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mpegaudioparse: Remove empty lines added by buggy indent.
This commit is contained in:
René Stadler
parent
d950699d2e
commit
be6292d4de
1 changed files with 0 additions and 63 deletions
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@ -102,26 +102,19 @@ enum
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static void gst_mp3parse_class_init (GstMPEGAudioParseClass * klass);
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static void gst_mp3parse_base_init (gpointer klass);
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static void gst_mp3parse_init (GstMPEGAudioParse * mp3parse,
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GstMPEGAudioParseClass * klass);
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static gboolean gst_mp3parse_sink_event (GstPad * pad, GstEvent * event);
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static GstFlowReturn gst_mp3parse_chain (GstPad * pad, GstBuffer * buffer);
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static gboolean mp3parse_src_query (GstPad * pad, GstQuery * query);
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static const GstQueryType *mp3parse_get_query_types (GstPad * pad);
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static gboolean mp3parse_src_event (GstPad * pad, GstEvent * event);
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static int head_check (GstMPEGAudioParse * mp3parse, unsigned long head);
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static void gst_mp3parse_dispose (GObject * object);
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static void gst_mp3parse_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_mp3parse_get_property (GObject * object, guint prop_id,
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@ -143,7 +136,6 @@ static GType
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gst_mp3_channel_mode_get_type (void)
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{
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static GType mp3_channel_mode_type = 0;
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static GEnumValue mp3_channel_mode[] = {
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{MP3_CHANNEL_MODE_UNKNOWN, "Unknown", "unknown"},
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{MP3_CHANNEL_MODE_MONO, "Mono", "mono"},
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@ -188,7 +180,6 @@ mp3_type_frame_length_from_header (GstMPEGAudioParse * mp3parse, guint32 header,
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gulong mode, samplerate, bitrate, layer, channels, padding, crc;
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gulong version;
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gint lsf, mpg25;
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GEnumValue *mode_enum;
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if (header & (1 << 20)) {
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@ -297,7 +288,6 @@ static void
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gst_mp3parse_class_init (GstMPEGAudioParseClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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gobject_class = (GObjectClass *) klass;
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@ -448,9 +438,7 @@ static gboolean
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gst_mp3parse_sink_event (GstPad * pad, GstEvent * event)
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{
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gboolean res = TRUE;
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GstMPEGAudioParse *mp3parse;
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GstEvent **eventp;
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mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad));
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@ -459,11 +447,8 @@ gst_mp3parse_sink_event (GstPad * pad, GstEvent * event)
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case GST_EVENT_NEWSEGMENT:
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{
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gdouble rate, applied_rate;
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GstFormat format;
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gint64 start, stop, pos;
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gboolean update;
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gst_event_parse_new_segment_full (event, &update, &rate, &applied_rate,
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@ -472,7 +457,6 @@ gst_mp3parse_sink_event (GstPad * pad, GstEvent * event)
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g_mutex_lock (mp3parse->pending_accurate_seeks_lock);
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if (format == GST_FORMAT_BYTES && mp3parse->pending_accurate_seeks) {
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MPEGAudioPendingAccurateSeek *seek = NULL;
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GSList *node;
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for (node = mp3parse->pending_accurate_seeks; node; node = node->next) {
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@ -623,13 +607,9 @@ gst_mp3parse_emit_frame (GstMPEGAudioParse * mp3parse, guint size,
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guint mode, guint crc)
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{
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GstBuffer *outbuf;
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guint bitrate;
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GstFlowReturn ret = GST_FLOW_OK;
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GstClockTime push_start;
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GstTagList *taglist;
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outbuf = gst_adapter_take_buffer (mp3parse->adapter, size);
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@ -859,21 +839,15 @@ static void
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gst_mp3parse_handle_first_frame (GstMPEGAudioParse * mp3parse)
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{
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GstTagList *taglist;
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gchar *codec;
