gst_plugins_good/gst/mpegaudioparse/gstmpegaudioparse.c
changeset 20 7e3786c5ed27
--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/gst_plugins_good/gst/mpegaudioparse/gstmpegaudioparse.c	Fri May 14 18:43:44 2010 -0500
@@ -0,0 +1,2199 @@
+/* GStreamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ * Copyright (C) <2006-2007> Jan Schmidt <thaytan@mad.scientist.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "../../config.h"
+#endif
+
+#include <string.h>
+
+#include "gstmpegaudioparse.h"
+
+GST_DEBUG_CATEGORY_STATIC (mp3parse_debug);
+#define GST_CAT_DEFAULT mp3parse_debug
+
+#define MP3_CHANNEL_MODE_UNKNOWN -1
+#define MP3_CHANNEL_MODE_STEREO 0
+#define MP3_CHANNEL_MODE_JOINT_STEREO 1
+#define MP3_CHANNEL_MODE_DUAL_CHANNEL 2
+#define MP3_CHANNEL_MODE_MONO 3
+
+#define CRC_UNKNOWN -1
+#define CRC_PROTECTED 0
+#define CRC_NOT_PROTECTED 1
+
+#define XING_FRAMES_FLAG     0x0001
+#define XING_BYTES_FLAG      0x0002
+#define XING_TOC_FLAG        0x0004
+#define XING_VBR_SCALE_FLAG  0x0008
+
+#ifndef GST_READ_UINT24_BE
+#define GST_READ_UINT24_BE(p) (p[2] | (p[1] << 8) | (p[0] << 16))
+#endif
+
+/* Minimum number of consecutive, valid-looking frames to consider
+   for resyncing */
+#define MIN_RESYNC_FRAMES 3
+
+static inline MPEGAudioSeekEntry *
+mpeg_audio_seek_entry_new ()
+{
+  return g_slice_new (MPEGAudioSeekEntry);
+}
+
+static inline void
+mpeg_audio_seek_entry_free (MPEGAudioSeekEntry * entry)
+{
+  g_slice_free (MPEGAudioSeekEntry, entry);
+}
+
+/* elementfactory information */
+static GstElementDetails mp3parse_details = {
+  "MPEG1 Audio Parser",
+  "Codec/Parser/Audio",
+  "Parses and frames mpeg1 audio streams (levels 1-3), provides seek",
+  "Jan Schmidt <thaytan@mad.scientist.com>\n"
+      "Erik Walthinsen <omega@cse.ogi.edu>"
+};
+
+static GstStaticPadTemplate mp3_src_template = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/mpeg, "
+        "mpegversion = (int) 1, "
+        "layer = (int) [ 1, 3 ], "
+        "rate = (int) [ 8000, 48000 ], channels = (int) [ 1, 2 ],"
+        "parsed=(boolean) true")
+    );
+
+static GstStaticPadTemplate mp3_sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/mpeg, mpegversion = (int) 1, parsed=(boolean)false")
+    );
+
+/* GstMPEGAudioParse signals and args */
+enum
+{
+  /* FILL ME */
+  LAST_SIGNAL
+};
+
+enum
+{
+  ARG_0,
+  ARG_SKIP,
+  ARG_BIT_RATE
+      /* FILL ME */
+};
+
+
+static void gst_mp3parse_class_init (GstMPEGAudioParseClass * klass);
+static void gst_mp3parse_base_init (gpointer klass);
+static void gst_mp3parse_init (GstMPEGAudioParse * mp3parse,
+    GstMPEGAudioParseClass * klass);
+
+static gboolean gst_mp3parse_sink_event (GstPad * pad, GstEvent * event);
+static GstFlowReturn gst_mp3parse_chain (GstPad * pad, GstBuffer * buffer);
+static gboolean mp3parse_src_query (GstPad * pad, GstQuery * query);
+static const GstQueryType *mp3parse_get_query_types (GstPad * pad);
+static gboolean mp3parse_src_event (GstPad * pad, GstEvent * event);
+
+static int head_check (GstMPEGAudioParse * mp3parse, unsigned long head);
+
+static void gst_mp3parse_dispose (GObject * object);
+static void gst_mp3parse_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_mp3parse_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+static GstStateChangeReturn gst_mp3parse_change_state (GstElement * element,
+    GstStateChange transition);
+static GstFlowReturn
+gst_mp3parse_handle_data (GstMPEGAudioParse * mp3parse, gboolean at_eos);
+
+static gboolean mp3parse_bytepos_to_time (GstMPEGAudioParse * mp3parse,
+    gint64 bytepos, GstClockTime * ts, gboolean from_total_time);
+static gboolean
+mp3parse_total_bytes (GstMPEGAudioParse * mp3parse, gint64 * total);
+static gboolean
+mp3parse_total_time (GstMPEGAudioParse * mp3parse, GstClockTime * total);
+
+GST_BOILERPLATE (GstMPEGAudioParse, gst_mp3parse, GstElement, GST_TYPE_ELEMENT);
+
+#define GST_TYPE_MP3_CHANNEL_MODE (gst_mp3_channel_mode_get_type())
+
+static const GEnumValue mp3_channel_mode[] = {
+  {MP3_CHANNEL_MODE_UNKNOWN, "Unknown", "unknown"},
+  {MP3_CHANNEL_MODE_MONO, "Mono", "mono"},
+  {MP3_CHANNEL_MODE_DUAL_CHANNEL, "Dual Channel", "dual-channel"},
+  {MP3_CHANNEL_MODE_JOINT_STEREO, "Joint Stereo", "joint-stereo"},
+  {MP3_CHANNEL_MODE_STEREO, "Stereo", "stereo"},
+  {0, NULL, NULL},
+};
+
+static GType
+gst_mp3_channel_mode_get_type (void)
+{
+  static GType mp3_channel_mode_type = 0;
+
+  if (!mp3_channel_mode_type) {
+    mp3_channel_mode_type =
+        g_enum_register_static ("GstMp3ChannelMode", mp3_channel_mode);
+  }
+  return mp3_channel_mode_type;
+}
+
+static const gchar *
+gst_mp3_channel_mode_get_nick (gint mode)
+{
+  guint i;
+  for (i = 0; i < G_N_ELEMENTS (mp3_channel_mode); i++) {
+    if (mp3_channel_mode[i].value == mode)
+      return mp3_channel_mode[i].value_nick;
+  }
+  return NULL;
+}
+
+static const guint mp3types_bitrates[2][3][16] = {
+  {
+        {0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448,},
+        {0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384,},
+        {0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320,}
+      },
+  {
+        {0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256,},
+        {0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,},
+        {0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,}
+      },
+};
+
+static const guint mp3types_freqs[3][3] = { {44100, 48000, 32000},
+{22050, 24000, 16000},
+{11025, 12000, 8000}
+};
+
+static inline guint
+mp3_type_frame_length_from_header (GstMPEGAudioParse * mp3parse, guint32 header,
+    guint * put_version, guint * put_layer, guint * put_channels,
+    guint * put_bitrate, guint * put_samplerate, guint * put_mode,
+    guint * put_crc)
+{
+  guint length;
+  gulong mode, samplerate, bitrate, layer, channels, padding, crc;
+  gulong version;
+  gint lsf, mpg25;
+
+  if (header & (1 << 20)) {
+    lsf = (header & (1 << 19)) ? 0 : 1;
+    mpg25 = 0;
+  } else {
+    lsf = 1;
+    mpg25 = 1;
+  }
+
+  version = 1 + lsf + mpg25;
+
+  layer = 4 - ((header >> 17) & 0x3);
+
+  crc = (header >> 16) & 0x1;
+
+  bitrate = (header >> 12) & 0xF;
+  bitrate = mp3types_bitrates[lsf][layer - 1][bitrate] * 1000;
+  /* The caller has ensured we have a valid header, so bitrate can't be
+     zero here. */
+  g_assert (bitrate != 0);
+
+  samplerate = (header >> 10) & 0x3;
+  samplerate = mp3types_freqs[lsf + mpg25][samplerate];
+
+  padding = (header >> 9) & 0x1;
+
+  mode = (header >> 6) & 0x3;
+  channels = (mode == 3) ? 1 : 2;
+
+  switch (layer) {
+    case 1:
+      length = 4 * ((bitrate * 12) / samplerate + padding);
+      break;
+    case 2:
+      length = (bitrate * 144) / samplerate + padding;
+      break;
+    default:
+    case 3:
+      length = (bitrate * 144) / (samplerate << lsf) + padding;
+      break;
+  }
+
+  GST_DEBUG_OBJECT (mp3parse, "Calculated mp3 frame length of %u bytes",
+      length);
+  GST_DEBUG_OBJECT (mp3parse, "samplerate = %lu, bitrate = %lu, version = %lu, "
+      "layer = %lu, channels = %lu, mode = %s", samplerate, bitrate, version,
+      layer, channels, gst_mp3_channel_mode_get_nick (mode));
+
+  if (put_version)
+    *put_version = version;
+  if (put_layer)
+    *put_layer = layer;
+  if (put_channels)
+    *put_channels = channels;
+  if (put_bitrate)
+    *put_bitrate = bitrate;
+  if (put_samplerate)
+    *put_samplerate = samplerate;
+  if (put_mode)
+    *put_mode = mode;
+  if (put_crc)
+    *put_crc = crc;
+
+  return length;
+}
+
+static GstCaps *
+mp3_caps_create (guint version, guint layer, guint channels, guint samplerate)
+{
+  GstCaps *new;
+
+  g_assert (version);
+  g_assert (layer);
+  g_assert (samplerate);
+  g_assert (channels);
+
+  new = gst_caps_new_simple ("audio/mpeg",
+      "mpegversion", G_TYPE_INT, 1,
+      "mpegaudioversion", G_TYPE_INT, version,
+      "layer", G_TYPE_INT, layer,
+      "rate", G_TYPE_INT, samplerate,
+      "channels", G_TYPE_INT, channels, "parsed", G_TYPE_BOOLEAN, TRUE, NULL);
+
+  return new;
+}
+
+static void
+gst_mp3parse_base_init (gpointer klass)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&mp3_sink_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&mp3_src_template));
+
+  GST_DEBUG_CATEGORY_INIT (mp3parse_debug, "mp3parse", 0, "MPEG Audio Parser");
+
+  gst_element_class_set_details (element_class, &mp3parse_details);
+}
+
+static void
+gst_mp3parse_class_init (GstMPEGAudioParseClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+
+  parent_class = g_type_class_peek_parent (klass);
+
+  gobject_class->set_property = gst_mp3parse_set_property;
+  gobject_class->get_property = gst_mp3parse_get_property;
+  gobject_class->dispose = gst_mp3parse_dispose;
+
+  g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_SKIP,
+      g_param_spec_int ("skip", "skip", "skip",
