gstreamer/gst-libs/gst/audio/gstaudiosink.c
Thomas Vander Stichele 9e8a11d3ce use overridable ERROR_CFLAGS; more macro splitting
Original commit message from CVS:
use overridable ERROR_CFLAGS; more macro splitting
2005-07-10 12:03:58 +00:00

419 lines
11 KiB
C

/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2005 Wim Taymans <wim@fluendo.com>
*
* gstaudiosink.c: simple audio sink base class
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include <string.h>
#include "gstaudiosink.h"
GST_DEBUG_CATEGORY_STATIC (gst_audio_sink_debug);
#define GST_CAT_DEFAULT gst_audio_sink_debug
#define GST_TYPE_AUDIORING_BUFFER \
(gst_audioringbuffer_get_type())
#define GST_AUDIORING_BUFFER(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIORING_BUFFER,GstAudioRingBuffer))
#define GST_AUDIORING_BUFFER_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIORING_BUFFER,GstAudioRingBufferClass))
#define GST_AUDIORING_BUFFER_GET_CLASS(obj) \
(G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_AUDIORING_BUFFER, GstAudioRingBufferClass))
#define GST_IS_AUDIORING_BUFFER(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIORING_BUFFER))
#define GST_IS_AUDIORING_BUFFER_CLASS(obj)\
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIORING_BUFFER))
typedef struct _GstAudioRingBuffer GstAudioRingBuffer;
typedef struct _GstAudioRingBufferClass GstAudioRingBufferClass;
#define GST_AUDIORING_BUFFER_GET_COND(buf) (((GstAudioRingBuffer *)buf)->cond)
#define GST_AUDIORING_BUFFER_WAIT(buf) (g_cond_wait (GST_AUDIORING_BUFFER_GET_COND (buf), GST_GET_LOCK (buf)))
#define GST_AUDIORING_BUFFER_SIGNAL(buf) (g_cond_signal (GST_AUDIORING_BUFFER_GET_COND (buf)))
#define GST_AUDIORING_BUFFER_BROADCAST(buf)(g_cond_broadcast (GST_AUDIORING_BUFFER_GET_COND (buf)))
struct _GstAudioRingBuffer
{
GstRingBuffer object;
gboolean running;
gint queuedseg;
GCond *cond;
};
struct _GstAudioRingBufferClass
{
GstRingBufferClass parent_class;
};
static void gst_audioringbuffer_class_init (GstAudioRingBufferClass * klass);
static void gst_audioringbuffer_init (GstAudioRingBuffer * ringbuffer);
static void gst_audioringbuffer_dispose (GObject * object);
static void gst_audioringbuffer_finalize (GObject * object);
static GstRingBufferClass *ring_parent_class = NULL;
static gboolean gst_audioringbuffer_acquire (GstRingBuffer * buf,
GstRingBufferSpec * spec);
static gboolean gst_audioringbuffer_release (GstRingBuffer * buf);
static gboolean gst_audioringbuffer_start (GstRingBuffer * buf);
static gboolean gst_audioringbuffer_stop (GstRingBuffer * buf);
static guint gst_audioringbuffer_delay (GstRingBuffer * buf);
/* ringbuffer abstract base class */
static GType
gst_audioringbuffer_get_type (void)
{
static GType ringbuffer_type = 0;
if (!ringbuffer_type) {
static const GTypeInfo ringbuffer_info = {
sizeof (GstAudioRingBufferClass),
NULL,
NULL,
(GClassInitFunc) gst_audioringbuffer_class_init,
NULL,
NULL,
sizeof (GstAudioRingBuffer),
0,
(GInstanceInitFunc) gst_audioringbuffer_init,
NULL
};
ringbuffer_type =
g_type_register_static (GST_TYPE_RING_BUFFER, "GstAudioSinkRingBuffer",
&ringbuffer_info, 0);
}
return ringbuffer_type;
}
static void
gst_audioringbuffer_class_init (GstAudioRingBufferClass * klass)
{
GObjectClass *gobject_class;
GstObjectClass *gstobject_class;
GstRingBufferClass *gstringbuffer_class;
