[
AC_MSG_WARN(building experimental plug-ins)
USE_TARKIN="yes"
- USE_RTP="yes"
USE_SHOUT2="yes"
],[
AC_MSG_NOTICE(not building experimental plug-ins)
USE_TARKIN="no"
- USE_RTP="no"
USE_SHOUT2="no"
])
AC_SUBST(RAW1394_LIBS)
])
-dnl *** rtp ***
-dnl FIXME : adapt and make it work
-translit(dnm, m, l) AM_CONDITIONAL(USE_RTP, true)
-GST_CHECK_FEATURE(RTP, [RTP library], rtpenc rtpdec,[
- AC_CHECK_LIB(ortp, ortp_init, HAVE_RTP=yes, HAVE_RTP=no, $GST_CFLAGS $GST_LIBS)
- RTP_LIBS="-lortp"
- AC_SUBST(RTP_LIBS)
-])
-
-dnl FIXME header check needs to use GLIB_CFLAGS in order to succeed for rtp
-dnl can use GST_* which should have GLIB_* info
-dnl AC_CHECK_HEADERS(rtp/rtp.h, HAVE_LIBRTP=yes, HAVE_LIBRTP=no)
-dnl AC_CHECK_HEADERS(rtp/rtp-packet.h, :, HAVE_LIBRTP=no)
-dnl AC_CHECK_HEADERS(rtp/rtcp-packet.h, :, HAVE_LIBRTP=no)
-dnl AC_CHECK_HEADERS(rtp/rtp-audio.h, :, HAVE_LIBRTP=no)
-
dnl *** SDL ***
translit(dnm, m, l) AM_CONDITIONAL(USE_SDL, true)
GST_CHECK_FEATURE(SDL, [SDL plug-in], sdlvideosink, [
gst/playondemand/Makefile
gst/qtdemux/Makefile
gst/rtjpeg/Makefile
+gst/rtp/Makefile
gst/silence/Makefile
gst/sine/Makefile
gst/smooth/Makefile
ext/mpeg2dec/Makefile
ext/openquicktime/Makefile
ext/raw1394/Makefile
-ext/rtp/Makefile
ext/sdl/Makefile
ext/shout/Makefile
ext/shout2/Makefile
RAW1394_DIR=
endif
-if USE_RTP
-RTP_DIR=rtp
-else
-RTP_DIR=
-endif
-
if USE_SDL
SDL_DIR=sdl
else
$(LADSPA_DIR) $(LAME_DIR) $(LCS_DIR) \
$(LIBDV_DIR) $(LIBFAME_DIR) $(LIBPNG_DIR) \
$(MAD_DIR) $(MIKMOD_DIR) $(MJPEGTOOLS_DIR) $(MPEG2DEC_DIR) \
- $(OPENQUICKTIME_DIR) $(RAW1394_DIR) $(RTP_DIR) \
+ $(OPENQUICKTIME_DIR) $(RAW1394_DIR) \
$(SDL_DIR) $(SHOUT_DIR) $(SIDPLAY_DIR) \
$(SMOOTHWAVE_DIR) $(SWFDEC_DIR) $(TARKIN_DIR) \
$(VORBIS_DIR) $(XMMS_DIR) $(SNAPSHOT_DIR)
hermes http ivorbis jack jpeg \
ladspa lame lcs libfame libpng \
mad mikmod mjpegtools mpeg2dec \
- openquicktime raw1394 rtp \
+ openquicktime raw1394 \
sdl snapshot shout shout2 sidplay \
smoothwave swfdec tarkin vorbis xmms
--- /dev/null
+plugindir = $(libdir)/gstreamer-@GST_MAJORMINOR@
+
+plugin_LTLIBRARIES = libgstrtp.la
+
+libgstrtp_la_SOURCES = gstrtp.c gstrtpL16enc.c gstrtpL16parse.c rtp-packet.c
+
+libgstrtp_la_CFLAGS = $(GST_CFLAGS)
+libgstrtp_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+
+noinst_HEADERS = gstrtpL16enc.h gstrtpL16parse.h gstrtp-common.h rtp-packet.h
--- /dev/null
+
+TODO
+----
+
+- implement packing up to the MTU.
+- discont events in the case of packet loss
+- figure out the clocking.
+- implement various RFCs dealing with different payload types.
+ (as modules?)
+- Throw-out the the caps-nego & other session control things to the
+ Application Developer( App ), by turning rtcp work into, signals
+ in gstrtpsend & props/args in gstrtprecv.
+ The App would then be free to use any sort of session control
+ protocal like RTSP.( done )
+
+
+Relevant RFCs
+-------------
+
+1889 RTP: A Transport Protocol for Real-Time Applications.
+
+2198 RTP Payload for Redundant Audio Data.
+3119 A More Loss-Tolerant RTP Payload Format for MP3 Audio.
+
+2793 RTP Payload for Text Conversation.
+
+2032 RTP Payload Format for H.261 Video Streams.
+2190 RTP Payload Format for H.263 Video Streams.
+2250 RTP Payload Format for MPEG1/MPEG2 Video.
+2343 RTP Payload Format for Bundled MPEG.
+2429 RTP Payload Format for the 1998 Version of ITU-T Rec. H.263 Video
+2431 RTP Payload Format for BT.656 Video Encoding.
+2435 RTP Payload Format for JPEG-compressed Video.
+3016 RTP Payload Format for MPEG-4 Audio/Visual Streams.
+3047 RTP Payload Format for ITU-T Recommendation G.722.1.
+
+2733 An RTP Payload Format for Generic Forward Error Correction.
+2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony
+ Signals.
+2862 RTP Payload Format for Real-Time Pointers.
+1890 RTP Profile for Audio and Video Conferences with Minimal Control.
+2508 Compressing IP/UDP/RTP Headers for Low-Speed Serial Links.
+
+
+do we care?
+-----------
+
+2029 RTP Payload Format of Sun's CellB Video Encoding.
+
+usefull
+-------
+
+http://www.iana.org/assignments/rtp-parameters
--- /dev/null
+*GstRtpRecv:
+ *gstrtprecv.c
+
+*For Sequencing:
+ * timestamp
+ * algorithm
+
+*For Video:
+ * payload_t
--- /dev/null
+/* Gnome-Streamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ *
+ * 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.
+ */
+
+
+#ifndef __GST_RTP_COMMON_H__
+#define __GST_RTP_COMMON_H__
+
+#define RTP_VERSION 2
+
+typedef enum
+{
+/* Audio: */
+ PAYLOAD_GSM = 3,
+ PAYLOAD_L16_STEREO = 10,
+ PAYLOAD_L16_MONO = 11,
+ PAYLOAD_MPA = 14, /* Audio MPEG 1-3 */
+ PAYLOAD_G723_63 = 16, /* Not standard */
+ PAYLOAD_G723_53 = 17, /* Not standard */
+ PAYLOAD_TS48 = 18, /* Not standard */
+ PAYLOAD_TS41 = 19, /* Not standard */
+ PAYLOAD_G728 = 20, /* Not standard */
+ PAYLOAD_G729 = 21, /* Not standard */
+
+/* Video: */
+ PAYLOAD_MPV = 32, /* Video MPEG 1 & 2 */
+
+/* BOTH */
+ PAYLOAD_BMPEG = 34 /* Not Standard */
+}
+rtp_payload_t;
+
+#endif
--- /dev/null
+/* Gnome-Streamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ *
+ * 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.
