dtmf: move dtmf plugin from -bad to -good

https://bugzilla.gnome.org/show_bug.cgi?id=687416

Android.mk

@@ -13,6 +13,7 @@ GST_PLUGINS_GOOD_BUILT_SOURCES = \
 	gst/cutter/Android.mk		\
 	gst/debugutils/Android.mk	\
 	gst/deinterlace/Android.mk	\
+	gst/dtmf/Android.mk         \
 	gst/effectv/Android.mk		\
 	gst/equalizer/Android.mk	\
 	gst/flv/Android.mk		\
@@ -79,6 +80,7 @@ CONFIGURE_TARGETS += gst-plugins-good-configure
 -include $(GST_PLUGINS_GOOD_TOP)/gst/cutter/Android.mk
 -include $(GST_PLUGINS_GOOD_TOP)/gst/debugutils/Android.mk
 -include $(GST_PLUGINS_GOOD_TOP)/gst/deinterlace/Android.mk
+-include $(GST_PLUGINS_GOOD_TOP)/gst/dtmf/Android.mk
 -include $(GST_PLUGINS_GOOD_TOP)/gst/effectv/Android.mk
 -include $(GST_PLUGINS_GOOD_TOP)/gst/equalizer/Android.mk
 -include $(GST_PLUGINS_GOOD_TOP)/gst/flv/Android.mk

configure.ac

@@ -317,6 +317,7 @@ AG_GST_CHECK_PLUGIN(avi)
 AG_GST_CHECK_PLUGIN(cutter)
 AG_GST_CHECK_PLUGIN(debugutils)
 AG_GST_CHECK_PLUGIN(deinterlace)
+AG_GST_CHECK_PLUGIN(dtmf)
 AG_GST_CHECK_PLUGIN(effectv)
 AG_GST_CHECK_PLUGIN(equalizer)
 AG_GST_CHECK_PLUGIN(flv)
@@ -1088,6 +1089,7 @@ gst/autodetect/Makefile
 gst/avi/Makefile
 gst/cutter/Makefile
 gst/deinterlace/Makefile
+gst/dtmf/Makefile
 gst/debugutils/Makefile
 gst/effectv/Makefile
 gst/equalizer/Makefile

docs/plugins/Makefile.am

@@ -115,6 +115,9 @@ EXTRA_HFILES = \
 	$(top_srcdir)/gst/debugutils/gsttaginject.h \
 	$(top_srcdir)/gst/debugutils/progressreport.h \
 	$(top_srcdir)/gst/deinterlace/gstdeinterlace.h \
+	$(top_srcdir)/gst/dtmf/gstdtmfsrc.h \
+	$(top_srcdir)/gst/dtmf/gstrtpdtmfsrc.h \
+	$(top_srcdir)/gst/dtmf/gstrtpdtmfdepay.h \
 	$(top_srcdir)/gst/effectv/gstaging.h \
 	$(top_srcdir)/gst/effectv/gstdice.h \
 	$(top_srcdir)/gst/effectv/gstedge.h \

docs/plugins/gst-plugins-good-plugins-docs.sgml

@@ -60,6 +60,7 @@
     <xi:include href="xml/element-deinterleave.xml" />
     <xi:include href="xml/element-dicetv.xml" />
     <xi:include href="xml/element-directsoundsink.xml" />
+    <xi:include href="xml/element-dtmfsrc.xml" />
     <xi:include href="xml/element-dv1394src.xml" />
     <xi:include href="xml/element-dvdec.xml" />
     <xi:include href="xml/element-dvdemux.xml" />
@@ -133,6 +134,7 @@
     <xi:include href="xml/element-rtspsrc.xml" />
     <xi:include href="xml/element-rtpbin.xml" />
     <xi:include href="xml/element-rtpdtmfmux.xml" />
+    <xi:include href="xml/element-rtpdtmfsrc.xml" />
     <xi:include href="xml/element-rtpjitterbuffer.xml" />
     <xi:include href="xml/element-rtpmux.xml" />
     <xi:include href="xml/element-rtpptdemux.xml" />
@@ -194,6 +196,7 @@
     <xi:include href="xml/plugin-debug.xml" />
     <xi:include href="xml/plugin-deinterlace.xml" />
     <xi:include href="xml/plugin-directsound.xml" />
+    <xi:include href="xml/plugin-dtmf.xml" />
     <xi:include href="xml/plugin-dv.xml" />
     <xi:include href="xml/plugin-equalizer.xml" />
     <xi:include href="xml/plugin-effectv.xml" />

