gstreamer/ext/resindvd/rsnaudiomunge.c
Jan Schmidt 4d5a48db97 resindvd: Create all the audio streams of all formats
Enable creation of the demuxer pads for all audio streams, even types we
don't yet support. This means that unsupported types (LPCM, DTS) are output,
but not linked to anything yet. If only unsupported streams are available,
the user hears silence instead of having the pipeline not pre-roll correctly.

This is a prerequisite for hooking up the automatic decoder switching.
2009-05-27 22:57:03 +01:00

386 lines
11 KiB
C

/* 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 "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 void
rsn_audiomunge_handle_dvd_event (RsnAudioMunge * munge, GstEvent * event)
{
const GstStructure *s;
const gchar *event_type;
s = gst_event_get_structure (event);
event_type = gst_structure_get_string (s, "event");
if (event_type == NULL)
return;
if (strcmp (event_type, "dvd-still") == 0) {
gboolean in_still;
if (!gst_structure_get_boolean (s, "still-state", &in_still))
return;
/* 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);
}
}
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:
{
const GstStructure *s = gst_event_get_structure (event);
if (s && gst_structure_has_name (s, "application/x-gst-dvd"))
rsn_audiomunge_handle_dvd_event (munge, event);
ret = gst_pad_push_event (munge->srcpad, event);
break;
}
default:
ret = gst_pad_push_event (munge->srcpad, event);
break;
}
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;
}