| /* GStreamer SRT plugin based on libsrt |
| * Copyright (C) 2017, Collabora Ltd. |
| * Author:Justin Kim <justin.kim@collabora.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:element-srtserversrc |
| * @title: srtserversrc |
| * |
| * srtserversrc is a network source that reads <ulink url="http://www.srtalliance.org/">SRT</ulink> |
| * packets from the network. Although SRT is a protocol based on UDP, srtserversrc works like |
| * a server socket of connection-oriented protocol, but it accepts to only one client connection. |
| * |
| * <refsect2> |
| * <title>Examples</title> |
| * |[ |
| * gst-launch-1.0 -v srtserversrc uri="srt://:7001" ! fakesink |
| * ]| This pipeline shows how to bind SRT server by setting #GstSRTServerSrc:uri property. |
| * </refsect2> |
| * |
| */ |
| |
| #ifdef HAVE_CONFIG_H |
| #include "config.h" |
| #endif |
| |
| #include "gstsrtserversrc.h" |
| #include "gstsrt.h" |
| #include <gio/gio.h> |
| |
| #define SRT_DEFAULT_POLL_TIMEOUT 100 |
| |
| static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", |
| GST_PAD_SRC, |
| GST_PAD_ALWAYS, |
| GST_STATIC_CAPS_ANY); |
| |
| #define GST_CAT_DEFAULT gst_debug_srt_server_src |
| GST_DEBUG_CATEGORY (GST_CAT_DEFAULT); |
| |
| struct _GstSRTServerSrcPrivate |
| { |
| SRTSOCKET sock; |
| SRTSOCKET client_sock; |
| GSocketAddress *client_sockaddr; |
| |
| gint poll_id; |
| gint poll_timeout; |
| |
| gboolean has_client; |
| gboolean cancelled; |
| }; |
| |
| #define GST_SRT_SERVER_SRC_GET_PRIVATE(obj) \ |
| (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_SRT_SERVER_SRC, GstSRTServerSrcPrivate)) |
| |
| enum |
| { |
| PROP_POLL_TIMEOUT = 1, |
| |
| /*< private > */ |
| PROP_LAST |
| }; |
| |
| static GParamSpec *properties[PROP_LAST]; |
| |
| enum |
| { |
| SIG_CLIENT_ADDED, |
| SIG_CLIENT_CLOSED, |
| |
| LAST_SIGNAL |
| }; |
| |
| static guint signals[LAST_SIGNAL] = { 0 }; |
| |
| #define gst_srt_server_src_parent_class parent_class |
| G_DEFINE_TYPE_WITH_CODE (GstSRTServerSrc, gst_srt_server_src, |
| GST_TYPE_SRT_BASE_SRC, G_ADD_PRIVATE (GstSRTServerSrc) |
| GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "srtserversrc", 0, |
| "SRT Server Source")); |
| |
| static void |
| gst_srt_server_src_get_property (GObject * object, |
| guint prop_id, GValue * value, GParamSpec * pspec) |
| { |
| GstSRTServerSrc *self = GST_SRT_SERVER_SRC (object); |
| GstSRTServerSrcPrivate *priv = GST_SRT_SERVER_SRC_GET_PRIVATE (self); |
| |
| switch (prop_id) { |
| case PROP_POLL_TIMEOUT: |
| g_value_set_int (value, priv->poll_timeout); |
| break; |
| default: |
| G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); |
| break; |
| } |
| } |
| |
| static void |
| gst_srt_server_src_set_property (GObject * object, |
| guint prop_id, const GValue * value, GParamSpec * pspec) |
| { |
| GstSRTServerSrc *self = GST_SRT_SERVER_SRC (object); |
| GstSRTServerSrcPrivate *priv = GST_SRT_SERVER_SRC_GET_PRIVATE (self); |
| |
| switch (prop_id) { |
| case PROP_POLL_TIMEOUT: |
| priv->poll_timeout = g_value_get_int (value); |
| break; |
| default: |
| G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); |
