Port to gst_print* family

g_print* would print broken string on Windows
See also https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/258

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/20>

8 files changed, 163 insertions, 162 deletions

diff --git a/network/http-launch/http-launch.c b/network/http-launch/http-launch.c
index 5eedf00..ac79548 100644
--- a/network/http-launch/http-launch.c
+++ b/network/http-launch/http-launch.c
@@ -56,7 +56,7 @@ static gboolean caps_resolved = FALSE;
 static void
 remove_client (Client * client)
 {
-  g_print ("Removing connection %s\n", client->name);
+  gst_print ("Removing connection %s\n", client->name);
 
   G_LOCK (clients);
   clients = g_list_remove (clients, client);
@@ -96,7 +96,7 @@ write_bytes (Client * client, const gchar * data, guint len)
 
   if (w <= 0) {
     if (err) {
-      g_print ("Write error %s\n", err->message);
+      gst_print ("Write error %s\n", err->message);
       g_clear_error (&err);
     }
     remove_client (client);
@@ -167,15 +167,15 @@ client_message (Client * client, const gchar * data, guint len)
         g_source_destroy (client->tosource);
         g_source_unref (client->tosource);
         client->tosource = NULL;
-        g_print ("Starting to stream to %s\n", client->name);
+        gst_print ("Starting to stream to %s\n", client->name);
         g_signal_emit_by_name (multisocketsink, "add", client->socket);
       }
 
       if (!started) {
-        g_print ("Starting pipeline\n");
+        gst_print ("Starting pipeline\n");
         if (gst_element_set_state (pipeline,
                 GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE) {
-          g_print ("Failed to start pipeline\n");
+          gst_print ("Failed to start pipeline\n");
           g_main_loop_quit (loop);
         }
         started = TRUE;
@@ -204,7 +204,7 @@ client_message (Client * client, const gchar * data, guint len)
 static gboolean
 on_timeout (Client * client)
 {
-  g_print ("Timeout\n");
+  gst_print ("Timeout\n");
   remove_client (client);
 
   return FALSE;
@@ -250,14 +250,14 @@ on_read_bytes (GPollableInputStream * stream, Client * client)
     }
 
     if (client->current_message->len >= 1024 * 1024) {
-      g_print ("No complete request after 1MB of data\n");
+      gst_print ("No complete request after 1MB of data\n");
       remove_client (client);
       return FALSE;
     }
 
     return TRUE;
   } else {
-    g_print ("Read error %s\n", err->message);
+    gst_print ("Read error %s\n", err->message);
     g_clear_error (&err);
     remove_client (client);
     return FALSE;
@@ -284,7 +284,7 @@ on_new_connection (GSocketService * service, GSocketConnection * connection,
   g_free (ip);
   g_object_unref (addr);
 
-  g_print ("New connection %s\n", client->name);
+  gst_print ("New connection %s\n", client->name);
 
   client->waiting_200_ok = FALSE;
   client->http_version = g_strdup ("");
@@ -324,7 +324,7 @@ on_message (GstBus * bus, GstMessage * message, gpointer user_data)
       GError *err;
 
       gst_message_parse_error (message, &err, &debug);
-      g_print ("Error %s\n", err->message);
+      gst_print ("Error %s\n", err->message);
       g_error_free (err);
       g_free (debug);
       g_main_loop_quit (loop);
@@ -335,13 +335,13 @@ on_message (GstBus * bus, GstMessage * message, gpointer user_data)
       GError *err;
 
       gst_message_parse_warning (message, &err, &debug);
-      g_print ("Warning %s\n", err->message);
+      gst_print ("Warning %s\n", err->message);
       g_error_free (err);
       g_free (debug);
       break;
     }
     case GST_MESSAGE_EOS:{
-      g_print ("EOS\n");
+      gst_print ("EOS\n");
       g_main_loop_quit (loop);
     }
     default:
@@ -404,7 +404,7 @@ on_stream_caps_changed (GObject * obj, GParamSpec * pspec, gpointer user_data)
       } else {
         content_type = g_strdup_printf ("Content-Type: %s\r\n", mimetype);
       }
-      g_print ("%s", content_type);
+      gst_print ("%s", content_type);
       break;
     }
     i++;
@@ -443,7 +443,7 @@ main (gint argc, gchar ** argv)
   gst_init (&argc, &argv);
 
   if (argc < 4) {
-    g_print ("usage: %s PORT <launch line>\n"
+    gst_print ("usage: %s PORT <launch line>\n"
         "example: %s 8080 ( videotestsrc ! theoraenc ! oggmux name=stream )\n",
         argv[0], argv[0]);
     return -1;
@@ -454,21 +454,21 @@ main (gint argc, gchar ** argv)
 
   bin = gst_parse_launchv ((const gchar **) argv + 2, &err);
   if (!bin) {
-    g_print ("invalid pipeline: %s\n", err->message);
+    gst_print ("invalid pipeline: %s\n", err->message);
     g_clear_error (&err);
     return -2;
   }
 
   stream = gst_bin_get_by_name (GST_BIN (bin), "stream");
   if (!stream) {
-    g_print ("no element with name \"stream\" found\n");
+    gst_print ("no element with name \"stream\" found\n");
     gst_object_unref (bin);
     return -3;
   }
 
   srcpad = gst_element_get_static_pad (stream, "src");
   if (!srcpad) {
-    g_print ("no \"src\" pad in element \"stream\" found\n");
+    gst_print ("no \"src\" pad in element \"stream\" found\n");
     gst_object_unref (stream);
     gst_object_unref (bin);
     return -4;
@@ -514,7 +514,7 @@ main (gint argc, gchar ** argv)
           GST_STATE_READY) == GST_STATE_CHANGE_FAILURE) {
     gst_object_unref (pipeline);
     g_main_loop_unref (loop);
-    g_print ("Failed to set pipeline to ready\n");
+    gst_print ("Failed to set pipeline to ready\n");
     return -5;
   }
 
@@ -526,7 +526,7 @@ main (gint argc, gchar ** argv)
 
   g_socket_service_start (service);
 
-  g_print ("Listening on http://127.0.0.1:%d/\n", port);
+  gst_print ("Listening on http://127.0.0.1:%d/\n", port);
 
   g_main_loop_run (loop);
 
diff --git a/playback/player/gst-play/gst-play.c b/playback/player/gst-play/gst-play.c
index d4c9e67..b7b98e1 100644
--- a/playback/player/gst-play/gst-play.c
+++ b/playback/player/gst-play/gst-play.c
@@ -59,10 +59,10 @@ static void play_set_relative_volume (GstPlay * play, gdouble volume_step);
 static void
 end_of_stream_cb (GstPlayer * player, GstPlay * play)
 {
-  g_print ("\n");
+  gst_print ("\n");
   /* and switch to next item in list */
   if (!play_next (play)) {
-    g_print ("Reached end of play list.\n");
+    gst_print ("Reached end of play list.\n");
     g_main_loop_quit (play->loop);
   }
 }
@@ -70,14 +70,14 @@ end_of_stream_cb (GstPlayer * player, GstPlay * play)
 static void
 error_cb (GstPlayer * player, GError * err, GstPlay * play)
 {
-  g_printerr ("ERROR %s for %s\n", err->message, play->uris[play->cur_idx]);
+  gst_printerr ("ERROR %s for %s\n", err->message, play->uris[play->cur_idx]);
 
