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
17 * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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 /* handle transport state requisitions */
219 if (src->transport == GST_JACK_TRANSPORT_SLAVE) {
220 GstState state = gst_jack_audio_client_get_transport_state (src->client);
222 if ((state != GST_STATE_VOID_PENDING) && (GST_STATE (src) != state)) {
223 GST_DEBUG_OBJECT (src, "requesting state change: %s",
224 gst_element_state_get_name (state));
225 gst_element_post_message (GST_ELEMENT (src),
226 gst_message_new_request_state (GST_OBJECT (src), state));
230 channels = buf->spec.channels;
232 /* get input buffers */
233 for (i = 0; i < channels; i++)
235 (sample_t *) jack_port_get_buffer (src->ports[i], nframes);
237 if (gst_ring_buffer_prepare_read (buf, &writeseg, &writeptr, &len)) {
238 flen = len / channels;
240 /* the number of samples must be exactly the segment size */
241 if (nframes * sizeof (sample_t) != flen)
244 /* the samples in the jack input buffers have to be interleaved into the
246 data = (sample_t *) writeptr;
247 for (i = 0; i < nframes; ++i)
248 for (j = 0; j < channels; ++j)
249 *data++ = src->buffers[j][i];
251 GST_DEBUG ("copy %d frames: %p, %d bytes, %d channels", nframes, writeptr,
252 len / channels, channels);
254 /* we wrote one segment */
255 gst_ring_buffer_advance (buf, 1);
262 GST_ERROR_OBJECT (src, "nbytes (%d) != flen (%d)",
263 (gint) (nframes * sizeof (sample_t)), flen);
270 jack_sample_rate_cb (jack_nframes_t nframes, void *arg)
272 GstJackAudioSrc *src;
273 GstJackRingBuffer *abuf;
275 abuf = GST_JACK_RING_BUFFER_CAST (arg);
276 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg));
278 if (abuf->sample_rate != -1 && abuf->sample_rate != nframes)
286 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS,
287 (NULL), ("Jack changed the sample rate, which is not supported"));
294 jack_buffer_size_cb (jack_nframes_t nframes, void *arg)
296 GstJackAudioSrc *src;
297 GstJackRingBuffer *abuf;
299 abuf = GST_JACK_RING_BUFFER_CAST (arg);
300 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg));
302 if (abuf->buffer_size != -1 && abuf->buffer_size != nframes)
310 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS,
311 (NULL), ("Jack changed the buffer size, which is not supported"));
317 jack_shutdown_cb (void *arg)
319 GstJackAudioSrc *src;
321 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg));
323 GST_DEBUG_OBJECT (src, "shutdown");
325 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND,
326 (NULL), ("Jack server shutdown"));
330 gst_jack_ring_buffer_init (GstJackRingBuffer * buf,
331 GstJackRingBufferClass * g_class)
334 buf->buffer_size = -1;
335 buf->sample_rate = -1;
338 /* the _open_device method should make a connection with the server
341 gst_jack_ring_buffer_open_device (GstRingBuffer * buf)
343 GstJackAudioSrc *src;
344 jack_status_t status = 0;
347 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
349 GST_DEBUG_OBJECT (src, "open");
351 if (src->client_name) {
352 name = src->client_name;
354 name = g_get_application_name ();
359 src->client = gst_jack_audio_client_new (name, src->server,
361 GST_JACK_CLIENT_SOURCE,
363 jack_process_cb, jack_buffer_size_cb, jack_sample_rate_cb, buf, &status);
364 if (src->client == NULL)
367 GST_DEBUG_OBJECT (src, "opened");
374 if (status & (JackServerFailed | JackFailure)) {
375 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND,
376 (_("Jack server not found")),
377 ("Cannot connect to the Jack server (status %d)", status));
379 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ,
380 (NULL), ("Jack client open error (status %d)", status));
386 /* close the connection with the server
389 gst_jack_ring_buffer_close_device (GstRingBuffer * buf)
391 GstJackAudioSrc *src;
393 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
395 GST_DEBUG_OBJECT (src, "close");
397 gst_jack_audio_src_free_channels (src);
398 gst_jack_audio_client_free (src->client);
405 /* allocate a buffer and setup resources to process the audio samples of
406 * the format as specified in @spec.