| break; |
| } |
| } |
| |
| static void |
| gst_srt_server_src_finalize (GObject * object) |
| { |
| GstSRTServerSrc *self = GST_SRT_SERVER_SRC (object); |
| GstSRTServerSrcPrivate *priv = GST_SRT_SERVER_SRC_GET_PRIVATE (self); |
| |
| if (priv->poll_id != SRT_ERROR) { |
| srt_epoll_release (priv->poll_id); |
| priv->poll_id = SRT_ERROR; |
| } |
| |
| if (priv->sock != SRT_ERROR) { |
| srt_close (priv->sock); |
| priv->sock = SRT_ERROR; |
| } |
| |
| G_OBJECT_CLASS (parent_class)->finalize (object); |
| } |
| |
| static GstFlowReturn |
| gst_srt_server_src_fill (GstPushSrc * src, GstBuffer * outbuf) |
| { |
| GstSRTServerSrc *self = GST_SRT_SERVER_SRC (src); |
| GstSRTServerSrcPrivate *priv = GST_SRT_SERVER_SRC_GET_PRIVATE (self); |
| GstFlowReturn ret = GST_FLOW_OK; |
| GstMapInfo info; |
| SRTSOCKET ready[2]; |
| gint recv_len; |
| struct sockaddr client_sa; |
| size_t client_sa_len; |
| |
| while (!priv->has_client) { |
| GST_DEBUG_OBJECT (self, "poll wait (timeout: %d)", priv->poll_timeout); |
| |
| /* Make SRT server socket non-blocking */ |
| srt_setsockopt (priv->sock, 0, SRTO_SNDSYN, &(int) { |
| 0}, sizeof (int)); |
| |
| if (srt_epoll_wait (priv->poll_id, ready, &(int) { |
| 2}, 0, 0, priv->poll_timeout, 0, 0, 0, 0) == -1) { |
| int srt_errno = srt_getlasterror (NULL); |
| |
| /* Assuming that timeout error is normal */ |
| if (srt_errno != SRT_ETIMEOUT) { |
| GST_ELEMENT_ERROR (src, RESOURCE, FAILED, |
| ("SRT error: %s", srt_getlasterror_str ()), (NULL)); |
| |
| return GST_FLOW_ERROR; |
| } |
| |
| /* Mimicking cancellable */ |
| if (srt_errno == SRT_ETIMEOUT && priv->cancelled) { |
| GST_DEBUG_OBJECT (self, "Cancelled waiting for client"); |
| return GST_FLOW_FLUSHING; |
| } |
| |
| continue; |
| } |
| |
| priv->client_sock = |
| srt_accept (priv->sock, &client_sa, (int *) &client_sa_len); |
| |
| GST_DEBUG_OBJECT (self, "checking client sock"); |
| if (priv->client_sock == SRT_INVALID_SOCK) { |
| GST_WARNING_OBJECT (self, |
| "detected invalid SRT client socket (reason: %s)", |
| srt_getlasterror_str ()); |
| srt_clearlasterror (); |
| } else { |
| priv->has_client = TRUE; |
| g_clear_object (&priv->client_sockaddr); |
| priv->client_sockaddr = g_socket_address_new_from_native (&client_sa, |
| client_sa_len); |
| g_signal_emit (self, signals[SIG_CLIENT_ADDED], 0, |
| priv->client_sock, priv->client_sockaddr); |
| } |
| } |
| |
| GST_DEBUG_OBJECT (self, "filling buffer"); |
| |
| if (!gst_buffer_map (outbuf, &info, GST_MAP_WRITE)) { |
| GST_ELEMENT_ERROR (src, RESOURCE, WRITE, |
| ("Could not map the output stream"), (NULL)); |
| ret = GST_FLOW_ERROR; |
| goto out; |
| } |
| |
| recv_len = srt_recvmsg (priv->client_sock, (char *) info.data, |
| gst_buffer_get_size (outbuf)); |
| |
| gst_buffer_unmap (outbuf, &info); |
| |
| if (recv_len == SRT_ERROR) { |
| GST_WARNING_OBJECT (self, "%s", srt_getlasterror_str ()); |
| |
| g_signal_emit (self, signals[SIG_CLIENT_CLOSED], 0, |
| priv->client_sock, priv->client_sockaddr); |
| |
| srt_close (priv->client_sock); |
| priv->client_sock = SRT_INVALID_SOCK; |
| g_clear_object (&priv->client_sockaddr); |
