Bump to GStreamer 1.24.11

And add a new patch scheduled to ship in 1.24.12.

images/wkdev_sdk/jhbuild/patches/0001-webrtcbin-create-and-associate-transceivers-earlier-.patch

@@ -1,7 +1,7 @@
-From 8edcb3957601ba1237d3f60bbfc0f115a614aa75 Mon Sep 17 00:00:00 2001
+From e194b6bb173a95dcfb84c328fff426e60e6ab8f9 Mon Sep 17 00:00:00 2001
 From: Carlos Bentzen <[email protected]>
 Date: Wed, 10 Jul 2024 10:34:19 +0200
-Subject: [PATCH 1/9] webrtcbin: create and associate transceivers earlier in
+Subject: [PATCH 1/8] webrtcbin: create and associate transceivers earlier in
  negotation
 
 According to https://w3c.github.io/webrtc-pc/#set-the-session-description
@@ -24,14 +24,14 @@ associated after every session description is set.
 
 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7156>
 ---
- .../gst-plugins-bad/ext/webrtc/gstwebrtcbin.c | 495 ++++++++++--------
+ .../gst-plugins-bad/ext/webrtc/gstwebrtcbin.c | 477 +++++++++++-------
  .../gst-plugins-bad/ext/webrtc/webrtcsdp.c    |  11 +
  .../gst-plugins-bad/ext/webrtc/webrtcsdp.h    |   2 +
  .../tests/check/elements/webrtcbin.c          | 120 ++++-
- 4 files changed, 388 insertions(+), 240 deletions(-)
+ 4 files changed, 388 insertions(+), 222 deletions(-)
 
 diff --git a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
-index f170f512bf..b4196e3435 100644
+index dce4820a6d..225d246576 100644
 --- a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
 +++ b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
 @@ -748,6 +748,13 @@ transceiver_match_for_mid (GstWebRTCRTPTransceiver * trans, const gchar * mid)
@@ -169,8 +169,8 @@ index f170f512bf..b4196e3435 100644
 -          }
 -          rtp_trans = NULL;
 -        }
-       }
- 
+-      }
+-
 -      if (rtp_trans) {
 -        answer_dir = rtp_trans->direction;
 -        g_assert (answer_caps != NULL);
@@ -180,8 +180,8 @@ index f170f512bf..b4196e3435 100644
 -        answer_dir = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY;
 -        GST_WARNING_OBJECT (webrtc, "did not find compatible transceiver for "
 -            "offer caps %" GST_PTR_FORMAT ", will only receive", offer_caps);
--      }
--
+       }
+ 
 -      if (!rtp_trans) {
 -        GstCaps *trans_caps;
 -        GstWebRTCKind kind = GST_WEBRTC_KIND_UNKNOWN;
@@ -253,40 +253,15 @@ index f170f512bf..b4196e3435 100644
      }
    }
 
-@@ -6079,24 +6006,6 @@ _update_data_channel_from_sdp_media (GstWebRTCBin * webrtc,
-   transport_receive_bin_set_receive_state (receive, RECEIVE_STATE_PASS);
- }
-
--static gboolean
--_find_compatible_unassociated_transceiver (GstWebRTCRTPTransceiver * p1,
--    gconstpointer data)
--{
--  GstWebRTCKind kind = GPOINTER_TO_INT (data);
--
--  if (p1->mid)
--    return FALSE;
--  if (p1->mline != -1)
--    return FALSE;
--  if (p1->stopped)
--    return FALSE;
--  if (p1->kind != GST_WEBRTC_KIND_UNKNOWN && p1->kind != kind)
--    return FALSE;
--
--  return TRUE;
--}
--
- static void
- _connect_rtpfunnel (GstWebRTCBin * webrtc, guint session_id)
- {
-@@ -6179,7 +6088,6 @@ _update_transceivers_from_sdp (GstWebRTCBin * webrtc, SDPSource source,
+@@ -6185,7 +6112,6 @@ _update_transceivers_from_sdp (GstWebRTCBin * webrtc, SDPSource source,
    for (i = 0; i < gst_sdp_message_medias_len (sdp->sdp); i++) {
      const GstSDPMedia *media = gst_sdp_message_get_media (sdp->sdp, i);
      TransportStream *stream;
 -    GstWebRTCRTPTransceiver *trans;
      guint transport_idx;
 
      /* skip rejected media */
-@@ -6191,8 +6099,6 @@ _update_transceivers_from_sdp (GstWebRTCBin * webrtc, SDPSource source,
+@@ -6197,8 +6123,6 @@ _update_transceivers_from_sdp (GstWebRTCBin * webrtc, SDPSource source,
      else
        transport_idx = i;
 
