mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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653 lines
17 KiB
C
653 lines
17 KiB
C
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/* GStreamer
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* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
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* 2000 Wim Taymans <wtay@chello.be>
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* 2001 Bastien Nocera <hadess@hadess.net>
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* 2002 Kristian Rietveld <kris@gtk.org>
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* 2002,2003 Colin Walters <walters@gnu.org>
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*
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* rtmpsrc.c:
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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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:element-rtmpsrc
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*
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* This plugin reads data from a local or remote location specified
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* by an URI. This location can be specified using any protocol supported by
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* the RTMP library. Common protocols are 'file', 'http', 'ftp', or 'smb'.
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*
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* In case the #GstRTMPSrc:iradio-mode property is set and the
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* location is a http resource, rtmpsrc will send special icecast http
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* headers to the server to request additional icecast metainformation. If
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* the server is not an icecast server, it will display the same behaviour
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* as if the #GstRTMPSrc:iradio-mode property was not set. However,
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* if the server is in fact an icecast server, rtmpsrc will output
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* data with a media type of application/x-icy, in which case you will
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* need to use the #GstICYDemux element as follow-up element to extract
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* the icecast meta data and to determine the underlying media type.
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*
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* <refsect2>
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* <title>Example launch lines</title>
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* |[
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* gst-launch -v rtmpsrc location=file:///home/joe/foo.xyz ! fakesink
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* ]| The above pipeline will simply read a local file and do nothing with the
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* data read. Instead of rtmpsrc, we could just as well have used the
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* filesrc element here.
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* |[
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* gst-launch -v rtmpsrc location=smb://othercomputer/foo.xyz ! filesink location=/home/joe/foo.xyz
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* ]| The above pipeline will copy a file from a remote host to the local file
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* system using the Samba protocol.
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* |[
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* gst-launch -v rtmpsrc location=http://music.foobar.com/demo.mp3 ! mad ! audioconvert ! audioresample ! alsasink
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* ]| The above pipeline will read and decode and play an mp3 file from a
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* web server using the http protocol.
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* </refsect2>
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*/
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#define DEFAULT_RTMP_PORT 1935
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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 <glib/gi18n-lib.h>
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#include "gstrtmpsrc.h"
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#include <stdio.h>
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#include <stdlib.h>
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#include <sys/types.h>
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#include <sys/socket.h>
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#include <sys/time.h>
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#include <netinet/in.h>
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#include <arpa/inet.h>
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#include <netdb.h>
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#include <sys/stat.h>
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#include <fcntl.h>
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#include <unistd.h>
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#include <sys/mman.h>
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#include <errno.h>
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#include <string.h>
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#include <gst/gst.h>
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#include <gst/tag/tag.h>
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GST_DEBUG_CATEGORY_STATIC (rtmpsrc_debug);
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#define GST_CAT_DEFAULT rtmpsrc_debug
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static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS_ANY);
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enum
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{
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ARG_0,
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ARG_LOCATION,
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};
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static void gst_rtmp_src_base_init (gpointer g_class);
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static void gst_rtmp_src_class_init (GstRTMPSrcClass * klass);
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static void gst_rtmp_src_init (GstRTMPSrc * rtmpsrc);
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static void gst_rtmp_src_finalize (GObject * object);
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static void gst_rtmp_src_uri_handler_init (gpointer g_iface,
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gpointer iface_data);
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static void gst_rtmp_src_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_rtmp_src_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static gboolean gst_rtmp_src_stop (GstBaseSrc * src);
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static gboolean gst_rtmp_src_start (GstBaseSrc * src);
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static gboolean gst_rtmp_src_is_seekable (GstBaseSrc * src);
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#if 0
