1 /*-*- Mode: C; c-basic-offset: 2 -*-*/
3 /* GStreamer pulseaudio plugin
5 * Copyright (c) 2004-2008 Lennart Poettering
8 * gst-pulse is free software; you can redistribute it and/or modify
9 * it under the terms of the GNU Lesser General Public License as
10 * published by the Free Software Foundation; either version 2.1 of the
11 * License, or (at your option) any later version.
13 * gst-pulse is distributed in the hope that it will be useful, but
14 * WITHOUT ANY WARRANTY; without even the implied warranty of
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16 * Lesser General Public License for more details.
18 * You should have received a copy of the GNU Lesser General Public
19 * License along with gst-pulse; if not, write to the Free Software
20 * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301
25 * SECTION:element-pulsesink
28 * This element outputs audio to a
29 * <ulink href="http://www.pulseaudio.org">PulseAudio sound server</ulink>.
32 * <title>Example pipelines</title>
34 * gst-launch-1.0 -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! pulsesink
35 * ]| Play an Ogg/Vorbis file.
37 * gst-launch-1.0 -v audiotestsrc ! audioconvert ! volume volume=0.4 ! pulsesink
38 * ]| Play a 440Hz sine wave.
40 * gst-launch-1.0 -v audiotestsrc ! pulsesink stream-properties="props,media.title=test"
41 * ]| Play a sine wave and set a stream property. The property can be checked
53 #include <gst/base/gstbasesink.h>
54 #include <gst/gsttaglist.h>
55 #include <gst/audio/audio.h>
56 #include <gst/gst-i18n-plugin.h>
58 #include <gst/pbutils/pbutils.h> /* only used for GST_PLUGINS_BASE_VERSION_* */
60 #include <gst/glib-compat-private.h>
61 #if defined(__TIZEN__) && defined(PCM_DUMP_ENABLE)
63 #endif /* __TIZEN__ && PCM_DUMP_ENABLE */
64 #include "pulsesink.h"
65 #include "pulseutil.h"
67 GST_DEBUG_CATEGORY_EXTERN (pulse_debug);
68 #define GST_CAT_DEFAULT pulse_debug
70 #define DEFAULT_SERVER NULL
71 #define DEFAULT_DEVICE NULL
72 #define DEFAULT_CURRENT_DEVICE NULL
73 #define DEFAULT_DEVICE_NAME NULL
74 #define DEFAULT_VOLUME 1.0
75 #define DEFAULT_MUTE FALSE
76 #define MAX_VOLUME 10.0
78 #define DEFAULT_AUDIO_LATENCY "mid"
79 #endif /* __TIZEN__ */
91 PROP_STREAM_PROPERTIES,
94 #endif /* __TIZEN__ */
98 #if defined(__TIZEN__) && defined(PCM_DUMP_ENABLE)
99 #define GST_PULSESINK_DUMP_VCONF_KEY "memory/private/sound/pcm_dump"
100 #define GST_PULSESINK_DUMP_INPUT_PATH_PREFIX "/tmp/dump_pulsesink_in_"
101 #define GST_PULSESINK_DUMP_OUTPUT_PATH_PREFIX "/tmp/dump_pulsesink_out_"
102 #define GST_PULSESINK_DUMP_INPUT_FLAG 0x00000400
103 #define GST_PULSESINK_DUMP_OUTPUT_FLAG 0x00000800
104 #endif /* __TIZEN__ && PCM_DUMP_ENABLE */
106 #define GST_TYPE_PULSERING_BUFFER \
107 (gst_pulseringbuffer_get_type())
108 #define GST_PULSERING_BUFFER(obj) \
109 (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_PULSERING_BUFFER,GstPulseRingBuffer))
110 #define GST_PULSERING_BUFFER_CLASS(klass) \
111 (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_PULSERING_BUFFER,GstPulseRingBufferClass))
112 #define GST_PULSERING_BUFFER_GET_CLASS(obj) \
113 (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_PULSERING_BUFFER, GstPulseRingBufferClass))
114 #define GST_PULSERING_BUFFER_CAST(obj) \
115 ((GstPulseRingBuffer *)obj)
116 #define GST_IS_PULSERING_BUFFER(obj) \
117 (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_PULSERING_BUFFER))
118 #define GST_IS_PULSERING_BUFFER_CLASS(klass)\
119 (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_PULSERING_BUFFER))
121 typedef struct _GstPulseRingBuffer GstPulseRingBuffer;
122 typedef struct _GstPulseRingBufferClass GstPulseRingBufferClass;
124 typedef struct _GstPulseContext GstPulseContext;
126 /* A note on threading.
128 * We use a pa_threaded_mainloop to interact with the PulseAudio server. This
129 * starts up a separate thread that runs a mainloop to carry back events,
130 * messages and timing updates from the PulseAudio server.
132 * In most cases, the PulseAudio API we use communicates with the server and
133 * processes replies asynchronously. Operations on PA objects that result in
134 * such communication are protected with a pa_threaded_mainloop_lock() and
135 * pa_threaded_mainloop_unlock(). These guarantee mutual exclusion with the
136 * mainloop thread -- when an iteration of the mainloop thread begins, it first
137 * tries to acquire this lock, and cannot do so if our code also holds that
140 * When we need to complete an operation synchronously, we use
141 * pa_threaded_mainloop_wait() and pa_threaded_mainloop_signal(). These work
142 * much as pthread conditionals do. pa_threaded_mainloop_wait() is called with
143 * the mainloop lock held. It releases the lock (thereby allowing the mainloop
144 * to execute), and waits till one of our callbacks to be executed by the
145 * mainloop thread calls pa_threaded_mainloop_signal(). At the end of the
146 * mainloop iteration, the pa_threaded_mainloop_wait() will reacquire the
147 * mainloop lock and return control to the caller.
150 /* Store the PA contexts in a hash table to allow easy sharing among
151 * multiple instances of the sink. Keys are $context_name@$server_name
152 * (strings) and values should be GstPulseContext pointers.
154 struct _GstPulseContext
157 GSList *ring_buffers;
160 static GHashTable *gst_pulse_shared_contexts = NULL;
162 /* use one static main-loop for all instances
163 * this is needed to make the context sharing work as the contexts are
164 * released when releasing their parent main-loop
166 static pa_threaded_mainloop *mainloop = NULL;
167 static guint mainloop_ref_ct = 0;
169 /* lock for access to shared resources */
170 static GMutex pa_shared_resource_mutex;
172 /* We keep a custom ringbuffer that is backed up by data allocated by
173 * pulseaudio. We must also overide the commit function to write into
174 * pulseaudio memory instead. */
175 struct _GstPulseRingBuffer
177 GstAudioRingBuffer object;
184 pa_stream *probe_stream;
186 pa_format_info *format;
197 gboolean in_commit:1;
200 struct _GstPulseRingBufferClass
202 GstAudioRingBufferClass parent_class;
205 static GType gst_pulseringbuffer_get_type (void);
206 static void gst_pulseringbuffer_finalize (GObject * object);
208 static GstAudioRingBufferClass *ring_parent_class = NULL;
210 static gboolean gst_pulseringbuffer_open_device (GstAudioRingBuffer * buf);
211 static gboolean gst_pulseringbuffer_close_device (GstAudioRingBuffer * buf);
212 static gboolean gst_pulseringbuffer_acquire (GstAudioRingBuffer * buf,
213 GstAudioRingBufferSpec * spec);
214 static gboolean gst_pulseringbuffer_release (GstAudioRingBuffer * buf);
215 static gboolean gst_pulseringbuffer_start (GstAudioRingBuffer * buf);
216 static gboolean gst_pulseringbuffer_pause (GstAudioRingBuffer * buf);
217 static gboolean gst_pulseringbuffer_stop (GstAudioRingBuffer * buf);
218 static void gst_pulseringbuffer_clear (GstAudioRingBuffer * buf);
219 static guint gst_pulseringbuffer_commit (GstAudioRingBuffer * buf,
220 guint64 * sample, guchar * data, gint in_samples, gint out_samples,
223 static gboolean gst_pulsering_set_corked (GstPulseRingBuffer * pbuf, gboolean corked,
227 G_DEFINE_TYPE (GstPulseRingBuffer, gst_pulseringbuffer,
228 GST_TYPE_AUDIO_RING_BUFFER);
231 gst_pulsesink_init_contexts (void)
233 g_mutex_init (&pa_shared_resource_mutex);
234 gst_pulse_shared_contexts = g_hash_table_new_full (g_str_hash, g_str_equal,
239 gst_pulseringbuffer_class_init (GstPulseRingBufferClass * klass)
241 GObjectClass *gobject_class;
242 GstAudioRingBufferClass *gstringbuffer_class;
244 gobject_class = (GObjectClass *) klass;
245 gstringbuffer_class = (GstAudioRingBufferClass *) klass;
247 ring_parent_class = g_type_class_peek_parent (klass);
249 gobject_class->finalize = gst_pulseringbuffer_finalize;
251 gstringbuffer_class->open_device =
252 GST_DEBUG_FUNCPTR (gst_pulseringbuffer_open_device);
253 gstringbuffer_class->close_device =
254 GST_DEBUG_FUNCPTR (gst_pulseringbuffer_close_device);
255 gstringbuffer_class->acquire =
256 GST_DEBUG_FUNCPTR (gst_pulseringbuffer_acquire);
257 gstringbuffer_class->release =
258 GST_DEBUG_FUNCPTR (gst_pulseringbuffer_release);
259 gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_pulseringbuffer_start);
260 gstringbuffer_class->pause = GST_DEBUG_FUNCPTR (gst_pulseringbuffer_pause);
261 gstringbuffer_class->resume = GST_DEBUG_FUNCPTR (gst_pulseringbuffer_start);
262 gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_pulseringbuffer_stop);
263 gstringbuffer_class->clear_all =
264 GST_DEBUG_FUNCPTR (gst_pulseringbuffer_clear);
266 gstringbuffer_class->commit = GST_DEBUG_FUNCPTR (gst_pulseringbuffer_commit);
270 gst_pulseringbuffer_init (GstPulseRingBuffer * pbuf)
272 pbuf->stream_name = NULL;
273 pbuf->context = NULL;
275 pbuf->probe_stream = NULL;
279 pbuf->is_pcm = FALSE;
283 pbuf->m_writable = 0;
285 pbuf->m_lastoffset = 0;
288 pbuf->in_commit = FALSE;
289 pbuf->paused = FALSE;
292 /* Call with mainloop lock held if wait == TRUE) */
294 gst_pulse_destroy_stream (pa_stream * stream, gboolean wait)
296 /* Make sure we don't get any further callbacks */
297 pa_stream_set_write_callback (stream, NULL, NULL);
298 pa_stream_set_underflow_callback (stream, NULL, NULL);
299 pa_stream_set_overflow_callback (stream, NULL, NULL);
301 pa_stream_disconnect (stream);
304 pa_threaded_mainloop_wait (mainloop);
306 pa_stream_set_state_callback (stream, NULL, NULL);
307 pa_stream_unref (stream);
311 gst_pulsering_destroy_stream (GstPulseRingBuffer * pbuf)
313 if (pbuf->probe_stream) {
314 gst_pulse_destroy_stream (pbuf->probe_stream, FALSE);
315 pbuf->probe_stream = NULL;
321 /* drop shm memory buffer */
322 pa_stream_cancel_write (pbuf->stream);
324 /* reset internal variables */
327 pbuf->m_writable = 0;
329 pbuf->m_lastoffset = 0;
332 pa_format_info_free (pbuf->format);
335 pbuf->is_pcm = FALSE;
338 pa_stream_disconnect (pbuf->stream);
340 /* Make sure we don't get any further callbacks */
341 pa_stream_set_state_callback (pbuf->stream, NULL, NULL);
342 pa_stream_set_write_callback (pbuf->stream, NULL, NULL);
343 pa_stream_set_underflow_callback (pbuf->stream, NULL, NULL);
344 pa_stream_set_overflow_callback (pbuf->stream, NULL, NULL);
346 pa_stream_unref (pbuf->stream);
350 g_free (pbuf->stream_name);
351 pbuf->stream_name = NULL;
355 gst_pulsering_destroy_context (GstPulseRingBuffer * pbuf)
357 g_mutex_lock (&pa_shared_resource_mutex);
359 GST_DEBUG_OBJECT (pbuf, "destroying ringbuffer %p", pbuf);
361 gst_pulsering_destroy_stream (pbuf);
364 pa_context_unref (pbuf->context);
365 pbuf->context = NULL;
368 if (pbuf->context_name) {
369 GstPulseContext *pctx;
371 pctx = g_hash_table_lookup (gst_pulse_shared_contexts, pbuf->context_name);
373 GST_DEBUG_OBJECT (pbuf, "releasing context with name %s, pbuf=%p, pctx=%p",
374 pbuf->context_name, pbuf, pctx);
377 pctx->ring_buffers = g_slist_remove (pctx->ring_buffers, pbuf);
378 if (pctx->ring_buffers == NULL) {
379 GST_DEBUG_OBJECT (pbuf,
380 "destroying final context with name %s, pbuf=%p, pctx=%p",
381 pbuf->context_name, pbuf, pctx);
383 pa_context_disconnect (pctx->context);
385 /* Make sure we don't get any further callbacks */
386 pa_context_set_state_callback (pctx->context, NULL, NULL);
387 pa_context_set_subscribe_callback (pctx->context, NULL, NULL);
389 g_hash_table_remove (gst_pulse_shared_contexts, pbuf->context_name);
