| /* 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-srtclientsrc |
| * @title: srtclientsrc |
| * |
| * srtclientsrc 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, srtclientsrc works like |
| * a client socket of connection-oriented protocol. |
| * |
| * <refsect2> |
| * <title>Examples</title> |
| * |[ |
| * gst-launch-1.0 -v srtclientsrc uri="srt://127.0.0.1:7001" ! fakesink |
| * ]| This pipeline shows how to connect SRT server by setting #GstSRTClientSrc:uri property. |
| * |
| * |[ |
| * gst-launch-1.0 -v srtclientsrc uri="srt://192.168.1.10:7001" rendez-vous ! fakesink |
| * ]| This pipeline shows how to connect SRT server by setting #GstSRTClientSrc:uri property and using the rendez-vous mode. |
| * </refsect2> |
| * |
| */ |
| |
| #ifdef HAVE_CONFIG_H |
| #include "config.h" |
| #endif |
| |
| #include "gstsrtclientsrc.h" |
| #include <srt/srt.h> |
| #include <gio/gio.h> |
| |
| #include "gstsrt.h" |
| |
| 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_client_src |
| GST_DEBUG_CATEGORY (GST_CAT_DEFAULT); |
| |
| struct _GstSRTClientSrcPrivate |
| { |
| SRTSOCKET sock; |
| gint poll_id; |
| gint poll_timeout; |
| |
| gboolean rendez_vous; |
| gchar *bind_address; |
| guint16 bind_port; |
| }; |
| |
| #define GST_SRT_CLIENT_SRC_GET_PRIVATE(obj) \ |
| (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_SRT_CLIENT_SRC, GstSRTClientSrcPrivate)) |
| |
| #define SRT_DEFAULT_POLL_TIMEOUT -1 |
| enum |
| { |
| PROP_POLL_TIMEOUT = 1, |
| PROP_BIND_ADDRESS, |
| PROP_BIND_PORT, |
| PROP_RENDEZ_VOUS, |
| |
| /*< private > */ |
| PROP_LAST |
| }; |
| |
| static GParamSpec *properties[PROP_LAST + 1]; |
| |
| #define gst_srt_client_src_parent_class parent_class |
| G_DEFINE_TYPE_WITH_CODE (GstSRTClientSrc, gst_srt_client_src, |
| GST_TYPE_SRT_BASE_SRC, G_ADD_PRIVATE (GstSRTClientSrc) |
| GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "srtclientsrc", 0, |
| "SRT Client Source")); |
| |
| static void |
| gst_srt_client_src_get_property (GObject * object, |
| guint prop_id, GValue * value, GParamSpec * pspec) |
| { |
| GstSRTClientSrc *self = GST_SRT_CLIENT_SRC (object); |
| GstSRTClientSrcPrivate *priv = GST_SRT_CLIENT_SRC_GET_PRIVATE (self); |
| |
| switch (prop_id) { |
| case PROP_POLL_TIMEOUT: |
| g_value_set_int (value, priv->poll_timeout); |
| break; |
| case PROP_BIND_PORT: |
| g_value_set_int (value, priv->rendez_vous); |
| break; |
| case PROP_BIND_ADDRESS: |
| g_value_set_string (value, priv->bind_address); |
| break; |
| case PROP_RENDEZ_VOUS: |
| g_value_set_boolean (value, priv->bind_port); |
| break; |
| default: |
| G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); |
| break; |
| } |
| } |
| |
| static void |
| gst_srt_client_src_set_property (GObject * object, |
| guint prop_id, const GValue * value, GParamSpec * pspec) |
| { |
| GstSRTBaseSrc *self = GST_SRT_BASE_SRC (object); |
| GstSRTClientSrcPrivate *priv = GST_SRT_CLIENT_SRC_GET_PRIVATE (self); |
| |
