diff --git a/audio/hermite.c b/audio/hermite.c
index 1dc80e3a6f..a74e66d76a 100644
--- a/audio/hermite.c
+++ b/audio/hermite.c
@@ -18,13 +18,20 @@
 #include "resampler.h"
 #include <stdlib.h>
 #include <math.h>
+#include <stdio.h>
 #include "../boolean.h"
-#include "../general.h"
 
 #ifdef HAVE_CONFIG_H
 #include "../config.h"
 #endif
 
+#ifndef RESAMPLER_TEST
+#include "../general.h"
+#else
+#define RARCH_LOG(...) fprintf(stderr, __VA_ARGS__)
+#define RARCH_WARN(...) fprintf(stderr, __VA_ARGS__)
+#endif
+
 #define CHANNELS 2
 
 typedef struct rarch_hermite_resampler
@@ -51,8 +58,11 @@ static inline float hermite_kernel(float mu1, float a, float b, float c, float d
    return (a0 * b) + (a1 * m0) + (a2 * m1) + (a3 * c);
 }
 
-void *resampler_hermite_new(void)
+void *resampler_hermite_new(double bandwidth_mod)
 {
+   if (bandwidth_mod < 1.0)
+      RARCH_WARN("Hermite resampler is likely to sound absolutely terrible when downsampling.\n");
+
 #ifndef RESAMPLER_TEST
    RARCH_LOG("Hermite resampler [C]\n");
 #endif
diff --git a/audio/resampler.c b/audio/resampler.c
index 8ee2f3e36a..644c23882a 100644
--- a/audio/resampler.c
+++ b/audio/resampler.c
@@ -29,7 +29,7 @@ static const rarch_resampler_t *backends[] = {
    &hermite_resampler,
 };
 
-bool rarch_resampler_realloc(void **re, const rarch_resampler_t **backend, const char *ident)
+bool rarch_resampler_realloc(void **re, const rarch_resampler_t **backend, const char *ident, double bw_ratio)
 {
    if (*re && *backend)
       (*backend)->free(*re);
@@ -54,7 +54,7 @@ bool rarch_resampler_realloc(void **re, const rarch_resampler_t **backend, const
    if (!*backend)
       return false;
 
-   *re = (*backend)->init();
+   *re = (*backend)->init(bw_ratio);
    if (!*re)
    {
       *backend = NULL;
diff --git a/audio/resampler.h b/audio/resampler.h
index a865f2ab5a..981c44f541 100644
--- a/audio/resampler.h
+++ b/audio/resampler.h
@@ -46,7 +46,7 @@ struct resampler_data
 
 typedef struct rarch_resampler
 {
-   void *(*init)(void);
+   void *(*init)(double bandwidth_mod); // Bandwidth factor. Will be < 1.0 for downsampling, > 1.0 for upsamling. Corresponds to expected resampling ratio.
    void (*process)(void *re, struct resampler_data *data);
    void (*free)(void *re);
    const char *ident;
@@ -57,7 +57,7 @@ extern const rarch_resampler_t sinc_resampler;
 
 // Reallocs resampler. Will free previous handle before allocating a new one.
 // If ident is NULL, first resampler will be used.
-bool rarch_resampler_realloc(void **re, const rarch_resampler_t **backend, const char *ident);
+bool rarch_resampler_realloc(void **re, const rarch_resampler_t **backend, const char *ident, double bw_ratio);
 
 // Convenience macros.
 // freep makes sure to set handles to NULL to avoid double-free in rarch_resampler_realloc.
diff --git a/audio/sinc.c b/audio/sinc.c
index b0e911762c..1200c4b0a1 100644
--- a/audio/sinc.c
+++ b/audio/sinc.c
@@ -45,7 +45,7 @@
 #if defined(SINC_LOWEST_QUALITY)
 #define SINC_WINDOW_LANCZOS
 #define CUTOFF 0.98
-#define PHASE_BITS 11
+#define PHASE_BITS 12
 #define SINC_COEFF_LERP 0
 #define SUBPHASE_BITS 10
 #define SIDELOBES 2
@@ -104,18 +104,18 @@
 
