2 * Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
4 * gstjackaudiosink.c: jack audio sink implementation
6 * This library is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Library General Public
8 * License as published by the Free Software Foundation; either
9 * version 2 of the License, or (at your option) any later version.
11 * This library is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Library General Public License for more details.
16 * You should have received a copy of the GNU Library General Public
17 * License along with this library; if not, write to the
18 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
19 * Boston, MA 02111-1307, USA.
23 * SECTION:element-jackaudiosink
24 * @see_also: #GstBaseAudioSink, #GstRingBuffer
26 * A Sink that outputs data to Jack ports.
28 * It will create N Jack ports named out_<name>_<num> where
29 * <name> is the element name and <num> is starting from 1.
30 * Each port corresponds to a gstreamer channel.
32 * The samplerate as exposed on the caps is always the same as the samplerate of
35 * When the #GstJackAudioSink:connect property is set to auto, this element
36 * will try to connect each output port to a random physical jack input pin. In
37 * this mode, the sink will expose the number of physical channels on its pad
40 * When the #GstJackAudioSink:connect property is set to none, the element will
41 * accept any number of input channels and will create (but not connect) an
42 * output port for each channel.
44 * The element will generate an error when the Jack server is shut down when it
45 * was PAUSED or PLAYING. This element does not support dynamic rate and buffer
46 * size changes at runtime.
49 * <title>Example launch line</title>
51 * gst-launch audiotestsrc ! jackaudiosink
52 * ]| Play a sine wave to using jack.
55 * Last reviewed on 2006-11-30 (0.10.4)
62 #include <gst/gst-i18n-plugin.h>
66 #include "gstjackaudiosink.h"
67 #include "gstjackringbuffer.h"
69 GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_sink_debug);
70 #define GST_CAT_DEFAULT gst_jack_audio_sink_debug
73 gst_jack_audio_sink_allocate_channels (GstJackAudioSink * sink, gint channels)
75 jack_client_t *client;
77 client = gst_jack_audio_client_get_client (sink->client);
79 /* remove ports we don't need */
80 while (sink->port_count > channels) {
81 jack_port_unregister (client, sink->ports[--sink->port_count]);
84 /* alloc enough output ports */
85 sink->ports = g_realloc (sink->ports, sizeof (jack_port_t *) * channels);
86 sink->buffers = g_realloc (sink->buffers, sizeof (sample_t *) * channels);
88 /* create an output port for each channel */
89 while (sink->port_count < channels) {
92 /* port names start from 1 and are local to the element */
94 g_strdup_printf ("out_%s_%d", GST_ELEMENT_NAME (sink),
95 sink->port_count + 1);
96 sink->ports[sink->port_count] =
97 jack_port_register (client, name, JACK_DEFAULT_AUDIO_TYPE,
99 if (sink->ports[sink->port_count] == NULL)
110 gst_jack_audio_sink_free_channels (GstJackAudioSink * sink)
113 jack_client_t *client;
115 client = gst_jack_audio_client_get_client (sink->client);
117 /* get rid of all ports */
118 while (sink->port_count) {
119 GST_LOG_OBJECT (sink, "unregister port %d", i);
120 if ((res = jack_port_unregister (client, sink->ports[i++]))) {
121 GST_DEBUG_OBJECT (sink, "unregister of port failed (%d)", res);
125 g_free (sink->ports);
127 g_free (sink->buffers);
128 sink->buffers = NULL;
131 /* ringbuffer abstract base class */
133 gst_jack_ring_buffer_get_type (void)
135 static volatile gsize ringbuffer_type = 0;
137 if (g_once_init_enter (&ringbuffer_type)) {
138 static const GTypeInfo ringbuffer_info = {
139 sizeof (GstJackRingBufferClass),
142 (GClassInitFunc) gst_jack_ring_buffer_class_init,
145 sizeof (GstJackRingBuffer),
147 (GInstanceInitFunc) gst_jack_ring_buffer_init,
150 GType tmp = g_type_register_static (GST_TYPE_RING_BUFFER,
151 "GstJackAudioSinkRingBuffer", &ringbuffer_info, 0);
152 g_once_init_leave (&ringbuffer_type, tmp);
155 return (GType) ringbuffer_type;
159 gst_jack_ring_buffer_class_init (GstJackRingBufferClass * klass)
161 GstRingBufferClass *gstringbuffer_class;
163 gstringbuffer_class = (GstRingBufferClass *) klass;
165 ring_parent_class = g_type_class_peek_parent (klass);
167 gstringbuffer_class->open_device =
168 GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_open_device);
169 gstringbuffer_class->close_device =
170 GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_close_device);
171 gstringbuffer_class->acquire =
172 GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_acquire);
173 gstringbuffer_class->release =
174 GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_release);
175 gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start);
176 gstringbuffer_class->pause = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_pause);
177 gstringbuffer_class->resume = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start);
178 gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_stop);
180 gstringbuffer_class->delay = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_delay);
183 /* this is the callback of jack. This should RT-safe.
