From ef71c1319a2f1e97bcf741bc430f58d2bc4db048 Mon Sep 17 00:00:00 2001
From: Jan Schmidt <jan@centricular.com>
Date: Thu, 25 Jan 2024 22:25:19 +1100
Subject: [PATCH] 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>
---
 .../gst-plugins-bad/ext/webrtc/gstwebrtcbin.c |  2 ++
 .../gst-plugins-bad/ext/webrtc/utils.c        | 25 +++++++++++++++++++
 .../gst-plugins-bad/ext/webrtc/utils.h        |  2 ++
 3 files changed, 29 insertions(+)

diff --git a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
index 1f728e386c..a38ff85ef6 100644
--- a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
+++ b/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 =
diff --git a/subprojects/gst-plugins-bad/ext/webrtc/utils.c b/subprojects/gst-plugins-bad/ext/webrtc/utils.c
index f0741d1e55..e3552b3814 100644
--- a/subprojects/gst-plugins-bad/ext/webrtc/utils.c
+++ b/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)
 {
diff --git a/subprojects/gst-plugins-bad/ext/webrtc/utils.h b/subprojects/gst-plugins-bad/ext/webrtc/utils.h
index e5d3d124ab..bcbe906a73 100644
--- a/subprojects/gst-plugins-bad/ext/webrtc/utils.h
+++ b/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);