408 * We allocate N jack ports, one for each channel. If we are asked to
409 * automatically make a connection with physical ports, we connect as many
410 * ports as there are physical ports, leaving leftover ports unconnected.
412 * It is assumed that samplerate and number of channels are acceptable since our
413 * getcaps method will always provide correct values. If unacceptable caps are
414 * received for some reason, we fail here.
417 gst_jack_ring_buffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
419 GstJackAudioSrc *src;
420 GstJackRingBuffer *abuf;
422 gint sample_rate, buffer_size;
423 gint i, channels, res;
424 jack_client_t *client;
426 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
427 abuf = GST_JACK_RING_BUFFER_CAST (buf);
429 GST_DEBUG_OBJECT (src, "acquire");
431 client = gst_jack_audio_client_get_client (src->client);
433 /* sample rate must be that of the server */
434 sample_rate = jack_get_sample_rate (client);
435 if (sample_rate != spec->rate)
436 goto wrong_samplerate;
438 channels = spec->channels;
440 if (!gst_jack_audio_src_allocate_channels (src, channels))
443 gst_jack_set_layout_on_caps (&spec->caps, channels);
445 buffer_size = jack_get_buffer_size (client);
447 /* the segment size in bytes, this is large enough to hold a buffer of 32bit floats
448 * for all channels */
449 spec->segsize = buffer_size * sizeof (gfloat) * channels;
450 spec->latency_time = gst_util_uint64_scale (spec->segsize,
451 (GST_SECOND / GST_USECOND), spec->rate * spec->bytes_per_sample);
452 /* segtotal based on buffer-time latency */
453 spec->segtotal = spec->buffer_time / spec->latency_time;
454 if (spec->segtotal < 2) {
456 spec->buffer_time = spec->latency_time * spec->segtotal;
459 GST_DEBUG_OBJECT (src, "buffer time: %" G_GINT64_FORMAT " usec",
461 GST_DEBUG_OBJECT (src, "latency time: %" G_GINT64_FORMAT " usec",
463 GST_DEBUG_OBJECT (src, "buffer_size %d, segsize %d, segtotal %d",
464 buffer_size, spec->segsize, spec->segtotal);
466 /* allocate the ringbuffer memory now */
467 buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize);
468 memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data));
470 if ((res = gst_jack_audio_client_set_active (src->client, TRUE)))
471 goto could_not_activate;
473 /* if we need to automatically connect the ports, do so now. We must do this
474 * after activating the client. */
475 if (src->connect == GST_JACK_CONNECT_AUTO
476 || src->connect == GST_JACK_CONNECT_AUTO_FORCED) {
477 /* find all the physical output ports. A physical output port is a port
478 * associated with a hardware device. Someone needs connect to a physical
479 * port in order to capture something. */
481 jack_get_ports (client, NULL, NULL,
482 JackPortIsPhysical | JackPortIsOutput);
484 /* no ports? fine then we don't do anything except for posting a warning
486 GST_ELEMENT_WARNING (src, RESOURCE, NOT_FOUND, (NULL),
487 ("No physical output ports found, leaving ports unconnected"));
491 for (i = 0; i < channels; i++) {
492 /* stop when all output ports are exhausted */
493 if (ports[i] == NULL) {
494 /* post a warning that we could not connect all ports */
495 GST_ELEMENT_WARNING (src, RESOURCE, NOT_FOUND, (NULL),
496 ("No more physical ports, leaving some ports unconnected"));
499 GST_DEBUG_OBJECT (src, "try connecting to %s",
500 jack_port_name (src->ports[i]));
502 /* connect the physical port to a port */
503 res = jack_connect (client, ports[i], jack_port_name (src->ports[i]));
504 if (res != 0 && res != EEXIST)