186 jack_process_cb (jack_nframes_t nframes, void *arg)
188 GstJackAudioSink *sink;
192 gint i, j, flen, channels;
195 buf = GST_RING_BUFFER_CAST (arg);
196 sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
198 channels = buf->spec.channels;
200 /* get target buffers */
201 for (i = 0; i < channels; i++) {
203 (sample_t *) jack_port_get_buffer (sink->ports[i], nframes);
206 if (gst_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) {
207 flen = len / channels;
209 /* the number of samples must be exactly the segment size */
210 if (nframes * sizeof (sample_t) != flen)
213 GST_DEBUG_OBJECT (sink, "copy %d frames: %p, %d bytes, %d channels",
214 nframes, readptr, flen, channels);
215 data = (sample_t *) readptr;
217 /* the samples in the ringbuffer have the channels interleaved, we need to
218 * deinterleave into the jack target buffers */
219 for (i = 0; i < nframes; i++) {
220 for (j = 0; j < channels; j++) {
221 sink->buffers[j][i] = *data++;
225 /* clear written samples in the ringbuffer */
226 gst_ring_buffer_clear (buf, readseg);
228 /* we wrote one segment */
229 gst_ring_buffer_advance (buf, 1);
231 GST_DEBUG_OBJECT (sink, "write %d frames silence", nframes);
232 /* We are not allowed to read from the ringbuffer, write silence to all
233 * jack output buffers */
234 for (i = 0; i < channels; i++) {
235 memset (sink->buffers[i], 0, nframes * sizeof (sample_t));
243 GST_ERROR_OBJECT (sink, "nbytes (%d) != flen (%d)",
244 (gint) (nframes * sizeof (sample_t)), flen);
251 jack_sample_rate_cb (jack_nframes_t nframes, void *arg)
253 GstJackAudioSink *sink;
254 GstJackRingBuffer *abuf;
256 abuf = GST_JACK_RING_BUFFER_CAST (arg);
257 sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg));
259 if (abuf->sample_rate != -1 && abuf->sample_rate != nframes)
267 GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS,
268 (NULL), ("Jack changed the sample rate, which is not supported"));
275 jack_buffer_size_cb (jack_nframes_t nframes, void *arg)
277 GstJackAudioSink *sink;
278 GstJackRingBuffer *abuf;
280 abuf = GST_JACK_RING_BUFFER_CAST (arg);
281 sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg));
283 if (abuf->buffer_size != -1 && abuf->buffer_size != nframes)
291 GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS,
292 (NULL), ("Jack changed the buffer size, which is not supported"));
298 jack_shutdown_cb (void *arg)
300 GstJackAudioSink *sink;
302 sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg));
304 GST_DEBUG_OBJECT (sink, "shutdown");
306 GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
307 (NULL), ("Jack server shutdown"));
311 gst_jack_ring_buffer_init (GstJackRingBuffer * buf,
312 GstJackRingBufferClass * g_class)
315 buf->buffer_size = -1;
316 buf->sample_rate = -1;
319 /* the _open_device method should make a connection with the server
322 gst_jack_ring_buffer_open_device (GstRingBuffer * buf)
324 GstJackAudioSink *sink;
325 jack_status_t status = 0;
328 sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
330 GST_DEBUG_OBJECT (sink, "open");
332 if (sink->client_name) {
333 name = sink->client_name;
335 name = g_get_application_name ();
340 sink->client = gst_jack_audio_client_new (name, sink->server,
342 GST_JACK_CLIENT_SINK,
