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Although RTMP_ConnectStream() was failed, librtmp's internal memory is not freed by RTMP_ConnectStream(), so RTMP_Close() should be called before RTMP_Free() https://bugzilla.gnome.org/show_bug.cgi?id=797058
490 lines
13 KiB
C
490 lines
13 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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-rtmpsink
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* @title: 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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* ## Example launch line
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* |[
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* gst-launch-1.0 -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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*
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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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#ifdef G_OS_WIN32
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#include <winsock2.h>
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#endif
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#include <stdlib.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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#define DEFAULT_LOCATION NULL
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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 void gst_rtmp_sink_finalize (GObject * object);
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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 gboolean gst_rtmp_sink_event (GstBaseSink * sink, GstEvent * event);
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static gboolean gst_rtmp_sink_setcaps (GstBaseSink * sink, GstCaps * caps);
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static GstFlowReturn gst_rtmp_sink_render (GstBaseSink * sink, GstBuffer * buf);
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#define gst_rtmp_sink_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstRTMPSink, gst_rtmp_sink, GST_TYPE_BASE_SINK,
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G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER,
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gst_rtmp_sink_uri_handler_init));
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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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GstElementClass *gstelement_class;
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GstBaseSinkClass *gstbasesink_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbasesink_class = (GstBaseSinkClass *) klass;
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gobject_class->finalize = gst_rtmp_sink_finalize;
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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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g_object_class_install_property (gobject_class, PROP_LOCATION,
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g_param_spec_string ("location", "RTMP Location", "RTMP url",
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DEFAULT_LOCATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gst_element_class_set_static_metadata (gstelement_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_static_pad_template (gstelement_class, &sink_template);
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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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gstbasesink_class->set_caps = GST_DEBUG_FUNCPTR (gst_rtmp_sink_setcaps);
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gstbasesink_class->event = GST_DEBUG_FUNCPTR (gst_rtmp_sink_event);
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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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/* 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)
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{
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#ifdef G_OS_WIN32
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WSADATA wsa_data;
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if (WSAStartup (MAKEWORD (2, 2), &wsa_data) != 0) {
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GST_ERROR_OBJECT (sink, "WSAStartup failed: 0x%08x", WSAGetLastError ());
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}
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#endif
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}
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static void
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gst_rtmp_sink_finalize (GObject * object)
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{
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GstRTMPSink *sink = GST_RTMP_SINK (object);
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#ifdef G_OS_WIN32
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WSACleanup ();
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#endif
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g_free (sink->uri);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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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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if (!sink->rtmp) {
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GST_ERROR_OBJECT (sink, "Could not allocate librtmp's RTMP context");
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goto error;
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}
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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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goto error;
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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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sink->first = TRUE;
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sink->have_write_error = FALSE;
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return TRUE;
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error:
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if (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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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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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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if (sink->header) {
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gst_buffer_unref (sink->header);
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sink->header = NULL;
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}
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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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gboolean need_unref = FALSE;
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GstMapInfo map = GST_MAP_INFO_INIT;
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if (sink->rtmp == NULL) {
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/* Do not crash */
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GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL), ("Failed to write data"));
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return GST_FLOW_ERROR;
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}
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/* Ignore buffers that are in the stream headers (caps) */
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if (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_HEADER)) {
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return GST_FLOW_OK;
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}
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if (sink->first) {
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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))
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goto connection_failed;
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if (!RTMP_ConnectStream (sink->rtmp, 0)) {
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RTMP_Close (sink->rtmp);
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goto connection_failed;
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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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/* Prepend the header from the caps to the first non header buffer */
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if (sink->header) {
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buf = gst_buffer_append (gst_buffer_ref (sink->header),
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gst_buffer_ref (buf));
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need_unref = TRUE;
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}
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sink->first = FALSE;
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}
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if (sink->have_write_error)
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goto write_failed;
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GST_LOG_OBJECT (sink, "Sending %" G_GSIZE_FORMAT " bytes to RTMP server",
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gst_buffer_get_size (buf));
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gst_buffer_map (buf, &map, GST_MAP_READ);
