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Remove DTMF plugin, moved to -good

Browse source 
https://bugzilla.gnome.org/show_bug.cgi?id=687416
This commit is contained in:
Tim-Philipp Müller 2013-03-09 01:06:31 +00:00
parent ae550222a8
commit c2446a70f6
19 changed files with 0 additions and 3792 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

2
Android.mk
View file

@ -25,7 +25,6 @@ GST_PLUGINS_BAD_BUILT_SOURCES := \
gst/hls/Android.mk \ gst/hls/Android.mk \
gst/jp2kdecimator/Android.mk \ gst/jp2kdecimator/Android.mk \
gst/segmentclip/Android.mk \ gst/segmentclip/Android.mk \
gst/dtmf/Android.mk \
gst/mpeg4videoparse/Android.mk \ gst/mpeg4videoparse/Android.mk \
gst/siren/Android.mk \ gst/siren/Android.mk \
gst/dataurisrc/Android.mk \ gst/dataurisrc/Android.mk \
@ -117,7 +116,6 @@ CONFIGURE_TARGETS += gst-plugins-bad-configure
-include $(GST_PLUGINS_BAD_TOP)/gst/hls/Android.mk -include $(GST_PLUGINS_BAD_TOP)/gst/hls/Android.mk
-include $(GST_PLUGINS_BAD_TOP)/gst/jp2kdecimator/Android.mk -include $(GST_PLUGINS_BAD_TOP)/gst/jp2kdecimator/Android.mk
-include $(GST_PLUGINS_BAD_TOP)/gst/segmentclip/Android.mk -include $(GST_PLUGINS_BAD_TOP)/gst/segmentclip/Android.mk
-include $(GST_PLUGINS_BAD_TOP)/gst/dtmf/Android.mk
-include $(GST_PLUGINS_BAD_TOP)/gst/mpeg4videoparse/Android.mk -include $(GST_PLUGINS_BAD_TOP)/gst/mpeg4videoparse/Android.mk
-include $(GST_PLUGINS_BAD_TOP)/gst/siren/Android.mk -include $(GST_PLUGINS_BAD_TOP)/gst/siren/Android.mk
-include $(GST_PLUGINS_BAD_TOP)/gst/dataurisrc/Android.mk -include $(GST_PLUGINS_BAD_TOP)/gst/dataurisrc/Android.mk

2
configure.ac
View file

@ -346,7 +346,6 @@ AG_GST_CHECK_PLUGIN(coloreffects)
AG_GST_CHECK_PLUGIN(dataurisrc) AG_GST_CHECK_PLUGIN(dataurisrc)
AG_GST_CHECK_PLUGIN(dccp) AG_GST_CHECK_PLUGIN(dccp)
AG_GST_CHECK_PLUGIN(debugutils) AG_GST_CHECK_PLUGIN(debugutils)
AG_GST_CHECK_PLUGIN(dtmf)
AG_GST_CHECK_PLUGIN(dvbsuboverlay) AG_GST_CHECK_PLUGIN(dvbsuboverlay)
AG_GST_CHECK_PLUGIN(dvdspu) AG_GST_CHECK_PLUGIN(dvdspu)
AG_GST_CHECK_PLUGIN(faceoverlay) AG_GST_CHECK_PLUGIN(faceoverlay)
@ -2282,7 +2281,6 @@ gst/coloreffects/Makefile
gst/dataurisrc/Makefile gst/dataurisrc/Makefile
gst/dccp/Makefile gst/dccp/Makefile
gst/debugutils/Makefile gst/debugutils/Makefile
gst/dtmf/Makefile
gst/dvbsuboverlay/Makefile gst/dvbsuboverlay/Makefile
gst/dvdspu/Makefile gst/dvdspu/Makefile
gst/faceoverlay/Makefile gst/faceoverlay/Makefile

3
docs/plugins/Makefile.am
View file

@ -129,9 +129,6 @@ EXTRA_HFILES = \
$(top_srcdir)/gst/dccp/gstdccpserversink.h \ $(top_srcdir)/gst/dccp/gstdccpserversink.h \
$(top_srcdir)/gst/dccp/gstdccpserversrc.h \ $(top_srcdir)/gst/dccp/gstdccpserversrc.h \
$(top_srcdir)/gst/debugutils/fpsdisplaysink.h \ $(top_srcdir)/gst/debugutils/fpsdisplaysink.h \
$(top_srcdir)/gst/dtmf/gstdtmfsrc.h \
$(top_srcdir)/gst/dtmf/gstrtpdtmfsrc.h \
$(top_srcdir)/gst/dtmf/gstrtpdtmfdepay.h \
$(top_srcdir)/gst/dvdspu/gstdvdspu.h \ $(top_srcdir)/gst/dvdspu/gstdvdspu.h \
$(top_srcdir)/gst/festival/gstfestival.h \ $(top_srcdir)/gst/festival/gstfestival.h \
$(top_srcdir)/gst/gaudieffects/gstburn.h \ $(top_srcdir)/gst/gaudieffects/gstburn.h \

3
docs/plugins/gst-plugins-bad-plugins-docs.sgml
View file

@ -40,7 +40,6 @@
<xi:include href="xml/element-dilate.xml" /> <xi:include href="xml/element-dilate.xml" />
<xi:include href="xml/element-dodge.xml" /> <xi:include href="xml/element-dodge.xml" />
<xi:include href="xml/element-dtmfdetect.xml" /> <xi:include href="xml/element-dtmfdetect.xml" />
<xi:include href="xml/element-dtmfsrc.xml" />
<xi:include href="xml/element-dtsdec.xml" /> <xi:include href="xml/element-dtsdec.xml" />
<xi:include href="xml/element-dvbsrc.xml" /> <xi:include href="xml/element-dvbsrc.xml" />
<xi:include href="xml/element-dvdspu.xml" /> <xi:include href="xml/element-dvdspu.xml" />
@ -71,7 +70,6 @@
<xi:include href="xml/element-pyramidsegment.xml" /> <xi:include href="xml/element-pyramidsegment.xml" />
<xi:include href="xml/element-rtmpsink.xml" /> <xi:include href="xml/element-rtmpsink.xml" />
<xi:include href="xml/element-rtmpsrc.xml" /> <xi:include href="xml/element-rtmpsrc.xml" />
<xi:include href="xml/element-rtpdtmfsrc.xml" />
<xi:include href="xml/element-shmsink.xml" /> <xi:include href="xml/element-shmsink.xml" />
<xi:include href="xml/element-shmsrc.xml" /> <xi:include href="xml/element-shmsrc.xml" />
<xi:include href="xml/element-sdpdemux.xml" /> <xi:include href="xml/element-sdpdemux.xml" />
@ -109,7 +107,6 @@
<xi:include href="xml/plugin-dataurisrc.xml" /> <xi:include href="xml/plugin-dataurisrc.xml" />
<xi:include href="xml/plugin-debugutilsbad.xml" /> <xi:include href="xml/plugin-debugutilsbad.xml" />
<xi:include href="xml/plugin-dirac.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-dtsdec.xml" />
<xi:include href="xml/plugin-dvb.xml" /> <xi:include href="xml/plugin-dvb.xml" />
<xi:include href="xml/plugin-dvdspu.xml" /> <xi:include href="xml/plugin-dvdspu.xml" />

53
docs/plugins/gst-plugins-bad-plugins-sections.txt
View file

@ -475,25 +475,6 @@ gst_dodge_get_type
gst_dodge_plugin_init gst_dodge_plugin_init
</SECTION> </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> <SECTION>
<FILE>element-dtmfdetect</FILE> <FILE>element-dtmfdetect</FILE>
<TITLE>dtmfdetect</TITLE> <TITLE>dtmfdetect</TITLE>
@ -1174,40 +1155,6 @@ GST_RTMP_SRC_CLASS
GST_IS_RTMP_SRC_CLASS GST_IS_RTMP_SRC_CLASS
</SECTION> </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> <SECTION>
<FILE>element-sdlaudiosink</FILE> <FILE>element-sdlaudiosink</FILE>
<TITLE>sdlaudiosink</TITLE> <TITLE>sdlaudiosink</TITLE>

85
docs/plugins/inspect/plugin-dtmf.xml
View file

@ -1,85 +0,0 @@
<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-bad</source>
<package>GStreamer Bad Plug-ins git</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<name>dtmfdetect</name>
<longname>DTMF detector element</longname>
<class>Filter/Analyzer/Audio</class>
<description>This element detects DTMF tones</description>
<author>Olivier Crete &lt;olivier.crete@collabora.com&gt;</author>
<pads>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>audio/x-raw, format=(string)S16LE, rate=(int)8000, channels=(int)1</details>
</caps>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>audio/x-raw, format=(string)S16LE, rate=(int)8000, channels=(int)1</details>
</caps>
</pads>
</element>
<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 ], ssrc=(int)[ 0, 2147483647 ], encoding-name=(string)TELEPHONE-EVENT</details>
</caps>
</pads>
</element>
</elements>
</plugin>

1
gst-plugins-bad.spec.in
View file

@ -213,7 +213,6 @@ make ERROR_CFLAGS='' ERROR_CXXFLAGS=''
# %{_libdir}/gstreamer-%{majorminor}/libgstcoloreffects.so # %{_libdir}/gstreamer-%{majorminor}/libgstcoloreffects.so
%{_libdir}/gstreamer-%{majorminor}/libgstdataurisrc.so %{_libdir}/gstreamer-%{majorminor}/libgstdataurisrc.so
# %{_libdir}/gstreamer-%{majorminor}/libgstdccp.so # %{_libdir}/gstreamer-%{majorminor}/libgstdccp.so
%{_libdir}/gstreamer-%{majorminor}/libgstdtmf.so
# %{_libdir}/gstreamer-%{majorminor}/libgstfestival.so # %{_libdir}/gstreamer-%{majorminor}/libgstfestival.so
# %{_libdir}/gstreamer-%{majorminor}/libgstfrei0r.so # %{_libdir}/gstreamer-%{majorminor}/libgstfrei0r.so
# %{_libdir}/gstreamer-%{majorminor}/libgstgaudieffects.so # %{_libdir}/gstreamer-%{majorminor}/libgstgaudieffects.so

31
gst/dtmf/Makefile.am
View file

@ -1,31 +0,0 @@
plugin_LTLIBRARIES = libgstdtmf.la
libgstdtmf_la_SOURCES = gstdtmfsrc.c \
gstrtpdtmfsrc.c \
gstrtpdtmfdepay.c \
gstdtmf.c
noinst_HEADERS = gstdtmfsrc.h \
gstrtpdtmfsrc.h \
gstrtpdtmfdepay.h \
gstdtmfcommon.h
libgstdtmf_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS)
libgstdtmf_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstrtp-@GST_API_VERSION@ \
$(GST_BASE_LIBS) $(GST_LIBS) $(LIBM)
libgstdtmf_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstdtmf_la_LIBTOOLFLAGS = --tag=disable-static
Android.mk: Makefile.am $(BUILT_SOURCES)
androgenizer \
-:PROJECT libgstdtmf -:SHARED libgstdtmf \
-:TAGS eng debug \
-:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
-:SOURCES $(libgstdtmf_la_SOURCES) \
-:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstdtmf_la_CFLAGS) \
-:LDFLAGS $(libgstdtmf_la_LDFLAGS) \
$(libgstdtmf_la_LIBADD) \
-ldl \
-:PASSTHROUGH LOCAL_ARM_MODE:=arm \
LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
> $@