344 jack_process_cb, jack_buffer_size_cb, jack_sample_rate_cb, buf, &status);
345 if (sink->client == NULL)
348 GST_DEBUG_OBJECT (sink, "opened");
355 if (status & JackServerFailed) {
356 GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
357 (_("Jack server not found")),
358 ("Cannot connect to the Jack server (status %d)", status));
360 GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
361 (NULL), ("Jack client open error (status %d)", status));
367 /* close the connection with the server
370 gst_jack_ring_buffer_close_device (GstRingBuffer * buf)
372 GstJackAudioSink *sink;
374 sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
376 GST_DEBUG_OBJECT (sink, "close");
378 gst_jack_audio_sink_free_channels (sink);
379 gst_jack_audio_client_free (sink->client);
385 /* allocate a buffer and setup resources to process the audio samples of
386 * the format as specified in @spec.
388 * We allocate N jack ports, one for each channel. If we are asked to
389 * automatically make a connection with physical ports, we connect as many
390 * ports as there are physical ports, leaving leftover ports unconnected.
392 * It is assumed that samplerate and number of channels are acceptable since our
393 * getcaps method will always provide correct values. If unacceptable caps are
394 * received for some reason, we fail here.
397 gst_jack_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
399 GstJackAudioSink *sink;
400 GstJackRingBuffer *abuf;
402 gint sample_rate, buffer_size;
403 gint i, channels, res;
404 jack_client_t *client;
406 sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
407 abuf = GST_JACK_RING_BUFFER_CAST (buf);
409 GST_DEBUG_OBJECT (sink, "acquire");
411 client = gst_jack_audio_client_get_client (sink->client);
413 /* sample rate must be that of the server */
414 sample_rate = jack_get_sample_rate (client);
415 if (sample_rate != spec->rate)
416 goto wrong_samplerate;
418 channels = spec->channels;
420 if (!gst_jack_audio_sink_allocate_channels (sink, channels))
423 buffer_size = jack_get_buffer_size (client);
425 /* the segment size in bytes, this is large enough to hold a buffer of 32bit floats
426 * for all channels */
427 spec->segsize = buffer_size * sizeof (gfloat) * channels;
428 spec->latency_time = gst_util_uint64_scale (spec->segsize,
429 (GST_SECOND / GST_USECOND), spec->rate * spec->bytes_per_sample);
430 /* segtotal based on buffer-time latency */
431 spec->segtotal = spec->buffer_time / spec->latency_time;
432 if (spec->segtotal < 2) {
434 spec->buffer_time = spec->latency_time * spec->segtotal;
437 GST_DEBUG_OBJECT (sink, "buffer time: %" G_GINT64_FORMAT " usec",
439 GST_DEBUG_OBJECT (sink, "latency time: %" G_GINT64_FORMAT " usec",
441 GST_DEBUG_OBJECT (sink, "buffer_size %d, segsize %d, segtotal %d",
442 buffer_size, spec->segsize, spec->segtotal);
444 /* allocate the ringbuffer memory now */
445 buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize);
446 memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data));
448 if ((res = gst_jack_audio_client_set_active (sink->client, TRUE)))
449 goto could_not_activate;
451 /* if we need to automatically connect the ports, do so now. We must do this
452 * after activating the client. */
