daScript/modules/dasAudio/src/dasAudio.cpp at master · WhyNot135/daScript · GitHub

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#include "daScript/misc/platform.h"

#include "daScript/ast/ast.h"

#include "daScript/ast/ast_interop.h"

#include "daScript/ast/ast_typefactory_bind.h"

#include "daScript/ast/ast_handle.h"

#include "daScript/simulate/bind_enum.h"

#include <atomic>

// include vorbis extras before miniaudio

#define STB_VORBIS_HEADER_ONLY

#ifdef __forceinline

#undef __forceinline

#define __forceinline inline

#endif

#include <extras/stb_vorbis.c> /* Enables Vorbis decoding. */

#ifdef _MSC_VER

#pragma warning(push)

#pragma warning(disable:4755)

#pragma warning(disable:4701)

#endif

#ifdef __APPLE__

#define MA_NO_RUNTIME_LINKING

#endif

#define MINIAUDIO_IMPLEMENTATION

#define MA_USE_STDINT

#include <miniaudio.h>

#ifdef _MSC_VER

#pragma warning(pop)

#endif

#include "volume_mixer.h"

#include "hrtf.h"

#define I3DL32_REVERB_IMPLEMENTATION 1

#include "reverb.h"

#define MA_CHORUS_IMPLEMENTATION

#include "chorus.h"

#define CONVOLUTION_REVERB_IMPLEMENTATION

#include "convolution_reverb.h"

#define MA_EFFECTS_IMPLEMENTATION

#include "effects.h"

#define MA_PHASER_IMPLEMENTATION

#include "phaser.h"

#define MA_TREMOLO_IMPLEMENTATION

#include "tremolo.h"

#define MA_COMPRESSOR_IMPLEMENTATION

#include "compressor.h"

#include "dasAudio.h"

#ifndef HRTF_SAMPLE_RATE

#define HRTF_SAMPLE_RATE 48000

#endif

// Non-static wrapper for ma_sf2_biquad_tick (static in sf2_voice.h)

float das_ma_sf2_biquad_tick ( ma_sf2_biquad * bq, float input ) {

return ma_sf2_biquad_tick(bq, input);

}

MAKE_EXTERNAL_TYPE_FACTORY(Context,Context);

das::Context* get_clone_context( das::Context * ctx, uint32_t category );//link time resolved dependencies

MAKE_TYPE_FACTORY(ma_resampler_config,ma_resampler_config);

MAKE_TYPE_FACTORY(ma_resampler,ma_resampler);

MAKE_TYPE_FACTORY(ma_channel_converter_config,ma_channel_converter_config);

MAKE_TYPE_FACTORY(ma_channel_converter,ma_channel_converter);

MAKE_TYPE_FACTORY(ma_volume_mixer,ma_volume_mixer);

MAKE_TYPE_FACTORY(ma_decoder_config,ma_decoder_config);

MAKE_TYPE_FACTORY(ma_decoder,ma_decoder);

MAKE_TYPE_FACTORY(ma_limiter,ma_limiter);

MAKE_TYPE_FACTORY(ma_sf2_envelope,ma_sf2_envelope);

MAKE_TYPE_FACTORY(ma_sf2_biquad,ma_sf2_biquad);

MAKE_TYPE_FACTORY(ma_sf2_voice,ma_sf2_voice);

MAKE_TYPE_FACTORY(ma_chorus_config,ma_chorus_config);

MAKE_TYPE_FACTORY(ma_chorus,ma_chorus);

MAKE_TYPE_FACTORY(ma_delay,ma_delay);

MAKE_TYPE_FACTORY(ma_bitcrush,ma_bitcrush);

MAKE_TYPE_FACTORY(ma_waveshaper,ma_waveshaper);

MAKE_TYPE_FACTORY(ma_djfilter,ma_djfilter);

MAKE_TYPE_FACTORY(ma_bandpass,ma_bandpass);

MAKE_TYPE_FACTORY(ma_phaser,ma_phaser);

MAKE_TYPE_FACTORY(ma_tremolo,ma_tremolo);

MAKE_TYPE_FACTORY(ma_compressor,ma_compressor);

DAS_BASE_BIND_ENUM ( ma_format, ma_format, \

ma_format_unknown, \

ma_format_u8, \

ma_format_s16, \

ma_format_s24, \

ma_format_s32, \

ma_format_f32 \

);

DAS_BIND_ENUM_CAST ( ma_format );

DAS_BASE_BIND_ENUM ( ma_resample_algorithm, ma_resample_algorithm, \

ma_resample_algorithm_linear, \

ma_resample_algorithm_custom \

);

DAS_BIND_ENUM_CAST ( ma_resample_algorithm );

DAS_BASE_BIND_ENUM ( ma_channel_mix_mode, ma_channel_mix_mode, \

ma_channel_mix_mode_rectangular, \

ma_channel_mix_mode_simple, \

ma_channel_mix_mode_custom_weights, \

ma_channel_mix_mode_default \

);

DAS_BIND_ENUM_CAST ( ma_channel_mix_mode );

DAS_BASE_BIND_ENUM ( ma_dither_mode, ma_dither_mode, \

ma_dither_mode_none, \

ma_dither_mode_rectangle, \

ma_dither_mode_triangle \

);

DAS_BIND_ENUM_CAST ( ma_dither_mode );

