/* GStreamer * Copyright (C) 1999,2000 Erik Walthinsen * 2000 Wim Taymans * 2002 Kristian Rietveld * 2002,2003 Colin Walters * 2001,2010 Bastien Nocera * 2010 Sebastian Dröge * * 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., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ /** * SECTION:element-rtmpsrc * @title: rtmpsrc * * This plugin reads data from a local or remote location specified * by an URI. This location can be specified using any protocol supported by * the RTMP library, i.e. rtmp, rtmpt, rtmps, rtmpe, rtmfp, rtmpte and rtmpts. * * ## Example launch lines * |[ * gst-launch-1.0 -v rtmpsrc location=rtmp://somehost/someurl ! fakesink * ]| Open an RTMP location and pass its content to fakesink. * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include "gstrtmpsrc.h" #include #include #include #include #ifdef G_OS_WIN32 #include #endif GST_DEBUG_CATEGORY_STATIC (rtmpsrc_debug); #define GST_CAT_DEFAULT rtmpsrc_debug static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS_ANY); enum { PROP_0, PROP_LOCATION, PROP_TIMEOUT #if 0 PROP_SWF_URL, PROP_PAGE_URL #endif }; #define DEFAULT_LOCATION NULL #define DEFAULT_TIMEOUT 120 static void gst_rtmp_src_uri_handler_init (gpointer g_iface, gpointer iface_data); static void gst_rtmp_src_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_rtmp_src_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static void gst_rtmp_src_finalize (GObject * object); static gboolean gst_rtmp_src_unlock (GstBaseSrc * src); static gboolean gst_rtmp_src_stop (GstBaseSrc * src); static gboolean gst_rtmp_src_start (GstBaseSrc * src); static gboolean gst_rtmp_src_is_seekable (GstBaseSrc * src); static gboolean gst_rtmp_src_prepare_seek_segment (GstBaseSrc * src, GstEvent * event, GstSegment * segment); static gboolean gst_rtmp_src_do_seek (GstBaseSrc * src, GstSegment * segment); static GstFlowReturn gst_rtmp_src_create (GstPushSrc * pushsrc, GstBuffer ** buffer); static gboolean gst_rtmp_src_query (GstBaseSrc * src, GstQuery * query); #define gst_rtmp_src_parent_class parent_class G_DEFINE_TYPE_WITH_CODE (GstRTMPSrc, gst_rtmp_src, GST_TYPE_PUSH_SRC, G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtmp_src_uri_handler_init)); static void gst_rtmp_src_class_init (GstRTMPSrcClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GstBaseSrcClass *gstbasesrc_class; GstPushSrcClass *gstpushsrc_class; gobject_class = G_OBJECT_CLASS (klass); gstelement_class = GST_ELEMENT_CLASS (klass); gstbasesrc_class = GST_BASE_SRC_CLASS (klass); gstpushsrc_class = GST_PUSH_SRC_CLASS (klass); gobject_class->finalize = gst_rtmp_src_finalize; gobject_class->set_property = gst_rtmp_src_set_property; gobject_class->get_property = gst_rtmp_src_get_property; /* properties */ g_object_class_install_property (gobject_class, PROP_LOCATION, g_param_spec_string ("location", "RTMP Location", "Location of the RTMP url to read", DEFAULT_LOCATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_TIMEOUT, g_param_spec_int ("timeout", "RTMP Timeout", "Time without receiving any data from the server to wait before to timeout the session", 0, G_MAXINT, DEFAULT_TIMEOUT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gst_element_class_add_static_pad_template (gstelement_class, &srctemplate); gst_element_class_set_static_metadata (gstelement_class, "RTMP Source", "Source/File", "Read RTMP streams", "Bastien Nocera , " "Sebastian Dröge "); gstbasesrc_class->start = GST_DEBUG_FUNCPTR (gst_rtmp_src_start); gstbasesrc_class->stop = GST_DEBUG_FUNCPTR (gst_rtmp_src_stop); gstbasesrc_class->unlock = GST_DEBUG_FUNCPTR (gst_rtmp_src_unlock); gstbasesrc_class->is_seekable = GST_DEBUG_FUNCPTR (gst_rtmp_src_is_seekable); gstbasesrc_class->prepare_seek_segment = GST_DEBUG_FUNCPTR (gst_rtmp_src_prepare_seek_segment); gstbasesrc_class->do_seek = GST_DEBUG_FUNCPTR (gst_rtmp_src_do_seek); gstpushsrc_class->create = GST_DEBUG_FUNCPTR (gst_rtmp_src_create); gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_rtmp_src_query); GST_DEBUG_CATEGORY_INIT (rtmpsrc_debug, "rtmpsrc", 0, "RTMP Source"); } static void gst_rtmp_src_init (GstRTMPSrc * rtmpsrc) { #ifdef G_OS_WIN32 WSADATA wsa_data; if (WSAStartup (MAKEWORD (2, 2), &wsa_data) != 0) { GST_ERROR_OBJECT (rtmpsrc, "WSAStartup failed: 0x%08x", WSAGetLastError ()); } #endif rtmpsrc->cur_offset = 0; rtmpsrc->last_timestamp = 0; rtmpsrc->timeout = DEFAULT_TIMEOUT; gst_base_src_set_format (GST_BASE_SRC (rtmpsrc), GST_FORMAT_TIME); } static void gst_rtmp_src_finalize (GObject * object) { GstRTMPSrc *rtmpsrc = GST_RTMP_SRC (object); g_free (rtmpsrc->uri); rtmpsrc->uri = NULL; #ifdef G_OS_WIN32 WSACleanup (); #endif G_OBJECT_CLASS (parent_class)->finalize (object); } /* * URI interface support. */ static GstURIType gst_rtmp_src_uri_get_type (GType type) { return GST_URI_SRC; } static const gchar *const * gst_rtmp_src_uri_get_protocols (GType type) { static const gchar *protocols[] = { "rtmp", "rtmpt", "rtmps", "rtmpe", "rtmfp", "rtmpte", "rtmpts", NULL }; return protocols; } static gchar * gst_rtmp_src_uri_get_uri (GstURIHandler * handler) { GstRTMPSrc *src = GST_RTMP_SRC (handler); /* FIXME: make thread-safe */ return g_strdup (src->uri); } static gboolean gst_rtmp_src_uri_set_uri (GstURIHandler * handler, const gchar * uri, GError ** error) { GstRTMPSrc *src = GST_RTMP_SRC (handler); if (GST_STATE (src) >= 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 (src->uri); src->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_ERROR_OBJECT (src, "Failed to parse URI %s", uri); g_set_error (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI, "Could not parse RTMP URI"); /* FIXME: we should not be freeing RTMP internals to avoid leaking */ free (playpath.av_val); return FALSE; } free (playpath.av_val); src->uri = g_strdup (uri); } GST_DEBUG_OBJECT (src, "Changed URI to %s", GST_STR_NULL (uri)); return TRUE; } static void gst_rtmp_src_uri_handler_init (gpointer g_iface, gpointer iface_data) { GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface; iface->get_type = gst_rtmp_src_uri_get_type; iface->get_protocols = gst_rtmp_src_uri_get_protocols; iface->get_uri = gst_rtmp_src_uri_get_uri; iface->set_uri = gst_rtmp_src_uri_set_uri; } static void gst_rtmp_src_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstRTMPSrc *src; src = GST_RTMP_SRC (object); switch (prop_id) { case PROP_LOCATION:{ gst_rtmp_src_uri_set_uri (GST_URI_HANDLER (src), g_value_get_string (value), NULL); break; } case PROP_TIMEOUT:{ src->timeout = g_value_get_int (value); break; } default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_rtmp_src_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstRTMPSrc *src; src = GST_RTMP_SRC (object); switch (prop_id) { case PROP_LOCATION: g_value_set_string (value, src->uri); break; case PROP_TIMEOUT: g_value_set_int (value, src->timeout); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } /* * Read a new buffer from src->reqoffset, takes care of events * and seeking and