Merge branch 'plugin-move-mpg123'

Move mpg123 plugin from -bad to -ugly.

https://bugzilla.gnome.org/show_bug.cgi?id=719849

4 files changed, 1241 insertions, 0 deletions

diff --git a/ext/mpg123/Makefile.am b/ext/mpg123/Makefile.am
new file mode 100644
index 0000000000..6c96207c86
--- /dev/null
+++ b/ext/mpg123/Makefile.am
@@ -0,0 +1,12 @@
+plugin_LTLIBRARIES = libgstmpg123.la
+
+libgstmpg123_la_SOURCES = gstmpg123audiodec.c
+libgstmpg123_la_CFLAGS = -DGST_USE_UNSTABLE_API \
+	$(GST_PLUGINS_BASE_CFLAGS) \
+	$(GST_BASE_CFLAGS) $(GST_CFLAGS) $(MPG123_CFLAGS)
+libgstmpg123_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-@GST_API_VERSION@ \
+	$(GST_BASE_LIBS) $(GST_LIBS) $(MPG123_LIBS)
+libgstmpg123_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+libgstmpg123_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
+
+noinst_HEADERS = gstmpg123audiodec.h
diff --git a/ext/mpg123/gstmpg123audiodec.c b/ext/mpg123/gstmpg123audiodec.c
new file mode 100644
index 0000000000..cfd017ea32
--- /dev/null
+++ b/ext/mpg123/gstmpg123audiodec.c
@@ -0,0 +1,633 @@
+/*  MP3 decoding plugin for GStreamer using the mpg123 library
+ *  Copyright (C) 2012 Carlos Rafael Giani
+ *
+ *  This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Lesser General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2.1 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
+ *  Lesser General Public License for more details.
+ *
+ *  You should have received a copy of the GNU Lesser General Public
+ *  License along with this library; if not, write to the Free Software
+ *  Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+ */
+
+/**
+ * SECTION: element-mpg123audiodec
+ * @see_also: lamemp3enc, mad
+ *
+ * Audio decoder for MPEG-1 layer 1/2/3 audio data.
+ *
+ * <refsect2>
+ * <title>Example pipelines</title>
+ * |[
+ * gst-launch-1.0 filesrc location=music.mp3 ! mpegaudioparse ! mpg123audiodec ! audioconvert ! audioresample ! autoaudiosink
+ * ]| Decode and play the mp3 file
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include "gstmpg123audiodec.h"
+
+#include <stdlib.h>
+#include <string.h>
+
+GST_DEBUG_CATEGORY_STATIC (mpg123_debug);
+#define GST_CAT_DEFAULT mpg123_debug
+
+/* Omitted sample formats that mpg123 supports (or at least can support):
+ *  - 8bit integer signed
+ *  - 8bit integer unsigned
+ *  - a-law
+ *  - mu-law
+ *  - 64bit float
+ *
+ * The first four formats are not supported by the GstAudioDecoder base class.
+ * (The internal gst_audio_format_from_caps_structure() call fails.)
+ *
+ * The 64bit float issue is tricky. mpg123 actually decodes to "real",
+ * not necessarily to "float".
+ *
+ * "real" can be fixed point, 32bit float, 64bit float. There seems to be
+ * no way how to find out which one of them is actually used.
+ *
+ * However, in all known installations, "real" equals 32bit float, so that's
+ * what is used. */
+
+static GstStaticPadTemplate static_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/mpeg, "
+        "mpegversion = (int) 1, "
+        "layer = (int) [ 1, 3 ], "
+        "rate = (int) { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }, "
+        "channels = (int) [ 1, 2 ], " "parsed = (boolean) true ")
+    );
+
+static gboolean gst_mpg123_audio_dec_start (GstAudioDecoder * dec);
+static gboolean gst_mpg123_audio_dec_stop (GstAudioDecoder * dec);
+static GstFlowReturn gst_mpg123_audio_dec_push_decoded_bytes (GstMpg123AudioDec
+    * mpg123_decoder, unsigned char const *decoded_bytes,
+    size_t const num_decoded_bytes);
+static GstFlowReturn gst_mpg123_audio_dec_handle_frame (GstAudioDecoder * dec,
+    GstBuffer * input_buffer);
+static gboolean gst_mpg123_audio_dec_set_format (GstAudioDecoder * dec,
+    GstCaps * input_caps);
+static void gst_mpg123_audio_dec_flush (GstAudioDecoder * dec, gboolean hard);
+
+G_DEFINE_TYPE (GstMpg123AudioDec, gst_mpg123_audio_dec, GST_TYPE_AUDIO_DECODER);
+
+static void
+gst_mpg123_audio_dec_class_init (GstMpg123AudioDecClass * klass)
+{
+  GstAudioDecoderClass *base_class;
+  GstElementClass *element_class;
+  GstPadTemplate *src_template, *sink_template;
+  int error;
+
+  GST_DEBUG_CATEGORY_INIT (mpg123_debug, "mpg123", 0, "mpg123 mp3 decoder");
+
+  base_class = GST_AUDIO_DECODER_CLASS (klass);
+  element_class = GST_ELEMENT_CLASS (klass);
+
+  gst_element_class_set_static_metadata (element_class,
+      "mpg123 mp3 decoder",
+      "Codec/Decoder/Audio",
+      "Decodes mp3 streams using the mpg123 library",
+      "Carlos Rafael Giani <dv@pseudoterminal.org>");
+
+  /* Not using static pad template for srccaps, since the comma-separated list
+   * of formats needs to be created depending on whatever mpg123 supports */
+  {
+    const int *format_list;
+    const long *rates_list;
+    size_t num, i;
+    GString *s;
+    GstCaps *src_template_caps;
+
+    s = g_string_new ("audio/x-raw, ");
+
+    mpg123_encodings (&format_list, &num);
+    g_string_append (s, "format = { ");
+    for (i = 0; i < num; ++i) {
+      switch (format_list[i]) {
+        case MPG123_ENC_SIGNED_16:
