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

567 lines
15 KiB
C

/* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#define GLIB_DISABLE_DEPRECATION_WARNINGS
/**
* SECTION:rtsp-sdp
* @short_description: Make SDP messages
* @see_also: #GstRTSPMedia
*
* Last reviewed on 2013-07-11 (1.0.0)
*/
#include <string.h>
#include <gst/net/net.h>
#include <gst/sdp/gstmikey.h>
#include "rtsp-sdp.h"
static gboolean
get_info_from_tags (GstPad * pad, GstEvent ** event, gpointer user_data)
{
GstSDPMedia *media = (GstSDPMedia *) user_data;
if (GST_EVENT_TYPE (*event) == GST_EVENT_TAG) {
GstTagList *tags;
guint bitrate = 0;
gst_event_parse_tag (*event, &tags);
if (gst_tag_list_get_scope (tags) != GST_TAG_SCOPE_STREAM)
return TRUE;
if (!gst_tag_list_get_uint (tags, GST_TAG_MAXIMUM_BITRATE,
&bitrate) || bitrate == 0)
if (!gst_tag_list_get_uint (tags, GST_TAG_BITRATE, &bitrate) ||
bitrate == 0)
return TRUE;
/* set bandwidth (kbits/s) */
gst_sdp_media_add_bandwidth (media, GST_SDP_BWTYPE_AS, bitrate / 1000);
return FALSE;
}
return TRUE;
}
static void
update_sdp_from_tags (GstRTSPStream * stream, GstSDPMedia * stream_media)
{
GstPad *src_pad;
src_pad = gst_rtsp_stream_get_srcpad (stream);
gst_pad_sticky_events_foreach (src_pad, get_info_from_tags, stream_media);
gst_object_unref (src_pad);
}
static guint
get_roc_from_stats (GstStructure * stats, guint ssrc)
{
const GValue *va, *v;
guint i, len;
/* initialize roc to something different than 0, so if we don't get
the proper ROC from the encoder, streaming should fail initially. */
guint roc = -1;
va = gst_structure_get_value (stats, "streams");
if (!va || !G_VALUE_HOLDS (va, GST_TYPE_ARRAY)) {
GST_WARNING ("stats doesn't have a valid 'streams' field");
return 0;
}
len = gst_value_array_get_size (va);
/* look if there's any SSRC that matches. */
for (i = 0; i < len; i++) {
GstStructure *stream;
v = gst_value_array_get_value (va, i);
if (v && (stream = g_value_get_boxed (v))) {
guint stream_ssrc;
gst_structure_get_uint (stream, "ssrc", &stream_ssrc);
if (stream_ssrc == ssrc) {
gst_structure_get_uint (stream, "roc", &roc);
break;
}
}
}
return roc;
}
static gboolean
mikey_add_crypto_sessions (GstRTSPStream * stream, GstMIKEYMessage * msg)
{
guint i;
GObject *session;
GstElement *encoder;
GValueArray *sources;
gboolean roc_found;
encoder = gst_rtsp_stream_get_srtp_encoder (stream);
if (encoder == NULL) {
GST_ERROR ("unable to get SRTP encoder from stream %p", stream);
return FALSE;
}
session = gst_rtsp_stream_get_rtpsession (stream);
if (session == NULL) {
GST_ERROR ("unable to get RTP session from stream %p", stream);
gst_object_unref (encoder);
return FALSE;
}
roc_found = FALSE;
g_object_get (session, "sources", &sources, NULL);
for (i = 0; sources && (i < sources->n_values); i++) {
GValue *val;
GObject *source;
guint32 ssrc;
gboolean is_sender;
val = g_value_array_get_nth (sources, i);
source = (GObject *) g_value_get_object (val);
