text: /* GStreamer
   * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
   * Copyright (C) <2004> Wim Taymans <wim@fluendo.com>
   * Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>
   * Copyright (C) <2009> Sebastian Dröge <sebastian.droege@collabora.co.uk>
   *
   * 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.
   */
  
  /**
   * SECTION:element-lamemp3enc
   * @see_also: lame, mad, vorbisenc
   *
   * This element encodes raw integer audio into an MPEG-1 layer 3 (MP3) stream.
   * Note that <ulink url="http://en.wikipedia.org/wiki/MP3">MP3</ulink> is not
   * a free format, there are licensing and patent issues to take into
   * consideration. See <ulink url="http://www.vorbis.com/">Ogg/Vorbis</ulink>
   * for a royalty free (and often higher quality) alternative.
   *
   * <refsect2>
   * <title>Output sample rate</title>
   * If no fixed output sample rate is negotiated on the element's src pad,
   * the element will choose an optimal sample rate to resample to internally.
   * For example, a 16-bit 44.1 KHz mono audio stream encoded at 48 kbit will
   * get resampled to 32 KHz.  Use filter caps on the src pad to force a
   * particular sample rate.
   * </refsect2>
   * <refsect2>
   * <title>Example pipelines</title>
   * |[
   * gst-launch-1.0 -v audiotestsrc wave=sine num-buffers=100 ! audioconvert ! lamemp3enc ! filesink location=sine.mp3
   * ]| Encode a test sine signal to MP3.
   * |[
   * gst-launch-1.0 -v autoaudiosrc ! audioconvert ! lamemp3enc target=bitrate bitrate=192 ! filesink location=alsasrc.mp3
   * ]| Record from a sound card using ALSA and encode to MP3 with an average bitrate of 192kbps
   * |[
   * gst-launch-1.0 -v filesrc location=music.wav ! decodebin ! audioconvert ! audioresample ! lamemp3enc target=quality quality=0 ! id3v2mux ! filesink location=music.mp3
   * ]| Transcode from a .wav file to MP3 (the id3v2mux element is optional) with best VBR quality
   * |[
   * gst-launch-1.0 -v cdda://5 ! audioconvert ! lamemp3enc target=bitrate cbr=true bitrate=192 ! filesink location=track5.mp3
   * ]| Encode Audio CD track 5 to MP3 with a constant bitrate of 192kbps
   * |[
   * gst-launch-1.0 -v audiotestsrc num-buffers=10 ! audio/x-raw,rate=44100,channels=1 ! lamemp3enc target=bitrate cbr=true bitrate=48 ! filesink location=test.mp3
   * ]| Encode to a fixed sample rate
   * </refsect2>
   */
  
  #ifdef HAVE_CONFIG_H
  #include "config.h"
  #endif
  
  #include <string.h>
  #include "gstlamemp3enc.h"
  #include <gst/gst-i18n-plugin.h>
  
  /* lame < 3.98 */
  #ifndef HAVE_LAME_SET_VBR_QUALITY
  #define lame_set_VBR_quality(flags,q) lame_set_VBR_q((flags),(int)(q))
  #endif
  
  GST_DEBUG_CATEGORY_STATIC (debug);
  #define GST_CAT_DEFAULT debug
  
  /* elementfactory information */
  
  /* LAMEMP3ENC can do MPEG-1, MPEG-2, and MPEG-2.5, so it has 9 possible
   * sample rates it supports */
  static GstStaticPadTemplate gst_lamemp3enc_sink_template =
      GST_STATIC_PAD_TEMPLATE ("sink",
      GST_PAD_SINK,
      GST_PAD_ALWAYS,
      GST_STATIC_CAPS ("audio/x-raw, "
          "format = (string) " GST_AUDIO_NE (S16) ", "
          "layout = (string) interleaved, "
          "rate = (int) { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }, "
          "channels = (int) 1; "
          "audio/x-raw, "
          "format = (string) " GST_AUDIO_NE (S16) ", "
          "layout = (string) interleaved, "
          "rate = (int) { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }, "
          "channels = (int) 2, " "channel-mask = (bitmask) 0x3")
      );
  
  static GstStaticPadTemplate gst_lamemp3enc_src_template =
  GST_STATIC_PAD_TEMPLATE ("src",
      GST_PAD_SRC,
      GST_PAD_ALWAYS,
      GST_STATIC_CAPS ("audio/mpeg, "
          "mpegversion = (int) 1, "
          "layer = (int) 3, "
          "rate = (int) { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }, "
          "channels = (int) [ 1, 2 ]")
      );
  
