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
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.
Trying to make gstreamer application from the pipeline: gst-launch-1.0 playbin uri=rtsp:// video-sink="videoconvert ! video/x-raw,width=720, height=480 ! ximagesink"
Getting error to link elements. Posting my code:
#include <gst/gst.h>
int main(int argc, char *argv[]) {
GstElement *source, *videosink, *pipeline, *videoconvert;
GstCaps *capsFilter;
GstBus *bus;
GstMessage *msg;
GstPad *pad;
gboolean link_ok;
/* Initialize GStreamer */
gst_init (&argc, &argv);
/* Create Elements */
pipeline = gst_pipeline_new("my-pipeline");
source = gst_element_factory_make ("playbin", "source");
videoconvert = gst_element_factory_make("videoconvert", "convert");
videosink = gst_element_factory_make("ximagesink", "autovideosink");
/* set property value */
g_object_set (source, "uri", "rtsp:<file location>", NULL);
if (!pipeline || !source || !videoconvert || !videosink)
{
g_printerr ("Not all elements could be created.\n");
return;
}
gst_bin_add_many (GST_BIN(pipeline), videoconvert, videosink, NULL);
capsFilter = gst_caps_new_simple("video/x-raw",
"width", G_TYPE_INT, 176,
"height", G_TYPE_INT, 144,
NULL);
link_ok = gst_element_link_filtered(videoconvert,videosink, capsFilter);
gst_caps_unref (capsFilter);
if (!link_ok) {
g_warning ("Failed to link element1 and element2!");
}
if (gst_element_link_many( videoconvert, videosink, NULL) != TRUE) {
g_print ("Failed to link some elements .....1 !\n");
gst_object_unref (pipeline);
return -1;
}
/* Start playing */
gst_element_set_state (source, GST_STATE_PLAYING);
/* Wait until error or EOS */
bus = gst_element_get_bus (source);
msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE,
GST_MESSAGE_ERROR | GST_MESSAGE_EOS);
It should be noted that the playbin element gives a full pipeline from src to sink, so the pipeline without the sink stuff would play perfectly:
gst-launch-1.0 playbin uri=rtsp://127.0.0.1:8551/test
The correct pipeline that you want to create is next:
gst-launch-1.0 playbin uri=rtsp://127.0.0.1:8551/test video-sink="videoconvert ! video/x-raw,width=320,height=240 ! ximagesink"
In order to embed this into GStreamer application, it is not necessary to link all the elements. Instead of that, the necessary steps to make are to build a custom video output bin and set it to playbin's video-sink property. In other words, the way to go is create a bin and link the necessary elements and then tell playbin to use it via it's video-sink property. It is also necessary to create a GhostPad for the bin and point it to the sink pad of the first element within the bin.
This is the result:
#include <gst/gst.h>
int main(int argc, char *argv[]) {
GstElement *source, *videosink, *pipeline, *videoconvert, *customoutput;
GstCaps *capsFilter;
GstBus *bus;
GstMessage *msg;
GstPad *pad;
gboolean add_ok;
gboolean link_ok;
GstStateChangeReturn ret;
GMainLoop *loop;
/* Initialize GStreamer */
gst_init (&argc, &argv);
loop = g_main_loop_new (NULL, FALSE);
/* Create Elements */
pipeline = gst_pipeline_new("my-pipeline");
source = gst_element_factory_make ("playbin", "source");
videoconvert = gst_element_factory_make("videoconvert", "videoconvert");
capsFilter = gst_caps_new_simple("video/x-raw",
"width", G_TYPE_INT, 320,
"height", G_TYPE_INT, 240,
NULL);
videosink = gst_element_factory_make("ximagesink", "videosink");
customoutput = gst_bin_new("customoutput");
// It is possible to create the bin like this
// Ghost pads on the bin for unlinked source or sink pads within the bin can automatically be created
// customoutput = gst_parse_bin_from_description ("videoconvert ! video/x-raw,width=320 ! ximagesink", TRUE, NULL);
gst_bin_add_many (GST_BIN (customoutput), videoconvert, videosink, NULL);
link_ok = gst_element_link_filtered(videoconvert,videosink, capsFilter);
gst_caps_unref (capsFilter);
if (!link_ok) {
g_warning ("Failed to link element1 and element2!");
}
GstPad *sinkpad,*ghost_sinkpad;
sinkpad = gst_element_get_static_pad (videoconvert, "sink");
ghost_sinkpad = gst_ghost_pad_new ("sink", sinkpad);
gst_pad_set_active (ghost_sinkpad, TRUE);
gst_element_add_pad (customoutput, ghost_sinkpad);
/* set property value */
g_object_set (source, "video-sink", customoutput, NULL);
g_object_set (source, "uri", "rtsp://127.0.0.1:8551/test", NULL);
if (!pipeline || !source || !videoconvert || !capsFilter || !videosink || !customoutput)
{
g_printerr ("Not all elements could be created.\n");
return -1;
}
gst_bin_add_many (GST_BIN(pipeline), source,NULL);
// Start playing */
ret = gst_element_set_state (pipeline, GST_STATE_PLAYING);
g_print ("Running...\n");
g_main_loop_run (loop);
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);
/* Free resources */
gst_object_unref (bus);
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (pipeline);
g_main_loop_unref (loop);
return 0;
}
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 having issues converting my working GStreamer pipeline from a command line version to C code. From the command line the following command will successfully play my headerless mu-law audio file:
gst-launch filesrc location=test.ulaw ! audio/x-mulaw, rate=8000, channels=1 ! mulawdec ! audioconvert ! audioresample ! autoaudiosink
However, my issues are arising when trying to add in the "audio/x-mulaw, rate=8000, channels=1" bit into my C program. The program started off playing wav files (using wavparse in place of mulawdec) so that I know my base C code works, and it must just be that I am misinterpreting how the caps bit needs to be added in to make it work with mu-law files.
