/* 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_<name>_<num> where * <name> is the element name and <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 #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) */ #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); 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 (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; } 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_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, 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, (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; 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; } /* 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 _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); /* 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->ports = NULL; sink->port_count = 0; } 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; 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; }