base: Fix fallthrough declarations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8229>

subprojects/gst-plugins-base/ext/theora/gsttheoradec.c

@@ -567,6 +567,7 @@ theora_handle_header_packet (GstTheoraDec * dec, ogg_packet * packet)
     default:
       /* ignore */
       GST_WARNING_OBJECT (dec, "unknown theora header packet found");
+      /* FALLTHROUGH */
     case 0x80:
       /* nothing special, this is the identification header */
       res = GST_FLOW_OK;

subprojects/gst-plugins-base/ext/theora/gsttheoraparse.c

@@ -769,6 +769,7 @@ theora_parse_src_convert (GstPad * pad,
       switch (*dest_format) {
         case GST_FORMAT_BYTES:
           scale = 3 * (parse->info.pic_width * parse->info.pic_height) / 2;
+          /* FALLTHROUGH */
         case GST_FORMAT_DEFAULT:
           *dest_value = scale * gst_util_uint64_scale (src_value,
               parse->info.fps_numerator,

subprojects/gst-plugins-base/ext/vorbis/gstvorbisparse.c

@@ -556,6 +556,7 @@ vorbis_parse_convert (GstPad * pad,
       switch (*dest_format) {
         case GST_FORMAT_BYTES:
           scale = sizeof (float) * parse->vi.channels;
+          /* FALLTHROUGH */
         case GST_FORMAT_DEFAULT:
           *dest_value =
               scale * gst_util_uint64_scale_int (src_value, parse->vi.rate,

subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c

@@ -2549,9 +2549,8 @@ gst_audio_decoder_sink_eventfunc (GstAudioDecoder * dec, GstEvent * event)
           break;
         }
       }
-
-      /* fall through */
     }
+      /* FALLTHROUGH */
     default:
       if (!GST_EVENT_IS_SERIALIZED (event)) {
         ret =
@@ -2908,8 +2907,8 @@ gst_audio_decoder_sink_query_default (GstAudioDecoder * dec, GstQuery * query)
         res = FALSE;
         break;
       }
-      /* fall-through */
     }
+      /* FALLTHROUGH */
     default:
       res = gst_pad_query_default (pad, GST_OBJECT_CAST (dec), query);
       break;

subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c

@@ -1721,8 +1721,8 @@ gst_audio_encoder_sink_event_default (GstAudioEncoder * enc, GstEvent * event)
           break;
         }
       }
-      /* fall through */
     }
+      /* FALLTHROUGH */
     default:
       /* Forward non-serialized events immediately. */
       if (!GST_EVENT_IS_SERIALIZED (event)) {

subprojects/gst-plugins-base/gst-libs/gst/pbutils/codec-utils.c

@@ -3410,6 +3410,7 @@ gst_codec_utils_caps_from_mime_codec_single (const gchar * codec)
       GST_FIXME
           ("Signalling of ATMOS ('ec+3') isn't defined yet. Falling back to EAC3 caps");
       /* withdrawn, unused, do not use (was enhanced AC-3 audio with JOC) */
+      /* FALLTHROUGH */
     case GST_MAKE_FOURCC ('e', 'c', '-', '3'):
       /* ETSI TS 102 366 v1.4.1 - Digital Audio Compression (AC-3, Enhanced AC-3) Standard, Annex F */
       caps = gst_caps_new_empty_simple ("audio/x-eac3");

subprojects/gst-plugins-base/gst-libs/gst/riff/riff-media.c

@@ -1232,18 +1232,24 @@ gst_riff_wavext_get_default_channel_mask (guint nchannels)
     case 11:
       channel_mask |= 0x00400;
       channel_mask |= 0x00200;
+      /* FALLTHROUGH */
     case 9:
       channel_mask |= 0x00100;
+      /* FALLTHROUGH */
     case 8:
       channel_mask |= 0x00080;
       channel_mask |= 0x00040;
+      /* FALLTHROUGH */
     case 6:
       channel_mask |= 0x00020;
       channel_mask |= 0x00010;
+      /* FALLTHROUGH */
     case 4:
       channel_mask |= 0x00008;
+      /* FALLTHROUGH */
     case 3:
       channel_mask |= 0x00004;
+      /* FALLTHROUGH */
     case 2:
       channel_mask |= 0x00002;
       channel_mask |= 0x00001;

