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Added fully functional jackaudiosink.

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Original commit message from CVS:
* configure.ac:
* ext/Makefile.am:
* ext/jack/Makefile.am:
* ext/jack/gstjack.c: (plugin_init):
* ext/jack/gstjack.h:
* ext/jack/gstjackaudiosink.c: (gst_jack_ring_buffer_get_type),
(gst_jack_ring_buffer_class_init), (jack_process_cb),
(jack_sample_rate_cb), (jack_buffer_size_cb), (jack_shutdown_cb),
(gst_jack_ring_buffer_init), (gst_jack_ring_buffer_dispose),
(gst_jack_ring_buffer_finalize),
(gst_jack_ring_buffer_open_device),
(gst_jack_ring_buffer_close_device),
(gst_jack_ring_buffer_acquire), (gst_jack_ring_buffer_release),
(gst_jack_ring_buffer_start), (gst_jack_ring_buffer_pause),
(gst_jack_ring_buffer_stop), (gst_jack_ring_buffer_delay),
(gst_jack_connect_get_type), (gst_jack_audio_sink_base_init),
(gst_jack_audio_sink_class_init), (gst_jack_audio_sink_init),
(gst_jack_audio_sink_set_property),
(gst_jack_audio_sink_get_property), (gst_jack_audio_sink_getcaps),
(gst_jack_audio_sink_create_ringbuffer):
* ext/jack/gstjackaudiosink.h:
Added fully functional jackaudiosink.
This commit is contained in:
Wim Taymans 2006-11-30 11:49:36 +00:00
parent cba358cb56
commit 76c1316131
8 changed files with 992 additions and 655 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

25
ChangeLog
View file

@ -1,3 +1,28 @@
2006-11-30 Wim Taymans <wim@fluendo.com>
* configure.ac:
* ext/Makefile.am:
* ext/jack/Makefile.am:
* ext/jack/gstjack.c: (plugin_init):
* ext/jack/gstjack.h:
* ext/jack/gstjackaudiosink.c: (gst_jack_ring_buffer_get_type),
(gst_jack_ring_buffer_class_init), (jack_process_cb),
(jack_sample_rate_cb), (jack_buffer_size_cb), (jack_shutdown_cb),
(gst_jack_ring_buffer_init), (gst_jack_ring_buffer_dispose),
(gst_jack_ring_buffer_finalize),
(gst_jack_ring_buffer_open_device),
(gst_jack_ring_buffer_close_device),
(gst_jack_ring_buffer_acquire), (gst_jack_ring_buffer_release),
(gst_jack_ring_buffer_start), (gst_jack_ring_buffer_pause),
(gst_jack_ring_buffer_stop), (gst_jack_ring_buffer_delay),
(gst_jack_connect_get_type), (gst_jack_audio_sink_base_init),
(gst_jack_audio_sink_class_init), (gst_jack_audio_sink_init),
(gst_jack_audio_sink_set_property),
(gst_jack_audio_sink_get_property), (gst_jack_audio_sink_getcaps),
(gst_jack_audio_sink_create_ringbuffer):
* ext/jack/gstjackaudiosink.h:
Added fully functional jackaudiosink.
2006-11-27 Wim Taymans <wim@fluendo.com>
* gst/qtdemux/qtdemux.c: (gst_qtdemux_get_duration),

9
configure.ac
View file

@ -490,6 +490,14 @@ GST_CHECK_FEATURE(IVORBIS, [integer vorbis plug-in], ivorbisdec, [
AC_SUBST(IVORBIS_CFLAGS)
])
dnl *** Jack ***
translit(dnm, m, l) AM_CONDITIONAL(USE_JACK, true)
GST_CHECK_FEATURE(JACK, Jack, jack, [
PKG_CHECK_MODULES(JACK, jack >= 0.29.0, HAVE_JACK="yes", HAVE_JACK="no")
AC_SUBST(JACK_CFLAGS)
AC_SUBST(JACK_LIBS)
])
dnl *** libmms ***
translit(dnm, m, l) AM_CONDITIONAL(USE_LIBMMS, true)
GST_CHECK_FEATURE(LIBMMS, [mms protocol library], libmms, [
@ -842,6 +850,7 @@ ext/faac/Makefile
ext/faad/Makefile
ext/gsm/Makefile
ext/ivorbis/Makefile
ext/jack/Makefile
ext/libmms/Makefile
ext/Makefile
ext/mpeg2enc/Makefile

