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);
87 /* create an output port for each channel */
88 while (sink->port_count < channels) {
91 /* port names start from 1 and are local to the element */
93 g_strdup_printf ("out_%s_%d", GST_ELEMENT_NAME (sink),
94 sink->port_count + 1);
95 sink->ports[sink->port_count] =
96 jack_port_register (client, name, JACK_DEFAULT_AUDIO_TYPE,
98 if (sink->ports[sink->port_count] == NULL)
109 gst_jack_audio_sink_free_channels (GstJackAudioSink * sink)
112 jack_client_t *client;
114 client = gst_jack_audio_client_get_client (sink->client);
116 /* get rid of all ports */
117 while (sink->port_count) {
118 GST_LOG_OBJECT (sink, "unregister port %d", i);
119 if ((res = jack_port_unregister (client, sink->ports[i++]))) {
120 GST_DEBUG_OBJECT (sink, "unregister of port failed (%d)", res);
124 g_free (sink->ports);
128 /* ringbuffer abstract base class */
130 gst_jack_ring_buffer_get_type (void)
132 static volatile gsize ringbuffer_type = 0;
134 if (g_once_init_enter (&ringbuffer_type)) {
135 static const GTypeInfo ringbuffer_info = {
136 sizeof (GstJackRingBufferClass),
139 (GClassInitFunc) gst_jack_ring_buffer_class_init,
142 sizeof (GstJackRingBuffer),
144 (GInstanceInitFunc) gst_jack_ring_buffer_init,
147 GType tmp = g_type_register_static (GST_TYPE_RING_BUFFER,
148 "GstJackAudioSinkRingBuffer", &ringbuffer_info, 0);
149 g_once_init_leave (&ringbuffer_type, tmp);
152 return (GType) ringbuffer_type;
156 gst_jack_ring_buffer_class_init (GstJackRingBufferClass * klass)
158 GstRingBufferClass *gstringbuffer_class;
160 gstringbuffer_class = (GstRingBufferClass *) klass;
162 ring_parent_class = g_type_class_peek_parent (klass);
164 gstringbuffer_class->open_device =
165 GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_open_device);
166 gstringbuffer_class->close_device =
167 GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_close_device);
168 gstringbuffer_class->acquire =
169 GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_acquire);
170 gstringbuffer_class->release =
171 GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_release);
172 gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start);
173 gstringbuffer_class->pause = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_pause);
174 gstringbuffer_class->resume = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start);
175 gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_stop);
177 gstringbuffer_class->delay = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_delay);
180 /* this is the callback of jack. This should RT-safe.
183 jack_process_cb (jack_nframes_t nframes, void *arg)
185 GstJackAudioSink *sink;
189 gint i, j, flen, channels;
190 sample_t **buffers, *data;
192 buf = GST_RING_BUFFER_CAST (arg);
193 sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
195 channels = buf->spec.channels;
197 /* alloc pointers to samples */
198 buffers = g_alloca (sizeof (sample_t *) * channels);
200 /* get target buffers */
201 for (i = 0; i < channels; i++) {
202 buffers[i] = (sample_t *) jack_port_get_buffer (sink->ports[i], nframes);
205 if (gst_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) {
206 flen = len / channels;
208 /* the number of samples must be exactly the segment size */
209 if (nframes * sizeof (sample_t) != flen)
212 GST_DEBUG_OBJECT (sink, "copy %d frames: %p, %d bytes, %d channels",
213 nframes, readptr, flen, channels);
214 data = (sample_t *) readptr;
216 /* the samples in the ringbuffer have the channels interleaved, we need to
217 * deinterleave into the jack target buffers */
218 for (i = 0; i < nframes; i++) {