29
gst/dtmf/gstdtmf.c
View file

@ -1,29 +0,0 @@
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstdtmfsrc.h"
#include "gstrtpdtmfsrc.h"
#include "gstrtpdtmfdepay.h"
static gboolean
plugin_init (GstPlugin * plugin)
{
if (!gst_dtmf_src_plugin_init (plugin))
return FALSE;
if (!gst_rtp_dtmf_src_plugin_init (plugin))
return FALSE;
if (!gst_rtp_dtmf_depay_plugin_init (plugin))
return FALSE;
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
dtmf, "DTMF plugins",
plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)

37
gst/dtmf/gstdtmfcommon.h
View file

@ -1,37 +0,0 @@
#ifndef __GST_RTP_DTMF_COMMON_H__
#define __GST_RTP_DTMF_COMMON_H__
#define MIN_INTER_DIGIT_INTERVAL 100 /* ms */
#define MIN_PULSE_DURATION 250 /* ms */
#define MIN_VOLUME 0
#define MAX_VOLUME 36
#define MIN_EVENT 0
#define MAX_EVENT 15
#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 */
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
unsigned volume:6; /* power level of the tone, in dBm0 */
unsigned r:1; /* Reserved-bit */
unsigned e:1; /* End-bit */
#elif G_BYTE_ORDER == G_BIG_ENDIAN
unsigned e:1; /* End-bit */
unsigned r:1; /* Reserved-bit */
unsigned volume:6; /* power level of the tone, in dBm0 */
#else
#error "G_BYTE_ORDER should be big or little endian."
#endif
unsigned duration:16; /* Duration of digit, in timestamp units */
} GstRTPDTMFPayload;
#endif /* __GST_RTP_DTMF_COMMON_H__ */