  /********** Define useful types for non-programmatic interfaces **********/
  enum
  {
    LAMEMP3ENC_TARGET_QUALITY = 0,
    LAMEMP3ENC_TARGET_BITRATE
  };
  
  #define GST_TYPE_LAMEMP3ENC_TARGET (gst_lamemp3enc_target_get_type())
  static GType
  gst_lamemp3enc_target_get_type (void)
  {
    static GType lame_target_type = 0;
    static const GEnumValue lame_targets[] = {
      {LAMEMP3ENC_TARGET_QUALITY, "Quality", "quality"},
      {LAMEMP3ENC_TARGET_BITRATE, "Bitrate", "bitrate"},
      {0, NULL, NULL}
    };
  
    if (!lame_target_type) {
      lame_target_type =
          g_enum_register_static ("GstLameMP3EncTarget", lame_targets);
    }
    return lame_target_type;
  }
  
  enum
  {
    LAMEMP3ENC_ENCODING_ENGINE_QUALITY_FAST = 0,
    LAMEMP3ENC_ENCODING_ENGINE_QUALITY_STANDARD,
    LAMEMP3ENC_ENCODING_ENGINE_QUALITY_HIGH
  };
  
  #define GST_TYPE_LAMEMP3ENC_ENCODING_ENGINE_QUALITY (gst_lamemp3enc_encoding_engine_quality_get_type())
  static GType
  gst_lamemp3enc_encoding_engine_quality_get_type (void)
  {
    static GType lame_encoding_engine_quality_type = 0;
    static const GEnumValue lame_encoding_engine_quality[] = {
      {0, "Fast", "fast"},
      {1, "Standard", "standard"},
      {2, "High", "high"},
      {0, NULL, NULL}
    };
  
    if (!lame_encoding_engine_quality_type) {
      lame_encoding_engine_quality_type =
          g_enum_register_static ("GstLameMP3EncEncodingEngineQuality",
          lame_encoding_engine_quality);
    }
    return lame_encoding_engine_quality_type;
  }
  
  /********** Standard stuff for signals and arguments **********/
  
  enum
  {
    ARG_0,
    ARG_TARGET,
    ARG_BITRATE,
    ARG_CBR,
    ARG_QUALITY,
    ARG_ENCODING_ENGINE_QUALITY,
    ARG_MONO
  };
  
  #define DEFAULT_TARGET LAMEMP3ENC_TARGET_QUALITY
  #define DEFAULT_BITRATE 128
  #define DEFAULT_CBR FALSE
  #define DEFAULT_QUALITY 4
  #define DEFAULT_ENCODING_ENGINE_QUALITY LAMEMP3ENC_ENCODING_ENGINE_QUALITY_STANDARD
  #define DEFAULT_MONO FALSE
  
  static gboolean gst_lamemp3enc_start (GstAudioEncoder * enc);
  static gboolean gst_lamemp3enc_stop (GstAudioEncoder * enc);
  static gboolean gst_lamemp3enc_set_format (GstAudioEncoder * enc,
      GstAudioInfo * info);
  static GstFlowReturn gst_lamemp3enc_handle_frame (GstAudioEncoder * enc,
      GstBuffer * in_buf);
  static void gst_lamemp3enc_flush (GstAudioEncoder * enc);
  
  static void gst_lamemp3enc_set_property (GObject * object, guint prop_id,
      const GValue * value, GParamSpec * pspec);
  static void gst_lamemp3enc_get_property (GObject * object, guint prop_id,
      GValue * value, GParamSpec * pspec);
  static gboolean gst_lamemp3enc_setup (GstLameMP3Enc * lame, GstTagList ** tags);
  