I am creating the caps, then using the gst_element_link_filtered to use this:
GstCaps *gstMuLawCaps = gst_caps_from_string("audio/x-mulaw, rate=8000, channels=1");
gst_element_link_filtered(gstFileSource, gstMuLawDecoder, gstMuLawCaps);
But this is not working, and running the program produces the following output:
>gst-mulaw.exe test.ulaw
Playing.
Error: Internal data flow error.
Playback Finished.
I would be grateful if anyone is able to help shed some light on what I am doing wrong. The full code is given below:
#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 *gstSourceElement, GstPad *gstSourcePad, gpointer data) {
g_print("Linking dynamic pad.\n");
GstPad *gstSinkPad;
GstElement *gstSinkElement = (GstElement *) data;
gstSinkPad = gst_element_get_static_pad (gstSinkElement, "sink");
gst_pad_link (gstSourcePad, gstSinkPad);
gst_object_unref (gstSinkPad);
}
int main (int argc, char *argv[]) {
GMainLoop *loop;
GstElement *gstPipeline, *gstFileSource, *gstMuLawDecoder, *gstAudioConvert, *gstAudioResample, *gstAudioSink;
GstBus *bus;
// GStreamer initialisation.
gst_init (&argc, &argv);
loop = g_main_loop_new (NULL, FALSE);
// Check input arguments.
if (argc != 2) {
g_printerr ("Usage: %s <mu-law File>\n", argv[0]);
return -1;
}
// Create the GStreamer elements.
gstPipeline = gst_pipeline_new ("player");
gstFileSource = gst_element_factory_make ("filesrc", "filesource");
gstMuLawDecoder = gst_element_factory_make ("mulawdec", "mulawdecoder");
gstAudioConvert = gst_element_factory_make ("audioconvert", "audioconverter");
gstAudioResample = gst_element_factory_make ("audioresample", "audioresampler");
gstAudioSink = gst_element_factory_make ("autoaudiosink", "audiosink");
if (!gstPipeline || !gstFileSource || !gstMuLawDecoder || !gstAudioConvert || !gstAudioResample || !gstAudioSink) {
g_printerr ("An element could not be created. Exiting.\n");
return -1;
}
// Linke the filesrc object to that passed on the command line.
g_object_set (G_OBJECT (gstFileSource), "location", argv[1], NULL);
// Setup the GStreamer bus.
bus = gst_pipeline_get_bus (GST_PIPELINE (gstPipeline));
gst_bus_add_watch (bus, bus_call, loop);
gst_object_unref (bus);
// Add the objects to the pipeline.
gst_bin_add_many (GST_BIN (gstPipeline), gstFileSource, gstMuLawDecoder, gstAudioConvert, gstAudioResample, gstAudioSink, NULL);
// Link the elements together.
GstCaps *gstMuLawCaps = gst_caps_from_string("audio/x-mulaw, rate=8000, channels=1");
gst_element_link_filtered(gstFileSource, gstMuLawDecoder, gstMuLawCaps);
gst_caps_unref(gstMuLawCaps);
gst_element_link_many (gstAudioConvert, gstAudioResample, gstAudioSink, NULL);
g_signal_connect (gstMuLawDecoder, "pad-added", G_CALLBACK (on_pad_added), gstAudioConvert);
// Set the pipeline to state playing, and run the main loop.
g_print ("Playing.\n");
gst_element_set_state (gstPipeline, GST_STATE_PLAYING);
g_main_loop_run (loop);
// Finished playback, cleanup.
g_print ("Playback Finished.\n");
gst_element_set_state (gstPipeline, GST_STATE_NULL);
gst_object_unref (GST_OBJECT (gstPipeline));
return 0;
}
Thanks.
Try changing
// Link the elements together.
GstCaps *gstMuLawCaps = gst_caps_from_string("audio/x-mulaw, rate=8000, channels=1");
gst_element_link_filtered(gstFileSource, gstMuLawDecoder, gstMuLawCaps);
gst_caps_unref(gstMuLawCaps);
gst_element_link_many (gstAudioConvert, gstAudioResample, gstAudioSink, NULL);
g_signal_connect (gstMuLawDecoder, "pad-added", G_CALLBACK (on_pad_added), gstAudioConvert);
to
// Link the elements together.
GstCaps *gstMuLawCaps = gst_caps_from_string("audio/x-mulaw, rate=8000, channels=1");
gst_element_link_filtered(gstFileSource, gstMuLawDecoder, gstMuLawCaps);
gst_caps_unref(gstMuLawCaps);
gst_element_link_many (gstMuLawDecoder, gstAudioConvert, gstAudioResample, gstAudioSink, NULL);
pad-added handling is needed for elements with sometimes pads (e.g. demuxers). You can remove the on_pad_added callback function (it was not called anyway, right?) The pad-type can be see in "gst-inspect mulawdec" output.