391 pa_context_unref (pctx->context);
392 g_slice_free (GstPulseContext, pctx);
395 g_free (pbuf->context_name);
396 pbuf->context_name = NULL;
398 g_mutex_unlock (&pa_shared_resource_mutex);
402 gst_pulseringbuffer_finalize (GObject * object)
404 GstPulseRingBuffer *ringbuffer;
406 ringbuffer = GST_PULSERING_BUFFER_CAST (object);
408 gst_pulsering_destroy_context (ringbuffer);
409 G_OBJECT_CLASS (ring_parent_class)->finalize (object);
413 #define CONTEXT_OK(c) ((c) && PA_CONTEXT_IS_GOOD (pa_context_get_state ((c))))
414 #define STREAM_OK(s) ((s) && PA_STREAM_IS_GOOD (pa_stream_get_state ((s))))
417 gst_pulsering_is_dead (GstPulseSink * psink, GstPulseRingBuffer * pbuf,
418 gboolean check_stream)
420 if (!CONTEXT_OK (pbuf->context))
423 if (check_stream && !STREAM_OK (pbuf->stream))
430 const gchar *err_str =
431 pbuf->context ? pa_strerror (pa_context_errno (pbuf->context)) : NULL;
432 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED, ("Disconnected: %s",
439 gst_pulsering_context_state_cb (pa_context * c, void *userdata)
441 pa_context_state_t state;
442 pa_threaded_mainloop *mainloop = (pa_threaded_mainloop *) userdata;
444 state = pa_context_get_state (c);
446 GST_LOG ("got new context state %d", state);
449 case PA_CONTEXT_READY:
450 case PA_CONTEXT_TERMINATED:
451 case PA_CONTEXT_FAILED:
452 GST_LOG ("signaling");
453 pa_threaded_mainloop_signal (mainloop, 0);
456 case PA_CONTEXT_UNCONNECTED:
457 case PA_CONTEXT_CONNECTING:
458 case PA_CONTEXT_AUTHORIZING:
459 case PA_CONTEXT_SETTING_NAME:
465 gst_pulsering_context_subscribe_cb (pa_context * c,
466 pa_subscription_event_type_t t, uint32_t idx, void *userdata)
469 GstPulseContext *pctx = (GstPulseContext *) userdata;
472 if (t != (PA_SUBSCRIPTION_EVENT_SINK_INPUT | PA_SUBSCRIPTION_EVENT_CHANGE) &&
473 t != (PA_SUBSCRIPTION_EVENT_SINK_INPUT | PA_SUBSCRIPTION_EVENT_NEW))
476 for (walk = pctx->ring_buffers; walk; walk = g_slist_next (walk)) {
477 GstPulseRingBuffer *pbuf = (GstPulseRingBuffer *) walk->data;
478 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
480 GST_LOG_OBJECT (psink, "type %04x, idx %u", t, idx);
485 if (idx != pa_stream_get_index (pbuf->stream))
488 if (psink->device && pbuf->is_pcm &&
489 !g_str_equal (psink->device,
490 pa_stream_get_device_name (pbuf->stream))) {
491 /* Underlying sink changed. And this is not a passthrough stream. Let's
492 * see if someone upstream wants to try to renegotiate. */
495 g_free (psink->device);
496 psink->device = g_strdup (pa_stream_get_device_name (pbuf->stream));
498 GST_INFO_OBJECT (psink, "emitting sink-changed");
500 /* FIXME: send reconfigure event instead and let decodebin/playbin
501 * handle that. Also take care of ac3 alignment. See "pulse-format-lost" */
502 renego = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
503 gst_structure_new_empty ("pulse-sink-changed"));
505 if (!gst_pad_push_event (GST_BASE_SINK (psink)->sinkpad, renego))
506 GST_DEBUG_OBJECT (psink, "Emitted sink-changed - nobody was listening");
509 /* Actually this event is also triggered when other properties of
510 * the stream change that are unrelated to the volume. However it is
511 * probably cheaper to signal the change here and check for the
512 * volume when the GObject property is read instead of querying it always. */
514 /* inform streaming thread to notify */
515 g_atomic_int_compare_and_exchange (&psink->notify, 0, 1);
519 /* will be called when the device should be opened. In this case we will connect
520 * to the server. We should not try to open any streams in this state. */
522 gst_pulseringbuffer_open_device (GstAudioRingBuffer * buf)
525 GstPulseRingBuffer *pbuf;
526 GstPulseContext *pctx;
527 pa_mainloop_api *api;
528 gboolean need_unlock_shared;
530 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (buf));
531 pbuf = GST_PULSERING_BUFFER_CAST (buf);
533 g_assert (!pbuf->stream);
534 g_assert (psink->client_name);
537 pbuf->context_name = g_strdup_printf ("%s@%s", psink->client_name,
540 pbuf->context_name = g_strdup (psink->client_name);
542 pa_threaded_mainloop_lock (mainloop);
544 g_mutex_lock (&pa_shared_resource_mutex);
545 need_unlock_shared = TRUE;
547 pctx = g_hash_table_lookup (gst_pulse_shared_contexts, pbuf->context_name);
549 pctx = g_slice_new0 (GstPulseContext);
551 /* get the mainloop api and create a context */
552 GST_INFO_OBJECT (psink, "new context with name %s, pbuf=%p, pctx=%p",
553 pbuf->context_name, pbuf, pctx);
554 api = pa_threaded_mainloop_get_api (mainloop);
555 if (!(pctx->context = pa_context_new (api, pbuf->context_name)))
558 pctx->ring_buffers = g_slist_prepend (pctx->ring_buffers, pbuf);
559 g_hash_table_insert (gst_pulse_shared_contexts,
560 g_strdup (pbuf->context_name), (gpointer) pctx);
561 /* register some essential callbacks */
562 pa_context_set_state_callback (pctx->context,
563 gst_pulsering_context_state_cb, mainloop);
564 pa_context_set_subscribe_callback (pctx->context,
565 gst_pulsering_context_subscribe_cb, pctx);
567 /* try to connect to the server and wait for completion, we don't want to
568 * autospawn a deamon */
569 GST_LOG_OBJECT (psink, "connect to server %s",
570 GST_STR_NULL (psink->server));
571 if (pa_context_connect (pctx->context, psink->server,
572 PA_CONTEXT_NOAUTOSPAWN, NULL) < 0)
575 GST_INFO_OBJECT (psink,
576 "reusing shared context with name %s, pbuf=%p, pctx=%p",
577 pbuf->context_name, pbuf, pctx);
578 pctx->ring_buffers = g_slist_prepend (pctx->ring_buffers, pbuf);
581 g_mutex_unlock (&pa_shared_resource_mutex);
582 need_unlock_shared = FALSE;
584 /* context created or shared okay */
585 pbuf->context = pa_context_ref (pctx->context);
588 pa_context_state_t state;
590 state = pa_context_get_state (pbuf->context);
592 GST_LOG_OBJECT (psink, "context state is now %d", state);
594 if (!PA_CONTEXT_IS_GOOD (state))
597 if (state == PA_CONTEXT_READY)
600 /* Wait until the context is ready */
601 GST_LOG_OBJECT (psink, "waiting..");
602 pa_threaded_mainloop_wait (mainloop);
605 if (pa_context_get_server_protocol_version (pbuf->context) < 22) {
606 /* We need PulseAudio >= 1.0 on the server side for the extended API */
607 goto bad_server_version;
610 GST_LOG_OBJECT (psink, "opened the device");
612 pa_threaded_mainloop_unlock (mainloop);
619 if (need_unlock_shared)
620 g_mutex_unlock (&pa_shared_resource_mutex);
621 gst_pulsering_destroy_context (pbuf);
622 pa_threaded_mainloop_unlock (mainloop);
627 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
628 ("Failed to create context"), (NULL));
629 g_slice_free (GstPulseContext, pctx);
630 goto unlock_and_fail;
634 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED, ("Failed to connect: %s",
635 pa_strerror (pa_context_errno (pctx->context))), (NULL));
636 goto unlock_and_fail;
640 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED, ("PulseAudio server version "
641 "is too old."), (NULL));
642 goto unlock_and_fail;
646 /* close the device */
648 gst_pulseringbuffer_close_device (GstAudioRingBuffer * buf)
651 GstPulseRingBuffer *pbuf;
653 pbuf = GST_PULSERING_BUFFER_CAST (buf);
654 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (buf));
656 GST_LOG_OBJECT (psink, "closing device");
658 pa_threaded_mainloop_lock (mainloop);
659 gst_pulsering_destroy_context (pbuf);
660 pa_threaded_mainloop_unlock (mainloop);
662 #if defined(__TIZEN__) && defined(PCM_DUMP_ENABLE)
663 if (psink->dump_fd_input) {
664 fclose(psink->dump_fd_input);
665 psink->dump_fd_input = NULL;
667 #endif /* __TIZEN__ && PCM_DUMP_ENABLE */
669 GST_LOG_OBJECT (psink, "closed device");
675 gst_pulsering_stream_state_cb (pa_stream * s, void *userdata)
678 GstPulseRingBuffer *pbuf;
679 pa_stream_state_t state;
681 pbuf = GST_PULSERING_BUFFER_CAST (userdata);
682 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
684 state = pa_stream_get_state (s);
685 GST_LOG_OBJECT (psink, "got new stream state %d", state);
688 case PA_STREAM_READY:
689 case PA_STREAM_FAILED:
690 case PA_STREAM_TERMINATED:
691 GST_LOG_OBJECT (psink, "signaling");
692 pa_threaded_mainloop_signal (mainloop, 0);
694 case PA_STREAM_UNCONNECTED:
695 case PA_STREAM_CREATING:
701 gst_pulsering_stream_request_cb (pa_stream * s, size_t length, void *userdata)
704 GstAudioRingBuffer *rbuf;
705 GstPulseRingBuffer *pbuf;
707 rbuf = GST_AUDIO_RING_BUFFER_CAST (userdata);
708 pbuf = GST_PULSERING_BUFFER_CAST (userdata);
709 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
711 GST_LOG_OBJECT (psink, "got request for length %" G_GSIZE_FORMAT, length);
713 if (pbuf->in_commit && (length >= rbuf->spec.segsize)) {
714 /* only signal when we are waiting in the commit thread
715 * and got request for atleast a segment */
716 pa_threaded_mainloop_signal (mainloop, 0);
721 gst_pulsering_stream_underflow_cb (pa_stream * s, void *userdata)
724 GstPulseRingBuffer *pbuf;
726 pbuf = GST_PULSERING_BUFFER_CAST (userdata);
727 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
729 GST_WARNING_OBJECT (psink, "Got underflow");
733 gst_pulsering_stream_overflow_cb (pa_stream * s, void *userdata)
736 GstPulseRingBuffer *pbuf;
738 pbuf = GST_PULSERING_BUFFER_CAST (userdata);
739 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
741 GST_WARNING_OBJECT (psink, "Got overflow");
745 gst_pulsering_stream_latency_cb (pa_stream * s, void *userdata)
748 GstPulseRingBuffer *pbuf;
749 GstAudioRingBuffer *ringbuf;
750 const pa_timing_info *info;
753 info = pa_stream_get_timing_info (s);
755 pbuf = GST_PULSERING_BUFFER_CAST (userdata);
756 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
757 ringbuf = GST_AUDIO_RING_BUFFER (pbuf);
760 GST_LOG_OBJECT (psink, "latency update (information unknown)");
764 if (!info->read_index_corrupt) {
765 /* Update segdone based on the read index. segdone is of segment
766 * granularity, while the read index is at byte granularity. We take the
767 * ceiling while converting the latter to the former since it is more
768 * conservative to report that we've read more than we have than to report
769 * less. One concern here is that latency updates happen every 100ms, which
770 * means segdone is not updated very often, but increasing the update
771 * frequency would mean more communication overhead. */
772 g_atomic_int_set (&ringbuf->segdone,
773 (int) gst_util_uint64_scale_ceil (info->read_index, 1,
774 ringbuf->spec.segsize));
777 sink_usec = info->configured_sink_usec;
779 GST_LOG_OBJECT (psink,
780 "latency_update, %" G_GUINT64_FORMAT ", %d:%" G_GINT64_FORMAT ", %d:%"
781 G_GUINT64_FORMAT ", %" G_GUINT64_FORMAT ", %" G_GUINT64_FORMAT,
782 GST_TIMEVAL_TO_TIME (info->timestamp), info->write_index_corrupt,
783 info->write_index, info->read_index_corrupt, info->read_index,
784 info->sink_usec, sink_usec);
788 gst_pulsering_stream_suspended_cb (pa_stream * p, void *userdata)
791 GstPulseRingBuffer *pbuf;
793 pbuf = GST_PULSERING_BUFFER_CAST (userdata);
794 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
796 if (pa_stream_is_suspended (p))
797 GST_DEBUG_OBJECT (psink, "stream suspended");
799 GST_DEBUG_OBJECT (psink, "stream resumed");
803 gst_pulsering_stream_started_cb (pa_stream * p, void *userdata)
806 GstPulseRingBuffer *pbuf;
808 pbuf = GST_PULSERING_BUFFER_CAST (userdata);
809 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
811 GST_DEBUG_OBJECT (psink, "stream started");
815 gst_pulsering_stream_event_cb (pa_stream * p, const char *name,
