/* -*- c-basic-offset: 2 -*-

 * 

 * GStreamer

 * Copyright (C) 1999-2001 Erik Walthinsen <omega@cse.ogi.edu>

 *               2006 Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>

 *               2007-2009 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.

 * 

 * 

 * TODO:  - Implement the convolution in place, probably only makes sense

 *          when using FFT convolution as currently the convolution itself

 *          is probably the bottleneck

 *        - Maybe allow cascading the filter to get a better stopband attenuation.

 *          Can be done by convolving a filter kernel with itself

 */

#ifdef HAVE_CONFIG_H

#include "config.h"

#endif

#include <string.h>

#include <math.h>

#include <gst/gst.h>

#include <gst/audio/gstaudiofilter.h>

#include <gst/controller/gstcontroller.h>

#include "audiofxbasefirfilter.h"

#define GST_CAT_DEFAULT gst_audio_fx_base_fir_filter_debug

GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);

#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_fx_base_fir_filter_debug, "audiofxbasefirfilter", 0, \

      "FIR filter base class");

GST_BOILERPLATE_FULL (GstAudioFXBaseFIRFilter, gst_audio_fx_base_fir_filter,

    GstAudioFilter, GST_TYPE_AUDIO_FILTER, DEBUG_INIT);

static GstFlowReturn gst_audio_fx_base_fir_filter_transform (GstBaseTransform *

    base, GstBuffer * inbuf, GstBuffer * outbuf);

static gboolean gst_audio_fx_base_fir_filter_start (GstBaseTransform * base);

static gboolean gst_audio_fx_base_fir_filter_stop (GstBaseTransform * base);

static gboolean gst_audio_fx_base_fir_filter_event (GstBaseTransform * base,

    GstEvent * event);

static gboolean gst_audio_fx_base_fir_filter_setup (GstAudioFilter * base,

    GstRingBufferSpec * format);

static gboolean gst_audio_fx_base_fir_filter_query (GstPad * pad,

    GstQuery * query);

static const GstQueryType *gst_audio_fx_base_fir_filter_query_type (GstPad *

    pad);

/* Element class */

static void

gst_audio_fx_base_fir_filter_dispose (GObject * object)

{

  GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (object);

  if (self->residue) {

    g_free (self->residue);

    self->residue = NULL;

  }

  if (self->kernel) {

    g_free (self->kernel);

    self->kernel = NULL;

  }

  G_OBJECT_CLASS (parent_class)->dispose (object);

}

static void

gst_audio_fx_base_fir_filter_base_init (gpointer g_class)

{

  GstCaps *caps;

  caps = gst_caps_from_string (ALLOWED_CAPS);

  gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (g_class),

      caps);

  gst_caps_unref (caps);

}

static void

gst_audio_fx_base_fir_filter_class_init (GstAudioFXBaseFIRFilterClass * klass)

{

  GObjectClass *gobject_class = (GObjectClass *) klass;

  GstBaseTransformClass *trans_class = (GstBaseTransformClass *) klass;

  GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;

  gobject_class->dispose = gst_audio_fx_base_fir_filter_dispose;

  trans_class->transform =

      GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_transform);

  trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_start);

  trans_class->stop = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_stop);

  trans_class->event = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_event);

  filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_setup);

}

static void

gst_audio_fx_base_fir_filter_init (GstAudioFXBaseFIRFilter * self,

    GstAudioFXBaseFIRFilterClass * g_class)

{

  self->kernel = NULL;

  self->residue = NULL;

  self->next_ts = GST_CLOCK_TIME_NONE;

  self->next_off = GST_BUFFER_OFFSET_NONE;

  gst_pad_set_query_function (GST_BASE_TRANSFORM (self)->srcpad,

      gst_audio_fx_base_fir_filter_query);

  gst_pad_set_query_type_function (GST_BASE_TRANSFORM (self)->srcpad,

      gst_audio_fx_base_fir_filter_query_type);

}

#define DEFINE_PROCESS_FUNC(width,ctype) \

static void \

process_##width (GstAudioFXBaseFIRFilter * self, g##ctype * src, g##ctype * dst, guint input_samples) \