   /* if looping is enabled, then disable it else will keep looping forever */
   play->repeat = FALSE;
 
   /* try next item in list then */
   if (!play_next (play)) {
-    g_print ("Reached end of play list.\n");
+    gst_print ("Reached end of play list.\n");
     g_main_loop_quit (play->loop);
   }
 }
@@ -100,20 +100,20 @@ position_updated_cb (GstPlayer * player, GstClockTime pos, GstPlay * play)
     pstr[9] = '\0';
     g_snprintf (dstr, 32, "%" GST_TIME_FORMAT, GST_TIME_ARGS (dur));
     dstr[9] = '\0';
-    g_print ("%s / %s %s\r", pstr, dstr, status);
+    gst_print ("%s / %s %s\r", pstr, dstr, status);
   }
 }
 
 static void
 state_changed_cb (GstPlayer * player, GstPlayerState state, GstPlay * play)
 {
-  g_print ("State changed: %s\n", gst_player_state_get_name (state));
+  gst_print ("State changed: %s\n", gst_player_state_get_name (state));
 }
 
 static void
 buffering_cb (GstPlayer * player, gint percent, GstPlay * play)
 {
-  g_print ("Buffering: %d\n", percent);
+  gst_print ("Buffering: %d\n", percent);
 }
 
 static void
@@ -127,22 +127,22 @@ print_one_tag (const GstTagList * list, const gchar * tag, gpointer user_data)
 
     val = gst_tag_list_get_value_index (list, tag, i);
     if (G_VALUE_HOLDS_STRING (val)) {
-      g_print ("    %s : %s \n", tag, g_value_get_string (val));
+      gst_print ("    %s : %s \n", tag, g_value_get_string (val));
     } else if (G_VALUE_HOLDS_UINT (val)) {
-      g_print ("    %s : %u \n", tag, g_value_get_uint (val));
+      gst_print ("    %s : %u \n", tag, g_value_get_uint (val));
     } else if (G_VALUE_HOLDS_DOUBLE (val)) {
-      g_print ("    %s : %g \n", tag, g_value_get_double (val));
+      gst_print ("    %s : %g \n", tag, g_value_get_double (val));
     } else if (G_VALUE_HOLDS_BOOLEAN (val)) {
-      g_print ("    %s : %s \n", tag,
+      gst_print ("    %s : %s \n", tag,
           g_value_get_boolean (val) ? "true" : "false");
     } else if (GST_VALUE_HOLDS_DATE_TIME (val)) {
       GstDateTime *dt = g_value_get_boxed (val);
       gchar *dt_str = gst_date_time_to_iso8601_string (dt);
 
-      g_print ("    %s : %s \n", tag, dt_str);
+      gst_print ("    %s : %s \n", tag, dt_str);
       g_free (dt_str);
     } else {
-      g_print ("    %s : tag of type '%s' \n", tag, G_VALUE_TYPE_NAME (val));
+      gst_print ("    %s : tag of type '%s' \n", tag, G_VALUE_TYPE_NAME (val));
     }
   }
 }
@@ -156,15 +156,15 @@ print_video_info (GstPlayerVideoInfo * info)
   if (info == NULL)
     return;
 
-  g_print ("  width : %d\n", gst_player_video_info_get_width (info));
-  g_print ("  height : %d\n", gst_player_video_info_get_height (info));
-  g_print ("  max_bitrate : %d\n",
+  gst_print ("  width : %d\n", gst_player_video_info_get_width (info));
+  gst_print ("  height : %d\n", gst_player_video_info_get_height (info));
+  gst_print ("  max_bitrate : %d\n",
       gst_player_video_info_get_max_bitrate (info));
-  g_print ("  bitrate : %d\n", gst_player_video_info_get_bitrate (info));
+  gst_print ("  bitrate : %d\n", gst_player_video_info_get_bitrate (info));
   gst_player_video_info_get_framerate (info, &fps_n, &fps_d);
-  g_print ("  framerate : %.2f\n", (gdouble) fps_n / fps_d);
+  gst_print ("  framerate : %.2f\n", (gdouble) fps_n / fps_d);
   gst_player_video_info_get_pixel_aspect_ratio (info, &par_n, &par_d);
-  g_print ("  pixel-aspect-ratio  %u:%u\n", par_n, par_d);
+  gst_print ("  pixel-aspect-ratio  %u:%u\n", par_n, par_d);
 }
 
 static void
@@ -173,13 +173,13 @@ print_audio_info (GstPlayerAudioInfo * info)
   if (info == NULL)
     return;
 
-  g_print ("  sample rate : %d\n",
+  gst_print ("  sample rate : %d\n",
       gst_player_audio_info_get_sample_rate (info));
-  g_print ("  channels : %d\n", gst_player_audio_info_get_channels (info));
-  g_print ("  max_bitrate : %d\n",
+  gst_print ("  channels : %d\n", gst_player_audio_info_get_channels (info));
+  gst_print ("  max_bitrate : %d\n",
       gst_player_audio_info_get_max_bitrate (info));
-  g_print ("  bitrate : %d\n", gst_player_audio_info_get_bitrate (info));
-  g_print ("  language : %s\n", gst_player_audio_info_get_language (info));
+  gst_print ("  bitrate : %d\n", gst_player_audio_info_get_bitrate (info));
+  gst_print ("  language : %s\n", gst_player_audio_info_get_language (info));
 }
 
 static void
@@ -188,7 +188,7 @@ print_subtitle_info (GstPlayerSubtitleInfo * info)
   if (info == NULL)
     return;
 
-  g_print ("  language : %s\n", gst_player_subtitle_info_get_language (info));
+  gst_print ("  language : %s\n", gst_player_subtitle_info_get_language (info));
 }
 
 static void
@@ -197,31 +197,31 @@ print_all_stream_info (GstPlayerMediaInfo * media_info)
   guint count = 0;
   GList *list, *l;
 
-  g_print ("URI : %s\n", gst_player_media_info_get_uri (media_info));
-  g_print ("Duration: %" GST_TIME_FORMAT "\n",
+  gst_print ("URI : %s\n", gst_player_media_info_get_uri (media_info));
+  gst_print ("Duration: %" GST_TIME_FORMAT "\n",
       GST_TIME_ARGS (gst_player_media_info_get_duration (media_info)));
-  g_print ("Global taglist:\n");
+  gst_print ("Global taglist:\n");
   if (gst_player_media_info_get_tags (media_info))
     gst_tag_list_foreach (gst_player_media_info_get_tags (media_info),
         print_one_tag, NULL);
   else
-    g_print ("  (nil) \n");
+    gst_print ("  (nil) \n");
 
   list = gst_player_media_info_get_stream_list (media_info);
   if (!list)
     return;
 
-  g_print ("All Stream information\n");
+  gst_print ("All Stream information\n");
   for (l = list; l != NULL; l = l->next) {
     GstTagList *tags = NULL;
     GstPlayerStreamInfo *stream = (GstPlayerStreamInfo *) l->data;
 