 typedef struct rarch_sinc_resampler
 {
-#if SINC_COEFF_LERP
-   sample_t phase_table[1 << PHASE_BITS][TAPS * 2];
-#else
-   sample_t phase_table[1 << PHASE_BITS][TAPS];
-#endif
-   sample_t buffer_l[2 * TAPS];
-   sample_t buffer_r[2 * TAPS];
+   sample_t *phase_table;
+   sample_t *buffer_l;
+   sample_t *buffer_r;
 
    unsigned taps;
 
    unsigned ptr;
    uint32_t time;
+
+   // A buffer for phase_table, buffer_l and buffer_r are created in a single calloc().
+   // Ensure that we get as good cache locality as we can hope for.
+   sample_t *main_buffer;
 } rarch_sinc_resampler_t;
 
 static inline double sinc(double val)
@@ -173,6 +173,7 @@ static void init_sinc_table(rarch_sinc_resampler_t *resamp, double cutoff,
    double window_mod = window_function(0.0); // Need to normalize w(0) to 1.0.
    int stride = calculate_delta ? 2 : 1;
 
+   double sidelobes = taps / 2.0;
    for (int i = 0; i < phases; i++)
    {
       for (int j = 0; j < taps; j++)
@@ -180,7 +181,7 @@ static void init_sinc_table(rarch_sinc_resampler_t *resamp, double cutoff,
          int n = j * phases + i;
          double window_phase = (double)n / (phases * taps); // [0, 1).
          window_phase = 2.0 * window_phase - 1.0; // [-1, 1)
-         double sinc_phase = SIDELOBES * window_phase;
+         double sinc_phase = sidelobes * window_phase;
 
          float val = cutoff * sinc(M_PI * sinc_phase * cutoff) * window_function(window_phase) / window_mod;
          phase_table[i * stride * taps + j] = val;
@@ -189,26 +190,26 @@ static void init_sinc_table(rarch_sinc_resampler_t *resamp, double cutoff,
 
    if (calculate_delta)
    {
-      for (int i = 0; i < phases - 1; i++)
+      for (int p = 0; p < phases - 1; p++)
       {
          for (int j = 0; j < taps; j++)
          {
-            float delta = phase_table[(i + 1) * stride * taps + j] - phase_table[i * stride * taps + j];
-            phase_table[(i * stride + 1) * taps + j] = delta;
+            float delta = phase_table[(p + 1) * stride * taps + j] - phase_table[p * stride * taps + j];
+            phase_table[(p * stride + 1) * taps + j] = delta;
          }
       }
 
-      int i = phases - 1;
+      int phase = phases - 1;
       for (int j = 0; j < taps; j++)
       {
-         int n = j * phases + (i + 1);
-         double window_phase = (double)n / (phases * taps); // [0, 1).
-         window_phase = 2.0 * window_phase - 1.0; // [-1, 1)
-         double sinc_phase = SIDELOBES * window_phase;
+         int n = j * phases + (phase + 1);
+         double window_phase = (double)n / (phases * taps); // (0, 1].
+         window_phase = 2.0 * window_phase - 1.0; // (-1, 1]
+         double sinc_phase = sidelobes * window_phase;
 
          float val = cutoff * sinc(M_PI * sinc_phase * cutoff) * window_function(window_phase) / window_mod;
-         float delta = (val - phase_table[i * stride * taps + j]);
-         phase_table[(i * stride + 1) * taps + j] = delta;
+         float delta = (val - phase_table[phase * stride * taps + j]);
+         phase_table[(phase * stride + 1) * taps + j] = delta;
       }
    }
 }
@@ -240,14 +241,16 @@ static inline void process_sinc_C(rarch_sinc_resampler_t *resamp, float *out_buf
    const float *buffer_l = resamp->buffer_l + resamp->ptr;
    const float *buffer_r = resamp->buffer_r + resamp->ptr;
 
