Attachment 308892 Details for Bug 171595 – Patch

[patch] Patch

wk-gst-1.10.diff (text/plain), 85.34 KB, created by Carlos Garcia Campos on 2017-05-03 03:41:34 PDT

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Creator: Carlos Garcia Campos

Created: 2017-05-03 03:41:34 PDT

Size: 85.34 KB

patch

obsolete

>diff --git a/Tools/ChangeLog b/Tools/ChangeLog
>index 4b1841eb82f..b064908f326 100644
>--- a/Tools/ChangeLog
>+++ b/Tools/ChangeLog
>@@ -1,3 +1,21 @@
>+2017-05-03  Carlos Garcia Campos  <cgarcia@igalia.com>
>+
>+        [GTK] Bump GStreamer version to 1.10.4 in jhbuild
>+        https://bugs.webkit.org/show_bug.cgi?id=171595
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        * gtk/jhbuild.modules:
>+        * gtk/patches/gst-plugins-bad-0001-dtls-port-to-OpenSSL-1.1.0.patch: Removed.
>+        * gtk/patches/gst-plugins-bad-0002-dtlscertificate-Fix-error-checking-in-RSA_generate_k.patch: Removed.
>+        * gtk/patches/gst-plugins-good-0001-rtpbin-pipeline-gets-an-EOS-when-any-rtpsources-byes.patch: Removed.
>+        * gtk/patches/gst-plugins-good-0002-rtpbin-avoid-generating-errors-when-rtcp-messages-ar.patch: Removed.
>+        * gtk/patches/gst-plugins-good-0003-rtpbin-receive-bundle-support.patch: Removed.
>+        * gtk/patches/gst-plugins-good-0004-qtdemux-add-context-for-a-preferred-protection.patch: Removed.
>+        * gtk/patches/gst-plugins-good-Revert-qtdemux-expose-streams-with-first-moof-for-fr.patch: Removed.
>+        * gtk/patches/gst-plugins-good-use-the-tfdt-decode-time.patch: Removed.
>+        * gtk/patches/gstreamer-0001-protection-added-function-to-filter-system-ids.patch: Removed.
>+
> 2017-05-02  David Kilzer  <ddkilzer@apple.com>
> 
>         check-webkit-style should also keep js-test-post[-async].js in sync
>diff --git a/Tools/gtk/jhbuild.modules b/Tools/gtk/jhbuild.modules
>index 6cd4865d5aa..449d93f6de8 100644
>--- a/Tools/gtk/jhbuild.modules
>+++ b/Tools/gtk/jhbuild.modules
>@@ -353,13 +353,6 @@
>     </dependencies>
>   </autotools>
> 
>-  <autotools id="orc" autogenargs="--disable-gtk-doc" autogen-sh="configure">
>-    <branch module="orc/orc-0.4.17.tar.gz" version="0.4.17"
>-            repo="gstreamer"
>-            hash="sha256:4fc7cca48c59fff23afee78fb642cdbde001f56401c8f47b95a16578d1d5d7e8"
>-            md5sum="af1bf3dab9e69f3c36f389285e2a12a1"/>
>-  </autotools>
>-
>   <autotools id="gstreamer" autogen-sh="configure" autogenargs="--disable-gtk-doc">
>     <if condition-set="macos">
>       <autogenargs value="--disable-introspection"/>
>@@ -367,12 +360,9 @@
>     <dependencies>
>       <dep package="orc"/>
>     </dependencies>
>-    <branch module="gstreamer/gstreamer-${version}.tar.xz" version="1.8.0"
>+    <branch module="gstreamer/gstreamer-${version}.tar.xz" version="1.10.4"
>             repo="gstreamer"
>-            hash="sha256:947a314a212b5d94985d89b43440dbe66b696e12bbdf9a2f78967b98d74abedc"
>-            md5sum="6846d7289ec323c38c49b818171e955a">
>-      <patch file="gstreamer-0001-protection-added-function-to-filter-system-ids.patch" strip="1"/>
>-    </branch>
>+            hash="sha256:50c2f5af50a6cc6c0a3f3ed43bdd8b5e2bff00bacfb766d4be139ec06d8b5218"/>
>   </autotools>
> 
>   <autotools id="gst-plugins-base"
>@@ -384,10 +374,9 @@
>     <dependencies>
>       <dep package="gstreamer"/>
>     </dependencies>
>-    <branch module="gst-plugins-base/gst-plugins-base-${version}.tar.xz" version="1.8.0"
>+    <branch module="gst-plugins-base/gst-plugins-base-${version}.tar.xz" version="1.10.4"
>             repo="gstreamer"
>-            hash="sha256:abc0acc1d15b4b9c97c65cd9689bd6400081853b9980ea428d3c8572dd791522"
>-            md5sum="20cc8231609318310f2a55f64c86cbb4"/>
>+            hash="sha256:f6d245b6b3d4cb733f81ebb021074c525ece83db0c10e932794b339b8d935eb7"/>
>   </autotools>
> 
>   <autotools id="gst-plugins-good" autogen-sh="configure" autogenargs="--disable-examples --disable-soup --disable-gtk-doc">
>@@ -399,17 +388,9 @@
>       <dep package="libvpx"/>
>     </dependencies>
> 
>-    <branch module="gst-plugins-good/gst-plugins-good-${version}.tar.xz" version="1.8.0"
>+    <branch module="gst-plugins-good/gst-plugins-good-${version}.tar.xz" version="1.10.4"
>             repo="gstreamer"
>-            hash="sha256:c20c134d47dbc238d921707a3b66da709c2b4dd89f9d267cec13d1ddf16e9f4d"
>-            md5sum="91ed4649c7c2e43a61f731d144f6f6d0">
>-      <patch file="gst-plugins-good-use-the-tfdt-decode-time.patch" strip="1"/>
>-      <patch file="gst-plugins-good-Revert-qtdemux-expose-streams-with-first-moof-for-fr.patch" strip="1"/>
>-      <patch file="gst-plugins-good-0001-rtpbin-pipeline-gets-an-EOS-when-any-rtpsources-byes.patch" strip="1"/>
>-      <patch file="gst-plugins-good-0002-rtpbin-avoid-generating-errors-when-rtcp-messages-ar.patch" strip="1"/>
>-      <patch file="gst-plugins-good-0003-rtpbin-receive-bundle-support.patch" strip="1"/>
>-      <patch file="gst-plugins-good-0004-qtdemux-add-context-for-a-preferred-protection.patch" strip="1"/>
>-    </branch>
>+            hash="sha256:8a86c61434a8c44665365bd0b3557a040937d1f44bf69caee4e9ea816ce74d7e"/>
>   </autotools>
> 
>   <autotools id="gst-plugins-bad" autogen-sh="configure" autogenargs="--disable-examples --disable-gtk-doc">
>@@ -420,23 +401,18 @@
>       <dep package="gst-plugins-base"/>
>       <dep package="openh264"/>
>     </dependencies>
>-    <branch module="gst-plugins-bad/gst-plugins-bad-${version}.tar.xz" version="1.8.0"
>+    <branch module="gst-plugins-bad/gst-plugins-bad-${version}.tar.xz" version="1.10.4"
>             repo="gstreamer"
>-            hash="sha256:116376dd1085082422e0b21b0ecd3d1cb345c469c58e32463167d4675f4ca90e"
>-            md5sum="1c2d797bb96a81e9ef570c7a0a37203e">
>-      <patch file="gst-plugins-bad-0001-dtls-port-to-OpenSSL-1.1.0.patch" strip="1"/>
>-      <patch file="gst-plugins-bad-0002-dtlscertificate-Fix-error-checking-in-RSA_generate_k.patch" strip="1"/>
>-    </branch>
>+            hash="sha256:23ddae506b3a223b94869a0d3eea3e9a12e847f94d2d0e0b97102ce13ecd6966"/>
>   </autotools>
> 
>   <autotools id="gst-libav" autogen-sh="configure" autogenargs="--with-libav-extra-configure='--disable-yasm' --disable-gtk-doc">
>     <dependencies>
>       <dep package="gst-plugins-base"/>
>     </dependencies>
>-    <branch module="gst-libav/gst-libav-${version}.tar.xz" version="1.8.0"
>+    <branch module="gst-libav/gst-libav-${version}.tar.xz" version="1.10.4"
>             repo="gstreamer"
>-            hash="sha256:5a1ce28876aee93cb4f3d090f0e807915a5d9bc1325e3480dd302b85aeb4291c"
>-            md5sum="361638fa45466c5050bcde6bfe10fa46"/>
>+            hash="sha256:6ca0feca75e3d48315e07f20ec37cf6260ed1e9dde58df355febd5016246268b"/>
>   </autotools>
> 
>   <autotools id="xserver" autogenargs="--disable-xinerama --enable-glx --enable-composite --disable-xorg --disable-dmx --disable-xnest --disable-xquartz --disable-xwin --disable-xephyr --disable-xfake --disable-xfbdev --disable-install-setuid --disable-unit-tests --disable-present --enable-unix-transport --enable-tcp-transport --disable-local-transport --with-xkb-path=/usr/share/X11/xkb --with-xkb-output=/var/lib/xkb --with-xkb-bin-directory=/usr/bin --without-dtrace">
>diff --git a/Tools/gtk/patches/gst-plugins-bad-0001-dtls-port-to-OpenSSL-1.1.0.patch b/Tools/gtk/patches/gst-plugins-bad-0001-dtls-port-to-OpenSSL-1.1.0.patch
>deleted file mode 100644
>index 5d1064ed5e9..00000000000
>--- a/Tools/gtk/patches/gst-plugins-bad-0001-dtls-port-to-OpenSSL-1.1.0.patch
>+++ /dev/null
>@@ -1,236 +0,0 @@
>-From e938933167c494cdca443334f658b02a03c4486b Mon Sep 17 00:00:00 2001
>-From: Daiki Ueno <dueno@redhat.com>
>-Date: Wed, 26 Oct 2016 14:51:01 +0200
>-Subject: [PATCH] dtls: port to OpenSSL 1.1.0
>-
>-Changes are:
>-
>-- Use the wrapper functions to access opaque data types.  To preserve
>-  backward compatibility, define fallback definitions
>-
>-- Remove the use of idiom "pqueue_size(ssl->d1->sent_messages)", since
>-  there is no replacement
>-
>-- Use RSA_generate_key_ex instead of the deprecated RSA_generate_key
>-
>-https://bugzilla.gnome.org/show_bug.cgi?id=773540
>----
>- ext/dtls/gstdtlscertificate.c | 15 ++++++++
>- ext/dtls/gstdtlsconnection.c  | 87 ++++++++++++++++++++++++++++++++++++++-----
>- 2 files changed, 93 insertions(+), 9 deletions(-)
>-
>-diff --git a/ext/dtls/gstdtlscertificate.c b/ext/dtls/gstdtlscertificate.c
>-index 95fbb83..c1c9602 100644
>---- a/ext/dtls/gstdtlscertificate.c
>-+++ b/ext/dtls/gstdtlscertificate.c
>-@@ -199,7 +199,22 @@ init_generated (GstDtlsCertificate * self)
>-     priv->private_key = NULL;
>-     return;
>-   }
>-+
>-+  /* XXX: RSA_generate_key is actually deprecated in 0.9.8 */
>-+#if OPENSSL_VERSION_NUMBER < 0x10100001L
>-   rsa = RSA_generate_key (2048, RSA_F4, NULL, NULL);
>-+#else
>-+  rsa = RSA_new ();
>-+  if (rsa != NULL) {
>-+    BIGNUM *e = BN_new ();
>-+    if (e != NULL && BN_set_word (e, RSA_F4)
>-+        && RSA_generate_key_ex (rsa, 2048, e, NULL)) {
>-+      RSA_free (rsa);
>-+      rsa = NULL;
>-+    }
>-+    BN_free (e);
>-+  }
>-+#endif
>- 
>-   if (!rsa) {
>-     GST_WARNING_OBJECT (self, "failed to generate RSA");
>-diff --git a/ext/dtls/gstdtlsconnection.c b/ext/dtls/gstdtlsconnection.c
>-index 36f6d63..728f5a7 100644
>---- a/ext/dtls/gstdtlsconnection.c
>-+++ b/ext/dtls/gstdtlsconnection.c
>-@@ -42,6 +42,8 @@
>- #include <openssl/err.h>
>- #include <openssl/ssl.h>
>- 
>-+#include <string.h>
>-+
>- GST_DEBUG_CATEGORY_STATIC (gst_dtls_connection_debug);
>- #define GST_CAT_DEFAULT gst_dtls_connection_debug
>- G_DEFINE_TYPE_WITH_CODE (GstDtlsConnection, gst_dtls_connection, G_TYPE_OBJECT,
>-@@ -216,6 +218,38 @@ gst_dtls_connection_finalize (GObject * gobject)
>-   G_OBJECT_CLASS (gst_dtls_connection_parent_class)->finalize (gobject);
>- }
>- 
>-+#if OPENSSL_VERSION_NUMBER < 0x10100001L
>-+static void
>-+BIO_set_data (BIO * bio, void *ptr)
>-+{
>-+  bio->ptr = ptr;
>-+}
>-+
>-+static void *
>-+BIO_get_data (BIO * bio)
>-+{
>-+  return bio->ptr;
>-+}
>-+
>-+static void
>-+BIO_set_shutdown (BIO * bio, int shutdown)
>-+{
>-+  bio->shutdown = shutdown;
>-+}
>-+
>-+static void
>-+BIO_set_init (BIO * bio, int init)
>-+{
>-+  bio->init = init;
>-+}
>-+
>-+static X509 *
>-+X509_STORE_CTX_get0_cert (X509_STORE_CTX * ctx)
>-+{
>-+  return ctx->cert;
>-+}
>-+#endif
>-+
>- static void
>- gst_dtls_connection_set_property (GObject * object, guint prop_id,
>-     const GValue * value, GParamSpec * pspec)
>-@@ -239,7 +273,7 @@ gst_dtls_connection_set_property (GObject * object, guint prop_id,
>-       priv->bio = BIO_new (BIO_s_gst_dtls_connection ());
>-       g_return_if_fail (priv->bio);
>- 
>--      priv->bio->ptr = self;
>-+      BIO_set_data (priv->bio, self);
>-       SSL_set_bio (priv->ssl, priv->bio, priv->bio);
>- 
>-       SSL_set_verify (priv->ssl,
>-@@ -573,6 +607,7 @@ log_state (GstDtlsConnection * self, const gchar * str)
>-   states |= (! !SSL_want_write (priv->ssl) << 20);
>-   states |= (! !SSL_want_read (priv->ssl) << 24);
>- 
>-+#if OPENSSL_VERSION_NUMBER < 0x10100001L
>-   GST_LOG_OBJECT (self, "%s: role=%s buf=(%d,%p:%d/%d) %x|%x %s",
>-       str,
>-       priv->is_client ? "client" : "server",
>-@@ -581,6 +616,15 @@ log_state (GstDtlsConnection * self, const gchar * str)
>-       priv->bio_buffer_offset,
>-       priv->bio_buffer_len,
>-       states, SSL_get_state (priv->ssl), SSL_state_string_long (priv->ssl));
>-+#else
>-+  GST_LOG_OBJECT (self, "%s: role=%s buf=(%p:%d/%d) %x|%x %s",
>-+      str,
>-+      priv->is_client ? "client" : "server",
>-+      priv->bio_buffer,
>-+      priv->bio_buffer_offset,
>-+      priv->bio_buffer_len,
>-+      states, SSL_get_state (priv->ssl), SSL_state_string_long (priv->ssl));
>-+#endif
