gstreamer/gst/rawparse/gstaudioparse.c
Stefan Kost 15b9246617 Do not use short_description in section docs for elements. We extract them from element details and there will be war...
Original commit message from CVS:
* ext/dc1394/gstdc1394.c:
* ext/ivorbis/vorbisdec.c:
* ext/jack/gstjackaudiosink.c:
* ext/metadata/gstmetadatademux.c:
* ext/mythtv/gstmythtvsrc.c:
* ext/theora/theoradec.c:
* gst-libs/gst/app/gstappsink.c:
* gst/bayer/gstbayer2rgb.c:
* gst/deinterlace/gstdeinterlace.c:
* gst/rawparse/gstaudioparse.c:
* gst/rawparse/gstvideoparse.c:
* gst/rtpmanager/gstrtpbin.c:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/gstrtpssrcdemux.c:
* gst/selector/gstinputselector.c:
* gst/selector/gstoutputselector.c:
* gst/videosignal/gstvideoanalyse.c:
* gst/videosignal/gstvideodetect.c:
* gst/videosignal/gstvideomark.c:
* sys/oss4/oss4-mixer.c:
* sys/oss4/oss4-sink.c:
* sys/oss4/oss4-source.c:
Do not use short_description in section docs for elements. We extract
them from element details and there will be warnings if they differ.
Also fixing up the ChangeLog order.
2008-06-12 14:49:18 +00:00

318 lines
9.2 KiB
C

/* GStreamer
* Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
*
* gstaudioparse.c:
*
* 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.
*/
/**
* SECTION:element-audioparse
*
* Converts a byte stream into audio frames.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include "gstaudioparse.h"
typedef enum
{
GST_AUDIO_PARSE_FORMAT_INT,
GST_AUDIO_PARSE_FORMAT_FLOAT
} GstAudioParseFormat;
typedef enum
{
GST_AUDIO_PARSE_ENDIANNESS_LITTLE = 1234,
GST_AUDIO_PARSE_ENDIANNESS_BIG = 4321
} GstAudioParseEndianness;
static void gst_audio_parse_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_audio_parse_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstCaps *gst_audio_parse_get_caps (GstRawParse * rp);
static void gst_audio_parse_update_frame_size (GstAudioParse * ap);
GST_DEBUG_CATEGORY_STATIC (gst_audio_parse_debug);
#define GST_CAT_DEFAULT gst_audio_parse_debug
static const GstElementDetails gst_audio_parse_details =
GST_ELEMENT_DETAILS ("Audio Parse",
"Filter/Audio",
"Converts stream into audio frames",
"Sebastian Dröge <slomo@circular-chaos.org>");
enum
{
ARG_0,
ARG_FORMAT,
ARG_RATE,
ARG_CHANNELS,
ARG_ENDIANNESS,
ARG_WIDTH,
ARG_DEPTH,
ARG_SIGNED,
};
#define GST_AUDIO_PARSE_FORMAT (gst_audio_parse_format_get_type ())
static GType
gst_audio_parse_format_get_type (void)
{
static GType audio_parse_format_type = 0;
static const GEnumValue format_types[] = {
{GST_AUDIO_PARSE_FORMAT_INT, "Integer", "int"},
{GST_AUDIO_PARSE_FORMAT_FLOAT, "Floating Point", "float"},
{0, NULL, NULL}
};
if (!audio_parse_format_type) {
audio_parse_format_type =
g_enum_register_static ("GstAudioParseFormat", format_types);
}
return audio_parse_format_type;
}
#define GST_AUDIO_PARSE_ENDIANNESS (gst_audio_parse_endianness_get_type ())
static GType
gst_audio_parse_endianness_get_type (void)
{
static GType audio_parse_endianness_type = 0;
static const GEnumValue endian_types[] = {
{GST_AUDIO_PARSE_ENDIANNESS_LITTLE, "Little Endian", "little"},
{GST_AUDIO_PARSE_ENDIANNESS_BIG, "Big Endian", "big"},
{0, NULL, NULL}
};
if (!audio_parse_endianness_type) {
audio_parse_endianness_type =
g_enum_register_static ("GstAudioParseEndianness", endian_types);
}
return audio_parse_endianness_type;
}
GST_BOILERPLATE (GstAudioParse, gst_audio_parse, GstRawParse,
GST_TYPE_RAW_PARSE);
static void
gst_audio_parse_base_init (gpointer g_class)
{
GstRawParseClass *rp_class = GST_RAW_PARSE_CLASS (g_class);
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
GstCaps *caps;
GST_DEBUG_CATEGORY_INIT (gst_audio_parse_debug, "audioparse", 0,
"audioparse element");
gst_element_class_set_details (gstelement_class, &gst_audio_parse_details);
caps =
gst_caps_from_string ("audio/x-raw-int,"
" depth=(int) [ 1, 32 ],"
" width=(int) { 8, 16, 24, 32 },"
" endianness=(int) { LITTLE_ENDIAN, BIG_ENDIAN }, "
" signed=(bool) { TRUE, FALSE },"
" rate=(int) [ 1, MAX ],"
" channels=(int) [ 1, MAX ]; "
"audio/x-raw-float,"
" width=(int) { 32, 64 },"
" endianness=(int) { LITTLE_ENDIAN, BIG_ENDIAN }, "
" rate=(int)[1,MAX]," " channels=(int)[1,MAX]");
gst_raw_parse_class_set_src_pad_template (rp_class, caps);
gst_raw_parse_class_set_multiple_frames_per_buffer (rp_class, TRUE);