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

@@ -603,6 +603,25 @@ GST_DSOUND_LOCK
 GST_DSOUND_UNLOCK
 </SECTION>
 
+<SECTION>
+<FILE>element-dtmfsrc</FILE>
+<TITLE>dtmfsrc</TITLE>
+GstDTMFSrc
+<SUBSECTION Standard>
+GstDTMFEventType
+GstDTMFSrcEvent
+GstDTMFSrcClass
+GST_TYPE_DTMF_SRC
+GST_DTMF_SRC
+GST_DTMF_SRC_CAST
+GST_DTMF_SRC_CLASS
+GST_DTMF_SRC_GET_CLASS
+GST_IS_DTMF_SRC
+GST_IS_DTMF_SRC_CLASS
+gst_dtmf_src_get_type
+gst_dtmf_src_plugin_init
+</SECTION>
+
 <SECTION>
 <FILE>element-dv1394src</FILE>
 <TITLE>dv1394src</TITLE>
@@ -1571,6 +1590,40 @@ GST_IS_RTP_DEC_CLASS
 gst_rtp_dec_get_type
 </SECTION>
 
+<SECTION>
+<FILE>element-rtpdtmfdepay</FILE>
+<TITLE>rtpdtmfdepay</TITLE>
+GstRtpDTMFDepay
+<SUBSECTION Standard>
+GstRtpDTMFDepayClass
+GST_TYPE_RTP_DTMF_DEPAY
+GST_IS_RTP_DTMF_DEPAY
+GST_IS_RTP_DTMF_DEPAY_CLASS
+GST_RTP_DTMF_DEPAY
+GST_RTP_DTMF_DEPAY_CLASS
+gst_rtp_dtmf_depay_plugin_init
+</SECTION>
+
+<SECTION>
+<FILE>element-rtpdtmfsrc</FILE>
+<TITLE>rtpdtmfsrc</TITLE>
+GstRTPDTMFSrc
+<SUBSECTION Standard>
+GstRTPDTMFSrcClass
+GST_TYPE_RTP_DTMF_SRC
+GST_IS_RTP_DTMF_SRC
+GST_IS_RTP_DTMF_SRC_CLASS
+GST_RTP_DTMF_SRC
+GST_RTP_DTMF_SRC_CAST
+GST_RTP_DTMF_SRC_CLASS
+GST_RTP_DTMF_SRC_GET_CLASS
+gst_rtp_dtmf_src_get_type
+gst_rtp_dtmf_src_plugin_init
+GstRTPDTMFPayload
+GstRTPDTMFSrcEvent
+GstRTPDTMFEventType
+</SECTION>
+
 <SECTION>
 <FILE>element-rtpj2kpay</FILE>
 <TITLE>rtpj2kpay</TITLE>

docs/plugins/gst-plugins-good-plugins.args

@@ -964,7 +964,7 @@
 <RANGE></RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>message</NICK>
-<BLURB>Post a level message for each passed interval.</BLURB>
+<BLURB>Post a 'level' message for each passed interval (deprecated).</BLURB>
 <DEFAULT>TRUE</DEFAULT>
 </ARG>
 
