Gstreamer: trickplay mode in rtsp-server
Igor
igor.vl.bondarenko at gmail.com
Tue Sep 4 10:08:22 UTC 2018
Here working example of trickplay mode for rtsp-server. Thanks to Olivier
CrĂȘte for suggestions!
seek-event should be passed into MediaElement, and appsrc should be seekable
with seek-data connected.
///////////////////////////////////////////////////////////////////////
#include <gst/gst.h>
#include <gst/rtsp-server/rtsp-server.h>
#include <string>
#include <fstream>
static GstElement *pMediaElement = NULL;
/* this timeout is periodically run to clean up the expired sessions from
the
* pool. This needs to be run explicitly currently but might be done
* automatically as part of the mainloop. */
static gboolean
timeout(GstRTSPServer * server)
{
GstRTSPSessionPool *pool;
pool = gst_rtsp_server_get_session_pool(server);
gst_rtsp_session_pool_cleanup(pool);
g_object_unref(pool);
return TRUE;
}
static int sFileSize(const std::string &filename)
{
std::ifstream in(filename, std::ifstream::ate | std::ifstream::binary);
return in.tellg();
}
static void onNeedVideoData(GstElement * appsrc)
{
static int NN = 0;
++NN;
std::string filename = "C:\\rtsp_files\\body" + std::to_string(NN) +
".bin";
int Size = sFileSize(filename);
GstBuffer* buf = gst_buffer_new_and_alloc(Size);
GstMapInfo map;
gst_buffer_map(buf, &map, GST_MAP_WRITE);
FILE *fp = fopen(filename.c_str(), "rb");
fread(map.data, sizeof(unsigned char), Size, fp);
fclose(fp);
gst_buffer_unmap(buf, &map);
//in random moment we send seek-event to MediaElement
if (NN == 300){
gint64 position;
if (!gst_element_query_position(pMediaElement, GST_FORMAT_TIME,
&position)) {
g_printerr("Unable to retrieve current position.\n");
return;
}
GstEvent * seek_event = gst_event_new_seek(4., GST_FORMAT_TIME,
(GstSeekFlags)(GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE),
GST_SEEK_TYPE_SET, position, GST_SEEK_TYPE_NONE, 0);
auto res = gst_element_send_event(pMediaElement, seek_event);
g_print("%d\n", res);
}
GstFlowReturn ret;
g_signal_emit_by_name(appsrc, "push-buffer", buf, &ret);
gst_buffer_unref(buf);
}
static void need_video_data(GstElement * appsrc, guint unused)
{
onNeedVideoData(appsrc);
}
/* called when appsrc wants us to return data from a new position with the
next
* call to push-buffer. */
static gboolean
seek_data(GstElement * appsrc, guint64 position)
{
g_print("seek_data call\n");
//GST_DEBUG("seek to offset %" G_GUINT64_FORMAT, position);
//app->offset = position;
return TRUE;
}
static void
media_constructed(GstRTSPMediaFactory * factory, GstRTSPMedia * media)
{
GstElement* element = pMediaElement = gst_rtsp_media_get_element(media);
GstElement* vsrc = gst_bin_get_by_name_recurse_up(GST_BIN(element),
"vsrc");
gst_util_set_object_arg(G_OBJECT(vsrc), "stream-type", "seekable");
g_signal_connect(vsrc, "need-data", (GCallback)need_video_data, NULL);
g_signal_connect(vsrc, "seek-data", G_CALLBACK(seek_data), NULL);
}
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=vsrc !"
"h264parse config-interval=1 ! rtph264pay pt=96 name=pay0 )");
gst_rtsp_media_factory_set_shared(factory, TRUE);
g_signal_connect(factory, "media-constructed", (GCallback)
media_constructed, 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 mapper anymore */
g_object_unref(mounts);
/* attach the server to the default maincontext */
if (gst_rtsp_server_attach(server, NULL) == 0)
goto failed;
/* add a timeout for the session cleanup */
g_timeout_add_seconds(2, (GSourceFunc)timeout, server);
/* start serving, this never stops */
g_print("stream ready at rtsp://127.0.0.1:8554/test\n");
g_main_loop_run(loop);
return 0;
/* ERRORS */
failed:
{
g_print("failed to attach the server\n");
return -1;
}
}
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