| priv->has_client = FALSE; |
| gst_buffer_resize (outbuf, 0, 0); |
| ret = GST_FLOW_OK; |
| goto out; |
| } else if (recv_len == 0) { |
| ret = GST_FLOW_EOS; |
| goto out; |
| } |
| |
| GST_BUFFER_PTS (outbuf) = |
| gst_clock_get_time (GST_ELEMENT_CLOCK (src)) - |
| GST_ELEMENT_CAST (src)->base_time; |
| |
| gst_buffer_resize (outbuf, 0, recv_len); |
| |
| GST_LOG_OBJECT (src, |
| "filled buffer from _get of size %" G_GSIZE_FORMAT ", ts %" |
| GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT |
| ", offset %" G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT, |
| gst_buffer_get_size (outbuf), |
| GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)), |
| GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), |
| GST_BUFFER_OFFSET (outbuf), GST_BUFFER_OFFSET_END (outbuf)); |
| |
| out: |
| return ret; |
| } |
| |
| static gboolean |
| gst_srt_server_src_start (GstBaseSrc * src) |
| { |
| GstSRTServerSrc *self = GST_SRT_SERVER_SRC (src); |
| GstSRTServerSrcPrivate *priv = GST_SRT_SERVER_SRC_GET_PRIVATE (self); |
| GstSRTBaseSrc *base = GST_SRT_BASE_SRC (src); |
| GstUri *uri = gst_uri_ref (base->uri); |
| GError *error = NULL; |
| struct sockaddr sa; |
| size_t sa_len; |
| GSocketAddress *socket_address; |
| const gchar *host; |
| int lat = base->latency; |
| |
| if (gst_uri_get_port (uri) == GST_URI_NO_PORT) { |
| GST_ELEMENT_ERROR (src, RESOURCE, OPEN_WRITE, NULL, (("Invalid port"))); |
| return FALSE; |
| } |
| |
| host = gst_uri_get_host (uri); |
| if (host == NULL) { |
| GInetAddress *any = g_inet_address_new_any (G_SOCKET_FAMILY_IPV4); |
| |
| socket_address = g_inet_socket_address_new (any, gst_uri_get_port (uri)); |
| g_object_unref (any); |
| } else { |
| socket_address = |
| g_inet_socket_address_new_from_string (host, gst_uri_get_port (uri)); |
| } |
| |
| if (socket_address == NULL) { |
| GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, ("Invalid URI"), |
| ("failed to extract host or port from the given URI")); |
| goto failed; |
| } |
| |
| sa_len = g_socket_address_get_native_size (socket_address); |
| if (!g_socket_address_to_native (socket_address, &sa, sa_len, &error)) { |
| GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, ("Invalid URI"), |
| ("cannot resolve address (reason: %s)", error->message)); |
| goto failed; |
| } |
| |
| priv->sock = srt_socket (sa.sa_family, SOCK_DGRAM, 0); |
| if (priv->sock == SRT_ERROR) { |
| GST_ELEMENT_ERROR (self, LIBRARY, INIT, (NULL), |
| ("failed to create poll id for SRT socket (reason: %s)", |
| srt_getlasterror_str ())); |
| goto failed; |
| } |
| |
| /* Make sure TSBPD mode is enable (SRT mode) */ |
| srt_setsockopt (priv->sock, 0, SRTO_TSBPDMODE, &(int) { |
| 1}, sizeof (int)); |
| |
| /* This is a sink, we're always a receiver */ |
| srt_setsockopt (priv->sock, 0, SRTO_SENDER, &(int) { |
| 0}, sizeof (int)); |
| |
| srt_setsockopt (priv->sock, 0, SRTO_TSBPDDELAY, &lat, sizeof (int)); |
| |
| if (base->passphrase != NULL && base->passphrase[0] != '\0') { |
| srt_setsockopt (priv->sock, 0, SRTO_PASSPHRASE, |
| base->passphrase, strlen (base->passphrase)); |