219 for (j = 0; j < channels; j++) {
220 buffers[j][i] = *data++;
224 /* clear written samples in the ringbuffer */
225 gst_ring_buffer_clear (buf, readseg);
227 /* we wrote one segment */
228 gst_ring_buffer_advance (buf, 1);
230 GST_DEBUG_OBJECT (sink, "write %d frames silence", nframes);
231 /* We are not allowed to read from the ringbuffer, write silence to all
232 * jack output buffers */
233 for (i = 0; i < channels; i++) {
234 memset (buffers[i], 0, nframes * sizeof (sample_t));
242 GST_ERROR_OBJECT (sink, "nbytes (%d) != flen (%d)",
243 (gint) (nframes * sizeof (sample_t)), flen);
250 jack_sample_rate_cb (jack_nframes_t nframes, void *arg)
252 GstJackAudioSink *sink;
253 GstJackRingBuffer *abuf;
255 abuf = GST_JACK_RING_BUFFER_CAST (arg);
256 sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg));
258 if (abuf->sample_rate != -1 && abuf->sample_rate != nframes)
266 GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS,
267 (NULL), ("Jack changed the sample rate, which is not supported"));
274 jack_buffer_size_cb (jack_nframes_t nframes, void *arg)
276 GstJackAudioSink *sink;
277 GstJackRingBuffer *abuf;
279 abuf = GST_JACK_RING_BUFFER_CAST (arg);
280 sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg));
282 if (abuf->buffer_size != -1 && abuf->buffer_size != nframes)
290 GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS,
291 (NULL), ("Jack changed the buffer size, which is not supported"));
297 jack_shutdown_cb (void *arg)
299 GstJackAudioSink *sink;
301 sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg));
303 GST_DEBUG_OBJECT (sink, "shutdown");
305 GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
306 (NULL), ("Jack server shutdown"));
310 gst_jack_ring_buffer_init (GstJackRingBuffer * buf,
311 GstJackRingBufferClass * g_class)
314 buf->buffer_size = -1;
315 buf->sample_rate = -1;
318 /* the _open_device method should make a connection with the server
321 gst_jack_ring_buffer_open_device (GstRingBuffer * buf)
323 GstJackAudioSink *sink;
324 jack_status_t status = 0;
327 sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
329 GST_DEBUG_OBJECT (sink, "open");
331 name = g_get_application_name ();
335 sink->client = gst_jack_audio_client_new (name, sink->server,
337 GST_JACK_CLIENT_SINK,
339 jack_process_cb, jack_buffer_size_cb, jack_sample_rate_cb, buf, &status);
340 if (sink->client == NULL)
343 GST_DEBUG_OBJECT (sink, "opened");
350 if (status & JackServerFailed) {
351 GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
352 (_("Jack server not found")),
353 ("Cannot connect to the Jack server (status %d)", status));
355 GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
356 (NULL), ("Jack client open error (status %d)", status));
362 /* close the connection with the server
365 gst_jack_ring_buffer_close_device (GstRingBuffer * buf)
367 GstJackAudioSink *sink;
369 sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
371 GST_DEBUG_OBJECT (sink, "close");
373 gst_jack_audio_sink_free_channels (sink);
374 gst_jack_audio_client_free (sink->client);
380 /* allocate a buffer and setup resources to process the audio samples of
381 * the format as specified in @spec.
383 * We allocate N jack ports, one for each channel. If we are asked to
384 * automatically make a connection with physical ports, we connect as many
385 * ports as there are physical ports, leaving leftover ports unconnected.
387 * It is assumed that samplerate and number of channels are acceptable since our
388 * getcaps method will always provide correct values. If unacceptable caps are
389 * received for some reason, we fail here.
