gstreamer/tests/examples/app/appsink-src.c

#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;
}