  #define gst_lamemp3enc_parent_class parent_class
  G_DEFINE_TYPE (GstLameMP3Enc, gst_lamemp3enc, GST_TYPE_AUDIO_ENCODER);
  
  static void
  gst_lamemp3enc_release_memory (GstLameMP3Enc * lame)
  {
    if (lame->lgf) {
      lame_close (lame->lgf);
      lame->lgf = NULL;
    }
  }
  
  static void
  gst_lamemp3enc_finalize (GObject * obj)
  {
    gst_lamemp3enc_release_memory (GST_LAMEMP3ENC (obj));
  
    G_OBJECT_CLASS (parent_class)->finalize (obj);
  }
  
  static void
  gst_lamemp3enc_class_init (GstLameMP3EncClass * klass)
  {
    GObjectClass *gobject_class;
    GstElementClass *gstelement_class;
    GstAudioEncoderClass *base_class;
  
    gobject_class = (GObjectClass *) klass;
    gstelement_class = (GstElementClass *) klass;
    base_class = (GstAudioEncoderClass *) klass;
  
    gobject_class->set_property = gst_lamemp3enc_set_property;
    gobject_class->get_property = gst_lamemp3enc_get_property;
    gobject_class->finalize = gst_lamemp3enc_finalize;
  
    gst_element_class_add_static_pad_template (gstelement_class,
        &gst_lamemp3enc_src_template);
    gst_element_class_add_static_pad_template (gstelement_class,
        &gst_lamemp3enc_sink_template);
  
    gst_element_class_set_static_metadata (gstelement_class,
        "L.A.M.E. mp3 encoder", "Codec/Encoder/Audio",
        "High-quality free MP3 encoder",
        "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
  
    base_class->start = GST_DEBUG_FUNCPTR (gst_lamemp3enc_start);
    base_class->stop = GST_DEBUG_FUNCPTR (gst_lamemp3enc_stop);
    base_class->set_format = GST_DEBUG_FUNCPTR (gst_lamemp3enc_set_format);
    base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_lamemp3enc_handle_frame);
    base_class->flush = GST_DEBUG_FUNCPTR (gst_lamemp3enc_flush);
  
    g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_TARGET,
        g_param_spec_enum ("target", "Target",
            "Optimize for quality or bitrate", GST_TYPE_LAMEMP3ENC_TARGET,
            DEFAULT_TARGET,
            G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
    g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_BITRATE,
        g_param_spec_int ("bitrate", "Bitrate (kb/s)",
            "Bitrate in kbit/sec (Only valid if target is bitrate, for CBR one "
            "of 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, "
            "256 or 320)", 8, 320, DEFAULT_BITRATE,
            G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
    g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_CBR,
        g_param_spec_boolean ("cbr", "CBR", "Enforce constant bitrate encoding "
            "(Only valid if target is bitrate)", DEFAULT_CBR,
            G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
    g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_QUALITY,
        g_param_spec_float ("quality", "Quality",
            "VBR Quality from 0 to 10, 0 being the best "
            "(Only valid if target is quality)", 0.0, 9.999,
            DEFAULT_QUALITY,
            G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
    g_object_class_install_property (G_OBJECT_CLASS (klass),
        ARG_ENCODING_ENGINE_QUALITY, g_param_spec_enum ("encoding-engine-quality",
            "Encoding Engine Quality", "Quality/speed of the encoding engine, "
            "this does not affect the bitrate!",
            GST_TYPE_LAMEMP3ENC_ENCODING_ENGINE_QUALITY,
            DEFAULT_ENCODING_ENGINE_QUALITY,
            G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
    g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_MONO,
        g_param_spec_boolean ("mono", "Mono", "Enforce mono encoding",
            DEFAULT_MONO,
            G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
  }
  
  static void
  gst_lamemp3enc_init (GstLameMP3Enc * lame)
  {
    GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (lame));
  }
  
  static gboolean
  gst_lamemp3enc_start (GstAudioEncoder * enc)
  {
    GstLameMP3Enc *lame = GST_LAMEMP3ENC (enc);
  
    GST_DEBUG_OBJECT (lame, "start");
  
    if (!lame->adapter)
      lame->adapter = gst_adapter_new ();
    gst_adapter_clear (lame->adapter);
  
    return TRUE;
  }
  
  static gboolean
  gst_lamemp3enc_stop (GstAudioEncoder * enc)
  {
    GstLameMP3Enc *lame = GST_LAMEMP3ENC (enc);
  