453 if (sink->connect == GST_JACK_CONNECT_AUTO
454 || sink->connect == GST_JACK_CONNECT_AUTO_FORCED) {
455 /* find all the physical input ports. A physical input port is a port
456 * associated with a hardware device. Someone needs connect to a physical
457 * port in order to hear something. */
458 ports = jack_get_ports (client, NULL, NULL,
459 JackPortIsPhysical | JackPortIsInput);
461 /* no ports? fine then we don't do anything except for posting a warning
463 GST_ELEMENT_WARNING (sink, RESOURCE, NOT_FOUND, (NULL),
464 ("No physical input ports found, leaving ports unconnected"));
468 for (i = 0; i < channels; i++) {
469 /* stop when all input ports are exhausted */
470 if (ports[i] == NULL) {
471 /* post a warning that we could not connect all ports */
472 GST_ELEMENT_WARNING (sink, RESOURCE, NOT_FOUND, (NULL),
473 ("No more physical ports, leaving some ports unconnected"));
476 GST_DEBUG_OBJECT (sink, "try connecting to %s",
477 jack_port_name (sink->ports[i]));
478 /* connect the port to a physical port */
479 res = jack_connect (client, jack_port_name (sink->ports[i]), ports[i]);
480 if (res != 0 && res != EEXIST)
487 abuf->sample_rate = sample_rate;
488 abuf->buffer_size = buffer_size;
489 abuf->channels = spec->channels;
496 GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
497 ("Wrong samplerate, server is running at %d and we received %d",
498 sample_rate, spec->rate));
503 GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
504 ("Cannot allocate more Jack ports"));
509 GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
510 ("Could not activate client (%d:%s)", res, g_strerror (res)));
515 GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
516 ("Could not connect output ports to physical ports (%d:%s)",
517 res, g_strerror (res)));
523 /* function is called with LOCK */
525 gst_jack_ring_buffer_release (GstRingBuffer * buf)
527 GstJackAudioSink *sink;
528 GstJackRingBuffer *abuf;
531 abuf = GST_JACK_RING_BUFFER_CAST (buf);
532 sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
534 GST_DEBUG_OBJECT (sink, "release");
536 if ((res = gst_jack_audio_client_set_active (sink->client, FALSE))) {
537 /* we only warn, this means the server is probably shut down and the client
539 GST_ELEMENT_WARNING (sink, RESOURCE, CLOSE, (NULL),
540 ("Could not deactivate Jack client (%d)", res));
544 abuf->buffer_size = -1;
545 abuf->sample_rate = -1;
547 /* free the buffer */
548 gst_buffer_unref (buf->data);
555 gst_jack_ring_buffer_start (GstRingBuffer * buf)
557 GstJackAudioSink *sink;
559 sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
561 GST_DEBUG_OBJECT (sink, "start");
567 gst_jack_ring_buffer_pause (GstRingBuffer * buf)
569 GstJackAudioSink *sink;
571 sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
573 GST_DEBUG_OBJECT (sink, "pause");
579 gst_jack_ring_buffer_stop (GstRingBuffer * buf)
581 GstJackAudioSink *sink;
583 sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
585 GST_DEBUG_OBJECT (sink, "stop");
590 #if defined (HAVE_JACK_0_120_1) || defined(HAVE_JACK_1_9_7)
592 gst_jack_ring_buffer_delay (GstRingBuffer * buf)
594 GstJackAudioSink *sink;
596 jack_latency_range_t range;
598 sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