>- }
>- 
>- static void
>-@@ -737,7 +781,7 @@ openssl_verify_callback (int preverify_ok, X509_STORE_CTX * x509_ctx)
>-   self = SSL_get_ex_data (ssl, connection_ex_index);
>-   g_return_val_if_fail (GST_IS_DTLS_CONNECTION (self), FALSE);
>- 
>--  pem = _gst_dtls_x509_to_pem (x509_ctx->cert);
>-+  pem = _gst_dtls_x509_to_pem (X509_STORE_CTX_get0_cert (x509_ctx));
>- 
>-   if (!pem) {
>-     GST_WARNING_OBJECT (self,
>-@@ -749,7 +793,8 @@ openssl_verify_callback (int preverify_ok, X509_STORE_CTX * x509_ctx)
>-       gint len;
>- 
>-       len =
>--          X509_NAME_print_ex (bio, X509_get_subject_name (x509_ctx->cert), 1,
>-+          X509_NAME_print_ex (bio,
>-+          X509_get_subject_name (X509_STORE_CTX_get0_cert (x509_ctx)), 1,
>-           XN_FLAG_MULTILINE);
>-       BIO_read (bio, buffer, len);
>-       buffer[len] = '\0';
>-@@ -777,6 +822,7 @@ openssl_verify_callback (int preverify_ok, X509_STORE_CTX * x509_ctx)
>-     ########  ####  #######
>- */
>- 
>-+#if OPENSSL_VERSION_NUMBER < 0x10100001L
>- static BIO_METHOD custom_bio_methods = {
>-   BIO_TYPE_BIO,
>-   "stream",
>-@@ -795,11 +841,34 @@ BIO_s_gst_dtls_connection (void)
>- {
>-   return &custom_bio_methods;
>- }
>-+#else
>-+static BIO_METHOD *custom_bio_methods;
>-+
>-+static BIO_METHOD *
>-+BIO_s_gst_dtls_connection (void)
>-+{
>-+  if (custom_bio_methods != NULL)
>-+    return custom_bio_methods;
>-+
>-+  custom_bio_methods = BIO_meth_new (BIO_TYPE_BIO, "stream");
>-+  if (custom_bio_methods == NULL
>-+      || !BIO_meth_set_write (custom_bio_methods, bio_method_write)
>-+      || !BIO_meth_set_read (custom_bio_methods, bio_method_read)
>-+      || !BIO_meth_set_ctrl (custom_bio_methods, bio_method_ctrl)
>-+      || !BIO_meth_set_create (custom_bio_methods, bio_method_new)
>-+      || !BIO_meth_set_destroy (custom_bio_methods, bio_method_free)) {
>-+    BIO_meth_free (custom_bio_methods);
>-+    return NULL;
>-+  }
>-+
>-+  return custom_bio_methods;
>-+}
>-+#endif
>- 
>- static int
>- bio_method_write (BIO * bio, const char *data, int size)
>- {
>--  GstDtlsConnection *self = GST_DTLS_CONNECTION (bio->ptr);
>-+  GstDtlsConnection *self = GST_DTLS_CONNECTION (BIO_get_data (bio));
>- 
>-   GST_LOG_OBJECT (self, "BIO: writing %d", size);
>- 
>-@@ -824,7 +893,7 @@ bio_method_write (BIO * bio, const char *data, int size)
>- static int
>- bio_method_read (BIO * bio, char *out_buffer, int size)
>- {
>--  GstDtlsConnection *self = GST_DTLS_CONNECTION (bio->ptr);
>-+  GstDtlsConnection *self = GST_DTLS_CONNECTION (BIO_get_data (bio));
>-   GstDtlsConnectionPrivate *priv = self->priv;
>-   guint internal_size;
>-   gint copy_size;
>-@@ -868,7 +937,7 @@ bio_method_read (BIO * bio, char *out_buffer, int size)
>- static long
>- bio_method_ctrl (BIO * bio, int cmd, long arg1, void *arg2)
>- {
>--  GstDtlsConnection *self = GST_DTLS_CONNECTION (bio->ptr);
>-+  GstDtlsConnection *self = GST_DTLS_CONNECTION (BIO_get_data (bio));
>-   GstDtlsConnectionPrivate *priv = self->priv;
>- 
>-   switch (cmd) {
>-@@ -916,8 +985,8 @@ bio_method_new (BIO * bio)
>- {
>-   GST_LOG_OBJECT (NULL, "BIO: new");
>- 
>--  bio->shutdown = 0;
>--  bio->init = 1;
>-+  BIO_set_shutdown (bio, 0);
>-+  BIO_set_init (bio, 1);
>- 
>-   return 1;
>- }
>-@@ -930,6 +999,6 @@ bio_method_free (BIO * bio)
>-     return 0;
>-   }
>- 
>--  GST_LOG_OBJECT (GST_DTLS_CONNECTION (bio->ptr), "BIO free");
>-+  GST_LOG_OBJECT (GST_DTLS_CONNECTION (BIO_get_data (bio)), "BIO free");
>-   return 0;
>- }
>--- 
>-2.10.2
>-
>diff --git a/Tools/gtk/patches/gst-plugins-bad-0002-dtlscertificate-Fix-error-checking-in-RSA_generate_k.patch b/Tools/gtk/patches/gst-plugins-bad-0002-dtlscertificate-Fix-error-checking-in-RSA_generate_k.patch
>deleted file mode 100644
>index c66877370c6..00000000000
>--- a/Tools/gtk/patches/gst-plugins-bad-0002-dtlscertificate-Fix-error-checking-in-RSA_generate_k.patch
>+++ /dev/null
>@@ -1,37 +0,0 @@
>-From 3a069193e25364ebdacac86f4b03022c151ea29c Mon Sep 17 00:00:00 2001
>-From: =?UTF-8?q?Sebastian=20Dr=C3=B6ge?= <sebastian@centricular.com>
>-Date: Mon, 14 Nov 2016 11:32:17 +0200
>-Subject: [PATCH] dtlscertificate: Fix error checking in RSA_generate_key_ex()
>- usage
>-
>-Was broken during the port for OpenSSL 1.1.
>-
>-https://bugzilla.gnome.org/show_bug.cgi?id=774328
>----
>- ext/dtls/gstdtlscertificate.c | 7 ++++---
>- 1 file changed, 4 insertions(+), 3 deletions(-)
>-
>-diff --git a/ext/dtls/gstdtlscertificate.c b/ext/dtls/gstdtlscertificate.c
>-index c1c9602..c2d9bb2 100644
>---- a/ext/dtls/gstdtlscertificate.c
>-+++ b/ext/dtls/gstdtlscertificate.c
>-@@ -207,12 +207,13 @@ init_generated (GstDtlsCertificate * self)
>-   rsa = RSA_new ();
>-   if (rsa != NULL) {
>-     BIGNUM *e = BN_new ();
>--    if (e != NULL && BN_set_word (e, RSA_F4)
>--        && RSA_generate_key_ex (rsa, 2048, e, NULL)) {
>-+    if (e == NULL || !BN_set_word (e, RSA_F4)
>-+        || !RSA_generate_key_ex (rsa, 2048, e, NULL)) {
>-       RSA_free (rsa);
>-       rsa = NULL;
>-     }
>--    BN_free (e);
>-+    if (e)
>-+      BN_free (e);
>-   }
>- #endif
>- 
>--- 
>-2.10.2
>-
>diff --git a/Tools/gtk/patches/gst-plugins-good-0001-rtpbin-pipeline-gets-an-EOS-when-any-rtpsources-byes.patch b/Tools/gtk/patches/gst-plugins-good-0001-rtpbin-pipeline-gets-an-EOS-when-any-rtpsources-byes.patch
>deleted file mode 100644
>index f0f90d10457..00000000000
>--- a/Tools/gtk/patches/gst-plugins-good-0001-rtpbin-pipeline-gets-an-EOS-when-any-rtpsources-byes.patch
>+++ /dev/null
>@@ -1,156 +0,0 @@
>-From eeea2a7fe88a17b15318d5b6ae6e190b2f777030 Mon Sep 17 00:00:00 2001
>-From: "Alejandro G. Castro" <alex@igalia.com>
>-Date: Wed, 26 Oct 2016 13:21:29 +0200
>-Subject: [PATCH] rtpbin: pipeline gets an EOS when any rtpsources byes
>-
>-Instead of sending EOS when a source byes we have to wait for
>-all the sources to be gone, which means they already sent BYE and
>-were removed from the session. We now handle the EOS in the rtcp
>-loop checking the amount of sources in the session.
>-
>-https://bugzilla.gnome.org/show_bug.cgi?id=773218
>----
>- gst/rtpmanager/gstrtpsession.c | 29 +++++++++++++++++++----------
>- gst/rtpmanager/rtpsession.c    |  6 +-----
>- gst/rtpmanager/rtpsession.h    |  3 +--
>- 3 files changed, 21 insertions(+), 17 deletions(-)
>-
>-diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c
>-index 3688e85..fd1f2b1 100644
>---- a/gst/rtpmanager/gstrtpsession.c
>-+++ b/gst/rtpmanager/gstrtpsession.c
>-@@ -293,7 +293,7 @@ static GstFlowReturn gst_rtp_session_process_rtp (RTPSession * sess,
>- static GstFlowReturn gst_rtp_session_send_rtp (RTPSession * sess,
>-     RTPSource * src, gpointer data, gpointer user_data);
>- static GstFlowReturn gst_rtp_session_send_rtcp (RTPSession * sess,
>--    RTPSource * src, GstBuffer * buffer, gboolean eos, gpointer user_data);
>-+    RTPSource * src, GstBuffer * buffer, gpointer user_data);
>- static GstFlowReturn gst_rtp_session_sync_rtcp (RTPSession * sess,
>-     GstBuffer * buffer, gpointer user_data);
>- static gint gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload,
>-@@ -1156,6 +1156,22 @@ rtcp_thread (GstRtpSession * rtpsession)
>-     GST_RTP_SESSION_UNLOCK (rtpsession);
>-     rtp_session_on_timeout (session, current_time, ntpnstime, running_time);
>-     GST_RTP_SESSION_LOCK (rtpsession);
>-+
>-+    if (!rtp_session_get_num_sources (session)) {
>-+      /* when no sources left in the session, all of the them have went
>-+       * BYE at some point and removed, we can send EOS to the
>-+       * pipeline. */
>-+      GstPad *rtcp_src = rtpsession->send_rtcp_src;
>-+
>-+      if (rtcp_src) {
>-+        gst_object_ref (rtcp_src);
>-+        GST_LOG_OBJECT (rtpsession, "sending EOS");
>-+        GST_RTP_SESSION_UNLOCK (rtpsession);
>-+        gst_pad_push_event (rtpsession->send_rtcp_src, gst_event_new_eos ());
>-+        GST_RTP_SESSION_LOCK (rtpsession);
>-+        gst_object_unref (rtcp_src);
>-+      }
>-+    }
>-   }
>-   /* mark the thread as stopped now */
>-   rtpsession->priv->thread_stopped = TRUE;
>-@@ -1413,11 +1429,10 @@ do_rtcp_events (GstRtpSession * rtpsession, GstPad * srcpad)
>- }
>- 
>- /* called when the session manager has an RTCP packet ready for further
>-- * sending. The eos flag is set when an EOS event should be sent downstream as
>-- * well. */
>-+ * sending. */
>- static GstFlowReturn
>- gst_rtp_session_send_rtcp (RTPSession * sess, RTPSource * src,
>--    GstBuffer * buffer, gboolean eos, gpointer user_data)
>-+    GstBuffer * buffer, gpointer user_data)
>- {
>-   GstFlowReturn result;
>-   GstRtpSession *rtpsession;
>-@@ -1440,11 +1455,6 @@ gst_rtp_session_send_rtcp (RTPSession * sess, RTPSource * src,
>-     GST_LOG_OBJECT (rtpsession, "sending RTCP");
>-     result = gst_pad_push (rtcp_src, buffer);
>- 
>--    /* we have to send EOS after this packet */
>--    if (eos) {
>--      GST_LOG_OBJECT (rtpsession, "sending EOS");
>--      gst_pad_push_event (rtcp_src, gst_event_new_eos ());
>--    }
>-     gst_object_unref (rtcp_src);
>-   } else {
>-     GST_RTP_SESSION_UNLOCK (rtpsession);
>-@@ -2056,7 +2066,6 @@ gst_rtp_session_event_send_rtcp_src (GstPad * pad, GstObject * parent,
>-   return ret;
>- }
>- 
>--
>- static gboolean
>- gst_rtp_session_event_send_rtp_sink (GstPad * pad, GstObject * parent,
>-     GstEvent * event)
>-diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c
>-index 8b33b6b..aa8b40b 100644
>---- a/gst/rtpmanager/rtpsession.c
>-+++ b/gst/rtpmanager/rtpsession.c
>-@@ -3263,7 +3263,6 @@ early_exit:
>- typedef struct
>- {
>-   RTPSource *source;
>--  gboolean is_bye;
>-   GstBuffer *buffer;
>- } ReportOutput;
>- 
>-@@ -3874,7 +3873,6 @@ static void
>- generate_rtcp (const gchar * key, RTPSource * source, ReportData * data)
>- {
>-   RTPSession *sess = data->sess;
>--  gboolean is_bye = FALSE;
>-   ReportOutput *output;
>- 
>-   /* only generate RTCP for active internal sources */
>-@@ -3893,7 +3891,6 @@ generate_rtcp (const gchar * key, RTPSource * source, ReportData * data)
>-   if (source->marked_bye) {
>-     /* send BYE */
>-     make_source_bye (sess, source, data);
>--    is_bye = TRUE;
>-   } else if (!data->is_early) {
>-     /* loop over all known sources and add report blocks. If we are early, we
>-      * just make a minimal RTCP packet and skip this step */
>-@@ -3918,7 +3915,6 @@ generate_rtcp (const gchar * key, RTPSource * source, ReportData * data)
>- 
>-   output = g_slice_new (ReportOutput);
>-   output->source = g_object_ref (source);
>--  output->is_bye = is_bye;
>-   output->buffer = data->rtcp;
>-   /* queue the RTCP packet to push later */
>-   g_queue_push_tail (&data->output, output);
>-@@ -4098,7 +4094,7 @@ done:
>-       GST_DEBUG ("%p, sending RTCP packet, avg size %u, %u", &sess->stats,
>-           sess->stats.avg_rtcp_packet_size, packet_size);
>-       result =
>--          sess->callbacks.send_rtcp (sess, source, buffer, output->is_bye,
>-+          sess->callbacks.send_rtcp (sess, source, buffer,
>-           sess->send_rtcp_user_data);
>- 
>-       RTP_SESSION_LOCK (sess);
>-diff --git a/gst/rtpmanager/rtpsession.h b/gst/rtpmanager/rtpsession.h
>-index 9fa9327..c25981a 100644
>---- a/gst/rtpmanager/rtpsession.h
>-+++ b/gst/rtpmanager/rtpsession.h
>-@@ -71,7 +71,6 @@ typedef GstFlowReturn (*RTPSessionSendRTP) (RTPSession *sess, RTPSource *src, gp
>-  * @sess: an #RTPSession
>-  * @src: the #RTPSource
>-  * @buffer: the RTCP buffer ready for sending
>-- * @eos: if an EOS event should be pushed
>-  * @user_data: user data specified when registering
>-  *
>-  * This callback will be called when @sess has @buffer ready for sending to
>-@@ -80,7 +79,7 @@ typedef GstFlowReturn (*RTPSessionSendRTP) (RTPSession *sess, RTPSource *src, gp
>-  * Returns: a #GstFlowReturn.