| srt_setsockopt (priv->sock, 0, SRTO_PBKEYLEN, |
| &base->key_length, sizeof (int)); |
| } |
| |
| priv->poll_id = srt_epoll_create (); |
| if (priv->poll_id == -1) { |
| GST_ELEMENT_ERROR (self, LIBRARY, INIT, (NULL), |
| ("failed to create poll id for SRT socket (reason: %s)", |
| srt_getlasterror_str ())); |
| goto failed; |
| } |
| |
| srt_epoll_add_usock (priv->poll_id, priv->sock, &(int) { |
| SRT_EPOLL_IN}); |
| |
| if (srt_bind (priv->sock, &sa, sa_len) == SRT_ERROR) { |
| GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL), |
| ("failed to bind SRT server socket (reason: %s)", |
| srt_getlasterror_str ())); |
| goto failed; |
| } |
| |
| if (srt_listen (priv->sock, 1) == SRT_ERROR) { |
| GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL), |
| ("failed to listen SRT socket (reason: %s)", srt_getlasterror_str ())); |
| goto failed; |
| } |
| |
| g_clear_pointer (&uri, gst_uri_unref); |
| g_clear_object (&socket_address); |
| |
| return TRUE; |
| |
| failed: |
| if (priv->poll_id != SRT_ERROR) { |
| srt_epoll_release (priv->poll_id); |
| priv->poll_id = SRT_ERROR; |
| } |
| |
| if (priv->sock != SRT_ERROR) { |
| srt_close (priv->sock); |
| priv->sock = SRT_ERROR; |
| } |
| |
| g_clear_error (&error); |
| g_clear_pointer (&uri, gst_uri_unref); |
| g_clear_object (&socket_address); |
| |
| return FALSE; |
| } |
| |
| static gboolean |
| gst_srt_server_src_stop (GstBaseSrc * src) |
| { |
| GstSRTServerSrc *self = GST_SRT_SERVER_SRC (src); |
| GstSRTServerSrcPrivate *priv = GST_SRT_SERVER_SRC_GET_PRIVATE (self); |
| |
| if (priv->client_sock != SRT_INVALID_SOCK) { |
| g_signal_emit (self, signals[SIG_CLIENT_ADDED], 0, |
| priv->client_sock, priv->client_sockaddr); |
| srt_close (priv->client_sock); |
| g_clear_object (&priv->client_sockaddr); |
| priv->client_sock = SRT_INVALID_SOCK; |
| priv->has_client = FALSE; |
| } |
| |
| if (priv->poll_id != SRT_ERROR) { |
| srt_epoll_remove_usock (priv->poll_id, priv->sock); |
| srt_epoll_release (priv->poll_id); |
| priv->poll_id = SRT_ERROR; |
| } |
| |
| if (priv->sock != SRT_INVALID_SOCK) { |
| GST_DEBUG_OBJECT (self, "closing SRT connection"); |
| srt_close (priv->sock); |
| priv->sock = SRT_INVALID_SOCK; |
| } |
| |
| priv->cancelled = FALSE; |
| |
| return TRUE; |
| } |
| |
| static gboolean |
| gst_srt_server_src_unlock (GstBaseSrc * src) |
| { |
| GstSRTServerSrc *self = GST_SRT_SERVER_SRC (src); |
| GstSRTServerSrcPrivate *priv = GST_SRT_SERVER_SRC_GET_PRIVATE (self); |
| |
| priv->cancelled = TRUE; |
| |
| return TRUE; |
| } |
| |
| static gboolean |
| gst_srt_server_src_unlock_stop (GstBaseSrc * src) |
| { |
| GstSRTServerSrc *self = GST_SRT_SERVER_SRC (src); |
| GstSRTServerSrcPrivate *priv = GST_SRT_SERVER_SRC_GET_PRIVATE (self); |
| |
| priv->cancelled = FALSE; |
| |
| return TRUE; |
| } |
| |
| static void |
| gst_srt_server_src_class_init (GstSRTServerSrcClass * klass) |
| { |
| GObjectClass *gobject_class = G_OBJECT_CLASS (klass); |
| GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass); |
| GstBaseSrcClass *gstbasesrc_class = GST_BASE_SRC_CLASS (klass); |
| GstPushSrcClass *gstpushsrc_class = GST_PUSH_SRC_CLASS (klass); |
