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diff --git a/gst/audioparsers/gstmpegaudioparse.c b/gst/audioparsers/gstmpegaudioparse.c
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+++ b/gst/audioparsers/gstmpegaudioparse.c
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+/* GStreamer MPEG audio parser
+ * Copyright (C) 2006-2007 Jan Schmidt <thaytan@mad.scientist.com>
+ * Copyright (C) 2010 Mark Nauwelaerts <mnauw users sf net>
+ * Copyright (C) 2010 Nokia Corporation. All rights reserved.
+ *   Contact: Stefan Kost <stefan.kost@nokia.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.
+ */
+/**
+ * SECTION:element-mpegaudioparse
+ * @short_description: MPEG audio parser
+ * @see_also: #GstAmrParse, #GstAACParse
+ *
+ * Parses and frames mpeg1 audio streams. Provides seeking.
+ *
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * gst-launch filesrc location=test.mp3 ! mpegaudioparse ! mad ! autoaudiosink
+ * ]|
+ * </refsect2>
+ */
+
+/* FIXME: we should make the base class (GstBaseParse) aware of the
+ * XING seek table somehow, so it can use it properly for things like
+ * accurate seeks. Currently it can only do a lookup via the convert function,
+ * but then doesn't know what the result represents exactly. One could either
+ * add a vfunc for index lookup, or just make mpegaudioparse populate the
+ * base class's index via the API provided.
+ */
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <string.h>
+
+#include "gstmpegaudioparse.h"
+#include <gst/base/gstbytereader.h>
+
+GST_DEBUG_CATEGORY_STATIC (mpeg_audio_parse_debug);
+#define GST_CAT_DEFAULT mpeg_audio_parse_debug
+
+#define MPEG_AUDIO_CHANNEL_MODE_UNKNOWN -1
+#define MPEG_AUDIO_CHANNEL_MODE_STEREO 0
+#define MPEG_AUDIO_CHANNEL_MODE_JOINT_STEREO 1
+#define MPEG_AUDIO_CHANNEL_MODE_DUAL_CHANNEL 2
+#define MPEG_AUDIO_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
+
+static GstStaticPadTemplate 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 sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/mpeg, mpegversion = (int) 1, parsed=(boolean)false")
+    );
+
+static void gst_mpeg_audio_parse_finalize (GObject * object);
+
+static gboolean gst_mpeg_audio_parse_start (GstBaseParse * parse);
+static gboolean gst_mpeg_audio_parse_stop (GstBaseParse * parse);
+static gboolean gst_mpeg_audio_parse_check_valid_frame (GstBaseParse * parse,
+    GstBaseParseFrame * frame, guint * size, gint * skipsize);
+static GstFlowReturn gst_mpeg_audio_parse_parse_frame (GstBaseParse * parse,
+    GstBaseParseFrame * frame);
+static GstFlowReturn gst_mpeg_audio_parse_pre_push_frame (GstBaseParse * parse,
+    GstBaseParseFrame * frame);
+static gboolean gst_mpeg_audio_parse_convert (GstBaseParse * parse,
+    GstFormat src_format, gint64 src_value,
+    GstFormat dest_format, gint64 * dest_value);
+
+GST_BOILERPLATE (GstMpegAudioParse, gst_mpeg_audio_parse, GstBaseParse,
+    GST_TYPE_BASE_PARSE);
+
+#define GST_TYPE_MPEG_AUDIO_CHANNEL_MODE  \
+    (gst_mpeg_audio_channel_mode_get_type())
+
+static const GEnumValue mpeg_audio_channel_mode[] = {
+  {MPEG_AUDIO_CHANNEL_MODE_UNKNOWN, "Unknown", "unknown"},
+  {MPEG_AUDIO_CHANNEL_MODE_MONO, "Mono", "mono"},
+  {MPEG_AUDIO_CHANNEL_MODE_DUAL_CHANNEL, "Dual Channel", "dual-channel"},
+  {MPEG_AUDIO_CHANNEL_MODE_JOINT_STEREO, "Joint Stereo", "joint-stereo"},
+  {MPEG_AUDIO_CHANNEL_MODE_STEREO, "Stereo", "stereo"},
+  {0, NULL, NULL},
+};
+
+static GType
+gst_mpeg_audio_channel_mode_get_type (void)
+{
+  static GType mpeg_audio_channel_mode_type = 0;
+
+  if (!mpeg_audio_channel_mode_type) {
