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;
919 pa_cvolume *pv = NULL;
920 pa_stream_flags_t flags;
922 GstAudioClock *clock;
923 pa_format_info *formats[1];
924 #ifndef GST_DISABLE_GST_DEBUG
925 gchar print_buf[PA_FORMAT_INFO_SNPRINT_MAX];
928 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (buf));
929 pbuf = GST_PULSERING_BUFFER_CAST (buf);
931 GST_LOG_OBJECT (psink, "creating sample spec");
932 /* convert the gstreamer sample spec to the pulseaudio format */
933 if (!gst_pulse_fill_format_info (spec, &pbuf->format, &pbuf->channels))
935 pbuf->is_pcm = pa_format_info_is_pcm (pbuf->format);
937 pa_threaded_mainloop_lock (mainloop);
939 /* we need a context and a no stream */
940 g_assert (pbuf->context);
941 g_assert (!pbuf->stream);
943 /* if we have a probe, disconnect it first so that if we're creating a
944 * compressed stream, it doesn't get blocked by a PCM stream */
945 if (pbuf->probe_stream) {
946 gst_pulse_destroy_stream (pbuf->probe_stream, TRUE);
947 pbuf->probe_stream = NULL;
950 /* enable event notifications */
951 GST_LOG_OBJECT (psink, "subscribing to context events");
952 if (!(o = pa_context_subscribe (pbuf->context,
953 PA_SUBSCRIPTION_MASK_SINK_INPUT, NULL, NULL)))
954 goto subscribe_failed;
956 pa_operation_unref (o);
958 /* initialize the channel map */
959 if (pbuf->is_pcm && gst_pulse_gst_to_channel_map (&channel_map, spec))
960 pa_format_info_set_channel_map (pbuf->format, &channel_map);
962 /* find a good name for the stream */
963 if (psink->stream_name)
964 name = psink->stream_name;
966 name = "Playback Stream";
968 #if defined(__TIZEN__) && defined(PCM_DUMP_ENABLE)
969 if (psink->need_dump_input == TRUE && psink->dump_fd_input == NULL) {
970 char *suffix , *dump_path;
971 GDateTime *time = g_date_time_new_now_local();
973 suffix = g_date_time_format(time, "%m%d_%H%M%S");
974 dump_path = g_strdup_printf("%s_%dch_%dhz_%s.pcm", GST_PULSESINK_DUMP_INPUT_PATH_PREFIX, pbuf->channels, spec->rate, suffix);
976 psink->dump_fd_input = fopen(dump_path, "w+");
980 g_date_time_unref(time);
982 #endif /* __TIZEN__ && PCM_DUMP_ENABLE */
984 /* create a stream */
985 formats[0] = pbuf->format;
986 if (!(pbuf->stream = pa_stream_new_extended (pbuf->context, name, formats, 1,
990 /* install essential callbacks */
991 pa_stream_set_state_callback (pbuf->stream,
992 gst_pulsering_stream_state_cb, pbuf);
993 pa_stream_set_write_callback (pbuf->stream,
994 gst_pulsering_stream_request_cb, pbuf);
995 pa_stream_set_underflow_callback (pbuf->stream,
996 gst_pulsering_stream_underflow_cb, pbuf);
997 pa_stream_set_overflow_callback (pbuf->stream,
998 gst_pulsering_stream_overflow_cb, pbuf);
999 pa_stream_set_latency_update_callback (pbuf->stream,
1000 gst_pulsering_stream_latency_cb, pbuf);
1001 pa_stream_set_suspended_callback (pbuf->stream,
1002 gst_pulsering_stream_suspended_cb, pbuf);
1003 pa_stream_set_started_callback (pbuf->stream,
1004 gst_pulsering_stream_started_cb, pbuf);
1005 pa_stream_set_event_callback (pbuf->stream,
1006 gst_pulsering_stream_event_cb, pbuf);
1008 /* buffering requirements. When setting prebuf to 0, the stream will not pause
1009 * when we cause an underrun, which causes time to continue. */
1010 memset (&wanted, 0, sizeof (wanted));
1011 wanted.tlength = spec->segtotal * spec->segsize;
1012 wanted.maxlength = -1;
1014 wanted.minreq = spec->segsize;
1016 GST_INFO_OBJECT (psink, "tlength: %d", wanted.tlength);
1017 GST_INFO_OBJECT (psink, "maxlength: %d", wanted.maxlength);
1018 GST_INFO_OBJECT (psink, "prebuf: %d", wanted.prebuf);
1019 GST_INFO_OBJECT (psink, "minreq: %d", wanted.minreq);
1021 /* configure volume when we changed it, else we leave the default */
1022 if (psink->volume_set) {
1023 GST_LOG_OBJECT (psink, "have volume of %f", psink->volume);
1026 gst_pulse_cvolume_from_linear (pv, pbuf->channels, psink->volume);
1028 GST_DEBUG_OBJECT (psink, "passthrough stream, not setting volume");
1035 /* construct the flags */
1036 flags = PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE |
1037 PA_STREAM_ADJUST_LATENCY | PA_STREAM_START_CORKED;
1039 if (psink->mute_set) {
1041 flags |= PA_STREAM_START_MUTED;
1043 flags |= PA_STREAM_START_UNMUTED;
1046 /* we always start corked (see flags above) */
1047 pbuf->corked = TRUE;
1049 /* try to connect now */
1050 GST_LOG_OBJECT (psink, "connect for playback to device %s",
1051 GST_STR_NULL (psink->device));
1052 if (pa_stream_connect_playback (pbuf->stream, psink->device,
1053 &wanted, flags, pv, NULL) < 0)
1054 goto connect_failed;
1056 /* our clock will now start from 0 again */
1057 clock = GST_AUDIO_CLOCK (GST_AUDIO_BASE_SINK (psink)->provided_clock);
1058 gst_audio_clock_reset (clock, 0);
1060 if (!gst_pulsering_wait_for_stream_ready (psink, pbuf->stream))
1061 goto connect_failed;
1063 g_free (psink->device);
1064 psink->device = g_strdup (pa_stream_get_device_name (pbuf->stream));
1066 #ifndef GST_DISABLE_GST_DEBUG
1067 pa_format_info_snprint (print_buf, sizeof (print_buf),
1068 pa_stream_get_format_info (pbuf->stream));
1069 GST_INFO_OBJECT (psink, "negotiated to: %s", print_buf);
1072 /* After we passed the volume off of to PA we never want to set it
1073 again, since it is PA's job to save/restore volumes. */
1074 psink->volume_set = psink->mute_set = FALSE;
1076 GST_LOG_OBJECT (psink, "stream is acquired now");
1078 /* get the actual buffering properties now */
1079 actual = pa_stream_get_buffer_attr (pbuf->stream);
1081 GST_INFO_OBJECT (psink, "tlength: %d (wanted: %d)", actual->tlength,
1083 GST_INFO_OBJECT (psink, "maxlength: %d", actual->maxlength);
1084 GST_INFO_OBJECT (psink, "prebuf: %d", actual->prebuf);
1085 GST_INFO_OBJECT (psink, "minreq: %d (wanted %d)", actual->minreq,
1088 spec->segsize = actual->minreq;
1089 spec->segtotal = actual->tlength / spec->segsize;
1091 pa_threaded_mainloop_unlock (mainloop);
1098 gst_pulsering_destroy_stream (pbuf);
1099 pa_threaded_mainloop_unlock (mainloop);
1105 GST_ELEMENT_ERROR (psink, RESOURCE, SETTINGS,
1106 ("Invalid sample specification."), (NULL));
1111 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
1112 ("pa_context_subscribe() failed: %s",
1113 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
1114 goto unlock_and_fail;
1118 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
1119 ("Failed to create stream: %s",
1120 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
1121 goto unlock_and_fail;
1125 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
1126 ("Failed to connect stream: %s",
1127 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
1128 goto unlock_and_fail;
1132 /* free the stream that we acquired before */
1134 gst_pulseringbuffer_release (GstAudioRingBuffer * buf)
1136 GstPulseRingBuffer *pbuf;
1138 pbuf = GST_PULSERING_BUFFER_CAST (buf);
1140 pa_threaded_mainloop_lock (mainloop);
1141 gst_pulsering_destroy_stream (pbuf);
1142 pa_threaded_mainloop_unlock (mainloop);
1145 GstPulseSink *psink;
1147 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
1148 g_atomic_int_set (&psink->format_lost, FALSE);
1149 psink->format_lost_time = GST_CLOCK_TIME_NONE;
1156 gst_pulsering_success_cb (pa_stream * s, int success, void *userdata)
1158 pa_threaded_mainloop_signal (mainloop, 0);
1161 /* update the corked state of a stream, must be called with the mainloop
1164 gst_pulsering_set_corked (GstPulseRingBuffer * pbuf, gboolean corked,
1167 pa_operation *o = NULL;
1168 GstPulseSink *psink;
1169 gboolean res = FALSE;
1171 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
1173 if (g_atomic_int_get (&psink->format_lost)) {
1174 /* Sink format changed, stream's gone so fake being paused */
1178 GST_DEBUG_OBJECT (psink, "setting corked state to %d", corked);
1179 if (pbuf->corked != corked) {
