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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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337 lines
10 KiB
C
337 lines
10 KiB
C
/* GStreamer SRT plugin based on libsrt
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* Copyright (C) 2017, Collabora Ltd.
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* Author:Justin Kim <justin.kim@collabora.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-srtclientsrc
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* @title: srtclientsrc
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*
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* srtclientsrc is a network source that reads <ulink url="http://www.srtalliance.org/">SRT</ulink>
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* packets from the network. Although SRT is a protocol based on UDP, srtclientsrc works like
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* a client socket of connection-oriented protocol.
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*
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* <refsect2>
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* <title>Examples</title>
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* |[
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* gst-launch-1.0 -v srtclientsrc uri="srt://127.0.0.1:7001" ! fakesink
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* ]| This pipeline shows how to connect SRT server by setting #GstSRTClientSrc:uri property.
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*
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* |[
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* gst-launch-1.0 -v srtclientsrc uri="srt://192.168.1.10:7001" rendez-vous ! fakesink
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* ]| This pipeline shows how to connect SRT server by setting #GstSRTClientSrc:uri property and using the rendez-vous mode.
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* </refsect2>
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*
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include "gstsrtclientsrc.h"
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#include <srt/srt.h>
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#include <gio/gio.h>
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#include "gstsrt.h"
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static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS_ANY);
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#define GST_CAT_DEFAULT gst_debug_srt_client_src
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GST_DEBUG_CATEGORY (GST_CAT_DEFAULT);
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struct _GstSRTClientSrcPrivate
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{
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SRTSOCKET sock;
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gint poll_id;
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gint poll_timeout;
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gboolean rendez_vous;
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gchar *bind_address;
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guint16 bind_port;
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};
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#define GST_SRT_CLIENT_SRC_GET_PRIVATE(obj) \
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(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_SRT_CLIENT_SRC, GstSRTClientSrcPrivate))
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#define SRT_DEFAULT_POLL_TIMEOUT -1
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enum
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{
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PROP_POLL_TIMEOUT = 1,
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PROP_BIND_ADDRESS,
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PROP_BIND_PORT,
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PROP_RENDEZ_VOUS,
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/*< private > */
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PROP_LAST
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};
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static GParamSpec *properties[PROP_LAST + 1];
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#define gst_srt_client_src_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstSRTClientSrc, gst_srt_client_src,
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GST_TYPE_SRT_BASE_SRC, G_ADD_PRIVATE (GstSRTClientSrc)
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GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "srtclientsrc", 0,
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"SRT Client Source"));
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static void
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gst_srt_client_src_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec)
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{
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GstSRTClientSrc *self = GST_SRT_CLIENT_SRC (object);
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GstSRTClientSrcPrivate *priv = GST_SRT_CLIENT_SRC_GET_PRIVATE (self);
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switch (prop_id) {
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case PROP_POLL_TIMEOUT:
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g_value_set_int (value, priv->poll_timeout);
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break;
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case PROP_BIND_PORT:
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g_value_set_int (value, priv->bind_port);
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break;
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case PROP_BIND_ADDRESS:
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g_value_set_string (value, priv->bind_address);
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break;
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case PROP_RENDEZ_VOUS:
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g_value_set_boolean (value, priv->rendez_vous);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_srt_client_src_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec)
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{
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GstSRTBaseSrc *self = GST_SRT_BASE_SRC (object);
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GstSRTClientSrcPrivate *priv = GST_SRT_CLIENT_SRC_GET_PRIVATE (self);
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switch (prop_id) {
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case PROP_POLL_TIMEOUT:
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priv->poll_timeout = g_value_get_int (value);
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break;
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case PROP_BIND_ADDRESS:
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g_free (priv->bind_address);
