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2ff63e563b
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.
194 lines
5.5 KiB
C
194 lines
5.5 KiB
C
/* GStreamer
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* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
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* 2000 Wim Taymans <wtay@chello.be>
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*
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* audioclock.c: Clock for use by audio plugins
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#include "audioclock.h"
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static void gst_audio_clock_class_init (GstAudioClockClass *klass);
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static void gst_audio_clock_init (GstAudioClock *clock);
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static GstClockTime gst_audio_clock_get_internal_time (GstClock *clock);
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static GstClockReturn gst_audio_clock_id_wait_async (GstClock *clock,
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GstClockEntry *entry);
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static void gst_audio_clock_id_unschedule (GstClock *clock,
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GstClockEntry *entry);
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static GstSystemClockClass *parent_class = NULL;
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/* static guint gst_audio_clock_signals[LAST_SIGNAL] = { 0 }; */
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GType
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gst_audio_clock_get_type (void)
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{
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static GType clock_type = 0;
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if (!clock_type) {
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static const GTypeInfo clock_info = {
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sizeof (GstAudioClockClass),
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NULL,
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NULL,
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(GClassInitFunc) gst_audio_clock_class_init,
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NULL,
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NULL,
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sizeof (GstAudioClock),
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4,
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(GInstanceInitFunc) gst_audio_clock_init,
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NULL
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};
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clock_type = g_type_register_static (GST_TYPE_SYSTEM_CLOCK, "GstAudioClock",
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&clock_info, 0);
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}
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return clock_type;
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}
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static void
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gst_audio_clock_class_init (GstAudioClockClass *klass)
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{
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GObjectClass *gobject_class;
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GstObjectClass *gstobject_class;
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GstClockClass *gstclock_class;
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gobject_class = (GObjectClass*) klass;
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gstobject_class = (GstObjectClass*) klass;
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gstclock_class = (GstClockClass*) klass;
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parent_class = g_type_class_ref (GST_TYPE_SYSTEM_CLOCK);
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gstclock_class->get_internal_time = gst_audio_clock_get_internal_time;
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gstclock_class->wait_async = gst_audio_clock_id_wait_async;
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gstclock_class->unschedule = gst_audio_clock_id_unschedule;
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}
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static void
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gst_audio_clock_init (GstAudioClock *clock)
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{
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gst_object_set_name (GST_OBJECT (clock), "GstAudioClock");
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clock->prev1 = 0;
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clock->prev2 = 0;
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}
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GstClock*
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gst_audio_clock_new (gchar *name, GstAudioClockGetTimeFunc func, gpointer user_data)
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{
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GstAudioClock *aclock = GST_AUDIO_CLOCK (g_object_new (GST_TYPE_AUDIO_CLOCK, NULL));
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aclock->func = func;
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aclock->user_data = user_data;
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aclock->adjust = 0;
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return (GstClock*)aclock;
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}
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void
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gst_audio_clock_set_active (GstAudioClock *aclock, gboolean active)
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{
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GTimeVal timeval;
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GstClockTime time;
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GstClockTime audiotime;
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g_get_current_time (&timeval);
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time = GST_TIMEVAL_TO_TIME (timeval);
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audiotime = aclock->func ((GstClock*)aclock, aclock->user_data);
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if (active) {
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aclock->adjust = time - audiotime;
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}
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else {
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aclock->adjust = audiotime - time;
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}
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aclock->active = active;
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}
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static GstClockTime
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gst_audio_clock_get_internal_time (GstClock *clock)
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{
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GstAudioClock *aclock = GST_AUDIO_CLOCK (clock);
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if (aclock->active) {
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GstClockTime audiotime;
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audiotime = aclock->func (clock, aclock->user_data) + aclock->adjust;
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return audiotime;
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}
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else {
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GstClockTime time;
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GTimeVal timeval;
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g_get_current_time (&timeval);
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time = GST_TIMEVAL_TO_TIME (timeval);
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return time;
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}
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}
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void
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gst_audio_clock_update_time (GstAudioClock *aclock, GstClockTime time)
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{
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/* I don't know of a purpose in updating these; perhaps they can be removed */
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aclock->prev2 = aclock->prev1;
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aclock->prev1 = time;
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/* FIXME: the wait_async subsystem should be made threadsafe, but I don't want
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* to lock and unlock a mutex on every iteration... */
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while (aclock->async_entries) {
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GstClockEntry *entry = (GstClockEntry*)aclock->async_entries->data;
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if (entry->time > time)
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break;
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entry->func ((GstClock*)aclock, time, entry, entry->user_data);
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aclock->async_entries = g_slist_delete_link (aclock->async_entries,
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aclock->async_entries);
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/* do I need to free the entry? */
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}
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}
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static gint
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compare_clock_entries (GstClockEntry *entry1, GstClockEntry *entry2)
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{
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return entry1->time - entry2->time;
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}
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static GstClockReturn
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gst_audio_clock_id_wait_async (GstClock *clock, GstClockEntry *entry)
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{
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GstAudioClock *aclock = (GstAudioClock*)clock;
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aclock->async_entries = g_slist_insert_sorted (aclock->async_entries,
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entry,
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(GCompareFunc)compare_clock_entries);
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/* is this the proper return val? */
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return GST_CLOCK_EARLY;
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}
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static void
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gst_audio_clock_id_unschedule (GstClock *clock, GstClockEntry *entry)
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{
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GstAudioClock *aclock = (GstAudioClock*)clock;
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aclock->async_entries = g_slist_remove (aclock->async_entries,
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entry);
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}
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