-    g_print (" Stream # %u \n", count++);
-    g_print ("  type : %s_%u\n",
+    gst_print (" Stream # %u \n", count++);
+    gst_print ("  type : %s_%u\n",
         gst_player_stream_info_get_stream_type (stream),
         gst_player_stream_info_get_index (stream));
     tags = gst_player_stream_info_get_tags (stream);
-    g_print ("  taglist : \n");
+    gst_print ("  taglist : \n");
     if (tags) {
       gst_tag_list_foreach (tags, print_one_tag, NULL);
     }
@@ -244,11 +244,11 @@ print_all_video_stream (GstPlayerMediaInfo * media_info)
   if (!list)
     return;
 
-  g_print ("All video streams\n");
+  gst_print ("All video streams\n");
   for (l = list; l != NULL; l = l->next) {
     GstPlayerVideoInfo *info = (GstPlayerVideoInfo *) l->data;
     GstPlayerStreamInfo *sinfo = (GstPlayerStreamInfo *) info;
-    g_print (" %s_%d #\n", gst_player_stream_info_get_stream_type (sinfo),
+    gst_print (" %s_%d #\n", gst_player_stream_info_get_stream_type (sinfo),
         gst_player_stream_info_get_index (sinfo));
     print_video_info (info);
   }
@@ -263,11 +263,11 @@ print_all_subtitle_stream (GstPlayerMediaInfo * media_info)
   if (!list)
     return;
 
-  g_print ("All subtitle streams:\n");
+  gst_print ("All subtitle streams:\n");
   for (l = list; l != NULL; l = l->next) {
     GstPlayerSubtitleInfo *info = (GstPlayerSubtitleInfo *) l->data;
     GstPlayerStreamInfo *sinfo = (GstPlayerStreamInfo *) info;
-    g_print (" %s_%d #\n", gst_player_stream_info_get_stream_type (sinfo),
+    gst_print (" %s_%d #\n", gst_player_stream_info_get_stream_type (sinfo),
         gst_player_stream_info_get_index (sinfo));
     print_subtitle_info (info);
   }
@@ -282,11 +282,11 @@ print_all_audio_stream (GstPlayerMediaInfo * media_info)
   if (!list)
     return;
 
-  g_print ("All audio streams: \n");
+  gst_print ("All audio streams: \n");
   for (l = list; l != NULL; l = l->next) {
     GstPlayerAudioInfo *info = (GstPlayerAudioInfo *) l->data;
     GstPlayerStreamInfo *sinfo = (GstPlayerStreamInfo *) info;
-    g_print (" %s_%d #\n", gst_player_stream_info_get_stream_type (sinfo),
+    gst_print (" %s_%d #\n", gst_player_stream_info_get_stream_type (sinfo),
         gst_player_stream_info_get_index (sinfo));
     print_audio_info (info);
   }
@@ -299,15 +299,15 @@ print_current_tracks (GstPlay * play)
   GstPlayerVideoInfo *video = NULL;
   GstPlayerSubtitleInfo *subtitle = NULL;
 
-  g_print ("Current video track: \n");
+  gst_print ("Current video track: \n");
   video = gst_player_get_current_video_track (play->player);
   print_video_info (video);
 
-  g_print ("Current audio track: \n");
+  gst_print ("Current audio track: \n");
   audio = gst_player_get_current_audio_track (play->player);
   print_audio_info (audio);
 
-  g_print ("Current subtitle track: \n");
+  gst_print ("Current subtitle track: \n");
   subtitle = gst_player_get_current_subtitle_track (play->player);
   print_subtitle_info (subtitle);
 
@@ -325,11 +325,11 @@ static void
 print_media_info (GstPlayerMediaInfo * media_info)
 {
   print_all_stream_info (media_info);
-  g_print ("\n");
+  gst_print ("\n");
   print_all_video_stream (media_info);
-  g_print ("\n");
+  gst_print ("\n");
   print_all_audio_stream (media_info);
-  g_print ("\n");
+  gst_print ("\n");
   print_all_subtitle_stream (media_info);
 }
 
@@ -410,7 +410,7 @@ play_set_relative_volume (GstPlay * play, gdouble volume_step)
 
   g_object_set (play->player, "volume", volume, NULL);
 
-  g_print ("Volume: %.0f%%                  \n", volume * 100);
+  gst_print ("Volume: %.0f%%                  \n", volume * 100);
 }
 
 static gchar *
@@ -438,7 +438,7 @@ play_uri (GstPlay * play, const gchar * next_uri)
   play_reset (play);
 
   loc = play_uri_get_display_name (play, next_uri);
-  g_print ("Now playing %s\n", loc);
+  gst_print ("Now playing %s\n", loc);
   g_free (loc);
 
   g_object_set (play->player, "uri", next_uri, NULL);
@@ -451,7 +451,7 @@ play_next (GstPlay * play)
 {
   if ((play->cur_idx + 1) >= play->num_uris) {
     if (play->repeat) {
-      g_print ("Looping playlist \n");
+      gst_print ("Looping playlist \n");
       play->cur_idx = -1;
     } else
       return FALSE;
@@ -567,7 +567,7 @@ relative_seek (GstPlay * play, gdouble percent)
   g_object_get (play->player, "position", &pos, "duration", &dur, NULL);
 
   if (dur <= 0) {
-    g_print ("\nCould not seek.\n");
+    gst_print ("\nCould not seek.\n");
     return;
   }
 
@@ -602,7 +602,7 @@ keyboard_cb (const gchar * key_input, gpointer user_data)
       break;
     case '>':
       if (!play_next (play)) {
-        g_print ("\nReached end of play list.\n");
+        gst_print ("\nReached end of play list.\n");
         g_main_loop_quit (play->loop);
       }
       break;
@@ -671,7 +671,7 @@ main (int argc, char **argv)
   g_option_context_add_main_entries (ctx, options, NULL);
   g_option_context_add_group (ctx, gst_init_get_option_group ());
   if (!g_option_context_parse (ctx, &argc, &argv, &err)) {
-    g_print ("Error initializing: %s\n", GST_STR_NULL (err->message));
+    gst_print ("Error initializing: %s\n", GST_STR_NULL (err->message));
     g_clear_error (&err);
     g_option_context_free (ctx);
     return 1;
@@ -684,8 +684,8 @@ main (int argc, char **argv)
     gchar *version_str;
 
     version_str = gst_version_string ();
-    g_print ("%s version %s\n", g_get_prgname (), "1.0");
-    g_print ("%s\n", version_str);
+    gst_print ("%s version %s\n", g_get_prgname (), "1.0");
+    gst_print ("%s\n", version_str);
     g_free (version_str);
 
     g_free (playlist_file);
@@ -712,7 +712,7 @@ main (int argc, char **argv)
       g_strfreev (lines);
       g_free (playlist_contents);
     } else {
-      g_printerr ("Could not read playlist: %s\n", err->message);
+      gst_printerr ("Could not read playlist: %s\n", err->message);
       g_clear_error (&err);
     }
     g_free (playlist_file);
@@ -720,10 +720,10 @@ main (int argc, char **argv)
   }
 
   if (playlist->len == 0 && (filenames == NULL || *filenames == NULL)) {
-    g_printerr ("Usage: %s FILE1|URI1 [FILE2|URI2] [FILE3|URI3] ...",
+    gst_printerr ("Usage: %s FILE1|URI1 [FILE2|URI2] [FILE3|URI3] ...",
         "gst-play");
-    g_printerr ("\n\n"),
-        g_printerr ("%s\n\n",
+    gst_printerr ("\n\n"),
+        gst_printerr ("%s\n\n",
         "You must provide at least one filename or URI to play.");
     /* No input provided. Free array */
     g_ptr_array_free (playlist, TRUE);
@@ -757,7 +757,7 @@ main (int argc, char **argv)
     if (gst_play_kb_set_key_handler (keyboard_cb, play)) {
       atexit (restore_terminal);
     } else {
-      g_print ("Interactive keyboard handling in terminal not available.\n");
+      gst_print ("Interactive keyboard handling in terminal not available.\n");
     }
   }
 
