ext/resindvd/rsnaudiomunge.c - mtk-gst-plugins-bad - Git at Google

 /* 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_setcaps_function (munge->sinkpad,
       GST_DEBUG_FUNCPTR (rsn_audiomunge_set_caps));
   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);

   ret = gst_pad_set_caps (otherpad, caps);
   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_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;
 }
	/* 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_setcaps_function (munge->sinkpad,
	GST_DEBUG_FUNCPTR (rsn_audiomunge_set_caps));
	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);

	ret = gst_pad_set_caps (otherpad, caps);
	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_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;
	}