/* 

 * 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., 59 Temple Place - Suite 330,

 * Boston, MA 02111-1307, USA.

 */

/**

 * SECTION:element-audioecho

 * @Since: 0.10.14

 *

 * audioecho adds an echo or (simple) reverb effect to an audio stream. The echo

 * delay, intensity and the percentage of feedback can be configured.

 *

 * For getting an echo effect you have to set the delay to a larger value,

 * for example 200ms and more. Everything below will result in a simple

 * reverb effect, which results in a slightly metallic sound.

 *

 * Use the max-delay property to set the maximum amount of delay that

 * will be used. This can only be set before going to the PAUSED or PLAYING

 * state and will be set to the current delay by default.

 *

 * <refsect2>

 * <title>Example launch line</title>

 * |[

 * gst-launch filesrc location="melo1.ogg" ! audioconvert ! audioecho delay=500000000 intensity=0.6 feedback=0.4 ! audioconvert ! autoaudiosink

 * gst-launch filesrc location="melo1.ogg" ! decodebin ! audioconvert ! audioecho delay=50000000 intensity=0.6 feedback=0.4 ! audioconvert ! autoaudiosink

 * ]|

 * </refsect2>

 */

#ifdef HAVE_CONFIG_H

#include "config.h"

#endif

#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 "audioecho.h"

#define GST_CAT_DEFAULT gst_audio_echo_debug

GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);

enum

{

  PROP_0,

  PROP_DELAY,

  PROP_MAX_DELAY,

  PROP_INTENSITY,

  PROP_FEEDBACK

};

#define ALLOWED_CAPS \

    "audio/x-raw-float,"                                              \

    " width=(int) { 32, 64 }, "                                       \

    " endianness=(int)BYTE_ORDER,"                                    \

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

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

#define DEBUG_INIT(bla) \

  GST_DEBUG_CATEGORY_INIT (gst_audio_echo_debug, "audioecho", 0, "audioecho element");

GST_BOILERPLATE_FULL (GstAudioEcho, gst_audio_echo, GstAudioFilter,

    GST_TYPE_AUDIO_FILTER, DEBUG_INIT);

static void gst_audio_echo_set_property (GObject * object, guint prop_id,

    const GValue * value, GParamSpec * pspec);

static void gst_audio_echo_get_property (GObject * object, guint prop_id,

    GValue * value, GParamSpec * pspec);

static void gst_audio_echo_finalize (GObject * object);

static gboolean gst_audio_echo_setup (GstAudioFilter * self,

    GstRingBufferSpec * format);

static gboolean gst_audio_echo_stop (GstBaseTransform * base);

static GstFlowReturn gst_audio_echo_transform_ip (GstBaseTransform * base,

    GstBuffer * buf);

static void gst_audio_echo_transform_float (GstAudioEcho * self,

    gfloat * data, guint num_samples);

static void gst_audio_echo_transform_double (GstAudioEcho * self,

    gdouble * data, guint num_samples);

/* GObject vmethod implementations */

static void

gst_audio_echo_base_init (gpointer klass)

{

  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);

  GstCaps *caps;

  gst_element_class_set_details_simple (element_class, "Audio echo",

      "Filter/Effect/Audio",

      "Adds an echo or reverb effect to an audio stream",

      "Sebastian Dröge <sebastian.droege@collabora.co.uk>");

  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_echo_class_init (GstAudioEchoClass * klass)

{

  GObjectClass *gobject_class = (GObjectClass *) klass;

  GstBaseTransformClass *basetransform_class = (GstBaseTransformClass *) klass;

  GstAudioFilterClass *audioself_class = (GstAudioFilterClass *) klass;

  gobject_class->set_property = gst_audio_echo_set_property;

  gobject_class->get_property = gst_audio_echo_get_property;

  gobject_class->finalize = gst_audio_echo_finalize;

  g_object_class_install_property (gobject_class, PROP_DELAY,

      g_param_spec_uint64 ("delay", "Delay",

          "Delay of the echo in nanoseconds", 1, G_MAXUINT64,

          1, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS

          | GST_PARAM_CONTROLLABLE));

  g_object_class_install_property (gobject_class, PROP_MAX_DELAY,

      g_param_spec_uint64 ("max-delay", "Maximum Delay",

          "Maximum delay of the echo in nanoseconds"