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const guint32 xing_id = 0x58696e67; /* 'Xing' in hex */
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const guint32 info_id = 0x496e666f; /* 'Info' in hex - found in LAME CBR files */
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const guint32 vbri_id = 0x56425249; /* 'VBRI' in hex */
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gint offset;
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guint64 avail;
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guint32 read_id;
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const guint8 *data;
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/* Output codec tag */
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@ -927,11 +901,8 @@ gst_mp3parse_handle_first_frame (GstMPEGAudioParse * mp3parse)
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read_id = GST_READ_UINT32_BE (data);
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if (read_id == xing_id || read_id == info_id) {
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guint32 xing_flags;
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guint bytes_needed = offset + 8;
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gint64 total_bytes;
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GstClockTime total_time;
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GST_DEBUG_OBJECT (mp3parse, "Found Xing header marker 0x%x", xing_id);
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@ -999,9 +970,7 @@ gst_mp3parse_handle_first_frame (GstMPEGAudioParse * mp3parse)
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if (xing_flags & XING_TOC_FLAG) {
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int i, percent = 0;
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guchar *table = mp3parse->xing_seek_table;
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guchar old = 0;
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if (data[0] != 0) {
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@ -1029,7 +998,6 @@ gst_mp3parse_handle_first_frame (GstMPEGAudioParse * mp3parse)
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mp3parse->xing_seek_table_inverse[i] = percent * 100;
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} else if (table[percent] < i && percent < 99) {
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gdouble fa, fb, fx;
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gint a = percent, b = percent + 1;
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fa = table[a];
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@ -1038,7 +1006,6 @@ gst_mp3parse_handle_first_frame (GstMPEGAudioParse * mp3parse)
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mp3parse->xing_seek_table_inverse[i] = (guint16) (fx * 100);
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} else if (percent == 98 && table[percent + 1] <= i) {
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gdouble fa, fb, fx;
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gint a = percent + 1, b = 100;
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fa = table[a];
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@ -1066,9 +1033,7 @@ gst_mp3parse_handle_first_frame (GstMPEGAudioParse * mp3parse)
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mp3parse->xing_vbr_scale);
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} else if (read_id == vbri_id) {
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gint64 total_bytes, total_frames;
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GstClockTime total_time;
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guint16 nseek_points;
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GST_DEBUG_OBJECT (mp3parse, "Found VBRI header marker 0x%x", vbri_id);
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@ -1123,7 +1088,6 @@ gst_mp3parse_handle_first_frame (GstMPEGAudioParse * mp3parse)
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if (nseek_points > 0) {
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guint scale, seek_bytes, seek_frames;
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gint i;
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mp3parse->vbri_seek_points = nseek_points;
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@ -1197,17 +1161,11 @@ static GstFlowReturn
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gst_mp3parse_chain (GstPad * pad, GstBuffer * buf)
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{
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GstFlowReturn flow = GST_FLOW_OK;
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GstMPEGAudioParse *mp3parse;
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const guchar *data;
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guint32 header;
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int bpf;
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guint available;
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GstClockTime timestamp;
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mp3parse = GST_MP3PARSE (GST_PAD_PARENT (pad));
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@ -1286,7 +1244,6 @@ gst_mp3parse_chain (GstPad * pad, GstBuffer * buf)
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*************************************************************************/
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if (mp3parse->resyncing) {
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guint32 header2;
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const guint8 *data2;
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/* wait until we have the the entire current frame as well as the next
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@ -1499,7 +1456,6 @@ static GstStateChangeReturn
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gst_mp3parse_change_state (GstElement * element, GstStateChange transition)
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{
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GstMPEGAudioParse *mp3parse;
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GstStateChangeReturn result;
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mp3parse = GST_MP3PARSE (element);
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@ -1574,7 +1530,6 @@ mp3parse_time_to_bytepos (GstMPEGAudioParse * mp3parse, GstClockTime ts,