+          g_string_append (s, (i > 0) ? ", " : "");
+          g_string_append (s, GST_AUDIO_NE (S16));
+          break;
+        case MPG123_ENC_UNSIGNED_16:
+          g_string_append (s, (i > 0) ? ", " : "");
+          g_string_append (s, GST_AUDIO_NE (U16));
+          break;
+        case MPG123_ENC_SIGNED_24:
+          g_string_append (s, (i > 0) ? ", " : "");
+          g_string_append (s, GST_AUDIO_NE (S24));
+          break;
+        case MPG123_ENC_UNSIGNED_24:
+          g_string_append (s, (i > 0) ? ", " : "");
+          g_string_append (s, GST_AUDIO_NE (U24));
+          break;
+        case MPG123_ENC_SIGNED_32:
+          g_string_append (s, (i > 0) ? ", " : "");
+          g_string_append (s, GST_AUDIO_NE (S32));
+          break;
+        case MPG123_ENC_UNSIGNED_32:
+          g_string_append (s, (i > 0) ? ", " : "");
+          g_string_append (s, GST_AUDIO_NE (U32));
+          break;
+        case MPG123_ENC_FLOAT_32:
+          g_string_append (s, (i > 0) ? ", " : "");
+          g_string_append (s, GST_AUDIO_NE (F32));
+          break;
+        default:
+          GST_DEBUG ("Ignoring mpg123 format %d", format_list[i]);
+          break;
+      }
+    }
+    g_string_append (s, " }, ");
+
+    mpg123_rates (&rates_list, &num);
+    g_string_append (s, "rate = (int) { ");
+    for (i = 0; i < num; ++i) {
+      g_string_append_printf (s, "%s%lu", (i > 0) ? ", " : "", rates_list[i]);
+    }
+    g_string_append (s, "}, ");
+
+    g_string_append (s, "channels = (int) [ 1, 2 ], ");
+    g_string_append (s, "layout = (string) interleaved");
+
+    src_template_caps = gst_caps_from_string (s->str);
+    src_template = gst_pad_template_new ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
+        src_template_caps);
+
+    g_string_free (s, TRUE);
+  }
+
+  sink_template = gst_static_pad_template_get (&static_sink_template);
+
+  gst_element_class_add_pad_template (element_class, sink_template);
+  gst_element_class_add_pad_template (element_class, src_template);
+
+  base_class->start = GST_DEBUG_FUNCPTR (gst_mpg123_audio_dec_start);
+  base_class->stop = GST_DEBUG_FUNCPTR (gst_mpg123_audio_dec_stop);
+  base_class->handle_frame =
+      GST_DEBUG_FUNCPTR (gst_mpg123_audio_dec_handle_frame);
+  base_class->set_format = GST_DEBUG_FUNCPTR (gst_mpg123_audio_dec_set_format);
+  base_class->flush = GST_DEBUG_FUNCPTR (gst_mpg123_audio_dec_flush);
+
+  error = mpg123_init ();
+  if (G_UNLIKELY (error != MPG123_OK))
+    GST_ERROR ("Could not initialize mpg123 library: %s",
+        mpg123_plain_strerror (error));
+  else
+    GST_INFO ("mpg123 library initialized");
+}
+
+
+void
+gst_mpg123_audio_dec_init (GstMpg123AudioDec * mpg123_decoder)
+{
+  mpg123_decoder->handle = NULL;
+  gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (mpg123_decoder), TRUE);
+  gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
+      (mpg123_decoder), TRUE);
+  GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (mpg123_decoder));
+}
+
+
+static gboolean
+gst_mpg123_audio_dec_start (GstAudioDecoder * dec)
+{
+  GstMpg123AudioDec *mpg123_decoder;
+  int error;
+
+  mpg123_decoder = GST_MPG123_AUDIO_DEC (dec);
+  error = 0;
+
+  mpg123_decoder->handle = mpg123_new (NULL, &error);
+  mpg123_decoder->has_next_audioinfo = FALSE;
+  mpg123_decoder->frame_offset = 0;
+
+  /* Initially, the mpg123 handle comes with a set of default formats
+   * supported. This clears this set.  This is necessary, since only one
+   * format shall be supported (see set_format for more). */
+  mpg123_format_none (mpg123_decoder->handle);
+
+  /* Built-in mpg123 support for gapless decoding is disabled for now,
+   * since it does not work well with seeking */
+  mpg123_param (mpg123_decoder->handle, MPG123_REMOVE_FLAGS, MPG123_GAPLESS, 0);
+  /* Tells mpg123 to use a small read-ahead buffer for better MPEG sync;
+   * essential for MP3 radio streams */
+  mpg123_param (mpg123_decoder->handle, MPG123_ADD_FLAGS, MPG123_SEEKBUFFER, 0);
+  /* Sets the resync limit to the end of the stream (otherwise mpg123 may give
+   * up on decoding prematurely, especially with mp3 web radios) */
+  mpg123_param (mpg123_decoder->handle, MPG123_RESYNC_LIMIT, -1, 0);
+#if MPG123_API_VERSION >= 36
+  /* The precise API version where MPG123_AUTO_RESAMPLE appeared is
+   * somewhere between 29 and 36 */
+  /* Don't let mpg123 resample output */
+  mpg123_param (mpg123_decoder->handle, MPG123_REMOVE_FLAGS,
+      MPG123_AUTO_RESAMPLE, 0);
+#endif
+  /* Don't let mpg123 print messages to stdout/stderr */
+  mpg123_param (mpg123_decoder->handle, MPG123_ADD_FLAGS, MPG123_QUIET, 0);
+
+  /* Open in feed mode (= encoded data is fed manually into the handle). */
+  error = mpg123_open_feed (mpg123_decoder->handle);
+
+  if (G_UNLIKELY (error != MPG123_OK)) {
+    GST_ELEMENT_ERROR (dec, LIBRARY, INIT, (NULL),
+        ("%s", mpg123_strerror (mpg123_decoder->handle)));
+    mpg123_close (mpg123_decoder->handle);
+    mpg123_delete (mpg123_decoder->handle);
+    mpg123_decoder->handle = NULL;
+    return FALSE;
+  }
+