@@ -767,7 +767,7 @@ main (int argc, char **argv)
   /* clean up */
   play_free (play);
 
-  g_print ("\n");
+  gst_print ("\n");
   gst_deinit ();
   return 0;
 }
diff --git a/playback/player/gtk/gtk-play.c b/playback/player/gtk/gtk-play.c
index 3cea7e6..786458e 100644
--- a/playback/player/gtk/gtk-play.c
+++ b/playback/player/gtk/gtk-play.c
@@ -162,7 +162,7 @@ load_from_builder (const gchar * filename, gboolean register_sig_handler,
 
   builder = gtk_builder_new_from_resource (filename);
   if (builder == NULL) {
-    g_print ("ERROR: failed to load %s \n", filename);
+    gst_print ("ERROR: failed to load %s \n", filename);
     return NULL;
   }
 
@@ -1358,7 +1358,7 @@ gtk_widget_apply_css (GtkWidget * widget, const gchar * filename)
   provider = GTK_STYLE_PROVIDER (gtk_css_provider_new ());
   bytes = g_resources_lookup_data (filename, 0, &err);
   if (err) {
-    g_print ("ERROR: failed to apply css %s '%s' \n", filename, err->message);
+    gst_print ("ERROR: failed to apply css %s '%s' \n", filename, err->message);
     return;
   }
   data = g_bytes_get_data (bytes, &data_size);
@@ -1612,12 +1612,12 @@ gtk_play_get_cover_image (GstPlayerMediaInfo * media_info)
   if ((type != GST_TAG_IMAGE_TYPE_FRONT_COVER) &&
       (type != GST_TAG_IMAGE_TYPE_UNDEFINED) &&
       (type != GST_TAG_IMAGE_TYPE_NONE)) {
-    g_print ("unsupport type ... %d \n", type);
+    gst_print ("unsupport type ... %d \n", type);
     return NULL;
   }
 
   if (!gst_buffer_map (buffer, &info, GST_MAP_READ)) {
-    g_print ("failed to map gst buffer \n");
+    gst_print ("failed to map gst buffer \n");
     return NULL;
   }
 
@@ -1628,7 +1628,7 @@ gtk_play_get_cover_image (GstPlayerMediaInfo * media_info)
     if (pixbuf) {
       g_object_ref (pixbuf);
     } else {
-      g_print ("failed to convert gst buffer to pixbuf %s \n", err->message);
+      gst_print ("failed to convert gst buffer to pixbuf %s \n", err->message);
       g_error_free (err);
     }
   }
diff --git a/playback/player/qt/extension/qgstplayer.cpp b/playback/player/qt/extension/qgstplayer.cpp
index 6667e8d..4ea16aa 100644
--- a/playback/player/qt/extension/qgstplayer.cpp
+++ b/playback/player/qt/extension/qgstplayer.cpp
@@ -184,12 +184,12 @@ void MediaInfo::update(GstPlayerMediaInfo *info)
     if ((type != GST_TAG_IMAGE_TYPE_FRONT_COVER) &&
         (type != GST_TAG_IMAGE_TYPE_UNDEFINED) &&
         (type != GST_TAG_IMAGE_TYPE_NONE)) {
-      g_print ("unsupport type ... %d \n", type);
+      gst_print ("unsupport type ... %d \n", type);
       return;
     }
 
     if (!gst_buffer_map (buffer, &map_info, GST_MAP_READ)) {
-      g_print ("failed to map gst buffer \n");
+      gst_print ("failed to map gst buffer \n");
       return;
     }
 
diff --git a/webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c b/webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c
index 1c5b0ec..5f8076e 100644
--- a/webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c
+++ b/webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c
@@ -78,7 +78,7 @@ static gboolean
 cleanup_and_quit_loop (const gchar * msg, enum AppState state)
 {
   if (msg)
-    g_printerr ("%s\n", msg);
+    gst_printerr ("%s\n", msg);
   if (state > 0)
     app_state = state;
 
@@ -153,7 +153,7 @@ on_incoming_decodebin_stream (GstElement * decodebin, GstPad * pad,
   const gchar *name;
 
   if (!gst_pad_has_current_caps (pad)) {
-    g_printerr ("Pad '%s' has no caps, can't do anything, ignoring\n",
+    gst_printerr ("Pad '%s' has no caps, can't do anything, ignoring\n",
         GST_PAD_NAME (pad));
     return;
   }
@@ -166,7 +166,7 @@ on_incoming_decodebin_stream (GstElement * decodebin, GstPad * pad,
   } else if (g_str_has_prefix (name, "audio")) {
     handle_media_stream (pad, pipe, "audioconvert", "autoaudiosink");
   } else {
-    g_printerr ("Unknown pad %s, ignoring", GST_PAD_NAME (pad));
+    gst_printerr ("Unknown pad %s, ignoring", GST_PAD_NAME (pad));
   }
 }
 
@@ -240,7 +240,7 @@ send_room_peer_sdp (GstWebRTCSessionDescription * desc, const gchar * peer_id)
     g_assert_not_reached ();
 
   text = gst_sdp_message_as_text (desc->sdp);
-  g_print ("Sending sdp %s to %s:\n%s\n", sdptype, peer_id, text);
+  gst_print ("Sending sdp %s to %s:\n%s\n", sdptype, peer_id, text);
 
   sdp = json_object_new ();
   json_object_set_string_member (sdp, "type", sdptype);
@@ -420,12 +420,12 @@ start_pipeline (void)
       "queue ! " RTP_CAPS_OPUS (96) " ! audiotee. ", &error);
 
   if (error) {
-    g_printerr ("Failed to parse launch: %s\n", error->message);
+    gst_printerr ("Failed to parse launch: %s\n", error->message);
     g_error_free (error);
     goto err;
   }
 
-  g_print ("Starting pipeline, not transmitting yet\n");
+  gst_print ("Starting pipeline, not transmitting yet\n");
   ret = gst_element_set_state (GST_ELEMENT (pipeline), GST_STATE_PLAYING);
   if (ret == GST_STATE_CHANGE_FAILURE)
     goto err;
@@ -433,7 +433,7 @@ start_pipeline (void)
   return TRUE;
 
 err:
-  g_print ("State change failure\n");
+  gst_print ("State change failure\n");
   if (pipeline)
     g_clear_object (&pipeline);
   return FALSE;
@@ -451,7 +451,7 @@ join_room_on_server (void)
   if (!room_id)
     return FALSE;
 
-  g_print ("Joining room %s\n", room_id);
+  gst_print ("Joining room %s\n", room_id);
   app_state = ROOM_JOINING;
   msg = g_strdup_printf ("ROOM %s", room_id);
   soup_websocket_connection_send_text (ws_conn, msg);
@@ -468,7 +468,7 @@ register_with_server (void)
       SOUP_WEBSOCKET_STATE_OPEN)
     return FALSE;
 
