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jack: move plugin to gst-plugins-good

Browse source 
https://bugzilla.gnome.org/show_bug.cgi?id=621929
This commit is contained in:
Tim-Philipp Müller 2011-01-02 15:11:52 +00:00
parent 992c05f840
commit a197901b82
28 changed files with 19 additions and 3154 deletions
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2
LICENSE_readme
View file

@ -37,8 +37,6 @@ xvid libxvidcore (http://www.xvid.org/)
Plugins derived from GPL code are as follows:
dvdreadsrc libdvdread (http://www.dtek.chalmers.se/groups/dvd/)
jack libjack (http://jackit.sourceforge.net/)
Note libjack is LGPL, but plugin is GPL.
monoscope None (Algorithm by Ralph Loader, Joerg Walter,
Richard Boulton, and Andy Lo A Foe)
rtjpeg None (Erik Walthinsen's algorithm)

4
Makefile.am
View file

@ -46,6 +46,7 @@ CRUFT_FILES = \
$(top_builddir)/common/shave-libtool \
$(top_builddir)/ext/alsaspdif/.libs/*.{so,dll,DLL,dylib} \
$(top_builddir)/ext/ivorbis/.libs/*.{so,dll,DLL,dylib} \
$(top_builddir)/ext/jack/.libs/*.{so,dll,DLL,dylib} \
$(top_builddir)/gst/aacparse/.libs/*.{so,dll,DLL,dylib} \
$(top_builddir)/gst/amrparse/.libs/*.{so,dll,DLL,dylib} \
$(top_builddir)/gst/flacparse/.libs/*.{so,dll,DLL,dylib} \
@ -60,6 +61,7 @@ CRUFT_FILES = \
$(top_builddir)/tests/check/elements/selector \
$(top_builddir)/tests/check/elements/valve \
$(top_builddir)/tests/check/pipelines/metadata \
$(top_builddir)/tests/examples/jack/jack_client \
$(top_builddir)/tests/examples/switch/switcher \
$(top_builddir)/tests/icles/output-selector-test \
$(top_builddir)/tests/icles/test-oss4
@ -74,8 +76,10 @@ CRUFT_DIRS = \
$(top_srcdir)/gst/valve \
$(top_srcdir)/tests/examples/shapewipe \
$(top_srcdir)/tests/examples/switch \
$(top_srcdir)/tests/examples/jack \
$(top_srcdir)/ext/alsaspdif \
$(top_srcdir)/ext/ivorbis \
$(top_srcdir)/ext/jack \
$(top_srcdir)/ext/metadata
include $(top_srcdir)/common/cruft.mak

11
configure.ac
View file

@ -888,14 +888,6 @@ AG_GST_CHECK_FEATURE(GSM, [GSM library], gsmenc gsmdec, [
AC_SUBST(GSM_LIBS)
])
dnl *** Jack ***
translit(dnm, m, l) AM_CONDITIONAL(USE_JACK, true)
AG_GST_CHECK_FEATURE(JACK, Jack, jack, [
PKG_CHECK_MODULES(JACK, jack >= 0.99.10, HAVE_JACK="yes", HAVE_JACK="no")
AC_SUBST(JACK_CFLAGS)
AC_SUBST(JACK_LIBS)
])
dnl *** jp2k ***
translit(dnm, m, l) AM_CONDITIONAL(USE_JP2K, true)
AG_GST_CHECK_FEATURE(JP2K, [jp2k], jp2kdec jp2kenc, [
@ -1607,7 +1599,6 @@ AM_CONDITIONAL(USE_FAAD, false)
AM_CONDITIONAL(USE_FBDEV, false)
AM_CONDITIONAL(USE_FLITE, false)
AM_CONDITIONAL(USE_GSM, false)
AM_CONDITIONAL(USE_JACK, false)
AM_CONDITIONAL(USE_JP2K, false)
AM_CONDITIONAL(USE_KATE, false)
AM_CONDITIONAL(USE_TIGER, false)
@ -1812,7 +1803,6 @@ tests/examples/camerabin2/Makefile
tests/examples/directfb/Makefile
tests/examples/mxf/Makefile
tests/examples/scaletempo/Makefile
tests/examples/jack/Makefile
tests/icles/Makefile
ext/amrwbenc/Makefile
ext/assrender/Makefile
@ -1830,7 +1820,6 @@ ext/faac/Makefile
ext/faad/Makefile
ext/flite/Makefile
ext/gsm/Makefile
ext/jack/Makefile
ext/jp2k/Makefile
ext/kate/Makefile
ext/ladspa/Makefile

2
docs/plugins/Makefile.am
View file

@ -99,8 +99,6 @@ EXTRA_HFILES = \
$(top_srcdir)/ext/dts/gstdtsdec.h \
$(top_srcdir)/ext/faac/gstfaac.h \
$(top_srcdir)/ext/faad/gstfaad.h \
$(top_srcdir)/ext/jack/gstjackaudiosrc.h \
$(top_srcdir)/ext/jack/gstjackaudiosink.h \
$(top_srcdir)/ext/kate/gstkateenc.h \
$(top_srcdir)/ext/kate/gstkatedec.h \
$(top_srcdir)/ext/kate/gstkateparse.h \

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

@ -69,8 +69,6 @@
<xi:include href="xml/element-freeze.xml" />
<xi:include href="xml/element-gaussianblur.xml" />
<xi:include href="xml/element-ivfparse.xml" />
<xi:include href="xml/element-jackaudiosrc.xml" />
<xi:include href="xml/element-jackaudiosink.xml" />
<xi:include href="xml/element-jpegparse.xml" />
<xi:include href="xml/element-kaleidoscope.xml" />
<xi:include href="xml/element-kateenc.xml" />
@ -169,7 +167,6 @@
<xi:include href="xml/plugin-gsm.xml" />
<xi:include href="xml/plugin-h264parse.xml" />
<xi:include href="xml/plugin-ivfparse.xml" />
<xi:include href="xml/plugin-jack.xml" />
<xi:include href="xml/plugin-jpegformat.xml" />
<xi:include href="xml/plugin-kate.xml" />
<xi:include href="xml/plugin-ladspa.xml" />

30
docs/plugins/gst-plugins-bad-plugins-sections.txt
View file

@ -806,36 +806,6 @@ GST_TYPE_IVF_PARSE
gst_ivf_parse_get_type
</SECTION>
<SECTION>
<FILE>element-jackaudiosrc</FILE>
<TITLE>jackaudiosrc</TITLE>
GstJackAudioSrc
<SUBSECTION Standard>
GstJackAudioSrcClass
GST_JACK_AUDIO_SRC
GST_JACK_AUDIO_SRC_CLASS
GST_JACK_AUDIO_SRC_GET_CLASS
GST_IS_JACK_AUDIO_SRC
GST_IS_JACK_AUDIO_SRC_CLASS
GST_TYPE_JACK_AUDIO_SRC
gst_jack_audio_src_get_type
</SECTION>
<SECTION>
<FILE>element-jackaudiosink</FILE>
<TITLE>jackaudiosink</TITLE>
GstJackAudioSink
<SUBSECTION Standard>
GstJackAudioSinkClass
GST_JACK_AUDIO_SINK
GST_JACK_AUDIO_SINK_CLASS
GST_JACK_AUDIO_SINK_GET_CLASS
GST_IS_JACK_AUDIO_SINK
GST_IS_JACK_AUDIO_SINK_CLASS
GST_TYPE_JACK_AUDIO_SINK
gst_jack_audio_sink_get_type
</SECTION>
<SECTION>
<FILE>element-jpegparse</FILE>
<TITLE>jpegparse</TITLE>

86
docs/plugins/gst-plugins-bad-plugins.args
View file

@ -1538,36 +1538,6 @@
<DEFAULT>1</DEFAULT>
</ARG>
<ARG>
<NAME>GstJackAudioSink::connect</NAME>
<TYPE>GstJackConnect</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Connect</NICK>
<BLURB>Specify how the output ports will be connected.</BLURB>
<DEFAULT>Automatically connect ports to physical ports</DEFAULT>
</ARG>
<ARG>
<NAME>GstJackAudioSink::server</NAME>
<TYPE>gchar*</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Server</NICK>
<BLURB>The Jack server to connect to (NULL = default).</BLURB>
<DEFAULT>NULL</DEFAULT>
</ARG>
<ARG>
<NAME>GstJackAudioSink::client</NAME>
<TYPE>JackClient*</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>JackClient</NICK>
<BLURB>Handle for jack client.</BLURB>
<DEFAULT></DEFAULT>
</ARG>
<ARG>
<NAME>GstDvbSrc::bandwidth</NAME>
<TYPE>GstDvbSrcBandwidth</TYPE>
@ -21953,36 +21923,6 @@
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstJackAudioSrc::connect</NAME>
<TYPE>GstJackConnect</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Connect</NICK>
<BLURB>Specify how the input ports will be connected.</BLURB>
<DEFAULT>Automatically connect ports to physical ports</DEFAULT>
</ARG>
<ARG>
<NAME>GstJackAudioSrc::server</NAME>
<TYPE>gchar*</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Server</NICK>
<BLURB>The Jack server to connect to (NULL = default).</BLURB>
<DEFAULT>NULL</DEFAULT>
</ARG>
<ARG>
<NAME>GstJackAudioSrc::client</NAME>
<TYPE>JackClient*</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>JackClient</NICK>
<BLURB>Handle for jack client.</BLURB>
<DEFAULT></DEFAULT>
</ARG>
<ARG>
<NAME>GstDCCPClientSrc::caps</NAME>
<TYPE>GstCaps*</TYPE>
@ -26530,7 +26470,7 @@
<FLAGS>rw</FLAGS>
<NICK>physics</NICK>
<BLURB>water density: from 1 to 4.</BLURB>
<DEFAULT>7.75038e-304</DEFAULT>
<DEFAULT>8.09774e-321</DEFAULT>
</ARG>
<ARG>
@ -26570,7 +26510,7 @@
<FLAGS>rw</FLAGS>
<NICK>splash</NICK>
<BLURB>make a big splash in the center.</BLURB>
<DEFAULT>0</DEFAULT>
<DEFAULT>2.34994e-310</DEFAULT>
</ARG>
<ARG>
@ -26580,7 +26520,7 @@
<FLAGS>rw</FLAGS>
<NICK>splash</NICK>
<BLURB>make a big splash in the center.</BLURB>
<DEFAULT>4.77773e-299</DEFAULT>
<DEFAULT>1.82574e-315</DEFAULT>
</ARG>
<ARG>
@ -26610,7 +26550,7 @@
<FLAGS>rw</FLAGS>
<NICK>ratiox</NICK>
<BLURB>x-ratio.</BLURB>
<DEFAULT>0</DEFAULT>
<DEFAULT>1.47273e-316</DEFAULT>
</ARG>
<ARG>
@ -26620,7 +26560,7 @@
<FLAGS>rw</FLAGS>
<NICK>ratioy</NICK>
<BLURB>y-ratio.</BLURB>
<DEFAULT>0</DEFAULT>
<DEFAULT>1.85891e-316</DEFAULT>
</ARG>
<ARG>
@ -26630,7 +26570,7 @@
<FLAGS>rw</FLAGS>
<NICK>DelayTime</NICK>
<BLURB>the delay time.</BLURB>
<DEFAULT>0</DEFAULT>
<DEFAULT>2.18476e-316</DEFAULT>
</ARG>
<ARG>
@ -26660,7 +26600,7 @@
<FLAGS>rw</FLAGS>
<NICK>Color</NICK>
<BLURB>the color of the image.</BLURB>
<DEFAULT>0</DEFAULT>
<DEFAULT>1.39669e-37</DEFAULT>
</ARG>
<ARG>
@ -26680,7 +26620,7 @@
<FLAGS>rw</FLAGS>
<NICK>Color-R</NICK>
<BLURB>the color of the image.</BLURB>
<DEFAULT>0</DEFAULT>
<DEFAULT>7.30424e-38</DEFAULT>
</ARG>
<ARG>
@ -27010,7 +26950,7 @@
<FLAGS>rw</FLAGS>
<NICK>lredscale</NICK>
<BLURB>multiplier for downscaling non-edge brightness.</BLURB>
<DEFAULT>0</DEFAULT>
<DEFAULT>3.40216e-111</DEFAULT>
</ARG>
<ARG>
@ -27020,7 +26960,7 @@
<FLAGS>rw</FLAGS>
<NICK>lthresh</NICK>
<BLURB>threshold for edge lightening.</BLURB>
<DEFAULT>0</DEFAULT>
<DEFAULT>6.9235e+228</DEFAULT>
</ARG>
<ARG>
@ -27030,7 +26970,7 @@
<FLAGS>rw</FLAGS>
<NICK>lupscale</NICK>
<BLURB>multiplier for upscaling edge brightness.</BLURB>
<DEFAULT>0</DEFAULT>
<DEFAULT>7.54985e-96</DEFAULT>
</ARG>
<ARG>
@ -27200,7 +27140,7 @@
<FLAGS>rw</FLAGS>
<NICK>blend</NICK>
<BLURB>blend factor.</BLURB>
<DEFAULT>4.74303e-322</DEFAULT>
<DEFAULT>4.77773e-299</DEFAULT>
</ARG>
<ARG>
@ -27390,7 +27330,7 @@
<FLAGS>rw</FLAGS>
<NICK>HSync</NICK>
<BLURB>the hsync offset.</BLURB>
<DEFAULT>0</DEFAULT>
<DEFAULT>1.86264e-09</DEFAULT>
</ARG>
<ARG>

