gstreamer appsrc: saving application-generated media to file saving application-generated media to file

Example

#include <string.h>

#include <gst/gst.h>

#include <gst/app/gstappsrc.h>

/*
 * an example application of using appsrc in push mode to create a file.
 * from buffers we push into the pipeline.
 */

/* S16LE 10ms frame audio */
#define BUFFER_SIZE 160

/* 300 frames = 5 minutes */
#define TOTAL_FRAMES 30000

#define QUEUED_FRAMES 30

typedef struct
{
  GMainLoop *loop;
  GstElement *pipeline;
  GstElement *source;
  guint source_id;
  guint num_frame;
} AppData;

/* This method is called by the idle GSource in the mainloop. We feed 1 buffer
 * of BUFFER_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_buffer (AppData * app)
{
  gpointer raw_buffer;
  GstBuffer *app_buffer;
  GstMemory *mem;
  GstFlowReturn ret;

  app->num_frame++;

  if (app->num_frame >= TOTAL_FRAMES) {
    /* we are EOS, send end-of-stream and remove the source */
    g_signal_emit_by_name (app->source, "end-of-stream", &ret);
    return FALSE;
  }

  app_buffer = gst_buffer_new();

  mem = gst_allocator_alloc (NULL, BUFFER_SIZE, NULL);

  gst_buffer_append_memory(app_buffer, mem);

  gst_buffer_set_size(app_buffer, BUFFER_SIZE);

  /* Setting the correct timestamp for the buffer is very important, otherwise the
   * resulting file won't be created correctly */
  GST_BUFFER_TIMESTAMP(app_buffer) = (GstClockTime)((app->num_frame / 100.0) * 1e9);

  /* push new buffer */
  g_signal_emit_by_name (app->source, "push-buffer", app_buffer, &ret);

  gst_buffer_unref(app_buffer);

  if (ret != GST_FLOW_OK) {
    /* some error, stop sending data */
    return FALSE;
  }

  return TRUE;
}

/* This signal callback is called when appsrc needs data, we add an idle handler
 * to the mainloop to start pushing data into the appsrc */
static void
start_feed (GstElement * pipeline, guint size, AppData * app)
{
  if (app->source_id == 0) {
        g_print ("start feeding at frame %i\n", app->num_frame);
    app->source_id = g_idle_add ((GSourceFunc) push_buffer, app);
  }
}

/* This callback is called when appsrc has enough data and we can stop sending.
 * We remove the idle handler from the mainloop */
static void
stop_feed (GstElement * pipeline, AppData * app)
{
  if (app->source_id != 0) {
        g_print ("stop feeding at frame %i\n", app->num_frame);
    g_source_remove (app->source_id);
    app->source_id = 0;
  }
}

/* called when we get a GstMessage from the pipeline when we get EOS, we
 * exit the mainloop and this testapp. */
static gboolean
on_pipeline_message (GstBus * bus, GstMessage * message, AppData * app)
{
  GstState state, pending;

  switch (GST_MESSAGE_TYPE (message)) {
    case GST_MESSAGE_EOS:
      g_print ("Received End of Stream message\n");
      g_main_loop_quit (app->loop);
      break;
    case GST_MESSAGE_ERROR:
      {
        g_print ("Received error\n");

        GError *err = NULL;
        gchar *dbg_info = NULL;

        gst_message_parse_error (message, &err, &dbg_info);
        g_printerr ("ERROR from element %s: %s\n",
            GST_OBJECT_NAME (message->src), err->message);
        g_printerr ("Debugging info: %s\n", (dbg_info) ? dbg_info : "none");
        g_error_free (err);
        g_free (dbg_info);
      }

      g_main_loop_quit (app->loop);
      break;
    case GST_MESSAGE_STATE_CHANGED:
          gst_element_get_state(app->source, &state, &pending, GST_CLOCK_TIME_NONE);
      /* g_print ("State changed from %i to %i\n", state, pending); */
      break;
        default:
      break;
  }
  return TRUE;
}

int
main (int argc, char *argv[])
{
  AppData *app = NULL;
  GstBus *bus = NULL;
  GstElement *appsrc = NULL;

  gst_init (&argc, &argv);

  app = g_new0 (AppData, 1);

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

  /* setting up pipeline, we push media data into this pipeline that will
   * then be recorded to a file, encoded with a codec*/
  app->pipeline = gst_parse_launch ("appsrc is-live=true name=source caps=audio/x-raw,format=S16LE,rate=8000,channels=1,layout=interleaved ! flacenc ! oggmux ! filesink location=flac-audio.ogg", NULL);

  if (app->pipeline == NULL) {
    g_print ("Bad pipeline\n");
    return -1;
  }

  appsrc = gst_bin_get_by_name (GST_BIN (app->pipeline), "source");

  /* setting maximum of bytes queued */
  gst_app_src_set_max_bytes((GstAppSrc *)appsrc, QUEUED_FRAMES * BUFFER_SIZE);

  /* uncomment the next line to block when appsrc has buffered enough */
  /* g_object_set (appsrc, "block", TRUE, NULL); */
  app->source = appsrc;

  /* add watch for messages */
  bus = gst_element_get_bus (app->pipeline);
  gst_bus_add_watch (bus, (GstBusFunc) on_pipeline_message, app);
  gst_object_unref (bus);

  /* configure the appsrc, we will push data into the appsrc from the
   * mainloop */
  g_signal_connect (app->source, "need-data", G_CALLBACK (start_feed), app);
  g_signal_connect (app->source, "enough-data", G_CALLBACK (stop_feed), app);

  /* go to playing and wait in a mainloop */
  gst_element_set_state (app->pipeline, GST_STATE_PLAYING);

  /* this mainloop is stopped when we receive an error or EOS */
  g_print ("Creating movie...\n");
  g_main_loop_run (app->loop);
  g_print ("Done.\n");

  gst_app_src_end_of_stream (GST_APP_SRC (app->source));

  gst_element_set_state (app->pipeline, GST_STATE_NULL);

  /* Cleaning up */
  gst_object_unref (app->source);
  gst_object_unref (app->pipeline);
  g_main_loop_unref (app->loop);
  g_free (app);

  return 0;
}