mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-10-05 18:22:23 +00:00
0a350c610d
We have srt{client,server}{src,sink} elements in accordance to the norm of the connection oriented protocols. However, SRT connection mode can be changed by uri parameters so it requires an integrated element to handle the parameters. fix: #740
391 lines
11 KiB
C
391 lines
11 KiB
C
/* GStreamer
|
|
* Copyright (C) 2018, Collabora Ltd.
|
|
* Copyright (C) 2018, SK Telecom, Co., Ltd.
|
|
* Author: Jeongseok Kim <jeongseok.kim@sk.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-srtsink
|
|
* @title: srtsink
|
|
*
|
|
* srtsink is a network sink that sends <ulink url="http://www.srtalliance.org/">SRT</ulink>
|
|
* packets to the network.
|
|
*
|
|
* <refsect2>
|
|
* <title>Examples</title>
|
|
* |[
|
|
* gst-launch-1.0 -v audiotestsrc ! srtsink uri://host?mode=caller
|
|
* ]| This pipeline shows how to serve SRT packets through the default port.
|
|
*
|
|
* |[
|
|
* gst-launch-1.0 -v audiotestsrc ! srtsink uri://host:port?mode=listener
|
|
* ]| This pipeline shows how to wait SRT callers.
|
|
* </refsect2>
|
|
*
|
|
*/
|
|
|
|
#ifdef HAVE_CONFIG_H
|
|
#include <config.h>
|
|
#endif
|
|
|
|
#include "gstsrtsink.h"
|
|
|
|
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
|
|
GST_PAD_SINK,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS_ANY);
|
|
|
|
#define GST_CAT_DEFAULT gst_debug_srt_sink
|
|
GST_DEBUG_CATEGORY (GST_CAT_DEFAULT);
|
|
|
|
enum
|
|
{
|
|
SIG_CALLER_ADDED,
|
|
SIG_CALLER_REMOVED,
|
|
|
|
LAST_SIGNAL
|
|
};
|
|
|
|
static guint signals[LAST_SIGNAL] = { 0 };
|
|
|
|
static void gst_srt_sink_uri_handler_init (gpointer g_iface,
|
|
gpointer iface_data);
|
|
static gchar *gst_srt_sink_uri_get_uri (GstURIHandler * handler);
|
|
static gboolean gst_srt_sink_uri_set_uri (GstURIHandler * handler,
|
|
const gchar * uri, GError ** error);
|
|
|
|
#define gst_srt_sink_parent_class parent_class
|
|
G_DEFINE_TYPE_WITH_CODE (GstSRTSink, gst_srt_sink,
|
|
GST_TYPE_BASE_SINK,
|
|
G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_srt_sink_uri_handler_init)
|
|
GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "srtsink", 0, "SRT Sink"));
|
|
|
|
static void
|
|
gst_srt_sink_set_property (GObject * object,
|
|
guint prop_id, const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstSRTSink *self = GST_SRT_SINK (object);
|
|
|
|
if (!gst_srt_object_set_property_helper (self->srtobject, prop_id, value,
|
|
pspec)) {
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_srt_sink_get_property (GObject * object,
|
|
guint prop_id, GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstSRTSink *self = GST_SRT_SINK (object);
|
|
|
|
if (!gst_srt_object_get_property_helper (self->srtobject, prop_id, value,
|
|
pspec)) {
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_srt_sink_finalize (GObject * object)
|
|
{
|
|
GstSRTSink *self = GST_SRT_SINK (object);
|
|
|
|
g_clear_object (&self->cancellable);
|
|
gst_srt_object_destroy (self->srtobject);
|
|
|
|
G_OBJECT_CLASS (parent_class)->finalize (object);
|
|
}
|
|
|
|
static void
|
|
gst_srt_sink_init (GstSRTSink * self)
|
|
{
|
|
self->srtobject = gst_srt_object_new (GST_ELEMENT (self));
|
|
self->cancellable = g_cancellable_new ();
|
|
|
|
gst_srt_object_set_uri (self->srtobject, GST_SRT_DEFAULT_URI, NULL);
|
|
}
|
|
|
|
static void
|
|
gst_srt_sink_caller_added_cb (int sock, GSocketAddress * addr,
|
|
GstSRTObject * srtobject)
|
|
{
|
|
g_signal_emit (srtobject->element, signals[SIG_CALLER_ADDED], 0, sock, addr);
|
|
}
|
|
|
|
static void
|
|
gst_srt_sink_caller_removed_cb (int sock, GSocketAddress * addr,
|
|
GstSRTObject * srtobject)
|
|
{
|
|
g_signal_emit (srtobject->element, signals[SIG_CALLER_REMOVED], 0, sock,
|
|
addr);
|
|
}
|
|
|
|
static gboolean
|
|
gst_srt_sink_start (GstBaseSink * bsink)
|
|
{
|
|
GstSRTSink *self = GST_SRT_SINK (bsink);
|
|
GstSRTConnectionMode connection_mode = GST_SRT_CONNECTION_MODE_NONE;
|
|
|
|
GError *error = NULL;
|
|
gboolean ret = FALSE;
|
|
|
|
gst_structure_get_enum (self->srtobject->parameters, "mode",
|
|
GST_TYPE_SRT_CONNECTION_MODE, (gint *) & connection_mode);
|
|
|
|
if (connection_mode == GST_SRT_CONNECTION_MODE_LISTENER) {
|
|
ret =
|
|
gst_srt_object_open_full (self->srtobject, gst_srt_sink_caller_added_cb,
|
|
gst_srt_sink_caller_removed_cb, self->cancellable, &error);
|
|
} else {
|
|
ret = gst_srt_object_open (self->srtobject, self->cancellable, &error);
|
|
}
|
|
|
|
if (!ret) {
|
|
GST_WARNING_OBJECT (self, "Failed to open SRT: %s", error->message);
|
|
