diff --git a/ext/minigb_apu/LICENSE b/ext/minigb_apu/LICENSE
new file mode 100644
index 0000000..af254b2
--- /dev/null
+++ b/ext/minigb_apu/LICENSE
@@ -0,0 +1,20 @@
+Copyright (c) 2017 Alex Baines <alex@abaines.me.uk>
+Copyright (c) 2019 Mahyar Koshkouei <mk@deltabeard.com>
+
+Permission is hereby granted, free of charge, to any person obtaining a copy
+of this software and associated documentation files (the "Software"), to deal
+in the Software without restriction, including without limitation the rights
+to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+copies of the Software, and to permit persons to whom the Software is
+furnished to do so, subject to the following conditions:
+
+The above copyright notice and this permission notice shall be included in all
+copies or substantial portions of the Software.
+
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+SOFTWARE.
diff --git a/ext/minigb_apu/minigb_apu.c b/ext/minigb_apu/minigb_apu.c
new file mode 100644
index 0000000..2a6ff66
--- /dev/null
+++ b/ext/minigb_apu/minigb_apu.c
@@ -0,0 +1,553 @@
+/**
+ * minigb_apu is released under the terms listed within the LICENSE file.
+ *
+ * minigb_apu emulates the audio processing unit (APU) of the Game Boy. This
+ * project is based on MiniGBS by Alex Baines: https://github.com/baines/MiniGBS
+ */
+
+#include <math.h>
+#include <stdbool.h>
+#include <stdint.h>
+#include <string.h>
+
+#include "minigb_apu.h"
+
+#define AUDIO_MEM_SIZE (0xFF3F - 0xFF10 + 1)
+#define AUDIO_ADDR_COMPENSATION 0xFF10
+
+#define MAX(a, b) ( a > b ? a : b )
+#define MIN(a, b) ( a <= b ? a : b )
+
+/**
+ * Memory holding audio registers between 0xFF10 and 0xFF3F inclusive.
+ */
+static uint8_t audio_mem[AUDIO_MEM_SIZE];
+
+struct chan_len_ctr {
+	uint8_t load;
+	bool enabled;
+	float counter;
+	float inc;
+};
+
+struct chan_vol_env {
+	uint8_t step;
+	bool up;
+	float counter;
+	float inc;
+};
+
+struct chan_freq_sweep {
+	uint_fast16_t freq;
+	uint8_t rate;
+	uint8_t shift;
+	bool up;
+	float counter;
+	float inc;
+};
+
+static struct chan {
+	bool enabled;
+	bool powered;
+	bool on_left;
+	bool on_right;
+	bool muted;
+
+	uint8_t volume;
+	uint8_t volume_init;
+
+	uint16_t freq;
+	float    freq_counter;
+	float    freq_inc;
+
+	int_fast8_t val;
+
+	struct chan_len_ctr    len;
+	struct chan_vol_env    env;
+	struct chan_freq_sweep sweep;
+
+	// square
+	uint8_t duty;
+	uint8_t duty_counter;
+
+	// noise
+	uint16_t lfsr_reg;
+	uint8_t     lfsr_wide;
+	uint8_t      lfsr_div;
+
+	// wave
+	uint8_t sample;
+
+#if ENABLE_HIPASS
+	float capacitor;
+#endif
+} chans[4];
+
+static float vol_l, vol_r;
+
+static float hipass(struct chan *c, float sample)
+{
+#if ENABLE_HIPASS
+	float out    = sample - c->capacitor;
+	c->capacitor = sample - out * 0.996f;
+	return out;
+#else
+	return sample;
+#endif
+}
+
+static void set_note_freq(struct chan *c, const uint_fast16_t freq)
+{
+	c->freq_inc = freq / AUDIO_SAMPLE_RATE;
+}
+
+static void chan_enable(const uint_fast8_t i, const bool enable)
+{
+	chans[i].enabled = enable;
+
+	uint8_t val = (audio_mem[0xFF26 - AUDIO_ADDR_COMPENSATION] & 0x80) |
