From 38bf3169ff0813d2efdf35333caa446441183917 Mon Sep 17 00:00:00 2001 From: Jan Schmidt Date: Sat, 18 Jun 2011 01:09:51 +1000 Subject: [PATCH] RTMP: add rtmpsink element for output to an RTMP server --- .../plugins/gst-plugins-bad-plugins-docs.sgml | 5 +- .../gst-plugins-bad-plugins-sections.txt | 28 ++ .../{plugin-rtmpsrc.xml => plugin-rtmp.xml} | 19 +- ext/rtmp/Makefile.am | 4 +- ext/rtmp/gstrtmp.c | 54 +++ ext/rtmp/gstrtmpsink.c | 347 ++++++++++++++++++ ext/rtmp/gstrtmpsink.h | 68 ++++ ext/rtmp/gstrtmpsrc.c | 17 +- 8 files changed, 522 insertions(+), 20 deletions(-) rename docs/plugins/inspect/{plugin-rtmpsrc.xml => plugin-rtmp.xml} (59%) create mode 100644 ext/rtmp/gstrtmp.c create mode 100644 ext/rtmp/gstrtmpsink.c create mode 100644 ext/rtmp/gstrtmpsink.h diff --git a/docs/plugins/gst-plugins-bad-plugins-docs.sgml b/docs/plugins/gst-plugins-bad-plugins-docs.sgml index 17acb5739b..db13207c38 100644 --- a/docs/plugins/gst-plugins-bad-plugins-docs.sgml +++ b/docs/plugins/gst-plugins-bad-plugins-docs.sgml @@ -89,7 +89,9 @@ - + + + @@ -196,6 +198,7 @@ + diff --git a/docs/plugins/gst-plugins-bad-plugins-sections.txt b/docs/plugins/gst-plugins-bad-plugins-sections.txt index 5efe2c1166..4a04d6a0cd 100644 --- a/docs/plugins/gst-plugins-bad-plugins-sections.txt +++ b/docs/plugins/gst-plugins-bad-plugins-sections.txt @@ -1160,6 +1160,34 @@ GST_TYPE_RSVG_DEC gst_rsvg_dec_get_type +
+element-rtmpsink +rtmpsink +GstRTMPSink + +GstRTMPSinkClass +GST_RTMP_SINK +GST_IS_RTMP_SINK +GST_TYPE_RTMP_SINK +gst_rtmp_sink_get_type +GST_RTMP_SINK_CLASS +GST_IS_RTMP_SINK_CLASS +
+ +
+element-rtmpsrc +rtmpsrc +GstRTMPSrc + +GstRTMPSrcClass +GST_RTMP_SRC +GST_IS_RTMP_SRC +GST_TYPE_RTMP_SRC +gst_rtmp_src_get_type +GST_RTMP_SRC_CLASS +GST_IS_RTMP_SRC_CLASS +
+
element-rtpdtmfdepay rtpdtmfdepay diff --git a/docs/plugins/inspect/plugin-rtmpsrc.xml b/docs/plugins/inspect/plugin-rtmp.xml similarity index 59% rename from docs/plugins/inspect/plugin-rtmpsrc.xml rename to docs/plugins/inspect/plugin-rtmp.xml index c85740938f..7d9ae4fdc9 100644 --- a/docs/plugins/inspect/plugin-rtmpsrc.xml +++ b/docs/plugins/inspect/plugin-rtmp.xml @@ -1,6 +1,6 @@ - rtmpsrc - RTMP source + rtmp + RTMP source and sink ../../ext/rtmp/.libs/libgstrtmp.so libgstrtmp.so 0.10.22.1 @@ -9,6 +9,21 @@ GStreamer Bad Plug-ins git Unknown package origin + + rtmpsink + RTMP output sink + Sink/Network + Sends FLV content to a server via RTMP + Jan Schmidt <thaytan@noraisin.net> + + + sink + sink + always +
video/x-flv
+ + + rtmpsrc RTMP Source diff --git a/ext/rtmp/Makefile.am b/ext/rtmp/Makefile.am index e97c7a7589..bd2398cab0 100644 --- a/ext/rtmp/Makefile.am +++ b/ext/rtmp/Makefile.am @@ -1,8 +1,8 @@ plugin_LTLIBRARIES = libgstrtmp.la -libgstrtmp_la_SOURCES = gstrtmpsrc.c +libgstrtmp_la_SOURCES = gstrtmpsrc.c gstrtmpsink.c gstrtmp.c -noinst_HEADERS = gstrtmpsrc.h +noinst_HEADERS = gstrtmpsrc.h gstrtmpsink.h libgstrtmp_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(RTMP_CFLAGS) libgstrtmp_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) $(GST_LIBS) $(RTMP_LIBS) libgstrtmp_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) diff --git a/ext/rtmp/gstrtmp.c b/ext/rtmp/gstrtmp.c new file mode 100644 index 0000000000..7acbea4a90 --- /dev/null +++ b/ext/rtmp/gstrtmp.c @@ -0,0 +1,54 @@ +/* GStreamer + * Copyright (C) 1999,2000 Erik Walthinsen + * 2000 Wim Taymans + * 2002 Kristian Rietveld + * 2002,2003 Colin Walters + * 2001,2010 Bastien Nocera + * 