@@ -988,6 +988,16 @@
 <DEFAULT>300000000</DEFAULT>
 </ARG>
 
+<ARG>
+<NAME>GstLevel::post-messages</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Post Messages</NICK>
+<BLURB>Whether to post a 'level' element message on the bus for each passed interval.</BLURB>
+<DEFAULT>TRUE</DEFAULT>
+</ARG>
+
 <ARG>
 <NAME>GstDiceTV::square-bits</NAME>
 <TYPE>gint</TYPE>
@@ -1114,7 +1124,7 @@
 <RANGE><= 255</RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>Target Blue</NICK>
-<BLURB>The Blue target.</BLURB>
+<BLURB>The blue color value for custom RGB chroma keying.</BLURB>
 <DEFAULT>0</DEFAULT>
 </ARG>
 
@@ -1124,7 +1134,7 @@
 <RANGE><= 255</RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>Target Green</NICK>
-<BLURB>The Green target.</BLURB>
+<BLURB>The green color value for custom RGB chroma keying.</BLURB>
 <DEFAULT>255</DEFAULT>
 </ARG>
 
@@ -1134,7 +1144,7 @@
 <RANGE><= 255</RANGE>
 <FLAGS>rw</FLAGS>
 <NICK>Target Red</NICK>
-<BLURB>The Red target.</BLURB>
+<BLURB>The red color value for custom RGB chroma keying.</BLURB>
 <DEFAULT>0</DEFAULT>
 </ARG>
 
@@ -1153,7 +1163,7 @@
 <TYPE>guint</TYPE>
 <RANGE><= 128</RANGE>
 <FLAGS>rw</FLAGS>
-<NICK>Sensitivity</NICK>
+<NICK>White Sensitivity</NICK>
 <BLURB>Sensitivity to bright colors.</BLURB>
 <DEFAULT>100</DEFAULT>
 </ARG>
@@ -23517,3 +23527,114 @@
 <BLURB>Minimum quantity of frames to send in one packet (-1 for maximum allowed by the mtu).</BLURB>
 <DEFAULT>0</DEFAULT>
 </ARG>
+
+<ARG>
+<NAME>GstRTPDTMFSrc::clock-rate</NAME>
+<TYPE>guint</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>clockrate</NICK>
+<BLURB>The clock-rate at which to generate the dtmf packets.</BLURB>
+<DEFAULT>8000</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRTPDTMFSrc::packet-redundancy</NAME>
+<TYPE>guint</TYPE>
+<RANGE>[1,5]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Packet Redundancy</NICK>
+<BLURB>Number of packets to send to indicate start and stop dtmf events.</BLURB>
+<DEFAULT>1</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRTPDTMFSrc::pt</NAME>
+<TYPE>guint</TYPE>
+<RANGE><= 128</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>payload type</NICK>
+<BLURB>The payload type of the packets.</BLURB>
+<DEFAULT>96</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRTPDTMFSrc::seqnum</NAME>
+<TYPE>guint</TYPE>
+<RANGE></RANGE>
+<FLAGS>r</FLAGS>
+<NICK>Sequence number</NICK>
+<BLURB>The RTP sequence number of the last processed packet.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRTPDTMFSrc::seqnum-offset</NAME>
+<TYPE>gint</TYPE>
+<RANGE>>= G_MAXULONG</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Sequence number Offset</NICK>
+<BLURB>Offset to add to all outgoing seqnum (-1 = random).</BLURB>
+<DEFAULT>-1</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRTPDTMFSrc::ssrc</NAME>
+<TYPE>guint</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>SSRC</NICK>
+<BLURB>The SSRC of the packets (-1 == random).</BLURB>
+<DEFAULT>4294967295</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRTPDTMFSrc::timestamp</NAME>
+<TYPE>guint</TYPE>
+<RANGE></RANGE>
+<FLAGS>r</FLAGS>
+<NICK>Timestamp</NICK>
+<BLURB>The RTP timestamp of the last processed packet.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRTPDTMFSrc::timestamp-offset</NAME>
+<TYPE>gint</TYPE>
+<RANGE>>= G_MAXULONG</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Timestamp Offset</NICK>
+<BLURB>Offset to add to all outgoing timestamps (-1 = random).</BLURB>
+<DEFAULT>-1</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpDTMFDepay::max-duration</NAME>
+<TYPE>guint</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Maximum duration</NICK>
+<BLURB>The maxumimum duration (ms) of the outgoing soundpacket. (0 = no limit).</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRtpDTMFDepay::unit-time</NAME>
+<TYPE>guint</TYPE>
+<RANGE><= 1000</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Duration unittime</NICK>
+<BLURB>The smallest unit (ms) the duration must be a multiple of (0 disables it).</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstDTMFSrc::interval</NAME>
+<TYPE>guint</TYPE>
+<RANGE>[10,50]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Interval between tone packets</NICK>
+<BLURB>Interval in ms between two tone packets.</BLURB>
+<DEFAULT>50</DEFAULT>
+</ARG>
+