1180 if (!(o = pa_stream_cork (pbuf->stream, corked,
1181 gst_pulsering_success_cb, pbuf)))
1184 while (wait && pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
1185 pa_threaded_mainloop_wait (mainloop);
1186 if (gst_pulsering_is_dead (psink, pbuf, TRUE))
1189 pbuf->corked = corked;
1191 GST_DEBUG_OBJECT (psink, "skipping, already in requested state");
1197 pa_operation_unref (o);
1204 GST_DEBUG_OBJECT (psink, "the server is dead");
1209 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
1210 ("pa_stream_cork() failed: %s",
1211 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
1217 gst_pulseringbuffer_clear (GstAudioRingBuffer * buf)
1219 GstPulseSink *psink;
1220 GstPulseRingBuffer *pbuf;
1221 pa_operation *o = NULL;
1223 pbuf = GST_PULSERING_BUFFER_CAST (buf);
1224 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
1226 pa_threaded_mainloop_lock (mainloop);
1227 GST_DEBUG_OBJECT (psink, "clearing");
1229 /* don't wait for the flush to complete */
1230 if ((o = pa_stream_flush (pbuf->stream, NULL, pbuf)))
1231 pa_operation_unref (o);
1233 pa_threaded_mainloop_unlock (mainloop);
1237 /* called from pulse thread with the mainloop lock */
1239 mainloop_enter_defer_cb (pa_mainloop_api * api, void *userdata)
1241 GstPulseSink *pulsesink = GST_PULSESINK (userdata);
1242 GstMessage *message;
1245 GST_DEBUG_OBJECT (pulsesink, "posting ENTER stream status");
1246 message = gst_message_new_stream_status (GST_OBJECT (pulsesink),
1247 GST_STREAM_STATUS_TYPE_ENTER, GST_ELEMENT (pulsesink));
1248 g_value_init (&val, GST_TYPE_G_THREAD);
1249 g_value_set_boxed (&val, g_thread_self ());
1250 gst_message_set_stream_status_object (message, &val);
1251 g_value_unset (&val);
1253 gst_element_post_message (GST_ELEMENT (pulsesink), message);
1255 g_return_if_fail (pulsesink->defer_pending);
1256 pulsesink->defer_pending--;
1257 pa_threaded_mainloop_signal (mainloop, 0);
1261 /* start/resume playback ASAP, we don't uncork here but in the commit method */
1263 gst_pulseringbuffer_start (GstAudioRingBuffer * buf)
1265 GstPulseSink *psink;
1266 GstPulseRingBuffer *pbuf;
1268 pbuf = GST_PULSERING_BUFFER_CAST (buf);
1269 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
1271 pa_threaded_mainloop_lock (mainloop);
1273 GST_DEBUG_OBJECT (psink, "starting");
1274 pbuf->paused = FALSE;
1276 /* EOS needs running clock */
1277 if (GST_BASE_SINK_CAST (psink)->eos ||
1278 g_atomic_int_get (&GST_AUDIO_BASE_SINK (psink)->eos_rendering))
1279 gst_pulsering_set_corked (pbuf, FALSE, FALSE);
1282 GST_DEBUG_OBJECT (psink, "scheduling stream status");
1283 psink->defer_pending++;
1284 pa_mainloop_api_once (pa_threaded_mainloop_get_api (mainloop),
1285 mainloop_enter_defer_cb, psink);
1287 /* Wait for the stream status message to be posted. This needs to be done
1288 * synchronously because the callback will take the mainloop lock
1289 * (implicitly) and then take the GST_OBJECT_LOCK. Everywhere else, we take
1290 * the locks in the reverse order, so not doing this synchronously could
1291 * cause a deadlock. */
1292 GST_DEBUG_OBJECT (psink, "waiting for stream status (ENTER) to be posted");
1293 pa_threaded_mainloop_wait (mainloop);
1296 pa_threaded_mainloop_unlock (mainloop);
1301 /* pause/stop playback ASAP */
1303 gst_pulseringbuffer_pause (GstAudioRingBuffer * buf)
1305 GstPulseSink *psink;
1306 GstPulseRingBuffer *pbuf;
1309 pbuf = GST_PULSERING_BUFFER_CAST (buf);
1310 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
1312 pa_threaded_mainloop_lock (mainloop);
1313 GST_DEBUG_OBJECT (psink, "pausing and corking");
1314 /* make sure the commit method stops writing */
1315 pbuf->paused = TRUE;
1316 res = gst_pulsering_set_corked (pbuf, TRUE, TRUE);
1317 if (pbuf->in_commit) {
1318 /* we are waiting in a commit, signal */
1319 GST_DEBUG_OBJECT (psink, "signal commit");
1320 pa_threaded_mainloop_signal (mainloop, 0);
1322 pa_threaded_mainloop_unlock (mainloop);
1328 /* called from pulse thread with the mainloop lock */
1330 mainloop_leave_defer_cb (pa_mainloop_api * api, void *userdata)
1332 GstPulseSink *pulsesink = GST_PULSESINK (userdata);
1333 GstMessage *message;
1336 GST_DEBUG_OBJECT (pulsesink, "posting LEAVE stream status");
1337 message = gst_message_new_stream_status (GST_OBJECT (pulsesink),
1338 GST_STREAM_STATUS_TYPE_LEAVE, GST_ELEMENT (pulsesink));
1339 g_value_init (&val, GST_TYPE_G_THREAD);
1340 g_value_set_boxed (&val, g_thread_self ());
1341 gst_message_set_stream_status_object (message, &val);
1342 g_value_unset (&val);
1344 gst_element_post_message (GST_ELEMENT (pulsesink), message);
1346 g_return_if_fail (pulsesink->defer_pending);
1347 pulsesink->defer_pending--;
1348 pa_threaded_mainloop_signal (mainloop, 0);
1352 /* stop playback, we flush everything. */
1354 gst_pulseringbuffer_stop (GstAudioRingBuffer * buf)
1356 GstPulseSink *psink;
1357 GstPulseRingBuffer *pbuf;
1358 gboolean res = FALSE;
1359 pa_operation *o = NULL;
1361 pbuf = GST_PULSERING_BUFFER_CAST (buf);
1362 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
1364 pa_threaded_mainloop_lock (mainloop);
1366 pbuf->paused = TRUE;
1367 res = gst_pulsering_set_corked (pbuf, TRUE, TRUE);
1369 /* Inform anyone waiting in _commit() call that it shall wakeup */
1370 if (pbuf->in_commit) {
1371 GST_DEBUG_OBJECT (psink, "signal commit thread");
1372 pa_threaded_mainloop_signal (mainloop, 0);
1374 if (g_atomic_int_get (&psink->format_lost)) {
1375 /* Don't try to flush, the stream's probably gone by now */
1380 /* then try to flush, it's not fatal when this fails */
1381 GST_DEBUG_OBJECT (psink, "flushing");
1382 if ((o = pa_stream_flush (pbuf->stream, gst_pulsering_success_cb, pbuf))) {
1383 while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
1384 GST_DEBUG_OBJECT (psink, "wait for completion");
1385 pa_threaded_mainloop_wait (mainloop);
1386 if (gst_pulsering_is_dead (psink, pbuf, TRUE))
1389 GST_DEBUG_OBJECT (psink, "flush completed");
1395 pa_operation_cancel (o);
1396 pa_operation_unref (o);
1399 GST_DEBUG_OBJECT (psink, "scheduling stream status");
1400 psink->defer_pending++;
1401 pa_mainloop_api_once (pa_threaded_mainloop_get_api (mainloop),
1402 mainloop_leave_defer_cb, psink);
1404 /* Wait for the stream status message to be posted. This needs to be done
1405 * synchronously because the callback will take the mainloop lock
1406 * (implicitly) and then take the GST_OBJECT_LOCK. Everywhere else, we take
1407 * the locks in the reverse order, so not doing this synchronously could
1408 * cause a deadlock. */
1409 GST_DEBUG_OBJECT (psink, "waiting for stream status (LEAVE) to be posted");
1410 pa_threaded_mainloop_wait (mainloop);
1413 pa_threaded_mainloop_unlock (mainloop);
1420 GST_DEBUG_OBJECT (psink, "the server is dead");
1425 /* in_samples >= out_samples, rate > 1.0 */
1426 #define FWD_UP_SAMPLES(s,se,d,de) \
1428 guint8 *sb = s, *db = d; \
1429 while (s <= se && d < de) { \
1430 memcpy (d, s, bpf); \
1433 if ((*accum << 1) >= inr) { \
1438 in_samples -= (s - sb)/bpf; \
1439 out_samples -= (d - db)/bpf; \
1440 GST_DEBUG ("fwd_up end %d/%d",*accum,*toprocess); \
1443 /* out_samples > in_samples, for rates smaller than 1.0 */
1444 #define FWD_DOWN_SAMPLES(s,se,d,de) \
1446 guint8 *sb = s, *db = d; \
1447 while (s <= se && d < de) { \
1448 memcpy (d, s, bpf); \
1451 if ((*accum << 1) >= outr) { \
1456 in_samples -= (s - sb)/bpf; \
1457 out_samples -= (d - db)/bpf; \
1458 GST_DEBUG ("fwd_down end %d/%d",*accum,*toprocess); \
1461 #define REV_UP_SAMPLES(s,se,d,de) \
1463 guint8 *sb = se, *db = d; \
1464 while (s <= se && d < de) { \
1465 memcpy (d, se, bpf); \
1468 while (d < de && (*accum << 1) >= inr) { \
1473 in_samples -= (sb - se)/bpf; \
1474 out_samples -= (d - db)/bpf; \
