diff --git a/py/circuitpy_defns.mk b/py/circuitpy_defns.mk
index c2c9ffa6736e..61516561822c 100644
--- a/py/circuitpy_defns.mk
+++ b/py/circuitpy_defns.mk
@@ -134,6 +134,9 @@ endif
 ifeq ($(CIRCUITPY_AUDIODELAYS),1)
 SRC_PATTERNS += audiodelays/%
 endif
+ifeq ($(CIRCUITPY_AUDIOFILTERS),1)
+SRC_PATTERNS += audiofilters/%
+endif
 ifeq ($(CIRCUITPY_AUDIOMIXER),1)
 SRC_PATTERNS += audiomixer/%
 endif
@@ -622,6 +625,8 @@ SRC_SHARED_MODULE_ALL = \
 	audiocore/__init__.c \
 	audiodelays/Echo.c \
 	audiodelays/__init__.c \
+	audiofilters/Filter.c \
+	audiofilters/__init__.c \
 	audioio/__init__.c \
 	audiomixer/Mixer.c \
 	audiomixer/MixerVoice.c \
diff --git a/py/circuitpy_mpconfig.mk b/py/circuitpy_mpconfig.mk
index a8e44a12c941..e42555291192 100644
--- a/py/circuitpy_mpconfig.mk
+++ b/py/circuitpy_mpconfig.mk
@@ -144,6 +144,8 @@ CFLAGS += -DCIRCUITPY_AUDIOMP3=$(CIRCUITPY_AUDIOMP3)
 CIRCUITPY_AUDIOEFFECTS ?= 0
 CIRCUITPY_AUDIODELAYS ?= $(CIRCUITPY_AUDIOEFFECTS)
 CFLAGS += -DCIRCUITPY_AUDIODELAYS=$(CIRCUITPY_AUDIODELAYS)
+CIRCUITPY_AUDIOFILTERS ?= $(CIRCUITPY_AUDIOEFFECTS)
+CFLAGS += -DCIRCUITPY_AUDIOFILTERS=$(CIRCUITPY_AUDIOFILTERS)
 
 CIRCUITPY_AURORA_EPAPER ?= 0
 CFLAGS += -DCIRCUITPY_AURORA_EPAPER=$(CIRCUITPY_AURORA_EPAPER)
diff --git a/shared-bindings/audiofilters/Filter.c b/shared-bindings/audiofilters/Filter.c
new file mode 100644
index 000000000000..457897909e0d
--- /dev/null
+++ b/shared-bindings/audiofilters/Filter.c
@@ -0,0 +1,266 @@
+// This file is part of the CircuitPython project: https://circuitpython.org
+//
+// SPDX-FileCopyrightText: Copyright (c) 2024 Cooper Dalrymple
+//
+// SPDX-License-Identifier: MIT
+
+#include <stdint.h>
+
+#include "shared-bindings/audiofilters/Filter.h"
+#include "shared-module/audiofilters/Filter.h"
+
+#include "shared/runtime/context_manager_helpers.h"
+#include "py/binary.h"
+#include "py/objproperty.h"
+#include "py/runtime.h"
+#include "shared-bindings/util.h"
+#include "shared-module/synthio/block.h"
+
+#define MIX_DEFAULT 1.0f
+
+//| class Filter:
+//|     """A Filter effect"""
+//|
+//|     def __init__(
+//|         self,
+//|         filter: Optional[synthio.Biquad] = None,
+//|         mix: synthio.BlockInput = 1.0,
+//|         buffer_size: int = 512,
+//|         sample_rate: int = 8000,
+//|         bits_per_sample: int = 16,
+//|         samples_signed: bool = True,
+//|         channel_count: int = 1,
+//|     ) -> None:
+//|         """Create a Filter effect where the original sample is processed through a biquad filter
+//|            created by a synthio.Synthesizer object. This can be used to generate a low-pass,
+//|            high-pass, or band-pass filter.
+//|
+//|            The mix parameter allows you to change how much of the unchanged sample passes through to
+//|            the output to how much of the effect audio you hear as the output.
+//|
+//|         :param Optional[synthio.Biquad] filter: The normalized biquad filter object used to process the signal.
+//|         :param synthio.BlockInput mix: The mix as a ratio of the sample (0.0) to the effect (1.0).
+//|         :param int buffer_size: The total size in bytes of each of the two playback buffers to use
+//|         :param int sample_rate: The sample rate to be used
+//|         :param int channel_count: The number of channels the source samples contain. 1 = mono; 2 = stereo.
+//|         :param int bits_per_sample: The bits per sample of the effect
+//|         :param bool samples_signed: Effect is signed (True) or unsigned (False)
+//|
+//|         Playing adding a filter to a synth::
+//|
+//|           import time
+//|           import board
+//|           import audiobusio
+//|           import synthio
+//|           import audiofilters
+//|
+//|           audio = audiobusio.I2SOut(bit_clock=board.GP20, word_select=board.GP21, data=board.GP22)
+//|           synth = synthio.Synthesizer(channel_count=1, sample_rate=44100)
+//|           filter = audiofilters.Filter(filter=synth.low_pass_filter(frequency=2000, Q=1.25), buffer_size=1024, channel_count=1, sample_rate=44100, mix=1.0)
+//|           filter.play(synth)
+//|           audio.play(filter)
+//|
+//|           note = synthio.Note(261)
+//|           while True:
+//|               synth.press(note)
+//|               time.sleep(0.25)
+//|               synth.release(note)
+//|               time.sleep(5)"""
+//|         ...
