2 * Copyright (C) 2008 Tristan Matthews <tristan@sat.qc.ca>
4 * Permission is hereby granted, free of charge, to any person obtaining a
5 * copy of this software and associated documentation files (the "Software"),
6 * to deal in the Software without restriction, including without limitation
7 * the rights to use, copy, modify, merge, publish, distribute, sublicense,
8 * and/or sell copies of the Software, and to permit persons to whom the
9 * Software is furnished to do so, subject to the following conditions:
11 * The above copyright notice and this permission notice shall be included in
12 * all copies or substantial portions of the Software.
14 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
15 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
16 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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18 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
19 * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
20 * DEALINGS IN THE SOFTWARE.
22 * Alternatively, the contents of this file may be used under the
23 * GNU Lesser General Public License Version 2.1 (the "LGPL"), in
24 * which case the following provisions apply instead of the ones
27 * This library is free software; you can redistribute it and/or
28 * modify it under the terms of the GNU Library General Public
29 * License as published by the Free Software Foundation; either
30 * version 2 of the License, or (at your option) any later version.
32 * This library is distributed in the hope that it will be useful,
33 * but WITHOUT ANY WARRANTY; without even the implied warranty of
34 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
35 * Library General Public License for more details.
37 * You should have received a copy of the GNU Library General Public
38 * License along with this library; if not, write to the
39 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
40 * Boston, MA 02111-1307, USA.
44 * SECTION:element-jackaudiosrc
45 * @see_also: #GstBaseAudioSrc, #GstRingBuffer
47 * A Src that inputs data from Jack ports.
49 * It will create N Jack ports named in_<name>_<num> where
50 * <name> is the element name and <num> is starting from 1.
51 * Each port corresponds to a gstreamer channel.
53 * The samplerate as exposed on the caps is always the same as the samplerate of
56 * When the #GstJackAudioSrc:connect property is set to auto, this element
57 * will try to connect each input port to a random physical jack output pin.
59 * When the #GstJackAudioSrc:connect property is set to none, the element will
60 * accept any number of output channels and will create (but not connect) an
61 * input port for each channel.
63 * The element will generate an error when the Jack server is shut down when it
64 * was PAUSED or PLAYING. This element does not support dynamic rate and buffer
65 * size changes at runtime.
68 * <title>Example launch line</title>
70 * gst-launch jackaudiosrc connect=0 ! jackaudiosink connect=0
71 * ]| Get audio input into gstreamer from jack.
74 * Last reviewed on 2008-07-22 (0.10.4)
81 #include <gst/gst-i18n-plugin.h>
85 #include "gstjackaudiosrc.h"
86 #include "gstjackringbuffer.h"
87 #include "gstjackutil.h"
89 GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_src_debug);
90 #define GST_CAT_DEFAULT gst_jack_audio_src_debug
93 gst_jack_audio_src_allocate_channels (GstJackAudioSrc * src, gint channels)
95 jack_client_t *client;
97 client = gst_jack_audio_client_get_client (src->client);
99 /* remove ports we don't need */
100 while (src->port_count > channels)
101 jack_port_unregister (client, src->ports[--src->port_count]);
103 /* alloc enough input ports */
104 src->ports = g_realloc (src->ports, sizeof (jack_port_t *) * channels);
105 src->buffers = g_realloc (src->buffers, sizeof (sample_t *) * channels);
107 /* create an input port for each channel */
108 while (src->port_count < channels) {
111 /* port names start from 1 and are local to the element */
113 g_strdup_printf ("in_%s_%d", GST_ELEMENT_NAME (src),
114 src->port_count + 1);
115 src->ports[src->port_count] =
116 jack_port_register (client, name, JACK_DEFAULT_AUDIO_TYPE,
118 if (src->ports[src->port_count] == NULL)
129 gst_jack_audio_src_free_channels (GstJackAudioSrc * src)
132 jack_client_t *client;
134 client = gst_jack_audio_client_get_client (src->client);
136 /* get rid of all ports */
137 while (src->port_count) {
138 GST_LOG_OBJECT (src, "unregister port %d", i);
139 if ((res = jack_port_unregister (client, src->ports[i++])))
140 GST_DEBUG_OBJECT (src, "unregister of port failed (%d)", res);
146 g_free (src->buffers);
150 /* ringbuffer abstract base class */
152 gst_jack_ring_buffer_get_type (void)
154 static volatile gsize ringbuffer_type = 0;
156 if (g_once_init_enter (&ringbuffer_type)) {
157 static const GTypeInfo ringbuffer_info = { sizeof (GstJackRingBufferClass),
160 (GClassInitFunc) gst_jack_ring_buffer_class_init,
163 sizeof (GstJackRingBuffer),
165 (GInstanceInitFunc) gst_jack_ring_buffer_init,
168 GType tmp = g_type_register_static (GST_TYPE_RING_BUFFER,
169 "GstJackAudioSrcRingBuffer", &ringbuffer_info, 0);
170 g_once_init_leave (&ringbuffer_type, tmp);
173 return (GType) ringbuffer_type;
177 gst_jack_ring_buffer_class_init (GstJackRingBufferClass * klass)
179 GstRingBufferClass *gstringbuffer_class;
181 gstringbuffer_class = (GstRingBufferClass *) klass;
183 ring_parent_class = g_type_class_peek_parent (klass);
185 gstringbuffer_class->open_device =
186 GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_open_device);
187 gstringbuffer_class->close_device =
188 GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_close_device);
189 gstringbuffer_class->acquire =
190 GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_acquire);
191 gstringbuffer_class->release =
192 GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_release);
193 gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start);
194 gstringbuffer_class->pause = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_pause);
195 gstringbuffer_class->resume = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start);
196 gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_stop);
198 gstringbuffer_class->delay = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_delay);
201 /* this is the callback of jack. This should be RT-safe.
