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resindvd: Add some GAP event stuff to make still-frames start to work

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And remove the dead audiomunge element. It isn't needed now, we just
send GAP events
This commit is contained in:
Jan Schmidt 2012-08-31 08:46:45 -07:00
parent 1218cff3dc
commit 211828979b
8 changed files with 34 additions and 486 deletions
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4
ext/resindvd/Makefile.am
View file

@ -12,8 +12,7 @@ libgstresindvd_la_SOURCES = \
gstpesfilter.c \
rsninputselector.c \
# rsnparsetter.c \
# rsnwrappedbuffer.c \
# rsnaudiomunge.c
# rsnwrappedbuffer.c
libgstresindvd_la_CFLAGS = $(GST_PLUGINS_BAD_CFLAGS) \
$(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) \
@ -25,7 +24,6 @@ libgstresindvd_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstresindvd_la_LIBTOOLFLAGS = --tag=disable-static
noinst_HEADERS = resindvdbin.h \
rsnaudiomunge.h \
rsndec.h \
rsninputselector.h \
resindvdsrc.h \

59
ext/resindvd/gstmpegdemux.c
View file

@ -24,6 +24,7 @@
#endif
#include <string.h>
#include <gst/video/video.h>
#include "gstmpegdefs.h"
#include "gstmpegdemux.h"
@ -122,7 +123,7 @@ static gboolean gst_flups_demux_src_query (GstPad * pad, GstObject * parent,
static GstStateChangeReturn gst_flups_demux_change_state (GstElement * element,
GstStateChange transition);
static inline void gst_flups_demux_send_segment_updates (GstFluPSDemux * demux,
static inline void gst_flups_demux_send_gap_updates (GstFluPSDemux * demux,
GstClockTime new_time);
static inline void gst_flups_demux_clear_times (GstFluPSDemux * demux);
@ -770,6 +771,7 @@ gst_flups_demux_flush (GstFluPSDemux * demux)
demux->current_scr = G_MAXUINT64;
demux->bytes_since_scr = 0;
demux->scr_adjust = GSTTIME_TO_MPEGTIME (SCR_MUNGE);
demux->in_still = FALSE;
}
static inline void
@ -788,8 +790,7 @@ gst_flups_demux_clear_times (GstFluPSDemux * demux)
}
static inline void
gst_flups_demux_send_segment_updates (GstFluPSDemux * demux,
GstClockTime new_time)
gst_flups_demux_send_gap_updates (GstFluPSDemux * demux, GstClockTime new_time)
{
gint id;
GstEvent *event = NULL;
@ -807,35 +808,20 @@ gst_flups_demux_send_segment_updates (GstFluPSDemux * demux,
stream->last_ts = demux->src_segment.start;
if (stream->last_ts + stream->segment_thresh < new_time) {
#if 0
g_print ("Segment update to pad %s time %" GST_TIME_FORMAT " stop now %"
GST_TIME_FORMAT " position %" GST_TIME_FORMAT "\n",
GST_PAD_NAME (stream->pad), GST_TIME_ARGS (new_time),
GST_TIME_ARGS (demux->src_segment.stop),
GST_TIME_ARGS (demux->src_segment.position));
g_print ("Gap event update to pad %s from time %" GST_TIME_FORMAT
" to %" GST_TIME_FORMAT "\n", GST_PAD_NAME (stream->pad),
GST_TIME_ARGS (stream->last_ts), GST_TIME_ARGS (new_time));
#endif
GST_DEBUG_OBJECT (demux,
"Segment update to pad %s time %" GST_TIME_FORMAT,
GST_PAD_NAME (stream->pad), GST_TIME_ARGS (new_time));
if (event == NULL) {
GstSegment segment;
gst_segment_init (&segment, GST_FORMAT_TIME);
segment.rate = demux->src_segment.rate;
segment.applied_rate = demux->src_segment.applied_rate;
segment.start = new_time;
segment.stop = demux->src_segment.stop;
segment.time =
demux->src_segment.time + (new_time - demux->src_segment.start);
event = gst_event_new_segment (&segment);
}
gst_event_ref (event);
"Gap event update to pad %s from time %" GST_TIME_FORMAT " to %"
GST_TIME_FORMAT, GST_PAD_NAME (stream->pad),
GST_TIME_ARGS (stream->last_ts), GST_TIME_ARGS (new_time));
event = gst_event_new_gap (stream->last_ts, new_time - stream->last_ts);
gst_pad_push_event (stream->pad, event);
stream->last_seg_start = stream->last_ts = new_time;
}
}
}
if (event)
gst_event_unref (event);
}
static inline void
@ -934,7 +920,6 @@ gst_flups_demux_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
"demux: received new segment start %" G_GINT64_FORMAT " stop %"
G_GINT64_FORMAT " time %" G_GINT64_FORMAT, start, stop, time);
adjust = base - start + SCR_MUNGE;
start = base + SCR_MUNGE;
@ -961,9 +946,14 @@ gst_flups_demux_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
demux->src_segment.rate = segment->rate;
demux->src_segment.applied_rate = segment->applied_rate;
demux->src_segment.format = segment->format;
demux->src_segment.start = segment->start;
demux->src_segment.stop = segment->stop;
demux->src_segment.time = segment->time;
demux->src_segment.start = start;
demux->src_segment.stop = stop;
demux->src_segment.time = time;
if (demux->in_still && stop != -1) {
/* Generate gap buffers, due to closing segment from a still-frame */
gst_flups_demux_send_gap_updates (demux, stop);
}
gst_event_unref (event);
event = gst_event_new_segment (&demux->src_segment);
@ -983,8 +973,15 @@ gst_flups_demux_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
case GST_EVENT_CUSTOM_DOWNSTREAM_OOB:
{
const GstStructure *structure = gst_event_get_structure (event);
gboolean in_still;
if (structure != NULL
if (gst_video_event_parse_still_frame (event, &in_still)) {
/* Remember the still-frame state, so we can generate a pre-roll
* GAP event when a segment event arrives */
demux->in_still = in_still;
GST_INFO_OBJECT (demux, "still-state now %d", demux->in_still);
gst_flups_demux_send_event (demux, event);
} else if (structure != NULL
&& gst_structure_has_name (structure, "application/x-gst-dvd")) {
res = gst_flups_demux_handle_dvd_event (demux, event);
} else {
@ -1473,7 +1470,7 @@ gst_flups_demux_parse_pack_start (GstFluPSDemux * demux)
if (new_time != GST_CLOCK_TIME_NONE) {
// g_print ("SCR now %" GST_TIME_FORMAT "\n", GST_TIME_ARGS (new_time));
gst_segment_set_position (&demux->src_segment, GST_FORMAT_TIME, new_time);
gst_flups_demux_send_segment_updates (demux, new_time);
gst_flups_demux_send_gap_updates (demux, new_time);
}
/* Reset the bytes_since_scr value to count the data remaining in the