+static mp_obj_t audiofilters_filter_make_new(const mp_obj_type_t *type, size_t n_args, size_t n_kw, const mp_obj_t *all_args) {
+    enum { ARG_filter, ARG_mix, ARG_buffer_size, ARG_sample_rate, ARG_bits_per_sample, ARG_samples_signed, ARG_channel_count, };
+    static const mp_arg_t allowed_args[] = {
+        { MP_QSTR_filter, MP_ARG_OBJ | MP_ARG_KW_ONLY,  {.u_obj = MP_OBJ_NULL} },
+        { MP_QSTR_mix, MP_ARG_OBJ | MP_ARG_KW_ONLY,  {.u_obj = MP_OBJ_NULL} },
+        { MP_QSTR_buffer_size, MP_ARG_INT | MP_ARG_KW_ONLY, {.u_int = 512} },
+        { MP_QSTR_sample_rate, MP_ARG_INT | MP_ARG_KW_ONLY, {.u_int = 8000} },
+        { MP_QSTR_bits_per_sample, MP_ARG_INT | MP_ARG_KW_ONLY, {.u_int = 16} },
+        { MP_QSTR_samples_signed, MP_ARG_BOOL | MP_ARG_KW_ONLY, {.u_bool = true} },
+        { MP_QSTR_channel_count, MP_ARG_INT | MP_ARG_KW_ONLY, {.u_int = 1 } },
+    };
+
+    mp_arg_val_t args[MP_ARRAY_SIZE(allowed_args)];
+    mp_arg_parse_all_kw_array(n_args, n_kw, all_args, MP_ARRAY_SIZE(allowed_args), allowed_args, args);
+
+    mp_int_t channel_count = mp_arg_validate_int_range(args[ARG_channel_count].u_int, 1, 2, MP_QSTR_channel_count);
+    mp_int_t sample_rate = mp_arg_validate_int_min(args[ARG_sample_rate].u_int, 1, MP_QSTR_sample_rate);
+    mp_int_t bits_per_sample = args[ARG_bits_per_sample].u_int;
+    if (bits_per_sample != 8 && bits_per_sample != 16) {
+        mp_raise_ValueError(MP_ERROR_TEXT("bits_per_sample must be 8 or 16"));
+    }
+
+    audiofilters_filter_obj_t *self = mp_obj_malloc(audiofilters_filter_obj_t, &audiofilters_filter_type);
+    common_hal_audiofilters_filter_construct(self, args[ARG_filter].u_obj, args[ARG_mix].u_obj, args[ARG_buffer_size].u_int, bits_per_sample, args[ARG_samples_signed].u_bool, channel_count, sample_rate);
+
+    return MP_OBJ_FROM_PTR(self);
+}
+
+//|     def deinit(self) -> None:
+//|         """Deinitialises the Filter."""
+//|         ...
+static mp_obj_t audiofilters_filter_deinit(mp_obj_t self_in) {
+    audiofilters_filter_obj_t *self = MP_OBJ_TO_PTR(self_in);
+    common_hal_audiofilters_filter_deinit(self);
+    return mp_const_none;
+}
+static MP_DEFINE_CONST_FUN_OBJ_1(audiofilters_filter_deinit_obj, audiofilters_filter_deinit);
+
+static void check_for_deinit(audiofilters_filter_obj_t *self) {
+    if (common_hal_audiofilters_filter_deinited(self)) {
+        raise_deinited_error();
+    }
+}
+
+//|     def __enter__(self) -> Filter:
+//|         """No-op used by Context Managers."""
+//|         ...
+//  Provided by context manager helper.
+
+//|     def __exit__(self) -> None:
+//|         """Automatically deinitializes when exiting a context. See
+//|         :ref:`lifetime-and-contextmanagers` for more info."""
+//|         ...
+static mp_obj_t audiofilters_filter_obj___exit__(size_t n_args, const mp_obj_t *args) {
+    (void)n_args;
+    common_hal_audiofilters_filter_deinit(args[0]);
+    return mp_const_none;
+}
+static MP_DEFINE_CONST_FUN_OBJ_VAR_BETWEEN(audiofilters_filter___exit___obj, 4, 4, audiofilters_filter_obj___exit__);
+
+
+//|     filter: Optional[synthio.Biquad]
+//|     """The normalized biquad filter object used to process the signal."""
