From 61d200a957125a5fc7d6d3536ce8341c1fedd5cd Mon Sep 17 00:00:00 2001 From: Seungha Yang Date: Mon, 27 Jul 2020 02:20:59 +0900 Subject: [PATCH] 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: --- network/http-launch/http-launch.c | 38 +++--- playback/player/gst-play/gst-play.c | 118 +++++++++--------- playback/player/gtk/gtk-play.c | 10 +- playback/player/qt/extension/qgstplayer.cpp | 4 +- .../gst/mp-webrtc-sendrecv.c | 61 ++++----- webrtc/sendonly/webrtc-recvonly-h264.c | 22 ++-- webrtc/sendonly/webrtc-unidirectional-h264.c | 16 +-- webrtc/sendrecv/gst/webrtc-sendrecv.c | 56 ++++----- 8 files changed, 163 insertions(+), 162 deletions(-) diff --git a/network/http-launch/http-launch.c b/network/http-launch/http-launch.c index 5eedf0062d..ac795483bb 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 \n" + gst_print ("usage: %s PORT \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 d4c9e67ed2..b7b98e1fd8 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 3cea7e6cfe..786458e298 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 6667e8d6c4..4ea16aab96 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 1c5b0ecc23..5f8076eeb5 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 5a5508afc2..1fc9a1dfa2 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 b8060145fe..e297a534d5 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 4453c954da..8c4c9f0be5 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); }