-  g_print ("Registering id %s with server\n", local_id);
+  gst_print ("Registering id %s with server\n", local_id);
   app_state = SERVER_REGISTERING;
 
   /* Register with the server with a random integer id. Reply will be received
@@ -497,7 +497,7 @@ do_registration (void)
     return FALSE;
   }
   app_state = SERVER_REGISTERED;
-  g_print ("Registered with server\n");
+  gst_print ("Registered with server\n");
   /* Ask signalling server that we want to join a room */
   if (!join_room_on_server ()) {
     cleanup_and_quit_loop ("ERROR: Failed to join room", ROOM_CALL_ERROR);
@@ -523,7 +523,7 @@ do_join_room (const gchar * text)
   }
 
   app_state = ROOM_JOINED;
-  g_print ("Room joined\n");
+  gst_print ("Room joined\n");
   /* Start recording, but not transmitting */
   if (!start_pipeline ()) {
     cleanup_and_quit_loop ("ERROR: Failed to start pipeline", ROOM_CALL_ERROR);
@@ -536,11 +536,11 @@ do_join_room (const gchar * text)
   /* There are peers in the room already. We need to start negotiation
    * (exchange SDP and ICE candidates) and transmission of media. */
   if (len > 1 && strlen (peer_ids[1]) > 0) {
-    g_print ("Found %i peers already in room\n", len - 1);
+    gst_print ("Found %i peers already in room\n", len - 1);
     app_state = ROOM_CALL_OFFERING;
     for (ii = 1; ii < len; ii++) {
       gchar *peer_id = g_strdup (peer_ids[ii]);
-      g_print ("Negotiating with peer %s\n", peer_id);
+      gst_print ("Negotiating with peer %s\n", peer_id);
       /* This might fail asynchronously */
       call_peer (peer_id);
       peers = g_list_prepend (peers, peer_id);
@@ -621,7 +621,7 @@ handle_sdp_offer (const gchar * peer_id, const gchar * text)
 
   g_assert_cmpint (app_state, ==, ROOM_CALL_ANSWERING);
 
-  g_print ("Received offer:\n%s\n", text);
+  gst_print ("Received offer:\n%s\n", text);
 
   ret = gst_sdp_message_new (&sdp);
   g_assert_cmpint (ret, ==, GST_SDP_OK);
@@ -661,7 +661,7 @@ handle_sdp_answer (const gchar * peer_id, const gchar * text)
 
   g_assert_cmpint (app_state, >=, ROOM_CALL_OFFERING);
 
-  g_print ("Received answer:\n%s\n", text);
+  gst_print ("Received answer:\n%s\n", text);
 
   ret = gst_sdp_message_new (&sdp);
   g_assert_cmpint (ret, ==, GST_SDP_OK);
@@ -690,19 +690,20 @@ handle_peer_message (const gchar * peer_id, const gchar * msg)
   JsonObject *object, *child;
   JsonParser *parser = json_parser_new ();
   if (!json_parser_load_from_data (parser, msg, -1, NULL)) {
-    g_printerr ("Unknown message '%s' from '%s', ignoring", msg, peer_id);
+    gst_printerr ("Unknown message '%s' from '%s', ignoring", msg, peer_id);
     g_object_unref (parser);
     return FALSE;
   }
 
   root = json_parser_get_root (parser);
   if (!JSON_NODE_HOLDS_OBJECT (root)) {
-    g_printerr ("Unknown json message '%s' from '%s', ignoring", msg, peer_id);
+    gst_printerr ("Unknown json message '%s' from '%s', ignoring", msg,
+        peer_id);
     g_object_unref (parser);
     return FALSE;
   }
 
-  g_print ("Message from peer %s: %s\n", peer_id, msg);
+  gst_print ("Message from peer %s: %s\n", peer_id, msg);
 
   object = json_node_get_object (root);
   /* Check type of JSON message */
@@ -750,7 +751,7 @@ handle_peer_message (const gchar * peer_id, const gchar * msg)
         candidate);
     gst_object_unref (webrtc);
   } else {
-    g_printerr ("Ignoring unknown JSON message:\n%s\n", msg);
+    gst_printerr ("Ignoring unknown JSON message:\n%s\n", msg);
   }
   g_object_unref (parser);
   return TRUE;
@@ -765,7 +766,7 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
 
   switch (type) {
     case SOUP_WEBSOCKET_DATA_BINARY:
-      g_printerr ("Received unknown binary message, ignoring\n");
+      gst_printerr ("Received unknown binary message, ignoring\n");
       return;
     case SOUP_WEBSOCKET_DATA_TEXT:{
       gsize size;
@@ -803,18 +804,18 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
         peers = g_list_prepend (peers, g_strdup (splitm[1]));
         peer_id = find_peer_from_list (splitm[1]);
         g_assert_nonnull (peer_id);
-        g_print ("Peer %s has joined the room\n", peer_id);
+        gst_print ("Peer %s has joined the room\n", peer_id);
       } else if (g_str_has_prefix (text, "ROOM_PEER_LEFT")) {
         splitm = g_strsplit (text, " ", 2);
         peer_id = find_peer_from_list (splitm[1]);
         g_assert_nonnull (peer_id);
         peers = g_list_remove (peers, peer_id);
-        g_print ("Peer %s has left the room\n", peer_id);
+        gst_print ("Peer %s has left the room\n", peer_id);
         remove_peer_from_pipeline (peer_id);
         g_free ((gchar *) peer_id);
         /* TODO: cleanup pipeline */
       } else {
-        g_printerr ("WARNING: Ignoring unknown message %s\n", text);
+        gst_printerr ("WARNING: Ignoring unknown message %s\n", text);
       }
       g_strfreev (splitm);
     } else {
@@ -856,7 +857,7 @@ on_server_connected (SoupSession * session, GAsyncResult * res,
   g_assert_nonnull (ws_conn);
 
   app_state = SERVER_CONNECTED;
-  g_print ("Connected to signalling server\n");
+  gst_print ("Connected to signalling server\n");
 
   g_signal_connect (ws_conn, "closed", G_CALLBACK (on_server_closed), NULL);
   g_signal_connect (ws_conn, "message", G_CALLBACK (on_server_message), NULL);
@@ -888,7 +889,7 @@ connect_to_websocket_server_async (void)
 
   message = soup_message_new (SOUP_METHOD_GET, server_url);
 
-  g_print ("Connecting to server...\n");
+  gst_print ("Connecting to server...\n");
 
   /* Once connected, we will register */
   soup_session_websocket_connect_async (session, message, NULL, NULL, NULL,
@@ -912,7 +913,7 @@ check_plugins (void)
     GstPlugin *plugin;
     plugin = gst_registry_find_plugin (registry, needed[i]);
     if (!plugin) {
-      g_print ("Required gstreamer plugin '%s' not found\n", needed[i]);
+      gst_print ("Required gstreamer plugin '%s' not found\n", needed[i]);
       ret = FALSE;
       continue;
     }
@@ -931,7 +932,7 @@ main (int argc, char *argv[])
   g_option_context_add_main_entries (context, entries, NULL);
   g_option_context_add_group (context, gst_init_get_option_group ());
   if (!g_option_context_parse (context, &argc, &argv, &error)) {
-    g_printerr ("Error initializing: %s\n", error->message);
+    gst_printerr ("Error initializing: %s\n", error->message);
     return -1;
   }
 
