text: //
  // GStreamer tutorial #8 from here
  // https://gstreamer.freedesktop.org/documentation/tutorials/basic/short-cutting-the-pipeline.html
  // 
  
  #include <gst/gst.h>
  #include <gst/audio/audio.h>
  #include <string.h>
  
  #define CHUNK_SIZE 1024   /* Amount of bytes we are sending in each buffer */
  #define SAMPLE_RATE 44100 /* Samples per second we are sending */
  
  
  /* Structure to contain all our information, so we can pass it to callbacks */
  typedef struct _CustomData {
    GstElement *pipeline, *app_source, *tee, *audio_queue, *audio_convert1, *audio_resample, *audio_sink;
    GstElement *video_queue, *audio_convert2, *visual, *video_convert, *video_sink;
    GstElement *app_queue, *app_sink;
  
    guint64 num_samples;   /* Number of samples generated so far (for timestamp generation) */
    gfloat a, b, c, d;     /* For waveform generation */
  
    guint sourceid;        /* To control the GSource */
  
    GMainLoop *main_loop;  /* GLib's Main Loop */
  } CustomData;
  
  
  /* This method is called by the idle GSource in the mainloop, to feed CHUNK_SIZE bytes into appsrc.
   * The ide handler is added to the mainloop when appsrc requests us to start sending data (need-data signal)
   * and is removed when appsrc has enough data (enough-data signal).
   */
  static gboolean push_data (CustomData *data) {
    GstBuffer *buffer;
    GstFlowReturn ret;
    int i;
    GstMapInfo map;
    gint16 *raw;
    gint num_samples = CHUNK_SIZE / 2; /* Because each sample is 16 bits */
    gfloat freq;
  
    /* Create a new empty buffer */
    buffer = gst_buffer_new_and_alloc (CHUNK_SIZE);
  
    /* Set its timestamp and duration */
    GST_BUFFER_TIMESTAMP (buffer) = gst_util_uint64_scale (data->num_samples, GST_SECOND, SAMPLE_RATE);
    GST_BUFFER_DURATION (buffer) = gst_util_uint64_scale (CHUNK_SIZE, GST_SECOND, SAMPLE_RATE);
  
    /* Generate some psychodelic waveforms */
    gst_buffer_map (buffer, &map, GST_MAP_WRITE);
    raw = (gint16 *)map.data;
    data->c += data->d;
    data->d -= data->c / 1000;
    freq = 1100 + 1000 * data->d;
    for (i = 0; i < num_samples; i++) {
      data->a += data->b;
      data->b -= data->a / freq;
      raw[i] = (gint16)(500 * data->a);
    }
    gst_buffer_unmap (buffer, &map);
    data->num_samples += num_samples;
  
    /* Push the buffer into the appsrc */
    g_signal_emit_by_name (data->app_source, "push-buffer", buffer, &ret);
  
    /* Free the buffer now that we are done with it */
    gst_buffer_unref (buffer);
  
    if (ret != GST_FLOW_OK) {
      /* We got some error, stop sending data */
      return FALSE;
    }
  
    return TRUE;
  }
  
  /* This signal callback triggers when appsrc needs data. Here, we add an idle handler
   * to the mainloop to start pushing data into the appsrc */
  static void start_feed (GstElement *source, guint size, CustomData *data) {
    if (data->sourceid == 0) {
      g_print ("Start feeding\n");
      data->sourceid = g_idle_add ((GSourceFunc) push_data, data);
    }
  }
  
  /* This callback triggers when appsrc has enough data and we can stop sending.
   * We remove the idle handler from the mainloop */
  static void stop_feed (GstElement *source, CustomData *data) {
    if (data->sourceid != 0) {
      g_print ("Stop feeding\n");
      g_source_remove (data->sourceid);
      data->sourceid = 0;
    }
  }
  
  /* The appsink has received a buffer */
  static void new_sample (GstElement *sink, CustomData *data) {
    GstSample *sample;
  
    /* Retrieve the buffer */
    g_signal_emit_by_name (sink, "pull-sample", &sample);
    if (sample) {
      /* The only thing we do in this example is print a * to indicate a received buffer */
      g_print ("*");
      gst_buffer_unref (sample);
    }
  }
  
  /* This function is called when an error message is posted on the bus */
  static void error_cb (GstBus *bus, GstMessage *msg, CustomData *data) {
    GError *err;
    gchar *debug_info;
  
    /* Print error details on the screen */
    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);
  
    g_main_loop_quit (data->main_loop);
  }
  
  
  
  int main(int argc, char *argv[])
  {
    CustomData data;
    GstPad *tee_audio_pad, *tee_video_pad, *tee_app_pad;
    GstPad *queue_audio_pad, *queue_video_pad, *queue_app_pad;
    GstAudioInfo info;
    GstCaps *audio_caps;
    GstBus *bus;
  
    /* Initialize cumstom data structure */
    memset (&data, 0, sizeof (data));
    data.b = 1; /* For waveform generation */
    data.d = 1;
  
    /* Initialize GStreamer */
    gst_init (&argc, &argv);
  