+static mp_obj_t audiofilters_filter_obj_get_filter(mp_obj_t self_in) {
+    return common_hal_audiofilters_filter_get_filter(self_in);
+}
+MP_DEFINE_CONST_FUN_OBJ_1(audiofilters_filter_get_filter_obj, audiofilters_filter_obj_get_filter);
+
+static mp_obj_t audiofilters_filter_obj_set_filter(size_t n_args, const mp_obj_t *pos_args, mp_map_t *kw_args) {
+    enum { ARG_filter };
+    static const mp_arg_t allowed_args[] = {
+        { MP_QSTR_filter,     MP_ARG_OBJ | MP_ARG_REQUIRED, {} },
+    };
+    audiofilters_filter_obj_t *self = MP_OBJ_TO_PTR(pos_args[0]);
+    mp_arg_val_t args[MP_ARRAY_SIZE(allowed_args)];
+    mp_arg_parse_all(n_args - 1, pos_args + 1, kw_args, MP_ARRAY_SIZE(allowed_args), allowed_args, args);
+
+    common_hal_audiofilters_filter_set_filter(self, args[ARG_filter].u_obj);
+
+    return mp_const_none;
+}
+MP_DEFINE_CONST_FUN_OBJ_KW(audiofilters_filter_set_filter_obj, 1, audiofilters_filter_obj_set_filter);
+
+MP_PROPERTY_GETSET(audiofilters_filter_filter_obj,
+    (mp_obj_t)&audiofilters_filter_get_filter_obj,
+    (mp_obj_t)&audiofilters_filter_set_filter_obj);
+
+
+//|     mix: synthio.BlockInput
+//|     """The rate the filtered signal mix between 0 and 1 where 0 is only sample and 1 is all effect."""
+static mp_obj_t audiofilters_filter_obj_get_mix(mp_obj_t self_in) {
+    return common_hal_audiofilters_filter_get_mix(self_in);
+}
+MP_DEFINE_CONST_FUN_OBJ_1(audiofilters_filter_get_mix_obj, audiofilters_filter_obj_get_mix);
+
+static mp_obj_t audiofilters_filter_obj_set_mix(size_t n_args, const mp_obj_t *pos_args, mp_map_t *kw_args) {
+    enum { ARG_mix };
+    static const mp_arg_t allowed_args[] = {
+        { MP_QSTR_mix,     MP_ARG_OBJ | MP_ARG_REQUIRED, {} },
+    };
+    audiofilters_filter_obj_t *self = MP_OBJ_TO_PTR(pos_args[0]);
+    mp_arg_val_t args[MP_ARRAY_SIZE(allowed_args)];
+    mp_arg_parse_all(n_args - 1, pos_args + 1, kw_args, MP_ARRAY_SIZE(allowed_args), allowed_args, args);
+
+    common_hal_audiofilters_filter_set_mix(self, args[ARG_mix].u_obj);
+
+    return mp_const_none;
+}
+MP_DEFINE_CONST_FUN_OBJ_KW(audiofilters_filter_set_mix_obj, 1, audiofilters_filter_obj_set_mix);
+
+MP_PROPERTY_GETSET(audiofilters_filter_mix_obj,
+    (mp_obj_t)&audiofilters_filter_get_mix_obj,
+    (mp_obj_t)&audiofilters_filter_set_mix_obj);
+
+
+//|     playing: bool
+//|     """True when the effect is playing a sample. (read-only)"""
+static mp_obj_t audiofilters_filter_obj_get_playing(mp_obj_t self_in) {
+    audiofilters_filter_obj_t *self = MP_OBJ_TO_PTR(self_in);
+    check_for_deinit(self);
+    return mp_obj_new_bool(common_hal_audiofilters_filter_get_playing(self));
+}
+MP_DEFINE_CONST_FUN_OBJ_1(audiofilters_filter_get_playing_obj, audiofilters_filter_obj_get_playing);
+
+MP_PROPERTY_GETTER(audiofilters_filter_playing_obj,
+    (mp_obj_t)&audiofilters_filter_get_playing_obj);
+
+//|     def play(self, sample: circuitpython_typing.AudioSample, *, loop: bool = False) -> None:
+//|         """Plays the sample once when loop=False and continuously when loop=True.
+//|         Does not block. Use `playing` to block.
+//|
+//|         The sample must match the encoding settings given in the constructor."""
+//|         ...
+static mp_obj_t audiofilters_filter_obj_play(size_t n_args, const mp_obj_t *pos_args, mp_map_t *kw_args) {
+    enum { ARG_sample, ARG_loop };
+    static const mp_arg_t allowed_args[] = {
+        { MP_QSTR_sample,    MP_ARG_OBJ | MP_ARG_REQUIRED, {} },
+        { MP_QSTR_loop,      MP_ARG_BOOL | MP_ARG_KW_ONLY, {.u_bool = false} },
+    };
+    audiofilters_filter_obj_t *self = MP_OBJ_TO_PTR(pos_args[0]);
+    check_for_deinit(self);
+    mp_arg_val_t args[MP_ARRAY_SIZE(allowed_args)];
+    mp_arg_parse_all(n_args - 1, pos_args + 1, kw_args, MP_ARRAY_SIZE(allowed_args), allowed_args, args);
+
+
+    mp_obj_t sample = args[ARG_sample].u_obj;
+    common_hal_audiofilters_filter_play(self, sample, args[ARG_loop].u_bool);
+
+    return mp_const_none;
+}
+MP_DEFINE_CONST_FUN_OBJ_KW(audiofilters_filter_play_obj, 1, audiofilters_filter_obj_play);
+
+//|     def stop(self) -> None:
+//|         """Stops playback of the sample."""
+//|         ...