{ \

  gint kernel_length = self->kernel_length; \

  gint i, j, k, l; \

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

  gint res_start; \

  \

  /* convolution */ \

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

    dst[i] = 0.0; \

    k = i % channels; \

    l = i / channels; \

    for (j = 0; j < kernel_length; j++) \

      if (l < j) \

        dst[i] += \

            self->residue[(kernel_length + l - j) * channels + \

            k] * self->kernel[j]; \

      else \

        dst[i] += src[(l - j) * channels + k] * self->kernel[j]; \

  } \

  \

  /* copy the tail of the current input buffer to the residue, while \

   * keeping parts of the residue if the input buffer is smaller than \

   * the kernel length */ \

  if (input_samples < kernel_length * channels) \

    res_start = kernel_length * channels - input_samples; \

  else \

    res_start = 0; \

  \

  for (i = 0; i < res_start; i++) \

    self->residue[i] = self->residue[i + input_samples]; \

  for (i = res_start; i < kernel_length * channels; i++) \

    self->residue[i] = src[input_samples - kernel_length * channels + i]; \

  \

  self->residue_length += kernel_length * channels - res_start; \

  if (self->residue_length > kernel_length * channels) \

    self->residue_length = kernel_length * channels; \

}

DEFINE_PROCESS_FUNC (32, float);

DEFINE_PROCESS_FUNC (64, double);

#undef DEFINE_PROCESS_FUNC

void

gst_audio_fx_base_fir_filter_push_residue (GstAudioFXBaseFIRFilter * self)

{

  GstBuffer *outbuf;

  GstFlowReturn res;

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

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

  gint outsize, outsamples;

  gint diffsize, diffsamples;

  guint8 *in, *out;

  if (channels == 0 || rate == 0) {

    self->residue_length = 0;

    return;

  }

  /* Calculate the number of samples and their memory size that

   * should be pushed from the residue */

  outsamples = MIN (self->latency, self->residue_length / channels);

  outsize = outsamples * channels * (GST_AUDIO_FILTER (self)->format.width / 8);

  if (outsize == 0) {

    self->residue_length = 0;

    return;

  }

  /* Process the difference between latency and residue_length samples

   * to start at the actual data instead of starting at the zeros before

   * when we only got one buffer smaller than latency */

  diffsamples = self->latency - self->residue_length / channels;

  diffsize =

      diffsamples * channels * (GST_AUDIO_FILTER (self)->format.width / 8);

  if (diffsize > 0) {

    in = g_new0 (guint8, diffsize);

    out = g_new0 (guint8, diffsize);

    self->process (self, in, out, diffsamples * channels);

    g_free (in);

    g_free (out);

  }

  res = gst_pad_alloc_buffer (GST_BASE_TRANSFORM (self)->srcpad,

      GST_BUFFER_OFFSET_NONE, outsize,

      GST_PAD_CAPS (GST_BASE_TRANSFORM (self)->srcpad), &outbuf);

  if (G_UNLIKELY (res != GST_FLOW_OK)) {

    GST_WARNING_OBJECT (self, "failed allocating buffer of %d bytes", outsize);

    self->residue_length = 0;

    return;

  }

  /* Convolve the residue with zeros to get the actual remaining data */

  in = g_new0 (guint8, outsize);

  self->process (self, in, GST_BUFFER_DATA (outbuf), outsamples * channels);

  g_free (in);

  /* Set timestamp, offset, etc from the values we

   * saved when processing the regular buffers */

  if (GST_CLOCK_TIME_IS_VALID (self->next_ts))

    GST_BUFFER_TIMESTAMP (outbuf) = self->next_ts;

  else

    GST_BUFFER_TIMESTAMP (outbuf) = 0;

  GST_BUFFER_DURATION (outbuf) =

      gst_util_uint64_scale (outsamples, GST_SECOND, rate);

  self->next_ts += gst_util_uint64_scale (outsamples, GST_SECOND, rate);

  if (self->next_off != GST_BUFFER_OFFSET_NONE) {

    GST_BUFFER_OFFSET (outbuf) = self->next_off;

    GST_BUFFER_OFFSET_END (outbuf) = self->next_off + outsamples;

    self->next_off = GST_BUFFER_OFFSET_END (outbuf);

  }

  GST_DEBUG_OBJECT (self, "Pushing residue buffer of size %d with timestamp: %"

      GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %lld,"

      " offset_end: %lld, nsamples: %d", GST_BUFFER_SIZE (outbuf),

      GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),

      GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf),

      GST_BUFFER_OFFSET_END (outbuf), outsamples);

  res = gst_pad_push (GST_BASE_TRANSFORM (self)->srcpad, outbuf);

  if (G_UNLIKELY (res != GST_FLOW_OK)) {

    GST_WARNING_OBJECT (self, "failed to push residue");