g_clear_error (&error);
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_srt_sink_stop (GstBaseSink * bsink)
|
|
{
|
|
GstSRTSink *self = GST_SRT_SINK (bsink);
|
|
|
|
gst_srt_object_close (self->srtobject);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_srt_sink_render (GstBaseSink * sink, GstBuffer * buffer)
|
|
{
|
|
GstSRTSink *self = GST_SRT_SINK (sink);
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
GstMapInfo info;
|
|
GError *error = NULL;
|
|
|
|
if (g_cancellable_is_cancelled (self->cancellable)) {
|
|
ret = GST_FLOW_FLUSHING;
|
|
}
|
|
|
|
if (self->headers && GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_HEADER)) {
|
|
GST_DEBUG_OBJECT (self, "Have streamheaders,"
|
|
" ignoring header %" GST_PTR_FORMAT, buffer);
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
if (!gst_buffer_map (buffer, &info, GST_MAP_READ)) {
|
|
GST_ELEMENT_ERROR (self, RESOURCE, READ,
|
|
("Could not map the input stream"), (NULL));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
|
|
if (gst_srt_object_write (self->srtobject, self->headers, &info,
|
|
self->cancellable, &error) < 0) {
|
|
ret = GST_FLOW_ERROR;
|
|
}
|
|
|
|
gst_buffer_unmap (buffer, &info);
|
|
|
|
GST_TRACE_OBJECT (self, "sending buffer %p, offset %"
|
|
G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT
|
|
", timestamp %" GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT
|
|
", size %" G_GSIZE_FORMAT,
|
|
buffer, GST_BUFFER_OFFSET (buffer),
|
|
GST_BUFFER_OFFSET_END (buffer),
|
|
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
|
|
GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)),
|
|
gst_buffer_get_size (buffer));
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_srt_sink_unlock (GstBaseSink * bsink)
|
|
{
|
|
GstSRTSink *self = GST_SRT_SINK (bsink);
|
|
|
|
g_cancellable_cancel (self->cancellable);
|
|
gst_srt_object_wakeup (self->srtobject);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_srt_sink_unlock_stop (GstBaseSink * bsink)
|
|
{
|
|
GstSRTSink *self = GST_SRT_SINK (bsink);
|
|
|
|
g_cancellable_reset (self->cancellable);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_srt_sink_set_caps (GstBaseSink * bsink, GstCaps * caps)
|
|
{
|
|
GstSRTSink *self = GST_SRT_SINK (bsink);
|
|
GstStructure *s;
|
|
const GValue *streamheader;
|
|
|
|
GST_DEBUG_OBJECT (self, "setcaps %" GST_PTR_FORMAT, caps);
|
|
|
|
g_clear_pointer (&self->headers, gst_buffer_list_unref);
|
|
|
|
s = gst_caps_get_structure (caps, 0);
|
|
streamheader = gst_structure_get_value (s, "streamheader");
|
|
|
|
if (!streamheader) {
|
|
GST_DEBUG_OBJECT (self, "'streamheader' field not present");
|
|
} else if (GST_VALUE_HOLDS_BUFFER (streamheader)) {
|
|
GST_DEBUG_OBJECT (self, "'streamheader' field holds buffer");
|
|
self->headers = gst_buffer_list_new_sized (1);
|
|
gst_buffer_list_add (self->headers, g_value_dup_boxed (streamheader));
|
|
} else if (GST_VALUE_HOLDS_ARRAY (streamheader)) {
|
|
guint i, size;
|
|
|
|
GST_DEBUG_OBJECT (self, "'streamheader' field holds array");
|
|
|
|
size = gst_value_array_get_size (streamheader);
|
|
self->headers = gst_buffer_list_new_sized (size);
|
|
|
|
for (i = 0; i < size; i++) {
|
|
const GValue *v = gst_value_array_get_value (streamheader, i);
|
|
if (!GST_VALUE_HOLDS_BUFFER (v)) {
|
|
GST_ERROR_OBJECT (self, "'streamheader' item of unexpected type '%s'",
|
|
G_VALUE_TYPE_NAME (v));
|
|
return FALSE;
|
|
}
|
|
|
|
gst_buffer_list_add (self->headers, g_value_dup_boxed (v));
|
|
}
|
|
} else {
|
|
GST_ERROR_OBJECT (self, "'streamheader' field has unexpected type '%s'",
|
|
G_VALUE_TYPE_NAME (streamheader));
|
|
return FALSE;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (self, "Collected streamheaders: %u buffers",
|
|
self->headers ? gst_buffer_list_length (self->headers) : 0);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_srt_sink_class_init (GstSRTSinkClass * klass)
|
|
{
|
|
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
|
|
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
|
|
GstBaseSinkClass *gstbasesink_class = GST_BASE_SINK_CLASS (klass);
|
|
|
|
gobject_class->set_property = gst_srt_sink_set_property;
|
|
gobject_class->get_property = gst_srt_sink_get_property;
|
|
gobject_class->finalize = gst_srt_sink_finalize;
|
|
|
|
/**
|
|
* GstSRTSink::caller-added:
|
|
* @gstsrtsink: the srtsink element that emitted this signal
|
|
* @sock: the client socket descriptor that was added to srtsink
|
|
* @addr: the #GSocketAddress that describes the @sock
|
|
*
|
|
* The given socket descriptor was added to srtsink.