969
gst/dtmf/gstdtmfsrc.c
View file

@ -1,969 +0,0 @@
/* GStreamer DTMF source
*
* gstdtmfsrc.c:
*
* Copyright (C) <2007> Collabora.
* Contact: Youness Alaoui <youness.alaoui@collabora.co.uk>
* Copyright (C) <2007> Nokia Corporation.
* Contact: Zeeshan Ali <zeeshan.ali@nokia.com>
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000,2005 Wim Taymans <wim@fluendo.com>
*
* 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.
*/
/**
* SECTION:element-dtmfsrc
* @see_also: rtpdtmsrc, rtpdtmfmuxx
*
* The DTMFSrc element generates DTMF (ITU-T Q.23 Specification) tone packets on request
* from application. The application communicates the beginning and end of a
* DTMF event using custom upstream gstreamer events. To report a DTMF event, an
* application must send an event of type GST_EVENT_CUSTOM_UPSTREAM, having a
* structure of name "dtmf-event" with fields set according to the following
* table:
*
* <informaltable>
* <tgroup cols='4'>
* <colspec colname='Name' />
* <colspec colname='Type' />
* <colspec colname='Possible values' />
* <colspec colname='Purpose' />
* <thead>
* <row>
* <entry>Name</entry>
* <entry>GType</entry>
* <entry>Possible values</entry>
* <entry>Purpose</entry>
* </row>
* </thead>
* <tbody>
* <row>
* <entry>type</entry>
* <entry>G_TYPE_INT</entry>
* <entry>0-1</entry>
* <entry>The application uses this field to specify which of the two methods
* specified in RFC 2833 to use. The value should be 0 for tones and 1 for
* named events. Tones are specified by their frequencies and events are specied
* by their number. This element can only take events as input. Do not confuse
* with "method" which specified the output.
* </entry>
* </row>
* <row>
* <entry>number</entry>
* <entry>G_TYPE_INT</entry>
* <entry>0-15</entry>
* <entry>The event number.</entry>
* </row>
* <row>
* <entry>volume</entry>
* <entry>G_TYPE_INT</entry>
* <entry>0-36</entry>
* <entry>This field describes the power level of the tone, expressed in dBm0
* after dropping the sign. Power levels range from 0 to -63 dBm0. The range of
* valid DTMF is from 0 to -36 dBm0. Can be omitted if start is set to FALSE.
* </entry>
* </row>
* <row>
* <entry>start</entry>
* <entry>G_TYPE_BOOLEAN</entry>
* <entry>True or False</entry>
* <entry>Whether the event is starting or ending.</entry>
* </row>
* <row>
* <entry>method</entry>
* <entry>G_TYPE_INT</entry>
* <entry>2</entry>
* <entry>The method used for sending event, this element will react if this
* field is absent or 2.
* </entry>
* </row>
* </tbody>
* </tgroup>
* </informaltable>
*
* For example, the following code informs the pipeline (and in turn, the
* DTMFSrc element inside the pipeline) about the start of a DTMF named
* event '1' of volume -25 dBm0:
*
* <programlisting>
* structure = gst_structure_new ("dtmf-event",
* "type", G_TYPE_INT, 1,
* "number", G_TYPE_INT, 1,
* "volume", G_TYPE_INT, 25,
* "start", G_TYPE_BOOLEAN, TRUE, NULL);
*
* event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, structure);
* gst_element_send_event (pipeline, event);
* </programlisting>
*
* When a DTMF tone actually starts or stop, a "dtmf-event-processed"
* element #GstMessage with the same fields as the "dtmf-event"
* #GstEvent that was used to request the event. Also, if any event
* has not been processed when the element goes from the PAUSED to the
* READY state, then a "dtmf-event-dropped" message is posted on the
* #GstBus in the order that they were received.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include <glib.h>
#include "gstdtmfcommon.h"
#include "gstdtmfsrc.h"
#include <gst/audio/audio.h>
#define GST_TONE_DTMF_TYPE_EVENT 1
#define DEFAULT_PACKET_INTERVAL 50 /* ms */
#define MIN_PACKET_INTERVAL 10 /* ms */
#define MAX_PACKET_INTERVAL 50 /* ms */
#define DEFAULT_SAMPLE_RATE 8000
#define SAMPLE_SIZE 16
#define CHANNELS 1
#define MIN_DUTY_CYCLE (MIN_INTER_DIGIT_INTERVAL + MIN_PULSE_DURATION)
typedef struct st_dtmf_key
{
const char *event_name;
int event_encoding;
float low_frequency;
float high_frequency;
} DTMF_KEY;
static const DTMF_KEY DTMF_KEYS[] = {
{"DTMF_KEY_EVENT_0", 0, 941, 1336},
{"DTMF_KEY_EVENT_1", 1, 697, 1209},
{"DTMF_KEY_EVENT_2", 2, 697, 1336},
{"DTMF_KEY_EVENT_3", 3, 697, 1477},
{"DTMF_KEY_EVENT_4", 4, 770, 1209},
{"DTMF_KEY_EVENT_5", 5, 770, 1336},
{"DTMF_KEY_EVENT_6", 6, 770, 1477},
{"DTMF_KEY_EVENT_7", 7, 852, 1209},
{"DTMF_KEY_EVENT_8", 8, 852, 1336},
{"DTMF_KEY_EVENT_9", 9, 852, 1477},
{"DTMF_KEY_EVENT_S", 10, 941, 1209},
{"DTMF_KEY_EVENT_P", 11, 941, 1477},
{"DTMF_KEY_EVENT_A", 12, 697, 1633},
{"DTMF_KEY_EVENT_B", 13, 770, 1633},
{"DTMF_KEY_EVENT_C", 14, 852, 1633},
{"DTMF_KEY_EVENT_D", 15, 941, 1633},
};
#define MAX_DTMF_EVENTS 16
enum
{
DTMF_KEY_EVENT_1 = 1,
DTMF_KEY_EVENT_2 = 2,
DTMF_KEY_EVENT_3 = 3,
DTMF_KEY_EVENT_4 = 4,
DTMF_KEY_EVENT_5 = 5,
DTMF_KEY_EVENT_6 = 6,
DTMF_KEY_EVENT_7 = 7,
DTMF_KEY_EVENT_8 = 8,
DTMF_KEY_EVENT_9 = 9,
DTMF_KEY_EVENT_0 = 0,
DTMF_KEY_EVENT_STAR = 10,
DTMF_KEY_EVENT_POUND = 11,
DTMF_KEY_EVENT_A = 12,
DTMF_KEY_EVENT_B = 13,
DTMF_KEY_EVENT_C = 14,
DTMF_KEY_EVENT_D = 15,
};
GST_DEBUG_CATEGORY_STATIC (gst_dtmf_src_debug);
#define GST_CAT_DEFAULT gst_dtmf_src_debug
enum
{
PROP_0,
PROP_INTERVAL,
};
static GstStaticPadTemplate gst_dtmf_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) \"" GST_AUDIO_NE (S16) "\", "
"rate = " GST_AUDIO_RATE_RANGE ", " "channels = (int) 1")
);
#define parent_class gst_dtmf_src_parent_class
G_DEFINE_TYPE (GstDTMFSrc, gst_dtmf_src, GST_TYPE_BASE_SRC);
static void gst_dtmf_src_finalize (GObject * object);
static void gst_dtmf_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_dtmf_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_dtmf_src_handle_event (GstBaseSrc * src, GstEvent * event);
static gboolean gst_dtmf_src_send_event (GstElement * src, GstEvent * event);
static GstStateChangeReturn gst_dtmf_src_change_state (GstElement * element,
GstStateChange transition);
static GstFlowReturn gst_dtmf_src_create (GstBaseSrc * basesrc,
guint64 offset, guint length, GstBuffer ** buffer);
static void gst_dtmf_src_add_start_event (GstDTMFSrc * dtmfsrc,
gint event_number, gint event_volume);
static void gst_dtmf_src_add_stop_event (GstDTMFSrc * dtmfsrc);
static gboolean gst_dtmf_src_unlock (GstBaseSrc * src);
static gboolean gst_dtmf_src_unlock_stop (GstBaseSrc * src);
static gboolean gst_dtmf_src_negotiate (GstBaseSrc * basesrc);
static void
gst_dtmf_src_class_init (GstDTMFSrcClass * klass)
{
GObjectClass *gobject_class;
GstBaseSrcClass *gstbasesrc_class;
GstElementClass *gstelement_class;
gobject_class = G_OBJECT_CLASS (klass);
gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
gstelement_class = GST_ELEMENT_CLASS (klass);
GST_DEBUG_CATEGORY_INIT (gst_dtmf_src_debug, "dtmfsrc", 0, "dtmfsrc element");
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_dtmf_src_template));
gst_element_class_set_static_metadata (gstelement_class,
"DTMF tone generator", "Source/Audio", "Generates DTMF tones",
"Youness Alaoui <youness.alaoui@collabora.co.uk>");
gobject_class->finalize = gst_dtmf_src_finalize;
gobject_class->set_property = gst_dtmf_src_set_property;
gobject_class->get_property = gst_dtmf_src_get_property;
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_INTERVAL,
g_param_spec_uint ("interval", "Interval between tone packets",
"Interval in ms between two tone packets", MIN_PACKET_INTERVAL,
MAX_PACKET_INTERVAL, DEFAULT_PACKET_INTERVAL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_dtmf_src_change_state);
gstelement_class->send_event = GST_DEBUG_FUNCPTR (gst_dtmf_src_send_event);
gstbasesrc_class->unlock = GST_DEBUG_FUNCPTR (gst_dtmf_src_unlock);
gstbasesrc_class->unlock_stop = GST_DEBUG_FUNCPTR (gst_dtmf_src_unlock_stop);
gstbasesrc_class->event = GST_DEBUG_FUNCPTR (gst_dtmf_src_handle_event);
gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_dtmf_src_create);
gstbasesrc_class->negotiate = GST_DEBUG_FUNCPTR (gst_dtmf_src_negotiate);
}
static void
event_free (GstDTMFSrcEvent * event)
{
if (event)
g_slice_free (GstDTMFSrcEvent, event);
}
static void
gst_dtmf_src_init (GstDTMFSrc * dtmfsrc)
{
/* we operate in time */
gst_base_src_set_format (GST_BASE_SRC (dtmfsrc), GST_FORMAT_TIME);
gst_base_src_set_live (GST_BASE_SRC (dtmfsrc), TRUE);
dtmfsrc->interval = DEFAULT_PACKET_INTERVAL;
dtmfsrc->event_queue = g_async_queue_new_full ((GDestroyNotify) event_free);
dtmfsrc->last_event = NULL;
dtmfsrc->sample_rate = DEFAULT_SAMPLE_RATE;
GST_DEBUG_OBJECT (dtmfsrc, "init done");
}
static void
gst_dtmf_src_finalize (GObject * object)
{
GstDTMFSrc *dtmfsrc;
dtmfsrc = GST_DTMF_SRC (object);
if (dtmfsrc->event_queue) {
g_async_queue_unref (dtmfsrc->event_queue);
dtmfsrc->event_queue = NULL;
}
G_OBJECT_CLASS (gst_dtmf_src_parent_class)->finalize (object);
}
static gboolean
gst_dtmf_src_handle_dtmf_event (GstDTMFSrc * dtmfsrc, GstEvent * event)
{
const GstStructure *event_structure;
GstStateChangeReturn sret;
GstState state;
gint event_type;
gboolean start;
gint method;
GstClockTime last_stop;
gint event_number;
gint event_volume;
gboolean correct_order;
sret = gst_element_get_state (GST_ELEMENT (dtmfsrc), &state, NULL, 0);
if (sret != GST_STATE_CHANGE_SUCCESS || state != GST_STATE_PLAYING) {
GST_DEBUG_OBJECT (dtmfsrc, "dtmf-event, but not in PLAYING state");
goto failure;
}
event_structure = gst_event_get_structure (event);
if (!gst_structure_get_int (event_structure, "type", &event_type) ||
!gst_structure_get_boolean (event_structure, "start", &start) ||
(start == TRUE && event_type != GST_TONE_DTMF_TYPE_EVENT))
goto failure;
if (gst_structure_get_int (event_structure, "method", &method)) {
if (method != 2) {
goto failure;
}
}
if (start)
if (!gst_structure_get_int (event_structure, "number", &event_number) ||
!gst_structure_get_int (event_structure, "volume", &event_volume))
goto failure;
GST_OBJECT_LOCK (dtmfsrc);
if (gst_structure_get_clock_time (event_structure, "last-stop", &last_stop))
dtmfsrc->last_stop = last_stop;
else
dtmfsrc->last_stop = GST_CLOCK_TIME_NONE;
correct_order = (start != dtmfsrc->last_event_was_start);
dtmfsrc->last_event_was_start = start;
GST_OBJECT_UNLOCK (dtmfsrc);
if (!correct_order)
goto failure;
if (start) {
GST_DEBUG_OBJECT (dtmfsrc, "Received start event %d with volume %d",
event_number, event_volume);
gst_dtmf_src_add_start_event (dtmfsrc, event_number, event_volume);
}
else {
GST_DEBUG_OBJECT (dtmfsrc, "Received stop event");
gst_dtmf_src_add_stop_event (dtmfsrc);
}
return TRUE;
failure:
return FALSE;
}
static gboolean
gst_dtmf_src_handle_event (GstBaseSrc * src, GstEvent * event)
{
GstDTMFSrc *dtmfsrc;
gboolean result = FALSE;
dtmfsrc = GST_DTMF_SRC (src);
GST_LOG_OBJECT (dtmfsrc, "Received an %s event on the src pad",
GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CUSTOM_UPSTREAM:
if (gst_event_has_name (event, "dtmf-event")) {
result = gst_dtmf_src_handle_dtmf_event (dtmfsrc, event);
break;
}
/* fall through */
default:
result = GST_BASE_SRC_CLASS (parent_class)->event (src, event);
break;
}
return result;
}
static gboolean
gst_dtmf_src_send_event (GstElement * element, GstEvent * event)
{
GstDTMFSrc *dtmfsrc = GST_DTMF_SRC (element);
gboolean ret;
GST_LOG_OBJECT (dtmfsrc, "Received an %s event via send_event",
GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CUSTOM_BOTH:
case GST_EVENT_CUSTOM_BOTH_OOB:
case GST_EVENT_CUSTOM_UPSTREAM:
case GST_EVENT_CUSTOM_DOWNSTREAM:
case GST_EVENT_CUSTOM_DOWNSTREAM_OOB:
if (gst_event_has_name (event, "dtmf-event")) {
ret = gst_dtmf_src_handle_dtmf_event (dtmfsrc, event);
break;
}
/* fall through */
default:
ret = GST_ELEMENT_CLASS (parent_class)->send_event (element, event);
break;
}
return ret;
}
static void
gst_dtmf_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstDTMFSrc *dtmfsrc;
dtmfsrc = GST_DTMF_SRC (object);
switch (prop_id) {
case PROP_INTERVAL:
dtmfsrc->interval = g_value_get_uint (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_dtmf_src_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstDTMFSrc *dtmfsrc;
dtmfsrc = GST_DTMF_SRC (object);
switch (prop_id) {
case PROP_INTERVAL:
g_value_set_uint (value, dtmfsrc->interval);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_dtmf_prepare_timestamps (GstDTMFSrc * dtmfsrc)
{
GstClockTime last_stop;
GstClockTime timestamp;
GST_OBJECT_LOCK (dtmfsrc);
last_stop = dtmfsrc->last_stop;
GST_OBJECT_UNLOCK (dtmfsrc);
if (GST_CLOCK_TIME_IS_VALID (last_stop)) {
timestamp = last_stop;
} else {
GstClock *clock;
/* If there is no valid start time, lets use now as the start time */
clock = gst_element_get_clock (GST_ELEMENT (dtmfsrc));
if (clock != NULL) {
timestamp = gst_clock_get_time (clock)
- gst_element_get_base_time (GST_ELEMENT (dtmfsrc));
gst_object_unref (clock);
} else {
gchar *dtmf_name = gst_element_get_name (dtmfsrc);
GST_ERROR_OBJECT (dtmfsrc, "No clock set for element %s", dtmf_name);
dtmfsrc->timestamp = GST_CLOCK_TIME_NONE;
g_free (dtmf_name);
return;
}
}
/* Make sure the timestamp always goes forward */
if (timestamp > dtmfsrc->timestamp)
dtmfsrc->timestamp = timestamp;
}
static void
gst_dtmf_src_add_start_event (GstDTMFSrc * dtmfsrc, gint event_number,
gint event_volume)
{
GstDTMFSrcEvent *event = g_slice_new0 (GstDTMFSrcEvent);
event->event_type = DTMF_EVENT_TYPE_START;
event->sample = 0;
event->event_number = CLAMP (event_number, MIN_EVENT, MAX_EVENT);
event->volume = CLAMP (event_volume, MIN_VOLUME, MAX_VOLUME);
g_async_queue_push (dtmfsrc->event_queue, event);
}
static void
gst_dtmf_src_add_stop_event (GstDTMFSrc * dtmfsrc)
{
GstDTMFSrcEvent *event = g_slice_new0 (GstDTMFSrcEvent);
event->event_type = DTMF_EVENT_TYPE_STOP;
event->sample = 0;
event->event_number = 0;
event->volume = 0;
g_async_queue_push (dtmfsrc->event_queue, event);
}
static GstBuffer *
gst_dtmf_src_generate_silence (float duration, gint sample_rate)
{
gint buf_size;
/* Create a buffer with data set to 0 */
buf_size = ((duration / 1000) * sample_rate * SAMPLE_SIZE * CHANNELS) / 8;
return gst_buffer_new_wrapped (g_malloc0 (buf_size), buf_size);
}
static GstBuffer *
gst_dtmf_src_generate_tone (GstDTMFSrcEvent * event, DTMF_KEY key,
float duration, gint sample_rate)
{
GstBuffer *buffer;
GstMapInfo map;
gint16 *p;
gint tone_size;
double i = 0;
double amplitude, f1, f2;
double volume_factor;
static GstAllocationParams params = { 0, 1, 0, 0, };
/* Create a buffer for the tone */
tone_size = ((duration / 1000) * sample_rate * SAMPLE_SIZE * CHANNELS) / 8;
buffer = gst_buffer_new_allocate (NULL, tone_size, &params);
gst_buffer_map (buffer, &map, GST_MAP_READWRITE);
p = (gint16 *) map.data;
volume_factor = pow (10, (-event->volume) / 20);
/*
* For each sample point we calculate 'x' as the
* the amplitude value.
*/
for (i = 0; i < (tone_size / (SAMPLE_SIZE / 8)); i++) {
/*
* We add the fundamental frequencies together.
*/
f1 = sin (2 * M_PI * key.low_frequency * (event->sample / sample_rate));
f2 = sin (2 * M_PI * key.high_frequency * (event->sample / sample_rate));
amplitude = (f1 + f2) / 2;
/* Adjust the volume */
amplitude *= volume_factor;
/* Make the [-1:1] interval into a [-32767:32767] interval */
amplitude *= 32767;
/* Store it in the data buffer */
*(p++) = (gint16) amplitude;
(event->sample)++;
}
gst_buffer_unmap (buffer, &map);
return buffer;
}
static GstBuffer *
gst_dtmf_src_create_next_tone_packet (GstDTMFSrc * dtmfsrc,
GstDTMFSrcEvent * event)
{
GstBuffer *buf = NULL;
gboolean send_silence = FALSE;
GST_LOG_OBJECT (dtmfsrc, "Creating buffer for tone %s",
DTMF_KEYS[event->event_number].event_name);
if (event->packet_count * dtmfsrc->interval < MIN_INTER_DIGIT_INTERVAL) {
send_silence = TRUE;
}
if (send_silence) {
GST_LOG_OBJECT (dtmfsrc, "Generating silence");
buf = gst_dtmf_src_generate_silence (dtmfsrc->interval,
dtmfsrc->sample_rate);
} else {
GST_LOG_OBJECT (dtmfsrc, "Generating tone");
buf = gst_dtmf_src_generate_tone (event, DTMF_KEYS[event->event_number],
dtmfsrc->interval, dtmfsrc->sample_rate);
}
event->packet_count++;
/* timestamp and duration of GstBuffer */
GST_BUFFER_DURATION (buf) = dtmfsrc->interval * GST_MSECOND;
GST_BUFFER_TIMESTAMP (buf) = dtmfsrc->timestamp;
GST_LOG_OBJECT (dtmfsrc, "Creating new buffer with event %u duration "
" gst: %" GST_TIME_FORMAT " at %" GST_TIME_FORMAT,
event->event_number, GST_TIME_ARGS (GST_BUFFER_DURATION (buf)),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
dtmfsrc->timestamp += GST_BUFFER_DURATION (buf);
return buf;
}
static void
gst_dtmf_src_post_message (GstDTMFSrc * dtmfsrc, const gchar * message_name,
GstDTMFSrcEvent * event)
{
GstStructure *s = NULL;
switch (event->event_type) {
case DTMF_EVENT_TYPE_START:
s = gst_structure_new (message_name,
"type", G_TYPE_INT, 1,
"method", G_TYPE_INT, 2,
"start", G_TYPE_BOOLEAN, TRUE,
"number", G_TYPE_INT, event->event_number,
"volume", G_TYPE_INT, event->volume, NULL);
break;
case DTMF_EVENT_TYPE_STOP:
s = gst_structure_new (message_name,
"type", G_TYPE_INT, 1, "method", G_TYPE_INT, 2,
"start", G_TYPE_BOOLEAN, FALSE, NULL);
break;
case DTMF_EVENT_TYPE_PAUSE_TASK:
return;
}
if (s)
gst_element_post_message (GST_ELEMENT (dtmfsrc),
gst_message_new_element (GST_OBJECT (dtmfsrc), s));
}
static GstFlowReturn
gst_dtmf_src_create (GstBaseSrc * basesrc, guint64 offset,
guint length, GstBuffer ** buffer)
{
GstBuffer *buf = NULL;
GstDTMFSrcEvent *event;
GstDTMFSrc *dtmfsrc;
GstClock *clock;
GstClockID *clockid;
GstClockReturn clockret;
dtmfsrc = GST_DTMF_SRC (basesrc);
do {
if (dtmfsrc->last_event == NULL) {
GST_DEBUG_OBJECT (dtmfsrc, "popping");
event = g_async_queue_pop (dtmfsrc->event_queue);
GST_DEBUG_OBJECT (dtmfsrc, "popped %d", event->event_type);
switch (event->event_type) {
case DTMF_EVENT_TYPE_STOP:
GST_WARNING_OBJECT (dtmfsrc,
"Received a DTMF stop event when already stopped");
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
break;
case DTMF_EVENT_TYPE_START:
gst_dtmf_prepare_timestamps (dtmfsrc);
event->packet_count = 0;
dtmfsrc->last_event = event;
event = NULL;
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-processed",
dtmfsrc->last_event);
break;
case DTMF_EVENT_TYPE_PAUSE_TASK:
/*
* We're pushing it back because it has to stay in there until
* the task is really paused (and the queue will then be flushed)
*/
GST_DEBUG_OBJECT (dtmfsrc, "pushing pause_task...");
GST_OBJECT_LOCK (dtmfsrc);
if (dtmfsrc->paused) {
g_async_queue_push (dtmfsrc->event_queue, event);
goto paused_locked;
}
GST_OBJECT_UNLOCK (dtmfsrc);
break;
}
if (event)
g_slice_free (GstDTMFSrcEvent, event);
} else if (dtmfsrc->last_event->packet_count * dtmfsrc->interval >=
MIN_DUTY_CYCLE) {
event = g_async_queue_try_pop (dtmfsrc->event_queue);
if (event != NULL) {
switch (event->event_type) {
case DTMF_EVENT_TYPE_START:
GST_WARNING_OBJECT (dtmfsrc,
"Received two consecutive DTMF start events");
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
break;
case DTMF_EVENT_TYPE_STOP:
g_slice_free (GstDTMFSrcEvent, dtmfsrc->last_event);
dtmfsrc->last_event = NULL;
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-processed", event);
break;
case DTMF_EVENT_TYPE_PAUSE_TASK:
/*
* We're pushing it back because it has to stay in there until
* the task is really paused (and the queue will then be flushed)
*/
GST_DEBUG_OBJECT (dtmfsrc, "pushing pause_task...");
GST_OBJECT_LOCK (dtmfsrc);
if (dtmfsrc->paused) {
g_async_queue_push (dtmfsrc->event_queue, event);
goto paused_locked;
}
GST_OBJECT_UNLOCK (dtmfsrc);
break;
}
g_slice_free (GstDTMFSrcEvent, event);
}
}
} while (dtmfsrc->last_event == NULL);
GST_LOG_OBJECT (dtmfsrc, "end event check, now wait for the proper time");
clock = gst_element_get_clock (GST_ELEMENT (basesrc));
clockid = gst_clock_new_single_shot_id (clock, dtmfsrc->timestamp +
gst_element_get_base_time (GST_ELEMENT (dtmfsrc)));
gst_object_unref (clock);
GST_OBJECT_LOCK (dtmfsrc);
if (!dtmfsrc->paused) {
dtmfsrc->clockid = clockid;
GST_OBJECT_UNLOCK (dtmfsrc);
clockret = gst_clock_id_wait (clockid, NULL);
GST_OBJECT_LOCK (dtmfsrc);
if (dtmfsrc->paused)
clockret = GST_CLOCK_UNSCHEDULED;
} else {
clockret = GST_CLOCK_UNSCHEDULED;
}
gst_clock_id_unref (clockid);
dtmfsrc->clockid = NULL;
GST_OBJECT_UNLOCK (dtmfsrc);
if (clockret == GST_CLOCK_UNSCHEDULED) {
goto paused;
}
buf = gst_dtmf_src_create_next_tone_packet (dtmfsrc, dtmfsrc->last_event);
GST_LOG_OBJECT (dtmfsrc, "Created buffer of size %" G_GSIZE_FORMAT,
gst_buffer_get_size (buf));
*buffer = buf;
return GST_FLOW_OK;
paused_locked:
GST_OBJECT_UNLOCK (dtmfsrc);
paused:
if (dtmfsrc->last_event) {
GST_DEBUG_OBJECT (dtmfsrc, "Stopping current event");
/* Don't forget to release the stream lock */
g_slice_free (GstDTMFSrcEvent, dtmfsrc->last_event);
dtmfsrc->last_event = NULL;
}
return GST_FLOW_FLUSHING;
}
static gboolean
gst_dtmf_src_unlock (GstBaseSrc * src)
{
GstDTMFSrc *dtmfsrc = GST_DTMF_SRC (src);
GstDTMFSrcEvent *event = NULL;
GST_DEBUG_OBJECT (dtmfsrc, "Called unlock");
GST_OBJECT_LOCK (dtmfsrc);
dtmfsrc->paused = TRUE;
if (dtmfsrc->clockid) {
gst_clock_id_unschedule (dtmfsrc->clockid);
}
GST_OBJECT_UNLOCK (dtmfsrc);
GST_DEBUG_OBJECT (dtmfsrc, "Pushing the PAUSE_TASK event on unlock request");
event = g_slice_new0 (GstDTMFSrcEvent);
event->event_type = DTMF_EVENT_TYPE_PAUSE_TASK;
g_async_queue_push (dtmfsrc->event_queue, event);
return TRUE;
}
static gboolean
gst_dtmf_src_unlock_stop (GstBaseSrc * src)
{
GstDTMFSrc *dtmfsrc = GST_DTMF_SRC (src);
GST_DEBUG_OBJECT (dtmfsrc, "Unlock stopped");
GST_OBJECT_LOCK (dtmfsrc);
dtmfsrc->paused = FALSE;
GST_OBJECT_UNLOCK (dtmfsrc);
return TRUE;
}
static gboolean
gst_dtmf_src_negotiate (GstBaseSrc * basesrc)
{
GstDTMFSrc *dtmfsrc = GST_DTMF_SRC (basesrc);
GstCaps *caps;
GstStructure *s;
gboolean ret;
caps = gst_pad_get_allowed_caps (GST_BASE_SRC_PAD (basesrc));
if (!caps)
caps = gst_pad_get_pad_template_caps (GST_BASE_SRC_PAD (basesrc));
if (gst_caps_is_empty (caps)) {
gst_caps_unref (caps);
return FALSE;
}
caps = gst_caps_truncate (caps);
caps = gst_caps_make_writable (caps);
s = gst_caps_get_structure (caps, 0);
gst_structure_fixate_field_nearest_int (s, "rate", DEFAULT_SAMPLE_RATE);
if (!gst_structure_get_int (s, "rate", &dtmfsrc->sample_rate)) {
GST_ERROR_OBJECT (dtmfsrc, "Could not get rate");
gst_caps_unref (caps);
return FALSE;
}
ret = gst_pad_set_caps (GST_BASE_SRC_PAD (basesrc), caps);
gst_caps_unref (caps);
return ret;
}
static GstStateChangeReturn
gst_dtmf_src_change_state (GstElement * element, GstStateChange transition)
{
GstDTMFSrc *dtmfsrc;
GstStateChangeReturn result;
gboolean no_preroll = FALSE;
GstDTMFSrcEvent *event = NULL;
dtmfsrc = GST_DTMF_SRC (element);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
/* Flushing the event queue */
event = g_async_queue_try_pop (dtmfsrc->event_queue);
while (event != NULL) {
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
g_slice_free (GstDTMFSrcEvent, event);
event = g_async_queue_try_pop (dtmfsrc->event_queue);
}
dtmfsrc->last_event_was_start = FALSE;
dtmfsrc->timestamp = 0;
no_preroll = TRUE;
break;
default:
break;
}
if ((result =
GST_ELEMENT_CLASS (gst_dtmf_src_parent_class)->change_state (element,
transition)) == GST_STATE_CHANGE_FAILURE)
goto failure;
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
no_preroll = TRUE;
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
GST_DEBUG_OBJECT (dtmfsrc, "Flushing event queue");
/* Flushing the event queue */
event = g_async_queue_try_pop (dtmfsrc->event_queue);
while (event != NULL) {
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
g_slice_free (GstDTMFSrcEvent, event);
event = g_async_queue_try_pop (dtmfsrc->event_queue);
}
dtmfsrc->last_event_was_start = FALSE;
break;
default:
break;
}
if (no_preroll && result == GST_STATE_CHANGE_SUCCESS)
result = GST_STATE_CHANGE_NO_PREROLL;
return result;
/* ERRORS */
failure:
{
GST_ERROR_OBJECT (dtmfsrc, "parent failed state change");
return result;
}
}
gboolean
gst_dtmf_src_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "dtmfsrc",
GST_RANK_NONE, GST_TYPE_DTMF_SRC);
}

