v1

basic-tutorial-3.cpp

@@ -0,0 +1,151 @@
+#include <gst/gst.h>
+
+/* Structure to contain all our information, so we can pass it to callbacks */
+typedef struct _CustomData {
+    GstElement *pipeline;
+    GstElement *source;
+    GstElement *convert;
+    GstElement *resample;
+    GstElement *sink;
+} CustomData;
+
+/* Handler for the pad-added signal */
+static void pad_added_handler (GstElement *src, GstPad *pad, CustomData *data);
+
+int main(int argc, char *argv[]) {
+    CustomData data;
+    GstBus *bus;
+    GstMessage *msg;
+    GstStateChangeReturn ret;
+    gboolean terminate = FALSE;
+
+    /* Initialize GStreamer */
+    gst_init (&argc, &argv);
+
+    /* Create the elements */
+    data.source = gst_element_factory_make ("uridecodebin", "source");
+    data.convert = gst_element_factory_make ("audioconvert", "convert");
+    data.resample = gst_element_factory_make ("audioresample", "resample");
+    data.sink = gst_element_factory_make ("autoaudiosink", "sink");
+
+    /* Create the empty pipeline */
+    data.pipeline = gst_pipeline_new ("test-pipeline");
+
+    if (!data.pipeline || !data.source || !data.convert || !data.resample || !data.sink) {
+        g_printerr ("Not all elements could be created.\n");
+        return -1;
+    }
+
+    /* Build the pipeline. Note that we are NOT linking the source at this
+     * point. We will do it later. */
+    gst_bin_add_many (GST_BIN (data.pipeline), data.source, data.convert, data.resample, data.sink, NULL);
+    if (!gst_element_link_many (data.convert, data.resample, data.sink, NULL)) {
+        g_printerr ("Elements could not be linked.\n");
+        gst_object_unref (data.pipeline);
+        return -1;
+    }
+
+    /* Set the URI to play */
+    g_object_set (data.source, "uri", "https://www.freedesktop.org/software/gstreamer-sdk/data/media/sintel_trailer-480p.webm", NULL);
+
+    /* Connect to the pad-added signal */
+    g_signal_connect (data.source, "pad-added", G_CALLBACK (pad_added_handler), &data);
+
+    /* Start playing */
+    ret = gst_element_set_state (data.pipeline, GST_STATE_PLAYING);
+    if (ret == GST_STATE_CHANGE_FAILURE) {
+        g_printerr ("Unable to set the pipeline to the playing state.\n");
+        gst_object_unref (data.pipeline);
+        return -1;
+    }
+
+    /* Listen to the bus */
+    bus = gst_element_get_bus (data.pipeline);
+    do {
+        msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE,
+                                          static_cast<GstMessageType>(GST_MESSAGE_STATE_CHANGED | GST_MESSAGE_ERROR | GST_MESSAGE_EOS));
+
+        /* Parse message */
+        if (msg != NULL) {
+            GError *err;
+            gchar *debug_info;
+
+            switch (GST_MESSAGE_TYPE (msg)) {
+                case GST_MESSAGE_ERROR:
+                    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);
+                    terminate = TRUE;
+                    break;
+                case GST_MESSAGE_EOS:
+                    g_print ("End-Of-Stream reached.\n");
+                    terminate = TRUE;
+                    break;
+                case GST_MESSAGE_STATE_CHANGED:
+                    /* We are only interested in state-changed messages from the pipeline */
+                    if (GST_MESSAGE_SRC (msg) == GST_OBJECT (data.pipeline)) {
+                        GstState old_state, new_state, pending_state;
+                        gst_message_parse_state_changed (msg, &old_state, &new_state, &pending_state);
+                        g_print ("Pipeline state changed from %s to %s:\n",
+                                 gst_element_state_get_name (old_state), gst_element_state_get_name (new_state));
+                    }
+                    break;
+                default:
+                    /* We should not reach here */
+                    g_printerr ("Unexpected message received.\n");
+                    break;
+            }
+            gst_message_unref (msg);
+        }
+    } while (!terminate);
+
+    /* Free resources */
+    gst_object_unref (bus);
+    gst_element_set_state (data.pipeline, GST_STATE_NULL);
+    gst_object_unref (data.pipeline);
+    return 0;
+}
+
+/* This function will be called by the pad-added signal */
+static void pad_added_handler (GstElement *src, GstPad *new_pad, CustomData *data) {
+    GstPad *sink_pad = gst_element_get_static_pad (data->convert, "sink");
+    GstPadLinkReturn ret;
+    GstCaps *new_pad_caps = NULL;
+    GstStructure *new_pad_struct = NULL;
+    const gchar *new_pad_type = NULL;
+
+    g_print ("Received new pad '%s' from '%s':\n", GST_PAD_NAME (new_pad), GST_ELEMENT_NAME (src));
+
+    /* If our converter is already linked, we have nothing to do here */
+    if (gst_pad_is_linked (sink_pad)) {
+        g_print ("We are already linked. Ignoring.\n");
+        goto exit;
+    }
+
+    /* Check the new pad's type */
+    new_pad_caps = gst_pad_get_current_caps (new_pad);
+    new_pad_struct = gst_caps_get_structure (new_pad_caps, 0);
+    new_pad_type = gst_structure_get_name (new_pad_struct);
+    if (!g_str_has_prefix (new_pad_type, "audio/x-raw")) {
+        g_print ("It has type '%s' which is not raw audio. Ignoring.\n", new_pad_type);
+        goto exit;
+    }
+
+    /* Attempt the link */
+    ret = gst_pad_link (new_pad, sink_pad);
+    if (GST_PAD_LINK_FAILED (ret)) {
+        g_print ("Type is '%s' but link failed.\n", new_pad_type);
+    } else {
+        g_print ("Link succeeded (type '%s').\n", new_pad_type);
+    }
+
+    exit:
+    /* Unreference the new pad's caps, if we got them */
+    if (new_pad_caps != NULL)
+        gst_caps_unref (new_pad_caps);
+
+    /* Unreference the sink pad */
+    gst_object_unref (sink_pad);
+}