+ */
+
+#include "gstrtpL16enc.h"
+#include "gstrtpL16parse.h"
+
+static gboolean
+plugin_init (GModule *module, GstPlugin *plugin)
+{
+ gst_rtpL16enc_plugin_init (module, plugin);
+ gst_rtpL16parse_plugin_init (module, plugin);
+
+ return TRUE;
+}
+
+GstPluginDesc plugin_desc = {
+ GST_VERSION_MAJOR,
+ GST_VERSION_MINOR,
+ "rtp",
+ plugin_init
+};
--- /dev/null
+/* GStreamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ *
+ * 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
+ */
+
+#include <string.h>
+#include "gstrtpL16parse.h"
+#include "gstrtp-common.h"
+
+/* elementfactory information */
+static GstElementDetails gst_rtp_L16parse_details = {
+ "RTP packet parser",
+ "RtpL16Parse",
+ "GPL",
+ "Extracts raw audio from RTP packets",
+ VERSION,
+ "Zeeshan Ali <zak147@yahoo.com>",
+ "(C) 2003",
+};
+
+/* RtpL16Parse signals and args */
+enum
+{
+ /* FILL ME */
+ LAST_SIGNAL
+};
+
+enum
+{
+ ARG_0,
+ ARG_FREQUENCY,
+ ARG_PAYLOAD_TYPE,
+};
+
+GST_PAD_TEMPLATE_FACTORY (src_factory,
+ "src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_CAPS_NEW (
+ "audio_raw",
+ "audio/raw",
+ "format", GST_PROPS_STRING ("int"),
+ "law", GST_PROPS_INT (0),
+ "endianness", GST_PROPS_INT (G_BYTE_ORDER),
+ "signed", GST_PROPS_BOOLEAN (TRUE),
+ "width", GST_PROPS_INT (16),
+ "depth", GST_PROPS_INT (16),
+ "rate", GST_PROPS_INT_RANGE (1000, 48000),
+ "channels", GST_PROPS_INT_RANGE (1, 2))
+)
+
+GST_PAD_TEMPLATE_FACTORY (sink_factory,
+ "sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_CAPS_NEW (
+ "rtp",
+ "application/x-rtp",
+ NULL)
+);
+
+static void gst_rtpL16parse_class_init (GstRtpL16ParseClass * klass);
+static void gst_rtpL16parse_init (GstRtpL16Parse * rtpL16parse);
+
+static void gst_rtpL16parse_chain (GstPad * pad, GstBuffer * buf);
+
+static void gst_rtpL16parse_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec);
+static void gst_rtpL16parse_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec);
+static GstElementStateReturn gst_rtpL16parse_change_state (GstElement * element);
+
+static GstElementClass *parent_class = NULL;
+
+static GType gst_rtpL16parse_get_type (void)
+{
+ static GType rtpL16parse_type = 0;
+
+ if (!rtpL16parse_type) {
+ static const GTypeInfo rtpL16parse_info = {
+ sizeof (GstRtpL16ParseClass),
+ NULL,
+ NULL,
+ (GClassInitFunc) gst_rtpL16parse_class_init,
+ NULL,
+ NULL,
+ sizeof (GstRtpL16Parse),
+ 0,
+ (GInstanceInitFunc) gst_rtpL16parse_init,
+ };
+
+ rtpL16parse_type = g_type_register_static (GST_TYPE_ELEMENT, "GstRtpL16Parse", &rtpL16parse_info, 0);
+ }
+ return rtpL16parse_type;
+}
+
+static void
+gst_rtpL16parse_class_init (GstRtpL16ParseClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
+
+ parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
+
+ g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_PAYLOAD_TYPE,
+ g_param_spec_int ("payload_type", "payload_type", "payload type",
+ G_MININT, G_MAXINT, PAYLOAD_L16_STEREO, G_PARAM_READABLE));
+ g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_FREQUENCY,
+ g_param_spec_int ("frequency", "frequency", "frequency",
+ G_MININT, G_MAXINT, 44100, G_PARAM_READWRITE));
+
+ gobject_class->set_property = gst_rtpL16parse_set_property;
+ gobject_class->get_property = gst_rtpL16parse_get_property;
+
+ gstelement_class->change_state = gst_rtpL16parse_change_state;
+}
+
+static void
+gst_rtpL16parse_init (GstRtpL16Parse * rtpL16parse)
+{
+ rtpL16parse->srcpad = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (src_factory), "src");
+ rtpL16parse->sinkpad = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (sink_factory), "sink");
+ gst_element_add_pad (GST_ELEMENT (rtpL16parse), rtpL16parse->srcpad);
+ gst_element_add_pad (GST_ELEMENT (rtpL16parse), rtpL16parse->sinkpad);
+ gst_pad_set_chain_function (rtpL16parse->sinkpad, gst_rtpL16parse_chain);
+
+ rtpL16parse->frequency = 44100;
+ rtpL16parse->channels = 2;
+
+ rtpL16parse->payload_type = PAYLOAD_L16_STEREO;
+}
+
+void
+gst_rtpL16parse_ntohs (GstBuffer *buf)
+{
+ guint16 *i, *len;
+
+ /* FIXME: is this code correct or even sane at all? */
+ i = (guint16 *) GST_BUFFER_DATA(buf);
+ len = i + GST_BUFFER_SIZE (buf) / sizeof (guint16 *);
+
+ for (; i<len; i++) {
+ *i = g_ntohs (*i);
+ }
+}
+
+void
+gst_rtpL16_caps_nego (GstRtpL16Parse *rtpL16parse)
+{
+ GstCaps *caps;
+
+ caps = GST_CAPS_NEW (
+ "audio_raw",
+ "audio/raw",
+ "format", GST_PROPS_STRING ("int"),
+ "law", GST_PROPS_INT (0),
+ "endianness", GST_PROPS_INT (G_BYTE_ORDER),
+ "signed", GST_PROPS_BOOLEAN (TRUE),
+ "width", GST_PROPS_INT (16),
+ "depth", GST_PROPS_INT (16),
+ "rate", GST_PROPS_INT (rtpL16parse->frequency),
+ "channels", GST_PROPS_INT (rtpL16parse->channels));
+
+ gst_pad_try_set_caps (rtpL16parse->srcpad, caps);
+}
+
+void
+gst_rtpL16parse_payloadtype_change (GstRtpL16Parse *rtpL16parse, rtp_payload_t pt)
+{
+ rtpL16parse->payload_type = pt;
+
+ switch (pt) {
+ case PAYLOAD_L16_MONO:
+ rtpL16parse->channels = 1;
+ break;
+ case PAYLOAD_L16_STEREO:
+ rtpL16parse->channels = 2;
+ break;
+ default:
+ g_warning ("unkown payload_t %d\n", pt);
+ }
+
+ gst_rtpL16_caps_nego (rtpL16parse);
+}
+
+static void
+gst_rtpL16parse_chain (GstPad * pad, GstBuffer * buf)
+{
+ GstRtpL16Parse *rtpL16parse;
+ GstBuffer *outbuf;
+ Rtp_Packet packet;
+ rtp_payload_t pt;
+
+ g_return_if_fail (pad != NULL);
+ g_return_if_fail (GST_IS_PAD (pad));
+ g_return_if_fail (buf != NULL);
+
+ rtpL16parse = GST_RTP_L16_PARSE (GST_OBJECT_PARENT (pad));
+
+ g_return_if_fail (rtpL16parse != NULL);
+ g_return_if_fail (GST_IS_RTP_L16_PARSE (rtpL16parse));
+
+ if (GST_IS_EVENT (buf)) {
+ GstEvent *event = GST_EVENT (buf);
+ gst_pad_event_default (pad, event);
+
+ return;
+ }
+
+ if (GST_PAD_CAPS (rtpL16parse->srcpad) == NULL) {
+ gst_rtpL16_caps_nego (rtpL16parse);
+ }
+
+ packet = rtp_packet_new_copy_data (GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
+
+ pt = rtp_packet_get_payload_type (packet);
+
+ if (pt != rtpL16parse->payload_type) {
+ gst_rtpL16parse_payloadtype_change (rtpL16parse, pt);
+ }
+
+ outbuf = gst_buffer_new ();
+ GST_BUFFER_SIZE (outbuf) = rtp_packet_get_payload_len (packet);
+ GST_BUFFER_DATA (outbuf) = g_malloc (GST_BUFFER_SIZE (outbuf));
+ GST_BUFFER_TIMESTAMP (outbuf) = g_ntohl (rtp_packet_get_timestamp (packet)) * GST_SECOND;
+
+ memcpy (GST_BUFFER_DATA (outbuf), rtp_packet_get_payload (packet), GST_BUFFER_SIZE (outbuf));
+
+ GST_DEBUG (0,"gst_rtpL16parse_chain: pushing buffer of size %d", GST_BUFFER_SIZE(outbuf));
+
+ /* FIXME: According to RFC 1890, this is required, right? */
+#if G_BYTE_ORDER == G_LITTLE_ENDIAN
+ gst_rtpL16parse_ntohs (outbuf);
+#endif
+
+ gst_pad_push (rtpL16parse->srcpad, outbuf);
+
+ rtp_packet_free (packet);
+ gst_buffer_unref (buf);
+}
+
+static void
+gst_rtpL16parse_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec)
+{
+ GstRtpL16Parse *rtpL16parse;
+
+ /* it's not null if we got it, but it might not be ours */
+ g_return_if_fail (GST_IS_RTP_L16_PARSE (object));
+ rtpL16parse = GST_RTP_L16_PARSE (object);
+
+ switch (prop_id) {
+ case ARG_PAYLOAD_TYPE:
+ gst_rtpL16parse_payloadtype_change (rtpL16parse, g_value_get_int (value));
+ break;
+ case ARG_FREQUENCY:
+ rtpL16parse->frequency = g_value_get_int (value);
+ break;
+ default:
+ break;
+ }
+}
+
+static void
+gst_rtpL16parse_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec)
+{
+ GstRtpL16Parse *rtpL16parse;
+
+ /* it's not null if we got it, but it might not be ours */
+ g_return_if_fail (GST_IS_RTP_L16_PARSE (object));
+ rtpL16parse = GST_RTP_L16_PARSE (object);
+
+ switch (prop_id) {
+ case ARG_PAYLOAD_TYPE:
+ g_value_set_int (value, rtpL16parse->payload_type);
+ break;
+ case ARG_FREQUENCY:
+ g_value_set_int (value, rtpL16parse->frequency);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static GstElementStateReturn
+gst_rtpL16parse_change_state (GstElement * element)
+{
+ GstRtpL16Parse *rtpL16parse;
+
+ g_return_val_if_fail (GST_IS_RTP_L16_PARSE (element), GST_STATE_FAILURE);
+
+ rtpL16parse = GST_RTP_L16_PARSE (element);
+
+ GST_DEBUG (0, "state pending %d\n", GST_STATE_PENDING (element));
+
+ switch (GST_STATE_TRANSITION (element)) {
+ case GST_STATE_NULL_TO_READY:
+ break;
+ case GST_STATE_READY_TO_NULL:
+ break;
+ default:
+ break;
+ }
+
+ /* if we haven't failed already, give the parent class a chance to ;-) */
+ if (GST_ELEMENT_CLASS (parent_class)->change_state)
+ return GST_ELEMENT_CLASS (parent_class)->change_state (element);
+
+ return GST_STATE_SUCCESS;
+}
+
+gboolean
+gst_rtpL16parse_plugin_init (GModule * module, GstPlugin * plugin)
+{
+ GstElementFactory *rtpL16parse;
+
+ rtpL16parse = gst_element_factory_new ("rtpL16parse", GST_TYPE_RTP_L16_PARSE, &gst_rtp_L16parse_details);
+ g_return_val_if_fail (rtpL16parse != NULL, FALSE);
+
+ gst_element_factory_add_pad_template (rtpL16parse, GST_PAD_TEMPLATE_GET (src_factory));
+ gst_element_factory_add_pad_template (rtpL16parse, GST_PAD_TEMPLATE_GET (sink_factory));
+
+ gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (rtpL16parse));
+
+ return TRUE;
+}
--- /dev/null
+/* Gnome-Streamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ *
+ * 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.