2
ext/resindvd/gstmpegdemux.h
View file

@ -97,7 +97,7 @@ struct _GstFluPSDemux {
GstPad * sinkpad;
gboolean random_access; /* If we operate in pull mode */
gboolean flushing;
gboolean in_still;
GstAdapter * adapter;
GstAdapter * rev_adapter;

20
ext/resindvd/resindvdbin.c
View file

@ -466,28 +466,12 @@ create_elements (RsnDvdBin * dvdbin)
RSN_TYPE_INPUT_SELECTOR, "audioselect", "Audio stream selector"))
return FALSE;
if (!try_create_piece (dvdbin, DVD_ELEM_AUD_MUNGE, "identity",
0 /* RSN_TYPE_AUDIOMUNGE */ , "audioearlymunge",
"Audio output filter"))
return FALSE;
if (!try_create_piece (dvdbin, DVD_ELEM_AUDDEC, NULL,
RSN_TYPE_AUDIODEC, "auddec", "audio decoder"))
return FALSE;
src = gst_element_get_static_pad (dvdbin->pieces[DVD_ELEM_AUD_MUNGE], "src");
sink = gst_element_get_static_pad (dvdbin->pieces[DVD_ELEM_AUDDEC], "sink");
if (src == NULL || sink == NULL)
goto failed_aud_connect;
if (GST_PAD_LINK_FAILED (gst_pad_link (src, sink)))
goto failed_aud_connect;
gst_object_unref (sink);
gst_object_unref (src);
src = sink = NULL;
src = gst_element_get_static_pad (dvdbin->pieces[DVD_ELEM_AUD_SELECT], "src");
sink =
gst_element_get_static_pad (dvdbin->pieces[DVD_ELEM_AUD_MUNGE], "sink");
sink = gst_element_get_static_pad (dvdbin->pieces[DVD_ELEM_AUDDEC], "sink");
if (src == NULL || sink == NULL)
goto failed_aud_connect;
if (GST_PAD_LINK_FAILED (gst_pad_link (src, sink)))
@ -701,7 +685,7 @@ demux_pad_added (GstElement * element, GstPad * pad, RsnDvdBin * dvdbin)
gst_element_get_request_pad (dvdbin->pieces[DVD_ELEM_SPU_SELECT],
"sink_%u");
skip_mq = TRUE;
} else if (can_sink_caps (dvdbin->pieces[DVD_ELEM_AUD_MUNGE], caps)) {
} else if (can_sink_caps (dvdbin->pieces[DVD_ELEM_AUDDEC], caps)) {
GST_LOG_OBJECT (dvdbin, "Found audio pad w/ caps %" GST_PTR_FORMAT, caps);
dest_pad =
gst_element_get_request_pad (dvdbin->pieces[DVD_ELEM_AUD_SELECT],