+          G_MININT, G_MAXINT, 0, G_PARAM_READWRITE));
+  g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_BIT_RATE,
+      g_param_spec_int ("bitrate", "Bitrate", "Bit Rate",
+          G_MININT, G_MAXINT, 0, G_PARAM_READABLE));
+
+  gstelement_class->change_state = gst_mp3parse_change_state;
+
+/* register tags */
+#define GST_TAG_CRC    "has-crc"
+#define GST_TAG_MODE     "channel-mode"
+
+  gst_tag_register (GST_TAG_CRC, GST_TAG_FLAG_META, G_TYPE_BOOLEAN,
+      "has crc", "Using CRC", NULL);
+  gst_tag_register (GST_TAG_MODE, GST_TAG_FLAG_ENCODED, G_TYPE_STRING,
+      "channel mode", "MPEG audio channel mode", NULL);
+
+  g_type_class_ref (GST_TYPE_MP3_CHANNEL_MODE);
+}
+
+static void
+gst_mp3parse_reset (GstMPEGAudioParse * mp3parse)
+{
+  mp3parse->skip = 0;
+  mp3parse->resyncing = TRUE;
+  mp3parse->next_ts = GST_CLOCK_TIME_NONE;
+  mp3parse->cur_offset = -1;
+
+  mp3parse->sync_offset = 0;
+  mp3parse->tracked_offset = 0;
+  mp3parse->pending_ts = GST_CLOCK_TIME_NONE;
+  mp3parse->pending_offset = -1;
+
+  gst_adapter_clear (mp3parse->adapter);
+
+  mp3parse->rate = mp3parse->channels = mp3parse->layer = -1;
+  mp3parse->version = 1;
+  mp3parse->max_bitreservoir = GST_CLOCK_TIME_NONE;
+
+  mp3parse->avg_bitrate = 0;
+  mp3parse->bitrate_sum = 0;
+  mp3parse->last_posted_bitrate = 0;
+  mp3parse->frame_count = 0;
+  mp3parse->sent_codec_tag = FALSE;
+
+  mp3parse->last_posted_crc = CRC_UNKNOWN;
+  mp3parse->last_posted_channel_mode = MP3_CHANNEL_MODE_UNKNOWN;
+
+  mp3parse->xing_flags = 0;
+  mp3parse->xing_bitrate = 0;
+  mp3parse->xing_frames = 0;
+  mp3parse->xing_total_time = 0;
+  mp3parse->xing_bytes = 0;
+  mp3parse->xing_vbr_scale = 0;
+  memset (mp3parse->xing_seek_table, 0, 100);
+  memset (mp3parse->xing_seek_table_inverse, 0, 256);
+
+  mp3parse->vbri_bitrate = 0;
+  mp3parse->vbri_frames = 0;
+  mp3parse->vbri_total_time = 0;
+  mp3parse->vbri_bytes = 0;
+  mp3parse->vbri_seek_points = 0;
+  g_free (mp3parse->vbri_seek_table);
+  mp3parse->vbri_seek_table = NULL;
+
+  if (mp3parse->seek_table) {
+    g_list_foreach (mp3parse->seek_table, (GFunc) mpeg_audio_seek_entry_free,
+        NULL);
+    g_list_free (mp3parse->seek_table);
+    mp3parse->seek_table = NULL;
+  }
+
+  g_mutex_lock (mp3parse->pending_seeks_lock);
+  if (mp3parse->pending_accurate_seeks) {
+    g_slist_foreach (mp3parse->pending_accurate_seeks, (GFunc) g_free, NULL);
+    g_slist_free (mp3parse->pending_accurate_seeks);
+    mp3parse->pending_accurate_seeks = NULL;
+  }
+  if (mp3parse->pending_nonaccurate_seeks) {
+    g_slist_foreach (mp3parse->pending_nonaccurate_seeks, (GFunc) g_free, NULL);
+    g_slist_free (mp3parse->pending_nonaccurate_seeks);
+    mp3parse->pending_nonaccurate_seeks = NULL;
+  }
+  g_mutex_unlock (mp3parse->pending_seeks_lock);
+
+  if (mp3parse->pending_segment) {
+    GstEvent **eventp = &mp3parse->pending_segment;
+
+    gst_event_replace (eventp, NULL);
+  }
+
+  mp3parse->exact_position = FALSE;
+  gst_segment_init (&mp3parse->segment, GST_FORMAT_TIME);
+}
+
+static void
+gst_mp3parse_init (GstMPEGAudioParse * mp3parse, GstMPEGAudioParseClass * klass)
+{
+  mp3parse->sinkpad =
+      gst_pad_new_from_static_template (&mp3_sink_template, "sink");
+  gst_pad_set_event_function (mp3parse->sinkpad, gst_mp3parse_sink_event);
+  gst_pad_set_chain_function (mp3parse->sinkpad, gst_mp3parse_chain);
+  gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->sinkpad);
+
+  mp3parse->srcpad =
+      gst_pad_new_from_static_template (&mp3_src_template, "src");
+  gst_pad_use_fixed_caps (mp3parse->srcpad);
+  gst_pad_set_event_function (mp3parse->srcpad, mp3parse_src_event);
+  gst_pad_set_query_function (mp3parse->srcpad, mp3parse_src_query);
+  gst_pad_set_query_type_function (mp3parse->srcpad, mp3parse_get_query_types);
+  gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->srcpad);
+
+  mp3parse->adapter = gst_adapter_new ();
+  mp3parse->pending_seeks_lock = g_mutex_new ();
+
+  gst_mp3parse_reset (mp3parse);
+}
+
+static void
+gst_mp3parse_dispose (GObject * object)
+{
+  GstMPEGAudioParse *mp3parse = GST_MP3PARSE (object);
+
+  gst_mp3parse_reset (mp3parse);
+
+  if (mp3parse->adapter) {
+    g_object_unref (mp3parse->adapter);
+    mp3parse->adapter = NULL;
+  }
+  g_mutex_free (mp3parse->pending_seeks_lock);
+  mp3parse->pending_seeks_lock = NULL;
+
+  g_list_foreach (mp3parse->pending_events, (GFunc) gst_mini_object_unref,
+      NULL);
+  g_list_free (mp3parse->pending_events);
+  mp3parse->pending_events = NULL;
+
+  G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static gboolean
+gst_mp3parse_sink_event (GstPad * pad, GstEvent * event)
+{
+  gboolean res = TRUE;
+  GstMPEGAudioParse *mp3parse;
+  GstEvent **eventp;
+
+  mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad));
+
+  switch (GST_EVENT_TYPE (event)) {
+    case GST_EVENT_NEWSEGMENT:
+    {
+      gdouble rate, applied_rate;
+      GstFormat format;
+      gint64 start, stop, pos;
+      gboolean update;
+      MPEGAudioPendingAccurateSeek *seek = NULL;
+      GSList *node;
+
+      gst_event_parse_new_segment_full (event, &update, &rate, &applied_rate,
+          &format, &start, &stop, &pos);
+
+      g_mutex_lock (mp3parse->pending_seeks_lock);
+      if (format == GST_FORMAT_BYTES && mp3parse->pending_accurate_seeks) {
+
+        for (node = mp3parse->pending_accurate_seeks; node; node = node->next) {
+          MPEGAudioPendingAccurateSeek *tmp = node->data;
+
+          if (tmp->upstream_start == pos) {
+            seek = tmp;
+            break;
+          }
+        }
+        if (seek) {
+          GstSegment *s = &seek->segment;
+
+          event =
+              gst_event_new_new_segment_full (FALSE, s->rate, s->applied_rate,
+              GST_FORMAT_TIME, s->start, s->stop, s->last_stop);
+
+          mp3parse->segment = seek->segment;
+
+          mp3parse->resyncing = FALSE;
+          mp3parse->cur_offset = pos;
+          mp3parse->next_ts = seek->timestamp_start;
+          mp3parse->pending_ts = GST_CLOCK_TIME_NONE;
+          mp3parse->tracked_offset = 0;
+          mp3parse->sync_offset = 0;
+
+          gst_event_parse_new_segment_full (event, &update, &rate,
+              &applied_rate, &format, &start, &stop, &pos);
+
+          GST_DEBUG_OBJECT (mp3parse,
+              "Pushing accurate newseg rate %g, applied rate %g, "
+              "format %d, start %" G_GINT64_FORMAT ", stop %" G_GINT64_FORMAT
+              ", pos %" G_GINT64_FORMAT, rate, applied_rate, format, start,
+              stop, pos);
+
+          g_free (seek);
+          mp3parse->pending_accurate_seeks =
+              g_slist_delete_link (mp3parse->pending_accurate_seeks, node);
+
+          g_mutex_unlock (mp3parse->pending_seeks_lock);
+          res = gst_pad_push_event (mp3parse->srcpad, event);
+
+          return res;
+        } else {
+          GST_WARNING_OBJECT (mp3parse,
+              "Accurate seek not possible, didn't get an appropiate upstream segment");
+        }
+      }
+      g_mutex_unlock (mp3parse->pending_seeks_lock);
+
+      mp3parse->exact_position = FALSE;
+
+      if (format == GST_FORMAT_BYTES) {
+        GstClockTime seg_start, seg_stop, seg_pos;
+
+        /* stop time is allowed to be open-ended, but not start & pos */
+        if (!mp3parse_bytepos_to_time (mp3parse, stop, &seg_stop, FALSE))
+          seg_stop = GST_CLOCK_TIME_NONE;
+        if (mp3parse_bytepos_to_time (mp3parse, start, &seg_start, FALSE) &&
+            mp3parse_bytepos_to_time (mp3parse, pos, &seg_pos, FALSE)) {
+          gst_event_unref (event);
+
+          /* search the pending nonaccurate seeks */
+          g_mutex_lock (mp3parse->pending_seeks_lock);
+          seek = NULL;
+          for (node = mp3parse->pending_nonaccurate_seeks; node;
+              node = node->next) {
+            MPEGAudioPendingAccurateSeek *tmp = node->data;
+
+            if (tmp->upstream_start == pos) {
+              seek = tmp;
+              break;
+            }
+          }
+
+          if (seek) {
+            if (seek->segment.stop == -1) {
+              /* corrent the segment end, because non-accurate seeks might make
+               * our streaming end earlier (see bug #603695) */
+              seg_stop = -1;
+            }
+            g_free (seek);
+            mp3parse->pending_nonaccurate_seeks =
+                g_slist_delete_link (mp3parse->pending_nonaccurate_seeks, node);
+          }
+          g_mutex_unlock (mp3parse->pending_seeks_lock);
+
+          event = gst_event_new_new_segment_full (update, rate, applied_rate,
+              GST_FORMAT_TIME, seg_start, seg_stop, seg_pos);
+          format = GST_FORMAT_TIME;
+          GST_DEBUG_OBJECT (mp3parse, "Converted incoming segment to TIME. "
+              "start = %" GST_TIME_FORMAT ", stop = %" GST_TIME_FORMAT
