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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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96b7059d24
rtmpsrc outputs truncated buffers because, when enough data is read to fill the buffer, the amount read that time (todo) is set to zero before it's added to the cumulative buffer size (bsize). The buffer is then truncated to bsize resulting in lost data. This patch adds todo to bsize before setting todo to zero. Fixes #678509
605 lines
15 KiB
C
605 lines
15 KiB
C
/* 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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* 2002 Kristian Rietveld <kris@gtk.org>
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* 2002,2003 Colin Walters <walters@gnu.org>
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* 2001,2010 Bastien Nocera <hadess@hadess.net>
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* 2010 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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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, i.e. rtmp, rtmpt, rtmps, rtmpe, rtmfp, rtmpte and rtmpts.
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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=rtmp://somehost/someurl ! fakesink
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* ]| Open an RTMP location and pass its content to fakesink.
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* </refsect2>
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*/
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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 <string.h>
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#include <gst/gst.h>
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#ifdef G_OS_WIN32
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#include <winsock2.h>
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#endif
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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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PROP_0,
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PROP_LOCATION
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#if 0
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PROP_SWF_URL,
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PROP_PAGE_URL
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#endif
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};
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#define DEFAULT_LOCATION NULL
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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 void gst_rtmp_src_finalize (GObject * object);
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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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static gboolean gst_rtmp_src_prepare_seek_segment (GstBaseSrc * src,
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GstEvent * event, GstSegment * segment);
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static gboolean gst_rtmp_src_do_seek (GstBaseSrc * src, GstSegment * segment);
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static GstFlowReturn gst_rtmp_src_create (GstPushSrc * pushsrc,
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GstBuffer ** buffer);
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static gboolean gst_rtmp_src_query (GstBaseSrc * src, GstQuery * query);
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#define gst_rtmp_src_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstRTMPSrc, gst_rtmp_src, GST_TYPE_PUSH_SRC,
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G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER,
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gst_rtmp_src_uri_handler_init));
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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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GstElementClass *gstelement_class;
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GstBaseSrcClass *gstbasesrc_class;
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GstPushSrcClass *gstpushsrc_class;
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gobject_class = G_OBJECT_CLASS (klass);
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gstelement_class = GST_ELEMENT_CLASS (klass);
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gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
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gstpushsrc_class = GST_PUSH_SRC_CLASS (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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g_object_class_install_property (gobject_class, PROP_LOCATION,
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g_param_spec_string ("location", "RTMP Location",
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"Location of the RTMP url to read",
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DEFAULT_LOCATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&srctemplate));
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gst_element_class_set_details_simple (gstelement_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>, "
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"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
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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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gstbasesrc_class->is_seekable = GST_DEBUG_FUNCPTR (gst_rtmp_src_is_seekable);
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gstbasesrc_class->prepare_seek_segment =
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GST_DEBUG_FUNCPTR (gst_rtmp_src_prepare_seek_segment);
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gstbasesrc_class->do_seek = GST_DEBUG_FUNCPTR (gst_rtmp_src_do_seek);
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gstpushsrc_class->create = GST_DEBUG_FUNCPTR (gst_rtmp_src_create);
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gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_rtmp_src_query);
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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_init (GstRTMPSrc * rtmpsrc)
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{
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#ifdef G_OS_WIN32
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WSADATA wsa_data;
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if (WSAStartup (MAKEWORD (2, 2), &wsa_data) != 0) {
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GST_ERROR_OBJECT (rtmpsrc, "WSAStartup failed: 0x%08x", WSAGetLastError ());
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}
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#endif
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rtmpsrc->cur_offset = 0;
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rtmpsrc->last_timestamp = 0;
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gst_base_src_set_format (GST_BASE_SRC (rtmpsrc), GST_FORMAT_TIME);
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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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#ifdef G_OS_WIN32
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WSACleanup ();
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#endif
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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 (GType type)
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{
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return GST_URI_SRC;
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}
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static const gchar *const *
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gst_rtmp_src_uri_get_protocols (GType type)
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{
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static const gchar *protocols[] =
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{ "rtmp", "rtmpt", "rtmps", "rtmpe", "rtmfp", "rtmpte", "rtmpts", NULL };
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return protocols;
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}