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if (RTMP_Write (sink->rtmp, (char *) map.data, map.size) <= 0)
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goto write_failed;
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gst_buffer_unmap (buf, &map);
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if (need_unref)
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gst_buffer_unref (buf);
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return GST_FLOW_OK;
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/* ERRORS */
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write_failed:
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{
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GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL), ("Failed to write data"));
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gst_buffer_unmap (buf, &map);
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if (need_unref)
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gst_buffer_unref (buf);
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sink->have_write_error = TRUE;
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return GST_FLOW_ERROR;
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}
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connection_failed:
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{
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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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sink->have_write_error = TRUE;
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return GST_FLOW_ERROR;
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}
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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 (GType type)
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{
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return GST_URI_SINK;
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}
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static const gchar *const *
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gst_rtmp_sink_uri_get_protocols (GType type)
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{
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static const gchar *protocols[] =
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{ "rtmp", "rtmpt", "rtmps", "rtmpe", "rtmfp", "rtmpte", "rtmpts", NULL };
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return protocols;
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}
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static 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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/* FIXME: make thread-safe */
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return g_strdup (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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GError ** error)
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{
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GstRTMPSink *sink = GST_RTMP_SINK (handler);
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gboolean ret = TRUE;
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if (GST_STATE (sink) >= GST_STATE_PAUSED) {
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g_set_error (error, GST_URI_ERROR, GST_URI_ERROR_BAD_STATE,
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"Changing the URI on rtmpsink when it is running is not supported");
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return FALSE;
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}
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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) {
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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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g_set_error (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
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"Could not parse RTMP URI");
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ret = FALSE;
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} else {
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sink->uri = g_strdup (uri);
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}
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if (playpath.av_val)
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free (playpath.av_val);
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}
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if (ret) {
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sink->have_write_error = FALSE;
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GST_DEBUG_OBJECT (sink, "Changed URI to %s", GST_STR_NULL (uri));
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}
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return ret;
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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), NULL);
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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 gboolean
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gst_rtmp_sink_setcaps (GstBaseSink * sink, GstCaps * caps)
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{
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GstRTMPSink *rtmpsink = GST_RTMP_SINK (sink);
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GstStructure *s;
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const GValue *sh;
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GST_DEBUG_OBJECT (sink, "caps set to %" GST_PTR_FORMAT, caps);
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/* Clear our current header buffer */
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if (rtmpsink->header) {
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gst_buffer_unref (rtmpsink->header);
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rtmpsink->header = NULL;
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}
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s = gst_caps_get_structure (caps, 0);
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sh = gst_structure_get_value (s, "streamheader");
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if (sh == NULL)
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goto out;
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if (GST_VALUE_HOLDS_BUFFER (sh)) {
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rtmpsink->header = gst_buffer_ref (gst_value_get_buffer (sh));
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} else if (GST_VALUE_HOLDS_ARRAY (sh)) {
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GArray *buffers;
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gint i;
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buffers = g_value_peek_pointer (sh);
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/* Concatenate all buffers in streamheader into one */
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rtmpsink->header = gst_buffer_new ();
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for (i = 0; i < buffers->len; ++i) {
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GValue *val;
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GstBuffer *buf;
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val = &g_array_index (buffers, GValue, i);
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buf = g_value_peek_pointer (val);
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gst_buffer_ref (buf);
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rtmpsink->header = gst_buffer_append (rtmpsink->header, buf);
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}
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} else {
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GST_ERROR_OBJECT (rtmpsink, "streamheader field has unexpected type %s",
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G_VALUE_TYPE_NAME (sh));
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}
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GST_DEBUG_OBJECT (rtmpsink, "have %" G_GSIZE_FORMAT " bytes of header data",
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gst_buffer_get_size (rtmpsink->header));
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out:
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return TRUE;
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}
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static gboolean
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gst_rtmp_sink_event (GstBaseSink * sink, GstEvent * event)
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{
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GstRTMPSink *rtmpsink = GST_RTMP_SINK (sink);
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switch (event->type) {
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case GST_EVENT_FLUSH_STOP:
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rtmpsink->have_write_error = FALSE;
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break;
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default:
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break;
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}
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return GST_BASE_SINK_CLASS (parent_class)->event (sink, event);
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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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