ext/mpg123/gstmpg123audiodec.c - mtk-gst-plugins-bad-debian - Git at Google

 /*  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 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);
 }


 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)
 {
 /* Using the parsed information upstream, and the list of allowed caps
  * downstream, this code tries to find a suitable audio info. It is important
  * to keep in mind that the rate and number of channels should never deviate
  * from the one the bitstream has, otherwise mpg123 has to mix channels and/or
  * resample (and as its docs say, its internal resampler is very crude). The
  * sample format, however, can be chosen freely, because the MPEG specs do not
  * mandate any special format. Therefore, rate and number of channels are taken
  * from upstream (which parsed the MPEG frames, so the input_caps contain
  * exactly the rate and number of channels the bitstream actually has), while
  * the sample format is chosen by trying out all caps that are allowed by
  * downstream. This way, the output is adjusted to what the downstream prefers.
  *
  * Also, the new output audio info is not set immediately. Instead, it is
  * considered the "next audioinfo". The code waits for mpg123 to notice the new
  * format (= when mpg123_decode_frame() returns MPG123_AUDIO_DEC_NEW_FORMAT),
  * and then sets the next audioinfo. Otherwise, the next audioinfo is set too
  * soon, which may cause problems with mp3s containing several format headers.
  * One example would be an mp3 with the first 30 seconds using 44.1 kHz, then
  * the next 30 seconds using 32 kHz. Rare, but possible.
  *
  * STEPS:
  *
  * 1. get rate and channels from input_caps
  * 2. get allowed caps from src pad
  * 3. for each structure in allowed caps:
  * 3.1. take format
  * 3.2. if the combination of format with rate and channels is unsupported by
  *      mpg123, go to (3), or exit with error if there are no more structures
  *      to try
  * 3.3. create next audioinfo out of rate,channels,format, and exit
  */


   int rate, channels;
   GstMpg123AudioDec *mpg123_decoder;
   GstCaps *allowed_srccaps;
   guint structure_nr;
   gboolean match_found = FALSE;

   mpg123_decoder = GST_MPG123_AUDIO_DEC (dec);

   g_assert (mpg123_decoder->handle != NULL);

   mpg123_decoder->has_next_audioinfo = FALSE;

   /* Get rate and 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", &rate)) {
       err = TRUE;
       GST_ERROR_OBJECT (dec, "Input caps do not have a rate value");
     }
     if (!gst_structure_get_int (structure, "channels", &channels)) {
       err = TRUE;
       GST_ERROR_OBJECT (dec, "Input caps do not have a channel value");
     }

     if (err)
       return FALSE;
   }

   /* Get the caps that are allowed by downstream */
   {
     GstCaps *allowed_srccaps_unnorm =
         gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (dec));
     if (!allowed_srccaps_unnorm) {
       GST_ERROR_OBJECT (dec, "Allowed src caps are NULL");
       return FALSE;
     }
     allowed_srccaps = gst_caps_normalize (allowed_srccaps_unnorm);
   }

   /* Go through all allowed caps, pick the first one that matches */
   for (structure_nr = 0; structure_nr < gst_caps_get_size (allowed_srccaps);
       ++structure_nr) {
     GstStructure *structure;
     gchar const *format_str;
     GstAudioFormat format;
     int encoding;

     structure = gst_caps_get_structure (allowed_srccaps, structure_nr);

     format_str = gst_structure_get_string (structure, "format");
     if (format_str == NULL) {
       GST_DEBUG_OBJECT (dec, "Could not get format from src caps");
       continue;
     }

     format = gst_audio_format_from_string (format_str);
     if (format == GST_AUDIO_FORMAT_UNKNOWN) {
       GST_DEBUG_OBJECT (dec, "Unknown format %s", format_str);
       continue;
     }

     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:
         GST_DEBUG_OBJECT (dec,
             "Format %s in srccaps is not supported", format_str);
         continue;
     }

     {
       int err;

       /* Cleanup old formats & set new one */
       mpg123_format_none (mpg123_decoder->handle);
       err = mpg123_format (mpg123_decoder->handle, rate, channels, encoding);
       if (err != MPG123_OK) {
         GST_DEBUG_OBJECT (dec,
             "mpg123 cannot use caps %" GST_PTR_FORMAT
             " because mpg123_format() failed: %s", structure,
             mpg123_strerror (mpg123_decoder->handle));
         continue;
       }
     }

     gst_audio_info_init (&(mpg123_decoder->next_audioinfo));
     gst_audio_info_set_format (&(mpg123_decoder->next_audioinfo), format, rate,
         channels, NULL);
     GST_LOG_OBJECT (dec, "The next audio format is: %s, %u Hz, %u channels",
         format_str, rate, channels);
     mpg123_decoder->has_next_audioinfo = TRUE;

     match_found = TRUE;

     break;
   }

   gst_caps_unref (allowed_srccaps);

   return match_found;
 }


 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;
   }

   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)
	/* 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 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);
	}