DAS_BASE_BIND_ENUM ( ma_result, ma_result, \

MA_SUCCESS ,\

MA_ERROR ,\

MA_INVALID_ARGS ,\

MA_INVALID_OPERATION ,\

MA_OUT_OF_MEMORY ,\

MA_OUT_OF_RANGE ,\

MA_ACCESS_DENIED ,\

MA_DOES_NOT_EXIST ,\

MA_ALREADY_EXISTS ,\

MA_TOO_MANY_OPEN_FILES ,\

MA_INVALID_FILE ,\

MA_TOO_BIG ,\

MA_PATH_TOO_LONG ,\

MA_NAME_TOO_LONG ,\

MA_NOT_DIRECTORY ,\

MA_IS_DIRECTORY ,\

MA_DIRECTORY_NOT_EMPTY ,\

MA_AT_END ,\

MA_NO_SPACE ,\

MA_BUSY ,\

MA_IO_ERROR ,\

MA_INTERRUPT ,\

MA_UNAVAILABLE ,\

MA_ALREADY_IN_USE ,\

MA_BAD_ADDRESS ,\

MA_BAD_SEEK ,\

MA_BAD_PIPE ,\

MA_DEADLOCK ,\

MA_TOO_MANY_LINKS ,\

MA_NOT_IMPLEMENTED ,\

MA_NO_MESSAGE ,\

MA_BAD_MESSAGE ,\

MA_NO_DATA_AVAILABLE ,\

MA_INVALID_DATA ,\

MA_TIMEOUT ,\

MA_NO_NETWORK ,\

MA_NOT_UNIQUE ,\

MA_NOT_SOCKET ,\

MA_NO_ADDRESS ,\

MA_BAD_PROTOCOL ,\

MA_PROTOCOL_UNAVAILABLE ,\

MA_PROTOCOL_NOT_SUPPORTED ,\

MA_PROTOCOL_FAMILY_NOT_SUPPORTED ,\

MA_ADDRESS_FAMILY_NOT_SUPPORTED ,\

MA_SOCKET_NOT_SUPPORTED ,\

MA_CONNECTION_RESET ,\

MA_ALREADY_CONNECTED ,\

MA_NOT_CONNECTED ,\

MA_CONNECTION_REFUSED ,\

MA_NO_HOST ,\

MA_IN_PROGRESS ,\

MA_CANCELLED ,\

MA_MEMORY_ALREADY_MAPPED ,\

MA_FORMAT_NOT_SUPPORTED ,\

MA_DEVICE_TYPE_NOT_SUPPORTED ,\

MA_SHARE_MODE_NOT_SUPPORTED ,\

MA_NO_BACKEND ,\

MA_NO_DEVICE ,\

MA_API_NOT_FOUND ,\

MA_INVALID_DEVICE_CONFIG ,\

MA_LOOP ,\

MA_DEVICE_NOT_INITIALIZED ,\

MA_DEVICE_ALREADY_INITIALIZED ,\

MA_DEVICE_NOT_STARTED ,\

MA_DEVICE_NOT_STOPPED ,\

MA_FAILED_TO_INIT_BACKEND ,\

MA_FAILED_TO_OPEN_BACKEND_DEVICE ,\

MA_FAILED_TO_START_BACKEND_DEVICE ,\

MA_FAILED_TO_STOP_BACKEND_DEVICE \

);

DAS_BIND_ENUM_CAST ( ma_result );

DAS_BASE_BIND_ENUM( das::I3DL2Preset, I3DL2Preset, \

Generic, \

PaddedCell, \

Room, \

Bathroom, \

LivingRoom, \

StoneRoom, \

Auditorium, \

ConcertHall, \

Cave, \

Arena, \

Hangar, \

CarpetedHallway, \

Hallway, \

StoneCorridor, \

Alley, \

Forest, \

City, \

Mountains, \

Quarry, \

Plain, \

ParkingLot, \

SewerPipe, \

Underwater \

);

DAS_BIND_ENUM_CAST ( das::I3DL2Preset );

// ConvReverbQualityEnum is defined in dasAudio.h (das-facing names for the C ConvReverbQuality enum)

DAS_BASE_BIND_ENUM( ConvReverbQualityEnum, ConvReverbQuality, high, medium, low );

DAS_BIND_ENUM_CAST ( ConvReverbQualityEnum );

MAKE_TYPE_FACTORY(I3DL2ReverbProperties,I3DL2ReverbProperties);

MAKE_TYPE_FACTORY(I3DL2Reverb,I3DL2Reverb);

MAKE_TYPE_FACTORY(ma_hrtf,ma_hrtf);

MAKE_TYPE_FACTORY(ConvolutionReverb,ConvolutionReverb);

namespace das {

static ma_device g_device;

static ma_log g_ma_log_struct;

static shared_ptr<Context> g_mixer_context;

static daScriptEnvironment * g_mixer_env = nullptr;

static Func g_mixer_function = (void *) nullptr;

static bool g_mixer_initialized = false;

static int g_rate = 0;

static int g_channels = 0;

// Null-backend mode (for headless tests / CI): force miniaudio's ma_backend_null

// so the device + audio thread run without real hardware. Set via sound_set_null_device.

static bool g_force_null_backend = false;

static ma_context g_null_context;

static bool g_null_context_inited = false;

// Recorded at dasAudio_init: whether this audio system came up single-threaded —

// i.e. the device callback (command_processor) runs on the MAIN thread, with no

// separate audio thread to drain the command queue. The teardown reads this to

// decide whether it must drain the queue itself (single-thread) or wait for the

// audio thread (multi-thread, the original path). Today it's the emscripten

// no-pthread build (ScriptProcessor callback on main); kept as a runtime flag,

// not a compile-time check at the call sites, so the audio code asks "how was I

// initialized" rather than "what platform is this".

