diff --git a/audio/dsp_filter.c b/audio/dsp_filter.c
index 939c6ce455..48ecaf2690 100644
--- a/audio/dsp_filter.c
+++ b/audio/dsp_filter.c
@@ -226,6 +226,7 @@ extern const struct dspfilter_implementation *iir_dspfilter_get_implementation(d
 extern const struct dspfilter_implementation *echo_dspfilter_get_implementation(dspfilter_simd_mask_t mask);
 extern const struct dspfilter_implementation *phaser_dspfilter_get_implementation(dspfilter_simd_mask_t mask);
 extern const struct dspfilter_implementation *wahwah_dspfilter_get_implementation(dspfilter_simd_mask_t mask);
+extern const struct dspfilter_implementation *eq_dspfilter_get_implementation(dspfilter_simd_mask_t mask);
 
 static const dspfilter_get_implementation_t dsp_plugs_builtin[] = {
    panning_dspfilter_get_implementation,
@@ -233,6 +234,7 @@ static const dspfilter_get_implementation_t dsp_plugs_builtin[] = {
    echo_dspfilter_get_implementation,
    phaser_dspfilter_get_implementation,
    wahwah_dspfilter_get_implementation,
+   eq_dspfilter_get_implementation,
 };
 
 static bool append_plugs(rarch_dsp_filter_t *dsp)
diff --git a/audio/filters/eq.c b/audio/filters/eq.c
new file mode 100644
index 0000000000..0b4a5d24f7
--- /dev/null
+++ b/audio/filters/eq.c
@@ -0,0 +1,362 @@
+/*  RetroArch - A frontend for libretro.
+ *  Copyright (C) 2010-2014 - Hans-Kristian Arntzen
+ *
+ *  RetroArch is free software: you can redistribute it and/or modify it under the terms
+ *  of the GNU General Public License as published by the Free Software Found-
+ *  ation, either version 3 of the License, or (at your option) any later version.
+ *
+ *  RetroArch 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 General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License along with RetroArch.
+ *  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include "dspfilter.h"
+#include <stdlib.h>
+#include <string.h>
+#include <stdio.h>
+
+#include "fft/fft.c"
+
+#ifndef M_PI
+#define M_PI 3.1415926535897932384626433832795
+#endif
+
+#ifndef min
+#define min(a, b) ((a) < (b) ? (a) : (b))
+#endif
+
+struct eq_data
+{
+   fft_t *fft;
+   float buffer[8 * 1024];
+
+   float *save;
+   float *block;
+   fft_complex_t *filter;
+   fft_complex_t *fftblock;
+   unsigned block_size;
+   unsigned block_ptr;
+};
+
+struct eq_gain
+{
+   float freq;
+   float gain; // Linear.
+};
+
+static void eq_free(void *data)
+{
+   struct eq_data *eq = (struct eq_data*)data;
+   if (!eq)
+      return;
+
+   fft_free(eq->fft);
+   free(eq->save);
+   free(eq->block);
+   free(eq->fftblock);
+   free(eq->filter);
+   free(eq);
+}
+
+static void eq_process(void *data, struct dspfilter_output *output,
+      const struct dspfilter_input *input)
+{
+   struct eq_data *eq = (struct eq_data*)data;
+
+   output->samples = eq->buffer;
+   output->frames  = 0;
+
+   float *out = eq->buffer;
+   const float *in = input->samples;
+   unsigned input_frames = input->frames;
+
+   while (input_frames)
+   {
+      unsigned write_avail = eq->block_size - eq->block_ptr;
+      if (input_frames < write_avail)
+         write_avail = input_frames;
+
+      memcpy(eq->block + eq->block_ptr * 2, in, write_avail * 2 * sizeof(float));
+
+      in += write_avail * 2;
+      input_frames -= write_avail;
+      eq->block_ptr += write_avail;
+
+      // Convolve a new block.
+      if (eq->block_ptr == eq->block_size)
+      {
+         unsigned i, c;
+
+         for (c = 0; c < 2; c++)
+         {
+            fft_process_forward(eq->fft, eq->fftblock, eq->block + c, 2);
+            for (i = 0; i < 2 * eq->block_size; i++)
+               eq->fftblock[i] = fft_complex_mul(eq->fftblock[i], eq->filter[i]);
+            fft_process_inverse(eq->fft, out + c, eq->fftblock, 2);
+         }
+
+         // Overlap add method, so add in saved block now.
+         for (i = 0; i < 2 * eq->block_size; i++)
+            out[i] += eq->save[i];
+
+         // Save block for later.
