gstreamer/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-onvif-media.c

Ignoring revisions in .git-blame-ignore-revs. Click here to bypass and see the normal blame view.

359 lines
10 KiB
C
Raw Normal View History

/* GStreamer
* Copyright (C) 2017 Sebastian Dröge <sebastian@centricular.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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:rtsp-onvif-media
* @short_description: The ONVIF media pipeline
* @see_also: #GstRTSPMedia, #GstRTSPOnvifMediaFactory, #GstRTSPStream, #GstRTSPSession,
* #GstRTSPSessionMedia
*
* a #GstRTSPOnvifMedia contains the complete GStreamer pipeline to manage the
* streaming to the clients. The actual data transfer is done by the
* #GstRTSPStream objects that are created and exposed by the #GstRTSPMedia.
*
* On top of #GstRTSPMedia this subclass adds special ONVIF features.
* Special ONVIF features that are currently supported is a backchannel for
* the client to send back media to the server in a normal PLAY media. To
* handle the ONVIF backchannel, a #GstRTSPOnvifMediaFactory and
* #GstRTSPOnvifServer has to be used.
*
* Since: 1.14
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "rtsp-onvif-media.h"
#include "rtsp-latency-bin.h"
struct GstRTSPOnvifMediaPrivate
{
GMutex lock;
guint backchannel_bandwidth;
};
G_DEFINE_TYPE_WITH_PRIVATE (GstRTSPOnvifMedia, gst_rtsp_onvif_media,
GST_TYPE_RTSP_MEDIA);
static gboolean
gst_rtsp_onvif_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
GstSDPInfo * info)
{
guint i, n_streams;
gchar *rangestr;
gboolean res;
/* Mostly a copy of gst_rtsp_sdp_from_media() which handles the backchannel
* stream separately and adds sendonly/recvonly attributes to each media
*/
n_streams = gst_rtsp_media_n_streams (media);
rangestr = gst_rtsp_media_get_range_string (media, FALSE, GST_RTSP_RANGE_NPT);
if (rangestr == NULL)
goto not_prepared;
gst_sdp_message_add_attribute (sdp, "range", rangestr);
g_free (rangestr);
res = TRUE;
for (i = 0; res && (i < n_streams); i++) {
GstRTSPStream *stream;
GstCaps *caps = NULL;
GstRTSPProfile profiles;
guint mask;
GstPad *sinkpad = NULL;
guint n_caps, j;
/* Mostly a copy of gst_rtsp_sdp_from_stream() which handles the
* backchannel stream separately */
stream = gst_rtsp_media_get_stream (media, i);
if ((sinkpad = gst_rtsp_stream_get_sinkpad (stream))) {
caps = gst_pad_query_caps (sinkpad, NULL);
} else {
caps = gst_rtsp_stream_get_caps (stream);
}
if (caps == NULL) {
GST_ERROR ("stream %p has no caps", stream);
res = FALSE;
if (sinkpad)
gst_object_unref (sinkpad);
break;
} else if (!sinkpad && !gst_caps_is_fixed (caps)) {
GST_ERROR ("stream %p has unfixed caps", stream);
res = FALSE;
gst_caps_unref (caps);
break;
}
n_caps = gst_caps_get_size (caps);
for (j = 0; res && j < n_caps; j++) {
GstStructure *s = gst_caps_get_structure (caps, j);
GstCaps *media_caps = gst_caps_new_full (gst_structure_copy (s), NULL);
if (!gst_caps_is_fixed (media_caps)) {
GST_ERROR ("media caps for stream %p are not all fixed", stream);
res = FALSE;
gst_caps_unref (media_caps);
break;
}
/* make a new media for each profile */
profiles = gst_rtsp_stream_get_profiles (stream);
mask = 1;
res = TRUE;
while (res && (profiles >= mask)) {
GstRTSPProfile prof = profiles & mask;
if (prof) {
res = gst_rtsp_sdp_make_media (sdp, info, stream, media_caps, prof);
if (res) {
GstSDPMedia *smedia =
&g_array_index (sdp->medias, GstSDPMedia, sdp->medias->len - 1);
gchar *x_onvif_track, *media_str;
media_str =
g_ascii_strup (gst_structure_get_string (s, "media"), -1);
x_onvif_track =
g_strdup_printf ("%s%03d", media_str, sdp->medias->len - 1);
gst_sdp_media_add_attribute (smedia, "x-onvif-track",
x_onvif_track);
g_free (x_onvif_track);
g_free (media_str);
if (sinkpad) {
GstRTSPOnvifMedia *onvif_media = GST_RTSP_ONVIF_MEDIA (media);
gst_sdp_media_add_attribute (smedia, "sendonly", "");
if (onvif_media->priv->backchannel_bandwidth > 0)
gst_sdp_media_add_bandwidth (smedia, GST_SDP_BWTYPE_AS,
onvif_media->priv->backchannel_bandwidth);
} else {
gst_sdp_media_add_attribute (smedia, "recvonly", "");
}
}
}
mask <<= 1;
}
if (sinkpad) {
GstStructure *s = gst_caps_get_structure (media_caps, 0);
gint pt = -1;
if (!gst_structure_get_int (s, "payload", &pt) || pt < 0) {
GST_ERROR ("stream %p has no payload type", stream);
res = FALSE;
gst_caps_unref (media_caps);
gst_object_unref (sinkpad);
break;
}
gst_rtsp_stream_set_pt_map (stream, pt, media_caps);
}
gst_caps_unref (media_caps);
}
gst_caps_unref (caps);
if (sinkpad)
