gstreamer/gst-libs/gst/audio/audioclock.c
Andy Wingo 7c6f49bf6b actually recurse into sndfile if we are able big ladspa cleanups, mainly to comply with the buffer-frames caps proper...
Original commit message from CVS:
* actually recurse into sndfile if we are able
* big ladspa cleanups, mainly to comply with the buffer-frames caps property, but also general
cleanups
- the samplerate prop is gone, if you want to set it explicitly (as in for get-based plugins)
you need to use a filtered connection, just like with buffer-frames
* big float2int and int2float changes for buffer-frames compatibility - I think it's quite a bit
simpler
* make the ossclock general, add it to gstaudio, and use it in sndfile as well

i need to update mimetypes, but that's coming soon. there are some other plugins that don't
support buffer-frames, i guess i need to get around to fixing them as well.
2003-07-16 16:08:13 +00:00

194 lines
5.5 KiB
C

/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000 Wim Taymans <wtay@chello.be>
*
* audioclock.c: Clock for use by audio plugins
*
* 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 "audioclock.h"
static void gst_audio_clock_class_init (GstAudioClockClass *klass);
static void gst_audio_clock_init (GstAudioClock *clock);
static GstClockTime gst_audio_clock_get_internal_time (GstClock *clock);
static GstClockReturn gst_audio_clock_id_wait_async (GstClock *clock,
GstClockEntry *entry);
static void gst_audio_clock_id_unschedule (GstClock *clock,
GstClockEntry *entry);
static GstSystemClockClass *parent_class = NULL;
/* static guint gst_audio_clock_signals[LAST_SIGNAL] = { 0 }; */
GType
gst_audio_clock_get_type (void)
{
static GType clock_type = 0;
if (!clock_type) {
static const GTypeInfo clock_info = {
sizeof (GstAudioClockClass),
NULL,
NULL,
(GClassInitFunc) gst_audio_clock_class_init,
NULL,
NULL,
sizeof (GstAudioClock),
4,
(GInstanceInitFunc) gst_audio_clock_init,
NULL
};
clock_type = g_type_register_static (GST_TYPE_SYSTEM_CLOCK, "GstAudioClock",
&clock_info, 0);
}
return clock_type;
}
static void
gst_audio_clock_class_init (GstAudioClockClass *klass)
{
GObjectClass *gobject_class;
GstObjectClass *gstobject_class;
GstClockClass *gstclock_class;
gobject_class = (GObjectClass*) klass;
gstobject_class = (GstObjectClass*) klass;
gstclock_class = (GstClockClass*) klass;
parent_class = g_type_class_ref (GST_TYPE_SYSTEM_CLOCK);
gstclock_class->get_internal_time = gst_audio_clock_get_internal_time;
gstclock_class->wait_async = gst_audio_clock_id_wait_async;
gstclock_class->unschedule = gst_audio_clock_id_unschedule;
}
static void
gst_audio_clock_init (GstAudioClock *clock)
{
gst_object_set_name (GST_OBJECT (clock), "GstAudioClock");
clock->prev1 = 0;
clock->prev2 = 0;
}
GstClock*
gst_audio_clock_new (gchar *name, GstAudioClockGetTimeFunc func, gpointer user_data)
{
GstAudioClock *aclock = GST_AUDIO_CLOCK (g_object_new (GST_TYPE_AUDIO_CLOCK, NULL));
aclock->func = func;
aclock->user_data = user_data;
aclock->adjust = 0;
return (GstClock*)aclock;
}
void
gst_audio_clock_set_active (GstAudioClock *aclock, gboolean active)
{
GTimeVal timeval;
GstClockTime time;
GstClockTime audiotime;
g_get_current_time (&timeval);
time = GST_TIMEVAL_TO_TIME (timeval);
audiotime = aclock->func ((GstClock*)aclock, aclock->user_data);
if (active) {
aclock->adjust = time - audiotime;
}
else {
aclock->adjust = audiotime - time;
}
aclock->active = active;
}
static GstClockTime
gst_audio_clock_get_internal_time (GstClock *clock)
{
GstAudioClock *aclock = GST_AUDIO_CLOCK (clock);
if (aclock->active) {
GstClockTime audiotime;
audiotime = aclock->func (clock, aclock->user_data) + aclock->adjust;
return audiotime;
}
else {
GstClockTime time;
GTimeVal timeval;
g_get_current_time (&timeval);
time = GST_TIMEVAL_TO_TIME (timeval);
return time;
}
}
void
gst_audio_clock_update_time (GstAudioClock *aclock, GstClockTime time)
{
/* I don't know of a purpose in updating these; perhaps they can be removed */
aclock->prev2 = aclock->prev1;
aclock->prev1 = time;
/* FIXME: the wait_async subsystem should be made threadsafe, but I don't want
* to lock and unlock a mutex on every iteration... */
while (aclock->async_entries) {
GstClockEntry *entry = (GstClockEntry*)aclock->async_entries->data;
if (entry->time > time)
break;
entry->func ((GstClock*)aclock, time, entry, entry->user_data);
aclock->async_entries = g_slist_delete_link (aclock->async_entries,
aclock->async_entries);
/* do I need to free the entry? */
}
}
static gint
compare_clock_entries (GstClockEntry *entry1, GstClockEntry *entry2)
{
return entry1->time - entry2->time;
}
static GstClockReturn
gst_audio_clock_id_wait_async (GstClock *clock, GstClockEntry *entry)
{
GstAudioClock *aclock = (GstAudioClock*)clock;
aclock->async_entries = g_slist_insert_sorted (aclock->async_entries,
entry,
(GCompareFunc)compare_clock_entries);
/* is this the proper return val? */
return GST_CLOCK_EARLY;
}
static void
gst_audio_clock_id_unschedule (GstClock *clock, GstClockEntry *entry)
{
GstAudioClock *aclock = (GstAudioClock*)clock;
aclock->async_entries = g_slist_remove (aclock->async_entries,
entry);
}