g_object_get (source, "ssrc", &ssrc, "is-sender", &is_sender, NULL);
if (is_sender) {
guint32 roc = -1;
GstStructure *stats;
g_object_get (encoder, "stats", &stats, NULL);
if (stats) {
roc = get_roc_from_stats (stats, ssrc);
gst_structure_free (stats);
}
roc_found = ! !(roc != -1);
if (!roc_found) {
GST_ERROR ("unable to obtain ROC for stream %p with SSRC %u",
stream, ssrc);
goto cleanup;
}
GST_INFO ("stream %p with SSRC %u has a ROC of %u", stream, ssrc, roc);
gst_mikey_message_add_cs_srtp (msg, 0, ssrc, roc);
}
}
cleanup:
{
g_value_array_free (sources);
gst_object_unref (encoder);
g_object_unref (session);
return roc_found;
}
}
static gboolean
make_media (GstSDPMessage * sdp, GstSDPInfo * info,
GstRTSPStream * stream, GstCaps * caps, GstRTSPProfile profile)
{
GstSDPMedia *smedia;
gchar *tmp;
GstRTSPLowerTrans ltrans;
GSocketFamily family;
const gchar *addrtype, *proto;
gchar *address;
guint ttl;
GstClockTime rtx_time;
gchar *base64;
GstMIKEYMessage *mikey_msg;
gst_sdp_media_new (&smedia);
if (gst_sdp_media_set_media_from_caps (caps, smedia) != GST_SDP_OK) {
goto caps_error;
}
gst_sdp_media_set_port_info (smedia, 0, 1);
switch (profile) {
case GST_RTSP_PROFILE_AVP:
proto = "RTP/AVP";
break;
case GST_RTSP_PROFILE_AVPF:
proto = "RTP/AVPF";
break;
case GST_RTSP_PROFILE_SAVP:
proto = "RTP/SAVP";
break;
case GST_RTSP_PROFILE_SAVPF:
proto = "RTP/SAVPF";
break;
default:
proto = "udp";
break;
}
gst_sdp_media_set_proto (smedia, proto);
if (info->is_ipv6) {
addrtype = "IP6";
family = G_SOCKET_FAMILY_IPV6;
} else {
addrtype = "IP4";
family = G_SOCKET_FAMILY_IPV4;
}
ltrans = gst_rtsp_stream_get_protocols (stream);
if (ltrans == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
GstRTSPAddress *addr;
addr = gst_rtsp_stream_get_multicast_address (stream, family);
if (addr == NULL)
goto no_multicast;
address = g_strdup (addr->address);
ttl = addr->ttl;
gst_rtsp_address_free (addr);
} else {
ttl = 16;
if (info->is_ipv6)
address = g_strdup ("::");
else
address = g_strdup ("0.0.0.0");
}
/* for the c= line */
gst_sdp_media_add_connection (smedia, "IN", addrtype, address, ttl, 1);
g_free (address);
/* the config uri */
tmp = gst_rtsp_stream_get_control (stream);
gst_sdp_media_add_attribute (smedia, "control", tmp);
g_free (tmp);
/* check for srtp */
mikey_msg = gst_mikey_message_new_from_caps (caps);
if (mikey_msg) {
/* add policy '0' for all sending SSRC */
if (!mikey_add_crypto_sessions (stream, mikey_msg))
goto crypto_sessions_error;
base64 = gst_mikey_message_base64_encode (mikey_msg);
if (base64) {
tmp = g_strdup_printf ("mikey %s", base64);
g_free (base64);
gst_sdp_media_add_attribute (smedia, "key-mgmt", tmp);
g_free (tmp);
}
gst_mikey_message_unref (mikey_msg);
}
/* RFC 7273 clock signalling */
{
GstBin *joined_bin = gst_rtsp_stream_get_joined_bin (stream);
GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (joined_bin));
gchar *ts_refclk = NULL;
gchar *mediaclk = NULL;