202 * Writes samples from the jack input port's buffer to the gst ring buffer.
205 jack_process_cb (jack_nframes_t nframes, void *arg)
207 GstJackAudioSrc *src;
212 gint channels, i, j, flen;
215 buf = GST_RING_BUFFER_CAST (arg);
216 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
218 channels = buf->spec.channels;
220 /* get input buffers */
221 for (i = 0; i < channels; i++)
223 (sample_t *) jack_port_get_buffer (src->ports[i], nframes);
225 if (gst_ring_buffer_prepare_read (buf, &writeseg, &writeptr, &len)) {
226 flen = len / channels;
228 /* the number of samples must be exactly the segment size */
229 if (nframes * sizeof (sample_t) != flen)
232 /* the samples in the jack input buffers have to be interleaved into the
234 data = (sample_t *) writeptr;
235 for (i = 0; i < nframes; ++i)
236 for (j = 0; j < channels; ++j)
237 *data++ = src->buffers[j][i];
239 GST_DEBUG ("copy %d frames: %p, %d bytes, %d channels", nframes, writeptr,
240 len / channels, channels);
242 /* we wrote one segment */
243 gst_ring_buffer_advance (buf, 1);
250 GST_ERROR_OBJECT (src, "nbytes (%d) != flen (%d)",
251 (gint) (nframes * sizeof (sample_t)), flen);
258 jack_sample_rate_cb (jack_nframes_t nframes, void *arg)
260 GstJackAudioSrc *src;
261 GstJackRingBuffer *abuf;
263 abuf = GST_JACK_RING_BUFFER_CAST (arg);
264 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg));
266 if (abuf->sample_rate != -1 && abuf->sample_rate != nframes)
274 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS,
275 (NULL), ("Jack changed the sample rate, which is not supported"));
282 jack_buffer_size_cb (jack_nframes_t nframes, void *arg)
284 GstJackAudioSrc *src;
285 GstJackRingBuffer *abuf;
287 abuf = GST_JACK_RING_BUFFER_CAST (arg);
288 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg));
290 if (abuf->buffer_size != -1 && abuf->buffer_size != nframes)
298 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS,
299 (NULL), ("Jack changed the buffer size, which is not supported"));
305 jack_shutdown_cb (void *arg)
307 GstJackAudioSrc *src;
309 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg));
311 GST_DEBUG_OBJECT (src, "shutdown");
313 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND,
314 (NULL), ("Jack server shutdown"));
318 gst_jack_ring_buffer_init (GstJackRingBuffer * buf,
319 GstJackRingBufferClass * g_class)
322 buf->buffer_size = -1;
323 buf->sample_rate = -1;
326 /* the _open_device method should make a connection with the server
329 gst_jack_ring_buffer_open_device (GstRingBuffer * buf)
331 GstJackAudioSrc *src;
332 jack_status_t status = 0;
335 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
337 GST_DEBUG_OBJECT (src, "open");
339 name = g_get_application_name ();
343 src->client = gst_jack_audio_client_new (name, src->server,
345 GST_JACK_CLIENT_SOURCE,
347 jack_process_cb, jack_buffer_size_cb, jack_sample_rate_cb, buf, &status);
348 if (src->client == NULL)
351 GST_DEBUG_OBJECT (src, "opened");
358 if (status & JackServerFailed) {
359 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND,
360 (_("Jack server not found")),
361 ("Cannot connect to the Jack server (status %d)", status));
363 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_WRITE,
364 (NULL), ("Jack client open error (status %d)", status));
370 /* close the connection with the server
373 gst_jack_ring_buffer_close_device (GstRingBuffer * buf)
375 GstJackAudioSrc *src;
377 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
379 GST_DEBUG_OBJECT (src, "close");
381 gst_jack_audio_src_free_channels (src);
382 gst_jack_audio_client_free (src->client);
389 /* allocate a buffer and setup resources to process the audio samples of
390 * the format as specified in @spec.