+              ", pos = %" GST_TIME_FORMAT, GST_TIME_ARGS (seg_start),
+              GST_TIME_ARGS (seg_stop), GST_TIME_ARGS (seg_pos));
+        }
+      }
+
+      if (format != GST_FORMAT_TIME) {
+        /* Unknown incoming segment format. Output a default open-ended 
+         * TIME segment */
+        gst_event_unref (event);
+        event = gst_event_new_new_segment_full (update, rate, applied_rate,
+            GST_FORMAT_TIME, 0, GST_CLOCK_TIME_NONE, 0);
+      }
+
+      mp3parse->resyncing = TRUE;
+      mp3parse->cur_offset = -1;
+      mp3parse->next_ts = GST_CLOCK_TIME_NONE;
+      mp3parse->pending_ts = GST_CLOCK_TIME_NONE;
+      mp3parse->tracked_offset = 0;
+      mp3parse->sync_offset = 0;
+      /* also clear leftover data if clearing so much state */
+      gst_adapter_clear (mp3parse->adapter);
+
+      gst_event_parse_new_segment_full (event, &update, &rate, &applied_rate,
+          &format, &start, &stop, &pos);
+      GST_DEBUG_OBJECT (mp3parse, "Pushing newseg rate %g, applied rate %g, "
+          "format %d, start %" G_GINT64_FORMAT ", stop %" G_GINT64_FORMAT
+          ", pos %" G_GINT64_FORMAT, rate, applied_rate, format, start, stop,
+          pos);
+
+      gst_segment_set_newsegment_full (&mp3parse->segment, update, rate,
+          applied_rate, format, start, stop, pos);
+
+      /* save the segment for later, right before we push a new buffer so that
+       * the caps are fixed and the next linked element can receive the segment. */
+      eventp = &mp3parse->pending_segment;
+      gst_event_replace (eventp, event);
+      gst_event_unref (event);
+      res = TRUE;
+      break;
+    }
+    case GST_EVENT_FLUSH_STOP:
+      /* Clear our adapter and set up for a new position */
+      gst_adapter_clear (mp3parse->adapter);
+      eventp = &mp3parse->pending_segment;
+      gst_event_replace (eventp, NULL);
+      res = gst_pad_push_event (mp3parse->srcpad, event);
+      break;
+    case GST_EVENT_EOS:
+      /* If we haven't processed any frames yet, then make sure we process
+         at least whatever's in our adapter */
+      if (mp3parse->frame_count == 0) {
+        gst_mp3parse_handle_data (mp3parse, TRUE);
+
+        /* If we STILL have zero frames processed, fire an error */
+        if (mp3parse->frame_count == 0) {
+          GST_ELEMENT_ERROR (mp3parse, STREAM, WRONG_TYPE,
+              ("No valid frames found before end of stream"), (NULL));
+        }
+      }
+      /* fall through */
+    default:
+      if (mp3parse->pending_segment &&
+          (GST_EVENT_TYPE (event) != GST_EVENT_EOS) &&
+          (GST_EVENT_TYPE (event) != GST_EVENT_FLUSH_START)) {
+        /* Cache all events except EOS and the ones above if we have
+         * a pending segment */
+        mp3parse->pending_events =
+            g_list_append (mp3parse->pending_events, event);
+      } else {
+        res = gst_pad_push_event (mp3parse->srcpad, event);
+      }
+      break;
+  }
+
+  gst_object_unref (mp3parse);
+
+  return res;
+}
+
+static void
+gst_mp3parse_add_index_entry (GstMPEGAudioParse * mp3parse, guint64 offset,
+    GstClockTime ts)
+{
+  MPEGAudioSeekEntry *entry, *last;
+
+  if (G_LIKELY (mp3parse->seek_table != NULL)) {
+    last = mp3parse->seek_table->data;
+
+    if (last->byte >= offset)
+      return;
+
+    if (GST_CLOCK_DIFF (last->timestamp, ts) < mp3parse->idx_interval)
+      return;
+  }
+
+  entry = mpeg_audio_seek_entry_new ();
+  entry->byte = offset;
+  entry->timestamp = ts;
+  mp3parse->seek_table = g_list_prepend (mp3parse->seek_table, entry);
+
+  GST_LOG_OBJECT (mp3parse, "Adding index entry %" GST_TIME_FORMAT " @ offset "
+      "0x%08" G_GINT64_MODIFIER "x", GST_TIME_ARGS (ts), offset);
+}
+
+/* Prepare a buffer of the indicated size, timestamp it and output */
+static GstFlowReturn
+gst_mp3parse_emit_frame (GstMPEGAudioParse * mp3parse, guint size,
+    guint mode, guint crc)
+{
+  GstBuffer *outbuf;
+  guint bitrate;
+  GstFlowReturn ret = GST_FLOW_OK;
+  GstClockTime push_start;
+  GstTagList *taglist;
+
+  outbuf = gst_adapter_take_buffer (mp3parse->adapter, size);
+
+  GST_BUFFER_DURATION (outbuf) =
+      gst_util_uint64_scale (GST_SECOND, mp3parse->spf, mp3parse->rate);
+
+  GST_BUFFER_OFFSET (outbuf) = mp3parse->cur_offset;
+
+  /* Check if we have a pending timestamp from an incoming buffer to apply
+   * here */
+  if (GST_CLOCK_TIME_IS_VALID (mp3parse->pending_ts)) {
+    if (mp3parse->tracked_offset >= mp3parse->pending_offset) {
+      /* If the incoming timestamp differs from our expected by more than 
+       * half a frame, then take it instead of our calculated timestamp.
+       * This avoids creating imperfect streams just because of 
+       * quantization in the container timestamping */
+      GstClockTimeDiff diff = mp3parse->next_ts - mp3parse->pending_ts;
+      GstClockTimeDiff thresh = GST_BUFFER_DURATION (outbuf) / 2;
+
+      if (diff < -thresh || diff > thresh) {
+        GST_DEBUG_OBJECT (mp3parse, "Updating next_ts from %" GST_TIME_FORMAT
+            " to pending ts %" GST_TIME_FORMAT
+            " at offset %" G_GINT64_FORMAT " (pending offset was %"
+            G_GINT64_FORMAT ")", GST_TIME_ARGS (mp3parse->next_ts),
+            GST_TIME_ARGS (mp3parse->pending_ts), mp3parse->tracked_offset,
+            mp3parse->pending_offset);
+        mp3parse->next_ts = mp3parse->pending_ts;
+      }
+      mp3parse->pending_ts = GST_CLOCK_TIME_NONE;
+    }
+  }
+
+  /* Decide what timestamp we're going to apply */
+  if (GST_CLOCK_TIME_IS_VALID (mp3parse->next_ts)) {
+    GST_BUFFER_TIMESTAMP (outbuf) = mp3parse->next_ts;
+  } else {
+    GstClockTime ts;
+
+    /* No timestamp yet, convert our offset to a timestamp if we can, or
+     * start at 0 */
+    if (mp3parse_bytepos_to_time (mp3parse, mp3parse->cur_offset, &ts, FALSE) &&
+        GST_CLOCK_TIME_IS_VALID (ts))
+      GST_BUFFER_TIMESTAMP (outbuf) = ts;
+    else {
+      GST_BUFFER_TIMESTAMP (outbuf) = 0;
+    }
+  }
+
+  if (GST_BUFFER_TIMESTAMP (outbuf) == 0)
+    mp3parse->exact_position = TRUE;
+
+  if (mp3parse->seekable &&
+      mp3parse->exact_position && GST_BUFFER_TIMESTAMP_IS_VALID (outbuf) &&
+      mp3parse->cur_offset != GST_BUFFER_OFFSET_NONE) {
+    gst_mp3parse_add_index_entry (mp3parse, mp3parse->cur_offset,
+        GST_BUFFER_TIMESTAMP (outbuf));
+  }
+
+  /* Update our byte offset tracking */
+  if (mp3parse->cur_offset != -1) {
+    mp3parse->cur_offset += size;
+  }
+  mp3parse->tracked_offset += size;
+
+  if (GST_BUFFER_TIMESTAMP_IS_VALID (outbuf))
+    mp3parse->next_ts =
+        GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf);
+
+  gst_buffer_set_caps (outbuf, GST_PAD_CAPS (mp3parse->srcpad));
+
+  /* Post a bitrate tag if we need to before pushing the buffer */
+  if (mp3parse->xing_bitrate != 0)
+    bitrate = mp3parse->xing_bitrate;
+  else if (mp3parse->vbri_bitrate != 0)
+    bitrate = mp3parse->vbri_bitrate;
+  else
+    bitrate = mp3parse->avg_bitrate;
+
+  /* we will create a taglist (if any of the parameters has changed)
+   * to add the tags that changed */
+  taglist = NULL;
+  if ((mp3parse->last_posted_bitrate / 10000) != (bitrate / 10000)) {
+    taglist = gst_tag_list_new ();
+    mp3parse->last_posted_bitrate = bitrate;
+    gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, GST_TAG_BITRATE,
+        mp3parse->last_posted_bitrate, NULL);
+
+    /* Post a new duration message if the average bitrate changes that much
+     * so applications can update their cached values
+     */
+    if ((mp3parse->xing_flags & XING_TOC_FLAG) == 0
+        && mp3parse->vbri_total_time == 0) {
+      gst_element_post_message (GST_ELEMENT (mp3parse),
+          gst_message_new_duration (GST_OBJECT (mp3parse), GST_FORMAT_TIME,
+              -1));
+    }
+  }
+
+  if (mp3parse->last_posted_crc != crc) {
+    gboolean using_crc;
+
+    if (!taglist) {
+      taglist = gst_tag_list_new ();
+    }
+    mp3parse->last_posted_crc = crc;
+    if (mp3parse->last_posted_crc == CRC_PROTECTED) {
+      using_crc = TRUE;
+    } else {
+      using_crc = FALSE;
+    }
+    gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, GST_TAG_CRC,
+        using_crc, NULL);
+  }
+
+  if (mp3parse->last_posted_channel_mode != mode) {
+    if (!taglist) {
+      taglist = gst_tag_list_new ();
+    }
+    mp3parse->last_posted_channel_mode = mode;
+
+    gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, GST_TAG_MODE,
+        gst_mp3_channel_mode_get_nick (mode), NULL);
+  }
+
+  /* if the taglist exists, we need to send it */
+  if (taglist) {
+    gst_element_found_tags_for_pad (GST_ELEMENT (mp3parse),
+        mp3parse->srcpad, taglist);
+  }
+
+  /* We start pushing 9 frames earlier (29 frames for MPEG2) than
+   * segment start to be able to decode the first frame we want.
