| /* |
| * 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., 51 Franklin St, Fifth Floor, |
| * Boston, MA 02110-1301, 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 |
| |
| #include <stdlib.h> |
| |
| GST_DEBUG_CATEGORY_STATIC (gst_rtmp_sink_debug); |
| #define GST_CAT_DEFAULT gst_rtmp_sink_debug |
| |
| #define DEFAULT_LOCATION NULL |
| |
| 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; |
| |
| g_object_class_install_property (gobject_class, PROP_LOCATION, |
| g_param_spec_string ("location", "RTMP Location", "RTMP url", |
| DEFAULT_LOCATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); |
| |
| gst_element_class_set_static_metadata (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) |
| { |
| GstRTMPSink *sink = GST_RTMP_SINK (object); |
| |
| #ifdef G_OS_WIN32 |
| WSACleanup (); |
| #endif |
| g_free (sink->uri); |
| |
| 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; |
| GstMapInfo map; |
| |
| 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_append (sink->cache, buf); |
| sink->cache = NULL; |
| } |
| |
| GST_LOG_OBJECT (sink, "Sending %" G_GSIZE_FORMAT " bytes to RTMP server", |
| gst_buffer_get_size (buf)); |
| |
| gst_buffer_map (buf, &map, GST_MAP_READ); |
| |
| if (RTMP_Write (sink->rtmp, (char *) map.data, map.size) <= 0) |
| goto write_failed; |
| |
| gst_buffer_unmap (buf, &map); |
| 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, &map); |
| 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); |
| gboolean ret = TRUE; |
| |
| if (GST_STATE (sink) >= GST_STATE_PAUSED) { |
| g_set_error (error, GST_URI_ERROR, GST_URI_ERROR_BAD_STATE, |
| "Changing the URI on rtmpsink 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) { |
| 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"); |
| ret = FALSE; |
| } else { |
| sink->uri = g_strdup (uri); |
| } |
| |
| if (playpath.av_val) |
| free (playpath.av_val); |
| } |
| |
| if (ret) |
| GST_DEBUG_OBJECT (sink, "Changed URI to %s", GST_STR_NULL (uri)); |
| |
| return ret; |
| } |
| |
| 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; |
| } |
| } |