subprojects/gst-plugins-base/gst-libs/gst/tag/gsttagdemux.c

@@ -639,7 +639,7 @@ gst_tag_demux_chain_buffer (GstTagDemux * demux, GstBuffer * buf,
       gst_tag_demux_chain_parse_tag (demux);
       if (demux->priv->state != GST_TAG_DEMUX_TYPEFINDING)
         break;
-      /* Fall-through */
+      /* FALLTHROUGH */
     case GST_TAG_DEMUX_TYPEFINDING:{
       GstTypeFindProbability probability = 0;
       GstBuffer *typefind_buf = NULL;
@@ -696,8 +696,8 @@ gst_tag_demux_chain_buffer (GstTagDemux * demux, GstBuffer * buf,
       /* Move onto streaming and fall-through to push out existing
        * data */
       demux->priv->state = GST_TAG_DEMUX_STREAMING;
-      /* fall-through */
     }
+      /* FALLTHROUGH */
     case GST_TAG_DEMUX_STREAMING:{
       GstBuffer *outbuf = NULL;
       gsize outbuf_size;

subprojects/gst-plugins-base/gst-libs/gst/tag/id3v2.c

@@ -525,6 +525,7 @@ id3v2_frames_to_tag_list (ID3TagsWorking * work)
         break;
       case 3:
         read_synch_size = FALSE;        /* 2.3 frame size is not synch-safe */
+        /* FALLTHROUGH */
       case 4:
       default:
         frame_id[0] = work->hdr.frame_data[0];

subprojects/gst-plugins-base/gst/playback/gstplaybin2.c

@@ -5839,6 +5839,7 @@ gst_play_bin_change_state (GstElement * element, GstStateChange transition)
       GST_PLAY_BIN_DYN_UNLOCK (playbin);
       if (!do_save)
         break;
+      /* FALLTHROUGH */
     case GST_STATE_CHANGE_READY_TO_NULL:
       /* we go async to PAUSED, so if that fails, we never make it to PAUSED
        * and no state change PAUSED to READY passes here,

subprojects/gst-plugins-base/gst/playback/gsturisourcebin.c

@@ -992,7 +992,7 @@ demux_pad_events (GstPad * pad, GstPadProbeInfo * info, OutputSlotInfo * slot)
         GST_PAD_PROBE_INFO_DATA (info) = add_stream_start_custom_flag (&ev);
       }
     }
-      /* PASSTHROUGH */
+      /* FALLTHROUGH */
     case GST_EVENT_FLUSH_STOP:
       BUFFERING_LOCK (urisrc);
       slot->is_eos = FALSE;

subprojects/gst-plugins-base/gst/videorate/gstvideorate.c

@@ -1256,8 +1256,8 @@ gst_video_rate_src_event (GstBaseTransform * trans, GstEvent * event)
         gst_event_unref (event);
         event = gst_event_new_qos (type, proportion, diff, timestamp);
       }
-      /* Fallthrough */
     }
+      /* FALLTHROUGH */
     default:
       res = gst_pad_push_event (sinkpad, event);
       break;

subprojects/gst-plugins-base/gst/videotestsrc/gstvideotestsrc.c

@@ -1179,8 +1179,8 @@ gst_video_test_src_query (GstBaseSrc * bsrc, GstQuery * query)
             break;
         }
       }
-      /* fall through */
     }
+      /* FALLTHROUGH */
     default:
       res = GST_BASE_SRC_CLASS (parent_class)->query (bsrc, query);
       break;

subprojects/gst-plugins-base/tests/examples/playback/playback-test.c

@@ -2039,6 +2039,7 @@ msg_buffering (GstBus * bus, GstMessage * message, PlaybackApp * app)
       break;
     case GST_BUFFERING_LIVE:
       app->is_live = TRUE;
+      /* FALLTHROUGH */
     case GST_BUFFERING_TIMESHIFT:
     case GST_BUFFERING_STREAM:
       do_stream_buffering (app, percent);