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gint64 * bytepos)
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{
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gint64 total_bytes;
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GstClockTime total_time;
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/* -1 always maps to -1 */
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@ -1588,7 +1543,6 @@ mp3parse_time_to_bytepos (GstMPEGAudioParse * mp3parse, GstClockTime ts,
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mp3parse_total_bytes (mp3parse, &total_bytes) &&
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mp3parse_total_time (mp3parse, &total_time)) {
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gdouble fa, fb, fx;
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gdouble percent =
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CLAMP ((100.0 * gst_util_guint64_to_gdouble (ts)) /
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gst_util_guint64_to_gdouble (total_time), 0.0, 100.0);
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@ -1611,7 +1565,6 @@ mp3parse_time_to_bytepos (GstMPEGAudioParse * mp3parse, GstClockTime ts,
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mp3parse_total_bytes (mp3parse, &total_bytes) &&
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mp3parse_total_time (mp3parse, &total_time)) {
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gint i, j;
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gdouble a, b, fa, fb;
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i = gst_util_uint64_scale (ts, mp3parse->vbri_seek_points - 1, total_time);
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@ -1653,7 +1606,6 @@ mp3parse_bytepos_to_time (GstMPEGAudioParse * mp3parse,
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gint64 bytepos, GstClockTime * ts, gboolean from_total_time)
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{
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gint64 total_bytes;
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GstClockTime total_time;
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if (bytepos == -1) {
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@ -1671,9 +1623,7 @@ mp3parse_bytepos_to_time (GstMPEGAudioParse * mp3parse,
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mp3parse_total_bytes (mp3parse, &total_bytes) &&
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mp3parse_total_time (mp3parse, &total_time)) {
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gdouble fa, fb, fx;
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gdouble pos = CLAMP ((bytepos * 256.0) / total_bytes, 0.0, 256.0);
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gint index = CLAMP (pos, 0, 255);
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fa = mp3parse->xing_seek_table_inverse[index];
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@ -1693,9 +1643,7 @@ mp3parse_bytepos_to_time (GstMPEGAudioParse * mp3parse,
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mp3parse_total_bytes (mp3parse, &total_bytes) &&
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mp3parse_total_time (mp3parse, &total_time)) {
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gint i = 0;
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guint64 sum = 0;
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gdouble a, b, fa, fb;
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@ -1737,15 +1685,10 @@ static gboolean
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mp3parse_handle_seek (GstMPEGAudioParse * mp3parse, GstEvent * event)
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{
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GstFormat format;
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gdouble rate;
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GstSeekFlags flags;
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GstSeekType cur_type, stop_type;
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gint64 cur, stop;
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gint64 byte_cur, byte_stop;
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gst_event_parse_seek (event, &rate, &format, &flags, &cur_type, &cur,
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@ -1792,7 +1735,6 @@ mp3parse_handle_seek (GstMPEGAudioParse * mp3parse, GstEvent * event)
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start = 0;
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} else {
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MPEGAudioSeekEntry *entry = NULL, *start_entry = NULL, *stop_entry = NULL;
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GList *start_node, *stop_node;
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gint64 seek_ts = (cur > mp3parse->max_bitreservoir) ?
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(cur - mp3parse->max_bitreservoir) : 0;
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@ -1880,7 +1822,6 @@ static gboolean
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mp3parse_src_event (GstPad * pad, GstEvent * event)
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{
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GstMPEGAudioParse *mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad));
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gboolean res = FALSE;
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g_return_val_if_fail (mp3parse != NULL, FALSE);
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@ -1903,13 +1844,9 @@ static gboolean
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mp3parse_src_query (GstPad * pad, GstQuery * query)
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{
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GstFormat format;
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GstClockTime total;
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GstMPEGAudioParse *mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad));
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gboolean res = FALSE;
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GstPad *peer;
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g_return_val_if_fail (mp3parse != NULL, FALSE);
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