gstreamer: Include additional patches

Out of the list only
0009-rtpfunnel-Ensure-segment-events-are-forwarded-after-.patch is going to ship
in 1.24(.10). The others will ship in 1.26.

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

@@ -1,7 +1,7 @@
-From 48ae40f477523bed4cd709d163f541e748356071 Mon Sep 17 00:00:00 2001
+From 8edcb3957601ba1237d3f60bbfc0f115a614aa75 Mon Sep 17 00:00:00 2001
 From: Carlos Bentzen <[email protected]>
 Date: Wed, 10 Jul 2024 10:34:19 +0200
-Subject: [PATCH] webrtcbin: create and associate transceivers earlier in
+Subject: [PATCH 1/9] webrtcbin: create and associate transceivers earlier in
  negotation
 
 According to https://w3c.github.io/webrtc-pc/#set-the-session-description
@@ -31,7 +31,7 @@ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/71
  4 files changed, 388 insertions(+), 240 deletions(-)
 
 diff --git a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
-index 9b84fa317d..7cb0eb6ffc 100644
+index f170f512bf..b4196e3435 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)
@@ -69,7 +69,7 @@ index 9b84fa317d..7cb0eb6ffc 100644
  typedef gboolean (*FindTransportFunc) (TransportStream * p1,
      gconstpointer data);
 
-@@ -4543,146 +4564,51 @@ _create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options,
+@@ -4553,146 +4574,51 @@ _create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options,
 
        _remove_optional_offer_fields (offer_caps);
 
@@ -660,7 +660,7 @@ index 80d21203c2..abeb5dba33 100644
                                                                               GstWebRTCRTPTransceiverDirection answer);
  G_GNUC_INTERNAL
 diff --git a/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c b/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c
-index 8fa8eeaf76..1dd2ddf3c2 100644
+index 7fa337e9ba..adf5014e02 100644
 --- a/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c
 +++ b/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c
 @@ -50,9 +50,11 @@ typedef enum
@@ -890,7 +890,7 @@ index 8fa8eeaf76..1dd2ddf3c2 100644
 
    p = gst_promise_new_with_change_func (_on_stats, t, NULL);
    g_signal_emit_by_name (t->webrtc1, "get-stats", NULL, p);
-@@ -5905,7 +5968,7 @@ GST_START_TEST (test_sdp_session_setup_attribute)
+@@ -5955,7 +6018,7 @@ GST_START_TEST (test_sdp_session_setup_attribute)
    fail_if (gst_element_set_state (t->webrtc2, GST_STATE_READY) ==
        GST_STATE_CHANGE_FAILURE);
    test_webrtc_create_offer (t);
@@ -899,7 +899,7 @@ index 8fa8eeaf76..1dd2ddf3c2 100644
 
    test_webrtc_wait_for_ice_gathering_complete (t);
 
-@@ -5943,6 +6006,7 @@ webrtcbin_suite (void)
+@@ -5993,6 +6056,7 @@ webrtcbin_suite (void)
      tcase_add_test (tc, test_media_direction);
      tcase_add_test (tc, test_add_transceiver);
      tcase_add_test (tc, test_get_transceivers);
@@ -908,5 +908,5 @@ index 8fa8eeaf76..1dd2ddf3c2 100644
      tcase_add_test (tc, test_recvonly_sendonly);
      tcase_add_test (tc, test_payload_types);
 -- 
-2.46.0
+2.47.0
 

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

@@ -1,7 +1,7 @@
-From cad3e63546b17570e284b544afebb5566d75c6d7 Mon Sep 17 00:00:00 2001
+From 8615a8ac712bc173c0c3585392683053b7b8ee94 Mon Sep 17 00:00:00 2001
 From: Carlos Bentzen <[email protected]>
 Date: Fri, 2 Aug 2024 11:19:56 +0200
-Subject: [PATCH] webrtcbin: reverse direction from remote media
+Subject: [PATCH 2/9] 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.
@@ -12,7 +12,7 @@ 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 7cb0eb6ffc..6433d123b2 100644
+index b4196e3435..5ad6550d88 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,
@@ -83,5 +83,5 @@ index 7cb0eb6ffc..6433d123b2 100644
        /* Let transport be the RTCDtlsTransport object representing the RTP/RTCP component of the media transport
         * used by transceiver's associated media description, according to [RFC8843]. */
 -- 
-2.46.0
+2.47.0
 