docs/plugins/gst-plugins-good-plugins.hierarchy

@@ -2,7 +2,7 @@ GObject
   GInitiallyUnowned
     GstObject
       GstAllocator
-        GstDefaultAllocator
+        GstAllocatorSysmem
       GstAudioRingBuffer
         GstAudioSinkRingBuffer
         GstAudioSrcRingBuffer
@@ -60,6 +60,7 @@ GObject
             GstGdkPixbufSink
             GstV4l2Sink
         GstBaseSrc
+          GstDTMFSrc
           GstPushSrc
             GstAudioBaseSrc
               GstAudioSrc
@@ -74,6 +75,7 @@ GObject
             GstUDPSrc
             GstV4l2Src
             GstXImageSrc
+          GstRTPDTMFSrc
           GstSplitFileSrc
         GstBaseTransform
           GstAudioFilter
@@ -180,6 +182,7 @@ GObject
           GstRtpAC3Depay
           GstRtpAMRDepay
           GstRtpCELTDepay
+          GstRtpDTMFDepay
           GstRtpG722Depay
           GstRtpG723Depay
           GstRtpG726Depay

docs/plugins/gst-plugins-good-plugins.signals

@@ -686,8 +686,6 @@ guint64  arg3
 <FLAGS>l</FLAGS>
 GstV4l2Src *gstv4l2src
 gint  arg1
-guint  arg2
-guint  arg3
-guint  arg4
+GstCaps *arg2
 </SIGNAL>
 

docs/plugins/inspect/plugin-audiofx.xml

@@ -209,13 +209,13 @@
           <name>sink</name>
           <direction>sink</direction>
           <presence>always</presence>
-          <details>audio/x-raw, format=(string){ F32LE, S16LE }, rate=(int)[ 1, 2147483647 ], channels=(int)1, layout=(string)interleaved; audio/x-raw, format=(string){ F32LE, S16LE }, rate=(int)[ 1, 2147483647 ], channels=(int)2, layout=(string)interleaved, channel-mask=(bitmask)0x0000000000000003</details>
+          <details>audio/x-raw, format=(string){ F32LE, S16LE }, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2 ], layout=(string)interleaved</details>
         </caps>
         <caps>
           <name>src</name>
           <direction>source</direction>
           <presence>always</presence>
-          <details>audio/x-raw, format=(string){ S32LE, S16LE }, rate=(int)[ 1, 2147483647 ], channels=(int)2, layout=(string)interleaved, channel-mask=(bitmask)0x0000000000000003</details>
+          <details>audio/x-raw, format=(string){ F32LE, S16LE }, rate=(int)[ 1, 2147483647 ], channels=(int)2, layout=(string)interleaved</details>
         </caps>
       </pads>
     </element>

