSent;
- unsigned long long responsesReceived;
- unsigned long long responsesSent;
- unsigned long long retransmissionsReceived;
- unsigned long long retransmissionsSent;
- unsigned long long consentRequestsSent;
- DOMHighResTimeStamp consentExpiredTimestamp;
-*/
+ _set_base_stats (stats, GST_WEBRTC_STATS_CANDIDATE_PAIR, ts, id);
+
+ /* RTCIceCandidatePairStats
+ DOMString transportId;
+ DOMString localCandidateId;
+ DOMString remoteCandidateId;
+
+ XXX: To be added:
+
+ RTCStatsIceCandidatePairState state;
+ boolean nominated;
+ unsigned long packetsSent;
+ unsigned long packetsReceived;
+ unsigned long long bytesSent;
+ unsigned long long bytesReceived;
+ DOMHighResTimeStamp lastPacketSentTimestamp;
+ DOMHighResTimeStamp lastPacketReceivedTimestamp;
+ DOMHighResTimeStamp firstRequestTimestamp;
+ DOMHighResTimeStamp lastRequestTimestamp;
+ DOMHighResTimeStamp lastResponseTimestamp;
+ double totalRoundTripTime;
+ double currentRoundTripTime;
+ double availableOutgoingBitrate;
+ double availableIncomingBitrate;
+ unsigned long circuitBreakerTriggerCount;
+ unsigned long long requestsReceived;
+ unsigned long long requestsSent;
+ unsigned long long responsesReceived;
+ unsigned long long responsesSent;
+ unsigned long long retransmissionsReceived;
+ unsigned long long retransmissionsSent;
+ unsigned long long consentRequestsSent;
+ DOMHighResTimeStamp consentExpiredTimestamp;
+ unsigned long packetsDiscardedOnSend;
+ unsigned long long bytesDiscardedOnSend;
+ unsigned long long requestBytesSent;
+ unsigned long long consentRequestBytesSent;
+ unsigned long long responseBytesSent;
+ */
-/* XXX: RTCIceCandidateStats
- DOMString transportId;
- boolean isRemote;
- RTCNetworkType networkType;
- DOMString ip;
- long port;
- DOMString protocol;
- RTCIceCandidateType candidateType;
- long priority;
- DOMString url;
- DOMString relayProtocol;
- boolean deleted = false;
-};
-*/
+ if (gst_webrtc_ice_get_selected_pair (webrtc->priv->ice, stream,
+ &local_cand, &remote_cand)) {
+ local_cand_id =
+ _get_stats_from_ice_candidates (webrtc, local_cand, transport_id,
+ "local", s);
+ remote_cand_id =
+ _get_stats_from_ice_candidates (webrtc, remote_cand, transport_id,
+ "remote", s);
+
+ gst_structure_set (stats, "local-candidate-id", G_TYPE_STRING,
+ local_cand_id, NULL);
+ gst_structure_set (stats, "remote-candidate-id", G_TYPE_STRING,
+ remote_cand_id, NULL);
+ } else
+ GST_INFO_OBJECT (webrtc,
+ "No selected ICE candidate pair was found for transport %s",
+ GST_OBJECT_NAME (transport));
/* XXX: these stats are at the rtp session level but there isn't a specific
* stats structure for that. The RTCIceCandidatePairStats is the closest with
NULL);
gst_structure_set (s, id, GST_TYPE_STRUCTURE, stats, NULL);
+
+ g_free (local_cand_id);
+ g_free (remote_cand_id);
+
+ if (local_cand) {
+ g_free (local_cand->ipaddr);
+ g_free (local_cand->url);
+ }
+ if (remote_cand)
+ g_free (remote_cand->ipaddr);
+
+ g_free (local_cand);
+ g_free (remote_cand);
+
gst_structure_free (stats);
return id;
/* https://www.w3.org/TR/webrtc-stats/#dom-rtctransportstats */
static gchar *
_get_stats_from_dtls_transport (GstWebRTCBin * webrtc,
- GstWebRTCDTLSTransport * transport, const GstStructure * twcc_stats,
- GstStructure * s)
+ GstWebRTCDTLSTransport * transport, GstWebRTCICEStream * stream,
+ const GstStructure * twcc_stats, GstStructure * s)
{
GstStructure *stats;
gchar *id;
DOMString issuerCertificateId;
*/
-/* XXX: RTCIceCandidateStats
- DOMString transportId;
- boolean isRemote;
- DOMString ip;
- long port;
- DOMString protocol;
- RTCIceCandidateType candidateType;