+   * 9 (29) frames are the theoretical maximum of frames that contain
+   * data for the current frame (bit reservoir).
+   */
+  if (mp3parse->segment.start == 0) {
+    push_start = 0;
+  } else if (GST_CLOCK_TIME_IS_VALID (mp3parse->max_bitreservoir)) {
+    if (GST_CLOCK_TIME_IS_VALID (mp3parse->segment.start) &&
+        mp3parse->segment.start > mp3parse->max_bitreservoir)
+      push_start = mp3parse->segment.start - mp3parse->max_bitreservoir;
+    else
+      push_start = 0;
+  } else {
+    push_start = mp3parse->segment.start;
+  }
+
+  if (G_UNLIKELY ((GST_CLOCK_TIME_IS_VALID (push_start) &&
+              GST_BUFFER_TIMESTAMP_IS_VALID (outbuf) &&
+              GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf)
+              < push_start))) {
+    GST_DEBUG_OBJECT (mp3parse,
+        "Buffer before configured segment range %" GST_TIME_FORMAT
+        " to %" GST_TIME_FORMAT ", dropping, timestamp %"
+        GST_TIME_FORMAT " duration %" GST_TIME_FORMAT
+        ", offset 0x%08" G_GINT64_MODIFIER "x", GST_TIME_ARGS (push_start),
+        GST_TIME_ARGS (mp3parse->segment.stop),
+        GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
+        GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)),
+        GST_BUFFER_OFFSET (outbuf));
+
+    gst_buffer_unref (outbuf);
+    ret = GST_FLOW_OK;
+  } else if (G_UNLIKELY (GST_BUFFER_TIMESTAMP_IS_VALID (outbuf) &&
+          GST_CLOCK_TIME_IS_VALID (mp3parse->segment.stop) &&
+          GST_BUFFER_TIMESTAMP (outbuf) >=
+          mp3parse->segment.stop + GST_BUFFER_DURATION (outbuf))) {
+    /* Some mp3 streams have an offset in the timestamps, for which we have to
+     * push the frame *after* the end position in order for the decoder to be
+     * able to decode everything up until the segment.stop position.
+     * That is the reason of the calculated offset */
+    GST_DEBUG_OBJECT (mp3parse,
+        "Buffer after configured segment range %" GST_TIME_FORMAT " to %"
+        GST_TIME_FORMAT ", returning GST_FLOW_UNEXPECTED, timestamp %"
+        GST_TIME_FORMAT " duration %" GST_TIME_FORMAT ", offset 0x%08"
+        G_GINT64_MODIFIER "x", GST_TIME_ARGS (push_start),
+        GST_TIME_ARGS (mp3parse->segment.stop),
+        GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
+        GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)),
+        GST_BUFFER_OFFSET (outbuf));
+
+    gst_buffer_unref (outbuf);
+    ret = GST_FLOW_UNEXPECTED;
+  } else {
+    GST_DEBUG_OBJECT (mp3parse,
+        "pushing buffer of %d bytes, timestamp %" GST_TIME_FORMAT
+        ", offset 0x%08" G_GINT64_MODIFIER "x", size,
+        GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
+        GST_BUFFER_OFFSET (outbuf));
+    mp3parse->segment.last_stop = GST_BUFFER_TIMESTAMP (outbuf);
+    /* push any pending segment now */
+    if (mp3parse->pending_segment) {
+      gst_pad_push_event (mp3parse->srcpad, mp3parse->pending_segment);
+      mp3parse->pending_segment = NULL;
+    }
+    if (mp3parse->pending_events) {
+      GList *l;
+
+      for (l = mp3parse->pending_events; l != NULL; l = l->next) {
+        gst_pad_push_event (mp3parse->srcpad, GST_EVENT (l->data));
+      }
+      g_list_free (mp3parse->pending_events);
+      mp3parse->pending_events = NULL;
+    }
+
+    /* set discont if needed */
+    if (mp3parse->discont) {
+      GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
+      mp3parse->discont = FALSE;
+    }
+
+    ret = gst_pad_push (mp3parse->srcpad, outbuf);
+  }
+
+  return ret;
+}
+
+static void
+gst_mp3parse_handle_first_frame (GstMPEGAudioParse * mp3parse)
+{
+  GstTagList *taglist;
+  gchar *codec;
+  const guint32 xing_id = 0x58696e67;   /* 'Xing' in hex */
+  const guint32 info_id = 0x496e666f;   /* 'Info' in hex - found in LAME CBR files */
+  const guint32 vbri_id = 0x56425249;   /* 'VBRI' in hex */
+
+  gint offset;
+
+  guint64 avail;
+  gint64 upstream_total_bytes = 0;
+  guint32 read_id;
+  const guint8 *data;
+
+  /* Output codec tag */
+  if (!mp3parse->sent_codec_tag) {
+    if (mp3parse->layer == 3) {
+      codec = g_strdup_printf ("MPEG %d Audio, Layer %d (MP3)",
+          mp3parse->version, mp3parse->layer);
+    } else {
+      codec = g_strdup_printf ("MPEG %d Audio, Layer %d",
+          mp3parse->version, mp3parse->layer);
+    }
+
+    taglist = gst_tag_list_new ();
+    gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE,
+        GST_TAG_AUDIO_CODEC, codec, NULL);
+    gst_element_found_tags_for_pad (GST_ELEMENT (mp3parse),
+        mp3parse->srcpad, taglist);
+    g_free (codec);
+
+    mp3parse->sent_codec_tag = TRUE;
+  }
+  /* end setting the tag */
+
+  /* Check first frame for Xing info */
+  if (mp3parse->version == 1) { /* MPEG-1 file */
+    if (mp3parse->channels == 1)
+      offset = 0x11;
+    else
+      offset = 0x20;
+  } else {                      /* MPEG-2 header */
+    if (mp3parse->channels == 1)
+      offset = 0x09;
+    else
+      offset = 0x11;
+  }
+  /* Skip the 4 bytes of the MP3 header too */
+  offset += 4;
+
+  /* Check if we have enough data to read the Xing header */
+  avail = gst_adapter_available (mp3parse->adapter);
+
+  if (avail < offset + 8)
+    return;
+
+  data = gst_adapter_peek (mp3parse->adapter, offset + 8);
+  if (data == NULL)
+    return;
+  /* The header starts at the provided offset */
+  data += offset;
+
+  /* obtain real upstream total bytes */
+  mp3parse_total_bytes (mp3parse, &upstream_total_bytes);
+
+  read_id = GST_READ_UINT32_BE (data);
+  if (read_id == xing_id || read_id == info_id) {
+    guint32 xing_flags;
+    guint bytes_needed = offset + 8;
+    gint64 total_bytes;
+    GstClockTime total_time;
+
+    GST_DEBUG_OBJECT (mp3parse, "Found Xing header marker 0x%x", xing_id);
+
+    /* Read 4 base bytes of flags, big-endian */
+    xing_flags = GST_READ_UINT32_BE (data + 4);
+    if (xing_flags & XING_FRAMES_FLAG)
+      bytes_needed += 4;
+    if (xing_flags & XING_BYTES_FLAG)
+      bytes_needed += 4;
+    if (xing_flags & XING_TOC_FLAG)
+      bytes_needed += 100;
+    if (xing_flags & XING_VBR_SCALE_FLAG)
+      bytes_needed += 4;
+    if (avail < bytes_needed) {
+      GST_DEBUG_OBJECT (mp3parse,
+          "Not enough data to read Xing header (need %d)", bytes_needed);
+      return;
+    }
+
+    GST_DEBUG_OBJECT (mp3parse, "Reading Xing header");
+    mp3parse->xing_flags = xing_flags;
+    data = gst_adapter_peek (mp3parse->adapter, bytes_needed);
+    data += offset + 8;
+
+    if (xing_flags & XING_FRAMES_FLAG) {
+      mp3parse->xing_frames = GST_READ_UINT32_BE (data);
+      if (mp3parse->xing_frames == 0) {
+        GST_WARNING_OBJECT (mp3parse,
+            "Invalid number of frames in Xing header");
+        mp3parse->xing_flags &= ~XING_FRAMES_FLAG;
+      } else {
+        mp3parse->xing_total_time = gst_util_uint64_scale (GST_SECOND,
+            (guint64) (mp3parse->xing_frames) * (mp3parse->spf),
+            mp3parse->rate);
+      }
+
+      data += 4;
+    } else {
+      mp3parse->xing_frames = 0;
+      mp3parse->xing_total_time = 0;
+    }
+
+    if (xing_flags & XING_BYTES_FLAG) {
+      mp3parse->xing_bytes = GST_READ_UINT32_BE (data);
+      if (mp3parse->xing_bytes == 0) {
+        GST_WARNING_OBJECT (mp3parse, "Invalid number of bytes in Xing header");
+        mp3parse->xing_flags &= ~XING_BYTES_FLAG;
+      }
+
+      data += 4;
+    } else {
+      mp3parse->xing_bytes = 0;
+    }
+
+    /* If we know the upstream size and duration, compute the
+     * total bitrate, rounded up to the nearest kbit/sec */
+    if ((total_time = mp3parse->xing_total_time) &&
+        (total_bytes = mp3parse->xing_bytes)) {
+      mp3parse->xing_bitrate = gst_util_uint64_scale (total_bytes,
+          8 * GST_SECOND, total_time);
+      mp3parse->xing_bitrate += 500;
+      mp3parse->xing_bitrate -= mp3parse->xing_bitrate % 1000;
+    }
+
+    if (xing_flags & XING_TOC_FLAG) {
+      int i, percent = 0;
+      guchar *table = mp3parse->xing_seek_table;
+      guchar old = 0, new;
+      guint first;
+
+      first = data[0];
+      GST_DEBUG_OBJECT (mp3parse,
+          "Subtracting initial offset of %d bytes from Xing TOC", first);
+
+      /* xing seek table: percent time -> 1/256 bytepos */
+      for (i = 0; i < 100; i++) {
+        new = data[i] - first;
+        if (old > new) {
+          GST_WARNING_OBJECT (mp3parse, "Skipping broken Xing TOC");
+          mp3parse->xing_flags &= ~XING_TOC_FLAG;
+          goto skip_toc;
+        }
+        mp3parse->xing_seek_table[i] = old = new;
+      }
+
+      /* build inverse table: 1/256 bytepos -> 1/100 percent time */
+      for (i = 0; i < 256; i++) {
+        while (percent < 99 && table[percent + 1] <= i)
+          percent++;
+
+        if (table[percent] == i) {