600 for (i = 0; i < sink->port_count; i++) {
601 jack_port_get_latency_range (sink->ports[i], JackPlaybackLatency, &range);
606 GST_LOG_OBJECT (sink, "delay %u", res);
610 #else /* !(defined (HAVE_JACK_0_120_1) || defined(HAVE_JACK_1_9_7)) */
612 gst_jack_ring_buffer_delay (GstRingBuffer * buf)
614 GstJackAudioSink *sink;
617 jack_client_t *client;
619 sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
620 client = gst_jack_audio_client_get_client (sink->client);
622 for (i = 0; i < sink->port_count; i++) {
623 latency = jack_port_get_total_latency (client, sink->ports[i]);
628 GST_LOG_OBJECT (sink, "delay %u", res);
634 static GstStaticPadTemplate jackaudiosink_sink_factory =
635 GST_STATIC_PAD_TEMPLATE ("sink",
638 GST_STATIC_CAPS ("audio/x-raw-float, "
639 "endianness = (int) BYTE_ORDER, "
641 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
644 /* AudioSink signals and args */
651 #define DEFAULT_PROP_CONNECT GST_JACK_CONNECT_AUTO
652 #define DEFAULT_PROP_SERVER NULL
653 #define DEFAULT_PROP_CLIENT_NAME NULL
665 #define _do_init(bla) \
666 GST_DEBUG_CATEGORY_INIT (gst_jack_audio_sink_debug, "jacksink", 0, "jacksink element");
668 GST_BOILERPLATE_FULL (GstJackAudioSink, gst_jack_audio_sink, GstBaseAudioSink,
669 GST_TYPE_BASE_AUDIO_SINK, _do_init);
671 static void gst_jack_audio_sink_dispose (GObject * object);
672 static void gst_jack_audio_sink_set_property (GObject * object, guint prop_id,
673 const GValue * value, GParamSpec * pspec);
674 static void gst_jack_audio_sink_get_property (GObject * object, guint prop_id,
675 GValue * value, GParamSpec * pspec);
677 static GstCaps *gst_jack_audio_sink_getcaps (GstBaseSink * bsink);
678 static GstRingBuffer *gst_jack_audio_sink_create_ringbuffer (GstBaseAudioSink *
682 gst_jack_audio_sink_base_init (gpointer g_class)
684 GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
686 gst_element_class_set_details_simple (element_class, "Audio Sink (Jack)",
687 "Sink/Audio", "Output audio to a JACK server",
688 "Wim Taymans <wim.taymans@gmail.com>");
690 gst_element_class_add_static_pad_template (element_class,
691 &jackaudiosink_sink_factory);
695 gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass)
697 GObjectClass *gobject_class;
698 GstBaseSinkClass *gstbasesink_class;
699 GstBaseAudioSinkClass *gstbaseaudiosink_class;
701 gobject_class = (GObjectClass *) klass;
702 gstbasesink_class = (GstBaseSinkClass *) klass;
703 gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass;
705 gobject_class->dispose = gst_jack_audio_sink_dispose;
706 gobject_class->get_property = gst_jack_audio_sink_get_property;
707 gobject_class->set_property = gst_jack_audio_sink_set_property;
709 g_object_class_install_property (gobject_class, PROP_CONNECT,
710 g_param_spec_enum ("connect", "Connect",
711 "Specify how the output ports will be connected",
712 GST_TYPE_JACK_CONNECT, DEFAULT_PROP_CONNECT,
713 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
715 g_object_class_install_property (gobject_class, PROP_SERVER,
716 g_param_spec_string ("server", "Server",
717 "The Jack server to connect to (NULL = default)",
718 DEFAULT_PROP_SERVER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
721 * GstJackAudioSink:client-name
723 * The client name to use.