+    mpeg_audio_channel_mode_type =
+        g_enum_register_static ("GstMpegAudioChannelMode",
+        mpeg_audio_channel_mode);
+  }
+  return mpeg_audio_channel_mode_type;
+}
+
+static const gchar *
+gst_mpeg_audio_channel_mode_get_nick (gint mode)
+{
+  guint i;
+  for (i = 0; i < G_N_ELEMENTS (mpeg_audio_channel_mode); i++) {
+    if (mpeg_audio_channel_mode[i].value == mode)
+      return mpeg_audio_channel_mode[i].value_nick;
+  }
+  return NULL;
+}
+
+static void
+gst_mpeg_audio_parse_base_init (gpointer klass)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&sink_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&src_template));
+
+  gst_element_class_set_details_simple (element_class, "MPEG1 Audio Parser",
+      "Codec/Parser/Audio",
+      "Parses and frames mpeg1 audio streams (levels 1-3), provides seek",
+      "Jan Schmidt <thaytan@mad.scientist.com>,"
+      "Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>");
+}
+
+static void
+gst_mpeg_audio_parse_class_init (GstMpegAudioParseClass * klass)
+{
+  GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
+  GObjectClass *object_class = G_OBJECT_CLASS (klass);
+
+  GST_DEBUG_CATEGORY_INIT (mpeg_audio_parse_debug, "mpegaudioparse", 0,
+      "MPEG1 audio stream parser");
+
+  object_class->finalize = gst_mpeg_audio_parse_finalize;
+
+  parse_class->start = GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_start);
+  parse_class->stop = GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_stop);
+  parse_class->check_valid_frame =
+      GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_check_valid_frame);
+  parse_class->parse_frame =
+      GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_parse_frame);
+  parse_class->pre_push_frame =
+      GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_pre_push_frame);
+  parse_class->convert = GST_DEBUG_FUNCPTR (gst_mpeg_audio_parse_convert);
+
+  /* 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_MPEG_AUDIO_CHANNEL_MODE);
+}
+
+static void
+gst_mpeg_audio_parse_reset (GstMpegAudioParse * mp3parse)
+{
+  mp3parse->channels = -1;
+  mp3parse->rate = -1;
+  mp3parse->sent_codec_tag = FALSE;
+  mp3parse->last_posted_crc = CRC_UNKNOWN;
+  mp3parse->last_posted_channel_mode = MPEG_AUDIO_CHANNEL_MODE_UNKNOWN;
+
+  mp3parse->hdr_bitrate = 0;
+
+  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;
+
+  mp3parse->encoder_delay = 0;
+  mp3parse->encoder_padding = 0;
+}
+
+static void
+gst_mpeg_audio_parse_init (GstMpegAudioParse * mp3parse,
+    GstMpegAudioParseClass * klass)
+{
+  gst_mpeg_audio_parse_reset (mp3parse);
+}
+
+static void
+gst_mpeg_audio_parse_finalize (GObject * object)
+{
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static gboolean
+gst_mpeg_audio_parse_start (GstBaseParse * parse)
+{
+  GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
+
+  gst_base_parse_set_min_frame_size (GST_BASE_PARSE (mp3parse), 1024);
+  GST_DEBUG_OBJECT (parse, "starting");
+
+  gst_mpeg_audio_parse_reset (mp3parse);
+
+  return TRUE;
+}
+
+static gboolean
+gst_mpeg_audio_parse_stop (GstBaseParse * parse)
+{
+  GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
+
+  GST_DEBUG_OBJECT (parse, "stopping");
+
+  gst_mpeg_audio_parse_reset (mp3parse);
+
+  return TRUE;
+}
+
+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_mpeg_audio_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;
+}
+
+/* Minimum number of consecutive, valid-looking frames to consider
+ * for resyncing */
+#define MIN_RESYNC_FRAMES 3
+
+/* 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.