101
gst/dtmf/gstdtmfsrc.h
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@ -1,101 +0,0 @@
/* GStreamer DTMF source
*
* gstdtmfsrc.h:
*
* Copyright (C) <2007> Collabora.
* Contact: Youness Alaoui <youness.alaoui@collabora.co.uk>
* Copyright (C) <2007> Nokia Corporation.
* Contact: Zeeshan Ali <zeeshan.ali@nokia.com>
* Copyright (C) <2005> Wim Taymans <wim@fluendo.com>
*
* 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_DTMF_SRC_H__
#define __GST_DTMF_SRC_H__
#include <gst/gst.h>
#include <gst/gstbuffer.h>
#include <gst/base/gstbasesrc.h>
G_BEGIN_DECLS
#define GST_TYPE_DTMF_SRC (gst_dtmf_src_get_type())
#define GST_DTMF_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_DTMF_SRC,GstDTMFSrc))
#define GST_DTMF_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_DTMF_SRC,GstDTMFSrcClass))
#define GST_DTMF_SRC_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_DTMF_SRC, GstDTMFSrcClass))
#define GST_IS_DTMF_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_DTMF_SRC))
#define GST_IS_DTMF_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_DTMF_SRC))
#define GST_DTMF_SRC_CAST(obj) ((GstDTMFSrc *)(obj))
typedef struct _GstDTMFSrc GstDTMFSrc;
typedef struct _GstDTMFSrcClass GstDTMFSrcClass;
enum _GstDTMFEventType
{
DTMF_EVENT_TYPE_START,
DTMF_EVENT_TYPE_STOP,
DTMF_EVENT_TYPE_PAUSE_TASK
};
typedef enum _GstDTMFEventType GstDTMFEventType;
struct _GstDTMFSrcEvent
{
GstDTMFEventType event_type;
double sample;
guint16 event_number;
guint16 volume;
guint32 packet_count;
};
typedef struct _GstDTMFSrcEvent GstDTMFSrcEvent;
/**
* GstDTMFSrc:
* @element: the parent element.
*
* The opaque #GstDTMFSrc data structure.
*/
struct _GstDTMFSrc
{
/*< private >*/
GstBaseSrc parent;
GAsyncQueue *event_queue;
GstDTMFSrcEvent *last_event;
gboolean last_event_was_start;
guint16 interval;
GstClockTime timestamp;
gboolean paused;
GstClockID clockid;
GstClockTime last_stop;
gint sample_rate;
};
struct _GstDTMFSrcClass
{
GstBaseSrcClass parent_class;
};
GType gst_dtmf_src_get_type (void);
gboolean gst_dtmf_src_plugin_init (GstPlugin * plugin);
G_END_DECLS
#endif /* __GST_DTMF_SRC_H__ */