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static gboolean gst_rtmp_src_check_get_range (GstBaseSrc * src);
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static gboolean gst_rtmp_src_get_size (GstBaseSrc * src, guint64 * size);
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#endif
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static GstFlowReturn gst_rtmp_src_create (GstBaseSrc * basesrc,
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guint64 offset, guint size, GstBuffer ** buffer);
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#if 0
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static gboolean gst_rtmp_src_query (GstBaseSrc * src, GstQuery * query);
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#endif
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static GstElementClass *parent_class = NULL;
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static gboolean
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plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtmpsrc", GST_RANK_NONE,
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GST_TYPE_RTMP_SRC);
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}
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GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
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GST_VERSION_MINOR,
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"rtmpsrc",
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"flvstreamer sources",
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plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);
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GType
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gst_rtmp_src_get_type (void)
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{
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static GType rtmpsrc_type = 0;
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if (!rtmpsrc_type) {
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static const GTypeInfo rtmpsrc_info = {
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sizeof (GstRTMPSrcClass),
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gst_rtmp_src_base_init,
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NULL,
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(GClassInitFunc) gst_rtmp_src_class_init,
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NULL,
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NULL,
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sizeof (GstRTMPSrc),
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0,
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(GInstanceInitFunc) gst_rtmp_src_init,
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};
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static const GInterfaceInfo urihandler_info = {
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gst_rtmp_src_uri_handler_init,
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NULL,
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NULL
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};
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rtmpsrc_type =
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g_type_register_static (GST_TYPE_BASE_SRC,
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"GstRTMPSrc", &rtmpsrc_info, (GTypeFlags) 0);
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g_type_add_interface_static (rtmpsrc_type, GST_TYPE_URI_HANDLER,
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&urihandler_info);
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}
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return rtmpsrc_type;
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}
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static void
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gst_rtmp_src_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&srctemplate));
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gst_element_class_set_details_simple (element_class,
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"RTMP Source",
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"Source/File",
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"Read RTMP streams",
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"Bastien Nocera <hadess@hadess.net>\n"
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"GStreamer maintainers <gstreamer-devel@lists.sourceforge.net>");
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GST_DEBUG_CATEGORY_INIT (rtmpsrc_debug, "rtmpsrc", 0, "RTMP Source");
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}
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static void
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gst_rtmp_src_class_init (GstRTMPSrcClass * klass)
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{
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GObjectClass *gobject_class;
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GstBaseSrcClass *gstbasesrc_class;
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gobject_class = G_OBJECT_CLASS (klass);
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gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
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parent_class = (GstElementClass *) g_type_class_peek_parent (klass);
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gobject_class->finalize = gst_rtmp_src_finalize;
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gobject_class->set_property = gst_rtmp_src_set_property;
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gobject_class->get_property = gst_rtmp_src_get_property;
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/* properties */
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gst_element_class_install_std_props (GST_ELEMENT_CLASS (klass),
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"location", ARG_LOCATION, G_PARAM_READWRITE, NULL);
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gstbasesrc_class->start = GST_DEBUG_FUNCPTR (gst_rtmp_src_start);
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gstbasesrc_class->stop = GST_DEBUG_FUNCPTR (gst_rtmp_src_stop);
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#if 0
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gstbasesrc_class->get_size = GST_DEBUG_FUNCPTR (gst_rtmp_src_get_size);
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#endif
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gstbasesrc_class->is_seekable = GST_DEBUG_FUNCPTR (gst_rtmp_src_is_seekable);
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#if 0
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gstbasesrc_class->check_get_range =
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GST_DEBUG_FUNCPTR (gst_rtmp_src_check_get_range);
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#endif
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gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_rtmp_src_create);
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#if 0
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gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_rtmp_src_query);
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#endif
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}
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static void
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gst_rtmp_src_init (GstRTMPSrc * rtmpsrc)
