Moving lame mp3 encoder plugin from -ugly

https://bugzilla.gnome.org/show_bug.cgi?id=774252

5 files changed, 1090 insertions, 0 deletions

diff --git a/ext/lame/Makefile.am b/ext/lame/Makefile.am
new file mode 100644
index 000000000..5e5b3a02d
--- /dev/null
+++ b/ext/lame/Makefile.am
@@ -0,0 +1,10 @@
+plugin_LTLIBRARIES = libgstlame.la
+
+libgstlame_la_SOURCES = gstlamemp3enc.c plugin.c
+libgstlame_la_CFLAGS = \
+        $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(LAME_CFLAGS)
+libgstlame_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) -lgstaudio-$(GST_API_VERSION) \
+	$(GST_BASE_LIBS) $(GST_LIBS) $(LAME_LIBS)
+libgstlame_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+
+noinst_HEADERS = gstlamemp3enc.h
diff --git a/ext/lame/gstlamemp3enc.c b/ext/lame/gstlamemp3enc.c
new file mode 100644
index 000000000..a4a637ad4
--- /dev/null
+++ b/ext/lame/gstlamemp3enc.c
@@ -0,0 +1,934 @@
+/* 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>
+ *
+ * Since: 0.10.12
+ */
+
+#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;
+}
diff --git a/ext/lame/gstlamemp3enc.h b/ext/lame/gstlamemp3enc.h
new file mode 100644
index 000000000..25dbc46f8
--- /dev/null
+++ b/ext/lame/gstlamemp3enc.h
@@ -0,0 +1,83 @@
+/* GStreamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ * 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.
+ */
+
+
+#ifndef __GST_LAMEMP3ENC_H__
+#define __GST_LAMEMP3ENC_H__
+
+
+#include <gst/gst.h>
+#include <gst/audio/gstaudioencoder.h>
+#include <gst/base/gstadapter.h>
+
+G_BEGIN_DECLS
+
+#include <lame/lame.h>
+
+#define GST_TYPE_LAMEMP3ENC \
+  (gst_lamemp3enc_get_type())
+#define GST_LAMEMP3ENC(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_LAMEMP3ENC,GstLameMP3Enc))
+#define GST_LAMEMP3ENC_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_LAMEMP3ENC,GstLameMP3EncClass))
+#define GST_IS_LAMEMP3ENC(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_LAMEMP3ENC))
+#define GST_IS_LAMEMP3ENC_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_LAMEMP3ENC))
+
+typedef struct _GstLameMP3Enc GstLameMP3Enc;
+typedef struct _GstLameMP3EncClass GstLameMP3EncClass;
+
+/**
+ * GstLameMP3Enc:
+ *
+ * Opaque data structure.
+ */
+struct _GstLameMP3Enc {
+  GstAudioEncoder element;
+
+  /*< private >*/
+  gint samplerate;
+  gint out_samplerate;
+  gint num_channels;
+
+  /* properties */
+  gint target;
+  gint bitrate;
+  gboolean cbr;
+  gfloat quality;
+  gint encoding_engine_quality;
+  gboolean mono;
+
+  lame_global_flags *lgf;
+
+  GstAdapter *adapter;
+};
+
+struct _GstLameMP3EncClass {
+  GstAudioEncoderClass parent_class;
+};
+
+GType gst_lamemp3enc_get_type(void);
+gboolean gst_lamemp3enc_register (GstPlugin * plugin);
+
+G_END_DECLS
+
+#endif /* __GST_LAMEMP3ENC_H__ */
diff --git a/ext/lame/meson.build b/ext/lame/meson.build
new file mode 100644
index 000000000..2812cb930
--- /dev/null
+++ b/ext/lame/meson.build
@@ -0,0 +1,19 @@
+lame_dep = cc.find_library('mp3lame', required : false)
+
+if lame_dep.found() and cc.has_header_symbol('lame/lame.h', 'lame_init')
+  lame_extra_c_args = []
+  if cc.has_header_symbol('lame/lame.h', 'lame_set_VBR_quality')
+    lame_extra_c_args += ['-DHAVE_LAME_SET_VBR_QUALITY']
+  endif
+  if cc.has_header_symbol('lame/lame.h', 'MEDIUM')
+    lame_extra_c_args += ['-DGSTLAME_PRESET']
+  endif
+  lame = library('gstlame',
+    ['gstlamemp3enc.c', 'plugin.c'],
+    c_args : ugly_args + lame_extra_c_args,
+    include_directories : [configinc, libsinc],
+    dependencies : [gstaudio_dep, lame_dep],
+    install : true,
+    install_dir : plugins_install_dir,
+  )
+endif
diff --git a/ext/lame/plugin.c b/ext/lame/plugin.c
new file mode 100644
index 000000000..6aae21ee3
--- /dev/null
+++ b/ext/lame/plugin.c
@@ -0,0 +1,44 @@
+/* GStreamer
+ * 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.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <gst/gst.h>
+#include <gst/gst-i18n-plugin.h>
+
+#include "gstlamemp3enc.h"
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+#ifdef ENABLE_NLS
+  bindtextdomain (GETTEXT_PACKAGE, LOCALEDIR);
+  bind_textdomain_codeset (GETTEXT_PACKAGE, "UTF-8");
+#endif /* ENABLE_NLS */
+
+  return gst_lamemp3enc_register (plugin);
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+    GST_VERSION_MINOR,
+    lame,
+    "Encode MP3s with LAME",
+    plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);