+		      (chans[3].enabled << 3) | (chans[2].enabled << 2) |
+		      (chans[1].enabled << 1) | (chans[0].enabled << 0);
+
+	audio_mem[0xFF26 - AUDIO_ADDR_COMPENSATION] = val;
+}
+
+static void update_env(struct chan *c)
+{
+	c->env.counter += c->env.inc;
+
+	while (c->env.counter > 1.0f) {
+		if (c->env.step) {
+			c->volume += c->env.up ? 1 : -1;
+			if (c->volume == 0 || c->volume == 15) {
+				c->env.inc = 0;
+			}
+			c->volume = MAX(0, MIN(15, c->volume));
+		}
+		c->env.counter -= 1.0f;
+	}
+}
+
+static void update_len(struct chan *c)
+{
+	if (c->len.enabled) {
+		c->len.counter += c->len.inc;
+		if (c->len.counter > 1.0f) {
+			chan_enable(c - chans, 0);
+			c->len.counter = 0.0f;
+		}
+	}
+}
+
+static bool update_freq(struct chan *c, float *pos)
+{
+	float inc = c->freq_inc - *pos;
+	c->freq_counter += inc;
+
+	if (c->freq_counter > 1.0f) {
+		*pos		= c->freq_inc - (c->freq_counter - 1.0f);
+		c->freq_counter = 0.0f;
+		return true;
+	} else {
+		*pos = c->freq_inc;
+		return false;
+	}
+}
+
+static void update_sweep(struct chan *c)
+{
+	c->sweep.counter += c->sweep.inc;
+
+	while (c->sweep.counter > 1.0f) {
+		if (c->sweep.shift) {
+			uint16_t inc = (c->sweep.freq >> c->sweep.shift);
+			if (!c->sweep.up)
+				inc *= -1;
+
+			c->freq += inc;
+			if (c->freq > 2047) {
+				c->enabled = 0;
+			} else {
+				set_note_freq(c,
+					4194304 / ((2048 - c->freq)<< 5));
+				c->freq_inc *= 8.0f;
+			}
+		} else if (c->sweep.rate) {
+			c->enabled = 0;
+		}
+		c->sweep.counter -= 1.0f;
+	}
+}
+
+static void update_square(float *restrict samples, const bool ch2)
+{
+	struct chan *c = chans + ch2;
+	if (!c->powered)
+		return;
+
+	set_note_freq(c, 4194304.0f / ((2048 - c->freq) << 5));
+	c->freq_inc *= 8.0f;
+
+	for (uint_fast16_t i = 0; i < AUDIO_NSAMPLES; i += 2) {
+		update_len(c);
+
+		if (c->enabled) {
+			update_env(c);
+			if (!ch2)
+				update_sweep(c);
+
+			float pos      = 0.0f;
+			float prev_pos = 0.0f;
+			float sample   = 0.0f;
+
+			while (update_freq(c, &pos)) {
+				c->duty_counter = (c->duty_counter + 1) & 7;
+				sample += ((pos - prev_pos) / c->freq_inc) *
+					  (float)c->val;
+				c->val = (c->duty & (1 << c->duty_counter)) ?
+						 1 :
+						 -1;
+				prev_pos = pos;
+			}
+			sample += ((pos - prev_pos) / c->freq_inc) *
+				  (float)c->val;
+			sample = hipass(c, sample * (c->volume / 15.0f));
+
+			if (!c->muted) {
+				samples[i + 0] +=
+					sample * 0.25f * c->on_left * vol_l;
+				samples[i + 1] +=
+					sample * 0.25f * c->on_right * vol_r;
+			}
+		}
+	}
+}
+
+static uint8_t wave_sample(const unsigned int pos, const unsigned int volume)
+{
+	uint8_t sample =
+		audio_mem[(0xFF30 + pos / 2) - AUDIO_ADDR_COMPENSATION];
+	if (pos & 1) {
+		sample &= 0xF;
+	} else {
+		sample >>= 4;
+	}
+	return volume ? (sample >> (volume - 1)) : 0;
+}
+
+static void update_wave(float *restrict samples)
+{
+	struct chan *c = chans + 2;
+	if (!c->powered)
+		return;
+
+	uint_fast16_t freq = 4194304.0f / ((2048 - c->freq) << 5);
+	set_note_freq(c, freq);
+
+	c->freq_inc *= 16.0f;
+
+	for (uint_fast16_t i = 0; i < AUDIO_NSAMPLES; i += 2) {