| |
| gobject_class->set_property = gst_srt_server_src_set_property; |
| gobject_class->get_property = gst_srt_server_src_get_property; |
| gobject_class->finalize = gst_srt_server_src_finalize; |
| |
| /** |
| * GstSRTServerSrc:poll-timeout: |
| * |
| * The timeout(ms) value when polling SRT socket. For #GstSRTServerSrc, |
| * this value shouldn't be set as -1 (infinite) because "srt_epoll_wait" |
| * isn't cancellable unless closing the socket. |
| */ |
| properties[PROP_POLL_TIMEOUT] = |
| g_param_spec_int ("poll-timeout", "Poll timeout", |
| "Return poll wait after timeout miliseconds", 0, G_MAXINT32, |
| SRT_DEFAULT_POLL_TIMEOUT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS); |
| |
| g_object_class_install_properties (gobject_class, PROP_LAST, properties); |
| |
| /** |
| * GstSRTServerSrc::client-added: |
| * @gstsrtserversrc: the srtserversrc element that emitted this signal |
| * @sock: the client socket descriptor that was added to srtserversrc |
| * @addr: the pointer of "struct sockaddr" that describes the @sock |
| * @addr_len: the length of @addr |
| * |
| * The given socket descriptor was added to srtserversrc. |
| */ |
| signals[SIG_CLIENT_ADDED] = |
| g_signal_new ("client-added", G_TYPE_FROM_CLASS (klass), |
| G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstSRTServerSrcClass, client_added), |
| NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, |
| 2, G_TYPE_INT, G_TYPE_SOCKET_ADDRESS); |
| |
| /** |
| * GstSRTServerSrc::client-closed: |
| * @gstsrtserversrc: the srtserversrc element that emitted this signal |
| * @sock: the client socket descriptor that was added to srtserversrc |
| * @addr: the pointer of "struct sockaddr" that describes the @sock |
| * @addr_len: the length of @addr |
| * |
| * The given socket descriptor was closed. |
| */ |
| signals[SIG_CLIENT_CLOSED] = |
| g_signal_new ("client-closed", G_TYPE_FROM_CLASS (klass), |
| G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstSRTServerSrcClass, client_closed), |
| NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, |
| 2, G_TYPE_INT, G_TYPE_SOCKET_ADDRESS); |
| |
| gst_element_class_add_static_pad_template (gstelement_class, &src_template); |
| gst_element_class_set_metadata (gstelement_class, |
| "SRT Server source", "Source/Network", |
| "Receive data over the network via SRT", |
| "Justin Kim <justin.kim@collabora.com>"); |
| |
| gstbasesrc_class->start = GST_DEBUG_FUNCPTR (gst_srt_server_src_start); |
| gstbasesrc_class->stop = GST_DEBUG_FUNCPTR (gst_srt_server_src_stop); |
| gstbasesrc_class->unlock = GST_DEBUG_FUNCPTR (gst_srt_server_src_unlock); |
| gstbasesrc_class->unlock_stop = |
| GST_DEBUG_FUNCPTR (gst_srt_server_src_unlock_stop); |
| |
| gstpushsrc_class->fill = GST_DEBUG_FUNCPTR (gst_srt_server_src_fill); |
| } |
| |
| static void |
| gst_srt_server_src_init (GstSRTServerSrc * self) |
| { |
| GstSRTServerSrcPrivate *priv = GST_SRT_SERVER_SRC_GET_PRIVATE (self); |
| |
| priv->sock = SRT_INVALID_SOCK; |
| priv->client_sock = SRT_INVALID_SOCK; |
| priv->poll_id = SRT_ERROR; |
| priv->poll_timeout = SRT_DEFAULT_POLL_TIMEOUT; |
| } |