docs/plugins/inspect/plugin-dtmf.xml

+<plugin>
+  <name>dtmf</name>
+  <description>DTMF plugins</description>
+  <filename>../../gst/dtmf/.libs/libgstdtmf.so</filename>
+  <basename>libgstdtmf.so</basename>
+  <version>1.1.0.1</version>
+  <license>LGPL</license>
+  <source>gst-plugins-good</source>
+  <package>GStreamer Good Plug-ins git</package>
+  <origin>Unknown package origin</origin>
+  <elements>
+    <element>
+      <name>dtmfsrc</name>
+      <longname>DTMF tone generator</longname>
+      <class>Source/Audio</class>
+      <description>Generates DTMF tones</description>
+      <author>Youness Alaoui &lt;youness.alaoui@collabora.co.uk&gt;</author>
+      <pads>
+        <caps>
+          <name>src</name>
+          <direction>source</direction>
+          <presence>always</presence>
+          <details>audio/x-raw, format=(string)S16LE, rate=(int)[ 1, 2147483647 ], channels=(int)1</details>
+        </caps>
+      </pads>
+    </element>
+    <element>
+      <name>rtpdtmfdepay</name>
+      <longname>RTP DTMF packet depayloader</longname>
+      <class>Codec/Depayloader/Network</class>
+      <description>Generates DTMF Sound from telephone-event RTP packets</description>
+      <author>Youness Alaoui &lt;youness.alaoui@collabora.co.uk&gt;</author>
+      <pads>
+        <caps>
+          <name>sink</name>
+          <direction>sink</direction>
+          <presence>always</presence>
+          <details>application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int)[ 0, 2147483647 ], encoding-name=(string)TELEPHONE-EVENT</details>
+        </caps>
+        <caps>
+          <name>src</name>
+          <direction>source</direction>
+          <presence>always</presence>
+          <details>audio/x-raw, format=(string)S16LE, rate=(int)[ 1, 2147483647 ], channels=(int)1</details>
+        </caps>
+      </pads>
+    </element>
+    <element>
+      <name>rtpdtmfsrc</name>
+      <longname>RTP DTMF packet generator</longname>
+      <class>Source/Network</class>
+      <description>Generates RTP DTMF packets</description>
+      <author>Zeeshan Ali &lt;zeeshan.ali@nokia.com&gt;</author>
+      <pads>
+        <caps>
+          <name>src</name>
+          <direction>source</direction>
+          <presence>always</presence>
+          <details>application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int)[ 0, 2147483647 ], encoding-name=(string)TELEPHONE-EVENT</details>
+        </caps>
+      </pads>
+    </element>
+  </elements>
+</plugin>
\ No newline at end of file

docs/plugins/inspect/plugin-jpeg.xml

@@ -20,7 +20,7 @@
           <name>sink</name>
           <direction>sink</direction>
           <presence>always</presence>
-          <details>image/jpeg, width=(int)[ 1, 65535 ], height=(int)[ 1, 65535 ], sof-marker=(int){ 0, 1, 2, 5, 6, 7, 9, 10, 13, 14 }</details>
+          <details>image/jpeg, width=(int)[ 1, 65535 ], height=(int)[ 1, 65535 ]</details>
         </caps>
         <caps>
           <name>src</name>
@@ -47,7 +47,7 @@
           <name>src</name>
           <direction>source</direction>
           <presence>always</presence>
-          <details>image/jpeg, width=(int)[ 16, 65535 ], height=(int)[ 16, 65535 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+          <details>image/jpeg, width=(int)[ 16, 65535 ], height=(int)[ 16, 65535 ], framerate=(fraction)[ 0/1, 2147483647/1 ], sof-marker=(int){ 0, 1, 2, 9 }</details>
         </caps>
       </pads>
     </element>

