#endif
#include <string.h>
+#include <gst/video/video.h>
#include "gstmpegdefs.h"
#include "gstmpegdemux.h"
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);
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
}
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;
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
"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;
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);
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 {
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
+++ /dev/null
-/* 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;
-}
projects / platform / upstream / gstreamer.git / commitdiff