+//|
+static mp_obj_t audiofilters_filter_obj_stop(mp_obj_t self_in) {
+    audiofilters_filter_obj_t *self = MP_OBJ_TO_PTR(self_in);
+
+    common_hal_audiofilters_filter_stop(self);
+    return mp_const_none;
+}
+MP_DEFINE_CONST_FUN_OBJ_1(audiofilters_filter_stop_obj, audiofilters_filter_obj_stop);
+
+static const mp_rom_map_elem_t audiofilters_filter_locals_dict_table[] = {
+    // Methods
+    { MP_ROM_QSTR(MP_QSTR_deinit), MP_ROM_PTR(&audiofilters_filter_deinit_obj) },
+    { MP_ROM_QSTR(MP_QSTR___enter__), MP_ROM_PTR(&default___enter___obj) },
+    { MP_ROM_QSTR(MP_QSTR___exit__), MP_ROM_PTR(&audiofilters_filter___exit___obj) },
+    { MP_ROM_QSTR(MP_QSTR_play), MP_ROM_PTR(&audiofilters_filter_play_obj) },
+    { MP_ROM_QSTR(MP_QSTR_stop), MP_ROM_PTR(&audiofilters_filter_stop_obj) },
+
+    // Properties
+    { MP_ROM_QSTR(MP_QSTR_playing), MP_ROM_PTR(&audiofilters_filter_playing_obj) },
+    { MP_ROM_QSTR(MP_QSTR_filter), MP_ROM_PTR(&audiofilters_filter_filter_obj) },
+    { MP_ROM_QSTR(MP_QSTR_mix), MP_ROM_PTR(&audiofilters_filter_mix_obj) },
+};
+static MP_DEFINE_CONST_DICT(audiofilters_filter_locals_dict, audiofilters_filter_locals_dict_table);
+
+static const audiosample_p_t audiofilters_filter_proto = {
+    MP_PROTO_IMPLEMENT(MP_QSTR_protocol_audiosample)
+    .sample_rate = (audiosample_sample_rate_fun)common_hal_audiofilters_filter_get_sample_rate,
+    .bits_per_sample = (audiosample_bits_per_sample_fun)common_hal_audiofilters_filter_get_bits_per_sample,
+    .channel_count = (audiosample_channel_count_fun)common_hal_audiofilters_filter_get_channel_count,
+    .reset_buffer = (audiosample_reset_buffer_fun)audiofilters_filter_reset_buffer,
+    .get_buffer = (audiosample_get_buffer_fun)audiofilters_filter_get_buffer,
+    .get_buffer_structure = (audiosample_get_buffer_structure_fun)audiofilters_filter_get_buffer_structure,
+};
+
+MP_DEFINE_CONST_OBJ_TYPE(
+    audiofilters_filter_type,
+    MP_QSTR_Filter,
+    MP_TYPE_FLAG_HAS_SPECIAL_ACCESSORS,
+    make_new, audiofilters_filter_make_new,
+    locals_dict, &audiofilters_filter_locals_dict,
+    protocol, &audiofilters_filter_proto
+    );
diff --git a/shared-bindings/audiofilters/Filter.h b/shared-bindings/audiofilters/Filter.h
new file mode 100644
index 000000000000..739b625ee6c5
--- /dev/null
+++ b/shared-bindings/audiofilters/Filter.h
@@ -0,0 +1,33 @@
+// This file is part of the CircuitPython project: https://circuitpython.org
+//
+// SPDX-FileCopyrightText: Copyright (c) 2024 Cooper Dalrymple
+//
+// SPDX-License-Identifier: MIT
+
+#pragma once
+
+#include "shared-module/audiofilters/Filter.h"
+
+extern const mp_obj_type_t audiofilters_filter_type;
+
+void common_hal_audiofilters_filter_construct(audiofilters_filter_obj_t *self,
+    mp_obj_t filter, mp_obj_t mix,
+    uint32_t buffer_size, uint8_t bits_per_sample, bool samples_signed,
+    uint8_t channel_count, uint32_t sample_rate);
+
+void common_hal_audiofilters_filter_deinit(audiofilters_filter_obj_t *self);
+bool common_hal_audiofilters_filter_deinited(audiofilters_filter_obj_t *self);
+
+uint32_t common_hal_audiofilters_filter_get_sample_rate(audiofilters_filter_obj_t *self);
+uint8_t common_hal_audiofilters_filter_get_channel_count(audiofilters_filter_obj_t *self);
+uint8_t common_hal_audiofilters_filter_get_bits_per_sample(audiofilters_filter_obj_t *self);
+
+mp_obj_t common_hal_audiofilters_filter_get_filter(audiofilters_filter_obj_t *self);
+void common_hal_audiofilters_filter_set_filter(audiofilters_filter_obj_t *self, mp_obj_t arg);
+
+mp_obj_t common_hal_audiofilters_filter_get_mix(audiofilters_filter_obj_t *self);
+void common_hal_audiofilters_filter_set_mix(audiofilters_filter_obj_t *self, mp_obj_t arg);
+
+bool common_hal_audiofilters_filter_get_playing(audiofilters_filter_obj_t *self);
+void common_hal_audiofilters_filter_play(audiofilters_filter_obj_t *self, mp_obj_t sample, bool loop);
+void common_hal_audiofilters_filter_stop(audiofilters_filter_obj_t *self);
diff --git a/shared-bindings/audiofilters/__init__.c b/shared-bindings/audiofilters/__init__.c
new file mode 100644
index 000000000000..c4124515b7d3
--- /dev/null
+++ b/shared-bindings/audiofilters/__init__.c
@@ -0,0 +1,33 @@
+// This file is part of the CircuitPython project: https://circuitpython.org
+//
+// SPDX-FileCopyrightText: Copyright (c) 2024 Cooper Dalrymple
+//
+// SPDX-License-Identifier: MIT
+
+#include <stdint.h>
+
+#include "py/obj.h"
+#include "py/runtime.h"
+
+#include "shared-bindings/audiofilters/__init__.h"
+#include "shared-bindings/audiofilters/Filter.h"
+
+//| """Support for audio filter effects
+//|
+//| The `audiofilters` module contains classes to provide access to audio filter effects.