8
ext/Makefile.am
View file

@ -106,11 +106,11 @@ else
IVORBIS_DIR=
endif
# if USE_JACK
# JACK_DIR=jack
# else
if USE_JACK
JACK_DIR=jack
else
JACK_DIR=
# endif
endif
# if USE_LCS
# LCS_DIR=lcs

8
ext/jack/Makefile.am
View file

@ -1,11 +1,11 @@
plugin_LTLIBRARIES = libgstjack.la
libgstjack_la_SOURCES = gstjack.c gstjackbin.c
libgstjack_la_CFLAGS = $(GST_CFLAGS) $(JACK_CFLAGS)
libgstjack_la_LIBADD = $(JACK_LIBS)
libgstjack_la_SOURCES = gstjack.c gstjackaudiosink.c
libgstjack_la_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)
noinst_HEADERS = gstjack.h
noinst_HEADERS = gstjackaudiosink.h
EXTRA_DIST = README

542
ext/jack/gstjack.c
View file

@ -1,528 +1,33 @@
/* -*- Mode: C; c-basic-offset: 4 -*- */
/*
Copyright (C) 2002, 2003 Andy Wingo <wingo@pobox.com>
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU 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
General Public License for more details.
You should have received a copy of the GNU General Public
License along with this library; if not, write to the Free
Software Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
/* 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 <stdlib.h>
#include <string.h>
#include "gstjack.h"
#include <gst/audio/audio.h>
/* TODO:
- work out the src side (caps setting, etc)
future core TODO:
- make a jack clock provider
- add GST_ELEMENT_FIXED_DATA_RATE, GST_ELEMENT_QOS,
GST_ELEMENT_CHANGES_DATA_RATE element flags, and make the scheduler
sensitive to them
*/
/* elementfactory information */
static GstElementDetails gst_jack_bin_details = {
"Jack Bin",
"Generic/Bin",
"Jack processing bin",
"Andy Wingo <wingo@pobox.com>",
};
static GstElementDetails gst_jack_sink_details = {
"Jack Sink",
"Sink/Audio",
"Output to a Jack processing network",
"Andy Wingo <wingo@pobox.com>",
};
static GstElementDetails gst_jack_src_details = {
"Jack Src",
"Source/Audio",
"Input from a Jack processing network",
"Andy Wingo <wingo@pobox.com>",
};
static GHashTable *port_name_counts = NULL;
static GstElementClass *parent_class = NULL;
static void gst_jack_base_init (gpointer g_class);
static void gst_jack_src_base_init (gpointer g_class);
static void gst_jack_sink_base_init (gpointer g_class);
static void gst_jack_init (GstJack * this);
static void gst_jack_class_init (GstJackClass * klass);
static void gst_jack_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_jack_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstPadTemplate *gst_jack_src_request_pad_factory ();
static GstPadTemplate *gst_jack_sink_request_pad_factory ();
static GstPad *gst_jack_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * name);
static GstStateChangeReturn gst_jack_change_state (GstElement * element,
GstStateChange transition);
static GstPadLinkReturn gst_jack_link (GstPad * pad, const GstCaps * caps);
static void gst_jack_loop (GstElement * element);
enum
{
ARG_0,
ARG_PORT_NAME_PREFIX
};
GType
gst_jack_get_type (void)
{
static GType jack_type = 0;
if (!jack_type) {
static const GTypeInfo jack_info = {
sizeof (GstJackClass),
gst_jack_base_init,
NULL,
NULL,
NULL,
NULL,
sizeof (GstJack),
0,
NULL,
};
jack_type =
g_type_register_static (GST_TYPE_ELEMENT, "GstJack", &jack_info, 0);
}
return jack_type;
}
GType