816 pa_proplist * pl, void *userdata)
819 GstPulseRingBuffer *pbuf;
821 pbuf = GST_PULSERING_BUFFER_CAST (userdata);
822 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
824 if (!strcmp (name, PA_STREAM_EVENT_REQUEST_CORK)) {
825 /* the stream wants to PAUSE, post a message for the application. */
826 GST_DEBUG_OBJECT (psink, "got request for CORK");
827 gst_element_post_message (GST_ELEMENT_CAST (psink),
828 gst_message_new_request_state (GST_OBJECT_CAST (psink),
831 } else if (!strcmp (name, PA_STREAM_EVENT_REQUEST_UNCORK)) {
832 GST_DEBUG_OBJECT (psink, "got request for UNCORK");
833 gst_element_post_message (GST_ELEMENT_CAST (psink),
834 gst_message_new_request_state (GST_OBJECT_CAST (psink),
836 } else if (!strcmp (name, PA_STREAM_EVENT_FORMAT_LOST)) {
839 if (g_atomic_int_get (&psink->format_lost)) {
840 /* Duplicate event before we're done reconfiguring, discard */
844 GST_DEBUG_OBJECT (psink, "got FORMAT LOST");
845 g_atomic_int_set (&psink->format_lost, 1);
846 psink->format_lost_time = g_ascii_strtoull (pa_proplist_gets (pl,
847 "stream-time"), NULL, 0) * 1000;
849 g_free (psink->device);
850 psink->device = g_strdup (pa_proplist_gets (pl, "device"));
852 /* FIXME: send reconfigure event instead and let decodebin/playbin
853 * handle that. Also take care of ac3 alignment */
854 renego = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
855 gst_structure_new_empty ("pulse-format-lost"));
858 if (g_str_equal (gst_structure_get_name (st), "audio/x-eac3")) {
859 GstStructure *event_st = gst_structure_new ("ac3parse-set-alignment",
860 "alignment", G_TYPE_STRING, pbin->dbin ? "frame" : "iec61937", NULL);
862 if (!gst_pad_push_event (pbin->sinkpad,
863 gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, event_st)))
864 GST_WARNING_OBJECT (pbin->sinkpad, "Could not update alignment");
868 if (!gst_pad_push_event (GST_BASE_SINK (psink)->sinkpad, renego)) {
869 /* Nobody handled the format change - emit an error */
870 GST_ELEMENT_ERROR (psink, STREAM, FORMAT, ("Sink format changed"),
871 ("Sink format changed"));
874 } else if (!strcmp (name, PA_STREAM_EVENT_POP_TIMEOUT)) {
875 GST_WARNING_OBJECT (psink, "got event [%s], cork stream now!!!!", name);
876 gst_pulsering_set_corked (pbuf, TRUE, FALSE);
879 GST_DEBUG_OBJECT (psink, "got unknown event %s", name);
883 /* Called with the mainloop locked */
885 gst_pulsering_wait_for_stream_ready (GstPulseSink * psink, pa_stream * stream)
887 pa_stream_state_t state;
890 state = pa_stream_get_state (stream);
892 GST_LOG_OBJECT (psink, "stream state is now %d", state);
894 if (!PA_STREAM_IS_GOOD (state))
897 if (state == PA_STREAM_READY)
900 /* Wait until the stream is ready */
901 pa_threaded_mainloop_wait (mainloop);
906 /* This method should create a new stream of the given @spec. No playback should
907 * start yet so we start in the corked state. */
909 gst_pulseringbuffer_acquire (GstAudioRingBuffer * buf,
910 GstAudioRingBufferSpec * spec)
913 GstPulseRingBuffer *pbuf;
914 pa_buffer_attr wanted;
915 const pa_buffer_attr *actual;
916 pa_channel_map channel_map;
917 pa_operation *o = NULL;
921 pa_cvolume *pv = NULL;
922 pa_stream_flags_t flags;
924 GstAudioClock *clock;
925 pa_format_info *formats[1];
926 #ifndef GST_DISABLE_GST_DEBUG
927 gchar print_buf[PA_FORMAT_INFO_SNPRINT_MAX];
930 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (buf));
931 pbuf = GST_PULSERING_BUFFER_CAST (buf);
933 GST_LOG_OBJECT (psink, "creating sample spec");
934 /* convert the gstreamer sample spec to the pulseaudio format */
935 if (!gst_pulse_fill_format_info (spec, &pbuf->format, &pbuf->channels))
937 pbuf->is_pcm = pa_format_info_is_pcm (pbuf->format);
939 pa_threaded_mainloop_lock (mainloop);
941 /* we need a context and a no stream */
942 g_assert (pbuf->context);
943 g_assert (!pbuf->stream);
945 /* if we have a probe, disconnect it first so that if we're creating a
946 * compressed stream, it doesn't get blocked by a PCM stream */
947 if (pbuf->probe_stream) {
948 gst_pulse_destroy_stream (pbuf->probe_stream, TRUE);
949 pbuf->probe_stream = NULL;
952 /* enable event notifications */
953 GST_LOG_OBJECT (psink, "subscribing to context events");
954 if (!(o = pa_context_subscribe (pbuf->context,
955 PA_SUBSCRIPTION_MASK_SINK_INPUT, NULL, NULL)))
956 goto subscribe_failed;
958 pa_operation_unref (o);
960 /* initialize the channel map */
961 if (pbuf->is_pcm && gst_pulse_gst_to_channel_map (&channel_map, spec))
962 pa_format_info_set_channel_map (pbuf->format, &channel_map);
964 /* find a good name for the stream */
965 if (psink->stream_name)
966 name = psink->stream_name;
968 name = "Playback Stream";
970 #if defined(__TIZEN__) && defined(PCM_DUMP_ENABLE)
971 if (psink->need_dump_input == TRUE && psink->dump_fd_input == NULL) {
972 char *suffix , *dump_path;
973 GDateTime *time = g_date_time_new_now_local();
975 suffix = g_date_time_format(time, "%m%d_%H%M%S");
976 dump_path = g_strdup_printf("%s%dch_%dhz_%s.pcm", GST_PULSESINK_DUMP_INPUT_PATH_PREFIX, pbuf->channels, spec->info.rate, suffix);
977 GST_WARNING_OBJECT(psink, "pulse-sink dumping enabled: dump path [%s]", dump_path);
978 psink->dump_fd_input = fopen(dump_path, "w+");
982 g_date_time_unref(time);
984 #endif /* __TIZEN__ && PCM_DUMP_ENABLE */
986 /* create a stream */
987 formats[0] = pbuf->format;
988 if (!(pbuf->stream = pa_stream_new_extended (pbuf->context, name, formats, 1,
992 /* install essential callbacks */
993 pa_stream_set_state_callback (pbuf->stream,
994 gst_pulsering_stream_state_cb, pbuf);
995 pa_stream_set_write_callback (pbuf->stream,
996 gst_pulsering_stream_request_cb, pbuf);
997 pa_stream_set_underflow_callback (pbuf->stream,
998 gst_pulsering_stream_underflow_cb, pbuf);
999 pa_stream_set_overflow_callback (pbuf->stream,
1000 gst_pulsering_stream_overflow_cb, pbuf);
1001 pa_stream_set_latency_update_callback (pbuf->stream,
1002 gst_pulsering_stream_latency_cb, pbuf);
1003 pa_stream_set_suspended_callback (pbuf->stream,
1004 gst_pulsering_stream_suspended_cb, pbuf);
1005 pa_stream_set_started_callback (pbuf->stream,
1006 gst_pulsering_stream_started_cb, pbuf);
1007 pa_stream_set_event_callback (pbuf->stream,
1008 gst_pulsering_stream_event_cb, pbuf);
1010 /* buffering requirements. When setting prebuf to 0, the stream will not pause
1011 * when we cause an underrun, which causes time to continue. */
1012 memset (&wanted, 0, sizeof (wanted));
1013 wanted.tlength = spec->segtotal * spec->segsize;
1014 wanted.maxlength = -1;
1016 wanted.minreq = spec->segsize;
1018 GST_INFO_OBJECT (psink, "tlength: %d", wanted.tlength);
1019 GST_INFO_OBJECT (psink, "maxlength: %d", wanted.maxlength);
1020 GST_INFO_OBJECT (psink, "prebuf: %d", wanted.prebuf);
1021 GST_INFO_OBJECT (psink, "minreq: %d", wanted.minreq);
1024 /* configure volume when we changed it, else we leave the default */
1025 if (psink->volume_set) {
1026 GST_LOG_OBJECT (psink, "have volume of %f", psink->volume);
1029 gst_pulse_cvolume_from_linear (pv, pbuf->channels, psink->volume);
1031 GST_DEBUG_OBJECT (psink, "passthrough stream, not setting volume");
1039 /* construct the flags */
1040 flags = PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE |
1041 PA_STREAM_ADJUST_LATENCY | PA_STREAM_START_CORKED;
1044 if (psink->mute_set) {
1046 flags |= PA_STREAM_START_MUTED;
1048 flags |= PA_STREAM_START_UNMUTED;
1052 /* we always start corked (see flags above) */
1053 pbuf->corked = TRUE;
1055 /* try to connect now */
1056 GST_LOG_OBJECT (psink, "connect for playback to device %s",
1057 GST_STR_NULL (psink->device));
1058 if (pa_stream_connect_playback (pbuf->stream, psink->device,
1059 &wanted, flags, pv, NULL) < 0)
1060 goto connect_failed;
1062 /* our clock will now start from 0 again */
1063 clock = GST_AUDIO_CLOCK (GST_AUDIO_BASE_SINK (psink)->provided_clock);
1064 gst_audio_clock_reset (clock, 0);
1066 if (!gst_pulsering_wait_for_stream_ready (psink, pbuf->stream))
1067 goto connect_failed;
1069 g_free (psink->device);
1070 psink->device = g_strdup (pa_stream_get_device_name (pbuf->stream));
1072 #ifndef GST_DISABLE_GST_DEBUG
1073 pa_format_info_snprint (print_buf, sizeof (print_buf),
1074 pa_stream_get_format_info (pbuf->stream));
1075 GST_INFO_OBJECT (psink, "negotiated to: %s", print_buf);
1081 if ((idx = pa_stream_get_index (pbuf->stream)) == PA_INVALID_INDEX)
1083 if (psink->volume_set)
1084 gst_pulse_set_volume_ratio (idx, "out", psink->volume);
1085 if (psink->mute_set)
1087 gst_pulse_set_volume_ratio (idx, "out", 0);
1090 /* After we passed the volume off of to PA we never want to set it
1091 again, since it is PA's job to save/restore volumes. */
1092 psink->volume_set = psink->mute_set = FALSE;
1094 GST_LOG_OBJECT (psink, "stream is acquired now");
1096 /* get the actual buffering properties now */
1097 actual = pa_stream_get_buffer_attr (pbuf->stream);
1099 GST_INFO_OBJECT (psink, "tlength: %d (wanted: %d)", actual->tlength,
1101 GST_INFO_OBJECT (psink, "maxlength: %d", actual->maxlength);
1102 GST_INFO_OBJECT (psink, "prebuf: %d", actual->prebuf);
1103 GST_INFO_OBJECT (psink, "minreq: %d (wanted %d)", actual->minreq,
1106 spec->segsize = actual->minreq;
1107 spec->segtotal = actual->tlength / spec->segsize;
1109 pa_threaded_mainloop_unlock (mainloop);
1116 gst_pulsering_destroy_stream (pbuf);
1117 pa_threaded_mainloop_unlock (mainloop);
1123 GST_ELEMENT_ERROR (psink, RESOURCE, SETTINGS,
1124 ("Invalid sample specification."), (NULL));
1129 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
1130 ("pa_context_subscribe() failed: %s",
1131 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
1132 goto unlock_and_fail;
1136 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
1137 ("Failed to create stream: %s",
1138 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
1139 goto unlock_and_fail;
1143 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
1144 ("Failed to connect stream: %s",
1145 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
1146 goto unlock_and_fail;
1151 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
1152 ("Failed to get stream index: %s",
1153 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
1154 goto unlock_and_fail;
1159 /* free the stream that we acquired before */
1161 gst_pulseringbuffer_release (GstAudioRingBuffer * buf)
1163 GstPulseRingBuffer *pbuf;
1165 pbuf = GST_PULSERING_BUFFER_CAST (buf);
1167 pa_threaded_mainloop_lock (mainloop);
1168 gst_pulsering_destroy_stream (pbuf);
1169 pa_threaded_mainloop_unlock (mainloop);
1172 GstPulseSink *psink;
1174 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
1175 g_atomic_int_set (&psink->format_lost, FALSE);
1176 psink->format_lost_time = GST_CLOCK_TIME_NONE;
1183 gst_pulsering_success_cb (pa_stream * s, int success, void *userdata)
1185 pa_threaded_mainloop_signal (mainloop, 0);
1188 /* update the corked state of a stream, must be called with the mainloop
1191 gst_pulsering_set_corked (GstPulseRingBuffer * pbuf, gboolean corked,
1194 pa_operation *o = NULL;
1195 GstPulseSink *psink;
1196 gboolean res = FALSE;
1198 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
1200 if (g_atomic_int_get (&psink->format_lost)) {
1201 /* Sink format changed, stream's gone so fake being paused */
1205 GST_DEBUG_OBJECT (psink, "setting corked state to %d", corked);
1206 if (pbuf->corked != corked) {
1207 if (!(o = pa_stream_cork (pbuf->stream, corked,
1208 gst_pulsering_success_cb, pbuf)))