| switch (prop_id) { |
| case PROP_POLL_TIMEOUT: |
| priv->poll_timeout = g_value_get_int (value); |
| break; |
| case PROP_BIND_ADDRESS: |
| g_free (priv->bind_address); |
| priv->bind_address = g_value_dup_string (value); |
| break; |
| case PROP_BIND_PORT: |
| priv->bind_port = g_value_get_int (value); |
| break; |
| case PROP_RENDEZ_VOUS: |
| priv->rendez_vous = g_value_get_boolean (value); |
| break; |
| default: |
| G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); |
| break; |
| } |
| } |
| |
| static void |
| gst_srt_client_src_finalize (GObject * object) |
| { |
| GstSRTClientSrc *self = GST_SRT_CLIENT_SRC (object); |
| GstSRTClientSrcPrivate *priv = GST_SRT_CLIENT_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_INVALID_SOCK) { |
| srt_close (priv->sock); |
| priv->sock = SRT_INVALID_SOCK; |
| } |
| |
| g_free (priv->bind_address); |
| |
| G_OBJECT_CLASS (parent_class)->finalize (object); |
| } |
| |
| static GstFlowReturn |
| gst_srt_client_src_fill (GstPushSrc * src, GstBuffer * outbuf) |
| { |
| GstSRTClientSrc *self = GST_SRT_CLIENT_SRC (src); |
| GstSRTClientSrcPrivate *priv = GST_SRT_CLIENT_SRC_GET_PRIVATE (self); |
| GstFlowReturn ret = GST_FLOW_OK; |
| GstMapInfo info; |
| SRTSOCKET ready[2]; |
| gint recv_len; |
| |
| if (srt_epoll_wait (priv->poll_id, 0, 0, ready, &(int) { |
| 2}, priv->poll_timeout, 0, 0, 0, 0) == -1) { |
| |
| /* Assuming that timeout error is normal */ |
| if (srt_getlasterror (NULL) != SRT_ETIMEOUT) { |
| GST_ELEMENT_ERROR (src, RESOURCE, READ, |
| (NULL), ("srt_epoll_wait error: %s", srt_getlasterror_str ())); |
| ret = GST_FLOW_ERROR; |
| } |
| srt_clearlasterror (); |
| goto out; |
| } |
| |
| if (!gst_buffer_map (outbuf, &info, GST_MAP_WRITE)) { |
| GST_ELEMENT_ERROR (src, RESOURCE, READ, |
| ("Could not map the buffer for writing "), (NULL)); |
| ret = GST_FLOW_ERROR; |
| goto out; |
| } |
| |
| recv_len = srt_recvmsg (priv->sock, (char *) info.data, |
| gst_buffer_get_size (outbuf)); |
| |
| gst_buffer_unmap (outbuf, &info); |
| |
| if (recv_len == SRT_ERROR) { |
| GST_ELEMENT_ERROR (src, RESOURCE, READ, |
| (NULL), ("srt_recvmsg error: %s", srt_getlasterror_str ())); |
| ret = GST_FLOW_ERROR; |
| 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_client_src_start (GstBaseSrc * src) |
| { |
| GstSRTClientSrc *self = GST_SRT_CLIENT_SRC (src); |
| GstSRTClientSrcPrivate *priv = GST_SRT_CLIENT_SRC_GET_PRIVATE (self); |
| GstSRTBaseSrc *base = GST_SRT_BASE_SRC (src); |
| GstUri *uri = gst_uri_ref (base->uri); |
| GSocketAddress *socket_address = NULL; |
| |
| priv->sock = gst_srt_client_connect_full (GST_ELEMENT (src), FALSE, |
| gst_uri_get_host (uri), gst_uri_get_port (uri), priv->rendez_vous, |
| priv->bind_address, priv->bind_port, base->latency, |
| &socket_address, &priv->poll_id, base->passphrase, base->key_length); |
| |
| g_clear_object (&socket_address); |
| g_clear_pointer (&uri, gst_uri_unref); |
| |
| return (priv->sock != SRT_INVALID_SOCK); |
| } |
| |