    GST_DEBUG_OBJECT (lame, "stop");
  
    if (lame->adapter) {
      g_object_unref (lame->adapter);
      lame->adapter = NULL;
    }
  
    gst_lamemp3enc_release_memory (lame);
    return TRUE;
  }
  
  static gboolean
  gst_lamemp3enc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
  {
    GstLameMP3Enc *lame;
    gint out_samplerate;
    gint version;
    GstCaps *othercaps;
    GstClockTime latency;
    GstTagList *tags = NULL;
  
    lame = GST_LAMEMP3ENC (enc);
  
    /* parameters already parsed for us */
    lame->samplerate = GST_AUDIO_INFO_RATE (info);
    lame->num_channels = GST_AUDIO_INFO_CHANNELS (info);
  
    /* but we might be asked to reconfigure, so reset */
    gst_lamemp3enc_release_memory (lame);
  
    GST_DEBUG_OBJECT (lame, "setting up lame");
    if (!gst_lamemp3enc_setup (lame, &tags))
      goto setup_failed;
  
    out_samplerate = lame_get_out_samplerate (lame->lgf);
    if (out_samplerate == 0)
      goto zero_output_rate;
    if (out_samplerate != lame->samplerate) {
      GST_WARNING_OBJECT (lame,
          "output samplerate %d is different from incoming samplerate %d",
          out_samplerate, lame->samplerate);
    }
    lame->out_samplerate = out_samplerate;
  
    version = lame_get_version (lame->lgf);
    if (version == 0)
      version = 2;
    else if (version == 1)
      version = 1;
    else if (version == 2)
      version = 3;
  
    othercaps =
        gst_caps_new_simple ("audio/mpeg",
        "mpegversion", G_TYPE_INT, 1,
        "mpegaudioversion", G_TYPE_INT, version,
        "layer", G_TYPE_INT, 3,
        "channels", G_TYPE_INT, lame->mono ? 1 : lame->num_channels,
        "rate", G_TYPE_INT, out_samplerate, NULL);
  
    /* and use these caps */
    gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (enc), othercaps);
    gst_caps_unref (othercaps);
  
    /* base class feedback:
     * - we will handle buffers, just hand us all available
     * - report latency */
    latency = gst_util_uint64_scale_int (lame_get_framesize (lame->lgf),
        GST_SECOND, lame->samplerate);
    gst_audio_encoder_set_latency (enc, latency, latency);
  
    if (tags) {
      gst_audio_encoder_merge_tags (enc, tags, GST_TAG_MERGE_REPLACE);
      gst_tag_list_unref (tags);
    }
  
    return TRUE;
  
  zero_output_rate:
    {
      if (tags)
        gst_tag_list_unref (tags);
      GST_ELEMENT_ERROR (lame, LIBRARY, SETTINGS, (NULL),
          ("LAME mp3 audio decided on a zero sample rate"));
      return FALSE;
    }
  setup_failed:
    {
      GST_ELEMENT_ERROR (lame, LIBRARY, SETTINGS,
          (_("Failed to configure LAME mp3 audio encoder. Check your encoding parameters.")), (NULL));
      return FALSE;
    }
  }
  
  /* <php-emulation-mode>three underscores for ___rate is really really really
   * private as opposed to one underscore<php-emulation-mode> */
  /* call this MACRO outside of the NULL state so that we have a higher chance
   * of actually having a pipeline and bus to get the message through */
  