511 abuf->sample_rate = sample_rate;
512 abuf->buffer_size = buffer_size;
513 abuf->channels = spec->channels;
520 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
521 ("Wrong samplerate, server is running at %d and we received %d",
522 sample_rate, spec->rate));
527 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
528 ("Cannot allocate more Jack ports"));
533 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
534 ("Could not activate client (%d:%s)", res, g_strerror (res)));
539 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
540 ("Could not connect input ports to physical ports (%d:%s)",
541 res, g_strerror (res)));
547 /* function is called with LOCK */
549 gst_jack_ring_buffer_release (GstRingBuffer * buf)
551 GstJackAudioSrc *src;
552 GstJackRingBuffer *abuf;
555 abuf = GST_JACK_RING_BUFFER_CAST (buf);
556 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
558 GST_DEBUG_OBJECT (src, "release");
560 if ((res = gst_jack_audio_client_set_active (src->client, FALSE))) {
561 /* we only warn, this means the server is probably shut down and the client
563 GST_ELEMENT_WARNING (src, RESOURCE, CLOSE, (NULL),
564 ("Could not deactivate Jack client (%d)", res));
568 abuf->buffer_size = -1;
569 abuf->sample_rate = -1;
571 /* free the buffer */
572 gst_buffer_unref (buf->data);
579 gst_jack_ring_buffer_start (GstRingBuffer * buf)
581 GstJackAudioSrc *src;
583 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
585 GST_DEBUG_OBJECT (src, "start");
587 if (src->transport == GST_JACK_TRANSPORT_MASTER) {
588 jack_client_t *client;
590 client = gst_jack_audio_client_get_client (src->client);
591 jack_transport_start (client);
598 gst_jack_ring_buffer_pause (GstRingBuffer * buf)
600 GstJackAudioSrc *src;
602 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
604 GST_DEBUG_OBJECT (src, "pause");
606 if (src->transport == GST_JACK_TRANSPORT_MASTER) {
607 jack_client_t *client;
609 client = gst_jack_audio_client_get_client (src->client);
610 jack_transport_stop (client);
617 gst_jack_ring_buffer_stop (GstRingBuffer * buf)
619 GstJackAudioSrc *src;
621 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
623 GST_DEBUG_OBJECT (src, "stop");
625 if (src->transport == GST_JACK_TRANSPORT_MASTER) {
626 jack_client_t *client;
628 client = gst_jack_audio_client_get_client (src->client);
629 jack_transport_stop (client);
635 #if defined (HAVE_JACK_0_120_1) || defined(HAVE_JACK_1_9_7)
637 gst_jack_ring_buffer_delay (GstRingBuffer * buf)
639 GstJackAudioSrc *src;
641 jack_latency_range_t range;
643 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
645 for (i = 0; i < src->port_count; i++) {
646 jack_port_get_latency_range (src->ports[i], JackCaptureLatency, &range);
651 GST_DEBUG_OBJECT (src, "delay %u", res);
655 #else /* !(defined (HAVE_JACK_0_120_1) || defined(HAVE_JACK_1_9_7)) */
657 gst_jack_ring_buffer_delay (GstRingBuffer * buf)
659 GstJackAudioSrc *src;
662 jack_client_t *client;
664 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
666 client = gst_jack_audio_client_get_client (src->client);
668 for (i = 0; i < src->port_count; i++) {
669 latency = jack_port_get_total_latency (client, src->ports[i]);
674 GST_DEBUG_OBJECT (src, "delay %u", res);
680 /* Audiosrc signals and args */
687 #define DEFAULT_PROP_CONNECT GST_JACK_CONNECT_AUTO
688 #define DEFAULT_PROP_SERVER NULL
689 #define DEFAULT_PROP_CLIENT_NAME NULL
690 #define DEFAULT_PROP_TRANSPORT GST_JACK_TRANSPORT_AUTONOMOUS
704 /* the capabilities of the inputs and outputs.
