gstreamer/ext/rtmp/gstrtmpsink.c
Tim-Philipp Müller 2a78a3010d Merge commit '26d6add9457f00ce8ec13844368466f0e3816e5d' into 0.11
Conflicts:
	ext/rtmp/gstrtmpsink.c
2011-11-28 23:20:02 +00:00

375 lines
10 KiB
C

/*
* GStreamer
* Copyright (C) 2010 Jan Schmidt <thaytan@noraisin.net>
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-rtmpsink
*
* This element delivers data to a streaming server via RTMP. It uses
* librtmp, and supports any protocols/urls that librtmp supports.
* The URL/location can contain extra connection or session parameters
* for librtmp, such as 'flashver=version'. See the librtmp documentation
* for more detail
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch -v videotestsrc ! ffenc_flv ! flvmux ! rtmpsink location='rtmp://localhost/path/to/stream live=1'
* ]| Encode a test video stream to FLV video format and stream it via RTMP.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include "gstrtmpsink.h"
#ifdef G_OS_WIN32
#include <winsock2.h>
#endif
GST_DEBUG_CATEGORY_STATIC (gst_rtmp_sink_debug);
#define GST_CAT_DEFAULT gst_rtmp_sink_debug
/* Filter signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_LOCATION
};
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("video/x-flv")
);
static void gst_rtmp_sink_uri_handler_init (gpointer g_iface,
gpointer iface_data);
static void gst_rtmp_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rtmp_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_rtmp_sink_finalize (GObject * object);
static gboolean gst_rtmp_sink_stop (GstBaseSink * sink);
static gboolean gst_rtmp_sink_start (GstBaseSink * sink);
static GstFlowReturn gst_rtmp_sink_render (GstBaseSink * sink, GstBuffer * buf);
#define gst_rtmp_sink_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstRTMPSink, gst_rtmp_sink, GST_TYPE_BASE_SINK,
G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER,
gst_rtmp_sink_uri_handler_init));
/* initialize the plugin's class */
static void
gst_rtmp_sink_class_init (GstRTMPSinkClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSinkClass *gstbasesink_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesink_class = (GstBaseSinkClass *) klass;
gobject_class->finalize = gst_rtmp_sink_finalize;
gobject_class->set_property = gst_rtmp_sink_set_property;
gobject_class->get_property = gst_rtmp_sink_get_property;
gst_element_class_install_std_props (gstelement_class,
"location", PROP_LOCATION, G_PARAM_READWRITE, NULL);
gst_element_class_set_details_simple (gstelement_class,
"RTMP output sink",
"Sink/Network", "Sends FLV content to a server via RTMP",
"Jan Schmidt <thaytan@noraisin.net>");
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&sink_template));
gstbasesink_class->start = GST_DEBUG_FUNCPTR (gst_rtmp_sink_start);
gstbasesink_class->stop = GST_DEBUG_FUNCPTR (gst_rtmp_sink_stop);
gstbasesink_class->render = GST_DEBUG_FUNCPTR (gst_rtmp_sink_render);
GST_DEBUG_CATEGORY_INIT (gst_rtmp_sink_debug, "rtmpsink", 0,
"RTMP server element");
}
/* initialize the new element
* initialize instance structure
*/
static void
gst_rtmp_sink_init (GstRTMPSink * sink)
{
#ifdef G_OS_WIN32
WSADATA wsa_data;
if (WSAStartup (MAKEWORD (2, 2), &wsa_data) != 0) {
GST_ERROR_OBJECT (sink, "WSAStartup failed: 0x%08x", WSAGetLastError ());
}
#endif
}
static void
gst_rtmp_sink_finalize (GObject * object)
{
#ifdef G_OS_WIN32
WSACleanup ();
#endif
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_rtmp_sink_start (GstBaseSink * basesink)
{
GstRTMPSink *sink = GST_RTMP_SINK (basesink);
if (!sink->uri) {
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
("Please set URI for RTMP output"), ("No URI set before starting"));
return FALSE;
}
sink->rtmp_uri = g_strdup (sink->uri);
sink->rtmp = RTMP_Alloc ();
RTMP_Init (sink->rtmp);
if (!RTMP_SetupURL (sink->rtmp, sink->rtmp_uri)) {
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE, (NULL),
("Failed to setup URL '%s'", sink->uri));
RTMP_Free (sink->rtmp);
sink->rtmp = NULL;
g_free (sink->rtmp_uri);
sink->rtmp_uri = NULL;
return FALSE;
}
GST_DEBUG_OBJECT (sink, "Created RTMP object");
/* Mark this as an output connection */
RTMP_EnableWrite (sink->rtmp);
sink->first = TRUE;
return TRUE;
}
static gboolean
gst_rtmp_sink_stop (GstBaseSink * basesink)
{