such. */ static GstFlowReturn gst_rtmp_src_create (GstPushSrc * pushsrc, GstBuffer ** buffer) { GstRTMPSrc *src; GstBuffer *buf; GstMapInfo map; guint8 *data; guint todo; gsize bsize; int size; src = GST_RTMP_SRC (pushsrc); g_return_val_if_fail (src->rtmp != NULL, GST_FLOW_ERROR); size = GST_BASE_SRC_CAST (pushsrc)->blocksize; GST_DEBUG ("reading from %" G_GUINT64_FORMAT ", size %u", src->cur_offset, size); buf = gst_buffer_new_allocate (NULL, size, NULL); if (G_UNLIKELY (buf == NULL)) { GST_ERROR_OBJECT (src, "Failed to allocate %u bytes", size); return GST_FLOW_ERROR; } todo = size; gst_buffer_map (buf, &map, GST_MAP_WRITE); data = map.data; bsize = 0; while (todo > 0) { int read = RTMP_Read (src->rtmp, (char *) data, todo); if (G_UNLIKELY (read == 0 && todo == size)) goto eos; if (G_UNLIKELY (read == 0)) break; if (G_UNLIKELY (read < 0)) goto read_failed; if (read < todo) { data += read; todo -= read; bsize += read; } else { bsize += todo; todo = 0; } GST_LOG (" got size %d", read); } gst_buffer_unmap (buf, &map); gst_buffer_resize (buf, 0, bsize); if (src->discont) { GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT); src->discont = FALSE; } GST_BUFFER_TIMESTAMP (buf) = src->last_timestamp; GST_BUFFER_OFFSET (buf) = src->cur_offset; src->cur_offset += size; if (src->last_timestamp == GST_CLOCK_TIME_NONE) src->last_timestamp = src->rtmp->m_mediaStamp * GST_MSECOND; else src->last_timestamp = MAX (src->last_timestamp, src->rtmp->m_mediaStamp * GST_MSECOND); GST_LOG_OBJECT (src, "Created buffer of size %u at %" G_GINT64_FORMAT " with timestamp %" GST_TIME_FORMAT, size, GST_BUFFER_OFFSET (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf))); /* we're done, return the buffer */ *buffer = buf; return GST_FLOW_OK; read_failed: { gst_buffer_unref (buf); GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL), ("Failed to read data")); return GST_FLOW_ERROR; } eos: { gst_buffer_unref (buf); if (src->cur_offset == 0) { GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL), ("Failed to read any data from stream, check your URL")); return GST_FLOW_ERROR; } else { GST_DEBUG_OBJECT (src, "Reading data gave EOS"); return GST_FLOW_EOS; } } } static gboolean gst_rtmp_src_query (GstBaseSrc * basesrc, GstQuery * query) { gboolean ret = FALSE; GstRTMPSrc *src = GST_RTMP_SRC (basesrc); switch (GST_QUERY_TYPE (query)) { case GST_QUERY_URI: gst_query_set_uri (query, src->uri); ret = TRUE; break; case GST_QUERY_POSITION:{ GstFormat format; gst_query_parse_position (query, &format, NULL); if (format == GST_FORMAT_TIME) { gst_query_set_position (query, format, src->last_timestamp); ret = TRUE; } break; } case GST_QUERY_DURATION:{ GstFormat format; gdouble duration; gst_query_parse_duration (query, &format, NULL); if (format == GST_FORMAT_TIME && src->rtmp) { duration = RTMP_GetDuration (src->rtmp); if (duration != 0.0) { gst_query_set_duration (query, format, duration * GST_SECOND); ret = TRUE; } } break; } case GST_QUERY_SCHEDULING:{ gst_query_set_scheduling (query, GST_SCHEDULING_FLAG_SEQUENTIAL | GST_SCHEDULING_FLAG_BANDWIDTH_LIMITED, 1, -1, 0); gst_query_add_scheduling_mode (query, GST_PAD_MODE_PUSH); ret = TRUE; break; } default: ret = FALSE; break; } if (!ret) ret = GST_BASE_SRC_CLASS (parent_class)->query (basesrc, query); return ret; } static gboolean gst_rtmp_src_is_seekable (GstBaseSrc * basesrc) { GstRTMPSrc *src; src = GST_RTMP_SRC (basesrc); return src->seekable; } static gboolean gst_rtmp_src_prepare_seek_segment (GstBaseSrc * basesrc, GstEvent * event, GstSegment * segment) { GstRTMPSrc *src; GstSeekType cur_type, stop_type; gint64 cur, stop; GstSeekFlags flags; GstFormat format; gdouble rate; src = GST_RTMP_SRC (basesrc); gst_event_parse_seek (event, &rate, &format, &flags, &cur_type, &cur, &stop_type, &stop); if (!src->seekable) { GST_LOG_OBJECT (src, "Not a seekable stream"); return FALSE; } if (!src->rtmp) { GST_LOG_OBJECT (src, "Not connected yet"); return FALSE; } if (format != GST_FORMAT_TIME) { GST_LOG_OBJECT (src, "Seeking only supported in TIME format"); return FALSE; } if (stop_type != GST_SEEK_TYPE_NONE) { GST_LOG_OBJECT (src, "Setting a stop position is not supported"); return FALSE; } gst_segment_init (segment, GST_FORMAT_TIME); gst_segment_do_seek (segment, rate, format, flags, cur_type, cur, stop_type, stop, NULL); return TRUE; } static gboolean gst_rtmp_src_do_seek (GstBaseSrc * basesrc, GstSegment * segment) { GstRTMPSrc *src; src = GST_RTMP_SRC (basesrc); if (segment->format != GST_FORMAT_TIME) { GST_LOG_OBJECT (src, "Only time based seeks are supported"); return FALSE; } if (!src->rtmp) { GST_LOG_OBJECT (src, "Not connected yet"); return FALSE; } src->discont = TRUE; /* Initial seek */ if (src->cur_offset == 0 && segment->start == 0) return TRUE; if (!src->seekable) { GST_LOG_OBJECT (src, "Not a seekable stream"); return FALSE; } src->last_timestamp = GST_CLOCK_TIME_NONE; if (!RTMP_SendSeek (src->rtmp, segment->start / GST_MSECOND)) { GST_ERROR_OBJECT (src, "Seeking failed"); src->seekable = FALSE; return FALSE; } GST_DEBUG_OBJECT (src, "Seek to %" GST_TIME_FORMAT " successfull", GST_TIME_ARGS (segment->start)); return TRUE; } #define STR2AVAL(av,str) G_STMT_START { \ av.av_val = str; \ av.av_len = strlen(av.av_val); \ } G_STMT_END; /* open the file, do stuff necessary to go to PAUSED state */ static gboolean gst_rtmp_src_start (GstBaseSrc * basesrc) { GstRTMPSrc *src; src = GST_RTMP_SRC (basesrc); if (!src->uri) { GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL), ("No filename given")); return FALSE; } src->cur_offset = 0; src->last_timestamp = 0; src->discont = TRUE; src->rtmp = RTMP_Alloc (); if (!src->rtmp) { GST_ERROR_OBJECT (src, "Could not allocate librtmp's RTMP context"); goto error; } RTMP_Init (src->rtmp); src->rtmp->Link.timeout = src->timeout; if (!RTMP_SetupURL (src->rtmp, src->uri)) { GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL), ("Failed to setup URL '%s'", src->uri)); goto error; } src->seekable = !(src->rtmp->Link.lFlags & RTMP_LF_LIVE); GST_INFO_OBJECT (src, "seekable %d", src->seekable); /* open if required */ if (!RTMP_IsConnected (src->rtmp)) { if (!RTMP_Connect (src->rtmp, NULL)) { GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL), ("Could not connect to RTMP stream \"%s\" for reading", src->uri)); goto error; } } return TRUE; error: if (src->rtmp) { RTMP_Free (src->rtmp); src->rtmp = NULL; } return FALSE; } #undef STR2AVAL static gboolean gst_rtmp_src_unlock (GstBaseSrc * basesrc) { GstRTMPSrc *rtmpsrc = GST_RTMP_SRC (basesrc); GST_DEBUG_OBJECT (rtmpsrc, "unlock"); /* This closes the socket, which means that any pending socket calls * error out. */ if (rtmpsrc->rtmp) { RTMP_Close (rtmpsrc->rtmp); } return TRUE; } static gboolean gst_rtmp_src_stop (GstBaseSrc * basesrc) { GstRTMPSrc *src; src = GST_RTMP_SRC (basesrc); if (src->rtmp) { RTMP_Free (src->rtmp); src->rtmp = NULL; } src->cur_offset = 0; src->last_timestamp = 0; src->discont = TRUE; return TRUE; }