gst_object_unref (sinkpad);
}
{
GstNetTimeProvider *provider;
if ((provider =
gst_rtsp_media_get_time_provider (media, info->server_ip, 0))) {
GstClock *clock;
gchar *address, *str;
gint port;
g_object_get (provider, "clock", &clock, "address", &address, "port",
&port, NULL);
str = g_strdup_printf ("GstNetTimeProvider %s %s:%d %" G_GUINT64_FORMAT,
g_type_name (G_TYPE_FROM_INSTANCE (clock)), address, port,
gst_clock_get_time (clock));
gst_sdp_message_add_attribute (sdp, "x-gst-clock", str);
g_free (str);
gst_object_unref (clock);
g_free (address);
gst_object_unref (provider);
}
}
return res;
/* ERRORS */
not_prepared:
{
GST_ERROR ("media %p is not prepared", media);
return FALSE;
}
}
static void
gst_rtsp_onvif_media_finalize (GObject * object)
{
GstRTSPOnvifMedia *media = GST_RTSP_ONVIF_MEDIA (object);
g_mutex_clear (&media->priv->lock);
G_OBJECT_CLASS (gst_rtsp_onvif_media_parent_class)->finalize (object);
}
static void
gst_rtsp_onvif_media_class_init (GstRTSPOnvifMediaClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstRTSPMediaClass *media_class = (GstRTSPMediaClass *) klass;
gobject_class->finalize = gst_rtsp_onvif_media_finalize;
media_class->setup_sdp = gst_rtsp_onvif_media_setup_sdp;
}
static void
gst_rtsp_onvif_media_init (GstRTSPOnvifMedia * media)
{
media->priv = gst_rtsp_onvif_media_get_instance_private (media);
g_mutex_init (&media->priv->lock);
}
/**
* gst_rtsp_onvif_media_collect_backchannel:
* @media: a #GstRTSPOnvifMedia
*
* Find the ONVIF backchannel depayloader element. It should be named
* 'depay_backchannel', be placed in a bin called 'onvif-backchannel'
* and return all supported RTP caps on a caps query. Complete RTP caps with
* at least the payload type, clock-rate and encoding-name are required.
*
* A new #GstRTSPStream is created for the backchannel if found.
*
* Returns: %TRUE if a backchannel stream could be found and created
*
* Since: 1.14
*/
gboolean
gst_rtsp_onvif_media_collect_backchannel (GstRTSPOnvifMedia * media)
{
GstElement *element, *backchannel_bin = NULL;
GstElement *latency_bin;
GstPad *pad = NULL;
gboolean ret = FALSE;
g_return_val_if_fail (GST_IS_RTSP_ONVIF_MEDIA (media), FALSE);
element = gst_rtsp_media_get_element (GST_RTSP_MEDIA (media));
if (!element)
return ret;
backchannel_bin =
gst_bin_get_by_name (GST_BIN (element), "onvif-backchannel");
if (!backchannel_bin)
goto out;
/* We don't want the backchannel element, which is a receiver, to affect
* latency on the complete pipeline. That's why we remove it from the
* pipeline and add it to a @GstRTSPLatencyBin which will prevent it from
* messing up pipelines latency. The extra reference is needed so that it
* is not freed in case the pipeline holds the the only ref to it.
*
* TODO: a more generic solution should be implemented in
* gst_rtsp_media_collect_streams() where all receivers are encapsulated
* in a @GstRTSPLatencyBin in cases when there are senders too. */
gst_object_ref (backchannel_bin);
gst_bin_remove (GST_BIN (element), backchannel_bin);
latency_bin = gst_rtsp_latency_bin_new (backchannel_bin);
g_assert (latency_bin);
gst_bin_add (GST_BIN (element), latency_bin);
pad = gst_element_get_static_pad (latency_bin, "sink");
if (!pad)
goto out;
gst_rtsp_media_create_stream (GST_RTSP_MEDIA (media), latency_bin, pad);
ret = TRUE;
out:
if (pad)
gst_object_unref (pad);
if (backchannel_bin)
gst_object_unref (backchannel_bin);
gst_object_unref (element);
return ret;
}
/**
* gst_rtsp_onvif_media_set_backchannel_bandwidth:
* @media: a #GstRTSPMedia
* @bandwidth: the bandwidth in bits per second
*
* Set the configured/supported bandwidth of the ONVIF backchannel pipeline in
* bits per second.
*
* Since: 1.14
*/
void
gst_rtsp_onvif_media_set_backchannel_bandwidth (GstRTSPOnvifMedia * media,
guint bandwidth)
{
g_return_if_fail (GST_IS_RTSP_ONVIF_MEDIA (media));
g_mutex_lock (&media->priv->lock);
media->priv->backchannel_bandwidth = bandwidth;
g_mutex_unlock (&media->priv->lock);
}
/**
* gst_rtsp_onvif_media_get_backchannel_bandwidth:
* @media: a #GstRTSPMedia
*
* Get the configured/supported bandwidth of the ONVIF backchannel pipeline in
* bits per second.
*
* Returns: the configured/supported backchannel bandwidth.
*
* Since: 1.14
*/
guint
gst_rtsp_onvif_media_get_backchannel_bandwidth (GstRTSPOnvifMedia * media)
{
guint bandwidth;
g_return_val_if_fail (GST_IS_RTSP_ONVIF_MEDIA (media), 0);
g_mutex_lock (&media->priv->lock);
bandwidth = media->priv->backchannel_bandwidth;
g_mutex_unlock (&media->priv->lock);
return bandwidth;
}