docs/plugins/inspect/plugin-level.xml

@@ -20,13 +20,13 @@
           <name>sink</name>
           <direction>sink</direction>
           <presence>always</presence>
-          <details>audio/x-raw, format=(string){ S8, S16LE, S32LEF32LE, F64LE }, layout=(string)interleaved, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
+          <details>audio/x-raw, format=(string){ S8, S16LE, S32LE, F32LE, F64LE }, layout=(string)interleaved, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
         </caps>
         <caps>
           <name>src</name>
           <direction>source</direction>
           <presence>always</presence>
-          <details>audio/x-raw, format=(string){ S8, S16LE, S32LEF32LE, F64LE }, layout=(string)interleaved, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
+          <details>audio/x-raw, format=(string){ S8, S16LE, S32LE, F32LE, F64LE }, layout=(string)interleaved, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
         </caps>
       </pads>
     </element>

docs/plugins/inspect/plugin-shout2send.xml

@@ -20,7 +20,7 @@
           <name>sink</name>
           <direction>sink</direction>
           <presence>always</presence>
-          <details>application/ogg; audio/ogg; video/ogg; audio/mpeg, mpegversion=(int)1, layer=(int)[ 1, 3 ]; video/webm; audio/webm</details>
+          <details>application/ogg; audio/ogg; video/ogg; audio/mpeg, mpegversion=(int)1, layer=(int)[ 1, 3 ]</details>
         </caps>
       </pads>
     </element>

gst-plugins-good.spec.in

@@ -129,6 +129,7 @@ rm -rf $RPM_BUILD_ROOT
 %{_libdir}/gstreamer-%{majorminor}/libgstshapewipe.so
 %{_libdir}/gstreamer-%{majorminor}/libgstvideofilter.so
 %{_libdir}/gstreamer-%{majorminor}/libgstaudioparsers.so
+%{_libdir}/gstreamer-%{majorminor}/libgstdtmf.so
 
 # sys plugins
 @USE_GST_V4L2_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstvideo4linux2.so

gst/dtmf/gstdtmf.c

+/* GStreamer DTMF plugin
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
 
 #ifdef HAVE_CONFIG_H
 #include "config.h"
@@ -7,7 +24,6 @@
 #include "gstrtpdtmfsrc.h"
 #include "gstrtpdtmfdepay.h"
 
-
 static gboolean
 plugin_init (GstPlugin * plugin)
 {

gst/dtmf/gstdtmfcommon.h

+/* GStreamer DTMF plugin
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
 
 #ifndef __GST_RTP_DTMF_COMMON_H__
 #define __GST_RTP_DTMF_COMMON_H__
 
+#include <gst/math-compat.h>
+
 #define MIN_INTER_DIGIT_INTERVAL 100     /* ms */
 #define MIN_PULSE_DURATION       250     /* ms */
 
@@ -13,10 +32,6 @@
 #define MIN_EVENT_STRING         "0"
 #define MAX_EVENT_STRING         "15"
 
-#ifndef M_PI
-#define M_PI           3.14159265358979323846  /* pi */
-#endif
-
 typedef struct
 {
   unsigned event:8;             /* Current DTMF event */

tests/check/Makefile.am

@@ -121,6 +121,7 @@ check_PROGRAMS = \
 	elements/capssetter \
 	elements/deinterlace \
 	elements/deinterleave \
+	elements/dtmf \
 	elements/equalizer \
 	elements/flacparse \
 	elements/flvdemux \
@@ -253,6 +254,11 @@ elements_alphacolor_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
 elements_deinterlace_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
 elements_deinterlace_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstvideo-$(GST_API_VERSION) $(LDADD)
 
+elements_dtmf_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) \
+	$(GST_BASE_CFLAGS) $(GST_CFLAGS) $(AM_CFLAGS)
+elements_dtmf_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstrtp-@GST_API_VERSION@ \
+    $(GST_BASE_LIBS) $(GST_LIBS) $(LDADD)
+
 elements_deinterleave_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
 elements_deinterleave_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_API_VERSION) $(LDADD)
 elements_interleave_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)

tests/check/elements/.gitignore

@@ -22,6 +22,7 @@ avisubtitle
 capssetter
 deinterlace
 deinterleave
+dtmf
 equalizer
 gdkpixbufsink
 flacparse