gst_jack_sink_get_type (void)
{
static GType jack_type = 0;
if (!jack_type) {
static const GTypeInfo jack_info = {
sizeof (GstJackClass),
gst_jack_sink_base_init,
NULL,
(GClassInitFunc) gst_jack_class_init,
NULL,
NULL,
sizeof (GstJack),
0,
(GInstanceInitFunc) gst_jack_init,
};
jack_type =
g_type_register_static (GST_TYPE_JACK, "GstJackSink", &jack_info, 0);
}
return jack_type;
}
GType
gst_jack_src_get_type (void)
{
static GType jack_type = 0;
if (!jack_type) {
static const GTypeInfo jack_info = {
sizeof (GstJackClass),
gst_jack_src_base_init,
NULL,
(GClassInitFunc) gst_jack_class_init,
NULL,
NULL,
sizeof (GstJack),
0,
(GInstanceInitFunc) gst_jack_init,
};
jack_type =
g_type_register_static (GST_TYPE_JACK, "GstJackSrc", &jack_info, 0);
}
return jack_type;
}
static void
gst_jack_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_set_details (element_class, &gst_jack_bin_details);
}
static void
gst_jack_src_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (element_class,
gst_jack_src_request_pad_factory ());
gst_element_class_set_details (element_class, &gst_jack_src_details);
}
static void
gst_jack_sink_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (element_class,
gst_jack_sink_request_pad_factory ());
gst_element_class_set_details (element_class, &gst_jack_sink_details);
}
static void
gst_jack_class_init (GstJackClass * klass)
{
GObjectClass *object_class;
GstElementClass *element_class;
GParamSpec *pspec;
gchar *prefix;
object_class = (GObjectClass *) klass;
element_class = (GstElementClass *) klass;
if (parent_class == NULL)
parent_class = g_type_class_peek_parent (klass);
object_class->get_property = gst_jack_get_property;
object_class->set_property = gst_jack_set_property;
if (GST_IS_JACK_SINK_CLASS (klass))
prefix = "gst-out-";
else
prefix = "gst-in-";
pspec = g_param_spec_string ("port-name-prefix", "Port name prefix",
"String to prepend to jack port names",
prefix, G_PARAM_READWRITE | G_PARAM_CONSTRUCT);
g_object_class_install_property (object_class, ARG_PORT_NAME_PREFIX, pspec);
element_class->change_state = gst_jack_change_state;
element_class->request_new_pad = gst_jack_request_new_pad;
}
static void
gst_jack_init (GstJack * this)
{
if (G_OBJECT_TYPE (this) == GST_TYPE_JACK_SRC)
this->direction = GST_PAD_SRC;
else if (G_OBJECT_TYPE (this) == GST_TYPE_JACK_SINK)
this->direction = GST_PAD_SINK;
else
g_assert_not_reached ();
gst_element_set_loop_function (GST_ELEMENT (this), gst_jack_loop);
}
static void
gst_jack_set_property (GObject * object, guint prop_id, const GValue * value,
GParamSpec * pspec)
{
GstJack *this = (GstJack *) object;
switch (prop_id) {
case ARG_PORT_NAME_PREFIX:
if (this->port_name_prefix)
g_free (this->port_name_prefix);
this->port_name_prefix = g_strdup (g_value_get_string (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
return;
}
}
static void
gst_jack_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstJack *this = (GstJack *) object;
switch (prop_id) {
case ARG_PORT_NAME_PREFIX:
g_value_set_string (value, this->port_name_prefix);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstPadTemplate *
gst_jack_src_request_pad_factory (void)
{
static GstPadTemplate *template = NULL;
if (!template) {
GstCaps *caps;