392 gst_jack_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
394 GstJackAudioSink *sink;
395 GstJackRingBuffer *abuf;
397 gint sample_rate, buffer_size;
398 gint i, channels, res;
399 jack_client_t *client;
401 sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
402 abuf = GST_JACK_RING_BUFFER_CAST (buf);
404 GST_DEBUG_OBJECT (sink, "acquire");
406 client = gst_jack_audio_client_get_client (sink->client);
408 /* sample rate must be that of the server */
409 sample_rate = jack_get_sample_rate (client);
410 if (sample_rate != spec->rate)
411 goto wrong_samplerate;
413 channels = spec->channels;
415 if (!gst_jack_audio_sink_allocate_channels (sink, channels))
418 buffer_size = jack_get_buffer_size (client);
420 /* the segment size in bytes, this is large enough to hold a buffer of 32bit floats
421 * for all channels */
422 spec->segsize = buffer_size * sizeof (gfloat) * channels;
423 spec->latency_time = gst_util_uint64_scale (spec->segsize,
424 (GST_SECOND / GST_USECOND), spec->rate * spec->bytes_per_sample);
425 /* segtotal based on buffer-time latency */
426 spec->segtotal = spec->buffer_time / spec->latency_time;
427 if (spec->segtotal < 2) {
429 spec->buffer_time = spec->latency_time * spec->segtotal;
432 GST_DEBUG_OBJECT (sink, "buffer time: %" G_GINT64_FORMAT " usec",
434 GST_DEBUG_OBJECT (sink, "latency time: %" G_GINT64_FORMAT " usec",
436 GST_DEBUG_OBJECT (sink, "buffer_size %d, segsize %d, segtotal %d",
437 buffer_size, spec->segsize, spec->segtotal);
439 /* allocate the ringbuffer memory now */
440 buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize);
441 memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data));
443 if ((res = gst_jack_audio_client_set_active (sink->client, TRUE)))
444 goto could_not_activate;
446 /* if we need to automatically connect the ports, do so now. We must do this
447 * after activating the client. */
448 if (sink->connect == GST_JACK_CONNECT_AUTO
449 || sink->connect == GST_JACK_CONNECT_AUTO_FORCED) {
450 /* find all the physical input ports. A physical input port is a port
451 * associated with a hardware device. Someone needs connect to a physical
452 * port in order to hear something. */
453 ports = jack_get_ports (client, NULL, NULL,
454 JackPortIsPhysical | JackPortIsInput);
456 /* no ports? fine then we don't do anything except for posting a warning
458 GST_ELEMENT_WARNING (sink, RESOURCE, NOT_FOUND, (NULL),
459 ("No physical input ports found, leaving ports unconnected"));
463 for (i = 0; i < channels; i++) {
464 /* stop when all input ports are exhausted */
465 if (ports[i] == NULL) {
466 /* post a warning that we could not connect all ports */
467 GST_ELEMENT_WARNING (sink, RESOURCE, NOT_FOUND, (NULL),
468 ("No more physical ports, leaving some ports unconnected"));
471 GST_DEBUG_OBJECT (sink, "try connecting to %s",
472 jack_port_name (sink->ports[i]));
473 /* connect the port to a physical port */
474 res = jack_connect (client, jack_port_name (sink->ports[i]), ports[i]);
475 if (res != 0 && res != EEXIST)
482 abuf->sample_rate = sample_rate;
483 abuf->buffer_size = buffer_size;
484 abuf->channels = spec->channels;
491 GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
492 ("Wrong samplerate, server is running at %d and we received %d",
493 sample_rate, spec->rate));
498 GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
499 ("Cannot allocate more Jack ports"));
504 GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
505 ("Could not activate client (%d:%s)", res, g_strerror (res)));
510 GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
511 ("Could not connect output ports to physical ports (%d:%s)",
512 res, g_strerror (res)));
518 /* function is called with LOCK */
520 gst_jack_ring_buffer_release (GstRingBuffer * buf)
522 GstJackAudioSink *sink;
523 GstJackRingBuffer *abuf;
526 abuf = GST_JACK_RING_BUFFER_CAST (buf);