guint rtptime, clock_rate;
GstClockTime running_time, base_time, clock_time;
GstRTSPPublishClockMode publish_clock_mode =
gst_rtsp_stream_get_publish_clock_mode (stream);
gst_rtsp_stream_get_rtpinfo (stream, &rtptime, NULL, &clock_rate,
&running_time);
base_time = gst_element_get_base_time (GST_ELEMENT_CAST (joined_bin));
g_assert (base_time != GST_CLOCK_TIME_NONE);
clock_time = running_time + base_time;
if (publish_clock_mode != GST_RTSP_PUBLISH_CLOCK_MODE_NONE && clock) {
if (GST_IS_NTP_CLOCK (clock) || GST_IS_PTP_CLOCK (clock)) {
if (publish_clock_mode == GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET) {
guint32 mediaclk_offset;
/* Calculate RTP time at the clock's epoch. That's the direct offset */
clock_time =
gst_util_uint64_scale (clock_time, clock_rate, GST_SECOND);
clock_time &= 0xffffffff;
mediaclk_offset =
G_GUINT64_CONSTANT (0xffffffff) + rtptime - clock_time;
mediaclk = g_strdup_printf ("direct=%u", (guint32) mediaclk_offset);
}
if (GST_IS_NTP_CLOCK (clock)) {
gchar *ntp_address;
guint ntp_port;
g_object_get (clock, "address", &ntp_address, "port", &ntp_port,
NULL);
if (ntp_port == 123)
ts_refclk = g_strdup_printf ("ntp=%s", ntp_address);
else
ts_refclk = g_strdup_printf ("ntp=%s:%u", ntp_address, ntp_port);
g_free (ntp_address);
} else {
guint64 ptp_clock_id;
guint ptp_domain;
g_object_get (clock, "grandmaster-clock-id", &ptp_clock_id, "domain",
&ptp_domain, NULL);
if (ptp_domain != 0)
ts_refclk =
g_strdup_printf
("ptp=IEEE1588-2008:%02X-%02X-%02X-%02X-%02X-%02X-%02X-%02X:%u",
(guint) (ptp_clock_id >> 56) & 0xff,
(guint) (ptp_clock_id >> 48) & 0xff,
(guint) (ptp_clock_id >> 40) & 0xff,
(guint) (ptp_clock_id >> 32) & 0xff,
(guint) (ptp_clock_id >> 24) & 0xff,
(guint) (ptp_clock_id >> 16) & 0xff,
(guint) (ptp_clock_id >> 8) & 0xff,
(guint) (ptp_clock_id >> 0) & 0xff, ptp_domain);
else
ts_refclk =
g_strdup_printf
("ptp=IEEE1588-2008:%02X-%02X-%02X-%02X-%02X-%02X-%02X-%02X",
(guint) (ptp_clock_id >> 56) & 0xff,
(guint) (ptp_clock_id >> 48) & 0xff,
(guint) (ptp_clock_id >> 40) & 0xff,
(guint) (ptp_clock_id >> 32) & 0xff,
(guint) (ptp_clock_id >> 24) & 0xff,
(guint) (ptp_clock_id >> 16) & 0xff,
(guint) (ptp_clock_id >> 8) & 0xff,
(guint) (ptp_clock_id >> 0) & 0xff);
}
}
}
if (clock)
gst_object_unref (clock);
if (!ts_refclk)
ts_refclk = g_strdup ("local");
if (!mediaclk)
mediaclk = g_strdup ("sender");
gst_sdp_media_add_attribute (smedia, "ts-refclk", ts_refclk);
gst_sdp_media_add_attribute (smedia, "mediaclk", mediaclk);
g_free (ts_refclk);
g_free (mediaclk);
gst_object_unref (joined_bin);
}
update_sdp_from_tags (stream, smedia);
if ((profile == GST_RTSP_PROFILE_AVPF || profile == GST_RTSP_PROFILE_SAVPF)
&& (rtx_time = gst_rtsp_stream_get_retransmission_time (stream))) {
/* ssrc multiplexed retransmit functionality */
guint rtx_pt = gst_rtsp_stream_get_retransmission_pt (stream);
if (rtx_pt == 0) {
g_warning ("failed to find an available dynamic payload type. "