+		update_len(c);
+
+		if (c->enabled) {
+			float pos      = 0.0f;
+			float prev_pos = 0.0f;
+			float sample   = 0.0f;
+
+			c->sample = wave_sample(c->val, c->volume);
+
+			while (update_freq(c, &pos)) {
+				c->val = (c->val + 1) & 31;
+				sample += ((pos - prev_pos) / c->freq_inc) *
+					  (float)c->sample;
+				c->sample = wave_sample(c->val, c->volume);
+				prev_pos  = pos;
+			}
+			sample += ((pos - prev_pos) / c->freq_inc) *
+				  (float)c->sample;
+
+			if (c->volume > 0) {
+				float diff = (float[]){ 7.5f, 3.75f,
+							1.5f }[c->volume - 1];
+				sample     = hipass(c, (sample - diff) / 7.5f);
+
+				if (!c->muted) {
+					samples[i + 0] += sample * 0.25f *
+							  c->on_left * vol_l;
+					samples[i + 1] += sample * 0.25f *
+							  c->on_right * vol_r;
+				}
+			}
+		}
+	}
+}
+
+static void update_noise(float *restrict samples)
+{
+	struct chan *c = chans + 3;
+	if (!c->powered)
+		return;
+
+	uint_fast16_t freq = 4194304 / ((uint_fast8_t[]){
+			8, 16, 32, 48, 64, 80, 96, 112
+		}[c->lfsr_div] << c->freq);
+	set_note_freq(c, freq);
+
+	if (c->freq >= 14)
+		c->enabled = 0;
+
+	for (uint_fast16_t i = 0; i < AUDIO_NSAMPLES; i += 2) {
+		update_len(c);
+
+		if (c->enabled) {
+			update_env(c);
+
+			float pos      = 0.0f;
+			float prev_pos = 0.0f;
+			float sample   = 0.0f;
+
+			while (update_freq(c, &pos)) {
+				c->lfsr_reg = (c->lfsr_reg << 1) |
+					      (c->val == 1);
+
+				if (c->lfsr_wide) {
+					c->val = !(((c->lfsr_reg >> 14) & 1) ^
+						   ((c->lfsr_reg >> 13) & 1)) ?
+							 1 :
+							 -1;
+				} else {
+					c->val = !(((c->lfsr_reg >> 6) & 1) ^
+						   ((c->lfsr_reg >> 5) & 1)) ?
+							 1 :
+							 -1;
+				}
+				sample += ((pos - prev_pos) / c->freq_inc) *
+					  c->val;
+				prev_pos = pos;
+			}
+			sample += ((pos - prev_pos) / c->freq_inc) * c->val;
+			sample = hipass(c, sample * (c->volume / 15.0f));
+
+			if (!c->muted) {
+				samples[i + 0] +=
+					sample * 0.25f * c->on_left * vol_l;
+				samples[i + 1] +=
+					sample * 0.25f * c->on_right * vol_r;
+			}
+		}
+	}
+}
+
+/**
+ * SDL2 style audio callback function.
+ */
+void audio_callback(void *userdata, void *restrict stream, int len)
+{
+	/* Appease unused variable warning. */
+	(void)userdata;
+
+	memset(stream, 0, len);
+
+	update_square(stream, 0);
+	update_square(stream, 1);
+	update_wave(stream);
+	update_noise(stream);
+}
+
+static void chan_trigger(uint_fast8_t i)
+{
+	struct chan *c = chans + i;
+
+	chan_enable(i, 1);
+	c->volume = c->volume_init;
+
+	// volume envelope
+	{
+		uint8_t val =
+			audio_mem[(0xFF12 + (i * 5)) - AUDIO_ADDR_COMPENSATION];
+
+		c->env.step = val & 0x07;
+		c->env.up   = val & 0x08 ? 1 : 0;
+		c->env.inc  = c->env.step ? (64.0f / (float)c->env.step) /
+						   AUDIO_SAMPLE_RATE :
+					   8.0f / AUDIO_SAMPLE_RATE;
+		c->env.counter = 0.0f;
+	}
+
+	// freq sweep
+	if (i == 0) {
+		uint8_t val = audio_mem[0xFF10 - AUDIO_ADDR_COMPENSATION];
+
+		c->sweep.freq  = c->freq;
+		c->sweep.rate  = (val >> 4) & 0x07;
+		c->sweep.up    = !(val & 0x08);
+		c->sweep.shift = (val & 0x07);
+		c->sweep.inc   = c->sweep.rate ?