static bool g_audio_single_threaded = false;

// ---- capture (microphone) ----

// A capture device independent of the playback device above: recording has its own lifecycle, so

// it does not share g_device / g_mixer_context. The RT capture callback writes interleaved f32 into

// g_capture_rb (a lock-free single-producer/single-consumer ring); scripts drain it on the main

// thread via dasAudio_record_read. No daScript code runs on the audio thread — that is what lets

// captured data cross into a context safely (see the threading notes in the module docs).

static ma_context g_capture_context;

static bool g_capture_context_inited = false;

static bool g_capture_context_is_null = false; // which backend g_capture_context was built with (tracks g_force_null_backend)

static ma_device g_capture_device;

static ma_pcm_rb g_capture_rb;

static bool g_capture_initialized = false;

static int g_capture_channels = 0;

// Frames dropped because the ring was full when the callback tried to write (drain loop too slow).

// Written on the audio thread, read on the main thread — atomic to stay data-race-clean under TSAN.

static std::atomic<uint64_t> g_capture_overflow_frames { 0 };

// Value-copies of the last capture enumeration, so device names/ids survive past the context-internal

// buffer that ma_context_get_devices returns. Populated by dasAudio_record_device_count (main thread).

static vector<ma_device_info> g_capture_device_cache;

void dasAudio_set_null_device ( bool enabled ) {

g_force_null_backend = enabled;

// The capture context is rebuilt lazily in ensure_capture_context when its backend no longer

// matches this flag — so a toggle here (even mid-capture) takes effect on the next idle use.

}

bool dasAudio_is_single_threaded () {

return g_audio_single_threaded;

}

void on_error_log ( void * , ma_uint32 level, const char * message ) {

if (level <= 1) {

LOG(LogLevel::error) << message;

} else {

LOG(LogLevel::info) << message;

}

void data_callback(ma_device*, void* pOutput, const void*, ma_uint32 frameCount) {

float fdt = 1.0f / float(g_rate);

Array buffer;

array_mark_locked(buffer, pOutput, frameCount * g_channels);

lock_guard<recursive_mutex> guard(*g_mixer_context->contextMutex);

auto saved = daScriptEnvironment::exchangeBound(g_mixer_env);

g_mixer_context->restart();

if ( !g_mixer_context.get()->runWithCatch([&](){

das_invoke_function<void>::invoke<Array&,int32_t,int32_t>(g_mixer_context.get(),nullptr,g_mixer_function,buffer,g_channels,g_rate,fdt);

}) ) {

g_mixer_context->to_err(&g_mixer_context->exceptionAt, g_mixer_context->getException());

g_mixer_context->clearException();

}

daScriptEnvironment::setBound(saved);

}

Context & dasAudio_mixerContext ( Context * context, LineInfoArg * at ) {

if ( !g_mixer_context ) context->throw_error_at(at,"sound mixer is not initialized");

return *g_mixer_context;

}

bool dasAudio_init ( TFunc<void,TTemporary<TArray<float>>,int32_t,int32_t,float> mixer, int32_t rate, int32_t channels, Context & context ) {

g_mixer_initialized = false;

// Record how this audio system comes up: single-threaded (device callback on

// the main thread, no separate audio thread) vs threaded. See g_audio_single_threaded.

#if defined(__EMSCRIPTEN__) && !defined(__EMSCRIPTEN_PTHREADS__)

g_audio_single_threaded = true;

#else

g_audio_single_threaded = false;

#endif

g_rate = rate;

g_channels = channels;

// log

ma_log_init(nullptr, &g_ma_log_struct);

ma_log_register_callback(&g_ma_log_struct, {on_error_log, nullptr});

// device

ma_device_config deviceConfig;

deviceConfig = ma_device_config_init(ma_device_type_playback);

deviceConfig.playback.format = ma_format_f32;

deviceConfig.playback.channels = g_channels;

deviceConfig.sampleRate = g_rate;

deviceConfig.dataCallback = data_callback;

deviceConfig.pUserData = NULL;

ma_context * pContext = nullptr;

if ( g_force_null_backend ) {

// Restrict to the null backend (timer-driven callback, no hardware).

ma_backend nullBackend = ma_backend_null;

if ( ma_context_init(&nullBackend, 1, nullptr, &g_null_context) != MA_SUCCESS ) {

LOG(LogLevel::error) << "failed to init null audio context.\n";

return false;

}

g_null_context_inited = true;

pContext = &g_null_context;

}

if ( ma_device_init(pContext, &deviceConfig, &g_device) != MA_SUCCESS ) {

LOG(LogLevel::error) << "failed to open playback device.\n";

if ( g_null_context_inited ) { ma_context_uninit(&g_null_context); g_null_context_inited = false; }

return false;

}

g_mixer_context.reset(get_clone_context(&context,uint32_t(ContextCategory::audio_context)));

g_mixer_context->verySafeContext = false;

g_mixer_function = mixer;

g_mixer_env = daScriptEnvironment::getBound();

if ( ma_device_start(&g_device) != MA_SUCCESS ) {

ma_device_uninit(&g_device);

if ( g_null_context_inited ) { ma_context_uninit(&g_null_context); g_null_context_inited = false; }

g_mixer_context.reset();

return false;

}

g_mixer_initialized = true;

return true;

}

// Bring up a persistent context for capture: enumeration hands back an opaque, context-owned

// ma_device_id, and the device we later open on it must outlive that context — so we keep one

// around rather than the transient (pContext==nullptr) form the playback path uses. Honors the

// null-backend override so headless/CI runs enumerate+capture without real hardware.

static bool ensure_capture_context () {

// If the cached context was built for a different backend than currently requested (e.g.