@@ -939,7 +940,7 @@ main (int argc, char *argv[])
     return -1;
 
   if (!room_id) {
-    g_printerr ("--room-id is a required argument\n");
+    gst_printerr ("--room-id is a required argument\n");
     return -1;
   }
 
@@ -949,7 +950,7 @@ main (int argc, char *argv[])
   /* Sanitize by removing whitespace, modifies string in-place */
   g_strdelimit (local_id, " \t\n\r", '-');
 
-  g_print ("Our local id is %s\n", local_id);
+  gst_print ("Our local id is %s\n", local_id);
 
   if (!server_url)
     server_url = g_strdup (default_server_url);
@@ -971,7 +972,7 @@ main (int argc, char *argv[])
   g_main_loop_run (loop);
 
   gst_element_set_state (GST_ELEMENT (pipeline), GST_STATE_NULL);
-  g_print ("Pipeline stopped\n");
+  gst_print ("Pipeline stopped\n");
 
   gst_object_unref (pipeline);
   g_free (server_url);
diff --git a/webrtc/sendonly/webrtc-recvonly-h264.c b/webrtc/sendonly/webrtc-recvonly-h264.c
index 5a5508a..1fc9a1d 100644
--- a/webrtc/sendonly/webrtc-recvonly-h264.c
+++ b/webrtc/sendonly/webrtc-recvonly-h264.c
@@ -190,7 +190,7 @@ handle_media_stream (GstPad * pad, GstElement * pipe, const char *convert_name,
   GstElement *q, *conv, *resample, *sink;
   GstPadLinkReturn ret;
 
-  g_print ("Trying to handle stream with %s ! %s", convert_name, sink_name);
+  gst_print ("Trying to handle stream with %s ! %s", convert_name, sink_name);
 
   q = gst_element_factory_make ("queue", NULL);
   g_assert_nonnull (q);
@@ -232,7 +232,7 @@ on_incoming_decodebin_stream (GstElement * decodebin, GstPad * pad,
   const gchar *name;
 
   if (!gst_pad_has_current_caps (pad)) {
-    g_printerr ("Pad '%s' has no caps, can't do anything, ignoring\n",
+    gst_printerr ("Pad '%s' has no caps, can't do anything, ignoring\n",
         GST_PAD_NAME (pad));
     return;
   }
@@ -245,7 +245,7 @@ on_incoming_decodebin_stream (GstElement * decodebin, GstPad * pad,
   } else if (g_str_has_prefix (name, "audio")) {
     handle_media_stream (pad, pipe, "audioconvert", "autoaudiosink");
   } else {
-    g_printerr ("Unknown pad %s, ignoring", GST_PAD_NAME (pad));
+    gst_printerr ("Unknown pad %s, ignoring", GST_PAD_NAME (pad));
   }
 }
 
@@ -386,7 +386,7 @@ on_offer_created_cb (GstPromise * promise, gpointer user_data)
   gst_promise_unref (local_desc_promise);
 
   sdp_string = gst_sdp_message_as_text (offer->sdp);
-  g_print ("Negotiation offer created:\n%s\n", sdp_string);
+  gst_print ("Negotiation offer created:\n%s\n", sdp_string);
 
   sdp_json = json_object_new ();
   json_object_set_string_member (sdp_json, "type", "sdp");
@@ -413,7 +413,7 @@ on_negotiation_needed_cb (GstElement * webrtcbin, gpointer user_data)
   GstPromise *promise;
   ReceiverEntry *receiver_entry = (ReceiverEntry *) user_data;
 
-  g_print ("Creating negotiation offer\n");
+  gst_print ("Creating negotiation offer\n");
 
   promise = gst_promise_new_with_change_func (on_offer_created_cb,
       (gpointer) receiver_entry, NULL);
@@ -525,7 +525,7 @@ soup_websocket_message_cb (G_GNUC_UNUSED SoupWebsocketConnection * connection,
     }
     sdp_string = json_object_get_string_member (data_json_object, "sdp");
 
-    g_print ("Received SDP:\n%s\n", sdp_string);
+    gst_print ("Received SDP:\n%s\n", sdp_string);
 
     ret = gst_sdp_message_new (&sdp);
     g_assert_cmphex (ret, ==, GST_SDP_OK);
@@ -566,7 +566,7 @@ soup_websocket_message_cb (G_GNUC_UNUSED SoupWebsocketConnection * connection,
     candidate_string = json_object_get_string_member (data_json_object,
         "candidate");
 
-    g_print ("Received ICE candidate with mline index %u; candidate: %s\n",
+    gst_print ("Received ICE candidate with mline index %u; candidate: %s\n",
         mline_index, candidate_string);
 
     g_signal_emit_by_name (receiver_entry->webrtcbin, "add-ice-candidate",
@@ -592,7 +592,7 @@ soup_websocket_closed_cb (SoupWebsocketConnection * connection,
 {
   GHashTable *receiver_entry_table = (GHashTable *) user_data;
   g_hash_table_remove (receiver_entry_table, connection);
-  g_print ("Closed websocket connection %p\n", (gpointer) connection);
+  gst_print ("Closed websocket connection %p\n", (gpointer) connection);
 }
 
 
@@ -629,7 +629,7 @@ soup_websocket_handler (G_GNUC_UNUSED SoupServer * server,
   ReceiverEntry *receiver_entry;
   GHashTable *receiver_entry_table = (GHashTable *) user_data;
 
-  g_print ("Processing new websocket connection %p", (gpointer) connection);
+  gst_print ("Processing new websocket connection %p", (gpointer) connection);
 
   g_signal_connect (G_OBJECT (connection), "closed",
       G_CALLBACK (soup_websocket_closed_cb), (gpointer) receiver_entry_table);
@@ -662,7 +662,7 @@ get_string_from_json_object (JsonObject * object)
 gboolean
 exit_sighandler (gpointer user_data)
 {
-  g_print ("Caught signal, stopping mainloop\n");
+  gst_print ("Caught signal, stopping mainloop\n");
   GMainLoop *mainloop = (GMainLoop *) user_data;
   g_main_loop_quit (mainloop);
   return TRUE;
@@ -699,7 +699,7 @@ main (int argc, char *argv[])
   soup_server_listen_all (soup_server, SOUP_HTTP_PORT,
       (SoupServerListenOptions) 0, NULL);
 
-  g_print ("WebRTC page link: http://127.0.0.1:%d/\n", (gint) SOUP_HTTP_PORT);
+  gst_print ("WebRTC page link: http://127.0.0.1:%d/\n", (gint) SOUP_HTTP_PORT);
 
   g_main_loop_run (mainloop);
 
diff --git a/webrtc/sendonly/webrtc-unidirectional-h264.c b/webrtc/sendonly/webrtc-unidirectional-h264.c
index b806014..e297a53 100644
--- a/webrtc/sendonly/webrtc-unidirectional-h264.c
+++ b/webrtc/sendonly/webrtc-unidirectional-h264.c
@@ -260,7 +260,7 @@ on_offer_created_cb (GstPromise * promise, gpointer user_data)
   gst_promise_unref (local_desc_promise);
 
   sdp_string = gst_sdp_message_as_text (offer->sdp);
-  g_print ("Negotiation offer created:\n%s\n", sdp_string);
+  gst_print ("Negotiation offer created:\n%s\n", sdp_string);
 
   sdp_json = json_object_new ();
   json_object_set_string_member (sdp_json, "type", "sdp");
@@ -287,7 +287,7 @@ on_negotiation_needed_cb (GstElement * webrtcbin, gpointer user_data)
   GstPromise *promise;
   ReceiverEntry *receiver_entry = (ReceiverEntry *) user_data;
 