"Not adding retransmission");
} else {
gchar *tmp;
GstStructure *s;
gint caps_pt, caps_rate;
s = gst_caps_get_structure (caps, 0);
if (s == NULL)
goto no_caps_info;
/* get payload type and clock rate */
gst_structure_get_int (s, "payload", &caps_pt);
gst_structure_get_int (s, "clock-rate", &caps_rate);
tmp = g_strdup_printf ("%d", rtx_pt);
gst_sdp_media_add_format (smedia, tmp);
g_free (tmp);
tmp = g_strdup_printf ("%d rtx/%d", rtx_pt, caps_rate);
gst_sdp_media_add_attribute (smedia, "rtpmap", tmp);
g_free (tmp);
tmp =
g_strdup_printf ("%d apt=%d;rtx-time=%" G_GINT64_FORMAT, rtx_pt,
caps_pt, GST_TIME_AS_MSECONDS (rtx_time));
gst_sdp_media_add_attribute (smedia, "fmtp", tmp);
g_free (tmp);
}
}
gst_sdp_message_add_media (sdp, smedia);
gst_sdp_media_free (smedia);
return TRUE;
/* ERRORS */
caps_error:
{
gst_sdp_media_free (smedia);
GST_ERROR ("unable to set media from caps for stream %d",
gst_rtsp_stream_get_index (stream));
return FALSE;
}
no_multicast:
{
gst_sdp_media_free (smedia);
GST_ERROR ("stream %d has no multicast address",
gst_rtsp_stream_get_index (stream));
return FALSE;
}
no_caps_info:
{
gst_sdp_media_free (smedia);
GST_ERROR ("caps for stream %d have no structure",
gst_rtsp_stream_get_index (stream));
return FALSE;
}
crypto_sessions_error:
{
gst_sdp_media_free (smedia);
GST_ERROR ("unable to add MIKEY crypto sessions for stream %d",
gst_rtsp_stream_get_index (stream));
return FALSE;
}
}
/**
* gst_rtsp_sdp_from_media:
* @sdp: a #GstSDPMessage
* @info: (transfer none): a #GstSDPInfo
* @media: (transfer none): a #GstRTSPMedia
*
* Add @media specific info to @sdp. @info is used to configure the connection
* information in the SDP.
*
* Returns: TRUE on success.
*/
gboolean
gst_rtsp_sdp_from_media (GstSDPMessage * sdp, GstSDPInfo * info,
GstRTSPMedia * media)
{
guint i, n_streams;
gchar *rangestr;
gboolean res;
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;
stream = gst_rtsp_media_get_stream (media, i);
res = gst_rtsp_sdp_from_stream (sdp, info, stream);
if (!res) {
GST_ERROR ("could not get SDP from stream %p", stream);
goto sdp_error;
}
}
{
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;
}
sdp_error:
{
GST_ERROR ("could not get SDP from media %p", media);
return FALSE;
}
}
/**
* gst_rtsp_sdp_from_stream:
* @sdp: a #GstSDPMessage
* @info: (transfer none): a #GstSDPInfo
* @stream: (transfer none): a #GstRTSPStream
*
* Add info from @stream to @sdp.
*
* Returns: TRUE on success.
*/
gboolean
gst_rtsp_sdp_from_stream (GstSDPMessage * sdp, GstSDPInfo * info,
GstRTSPStream * stream)
{
GstCaps *caps;
GstRTSPProfile profiles;
guint mask;
gboolean res;
caps = gst_rtsp_stream_get_caps (stream);
if (caps == NULL) {
GST_ERROR ("stream %p has no caps", stream);
return FALSE;
}
/* 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 = make_media (sdp, info, stream, caps, prof);
mask <<= 1;
}
gst_caps_unref (caps);
return res;
}