          " (can't be changed in PLAYING or PAUSED state)",

          1, G_MAXUINT64, 1,

          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | GST_PARAM_CONTROLLABLE));

  g_object_class_install_property (gobject_class, PROP_INTENSITY,

      g_param_spec_float ("intensity", "Intensity",

          "Intensity of the echo", 0.0, 1.0,

          0.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS

          | GST_PARAM_CONTROLLABLE));

  g_object_class_install_property (gobject_class, PROP_FEEDBACK,

      g_param_spec_float ("feedback", "Feedback",

          "Amount of feedback", 0.0, 1.0,

          0.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS

          | GST_PARAM_CONTROLLABLE));

  audioself_class->setup = GST_DEBUG_FUNCPTR (gst_audio_echo_setup);

  basetransform_class->transform_ip =

      GST_DEBUG_FUNCPTR (gst_audio_echo_transform_ip);

  basetransform_class->stop = GST_DEBUG_FUNCPTR (gst_audio_echo_stop);

}

static void

gst_audio_echo_init (GstAudioEcho * self, GstAudioEchoClass * klass)

{

  self->delay = 1;

  self->max_delay = 1;

  self->intensity = 0.0;

  self->feedback = 0.0;

  gst_base_transform_set_in_place (GST_BASE_TRANSFORM (self), TRUE);

}

static void

gst_audio_echo_finalize (GObject * object)

{

  GstAudioEcho *self = GST_AUDIO_ECHO (object);

  g_free (self->buffer);

  self->buffer = NULL;

  G_OBJECT_CLASS (parent_class)->finalize (object);

}

static void

gst_audio_echo_set_property (GObject * object, guint prop_id,

    const GValue * value, GParamSpec * pspec)

{

  GstAudioEcho *self = GST_AUDIO_ECHO (object);

  switch (prop_id) {

    case PROP_DELAY:{

      guint64 max_delay, delay;

      GST_BASE_TRANSFORM_LOCK (self);

      delay = g_value_get_uint64 (value);

      max_delay = self->max_delay;

      if (delay > max_delay && GST_STATE (self) > GST_STATE_READY) {

        GST_WARNING_OBJECT (self, "New delay (%" GST_TIME_FORMAT ") "

            "is larger than maximum delay (%" GST_TIME_FORMAT ")",

            GST_TIME_ARGS (delay), GST_TIME_ARGS (max_delay));

        self->delay = max_delay;

      } else {

        self->delay = delay;

        self->max_delay = MAX (delay, max_delay);

      }

      GST_BASE_TRANSFORM_UNLOCK (self);

    }

      break;

    case PROP_MAX_DELAY:{

      guint64 max_delay, delay;

      GST_BASE_TRANSFORM_LOCK (self);

      max_delay = g_value_get_uint64 (value);

      delay = self->delay;

      if (GST_STATE (self) > GST_STATE_READY) {

        GST_ERROR_OBJECT (self, "Can't change maximum delay in"

            " PLAYING or PAUSED state");

      } else {

        self->delay = delay;

        self->max_delay = max_delay;

      }

      GST_BASE_TRANSFORM_UNLOCK (self);

    }

      break;

    case PROP_INTENSITY:{

      GST_BASE_TRANSFORM_LOCK (self);

      self->intensity = g_value_get_float (value);

      GST_BASE_TRANSFORM_UNLOCK (self);

    }

      break;

    case PROP_FEEDBACK:{

      GST_BASE_TRANSFORM_LOCK (self);

      self->feedback = g_value_get_float (value);

      GST_BASE_TRANSFORM_UNLOCK (self);

    }

      break;

    default:

      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);

      break;

  }

}

static void

gst_audio_echo_get_property (GObject * object, guint prop_id,

    GValue * value, GParamSpec * pspec)

{

  GstAudioEcho *self = GST_AUDIO_ECHO (object);

  switch (prop_id) {

    case PROP_DELAY:

      GST_BASE_TRANSFORM_LOCK (self);

      g_value_set_uint64 (value, self->delay);

      GST_BASE_TRANSFORM_UNLOCK (self);

      break;

    case PROP_MAX_DELAY:

      GST_BASE_TRANSFORM_LOCK (self);

      g_value_set_uint64 (value, self->max_delay);

      GST_BASE_TRANSFORM_UNLOCK (self);

      break;

    case PROP_INTENSITY:

      GST_BASE_TRANSFORM_LOCK (self);

      g_value_set_float (value, self->intensity);