-  g_print ("Creating negotiation offer\n");
+  gst_print ("Creating negotiation offer\n");
 
   promise = gst_promise_new_with_change_func (on_offer_created_cb,
       (gpointer) receiver_entry, NULL);
@@ -399,7 +399,7 @@ soup_websocket_message_cb (G_GNUC_UNUSED SoupWebsocketConnection * connection,
     }
     sdp_string = json_object_get_string_member (data_json_object, "sdp");
 
-    g_print ("Received SDP:\n%s\n", sdp_string);
+    gst_print ("Received SDP:\n%s\n", sdp_string);
 
     ret = gst_sdp_message_new (&sdp);
     g_assert_cmphex (ret, ==, GST_SDP_OK);
@@ -440,7 +440,7 @@ soup_websocket_message_cb (G_GNUC_UNUSED SoupWebsocketConnection * connection,
     candidate_string = json_object_get_string_member (data_json_object,
         "candidate");
 
-    g_print ("Received ICE candidate with mline index %u; candidate: %s\n",
+    gst_print ("Received ICE candidate with mline index %u; candidate: %s\n",
         mline_index, candidate_string);
 
     g_signal_emit_by_name (receiver_entry->webrtcbin, "add-ice-candidate",
@@ -466,7 +466,7 @@ soup_websocket_closed_cb (SoupWebsocketConnection * connection,
 {
   GHashTable *receiver_entry_table = (GHashTable *) user_data;
   g_hash_table_remove (receiver_entry_table, connection);
-  g_print ("Closed websocket connection %p\n", (gpointer) connection);
+  gst_print ("Closed websocket connection %p\n", (gpointer) connection);
 }
 
 
@@ -503,7 +503,7 @@ soup_websocket_handler (G_GNUC_UNUSED SoupServer * server,
   ReceiverEntry *receiver_entry;
   GHashTable *receiver_entry_table = (GHashTable *) user_data;
 
-  g_print ("Processing new websocket connection %p", (gpointer) connection);
+  gst_print ("Processing new websocket connection %p", (gpointer) connection);
 
   g_signal_connect (G_OBJECT (connection), "closed",
       G_CALLBACK (soup_websocket_closed_cb), (gpointer) receiver_entry_table);
@@ -536,7 +536,7 @@ get_string_from_json_object (JsonObject * object)
 gboolean
 exit_sighandler (gpointer user_data)
 {
-  g_print ("Caught signal, stopping mainloop\n");
+  gst_print ("Caught signal, stopping mainloop\n");
   GMainLoop *mainloop = (GMainLoop *) user_data;
   g_main_loop_quit (mainloop);
   return TRUE;
@@ -573,7 +573,7 @@ main (int argc, char *argv[])
   soup_server_listen_all (soup_server, SOUP_HTTP_PORT,
       (SoupServerListenOptions) 0, NULL);
 
-  g_print ("WebRTC page link: http://127.0.0.1:%d/\n", (gint) SOUP_HTTP_PORT);
+  gst_print ("WebRTC page link: http://127.0.0.1:%d/\n", (gint) SOUP_HTTP_PORT);
 
   g_main_loop_run (mainloop);
 
diff --git a/webrtc/sendrecv/gst/webrtc-sendrecv.c b/webrtc/sendrecv/gst/webrtc-sendrecv.c
index 4453c95..8c4c9f0 100644
--- a/webrtc/sendrecv/gst/webrtc-sendrecv.c
+++ b/webrtc/sendrecv/gst/webrtc-sendrecv.c
@@ -65,7 +65,7 @@ static gboolean
 cleanup_and_quit_loop (const gchar * msg, enum AppState state)
 {
   if (msg)
-    g_printerr ("%s\n", msg);
+    gst_printerr ("%s\n", msg);
   if (state > 0)
     app_state = state;
 
@@ -114,7 +114,7 @@ handle_media_stream (GstPad * pad, GstElement * pipe, const char *convert_name,
   GstElement *q, *conv, *resample, *sink;
   GstPadLinkReturn ret;
 
-  g_print ("Trying to handle stream with %s ! %s", convert_name, sink_name);
+  gst_print ("Trying to handle stream with %s ! %s", convert_name, sink_name);
 
   q = gst_element_factory_make ("queue", NULL);
   g_assert_nonnull (q);
@@ -156,7 +156,7 @@ on_incoming_decodebin_stream (GstElement * decodebin, GstPad * pad,
   const gchar *name;
 
   if (!gst_pad_has_current_caps (pad)) {
-    g_printerr ("Pad '%s' has no caps, can't do anything, ignoring\n",
+    gst_printerr ("Pad '%s' has no caps, can't do anything, ignoring\n",
         GST_PAD_NAME (pad));
     return;
   }
@@ -169,7 +169,7 @@ on_incoming_decodebin_stream (GstElement * decodebin, GstPad * pad,
   } else if (g_str_has_prefix (name, "audio")) {
     handle_media_stream (pad, pipe, "audioconvert", "autoaudiosink");
   } else {
-    g_printerr ("Unknown pad %s, ignoring", GST_PAD_NAME (pad));
+    gst_printerr ("Unknown pad %s, ignoring", GST_PAD_NAME (pad));
   }
 }
 
@@ -233,10 +233,10 @@ send_sdp_to_peer (GstWebRTCSessionDescription * desc)
   sdp = json_object_new ();
 
   if (desc->type == GST_WEBRTC_SDP_TYPE_OFFER) {
-    g_print ("Sending offer:\n%s\n", text);
+    gst_print ("Sending offer:\n%s\n", text);
     json_object_set_string_member (sdp, "type", "offer");
   } else if (desc->type == GST_WEBRTC_SDP_TYPE_ANSWER) {
-    g_print ("Sending answer:\n%s\n", text);
+    gst_print ("Sending answer:\n%s\n", text);
     json_object_set_string_member (sdp, "type", "answer");
   } else {
     g_assert_not_reached ();
@@ -310,7 +310,7 @@ static void
 data_channel_on_open (GObject * dc, gpointer user_data)
 {
   GBytes *bytes = g_bytes_new ("data", strlen ("data"));
-  g_print ("data channel opened\n");
+  gst_print ("data channel opened\n");
   g_signal_emit_by_name (dc, "send-string", "Hi! from GStreamer");
   g_signal_emit_by_name (dc, "send-data", bytes);
   g_bytes_unref (bytes);
@@ -325,7 +325,7 @@ data_channel_on_close (GObject * dc, gpointer user_data)
 static void
 data_channel_on_message_string (GObject * dc, gchar * str, gpointer user_data)
 {
-  g_print ("Received data channel message: %s\n", str);
+  gst_print ("Received data channel message: %s\n", str);
 }
 