1475 GST_DEBUG ("rev_up end %d/%d",*accum,*toprocess); \
1478 #define REV_DOWN_SAMPLES(s,se,d,de) \
1480 guint8 *sb = se, *db = d; \
1481 while (s <= se && d < de) { \
1482 memcpy (d, se, bpf); \
1485 while (s <= se && (*accum << 1) >= outr) { \
1490 in_samples -= (sb - se)/bpf; \
1491 out_samples -= (d - db)/bpf; \
1492 GST_DEBUG ("rev_down end %d/%d",*accum,*toprocess); \
1495 /* our custom commit function because we write into the buffer of pulseaudio
1496 * instead of keeping our own buffer */
1498 gst_pulseringbuffer_commit (GstAudioRingBuffer * buf, guint64 * sample,
1499 guchar * data, gint in_samples, gint out_samples, gint * accum)
1501 GstPulseSink *psink;
1502 GstPulseRingBuffer *pbuf;
1507 gint inr, outr, bpf;
1511 pbuf = GST_PULSERING_BUFFER_CAST (buf);
1512 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
1514 /* FIXME post message rather than using a signal (as mixer interface) */
1515 if (g_atomic_int_compare_and_exchange (&psink->notify, 1, 0)) {
1516 g_object_notify (G_OBJECT (psink), "volume");
1517 g_object_notify (G_OBJECT (psink), "mute");
1518 g_object_notify (G_OBJECT (psink), "current-device");
1521 /* make sure the ringbuffer is started */
1522 if (G_UNLIKELY (g_atomic_int_get (&buf->state) !=
1523 GST_AUDIO_RING_BUFFER_STATE_STARTED)) {
1524 /* see if we are allowed to start it */
1525 if (G_UNLIKELY (g_atomic_int_get (&buf->may_start) == FALSE))
1528 GST_DEBUG_OBJECT (buf, "start!");
1529 if (!gst_audio_ring_buffer_start (buf))
1533 pa_threaded_mainloop_lock (mainloop);
1535 GST_DEBUG_OBJECT (psink, "entering commit");
1536 pbuf->in_commit = TRUE;
1538 bpf = GST_AUDIO_INFO_BPF (&buf->spec.info);
1539 bufsize = buf->spec.segsize * buf->spec.segtotal;
1541 /* our toy resampler for trick modes */
1542 reverse = out_samples < 0;
1543 out_samples = ABS (out_samples);
1545 if (in_samples >= out_samples)
1546 toprocess = &in_samples;
1548 toprocess = &out_samples;
1550 inr = in_samples - 1;
1551 outr = out_samples - 1;
1553 GST_DEBUG_OBJECT (psink, "in %d, out %d", inr, outr);
1555 /* data_end points to the last sample we have to write, not past it. This is
1556 * needed to properly handle reverse playback: it points to the last sample. */
1557 data_end = data + (bpf * inr);
1559 if (g_atomic_int_get (&psink->format_lost)) {
1560 /* Sink format changed, drop the data and hope upstream renegotiates */
1567 /* offset is in bytes */
1568 offset = *sample * bpf;
1570 while (*toprocess > 0) {
1574 GST_LOG_OBJECT (psink,
1575 "need to write %d samples at offset %" G_GINT64_FORMAT, *toprocess,
1578 if (offset != pbuf->m_lastoffset)
1579 GST_LOG_OBJECT (psink, "discontinuity, offset is %" G_GINT64_FORMAT ", "
1580 "last offset was %" G_GINT64_FORMAT, offset, pbuf->m_lastoffset);
1582 towrite = out_samples * bpf;
1584 /* Wait for at least segsize bytes to become available */
1585 if (towrite > buf->spec.segsize)
1586 towrite = buf->spec.segsize;
1588 if ((pbuf->m_writable < towrite) || (offset != pbuf->m_lastoffset)) {
1589 /* if no room left or discontinuity in offset,
1590 we need to flush data and get a new buffer */
1592 /* flush the buffer if possible */
1593 if ((pbuf->m_data != NULL) && (pbuf->m_towrite > 0)) {
1595 GST_LOG_OBJECT (psink,
1596 "flushing %u samples at offset %" G_GINT64_FORMAT,
1597 (guint) pbuf->m_towrite / bpf, pbuf->m_offset);
1599 if (pa_stream_write (pbuf->stream, (uint8_t *) pbuf->m_data,
1600 pbuf->m_towrite, NULL, pbuf->m_offset, PA_SEEK_ABSOLUTE) < 0) {
1604 pbuf->m_towrite = 0;
1605 pbuf->m_offset = offset; /* keep track of current offset */
1607 /* get a buffer to write in for now on */
1609 pbuf->m_writable = pa_stream_writable_size (pbuf->stream);
1611 if (g_atomic_int_get (&psink->format_lost)) {
1612 /* Sink format changed, give up and hope upstream renegotiates */
1616 if (pbuf->m_writable == (size_t) - 1)
1617 goto writable_size_failed;
1619 pbuf->m_writable /= bpf;
1620 pbuf->m_writable *= bpf; /* handle only complete samples */
1622 if (pbuf->m_writable >= towrite)
1625 /* see if we need to uncork because we have no free space */
1627 if (!gst_pulsering_set_corked (pbuf, FALSE, FALSE))
1631 /* we can't write segsize bytes, wait a bit */
1632 GST_LOG_OBJECT (psink, "waiting for free space");
1633 pa_threaded_mainloop_wait (mainloop);
1639 /* Recalculate what we can write in the next chunk */
1640 towrite = out_samples * bpf;
1641 if (pbuf->m_writable > towrite)
1642 pbuf->m_writable = towrite;
1644 GST_LOG_OBJECT (psink, "requesting %" G_GSIZE_FORMAT " bytes of "
1645 "shared memory", pbuf->m_writable);
1647 if (pa_stream_begin_write (pbuf->stream, &pbuf->m_data,
1648 &pbuf->m_writable) < 0) {
1649 GST_LOG_OBJECT (psink, "pa_stream_begin_write() failed");
1650 goto writable_size_failed;
1653 GST_LOG_OBJECT (psink, "got %" G_GSIZE_FORMAT " bytes of shared memory",
1658 if (towrite > pbuf->m_writable)
1659 towrite = pbuf->m_writable;
1660 avail = towrite / bpf;
1662 GST_LOG_OBJECT (psink, "writing %u samples at offset %" G_GUINT64_FORMAT,
1663 (guint) avail, offset);
1665 /* No trick modes for passthrough streams */
1666 if (G_UNLIKELY (!pbuf->is_pcm && (inr != outr || reverse))) {
1667 GST_WARNING_OBJECT (psink, "Passthrough stream can't run in trick mode");
1668 goto unlock_and_fail;
1671 if (G_LIKELY (inr == outr && !reverse)) {
1672 /* no rate conversion, simply write out the samples */
1673 /* copy the data into internal buffer */
1675 memcpy ((guint8 *) pbuf->m_data + pbuf->m_towrite, data, towrite);
1676 pbuf->m_towrite += towrite;
1677 pbuf->m_writable -= towrite;
1680 in_samples -= avail;
1681 out_samples -= avail;
1683 guint8 *dest, *d, *d_end;
1685 /* write into the PulseAudio shm buffer */
1686 dest = d = (guint8 *) pbuf->m_data + pbuf->m_towrite;
1687 d_end = d + towrite;
1691 /* forward speed up */
1692 FWD_UP_SAMPLES (data, data_end, d, d_end);
1694 /* forward slow down */
1695 FWD_DOWN_SAMPLES (data, data_end, d, d_end);
1698 /* reverse speed up */
1699 REV_UP_SAMPLES (data, data_end, d, d_end);
1701 /* reverse slow down */
1702 REV_DOWN_SAMPLES (data, data_end, d, d_end);
1704 /* see what we have left to write */
1705 towrite = (d - dest);
1706 pbuf->m_towrite += towrite;
1707 pbuf->m_writable -= towrite;
1709 avail = towrite / bpf;
1712 /* flush the buffer if it's full */
1713 if ((pbuf->m_data != NULL) && (pbuf->m_towrite > 0)
1714 && (pbuf->m_writable == 0)) {
1715 GST_LOG_OBJECT (psink, "flushing %u samples at offset %" G_GINT64_FORMAT,
1716 (guint) pbuf->m_towrite / bpf, pbuf->m_offset);
1718 if (pa_stream_write (pbuf->stream, (uint8_t *) pbuf->m_data,
1719 pbuf->m_towrite, NULL, pbuf->m_offset, PA_SEEK_ABSOLUTE) < 0) {
1722 pbuf->m_towrite = 0;
1723 pbuf->m_offset = offset + towrite; /* keep track of current offset */
1727 offset += avail * bpf;
1728 pbuf->m_lastoffset = offset;
1730 /* check if we need to uncork after writing the samples */
1732 const pa_timing_info *info;
1734 if ((info = pa_stream_get_timing_info (pbuf->stream))) {
1735 GST_LOG_OBJECT (psink,
1736 "read_index at %" G_GUINT64_FORMAT ", offset %" G_GINT64_FORMAT,
1737 info->read_index, offset);
1739 /* we uncork when the read_index is too far behind the offset we need
1741 if (info->read_index + bufsize <= offset) {
1742 if (!gst_pulsering_set_corked (pbuf, FALSE, FALSE))
1746 GST_LOG_OBJECT (psink, "no timing info available yet");
1752 /* we consumed all samples here */
1753 data = data_end + bpf;
1755 pbuf->in_commit = FALSE;
1756 pa_threaded_mainloop_unlock (mainloop);
1759 result = inr - ((data_end - data) / bpf);
1760 GST_LOG_OBJECT (psink, "wrote %d samples", result);
1767 pbuf->in_commit = FALSE;
1768 GST_LOG_OBJECT (psink, "we are reset");
1769 pa_threaded_mainloop_unlock (mainloop);
1774 GST_LOG_OBJECT (psink, "we can not start");