543
gst/dtmf/gstrtpdtmfdepay.c
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@ -1,543 +0,0 @@
/* GstRtpDtmfDepay
*
* Copyright (C) 2008 Collabora Limited
* Copyright (C) 2008 Nokia Corporation
* Contact: Youness Alaoui <youness.alaoui@collabora.co.uk>
*
* 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.
*/
/**
* SECTION:element-rtpdtmfdepay
* @see_also: rtpdtmfsrc, rtpdtmfmux
*
* This element takes RTP DTMF packets and produces sound. It also emits a
* message on the #GstBus.
*
* The message is called "dtmf-event" and has the following fields
* <informaltable>
* <tgroup cols='4'>
* <colspec colname='Name' />
* <colspec colname='Type' />
* <colspec colname='Possible values' />
* <colspec colname='Purpose' />
* <thead>
* <row>
* <entry>Name</entry>
* <entry>GType</entry>
* <entry>Possible values</entry>
* <entry>Purpose</entry>
* </row>
* </thead>
* <tbody>
* <row>
* <entry>type</entry>
* <entry>G_TYPE_INT</entry>
* <entry>0-1</entry>
* <entry>Which of the two methods
* specified in RFC 2833 to use. The value should be 0 for tones and 1 for
* named events. Tones are specified by their frequencies and events are specied
* by their number. This element currently only recognizes events.
* Do not confuse with "method" which specified the output.
* </entry>
* </row>
* <row>
* <entry>number</entry>
* <entry>G_TYPE_INT</entry>
* <entry>0-16</entry>
* <entry>The event number.</entry>
* </row>
* <row>
* <entry>volume</entry>
* <entry>G_TYPE_INT</entry>
* <entry>0-36</entry>
* <entry>This field describes the power level of the tone, expressed in dBm0
* after dropping the sign. Power levels range from 0 to -63 dBm0. The range of
* valid DTMF is from 0 to -36 dBm0.
* </entry>
* </row>
* <row>
* <entry>method</entry>
* <entry>G_TYPE_INT</entry>
* <entry>1</entry>
* <entry>This field will always been 1 (ie RTP event) from this element.
* </entry>
* </row>
* </tbody>
* </tgroup>
* </informaltable>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstrtpdtmfdepay.h"
#include <string.h>
#include <math.h>
#include <gst/audio/audio.h>
#include <gst/rtp/gstrtpbuffer.h>
#define DEFAULT_PACKET_INTERVAL 50 /* ms */
#define MIN_PACKET_INTERVAL 10 /* ms */
#define MAX_PACKET_INTERVAL 50 /* ms */
#define SAMPLE_RATE 8000
#define SAMPLE_SIZE 16
#define CHANNELS 1
#define MIN_DUTY_CYCLE (MIN_INTER_DIGIT_INTERVAL + MIN_PULSE_DURATION)
#define MIN_UNIT_TIME 0
#define MAX_UNIT_TIME 1000
#define DEFAULT_UNIT_TIME 0
#define DEFAULT_MAX_DURATION 0
typedef struct st_dtmf_key
{
const char *event_name;
int event_encoding;
float low_frequency;
float high_frequency;
} DTMF_KEY;
static const DTMF_KEY DTMF_KEYS[] = {
{"DTMF_KEY_EVENT_0", 0, 941, 1336},
{"DTMF_KEY_EVENT_1", 1, 697, 1209},
{"DTMF_KEY_EVENT_2", 2, 697, 1336},
{"DTMF_KEY_EVENT_3", 3, 697, 1477},
{"DTMF_KEY_EVENT_4", 4, 770, 1209},
{"DTMF_KEY_EVENT_5", 5, 770, 1336},
{"DTMF_KEY_EVENT_6", 6, 770, 1477},
{"DTMF_KEY_EVENT_7", 7, 852, 1209},
{"DTMF_KEY_EVENT_8", 8, 852, 1336},
{"DTMF_KEY_EVENT_9", 9, 852, 1477},
{"DTMF_KEY_EVENT_S", 10, 941, 1209},
{"DTMF_KEY_EVENT_P", 11, 941, 1477},
{"DTMF_KEY_EVENT_A", 12, 697, 1633},
{"DTMF_KEY_EVENT_B", 13, 770, 1633},
{"DTMF_KEY_EVENT_C", 14, 852, 1633},
{"DTMF_KEY_EVENT_D", 15, 941, 1633},
};
#define MAX_DTMF_EVENTS 16
enum
{
DTMF_KEY_EVENT_1 = 1,
DTMF_KEY_EVENT_2 = 2,
DTMF_KEY_EVENT_3 = 3,
DTMF_KEY_EVENT_4 = 4,
DTMF_KEY_EVENT_5 = 5,
DTMF_KEY_EVENT_6 = 6,
DTMF_KEY_EVENT_7 = 7,
DTMF_KEY_EVENT_8 = 8,
DTMF_KEY_EVENT_9 = 9,
DTMF_KEY_EVENT_0 = 0,
DTMF_KEY_EVENT_STAR = 10,
DTMF_KEY_EVENT_POUND = 11,
DTMF_KEY_EVENT_A = 12,
DTMF_KEY_EVENT_B = 13,
DTMF_KEY_EVENT_C = 14,
DTMF_KEY_EVENT_D = 15,
};
GST_DEBUG_CATEGORY_STATIC (gst_rtp_dtmf_depay_debug);
#define GST_CAT_DEFAULT gst_rtp_dtmf_depay_debug
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_UNIT_TIME,
PROP_MAX_DURATION
};
enum
{
ARG_0
};
static GstStaticPadTemplate gst_rtp_dtmf_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) \"" GST_AUDIO_NE (S16) "\", "
"rate = " GST_AUDIO_RATE_RANGE ", " "channels = (int) 1")
);
static GstStaticPadTemplate gst_rtp_dtmf_depay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) [ 0, MAX ], "
"encoding-name = (string) \"TELEPHONE-EVENT\"")
);
G_DEFINE_TYPE (GstRtpDTMFDepay, gst_rtp_dtmf_depay,
GST_TYPE_RTP_BASE_DEPAYLOAD);
static void gst_rtp_dtmf_depay_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rtp_dtmf_depay_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstBuffer *gst_rtp_dtmf_depay_process (GstRTPBaseDepayload * depayload,
GstBuffer * buf);
gboolean gst_rtp_dtmf_depay_setcaps (GstRTPBaseDepayload * filter,
GstCaps * caps);
static void
gst_rtp_dtmf_depay_class_init (GstRtpDTMFDepayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
gobject_class = G_OBJECT_CLASS (klass);
gstelement_class = GST_ELEMENT_CLASS (klass);
gstrtpbasedepayload_class = GST_RTP_BASE_DEPAYLOAD_CLASS (klass);
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_dtmf_depay_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_dtmf_depay_sink_template));
GST_DEBUG_CATEGORY_INIT (gst_rtp_dtmf_depay_debug,
"rtpdtmfdepay", 0, "rtpdtmfdepay element");
gst_element_class_set_static_metadata (gstelement_class,
"RTP DTMF packet depayloader", "Codec/Depayloader/Network",
"Generates DTMF Sound from telephone-event RTP packets",
"Youness Alaoui <youness.alaoui@collabora.co.uk>");
gobject_class->set_property =
GST_DEBUG_FUNCPTR (gst_rtp_dtmf_depay_set_property);
gobject_class->get_property =
GST_DEBUG_FUNCPTR (gst_rtp_dtmf_depay_get_property);
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_UNIT_TIME,
g_param_spec_uint ("unit-time", "Duration unittime",
"The smallest unit (ms) the duration must be a multiple of (0 disables it)",
MIN_UNIT_TIME, MAX_UNIT_TIME, DEFAULT_UNIT_TIME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_MAX_DURATION,
g_param_spec_uint ("max-duration", "Maximum duration",
"The maxumimum duration (ms) of the outgoing soundpacket. "
"(0 = no limit)", 0, G_MAXUINT, DEFAULT_MAX_DURATION,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gstrtpbasedepayload_class->process =
GST_DEBUG_FUNCPTR (gst_rtp_dtmf_depay_process);
gstrtpbasedepayload_class->set_caps =
GST_DEBUG_FUNCPTR (gst_rtp_dtmf_depay_setcaps);
}
static void
gst_rtp_dtmf_depay_init (GstRtpDTMFDepay * rtpdtmfdepay)
{
rtpdtmfdepay->unit_time = DEFAULT_UNIT_TIME;
}
static void
gst_rtp_dtmf_depay_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRtpDTMFDepay *rtpdtmfdepay;
rtpdtmfdepay = GST_RTP_DTMF_DEPAY (object);
switch (prop_id) {
case PROP_UNIT_TIME:
rtpdtmfdepay->unit_time = g_value_get_uint (value);
break;
case PROP_MAX_DURATION:
rtpdtmfdepay->max_duration = g_value_get_uint (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_dtmf_depay_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRtpDTMFDepay *rtpdtmfdepay;
rtpdtmfdepay = GST_RTP_DTMF_DEPAY (object);
switch (prop_id) {
case PROP_UNIT_TIME:
g_value_set_uint (value, rtpdtmfdepay->unit_time);
break;
case PROP_MAX_DURATION:
g_value_set_uint (value, rtpdtmfdepay->max_duration);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
gboolean
gst_rtp_dtmf_depay_setcaps (GstRTPBaseDepayload * filter, GstCaps * caps)
{
GstCaps *filtercaps, *srccaps;
GstStructure *structure = gst_caps_get_structure (caps, 0);
gint clock_rate = 8000; /* default */
gst_structure_get_int (structure, "clock-rate", &clock_rate);
filter->clock_rate = clock_rate;
filtercaps =
gst_pad_get_pad_template_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (filter));
filtercaps = gst_caps_make_writable (filtercaps);
gst_caps_set_simple (filtercaps, "rate", G_TYPE_INT, clock_rate, NULL);
srccaps = gst_pad_peer_query_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (filter),
filtercaps);
gst_caps_unref (filtercaps);
gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (filter), srccaps);
gst_caps_unref (srccaps);
return TRUE;
}
static GstBuffer *
gst_dtmf_src_generate_tone (GstRtpDTMFDepay * rtpdtmfdepay,
GstRTPDTMFPayload payload)
{
GstBuffer *buf;
GstMapInfo map;
gint16 *p;
gint tone_size;
double i = 0;
double amplitude, f1, f2;
double volume_factor;
DTMF_KEY key = DTMF_KEYS[payload.event];
guint32 clock_rate = 8000 /* default */ ;
GstRTPBaseDepayload *depayload = GST_RTP_BASE_DEPAYLOAD (rtpdtmfdepay);
gint volume;
static GstAllocationParams params = { 0, 1, 0, 0, };
clock_rate = depayload->clock_rate;
/* Create a buffer for the tone */
tone_size = (payload.duration * SAMPLE_SIZE * CHANNELS) / 8;
buf = gst_buffer_new_allocate (NULL, tone_size, &params);
GST_BUFFER_DURATION (buf) = payload.duration * GST_SECOND / clock_rate;
volume = payload.volume;
gst_buffer_map (buf, &map, GST_MAP_WRITE);
p = (gint16 *) map.data;
volume_factor = pow (10, (-volume) / 20);
/*
* For each sample point we calculate 'x' as the
* the amplitude value.
*/
for (i = 0; i < (tone_size / (SAMPLE_SIZE / 8)); i++) {
/*
* We add the fundamental frequencies together.
*/
f1 = sin (2 * M_PI * key.low_frequency * (rtpdtmfdepay->sample /
clock_rate));
f2 = sin (2 * M_PI * key.high_frequency * (rtpdtmfdepay->sample /
clock_rate));
amplitude = (f1 + f2) / 2;
/* Adjust the volume */
amplitude *= volume_factor;
/* Make the [-1:1] interval into a [-32767:32767] interval */
amplitude *= 32767;
/* Store it in the data buffer */
*(p++) = (gint16) amplitude;
(rtpdtmfdepay->sample)++;
}
gst_buffer_unmap (buf, &map);
return buf;
}
static GstBuffer *
gst_rtp_dtmf_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
{
GstRtpDTMFDepay *rtpdtmfdepay = NULL;
GstBuffer *outbuf = NULL;
gint payload_len;
guint8 *payload = NULL;
guint32 timestamp;
GstRTPDTMFPayload dtmf_payload;
gboolean marker;
GstStructure *structure = NULL;
GstMessage *dtmf_message = NULL;
GstRTPBuffer rtpbuffer = GST_RTP_BUFFER_INIT;
rtpdtmfdepay = GST_RTP_DTMF_DEPAY (depayload);
gst_rtp_buffer_map (buf, GST_MAP_READ, &rtpbuffer);
payload_len = gst_rtp_buffer_get_payload_len (&rtpbuffer);
payload = gst_rtp_buffer_get_payload (&rtpbuffer);
if (payload_len != sizeof (GstRTPDTMFPayload))
goto bad_packet;
memcpy (&dtmf_payload, payload, sizeof (GstRTPDTMFPayload));
if (dtmf_payload.event > MAX_EVENT)
goto bad_packet;
marker = gst_rtp_buffer_get_marker (&rtpbuffer);
timestamp = gst_rtp_buffer_get_timestamp (&rtpbuffer);
dtmf_payload.duration = g_ntohs (dtmf_payload.duration);
/* clip to whole units of unit_time */
if (rtpdtmfdepay->unit_time) {
guint unit_time_clock =
(rtpdtmfdepay->unit_time * depayload->clock_rate) / 1000;
if (dtmf_payload.duration % unit_time_clock) {
/* Make sure we don't overflow the duration */
if (dtmf_payload.duration < G_MAXUINT16 - unit_time_clock)
dtmf_payload.duration += unit_time_clock -
(dtmf_payload.duration % unit_time_clock);
else
dtmf_payload.duration -= dtmf_payload.duration % unit_time_clock;
}
}
/* clip to max duration */
if (rtpdtmfdepay->max_duration) {
guint max_duration_clock =
(rtpdtmfdepay->max_duration * depayload->clock_rate) / 1000;
if (max_duration_clock < G_MAXUINT16 &&
dtmf_payload.duration > max_duration_clock)
dtmf_payload.duration = max_duration_clock;
}
GST_DEBUG_OBJECT (depayload, "Received new RTP DTMF packet : "
"marker=%d - timestamp=%u - event=%d - duration=%d",
marker, timestamp, dtmf_payload.event, dtmf_payload.duration);
GST_DEBUG_OBJECT (depayload,
"Previous information : timestamp=%u - duration=%d",
rtpdtmfdepay->previous_ts, rtpdtmfdepay->previous_duration);
/* First packet */
if (marker || rtpdtmfdepay->previous_ts != timestamp) {
rtpdtmfdepay->sample = 0;
rtpdtmfdepay->previous_ts = timestamp;
rtpdtmfdepay->previous_duration = dtmf_payload.duration;
rtpdtmfdepay->first_gst_ts = GST_BUFFER_PTS (buf);
structure = gst_structure_new ("dtmf-event",
"number", G_TYPE_INT, dtmf_payload.event,
"volume", G_TYPE_INT, dtmf_payload.volume,
"type", G_TYPE_INT, 1, "method", G_TYPE_INT, 1, NULL);
if (structure) {
dtmf_message =
gst_message_new_element (GST_OBJECT (depayload), structure);
if (dtmf_message) {
if (!gst_element_post_message (GST_ELEMENT (depayload), dtmf_message)) {
GST_ERROR_OBJECT (depayload,
"Unable to send dtmf-event message to bus");
}
} else {
GST_ERROR_OBJECT (depayload, "Unable to create dtmf-event message");
}
} else {
GST_ERROR_OBJECT (depayload, "Unable to create dtmf-event structure");
}
} else {
guint16 duration = dtmf_payload.duration;
dtmf_payload.duration -= rtpdtmfdepay->previous_duration;
/* If late buffer, ignore */
if (duration > rtpdtmfdepay->previous_duration)
rtpdtmfdepay->previous_duration = duration;
}
GST_DEBUG_OBJECT (depayload, "new previous duration : %d - new duration : %d"
" - diff : %d - clock rate : %d - timestamp : %" G_GUINT64_FORMAT,
rtpdtmfdepay->previous_duration, dtmf_payload.duration,
(rtpdtmfdepay->previous_duration - dtmf_payload.duration),
depayload->clock_rate, GST_BUFFER_TIMESTAMP (buf));
/* If late or duplicate packet (like the redundant end packet). Ignore */
if (dtmf_payload.duration > 0) {
outbuf = gst_dtmf_src_generate_tone (rtpdtmfdepay, dtmf_payload);
GST_BUFFER_PTS (outbuf) = rtpdtmfdepay->first_gst_ts +
(rtpdtmfdepay->previous_duration - dtmf_payload.duration) *
GST_SECOND / depayload->clock_rate;
GST_BUFFER_OFFSET (outbuf) =
(rtpdtmfdepay->previous_duration - dtmf_payload.duration) *
GST_SECOND / depayload->clock_rate;
GST_BUFFER_OFFSET_END (outbuf) = rtpdtmfdepay->previous_duration *
GST_SECOND / depayload->clock_rate;
GST_DEBUG_OBJECT (depayload,
"timestamp : %" G_GUINT64_FORMAT " - time %" GST_TIME_FORMAT,
GST_BUFFER_TIMESTAMP (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
}
gst_rtp_buffer_unmap (&rtpbuffer);
return outbuf;
bad_packet:
GST_ELEMENT_WARNING (rtpdtmfdepay, STREAM, DECODE,
("Packet did not validate"), (NULL));
if (rtpbuffer.buffer != NULL)
gst_rtp_buffer_unmap (&rtpbuffer);
return NULL;
}
gboolean
gst_rtp_dtmf_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpdtmfdepay",
GST_RANK_MARGINAL, GST_TYPE_RTP_DTMF_DEPAY);
}