 static void
@@ -368,7 +368,7 @@ on_ice_gathering_state_notify (GstElement * webrtcbin, GParamSpec * pspec,
       new_state = "complete";
       break;
   }
-  g_print ("ICE gathering state changed to %s\n", new_state);
+  gst_print ("ICE gathering state changed to %s\n", new_state);
 }
 
 static gboolean
@@ -386,7 +386,7 @@ start_pipeline (void)
       "queue ! " RTP_CAPS_OPUS "97 ! sendrecv. ", &error);
 
   if (error) {
-    g_printerr ("Failed to parse launch: %s\n", error->message);
+    gst_printerr ("Failed to parse launch: %s\n", error->message);
     g_error_free (error);
     goto err;
   }
@@ -411,10 +411,10 @@ start_pipeline (void)
   g_signal_emit_by_name (webrtc1, "create-data-channel", "channel", NULL,
       &send_channel);
   if (send_channel) {
-    g_print ("Created data channel\n");
+    gst_print ("Created data channel\n");
     connect_data_channel_signals (send_channel);
   } else {
-    g_print ("Could not create data channel, is usrsctp available?\n");
+    gst_print ("Could not create data channel, is usrsctp available?\n");
   }
 
   g_signal_connect (webrtc1, "on-data-channel", G_CALLBACK (on_data_channel),
@@ -425,7 +425,7 @@ start_pipeline (void)
   /* Lifetime is the same as the pipeline itself */
   gst_object_unref (webrtc1);
 
-  g_print ("Starting pipeline\n");
+  gst_print ("Starting pipeline\n");
   ret = gst_element_set_state (GST_ELEMENT (pipe1), GST_STATE_PLAYING);
   if (ret == GST_STATE_CHANGE_FAILURE)
     goto err;
@@ -452,7 +452,7 @@ setup_call (void)
   if (!peer_id)
     return FALSE;
 
-  g_print ("Setting up signalling server call with %s\n", peer_id);
+  gst_print ("Setting up signalling server call with %s\n", peer_id);
   app_state = PEER_CONNECTING;
   msg = g_strdup_printf ("SESSION %s", peer_id);
   soup_websocket_connection_send_text (ws_conn, msg);
@@ -471,7 +471,7 @@ register_with_server (void)
     return FALSE;
 
   our_id = g_random_int_range (10, 10000);
-  g_print ("Registering id %i with server\n", our_id);
+  gst_print ("Registering id %i with server\n", our_id);
   app_state = SERVER_REGISTERING;
 
   /* Register with the server with a random integer id. Reply will be received
@@ -550,7 +550,7 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
 
   switch (type) {
     case SOUP_WEBSOCKET_DATA_BINARY:
-      g_printerr ("Received unknown binary message, ignoring\n");
+      gst_printerr ("Received unknown binary message, ignoring\n");
       return;
     case SOUP_WEBSOCKET_DATA_TEXT:{
       gsize size;
@@ -571,7 +571,7 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
       goto out;
     }
     app_state = SERVER_REGISTERED;
-    g_print ("Registered with server\n");
+    gst_print ("Registered with server\n");
     /* Ask signalling server to connect us with a specific peer */
     if (!setup_call ()) {
       cleanup_and_quit_loop ("ERROR: Failed to setup call", PEER_CALL_ERROR);
@@ -616,14 +616,14 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
     JsonObject *object, *child;
     JsonParser *parser = json_parser_new ();
     if (!json_parser_load_from_data (parser, text, -1, NULL)) {
-      g_printerr ("Unknown message '%s', ignoring", text);
+      gst_printerr ("Unknown message '%s', ignoring", text);
       g_object_unref (parser);
       goto out;
     }
 
     root = json_parser_get_root (parser);
     if (!JSON_NODE_HOLDS_OBJECT (root)) {
-      g_printerr ("Unknown json message '%s', ignoring", text);
+      gst_printerr ("Unknown json message '%s', ignoring", text);
       g_object_unref (parser);
       goto out;
     }
@@ -660,7 +660,7 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
       g_assert_cmphex (ret, ==, GST_SDP_OK);
 
       if (g_str_equal (sdptype, "answer")) {
-        g_print ("Received answer:\n%s\n", text);
+        gst_print ("Received answer:\n%s\n", text);
         answer = gst_webrtc_session_description_new (GST_WEBRTC_SDP_TYPE_ANSWER,
             sdp);
         g_assert_nonnull (answer);
@@ -675,7 +675,7 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
         }
         app_state = PEER_CALL_STARTED;
       } else {
-        g_print ("Received offer:\n%s\n", text);
+        gst_print ("Received offer:\n%s\n", text);
         on_offer_received (sdp);
       }
 
@@ -691,7 +691,7 @@ on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
       g_signal_emit_by_name (webrtc1, "add-ice-candidate", sdpmlineindex,
           candidate);
     } else {
-      g_printerr ("Ignoring unknown JSON message:\n%s\n", text);
+      gst_printerr ("Ignoring unknown JSON message:\n%s\n", text);
     }
     g_object_unref (parser);
   }
@@ -716,7 +716,7 @@ on_server_connected (SoupSession * session, GAsyncResult * res,
   g_assert_nonnull (ws_conn);
 
   app_state = SERVER_CONNECTED;
-  g_print ("Connected to signalling server\n");
+  gst_print ("Connected to signalling server\n");
 
   g_signal_connect (ws_conn, "closed", G_CALLBACK (on_server_closed), NULL);
   g_signal_connect (ws_conn, "message", G_CALLBACK (on_server_message), NULL);
@@ -748,7 +748,7 @@ connect_to_websocket_server_async (void)
 
   message = soup_message_new (SOUP_METHOD_GET, server_url);
 
-  g_print ("Connecting to server...\n");
+  gst_print ("Connecting to server...\n");
 
   /* Once connected, we will register */
   soup_session_websocket_connect_async (session, message, NULL, NULL, NULL,
@@ -772,7 +772,7 @@ check_plugins (void)
   for (i = 0; i < g_strv_length ((gchar **) needed); i++) {
     plugin = gst_registry_find_plugin (registry, needed[i]);
     if (!plugin) {
-      g_print ("Required gstreamer plugin '%s' not found\n", needed[i]);
+      gst_print ("Required gstreamer plugin '%s' not found\n", needed[i]);
       ret = FALSE;
       continue;
     }
@@ -791,7 +791,7 @@ main (int argc, char *argv[])
   g_option_context_add_main_entries (context, entries, NULL);
   g_option_context_add_group (context, gst_init_get_option_group ());
   if (!g_option_context_parse (context, &argc, &argv, &error)) {
-    g_printerr ("Error initializing: %s\n", error->message);
+    gst_printerr ("Error initializing: %s\n", error->message);
     return -1;
   }
 
@@ -799,7 +799,7 @@ main (int argc, char *argv[])
     return -1;
 
   if (!peer_id) {
-    g_printerr ("--peer-id is a required argument\n");
+    gst_printerr ("--peer-id is a required argument\n");
     return -1;
   }
 
@@ -822,7 +822,7 @@ main (int argc, char *argv[])
 
   if (pipe1) {
     gst_element_set_state (GST_ELEMENT (pipe1), GST_STATE_NULL);
-    g_print ("Pipeline stopped\n");
+    gst_print ("Pipeline stopped\n");
     gst_object_unref (pipe1);
   }