5
docs/plugins/gst-plugins-bad-plugins.hierarchy
View file

@ -31,7 +31,6 @@ GObject
GstApExSink
GstNasSink
GstSDLAudioSink
GstJackAudioSink
GstChecksumSink
GstDCCPClientSink
GstDCCPServerSink
@ -46,8 +45,6 @@ GObject
GstDTMFSrc
GstDataURISrc
GstPushSrc
GstBaseAudioSrc
GstJackAudioSrc
GstDCCPClientSrc
GstDCCPServerSrc
GstDc1394
@ -490,8 +487,6 @@ GObject
GstRegistry
GstRingBuffer
GstAudioSinkRingBuffer
GstJackAudioSinkRingBuffer
GstJackAudioSrcRingBuffer
GstTask
GstTaskPool
GstSignalObject

43
docs/plugins/inspect/plugin-jack.xml
View file

@ -1,43 +0,0 @@
<plugin>
<name>jack</name>
<description>Jack elements</description>
<filename>../../ext/jack/.libs/libgstjack.so</filename>
<basename>libgstjack.so</basename>
<version>0.10.20.1</version>
<license>LGPL</license>
<source>gst-plugins-bad</source>
<package>GStreamer Bad Plug-ins git</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<name>jackaudiosink</name>
<longname>Audio Sink (Jack)</longname>
<class>Sink/Audio</class>
<description>Output to Jack</description>
<author>Wim Taymans &lt;wim@fluendo.com&gt;</author>
<pads>
<caps>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>audio/x-raw-float, endianness=(int){ 1234 }, width=(int)32, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
</caps>
</pads>
</element>
<element>
<name>jackaudiosrc</name>
<longname>Audio Source (Jack)</longname>
<class>Source/Audio</class>
<description>Input from Jack</description>
<author>Tristan Matthews &lt;tristan@sat.qc.ca&gt;</author>
<pads>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>audio/x-raw-float, endianness=(int){ 1234 }, width=(int)32, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
</caps>
</pads>
</element>
</elements>
</plugin>

8
ext/Makefile.am
View file

@ -136,12 +136,6 @@ endif
HERMES_DIR=
# endif
if USE_JACK
JACK_DIR=jack
else
JACK_DIR=
endif
if USE_JP2K
JP2K_DIR = jp2k
else
@ -398,7 +392,6 @@ SUBDIRS=\
$(GSM_DIR) \
$(G729_DIR) \
$(HERMES_DIR) \
$(JACK_DIR) \
$(JP2K_DIR) \
$(KATE_DIR) \
$(LADSPA_DIR) \
@ -453,7 +446,6 @@ DIST_SUBDIRS = \
gsettings \
gsm \
ladspa \
jack \
jp2k \
kate \
libmms \

1
ext/jack/.gitignore vendored
View file

@ -1 +0,0 @@
*.loT

12
ext/jack/Makefile.am
View file

@ -1,12 +0,0 @@
plugin_LTLIBRARIES = libgstjack.la
libgstjack_la_SOURCES = gstjackutil.c gstjack.c gstjackaudiosrc.c gstjackaudiosink.c gstjackaudioclient.c
libgstjack_la_CFLAGS = $(GST_PLUGINS_BAD_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(JACK_CFLAGS)
libgstjack_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) $(JACK_LIBS)
libgstjack_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstjack_la_LIBTOOLFLAGS = --tag=disable-static
noinst_HEADERS = gstjackutil.h gstjackaudiosrc.h gstjackaudiosink.h gstjackaudioclient.h gstjack.h gstjackringbuffer.h
EXTRA_DIST = README

4
ext/jack/README
View file

@ -1,4 +0,0 @@
to be written, la dee da
jackit.sf.net

95
ext/jack/gstjack.c
View file

@ -1,95 +0,0 @@
/* GStreamer Jack plugins
* Copyright (C) 2006 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstjackaudiosrc.h"
#include "gstjackaudiosink.h"
GType
gst_jack_connect_get_type (void)
{
static GType jack_connect_type = 0;
static const GEnumValue jack_connect[] = {
{GST_JACK_CONNECT_NONE,
"Don't automatically connect ports to physical ports", "none"},
{GST_JACK_CONNECT_AUTO,
"Automatically connect ports to physical ports", "auto"},
{GST_JACK_CONNECT_AUTO_FORCED,
"Automatically connect ports to as many physical ports as possible",
"auto-forced"},
{0, NULL, NULL},
};
if (!jack_connect_type) {
jack_connect_type = g_enum_register_static ("GstJackConnect", jack_connect);
}
return jack_connect_type;
}
static gpointer
gst_jack_client_copy (gpointer jclient)
{
return jclient;
}
static void
gst_jack_client_free (gpointer jclient)
{
return;
}
GType
gst_jack_client_get_type (void)
{
static GType type; /* 0 */
if (type == 0) {
/* hackish, but makes it show up nicely in gst-inspect */
type = g_boxed_type_register_static ("JackClient",
(GBoxedCopyFunc) gst_jack_client_copy,
(GBoxedFreeFunc) gst_jack_client_free);
}
return type;
}
static gboolean
plugin_init (GstPlugin * plugin)
{
if (!gst_element_register (plugin, "jackaudiosrc", GST_RANK_PRIMARY,
GST_TYPE_JACK_AUDIO_SRC))
return FALSE;
if (!gst_element_register (plugin, "jackaudiosink", GST_RANK_PRIMARY,
GST_TYPE_JACK_AUDIO_SINK))
return FALSE;
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"jack",
"Jack elements",
plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)

55
ext/jack/gstjack.h
View file

@ -1,55 +0,0 @@
/* GStreamer
* Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
*
* gstjack.h:
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef _GST_JACK_H_
#define _GST_JACK_H_
/**
* GstJackConnect:
* @GST_JACK_CONNECT_NONE: Don't automatically connect to physical ports.
* In this mode, the element will accept any number of input channels and will
* create (but not connect) an output port for each channel.
* @GST_JACK_CONNECT_AUTO: In this mode, the element will try to connect each
* output port to a random physical jack input pin. The sink will
* expose the number of physical channels on its pad caps.
* @GST_JACK_CONNECT_AUTO_FORCED: In this mode, the element will try to connect each
* output port to a random physical jack input pin. The element will accept any number
* of input channels.
*
* Specify how the output ports will be connected.
*/
typedef enum {
GST_JACK_CONNECT_NONE,
GST_JACK_CONNECT_AUTO,
GST_JACK_CONNECT_AUTO_FORCED
} GstJackConnect;
typedef jack_default_audio_sample_t sample_t;
#define GST_TYPE_JACK_CONNECT (gst_jack_connect_get_type())
#define GST_TYPE_JACK_CLIENT (gst_jack_client_get_type ())
GType gst_jack_client_get_type(void);
GType gst_jack_connect_get_type(void);
#endif // _GST_JACK_H_