caps = gst_caps_from_string (GST_AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_CAPS);
template = gst_pad_template_new ("%s", GST_PAD_SRC, GST_PAD_REQUEST, caps);
}
return template;
}
static GstPadTemplate *
gst_jack_sink_request_pad_factory (void)
{
static GstPadTemplate *template = NULL;
if (!template) {
GstCaps *caps;
caps = gst_caps_from_string (GST_AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_CAPS);
template = gst_pad_template_new ("%s", GST_PAD_SINK, GST_PAD_REQUEST, caps);
}
return template;
}
static GstPad *
gst_jack_request_new_pad (GstElement * element, GstPadTemplate * templ,
const gchar * name)
{
GstJack *this;
gchar *newname;
GList *l, **pad_list;
GstJackPad *pad;
gint count;
g_return_val_if_fail (GST_IS_JACK (element), NULL);
this = GST_JACK (element);
if (!this->bin)
pad_list = &this->pads;
else if (this->direction == GST_PAD_SRC)
pad_list = &this->bin->src_pads;
else
pad_list = &this->bin->sink_pads;
if (name) {
l = *pad_list;
while (l) {
if (strcmp (GST_JACK_PAD (l)->name, name) == 0) {
g_warning ("requested port name %s already in use.", name);
return NULL;
}
l = l->next;
}
newname = g_strdup (name);
} else {
if (this->direction == GST_PAD_SINK)
newname = g_strdup ("alsa_pcm:playback_1");
else
newname = g_strdup ("alsa_pcm:capture_1");
}
pad = g_new0 (GstJackPad, 1);
if (!port_name_counts)
port_name_counts = g_hash_table_new (g_str_hash, g_str_equal);
count =
GPOINTER_TO_INT (g_hash_table_lookup (port_name_counts,
this->port_name_prefix));
g_hash_table_insert (port_name_counts, g_strdup (this->port_name_prefix),
GINT_TO_POINTER (count + 1));
pad->name = g_strdup_printf ("%s%d", this->port_name_prefix, count);
pad->peer_name = newname;
pad->pad = gst_pad_new_from_template (templ, newname);
gst_element_add_pad (GST_ELEMENT (this), pad->pad);
gst_pad_set_link_function (pad->pad, gst_jack_link);
this->pads = g_list_append (this->pads, pad);
g_print ("returning from request_new_pad, pad %s created, to connect to %s\n",
pad->name, pad->peer_name);
return pad->pad;
}
static GstStateChangeReturn
gst_jack_change_state (GstElement * element, GstStateChange transition)
{
GstJack *this;
GList *l = NULL, **pads;
GstJackPad *pad;
GstCaps *caps;
g_return_val_if_fail (element != NULL, FALSE);
this = GST_JACK (element);
switch (GST_STATE_PENDING (element)) {
case GST_STATE_NULL:
JACK_DEBUG ("%s: NULL", GST_OBJECT_NAME (GST_OBJECT (this)));
break;
case GST_STATE_READY:
JACK_DEBUG ("%s: READY", GST_OBJECT_NAME (GST_OBJECT (this)));
if (!this->bin) {
if (!(this->bin = (GstJackBin *) gst_element_get_managing_bin (element))
|| !GST_IS_JACK_BIN (this->bin)) {
this->bin = NULL;
g_warning ("jack element %s needs to be contained in a jack bin.",
GST_OBJECT_NAME (element));
return GST_STATE_CHANGE_FAILURE;
}
/* fixme: verify that all names are unique */
l = this->pads;
pads =
(this->direction ==
GST_PAD_SRC) ? &this->bin->src_pads : &this->bin->sink_pads;
while (l) {
pad = GST_JACK_PAD (l);
JACK_DEBUG ("%s: appending pad %s:%s to list", GST_OBJECT_NAME (this),
pad->name, pad->peer_name);
*pads = g_list_append (*pads, pad);
l = g_list_next (l);
}
}
break;
case GST_STATE_PAUSED:
JACK_DEBUG ("%s: PAUSED", GST_OBJECT_NAME (GST_OBJECT (this)));