// sound_set_null_device was toggled since), drop it and rebuild — but only while idle; a live

// capture device holds this context, so a mismatch during recording waits until the next use.

if ( g_capture_context_inited && g_capture_context_is_null != g_force_null_backend && !g_capture_initialized ) {

ma_context_uninit(&g_capture_context);

g_capture_context_inited = false;

g_capture_device_cache.clear();

}

if ( g_capture_context_inited ) return true;

ma_result r;

if ( g_force_null_backend ) {

ma_backend nullBackend = ma_backend_null;

r = ma_context_init(&nullBackend, 1, nullptr, &g_capture_context);

} else {

r = ma_context_init(nullptr, 0, nullptr, &g_capture_context);

}

if ( r != MA_SUCCESS ) {

LOG(LogLevel::error) << "failed to init capture context.\n";

return false;

}

g_capture_context_inited = true;

g_capture_context_is_null = g_force_null_backend;

return true;

}

void dasAudio_finalize ( void ) {

if ( g_mixer_initialized ) {

ma_device_uninit(&g_device);

g_mixer_context.reset();

g_mixer_initialized = false;

}

// capture teardown, in case a script never called sound_record_stop

if ( g_capture_initialized ) {

ma_device_uninit(&g_capture_device);

g_capture_initialized = false;

ma_pcm_rb_uninit(&g_capture_rb);

}

if ( g_capture_context_inited ) {

ma_context_uninit(&g_capture_context);

g_capture_context_inited = false;

}

g_capture_device_cache.clear();

if ( g_null_context_inited ) {

ma_context_uninit(&g_null_context);

g_null_context_inited = false;

}

// RT-thread capture callback: append the just-captured interleaved f32 frames to the ring. Pure

// C++ — no daScript, no allocation, no lock. On overflow (drain loop fell behind) the excess is

// dropped and counted; recording never blocks the audio thread.

static void capture_callback ( ma_device *, void *, const void * pInput, ma_uint32 frameCount ) {

if ( !g_capture_initialized || pInput == nullptr ) return;

const float * src = (const float *) pInput;

ma_uint32 channels = (ma_uint32) g_capture_channels;

ma_uint32 framesLeft = frameCount;

ma_uint32 srcFrame = 0;

while ( framesLeft > 0 ) {

ma_uint32 chunk = framesLeft;

void * pWrite = nullptr;

if ( ma_pcm_rb_acquire_write(&g_capture_rb, &chunk, &pWrite) != MA_SUCCESS ) break;

if ( chunk == 0 ) {

ma_pcm_rb_commit_write(&g_capture_rb, 0);

g_capture_overflow_frames.fetch_add(framesLeft, std::memory_order_relaxed);

break;

}

memcpy(pWrite, src + (size_t)srcFrame * channels, (size_t)chunk * channels * sizeof(float));

ma_pcm_rb_commit_write(&g_capture_rb, chunk);

srcFrame += chunk;

framesLeft -= chunk;

}

bool dasAudio_record_start ( int32_t rate, int32_t channels, int32_t rb_frames, int32_t device_index, Context * context, LineInfoArg * at ) {

if ( g_capture_initialized ) context->throw_error_at(at, "already recording; call sound_record_stop first");

if ( rate <= 0 || channels <= 0 ) context->throw_error_at(at, "record rate and channels must be positive");

if ( rb_frames <= 0 ) rb_frames = rate; // ~1s ring by default

if ( !ensure_capture_context() ) return false;

if ( ma_pcm_rb_init(ma_format_f32, (ma_uint32)channels, (ma_uint32)rb_frames, nullptr, nullptr, &g_capture_rb) != MA_SUCCESS ) {

LOG(LogLevel::error) << "failed to init capture ring buffer.\n";

return false;

}

ma_device_config cfg = ma_device_config_init(ma_device_type_capture);

cfg.capture.format = ma_format_f32;

cfg.capture.channels = (ma_uint32)channels;

cfg.sampleRate = (ma_uint32)rate;

cfg.dataCallback = capture_callback;

cfg.pUserData = nullptr;

if ( device_index >= 0 ) {

// Re-enumerate for a fresh, valid id; the pointer only needs to survive until ma_device_init copies it.

ma_device_info * pCapture = nullptr;

ma_uint32 captureCount = 0;

if ( ma_context_get_devices(&g_capture_context, nullptr, nullptr, &pCapture, &captureCount) == MA_SUCCESS

&& device_index < (int32_t)captureCount ) {

cfg.capture.pDeviceID = &pCapture[device_index].id;

} else {

LOG(LogLevel::error) << "capture device index out of range; falling back to default device.\n";

}

g_capture_overflow_frames.store(0, std::memory_order_relaxed);

if ( ma_device_init(&g_capture_context, &cfg, &g_capture_device) != MA_SUCCESS ) {

LOG(LogLevel::error) << "failed to open capture device.\n";

ma_pcm_rb_uninit(&g_capture_rb);

return false;

}

g_capture_channels = channels;

g_capture_initialized = true; // set before start: the callback may fire immediately

if ( ma_device_start(&g_capture_device) != MA_SUCCESS ) {

LOG(LogLevel::error) << "failed to start capture device.\n";

g_capture_initialized = false;

ma_device_uninit(&g_capture_device);

ma_pcm_rb_uninit(&g_capture_rb);

return false;