2010 Sebastian Dröge + * 2010 Jan Schmidt + * + * rtmpsrc.c: + * + * 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. + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include + +#include "gstrtmpsrc.h" +#include "gstrtmpsink.h" + +static gboolean +plugin_init (GstPlugin * plugin) +{ + gboolean ret; + + ret = gst_element_register (plugin, "rtmpsrc", GST_RANK_PRIMARY, + GST_TYPE_RTMP_SRC); + ret &= gst_element_register (plugin, "rtmpsink", GST_RANK_PRIMARY, + GST_TYPE_RTMP_SINK); + + return ret; +} + +GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, + GST_VERSION_MINOR, + "rtmp", + "RTMP source and sink", + plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN); diff --git a/ext/rtmp/gstrtmpsink.c b/ext/rtmp/gstrtmpsink.c new file mode 100644 index 0000000000..e3933b1503 --- /dev/null +++ b/ext/rtmp/gstrtmpsink.c @@ -0,0 +1,347 @@ +/* + * GStreamer + * Copyright (C) 2010 Jan Schmidt + * + * 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 + * + * + * Example launch line + * |[ + * 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. + * + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include + +#include "gstrtmpsink.h" + +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 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); + +static void +_do_init (GType gtype) +{ + static const GInterfaceInfo urihandler_info = { + gst_rtmp_sink_uri_handler_init, + NULL, + NULL + }; + + g_type_add_interface_static (gtype, GST_TYPE_URI_HANDLER, &urihandler_info); + + GST_DEBUG_CATEGORY_INIT (gst_rtmp_sink_debug, "rtmpsink", 0, + "RTMP server element"); +} + +GST_BOILERPLATE_FULL (GstRTMPSink, gst_rtmp_sink, GstBaseSink, + GST_TYPE_BASE_SINK, _do_init); + + +static void +gst_rtmp_sink_base_init (gpointer klass) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (klass); + + gst_element_class_set_details_simple (element_class, + "RTMP output sink", + "Sink/Network", "Sends FLV content to a server via RTMP", + "Jan Schmidt "); + + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&sink_template)); +} + +/* initialize the plugin's class */ +static void +gst_rtmp_sink_class_init (GstRTMPSinkClass * klass) +{ + GObjectClass *gobject_class; + GstBaseSinkClass *gstbasesink_class = (GstBaseSinkClass *) klass; + + gobject_class = (GObjectClass *) klass; + gobject_class->set_property = gst_rtmp_sink_set_property; + gobject_class->get_property = gst_rtmp_sink_get_property; + + 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_element_class_install_std_props (GST_ELEMENT_CLASS (klass), + "location", PROP_LOCATION, G_PARAM_READWRITE, NULL); +} + +/* initialize the new element + * initialize instance structure + */ +static void +gst_rtmp_sink_init (GstRTMPSink * sink, GstRTMPSinkClass * klass) +{ +} + +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); + + /* 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 FALSE; + } + GST_DEBUG_OBJECT (sink, "Opened connection to %s", sink->rtmp_uri); + } + + 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; + + if (sink->first) { + /* 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 %d for concatenation", + GST_BUFFER_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 %d to cached buf", + GST_BUFFER_SIZE (buf)); + gst_buffer_ref (buf); + reffed_buf = buf = gst_buffer_join (sink->cache, buf); + sink->cache = NULL; + } + + GST_LOG_OBJECT (sink, "Sending %d bytes to RTMP server", + GST_BUFFER_SIZE (buf)); + + if (!RTMP_Write (sink->rtmp, + (char *) GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf))) { + GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL), ("Failed to write data")); + if (reffed_buf) + gst_buffer_unref (reffed_buf); + return GST_FLOW_ERROR; + } + + if (reffed_buf) + gst_buffer_unref (reffed_buf); + + return GST_FLOW_OK; +} + +/* + * URI interface support. + */ +static GstURIType +gst_rtmp_sink_uri_get_type (void) +{ + return GST_URI_SINK; +} + +static gchar ** +gst_rtmp_sink_uri_get_protocols (void) +{ + static gchar *protocols[] = + { (char *) "rtmp", (char *) "rtmpt", (char *) "rtmps", (char *) "rtmpe", + (char *) "rtmfp", (char *) "rtmpte", (char *) "rtmpts", NULL + }; + return protocols; +} + +static const gchar * +gst_rtmp_sink_uri_get_uri (GstURIHandler * handler) +{ + GstRTMPSink *sink = GST_RTMP_SINK (handler); + + return sink->uri; +} + +static gboolean +gst_rtmp_sink_uri_set_uri (GstURIHandler * handler, const gchar * uri) +{ + GstRTMPSink *sink = GST_RTMP_SINK (handler); + + if (GST_STATE (sink) >= GST_STATE_PAUSED) + 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)); + 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)); + 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; + } +} diff --git a/ext/rtmp/gstrtmpsink.h b/ext/rtmp/gstrtmpsink.h new file mode 100644 index 0000000000..cb9315ebb2 --- /dev/null +++ b/ext/rtmp/gstrtmpsink.h @@ -0,0 +1,68 @@ +/* + * GStreamer + * Copyright (C) 2010 Jan Schmidt + * + * 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. + */ + +#ifndef __GST_RTMP_SINK_H__ +#define __GST_RTMP_SINK_H__ + +#include +#include + +#include +#include +#include + +G_BEGIN_DECLS + +#define GST_TYPE_RTMP_SINK \ + (gst_rtmp_sink_get_type()) +#define GST_RTMP_SINK(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTMP_SINK,GstRTMPSink)) +#define GST_RTMP_SINK_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTMP_SINK,GstRTMPSinkClass)) +#define GST_IS_RTMP_SINK(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTMP_SINK)) +#define GST_IS_RTMP_SINK_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTMP_SINK)) + +typedef struct _GstRTMPSink GstRTMPSink; +typedef struct _GstRTMPSinkClass GstRTMPSinkClass; + +struct _GstRTMPSink { + GstBaseSink parent; + + /* < private > */ + gchar *uri; + + RTMP *rtmp; + gchar *rtmp_uri; /* copy of url for librtmp */ + + GstBuffer *cache; /* Cached buffer */ + gboolean first; +}; + +struct _GstRTMPSinkClass { + GstBaseSinkClass parent_class; +}; + +GType gst_rtmp_sink_get_type (void); + +G_END_DECLS + +#endif /* __GST_RTMP_SINK_H__ */ diff --git a/ext/rtmp/gstrtmpsrc.c b/ext/rtmp/gstrtmpsrc.c index 2376ccef14..e37ac06b73 100644 --- a/ext/rtmp/gstrtmpsrc.c +++ b/ext/rtmp/gstrtmpsrc.c @@ -98,6 +98,8 @@ _do_init (GType gtype) }; g_type_add_interface_static (gtype, GST_TYPE_URI_HANDLER, &urihandler_info); + + GST_DEBUG_CATEGORY_INIT (rtmpsrc_debug, "rtmpsrc", 0, "RTMP Source"); } GST_BOILERPLATE_FULL (GstRTMPSrc, gst_rtmp_src, GstPushSrc, GST_TYPE_PUSH_SRC, @@ -581,18 +583,3 @@ gst_rtmp_src_stop (GstBaseSrc * basesrc) return TRUE; } - -static gboolean -plugin_init (GstPlugin * plugin) -{ - GST_DEBUG_CATEGORY_INIT (rtmpsrc_debug, "rtmpsrc", 0, "RTMP Source"); - - return gst_element_register (plugin, "rtmpsrc", GST_RANK_PRIMARY, - GST_TYPE_RTMP_SRC); -} - -GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, - GST_VERSION_MINOR, - "rtmpsrc", - "RTMP source", - plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);