examples: Add test-appsrc2

Add an example of feeding both audio and video into an RTSP
pipeline via appsrc.

examples/.gitignore

@@ -1,4 +1,5 @@
 test-appsrc
+test-appsrc2
 test-cgroups
 test-launch
 test-mp4

examples/Makefile.am

@@ -1,6 +1,6 @@
 noinst_PROGRAMS = test-video test-ogg test-mp4 test-readme \
 		  test-launch test-sdp test-uri test-auth test-auth-digest \
-		  test-multicast test-multicast2 test-appsrc \
+		  test-multicast test-multicast2 test-appsrc test-appsrc2 \
 		  test-video-rtx test-record test-record-auth \
 		  test-netclock test-netclock-client \
 		  test-onvif-backchannel test-video-disconnect

examples/test-appsrc2.c

@@ -0,0 +1,196 @@
+/* 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/app/app.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+typedef struct
+{
+  GstElement *generator_pipe;
+  GstElement *vid_appsink;
+  GstElement *vid_appsrc;
+  GstElement *aud_appsink;
+  GstElement *aud_appsrc;
+} MyContext;
+
+/* called when we need to give data to an appsrc */
+static void
+need_data (GstElement * appsrc, guint unused, MyContext * ctx)
+{
+  GstSample *sample;
+  GstFlowReturn ret;
+
+  if (appsrc == ctx->vid_appsrc)
+    sample = gst_app_sink_pull_sample (GST_APP_SINK (ctx->vid_appsink));
+  else
+    sample = gst_app_sink_pull_sample (GST_APP_SINK (ctx->aud_appsink));
+
+  if (sample) {
+    GstBuffer *buffer = gst_sample_get_buffer (sample);
+    GstSegment *seg = gst_sample_get_segment (sample);
+    GstClockTime pts, dts;
+
+    /* Convert the PTS/DTS to running time so they start from 0 */
+    pts = GST_BUFFER_PTS (buffer);
+    if (GST_CLOCK_TIME_IS_VALID (pts))
+      pts = gst_segment_to_running_time (seg, GST_FORMAT_TIME, pts);
+
+    dts = GST_BUFFER_DTS (buffer);
+    if (GST_CLOCK_TIME_IS_VALID (dts))
+      dts = gst_segment_to_running_time (seg, GST_FORMAT_TIME, dts);
+
+    if (buffer) {
+      /* Make writable so we can adjust the timestamps */
+      buffer = gst_buffer_copy (buffer);
+      GST_BUFFER_PTS (buffer) = pts;
+      GST_BUFFER_DTS (buffer) = dts;
+      g_signal_emit_by_name (appsrc, "push-buffer", buffer, &ret);
+    }
+
+    /* we don't need the appsink sample anymore */
+    gst_sample_unref (sample);
+  }
+}
+
+static void
+ctx_free (MyContext * ctx)
+{
+  gst_element_set_state (ctx->generator_pipe, GST_STATE_NULL);
+
+  gst_object_unref (ctx->generator_pipe);
+  gst_object_unref (ctx->vid_appsrc);
+  gst_object_unref (ctx->vid_appsink);
+  gst_object_unref (ctx->aud_appsrc);
+  gst_object_unref (ctx->aud_appsink);
+
+  g_free (ctx);
+}
+
+/* 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, *appsink;
+  GstCaps *caps;
+  MyContext *ctx;
+
+  ctx = g_new0 (MyContext, 1);
+  /* This pipeline generates H264 video and PCM audio. The appsinks are kept small so that if delivery is slow,
+   * encoded buffers are dropped as needed. There's slightly more buffers (32) allowed for audio */
+  ctx->generator_pipe =
+      gst_parse_launch
+      ("videotestsrc is-live=true ! x264enc speed-preset=superfast tune=zerolatency ! h264parse ! appsink name=vid max-buffers=1 drop=true "
+      "audiotestsrc is-live=true ! appsink name=aud max-buffers=32 drop=true",
+      NULL);
+
+  /* make sure the data is freed when the media is gone */
+  g_object_set_data_full (G_OBJECT (media), "rtsp-extra-data", ctx,
+      (GDestroyNotify) ctx_free);
+
+  /* get the element (bin) used for providing the streams of the media */
+  element = gst_rtsp_media_get_element (media);
+
+  /* Find the 2 app sources (video / audio), and configure them, connect to the
+   * signals to request data */
+  /* configure the caps of the video */
+  caps = gst_caps_new_simple ("video/x-h264",
+      "stream-format", G_TYPE_STRING, "byte-stream",
+      "alignment", G_TYPE_STRING, "au",
+      "width", G_TYPE_INT, 384, "height", G_TYPE_INT, 288,
+      "framerate", GST_TYPE_FRACTION, 15, 1, NULL);
+  ctx->vid_appsrc = appsrc =
+      gst_bin_get_by_name_recurse_up (GST_BIN (element), "videosrc");
+  ctx->vid_appsink = appsink =
+      gst_bin_get_by_name (GST_BIN (ctx->generator_pipe), "vid");
+  gst_util_set_object_arg (G_OBJECT (appsrc), "format", "time");
+  g_object_set (G_OBJECT (appsrc), "caps", caps, NULL);
+  g_object_set (G_OBJECT (appsink), "caps", caps, NULL);
+  /* install the callback that will be called when a buffer is needed */
+  g_signal_connect (appsrc, "need-data", (GCallback) need_data, ctx);
+  gst_caps_unref (caps);
+
+  caps = gst_caps_new_simple ("audio/x-raw", "format", G_TYPE_STRING, "S24BE",
+      "layout", G_TYPE_STRING, "interleaved", "rate", G_TYPE_INT, 48000,
+      "channels", G_TYPE_INT, 2, NULL);
+  ctx->aud_appsrc = appsrc =
+      gst_bin_get_by_name_recurse_up (GST_BIN (element), "audiosrc");
+  ctx->aud_appsink = appsink =
+      gst_bin_get_by_name (GST_BIN (ctx->generator_pipe), "aud");
+  gst_util_set_object_arg (G_OBJECT (appsrc), "format", "time");
+  g_object_set (G_OBJECT (appsrc), "caps", caps, NULL);
+  g_object_set (G_OBJECT (appsink), "caps", caps, NULL);
+  g_signal_connect (appsrc, "need-data", (GCallback) need_data, ctx);
+  gst_caps_unref (caps);
+
+  gst_element_set_state (ctx->generator_pipe, GST_STATE_PLAYING);
+  gst_object_unref (element);
+}
+
+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=videosrc ! h264parse ! rtph264pay name=pay0 pt=96 "
+      "  appsrc name=audiosrc ! audioconvert ! rtpL24pay name=pay1 pt=97 )");
+
+  /* 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;
+}