2 * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
3 * 2005 Wim Taymans <wim@fluendo.com>
5 * gstaudiosink.c: simple audio sink base class
7 * This library is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Library General Public
9 * License as published by the Free Software Foundation; either
10 * version 2 of the License, or (at your option) any later version.
12 * This library is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Library General Public License for more details.
17 * You should have received a copy of the GNU Library General Public
18 * License along with this library; if not, write to the
19 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
20 * Boston, MA 02111-1307, USA.
24 * SECTION:gstaudiosink
25 * @short_description: Simple base class for audio sinks
26 * @see_also: #GstBaseAudioSink, #GstRingBuffer, #GstAudioSink.
28 * This is the most simple base class for audio sinks that only requires
29 * subclasses to implement a set of simple functions:
34 * <listitem><para>Open the device.</para></listitem>
37 * <term>prepare()</term>
38 * <listitem><para>Configure the device with the specified format.</para></listitem>
41 * <term>write()</term>
42 * <listitem><para>Write samples to the device.</para></listitem>
45 * <term>reset()</term>
46 * <listitem><para>Unblock writes and flush the device.</para></listitem>
49 * <term>delay()</term>
50 * <listitem><para>Get the number of samples written but not yet played
51 * by the device.</para></listitem>
54 * <term>unprepare()</term>
55 * <listitem><para>Undo operations done by prepare.</para></listitem>
58 * <term>close()</term>
59 * <listitem><para>Close the device.</para></listitem>
63 * All scheduling of samples and timestamps is done in this base class
64 * together with #GstBaseAudioSink using a default implementation of a
65 * #GstRingBuffer that uses threads.
67 * Last reviewed on 2006-09-27 (0.10.12)
72 #include "gstaudiosink.h"
74 GST_DEBUG_CATEGORY_STATIC (gst_audio_sink_debug);
75 #define GST_CAT_DEFAULT gst_audio_sink_debug
77 #define GST_TYPE_AUDIORING_BUFFER \
78 (gst_audioringbuffer_get_type())
79 #define GST_AUDIORING_BUFFER(obj) \
80 (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIORING_BUFFER,GstAudioRingBuffer))
81 #define GST_AUDIORING_BUFFER_CLASS(klass) \
82 (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIORING_BUFFER,GstAudioRingBufferClass))
83 #define GST_AUDIORING_BUFFER_GET_CLASS(obj) \
84 (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_AUDIORING_BUFFER, GstAudioRingBufferClass))
85 #define GST_AUDIORING_BUFFER_CAST(obj) \
86 ((GstAudioRingBuffer *)obj)
87 #define GST_IS_AUDIORING_BUFFER(obj) \
88 (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIORING_BUFFER))
89 #define GST_IS_AUDIORING_BUFFER_CLASS(klass)\
90 (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIORING_BUFFER))
92 typedef struct _GstAudioRingBuffer GstAudioRingBuffer;
93 typedef struct _GstAudioRingBufferClass GstAudioRingBufferClass;
95 #define GST_AUDIORING_BUFFER_GET_COND(buf) (((GstAudioRingBuffer *)buf)->cond)
96 #define GST_AUDIORING_BUFFER_WAIT(buf) (g_cond_wait (GST_AUDIORING_BUFFER_GET_COND (buf), GST_OBJECT_GET_LOCK (buf)))
97 #define GST_AUDIORING_BUFFER_SIGNAL(buf) (g_cond_signal (GST_AUDIORING_BUFFER_GET_COND (buf)))
