mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-24 10:41:04 +00:00
4eeff95f92
`delay` should be a GstClockTimeDiff since SRT time is int64_t. All values are in local time so we should never see a srctime that's in the future. If we do, clamp the delay to 0 and warn about it. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1674>
431 lines
12 KiB
C
431 lines
12 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-srtsrc
|
|
* @title: srtsrc
|
|
*
|
|
* srtsrc is a network source that reads [SRT](http://www.srtalliance.org/)
|
|
* packets from the network.
|
|
*
|
|
* ## 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,
|
|
|
|
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);
|
|
|
|
#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
|
|
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;
|
|
|
|
/**
|
|
* GstSRTSrc::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 (GstSRTSrcClass, caller_added),
|
|
NULL, NULL, NULL, G_TYPE_NONE, 2, G_TYPE_INT, G_TYPE_SOCKET_ADDRESS);
|
|
|
|
/**
|
|
* GstSRTSrc::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 (GstSRTSrcClass,
|
|
caller_added), NULL, NULL, NULL, 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, &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;
|
|
}
|