1 /* GStreamer
2 * Copyright (C) 2008 Tristan Matthews <tristan@sat.qc.ca>
3 *
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:
10 *
11 * The above copyright notice and this permission notice shall be included in
12 * all copies or substantial portions of the Software.
13 *
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.
21 *
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
25 * mentioned above:
26 *
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.
31 *
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.
36 *
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., 51 Franklin St, Fifth Floor,
40 * Boston, MA 02110-1301, USA.
41 */
42
43 /**
44 * SECTION:element-jackaudiosrc
45 * @see_also: #GstAudioBaseSrc, #GstAudioRingBuffer
46 *
47 * A Src that inputs data from Jack ports.
48 *
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.
52 *
53 * The samplerate as exposed on the caps is always the same as the samplerate of
54 * the jack server.
55 *
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.
58 *
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.
62 *
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.
66 *
67 * <refsect2>
68 * <title>Example launch line</title>
69 * |[
70 * gst-launch-1.0 jackaudiosrc connect=0 ! jackaudiosink connect=0
71 * ]| Get audio input into gstreamer from jack.
72 * </refsect2>
73 */
74
75 #ifdef HAVE_CONFIG_H
76 #include "config.h"
77 #endif
78
79 #include <gst/gst-i18n-plugin.h>
80 #include <stdlib.h>
81 #include <string.h>
82
83 #include <gst/audio/audio.h>
84
85 #include "gstjackaudiosrc.h"
86 #include "gstjackringbuffer.h"
87 #include "gstjackutil.h"
88
89 GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_src_debug);
90 #define GST_CAT_DEFAULT gst_jack_audio_src_debug
91
92 static gboolean
gst_jack_audio_src_allocate_channels(GstJackAudioSrc * src,gint channels)93 gst_jack_audio_src_allocate_channels (GstJackAudioSrc * src, gint channels)
94 {
95 jack_client_t *client;
96
97 client = gst_jack_audio_client_get_client (src->client);
98
99 /* remove ports we don't need */
100 while (src->port_count > channels)
101 jack_port_unregister (client, src->ports[--src->port_count]);
102
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);
106
107 /* create an input port for each channel */
108 while (src->port_count < channels) {
109 gchar *name;
110
111 /* port names start from 1 and are local to the element */
112 name =
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,
117 JackPortIsInput, 0);
118 if (src->ports[src->port_count] == NULL)
119 return FALSE;
120
121 src->port_count++;
122
123 g_free (name);
124 }
125 return TRUE;
126 }
127
128 static void
gst_jack_audio_src_free_channels(GstJackAudioSrc * src)129 gst_jack_audio_src_free_channels (GstJackAudioSrc * src)
130 {
131 gint res, i = 0;
132 jack_client_t *client;
133
134 client = gst_jack_audio_client_get_client (src->client);
135
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);
141
142 src->port_count--;
143 }
144 g_free (src->ports);
145 src->ports = NULL;
146 g_free (src->buffers);
147 src->buffers = NULL;
148 }
149
150 /* ringbuffer abstract base class */
151 static GType
gst_jack_ring_buffer_get_type(void)152 gst_jack_ring_buffer_get_type (void)
153 {
154 static volatile gsize ringbuffer_type = 0;
155
156 if (g_once_init_enter (&ringbuffer_type)) {
157 static const GTypeInfo ringbuffer_info = { sizeof (GstJackRingBufferClass),
158 NULL,
159 NULL,
160 (GClassInitFunc) gst_jack_ring_buffer_class_init,
161 NULL,
162 NULL,
163 sizeof (GstJackRingBuffer),
164 0,
165 (GInstanceInitFunc) gst_jack_ring_buffer_init,
166 NULL
167 };
168 GType tmp = g_type_register_static (GST_TYPE_AUDIO_RING_BUFFER,
169 "GstJackAudioSrcRingBuffer", &ringbuffer_info, 0);
170 g_once_init_leave (&ringbuffer_type, tmp);
171 }
172
173 return (GType) ringbuffer_type;
174 }
175
176 static void
gst_jack_ring_buffer_class_init(GstJackRingBufferClass * klass)177 gst_jack_ring_buffer_class_init (GstJackRingBufferClass * klass)
178 {
179 GstAudioRingBufferClass *gstringbuffer_class;
180
181 gstringbuffer_class = (GstAudioRingBufferClass *) klass;
182
183 ring_parent_class = g_type_class_peek_parent (klass);
184
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);
197
198 gstringbuffer_class->delay = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_delay);
199 }
200
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.
