1 files changed, 154 insertions, 0 deletions
diff --git a/tutorials/playback-tutorial-3.c b/tutorials/playback-tutorial-3.c
new file mode 100644
index 0000000..4f8b809
--- /dev/null
+++ b/tutorials/playback-tutorial-3.c
@@ -0,0 +1,154 @@
+#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 (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;
+  }
+}
+  
+/* 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 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 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;
+}