+  GST_INFO_OBJECT (dec, "mpg123 decoder started");
+
+  return TRUE;
+}
+
+
+static gboolean
+gst_mpg123_audio_dec_stop (GstAudioDecoder * dec)
+{
+  GstMpg123AudioDec *mpg123_decoder = GST_MPG123_AUDIO_DEC (dec);
+
+  if (G_LIKELY (mpg123_decoder->handle != NULL)) {
+    mpg123_close (mpg123_decoder->handle);
+    mpg123_delete (mpg123_decoder->handle);
+    mpg123_decoder->handle = NULL;
+  }
+
+  GST_INFO_OBJECT (dec, "mpg123 decoder stopped");
+
+  return TRUE;
+}
+
+
+static GstFlowReturn
+gst_mpg123_audio_dec_push_decoded_bytes (GstMpg123AudioDec * mpg123_decoder,
+    unsigned char const *decoded_bytes, size_t const num_decoded_bytes)
+{
+  GstBuffer *output_buffer;
+  GstAudioDecoder *dec;
+
+  output_buffer = NULL;
+  dec = GST_AUDIO_DECODER (mpg123_decoder);
+
+  if ((num_decoded_bytes == 0) || (decoded_bytes == NULL)) {
+    /* This occurs in the first few frames, which do not carry data; once
+     * MPG123_AUDIO_DEC_NEW_FORMAT is received, the empty frames stop occurring */
+    GST_DEBUG_OBJECT (mpg123_decoder,
+        "cannot decode yet, need more data -> no output buffer to push");
+    return GST_FLOW_OK;
+  }
+
+  output_buffer = gst_buffer_new_allocate (NULL, num_decoded_bytes, NULL);
+
+  if (output_buffer == NULL) {
+    /* This is necessary to advance playback in time,
+     * even when nothing was decoded. */
+    return gst_audio_decoder_finish_frame (dec, NULL, 1);
+  } else {
+    GstMapInfo info;
+
+    if (gst_buffer_map (output_buffer, &info, GST_MAP_WRITE)) {
+      memcpy (info.data, decoded_bytes, num_decoded_bytes);
+      gst_buffer_unmap (output_buffer, &info);
+    } else {
+      GST_ERROR_OBJECT (mpg123_decoder, "gst_buffer_map() returned NULL");
+      gst_buffer_unref (output_buffer);
+      output_buffer = NULL;
+    }
+
+    return gst_audio_decoder_finish_frame (dec, output_buffer, 1);
+  }
+}
+
+
+static GstFlowReturn
+gst_mpg123_audio_dec_handle_frame (GstAudioDecoder * dec,
+    GstBuffer * input_buffer)
+{
+  GstMpg123AudioDec *mpg123_decoder;
+  int decode_error;
+  unsigned char *decoded_bytes;
+  size_t num_decoded_bytes;
+  GstFlowReturn retval;
+
+  mpg123_decoder = GST_MPG123_AUDIO_DEC (dec);
+
+  g_assert (mpg123_decoder->handle != NULL);
+
+  /* The actual decoding */
+  {
+    /* feed input data (if there is any) */
+    if (G_LIKELY (input_buffer != NULL)) {
+      GstMapInfo info;
+
+      if (gst_buffer_map (input_buffer, &info, GST_MAP_READ)) {
+        mpg123_feed (mpg123_decoder->handle, info.data, info.size);
+        gst_buffer_unmap (input_buffer, &info);
+      } else {
+        GST_AUDIO_DECODER_ERROR (mpg123_decoder, 1, RESOURCE, READ, (NULL),
+            ("gst_memory_map() failed"), retval);
+        return retval;
+      }
+    }
+
+    /* Try to decode a frame */
+    decoded_bytes = NULL;
+    num_decoded_bytes = 0;
+    decode_error = mpg123_decode_frame (mpg123_decoder->handle,
+        &mpg123_decoder->frame_offset, &decoded_bytes, &num_decoded_bytes);
+  }
+
+  retval = GST_FLOW_OK;
+
+  switch (decode_error) {
+    case MPG123_NEW_FORMAT:
+      /* As mentioned in gst_mpg123_audio_dec_set_format(), the next audioinfo
+       * is not set immediately; instead, the code waits for mpg123 to take
+       * note of the new format, and then sets the audioinfo. This fixes glitches
+       * with mp3s containing several format headers (for example, first half
+       * using 44.1kHz, second half 32 kHz) */
+
+      GST_LOG_OBJECT (dec,
+          "mpg123 reported a new format -> setting next srccaps");
+
+      gst_mpg123_audio_dec_push_decoded_bytes (mpg123_decoder, decoded_bytes,
+          num_decoded_bytes);
+
+      /* If there is a next audioinfo, use it, then set has_next_audioinfo to
+       * FALSE, to make sure gst_audio_decoder_set_output_format() isn't called
+       * again until set_format is called by the base class */
+      if (mpg123_decoder->has_next_audioinfo) {
+        if (!gst_audio_decoder_set_output_format (dec,
+                &(mpg123_decoder->next_audioinfo))) {
+          GST_WARNING_OBJECT (dec, "Unable to set output format");
+          retval = GST_FLOW_NOT_NEGOTIATED;
+        }
+        mpg123_decoder->has_next_audioinfo = FALSE;
+      }
+
+      break;
+
+    case MPG123_NEED_MORE:
+    case MPG123_OK:
+      retval = gst_mpg123_audio_dec_push_decoded_bytes (mpg123_decoder,
+          decoded_bytes, num_decoded_bytes);
+      break;
+
+    case MPG123_DONE:
+      /* If this happens, then the upstream parser somehow missed the ending
+       * of the bitstream */
+      GST_LOG_OBJECT (dec, "mpg123 is done decoding");
+      gst_mpg123_audio_dec_push_decoded_bytes (mpg123_decoder, decoded_bytes,
+          num_decoded_bytes);
+      retval = GST_FLOW_EOS;
+      break;
+
+    default:
+    {
+      /* Anything else is considered an error */
+      int errcode;
+      retval = GST_FLOW_ERROR;  /* use error by default */
+      switch (decode_error) {
+        case MPG123_ERR:
+          errcode = mpg123_errcode (mpg123_decoder->handle);
+          break;
+        default:
+          errcode = decode_error;
+      }
+      switch (errcode) {
+        case MPG123_BAD_OUTFORMAT:{
+          GstCaps *input_caps =
+              gst_pad_get_current_caps (GST_AUDIO_DECODER_SINK_PAD (dec));
+          GST_ELEMENT_ERROR (dec, STREAM, FORMAT, (NULL),
+              ("Output sample format could not be used when trying to decode frame. "
+                  "This is typically caused when the input caps (often the sample "
+                  "rate) do not match the actual format of the audio data. "
+                  "Input caps: %" GST_PTR_FORMAT, input_caps)
+              );
+          gst_caps_unref (input_caps);
+          break;
+        }
+        default:{
+          char const *errmsg = mpg123_plain_strerror (errcode);
+          /* GST_AUDIO_DECODER_ERROR sets a new return value according to
+           * its estimations */
+          GST_AUDIO_DECODER_ERROR (mpg123_decoder, 1, STREAM, DECODE, (NULL),
+              ("mpg123 decoding error: %s", errmsg), retval);
+        }
+      }
+    }
+  }
+
+  return retval;
+}
+
+
+static gboolean
+gst_mpg123_audio_dec_set_format (GstAudioDecoder * dec, GstCaps * input_caps)
+{
+  /* "encoding" is the sample format specifier for mpg123 */
+  int encoding;
+  int sample_rate, num_channels;
+  GstAudioFormat format;
+  GstMpg123AudioDec *mpg123_decoder;
+  gboolean retval = FALSE;
+
+  mpg123_decoder = GST_MPG123_AUDIO_DEC (dec);
+
+  g_assert (mpg123_decoder->handle != NULL);
+
+  mpg123_decoder->has_next_audioinfo = FALSE;
+
+  /* Get sample rate and number of channels from input_caps */
+  {
+    GstStructure *structure;
+    gboolean err = FALSE;
+
+    /* Only the first structure is used (multiple
+     * input caps structures don't make sense */
+    structure = gst_caps_get_structure (input_caps, 0);
+
+    if (!gst_structure_get_int (structure, "rate", &sample_rate)) {
+      err = TRUE;
+      GST_ERROR_OBJECT (dec, "Input caps do not have a rate value");
+    }
+    if (!gst_structure_get_int (structure, "channels", &num_channels)) {
+      err = TRUE;
+      GST_ERROR_OBJECT (dec, "Input caps do not have a channel value");
+    }
+
+    if (G_UNLIKELY (err))
+      goto done;
+  }
+
+  /* Get sample format from the allowed src caps */
+  {
+    GstCaps *allowed_srccaps =
+        gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (dec));
+
+    if (allowed_srccaps == NULL) {
+      /* srcpad is not linked (yet), so no peer information is available;
+       * just use the default sample format (16 bit signed integer) */
+      GST_DEBUG_OBJECT (mpg123_decoder,
+          "srcpad is not linked (yet) -> using S16 sample format");
+      format = GST_AUDIO_FORMAT_S16;
+      encoding = MPG123_ENC_SIGNED_16;
+    } else if (gst_caps_is_empty (allowed_srccaps)) {
+      gst_caps_unref (allowed_srccaps);
+      goto done;
+    } else {
+      gchar const *format_str;
+      GValue const *format_value;
+
+      /* Look at the sample format values from the first structure */
+      GstStructure *structure = gst_caps_get_structure (allowed_srccaps, 0);
+      format_value = gst_structure_get_value (structure, "format");
+
+      if (format_value == NULL) {
+        gst_caps_unref (allowed_srccaps);
+        goto done;
+      } else if (GST_VALUE_HOLDS_LIST (format_value)) {
+        /* if value is a format list, pick the first entry */
+        GValue const *fmt_list_value =
+            gst_value_list_get_value (format_value, 0);
+        format_str = g_value_get_string (fmt_list_value);
+      } else if (G_VALUE_HOLDS_STRING (format_value)) {
+        /* if value is a string, use it directly */
+        format_str = g_value_get_string (format_value);
+      } else {
+        GST_ERROR_OBJECT (mpg123_decoder, "unexpected type for 'format' field "
+            "in caps structure %" GST_PTR_FORMAT, structure);
+        gst_caps_unref (allowed_srccaps);
+        goto done;
+      }
+
+      /* get the format value from the string */
+      format = gst_audio_format_from_string (format_str);
+      gst_caps_unref (allowed_srccaps);
+
+      g_assert (format != GST_AUDIO_FORMAT_UNKNOWN);
+
+      /* convert format to mpg123 encoding */
+      switch (format) {
+        case GST_AUDIO_FORMAT_S16:
+          encoding = MPG123_ENC_SIGNED_16;
+          break;
+        case GST_AUDIO_FORMAT_S24:
+          encoding = MPG123_ENC_SIGNED_24;
+          break;
+        case GST_AUDIO_FORMAT_S32:
+          encoding = MPG123_ENC_SIGNED_32;
+          break;
+        case GST_AUDIO_FORMAT_U16:
+          encoding = MPG123_ENC_UNSIGNED_16;
+          break;
+        case GST_AUDIO_FORMAT_U24:
+          encoding = MPG123_ENC_UNSIGNED_24;
+          break;
+        case GST_AUDIO_FORMAT_U32:
+          encoding = MPG123_ENC_UNSIGNED_32;
+          break;
+        case GST_AUDIO_FORMAT_F32:
+          encoding = MPG123_ENC_FLOAT_32;
+          break;
+        default:
+          g_assert_not_reached ();
+          goto done;
+      }
+    }
+  }
+
+  /* Sample rate, number of channels, and sample format are known at this point.
