/* 

 * 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.

 */

/**

 * SECTION:element-audioecho

 *

 * 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.

 *

 * audioecho can also be used to apply a configurable delay to audio channels

 * by setting surround-delay=true. In that mode, it just delays "surround

 * channels" by the delay amount instead of performing an echo. The

 * channels that are configured surround channels for the delay are

 * selected using the surround-channels mask property.

 *

 * <refsect2>

 * <title>Example launch lines</title>

 * |[

 * gst-launch-1.0 autoaudiosrc ! audioconvert ! audioecho delay=500000000 intensity=0.6 feedback=0.4 ! audioconvert ! autoaudiosink

 * gst-launch-1.0 filesrc location="melo1.ogg" ! decodebin ! audioconvert ! audioecho delay=50000000 intensity=0.6 feedback=0.4 ! audioconvert ! autoaudiosink

 * gst-launch-1.0 audiotestsrc ! audioconvert ! audio/x-raw,channels=4 ! audioecho surround-delay=true delay=500000000 ! 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 "audioecho.h"

#define GST_CAT_DEFAULT gst_audio_echo_debug

GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);

/* Everything except the first 2 channels are considered surround */

#define DEFAULT_SURROUND_MASK ~((guint64)(0x3))

enum

{

  PROP_0,

  PROP_DELAY,

  PROP_MAX_DELAY,

  PROP_INTENSITY,

  PROP_FEEDBACK,

  PROP_SUR_DELAY,

  PROP_SUR_MASK

};

#define ALLOWED_CAPS \

    "audio/x-raw,"                                                 \

    " format=(string) {"GST_AUDIO_NE(F32)","GST_AUDIO_NE(F64)"}, " \

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

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

    " layout=(string) interleaved"

#define gst_audio_echo_parent_class parent_class

G_DEFINE_TYPE (GstAudioEcho, gst_audio_echo, GST_TYPE_AUDIO_FILTER);

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,

    const GstAudioInfo * info);

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_class_init (GstAudioEchoClass * klass)

{

  GObjectClass *gobject_class = (GObjectClass *) klass;

  GstElementClass *gstelement_class = (GstElementClass *) klass;

  GstBaseTransformClass *basetransform_class = (GstBaseTransformClass *) klass;

  GstAudioFilterClass *audioself_class = (GstAudioFilterClass *) klass;

  GstCaps *caps;

  GST_DEBUG_CATEGORY_INIT (gst_audio_echo_debug, "audioecho", 0,

      "audioecho element");

  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_MUTABLE_READY));

  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));

  g_object_class_install_property (gobject_class, PROP_SUR_DELAY,

      g_param_spec_boolean ("surround-delay", "Enable Surround Delay",

          "Delay Surround Channels when TRUE instead of applying an echo effect",

          FALSE,

          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | GST_PARAM_CONTROLLABLE));

  g_object_class_install_property (gobject_class, PROP_SUR_MASK,

      g_param_spec_uint64 ("surround-mask", "Surround Mask",

          "A bitmask of channels that are considered surround and delayed when surround-delay = TRUE",

          1, G_MAXUINT64, DEFAULT_SURROUND_MASK,

          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |

          GST_PARAM_MUTABLE_READY));

  gst_element_class_set_static_metadata (gstelement_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);

  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)

{

  self->delay = 1;

  self->max_delay = 1;

  self->intensity = 0.0;

  self->feedback = 0.0;

  self->surdelay = FALSE;

  self->surround_mask = DEFAULT_SURROUND_MASK;

  g_mutex_init (&self->lock);

  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_mutex_clear (&self->lock);

  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;

      guint rate;

      g_mutex_lock (&self->lock);

      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);

        if (delay > max_delay) {

          g_free (self->buffer);

          self->buffer = NULL;

        }

      }

      rate = GST_AUDIO_FILTER_RATE (self);

      if (rate > 0)

        self->delay_frames =

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

      g_mutex_unlock (&self->lock);

      break;

    }

    case PROP_MAX_DELAY:{

      guint64 max_delay;

      g_mutex_lock (&self->lock);

      max_delay = g_value_get_uint64 (value);

      if (GST_STATE (self) > GST_STATE_READY) {

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

            " PLAYING or PAUSED state");

      } else {

        self->max_delay = max_delay;

        g_free (self->buffer);

        self->buffer = NULL;

      }

      g_mutex_unlock (&self->lock);

      break;

    }

    case PROP_INTENSITY:{

      g_mutex_lock (&self->lock);

      self->intensity = g_value_get_float (value);

      g_mutex_unlock (&self->lock);

      break;

    }

    case PROP_FEEDBACK:{

      g_mutex_lock (&self->lock);

      self->feedback = g_value_get_float (value);

      g_mutex_unlock (&self->lock);

      break;

    }

    case PROP_SUR_DELAY:{

      g_mutex_lock (&self->lock);

      self->surdelay = g_value_get_boolean (value);

      g_mutex_unlock (&self->lock);

      break;

    }

    case PROP_SUR_MASK:{

      g_mutex_lock (&self->lock);

      self->surround_mask = g_value_get_uint64 (value);

      g_mutex_unlock (&self->lock);