  #define CHECK_AND_FIXUP_BITRATE(obj,param,rate)		 		  \
  G_STMT_START {                                                            \
    gint ___rate = rate;                                                    \
    gint maxrate = 320;							  \
    gint multiplier = 64;							  \
    if (rate == 0) {                                                        \
      ___rate = rate;                                                       \
    } else if (rate <= 64) {				                  \
      maxrate = 64; multiplier = 8;                                         \
      if ((rate % 8) != 0) ___rate = GST_ROUND_UP_8 (rate); 		  \
    } else if (rate <= 128) {						  \
      maxrate = 128; multiplier = 16;                                       \
      if ((rate % 16) != 0) ___rate = GST_ROUND_UP_16 (rate);               \
    } else if (rate <= 256) {						  \
      maxrate = 256; multiplier = 32;                                       \
      if ((rate % 32) != 0) ___rate = GST_ROUND_UP_32 (rate);               \
    } else if (rate <= 320) { 						  \
      maxrate = 320; multiplier = 64;                                       \
      if ((rate % 64) != 0) ___rate = GST_ROUND_UP_64 (rate);               \
    }                                                                       \
    if (___rate != rate) {                                                  \
      GST_ELEMENT_WARNING (obj, LIBRARY, SETTINGS,			  \
        (_("The requested bitrate %d kbit/s for property '%s' "             \
         "is not allowed. "  					          \
         "The bitrate was changed to %d kbit/s."), rate,		          \
           param,  ___rate), 					          \
         ("A bitrate below %d should be a multiple of %d.", 		  \
            maxrate, multiplier));		  			  \
      rate = ___rate;                                                       \
    }                                                                       \
  } G_STMT_END
  
  static void
  gst_lamemp3enc_set_property (GObject * object, guint prop_id,
      const GValue * value, GParamSpec * pspec)
  {
    GstLameMP3Enc *lame;
  
    lame = GST_LAMEMP3ENC (object);
  
    switch (prop_id) {
      case ARG_TARGET:
        lame->target = g_value_get_enum (value);
        break;
      case ARG_BITRATE:
        lame->bitrate = g_value_get_int (value);
        break;
      case ARG_CBR:
        lame->cbr = g_value_get_boolean (value);
        break;
      case ARG_QUALITY:
        lame->quality = g_value_get_float (value);
        break;
      case ARG_ENCODING_ENGINE_QUALITY:
        lame->encoding_engine_quality = g_value_get_enum (value);
        break;
      case ARG_MONO:
        lame->mono = g_value_get_boolean (value);
        break;
      default:
        G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
        break;
    }
  }
  
  static void
  gst_lamemp3enc_get_property (GObject * object, guint prop_id, GValue * value,
      GParamSpec * pspec)
  {
    GstLameMP3Enc *lame;
  
    lame = GST_LAMEMP3ENC (object);
  
    switch (prop_id) {
      case ARG_TARGET:
        g_value_set_enum (value, lame->target);
        break;
      case ARG_BITRATE:
        g_value_set_int (value, lame->bitrate);
        break;
      case ARG_CBR:
        g_value_set_boolean (value, lame->cbr);
        break;
      case ARG_QUALITY:
        g_value_set_float (value, lame->quality);
        break;
      case ARG_ENCODING_ENGINE_QUALITY:
        g_value_set_enum (value, lame->encoding_engine_quality);
        break;
      case ARG_MONO:
        g_value_set_boolean (value, lame->mono);
        break;
      default:
        G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
        break;
    }
  }
  
  /* **** credits go to mpegaudioparse **** */
  
  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 (GstLameMP3Enc * lame, 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 (lame, "Calculated mp3 frame length of %u bytes", length);
    GST_DEBUG_OBJECT (lame, "samplerate = %lu, bitrate = %lu, version = %lu, "
        "layer = %lu, channels = %lu", samplerate, bitrate, version,
        layer, channels);
  
    if (put_version)
      *put_version = version;
    if (put_layer)
      *put_layer = layer;
    if (put_channels)
      *put_channels = channels;
    if (put_bitrate)
      *put_bitrate = bitrate;
    if (put_samplerate)
      *put_samplerate = samplerate;
    if (put_mode)
      *put_mode = mode;
    if (put_crc)
      *put_crc = crc;
  