if (GST_STATE (element) == GST_STATE_READY) {
/* we're in READY->PAUSED */
l = this->pads;
while (l) {
pad = GST_JACK_PAD (l);
caps = gst_caps_copy (gst_pad_get_negotiated_caps (pad->pad));
gst_caps_set_simple (caps,
"rate", G_TYPE_INT, (int) this->bin->rate,
"buffer-frames", G_TYPE_INT, (gint) this->bin->nframes, NULL);
if (gst_pad_try_set_caps (pad->pad, caps) <= 0)
return GST_STATE_CHANGE_FAILURE;
l = g_list_next (l);
}
}
break;
case GST_STATE_PLAYING:
JACK_DEBUG ("%s: PLAYING", GST_OBJECT_NAME (GST_OBJECT (this)));
break;
}
JACK_DEBUG ("%s: state change finished", GST_OBJECT_NAME (this));
if (GST_ELEMENT_CLASS (parent_class)->change_state)
return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
return GST_STATE_CHANGE_SUCCESS;
}
static GstPadLinkReturn
gst_jack_link (GstPad * pad, const GstCaps * caps)
{
GstJack *this;
gint rate, buffer_frames;
GstStructure *structure;
this = GST_JACK (GST_OBJECT_PARENT (pad));
structure = gst_caps_get_structure (caps, 0);
gst_structure_get_int (structure, "rate", &rate);
gst_structure_get_int (structure, "buffer-frames", &buffer_frames);
if (this->bin && (rate != this->bin->rate ||
buffer_frames != this->bin->nframes))
return GST_PAD_LINK_REFUSED;
return GST_PAD_LINK_OK;
}
static void
gst_jack_loop (GstElement * element)
{
GstJack *this;
GList *pads;
gint len;
GstJackPad *pad;
GstBuffer *buffer;
this = GST_JACK (element);
len = this->bin->nframes * sizeof (sample_t);
pads = this->pads;
while (pads) {
pad = GST_JACK_PAD (pads);
if (this->direction == GST_PAD_SINK) {
buffer = GST_BUFFER (gst_pad_pull (pad->pad));
if (GST_IS_EVENT (buffer)) {
GstEvent *event = GST_EVENT (buffer);
switch (GST_EVENT_TYPE (buffer)) {
case GST_EVENT_EOS:
gst_element_set_eos (element);
gst_event_unref (event);
return;
default:
gst_pad_event_default (pad->pad, event);
return;
}
}
/* if the other plugins only give out buffer-frames or less (as
they should), if the length of the GstBuffer is different
from nframes then the buffer is short and we will get EOS
next */
memcpy (pad->data, GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer));
if (len != GST_BUFFER_SIZE (buffer))
memset (pad->data + GST_BUFFER_SIZE (buffer), 0,
len - GST_BUFFER_SIZE (buffer));
gst_buffer_unref (buffer);
} else {
buffer = gst_buffer_new ();
gst_buffer_set_data (buffer, pad->data, len);
GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_DONTFREE);
gst_pad_push (pad->pad, GST_DATA (buffer));
}
pads = g_list_next (pads);
}
}
#include "gstjackaudiosink.h"
static gboolean
plugin_init (GstPlugin * plugin)
{
if (!gst_element_register (plugin, "jackbin", GST_RANK_NONE,
GST_TYPE_JACK_BIN))
return FALSE;
if (!gst_element_register (plugin, "jacksrc", GST_RANK_NONE,
GST_TYPE_JACK_SRC))
return FALSE;
if (!gst_element_register (plugin, "jacksink", GST_RANK_NONE,
GST_TYPE_JACK_SINK))
if (!gst_element_register (plugin, "jackaudiosink", GST_RANK_PRIMARY,
GST_TYPE_JACK_AUDIO_SINK))
return FALSE;
return TRUE;
@ -531,5 +36,6 @@ plugin_init (GstPlugin * plugin)
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"jack",
"Jack Plugin Library", plugin_init, VERSION, "GPL", GST_PACKAGE_NAME,
GST_PACKAGE_ORIGIN)
"Jack elements",
plugin_init,
VERSION, "LGPL", "Gstreamer", "http://gstreamer.freedesktop.org")