+   * Set the audioinfo structure's values and the mpg123 format. */
+  {
+    int err;
+
+    /* clear all existing format settings from the mpg123 instance */
+    mpg123_format_none (mpg123_decoder->handle);
+    /* set the chosen format */
+    err =
+        mpg123_format (mpg123_decoder->handle, sample_rate, num_channels,
+        encoding);
+
+    if (err != MPG123_OK) {
+      GST_WARNING_OBJECT (dec,
+          "mpg123_format() failed: %s",
+          mpg123_strerror (mpg123_decoder->handle));
+    } else {
+      gst_audio_info_init (&(mpg123_decoder->next_audioinfo));
+      gst_audio_info_set_format (&(mpg123_decoder->next_audioinfo), format,
+          sample_rate, num_channels, NULL);
+      GST_LOG_OBJECT (dec, "The next audio format is: %s, %u Hz, %u channels",
+          gst_audio_format_to_string (format), sample_rate, num_channels);
+      mpg123_decoder->has_next_audioinfo = TRUE;
+
+      retval = TRUE;
+    }
+  }
+
+done:
+  return retval;
+}
+
+
+static void
+gst_mpg123_audio_dec_flush (GstAudioDecoder * dec, gboolean hard)
+{
+  int error;
+  GstMpg123AudioDec *mpg123_decoder;
+
+  GST_LOG_OBJECT (dec, "Flushing decoder");
+
+  mpg123_decoder = GST_MPG123_AUDIO_DEC (dec);
+
+  g_assert (mpg123_decoder->handle != NULL);
+
+  /* Flush by reopening the feed */
+  mpg123_close (mpg123_decoder->handle);
+  error = mpg123_open_feed (mpg123_decoder->handle);
+
+  if (G_UNLIKELY (error != MPG123_OK)) {
+    GST_ELEMENT_ERROR (dec, LIBRARY, INIT, (NULL),
+        ("Error while reopening mpg123 feed: %s",
+            mpg123_plain_strerror (error)));
+    mpg123_close (mpg123_decoder->handle);
+    mpg123_delete (mpg123_decoder->handle);
+    mpg123_decoder->handle = NULL;
+  }
+
+  if (hard)
+    mpg123_decoder->has_next_audioinfo = FALSE;
+
+  /* opening/closing feeds do not affect the format defined by the
+   * mpg123_format() call that was made in gst_mpg123_audio_dec_set_format(),
+   * and since the up/downstream caps are not expected to change here, no
+   * mpg123_format() calls are done */
+}
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+  return gst_element_register (plugin, "mpg123audiodec",
+      GST_RANK_MARGINAL, gst_mpg123_audio_dec_get_type ());
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+    GST_VERSION_MINOR,
+    mpg123, "mp3 decoding based on the mpg123 library",
+    plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
diff --git a/ext/mpg123/gstmpg123audiodec.h b/ext/mpg123/gstmpg123audiodec.h
new file mode 100644
index 0000000000..e837a56b0c
--- /dev/null
+++ b/ext/mpg123/gstmpg123audiodec.h
@@ -0,0 +1,62 @@
+/*  MP3 decoding plugin for GStreamer using the mpg123 library
+ *  Copyright (C) 2012 Carlos Rafael Giani
+ *
+ *  This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Lesser General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2.1 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
+ *  Lesser General Public License for more details.
+ *
+ *  You should have received a copy of the GNU Lesser General Public
+ *  License along with this library; if not, write to the Free Software
+ *  Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+ */
+
+#ifndef __GST_MPG123_AUDIO_DEC_H__
+#define __GST_MPG123_AUDIO_DEC_H__
+
+#include <gst/gst.h>
+#include <gst/audio/gstaudiodecoder.h>
+#include <mpg123.h>
+
+
+G_BEGIN_DECLS
+
+
+typedef struct _GstMpg123AudioDec GstMpg123AudioDec;
+typedef struct _GstMpg123AudioDecClass GstMpg123AudioDecClass;
+
+
+#define GST_TYPE_MPG123_AUDIO_DEC             (gst_mpg123_audio_dec_get_type())
+#define GST_MPG123_AUDIO_DEC(obj)             (G_TYPE_CHECK_INSTANCE_CAST((obj), GST_TYPE_MPG123_AUDIO_DEC,GstMpg123AudioDec))
+#define GST_MPG123_AUDIO_DEC_CLASS(klass)     (G_TYPE_CHECK_CLASS_CAST((klass), GST_TYPE_MPG123_AUDIO_DEC,GstMpg123AudioDecClass))
+#define GST_IS_MPG123_AUDIO_DEC(obj)          (G_TYPE_CHECK_INSTANCE_TYPE((obj), GST_TYPE_MPG123_AUDIO_DEC))
+#define GST_IS_MPG123_AUDIO_DEC_CLASS(klass)  (G_TYPE_CHECK_CLASS_TYPE((klass), GST_TYPE_MPG123_AUDIO_DEC))
+
+struct _GstMpg123AudioDec
+{
+  GstAudioDecoder parent;
+
+  mpg123_handle *handle;
+
+  GstAudioInfo next_audioinfo;
+  gboolean has_next_audioinfo;
+
+  off_t frame_offset;
+};
+
+
+struct _GstMpg123AudioDecClass
+{
+  GstAudioDecoderClass parent_class;
+};
+
+G_GNUC_INTERNAL GType gst_mpg123_audio_dec_get_type (void);
+
+G_END_DECLS
+
+#endif
diff --git a/tests/check/elements/mpg123audiodec.c b/tests/check/elements/mpg123audiodec.c
new file mode 100644
index 0000000000..20d6e779dd
--- /dev/null
+++ b/tests/check/elements/mpg123audiodec.c
@@ -0,0 +1,534 @@
+/* GStreamer
+ *
+ * unit test for mpg123audiodec
+ *
+ * Copyright (c) 2012 Carlos Rafael Giani <dv@pseudoterminal.org>
+ *
+ * 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., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <unistd.h>
+
+#include <gst/check/gstcheck.h>
+#include <gst/audio/audio.h>
+
+#include <gst/fft/gstfft.h>
+#include <gst/fft/gstffts16.h>
+#include <gst/fft/gstffts32.h>
+#include <gst/fft/gstfftf32.h>
+#include <gst/fft/gstfftf64.h>
+
+#include <gst/app/gstappsink.h>
+
+/* For ease of programming we use globals to keep refs for our floating
+ * src and sink pads we create; otherwise we always have to do get_pad,
+ * get_peer, and then remove references in every test function */
+static GstPad *mysrcpad, *mysinkpad;
+
+
+#define MP2_STREAM_FILENAME "stream.mp2"
+#define MP3_CBR_STREAM_FILENAME "cbr_stream.mp3"
+#define MP3_VBR_STREAM_FILENAME "vbr_stream.mp3"
+
+
+/* mpeg 1 layer 2 stream created with:
+ * gst-launch-1.0 -v audiotestsrc wave=sine freq=440 volume=1 num-buffers=32 ! \
+ *   "audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int)44100, channels=(int)1" ! \
+ *   avenc_mp2 bitrate=32000 ! tee name=t \
+ *   t. ! queue ! fakesink silent=false \
+ *   t. ! queue ! filesink location=test.mp2
+ *
+ * mpeg 1 layer 3 CBR stream created with:
+ * gst-launch-1.0 -v audiotestsrc wave=sine freq=440 volume=1 num-buffers=32 ! \
+ *   "audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int)44100, channels=(int)1" ! \