+   unsigned taps  = resamp->taps;
    unsigned phase = resamp->time >> SUBPHASE_BITS;
-   const float *phase_table = resamp->phase_table[phase];
 #if SINC_COEFF_LERP
-   const float *delta_table = phase_table + TAPS;
+   const float *phase_table = resamp->phase_table + phase * taps * 2;
+   const float *delta_table = phase_table + taps;
    float delta = (float)(resamp->time & SUBPHASE_MASK) * SUBPHASE_MOD;
+#else
+   const float *phase_table = resamp->phase_table + phase * taps;
 #endif
 
-   unsigned taps = resamp->taps;
    for (unsigned i = 0; i < taps; i++)
    {
 #if SINC_COEFF_LERP
@@ -273,14 +276,16 @@ static void process_sinc(rarch_sinc_resampler_t *resamp, float *out_buffer)
    const float *buffer_l = resamp->buffer_l + resamp->ptr;
    const float *buffer_r = resamp->buffer_r + resamp->ptr;
 
+   unsigned taps = resamp->taps;
    unsigned phase = resamp->time >> SUBPHASE_BITS;
-   const float *phase_table = resamp->phase_table[phase];
 #if SINC_COEFF_LERP
-   const float *delta_table = phase_table + TAPS;
+   const float *phase_table = resamp->phase_table + phase * taps * 2;
+   const float *delta_table = phase_table + taps;
    __m256 delta = _mm256_set1_ps((float)(resamp->time & SUBPHASE_MASK) * SUBPHASE_MOD);
+#else
+   const float *phase_table = resamp->phase_table + phase * taps;
 #endif
 
-   unsigned taps = resamp->taps;
    for (unsigned i = 0; i < taps; i += 8)
    {
       __m256 buf_l = _mm256_loadu_ps(buffer_l + i);
@@ -321,10 +326,12 @@ static void process_sinc(rarch_sinc_resampler_t *resamp, float *out_buffer)
 
    unsigned taps = resamp->taps;
    unsigned phase = resamp->time >> SUBPHASE_BITS;
-   const float *phase_table = resamp->phase_table[phase];
 #if SINC_COEFF_LERP
+   const float *phase_table = resamp->phase_table + phase * taps * 2;
    const float *delta_table = phase_table + taps;
    __m128 delta = _mm_set1_ps((float)(resamp->time & SUBPHASE_MASK) * SUBPHASE_MOD);
+#else
+   const float *phase_table = resamp->phase_table + phase * taps;
 #endif
 
    for (unsigned i = 0; i < taps; i += 4)
@@ -365,10 +372,6 @@ static void process_sinc(rarch_sinc_resampler_t *resamp, float *out_buffer)
 }
 #elif defined(HAVE_NEON)
 
-#if TAPS < 8
-#error "NEON asm requires at least 8 taps (for now)."
-#endif
-
 #if SINC_COEFF_LERP
 #error "NEON asm does not support SINC lerp."
 #endif
@@ -398,9 +401,6 @@ static void resampler_sinc_process(void *re_, struct resampler_data *data)
 {
    rarch_sinc_resampler_t *re = (rarch_sinc_resampler_t*)re_;
 
-   // If data->ratio is < 1, we are downsampling.
-   // The sinc table is not set up for this, as it always assumes upsampling.
-   // Downsampling will work, but with some added noise due to aliasing might be present.
    uint32_t ratio = PHASES / data->ratio;
 
    const sample_t *input = data->data_in;
@@ -412,9 +412,13 @@ static void resampler_sinc_process(void *re_, struct resampler_data *data)
    {
       while (frames && re->time >= PHASES)
       {
-         re->ptr = (re->ptr - 1) & (TAPS - 1);
-         re->buffer_l[re->ptr + TAPS] = re->buffer_l[re->ptr] = *input++;
-         re->buffer_r[re->ptr + TAPS] = re->buffer_r[re->ptr] = *input++;
+         // Push in reverse to make filter more obvious.
+         if (!re->ptr)
+            re->ptr = re->taps;
+         re->ptr--;
+
+         re->buffer_l[re->ptr + re->taps] = re->buffer_l[re->ptr] = *input++;
+         re->buffer_r[re->ptr + re->taps] = re->buffer_r[re->ptr] = *input++;
 
          re->time -= PHASES;
          frames--;
@@ -434,19 +438,52 @@ static void resampler_sinc_process(void *re_, struct resampler_data *data)
 
 static void resampler_sinc_free(void *re)
 {
-   aligned_free__(re);
+   rarch_sinc_resampler_t *resampler = (rarch_sinc_resampler_t*)re;
+   if (resampler)
+      aligned_free__(resampler->main_buffer);
+   free(resampler);
 }
 