3
ext/resindvd/resindvdbin.h
View file

@ -49,8 +49,7 @@ typedef struct _RsnDvdBinClass RsnDvdBinClass;
#define DVD_ELEM_VIDQ 7
#define DVD_ELEM_SPU_SELECT 8
#define DVD_ELEM_AUD_SELECT 9
#define DVD_ELEM_AUD_MUNGE 10
#define DVD_ELEM_LAST 11
#define DVD_ELEM_LAST 10
struct _RsnDvdBin
{

2
ext/resindvd/resindvdsrc.c
View file

@ -630,7 +630,7 @@ rsn_dvdsrc_do_still (resinDvdSrc * src, int duration)
* event, then sleep */
still_event = gst_video_event_new_still_frame (TRUE);
segment->position = src->cur_end_ts;
segment->stop = segment->position = src->cur_end_ts;
seg_event = gst_event_new_segment (segment);

369
ext/resindvd/rsnaudiomunge.c
View file

@ -1,369 +0,0 @@
/* GStreamer
* Copyright (C) 2008 Jan Schmidt <thaytan@noraisin.net>
*
* 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.
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include <string.h>
#include <gst/gst.h>
#include <gst/video/video.h>
#include "rsnaudiomunge.h"
GST_DEBUG_CATEGORY_STATIC (rsn_audiomunge_debug);
#define GST_CAT_DEFAULT rsn_audiomunge_debug
#define AUDIO_FILL_THRESHOLD (GST_SECOND/5)
/* Filter signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_SILENT
};
/* the capabilities of the inputs and outputs.
*
* describe the real formats here.
*/
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("ANY")
);
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("ANY")
);
G_DEFINE_TYPE (RsnAudioMunge, rsn_audiomunge, GST_TYPE_ELEMENT);
static void rsn_audiomunge_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void rsn_audiomunge_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean rsn_audiomunge_set_caps (GstPad * pad, GstCaps * caps);
static GstFlowReturn rsn_audiomunge_chain (GstPad * pad, GstBuffer * buf);
static gboolean rsn_audiomunge_sink_event (GstPad * pad, GstEvent * event);
static GstStateChangeReturn
rsn_audiomunge_change_state (GstElement * element, GstStateChange transition);
static void
rsn_audiomunge_class_init (RsnAudioMungeClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) (klass);
GstElementClass *element_class = (GstElementClass *) (klass);
GST_DEBUG_CATEGORY_INIT (rsn_audiomunge_debug, "rsnaudiomunge",
0, "ResinDVD audio stream regulator");
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_template));
gst_element_class_set_details_simple (element_class, "RsnAudioMunge",
"Audio/Filter",
"Resin DVD audio stream regulator", "Jan Schmidt <thaytan@noraisin.net>");
gobject_class->set_property = rsn_audiomunge_set_property;
gobject_class->get_property = rsn_audiomunge_get_property;
element_class->change_state = rsn_audiomunge_change_state;
}
static void
rsn_audiomunge_init (RsnAudioMunge * munge)
{
munge->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
gst_pad_set_getcaps_function (munge->sinkpad,
GST_DEBUG_FUNCPTR (gst_pad_proxy_getcaps));
gst_pad_set_chain_function (munge->sinkpad,
GST_DEBUG_FUNCPTR (rsn_audiomunge_chain));
gst_pad_set_event_function (munge->sinkpad,
GST_DEBUG_FUNCPTR (rsn_audiomunge_sink_event));
gst_element_add_pad (GST_ELEMENT (munge), munge->sinkpad);
munge->srcpad = gst_pad_new_from_static_template (&src_template, "src");
gst_pad_set_getcaps_function (munge->srcpad,
GST_DEBUG_FUNCPTR (gst_pad_proxy_getcaps));
gst_element_add_pad (GST_ELEMENT (munge), munge->srcpad);
}
static void
rsn_audiomunge_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
//RsnAudioMunge *munge = RSN_AUDIOMUNGE (object);
switch (prop_id) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
rsn_audiomunge_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
//RsnAudioMunge *munge = RSN_AUDIOMUNGE (object);
switch (prop_id) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
rsn_audiomunge_set_caps (GstPad * pad, GstCaps * caps)
{
RsnAudioMunge *munge = RSN_AUDIOMUNGE (gst_pad_get_parent (pad));
GstPad *otherpad;
gboolean ret;
g_return_val_if_fail (munge != NULL, FALSE);
otherpad = (pad == munge->srcpad) ? munge->sinkpad : munge->srcpad;
gst_object_unref (munge);
return ret;
}
static void
rsn_audiomunge_reset (RsnAudioMunge * munge)
{
munge->have_audio = FALSE;
munge->in_still = FALSE;
gst_segment_init (&munge->sink_segment, GST_FORMAT_TIME);
}
static GstFlowReturn
rsn_audiomunge_chain (GstPad * pad, GstBuffer * buf)
{
RsnAudioMunge *munge = RSN_AUDIOMUNGE (GST_OBJECT_PARENT (pad));
if (!munge->have_audio) {
GST_INFO_OBJECT (munge,
"First audio after flush has TS %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
}
munge->have_audio = TRUE;
/* just push out the incoming buffer without touching it */
return gst_pad_push (munge->srcpad, buf);
}
/* Create and send a silence buffer downstream */
static GstFlowReturn
rsn_audiomunge_make_audio (RsnAudioMunge * munge,
GstClockTime start, GstClockTime fill_time)
{
GstFlowReturn ret;