gst_caps_unref (caps);
}
static void
gst_audio_parse_class_init (GstAudioParseClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstRawParseClass *rp_class = GST_RAW_PARSE_CLASS (klass);
gobject_class->set_property = gst_audio_parse_set_property;
gobject_class->get_property = gst_audio_parse_get_property;
rp_class->get_caps = gst_audio_parse_get_caps;
g_object_class_install_property (gobject_class, ARG_FORMAT,
g_param_spec_enum ("format", "Format",
"Format of audio samples in raw stream", GST_AUDIO_PARSE_FORMAT,
GST_AUDIO_PARSE_FORMAT_INT, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_RATE,
g_param_spec_int ("rate", "Rate", "Rate of audio samples in raw stream",
1, INT_MAX, 44100, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_CHANNELS,
g_param_spec_int ("channels", "Channels",
"Number of channels in raw stream", 1, INT_MAX, 2,
G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_WIDTH,
g_param_spec_int ("width", "Width",
"Width of audio samples in raw stream", 1, INT_MAX, 16,
G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_DEPTH,
g_param_spec_int ("depth", "Depth",
"Depth of audio samples in raw stream", 1, INT_MAX, 16,
G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_SIGNED,
g_param_spec_boolean ("signed", "signed",
"Sign of audio samples in raw stream", TRUE, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_ENDIANNESS,
g_param_spec_enum ("endianness", "Endianness",
"Endianness of audio samples in raw stream",
GST_AUDIO_PARSE_ENDIANNESS, G_BYTE_ORDER, G_PARAM_READWRITE));
}
static void
gst_audio_parse_init (GstAudioParse * ap, GstAudioParseClass * g_class)
{
ap->format = GST_AUDIO_PARSE_FORMAT_INT;
ap->channels = 2;
ap->width = 16;
ap->depth = 16;
ap->signedness = TRUE;
ap->endianness = G_BYTE_ORDER;
gst_audio_parse_update_frame_size (ap);
gst_raw_parse_set_fps (GST_RAW_PARSE (ap), 44100, 1);
}
static void
gst_audio_parse_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioParse *ap = GST_AUDIO_PARSE (object);
g_return_if_fail (!gst_raw_parse_is_negotiated (GST_RAW_PARSE (ap)));
switch (prop_id) {
case ARG_FORMAT:
ap->format = g_value_get_enum (value);
break;
case ARG_RATE:
gst_raw_parse_set_fps (GST_RAW_PARSE (ap), g_value_get_int (value), 1);
break;
case ARG_CHANNELS:
ap->channels = g_value_get_int (value);
break;
case ARG_WIDTH:
ap->width = g_value_get_int (value);
break;
case ARG_DEPTH:
ap->depth = g_value_get_int (value);
break;
case ARG_SIGNED:
ap->signedness = g_value_get_boolean (value);
break;
case ARG_ENDIANNESS:
ap->endianness = g_value_get_enum (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
gst_audio_parse_update_frame_size (ap);
}
static void
gst_audio_parse_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstAudioParse *ap = GST_AUDIO_PARSE (object);
switch (prop_id) {
case ARG_FORMAT:
g_value_set_enum (value, ap->format);
break;
case ARG_RATE:{
gint fps_n, fps_d;
gst_raw_parse_get_fps (GST_RAW_PARSE (ap), &fps_n, &fps_d);
g_value_set_int (value, fps_n);
break;
}
case ARG_CHANNELS:
g_value_set_int (value, ap->channels);
break;
case ARG_WIDTH:
g_value_set_int (value, ap->width);
break;
case ARG_DEPTH:
g_value_set_int (value, ap->depth);
break;
case ARG_SIGNED:
g_value_set_boolean (value, ap->signedness);
break;
case ARG_ENDIANNESS:
g_value_set_enum (value, ap->endianness);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
void
gst_audio_parse_update_frame_size (GstAudioParse * ap)
{
gint framesize;
framesize = (ap->width / 8) * ap->channels;
gst_raw_parse_set_framesize (GST_RAW_PARSE (ap), framesize);
}
static GstCaps *
gst_audio_parse_get_caps (GstRawParse * rp)
{
GstAudioParse *ap = GST_AUDIO_PARSE (rp);
GstCaps *caps;
gint fps_n, fps_d;
gst_raw_parse_get_fps (rp, &fps_n, &fps_d);
if (ap->format == GST_AUDIO_PARSE_FORMAT_INT) {
caps = gst_caps_new_simple ("audio/x-raw-int",
"rate", G_TYPE_INT, fps_n,
"channels", G_TYPE_INT, ap->channels,
"width", G_TYPE_INT, ap->width,
"depth", G_TYPE_INT, ap->depth,
"signed", G_TYPE_BOOLEAN, ap->signedness,
"endianness", G_TYPE_INT, ap->endianness, NULL);
} else {
caps = gst_caps_new_simple ("audio/x-raw-float",
"rate", G_TYPE_INT, fps_n,
"channels", G_TYPE_INT, ap->channels,
"width", G_TYPE_INT, ap->width,
"endianness", G_TYPE_INT, ap->endianness, NULL);
}
return caps;
}