gstreamer audiomixer command to code converting - c++

Question

I want to use audiomixer in my application which receives audios from different sources and should play them together in speaker.
my final application should do something like this command:
gst-launch-1.0 audiomixer name=mix ! autoaudiosink autoaudiosrc ! \
audioconvert ! mix. udpsrc port=5001 caps="application/x-rtp" ! queue !\
rtppcmudepay ! mulawdec ! audioconvert ! audioresample ! mix.
I already wrote a code to use tee and queues and know how to work with tee and queues in code based on this code. but I don't know how to use mixer in my code.
so for simplicity I just want to write a code to work as this command does:
gst-launch-1.0 audiotestsrc freq=100 ! audiomixer name=mix ! audioconvert ! autoaudiosink autoaudiosrc ! mix.
I didn't find any useful example to reach this goal, how can I write a C code to do this?

Answer 1

for the second part:
gst-launch-1.0 audiotestsrc freq=100 ! audiomixer name=mix ! audioconvert ! autoaudiosink autoaudiosrc ! mix.
this code works:
#include <gst/gst.h>
static GMainLoop *loop;
int bus_callback (GstBus *bus, GstMessage *message, gpointer data)
{
g_print ("Got %s message\n", GST_MESSAGE_TYPE_NAME (message));
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_ERROR: {
GError *err;
gchar *debug;
gst_message_parse_error (message, &err, &debug);
g_print ("Error: %s\n", err->message);
g_error_free (err);
g_free (debug);
g_main_loop_quit (loop);
break;
}
case GST_MESSAGE_EOS:
/* end-of-stream */
g_main_loop_quit (loop);
break;
default:
/* unhandled message */
break;
}
/* we want to be notified again the next time there is a message
* on the bus, so returning TRUE (FALSE means we want to stop watching
* for messages on the bus and our callback should not be called again)
*/
return TRUE;
}
int main(int argc, char *argv[])
{
/* Initialize GStreamer */
gst_init (nullptr, nullptr);
GstElement *pipeline, *src1,*src2, *sink, *convert1,*convert2,*audiomixer;
GstPad *conv_pad1, *conv_pad2, *mixer1_sinkpad,*mixer2_sinkpad;
gint i;
static GstBus *bus;
static guint bus_watch_id;
pipeline = gst_pipeline_new ("pipeline");
audiomixer = gst_element_factory_make ("adder", "mixer");
sink = gst_element_factory_make ("autoaudiosink", "sink");
src1 = gst_element_factory_make ("audiotestsrc", "src1");
convert1 = gst_element_factory_make ("audioconvert", "convert1");
src2 = gst_element_factory_make ("autoaudiosrc", "src2");
convert2 = gst_element_factory_make ("audioconvert", "convert2");
//g_object_set (sink, "async-handling", TRUE, NULL);
gst_bin_add_many (GST_BIN (pipeline), audiomixer ,sink, NULL);
gst_bin_add_many (GST_BIN (pipeline), src1 , convert1 , NULL);
gst_bin_add_many (GST_BIN (pipeline), src2 , convert2 , NULL);
gst_element_link (src1, convert1 );
gst_element_link (src2, convert2 );
gst_element_link(audiomixer , sink);
conv_pad1= gst_element_get_static_pad (convert1, "src");
mixer1_sinkpad = gst_element_get_request_pad (audiomixer, "sink_%u");
gst_pad_link (conv_pad1, mixer1_sinkpad);
g_object_unref(mixer1_sinkpad);
conv_pad2= gst_element_get_static_pad (convert2, "src");
mixer2_sinkpad = gst_element_get_request_pad (audiomixer, "sink_%u");
gst_pad_link (conv_pad2, mixer2_sinkpad);
g_object_unref(mixer2_sinkpad);
/* adds a watch for new message on our pipeline’s message bus to
* the default GLib main context, which is the main context that our
* GLib main loop is attached to below
*/
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
bus_watch_id = gst_bus_add_watch (bus, bus_callback, NULL);
gst_object_unref (bus);
/* Start playing */
gst_element_set_state (pipeline, GST_STATE_PLAYING);
loop = g_main_loop_new (NULL, FALSE);
g_main_loop_run (loop);
g_object_unref(conv_pad1);
g_object_unref(conv_pad2);
gst_element_set_state (pipeline, GST_STATE_NULL);
g_source_remove (bus_watch_id);
}