525
ext/jack/gstjackaudioclient.c
View file

@ -1,525 +0,0 @@
/* GStreamer
* Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
*
* gstjackaudioclient.c: jack audio client implementation
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include <string.h>
#include "gstjackaudioclient.h"
GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_client_debug);
#define GST_CAT_DEFAULT gst_jack_audio_client_debug
void
gst_jack_audio_client_init (void)
{
GST_DEBUG_CATEGORY_INIT (gst_jack_audio_client_debug, "jackclient", 0,
"jackclient helpers");
}
/* a list of global connections indexed by id and server. */
G_LOCK_DEFINE_STATIC (connections_lock);
static GList *connections;
/* the connection to a server */
typedef struct
{
gint refcount;
GMutex *lock;
GCond *flush_cond;
/* id/server pair and the connection */
gchar *id;
gchar *server;
jack_client_t *client;
/* lists of GstJackAudioClients */
gint n_clients;
GList *src_clients;
GList *sink_clients;
} GstJackAudioConnection;
/* an object sharing a jack_client_t connection. */
struct _GstJackAudioClient
{
GstJackAudioConnection *conn;
GstJackClientType type;
gboolean active;
gboolean deactivate;
void (*shutdown) (void *arg);
JackProcessCallback process;
JackBufferSizeCallback buffer_size;
JackSampleRateCallback sample_rate;
gpointer user_data;
};
typedef jack_default_audio_sample_t sample_t;
typedef struct
{
jack_nframes_t nframes;
gpointer user_data;
} JackCB;
static int
jack_process_cb (jack_nframes_t nframes, void *arg)
{
GstJackAudioConnection *conn = (GstJackAudioConnection *) arg;
GList *walk;
int res = 0;
g_mutex_lock (conn->lock);
/* call sources first, then sinks. Sources will either push data into the
* ringbuffer of the sinks, which will then pull the data out of it, or
* sinks will pull the data from the sources. */
for (walk = conn->src_clients; walk; walk = g_list_next (walk)) {
GstJackAudioClient *client = (GstJackAudioClient *) walk->data;
/* only call active clients */
if ((client->active || client->deactivate) && client->process) {
res = client->process (nframes, client->user_data);
if (client->deactivate) {
client->deactivate = FALSE;
g_cond_signal (conn->flush_cond);
}
}
}
for (walk = conn->sink_clients; walk; walk = g_list_next (walk)) {
GstJackAudioClient *client = (GstJackAudioClient *) walk->data;
/* only call active clients */
if ((client->active || client->deactivate) && client->process) {
res = client->process (nframes, client->user_data);
if (client->deactivate) {
client->deactivate = FALSE;
g_cond_signal (conn->flush_cond);
}
}
}
g_mutex_unlock (conn->lock);
return res;
}
/* we error out */
static int
jack_sample_rate_cb (jack_nframes_t nframes, void *arg)
{
return 0;
}
/* we error out */
static int
jack_buffer_size_cb (jack_nframes_t nframes, void *arg)
{
return 0;
}
static void
jack_shutdown_cb (void *arg)
{
GstJackAudioConnection *conn = (GstJackAudioConnection *) arg;
GList *walk;
GST_DEBUG ("disconnect client %s from server %s", conn->id,
GST_STR_NULL (conn->server));
g_mutex_lock (conn->lock);
for (walk = conn->src_clients; walk; walk = g_list_next (walk)) {
GstJackAudioClient *client = (GstJackAudioClient *) walk->data;
if (client->shutdown)
client->shutdown (client->user_data);
}
for (walk = conn->sink_clients; walk; walk = g_list_next (walk)) {
GstJackAudioClient *client = (GstJackAudioClient *) walk->data;
if (client->shutdown)
client->shutdown (client->user_data);
}
g_mutex_unlock (conn->lock);
}
typedef struct
{
const gchar *id;
const gchar *server;
} FindData;
static gint
connection_find (GstJackAudioConnection * conn, FindData * data)
{
/* id's must match */
if (strcmp (conn->id, data->id))
return 1;
/* both the same or NULL */
if (conn->server == data->server)
return 0;
/* we cannot compare NULL */
if (conn->server == NULL || data->server == NULL)
return 1;
if (strcmp (conn->server, data->server))
return 1;
return 0;
}
/* make a connection with @id and @server. Returns NULL on failure with the
* status set. */
static GstJackAudioConnection *
gst_jack_audio_make_connection (const gchar * id, const gchar * server,
jack_client_t * jclient, jack_status_t * status)
{
GstJackAudioConnection *conn;
jack_options_t options;
gint res;
*status = 0;
GST_DEBUG ("new client %s, connecting to server %s", id,
GST_STR_NULL (server));
/* never start a server */
options = JackNoStartServer;
/* if we have a servername, use it */
if (server != NULL)
options |= JackServerName;
/* open the client */
if (jclient == NULL)
jclient = jack_client_open (id, options, status, server);
if (jclient == NULL)
goto could_not_open;
/* now create object */
conn = g_new (GstJackAudioConnection, 1);
conn->refcount = 1;
conn->lock = g_mutex_new ();
conn->flush_cond = g_cond_new ();
conn->id = g_strdup (id);
conn->server = g_strdup (server);
conn->client = jclient;
conn->n_clients = 0;
conn->src_clients = NULL;
conn->sink_clients = NULL;
/* set our callbacks */
jack_set_process_callback (jclient, jack_process_cb, conn);
/* these callbacks cause us to error */
jack_set_buffer_size_callback (jclient, jack_buffer_size_cb, conn);
jack_set_sample_rate_callback (jclient, jack_sample_rate_cb, conn);
jack_on_shutdown (jclient, jack_shutdown_cb, conn);
/* all callbacks are set, activate the client */
if ((res = jack_activate (jclient)))
goto could_not_activate;
GST_DEBUG ("opened connection %p", conn);
return conn;
/* ERRORS */
could_not_open:
{
GST_DEBUG ("failed to open jack client, %d", *status);
return NULL;
}
could_not_activate:
{
GST_ERROR ("Could not activate client (%d)", res);
*status = JackFailure;
g_mutex_free (conn->lock);
g_free (conn->id);
g_free (conn->server);
g_free (conn);
return NULL;
}
}
static GstJackAudioConnection *
gst_jack_audio_get_connection (const gchar * id, const gchar * server,
jack_client_t * jclient, jack_status_t * status)
{
GstJackAudioConnection *conn;
GList *found;
FindData data;
GST_DEBUG ("getting connection for id %s, server %s", id,
GST_STR_NULL (server));
data.id = id;
data.server = server;
G_LOCK (connections_lock);
found =
g_list_find_custom (connections, &data, (GCompareFunc) connection_find);
if (found != NULL && jclient != NULL) {
/* we found it, increase refcount and return it */
conn = (GstJackAudioConnection *) found->data;
conn->refcount++;
GST_DEBUG ("found connection %p", conn);
} else {
/* make new connection */
conn = gst_jack_audio_make_connection (id, server, jclient, status);
if (conn != NULL) {
GST_DEBUG ("created connection %p", conn);
/* add to list on success */
connections = g_list_prepend (connections, conn);
} else {
GST_WARNING ("could not create connection");
}
}
G_UNLOCK (connections_lock);
return conn;
}
static void
gst_jack_audio_unref_connection (GstJackAudioConnection * conn)
{
gint res;
gboolean zero;
GST_DEBUG ("unref connection %p refcnt %d", conn, conn->refcount);
G_LOCK (connections_lock);
conn->refcount--;
if ((zero = (conn->refcount == 0))) {
GST_DEBUG ("closing connection %p", conn);
/* remove from list, we can release the mutex after removing the connection
* from the list because after that, nobody can access the connection anymore. */
connections = g_list_remove (connections, conn);
}
G_UNLOCK (connections_lock);
/* if we are zero, close and cleanup the connection */
if (zero) {
/* don't use conn->lock here. two reasons:
*
* 1) its not necessary: jack_deactivate() will not return until the JACK thread
* associated with this connection is cleaned up by a thread join, hence
* no more callbacks can occur or be in progress.
*
* 2) it would deadlock anyway, because jack_deactivate() will sleep
* waiting for the JACK thread, and can thus cause deadlock in
* jack_process_cb()
*/
if ((res = jack_deactivate (conn->client))) {
/* we only warn, this means the server is probably shut down and the client
* is gone anyway. */
GST_WARNING ("Could not deactivate Jack client (%d)", res);
}
/* close connection */
if ((res = jack_client_close (conn->client))) {
/* we assume the client is gone. */
GST_WARNING ("close failed (%d)", res);
}
/* free resources */
g_mutex_free (conn->lock);
g_cond_free (conn->flush_cond);
g_free (conn->id);
g_free (conn->server);
g_free (conn);
}
}
static void
gst_jack_audio_connection_add_client (GstJackAudioConnection * conn,
GstJackAudioClient * client)
{
g_mutex_lock (conn->lock);
switch (client->type) {
case GST_JACK_CLIENT_SOURCE:
conn->src_clients = g_list_append (conn->src_clients, client);
conn->n_clients++;
break;
case GST_JACK_CLIENT_SINK:
conn->sink_clients = g_list_append (conn->sink_clients, client);
conn->n_clients++;
break;
default:
g_warning ("trying to add unknown client type");
break;
}
g_mutex_unlock (conn->lock);
}
static void
gst_jack_audio_connection_remove_client (GstJackAudioConnection * conn,
GstJackAudioClient * client)
{
g_mutex_lock (conn->lock);
switch (client->type) {
case GST_JACK_CLIENT_SOURCE:
conn->src_clients = g_list_remove (conn->src_clients, client);
conn->n_clients--;
break;
case GST_JACK_CLIENT_SINK:
conn->sink_clients = g_list_remove (conn->sink_clients, client);
conn->n_clients--;
break;
default:
g_warning ("trying to remove unknown client type");
break;
}
g_mutex_unlock (conn->lock);
}
/**
* gst_jack_audio_client_get:
* @id: the client id
* @server: the server to connect to or NULL for the default server
* @type: the client type
* @shutdown: a callback when the jack server shuts down
* @process: a callback when samples are available
* @buffer_size: a callback when the buffer_size changes
* @sample_rate: a callback when the sample_rate changes
* @user_data: user data passed to the callbacks
* @status: pointer to hold the jack status code in case of errors
*
* Get the jack client connection for @id and @server. Connections to the same
* @id and @server will receive the same physical Jack client connection and
* will therefore be scheduled in the same process callback.
*
* Returns: a #GstJackAudioClient.
*/
GstJackAudioClient *
gst_jack_audio_client_new (const gchar * id, const gchar * server,
jack_client_t * jclient, GstJackClientType type,
void (*shutdown) (void *arg), JackProcessCallback process,
JackBufferSizeCallback buffer_size, JackSampleRateCallback sample_rate,
gpointer user_data, jack_status_t * status)
{
GstJackAudioClient *client;
GstJackAudioConnection *conn;
g_return_val_if_fail (id != NULL, NULL);
g_return_val_if_fail (status != NULL, NULL);
/* first get a connection for the id/server pair */
conn = gst_jack_audio_get_connection (id, server, jclient, status);
if (conn == NULL)
goto no_connection;
GST_INFO ("new client %s", id);
/* make new client using the connection */
client = g_new (GstJackAudioClient, 1);
client->active = client->deactivate = FALSE;
client->conn = conn;
client->type = type;
client->shutdown = shutdown;
client->process = process;
client->buffer_size = buffer_size;
client->sample_rate = sample_rate;
client->user_data = user_data;
/* add the client to the connection */
gst_jack_audio_connection_add_client (conn, client);
return client;
/* ERRORS */
no_connection:
{
GST_DEBUG ("Could not get server connection (%d)", *status);
return NULL;
}
}
/**
* gst_jack_audio_client_free:
* @client: a #GstJackAudioClient
*
* Free the resources used by @client.
*/
void
gst_jack_audio_client_free (GstJackAudioClient * client)
{
GstJackAudioConnection *conn;
g_return_if_fail (client != NULL);
GST_INFO ("free client");
conn = client->conn;
/* remove from connection first so that it's not scheduled anymore after this
* call */
gst_jack_audio_connection_remove_client (conn, client);
gst_jack_audio_unref_connection (conn);
g_free (client);
}
/**
* gst_jack_audio_client_get_client:
* @client: a #GstJackAudioClient
*
* Get the jack audio client for @client. This function is used to perform
* operations on the jack server from this client.
*
* Returns: The jack audio client.
*/
jack_client_t *
gst_jack_audio_client_get_client (GstJackAudioClient * client)
{
g_return_val_if_fail (client != NULL, NULL);
/* no lock needed, the connection and the client does not change
* once the client is created. */
return client->conn->client;
}
/**
* gst_jack_audio_client_set_active:
* @client: a #GstJackAudioClient
* @active: new mode for the client
*
* Activate or deactive @client. When a client is activated it will receive
* callbacks when data should be processed.
*
* Returns: 0 if all ok.
*/
gint
gst_jack_audio_client_set_active (GstJackAudioClient * client, gboolean active)
{
g_return_val_if_fail (client != NULL, -1);
/* make sure that we are not dispatching the client */
g_mutex_lock (client->conn->lock);
if (client->active && !active) {
/* we need to process once more to flush the port */
client->deactivate = TRUE;
/* need to wait for process_cb run once more */
while (client->deactivate)
g_cond_wait (client->conn->flush_cond, client->conn->lock);
}
client->active = active;
g_mutex_unlock (client->conn->lock);
return 0;
}