98 #define GST_AUDIORING_BUFFER_BROADCAST(buf)(g_cond_broadcast (GST_AUDIORING_BUFFER_GET_COND (buf)))
100 struct _GstAudioRingBuffer
102 GstRingBuffer object;
110 struct _GstAudioRingBufferClass
112 GstRingBufferClass parent_class;
115 static void gst_audioringbuffer_class_init (GstAudioRingBufferClass * klass);
116 static void gst_audioringbuffer_init (GstAudioRingBuffer * ringbuffer,
117 GstAudioRingBufferClass * klass);
118 static void gst_audioringbuffer_dispose (GObject * object);
119 static void gst_audioringbuffer_finalize (GObject * object);
121 static GstRingBufferClass *ring_parent_class = NULL;
123 static gboolean gst_audioringbuffer_open_device (GstRingBuffer * buf);
124 static gboolean gst_audioringbuffer_close_device (GstRingBuffer * buf);
125 static gboolean gst_audioringbuffer_acquire (GstRingBuffer * buf,
126 GstRingBufferSpec * spec);
127 static gboolean gst_audioringbuffer_release (GstRingBuffer * buf);
128 static gboolean gst_audioringbuffer_start (GstRingBuffer * buf);
129 static gboolean gst_audioringbuffer_pause (GstRingBuffer * buf);
130 static gboolean gst_audioringbuffer_stop (GstRingBuffer * buf);
131 static guint gst_audioringbuffer_delay (GstRingBuffer * buf);
132 static gboolean gst_audioringbuffer_activate (GstRingBuffer * buf,
135 /* ringbuffer abstract base class */
137 gst_audioringbuffer_get_type (void)
139 static GType ringbuffer_type = 0;
141 if (!ringbuffer_type) {
142 static const GTypeInfo ringbuffer_info = {
143 sizeof (GstAudioRingBufferClass),
146 (GClassInitFunc) gst_audioringbuffer_class_init,
149 sizeof (GstAudioRingBuffer),
151 (GInstanceInitFunc) gst_audioringbuffer_init,
156 g_type_register_static (GST_TYPE_RING_BUFFER, "GstAudioSinkRingBuffer",
157 &ringbuffer_info, 0);
159 return ringbuffer_type;
163 gst_audioringbuffer_class_init (GstAudioRingBufferClass * klass)
165 GObjectClass *gobject_class;
166 GstRingBufferClass *gstringbuffer_class;
168 gobject_class = (GObjectClass *) klass;
169 gstringbuffer_class = (GstRingBufferClass *) klass;
171 ring_parent_class = g_type_class_peek_parent (klass);
173 gobject_class->dispose = gst_audioringbuffer_dispose;
174 gobject_class->finalize = gst_audioringbuffer_finalize;
176 gstringbuffer_class->open_device =
177 GST_DEBUG_FUNCPTR (gst_audioringbuffer_open_device);
178 gstringbuffer_class->close_device =
179 GST_DEBUG_FUNCPTR (gst_audioringbuffer_close_device);
180 gstringbuffer_class->acquire =
181 GST_DEBUG_FUNCPTR (gst_audioringbuffer_acquire);
182 gstringbuffer_class->release =
183 GST_DEBUG_FUNCPTR (gst_audioringbuffer_release);
184 gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_audioringbuffer_start);
185 gstringbuffer_class->pause = GST_DEBUG_FUNCPTR (gst_audioringbuffer_pause);
186 gstringbuffer_class->resume = GST_DEBUG_FUNCPTR (gst_audioringbuffer_start);
187 gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_audioringbuffer_stop);
189 gstringbuffer_class->delay = GST_DEBUG_FUNCPTR (gst_audioringbuffer_delay);
190 gstringbuffer_class->activate =
191 GST_DEBUG_FUNCPTR (gst_audioringbuffer_activate);
194 typedef guint (*WriteFunc) (GstAudioSink * sink, gpointer data, guint length);
196 /* this internal thread does nothing else but write samples to the audio device.
197 * It will write each segment in the ringbuffer and will update the play
199 * The start/stop methods control the thread.
