gstreamer memory usage

#include <gst/gst.h>

#include <cstdio>

#include <string>
#include <sstream>
#include <iostream>


void log(const char * chaine,...) {
    va_list ap;
    va_start( ap, chaine );

    vfprintf( stdout, chaine, ap );

    va_end( ap );
}

GstElement* _pipeline;
GMainLoop*  _loop;
GstBus* bus = NULL;
guint _timeout = 0;
guint _busWatchId = 0;
bool _useAudio = false;

int _duration = 10;

std::string _filename = "test_04.mp4";
std::string _rtmpLocation = "";
std::string _login = "";
std::string _password = "";

gboolean stop_encoding( void * data ) {
    if( g_main_loop_is_running( _loop ) ) {
        gst_element_send_event ( _pipeline, gst_event_new_eos() );
    }

    return FALSE;
}

gboolean bus_call ( GstBus* bus, GstMessage* msg, void* data ){

    switch ( GST_MESSAGE_TYPE ( msg ) ){
    case GST_MESSAGE_EOS: {
        log ( "bus_call: 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 );

        log( "bus_call: Error: %s\n", error->message );
        g_error_free ( error );
        g_main_loop_quit ( _loop );

        break;
    }
    case GST_MESSAGE_STATE_CHANGED : {
        GstObject * obj = GST_MESSAGE_SRC( msg );
        if( GST_IS_PIPELINE( obj ) ) {
            if( GST_ELEMENT( obj ) == _pipeline ) {
                GstState oldstate, newstate, pending;
                gst_message_parse_state_changed ( msg, &oldstate, &newstate, &pending );
                if( newstate == GST_STATE_PLAYING ) {
                    if( _duration ) {
                        log("bus_call: pipeline state changed to PLAYING, setting up timeout %ds\n", _duration );
                        _timeout = g_timeout_add_seconds( _duration, (GSourceFunc)stop_encoding, NULL);
                    }
                    else {
                        log("bus_call: pipeline state changed to PLAYING but no timeout needed (duration=0)\n");
                    }
                }
                else if( newstate == GST_STATE_PAUSED ) {
                    log("bus_call: pipeline state changed to PAUSED\n" );
                }
            }
        }
        break;
    }
    default:
        break;
    }

    return true;
}

void on_pad_added( GstElement *decoder, GstPad *pad, gpointer data ) {
    GstCaps *caps = NULL;
#if GST_CHECK_VERSION(1,0,0)
    caps = gst_pad_query_caps( pad, NULL );
#else
    caps = gst_pad_get_caps( pad );
#endif
    GstBin *bin = (GstBin *) data;
    GstStructure *structure = gst_caps_get_structure( caps, 0 );
    int ret = 0;

    if(g_strrstr(gst_structure_get_string(structure,"media"), "video")) {
        GstElement * element = gst_bin_get_by_name( bin,"rtph264depay");
        if( element ) {
            GstPad * targetsink = gst_element_get_static_pad(element,"sink");

            ret = gst_pad_link( pad, targetsink );
            if( ret != GST_PAD_LINK_OK ) {
                GstCaps *sinkcaps = NULL;
#if GST_CHECK_VERSION(1,0,0)
                sinkcaps = gst_pad_query_caps( targetsink, NULL );
#else
                sinkcaps = gst_pad_get_caps( targetsink );
#endif
                log("on_pad_added: %d LINKING RETURN: %d\n", __LINE__, ret );
                log("on_pad_added: Src Caps: %s\n", gst_caps_to_string( caps ) );
                log("on_pad_added: Target Caps: %s\n", gst_caps_to_string( sinkcaps ) );
                gst_caps_unref( sinkcaps );
            }
            else
                log("on_pad_added: linked video\n" );

            gst_object_unref( targetsink );
            gst_object_unref( element );
        }
    }

    if( g_strrstr( gst_structure_get_string( structure, "media" ), "audio" ) ) {
        GstElement * element = gst_bin_get_by_name( bin, "rtpmp4gdepay" );

        if( element ) {
            GstPad * targetsink = gst_element_get_static_pad( element, "sink" );
            ret = gst_pad_link( pad, targetsink );
            if( ret != GST_PAD_LINK_OK ) {
                GstCaps *sinkcaps = NULL;
#if GST_CHECK_VERSION(1,0,0)
                sinkcaps = gst_pad_query_caps( targetsink, NULL );
#else
                sinkcaps = gst_pad_get_caps( targetsink );
#endif
                log("on_pad_added: %d LINKING RETURN: %d\n", __LINE__, ret );
                log("on_pad_added: Src Caps: %s\n", gst_caps_to_string( caps ) );
                log("on_pad_added: Target Caps: %s\n", gst_caps_to_string( sinkcaps ) );
                gst_caps_unref( sinkcaps );
            }
            else
                log("on_pad_added: linked audio\n" );

            gst_object_unref( targetsink );
            gst_object_unref( element );
        }
    }

    gst_caps_unref( caps );

    return;
}

bool setState( GstState state ) {
    GstStateChangeReturn sret;

