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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.
318 lines
9.2 KiB
C
318 lines
9.2 KiB
C
/* GStreamer
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* Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
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*
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* gstaudioparse.c:
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:element-audioparse
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*
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* Converts a byte stream into audio frames.
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include "gstaudioparse.h"
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typedef enum
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{
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GST_AUDIO_PARSE_FORMAT_INT,
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GST_AUDIO_PARSE_FORMAT_FLOAT
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} GstAudioParseFormat;
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typedef enum
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{
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GST_AUDIO_PARSE_ENDIANNESS_LITTLE = 1234,
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GST_AUDIO_PARSE_ENDIANNESS_BIG = 4321
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} GstAudioParseEndianness;
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static void gst_audio_parse_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_audio_parse_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static GstCaps *gst_audio_parse_get_caps (GstRawParse * rp);
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static void gst_audio_parse_update_frame_size (GstAudioParse * ap);
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GST_DEBUG_CATEGORY_STATIC (gst_audio_parse_debug);
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#define GST_CAT_DEFAULT gst_audio_parse_debug
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static const GstElementDetails gst_audio_parse_details =
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GST_ELEMENT_DETAILS ("Audio Parse",
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"Filter/Audio",
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"Converts stream into audio frames",
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"Sebastian Dröge <slomo@circular-chaos.org>");
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enum
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{
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ARG_0,
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ARG_FORMAT,
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ARG_RATE,
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ARG_CHANNELS,
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ARG_ENDIANNESS,
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ARG_WIDTH,
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ARG_DEPTH,
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ARG_SIGNED,
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};
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#define GST_AUDIO_PARSE_FORMAT (gst_audio_parse_format_get_type ())
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static GType
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gst_audio_parse_format_get_type (void)
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{
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static GType audio_parse_format_type = 0;
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static const GEnumValue format_types[] = {
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{GST_AUDIO_PARSE_FORMAT_INT, "Integer", "int"},
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{GST_AUDIO_PARSE_FORMAT_FLOAT, "Floating Point", "float"},
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{0, NULL, NULL}
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};
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if (!audio_parse_format_type) {
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audio_parse_format_type =
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g_enum_register_static ("GstAudioParseFormat", format_types);
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}
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return audio_parse_format_type;
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}
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#define GST_AUDIO_PARSE_ENDIANNESS (gst_audio_parse_endianness_get_type ())
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static GType
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gst_audio_parse_endianness_get_type (void)
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{
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static GType audio_parse_endianness_type = 0;
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static const GEnumValue endian_types[] = {
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{GST_AUDIO_PARSE_ENDIANNESS_LITTLE, "Little Endian", "little"},
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{GST_AUDIO_PARSE_ENDIANNESS_BIG, "Big Endian", "big"},
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{0, NULL, NULL}
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};
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if (!audio_parse_endianness_type) {
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audio_parse_endianness_type =
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g_enum_register_static ("GstAudioParseEndianness", endian_types);
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}
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return audio_parse_endianness_type;
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}
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GST_BOILERPLATE (GstAudioParse, gst_audio_parse, GstRawParse,
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GST_TYPE_RAW_PARSE);
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static void
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gst_audio_parse_base_init (gpointer g_class)
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{
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GstRawParseClass *rp_class = GST_RAW_PARSE_CLASS (g_class);
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GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
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GstCaps *caps;
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GST_DEBUG_CATEGORY_INIT (gst_audio_parse_debug, "audioparse", 0,
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"audioparse element");
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gst_element_class_set_details (gstelement_class, &gst_audio_parse_details);
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caps =
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gst_caps_from_string ("audio/x-raw-int,"
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" depth=(int) [ 1, 32 ],"
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" width=(int) { 8, 16, 24, 32 },"
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" endianness=(int) { LITTLE_ENDIAN, BIG_ENDIAN }, "
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" signed=(bool) { TRUE, FALSE },"
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" rate=(int) [ 1, MAX ],"
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" channels=(int) [ 1, MAX ]; "
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"audio/x-raw-float,"
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" width=(int) { 32, 64 },"
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" endianness=(int) { LITTLE_ENDIAN, BIG_ENDIAN }, "
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" rate=(int)[1,MAX]," " channels=(int)[1,MAX]");
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gst_raw_parse_class_set_src_pad_template (rp_class, caps);
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gst_raw_parse_class_set_multiple_frames_per_buffer (rp_class, TRUE);
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gst_caps_unref (caps);
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}
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static void
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gst_audio_parse_class_init (GstAudioParseClass * klass)
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{
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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GstRawParseClass *rp_class = GST_RAW_PARSE_CLASS (klass);
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gobject_class->set_property = gst_audio_parse_set_property;
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gobject_class->get_property = gst_audio_parse_get_property;
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rp_class->get_caps = gst_audio_parse_get_caps;
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g_object_class_install_property (gobject_class, ARG_FORMAT,
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g_param_spec_enum ("format", "Format",
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"Format of audio samples in raw stream", GST_AUDIO_PARSE_FORMAT,
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GST_AUDIO_PARSE_FORMAT_INT, G_PARAM_READWRITE));
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g_object_class_install_property (gobject_class, ARG_RATE,
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g_param_spec_int ("rate", "Rate", "Rate of audio samples in raw stream",
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1, INT_MAX, 44100, G_PARAM_READWRITE));
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g_object_class_install_property (gobject_class, ARG_CHANNELS,
