webrtcbin: Improve SDP intersection for Opus

Remove optional sprop-stereo and sprop-maxcapture fields from Opus
remote offer caps before intersecting with local codec preferences.

According to https://datatracker.ietf.org/doc/html/rfc7587#section-7.1
those fields are sender-only informative, and don't affect
interoperability.

Fixes cases where the webrtc media will end up receive-only if the
local side wants to send stereo but the remote is sending mono, or
vice versa.

There may be other fields in other codecs, so the implementation
anticipates needing to add further fields and codecs in the future.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5993>

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

@@ -4542,6 +4542,8 @@ _create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options,
       gst_sdp_media_set_proto (media, "UDP/TLS/RTP/SAVPF");
       offer_caps = _rtp_caps_from_media (offer_media);
 
+      _remove_optional_offer_fields (offer_caps);
+
       if (last_answer && i < gst_sdp_message_medias_len (last_answer)
           && (rtp_trans = _find_transceiver_for_mid (webrtc, mid))) {
         const GstSDPMedia *last_media =

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

@@ -170,6 +170,31 @@ _g_checksum_to_webrtc_string (GChecksumType type)
   }
 }
 
+void
+_remove_optional_offer_fields (GstCaps * offer_caps)
+{
+  int i;
+
+  for (i = 0; i < gst_caps_get_size (offer_caps); i++) {
+    GstStructure *s = gst_caps_get_structure (offer_caps, i);
+    const gchar *mtype = gst_structure_get_string (s, "media");
+    const gchar *encoding_name = gst_structure_get_string (s, "encoding-name");
+
+    if (mtype == NULL || encoding_name == NULL) {
+      continue;
+    }
+
+    /* Special cases for different codecs - sender-only fields
+     * that we don't need to care about for SDP intersection */
+    if (g_str_equal (mtype, "audio")) {
+      if (g_str_equal (encoding_name, "OPUS")) {
+        gst_structure_remove_fields (s, "sprop-stereo", "sprop-maxcapturerate",
+            NULL);
+      }
+    }
+  }
+}
+
 GstCaps *
 _rtp_caps_from_media (const GstSDPMedia * media)
 {

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

@@ -63,6 +63,8 @@ const gchar *                 _enum_value_to_string       (GType type, guint val
 G_GNUC_INTERNAL
 const gchar *           _g_checksum_to_webrtc_string (GChecksumType type);
 G_GNUC_INTERNAL
+void                    _remove_optional_offer_fields (GstCaps *offer_caps);
+G_GNUC_INTERNAL
 GstCaps *               _rtp_caps_from_media        (const GstSDPMedia * media);
 G_GNUC_INTERNAL
 GstWebRTCKind           webrtc_kind_from_caps       (const GstCaps * caps);