>-  */
>- typedef GstFlowReturn (*RTPSessionSendRTCP) (RTPSession *sess, RTPSource *src, GstBuffer *buffer,
>--    gboolean eos, gpointer user_data);
>-+    gpointer user_data);
>- 
>- /**
>-  * RTPSessionSyncRTCP:
>--- 
>-2.10.2
>-
>diff --git a/Tools/gtk/patches/gst-plugins-good-0002-rtpbin-avoid-generating-errors-when-rtcp-messages-ar.patch b/Tools/gtk/patches/gst-plugins-good-0002-rtpbin-avoid-generating-errors-when-rtcp-messages-ar.patch
>deleted file mode 100644
>index aad048300cf..00000000000
>--- a/Tools/gtk/patches/gst-plugins-good-0002-rtpbin-avoid-generating-errors-when-rtcp-messages-ar.patch
>+++ /dev/null
>@@ -1,61 +0,0 @@
>-From dab6c473c3e1b2b400ee18afc70140c2f842debf Mon Sep 17 00:00:00 2001
>-From: "Alejandro G. Castro" <alex@igalia.com>
>-Date: Thu, 20 Oct 2016 13:14:13 +0200
>-Subject: [PATCH] rtpbin: avoid generating errors when rtcp messages are empty
>- and check the queue is not empty
>-
>-Add a check to verify all the output buffers were empty for the
>-session in a timout and log an error.
>-
>-https://bugzilla.gnome.org/show_bug.cgi?id=773269
>----
>- gst/rtpmanager/rtpsession.c | 12 ++++++++++--
>- 1 file changed, 10 insertions(+), 2 deletions(-)
>-
>-diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c
>-index 75908c0..f1d9210 100644
>---- a/gst/rtpmanager/rtpsession.c
>-+++ b/gst/rtpmanager/rtpsession.c
>-@@ -3923,6 +3923,7 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
>-   ReportData data = { GST_RTCP_BUFFER_INIT };
>-   GHashTable *table_copy;
>-   ReportOutput *output;
>-+  gboolean all_empty = FALSE;
>- 
>-   g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
>- 
>-@@ -3989,6 +3990,9 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
>-   if (!is_rtcp_time (sess, current_time, &data))
>-     goto done;
>- 
>-+  /* check if all the buffers are empty afer generation */
>-+  all_empty = TRUE;
>-+
>-   GST_DEBUG
>-       ("doing RTCP generation %u for %u sources, early %d, may suppress %d",
>-       sess->generation, data.num_to_report, data.is_early, data.may_suppress);
>-@@ -4036,8 +4040,8 @@ done:
>- 
>-     empty_buffer = gst_buffer_get_size (buffer) == 0;
>- 
>--    if (empty_buffer)
>--      g_warning ("rtpsession: Trying to send an empty RTCP packet");
>-+    if (!empty_buffer)
>-+      all_empty = FALSE;
>- 
>-     if (sess->callbacks.send_rtcp &&
>-         !empty_buffer && (do_not_suppress || !data.may_suppress)) {
>-@@ -4068,6 +4072,10 @@ done:
>-     g_object_unref (source);
>-     g_slice_free (ReportOutput, output);
>-   }
>-+
>-+  if (all_empty)
>-+    GST_ERROR ("generated empty RTCP messages for all the sources");
>-+
>-   return result;
>- }
>- 
>--- 
>-2.10.2
>-
>diff --git a/Tools/gtk/patches/gst-plugins-good-0003-rtpbin-receive-bundle-support.patch b/Tools/gtk/patches/gst-plugins-good-0003-rtpbin-receive-bundle-support.patch
>deleted file mode 100644
>index ff89c81998b..00000000000
>--- a/Tools/gtk/patches/gst-plugins-good-0003-rtpbin-receive-bundle-support.patch
>+++ /dev/null
>@@ -1,1018 +0,0 @@
>-From dcd3ce9751cdef0b5ab1fa118355f92bdfe82cb3 Mon Sep 17 00:00:00 2001
>-From: Philippe Normand <philn@igalia.com>
>-Date: Wed, 16 Nov 2016 08:56:34 +0100
>-Subject: [PATCH] rtpbin: receive bundle support
>-
>-A new signal named on-bundled-ssrc is provided and can be
>-used by the application to redirect a stream to a different
>-GstRtpSession or to keep the RTX stream grouped within the
>-GstRtpSession of the same media type.
>-
>-https://bugzilla.gnome.org/show_bug.cgi?id=772740
>----
>- docs/plugins/gst-plugins-good-plugins.signals |   8 +
>- gst/rtpmanager/gstrtpbin.c                    | 562 ++++++++++++++++++--------
>- gst/rtpmanager/gstrtpbin.h                    |   2 +
>- tests/check/Makefile.am                       |   4 +
>- tests/check/elements/.gitignore               |   1 +
>- tests/check/elements/rtpbundle.c              | 390 ++++++++++++++++++
>- tests/check/meson.build                       |   1 +
>- tests/examples/rtp/.gitignore                 |   2 +
>- tests/examples/rtp/Makefile.am                |  10 +-
>- tests/examples/rtp/client-rtpbundle.c         | 266 ++++++++++++
>- tests/examples/rtp/server-rtpbundle.c         | 179 ++++++++
>- 11 files changed, 1265 insertions(+), 160 deletions(-)
>- create mode 100644 tests/check/elements/rtpbundle.c
>- create mode 100644 tests/examples/rtp/client-rtpbundle.c
>- create mode 100644 tests/examples/rtp/server-rtpbundle.c
>-
>-diff --git a/docs/plugins/gst-plugins-good-plugins.signals b/docs/plugins/gst-plugins-good-plugins.signals
>-index 3db17e9..44bbdda 100644
>---- a/docs/plugins/gst-plugins-good-plugins.signals
>-+++ b/docs/plugins/gst-plugins-good-plugins.signals
>-@@ -375,6 +375,14 @@ guint  arg1
>- </SIGNAL>
>- 
>- <SIGNAL>
>-+<NAME>GstRtpBin::on-bundled-ssrc</NAME>
>-+<RETURNS>guint</RETURNS>
>-+<FLAGS>l</FLAGS>
>-+GstRtpBin *gstrtpbin
>-+guint  arg1
>-+</SIGNAL>
>-+
>-+<SIGNAL>
>- <NAME>GstRtpJitterBuffer::clear-pt-map</NAME>
>- <RETURNS>void</RETURNS>
>- <FLAGS>la</FLAGS>
>-diff --git a/gst/rtpmanager/gstrtpbin.c b/gst/rtpmanager/gstrtpbin.c
>-index 648adb9..f58de01 100644
>---- a/gst/rtpmanager/gstrtpbin.c
>-+++ b/gst/rtpmanager/gstrtpbin.c
>-@@ -53,6 +53,13 @@
>-  * SSRC in the RTP packets to its own SSRC and wil forward the packets on the
>-  * send_rtp_src_\%u pad after updating its internal state.
>-  *
>-+ * #GstRtpBin can also demultiplex incoming bundled streams. The first
>-+ * #GstRtpSession will have a #GstRtpSsrcDemux element splitting the streams
>-+ * based on their SSRC and potentially dispatched to a different #GstRtpSession.
>-+ * Because retransmission SSRCs need to be merged with the corresponding media
>-+ * stream the #GstRtpBin::on-bundled-ssrc signal is emitted so that the
>-+ * application can find out to which session the SSRC belongs.
>-+ *
>-  * The session manager needs the clock-rate of the payload types it is handling
>-  * and will signal the #GstRtpSession::request-pt-map signal when it needs such a
>-  * mapping. One can clear the cached values with the #GstRtpSession::clear-pt-map
>-@@ -276,6 +283,8 @@ enum
>-   SIGNAL_ON_NEW_SENDER_SSRC,
>-   SIGNAL_ON_SENDER_SSRC_ACTIVE,
>- 
>-+  SIGNAL_ON_BUNDLED_SSRC,
>-+
>-   LAST_SIGNAL
>- };
>- 
>-@@ -362,6 +371,14 @@ static void remove_send_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session);
>- static void remove_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session);
>- static void free_client (GstRtpBinClient * client, GstRtpBin * bin);
>- static void free_stream (GstRtpBinStream * stream, GstRtpBin * bin);
>-+static GstRtpBinSession *create_session (GstRtpBin * rtpbin, gint id);
>-+static GstPad *complete_session_sink (GstRtpBin * rtpbin,
>-+    GstRtpBinSession * session, gboolean bundle_demuxer_needed);
>-+static void
>-+complete_session_receiver (GstRtpBin * rtpbin, GstRtpBinSession * session,
>-+    guint sessid);
>-+static GstPad *complete_session_rtcp (GstRtpBin * rtpbin,
>-+    GstRtpBinSession * session, guint sessid, gboolean bundle_demuxer_needed);
>- 
>- /* Manages the RTP stream for one SSRC.