+				       (128.0f / (float)(c->sweep.rate)) /
+					       AUDIO_SAMPLE_RATE :
+				       0;
+		c->sweep.counter = nexttowardf(1.0f, 1.1f);
+	}
+
+	int len_max = 64;
+
+	if (i == 2) { // wave
+		len_max = 256;
+		c->val  = 0;
+	} else if (i == 3) { // noise
+		c->lfsr_reg = 0xFFFF;
+		c->val      = -1;
+	}
+
+	c->len.inc =
+		(256.0f / (float)(len_max - c->len.load)) / AUDIO_SAMPLE_RATE;
+	c->len.counter = 0.0f;
+}
+
+/**
+ * Read audio register.
+ * \param addr	Address of audio register. Must be 0xFF10 <= addr <= 0xFF3F.
+ *				This is not checked in this function.
+ * \return		Byte at address.
+ */
+uint8_t audio_read(const uint16_t addr)
+{
+	static uint8_t ortab[] = { 0x80, 0x3f, 0x00, 0xff, 0xbf, 0xff,
+				   0x3f, 0x00, 0xff, 0xbf, 0x7f, 0xff,
+				   0x9f, 0xff, 0xbf, 0xff, 0xff, 0x00,
+				   0x00, 0xbf, 0x00, 0x00, 0x70 };
+
+	if (addr > 0xFF26)
+		return audio_mem[addr - AUDIO_ADDR_COMPENSATION];
+	
+	return audio_mem[addr - AUDIO_ADDR_COMPENSATION] | ortab[addr - 0xFF10];
+}
+
+/**
+ * Write audio register.
+ * \param addr	Address of audio register. Must be 0xFF10 <= addr <= 0xFF3F.
+ *				This is not checked in this function.
+ * \param val	Byte to write at address.
+ */
+void audio_write(const uint16_t addr, const uint8_t val)
+{
+	/* Find sound channel corresponding to register address. */
+	uint_fast8_t i = (addr - 0xFF10) / 5;
+	audio_mem[addr - AUDIO_ADDR_COMPENSATION] = val;
+
+	switch (addr) {
+	case 0xFF12:
+	case 0xFF17:
+	case 0xFF21: {
+		chans[i].volume_init = val >> 4;
+		chans[i].powered     = (val >> 3) != 0;
+
+		// "zombie mode" stuff, needed for Prehistorik Man and probably
+		// others
+		if (chans[i].powered && chans[i].enabled) {
+			if ((chans[i].env.step == 0 && chans[i].env.inc != 0)) {
+				if (val & 0x08) {
+					chans[i].volume++;
+				} else {
+					chans[i].volume += 2;
+				}
+			} else {
+				chans[i].volume = 16 - chans[i].volume;
+			}
+
+			chans[i].volume &= 0x0F;
+			chans[i].env.step = val & 0x07;
+		}
+	} break;
+
+	case 0xFF1C:
+		chans[i].volume = chans[i].volume_init = (val >> 5) & 0x03;
+		break;
+
+	case 0xFF11:
+	case 0xFF16:
+	case 0xFF20: {
+		const uint8_t duty_lookup[] = { 0x10, 0x30, 0x3C, 0xCF };
+		chans[i].len.load	   = val & 0x3f;
+		chans[i].duty		    = duty_lookup[val >> 6];
+		break;
+	}
+
+	case 0xFF1B:
+		chans[i].len.load = val;
+		break;
+
+	case 0xFF13:
+	case 0xFF18:
+	case 0xFF1D:
+		chans[i].freq &= 0xFF00;
+		chans[i].freq |= val;
+		break;
+
+	case 0xFF1A:
+		chans[i].powered = (val & 0x80) != 0;
+		chan_enable(i, val & 0x80);
+		break;
+
+	case 0xFF14:
+	case 0xFF19:
+	case 0xFF1E:
+		chans[i].freq &= 0x00FF;
+		chans[i].freq |= ((val & 0x07) << 8);
+		/* Intentional fall-through. */
