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gstreamer/gst/rtsp-server/rtsp-stream.c

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/* GStreamer
* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.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.
*/
#include <string.h>
#include <stdlib.h>
#include <gio/gio.h>
#include <gst/app/gstappsrc.h>
#include <gst/app/gstappsink.h>
#include "rtsp-stream.h"
enum
{
PROP_0,
PROP_LAST
};
GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
#define GST_CAT_DEFAULT rtsp_stream_debug
static GQuark ssrc_stream_map_key;
static void gst_rtsp_stream_finalize (GObject * obj);
G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
static void
gst_rtsp_stream_class_init (GstRTSPStreamClass * klass)
{
GObjectClass *gobject_class;
gobject_class = G_OBJECT_CLASS (klass);
gobject_class->finalize = gst_rtsp_stream_finalize;
GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
}
static void
gst_rtsp_stream_init (GstRTSPStream * media)
{
}
static void
gst_rtsp_stream_finalize (GObject * obj)
{
GstRTSPStream *stream;
stream = GST_RTSP_STREAM (obj);
/* we really need to be unjoined now */
g_return_if_fail (!stream->is_joined);
gst_object_unref (stream->payloader);
gst_object_unref (stream->srcpad);
G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
}
/**
* gst_rtsp_stream_new:
* @idx: an index
* @srcpad: a #GstPad
* @payloader: a #GstElement
*
* Create a new media stream with index @idx that handles RTP data on
* @srcpad and has a payloader element @payloader.
*
* Returns: a new #GstRTSPStream
*/
GstRTSPStream *
gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * srcpad)
{
GstRTSPStream *stream;
g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
g_return_val_if_fail (GST_IS_PAD (srcpad), NULL);
g_return_val_if_fail (GST_PAD_IS_SRC (srcpad), NULL);
stream = g_object_new (GST_TYPE_RTSP_STREAM, NULL);
stream->idx = idx;
stream->payloader = gst_object_ref (payloader);
stream->srcpad = gst_object_ref (srcpad);
return stream;
}
/**
* gst_rtsp_stream_set_mtu:
* @stream: a #GstRTSPStream
* @mtu: a new MTU
*
* Configure the mtu in the payloader of @stream to @mtu.
*/
void
gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu)
{
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
g_object_set (G_OBJECT (stream->payloader), "mtu", mtu, NULL);
}
/**
* gst_rtsp_stream_get_mtu:
* @stream: a #GstRTSPStream
*
* Get the configured MTU in the payloader of @stream.
*
* Returns: the MTU of the payloader.
*/
guint
gst_rtsp_stream_get_mtu (GstRTSPStream * stream)
{
guint mtu;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
g_object_get (G_OBJECT (stream->payloader), "mtu", &mtu, NULL);
return mtu;
}
static gboolean
alloc_ports (GstRTSPStream * stream)
{
GstStateChangeReturn ret;
GstElement *udpsrc0, *udpsrc1;
GstElement *udpsink0, *udpsink1;
gint tmp_rtp, tmp_rtcp;
guint count;
gint rtpport, rtcpport;
GSocket *socket;
const gchar *host;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
udpsrc0 = NULL;
udpsrc1 = NULL;
udpsink0 = NULL;
udpsink1 = NULL;
count = 0;
/* Start with random port */
tmp_rtp = 0;
if (stream->is_ipv6)
host = "udp://[::0]";
else
host = "udp://0.0.0.0";
/* try to allocate 2 UDP ports, the RTP port should be an even
* number and the RTCP port should be the next (uneven) port */
again:
udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
if (udpsrc0 == NULL)
goto no_udp_protocol;
g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, NULL);
ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
if (ret == GST_STATE_CHANGE_FAILURE) {
if (tmp_rtp != 0) {
tmp_rtp += 2;
if (++count > 20)
goto no_ports;
gst_element_set_state (udpsrc0, GST_STATE_NULL);
gst_object_unref (udpsrc0);
goto again;
}
goto no_udp_protocol;
}
