gst-launch-1.0 --gst-debug-level=3 dvbsrc modulation='QAM 256'
frequency=147000000 ! decodebin name=demux demux. ! queue !
audioresample ! audioconvert ! voaacenc ! mux. mpegtsmux name=mux
! udpsink host=127.0.0.0 port=22 demux. ! queue ! videoconvert !
videoscale ! x264enc bitrate=1240 tune=zerolatency
!video/x-h264,stream-format=byte-stream,profile=high,width=540,height=380,key-int-max=15
! mux.
I want to convert into c with the same thing.
I am trying -:
#include <gst/gst.h>
#include <glib.h>
static gboolean bus_call (GstBus *bus, GstMessage *msg, gpointer data)
{
GMainLoop *loop = (GMainLoop *) data;
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_EOS:
g_print ("End of stream\n");
g_main_loop_quit (loop);
break;
case GST_MESSAGE_ERROR: {
gchar *debug;
GError *error;
gst_message_parse_error (msg, &error, &debug);
g_free (debug);
g_printerr ("Error: %s\n", error->message);
g_error_free (error);
g_main_loop_quit (loop);
break;
}
default:
break;
}
return TRUE;
}
int main (gint argc, gchar *argv[])
{
GMainLoop *loop;
GstElement *pipeline, *source, *demuxer,*dec,*vdqueue,*adqueue, *conv, *sink;
GstBus *bus;
/* init GStreamer */
gst_init (&argc, &argv);
loop = g_main_loop_new (NULL, FALSE);
/* setup */
pipeline = gst_pipeline_new ("pipeline");
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
gst_bus_add_watch (bus, bus_call, loop);
gst_object_unref (bus);
source = gst_element_factory_make ("filesrc", "filesource");
g_object_set (G_OBJECT (source), "location", argv[1], NULL);
g_printerr ("source is: %s \n",argv[1]);
//demuxer = gst_element_factory_make ("avidemux", "avi-demuxer");
demuxer = gst_element_factory_make ("oggdemux", "ogg-demuxer");
vdqueue = gst_element_factory_make ("queue", "video-queue");
adqueue = gst_element_factory_make ("queue", "audio-queue");
dec = gst_element_factory_make ("vorbisdec", "vorbis-decoder");
conv = gst_element_factory_make ("audioconvert", "converter");
sink = gst_element_factory_make ("autoaudiosink", "audio-output");
/* g_object_set (pipeline->mux,"max-delay", (guint64)0,NULL);
g_object_set (pipeline->mux,"max-page-delay",(guint64)0,NULL);
*/
g_object_set (G_OBJECT (source), "caps",
gst_caps_new_simple ("video/x-raw",
"format", G_TYPE_STRING, "RGB16",
"width", G_TYPE_INT, 640,
"height", G_TYPE_INT, 480,
"framerate", GST_TYPE_FRACTION, 0, 1,
NULL), NULL);
gst_bin_add_many (GST_BIN (pipeline),source,demuxer,vdqueue,adqueue,dec,conv,sink, NULL);
gst_element_link (source, demuxer);
gst_element_link (demuxer, vdqueue);
gst_element_link (vdqueue, adqueue);
gst_element_link (adqueue, dec);
gst_element_link (dec, conv);
gst_element_link (conv, sink);
/* run */
// g_printerr ("pipeline is: %s \n",pipeline);
gst_element_set_state (pipeline, GST_STATE_PLAYING);
g_main_loop_run (loop);
/* cleanup */
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (GST_OBJECT (pipeline));
return 0;
}
Related
I've been studying about gstreamer and I don't understand where I got it wrong but every pipeline I tried turning code would give me 'Elements could not be linked' and I'm running this code on ubuntu 20.04 which has an upstream kernel version 5.13.0-35(which I think is not a concern) with gcc 9 and here is the code:
#include <gst/gst.h>
#include <stdio.h>
/* Structure to contain all our information, so we can pass it to callbacks */
typedef struct _CustomData {
GstElement *pipeline;
GstElement *source;
GstCaps *caps;
GstElement *depay;
GstElement *parse;
GstElement *decode;
GstElement *convert;
GstElement *sink;
} CustomData;
int main (int argc, char *argv[])
{
CustomData data;
GstBus *bus;
GstMessage *msg;
GstStateChangeReturn ret;
/* Initialize GStreamer */
gst_init (&argc, &argv);