      GST_BASE_TRANSFORM_UNLOCK (self);

      break;

    case PROP_FEEDBACK:

      GST_BASE_TRANSFORM_LOCK (self);

      g_value_set_float (value, self->feedback);

      GST_BASE_TRANSFORM_UNLOCK (self);

      break;

    default:

      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);

      break;

  }

}

/* GstAudioFilter vmethod implementations */

static gboolean

gst_audio_echo_setup (GstAudioFilter * base, GstRingBufferSpec * format)

{

  GstAudioEcho *self = GST_AUDIO_ECHO (base);

  gboolean ret = TRUE;

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

    self->process = (GstAudioEchoProcessFunc)

        gst_audio_echo_transform_float;

  else if (format->type == GST_BUFTYPE_FLOAT && format->width == 64)

    self->process = (GstAudioEchoProcessFunc)

        gst_audio_echo_transform_double;

  else

    ret = FALSE;

  g_free (self->buffer);

  self->buffer = NULL;

  self->buffer_pos = 0;

  self->buffer_size = 0;

  self->buffer_size_frames = 0;

  return ret;

}

static gboolean

gst_audio_echo_stop (GstBaseTransform * base)

{

  GstAudioEcho *self = GST_AUDIO_ECHO (base);

  g_free (self->buffer);

  self->buffer = NULL;

  self->buffer_pos = 0;

  self->buffer_size = 0;

  self->buffer_size_frames = 0;

  return TRUE;

}

#define TRANSFORM_FUNC(name, type) \

static void \

gst_audio_echo_transform_##name (GstAudioEcho * self, \

    type * data, guint num_samples) \

{ \

  type *buffer = (type *) self->buffer; \

  guint channels = GST_AUDIO_FILTER (self)->format.channels; \

  guint rate = GST_AUDIO_FILTER (self)->format.rate; \

  guint i, j; \

  guint echo_index = self->buffer_size_frames - self->delay_frames; \

  gdouble echo_off = ((((gdouble) self->delay) * rate) / GST_SECOND) - self->delay_frames; \

  \

  if (echo_off < 0.0) \

    echo_off = 0.0; \

  \

  num_samples /= channels; \

  \

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

    guint echo0_index = ((echo_index + self->buffer_pos) % self->buffer_size_frames) * channels; \

    guint echo1_index = ((echo_index + self->buffer_pos +1) % self->buffer_size_frames) * channels; \

    guint rbout_index = (self->buffer_pos % self->buffer_size_frames) * channels; \

    for (j = 0; j < channels; j++) { \

      gdouble in = data[i*channels + j]; \

      gdouble echo0 = buffer[echo0_index + j]; \

      gdouble echo1 = buffer[echo1_index + j]; \

      gdouble echo = echo0 + (echo1-echo0)*echo_off; \

      type out = in + self->intensity * echo; \

      \

      data[i*channels + j] = out; \

      \

      buffer[rbout_index + j] = in + self->feedback * echo; \

    } \

    self->buffer_pos = (self->buffer_pos + 1) % self->buffer_size_frames; \

  } \

}

TRANSFORM_FUNC (float, gfloat);

TRANSFORM_FUNC (double, gdouble);

/* GstBaseTransform vmethod implementations */

static GstFlowReturn

gst_audio_echo_transform_ip (GstBaseTransform * base, GstBuffer * buf)

{

  GstAudioEcho *self = GST_AUDIO_ECHO (base);

  guint num_samples =

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

  if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))

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

  if (self->buffer == NULL) {

    guint width, rate, channels;

    width = GST_AUDIO_FILTER (self)->format.width / 8;

    rate = GST_AUDIO_FILTER (self)->format.rate;

    channels = GST_AUDIO_FILTER (self)->format.channels;

    self->delay_frames =

        MAX (gst_util_uint64_scale (self->delay, rate, GST_SECOND), 1);

    self->buffer_size_frames =

        MAX (gst_util_uint64_scale (self->max_delay, rate, GST_SECOND), 1);

    self->buffer_size = self->buffer_size_frames * width * channels;

    self->buffer = g_try_malloc0 (self->buffer_size);

    self->buffer_pos = 0;

    if (self->buffer == NULL) {

      GST_ERROR_OBJECT (self, "Failed to allocate %u bytes", self->buffer_size);

      return GST_FLOW_ERROR;

    }

  }

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

  return GST_FLOW_OK;

}