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g_param_spec_int ("channels", "Channels",
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"Number of channels in raw stream", 1, INT_MAX, 2,
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G_PARAM_READWRITE));
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g_object_class_install_property (gobject_class, ARG_WIDTH,
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g_param_spec_int ("width", "Width",
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"Width of audio samples in raw stream", 1, INT_MAX, 16,
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G_PARAM_READWRITE));
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g_object_class_install_property (gobject_class, ARG_DEPTH,
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g_param_spec_int ("depth", "Depth",
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"Depth of audio samples in raw stream", 1, INT_MAX, 16,
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G_PARAM_READWRITE));
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g_object_class_install_property (gobject_class, ARG_SIGNED,
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g_param_spec_boolean ("signed", "signed",
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"Sign of audio samples in raw stream", TRUE, G_PARAM_READWRITE));
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g_object_class_install_property (gobject_class, ARG_ENDIANNESS,
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g_param_spec_enum ("endianness", "Endianness",
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"Endianness of audio samples in raw stream",
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GST_AUDIO_PARSE_ENDIANNESS, G_BYTE_ORDER, G_PARAM_READWRITE));
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}
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static void
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gst_audio_parse_init (GstAudioParse * ap, GstAudioParseClass * g_class)
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{
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ap->format = GST_AUDIO_PARSE_FORMAT_INT;
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ap->channels = 2;
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ap->width = 16;
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ap->depth = 16;
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ap->signedness = TRUE;
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ap->endianness = G_BYTE_ORDER;
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gst_audio_parse_update_frame_size (ap);
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gst_raw_parse_set_fps (GST_RAW_PARSE (ap), 44100, 1);
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}
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static void
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gst_audio_parse_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstAudioParse *ap = GST_AUDIO_PARSE (object);
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g_return_if_fail (!gst_raw_parse_is_negotiated (GST_RAW_PARSE (ap)));
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switch (prop_id) {
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case ARG_FORMAT:
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ap->format = g_value_get_enum (value);
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break;
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case ARG_RATE:
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gst_raw_parse_set_fps (GST_RAW_PARSE (ap), g_value_get_int (value), 1);
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break;
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case ARG_CHANNELS:
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ap->channels = g_value_get_int (value);
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break;
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case ARG_WIDTH:
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ap->width = g_value_get_int (value);
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break;
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case ARG_DEPTH:
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ap->depth = g_value_get_int (value);
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break;
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case ARG_SIGNED:
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ap->signedness = g_value_get_boolean (value);
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break;
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case ARG_ENDIANNESS:
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ap->endianness = g_value_get_enum (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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gst_audio_parse_update_frame_size (ap);
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}
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static void
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gst_audio_parse_get_property (GObject * object, guint prop_id, GValue * value,
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GParamSpec * pspec)
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{
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GstAudioParse *ap = GST_AUDIO_PARSE (object);
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switch (prop_id) {
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case ARG_FORMAT:
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g_value_set_enum (value, ap->format);
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break;
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case ARG_RATE:{
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gint fps_n, fps_d;
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gst_raw_parse_get_fps (GST_RAW_PARSE (ap), &fps_n, &fps_d);
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g_value_set_int (value, fps_n);
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break;
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}
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case ARG_CHANNELS:
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g_value_set_int (value, ap->channels);
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break;
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case ARG_WIDTH:
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g_value_set_int (value, ap->width);
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break;
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case ARG_DEPTH:
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g_value_set_int (value, ap->depth);
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break;
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case ARG_SIGNED:
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g_value_set_boolean (value, ap->signedness);
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break;
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case ARG_ENDIANNESS:
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g_value_set_enum (value, ap->endianness);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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void
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gst_audio_parse_update_frame_size (GstAudioParse * ap)
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{
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gint framesize;
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framesize = (ap->width / 8) * ap->channels;
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gst_raw_parse_set_framesize (GST_RAW_PARSE (ap), framesize);
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}
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static GstCaps *
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gst_audio_parse_get_caps (GstRawParse * rp)
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{
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GstAudioParse *ap = GST_AUDIO_PARSE (rp);
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GstCaps *caps;
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gint fps_n, fps_d;
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gst_raw_parse_get_fps (rp, &fps_n, &fps_d);
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if (ap->format == GST_AUDIO_PARSE_FORMAT_INT) {
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caps = gst_caps_new_simple ("audio/x-raw-int",
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"rate", G_TYPE_INT, fps_n,
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"channels", G_TYPE_INT, ap->channels,
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"width", G_TYPE_INT, ap->width,
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"depth", G_TYPE_INT, ap->depth,
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"signed", G_TYPE_BOOLEAN, ap->signedness,
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"endianness", G_TYPE_INT, ap->endianness, NULL);
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} else {
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caps = gst_caps_new_simple ("audio/x-raw-float",
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"rate", G_TYPE_INT, fps_n,
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"channels", G_TYPE_INT, ap->channels,
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"width", G_TYPE_INT, ap->width,
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"endianness", G_TYPE_INT, ap->endianness, NULL);
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}
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return caps;
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}
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