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Diffstat (limited to 'gst/rtpmanager/gstrtpclient.c')
| -rw-r--r-- | gst/rtpmanager/gstrtpclient.c | 482 |
1 files changed, 482 insertions, 0 deletions
diff --git a/gst/rtpmanager/gstrtpclient.c b/gst/rtpmanager/gstrtpclient.c new file mode 100644 index 00000000..2984d3f9 --- /dev/null +++ b/gst/rtpmanager/gstrtpclient.c @@ -0,0 +1,482 @@ +/* GStreamer + * Copyright (C) <2007> Wim Taymans <wim@fluendo.com> + * + * 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-rtpclient + * @short_description: handle media from one RTP client + * @see_also: rtpjitterbuffer, rtpbin, rtpsession + * + * <refsect2> + * <para> + * This element handles RTP data from one client. It accepts multiple RTP streams that + * should be synchronized together. + * </para> + * <title>Example pipelines</title> + * <para> + * <programlisting> + * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! alsasink + * </programlisting> + * </para> + * </refsect2> + * + * Last reviewed on 2007-04-02 (0.10.6) + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif +#include <string.h> + +#include "gstrtpclient.h" + +/* elementfactory information */ +static const GstElementDetails rtpclient_details = +GST_ELEMENT_DETAILS ("RTP Client", + "Filter/Editor/Video", + "Implement an RTP client", + "Wim Taymans <wim@fluendo.com>"); + +/* sink pads */ +static GstStaticPadTemplate rtpclient_rtp_sink_template = +GST_STATIC_PAD_TEMPLATE ("rtp_sink_%d", + GST_PAD_SINK, + GST_PAD_REQUEST, + GST_STATIC_CAPS ("application/x-rtp") + ); + +static GstStaticPadTemplate rtpclient_sync_sink_template = +GST_STATIC_PAD_TEMPLATE ("sync_sink_%d", + GST_PAD_SINK, + GST_PAD_REQUEST, + GST_STATIC_CAPS ("application/x-rtcp") + ); + +/* src pads */ +static GstStaticPadTemplate rtpclient_rtp_src_template = +GST_STATIC_PAD_TEMPLATE ("rtp_src_%d_%d", + GST_PAD_SRC, + GST_PAD_SOMETIMES, + GST_STATIC_CAPS ("application/x-rtp") + ); + +#define GST_RTP_CLIENT_GET_PRIVATE(obj) \ + (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_CLIENT, GstRTPClientPrivate)) + +struct _GstRTPClientPrivate +{ +}; + +/* all the info needed to handle the stream with SSRC */ +typedef struct +{ + GstRTPClient *client; + + /* the SSRC of this stream */ + guint32 ssrc; + + /* RTP and RTCP in */ + GstPad *rtp_sink; + GstPad *sync_sink; + + /* the jitterbuffer */ + GstElement *jitterbuffer; + /* the payload demuxer */ + GstElement *ptdemux; + /* the new-pad signal */ + gulong new_pad_sig; +} GstRTPClientStream; + +/* the PT demuxer found a new payload type */ +static void +new_pad (GstElement * element, GstPad * pad, GstRTPClientStream * stream) +{ +} + +/* create a new stream for SSRC. + * + * We create a jitterbuffer and an payload demuxer for the SSRC. The sinkpad of + * the jitterbuffer is ghosted to the bin. We connect a pad-added signal to + * rtpptdemux so that we can ghost the payload pads outside. + * + * +-----------------+ +---------------+ + * | rtpjitterbuffer | | rtpptdemux | + * +- sink src - sink | + * / +-----------------+ +---------------+ + * + */ +static GstRTPClientStream * +create_stream (GstRTPClient * rtpclient, guint32 ssrc) +{ + GstRTPClientStream *stream; + gchar *name; + GstPad *srcpad, *sinkpad; + GstPadLinkReturn res; + + stream = g_new0 (GstRTPClientStream, 1); + stream->ssrc = ssrc; + stream->client = rtpclient; + + stream->jitterbuffer = gst_element_factory_make ("rtpjitterbuffer", NULL); + if (!stream->jitterbuffer) + goto no_jitterbuffer; + + stream->ptdemux = gst_element_factory_make ("rtpptdemux", NULL); + if (!stream->ptdemux) + goto no_ptdemux; + + /* add elements to bin */ + gst_bin_add (GST_BIN_CAST (rtpclient), stream->jitterbuffer); + gst_bin_add (GST_BIN_CAST (rtpclient), stream->ptdemux); + + /* link jitterbuffer and PT demuxer */ + srcpad = gst_element_get_pad (stream->jitterbuffer, "src"); + sinkpad = gst_element_get_pad (stream->ptdemux, "sink"); + res = gst_pad_link (srcpad, sinkpad); + gst_object_unref (srcpad); + gst_object_unref (sinkpad); + + if (res != GST_PAD_LINK_OK) + goto could_not_link; + + /* add