203 */
204 static int
jack_process_cb(jack_nframes_t nframes,void * arg)205 jack_process_cb (jack_nframes_t nframes, void *arg)
206 {
207 GstJackAudioSrc *src;
208 GstAudioRingBuffer *buf;
209 gint len;
210 guint8 *writeptr;
211 gint writeseg;
212 gint channels, i, j, flen;
213 sample_t *data;
214
215 buf = GST_AUDIO_RING_BUFFER_CAST (arg);
216 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
217
218 channels = GST_AUDIO_INFO_CHANNELS (&buf->spec.info);
219
220 /* get input buffers */
221 for (i = 0; i < channels; i++)
222 src->buffers[i] =
223 (sample_t *) jack_port_get_buffer (src->ports[i], nframes);
224
225 if (gst_audio_ring_buffer_prepare_read (buf, &writeseg, &writeptr, &len)) {
226 flen = len / channels;
227
228 /* the number of samples must be exactly the segment size */
229 if (nframes * sizeof (sample_t) != flen)
230 goto wrong_size;
231
232 /* the samples in the jack input buffers have to be interleaved into the
233 * ringbuffer */
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];
238
239 GST_DEBUG ("copy %d frames: %p, %d bytes, %d channels", nframes, writeptr,
240 len / channels, channels);
241
242 /* we wrote one segment */
243 gst_audio_ring_buffer_advance (buf, 1);
244 }
245 return 0;
246
247 /* ERRORS */
248 wrong_size:
249 {
250 GST_ERROR_OBJECT (src, "nbytes (%d) != flen (%d)",
251 (gint) (nframes * sizeof (sample_t)), flen);
252 return 1;
253 }
254 }
255
256 /* we error out */
257 static int
jack_sample_rate_cb(jack_nframes_t nframes,void * arg)258 jack_sample_rate_cb (jack_nframes_t nframes, void *arg)
259 {
260 GstJackAudioSrc *src;
261 GstJackRingBuffer *abuf;
262
263 abuf = GST_JACK_RING_BUFFER_CAST (arg);
264 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg));
265
266 if (abuf->sample_rate != -1 && abuf->sample_rate != nframes)
267 goto not_supported;
268
269 return 0;
270
271 /* ERRORS */
272 not_supported:
273 {
274 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS,
275 (NULL), ("Jack changed the sample rate, which is not supported"));
276 return 1;
277 }
278 }
279
280 /* we error out */
281 static int
jack_buffer_size_cb(jack_nframes_t nframes,void * arg)282 jack_buffer_size_cb (jack_nframes_t nframes, void *arg)
283 {
284 GstJackAudioSrc *src;
285 GstJackRingBuffer *abuf;
286
287 abuf = GST_JACK_RING_BUFFER_CAST (arg);
288 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg));
289
290 if (abuf->buffer_size != -1 && abuf->buffer_size != nframes)
291 goto not_supported;
292
293 return 0;
294
295 /* ERRORS */
296 not_supported:
297 {
298 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS,
299 (NULL), ("Jack changed the buffer size, which is not supported"));
300 return 1;
301 }
302 }
303
304 static void
jack_shutdown_cb(void * arg)305 jack_shutdown_cb (void *arg)
306 {
307 GstJackAudioSrc *src;
308
309 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg));
310
311 GST_DEBUG_OBJECT (src, "shutdown");
312
313 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND,
314 (NULL), ("Jack server shutdown"));
315 }
316
317 static void
gst_jack_ring_buffer_init(GstJackRingBuffer * buf,GstJackRingBufferClass * g_class)318 gst_jack_ring_buffer_init (GstJackRingBuffer * buf,
319 GstJackRingBufferClass * g_class)
320 {
321 buf->channels = -1;
322 buf->buffer_size = -1;
323 buf->sample_rate = -1;
324 }
325
326 /* the _open_device method should make a connection with the server
327 */
328 static gboolean
gst_jack_ring_buffer_open_device(GstAudioRingBuffer * buf)329 gst_jack_ring_buffer_open_device (GstAudioRingBuffer * buf)
330 {
331 GstJackAudioSrc *src;
332 jack_status_t status = 0;
333 const gchar *name;
334