|
|
*/
|
|
signals[SIG_CALLER_ADDED] =
|
|
g_signal_new ("caller-added", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstSRTSinkClass, caller_added),
|
|
NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE,
|
|
2, G_TYPE_INT, G_TYPE_SOCKET_ADDRESS);
|
|
|
|
/**
|
|
* GstSRTSink::caller-removed:
|
|
* @gstsrtsink: the srtsink element that emitted this signal
|
|
* @sock: the client socket descriptor that was added to srtsink
|
|
* @addr: the #GSocketAddress that describes the @sock
|
|
*
|
|
* The given socket descriptor was removed from srtsink.
|
|
*/
|
|
signals[SIG_CALLER_REMOVED] =
|
|
g_signal_new ("caller-removed", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstSRTSinkClass,
|
|
caller_added), NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE,
|
|
2, G_TYPE_INT, G_TYPE_SOCKET_ADDRESS);
|
|
|
|
gst_srt_object_install_properties_helper (gobject_class);
|
|
|
|
gst_element_class_add_static_pad_template (gstelement_class, &sink_template);
|
|
gst_element_class_set_metadata (gstelement_class,
|
|
"SRT sink", "Sink/Network",
|
|
"Send data over the network via SRT",
|
|
"Justin Kim <justin.joy.9to5@gmail.com>");
|
|
|
|
gstbasesink_class->start = GST_DEBUG_FUNCPTR (gst_srt_sink_start);
|
|
gstbasesink_class->stop = GST_DEBUG_FUNCPTR (gst_srt_sink_stop);
|
|
gstbasesink_class->render = GST_DEBUG_FUNCPTR (gst_srt_sink_render);
|
|
gstbasesink_class->unlock = GST_DEBUG_FUNCPTR (gst_srt_sink_unlock);
|
|
gstbasesink_class->unlock_stop = GST_DEBUG_FUNCPTR (gst_srt_sink_unlock_stop);
|
|
gstbasesink_class->set_caps = GST_DEBUG_FUNCPTR (gst_srt_sink_set_caps);
|
|
|
|
}
|
|
|
|
static GstURIType
|
|
gst_srt_sink_uri_get_type (GType type)
|
|
{
|
|
return GST_URI_SINK;
|
|
}
|
|
|
|
static const gchar *const *
|
|
gst_srt_sink_uri_get_protocols (GType type)
|
|
{
|
|
static const gchar *protocols[] = { GST_SRT_DEFAULT_URI_SCHEME, NULL };
|
|
|
|
return protocols;
|
|
}
|
|
|
|
static gchar *
|
|
gst_srt_sink_uri_get_uri (GstURIHandler * handler)
|
|
{
|
|
gchar *uri_str;
|
|
GstSRTSink *self = GST_SRT_SINK (handler);
|
|
|
|
GST_OBJECT_LOCK (self);
|
|
uri_str = gst_uri_to_string (self->srtobject->uri);
|
|
GST_OBJECT_UNLOCK (self);
|
|
|
|
return uri_str;
|
|
}
|
|
|
|
static gboolean
|
|
gst_srt_sink_uri_set_uri (GstURIHandler * handler,
|
|
const gchar * uri, GError ** error)
|
|
{
|
|
GstSRTSink *self = GST_SRT_SINK (handler);
|
|
|
|
return gst_srt_object_set_uri (self->srtobject, uri, error);
|
|
}
|
|
|
|
static void
|
|
gst_srt_sink_uri_handler_init (gpointer g_iface, gpointer iface_data)
|
|
{
|
|
GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
|
|
|
|
iface->get_type = gst_srt_sink_uri_get_type;
|
|
iface->get_protocols = gst_srt_sink_uri_get_protocols;
|
|
iface->get_uri = gst_srt_sink_uri_get_uri;
|
|
iface->set_uri = gst_srt_sink_uri_set_uri;
|
|
}
|