gstreamer/ext/resindvd/rsnaudiomunge.c

369 lines
11 KiB
C
Raw Normal View History

/* 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);
static GstElementDetails element_details = {
"RsnAudioMunge",
"Audio/Filter",
"Resin DVD audio stream regulator",
"Jan Schmidt <thaytan@noraisin.net>"
};
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 (element_class, &element_details);
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;
}