diff options
| -rw-r--r-- | ext/rtmp/gstrtmpsrc.c | 426 | ||||
| -rw-r--r-- | ext/rtmp/gstrtmpsrc.h | 14 |
2 files changed, 126 insertions, 314 deletions
diff --git a/ext/rtmp/gstrtmpsrc.c b/ext/rtmp/gstrtmpsrc.c index c8aceb5fc..a8d559bb6 100644 --- a/ext/rtmp/gstrtmpsrc.c +++ b/ext/rtmp/gstrtmpsrc.c @@ -1,9 +1,10 @@ /* GStreamer * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> * 2000 Wim Taymans <wtay@chello.be> - * 2001 Bastien Nocera <hadess@hadess.net> * 2002 Kristian Rietveld <kris@gtk.org> * 2002,2003 Colin Walters <walters@gnu.org> + * 2001,2010 Bastien Nocera <hadess@hadess.net> + * 2010 Sebastian Dröge <sebastian.droege@collabora.co.uk> * * rtmpsrc.c: * @@ -28,38 +29,16 @@ * * 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. Common protocols are 'file', 'http', 'ftp', or 'smb'. - * - * In case the #GstRTMPSrc:iradio-mode property is set and the - * location is a http resource, rtmpsrc will send special icecast http - * headers to the server to request additional icecast metainformation. If - * the server is not an icecast server, it will display the same behaviour - * as if the #GstRTMPSrc:iradio-mode property was not set. However, - * if the server is in fact an icecast server, rtmpsrc will output - * data with a media type of application/x-icy, in which case you will - * need to use the #GstICYDemux element as follow-up element to extract - * the icecast meta data and to determine the underlying media type. + * the RTMP library, i.e. rtmp, rtmpt, rtmps, rtmpe, rtmfp, rtmpte and rtmpts. * * <refsect2> * <title>Example launch lines</title> * |[ - * gst-launch -v rtmpsrc location=file:///home/joe/foo.xyz ! fakesink - * ]| The above pipeline will simply read a local file and do nothing with the - * data read. Instead of rtmpsrc, we could just as well have used the - * filesrc element here. - * |[ - * gst-launch -v rtmpsrc location=smb://othercomputer/foo.xyz ! filesink location=/home/joe/foo.xyz - * ]| The above pipeline will copy a file from a remote host to the local file - * system using the Samba protocol. - * |[ - * gst-launch -v rtmpsrc location=http://music.foobar.com/demo.mp3 ! mad ! audioconvert ! audioresample ! alsasink - * ]| The above pipeline will read and decode and play an mp3 file from a - * web server using the http protocol. + * gst-launch -v rtmpsrc location=rtmp://somehost/someurl ! fakesink + * ]| Open an RTMP location and pass its content to fakesink. * </refsect2> */ -#define DEFAULT_RTMP_PORT 1935 - #ifdef HAVE_CONFIG_H #include "config.h" #endif @@ -70,21 +49,9 @@ #include <stdio.h> #include <stdlib.h> -#include <sys/types.h> -#include <sys/socket.h> -#include <sys/time.h> -#include <netinet/in.h> -#include <arpa/inet.h> -#include <netdb.h> -#include <sys/stat.h> -#include <fcntl.h> -#include <unistd.h> -#include <sys/mman.h> -#include <errno.h> #include <string.h> #include <gst/gst.h> -#include <gst/tag/tag.h> GST_DEBUG_CATEGORY_STATIC (rtmpsrc_debug); #define GST_CAT_DEFAULT rtmpsrc_debug @@ -96,14 +63,10 @@ static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src", enum { - ARG_0, - ARG_LOCATION, + PROP_0, + PROP_LOCATION, }; -static