1211 while (wait && pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
1212 pa_threaded_mainloop_wait (mainloop);
1213 if (gst_pulsering_is_dead (psink, pbuf, TRUE))
1216 pbuf->corked = corked;
1218 GST_DEBUG_OBJECT (psink, "skipping, already in requested state");
1224 pa_operation_unref (o);
1231 GST_DEBUG_OBJECT (psink, "the server is dead");
1236 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
1237 ("pa_stream_cork() failed: %s",
1238 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
1244 gst_pulseringbuffer_clear (GstAudioRingBuffer * buf)
1246 GstPulseSink *psink;
1247 GstPulseRingBuffer *pbuf;
1248 pa_operation *o = NULL;
1250 pbuf = GST_PULSERING_BUFFER_CAST (buf);
1251 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
1253 pa_threaded_mainloop_lock (mainloop);
1254 GST_DEBUG_OBJECT (psink, "clearing");
1256 /* don't wait for the flush to complete */
1257 if ((o = pa_stream_flush (pbuf->stream, NULL, pbuf)))
1258 pa_operation_unref (o);
1260 pa_threaded_mainloop_unlock (mainloop);
1264 /* called from pulse thread with the mainloop lock */
1266 mainloop_enter_defer_cb (pa_mainloop_api * api, void *userdata)
1268 GstPulseSink *pulsesink = GST_PULSESINK (userdata);
1269 GstMessage *message;
1272 GST_DEBUG_OBJECT (pulsesink, "posting ENTER stream status");
1273 message = gst_message_new_stream_status (GST_OBJECT (pulsesink),
1274 GST_STREAM_STATUS_TYPE_ENTER, GST_ELEMENT (pulsesink));
1275 g_value_init (&val, GST_TYPE_G_THREAD);
1276 g_value_set_boxed (&val, g_thread_self ());
1277 gst_message_set_stream_status_object (message, &val);
1278 g_value_unset (&val);
1280 gst_element_post_message (GST_ELEMENT (pulsesink), message);
1282 g_return_if_fail (pulsesink->defer_pending);
1283 pulsesink->defer_pending--;
1284 pa_threaded_mainloop_signal (mainloop, 0);
1288 /* start/resume playback ASAP, we don't uncork here but in the commit method */
1290 gst_pulseringbuffer_start (GstAudioRingBuffer * buf)
1292 GstPulseSink *psink;
1293 GstPulseRingBuffer *pbuf;
1295 pbuf = GST_PULSERING_BUFFER_CAST (buf);
1296 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
1298 pa_threaded_mainloop_lock (mainloop);
1300 GST_DEBUG_OBJECT (psink, "starting");
1301 pbuf->paused = FALSE;
1303 /* EOS needs running clock */
1304 if (GST_BASE_SINK_CAST (psink)->eos ||
1305 g_atomic_int_get (&GST_AUDIO_BASE_SINK (psink)->eos_rendering))
1306 gst_pulsering_set_corked (pbuf, FALSE, FALSE);
1309 GST_DEBUG_OBJECT (psink, "scheduling stream status");
1310 psink->defer_pending++;
1311 pa_mainloop_api_once (pa_threaded_mainloop_get_api (mainloop),
1312 mainloop_enter_defer_cb, psink);
1314 /* Wait for the stream status message to be posted. This needs to be done
1315 * synchronously because the callback will take the mainloop lock
1316 * (implicitly) and then take the GST_OBJECT_LOCK. Everywhere else, we take
1317 * the locks in the reverse order, so not doing this synchronously could
1318 * cause a deadlock. */
1319 GST_DEBUG_OBJECT (psink, "waiting for stream status (ENTER) to be posted");
1320 pa_threaded_mainloop_wait (mainloop);
1323 pa_threaded_mainloop_unlock (mainloop);
1328 /* pause/stop playback ASAP */
1330 gst_pulseringbuffer_pause (GstAudioRingBuffer * buf)
1332 GstPulseSink *psink;
1333 GstPulseRingBuffer *pbuf;
1336 pbuf = GST_PULSERING_BUFFER_CAST (buf);
1337 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
1339 pa_threaded_mainloop_lock (mainloop);
1340 GST_DEBUG_OBJECT (psink, "pausing and corking");
1341 /* make sure the commit method stops writing */
1342 pbuf->paused = TRUE;
1343 res = gst_pulsering_set_corked (pbuf, TRUE, TRUE);
1344 if (pbuf->in_commit) {
1345 /* we are waiting in a commit, signal */
1346 GST_DEBUG_OBJECT (psink, "signal commit");
1347 pa_threaded_mainloop_signal (mainloop, 0);
1349 pa_threaded_mainloop_unlock (mainloop);
1355 /* called from pulse thread with the mainloop lock */
1357 mainloop_leave_defer_cb (pa_mainloop_api * api, void *userdata)
1359 GstPulseSink *pulsesink = GST_PULSESINK (userdata);
1360 GstMessage *message;
1363 GST_DEBUG_OBJECT (pulsesink, "posting LEAVE stream status");
1364 message = gst_message_new_stream_status (GST_OBJECT (pulsesink),
1365 GST_STREAM_STATUS_TYPE_LEAVE, GST_ELEMENT (pulsesink));
1366 g_value_init (&val, GST_TYPE_G_THREAD);
1367 g_value_set_boxed (&val, g_thread_self ());
1368 gst_message_set_stream_status_object (message, &val);
1369 g_value_unset (&val);
1371 gst_element_post_message (GST_ELEMENT (pulsesink), message);
1373 g_return_if_fail (pulsesink->defer_pending);
1374 pulsesink->defer_pending--;
1375 pa_threaded_mainloop_signal (mainloop, 0);
1379 /* stop playback, we flush everything. */
1381 gst_pulseringbuffer_stop (GstAudioRingBuffer * buf)
1383 GstPulseSink *psink;
1384 GstPulseRingBuffer *pbuf;
1385 gboolean res = FALSE;
1386 pa_operation *o = NULL;
1388 pbuf = GST_PULSERING_BUFFER_CAST (buf);
1389 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
1391 pa_threaded_mainloop_lock (mainloop);
1393 pbuf->paused = TRUE;
1394 res = gst_pulsering_set_corked (pbuf, TRUE, TRUE);
1396 /* Inform anyone waiting in _commit() call that it shall wakeup */
1397 if (pbuf->in_commit) {
1398 GST_DEBUG_OBJECT (psink, "signal commit thread");
1399 pa_threaded_mainloop_signal (mainloop, 0);
1401 if (g_atomic_int_get (&psink->format_lost)) {
1402 /* Don't try to flush, the stream's probably gone by now */
1407 /* then try to flush, it's not fatal when this fails */
1408 GST_DEBUG_OBJECT (psink, "flushing");
1409 if ((o = pa_stream_flush (pbuf->stream, gst_pulsering_success_cb, pbuf))) {
1410 while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
1411 GST_DEBUG_OBJECT (psink, "wait for completion");
1412 pa_threaded_mainloop_wait (mainloop);
1413 if (gst_pulsering_is_dead (psink, pbuf, TRUE))
1416 GST_DEBUG_OBJECT (psink, "flush completed");
1422 pa_operation_cancel (o);
1423 pa_operation_unref (o);
1426 GST_DEBUG_OBJECT (psink, "scheduling stream status");
1427 psink->defer_pending++;
1428 pa_mainloop_api_once (pa_threaded_mainloop_get_api (mainloop),
1429 mainloop_leave_defer_cb, psink);
1431 /* Wait for the stream status message to be posted. This needs to be done
1432 * synchronously because the callback will take the mainloop lock
1433 * (implicitly) and then take the GST_OBJECT_LOCK. Everywhere else, we take
1434 * the locks in the reverse order, so not doing this synchronously could
1435 * cause a deadlock. */
1436 GST_DEBUG_OBJECT (psink, "waiting for stream status (LEAVE) to be posted");
1437 pa_threaded_mainloop_wait (mainloop);
1440 pa_threaded_mainloop_unlock (mainloop);
1447 GST_DEBUG_OBJECT (psink, "the server is dead");
1452 /* in_samples >= out_samples, rate > 1.0 */
1453 #define FWD_UP_SAMPLES(s,se,d,de) \
1455 guint8 *sb = s, *db = d; \
1456 while (s <= se && d < de) { \
1457 memcpy (d, s, bpf); \
1460 if ((*accum << 1) >= inr) { \
1465 in_samples -= (s - sb)/bpf; \
1466 out_samples -= (d - db)/bpf; \
1467 GST_DEBUG ("fwd_up end %d/%d",*accum,*toprocess); \
1470 /* out_samples > in_samples, for rates smaller than 1.0 */
1471 #define FWD_DOWN_SAMPLES(s,se,d,de) \
1473 guint8 *sb = s, *db = d; \
1474 while (s <= se && d < de) { \
1475 memcpy (d, s, bpf); \
1478 if ((*accum << 1) >= outr) { \
1483 in_samples -= (s - sb)/bpf; \
1484 out_samples -= (d - db)/bpf; \
1485 GST_DEBUG ("fwd_down end %d/%d",*accum,*toprocess); \
1488 #define REV_UP_SAMPLES(s,se,d,de) \
1490 guint8 *sb = se, *db = d; \
1491 while (s <= se && d < de) { \
1492 memcpy (d, se, bpf); \
1495 while (d < de && (*accum << 1) >= inr) { \
1500 in_samples -= (sb - se)/bpf; \
1501 out_samples -= (d - db)/bpf; \
1502 GST_DEBUG ("rev_up end %d/%d",*accum,*toprocess); \
1505 #define REV_DOWN_SAMPLES(s,se,d,de) \
1507 guint8 *sb = se, *db = d; \
1508 while (s <= se && d < de) { \
1509 memcpy (d, se, bpf); \
1512 while (s <= se && (*accum << 1) >= outr) { \
1517 in_samples -= (sb - se)/bpf; \
1518 out_samples -= (d - db)/bpf; \
1519 GST_DEBUG ("rev_down end %d/%d",*accum,*toprocess); \
1522 /* our custom commit function because we write into the buffer of pulseaudio
1523 * instead of keeping our own buffer */
1525 gst_pulseringbuffer_commit (GstAudioRingBuffer * buf, guint64 * sample,
1526 guchar * data, gint in_samples, gint out_samples, gint * accum)
1528 GstPulseSink *psink;
1529 GstPulseRingBuffer *pbuf;
1534 gint inr, outr, bpf;
1538 pbuf = GST_PULSERING_BUFFER_CAST (buf);
1539 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
1541 /* FIXME post message rather than using a signal (as mixer interface) */
1542 if (g_atomic_int_compare_and_exchange (&psink->notify, 1, 0)) {
1543 g_object_notify (G_OBJECT (psink), "volume");
1544 g_object_notify (G_OBJECT (psink), "mute");
1545 g_object_notify (G_OBJECT (psink), "current-device");
1548 /* make sure the ringbuffer is started */
1549 if (G_UNLIKELY (g_atomic_int_get (&buf->state) !=
1550 GST_AUDIO_RING_BUFFER_STATE_STARTED)) {
1551 /* see if we are allowed to start it */
1552 if (G_UNLIKELY (g_atomic_int_get (&buf->may_start) == FALSE))
1555 GST_DEBUG_OBJECT (buf, "start!");
1556 if (!gst_audio_ring_buffer_start (buf))
1560 pa_threaded_mainloop_lock (mainloop);
1562 GST_DEBUG_OBJECT (psink, "entering commit");
1563 pbuf->in_commit = TRUE;
1565 bpf = GST_AUDIO_INFO_BPF (&buf->spec.info);
1566 bufsize = buf->spec.segsize * buf->spec.segtotal;
1568 /* our toy resampler for trick modes */
1569 reverse = out_samples < 0;
1570 out_samples = ABS (out_samples);
1572 if (in_samples >= out_samples)
1573 toprocess = &in_samples;
1575 toprocess = &out_samples;
1577 inr = in_samples - 1;
1578 outr = out_samples - 1;
1580 GST_DEBUG_OBJECT (psink, "in %d, out %d", inr, outr);
1582 /* data_end points to the last sample we have to write, not past it. This is
1583 * needed to properly handle reverse playback: it points to the last sample. */
1584 data_end = data + (bpf * inr);
1586 if (g_atomic_int_get (&psink->format_lost)) {
1587 /* Sink format changed, drop the data and hope upstream renegotiates */
1595 /* ensure running clock for whatever out there */
1597 if (!gst_pulsering_set_corked (pbuf, FALSE, FALSE))
1601 /* offset is in bytes */
1602 offset = *sample * bpf;
1604 while (*toprocess > 0) {
1608 GST_LOG_OBJECT (psink,
1609 "need to write %d samples at offset %" G_GINT64_FORMAT, *toprocess,
1612 if (offset != pbuf->m_lastoffset)
1613 GST_LOG_OBJECT (psink, "discontinuity, offset is %" G_GINT64_FORMAT ", "
1614 "last offset was %" G_GINT64_FORMAT, offset, pbuf->m_lastoffset);
1616 towrite = out_samples * bpf;
1618 /* Wait for at least segsize bytes to become available */
1619 if (towrite > buf->spec.segsize)
1620 towrite = buf->spec.segsize;
1622 if ((pbuf->m_writable < towrite) || (offset != pbuf->m_lastoffset)) {
1623 /* if no room left or discontinuity in offset,
1624 we need to flush data and get a new buffer */
1626 /* flush the buffer if possible */
1627 if ((pbuf->m_data != NULL) && (pbuf->m_towrite > 0)) {
1629 GST_LOG_OBJECT (psink,
1630 "flushing %u samples at offset %" G_GINT64_FORMAT,
1631 (guint) pbuf->m_towrite / bpf, pbuf->m_offset);
1633 if (pa_stream_write (pbuf->stream, (uint8_t *) pbuf->m_data,
1634 pbuf->m_towrite, NULL, pbuf->m_offset, PA_SEEK_ABSOLUTE) < 0) {
1638 pbuf->m_towrite = 0;
1639 pbuf->m_offset = offset; /* keep track of current offset */
1641 /* get a buffer to write in for now on */
1643 pbuf->m_writable = pa_stream_writable_size (pbuf->stream);