@@ -295,7 +270,7 @@ index f170f512bf..b4196e3435 100644
      stream = _get_or_create_transport_stream (webrtc, transport_idx,
          _message_media_is_datachannel (sdp->sdp, transport_idx));
      if (!bundled) {
-@@ -6203,60 +6109,28 @@ _update_transceivers_from_sdp (GstWebRTCBin * webrtc, SDPSource source,
+@@ -6209,60 +6133,28 @@ _update_transceivers_from_sdp (GstWebRTCBin * webrtc, SDPSource source,
        ensure_rtx_hdr_ext (stream);
      }
 
@@ -310,10 +285,7 @@ index f170f512bf..b4196e3435 100644
 -      if (g_strcmp0 (gst_sdp_media_get_media (media), "audio") == 0 ||
 -          g_strcmp0 (gst_sdp_media_get_media (media), "video") == 0) {
 -        GstWebRTCKind kind = GST_WEBRTC_KIND_UNKNOWN;
-+    if (g_strcmp0 (gst_sdp_media_get_media (media), "audio") == 0 ||
-+        g_strcmp0 (gst_sdp_media_get_media (media), "video") == 0) {
-+      GstWebRTCRTPTransceiver *trans;
-
+-
 -        /* No existing transceiver, find an unused one */
 -        if (!trans) {
 -          if (g_strcmp0 (gst_sdp_media_get_media (media), "audio") == 0)
@@ -343,7 +315,10 @@ index f170f512bf..b4196e3435 100644
 -              gst_webrtc_bin_signals[ON_NEW_TRANSCEIVER_SIGNAL], 0, trans);
 -          PC_LOCK (webrtc);
 -        }
--
++    if (g_strcmp0 (gst_sdp_media_get_media (media), "audio") == 0 ||
++        g_strcmp0 (gst_sdp_media_get_media (media), "video") == 0) {
++      GstWebRTCRTPTransceiver *trans;
+
 -        _update_transceiver_from_sdp_media (webrtc, sdp->sdp, i, stream,
 -            trans, bundled, bundle_idx, error);
 -        if (error && *error)
@@ -376,7 +351,7 @@ index f170f512bf..b4196e3435 100644
      }
    }
 
-@@ -6406,6 +6280,210 @@ get_last_generated_description (GstWebRTCBin * webrtc, SDPSource source,
+@@ -6412,6 +6304,210 @@ get_last_generated_description (GstWebRTCBin * webrtc, SDPSource source,
    return NULL;
  }
 
@@ -587,7 +562,7 @@ index f170f512bf..b4196e3435 100644
 
  /* http://w3c.github.io/webrtc-pc/#set-description */
  static GstStructure *
-@@ -6568,21 +6646,8 @@ _set_description_task (GstWebRTCBin * webrtc, struct set_description *sd)
+@@ -6574,21 +6670,8 @@ _set_description_task (GstWebRTCBin * webrtc, struct set_description *sd)
      }
    }
 
@@ -611,7 +586,7 @@ index f170f512bf..b4196e3435 100644
 
    if (webrtc->signaling_state != new_signaling_state) {
      webrtc->signaling_state = new_signaling_state;
-@@ -6642,6 +6707,12 @@ _set_description_task (GstWebRTCBin * webrtc, struct set_description *sd)
+@@ -6648,6 +6731,12 @@ _set_description_task (GstWebRTCBin * webrtc, struct set_description *sd)
          continue;
        }
 

images/wkdev_sdk/jhbuild/patches/0002-webrtcbin-reverse-direction-from-remote-media.patch

@@ -1,7 +1,7 @@
-From 8615a8ac712bc173c0c3585392683053b7b8ee94 Mon Sep 17 00:00:00 2001
+From 2e69c18ecb12a66ce92f7e3c591343916856c1b1 Mon Sep 17 00:00:00 2001
 From: Carlos Bentzen <[email protected]>
 Date: Fri, 2 Aug 2024 11:19:56 +0200
-Subject: [PATCH 2/9] webrtcbin: reverse direction from remote media
+Subject: [PATCH 2/8] webrtcbin: reverse direction from remote media
 