335 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
336
337 GST_DEBUG_OBJECT (src, "open");
338
339 if (src->client_name) {
340 name = src->client_name;
341 } else {
342 name = g_get_application_name ();
343 }
344 if (!name)
345 name = "GStreamer";
346
347 src->client = gst_jack_audio_client_new (name, src->server,
348 src->jclient,
349 GST_JACK_CLIENT_SOURCE,
350 jack_shutdown_cb,
351 jack_process_cb, jack_buffer_size_cb, jack_sample_rate_cb, buf, &status);
352 if (src->client == NULL)
353 goto could_not_open;
354
355 GST_DEBUG_OBJECT (src, "opened");
356
357 return TRUE;
358
359 /* ERRORS */
360 could_not_open:
361 {
362 if (status & (JackServerFailed | JackFailure)) {
363 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND,
364 (_("Jack server not found")),
365 ("Cannot connect to the Jack server (status %d)", status));
366 } else {
367 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ,
368 (NULL), ("Jack client open error (status %d)", status));
369 }
370 return FALSE;
371 }
372 }
373
374 /* close the connection with the server
375 */
376 static gboolean
gst_jack_ring_buffer_close_device(GstAudioRingBuffer * buf)377 gst_jack_ring_buffer_close_device (GstAudioRingBuffer * buf)
378 {
379 GstJackAudioSrc *src;
380
381 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
382
383 GST_DEBUG_OBJECT (src, "close");
384
385 gst_jack_audio_src_free_channels (src);
386 gst_jack_audio_client_free (src->client);
387 src->client = NULL;
388
389 return TRUE;
390 }
391
392
393 /* allocate a buffer and setup resources to process the audio samples of
394 * the format as specified in @spec.
395 *
396 * We allocate N jack ports, one for each channel. If we are asked to
397 * automatically make a connection with physical ports, we connect as many
398 * ports as there are physical ports, leaving leftover ports unconnected.
399 *
400 * It is assumed that samplerate and number of channels are acceptable since our
401 * getcaps method will always provide correct values. If unacceptable caps are
402 * received for some reason, we fail here.
403 */
404 static gboolean
gst_jack_ring_buffer_acquire(GstAudioRingBuffer * buf,GstAudioRingBufferSpec * spec)405 gst_jack_ring_buffer_acquire (GstAudioRingBuffer * buf,
406 GstAudioRingBufferSpec * spec)
407 {
408 GstJackAudioSrc *src;
409 GstJackRingBuffer *abuf;
410 const char **ports;
411 gint sample_rate, buffer_size;
412 gint i, bpf, rate, channels, res;
413 jack_client_t *client;
414
415 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
416 abuf = GST_JACK_RING_BUFFER_CAST (buf);
417
418 GST_DEBUG_OBJECT (src, "acquire");
419
420 client = gst_jack_audio_client_get_client (src->client);
421
422 rate = GST_AUDIO_INFO_RATE (&spec->info);
423
424 /* sample rate must be that of the server */
425 sample_rate = jack_get_sample_rate (client);
426 if (sample_rate != rate)
427 goto wrong_samplerate;
428
429 bpf = GST_AUDIO_INFO_BPF (&spec->info);
430 channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
431
432 if (!gst_jack_audio_src_allocate_channels (src, channels))
433 goto out_of_ports;
434
435 gst_jack_set_layout (buf, spec);
436
437 buffer_size = jack_get_buffer_size (client);
438
439 /* the segment size in bytes, this is large enough to hold a buffer of 32bit floats
440 * for all channels */
441 spec->segsize = buffer_size * sizeof (gfloat) * channels;
442 spec->latency_time = gst_util_uint64_scale (spec->segsize,
443 (GST_SECOND / GST_USECOND), rate * bpf);
444 /* segtotal based on buffer-time latency */
445 spec->segtotal = spec->buffer_time / spec->latency_time;
446 if (spec->segtotal < 2) {
447 spec->segtotal = 2;