+         memcpy(eq->save, out + 2 * eq->block_size, 2 * eq->block_size * sizeof(float));
+
+         out += eq->block_size * 2;
+         output->frames += eq->block_size;
+         eq->block_ptr = 0;
+      }
+   }
+}
+
+static int gains_cmp(const void *a_, const void *b_)
+{
+   const struct eq_gain *a = (const struct eq_gain*)a_;
+   const struct eq_gain *b = (const struct eq_gain*)b_;
+   if (a->freq < b->freq)
+      return -1;
+   else if (a->freq > b->freq)
+      return 1;
+   else
+      return 0;
+}
+
+static void generate_response(fft_complex_t *response,
+      const struct eq_gain *gains, unsigned num_gains, unsigned samples)
+{
+   unsigned i;
+
+   float start_freq = 0.0f;
+   float start_gain = 1.0f;
+
+   float end_freq = 1.0f;
+   float end_gain = 1.0f;
+
+   if (num_gains)
+   {
+      end_freq = gains->freq;
+      end_gain = gains->gain;
+      num_gains--;
+      gains++;
+   }
+
+   // Create a response by linear interpolation between known frequency sample points.
+   for (i = 0; i <= samples; i++)
+   {
+      float freq = (float)i / samples;
+
+      while (freq >= end_freq)
+      {
+         if (num_gains)
+         {
+            start_freq = end_freq;
+            start_gain = end_gain;
+            end_freq = gains->freq;
+            end_gain = gains->gain;
+
+            gains++;
+            num_gains--;
+         }
+         else
+         {
+            start_freq = end_freq;
+            start_gain = end_gain;
+            end_freq = 1.0f;
+            end_gain = 1.0f;
+            break;
+         }
+      }
+
+      float lerp = 0.5f;
+      // Edge case where i == samples.
+      if (end_freq > start_freq)
+         lerp = (freq - start_freq) / (end_freq - start_freq);
+      float gain = (1.0f - lerp) * start_gain + lerp * end_gain;
+
+      response[i].real = gain;
+      response[i].imag = 0.0f;
+      response[2 * samples - i].real = gain;
+      response[2 * samples - i].imag = 0.0f;
+   }
+}
+
+// Modified Bessel function of first order.
+// Check Wiki for mathematical definition ...
+static inline double kaiser_besseli0(double x)
+{
+   unsigned i;
+   double sum = 0.0;
+
+   double factorial = 1.0;
+   double factorial_mult = 0.0;
+   double x_pow = 1.0;
+   double two_div_pow = 1.0;
+   double x_sqr = x * x;
+
+   // Approximate. This is an infinite sum.
+   // Luckily, it converges rather fast.
+   for (i = 0; i < 18; i++)
+   {
+      sum += x_pow * two_div_pow / (factorial * factorial);
+
+      factorial_mult += 1.0;
+      x_pow *= x_sqr;
+      two_div_pow *= 0.25;
+      factorial *= factorial_mult;
+   }
+
+   return sum;
+}
+
+static inline double kaiser_window(double index, double beta)
+{
+   return kaiser_besseli0(beta * sqrt(1 - index * index));
+}
+
+static void create_filter(struct eq_data *eq, unsigned size_log2,
+      struct eq_gain *gains, unsigned num_gains, double beta)
+{
+   int i;
+   int half_block_size = eq->block_size >> 1;
+   double window_mod = 1.0 / kaiser_window(0.0, beta);
+
+   fft_t *fft = fft_new(size_log2);
+   float *time_filter = (float*)calloc(eq->block_size * 2, sizeof(*time_filter));
+   if (!fft || !time_filter)
+      goto end;
+
+   // Make sure bands are in correct order.
+   qsort(gains, num_gains, sizeof(*gains), gains_cmp);
+
+   // Compute desired filter response.
+   generate_response(eq->filter, gains, num_gains, half_block_size);
+
+   // Get equivalent time-domain filter.
+   fft_process_inverse(fft, time_filter, eq->filter, 1);
+
+   // ifftshift() to create the correct linear phase filter.
+   // The filter response was designed with zero phase, which won't work unless we compensate
+   // for the repeating property of the FFT here by flipping left and right blocks.
+   for (i = 0; i < half_block_size; i++)
+   {
+      float tmp = time_filter[i + half_block_size];
+      time_filter[i + half_block_size] = time_filter[i];
+      time_filter[i] = tmp;
+   }
+
+   // Apply a window to smooth out the frequency repsonse.
