mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-18 14:26:43 +00:00
d8f61515d8
By passing NULL to `g_signal_new` instead of a marshaller, GLib will actually internally optimize the signal (if the marshaller is available in GLib itself) by also setting the valist marshaller. This makes the signal emission a bit more performant than the regular marshalling, which still needs to box into `GValue` and call libffi in case of a generic marshaller. Note that for custom marshallers, one would use `g_signal_set_va_marshaller()` with the valist marshaller instead.
358 lines
10 KiB
C
358 lines
10 KiB
C
/* GStreamer
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* Copyright (C) 2018, Collabora Ltd.
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* Copyright (C) 2018, SK Telecom, Co., Ltd.
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* Author: Jeongseok Kim <jeongseok.kim@sk.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-srtsrc
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* @title: srtsrc
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*
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* srtsrc is a network source that reads [SRT](http://www.srtalliance.org/)
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* packets from the network.
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*
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* ## Examples
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* |[
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* gst-launch-1.0 -v srtsrc uri="srt://127.0.0.1:7001" ! fakesink
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* ]| This pipeline shows how to connect SRT server by setting #GstSRTSrc:uri property.
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*
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* |[
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* gst-launch-1.0 -v srtsrc uri="srt://:7001?mode=listener" ! fakesink
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* ]| This pipeline shows how to wait SRT connection by setting #GstSRTSrc: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?mode=rendez-vous" ! fakesink
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* ]| This pipeline shows how to connect SRT server by setting #GstSRTSrc:uri property and using the rendez-vous mode.
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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 "gstsrtsrc.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_src
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GST_DEBUG_CATEGORY (GST_CAT_DEFAULT);
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enum
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{
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SIG_CALLER_ADDED,
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SIG_CALLER_REMOVED,
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LAST_SIGNAL
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};
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static guint signals[LAST_SIGNAL] = { 0 };
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static void gst_srt_src_uri_handler_init (gpointer g_iface,
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gpointer iface_data);
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static gchar *gst_srt_src_uri_get_uri (GstURIHandler * handler);
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static gboolean gst_srt_src_uri_set_uri (GstURIHandler * handler,
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const gchar * uri, GError ** error);
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#define gst_srt_src_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstSRTSrc, gst_srt_src,
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GST_TYPE_PUSH_SRC,
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G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_srt_src_uri_handler_init)
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GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "srtsrc", 0, "SRT Source"));
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static void
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gst_srt_src_caller_added_cb (int sock, GSocketAddress * addr,
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GstSRTObject * srtobject)
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{
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g_signal_emit (srtobject->element, signals[SIG_CALLER_ADDED], 0, sock, addr);
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}
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static void
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gst_srt_src_caller_removed_cb (int sock, GSocketAddress * addr,
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GstSRTObject * srtobject)
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{
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g_signal_emit (srtobject->element, signals[SIG_CALLER_REMOVED], 0, sock,
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addr);
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}
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static gboolean
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gst_srt_src_start (GstBaseSrc * bsrc)
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{
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GstSRTSrc *self = GST_SRT_SRC (bsrc);
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GError *error = NULL;
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gboolean ret = FALSE;
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GstSRTConnectionMode connection_mode = GST_SRT_CONNECTION_MODE_NONE;
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gst_structure_get_enum (self->srtobject->parameters, "mode",
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GST_TYPE_SRT_CONNECTION_MODE, (gint *) & connection_mode);
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if (connection_mode == GST_SRT_CONNECTION_MODE_LISTENER) {
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ret =
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gst_srt_object_open_full (self->srtobject, gst_srt_src_caller_added_cb,
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gst_srt_src_caller_removed_cb, self->cancellable, &error);
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} else {
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ret = gst_srt_object_open (self->srtobject, self->cancellable, &error);
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}
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if (!ret) {
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/* ensure error is posted since state change will fail */
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GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL),
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("Failed to open SRT: %s", error->message));
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g_clear_error (&error);
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}
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return ret;
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}
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static gboolean
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gst_srt_src_stop (GstBaseSrc * bsrc)
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{
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GstSRTSrc *self = GST_SRT_SRC (bsrc);
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gst_srt_object_close (self->srtobject);
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return TRUE;
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}
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static GstFlowReturn
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gst_srt_src_fill (GstPushSrc * src, GstBuffer * outbuf)
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{
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GstSRTSrc *self = GST_SRT_SRC (src);
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GstFlowReturn ret = GST_FLOW_OK;
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GstMapInfo info;
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GError *err = NULL;
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gssize recv_len;
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if (g_cancellable_is_cancelled (self->cancellable)) {
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ret = GST_FLOW_FLUSHING;
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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 = gst_srt_object_read (self->srtobject, info.data,
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gst_buffer_get_size (outbuf), self->cancellable, &err);
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gst_buffer_unmap (outbuf, &info);
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if (g_cancellable_is_cancelled (self->cancellable)) {
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ret = GST_FLOW_FLUSHING;
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goto out;
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}
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if (recv_len < 0) {
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GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL), ("%s", err->message));
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ret = GST_FLOW_ERROR;
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g_clear_error (&err);
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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_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 void
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gst_srt_src_init (GstSRTSrc * self)
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{
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self->srtobject = gst_srt_object_new (GST_ELEMENT (self));
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self->cancellable = g_cancellable_new ();