+ *   lamemp3enc encoding-engine-quality=high cbr=true target=bitrate bitrate=32 ! \
+ *   "audio/mpeg, rate=(int)44100, channels=(int)1" ! tee name=t \
+ *   t. ! queue ! fakesink silent=false \
+ *   t. ! queue ! filesink location=test.mp3
+ *
+ * mpeg 1 layer 3 VBR stream created with:
+ * gst-launch-1.0 -v audiotestsrc wave=sine freq=440 volume=1 num-buffers=32 ! \
+ *   "audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int)44100, channels=(int)1" ! \
+ *   lamemp3enc encoding-engine-quality=high cbr=false target=quality quality=7 ! \
+ *   "audio/mpeg, rate=(int)44100, channels=(int)1" ! tee name=t \
+ *   t. ! queue ! fakesink silent=false \
+ *   t. ! queue ! filesink location=test.mp3
+ */
+
+
+/* FFT test helpers taken from gst-plugins-base tests/check/audioresample.c */
+
+#define FFT_HELPERS(type,ffttag,ffttag2,scale)                                \
+static gdouble magnitude##ffttag (const GstFFT##ffttag##Complex *c)           \
+{                                                                             \
+  gdouble mag = (gdouble) c->r * (gdouble) c->r;                              \
+  mag += (gdouble) c->i * (gdouble) c->i;                                     \
+  mag /= scale * scale;                                                       \
+  mag = 10.0 * log10 (mag);                                                   \
+  return mag;                                                                 \
+}                                                                             \
+static gdouble find_main_frequency_spot_##ffttag (                            \
+    const GstFFT##ffttag##Complex *v, int elements)                           \
+{                                                                             \
+  int i;                                                                      \
+  gdouble maxmag = -9999;                                                     \
+  int maxidx = 0;                                                             \
+  for (i=0; i<elements; ++i) {                                                \
+    gdouble mag = magnitude##ffttag (v+i);                                    \
+    if (mag > maxmag) {                                                       \
+      maxmag = mag;                                                           \
+      maxidx = i;                                                             \
+    }                                                                         \
+  }                                                                           \
+  return maxidx / (gdouble) elements;                                         \
+}                                                                             \
+static gboolean is_zero_except_##ffttag (const GstFFT##ffttag##Complex *v,    \
+    int elements, gdouble spot)                                               \
+{                                                                             \
+  int i;                                                                      \
+  for (i=0; i<elements; ++i) {                                                \
+    gdouble pos = i / (gdouble) elements;                                     \
+    gdouble mag = magnitude##ffttag (v+i);                                    \
+    if (fabs (pos - spot) > 0.01) {                                           \
+      if (mag > -35.0) {                                                      \
+        GST_LOG("Found magnitude at %f : %f (peak at %f)\n", pos, mag, spot); \
+        return FALSE;                                                         \
+      }                                                                       \
+    }                                                                         \
+  }                                                                           \
+  return TRUE;                                                                \
+}                                                                             \
+static void check_main_frequency_spot_##ffttag (GstBuffer *buffer, gdouble    \
+    expected_spot)                                                            \
+{                                                                             \
+  GstMapInfo map;                                                             \
+  int num_samples;                                                            \
+  gdouble actual_spot;                                                        \
+  GstFFT##ffttag *ctx;                                                        \
+  GstFFT##ffttag##Complex *fftdata;                                           \
+                                                                              \
+  gst_buffer_map (buffer, &map, GST_MAP_READ);                                \
+                                                                              \
+  num_samples = map.size / sizeof(type) & ~1;                                 \
+  ctx = gst_fft_##ffttag2##_new (num_samples, FALSE);                         \
+  fftdata = g_new (GstFFT##ffttag##Complex, num_samples / 2 + 1);             \
+                                                                              \
+  gst_fft_##ffttag2##_window (ctx, (type*)map.data,                           \
+    GST_FFT_WINDOW_HAMMING);                                                  \
+  gst_fft_##ffttag2##_fft (ctx, (type*)map.data, fftdata);                    \
+                                                                              \
+  actual_spot = find_main_frequency_spot_##ffttag (fftdata,                   \
+    num_samples / 2 + 1);                                                     \
+  GST_LOG ("Expected spot: %.3f actual: %.3f %f", expected_spot, actual_spot, \
+    fabs (expected_spot - actual_spot));                                      \
+  fail_unless (fabs (expected_spot - actual_spot) < 0.05,                     \
+    "Actual main frequency spot is too far away from expected one");          \
