webrtcbin: Support for setting kind attribute on RTCRtpStreamStats

The attribute maps the `kind` property of the associated transceiver.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3630>

diff --git a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
index 5377aa3..6a8f722 100644 (file)
--- a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
+++ b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
@@ -8746,6 +8746,7 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
    *  "ssrc"                G_TYPE_STRING               the rtp sequence src in use
    *  "transport-id"        G_TYPE_STRING               identifier for the associated RTCTransportStats for this stream
    *  "codec-id"            G_TYPE_STRING               identifier for the associated RTCCodecStats for this stream
+   *  "kind"                G_TYPE_STRING               either "audio" or "video", depending on the associated transceiver (Since: 1.22)
    *
    * RTCReceivedStreamStats supported fields (https://w3c.github.io/webrtc-stats/#receivedrtpstats-dict*)
    *

diff --git a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcstats.c b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcstats.c
index e61ef59..5ff2bd6 100644 (file)
--- a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcstats.c
+++ b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcstats.c
@@ -105,7 +105,7 @@ _gst_structure_take_structure (GstStructure * s, const char *fieldname,
 static gboolean
 _get_stats_from_remote_rtp_source_stats (GstWebRTCBin * webrtc,
     TransportStream * stream, const GstStructure * source_stats,
-    guint ssrc, guint clock_rate, const gchar * codec_id,
+    guint ssrc, guint clock_rate, const gchar * codec_id, const gchar * kind,
     const gchar * transport_id, GstStructure * s)
 {
   gboolean have_rb = FALSE, internal = FALSE;
@@ -135,7 +135,8 @@ _get_stats_from_remote_rtp_source_stats (GstWebRTCBin * webrtc,
   gst_structure_set (r_in, "ssrc", G_TYPE_UINT, ssrc, NULL);
   gst_structure_set (r_in, "codec-id", G_TYPE_STRING, codec_id, NULL);
   gst_structure_set (r_in, "transport-id", G_TYPE_STRING, transport_id, NULL);
-  /* To be added: kind */
+  if (kind)
+    gst_structure_set (r_in, "kind", G_TYPE_STRING, kind, NULL);
 
   /* RTCReceivedRtpStreamStats */
 
@@ -205,7 +206,8 @@ _get_stats_from_remote_rtp_source_stats (GstWebRTCBin * webrtc,
 static void
 _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
     TransportStream * stream, const GstStructure * source_stats,
-    const gchar * codec_id, const gchar * transport_id, GstStructure * s)
+    const gchar * codec_id, const gchar * kind, const gchar * transport_id,
+    GstStructure * s)
 {
   guint ssrc, fir, pli, nack, jitter;
   int clock_rate;
@@ -230,8 +232,8 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
     gst_structure_set (out, "ssrc", G_TYPE_UINT, ssrc, NULL);
     gst_structure_set (out, "codec-id", G_TYPE_STRING, codec_id, NULL);
     gst_structure_set (out, "transport-id", G_TYPE_STRING, transport_id, NULL);
-    /* To be added: kind */
-
+    if (kind)
+      gst_structure_set (out, "kind", G_TYPE_STRING, kind, NULL);
 
     /* RTCSentRtpStreamStats  */
     if (gst_structure_get_uint64 (source_stats, "octets-sent", &bytes))
@@ -350,8 +352,8 @@ _get_stats_from_rtp_source_stats (GstWebRTCBin * webrtc,
     gst_structure_set (in, "ssrc", G_TYPE_UINT, ssrc, NULL);
     gst_structure_set (in, "codec-id", G_TYPE_STRING, codec_id, NULL);
     gst_structure_set (in, "transport-id", G_TYPE_STRING, transport_id, NULL);
-    /* To be added: kind */
-
+    if (kind)
+      gst_structure_set (in, "kind", G_TYPE_STRING, kind, NULL);
 
     /* RTCReceivedRtpStreamStats */
 
@@ -871,6 +873,7 @@ struct transport_stream_stats
   TransportStream *stream;
   char *transport_id;
   char *codec_id;
+  const char *kind;
   guint clock_rate;
   GValueArray *source_stats;
   GstStructure *s;
@@ -897,12 +900,14 @@ webrtc_stats_get_from_transport (SsrcMapItem * entry,
     if (gst_structure_get_uint (stats, "ssrc", &stats_ssrc) &&
         entry->ssrc == stats_ssrc)
       _get_stats_from_rtp_source_stats (ts_stats->webrtc, ts_stats->stream,
-          stats, ts_stats->codec_id, ts_stats->transport_id, ts_stats->s);
-    else if (gst_structure_get_uint (stats, "rb-ssrc", &stats_ssrc) &&
-        entry->ssrc == stats_ssrc)
+          stats, ts_stats->codec_id, ts_stats->kind, ts_stats->transport_id,
+          ts_stats->s);
+    else if (gst_structure_get_uint (stats, "rb-ssrc", &stats_ssrc)
+        && entry->ssrc == stats_ssrc)
       _get_stats_from_remote_rtp_source_stats (ts_stats->webrtc,
           ts_stats->stream, stats, entry->ssrc, ts_stats->clock_rate,
-          ts_stats->codec_id, ts_stats->transport_id, ts_stats->s);
+          ts_stats->codec_id, ts_stats->kind, ts_stats->transport_id,
+          ts_stats->s);
   }
 
   /* we want to look at all the entries */
@@ -918,6 +923,7 @@ _get_stats_from_pad (GstWebRTCBin * webrtc, GstPad * pad, GstStructure * s)
   GObject *rtp_session;
   GObject *gst_rtp_session;
   GstStructure *rtp_stats, *twcc_stats;
+  GstWebRTCKind kind;
 
   _get_codec_stats_from_pad (webrtc, pad, s, &ts_stats.codec_id, &ssrc,
       &clock_rate);
@@ -925,6 +931,19 @@ _get_stats_from_pad (GstWebRTCBin * webrtc, GstPad * pad, GstStructure * s)
   if (!wpad->trans)
     goto out;
 
+  g_object_get (wpad->trans, "kind", &kind, NULL);
+  switch (kind) {
+    case GST_WEBRTC_KIND_AUDIO:
+      ts_stats.kind = "audio";
+      break;
+    case GST_WEBRTC_KIND_VIDEO:
+      ts_stats.kind = "video";
+      break;
+    case GST_WEBRTC_KIND_UNKNOWN:
+      ts_stats.kind = NULL;
+      break;
+  };
+
   ts_stats.stream = WEBRTC_TRANSCEIVER (wpad->trans)->stream;
   if (!ts_stats.stream)
     goto out;

diff --git a/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c b/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c
index b9d1b83..617dee5 100644 (file)
--- a/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c
+++ b/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c
@@ -1517,7 +1517,7 @@ validate_codec_stats (const GstStructure * s)
 static void
 validate_rtc_stream_stats (const GstStructure * s, const GstStructure * stats)
 {
-  gchar *codec_id, *transport_id;
+  gchar *codec_id, *transport_id, *kind;
   GstStructure *codec, *transport;
 
   fail_unless (gst_structure_get (s, "codec-id", G_TYPE_STRING, &codec_id,
@@ -1536,8 +1536,12 @@ validate_rtc_stream_stats (const GstStructure * s, const GstStructure * stats)
   gst_structure_free (transport);
   gst_structure_free (codec);
 
+  fail_unless (gst_structure_get (s, "kind", G_TYPE_STRING, &kind, NULL));
+  fail_unless (g_str_equal (kind, "audio") || g_str_equal (kind, "video"));
+
   g_free (codec_id);
   g_free (transport_id);
+  g_free (kind);
 }
 
 static void