-/* -*- 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;
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")
--- /dev/null
+/* 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;
+}
projects / platform / upstream / gstreamer.git / commitdiff