void gst_rtmp_src_base_init (gpointer g_class); -static void gst_rtmp_src_class_init (GstRTMPSrcClass * klass); -static void gst_rtmp_src_init (GstRTMPSrc * rtmpsrc); -static void gst_rtmp_src_finalize (GObject * object); static void gst_rtmp_src_uri_handler_init (gpointer g_iface, gpointer iface_data); @@ -111,67 +74,30 @@ 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_stop (GstBaseSrc * src); static gboolean gst_rtmp_src_start (GstBaseSrc * src); static gboolean gst_rtmp_src_is_seekable (GstBaseSrc * src); -#if 0 -static gboolean gst_rtmp_src_check_get_range (GstBaseSrc * src); -static gboolean gst_rtmp_src_get_size (GstBaseSrc * src, guint64 * size); -#endif -static GstFlowReturn gst_rtmp_src_create (GstBaseSrc * basesrc, - guint64 offset, guint size, GstBuffer ** buffer); -#if 0 +static GstFlowReturn gst_rtmp_src_create (GstPushSrc * pushsrc, + GstBuffer ** buffer); static gboolean gst_rtmp_src_query (GstBaseSrc * src, GstQuery * query); -#endif - -static GstElementClass *parent_class = NULL; -static gboolean -plugin_init (GstPlugin * plugin) +static void +_do_init (GType gtype) { - return gst_element_register (plugin, "rtmpsrc", GST_RANK_NONE, - GST_TYPE_RTMP_SRC); -} + static const GInterfaceInfo urihandler_info = { + gst_rtmp_src_uri_handler_init, + NULL, + NULL + }; -GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, - GST_VERSION_MINOR, - "rtmpsrc", - "flvstreamer sources", - plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN); - -GType -gst_rtmp_src_get_type (void) -{ - static GType rtmpsrc_type = 0; - - if (!rtmpsrc_type) { - static const GTypeInfo rtmpsrc_info = { - sizeof (GstRTMPSrcClass), - gst_rtmp_src_base_init, - NULL, - (GClassInitFunc) gst_rtmp_src_class_init, - NULL, - NULL, - sizeof (GstRTMPSrc), - 0, - (GInstanceInitFunc) gst_rtmp_src_init, - }; - static const GInterfaceInfo urihandler_info = { - gst_rtmp_src_uri_handler_init, - NULL, - NULL - }; - - rtmpsrc_type = - g_type_register_static (GST_TYPE_BASE_SRC, - "GstRTMPSrc", &rtmpsrc_info, (GTypeFlags) 0); - g_type_add_interface_static (rtmpsrc_type, GST_TYPE_URI_HANDLER, - &urihandler_info); - } - return rtmpsrc_type; + g_type_add_interface_static (gtype, GST_TYPE_URI_HANDLER, &urihandler_info); } +GST_BOILERPLATE_FULL (GstRTMPSrc, gst_rtmp_src, GstPushSrc, GST_TYPE_PUSH_SRC, + _do_init); + static void gst_rtmp_src_base_init (gpointer g_class) { @@ -184,10 +110,8 @@ gst_rtmp_src_base_init (gpointer g_class) "RTMP Source", "Source/File", "Read RTMP streams", - "Bastien Nocera <hadess@hadess.net>\n" - "GStreamer maintainers <gstreamer-devel@lists.sourceforge.net>"); - - GST_DEBUG_CATEGORY_INIT (rtmpsrc_debug, "rtmpsrc", 0, "RTMP Source"); + "Bastien Nocera <hadess@hadess.net>, " + "Sebastian Dröge <sebastian.droege@collabora.co.uk>"); } static void @@ -195,11 +119,11 @@ gst_rtmp_src_class_init (GstRTMPSrcClass * klass) { GObjectClass *gobject_class; GstBaseSrcClass *gstbasesrc_class; + GstPushSrcClass *gstpushsrc_class; gobject_class = G_OBJECT_CLASS (klass); gstbasesrc_class = GST_BASE_SRC_CLASS (klass); - - parent_class = (GstElementClass *) g_type_class_peek_parent (klass); + gstpushsrc_class = GST_PUSH_SRC_CLASS (klass); gobject_class->finalize = gst_rtmp_src_finalize; gobject_class->set_property = gst_rtmp_src_set_property; @@ -207,29 +131,19 @@ gst_rtmp_src_class_init (GstRTMPSrcClass * klass) /* properties */ gst_element_class_install_std_props (GST_ELEMENT_CLASS (klass), - "location", ARG_LOCATION, G_PARAM_READWRITE, NULL); + "location", PROP_LOCATION, G_PARAM_READWRITE, NULL); gstbasesrc_class->start = GST_DEBUG_FUNCPTR (gst_rtmp_src_start); gstbasesrc_class->stop = GST_DEBUG_FUNCPTR (gst_rtmp_src_stop); -#if 0 - gstbasesrc_class->get_size = GST_DEBUG_FUNCPTR (gst_rtmp_src_get_size); -#endif gstbasesrc_class->is_seekable = GST_DEBUG_FUNCPTR (gst_rtmp_src_is_seekable); -#if 0 - gstbasesrc_class->check_get_range = - GST_DEBUG_FUNCPTR (gst_rtmp_src_check_get_range); -#endif - gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_rtmp_src_create); -#if 0 + gstpushsrc_class->create = GST_DEBUG_FUNCPTR (gst_rtmp_src_create); gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_rtmp_src_query); -#endif } static void -gst_rtmp_src_init (GstRTMPSrc * rtmpsrc) +gst_rtmp_src_init (GstRTMPSrc * rtmpsrc, GstRTMPSrcClass * klass) { rtmpsrc->curoffset = 0; - rtmpsrc->seekable = FALSE; } static void @@ -262,7 +176,10 @@ gst_rtmp_src_uri_get_type (void) static gchar ** gst_rtmp_src_uri_get_protocols (void) { - static gchar *protocols[] = { (char *) "rtmp", NULL }; + static gchar *protocols[] = + { (char *) "rtmp", (char *) "rtmpt", (char *) "rtmps", (char *) "rtmpe", + (char *) "rtmfp", (char *) "rtmpte", (char *) "rtmpts", NULL + }; return protocols; } @@ -278,13 +195,40 @@ static gboolean gst_rtmp_src_uri_set_uri (GstURIHandler * handler, const gchar * uri) { GstRTMPSrc *src = GST_RTMP_SRC (handler); + gchar *new_location; - if (GST_STATE (src) == GST_STATE_PLAYING || - GST_STATE (src) == GST_STATE_PAUSED) + if (GST_STATE (src) >= GST_STATE_PAUSED) return FALSE; - g_object_set (G_OBJECT (src), "location", uri, NULL); - g_message ("just set uri to %s", uri); + g_free (src->uri); + src->uri = NULL; + + if (src->rtmp) { + RTMP_Close (src->rtmp); + RTMP_Free (src->rtmp); + src->rtmp = NULL; + } + + if (uri != NULL) { + + new_location = g_strdup (uri); + + src->rtmp = RTMP_Alloc (); + RTMP_Init (src->rtmp); + if (!RTMP_SetupURL (src->rtmp, new_location)) { + GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, NULL, + ("Failed to setup URL '%s'", src->uri)); + g_free (new_location); + RTMP_Free (src->rtmp); + src->rtmp = NULL; + return FALSE; + } else { + src->uri = g_strdup (uri); + GST_DEBUG_OBJECT (src, "parsed uri '%s' properly", src->uri); + } + } + + GST_DEBUG_OBJECT (src, "Changed URI to %s", GST_STR_NULL (uri)); return TRUE; } @@ -309,36 +253,9 @@ gst_rtmp_src_set_property (GObject * object, guint prop_id, src = GST_RTMP_SRC (object); switch (prop_id) { - case ARG_LOCATION:{ - char *new_location; - /* the element must be stopped or paused in order to do this */ - if (GST_STATE (src) == GST_STATE_PLAYING || - GST_STATE (src) == GST_STATE_PAUSED) - break; - - g_free (src->uri); - src->uri = NULL; - - if (src->rtmp) { - RTMP_Close (src->rtmp); - RTMP_Free (src->rtmp); - src->rtmp = NULL; - } - - new_location = g_value_dup_string (value); - - src->rtmp = RTMP_Alloc (); - RTMP_Init (src->rtmp); - if (!RTMP_SetupURL (src->rtmp, new_location)) { - GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, NULL, - ("Failed to setup URL '%s'", src->uri)); - g_free (new_location); - RTMP_Free (src->rtmp); - src->rtmp = NULL; - } else { - src->uri = g_value_dup_string (value); - g_message ("parsed uri '%s' properly", src->uri); - } + case PROP_LOCATION:{ + gst_rtmp_src_uri_set_uri (GST_URI_HANDLER (src), + g_value_get_string (value)); break; } default: @@ -356,7 +273,7 @@ gst_rtmp_src_get_property (GObject * object, guint prop_id, GValue * value, src = GST_RTMP_SRC (object); switch (prop_id) { - case ARG_LOCATION: + case PROP_LOCATION: g_value_set_string (value, src->uri); break; default: @@ -370,50 +287,26 @@ gst_rtmp_src_get_property (GObject * object, guint prop_id, GValue * value, * and seeking and such. */ static GstFlowReturn -gst_rtmp_src_create (GstBaseSrc * basesrc, guint64 offset, guint size, - GstBuffer ** buffer) +gst_rtmp_src_create (GstPushSrc * pushsrc, GstBuffer ** buffer) { GstRTMPSrc *src; GstBuffer *buf; guint8 *data; guint todo; int read; + int size; - src = GST_RTMP_SRC (basesrc); + src = GST_RTMP_SRC (pushsrc); g_return_val_if_fail (src->rtmp != NULL, GST_FLOW_ERROR); - GST_DEBUG ("now at %" G_GINT64_FORMAT ", reading from %" G_GUINT64_FORMAT - ", size %u", src->curoffset, offset, size); + size = GST_BASE_SRC_CAST (pushsrc)->blocksize; - /* open if required */ - if (G_UNLIKELY (!RTMP_IsConnected (src->rtmp))) { - if (!RTMP_Connect (src->rtmp, NULL)) { - GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL), - ("Could not connect to RTMP stream \"%s\" for reading: %s (%d)", - src->uri, "FIXME", 0)); - return GST_FLOW_ERROR; - } - } - - /* seek if required */ - if (G_UNLIKELY (src->curoffset != offset)) { - GST_DEBUG ("need to seek"); - if (src->seekable) { -#if 0 - GST_DEBUG ("seeking to %" G_GUINT64_FORMAT, offset); - res = rtmp_seek (src->handle, RTMP_SEEK_START, offset); - if (res != RTMP_OK) - goto seek_failed; - src->curoffset = offset; -#endif - } else { - goto cannot_seek; - } - } + GST_DEBUG ("reading from %" G_GUINT64_FORMAT + ", size %u", src->curoffset, size); buf = gst_buffer_try_new_and_alloc (size); - if (G_UNLIKELY (buf == NULL && size == 0)) { + if (G_UNLIKELY (buf == NULL)) { GST_ERROR_OBJECT (src, "Failed to allocate %u bytes", size); return GST_FLOW_ERROR; } @@ -425,16 +318,14 @@ gst_rtmp_src_create (GstBaseSrc * basesrc, guint64 offset, guint size, todo = size; while (todo > 0) { - read = RTMP_Read (src->rtmp, (char *) &data, todo); + read = RTMP_Read (src->rtmp, (char *) data, todo); - if (G_UNLIKELY (read == -1)) + if (G_UNLIKELY (read == 0)) goto eos; - if (G_UNLIKELY (read == -2)) + if (G_UNLIKELY (read == -1)) goto read_failed; - /* FIXME handle -3 ? */ - if (read < todo) { data = &data[read]; todo -= read; @@ -449,30 +340,8 @@ gst_rtmp_src_create (GstBaseSrc * basesrc, guint64 offset, guint size, /* we're done, return the buffer */ *buffer = buf; -#if 0 - RTMPFileSize readbytes; - guint todo; - - - - return GST_FLOW_OK; -#endif