59
ext/jack/gstjackaudioclient.h
View file

@ -1,59 +0,0 @@
/* GStreamer
* Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
*
* gstjackaudioclient.h:
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __GST_JACK_AUDIO_CLIENT_H__
#define __GST_JACK_AUDIO_CLIENT_H__
#include <jack/jack.h>
#include <gst/gst.h>
G_BEGIN_DECLS
typedef enum
{
GST_JACK_CLIENT_SOURCE,
GST_JACK_CLIENT_SINK
} GstJackClientType;
typedef struct _GstJackAudioClient GstJackAudioClient;
void gst_jack_audio_client_init (void);
GstJackAudioClient * gst_jack_audio_client_new (const gchar *id, const gchar *server,
jack_client_t *jclient,
GstJackClientType type,
void (*shutdown) (void *arg),
JackProcessCallback process,
JackBufferSizeCallback buffer_size,
JackSampleRateCallback sample_rate,
gpointer user_data,
jack_status_t *status);
void gst_jack_audio_client_free (GstJackAudioClient *client);
jack_client_t * gst_jack_audio_client_get_client (GstJackAudioClient *client);
gboolean gst_jack_audio_client_set_active (GstJackAudioClient *client, gboolean active);
G_END_DECLS
#endif /* __GST_JACK_AUDIO_CLIENT_H__ */

