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