gstreamer/ext/srt/gstsrtsrc.c

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/* GStreamer
* Copyright (C) 2018, Collabora Ltd.
* Copyright (C) 2018, SK Telecom, Co., Ltd.
* Author: Jeongseok Kim <jeongseok.kim@sk.com>
2020-03-18 16:54:29 +00:00
*
* 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-srtsrc
* @title: srtsrc
*
* srtsrc is a network source that reads [SRT](http://www.srtalliance.org/)
* packets from the network.
*
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* ## Examples
* |[
* gst-launch-1.0 -v srtsrc uri="srt://127.0.0.1:7001" ! fakesink
* ]| This pipeline shows how to connect SRT server by setting #GstSRTSrc:uri property.
*
* |[
* gst-launch-1.0 -v srtsrc uri="srt://:7001?mode=listener" ! fakesink
* ]| This pipeline shows how to wait SRT connection by setting #GstSRTSrc:uri property.
*
* |[
* gst-launch-1.0 -v srtclientsrc uri="srt://192.168.1.10:7001?mode=rendez-vous" ! fakesink
* ]| This pipeline shows how to connect SRT server by setting #GstSRTSrc:uri property and using the rendez-vous mode.
*
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include "gstsrtsrc.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_src
GST_DEBUG_CATEGORY (GST_CAT_DEFAULT);
enum
{
SIG_CALLER_ADDED,
SIG_CALLER_REMOVED,
SIG_CALLER_REJECTED,
SIG_CALLER_CONNECTING,
LAST_SIGNAL
};
static guint signals[LAST_SIGNAL] = { 0 };
static void gst_srt_src_uri_handler_init (gpointer g_iface,
gpointer iface_data);
static gchar *gst_srt_src_uri_get_uri (GstURIHandler * handler);
static gboolean gst_srt_src_uri_set_uri (GstURIHandler * handler,
const gchar * uri, GError ** error);
static gboolean src_default_caller_connecting (GstSRTSrc * self,
GSocketAddress * addr, const gchar * username, gpointer data);
static gboolean src_authentication_accumulator (GSignalInvocationHint * ihint,
GValue * return_accu, const GValue * handler_return, gpointer data);
#define gst_srt_src_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstSRTSrc, gst_srt_src,
GST_TYPE_PUSH_SRC,
G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_srt_src_uri_handler_init)
GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "srtsrc", 0, "SRT Source"));
static gboolean
src_default_caller_connecting (GstSRTSrc * self,
GSocketAddress * addr, const gchar * stream_id, gpointer data)
{
/* Accept all connections. */
return TRUE;
}
static gboolean
src_authentication_accumulator (GSignalInvocationHint * ihint,
GValue * return_accu, const GValue * handler_return, gpointer data)
{
gboolean ret = g_value_get_boolean (handler_return);
/* Handlers return TRUE on authentication success and we want to stop on
* the first failure. */
g_value_set_boolean (return_accu, ret);
return ret;
}
static gboolean
gst_srt_src_start (GstBaseSrc * bsrc)
{
GstSRTSrc *self = GST_SRT_SRC (bsrc);
GError *error = NULL;
gboolean ret = FALSE;
GstSRTConnectionMode connection_mode = GST_SRT_CONNECTION_MODE_NONE;
gst_structure_get_enum (self->srtobject->parameters, "mode",
GST_TYPE_SRT_CONNECTION_MODE, (gint *) & connection_mode);
ret = gst_srt_object_open (self->srtobject, self->cancellable, &error);
if (!ret) {
/* ensure error is posted since state change will fail */
GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL),
("Failed to open SRT: %s", error->message));
g_clear_error (&error);
}
/* Reset expected pktseq */
self->next_pktseq = 0;
return ret;
}
static gboolean
gst_srt_src_stop (GstBaseSrc * bsrc)
{
GstSRTSrc *self = GST_SRT_SRC (bsrc);
gst_srt_object_close (self->srtobject);
return TRUE;
}
static GstFlowReturn
gst_srt_src_fill (GstPushSrc * src, GstBuffer * outbuf)
{
GstSRTSrc *self = GST_SRT_SRC (src);
GstFlowReturn ret = GST_FLOW_OK;
GstMapInfo info;
GError *err = NULL;
gssize recv_len;
GstClock *clock;
GstClockTime base_time;
GstClockTime capture_time;
GstClockTimeDiff delay;
int64_t srt_time;
SRT_MSGCTRL mctrl;
if (g_cancellable_is_cancelled (self->cancellable)) {
ret = GST_FLOW_FLUSHING;
}
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;
}
/* Get clock and values */
clock = gst_element_get_clock (GST_ELEMENT (src));
if (!clock) {
GST_DEBUG_OBJECT (src, "Clock missing, flushing");
return GST_FLOW_FLUSHING;
}
base_time = gst_element_get_base_time (GST_ELEMENT (src));
recv_len = gst_srt_object_read (self->srtobject, info.data,
gst_buffer_get_size (outbuf), self->cancellable, &err, &mctrl);
/* Capture clock values ASAP */
capture_time = gst_clock_get_time (clock);