852
ext/jack/gstjackaudiosink.c
View file

@ -1,852 +0,0 @@
/* GStreamer
* Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
*
* gstjackaudiosink.c: jack audio sink implementation
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-jackaudiosink
* @see_also: #GstBaseAudioSink, #GstRingBuffer
*
* A Sink that outputs data to Jack ports.
*
* It will create N Jack ports named out_&lt;name&gt;_&lt;num&gt; where
* &lt;name&gt; is the element name and &lt;num&gt; is starting from 1.
* Each port corresponds to a gstreamer channel.
*
* The samplerate as exposed on the caps is always the same as the samplerate of
* the jack server.
*
* When the #GstJackAudioSink:connect property is set to auto, this element
* will try to connect each output port to a random physical jack input pin. In
* this mode, the sink will expose the number of physical channels on its pad
* caps.
*
* When the #GstJackAudioSink:connect property is set to none, the element will
* accept any number of input channels and will create (but not connect) an
* output port for each channel.
*
* The element will generate an error when the Jack server is shut down when it
* was PAUSED or PLAYING. This element does not support dynamic rate and buffer
* size changes at runtime.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch audiotestsrc ! jackaudiosink
* ]| Play a sine wave to using jack.
* </refsect2>
*
* Last reviewed on 2006-11-30 (0.10.4)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst-i18n-plugin.h>
#include <stdlib.h>
#include <string.h>
#include "gstjackaudiosink.h"
#include "gstjackringbuffer.h"
GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_sink_debug);
#define GST_CAT_DEFAULT gst_jack_audio_sink_debug
static gboolean
gst_jack_audio_sink_allocate_channels (GstJackAudioSink * sink, gint channels)
{
jack_client_t *client;
client = gst_jack_audio_client_get_client (sink->client);
/* remove ports we don't need */
while (sink->port_count > channels) {
jack_port_unregister (client, sink->ports[--sink->port_count]);
}
/* alloc enough output ports */
sink->ports = g_realloc (sink->ports, sizeof (jack_port_t *) * channels);
/* create an output port for each channel */
while (sink->port_count < channels) {
gchar *name;
/* port names start from 1 and are local to the element */
name =
g_strdup_printf ("out_%s_%d", GST_ELEMENT_NAME (sink),
sink->port_count + 1);
sink->ports[sink->port_count] =
jack_port_register (client, name, JACK_DEFAULT_AUDIO_TYPE,
JackPortIsOutput, 0);
if (sink->ports[sink->port_count] == NULL)
return FALSE;
sink->port_count++;
g_free (name);
}
return TRUE;
}
static void
gst_jack_audio_sink_free_channels (GstJackAudioSink * sink)
{
gint res, i = 0;
jack_client_t *client;
client = gst_jack_audio_client_get_client (sink->client);
/* get rid of all ports */
while (sink->port_count) {
GST_LOG_OBJECT (sink, "unregister port %d", i);
if ((res = jack_port_unregister (client, sink->ports[i++]))) {
GST_DEBUG_OBJECT (sink, "unregister of port failed (%d)", res);
}
sink->port_count--;
}
g_free (sink->ports);
sink->ports = NULL;
}
/* ringbuffer abstract base class */
static GType
gst_jack_ring_buffer_get_type (void)
{
static GType ringbuffer_type = 0;
if (!ringbuffer_type) {
static const GTypeInfo ringbuffer_info = {
sizeof (GstJackRingBufferClass),
NULL,
NULL,
(GClassInitFunc) gst_jack_ring_buffer_class_init,
NULL,
NULL,
sizeof (GstJackRingBuffer),
0,
(GInstanceInitFunc) gst_jack_ring_buffer_init,
NULL
};
ringbuffer_type =
g_type_register_static (GST_TYPE_RING_BUFFER,
"GstJackAudioSinkRingBuffer", &ringbuffer_info, 0);
}
return ringbuffer_type;
}
static void
gst_jack_ring_buffer_class_init (GstJackRingBufferClass * klass)
{
GObjectClass *gobject_class;
GstObjectClass *gstobject_class;
GstRingBufferClass *gstringbuffer_class;
gobject_class = (GObjectClass *) klass;
gstobject_class = (GstObjectClass *) klass;
gstringbuffer_class = (GstRingBufferClass *) klass;
ring_parent_class = g_type_class_peek_parent (klass);
gstringbuffer_class->open_device =
GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_open_device);
gstringbuffer_class->close_device =
GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_close_device);
gstringbuffer_class->acquire =
GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_acquire);
gstringbuffer_class->release =
GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_release);
gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start);
gstringbuffer_class->pause = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_pause);
gstringbuffer_class->resume = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start);
gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_stop);
gstringbuffer_class->delay = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_delay);
}
/* this is the callback of jack. This should RT-safe.
*/
static int
jack_process_cb (jack_nframes_t nframes, void *arg)
{
GstJackAudioSink *sink;
GstRingBuffer *buf;
GstJackRingBuffer *abuf;
gint readseg, len;
guint8 *readptr;
gint i, j, flen, channels;
sample_t **buffers, *data;
buf = GST_RING_BUFFER_CAST (arg);
abuf = GST_JACK_RING_BUFFER_CAST (arg);
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
channels = buf->spec.channels;
/* alloc pointers to samples */
buffers = g_alloca (sizeof (sample_t *) * channels);
/* get target buffers */
for (i = 0; i < channels; i++) {
buffers[i] = (sample_t *) jack_port_get_buffer (sink->ports[i], nframes);
}
if (gst_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) {
flen = len / channels;
/* the number of samples must be exactly the segment size */
if (nframes * sizeof (sample_t) != flen)
goto wrong_size;
GST_DEBUG_OBJECT (sink, "copy %d frames: %p, %d bytes, %d channels",
nframes, readptr, flen, channels);
data = (sample_t *) readptr;
/* the samples in the ringbuffer have the channels interleaved, we need to
* deinterleave into the jack target buffers */
for (i = 0; i < nframes; i++) {
for (j = 0; j < channels; j++) {
buffers[j][i] = *data++;
}
}
/* clear written samples in the ringbuffer */
gst_ring_buffer_clear (buf, readseg);
/* we wrote one segment */
gst_ring_buffer_advance (buf, 1);
} else {
GST_DEBUG_OBJECT (sink, "write %d frames silence", nframes);
/* We are not allowed to read from the ringbuffer, write silence to all
* jack output buffers */
for (i = 0; i < channels; i++) {
memset (buffers[i], 0, nframes * sizeof (sample_t));
}
}
return 0;
/* ERRORS */
wrong_size:
{
GST_ERROR_OBJECT (sink, "nbytes (%d) != flen (%d)",
(gint) (nframes * sizeof (sample_t)), flen);
return 1;
}
}
/* we error out */
static int
jack_sample_rate_cb (jack_nframes_t nframes, void *arg)
{
GstJackAudioSink *sink;
GstJackRingBuffer *abuf;
abuf = GST_JACK_RING_BUFFER_CAST (arg);
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg));
if (abuf->sample_rate != -1 && abuf->sample_rate != nframes)
goto not_supported;
return 0;
/* ERRORS */
not_supported:
{
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS,
(NULL), ("Jack changed the sample rate, which is not supported"));
return 1;
}
}
/* we error out */
static int
jack_buffer_size_cb (jack_nframes_t nframes, void *arg)
{
GstJackAudioSink *sink;
GstJackRingBuffer *abuf;
abuf = GST_JACK_RING_BUFFER_CAST (arg);
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg));
if (abuf->buffer_size != -1 && abuf->buffer_size != nframes)
goto not_supported;
return 0;
/* ERRORS */
not_supported:
{
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS,
(NULL), ("Jack changed the buffer size, which is not supported"));
return 1;
}
}
static void
jack_shutdown_cb (void *arg)
{
GstJackAudioSink *sink;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg));
GST_DEBUG_OBJECT (sink, "shutdown");
GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
(NULL), ("Jack server shutdown"));
}
static void
gst_jack_ring_buffer_init (GstJackRingBuffer * buf,
GstJackRingBufferClass * g_class)
{
buf->channels = -1;
buf->buffer_size = -1;
buf->sample_rate = -1;
}
/* the _open_device method should make a connection with the server
*/
static gboolean
gst_jack_ring_buffer_open_device (GstRingBuffer * buf)
{
GstJackAudioSink *sink;
jack_status_t status = 0;
const gchar *name;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (sink, "open");
name = g_get_application_name ();
if (!name)
name = "GStreamer";
sink->client = gst_jack_audio_client_new (name, sink->server,
sink->jclient,
GST_JACK_CLIENT_SINK,
jack_shutdown_cb,
jack_process_cb, jack_buffer_size_cb, jack_sample_rate_cb, buf, &status);
if (sink->client == NULL)
goto could_not_open;
GST_DEBUG_OBJECT (sink, "opened");
return TRUE;
/* ERRORS */
could_not_open:
{
if (status & JackServerFailed) {
GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
(_("Jack server not found")),
("Cannot connect to the Jack server (status %d)", status));
} else {
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
(NULL), ("Jack client open error (status %d)", status));
}
return FALSE;
}
}
/* close the connection with the server
*/
static gboolean
gst_jack_ring_buffer_close_device (GstRingBuffer * buf)
{
GstJackAudioSink *sink;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (sink, "close");
gst_jack_audio_sink_free_channels (sink);
gst_jack_audio_client_free (sink->client);
sink->client = NULL;
return TRUE;
}
/* allocate a buffer and setup resources to process the audio samples of
* the format as specified in @spec.
*
* We allocate N jack ports, one for each channel. If we are asked to
* automatically make a connection with physical ports, we connect as many
* ports as there are physical ports, leaving leftover ports unconnected.
*
* It is assumed that samplerate and number of channels are acceptable since our
* getcaps method will always provide correct values. If unacceptable caps are
* received for some reason, we fail here.
*/
static gboolean
gst_jack_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
{
GstJackAudioSink *sink;
GstJackRingBuffer *abuf;
const char **ports;
gint sample_rate, buffer_size;
gint i, channels, res;
jack_client_t *client;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
abuf = GST_JACK_RING_BUFFER_CAST (buf);
GST_DEBUG_OBJECT (sink, "acquire");
client = gst_jack_audio_client_get_client (sink->client);
/* sample rate must be that of the server */
sample_rate = jack_get_sample_rate (client);
if (sample_rate != spec->rate)
goto wrong_samplerate;
channels = spec->channels;
if (!gst_jack_audio_sink_allocate_channels (sink, channels))
goto out_of_ports;
buffer_size = jack_get_buffer_size (client);
/* the segment size in bytes, this is large enough to hold a buffer of 32bit floats
* for all channels */
spec->segsize = buffer_size * sizeof (gfloat) * channels;
spec->latency_time = gst_util_uint64_scale (spec->segsize,
(GST_SECOND / GST_USECOND), spec->rate * spec->bytes_per_sample);
/* segtotal based on buffer-time latency */
spec->segtotal = spec->buffer_time / spec->latency_time;
if (spec->segtotal < 2) {
spec->segtotal = 2;
spec->buffer_time = spec->latency_time * spec->segtotal;
}
GST_DEBUG_OBJECT (sink, "buffer time: %" G_GINT64_FORMAT " usec",
spec->buffer_time);
GST_DEBUG_OBJECT (sink, "latency time: %" G_GINT64_FORMAT " usec",
spec->latency_time);
GST_DEBUG_OBJECT (sink, "buffer_size %d, segsize %d, segtotal %d",
buffer_size, spec->segsize, spec->segtotal);
/* allocate the ringbuffer memory now */
buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize);
memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data));
if ((res = gst_jack_audio_client_set_active (sink->client, TRUE)))
goto could_not_activate;
/* if we need to automatically connect the ports, do so now. We must do this
* after activating the client. */
if (sink->connect == GST_JACK_CONNECT_AUTO
|| sink->connect == GST_JACK_CONNECT_AUTO_FORCED) {
/* find all the physical input ports. A physical input port is a port
* associated with a hardware device. Someone needs connect to a physical
* port in order to hear something. */
ports = jack_get_ports (client, NULL, NULL,
JackPortIsPhysical | JackPortIsInput);
if (ports == NULL) {
/* no ports? fine then we don't do anything except for posting a warning
* message. */
GST_ELEMENT_WARNING (sink, RESOURCE, NOT_FOUND, (NULL),
("No physical input ports found, leaving ports unconnected"));
goto done;
}
for (i = 0; i < channels; i++) {
/* stop when all input ports are exhausted */
if (ports[i] == NULL) {
/* post a warning that we could not connect all ports */
GST_ELEMENT_WARNING (sink, RESOURCE, NOT_FOUND, (NULL),
("No more physical ports, leaving some ports unconnected"));
break;
}
GST_DEBUG_OBJECT (sink, "try connecting to %s",
jack_port_name (sink->ports[i]));
/* connect the port to a physical port */
res = jack_connect (client, jack_port_name (sink->ports[i]), ports[i]);
if (res != 0 && res != EEXIST)
goto cannot_connect;
}
free (ports);
}
done:
abuf->sample_rate = sample_rate;
abuf->buffer_size = buffer_size;
abuf->channels = spec->channels;
return TRUE;
/* ERRORS */
wrong_samplerate:
{
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("Wrong samplerate, server is running at %d and we received %d",
sample_rate, spec->rate));
return FALSE;
}
out_of_ports:
{
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("Cannot allocate more Jack ports"));
return FALSE;
}
could_not_activate:
{
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("Could not activate client (%d:%s)", res, g_strerror (res)));
return FALSE;
}
cannot_connect:
{
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("Could not connect output ports to physical ports (%d:%s)",
res, g_strerror (res)));
free (ports);
return FALSE;
}
}
/* function is called with LOCK */
static gboolean
gst_jack_ring_buffer_release (GstRingBuffer * buf)
{
GstJackAudioSink *sink;
GstJackRingBuffer *abuf;
gint res;
abuf = GST_JACK_RING_BUFFER_CAST (buf);
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (sink, "release");
if ((res = gst_jack_audio_client_set_active (sink->client, FALSE))) {
/* we only warn, this means the server is probably shut down and the client
* is gone anyway. */
GST_ELEMENT_WARNING (sink, RESOURCE, CLOSE, (NULL),
("Could not deactivate Jack client (%d)", res));
}
abuf->channels = -1;
abuf->buffer_size = -1;
abuf->sample_rate = -1;
/* free the buffer */
gst_buffer_unref (buf->data);
buf->data = NULL;
return TRUE;
}
static gboolean
gst_jack_ring_buffer_start (GstRingBuffer * buf)
{
GstJackAudioSink *sink;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (sink, "start");
return TRUE;
}
static gboolean
gst_jack_ring_buffer_pause (GstRingBuffer * buf)
{
GstJackAudioSink *sink;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (sink, "pause");
return TRUE;
}
static gboolean
gst_jack_ring_buffer_stop (GstRingBuffer * buf)
{
GstJackAudioSink *sink;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (sink, "stop");
return TRUE;
}
static guint
gst_jack_ring_buffer_delay (GstRingBuffer * buf)
{
GstJackAudioSink *sink;
guint i, res = 0, latency;
jack_client_t *client;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
client = gst_jack_audio_client_get_client (sink->client);
for (i = 0; i < sink->port_count; i++) {
latency = jack_port_get_total_latency (client, sink->ports[i]);
if (latency > res)
res = latency;
}
GST_LOG_OBJECT (sink, "delay %u", res);
return res;
}
static GstStaticPadTemplate jackaudiosink_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-float, "
"endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, "
"width = (int) 32, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
);
/* AudioSink signals and args */
enum
{
/* FILL ME */
SIGNAL_LAST
};
#define DEFAULT_PROP_CONNECT GST_JACK_CONNECT_AUTO
#define DEFAULT_PROP_SERVER NULL
enum
{
PROP_0,
PROP_CONNECT,
PROP_SERVER,
PROP_CLIENT,
PROP_LAST
};
#define _do_init(bla) \
GST_DEBUG_CATEGORY_INIT (gst_jack_audio_sink_debug, "jacksink", 0, "jacksink element");
GST_BOILERPLATE_FULL (GstJackAudioSink, gst_jack_audio_sink, GstBaseAudioSink,
GST_TYPE_BASE_AUDIO_SINK, _do_init);
static void gst_jack_audio_sink_dispose (GObject * object);
static void gst_jack_audio_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_jack_audio_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstCaps *gst_jack_audio_sink_getcaps (GstBaseSink * bsink);
static GstRingBuffer *gst_jack_audio_sink_create_ringbuffer (GstBaseAudioSink *
sink);
static void
gst_jack_audio_sink_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_set_details_simple (element_class, "Audio Sink (Jack)",
"Sink/Audio", "Output to Jack", "Wim Taymans <wim@fluendo.com>");
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&jackaudiosink_sink_factory));
}
static void
gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSinkClass *gstbasesink_class;
GstBaseAudioSinkClass *gstbaseaudiosink_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesink_class = (GstBaseSinkClass *) klass;
gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass;
gobject_class->dispose = gst_jack_audio_sink_dispose;
gobject_class->get_property = gst_jack_audio_sink_get_property;
gobject_class->set_property = gst_jack_audio_sink_set_property;
g_object_class_install_property (gobject_class, PROP_CONNECT,
g_param_spec_enum ("connect", "Connect",
"Specify how the output ports will be connected",
GST_TYPE_JACK_CONNECT, DEFAULT_PROP_CONNECT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_SERVER,
g_param_spec_string ("server", "Server",
"The Jack server to connect to (NULL = default)",
DEFAULT_PROP_SERVER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_CLIENT,
g_param_spec_boxed ("client", "JackClient", "Handle for jack client",
GST_TYPE_JACK_CLIENT,
GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
G_PARAM_STATIC_STRINGS));
gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_sink_getcaps);
gstbaseaudiosink_class->create_ringbuffer =
GST_DEBUG_FUNCPTR (gst_jack_audio_sink_create_ringbuffer);
/* ref class from a thread-safe context to work around missing bit of
* thread-safety in GObject */
g_type_class_ref (GST_TYPE_JACK_RING_BUFFER);
gst_jack_audio_client_init ();
}
static void
gst_jack_audio_sink_init (GstJackAudioSink * sink,
GstJackAudioSinkClass * g_class)
{
sink->connect = DEFAULT_PROP_CONNECT;
sink->server = g_strdup (DEFAULT_PROP_SERVER);
sink->jclient = NULL;
sink->ports = NULL;
sink->port_count = 0;
}
static void
gst_jack_audio_sink_dispose (GObject * object)
{
GstJackAudioSink *sink = GST_JACK_AUDIO_SINK (object);
gst_caps_replace (&sink->caps, NULL);
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_jack_audio_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstJackAudioSink *sink;
sink = GST_JACK_AUDIO_SINK (object);
switch (prop_id) {
case PROP_CONNECT:
sink->connect = g_value_get_enum (value);
break;
case PROP_SERVER:
g_free (sink->server);
sink->server = g_value_dup_string (value);
break;
case PROP_CLIENT:
if (GST_STATE (sink) == GST_STATE_NULL ||
GST_STATE (sink) == GST_STATE_READY) {
sink->jclient = g_value_get_boxed (value);
}
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_jack_audio_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstJackAudioSink *sink;
sink = GST_JACK_AUDIO_SINK (object);
switch (prop_id) {
case PROP_CONNECT:
g_value_set_enum (value, sink->connect);
break;
case PROP_SERVER:
g_value_set_string (value, sink->server);
break;
case PROP_CLIENT:
g_value_set_boxed (value, sink->jclient);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstCaps *
gst_jack_audio_sink_getcaps (GstBaseSink * bsink)
{
GstJackAudioSink *sink = GST_JACK_AUDIO_SINK (bsink);
const char **ports;
gint min, max;
gint rate;
jack_client_t *client;
if (sink->client == NULL)
goto no_client;
client = gst_jack_audio_client_get_client (sink->client);
if (sink->connect == GST_JACK_CONNECT_AUTO) {
/* get a port count, this is the number of channels we can automatically
* connect. */
ports = jack_get_ports (client, NULL, NULL,
JackPortIsPhysical | JackPortIsInput);
max = 0;
if (ports != NULL) {
for (; ports[max]; max++);
free (ports);
} else
max = 0;
} else {
/* we allow any number of pads, something else is going to connect the
* pads. */
max = G_MAXINT;
}
min = MIN (1, max);
rate = jack_get_sample_rate (client);
GST_DEBUG_OBJECT (sink, "got %d-%d ports, samplerate: %d", min, max, rate);
if (!sink->caps) {
sink->caps = gst_caps_new_simple ("audio/x-raw-float",
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"width", G_TYPE_INT, 32,
"rate", G_TYPE_INT, rate,
"channels", GST_TYPE_INT_RANGE, min, max, NULL);
}
GST_INFO_OBJECT (sink, "returning caps %" GST_PTR_FORMAT, sink->caps);
return gst_caps_ref (sink->caps);
/* ERRORS */
no_client:
{
GST_DEBUG_OBJECT (sink, "device not open, using template caps");
/* base class will get template caps for us when we return NULL */
return NULL;
}
}
static GstRingBuffer *
gst_jack_audio_sink_create_ringbuffer (GstBaseAudioSink * sink)
{
GstRingBuffer *buffer;
buffer = g_object_new (GST_TYPE_JACK_RING_BUFFER, NULL);
GST_DEBUG_OBJECT (sink, "created ringbuffer @%p", buffer);
return buffer;
}