GstBuffer *audio_buf;
GstCaps *caps;
guint buf_size;
/* Just generate a 48khz stereo buffer for now */
/* FIXME: Adapt to the allowed formats, according to the currently
* plugged decoder, or at least add a source pad that accepts the
* caps we're outputting if the upstream decoder does not */
#if 0
caps =
gst_caps_from_string
("audio/x-raw-int,rate=48000,channels=2,width=16,depth=16,signed=(boolean)true,endianness=4321");
buf_size = 4 * (48000 * fill_time / GST_SECOND);
#else
caps = gst_caps_from_string ("audio/x-raw-float, endianness=(int)1234,"
"width=(int)32, channels=(int)2, rate=(int)48000");
buf_size = 2 * 4 * (48000 * fill_time / GST_SECOND);
#endif
audio_buf = gst_buffer_new_and_alloc (buf_size);
gst_buffer_set_caps (audio_buf, caps);
gst_caps_unref (caps);
GST_BUFFER_TIMESTAMP (audio_buf) = start;
GST_BUFFER_DURATION (audio_buf) = fill_time;
GST_BUFFER_FLAG_SET (audio_buf, GST_BUFFER_FLAG_DISCONT);
memset (GST_BUFFER_DATA (audio_buf), 0, buf_size);
GST_LOG_OBJECT (munge, "Sending %u bytes (%" GST_TIME_FORMAT
") of audio data with TS %" GST_TIME_FORMAT,
buf_size, GST_TIME_ARGS (fill_time), GST_TIME_ARGS (start));
ret = gst_pad_push (munge->srcpad, audio_buf);
return ret;
}
static gboolean
rsn_audiomunge_sink_event (GstPad * pad, GstEvent * event)
{
gboolean ret = FALSE;
RsnAudioMunge *munge = RSN_AUDIOMUNGE (gst_pad_get_parent (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CAPS:
{
GstCaps *caps;
gst_event_parse_caps (event, &caps);
ret = gst_pad_set_caps (munge->src_pad, caps);
gst_event_unref (caps);
}
case GST_EVENT_FLUSH_STOP:
rsn_audiomunge_reset (munge);
ret = gst_pad_push_event (munge->srcpad, event);
break;
case GST_EVENT_NEWSEGMENT:
{
GstSegment *segment;
gboolean update;
GstFormat format;
gdouble rate, arate;
gint64 start, stop, time;
gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
&start, &stop, &time);
/* we need TIME format */
if (format != GST_FORMAT_TIME)
goto newseg_wrong_format;
/* now configure the values */
segment = &munge->sink_segment;
gst_segment_set_newsegment_full (segment, update,
rate, arate, format, start, stop, time);
/*
* FIXME:
* If this is a segment update and accum >= threshold,
* or we're in a still frame and there's been no audio received,
* then we need to generate some audio data.
*
* If caused by a segment start update (time advancing in a gap) adjust
* the new-segment and send the buffer.
*
* Otherwise, send the buffer before the newsegment, so that it appears
* in the closing segment.
*/
if (!update) {
GST_DEBUG_OBJECT (munge,
"Sending newsegment: update %d start %" GST_TIME_FORMAT " stop %"
GST_TIME_FORMAT " accum now %" GST_TIME_FORMAT, update,
GST_TIME_ARGS (start), GST_TIME_ARGS (stop),
GST_TIME_ARGS (segment->accum));
ret = gst_pad_push_event (munge->srcpad, event);
}
if (!munge->have_audio) {
if ((update && segment->accum >= AUDIO_FILL_THRESHOLD)
|| munge->in_still) {
GST_DEBUG_OBJECT (munge,
"Sending audio fill with ts %" GST_TIME_FORMAT ": accum = %"
GST_TIME_FORMAT " still-state=%d", GST_TIME_ARGS (segment->start),
GST_TIME_ARGS (segment->accum), munge->in_still);
/* Just generate a 200ms silence buffer for now. FIXME: Fill the gap */
if (rsn_audiomunge_make_audio (munge, segment->start,
GST_SECOND / 5) == GST_FLOW_OK)
munge->have_audio = TRUE;
} else {
GST_LOG_OBJECT (munge, "Not sending audio fill buffer: "
"Not segment update, or segment accum below thresh: accum = %"
GST_TIME_FORMAT, GST_TIME_ARGS (segment->accum));
}
}
if (update) {
GST_DEBUG_OBJECT (munge,
"Sending newsegment: update %d start %" GST_TIME_FORMAT " stop %"
GST_TIME_FORMAT " accum now %" GST_TIME_FORMAT, update,
GST_TIME_ARGS (start), GST_TIME_ARGS (stop),
GST_TIME_ARGS (segment->accum));
ret = gst_pad_push_event (munge->srcpad, event);
}
break;
}
case GST_EVENT_CUSTOM_DOWNSTREAM:
{
gboolean in_still;
if (gst_video_event_parse_still_frame (event, &in_still)) {
/* Remember the still-frame state, so we can generate a pre-roll
* buffer when a new-segment arrives */
munge->in_still = in_still;
GST_INFO_OBJECT (munge, "AUDIO MUNGE: still-state now %d",
munge->in_still);
}
ret = gst_pad_push_event (munge->srcpad, event);
break;
}
default:
ret = gst_pad_push_event (munge->srcpad, event);
break;
}
gst_object_unref (munge);
return ret;
newseg_wrong_format:
GST_DEBUG_OBJECT (munge, "received non TIME newsegment");
gst_event_unref (event);
gst_object_unref (munge);
return FALSE;
}
static GstStateChangeReturn
rsn_audiomunge_change_state (GstElement * element, GstStateChange transition)
{
RsnAudioMunge *munge = RSN_AUDIOMUNGE (element);
GstStateChangeReturn ret;
if (transition == GST_STATE_CHANGE_READY_TO_PAUSED)
rsn_audiomunge_reset (munge);
ret =
GST_ELEMENT_CLASS (rsn_audiomunge_parent_class)->change_state (element,
transition);
return ret;
}