#if SRT_VERSION_VALUE >= 0x10402
/* Use SRT clock value if available (SRT > 1.4.2) */
srt_time = srt_time_now ();
#else
/* Else use the unix epoch monotonic clock */
srt_time = g_get_real_time ();
#endif
gst_object_unref (clock);
gst_buffer_unmap (outbuf, &info);
GST_LOG_OBJECT (src,
"recv_len:%" G_GSIZE_FORMAT " pktseq:%d msgno:%d srctime:%"
G_GINT64_FORMAT, recv_len, mctrl.pktseq, mctrl.msgno, mctrl.srctime);
if (g_cancellable_is_cancelled (self->cancellable)) {
ret = GST_FLOW_FLUSHING;
goto out;
}
if (recv_len < 0) {
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL), ("%s", err->message));
ret = GST_FLOW_ERROR;
g_clear_error (&err);
goto out;
} else if (recv_len == 0) {
ret = GST_FLOW_EOS;
goto out;
}
/* Detect discontinuities */
if (mctrl.pktseq != self->next_pktseq) {
GST_WARNING_OBJECT (src, "discont detected %d (expected: %d)",
mctrl.pktseq, self->next_pktseq);
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
}
/* pktseq is a 31bit field */
self->next_pktseq = (mctrl.pktseq + 1) % G_MAXINT32;
/* 0 means we do not have a srctime */
if (mctrl.srctime != 0)
delay = (srt_time - mctrl.srctime) * GST_USECOND;
else
delay = 0;
GST_LOG_OBJECT (src, "delay: %" GST_STIME_FORMAT, GST_STIME_ARGS (delay));
if (delay < 0) {
GST_WARNING_OBJECT (src,
"Calculated SRT delay %" GST_STIME_FORMAT " is negative, clamping to 0",
GST_STIME_ARGS (delay));
delay = 0;
}
/* Subtract the base_time (since the pipeline started) ... */
if (capture_time > base_time)
capture_time -= base_time;
else
capture_time = 0;
/* And adjust by the delay */
if (capture_time > delay)
capture_time -= delay;
else
capture_time = 0;
GST_BUFFER_TIMESTAMP (outbuf) = capture_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 void
gst_srt_src_init (GstSRTSrc * self)
{
self->srtobject = gst_srt_object_new (GST_ELEMENT (self));
self->cancellable = g_cancellable_new ();
gst_base_src_set_format (GST_BASE_SRC (self), GST_FORMAT_TIME);
gst_base_src_set_live (GST_BASE_SRC (self), TRUE);
/* We do the timing ourselves */
gst_base_src_set_do_timestamp (GST_BASE_SRC (self), FALSE);
gst_srt_object_set_uri (self->srtobject, GST_SRT_DEFAULT_URI, NULL);
}
static void
gst_srt_src_finalize (GObject * object)
{
GstSRTSrc *self = GST_SRT_SRC (object);
g_clear_object (&self->cancellable);
gst_srt_object_destroy (self->srtobject);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_srt_src_unlock (GstBaseSrc * bsrc)
{
GstSRTSrc *self = GST_SRT_SRC (bsrc);
gst_srt_object_wakeup (self->srtobject, self->cancellable);
return TRUE;
}
static gboolean
gst_srt_src_unlock_stop (GstBaseSrc * bsrc)
{
GstSRTSrc *self = GST_SRT_SRC (bsrc);
g_cancellable_reset (self->cancellable);
return TRUE;
}
static void
gst_srt_src_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstSRTSrc *self = GST_SRT_SRC (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_src_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstSRTSrc *self = GST_SRT_SRC (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 gboolean
gst_srt_src_query (GstBaseSrc * basesrc, GstQuery * query)
{
GstSRTSrc *self = GST_SRT_SRC (basesrc);
if (GST_QUERY_TYPE (query) == GST_QUERY_LATENCY) {
gint latency;
if (!gst_structure_get_int (self->srtobject->parameters, "latency",
&latency))
latency = GST_SRT_DEFAULT_LATENCY;
gst_query_set_latency (query, TRUE, latency * GST_MSECOND,
latency * GST_MSECOND);
return TRUE;
} else {
return GST_BASE_SRC_CLASS (parent_class)->query (basesrc, query);
}
}
static void
gst_srt_src_class_init (GstSRTSrcClass * 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_src_set_property;
gobject_class->get_property = gst_srt_src_get_property;
gobject_class->finalize = gst_srt_src_finalize;
klass->caller_connecting = src_default_caller_connecting;
/**
* GstSRTSrc::caller-added:
* @gstsrtsrc: the srtsrc element that emitted this signal
* @sock: the client socket descriptor that was added to srtsrc
* @addr: the #GSocketAddress that describes the @sock
2020-03-18 16:54:29 +00:00
*
* 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 (GstSRTSrcClass, caller_added),
NULL, NULL, NULL, G_TYPE_NONE, 2, G_TYPE_INT, G_TYPE_SOCKET_ADDRESS);
/**
* GstSRTSrc::caller-removed:
* @gstsrtsrc: the srtsrc element that emitted this signal
* @sock: the client socket descriptor that was added to srtsrc
* @addr: the #GSocketAddress that describes the @sock
*
* The given socket descriptor was removed from srtsrc.