    return length;
  }
  
  static gboolean
  mp3_sync_check (GstLameMP3Enc * lame, unsigned long head)
  {
    GST_DEBUG_OBJECT (lame, "checking mp3 header 0x%08lx", head);
    /* if it's not a valid sync */
    if ((head & 0xffe00000) != 0xffe00000) {
      GST_WARNING_OBJECT (lame, "invalid sync");
      return FALSE;
    }
    /* if it's an invalid MPEG version */
    if (((head >> 19) & 3) == 0x1) {
      GST_WARNING_OBJECT (lame, "invalid MPEG version: 0x%lx", (head >> 19) & 3);
      return FALSE;
    }
    /* if it's an invalid layer */
    if (!((head >> 17) & 3)) {
      GST_WARNING_OBJECT (lame, "invalid layer: 0x%lx", (head >> 17) & 3);
      return FALSE;
    }
    /* if it's an invalid bitrate */
    if (((head >> 12) & 0xf) == 0x0) {
      GST_WARNING_OBJECT (lame, "invalid bitrate: 0x%lx."
          "Free format files are not supported yet", (head >> 12) & 0xf);
      return FALSE;
    }
    if (((head >> 12) & 0xf) == 0xf) {
      GST_WARNING_OBJECT (lame, "invalid bitrate: 0x%lx", (head >> 12) & 0xf);
      return FALSE;
    }
    /* if it's an invalid samplerate */
    if (((head >> 10) & 0x3) == 0x3) {
      GST_WARNING_OBJECT (lame, "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 (lame, "invalid emphasis: 0x%lx", head & 0x3);
    }
  
    return TRUE;
  }
  
  /* **** end mpegaudioparse **** */
  
  static GstFlowReturn
  gst_lamemp3enc_finish_frames (GstLameMP3Enc * lame)
  {
    gint av;
    guint header;
    GstFlowReturn result = GST_FLOW_OK;
  
    /* limited parsing, we don't expect to lose sync here */
    while ((result == GST_FLOW_OK) &&
        ((av = gst_adapter_available (lame->adapter)) > 4)) {
      guint rate, version, layer, size;
      GstBuffer *mp3_buf;
      const guint8 *data;
      guint samples_per_frame;
  
      data = gst_adapter_map (lame->adapter, 4);
      header = GST_READ_UINT32_BE (data);
      gst_adapter_unmap (lame->adapter);
  
      if (!mp3_sync_check (lame, header))
        goto invalid_header;
  
      size = mp3_type_frame_length_from_header (lame, header, &version, &layer,
          NULL, NULL, &rate, NULL, NULL);
  
      if (G_UNLIKELY (layer != 3 || rate != lame->out_samplerate)) {
        GST_DEBUG_OBJECT (lame,
            "unexpected mp3 header with rate %u, version %u, layer %u",
            rate, version, layer);
        goto invalid_header;
      }
  
      if (size > av) {
        /* pretty likely to occur when lame is holding back on us */
        GST_LOG_OBJECT (lame, "frame size %u (> %d)", size, av);
        break;
      }
  
      /* Account for the internal resampling, finish frame really wants to
       * know about the number of incoming samples
       */
      samples_per_frame = (version == 1) ? 1152 : 576;
      samples_per_frame *= lame->samplerate;
      samples_per_frame /= lame->out_samplerate;
  
      /* should be ok now */
      mp3_buf = gst_adapter_take_buffer (lame->adapter, size);
      /* number of samples for MPEG-1, layer 3 */
      result = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (lame),
          mp3_buf, samples_per_frame);
    }
  
  exit:
    return result;
  
    /* ERRORS */
  invalid_header:
    {
      GST_ELEMENT_ERROR (lame, STREAM, ENCODE,
          ("invalid lame mp3 sync header %08X", header), (NULL));
      result = GST_FLOW_ERROR;
      goto exit;
    }
  }
  
  static GstFlowReturn
  gst_lamemp3enc_flush_full (GstLameMP3Enc * lame, gboolean push)
  {
    GstBuffer *buf;
    GstMapInfo map;
    gint size;
    GstFlowReturn result = GST_FLOW_OK;
    gint av;
  
    if (!lame->lgf)
      return GST_FLOW_OK;
  
    buf = gst_buffer_new_and_alloc (7200);
    gst_buffer_map (buf, &map, GST_MAP_WRITE);
    size = lame_encode_flush (lame->lgf, map.data, 7200);
  
    if (size > 0) {
      gst_buffer_unmap (buf, &map);
      gst_buffer_resize (buf, 0, size);
      GST_DEBUG_OBJECT (lame, "collecting final %d bytes", size);
      gst_adapter_push (lame->adapter, buf);
    } else {
      gst_buffer_unmap (buf, &map);
      GST_DEBUG_OBJECT (lame, "no final packet (size=%d, push=%d)", size, push);
      gst_buffer_unref (buf);
      result = GST_FLOW_OK;
    }
  
    if (push) {
      result = gst_lamemp3enc_finish_frames (lame);
    } else {
      /* never mind */
      gst_adapter_clear (lame->adapter);
    }
  