    GstClockTime timeout = 10 * GST_SECOND; //10s

    log( "setState: Setting pipeline state to %s\n", gst_element_state_get_name( state ) );
    sret = gst_element_set_state ( _pipeline, state );
    if( sret == GST_STATE_CHANGE_ASYNC )
        log( "setState: going to %s state is ASYNC\n", gst_element_state_get_name( state ) );

    log( "setState: getting state\n" );
    sret = gst_element_get_state( _pipeline, NULL, NULL, timeout);

    if( sret != GST_STATE_CHANGE_SUCCESS && sret != GST_STATE_CHANGE_NO_PREROLL ) {
        if( sret == GST_STATE_CHANGE_FAILURE )
            log( "setState: failed to go into %s state (failure returned)\n", gst_element_state_get_name( state ) );
        else //ASYNC
            log( "setState: failed to go into %s state (async returned)\n", gst_element_state_get_name( state ) );

        if( state != GST_STATE_NULL ) {
            log( "setState: fallback to pipeline NULL state\n");

            gst_element_set_state ( _pipeline, GST_STATE_NULL );

            /* wait until state change either completes or fails */
            sret = gst_element_get_state( _pipeline, &state, NULL, timeout );
            if( sret == GST_STATE_CHANGE_SUCCESS )
                log( "setState: fallback to pipeline NULL state succeed\n" );
            if( sret == GST_STATE_CHANGE_FAILURE )
                log( "setState: fallback to pipeline NULL state failed (failure returned)\n" );
            else //ASYNC
                log( "setState: fallback to pipeline NULL state failed (async returned)\n" );
        }

        return false;
    }

    log( "setState: Setting pipeline state to %s DONE successfully\n", gst_element_state_get_name( state ) );

    return true;
}

/*
 * pipeline without audio:
 * rtspsrc location="..." ! rtph264depay ! h264parse ! mp4mux ! filesink location=".."
 */
bool init() {
    GstBus* bus = NULL;
    _loop = g_main_loop_new( NULL, false );

    if( _filename.empty( ) && _rtmpLocation.empty( ) ) {
        log( "init: Can't init pipeline, no filename nor rtmpLocation provided\n");
        return false;
    }

    _pipeline = gst_pipeline_new( "mypipeline" );
    gst_pipeline_set_auto_flush_bus( GST_PIPELINE( _pipeline ), false );

    if( !_pipeline ) {
        log( "init: Failed to init pipeline\n");
        return false;
    }

    GstElement *rtspsrc = NULL,
        *rtph264depay = NULL,
        *h264parse = NULL,
        *aacparse = NULL,
        *rtpmp4gdepay = NULL,
        *muxer = NULL,
        *sink = NULL;

    bus = gst_pipeline_get_bus( GST_PIPELINE( _pipeline ) );
    _busWatchId = gst_bus_add_watch( bus, bus_call, NULL );
    gst_object_unref( bus );

    rtspsrc = gst_element_factory_make( "rtspsrc", "src" );
    rtph264depay = gst_element_factory_make( "rtph264depay", "rtph264depay" );
    h264parse = gst_element_factory_make( "h264parse", "h264parse" );

    if( !rtspsrc || !rtph264depay || !h264parse ) {
        gst_object_unref( GST_OBJECT( _pipeline ) );
        _pipeline = NULL;
        g_main_loop_unref( _loop );
        _loop = NULL;
        log( "init: Failed to create element rtspsrc or rtph264depay or h264parse\n");
        return false;
    }

    //rtsp source
    std::ostringstream rtspLocation;
    rtspLocation << "rtsp://192.168.10.26/axis-media/media.amp?videocodec=h264&resolution=320x180";

    g_object_set( G_OBJECT( rtspsrc ), "location", rtspLocation.str( ).c_str( ), NULL );
    if( !_login.empty( ) )
        g_object_set( G_OBJECT( rtspsrc ), "user-id", _login.c_str( ), NULL );
    if( !_password.empty( ) )
        g_object_set( G_OBJECT( rtspsrc ), "user-pw", _password.c_str( ), NULL );

    if( !_filename.empty( ) ) {
        muxer = gst_element_factory_make( "mp4mux", "mux" );
        if( !muxer ) {
            gst_object_unref( GST_OBJECT( _pipeline ) );
            _pipeline = NULL;
            g_main_loop_unref( _loop );
            _loop = NULL;
            log( "init: Failed to create element mp4mux\n" );
            return false;
        }

        if( !_filename.compare( "stdout" ) ) {
            sink = gst_element_factory_make( "fdsink", NULL );
            if( sink )
                g_object_set( G_OBJECT( sink ), "fd", 1, NULL );
        }
        else {
            sink = gst_element_factory_make( "filesink", NULL );
            if( sink )
                g_object_set( G_OBJECT( sink ), "location", _filename.c_str( ), NULL );
        }

        if( !sink ) {
            gst_object_unref( GST_OBJECT( _pipeline ) );
            _pipeline = NULL;
            g_main_loop_unref( _loop );
            _loop = NULL;

            log( "init: Failed to create element fdsink/filesink\n");
            return false;
        }