+//|
+//| """
+
+static const mp_rom_map_elem_t audiofilters_module_globals_table[] = {
+    { MP_ROM_QSTR(MP_QSTR___name__), MP_ROM_QSTR(MP_QSTR_audiofilters) },
+    { MP_ROM_QSTR(MP_QSTR_Filter), MP_ROM_PTR(&audiofilters_filter_type) },
+};
+
+static MP_DEFINE_CONST_DICT(audiofilters_module_globals, audiofilters_module_globals_table);
+
+const mp_obj_module_t audiofilters_module = {
+    .base = { &mp_type_module },
+    .globals = (mp_obj_dict_t *)&audiofilters_module_globals,
+};
+
+MP_REGISTER_MODULE(MP_QSTR_audiofilters, audiofilters_module);
diff --git a/shared-bindings/audiofilters/__init__.h b/shared-bindings/audiofilters/__init__.h
new file mode 100644
index 000000000000..29d2f6726559
--- /dev/null
+++ b/shared-bindings/audiofilters/__init__.h
@@ -0,0 +1,7 @@
+// This file is part of the CircuitPython project: https://circuitpython.org
+//
+// SPDX-FileCopyrightText: Copyright (c) 2024 Cooper Dalrymple
+//
+// SPDX-License-Identifier: MIT
+
+#pragma once
diff --git a/shared-module/audiofilters/Filter.c b/shared-module/audiofilters/Filter.c
new file mode 100644
index 000000000000..86edb6cb0418
--- /dev/null
+++ b/shared-module/audiofilters/Filter.c
@@ -0,0 +1,331 @@
+// This file is part of the CircuitPython project: https://circuitpython.org
+//
+// SPDX-FileCopyrightText: Copyright (c) 2024 Cooper Dalrymple
+//
+// SPDX-License-Identifier: MIT
+#include "shared-bindings/audiofilters/Filter.h"
+
+#include <stdint.h>
+#include "py/runtime.h"
+
+void common_hal_audiofilters_filter_construct(audiofilters_filter_obj_t *self,
+    mp_obj_t filter, mp_obj_t mix,
+    uint32_t buffer_size, uint8_t bits_per_sample,
+    bool samples_signed, uint8_t channel_count, uint32_t sample_rate) {
+
+    // Basic settings every effect and audio sample has
+    // These are the effects values, not the source sample(s)
+    self->bits_per_sample = bits_per_sample; // Most common is 16, but 8 is also supported in many places
+    self->samples_signed = samples_signed; // Are the samples we provide signed (common is true)
+    self->channel_count = channel_count; // Channels can be 1 for mono or 2 for stereo
+    self->sample_rate = sample_rate; // Sample rate for the effect, this generally needs to match all audio objects
+
+    // To smooth things out as CircuitPython is doing other tasks most audio objects have a buffer
+    // A double buffer is set up here so the audio output can use DMA on buffer 1 while we
+    // write to and create buffer 2.
+    // This buffer is what is passed to the audio component that plays the effect.
+    // Samples are set sequentially. For stereo audio they are passed L/R/L/R/...
+    self->buffer_len = buffer_size; // in bytes
+
+    self->buffer[0] = m_malloc(self->buffer_len);
+    if (self->buffer[0] == NULL) {
+        common_hal_audiofilters_filter_deinit(self);
+        m_malloc_fail(self->buffer_len);
+    }
+    memset(self->buffer[0], 0, self->buffer_len);
+
+    self->buffer[1] = m_malloc(self->buffer_len);
+    if (self->buffer[1] == NULL) {
+        common_hal_audiofilters_filter_deinit(self);
+        m_malloc_fail(self->buffer_len);
+    }
+    memset(self->buffer[1], 0, self->buffer_len);
+
+    self->last_buf_idx = 1; // Which buffer to use first, toggle between 0 and 1
+
+    // This buffer will be used to process samples through the biquad filter
+    self->filter_buffer = m_malloc(SYNTHIO_MAX_DUR * sizeof(int32_t));
+    if (self->filter_buffer == NULL) {
+        common_hal_audiofilters_filter_deinit(self);
+        m_malloc_fail(SYNTHIO_MAX_DUR * sizeof(int32_t));
+    }
+    memset(self->filter_buffer, 0, SYNTHIO_MAX_DUR * sizeof(int32_t));
+
+    // Initialize other values most effects will need.
+    self->sample = NULL; // The current playing sample
+    self->sample_remaining_buffer = NULL; // Pointer to the start of the sample buffer we have not played
+    self->sample_buffer_length = 0; // How many samples do we have left to play (these may be 16 bit!)
+    self->loop = false; // When the sample is done do we loop to the start again or stop (e.g. in a wav file)
+    self->more_data = false; // Is there still more data to read from the sample or did we finish
+
+    // The below section sets up the effect's starting values.