>-  *
>-@@ -428,6 +445,12 @@ struct _GstRtpBinSession
>-   gulong demux_newpad_sig;
>-   gulong demux_padremoved_sig;
>- 
>-+  /* Bundling support */
>-+  GstElement *rtp_funnel;
>-+  GstElement *rtcp_funnel;
>-+  GstElement *bundle_demux;
>-+  gulong bundle_demux_newpad_sig;
>-+
>-   GMutex lock;
>- 
>-   /* list of GstRtpBinStream */
>-@@ -629,6 +652,96 @@ ssrc_demux_pad_removed (GstElement * element, guint ssrc, GstPad * pad,
>-   GST_RTP_BIN_UNLOCK (rtpbin);
>- }
>- 
>-+static void
>-+new_bundled_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
>-+    GstRtpBinSession * session)
>-+{
>-+  GValue result = G_VALUE_INIT;
>-+  GValue params[2] = { G_VALUE_INIT, G_VALUE_INIT };
>-+  guint session_id = 0;
>-+  GstRtpBinSession *target_session = NULL;
>-+  GstRtpBin *rtpbin = session->bin;
>-+  gchar *name;
>-+  GstPad *src_pad;
>-+  GstPad *recv_rtp_sink = NULL;
>-+  GstPad *recv_rtcp_sink = NULL;
>-+  GstPadLinkReturn ret;
>-+
>-+  GST_RTP_BIN_DYN_LOCK (rtpbin);
>-+  GST_DEBUG_OBJECT (rtpbin, "new bundled SSRC pad %08x, %s:%s", ssrc,
>-+      GST_DEBUG_PAD_NAME (pad));
>-+
>-+  g_value_init (&result, G_TYPE_UINT);
>-+  g_value_init (&params[0], GST_TYPE_ELEMENT);
>-+  g_value_set_object (&params[0], rtpbin);
>-+  g_value_init (&params[1], G_TYPE_UINT);
>-+  g_value_set_uint (&params[1], ssrc);
>-+
>-+  g_signal_emitv (params,
>-+      gst_rtp_bin_signals[SIGNAL_ON_BUNDLED_SSRC], 0, &result);
>-+  g_value_unset (&params[0]);
>-+
>-+  session_id = g_value_get_uint (&result);
>-+  if (session_id == 0) {
>-+    target_session = session;
>-+  } else {
>-+    target_session = find_session_by_id (rtpbin, (gint) session_id);
>-+    if (!target_session) {
>-+      target_session = create_session (rtpbin, session_id);
>-+    }
>-+    if (!target_session->recv_rtp_sink) {
>-+      recv_rtp_sink = complete_session_sink (rtpbin, target_session, FALSE);
>-+    }
>-+
>-+    if (!target_session->recv_rtp_src)
>-+      complete_session_receiver (rtpbin, target_session, session_id);
>-+
>-+    if (!target_session->recv_rtcp_sink) {
>-+      recv_rtcp_sink =
>-+          complete_session_rtcp (rtpbin, target_session, session_id, FALSE);
>-+    }
>-+  }
>-+
>-+  GST_DEBUG_OBJECT (rtpbin, "Assigning bundled ssrc %u to session %u", ssrc,
>-+      session_id);
>-+
>-+  if (!recv_rtp_sink) {
>-+    recv_rtp_sink =
>-+        gst_element_get_request_pad (target_session->rtp_funnel, "sink_%u");
>-+  }
>-+
>-+  if (!recv_rtcp_sink) {
>-+    recv_rtcp_sink =
>-+        gst_element_get_request_pad (target_session->rtcp_funnel, "sink_%u");
>-+  }
>-+
>-+  name = g_strdup_printf ("src_%u", ssrc);
>-+  src_pad = gst_element_get_static_pad (element, name);
>-+  ret = gst_pad_link (src_pad, recv_rtp_sink);
>-+  g_free (name);
>-+  gst_object_unref (src_pad);
>-+  gst_object_unref (recv_rtp_sink);
>-+  if (ret != GST_PAD_LINK_OK) {
>-+    g_warning
>-+        ("rtpbin: failed to link bundle demuxer to receive rtp funnel for session %u",
>-+        session_id);
>-+  }
>-+
>-+  name = g_strdup_printf ("rtcp_src_%u", ssrc);
>-+  src_pad = gst_element_get_static_pad (element, name);
>-+  gst_pad_link (src_pad, recv_rtcp_sink);
>-+  g_free (name);
>-+  gst_object_unref (src_pad);
>-+  gst_object_unref (recv_rtcp_sink);
>-+  if (ret != GST_PAD_LINK_OK) {
>-+    g_warning
>-+        ("rtpbin: failed to link bundle demuxer to receive rtcp sink pad for session %u",
>-+        session_id);
>-+  }
>-+
>-+  GST_RTP_BIN_DYN_UNLOCK (rtpbin);
>-+}
>-+
>- /* create a session with the given id.  Must be called with RTP_BIN_LOCK */
>- static GstRtpBinSession *
>- create_session (GstRtpBin * rtpbin, gint id)
>-@@ -649,6 +762,10 @@ create_session (GstRtpBin * rtpbin, gint id)
>-   sess->bin = rtpbin;
>-   sess->session = session;
>-   sess->demux = demux;
>-+
>-+  sess->rtp_funnel = gst_element_factory_make ("funnel", NULL);
>-+  sess->rtcp_funnel = gst_element_factory_make ("funnel", NULL);
>-+
>-   sess->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
>-       (GDestroyNotify) gst_caps_unref);
>-   rtpbin->sessions = g_slist_prepend (rtpbin->sessions, sess);
>-@@ -696,6 +813,8 @@ create_session (GstRtpBin * rtpbin, gint id)
>- 
>-   gst_bin_add (GST_BIN_CAST (rtpbin), session);
>-   gst_bin_add (GST_BIN_CAST (rtpbin), demux);
>-+  gst_bin_add (GST_BIN_CAST (rtpbin), sess->rtp_funnel);
>-+  gst_bin_add (GST_BIN_CAST (rtpbin), sess->rtcp_funnel);
>- 
>-   GST_OBJECT_LOCK (rtpbin);
>-   target = GST_STATE_TARGET (rtpbin);
>-@@ -704,6 +823,8 @@ create_session (GstRtpBin * rtpbin, gint id)
>-   /* change state only to what's needed */
>-   gst_element_set_state (demux, target);
>-   gst_element_set_state (session, target);
>-+  gst_element_set_state (sess->rtp_funnel, target);
>-+  gst_element_set_state (sess->rtcp_funnel, target);
>- 
>-   return sess;
>- 
>-@@ -807,7 +928,7 @@ get_pt_map (GstRtpBinSession * session, guint pt)
>-   GValue ret = { 0 };
>-   GValue args[3] = { {0}, {0}, {0} };
>- 
>--  GST_DEBUG ("searching pt %d in cache", pt);
>-+  GST_DEBUG ("searching pt %u in cache", pt);
>- 
>-   GST_RTP_SESSION_LOCK (session);
>- 
>-@@ -820,7 +941,7 @@ get_pt_map (GstRtpBinSession * session, guint pt)
>- 
>-   bin = session->bin;
>- 
>--  GST_DEBUG ("emiting signal for pt %d in session %d", pt, session->id);
>-+  GST_DEBUG ("emiting signal for pt %u in session %u", pt, session->id);
>- 
>-   /* not in cache, send signal to request caps */
>-   g_value_init (&args[0], GST_TYPE_ELEMENT);
>-@@ -856,7 +977,7 @@ get_pt_map (GstRtpBinSession * session, guint pt)
>-   if (!caps)
>-     goto no_caps;
>- 
>--  GST_DEBUG ("caching pt %d as %" GST_PTR_FORMAT, pt, caps);
>-+  GST_DEBUG ("caching pt %u as %" GST_PTR_FORMAT, pt, caps);
>- 
>-   /* store in cache, take additional ref */
>-   g_hash_table_insert (session->ptmap, GINT_TO_POINTER (pt),
>-@@ -947,7 +1068,7 @@ gst_rtp_bin_get_session (GstRtpBin * bin, guint session_id)
>-   GstElement *ret = NULL;
>- 
>-   GST_RTP_BIN_LOCK (bin);
>--  GST_DEBUG_OBJECT (bin, "retrieving GstRtpSession, index: %d", session_id);
>-+  GST_DEBUG_OBJECT (bin, "retrieving GstRtpSession, index: %u", session_id);
>-   session = find_session_by_id (bin, (gint) session_id);
>-   if (session) {
>-     ret = gst_object_ref (session->session);
>-@@ -964,7 +1085,7 @@ gst_rtp_bin_get_internal_session (GstRtpBin * bin, guint session_id)
>-   GstRtpBinSession *session;
>- 
>-   GST_RTP_BIN_LOCK (bin);
>--  GST_DEBUG_OBJECT (bin, "retrieving internal RTPSession object, index: %d",
>-+  GST_DEBUG_OBJECT (bin, "retrieving internal RTPSession object, index: %u",
>-       session_id);
>-   session = find_session_by_id (bin, (gint) session_id);
>-   if (session) {
>-@@ -2194,6 +2315,29 @@ gst_rtp_bin_class_init (GstRtpBinClass * klass)
>-           on_sender_ssrc_active), NULL, NULL, g_cclosure_marshal_generic,
>-       G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT);
>- 
>-+
>-+  /**
>-+   * GstRtpBin::on-bundled-ssrc:
>-+   * @rtpbin: the object which received the signal
>-+   * @ssrc: the bundled SSRC
>-+   *
>-+   * Notify of a new incoming bundled SSRC. If no handler is connected to the
>-+   * signal then the #GstRtpSession created for the recv_rtp_sink_\%u
>-+   * request pad will be managing this new SSRC. However if there is a handler
>-+   * connected then the application can decided to dispatch this new stream to
>-+   * another session by providing its ID as return value of the handler. This
>-+   * can be particularly useful to keep retransmission SSRCs grouped with the
>-+   * session for which they handle retransmission.
>-+   *
>-+   * Since: 1.12
>-+   */
>-+  gst_rtp_bin_signals[SIGNAL_ON_BUNDLED_SSRC] =
>-+      g_signal_new ("on-bundled-ssrc", G_TYPE_FROM_CLASS (klass),
>-+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
>-+          on_bundled_ssrc), NULL, NULL,
>-+      g_cclosure_marshal_generic, G_TYPE_UINT, 1, G_TYPE_UINT);
>-+
>-+
>-   g_object_class_install_property (gobject_class, PROP_SDES,
>-       g_param_spec_boxed ("sdes", "SDES",
>-           "The SDES items of this session",
>-@@ -3021,7 +3165,7 @@ new_payload_found (GstElement * element, guint pt, GstPad * pad,
>- 
>-   rtpbin = stream->bin;
>- 
>--  GST_DEBUG ("new payload pad %d", pt);
>-+  GST_DEBUG_OBJECT (rtpbin, "new payload pad %u", pt);
>- 
>-   GST_RTP_BIN_SHUTDOWN_LOCK (rtpbin, shutdown);
>- 
>-@@ -3078,7 +3222,7 @@ pt_map_requested (GstElement * element, guint pt, GstRtpBinSession * session)
>- 
>-   rtpbin = session->bin;
>- 
>--  GST_DEBUG_OBJECT (rtpbin, "payload map requested for pt %d in session %d", pt,
>-+  GST_DEBUG_OBJECT (rtpbin, "payload map requested for pt %u in session %u", pt,
>-       session->id);
>- 
>-   caps = get_pt_map (session, pt);
>-@@ -3099,7 +3243,7 @@ static void
>- payload_type_change (GstElement * element, guint pt, GstRtpBinSession * session)
>- {
>-   GST_DEBUG_OBJECT (session->bin,
>--      "emiting signal for pt type changed to %d in session %d", pt,
>-+      "emiting signal for pt type changed to %u in session %u", pt,
>-       session->id);
>- 
>-   g_signal_emit (session->bin, gst_rtp_bin_signals[SIGNAL_PAYLOAD_TYPE_CHANGE],
>-@@ -3246,15 +3390,42 @@ no_stream:
>-   }
>- }
>- 
>--static gboolean
>--complete_session_sink (GstRtpBin * rtpbin, GstRtpBinSession * session)
>-+static void
>-+session_maybe_create_bundle_demuxer (GstRtpBinSession * session)
>-+{
>-+  GstRtpBin *rtpbin;
>-+
>-+  if (session->bundle_demux)
>-+    return;
>-+
>-+  rtpbin = session->bin;
>-+  if (g_signal_has_handler_pending (rtpbin,
>-+          gst_rtp_bin_signals[SIGNAL_ON_BUNDLED_SSRC], 0, TRUE)) {
>-+    GST_DEBUG_OBJECT (rtpbin, "Adding a bundle SSRC demuxer to session %u",
>-+        session->id);
>-+    session->bundle_demux = gst_element_factory_make ("rtpssrcdemux", NULL);
>-+    session->bundle_demux_newpad_sig = g_signal_connect (session->bundle_demux,
>-+        "new-ssrc-pad", (GCallback) new_bundled_ssrc_pad_found, session);
>-+
>-+    gst_bin_add (GST_BIN_CAST (rtpbin), session->bundle_demux);
>-+    gst_element_sync_state_with_parent (session->bundle_demux);
>-+  } else {
>-+    GST_DEBUG_OBJECT (rtpbin,
>-+        "No handler for the on-bundled-ssrc signal so no need for a bundle SSRC demuxer in session %u",
>-+        session->id);
>-+  }
>-+}
>-+
>-+static GstPad *
>-+complete_session_sink (GstRtpBin * rtpbin, GstRtpBinSession * session,
>-+    gboolean bundle_demuxer_needed)
>- {
>--  gchar *gname;
>-   guint sessid = session->id;
>-   GstPad *recv_rtp_sink;
>-+  GstPad *funnel_src;
>-   GstElement *decoder;
>--  GstElementClass *klass;
>--  GstPadTemplate *templ;
>-+
>-+  g_assert (!session->recv_rtp_sink);
>- 
>-   /* get recv_rtp pad and store */
>-   session->recv_rtp_sink =
>-@@ -3265,6 +3436,9 @@ complete_session_sink (GstRtpBin * rtpbin, GstRtpBinSession * session)
>-   g_signal_connect (session->recv_rtp_sink, "notify::caps",
>-       (GCallback) caps_changed, session);
>- 
>-+  if (bundle_demuxer_needed)
>-+    session_maybe_create_bundle_demuxer (session);
>-+
>-   GST_DEBUG_OBJECT (rtpbin, "requesting RTP decoder");
>-   decoder = session_request_element (session, SIGNAL_REQUEST_RTP_DECODER);
>-   if (decoder) {
>-@@ -3282,7 +3456,14 @@ complete_session_sink (GstRtpBin * rtpbin, GstRtpBinSession * session)
>-     if (decsrc == NULL)
>-       goto dec_src_failed;
>- 
>--    ret = gst_pad_link (decsrc, session->recv_rtp_sink);
>-+    if (session->bundle_demux) {
>-+      GstPad *demux_sink;
>-+      demux_sink = gst_element_get_static_pad (session->bundle_demux, "sink");
>-+      ret = gst_pad_link (decsrc, demux_sink);
>-+      gst_object_unref (demux_sink);
>-+    } else {
>-+      ret = gst_pad_link (decsrc, session->recv_rtp_sink);
>-+    }
>-     gst_object_unref (decsrc);
>- 
>-     if (ret != GST_PAD_LINK_OK)
>-@@ -3290,81 +3471,54 @@ complete_session_sink (GstRtpBin * rtpbin, GstRtpBinSession * session)
>- 
>-   } else {
>-     GST_DEBUG_OBJECT (rtpbin, "no RTP decoder given");
>--    recv_rtp_sink = gst_object_ref (session->recv_rtp_sink);
>-+    if (session->bundle_demux) {
>-+      recv_rtp_sink =
>-+          gst_element_get_static_pad (session->bundle_demux, "sink");
>-+    } else {
>-+      recv_rtp_sink =
>-+          gst_element_get_request_pad (session->rtp_funnel, "sink_%u");
>-+    }
>-   }
>- 
>--  GST_DEBUG_OBJECT (rtpbin, "ghosting session sink pad");
>--  klass = GST_ELEMENT_GET_CLASS (rtpbin);
>--  gname = g_strdup_printf ("recv_rtp_sink_%u", sessid);
>--  templ = gst_element_class_get_pad_template (klass, "recv_rtp_sink_%u");
>--  session->recv_rtp_sink_ghost =
>--      gst_ghost_pad_new_from_template (gname, recv_rtp_sink, templ);
>--  gst_object_unref (recv_rtp_sink);
>--  gst_pad_set_active (session->recv_rtp_sink_ghost, TRUE);
>--  gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->recv_rtp_sink_ghost);
>--  g_free (gname);
>-+  funnel_src = gst_element_get_static_pad (session->rtp_funnel, "src");
>-+  gst_pad_link (funnel_src, session->recv_rtp_sink);
>-+  gst_object_unref (funnel_src);
>- 
>--  return TRUE;
>-+  return recv_rtp_sink;
>- 
>-   /* ERRORS */
>- pad_failed:
>-   {
>-     g_warning ("rtpbin: failed to get session recv_rtp_sink pad");
>--    return FALSE;
>-+    return NULL;
>-   }
>- dec_sink_failed:
>-   {
>--    g_warning ("rtpbin: failed to get decoder sink pad for session %d", sessid);
>--    return FALSE;
>-+    g_warning ("rtpbin: failed to get decoder sink pad for session %u", sessid);
>-+    return NULL;
>-   }
>- dec_src_failed:
>-   {
>--    g_warning ("rtpbin: failed to get decoder src pad for session %d", sessid);
>-+    g_warning ("rtpbin: failed to get decoder src pad for session %u", sessid);
>-     gst_object_unref (recv_rtp_sink);
>--    return FALSE;
>-+    return NULL;
>-   }
>- dec_link_failed:
>-   {
>--    g_warning ("rtpbin: failed to link rtp decoder for session %d", sessid);
>-+    g_warning ("rtpbin: failed to link rtp decoder for session %u", sessid);
>-     gst_object_unref (recv_rtp_sink);
>--    return FALSE;
>-+    return NULL;
>-   }
>- }
>- 
>--/* Create a pad for receiving RTP for the session in @name. Must be called with
>-- * RTP_BIN_LOCK.