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static 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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/* FIXME: make thread-safe */
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return g_strdup (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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GError ** error)
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{
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GstRTMPSrc *src = GST_RTMP_SRC (handler);
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if (GST_STATE (src) >= GST_STATE_PAUSED) {
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g_set_error (error, GST_URI_ERROR, GST_URI_ERROR_BAD_STATE,
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"Changing the URI on rtmpsrc when it is running is not supported");
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return FALSE;
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}
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g_free (src->uri);
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src->uri = NULL;
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if (uri != NULL) {
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int protocol;
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AVal host;
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unsigned int port;
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AVal playpath, app;
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if (!RTMP_ParseURL (uri, &protocol, &host, &port, &playpath, &app) ||
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!host.av_len || !playpath.av_len) {
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GST_ERROR_OBJECT (src, "Failed to parse URI %s", uri);
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g_set_error (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
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"Could not parse RTMP URI");
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return FALSE;
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}
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src->uri = g_strdup (uri);
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}
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GST_DEBUG_OBJECT (src, "Changed URI to %s", GST_STR_NULL (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 PROP_LOCATION:{
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gst_rtmp_src_uri_set_uri (GST_URI_HANDLER (src),
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g_value_get_string (value), NULL);
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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 PROP_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 (GstPushSrc * pushsrc, GstBuffer ** buffer)
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{
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GstRTMPSrc *src;
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GstBuffer *buf;
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GstMapInfo map;
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guint8 *data;
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guint todo;
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gsize bsize;
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int read;
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int size;
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src = GST_RTMP_SRC (pushsrc);
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g_return_val_if_fail (src->rtmp != NULL, GST_FLOW_ERROR);
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size = GST_BASE_SRC_CAST (pushsrc)->blocksize;
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GST_DEBUG ("reading from %" G_GUINT64_FORMAT
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", size %u", src->cur_offset, size);
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buf = gst_buffer_new_allocate (NULL, size, NULL);
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if (G_UNLIKELY (buf == NULL)) {
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GST_ERROR_OBJECT (src, "Failed to allocate %u bytes", size);
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return GST_FLOW_ERROR;
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}
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bsize = todo = size;
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gst_buffer_map (buf, &map, GST_MAP_WRITE);
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data = map.data;
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read = bsize = 0;
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while (todo > 0) {
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read = RTMP_Read (src->rtmp, (char *) data, todo);
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if (G_UNLIKELY (read == 0 && todo == size)) {
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goto eos;
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} else if (G_UNLIKELY (read == 0)) {
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todo = 0;
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break;
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}
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if (G_UNLIKELY (read < 0))
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goto read_failed;
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if (read < todo) {
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data += read;
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todo -= read;
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bsize += read;
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} else {
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bsize += todo;
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todo = 0;
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}
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GST_LOG (" got size %d", read);
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}
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gst_buffer_unmap (buf, &map);
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gst_buffer_resize (buf, 0, bsize);
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if (src->discont) {
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GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
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src->discont = FALSE;
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}
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GST_BUFFER_TIMESTAMP (buf) = src->last_timestamp;
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GST_BUFFER_OFFSET (buf) = src->cur_offset;
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src->cur_offset += size;
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if (src->last_timestamp == GST_CLOCK_TIME_NONE)
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src->last_timestamp = src->rtmp->m_mediaStamp * GST_MSECOND;
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else
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src->last_timestamp =
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MAX (src->last_timestamp, src->rtmp->m_mediaStamp * GST_MSECOND);
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GST_LOG_OBJECT (src, "Created buffer of size %u at %" G_GINT64_FORMAT
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" with timestamp %" GST_TIME_FORMAT, size, GST_BUFFER_OFFSET (buf),
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GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
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/* we're done, return the buffer */
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*buffer = buf;
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return GST_FLOW_OK;
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read_failed:
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{
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gst_buffer_unref (buf);
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GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL), ("Failed to read data"));
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return GST_FLOW_ERROR;
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}
