diff --git a/locale/circuitpython.pot b/locale/circuitpython.pot index 486d4eb149be3..6beb320527e43 100644 --- a/locale/circuitpython.pot +++ b/locale/circuitpython.pot @@ -2595,6 +2595,11 @@ msgstr "" #: shared-bindings/audiodelays/MultiTapDelay.c #: shared-bindings/audiodelays/PitchShift.c #: shared-bindings/audiofilters/Distortion.c +#: shared-bindings/audiodelays/Reverb.c +msgid "bits_per_sample must be 16" +msgstr "" + +#: shared-bindings/audiodelays/Echo.c shared-bindings/audiofilters/Distortion.c #: shared-bindings/audiofilters/Filter.c shared-bindings/audiomixer/Mixer.c msgid "bits_per_sample must be 8 or 16" msgstr "" @@ -4012,6 +4017,10 @@ msgstr "" msgid "rsplit(None,n)" msgstr "" +#: shared-bindings/audiodelays/Reverb.c +msgid "samples_signed must be true" +msgstr "" + #: ports/atmel-samd/common-hal/audiobusio/PDMIn.c #: ports/raspberrypi/common-hal/audiobusio/PDMIn.c msgid "sampling rate out of range" diff --git a/ports/unix/variants/coverage/mpconfigvariant.mk b/ports/unix/variants/coverage/mpconfigvariant.mk index e18645409fdd8..e1924479bbf53 100644 --- a/ports/unix/variants/coverage/mpconfigvariant.mk +++ b/ports/unix/variants/coverage/mpconfigvariant.mk @@ -41,6 +41,8 @@ SRC_BITMAP := \ shared-bindings/audiofilters/Distortion.c \ shared-bindings/audiofilters/Filter.c \ shared-bindings/audiofilters/__init__.c \ + shared-bindings/audiofreeverb/Freeverb.c \ + shared-bindings/audiofreeverb/__init__.c \ shared-bindings/audiomixer/__init__.c \ shared-bindings/audiomixer/Mixer.c \ shared-bindings/audiomixer/MixerVoice.c \ @@ -86,6 +88,8 @@ SRC_BITMAP := \ shared-module/audiofilters/Distortion.c \ shared-module/audiofilters/Filter.c \ shared-module/audiofilters/__init__.c \ + shared-module/audiofreeverb/Freeverb.c \ + shared-module/audiofreeverb/__init__.c \ shared-module/audiomixer/__init__.c \ shared-module/audiomp3/MP3Decoder.c \ shared-module/audiomixer/Mixer.c \ diff --git a/ports/zephyr-cp/boards/nordic/nrf5340dk/autogen_board_info.toml b/ports/zephyr-cp/boards/nordic/nrf5340dk/autogen_board_info.toml index 624e7e69a3fd1..aecccebb9c850 100644 --- a/ports/zephyr-cp/boards/nordic/nrf5340dk/autogen_board_info.toml +++ b/ports/zephyr-cp/boards/nordic/nrf5340dk/autogen_board_info.toml @@ -19,6 +19,7 @@ audiobusio = false audiocore = false audiodelays = false audiofilters = false +audiofreeverb = false audioio = false audiomixer = false audiomp3 = false diff --git a/ports/zephyr-cp/boards/nordic/nrf54l15dk/autogen_board_info.toml b/ports/zephyr-cp/boards/nordic/nrf54l15dk/autogen_board_info.toml index 07ab1f7f4dfc6..27e4f35f22366 100644 --- a/ports/zephyr-cp/boards/nordic/nrf54l15dk/autogen_board_info.toml +++ b/ports/zephyr-cp/boards/nordic/nrf54l15dk/autogen_board_info.toml @@ -19,6 +19,7 @@ audiobusio = false audiocore = false audiodelays = false audiofilters = false +audiofreeverb = false audioio = false audiomixer = false audiomp3 = false diff --git a/ports/zephyr-cp/boards/nordic/nrf7002dk/autogen_board_info.toml b/ports/zephyr-cp/boards/nordic/nrf7002dk/autogen_board_info.toml index b10bbe69a0a10..35e3791507023 100644 --- a/ports/zephyr-cp/boards/nordic/nrf7002dk/autogen_board_info.toml +++ b/ports/zephyr-cp/boards/nordic/nrf7002dk/autogen_board_info.toml @@ -19,6 +19,7 @@ audiobusio = false audiocore = false audiodelays = false audiofilters = false +audiofreeverb = false audioio = false audiomixer = false audiomp3 = false diff --git a/ports/zephyr-cp/boards/renesas/ek_ra6m5/autogen_board_info.toml b/ports/zephyr-cp/boards/renesas/ek_ra6m5/autogen_board_info.toml index fcfeec097d124..4f4ee95bd0296 100644 --- a/ports/zephyr-cp/boards/renesas/ek_ra6m5/autogen_board_info.toml +++ b/ports/zephyr-cp/boards/renesas/ek_ra6m5/autogen_board_info.toml @@ -19,6 +19,7 @@ audiobusio = false audiocore = false audiodelays = false audiofilters = false +audiofreeverb = false audioio = false audiomixer = false audiomp3 = false diff --git a/ports/zephyr-cp/boards/renesas/ek_ra8d1/autogen_board_info.toml b/ports/zephyr-cp/boards/renesas/ek_ra8d1/autogen_board_info.toml index 69481e904fea4..90bb28c341ef6 100644 --- a/ports/zephyr-cp/boards/renesas/ek_ra8d1/autogen_board_info.toml +++ b/ports/zephyr-cp/boards/renesas/ek_ra8d1/autogen_board_info.toml @@ -19,6 +19,7 @@ audiobusio = false audiocore = false audiodelays = false audiofilters = false +audiofreeverb = false audioio = false audiomixer = false audiomp3 = false diff --git a/ports/zephyr-cp/boards/st/nucleo_u575zi_q/autogen_board_info.toml b/ports/zephyr-cp/boards/st/nucleo_u575zi_q/autogen_board_info.toml index e7e179e4c2cc9..c6fa66037a9ac 100644 --- a/ports/zephyr-cp/boards/st/nucleo_u575zi_q/autogen_board_info.toml +++ b/ports/zephyr-cp/boards/st/nucleo_u575zi_q/autogen_board_info.toml @@ -19,6 +19,7 @@ audiobusio = false audiocore = false audiodelays = false audiofilters = false +audiofreeverb = false audioio = false audiomixer = false audiomp3 = false diff --git a/ports/zephyr-cp/boards/st/stm32h7b3i_dk/autogen_board_info.toml b/ports/zephyr-cp/boards/st/stm32h7b3i_dk/autogen_board_info.toml index d4fa229b7e67d..9e117262b3f91 100644 --- a/ports/zephyr-cp/boards/st/stm32h7b3i_dk/autogen_board_info.toml +++ b/ports/zephyr-cp/boards/st/stm32h7b3i_dk/autogen_board_info.toml @@ -19,6 +19,7 @@ audiobusio = false audiocore = false audiodelays = false audiofilters = false +audiofreeverb = false audioio = false audiomixer = false audiomp3 = false diff --git a/py/circuitpy_defns.mk b/py/circuitpy_defns.mk index 03431f392f1b0..11999ac515533 100644 --- a/py/circuitpy_defns.mk +++ b/py/circuitpy_defns.mk @@ -137,6 +137,9 @@ endif ifeq ($(CIRCUITPY_AUDIOFILTERS),1) SRC_PATTERNS += audiofilters/% endif +ifeq ($(CIRCUITPY_AUDIOFREEVERB),1) +SRC_PATTERNS += audiofreeverb/% +endif ifeq ($(CIRCUITPY_AUDIOMIXER),1) SRC_PATTERNS += audiomixer/% endif @@ -671,6 +674,8 @@ SRC_SHARED_MODULE_ALL = \ audiofilters/Distortion.c \ audiofilters/Filter.c \ audiofilters/__init__.c \ + audiofreeverb/__init__.c \ + audiofreeverb/Freeverb.c \ audioio/__init__.c \ audiomixer/Mixer.c \ audiomixer/MixerVoice.c \ diff --git a/py/circuitpy_mpconfig.mk b/py/circuitpy_mpconfig.mk index c277c04c9d4d5..32ed205ec211b 100644 --- a/py/circuitpy_mpconfig.mk +++ b/py/circuitpy_mpconfig.mk @@ -146,6 +146,8 @@ CIRCUITPY_AUDIODELAYS ?= $(CIRCUITPY_AUDIOEFFECTS) CFLAGS += -DCIRCUITPY_AUDIODELAYS=$(CIRCUITPY_AUDIODELAYS) CIRCUITPY_AUDIOFILTERS ?= $(CIRCUITPY_AUDIOEFFECTS) CFLAGS += -DCIRCUITPY_AUDIOFILTERS=$(CIRCUITPY_AUDIOFILTERS) +CIRCUITPY_AUDIOFREEVERB ?= $(CIRCUITPY_AUDIOEFFECTS) +CFLAGS += -DCIRCUITPY_AUDIOFREEVERB=$(CIRCUITPY_AUDIOFREEVERB) CIRCUITPY_AURORA_EPAPER ?= 0 CFLAGS += -DCIRCUITPY_AURORA_EPAPER=$(CIRCUITPY_AURORA_EPAPER) diff --git a/shared-bindings/audiofreeverb/Freeverb.c b/shared-bindings/audiofreeverb/Freeverb.c new file mode 100644 index 0000000000000..62c9237a0d271 --- /dev/null +++ b/shared-bindings/audiofreeverb/Freeverb.c @@ -0,0 +1,268 @@ +// This file is part of the CircuitPython project: https://circuitpython.org +// +// SPDX-FileCopyrightText: Copyright (c) 2025 Mark Komus +// +// SPDX-License-Identifier: MIT + +#include + +#include "shared-bindings/audiofreeverb/Freeverb.h" +#include "shared-bindings/audiocore/__init__.h" +#include "shared-module/audiofreeverb/Freeverb.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" + +//| class Freeverb: +//| """An Freeverb effect""" +//| +//| def __init__( +//| self, +//| roomsize: synthio.BlockInput = 0.5, +//| damp: synthio.BlockInput = 0.5, +//| mix: synthio.BlockInput = 0.5, +//| buffer_size: int = 512, +//| sample_rate: int = 8000, +//| bits_per_sample: int = 16, +//| samples_signed: bool = True, +//| channel_count: int = 1, +//| ) -> None: +//| """Create a Reverb effect simulating the audio taking place in a large room where you get echos +//| off of various surfaces at various times. The size of the room can be adjusted as well as how +//| much the higher frequencies get absorbed by the walls. +//| +//| 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 synthio.BlockInput roomsize: The size of the room. 0.0 = smallest; 1.0 = largest. +//| :param synthio.BlockInput damp: How much the walls absorb. 