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

@@ -1,7 +1,7 @@
-From f3bf3ae53c7705823a9179f4df1f279d0342bd63 Mon Sep 17 00:00:00 2001
+From cd27befcb7a75a1cca00027a195c1f40aa8f0c26 Mon Sep 17 00:00:00 2001
 From: Carlos Bentzen <[email protected]>
 Date: Fri, 2 Aug 2024 11:21:13 +0200
-Subject: [PATCH] webrtcbin: connect output stream on recv transceivers
+Subject: [PATCH 3/9] 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
@@ -15,14 +15,14 @@ adds a test for this, so that this doesn't regress anymore.
 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7294>
 ---
  .../gst-plugins-bad/ext/webrtc/gstwebrtcbin.c |  6 ++
- .../tests/check/elements/webrtcbin.c          | 64 +++++++++++++++++++
- 2 files changed, 70 insertions(+)
+ .../tests/check/elements/webrtcbin.c          | 62 +++++++++++++++++++
+ 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 8c8a6ab563..6861b50845 100644
+index 5ad6550d88..ec8fc47490 100644
 --- a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
 +++ b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
-@@ -6490,6 +6490,12 @@ _create_and_associate_transceivers_from_sdp (GstWebRTCBin * webrtc,
+@@ -6500,6 +6500,12 @@ _create_and_associate_transceivers_from_sdp (GstWebRTCBin * webrtc,
          webrtc_transceiver_set_transport (wtrans, stream);
        }
      }
@@ -36,12 +36,12 @@ index 8c8a6ab563..6861b50845 100644
 
    ret = TRUE;
 diff --git a/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c b/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c
-index adf5014e02..bb13887422 100644
+index adf5014e02..2272943a27 100644
 --- a/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c
 +++ b/subprojects/gst-plugins-bad/tests/check/elements/webrtcbin.c
-@@ -4653,6 +4653,69 @@ a=group:BUNDLE \r\n\
-
- GST_END_TEST;
+@@ -4651,6 +4651,67 @@ a=group:BUNDLE \r\n\
+   test_webrtc_free (t);
+ }
 
 +GST_START_TEST (test_audio_sendrecv)
 +{
@@ -104,12 +104,10 @@ index adf5014e02..bb13887422 100644
 +  test_webrtc_free (t);
 +}
 +
-+GST_END_TEST;
-+
+ GST_END_TEST;
+ 
  static void
- new_jitterbuffer_set_fast_start (GstElement * rtpbin,
-     GstElement * rtpjitterbuffer, guint session_id, guint ssrc,
-@@ -6051,6 +6114,7 @@ webrtcbin_suite (void)
+@@ -6051,6 +6112,7 @@ webrtcbin_suite (void)
      tcase_add_test (tc, test_session_stats);
      tcase_add_test (tc, test_stats_with_stream);
      tcase_add_test (tc, test_audio);

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

@@ -0,0 +1,46 @@
+From aaf06f221975d4ef771e81438da7179b4b3bdd00 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
+ transceivers
+
+Fix an inverted condition when checking if sink pad caps match
+the codec-preference of an unassociated transceiver, and
+fix a condition check for transceiver media kind to
+avoid matching sinkpad requests where caps aren't provided
+against unassociated transceivers where the caps might
+not match later.
+
+Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7237>
+---
+ subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c | 6 +++---
+ 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
+--- 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,
+       GstWebRTCBinPad *pad2;
+       gboolean has_matching_caps;
+
+-      /* Ignore transceivers with a non-matching kind */
++      /* Ignore transceivers with a non-matching kind or where we don't know the kind we want */
+       if (tmptrans->kind != GST_WEBRTC_KIND_UNKNOWN &&
+-          kind != GST_WEBRTC_KIND_UNKNOWN && tmptrans->kind != kind)
++          (kind == GST_WEBRTC_KIND_UNKNOWN || tmptrans->kind != kind))
+         continue;
+
+       /* Ignore stopped transmitters */
+@@ -8379,7 +8379,7 @@ gst_webrtc_bin_request_new_pad (GstElement * element, GstPadTemplate * templ,
+
+       GST_OBJECT_LOCK (tmptrans);
+       has_matching_caps = (caps && tmptrans->codec_preferences &&
+-          !gst_caps_can_intersect (caps, tmptrans->codec_preferences));
++          gst_caps_can_intersect (caps, tmptrans->codec_preferences));
+       GST_OBJECT_UNLOCK (tmptrans);
+       /* Ignore transceivers with non-matching caps */
+       if (!has_matching_caps)
+-- 
+2.47.0
+