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{
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rtmpsrc->curoffset = 0;
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rtmpsrc->seekable = FALSE;
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}
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static void
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gst_rtmp_src_finalize (GObject * object)
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{
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GstRTMPSrc *rtmpsrc = GST_RTMP_SRC (object);
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g_free (rtmpsrc->uri);
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rtmpsrc->uri = NULL;
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if (rtmpsrc->rtmp) {
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RTMP_Close (rtmpsrc->rtmp);
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RTMP_Free (rtmpsrc->rtmp);
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rtmpsrc->rtmp = NULL;
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}
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G_OBJECT_CLASS (parent_class)->finalize (object);
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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_src_uri_get_type (void)
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{
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return GST_URI_SRC;
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}
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static gchar **
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gst_rtmp_src_uri_get_protocols (void)
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{
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static gchar *protocols[] = { (char *) "rtmp", NULL };
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return protocols;
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}
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static const gchar *
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gst_rtmp_src_uri_get_uri (GstURIHandler * handler)
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{
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GstRTMPSrc *src = GST_RTMP_SRC (handler);
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return src->uri;
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}
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static gboolean
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gst_rtmp_src_uri_set_uri (GstURIHandler * handler, const gchar * uri)
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{
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GstRTMPSrc *src = GST_RTMP_SRC (handler);
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if (GST_STATE (src) == GST_STATE_PLAYING ||
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GST_STATE (src) == GST_STATE_PAUSED)
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return FALSE;
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g_object_set (G_OBJECT (src), "location", uri, NULL);
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g_message ("just set uri to %s", uri);
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return TRUE;
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}
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static void
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gst_rtmp_src_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_src_uri_get_type;
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iface->get_protocols = gst_rtmp_src_uri_get_protocols;
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iface->get_uri = gst_rtmp_src_uri_get_uri;
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iface->set_uri = gst_rtmp_src_uri_set_uri;
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}
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static void
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gst_rtmp_src_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstRTMPSrc *src;
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src = GST_RTMP_SRC (object);
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switch (prop_id) {
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case ARG_LOCATION:{
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char *new_location;
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/* the element must be stopped or paused in order to do this */
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if (GST_STATE (src) == GST_STATE_PLAYING ||
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GST_STATE (src) == GST_STATE_PAUSED)
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break;
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g_free (src->uri);
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src->uri = NULL;
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if (src->rtmp) {
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RTMP_Close (src->rtmp);
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RTMP_Free (src->rtmp);
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src->rtmp = NULL;
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}
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new_location = g_value_dup_string (value);
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src->rtmp = RTMP_Alloc ();
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RTMP_Init (src->rtmp);
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if (!RTMP_SetupURL (src->rtmp, new_location)) {
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GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, NULL,
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("Failed to setup URL '%s'", src->uri));
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g_free (new_location);
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RTMP_Free (src->rtmp);
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src->rtmp = NULL;
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} else {
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src->uri = g_value_dup_string (value);
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g_message ("parsed uri '%s' properly", src->uri);
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}
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break;
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}
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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 void
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gst_rtmp_src_get_property (GObject * object, guint prop_id, GValue * value,
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GParamSpec * pspec)
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{
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GstRTMPSrc *src;
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src = GST_RTMP_SRC (object);
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switch (prop_id) {
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case ARG_LOCATION:
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g_value_set_string (value, src->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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/*
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* Read a new buffer from src->reqoffset, takes care of events
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* and seeking and such.
