#include <gst/gst.h>

#include <string.h>

#include <gst/app/gstappsrc.h>

#include <gst/app/gstappsink.h>

#include <gst/app/gstappbuffer.h>

/* these are the caps we are going to pass through the appsink and appsrc */

const gchar *audio_caps =

    "audio/x-raw-int,channels=1,rate=8000,signed=(boolean)true,width=16,depth=16,endianness=1234";

typedef struct

{

  GMainLoop *loop;

  GstElement *source;

  GstElement *sink;

} ProgramData;

/* called when the appsink notifies us that there is a new buffer ready for

 * processing */

static void

on_new_buffer_from_source (GstElement * elt, ProgramData * data)

{

  guint size;

  gpointer raw_buffer;

  GstBuffer *app_buffer, *buffer;

  GstElement *source;

  /* get the buffer from appsink */

  buffer = gst_app_sink_pull_buffer (GST_APP_SINK (elt));

  /* turn it into an app buffer, it's not really needed, we could simply push

   * the retrieved buffer from appsink into appsrc just fine.  */

  size = GST_BUFFER_SIZE (buffer);

  g_print ("Pushing a buffer of size %d\n", size);

  raw_buffer = g_malloc0 (size);

  memcpy (raw_buffer, GST_BUFFER_DATA (buffer), size);

  app_buffer = gst_app_buffer_new (raw_buffer, size, g_free, raw_buffer);

  /* newer basesrc will set caps for use automatically but it does not really

   * hurt to set it on the buffer again */

  gst_buffer_set_caps (app_buffer, GST_BUFFER_CAPS (buffer));

  /* we don't need the appsink buffer anymore */

  gst_buffer_unref (buffer);

  /* get source an push new buffer */

  source = gst_bin_get_by_name (GST_BIN (data->sink), "testsource");

  gst_app_src_push_buffer (GST_APP_SRC (source), app_buffer);

}

/* called when we get a GstMessage from the source pipeline when we get EOS, we

 * notify the appsrc of it. */

static gboolean

on_source_message (GstBus * bus, GstMessage * message, ProgramData * data)

{

  GstElement *source;

  switch (GST_MESSAGE_TYPE (message)) {

    case GST_MESSAGE_EOS:

      g_print ("The source got dry\n");

      source = gst_bin_get_by_name (GST_BIN (data->sink), "testsource");

      gst_app_src_end_of_stream (GST_APP_SRC (source));

      break;

    case GST_MESSAGE_ERROR:

      g_print ("Received error\n");

      g_main_loop_quit (data->loop);

      break;

    default:

      break;

  }

  return TRUE;

}

/* called when we get a GstMessage from the sink pipeline when we get EOS, we

 * exit the mainloop and this testapp. */

static gboolean

on_sink_message (GstBus * bus, GstMessage * message, ProgramData * data)

{

  /* nil */

  switch (GST_MESSAGE_TYPE (message)) {

    case GST_MESSAGE_EOS:

      g_print ("Finished playback\n");

      g_main_loop_quit (data->loop);

      break;

    case GST_MESSAGE_ERROR:

      g_print ("Received error\n");

      g_main_loop_quit (data->loop);

      break;

    default:

      break;

  }

  return TRUE;

}

int

main (int argc, char *argv[])

{

  gchar *filename = NULL;

  ProgramData *data = NULL;

  gchar *string = NULL;

  GstBus *bus = NULL;

  GstElement *testsink = NULL;

  GstElement *testsource = NULL;

  gst_init (&argc, &argv);

  if (argc == 2)

    filename = g_strdup (argv[1]);

  else

    filename = g_strdup ("/usr/share/sounds/ekiga/ring.wav");

  data = g_new0 (ProgramData, 1);

  data->loop = g_main_loop_new (NULL, FALSE);

  /* setting up source pipeline, we read from a file and convert to our desired

   * caps. */

  string =

      g_strdup_printf

      ("filesrc location=\"%s\" ! wavparse ! audioconvert ! audioresample ! appsink caps=\"%s\" name=testsink",

      filename, audio_caps);

  g_free (filename);

  data->source = gst_parse_launch (string, NULL);

  g_free (string);

  if (data->source == NULL) {

    g_print ("Bad source\n");

    return -1;

  }

  /* to be notified of messages from this pipeline, mostly EOS */

  bus = gst_element_get_bus (data->source);

  gst_bus_add_watch (bus, (GstBusFunc) on_source_message, data);

  gst_object_unref (bus);

  /* we use appsink in push mode, it sends us a signal when data is available

   * and we pull out the data in the signal callback. We want the appsink to

   * push as fast as it can, hence the sync=false */

  testsink = gst_bin_get_by_name (GST_BIN (data->source), "testsink");

  g_object_set (G_OBJECT (testsink), "emit-signals", TRUE, "sync", FALSE, NULL);

  g_signal_connect (testsink, "new-buffer",

      G_CALLBACK (on_new_buffer_from_source), data);

  gst_object_unref (testsink);

  /* setting up sink pipeline, we push audio data into this pipeline that will

   * then play it back using the default audio sink. We have no blocking

   * behaviour on the src which means that we will push the entire file into

   * memory. */

  string =

      g_strdup_printf ("appsrc name=testsource caps=\"%s\" ! autoaudiosink",

      audio_caps);

  data->sink = gst_parse_launch (string, NULL);

  g_free (string);

  if (data->sink == NULL) {

    g_print ("Bad sink\n");

    return -1;

  }

  testsource = gst_bin_get_by_name (GST_BIN (data->sink), "testsource");

  /* configure for time-based format */

  g_object_set (testsource, "format", GST_FORMAT_TIME, NULL);

  /* uncomment the next line to block when appsrc has buffered enough */

  /* g_object_set (testsource, "block", TRUE, NULL); */

  gst_object_unref (testsource);

  bus = gst_element_get_bus (data->sink);

  gst_bus_add_watch (bus, (GstBusFunc) on_sink_message, data);

  gst_object_unref (bus);

  /* launching things */

  gst_element_set_state (data->sink, GST_STATE_PLAYING);

  gst_element_set_state (data->source, GST_STATE_PLAYING);

  /* let's run !, this loop will quit when the sink pipeline goes EOS or when an

   * error occurs in the source or sink pipelines. */

  g_print ("Let's run!\n");

  g_main_loop_run (data->loop);

  g_print ("Going out\n");

  gst_element_set_state (data->source, GST_STATE_NULL);

  gst_element_set_state (data->sink, GST_STATE_NULL);

  gst_object_unref (data->source);

  gst_object_unref (data->sink);

  g_main_loop_unref (data->loop);

  g_free (data);

  return 0;

}