}

return true;

}

void dasAudio_record_stop ( void ) {

if ( !g_capture_initialized ) return;

ma_device_uninit(&g_capture_device); // stops the device and joins the audio thread

g_capture_initialized = false;

ma_pcm_rb_uninit(&g_capture_rb);

}

// Drain up to out's length of interleaved f32 from the ring into out; returns FRAMES read

// (0 if not recording or nothing buffered). out is caller-sized; frames = out.size / channels.

int32_t dasAudio_record_read ( TArray<float> & out, Context *, LineInfoArg * ) {

if ( !g_capture_initialized ) return 0;

ma_uint32 channels = (ma_uint32) g_capture_channels;

if ( channels == 0 || out.size == 0 ) return 0;

ma_uint32 wantFrames = (ma_uint32)(out.size / channels);

float * dst = (float *) out.data;

ma_uint32 framesRead = 0;

while ( framesRead < wantFrames ) {

ma_uint32 chunk = wantFrames - framesRead;

void * pRead = nullptr;

if ( ma_pcm_rb_acquire_read(&g_capture_rb, &chunk, &pRead) != MA_SUCCESS ) break;

if ( chunk == 0 ) { ma_pcm_rb_commit_read(&g_capture_rb, 0); break; }

memcpy(dst + (size_t)framesRead * channels, pRead, (size_t)chunk * channels * sizeof(float));

ma_pcm_rb_commit_read(&g_capture_rb, chunk);

framesRead += chunk;

}

return (int32_t) framesRead;

}

int32_t dasAudio_record_available ( void ) {

if ( !g_capture_initialized ) return 0;

return (int32_t) ma_pcm_rb_available_read(&g_capture_rb);

}

bool dasAudio_is_recording ( void ) {

return g_capture_initialized;

}

int64_t dasAudio_record_overflow_frames ( void ) {

return (int64_t) g_capture_overflow_frames.load(std::memory_order_relaxed);

}

int32_t dasAudio_record_device_count ( Context *, LineInfoArg * ) {

g_capture_device_cache.clear(); // a failed enumeration must not leave stale devices readable by name/is_default

if ( !ensure_capture_context() ) return 0;

ma_device_info * pCapture = nullptr;

ma_uint32 captureCount = 0;

if ( ma_context_get_devices(&g_capture_context, nullptr, nullptr, &pCapture, &captureCount) != MA_SUCCESS ) return 0;

g_capture_device_cache.assign(pCapture, pCapture + captureCount); // value-copy; ids survive next enumeration

return (int32_t) captureCount;

}

char * dasAudio_record_device_name ( int32_t index, Context * context, LineInfoArg * at ) {

if ( index < 0 || index >= (int32_t) g_capture_device_cache.size() ) return nullptr;

const char * name = g_capture_device_cache[index].name;

return context->allocateString(name, (uint64_t) strlen(name), at);

}

bool dasAudio_record_device_is_default ( int32_t index ) {

if ( index < 0 || index >= (int32_t) g_capture_device_cache.size() ) return false;

return g_capture_device_cache[index].isDefault != 0;

}

MA_API ma_result dasAudio_ma_resampler_init(const ma_resampler_config* pConfig, ma_resampler* pResampler) {

return ma_resampler_init(pConfig, nullptr, pResampler);

}

MA_API void dasAudio_ma_resampler_uninit(ma_resampler* pResampler) {

return ma_resampler_uninit(pResampler,nullptr);

}

MA_API ma_result dasAudio_ma_channel_converter_init(const ma_channel_converter_config* pConfig, ma_channel_converter* pConverter) {

return ma_channel_converter_init(pConfig, nullptr, pConverter);

}

MA_API void dasAudio_ma_channel_converter_uninit(ma_channel_converter* pConverter) {

return ma_channel_converter_uninit(pConverter, nullptr);

}

MA_API ma_uint64 dasAudio_ma_resampler_get_required_input_frame_count(const ma_resampler* pResampler, ma_uint64 outputFrameCount) {

ma_uint64 inputFrameCount = 0ul;

ma_resampler_get_required_input_frame_count(pResampler, outputFrameCount, &inputFrameCount);

return inputFrameCount;

}

MA_API ma_uint64 dasAudio_ma_resampler_get_expected_output_frame_count(const ma_resampler* pResampler, ma_uint64 inputFrameCount) {

ma_uint64 outputFrameCount = 0ul;

ma_resampler_get_expected_output_frame_count(pResampler, inputFrameCount, &outputFrameCount);

return outputFrameCount;

}

MA_API ma_result dasAudio_ma_decoder_init_memory(const void* pData, ma_uint64 dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder) {

return ma_decoder_init_memory(pData, (size_t)dataSize, pConfig, pDecoder);

}

MA_API ma_uint64 dasAudio_ma_decoder_get_length_in_pcm_frames(ma_decoder* pDecoder) {

ma_uint64 frameCount = 0;

ma_decoder_get_length_in_pcm_frames(pDecoder, &frameCount);

return frameCount;

}

MA_API ma_uint64 dasAudio_ma_decoder_read_pcm_frames(ma_decoder* pDecoder, void* pFramesOut, ma_uint64 frameCount) {

ma_uint64 framesRead = 0;

ma_decoder_read_pcm_frames(pDecoder, pFramesOut, frameCount, &framesRead);

return framesRead;