+ */
+
+#ifndef __GST_RTP_L16_PARSE_H__
+#define __GST_RTP_L16_PARSE_H__
+
+#include <gst/gst.h>
+#include "rtp-packet.h"
+#include "gstrtp-common.h"
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif /* __cplusplus */
+
+/* Definition of structure storing data for this element. */
+typedef struct _GstRtpL16Parse GstRtpL16Parse;
+struct _GstRtpL16Parse
+{
+ GstElement element;
+
+ GstPad *sinkpad;
+ GstPad *srcpad;
+
+ guint frequency;
+ guint channels;
+
+ rtp_payload_t payload_type;
+};
+
+/* Standard definition defining a class for this element. */
+typedef struct _GstRtpL16ParseClass GstRtpL16ParseClass;
+struct _GstRtpL16ParseClass
+{
+ GstElementClass parent_class;
+};
+
+/* Standard macros for defining types for this element. */
+#define GST_TYPE_RTP_L16_PARSE \
+ (gst_rtpL16parse_get_type())
+#define GST_RTP_L16_PARSE(obj) \
+ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_L16_PARSE,GstRtpL16Parse))
+#define GST_RTP_L16_PARSE_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_L16_PARSE,GstRtpL16Parse))
+#define GST_IS_RTP_L16_PARSE(obj) \
+ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_L16_PARSE))
+#define GST_IS_RTP_L16_PARSE_CLASS(obj) \
+ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_L16_PARSE))
+
+gboolean gst_rtpL16parse_plugin_init (GModule * module, GstPlugin * plugin);
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+
+
+#endif /* __GST_RTP_L16_PARSE_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ *
+ * 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.
+ */
+
+#include <string.h>
+#include "gstrtpL16enc.h"
+
+/* elementfactory information */
+static GstElementDetails gst_rtpL16enc_details = {
+ "RTP RAW Audio Encoder",
+ "RtpL16Enc",
+ "LGPL",
+ "Encodes Raw Audio into an RTP packet",
+ VERSION,
+ "Zeeshan Ali <zak147@yahoo.com>",
+ "(C) 2003",
+};
+
+/* RtpL16Enc signals and args */
+enum
+{
+ /* FILL ME */
+ LAST_SIGNAL
+};
+
+enum
+{
+ /* FILL ME */
+ ARG_0,
+};
+
+GST_PAD_TEMPLATE_FACTORY (sink_factory,
+ "sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_CAPS_NEW (
+ "audio_raw",
+ "audio/raw",
+ "format", GST_PROPS_STRING ("int"),
+ "law", GST_PROPS_INT (0),
+ "endianness", GST_PROPS_INT (G_BYTE_ORDER),
+ "signed", GST_PROPS_BOOLEAN (TRUE),
+ "width", GST_PROPS_INT (16),
+ "depth", GST_PROPS_INT (16),
+ "rate", GST_PROPS_INT_RANGE (1000, 48000),
+ "channels", GST_PROPS_INT_RANGE (1, 2)
+ )
+);
+
+GST_PAD_TEMPLATE_FACTORY (src_factory,
+ "src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_CAPS_NEW (
+ "rtp",
+ "application/x-rtp",
+ NULL)
+);
+
+static void gst_rtpL16enc_class_init (GstRtpL16EncClass * klass);
+static void gst_rtpL16enc_init (GstRtpL16Enc * rtpL16enc);
+static void gst_rtpL16enc_chain (GstPad * pad, GstBuffer * buf);
+static void gst_rtpL16enc_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec);
+static void gst_rtpL16enc_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec);
+static GstPadLinkReturn gst_rtpL16enc_sinkconnect (GstPad * pad, GstCaps * caps);
+static GstElementStateReturn gst_rtpL16enc_change_state (GstElement * element);
+
+static GstElementClass *parent_class = NULL;
+
+static GType gst_rtpL16enc_get_type (void)
+{
+ static GType rtpL16enc_type = 0;
+
+ if (!rtpL16enc_type) {
+ static const GTypeInfo rtpL16enc_info = {
+ sizeof (GstRtpL16EncClass),
+ NULL,
+ NULL,
+ (GClassInitFunc) gst_rtpL16enc_class_init,
+ NULL,
+ NULL,
+ sizeof (GstRtpL16Enc),
+ 0,
+ (GInstanceInitFunc) gst_rtpL16enc_init,
+ };
+
+ rtpL16enc_type = g_type_register_static (GST_TYPE_ELEMENT, "GstRtpL16Enc", &rtpL16enc_info, 0);
+ }
+ return rtpL16enc_type;
+}
+
+static void
+gst_rtpL16enc_class_init (GstRtpL16EncClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
+
+ parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
+
+ gobject_class->set_property = gst_rtpL16enc_set_property;
+ gobject_class->get_property = gst_rtpL16enc_get_property;
+
+ gstelement_class->change_state = gst_rtpL16enc_change_state;
+}
+
+static void
+gst_rtpL16enc_init (GstRtpL16Enc * rtpL16enc)
+{
+ rtpL16enc->sinkpad = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (sink_factory), "sink");
+ rtpL16enc->srcpad = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (src_factory), "src");
+ gst_element_add_pad (GST_ELEMENT (rtpL16enc), rtpL16enc->sinkpad);
+ gst_element_add_pad (GST_ELEMENT (rtpL16enc), rtpL16enc->srcpad);
+ gst_pad_set_chain_function (rtpL16enc->sinkpad, gst_rtpL16enc_chain);
+ gst_pad_set_link_function (rtpL16enc->sinkpad, gst_rtpL16enc_sinkconnect);
+
+ rtpL16enc->frequency = 44100;
+ rtpL16enc->channels = 2;
+
+ rtpL16enc->next_time = 0;
+ rtpL16enc->time_interval = 0;
+
+ rtpL16enc->seq = 0;
+ rtpL16enc->ssrc = random ();
+}
+
+static GstPadLinkReturn
+gst_rtpL16enc_sinkconnect (GstPad * pad, GstCaps * caps)
+{
+ GstRtpL16Enc *rtpL16enc;
+
+ rtpL16enc = GST_RTP_L16_ENC (gst_pad_get_parent (pad));
+
+ gst_caps_get_int (caps, "rate", &rtpL16enc->frequency);
+ gst_caps_get_int (caps, "channels", &rtpL16enc->channels);
+
+ /* Pre-calculate what we can */
+ rtpL16enc->time_interval = GST_SECOND / (2 * rtpL16enc->channels * rtpL16enc->frequency);
+
+ return GST_PAD_LINK_OK;
+}
+
+
+void
+gst_rtpL16enc_htons (GstBuffer *buf)
+{
+ guint16 *i, *len;
+
+ /* FIXME: is this code correct or even sane at all? */
+ i = (guint16 *) GST_BUFFER_DATA(buf);
+ len = i + GST_BUFFER_SIZE (buf) / sizeof (guint16 *);
+
+ for (; i<len; i++) {
+ *i = g_htons (*i);
+ }
+}
+
+static void
+gst_rtpL16enc_chain (GstPad * pad, GstBuffer * buf)
+{
+ GstRtpL16Enc *rtpL16enc;
+ GstBuffer *outbuf;
+ Rtp_Packet packet;
+
+ g_return_if_fail (pad != NULL);
+ g_return_if_fail (GST_IS_PAD (pad));
+ g_return_if_fail (buf != NULL);
+
+ rtpL16enc = GST_RTP_L16_ENC (GST_OBJECT_PARENT (pad));
+
+ g_return_if_fail (rtpL16enc != NULL);
+ g_return_if_fail (GST_IS_RTP_L16_ENC (rtpL16enc));
+
+ if (GST_IS_EVENT (buf)) {
+ GstEvent *event = GST_EVENT (buf);
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_DISCONTINUOUS:
+ GST_DEBUG (GST_CAT_EVENT, "discont");
+ rtpL16enc->next_time = 0;
+ gst_pad_event_default (pad, event);
+ return;
+ default:
+ gst_pad_event_default (pad, event);
+ return;
+ }
+ }
+
+ /* We only need the header */
+ packet = rtp_packet_new_allocate (0, 0, 0);
+
+ rtp_packet_set_csrc_count (packet, 0);
+ rtp_packet_set_extension (packet, 0);
+ rtp_packet_set_padding (packet, 0);
+ rtp_packet_set_version (packet, RTP_VERSION);
+ rtp_packet_set_marker (packet, 0);
+ rtp_packet_set_ssrc (packet, g_htonl (rtpL16enc->ssrc));
+ rtp_packet_set_seq (packet, g_htons (rtpL16enc->seq));
+ rtp_packet_set_timestamp (packet, g_htonl ((guint32) rtpL16enc->next_time / GST_SECOND));
+
+ if (rtpL16enc->channels == 1) {