78
ext/jack/gstjackaudiosink.h
View file

@ -1,78 +0,0 @@
/* GStreamer
* Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
*
* gstjacksink.h:
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __GST_JACK_AUDIO_SINK_H__
#define __GST_JACK_AUDIO_SINK_H__
#include <jack/jack.h>
#include <gst/gst.h>
#include <gst/audio/gstbaseaudiosink.h>
#include "gstjack.h"
#include "gstjackaudioclient.h"
G_BEGIN_DECLS
#define GST_TYPE_JACK_AUDIO_SINK (gst_jack_audio_sink_get_type())
#define GST_JACK_AUDIO_SINK(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_JACK_AUDIO_SINK,GstJackAudioSink))
#define GST_JACK_AUDIO_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_JACK_AUDIO_SINK,GstJackAudioSinkClass))
#define GST_JACK_AUDIO_SINK_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_JACK_AUDIO_SINK,GstJackAudioSinkClass))
#define GST_IS_JACK_AUDIO_SINK(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_JACK_AUDIO_SINK))
#define GST_IS_JACK_AUDIO_SINK_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_JACK_AUDIO_SINK))
typedef struct _GstJackAudioSink GstJackAudioSink;
typedef struct _GstJackAudioSinkClass GstJackAudioSinkClass;
/**
* GstJackAudioSink:
*
* Opaque #GstJackAudioSink.
*/
struct _GstJackAudioSink {
GstBaseAudioSink element;
/*< private >*/
/* cached caps */
GstCaps *caps;
/* properties */
GstJackConnect connect;
gchar *server;
jack_client_t *jclient;
/* our client */
GstJackAudioClient *client;
/* our ports */
jack_port_t **ports;
int port_count;
};
struct _GstJackAudioSinkClass {
GstBaseAudioSinkClass parent_class;
};
GType gst_jack_audio_sink_get_type (void);
G_END_DECLS
#endif /* __GST_JACK_AUDIO_SINK_H__ */