/* Create the elements */
data.source = gst_element_factory_make ("udpsrc", "source");
data.caps = gst_caps_new_simple("application/x-rtp",
"media", G_TYPE_STRING, "video",
"clock-rate", G_TYPE_INT, 90000,
"encoding-name", G_TYPE_STRING, "H264",
"payload", G_TYPE_INT, 96,
NULL);
data.depay = gst_element_factory_make ("rtph264depay", "depay");
data.parse = gst_element_factory_make ("h264parse", "parse");
data.decode = gst_element_factory_make ("decodebin", "decode");
data.convert = gst_element_factory_make ("videoconvert", "convert");
data.sink = gst_element_factory_make ("autovideosink", "sink");
/* Create the empty pipeline */
data.pipeline = gst_pipeline_new ("test-pipeline");
if (!data.pipeline || !data.source || !data.depay || !data.parse || !data.decode || !data.convert || !data.sink) {
g_printerr ("Not all elements could be created.\n");
return -1;
}
gst_bin_add_many (GST_BIN (data.pipeline), data.source, data.depay, data.parse, data.decode, data.convert, data.sink, NULL);
if (gst_element_link_many (data.source, data.depay, data.parse, data.decode, data.convert, data.sink, NULL) != TRUE) {
g_printerr ("Elements could not be linked.\n");
gst_object_unref (data.pipeline);
return -1;
}
/* Set the port and caps to play */
g_object_set (data.source, "port", 5000, NULL);
g_object_set (data.source, "caps", data.caps, NULL);
g_object_set (data.sink, "sync", FALSE, NULL);
/* Start playing */
ret = gst_element_set_state (data.pipeline, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE) {
g_printerr ("Unable to set the pipeline to the playing state.\n");
gst_object_unref (data.pipeline);
return -1;
}
/* Wait until error or EOS */
bus = gst_element_get_bus (data.pipeline);
msg =
gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE,
GST_MESSAGE_ERROR | GST_MESSAGE_EOS);
/* Parse message */
if (msg != NULL) {
GError *err;
gchar *debug_info;
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_ERROR:
gst_message_parse_error (msg, &err, &debug_info);
g_printerr ("Error received from element %s: %s\n",
GST_OBJECT_NAME (msg->src), err->message);
g_printerr ("Debugging information: %s\n",
debug_info ? debug_info : "none");
g_clear_error (&err);
g_free (debug_info);
break;
case GST_MESSAGE_EOS:
g_print ("End-Of-Stream reached.\n");
break;
default:
/* We should not reach here because we only asked for ERRORs and EOS */
g_printerr ("Unexpected message received.\n");
break;
}
gst_message_unref (msg);
}
/* Free resources */
gst_object_unref (bus);
gst_element_set_state (data.pipeline, GST_STATE_NULL);
gst_object_unref (data.pipeline);
return 0;
}
the recieving pipeline is:
gst-launch-1.0 -v udpsrc port=5000 ! "application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264, payload=(int)96" ! rtph264depay ! h264parse ! decodebin ! videoconvert ! autovideosink sync=false
And the sender is:
gst-launch-1.0 -v filesrc location=test.mp4 ! qtdemux ! h264parse ! avdec_h264 ! x264enc ! rtph264pay ! udpsink host=$HOST port=5000
Thanks in advance.
You may try using gst_parse_launch() that will negociate caps as gst-launch does:
const gchar *pipeline_str = "udpsrc port=5000 ! application/x-rtp,encoding-name=H264 ! rtph264depay ! h264parse ! decodebin ! videoconvert ! autovideosink sync=false";
GstElement *pipeline = gst_parse_launch (pipeline_str, NULL);
if (!pipeline) {
g_error ("Failed to create pipeline\n");
exit(-1);
}
...
Otherwise, you may have to add capsfilters.
I have a basic gstreamer command for playing audio stream received from network:
gst-launch-1.0 tcpserversrc host=127.0.0.1 port=5000 ! decodebin ! audioconvert ! alsasink
I tried to convert it to a C program, but when I'm running it I get "Internal data flow error".