return GST_FLOW_OK; -//seek_failed: - { - GST_ELEMENT_ERROR (src, RESOURCE, SEEK, (NULL), - ("Failed to seek to requested position %" G_GINT64_FORMAT ": %s", - offset, "FIXME")); - return GST_FLOW_ERROR; - } -cannot_seek: - { - GST_ELEMENT_ERROR (src, RESOURCE, SEEK, (NULL), - ("Requested seek from %" G_GINT64_FORMAT " to %" G_GINT64_FORMAT - " on non-seekable stream", src->curoffset, offset)); - return GST_FLOW_ERROR; - } read_failed: { gst_buffer_unref (buf); @@ -488,7 +357,6 @@ eos: } } -#if 0 static gboolean gst_rtmp_src_query (GstBaseSrc * basesrc, GstQuery * query) { @@ -500,6 +368,20 @@ gst_rtmp_src_query (GstBaseSrc * basesrc, GstQuery * query) gst_query_set_uri (query, src->uri); 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; + } default: ret = FALSE; break; @@ -510,7 +392,7 @@ gst_rtmp_src_query (GstBaseSrc * basesrc, GstQuery * query) return ret; } -#endif + static gboolean gst_rtmp_src_is_seekable (GstBaseSrc * basesrc) { @@ -518,135 +400,63 @@ gst_rtmp_src_is_seekable (GstBaseSrc * basesrc) src = GST_RTMP_SRC (basesrc); - return src->seekable; + return FALSE; } -#if 0 +/* open the file, do stuff necessary to go to PAUSED state */ static gboolean -gst_rtmp_src_check_get_range (GstBaseSrc * basesrc) +gst_rtmp_src_start (GstBaseSrc * basesrc) { GstRTMPSrc *src; - const gchar *protocol; src = GST_RTMP_SRC (basesrc); - if (src->uri == NULL) { - GST_WARNING_OBJECT (src, "no URI set yet"); - return FALSE; - } - - if (rtmp_uri_is_local (src->uri)) { - GST_LOG_OBJECT (src, "local URI (%s), assuming random access is possible", - GST_STR_NULL (src->uri_name)); - return TRUE; - } - - /* blacklist certain protocols we know won't work getrange-based */ - protocol = rtmp_uri_get_scheme (src->uri); - if (protocol == NULL) - goto undecided; - - if (strcmp (protocol, "http") == 0 || strcmp (protocol, "https") == 0) { - GST_LOG_OBJECT (src, "blacklisted protocol '%s', no random access possible" - " (URI=%s)", protocol, GST_STR_NULL (src->uri_name)); - return FALSE; - } - - /* fall through to undecided */ - -undecided: - { - /* don't know what to do, let the basesrc class decide for us */ - GST_LOG_OBJECT (src, "undecided about URI '%s', let base class handle it", - GST_STR_NULL (src->uri_name)); - - if (GST_BASE_SRC_CLASS (parent_class)->check_get_range) - return GST_BASE_SRC_CLASS (parent_class)->check_get_range (basesrc); - + if (!src->uri) { + GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL), ("No filename given")); return FALSE; } -} -#endif - -#if 0 -static gboolean -gst_rtmp_src_get_size (GstBaseSrc * basesrc, guint64 * size) -{ - GstRTMPSrc *src; - RTMPFileInfo *info; - RTMPFileInfoOptions options; - RTMPResult res; - src = GST_RTMP_SRC (basesrc); + src->curoffset = 0; - *size = -1; - info = rtmp_file_info_new (); - options = RTMP_FILE_INFO_DEFAULT | RTMP_FILE_INFO_FOLLOW_LINKS; - res = rtmp_get_file_info_from_handle (src->handle, info, options); - if (res == RTMP_OK) { - if ((info->valid_fields & RTMP_FILE_INFO_FIELDS_SIZE) != 0) { - *size = info->size; - GST_DEBUG_OBJECT (src, "from handle: %" G_GUINT64_FORMAT " bytes", *size); - } else if (src->own_handle && rtmp_uri_is_local (src->uri)) { - GST_DEBUG_OBJECT (src, - "file size not known, file local, trying fallback"); - res = rtmp_get_file_info_uri (src->uri, info, options); - if (res == RTMP_OK && - (info->valid_fields & RTMP_FILE_INFO_FIELDS_SIZE) != 0) { - *size = info->size; - GST_DEBUG_OBJECT (src, "from uri: %" G_GUINT64_FORMAT " bytes", *size); - } + /* open if required */ + if (!RTMP_IsConnected (src->rtmp)) { + if (!RTMP_Connect (src->rtmp, NULL)) { + GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL), + ("Could not connect to RTMP stream \"%s\" for reading: %s (%d)", + src->uri, "FIXME", 0)); + return FALSE; } - } else { - GST_WARNING_OBJECT (src, "getting info failed: %s", - rtmp_result_to_string (res)); } - rtmp_file_info_unref (info); - - if (*size == (RTMPFileSize) - 1) - return FALSE; - - GST_DEBUG_OBJECT (src, "return size %" G_GUINT64_FORMAT, *size); return TRUE; } -#endif -/* open the file, do stuff necessary to go to PAUSED state */ static gboolean -gst_rtmp_src_start (GstBaseSrc * basesrc) +gst_rtmp_src_stop (GstBaseSrc * basesrc) { GstRTMPSrc *src; src = GST_RTMP_SRC (basesrc); - g_message ("start called!"); +//FIXME you can't run RTMP_Close multiple times +// RTMP_Close (src->rtmp); - if (!src->uri) { - GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL), ("No filename given")); - return FALSE; - } + src->curoffset = 0; return TRUE; } static gboolean -gst_rtmp_src_stop (GstBaseSrc * basesrc) +plugin_init (GstPlugin * plugin) { - GstRTMPSrc *src; - - src = GST_RTMP_SRC (basesrc); - -//FIXME you can't run RTMP_Close multiple times -// RTMP_Close (src->rtmp); - - g_message ("stop called!"); - - src->curoffset = 0; + GST_DEBUG_CATEGORY_INIT (rtmpsrc_debug, "rtmpsrc", 0, "RTMP Source"); - return TRUE; + return gst_element_register (plugin, "rtmpsrc", GST_RANK_PRIMARY, + GST_TYPE_RTMP_SRC); } -/* - * vim: sw=2 ts=8 cindent noai bs=2 - */ +GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, + GST_VERSION_MINOR, + "rtmpsrc", + "RTMP source", + plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN); diff --git a/ext/rtmp/gstrtmpsrc.h b/ext/rtmp/gstrtmpsrc.h index c16ac90ef..19d171064 100644 --- a/ext/rtmp/gstrtmpsrc.h +++ b/ext/rtmp/gstrtmpsrc.h @@ -1,9 +1,10 @@ /* GStreamer * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> * 2000 Wim Taymans <wtay@chello.be> - * 2001 Bastien Nocera <hadess@hadess.net> * 2002 Kristian Rietveld <kris@gtk.org> * 2002,2003 Colin Walters <walters@gnu.org> + * 2001,2010 Bastien Nocera <hadess@hadess.net> + * 2010 Sebastian Dröge <sebastian.droege@collabora.co.uk> * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public @@ -25,6 +26,7 @@ #define __GST_RTMP_SRC_H__ #include <gst/base/gstbasesrc.h> +#include <gst/base/gstpushsrc.h> #include <librtmp/rtmp.h> #include <librtmp/log.h> @@ -53,19 +55,19 @@ typedef struct _GstRTMPSrcClass GstRTMPSrcClass; */ struct _GstRTMPSrc { - GstBaseSrc basesrc; - - char *uri; + GstPushSrc parent; + + /* < private > */ + gchar *uri; RTMP *rtmp; - gboolean seekable; gint64 curoffset; }; struct _GstRTMPSrcClass { - GstBaseSrcClass basesrc_class; + GstPushSrcClass parent; }; GType gst_rtmp_src_get_type (void); |