| static gboolean |
| gst_srt_client_src_stop (GstBaseSrc * src) |
| { |
| GstSRTClientSrc *self = GST_SRT_CLIENT_SRC (src); |
| GstSRTClientSrcPrivate *priv = GST_SRT_CLIENT_SRC_GET_PRIVATE (self); |
| |
| if (priv->poll_id != SRT_ERROR) { |
| if (priv->sock != SRT_INVALID_SOCK) |
| srt_epoll_remove_usock (priv->poll_id, priv->sock); |
| srt_epoll_release (priv->poll_id); |
| } |
| priv->poll_id = SRT_ERROR; |
| |
| GST_DEBUG_OBJECT (self, "closing SRT connection"); |
| if (priv->sock != SRT_INVALID_SOCK) |
| srt_close (priv->sock); |
| priv->sock = SRT_INVALID_SOCK; |
| |
| return TRUE; |
| } |
| |
| static void |
| gst_srt_client_src_class_init (GstSRTClientSrcClass * 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_client_src_set_property; |
| gobject_class->get_property = gst_srt_client_src_get_property; |
| gobject_class->finalize = gst_srt_client_src_finalize; |
| |
| /** |
| * GstSRTClientSrc:poll-timeout: |
| * |
| * The timeout(ms) value when polling SRT socket. |
| */ |
| properties[PROP_POLL_TIMEOUT] = |
| g_param_spec_int ("poll-timeout", "Poll timeout", |
| "Return poll wait after timeout miliseconds (-1 = infinite)", -1, |
| G_MAXINT32, SRT_DEFAULT_POLL_TIMEOUT, |
| G_PARAM_READWRITE | GST_PARAM_MUTABLE_READY | G_PARAM_STATIC_STRINGS); |
| |
| properties[PROP_BIND_ADDRESS] = |
| g_param_spec_string ("bind-address", "Bind Address", |
| "Address to bind socket to (required for rendez-vous mode) ", NULL, |
| G_PARAM_READWRITE | GST_PARAM_MUTABLE_READY | G_PARAM_STATIC_STRINGS); |
| |
| properties[PROP_BIND_PORT] = |
| g_param_spec_int ("bind-port", "Bind Port", |
| "Port to bind socket to (Ignored in rendez-vous mode)", 0, |
| G_MAXUINT16, 0, |
| G_PARAM_READWRITE | GST_PARAM_MUTABLE_READY | G_PARAM_STATIC_STRINGS); |
| |
| properties[PROP_RENDEZ_VOUS] = |
| g_param_spec_boolean ("rendez-vous", "Rendez Vous", |
| "Work in Rendez-Vous mode instead of client/caller mode", FALSE, |
| G_PARAM_READWRITE | GST_PARAM_MUTABLE_READY | G_PARAM_STATIC_STRINGS); |
| |
| g_object_class_install_properties (gobject_class, PROP_LAST, properties); |
| |
| gst_element_class_add_static_pad_template (gstelement_class, &src_template); |
| gst_element_class_set_metadata (gstelement_class, |
| "SRT client source", "Source/Network", |
| "Receive data over the network via SRT", |
| "Justin Kim <justin.kim@collabora.com>"); |
| |
| gstbasesrc_class->start = GST_DEBUG_FUNCPTR (gst_srt_client_src_start); |
| gstbasesrc_class->stop = GST_DEBUG_FUNCPTR (gst_srt_client_src_stop); |
| |
| gstpushsrc_class->fill = GST_DEBUG_FUNCPTR (gst_srt_client_src_fill); |
| } |
| |
| static void |
| gst_srt_client_src_init (GstSRTClientSrc * self) |
| { |
| GstSRTClientSrcPrivate *priv = GST_SRT_CLIENT_SRC_GET_PRIVATE (self); |
| |
| priv->sock = SRT_INVALID_SOCK; |
| priv->poll_id = SRT_ERROR; |
| priv->poll_timeout = SRT_DEFAULT_POLL_TIMEOUT; |
| priv->rendez_vous = FALSE; |
| priv->bind_address = NULL; |
| priv->bind_port = 0; |
| } |