0.0 = least; 1.0 = most. +//| :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. Freeverb requires 16 bits. +//| :param bool samples_signed: Effect is signed (True) or unsigned (False). Freeverb requires signed (True). +//| +//| Playing adding reverb to a synth:: +//| +//| import time +//| import board +//| import audiobusio +//| import synthio +//| import audiofreeverb +//| +//| audio = audiobusio.I2SOut(bit_clock=board.GP20, word_select=board.GP21, data=board.GP22) +//| synth = synthio.Synthesizer(channel_count=1, sample_rate=44100) +//| reverb = audiofreeverb.Freeverb(roomsize=0.7, damp=0.3, buffer_size=1024, channel_count=1, sample_rate=44100, mix=0.7) +//| reverb.play(synth) +//| audio.play(reverb) +//| +//| note = synthio.Note(261) +//| while True: +//| synth.press(note) +//| time.sleep(0.55) +//| synth.release(note) +//| time.sleep(5)""" +//| ... +//| +static mp_obj_t audiofreeverb_freeverb_make_new(const mp_obj_type_t *type, size_t n_args, size_t n_kw, const mp_obj_t *all_args) { + enum { ARG_roomsize, ARG_damp, 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_roomsize, MP_ARG_OBJ | MP_ARG_KW_ONLY, {.u_obj = MP_OBJ_NULL} }, + { MP_QSTR_damp, 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); + if (args[ARG_samples_signed].u_bool != true) { + mp_raise_ValueError(MP_ERROR_TEXT("samples_signed must be true")); + } + mp_int_t bits_per_sample = args[ARG_bits_per_sample].u_int; + if (bits_per_sample != 16) { + mp_raise_ValueError(MP_ERROR_TEXT("bits_per_sample must be 16")); + } + + audiofreeverb_freeverb_obj_t *self = mp_obj_malloc(audiofreeverb_freeverb_obj_t, &audiofreeverb_freeverb_type); + common_hal_audiofreeverb_freeverb_construct(self, args[ARG_roomsize].u_obj, args[ARG_damp].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 Freeverb.""" +//| ... +//| +static mp_obj_t audiofreeverb_freeverb_deinit(mp_obj_t self_in) { + audiofreeverb_freeverb_obj_t *self = MP_OBJ_TO_PTR(self_in); + common_hal_audiofreeverb_freeverb_deinit(self); + return mp_const_none; +} +static MP_DEFINE_CONST_FUN_OBJ_1(audiofreeverb_freeverb_deinit_obj, audiofreeverb_freeverb_deinit); + +static void check_for_deinit(audiofreeverb_freeverb_obj_t *self) { + audiosample_check_for_deinit(&self->base); +} + +//| def __enter__(self) -> Freeverb: +//| """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.""" +//| ... +//| +// Provided by context manager helper. + +//| roomsize: synthio.BlockInput +//| """Apparent size of the room 0.0-1.0""" +static mp_obj_t audiofreeverb_freeverb_obj_get_roomsize(mp_obj_t self_in) { + return common_hal_audiofreeverb_freeverb_get_roomsize(self_in); +} +MP_DEFINE_CONST_FUN_OBJ_1(audiofreeverb_freeverb_get_roomsize_obj, audiofreeverb_freeverb_obj_get_roomsize); + +static mp_obj_t audiofreeverb_freeverb_obj_set_roomsize(mp_obj_t self_in, mp_obj_t roomsize) { + audiofreeverb_freeverb_obj_t *self = MP_OBJ_TO_PTR(self_in); + common_hal_audiofreeverb_freeverb_set_roomsize(self, roomsize); + return mp_const_none; +} +MP_DEFINE_CONST_FUN_OBJ_2(audiofreeverb_freeverb_set_roomsize_obj, audiofreeverb_freeverb_obj_set_roomsize); + +MP_PROPERTY_GETSET(audiofreeverb_freeverb_roomsize_obj, + (mp_obj_t)&audiofreeverb_freeverb_get_roomsize_obj, + (mp_obj_t)&audiofreeverb_freeverb_set_roomsize_obj); + +//| damp: synthio.BlockInput +//| """How much the high frequencies are dampened in the area. 0.0-1.0""" +static mp_obj_t audiofreeverb_freeverb_obj_get_damp(mp_obj_t self_in) { + return common_hal_audiofreeverb_freeverb_get_damp(self_in); +} +MP_DEFINE_CONST_FUN_OBJ_1(audiofreeverb_freeverb_get_damp_obj, audiofreeverb_freeverb_obj_get_damp); + +static mp_obj_t audiofreeverb_freeverb_obj_set_damp(mp_obj_t self_in, mp_obj_t damp) { + audiofreeverb_freeverb_obj_t *self = MP_OBJ_TO_PTR(self_in); + common_hal_audiofreeverb_freeverb_set_damp(self, damp); + return mp_const_none; +} +MP_DEFINE_CONST_FUN_OBJ_2(audiofreeverb_freeverb_set_damp_obj, audiofreeverb_freeverb_obj_set_damp); + +MP_PROPERTY_GETSET(audiofreeverb_freeverb_damp_obj, + (mp_obj_t)&audiofreeverb_freeverb_get_damp_obj, + (mp_obj_t)&audiofreeverb_freeverb_set_damp_obj); + +//| mix: synthio.BlockInput +//| """The rate the reverb mix between 0 and 1 where 0 is only sample and 1 is all effect.""" +static mp_obj_t audiofreeverb_freeverb_obj_get_mix(mp_obj_t self_in) { + return common_hal_audiofreeverb_freeverb_get_mix(self_in); +} +MP_DEFINE_CONST_FUN_OBJ_1(audiofreeverb_freeverb_get_mix_obj, audiofreeverb_freeverb_obj_get_mix); + +static mp_obj_t audiofreeverb_freeverb_obj_set_mix(mp_obj_t self_in, mp_obj_t mix_in) { + audiofreeverb_freeverb_obj_t *self = MP_OBJ_TO_PTR(self_in); + common_hal_audiofreeverb_freeverb_set_mix(self, mix_in); + return mp_const_none; +} +MP_DEFINE_CONST_FUN_OBJ_2(audiofreeverb_freeverb_set_mix_obj, audiofreeverb_freeverb_obj_set_mix); + +MP_PROPERTY_GETSET(audiofreeverb_freeverb_mix_obj, + (mp_obj_t)&audiofreeverb_freeverb_get_mix_obj, + (mp_obj_t)&audiofreeverb_freeverb_set_mix_obj); + +//| playing: bool +//| """True when the effect is playing a sample. (read-only)""" +//| +static mp_obj_t audiofreeverb_freeverb_obj_get_playing(mp_obj_t self_in) { + audiofreeverb_freeverb_obj_t *self = MP_OBJ_TO_PTR(self_in); + check_for_deinit(self); + return mp_obj_new_bool(common_hal_audiofreeverb_freeverb_get_playing(self)); +} +MP_DEFINE_CONST_FUN_OBJ_1(audiofreeverb_freeverb_get_playing_obj, audiofreeverb_freeverb_obj_get_playing); + +MP_PROPERTY_GETTER(audiofreeverb_freeverb_playing_obj, + (mp_obj_t)&audiofreeverb_freeverb_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 audiofreeverb_freeverb_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} }, + }; + audiofreeverb_freeverb_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_audiofreeverb_freeverb_play(self, sample, args[ARG_loop].u_bool); + + return mp_const_none; +} +MP_DEFINE_CONST_FUN_OBJ_KW(audiofreeverb_freeverb_play_obj, 1, audiofreeverb_freeverb_obj_play); + +//| def stop(self) -> None: +//| """Stops playback of the sample. The reverb continues playing.""" +//| ... +//| +//| +static mp_obj_t audiofreeverb_freeverb_obj_stop(mp_obj_t self_in) { + audiofreeverb_freeverb_obj_t *self = MP_OBJ_TO_PTR(self_in); + + common_hal_audiofreeverb_freeverb_stop(self); + return mp_const_none; +} +MP_DEFINE_CONST_FUN_OBJ_1(audiofreeverb_freeverb_stop_obj, audiofreeverb_freeverb_obj_stop); + +static const mp_rom_map_elem_t audiofreeverb_freeverb_locals_dict_table[] = { + // Methods + { MP_ROM_QSTR(MP_QSTR_deinit), MP_ROM_PTR(&audiofreeverb_freeverb_deinit_obj) }, + { MP_ROM_QSTR(MP_QSTR___enter__), MP_ROM_PTR(&default___enter___obj) }, + { MP_ROM_QSTR(MP_QSTR___exit__), MP_ROM_PTR(&default___exit___obj) }, + { MP_ROM_QSTR(MP_QSTR_play), MP_ROM_PTR(&audiofreeverb_freeverb_play_obj) }, + { MP_ROM_QSTR(MP_QSTR_stop), MP_ROM_PTR(&audiofreeverb_freeverb_stop_obj) }, + + // Properties + { MP_ROM_QSTR(MP_QSTR_playing), MP_ROM_PTR(&audiofreeverb_freeverb_playing_obj) }, + { MP_ROM_QSTR(MP_QSTR_roomsize), MP_ROM_PTR(&audiofreeverb_freeverb_roomsize_obj) }, + { MP_ROM_QSTR(MP_QSTR_damp), MP_ROM_PTR(&audiofreeverb_freeverb_damp_obj) }, + { MP_ROM_QSTR(MP_QSTR_mix), MP_ROM_PTR(&audiofreeverb_freeverb_mix_obj) }, + AUDIOSAMPLE_FIELDS, +}; +static MP_DEFINE_CONST_DICT(audiofreeverb_freeverb_locals_dict, audiofreeverb_freeverb_locals_dict_table); + +static const audiosample_p_t audiofreeverb_freeverb_proto = { + MP_PROTO_IMPLEMENT(MP_QSTR_protocol_audiosample) + .reset_buffer = (audiosample_reset_buffer_fun)audiofreeverb_freeverb_reset_buffer, + .get_buffer = (audiosample_get_buffer_fun)audiofreeverb_freeverb_get_buffer, +}; + +MP_DEFINE_CONST_OBJ_TYPE( + audiofreeverb_freeverb_type, + MP_QSTR_freeverb, + MP_TYPE_FLAG_HAS_SPECIAL_ACCESSORS, + make_new, audiofreeverb_freeverb_make_new, + locals_dict, &audiofreeverb_freeverb_locals_dict, + protocol, &audiofreeverb_freeverb_proto + ); diff --git a/shared-bindings/audiofreeverb/Freeverb.h b/shared-bindings/audiofreeverb/Freeverb.h new file mode 100644 index 0000000000000..913953ebecf62 --- /dev/null +++ b/shared-bindings/audiofreeverb/Freeverb.h @@ -0,0 +1,36 @@ +// This file is part of the CircuitPython project: https://circuitpython.org +// +// SPDX-FileCopyrightText: Copyright (c) 2025 Mark Komus +// +// SPDX-License-Identifier: MIT + +#pragma once + +#include "shared-module/audiofreeverb/Freeverb.h" + +extern const mp_obj_type_t audiofreeverb_freeverb_type; + +void common_hal_audiofreeverb_freeverb_construct(audiofreeverb_freeverb_obj_t *self, + mp_obj_t roomsize, mp_obj_t damp, 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_audiofreeverb_freeverb_deinit(audiofreeverb_freeverb_obj_t *self); +bool common_hal_audiofreeverb_freeverb_deinited(audiofreeverb_freeverb_obj_t *self); + +uint32_t common_hal_audiofreeverb_freeverb_get_sample_rate(audiofreeverb_freeverb_obj_t *self); +uint8_t common_hal_audiofreeverb_freeverb_get_channel_count(audiofreeverb_freeverb_obj_t *self); +uint8_t common_hal_audiofreeverb_freeverb_get_bits_per_sample(audiofreeverb_freeverb_obj_t *self); + +mp_obj_t common_hal_audiofreeverb_freeverb_get_roomsize(audiofreeverb_freeverb_obj_t *self); +void common_hal_audiofreeverb_freeverb_set_roomsize(audiofreeverb_freeverb_obj_t *self, mp_obj_t feedback); + +mp_obj_t common_hal_audiofreeverb_freeverb_get_damp(audiofreeverb_freeverb_obj_t *self); +void common_hal_audiofreeverb_freeverb_set_damp(audiofreeverb_freeverb_obj_t *self, mp_obj_t damp); + +mp_obj_t common_hal_audiofreeverb_freeverb_get_mix(audiofreeverb_freeverb_obj_t *self); +void common_hal_audiofreeverb_freeverb_set_mix(audiofreeverb_freeverb_obj_t *self, mp_obj_t mix); + +bool common_hal_audiofreeverb_freeverb_get_playing(audiofreeverb_freeverb_obj_t *self); +void common_hal_audiofreeverb_freeverb_play(audiofreeverb_freeverb_obj_t *self, mp_obj_t sample, bool loop); +void common_hal_audiofreeverb_freeverb_stop(audiofreeverb_freeverb_obj_t *self); diff --git a/shared-bindings/audiofreeverb/__init__.c b/shared-bindings/audiofreeverb/__init__.c new file mode 100644 index 0000000000000..cb8c979c8cfec --- /dev/null +++ b/shared-bindings/audiofreeverb/__init__.c @@ -0,0 +1,34 @@ +// This file is part of the CircuitPython project: https://circuitpython.org +// +// SPDX-FileCopyrightText: Copyright (c) 2024 Mark Komus +// +// SPDX-License-Identifier: MIT + +#include + +#include "py/obj.h" +#include "py/runtime.h" + +#include "shared-bindings/audiofreeverb/__init__.h" +#include "shared-bindings/audiofreeverb/Freeverb.h" + + +//| """Support for audio freeverb effect +//| +//| The `audiofreeverb` module contains classes to provide access to audio freeverb effects. +//| +//| """ + +static const mp_rom_map_elem_t audiofreeverb_module_globals_table[] = { + { MP_ROM_QSTR(MP_QSTR___name__), MP_ROM_QSTR(MP_QSTR_audiofreeverb) }, + { MP_ROM_QSTR(MP_QSTR_Freeverb), MP_ROM_PTR(&audiofreeverb_freeverb_type) }, +}; + +static MP_DEFINE_CONST_DICT(audiofreeverb_module_globals, audiofreeverb_module_globals_table); + +const mp_obj_module_t audiofreeverb_module = { + .base = { &mp_type_module }, + .globals = (mp_obj_dict_t *)&audiofreeverb_module_globals, +}; + +MP_REGISTER_MODULE(MP_QSTR_audiofreeverb, audiofreeverb_module); diff --git a/shared-bindings/audiofreeverb/__init__.h b/shared-bindings/audiofreeverb/__init__.h new file mode 100644 index 0000000000000..66463561f5443 --- /dev/null +++ b/shared-bindings/audiofreeverb/__init__.h @@ -0,0 +1,7 @@ +// This file is part of the CircuitPython project: https://circuitpython.org +// +// SPDX-FileCopyrightText: Copyright (c) 2025 Mark Komus +// +// SPDX-License-Identifier: MIT + +#pragma once diff --git a/shared-module/audiofreeverb/Freeverb.c b/shared-module/audiofreeverb/Freeverb.c new file mode 100644 index 0000000000000..d5c4671595e1c --- /dev/null +++ b/shared-module/audiofreeverb/Freeverb.c @@ -0,0 +1,358 @@ +// This file is part of the CircuitPython project: https://circuitpython.org +// +// SPDX-FileCopyrightText: Copyright (c) 2025 Mark Komus +// +// SPDX-License-Identifier: MIT +// +// Based on FreeVerb - https://github.com/sinshu/freeverb/tree/main +// Fixed point ideas from - Paul Stoffregen in the Teensy audio library https://github.com/PaulStoffregen/Audio/blob/master/effect_freeverb.cpp +// +#include "shared-bindings/audiofreeverb/Freeverb.h" + +#include +#include "py/runtime.h" +#include + +void common_hal_audiofreeverb_freeverb_construct(audiofreeverb_freeverb_obj_t *self, mp_obj_t roomsize, mp_obj_t damp, 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->base.bits_per_sample = bits_per_sample; // Most common is 16, but 8 is also supported in many places + self->base.samples_signed = samples_signed; // Are the samples we provide signed (common is true) + self->base.channel_count = channel_count; // Channels can be 1 for mono or 2 for stereo + self->base.sample_rate = sample_rate; // Sample rate for the effect, this generally needs to match all audio objects + self->base.single_buffer = false; + self->base.max_buffer_length = buffer_size; + + // 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_maybe(self->buffer_len); + if (self->buffer[0] == NULL) { + common_hal_audiofreeverb_freeverb_deinit(self); + m_malloc_fail(self->buffer_len); + } + memset(self->buffer[0], 0, self->buffer_len); + + self->buffer[1] = m_malloc_maybe(self->buffer_len); + if (self->buffer[1] == NULL) { + common_hal_audiofreeverb_freeverb_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 + + // 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 reverb effect's