1779 GST_LOG_OBJECT (psink, "failed to start the ringbuffer");
1784 pbuf->in_commit = FALSE;
1785 GST_ERROR_OBJECT (psink, "uncork failed");
1786 pa_threaded_mainloop_unlock (mainloop);
1791 pbuf->in_commit = FALSE;
1792 GST_LOG_OBJECT (psink, "we are paused");
1793 pa_threaded_mainloop_unlock (mainloop);
1796 writable_size_failed:
1798 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
1799 ("pa_stream_writable_size() failed: %s",
1800 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
1801 goto unlock_and_fail;
1805 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
1806 ("pa_stream_write() failed: %s",
1807 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
1808 goto unlock_and_fail;
1812 /* write pending local samples, must be called with the mainloop lock */
1814 gst_pulsering_flush (GstPulseRingBuffer * pbuf)
1816 GstPulseSink *psink;
1818 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
1819 GST_DEBUG_OBJECT (psink, "entering flush");
1821 /* flush the buffer if possible */
1822 if (pbuf->stream && (pbuf->m_data != NULL) && (pbuf->m_towrite > 0)) {
1823 #ifndef GST_DISABLE_GST_DEBUG
1826 bpf = (GST_AUDIO_RING_BUFFER_CAST (pbuf))->spec.info.bpf;
1827 GST_LOG_OBJECT (psink,
1828 "flushing %u samples at offset %" G_GINT64_FORMAT,
1829 (guint) pbuf->m_towrite / bpf, pbuf->m_offset);
1832 if (pa_stream_write (pbuf->stream, (uint8_t *) pbuf->m_data,
1833 pbuf->m_towrite, NULL, pbuf->m_offset, PA_SEEK_ABSOLUTE) < 0) {
1837 pbuf->m_towrite = 0;
1838 pbuf->m_offset += pbuf->m_towrite; /* keep track of current offset */
1847 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
1848 ("pa_stream_write() failed: %s",
1849 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
1854 static void gst_pulsesink_set_property (GObject * object, guint prop_id,
1855 const GValue * value, GParamSpec * pspec);
1856 static void gst_pulsesink_get_property (GObject * object, guint prop_id,
1857 GValue * value, GParamSpec * pspec);
1858 static void gst_pulsesink_finalize (GObject * object);
1860 static gboolean gst_pulsesink_event (GstBaseSink * sink, GstEvent * event);
1861 static gboolean gst_pulsesink_query (GstBaseSink * sink, GstQuery * query);
1863 static GstStateChangeReturn gst_pulsesink_change_state (GstElement * element,
1864 GstStateChange transition);
1866 static GstStaticPadTemplate pad_template = GST_STATIC_PAD_TEMPLATE ("sink",
1869 GST_STATIC_CAPS (PULSE_SINK_TEMPLATE_CAPS));
1871 #define gst_pulsesink_parent_class parent_class
1872 G_DEFINE_TYPE_WITH_CODE (GstPulseSink, gst_pulsesink, GST_TYPE_AUDIO_BASE_SINK,
1873 gst_pulsesink_init_contexts ();
1874 G_IMPLEMENT_INTERFACE (GST_TYPE_STREAM_VOLUME, NULL)
1877 static GstAudioRingBuffer *
1878 gst_pulsesink_create_ringbuffer (GstAudioBaseSink * sink)
1880 GstAudioRingBuffer *buffer;
1882 GST_DEBUG_OBJECT (sink, "creating ringbuffer");
1883 buffer = g_object_new (GST_TYPE_PULSERING_BUFFER, NULL);
1884 GST_DEBUG_OBJECT (sink, "created ringbuffer @%p", buffer);
1890 gst_pulsesink_payload (GstAudioBaseSink * sink, GstBuffer * buf)
1892 switch (sink->ringbuffer->spec.type) {
1893 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3:
1894 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3:
1895 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS:
1896 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG:
1897 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC:
1898 case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC:
1900 /* FIXME: alloc memory from PA if possible */
1901 gint framesize = gst_audio_iec61937_frame_size (&sink->ringbuffer->spec);
1903 GstMapInfo inmap, outmap;
1909 out = gst_buffer_new_and_alloc (framesize);
1911 gst_buffer_map (buf, &inmap, GST_MAP_READ);
1912 gst_buffer_map (out, &outmap, GST_MAP_WRITE);
1914 res = gst_audio_iec61937_payload (inmap.data, inmap.size,
1915 outmap.data, outmap.size, &sink->ringbuffer->spec, G_BIG_ENDIAN);
1917 gst_buffer_unmap (buf, &inmap);
1918 gst_buffer_unmap (out, &outmap);
1921 gst_buffer_unref (out);
1925 gst_buffer_copy_into (out, buf, GST_BUFFER_COPY_METADATA, 0, -1);
1930 return gst_buffer_ref (buf);
1934 #if defined(__TIZEN__) && defined(PCM_DUMP_ENABLE)
1936 gst_pulsesink_pad_dump_handler (GstPad *pad, GstBuffer *buffer, gpointer data)
1938 GstPulseSink *psink = GST_PULSESINK_CAST (data);
1941 if (psink->dump_fd_input)
1942 ret = fwrite(GST_BUFFER_DATA(buffer), 1, GST_BUFFER_SIZE(buffer), psink->dump_fd_input);
1946 #endif /* __TIZEN__ && PCM_DUMP_ENABLE */
1949 gst_pulsesink_class_init (GstPulseSinkClass * klass)
1951 GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
1952 GstBaseSinkClass *gstbasesink_class = GST_BASE_SINK_CLASS (klass);
1953 GstBaseSinkClass *bc;
1954 GstAudioBaseSinkClass *gstaudiosink_class = GST_AUDIO_BASE_SINK_CLASS (klass);
1955 GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
1958 gobject_class->finalize = gst_pulsesink_finalize;
1959 gobject_class->set_property = gst_pulsesink_set_property;
1960 gobject_class->get_property = gst_pulsesink_get_property;
1962 gstbasesink_class->event = GST_DEBUG_FUNCPTR (gst_pulsesink_event);
1963 gstbasesink_class->query = GST_DEBUG_FUNCPTR (gst_pulsesink_query);
1965 /* restore the original basesink pull methods */
1966 bc = g_type_class_peek (GST_TYPE_BASE_SINK);
1967 gstbasesink_class->activate_pull = GST_DEBUG_FUNCPTR (bc->activate_pull);
1969 gstelement_class->change_state =
1970 GST_DEBUG_FUNCPTR (gst_pulsesink_change_state);
1972 gstaudiosink_class->create_ringbuffer =
1973 GST_DEBUG_FUNCPTR (gst_pulsesink_create_ringbuffer);
1974 gstaudiosink_class->payload = GST_DEBUG_FUNCPTR (gst_pulsesink_payload);
1976 /* Overwrite GObject fields */
1977 g_object_class_install_property (gobject_class,
1979 g_param_spec_string ("server", "Server",
1980 "The PulseAudio server to connect to", DEFAULT_SERVER,
1981 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
1983 g_object_class_install_property (gobject_class, PROP_DEVICE,
1984 g_param_spec_string ("device", "Device",
1985 "The PulseAudio sink device to connect to", DEFAULT_DEVICE,
1986 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
1988 g_object_class_install_property (gobject_class, PROP_CURRENT_DEVICE,
1989 g_param_spec_string ("current-device", "Current Device",
1990 "The current PulseAudio sink device", DEFAULT_CURRENT_DEVICE,
1991 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
1993 g_object_class_install_property (gobject_class,
1995 g_param_spec_string ("device-name", "Device name",
1996 "Human-readable name of the sound device", DEFAULT_DEVICE_NAME,
1997 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
1999 g_object_class_install_property (gobject_class,
2001 g_param_spec_double ("volume", "Volume",
2002 "Linear volume of this stream, 1.0=100%", 0.0, MAX_VOLUME,
2003 DEFAULT_VOLUME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2004 g_object_class_install_property (gobject_class,
2006 g_param_spec_boolean ("mute", "Mute",
2007 "Mute state of this stream", DEFAULT_MUTE,
2008 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2011 * GstPulseSink:client-name:
2013 * The PulseAudio client name to use.
2015 clientname = gst_pulse_client_name ();
2016 g_object_class_install_property (gobject_class,
2018 g_param_spec_string ("client-name", "Client Name",
2019 "The PulseAudio client name to use", clientname,
2020 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
2021 GST_PARAM_MUTABLE_READY));
2022 g_free (clientname);
2025 * GstPulseSink:stream-properties:
2027 * List of pulseaudio stream properties. A list of defined properties can be
2028 * found in the <ulink url="http://0pointer.de/lennart/projects/pulseaudio/doxygen/proplist_8h.html">pulseaudio api docs</ulink>.