	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)
	{
	/* Using the parsed information upstream, and the list of allowed caps
	* downstream, this code tries to find a suitable audio info. It is important
	* to keep in mind that the rate and number of channels should never deviate
	* from the one the bitstream has, otherwise mpg123 has to mix channels and/or
	* resample (and as its docs say, its internal resampler is very crude). The
	* sample format, however, can be chosen freely, because the MPEG specs do not
	* mandate any special format. Therefore, rate and number of channels are taken
	* from upstream (which parsed the MPEG frames, so the input_caps contain
	* exactly the rate and number of channels the bitstream actually has), while
	* the sample format is chosen by trying out all caps that are allowed by
	* downstream. This way, the output is adjusted to what the downstream prefers.
	*
	* Also, the new output audio info is not set immediately. Instead, it is
	* considered the "next audioinfo". The code waits for mpg123 to notice the new
	* format (= when mpg123_decode_frame() returns MPG123_AUDIO_DEC_NEW_FORMAT),
	* and then sets the next audioinfo. Otherwise, the next audioinfo is set too
	* soon, which may cause problems with mp3s containing several format headers.
	* One example would be an mp3 with the first 30 seconds using 44.1 kHz, then
	* the next 30 seconds using 32 kHz. Rare, but possible.
	*
	* STEPS:
	*
	* 1. get rate and channels from input_caps
	* 2. get allowed caps from src pad
	* 3. for each structure in allowed caps:
	* 3.1. take format
	* 3.2. if the combination of format with rate and channels is unsupported by
	* mpg123, go to (3), or exit with error if there are no more structures
	* to try
	* 3.3. create next audioinfo out of rate,channels,format, and exit
	*/


	int rate, channels;
	GstMpg123AudioDec *mpg123_decoder;
	GstCaps *allowed_srccaps;
	guint structure_nr;
	gboolean match_found = FALSE;

	mpg123_decoder = GST_MPG123_AUDIO_DEC (dec);

	g_assert (mpg123_decoder->handle != NULL);

	mpg123_decoder->has_next_audioinfo = FALSE;

	/* Get rate and 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", &rate)) {
	err = TRUE;
	GST_ERROR_OBJECT (dec, "Input caps do not have a rate value");
	}
	if (!gst_structure_get_int (structure, "channels", &channels)) {
	err = TRUE;
	GST_ERROR_OBJECT (dec, "Input caps do not have a channel value");
	}

	if (err)
	return FALSE;
	}

	/* Get the caps that are allowed by downstream */
	{
	GstCaps *allowed_srccaps_unnorm =
	gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (dec));
	if (!allowed_srccaps_unnorm) {
	GST_ERROR_OBJECT (dec, "Allowed src caps are NULL");
	return FALSE;
	}
	allowed_srccaps = gst_caps_normalize (allowed_srccaps_unnorm);
	}

	/* Go through all allowed caps, pick the first one that matches */
	for (structure_nr = 0; structure_nr < gst_caps_get_size (allowed_srccaps);
	++structure_nr) {
	GstStructure *structure;
	gchar const *format_str;
	GstAudioFormat format;
	int encoding;

	structure = gst_caps_get_structure (allowed_srccaps, structure_nr);

	format_str = gst_structure_get_string (structure, "format");
	if (format_str == NULL) {
	GST_DEBUG_OBJECT (dec, "Could not get format from src caps");
	continue;
	}

	format = gst_audio_format_from_string (format_str);
	if (format == GST_AUDIO_FORMAT_UNKNOWN) {
	GST_DEBUG_OBJECT (dec, "Unknown format %s", format_str);
	continue;
	}

	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:
	GST_DEBUG_OBJECT (dec,
	"Format %s in srccaps is not supported", format_str);
	continue;
	}

	{
	int err;

	/* Cleanup old formats & set new one */
	mpg123_format_none (mpg123_decoder->handle);
	err = mpg123_format (mpg123_decoder->handle, rate, channels, encoding);
	if (err != MPG123_OK) {
	GST_DEBUG_OBJECT (dec,
	"mpg123 cannot use caps %" GST_PTR_FORMAT
	" because mpg123_format() failed: %s", structure,
	mpg123_strerror (mpg123_decoder->handle));
	continue;
	}
	}

	gst_audio_info_init (&(mpg123_decoder->next_audioinfo));
	gst_audio_info_set_format (&(mpg123_decoder->next_audioinfo), format, rate,
	channels, NULL);
	GST_LOG_OBJECT (dec, "The next audio format is: %s, %u Hz, %u channels",
	format_str, rate, channels);
	mpg123_decoder->has_next_audioinfo = TRUE;

	match_found = TRUE;

	break;
	}

	gst_caps_unref (allowed_srccaps);

	return match_found;
	}


	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;
	}

	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)