 This had been overlooked from the spec. We need to reverse
 the remote media direction when setting the transceiver direction.
@@ -12,10 +12,10 @@ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/72
  1 file changed, 26 insertions(+), 3 deletions(-)
 
 diff --git a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
-index b4196e3435..5ad6550d88 100644
+index 225d246576..bc5ebb1909 100644
 --- a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
 +++ b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
-@@ -6280,6 +6280,22 @@ get_last_generated_description (GstWebRTCBin * webrtc, SDPSource source,
+@@ -6304,6 +6304,22 @@ get_last_generated_description (GstWebRTCBin * webrtc, SDPSource source,
    return NULL;
  }
 
@@ -38,7 +38,7 @@ index b4196e3435..5ad6550d88 100644
  /* https://w3c.github.io/webrtc-pc/#set-description (steps in 4.6.10.) */
  static gboolean
  _create_and_associate_transceivers_from_sdp (GstWebRTCBin * webrtc,
-@@ -6328,12 +6344,14 @@ _create_and_associate_transceivers_from_sdp (GstWebRTCBin * webrtc,
+@@ -6352,12 +6368,14 @@ _create_and_associate_transceivers_from_sdp (GstWebRTCBin * webrtc,
      const gchar *mid;
      guint transport_idx;
      TransportStream *stream;
@@ -53,7 +53,7 @@ index b4196e3435..5ad6550d88 100644
 
      /* XXX: not strictly required but a lot of functionality requires a mid */
      if (!mid) {
-@@ -6390,8 +6408,6 @@ _create_and_associate_transceivers_from_sdp (GstWebRTCBin * webrtc,
+@@ -6414,8 +6432,6 @@ _create_and_associate_transceivers_from_sdp (GstWebRTCBin * webrtc,
           * that were added to the PeerConnection by addTrack and are not associated with any "m=" section
           * and are not stopped, find the first (according to the canonical order described in Section 5.2.1)
           * such RtpTransceiver. */
@@ -62,7 +62,7 @@ index b4196e3435..5ad6550d88 100644
          if (direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV
              || direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY) {
            int j;
-@@ -6464,11 +6480,18 @@ _create_and_associate_transceivers_from_sdp (GstWebRTCBin * webrtc,
+@@ -6488,11 +6504,18 @@ _create_and_associate_transceivers_from_sdp (GstWebRTCBin * webrtc,
        trans->mid = g_strdup (mid);
        g_object_notify (G_OBJECT (trans), "mid");
 

images/wkdev_sdk/jhbuild/patches/0003-webrtcbin-connect-output-stream-on-recv-transceivers.patch

@@ -1,7 +1,7 @@
-From cd27befcb7a75a1cca00027a195c1f40aa8f0c26 Mon Sep 17 00:00:00 2001
+From 5bd0033bc9683610a74f668e67672fc524628fb2 Mon Sep 17 00:00:00 2001
 From: Carlos Bentzen <[email protected]>
 Date: Fri, 2 Aug 2024 11:21:13 +0200
-Subject: [PATCH 3/9] webrtcbin: connect output stream on recv transceivers
+Subject: [PATCH 3/8] webrtcbin: connect output stream on recv transceivers
 
 With MR 7156, transceivers and transports are created earlier,
 but for sendrecv media we could get `not-linked` errors due to
@@ -19,10 +19,10 @@ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/72
  2 files changed, 68 insertions(+)
 
 diff --git a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
-index 5ad6550d88..ec8fc47490 100644
+index bc5ebb1909..02652c0362 100644
 --- a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
 +++ b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
-@@ -6500,6 +6500,12 @@ _create_and_associate_transceivers_from_sdp (GstWebRTCBin * webrtc,
+@@ -6524,6 +6524,12 @@ _create_and_associate_transceivers_from_sdp (GstWebRTCBin * webrtc,
          webrtc_transceiver_set_transport (wtrans, stream);
        }
      }

images/wkdev_sdk/jhbuild/patches/0004-webrtc-Fixes-for-matching-pads-to-unassociated-trans.patch

@@ -1,7 +1,7 @@
-From aaf06f221975d4ef771e81438da7179b4b3bdd00 Mon Sep 17 00:00:00 2001
+From 11d381a3d77848e62189df20ba4e7fe8c9dadd45 Mon Sep 17 00:00:00 2001
 From: Jan Schmidt <[email protected]>
 Date: Wed, 24 Jul 2024 20:59:51 +1000
-Subject: [PATCH 4/9] webrtc: Fixes for matching pads to unassociated
+Subject: [PATCH 4/8] webrtc: Fixes for matching pads to unassociated
  transceivers
 