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eos:
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{
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gst_buffer_unref (buf);
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GST_DEBUG_OBJECT (src, "Reading data gave EOS");
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return GST_FLOW_EOS;
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}
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}
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static gboolean
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gst_rtmp_src_query (GstBaseSrc * basesrc, GstQuery * query)
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{
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gboolean ret = FALSE;
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GstRTMPSrc *src = GST_RTMP_SRC (basesrc);
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switch (GST_QUERY_TYPE (query)) {
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case GST_QUERY_URI:
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gst_query_set_uri (query, src->uri);
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ret = TRUE;
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break;
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case GST_QUERY_POSITION:{
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GstFormat format;
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gst_query_parse_position (query, &format, NULL);
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if (format == GST_FORMAT_TIME) {
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gst_query_set_duration (query, format, src->last_timestamp);
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ret = TRUE;
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}
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break;
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}
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case GST_QUERY_DURATION:{
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GstFormat format;
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gdouble duration;
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gst_query_parse_duration (query, &format, NULL);
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if (format == GST_FORMAT_TIME && src->rtmp) {
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duration = RTMP_GetDuration (src->rtmp);
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if (duration != 0.0) {
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gst_query_set_duration (query, format, duration * GST_SECOND);
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ret = TRUE;
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}
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}
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break;
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}
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default:
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ret = FALSE;
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break;
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}
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if (!ret)
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ret = GST_BASE_SRC_CLASS (parent_class)->query (basesrc, query);
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return ret;
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}
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static gboolean
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gst_rtmp_src_is_seekable (GstBaseSrc * basesrc)
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{
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GstRTMPSrc *src;
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src = GST_RTMP_SRC (basesrc);
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return src->seekable;
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}
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static gboolean
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gst_rtmp_src_prepare_seek_segment (GstBaseSrc * basesrc, GstEvent * event,
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GstSegment * segment)
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{
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GstRTMPSrc *src;
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GstSeekType cur_type, stop_type;
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gint64 cur, stop;
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GstSeekFlags flags;
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GstFormat format;
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gdouble rate;
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src = GST_RTMP_SRC (basesrc);
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gst_event_parse_seek (event, &rate, &format, &flags,
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&cur_type, &cur, &stop_type, &stop);
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if (!src->seekable) {
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GST_LOG_OBJECT (src, "Not a seekable stream");
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return FALSE;
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}
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if (!src->rtmp) {
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GST_LOG_OBJECT (src, "Not connected yet");
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return FALSE;
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}
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if (format != GST_FORMAT_TIME) {
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GST_LOG_OBJECT (src, "Seeking only supported in TIME format");
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return FALSE;
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}
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if (stop_type != GST_SEEK_TYPE_NONE) {
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GST_LOG_OBJECT (src, "Setting a stop position is not supported");
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return FALSE;
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}
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gst_segment_init (segment, GST_FORMAT_TIME);
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gst_segment_do_seek (segment, rate, format, flags, cur_type, cur, stop_type,
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stop, NULL);
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return TRUE;
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}
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static gboolean
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gst_rtmp_src_do_seek (GstBaseSrc * basesrc, GstSegment * segment)
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{
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GstRTMPSrc *src;
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|
|
|
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->seekable) {
|
|
GST_LOG_OBJECT (src, "Not a seekable stream");
|
|
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;
|
|
|
|
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;
|
|
gchar *uri_copy;
|
|
|
|
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->seekable = TRUE;
|
|
src->discont = TRUE;
|
|
|
|
uri_copy = g_strdup (src->uri);
|
|
src->rtmp = RTMP_Alloc ();
|
|
RTMP_Init (src->rtmp);
|
|
if (!RTMP_SetupURL (src->rtmp, uri_copy)) {
|
|
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
|
|
("Failed to setup URL '%s'", src->uri));
|
|
g_free (uri_copy);
|
|
RTMP_Free (src->rtmp);
|
|
src->rtmp = NULL;
|
|
return FALSE;
|
|
}
|
|
|
|
/* 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));
|
|
RTMP_Free (src->rtmp);
|
|
src->rtmp = NULL;
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
#undef STR2AVAL
|
|
|
|
static gboolean
|
|
gst_rtmp_src_stop (GstBaseSrc * basesrc)
|
|
{
|
|
GstRTMPSrc *src;
|
|
|
|
src = GST_RTMP_SRC (basesrc);
|
|
|
|
if (src->rtmp) {
|
|
RTMP_Close (src->rtmp);
|
|
RTMP_Free (src->rtmp);
|
|
src->rtmp = NULL;
|
|
}
|
|
|
|
src->cur_offset = 0;
|
|
src->last_timestamp = 0;
|
|
src->discont = TRUE;
|
|
|
|
return TRUE;
|
|
}
|