        //gstreamer-1.0: no more available option
#if ! GST_CHECK_VERSION(1,0,0)
        g_object_set( G_OBJECT( rtph264depay ), "byte-stream", false, NULL );
#endif
    }
    else {
        muxer = gst_element_factory_make( "flvmux", "mux" );
        sink = gst_element_factory_make( "rtmpsink", NULL );

        if( !muxer || !sink ) {
            gst_object_unref( GST_OBJECT( _pipeline ) );
            _pipeline = NULL;
            g_main_loop_unref( _loop );
            _loop = NULL;

            log( "init: Failed to create element rtmpsink or flvmux\n");
            return false;
        }

        g_object_set( G_OBJECT( muxer ), "streamable", true, NULL );

        //gstreamer-1.0: no more available option
#if ! GST_CHECK_VERSION(1,0,0)
        g_object_set( G_OBJECT( rtph264depay ), "byte-stream", true, NULL );
#endif
        std::string sinkLocation = _rtmpLocation + " live=1 timeout=10";

        g_object_set( G_OBJECT( sink ), "location", sinkLocation.c_str( ), NULL );
        g_object_set( G_OBJECT( sink ), "max-lateness", (gint64)1000000000, NULL);

        if( _useAudio ) {
            rtpmp4gdepay = gst_element_factory_make( "rtpmp4gdepay", "rtpmp4gdepay" );
            aacparse = gst_element_factory_make( "aacparse", "aacparse" );
            if( !rtpmp4gdepay || !aacparse ) {
                gst_object_unref( GST_OBJECT( _pipeline ) );
                _pipeline = NULL;
                g_main_loop_unref( _loop );
                _loop = NULL;

                log( "init: Failed to create element rtmp4gdepay or aacparse\n" );
                return false;
            }
        }
    }

    gst_bin_add_many( GST_BIN( _pipeline ), rtspsrc, rtph264depay, h264parse, muxer, sink, NULL );
    if( ! gst_element_link_many( rtph264depay, h264parse, muxer, sink, NULL ) ) {
        log( "init: Failed to link elements rtph264depay, h264parse, muxer, sink\n" );
        gst_object_unref( GST_OBJECT( _pipeline ) );
        _pipeline = NULL;
        g_main_loop_unref( _loop );
        _loop = NULL;

        return false;
    }

    if( _useAudio && rtpmp4gdepay ) {
        gst_bin_add_many( GST_BIN( _pipeline ), rtpmp4gdepay, aacparse, NULL );
        if( ! gst_element_link_many( rtpmp4gdepay, aacparse, muxer, NULL ) ) {
            log( "init: Failed to link elements rtpmp4gdepay, aacparse, muxer\n" );
            gst_object_unref( GST_OBJECT( _pipeline ) );
            _pipeline = NULL;
            g_main_loop_unref( _loop );
            _loop = NULL;

            return false;
        }
    }

    g_signal_connect( rtspsrc, "pad-added", G_CALLBACK( on_pad_added ), _pipeline );

    if( ! setState( GST_STATE_PAUSED ) ) {
        GstMessage *msg;

        // check if there is an error message with details on the bus
        msg = gst_bus_poll( bus, GST_MESSAGE_ERROR, 0 );
        if( msg ) {
            GError *err = NULL;

            gst_message_parse_error( msg, &err, NULL );
            log( "init: setState to PAUSED failed, error message: %s\n", err->message );
            g_error_free( err );
            gst_message_unref( msg );
        }

        gst_object_unref( GST_OBJECT( _pipeline ) );
        _pipeline = NULL;
        g_main_loop_unref( _loop );
        _loop = NULL;

        return false;
    }
    log( "init: Pipeline state change to PAUSED succeed\n");

    return true;
}

bool start() {
    bool rv = setState( GST_STATE_PLAYING );

    if( rv ) {
        log( "start: Running...\n" );
        g_main_loop_run( _loop );

        /* Out of the main loop, clean up nicely */
        log( "start: Returned, stopping playback, setting state to NULL\n" );
        rv = setState( GST_STATE_NULL );
    }

    log( "start: Deleting pipeline\n");
    gst_object_unref ( GST_OBJECT( _pipeline ) );
    _pipeline = NULL;
    g_source_remove( _busWatchId );
    _busWatchId = 0;
    if( _timeout )
        g_source_remove( _timeout );

    g_main_loop_unref( _loop );
    _loop = NULL;

    return rv;
}

bool die = false;
static void theEnd( int signal ) {
    log( "theEnd: end signal catched => %d\n", signal );
    die = true;
}

int main( int argc, char ** argv ) {
    signal( SIGTERM, theEnd ); /* default signal sent with kill */
    signal( SIGINT, theEnd ); /* Ctrl-C */
    signal( SIGHUP, theEnd );

    {
        gst_init_check(NULL, NULL, NULL);
        gst_debug_set_active(false);
        gst_debug_set_default_threshold(GST_LEVEL_NONE);
    }

    int i=0;
    while( ! die ) {
        log("\n\n%d\n\n", i);
        init();
        start();
        i++;
    }

    gst_deinit();

    return 0;
}