  }

  self->residue_length = 0;

}

/* GstAudioFilter vmethod implementations */

/* get notified of caps and plug in the correct process function */

static gboolean

gst_audio_fx_base_fir_filter_setup (GstAudioFilter * base,

    GstRingBufferSpec * format)

{

  GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);

  gboolean ret = TRUE;

  if (self->residue) {

    gst_audio_fx_base_fir_filter_push_residue (self);

    g_free (self->residue);

    self->residue = NULL;

    self->residue_length = 0;

    self->next_ts = GST_CLOCK_TIME_NONE;

    self->next_off = GST_BUFFER_OFFSET_NONE;

  }

  if (format->width == 32)

    self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_32;

  else if (format->width == 64)

    self->process = (GstAudioFXBaseFIRFilterProcessFunc) process_64;

  else

    ret = FALSE;

  return TRUE;

}

/* GstBaseTransform vmethod implementations */

static GstFlowReturn

gst_audio_fx_base_fir_filter_transform (GstBaseTransform * base,

    GstBuffer * inbuf, GstBuffer * outbuf)

{

  GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);

  GstClockTime timestamp;

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

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

  gint input_samples =

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

  gint output_samples = input_samples;

  gint diff = 0;

  timestamp = GST_BUFFER_TIMESTAMP (outbuf);

  if (!GST_CLOCK_TIME_IS_VALID (timestamp)) {

    GST_ERROR_OBJECT (self, "Invalid timestamp");

    return GST_FLOW_ERROR;

  }

  gst_object_sync_values (G_OBJECT (self), timestamp);

  g_return_val_if_fail (self->kernel != NULL, GST_FLOW_ERROR);

  g_return_val_if_fail (channels != 0, GST_FLOW_ERROR);

  if (!self->residue)

    self->residue = g_new0 (gdouble, self->kernel_length * channels);

  /* Reset the residue if already existing on discont buffers */

  if (GST_BUFFER_IS_DISCONT (inbuf) || (GST_CLOCK_TIME_IS_VALID (self->next_ts)

          && timestamp - gst_util_uint64_scale (MIN (self->latency,

                  self->residue_length / channels), GST_SECOND,

              rate) - self->next_ts > 5 * GST_MSECOND)) {

    GST_DEBUG_OBJECT (self, "Discontinuity detected - flushing");

    if (GST_CLOCK_TIME_IS_VALID (self->next_ts))

      gst_audio_fx_base_fir_filter_push_residue (self);

    self->residue_length = 0;

    self->next_ts = timestamp;

    self->next_off = GST_BUFFER_OFFSET (inbuf);

  } else if (!GST_CLOCK_TIME_IS_VALID (self->next_ts)) {

    self->next_ts = timestamp;

    self->next_off = GST_BUFFER_OFFSET (inbuf);

  }

  /* Calculate the number of samples we can push out now without outputting

   * latency zeros in the beginning */

  diff = self->latency * channels - self->residue_length;

  if (diff > 0)

    output_samples -= diff;

  self->process (self, GST_BUFFER_DATA (inbuf), GST_BUFFER_DATA (outbuf),

      input_samples);

  if (output_samples <= 0) {

    return GST_BASE_TRANSFORM_FLOW_DROPPED;

  }

  GST_BUFFER_TIMESTAMP (outbuf) = self->next_ts;

  GST_BUFFER_DURATION (outbuf) =

      gst_util_uint64_scale (output_samples / channels, GST_SECOND, rate);

  GST_BUFFER_OFFSET (outbuf) = self->next_off;

  if (GST_BUFFER_OFFSET_IS_VALID (outbuf))

    GST_BUFFER_OFFSET_END (outbuf) = self->next_off + output_samples / channels;

  else

    GST_BUFFER_OFFSET_END (outbuf) = GST_BUFFER_OFFSET_NONE;

  if (output_samples < input_samples) {

    GST_BUFFER_DATA (outbuf) +=

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

    GST_BUFFER_SIZE (outbuf) -=

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

  }

  self->next_ts += GST_BUFFER_DURATION (outbuf);

  self->next_off = GST_BUFFER_OFFSET_END (outbuf);

  GST_DEBUG_OBJECT (self, "Pushing buffer of size %d with timestamp: %"

      GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %lld,"

