tests: add appsrc example

Add an example on how to use appsrc to feed the server pipeline with data.
2024-12-23 16:50:47 +00:00 · 2013-08-23 15:14:29 +02:00 · 2013-08-23 15:14:29 +02:00 · 94ed18008a
commit 94ed18008a
parent ff10d24130
1 changed files with 137 additions and 0 deletions
--- a/examples/test-appsrc.c
+++ b/examples/test-appsrc.c
@ -0,0 +1,137 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+typedef struct
+{
+  gboolean white;
+  GstClockTime timestamp;
+} MyContext;
+
+/* called when we need to give data to appsrc */
+static void
+need_data (GstElement * appsrc, guint unused, MyContext * ctx)
+{
+  GstBuffer *buffer;
+  guint size;
+  GstFlowReturn ret;
+
+  size = 385 * 288 * 2;
+
+  buffer = gst_buffer_new_allocate (NULL, size, NULL);
+
+  /* this makes the image black/white */
+  gst_buffer_memset (buffer, 0, ctx->white ? 0xff : 0x0, size);
+
+  ctx->white = !ctx->white;
+
+  /* increment the timestamp every 1/2 second */
+  GST_BUFFER_PTS (buffer) = ctx->timestamp;
+  GST_BUFFER_DURATION (buffer) = gst_util_uint64_scale_int (1, GST_SECOND, 2);
+  ctx->timestamp += GST_BUFFER_DURATION (buffer);
+
+  g_signal_emit_by_name (appsrc, "push-buffer", buffer, &ret);
+}
+
+/* called when a new media pipeline is constructed. We can query the
+ * pipeline and configure our appsrc */
+static void
+media_configure (GstRTSPMediaFactory * factory, GstRTSPMedia * media,
+    gpointer user_data)
+{
+  GstElement *element, *appsrc;
+  MyContext *ctx;
+
+  /* get the element used for providing the streams of the media */
+  element = gst_rtsp_media_get_element (media);
+
+  /* get our appsrc, we named it 'mysrc' with the name property */
+  appsrc = gst_bin_get_by_name_recurse_up (GST_BIN (element), "mysrc");
+
+  /* this instructs appsrc that we will be dealing with timed buffer */
+  gst_util_set_object_arg (G_OBJECT (appsrc), "format", "time");
+  /* configure the caps of the video */
+  g_object_set (G_OBJECT (appsrc), "caps",
+      gst_caps_new_simple ("video/x-raw",
+          "format", G_TYPE_STRING, "RGB16",
+          "width", G_TYPE_INT, 384,
+          "height", G_TYPE_INT, 288,
+          "framerate", GST_TYPE_FRACTION, 0, 1, NULL), NULL);
+
+  ctx = g_new0 (MyContext, 1);
+  ctx->white = FALSE;
+  ctx->timestamp = 0;
+  /* make sure ther datais freed when the media is gone */
+  g_object_set_data_full (G_OBJECT (media), "my-extra-data", ctx,
+      (GDestroyNotify) g_free);
+
+  /* install the callback that will be called when a buffer is needed */
+  g_signal_connect (appsrc, "need-data", (GCallback) need_data, ctx);
+}
+
+int
+main (int argc, char *argv[])
+{
+  GMainLoop *loop;
+  GstRTSPServer *server;
+  GstRTSPMountPoints *mounts;
+  GstRTSPMediaFactory *factory;
+
+  gst_init (&argc, &argv);
+
+  loop = g_main_loop_new (NULL, FALSE);
+
+  /* create a server instance */
+  server = gst_rtsp_server_new ();
+
+  /* get the mount points for this server, every server has a default object
+   * that be used to map uri mount points to media factories */
+  mounts = gst_rtsp_server_get_mount_points (server);
+
+  /* make a media factory for a test stream. The default media factory can use
+   * gst-launch syntax to create pipelines.
+   * any launch line works as long as it contains elements named pay%d. Each
+   * element with pay%d names will be a stream */
+  factory = gst_rtsp_media_factory_new ();
+  gst_rtsp_media_factory_set_launch (factory,
+      "( appsrc name=mysrc ! videoconvert ! x264enc ! rtph264pay name=pay0 pt=96 )");
+
+  /* notify when our media is ready, This is called whenever someone asks for
+   * the media and a new pipeline with our appsrc is created */
+  g_signal_connect (factory, "media-configure", (GCallback) media_configure,
+      NULL);
+
+  /* attach the test factory to the /test url */
+  gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+  /* don't need the ref to the mounts anymore */
+  g_object_unref (mounts);
+
+  /* attach the server to the default maincontext */
+  gst_rtsp_server_attach (server, NULL);
+
+  /* start serving */
+  g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
+  g_main_loop_run (loop);
+
+  return 0;
+}