#include <gst/gst.h>
static gboolean bus_call (GstBus *bus, GstMessage *msg, gpointer data)
{
GMainLoop *loop = (GMainLoop *) data;
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_EOS:
g_print ("End of stream\n");
g_main_loop_quit (loop);
break;
case GST_MESSAGE_ERROR: {
gchar *debug;
GError *error;
gst_message_parse_error (msg, &error, &debug);
g_free (debug);
g_printerr ("Error: %s\n", error->message);
g_error_free (error);
g_main_loop_quit (loop);
break;
}
default:
break;
}
return TRUE;
}
gint main (gint argc, gchar *argv[])
{
GMainLoop *loop;
GstElement *pipeline, *src, *dec, *conv, *sink;
GstBus *bus;
/* init GStreamer */
gst_init (&argc, &argv);
loop = g_main_loop_new (NULL, FALSE);
/* setup */
pipeline = gst_pipeline_new ("pipeline");
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
gst_bus_add_watch (bus, bus_call, loop);
gst_object_unref (bus);
src = gst_element_factory_make ("tcpserversrc", "source");
g_object_set (G_OBJECT (src), "host", "127.0.0.1",NULL);
g_object_set (G_OBJECT (src), "port", 5000 ,NULL);
dec = gst_element_factory_make ("decodebin", "decoder");
conv = gst_element_factory_make ("audioconvert", "aconv");
sink = gst_element_factory_make ("alsasink", "sink");
gst_bin_add_many (GST_BIN (pipeline), src, dec, conv, sink, NULL);
gst_element_link (src, dec);
gst_element_link (dec, conv);
gst_element_link (conv, sink);
/* run */
gst_element_set_state (pipeline, GST_STATE_PLAYING);
g_main_loop_run (loop);
/* cleanup */
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (GST_OBJECT (pipeline));
return 0;
}
Here is the command which I'm using to test the server:
gst-launch-1.0 filesrc location="file.wav" ! tcpclientsink host=127.0.0.1 port=5000
Thanks.
Because decodebin's src pad is a dynamic (sometimes) pad. You have to connect decodebin to audioconvert when decodebin got its source pad.
You can see it with gst-inspect-1.0:
$ gst-inspect-1.0 decodebin
:
:
Pad Templates:
SRC template: 'src_%u'
Availability: Sometimes
Capabilities:
ANY
:
:
Add a callback function for pad-added to decodebin and link to audioconvert in the callback. A required change is basically this:
--- orig.c 2017-01-18 13:35:50.434605255 +0900
+++ new.c 2017-01-18 14:04:16.428847528 +0900
## -31,6 +31,21 ##
return TRUE;
}
+static void cb_new_pad (GstElement *element, GstPad *pad, gpointer data)
+{
+ gchar *name;
+ GstElement *other = data;
+
+ name = gst_pad_get_name (pad);
+ g_print ("A new pad %s was created for %s\n", name, gst_element_get_name(element));
+ g_free (name);
+
+ g_print ("element %s will be linked to %s\n",
+ gst_element_get_name(element),
+ gst_element_get_name(other));
+ gst_element_link(element, other);
+}
+
gint main (gint argc, gchar *argv[])
{
GMainLoop *loop;
## -59,9 +74,13 ##
gst_bin_add_many (GST_BIN (pipeline), src, dec, conv, sink, NULL);
gst_element_link (src, dec);
- gst_element_link (dec, conv);
gst_element_link (conv, sink);
+ /* you don't link them here */
+ /* gst_element_link (dec, conv); */
+ /* add call-back, instead */
+ g_signal_connect (dec, "pad-added", G_CALLBACK (cb_new_pad), conv);
+
/* run */
gst_element_set_state (pipeline, GST_STATE_PLAYING);
g_main_loop_run (loop);
Here is a link to a working code.
BTW, you don't have to do it by yourself but let gst_parse_launch() handle all of the above.
int main(int argc, char *argv[])
{
GstElement *pipeline;
GError *err = NULL;
GstBus *bus;
GMainLoop *loop;
gst_init(&argc, &argv);
loop = g_main_loop_new(NULL, FALSE);
pipeline = gst_parse_launch("tcpserversrc host=127.0.0.1 port=5000 ! decodebin ! audioconvert ! alsasink", &err);
gst_element_set_state(pipeline, GST_STATE_PLAYING);
bus = gst_element_get_bus(pipeline);
gst_bus_add_watch (bus, bus_call, loop);
g_main_loop_run(loop);
return 0;
}
There are some questions about dynamic pads on stackoverflow:
GStreamer: how to connect dynamic pads
g_signal_connect "pad-added" doesn't work
i have the following pipelines that one of them sends voice signals on udp port and the other receives them on the same port number on the receiver side
gst-launch-1.0 -v alsasrc ! audioconvert
! audio/x-raw,channels=2,depth=16,width=16,rate=44100 !