g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
/* check if port is even */
if ((tmp_rtp & 1) != 0) {
/* port not even, close and allocate another */
if (++count > 20)
goto no_ports;
gst_element_set_state (udpsrc0, GST_STATE_NULL);
gst_object_unref (udpsrc0);
tmp_rtp++;
goto again;
}
/* allocate port+1 for RTCP now */
udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
if (udpsrc1 == NULL)
goto no_udp_rtcp_protocol;
/* set port */
tmp_rtcp = tmp_rtp + 1;
g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, NULL);
ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
/* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
if (ret == GST_STATE_CHANGE_FAILURE) {
if (++count > 20)
goto no_ports;
gst_element_set_state (udpsrc0, GST_STATE_NULL);
gst_object_unref (udpsrc0);
gst_element_set_state (udpsrc1, GST_STATE_NULL);
gst_object_unref (udpsrc1);
tmp_rtp += 2;
goto again;
}
/* all fine, do port check */
g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
/* this should not happen... */
if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
goto port_error;
udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
if (!udpsink0)
goto no_udp_protocol;
g_object_get (G_OBJECT (udpsrc0), "socket", &socket, NULL);
g_object_set (G_OBJECT (udpsink0), "socket", socket, NULL);
g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
if (!udpsink1)
goto no_udp_protocol;
if (g_object_class_find_property (G_OBJECT_GET_CLASS (udpsink0),
"send-duplicates")) {
g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
} else {
g_warning
("old multiudpsink version found without send-duplicates property");
}
if (g_object_class_find_property (G_OBJECT_GET_CLASS (udpsink0),
"buffer-size")) {
g_object_set (G_OBJECT (udpsink0), "buffer-size", stream->buffer_size,
NULL);
} else {
GST_WARNING ("multiudpsink version found without buffer-size property");
}
g_object_get (G_OBJECT (udpsrc1), "socket", &socket, NULL);
g_object_set (G_OBJECT (udpsink1), "socket", socket, NULL);
g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
/* we keep these elements, we will further configure them when the
* client told us to really use the UDP ports. */
stream->udpsrc[0] = udpsrc0;
stream->udpsrc[1] = udpsrc1;
stream->udpsink[0] = udpsink0;
stream->udpsink[1] = udpsink1;
stream->server_port.min = rtpport;
stream->server_port.max = rtcpport;
return TRUE;
/* ERRORS */
no_udp_protocol:
{
goto cleanup;
}
no_ports:
{
goto cleanup;
}
no_udp_rtcp_protocol:
{
goto cleanup;
}
port_error:
{
goto cleanup;
}
cleanup:
{
if (udpsrc0) {
gst_element_set_state (udpsrc0, GST_STATE_NULL);
gst_object_unref (udpsrc0);
}
if (udpsrc1) {
gst_element_set_state (udpsrc1, GST_STATE_NULL);
gst_object_unref (udpsrc1);
}
if (udpsink0) {
gst_element_set_state (udpsink0, GST_STATE_NULL);
gst_object_unref (udpsink0);
}
if (udpsink1) {
gst_element_set_state (udpsink1, GST_STATE_NULL);
gst_object_unref (udpsink1);
}
return FALSE;
}
}
/* executed from streaming thread */
static void
caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
{
GstCaps *newcaps, *oldcaps;
newcaps = gst_pad_get_current_caps (pad);
oldcaps = stream->caps;
stream->caps = newcaps;
if (oldcaps)
gst_caps_unref (oldcaps);
GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
newcaps);
}
static void
dump_structure (const GstStructure * s)
{
gchar *sstr;
sstr = gst_structure_to_string (s);
GST_INFO ("structure: %s", sstr);
g_free (sstr);
}
static GstRTSPStreamTransport *
find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
{
GList *walk;
GstRTSPStreamTransport *result = NULL;
const gchar *tmp;
gchar *dest;
guint port;
if (rtcp_from == NULL)
return NULL;
tmp = g_strrstr (rtcp_from, ":");
if (tmp == NULL)
return NULL;
port = atoi (tmp + 1);
dest = g_strndup (rtcp_from, tmp - rtcp_from);