129
ext/jack/gstjack.h
View file

@ -1,129 +0,0 @@
/* -*- Mode: C; c-basic-offset: 4 -*- */
/*
Copyright (C) 2002 Andy Wingo <wingo@pobox.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., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#ifndef __GST_JACK_H__
#define __GST_JACK_H__
#include <jack/jack.h>
#include <gst/gst.h>
#include <gst/bytestream/bytestream.h>
//#define JACK_DEBUG(str, a...) g_message (str, ##a)
#define JACK_DEBUG(str, a...)
#define GST_JACK(obj) G_TYPE_CHECK_INSTANCE_CAST(obj, GST_TYPE_JACK, GstJack)
#define GST_JACK_CLASS(klass) G_TYPE_CHECK_CLASS_CAST(klass, GST_TYPE_JACK, GstJackClass)
#define GST_IS_JACK(obj) G_TYPE_CHECK_INSTANCE_TYPE(obj, GST_TYPE_JACK)
#define GST_IS_JACK_CLASS(klass) G_TYPE_CHECK_CLASS_TYPE(klass, GST_TYPE_JACK)
#define GST_TYPE_JACK gst_jack_get_type()
#define GST_JACK_SINK(obj) G_TYPE_CHECK_INSTANCE_CAST(obj, GST_TYPE_JACK_SINK, GstJack)
#define GST_JACK_SINK_CLASS(klass) G_TYPE_CHECK_CLASS_CAST(klass, GST_TYPE_JACK_SINK, GstJackClass)
#define GST_IS_JACK_SINK(obj) G_TYPE_CHECK_INSTANCE_TYPE(obj, GST_TYPE_JACK_SINK)
#define GST_IS_JACK_SINK_CLASS(klass) G_TYPE_CHECK_CLASS_TYPE(klass, GST_TYPE_JACK_SINK)
#define GST_TYPE_JACK_SINK gst_jack_sink_get_type()
#define GST_JACK_SRC(obj) G_TYPE_CHECK_INSTANCE_CAST(obj, GST_TYPE_JACK_SRC, GstJack)
#define GST_JACK_SRC_CLASS(klass) G_TYPE_CHECK_CLASS_CAST(klass, GST_TYPE_JACK_SRC, GstJackClass)
#define GST_IS_JACK_SRC(obj) G_TYPE_CHECK_INSTANCE_TYPE(obj, GST_TYPE_JACK_SRC)
#define GST_IS_JACK_SRC_CLASS(klass) G_TYPE_CHECK_CLASS_TYPE(klass, GST_TYPE_JACK_SRC)
#define GST_TYPE_JACK_SRC gst_jack_src_get_type()
#define GST_JACK_BIN(obj) G_TYPE_CHECK_INSTANCE_CAST(obj, GST_TYPE_JACK_BIN, GstJackBin)
#define GST_JACK_BIN_CLASS(klass) G_TYPE_CHECK_CLASS_CAST(klass, GST_TYPE_JACK_BIN, GstJackClass)
#define GST_IS_JACK_BIN(obj) G_TYPE_CHECK_INSTANCE_TYPE(obj, GST_TYPE_JACK_BIN)
#define GST_IS_JACK_BIN_CLASS(klass) G_TYPE_CHECK_CLASS_TYPE(klass, GST_TYPE_JACK_BIN)
#define GST_TYPE_JACK_BIN gst_jack_bin_get_type()
#define GST_JACK_PAD(l) ((GstJackPad*)l->data) /* l is a GList */
typedef struct _GstJack GstJack;
typedef struct _GstJackClass GstJackClass;
typedef struct _GstJackBin GstJackBin;
typedef struct _GstJackBinClass GstJackBinClass;
typedef GstJack GstJackSink;
typedef GstJackClass GstJackSinkClass;
typedef GstJack GstJackSrc;
typedef GstJackClass GstJackSrcClass;
enum {
GST_JACK_OPEN = (GST_BIN_FLAG_LAST << 0),
GST_JACK_ACTIVE = (GST_BIN_FLAG_LAST << 1),
GST_JACK_FLAG_LAST = (GST_BIN_FLAG_LAST << 3)
};
typedef jack_default_audio_sample_t sample_t;
typedef struct {
GstPad *pad;
void *data;
const gchar *name;
const gchar *peer_name;
jack_port_t *port;
} GstJackPad;
struct _GstJack {
GstElement element;
/* list of GstJackPads */
GList *pads;
/* for convenience */
GstPadDirection direction;
gchar *port_name_prefix;
GstJackBin *bin;
};
struct _GstJackClass {
GstElementClass parent_class;
};
struct _GstJackBin {
GstBin bin;
jack_client_t *client;
gint default_new_port_number;
/* lists of GstJackPads */
GList *sink_pads;
GList *src_pads;
gchar *client_name;
guint rate;
jack_nframes_t nframes;
};
struct _GstJackBinClass {
GstBinClass parent_class;
};
GType gst_jack_get_type (void);
GType gst_jack_bin_get_type (void);
GType gst_jack_sink_get_type (void);
GType gst_jack_src_get_type (void);
#endif /* __GST_JACK_H__ */

