webrtc: implement initial simulcast fec/rtx usage

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1664>

subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c

@@ -3052,19 +3052,20 @@ _gather_extmap (GstCaps * caps, GError ** error)
   return edata.extmap;
 }
 
-struct has_hdrext
+struct hdrext_id
 {
-  const char *rtphdrext;
-  gboolean result;
+  const char *rtphdrext_uri;
+  guint ext_id;
 };
 
 static gboolean
-structure_value_has_rtphdrext (GQuark field_id, const GValue * value,
+structure_value_get_rtphdrext_id (GQuark field_id, const GValue * value,
     gpointer user_data)
 {
-  struct has_hdrext *rtphdrext = user_data;
+  struct hdrext_id *rtphdrext = user_data;
+  const char *field_name = g_quark_to_string (field_id);
 
-  if (g_str_has_prefix (g_quark_to_string (field_id), "extmap-")) {
+  if (g_str_has_prefix (field_name, "extmap-")) {
     const char *val = NULL;
 
     if (GST_VALUE_HOLDS_ARRAY (value) && gst_value_array_get_size (value) >= 2) {
@@ -3074,8 +3075,12 @@ structure_value_has_rtphdrext (GQuark field_id, const GValue * value,
       val = g_value_get_string (value);
     }
 
-    if (g_strcmp0 (val, rtphdrext->rtphdrext) == 0) {
-      rtphdrext->result = TRUE;
+    if (g_strcmp0 (val, rtphdrext->rtphdrext_uri) == 0) {
+      gint64 id = g_ascii_strtoll (&field_name[strlen ("extmap-")], NULL, 10);
+
+      if (id > 0 && id < 256)
+        rtphdrext->ext_id = id;
+
       return FALSE;
     }
   }
@@ -3083,16 +3088,32 @@ structure_value_has_rtphdrext (GQuark field_id, const GValue * value,
   return TRUE;
 }
 
+// Returns -1 when not found
+static guint
+caps_get_rtp_header_extension_id (const GstCaps * caps,
+    const char *rtphdrext_uri)
+{
+  guint i, n;
+
+  n = gst_caps_get_size (caps);
+  for (i = 0; i < n; i++) {
+    const GstStructure *s = gst_caps_get_structure (caps, i);
+    struct hdrext_id data = { rtphdrext_uri, -1 };
+
+    gst_structure_foreach (s, structure_value_get_rtphdrext_id, &data);
+
+    if (data.ext_id != -1)
+      return data.ext_id;
+  }
+
+  return -1;
+}
+
 static gboolean
 caps_contain_rtp_header_extension (const GstCaps * caps,
-    const char *rtphdrextname)
+    const char *rtphdrext_uri)
 {
-  const GstStructure *s = gst_caps_get_structure (caps, 0);
-  struct has_hdrext data = { rtphdrextname, FALSE };
-
-  gst_structure_foreach (s, structure_value_has_rtphdrext, &data);
-
-  return data.result;
+  return caps_get_rtp_header_extension_id (caps, rtphdrext_uri) != -1;
 }
 
 static gboolean
@@ -5108,6 +5129,109 @@ _filter_sdp_fields (GQuark field_id, const GValue * value,
   return TRUE;
 }
 
