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196 lines
6.9 KiB
C
196 lines
6.9 KiB
C
/* GStreamer
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* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#include <gst/gst.h>
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#include <gst/app/app.h>
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#include <gst/rtsp-server/rtsp-server.h>
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typedef struct
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{
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GstElement *generator_pipe;
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GstElement *vid_appsink;
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GstElement *vid_appsrc;
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GstElement *aud_appsink;
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GstElement *aud_appsrc;
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} MyContext;
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/* called when we need to give data to an appsrc */
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static void
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need_data (GstElement * appsrc, guint unused, MyContext * ctx)
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{
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GstSample *sample;
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GstFlowReturn ret;
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if (appsrc == ctx->vid_appsrc)
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sample = gst_app_sink_pull_sample (GST_APP_SINK (ctx->vid_appsink));
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else
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sample = gst_app_sink_pull_sample (GST_APP_SINK (ctx->aud_appsink));
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if (sample) {
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GstBuffer *buffer = gst_sample_get_buffer (sample);
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GstSegment *seg = gst_sample_get_segment (sample);
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GstClockTime pts, dts;
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/* Convert the PTS/DTS to running time so they start from 0 */
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pts = GST_BUFFER_PTS (buffer);
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if (GST_CLOCK_TIME_IS_VALID (pts))
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pts = gst_segment_to_running_time (seg, GST_FORMAT_TIME, pts);
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dts = GST_BUFFER_DTS (buffer);
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if (GST_CLOCK_TIME_IS_VALID (dts))
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dts = gst_segment_to_running_time (seg, GST_FORMAT_TIME, dts);
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if (buffer) {
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/* Make writable so we can adjust the timestamps */
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buffer = gst_buffer_copy (buffer);
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GST_BUFFER_PTS (buffer) = pts;
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GST_BUFFER_DTS (buffer) = dts;
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g_signal_emit_by_name (appsrc, "push-buffer", buffer, &ret);
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}
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/* we don't need the appsink sample anymore */
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gst_sample_unref (sample);
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}
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}
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static void
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ctx_free (MyContext * ctx)
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{
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gst_element_set_state (ctx->generator_pipe, GST_STATE_NULL);
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gst_object_unref (ctx->generator_pipe);
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gst_object_unref (ctx->vid_appsrc);
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gst_object_unref (ctx->vid_appsink);
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gst_object_unref (ctx->aud_appsrc);
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gst_object_unref (ctx->aud_appsink);
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g_free (ctx);
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}
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/* called when a new media pipeline is constructed. We can query the
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* pipeline and configure our appsrc */
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static void
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media_configure (GstRTSPMediaFactory * factory, GstRTSPMedia * media,
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gpointer user_data)
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{
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GstElement *element, *appsrc, *appsink;
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GstCaps *caps;
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MyContext *ctx;
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ctx = g_new0 (MyContext, 1);
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/* This pipeline generates H264 video and PCM audio. The appsinks are kept small so that if delivery is slow,
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* encoded buffers are dropped as needed. There's slightly more buffers (32) allowed for audio */
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ctx->generator_pipe =
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gst_parse_launch
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("videotestsrc is-live=true ! x264enc speed-preset=superfast tune=zerolatency ! h264parse ! appsink name=vid max-buffers=1 drop=true "
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"audiotestsrc is-live=true ! appsink name=aud max-buffers=32 drop=true",
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NULL);
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/* make sure the data is freed when the media is gone */
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g_object_set_data_full (G_OBJECT (media), "rtsp-extra-data", ctx,
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(GDestroyNotify) ctx_free);
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/* get the element (bin) used for providing the streams of the media */
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element = gst_rtsp_media_get_element (media);
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/* Find the 2 app sources (video / audio), and configure them, connect to the
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* signals to request data */
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/* configure the caps of the video */
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caps = gst_caps_new_simple ("video/x-h264",
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"stream-format", G_TYPE_STRING, "byte-stream",
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"alignment", G_TYPE_STRING, "au",
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"width", G_TYPE_INT, 384, "height", G_TYPE_INT, 288,
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"framerate", GST_TYPE_FRACTION, 15, 1, NULL);
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ctx->vid_appsrc = appsrc =
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gst_bin_get_by_name_recurse_up (GST_BIN (element), "videosrc");
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ctx->vid_appsink = appsink =
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gst_bin_get_by_name (GST_BIN (ctx->generator_pipe), "vid");
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gst_util_set_object_arg (G_OBJECT (appsrc), "format", "time");
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g_object_set (G_OBJECT (appsrc), "caps", caps, NULL);
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g_object_set (G_OBJECT (appsink), "caps", caps, NULL);
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/* install the callback that will be called when a buffer is needed */
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g_signal_connect (appsrc, "need-data", (GCallback) need_data, ctx);
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gst_caps_unref (caps);
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caps = gst_caps_new_simple ("audio/x-raw", "format", G_TYPE_STRING, "S24BE",
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"layout", G_TYPE_STRING, "interleaved", "rate", G_TYPE_INT, 48000,
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"channels", G_TYPE_INT, 2, NULL);
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ctx->aud_appsrc = appsrc =
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gst_bin_get_by_name_recurse_up (GST_BIN (element), "audiosrc");
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ctx->aud_appsink = appsink =
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gst_bin_get_by_name (GST_BIN (ctx->generator_pipe), "aud");
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gst_util_set_object_arg (G_OBJECT (appsrc), "format", "time");
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g_object_set (G_OBJECT (appsrc), "caps", caps, NULL);
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g_object_set (G_OBJECT (appsink), "caps", caps, NULL);
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g_signal_connect (appsrc, "need-data", (GCallback) need_data, ctx);
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gst_caps_unref (caps);
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gst_element_set_state (ctx->generator_pipe, GST_STATE_PLAYING);
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gst_object_unref (element);
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}
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int
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main (int argc, char *argv[])
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{
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GMainLoop *loop;
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GstRTSPServer *server;
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GstRTSPMountPoints *mounts;
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GstRTSPMediaFactory *factory;
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gst_init (&argc, &argv);
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loop = g_main_loop_new (NULL, FALSE);
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/* create a server instance */
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server = gst_rtsp_server_new ();
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/* get the mount points for this server, every server has a default object
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* that be used to map uri mount points to media factories */
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mounts = gst_rtsp_server_get_mount_points (server);
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/* make a media factory for a test stream. The default media factory can use
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* gst-launch syntax to create pipelines.
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* any launch line works as long as it contains elements named pay%d. Each
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* element with pay%d names will be a stream */
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factory = gst_rtsp_media_factory_new ();
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gst_rtsp_media_factory_set_launch (factory,
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"( appsrc name=videosrc ! h264parse ! rtph264pay name=pay0 pt=96 "
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" appsrc name=audiosrc ! audioconvert ! rtpL24pay name=pay1 pt=97 )");
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/* notify when our media is ready, This is called whenever someone asks for
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* the media and a new pipeline with our appsrc is created */
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g_signal_connect (factory, "media-configure", (GCallback) media_configure,
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NULL);
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/* attach the test factory to the /test url */
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gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
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/* don't need the ref to the mounts anymore */
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g_object_unref (mounts);
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/* attach the server to the default maincontext */
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gst_rtsp_server_attach (server, NULL);
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/* start serving */
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g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
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g_main_loop_run (loop);
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return 0;
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}
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