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priv->bind_address = g_value_dup_string (value);
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break;
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case PROP_BIND_PORT:
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priv->bind_port = g_value_get_int (value);
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break;
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case PROP_RENDEZ_VOUS:
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priv->rendez_vous = g_value_get_boolean (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_srt_client_src_finalize (GObject * object)
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{
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GstSRTClientSrc *self = GST_SRT_CLIENT_SRC (object);
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GstSRTClientSrcPrivate *priv = GST_SRT_CLIENT_SRC_GET_PRIVATE (self);
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if (priv->poll_id != SRT_ERROR) {
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srt_epoll_release (priv->poll_id);
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priv->poll_id = SRT_ERROR;
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}
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if (priv->sock != SRT_INVALID_SOCK) {
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srt_close (priv->sock);
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priv->sock = SRT_INVALID_SOCK;
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}
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g_free (priv->bind_address);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static GstFlowReturn
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gst_srt_client_src_fill (GstPushSrc * src, GstBuffer * outbuf)
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{
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GstSRTClientSrc *self = GST_SRT_CLIENT_SRC (src);
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GstSRTClientSrcPrivate *priv = GST_SRT_CLIENT_SRC_GET_PRIVATE (self);
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GstFlowReturn ret = GST_FLOW_OK;
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GstMapInfo info;
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SRTSOCKET ready[2];
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gint recv_len;
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if (srt_epoll_wait (priv->poll_id, 0, 0, ready, &(int) {
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2}, priv->poll_timeout, 0, 0, 0, 0) == -1) {
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/* Assuming that timeout error is normal */
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if (srt_getlasterror (NULL) != SRT_ETIMEOUT) {
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GST_ELEMENT_ERROR (src, RESOURCE, READ,
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(NULL), ("srt_epoll_wait error: %s", srt_getlasterror_str ()));
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ret = GST_FLOW_ERROR;
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}
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srt_clearlasterror ();
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goto out;
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}
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if (!gst_buffer_map (outbuf, &info, GST_MAP_WRITE)) {
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GST_ELEMENT_ERROR (src, RESOURCE, READ,
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("Could not map the buffer for writing "), (NULL));
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ret = GST_FLOW_ERROR;
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goto out;
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}
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recv_len = srt_recvmsg (priv->sock, (char *) info.data,
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gst_buffer_get_size (outbuf));
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gst_buffer_unmap (outbuf, &info);
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if (recv_len == SRT_ERROR) {
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GST_ELEMENT_ERROR (src, RESOURCE, READ,
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(NULL), ("srt_recvmsg error: %s", srt_getlasterror_str ()));
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ret = GST_FLOW_ERROR;
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goto out;
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} else if (recv_len == 0) {
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ret = GST_FLOW_EOS;
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goto out;
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}
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GST_BUFFER_PTS (outbuf) =
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gst_clock_get_time (GST_ELEMENT_CLOCK (src)) -
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GST_ELEMENT_CAST (src)->base_time;
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gst_buffer_resize (outbuf, 0, recv_len);
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GST_LOG_OBJECT (src,
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"filled buffer from _get of size %" G_GSIZE_FORMAT ", ts %"
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GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT
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", offset %" G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
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gst_buffer_get_size (outbuf),
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GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
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GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)),
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GST_BUFFER_OFFSET (outbuf), GST_BUFFER_OFFSET_END (outbuf));
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out:
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return ret;
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}
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static gboolean
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gst_srt_client_src_start (GstBaseSrc * src)
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{
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GstSRTClientSrc *self = GST_SRT_CLIENT_SRC (src);
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GstSRTClientSrcPrivate *priv = GST_SRT_CLIENT_SRC_GET_PRIVATE (self);
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GstSRTBaseSrc *base = GST_SRT_BASE_SRC (src);
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GstUri *uri = gst_uri_ref (base->uri);
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GSocketAddress *socket_address = NULL;
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priv->sock = gst_srt_client_connect (GST_ELEMENT (src), FALSE,
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gst_uri_get_host (uri), gst_uri_get_port (uri), priv->rendez_vous,