68
gst/dtmf/gstrtpdtmfdepay.h
View file

@ -1,68 +0,0 @@
/* GstRtpDtmfDepay
*
* Copyright (C) 2008 Collabora Limited
* Copyright (C) 2008 Nokia Corporation
* Contact: Youness Alaoui <youness.alaoui@collabora.co.uk>
*
* 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_DEPAY_H__
#define __GST_RTP_DTMF_DEPAY_H__
#include <gst/gst.h>
#include <gst/base/gstadapter.h>
#include <gst/rtp/gstrtpbasedepayload.h>
#include "gstdtmfcommon.h"
G_BEGIN_DECLS
#define GST_TYPE_RTP_DTMF_DEPAY \
(gst_rtp_dtmf_depay_get_type())
#define GST_RTP_DTMF_DEPAY(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_DTMF_DEPAY,GstRtpDTMFDepay))
#define GST_RTP_DTMF_DEPAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_DTMF_DEPAY,GstRtpDTMFDepayClass))
#define GST_IS_RTP_DTMF_DEPAY(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_DTMF_DEPAY))
#define GST_IS_RTP_DTMF_DEPAY_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_DTMF_DEPAY))
typedef struct _GstRtpDTMFDepay GstRtpDTMFDepay;
typedef struct _GstRtpDTMFDepayClass GstRtpDTMFDepayClass;
struct _GstRtpDTMFDepay
{
/*< private >*/
GstRTPBaseDepayload depayload;
double sample;
guint32 previous_ts;
guint16 previous_duration;
GstClockTime first_gst_ts;
guint unit_time;
guint max_duration;
};
struct _GstRtpDTMFDepayClass
{
GstRTPBaseDepayloadClass parent_class;
};
GType gst_rtp_dtmf_depay_get_type (void);
gboolean gst_rtp_dtmf_depay_plugin_init (GstPlugin * plugin);
G_END_DECLS
#endif /* __GST_RTP_DTMF_DEPAY_H__ */

1156
gst/dtmf/gstrtpdtmfsrc.c
View file

File diff suppressed because it is too large Load diff

115
gst/dtmf/gstrtpdtmfsrc.h
View file

@ -1,115 +0,0 @@
/* GStreamer RTP DTMF source
*
* gstrtpdtmfsrc.h:
*
* Copyright (C) <2007> Nokia Corporation.
* Contact: Zeeshan Ali <zeeshan.ali@nokia.com>
* Copyright (C) <2005> Wim Taymans <wim@fluendo.com>
*
* 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_SRC_H__
#define __GST_RTP_DTMF_SRC_H__
#include <gst/gst.h>
#include <gst/base/gstbasesrc.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstdtmfcommon.h"
G_BEGIN_DECLS
#define GST_TYPE_RTP_DTMF_SRC (gst_rtp_dtmf_src_get_type())
#define GST_RTP_DTMF_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_DTMF_SRC,GstRTPDTMFSrc))
#define GST_RTP_DTMF_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_DTMF_SRC,GstRTPDTMFSrcClass))
#define GST_RTP_DTMF_SRC_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTP_DTMF_SRC, GstRTPDTMFSrcClass))
#define GST_IS_RTP_DTMF_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_DTMF_SRC))
#define GST_IS_RTP_DTMF_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_DTMF_SRC))
#define GST_RTP_DTMF_SRC_CAST(obj) ((GstRTPDTMFSrc *)(obj))
typedef struct _GstRTPDTMFSrc GstRTPDTMFSrc;
typedef struct _GstRTPDTMFSrcClass GstRTPDTMFSrcClass;
enum _GstRTPDTMFEventType
{
RTP_DTMF_EVENT_TYPE_START,
RTP_DTMF_EVENT_TYPE_STOP,
RTP_DTMF_EVENT_TYPE_PAUSE_TASK
};
typedef enum _GstRTPDTMFEventType GstRTPDTMFEventType;
struct _GstRTPDTMFSrcEvent
{
GstRTPDTMFEventType event_type;
GstRTPDTMFPayload *payload;
};
typedef struct _GstRTPDTMFSrcEvent GstRTPDTMFSrcEvent;
/**
* GstRTPDTMFSrc:
* @element: the parent element.
*
* The opaque #GstRTPDTMFSrc data structure.
*/
struct _GstRTPDTMFSrc
{
/*< private >*/
GstBaseSrc basesrc;
GAsyncQueue *event_queue;
GstClockID clockid;
gboolean paused;
GstRTPDTMFPayload *payload;
GstClockTime timestamp;
GstClockTime start_timestamp;
gboolean first_packet;
gboolean last_packet;
guint32 ts_base;
guint16 seqnum_base;
gint16 seqnum_offset;
guint16 seqnum;
gint32 ts_offset;
guint32 rtp_timestamp;
guint pt;
guint ssrc;
guint current_ssrc;
guint16 ptime;
guint16 packet_redundancy;
guint32 clock_rate;
gboolean last_event_was_start;
GstClockTime last_stop;
gboolean dirty;
guint16 redundancy_count;
};
struct _GstRTPDTMFSrcClass
{
GstBaseSrcClass parent_class;
};
GType gst_rtp_dtmf_src_get_type (void);
gboolean gst_rtp_dtmf_src_plugin_init (GstPlugin * plugin);
G_END_DECLS
#endif /* __GST_RTP_DTMF_SRC_H__ */

