base: Fix fallthrough declarations

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8229>
2025-03-28 20:05:38 +00:00 · 2025-01-03 10:35:57 +01:00 · 2025-01-03 10:35:57 +01:00 · 319ff17300
commit 319ff17300
parent 0e8f8dfe15
14 changed files with 21 additions and 9 deletions
--- a/subprojects/gst-plugins-base/ext/theora/gsttheoradec.c
+++ b/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;
--- a/subprojects/gst-plugins-base/ext/theora/gsttheoraparse.c
+++ b/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,
--- a/subprojects/gst-plugins-base/ext/vorbis/gstvorbisparse.c
+++ b/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,
--- a/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudiodecoder.c
+++ b/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;
--- a/subprojects/gst-plugins-base/gst-libs/gst/audio/gstaudioencoder.c
+++ b/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)) {
--- a/subprojects/gst-plugins-base/gst-libs/gst/pbutils/codec-utils.c
+++ b/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");
--- a/subprojects/gst-plugins-base/gst-libs/gst/riff/riff-media.c
+++ b/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;
--- a/subprojects/gst-plugins-base/gst-libs/gst/tag/gsttagdemux.c
+++ b/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;
--- a/subprojects/gst-plugins-base/gst-libs/gst/tag/id3v2.c
+++ b/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];
--- a/subprojects/gst-plugins-base/gst/playback/gstplaybin2.c
+++ b/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,
--- a/subprojects/gst-plugins-base/gst/playback/gsturisourcebin.c
+++ b/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;
--- a/subprojects/gst-plugins-base/gst/videorate/gstvideorate.c
+++ b/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;
--- a/subprojects/gst-plugins-base/gst/videotestsrc/gstvideotestsrc.c
+++ b/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;
--- a/subprojects/gst-plugins-base/tests/examples/playback/playback-test.c
+++ b/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);