+
+    if (filter == MP_OBJ_NULL) {
+        filter = mp_const_none;
+    }
+    synthio_biquad_filter_assign(&self->filter_state, filter);
+    self->filter_obj = filter;
+
+    // If we did not receive a BlockInput we need to create a default float value
+    if (mix == MP_OBJ_NULL) {
+        mix = mp_obj_new_float(1.0);
+    }
+    synthio_block_assign_slot(mix, &self->mix, MP_QSTR_mix);
+}
+
+bool common_hal_audiofilters_filter_deinited(audiofilters_filter_obj_t *self) {
+    if (self->buffer[0] == NULL) {
+        return true;
+    }
+    return false;
+}
+
+void common_hal_audiofilters_filter_deinit(audiofilters_filter_obj_t *self) {
+    if (common_hal_audiofilters_filter_deinited(self)) {
+        return;
+    }
+    self->buffer[0] = NULL;
+    self->buffer[1] = NULL;
+    self->filter_buffer = NULL;
+}
+
+mp_obj_t common_hal_audiofilters_filter_get_filter(audiofilters_filter_obj_t *self) {
+    return self->filter_obj;
+}
+
+void common_hal_audiofilters_filter_set_filter(audiofilters_filter_obj_t *self, mp_obj_t arg) {
+    synthio_biquad_filter_assign(&self->filter_state, arg);
+    self->filter_obj = arg;
+}
+
+mp_obj_t common_hal_audiofilters_filter_get_mix(audiofilters_filter_obj_t *self) {
+    return self->mix.obj;
+}
+
+void common_hal_audiofilters_filter_set_mix(audiofilters_filter_obj_t *self, mp_obj_t arg) {
+    synthio_block_assign_slot(arg, &self->mix, MP_QSTR_mix);
+}
+
+uint32_t common_hal_audiofilters_filter_get_sample_rate(audiofilters_filter_obj_t *self) {
+    return self->sample_rate;
+}
+
+uint8_t common_hal_audiofilters_filter_get_channel_count(audiofilters_filter_obj_t *self) {
+    return self->channel_count;
+}
+
+uint8_t common_hal_audiofilters_filter_get_bits_per_sample(audiofilters_filter_obj_t *self) {
+    return self->bits_per_sample;
+}
+
+void audiofilters_filter_reset_buffer(audiofilters_filter_obj_t *self,
+    bool single_channel_output,
+    uint8_t channel) {
+
+    memset(self->buffer[0], 0, self->buffer_len);
+    memset(self->buffer[1], 0, self->buffer_len);
+    memset(self->filter_buffer, 0, SYNTHIO_MAX_DUR * sizeof(int32_t));
+
+    synthio_biquad_filter_reset(&self->filter_state);
+}
+
+bool common_hal_audiofilters_filter_get_playing(audiofilters_filter_obj_t *self) {
+    return self->sample != NULL;
+}
+
+void common_hal_audiofilters_filter_play(audiofilters_filter_obj_t *self, mp_obj_t sample, bool loop) {
+    // When a sample is to be played we must ensure the samples values matches what we expect
+    // Then we reset the sample and get the first buffer to play
+    // The get_buffer function will actually process that data
+
+    if (audiosample_sample_rate(sample) != self->sample_rate) {
+        mp_raise_ValueError_varg(MP_ERROR_TEXT("The sample's %q does not match"), MP_QSTR_sample_rate);
+    }
+    if (audiosample_channel_count(sample) != self->channel_count) {
+        mp_raise_ValueError_varg(MP_ERROR_TEXT("The sample's %q does not match"), MP_QSTR_channel_count);
+    }
+    if (audiosample_bits_per_sample(sample) != self->bits_per_sample) {
+        mp_raise_ValueError_varg(MP_ERROR_TEXT("The sample's %q does not match"), MP_QSTR_bits_per_sample);
+    }
+    bool single_buffer;
+    bool samples_signed;
+    uint32_t max_buffer_length;
+    uint8_t spacing;
+    audiosample_get_buffer_structure(sample, false, &single_buffer, &samples_signed, &max_buffer_length, &spacing);
+    if (samples_signed != self->samples_signed) {
+        mp_raise_ValueError_varg(MP_ERROR_TEXT("The sample's %q does not match"), MP_QSTR_signedness);
+    }
+
+    self->sample = sample;
+    self->loop = loop;
+
+    audiosample_reset_buffer(self->sample, false, 0);
+    audioio_get_buffer_result_t result = audiosample_get_buffer(self->sample, false, 0, (uint8_t **)&self->sample_remaining_buffer, &self->sample_buffer_length);
+
+    // Track remaining sample length in terms of bytes per sample
+    self->sample_buffer_length /= (self->bits_per_sample / 8);
+    // Store if we have more data in the sample to retrieve
+    self->more_data = result == GET_BUFFER_MORE_DATA;
+
+    return;
+}
+
+void common_hal_audiofilters_filter_stop(audiofilters_filter_obj_t *self) {
+    // When the sample is set to stop playing do any cleanup here
+    self->sample = NULL;
+    return;
+}
+
+#define RANGE_LOW_16 (-28000)
+#define RANGE_HIGH_16 (28000)
+#define RANGE_SHIFT_16 (16)
+#define RANGE_SCALE_16 (0xfffffff / (32768 * 2 - RANGE_HIGH_16)) // 2 for echo+sample
+
+// dynamic range compression via a downward compressor with hard knee
+//
+// When the output value is within the range +-28000 (about 85% of full scale),
+// it is unchanged. Otherwise, it undergoes a gain reduction so that the
+// largest possible values, (+32768,-32767) * 2 (2 for echo and sample),
+// still fit within the output range
+//
+// This produces a much louder overall volume with multiple voices, without
+// much additional processing.