>-- */
>--static GstPad *
>--create_recv_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
>-+static void
>-+complete_session_receiver (GstRtpBin * rtpbin, GstRtpBinSession * session,
>-+    guint sessid)
>- {
>--  guint sessid;
>-   GstElement *aux;
>-   GstPad *recv_rtp_src;
>--  GstRtpBinSession *session;
>--
>--  /* first get the session number */
>--  if (name == NULL || sscanf (name, "recv_rtp_sink_%u", &sessid) != 1)
>--    goto no_name;
>--
>--  GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid);
>- 
>--  /* get or create session */
>--  session = find_session_by_id (rtpbin, sessid);
>--  if (!session) {
>--    GST_DEBUG_OBJECT (rtpbin, "creating session %d", sessid);
>--    /* create session now */
>--    session = create_session (rtpbin, sessid);
>--    if (session == NULL)
>--      goto create_error;
>--  }
>--
>--  /* check if pad was requested */
>--  if (session->recv_rtp_sink_ghost != NULL)
>--    return session->recv_rtp_sink_ghost;
>--
>--  /* setup the session sink pad */
>--  if (!complete_session_sink (rtpbin, session))
>--    goto session_sink_failed;
>-+  g_assert (!session->recv_rtp_src);
>- 
>-   session->recv_rtp_src =
>-       gst_element_get_static_pad (session->session, "recv_rtp_src");
>-@@ -3381,7 +3535,7 @@ create_recv_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
>- 
>-     GST_DEBUG_OBJECT (rtpbin, "linking AUX receiver");
>- 
>--    pname = g_strdup_printf ("sink_%d", sessid);
>-+    pname = g_strdup_printf ("sink_%u", sessid);
>-     auxsink = gst_element_get_static_pad (aux, pname);
>-     g_free (pname);
>-     if (auxsink == NULL)
>-@@ -3394,7 +3548,7 @@ create_recv_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
>- 
>-     /* this can be NULL when this AUX element is not to be linked to
>-      * an SSRC demuxer */
>--    pname = g_strdup_printf ("src_%d", sessid);
>-+    pname = g_strdup_printf ("src_%u", sessid);
>-     recv_rtp_src = gst_element_get_static_pad (aux, pname);
>-     g_free (pname);
>-   } else {
>-@@ -3408,8 +3562,8 @@ create_recv_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
>-     sinkdpad = gst_element_get_static_pad (session->demux, "sink");
>-     GST_DEBUG_OBJECT (rtpbin, "linking demuxer RTP sink pad");
>-     gst_pad_link_full (recv_rtp_src, sinkdpad, GST_PAD_LINK_CHECK_NOTHING);
>--    gst_object_unref (recv_rtp_src);
>-     gst_object_unref (sinkdpad);
>-+    gst_object_unref (recv_rtp_src);
>- 
>-     /* connect to the new-ssrc-pad signal of the SSRC demuxer */
>-     session->demux_newpad_sig = g_signal_connect (session->demux,
>-@@ -3417,6 +3571,71 @@ create_recv_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
>-     session->demux_padremoved_sig = g_signal_connect (session->demux,
>-         "removed-ssrc-pad", (GCallback) ssrc_demux_pad_removed, session);
>-   }
>-+
>-+  return;
>-+
>-+pad_failed:
>-+  {
>-+    g_warning ("rtpbin: failed to get session recv_rtp_src pad");
>-+    return;
>-+  }
>-+aux_sink_failed:
>-+  {
>-+    g_warning ("rtpbin: failed to get AUX sink pad for session %u", sessid);
>-+    return;
>-+  }
>-+aux_link_failed:
>-+  {
>-+    g_warning ("rtpbin: failed to link AUX pad to session %u", sessid);
>-+    return;
>-+  }
>-+}
>-+
>-+/* Create a pad for receiving RTP for the session in @name. Must be called with
>-+ * RTP_BIN_LOCK.
>-+ */
>-+static GstPad *
>-+create_recv_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
>-+{
>-+  guint sessid;
>-+  GstRtpBinSession *session;
>-+  GstPad *recv_rtp_sink;
>-+
>-+  /* first get the session number */
>-+  if (name == NULL || sscanf (name, "recv_rtp_sink_%u", &sessid) != 1)
>-+    goto no_name;
>-+
>-+  GST_DEBUG_OBJECT (rtpbin, "finding session %u", sessid);
>-+
>-+  /* get or create session */
>-+  session = find_session_by_id (rtpbin, sessid);
>-+  if (!session) {
>-+    GST_DEBUG_OBJECT (rtpbin, "creating session %u", sessid);
>-+    /* create session now */
>-+    session = create_session (rtpbin, sessid);
>-+    if (session == NULL)
>-+      goto create_error;
>-+  }
>-+
>-+  /* check if pad was requested */
>-+  if (session->recv_rtp_sink_ghost != NULL)
>-+    return session->recv_rtp_sink_ghost;
>-+
>-+  /* setup the session sink pad */
>-+  recv_rtp_sink = complete_session_sink (rtpbin, session, TRUE);
>-+  if (!recv_rtp_sink)
>-+    goto session_sink_failed;
>-+
>-+
>-+  GST_DEBUG_OBJECT (rtpbin, "ghosting session sink pad");
>-+  session->recv_rtp_sink_ghost =
>-+      gst_ghost_pad_new_from_template (name, recv_rtp_sink, templ);
>-+  gst_object_unref (recv_rtp_sink);
>-+  gst_pad_set_active (session->recv_rtp_sink_ghost, TRUE);
>-+  gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->recv_rtp_sink_ghost);
>-+
>-+  complete_session_receiver (rtpbin, session, sessid);
>-+
>-   return session->recv_rtp_sink_ghost;
>- 
>-   /* ERRORS */
>-@@ -3435,21 +3654,6 @@ session_sink_failed:
>-     /* warning already done */
>-     return NULL;
>-   }
>--pad_failed:
>--  {
>--    g_warning ("rtpbin: failed to get session recv_rtp_src pad");
>--    return NULL;
>--  }
>--aux_sink_failed:
>--  {
>--    g_warning ("rtpbin: failed to get AUX sink pad for session %d", sessid);
>--    return NULL;
>--  }
>--aux_link_failed:
>--  {
>--    g_warning ("rtpbin: failed to link AUX pad to session %d", sessid);
>--    return NULL;
>--  }
>- }
>- 
>- static void
>-@@ -3463,6 +3667,11 @@ remove_recv_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session)
>-     g_signal_handler_disconnect (session->demux, session->demux_padremoved_sig);
>-     session->demux_padremoved_sig = 0;
>-   }
>-+  if (session->bundle_demux_newpad_sig) {
>-+    g_signal_handler_disconnect (session->bundle_demux,
>-+        session->bundle_demux_newpad_sig);
>-+    session->bundle_demux_newpad_sig = 0;
>-+  }
>-   if (session->recv_rtp_src) {
>-     gst_object_unref (session->recv_rtp_src);
>-     session->recv_rtp_src = NULL;
>-@@ -3480,37 +3689,14 @@ remove_recv_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session)
>-   }
>- }
>- 
>--/* Create a pad for receiving RTCP for the session in @name. Must be called with
>-- * RTP_BIN_LOCK.
>-- */
>- static GstPad *
>--create_recv_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
>--    const gchar * name)
>-+complete_session_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session,
>-+    guint sessid, gboolean bundle_demuxer_needed)
>- {
>--  guint sessid;
>-   GstElement *decoder;
>--  GstRtpBinSession *session;
>--  GstPad *sinkdpad, *decsink;
>--
>--  /* first get the session number */
>--  if (name == NULL || sscanf (name, "recv_rtcp_sink_%u", &sessid) != 1)
>--    goto no_name;
>--
>--  GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid);
>--
>--  /* get or create the session */
>--  session = find_session_by_id (rtpbin, sessid);
>--  if (!session) {
>--    GST_DEBUG_OBJECT (rtpbin, "creating session %d", sessid);
>--    /* create session now */
>--    session = create_session (rtpbin, sessid);
>--    if (session == NULL)
>--      goto create_error;
>--  }
>--
>--  /* check if pad was requested */
>--  if (session->recv_rtcp_sink_ghost != NULL)
>--    return session->recv_rtcp_sink_ghost;
>-+  GstPad *sinkdpad;
>-+  GstPad *decsink = NULL;
>-+  GstPad *funnel_src;
>- 
>-   /* get recv_rtp pad and store */
>-   GST_DEBUG_OBJECT (rtpbin, "getting RTCP sink pad");
>-@@ -3519,6 +3705,9 @@ create_recv_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
>-   if (session->recv_rtcp_sink == NULL)
>-     goto pad_failed;
>- 
>-+  if (bundle_demuxer_needed)
>-+    session_maybe_create_bundle_demuxer (session);
>-+
>-   GST_DEBUG_OBJECT (rtpbin, "getting RTCP decoder");
>-   decoder = session_request_element (session, SIGNAL_REQUEST_RTCP_DECODER);
>-   if (decoder) {
>-@@ -3535,14 +3724,26 @@ create_recv_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
>-     if (decsrc == NULL)
>-       goto dec_src_failed;
>- 
>--    ret = gst_pad_link (decsrc, session->recv_rtcp_sink);
>-+    if (session->bundle_demux) {
>-+      GstPad *demux_sink;
>-+      demux_sink =
>-+          gst_element_get_static_pad (session->bundle_demux, "rtcp_sink");
>-+      ret = gst_pad_link (decsrc, demux_sink);
>-+      gst_object_unref (demux_sink);
>-+    } else {
>-+      ret = gst_pad_link (decsrc, session->recv_rtcp_sink);
>-+    }
>-     gst_object_unref (decsrc);
>- 
>-     if (ret != GST_PAD_LINK_OK)
>-       goto dec_link_failed;
>-   } else {
>-     GST_DEBUG_OBJECT (rtpbin, "no RTCP decoder given");
>--    decsink = gst_object_ref (session->recv_rtcp_sink);
>-+    if (session->bundle_demux) {
>-+      decsink = gst_element_get_static_pad (session->bundle_demux, "rtcp_sink");
>-+    } else {
>-+      decsink = gst_element_get_request_pad (session->rtcp_funnel, "sink_%u");
>-+    }
>-   }
>- 
>-   /* get srcpad, link to SSRCDemux */
>-@@ -3556,26 +3757,12 @@ create_recv_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
>-   gst_pad_link_full (session->sync_src, sinkdpad, GST_PAD_LINK_CHECK_NOTHING);
>-   gst_object_unref (sinkdpad);
>- 
>--  session->recv_rtcp_sink_ghost =
>--      gst_ghost_pad_new_from_template (name, decsink, templ);
>--  gst_object_unref (decsink);
>--  gst_pad_set_active (session->recv_rtcp_sink_ghost, TRUE);
>--  gst_element_add_pad (GST_ELEMENT_CAST (rtpbin),
>--      session->recv_rtcp_sink_ghost);
>-+  funnel_src = gst_element_get_static_pad (session->rtcp_funnel, "src");
>-+  gst_pad_link (funnel_src, session->recv_rtcp_sink);
>-+  gst_object_unref (funnel_src);
>- 
>--  return session->recv_rtcp_sink_ghost;
>-+  return decsink;
>- 
>--  /* ERRORS */
>--no_name:
>--  {
>--    g_warning ("rtpbin: invalid name given");
>--    return NULL;
>--  }
>--create_error:
>--  {
>--    /* create_session already warned */
>--    return NULL;
>--  }
>- pad_failed:
>-   {
>-     g_warning ("rtpbin: failed to get session rtcp_sink pad");
>-@@ -3583,25 +3770,82 @@ pad_failed:
>-   }
>- dec_sink_failed:
>-   {
>--    g_warning ("rtpbin: failed to get decoder sink pad for session %d", sessid);
>-+    g_warning ("rtpbin: failed to get decoder sink pad for session %u", sessid);
>-     return NULL;
>-   }
>- dec_src_failed:
>-   {
>--    g_warning ("rtpbin: failed to get decoder src pad for session %d", sessid);
>--    gst_object_unref (decsink);
>--    return NULL;
>-+    g_warning ("rtpbin: failed to get decoder src pad for session %u", sessid);
>-+    goto cleanup;
>-   }
>- dec_link_failed:
>-   {
>--    g_warning ("rtpbin: failed to link rtcp decoder for session %d", sessid);
>--    gst_object_unref (decsink);
>--    return NULL;
>-+    g_warning ("rtpbin: failed to link rtcp decoder for session %u", sessid);
>-+    goto cleanup;
>-   }
>- src_pad_failed:
>-   {
>-     g_warning ("rtpbin: failed to get session sync_src pad");
>--    gst_object_unref (decsink);
>-+  }
>-+
>-+cleanup:
>-+  gst_object_unref (decsink);
>-+  return NULL;
>-+}
>-+
>-+/* Create a pad for receiving RTCP for the session in @name. Must be called with
>-+ * RTP_BIN_LOCK.
