/* 

 * GStreamer

 * Copyright (C) 2008 Wim Taymans <wim.taymans@gmail.com>

 *

 * This library is free software; you can redistribute it and/or

 * modify it under the terms of the GNU Library General Public

 * License as published by the Free Software Foundation; either

 * version 2 of the License, or (at your option) any later version.

 *

 * This library is distributed in the hope that it will be useful,

 * but WITHOUT ANY WARRANTY; without even the implied warranty of

 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU

 * Library General Public License for more details.

 *

 * You should have received a copy of the GNU Library General Public

 * License along with this library; if not, write to the

 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,

 * Boston, MA 02111-1307, USA.

 */

/**

 * SECTION:element-audiokaraoke

 *

 * Remove the voice from audio by filtering the center channel.

 * This plugin is useful for karaoke applications.

 *

 * <refsect2>

 * <title>Example launch line</title>

 * |[

 * gst-launch filesrc location=song.ogg ! oggdemux ! vorbisdec ! audiokaraoke ! audioconvert ! alsasink

 * ]|

 * </refsect2>

 */

#ifdef HAVE_CONFIG_H

#include "config.h"

#endif

#include <math.h>

#include <gst/gst.h>

#include <gst/base/gstbasetransform.h>

#include <gst/audio/audio.h>

#include <gst/audio/gstaudiofilter.h>

#include <gst/controller/gstcontroller.h>

#include "audiokaraoke.h"

#define GST_CAT_DEFAULT gst_audio_karaoke_debug

GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);

static const GstElementDetails element_details =

GST_ELEMENT_DETAILS ("AudioKaraoke",

    "Filter/Effect/Audio",

    "Removes voice from sound",

    "Wim Taymans <wim.taymans@gmail.com>");

/* Filter signals and args */

enum

{

  /* FILL ME */

  LAST_SIGNAL

};

#define DEFAULT_LEVEL		1.0

#define DEFAULT_MONO_LEVEL	1.0

#define DEFAULT_FILTER_BAND	220.0

#define DEFAULT_FILTER_WIDTH	100.0

enum

{

  PROP_0,

  PROP_LEVEL,

  PROP_MONO_LEVEL,

  PROP_FILTER_BAND,

  PROP_FILTER_WIDTH,

  PROP_LAST

};

#define ALLOWED_CAPS \

    "audio/x-raw-int,"                                                \

    " depth=(int)16,"                                                 \

    " width=(int)16,"                                                 \

    " endianness=(int)BYTE_ORDER,"                                    \

    " signed=(bool)TRUE,"                                             \

    " rate=(int)[1,MAX],"                                             \

    " channels=(int)[1,MAX]; "                                        \

    "audio/x-raw-float,"                                              \

    " width=(int)32,"                                                 \

    " endianness=(int)BYTE_ORDER,"                                    \

    " rate=(int)[1,MAX],"                                             \

    " channels=(int)[1,MAX]"

#define DEBUG_INIT(bla) \

  GST_DEBUG_CATEGORY_INIT (gst_audio_karaoke_debug, "audiokaraoke", 0, "audiokaraoke element");

GST_BOILERPLATE_FULL (GstAudioKaraoke, gst_audio_karaoke, GstAudioFilter,

    GST_TYPE_AUDIO_FILTER, DEBUG_INIT);

static void gst_audio_karaoke_set_property (GObject * object, guint prop_id,

    const GValue * value, GParamSpec * pspec);

static void gst_audio_karaoke_get_property (GObject * object, guint prop_id,

    GValue * value, GParamSpec * pspec);

static gboolean gst_audio_karaoke_setup (GstAudioFilter * filter,

    GstRingBufferSpec * format);

static GstFlowReturn gst_audio_karaoke_transform_ip (GstBaseTransform * base,

    GstBuffer * buf);

static void gst_audio_karaoke_transform_int (GstAudioKaraoke * filter,

    gint16 * data, guint num_samples);

static void gst_audio_karaoke_transform_float (GstAudioKaraoke * filter,

    gfloat * data, guint num_samples);