706 * describe the real formats here.
709 static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
712 GST_STATIC_CAPS ("audio/x-raw-float, "
713 "endianness = (int) BYTE_ORDER, "
715 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
718 #define _do_init(bla) \
719 GST_DEBUG_CATEGORY_INIT(gst_jack_audio_src_debug, "jacksrc", 0, "jacksrc element");
721 GST_BOILERPLATE_FULL (GstJackAudioSrc, gst_jack_audio_src, GstBaseAudioSrc,
722 GST_TYPE_BASE_AUDIO_SRC, _do_init);
724 static void gst_jack_audio_src_dispose (GObject * object);
725 static void gst_jack_audio_src_set_property (GObject * object, guint prop_id,
726 const GValue * value, GParamSpec * pspec);
727 static void gst_jack_audio_src_get_property (GObject * object, guint prop_id,
728 GValue * value, GParamSpec * pspec);
730 static GstCaps *gst_jack_audio_src_getcaps (GstBaseSrc * bsrc);
731 static GstRingBuffer *gst_jack_audio_src_create_ringbuffer (GstBaseAudioSrc *
734 /* GObject vmethod implementations */
737 gst_jack_audio_src_base_init (gpointer gclass)
739 GstElementClass *element_class = GST_ELEMENT_CLASS (gclass);
741 gst_element_class_add_static_pad_template (element_class, &src_factory);
742 gst_element_class_set_details_simple (element_class, "Audio Source (Jack)",
743 "Source/Audio", "Captures audio from a JACK server",
744 "Tristan Matthews <tristan@sat.qc.ca>");
747 /* initialize the jack_audio_src's class */
749 gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass)
751 GObjectClass *gobject_class;
752 GstBaseSrcClass *gstbasesrc_class;
753 GstBaseAudioSrcClass *gstbaseaudiosrc_class;
755 gobject_class = (GObjectClass *) klass;
757 gstbasesrc_class = (GstBaseSrcClass *) klass;
758 gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass;
760 gobject_class->dispose = gst_jack_audio_src_dispose;
761 gobject_class->set_property = gst_jack_audio_src_set_property;
762 gobject_class->get_property = gst_jack_audio_src_get_property;
764 g_object_class_install_property (gobject_class, PROP_CONNECT,
765 g_param_spec_enum ("connect", "Connect",
766 "Specify how the input ports will be connected",
767 GST_TYPE_JACK_CONNECT, DEFAULT_PROP_CONNECT,
768 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
770 g_object_class_install_property (gobject_class, PROP_SERVER,
771 g_param_spec_string ("server", "Server",
772 "The Jack server to connect to (NULL = default)",
773 DEFAULT_PROP_SERVER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
776 * GstJackAudioSrc:client-name
778 * The client name to use.
782 g_object_class_install_property (gobject_class, PROP_CLIENT_NAME,
783 g_param_spec_string ("client-name", "Client name",
784 "The client name of the Jack instance (NULL = default)",
785 DEFAULT_PROP_CLIENT_NAME,
786 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
788 g_object_class_install_property (gobject_class, PROP_CLIENT,
789 g_param_spec_boxed ("client", "JackClient", "Handle for jack client",
790 GST_TYPE_JACK_CLIENT,
791 GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
792 G_PARAM_STATIC_STRINGS));
795 * GstJackAudioSink:transport
797 * The jack transport behaviour for the client.