874
ext/jack/gstjackaudiosrc.c
View file

@ -1,874 +0,0 @@
/* GStreamer
* Copyright (C) 2008 Tristan Matthews <tristan@sat.qc.ca>
*
* Permission is hereby granted, free of charge, to any person obtaining a
* copy of this software and associated documentation files (the "Software"),
* to deal in the Software without restriction, including without limitation
* the rights to use, copy, modify, merge, publish, distribute, sublicense,
* and/or sell copies of the Software, and to permit persons to whom the
* Software is furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
* DEALINGS IN THE SOFTWARE.
*
* Alternatively, the contents of this file may be used under the
* GNU Lesser General Public License Version 2.1 (the "LGPL"), in
* which case the following provisions apply instead of the ones
* mentioned above:
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-jackaudiosrc
* @see_also: #GstBaseAudioSrc, #GstRingBuffer
*
* A Src that inputs data from Jack ports.
*
* It will create N Jack ports named in_&lt;name&gt;_&lt;num&gt; where
* &lt;name&gt; is the element name and &lt;num&gt; is starting from 1.
* Each port corresponds to a gstreamer channel.
*
* The samplerate as exposed on the caps is always the same as the samplerate of
* the jack server.
*
* When the #GstJackAudioSrc:connect property is set to auto, this element
* will try to connect each input port to a random physical jack output pin.
*
* When the #GstJackAudioSrc:connect property is set to none, the element will
* accept any number of output channels and will create (but not connect) an
* input port for each channel.
*
* The element will generate an error when the Jack server is shut down when it
* was PAUSED or PLAYING. This element does not support dynamic rate and buffer
* size changes at runtime.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch jackaudiosrc connect=0 ! jackaudiosink connect=0
* ]| Get audio input into gstreamer from jack.
* </refsect2>
*
* Last reviewed on 2008-07-22 (0.10.4)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst-i18n-plugin.h>
#include <stdlib.h>
#include <string.h>
#include "gstjackaudiosrc.h"
#include "gstjackringbuffer.h"
#include "gstjackutil.h"
GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_src_debug);
#define GST_CAT_DEFAULT gst_jack_audio_src_debug
static gboolean
gst_jack_audio_src_allocate_channels (GstJackAudioSrc * src, gint channels)
{
jack_client_t *client;
client = gst_jack_audio_client_get_client (src->client);
/* remove ports we don't need */
while (src->port_count > channels)
jack_port_unregister (client, src->ports[--src->port_count]);
/* alloc enough input ports */
src->ports = g_realloc (src->ports, sizeof (jack_port_t *) * channels);
src->buffers = g_realloc (src->buffers, sizeof (sample_t *) * channels);
/* create an input port for each channel */
while (src->port_count < channels) {
gchar *name;
/* port names start from 1 and are local to the element */
name =
g_strdup_printf ("in_%s_%d", GST_ELEMENT_NAME (src),
src->port_count + 1);
src->ports[src->port_count] =
jack_port_register (client, name, JACK_DEFAULT_AUDIO_TYPE,
JackPortIsInput, 0);
if (src->ports[src->port_count] == NULL)
return FALSE;
src->port_count++;
g_free (name);
}
return TRUE;
}
static void
gst_jack_audio_src_free_channels (GstJackAudioSrc * src)
{
gint res, i = 0;
jack_client_t *client;
client = gst_jack_audio_client_get_client (src->client);
/* get rid of all ports */
while (src->port_count) {
GST_LOG_OBJECT (src, "unregister port %d", i);
if ((res = jack_port_unregister (client, src->ports[i++])))
GST_DEBUG_OBJECT (src, "unregister of port failed (%d)", res);
src->port_count--;
}
g_free (src->ports);
src->ports = NULL;
g_free (src->buffers);
src->buffers = NULL;
}
/* ringbuffer abstract base class */
static GType
gst_jack_ring_buffer_get_type (void)
{
static GType ringbuffer_type = 0;
if (!ringbuffer_type) {
static const GTypeInfo ringbuffer_info = { sizeof (GstJackRingBufferClass),
NULL,
NULL,
(GClassInitFunc) gst_jack_ring_buffer_class_init,
NULL,
NULL,
sizeof (GstJackRingBuffer),
0,
(GInstanceInitFunc) gst_jack_ring_buffer_init,
NULL
};
ringbuffer_type =
g_type_register_static (GST_TYPE_RING_BUFFER,
"GstJackAudioSrcRingBuffer", &ringbuffer_info, 0);
}
return ringbuffer_type;
}
static void
gst_jack_ring_buffer_class_init (GstJackRingBufferClass * klass)
{
GObjectClass *gobject_class;
GstObjectClass *gstobject_class;
GstRingBufferClass *gstringbuffer_class;
gobject_class = (GObjectClass *) klass;
gstobject_class = (GstObjectClass *) klass;
gstringbuffer_class = (GstRingBufferClass *) klass;
ring_parent_class = g_type_class_peek_parent (klass);
gstringbuffer_class->open_device =
GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_open_device);
gstringbuffer_class->close_device =
GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_close_device);
gstringbuffer_class->acquire =
GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_acquire);
gstringbuffer_class->release =
GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_release);
gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start);
gstringbuffer_class->pause = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_pause);
gstringbuffer_class->resume = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start);
gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_stop);
gstringbuffer_class->delay = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_delay);
}
/* this is the callback of jack. This should be RT-safe.
* Writes samples from the jack input port's buffer to the gst ring buffer.
*/
static int
jack_process_cb (jack_nframes_t nframes, void *arg)
{
GstJackAudioSrc *src;
GstRingBuffer *buf;
gint len;
guint8 *writeptr;
gint writeseg;
gint channels, i, j, flen;
sample_t *data;
buf = GST_RING_BUFFER_CAST (arg);
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
channels = buf->spec.channels;
/* get input buffers */
for (i = 0; i < channels; i++)
src->buffers[i] =
(sample_t *) jack_port_get_buffer (src->ports[i], nframes);
if (gst_ring_buffer_prepare_read (buf, &writeseg, &writeptr, &len)) {
flen = len / channels;
/* the number of samples must be exactly the segment size */
if (nframes * sizeof (sample_t) != flen)
goto wrong_size;
/* the samples in the jack input buffers have to be interleaved into the
* ringbuffer */
data = (sample_t *) writeptr;
for (i = 0; i < nframes; ++i)
for (j = 0; j < channels; ++j)
*data++ = src->buffers[j][i];
GST_DEBUG ("copy %d frames: %p, %d bytes, %d channels", nframes, writeptr,
len / channels, channels);
/* we wrote one segment */
gst_ring_buffer_advance (buf, 1);
}
return 0;
/* ERRORS */
wrong_size:
{
GST_ERROR_OBJECT (src, "nbytes (%d) != flen (%d)",
(gint) (nframes * sizeof (sample_t)), flen);
return 1;
}
}
/* we error out */
static int
jack_sample_rate_cb (jack_nframes_t nframes, void *arg)
{
GstJackAudioSrc *src;
GstJackRingBuffer *abuf;
abuf = GST_JACK_RING_BUFFER_CAST (arg);
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg));
if (abuf->sample_rate != -1 && abuf->sample_rate != nframes)
goto not_supported;
return 0;
/* ERRORS */
not_supported:
{
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS,
(NULL), ("Jack changed the sample rate, which is not supported"));
return 1;
}
}
/* we error out */
static int
jack_buffer_size_cb (jack_nframes_t nframes, void *arg)
{
GstJackAudioSrc *src;
GstJackRingBuffer *abuf;
abuf = GST_JACK_RING_BUFFER_CAST (arg);
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg));
if (abuf->buffer_size != -1 && abuf->buffer_size != nframes)
goto not_supported;
return 0;
/* ERRORS */
not_supported:
{
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS,
(NULL), ("Jack changed the buffer size, which is not supported"));
return 1;
}
}
static void
jack_shutdown_cb (void *arg)
{
GstJackAudioSrc *src;
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg));
GST_DEBUG_OBJECT (src, "shutdown");
GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND,
(NULL), ("Jack server shutdown"));
}
static void
gst_jack_ring_buffer_init (GstJackRingBuffer * buf,
GstJackRingBufferClass * g_class)
{
buf->channels = -1;
buf->buffer_size = -1;
buf->sample_rate = -1;
}
/* the _open_device method should make a connection with the server
*/
static gboolean
gst_jack_ring_buffer_open_device (GstRingBuffer * buf)
{
GstJackAudioSrc *src;
jack_status_t status = 0;
const gchar *name;
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (src, "open");
name = g_get_application_name ();
if (!name)
name = "GStreamer";
src->client = gst_jack_audio_client_new (name, src->server,
src->jclient,
GST_JACK_CLIENT_SOURCE,
jack_shutdown_cb,
jack_process_cb, jack_buffer_size_cb, jack_sample_rate_cb, buf, &status);
if (src->client == NULL)
goto could_not_open;
GST_DEBUG_OBJECT (src, "opened");
return TRUE;
/* ERRORS */
could_not_open:
{
if (status & JackServerFailed) {
GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND,
(_("Jack server not found")),
("Cannot connect to the Jack server (status %d)", status));
} else {
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_WRITE,
(NULL), ("Jack client open error (status %d)", status));
}
return FALSE;
}
}
/* close the connection with the server
*/
static gboolean
gst_jack_ring_buffer_close_device (GstRingBuffer * buf)
{
GstJackAudioSrc *src;
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (src, "close");
gst_jack_audio_src_free_channels (src);
gst_jack_audio_client_free (src->client);
src->client = NULL;
return TRUE;
}
/* allocate a buffer and setup resources to process the audio samples of
* the format as specified in @spec.
*
* We allocate N jack ports, one for each channel. If we are asked to
* automatically make a connection with physical ports, we connect as many
* ports as there are physical ports, leaving leftover ports unconnected.
*
* It is assumed that samplerate and number of channels are acceptable since our
* getcaps method will always provide correct values. If unacceptable caps are
* received for some reason, we fail here.
*/
static gboolean
gst_jack_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
{
GstJackAudioSrc *src;
GstJackRingBuffer *abuf;
const char **ports;
gint sample_rate, buffer_size;
gint i, channels, res;
jack_client_t *client;
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
abuf = GST_JACK_RING_BUFFER_CAST (buf);
GST_DEBUG_OBJECT (src, "acquire");
client = gst_jack_audio_client_get_client (src->client);
/* sample rate must be that of the server */
sample_rate = jack_get_sample_rate (client);
if (sample_rate != spec->rate)
goto wrong_samplerate;
channels = spec->channels;
if (!gst_jack_audio_src_allocate_channels (src, channels))
goto out_of_ports;
gst_jack_set_layout_on_caps (&spec->caps, channels);
buffer_size = jack_get_buffer_size (client);
/* the segment size in bytes, this is large enough to hold a buffer of 32bit floats
* for all channels */
spec->segsize = buffer_size * sizeof (gfloat) * channels;
spec->latency_time = gst_util_uint64_scale (spec->segsize,
(GST_SECOND / GST_USECOND), spec->rate * spec->bytes_per_sample);
/* segtotal based on buffer-time latency */
spec->segtotal = spec->buffer_time / spec->latency_time;
if (spec->segtotal < 2) {
spec->segtotal = 2;
spec->buffer_time = spec->latency_time * spec->segtotal;
}
GST_DEBUG_OBJECT (src, "buffer time: %" G_GINT64_FORMAT " usec",
spec->buffer_time);
GST_DEBUG_OBJECT (src, "latency time: %" G_GINT64_FORMAT " usec",
spec->latency_time);
GST_DEBUG_OBJECT (src, "buffer_size %d, segsize %d, segtotal %d",
buffer_size, spec->segsize, spec->segtotal);
/* allocate the ringbuffer memory now */
buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize);
memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data));
if ((res = gst_jack_audio_client_set_active (src->client, TRUE)))
goto could_not_activate;
/* if we need to automatically connect the ports, do so now. We must do this
* after activating the client. */
if (src->connect == GST_JACK_CONNECT_AUTO
|| src->connect == GST_JACK_CONNECT_AUTO_FORCED) {
/* find all the physical output ports. A physical output port is a port
* associated with a hardware device. Someone needs connect to a physical
* port in order to capture something. */
ports =
jack_get_ports (client, NULL, NULL,
JackPortIsPhysical | JackPortIsOutput);
if (ports == NULL) {
/* no ports? fine then we don't do anything except for posting a warning
* message. */
GST_ELEMENT_WARNING (src, RESOURCE, NOT_FOUND, (NULL),
("No physical output ports found, leaving ports unconnected"));
goto done;
}
for (i = 0; i < channels; i++) {
/* stop when all output ports are exhausted */
if (ports[i] == NULL) {
/* post a warning that we could not connect all ports */
GST_ELEMENT_WARNING (src, RESOURCE, NOT_FOUND, (NULL),
("No more physical ports, leaving some ports unconnected"));
break;
}
GST_DEBUG_OBJECT (src, "try connecting to %s",
jack_port_name (src->ports[i]));
/* connect the physical port to a port */
res = jack_connect (client, ports[i], jack_port_name (src->ports[i]));
if (res != 0 && res != EEXIST)
goto cannot_connect;
}
free (ports);
}
done:
abuf->sample_rate = sample_rate;
abuf->buffer_size = buffer_size;
abuf->channels = spec->channels;
return TRUE;
/* ERRORS */
wrong_samplerate:
{
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("Wrong samplerate, server is running at %d and we received %d",
sample_rate, spec->rate));
return FALSE;
}
out_of_ports:
{
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("Cannot allocate more Jack ports"));
return FALSE;
}
could_not_activate:
{
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("Could not activate client (%d:%s)", res, g_strerror (res)));
return FALSE;
}
cannot_connect:
{
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("Could not connect input ports to physical ports (%d:%s)",
res, g_strerror (res)));
free (ports);
return FALSE;
}
}
/* function is called with LOCK */
static gboolean
gst_jack_ring_buffer_release (GstRingBuffer * buf)
{
GstJackAudioSrc *src;
GstJackRingBuffer *abuf;
gint res;
abuf = GST_JACK_RING_BUFFER_CAST (buf);
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (src, "release");
if ((res = gst_jack_audio_client_set_active (src->client, FALSE))) {
/* we only warn, this means the server is probably shut down and the client
* is gone anyway. */
GST_ELEMENT_WARNING (src, RESOURCE, CLOSE, (NULL),
("Could not deactivate Jack client (%d)", res));
}
abuf->channels = -1;
abuf->buffer_size = -1;
abuf->sample_rate = -1;
/* free the buffer */
gst_buffer_unref (buf->data);
buf->data = NULL;
return TRUE;
}
static gboolean
gst_jack_ring_buffer_start (GstRingBuffer * buf)
{
GstJackAudioSrc *src;
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (src, "start");
return TRUE;
}
static gboolean
gst_jack_ring_buffer_pause (GstRingBuffer * buf)
{
GstJackAudioSrc *src;
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (src, "pause");
return TRUE;
}
static gboolean
gst_jack_ring_buffer_stop (GstRingBuffer * buf)
{
GstJackAudioSrc *src;
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (src, "stop");
return TRUE;
}
static guint
gst_jack_ring_buffer_delay (GstRingBuffer * buf)
{
GstJackAudioSrc *src;
guint i, res = 0, latency;
jack_client_t *client;
src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
client = gst_jack_audio_client_get_client (src->client);
for (i = 0; i < src->port_count; i++) {
latency = jack_port_get_total_latency (client, src->ports[i]);
if (latency > res)
res = latency;
}
GST_DEBUG_OBJECT (src, "delay %u", res);
return res;
}
/* Audiosrc signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
#define DEFAULT_PROP_CONNECT GST_JACK_CONNECT_AUTO
#define DEFAULT_PROP_SERVER NULL
enum
{
PROP_0,
PROP_CONNECT,
PROP_SERVER,
PROP_CLIENT,
PROP_LAST
};
/* the capabilities of the inputs and outputs.
*
* describe the real formats here.
*/
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-float, "
"endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, "
"width = (int) 32, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
);
#define _do_init(bla) \
GST_DEBUG_CATEGORY_INIT(gst_jack_audio_src_debug, "jacksrc", 0, "jacksrc element");
GST_BOILERPLATE_FULL (GstJackAudioSrc, gst_jack_audio_src, GstBaseAudioSrc,
GST_TYPE_BASE_AUDIO_SRC, _do_init);
static void gst_jack_audio_src_dispose (GObject * object);
static void gst_jack_audio_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_jack_audio_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstCaps *gst_jack_audio_src_getcaps (GstBaseSrc * bsrc);
static GstRingBuffer *gst_jack_audio_src_create_ringbuffer (GstBaseAudioSrc *
src);
/* GObject vmethod implementations */
static void
gst_jack_audio_src_base_init (gpointer gclass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (gclass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_factory));
gst_element_class_set_details_simple (element_class, "Audio Source (Jack)",
"Source/Audio",
"Input from Jack", "Tristan Matthews <tristan@sat.qc.ca>");
}
/* initialize the jack_audio_src's class */
static void
gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSrcClass *gstbasesrc_class;
GstBaseAudioSrcClass *gstbaseaudiosrc_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesrc_class = (GstBaseSrcClass *) klass;
gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass;
gobject_class->dispose = gst_jack_audio_src_dispose;
gobject_class->set_property = gst_jack_audio_src_set_property;
gobject_class->get_property = gst_jack_audio_src_get_property;
g_object_class_install_property (gobject_class, PROP_CONNECT,
g_param_spec_enum ("connect", "Connect",
"Specify how the input ports will be connected",
GST_TYPE_JACK_CONNECT, DEFAULT_PROP_CONNECT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_SERVER,
g_param_spec_string ("server", "Server",
"The Jack server to connect to (NULL = default)",
DEFAULT_PROP_SERVER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_CLIENT,
g_param_spec_boxed ("client", "JackClient", "Handle for jack client",
GST_TYPE_JACK_CLIENT,
GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
G_PARAM_STATIC_STRINGS));
gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_src_getcaps);
gstbaseaudiosrc_class->create_ringbuffer =
GST_DEBUG_FUNCPTR (gst_jack_audio_src_create_ringbuffer);
/* ref class from a thread-safe context to work around missing bit of
* thread-safety in GObject */
g_type_class_ref (GST_TYPE_JACK_RING_BUFFER);
gst_jack_audio_client_init ();
}
/* initialize the new element
* instantiate pads and add them to element
* set pad calback functions
* initialize instance structure
*/
static void
gst_jack_audio_src_init (GstJackAudioSrc * src, GstJackAudioSrcClass * gclass)
{
//gst_base_src_set_live(GST_BASE_SRC (src), TRUE);
src->connect = DEFAULT_PROP_CONNECT;
src->server = g_strdup (DEFAULT_PROP_SERVER);
src->jclient = NULL;
src->ports = NULL;
src->port_count = 0;
src->buffers = NULL;
}
static void
gst_jack_audio_src_dispose (GObject * object)
{
GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
gst_caps_replace (&src->caps, NULL);
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_jack_audio_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
switch (prop_id) {
case PROP_CONNECT:
src->connect = g_value_get_enum (value);
break;
case PROP_SERVER:
g_free (src->server);
src->server = g_value_dup_string (value);
break;
case PROP_CLIENT:
if (GST_STATE (src) == GST_STATE_NULL ||
GST_STATE (src) == GST_STATE_READY) {
src->jclient = g_value_get_boxed (value);
}
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_jack_audio_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
switch (prop_id) {
case PROP_CONNECT:
g_value_set_enum (value, src->connect);
break;
case PROP_SERVER:
g_value_set_string (value, src->server);
break;
case PROP_CLIENT:
g_value_set_boxed (value, src->jclient);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstCaps *
gst_jack_audio_src_getcaps (GstBaseSrc * bsrc)
{
GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (bsrc);
const char **ports;
gint min, max;
gint rate;
jack_client_t *client;
if (src->client == NULL)
goto no_client;
client = gst_jack_audio_client_get_client (src->client);
if (src->connect == GST_JACK_CONNECT_AUTO) {
/* get a port count, this is the number of channels we can automatically
* connect. */
ports = jack_get_ports (client, NULL, NULL,
JackPortIsPhysical | JackPortIsOutput);
max = 0;
if (ports != NULL) {
for (; ports[max]; max++);
free (ports);
} else
max = 0;
} else {
/* we allow any number of pads, something else is going to connect the
* pads. */
max = G_MAXINT;
}
min = MIN (1, max);
rate = jack_get_sample_rate (client);
GST_DEBUG_OBJECT (src, "got %d-%d ports, samplerate: %d", min, max, rate);
if (!src->caps) {
src->caps = gst_caps_new_simple ("audio/x-raw-float",
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"width", G_TYPE_INT, 32,
"rate", G_TYPE_INT, rate,
"channels", GST_TYPE_INT_RANGE, min, max, NULL);
}
GST_INFO_OBJECT (src, "returning caps %" GST_PTR_FORMAT, src->caps);
return gst_caps_ref (src->caps);
/* ERRORS */
no_client:
{
GST_DEBUG_OBJECT (src, "device not open, using template caps");
/* base class will get template caps for us when we return NULL */
return NULL;
}
}
static GstRingBuffer *
gst_jack_audio_src_create_ringbuffer (GstBaseAudioSrc * src)
{
GstRingBuffer *buffer;
buffer = g_object_new (GST_TYPE_JACK_RING_BUFFER, NULL);
GST_DEBUG_OBJECT (src, "created ringbuffer @%p", buffer);
return buffer;
}