/* GObject vmethod implementations */

static void

gst_audio_karaoke_base_init (gpointer klass)

{

  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);

  GstCaps *caps;

  gst_element_class_set_details (element_class, &element_details);

  caps = gst_caps_from_string (ALLOWED_CAPS);

  gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),

      caps);

  gst_caps_unref (caps);

}

static void

gst_audio_karaoke_class_init (GstAudioKaraokeClass * klass)

{

  GObjectClass *gobject_class;

  gobject_class = (GObjectClass *) klass;

  gobject_class->set_property = gst_audio_karaoke_set_property;

  gobject_class->get_property = gst_audio_karaoke_get_property;

  g_object_class_install_property (gobject_class, PROP_LEVEL,

      g_param_spec_float ("level", "Level",

          "Level of the effect (1.0 = full)", 0.0, 1.0, DEFAULT_LEVEL,

          G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));

  g_object_class_install_property (gobject_class, PROP_MONO_LEVEL,

      g_param_spec_float ("mono-level", "Mono Level",

          "Level of the mono channel (1.0 = full)", 0.0, 1.0, DEFAULT_LEVEL,

          G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));

  g_object_class_install_property (gobject_class, PROP_FILTER_BAND,

      g_param_spec_float ("filter-band", "Filter Band",

          "The Frequency band of the filter", 0.0, 441.0, DEFAULT_FILTER_BAND,

          G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));

  g_object_class_install_property (gobject_class, PROP_FILTER_WIDTH,

      g_param_spec_float ("filter-width", "Filter Width",

          "The Frequency width of the filter", 0.0, 100.0, DEFAULT_FILTER_WIDTH,

          G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));

  GST_AUDIO_FILTER_CLASS (klass)->setup =

      GST_DEBUG_FUNCPTR (gst_audio_karaoke_setup);

  GST_BASE_TRANSFORM_CLASS (klass)->transform_ip =

      GST_DEBUG_FUNCPTR (gst_audio_karaoke_transform_ip);

}

static void

gst_audio_karaoke_init (GstAudioKaraoke * filter, GstAudioKaraokeClass * klass)

{

  gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);

  gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (filter), TRUE);

  filter->level = DEFAULT_LEVEL;

  filter->mono_level = DEFAULT_MONO_LEVEL;

  filter->filter_band = DEFAULT_FILTER_BAND;

  filter->filter_width = DEFAULT_FILTER_WIDTH;

}

static void

update_filter (GstAudioKaraoke * filter, gint rate)

{

  gfloat A, B, C;

  if (rate == 0)

    return;

  C = exp (-2 * M_PI * filter->filter_width / rate);

  B = -4 * C / (1 + C) * cos (2 * M_PI * filter->filter_band / rate);

  A = sqrt (1 - B * B / (4 * C)) * (1 - C);

  filter->A = A;

  filter->B = B;

  filter->C = C;

  filter->y1 = 0.0;

  filter->y2 = 0.0;

}

static void

gst_audio_karaoke_set_property (GObject * object, guint prop_id,

    const GValue * value, GParamSpec * pspec)

{

  GstAudioKaraoke *filter;

  filter = GST_AUDIO_KARAOKE (object);

  switch (prop_id) {

    case PROP_LEVEL:

      filter->level = g_value_get_float (value);

      break;

    case PROP_MONO_LEVEL:

      filter->mono_level = g_value_get_float (value);

      break;

    case PROP_FILTER_BAND:

      filter->filter_band = g_value_get_float (value);

      update_filter (filter, filter->rate);

      break;

    case PROP_FILTER_WIDTH:

      filter->filter_width = g_value_get_float (value);

      update_filter (filter, filter->rate);

      break;

    default:

      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);

      break;

  }

}

static void

gst_audio_karaoke_get_property (GObject * object, guint prop_id,

    GValue * value, GParamSpec * pspec)