      " offset_end: %lld, nsamples: %d", GST_BUFFER_SIZE (outbuf),

      GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),

      GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf),

      GST_BUFFER_OFFSET_END (outbuf), output_samples / channels);

  return GST_FLOW_OK;

}

static gboolean

gst_audio_fx_base_fir_filter_start (GstBaseTransform * base)

{

  GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);

  self->residue_length = 0;

  self->next_ts = GST_CLOCK_TIME_NONE;

  self->next_off = GST_BUFFER_OFFSET_NONE;

  return TRUE;

}

static gboolean

gst_audio_fx_base_fir_filter_stop (GstBaseTransform * base)

{

  GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);

  g_free (self->residue);

  self->residue = NULL;

  return TRUE;

}

static gboolean

gst_audio_fx_base_fir_filter_query (GstPad * pad, GstQuery * query)

{

  GstAudioFXBaseFIRFilter *self =

      GST_AUDIO_FX_BASE_FIR_FILTER (gst_pad_get_parent (pad));

  gboolean res = TRUE;

  switch (GST_QUERY_TYPE (query)) {

    case GST_QUERY_LATENCY:

    {

      GstClockTime min, max;

      gboolean live;

      guint64 latency;

      GstPad *peer;

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

      if (rate == 0) {

        res = FALSE;

      } else if ((peer = gst_pad_get_peer (GST_BASE_TRANSFORM (self)->sinkpad))) {

        if ((res = gst_pad_query (peer, query))) {

          gst_query_parse_latency (query, &live, &min, &max);

          GST_DEBUG_OBJECT (self, "Peer latency: min %"

              GST_TIME_FORMAT " max %" GST_TIME_FORMAT,

              GST_TIME_ARGS (min), GST_TIME_ARGS (max));

          /* add our own latency */

          latency = gst_util_uint64_scale (self->latency, GST_SECOND, rate);

          GST_DEBUG_OBJECT (self, "Our latency: %"

              GST_TIME_FORMAT, GST_TIME_ARGS (latency));

          min += latency;

          if (max != GST_CLOCK_TIME_NONE)

            max += latency;

          GST_DEBUG_OBJECT (self, "Calculated total latency : min %"

              GST_TIME_FORMAT " max %" GST_TIME_FORMAT,

              GST_TIME_ARGS (min), GST_TIME_ARGS (max));

          gst_query_set_latency (query, live, min, max);

        }

        gst_object_unref (peer);

      }

      break;

    }

    default:

      res = gst_pad_query_default (pad, query);

      break;

  }

  gst_object_unref (self);

  return res;

}

static const GstQueryType *

gst_audio_fx_base_fir_filter_query_type (GstPad * pad)

{

  static const GstQueryType types[] = {

    GST_QUERY_LATENCY,

    0

  };

  return types;

}

static gboolean

gst_audio_fx_base_fir_filter_event (GstBaseTransform * base, GstEvent * event)

{

  GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);

  switch (GST_EVENT_TYPE (event)) {

    case GST_EVENT_EOS:

      gst_audio_fx_base_fir_filter_push_residue (self);

      self->next_ts = GST_CLOCK_TIME_NONE;

      self->next_off = GST_BUFFER_OFFSET_NONE;

      break;

    default:

      break;

  }

  return GST_BASE_TRANSFORM_CLASS (parent_class)->event (base, event);

}

void

gst_audio_fx_base_fir_filter_set_kernel (GstAudioFXBaseFIRFilter * self,

    gdouble * kernel, guint kernel_length, guint64 latency)

{

  g_return_if_fail (kernel != NULL);

  g_return_if_fail (self != NULL);

  GST_BASE_TRANSFORM_LOCK (self);

  if (self->residue) {

    gst_audio_fx_base_fir_filter_push_residue (self);

    self->next_ts = GST_CLOCK_TIME_NONE;

    self->next_off = GST_BUFFER_OFFSET_NONE;

    self->residue_length = 0;

  }

  g_free (self->kernel);

  g_free (self->residue);

  self->kernel = kernel;

  self->kernel_length = kernel_length;

  if (GST_AUDIO_FILTER (self)->format.channels) {

    self->residue =

        g_new0 (gdouble,

        kernel_length * GST_AUDIO_FILTER (self)->format.channels);

    self->residue_length = 0;

  }

  if (self->latency != latency) {

    self->latency = latency;

    gst_element_post_message (GST_ELEMENT (self),

        gst_message_new_latency (GST_OBJECT (self)));

  }

  GST_BASE_TRANSFORM_UNLOCK (self);

}