1645 if (g_atomic_int_get (&psink->format_lost)) {
1646 /* Sink format changed, give up and hope upstream renegotiates */
1650 if (pbuf->m_writable == (size_t) - 1)
1651 goto writable_size_failed;
1653 pbuf->m_writable /= bpf;
1654 pbuf->m_writable *= bpf; /* handle only complete samples */
1656 if (pbuf->m_writable >= towrite)
1659 /* see if we need to uncork because we have no free space */
1661 if (!gst_pulsering_set_corked (pbuf, FALSE, FALSE))
1665 /* we can't write segsize bytes, wait a bit */
1666 GST_LOG_OBJECT (psink, "waiting for free space");
1667 pa_threaded_mainloop_wait (mainloop);
1673 /* Recalculate what we can write in the next chunk */
1674 towrite = out_samples * bpf;
1675 if (pbuf->m_writable > towrite)
1676 pbuf->m_writable = towrite;
1678 GST_LOG_OBJECT (psink, "requesting %" G_GSIZE_FORMAT " bytes of "
1679 "shared memory", pbuf->m_writable);
1681 if (pa_stream_begin_write (pbuf->stream, &pbuf->m_data,
1682 &pbuf->m_writable) < 0) {
1683 GST_LOG_OBJECT (psink, "pa_stream_begin_write() failed");
1684 goto writable_size_failed;
1687 GST_LOG_OBJECT (psink, "got %" G_GSIZE_FORMAT " bytes of shared memory",
1692 if (towrite > pbuf->m_writable)
1693 towrite = pbuf->m_writable;
1694 avail = towrite / bpf;
1696 GST_LOG_OBJECT (psink, "writing %u samples at offset %" G_GUINT64_FORMAT,
1697 (guint) avail, offset);
1699 /* No trick modes for passthrough streams */
1700 if (G_UNLIKELY (!pbuf->is_pcm && (inr != outr || reverse))) {
1701 GST_WARNING_OBJECT (psink, "Passthrough stream can't run in trick mode");
1702 goto unlock_and_fail;
1705 if (G_LIKELY (inr == outr && !reverse)) {
1706 /* no rate conversion, simply write out the samples */
1707 /* copy the data into internal buffer */
1709 memcpy ((guint8 *) pbuf->m_data + pbuf->m_towrite, data, towrite);
1710 pbuf->m_towrite += towrite;
1711 pbuf->m_writable -= towrite;
1714 in_samples -= avail;
1715 out_samples -= avail;
1717 guint8 *dest, *d, *d_end;
1719 /* write into the PulseAudio shm buffer */
1720 dest = d = (guint8 *) pbuf->m_data + pbuf->m_towrite;
1721 d_end = d + towrite;
1725 /* forward speed up */
1726 FWD_UP_SAMPLES (data, data_end, d, d_end);
1728 /* forward slow down */
1729 FWD_DOWN_SAMPLES (data, data_end, d, d_end);
1732 /* reverse speed up */
1733 REV_UP_SAMPLES (data, data_end, d, d_end);
1735 /* reverse slow down */
1736 REV_DOWN_SAMPLES (data, data_end, d, d_end);
1738 /* see what we have left to write */
1739 towrite = (d - dest);
1740 pbuf->m_towrite += towrite;
1741 pbuf->m_writable -= towrite;
1743 avail = towrite / bpf;
1746 /* flush the buffer if it's full */
1747 if ((pbuf->m_data != NULL) && (pbuf->m_towrite > 0)
1748 && (pbuf->m_writable == 0)) {
1749 GST_LOG_OBJECT (psink, "flushing %u samples at offset %" G_GINT64_FORMAT,
1750 (guint) pbuf->m_towrite / bpf, pbuf->m_offset);
1752 if (pa_stream_write (pbuf->stream, (uint8_t *) pbuf->m_data,
1753 pbuf->m_towrite, NULL, pbuf->m_offset, PA_SEEK_ABSOLUTE) < 0) {
1756 pbuf->m_towrite = 0;
1757 pbuf->m_offset = offset + towrite; /* keep track of current offset */
1761 offset += avail * bpf;
1762 pbuf->m_lastoffset = offset;
1764 /* check if we need to uncork after writing the samples */
1766 const pa_timing_info *info;
1768 if ((info = pa_stream_get_timing_info (pbuf->stream))) {
1769 GST_LOG_OBJECT (psink,
1770 "read_index at %" G_GUINT64_FORMAT ", offset %" G_GINT64_FORMAT,
1771 info->read_index, offset);
1773 /* we uncork when the read_index is too far behind the offset we need
1775 if (info->read_index + bufsize <= offset) {
1776 if (!gst_pulsering_set_corked (pbuf, FALSE, FALSE))
1780 GST_LOG_OBJECT (psink, "no timing info available yet");
1786 /* we consumed all samples here */
1787 data = data_end + bpf;
1789 pbuf->in_commit = FALSE;
1790 pa_threaded_mainloop_unlock (mainloop);
1793 result = inr - ((data_end - data) / bpf);
1794 GST_LOG_OBJECT (psink, "wrote %d samples", result);
1801 pbuf->in_commit = FALSE;
1802 GST_LOG_OBJECT (psink, "we are reset");
1803 pa_threaded_mainloop_unlock (mainloop);
1808 GST_LOG_OBJECT (psink, "we can not start");
1813 GST_LOG_OBJECT (psink, "failed to start the ringbuffer");
1818 pbuf->in_commit = FALSE;
1819 GST_ERROR_OBJECT (psink, "uncork failed");
1820 pa_threaded_mainloop_unlock (mainloop);
1825 pbuf->in_commit = FALSE;
1826 GST_LOG_OBJECT (psink, "we are paused");
1827 pa_threaded_mainloop_unlock (mainloop);
1830 writable_size_failed:
1832 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
1833 ("pa_stream_writable_size() failed: %s",
1834 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
1835 goto unlock_and_fail;
1839 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
1840 ("pa_stream_write() failed: %s",
1841 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
1842 goto unlock_and_fail;
1846 /* write pending local samples, must be called with the mainloop lock */
1848 gst_pulsering_flush (GstPulseRingBuffer * pbuf)
1850 GstPulseSink *psink;
1852 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
1853 GST_DEBUG_OBJECT (psink, "entering flush");
1855 /* flush the buffer if possible */
1856 if (pbuf->stream && (pbuf->m_data != NULL) && (pbuf->m_towrite > 0)) {
1857 #ifndef GST_DISABLE_GST_DEBUG
1860 bpf = (GST_AUDIO_RING_BUFFER_CAST (pbuf))->spec.info.bpf;
1861 GST_LOG_OBJECT (psink,
1862 "flushing %u samples at offset %" G_GINT64_FORMAT,
1863 (guint) pbuf->m_towrite / bpf, pbuf->m_offset);
1866 if (pa_stream_write (pbuf->stream, (uint8_t *) pbuf->m_data,
1867 pbuf->m_towrite, NULL, pbuf->m_offset, PA_SEEK_ABSOLUTE) < 0) {
1871 pbuf->m_towrite = 0;
1872 pbuf->m_offset += pbuf->m_towrite; /* keep track of current offset */
1881 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
1882 ("pa_stream_write() failed: %s",
1883 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
1888 static void gst_pulsesink_set_property (GObject * object, guint prop_id,
1889 const GValue * value, GParamSpec * pspec);
1890 static void gst_pulsesink_get_property (GObject * object, guint prop_id,
1891 GValue * value, GParamSpec * pspec);
1892 static void gst_pulsesink_finalize (GObject * object);
1894 static gboolean gst_pulsesink_event (GstBaseSink * sink, GstEvent * event);
1895 static gboolean gst_pulsesink_query (GstBaseSink * sink, GstQuery * query);
1897 static GstStateChangeReturn gst_pulsesink_change_state (GstElement * element,
1898 GstStateChange transition);
1900 #define gst_pulsesink_parent_class parent_class
1901 G_DEFINE_TYPE_WITH_CODE (GstPulseSink, gst_pulsesink, GST_TYPE_AUDIO_BASE_SINK,
1902 gst_pulsesink_init_contexts ();
1903 G_IMPLEMENT_INTERFACE (GST_TYPE_STREAM_VOLUME, NULL)
1906 static GstAudioRingBuffer *
1907 gst_pulsesink_create_ringbuffer (GstAudioBaseSink * sink)
1909 GstAudioRingBuffer *buffer;
1911 GST_DEBUG_OBJECT (sink, "creating ringbuffer");
1912 buffer = g_object_new (GST_TYPE_PULSERING_BUFFER, NULL);
1913 GST_DEBUG_OBJECT (sink, "created ringbuffer @%p", buffer);
1919 gst_pulsesink_payload (GstAudioBaseSink * sink, GstBuffer * buf)
1921 switch (sink->ringbuffer->spec.type) {
1922 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3:
1923 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3:
1924 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS:
1925 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG:
1926 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC:
1927 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC:
1929 /* FIXME: alloc memory from PA if possible */
1930 gint framesize = gst_audio_iec61937_frame_size (&sink->ringbuffer->spec);
1932 GstMapInfo inmap, outmap;
1938 out = gst_buffer_new_and_alloc (framesize);
1940 gst_buffer_map (buf, &inmap, GST_MAP_READ);
1941 gst_buffer_map (out, &outmap, GST_MAP_WRITE);
1943 res = gst_audio_iec61937_payload (inmap.data, inmap.size,
1944 outmap.data, outmap.size, &sink->ringbuffer->spec, G_BIG_ENDIAN);
1946 gst_buffer_unmap (buf, &inmap);
1947 gst_buffer_unmap (out, &outmap);
1950 gst_buffer_unref (out);
1954 gst_buffer_copy_into (out, buf, GST_BUFFER_COPY_METADATA, 0, -1);
1959 return gst_buffer_ref (buf);
1963 #if defined(__TIZEN__) && defined(PCM_DUMP_ENABLE)
1964 static GstPadProbeReturn
1965 gst_pulsesink_pad_dump_probe (GstPad * pad, GstPadProbeInfo * info, gpointer data)
1967 GstPulseSink *psink = GST_PULSESINK_CAST (data);
1969 GstBuffer *buffer = GST_PAD_PROBE_INFO_BUFFER (info);
1971 if (psink->dump_fd_input) {
1972 gst_buffer_map(buffer, &in_map, GST_MAP_READ);
1973 written = fwrite(in_map.data, 1, in_map.size, psink->dump_fd_input);
1974 if (written != in_map.size)
1975 GST_WARNING("failed to write!!! ferror=%d", ferror(psink->dump_fd_input));
1976 gst_buffer_unmap(buffer, &in_map);
1978 return GST_PAD_PROBE_OK;
1980 #endif /* __TIZEN__ && PCM_DUMP_ENABLE */
1983 gst_pulsesink_class_init (GstPulseSinkClass * klass)
1985 GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
1986 GstBaseSinkClass *gstbasesink_class = GST_BASE_SINK_CLASS (klass);
1987 GstBaseSinkClass *bc;
1988 GstAudioBaseSinkClass *gstaudiosink_class = GST_AUDIO_BASE_SINK_CLASS (klass);
1989 GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
1993 gobject_class->finalize = gst_pulsesink_finalize;
1994 gobject_class->set_property = gst_pulsesink_set_property;
1995 gobject_class->get_property = gst_pulsesink_get_property;
1997 gstbasesink_class->event = GST_DEBUG_FUNCPTR (gst_pulsesink_event);
1998 gstbasesink_class->query = GST_DEBUG_FUNCPTR (gst_pulsesink_query);
2000 /* restore the original basesink pull methods */
2001 bc = g_type_class_peek (GST_TYPE_BASE_SINK);
2002 gstbasesink_class->activate_pull = GST_DEBUG_FUNCPTR (bc->activate_pull);
2004 gstelement_class->change_state =
2005 GST_DEBUG_FUNCPTR (gst_pulsesink_change_state);
2007 gstaudiosink_class->create_ringbuffer =
2008 GST_DEBUG_FUNCPTR (gst_pulsesink_create_ringbuffer);
2009 gstaudiosink_class->payload = GST_DEBUG_FUNCPTR (gst_pulsesink_payload);
2011 /* Overwrite GObject fields */
2012 g_object_class_install_property (gobject_class,
2014 g_param_spec_string ("server", "Server",
2015 "The PulseAudio server to connect to", DEFAULT_SERVER,
2016 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2018 g_object_class_install_property (gobject_class, PROP_DEVICE,
2019 g_param_spec_string ("device", "Device",
2020 "The PulseAudio sink device to connect to", DEFAULT_DEVICE,
2021 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2023 g_object_class_install_property (gobject_class, PROP_CURRENT_DEVICE,
2024 g_param_spec_string ("current-device", "Current Device",
2025 "The current PulseAudio sink device", DEFAULT_CURRENT_DEVICE,
2026 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
2028 g_object_class_install_property (gobject_class,
2030 g_param_spec_string ("device-name", "Device name",
2031 "Human-readable name of the sound device", DEFAULT_DEVICE_NAME,
2032 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
2034 g_object_class_install_property (gobject_class,
2036 g_param_spec_double ("volume", "Volume",
2037 "Linear volume of this stream, 1.0=100%", 0.0, MAX_VOLUME,
2038 DEFAULT_VOLUME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2039 g_object_class_install_property (gobject_class,
2041 g_param_spec_boolean ("mute", "Mute",
2042 "Mute state of this stream", DEFAULT_MUTE,
2043 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2046 * GstPulseSink:client-name:
2048 * The PulseAudio client name to use.