727 g_object_class_install_property (gobject_class, PROP_CLIENT_NAME,
728 g_param_spec_string ("client-name", "Client name",
729 "The client name of the Jack instance (NULL = default)",
730 DEFAULT_PROP_CLIENT_NAME,
731 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
733 g_object_class_install_property (gobject_class, PROP_CLIENT,
734 g_param_spec_boxed ("client", "JackClient", "Handle for jack client",
735 GST_TYPE_JACK_CLIENT,
736 GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
737 G_PARAM_STATIC_STRINGS));
739 gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_sink_getcaps);
741 gstbaseaudiosink_class->create_ringbuffer =
742 GST_DEBUG_FUNCPTR (gst_jack_audio_sink_create_ringbuffer);
744 /* ref class from a thread-safe context to work around missing bit of
745 * thread-safety in GObject */
746 g_type_class_ref (GST_TYPE_JACK_RING_BUFFER);
748 gst_jack_audio_client_init ();
752 gst_jack_audio_sink_init (GstJackAudioSink * sink,
753 GstJackAudioSinkClass * g_class)
755 sink->connect = DEFAULT_PROP_CONNECT;
756 sink->server = g_strdup (DEFAULT_PROP_SERVER);
757 sink->jclient = NULL;
759 sink->port_count = 0;
760 sink->client_name = g_strdup (DEFAULT_PROP_CLIENT_NAME);
761 sink->buffers = NULL;
765 gst_jack_audio_sink_dispose (GObject * object)
767 GstJackAudioSink *sink = GST_JACK_AUDIO_SINK (object);
769 gst_caps_replace (&sink->caps, NULL);
771 if (sink->client_name != NULL) {
772 g_free (sink->client_name);
773 sink->client_name = NULL;
776 G_OBJECT_CLASS (parent_class)->dispose (object);
780 gst_jack_audio_sink_set_property (GObject * object, guint prop_id,
781 const GValue * value, GParamSpec * pspec)
783 GstJackAudioSink *sink;
785 sink = GST_JACK_AUDIO_SINK (object);
788 case PROP_CLIENT_NAME:
789 g_free (sink->client_name);
790 sink->client_name = g_value_dup_string (value);
793 sink->connect = g_value_get_enum (value);
796 g_free (sink->server);
797 sink->server = g_value_dup_string (value);
800 if (GST_STATE (sink) == GST_STATE_NULL ||
801 GST_STATE (sink) == GST_STATE_READY) {
802 sink->jclient = g_value_get_boxed (value);
806 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
812 gst_jack_audio_sink_get_property (GObject * object, guint prop_id,
813 GValue * value, GParamSpec * pspec)
815 GstJackAudioSink *sink;
817 sink = GST_JACK_AUDIO_SINK (object);
820 case PROP_CLIENT_NAME:
821 g_value_set_string (value, sink->client_name);
824 g_value_set_enum (value, sink->connect);
827 g_value_set_string (value, sink->server);
830 g_value_set_boxed (value, sink->jclient);
833 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
839 gst_jack_audio_sink_getcaps (GstBaseSink * bsink)
841 GstJackAudioSink *sink = GST_JACK_AUDIO_SINK (bsink);
845 jack_client_t *client;
847 if (sink->client == NULL)
850 client = gst_jack_audio_client_get_client (sink->client);
852 if (sink->connect == GST_JACK_CONNECT_AUTO) {
853 /* get a port count, this is the number of channels we can automatically
855 ports = jack_get_ports (client, NULL, NULL,
856 JackPortIsPhysical | JackPortIsInput);
859 for (; ports[max]; max++);
864 /* we allow any number of pads, something else is going to connect the
870 rate = jack_get_sample_rate (client);
872 GST_DEBUG_OBJECT (sink, "got %d-%d ports, samplerate: %d", min, max, rate);
875 sink->caps = gst_caps_new_simple ("audio/x-raw-float",
876 "endianness", G_TYPE_INT, G_BYTE_ORDER,
877 "width", G_TYPE_INT, 32,
878 "rate", G_TYPE_INT, rate,
879 "channels", GST_TYPE_INT_RANGE, min, max, NULL);
881 GST_INFO_OBJECT (sink, "returning caps %" GST_PTR_FORMAT, sink->caps);
883 return gst_caps_ref (sink->caps);
888 GST_DEBUG_OBJECT (sink, "device not open, using template caps");
889 /* base class will get template caps for us when we return NULL */
894 static GstRingBuffer *
895 gst_jack_audio_sink_create_ringbuffer (GstBaseAudioSink * sink)
897 GstRingBuffer *buffer;
899 buffer = g_object_new (GST_TYPE_JACK_RING_BUFFER, NULL);
900 GST_DEBUG_OBJECT (sink, "created ringbuffer @%p", buffer);