5
tests/check/Makefile.am
View file

@ -207,7 +207,6 @@ check_PROGRAMS = \
elements/baseaudiovisualizer \ elements/baseaudiovisualizer \
elements/camerabin \ elements/camerabin \
elements/dataurisrc \ elements/dataurisrc \
elements/dtmf \
elements/gdppay \ elements/gdppay \
elements/gdpdepay \ elements/gdpdepay \
$(check_jifmux) \ $(check_jifmux) \
@ -338,10 +337,6 @@ elements_camerabin_LDADD = \
$(GST_BASE_LIBS) $(GST_LIBS) $(LDADD) $(GST_BASE_LIBS) $(GST_LIBS) $(LDADD)
elements_camerabin_SOURCES = elements/camerabin.c elements_camerabin_SOURCES = elements/camerabin.c
elements_dtmf_CFLAGS = $(GST_PLUGINS_BAD_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS) \
$(GST_BASE_CFLAGS) $(GST_CFLAGS) $(AM_CFLAGS)
elements_dtmf_LDADD = $(GST_BASE_LIBS) $(LDADD) -lgstrtp-@GST_API_VERSION@
elements_jifmux_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(EXIF_CFLAGS) $(AM_CFLAGS) elements_jifmux_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(EXIF_CFLAGS) $(AM_CFLAGS)
elements_jifmux_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgsttag-$(GST_API_VERSION) $(GST_CHECK_LIBS) $(EXIF_LIBS) $(LDADD) elements_jifmux_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgsttag-$(GST_API_VERSION) $(GST_CHECK_LIBS) $(EXIF_LIBS) $(LDADD)
elements_jifmux_SOURCES = elements/jifmux.c elements_jifmux_SOURCES = elements/jifmux.c

1
tests/check/elements/.gitignore vendored
View file

@ -12,7 +12,6 @@ curlhttpsink
curlsmtpsink curlsmtpsink
deinterleave deinterleave
dataurisrc dataurisrc
dtmf
faac faac
faad faad
gdpdepay gdpdepay