rtpL16pay ! udpsink
host=127.0.0.1 port=5000 //sender
and
gst-launch-1.0 udpsrc port=5000 ! "application/x-rtp,
media=(string)audio, clock-rate=(int)44100,
encoding-name=(string)L16, channels=(int)2,
payload=(int)96" ! rtpL16depay ! audioconvert
! alsasink //receiver
These pipelines work perfectly.
now i am trying to write a source code using Gstreamer SDK that does the same thing. I have come so far:
#include <gst/gst.h>
#include <string.h>
int main(int argc, char *argv[]) {
GstElement *pipeline, *source, *audiosink,*rtppay,*rtpdepay,*filter,*filter1,*conv,*conv1,*udpsink,*udpsrc,*receive_resample;
GstBus *bus;
GstMessage *msg;
GstCaps *filtercaps;
GstStateChangeReturn ret;
/* Initialize GStreamer */
gst_init (&argc, &argv);
/* Create the elements */
source = gst_element_factory_make ("alsasrc", "source");
conv= gst_element_factory_make ("audioconvert", "conv");
conv1= gst_element_factory_make ("audioconvert", "conv1");
filter=gst_element_factory_make("capsfilter","filter");
rtppay=gst_element_factory_make("rtpL16pay","rtppay");
rtpdepay=gst_element_factory_make("rtpL16depay","rtpdepay");
udpsink=gst_element_factory_make("udpsink","udpsink");
audiosink = gst_element_factory_make ("autoaudiosink", "audiosink");
receive_resample = gst_element_factory_make("audioresample", NULL);
udpsrc=gst_element_factory_make("udpsrc",NULL);
filter1=gst_element_factory_make("capsfilter","filter");
g_object_set(udpsrc,"port",5000,NULL);
g_object_set (G_OBJECT (udpsrc), "caps", gst_caps_from_string("application/x-rtp,media=audio,payload=96,clock-rate=44100,encoding-name=L16,channels=2"), NULL);
/* Create the empty pipeline */
pipeline = gst_pipeline_new ("test-pipeline");
if (!pipeline || !source || !filter || !conv || !rtppay || !udpsink ) {
g_printerr ("Not all elements could be created.\n");
return -1;
}
g_object_set(G_OBJECT(udpsink),"host","127.0.0.1",NULL);
g_object_set(G_OBJECT(udpsink),"port",5000,NULL);
filtercaps = gst_caps_new_simple ("audio/x-raw",
"channels", G_TYPE_INT, 2,
"width", G_TYPE_INT, 16,
"depth", G_TYPE_INT, 16,
"rate", G_TYPE_INT, 44100,
NULL);
g_object_set (G_OBJECT (filter), "caps", filtercaps, NULL);
gst_caps_unref (filtercaps);
filtercaps = gst_caps_new_simple ("application/x-rtp",
"media",G_TYPE_STRING,"audio",
"clock-rate",G_TYPE_INT,44100,
"encoding-name",G_TYPE_STRING,"L16",
"channels", G_TYPE_INT, 2,
"payload",G_TYPE_INT,96,
NULL);
g_object_set (G_OBJECT (filter1), "caps", filtercaps, NULL);
gst_caps_unref (filtercaps);
/* Build the pipeline */
gst_bin_add_many (GST_BIN (pipeline), source,filter,conv,rtppay,udpsink, NULL);
if (gst_element_link_many (source,filter,conv,rtppay,udpsink, NULL) != TRUE) {
g_printerr ("Elements could not be linked.\n");
gst_object_unref (pipeline);
return -1;
}
gst_bin_add_many (GST_BIN (pipeline),udpsrc,rtpdepay,conv1,receive_resample,audiosink,NULL);
if (gst_element_link_many (udpsrc,rtpdepay,conv1,receive_resample,audiosink,NULL) != TRUE) {
g_printerr ("Elements could not be linked.\n");
gst_object_unref (pipeline);
return -1;
}
/* Modify the source's properties */
// g_object_set (source, "pattern", 0, NULL);
/* Start playing */