392 * We allocate N jack ports, one for each channel. If we are asked to
393 * automatically make a connection with physical ports, we connect as many
394 * ports as there are physical ports, leaving leftover ports unconnected.
396 * It is assumed that samplerate and number of channels are acceptable since our
397 * getcaps method will always provide correct values. If unacceptable caps are
398 * received for some reason, we fail here.
401 gst_jack_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
403 GstJackAudioSrc *src;
404 GstJackRingBuffer *abuf;
406 gint sample_rate, buffer_size;
407 gint i, channels, res;
408 jack_client_t *client;
410 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
411 abuf = GST_JACK_RING_BUFFER_CAST (buf);
413 GST_DEBUG_OBJECT (src, "acquire");
415 client = gst_jack_audio_client_get_client (src->client);
417 /* sample rate must be that of the server */
418 sample_rate = jack_get_sample_rate (client);
419 if (sample_rate != spec->rate)
420 goto wrong_samplerate;
422 channels = spec->channels;
424 if (!gst_jack_audio_src_allocate_channels (src, channels))
427 gst_jack_set_layout_on_caps (&spec->caps, channels);
429 buffer_size = jack_get_buffer_size (client);
431 /* the segment size in bytes, this is large enough to hold a buffer of 32bit floats
432 * for all channels */
433 spec->segsize = buffer_size * sizeof (gfloat) * channels;
434 spec->latency_time = gst_util_uint64_scale (spec->segsize,
435 (GST_SECOND / GST_USECOND), spec->rate * spec->bytes_per_sample);
436 /* segtotal based on buffer-time latency */
437 spec->segtotal = spec->buffer_time / spec->latency_time;
438 if (spec->segtotal < 2) {
440 spec->buffer_time = spec->latency_time * spec->segtotal;
443 GST_DEBUG_OBJECT (src, "buffer time: %" G_GINT64_FORMAT " usec",
445 GST_DEBUG_OBJECT (src, "latency time: %" G_GINT64_FORMAT " usec",
447 GST_DEBUG_OBJECT (src, "buffer_size %d, segsize %d, segtotal %d",
448 buffer_size, spec->segsize, spec->segtotal);
450 /* allocate the ringbuffer memory now */
451 buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize);
452 memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data));
454 if ((res = gst_jack_audio_client_set_active (src->client, TRUE)))
455 goto could_not_activate;
457 /* if we need to automatically connect the ports, do so now. We must do this
458 * after activating the client. */
459 if (src->connect == GST_JACK_CONNECT_AUTO
460 || src->connect == GST_JACK_CONNECT_AUTO_FORCED) {
461 /* find all the physical output ports. A physical output port is a port
462 * associated with a hardware device. Someone needs connect to a physical
463 * port in order to capture something. */
465 jack_get_ports (client, NULL, NULL,
466 JackPortIsPhysical | JackPortIsOutput);
468 /* no ports? fine then we don't do anything except for posting a warning
470 GST_ELEMENT_WARNING (src, RESOURCE, NOT_FOUND, (NULL),
471 ("No physical output ports found, leaving ports unconnected"));
475 for (i = 0; i < channels; i++) {
476 /* stop when all output ports are exhausted */
477 if (ports[i] == NULL) {
478 /* post a warning that we could not connect all ports */
479 GST_ELEMENT_WARNING (src, RESOURCE, NOT_FOUND, (NULL),
480 ("No more physical ports, leaving some ports unconnected"));
483 GST_DEBUG_OBJECT (src, "try connecting to %s",
484 jack_port_name (src->ports[i]));
486 /* connect the physical port to a port */
487 res = jack_connect (client, ports[i], jack_port_name (src->ports[i]));
488 if (res != 0 && res != EEXIST)
495 abuf->sample_rate = sample_rate;
496 abuf->buffer_size = buffer_size;
497 abuf->channels = spec->channels;
504 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
505 ("Wrong samplerate, server is running at %d and we received %d",
506 sample_rate, spec->rate));
511 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
512 ("Cannot allocate more Jack ports"));
517 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
518 ("Could not activate client (%d:%s)", res, g_strerror (res)));
523 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
524 ("Could not connect input ports to physical ports (%d:%s)",
