Gstreamer를 수정하여 마이크 오디오를 캡처하고 버퍼링하거나 원시 파일로 덤프하면 아무 내용도 저장되지 않습니다.

Question

마이크 오디오를 캡처하여 파일로 저장하려고합니다. 하지만 작동하지 않는, 나는 할당하는 동안 파일을 재생할 수 있습니다. 어떻게 마이크를 활성화하고 버퍼링하거나 원시 .odd/vorbis로 저장하거나 덤프 할 수 있습니까?

#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, *source, *demuxer, *decoder, *conv, *sink; 
    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; 
    } 


    /* Create gstreamer elements */ 
    pipeline = gst_pipeline_new ("audio-player"); 
    source = gst_element_factory_make ("filesrc",  "file-source"); 
    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 ("autoaudiosink", "audio-output"); 

    if (!pipeline || !source || !demuxer || !decoder || !conv || !sink) { 
    g_printerr ("One element could not be created. Exiting.\n"); 
    return -1; 
    } 

    /* Set up the pipeline */ 

    /* we set the input filename to the source element */ 
    g_object_set (G_OBJECT (source), "location", argv[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 */ 
    /* file-source | ogg-demuxer | vorbis-decoder | converter | alsa-output */ 
    gst_bin_add_many (GST_BIN (pipeline), 
        source, demuxer, decoder, conv, sink, NULL); 

    /* we link the elements together */ 
    /* file-source -> ogg-demuxer ~> vorbis-decoder -> converter -> alsa-output */ 
    gst_element_link (source, demuxer); 
    gst_element_link_many (decoder, conv, sink, NULL); 
    g_signal_connect (demuxer, "pad-added", G_CALLBACK (on_pad_added), decoder); 

    /* note that the demuxer will be linked to the decoder dynamically. 
    The reason is that Ogg may contain various streams (for example 
    audio and video). The source pad(s) will be created at run time, 
    by the demuxer when it detects the amount and nature of streams. 
    Therefore we connect a callback function which will be executed 
    when the "pad-added" is emitted.*/ 


    /* Set the pipeline to "playing" state*/ 
    g_print ("Now playing: %s\n", argv[1]); 
    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; 
} 

Answer 1

실제로 무엇이 문제입니까?

gst-launch osssrc device=<mic i/p dev> ! audioconvert ! vorbisenc ! oggmux ! filesink location=dump.ogg 

Answer 2

으로합니다. 천재, 작동 해. 방금 질문에 대답했습니다. 하지만 불행히도 크로스 플랫폼 애플리케이션을 작성해야한다. 리눅스가 아니라면 어떻게해야합니까?