+ rtp_packet_set_payload_type (packet, (guint8) PAYLOAD_L16_MONO);
+ }
+
+ else {
+ rtp_packet_set_payload_type (packet, (guint8) PAYLOAD_L16_STEREO);
+ }
+
+ /* FIXME: According to RFC 1890, this is required, right? */
+#if G_BYTE_ORDER == G_LITTLE_ENDIAN
+ gst_rtpL16enc_htons (buf);
+#endif
+
+ outbuf = gst_buffer_new ();
+ GST_BUFFER_SIZE (outbuf) = rtp_packet_get_packet_len (packet) + GST_BUFFER_SIZE (buf);
+ GST_BUFFER_DATA (outbuf) = g_malloc (GST_BUFFER_SIZE (outbuf));
+ GST_BUFFER_TIMESTAMP (outbuf) = rtpL16enc->next_time;
+
+ memcpy (GST_BUFFER_DATA (outbuf), packet->data, rtp_packet_get_packet_len (packet));
+ memcpy (GST_BUFFER_DATA (outbuf) + rtp_packet_get_packet_len(packet), GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
+
+ GST_DEBUG (0,"gst_rtpL16enc_chain: pushing buffer of size %d", GST_BUFFER_SIZE(outbuf));
+ gst_pad_push (rtpL16enc->srcpad, outbuf);
+
+ ++rtpL16enc->seq;
+ rtpL16enc->next_time += rtpL16enc->time_interval * GST_BUFFER_SIZE (buf);
+
+ rtp_packet_free (packet);
+ gst_buffer_unref (buf);
+}
+
+static void
+gst_rtpL16enc_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec)
+{
+ GstRtpL16Enc *rtpL16enc;
+
+ /* it's not null if we got it, but it might not be ours */
+ g_return_if_fail (GST_IS_RTP_L16_ENC (object));
+ rtpL16enc = GST_RTP_L16_ENC (object);
+
+ switch (prop_id) {
+ default:
+ break;
+ }
+}
+
+static void
+gst_rtpL16enc_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec)
+{
+ GstRtpL16Enc *rtpL16enc;
+
+ /* it's not null if we got it, but it might not be ours */
+ g_return_if_fail (GST_IS_RTP_L16_ENC (object));
+ rtpL16enc = GST_RTP_L16_ENC (object);
+
+ switch (prop_id) {
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static GstElementStateReturn
+gst_rtpL16enc_change_state (GstElement * element)
+{
+ GstRtpL16Enc *rtpL16enc;
+
+ g_return_val_if_fail (GST_IS_RTP_L16_ENC (element), GST_STATE_FAILURE);
+
+ rtpL16enc = GST_RTP_L16_ENC (element);
+
+ GST_DEBUG (0, "state pending %d\n", GST_STATE_PENDING (element));
+
+ /* if going down into NULL state, close the file if it's open */
+ switch (GST_STATE_TRANSITION (element)) {
+ case GST_STATE_NULL_TO_READY:
+ break;
+
+ case GST_STATE_READY_TO_NULL:
+ break;
+
+ default:
+ break;
+ }
+
+ /* if we haven't failed already, give the parent class a chance to ;-) */
+ if (GST_ELEMENT_CLASS (parent_class)->change_state)
+ return GST_ELEMENT_CLASS (parent_class)->change_state (element);
+
+ return GST_STATE_SUCCESS;
+}
+
+gboolean
+gst_rtpL16enc_plugin_init (GModule * module, GstPlugin * plugin)
+{
+ GstElementFactory *rtpL16enc;
+
+ rtpL16enc = gst_element_factory_new ("rtpL16enc", GST_TYPE_RTP_L16_ENC, &gst_rtpL16enc_details);
+ g_return_val_if_fail (rtpL16enc != NULL, FALSE);
+
+ gst_element_factory_add_pad_template (rtpL16enc, GST_PAD_TEMPLATE_GET (sink_factory));
+ gst_element_factory_add_pad_template (rtpL16enc, GST_PAD_TEMPLATE_GET (src_factory));
+
+ gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (rtpL16enc));
+
+ return TRUE;
+}
--- /dev/null
+/* Gnome-Streamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ *
+ * 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.
+ */
+
+
+#ifndef __GST_RTP_L16_ENC_H__
+#define __GST_RTP_L16_ENC_H__
+
+#include <gst/gst.h>
+#include "rtp-packet.h"
+#include "gstrtp-common.h"
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif /* __cplusplus */
+
+/* Definition of structure storing data for this element. */
+typedef struct _GstRtpL16Enc GstRtpL16Enc;
+struct _GstRtpL16Enc
+{
+ GstElement element;
+
+ GstPad *sinkpad;
+ GstPad *srcpad;
+
+ guint frequency;
+ guint channels;
+
+ /* the timestamp of the next frame */
+ guint64 next_time;
+ /* the interval between frames */
+ guint64 time_interval;
+
+ guint32 ssrc;
+ guint16 seq;
+};
+
+/* Standard definition defining a class for this element. */
+typedef struct _GstRtpL16EncClass GstRtpL16EncClass;
+struct _GstRtpL16EncClass
+{
+ GstElementClass parent_class;
+};
+
+/* Standard macros for defining types for this element. */
+#define GST_TYPE_RTP_L16_ENC \
+ (gst_rtpL16enc_get_type())
+#define GST_RTP_L16_ENC(obj) \
+ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_L16_ENC,GstRtpL16Enc))
+#define GST_RTP_L16_ENC_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_L16_ENC,GstRtpL16Enc))
+#define GST_IS_RTP_L16_ENC(obj) \
+ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_L16_ENC))
+#define GST_IS_RTP_L16_ENC_CLASS(obj) \
+ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_L16_ENC))
+
+gboolean gst_rtpL16enc_plugin_init (GModule * module, GstPlugin * plugin);
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+
+
+#endif /* __GST_RTP_L16_ENC_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ *
+ * 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
+ */
+
+#include <string.h>
+#include "gstrtpL16parse.h"
+#include "gstrtp-common.h"
+
+/* elementfactory information */
+static GstElementDetails gst_rtp_L16parse_details = {
+ "RTP packet parser",
+ "RtpL16Parse",
+ "GPL",
+ "Extracts raw audio from RTP packets",
+ VERSION,
+ "Zeeshan Ali <zak147@yahoo.com>",
+ "(C) 2003",
+};
+
+/* RtpL16Parse signals and args */
+enum
+{
+ /* FILL ME */
+ LAST_SIGNAL
+};
+
+enum
+{
+ ARG_0,
+ ARG_FREQUENCY,
+ ARG_PAYLOAD_TYPE,
+};
+
+GST_PAD_TEMPLATE_FACTORY (src_factory,
+ "src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_CAPS_NEW (
+ "audio_raw",
+ "audio/raw",
+ "format", GST_PROPS_STRING ("int"),
+ "law", GST_PROPS_INT (0),
+ "endianness", GST_PROPS_INT (G_BYTE_ORDER),
+ "signed", GST_PROPS_BOOLEAN (TRUE),
+ "width", GST_PROPS_INT (16),
+ "depth", GST_PROPS_INT (16),
+ "rate", GST_PROPS_INT_RANGE (1000, 48000),
+ "channels", GST_PROPS_INT_RANGE (1, 2))
+)
+
+GST_PAD_TEMPLATE_FACTORY (sink_factory,
+ "sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_CAPS_NEW (
+ "rtp",
+ "application/x-rtp",
+ NULL)
+);
+
+static void gst_rtpL16parse_class_init (GstRtpL16ParseClass * klass);
+static void gst_rtpL16parse_init (GstRtpL16Parse * rtpL16parse);
+
+static void gst_rtpL16parse_chain (GstPad * pad, GstBuffer * buf);
+
+static void gst_rtpL16parse_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec);
+static void gst_rtpL16parse_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec);
+static GstElementStateReturn gst_rtpL16parse_change_state (GstElement * element);
+
+static GstElementClass *parent_class = NULL;
+
+static GType gst_rtpL16parse_get_type (void)
+{
+ static GType rtpL16parse_type = 0;
+
+ if (!rtpL16parse_type) {
+ static const GTypeInfo rtpL16parse_info = {
+ sizeof (GstRtpL16ParseClass),
+ NULL,
+ NULL,
+ (GClassInitFunc) gst_rtpL16parse_class_init,
+ NULL,
+ NULL,
+ sizeof (GstRtpL16Parse),
+ 0,
+ (GInstanceInitFunc) gst_rtpL16parse_init,
+ };
+