starting values. For a different effect this section will change + if (roomsize == MP_OBJ_NULL) { + roomsize = mp_obj_new_float(MICROPY_FLOAT_CONST(0.5)); + } + synthio_block_assign_slot(roomsize, &self->roomsize, MP_QSTR_roomsize); + common_hal_audiofreeverb_freeverb_set_roomsize(self, roomsize); + + if (damp == MP_OBJ_NULL) { + damp = mp_obj_new_float(MICROPY_FLOAT_CONST(0.5)); + } + synthio_block_assign_slot(damp, &self->damp, MP_QSTR_damp); + common_hal_audiofreeverb_freeverb_set_damp(self, damp); + + if (mix == MP_OBJ_NULL) { + mix = mp_obj_new_float(MICROPY_FLOAT_CONST(0.5)); + } + synthio_block_assign_slot(mix, &self->mix, MP_QSTR_mix); + common_hal_audiofreeverb_freeverb_set_mix(self, mix); + + // Set up the comb filters + // These values come from FreeVerb and are selected for the best reverb sound + self->combbuffersizes[0] = self->combbuffersizes[8] = 1116; + self->combbuffersizes[1] = self->combbuffersizes[9] = 1188; + self->combbuffersizes[2] = self->combbuffersizes[10] = 1277; + self->combbuffersizes[3] = self->combbuffersizes[11] = 1356; + self->combbuffersizes[4] = self->combbuffersizes[12] = 1422; + self->combbuffersizes[5] = self->combbuffersizes[13] = 1491; + self->combbuffersizes[6] = self->combbuffersizes[14] = 1557; + self->combbuffersizes[7] = self->combbuffersizes[15] = 1617; + for (uint32_t i = 0; i < 8 * channel_count; i++) { + self->combbuffers[i] = m_malloc_maybe(self->combbuffersizes[i] * sizeof(uint16_t)); + if (self->combbuffers[i] == NULL) { + common_hal_audiofreeverb_freeverb_deinit(self); + m_malloc_fail(self->combbuffersizes[i]); + } + memset(self->combbuffers[i], 0, self->combbuffersizes[i]); + + self->combbufferindex[i] = 0; + self->combfitlers[i] = 0; + } + + // Set up the allpass filters + // These values come from FreeVerb and are selected for the best reverb sound + self->allpassbuffersizes[0] = self->allpassbuffersizes[4] = 556; + self->allpassbuffersizes[1] = self->allpassbuffersizes[5] = 441; + self->allpassbuffersizes[2] = self->allpassbuffersizes[6] = 341; + self->allpassbuffersizes[3] = self->allpassbuffersizes[7] = 225; + for (uint32_t i = 0; i < 4 * channel_count; i++) { + self->allpassbuffers[i] = m_malloc_maybe(self->allpassbuffersizes[i] * sizeof(uint16_t)); + if (self->allpassbuffers[i] == NULL) { + common_hal_audiofreeverb_freeverb_deinit(self); + m_malloc_fail(self->allpassbuffersizes[i]); + } + memset(self->allpassbuffers[i], 0, self->allpassbuffersizes[i]); + + self->allpassbufferindex[i] = 0; + } +} + +bool common_hal_audiofreeverb_freeverb_deinited(audiofreeverb_freeverb_obj_t *self) { + if (self->buffer[0] == NULL) { + return true; + } + return false; +} + +void common_hal_audiofreeverb_freeverb_deinit(audiofreeverb_freeverb_obj_t *self) { + if (common_hal_audiofreeverb_freeverb_deinited(self)) { + return; + } + self->buffer[0] = NULL; + self->buffer[1] = NULL; +} + +mp_obj_t common_hal_audiofreeverb_freeverb_get_roomsize(audiofreeverb_freeverb_obj_t *self) { + return self->roomsize.obj; +} + +void common_hal_audiofreeverb_freeverb_set_roomsize(audiofreeverb_freeverb_obj_t *self, mp_obj_t roomsize_obj) { + synthio_block_assign_slot(roomsize_obj, &self->roomsize, MP_QSTR_roomsize); +} + +int16_t audiofreeverb_freeverb_get_roomsize_fixedpoint(mp_float_t n) { + if (n > (mp_float_t)MICROPY_FLOAT_CONST(1.0)) { + n = MICROPY_FLOAT_CONST(1.0); + } else if (n < (mp_float_t)MICROPY_FLOAT_CONST(0.0)) { + n = MICROPY_FLOAT_CONST(0.0); + } + + return (int16_t)(n * (mp_float_t)MICROPY_FLOAT_CONST(9175.04)) + 22937; // 9175.04 = 0.28f in fixed point 22937 = 0.7f +} + +mp_obj_t common_hal_audiofreeverb_freeverb_get_damp(audiofreeverb_freeverb_obj_t *self) { + return self->damp.obj; +} + +void common_hal_audiofreeverb_freeverb_set_damp(audiofreeverb_freeverb_obj_t *self, mp_obj_t damp) { + synthio_block_assign_slot(damp, &self->damp, MP_QSTR_damp); +} + +void audiofreeverb_freeverb_get_damp_fixedpoint(mp_float_t n, int16_t *damp1, int16_t *damp2) { + if (n > (mp_float_t)MICROPY_FLOAT_CONST(1.0)) { + n = MICROPY_FLOAT_CONST(1.0); + } else if (n < (mp_float_t)MICROPY_FLOAT_CONST(0.0)) { + n = MICROPY_FLOAT_CONST(0.0); + } + + *damp1 = (int16_t)(n * (mp_float_t)MICROPY_FLOAT_CONST(13107.2)); // 13107.2 = 0.4f scaling factor + *damp2 = (int16_t)(32768 - *damp1); // inverse of x1 damp2 = 1.0 - damp1 +} + +mp_obj_t