+//
+// https://en.wikipedia.org/wiki/Dynamic_range_compression
+static
+int16_t mix_down_sample(int32_t sample) {
+    if (sample < RANGE_LOW_16) {
+        sample = (((sample - RANGE_LOW_16) * RANGE_SCALE_16) >> RANGE_SHIFT_16) + RANGE_LOW_16;
+    } else if (sample > RANGE_HIGH_16) {
+        sample = (((sample - RANGE_HIGH_16) * RANGE_SCALE_16) >> RANGE_SHIFT_16) + RANGE_HIGH_16;
+    }
+    return sample;
+}
+
+audioio_get_buffer_result_t audiofilters_filter_get_buffer(audiofilters_filter_obj_t *self, bool single_channel_output, uint8_t channel,
+    uint8_t **buffer, uint32_t *buffer_length) {
+
+    if (!single_channel_output) {
+        channel = 0;
+    }
+
+    // get the effect values we need from the BlockInput. These may change at run time so you need to do bounds checking if required
+    mp_float_t mix = MIN(1.0, MAX(synthio_block_slot_get(&self->mix), 0.0));
+
+    // Switch our buffers to the other buffer
+    self->last_buf_idx = !self->last_buf_idx;
+
+    // If we are using 16 bit samples we need a 16 bit pointer, 8 bit needs an 8 bit pointer
+    int16_t *word_buffer = (int16_t *)self->buffer[self->last_buf_idx];
+    int8_t *hword_buffer = self->buffer[self->last_buf_idx];
+    uint32_t length = self->buffer_len / (self->bits_per_sample / 8);
+
+    // Loop over the entire length of our buffer to fill it, this may require several calls to get data from the sample
+    while (length != 0) {
+        // Check if there is no more sample to play, we will either load more data, reset the sample if loop is on or clear the sample
+        if (self->sample_buffer_length == 0) {
+            if (!self->more_data) { // The sample has indicated it has no more data to play
+                if (self->loop && self->sample) { // If we are supposed to loop reset the sample to the start
+                    audiosample_reset_buffer(self->sample, false, 0);
+                } else { // If we were not supposed to loop the sample, stop playing it
+                    self->sample = NULL;
+                }
+            }
+            if (self->sample) {
+                // Load another sample buffer to play
+                audioio_get_buffer_result_t result = audiosample_get_buffer(self->sample, false, 0, (uint8_t **)&self->sample_remaining_buffer, &self->sample_buffer_length);
+                // Track length in terms of words.
+                self->sample_buffer_length /= (self->bits_per_sample / 8);
+                self->more_data = result == GET_BUFFER_MORE_DATA;
+            }
+        }
+
+        // If we have a sample, filter it
+        if (self->sample != NULL) {
+            // Determine how many bytes we can process to our buffer, the less of the sample we have left and our buffer remaining
+            uint32_t n = MIN(self->sample_buffer_length, length);
+
+            int16_t *sample_src = (int16_t *)self->sample_remaining_buffer; // for 16-bit samples
+            int8_t *sample_hsrc = (int8_t *)self->sample_remaining_buffer; // for 8-bit samples
+
+            if (mix <= 0.01 || self->filter_obj == mp_const_none) { // if mix is zero pure sample only or no biquad filter object is provided
+                for (uint32_t i = 0; i < n; i++) {
+                    if (MP_LIKELY(self->bits_per_sample == 16)) {
+                        word_buffer[i] = sample_src[i];
+                    } else {
+                        hword_buffer[i] = sample_hsrc[i];
+                    }
+                }
+            } else {
+                uint32_t i = 0;
+                while (i < n) {
+                    uint32_t n_samples = MIN(SYNTHIO_MAX_DUR, n - i);
+
+                    // Fill filter buffer with samples
+                    for (uint32_t j = 0; j < n_samples; j++) {
+                        if (MP_LIKELY(self->bits_per_sample == 16)) {
+                            self->filter_buffer[j] = sample_src[i + j];
+                        } else {
+                            if (self->samples_signed) {
+                                self->filter_buffer[j] = sample_hsrc[i + j];
+                            } else {
+                                // Be careful here changing from an 8 bit unsigned to signed into a 32-bit signed
+                                self->filter_buffer[j] = (int8_t)(((uint8_t)sample_hsrc[i + j]) ^ 0x80);
+                            }
+                        }
+                    }
+
+                    // Process biquad filter
+                    synthio_biquad_filter_samples(&self->filter_state, self->filter_buffer, n_samples);
+
+                    // Mix processed signal with original sample and transfer to output buffer
+                    for (uint32_t j = 0; j < n_samples; j++) {
+                        if (MP_LIKELY(self->bits_per_sample == 16)) {