97
ext/jack/gstjackaudiosrc.h
View file

@ -1,97 +0,0 @@
/* GStreamer
* Copyright (C) 2008 Tristan Matthews <tristan@sat.qc.ca>
*
* Permission is hereby granted, free of charge, to any person obtaining a
* copy of this software and associated documentation files (the "Software"),
* to deal in the Software without restriction, including without limitation
* the rights to use, copy, modify, merge, publish, distribute, sublicense,
* and/or sell copies of the Software, and to permit persons to whom the
* Software is furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
* DEALINGS IN THE SOFTWARE.
*
* Alternatively, the contents of this file may be used under the
* GNU Lesser General Public License Version 2.1 (the "LGPL"), in
* which case the following provisions apply instead of the ones
* mentioned above:
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __GST_JACK_AUDIO_SRC_H__
#define __GST_JACK_AUDIO_SRC_H__
#include <jack/jack.h>
#include <gst/gst.h>
#include <gst/audio/gstaudiosrc.h>
#include "gstjackaudioclient.h"
#include "gstjack.h"
G_BEGIN_DECLS
#define GST_TYPE_JACK_AUDIO_SRC (gst_jack_audio_src_get_type())
#define GST_JACK_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_JACK_AUDIO_SRC,GstJackAudioSrc))
#define GST_JACK_AUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_JACK_AUDIO_SRC,GstJackAudioSrcClass))
#define GST_JACK_AUDIO_SRC_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_JACK_AUDIO_SRC,GstJackAudioSrcClass))
#define GST_IS_JACK_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_JACK_AUDIO_SRC))
#define GST_IS_JACK_AUDIO_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_JACK_AUDIO_SRC))
typedef struct _GstJackAudioSrc GstJackAudioSrc;
typedef struct _GstJackAudioSrcClass GstJackAudioSrcClass;
struct _GstJackAudioSrc
{
GstBaseAudioSrc src;
/*< private >*/
/* cached caps */
GstCaps *caps;
/* properties */
GstJackConnect connect;
gchar *server;
jack_client_t *jclient;
/* our client */
GstJackAudioClient *client;
/* our ports */
jack_port_t **ports;
int port_count;
sample_t **buffers;
};
struct _GstJackAudioSrcClass
{
GstBaseAudioSrcClass parent_class;
};
GType gst_jack_audio_src_get_type (void);
G_END_DECLS
#endif /* __GST_JACK_AUDIO_SRC_H__ */

88
ext/jack/gstjackringbuffer.h
View file

@ -1,88 +0,0 @@
/*
* GStreamer
* Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
* Copyright (C) 2008 Tristan Matthews <tristan@sat.qc.ca>
*
* Permission is hereby granted, free of charge, to any person obtaining a
* copy of this software and associated documentation files (the "Software"),
* to deal in the Software without restriction, including without limitation
* the rights to use, copy, modify, merge, publish, distribute, sublicense,
* and/or sell copies of the Software, and to permit persons to whom the
* Software is furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
* DEALINGS IN THE SOFTWARE.
*
* Alternatively, the contents of this file may be used under the
* GNU Lesser General Public License Version 2.1 (the "LGPL"), in
* which case the following provisions apply instead of the ones
* mentioned above:
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __GST_JACK_RING_BUFFER_H__
#define __GST_JACK_RING_BUFFER_H__
#define GST_TYPE_JACK_RING_BUFFER (gst_jack_ring_buffer_get_type())
#define GST_JACK_RING_BUFFER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_JACK_RING_BUFFER,GstJackRingBuffer))
#define GST_JACK_RING_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_JACK_RING_BUFFER,GstJackRingBufferClass))
#define GST_JACK_RING_BUFFER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_JACK_RING_BUFFER,GstJackRingBufferClass))
#define GST_JACK_RING_BUFFER_CAST(obj) ((GstJackRingBuffer *)obj)
#define GST_IS_JACK_RING_BUFFER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_JACK_RING_BUFFER))
#define GST_IS_JACK_RING_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_JACK_RING_BUFFER))
typedef struct _GstJackRingBuffer GstJackRingBuffer;
typedef struct _GstJackRingBufferClass GstJackRingBufferClass;
struct _GstJackRingBuffer
{
GstRingBuffer object;
gint sample_rate;
gint buffer_size;
gint channels;
};
struct _GstJackRingBufferClass
{
GstRingBufferClass parent_class;
};
static void gst_jack_ring_buffer_class_init(GstJackRingBufferClass * klass);
static void gst_jack_ring_buffer_init(GstJackRingBuffer * ringbuffer,
GstJackRingBufferClass * klass);
static GstRingBufferClass *ring_parent_class = NULL;
static gboolean gst_jack_ring_buffer_open_device(GstRingBuffer * buf);
static gboolean gst_jack_ring_buffer_close_device(GstRingBuffer * buf);
static gboolean gst_jack_ring_buffer_acquire(GstRingBuffer * buf,GstRingBufferSpec * spec);
static gboolean gst_jack_ring_buffer_release(GstRingBuffer * buf);
static gboolean gst_jack_ring_buffer_start(GstRingBuffer * buf);
static gboolean gst_jack_ring_buffer_pause(GstRingBuffer * buf);
static gboolean gst_jack_ring_buffer_stop(GstRingBuffer * buf);
static guint gst_jack_ring_buffer_delay(GstRingBuffer * buf);
#endif

114
ext/jack/gstjackutil.c
View file

@ -1,114 +0,0 @@
/* GStreamer Jack utility functions
* Copyright (C) 2010 Tristan Matthews <tristan@sat.qc.ca>
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include "gstjackutil.h"
#include <gst/audio/multichannel.h>
static const GstAudioChannelPosition default_positions[8][8] = {
/* 1 channel */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_MONO,
},
/* 2 channels */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
},
/* 3 channels (2.1) */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_LFE, /* or FRONT_CENTER for 3.0? */
},
/* 4 channels (4.0 or 3.1?) */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
},
/* 5 channels */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
},
/* 6 channels */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE,
},
/* 7 channels */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE,
GST_AUDIO_CHANNEL_POSITION_REAR_CENTER,
},
/* 8 channels */
{
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE,
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
}
};
/* if channels are less than or equal to 8, we set a default layout,
* otherwise set layout to an array of GST_AUDIO_CHANNEL_POSITION_NONE */
void
gst_jack_set_layout_on_caps (GstCaps ** caps, gint channels)
{
int c;
GValue pos = { 0 };
GValue chanpos = { 0 };
gst_caps_unref (*caps);
if (channels <= 8) {
g_assert (channels >= 1);
gst_audio_set_channel_positions (gst_caps_get_structure (*caps, 0),
default_positions[channels - 1]);
} else {
g_value_init (&chanpos, GST_TYPE_ARRAY);
g_value_init (&pos, GST_TYPE_AUDIO_CHANNEL_POSITION);
for (c = 0; c < channels; c++) {
g_value_set_enum (&pos, GST_AUDIO_CHANNEL_POSITION_NONE);
gst_value_array_append_value (&chanpos, &pos);
}
g_value_unset (&pos);
gst_structure_set_value (gst_caps_get_structure (*caps, 0),
"channel-positions", &chanpos);
g_value_unset (&chanpos);
}
gst_caps_ref (*caps);
}

30
ext/jack/gstjackutil.h
View file

@ -1,30 +0,0 @@
/* GStreamer
* Copyright (C) 2010 Tristan Matthews <tristan@sat.qc.ca>
*
* gstjackutil.h:
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef _GST_JACK_UTIL_H_
#define _GST_JACK_UTIL_H_
#include <gst/gst.h>
void
gst_jack_set_layout_on_caps (GstCaps **caps, gint channels);
#endif // _GST_JACK_UTIL_H_

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

@ -165,7 +165,6 @@ rm -rf $RPM_BUILD_ROOT
@USE_DC1394_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstdc1394.so
@USE_TIMIDITY_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgsttimidity.so
@USE_WILDMIDI_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstwildmidi.so
@USE_JACK_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstjack.so
@USE_SNDFILE_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstsndfile.so
@USE_CELT_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstcelt.so
@USE_MPEG2ENC_TRUE@%{_libdir}/gstreamer-%{majorminor}/libgstmpeg2enc.so

9
tests/examples/Makefile.am
View file

@ -1,9 +1,4 @@
if HAVE_GTK
if USE_JACK
JACK_EXAMPLES=jack
else
JACK_EXAMPLES=
endif
GTK_EXAMPLES=camerabin mxf scaletempo camerabin2
else
GTK_EXAMPLES=
@ -21,7 +16,7 @@ else
CAMERABIN2=
endif
SUBDIRS= $(DIRECTFB_DIR) $(GTK_EXAMPLES) $(JACK_EXAMPLES)
DIST_SUBDIRS= camerabin $(CAMERABIN2) directfb jack mxf scaletempo
SUBDIRS= $(DIRECTFB_DIR) $(GTK_EXAMPLES)
DIST_SUBDIRS= camerabin $(CAMERABIN2) directfb mxf scaletempo
include $(top_srcdir)/common/parallel-subdirs.mak

6
tests/examples/jack/Makefile.am
View file

@ -1,6 +0,0 @@
noinst_PROGRAMS = jack_client
jack_client_SOURCES = jack_client.c
jack_client_CFLAGS = $(GST_CFLAGS) $(GTK_CFLAGS) $(JACK_CFLAGS)
jack_client_LDFLAGS = $(GST_LIBS) $(GTK_LIBS) $(JACK_LIBS)

79
tests/examples/jack/jack_client.c
View file

@ -1,79 +0,0 @@
/* This app demonstrates the creation and use of a jack client in conjunction
* with the jack plugins. This way, an application can control the jack client
* directly.
*/
#include <gst/gst.h>
#include <gtk/gtk.h>
#include <jack/jack.h>
static gboolean
quit_cb (gpointer data)
{
gtk_main_quit ();
return FALSE;
}
int
main (int argc, char **argv)
{
jack_client_t *src_client, *sink_client;
jack_status_t status;
GstElement *pipeline, *src, *sink;
GstStateChangeReturn ret;
gst_init (&argc, &argv);
/* create jack clients */
src_client = jack_client_open ("src_client", JackNoStartServer, &status);
if (src_client == NULL) {
if (status & JackServerFailed)
g_print ("JACK server not running\n");
else
g_print ("jack_client_open() failed, status = 0x%2.0x\n", status);
return 1;
}
sink_client = jack_client_open ("sink_client", JackNoStartServer, &status);
if (sink_client == NULL) {
if (status & JackServerFailed)
g_print ("JACK server not running\n");
else
g_print ("jack_client_open() failed, status = 0x%2.0x\n", status);
return 1;
}
/* create gst elements */
pipeline = gst_pipeline_new ("my_pipeline");
src = gst_element_factory_make ("jackaudiosrc", NULL);
sink = gst_element_factory_make ("jackaudiosink", NULL);
g_object_set (src, "client", src_client, NULL);
g_object_set (sink, "client", sink_client, NULL);
gst_bin_add_many (GST_BIN (pipeline), src, sink, NULL);
/* link everything together */
if (!gst_element_link (src, sink)) {
g_print ("Failed to link elements!\n");
return 1;
}
/* run */
ret = gst_element_set_state (pipeline, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE) {
g_print ("Failed to start up pipeline!\n");
return 1;
}
/* quit after 5 seconds */
g_timeout_add (5000, (GSourceFunc) quit_cb, NULL);
gtk_main ();
/* clean up */
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (pipeline);
return 0;
}