common_hal_audiofreeverb_freeverb_get_mix(audiofreeverb_freeverb_obj_t *self) { + return self->mix.obj; +} + +void common_hal_audiofreeverb_freeverb_set_mix(audiofreeverb_freeverb_obj_t *self, mp_obj_t mix) { + synthio_block_assign_slot(mix, &self->mix, MP_QSTR_mix); +} + +void audiofreeverb_freeverb_get_mix_fixedpoint(mp_float_t mix, int16_t *mix_sample, int16_t *mix_effect) { + mix = mix * (mp_float_t)MICROPY_FLOAT_CONST(2.0); + *mix_sample = (int16_t)(MIN((mp_float_t)MICROPY_FLOAT_CONST(2.0) - mix, (mp_float_t)MICROPY_FLOAT_CONST(1.0)) * 32767); + *mix_effect = (int16_t)(MIN(mix, (mp_float_t)MICROPY_FLOAT_CONST(1.0)) * 32767); +} + +void audiofreeverb_freeverb_reset_buffer(audiofreeverb_freeverb_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); +} + +bool common_hal_audiofreeverb_freeverb_get_playing(audiofreeverb_freeverb_obj_t *self) { + return self->sample != NULL; +} + +void common_hal_audiofreeverb_freeverb_play(audiofreeverb_freeverb_obj_t *self, mp_obj_t sample, bool loop) { + audiosample_must_match(&self->base, sample); + + 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->base.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_audiofreeverb_freeverb_stop(audiofreeverb_freeverb_obj_t *self) { + // When the sample is set to stop playing do any cleanup here + // For reverb we clear the sample but the reverb continues until the object reading our effect stops + self->sample = NULL; + return; +} + +// cleaner sat16 by http://www.moseleyinstruments.com/ +static int16_t sat16(int32_t n, int rshift) { + // we should always round towards 0 + // to avoid recirculating round-off noise + // + // a 2s complement positive number is always + // rounded down, so we only need to take + // care of negative numbers + if (n < 0) { + n = n + (~(0xFFFFFFFFUL << rshift)); + } + n = n >> rshift; + if (n > 32767) { + return 32767; + } + if (n < -32768) { + return -32768; + } + return n; +} + +audioio_get_buffer_result_t audiofreeverb_freeverb_get_buffer(audiofreeverb_freeverb_obj_t *self, bool single_channel_output, uint8_t channel, + uint8_t **buffer, uint32_t *buffer_length) { + + // Switch our buffers to the other buffer + self->last_buf_idx = !self->last_buf_idx; + + // 16 bit samples we need a 16 bit pointer + int16_t *word_buffer = (int16_t *)self->buffer[self->last_buf_idx]; + uint32_t length = self->buffer_len / (self->base.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 but we still need to play the reverb + 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->base.bits_per_sample / 8); + self->more_data = result == GET_BUFFER_MORE_DATA; + } + } + + // 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; + if (self->sample == NULL) { + n = MIN(length, SYNTHIO_MAX_DUR * self->base.channel_count); + } else { + n = MIN(MIN(self->sample_buffer_length, length), SYNTHIO_MAX_DUR * self->base.channel_count); + } + + // get the effect values we need from the BlockInput. These may change at run time so you need to do bounds checking if required + shared_bindings_synthio_lfo_tick(self->base.sample_rate, n / self->base.channel_count); + mp_float_t damp = synthio_block_slot_get_limited(&self->damp, MICROPY_FLOAT_CONST(0.0), MICROPY_FLOAT_CONST(1.0)); + int16_t damp1, damp2; + audiofreeverb_freeverb_get_damp_fixedpoint(damp, &damp1, &damp2); + + mp_float_t mix = synthio_block_slot_get_limited(&self->mix, MICROPY_FLOAT_CONST(0.0), MICROPY_FLOAT_CONST(1.0)); + int16_t mix_sample, mix_effect; + audiofreeverb_freeverb_get_mix_fixedpoint(mix, &mix_sample, &mix_effect); + + mp_float_t roomsize = synthio_block_slot_get_limited(&self->roomsize, MICROPY_FLOAT_CONST(0.0), MICROPY_FLOAT_CONST(1.0)); + int16_t feedback = audiofreeverb_freeverb_get_roomsize_fixedpoint(roomsize); + + int16_t *sample_src = (int16_t *)self->sample_remaining_buffer; + + for (uint32_t i = 0; i < n; i++) { + int32_t sample_word = 0; + if (self->sample != NULL) { + sample_word = sample_src[i]; + } + + int32_t word, sum; + int16_t input, bufout, output; + uint32_t channel_comb_offset = 0, channel_allpass_offset = 0; + + input = sat16(sample_word * 8738, 17); // Initial input scaled down so we can add reverb + sum = 0; + + // Calculate each of the 8 comb buffers + for (uint32_t j = 0 + channel_comb_offset; j < 8 + channel_comb_offset; j++) { + bufout = self->combbuffers[j][self->combbufferindex[j]]; + sum += bufout; + self->combfitlers[j] = sat16(bufout * damp2 + self->combfitlers[j] * damp1, 15); + self->combbuffers[j][self->combbufferindex[j]] = sat16(input + sat16(self->combfitlers[j] * feedback, 15), 0); + if (++self->combbufferindex[j] >= self->combbuffersizes[j]) { + self->combbufferindex[j] = 0; + } + } + + output = sat16(sum * 31457, 17); // 31457 = 0.24f with shift of 17 + + // Calculate each of the 4 all pass buffers + for (uint32_t j = 0 + channel_allpass_offset; j < 4 + channel_allpass_offset; j++) { + bufout = self->allpassbuffers[j][self->allpassbufferindex[j]]; + self->allpassbuffers[j][self->allpassbufferindex[j]] = output + (bufout >> 1); // bufout >> 1 same as bufout*0.5f + output = sat16(bufout - output, 1); + if (++self->allpassbufferindex[j] >= self->allpassbuffersizes[j]) { + self->allpassbufferindex[j] = 0; + } + } + + word = output * 30; // Add some volume back don't have to saturate as next step will + + word = sat16(sample_word * mix_sample, 15) + sat16(word * mix_effect, 15); + word = synthio_mix_down_sample(word, SYNTHIO_MIX_DOWN_SCALE(2)); + word_buffer[i] = (int16_t)word; + + if ((self->base.channel_count == 2) && (channel_comb_offset == 0)) { + channel_comb_offset = 8; + channel_allpass_offset = 4; + } else { + channel_comb_offset = 0; + channel_allpass_offset = 0; + } + } + + // Update the remaining length and the buffer positions based on how much we wrote into our buffer + length -= n; + word_buffer += n; + self->sample_remaining_buffer += (n * (self->base.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; + + // Reverb 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; +} diff --git a/shared-module/audiofreeverb/Freeverb.h b/shared-module/audiofreeverb/Freeverb.h new file mode 100644 index 0000000000000..44747f0fc951d --- /dev/null +++ b/shared-module/audiofreeverb/Freeverb.h @@ -0,0 +1,56 @@ +// This file is part of the CircuitPython project: https://circuitpython.org +// +// SPDX-FileCopyrightText: Copyright (c) 2025 Mark Komus +// +// SPDX-License-Identifier: MIT +#pragma once + +#include "py/obj.h" + +#include "shared-module/audiocore/__init__.h" +#include "shared-module/synthio/__init__.h" +#include "shared-module/synthio/block.h" + +extern const mp_obj_type_t audiofreeverb_freeverb_type; + +typedef struct { + audiosample_base_t base; + synthio_block_slot_t roomsize; + synthio_block_slot_t damp; + synthio_block_slot_t mix; + + 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; + + bool loop; + bool more_data; + + int16_t combbuffersizes[16]; + int16_t *combbuffers[16]; + int16_t combbufferindex[16]; + int16_t combfitlers[16]; + + int16_t allpassbuffersizes[8]; + int16_t *allpassbuffers[8]; + int16_t allpassbufferindex[8]; + + mp_obj_t sample; +} audiofreeverb_freeverb_obj_t; + +void audiofreeverb_freeverb_reset_buffer(audiofreeverb_freeverb_obj_t *self, + bool single_channel_output, + uint8_t channel); + +audioio_get_buffer_result_t audiofreeverb_freeverb_get_buffer(audiofreeverb_freeverb_obj_t *self, + bool single_channel_output, + uint8_t channel, + uint8_t **buffer, + uint32_t *buffer_length); // length in bytes + +int16_t audiofreeverb_freeverb_get_roomsize_fixedpoint(mp_float_t n); +void audiofreeverb_freeverb_get_damp_fixedpoint(mp_float_t n, int16_t *damp1, int16_t *damp2); +void audiofreeverb_freeverb_get_mix_fixedpoint(mp_float_t mix, int16_t *mix_sample, int16_t *mix_effect); diff --git a/shared-module/audiofreeverb/__init__.c b/shared-module/audiofreeverb/__init__.c new file mode 100644 index 0000000000000..94cd4caa3bd1b --- /dev/null +++ b/shared-module/audiofreeverb/__init__.c @@ -0,0 +1,5 @@ +// This file is part of the CircuitPython project: https://circuitpython.org +// +// SPDX-FileCopyrightText: Copyright (c) 2025 Mark Komus +// +// SPDX-License-Identifier: MIT diff --git a/shared-module/audiofreeverb/__init__.h b/shared-module/audiofreeverb/__init__.h new file mode 100644 index 0000000000000..66463561f5443 --- /dev/null +++ b/shared-module/audiofreeverb/__init__.h @@ -0,0 +1,7 @@ +// This file is part of the CircuitPython project: https://circuitpython.org +// +// SPDX-FileCopyrightText: Copyright (c) 2025 Mark Komus +// +// SPDX-License-Identifier: MIT + +#pragma once pFad - Phonifier reborn

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