GstRTMPSink *sink = GST_RTMP_SINK (basesink);
gst_buffer_replace (&sink->cache, NULL);
if (sink->rtmp) {
RTMP_Close (sink->rtmp);
RTMP_Free (sink->rtmp);
sink->rtmp = NULL;
}
if (sink->rtmp_uri) {
g_free (sink->rtmp_uri);
sink->rtmp_uri = NULL;
}
return TRUE;
}
static GstFlowReturn
gst_rtmp_sink_render (GstBaseSink * bsink, GstBuffer * buf)
{
GstRTMPSink *sink = GST_RTMP_SINK (bsink);
GstBuffer *reffed_buf = NULL;
guint8 *data;
gsize size;
if (sink->first) {
/* open the connection */
if (!RTMP_IsConnected (sink->rtmp)) {
if (!RTMP_Connect (sink->rtmp, NULL)
|| !RTMP_ConnectStream (sink->rtmp, 0)) {
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE, (NULL),
("Could not connect to RTMP stream \"%s\" for writing", sink->uri));
RTMP_Free (sink->rtmp);
sink->rtmp = NULL;
g_free (sink->rtmp_uri);
sink->rtmp_uri = NULL;
return GST_FLOW_ERROR;
}
GST_DEBUG_OBJECT (sink, "Opened connection to %s", sink->rtmp_uri);
}
/* FIXME: Parse the first buffer and see if it contains a header plus a packet instead
* of just assuming it's only the header */
GST_LOG_OBJECT (sink, "Caching first buffer of size %" G_GSIZE_FORMAT
" for concatenation", gst_buffer_get_size (buf));
gst_buffer_replace (&sink->cache, buf);
sink->first = FALSE;
return GST_FLOW_OK;
}
if (sink->cache) {
GST_LOG_OBJECT (sink, "Joining 2nd buffer of size %" G_GSIZE_FORMAT
" to cached buf", gst_buffer_get_size (buf));
gst_buffer_ref (buf);
reffed_buf = buf = gst_buffer_join (sink->cache, buf);
sink->cache = NULL;
}
GST_LOG_OBJECT (sink, "Sending %" G_GSIZE_FORMAT " bytes to RTMP server",
gst_buffer_get_size (buf));
data = gst_buffer_map (buf, &size, NULL, GST_MAP_READ);
if (!RTMP_Write (sink->rtmp, (char *) data, size))
goto write_failed;
gst_buffer_unmap (buf, data, size);
if (reffed_buf)
gst_buffer_unref (reffed_buf);
return GST_FLOW_OK;
/* ERRORS */
write_failed:
{
GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL), ("Failed to write data"));
gst_buffer_unmap (buf, data, size);
if (reffed_buf)
gst_buffer_unref (reffed_buf);
return GST_FLOW_ERROR;
}
}
/*
* URI interface support.
*/
static GstURIType
gst_rtmp_sink_uri_get_type (GType type)
{
return GST_URI_SINK;
}
static const gchar *const *
gst_rtmp_sink_uri_get_protocols (GType type)
{
static const gchar *protocols[] =
{ "rtmp", "rtmpt", "rtmps", "rtmpe", "rtmfp", "rtmpte", "rtmpts", NULL };
return protocols;
}
static gchar *
gst_rtmp_sink_uri_get_uri (GstURIHandler * handler)
{
GstRTMPSink *sink = GST_RTMP_SINK (handler);
/* FIXME: make thread-safe */
return g_strdup (sink->uri);
}
static gboolean
gst_rtmp_sink_uri_set_uri (GstURIHandler * handler, const gchar * uri,
GError ** error)
{
GstRTMPSink *sink = GST_RTMP_SINK (handler);
if (GST_STATE (sink) >= GST_STATE_PAUSED) {
g_set_error (error, GST_URI_ERROR, GST_URI_ERROR_BAD_STATE,
"Changing the URI on rtmpsrc when it is running is not supported");
return FALSE;
}
g_free (sink->uri);
sink->uri = NULL;
if (uri != NULL) {
int protocol;
AVal host;
unsigned int port;
AVal playpath, app;
if (!RTMP_ParseURL (uri, &protocol, &host, &port, &playpath, &app) ||
!host.av_len || !playpath.av_len) {
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
("Failed to parse URI %s", uri), (NULL));
g_set_error (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
"Could not parse RTMP URI");
return FALSE;
}
sink->uri = g_strdup (uri);
}
GST_DEBUG_OBJECT (sink, "Changed URI to %s", GST_STR_NULL (uri));
return TRUE;
}
static void
gst_rtmp_sink_uri_handler_init (gpointer g_iface, gpointer iface_data)
{
GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
iface->get_type = gst_rtmp_sink_uri_get_type;
iface->get_protocols = gst_rtmp_sink_uri_get_protocols;
iface->get_uri = gst_rtmp_sink_uri_get_uri;
iface->set_uri = gst_rtmp_sink_uri_set_uri;
}
static void
gst_rtmp_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRTMPSink *sink = GST_RTMP_SINK (object);
switch (prop_id) {
case PROP_LOCATION:
gst_rtmp_sink_uri_set_uri (GST_URI_HANDLER (sink),
g_value_get_string (value), NULL);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtmp_sink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRTMPSink *sink = GST_RTMP_SINK (object);
switch (prop_id) {
case PROP_LOCATION:
g_value_set_string (value, sink->uri);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}