525 res, g_strerror (res)));
531 /* function is called with LOCK */
533 gst_jack_ring_buffer_release (GstRingBuffer * buf)
535 GstJackAudioSrc *src;
536 GstJackRingBuffer *abuf;
539 abuf = GST_JACK_RING_BUFFER_CAST (buf);
540 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
542 GST_DEBUG_OBJECT (src, "release");
544 if ((res = gst_jack_audio_client_set_active (src->client, FALSE))) {
545 /* we only warn, this means the server is probably shut down and the client
547 GST_ELEMENT_WARNING (src, RESOURCE, CLOSE, (NULL),
548 ("Could not deactivate Jack client (%d)", res));
552 abuf->buffer_size = -1;
553 abuf->sample_rate = -1;
555 /* free the buffer */
556 gst_buffer_unref (buf->data);
563 gst_jack_ring_buffer_start (GstRingBuffer * buf)
565 GstJackAudioSrc *src;
567 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
569 GST_DEBUG_OBJECT (src, "start");
575 gst_jack_ring_buffer_pause (GstRingBuffer * buf)
577 GstJackAudioSrc *src;
579 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
581 GST_DEBUG_OBJECT (src, "pause");
587 gst_jack_ring_buffer_stop (GstRingBuffer * buf)
589 GstJackAudioSrc *src;
591 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
593 GST_DEBUG_OBJECT (src, "stop");
598 #if defined (HAVE_JACK_0_120_1) || defined(HAVE_JACK_1_9_7)
600 gst_jack_ring_buffer_delay (GstRingBuffer * buf)
602 GstJackAudioSrc *src;
604 jack_latency_range_t range;
606 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
608 for (i = 0; i < src->port_count; i++) {
609 jack_port_get_latency_range (src->ports[i], JackCaptureLatency, &range);
614 GST_DEBUG_OBJECT (src, "delay %u", res);
618 #else /* !(defined (HAVE_JACK_0_120_1) || defined(HAVE_JACK_1_9_7)) */
620 gst_jack_ring_buffer_delay (GstRingBuffer * buf)
622 GstJackAudioSrc *src;
625 jack_client_t *client;
627 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
629 client = gst_jack_audio_client_get_client (src->client);
631 for (i = 0; i < src->port_count; i++) {
632 latency = jack_port_get_total_latency (client, src->ports[i]);
637 GST_DEBUG_OBJECT (src, "delay %u", res);
643 /* Audiosrc signals and args */
650 #define DEFAULT_PROP_CONNECT GST_JACK_CONNECT_AUTO
651 #define DEFAULT_PROP_SERVER NULL
663 /* the capabilities of the inputs and outputs.
665 * describe the real formats here.
668 static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
671 GST_STATIC_CAPS ("audio/x-raw-float, "
672 "endianness = (int) BYTE_ORDER, "
674 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
677 #define _do_init(bla) \
678 GST_DEBUG_CATEGORY_INIT(gst_jack_audio_src_debug, "jacksrc", 0, "jacksrc element");
680 GST_BOILERPLATE_FULL (GstJackAudioSrc, gst_jack_audio_src, GstBaseAudioSrc,
681 GST_TYPE_BASE_AUDIO_SRC, _do_init);
683 static void gst_jack_audio_src_dispose (GObject * object);
684 static void gst_jack_audio_src_set_property (GObject * object, guint prop_id,
685 const GValue * value, GParamSpec * pspec);
686 static void gst_jack_audio_src_get_property (GObject * object, guint prop_id,
687 GValue * value, GParamSpec * pspec);
689 static GstCaps *gst_jack_audio_src_getcaps (GstBaseSrc * bsrc);
690 static GstRingBuffer *gst_jack_audio_src_create_ringbuffer (GstBaseAudioSrc *
693 /* GObject vmethod implementations */
696 gst_jack_audio_src_base_init (gpointer gclass)
698 GstElementClass *element_class = GST_ELEMENT_CLASS (gclass);
700 gst_element_class_add_pad_template (element_class,
701 gst_static_pad_template_get (&src_factory));
702 gst_element_class_set_details_simple (element_class, "Audio Source (Jack)",
703 "Source/Audio", "Captures audio from a JACK server",
704 "Tristan Matthews <tristan@sat.qc.ca>");
707 /* initialize the jack_audio_src's class */
709 gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass)
711 GObjectClass *gobject_class;
712 GstBaseSrcClass *gstbasesrc_class;
713 GstBaseAudioSrcClass *gstbaseaudiosrc_class;
715 gobject_class = (GObjectClass *) klass;
717 gstbasesrc_class = (GstBaseSrcClass *) klass;
718 gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass;
720 gobject_class->dispose = gst_jack_audio_src_dispose;