202 audioringbuffer_thread_func (GstRingBuffer * buf)
205 GstAudioSinkClass *csink;
206 GstAudioRingBuffer *abuf = GST_AUDIORING_BUFFER_CAST (buf);
211 sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
212 csink = GST_AUDIO_SINK_GET_CLASS (sink);
214 GST_DEBUG_OBJECT (sink, "enter thread");
216 GST_OBJECT_LOCK (abuf);
217 GST_DEBUG_OBJECT (sink, "signal wait");
218 GST_AUDIORING_BUFFER_SIGNAL (buf);
219 GST_OBJECT_UNLOCK (abuf);
221 writefunc = csink->write;
222 if (writefunc == NULL)
225 g_value_init (&val, G_TYPE_POINTER);
226 g_value_set_pointer (&val, sink->thread);
227 message = gst_message_new_stream_status (GST_OBJECT_CAST (buf),
228 GST_STREAM_STATUS_TYPE_ENTER, GST_ELEMENT_CAST (sink));
229 gst_message_set_stream_status_object (message, &val);
230 GST_DEBUG_OBJECT (sink, "posting ENTER stream status");
231 gst_element_post_message (GST_ELEMENT_CAST (sink), message);
238 /* buffer must be started */
239 if (gst_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) {
244 written = writefunc (sink, readptr, left);
245 GST_LOG_OBJECT (sink, "transfered %d bytes of %d from segment %d",
246 written, left, readseg);
247 if (written < 0 || written > left) {
248 /* might not be critical, it e.g. happens when aborting playback */
249 GST_WARNING_OBJECT (sink,
250 "error writing data in %s (reason: %s), skipping segment (left: %d, written: %d)",
251 GST_DEBUG_FUNCPTR_NAME (writefunc),
252 (errno > 1 ? g_strerror (errno) : "unknown"), left, written);
259 /* clear written samples */
260 gst_ring_buffer_clear (buf, readseg);
262 /* we wrote one segment */
263 gst_ring_buffer_advance (buf, 1);
265 GST_OBJECT_LOCK (abuf);
268 GST_DEBUG_OBJECT (sink, "signal wait");
269 GST_AUDIORING_BUFFER_SIGNAL (buf);
270 GST_DEBUG_OBJECT (sink, "wait for action");
271 GST_AUDIORING_BUFFER_WAIT (buf);
272 GST_DEBUG_OBJECT (sink, "got signal");
275 GST_DEBUG_OBJECT (sink, "continue running");
276 GST_OBJECT_UNLOCK (abuf);
280 /* Will never be reached */
281 g_assert_not_reached ();
287 GST_DEBUG_OBJECT (sink, "no write function, exit thread");
292 GST_OBJECT_UNLOCK (abuf);
293 GST_DEBUG_OBJECT (sink, "stop running, exit thread");
294 message = gst_message_new_stream_status (GST_OBJECT_CAST (buf),
295 GST_STREAM_STATUS_TYPE_LEAVE, GST_ELEMENT_CAST (sink));
296 gst_message_set_stream_status_object (message, &val);
297 GST_DEBUG_OBJECT (sink, "posting LEAVE stream status");
298 gst_element_post_message (GST_ELEMENT_CAST (sink), message);
304 gst_audioringbuffer_init (GstAudioRingBuffer * ringbuffer,
305 GstAudioRingBufferClass * g_class)
307 ringbuffer->running = FALSE;
308 ringbuffer->queuedseg = 0;
310 ringbuffer->cond = g_cond_new ();
314 gst_audioringbuffer_dispose (GObject * object)
316 G_OBJECT_CLASS (ring_parent_class)->dispose (object);
320 gst_audioringbuffer_finalize (GObject * object)
322 GstAudioRingBuffer *ringbuffer = GST_AUDIORING_BUFFER_CAST (object);
324 g_cond_free (ringbuffer->cond);
326 G_OBJECT_CLASS (ring_parent_class)->finalize (object);
330 gst_audioringbuffer_open_device (GstRingBuffer * buf)
333 GstAudioSinkClass *csink;
334 gboolean result = TRUE;
336 sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
337 csink = GST_AUDIO_SINK_GET_CLASS (sink);
340 result = csink->open (sink);
349 GST_DEBUG_OBJECT (sink, "could not open device");
355 gst_audioringbuffer_close_device (GstRingBuffer * buf)
358 GstAudioSinkClass *csink;
359 gboolean result = TRUE;
361 sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
362 csink = GST_AUDIO_SINK_GET_CLASS (sink);
365 result = csink->close (sink);
368 goto could_not_close;
374 GST_DEBUG_OBJECT (sink, "could not close device");
380 gst_audioringbuffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
383 GstAudioSinkClass *csink;
384 gboolean result = FALSE;
386 sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
387 csink = GST_AUDIO_SINK_GET_CLASS (sink);
390 result = csink->prepare (sink, spec);
392 goto could_not_prepare;
394 /* set latency to one more segment as we need some headroom */