448 spec->buffer_time = spec->latency_time * spec->segtotal;
449 }
450
451 GST_DEBUG_OBJECT (src, "buffer time: %" G_GINT64_FORMAT " usec",
452 spec->buffer_time);
453 GST_DEBUG_OBJECT (src, "latency time: %" G_GINT64_FORMAT " usec",
454 spec->latency_time);
455 GST_DEBUG_OBJECT (src, "buffer_size %d, segsize %d, segtotal %d",
456 buffer_size, spec->segsize, spec->segtotal);
457
458 /* allocate the ringbuffer memory now */
459 buf->size = spec->segtotal * spec->segsize;
460 buf->memory = g_malloc0 (buf->size);
461
462 if ((res = gst_jack_audio_client_set_active (src->client, TRUE)))
463 goto could_not_activate;
464
465 /* if we need to automatically connect the ports, do so now. We must do this
466 * after activating the client. */
467 if (src->connect == GST_JACK_CONNECT_AUTO
468 || src->connect == GST_JACK_CONNECT_AUTO_FORCED) {
469 /* find all the physical output ports. A physical output port is a port
470 * associated with a hardware device. Someone needs connect to a physical
471 * port in order to capture something. */
472
473 if (src->port_pattern == NULL) {
474 ports = jack_get_ports (client, NULL, NULL,
475 JackPortIsPhysical | JackPortIsOutput);
476 } else {
477 ports = jack_get_ports (client, src->port_pattern, NULL,
478 JackPortIsOutput);
479 }
480
481 if (ports == NULL) {
482 /* no ports? fine then we don't do anything except for posting a warning
483 * message. */
484 GST_ELEMENT_WARNING (src, RESOURCE, NOT_FOUND, (NULL),
485 ("No physical output ports found, leaving ports unconnected"));
486 goto done;
487 }
488
489 for (i = 0; i < channels; i++) {
490 /* stop when all output ports are exhausted */
491 if (ports[i] == NULL) {
492 /* post a warning that we could not connect all ports */
493 GST_ELEMENT_WARNING (src, RESOURCE, NOT_FOUND, (NULL),
494 ("No more physical ports, leaving some ports unconnected"));
495 break;
496 }
497 GST_DEBUG_OBJECT (src, "try connecting to %s",
498 jack_port_name (src->ports[i]));
499
500 /* connect the physical port to a port */
501 res = jack_connect (client, ports[i], jack_port_name (src->ports[i]));
502 if (res != 0 && res != EEXIST)
503 goto cannot_connect;
504 }
505 free (ports);
506 }
507 done:
508
509 abuf->sample_rate = sample_rate;
510 abuf->buffer_size = buffer_size;
511 abuf->channels = channels;
512
513 return TRUE;
514
515 /* ERRORS */
516 wrong_samplerate:
517 {
518 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
519 ("Wrong samplerate, server is running at %d and we received %d",
520 sample_rate, rate));
521 return FALSE;
522 }
523 out_of_ports:
524 {
525 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
526 ("Cannot allocate more Jack ports"));
527 return FALSE;
528 }
529 could_not_activate:
530 {
531 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
532 ("Could not activate client (%d:%s)", res, g_strerror (res)));
533 return FALSE;
534 }
535 cannot_connect:
536 {
537 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
538 ("Could not connect input ports to physical ports (%d:%s)",
539 res, g_strerror (res)));
540 free (ports);
541 return FALSE;
542 }
543 }
544
545 /* function is called with LOCK */
546 static gboolean
gst_jack_ring_buffer_release(GstAudioRingBuffer * buf)547 gst_jack_ring_buffer_release (GstAudioRingBuffer * buf)
548 {
549 GstJackAudioSrc *src;
550 GstJackRingBuffer *abuf;
551 gint res;
552
553 abuf = GST_JACK_RING_BUFFER_CAST (buf);
554 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
555
556 GST_DEBUG_OBJECT (src, "release");
557
558 if ((res = gst_jack_audio_client_set_active (src->client, FALSE))) {
559 /* we only warn, this means the server is probably shut down and the client