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gst_base_src_set_format (GST_BASE_SRC (self), GST_FORMAT_TIME);
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gst_base_src_set_live (GST_BASE_SRC (self), TRUE);
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gst_base_src_set_do_timestamp (GST_BASE_SRC (self), TRUE);
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gst_srt_object_set_uri (self->srtobject, GST_SRT_DEFAULT_URI, NULL);
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}
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static void
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gst_srt_src_finalize (GObject * object)
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{
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GstSRTSrc *self = GST_SRT_SRC (object);
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g_clear_object (&self->cancellable);
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gst_srt_object_destroy (self->srtobject);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static gboolean
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gst_srt_src_unlock (GstBaseSrc * bsrc)
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{
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GstSRTSrc *self = GST_SRT_SRC (bsrc);
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gst_srt_object_wakeup (self->srtobject, self->cancellable);
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return TRUE;
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}
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static gboolean
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gst_srt_src_unlock_stop (GstBaseSrc * bsrc)
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{
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GstSRTSrc *self = GST_SRT_SRC (bsrc);
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g_cancellable_reset (self->cancellable);
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return TRUE;
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}
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static void
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gst_srt_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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GstSRTSrc *self = GST_SRT_SRC (object);
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if (!gst_srt_object_set_property_helper (self->srtobject, prop_id, value,
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pspec)) {
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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}
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}
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static void
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gst_srt_src_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec)
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{
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GstSRTSrc *self = GST_SRT_SRC (object);
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if (!gst_srt_object_get_property_helper (self->srtobject, prop_id, value,
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pspec)) {
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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}
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}
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static void
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gst_srt_src_class_init (GstSRTSrcClass * 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_src_set_property;
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gobject_class->get_property = gst_srt_src_get_property;
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gobject_class->finalize = gst_srt_src_finalize;
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/**
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* GstSRTSrc::caller-added:
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* @gstsrtsink: the srtsink element that emitted this signal
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* @sock: the client socket descriptor that was added to srtsink
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* @addr: the #GSocketAddress that describes the @sock
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*
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* The given socket descriptor was added to srtsink.
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*/
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signals[SIG_CALLER_ADDED] =
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g_signal_new ("caller-added", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstSRTSrcClass, caller_added),
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NULL, NULL, NULL, G_TYPE_NONE, 2, G_TYPE_INT, G_TYPE_SOCKET_ADDRESS);
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/**
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* GstSRTSrc::caller-removed:
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* @gstsrtsink: the srtsink element that emitted this signal
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* @sock: the client socket descriptor that was added to srtsink
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* @addr: the #GSocketAddress that describes the @sock
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*
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* The given socket descriptor was removed from srtsink.
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*/
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signals[SIG_CALLER_REMOVED] =
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g_signal_new ("caller-removed", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstSRTSrcClass,
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caller_added), NULL, NULL, NULL, G_TYPE_NONE,
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2, G_TYPE_INT, G_TYPE_SOCKET_ADDRESS);
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gst_srt_object_install_properties_helper (gobject_class);
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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 source", "Source/Network",
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"Receive data over the network via SRT",
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"Justin Kim <justin.joy.9to5@gmail.com>");
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gstbasesrc_class->start = GST_DEBUG_FUNCPTR (gst_srt_src_start);
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gstbasesrc_class->stop = GST_DEBUG_FUNCPTR (gst_srt_src_stop);
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gstbasesrc_class->unlock = GST_DEBUG_FUNCPTR (gst_srt_src_unlock);
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gstbasesrc_class->unlock_stop = GST_DEBUG_FUNCPTR (gst_srt_src_unlock_stop);
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gstpushsrc_class->fill = GST_DEBUG_FUNCPTR (gst_srt_src_fill);
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}
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static GstURIType
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gst_srt_src_uri_get_type (GType type)
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{
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return GST_URI_SRC;
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}
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static const gchar *const *
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gst_srt_src_uri_get_protocols (GType type)
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{
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static const gchar *protocols[] = { GST_SRT_DEFAULT_URI_SCHEME, NULL };
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return protocols;
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}
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static gchar *
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gst_srt_src_uri_get_uri (GstURIHandler * handler)
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{
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gchar *uri_str;
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GstSRTSrc *self = GST_SRT_SRC (handler);
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GST_OBJECT_LOCK (self);
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uri_str = gst_uri_to_string (self->srtobject->uri);
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GST_OBJECT_UNLOCK (self);
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return uri_str;
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}
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static gboolean
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gst_srt_src_uri_set_uri (GstURIHandler * handler,
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const gchar * uri, GError ** error)
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{
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GstSRTSrc *self = GST_SRT_SRC (handler);
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return gst_srt_object_set_uri (self->srtobject, uri, error);
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}
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static void
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gst_srt_src_uri_handler_init (gpointer g_iface, gpointer iface_data)
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{
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GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
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iface->get_type = gst_srt_src_uri_get_type;
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iface->get_protocols = gst_srt_src_uri_get_protocols;
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iface->get_uri = gst_srt_src_uri_get_uri;
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iface->set_uri = gst_srt_src_uri_set_uri;
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}
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