gobject_class = (GObjectClass *) klass;
gstobject_class = (GstObjectClass *) klass;
gstringbuffer_class = (GstRingBufferClass *) klass;
ring_parent_class = g_type_class_ref (GST_TYPE_RING_BUFFER);
gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_audioringbuffer_dispose);
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_audioringbuffer_finalize);
gstringbuffer_class->acquire =
GST_DEBUG_FUNCPTR (gst_audioringbuffer_acquire);
gstringbuffer_class->release =
GST_DEBUG_FUNCPTR (gst_audioringbuffer_release);
gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_audioringbuffer_start);
gstringbuffer_class->resume = GST_DEBUG_FUNCPTR (gst_audioringbuffer_start);
gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_audioringbuffer_stop);
gstringbuffer_class->delay = GST_DEBUG_FUNCPTR (gst_audioringbuffer_delay);
}
typedef guint (*WriteFunc) (GstAudioSink * sink, gpointer data, guint length);
/* this internal thread does nothing else but write samples to the audio device.
* It will write each segment in the ringbuffer and will update the play
* pointer.
* The start/stop methods control the thread.
*/
static void
audioringbuffer_thread_func (GstRingBuffer * buf)
{
GstAudioSink *sink;
GstAudioSinkClass *csink;
GstAudioRingBuffer *abuf = GST_AUDIORING_BUFFER (buf);
WriteFunc writefunc;
sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
csink = GST_AUDIO_SINK_GET_CLASS (sink);
GST_DEBUG ("enter thread");
writefunc = csink->write;
if (writefunc == NULL)
goto no_function;
while (TRUE) {
gint left, len;
guint8 *readptr;
gint readseg;
if (gst_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) {
gint written = 0;
left = len;
do {
GST_DEBUG ("transfer %d bytes from segment %d", left, readseg);
written = writefunc (sink, readptr + written, left);
GST_DEBUG ("transfered %d bytes", written);
if (written < 0 || written > left) {
GST_WARNING ("error writing data (reason: %s), skipping segment\n",
strerror (errno));
break;
}
left -= written;
} while (left > 0);
/* clear written samples */
gst_ring_buffer_clear (buf, readseg);
/* we wrote one segment */
gst_ring_buffer_advance (buf, 1);
} else {
GST_LOCK (abuf);
if (!abuf->running)
goto stop_running;
GST_DEBUG ("signal wait");
GST_AUDIORING_BUFFER_SIGNAL (buf);
GST_DEBUG ("wait for action");
GST_AUDIORING_BUFFER_WAIT (buf);
GST_DEBUG ("got signal");
if (!abuf->running)
goto stop_running;
GST_DEBUG ("continue running");
GST_UNLOCK (abuf);
}
}
GST_DEBUG ("exit thread");
return;
/* ERROR */
no_function:
{
GST_DEBUG ("no write function, exit thread");
return;
}
stop_running:
{
GST_UNLOCK (abuf);
GST_DEBUG ("stop running, exit thread");
return;
}
}
static void
gst_audioringbuffer_init (GstAudioRingBuffer * ringbuffer)
{
ringbuffer->running = FALSE;
ringbuffer->queuedseg = 0;
ringbuffer->cond = g_cond_new ();
}
static void
gst_audioringbuffer_dispose (GObject * object)
{
G_OBJECT_CLASS (ring_parent_class)->dispose (object);
}
static void
gst_audioringbuffer_finalize (GObject * object)
{
G_OBJECT_CLASS (ring_parent_class)->finalize (object);
}
static gboolean
gst_audioringbuffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
{
GstAudioSink *sink;
GstAudioSinkClass *csink;
GstAudioRingBuffer *abuf;
gboolean result = FALSE;
sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