    /* either way, we expect nothing left */
    if ((av = gst_adapter_available (lame->adapter))) {
      /* should this be more fatal ?? */
      GST_WARNING_OBJECT (lame, "unparsed %d bytes left after flushing", av);
      /* clean up anyway */
      gst_adapter_clear (lame->adapter);
    }
  
    return result;
  }
  
  static void
  gst_lamemp3enc_flush (GstAudioEncoder * enc)
  {
    gst_lamemp3enc_flush_full (GST_LAMEMP3ENC (enc), FALSE);
  }
  
  static GstFlowReturn
  gst_lamemp3enc_handle_frame (GstAudioEncoder * enc, GstBuffer * in_buf)
  {
    GstLameMP3Enc *lame;
    gint mp3_buffer_size, mp3_size;
    GstBuffer *mp3_buf;
    GstFlowReturn result;
    gint num_samples;
    GstMapInfo in_map, mp3_map;
  
    lame = GST_LAMEMP3ENC (enc);
  
    /* squeeze remaining and push */
    if (G_UNLIKELY (in_buf == NULL))
      return gst_lamemp3enc_flush_full (lame, TRUE);
  
    gst_buffer_map (in_buf, &in_map, GST_MAP_READ);
  
    num_samples = in_map.size / 2;
  
    /* allocate space for output */
    mp3_buffer_size = 1.25 * num_samples + 7200;
    mp3_buf = gst_buffer_new_allocate (NULL, mp3_buffer_size, NULL);
    gst_buffer_map (mp3_buf, &mp3_map, GST_MAP_WRITE);
  
    /* lame seems to be too stupid to get mono interleaved going */
    if (lame->num_channels == 1) {
      mp3_size = lame_encode_buffer (lame->lgf,
          (short int *) in_map.data,
          (short int *) in_map.data, num_samples, mp3_map.data, mp3_buffer_size);
    } else {
      mp3_size = lame_encode_buffer_interleaved (lame->lgf,
          (short int *) in_map.data,
          num_samples / lame->num_channels, mp3_map.data, mp3_buffer_size);
    }
    gst_buffer_unmap (in_buf, &in_map);
  
    GST_LOG_OBJECT (lame, "encoded %" G_GSIZE_FORMAT " bytes of audio "
        "to %d bytes of mp3", in_map.size, mp3_size);
  
    if (G_LIKELY (mp3_size > 0)) {
      /* unfortunately lame does not provide frame delineated output,
       * so collect output and parse into frames ... */
      gst_buffer_unmap (mp3_buf, &mp3_map);
      gst_buffer_resize (mp3_buf, 0, mp3_size);
      gst_adapter_push (lame->adapter, mp3_buf);
      result = gst_lamemp3enc_finish_frames (lame);
    } else {
      gst_buffer_unmap (mp3_buf, &mp3_map);
      if (mp3_size < 0) {
        /* eat error ? */
        g_warning ("error %d", mp3_size);
      }
      gst_buffer_unref (mp3_buf);
      result = GST_FLOW_OK;
    }
  
    return result;
  }
  
  /* set up the encoder state */
  static gboolean
  gst_lamemp3enc_setup (GstLameMP3Enc * lame, GstTagList ** tags)
  {
    gboolean res;
  
  #define CHECK_ERROR(command) G_STMT_START {\
    if ((command) < 0) { \
      GST_ERROR_OBJECT (lame, "setup failed: " G_STRINGIFY (command)); \
      if (*tags) { \
        gst_tag_list_unref (*tags); \
        *tags = NULL; \
      } \
      return FALSE; \
    } \
  }G_STMT_END
  
    int retval;
    GstCaps *allowed_caps;
  
    GST_DEBUG_OBJECT (lame, "starting setup");
  
    lame->lgf = lame_init ();
  
    if (lame->lgf == NULL)
      return FALSE;
  