2050 clientname = gst_pulse_client_name ();
2051 g_object_class_install_property (gobject_class,
2053 g_param_spec_string ("client-name", "Client Name",
2054 "The PulseAudio client name to use", clientname,
2055 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
2056 GST_PARAM_MUTABLE_READY));
2057 g_free (clientname);
2060 * GstPulseSink:stream-properties:
2062 * List of pulseaudio stream properties. A list of defined properties can be
2063 * found in the <ulink url="http://0pointer.de/lennart/projects/pulseaudio/doxygen/proplist_8h.html">pulseaudio api docs</ulink>.
2065 * Below is an example for registering as a music application to pulseaudio.
2067 * GstStructure *props;
2069 * props = gst_structure_from_string ("props,media.role=music", NULL);
2070 * g_object_set (pulse, "stream-properties", props, NULL);
2071 * gst_structure_free
2074 g_object_class_install_property (gobject_class,
2075 PROP_STREAM_PROPERTIES,
2076 g_param_spec_boxed ("stream-properties", "stream properties",
2077 "list of pulseaudio stream properties",
2078 GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2081 g_object_class_install_property (gobject_class,
2083 g_param_spec_string ("latency", "Audio Backend Latency",
2084 "Audio Backend Latency (\"low\": Low Latency, \"mid\": Mid Latency, \"high\": High Latency)",
2085 DEFAULT_AUDIO_LATENCY,
2086 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2087 #endif /* __TIZEN__ */
2089 gst_element_class_set_static_metadata (gstelement_class,
2090 "PulseAudio Audio Sink",
2091 "Sink/Audio", "Plays audio to a PulseAudio server", "Lennart Poettering");
2094 gst_pulse_fix_pcm_caps (gst_caps_from_string (PULSE_SINK_TEMPLATE_CAPS));
2095 gst_element_class_add_pad_template (gstelement_class,
2096 gst_pad_template_new ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, caps));
2097 gst_caps_unref (caps);
2101 free_device_info (GstPulseDeviceInfo * device_info)
2105 g_free (device_info->description);
2107 for (l = g_list_first (device_info->formats); l; l = g_list_next (l))
2108 pa_format_info_free ((pa_format_info *) l->data);
2110 g_list_free (device_info->formats);
2113 /* Returns the current time of the sink ringbuffer. The timing_info is updated
2114 * on every data write/flush and every 100ms (PA_STREAM_AUTO_TIMING_UPDATE).
2117 gst_pulsesink_get_time (GstClock * clock, GstAudioBaseSink * sink)
2119 GstPulseSink *psink;
2120 GstPulseRingBuffer *pbuf;
2123 if (!sink->ringbuffer || !sink->ringbuffer->acquired)
2124 return GST_CLOCK_TIME_NONE;
2126 pbuf = GST_PULSERING_BUFFER_CAST (sink->ringbuffer);
2127 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
2129 if (g_atomic_int_get (&psink->format_lost)) {
2130 /* Stream was lost in a format change, it'll get set up again once
2131 * upstream renegotiates */
2132 return psink->format_lost_time;
2135 pa_threaded_mainloop_lock (mainloop);
2136 if (gst_pulsering_is_dead (psink, pbuf, TRUE))
2139 /* if we don't have enough data to get a timestamp, just return NONE, which
2140 * will return the last reported time */
2141 if (pa_stream_get_time (pbuf->stream, &time) < 0) {
2142 GST_DEBUG_OBJECT (psink, "could not get time");
2143 time = GST_CLOCK_TIME_NONE;
2146 pa_threaded_mainloop_unlock (mainloop);
2148 GST_LOG_OBJECT (psink, "current time is %" GST_TIME_FORMAT,
2149 GST_TIME_ARGS (time));
2156 GST_DEBUG_OBJECT (psink, "the server is dead");
2157 pa_threaded_mainloop_unlock (mainloop);
2159 return GST_CLOCK_TIME_NONE;
2164 gst_pulsesink_sink_info_cb (pa_context * c, const pa_sink_info * i, int eol,
2167 GstPulseDeviceInfo *device_info = (GstPulseDeviceInfo *) userdata;
2173 device_info->description = g_strdup (i->description);
2175 device_info->formats = NULL;
2176 for (j = 0; j < i->n_formats; j++)
2177 device_info->formats = g_list_prepend (device_info->formats,
2178 pa_format_info_copy (i->formats[j]));
2181 pa_threaded_mainloop_signal (mainloop, 0);
2184 /* Call with mainloop lock held */
2186 gst_pulsesink_create_probe_stream (GstPulseSink * psink,
2187 GstPulseRingBuffer * pbuf, pa_format_info * format)
2189 pa_format_info *formats[1] = { format };
2191 pa_stream_flags_t flags;
2193 GST_LOG_OBJECT (psink, "Creating probe stream");
2195 if (!(stream = pa_stream_new_extended (pbuf->context, "pulsesink probe",
2196 formats, 1, psink->proplist)))
2199 /* construct the flags */
2200 flags = PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE |
2201 PA_STREAM_ADJUST_LATENCY | PA_STREAM_START_CORKED;
2203 pa_stream_set_state_callback (stream, gst_pulsering_stream_state_cb, pbuf);
2205 if (pa_stream_connect_playback (stream, psink->device, NULL, flags, NULL,
2209 if (!gst_pulsering_wait_for_stream_ready (psink, stream))
2216 pa_stream_unref (stream);
2221 gst_pulsesink_query_getcaps (GstPulseSink * psink, GstCaps * filter)
2223 GstPulseRingBuffer *pbuf = NULL;
2224 GstPulseDeviceInfo device_info = { NULL, NULL };
2225 GstCaps *ret = NULL;
2227 pa_operation *o = NULL;
2230 GST_OBJECT_LOCK (psink);
2231 pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
2233 gst_object_ref (pbuf);
2234 GST_OBJECT_UNLOCK (psink);
2237 ret = gst_pad_get_pad_template_caps (GST_AUDIO_BASE_SINK_PAD (psink));
2241 GST_OBJECT_LOCK (pbuf);
2242 pa_threaded_mainloop_lock (mainloop);
2244 if (!pbuf->context) {
2245 ret = gst_pad_get_pad_template_caps (GST_AUDIO_BASE_SINK_PAD (psink));
2249 ret = gst_caps_new_empty ();
2252 /* We're in PAUSED or higher */
2253 stream = pbuf->stream;
2255 } else if (pbuf->probe_stream) {
2256 /* We're not paused, but have a cached probe stream */
2257 stream = pbuf->probe_stream;
2260 /* We're not yet in PAUSED and still need to create a probe stream.
2262 * FIXME: PA doesn't accept "any" format. We fix something reasonable since
2263 * this is merely a probe. This should eventually be fixed in PA and
2264 * hard-coding the format should be dropped. */
2265 pa_format_info *format = pa_format_info_new ();
2266 format->encoding = PA_ENCODING_PCM;
2267 pa_format_info_set_sample_format (format, PA_SAMPLE_S16LE);
2268 pa_format_info_set_rate (format, GST_AUDIO_DEF_RATE);
2269 pa_format_info_set_channels (format, GST_AUDIO_DEF_CHANNELS);
2271 pbuf->probe_stream = gst_pulsesink_create_probe_stream (psink, pbuf,
2274 pa_format_info_free (format);
2276 if (!pbuf->probe_stream) {
2277 GST_WARNING_OBJECT (psink, "Could not create probe stream");
2281 stream = pbuf->probe_stream;
2284 if (!(o = pa_context_get_sink_info_by_name (pbuf->context,
2285 pa_stream_get_device_name (stream), gst_pulsesink_sink_info_cb,
2289 while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
2290 pa_threaded_mainloop_wait (mainloop);
2291 if (gst_pulsering_is_dead (psink, pbuf, FALSE))
2295 for (i = g_list_first (device_info.formats); i; i = g_list_next (i)) {
2296 GstCaps *caps = gst_pulse_format_info_to_caps ((pa_format_info *) i->data);
2298 gst_caps_append (ret, caps);
2302 pa_threaded_mainloop_unlock (mainloop);
2303 /* FIXME: this could be freed after device_name is got */
2304 GST_OBJECT_UNLOCK (pbuf);
2307 GstCaps *tmp = gst_caps_intersect_full (filter, ret,
2308 GST_CAPS_INTERSECT_FIRST);
2309 gst_caps_unref (ret);
2314 free_device_info (&device_info);
2317 pa_operation_unref (o);
2320 gst_object_unref (pbuf);
2322 GST_DEBUG_OBJECT (psink, "caps %" GST_PTR_FORMAT, ret);
2328 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
2329 ("pa_context_get_sink_input_info() failed: %s",
2330 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
2336 gst_pulsesink_query_acceptcaps (GstPulseSink * psink, GstCaps * caps)
2338 GstPulseRingBuffer *pbuf = NULL;
2339 GstPulseDeviceInfo device_info = { NULL, NULL };
2342 gboolean ret = FALSE;
2344 GstAudioRingBufferSpec spec = { 0 };
2345 pa_operation *o = NULL;
2346 pa_channel_map channel_map;
2347 pa_format_info *format = NULL;
2350 pad_caps = gst_pad_get_pad_template_caps (GST_BASE_SINK_PAD (psink));
2351 ret = gst_caps_is_subset (caps, pad_caps);
2352 gst_caps_unref (pad_caps);
2354 GST_DEBUG_OBJECT (psink, "caps %" GST_PTR_FORMAT, caps);
2356 /* Template caps didn't match */
2360 /* If we've not got fixed caps, creating a stream might fail, so let's just
2361 * return from here with default acceptcaps behaviour */
2362 if (!gst_caps_is_fixed (caps))
2365 GST_OBJECT_LOCK (psink);
2366 pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
2368 gst_object_ref (pbuf);
2369 GST_OBJECT_UNLOCK (psink);
2371 /* We're still in NULL state */
2375 GST_OBJECT_LOCK (pbuf);
2376 pa_threaded_mainloop_lock (mainloop);
2378 if (pbuf->context == NULL)
2383 spec.latency_time = GST_AUDIO_BASE_SINK (psink)->latency_time;
2384 if (!gst_audio_ring_buffer_parse_caps (&spec, caps))
2387 if (!gst_pulse_fill_format_info (&spec, &format, &channels))
2390 /* Make sure input is framed (one frame per buffer) and can be payloaded */
2391 if (!pa_format_info_is_pcm (format)) {
2392 gboolean framed = FALSE, parsed = FALSE;
2393 st = gst_caps_get_structure (caps, 0);
2395 gst_structure_get_boolean (st, "framed", &framed);
2396 gst_structure_get_boolean (st, "parsed", &parsed);
2397 if ((!framed && !parsed) || gst_audio_iec61937_frame_size (&spec) <= 0)
2401 /* initialize the channel map */
2402 if (pa_format_info_is_pcm (format) &&
2403 gst_pulse_gst_to_channel_map (&channel_map, &spec))
2404 pa_format_info_set_channel_map (format, &channel_map);
2406 if (pbuf->stream || pbuf->probe_stream) {
2407 /* We're already in PAUSED or above, so just reuse this stream to query
2408 * sink formats and use those. */
2410 const char *device_name = pa_stream_get_device_name (pbuf->stream ?