+ * If FALSE is returned, then *valid contains minimum needed data.
+ */
+static gboolean
+gst_mp3parse_validate_extended (GstMpegAudioParse * mp3parse, GstBuffer * buf,
+    guint32 header, int bpf, gboolean at_eos, gint * valid)
+{
+  guint32 next_header;
+  const guint8 *data;
+  guint available;
+  int frames_found = 1;
+  int offset = bpf;
+
+  available = GST_BUFFER_SIZE (buf);
+  data = GST_BUFFER_DATA (buf);
+
+  while (frames_found < MIN_RESYNC_FRAMES) {
+    /* Check if we have enough data for all these frames, plus the next
+       frame header. */
+    if (available < offset + 4) {
+      if (at_eos) {
+        /* Running out of data at EOS is fine; just accept it */
+        *valid = TRUE;
+        return TRUE;
+      } else {
+        *valid = offset + 4;
+        return FALSE;
+      }
+    }
+
+    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 gboolean
+gst_mpeg_audio_parse_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 gboolean
+gst_mpeg_audio_parse_check_valid_frame (GstBaseParse * parse,
+    GstBaseParseFrame * frame, guint * framesize, gint * skipsize)
+{
+  GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
+  GstBuffer *buf = frame->buffer;
+  GstByteReader reader = GST_BYTE_READER_INIT_FROM_BUFFER (buf);
+  gint off, bpf;
+  gboolean lost_sync, draining, valid, caps_change;
+  guint32 header;
+  guint bitrate, layer, rate, channels, version, mode, crc;
+
+  if (G_UNLIKELY (GST_BUFFER_SIZE (buf) < 6))
+    return FALSE;
+
+  off = gst_byte_reader_masked_scan_uint32 (&reader, 0xffe00000, 0xffe00000,
+      0, GST_BUFFER_SIZE (buf));
+
+  GST_LOG_OBJECT (parse, "possible sync at buffer offset %d", off);
+
+  /* didn't find anything that looks like a sync word, skip */
+  if (off < 0) {
+    *skipsize = GST_BUFFER_SIZE (buf) - 3;
+    return FALSE;
+  }
+
+  /* possible frame header, but not at offset 0? skip bytes before sync */
+  if (off > 0) {
+    *skipsize = off;
+    return FALSE;
+  }
+
+  /* make sure the values in the frame header look sane */
+  header = GST_READ_UINT32_BE (GST_BUFFER_DATA (buf));
+  if (!gst_mpeg_audio_parse_head_check (mp3parse, header)) {
+    *skipsize = 1;
+    return FALSE;
+  }
+
+  GST_LOG_OBJECT (parse, "got frame");
+
+  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;
+
+  lost_sync = GST_BASE_PARSE_LOST_SYNC (parse);
+  draining = GST_BASE_PARSE_DRAINING (parse);
+
+  if (!draining && (lost_sync || caps_change)) {
+    if (!gst_mp3parse_validate_extended (mp3parse, buf, header, bpf, draining,
+            &valid)) {
+      /* not enough data */
+      gst_base_parse_set_min_frame_size (parse, valid);
+      *skipsize = 0;
+      return FALSE;
+    } else {
+      if (!valid) {
+        *skipsize = off + 2;
+        return FALSE;
+      }
+    }
+  } else if (draining && lost_sync && caps_change && mp3parse->rate > 0) {
+    /* avoid caps jitter that we can't be sure of */
+    *skipsize = off + 2;
+    return FALSE;
+  }
+
+  *framesize = bpf;
+  return TRUE;
+}
+
+static void
+gst_mpeg_audio_parse_handle_first_frame (GstMpegAudioParse * mp3parse,
+    GstBuffer * buf)
+{
+  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 */