+	case 0xFF23:
+		chans[i].len.enabled = val & 0x40 ? 1 : 0;
+		if (val & 0x80)
+			chan_trigger(i);
+
+		break;
+
+	case 0xFF22:
+		chans[3].freq      = val >> 4;
+		chans[3].lfsr_wide = !(val & 0x08);
+		chans[3].lfsr_div  = val & 0x07;
+		break;
+
+	case 0xFF24:
+		vol_l = ((val >> 4) & 0x07) / 7.0f;
+		vol_r = (val & 0x07) / 7.0f;
+		break;
+
+	case 0xFF25:
+		for (uint_fast8_t i = 0; i < 4; ++i) {
+			chans[i].on_left  = (val >> (4 + i)) & 1;
+			chans[i].on_right = (val >> i) & 1;
+		}
+		break;
+	}
+}
+
+void audio_init(void)
+{
+	/* Initialise channels and samples. */
+	memset(chans, 0, sizeof(chans));
+	chans[0].val = chans[1].val = -1;
+
+	/* Initialise IO registers. */
+	{
+		const uint8_t regs_init[] = { 0x80, 0xBF, 0xF3, 0xFF, 0x3F,
+					      0xFF, 0x3F, 0x00, 0xFF, 0x3F,
+					      0x7F, 0xFF, 0x9F, 0xFF, 0x3F,
+					      0xFF, 0xFF, 0x00, 0x00, 0x3F,
+					      0x77, 0xF3, 0xF1 };
+
+		for(uint_fast8_t i = 0; i < sizeof(regs_init); ++i)
+			audio_write(0xFF10 + i, regs_init[i]);
+	}
+
+	/* Initialise Wave Pattern RAM. */
+	{
+		const uint8_t wave_init[] = { 0xac, 0xdd, 0xda, 0x48,
+					      0x36, 0x02, 0xcf, 0x16,
+					      0x2c, 0x04, 0xe5, 0x2c,
+					      0xac, 0xdd, 0xda, 0x48 };
+
+		for(uint_fast8_t i = 0; i < sizeof(wave_init); ++i)
+			audio_write(0xFF30 + i, wave_init[i]);
+	}
+}
diff --git a/ext/minigb_apu/minigb_apu.h b/ext/minigb_apu/minigb_apu.h
new file mode 100644
index 0000000..7cab051
--- /dev/null
+++ b/ext/minigb_apu/minigb_apu.h
@@ -0,0 +1,48 @@
+/**
+ * minigb_apu is released under the terms listed within the LICENSE file.
+ *
+ * minigb_apu emulates the audio processing unit (APU) of the Game Boy. This
+ * project is based on MiniGBS by Alex Baines: https://github.com/baines/MiniGBS
+ */
+
+#pragma once
+
+#include <stdint.h>
+
+#define DMG_CLOCK_FREQ		4194304.0
+#define SCREEN_REFRESH_CYCLES	70224.0
+#define VERTICAL_SYNC		(DMG_CLOCK_FREQ/SCREEN_REFRESH_CYCLES)
+
+#ifndef ENABLE_HIPASS
+# define ENABLE_HIPASS 1
+#endif
+
+#ifndef AUDIO_SAMPLE_RATE
+# define AUDIO_SAMPLE_RATE	8000.0
+#endif
+
+#ifndef AUDIO_NSAMPLES
+# define AUDIO_NSAMPLES		256
+//((unsigned)(AUDIO_SAMPLE_RATE / VERTICAL_SYNC) * 2)
+#endif
+
+/**
+ * Fill allocated buffer "data" with "len" number of 32-bit floating point
+ * samples (native endian order) in stereo interleaved format.
+ */
+void audio_callback(void *ptr, void *data, int len);
+
+/**
+ * Read audio register at given address "addr".
+ */
+uint8_t audio_read(const uint16_t addr);
+
+/**
+ * Write "val" to audio register at given address "addr".
+ */
+void audio_write(const uint16_t addr, const uint8_t val);
+
+/**
+ * Initialise audio driver.
+ */
+void audio_init(void);