527 sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
529 GST_DEBUG_OBJECT (sink, "release");
531 if ((res = gst_jack_audio_client_set_active (sink->client, FALSE))) {
532 /* we only warn, this means the server is probably shut down and the client
534 GST_ELEMENT_WARNING (sink, RESOURCE, CLOSE, (NULL),
535 ("Could not deactivate Jack client (%d)", res));
539 abuf->buffer_size = -1;
540 abuf->sample_rate = -1;
542 /* free the buffer */
543 gst_buffer_unref (buf->data);
550 gst_jack_ring_buffer_start (GstRingBuffer * buf)
552 GstJackAudioSink *sink;
554 sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
556 GST_DEBUG_OBJECT (sink, "start");
562 gst_jack_ring_buffer_pause (GstRingBuffer * buf)
564 GstJackAudioSink *sink;
566 sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
568 GST_DEBUG_OBJECT (sink, "pause");
574 gst_jack_ring_buffer_stop (GstRingBuffer * buf)
576 GstJackAudioSink *sink;
578 sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
580 GST_DEBUG_OBJECT (sink, "stop");
585 #if defined (HAVE_JACK_0_120_1) || defined(HAVE_JACK_1_9_7)
587 gst_jack_ring_buffer_delay (GstRingBuffer * buf)
589 GstJackAudioSink *sink;
591 jack_latency_range_t range;
593 sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
595 for (i = 0; i < sink->port_count; i++) {
596 jack_port_get_latency_range (sink->ports[i], JackPlaybackLatency, &range);
601 GST_LOG_OBJECT (sink, "delay %u", res);
605 #else /* !(defined (HAVE_JACK_0_120_1) || defined(HAVE_JACK_1_9_7)) */
607 gst_jack_ring_buffer_delay (GstRingBuffer * buf)
609 GstJackAudioSink *sink;
612 jack_client_t *client;
614 sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
615 client = gst_jack_audio_client_get_client (sink->client);
617 for (i = 0; i < sink->port_count; i++) {
618 latency = jack_port_get_total_latency (client, sink->ports[i]);
623 GST_LOG_OBJECT (sink, "delay %u", res);
629 static GstStaticPadTemplate jackaudiosink_sink_factory =
630 GST_STATIC_PAD_TEMPLATE ("sink",
633 GST_STATIC_CAPS ("audio/x-raw-float, "
634 "endianness = (int) BYTE_ORDER, "
636 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
639 /* AudioSink signals and args */
646 #define DEFAULT_PROP_CONNECT GST_JACK_CONNECT_AUTO
647 #define DEFAULT_PROP_SERVER NULL
658 #define _do_init(bla) \
659 GST_DEBUG_CATEGORY_INIT (gst_jack_audio_sink_debug, "jacksink", 0, "jacksink element");
661 GST_BOILERPLATE_FULL (GstJackAudioSink, gst_jack_audio_sink, GstBaseAudioSink,
662 GST_TYPE_BASE_AUDIO_SINK, _do_init);
664 static void gst_jack_audio_sink_dispose (GObject * object);
665 static void gst_jack_audio_sink_set_property (GObject * object, guint prop_id,
666 const GValue * value, GParamSpec * pspec);
667 static void gst_jack_audio_sink_get_property (GObject * object, guint prop_id,
668 GValue * value, GParamSpec * pspec);
670 static GstCaps *gst_jack_audio_sink_getcaps (GstBaseSink * bsink);
671 static GstRingBuffer *gst_jack_audio_sink_create_ringbuffer (GstBaseAudioSink *
675 gst_jack_audio_sink_base_init (gpointer g_class)
677 GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
679 gst_element_class_set_details_simple (element_class, "Audio Sink (Jack)",
680 "Sink/Audio", "Output audio to a JACK server",
681 "Wim Taymans <wim.taymans@gmail.com>");
683 gst_element_class_add_pad_template (element_class,
684 gst_static_pad_template_get (&jackaudiosink_sink_factory));
688 gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass)
690 GObjectClass *gobject_class;
691 GstBaseSinkClass *gstbasesink_class;
692 GstBaseAudioSinkClass *gstbaseaudiosink_class;
694 gobject_class = (GObjectClass *) klass;
695 gstbasesink_class = (GstBaseSinkClass *) klass;
696 gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass;
698 gobject_class->dispose = gst_jack_audio_sink_dispose;
699 gobject_class->get_property = gst_jack_audio_sink_get_property;
700 gobject_class->set_property = gst_jack_audio_sink_set_property;