+  const guint32 lame_id = 0x4c414d45;   /* 'LAME' in hex */
+  gint offset_xing, offset_vbri;
+  guint64 avail;
+  gint64 upstream_total_bytes = 0;
+  GstFormat fmt = GST_FORMAT_BYTES;
+  guint32 read_id_xing = 0, read_id_vbri = 0;
+  const guint8 *data;
+  guint bitrate;
+
+  if (mp3parse->sent_codec_tag)
+    return;
+
+  /* Check first frame for Xing info */
+  if (mp3parse->version == 1) { /* MPEG-1 file */
+    if (mp3parse->channels == 1)
+      offset_xing = 0x11;
+    else
+      offset_xing = 0x20;
+  } else {                      /* MPEG-2 header */
+    if (mp3parse->channels == 1)
+      offset_xing = 0x09;
+    else
+      offset_xing = 0x11;
+  }
+
+  /* The VBRI tag is always at offset 0x20 */
+  offset_vbri = 0x20;
+
+  /* Skip the 4 bytes of the MP3 header too */
+  offset_xing += 4;
+  offset_vbri += 4;
+
+  /* Check if we have enough data to read the Xing header */
+  avail = GST_BUFFER_SIZE (buf);
+  data = GST_BUFFER_DATA (buf);
+
+  if (avail >= offset_xing + 4) {
+    read_id_xing = GST_READ_UINT32_BE (data + offset_xing);
+  }
+  if (avail >= offset_vbri + 4) {
+    read_id_vbri = GST_READ_UINT32_BE (data + offset_vbri);
+  }
+
+  /* obtain real upstream total bytes */
+  fmt = GST_FORMAT_BYTES;
+  if (!gst_pad_query_peer_duration (GST_BASE_PARSE_SINK_PAD (GST_BASE_PARSE
+              (mp3parse)), &fmt, &upstream_total_bytes))
+    upstream_total_bytes = 0;
+
+  if (read_id_xing == xing_id || read_id_xing == info_id) {
+    guint32 xing_flags;
+    guint bytes_needed = offset_xing + 8;
+    gint64 total_bytes;
+    GstClockTime total_time;
+
+    GST_DEBUG_OBJECT (mp3parse, "Found Xing header marker 0x%x", xing_id);
+
+    /* Move data after Xing header */
+    data += offset_xing + 4;
+
+    /* Read 4 base bytes of flags, big-endian */
+    xing_flags = GST_READ_UINT32_BE (data);
+    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;
+
+    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);
+      data += 4;
+    } 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;
+    }
+
+    /* Optional LAME tag? */
+    if (avail - bytes_needed >= 36 && GST_READ_UINT32_BE (data) == lame_id) {
+      gchar lame_version[10] = { 0, };
+      guint tag_rev;
+      guint32 encoder_delay, encoder_padding;
+
+      memcpy (lame_version, data, 9);
+      data += 9;
+      tag_rev = data[0] >> 4;
+      GST_DEBUG_OBJECT (mp3parse, "Found LAME tag revision %d created by '%s'",
+          tag_rev, lame_version);
+
+      /* Skip all the information we're not interested in */
+      data += 12;
+      /* Encoder delay and end padding */
+      encoder_delay = GST_READ_UINT24_BE (data);
+      encoder_delay >>= 12;
+      encoder_padding = GST_READ_UINT24_BE (data);
+      encoder_padding &= 0x000fff;
+
+      mp3parse->encoder_delay = encoder_delay;
+      mp3parse->encoder_padding = encoder_padding;
+
+      GST_DEBUG_OBJECT (mp3parse, "Encoder delay %u, encoder padding %u",
+          encoder_delay, encoder_padding);
+    }
+  }
+
+  if (read_id_vbri == 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_vbri + 26) {
+      GST_DEBUG_OBJECT (mp3parse,
+          "Not enough data to read VBRI header (need %d)", offset_vbri + 26);