ret = gst_element_set_state (pipeline, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE) {
g_printerr ("Unable to set the pipeline to the playing state.\n");
gst_object_unref (pipeline);
return -1;
}
/* Wait until error or EOS */
bus = gst_element_get_bus (pipeline);
msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE, GST_MESSAGE_ERROR | GST_MESSAGE_EOS);
/* Parse message */
if (msg != NULL) {
GError *err;
gchar *debug_info;
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_ERROR:
gst_message_parse_error (msg, &err, &debug_info);
g_printerr ("Error received from element %s: %s\n", GST_OBJECT_NAME (msg->src), err->message);
g_printerr ("Debugging information: %s\n", debug_info ? debug_info : "none");
g_clear_error (&err);
g_free (debug_info);
break;
case GST_MESSAGE_EOS:
g_print ("End-Of-Stream reached.\n");
break;
default:
/* We should not reach here because we only asked for ERRORs and EOS */
g_printerr ("Unexpected message received.\n");
break;
}
gst_message_unref (msg);
}
/* Free resources */
gst_object_unref (bus);
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (pipeline);
return 0;
}
but somehow i dont receive any voice on the receiver. i dont get any errors of any kind. Any ideas why this is happening?
I want to display two pictures in the same window. This code displays them in two different windows.
Is there a solution for joining the two pictures in the same window?
#include <gst/gst.h>
#include <glib.h>
static gboolean bus_call (GstBus *bus, GstMessage *msg, gpointer data)
{
GMainLoop *loop = (GMainLoop *) data;
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_EOS:
g_print ("End of stream\n");
g_main_loop_quit (loop);
break;
case GST_MESSAGE_ERROR: {
gchar *debug;
GError *error;
gst_message_parse_error (msg, &error, &debug);
g_free (debug);
g_printerr ("Error: %s\n", error->message);
g_error_free (error);
g_main_loop_quit (loop);
break;
}
default:
break;
}
return TRUE;
}
static void on_pad_added (GstElement *element, GstPad *pad, gpointer data)
{
GstPad *sinkpad;
GstElement *decoder = (GstElement *) data;
/* We can now link this pad with the vorbis-decoder sink pad */
g_print ("Dynamic pad created, linking demuxer/decoder\n");
sinkpad = gst_element_get_static_pad (decoder, "sink");
gst_pad_link (pad, sinkpad);
gst_object_unref (sinkpad);
}
int main (int argc, char *argv[]) {
GMainLoop *loop;
GstElement *pipeline,*freeze,*clrspace, *source1, *source2, *videobox1,*videobox2, *mixer,*sink,*queuevideo;
GstBus *bus;
loop = g_main_loop_new (NULL, FALSE);
gst_init (&argc, &argv);
/* Create gstreamer elements */
pipeline = gst_pipeline_new ("player");
source1 = gst_element_factory_make ("playbin2", "dec1");
source2 = gst_element_factory_make ("playbin2", "dec2");
freeze = gst_element_factory_make ("imagefreeze", "fr");
videobox1 = gst_element_factory_make ("videobox", "videobox1");
videobox2 = gst_element_factory_make ("videobox", "videobox2");
clrspace = gst_element_factory_make ("ffmpegcolorspace", "clrspace");
mixer = gst_element_factory_make ("videomixer", "mixer");
queuevideo = gst_element_factory_make ("queue", "queue-video");
sink = gst_element_factory_make ("autovideosink", "sink");