}

struct MAResamplerConfigAnnotation : ManagedStructureAnnotation<ma_resampler_config> {

MAResamplerConfigAnnotation ( ModuleLibrary & mlib )

: ManagedStructureAnnotation("ma_resampler_config", mlib, "ma_resampler_config") {

addField<DAS_BIND_MANAGED_FIELD(format)>("format","format");

addField<DAS_BIND_MANAGED_FIELD(channels)>("channels","channels");

addField<DAS_BIND_MANAGED_FIELD(sampleRateIn)>("sampleRateIn","sampleRateIn");

addField<DAS_BIND_MANAGED_FIELD(sampleRateOut)>("sampleRateOut","sampleRateOut");

addField<DAS_BIND_MANAGED_FIELD(lpfOrder)>("lpfOrder","lpfOrder");

addField<DAS_BIND_MANAGED_FIELD(lpfNyquistFactor)>("lpfNyquistFactor","lpfNyquistFactor");

addField<DAS_BIND_MANAGED_FIELD(quality)>("quality","quality");

}

};

struct MAResamplerAnnotation : ManagedStructureAnnotation<ma_resampler> {

MAResamplerAnnotation ( ModuleLibrary & mlib )

: ManagedStructureAnnotation("ma_resampler", mlib, "ma_resampler") {

addField<DAS_BIND_MANAGED_FIELD(format)>("format","format");

addField<DAS_BIND_MANAGED_FIELD(channels)>("channels","channels");

addField<DAS_BIND_MANAGED_FIELD(sampleRateIn)>("sampleRateIn","sampleRateIn");

addField<DAS_BIND_MANAGED_FIELD(sampleRateOut)>("sampleRateOut","sampleRateOut");

// addField<DAS_BIND_MANAGED_FIELD(state)>("state","state");

}

};

struct MAChannelConvertorConfigAnnotation : ManagedStructureAnnotation<ma_channel_converter_config> {

MAChannelConvertorConfigAnnotation ( ModuleLibrary & mlib )

: ManagedStructureAnnotation("ma_channel_converter_config", mlib, "ma_channel_converter_config") {

addField<DAS_BIND_MANAGED_FIELD(format)>("format","format");

addField<DAS_BIND_MANAGED_FIELD(channelsIn)>("channelsIn","channelsIn");

addField<DAS_BIND_MANAGED_FIELD(channelsOut)>("channelsOut","channelsOut");

addField<DAS_BIND_MANAGED_FIELD(pChannelMapIn)>("channelMapIn","channelMapIn");

addField<DAS_BIND_MANAGED_FIELD(pChannelMapOut)>("channelMapOut","channelMapOut");

addField<DAS_BIND_MANAGED_FIELD(mixingMode)>("mixingMode","mixingMode");

addField<DAS_BIND_MANAGED_FIELD(ppWeights)>("weights","weights");

}

};

struct MAChannelConverterAnnotation : ManagedStructureAnnotation<ma_channel_converter> {

MAChannelConverterAnnotation ( ModuleLibrary & mlib )

: ManagedStructureAnnotation("ma_channel_converter", mlib, "ma_channel_converter") {

}

};

struct MAVolumeMixerAnnotation : ManagedStructureAnnotation<ma_volume_mixer> {

MAVolumeMixerAnnotation ( ModuleLibrary & mlib )

: ManagedStructureAnnotation("ma_volume_mixer", mlib, "ma_volume_mixer") {

addField<DAS_BIND_MANAGED_FIELD(volume)>("volume","volume");

addField<DAS_BIND_MANAGED_FIELD(dvolume)>("dvolume","dvolume");

addField<DAS_BIND_MANAGED_FIELD(tvolume)>("tvolume","tvolume");

addField<DAS_BIND_MANAGED_FIELD(pan)>("pan","pan");

addField<DAS_BIND_MANAGED_FIELD(nChannels)>("channels","nChannels");

}

};

struct MADecoderConfigAnnotation : ManagedStructureAnnotation<ma_decoder_config> {

MADecoderConfigAnnotation ( ModuleLibrary & mlib )

: ManagedStructureAnnotation("ma_decoder_config", mlib, "ma_decoder_config") {

addField<DAS_BIND_MANAGED_FIELD(format)>("format","format");

addField<DAS_BIND_MANAGED_FIELD(channels)>("channels","channels");

addField<DAS_BIND_MANAGED_FIELD(pChannelMap)>("channelMap","channelMap");

addField<DAS_BIND_MANAGED_FIELD(channelMixMode)>("channelMixMode","channelMixMode");

addField<DAS_BIND_MANAGED_FIELD(ditherMode)>("ditherMode","ditherMode");

}

};

struct MADecoderAnnotation : ManagedStructureAnnotation<ma_decoder> {

MADecoderAnnotation ( ModuleLibrary & mlib )

: ManagedStructureAnnotation("ma_decoder", mlib, "ma_decoder") {

addField<DAS_BIND_MANAGED_FIELD(outputFormat)>("outputFormat","outputFormat");

addField<DAS_BIND_MANAGED_FIELD(outputChannels)>("outputChannels","outputChannels");

addField<DAS_BIND_MANAGED_FIELD(outputSampleRate)>("outputSampleRate","outputSampleRate");

}

};

struct MALimiterAnnotation : ManagedStructureAnnotation<ma_limiter> {

MALimiterAnnotation ( ModuleLibrary & mlib )