588
tests/check/elements/dtmf.c
View file

@ -1,588 +0,0 @@
/* GStreamer
*
* unit test for dtmf elements
* Copyright (C) 2013 Collabora Ltd
* @author: Olivier Crete <olivier.crete@collabora.com>
*
* 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"
#endif
#include <gst/gst.h>
#include <gst/audio/audio.h>
#include <gst/check/gstcheck.h>
#include <gst/check/gsttestclock.h>
#include <gst/rtp/gstrtpbuffer.h>
/* Include this from the plugin to get the defines */
#include "../../gst/dtmf/gstdtmfcommon.h"
#define END_BIT (1<<7)
static GstStaticPadTemplate audio_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) \"" GST_AUDIO_NE (S16) "\", "
"rate = " GST_AUDIO_RATE_RANGE ", " "channels = (int) 1")
);
static GstStaticPadTemplate rtp_dtmf_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) [ 0, MAX ], "
"encoding-name = (string) \"TELEPHONE-EVENT\"")
);
static void
check_get_dtmf_event_message (GstBus * bus, gint number, gint volume)
{
GstMessage *message;
gboolean have_message = FALSE;
while (!have_message &&
(message = gst_bus_pop_filtered (bus, GST_MESSAGE_ELEMENT)) != NULL) {
if (gst_message_has_name (message, "dtmf-event")) {
const GstStructure *s = gst_message_get_structure (message);
gint stype, snumber, smethod, svolume;
fail_unless (gst_structure_get (s,
"type", G_TYPE_INT, &stype,
"number", G_TYPE_INT, &snumber,
"method", G_TYPE_INT, &smethod,
"volume", G_TYPE_INT, &svolume, NULL));
fail_unless (stype == 1);
fail_unless (smethod == 1);
fail_unless (snumber == number);
fail_unless (svolume == volume);
have_message = TRUE;
}
gst_message_unref (message);
}
fail_unless (have_message);
}
static void
check_no_dtmf_event_message (GstBus * bus)
{
GstMessage *message;
gboolean have_message = FALSE;
while (!have_message &&
(message = gst_bus_pop_filtered (bus, GST_MESSAGE_ELEMENT)) != NULL) {
if (gst_message_has_name (message, "dtmf-event") ||
gst_message_has_name (message, "dtmf-event-processed") ||
gst_message_has_name (message, "dtmf-event-dropped")) {
have_message = TRUE;
}
gst_message_unref (message);
}
fail_unless (!have_message);
}
static void
check_buffers_duration (GstClockTime expected_duration)
{
GstClockTime duration = 0;
while (buffers) {
GstBuffer *buf = buffers->data;
buffers = g_list_delete_link (buffers, buffers);
fail_unless (GST_BUFFER_DURATION_IS_VALID (buf));
duration += GST_BUFFER_DURATION (buf);
gst_buffer_unref (buf);
}
fail_unless (duration == expected_duration);
}
static void
send_rtp_packet (GstPad * src, guint timestamp, gboolean marker, gboolean end,
guint number, guint volume, guint duration)
{
GstBuffer *buf;
GstRTPBuffer rtpbuf = GST_RTP_BUFFER_INIT;
gchar *payload;
static guint seqnum = 1;
buf = gst_rtp_buffer_new_allocate (4, 0, 0);
fail_unless (gst_rtp_buffer_map (buf, GST_MAP_READWRITE, &rtpbuf));
gst_rtp_buffer_set_seq (&rtpbuf, seqnum++);
gst_rtp_buffer_set_timestamp (&rtpbuf, timestamp);
gst_rtp_buffer_set_marker (&rtpbuf, marker);
payload = gst_rtp_buffer_get_payload (&rtpbuf);
payload[0] = number;
payload[1] = volume | (end ? END_BIT : 0);
GST_WRITE_UINT16_BE (payload + 2, duration);
gst_rtp_buffer_unmap (&rtpbuf);
fail_unless (gst_pad_push (src, buf) == GST_FLOW_OK);
}
GST_START_TEST (test_rtpdtmfdepay)
{
GstElement *dtmfdepay;
GstPad *src, *sink;
GstBus *bus;
GstCaps *caps_in;
GstCaps *expected_caps_out;
GstCaps *caps_out;
dtmfdepay = gst_check_setup_element ("rtpdtmfdepay");
sink = gst_check_setup_sink_pad (dtmfdepay, &audio_sink_template);
src = gst_check_setup_src_pad (dtmfdepay, &rtp_dtmf_src_template);
bus = gst_bus_new ();
gst_element_set_bus (dtmfdepay, bus);
gst_pad_set_active (src, TRUE);
gst_pad_set_active (sink, TRUE);
gst_element_set_state (dtmfdepay, GST_STATE_PLAYING);
caps_in = gst_caps_new_simple ("application/x-rtp",
"encoding-name", G_TYPE_STRING, "TELEPHONE-EVENT",
"media", G_TYPE_STRING, "audio",
"clock-rate", G_TYPE_INT, 1000, "payload", G_TYPE_INT, 99, NULL);
fail_unless (gst_pad_set_caps (src, caps_in));
gst_caps_unref (caps_in);
caps_out = gst_pad_get_current_caps (sink);
expected_caps_out = gst_caps_new_simple ("audio/x-raw",
"format", G_TYPE_STRING, GST_AUDIO_NE (S16),
"rate", G_TYPE_INT, 1000, "channels", G_TYPE_INT, 1, NULL);
fail_unless (gst_caps_is_equal_fixed (caps_out, expected_caps_out));
gst_caps_unref (expected_caps_out);
gst_caps_unref (caps_out);
/* Single packet DTMF */
send_rtp_packet (src, 200, TRUE, TRUE, 1, 5, 250);
check_get_dtmf_event_message (bus, 1, 5);
check_buffers_duration (250 * GST_MSECOND);
/* Two packet DTMF */
send_rtp_packet (src, 800, TRUE, FALSE, 1, 5, 200);
send_rtp_packet (src, 800, FALSE, TRUE, 1, 5, 400);
check_buffers_duration (400 * GST_MSECOND);
check_get_dtmf_event_message (bus, 1, 5);
/* Long DTMF */
send_rtp_packet (src, 3000, TRUE, FALSE, 1, 5, 200);
check_get_dtmf_event_message (bus, 1, 5);
check_buffers_duration (200 * GST_MSECOND);
send_rtp_packet (src, 3000, FALSE, FALSE, 1, 5, 400);
check_no_dtmf_event_message (bus);
check_buffers_duration (200 * GST_MSECOND);
send_rtp_packet (src, 3000, FALSE, FALSE, 1, 5, 600);
check_no_dtmf_event_message (bus);
check_buffers_duration (200 * GST_MSECOND);
/* New without end to last */
send_rtp_packet (src, 4000, TRUE, TRUE, 1, 5, 250);
check_get_dtmf_event_message (bus, 1, 5);
check_buffers_duration (250 * GST_MSECOND);
check_no_dtmf_event_message (bus);
fail_unless (buffers == NULL);
gst_element_set_bus (dtmfdepay, NULL);
gst_object_unref (bus);
gst_pad_set_active (src, FALSE);
gst_pad_set_active (sink, FALSE);
gst_check_teardown_sink_pad (dtmfdepay);
gst_check_teardown_src_pad (dtmfdepay);
gst_check_teardown_element (dtmfdepay);
}
GST_END_TEST;
static GstStaticPadTemplate rtp_dtmf_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) 99, "
"clock-rate = (int) 1000, "
"seqnum-base = (uint) 333, "
"clock-base = (uint) 666, "
"ssrc = (uint) 999, "
"maxptime = (uint) 20, encoding-name = (string) \"TELEPHONE-EVENT\"")
);
GstElement *dtmfsrc;
GstPad *sink;
GstClock *testclock;
GstBus *bus;
static void
check_message_structure (GstStructure * expected_s)
{
GstMessage *message;
gboolean have_message = FALSE;
while (!have_message &&
(message = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE,
GST_MESSAGE_ELEMENT)) != NULL) {
if (gst_message_has_name (message, gst_structure_get_name (expected_s))) {
const GstStructure *s = gst_message_get_structure (message);
fail_unless (gst_structure_is_equal (s, expected_s));
have_message = TRUE;
}
gst_message_unref (message);
}
fail_unless (have_message);
gst_structure_free (expected_s);
}
static void
check_rtp_buffer (GstClockTime ts, GstClockTime duration, gboolean start,
gboolean end, guint rtpts, guint ssrc, guint volume, guint number,
guint rtpduration)
{
GstBuffer *buffer;
GstRTPBuffer rtpbuffer = GST_RTP_BUFFER_INIT;
gchar *payload;
g_mutex_lock (&check_mutex);
while (buffers == NULL)
g_cond_wait (&check_cond, &check_mutex);
g_mutex_unlock (&check_mutex);
fail_unless (buffers != NULL);
buffer = buffers->data;
buffers = g_list_delete_link (buffers, buffers);
fail_unless (GST_BUFFER_PTS (buffer) == ts);
fail_unless (GST_BUFFER_DURATION (buffer) == duration);
fail_unless (gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtpbuffer));
fail_unless (gst_rtp_buffer_get_marker (&rtpbuffer) == start);
fail_unless (gst_rtp_buffer_get_timestamp (&rtpbuffer) == rtpts);
payload = gst_rtp_buffer_get_payload (&rtpbuffer);
fail_unless (payload[0] == number);
fail_unless ((payload[1] & 0x7F) == volume);
fail_unless (! !(payload[1] & 0x80) == end);
fail_unless (GST_READ_UINT16_BE (payload + 2) == rtpduration);
gst_rtp_buffer_unmap (&rtpbuffer);
gst_buffer_unref (buffer);
}
gint method;
static void
setup_rtpdtmfsrc (void)
{
testclock = gst_test_clock_new ();
bus = gst_bus_new ();
method = 1;
dtmfsrc = gst_check_setup_element ("rtpdtmfsrc");
sink = gst_check_setup_sink_pad (dtmfsrc, &rtp_dtmf_sink_template);
gst_element_set_bus (dtmfsrc, bus);
fail_unless (gst_element_set_clock (dtmfsrc, testclock));
gst_pad_set_active (sink, TRUE);
fail_unless (gst_element_set_state (dtmfsrc, GST_STATE_PLAYING) ==
GST_STATE_CHANGE_SUCCESS);
}
static void
teardown_dtmfsrc (void)
{
gst_object_unref (testclock);
gst_pad_set_active (sink, FALSE);
gst_element_set_bus (dtmfsrc, NULL);
gst_object_unref (bus);
gst_check_teardown_sink_pad (dtmfsrc);
gst_check_teardown_element (dtmfsrc);
}
GST_START_TEST (test_dtmfsrc_invalid_events)
{
GstStructure *s;
/* Missing start */
s = gst_structure_new ("dtmf-event",
"type", G_TYPE_INT, 1, "number", G_TYPE_INT, 3,
"method", G_TYPE_INT, method, "volume", G_TYPE_INT, 8, NULL);
fail_unless (gst_pad_push_event (sink,
gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, s)) == FALSE);
/* Missing volume */
s = gst_structure_new ("dtmf-event",
"type", G_TYPE_INT, 1, "number", G_TYPE_INT, 3,
"method", G_TYPE_INT, method, "start", G_TYPE_BOOLEAN, TRUE, NULL);
fail_unless (gst_pad_push_event (sink,
gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, s)) == FALSE);
/* Missing number */
s = gst_structure_new ("dtmf-event",
"type", G_TYPE_INT, 1, "method", G_TYPE_INT, method,
"volume", G_TYPE_INT, 8, "start", G_TYPE_BOOLEAN, TRUE, NULL);
fail_unless (gst_pad_push_event (sink,
gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, s)) == FALSE);
/* Missing type */
s = gst_structure_new ("dtmf-event",
"number", G_TYPE_INT, 3, "method", G_TYPE_INT, method,
"volume", G_TYPE_INT, 8, "start", G_TYPE_BOOLEAN, TRUE, NULL);
fail_unless (gst_pad_push_event (sink,
gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, s)) == FALSE);
/* Stop before start */
s = gst_structure_new ("dtmf-event",
"type", G_TYPE_INT, 1, "number", G_TYPE_INT, 3,
"method", G_TYPE_INT, method, "volume", G_TYPE_INT, 8,
"start", G_TYPE_BOOLEAN, FALSE, NULL);
fail_unless (gst_pad_push_event (sink,
gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, s)) == FALSE);
gst_element_set_state (dtmfsrc, GST_STATE_NULL);
}
GST_END_TEST;
GST_START_TEST (test_rtpdtmfsrc_min_duration)
{
GstStructure *s;
GstClockID id;
guint timestamp = 0;
GstCaps *expected_caps, *caps;
/* Minimum duration dtmf */
s = gst_structure_new ("dtmf-event",
"type", G_TYPE_INT, 1, "number", G_TYPE_INT, 3,
"method", G_TYPE_INT, 1, "volume", G_TYPE_INT, 8,
"start", G_TYPE_BOOLEAN, TRUE, NULL);
fail_unless (gst_pad_push_event (sink,
gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
gst_structure_copy (s))));
check_no_dtmf_event_message (bus);
gst_test_clock_wait_for_next_pending_id (GST_TEST_CLOCK (testclock), NULL);
fail_unless (buffers == NULL);
id = gst_test_clock_process_next_clock_id (GST_TEST_CLOCK (testclock));
fail_unless (gst_clock_id_get_time (id) == 0);
gst_clock_id_unref (id);
gst_structure_set_name (s, "dtmf-event-processed");
check_message_structure (s);
s = gst_structure_new ("dtmf-event",
"type", G_TYPE_INT, 1, "method", G_TYPE_INT, 1,
"start", G_TYPE_BOOLEAN, FALSE, NULL);
fail_unless (gst_pad_push_event (sink,
gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
gst_structure_copy (s))));
check_rtp_buffer (0, 20 * GST_MSECOND, TRUE, FALSE, 666, 999, 8, 3, 20);
for (timestamp = 20; timestamp < MIN_PULSE_DURATION + 20; timestamp += 20) {
gst_test_clock_advance_time (GST_TEST_CLOCK (testclock),
20 * GST_MSECOND + 1);
gst_test_clock_wait_for_next_pending_id (GST_TEST_CLOCK (testclock), NULL);
fail_unless (buffers == NULL);
id = gst_test_clock_process_next_clock_id (GST_TEST_CLOCK (testclock));
fail_unless (gst_clock_id_get_time (id) == timestamp * GST_MSECOND);
gst_clock_id_unref (id);
if (timestamp < MIN_PULSE_DURATION) {
check_rtp_buffer (timestamp * GST_MSECOND, 20 * GST_MSECOND, FALSE,
FALSE, 666, 999, 8, 3, timestamp + 20);
check_no_dtmf_event_message (bus);
} else {
gst_structure_set_name (s, "dtmf-event-processed");
check_message_structure (s);
check_rtp_buffer (timestamp * GST_MSECOND,
(20 + MIN_INTER_DIGIT_INTERVAL) * GST_MSECOND, FALSE, TRUE, 666,
999, 8, 3, timestamp + 20);
}
fail_unless (buffers == NULL);
}
fail_unless (gst_test_clock_peek_id_count (GST_TEST_CLOCK (testclock)) == 0);
/* caps check */
expected_caps = gst_caps_new_simple ("application/x-rtp",
"encoding-name", G_TYPE_STRING, "TELEPHONE-EVENT",
"media", G_TYPE_STRING, "audio",
"clock-rate", G_TYPE_INT, 1000, "payload", G_TYPE_INT, 99,
"seqnum-base", G_TYPE_UINT, 333,
"clock-base", G_TYPE_UINT, 666,
"ssrc", G_TYPE_UINT, 999, "ptime", G_TYPE_UINT, 20, NULL);
caps = gst_pad_get_current_caps (sink);
fail_unless (gst_caps_can_intersect (caps, expected_caps));
gst_caps_unref (caps);
gst_caps_unref (expected_caps);
gst_element_set_state (dtmfsrc, GST_STATE_NULL);
check_no_dtmf_event_message (bus);
}
GST_END_TEST;
static GstStaticPadTemplate audio_dtmfsrc_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) \"" GST_AUDIO_NE (S16) "\", "
"rate = (int) 8003, " "channels = (int) 1")
);
static void
setup_dtmfsrc (void)
{
testclock = gst_test_clock_new ();
bus = gst_bus_new ();
method = 2;
dtmfsrc = gst_check_setup_element ("dtmfsrc");
sink = gst_check_setup_sink_pad (dtmfsrc, &audio_dtmfsrc_sink_template);
gst_element_set_bus (dtmfsrc, bus);
fail_unless (gst_element_set_clock (dtmfsrc, testclock));
gst_pad_set_active (sink, TRUE);
fail_unless (gst_element_set_state (dtmfsrc, GST_STATE_PLAYING) ==
GST_STATE_CHANGE_SUCCESS);
}
GST_START_TEST (test_dtmfsrc_min_duration)
{
GstStructure *s;
GstClockID id;
GstClockTime timestamp = 0;
GstCaps *expected_caps, *caps;
guint interval;
g_object_get (dtmfsrc, "interval", &interval, NULL);
fail_unless (interval == 50);
/* Minimum duration dtmf */
gst_test_clock_set_time (GST_TEST_CLOCK (testclock), 0);
s = gst_structure_new ("dtmf-event",
"type", G_TYPE_INT, 1, "number", G_TYPE_INT, 3,
"method", G_TYPE_INT, 2, "volume", G_TYPE_INT, 8,
"start", G_TYPE_BOOLEAN, TRUE, NULL);
fail_unless (gst_pad_push_event (sink,
gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
gst_structure_copy (s))));
gst_test_clock_wait_for_next_pending_id (GST_TEST_CLOCK (testclock), NULL);
id = gst_test_clock_process_next_clock_id (GST_TEST_CLOCK (testclock));
fail_unless (gst_clock_id_get_time (id) == 0);
gst_clock_id_unref (id);
gst_structure_set_name (s, "dtmf-event-processed");
check_message_structure (s);
s = gst_structure_new ("dtmf-event",
"type", G_TYPE_INT, 1, "method", G_TYPE_INT, 2,
"start", G_TYPE_BOOLEAN, FALSE, NULL);
fail_unless (gst_pad_push_event (sink,
gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
gst_structure_copy (s))));
for (timestamp = interval * GST_MSECOND;
timestamp < (MIN_PULSE_DURATION + MIN_INTER_DIGIT_INTERVAL) *
GST_MSECOND; timestamp += GST_MSECOND * interval) {
gst_test_clock_wait_for_next_pending_id (GST_TEST_CLOCK (testclock), NULL);
gst_test_clock_advance_time (GST_TEST_CLOCK (testclock),
interval * GST_MSECOND);
id = gst_test_clock_process_next_clock_id (GST_TEST_CLOCK (testclock));
fail_unless (gst_clock_id_get_time (id) == timestamp);
gst_clock_id_unref (id);
}
gst_structure_set_name (s, "dtmf-event-processed");
check_message_structure (s);
check_buffers_duration ((MIN_PULSE_DURATION + MIN_INTER_DIGIT_INTERVAL) *
GST_MSECOND);
fail_unless (gst_test_clock_peek_id_count (GST_TEST_CLOCK (testclock)) == 0);
/* caps check */
expected_caps = gst_caps_new_simple ("audio/x-raw",
"format", G_TYPE_STRING, GST_AUDIO_NE (S16),
"rate", G_TYPE_INT, 8003, "channels", G_TYPE_INT, 1, NULL);
caps = gst_pad_get_current_caps (sink);
fail_unless (gst_caps_can_intersect (caps, expected_caps));
gst_caps_unref (caps);
gst_caps_unref (expected_caps);
gst_element_set_state (dtmfsrc, GST_STATE_NULL);
check_no_dtmf_event_message (bus);
}
GST_END_TEST;
static Suite *
dtmf_suite (void)
{
Suite *s = suite_create ("dtmf");
TCase *tc;
tc = tcase_create ("rtpdtmfdepay");
tcase_add_test (tc, test_rtpdtmfdepay);
suite_add_tcase (s, tc);
tc = tcase_create ("rtpdtmfsrc");
tcase_add_checked_fixture (tc, setup_rtpdtmfsrc, teardown_dtmfsrc);
tcase_add_test (tc, test_dtmfsrc_invalid_events);
tcase_add_test (tc, test_rtpdtmfsrc_min_duration);
suite_add_tcase (s, tc);
tc = tcase_create ("dtmfsrc");
tcase_add_checked_fixture (tc, setup_dtmfsrc, teardown_dtmfsrc);
tcase_add_test (tc, test_dtmfsrc_invalid_events);
tcase_add_test (tc, test_dtmfsrc_min_duration);
suite_add_tcase (s, tc);
return s;
}
GST_CHECK_MAIN (dtmf);