>-+ */
>-+static GstPad *
>-+create_recv_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
>-+    const gchar * name)
>-+{
>-+  guint sessid;
>-+  GstRtpBinSession *session;
>-+  GstPad *decsink = NULL;
>-+
>-+  /* first get the session number */
>-+  if (name == NULL || sscanf (name, "recv_rtcp_sink_%u", &sessid) != 1)
>-+    goto no_name;
>-+
>-+  GST_DEBUG_OBJECT (rtpbin, "finding session %u", sessid);
>-+
>-+  /* get or create the session */
>-+  session = find_session_by_id (rtpbin, sessid);
>-+  if (!session) {
>-+    GST_DEBUG_OBJECT (rtpbin, "creating session %u", sessid);
>-+    /* create session now */
>-+    session = create_session (rtpbin, sessid);
>-+    if (session == NULL)
>-+      goto create_error;
>-+  }
>-+
>-+  /* check if pad was requested */
>-+  if (session->recv_rtcp_sink_ghost != NULL)
>-+    return session->recv_rtcp_sink_ghost;
>-+
>-+  decsink = complete_session_rtcp (rtpbin, session, sessid, TRUE);
>-+  if (!decsink)
>-+    goto create_error;
>-+
>-+  session->recv_rtcp_sink_ghost =
>-+      gst_ghost_pad_new_from_template (name, decsink, templ);
>-+  gst_object_unref (decsink);
>-+  gst_pad_set_active (session->recv_rtcp_sink_ghost, TRUE);
>-+  gst_element_add_pad (GST_ELEMENT_CAST (rtpbin),
>-+      session->recv_rtcp_sink_ghost);
>-+
>-+  return session->recv_rtcp_sink_ghost;
>-+
>-+  /* ERRORS */
>-+no_name:
>-+  {
>-+    g_warning ("rtpbin: invalid name given");
>-+    return NULL;
>-+  }
>-+create_error:
>-+  {
>-+    /* create_session already warned */
>-     return NULL;
>-   }
>- }
>-@@ -3651,7 +3895,7 @@ complete_session_src (GstRtpBin * rtpbin, GstRtpBinSession * session)
>-     GstPadLinkReturn ret;
>- 
>-     GST_DEBUG_OBJECT (rtpbin, "linking RTP encoder");
>--    ename = g_strdup_printf ("rtp_src_%d", sessid);
>-+    ename = g_strdup_printf ("rtp_src_%u", sessid);
>-     encsrc = gst_element_get_static_pad (encoder, ename);
>-     g_free (ename);
>- 
>-@@ -3660,7 +3904,7 @@ complete_session_src (GstRtpBin * rtpbin, GstRtpBinSession * session)
>- 
>-     send_rtp_src = encsrc;
>- 
>--    ename = g_strdup_printf ("rtp_sink_%d", sessid);
>-+    ename = g_strdup_printf ("rtp_sink_%u", sessid);
>-     encsink = gst_element_get_static_pad (encoder, ename);
>-     g_free (ename);
>-     if (encsink == NULL)
>-@@ -3694,23 +3938,23 @@ complete_session_src (GstRtpBin * rtpbin, GstRtpBinSession * session)
>-   /* ERRORS */
>- no_srcpad:
>-   {
>--    g_warning ("rtpbin: failed to get rtp source pad for session %d", sessid);
>-+    g_warning ("rtpbin: failed to get rtp source pad for session %u", sessid);
>-     return FALSE;
>-   }
>- enc_src_failed:
>-   {
>--    g_warning ("rtpbin: failed to get encoder src pad for session %d", sessid);
>-+    g_warning ("rtpbin: failed to get encoder src pad for session %u", sessid);
>-     return FALSE;
>-   }
>- enc_sink_failed:
>-   {
>--    g_warning ("rtpbin: failed to get encoder sink pad for session %d", sessid);
>-+    g_warning ("rtpbin: failed to get encoder sink pad for session %u", sessid);
>-     gst_object_unref (send_rtp_src);
>-     return FALSE;
>-   }
>- enc_link_failed:
>-   {
>--    g_warning ("rtpbin: failed to link rtp encoder for session %d", sessid);
>-+    g_warning ("rtpbin: failed to link rtp encoder for session %u", sessid);
>-     gst_object_unref (send_rtp_src);
>-     return FALSE;
>-   }
>-@@ -3772,22 +4016,22 @@ create_error:
>-   }
>- existing_session:
>-   {
>--    g_warning ("rtpbin: session %d is already a sender", sessid);
>-+    g_warning ("rtpbin: session %u is already a sender", sessid);
>-     return FALSE;
>-   }
>- pad_failed:
>-   {
>--    g_warning ("rtpbin: failed to get session pad for session %d", sessid);
>-+    g_warning ("rtpbin: failed to get session pad for session %u", sessid);
>-     return FALSE;
>-   }
>- aux_link_failed:
>-   {
>--    g_warning ("rtpbin: failed to link AUX for session %d", sessid);
>-+    g_warning ("rtpbin: failed to link AUX for session %u", sessid);
>-     return FALSE;
>-   }
>- session_src_failed:
>-   {
>--    g_warning ("rtpbin: failed to complete AUX for session %d", sessid);
>-+    g_warning ("rtpbin: failed to complete AUX for session %u", sessid);
>-     return FALSE;
>-   }
>- }
>-@@ -3847,7 +4091,7 @@ create_send_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
>-     if (!setup_aux_sender (rtpbin, session, aux))
>-       goto aux_session_failed;
>- 
>--    pname = g_strdup_printf ("sink_%d", sessid);
>-+    pname = g_strdup_printf ("sink_%u", sessid);
>-     send_rtp_sink = gst_element_get_static_pad (aux, pname);
>-     g_free (pname);
>- 
>-@@ -3887,27 +4131,27 @@ create_error:
>-   }
>- existing_session:
>-   {
>--    g_warning ("rtpbin: session %d is already in use", sessid);
>-+    g_warning ("rtpbin: session %u is already in use", sessid);
>-     return NULL;
>-   }
>- aux_session_failed:
>-   {
>--    g_warning ("rtpbin: failed to get AUX sink pad for session %d", sessid);
>-+    g_warning ("rtpbin: failed to get AUX sink pad for session %u", sessid);
>-     return NULL;
>-   }
>- aux_sink_failed:
>-   {
>--    g_warning ("rtpbin: failed to get AUX sink pad for session %d", sessid);
>-+    g_warning ("rtpbin: failed to get AUX sink pad for session %u", sessid);
>-     return NULL;
>-   }
>- pad_failed:
>-   {
>--    g_warning ("rtpbin: failed to get session pad for session %d", sessid);
>-+    g_warning ("rtpbin: failed to get session pad for session %u", sessid);
>-     return NULL;
>-   }
>- session_src_failed:
>-   {
>--    g_warning ("rtpbin: failed to setup source pads for session %d", sessid);
>-+    g_warning ("rtpbin: failed to setup source pads for session %u", sessid);
>-     return NULL;
>-   }
>- }
>-@@ -3978,13 +4222,13 @@ create_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
>- 
>-     GST_DEBUG_OBJECT (rtpbin, "linking RTCP encoder");
>- 
>--    ename = g_strdup_printf ("rtcp_src_%d", sessid);
>-+    ename = g_strdup_printf ("rtcp_src_%u", sessid);
>-     encsrc = gst_element_get_static_pad (encoder, ename);
>-     g_free (ename);
>-     if (encsrc == NULL)
>-       goto enc_src_failed;
>- 
>--    ename = g_strdup_printf ("rtcp_sink_%d", sessid);
>-+    ename = g_strdup_printf ("rtcp_sink_%u", sessid);
>-     encsink = gst_element_get_static_pad (encoder, ename);
>-     g_free (ename);
>-     if (encsink == NULL)
>-@@ -4021,23 +4265,23 @@ no_session:
>-   }
>- pad_failed:
>-   {
>--    g_warning ("rtpbin: failed to get rtcp pad for session %d", sessid);
>-+    g_warning ("rtpbin: failed to get rtcp pad for session %u", sessid);
>-     return NULL;
>-   }
>- enc_src_failed:
>-   {
>--    g_warning ("rtpbin: failed to get encoder src pad for session %d", sessid);
>-+    g_warning ("rtpbin: failed to get encoder src pad for session %u", sessid);
>-     return NULL;
>-   }
>- enc_sink_failed:
>-   {
>--    g_warning ("rtpbin: failed to get encoder sink pad for session %d", sessid);
>-+    g_warning ("rtpbin: failed to get encoder sink pad for session %u", sessid);
>-     gst_object_unref (encsrc);
>-     return NULL;
>-   }
>- enc_link_failed:
>-   {
>--    g_warning ("rtpbin: failed to link rtcp encoder for session %d", sessid);
>-+    g_warning ("rtpbin: failed to link rtcp encoder for session %u", sessid);
>-     gst_object_unref (encsrc);
>-     return NULL;
>-   }
>-diff --git a/gst/rtpmanager/gstrtpbin.h b/gst/rtpmanager/gstrtpbin.h
>-index fb13a47..384b76d 100644
>---- a/gst/rtpmanager/gstrtpbin.h
>-+++ b/gst/rtpmanager/gstrtpbin.h
>-@@ -127,6 +127,8 @@ struct _GstRtpBinClass {
>- 
>-   void     (*on_new_sender_ssrc)      (GstRtpBin *rtpbin, guint session, guint32 ssrc);
>-   void     (*on_sender_ssrc_active)   (GstRtpBin *rtpbin, guint session, guint32 ssrc);
>-+
>-+  guint    (*on_bundled_ssrc)         (GstRtpBin *rtpbin, guint ssrc);
>- };
>- 
>- GType gst_rtp_bin_get_type (void);
>diff --git a/Tools/gtk/patches/gst-plugins-good-0004-qtdemux-add-context-for-a-preferred-protection.patch b/Tools/gtk/patches/gst-plugins-good-0004-qtdemux-add-context-for-a-preferred-protection.patch
>deleted file mode 100644
>index fb15e018f09..00000000000
>--- a/Tools/gtk/patches/gst-plugins-good-0004-qtdemux-add-context-for-a-preferred-protection.patch
>+++ /dev/null
>@@ -1,320 +0,0 @@
>-From ae4f7d4f09a051c2fbbd05e4df9f79fa6522104f Mon Sep 17 00:00:00 2001
>-From: Xabier Rodriguez Calvar <calvaris@igalia.com>
>-Date: Fri, 16 Sep 2016 16:08:18 +0200
>-Subject: [PATCH] qtdemux: add context for a preferred protection
>-
>-qtdemux selected the first system corresponding to a working GStreamer
>-decryptor. With this change, before selecting that decryptor, qtdemux
>-will check if it has context (a preferred decryptor id) and if not, it
>-will request it.
>-
>-The request includes track-id, available key system ids for the
>-available decryptors and event the events so that the init data is
>-accessible.