+      return;
+    }
+
+    GST_DEBUG_OBJECT (mp3parse, "Reading VBRI header");
+
+    /* Move data after VBRI header */
+    data += offset_vbri + 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_vbri + 26 + nseek_points * seek_bytes) {
+        GST_WARNING_OBJECT (mp3parse,
+            "Not enough data to read VBRI seek table (need %d)",
+            offset_vbri + 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;
+      }
+
+      if (avail < offset_vbri + 26) {
+        GST_DEBUG_OBJECT (mp3parse,
+            "Not enough data to read VBRI header (need %d)",
+            offset_vbri + 26 + nseek_points * seek_bytes);
+        return;
+      }
+
+      data = GST_BUFFER_DATA (buf);
+      data += offset_vbri + 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");
+  }
+
+  /* set duration if tables provided a valid one */
+  if (mp3parse->xing_flags & XING_FRAMES_FLAG) {
+    gst_base_parse_set_duration (GST_BASE_PARSE (mp3parse), GST_FORMAT_TIME,
+        mp3parse->xing_total_time, 0);
+  }
+  if (mp3parse->vbri_total_time != 0 && mp3parse->vbri_valid) {
+    gst_base_parse_set_duration (GST_BASE_PARSE (mp3parse), GST_FORMAT_TIME,
+        mp3parse->vbri_total_time, 0);
+  }
+
+  /* tell baseclass how nicely we can seek, and a bitrate if one found */
+  /* FIXME: fill index with seek table */
+#if 0
+  seekable = GST_BASE_PARSE_SEEK_DEFAULT;
+  if ((mp3parse->xing_flags & XING_TOC_FLAG) && mp3parse->xing_bytes &&
+      mp3parse->xing_total_time)
+    seekable = GST_BASE_PARSE_SEEK_TABLE;
+
+  if (mp3parse->vbri_seek_table && mp3parse->vbri_bytes &&
+      mp3parse->vbri_total_time)
+    seekable = GST_BASE_PARSE_SEEK_TABLE;
+#endif
+
+  if (mp3parse->xing_bitrate)
+    bitrate = mp3parse->xing_bitrate;
+  else if (mp3parse->vbri_bitrate)
+    bitrate = mp3parse->vbri_bitrate;
+  else
+    bitrate = 0;
+
+  gst_base_parse_set_average_bitrate (GST_BASE_PARSE (mp3parse), bitrate);
+}
+
+static GstFlowReturn
+gst_mpeg_audio_parse_parse_frame (GstBaseParse * parse,
+    GstBaseParseFrame * frame)
+{
+  GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
+  GstBuffer *buf = frame->buffer;
+  guint bitrate, layer, rate, channels, version, mode, crc;
+
+  g_return_val_if_fail (GST_BUFFER_SIZE (buf) >= 4, GST_FLOW_ERROR);
+
+  if (!mp3_type_frame_length_from_header (mp3parse,
+          GST_READ_UINT32_BE (GST_BUFFER_DATA (buf)),
+          &version, &layer, &channels, &bitrate, &rate, &mode, &crc))
+    goto broken_header;
+
+  if (G_UNLIKELY (channels != mp3parse->channels || rate != mp3parse->rate ||
+          layer != mp3parse->layer || version != mp3parse->version)) {
+    GstCaps *caps = 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, rate,
+        "channels", G_TYPE_INT, channels, "parsed", G_TYPE_BOOLEAN, TRUE, NULL);
+    gst_buffer_set_caps (buf, caps);
+    gst_pad_set_caps (GST_BASE_PARSE_SRC_PAD (parse), caps);
+    gst_caps_unref (caps);
+
+    mp3parse->rate = rate;
+    mp3parse->channels = channels;
+    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;
+    }
+
+    /* lead_in:
+     * 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).