GST_INFO ("finding %s:%d in %d transports", dest, port,
g_list_length (stream->transports));
for (walk = stream->transports; walk; walk = g_list_next (walk)) {
GstRTSPStreamTransport *trans = walk->data;
gint min, max;
min = trans->transport->client_port.min;
max = trans->transport->client_port.max;
if ((strcmp (trans->transport->destination, dest) == 0) && (min == port
|| max == port)) {
result = trans;
break;
}
}
g_free (dest);
return result;
}
static GstRTSPStreamTransport *
check_transport (GObject * source, GstRTSPStream * stream)
{
GstStructure *stats;
GstRTSPStreamTransport *trans;
/* see if we have a stream to match with the origin of the RTCP packet */
trans = g_object_get_qdata (source, ssrc_stream_map_key);
if (trans == NULL) {
g_object_get (source, "stats", &stats, NULL);
if (stats) {
const gchar *rtcp_from;
dump_structure (stats);
rtcp_from = gst_structure_get_string (stats, "rtcp-from");
if ((trans = find_transport (stream, rtcp_from))) {
GST_INFO ("%p: found transport %p for source %p", stream, trans,
source);
/* keep ref to the source */
trans->rtpsource = source;
g_object_set_qdata (source, ssrc_stream_map_key, trans);
}
gst_structure_free (stats);
}
}
return trans;
}
static void
on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
{
GstRTSPStreamTransport *trans;
GST_INFO ("%p: new source %p", stream, source);
trans = check_transport (source, stream);
if (trans)
GST_INFO ("%p: source %p for transport %p", stream, source, trans);
}
static void
on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
{
GST_INFO ("%p: new SDES %p", stream, source);
}
static void
on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
{
GstRTSPStreamTransport *trans;
trans = check_transport (source, stream);
if (trans)
GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
if (trans && trans->keep_alive)
trans->keep_alive (trans->ka_user_data);
#ifdef DUMP_STATS
{
GstStructure *stats;
g_object_get (source, "stats", &stats, NULL);
if (stats) {
dump_structure (stats);
gst_structure_free (stats);
}
}
#endif
}
static void
on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
{
GST_INFO ("%p: source %p bye", stream, source);
}
static void
on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
{
GstRTSPStreamTransport *trans;
GST_INFO ("%p: source %p bye timeout", stream, source);
if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
trans->rtpsource = NULL;
trans->timeout = TRUE;
}
}
static void
on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
{
GstRTSPStreamTransport *trans;
GST_INFO ("%p: source %p timeout", stream, source);
if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
trans->rtpsource = NULL;
trans->timeout = TRUE;
}
}
static GstFlowReturn
handle_new_sample (GstAppSink * sink, gpointer user_data)
{
GList *walk;
GstSample *sample;
GstBuffer *buffer;
GstRTSPStream *stream;
sample = gst_app_sink_pull_sample (sink);
if (!sample)
return GST_FLOW_OK;
stream = (GstRTSPStream *) user_data;
buffer = gst_sample_get_buffer (sample);
for (walk = stream->transports; walk; walk = g_list_next (walk)) {
GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
if (GST_ELEMENT_CAST (sink) == stream->appsink[0]) {
if (tr->send_rtp)
tr->send_rtp (buffer, tr->transport->interleaved.min, tr->user_data);
} else {
if (tr->send_rtcp)
tr->send_rtcp (buffer, tr->transport->interleaved.max, tr->user_data);
}
}
gst_sample_unref (sample);
return GST_FLOW_OK;
}
static GstAppSinkCallbacks sink_cb = {
NULL, /* not interested in EOS */
NULL, /* not interested in preroll samples */
handle_new_sample,
};
/**
* gst_rtsp_stream_join_bin:
* @stream: a #GstRTSPStream
* @bin: a #GstBin to join
* @rtpbin: a rtpbin element in @bin
* @state: the target state of the new elements
*
* Join the #Gstbin @bin that contains the element @rtpbin.
