mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-20 23:36:38 +00:00
9e8a11d3ce
Original commit message from CVS: use overridable ERROR_CFLAGS; more macro splitting
419 lines
11 KiB
C
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;
|
|
}
|