[Gstreamer] a cpp implementation to receive two audio from udp sink

Readme.md

Tested with gstreamer GStreamer 1.14.5 (on aarch64 and amd64)

To stream system mic with udp, on Windows:

.\gst-launch-1.0.exe -v directsoundsrc ! audioresample ! audioconvert ! rtpL24pay ! udpsink host=192.168.1.3 port=4343

To stream system mic with udp, on Linux:

gst-launch-1.0 -v alsasrc ! queue ! audioresample ! audioconvert ! rtpL24pay ! udpsink host=192.168.1.3 port=4343

CMakeLists.txt

cmake_minimum_required(VERSION 3.10.2)
                               
project(RTSP-Test)             

find_package(PkgConfig REQUIRED)
#pkg_check_modules(gtk3 REQUIRED IMPORTED_TARGET gtk+-3.0)
pkg_search_module(gstreamer REQUIRED IMPORTED_TARGET gstreamer-1.0)
pkg_search_module(gstreamer-sdp REQUIRED IMPORTED_TARGET gstreamer-sdp-1.0)
pkg_search_module(gstreamer-app REQUIRED IMPORTED_TARGET gstreamer-app-1.0)
pkg_search_module(gstreamer-audio REQUIRED IMPORTED_TARGET gstreamer-audio-1.0)

add_executable(rtps-test udpsrc_filesink.cpp)

target_link_libraries(rtps-test
    PkgConfig::gstreamer
    PkgConfig::gstreamer-sdp
    PkgConfig::gstreamer-app
    PkgConfig::gstreamer-audio
)

udpsrc_filesink.cpp

#include <gst/gst.h>

static GMainLoop *loop;

int main (int argc, char *argv[]) {

    GstElement *pipeline, *udpsrc, *rtpL2depay, *audioconvert, *wavenc, *filesink;

    gst_init (&argc, &argv);
    loop = g_main_loop_new (NULL, FALSE);

    pipeline = gst_pipeline_new ("pipeline");
    udpsrc = gst_element_factory_make ("udpsrc", NULL); 
    rtpL2depay = gst_element_factory_make ("rtpL24depay", NULL);
    audioconvert = gst_element_factory_make ("audioconvert", NULL);
    wavenc = gst_element_factory_make("wavenc", NULL);
    filesink = gst_element_factory_make ("filesink", NULL);

    g_object_set(G_OBJECT(udpsrc),
            "address", "192.168.1.3", /* UDP IP Address */
            "port", 4343, /* UDP port */
            NULL);

    g_object_set (G_OBJECT (udpsrc), "caps",
            gst_caps_new_simple ("application/x-rtp",
                "channels", G_TYPE_INT, 2,
                "format", G_TYPE_STRING, "S16LE",
                "media", G_TYPE_STRING, "audio",
                "payload", G_TYPE_INT, 96,
                "clock-rate", G_TYPE_INT, 44100,
                "encoding-name", G_TYPE_STRING, "L24",
                NULL), NULL);

     g_object_set(G_OBJECT(filesink), "location", "capture.wave", NULL);

    gst_bin_add_many (GST_BIN (pipeline), udpsrc, rtpL2depay, audioconvert, wavenc, filesink, NULL);
    gst_element_link_many (udpsrc, rtpL2depay, audioconvert, wavenc, filesink, NULL);

    gst_element_set_state (pipeline, GST_STATE_PLAYING);
    g_main_loop_run (loop);

    gst_element_set_state (pipeline, GST_STATE_NULL);
    gst_object_unref (GST_OBJECT (pipeline));
    g_main_loop_unref (loop);

    return 0;
}