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ext/jack/gstjackaudiosink.c Normal file
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/* 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:gstjacksink
* @short_description: JACK audio sink
* @see_also: #GstBaseAudioSink, #GstRingBuffer
*
* A Sink that outputs data to Jack ports.
*
* It will create N Jack ports named out_<num> where <num> 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 ::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 ::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.
*
* Last reviewed on 2006-11-30 (0.10.4)
*/
#include <string.h>
#include "gstjackaudiosink.h"
GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_sink_debug);
#define GST_CAT_DEFAULT gst_jack_audio_sink_debug
typedef jack_default_audio_sample_t sample_t;
#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;
jack_port_t **outport;
};
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 void gst_jack_ring_buffer_dispose (GObject * object);
static void gst_jack_ring_buffer_finalize (GObject * object);
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);
/* 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);
gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_dispose);
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_finalize);
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 (abuf->outport[i], nframes);
}
if (gst_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) {
flen = len / channels;
if (nframes * sizeof (sample_t) != flen)
goto wrong_size;
/* copy samples */
GST_DEBUG ("copy %d frames: %p, %d bytes, %d channels", nframes, readptr,
flen, channels);
data = (sample_t *) readptr;
/* copy and interleave into target buffers */
for (i = 0; i < nframes; i++) {
for (j = 0; j < channels; j++) {
buffers[j][i] = *data++;
}
}
/* clear written samples */
gst_ring_buffer_clear (buf, readseg);
/* we wrote one segment */
gst_ring_buffer_advance (buf, 1);
} else {
/* write silence to all 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)",
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;
}
static void
gst_jack_ring_buffer_dispose (GObject * object)
{
G_OBJECT_CLASS (ring_parent_class)->dispose (object);
}
static void
gst_jack_ring_buffer_finalize (GObject * object)
{
GstJackRingBuffer *ringbuffer;
ringbuffer = GST_JACK_RING_BUFFER_CAST (object);
G_OBJECT_CLASS (ring_parent_class)->finalize (object);
}
/* the _open_device method should make a connection with the server
*/
static gboolean
gst_jack_ring_buffer_open_device (GstRingBuffer * buf)
{
GstJackAudioSink *sink;
jack_options_t options;
jack_status_t status = 0;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (sink, "open");
/* never start a server */
options = JackNoStartServer;
/* if we have a servername, use it */
if (sink->server != NULL)
options |= JackServerName;
/* open the client */
sink->client = jack_client_open ("GStreamer", options, &status, sink->server);
if (sink->client == NULL)
goto could_not_open;
/* set our callbacks */
jack_set_process_callback (sink->client, jack_process_cb, buf);
/* these callbacks cause us to error */
jack_set_buffer_size_callback (sink->client, jack_buffer_size_cb, buf);
jack_set_sample_rate_callback (sink->client, jack_sample_rate_cb, buf);
jack_on_shutdown (sink->client, jack_shutdown_cb, buf);
GST_DEBUG_OBJECT (sink, "opened");
return TRUE;
/* ERRORS */
could_not_open:
{
if (status & JackServerFailed) {
GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
(NULL), ("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;
gint res;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (sink, "close");
if ((res = jack_client_close (sink->client))) {
/* just a warning, we assume the client is gone. */
GST_ELEMENT_WARNING (sink, RESOURCE, CLOSE,
(NULL), ("Jack client close error (%d)", res));
}
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 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;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
abuf = GST_JACK_RING_BUFFER_CAST (buf);
GST_DEBUG_OBJECT (sink, "acquire");
/* sample rate must be that of the server */
sample_rate = jack_get_sample_rate (sink->client);
if (sample_rate != spec->rate)
goto wrong_samplerate;
channels = spec->channels;
/* alloc enough output ports */
abuf->outport = g_new (jack_port_t *, channels);
/* create an output port for each channel */
for (i = 0; i < channels; i++) {
gchar *name;
/* port names start from 1 */
