gstreamer/subprojects/gst-docs/markdown/tutorials/playback/short-cutting-the-pipeline.md

Playback tutorial 3: Short-cutting the pipeline

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Goal

showed how an application can manually extract or inject data into a pipeline by using two special elements called appsrc and appsink. playbin allows using these elements too, but the method to connect them is different. To connect an appsink to playbin see . This tutorial shows:

• How to connect appsrc with playbin
• How to configure the appsrc

A playbin waveform generator

Copy this code into a text file named playback-tutorial-3.c.

playback-tutorial-3.c

#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;
  GstElement *app_source;

  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 (num_samples, 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;
  }
}

/* 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);
}

/* This function is called when playbin has created the appsrc element, so we have
 * a chance to configure it. */
static void source_setup (GstElement *pipeline, GstElement *source, CustomData *data) {
  GstAudioInfo info;
  GstCaps *audio_caps;

  g_print ("Source has been created. Configuring.\n");
  data->app_source = source;

  /* 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 (source, "caps", audio_caps, "format", GST_FORMAT_TIME, NULL);
  g_signal_connect (source, "need-data", G_CALLBACK (start_feed), data);
  g_signal_connect (source, "enough-data", G_CALLBACK (stop_feed), data);
  gst_caps_unref (audio_caps);
}

int main(int argc, char *argv[]) {
  CustomData data;
  GstBus *bus;

  /* Initialize custom data structure */
  memset (&data, 0, sizeof (data));
  data.b = 1; /* For waveform generation */
  data.d = 1;

  /* Initialize GStreamer */
  gst_init (&argc, &argv);

  /* Create the playbin element */
  data.pipeline = gst_parse_launch ("playbin uri=appsrc://", NULL);
  g_signal_connect (data.pipeline, "source-setup", G_CALLBACK (source_setup), &data);

  /* 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);

  /* Free resources */
  gst_element_set_state (data.pipeline, GST_STATE_NULL);
  gst_object_unref (data.pipeline);
  return 0;
}

![information] If you need help to compile this code, refer to the Building the tutorials section for your platform: [Mac] or [Windows] or use this specific command on Linux:

gcc playback-tutorial-3.c -o playback-tutorial-3 `pkg-config --cflags --libs gstreamer-1.0 gstreamer-audio-1.0`

If you need help to run this code, refer to the Running the tutorials section for your platform: [Mac OS X], [Windows][1], for [iOS] or for [android].

This tutorial opens a window and displays a movie, with accompanying audio. The media is fetched from the Internet, so the window might take a few seconds to appear, depending on your connection speed. In the console window, you should see a message indicating where the media is being stored, and a text graph representing the downloaded portions and the current position. A buffering message appears whenever buffering is required, which might never happen is your network connection is fast enough

Required libraries: gstreamer-1.0 gstreamer-audio-1.0

To use an appsrc as the source for the pipeline, simply instantiate a playbin and set its URI to appsrc://

/* Create the playbin element */
data.pipeline = gst_parse_launch ("playbin uri=appsrc://", NULL);

playbin will create an internal appsrc element and fire the source-setup signal to allow the application to configure it:

g_signal_connect (data.pipeline, "source-setup", G_CALLBACK (source_setup), &data);

In particular, it is important to set the caps property of appsrc, since, once the signal handler returns, playbin will instantiate the next element in the pipeline according to these caps:

/* This function is called when playbin has created the appsrc element, so we have
 * a chance to configure it. */
static void source_setup (GstElement *pipeline, GstElement *source, CustomData *data) {
  GstAudioInfo info;
  GstCaps *audio_caps;

  g_print ("Source has been created. Configuring.\n");
  data->app_source = source;

  /* 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 (source, "caps", audio_caps, "format", GST_FORMAT_TIME, NULL);
  g_signal_connect (source, "need-data", G_CALLBACK (start_feed), data);
  g_signal_connect (source, "enough-data", G_CALLBACK (stop_feed), data);
  gst_caps_unref (audio_caps);
}

The configuration of the appsrc is exactly the same as in : the caps are set to audio/x-raw, and two callbacks are registered, so the element can tell the application when it needs to start and stop pushing data. See for more details.

From this point onwards, playbin takes care of the rest of the pipeline, and the application only needs to worry about generating more data when told so.

To learn how data can be extracted from playbin using the appsink element, see .

Conclusion

This tutorial applies the concepts shown in to playbin. In particular, it has shown:

• How to connect appsrc with playbin using the special URI appsrc://
• How to configure the appsrc using the source-setup signal

It has been a pleasure having you here, and see you soon!