: ManagedStructureAnnotation("ma_limiter", mlib, "ma_limiter") {

addField<DAS_BIND_MANAGED_FIELD(gain)>("gain","gain");

addField<DAS_BIND_MANAGED_FIELD(nChannels)>("nChannels","nChannels");

addField<DAS_BIND_MANAGED_FIELD(attack_samples)>("attack_samples","attack_samples");

addField<DAS_BIND_MANAGED_FIELD(threshold)>("threshold","threshold");

addField<DAS_BIND_MANAGED_FIELD(attack_coeff)>("attack_coeff","attack_coeff");

addField<DAS_BIND_MANAGED_FIELD(release_coeff)>("release_coeff","release_coeff");

addField<DAS_BIND_MANAGED_FIELD(linear_limiter)>("linear_limiter","linear_limiter");

}

};

// SF2 voice

struct MASF2EnvelopeAnnotation : ManagedStructureAnnotation<ma_sf2_envelope> {

MASF2EnvelopeAnnotation ( ModuleLibrary & mlib )

: ManagedStructureAnnotation("ma_sf2_envelope", mlib, "ma_sf2_envelope") {

addField<DAS_BIND_MANAGED_FIELD(stage)>("stage","stage");

addField<DAS_BIND_MANAGED_FIELD(level)>("level","level");

addField<DAS_BIND_MANAGED_FIELD(release_level)>("release_level","release_level");

addField<DAS_BIND_MANAGED_FIELD(slope)>("slope","slope");

addField<DAS_BIND_MANAGED_FIELD(is_exponential)>("is_exponential","is_exponential");

addField<DAS_BIND_MANAGED_FIELD(is_amp_env)>("is_amp_env","is_amp_env");

addField<DAS_BIND_MANAGED_FIELD(midi_velocity)>("midi_velocity","midi_velocity");

addField<DAS_BIND_MANAGED_FIELD(samples_until_next)>("samples_until_next","samples_until_next");

addField<DAS_BIND_MANAGED_FIELD(delay_sec)>("delay_sec","delay_sec");

addField<DAS_BIND_MANAGED_FIELD(attack_sec)>("attack_sec","attack_sec");

addField<DAS_BIND_MANAGED_FIELD(hold_sec)>("hold_sec","hold_sec");

addField<DAS_BIND_MANAGED_FIELD(decay_sec)>("decay_sec","decay_sec");

addField<DAS_BIND_MANAGED_FIELD(sustain_level)>("sustain_level","sustain_level");

addField<DAS_BIND_MANAGED_FIELD(release_sec)>("release_sec","release_sec");

}

};

struct MASF2BiquadAnnotation : ManagedStructureAnnotation<ma_sf2_biquad> {

MASF2BiquadAnnotation ( ModuleLibrary & mlib )

: ManagedStructureAnnotation("ma_sf2_biquad", mlib, "ma_sf2_biquad") {

addField<DAS_BIND_MANAGED_FIELD(q_inv)>("q_inv","q_inv");

addField<DAS_BIND_MANAGED_FIELD(a0)>("a0","a0");

addField<DAS_BIND_MANAGED_FIELD(a1)>("a1","a1");

addField<DAS_BIND_MANAGED_FIELD(b1)>("b1","b1");

addField<DAS_BIND_MANAGED_FIELD(b2)>("b2","b2");

addField<DAS_BIND_MANAGED_FIELD(z1)>("z1","z1");

addField<DAS_BIND_MANAGED_FIELD(z2)>("z2","z2");

addField<DAS_BIND_MANAGED_FIELD(active)>("active","active");

}

};

struct MASF2VoiceAnnotation : ManagedStructureAnnotation<ma_sf2_voice> {

MASF2VoiceAnnotation ( ModuleLibrary & mlib )

: ManagedStructureAnnotation("ma_sf2_voice", mlib, "ma_sf2_voice") {

addField<DAS_BIND_MANAGED_FIELD(sample_start)>("sample_start","sample_start");

addField<DAS_BIND_MANAGED_FIELD(sample_end)>("sample_end","sample_end");

addField<DAS_BIND_MANAGED_FIELD(loop_start)>("loop_start","loop_start");

addField<DAS_BIND_MANAGED_FIELD(loop_end)>("loop_end","loop_end");

addField<DAS_BIND_MANAGED_FIELD(loop_mode)>("loop_mode","loop_mode");

addField<DAS_BIND_MANAGED_FIELD(sample_start_r)>("sample_start_r","sample_start_r");

addField<DAS_BIND_MANAGED_FIELD(sample_end_r)>("sample_end_r","sample_end_r");

addField<DAS_BIND_MANAGED_FIELD(loop_start_r)>("loop_start_r","loop_start_r");

addField<DAS_BIND_MANAGED_FIELD(loop_end_r)>("loop_end_r","loop_end_r");

addField<DAS_BIND_MANAGED_FIELD(stereo)>("stereo","stereo");

addField<DAS_BIND_MANAGED_FIELD(position)>("position","position");

addField<DAS_BIND_MANAGED_FIELD(phase_inc)>("phase_inc","phase_inc");

addField<DAS_BIND_MANAGED_FIELD(vol_env)>("vol_env","vol_env");

addField<DAS_BIND_MANAGED_FIELD(mod_env)>("mod_env","mod_env");

addField<DAS_BIND_MANAGED_FIELD(filter)>("filter","filter");

addField<DAS_BIND_MANAGED_FIELD(filter_r)>("filter_r","filter_r");

addField<DAS_BIND_MANAGED_FIELD(initial_filter_fc)>("initial_filter_fc","initial_filter_fc");