csink = GST_AUDIO_SINK_GET_CLASS (sink);
if (csink->open)
result = csink->open (sink, spec);
if (!result)
goto could_not_open;
/* allocate one more segment as we need some headroom */
spec->segtotal++;
buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize);
memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data));
abuf = GST_AUDIORING_BUFFER (buf);
abuf->running = TRUE;
sink->thread =
g_thread_create ((GThreadFunc) audioringbuffer_thread_func, buf, TRUE,
NULL);
GST_AUDIORING_BUFFER_WAIT (buf);
return result;
could_not_open:
{
return FALSE;
}
}
/* function is called with LOCK */
static gboolean
gst_audioringbuffer_release (GstRingBuffer * buf)
{
GstAudioSink *sink;
GstAudioSinkClass *csink;
GstAudioRingBuffer *abuf;
gboolean result = FALSE;
sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
csink = GST_AUDIO_SINK_GET_CLASS (sink);
abuf = GST_AUDIORING_BUFFER (buf);
abuf->running = FALSE;
GST_AUDIORING_BUFFER_SIGNAL (buf);
GST_UNLOCK (buf);
/* join the thread */
g_thread_join (sink->thread);
GST_LOCK (buf);
/* free the buffer */
gst_buffer_unref (buf->data);
buf->data = NULL;
if (csink->close)
result = csink->close (sink);
return result;
}
static gboolean
gst_audioringbuffer_start (GstRingBuffer * buf)
{
GstAudioSink *sink;
sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
GST_DEBUG ("start, sending signal");
GST_AUDIORING_BUFFER_SIGNAL (buf);
return TRUE;
}
static gboolean
gst_audioringbuffer_stop (GstRingBuffer * buf)
{
GstAudioSink *sink;
GstAudioSinkClass *csink;
sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
csink = GST_AUDIO_SINK_GET_CLASS (sink);
/* unblock any pending writes to the audio device */
if (csink->reset) {
GST_DEBUG ("reset...");
csink->reset (sink);
GST_DEBUG ("reset done");
}
GST_DEBUG ("stop, waiting...");
GST_AUDIORING_BUFFER_WAIT (buf);
GST_DEBUG ("stoped");
return TRUE;
}
static guint
gst_audioringbuffer_delay (GstRingBuffer * buf)
{
GstAudioSink *sink;
GstAudioSinkClass *csink;
guint res = 0;
sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
csink = GST_AUDIO_SINK_GET_CLASS (sink);
if (csink->delay)
res = csink->delay (sink);
return res;
}
/* AudioSink signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0,
};
#define _do_init(bla) \
GST_DEBUG_CATEGORY_INIT (gst_audio_sink_debug, "audiosink", 0, "audiosink element");
GST_BOILERPLATE_FULL (GstAudioSink, gst_audio_sink, GstBaseAudioSink,
GST_TYPE_BASE_AUDIO_SINK, _do_init);
static GstRingBuffer *gst_audio_sink_create_ringbuffer (GstBaseAudioSink *
sink);
static void
gst_audio_sink_base_init (gpointer g_class)
{
}
static void
gst_audio_sink_class_init (GstAudioSinkClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSinkClass *gstbasesink_class;
GstBaseAudioSinkClass *gstbaseaudiosink_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesink_class = (GstBaseSinkClass *) klass;
gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass;
gstbaseaudiosink_class->create_ringbuffer =
GST_DEBUG_FUNCPTR (gst_audio_sink_create_ringbuffer);
}
static void
gst_audio_sink_init (GstAudioSink * audiosink)
{
}
static GstRingBuffer *
gst_audio_sink_create_ringbuffer (GstBaseAudioSink * sink)
{
GstRingBuffer *buffer;
GST_DEBUG ("creating ringbuffer");
buffer = g_object_new (GST_TYPE_AUDIORING_BUFFER, NULL);
GST_DEBUG ("created ringbuffer @%p", buffer);
return buffer;
}