Merge branch 'work'

docs/plugins/.gitignore

@@ -6,6 +6,7 @@ Makefile.in
 *-undocumented.txt
 *-undeclared.txt
 *-unused.txt
+*.bak
 
 tmpl
 xml
@@ -13,7 +14,7 @@ html
 
 gst-plugins-bad-plugins-scan
 gst-plugins-bad-plugins-scan.c
-inspect-registry.
+inspect-registry.*
 
 html-build.stamp
 html.stamp

docs/plugins/Makefile.am

@@ -96,6 +96,7 @@ EXTRA_HFILES = \
 	$(top_srcdir)/ext/celt/gstceltenc.h \
 	$(top_srcdir)/ext/dc1394/gstdc1394.h \
 	$(top_srcdir)/ext/directfb/dfbvideosink.h \
+	$(top_srcdir)/ext/dts/gstdtsdec.h \
 	$(top_srcdir)/ext/ivorbis/vorbisdec.h \
 	$(top_srcdir)/ext/jack/gstjackaudiosrc.h \
 	$(top_srcdir)/ext/jack/gstjackaudiosink.h \

docs/plugins/gst-plugins-bad-plugins-docs.sgml

@@ -34,6 +34,7 @@
     <xi:include href="xml/element-deinterlace.xml" />
     <xi:include href="xml/element-dfbvideosink.xml" />
     <xi:include href="xml/element-dtmfsrc.xml" />
+    <xi:include href="xml/element-dtsdec.xml" />
     <xi:include href="xml/element-dvbsrc.xml" />
     <xi:include href="xml/element-dvdspu.xml" />
     <xi:include href="xml/element-festival.xml" />
@@ -97,6 +98,7 @@
     <xi:include href="xml/plugin-celt.xml" />
     <xi:include href="xml/plugin-dccp.xml" />
     <xi:include href="xml/plugin-dfbvideosink.xml" />
+    <xi:include href="xml/plugin-dirac.xml" />
     <xi:include href="xml/plugin-dtmf.xml" />
     <xi:include href="xml/plugin-dtsdec.xml" />
     <xi:include href="xml/plugin-dvbsrc.xml" />

docs/plugins/gst-plugins-bad-plugins-sections.txt

@@ -248,6 +248,20 @@ gst_dtmf_src_get_type
 gst_dtmf_src_plugin_init
 </SECTION>
 
+<SECTION>
+<FILE>element-dtsdec</FILE>
+<TITLE>dtsdec</TITLE>
+GstDtsDec
+<SUBSECTION Standard>
+GstDtsDecClass
+gst_dtsdec_get_type
+GST_TYPE_DTSDEC
+GST_DTSDEC
+GST_DTSDEC_CLASS
+GST_IS_DTSDEC
+GST_IS_DTSDEC_CLASS
+</SECTION>
+
 <SECTION>
 <FILE>element-dvbsrc</FILE>
 <TITLE>dvbsrc</TITLE>

docs/plugins/inspect/plugin-dirac.xml

@@ -0,0 +1,34 @@
+<plugin>
+  <name>dirac</name>
+  <description>Dirac plugin</description>
+  <filename>../../ext/dirac/.libs/libgstdirac.so</filename>
+  <basename>libgstdirac.so</basename>
+  <version>0.10.10.1</version>
+  <license>LGPL</license>
+  <source>gst-plugins-bad</source>
+  <package>GStreamer Bad Plug-ins CVS/prerelease</package>
+  <origin>Unknown package origin</origin>
+  <elements>
+    <element>
+      <name>diracenc</name>
+      <longname>Dirac Encoder</longname>
+      <class>Codec/Encoder/Video</class>
+      <description>Encode raw YUV video into Dirac stream</description>
+      <author>David Schleef &lt;ds@schleef.org&gt;</author>
+      <pads>
+        <caps>
+          <name>sink</name>
+          <direction>sink</direction>
+          <presence>always</presence>
+          <details>video/x-raw-yuv, format=(fourcc){ I420, YUY2, UYVY, AYUV }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+        </caps>
+        <caps>
+          <name>src</name>
+          <direction>source</direction>
+          <presence>always</presence>
+          <details>video/x-dirac</details>
+        </caps>
+      </pads>
+    </element>
+  </elements>
+</plugin>