560 * is gone anyway. */
561 GST_ELEMENT_WARNING (src, RESOURCE, CLOSE, (NULL),
562 ("Could not deactivate Jack client (%d)", res));
563 }
564
565 abuf->channels = -1;
566 abuf->buffer_size = -1;
567 abuf->sample_rate = -1;
568
569 /* free the buffer */
570 g_free (buf->memory);
571 buf->memory = NULL;
572
573 return TRUE;
574 }
575
576 static gboolean
gst_jack_ring_buffer_start(GstAudioRingBuffer * buf)577 gst_jack_ring_buffer_start (GstAudioRingBuffer * buf)
578 {
579 GstJackAudioSrc *src;
580
581 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
582
583 GST_DEBUG_OBJECT (src, "start");
584
585 if (src->transport & GST_JACK_TRANSPORT_MASTER) {
586 jack_client_t *client;
587
588 client = gst_jack_audio_client_get_client (src->client);
589 jack_transport_start (client);
590 }
591
592 return TRUE;
593 }
594
595 static gboolean
gst_jack_ring_buffer_pause(GstAudioRingBuffer * buf)596 gst_jack_ring_buffer_pause (GstAudioRingBuffer * buf)
597 {
598 GstJackAudioSrc *src;
599
600 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
601
602 GST_DEBUG_OBJECT (src, "pause");
603
604 if (src->transport & GST_JACK_TRANSPORT_MASTER) {
605 jack_client_t *client;
606
607 client = gst_jack_audio_client_get_client (src->client);
608 jack_transport_stop (client);
609 }
610
611 return TRUE;
612 }
613
614 static gboolean
gst_jack_ring_buffer_stop(GstAudioRingBuffer * buf)615 gst_jack_ring_buffer_stop (GstAudioRingBuffer * buf)
616 {
617 GstJackAudioSrc *src;
618
619 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
620
621 GST_DEBUG_OBJECT (src, "stop");
622
623 if (src->transport & GST_JACK_TRANSPORT_MASTER) {
624 jack_client_t *client;
625
626 client = gst_jack_audio_client_get_client (src->client);
627 jack_transport_stop (client);
628 }
629
630 return TRUE;
631 }
632
633 #if defined (HAVE_JACK_0_120_1) || defined(HAVE_JACK_1_9_7)
634 static guint
gst_jack_ring_buffer_delay(GstAudioRingBuffer * buf)635 gst_jack_ring_buffer_delay (GstAudioRingBuffer * buf)
636 {
637 GstJackAudioSrc *src;
638 guint i, res = 0;
639 jack_latency_range_t range;
640
641 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
642
643 for (i = 0; i < src->port_count; i++) {
644 jack_port_get_latency_range (src->ports[i], JackCaptureLatency, &range);
645 if (range.max > res)
646 res = range.max;
647 }
648
649 GST_DEBUG_OBJECT (src, "delay %u", res);
650
651 return res;
652 }
653 #else /* !(defined (HAVE_JACK_0_120_1) || defined(HAVE_JACK_1_9_7)) */
654 static guint
gst_jack_ring_buffer_delay(GstAudioRingBuffer * buf)655 gst_jack_ring_buffer_delay (GstAudioRingBuffer * buf)
656 {
657 GstJackAudioSrc *src;
658 guint i, res = 0;
659 guint latency;
660 jack_client_t *client;
661
662 src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
663
664 client = gst_jack_audio_client_get_client (src->client);
665
666 for (i = 0; i < src->port_count; i++) {
667 latency = jack_port_get_total_latency (client, src->ports[i]);
668 if (latency > res)
669 res = latency;
670 }
671
672 GST_DEBUG_OBJECT (src, "delay %u", res);
673
674 return res;
675 }
676 #endif
677
678 /* Audiosrc signals and args */
679 enum
680 {
681 /* FILL ME */
682 LAST_SIGNAL
683 };
684
685 #define DEFAULT_PROP_CONNECT GST_JACK_CONNECT_AUTO
686 #define DEFAULT_PROP_SERVER NULL
687 #define DEFAULT_PROP_CLIENT_NAME NULL
688 #define DEFAULT_PROP_TRANSPORT GST_JACK_TRANSPORT_AUTONOMOUS
689 #define DEFAULT_PROP_PORT_PATTERN NULL
690 enum
691 {
692 PROP_0,
693 PROP_CONNECT,
694 PROP_SERVER,
695 PROP_CLIENT,
696 PROP_CLIENT_NAME,
697 PROP_PORT_PATTERN,
698 PROP_TRANSPORT,
699 PROP_LAST
700 };
701
702 /* the capabilities of the inputs and outputs.