+          mp3parse->xing_seek_table_inverse[i] = percent * 100;
+        } else if (table[percent] < i && percent < 99) {
+          gdouble fa, fb, fx;
+          gint a = percent, b = percent + 1;
+
+          fa = table[a];
+          fb = table[b];
+          fx = (b - a) / (fb - fa) * (i - fa) + a;
+          mp3parse->xing_seek_table_inverse[i] = (guint16) (fx * 100);
+        } else if (percent == 99) {
+          gdouble fa, fb, fx;
+          gint a = percent, b = 100;
+
+          fa = table[a];
+          fb = 256.0;
+          fx = (b - a) / (fb - fa) * (i - fa) + a;
+          mp3parse->xing_seek_table_inverse[i] = (guint16) (fx * 100);
+        }
+      }
+    skip_toc:
+      data += 100;
+    } else {
+      memset (mp3parse->xing_seek_table, 0, 100);
+      memset (mp3parse->xing_seek_table_inverse, 0, 256);
+    }
+
+    if (xing_flags & XING_VBR_SCALE_FLAG) {
+      mp3parse->xing_vbr_scale = GST_READ_UINT32_BE (data);
+    } else
+      mp3parse->xing_vbr_scale = 0;
+
+    GST_DEBUG_OBJECT (mp3parse, "Xing header reported %u frames, time %"
+        GST_TIME_FORMAT ", %u bytes, vbr scale %u", mp3parse->xing_frames,
+        GST_TIME_ARGS (mp3parse->xing_total_time), mp3parse->xing_bytes,
+        mp3parse->xing_vbr_scale);
+
+    /* check for truncated file */
+    if (upstream_total_bytes && mp3parse->xing_bytes &&
+        mp3parse->xing_bytes * 0.8 > upstream_total_bytes) {
+      GST_WARNING_OBJECT (mp3parse, "File appears to have been truncated; "
+          "invalidating Xing header duration and size");
+      mp3parse->xing_flags &= ~XING_BYTES_FLAG;
+      mp3parse->xing_flags &= ~XING_FRAMES_FLAG;
+    }
+  } else if (read_id == vbri_id) {
+    gint64 total_bytes, total_frames;
+    GstClockTime total_time;
+    guint16 nseek_points;
+
+    GST_DEBUG_OBJECT (mp3parse, "Found VBRI header marker 0x%x", vbri_id);
+    if (avail < offset + 26) {
+      GST_DEBUG_OBJECT (mp3parse,
+          "Not enough data to read VBRI header (need %d)", offset + 26);
+      return;
+    }
+
+    GST_DEBUG_OBJECT (mp3parse, "Reading VBRI header");
+    data = gst_adapter_peek (mp3parse->adapter, offset + 26);
+    data += offset + 4;
+
+    if (GST_READ_UINT16_BE (data) != 0x0001) {
+      GST_WARNING_OBJECT (mp3parse,
+          "Unsupported VBRI version 0x%x", GST_READ_UINT16_BE (data));
+      return;
+    }
+    data += 2;
+
+    /* Skip encoder delay */
+    data += 2;
+
+    /* Skip quality */
+    data += 2;
+
+    total_bytes = GST_READ_UINT32_BE (data);
+    if (total_bytes != 0)
+      mp3parse->vbri_bytes = total_bytes;
+    data += 4;
+
+    total_frames = GST_READ_UINT32_BE (data);
+    if (total_frames != 0) {
+      mp3parse->vbri_frames = total_frames;
+      mp3parse->vbri_total_time = gst_util_uint64_scale (GST_SECOND,
+          (guint64) (mp3parse->vbri_frames) * (mp3parse->spf), mp3parse->rate);
+    }
+    data += 4;
+
+    /* If we know the upstream size and duration, compute the 
+     * total bitrate, rounded up to the nearest kbit/sec */
+    if ((total_time = mp3parse->vbri_total_time) &&
+        (total_bytes = mp3parse->vbri_bytes)) {
+      mp3parse->vbri_bitrate = gst_util_uint64_scale (total_bytes,
+          8 * GST_SECOND, total_time);
+      mp3parse->vbri_bitrate += 500;
+      mp3parse->vbri_bitrate -= mp3parse->vbri_bitrate % 1000;
+    }
+
+    nseek_points = GST_READ_UINT16_BE (data);
+    data += 2;
+
+    if (nseek_points > 0) {
+      guint scale, seek_bytes, seek_frames;
+      gint i;
+
+      mp3parse->vbri_seek_points = nseek_points;
+
+      scale = GST_READ_UINT16_BE (data);
+      data += 2;
+
+      seek_bytes = GST_READ_UINT16_BE (data);
+      data += 2;
+
+      seek_frames = GST_READ_UINT16_BE (data);
+
+      if (scale == 0 || seek_bytes == 0 || seek_bytes > 4 || seek_frames == 0) {
+        GST_WARNING_OBJECT (mp3parse, "Unsupported VBRI seek table");
+        goto out_vbri;
+      }
+
+      if (avail < offset + 26 + nseek_points * seek_bytes) {
+        GST_WARNING_OBJECT (mp3parse,
+            "Not enough data to read VBRI seek table (need %d)",
+            offset + 26 + nseek_points * seek_bytes);
+        goto out_vbri;
+      }
+
+      if (seek_frames * nseek_points < total_frames - seek_frames ||
+          seek_frames * nseek_points > total_frames + seek_frames) {
+        GST_WARNING_OBJECT (mp3parse,
+            "VBRI seek table doesn't cover the complete file");
+        goto out_vbri;
+      }
+
+      data =
+          gst_adapter_peek (mp3parse->adapter,
+          offset + 26 + nseek_points * seek_bytes);
+      data += offset + 26;
+
+
+      /* VBRI seek table: frame/seek_frames -> byte */
+      mp3parse->vbri_seek_table = g_new (guint32, nseek_points);
+      if (seek_bytes == 4)
+        for (i = 0; i < nseek_points; i++) {
+          mp3parse->vbri_seek_table[i] = GST_READ_UINT32_BE (data) * scale;
+          data += 4;
+      } else if (seek_bytes == 3)
+        for (i = 0; i < nseek_points; i++) {
+          mp3parse->vbri_seek_table[i] = GST_READ_UINT24_BE (data) * scale;
+          data += 3;
+      } else if (seek_bytes == 2)
+        for (i = 0; i < nseek_points; i++) {
+          mp3parse->vbri_seek_table[i] = GST_READ_UINT16_BE (data) * scale;
+          data += 2;
+      } else                    /* seek_bytes == 1 */
+        for (i = 0; i < nseek_points; i++) {
+          mp3parse->vbri_seek_table[i] = GST_READ_UINT8 (data) * scale;
+          data += 1;
+        }
+    }
+  out_vbri:
+
+    GST_DEBUG_OBJECT (mp3parse, "VBRI header reported %u frames, time %"
+        GST_TIME_FORMAT ", bytes %u", mp3parse->vbri_frames,
+        GST_TIME_ARGS (mp3parse->vbri_total_time), mp3parse->vbri_bytes);
+
+    /* check for truncated file */
+    if (upstream_total_bytes && mp3parse->vbri_bytes &&
+        mp3parse->vbri_bytes * 0.8 > upstream_total_bytes) {
+      GST_WARNING_OBJECT (mp3parse, "File appears to have been truncated; "
+          "invalidating VBRI header duration and size");
+      mp3parse->vbri_valid = FALSE;
+    } else {
+      mp3parse->vbri_valid = TRUE;
+    }
+  } else {
+    GST_DEBUG_OBJECT (mp3parse,
+        "Xing, LAME or VBRI header not found in first frame");
+  }
+}
+
+static void
+gst_mp3parse_check_seekability (GstMPEGAudioParse * mp3parse)
+{
+  GstQuery *query;
+  gboolean seekable = FALSE;
+  gint64 start = -1, stop = -1;
+  guint idx_interval = 0;
+
+  query = gst_query_new_seeking (GST_FORMAT_BYTES);
+  if (!gst_pad_peer_query (mp3parse->sinkpad, query)) {
+    GST_DEBUG_OBJECT (mp3parse, "seeking query failed");
+    goto done;
+  }
+
+  gst_query_parse_seeking (query, NULL, &seekable, &start, &stop);
+
+  /* try harder to query upstream size if we didn't get it the first time */
+  if (seekable && stop == -1) {
+    GstFormat fmt = GST_FORMAT_BYTES;
+
+    GST_DEBUG_OBJECT (mp3parse, "doing duration query to fix up unset stop");
+    gst_pad_query_peer_duration (mp3parse->sinkpad, &fmt, &stop);
+  }
+
+  /* if upstream doesn't know the size, it's likely that it's not seekable in
+   * practice even if it technically may be seekable */
+  if (seekable && (start != 0 || stop <= start)) {
+    GST_DEBUG_OBJECT (mp3parse, "seekable but unknown start/stop -> disable");
+    seekable = FALSE;
+  }
+
+  /* let's not put every single frame into our index */
+  if (seekable) {
+    if (stop < 10 * 1024 * 1024)
+      idx_interval = 100;
+    else if (stop < 100 * 1024 * 1024)
+      idx_interval = 500;
+    else
+      idx_interval = 1000;
+  }
+
+done:
+
+  GST_INFO_OBJECT (mp3parse, "seekable: %d (%" G_GUINT64_FORMAT " - %"
+      G_GUINT64_FORMAT ")", seekable, start, stop);
+  mp3parse->seekable = seekable;
+
+  GST_INFO_OBJECT (mp3parse, "idx_interval: %ums", idx_interval);
+  mp3parse->idx_interval = idx_interval * GST_MSECOND;
+
+  gst_query_unref (query);
+}
+
+/* Flush some number of bytes and update tracked offsets */
+static void
+gst_mp3parse_flush_bytes (GstMPEGAudioParse * mp3parse, int bytes)
+{
+  gst_adapter_flush (mp3parse->adapter, bytes);
+  if (mp3parse->cur_offset != -1)
+    mp3parse->cur_offset += bytes;
+  mp3parse->tracked_offset += bytes;
+}
+
+/* Perform extended validation to check that subsequent headers match
+   the first header given here in important characteristics, to avoid
+   false sync. We look for a minimum of MIN_RESYNC_FRAMES consecutive
+   frames to match their major characteristics.
+
+   If at_eos is set to TRUE, we just check that we don't find any invalid
+   frames in whatever data is available, rather than requiring a full
+   MIN_RESYNC_FRAMES of data.
+
+   Returns TRUE if we've seen enough data to validate or reject the frame.
+   If TRUE is returned, then *valid contains TRUE if it validated, or false
+   if we decided it was false sync.