>----
>- gst/isomp4/qtdemux.c | 209 +++++++++++++++++++++++++++++++++++++++++++++++++--
>- gst/isomp4/qtdemux.h |   2 +-
>- 2 files changed, 204 insertions(+), 7 deletions(-)
>-
>-diff --git a/gst/isomp4/qtdemux.c b/gst/isomp4/qtdemux.c
>-index 0425e66..e05e75e 100644
>---- a/gst/isomp4/qtdemux.c
>-+++ b/gst/isomp4/qtdemux.c
>-@@ -480,6 +480,8 @@ static GstIndex *gst_qtdemux_get_index (GstElement * element);
>- #endif
>- static GstStateChangeReturn gst_qtdemux_change_state (GstElement * element,
>-     GstStateChange transition);
>-+static void gst_qtdemux_set_context (GstElement * element,
>-+    GstContext * context);
>- static gboolean qtdemux_sink_activate (GstPad * sinkpad, GstObject * parent);
>- static gboolean qtdemux_sink_activate_mode (GstPad * sinkpad,
>-     GstObject * parent, GstPadMode mode, gboolean active);
>-@@ -554,6 +556,7 @@ gst_qtdemux_class_init (GstQTDemuxClass * klass)
>-   gstelement_class->set_index = GST_DEBUG_FUNCPTR (gst_qtdemux_set_index);
>-   gstelement_class->get_index = GST_DEBUG_FUNCPTR (gst_qtdemux_get_index);
>- #endif
>-+  gstelement_class->set_context = GST_DEBUG_FUNCPTR (gst_qtdemux_set_context);
>- 
>-   gst_tag_register_musicbrainz_tags ();
>- 
>-@@ -612,6 +615,7 @@ gst_qtdemux_init (GstQTDemux * qtdemux)
>-   qtdemux->cenc_aux_info_sizes = NULL;
>-   qtdemux->cenc_aux_sample_count = 0;
>-   qtdemux->protection_system_ids = NULL;
>-+  qtdemux->preferred_protection_system_id = NULL;
>-   g_queue_init (&qtdemux->protection_event_queue);
>-   gst_segment_init (&qtdemux->segment, GST_FORMAT_TIME);
>-   qtdemux->flowcombiner = gst_flow_combiner_new ();
>-@@ -1972,6 +1976,10 @@ gst_qtdemux_reset (GstQTDemux * qtdemux, gboolean hard)
>-     g_queue_foreach (&qtdemux->protection_event_queue, (GFunc) gst_event_unref,
>-         NULL);
>-     g_queue_clear (&qtdemux->protection_event_queue);
>-+    if (qtdemux->preferred_protection_system_id) {
>-+      g_free (qtdemux->preferred_protection_system_id);
>-+      qtdemux->preferred_protection_system_id = NULL;
>-+    }
>-   }
>-   qtdemux->offset = 0;
>-   gst_adapter_clear (qtdemux->adapter);
>-@@ -2404,6 +2412,29 @@ gst_qtdemux_change_state (GstElement * element, GstStateChange transition)
>- }
>- 
>- static void
>-+gst_qtdemux_set_context (GstElement * element, GstContext * context)
>-+{
>-+  GstQTDemux *qtdemux = GST_QTDEMUX (element);
>-+
>-+  g_return_if_fail (GST_IS_CONTEXT (context));
>-+
>-+  if (g_strcmp0 (gst_context_get_context_type (context),
>-+          "drm-preferred-decryption-system-id") == 0) {
>-+    const GstStructure *s;
>-+
>-+    s = gst_context_get_structure (context);
>-+    qtdemux->preferred_protection_system_id =
>-+        g_strdup (gst_structure_get_string (s, "decryption-system-id"));
>-+    GST_DEBUG_OBJECT (element, "set preferred decryption system to %s",
>-+        qtdemux->preferred_protection_system_id);
>-+  }
>-+
>-+  GST_TRACE_OBJECT (element, "chaining set_context to superclass %p or %p",
>-+      GST_ELEMENT_GET_CLASS (element), parent_class);
>-+  GST_ELEMENT_CLASS (parent_class)->set_context (element, context);
>-+}
>-+
>-+static void
>- qtdemux_parse_ftyp (GstQTDemux * qtdemux, const guint8 * buffer, gint length)
>- {
>-   /* counts as header data */
>-@@ -3392,6 +3423,8 @@ qtdemux_parse_pssh (GstQTDemux * qtdemux, GNode * node)
>-   event = gst_event_new_protection (sysid_string, pssh,
>-       (parent_box_type == FOURCC_moov) ? "isobmff/moov" : "isobmff/moof");
>-   for (i = 0; i < qtdemux->n_streams; ++i) {
>-+    GST_TRACE_OBJECT (qtdemux,
>-+        "adding protection event for stream %d and system %s", i, sysid_string);
>-     g_queue_push_tail (&qtdemux->streams[i]->protection_scheme_event_queue,
>-         gst_event_ref (event));
>-   }
>-@@ -4993,6 +5026,12 @@ gst_qtdemux_decorate_and_push_buffer (GstQTDemux * qtdemux,
>-     GstEvent *event;
>- 
>-     while ((event = g_queue_pop_head (&stream->protection_scheme_event_queue))) {
>-+#if (!GST_DISABLE_GST_DEBUG)
>-+      const gchar *system_id = NULL;
>-+      gst_event_parse_protection (event, &system_id, NULL, NULL);
>-+      GST_TRACE_OBJECT (qtdemux, "pushing again protection event for system %s",
>-+          system_id);
>-+#endif
>-       gst_pad_push_event (stream->pad, event);
>-     }
>- 
>-@@ -6947,11 +6986,148 @@ qtdemux_do_allocation (GstQTDemux * qtdemux, QtDemuxStream * stream)
>- }
>- 
>- static gboolean
>-+pad_query (const GValue * item, GValue * value, gpointer user_data)
>-+{
>-+  GstPad *pad = g_value_get_object (item);
>-+  GstQuery *query = user_data;
>-+  gboolean res;
>-+
>-+  res = gst_pad_peer_query (pad, query);
>-+
>-+  if (res) {
>-+    g_value_set_boolean (value, TRUE);
>-+    return FALSE;
>-+  }
>-+
>-+  GST_INFO_OBJECT (pad, "pad peer query failed");
>-+  return TRUE;
>-+}
>-+
>-+static gboolean
>-+gst_qtdemux_run_query (GstElement * element, GstQuery * query,
>-+    GstPadDirection direction)
>-+{
>-+  GstIterator *it;
>-+  GstIteratorFoldFunction func = pad_query;
>-+  GValue res = { 0, };
>-+
>-+  g_value_init (&res, G_TYPE_BOOLEAN);
>-+  g_value_set_boolean (&res, FALSE);
>-+
>-+  /* Ask neighbor */
>-+  if (direction == GST_PAD_SRC)
>-+    it = gst_element_iterate_src_pads (element);
>-+  else
>-+    it = gst_element_iterate_sink_pads (element);
>-+
>-+  while (gst_iterator_fold (it, func, &res, query) == GST_ITERATOR_RESYNC)
>-+    gst_iterator_resync (it);
>-+
>-+  gst_iterator_free (it);
>-+
>-+  return g_value_get_boolean (&res);
>-+}
>-+
>-+static void
>-+gst_qtdemux_request_protection_context_if_needed (GstQTDemux * qtdemux,
>-+    QtDemuxStream * stream)
>-+{
>-+  GstQuery *query;
>-+  GstContext *ctxt;
>-+  GstElement *element = GST_ELEMENT (qtdemux);
>-+  GstStructure *st;
>-+  gchar **filtered_sys_ids;
>-+  GValue event_list = G_VALUE_INIT;
>-+  GList *walk;
>-+
>-+  /* 1. Check if we already have the context. */
>-+  if (qtdemux->preferred_protection_system_id != NULL) {
>-+    GST_LOG_OBJECT (element,
>-+        "already have the protection context, no need to request it again");
>-+    return;
>-+  }
>-+
>-+  GST_TRACE_OBJECT (qtdemux, "currently we have detected %u protection systems",
>-+      qtdemux->protection_system_ids->len);
>-+  g_ptr_array_add (qtdemux->protection_system_ids, NULL);
>-+  filtered_sys_ids = gst_protection_filter_systems_by_available_decryptors (
>-+      (const gchar **) qtdemux->protection_system_ids->pdata);
>-+  g_ptr_array_remove_index (qtdemux->protection_system_ids,
>-+      qtdemux->protection_system_ids->len - 1);
>-+  if (filtered_sys_ids == NULL || filtered_sys_ids[0] == NULL) {
>-+    GST_LOG_OBJECT (qtdemux, "no suitable decryptors found, not issuing the "
>-+        "context request");
>-+    g_strfreev (filtered_sys_ids);
>-+    return;
>-+  }
>-+  GST_TRACE_OBJECT (qtdemux, "found suitable decryptors, running the context "
>-+      "request");
>-+
>-+  if (stream->protection_scheme_event_queue.length) {
>-+    GST_TRACE_OBJECT (qtdemux, "using stream event queue, length %u",
>-+        stream->protection_scheme_event_queue.length);
>-+    walk = stream->protection_scheme_event_queue.tail;
>-+  } else {
>-+    GST_TRACE_OBJECT (qtdemux, "using demuxer event queue, length %u",
>-+        qtdemux->protection_event_queue.length);
>-+    walk = qtdemux->protection_event_queue.tail;
>-+  }
>-+
>-+  g_value_init (&event_list, GST_TYPE_LIST);
>-+  for (; walk; walk = g_list_previous (walk)) {
>-+    GValue *event_value = g_new0 (GValue, 1);
>-+    g_value_init (event_value, GST_TYPE_EVENT);
>-+    g_value_set_boxed (event_value, walk->data);
>-+    gst_value_list_append_and_take_value (&event_list, event_value);
>-+  }
>-+
>-+  /*  2a) Query downstream with GST_QUERY_CONTEXT for the context and
>-+   *      check if downstream already has a context of the specific type
>-+   *  2b) Query upstream as above.
>-+   */
>-+  query = gst_query_new_context ("drm-preferred-decryption-system-id");
>-+  st = (GstStructure *) gst_query_get_structure (query);
>-+  gst_structure_set (st, "track-id", G_TYPE_UINT, stream->track_id,
>-+      "stream-encryption-systems", G_TYPE_STRV, filtered_sys_ids, NULL);
>-+  gst_structure_set_value (st, "stream-encryption-events", &event_list);
>-+  if (gst_qtdemux_run_query (element, query, GST_PAD_SRC)) {
>-+    gst_query_parse_context (query, &ctxt);
>-+    GST_INFO_OBJECT (element, "found context (%p) in downstream query", ctxt);
>-+    gst_element_set_context (element, ctxt);
>-+  } else if (gst_qtdemux_run_query (element, query, GST_PAD_SINK)) {
>-+    gst_query_parse_context (query, &ctxt);
>-+    GST_INFO_OBJECT (element, "found context (%p) in upstream query", ctxt);
>-+    gst_element_set_context (element, ctxt);
>-+  } else {
>-+    /* 3) Post a GST_MESSAGE_NEED_CONTEXT message on the bus with
>-+     *    the required context type and afterwards check if a
>-+     *    usable context was set now as in 1). The message could
>-+     *    be handled by the parent bins of the element and the
>-+     *    application.
>-+     */
>-+    GstMessage *msg;
>-+
>-+    GST_INFO_OBJECT (element, "posting need context message");
>-+    msg = gst_message_new_need_context (GST_OBJECT_CAST (element),
>-+        "drm-preferred-decryption-system-id");
>-+    st = (GstStructure *) gst_message_get_structure (msg);
>-+    gst_structure_set (st, "track-id", G_TYPE_UINT, stream->track_id,
>-+        "stream-encryption-systems", G_TYPE_STRV, filtered_sys_ids, NULL);
>-+    gst_structure_set_value (st, "stream-encryption-events", &event_list);
>-+    gst_element_post_message (element, msg);
>-+  }
>-+
>-+  g_strfreev (filtered_sys_ids);
>-+  g_value_unset (&event_list);
>-+  gst_query_unref (query);
>-+}
>-+
>-+static gboolean
>- gst_qtdemux_configure_protected_caps (GstQTDemux * qtdemux,
>-     QtDemuxStream * stream)
>- {
>-   GstStructure *s;
>--  const gchar *selected_system;
>-+  const gchar *selected_system = NULL;
>- 
>-   g_return_val_if_fail (qtdemux != NULL, FALSE);
>-   g_return_val_if_fail (stream != NULL, FALSE);
>-@@ -6966,17 +7142,38 @@ gst_qtdemux_configure_protected_caps (GstQTDemux * qtdemux,
>-         "cenc protection system information has been found");
>-     return FALSE;
>-   }
>--  g_ptr_array_add (qtdemux->protection_system_ids, NULL);
>--  selected_system = gst_protection_select_system ((const gchar **)
>--      qtdemux->protection_system_ids->pdata);
>--  g_ptr_array_remove_index (qtdemux->protection_system_ids,
>--      qtdemux->protection_system_ids->len - 1);
>-+
>-+  gst_qtdemux_request_protection_context_if_needed (qtdemux, stream);
>-+  if (qtdemux->preferred_protection_system_id != NULL) {
>-+    guint i;
>-+    for (i = 0; i < qtdemux->protection_system_ids->len; i++) {
>-+      if (g_strcmp0 (g_ptr_array_index (qtdemux->protection_system_ids, i),
>-+              qtdemux->preferred_protection_system_id) == 0) {
>-+        const gchar *preferred_system_array[] =
>-+            { qtdemux->preferred_protection_system_id, NULL };
>-+        selected_system = gst_protection_select_system (preferred_system_array);
>-+        break;
>-+      }
>-+    }
>-+  }
>-+
>-+  if (!selected_system) {
>-+    g_ptr_array_add (qtdemux->protection_system_ids, NULL);
>-+    selected_system = gst_protection_select_system ((const gchar **)
>-+        qtdemux->protection_system_ids->pdata);
>-+    g_ptr_array_remove_index (qtdemux->protection_system_ids,
>-+        qtdemux->protection_system_ids->len - 1);
>-+  }
>-+
>-   if (!selected_system) {
>-     GST_ERROR_OBJECT (qtdemux, "stream is protected, but no "
>-         "suitable decryptor element has been found");
>-     return FALSE;
>-   }
>- 
>-+  GST_DEBUG_OBJECT (qtdemux, "selected protection system is %s",
>-+      selected_system);
>-+
>-   s = gst_caps_get_structure (stream->caps, 0);
>-   if (!gst_structure_has_name (s, "application/x-cenc")) {
>-     gst_structure_set (s,
>-diff --git a/gst/isomp4/qtdemux.h b/gst/isomp4/qtdemux.h
>-index 53bd071..55c4f63 100644
>---- a/gst/isomp4/qtdemux.h
>-+++ b/gst/isomp4/qtdemux.h
>-@@ -153,7 +153,7 @@ struct _GstQTDemux {
>-   guint64 cenc_aux_info_offset;
>-   guint8 *cenc_aux_info_sizes;
>-   guint32 cenc_aux_sample_count;
>--
>-+  gchar *preferred_protection_system_id;
>- };
>- 
>- struct _GstQTDemuxClass {
>--- 
>-2.11.0
>-
>diff --git a/Tools/gtk/patches/gst-plugins-good-Revert-qtdemux-expose-streams-with-first-moof-for-fr.patch b/Tools/gtk/patches/gst-plugins-good-Revert-qtdemux-expose-streams-with-first-moof-for-fr.patch
>deleted file mode 100644
>index 3a60db4772b..00000000000
>--- a/Tools/gtk/patches/gst-plugins-good-Revert-qtdemux-expose-streams-with-first-moof-for-fr.patch
>+++ /dev/null
>@@ -1,133 +0,0 @@
>-From 1a81bd90d4a3e59d6669a0bbfa456f1ed4e5db48 Mon Sep 17 00:00:00 2001
>-From: Xabier Rodriguez Calvar <calvaris@igalia.com>
>-Date: Thu, 7 Apr 2016 13:57:16 +0200
>-Subject: [PATCH] Revert "qtdemux: expose streams with first moof for
>- fragmented format"
>-
>-This reverts commit d8bb6687ea251570c331038279a43d448167d6ad.