ext/apexsink/gstapexraop.c

@@ -231,7 +231,17 @@ gst_apexraop_connect (GstApExRAOP * con)
       creq[GST_APEX_RAOP_SDP_DEFAULT_LENGTH],
       hreq[GST_APEX_RAOP_HDR_DEFAULT_LENGTH], *req;
   RSA *rsa;
-  guchar *mod, *exp, buf[4 + 8 + 16], rsakey[512];
+  guchar *mod, *exp, rsakey[512];
+  union gst_randbytes
+  {
+    struct asvals
+    {
+      gulong url_key;
+      guint64 conn_id;
+      guchar challenge[16];
+    } v;
+    guchar buf[4 + 8 + 16];
+  } randbuf;
   gsize size;
   struct sockaddr_in ioaddr;
   socklen_t iolen;
@@ -257,12 +267,9 @@ gst_apexraop_connect (GstApExRAOP * con)
           sizeof (conn->ctrl_sd_in)) < 0)
     return GST_RTSP_STS_DESTINATION_UNREACHABLE;
 
-  RAND_bytes (buf, sizeof (buf));
+  RAND_bytes (randbuf.buf, sizeof (randbuf));
-  sprintf ((gchar *) conn->url_abspath, "%lu", *((gulong *) buf));
+  sprintf ((gchar *) conn->url_abspath, "%lu", randbuf.v.url_key);
-  ac = g_base64_encode (buf + 12, 16);
+  sprintf ((char *) conn->cid, "%16" G_GINT64_MODIFIER "x", randbuf.v.conn_id);
-  g_strdel (ac, '=');
-  sprintf ((char *) conn->cid, "%08lx%08lx", *((gulong *) (buf + 4)),
-      *((gulong *) (buf + 8)));
 
   RAND_bytes (conn->aes_ky, AES_BLOCK_SIZE);
   RAND_bytes (conn->aes_iv, AES_BLOCK_SIZE);
@@ -285,6 +292,9 @@ gst_apexraop_connect (GstApExRAOP * con)
   getsockname (conn->ctrl_sd, (struct sockaddr *) &ioaddr, &iolen);
   inet_ntop (AF_INET, &(ioaddr.sin_addr), inaddr, INET_ADDRSTRLEN);
 
+  ac = g_base64_encode (randbuf.v.challenge, 16);
+  g_strdel (ac, '=');
+
   sprintf (creq,
       "v=0\r\n"
       "o=iTunes %s 0 IN IP4 %s\r\n"

ext/celt/gstceltdec.c

@@ -48,6 +48,8 @@
 GST_DEBUG_CATEGORY_STATIC (celtdec_debug);
 #define GST_CAT_DEFAULT celtdec_debug
 
+#define DEC_MAX_FRAME_SIZE 2000
+
 static GstStaticPadTemplate celt_dec_src_factory =
 GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,

ext/celt/gstceltdec.h

@@ -41,8 +41,6 @@ G_BEGIN_DECLS
 typedef struct _GstCeltDec GstCeltDec;
 typedef struct _GstCeltDecClass GstCeltDecClass;
 
-#define DEC_MAX_FRAME_SIZE 2000
-
 struct _GstCeltDec {
   GstElement            element;
 

ext/dts/gstdtsdec.c

@@ -17,9 +17,20 @@
  * Boston, MA 02111-1307, USA.
  */
 
-/* TODO: - Port to libdca API instead of relying on the compat header.
+/**
- *         libdca is the successor of libdts:
+ * SECTION:element-dtsdec
- *         http://www.videolan.org/developers/libdca.html
+ *
+ * Digital Theatre System (DTS) audio decoder
+ * 
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * gst-launch dvdreadsrc title=1 ! mpegpsdemux ! dtsdec ! audioresample ! audioconvert ! alsasink
+ * ]| Play a DTS audio track from a dvd.
+ * |[
+ * gst-launch filesrc location=abc.dts ! dtsdec ! audioresample ! audioconvert ! alsasink
+ * ]| Decode a standalone file and play it.
+ * </refsect2>
  */
 
 #ifdef HAVE_CONFIG_H
@@ -162,6 +173,14 @@ gst_dtsdec_class_init (GstDtsDecClass * klass)
 
   gstelement_class->change_state = gst_dtsdec_change_state;
 
+  /**
+   * GstDtsDec::drc
+   *
+   * Set to true to apply the recommended DTS dynamic range compression
+   * to the audio stream. Dynamic range compression makes loud sounds
+   * softer and soft sounds louder, so you can more easily listen
+   * to the stream without disturbing other people.
+   */
   g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_DRC,
       g_param_spec_boolean ("drc", "Dynamic Range Compression",
           "Use Dynamic Range Compression", FALSE, G_PARAM_READWRITE));