2411 pbuf->stream : pbuf->probe_stream);
2413 if (!(o = pa_context_get_sink_info_by_name (pbuf->context, device_name,
2414 gst_pulsesink_sink_info_cb, &device_info)))
2417 while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
2418 pa_threaded_mainloop_wait (mainloop);
2419 if (gst_pulsering_is_dead (psink, pbuf, FALSE))
2423 for (i = g_list_first (device_info.formats); i; i = g_list_next (i)) {
2424 if (pa_format_info_is_compatible ((pa_format_info *) i->data, format)) {
2430 /* We're in READY, let's connect a stream to see if the format is
2431 * accepted by whatever sink we're routed to */
2432 pbuf->probe_stream = gst_pulsesink_create_probe_stream (psink, pbuf,
2434 if (pbuf->probe_stream)
2440 pa_format_info_free (format);
2442 free_device_info (&device_info);
2445 pa_operation_unref (o);
2447 pa_threaded_mainloop_unlock (mainloop);
2448 GST_OBJECT_UNLOCK (pbuf);
2450 gst_caps_replace (&spec.caps, NULL);
2451 gst_object_unref (pbuf);
2459 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
2460 ("pa_context_get_sink_input_info() failed: %s",
2461 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
2467 gst_pulsesink_init (GstPulseSink * pulsesink)
2469 #if defined(__TIZEN__) && defined(PCM_DUMP_ENABLE)
2470 GstPad *sinkpad = NULL;
2472 #endif /* __TIZEN__ && PCM_DUMP_ENABLE */
2474 pulsesink->server = NULL;
2475 pulsesink->device = NULL;
2476 pulsesink->device_info.description = NULL;
2477 pulsesink->client_name = gst_pulse_client_name ();
2479 pulsesink->device_info.formats = NULL;
2481 pulsesink->volume = DEFAULT_VOLUME;
2482 pulsesink->volume_set = FALSE;
2484 pulsesink->mute = DEFAULT_MUTE;
2485 pulsesink->mute_set = FALSE;
2487 pulsesink->notify = 0;
2489 g_atomic_int_set (&pulsesink->format_lost, FALSE);
2490 pulsesink->format_lost_time = GST_CLOCK_TIME_NONE;
2492 pulsesink->properties = NULL;
2493 pulsesink->proplist = NULL;
2495 pulsesink->latency = g_strdup (DEFAULT_AUDIO_LATENCY);
2496 pulsesink->proplist = pa_proplist_new();
2497 pa_proplist_sets(pulsesink->proplist, PA_PROP_MEDIA_TIZEN_AUDIO_LATENCY, pulsesink->latency);
2498 #ifdef PCM_DUMP_ENABLE
2499 if (vconf_get_int(GST_PULSESINK_DUMP_VCONF_KEY, &vconf_dump)) {
2500 GST_WARNING("vconf_get_int %s failed", GST_PULSESINK_DUMP_VCONF_KEY);
2502 pulsesink->need_dump_input = vconf_dump & GST_PULSESINK_DUMP_INPUT_FLAG ? TRUE : FALSE;
2503 pulsesink->dump_fd_input = NULL;
2504 if (pulsesink->need_dump_input) {
2505 sinkpad = gst_element_get_static_pad((GstElement *)pulsesink, "sink");
2507 gst_pad_add_probe (sinkpad, GST_PAD_PROBE_TYPE_BUFFER, gst_pulsesink_pad_dump_probe, pulsesink, NULL);
2508 gst_object_unref (GST_OBJECT(sinkpad));
2512 #endif /* __TIZEN__ */
2514 /* override with a custom clock */
2515 if (GST_AUDIO_BASE_SINK (pulsesink)->provided_clock)
2516 gst_object_unref (GST_AUDIO_BASE_SINK (pulsesink)->provided_clock);
2518 GST_AUDIO_BASE_SINK (pulsesink)->provided_clock =
2519 gst_audio_clock_new ("GstPulseSinkClock",
2520 (GstAudioClockGetTimeFunc) gst_pulsesink_get_time, pulsesink, NULL);
2524 gst_pulsesink_finalize (GObject * object)
2526 GstPulseSink *pulsesink = GST_PULSESINK_CAST (object);
2528 g_free (pulsesink->server);
2529 g_free (pulsesink->device);
2530 g_free (pulsesink->client_name);
2531 g_free (pulsesink->current_sink_name);
2533 free_device_info (&pulsesink->device_info);
2535 if (pulsesink->properties)
2536 gst_structure_free (pulsesink->properties);
2537 if (pulsesink->proplist)
2538 pa_proplist_free (pulsesink->proplist);
2541 g_free (pulsesink->latency);
2542 #endif /* __TIZEN__ */
2544 G_OBJECT_CLASS (parent_class)->finalize (object);
2548 gst_pulsesink_set_volume (GstPulseSink * psink, gdouble volume)
2552 pa_operation *o = NULL;
2554 GstPulseRingBuffer *pbuf;
2561 pa_threaded_mainloop_lock (mainloop);
2564 GST_DEBUG_OBJECT (psink, "setting volume to %f", volume);
2566 pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
2567 if (pbuf == NULL || pbuf->stream == NULL)
2570 if ((idx = pa_stream_get_index (pbuf->stream)) == PA_INVALID_INDEX)
2575 gst_pulse_cvolume_from_linear (&v, pbuf->channels, volume);
2577 /* FIXME: this will eventually be superceded by checks to see if the volume
2578 * is readable/writable */
2581 if (!(o = pa_context_set_sink_input_volume (pbuf->context, idx,
2587 gst_pulse_set_volume_ratio (idx, "out", volume);
2588 psink->volume = volume;
2591 /* We don't really care about the result of this call */
2596 pa_operation_unref (o);
2598 pa_threaded_mainloop_unlock (mainloop);
2607 psink->volume = volume;
2608 psink->volume_set = TRUE;
2610 GST_DEBUG_OBJECT (psink, "we have no mainloop");
2616 psink->volume = volume;
2617 psink->volume_set = TRUE;
2619 GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
2624 GST_DEBUG_OBJECT (psink, "we don't have a stream index");
2630 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
2631 ("pa_stream_set_sink_input_volume() failed: %s",
2632 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
2639 gst_pulsesink_set_mute (GstPulseSink * psink, gboolean mute)
2642 pa_operation *o = NULL;
2644 GstPulseRingBuffer *pbuf;
2651 pa_threaded_mainloop_lock (mainloop);
2654 GST_DEBUG_OBJECT (psink, "setting mute state to %d", mute);
2656 pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
2657 if (pbuf == NULL || pbuf->stream == NULL)
2660 if ((idx = pa_stream_get_index (pbuf->stream)) == PA_INVALID_INDEX)
2664 if (!(o = pa_context_set_sink_input_mute (pbuf->context, idx,
2668 gst_pulse_set_volume_ratio (idx, "out", mute ? 0 : psink->volume);
2672 /* We don't really care about the result of this call */
2677 pa_operation_unref (o);
2679 pa_threaded_mainloop_unlock (mainloop);
2689 psink->mute_set = TRUE;
2691 GST_DEBUG_OBJECT (psink, "we have no mainloop");
2698 psink->mute_set = TRUE;
2700 GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
2705 GST_DEBUG_OBJECT (psink, "we don't have a stream index");
2711 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
2712 ("pa_stream_set_sink_input_mute() failed: %s",
2713 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
2720 gst_pulsesink_sink_input_info_cb (pa_context * c, const pa_sink_input_info * i,
2721 int eol, void *userdata)
2723 GstPulseRingBuffer *pbuf;
2724 GstPulseSink *psink;
2726 pbuf = GST_PULSERING_BUFFER_CAST (userdata);
2727 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
2735 /* If the index doesn't match our current stream,
2736 * it implies we just recreated the stream (caps change)
2738 if (i->index == pa_stream_get_index (pbuf->stream)) {
2739 psink->volume = pa_sw_volume_to_linear (pa_cvolume_max (&i->volume));
2740 psink->mute = i->mute;
2741 psink->current_sink_idx = i->sink;
2743 if (psink->volume > MAX_VOLUME) {
2744 GST_WARNING_OBJECT (psink, "Clipped volume from %f to %f", psink->volume,
2746 psink->volume = MAX_VOLUME;
2751 pa_threaded_mainloop_signal (mainloop, 0);
2755 gst_pulsesink_get_sink_input_info (GstPulseSink * psink, gdouble * volume,
2758 GstPulseRingBuffer *pbuf;
2759 pa_operation *o = NULL;
2765 pa_threaded_mainloop_lock (mainloop);
2767 pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
2768 if (pbuf == NULL || pbuf->stream == NULL)
2771 if ((idx = pa_stream_get_index (pbuf->stream)) == PA_INVALID_INDEX)
2774 if (!(o = pa_context_get_sink_input_info (pbuf->context, idx,
2775 gst_pulsesink_sink_input_info_cb, pbuf)))
2778 while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
2779 pa_threaded_mainloop_wait (mainloop);
2780 if (gst_pulsering_is_dead (psink, pbuf, TRUE))
2786 *volume = psink->volume;
2788 *mute = psink->mute;
2791 pa_operation_unref (o);
2793 pa_threaded_mainloop_unlock (mainloop);
2801 *volume = psink->volume;
2803 *mute = psink->mute;
2805 GST_DEBUG_OBJECT (psink, "we have no mainloop");
2810 GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
2815 GST_DEBUG_OBJECT (psink, "we don't have a stream index");
2820 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
2821 ("pa_context_get_sink_input_info() failed: %s",
2822 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
2828 gst_pulsesink_current_sink_info_cb (pa_context * c, const pa_sink_info * i,
2829 int eol, void *userdata)
2831 GstPulseSink *psink;
2833 psink = GST_PULSESINK_CAST (userdata);
2838 /* If the index doesn't match our current stream,
2839 * it implies we just recreated the stream (caps change)
2841 if (i->index == psink->current_sink_idx) {
2842 g_free (psink->current_sink_name);
2843 psink->current_sink_name = g_strdup (i->name);
2847 pa_threaded_mainloop_signal (mainloop, 0);
2851 gst_pulsesink_get_current_device (GstPulseSink * pulsesink)
2853 pa_operation *o = NULL;
2854 GstPulseRingBuffer *pbuf;
2855 gchar *current_sink;
2861 GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (pulsesink)->ringbuffer);
2862 if (pbuf == NULL || pbuf->stream == NULL)
2865 gst_pulsesink_get_sink_input_info (pulsesink, NULL, NULL);
2867 pa_threaded_mainloop_lock (mainloop);
2869 if (!(o = pa_context_get_sink_info_by_index (pbuf->context,
2870 pulsesink->current_sink_idx, gst_pulsesink_current_sink_info_cb,
2874 while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
2875 pa_threaded_mainloop_wait (mainloop);
2876 if (gst_pulsering_is_dead (pulsesink, pbuf, TRUE))
2882 current_sink = g_strdup (pulsesink->current_sink_name);
2885 pa_operation_unref (o);
2887 pa_threaded_mainloop_unlock (mainloop);
2889 return current_sink;
2894 GST_DEBUG_OBJECT (pulsesink, "we have no mainloop");
2899 GST_DEBUG_OBJECT (pulsesink, "we have no ringbuffer");
2904 GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED,
2905 ("pa_context_get_sink_input_info() failed: %s",
2906 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
2912 gst_pulsesink_device_description (GstPulseSink * psink)
2914 GstPulseRingBuffer *pbuf;
2915 pa_operation *o = NULL;
2921 pa_threaded_mainloop_lock (mainloop);
2922 pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
2926 free_device_info (&psink->device_info);
2927 if (!(o = pa_context_get_sink_info_by_name (pbuf->context,
2928 psink->device, gst_pulsesink_sink_info_cb, &psink->device_info)))
2931 while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
2932 pa_threaded_mainloop_wait (mainloop);
2933 if (gst_pulsering_is_dead (psink, pbuf, FALSE))
2939 pa_operation_unref (o);
2941 t = g_strdup (psink->device_info.description);
2942 pa_threaded_mainloop_unlock (mainloop);
2949 GST_DEBUG_OBJECT (psink, "we have no mainloop");
2954 GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
2959 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
2960 ("pa_context_get_sink_info_by_index() failed: %s",
2961 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
2967 gst_pulsesink_set_stream_device (GstPulseSink * psink, const gchar * device)
2969 pa_operation *o = NULL;
2970 GstPulseRingBuffer *pbuf;
2976 pa_threaded_mainloop_lock (mainloop);
2978 pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
2979 if (pbuf == NULL || pbuf->stream == NULL)
2982 if ((idx = pa_stream_get_index (pbuf->stream)) == PA_INVALID_INDEX)
2986 GST_DEBUG_OBJECT (psink, "setting stream device to %s", device);