*
* @stream will link to @rtpbin, which must be inside @bin. The elements
* added to @bin will be set to the state given in @state.
*
* Returns: %TRUE on success.
*/
gboolean
gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
GstElement * rtpbin, GstState state)
{
gint i, idx;
gchar *name;
GstPad *pad, *teepad, *queuepad, *selpad;
GstPadLinkReturn ret;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
if (stream->is_joined)
return TRUE;
/* create a session with the same index as the stream */
idx = stream->idx;
GST_INFO ("stream %p joining bin as session %d", stream, idx);
if (!alloc_ports (stream))
goto no_ports;
/* get a pad for sending RTP */
name = g_strdup_printf ("send_rtp_sink_%u", idx);
stream->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
g_free (name);
/* link the RTP pad to the session manager, it should not really fail unless
* this is not really an RTP pad */
ret = gst_pad_link (stream->srcpad, stream->send_rtp_sink);
if (ret != GST_PAD_LINK_OK)
goto link_failed;
/* get pads from the RTP session element for sending and receiving
* RTP/RTCP*/
name = g_strdup_printf ("send_rtp_src_%u", idx);
stream->send_src[0] = gst_element_get_static_pad (rtpbin, name);
g_free (name);
name = g_strdup_printf ("send_rtcp_src_%u", idx);
stream->send_src[1] = gst_element_get_request_pad (rtpbin, name);
g_free (name);
name = g_strdup_printf ("recv_rtp_sink_%u", idx);
stream->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
g_free (name);
name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
stream->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
g_free (name);
/* get the session */
g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &stream->session);
g_signal_connect (stream->session, "on-new-ssrc", (GCallback) on_new_ssrc,
stream);
g_signal_connect (stream->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
stream);
g_signal_connect (stream->session, "on-ssrc-active",
(GCallback) on_ssrc_active, stream);
g_signal_connect (stream->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
stream);
g_signal_connect (stream->session, "on-bye-timeout",
(GCallback) on_bye_timeout, stream);
g_signal_connect (stream->session, "on-timeout", (GCallback) on_timeout,
stream);
for (i = 0; i < 2; i++) {
/* For the sender we create this bit of pipeline for both
* RTP and RTCP. Sync and preroll are enabled on udpsink so
* we need to add a queue before appsink to make the pipeline
* not block. For the TCP case, we want to pump data to the
* client as fast as possible anyway.
*
* .--------. .-----. .---------.
* | rtpbin | | tee | | udpsink |
* | send->sink src->sink |
* '--------' | | '---------'
* | | .---------. .---------.