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*/
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static GstFlowReturn
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gst_rtmp_src_create (GstBaseSrc * basesrc, guint64 offset, guint size,
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GstBuffer ** buffer)
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{
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GstRTMPSrc *src;
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GstBuffer *buf;
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guint8 *data;
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guint todo;
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int read;
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src = GST_RTMP_SRC (basesrc);
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g_return_val_if_fail (src->rtmp != NULL, GST_FLOW_ERROR);
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GST_DEBUG ("now at %" G_GINT64_FORMAT ", reading from %" G_GUINT64_FORMAT
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", size %u", src->curoffset, offset, size);
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/* open if required */
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if (G_UNLIKELY (!RTMP_IsConnected (src->rtmp))) {
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if (!RTMP_Connect (src->rtmp, NULL)) {
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GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL),
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("Could not connect to RTMP stream \"%s\" for reading: %s (%d)",
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src->uri, "FIXME", 0));
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return GST_FLOW_ERROR;
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}
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}
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/* seek if required */
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if (G_UNLIKELY (src->curoffset != offset)) {
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GST_DEBUG ("need to seek");
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if (src->seekable) {
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#if 0
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GST_DEBUG ("seeking to %" G_GUINT64_FORMAT, offset);
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res = rtmp_seek (src->handle, RTMP_SEEK_START, offset);
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if (res != RTMP_OK)
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||
|
goto seek_failed;
|
||
|
src->curoffset = offset;
|
||
|
#endif
|
||
|
} else {
|
||
|
goto cannot_seek;
|
||
|
}
|
||
|
}
|
||
|
|
||
|
buf = gst_buffer_try_new_and_alloc (size);
|
||
|
if (G_UNLIKELY (buf == NULL && size == 0)) {
|
||
|
GST_ERROR_OBJECT (src, "Failed to allocate %u bytes", size);
|
||
|
return GST_FLOW_ERROR;
|
||
|
}
|
||
|
|
||
|
data = GST_BUFFER_DATA (buf);
|
||
|
|
||
|
/* FIXME add FLV header first time around? */
|
||
|
read = 0;
|
||
|
|
||
|
todo = size;
|
||
|
while (todo > 0) {
|
||
|
read = RTMP_Read (src->rtmp, (char *) &data, todo);
|
||
|
|
||
|
if (G_UNLIKELY (read == -1))
|
||
|
goto eos;
|
||
|
|
||
|
if (G_UNLIKELY (read == -2))
|
||
|
goto read_failed;
|
||
|
|
||
|
/* FIXME handle -3 ? */
|
||
|
|
||
|
if (read < todo) {
|
||
|
data = &data[read];
|
||
|
todo -= read;
|
||
|
} else {
|
||
|
todo = 0;
|
||
|
}
|
||
|
GST_LOG (" got size %" G_GUINT64_FORMAT, read);
|
||
|
}
|
||
|
GST_BUFFER_OFFSET (buf) = src->curoffset;
|
||
|
src->curoffset += 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);
|
||
|
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
|
||
|
("Failed to read data: %s", "FIXME"));
|
||
|
return GST_FLOW_ERROR;
|
||
|
}
|
||
|
eos:
|
||
|
{
|
||
|
gst_buffer_unref (buf);
|
||
|
GST_DEBUG_OBJECT (src, "Reading data gave EOS");
|
||
|
return GST_FLOW_UNEXPECTED;
|
||
|
}
|
||
|
}
|
||
|
|
||
|
#if 0
|
||
|
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;
|
||
|
default:
|
||
|
ret = FALSE;
|
||
|
break;
|
||
|
}
|
||
|
|
||
|
if (!ret)
|
||
|
ret = GST_BASE_SRC_CLASS (parent_class)->query (basesrc, query);
|
||
|
|
||
|
return ret;
|
||
|
}
|
||
|
#endif
|
||
|
static gboolean
|
||
|
gst_rtmp_src_is_seekable (GstBaseSrc * basesrc)
|
||
|
{
|
||
|
GstRTMPSrc *src;
|
||
|
|
||
|
src = GST_RTMP_SRC (basesrc);
|
||
|
|
||
|
return src->seekable;
|
||
|
}
|
||
|
|
||
|
#if 0
|
||
|
static gboolean
|
||
|
gst_rtmp_src_check_get_range (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);
|
||
|
|
||
|
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);
|
||
|
|
||
|
*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);
|
||
|
}
|
||
|
}
|
||
|
} 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)
|
||
|
{
|
||
|
GstRTMPSrc *src;
|
||
|
|
||
|
src = GST_RTMP_SRC (basesrc);
|
||
|
|
||
|
g_message ("start called!");
|
||
|
|
||
|
if (!src->uri) {
|
||
|
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL), ("No filename given"));
|
||
|
return FALSE;
|
||
|
}
|
||
|
|
||
|
return TRUE;
|
||
|
}
|
||
|
|
||
|
static gboolean
|
||
|
gst_rtmp_src_stop (GstBaseSrc * basesrc)
|
||
|
{
|
||
|
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;
|
||
|
|
||
|
return TRUE;
|
||
|
}
|
||
|
|
||
|
/*
|
||
|
* vim: sw=2 ts=8 cindent noai bs=2
|
||
|
*/
|