+   for (i = 0; i < (int)eq->block_size; i++)
+   {
+      // Simple cosine window.
+      double phase = (double)i / (eq->block_size - 1);
+      phase = 2.0 * (phase - 0.5);
+      time_filter[i] *= window_mod * kaiser_window(phase, beta);
+   }
+
+   // Debugging.
+#if 0
+   FILE *file = fopen("/tmp/test.txt", "w");
+   if (file)
+   {
+      for (i = 0; i < (int)eq->block_size; i++)
+         fprintf(file, "%.6f\n", time_filter[i]);
+      fclose(file);
+   }
+#endif
+
+   // Padded FFT to create our FFT filter.
+   fft_process_forward(eq->fft, eq->filter, time_filter, 1);
+
+end:
+   fft_free(fft);
+   free(time_filter);
+}
+
+static void *eq_init(const struct dspfilter_info *info,
+      const struct dspfilter_config *config, void *userdata)
+{
+   unsigned i;
+   struct eq_data *eq = (struct eq_data*)calloc(1, sizeof(*eq));
+   if (!eq)
+      return NULL;
+
+   const float default_freq[] = { 0.0f, info->input_rate };
+   const float default_gain[] = { 0.0f, 0.0f };
+
+   float beta;
+   config->get_float(userdata, "window_beta", &beta, 4.0f);
+
+   int size_log2;
+   config->get_int(userdata, "block_size_log2", &size_log2, 8);
+   unsigned size = 1 << size_log2;
+
+   struct eq_gain *gains = NULL;
+   float *frequencies, *gain;
+   unsigned num_freq, num_gain;
+   config->get_float_array(userdata, "frequencies", &frequencies, &num_freq, default_freq, 2);
+   config->get_float_array(userdata, "gains", &gain, &num_gain, default_gain, 2);
+
+   num_gain = num_freq = min(num_gain, num_freq);
+
+   gains = (struct eq_gain*)calloc(num_gain, sizeof(*gains));
+   if (!gains)
+      goto error;
+
+   for (i = 0; i < num_gain; i++)
+   {
+      gains[i].freq = frequencies[i] / (0.5f * info->input_rate);
+      gains[i].gain = pow(10.0, gain[i] / 20.0);
+   }
+   config->free(frequencies);
+   config->free(gain);
+
+   eq->block_size = size;
+
+   eq->save     = (float*)calloc(    size, 2 * sizeof(*eq->save));
+   eq->block    = (float*)calloc(2 * size, 2 * sizeof(*eq->block));
+   eq->fftblock = (fft_complex_t*)calloc(2 * size, sizeof(*eq->fftblock));
+   eq->filter   = (fft_complex_t*)calloc(2 * size, sizeof(*eq->filter));
+
+   // Use an FFT which is twice the block size with zero-padding
+   // to make circular convolution => proper convolution.
+   eq->fft = fft_new(size_log2 + 1);
+
+   if (!eq->fft || !eq->fftblock || !eq->save || !eq->block || !eq->filter)
+      goto error;
+
+   create_filter(eq, size_log2, gains, num_gain, beta);
+
+   free(gains);
+   return eq;
+
+error:
+   free(gains);
+   eq_free(eq);
+   return NULL;
+}
+
+static const struct dspfilter_implementation eq_plug = {
+   eq_init,
+   eq_process,
+   eq_free,
+
+   DSPFILTER_API_VERSION,
+   "Linear-Phase FFT Equalizer",
+   "eq",
+};
+
+#ifdef HAVE_FILTERS_BUILTIN
+#define dspfilter_get_implementation eq_dspfilter_get_implementation
+#endif
+
+const struct dspfilter_implementation *dspfilter_get_implementation(dspfilter_simd_mask_t mask)
+{
+   (void)mask;
+   return &eq_plug;
+}
+
+#undef dspfilter_get_implementation
+
diff --git a/audio/filters/fft/fft.c b/audio/filters/fft/fft.c
new file mode 100644
index 0000000000..aac3427ee9
--- /dev/null
+++ b/audio/filters/fft/fft.c
@@ -0,0 +1,196 @@
+/*  RetroArch - A frontend for libretro.
+ *  Copyright (C) 2010-2014 - Hans-Kristian Arntzen
+ *
+ *  RetroArch is free software: you can redistribute it and/or modify it under the terms
+ *  of the GNU General Public License as published by the Free Software Found-
+ *  ation, either version 3 of the License, or (at your option) any later version.