* | | | queue | | appsink |
* | src->sink src->sink |
* '-----' '---------' '---------'
*/
/* make tee for RTP/RTCP */
stream->tee[i] = gst_element_factory_make ("tee", NULL);
gst_bin_add (bin, stream->tee[i]);
/* and link to rtpbin send pad */
pad = gst_element_get_static_pad (stream->tee[i], "sink");
gst_pad_link (stream->send_src[i], pad);
gst_object_unref (pad);
/* add udpsink */
gst_bin_add (bin, stream->udpsink[i]);
/* link tee to udpsink */
teepad = gst_element_get_request_pad (stream->tee[i], "src_%u");
pad = gst_element_get_static_pad (stream->udpsink[i], "sink");
gst_pad_link (teepad, pad);
gst_object_unref (pad);
gst_object_unref (teepad);
/* make queue */
stream->appqueue[i] = gst_element_factory_make ("queue", NULL);
gst_bin_add (bin, stream->appqueue[i]);
/* and link to tee */
teepad = gst_element_get_request_pad (stream->tee[i], "src_%u");
pad = gst_element_get_static_pad (stream->appqueue[i], "sink");
gst_pad_link (teepad, pad);
gst_object_unref (pad);
gst_object_unref (teepad);
/* make appsink */
stream->appsink[i] = gst_element_factory_make ("appsink", NULL);
g_object_set (stream->appsink[i], "async", FALSE, "sync", FALSE, NULL);
g_object_set (stream->appsink[i], "emit-signals", FALSE, NULL);
gst_bin_add (bin, stream->appsink[i]);
gst_app_sink_set_callbacks (GST_APP_SINK_CAST (stream->appsink[i]),
&sink_cb, stream, NULL);
/* and link to queue */
queuepad = gst_element_get_static_pad (stream->appqueue[i], "src");
pad = gst_element_get_static_pad (stream->appsink[i], "sink");
gst_pad_link (queuepad, pad);
gst_object_unref (pad);
gst_object_unref (queuepad);
/* For the receiver we create this bit of pipeline for both
* RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
* and it is all funneled into the rtpbin receive pad.
*
* .--------. .--------. .--------.
* | udpsrc | | funnel | | rtpbin |
* | src->sink src->sink |
* '--------' | | '--------'
* .--------. | |
* | appsrc | | |
* | src->sink |
* '--------' '--------'
*/
/* make funnel for the RTP/RTCP receivers */
stream->funnel[i] = gst_element_factory_make ("funnel", NULL);
gst_bin_add (bin, stream->funnel[i]);
pad = gst_element_get_static_pad (stream->funnel[i], "src");
gst_pad_link (pad, stream->recv_sink[i]);
gst_object_unref (pad);
/* add udpsrc */
gst_bin_add (bin, stream->udpsrc[i]);
/* and link to the funnel */
selpad = gst_element_get_request_pad (stream->funnel[i], "sink_%u");
pad = gst_element_get_static_pad (stream->udpsrc[i], "src");
gst_pad_link (pad, selpad);
gst_object_unref (pad);
gst_object_unref (selpad);
/* make and add appsrc */
stream->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
gst_bin_add (bin, stream->appsrc[i]);
/* and link to the funnel */
selpad = gst_element_get_request_pad (stream->funnel[i], "sink_%u");
pad = gst_element_get_static_pad (stream->appsrc[i], "src");
gst_pad_link (pad, selpad);
gst_object_unref (pad);
gst_object_unref (selpad);
/* check if we need to set to a special state */
if (state != GST_STATE_NULL) {
gst_element_set_state (stream->udpsink[i], state);
gst_element_set_state (stream->appsink[i], state);
gst_element_set_state (stream->appqueue[i], state);
gst_element_set_state (stream->tee[i], state);
gst_element_set_state (stream->funnel[i], state);
gst_element_set_state (stream->appsrc[i], state);
}
/* we set and keep these to playing so that they don't cause NO_PREROLL return
* values */
gst_element_set_state (stream->udpsrc[i], GST_STATE_PLAYING);
gst_element_set_locked_state (stream->udpsrc[i], TRUE);
}
/* be notified of caps changes */
stream->caps_sig = g_signal_connect (stream->send_rtp_sink, "notify::caps",
(GCallback) caps_notify, stream);
stream->is_joined = TRUE;
return TRUE;
/* ERRORS */
no_ports:
{
GST_WARNING ("failed to allocate ports %d", idx);
return FALSE;
}
link_failed:
{
GST_WARNING ("failed to link stream %d", idx);
gst_object_unref (stream->send_rtp_sink);
stream->send_rtp_sink = NULL;
return FALSE;
}
}
/**
* gst_rtsp_stream_leave_bin:
* @stream: a #GstRTSPStream
* @bin: a #GstBin
* @rtpbin: a rtpbin #GstElement
*
* Remove the elements of @stream from @bin. @bin must be set
* to the NULL state before calling this.