stream to list */ + rtpclient->streams = g_list_prepend (rtpclient->streams, stream); + + /* ghost sinkpad */ + name = g_strdup_printf ("rtp_sink_%d", ssrc); + sinkpad = gst_element_get_pad (stream->jitterbuffer, "sink"); + stream->rtp_sink = gst_ghost_pad_new (name, sinkpad); + gst_object_unref (sinkpad); + g_free (name); + gst_element_add_pad (GST_ELEMENT_CAST (rtpclient), stream->rtp_sink); + + /* add signal to ptdemuxer */ + stream->new_pad_sig = + g_signal_connect (G_OBJECT (stream->ptdemux), "pad-added", + G_CALLBACK (new_pad), stream); + + return stream; + + /* ERRORS */ +no_jitterbuffer: + { + g_free (stream); + g_warning ("could not create rtpjitterbuffer element"); + return NULL; + } +no_ptdemux: + { + gst_object_unref (stream->jitterbuffer); + g_free (stream); + g_warning ("could not create rtpptdemux element"); + return NULL; + } +could_not_link: + { + gst_bin_remove (GST_BIN_CAST (rtpclient), stream->jitterbuffer); + gst_bin_remove (GST_BIN_CAST (rtpclient), stream->ptdemux); + g_free (stream); + g_warning ("could not link jitterbuffer and rtpptdemux element"); + return NULL; + } +} + +#if 0 +static void +free_stream (GstRTPClientStream * stream) +{ + gst_object_unref (stream->jitterbuffer); + g_free (stream); +} +#endif + +/* find the stream for the given SSRC, return NULL if the stream did not exist + */ +static GstRTPClientStream * +find_stream_by_ssrc (GstRTPClient * client, guint32 ssrc) +{ + GstRTPClientStream *stream; + GList *walk; + + for (walk = client->streams; walk; walk = g_list_next (walk)) { + stream = (GstRTPClientStream *) walk->data; + if (stream->ssrc == ssrc) + return stream; + } + return NULL; +} + +/* signals and args */ +enum +{ + /* FILL ME */ + LAST_SIGNAL +}; + +enum +{ + PROP_0 +}; + +/* GObject vmethods */ +static void gst_rtp_client_finalize (GObject * object); +static void gst_rtp_client_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec); +static void gst_rtp_client_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec); + +/* GstElement vmethods */ +static GstStateChangeReturn gst_rtp_client_change_state (GstElement * element, + GstStateChange transition); +static GstPad *gst_rtp_client_request_new_pad (GstElement * element, + GstPadTemplate * templ, const gchar * name); +static void gst_rtp_client_release_pad (GstElement * element, GstPad * pad); + +/*static guint gst_rtp_client_signals[LAST_SIGNAL] = { 0 }; */ + +GST_BOILERPLATE (GstRTPClient, gst_rtp_client, GstBin, GST_TYPE_BIN); + +static void +gst_rtp_client_base_init (gpointer klass) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (klass); + + /* sink pads */ + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&rtpclient_rtp_sink_template)); + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&rtpclient_sync_sink_template)); + + /* src pads */ + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&rtpclient_rtp_src_template)); + + gst_element_class_set_details (element_class, &rtpclient_details); +} + +static void +gst_rtp_client_class_init (GstRTPClientClass * klass) +{ + GObjectClass *gobject_class; + GstElementClass *gstelement_class; + + gobject_class = (GObjectClass *) klass; + gstelement_class = (GstElementClass *) klass; + + g_type_class_add_private (klass, sizeof (GstRTPClientPrivate)); + + gobject_class->finalize = gst_rtp_client_finalize; + gobject_class->set_property = gst_rtp_client_set_property; + gobject_class->get_property = gst_rtp_client_get_property; + + gstelement_class->change_state = + GST_DEBUG_FUNCPTR (gst_rtp_client_change_state); + gstelement_class->request_new_pad = + GST_DEBUG_FUNCPTR (gst_rtp_client_request_new_pad); + gstelement_class->release_pad = + GST_DEBUG_FUNCPTR (gst_rtp_client_release_pad); +} + +static void +gst_rtp_client_init (GstRTPClient * rtpclient, GstRTPClientClass * klass) +{ + rtpclient->priv = GST_RTP_CLIENT_GET_PRIVATE (rtpclient); +} + +static void +gst_rtp_client_finalize (GObject * object) +{ + GstRTPClient *rtpclient; + + rtpclient = GST_RTP_CLIENT (object); + + G_OBJECT_CLASS (parent_class)->finalize (object); +} + +static void +gst_rtp_client_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec) +{ + GstRTPClient *rtpclient; + + rtpclient = GST_RTP_CLIENT (object); + + switch (prop_id) { + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +gst_rtp_client_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec) +{ + GstRTPClient *rtpclient; + + rtpclient = GST_RTP_CLIENT (object); + + switch (prop_id) { + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static GstStateChangeReturn +gst_rtp_client_change_state (GstElement * element, GstStateChange transition) +{ + GstStateChangeReturn res; + GstRTPClient *rtpclient; + + rtpclient = GST_RTP_CLIENT (element); + + switch (transition) { + case GST_STATE_CHANGE_NULL_TO_READY: + break; + case GST_STATE_CHANGE_READY_TO_PAUSED: + break; + case GST_STATE_CHANGE_PAUSED_TO_PLAYING: + break; + default: + break; + } + + res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); + + switch (transition) { + case GST_STATE_CHANGE_PLAYING_TO_PAUSED: + break; + case GST_STATE_CHANGE_PAUSED_TO_READY: + break; + case GST_STATE_CHANGE_READY_TO_NULL: + break; + default: + break; + } + return res; +} + +/* We have 2 request pads (rtp_sink_%d and sync_sink_%d), the %d is assumed to + * be the SSRC of the stream. + * + * We require that the rtp pad is requested first for a particular SSRC, then + * (optionaly) the sync pad can be requested. If no sync pad is requested, no + * sync information can be exchanged for this stream. + */ +static GstPad * +gst_rtp_client_request_new_pad (GstElement * element, + GstPadTemplate * templ, const gchar * name) +{ + GstRTPClient *rtpclient; + GstElementClass *klass; + GstPadTemplate *rtp_sink_templ, *sync_sink_templ; + guint32 ssrc; + GstRTPClientStream *stream; + GstPad *result; + + g_return_val_if_fail (templ != NULL, NULL); + g_return_val_if_fail (GST_IS_RTP_CLIENT (element), NULL); + + if (templ->direction != GST_PAD_SINK) + goto wrong_direction; + + rtpclient = GST_RTP_CLIENT (element); + klass = GST_ELEMENT_GET_CLASS (element); + + /* figure out the template */ + rtp_sink_templ = gst_element_class_get_pad_template (klass, "rtp_sink_%d"); + sync_sink_templ = gst_element_class_get_pad_template (klass, "sync_sink_%d"); + + if (templ != rtp_sink_templ && templ != sync_sink_templ) + goto wrong_template; + + if (templ == rtp_sink_templ) { + /* create new rtp sink pad. If a stream with the pad number already exists + * we have an error, else we create the sinkpad, add a jitterbuffer and + * ptdemuxer. */ + if (name == NULL || strlen (name) < 9) + goto no_name; + + ssrc = atoi (&name[9]); + + /* see if a stream with that name exists, if so we have an error. */ + stream = find_stream_by_ssrc (rtpclient, ssrc); + if (stream != NULL) + goto stream_exists; + + /* ok, create new stream */ + stream = create_stream (rtpclient, ssrc); + if (stream == NULL) + goto stream_not_found; + + result = stream->rtp_sink; + } else { + /* create new rtp sink pad. We can only do this if the RTP pad was + * requested before, meaning the session with the padnumber must exist. */ + if (name == NULL || strlen (name) < 10) + goto no_name; + + ssrc = atoi (&name[10]); + + /* find stream */ + stream = find_stream_by_ssrc (rtpclient, ssrc); + if (stream == NULL) + goto stream_not_found; + + stream->sync_sink = + gst_pad_new_from_static_template (&rtpclient_sync_sink_template, name); + gst_element_add_pad (GST_ELEMENT_CAST (rtpclient), stream->sync_sink); + + result = stream->sync_sink; + } + + return result; + + /* ERRORS */ +wrong_direction: + { + g_warning ("rtpclient: request pad that is not a SINK pad"); + return NULL; + } +wrong_template: + { + g_warning ("rtpclient: this is not our template"); + return NULL; + } +no_name: + { + g_warning ("rtpclient: no padname was specified"); + return NULL; + } +stream_exists: + { + g_warning ("rtpclient: stream with SSRC %d already registered", ssrc); + return NULL; + } +stream_not_found: + { + g_warning ("rtpclient: stream with SSRC %d not yet registered", ssrc); + return NULL; + } +} + +static void +gst_rtp_client_release_pad (GstElement * element, GstPad * pad) +{ +} |