+ rtpL16parse_type = g_type_register_static (GST_TYPE_ELEMENT, "GstRtpL16Parse", &rtpL16parse_info, 0);
+ }
+ return rtpL16parse_type;
+}
+
+static void
+gst_rtpL16parse_class_init (GstRtpL16ParseClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
+
+ parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
+
+ g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_PAYLOAD_TYPE,
+ g_param_spec_int ("payload_type", "payload_type", "payload type",
+ G_MININT, G_MAXINT, PAYLOAD_L16_STEREO, G_PARAM_READABLE));
+ g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_FREQUENCY,
+ g_param_spec_int ("frequency", "frequency", "frequency",
+ G_MININT, G_MAXINT, 44100, G_PARAM_READWRITE));
+
+ gobject_class->set_property = gst_rtpL16parse_set_property;
+ gobject_class->get_property = gst_rtpL16parse_get_property;
+
+ gstelement_class->change_state = gst_rtpL16parse_change_state;
+}
+
+static void
+gst_rtpL16parse_init (GstRtpL16Parse * rtpL16parse)
+{
+ rtpL16parse->srcpad = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (src_factory), "src");
+ rtpL16parse->sinkpad = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (sink_factory), "sink");
+ gst_element_add_pad (GST_ELEMENT (rtpL16parse), rtpL16parse->srcpad);
+ gst_element_add_pad (GST_ELEMENT (rtpL16parse), rtpL16parse->sinkpad);
+ gst_pad_set_chain_function (rtpL16parse->sinkpad, gst_rtpL16parse_chain);
+
+ rtpL16parse->frequency = 44100;
+ rtpL16parse->channels = 2;
+
+ rtpL16parse->payload_type = PAYLOAD_L16_STEREO;
+}
+
+void
+gst_rtpL16parse_ntohs (GstBuffer *buf)
+{
+ guint16 *i, *len;
+
+ /* FIXME: is this code correct or even sane at all? */
+ i = (guint16 *) GST_BUFFER_DATA(buf);
+ len = i + GST_BUFFER_SIZE (buf) / sizeof (guint16 *);
+
+ for (; i<len; i++) {
+ *i = g_ntohs (*i);
+ }
+}
+
+void
+gst_rtpL16_caps_nego (GstRtpL16Parse *rtpL16parse)
+{
+ GstCaps *caps;
+
+ caps = GST_CAPS_NEW (
+ "audio_raw",
+ "audio/raw",
+ "format", GST_PROPS_STRING ("int"),
+ "law", GST_PROPS_INT (0),
+ "endianness", GST_PROPS_INT (G_BYTE_ORDER),
+ "signed", GST_PROPS_BOOLEAN (TRUE),
+ "width", GST_PROPS_INT (16),
+ "depth", GST_PROPS_INT (16),
+ "rate", GST_PROPS_INT (rtpL16parse->frequency),
+ "channels", GST_PROPS_INT (rtpL16parse->channels));
+
+ gst_pad_try_set_caps (rtpL16parse->srcpad, caps);
+}
+
+void
+gst_rtpL16parse_payloadtype_change (GstRtpL16Parse *rtpL16parse, rtp_payload_t pt)
+{
+ rtpL16parse->payload_type = pt;
+
+ switch (pt) {
+ case PAYLOAD_L16_MONO:
+ rtpL16parse->channels = 1;
+ break;
+ case PAYLOAD_L16_STEREO:
+ rtpL16parse->channels = 2;
+ break;
+ default:
+ g_warning ("unkown payload_t %d\n", pt);
+ }
+
+ gst_rtpL16_caps_nego (rtpL16parse);
+}
+
+static void
+gst_rtpL16parse_chain (GstPad * pad, GstBuffer * buf)
+{
+ GstRtpL16Parse *rtpL16parse;
+ GstBuffer *outbuf;
+ Rtp_Packet packet;
+ rtp_payload_t pt;
+
+ g_return_if_fail (pad != NULL);
+ g_return_if_fail (GST_IS_PAD (pad));
+ g_return_if_fail (buf != NULL);
+
+ rtpL16parse = GST_RTP_L16_PARSE (GST_OBJECT_PARENT (pad));
+
+ g_return_if_fail (rtpL16parse != NULL);
+ g_return_if_fail (GST_IS_RTP_L16_PARSE (rtpL16parse));
+
+ if (GST_IS_EVENT (buf)) {
+ GstEvent *event = GST_EVENT (buf);
+ gst_pad_event_default (pad, event);
+
+ return;
+ }
+
+ if (GST_PAD_CAPS (rtpL16parse->srcpad) == NULL) {
+ gst_rtpL16_caps_nego (rtpL16parse);
+ }
+
+ packet = rtp_packet_new_copy_data (GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
+
+ pt = rtp_packet_get_payload_type (packet);
+
+ if (pt != rtpL16parse->payload_type) {
+ gst_rtpL16parse_payloadtype_change (rtpL16parse, pt);
+ }
+
+ outbuf = gst_buffer_new ();
+ GST_BUFFER_SIZE (outbuf) = rtp_packet_get_payload_len (packet);
+ GST_BUFFER_DATA (outbuf) = g_malloc (GST_BUFFER_SIZE (outbuf));
+ GST_BUFFER_TIMESTAMP (outbuf) = g_ntohl (rtp_packet_get_timestamp (packet)) * GST_SECOND;
+
+ memcpy (GST_BUFFER_DATA (outbuf), rtp_packet_get_payload (packet), GST_BUFFER_SIZE (outbuf));
+
+ GST_DEBUG (0,"gst_rtpL16parse_chain: pushing buffer of size %d", GST_BUFFER_SIZE(outbuf));
+
+ /* FIXME: According to RFC 1890, this is required, right? */
+#if G_BYTE_ORDER == G_LITTLE_ENDIAN
+ gst_rtpL16parse_ntohs (outbuf);
+#endif
+
+ gst_pad_push (rtpL16parse->srcpad, outbuf);
+
+ rtp_packet_free (packet);
+ gst_buffer_unref (buf);
+}
+
+static void
+gst_rtpL16parse_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec)
+{
+ GstRtpL16Parse *rtpL16parse;
+
+ /* it's not null if we got it, but it might not be ours */
+ g_return_if_fail (GST_IS_RTP_L16_PARSE (object));
+ rtpL16parse = GST_RTP_L16_PARSE (object);
+
+ switch (prop_id) {
+ case ARG_PAYLOAD_TYPE:
+ gst_rtpL16parse_payloadtype_change (rtpL16parse, g_value_get_int (value));
+ break;
+ case ARG_FREQUENCY:
+ rtpL16parse->frequency = g_value_get_int (value);
+ break;
+ default:
+ break;
+ }
+}
+
+static void
+gst_rtpL16parse_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec)
+{
+ GstRtpL16Parse *rtpL16parse;
+
+ /* it's not null if we got it, but it might not be ours */
+ g_return_if_fail (GST_IS_RTP_L16_PARSE (object));
+ rtpL16parse = GST_RTP_L16_PARSE (object);
+
+ switch (prop_id) {
+ case ARG_PAYLOAD_TYPE:
+ g_value_set_int (value, rtpL16parse->payload_type);
+ break;
+ case ARG_FREQUENCY:
+ g_value_set_int (value, rtpL16parse->frequency);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static GstElementStateReturn
+gst_rtpL16parse_change_state (GstElement * element)
+{
+ GstRtpL16Parse *rtpL16parse;
+
+ g_return_val_if_fail (GST_IS_RTP_L16_PARSE (element), GST_STATE_FAILURE);
+
+ rtpL16parse = GST_RTP_L16_PARSE (element);
+
+ GST_DEBUG (0, "state pending %d\n", GST_STATE_PENDING (element));
+
+ switch (GST_STATE_TRANSITION (element)) {
+ case GST_STATE_NULL_TO_READY:
+ break;
+ case GST_STATE_READY_TO_NULL:
+ break;
+ default:
+ break;
+ }
+
+ /* if we haven't failed already, give the parent class a chance to ;-) */
+ if (GST_ELEMENT_CLASS (parent_class)->change_state)
+ return GST_ELEMENT_CLASS (parent_class)->change_state (element);
+
+ return GST_STATE_SUCCESS;
+}
+
+gboolean
+gst_rtpL16parse_plugin_init (GModule * module, GstPlugin * plugin)
+{
+ GstElementFactory *rtpL16parse;
+
+ rtpL16parse = gst_element_factory_new ("rtpL16parse", GST_TYPE_RTP_L16_PARSE, &gst_rtp_L16parse_details);
+ g_return_val_if_fail (rtpL16parse != NULL, FALSE);
+
+ gst_element_factory_add_pad_template (rtpL16parse, GST_PAD_TEMPLATE_GET (src_factory));
+ gst_element_factory_add_pad_template (rtpL16parse, GST_PAD_TEMPLATE_GET (sink_factory));
+
+ gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (rtpL16parse));
+
+ return TRUE;
+}
--- /dev/null
+/* Gnome-Streamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ *
+ * 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.