+ */
+static gboolean
+gst_mp3parse_validate_extended (GstMPEGAudioParse * mp3parse, guint32 header,
+    int bpf, gboolean at_eos, gboolean * valid)
+{
+  guint32 next_header;
+  const guint8 *data;
+  guint available;
+  int frames_found = 1;
+  int offset = bpf;
+
+  while (frames_found < MIN_RESYNC_FRAMES) {
+    /* Check if we have enough data for all these frames, plus the next
+       frame header. */
+    available = gst_adapter_available (mp3parse->adapter);
+    if (available < offset + 4) {
+      if (at_eos) {
+        /* Running out of data at EOS is fine; just accept it */
+        *valid = TRUE;
+        return TRUE;
+      } else {
+        return FALSE;
+      }
+    }
+
+    data = gst_adapter_peek (mp3parse->adapter, offset + 4);
+    next_header = GST_READ_UINT32_BE (data + offset);
+    GST_DEBUG_OBJECT (mp3parse, "At %d: header=%08X, header2=%08X, bpf=%d",
+        offset, (unsigned int) header, (unsigned int) next_header, bpf);
+
+/* mask the bits which are allowed to differ between frames */
+#define HDRMASK ~((0xF << 12)  /* bitrate */ | \
+                  (0x1 <<  9)  /* padding */ | \
+                  (0xf <<  4)  /* mode|mode extension */ | \
+                  (0xf))        /* copyright|emphasis */
+
+    if ((next_header & HDRMASK) != (header & HDRMASK)) {
+      /* If any of the unmasked bits don't match, then it's not valid */
+      GST_DEBUG_OBJECT (mp3parse, "next header doesn't match "
+          "(header=%08X (%08X), header2=%08X (%08X), bpf=%d)",
+          (guint) header, (guint) header & HDRMASK, (guint) next_header,
+          (guint) next_header & HDRMASK, bpf);
+      *valid = FALSE;
+      return TRUE;
+    } else if ((((next_header >> 12) & 0xf) == 0) ||
+        (((next_header >> 12) & 0xf) == 0xf)) {
+      /* The essential parts were the same, but the bitrate held an
+         invalid value - also reject */
+      GST_DEBUG_OBJECT (mp3parse, "next header invalid (bitrate)");
+      *valid = FALSE;
+      return TRUE;
+    }
+
+    bpf = mp3_type_frame_length_from_header (mp3parse, next_header,
+        NULL, NULL, NULL, NULL, NULL, NULL, NULL);
+
+    offset += bpf;
+    frames_found++;
+  }
+
+  *valid = TRUE;
+  return TRUE;
+}
+
+static GstFlowReturn
+gst_mp3parse_handle_data (GstMPEGAudioParse * mp3parse, gboolean at_eos)
+{
+  GstFlowReturn flow = GST_FLOW_OK;
+  const guchar *data;
+  guint32 header;
+  int bpf;
+  guint available;
+  guint bitrate, layer, rate, channels, version, mode, crc;
+  gboolean caps_change;
+
+  /* while we still have at least 4 bytes (for the header) available */
+  while (gst_adapter_available (mp3parse->adapter) >= 4) {
+    /* Get the header bytes, check if they're potentially valid */
+    data = gst_adapter_peek (mp3parse->adapter, 4);
+    header = GST_READ_UINT32_BE (data);
+
+    if (!head_check (mp3parse, header)) {
+      /* Not a valid MP3 header; we start looking forward byte-by-byte trying to
+         find a place to resync */
+      if (!mp3parse->resyncing)
+        mp3parse->sync_offset = mp3parse->tracked_offset;
+      mp3parse->resyncing = TRUE;
+      gst_mp3parse_flush_bytes (mp3parse, 1);
+      GST_DEBUG_OBJECT (mp3parse, "wrong header, skipping byte");
+      continue;
+    }
+
+    /* We have a potentially valid header.
+       If this is just a normal 'next frame', we go ahead and output it.
+
+       However, sometimes, we do additional validation to ensure we haven't
+       got false sync (common with mp3 due to the short sync word).
+       The additional validation requires that we find several consecutive mp3
+       frames with the same major parameters, or reach EOS with a smaller
+       number of valid-looking frames.
+
+       We do this if:
+       - This is the very first frame we've processed
+       - We're resyncing after a non-accurate seek, or after losing sync
+       due to invalid data.
+       - The format of the stream changes in a major way (number of channels,
+       sample rate, layer, or mpeg version).
+     */
+    available = gst_adapter_available (mp3parse->adapter);
+
+    if (G_UNLIKELY (mp3parse->resyncing &&
+            mp3parse->tracked_offset - mp3parse->sync_offset > 2 * 1024 * 1024))
+      goto sync_failure;
+
+    bpf = mp3_type_frame_length_from_header (mp3parse, header,
+        &version, &layer, &channels, &bitrate, &rate, &mode, &crc);
+    g_assert (bpf != 0);
+
+    if (channels != mp3parse->channels ||
+        rate != mp3parse->rate || layer != mp3parse->layer ||
+        version != mp3parse->version)
+      caps_change = TRUE;
+    else
+      caps_change = FALSE;
+
+    if (mp3parse->resyncing || caps_change) {
+      gboolean valid;
+      if (!gst_mp3parse_validate_extended (mp3parse, header, bpf, at_eos,
+              &valid)) {
+        /* Not enough data to validate; wait for more */
+        break;
+      }
+
+      if (!valid) {
+        /* Extended validation failed; we probably got false sync.
+           Continue searching from the next byte in the stream */
+        if (!mp3parse->resyncing)
+          mp3parse->sync_offset = mp3parse->tracked_offset;
+        mp3parse->resyncing = TRUE;
+        gst_mp3parse_flush_bytes (mp3parse, 1);
+        continue;
+      }
+    }
+
+    /* if we don't have the whole frame... */
+    if (available < bpf) {
+      GST_DEBUG_OBJECT (mp3parse, "insufficient data available, need "
+          "%d bytes, have %d", bpf, available);
+      break;
+    }
+
+    if (caps_change) {
+      GstCaps *caps;
+
+      caps = mp3_caps_create (version, layer, channels, rate);
+      gst_pad_set_caps (mp3parse->srcpad, caps);
+      gst_caps_unref (caps);
+
+      mp3parse->channels = channels;
+      mp3parse->rate = rate;
+
+      mp3parse->layer = layer;
+      mp3parse->version = version;
+
+      /* see http://www.codeproject.com/audio/MPEGAudioInfo.asp */
+      if (mp3parse->layer == 1)
+        mp3parse->spf = 384;
+      else if (mp3parse->layer == 2)
+        mp3parse->spf = 1152;
+      else if (mp3parse->version == 1) {
+        mp3parse->spf = 1152;
+      } else {
+        /* MPEG-2 or "2.5" */
+        mp3parse->spf = 576;
+      }
+
+      mp3parse->max_bitreservoir = gst_util_uint64_scale (GST_SECOND,
+          ((version == 1) ? 10 : 30) * mp3parse->spf, mp3parse->rate);
+    }
+
+    mp3parse->bit_rate = bitrate;
+
+    /* Check the first frame for a Xing header to get our total length */
+    if (mp3parse->frame_count == 0) {
+      /* For the first frame in the file, look for a Xing frame after 
+       * the header, and output a codec tag */
+      gst_mp3parse_handle_first_frame (mp3parse);
+
+      /* Check if we're seekable */
+      gst_mp3parse_check_seekability (mp3parse);
+    }
+
+    /* Update VBR stats */
+    mp3parse->bitrate_sum += mp3parse->bit_rate;
+    mp3parse->frame_count++;
+    /* Compute the average bitrate, rounded up to the nearest 1000 bits */
+    mp3parse->avg_bitrate =
+        (mp3parse->bitrate_sum / mp3parse->frame_count + 500);
+    mp3parse->avg_bitrate -= mp3parse->avg_bitrate % 1000;
+
+    if (!mp3parse->skip) {
+      mp3parse->resyncing = FALSE;
+      flow = gst_mp3parse_emit_frame (mp3parse, bpf, mode, crc);
+      if (GST_FLOW_IS_FATAL (flow))
+        break;
+    } else {
+      GST_DEBUG_OBJECT (mp3parse, "skipping buffer of %d bytes", bpf);
+      gst_mp3parse_flush_bytes (mp3parse, bpf);
+      mp3parse->skip--;
+    }
+  }
+
+  return flow;
+
+  /* ERRORS */
+sync_failure:
+  {
+    GST_ELEMENT_ERROR (mp3parse, STREAM, DECODE,
+        ("Failed to parse stream"), (NULL));
+    return GST_FLOW_ERROR;
+  }
+}
+
+static GstFlowReturn
+gst_mp3parse_chain (GstPad * pad, GstBuffer * buf)
+{
+  GstMPEGAudioParse *mp3parse;
+  GstClockTime timestamp;
+
+  mp3parse = GST_MP3PARSE (GST_PAD_PARENT (pad));
+
+  GST_LOG_OBJECT (mp3parse, "buffer of %d bytes", GST_BUFFER_SIZE (buf));
+
+  timestamp = GST_BUFFER_TIMESTAMP (buf);
+
+  mp3parse->discont |= GST_BUFFER_IS_DISCONT (buf);
+
+  /* If we don't yet have a next timestamp, save it and the incoming offset
+   * so we can apply it to the right outgoing buffer */
+  if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
+    gint64 avail = gst_adapter_available (mp3parse->adapter);
+
+    mp3parse->pending_ts = timestamp;
+    mp3parse->pending_offset = mp3parse->tracked_offset + avail;
+
+    /* If we have no data pending and the next timestamp is
+     * invalid we can use the upstream timestamp for the next frame.
+     *
+     * This will give us a timestamp if we're resyncing and upstream
+     * gave us -1 as offset. */
+    if (avail == 0 && !GST_CLOCK_TIME_IS_VALID (mp3parse->next_ts))
+      mp3parse->next_ts = timestamp;
+
+    GST_LOG_OBJECT (mp3parse, "Have pending ts %" GST_TIME_FORMAT
+        " to apply in %" G_GINT64_FORMAT " bytes (@ off %" G_GINT64_FORMAT ")",
+        GST_TIME_ARGS (mp3parse->pending_ts), avail, mp3parse->pending_offset);
+  }
+
+  /* Update the cur_offset we'll apply to outgoing buffers */
+  if (mp3parse->cur_offset == -1 && GST_BUFFER_OFFSET (buf) != -1)
+    mp3parse->cur_offset = GST_BUFFER_OFFSET (buf);
+
+  /* And add the data to the pool */
+  gst_adapter_push (mp3parse->adapter, buf);
+
+  return gst_mp3parse_handle_data (mp3parse, FALSE);
+}
+
+static gboolean
+head_check (GstMPEGAudioParse * mp3parse, unsigned long head)
+{
+  GST_DEBUG_OBJECT (mp3parse, "checking mp3 header 0x%08lx", head);
+  /* if it's not a valid sync */
+  if ((head & 0xffe00000) != 0xffe00000) {
+    GST_WARNING_OBJECT (mp3parse, "invalid sync");
+    return FALSE;
+  }
+  /* if it's an invalid MPEG version */
+  if (((head >> 19) & 3) == 0x1) {
+    GST_WARNING_OBJECT (mp3parse, "invalid MPEG version: 0x%lx",
+        (head >> 19) & 3);
+    return FALSE;
+  }
+  /* if it's an invalid layer */
+  if (!((head >> 17) & 3)) {
+    GST_WARNING_OBJECT (mp3parse, "invalid layer: 0x%lx", (head >> 17) & 3);
+    return FALSE;
+  }
+  /* if it's an invalid bitrate */
+  if (((head >> 12) & 0xf) == 0x0) {
+    GST_WARNING_OBJECT (mp3parse, "invalid bitrate: 0x%lx."
