rtmp: Move to ext and drop internal librtmp copy

We really don't want this in gst-plugins-bad because of
legal complexities around RTMP and possible problems
for distributions.

Add README that explains how to build librtmp to be suitable
for linking to the GStreamer plugin.

1 files changed, 652 insertions, 0 deletions

diff --git a/ext/rtmp/gstrtmpsrc.c b/ext/rtmp/gstrtmpsrc.c
new file mode 100644
index 000000000..c8aceb5fc
--- /dev/null
+++ b/ext/rtmp/gstrtmpsrc.c
@@ -0,0 +1,652 @@
+/* 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>
+ *
+ * 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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+/**
+ * SECTION:element-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. 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.
+ *
+ * <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.
+ * </refsect2>
+ */
+
+#define DEFAULT_RTMP_PORT 1935
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <glib/gi18n-lib.h>
+
+#include "gstrtmpsrc.h"
+
+#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
+
+static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS_ANY);
+
+enum
+{
+  ARG_0,
+  ARG_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);
+
+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 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 gboolean gst_rtmp_src_query (GstBaseSrc * src, GstQuery * query);
+#endif
+
+static GstElementClass *parent_class = NULL;
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+  return gst_element_register (plugin, "rtmpsrc", GST_RANK_NONE,
+      GST_TYPE_RTMP_SRC);
+}
+
+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;
+}
+
+static void
+gst_rtmp_src_base_init (gpointer g_class)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&srctemplate));
+
+  gst_element_class_set_details_simple (element_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");
+}
+
+static void
+gst_rtmp_src_class_init (GstRTMPSrcClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstBaseSrcClass *gstbasesrc_class;
+
+  gobject_class = G_OBJECT_CLASS (klass);
+  gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
+
+  parent_class = (GstElementClass *) g_type_class_peek_parent (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 */
+  gst_element_class_install_std_props (GST_ELEMENT_CLASS (klass),
+      "location", ARG_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
+  gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_rtmp_src_query);
+#endif
+}
+
+static void
+gst_rtmp_src_init (GstRTMPSrc * rtmpsrc)
+{
+  rtmpsrc->curoffset = 0;
+  rtmpsrc->seekable = FALSE;
+}
+
+static void
+gst_rtmp_src_finalize (GObject * object)
+{
+  GstRTMPSrc *rtmpsrc = GST_RTMP_SRC (object);
+
+  g_free (rtmpsrc->uri);
+  rtmpsrc->uri = NULL;
+
+  if (rtmpsrc->rtmp) {
+    RTMP_Close (rtmpsrc->rtmp);
+    RTMP_Free (rtmpsrc->rtmp);
+    rtmpsrc->rtmp = NULL;
+  }
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+/*
+ * URI interface support.
+ */
+
+static GstURIType
+gst_rtmp_src_uri_get_type (void)
+{
+  return GST_URI_SRC;
+}
+
+static gchar **
+gst_rtmp_src_uri_get_protocols (void)
+{
+  static gchar *protocols[] = { (char *) "rtmp", NULL };
+  return protocols;
+}
+
+static const gchar *
+gst_rtmp_src_uri_get_uri (GstURIHandler * handler)
+{
+  GstRTMPSrc *src = GST_RTMP_SRC (handler);
+
+  return src->uri;
+}
+
+static gboolean
+gst_rtmp_src_uri_set_uri (GstURIHandler * handler, const gchar * uri)
+{
+  GstRTMPSrc *src = GST_RTMP_SRC (handler);
+
+  if (GST_STATE (src) == GST_STATE_PLAYING ||
+      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);
+
+  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 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);
+      }
+      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 ARG_LOCATION:
+      g_value_set_string (value, src->uri);
+      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 (GstBaseSrc * basesrc, guint64 offset, guint size,
+    GstBuffer ** buffer)
+{
+  GstRTMPSrc *src;
+  GstBuffer *buf;
+  guint8 *data;
+  guint todo;
+  int read;
+
+  src = GST_RTMP_SRC (basesrc);
+
+  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);
+
+  /* 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;
+    }
+  }
+
+  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
+ */