    *tags = gst_tag_list_new_empty ();
  
    /* copy the parameters over */
    lame_set_in_samplerate (lame->lgf, lame->samplerate);
  
    /* let lame choose default samplerate unless outgoing sample rate is fixed */
    allowed_caps = gst_pad_get_allowed_caps (GST_AUDIO_ENCODER_SRC_PAD (lame));
  
    if (allowed_caps != NULL) {
      GstStructure *structure;
      gint samplerate;
  
      structure = gst_caps_get_structure (allowed_caps, 0);
  
      if (gst_structure_get_int (structure, "rate", &samplerate)) {
        GST_DEBUG_OBJECT (lame, "Setting sample rate to %d as fixed in src caps",
            samplerate);
        lame_set_out_samplerate (lame->lgf, samplerate);
      } else {
        GST_DEBUG_OBJECT (lame, "Letting lame choose sample rate");
        lame_set_out_samplerate (lame->lgf, 0);
      }
      gst_caps_unref (allowed_caps);
      allowed_caps = NULL;
    } else {
      GST_DEBUG_OBJECT (lame, "No peer yet, letting lame choose sample rate");
      lame_set_out_samplerate (lame->lgf, 0);
    }
  
    CHECK_ERROR (lame_set_num_channels (lame->lgf, lame->num_channels));
    CHECK_ERROR (lame_set_bWriteVbrTag (lame->lgf, 0));
  
    if (lame->target == LAMEMP3ENC_TARGET_QUALITY) {
      CHECK_ERROR (lame_set_VBR (lame->lgf, vbr_default));
      CHECK_ERROR (lame_set_VBR_quality (lame->lgf, lame->quality));
    } else {
      if (lame->cbr) {
        CHECK_AND_FIXUP_BITRATE (lame, "bitrate", lame->bitrate);
        CHECK_ERROR (lame_set_VBR (lame->lgf, vbr_off));
        CHECK_ERROR (lame_set_brate (lame->lgf, lame->bitrate));
      } else {
        CHECK_ERROR (lame_set_VBR (lame->lgf, vbr_abr));
        CHECK_ERROR (lame_set_VBR_mean_bitrate_kbps (lame->lgf, lame->bitrate));
      }
      gst_tag_list_add (*tags, GST_TAG_MERGE_REPLACE, GST_TAG_BITRATE,
          lame->bitrate * 1000, NULL);
    }
  
    if (lame->encoding_engine_quality == LAMEMP3ENC_ENCODING_ENGINE_QUALITY_FAST)
      CHECK_ERROR (lame_set_quality (lame->lgf, 7));
    else if (lame->encoding_engine_quality ==
        LAMEMP3ENC_ENCODING_ENGINE_QUALITY_HIGH)
      CHECK_ERROR (lame_set_quality (lame->lgf, 2));
    /* else default */
  
    if (lame->mono)
      CHECK_ERROR (lame_set_mode (lame->lgf, MONO));
  
    /* initialize the lame encoder */
    if ((retval = lame_init_params (lame->lgf)) >= 0) {
      /* FIXME: it would be nice to print out the mode here */
      GST_INFO
          ("lame encoder setup (target %s, quality %f, bitrate %d, %d Hz, %d channels)",
          (lame->target == LAMEMP3ENC_TARGET_QUALITY) ? "quality" : "bitrate",
          lame->quality, lame->bitrate, lame->samplerate, lame->num_channels);
      res = TRUE;
    } else {
      GST_ERROR_OBJECT (lame, "lame_init_params returned %d", retval);
      res = FALSE;
    }
  
    GST_DEBUG_OBJECT (lame, "done with setup");
    return res;
  #undef CHECK_ERROR
  }
  
  gboolean
  gst_lamemp3enc_register (GstPlugin * plugin)
  {
    GST_DEBUG_CATEGORY_INIT (debug, "lamemp3enc", 0, "lame mp3 encoder");
  
    if (!gst_element_register (plugin, "lamemp3enc", GST_RANK_PRIMARY,
            GST_TYPE_LAMEMP3ENC))
      return FALSE;
  
    return TRUE;
  }
  