+static guint
+transport_stream_ptmap_get_rtp_header_extension_id (TransportStream * stream,
+    const char *rtphdrext_uri)
+{
+  guint i;
+
+  for (i = 0; i < stream->ptmap->len; i++) {
+    PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
+    guint id;
+
+    id = caps_get_rtp_header_extension_id (item->caps, rtphdrext_uri);
+    if (id != -1)
+      return id;
+  }
+
+  return -1;
+}
+
+static void
+ensure_rtx_hdr_ext (TransportStream * stream)
+{
+  stream->rtphdrext_id_stream_id =
+      transport_stream_ptmap_get_rtp_header_extension_id (stream,
+      RTPHDREXT_STREAM_ID);
+  stream->rtphdrext_id_repaired_stream_id =
+      transport_stream_ptmap_get_rtp_header_extension_id (stream,
+      RTPHDREXT_REPAIRED_STREAM_ID);
+
+  /* TODO: removing header extensions usage from rtx on renegotiation */
+
+  if (stream->rtxsend) {
+    if (stream->rtphdrext_id_stream_id != -1 && !stream->rtxsend_stream_id) {
+      stream->rtxsend_stream_id =
+          gst_rtp_header_extension_create_from_uri (RTPHDREXT_STREAM_ID);
+      if (!stream->rtxsend_stream_id)
+        g_warn_if_reached ();
+      gst_rtp_header_extension_set_id (stream->rtxsend_stream_id,
+          stream->rtphdrext_id_stream_id);
+
+      GST_DEBUG_OBJECT (stream, "adding rtp header extension %" GST_PTR_FORMAT
+          " with id %u to %" GST_PTR_FORMAT, stream->rtxsend_stream_id,
+          stream->rtphdrext_id_stream_id, stream->rtxsend);
+
+      g_signal_emit_by_name (stream->rtxsend, "add-extension",
+          stream->rtxsend_stream_id);
+    }
+
+    if (stream->rtphdrext_id_repaired_stream_id != -1
+        && !stream->rtxsend_repaired_stream_id) {
+      stream->rtxsend_repaired_stream_id =
+          gst_rtp_header_extension_create_from_uri
+          (RTPHDREXT_REPAIRED_STREAM_ID);
+      if (!stream->rtxsend_repaired_stream_id)
+        g_warn_if_reached ();
+      gst_rtp_header_extension_set_id (stream->rtxsend_repaired_stream_id,
+          stream->rtphdrext_id_repaired_stream_id);
+
+      GST_DEBUG_OBJECT (stream, "adding rtp header extension %" GST_PTR_FORMAT
+          " with id %u to %" GST_PTR_FORMAT, stream->rtxsend_repaired_stream_id,
+          stream->rtphdrext_id_repaired_stream_id, stream->rtxsend);
+
+      g_signal_emit_by_name (stream->rtxsend, "add-extension",
+          stream->rtxsend_repaired_stream_id);
+    }
+  }
+
+  if (stream->rtxreceive) {
+    if (stream->rtphdrext_id_stream_id != -1 && !stream->rtxreceive_stream_id) {
+      stream->rtxreceive_stream_id =
+          gst_rtp_header_extension_create_from_uri (RTPHDREXT_STREAM_ID);
+      if (!stream->rtxreceive_stream_id)
+        g_warn_if_reached ();
+      gst_rtp_header_extension_set_id (stream->rtxreceive_stream_id,
+          stream->rtphdrext_id_stream_id);
+
+      GST_DEBUG_OBJECT (stream, "adding rtp header extension %" GST_PTR_FORMAT
+          " with id %u to %" GST_PTR_FORMAT, stream->rtxsend_stream_id,
+          stream->rtphdrext_id_stream_id, stream->rtxreceive);
+
+      g_signal_emit_by_name (stream->rtxreceive, "add-extension",
+          stream->rtxreceive_stream_id);
+    }
+
+    if (stream->rtphdrext_id_repaired_stream_id != -1
+        && !stream->rtxreceive_repaired_stream_id) {
+      stream->rtxreceive_repaired_stream_id =
+          gst_rtp_header_extension_create_from_uri
+          (RTPHDREXT_REPAIRED_STREAM_ID);
+      if (!stream->rtxreceive_repaired_stream_id)
+        g_warn_if_reached ();
+      gst_rtp_header_extension_set_id (stream->rtxreceive_repaired_stream_id,
+          stream->rtphdrext_id_repaired_stream_id);
+
+      GST_DEBUG_OBJECT (stream, "adding rtp header extension %" GST_PTR_FORMAT
+          " with id %u to %" GST_PTR_FORMAT, stream->rtxsend_repaired_stream_id,
+          stream->rtphdrext_id_repaired_stream_id, stream->rtxreceive);
+
+      g_signal_emit_by_name (stream->rtxreceive, "add-extension",
+          stream->rtxreceive_repaired_stream_id);
+    }
+  }
+}
+
 static void
 _update_transport_ptmap_from_media (GstWebRTCBin * webrtc,
     TransportStream * stream, const GstSDPMessage * sdp, guint media_idx)
@@ -5636,6 +5760,7 @@ _update_transceivers_from_sdp (GstWebRTCBin * webrtc, SDPSource source,
        * parameters aren't set up properly for the bundled streams */
       _update_transport_ptmap_from_media (webrtc, bundle_stream, sdp->sdp, i);
     }
+    ensure_rtx_hdr_ext (bundle_stream);
 
     _connect_rtpfunnel (webrtc, bundle_idx);
   }
@@ -5664,6 +5789,7 @@ _update_transceivers_from_sdp (GstWebRTCBin * webrtc, SDPSource source,
        * bundling we need to do it now */
       g_array_set_size (stream->ptmap, 0);
       _update_transport_ptmap_from_media (webrtc, stream, sdp->sdp, i);
+      ensure_rtx_hdr_ext (stream);
     }
 
     if (trans)
@@ -6895,6 +7021,7 @@ on_rtpbin_request_aux_sender (GstElement * rtpbin, guint session_id,
 
   if (!gst_bin_add (GST_BIN (ret), rtx))
     g_warn_if_reached ();
+  ensure_rtx_hdr_ext (stream);
 
   stream->rtxsend = gst_object_ref (rtx);
   _set_internal_rtpbin_element_props_from_stream (webrtc, stream);
@@ -6955,6 +7082,8 @@ on_rtpbin_request_aux_receiver (GstElement * rtpbin, guint session_id,
   if (!gst_bin_add (GST_BIN (ret), stream->rtxreceive))
     g_warn_if_reached ();
 