{

  GstAudioKaraoke *filter;

  filter = GST_AUDIO_KARAOKE (object);

  switch (prop_id) {

    case PROP_LEVEL:

      g_value_set_float (value, filter->level);

      break;

    case PROP_MONO_LEVEL:

      g_value_set_float (value, filter->mono_level);

      break;

    case PROP_FILTER_BAND:

      g_value_set_float (value, filter->filter_band);

      break;

    case PROP_FILTER_WIDTH:

      g_value_set_float (value, filter->filter_width);

      break;

    default:

      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);

      break;

  }

}

/* GstAudioFilter vmethod implementations */

static gboolean

gst_audio_karaoke_setup (GstAudioFilter * base, GstRingBufferSpec * format)

{

  GstAudioKaraoke *filter = GST_AUDIO_KARAOKE (base);

  gboolean ret = TRUE;

  filter->channels = format->channels;

  filter->rate = format->rate;

  if (format->type == GST_BUFTYPE_FLOAT && format->width == 32)

    filter->process = (GstAudioKaraokeProcessFunc)

        gst_audio_karaoke_transform_float;

  else if (format->type == GST_BUFTYPE_LINEAR && format->width == 16)

    filter->process = (GstAudioKaraokeProcessFunc)

        gst_audio_karaoke_transform_int;

  else

    ret = FALSE;

  update_filter (filter, format->rate);

  return ret;

}

static void

gst_audio_karaoke_transform_int (GstAudioKaraoke * filter,

    gint16 * data, guint num_samples)

{

  gint i, l, r, o, x;

  gint channels;

  gdouble y;

  gint level;

  channels = filter->channels;

  level = filter->level * 256;

  for (i = 0; i < num_samples; i += channels) {

    /* get left and right inputs */

    l = data[i];

    r = data[i + 1];

    /* do filtering */

    x = (l + r) / 2;

    y = (filter->A * x - filter->B * filter->y1) - filter->C * filter->y2;

    filter->y2 = filter->y1;

    filter->y1 = y;

    /* filter mono signal */

    o = (int) (y * filter->mono_level);

    o = CLAMP (o, G_MININT16, G_MAXINT16);

    o = (o * level) >> 8;

    /* now cut the center */

    x = l - ((r * level) >> 8) + o;

    r = r - ((l * level) >> 8) + o;

    data[i] = CLAMP (x, G_MININT16, G_MAXINT16);

    data[i + 1] = CLAMP (r, G_MININT16, G_MAXINT16);

  }

}

static void

gst_audio_karaoke_transform_float (GstAudioKaraoke * filter,

    gfloat * data, guint num_samples)

{

  gint i;

  gint channels;

  gdouble l, r, o;

  gdouble y;

  channels = filter->channels;

  for (i = 0; i < num_samples; i += channels) {

    /* get left and right inputs */

    l = data[i];

    r = data[i + 1];

    /* do filtering */

    y = (filter->A * ((l + r) / 2.0) - filter->B * filter->y1) -

        filter->C * filter->y2;

    filter->y2 = filter->y1;

    filter->y1 = y;

    /* filter mono signal */

    o = y * filter->mono_level * filter->level;

    /* now cut the center */

    data[i] = l - (r * filter->level) + o;

    data[i + 1] = r - (l * filter->level) + o;

  }

}

/* GstBaseTransform vmethod implementations */

static GstFlowReturn

gst_audio_karaoke_transform_ip (GstBaseTransform * base, GstBuffer * buf)

{

  GstAudioKaraoke *filter = GST_AUDIO_KARAOKE (base);

  guint num_samples =

      GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);

  if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))

    gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));

  if (gst_base_transform_is_passthrough (base) ||

      G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))

    return GST_FLOW_OK;

  filter->process (filter, GST_BUFFER_DATA (buf), num_samples);

  return GST_FLOW_OK;

}