703 *
704 * describe the real formats here.
705 */
706
707 static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
708 GST_PAD_SRC,
709 GST_PAD_ALWAYS,
710 GST_STATIC_CAPS ("audio/x-raw, "
711 "format = (string) " GST_JACK_FORMAT_STR ", "
712 "layout = (string) interleaved, "
713 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
714 );
715
716 #define gst_jack_audio_src_parent_class parent_class
717 G_DEFINE_TYPE (GstJackAudioSrc, gst_jack_audio_src, GST_TYPE_AUDIO_BASE_SRC);
718
719 static void gst_jack_audio_src_dispose (GObject * object);
720 static void gst_jack_audio_src_set_property (GObject * object, guint prop_id,
721 const GValue * value, GParamSpec * pspec);
722 static void gst_jack_audio_src_get_property (GObject * object, guint prop_id,
723 GValue * value, GParamSpec * pspec);
724
725 static GstCaps *gst_jack_audio_src_getcaps (GstBaseSrc * bsrc,
726 GstCaps * filter);
727 static GstAudioRingBuffer *gst_jack_audio_src_create_ringbuffer (GstAudioBaseSrc
728 * src);
729
730 /* GObject vmethod implementations */
731
732 /* initialize the jack_audio_src's class */
733 static void
gst_jack_audio_src_class_init(GstJackAudioSrcClass * klass)734 gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass)
735 {
736 GObjectClass *gobject_class;
737 GstElementClass *gstelement_class;
738 GstBaseSrcClass *gstbasesrc_class;
739 GstAudioBaseSrcClass *gstaudiobasesrc_class;
740
741 GST_DEBUG_CATEGORY_INIT (gst_jack_audio_src_debug, "jacksrc", 0,
742 "jacksrc element");
743
744 gobject_class = (GObjectClass *) klass;
745 gstelement_class = (GstElementClass *) klass;
746 gstbasesrc_class = (GstBaseSrcClass *) klass;
747 gstaudiobasesrc_class = (GstAudioBaseSrcClass *) klass;
748
749 gobject_class->dispose = gst_jack_audio_src_dispose;
750 gobject_class->set_property = gst_jack_audio_src_set_property;
751 gobject_class->get_property = gst_jack_audio_src_get_property;
752
753 g_object_class_install_property (gobject_class, PROP_CONNECT,
754 g_param_spec_enum ("connect", "Connect",
755 "Specify how the input ports will be connected",
756 GST_TYPE_JACK_CONNECT, DEFAULT_PROP_CONNECT,
757 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
758
759 g_object_class_install_property (gobject_class, PROP_SERVER,
760 g_param_spec_string ("server", "Server",
761 "The Jack server to connect to (NULL = default)",
762 DEFAULT_PROP_SERVER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
763
764 /**
765 * GstJackAudioSrc:client-name:
766 *
767 * The client name to use.