*
* Return: %TRUE on success.
*/
gboolean
gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
GstElement * rtpbin)
{
gint i;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
if (!stream->is_joined)
return TRUE;
/* all transports must be removed by now */
g_return_val_if_fail (stream->transports == NULL, FALSE);
GST_INFO ("stream %p leaving bin", stream);
gst_pad_unlink (stream->srcpad, stream->send_rtp_sink);
g_signal_handler_disconnect (stream->send_rtp_sink, stream->caps_sig);
gst_element_release_request_pad (rtpbin, stream->send_rtp_sink);
gst_object_unref (stream->send_rtp_sink);
stream->send_rtp_sink = NULL;
for (i = 0; i < 2; i++) {
/* and set udpsrc to NULL now before removing */
gst_element_set_locked_state (stream->udpsrc[i], FALSE);
gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
/* removing them should also nicely release the request
* pads when they finalize */
gst_bin_remove (bin, stream->udpsrc[i]);
gst_bin_remove (bin, stream->udpsink[i]);
gst_bin_remove (bin, stream->appsrc[i]);
gst_bin_remove (bin, stream->appsink[i]);
gst_bin_remove (bin, stream->appqueue[i]);
gst_bin_remove (bin, stream->tee[i]);
gst_bin_remove (bin, stream->funnel[i]);
gst_element_release_request_pad (rtpbin, stream->recv_sink[i]);
gst_object_unref (stream->recv_sink[i]);
stream->recv_sink[i] = NULL;
stream->udpsrc[i] = NULL;
stream->udpsink[i] = NULL;
stream->appsrc[i] = NULL;
stream->appsink[i] = NULL;
stream->appqueue[i] = NULL;
stream->tee[i] = NULL;
stream->funnel[i] = NULL;
}
gst_object_unref (stream->send_src[0]);
stream->send_src[0] = NULL;
gst_element_release_request_pad (rtpbin, stream->send_src[1]);
gst_object_unref (stream->send_src[1]);
stream->send_src[1] = NULL;
g_object_unref (stream->session);
if (stream->caps)
gst_caps_unref (stream->caps);
stream->is_joined = FALSE;
return TRUE;
}
/**
* gst_rtsp_stream_get_rtpinfo:
* @stream: a #GstRTSPStream
* @rtptime: result RTP timestamp
* @seq: result RTP seqnum
*
* Retrieve the current rtptime and seq. This is used to
* construct a RTPInfo reply header.
*
* Returns: %TRUE when rtptime and seq could be determined.
*/
gboolean
gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
guint * rtptime, guint * seq)
{
GObjectClass *payobjclass;
payobjclass = G_OBJECT_GET_CLASS (stream->payloader);
if (!g_object_class_find_property (payobjclass, "seqnum") ||
!g_object_class_find_property (payobjclass, "timestamp"))
return FALSE;
g_object_get (stream->payloader, "seqnum", seq, "timestamp", rtptime, NULL);
return TRUE;
}
/**
* gst_rtsp_stream_recv_rtp:
* @stream: a #GstRTSPStream
* @buffer: (transfer full): a #GstBuffer
*
* Handle an RTP buffer for the stream. This method is usually called when a
* message has been received from a client using the TCP transport.
*
* This function takes ownership of @buffer.
*
* Returns: a GstFlowReturn.
*/
GstFlowReturn
gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
{
GstFlowReturn ret;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
g_return_val_if_fail (stream->is_joined, FALSE);
ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (stream->appsrc[0]), buffer);
return ret;
}
/**
* gst_rtsp_stream_recv_rtcp:
* @stream: a #GstRTSPStream
* @buffer: (transfer full): a #GstBuffer
*
* Handle an RTCP buffer for the stream. This method is usually called when a
* message has been received from a client using the TCP transport.
*
* This function takes ownership of @buffer.