+ */
+
+#ifndef __GST_RTP_L16_PARSE_H__
+#define __GST_RTP_L16_PARSE_H__
+
+#include <gst/gst.h>
+#include "rtp-packet.h"
+#include "gstrtp-common.h"
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif /* __cplusplus */
+
+/* Definition of structure storing data for this element. */
+typedef struct _GstRtpL16Parse GstRtpL16Parse;
+struct _GstRtpL16Parse
+{
+ GstElement element;
+
+ GstPad *sinkpad;
+ GstPad *srcpad;
+
+ guint frequency;
+ guint channels;
+
+ rtp_payload_t payload_type;
+};
+
+/* Standard definition defining a class for this element. */
+typedef struct _GstRtpL16ParseClass GstRtpL16ParseClass;
+struct _GstRtpL16ParseClass
+{
+ GstElementClass parent_class;
+};
+
+/* Standard macros for defining types for this element. */
+#define GST_TYPE_RTP_L16_PARSE \
+ (gst_rtpL16parse_get_type())
+#define GST_RTP_L16_PARSE(obj) \
+ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_L16_PARSE,GstRtpL16Parse))
+#define GST_RTP_L16_PARSE_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_L16_PARSE,GstRtpL16Parse))
+#define GST_IS_RTP_L16_PARSE(obj) \
+ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_L16_PARSE))
+#define GST_IS_RTP_L16_PARSE_CLASS(obj) \
+ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_L16_PARSE))
+
+gboolean gst_rtpL16parse_plugin_init (GModule * module, GstPlugin * plugin);
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+
+
+#endif /* __GST_RTP_L16_PARSE_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ *
+ * 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.
+ */
+
+#include <string.h>
+#include "gstrtpL16enc.h"
+
+/* elementfactory information */
+static GstElementDetails gst_rtpL16enc_details = {
+ "RTP RAW Audio Encoder",
+ "RtpL16Enc",
+ "LGPL",
+ "Encodes Raw Audio into an RTP packet",
+ VERSION,
+ "Zeeshan Ali <zak147@yahoo.com>",
+ "(C) 2003",
+};
+
+/* RtpL16Enc signals and args */
+enum
+{
+ /* FILL ME */
+ LAST_SIGNAL
+};
+
+enum
+{
+ /* FILL ME */
+ ARG_0,
+};
+
+GST_PAD_TEMPLATE_FACTORY (sink_factory,
+ "sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_CAPS_NEW (
+ "audio_raw",
+ "audio/raw",
+ "format", GST_PROPS_STRING ("int"),
+ "law", GST_PROPS_INT (0),
+ "endianness", GST_PROPS_INT (G_BYTE_ORDER),
+ "signed", GST_PROPS_BOOLEAN (TRUE),
+ "width", GST_PROPS_INT (16),
+ "depth", GST_PROPS_INT (16),
+ "rate", GST_PROPS_INT_RANGE (1000, 48000),
+ "channels", GST_PROPS_INT_RANGE (1, 2)
+ )
+);
+
+GST_PAD_TEMPLATE_FACTORY (src_factory,
+ "src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_CAPS_NEW (
+ "rtp",
+ "application/x-rtp",
+ NULL)
+);
+
+static void gst_rtpL16enc_class_init (GstRtpL16EncClass * klass);
+static void gst_rtpL16enc_init (GstRtpL16Enc * rtpL16enc);
+static void gst_rtpL16enc_chain (GstPad * pad, GstBuffer * buf);
+static void gst_rtpL16enc_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec);
+static void gst_rtpL16enc_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec);
+static GstPadLinkReturn gst_rtpL16enc_sinkconnect (GstPad * pad, GstCaps * caps);
+static GstElementStateReturn gst_rtpL16enc_change_state (GstElement * element);
+
+static GstElementClass *parent_class = NULL;
+
+static GType gst_rtpL16enc_get_type (void)
+{
+ static GType rtpL16enc_type = 0;
+
+ if (!rtpL16enc_type) {
+ static const GTypeInfo rtpL16enc_info = {
+ sizeof (GstRtpL16EncClass),
+ NULL,
+ NULL,
+ (GClassInitFunc) gst_rtpL16enc_class_init,
+ NULL,
+ NULL,
+ sizeof (GstRtpL16Enc),
+ 0,
+ (GInstanceInitFunc) gst_rtpL16enc_init,
+ };
+
+ rtpL16enc_type = g_type_register_static (GST_TYPE_ELEMENT, "GstRtpL16Enc", &rtpL16enc_info, 0);
+ }
+ return rtpL16enc_type;
+}
+
+static void
+gst_rtpL16enc_class_init (GstRtpL16EncClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
+
+ parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
+
+ gobject_class->set_property = gst_rtpL16enc_set_property;
+ gobject_class->get_property = gst_rtpL16enc_get_property;
+
+ gstelement_class->change_state = gst_rtpL16enc_change_state;
+}
+
+static void
+gst_rtpL16enc_init (GstRtpL16Enc * rtpL16enc)
+{
+ rtpL16enc->sinkpad = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (sink_factory), "sink");
+ rtpL16enc->srcpad = gst_pad_new_from_template (GST_PAD_TEMPLATE_GET (src_factory), "src");
+ gst_element_add_pad (GST_ELEMENT (rtpL16enc), rtpL16enc->sinkpad);
+ gst_element_add_pad (GST_ELEMENT (rtpL16enc), rtpL16enc->srcpad);
+ gst_pad_set_chain_function (rtpL16enc->sinkpad, gst_rtpL16enc_chain);
+ gst_pad_set_link_function (rtpL16enc->sinkpad, gst_rtpL16enc_sinkconnect);
+
+ rtpL16enc->frequency = 44100;
+ rtpL16enc->channels = 2;
+
+ rtpL16enc->next_time = 0;
+ rtpL16enc->time_interval = 0;
+
+ rtpL16enc->seq = 0;
+ rtpL16enc->ssrc = random ();
+}
+
+static GstPadLinkReturn
+gst_rtpL16enc_sinkconnect (GstPad * pad, GstCaps * caps)
+{
+ GstRtpL16Enc *rtpL16enc;
+
+ rtpL16enc = GST_RTP_L16_ENC (gst_pad_get_parent (pad));
+
+ gst_caps_get_int (caps, "rate", &rtpL16enc->frequency);
+ gst_caps_get_int (caps, "channels", &rtpL16enc->channels);
+
+ /* Pre-calculate what we can */
+ rtpL16enc->time_interval = GST_SECOND / (2 * rtpL16enc->channels * rtpL16enc->frequency);
+
+ return GST_PAD_LINK_OK;
+}
+
+
+void
+gst_rtpL16enc_htons (GstBuffer *buf)
+{
+ guint16 *i, *len;
+
+ /* FIXME: is this code correct or even sane at all? */
+ i = (guint16 *) GST_BUFFER_DATA(buf);
+ len = i + GST_BUFFER_SIZE (buf) / sizeof (guint16 *);
+
+ for (; i<len; i++) {
+ *i = g_htons (*i);
+ }
+}
+
+static void
+gst_rtpL16enc_chain (GstPad * pad, GstBuffer * buf)
+{
+ GstRtpL16Enc *rtpL16enc;
+ GstBuffer *outbuf;
+ Rtp_Packet packet;
+
+ g_return_if_fail (pad != NULL);
+ g_return_if_fail (GST_IS_PAD (pad));
+ g_return_if_fail (buf != NULL);
+
+ rtpL16enc = GST_RTP_L16_ENC (GST_OBJECT_PARENT (pad));
+
+ g_return_if_fail (rtpL16enc != NULL);
+ g_return_if_fail (GST_IS_RTP_L16_ENC (rtpL16enc));
+
+ if (GST_IS_EVENT (buf)) {
+ GstEvent *event = GST_EVENT (buf);
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_DISCONTINUOUS:
+ GST_DEBUG (GST_CAT_EVENT, "discont");
+ rtpL16enc->next_time = 0;
+ gst_pad_event_default (pad, event);
+ return;
+ default:
+ gst_pad_event_default (pad, event);
+ return;
+ }
+ }
+
+ /* We only need the header */
+ packet = rtp_packet_new_allocate (0, 0, 0);
+
+ rtp_packet_set_csrc_count (packet, 0);
+ rtp_packet_set_extension (packet, 0);
+ rtp_packet_set_padding (packet, 0);
+ rtp_packet_set_version (packet, RTP_VERSION);
+ rtp_packet_set_marker (packet, 0);
+ rtp_packet_set_ssrc (packet, g_htonl (rtpL16enc->ssrc));
+ rtp_packet_set_seq (packet, g_htons (rtpL16enc->seq));
+ rtp_packet_set_timestamp (packet, g_htonl ((guint32) rtpL16enc->next_time / GST_SECOND));
+
+ if (rtpL16enc->channels == 1) {
+ rtp_packet_set_payload_type (packet, (guint8) PAYLOAD_L16_MONO);
+ }
+
+ else {
+ rtp_packet_set_payload_type (packet, (guint8) PAYLOAD_L16_STEREO);
+ }
+
+ /* FIXME: According to RFC 1890, this is required, right? */
+#if G_BYTE_ORDER == G_LITTLE_ENDIAN
+ gst_rtpL16enc_htons (buf);
+#endif
+
+ outbuf = gst_buffer_new ();
+ GST_BUFFER_SIZE (outbuf) = rtp_packet_get_packet_len (packet) + GST_BUFFER_SIZE (buf);