61
ext/resindvd/rsnaudiomunge.h
View file

@ -1,61 +0,0 @@
/* GStreamer
* Copyright (C) 2008 Jan Schmidt <thaytan@noraisin.net>
*
* 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.
*/
#ifndef __RSNAUDIOMUNGE_H__
#define __RSNAUDIOMUNGE_H__
#include <gst/gst.h>
G_BEGIN_DECLS
/* #defines don't like whitespacey bits */
#define RSN_TYPE_AUDIOMUNGE (rsn_audiomunge_get_type())
#define RSN_AUDIOMUNGE(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),RSN_TYPE_AUDIOMUNGE,RsnAudioMunge))
#define RSN_AUDIOMUNGE_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),RSN_TYPE_AUDIOMUNGE,RsnAudioMungeClass))
#define RSN_IS_AUDIOMUNGE(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),RSN_TYPE_AUDIOMUNGE))
#define RSN_IS_AUDIOMUNGE_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),RSN_TYPE_AUDIOMUNGE))
typedef struct _RsnAudioMunge RsnAudioMunge;
typedef struct _RsnAudioMungeClass RsnAudioMungeClass;
struct _RsnAudioMunge
{
GstElement element;
GstPad *sinkpad, *srcpad;
GstSegment sink_segment;
gboolean have_audio;
gboolean in_still;
};
struct _RsnAudioMungeClass
{
GstElementClass parent_class;
};
GType rsn_audiomunge_get_type (void);
G_END_DECLS
#endif /* __RSNAUDIOMUNGE_H__ */