2030 * Below is an example for registering as a music application to pulseaudio.
2032 * GstStructure *props;
2034 * props = gst_structure_from_string ("props,media.role=music", NULL);
2035 * g_object_set (pulse, "stream-properties", props, NULL);
2036 * gst_structure_free
2039 g_object_class_install_property (gobject_class,
2040 PROP_STREAM_PROPERTIES,
2041 g_param_spec_boxed ("stream-properties", "stream properties",
2042 "list of pulseaudio stream properties",
2043 GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2046 g_object_class_install_property (gobject_class,
2048 g_param_spec_string ("latency", "Audio Backend Latency",
2049 "Audio Backend Latency (\"low\": Low Latency, \"mid\": Mid Latency, \"high\": High Latency)",
2050 DEFAULT_AUDIO_LATENCY,
2051 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2052 #endif /* __TIZEN__ */
2054 gst_element_class_set_static_metadata (gstelement_class,
2055 "PulseAudio Audio Sink",
2056 "Sink/Audio", "Plays audio to a PulseAudio server", "Lennart Poettering");
2057 gst_element_class_add_static_pad_template (gstelement_class, &pad_template);
2061 free_device_info (GstPulseDeviceInfo * device_info)
2065 g_free (device_info->description);
2067 for (l = g_list_first (device_info->formats); l; l = g_list_next (l))
2068 pa_format_info_free ((pa_format_info *) l->data);
2070 g_list_free (device_info->formats);
2073 /* Returns the current time of the sink ringbuffer. The timing_info is updated
2074 * on every data write/flush and every 100ms (PA_STREAM_AUTO_TIMING_UPDATE).
2077 gst_pulsesink_get_time (GstClock * clock, GstAudioBaseSink * sink)
2079 GstPulseSink *psink;
2080 GstPulseRingBuffer *pbuf;
2083 if (!sink->ringbuffer || !sink->ringbuffer->acquired)
2084 return GST_CLOCK_TIME_NONE;
2086 pbuf = GST_PULSERING_BUFFER_CAST (sink->ringbuffer);
2087 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
2089 if (g_atomic_int_get (&psink->format_lost)) {
2090 /* Stream was lost in a format change, it'll get set up again once
2091 * upstream renegotiates */
2092 return psink->format_lost_time;
2095 pa_threaded_mainloop_lock (mainloop);
2096 if (gst_pulsering_is_dead (psink, pbuf, TRUE))
2099 /* if we don't have enough data to get a timestamp, just return NONE, which
2100 * will return the last reported time */
2101 if (pa_stream_get_time (pbuf->stream, &time) < 0) {
2102 GST_DEBUG_OBJECT (psink, "could not get time");
2103 time = GST_CLOCK_TIME_NONE;
2106 pa_threaded_mainloop_unlock (mainloop);
2108 GST_LOG_OBJECT (psink, "current time is %" GST_TIME_FORMAT,
2109 GST_TIME_ARGS (time));
2116 GST_DEBUG_OBJECT (psink, "the server is dead");
2117 pa_threaded_mainloop_unlock (mainloop);
2119 return GST_CLOCK_TIME_NONE;
2124 gst_pulsesink_sink_info_cb (pa_context * c, const pa_sink_info * i, int eol,
2127 GstPulseDeviceInfo *device_info = (GstPulseDeviceInfo *) userdata;
2133 device_info->description = g_strdup (i->description);
2135 device_info->formats = NULL;
2136 for (j = 0; j < i->n_formats; j++)
2137 device_info->formats = g_list_prepend (device_info->formats,
2138 pa_format_info_copy (i->formats[j]));
2141 pa_threaded_mainloop_signal (mainloop, 0);
2144 /* Call with mainloop lock held */
2146 gst_pulsesink_create_probe_stream (GstPulseSink * psink,
2147 GstPulseRingBuffer * pbuf, pa_format_info * format)
2149 pa_format_info *formats[1] = { format };
2151 pa_stream_flags_t flags;
2153 GST_LOG_OBJECT (psink, "Creating probe stream");
2155 if (!(stream = pa_stream_new_extended (pbuf->context, "pulsesink probe",
2156 formats, 1, psink->proplist)))
2159 /* construct the flags */
2160 flags = PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE |
2161 PA_STREAM_ADJUST_LATENCY | PA_STREAM_START_CORKED;
2163 pa_stream_set_state_callback (stream, gst_pulsering_stream_state_cb, pbuf);
2165 if (pa_stream_connect_playback (stream, psink->device, NULL, flags, NULL,
2169 if (!gst_pulsering_wait_for_stream_ready (psink, stream))
2176 pa_stream_unref (stream);
2181 gst_pulsesink_query_getcaps (GstPulseSink * psink, GstCaps * filter)
2183 GstPulseRingBuffer *pbuf = NULL;
2184 GstPulseDeviceInfo device_info = { NULL, NULL };
2185 GstCaps *ret = NULL;
2187 pa_operation *o = NULL;
2190 GST_OBJECT_LOCK (psink);
2191 pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
2193 gst_object_ref (pbuf);
2194 GST_OBJECT_UNLOCK (psink);
2197 ret = gst_pad_get_pad_template_caps (GST_AUDIO_BASE_SINK_PAD (psink));
2201 GST_OBJECT_LOCK (pbuf);
2202 pa_threaded_mainloop_lock (mainloop);
2204 if (!pbuf->context) {
2205 ret = gst_pad_get_pad_template_caps (GST_AUDIO_BASE_SINK_PAD (psink));
2209 ret = gst_caps_new_empty ();
2212 /* We're in PAUSED or higher */
2213 stream = pbuf->stream;
2215 } else if (pbuf->probe_stream) {
2216 /* We're not paused, but have a cached probe stream */
2217 stream = pbuf->probe_stream;
2220 /* We're not yet in PAUSED and still need to create a probe stream.
2222 * FIXME: PA doesn't accept "any" format. We fix something reasonable since
2223 * this is merely a probe. This should eventually be fixed in PA and
2224 * hard-coding the format should be dropped. */
2225 pa_format_info *format = pa_format_info_new ();
2226 format->encoding = PA_ENCODING_PCM;
2227 pa_format_info_set_sample_format (format, PA_SAMPLE_S16LE);
2228 pa_format_info_set_rate (format, GST_AUDIO_DEF_RATE);
2229 pa_format_info_set_channels (format, GST_AUDIO_DEF_CHANNELS);
2231 pbuf->probe_stream = gst_pulsesink_create_probe_stream (psink, pbuf,
2234 pa_format_info_free (format);
2236 if (!pbuf->probe_stream) {
2237 GST_WARNING_OBJECT (psink, "Could not create probe stream");
2241 stream = pbuf->probe_stream;
2244 if (!(o = pa_context_get_sink_info_by_name (pbuf->context,
2245 pa_stream_get_device_name (stream), gst_pulsesink_sink_info_cb,
2249 while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
2250 pa_threaded_mainloop_wait (mainloop);
2251 if (gst_pulsering_is_dead (psink, pbuf, FALSE))
2255 for (i = g_list_first (device_info.formats); i; i = g_list_next (i)) {
2256 gst_caps_append (ret,
2257 gst_pulse_format_info_to_caps ((pa_format_info *) i->data));
2261 pa_threaded_mainloop_unlock (mainloop);
2262 /* FIXME: this could be freed after device_name is got */
2263 GST_OBJECT_UNLOCK (pbuf);
2266 GstCaps *tmp = gst_caps_intersect_full (filter, ret,
2267 GST_CAPS_INTERSECT_FIRST);
2268 gst_caps_unref (ret);
2273 free_device_info (&device_info);
2276 pa_operation_unref (o);
2279 gst_object_unref (pbuf);
2281 GST_DEBUG_OBJECT (psink, "caps %" GST_PTR_FORMAT, ret);
2287 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
2288 ("pa_context_get_sink_input_info() failed: %s",
2289 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
2295 gst_pulsesink_query_acceptcaps (GstPulseSink * psink, GstCaps * caps)
2297 GstPulseRingBuffer *pbuf = NULL;
2298 GstPulseDeviceInfo device_info = { NULL, NULL };
2301 gboolean ret = FALSE;
2303 GstAudioRingBufferSpec spec = { 0 };
2304 pa_operation *o = NULL;
2305 pa_channel_map channel_map;
2306 pa_format_info *format = NULL;
2309 pad_caps = gst_pad_get_pad_template_caps (GST_BASE_SINK_PAD (psink));
2310 ret = gst_caps_is_subset (caps, pad_caps);
2311 gst_caps_unref (pad_caps);
2313 GST_DEBUG_OBJECT (psink, "caps %" GST_PTR_FORMAT, caps);
2315 /* Template caps didn't match */
2319 /* If we've not got fixed caps, creating a stream might fail, so let's just
2320 * return from here with default acceptcaps behaviour */
2321 if (!gst_caps_is_fixed (caps))
2324 GST_OBJECT_LOCK (psink);
2325 pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
2327 gst_object_ref (pbuf);
2328 GST_OBJECT_UNLOCK (psink);
2330 /* We're still in NULL state */
2334 GST_OBJECT_LOCK (pbuf);
2335 pa_threaded_mainloop_lock (mainloop);
2337 if (pbuf->context == NULL)
2342 spec.latency_time = GST_AUDIO_BASE_SINK (psink)->latency_time;
2343 if (!gst_audio_ring_buffer_parse_caps (&spec, caps))
2346 if (!gst_pulse_fill_format_info (&spec, &format, &channels))
2349 /* Make sure input is framed (one frame per buffer) and can be payloaded */
2350 if (!pa_format_info_is_pcm (format)) {
2351 gboolean framed = FALSE, parsed = FALSE;
2352 st = gst_caps_get_structure (caps, 0);
2354 gst_structure_get_boolean (st, "framed", &framed);
2355 gst_structure_get_boolean (st, "parsed", &parsed);
2356 if ((!framed && !parsed) || gst_audio_iec61937_frame_size (&spec) <= 0)
2360 /* initialize the channel map */
2361 if (pa_format_info_is_pcm (format) &&
2362 gst_pulse_gst_to_channel_map (&channel_map, &spec))
2363 pa_format_info_set_channel_map (format, &channel_map);
2365 if (pbuf->stream || pbuf->probe_stream) {
2366 /* We're already in PAUSED or above, so just reuse this stream to query
2367 * sink formats and use those. */
2369 const char *device_name = pa_stream_get_device_name (pbuf->stream ?