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priv->bind_address, priv->bind_port, base->latency,
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&socket_address, &priv->poll_id, base->passphrase, base->key_length);
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g_clear_object (&socket_address);
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g_clear_pointer (&uri, gst_uri_unref);
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return (priv->sock != SRT_INVALID_SOCK);
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}
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static gboolean
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gst_srt_client_src_stop (GstBaseSrc * src)
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{
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GstSRTClientSrc *self = GST_SRT_CLIENT_SRC (src);
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GstSRTClientSrcPrivate *priv = GST_SRT_CLIENT_SRC_GET_PRIVATE (self);
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if (priv->poll_id != SRT_ERROR) {
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if (priv->sock != SRT_INVALID_SOCK)
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srt_epoll_remove_usock (priv->poll_id, priv->sock);
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srt_epoll_release (priv->poll_id);
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}
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priv->poll_id = SRT_ERROR;
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GST_DEBUG_OBJECT (self, "closing SRT connection");
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if (priv->sock != SRT_INVALID_SOCK)
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srt_close (priv->sock);
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priv->sock = SRT_INVALID_SOCK;
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return TRUE;
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}
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static void
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gst_srt_client_src_class_init (GstSRTClientSrcClass * klass)
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{
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
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GstBaseSrcClass *gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
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GstPushSrcClass *gstpushsrc_class = GST_PUSH_SRC_CLASS (klass);
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gobject_class->set_property = gst_srt_client_src_set_property;
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gobject_class->get_property = gst_srt_client_src_get_property;
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gobject_class->finalize = gst_srt_client_src_finalize;
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/**
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* GstSRTClientSrc:poll-timeout:
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*
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* The timeout(ms) value when polling SRT socket.
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*/
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properties[PROP_POLL_TIMEOUT] =
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g_param_spec_int ("poll-timeout", "Poll timeout",
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"Return poll wait after timeout miliseconds (-1 = infinite)", -1,
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G_MAXINT32, SRT_DEFAULT_POLL_TIMEOUT,
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G_PARAM_READWRITE | GST_PARAM_MUTABLE_READY | G_PARAM_STATIC_STRINGS);
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properties[PROP_BIND_ADDRESS] =
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g_param_spec_string ("bind-address", "Bind Address",
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"Address to bind socket to (required for rendez-vous mode) ", NULL,
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G_PARAM_READWRITE | GST_PARAM_MUTABLE_READY | G_PARAM_STATIC_STRINGS);
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properties[PROP_BIND_PORT] =
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g_param_spec_int ("bind-port", "Bind Port",
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"Port to bind socket to (Ignored in rendez-vous mode)", 0,
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G_MAXUINT16, 0,
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G_PARAM_READWRITE | GST_PARAM_MUTABLE_READY | G_PARAM_STATIC_STRINGS);
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properties[PROP_RENDEZ_VOUS] =
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g_param_spec_boolean ("rendez-vous", "Rendez Vous",
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"Work in Rendez-Vous mode instead of client/caller mode", FALSE,
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G_PARAM_READWRITE | GST_PARAM_MUTABLE_READY | G_PARAM_STATIC_STRINGS);
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g_object_class_install_properties (gobject_class, PROP_LAST, properties);
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gst_element_class_add_static_pad_template (gstelement_class, &src_template);
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gst_element_class_set_metadata (gstelement_class,
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"SRT client source", "Source/Network",
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"Receive data over the network via SRT",
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"Justin Kim <justin.kim@collabora.com>");
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gstbasesrc_class->start = GST_DEBUG_FUNCPTR (gst_srt_client_src_start);
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gstbasesrc_class->stop = GST_DEBUG_FUNCPTR (gst_srt_client_src_stop);
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gstpushsrc_class->fill = GST_DEBUG_FUNCPTR (gst_srt_client_src_fill);
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}
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static void
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gst_srt_client_src_init (GstSRTClientSrc * self)
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{
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GstSRTClientSrcPrivate *priv = GST_SRT_CLIENT_SRC_GET_PRIVATE (self);
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priv->sock = SRT_INVALID_SOCK;
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priv->poll_id = SRT_ERROR;
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priv->poll_timeout = SRT_DEFAULT_POLL_TIMEOUT;
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priv->rendez_vous = FALSE;
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priv->bind_address = NULL;
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priv->bind_port = 0;
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}
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