if (!pipeline || !source1 || !source2 || !sink || !mixer ||!freeze || !clrspace || !queuevideo ) {
g_printerr ("One element could not be created. Exiting.\n");
exit(1);
}
g_object_set (source1, "uri", "http://www.logotheque.fr/6396-2/logo+RMC+INFO.jpg", NULL);
g_object_set (source2, "uri", "http://www.logotheque.fr/6396-2/logo+RMC+INFO.jpg", NULL);
g_object_set(videobox1,"border-alpha",0,"top",0,"left",0,NULL);
g_object_set(videobox2,"border-alpha",0,"top",0,"left",-200,NULL);
/* we add a message handler */
bus = gst_pipeline_get_bus(GST_PIPELINE (pipeline));
gst_bus_add_watch (bus, bus_call, loop);
gst_object_unref (bus);
/* we add all elements into the pipeline */
gst_bin_add_many (GST_BIN(pipeline), source1,mixer, clrspace, freeze,videobox1, sink, NULL);
/* we link the elements together */
gst_element_link_many (source2, mixer, clrspace, freeze,videobox2,sink, NULL);
//gst_element_link_many(source[1], mixer, NULL);
g_signal_connect (source1, "pad-added", G_CALLBACK (on_pad_added), queuevideo);
g_signal_connect (source2, "pad-added", G_CALLBACK (on_pad_added), queuevideo);
/* Set the pipeline to "playing" state*/
gst_element_set_state(pipeline, GST_STATE_PLAYING);
/* Iterate */
g_print ("Running...\n");
g_main_loop_run (loop);
/* Out of the main loop, clean up nicely */
g_print ("Returned, stopping playback\n");
gst_element_set_state (pipeline, GST_STATE_NULL);
g_print ("Deleting pipeline\n");
gst_object_unref (GST_OBJECT(pipeline));
}
Here is how to combine two images with gstreamer from the command line :
gst-launch-1.0 uridecodebin uri=file:///home/meh/Pictures/questions.jpg ! videoscale ! video/x-raw, width=320, height=240 ! imagefreeze ! videomixer name=m sink_1::xpos=320 ! autovideosink uridecodebin uri=file:///home/meh/Pictures/testsrc.png ! videoscale ! video/x-raw, width=320, height=240 ! imagefreeze ! m.
Explanation :
We create two decoders for the images, resize them with videoscale to an arbitrary size (here 320 x 240), freeze them and send them to videomixer. videomixer has the x position of sink_1 set to 320, which offsets the first image so that the second one appears as well.
I am trying to simulate the following gstreamer pipeline using the C API:
gst-launch -e videomixer name=mix ! ffmpegcolorspace ! xvimagesink \
videotestsrc pattern=1 ! video/x-raw-yuv,width=100,height=100 ! videobox border-alpha=0 top=0 left=0 ! mix. \
videotestsrc pattern=0 ! video/x-raw-yuv,width=100,height=100 ! videobox border-alpha=0 top=0 left=-100 ! mix.
So far I have:
#include <gst/gst.h>
#include <glib.h>
static gboolean
bus_call (GstBus *bus,
GstMessage *msg,
gpointer data)
{
GMainLoop *loop = (GMainLoop *) data;
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_EOS:
g_print ("End of stream\n");
g_main_loop_quit (loop);
break;
case GST_MESSAGE_ERROR: {
gchar *debug;
GError *error;
gst_message_parse_error (msg, &error, &debug);
g_free (debug);
g_printerr ("Error: %s\n", error->message);
g_error_free (error);
g_main_loop_quit (loop);
break;
}
default:
break;
}
return TRUE;
}
int
main (int argc,
char *argv[])
{
GMainLoop *loop;
GstElement *pipeline;
GstElement *source1,*source2;
GstElement *scale,*filter;
GstElement *videobox1,*videobox2; //just one.
