[Gst Example] tutorial-8

GST Official Tutorial 8

Signed-off-by: MyungJoo Ham <myungjoo.ham@samsung.com>

diff --git a/gst_example/tutorial7/basic-tutorial-7.c b/gst_example/tutorial7/basic-tutorial-7.c
index 735d636..208bd72 100644 (file)
--- a/gst_example/tutorial7/basic-tutorial-7.c
+++ b/gst_example/tutorial7/basic-tutorial-7.c
@@ -65,11 +65,30 @@ int main(int argc, char *argv[]) {
 
   /* Start playing the pipeline */
   gst_element_set_state (pipeline, GST_STATE_PLAYING);
+  g_print ("Starting Playing\n");
 
   /* Wait until error or EOS */
   bus = gst_element_get_bus (pipeline);
   msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE, GST_MESSAGE_ERROR | GST_MESSAGE_EOS);
 
+  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);
+        break;
+      case GST_MESSAGE_EOS:
+        g_print ("End-Of-Stream reached.\n");
+        break;
+    }
+  }
+
   /* Release the request pads from the Tee, and unref them */
   gst_element_release_request_pad (tee, tee_audio_pad);
   gst_element_release_request_pad (tee, tee_video_pad);

diff --git a/gst_example/tutorial8/basic-tutorial-8.c b/gst_example/tutorial8/basic-tutorial-8.c
new file mode 100644 (file)
index 0000000..ace7bbd
--- /dev/null
+++ b/gst_example/tutorial8/basic-tutorial-8.c
@@ -0,0 +1,233 @@
+#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, *app_source, *tee, *audio_queue, *audio_convert1, *audio_resample, *audio_sink;
+  GstElement *video_queue, *audio_convert2, *visual, *video_convert, *video_sink;
+  GstElement *app_queue, *app_sink;
+
+  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;
+  }
+}
+
+/* The appsink has received a buffer */
+static void new_sample (GstElement *sink, CustomData *data) {
+  GstSample *sample;
+
+  /* Retrieve the buffer */
+  g_signal_emit_by_name (sink, "pull-sample", &sample);
+  if (sample) {
+    /* The only thing we do in this example is print a * to indicate a received buffer */
+    g_print ("*");
+    gst_buffer_unref (sample);
+  }
+}
+
+/* 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);
+}
+
+int main(int argc, char *argv[]) {
+  CustomData data;
+  GstPad *tee_audio_pad, *tee_video_pad, *tee_app_pad;
+  GstPad *queue_audio_pad, *queue_video_pad, *queue_app_pad;
+  GstAudioInfo info;
+  GstCaps *audio_caps;
+  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 elements */
+  data.app_source = gst_element_factory_make ("appsrc", "audio_source");
+  data.tee = gst_element_factory_make ("tee", "tee");
+  data.audio_queue = gst_element_factory_make ("queue", "audio_queue");
+  data.audio_convert1 = gst_element_factory_make ("audioconvert", "audio_convert1");
+  data.audio_resample = gst_element_factory_make ("audioresample", "audio_resample");
+  data.audio_sink = gst_element_factory_make ("autoaudiosink", "audio_sink");
+  data.video_queue = gst_element_factory_make ("queue", "video_queue");
+  data.audio_convert2 = gst_element_factory_make ("audioconvert", "audio_convert2");
+  data.visual = gst_element_factory_make ("wavescope", "visual");
+  data.video_convert = gst_element_factory_make ("videoconvert", "csp");
+  data.video_sink = gst_element_factory_make ("autovideosink", "video_sink");
+  data.app_queue = gst_element_factory_make ("queue", "app_queue");
+  data.app_sink = gst_element_factory_make ("appsink", "app_sink");
+
+  /* Create the empty pipeline */
+  data.pipeline = gst_pipeline_new ("test-pipeline");
+
+  if (!data.pipeline || !data.app_source || !data.tee || !data.audio_queue || !data.audio_convert1 ||