2370 pbuf->stream : pbuf->probe_stream);
2372 if (!(o = pa_context_get_sink_info_by_name (pbuf->context, device_name,
2373 gst_pulsesink_sink_info_cb, &device_info)))
2376 while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
2377 pa_threaded_mainloop_wait (mainloop);
2378 if (gst_pulsering_is_dead (psink, pbuf, FALSE))
2382 for (i = g_list_first (device_info.formats); i; i = g_list_next (i)) {
2383 if (pa_format_info_is_compatible ((pa_format_info *) i->data, format)) {
2389 /* We're in READY, let's connect a stream to see if the format is
2390 * accepted by whatever sink we're routed to */
2391 pbuf->probe_stream = gst_pulsesink_create_probe_stream (psink, pbuf,
2393 if (pbuf->probe_stream)
2399 pa_format_info_free (format);
2401 free_device_info (&device_info);
2404 pa_operation_unref (o);
2406 pa_threaded_mainloop_unlock (mainloop);
2407 GST_OBJECT_UNLOCK (pbuf);
2409 gst_caps_replace (&spec.caps, NULL);
2410 gst_object_unref (pbuf);
2418 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
2419 ("pa_context_get_sink_input_info() failed: %s",
2420 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
2426 gst_pulsesink_init (GstPulseSink * pulsesink)
2428 #if defined(__TIZEN__) && defined(PCM_DUMP_ENABLE)
2429 GstPad *sinkpad = NULL;
2431 #endif /* __TIZEN__ && PCM_DUMP_ENABLE */
2433 pulsesink->server = NULL;
2434 pulsesink->device = NULL;
2435 pulsesink->device_info.description = NULL;
2436 pulsesink->client_name = gst_pulse_client_name ();
2438 pulsesink->device_info.formats = NULL;
2440 pulsesink->volume = DEFAULT_VOLUME;
2441 pulsesink->volume_set = FALSE;
2443 pulsesink->mute = DEFAULT_MUTE;
2444 pulsesink->mute_set = FALSE;
2446 pulsesink->notify = 0;
2448 g_atomic_int_set (&pulsesink->format_lost, FALSE);
2449 pulsesink->format_lost_time = GST_CLOCK_TIME_NONE;
2451 pulsesink->properties = NULL;
2452 pulsesink->proplist = NULL;
2454 pulsesink->latency = g_strdup (DEFAULT_AUDIO_LATENCY);
2455 pulsesink->proplist = pa_proplist_new();
2456 pa_proplist_sets(pulsesink->proplist, PA_PROP_MEDIA_TIZEN_AUDIO_LATENCY, pulsesink->latency);
2457 #ifdef PCM_DUMP_ENABLE
2458 if (vconf_get_int(GST_PULSESINK_DUMP_VCONF_KEY, &vconf_dump)) {
2459 GST_WARNING("vconf_get_int %s failed", GST_PULSESINK_DUMP_VCONF_KEY);
2461 pulsesink->need_dump_input = vconf_dump & GST_PULSESINK_DUMP_INPUT_FLAG ? TRUE : FALSE;
2462 pulsesink->dump_fd_input = NULL;
2463 if (pulsesink->need_dump_input) {
2464 sinkpad = gst_element_get_static_pad((GstElement *)pulsesink, "sink");
2466 gst_pad_add_buffer_probe (sinkpad, G_CALLBACK (gst_pulsesink_pad_dump_handler), pulsesink);
2467 gst_object_unref (GST_OBJECT(sinkpad));
2471 #endif /* __TIZEN__ */
2473 /* override with a custom clock */
2474 if (GST_AUDIO_BASE_SINK (pulsesink)->provided_clock)
2475 gst_object_unref (GST_AUDIO_BASE_SINK (pulsesink)->provided_clock);
2477 GST_AUDIO_BASE_SINK (pulsesink)->provided_clock =
2478 gst_audio_clock_new ("GstPulseSinkClock",
2479 (GstAudioClockGetTimeFunc) gst_pulsesink_get_time, pulsesink, NULL);
2483 gst_pulsesink_finalize (GObject * object)
2485 GstPulseSink *pulsesink = GST_PULSESINK_CAST (object);
2487 g_free (pulsesink->server);
2488 g_free (pulsesink->device);
2489 g_free (pulsesink->client_name);
2490 g_free (pulsesink->current_sink_name);
2492 free_device_info (&pulsesink->device_info);
2494 if (pulsesink->properties)
2495 gst_structure_free (pulsesink->properties);
2496 if (pulsesink->proplist)
2497 pa_proplist_free (pulsesink->proplist);
2500 g_free (pulsesink->latency);
2501 #endif /* __TIZEN__ */
2503 G_OBJECT_CLASS (parent_class)->finalize (object);
2507 gst_pulsesink_set_volume (GstPulseSink * psink, gdouble volume)
2510 pa_operation *o = NULL;
2511 GstPulseRingBuffer *pbuf;
2517 pa_threaded_mainloop_lock (mainloop);
2519 GST_DEBUG_OBJECT (psink, "setting volume to %f", volume);
2521 pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
2522 if (pbuf == NULL || pbuf->stream == NULL)
2525 if ((idx = pa_stream_get_index (pbuf->stream)) == PA_INVALID_INDEX)
2529 gst_pulse_cvolume_from_linear (&v, pbuf->channels, volume);
2531 /* FIXME: this will eventually be superceded by checks to see if the volume
2532 * is readable/writable */
2535 if (!(o = pa_context_set_sink_input_volume (pbuf->context, idx,
2539 /* We don't really care about the result of this call */
2543 pa_operation_unref (o);
2545 pa_threaded_mainloop_unlock (mainloop);
2552 psink->volume = volume;
2553 psink->volume_set = TRUE;
2555 GST_DEBUG_OBJECT (psink, "we have no mainloop");
2560 psink->volume = volume;
2561 psink->volume_set = TRUE;
2563 GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
2568 GST_DEBUG_OBJECT (psink, "we don't have a stream index");
2573 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
2574 ("pa_stream_set_sink_input_volume() failed: %s",
2575 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
2581 gst_pulsesink_set_mute (GstPulseSink * psink, gboolean mute)
2583 pa_operation *o = NULL;
2584 GstPulseRingBuffer *pbuf;
2590 pa_threaded_mainloop_lock (mainloop);
2592 GST_DEBUG_OBJECT (psink, "setting mute state to %d", mute);
2594 pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
2595 if (pbuf == NULL || pbuf->stream == NULL)
2598 if ((idx = pa_stream_get_index (pbuf->stream)) == PA_INVALID_INDEX)
2601 if (!(o = pa_context_set_sink_input_mute (pbuf->context, idx,
2605 /* We don't really care about the result of this call */
2609 pa_operation_unref (o);
2611 pa_threaded_mainloop_unlock (mainloop);
2619 psink->mute_set = TRUE;
2621 GST_DEBUG_OBJECT (psink, "we have no mainloop");
2627 psink->mute_set = TRUE;
2629 GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
2634 GST_DEBUG_OBJECT (psink, "we don't have a stream index");
2639 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
2640 ("pa_stream_set_sink_input_mute() failed: %s",
2641 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
2647 gst_pulsesink_sink_input_info_cb (pa_context * c, const pa_sink_input_info * i,
2648 int eol, void *userdata)
2650 GstPulseRingBuffer *pbuf;
2651 GstPulseSink *psink;
2653 pbuf = GST_PULSERING_BUFFER_CAST (userdata);
2654 psink = GST_PULSESINK_CAST (GST_OBJECT_PARENT (pbuf));
2662 /* If the index doesn't match our current stream,
2663 * it implies we just recreated the stream (caps change)
2665 if (i->index == pa_stream_get_index (pbuf->stream)) {
2666 psink->volume = pa_sw_volume_to_linear (pa_cvolume_max (&i->volume));
2667 psink->mute = i->mute;
2668 psink->current_sink_idx = i->sink;
2670 if (psink->volume > MAX_VOLUME) {
2671 GST_WARNING_OBJECT (psink, "Clipped volume from %f to %f", psink->volume,
2673 psink->volume = MAX_VOLUME;
2678 pa_threaded_mainloop_signal (mainloop, 0);
2682 gst_pulsesink_get_sink_input_info (GstPulseSink * psink, gdouble * volume,
2685 GstPulseRingBuffer *pbuf;
2686 pa_operation *o = NULL;
2692 pa_threaded_mainloop_lock (mainloop);
2694 pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
2695 if (pbuf == NULL || pbuf->stream == NULL)
2698 if ((idx = pa_stream_get_index (pbuf->stream)) == PA_INVALID_INDEX)
2701 if (!(o = pa_context_get_sink_input_info (pbuf->context, idx,
2702 gst_pulsesink_sink_input_info_cb, pbuf)))
2705 while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
2706 pa_threaded_mainloop_wait (mainloop);
2707 if (gst_pulsering_is_dead (psink, pbuf, TRUE))