*
* Returns: a GstFlowReturn.
*/
GstFlowReturn
gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
{
GstFlowReturn ret;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
g_return_val_if_fail (stream->is_joined, FALSE);
ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (stream->appsrc[1]), buffer);
return ret;
}
static gboolean
update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
gboolean add)
{
GstRTSPTransport *tr;
gboolean updated;
updated = FALSE;
tr = trans->transport;
switch (tr->lower_transport) {
case GST_RTSP_LOWER_TRANS_UDP:
case GST_RTSP_LOWER_TRANS_UDP_MCAST:
{
gchar *dest;
gint min, max;
guint ttl = 0;
dest = tr->destination;
if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
min = tr->port.min;
max = tr->port.max;
ttl = tr->ttl;
} else {
min = tr->client_port.min;
max = tr->client_port.max;
}
if (add && !trans->active) {
GST_INFO ("adding %s:%d-%d", dest, min, max);
g_signal_emit_by_name (stream->udpsink[0], "add", dest, min, NULL);
g_signal_emit_by_name (stream->udpsink[1], "add", dest, max, NULL);
if (ttl > 0) {
GST_INFO ("setting ttl-mc %d", ttl);
g_object_set (G_OBJECT (stream->udpsink[0]), "ttl-mc", ttl, NULL);
g_object_set (G_OBJECT (stream->udpsink[1]), "ttl-mc", ttl, NULL);
}
stream->transports = g_list_prepend (stream->transports, trans);
trans->active = TRUE;
updated = TRUE;
} else if (trans->active) {
GST_INFO ("removing %s:%d-%d", dest, min, max);
g_signal_emit_by_name (stream->udpsink[0], "remove", dest, min, NULL);
g_signal_emit_by_name (stream->udpsink[1], "remove", dest, max, NULL);
stream->transports = g_list_remove (stream->transports, trans);
trans->active = FALSE;
updated = TRUE;
}
break;
}
case GST_RTSP_LOWER_TRANS_TCP:
if (add && !trans->active) {
GST_INFO ("adding TCP %s", tr->destination);
stream->transports = g_list_prepend (stream->transports, trans);
trans->active = TRUE;
updated = TRUE;
} else if (trans->active) {
GST_INFO ("removing TCP %s", tr->destination);
stream->transports = g_list_remove (stream->transports, trans);
trans->active = FALSE;
updated = TRUE;
}
break;
default:
GST_INFO ("Unknown transport %d", tr->lower_transport);
break;
}
return updated;
}
/**
* gst_rtsp_stream_add_transport:
* @stream: a #GstRTSPStream
* @trans: a #GstRTSPStreamTransport
*
* Add the transport in @trans to @stream. The media of @stream will
* then also be send to the values configured in @trans.
*
* @stream must be joined to a bin.
*
* @trans must contain a valid #GstRTSPTransport.
*
* Returns: %TRUE if @trans was added
*/
gboolean
gst_rtsp_stream_add_transport (GstRTSPStream * stream,
GstRTSPStreamTransport * trans)
{
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
g_return_val_if_fail (stream->is_joined, FALSE);
g_return_val_if_fail (trans->transport != NULL, FALSE);
return update_transport (stream, trans, TRUE);
}
/**
* gst_rtsp_stream_remove_transport:
* @stream: a #GstRTSPStream
* @trans: a #GstRTSPStreamTransport
*
* Remove the transport in @trans from @stream. The media of @stream will
* not be sent to the values configured in @trans.
*
* @stream must be joined to a bin.
*
* @trans must contain a valid #GstRTSPTransport.
*
* Returns: %TRUE if @trans was removed
*/
gboolean
gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
GstRTSPStreamTransport * trans)
{
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
g_return_val_if_fail (stream->is_joined, FALSE);
g_return_val_if_fail (trans->transport != NULL, FALSE);
return update_transport (stream, trans, FALSE);
}
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