702 g_object_class_install_property (gobject_class, PROP_CONNECT,
703 g_param_spec_enum ("connect", "Connect",
704 "Specify how the output ports will be connected",
705 GST_TYPE_JACK_CONNECT, DEFAULT_PROP_CONNECT,
706 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
708 g_object_class_install_property (gobject_class, PROP_SERVER,
709 g_param_spec_string ("server", "Server",
710 "The Jack server to connect to (NULL = default)",
711 DEFAULT_PROP_SERVER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
713 g_object_class_install_property (gobject_class, PROP_CLIENT,
714 g_param_spec_boxed ("client", "JackClient", "Handle for jack client",
715 GST_TYPE_JACK_CLIENT,
716 GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
717 G_PARAM_STATIC_STRINGS));
719 gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_sink_getcaps);
721 gstbaseaudiosink_class->create_ringbuffer =
722 GST_DEBUG_FUNCPTR (gst_jack_audio_sink_create_ringbuffer);
724 /* ref class from a thread-safe context to work around missing bit of
725 * thread-safety in GObject */
726 g_type_class_ref (GST_TYPE_JACK_RING_BUFFER);
728 gst_jack_audio_client_init ();
732 gst_jack_audio_sink_init (GstJackAudioSink * sink,
733 GstJackAudioSinkClass * g_class)
735 sink->connect = DEFAULT_PROP_CONNECT;
736 sink->server = g_strdup (DEFAULT_PROP_SERVER);
737 sink->jclient = NULL;
739 sink->port_count = 0;
743 gst_jack_audio_sink_dispose (GObject * object)
745 GstJackAudioSink *sink = GST_JACK_AUDIO_SINK (object);
747 gst_caps_replace (&sink->caps, NULL);
748 G_OBJECT_CLASS (parent_class)->dispose (object);
752 gst_jack_audio_sink_set_property (GObject * object, guint prop_id,
753 const GValue * value, GParamSpec * pspec)
755 GstJackAudioSink *sink;
757 sink = GST_JACK_AUDIO_SINK (object);
761 sink->connect = g_value_get_enum (value);
764 g_free (sink->server);
765 sink->server = g_value_dup_string (value);
768 if (GST_STATE (sink) == GST_STATE_NULL ||
769 GST_STATE (sink) == GST_STATE_READY) {
770 sink->jclient = g_value_get_boxed (value);
774 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
780 gst_jack_audio_sink_get_property (GObject * object, guint prop_id,
781 GValue * value, GParamSpec * pspec)
783 GstJackAudioSink *sink;
785 sink = GST_JACK_AUDIO_SINK (object);
789 g_value_set_enum (value, sink->connect);
792 g_value_set_string (value, sink->server);
795 g_value_set_boxed (value, sink->jclient);
798 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
804 gst_jack_audio_sink_getcaps (GstBaseSink * bsink)
806 GstJackAudioSink *sink = GST_JACK_AUDIO_SINK (bsink);
810 jack_client_t *client;
812 if (sink->client == NULL)
815 client = gst_jack_audio_client_get_client (sink->client);
817 if (sink->connect == GST_JACK_CONNECT_AUTO) {
818 /* get a port count, this is the number of channels we can automatically
820 ports = jack_get_ports (client, NULL, NULL,
821 JackPortIsPhysical | JackPortIsInput);
824 for (; ports[max]; max++);
829 /* we allow any number of pads, something else is going to connect the
835 rate = jack_get_sample_rate (client);
837 GST_DEBUG_OBJECT (sink, "got %d-%d ports, samplerate: %d", min, max, rate);
840 sink->caps = gst_caps_new_simple ("audio/x-raw-float",
841 "endianness", G_TYPE_INT, G_BYTE_ORDER,
842 "width", G_TYPE_INT, 32,
843 "rate", G_TYPE_INT, rate,
844 "channels", GST_TYPE_INT_RANGE, min, max, NULL);
846 GST_INFO_OBJECT (sink, "returning caps %" GST_PTR_FORMAT, sink->caps);
848 return gst_caps_ref (sink->caps);
853 GST_DEBUG_OBJECT (sink, "device not open, using template caps");
854 /* base class will get template caps for us when we return NULL */
859 static GstRingBuffer *
860 gst_jack_audio_sink_create_ringbuffer (GstBaseAudioSink * sink)
862 GstRingBuffer *buffer;
864 buffer = g_object_new (GST_TYPE_JACK_RING_BUFFER, NULL);
865 GST_DEBUG_OBJECT (sink, "created ringbuffer @%p", buffer);