+        "Free format files are not supported yet", (head >> 12) & 0xf);
+    return FALSE;
+  }
+  if (((head >> 12) & 0xf) == 0xf) {
+    GST_WARNING_OBJECT (mp3parse, "invalid bitrate: 0x%lx", (head >> 12) & 0xf);
+    return FALSE;
+  }
+  /* if it's an invalid samplerate */
+  if (((head >> 10) & 0x3) == 0x3) {
+    GST_WARNING_OBJECT (mp3parse, "invalid samplerate: 0x%lx",
+        (head >> 10) & 0x3);
+    return FALSE;
+  }
+
+  if ((head & 0x3) == 0x2) {
+    /* Ignore this as there are some files with emphasis 0x2 that can
+     * be played fine. See BGO #537235 */
+    GST_WARNING_OBJECT (mp3parse, "invalid emphasis: 0x%lx", head & 0x3);
+  }
+
+  return TRUE;
+}
+
+static void
+gst_mp3parse_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstMPEGAudioParse *src;
+
+  src = GST_MP3PARSE (object);
+
+  switch (prop_id) {
+    case ARG_SKIP:
+      src->skip = g_value_get_int (value);
+      break;
+    default:
+      break;
+  }
+}
+
+static void
+gst_mp3parse_get_property (GObject * object, guint prop_id, GValue * value,
+    GParamSpec * pspec)
+{
+  GstMPEGAudioParse *src;
+
+  src = GST_MP3PARSE (object);
+
+  switch (prop_id) {
+    case ARG_SKIP:
+      g_value_set_int (value, src->skip);
+      break;
+    case ARG_BIT_RATE:
+      g_value_set_int (value, src->bit_rate * 1000);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static GstStateChangeReturn
+gst_mp3parse_change_state (GstElement * element, GstStateChange transition)
+{
+  GstMPEGAudioParse *mp3parse;
+  GstStateChangeReturn result;
+
+  mp3parse = GST_MP3PARSE (element);
+
+  result = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+
+  switch (transition) {
+    case GST_STATE_CHANGE_PAUSED_TO_READY:
+      gst_mp3parse_reset (mp3parse);
+      break;
+    default:
+      break;
+  }
+
+  return result;
+}
+
+static gboolean
+mp3parse_total_bytes (GstMPEGAudioParse * mp3parse, gint64 * total)
+{
+  GstFormat fmt = GST_FORMAT_BYTES;
+
+  if (gst_pad_query_peer_duration (mp3parse->sinkpad, &fmt, total))
+    return TRUE;
+
+  if (mp3parse->xing_flags & XING_BYTES_FLAG) {
+    *total = mp3parse->xing_bytes;
+    return TRUE;
+  }
+
+  if (mp3parse->vbri_bytes != 0 && mp3parse->vbri_valid) {
+    *total = mp3parse->vbri_bytes;
+    return TRUE;
+  }
+
+  return FALSE;
+}
+
+static gboolean
+mp3parse_total_time (GstMPEGAudioParse * mp3parse, GstClockTime * total)
+{
+  gint64 total_bytes;
+
+  *total = GST_CLOCK_TIME_NONE;
+
+  if (mp3parse->xing_flags & XING_FRAMES_FLAG) {
+    *total = mp3parse->xing_total_time;
+    return TRUE;
+  }
+
+  if (mp3parse->vbri_total_time != 0 && mp3parse->vbri_valid) {
+    *total = mp3parse->vbri_total_time;
+    return TRUE;
+  }
+
+  /* Calculate time from the measured bitrate */
+  if (!mp3parse_total_bytes (mp3parse, &total_bytes))
+    return FALSE;
+
+  if (total_bytes != -1
+      && !mp3parse_bytepos_to_time (mp3parse, total_bytes, total, TRUE))
+    return FALSE;
+
+  return TRUE;
+}
+
+/* Convert a timestamp to the file position required to start decoding that
+ * timestamp. For now, this just uses the avg bitrate. Later, use an 
+ * incrementally accumulated seek table */
+static gboolean
+mp3parse_time_to_bytepos (GstMPEGAudioParse * mp3parse, GstClockTime ts,
+    gint64 * bytepos)
+{
+  gint64 total_bytes;
+  GstClockTime total_time;
+
+  /* -1 always maps to -1 */
+  if (ts == -1) {
+    *bytepos = -1;
+    return TRUE;
+  }
+
+  /* If XING seek table exists use this for time->byte conversion */
+  if ((mp3parse->xing_flags & XING_TOC_FLAG) &&
+      (total_bytes = mp3parse->xing_bytes) &&
+      (total_time = mp3parse->xing_total_time)) {
+    gdouble fa, fb, fx;
+    gdouble percent =
+        CLAMP ((100.0 * gst_util_guint64_to_gdouble (ts)) /
+        gst_util_guint64_to_gdouble (total_time), 0.0, 100.0);
+    gint index = CLAMP (percent, 0, 99);
+
+    fa = mp3parse->xing_seek_table[index];
+    if (index < 99)
+      fb = mp3parse->xing_seek_table[index + 1];
+    else
+      fb = 256.0;
+
+    fx = fa + (fb - fa) * (percent - index);
+
+    *bytepos = (1.0 / 256.0) * fx * total_bytes;
+
+    return TRUE;
+  }
+
+  if (mp3parse->vbri_seek_table && (total_bytes = mp3parse->vbri_bytes) &&
+      (total_time = mp3parse->vbri_total_time)) {
+    gint i, j;
+    gdouble a, b, fa, fb;
+
+    i = gst_util_uint64_scale (ts, mp3parse->vbri_seek_points - 1, total_time);
+    i = CLAMP (i, 0, mp3parse->vbri_seek_points - 1);
+
+    a = gst_guint64_to_gdouble (gst_util_uint64_scale (i, total_time,
+            mp3parse->vbri_seek_points));
+    fa = 0.0;
+    for (j = i; j >= 0; j--)
+      fa += mp3parse->vbri_seek_table[j];
+
+    if (i + 1 < mp3parse->vbri_seek_points) {
+      b = gst_guint64_to_gdouble (gst_util_uint64_scale (i + 1, total_time,
+              mp3parse->vbri_seek_points));
+      fb = fa + mp3parse->vbri_seek_table[i + 1];
+    } else {
+      b = gst_guint64_to_gdouble (total_time);
+      fb = total_bytes;
+    }
+
+    *bytepos = fa + ((fb - fa) / (b - a)) * (gst_guint64_to_gdouble (ts) - a);
+
+    return TRUE;
+  }
+
+  if (mp3parse->avg_bitrate == 0)
+    goto no_bitrate;
+
+  *bytepos =
+      gst_util_uint64_scale (ts, mp3parse->avg_bitrate, (8 * GST_SECOND));
+  return TRUE;
+no_bitrate:
+  GST_DEBUG_OBJECT (mp3parse, "Cannot seek yet - no average bitrate");
+  return FALSE;
+}
+
+static gboolean
+mp3parse_bytepos_to_time (GstMPEGAudioParse * mp3parse,
+    gint64 bytepos, GstClockTime * ts, gboolean from_total_time)
+{
+  gint64 total_bytes;
+  GstClockTime total_time;
+
+  if (bytepos == -1) {
+    *ts = GST_CLOCK_TIME_NONE;
+    return TRUE;
+  }
+
+  if (bytepos == 0) {
+    *ts = 0;
+    return TRUE;
+  }
+
+  /* If XING seek table exists use this for byte->time conversion */
+  if (!from_total_time && (mp3parse->xing_flags & XING_TOC_FLAG) &&
+      (total_bytes = mp3parse->xing_bytes) &&
+      (total_time = mp3parse->xing_total_time)) {
+    gdouble fa, fb, fx;
+    gdouble pos;
+    gint index;
+
+    pos = CLAMP ((bytepos * 256.0) / total_bytes, 0.0, 256.0);
+    index = CLAMP (pos, 0, 255);
+    fa = mp3parse->xing_seek_table_inverse[index];
+    if (index < 255)
+      fb = mp3parse->xing_seek_table_inverse[index + 1];
+    else
+      fb = 10000.0;
+
+    fx = fa + (fb - fa) * (pos - index);
+
+    *ts = (1.0 / 10000.0) * fx * gst_util_guint64_to_gdouble (total_time);
+
+    return TRUE;
+  }
+
+  if (!from_total_time && mp3parse->vbri_seek_table &&
+      (total_bytes = mp3parse->vbri_bytes) &&
+      (total_time = mp3parse->vbri_total_time)) {
+    gint i = 0;
+    guint64 sum = 0;
+    gdouble a, b, fa, fb;
+
+    do {
+      sum += mp3parse->vbri_seek_table[i];
+      i++;
+    } while (i + 1 < mp3parse->vbri_seek_points
+        && sum + mp3parse->vbri_seek_table[i] < bytepos);
+    i--;
+
+    a = gst_guint64_to_gdouble (sum);
+    fa = gst_guint64_to_gdouble (gst_util_uint64_scale (i, total_time,
+            mp3parse->vbri_seek_points));
+
+    if (i + 1 < mp3parse->vbri_seek_points) {
+      b = a + mp3parse->vbri_seek_table[i + 1];
+      fb = gst_guint64_to_gdouble (gst_util_uint64_scale (i + 1, total_time,
+              mp3parse->vbri_seek_points));
+    } else {
+      b = total_bytes;
+      fb = gst_guint64_to_gdouble (total_time);
+    }
+
+    *ts = gst_gdouble_to_guint64 (fa + ((fb - fa) / (b - a)) * (bytepos - a));
+
+    return TRUE;
+  }
+
+  /* Cannot convert anything except 0 if we don't have a bitrate yet */
+  if (mp3parse->avg_bitrate == 0)
+    return FALSE;
+
+  *ts = (GstClockTime) gst_util_uint64_scale (GST_SECOND, bytepos * 8,
+      mp3parse->avg_bitrate);
+  return TRUE;
+}
+
+static gboolean
+mp3parse_handle_seek (GstMPEGAudioParse * mp3parse, GstEvent * event)
+{
+  GstFormat format;
+  gdouble rate;
+  GstSeekFlags flags;
+  GstSeekType cur_type, stop_type;
+  gint64 cur, stop;
+  gint64 byte_cur, byte_stop;
+  MPEGAudioPendingAccurateSeek *seek;
+  GstClockTime start;
+
+  gst_event_parse_seek (event, &rate, &format, &flags, &cur_type, &cur,
+      &stop_type, &stop);
+
+  GST_DEBUG_OBJECT (mp3parse, "Performing seek to %" GST_TIME_FORMAT,
+      GST_TIME_ARGS (cur));
+
+  /* For any format other than TIME, see if upstream handles
+   * it directly or fail. For TIME, try upstream, but do it ourselves if
+   * it fails upstream */
+  if (format != GST_FORMAT_TIME) {
+    gst_event_ref (event);
+    return gst_pad_push_event (mp3parse->sinkpad, event);
+  } else {
+    gst_event_ref (event);
+    if (gst_pad_push_event (mp3parse->sinkpad, event))
+      return TRUE;
+  }
+
+  seek = g_new0 (MPEGAudioPendingAccurateSeek, 1);
+
+  seek->segment = mp3parse->segment;
+
+  gst_segment_set_seek (&seek->segment, rate, GST_FORMAT_TIME,
+      flags, cur_type, cur, stop_type, stop, NULL);
+
+  /* Handle TIME based seeks by converting to a BYTE position */
+
+  /* For accurate seeking get the frame 9 (MPEG1) or 29 (MPEG2) frames
+   * before the one we want to seek to and push them all to the decoder.