+     *
+     * lead_out:
+     * 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. */
+    gst_base_parse_set_frame_rate (parse, mp3parse->rate, mp3parse->spf,
+        (version == 1) ? 10 : 30, 2);
+  }
+
+  mp3parse->hdr_bitrate = bitrate;
+
+  /* For first frame; check for seek tables and output a codec tag */
+  gst_mpeg_audio_parse_handle_first_frame (mp3parse, buf);
+
+  /* store some frame info for later processing */
+  mp3parse->last_crc = crc;
+  mp3parse->last_mode = mode;
+
+  return GST_FLOW_OK;
+
+/* ERRORS */
+broken_header:
+  {
+    /* this really shouldn't ever happen */
+    GST_ELEMENT_ERROR (parse, STREAM, DECODE, (NULL), (NULL));
+    return GST_FLOW_ERROR;
+  }
+}
+
+static gboolean
+gst_mpeg_audio_parse_time_to_bytepos (GstMpegAudioParse * mp3parse,
+    GstClockTime ts, gint64 * bytepos)
+{
+  gint64 total_bytes;
+  GstClockTime total_time;
+
+  /* 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;
+  }
+
+  return FALSE;
+}
+
+static gboolean
+gst_mpeg_audio_parse_bytepos_to_time (GstMpegAudioParse * mp3parse,
+    gint64 bytepos, GstClockTime * ts)
+{
+  gint64 total_bytes;
+  GstClockTime total_time;
+
+  /* If XING seek table exists use this for byte->time conversion */
+  if ((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 (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;
+  }
+
+  return FALSE;
+}
+
+static gboolean
+gst_mpeg_audio_parse_convert (GstBaseParse * parse, GstFormat src_format,
+    gint64 src_value, GstFormat dest_format, gint64 * dest_value)
+{
+  GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
+  gboolean res = FALSE;
+
+  if (src_format == GST_FORMAT_TIME && dest_format == GST_FORMAT_BYTES)
+    res =
+        gst_mpeg_audio_parse_time_to_bytepos (mp3parse, src_value, dest_value);
+  else if (src_format == GST_FORMAT_BYTES && dest_format == GST_FORMAT_TIME)
+    res = gst_mpeg_audio_parse_bytepos_to_time (mp3parse, src_value,
+        (GstClockTime *) dest_value);
+
+  /* if no tables, fall back to default estimated rate based conversion */
+  if (!res)
+    return gst_base_parse_convert_default (parse, src_format, src_value,
+        dest_format, dest_value);
+
+  return res;
+}
+
+static GstFlowReturn
+gst_mpeg_audio_parse_pre_push_frame (GstBaseParse * parse,
+    GstBaseParseFrame * frame)
+{
+  GstMpegAudioParse *mp3parse = GST_MPEG_AUDIO_PARSE (parse);
+  GstTagList *taglist;
+
+  /* tag sending done late enough in hook to ensure pending events
+   * have already been sent */
+
+  if (!mp3parse->sent_codec_tag) {
+    gchar *codec;
+
+    /* 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);
+    if (mp3parse->hdr_bitrate > 0 && mp3parse->xing_bitrate == 0 &&
+        mp3parse->vbri_bitrate == 0) {
+      /* We don't have a VBR bitrate, so post the available bitrate as
+       * nominal and let baseparse calculate the real bitrate */
+      gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE,
+          GST_TAG_NOMINAL_BITRATE, mp3parse->hdr_bitrate, NULL);
+    }
+    gst_element_found_tags_for_pad (GST_ELEMENT (mp3parse),
+        GST_BASE_PARSE_SRC_PAD (mp3parse), taglist);
+    g_free (codec);
+
+    /* also signals the end of first-frame processing */
+    mp3parse->sent_codec_tag = TRUE;
+  }
+
+  /* we will create a taglist (if any of the parameters has changed)
+   * to add the tags that changed */
+  taglist = NULL;
+  if (mp3parse->last_posted_crc != mp3parse->last_crc) {
+    gboolean using_crc;
+
+    if (!taglist) {
+      taglist = gst_tag_list_new ();
+    }
+    mp3parse->last_posted_crc = mp3parse->last_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 != mp3parse->last_mode) {
+    if (!taglist) {
+      taglist = gst_tag_list_new ();
+    }
+    mp3parse->last_posted_channel_mode = mp3parse->last_mode;
+
+    gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, GST_TAG_MODE,
+        gst_mpeg_audio_channel_mode_get_nick (mp3parse->last_mode), NULL);
+  }
+
+  /* if the taglist exists, we need to send it */
+  if (taglist) {
+    gst_element_found_tags_for_pad (GST_ELEMENT (mp3parse),
+        GST_BASE_PARSE_SRC_PAD (mp3parse), taglist);
+  }
+
+  /* usual clipping applies */
+  frame->flags |= GST_BASE_PARSE_FRAME_FLAG_CLIP;
+
+  return GST_FLOW_OK;
+}