2713 *volume = psink->volume;
2715 *mute = psink->mute;
2718 pa_operation_unref (o);
2720 pa_threaded_mainloop_unlock (mainloop);
2728 *volume = psink->volume;
2730 *mute = psink->mute;
2732 GST_DEBUG_OBJECT (psink, "we have no mainloop");
2737 GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
2742 GST_DEBUG_OBJECT (psink, "we don't have a stream index");
2747 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
2748 ("pa_context_get_sink_input_info() failed: %s",
2749 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
2755 gst_pulsesink_current_sink_info_cb (pa_context * c, const pa_sink_info * i,
2756 int eol, void *userdata)
2758 GstPulseSink *psink;
2760 psink = GST_PULSESINK_CAST (userdata);
2765 /* If the index doesn't match our current stream,
2766 * it implies we just recreated the stream (caps change)
2768 if (i->index == psink->current_sink_idx) {
2769 g_free (psink->current_sink_name);
2770 psink->current_sink_name = g_strdup (i->name);
2774 pa_threaded_mainloop_signal (mainloop, 0);
2778 gst_pulsesink_get_current_device (GstPulseSink * pulsesink)
2780 pa_operation *o = NULL;
2781 GstPulseRingBuffer *pbuf;
2782 gchar *current_sink;
2788 GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (pulsesink)->ringbuffer);
2789 if (pbuf == NULL || pbuf->stream == NULL)
2792 gst_pulsesink_get_sink_input_info (pulsesink, NULL, NULL);
2794 pa_threaded_mainloop_lock (mainloop);
2796 if (!(o = pa_context_get_sink_info_by_index (pbuf->context,
2797 pulsesink->current_sink_idx, gst_pulsesink_current_sink_info_cb,
2801 while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
2802 pa_threaded_mainloop_wait (mainloop);
2803 if (gst_pulsering_is_dead (pulsesink, pbuf, TRUE))
2809 current_sink = g_strdup (pulsesink->current_sink_name);
2812 pa_operation_unref (o);
2814 pa_threaded_mainloop_unlock (mainloop);
2816 return current_sink;
2821 GST_DEBUG_OBJECT (pulsesink, "we have no mainloop");
2826 GST_DEBUG_OBJECT (pulsesink, "we have no ringbuffer");
2831 GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED,
2832 ("pa_context_get_sink_input_info() failed: %s",
2833 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
2839 gst_pulsesink_device_description (GstPulseSink * psink)
2841 GstPulseRingBuffer *pbuf;
2842 pa_operation *o = NULL;
2848 pa_threaded_mainloop_lock (mainloop);
2849 pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
2853 free_device_info (&psink->device_info);
2854 if (!(o = pa_context_get_sink_info_by_name (pbuf->context,
2855 psink->device, gst_pulsesink_sink_info_cb, &psink->device_info)))
2858 while (pa_operation_get_state (o) == PA_OPERATION_RUNNING) {
2859 pa_threaded_mainloop_wait (mainloop);
2860 if (gst_pulsering_is_dead (psink, pbuf, FALSE))
2866 pa_operation_unref (o);
2868 t = g_strdup (psink->device_info.description);
2869 pa_threaded_mainloop_unlock (mainloop);
2876 GST_DEBUG_OBJECT (psink, "we have no mainloop");
2881 GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
2886 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
2887 ("pa_context_get_sink_info_by_index() failed: %s",
2888 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
2894 gst_pulsesink_set_stream_device (GstPulseSink * psink, const gchar * device)
2896 pa_operation *o = NULL;
2897 GstPulseRingBuffer *pbuf;
2903 pa_threaded_mainloop_lock (mainloop);
2905 pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
2906 if (pbuf == NULL || pbuf->stream == NULL)
2909 if ((idx = pa_stream_get_index (pbuf->stream)) == PA_INVALID_INDEX)
2913 GST_DEBUG_OBJECT (psink, "setting stream device to %s", device);
2915 if (!(o = pa_context_move_sink_input_by_name (pbuf->context, idx, device,
2922 pa_operation_unref (o);
2924 pa_threaded_mainloop_unlock (mainloop);
2931 GST_DEBUG_OBJECT (psink, "we have no mainloop");
2936 GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
2941 GST_DEBUG_OBJECT (psink, "we don't have a stream index");
2946 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
2947 ("pa_context_move_sink_input_by_name(%s) failed: %s", device,
2948 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
2955 gst_pulsesink_set_property (GObject * object,
2956 guint prop_id, const GValue * value, GParamSpec * pspec)
2958 GstPulseSink *pulsesink = GST_PULSESINK_CAST (object);
2962 g_free (pulsesink->server);
2963 pulsesink->server = g_value_dup_string (value);
2966 g_free (pulsesink->device);
2967 pulsesink->device = g_value_dup_string (value);
2968 gst_pulsesink_set_stream_device (pulsesink, pulsesink->device);
2971 gst_pulsesink_set_volume (pulsesink, g_value_get_double (value));
2974 gst_pulsesink_set_mute (pulsesink, g_value_get_boolean (value));
2976 case PROP_CLIENT_NAME:
2977 g_free (pulsesink->client_name);
2978 if (!g_value_get_string (value)) {
2979 GST_WARNING_OBJECT (pulsesink,
2980 "Empty PulseAudio client name not allowed. Resetting to default value");
2981 pulsesink->client_name = gst_pulse_client_name ();
2983 pulsesink->client_name = g_value_dup_string (value);
2985 case PROP_STREAM_PROPERTIES:
2986 if (pulsesink->properties)
2987 gst_structure_free (pulsesink->properties);
2988 pulsesink->properties =
2989 gst_structure_copy (gst_value_get_structure (value));
2990 if (pulsesink->proplist)
2991 pa_proplist_free (pulsesink->proplist);
2992 pulsesink->proplist = gst_pulse_make_proplist (pulsesink->properties);
2995 case PROP_AUDIO_LATENCY:
2996 g_free (pulsesink->latency);
2997 pulsesink->latency = g_value_dup_string (value);
2998 /* setting NULL restores the default latency */
2999 if (pulsesink->latency == NULL) {
3000 pulsesink->latency = g_strdup (DEFAULT_AUDIO_LATENCY);
3002 if (!pulsesink->proplist) {
3003 pulsesink->proplist = pa_proplist_new();
3005 pa_proplist_sets(pulsesink->proplist, PA_PROP_MEDIA_TIZEN_AUDIO_LATENCY, pulsesink->latency);
3006 GST_DEBUG_OBJECT(pulsesink, "latency(%s)", pulsesink->latency);
3008 #endif /* __TIZEN__ */
3010 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
3016 gst_pulsesink_get_property (GObject * object,
3017 guint prop_id, GValue * value, GParamSpec * pspec)
3020 GstPulseSink *pulsesink = GST_PULSESINK_CAST (object);
3024 g_value_set_string (value, pulsesink->server);
3027 g_value_set_string (value, pulsesink->device);
3029 case PROP_CURRENT_DEVICE:
3031 gchar *current_device = gst_pulsesink_get_current_device (pulsesink);
3033 g_value_take_string (value, current_device);
3035 g_value_set_string (value, "");
3038 case PROP_DEVICE_NAME:
3039 g_value_take_string (value, gst_pulsesink_device_description (pulsesink));
3045 gst_pulsesink_get_sink_input_info (pulsesink, &volume, NULL);
3046 g_value_set_double (value, volume);
3053 gst_pulsesink_get_sink_input_info (pulsesink, NULL, &mute);
3054 g_value_set_boolean (value, mute);
3057 case PROP_CLIENT_NAME:
3058 g_value_set_string (value, pulsesink->client_name);
3060 case PROP_STREAM_PROPERTIES:
3061 gst_value_set_structure (value, pulsesink->properties);
3064 case PROP_AUDIO_LATENCY:
3065 g_value_set_string (value, pulsesink->latency);
3067 #endif /* __TIZEN__ */
3069 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
3075 gst_pulsesink_change_title (GstPulseSink * psink, const gchar * t)
3077 pa_operation *o = NULL;
3078 GstPulseRingBuffer *pbuf;
3080 pa_threaded_mainloop_lock (mainloop);