+ GST_BUFFER_DATA (outbuf) = g_malloc (GST_BUFFER_SIZE (outbuf));
+ GST_BUFFER_TIMESTAMP (outbuf) = rtpL16enc->next_time;
+
+ memcpy (GST_BUFFER_DATA (outbuf), packet->data, rtp_packet_get_packet_len (packet));
+ memcpy (GST_BUFFER_DATA (outbuf) + rtp_packet_get_packet_len(packet), GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
+
+ GST_DEBUG (0,"gst_rtpL16enc_chain: pushing buffer of size %d", GST_BUFFER_SIZE(outbuf));
+ gst_pad_push (rtpL16enc->srcpad, outbuf);
+
+ ++rtpL16enc->seq;
+ rtpL16enc->next_time += rtpL16enc->time_interval * GST_BUFFER_SIZE (buf);
+
+ rtp_packet_free (packet);
+ gst_buffer_unref (buf);
+}
+
+static void
+gst_rtpL16enc_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec)
+{
+ GstRtpL16Enc *rtpL16enc;
+
+ /* it's not null if we got it, but it might not be ours */
+ g_return_if_fail (GST_IS_RTP_L16_ENC (object));
+ rtpL16enc = GST_RTP_L16_ENC (object);
+
+ switch (prop_id) {
+ default:
+ break;
+ }
+}
+
+static void
+gst_rtpL16enc_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec)
+{
+ GstRtpL16Enc *rtpL16enc;
+
+ /* it's not null if we got it, but it might not be ours */
+ g_return_if_fail (GST_IS_RTP_L16_ENC (object));
+ rtpL16enc = GST_RTP_L16_ENC (object);
+
+ switch (prop_id) {
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static GstElementStateReturn
+gst_rtpL16enc_change_state (GstElement * element)
+{
+ GstRtpL16Enc *rtpL16enc;
+
+ g_return_val_if_fail (GST_IS_RTP_L16_ENC (element), GST_STATE_FAILURE);
+
+ rtpL16enc = GST_RTP_L16_ENC (element);
+
+ GST_DEBUG (0, "state pending %d\n", GST_STATE_PENDING (element));
+
+ /* if going down into NULL state, close the file if it's open */
+ switch (GST_STATE_TRANSITION (element)) {
+ case GST_STATE_NULL_TO_READY:
+ break;
+
+ case GST_STATE_READY_TO_NULL:
+ break;
+
+ default:
+ break;
+ }
+
+ /* if we haven't failed already, give the parent class a chance to ;-) */
+ if (GST_ELEMENT_CLASS (parent_class)->change_state)
+ return GST_ELEMENT_CLASS (parent_class)->change_state (element);
+
+ return GST_STATE_SUCCESS;
+}
+
+gboolean
+gst_rtpL16enc_plugin_init (GModule * module, GstPlugin * plugin)
+{
+ GstElementFactory *rtpL16enc;
+
+ rtpL16enc = gst_element_factory_new ("rtpL16enc", GST_TYPE_RTP_L16_ENC, &gst_rtpL16enc_details);
+ g_return_val_if_fail (rtpL16enc != NULL, FALSE);
+
+ gst_element_factory_add_pad_template (rtpL16enc, GST_PAD_TEMPLATE_GET (sink_factory));
+ gst_element_factory_add_pad_template (rtpL16enc, GST_PAD_TEMPLATE_GET (src_factory));
+
+ gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (rtpL16enc));
+
+ return TRUE;
+}
--- /dev/null
+/* Gnome-Streamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ *
+ * 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.
+ */
+
+
+#ifndef __GST_RTP_L16_ENC_H__
+#define __GST_RTP_L16_ENC_H__
+
+#include <gst/gst.h>
+#include "rtp-packet.h"
+#include "gstrtp-common.h"
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif /* __cplusplus */
+
+/* Definition of structure storing data for this element. */
+typedef struct _GstRtpL16Enc GstRtpL16Enc;
+struct _GstRtpL16Enc
+{
+ GstElement element;
+
+ GstPad *sinkpad;
+ GstPad *srcpad;
+
+ guint frequency;
+ guint channels;
+
+ /* the timestamp of the next frame */
+ guint64 next_time;
+ /* the interval between frames */
+ guint64 time_interval;
+
+ guint32 ssrc;
+ guint16 seq;
+};
+
+/* Standard definition defining a class for this element. */
+typedef struct _GstRtpL16EncClass GstRtpL16EncClass;
+struct _GstRtpL16EncClass
+{
+ GstElementClass parent_class;
+};
+
+/* Standard macros for defining types for this element. */
+#define GST_TYPE_RTP_L16_ENC \
+ (gst_rtpL16enc_get_type())
+#define GST_RTP_L16_ENC(obj) \
+ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_L16_ENC,GstRtpL16Enc))
+#define GST_RTP_L16_ENC_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_L16_ENC,GstRtpL16Enc))
+#define GST_IS_RTP_L16_ENC(obj) \
+ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_L16_ENC))
+#define GST_IS_RTP_L16_ENC_CLASS(obj) \
+ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_L16_ENC))
+
+gboolean gst_rtpL16enc_plugin_init (GModule * module, GstPlugin * plugin);
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+
+
+#endif /* __GST_RTP_L16_ENC_H__ */
--- /dev/null
+/*
+ Librtp - a library for the RTP/RTCP protocol
+ Copyright (C) 2000 Roland Dreier
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of the GNU General Public License as published by
+ the Free Software Foundation; either version 2 of the License, or
+ (at your option) any later version.
+
+ This program 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 General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+
+ $Id$
+*/
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include "rtp-packet.h"
+
+#include <glib.h>
+#include <stdlib.h>
+#include <sys/types.h>
+#include <sys/socket.h>
+#include <netinet/in.h>
+#include <string.h>
+#include <errno.h>
+
+Rtp_Packet
+rtp_packet_new_take_data(gpointer data, guint data_len)
+{
+ Rtp_Packet packet;
+
+ //g_return_val_if_fail(data_len < RTP_MTU, NULL);
+
+ packet = g_malloc(sizeof *packet);
+
+ packet -> data = data;
+ packet -> data_len = data_len;
+
+ return packet;
+}
+
+Rtp_Packet
+rtp_packet_new_copy_data(gpointer data, guint data_len)
+{
+ Rtp_Packet packet;
+
+ //g_return_val_if_fail(data_len < RTP_MTU, NULL);
+
+ packet = g_malloc(sizeof *packet);
+
+ packet -> data = g_memdup(data, data_len);
+ packet -> data_len = data_len;
+
+ return packet;
+}
+
+Rtp_Packet
+rtp_packet_new_allocate(guint payload_len, guint pad_len, guint csrc_count)
+{
+ guint len;
+ Rtp_Packet packet;
+
+ g_return_val_if_fail(csrc_count <= 15, NULL);
+
+ len = RTP_HEADER_LEN
+ + csrc_count * sizeof(guint32)
+ + payload_len + pad_len;
+
+ //g_return_val_if_fail(len < RTP_MTU, NULL);
+
+ packet = g_malloc(sizeof *packet);
+
+ packet -> data_len = len;
+ packet -> data = g_malloc(len);
+
+ return(packet);
+}
+
+
+void
+rtp_packet_free(Rtp_Packet packet)
+{
+ g_return_if_fail(packet != NULL);
+
+ g_free(packet -> data);
+ g_free(packet);
+}
+
+Rtp_Packet
+rtp_packet_read(int fd, struct sockaddr *fromaddr, socklen_t *fromlen)
+{
+ int packlen;
+ gpointer buf;
+
+ buf = g_malloc(RTP_MTU);
+
+ packlen = recvfrom(fd, buf, RTP_MTU, 0, fromaddr, fromlen);
+
+ if (packlen < 0) {
+ g_error("rtp_packet_read: recvfrom: %d %s", errno, strerror(errno));
+ /*exit(1);*/
+ return NULL;
+ }
+
+ return rtp_packet_new_take_data(buf, packlen);
+}
+
+void
+rtp_packet_send(Rtp_Packet packet, int fd, struct sockaddr *toaddr, socklen_t tolen)
+{
+ g_return_if_fail(packet != NULL);
+
+ sendto(fd, (void *) packet -> data,
+ packet -> data_len, 0,
+ toaddr, tolen);
+}
+
+guint8
+rtp_packet_get_version(Rtp_Packet packet)
+{
+ g_return_val_if_fail(packet != NULL, 0);
+
+ return ((Rtp_Header) packet -> data) -> version;
+}
+
+void
+rtp_packet_set_version(Rtp_Packet packet, guint8 version)
+{
+ g_return_if_fail(packet != NULL);
+ g_return_if_fail(version < 0x04);
+
+ ((Rtp_Header) packet -> data) -> version = version;
+}
+
+guint8
+rtp_packet_get_padding(Rtp_Packet packet)
+{
