/* GStreamer * Copyright (C) 2008 Wim Taymans * * 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 #include #include 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); gst_buffer_unref (buffer); } /* 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; }