801 g_object_class_install_property (gobject_class, PROP_TRANSPORT,
802 g_param_spec_enum ("transport", "Transport mode",
803 "Jack transport behaviour of the client",
804 GST_TYPE_JACK_TRANSPORT, DEFAULT_PROP_TRANSPORT,
805 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
807 gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_src_getcaps);
808 gstbaseaudiosrc_class->create_ringbuffer =
809 GST_DEBUG_FUNCPTR (gst_jack_audio_src_create_ringbuffer);
811 /* ref class from a thread-safe context to work around missing bit of
812 * thread-safety in GObject */
813 g_type_class_ref (GST_TYPE_JACK_RING_BUFFER);
815 gst_jack_audio_client_init ();
819 gst_jack_audio_src_init (GstJackAudioSrc * src, GstJackAudioSrcClass * gclass)
821 //gst_base_src_set_live(GST_BASE_SRC (src), TRUE);
822 src->connect = DEFAULT_PROP_CONNECT;
823 src->server = g_strdup (DEFAULT_PROP_SERVER);
828 src->client_name = g_strdup (DEFAULT_PROP_CLIENT_NAME);
829 src->transport = DEFAULT_PROP_TRANSPORT;
833 gst_jack_audio_src_dispose (GObject * object)
835 GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
837 gst_caps_replace (&src->caps, NULL);
839 if (src->client_name != NULL) {
840 g_free (src->client_name);
841 src->client_name = NULL;
844 G_OBJECT_CLASS (parent_class)->dispose (object);
848 gst_jack_audio_src_set_property (GObject * object, guint prop_id,
849 const GValue * value, GParamSpec * pspec)
851 GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
854 case PROP_CLIENT_NAME:
855 g_free (src->client_name);
856 src->client_name = g_value_dup_string (value);
859 src->connect = g_value_get_enum (value);
862 g_free (src->server);
863 src->server = g_value_dup_string (value);
866 if (GST_STATE (src) == GST_STATE_NULL ||
867 GST_STATE (src) == GST_STATE_READY) {
868 src->jclient = g_value_get_boxed (value);
872 src->transport = g_value_get_enum (value);
875 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
881 gst_jack_audio_src_get_property (GObject * object, guint prop_id,
882 GValue * value, GParamSpec * pspec)
884 GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
887 case PROP_CLIENT_NAME:
888 g_value_set_string (value, src->client_name);
891 g_value_set_enum (value, src->connect);
894 g_value_set_string (value, src->server);
897 g_value_set_boxed (value, src->jclient);
900 g_value_set_enum (value, src->transport);
903 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
909 gst_jack_audio_src_getcaps (GstBaseSrc * bsrc)
911 GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (bsrc);
915 jack_client_t *client;
917 if (src->client == NULL)
920 client = gst_jack_audio_client_get_client (src->client);
922 if (src->connect == GST_JACK_CONNECT_AUTO) {
923 /* get a port count, this is the number of channels we can automatically
925 ports = jack_get_ports (client, NULL, NULL,
926 JackPortIsPhysical | JackPortIsOutput);
929 for (; ports[max]; max++);
935 /* we allow any number of pads, something else is going to connect the
941 rate = jack_get_sample_rate (client);
943 GST_DEBUG_OBJECT (src, "got %d-%d ports, samplerate: %d", min, max, rate);
946 src->caps = gst_caps_new_simple ("audio/x-raw-float",
947 "endianness", G_TYPE_INT, G_BYTE_ORDER,
948 "width", G_TYPE_INT, 32,
949 "rate", G_TYPE_INT, rate,
950 "channels", GST_TYPE_INT_RANGE, min, max, NULL);
952 GST_INFO_OBJECT (src, "returning caps %" GST_PTR_FORMAT, src->caps);
954 return gst_caps_ref (src->caps);
959 GST_DEBUG_OBJECT (src, "device not open, using template caps");
960 /* base class will get template caps for us when we return NULL */
965 static GstRingBuffer *
966 gst_jack_audio_src_create_ringbuffer (GstBaseAudioSrc * src)
968 GstRingBuffer *buffer;
970 buffer = g_object_new (GST_TYPE_JACK_RING_BUFFER, NULL);
971 GST_DEBUG_OBJECT (src, "created ringbuffer @%p", buffer);