+ *
+ *  RetroArch 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 General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License along with RetroArch.
+ *  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include "fft.h"
+#include <math.h>
+#include <stdlib.h>
+
+#ifndef M_PI
+#define M_PI 3.1415926535897932384626433832795
+#endif
+
+struct fft
+{
+   fft_complex_t *interleave_buffer;
+   fft_complex_t *phase_lut;
+   unsigned *bitinverse_buffer;
+   unsigned size;
+};
+
+static unsigned bitswap(unsigned x, unsigned size_log2)
+{
+   unsigned i;
+   unsigned ret = 0;
+   for (i = 0; i < size_log2; i++)
+      ret |= ((x >> i) & 1) << (size_log2 - i - 1);
+   return ret;
+}
+
+static void build_bitinverse(unsigned *bitinverse, unsigned size_log2)
+{
+   unsigned i;
+   unsigned size = 1 << size_log2;
+   for (i = 0; i < size; i++)
+      bitinverse[i] = bitswap(i, size_log2);
+}
+
+static fft_complex_t exp_imag(double phase)
+{
+   fft_complex_t out = { cos(phase), sin(phase) };
+   return out;
+}
+
+static void build_phase_lut(fft_complex_t *out, int size)
+{
+   int i;
+   out += size;
+   for (i = -size; i <= size; i++)
+      out[i] = exp_imag((M_PI * i) / size);
+}
+
+static void interleave_complex(const unsigned *bitinverse,
+      fft_complex_t *out, const fft_complex_t *in,
+      unsigned samples, unsigned step)
+{
+   unsigned i;
+   for (i = 0; i < samples; i++, in += step)
+      out[bitinverse[i]] = *in;
+}
+
+static void interleave_float(const unsigned *bitinverse,
+      fft_complex_t *out, const float *in,
+      unsigned samples, unsigned step)
+{
+   unsigned i;
+   for (i = 0; i < samples; i++, in += step)
+   {
+      unsigned inv_i = bitinverse[i];
+      out[inv_i].real = *in;
+      out[inv_i].imag = 0.0f;
+   }
+}
+
+static void resolve_float(float *out, const fft_complex_t *in, unsigned samples,
+      float gain, unsigned step)
+{
+   unsigned i;
+   for (i = 0; i < samples; i++, in++, out += step)
+      *out = gain * in->real;
+}
+
+fft_t *fft_new(unsigned block_size_log2)
+{
+   fft_t *fft = (fft_t*)calloc(1, sizeof(*fft));
+   if (!fft)
+      return NULL;
+
+   unsigned size = 1 << block_size_log2;
+
+   fft->interleave_buffer = (fft_complex_t*)calloc(size, sizeof(*fft->interleave_buffer));
+   fft->bitinverse_buffer = (unsigned*)calloc(size, sizeof(*fft->bitinverse_buffer));
+   fft->phase_lut         = (fft_complex_t*)calloc(2 * size + 1, sizeof(*fft->phase_lut));
+
+   if (!fft->interleave_buffer || !fft->bitinverse_buffer || !fft->phase_lut)
+      goto error;
+
+   fft->size = size;
+
+   build_bitinverse(fft->bitinverse_buffer, block_size_log2);
+   build_phase_lut(fft->phase_lut, size);
+   return fft;
+
+error:
+   fft_free(fft);
+   return NULL;
+}
+
+void fft_free(fft_t *fft)
+{
+   if (!fft)
+      return;
+
+   free(fft->interleave_buffer);
+   free(fft->bitinverse_buffer);
+   free(fft->phase_lut);
+   free(fft);
+}
+
+static void butterfly(fft_complex_t *a, fft_complex_t *b, fft_complex_t mod)
+{
+   mod = fft_complex_mul(mod, *b);
+   *b = fft_complex_sub(*a, mod);
+   *a = fft_complex_add(*a, mod);
+}
+
+static void butterflies(fft_complex_t *butterfly_buf,
+      const fft_complex_t *phase_lut,
+      int phase_dir, unsigned step_size, unsigned samples)
+{
+   unsigned i, j;
+   for (i = 0; i < samples; i += step_size << 1)
+   {
+      int phase_step = (int)samples * phase_dir / (int)step_size;
+      for (j = i; j < i + step_size; j++)
+         butterfly(&butterfly_buf[j], &butterfly_buf[j + step_size], phase_lut[phase_step * (int)(j - i)]);
+   }
+}
+
+void fft_process_forward_complex(fft_t *fft,
+      fft_complex_t *out, const fft_complex_t *in, unsigned step)
+{
+   unsigned step_size;