*/
signals[SIG_CALLER_REMOVED] =
g_signal_new ("caller-removed", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstSRTSrcClass,
caller_added), NULL, NULL, NULL, G_TYPE_NONE,
2, G_TYPE_INT, G_TYPE_SOCKET_ADDRESS);
/**
* GstSRTSrc::caller-rejected:
* @gstsrtsrc: the srtsrc element that emitted this signal
* @addr: the #GSocketAddress that describes the client socket
* @stream_id: the stream Id to which the caller wants to connect
*
* A caller's connection to srtsrc in listener mode has been rejected.
*
* Since: 1.20
*
*/
signals[SIG_CALLER_REJECTED] =
g_signal_new ("caller-rejected", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstSRTSrcClass, caller_rejected),
NULL, NULL, NULL, G_TYPE_NONE, 2, G_TYPE_SOCKET_ADDRESS, G_TYPE_STRING);
/**
* GstSRTSrc::caller-connecting:
* @gstsrtsrc: the srtsrc element that emitted this signal
* @addr: the #GSocketAddress that describes the client socket
* @stream_id: the stream Id to which the caller wants to connect
*
* Whether to accept or reject a caller's connection to srtsrc in listener mode.
* The Caller's connection is rejected if the callback returns FALSE, else
* the connection is accepeted.
*
* Since: 1.20
*
*/
signals[SIG_CALLER_CONNECTING] =
g_signal_new ("caller-connecting", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstSRTSrcClass, caller_connecting),
src_authentication_accumulator, NULL, NULL, G_TYPE_BOOLEAN,
2, G_TYPE_SOCKET_ADDRESS, G_TYPE_STRING);
gst_srt_object_install_properties_helper (gobject_class);
gst_element_class_add_static_pad_template (gstelement_class, &src_template);
gst_element_class_set_metadata (gstelement_class,
"SRT source", "Source/Network",
"Receive data over the network via SRT",
"Justin Kim <justin.joy.9to5@gmail.com>");
gstbasesrc_class->start = GST_DEBUG_FUNCPTR (gst_srt_src_start);
gstbasesrc_class->stop = GST_DEBUG_FUNCPTR (gst_srt_src_stop);
gstbasesrc_class->unlock = GST_DEBUG_FUNCPTR (gst_srt_src_unlock);
gstbasesrc_class->unlock_stop = GST_DEBUG_FUNCPTR (gst_srt_src_unlock_stop);
gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_srt_src_query);
gstpushsrc_class->fill = GST_DEBUG_FUNCPTR (gst_srt_src_fill);
}
static GstURIType
gst_srt_src_uri_get_type (GType type)
{
return GST_URI_SRC;
}
static const gchar *const *
gst_srt_src_uri_get_protocols (GType type)
{
static const gchar *protocols[] = { GST_SRT_DEFAULT_URI_SCHEME, NULL };
return protocols;
}
static gchar *
gst_srt_src_uri_get_uri (GstURIHandler * handler)
{
gchar *uri_str;
GstSRTSrc *self = GST_SRT_SRC (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_src_uri_set_uri (GstURIHandler * handler,
const gchar * uri, GError ** error)
{
GstSRTSrc *self = GST_SRT_SRC (handler);
gboolean ret;
GST_OBJECT_LOCK (self);
ret = gst_srt_object_set_uri (self->srtobject, uri, error);
GST_OBJECT_UNLOCK (self);
return ret;
}
static void
gst_srt_src_uri_handler_init (gpointer g_iface, gpointer iface_data)
{
GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
iface->get_type = gst_srt_src_uri_get_type;
iface->get_protocols = gst_srt_src_uri_get_protocols;
iface->get_uri = gst_srt_src_uri_get_uri;
iface->set_uri = gst_srt_src_uri_set_uri;
}