name = g_strdup_printf ("out_%d", i + 1);
abuf->outport[i] = jack_port_register (sink->client, name,
JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0);
if (abuf->outport[i] == NULL)
goto out_of_ports;
g_free (name);
}
buffer_size = jack_get_buffer_size (sink->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;
GST_DEBUG_OBJECT (sink, "segsize %d, segtotal %d", 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 = jack_activate (sink->client)))
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) {
/* 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 (sink->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;
}
/* connect the port to a physical port */
if ((res = jack_connect (sink->client, jack_port_name (abuf->outport[i]),
ports[i])))
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)", res));
return FALSE;
}
cannot_connect:
{
GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
("Could not connect output ports to physical ports (%d)", res));
free (ports);
return FALSE;
}
}
/* function is called with LOCK */
static gboolean
gst_jack_ring_buffer_release (GstRingBuffer * buf)
{
GstJackAudioSink *sink;
GstJackRingBuffer *abuf;
gint i, res;
abuf = GST_JACK_RING_BUFFER_CAST (buf);
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (sink, "release");
if ((res = jack_deactivate (sink->client))) {
/* 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));
}
/* remove all ports */
for (i = 0; i < abuf->channels; i++) {
GST_LOG_OBJECT (sink, "unregister port %d", i);
if ((res = jack_port_unregister (sink->client, abuf->outport[i]))) {
GST_DEBUG_OBJECT (sink, "unregister of port failed (%d)", res);
}
abuf->outport[i] = NULL;
}
g_free (abuf->outport);
abuf->outport = NULL;
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 res = 0;
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
GST_DEBUG_OBJECT (sink, "delay %u", res);
return res;
}
/* elementfactory information */
static const GstElementDetails gst_jack_audio_sink_details =
GST_ELEMENT_DETAILS ("Audio Sink (Jack)",
"Sink/Audio",
"Output to Jack",
"Wim Taymans <wim@fluendo.com>");
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_LAST
};
#define GST_TYPE_JACK_CONNECT (gst_jack_connect_get_type())
static 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"},
{0, NULL, NULL},
};
if (!jack_connect_type) {
jack_connect_type = g_enum_register_static ("GstJackConnect", jack_connect);
}
return jack_connect_type;
}
#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_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 (element_class, &gst_jack_audio_sink_details);
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->get_property =
GST_DEBUG_FUNCPTR (gst_jack_audio_sink_get_property);
gobject_class->set_property =
GST_DEBUG_FUNCPTR (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_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));
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);
}
static void
gst_jack_audio_sink_init (GstJackAudioSink * sink,
GstJackAudioSinkClass * g_class)
{
sink->connect = DEFAULT_PROP_CONNECT;
sink->server = g_strdup (DEFAULT_PROP_SERVER);
}
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;
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;
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;
if (sink->client == NULL)
goto no_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 (sink->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, somoething else is going to connect the
* pads. */
max = G_MAXINT;
}
min = MIN (1, max);
rate = jack_get_sample_rate (sink->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;
}

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ext/jack/gstjackaudiosink.h Normal file
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/* 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>
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;
typedef enum {
GST_JACK_CONNECT_NONE,
GST_JACK_CONNECT_AUTO
} GstJackConnect;
/**
* GstJackAudioSink:
*
* Opaque #GstJackAudioSink.
*/
struct _GstJackAudioSink {
GstBaseAudioSink element;
/* cached caps */
GstCaps *caps;
/* properties */
GstJackConnect connect;
gchar *server;
/* our client */
jack_client_t *client;
/*< private >*/
gpointer _gst_reserved[GST_PADDING];
};
struct _GstJackAudioSinkClass {
GstBaseAudioSinkClass parent_class;
/*< private >*/
gpointer _gst_reserved[GST_PADDING];
};
GType gst_jack_audio_sink_get_type (void);
G_END_DECLS
#endif /* __GST_JACK_AUDIO_SINK_H__ */