+   unsigned samples = fft->size;
+   interleave_complex(fft->bitinverse_buffer, out, in, samples, step);
+
+   for (step_size = 1; step_size < samples; step_size <<= 1)
+   {
+      butterflies(out,
+            fft->phase_lut + samples,
+            -1, step_size, samples);
+   }
+}
+
+void fft_process_forward(fft_t *fft,
+      fft_complex_t *out, const float *in, unsigned step)
+{
+   unsigned step_size;
+   unsigned samples = fft->size;
+   interleave_float(fft->bitinverse_buffer, out, in, samples, step);
+
+   for (step_size = 1; step_size < fft->size; step_size <<= 1)
+   {
+      butterflies(out,
+            fft->phase_lut + samples,
+            -1, step_size, samples);
+   }
+}
+
+void fft_process_inverse(fft_t *fft,
+      float *out, const fft_complex_t *in, unsigned step)
+{
+   unsigned step_size;
+   unsigned samples = fft->size;
+   interleave_complex(fft->bitinverse_buffer, fft->interleave_buffer, in, samples, 1);
+
+   for (step_size = 1; step_size < samples; step_size <<= 1)
+   {
+      butterflies(fft->interleave_buffer,
+            fft->phase_lut + samples,
+            1, step_size, samples);
+   }
+
+   resolve_float(out, fft->interleave_buffer, samples, 1.0f / samples, step);
+}
+
diff --git a/audio/filters/fft/fft.h b/audio/filters/fft/fft.h
new file mode 100644
index 0000000000..3e54ddcfa5
--- /dev/null
+++ b/audio/filters/fft/fft.h
@@ -0,0 +1,88 @@
+/*  RetroArch - A frontend for libretro.
+ *  Copyright (C) 2010-2014 - Hans-Kristian Arntzen
+ *
+ *  RetroArch is free software: you can redistribute it and/or modify it under the terms
+ *  of the GNU General Public License as published by the Free Software Found-
+ *  ation, either version 3 of the License, or (at your option) any later version.
+ *
+ *  RetroArch 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 General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License along with RetroArch.
+ *  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#ifndef RARCH_FFT_H__
+#define RARCH_FFT_H__
+
+typedef struct fft fft_t;
+
+// C99 <complex.h> would be nice.
+typedef struct
+{
+   float real;
+   float imag;
+} fft_complex_t;
+
+static inline fft_complex_t fft_complex_mul(fft_complex_t a,
+      fft_complex_t b)
+{
+   fft_complex_t out = {
+      a.real * b.real - a.imag * b.imag,
+      a.imag * b.real + a.real * b.imag,
+   };
+
+   return out;
+
+}
+
+static inline fft_complex_t fft_complex_add(fft_complex_t a,
+      fft_complex_t b)
+{
+   fft_complex_t out = {
+      a.real + b.real,
+      a.imag + b.imag,
+   };
+
+   return out;
+
+}
+
+static inline fft_complex_t fft_complex_sub(fft_complex_t a,
+      fft_complex_t b)
+{
+   fft_complex_t out = {
+      a.real - b.real,
+      a.imag - b.imag,
+   };
+
+   return out;
+
+}
+
+static inline fft_complex_t fft_complex_conj(fft_complex_t a)
+{
+   fft_complex_t out = {
+      a.real, -a.imag,
+   };
+
+   return out;
+}
+
+fft_t *fft_new(unsigned block_size_log2);
+
+void fft_free(fft_t *fft);
+
+void fft_process_forward_complex(fft_t *fft,
+      fft_complex_t *out, const fft_complex_t *in, unsigned step);
+
+void fft_process_forward(fft_t *fft,
+      fft_complex_t *out, const float *in, unsigned step);
+
+void fft_process_inverse(fft_t *fft,
+      float *out, const fft_complex_t *in, unsigned step);
+
+
+#endif
+
diff --git a/griffin/griffin.c b/griffin/griffin.c
index 930a7d59a4..4a1638f4e5 100644
--- a/griffin/griffin.c
+++ b/griffin/griffin.c
@@ -481,6 +481,7 @@ FILTERS
 #include "../gfx/filters/phosphor2x.c"
 
 #include "../audio/filters/echo.c"
+#include "../audio/filters/eq.c"
 #include "../audio/filters/iir.c"
 #include "../audio/filters/panning.c"
 #include "../audio/filters/phaser.c"