-static void *resampler_sinc_new(void)
+static void *resampler_sinc_new(double bandwidth_mod)
 {
-   rarch_sinc_resampler_t *re = (rarch_sinc_resampler_t*)aligned_alloc__(128, sizeof(*re));
+   rarch_sinc_resampler_t *re = (rarch_sinc_resampler_t*)calloc(1, sizeof(*re));
    if (!re)
       return NULL;
 
    memset(re, 0, sizeof(*re));
 
    re->taps = TAPS;
-   init_sinc_table(re, CUTOFF, &re->phase_table[0][0], 1 << PHASE_BITS, re->taps, SINC_COEFF_LERP);
+   double cutoff = CUTOFF;
+
+   // Downsampling, must lower cutoff, and extend number of taps accordingly to keep same stopband attenuation.
+   if (bandwidth_mod < 1.0)
+   {
+      cutoff *= bandwidth_mod;
+      re->taps = (unsigned)ceil(re->taps / bandwidth_mod);
+   }
+
+   // Be SIMD-friendly.
+#if (defined(__AVX__) && ENABLE_AVX) || defined(HAVE_NEON)
+   re->taps = (re->taps + 7) & ~7;
+#else
+   re->taps = (re->taps + 3) & ~3;
+#endif
+
+   size_t phase_elems = (1 << PHASE_BITS) * re->taps;
+#if SINC_COEFF_LERP
+   phase_elems *= 2;
+#endif
+   size_t elems = phase_elems + 4 * re->taps;
+
+   re->main_buffer = (sample_t*)aligned_alloc__(128, sizeof(sample_t) * elems);
+   if (!re->main_buffer)
+      goto error;
+
+   re->phase_table = re->main_buffer;
+   re->buffer_l = re->main_buffer + phase_elems;
+   re->buffer_r = re->buffer_l + 2 * re->taps;
+
+   init_sinc_table(re, cutoff, re->phase_table, 1 << PHASE_BITS, re->taps, SINC_COEFF_LERP);
 
 #if defined(__AVX__) && ENABLE_AVX
    RARCH_LOG("Sinc resampler [AVX]\n");
@@ -462,8 +499,11 @@ static void *resampler_sinc_new(void)
 #endif
 
    RARCH_LOG("SINC params (%u phase bits, %u taps).\n", PHASE_BITS, re->taps);
-
    return re;
+
+error:
+   resampler_sinc_free(re);
+   return NULL;
 }
 
 const rarch_resampler_t sinc_resampler = {
diff --git a/audio/test/Makefile b/audio/test/Makefile
index d2634f948c..26089ba76c 100644
--- a/audio/test/Makefile
+++ b/audio/test/Makefile
@@ -16,10 +16,10 @@ LDFLAGS += -lm
 
 all: $(TESTS)
 
-test-hermite: ../hermite.o ../utils.o main.o resampler-hermite.o
+test-hermite: hermite.o ../utils.o main.o resampler-hermite.o
 	$(CC) -o $@ $^ $(LDFLAGS)
 
-test-snr-hermite: ../hermite.o ../utils.o snr.o resampler-hermite.o
+test-snr-hermite: hermite.o ../utils.o snr.o resampler-hermite.o
 	$(CC) -o $@ $^ $(LDFLAGS)
 
 resampler-sinc.o: ../resampler.c
@@ -28,6 +28,9 @@ resampler-sinc.o: ../resampler.c
 resampler-hermite.o: ../resampler.c
 	$(CC) -c -o $@ $< $(CFLAGS)
 