    /* Create the elements */
    data.app_source = gst_element_factory_make ("appsrc", "audio_source");
    data.tee = gst_element_factory_make ("tee", "tee");
    data.audio_queue = gst_element_factory_make ("queue", "audio_queue");
    data.audio_convert1 = gst_element_factory_make ("audioconvert", "audio_convert1");
    data.audio_resample = gst_element_factory_make ("audioresample", "audio_resample");
    data.audio_sink = gst_element_factory_make ("autoaudiosink", "audio_sink");
    data.video_queue = gst_element_factory_make ("queue", "video_queue");
    data.audio_convert2 = gst_element_factory_make ("audioconvert", "audio_convert2");
    data.visual = gst_element_factory_make ("wavescope", "visual");
    data.video_convert = gst_element_factory_make ("videoconvert", "csp");
    data.video_sink = gst_element_factory_make ("autovideosink", "video_sink");
    data.app_queue = gst_element_factory_make ("queue", "app_queue");
    data.app_sink = gst_element_factory_make ("appsink", "app_sink");
  
    /* Create the empty pipeline */
    data.pipeline = gst_pipeline_new ("test-pipeline");
  
    if (!data.pipeline || !data.app_source || !data.tee || !data.audio_queue || !data.audio_convert1 ||
        !data.audio_resample || !data.audio_sink || !data.video_queue || !data.audio_convert2 || !data.visual ||
        !data.video_convert || !data.video_sink || !data.app_queue || !data.app_sink) {
      g_printerr ("Not all elements could be created.\n");
      return -1;
    }
  
    /* Configure wavescope */
    g_object_set (data.visual, "shader", 0, "style", 0, NULL);
  
    /* Configure appsrc */
    gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16, SAMPLE_RATE, 1, NULL);
    audio_caps = gst_audio_info_to_caps (&info);
    g_object_set (data.app_source, "caps", audio_caps, "format", GST_FORMAT_TIME, NULL);
    g_signal_connect (data.app_source, "need-data", G_CALLBACK (start_feed), &data);
    g_signal_connect (data.app_source, "enough-data", G_CALLBACK (stop_feed), &data);
  
    /* Configure appsink */
    g_object_set (data.app_sink, "emit-signals", TRUE, "caps", audio_caps, NULL);
    g_signal_connect (data.app_sink, "new-sample", G_CALLBACK (new_sample), &data);
    gst_caps_unref (audio_caps);
  //  g_free (audio_caps_text);
  
    /* Link all elements that can be automatically linked because they have "Always" pads */
    gst_bin_add_many (GST_BIN (data.pipeline), data.app_source, data.tee, data.audio_queue, data.audio_convert1, data.audio_resample,
        data.audio_sink, data.video_queue, data.audio_convert2, data.visual, data.video_convert, data.video_sink, data.app_queue,
        data.app_sink, NULL);
    if (gst_element_link_many (data.app_source, data.tee, NULL) != TRUE ||
        gst_element_link_many (data.audio_queue, data.audio_convert1, data.audio_resample, data.audio_sink, NULL) != TRUE ||
        gst_element_link_many (data.video_queue, data.audio_convert2, data.visual, data.video_convert, data.video_sink, NULL) != TRUE ||
        gst_element_link_many (data.app_queue, data.app_sink, NULL) != TRUE) {
      g_printerr ("Elements could not be linked.\n");
      gst_object_unref (data.pipeline);
      return -1;
    }
  
    /* Manually link the Tee, which has "Request" pads */
    tee_audio_pad = gst_element_get_request_pad (data.tee, "src_%u");
    g_print ("Obtained request pad %s for audio branch.\n", gst_pad_get_name (tee_audio_pad));
    queue_audio_pad = gst_element_get_static_pad (data.audio_queue, "sink");
    tee_video_pad = gst_element_get_request_pad (data.tee, "src_%u");
    g_print ("Obtained request pad %s for video branch.\n", gst_pad_get_name (tee_video_pad));
    queue_video_pad = gst_element_get_static_pad (data.video_queue, "sink");
    tee_app_pad = gst_element_get_request_pad (data.tee, "src_%u");
    g_print ("Obtained request pad %s for app branch.\n", gst_pad_get_name (tee_app_pad));
    queue_app_pad = gst_element_get_static_pad (data.app_queue, "sink");
    if (gst_pad_link (tee_audio_pad, queue_audio_pad) != GST_PAD_LINK_OK ||
        gst_pad_link (tee_video_pad, queue_video_pad) != GST_PAD_LINK_OK ||
        gst_pad_link (tee_app_pad, queue_app_pad) != GST_PAD_LINK_OK) {
      g_printerr ("Tee could not be linked\n");
      gst_object_unref (data.pipeline);
      return -1;
    }
    gst_object_unref (queue_audio_pad);
    gst_object_unref (queue_video_pad);
    gst_object_unref (queue_app_pad);
  
    /* Instruct the bus to emit signals for each received message, and connect to the interesting signals */
    bus = gst_element_get_bus (data.pipeline);
    gst_bus_add_signal_watch (bus);
    g_signal_connect (G_OBJECT (bus), "message::error", (GCallback)error_cb, &data);
    gst_object_unref (bus);
  
    /* Start playing the pipeline */
    gst_element_set_state (data.pipeline, GST_STATE_PLAYING);
  
    /* Create a GLib Main Loop and set it to run */
    data.main_loop = g_main_loop_new (NULL, FALSE);
    g_main_loop_run (data.main_loop);
  
    /* Release the request pads from the Tee, and unref them */
    gst_element_release_request_pad (data.tee, tee_audio_pad);
    gst_element_release_request_pad (data.tee, tee_video_pad);
    gst_element_release_request_pad (data.tee, tee_app_pad);
    gst_object_unref (tee_audio_pad);
    gst_object_unref (tee_video_pad);
    gst_object_unref (tee_app_pad);
  
    /* Free resources */
    gst_element_set_state (data.pipeline, GST_STATE_NULL);
    gst_object_unref (data.pipeline);
    return 0;
  }
  