+                            word_buffer[i + j] = mix_down_sample((sample_src[i + j] * (1.0 - mix)) + (self->filter_buffer[j] * mix));
+                            if (!self->samples_signed) {
+                                word_buffer[i + j] ^= 0x8000;
+                            }
+                        } else {
+                            if (self->samples_signed) {
+                                hword_buffer[i + j] = (int8_t)((sample_hsrc[i + j] * (1.0 - mix)) + (self->filter_buffer[j] * mix));
+                            } else {
+                                hword_buffer[i + j] = (uint8_t)(((int8_t)(((uint8_t)sample_hsrc[i + j]) ^ 0x80) * (1.0 - mix)) + (self->filter_buffer[j] * mix)) ^ 0x80;
+                            }
+                        }
+                    }
+
+                    i += n_samples;
+                }
+            }
+
+            // Update the remaining length and the buffer positions based on how much we wrote into our buffer
+            length -= n;
+            word_buffer += n;
+            hword_buffer += n;
+            self->sample_remaining_buffer += (n * (self->bits_per_sample / 8));
+            self->sample_buffer_length -= n;
+        }
+    }
+
+    // Finally pass our buffer and length to the calling audio function
+    *buffer = (uint8_t *)self->buffer[self->last_buf_idx];
+    *buffer_length = self->buffer_len;
+
+    // Filter always returns more data but some effects may return GET_BUFFER_DONE or GET_BUFFER_ERROR (see audiocore/__init__.h)
+    return GET_BUFFER_MORE_DATA;
+}
+
+void audiofilters_filter_get_buffer_structure(audiofilters_filter_obj_t *self, bool single_channel_output,
+    bool *single_buffer, bool *samples_signed, uint32_t *max_buffer_length, uint8_t *spacing) {
+
+    // Return information about the effect's buffer (not the sample's)
+    // These are used by calling audio objects to determine how to handle the effect's buffer
+    *single_buffer = false;
+    *samples_signed = self->samples_signed;
+    *max_buffer_length = self->buffer_len;
+    if (single_channel_output) {
+        *spacing = self->channel_count;
+    } else {
+        *spacing = 1;
+    }
+}
diff --git a/shared-module/audiofilters/Filter.h b/shared-module/audiofilters/Filter.h
new file mode 100644
index 000000000000..b5d743a6c191
--- /dev/null
+++ b/shared-module/audiofilters/Filter.h
@@ -0,0 +1,55 @@
+// This file is part of the CircuitPython project: https://circuitpython.org
+//
+// SPDX-FileCopyrightText: Copyright (c) 2024 Cooper Dalrymple
+//
+// SPDX-License-Identifier: MIT
+#pragma once
+
+#include "py/obj.h"
+
+#include "shared-module/audiocore/__init__.h"
+#include "shared-module/synthio/block.h"
+#include "shared-module/synthio/Biquad.h"
+
+extern const mp_obj_type_t audiofilters_filter_type;
+
+typedef struct {
+    mp_obj_base_t base;
+    mp_obj_t filter_obj;
+    synthio_block_slot_t mix;
+
+    biquad_filter_state filter_state;
+
+    uint8_t bits_per_sample;
+    bool samples_signed;
+    uint8_t channel_count;
+    uint32_t sample_rate;
+
+    int8_t *buffer[2];
+    uint8_t last_buf_idx;
+    uint32_t buffer_len; // max buffer in bytes
+
+    uint8_t *sample_remaining_buffer;
+    uint32_t sample_buffer_length;
+
+    int32_t *filter_buffer;
+
+    bool loop;
+    bool more_data;
+
+    mp_obj_t sample;
+} audiofilters_filter_obj_t;
+
+void audiofilters_filter_reset_buffer(audiofilters_filter_obj_t *self,
+    bool single_channel_output,
+    uint8_t channel);
+
+audioio_get_buffer_result_t audiofilters_filter_get_buffer(audiofilters_filter_obj_t *self,
+    bool single_channel_output,
+    uint8_t channel,
+    uint8_t **buffer,
+    uint32_t *buffer_length);  // length in bytes
+
+void audiofilters_filter_get_buffer_structure(audiofilters_filter_obj_t *self, bool single_channel_output,
+    bool *single_buffer, bool *samples_signed,
+    uint32_t *max_buffer_length, uint8_t *spacing);
diff --git a/shared-module/audiofilters/__init__.c b/shared-module/audiofilters/__init__.c
new file mode 100644
index 000000000000..83929b4c4fb8
--- /dev/null
+++ b/shared-module/audiofilters/__init__.c
@@ -0,0 +1,5 @@
+// This file is part of the CircuitPython project: https://circuitpython.org
+//
+// SPDX-FileCopyrightText: Copyright (c) 2024 Cooper Dalrymple
+//
+// SPDX-License-Identifier: MIT
diff --git a/shared-module/audiofilters/__init__.h b/shared-module/audiofilters/__init__.h
new file mode 100644
index 000000000000..29d2f6726559
--- /dev/null
+++ b/shared-module/audiofilters/__init__.h
@@ -0,0 +1,7 @@
+// This file is part of the CircuitPython project: https://circuitpython.org
+//
+// SPDX-FileCopyrightText: Copyright (c) 2024 Cooper Dalrymple
+//
+// SPDX-License-Identifier: MIT
+
+#pragma once