+hermite.o: ../hermite.c
+	$(CC) -c -o $@ $< $(CFLAGS)
+
 sinc-lowest.o: ../sinc.c
 	$(CC) -c -o $@ $< $(CFLAGS) -DHAVE_SINC -DSINC_LOWEST_QUALITY
 
diff --git a/audio/test/main.c b/audio/test/main.c
index b51cdb4e04..531efd80cd 100644
--- a/audio/test/main.c
+++ b/audio/test/main.c
@@ -47,7 +47,7 @@ int main(int argc, char *argv[])
 
    const rarch_resampler_t *resampler = NULL;
    void *re = NULL;
-   if (!rarch_resampler_realloc(&re, &resampler, NULL))
+   if (!rarch_resampler_realloc(&re, &resampler, NULL, out_rate / in_rate))
    {
       fprintf(stderr, "Failed to allocate resampler ...\n");
       return 1;
diff --git a/audio/test/sinc_test.m b/audio/test/sinc_test.m
index 24584cc5aa..bbc0200696 100644
--- a/audio/test/sinc_test.m
+++ b/audio/test/sinc_test.m
@@ -2,14 +2,16 @@
 close all;
 
 % 4-tap and 8-tap are Lanczos windowed, but include here for completeness.
-sidelobes = [2 4 8 32 128];
-taps = sidelobes * 2;
-
 phases = 256;
-cutoffs = [0.65 0.75 0.825 0.90 0.95];
+bw = 0.375;
+downsample = round(phases / bw);
+cutoffs = bw * [0.65 0.75 0.825 0.90 0.95];
 betas = [2.0 3.0 5.5 10.5 14.5];
 
-freqs = 0.05 : 0.05 : 0.99;
+sidelobes = round([2 4 8 32 128] / bw);
+taps = sidelobes * 2;
+
+freqs = 0.05 : 0.02 : 0.99;
 
 filters = length(taps);
 for i = 1 : filters
@@ -21,17 +23,17 @@ for i = 1 : filters
     win = kaiser(filter_length, betas(i))';
     filter = s .* win;
 
-    impulse_response_half = 0.5 * upfirdn(1, filter, phases, phases / 2);
+    impulse_response_half = 0.5 * upfirdn(1, filter, phases, downsample);
     figure('name', sprintf('Response SINC: %d taps', taps(i)));
     freqz(impulse_response_half);
     ylim([-200 0]);
 
-    signal = zeros(1, 4001);
+    signal = zeros(1, 8001);
     for freq = freqs
-        signal = signal + sin(pi * freq * (0 : 4000));
+        signal = signal + sin(pi * freq * (0 : 8000));
     end
     
-    resampled = upfirdn(signal, filter, phases, round(phases * 44100 / 48000));
+    resampled = upfirdn(signal, filter, phases, downsample);
     figure('name', sprintf('Kaiser SINC: %d taps, w = %.f', taps(i), freq));
     freqz(resampled .* kaiser(length(resampled), 40.0)');
     ylim([-80 70]);
diff --git a/audio/test/snr.c b/audio/test/snr.c
index ab55461528..b3383d471f 100644
--- a/audio/test/snr.c
+++ b/audio/test/snr.c
@@ -23,6 +23,9 @@
 #include <assert.h>
 #include <stdbool.h>
 
+#undef min
+#define min(a, b) ((a) < (b)) ? (a) : (b)
+
 static void gen_signal(float *out, double omega, double bias_samples, size_t samples)
 {
    for (size_t i = 0; i < samples; i += 2)
@@ -244,15 +247,9 @@ int main(int argc, char *argv[])
 
    const unsigned fft_samples = 1024 * 128;
    unsigned out_rate = fft_samples / 2;
-   unsigned in_rate = out_rate / ratio;
+   unsigned in_rate = round(out_rate / ratio);
    ratio = (double)out_rate / in_rate;
 