- long priority;
- DOMString url;
- boolean deleted = false;
-*/
+ ice_id =
+ _get_stats_from_ice_transport (webrtc, transport->transport, stream,
+ twcc_stats, id, s);
+ if (ice_id) {
+ gst_structure_set (stats, "selected-candidate-pair-id", G_TYPE_STRING,
+ ice_id, NULL);
+ g_free (ice_id);
+ }
gst_structure_set (s, id, GST_TYPE_STRUCTURE, stats, NULL);
gst_structure_free (stats);
- ice_id =
- _get_stats_from_ice_transport (webrtc, transport->transport, twcc_stats,
- s);
- g_free (ice_id);
-
return id;
}
ts_stats.transport_id =
_get_stats_from_dtls_transport (webrtc, ts_stats.stream->transport,
- twcc_stats, s);
+ GST_WEBRTC_ICE_STREAM (ts_stats.stream->stream), twcc_stats, s);
GST_DEBUG_OBJECT (webrtc, "retrieving rtp stream stats from transport %"
GST_PTR_FORMAT " rtp session %" GST_PTR_FORMAT " with %u rtp sources, "
g_free (local_id);
}
+static void
+validate_candidate_stats (const GstStructure * s, const GstStructure * stats)
+{
+ guint port;
+ guint64 priority;
+ gchar *address, *candidateType, *protocol;
+
+ fail_unless (gst_structure_get (s, "address", G_TYPE_STRING, &address, NULL));
+ fail_unless (gst_structure_get (s, "port", G_TYPE_UINT, &port, NULL));
+ fail_unless (gst_structure_get (s, "candidate-type", G_TYPE_STRING,
+ &candidateType, NULL));
+ fail_unless (gst_structure_get (s, "priority", G_TYPE_UINT64, &priority,
+ NULL));
+ fail_unless (gst_structure_get (s, "protocol", G_TYPE_STRING, &protocol,
+ NULL));
+
+ fail_unless (strcmp (protocol, "udp") || strcmp (protocol, "tcp"));
+
+ g_free (address);
+ g_free (candidateType);
+ g_free (protocol);
+}
+
static gboolean
validate_stats_foreach (GQuark field_id, const GValue * value,
const GstStructure * stats)
} else if (type == GST_WEBRTC_STATS_TRANSPORT) {
} else if (type == GST_WEBRTC_STATS_CANDIDATE_PAIR) {
} else if (type == GST_WEBRTC_STATS_LOCAL_CANDIDATE) {
+ validate_candidate_stats (s, stats);
} else if (type == GST_WEBRTC_STATS_REMOTE_CANDIDATE) {
+ validate_candidate_stats (s, stats);
} else if (type == GST_WEBRTC_STATS_CERTIFICATE) {
} else {
g_assert_not_reached ();
GST_END_TEST;
+GST_START_TEST (test_stats_with_stream)
+{
+ struct test_webrtc *t = create_audio_test ();
+ GstPromise *p;
+ GstCaps *caps;
+ GstPad *pad;
+
+ /* test that the stats generated with stream are sane */
+
+ t->on_offer_created = NULL;
+ t->on_answer_created = NULL;
+ t->on_negotiation_needed = NULL;
+
+ 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);
+
+ test_webrtc_create_offer (t);
+
+ 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);
+
+ /* set caps for webrtcbin sink to validate codec stats */
+ caps = gst_caps_from_string (OPUS_RTP_CAPS (96));
+ pad = gst_element_get_static_pad (t->webrtc1, "sink_0");
+ gst_pad_set_caps (pad, caps);
+ gst_caps_unref (caps);
+
+ test_webrtc_wait_for_answer_error_eos (t);
+ fail_unless (t->state == STATE_ANSWER_SET);
+
+ p = gst_promise_new_with_change_func (_on_stats, t, NULL);
+ g_signal_emit_by_name (t->webrtc1, "get-stats", NULL, p);
+ p = gst_promise_new_with_change_func (_on_stats, t, NULL);
+ g_signal_emit_by_name (t->webrtc2, "get-stats", NULL, p);
+
+ test_webrtc_wait_for_state_mask (t, 1 << STATE_CUSTOM);
+
+ gst_object_unref (pad);
+ test_webrtc_free (t);
+}
+
+GST_END_TEST;
+
GST_START_TEST (test_add_transceiver)
{
struct test_webrtc *t = test_webrtc_new ();
if (nicesrc && nicesink && dtlssrtpenc && dtlssrtpdec) {
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_audio);
tcase_add_test (tc, test_ice_port_restriction);
tcase_add_test (tc, test_audio_video);
projects / platform / upstream / gstreamer.git / commitdiff