 Fix an inverted condition when checking if sink pad caps match
@@ -17,10 +17,10 @@ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/72
  1 file changed, 3 insertions(+), 3 deletions(-)
 
 diff --git a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
-index ec8fc47490..6a9484a2bc 100644
+index 02652c0362..6fa560b554 100644
 --- a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
 +++ b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
-@@ -8355,9 +8355,9 @@ gst_webrtc_bin_request_new_pad (GstElement * element, GstPadTemplate * templ,
+@@ -8379,9 +8379,9 @@ gst_webrtc_bin_request_new_pad (GstElement * element, GstPadTemplate * templ,
        GstWebRTCBinPad *pad2;
        gboolean has_matching_caps;
 
@@ -32,7 +32,7 @@ index ec8fc47490..6a9484a2bc 100644
          continue;
 
        /* Ignore stopped transmitters */
-@@ -8379,7 +8379,7 @@ gst_webrtc_bin_request_new_pad (GstElement * element, GstPadTemplate * templ,
+@@ -8403,7 +8403,7 @@ gst_webrtc_bin_request_new_pad (GstElement * element, GstPadTemplate * templ,
 
        GST_OBJECT_LOCK (tmptrans);
        has_matching_caps = (caps && tmptrans->codec_preferences &&

images/wkdev_sdk/jhbuild/patches/0005-webrtcbin-Fix-renegotiation-checks.patch

@@ -1,7 +1,7 @@
-From 22e49d68ab4095379df2f0e3fa09ee61f3fb5624 Mon Sep 17 00:00:00 2001
+From 8be67d014ff50275ad0dd0a70f81bb93b24033c6 Mon Sep 17 00:00:00 2001
 From: Jan Schmidt <[email protected]>
 Date: Thu, 1 Aug 2024 13:42:52 +1000
-Subject: [PATCH 5/9] webrtcbin: Fix renegotiation checks
+Subject: [PATCH 5/8] webrtcbin: Fix renegotiation checks
 
 When checking for renegotiation against a local offer,
 reverse the remote direction in the corresponding answer
@@ -27,7 +27,7 @@ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/73
  2 files changed, 151 insertions(+), 32 deletions(-)
 
 diff --git a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
-index 6a9484a2bc..d8cf348f99 100644
+index 6fa560b554..6fb42da6f4 100644
 --- a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
 +++ b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
 @@ -1772,6 +1772,22 @@ done:
@@ -109,7 +109,7 @@ index 6a9484a2bc..d8cf348f99 100644
                gst_webrtc_rtp_transceiver_direction_to_string (remote_dir));
            return TRUE;
          }
-@@ -6280,22 +6302,6 @@ get_last_generated_description (GstWebRTCBin * webrtc, SDPSource source,
+@@ -6304,22 +6326,6 @@ get_last_generated_description (GstWebRTCBin * webrtc, SDPSource source,
    return NULL;
  }
 

images/wkdev_sdk/jhbuild/patches/0006-webrtc-add-all-SSRC-attributes-getting-CAPS-for-a-PT.patch

@@ -1,7 +1,7 @@
-From e2db93c386bb133ca7e4d3641007771de573695b Mon Sep 17 00:00:00 2001
+From 41fa9831893c9ee434840b4d1bf1150ab124f762 Mon Sep 17 00:00:00 2001
 From: =?UTF-8?q?Fran=C3=A7ois=20Laignel?= <[email protected]>
 Date: Thu, 7 Mar 2024 17:36:33 +0100
-Subject: [PATCH 6/9] webrtc: add all SSRC attributes getting CAPS for a PT
+Subject: [PATCH 6/8] webrtc: add all SSRC attributes getting CAPS for a PT
 
 The transport stream only returned the CAPS for the first matching PT entry
 from the `ptmap`. Other SSRC with the same PT where not included. For a stream
@@ -24,7 +24,7 @@ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/61
  3 files changed, 57 insertions(+), 6 deletions(-)
 
 diff --git a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
-index d8cf348f99..2b920313d6 100644
+index 6fb42da6f4..8ac9850a62 100644
 --- a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
 +++ b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
 @@ -5080,6 +5080,8 @@ _set_internal_rtpbin_element_props_from_stream (GstWebRTCBin * webrtc,
@@ -36,7 +36,7 @@ index d8cf348f99..2b920313d6 100644
    }
 
    GST_DEBUG_OBJECT (stream, "setting payload map on %" GST_PTR_FORMAT " : %"
-@@ -7562,8 +7564,7 @@ on_rtpbin_request_pt_map (GstElement * rtpbin, guint session_id, guint pt,
+@@ -7586,8 +7588,7 @@ on_rtpbin_request_pt_map (GstElement * rtpbin, guint session_id, guint pt,
    if (!stream)
      goto unknown_session;