GstElement *mixer,*clrspace,*sink;
GstCaps *filtercaps;
GstBus *bus;
/* Initialisation */
gst_init (&argc, &argv);
loop = g_main_loop_new (NULL, FALSE);
/* Check input arguments */
/*if (argc != 2) {
g_printerr ("Usage: %s <Ogg/Vorbis filename>\n", argv[0]);
return -1;
}*/
//gst-launch videotestsrc pattern=snow ! ximagesink
/* Create gstreamer elements */
pipeline = gst_pipeline_new ("player");
source1 = gst_element_factory_make ("videotestsrc", "source1");
source2 = gst_element_factory_make ("videotestsrc", "source2");
// source2 = gst_element_factory_make ("uridecodebin", "file-source2");
scale = gst_element_factory_make ("videoscale", "scale");
filter = gst_element_factory_make("capsfilter","filter");
videobox1 = gst_element_factory_make ("videobox", "videobox1");
videobox2 = gst_element_factory_make ("videobox", "videobox2");
mixer = gst_element_factory_make ("videomixer", "mixer");
clrspace = gst_element_factory_make ("ffmpegcolorspace", "clrspace");
// demuxer = gst_element_factory_make ("oggdemux", "ogg-demuxer");
// decoder = gst_element_factory_make ("vorbisdec", "vorbis-decoder");
// conv = gst_element_factory_make ("audioconvert", "converter");
sink = gst_element_factory_make ("xvimagesink", "sink");
/*if (!pipeline || !source || !demuxer || !decoder || !conv || !sink) {
g_printerr ("One element could not be created. Exiting.\n");
return -1;
}*/
if (!pipeline || !source1 || !source2 || !sink) {
g_printerr ("One element could not be created. Exiting.\n");
return -1;
}
filtercaps = gst_caps_new_simple ("video/x-raw-yuv",
"width", G_TYPE_INT, 200,
"height", G_TYPE_INT, 100,
NULL);
g_object_set (G_OBJECT (filter), "caps", filtercaps, NULL);
//gst_caps_unref (filtercaps);
g_object_set(videobox1,"border-alpha",0,"top",0,"left",0,NULL);
g_object_set(videobox2,"border-alpha",0,"top",0,"left",-200,NULL);
/* Set up the pipeline */
/* we set the input filename to the source element */
g_object_set (G_OBJECT (source1), "pattern", 0, NULL);
g_object_set (G_OBJECT (source2), "pattern", 1, NULL);
/* we add a message handler */
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
gst_bus_add_watch (bus, bus_call, loop);
gst_object_unref (bus);
/* we add all elements into the pipeline */
gst_bin_add_many (GST_BIN (pipeline),
source1,filter,videobox1,mixer,clrspace, sink, source2,videobox2, NULL);
/* we link the elements together */
//gst_element_link_many (source1, scale, filter, videobox1, mixer, clrspace, sink);
//gst_element_link_many (source2, scale, filter, videobox2, mixer, clrspace, sink);
gst_element_link_many (source1, filter, videobox1, mixer, clrspace, sink);
gst_element_link_many (source2, filter, videobox2, mixer, clrspace, sink);
/* Set the pipeline to "playing" state*/
gst_element_set_state (pipeline, GST_STATE_PLAYING);
/* Iterate */
g_print ("Running...\n");
g_main_loop_run (loop);
/* Out of the main loop, clean up nicely */
g_print ("Returned, stopping playback\n");
gst_element_set_state (pipeline, GST_STATE_NULL);
g_print ("Deleting pipeline\n");
gst_object_unref (GST_OBJECT (pipeline));
return 0;
}
I have also set debugging on: export GST_DEBUG=3
When I run my program I get the following error:
Running...
0:00:00.178663884 4797 0x8937020 WARN basesrc gstbasesrc.c:2582:gst_base_src_loop:<source2> error: Internal data flow error.
0:00:00.178766444 4797 0x8937020 WARN basesrc gstbasesrc.c:2582:gst_base_src_loop:<source2> error: streaming task paused, reason not-linked (-1)
Error: Internal data flow error.
Returned, stopping playback
0:00:00.202571746 4797 0x893ae00 WARN basetransform gstbasetransform.c:1627:gst_base_transform_prepare_output_buffer:<clrspace> pad-alloc failed: wrong-state
0:00:00.202645907 4797 0x893ae00 WARN basetransform gstbasetransform.c:2335:gst_base_transform_handle_buffer:<clrspace> could not get buffer from pool: wrong-state
Deleting pipeline
Why is it complaining about source2 not-linked?
A little late, but may be helpful:
If you look at the documentation for the videomixer element, you'll see that videomixer's sink pads are request pads. You need to create these pads before linking them.
/* Manually link the mixer, which has "Request" pads */
mixer_sink_pad_template = gst_element_class_get_pad_template (GST_ELEMENT_GET_CLASS (mixer), "sink_%u");
mixer_sink_pad = gst_element_request_pad (mixer, mixer_sink_pad_template, NULL, NULL);
sink_pad = gst_element_get_static_pad (clrspace, "src");
gst_pad_link ( sink_pad,mixer_sink_pad);
Similarly request pads are created for as many streams as you want.
gst_element_link_many (source1, filter, videobox1, mixer, clrspace, sink, NULL);
gst_element_link_many (source2, filter, videobox2, mixer, NULL);
Please also learn about the x,y,z properties on videomixerpad, you can spare the videobox elements this way and gain performance