>----
>- gst/isomp4/qtdemux.c | 54 ++++++++++++++++------------------------------------
>- gst/isomp4/qtdemux.h |  1 -
>- 2 files changed, 16 insertions(+), 39 deletions(-)
>-
>-diff --git a/gst/isomp4/qtdemux.c b/gst/isomp4/qtdemux.c
>-index 39be163..9636b4b 100644
>---- a/gst/isomp4/qtdemux.c
>-+++ b/gst/isomp4/qtdemux.c
>-@@ -609,7 +609,6 @@ gst_qtdemux_init (GstQTDemux * qtdemux)
>-   qtdemux->state = QTDEMUX_STATE_INITIAL;
>-   qtdemux->pullbased = FALSE;
>-   qtdemux->posted_redirect = FALSE;
>--  qtdemux->pending_configure = FALSE;
>-   qtdemux->neededbytes = 16;
>-   qtdemux->todrop = 0;
>-   qtdemux->adapter = gst_adapter_new ();
>-@@ -2049,7 +2048,6 @@ gst_qtdemux_reset (GstQTDemux * qtdemux, gboolean hard)
>-     gst_caps_replace (&qtdemux->media_caps, NULL);
>-     qtdemux->timescale = 0;
>-     qtdemux->got_moov = FALSE;
>--    qtdemux->pending_configure = FALSE;
>-   } else if (qtdemux->mss_mode) {
>-     gst_flow_combiner_reset (qtdemux->flowcombiner);
>-     for (n = 0; n < qtdemux->n_streams; n++)
>-@@ -6104,7 +6102,6 @@ gst_qtdemux_process_adapter (GstQTDemux * demux, gboolean force)
>-             &fourcc);
>-         if (fourcc == FOURCC_moov) {
>-           gint n;
>--          gboolean got_samples = FALSE;
>- 
>-           /* in usual fragmented setup we could try to scan for more
>-            * and end up at the the moov (after mdat) again */
>-@@ -6136,27 +6133,19 @@ gst_qtdemux_process_adapter (GstQTDemux * demux, gboolean force)
>-             qtdemux_node_dump (demux, demux->moov_node);
>-             qtdemux_parse_tree (demux);
>-             qtdemux_prepare_streams (demux);
>-+            if (!demux->got_moov)
>-+              qtdemux_expose_streams (demux);
>-+            else {
>- 
>--            for (n = 0; n < demux->n_streams; n++) {
>--              QtDemuxStream *stream = demux->streams[n];
>--              got_samples |= stream->stbl_index >= 0;
>--            }
>--            if (!demux->fragmented || got_samples) {
>--              if (!demux->got_moov) {
>--                qtdemux_expose_streams (demux);
>--              } else {
>--                for (n = 0; n < demux->n_streams; n++) {
>--                  QtDemuxStream *stream = demux->streams[n];
>--                  gst_qtdemux_configure_stream (demux, stream);
>--                }
>-+              for (n = 0; n < demux->n_streams; n++) {
>-+                QtDemuxStream *stream = demux->streams[n];
>-+
>-+                gst_qtdemux_configure_stream (demux, stream);
>-               }
>--              gst_qtdemux_check_send_pending_segment (demux);
>--              demux->pending_configure = FALSE;
>--            } else {
>--              demux->pending_configure = TRUE;
>-             }
>- 
>-             demux->got_moov = TRUE;
>-+            gst_qtdemux_check_send_pending_segment (demux);
>- 
>-             /* fragmented streams headers shouldn't contain edts atoms */
>-             if (!demux->fragmented) {
>-@@ -6175,7 +6164,6 @@ gst_qtdemux_process_adapter (GstQTDemux * demux, gboolean force)
>-             guint64 dist = 0;
>-             GstClockTime prev_pts;
>-             guint64 prev_offset;
>--            gint n;
>- 
>-             GST_DEBUG_OBJECT (demux, "Parsing [moof]");
>- 
>-@@ -6209,25 +6197,15 @@ gst_qtdemux_process_adapter (GstQTDemux * demux, gboolean force)
>-               ret = GST_FLOW_ERROR;
>-               goto done;
>-             }
>--            /* in MSS we need to expose the pads after the first moof as we won't get a moov 
>--             * Also, fragmented format need to be exposed if a moov have no valid sample data */
>--            if (demux->mss_mode || demux->pending_configure) {
>--              if (!demux->exposed) {
>--                if (!demux->pending_newsegment) {
>--                  GstSegment segment;
>--                  gst_segment_init (&segment, GST_FORMAT_TIME);
>--                  GST_DEBUG_OBJECT (demux, "new pending_newsegment");
>--                  demux->pending_newsegment = gst_event_new_segment (&segment);
>--                }
>--                qtdemux_expose_streams (demux);
>--              } else {
>--                for (n = 0; n < demux->n_streams; n++) {
>--                  QtDemuxStream *stream = demux->streams[n];
>--                  gst_qtdemux_configure_stream (demux, stream);
>--                }
>-+            /* in MSS we need to expose the pads after the first moof as we won't get a moov */
>-+            if (demux->mss_mode && !demux->exposed) {
>-+              if (!demux->pending_newsegment) {
>-+                GstSegment segment;
>-+                gst_segment_init (&segment, GST_FORMAT_TIME);
>-+                GST_DEBUG_OBJECT (demux, "new pending_newsegment");
>-+                demux->pending_newsegment = gst_event_new_segment (&segment);
>-               }
>--              gst_qtdemux_check_send_pending_segment (demux);
>--              demux->pending_configure = FALSE;
>-+              qtdemux_expose_streams (demux);
>-             }
>-           } else {
>-             GST_DEBUG_OBJECT (demux, "Discarding [moof]");
>-diff --git a/gst/isomp4/qtdemux.h b/gst/isomp4/qtdemux.h
>-index 6061215..ecf0c63 100644
>---- a/gst/isomp4/qtdemux.h
>-+++ b/gst/isomp4/qtdemux.h
>-@@ -89,7 +89,6 @@ struct _GstQTDemux {
>-   gboolean posted_redirect;
>- 
>-   /* push based variables */
>--  gboolean pending_configure;
>-   guint neededbytes;
>-   guint todrop;
>-   GstAdapter *adapter;
>--- 
>-2.8.0.rc3
>-
>diff --git a/Tools/gtk/patches/gst-plugins-good-use-the-tfdt-decode-time.patch b/Tools/gtk/patches/gst-plugins-good-use-the-tfdt-decode-time.patch
>deleted file mode 100644
>index c3c08dfed9e..00000000000
>--- a/Tools/gtk/patches/gst-plugins-good-use-the-tfdt-decode-time.patch
>+++ /dev/null
>@@ -1,89 +0,0 @@
>-From 1556043c00eb60d3871b4baa8b029175c16c7097 Mon Sep 17 00:00:00 2001
>-From: =?UTF-8?q?Enrique=20Oca=C3=B1a=20Gonz=C3=A1lez?= <eocanha@igalia.com>
>-Date: Mon, 24 Oct 2016 16:56:31 +0000
>-Subject: [PATCH] Use the tfdt decode time on byte streams when it's
>- significantly different than the time in the last sample
>-
>-We consider there's a sifnificant difference when it's larger than on second
>-or than half the duration of the last processed fragment in case the latter is
>-larger.
>-
>-https://bugzilla.gnome.org/show_bug.cgi?id=754230
>----
>- gst/isomp4/qtdemux.c | 25 +++++++++++++++++++++++++
>- 1 file changed, 25 insertions(+)
>-
>-diff --git a/gst/isomp4/qtdemux.c b/gst/isomp4/qtdemux.c
>-index db2d361..5430a39 100644
>---- a/gst/isomp4/qtdemux.c
>-+++ b/gst/isomp4/qtdemux.c
>-@@ -95,6 +95,8 @@
>- 
>- #define STREAM_IS_EOS(s) (s->time_position == GST_CLOCK_TIME_NONE)
>- 
>-+#define ABSDIFF(x, y) ( (x) > (y) ? ((x) - (y)) : ((y) - (x)) )
>-+
>- GST_DEBUG_CATEGORY (qtdemux_debug);
>- 
>- /*typedef struct _QtNode QtNode; */
>-@@ -256,6 +258,7 @@ struct _QtDemuxStream
>-   guint32 n_samples_moof;       /* sample count in a moof */
>-   guint64 duration_moof;        /* duration in timescale of a moof, used for figure out
>-                                  * the framerate of fragmented format stream */
>-+  guint64 duration_last_moof;
>-   guint32 offset_in_sample;
>-   guint32 max_buffer_size;
>- 
>-@@ -1828,6 +1831,7 @@ _create_stream (void)
>-   stream->protection_scheme_info = NULL;
>-   stream->n_samples_moof = 0;
>-   stream->duration_moof = 0;
>-+  stream->duration_last_moof = 0;
>-   g_queue_init (&stream->protection_scheme_event_queue);
>-   return stream;
>- }
>-@@ -2315,6 +2319,7 @@ gst_qtdemux_stream_flush_samples_data (GstQTDemux * qtdemux,
>- 
>-   stream->n_samples_moof = 0;
>-   stream->duration_moof = 0;
>-+  stream->duration_last_moof = 0;
>- }
>- 
>- static void
>-@@ -2883,6 +2888,25 @@ qtdemux_parse_trun (GstQTDemux * qtdemux, GstByteReader * trun,
>-           stream->samples[stream->n_samples - 1].timestamp +
>-           stream->samples[stream->n_samples - 1].duration;
>- 
>-+      /* If this is a GST_FORMAT_BYTES stream and there's a significant
>-+       * difference (1 sec.) between decode_ts and timestamp, prefer the
>-+       * former */
>-+      if (!qtdemux->upstream_format_is_time
>-+          && ABSDIFF (decode_ts, timestamp) >
>-+          MAX (stream->duration_last_moof / 2,
>-+              GSTTIME_TO_QTSTREAMTIME (stream, GST_SECOND))) {
>-+        GST_INFO_OBJECT (qtdemux,
>-+            "decode_ts (%" GST_TIME_FORMAT ") and timestamp (%" GST_TIME_FORMAT
>-+            ") are significantly different (more than %" GST_TIME_FORMAT
>-+            "), using decode_ts",
>-+            GST_TIME_ARGS (QTSTREAMTIME_TO_GSTTIME (stream, decode_ts)),
>-+            GST_TIME_ARGS (QTSTREAMTIME_TO_GSTTIME (stream, timestamp)),
>-+            GST_TIME_ARGS (QTSTREAMTIME_TO_GSTTIME (stream,
>-+                    MAX (stream->duration_last_moof / 2,
>-+                        GSTTIME_TO_QTSTREAMTIME (stream, GST_SECOND)))));
>-+        timestamp = decode_ts;
>-+      }
>-+
>-       gst_ts = QTSTREAMTIME_TO_GSTTIME (stream, timestamp);
>-       GST_INFO_OBJECT (qtdemux, "first sample ts %" GST_TIME_FORMAT
>-           " (extends previous samples)", GST_TIME_ARGS (gst_ts));
>-@@ -3544,6 +3568,7 @@ qtdemux_parse_moof (GstQTDemux * qtdemux, const guint8 * buffer, guint length,
>- 
>-     /* initialise moof sample data */
>-     stream->n_samples_moof = 0;
>-+    stream->duration_last_moof = stream->duration_moof;
>-     stream->duration_moof = 0;
>- 
>-     /* Track Run node */
>--- 
>-2.1.4
>-
>diff --git a/Tools/gtk/patches/gstreamer-0001-protection-added-function-to-filter-system-ids.patch b/Tools/gtk/patches/gstreamer-0001-protection-added-function-to-filter-system-ids.patch
>deleted file mode 100644
>index e72ef6c132c..00000000000
>--- a/Tools/gtk/patches/gstreamer-0001-protection-added-function-to-filter-system-ids.patch
>+++ /dev/null
>@@ -1,77 +0,0 @@
>-From 7772eb350591682b6a315c8a87a58131f731f1d4 Mon Sep 17 00:00:00 2001
>-From: Xabier Rodriguez Calvar <calvaris@igalia.com>
>-Date: Wed, 19 Oct 2016 16:44:16 +0200
>-Subject: [PATCH] protection: added function to filter system ids
>-
>-gst_protection_filter_systems_by_available_decryptors takes an array of
>-strings and returns a new array of strings filtered by the avaible
>-decryptors for them so the ones you get are the ones that you should be
>-able to decrypt.
>----
>- gst/gstprotection.c | 36 ++++++++++++++++++++++++++++++++++++
>- gst/gstprotection.h |  2 ++
>- 2 files changed, 38 insertions(+)
>-
>-diff --git a/gst/gstprotection.c b/gst/gstprotection.c
>-index 8ee52ea..2d7e5e0 100644
>---- a/gst/gstprotection.c
>-+++ b/gst/gstprotection.c
>-@@ -191,6 +191,42 @@ gst_protection_select_system (const gchar ** system_identifiers)
>-   return retval;
>- }
>- 
>-+gchar **
>-+gst_protection_filter_systems_by_available_decryptors (const gchar **
>-+    system_identifiers)
>-+{
>-+  GList *decryptors, *walk;
>-+  gchar **retval;
>-+  guint i = 0, decryptors_number;
>-+
>-+  decryptors =
>-+      gst_element_factory_list_get_elements (GST_ELEMENT_FACTORY_TYPE_DECRYPTOR,
>-+      GST_RANK_MARGINAL);
>-+
>-+  decryptors_number = g_list_length (decryptors);
>-+  retval = g_new (gchar *, decryptors_number + 1);
>-+
>-+  GST_TRACE ("found %u decrytors", decryptors_number);
>-+
>-+  for (walk = decryptors; walk; walk = g_list_next (walk)) {
>-+    GstElementFactory *fact = (GstElementFactory *) walk->data;
>-+
>-+    const char *found_sys_id =
>-+        gst_protection_factory_check (fact, system_identifiers);
>-+    GST_TRACE ("factory %s is valid for %s", GST_OBJECT_NAME (fact),
>-+        found_sys_id);
>-+
>-+    if (found_sys_id) {
>-+      retval[i++] = g_strdup (found_sys_id);
>-+    }
>-+  }
>-+  retval[i] = NULL;
>-+
>-+  gst_plugin_feature_list_free (decryptors);
>-+
>-+  return retval;
>-+}
>-+
>- static const gchar *
>- gst_protection_factory_check (GstElementFactory * fact,
>-     const gchar ** system_identifiers)
>-diff --git a/gst/gstprotection.h b/gst/gstprotection.h
>-index f2f54c4..95976c5 100644
>---- a/gst/gstprotection.h
>-+++ b/gst/gstprotection.h
>-@@ -66,6 +66,8 @@ GstProtectionMeta *gst_buffer_add_protection_meta (GstBuffer * buffer,
>-     GstStructure * info);
>- 
>- const gchar *gst_protection_select_system (const gchar ** system_identifiers);
>-+gchar ** gst_protection_filter_systems_by_available_decryptors (
>-+    const gchar ** system_identifiers);
>- 
>- G_END_DECLS
>- #endif /* __GST_PROTECTION_META_H__ */
>--- 
>-2.10.2
>-

Flags:

calvaris: review-

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Attachments on bug 171595: 308892 | 309026