+   *
+   * This is necessary because of the bit reservoir. See
+   * http://www.mars.org/mailman/public/mad-dev/2002-May/000634.html
+   *
+   */
+
+  if (flags & GST_SEEK_FLAG_ACCURATE) {
+    if (!mp3parse->seek_table) {
+      byte_cur = 0;
+      byte_stop = -1;
+      start = 0;
+    } else {
+      MPEGAudioSeekEntry *entry = NULL, *start_entry = NULL, *stop_entry = NULL;
+      GList *start_node, *stop_node;
+      gint64 seek_ts = (cur > mp3parse->max_bitreservoir) ?
+          (cur - mp3parse->max_bitreservoir) : 0;
+
+      for (start_node = mp3parse->seek_table; start_node;
+          start_node = start_node->next) {
+        entry = start_node->data;
+
+        if (seek_ts >= entry->timestamp) {
+          start_entry = entry;
+          break;
+        }
+      }
+
+      if (!start_entry) {
+        start_entry = mp3parse->seek_table->data;
+        start = start_entry->timestamp;
+        byte_cur = start_entry->byte;
+      } else {
+        start = start_entry->timestamp;
+        byte_cur = start_entry->byte;
+      }
+
+      for (stop_node = mp3parse->seek_table; stop_node;
+          stop_node = stop_node->next) {
+        entry = stop_node->data;
+
+        if (stop >= entry->timestamp) {
+          stop_node = stop_node->prev;
+          stop_entry = (stop_node) ? stop_node->data : NULL;
+          break;
+        }
+      }
+
+      if (!stop_entry) {
+        byte_stop = -1;
+      } else {
+        byte_stop = stop_entry->byte;
+      }
+
+    }
+    event = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, cur_type,
+        byte_cur, stop_type, byte_stop);
+    g_mutex_lock (mp3parse->pending_seeks_lock);
+    seek->upstream_start = byte_cur;
+    seek->timestamp_start = start;
+    mp3parse->pending_accurate_seeks =
+        g_slist_prepend (mp3parse->pending_accurate_seeks, seek);
+    g_mutex_unlock (mp3parse->pending_seeks_lock);
+    if (gst_pad_push_event (mp3parse->sinkpad, event)) {
+      mp3parse->exact_position = TRUE;
+      return TRUE;
+    } else {
+      mp3parse->exact_position = TRUE;
+      g_mutex_lock (mp3parse->pending_seeks_lock);
+      mp3parse->pending_accurate_seeks =
+          g_slist_remove (mp3parse->pending_accurate_seeks, seek);
+      g_mutex_unlock (mp3parse->pending_seeks_lock);
+      g_free (seek);
+      return FALSE;
+    }
+  }
+
+  mp3parse->exact_position = FALSE;
+
+  /* Convert the TIME to the appropriate BYTE position at which to resume
+   * decoding. */
+  if (!mp3parse_time_to_bytepos (mp3parse, (GstClockTime) cur, &byte_cur))
+    goto no_pos;
+  if (!mp3parse_time_to_bytepos (mp3parse, (GstClockTime) stop, &byte_stop))
+    goto no_pos;
+
+  GST_DEBUG_OBJECT (mp3parse, "Seeking to byte range %" G_GINT64_FORMAT
+      " to %" G_GINT64_FORMAT, byte_cur, byte_stop);
+
+  /* Send BYTE based seek upstream */
+  event = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, cur_type,
+      byte_cur, stop_type, byte_stop);
+
+  GST_LOG_OBJECT (mp3parse, "Storing pending seek");
+  g_mutex_lock (mp3parse->pending_seeks_lock);
+  seek->upstream_start = byte_cur;
+  seek->timestamp_start = cur;
+  mp3parse->pending_nonaccurate_seeks =
+      g_slist_prepend (mp3parse->pending_nonaccurate_seeks, seek);
+  g_mutex_unlock (mp3parse->pending_seeks_lock);
+  if (gst_pad_push_event (mp3parse->sinkpad, event)) {
+    return TRUE;
+  } else {
+    g_mutex_lock (mp3parse->pending_seeks_lock);
+    mp3parse->pending_nonaccurate_seeks =
+        g_slist_remove (mp3parse->pending_nonaccurate_seeks, seek);
+    g_mutex_unlock (mp3parse->pending_seeks_lock);
+    g_free (seek);
+    return FALSE;
+  }
+
+no_pos:
+  GST_DEBUG_OBJECT (mp3parse,
+      "Could not determine byte position for desired time");
+  return FALSE;
+}
+
+static gboolean
+mp3parse_src_event (GstPad * pad, GstEvent * event)
+{
+  GstMPEGAudioParse *mp3parse;
+  gboolean res = FALSE;
+
+  mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad));
+
+  switch (GST_EVENT_TYPE (event)) {
+    case GST_EVENT_SEEK:
+      res = mp3parse_handle_seek (mp3parse, event);
+      gst_event_unref (event);
+      break;
+    default:
+      res = gst_pad_event_default (pad, event);
+      break;
+  }
+
+  gst_object_unref (mp3parse);
+  return res;
+}
+
+static gboolean
+mp3parse_src_query (GstPad * pad, GstQuery * query)
+{
+  GstFormat format;
+  GstClockTime total;
+  GstMPEGAudioParse *mp3parse;
+  gboolean res = FALSE;
+  GstPad *peer;
+
+  mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad));
+
+  GST_LOG_OBJECT (pad, "%s query", GST_QUERY_TYPE_NAME (query));
+
+  switch (GST_QUERY_TYPE (query)) {
+    case GST_QUERY_POSITION:
+      gst_query_parse_position (query, &format, NULL);
+
+      if (format == GST_FORMAT_BYTES || format == GST_FORMAT_DEFAULT) {
+        if (mp3parse->cur_offset != -1) {
+          gst_query_set_position (query, GST_FORMAT_BYTES,
+              mp3parse->cur_offset);
+          res = TRUE;
+        }
+      } else if (format == GST_FORMAT_TIME) {
+        if (mp3parse->next_ts == GST_CLOCK_TIME_NONE)
+          goto out;
+        gst_query_set_position (query, GST_FORMAT_TIME, mp3parse->next_ts);
+        res = TRUE;
+      }
+
+      /* If no answer above, see if upstream knows */
+      if (!res) {
+        if ((peer = gst_pad_get_peer (mp3parse->sinkpad)) != NULL) {
+          res = gst_pad_query (peer, query);
+          gst_object_unref (peer);
+          if (res)
+            goto out;
+        }
+      }
+      break;
+    case GST_QUERY_DURATION:
+      gst_query_parse_duration (query, &format, NULL);
+
+      /* First, see if upstream knows */
+      if ((peer = gst_pad_get_peer (mp3parse->sinkpad)) != NULL) {
+        res = gst_pad_query (peer, query);
+        gst_object_unref (peer);
+        if (res)
+          goto out;
+      }
+
+      if (format == GST_FORMAT_TIME) {
+        if (!mp3parse_total_time (mp3parse, &total) || total == -1)
+          goto out;
+        gst_query_set_duration (query, format, total);
+        res = TRUE;
+      }
+      break;
+    case GST_QUERY_SEEKING:
+      gst_query_parse_seeking (query, &format, NULL, NULL, NULL);
+
+      /* does upstream handle ? */
+      if ((peer = gst_pad_get_peer (mp3parse->sinkpad)) != NULL) {
+        res = gst_pad_query (peer, query);
+        gst_object_unref (peer);
+      }
+      /* we may be able to help if in TIME */
+      if (format == GST_FORMAT_TIME) {
+        gboolean seekable;
+
+        gst_query_parse_seeking (query, &format, &seekable, NULL, NULL);
+        /* already OK if upstream takes care */
+        if (!(res && seekable)) {
+          gint64 pos;
+
+          seekable = TRUE;
+          if (!mp3parse_total_time (mp3parse, &total) || total == -1) {
+            seekable = FALSE;
+          } else if (!mp3parse_time_to_bytepos (mp3parse, 0, &pos)) {
+            seekable = FALSE;
+          } else {
+            GstQuery *q;
+
+            q = gst_query_new_seeking (GST_FORMAT_BYTES);
+            if (!gst_pad_peer_query (mp3parse->sinkpad, q)) {
+              seekable = FALSE;
+            } else {
+              gst_query_parse_seeking (q, &format, &seekable, NULL, NULL);
+            }
+            gst_query_unref (q);
+          }
+          gst_query_set_seeking (query, GST_FORMAT_TIME, seekable, 0, total);
+          res = TRUE;
+        }
+      }
+      break;
+    default:
+      res = gst_pad_query_default (pad, query);
+      break;
+  }
+
+out:
+  gst_object_unref (mp3parse);
+  return res;
+}
+
+static const GstQueryType *
+mp3parse_get_query_types (GstPad * pad G_GNUC_UNUSED)
+{
+  static const GstQueryType query_types[] = {
+    GST_QUERY_POSITION,
+    GST_QUERY_DURATION,
+    0
+  };
+
+  return query_types;
+}