768 */
769 g_object_class_install_property (gobject_class, PROP_CLIENT_NAME,
770 g_param_spec_string ("client-name", "Client name",
771 "The client name of the Jack instance (NULL = default)",
772 DEFAULT_PROP_CLIENT_NAME,
773 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
774
775 g_object_class_install_property (gobject_class, PROP_CLIENT,
776 g_param_spec_boxed ("client", "JackClient", "Handle for jack client",
777 GST_TYPE_JACK_CLIENT,
778 GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
779 G_PARAM_STATIC_STRINGS));
780 /**
781 * GstJackAudioSrc:port-pattern
782 *
783 * autoconnect to ports matching pattern, when NULL connect to physical ports
784 *
785 * Since: 1.6
786 */
787 g_object_class_install_property (gobject_class, PROP_PORT_PATTERN,
788 g_param_spec_string ("port-pattern", "port pattern",
789 "A pattern to select which ports to connect to (NULL = first physical ports)",
790 DEFAULT_PROP_PORT_PATTERN,
791 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
792
793 /**
794 * GstJackAudioSink:transport:
795 *
796 * The jack transport behaviour for the client.
797 */
798 g_object_class_install_property (gobject_class, PROP_TRANSPORT,
799 g_param_spec_flags ("transport", "Transport mode",
800 "Jack transport behaviour of the client",
801 GST_TYPE_JACK_TRANSPORT, DEFAULT_PROP_TRANSPORT,
802 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
803
804 gst_element_class_add_static_pad_template (gstelement_class, &src_factory);
805
806 gst_element_class_set_static_metadata (gstelement_class,
807 "Audio Source (Jack)", "Source/Audio",
808 "Captures audio from a JACK server",
809 "Tristan Matthews <tristan@sat.qc.ca>");
810
811 gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_src_getcaps);
812 gstaudiobasesrc_class->create_ringbuffer =
813 GST_DEBUG_FUNCPTR (gst_jack_audio_src_create_ringbuffer);
814
815 /* ref class from a thread-safe context to work around missing bit of
816 * thread-safety in GObject */
817 g_type_class_ref (GST_TYPE_JACK_RING_BUFFER);
818
819 gst_jack_audio_client_init ();
820 }
821
822 static void
gst_jack_audio_src_init(GstJackAudioSrc * src)823 gst_jack_audio_src_init (GstJackAudioSrc * src)
824 {
825 //gst_base_src_set_live(GST_BASE_SRC (src), TRUE);
826 src->connect = DEFAULT_PROP_CONNECT;
827 src->server = g_strdup (DEFAULT_PROP_SERVER);
828 src->jclient = NULL;
829 src->ports = NULL;
830 src->port_count = 0;
831 src->buffers = NULL;
832 src->client_name = g_strdup (DEFAULT_PROP_CLIENT_NAME);
833 src->transport = DEFAULT_PROP_TRANSPORT;
834 }
835
836 static void
gst_jack_audio_src_dispose(GObject * object)837 gst_jack_audio_src_dispose (GObject * object)
838 {
839 GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
840
841 gst_caps_replace (&src->caps, NULL);
842
843 if (src->client_name != NULL) {
844 g_free (src->client_name);
845 src->client_name = NULL;
846 }
847
848 if (src->port_pattern != NULL) {
849 g_free (src->port_pattern);
850 src->port_pattern = NULL;
851 }
852
853 G_OBJECT_CLASS (parent_class)->dispose (object);
854 }
855
856 static void
gst_jack_audio_src_set_property(GObject * object,guint prop_id,const GValue * value,GParamSpec * pspec)857 gst_jack_audio_src_set_property (GObject * object, guint prop_id,
858 const GValue * value, GParamSpec * pspec)
859 {
860 GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
861
862 switch (prop_id) {
863 case PROP_CLIENT_NAME:
864 g_free (src->client_name);
865 src->client_name = g_value_dup_string (value);
866 break;
867 case PROP_PORT_PATTERN:
868 g_free (src->port_pattern);
869 src->port_pattern = g_value_dup_string (value);
870 break;
871 case PROP_CONNECT:
872 src->connect = g_value_get_enum (value);
873 break;
874 case PROP_SERVER:
875 g_free (src->server);
876 src->server = g_value_dup_string (value);
877 break;
878 case PROP_CLIENT:
879 if (GST_STATE (src) == GST_STATE_NULL ||
880 GST_STATE (src) == GST_STATE_READY) {
881 src->jclient = g_value_get_boxed (value);
882 }
883 break;
884 case PROP_TRANSPORT:
885 src->transport = g_value_get_flags (value);
886 break;
887 default:
888 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
889 break;
890 }
891 }
892
893 static void
gst_jack_audio_src_get_property(GObject * object,guint prop_id,GValue * value,GParamSpec * pspec)894 gst_jack_audio_src_get_property (GObject * object, guint prop_id,
895 GValue * value, GParamSpec * pspec)
896 {
897 GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
898
899 switch (prop_id) {
900 case PROP_CLIENT_NAME:
901 g_value_set_string (value, src->client_name);
902 break;
903 case PROP_PORT_PATTERN:
904 g_value_set_string (value, src->port_pattern);
905 break;
906 case PROP_CONNECT:
907 g_value_set_enum (value, src->connect);
908 break;
909 case PROP_SERVER:
910 g_value_set_string (value, src->server);
911 break;
912 case PROP_CLIENT:
913 g_value_set_boxed (value, src->jclient);
914 break;
915 case PROP_TRANSPORT:
916 g_value_set_flags (value, src->transport);
917 break;
918 default:
919 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
920 break;
921 }
922 }
923
924 static GstCaps *
gst_jack_audio_src_getcaps(GstBaseSrc * bsrc,GstCaps * filter)925 gst_jack_audio_src_getcaps (GstBaseSrc * bsrc, GstCaps * filter)
926 {
927 GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (bsrc);
928 const char **ports;
929 gint min, max;
930 gint rate;
931 jack_client_t *client;
932
933 if (src->client == NULL)
934 goto no_client;
935
936 client = gst_jack_audio_client_get_client (src->client);
937
938 if (src->connect == GST_JACK_CONNECT_AUTO) {
939 /* get a port count, this is the number of channels we can automatically
940 * connect. */
941 ports = jack_get_ports (client, NULL, NULL,
942 JackPortIsPhysical | JackPortIsOutput);
943 max = 0;
944 if (ports != NULL) {
945 for (; ports[max]; max++);
946
947 free (ports);
948 } else
949 max = 0;
950 } else {
951 /* we allow any number of pads, something else is going to connect the
952 * pads. */
953 max = G_MAXINT;
954 }
955 min = MIN (1, max);
956
957 rate = jack_get_sample_rate (client);
958
959 GST_DEBUG_OBJECT (src, "got %d-%d ports, samplerate: %d", min, max, rate);
960
961 if (!src->caps) {
962 src->caps = gst_caps_new_simple ("audio/x-raw",
963 "format", G_TYPE_STRING, GST_JACK_FORMAT_STR,
964 "layout", G_TYPE_STRING, "interleaved",
965 "rate", G_TYPE_INT, rate,
966 "channels", GST_TYPE_INT_RANGE, min, max, NULL);
967 }
968 GST_INFO_OBJECT (src, "returning caps %" GST_PTR_FORMAT, src->caps);
969
970 return gst_caps_ref (src->caps);
971
972 /* ERRORS */
973 no_client:
974 {
975 GST_DEBUG_OBJECT (src, "device not open, using template caps");
976 /* base class will get template caps for us when we return NULL */
977 return NULL;
978 }
979 }
980
981 static GstAudioRingBuffer *
gst_jack_audio_src_create_ringbuffer(GstAudioBaseSrc * src)982 gst_jack_audio_src_create_ringbuffer (GstAudioBaseSrc * src)
983 {
984 GstAudioRingBuffer *buffer;
985
986 buffer = g_object_new (GST_TYPE_JACK_RING_BUFFER, NULL);
987 GST_DEBUG_OBJECT (src, "created ringbuffer @%p", buffer);
988
989 return buffer;
990 }
991