+  fail_unless (is_zero_except_##ffttag (fftdata, num_samples / 2 + 1,         \
+    actual_spot), "One secondary peak in spectrum exceeds threshold");        \
+                                                                              \
+  gst_buffer_unmap (buffer, &map);                                            \
+                                                                              \
+  gst_fft_##ffttag2##_free (ctx);                                             \
+  g_free (fftdata);                                                           \
+}
+FFT_HELPERS (gint32, S32, s32, 2147483647.0);
+
+
+static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw, format = (string) " GST_AUDIO_NE (S32))
+    );
+static GstStaticPadTemplate layer2_srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS_ANY);
+static GstStaticPadTemplate layer3_srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS_ANY);
+
+
+static void
+setup_input_pipeline (gchar const *stream_filename, GstElement ** pipeline,
+    GstElement ** appsink)
+{
+  GstElement *source, *parser;
+
+  *pipeline = gst_pipeline_new (NULL);
+  source = gst_element_factory_make ("filesrc", NULL);
+  parser = gst_element_factory_make ("mpegaudioparse", NULL);
+  *appsink = gst_element_factory_make ("appsink", NULL);
+
+  gst_bin_add_many (GST_BIN (*pipeline), source, parser, *appsink, NULL);
+  gst_element_link_many (source, parser, *appsink, NULL);
+
+  {
+    char *full_filename =
+        g_build_filename (GST_TEST_FILES_PATH, stream_filename, NULL);
+    g_object_set (G_OBJECT (source), "location", full_filename, NULL);
+    g_free (full_filename);
+  }
+
+  gst_element_set_state (*pipeline, GST_STATE_PLAYING);
+}
+
+static void
+cleanup_input_pipeline (GstElement * pipeline)
+{
+  gst_element_set_state (pipeline, GST_STATE_NULL);
+  gst_object_unref (pipeline);
+}
+
+static GstElement *
+setup_mpeg1layer2dec (void)
+{
+  GstElement *mpg123audiodec;
+  GstCaps *caps;
+
+  GST_DEBUG ("setup_mpeg1layer2dec");
+  mpg123audiodec = gst_check_setup_element ("mpg123audiodec");
+  mysrcpad = gst_check_setup_src_pad (mpg123audiodec, &layer2_srctemplate);
+  mysinkpad = gst_check_setup_sink_pad (mpg123audiodec, &sinktemplate);
+  gst_pad_set_active (mysrcpad, TRUE);
+  gst_pad_set_active (mysinkpad, TRUE);
+
+  /* This is necessary to trigger a set_format call in the decoder;
+   * fixed caps don't trigger it */
+  caps = gst_caps_new_simple ("audio/mpeg",
+      "mpegversion", G_TYPE_INT, 1,
+      "layer", G_TYPE_INT, 2,
+      "rate", G_TYPE_INT, 44100,
+      "channels", G_TYPE_INT, 1, "parsed", G_TYPE_BOOLEAN, TRUE, NULL);
+  gst_check_setup_events (mysrcpad, mpg123audiodec, caps, GST_FORMAT_TIME);
+  gst_caps_unref (caps);
+
+  return mpg123audiodec;
+}
+
+static GstElement *
+setup_mpeg1layer3dec (void)
+{
+  GstElement *mpg123audiodec;
+  GstCaps *caps;
+
+  GST_DEBUG ("setup_mpeg1layer3dec");
+  mpg123audiodec = gst_check_setup_element ("mpg123audiodec");
+  mysrcpad = gst_check_setup_src_pad (mpg123audiodec, &layer3_srctemplate);
+  mysinkpad = gst_check_setup_sink_pad (mpg123audiodec, &sinktemplate);
+  gst_pad_set_active (mysrcpad, TRUE);
+  gst_pad_set_active (mysinkpad, TRUE);
+
+  /* This is necessary to trigger a set_format call in the decoder;
+   * fixed caps don't trigger it */
+  caps = gst_caps_new_simple ("audio/mpeg",
+      "mpegversion", G_TYPE_INT, 1,
+      "layer", G_TYPE_INT, 3,
+      "rate", G_TYPE_INT, 44100,
+      "channels", G_TYPE_INT, 1, "parsed", G_TYPE_BOOLEAN, TRUE, NULL);
+  gst_check_setup_events (mysrcpad, mpg123audiodec, caps, GST_FORMAT_TIME);
+  gst_caps_unref (caps);
+
+  return mpg123audiodec;
+}
+
+static void
+cleanup_mpg123audiodec (GstElement * mpg123audiodec)
+{
+  GST_DEBUG ("cleanup_mpeg1layer2dec");
+  gst_element_set_state (mpg123audiodec, GST_STATE_NULL);
+
+  gst_pad_set_active (mysrcpad, FALSE);
+  gst_pad_set_active (mysinkpad, FALSE);
+  gst_check_teardown_src_pad (mpg123audiodec);
+  gst_check_teardown_sink_pad (mpg123audiodec);
+  gst_check_teardown_element (mpg123audiodec);
+}
+
+static void
+run_decoding_test (GstElement * mpg123audiodec, gchar const *filename)
+{
+  GstBus *bus;
+  unsigned int num_input_buffers, num_decoded_buffers;
+  gint expected_size;
+  GstCaps *out_caps, *caps;
+  GstAudioInfo audioinfo;
+  GstElement *input_pipeline, *input_appsink;
+  int i;
+  GstBuffer *outbuffer;
+
+  /* 440 Hz = frequency of sine wave in audio data
+   * 44100 Hz = sample rate
+   * (44100 / 2) Hz = Nyquist frequency */
+  static double const expected_frequency_spot = 440.0 / (44100.0 / 2.0);
+
+  fail_unless (gst_element_set_state (mpg123audiodec,
+          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+      "could not set to playing");
+  bus = gst_bus_new ();
+
+  gst_element_set_bus (mpg123audiodec, bus);
+
+  setup_input_pipeline (filename, &input_pipeline, &input_appsink);
+
+  num_input_buffers = 0;
+  while (TRUE) {
+    GstSample *sample;
+    GstBuffer *input_buffer;
+
+    sample = gst_app_sink_pull_sample (GST_APP_SINK (input_appsink));
+    if (sample == NULL)
+      break;
+
+    fail_unless (GST_IS_SAMPLE (sample));
+
+    input_buffer = gst_sample_get_buffer (sample);
+    fail_if (input_buffer == NULL);
+
+    /* This is done to be on the safe side - docs say lifetime of the input buffer
+     * depends *solely* on the sample */
+    input_buffer = gst_buffer_copy (input_buffer);
+
+    fail_unless_equals_int (gst_pad_push (mysrcpad, input_buffer), GST_FLOW_OK);
+
+    ++num_input_buffers;
+
+    gst_sample_unref (sample);
+  }
+
+  num_decoded_buffers = g_list_length (buffers);
+
+  /* check number of decoded buffers */
+  fail_unless_equals_int (num_decoded_buffers, num_input_buffers - 2);
+
+  caps = gst_pad_get_current_caps (mysinkpad);
+  GST_LOG ("output caps %" GST_PTR_FORMAT, caps);
+  fail_unless (gst_audio_info_from_caps (&audioinfo, caps),
+      "Getting audio info from caps failed");
+
+  /* check caps */
+  out_caps = gst_caps_new_simple ("audio/x-raw",
+      "format", G_TYPE_STRING, GST_AUDIO_NE (S32),