      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:

      g_mutex_lock (&self->lock);

      g_value_set_uint64 (value, self->delay);

      g_mutex_unlock (&self->lock);

      break;

    case PROP_MAX_DELAY:

      g_mutex_lock (&self->lock);

      g_value_set_uint64 (value, self->max_delay);

      g_mutex_unlock (&self->lock);

      break;

    case PROP_INTENSITY:

      g_mutex_lock (&self->lock);

      g_value_set_float (value, self->intensity);

      g_mutex_unlock (&self->lock);

      break;

    case PROP_FEEDBACK:

      g_mutex_lock (&self->lock);

      g_value_set_float (value, self->feedback);

      g_mutex_unlock (&self->lock);

      break;

    case PROP_SUR_DELAY:

      g_mutex_lock (&self->lock);

      g_value_set_boolean (value, self->surdelay);

      g_mutex_unlock (&self->lock);

      break;

    case PROP_SUR_MASK:{

      g_mutex_lock (&self->lock);

      g_value_set_uint64 (value, self->surround_mask);

      g_mutex_unlock (&self->lock);

      break;

    }

    default:

      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);

      break;

  }

}

/* GstAudioFilter vmethod implementations */

static gboolean

gst_audio_echo_setup (GstAudioFilter * base, const GstAudioInfo * info)

{

  GstAudioEcho *self = GST_AUDIO_ECHO (base);

  gboolean ret = TRUE;

  switch (GST_AUDIO_INFO_FORMAT (info)) {

    case GST_AUDIO_FORMAT_F32:

      self->process = (GstAudioEchoProcessFunc)

          gst_audio_echo_transform_float;

      break;

    case GST_AUDIO_FORMAT_F64:

      self->process = (GstAudioEchoProcessFunc)

          gst_audio_echo_transform_double;

      break;

    default:

      ret = FALSE;

      break;

  }

  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_CHANNELS (self); \

  guint i, j; \

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

  gdouble intensity = self->intensity; \

  gdouble feedback = self->feedback; \

  guint buffer_pos = self->buffer_pos; \

  guint buffer_size_frames = self->buffer_size_frames; \

  \

  if (self->surdelay == FALSE) { \

    guint read_pos = ((echo_offset + buffer_pos) % buffer_size_frames) * channels; \

    guint write_pos = (buffer_pos % buffer_size_frames) * channels; \

    guint buffer_size = buffer_size_frames * channels; \

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

      gdouble in = *data; \

      gdouble echo = buffer[read_pos]; \

      type out = in + intensity * echo; \

      \

      *data = out; \

      \

      buffer[write_pos] = in + feedback * echo; \

      read_pos = (read_pos + 1) % buffer_size; \

      write_pos = (write_pos + 1) % buffer_size; \

      data++; \

    } \

    buffer_pos = write_pos / channels; \

  } else { \

    guint64 surround_mask = self->surround_mask; \

    guint read_pos = ((echo_offset + buffer_pos) % buffer_size_frames) * channels; \

    guint write_pos = (buffer_pos % buffer_size_frames) * channels; \

    guint buffer_size = buffer_size_frames * channels; \

    \

    num_samples /= channels; \

    \

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

      guint64 channel_mask = 1; \

      \

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

        if (channel_mask & surround_mask) { \

          gdouble in = data[j]; \

          gdouble echo = buffer[read_pos + j]; \

          type out = echo; \

          \

          data[j] = out; \

          \

          buffer[write_pos + j] = in; \

        } else { \

          gdouble in = data[j]; \

          gdouble echo = buffer[read_pos + j]; \

          type out = in + intensity * echo; \

          \

          data[j] = out; \

          \

          buffer[write_pos + j] = in + feedback * echo; \

        } \

        channel_mask <<= 1; \

      } \

      read_pos = (read_pos + channels) % buffer_size; \

      write_pos = (write_pos + channels) % buffer_size; \

      data += channels; \

    } \

    buffer_pos = write_pos / channels; \

  } \

  self->buffer_pos = buffer_pos; \

}

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;

  GstClockTime timestamp, stream_time;

  GstMapInfo map;

  g_mutex_lock (&self->lock);

  timestamp = GST_BUFFER_TIMESTAMP (buf);

  stream_time =

      gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);

  GST_DEBUG_OBJECT (self, "sync to %" GST_TIME_FORMAT,

      GST_TIME_ARGS (timestamp));

  if (GST_CLOCK_TIME_IS_VALID (stream_time))

    gst_object_sync_values (GST_OBJECT (self), stream_time);

  if (self->buffer == NULL) {

    guint bpf, rate;

    bpf = GST_AUDIO_FILTER_BPF (self);

    rate = GST_AUDIO_FILTER_RATE (self);

    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 * bpf;

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

    self->buffer_pos = 0;

    if (self->buffer == NULL) {

      g_mutex_unlock (&self->lock);

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

      return GST_FLOW_ERROR;

    }

  }

  gst_buffer_map (buf, &map, GST_MAP_READWRITE);

  num_samples = map.size / GST_AUDIO_FILTER_BPS (self);

  self->process (self, map.data, num_samples);

  gst_buffer_unmap (buf, &map);

  g_mutex_unlock (&self->lock);

  return GST_FLOW_OK;

}