395 spec->seglatency = spec->segtotal + 1;
397 buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize);
398 memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data));
405 GST_DEBUG_OBJECT (sink, "could not prepare device");
411 gst_audioringbuffer_activate (GstRingBuffer * buf, gboolean active)
414 GstAudioRingBuffer *abuf;
415 GError *error = NULL;
417 sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
418 abuf = GST_AUDIORING_BUFFER_CAST (buf);
421 abuf->running = TRUE;
423 GST_DEBUG_OBJECT (sink, "starting thread");
425 g_thread_create ((GThreadFunc) audioringbuffer_thread_func, buf, TRUE,
427 if (!sink->thread || error != NULL)
430 GST_DEBUG_OBJECT (sink, "waiting for thread");
431 /* the object lock is taken */
432 GST_AUDIORING_BUFFER_WAIT (buf);
433 GST_DEBUG_OBJECT (sink, "thread is started");
435 abuf->running = FALSE;
436 GST_DEBUG_OBJECT (sink, "signal wait");
437 GST_AUDIORING_BUFFER_SIGNAL (buf);
439 GST_OBJECT_UNLOCK (buf);
441 /* join the thread */
442 g_thread_join (sink->thread);
444 GST_OBJECT_LOCK (buf);
452 GST_ERROR_OBJECT (sink, "could not create thread %s", error->message);
454 GST_ERROR_OBJECT (sink, "could not create thread for unknown reason");
459 /* function is called with LOCK */
461 gst_audioringbuffer_release (GstRingBuffer * buf)
464 GstAudioSinkClass *csink;
465 gboolean result = FALSE;
467 sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
468 csink = GST_AUDIO_SINK_GET_CLASS (sink);
470 /* free the buffer */
471 gst_buffer_unref (buf->data);
474 if (csink->unprepare)
475 result = csink->unprepare (sink);
478 goto could_not_unprepare;
480 GST_DEBUG_OBJECT (sink, "unprepared");
486 GST_DEBUG_OBJECT (sink, "could not unprepare device");
492 gst_audioringbuffer_start (GstRingBuffer * buf)
496 sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
498 GST_DEBUG_OBJECT (sink, "start, sending signal");
499 GST_AUDIORING_BUFFER_SIGNAL (buf);
505 gst_audioringbuffer_pause (GstRingBuffer * buf)
508 GstAudioSinkClass *csink;
510 sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
511 csink = GST_AUDIO_SINK_GET_CLASS (sink);
513 /* unblock any pending writes to the audio device */
515 GST_DEBUG_OBJECT (sink, "reset...");
517 GST_DEBUG_OBJECT (sink, "reset done");
524 gst_audioringbuffer_stop (GstRingBuffer * buf)
527 GstAudioSinkClass *csink;
529 sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
530 csink = GST_AUDIO_SINK_GET_CLASS (sink);
532 /* unblock any pending writes to the audio device */
534 GST_DEBUG_OBJECT (sink, "reset...");
536 GST_DEBUG_OBJECT (sink, "reset done");
540 GST_DEBUG_OBJECT (sink, "stop, waiting...");
541 GST_AUDIORING_BUFFER_WAIT (buf);
542 GST_DEBUG_OBJECT (sink, "stopped");
550 gst_audioringbuffer_delay (GstRingBuffer * buf)
553 GstAudioSinkClass *csink;
556 sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
557 csink = GST_AUDIO_SINK_GET_CLASS (sink);
560 res = csink->delay (sink);
565 /* AudioSink signals and args */
577 #define _do_init(bla) \
578 GST_DEBUG_CATEGORY_INIT (gst_audio_sink_debug, "audiosink", 0, "audiosink element");
580 GST_BOILERPLATE_FULL (GstAudioSink, gst_audio_sink, GstBaseAudioSink,
581 GST_TYPE_BASE_AUDIO_SINK, _do_init);
583 static GstRingBuffer *gst_audio_sink_create_ringbuffer (GstBaseAudioSink *
587 gst_audio_sink_base_init (gpointer g_class)
592 gst_audio_sink_class_init (GstAudioSinkClass * klass)
594 GstBaseAudioSinkClass *gstbaseaudiosink_class;
596 gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass;
598 gstbaseaudiosink_class->create_ringbuffer =
599 GST_DEBUG_FUNCPTR (gst_audio_sink_create_ringbuffer);
601 g_type_class_ref (GST_TYPE_AUDIORING_BUFFER);
605 gst_audio_sink_init (GstAudioSink * audiosink, GstAudioSinkClass * g_class)
609 static GstRingBuffer *
610 gst_audio_sink_create_ringbuffer (GstBaseAudioSink * sink)
612 GstRingBuffer *buffer;
614 GST_DEBUG_OBJECT (sink, "creating ringbuffer");
615 buffer = g_object_new (GST_TYPE_AUDIORING_BUFFER, NULL);
616 GST_DEBUG_OBJECT (sink, "created ringbuffer @%p", buffer);