3082 pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
3084 if (pbuf == NULL || pbuf->stream == NULL)
3087 g_free (pbuf->stream_name);
3088 pbuf->stream_name = g_strdup (t);
3090 if (!(o = pa_stream_set_name (pbuf->stream, pbuf->stream_name, NULL, NULL)))
3093 /* We're not interested if this operation failed or not */
3097 pa_operation_unref (o);
3098 pa_threaded_mainloop_unlock (mainloop);
3105 GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
3110 GST_ELEMENT_ERROR (psink, RESOURCE, FAILED,
3111 ("pa_stream_set_name() failed: %s",
3112 pa_strerror (pa_context_errno (pbuf->context))), (NULL));
3118 gst_pulsesink_change_props (GstPulseSink * psink, GstTagList * l)
3120 static const gchar *const map[] = {
3121 GST_TAG_TITLE, PA_PROP_MEDIA_TITLE,
3123 /* might get overriden in the next iteration by GST_TAG_ARTIST */
3124 GST_TAG_PERFORMER, PA_PROP_MEDIA_ARTIST,
3126 GST_TAG_ARTIST, PA_PROP_MEDIA_ARTIST,
3127 GST_TAG_LANGUAGE_CODE, PA_PROP_MEDIA_LANGUAGE,
3128 GST_TAG_LOCATION, PA_PROP_MEDIA_FILENAME,
3129 /* We might add more here later on ... */
3132 pa_proplist *pl = NULL;
3133 const gchar *const *t;
3134 gboolean empty = TRUE;
3135 pa_operation *o = NULL;
3136 GstPulseRingBuffer *pbuf;
3138 pl = pa_proplist_new ();
3140 for (t = map; *t; t += 2) {
3143 if (gst_tag_list_get_string (l, *t, &n)) {
3146 pa_proplist_sets (pl, *(t + 1), n);
3156 pa_threaded_mainloop_lock (mainloop);
3157 pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
3158 if (pbuf == NULL || pbuf->stream == NULL)
3161 /* We're not interested if this operation failed or not */
3162 if (!(o = pa_stream_proplist_update (pbuf->stream, PA_UPDATE_REPLACE,
3164 GST_DEBUG_OBJECT (psink, "pa_stream_proplist_update() failed");
3170 pa_operation_unref (o);
3172 pa_threaded_mainloop_unlock (mainloop);
3177 pa_proplist_free (pl);
3184 GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
3190 gst_pulsesink_flush_ringbuffer (GstPulseSink * psink)
3192 GstPulseRingBuffer *pbuf;
3194 pa_threaded_mainloop_lock (mainloop);
3196 pbuf = GST_PULSERING_BUFFER_CAST (GST_AUDIO_BASE_SINK (psink)->ringbuffer);
3198 if (pbuf == NULL || pbuf->stream == NULL)
3201 gst_pulsering_flush (pbuf);
3203 /* Uncork if we haven't already (happens when waiting to get enough data
3204 * to send out the first time) */
3206 gst_pulsering_set_corked (pbuf, FALSE, FALSE);
3208 /* We're not interested if this operation failed or not */
3210 pa_threaded_mainloop_unlock (mainloop);
3217 GST_DEBUG_OBJECT (psink, "we have no ringbuffer");
3223 gst_pulsesink_event (GstBaseSink * sink, GstEvent * event)
3225 GstPulseSink *pulsesink = GST_PULSESINK_CAST (sink);
3227 switch (GST_EVENT_TYPE (event)) {
3228 case GST_EVENT_TAG:{
3229 gchar *title = NULL, *artist = NULL, *location = NULL, *description =
3230 NULL, *t = NULL, *buf = NULL;
3233 gst_event_parse_tag (event, &l);
3235 gst_tag_list_get_string (l, GST_TAG_TITLE, &title);
3236 gst_tag_list_get_string (l, GST_TAG_ARTIST, &artist);
3237 gst_tag_list_get_string (l, GST_TAG_LOCATION, &location);
3238 gst_tag_list_get_string (l, GST_TAG_DESCRIPTION, &description);
3241 gst_tag_list_get_string (l, GST_TAG_PERFORMER, &artist);
3243 if (title && artist)
3244 /* TRANSLATORS: 'song title' by 'artist name' */
3245 t = buf = g_strdup_printf (_("'%s' by '%s'"), g_strstrip (title),
3246 g_strstrip (artist));
3248 t = g_strstrip (title);
3249 else if (description)
3250 t = g_strstrip (description);
3252 t = g_strstrip (location);
3255 gst_pulsesink_change_title (pulsesink, t);
3260 g_free (description);
3263 gst_pulsesink_change_props (pulsesink, l);
3267 case GST_EVENT_GAP:{
3268 GstClockTime timestamp, duration;
3270 gst_event_parse_gap (event, ×tamp, &duration);
3271 if (duration == GST_CLOCK_TIME_NONE)
3272 gst_pulsesink_flush_ringbuffer (pulsesink);
3276 gst_pulsesink_flush_ringbuffer (pulsesink);
3282 return GST_BASE_SINK_CLASS (parent_class)->event (sink, event);
3286 gst_pulsesink_query (GstBaseSink * sink, GstQuery * query)
3288 GstPulseSink *pulsesink = GST_PULSESINK_CAST (sink);
3289 gboolean ret = FALSE;
3291 switch (GST_QUERY_TYPE (query)) {
3292 case GST_QUERY_CAPS:
3294 GstCaps *caps, *filter;
3296 gst_query_parse_caps (query, &filter);
3297 caps = gst_pulsesink_query_getcaps (pulsesink, filter);
3300 gst_query_set_caps_result (query, caps);
3301 gst_caps_unref (caps);
3306 case GST_QUERY_ACCEPT_CAPS:
3310 gst_query_parse_accept_caps (query, &caps);
3311 ret = gst_pulsesink_query_acceptcaps (pulsesink, caps);
3312 gst_query_set_accept_caps_result (query, ret);
3317 ret = GST_BASE_SINK_CLASS (parent_class)->query (sink, query);
3324 gst_pulsesink_release_mainloop (GstPulseSink * psink)
3329 pa_threaded_mainloop_lock (mainloop);
3330 while (psink->defer_pending) {
3331 GST_DEBUG_OBJECT (psink, "waiting for stream status message emission");
3332 pa_threaded_mainloop_wait (mainloop);
3334 pa_threaded_mainloop_unlock (mainloop);
3336 g_mutex_lock (&pa_shared_resource_mutex);
3338 if (!mainloop_ref_ct) {
3339 GST_INFO_OBJECT (psink, "terminating pa main loop thread");
3340 pa_threaded_mainloop_stop (mainloop);
3341 pa_threaded_mainloop_free (mainloop);
3344 g_mutex_unlock (&pa_shared_resource_mutex);
3347 static GstStateChangeReturn
3348 gst_pulsesink_change_state (GstElement * element, GstStateChange transition)
3350 GstPulseSink *pulsesink = GST_PULSESINK (element);
3351 GstStateChangeReturn ret;
3353 switch (transition) {
3354 case GST_STATE_CHANGE_NULL_TO_READY:
3355 g_mutex_lock (&pa_shared_resource_mutex);
3356 if (!mainloop_ref_ct) {
3357 GST_INFO_OBJECT (element, "new pa main loop thread");
3358 if (!(mainloop = pa_threaded_mainloop_new ()))
3359 goto mainloop_failed;
3360 if (pa_threaded_mainloop_start (mainloop) < 0) {
3361 pa_threaded_mainloop_free (mainloop);
3362 goto mainloop_start_failed;
3364 mainloop_ref_ct = 1;
3365 g_mutex_unlock (&pa_shared_resource_mutex);
3367 GST_INFO_OBJECT (element, "reusing pa main loop thread");
3369 g_mutex_unlock (&pa_shared_resource_mutex);
3372 case GST_STATE_CHANGE_READY_TO_PAUSED:
3373 gst_element_post_message (element,
3374 gst_message_new_clock_provide (GST_OBJECT_CAST (element),
3375 GST_AUDIO_BASE_SINK (pulsesink)->provided_clock, TRUE));
3382 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
3383 if (ret == GST_STATE_CHANGE_FAILURE)
3386 switch (transition) {
3387 case GST_STATE_CHANGE_PAUSED_TO_READY:
3388 /* format_lost is reset in release() in audiobasesink */
3389 gst_element_post_message (element,
3390 gst_message_new_clock_lost (GST_OBJECT_CAST (element),
3391 GST_AUDIO_BASE_SINK (pulsesink)->provided_clock));
3393 case GST_STATE_CHANGE_READY_TO_NULL:
3394 gst_pulsesink_release_mainloop (pulsesink);
3405 g_mutex_unlock (&pa_shared_resource_mutex);
3406 GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED,
3407 ("pa_threaded_mainloop_new() failed"), (NULL));
3408 return GST_STATE_CHANGE_FAILURE;
3410 mainloop_start_failed:
3412 g_mutex_unlock (&pa_shared_resource_mutex);
3413 GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED,
3414 ("pa_threaded_mainloop_start() failed"), (NULL));
3415 return GST_STATE_CHANGE_FAILURE;
3419 if (transition == GST_STATE_CHANGE_NULL_TO_READY) {
3420 /* Clear the PA mainloop if audiobasesink failed to open the ring_buffer */
3421 g_assert (mainloop);
3422 gst_pulsesink_release_mainloop (pulsesink);