--- /dev/null
+/* 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;
+}
projects / platform / upstream / gstreamer.git / commitdiff