2988 if (!(o = pa_context_move_sink_input_by_name (pbuf->context, idx, device,
2995 pa_operation_unref (o);
2997 pa_threaded_mainloop_unlock (mainloop);
3004 GST_DEBUG_OBJECT (psink, "we have no mainloop");
3009 GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
3014 GST_DEBUG_OBJECT (psink, "we don't have a stream index");
3019 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
3020 ("pa_context_move_sink_input_by_name(%s) failed: %s", device,
3021 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
3028 gst_pulsesink_set_property (GObject * object,
3029 guint prop_id, const GValue * value, GParamSpec * pspec)
3031 GstPulseSink *pulsesink = GST_PULSESINK_CAST (object);
3035 g_free (pulsesink->server);
3036 pulsesink->server = g_value_dup_string (value);
3039 g_free (pulsesink->device);
3040 pulsesink->device = g_value_dup_string (value);
3041 gst_pulsesink_set_stream_device (pulsesink, pulsesink->device);
3044 gst_pulsesink_set_volume (pulsesink, g_value_get_double (value));
3047 gst_pulsesink_set_mute (pulsesink, g_value_get_boolean (value));
3049 case PROP_CLIENT_NAME:
3050 g_free (pulsesink->client_name);
3051 if (!g_value_get_string (value)) {
3052 GST_WARNING_OBJECT (pulsesink,
3053 "Empty PulseAudio client name not allowed. Resetting to default value");
3054 pulsesink->client_name = gst_pulse_client_name ();
3056 pulsesink->client_name = g_value_dup_string (value);
3058 case PROP_STREAM_PROPERTIES:
3059 if (pulsesink->properties)
3060 gst_structure_free (pulsesink->properties);
3061 pulsesink->properties =
3062 gst_structure_copy (gst_value_get_structure (value));
3063 if (pulsesink->proplist)
3064 pa_proplist_free (pulsesink->proplist);
3065 pulsesink->proplist = gst_pulse_make_proplist (pulsesink->properties);
3068 case PROP_AUDIO_LATENCY:
3069 g_free (pulsesink->latency);
3070 pulsesink->latency = g_value_dup_string (value);
3071 /* setting NULL restores the default latency */
3072 if (pulsesink->latency == NULL) {
3073 pulsesink->latency = g_strdup (DEFAULT_AUDIO_LATENCY);
3075 if (!pulsesink->proplist) {
3076 pulsesink->proplist = pa_proplist_new();
3078 pa_proplist_sets(pulsesink->proplist, PA_PROP_MEDIA_TIZEN_AUDIO_LATENCY, pulsesink->latency);
3079 GST_DEBUG_OBJECT(pulsesink, "latency(%s)", pulsesink->latency);
3081 #endif /* __TIZEN__ */
3083 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
3089 gst_pulsesink_get_property (GObject * object,
3090 guint prop_id, GValue * value, GParamSpec * pspec)
3093 GstPulseSink *pulsesink = GST_PULSESINK_CAST (object);
3097 g_value_set_string (value, pulsesink->server);
3100 g_value_set_string (value, pulsesink->device);
3102 case PROP_CURRENT_DEVICE:
3104 gchar *current_device = gst_pulsesink_get_current_device (pulsesink);
3106 g_value_take_string (value, current_device);
3108 g_value_set_string (value, "");
3111 case PROP_DEVICE_NAME:
3112 g_value_take_string (value, gst_pulsesink_device_description (pulsesink));
3119 gst_pulsesink_get_sink_input_info (pulsesink, &volume, NULL);
3120 g_value_set_double (value, volume);
3122 g_value_set_double (value, pulsesink->volume);
3131 gst_pulsesink_get_sink_input_info (pulsesink, NULL, &mute);
3132 g_value_set_boolean (value, mute);
3134 g_value_set_boolean (value, pulsesink->mute);
3138 case PROP_CLIENT_NAME:
3139 g_value_set_string (value, pulsesink->client_name);
3141 case PROP_STREAM_PROPERTIES:
3142 gst_value_set_structure (value, pulsesink->properties);
3145 case PROP_AUDIO_LATENCY:
3146 g_value_set_string (value, pulsesink->latency);
3148 #endif /* __TIZEN__ */
3150 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
3156 gst_pulsesink_change_title (GstPulseSink * psink, const gchar * t)
3158 pa_operation *o = NULL;
3159 GstPulseRingBuffer *pbuf;
3161 pa_threaded_mainloop_lock (mainloop);
3163 pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
3165 if (pbuf == NULL || pbuf->stream == NULL)
3168 g_free (pbuf->stream_name);
3169 pbuf->stream_name = g_strdup (t);
3171 if (!(o = pa_stream_set_name (pbuf->stream, pbuf->stream_name, NULL, NULL)))
3174 /* We're not interested if this operation failed or not */
3178 pa_operation_unref (o);
3179 pa_threaded_mainloop_unlock (mainloop);
3186 GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
3191 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
3192 ("pa_stream_set_name() failed: %s",
3193 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
3199 gst_pulsesink_change_props (GstPulseSink * psink, GstTagList * l)
3201 static const gchar *const map[] = {
3202 GST_TAG_TITLE, PA_PROP_MEDIA_TITLE,
3204 /* might get overriden in the next iteration by GST_TAG_ARTIST */
3205 GST_TAG_PERFORMER, PA_PROP_MEDIA_ARTIST,
3207 GST_TAG_ARTIST, PA_PROP_MEDIA_ARTIST,
3208 GST_TAG_LANGUAGE_CODE, PA_PROP_MEDIA_LANGUAGE,
3209 GST_TAG_LOCATION, PA_PROP_MEDIA_FILENAME,
3210 /* We might add more here later on ... */
3213 pa_proplist *pl = NULL;
3214 const gchar *const *t;
3215 gboolean empty = TRUE;
3216 pa_operation *o = NULL;
3217 GstPulseRingBuffer *pbuf;
3219 pl = pa_proplist_new ();
3221 for (t = map; *t; t += 2) {
3224 if (gst_tag_list_get_string (l, *t, &n)) {
3227 pa_proplist_sets (pl, *(t + 1), n);
3237 pa_threaded_mainloop_lock (mainloop);
3238 pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
3239 if (pbuf == NULL || pbuf->stream == NULL)
3242 /* We're not interested if this operation failed or not */
3243 if (!(o = pa_stream_proplist_update (pbuf->stream, PA_UPDATE_REPLACE,
3245 GST_DEBUG_OBJECT (psink, "pa_stream_proplist_update() failed");
3251 pa_operation_unref (o);
3253 pa_threaded_mainloop_unlock (mainloop);
3258 pa_proplist_free (pl);
3265 GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
3271 gst_pulsesink_flush_ringbuffer (GstPulseSink * psink)
3273 GstPulseRingBuffer *pbuf;
3275 pa_threaded_mainloop_lock (mainloop);
3277 pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
3279 if (pbuf == NULL || pbuf->stream == NULL)
3282 gst_pulsering_flush (pbuf);
3284 /* Uncork if we haven't already (happens when waiting to get enough data
3285 * to send out the first time) */
3287 gst_pulsering_set_corked (pbuf, FALSE, FALSE);
3289 /* We're not interested if this operation failed or not */
3291 pa_threaded_mainloop_unlock (mainloop);
3298 GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
3304 gst_pulsesink_event (GstBaseSink * sink, GstEvent * event)
3306 GstPulseSink *pulsesink = GST_PULSESINK_CAST (sink);
3308 switch (GST_EVENT_TYPE (event)) {
3309 case GST_EVENT_TAG:{
3310 gchar *title = NULL, *artist = NULL, *location = NULL, *description =
3311 NULL, *t = NULL, *buf = NULL;
3314 gst_event_parse_tag (event, &l);
3316 gst_tag_list_get_string (l, GST_TAG_TITLE, &title);
3317 gst_tag_list_get_string (l, GST_TAG_ARTIST, &artist);
3318 gst_tag_list_get_string (l, GST_TAG_LOCATION, &location);
3319 gst_tag_list_get_string (l, GST_TAG_DESCRIPTION, &description);
3322 gst_tag_list_get_string (l, GST_TAG_PERFORMER, &artist);
3324 if (title && artist)
3325 /* TRANSLATORS: 'song title' by 'artist name' */
3326 t = buf = g_strdup_printf (_("'%s' by '%s'"), g_strstrip (title),
3327 g_strstrip (artist));
3329 t = g_strstrip (title);
3330 else if (description)
3331 t = g_strstrip (description);
3333 t = g_strstrip (location);
3336 gst_pulsesink_change_title (pulsesink, t);
3341 g_free (description);
3344 gst_pulsesink_change_props (pulsesink, l);
3348 case GST_EVENT_GAP:{
3349 GstClockTime timestamp, duration;
3351 gst_event_parse_gap (event, ×tamp, &duration);
3352 if (duration == GST_CLOCK_TIME_NONE)
3353 gst_pulsesink_flush_ringbuffer (pulsesink);
3357 gst_pulsesink_flush_ringbuffer (pulsesink);
3363 return GST_BASE_SINK_CLASS (parent_class)->event (sink, event);
3367 gst_pulsesink_query (GstBaseSink * sink, GstQuery * query)
3369 GstPulseSink *pulsesink = GST_PULSESINK_CAST (sink);
3370 gboolean ret = FALSE;
3372 switch (GST_QUERY_TYPE (query)) {
3373 case GST_QUERY_CAPS:
3375 GstCaps *caps, *filter;
3377 gst_query_parse_caps (query, &filter);
3378 caps = gst_pulsesink_query_getcaps (pulsesink, filter);
3381 gst_query_set_caps_result (query, caps);
3382 gst_caps_unref (caps);
3387 case GST_QUERY_ACCEPT_CAPS:
3391 gst_query_parse_accept_caps (query, &caps);
3392 ret = gst_pulsesink_query_acceptcaps (pulsesink, caps);
3393 gst_query_set_accept_caps_result (query, ret);
3398 ret = GST_BASE_SINK_CLASS (parent_class)->query (sink, query);
3405 gst_pulsesink_release_mainloop (GstPulseSink * psink)
3410 pa_threaded_mainloop_lock (mainloop);
3411 while (psink->defer_pending) {
3412 GST_DEBUG_OBJECT (psink, "waiting for stream status message emission");
3413 pa_threaded_mainloop_wait (mainloop);
3415 pa_threaded_mainloop_unlock (mainloop);
3417 g_mutex_lock (&pa_shared_resource_mutex);
3419 if (!mainloop_ref_ct) {
3420 GST_INFO_OBJECT (psink, "terminating pa main loop thread");
3421 pa_threaded_mainloop_stop (mainloop);
3422 pa_threaded_mainloop_free (mainloop);
3425 g_mutex_unlock (&pa_shared_resource_mutex);
3428 static GstStateChangeReturn
3429 gst_pulsesink_change_state (GstElement * element, GstStateChange transition)
3431 GstPulseSink *pulsesink = GST_PULSESINK (element);
3432 GstStateChangeReturn ret;
3434 switch (transition) {
3435 case GST_STATE_CHANGE_NULL_TO_READY:
3436 g_mutex_lock (&pa_shared_resource_mutex);
3437 if (!mainloop_ref_ct) {
3438 GST_INFO_OBJECT (element, "new pa main loop thread");
3439 if (!(mainloop = pa_threaded_mainloop_new ()))
3440 goto mainloop_failed;
3441 if (pa_threaded_mainloop_start (mainloop) < 0) {
3442 pa_threaded_mainloop_free (mainloop);
3443 goto mainloop_start_failed;
3445 mainloop_ref_ct = 1;
3446 g_mutex_unlock (&pa_shared_resource_mutex);
3448 GST_INFO_OBJECT (element, "reusing pa main loop thread");
3450 g_mutex_unlock (&pa_shared_resource_mutex);
3453 case GST_STATE_CHANGE_READY_TO_PAUSED:
3454 gst_element_post_message (element,
3455 gst_message_new_clock_provide (GST_OBJECT_CAST (element),
3456 GST_AUDIO_BASE_SINK (pulsesink)->provided_clock, TRUE));
3463 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
3464 if (ret == GST_STATE_CHANGE_FAILURE)
3467 switch (transition) {
3468 case GST_STATE_CHANGE_PAUSED_TO_READY:
3469 /* format_lost is reset in release() in audiobasesink */
3470 gst_element_post_message (element,
3471 gst_message_new_clock_lost (GST_OBJECT_CAST (element),
3472 GST_AUDIO_BASE_SINK (pulsesink)->provided_clock));
3474 case GST_STATE_CHANGE_READY_TO_NULL:
3475 gst_pulsesink_release_mainloop (pulsesink);
3486 g_mutex_unlock (&pa_shared_resource_mutex);
3487 GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED,
3488 ("pa_threaded_mainloop_new() failed"), (NULL));
3489 return GST_STATE_CHANGE_FAILURE;
3491 mainloop_start_failed:
3493 g_mutex_unlock (&pa_shared_resource_mutex);
3494 GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED,
3495 ("pa_threaded_mainloop_start() failed"), (NULL));
3496 return GST_STATE_CHANGE_FAILURE;
3500 if (transition == GST_STATE_CHANGE_NULL_TO_READY) {
3501 /* Clear the PA mainloop if audiobasesink failed to open the ring_buffer */
3502 g_assert (mainloop);
3503 gst_pulsesink_release_mainloop (pulsesink);