721 gobject_class->set_property = gst_jack_audio_src_set_property;
722 gobject_class->get_property = gst_jack_audio_src_get_property;
724 g_object_class_install_property (gobject_class, PROP_CONNECT,
725 g_param_spec_enum ("connect", "Connect",
726 "Specify how the input ports will be connected",
727 GST_TYPE_JACK_CONNECT, DEFAULT_PROP_CONNECT,
728 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
730 g_object_class_install_property (gobject_class, PROP_SERVER,
731 g_param_spec_string ("server", "Server",
732 "The Jack server to connect to (NULL = default)",
733 DEFAULT_PROP_SERVER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
735 g_object_class_install_property (gobject_class, PROP_CLIENT,
736 g_param_spec_boxed ("client", "JackClient", "Handle for jack client",
737 GST_TYPE_JACK_CLIENT,
738 GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
739 G_PARAM_STATIC_STRINGS));
741 gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_src_getcaps);
742 gstbaseaudiosrc_class->create_ringbuffer =
743 GST_DEBUG_FUNCPTR (gst_jack_audio_src_create_ringbuffer);
745 /* ref class from a thread-safe context to work around missing bit of
746 * thread-safety in GObject */
747 g_type_class_ref (GST_TYPE_JACK_RING_BUFFER);
749 gst_jack_audio_client_init ();
752 /* initialize the new element
753 * instantiate pads and add them to element
754 * set pad calback functions
755 * initialize instance structure
758 gst_jack_audio_src_init (GstJackAudioSrc * src, GstJackAudioSrcClass * gclass)
760 //gst_base_src_set_live(GST_BASE_SRC (src), TRUE);
761 src->connect = DEFAULT_PROP_CONNECT;
762 src->server = g_strdup (DEFAULT_PROP_SERVER);
770 gst_jack_audio_src_dispose (GObject * object)
772 GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
774 gst_caps_replace (&src->caps, NULL);
775 G_OBJECT_CLASS (parent_class)->dispose (object);
779 gst_jack_audio_src_set_property (GObject * object, guint prop_id,
780 const GValue * value, GParamSpec * pspec)
782 GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
786 src->connect = g_value_get_enum (value);
789 g_free (src->server);
790 src->server = g_value_dup_string (value);
793 if (GST_STATE (src) == GST_STATE_NULL ||
794 GST_STATE (src) == GST_STATE_READY) {
795 src->jclient = g_value_get_boxed (value);
799 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
805 gst_jack_audio_src_get_property (GObject * object, guint prop_id,
806 GValue * value, GParamSpec * pspec)
808 GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
812 g_value_set_enum (value, src->connect);
815 g_value_set_string (value, src->server);
818 g_value_set_boxed (value, src->jclient);
821 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
827 gst_jack_audio_src_getcaps (GstBaseSrc * bsrc)
829 GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (bsrc);
833 jack_client_t *client;
835 if (src->client == NULL)
838 client = gst_jack_audio_client_get_client (src->client);
840 if (src->connect == GST_JACK_CONNECT_AUTO) {
841 /* get a port count, this is the number of channels we can automatically
843 ports = jack_get_ports (client, NULL, NULL,
844 JackPortIsPhysical | JackPortIsOutput);
847 for (; ports[max]; max++);
853 /* we allow any number of pads, something else is going to connect the
859 rate = jack_get_sample_rate (client);
861 GST_DEBUG_OBJECT (src, "got %d-%d ports, samplerate: %d", min, max, rate);
864 src->caps = gst_caps_new_simple ("audio/x-raw-float",
865 "endianness", G_TYPE_INT, G_BYTE_ORDER,
866 "width", G_TYPE_INT, 32,
867 "rate", G_TYPE_INT, rate,
868 "channels", GST_TYPE_INT_RANGE, min, max, NULL);
870 GST_INFO_OBJECT (src, "returning caps %" GST_PTR_FORMAT, src->caps);
872 return gst_caps_ref (src->caps);
877 GST_DEBUG_OBJECT (src, "device not open, using template caps");
878 /* base class will get template caps for us when we return NULL */
883 static GstRingBuffer *
884 gst_jack_audio_src_create_ringbuffer (GstBaseAudioSrc * src)
886 GstRingBuffer *buffer;
888 buffer = g_object_new (GST_TYPE_JACK_RING_BUFFER, NULL);
889 GST_DEBUG_OBJECT (src, "created ringbuffer @%p", buffer);