-   if (ratio <= 1.0)
-   {
-      fprintf(stderr, "Ratio too low ...\n");
-      return 1;
-   }
-
    static const float freq_list[] = {
       0.001, 0.002, 0.003, 0.004, 0.005, 0.006, 0.007, 0.008, 0.009,
       0.010, 0.015, 0.020, 0.025, 0.030, 0.035, 0.040, 0.045, 0.050,
@@ -265,19 +262,18 @@ int main(int argc, char *argv[])
 
    unsigned samples = in_rate * 4;
    float *input = calloc(sizeof(float), samples);
-   float *output = calloc(sizeof(float), (fft_samples + 1) * 2);
+   float *output = calloc(sizeof(float), (fft_samples + 16) * 2);
    complex double *butterfly_buf = calloc(sizeof(complex double), fft_samples / 2);
    assert(input);
    assert(output);
 
    void *re = NULL;
    const rarch_resampler_t *resampler = NULL;
-   if (!rarch_resampler_realloc(&re, &resampler, NULL))
+   if (!rarch_resampler_realloc(&re, &resampler, NULL, ratio))
       return 1;
 
    test_fft();
 
-   printf("Omega,SNR,Gain\n");
    for (unsigned i = 0; i < sizeof(freq_list) / sizeof(freq_list[0]); i++)
    {
       unsigned freq = freq_list[i] * in_rate;
@@ -293,12 +289,13 @@ int main(int argc, char *argv[])
 
       rarch_resampler_process(resampler, re, &data);
 
-      unsigned out_samples = data.output_frames * 2;
-      assert(out_samples >= fft_samples * 2);
-
       // We generate 2 seconds worth of audio, however, only the last second is considered so phase has stabilized.
       struct snr_result res;
-      calculate_snr(&res, freq, in_rate / 2, output + fft_samples, butterfly_buf, fft_samples);
+      unsigned max_freq = min(in_rate, out_rate) / 2;
+      if (freq > max_freq)
+         continue;
+
+      calculate_snr(&res, freq, max_freq, output + fft_samples - 2048, butterfly_buf, fft_samples);
 
       printf("SNR @ w = %5.3f : %6.2lf dB, Gain: %6.1lf dB\n",
             freq_list[i], res.snr, res.gain);
diff --git a/driver.c b/driver.c
index 48eb414e13..364ecba733 100644
--- a/driver.c
+++ b/driver.c
@@ -406,9 +406,13 @@ void init_audio(void)
       g_extern.audio_data.chunk_size = g_extern.audio_data.nonblock_chunk_size;
    }
 
+   g_extern.audio_data.orig_src_ratio =
+      g_extern.audio_data.src_ratio =
+      (double)g_settings.audio.out_rate / g_settings.audio.in_rate;
+
    const char *resampler = *g_settings.audio.resampler ? g_settings.audio.resampler : NULL;
    if (!rarch_resampler_realloc(&g_extern.audio_data.resampler_data, &g_extern.audio_data.resampler,
-         resampler))
+         resampler, g_extern.audio_data.orig_src_ratio))
    {
       RARCH_ERR("Failed to initialize resampler \"%s\".\n", resampler ? resampler : "(default)");
       g_extern.audio_active = false;
@@ -421,10 +425,6 @@ void init_audio(void)
    rarch_assert(g_settings.audio.out_rate < g_settings.audio.in_rate * AUDIO_MAX_RATIO);
    rarch_assert(g_extern.audio_data.outsamples = (sample_t*)malloc(outsamples_max * sizeof(sample_t)));
 
-   g_extern.audio_data.orig_src_ratio =
-      g_extern.audio_data.src_ratio =
-      (double)g_settings.audio.out_rate / g_settings.audio.in_rate;
-
    if (g_extern.audio_active && g_settings.audio.rate_control)
    {
       if (driver.audio->buffer_size && driver.audio->write_avail)
diff --git a/record/ffemu.c b/record/ffemu.c
index 4728c7ac44..a7ffffe4ff 100644
--- a/record/ffemu.c
+++ b/record/ffemu.c
@@ -281,7 +281,8 @@ static bool ffemu_init_audio(ffemu_t *handle)
 
       rarch_resampler_realloc(&audio->resampler_data,
             &audio->resampler,
-            *g_settings.audio.resampler ? g_settings.audio.resampler : NULL);
+            *g_settings.audio.resampler ? g_settings.audio.resampler : NULL,
+            audio->ratio);
    }
    else
    {