+      "layout", G_TYPE_STRING, "interleaved",
+      "rate", G_TYPE_INT, 44100, "channels", G_TYPE_INT, 1, NULL);
+
+  fail_unless (gst_caps_is_equal_fixed (caps, out_caps), "Incorrect out caps");
+
+  gst_caps_unref (out_caps);
+  gst_caps_unref (caps);
+
+  /* here, test if decoded data is a sine tone, and if the sine frequency is at the
+   * right spot in the spectrum */
+  for (i = 0; i < num_decoded_buffers; ++i) {
+    outbuffer = GST_BUFFER (buffers->data);
+    fail_if (outbuffer == NULL, "Invalid buffer retrieved");
+
+    /* MPEG 1 layer 2 uses 1152 samples per frame */
+    expected_size = 1152 * GST_AUDIO_INFO_BPF (&audioinfo);
+    fail_unless_equals_int (gst_buffer_get_size (outbuffer), expected_size);
+
+    check_main_frequency_spot_S32 (outbuffer, expected_frequency_spot);
+
+    buffers = g_list_remove (buffers, outbuffer);
+    gst_buffer_unref (outbuffer);
+    outbuffer = NULL;
+  }
+
+  g_list_free (buffers);
+  buffers = NULL;
+
+  cleanup_input_pipeline (input_pipeline);
+  gst_bus_set_flushing (bus, TRUE);
+  gst_element_set_bus (mpg123audiodec, NULL);
+  gst_object_unref (GST_OBJECT (bus));
+}
+
+
+GST_START_TEST (test_decode_mpeg1layer2)
+{
+  GstElement *mpg123audiodec;
+  mpg123audiodec = setup_mpeg1layer2dec ();
+  run_decoding_test (mpg123audiodec, MP2_STREAM_FILENAME);
+  cleanup_mpg123audiodec (mpg123audiodec);
+  mpg123audiodec = NULL;
+}
+
+GST_END_TEST;
+
+
+GST_START_TEST (test_decode_mpeg1layer3_cbr)
+{
+  GstElement *mpg123audiodec;
+  mpg123audiodec = setup_mpeg1layer3dec ();
+  run_decoding_test (mpg123audiodec, MP3_CBR_STREAM_FILENAME);
+  cleanup_mpg123audiodec (mpg123audiodec);
+}
+
+GST_END_TEST;
+
+
+GST_START_TEST (test_decode_mpeg1layer3_vbr)
+{
+  GstElement *mpg123audiodec;
+  mpg123audiodec = setup_mpeg1layer3dec ();
+  run_decoding_test (mpg123audiodec, MP3_VBR_STREAM_FILENAME);
+  cleanup_mpg123audiodec (mpg123audiodec);
+}
+
+GST_END_TEST;
+
+
+GST_START_TEST (test_decode_garbage_mpeg1layer2)
+{
+  GstElement *mpg123audiodec;
+  GstBuffer *inbuffer;
+  GstBus *bus;
+  int i, num_buffers;
+  guint32 *tmpbuf;
+
+  mpg123audiodec = setup_mpeg1layer2dec ();
+
+  fail_unless (gst_element_set_state (mpg123audiodec,
+          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+      "could not set to playing");
+  bus = gst_bus_new ();
+
+  /* initialize the buffer with something that is no mpeg2 */
+  tmpbuf = g_new (guint32, 4096);
+  for (i = 0; i < 4096; i++) {
+    tmpbuf[i] = i;
+  }
+  inbuffer = gst_buffer_new_wrapped (tmpbuf, 4096 * sizeof (guint32));
+
+  ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+  gst_element_set_bus (mpg123audiodec, bus);
+
+  /* should be possible to push without problems but nothing gets decoded */
+  fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
+
+  num_buffers = g_list_length (buffers);
+
+  /* should be 0 buffers as decoding should've been impossible */
+  fail_unless_equals_int (num_buffers, 0);
+
+  g_list_free (buffers);
+  buffers = NULL;
+
+  gst_bus_set_flushing (bus, TRUE);
+  gst_element_set_bus (mpg123audiodec, NULL);
+  gst_object_unref (GST_OBJECT (bus));
+  cleanup_mpg123audiodec (mpg123audiodec);
+  mpg123audiodec = NULL;
+}
+
+GST_END_TEST;
+
+
+GST_START_TEST (test_decode_garbage_mpeg1layer3)
+{
+  GstElement *mpg123audiodec;
+  GstBuffer *inbuffer;
+  GstBus *bus;
+  int i, num_buffers;
+  guint32 *tmpbuf;
+
+  mpg123audiodec = setup_mpeg1layer3dec ();
+
+  fail_unless (gst_element_set_state (mpg123audiodec,
+          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+      "could not set to playing");
+  bus = gst_bus_new ();
+
+  /* initialize the buffer with something that is no mpeg2 */
+  tmpbuf = g_new (guint32, 4096);
+  for (i = 0; i < 4096; i++) {
+    tmpbuf[i] = i;
+  }
+  inbuffer = gst_buffer_new_wrapped (tmpbuf, 4096 * sizeof (guint32));
+
+  ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+  gst_element_set_bus (mpg123audiodec, bus);
+
+  /* should be possible to push without problems but nothing gets decoded */
+  fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
+
+  num_buffers = g_list_length (buffers);
+
+  /* should be 0 buffers as decoding should've been impossible */
+  fail_unless_equals_int (num_buffers, 0);
+
+  g_list_free (buffers);
+  buffers = NULL;
+
+  gst_bus_set_flushing (bus, TRUE);
+  gst_element_set_bus (mpg123audiodec, NULL);
+  gst_object_unref (GST_OBJECT (bus));
+  cleanup_mpg123audiodec (mpg123audiodec);
+  mpg123audiodec = NULL;
+}
+
+GST_END_TEST;
+
+
+static gboolean
+is_test_file_available (gchar const *filename)
+{
+  gboolean ret;
+  gchar *full_filename;
+  gchar *cwd;
+
+  cwd = g_get_current_dir ();
+  full_filename = g_build_filename (cwd, GST_TEST_FILES_PATH, filename, NULL);
+  ret =
+      g_file_test (full_filename, G_FILE_TEST_IS_REGULAR | G_FILE_TEST_EXISTS);
+  g_free (full_filename);
+  g_free (cwd);
+  return ret;
+}
+
+static Suite *
+mpg123audiodec_suite (void)
+{
+  GstRegistry *registry;
+  Suite *s = suite_create ("mpg123audiodec");
+  TCase *tc_chain = tcase_create ("general");
+
+  registry = gst_registry_get ();
+
+  suite_add_tcase (s, tc_chain);
+  if (gst_registry_check_feature_version (registry, "filesrc",
+          GST_VERSION_MAJOR, GST_VERSION_MINOR, 0) &&
+      gst_registry_check_feature_version (registry, "mpegaudioparse",
+          GST_VERSION_MAJOR, GST_VERSION_MINOR, 0) &&
+      gst_registry_check_feature_version (registry, "appsrc",
+          GST_VERSION_MAJOR, GST_VERSION_MINOR, 0)) {
+    if (is_test_file_available (MP2_STREAM_FILENAME))
+      tcase_add_test (tc_chain, test_decode_mpeg1layer2);
+    if (is_test_file_available (MP3_CBR_STREAM_FILENAME))
+      tcase_add_test (tc_chain, test_decode_mpeg1layer3_cbr);
+    if (is_test_file_available (MP3_VBR_STREAM_FILENAME))
+      tcase_add_test (tc_chain, test_decode_mpeg1layer3_vbr);
+  }
+  tcase_add_test (tc_chain, test_decode_garbage_mpeg1layer2);
+  tcase_add_test (tc_chain, test_decode_garbage_mpeg1layer3);
+
+  return s;
+}
+
+
+GST_CHECK_MAIN (mpg123audiodec)