+      !data.audio_resample || !data.audio_sink || !data.video_queue || !data.audio_convert2 || !data.visual ||
+      !data.video_convert || !data.video_sink || !data.app_queue || !data.app_sink) {
+    g_printerr ("Not all elements could be created.\n");
+    return -1;
+  }
+
+  /* Configure wavescope */
+  g_object_set (data.visual, "shader", 0, "style", 0, NULL);
+
+  /* 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 (data.app_source, "caps", audio_caps, "format", GST_FORMAT_TIME, NULL);
+  g_signal_connect (data.app_source, "need-data", G_CALLBACK (start_feed), &data);
+  g_signal_connect (data.app_source, "enough-data", G_CALLBACK (stop_feed), &data);
+
+  /* Configure appsink */
+  g_object_set (data.app_sink, "emit-signals", TRUE, "caps", audio_caps, NULL);
+  g_signal_connect (data.app_sink, "new-sample", G_CALLBACK (new_sample), &data);
+  gst_caps_unref (audio_caps);
+
+  /* Link all elements that can be automatically linked because they have "Always" pads */
+  gst_bin_add_many (GST_BIN (data.pipeline), data.app_source, data.tee, data.audio_queue, data.audio_convert1, data.audio_resample,
+      data.audio_sink, data.video_queue, data.audio_convert2, data.visual, data.video_convert, data.video_sink, data.app_queue,
+      data.app_sink, NULL);
+  if (gst_element_link_many (data.app_source, data.tee, NULL) != TRUE ||
+      gst_element_link_many (data.audio_queue, data.audio_convert1, data.audio_resample, data.audio_sink, NULL) != TRUE ||
+      gst_element_link_many (data.video_queue, data.audio_convert2, data.visual, data.video_convert, data.video_sink, NULL) != TRUE ||
+      gst_element_link_many (data.app_queue, data.app_sink, NULL) != TRUE) {
+    g_printerr ("Elements could not be linked.\n");
+    gst_object_unref (data.pipeline);
+    return -1;
+  }
+
+  /* Manually link the Tee, which has "Request" pads */
+  tee_audio_pad = gst_element_get_request_pad (data.tee, "src_%u");
+  g_print ("Obtained request pad %s for audio branch.\n", gst_pad_get_name (tee_audio_pad));
+  queue_audio_pad = gst_element_get_static_pad (data.audio_queue, "sink");
+  tee_video_pad = gst_element_get_request_pad (data.tee, "src_%u");
+  g_print ("Obtained request pad %s for video branch.\n", gst_pad_get_name (tee_video_pad));
+  queue_video_pad = gst_element_get_static_pad (data.video_queue, "sink");
+  tee_app_pad = gst_element_get_request_pad (data.tee, "src_%u");
+  g_print ("Obtained request pad %s for app branch.\n", gst_pad_get_name (tee_app_pad));
+  queue_app_pad = gst_element_get_static_pad (data.app_queue, "sink");
+  if (gst_pad_link (tee_audio_pad, queue_audio_pad) != GST_PAD_LINK_OK ||
+      gst_pad_link (tee_video_pad, queue_video_pad) != GST_PAD_LINK_OK ||
+      gst_pad_link (tee_app_pad, queue_app_pad) != GST_PAD_LINK_OK) {
+    g_printerr ("Tee could not be linked\n");
+    gst_object_unref (data.pipeline);
+    return -1;
+  }
+  gst_object_unref (queue_audio_pad);
+  gst_object_unref (queue_video_pad);
+  gst_object_unref (queue_app_pad);
+
+  /* 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);
+
+  /* Release the request pads from the Tee, and unref them */
+  gst_element_release_request_pad (data.tee, tee_audio_pad);
+  gst_element_release_request_pad (data.tee, tee_video_pad);
+  gst_element_release_request_pad (data.tee, tee_app_pad);
+  gst_object_unref (tee_audio_pad);
+  gst_object_unref (tee_video_pad);
+  gst_object_unref (tee_app_pad);
+
+  /* Free resources */
+  gst_element_set_state (data.pipeline, GST_STATE_NULL);
+  gst_object_unref (data.pipeline);
+  return 0;
+}
+