+  ensure_rtx_hdr_ext (stream);
+
   stream->reddec = gst_element_factory_make ("rtpreddec", NULL);
   gst_object_ref (stream->reddec);
   if (!gst_bin_add (GST_BIN (ret), stream->reddec))

subprojects/gst-plugins-bad/ext/webrtc/transportstream.c

@@ -192,6 +192,11 @@ transport_stream_finalize (GObject * object)
   g_array_free (stream->ptmap, TRUE);
   g_ptr_array_free (stream->ssrcmap, TRUE);
 
+  gst_clear_object (&stream->rtxsend_stream_id);
+  gst_clear_object (&stream->rtxsend_repaired_stream_id);
+  gst_clear_object (&stream->rtxreceive_stream_id);
+  gst_clear_object (&stream->rtxreceive_repaired_stream_id);
+
   G_OBJECT_CLASS (parent_class)->finalize (object);
 }
 
@@ -365,6 +370,9 @@ transport_stream_init (TransportStream * stream)
   g_array_set_clear_func (stream->ptmap, (GDestroyNotify) clear_ptmap_item);
   stream->ssrcmap = g_ptr_array_new_with_free_func (
       (GDestroyNotify) ssrcmap_item_free);
+
+  stream->rtphdrext_id_stream_id = -1;
+  stream->rtphdrext_id_repaired_stream_id = -1;
 }
 
 TransportStream *

subprojects/gst-plugins-bad/ext/webrtc/transportstream.h

@@ -21,6 +21,7 @@
 #define __TRANSPORT_STREAM_H__
 
 #include "fwd.h"
+#include <gst/rtp/rtp.h>
 #include <gst/webrtc/rtptransceiver.h>
 
 G_BEGIN_DECLS
@@ -65,8 +66,14 @@ struct _TransportStream
   GPtrArray                *ssrcmap;                /* array of SsrcMapItem's */
   gboolean                  output_connected;       /* whether receive bin is connected to rtpbin */
 
+  guint                     rtphdrext_id_stream_id;
+  guint                     rtphdrext_id_repaired_stream_id;
   GstElement               *rtxsend;
+  GstRTPHeaderExtension    *rtxsend_stream_id;
+  GstRTPHeaderExtension    *rtxsend_repaired_stream_id;
   GstElement               *rtxreceive;
+  GstRTPHeaderExtension    *rtxreceive_stream_id;
+  GstRTPHeaderExtension    *rtxreceive_repaired_stream_id;
 
   GstElement               *reddec;
   GList                    *fecdecs;

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

@@ -5014,10 +5014,11 @@ on_sdp_media_rid (struct test_webrtc *t, GstElement * element,
   }
 }
 
-GST_START_TEST (test_simulcast)
+static void
+do_test_simulcast (gboolean enable_fec_rtx)
 {
   struct test_webrtc *t = test_webrtc_new ();
-  guint media_format_count[] = { 1, };
+  guint media_format_count[] = { enable_fec_rtx ? 5 : 1, };
   VAL_SDP_INIT (media_formats, on_sdp_media_count_formats,
       media_format_count, NULL);
   VAL_SDP_INIT (payloads, on_sdp_media_no_duplicate_payloads, NULL,
@@ -5068,6 +5069,13 @@ GST_START_TEST (test_simulcast)
   gst_util_set_object_arg (G_OBJECT (t->webrtc2), "bundle-policy",
       "max-bundle");
 
+  if (enable_fec_rtx) {
+    g_signal_connect (t->webrtc1, "on-new-transceiver",
+        G_CALLBACK (on_new_transceiver_set_rtx_fec), NULL);
+    g_signal_connect (t->webrtc2, "on-new-transceiver",
+        G_CALLBACK (on_new_transceiver_set_rtx_fec), NULL);
+  }
+
   rtpbin2 = gst_bin_get_by_name (GST_BIN (t->webrtc2), "rtpbin");
   fail_unless (rtpbin2 != NULL);
   g_signal_connect (rtpbin2, "new-jitterbuffer",
@@ -5157,6 +5165,18 @@ GST_START_TEST (test_simulcast)
   g_array_unref (ssrcs_received);
 }
 
+GST_START_TEST (test_simulcast)
+{
+  do_test_simulcast (FALSE);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_simulcast_fec_rtx)
+{
+  do_test_simulcast (TRUE);
+}
+
 GST_END_TEST;
 
 static Suite *
@@ -5217,6 +5237,7 @@ webrtcbin_suite (void)
     tcase_add_test (tc, test_bundle_mid_header_extension);
     tcase_add_test (tc, test_max_bundle_fec);
     tcase_add_test (tc, test_simulcast);
+    tcase_add_test (tc, test_simulcast_fec_rtx);
     if (sctpenc && sctpdec) {
       tcase_add_test (tc, test_data_channel_create);
       tcase_add_test (tc, test_data_channel_remote_notify);