+ g_return_val_if_fail(packet != NULL, 0);
+
+ return ((Rtp_Header) packet -> data) -> padding;
+}
+
+void
+rtp_packet_set_padding(Rtp_Packet packet, guint8 padding)
+{
+ g_return_if_fail(packet != NULL);
+ g_return_if_fail(padding < 0x02);
+
+ ((Rtp_Header) packet -> data) -> padding = padding;
+}
+
+guint8
+rtp_packet_get_csrc_count(Rtp_Packet packet)
+{
+ g_return_val_if_fail(packet != NULL, 0);
+
+ return ((Rtp_Header) packet -> data) -> csrc_count;
+}
+
+guint8
+rtp_packet_get_extension(Rtp_Packet packet)
+{
+ g_return_val_if_fail(packet != NULL, 0);
+
+ return ((Rtp_Header) packet -> data) -> extension;
+}
+
+void
+rtp_packet_set_extension(Rtp_Packet packet, guint8 extension)
+{
+ g_return_if_fail(packet != NULL);
+ g_return_if_fail(extension < 0x02);
+
+ ((Rtp_Header) packet -> data) -> extension = extension;
+}
+
+void
+rtp_packet_set_csrc_count(Rtp_Packet packet, guint8 csrc_count)
+{
+ g_return_if_fail(packet != NULL);
+ g_return_if_fail(csrc_count < 0x04);
+
+ ((Rtp_Header) packet -> data) -> csrc_count = csrc_count;
+}
+
+guint8
+rtp_packet_get_marker(Rtp_Packet packet)
+{
+ g_return_val_if_fail(packet != NULL, 0);
+
+ return ((Rtp_Header) packet -> data) -> marker;
+}
+
+void
+rtp_packet_set_marker(Rtp_Packet packet, guint8 marker)
+{
+ g_return_if_fail(packet != NULL);
+ g_return_if_fail(marker < 0x02);
+
+ ((Rtp_Header) packet -> data) -> marker = marker;
+}
+
+guint8
+rtp_packet_get_payload_type(Rtp_Packet packet)
+{
+ g_return_val_if_fail(packet != NULL, 0);
+
+ return ((Rtp_Header) packet -> data) -> payload_type;
+}
+
+void
+rtp_packet_set_payload_type(Rtp_Packet packet, guint8 payload_type)
+{
+ g_return_if_fail(packet != NULL);
+ g_return_if_fail(payload_type < 0x80);
+
+ ((Rtp_Header) packet -> data) -> payload_type = payload_type;
+}
+
+guint16
+rtp_packet_get_seq(Rtp_Packet packet)
+{
+ g_return_val_if_fail(packet != NULL, 0);
+
+ return g_ntohs(((Rtp_Header) packet -> data) -> seq);
+}
+
+void
+rtp_packet_set_seq(Rtp_Packet packet, guint16 seq)
+{
+ g_return_if_fail(packet != NULL);
+
+ ((Rtp_Header) packet -> data) -> seq = g_htons(seq);
+}
+
+guint32
+rtp_packet_get_timestamp(Rtp_Packet packet)
+{
+ g_return_val_if_fail(packet != NULL, 0);
+
+ return g_ntohl(((Rtp_Header) packet -> data) -> timestamp);
+}
+
+void
+rtp_packet_set_timestamp(Rtp_Packet packet, guint32 timestamp)
+{
+ g_return_if_fail(packet != NULL);
+
+ ((Rtp_Header) packet -> data) -> timestamp = g_htonl(timestamp);
+}
+
+guint32
+rtp_packet_get_ssrc(Rtp_Packet packet)
+{
+ g_return_val_if_fail(packet != NULL, 0);
+
+ return g_ntohl(((Rtp_Header) packet -> data) -> ssrc);
+}
+
+void
+rtp_packet_set_ssrc(Rtp_Packet packet, guint32 ssrc)
+{
+ g_return_if_fail(packet != NULL);
+
+ ((Rtp_Header) packet -> data) -> ssrc = g_htonl(ssrc);
+}
+
+guint
+rtp_packet_get_payload_len(Rtp_Packet packet)
+{
+ guint len;
+
+ g_return_val_if_fail(packet != NULL, 0);
+
+ len = packet -> data_len
+ - RTP_HEADER_LEN
+ - rtp_packet_get_csrc_count(packet) * sizeof(guint32);
+
+ if (rtp_packet_get_padding(packet)) {
+ len -= ((guint8 *) packet -> data)[packet -> data_len - 1];
+ }
+
+ return len;
+}
+
+gpointer
+rtp_packet_get_payload(Rtp_Packet packet)
+{
+ g_return_val_if_fail(packet != NULL, NULL);
+
+ return ((char *) packet -> data)
+ + RTP_HEADER_LEN
+ + rtp_packet_get_csrc_count(packet) * sizeof(guint32);
+}
+
+guint
+rtp_packet_get_packet_len(Rtp_Packet packet)
+{
+ g_return_val_if_fail(packet != NULL, 0);
+
+ return packet -> data_len;
+}
--- /dev/null
+/*
+ Gnome-o-Phone - A program for internet telephony
+ Copyright (C) 1999 Roland Dreier
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of the GNU General Public License as published by
+ the Free Software Foundation; either version 2 of the License, or
+ (at your option) any later version.
+
+ This program 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 General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program; if not, write to the Free Software
+ Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+
+ $Id$
+*/
+
+#ifndef _RTP_PACKET_H
+#define _RTP_PACKET_H 1
+
+#include <glib.h>
+#include <netinet/in.h>
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+enum {
+ RTP_VERSION = 2,
+ RTP_HEADER_LEN = 12,
+ RTP_MTU = 2048
+};
+
+typedef struct Rtp_Header *Rtp_Header;
+
+struct Rtp_Packet_Struct {
+ gpointer data;
+ guint data_len;
+};
+
+struct Rtp_Header {
+#if G_BYTE_ORDER == G_LITTLE_ENDIAN
+ unsigned int csrc_count:4; /* CSRC count */
+ unsigned int extension:1; /* header extension flag */
+ unsigned int padding:1; /* padding flag */
+ unsigned int version:2; /* protocol version */
+ unsigned int payload_type:7; /* payload type */
+ unsigned int marker:1; /* marker bit */
+#elif G_BYTE_ORDER == G_BIG_ENDIAN
+ unsigned int version:2; /* protocol version */
+ unsigned int padding:1; /* padding flag */
+ unsigned int extension:1; /* header extension flag */
+ unsigned int csrc_count:4; /* CSRC count */
+ unsigned int marker:1; /* marker bit */
+ unsigned int payload_type:7; /* payload type */
+#else
+#error "G_BYTE_ORDER should be big or little endian."
+#endif
+ guint16 seq; /* sequence number */
+ guint32 timestamp; /* timestamp */
+ guint32 ssrc; /* synchronization source */
+ guint32 csrc[1]; /* optional CSRC list */
+};
+
+typedef struct Rtp_Packet_Struct *Rtp_Packet;
+
+Rtp_Packet rtp_packet_new_take_data(gpointer data, guint data_len);
+Rtp_Packet rtp_packet_new_copy_data(gpointer data, guint data_len);
+Rtp_Packet rtp_packet_new_allocate(guint payload_len,
+ guint pad_len, guint csrc_count);
+void rtp_packet_free(Rtp_Packet packet);
+Rtp_Packet rtp_packet_read(int fd, struct sockaddr *fromaddr, socklen_t *fromlen);
+void rtp_packet_send(Rtp_Packet packet, int fd, struct sockaddr *toaddr, socklen_t tolen);
+guint8 rtp_packet_get_version(Rtp_Packet packet);
+void rtp_packet_set_version(Rtp_Packet packet, guint8 version);
+guint8 rtp_packet_get_padding(Rtp_Packet packet);
+void rtp_packet_set_padding(Rtp_Packet packet, guint8 padding);
+guint8 rtp_packet_get_csrc_count(Rtp_Packet packet);
+guint8 rtp_packet_get_extension(Rtp_Packet packet);
+void rtp_packet_set_extension(Rtp_Packet packet, guint8 extension);
+void rtp_packet_set_csrc_count(Rtp_Packet packet, guint8 csrc_count);
+guint8 rtp_packet_get_marker(Rtp_Packet packet);
+void rtp_packet_set_marker(Rtp_Packet packet, guint8 marker);
+guint8 rtp_packet_get_payload_type(Rtp_Packet packet);
+void rtp_packet_set_payload_type(Rtp_Packet packet, guint8 payload_type);
+guint16 rtp_packet_get_seq(Rtp_Packet packet);
+void rtp_packet_set_seq(Rtp_Packet packet, guint16 seq);
+guint32 rtp_packet_get_timestamp(Rtp_Packet packet);
+void rtp_packet_set_timestamp(Rtp_Packet packet, guint32 timestamp);
+guint32 rtp_packet_get_ssrc(Rtp_Packet packet);
+void rtp_packet_set_ssrc(Rtp_Packet packet, guint32 ssrc);
+guint rtp_packet_get_payload_len(Rtp_Packet packet);
+gpointer rtp_packet_get_payload(Rtp_Packet packet);
+guint rtp_packet_get_packet_len(Rtp_Packet packet);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* rtp-packet.h */
projects / platform / upstream / gstreamer.git / commitdiff