addField<DAS_BIND_MANAGED_FIELD(initial_filter_q)>("initial_filter_q","initial_filter_q");

addField<DAS_BIND_MANAGED_FIELD(mod_env_to_pitch)>("mod_env_to_pitch","mod_env_to_pitch");

addField<DAS_BIND_MANAGED_FIELD(mod_lfo_to_pitch)>("mod_lfo_to_pitch","mod_lfo_to_pitch");

addField<DAS_BIND_MANAGED_FIELD(vib_lfo_to_pitch)>("vib_lfo_to_pitch","vib_lfo_to_pitch");

addField<DAS_BIND_MANAGED_FIELD(mod_env_to_filter_fc)>("mod_env_to_filter_fc","mod_env_to_filter_fc");

addField<DAS_BIND_MANAGED_FIELD(mod_lfo_to_filter_fc)>("mod_lfo_to_filter_fc","mod_lfo_to_filter_fc");

addField<DAS_BIND_MANAGED_FIELD(mod_lfo_to_volume)>("mod_lfo_to_volume","mod_lfo_to_volume");

addField<DAS_BIND_MANAGED_FIELD(mod_lfo_phase)>("mod_lfo_phase","mod_lfo_phase");

addField<DAS_BIND_MANAGED_FIELD(mod_lfo_freq)>("mod_lfo_freq","mod_lfo_freq");

addField<DAS_BIND_MANAGED_FIELD(mod_lfo_delay)>("mod_lfo_delay","mod_lfo_delay");

addField<DAS_BIND_MANAGED_FIELD(mod_lfo_elapsed)>("mod_lfo_elapsed","mod_lfo_elapsed");

addField<DAS_BIND_MANAGED_FIELD(vib_lfo_phase)>("vib_lfo_phase","vib_lfo_phase");

addField<DAS_BIND_MANAGED_FIELD(vib_lfo_freq)>("vib_lfo_freq","vib_lfo_freq");

addField<DAS_BIND_MANAGED_FIELD(vib_lfo_delay)>("vib_lfo_delay","vib_lfo_delay");

addField<DAS_BIND_MANAGED_FIELD(vib_lfo_elapsed)>("vib_lfo_elapsed","vib_lfo_elapsed");

addField<DAS_BIND_MANAGED_FIELD(pitch_bend_cents)>("pitch_bend_cents","pitch_bend_cents");

addField<DAS_BIND_MANAGED_FIELD(attenuation)>("attenuation","attenuation");

addField<DAS_BIND_MANAGED_FIELD(pan)>("pan","pan");

addField<DAS_BIND_MANAGED_FIELD(released)>("released","released");

addField<DAS_BIND_MANAGED_FIELD(finished)>("finished","finished");

addField<DAS_BIND_MANAGED_FIELD(sample_rate)>("sample_rate","sample_rate");

}

};

struct I3DL2ReverbPropertiesAnnotation : ManagedStructureAnnotation<I3DL2ReverbProperties> {

I3DL2ReverbPropertiesAnnotation ( ModuleLibrary & mlib )

: ManagedStructureAnnotation("I3DL2ReverbProperties", mlib, "I3DL2ReverbProperties") {

addField<DAS_BIND_MANAGED_FIELD(lRoom)>("lRoom","lRoom");

addField<DAS_BIND_MANAGED_FIELD(lRoomHF)>("lRoomHF","lRoomHF");

addField<DAS_BIND_MANAGED_FIELD(flDecayTime)>("flDecayTime","flDecayTime");

addField<DAS_BIND_MANAGED_FIELD(flDecayHFRatio)>("flDecayHFRatio","flDecayHFRatio");

addField<DAS_BIND_MANAGED_FIELD(lReflections)>("lReflections","lReflections");

addField<DAS_BIND_MANAGED_FIELD(flReflectionsDelay)>("flReflectionsDelay","flReflectionsDelay");

addField<DAS_BIND_MANAGED_FIELD(lReverb)>("lReverb","lReverb");

addField<DAS_BIND_MANAGED_FIELD(flReverbDelay)>("flReverbDelay","flReverbDelay");

addField<DAS_BIND_MANAGED_FIELD(flDiffusion)>("flDiffusion","flDiffusion");

addField<DAS_BIND_MANAGED_FIELD(flDensity)>("flDensity","flDensity");

}

};

struct I3DL2ReverbAnnotation : ManagedStructureAnnotation<I3DL2Reverb,true,true> {

I3DL2ReverbAnnotation ( ModuleLibrary & mlib )

: ManagedStructureAnnotation("I3DL2Reverb", mlib, "I3DL2Reverb") {

}

};

// ─── Chorus ───

struct MaChorusConfigAnnotation : ManagedStructureAnnotation<ma_chorus_config> {

MaChorusConfigAnnotation ( ModuleLibrary & mlib )

: ManagedStructureAnnotation("ma_chorus_config", mlib, "ma_chorus_config") {

addField<DAS_BIND_MANAGED_FIELD(rate)>("rate","rate");

addField<DAS_BIND_MANAGED_FIELD(depth)>("depth","depth");

addField<DAS_BIND_MANAGED_FIELD(feedback)>("feedback","feedback");

addField<DAS_BIND_MANAGED_FIELD(delay_ms)>("delay_ms","delay_ms");

addField<DAS_BIND_MANAGED_FIELD(wet)>("wet","wet");

}

};

struct MaChorusAnnotation : ManagedStructureAnnotation<ma_chorus,true,true> {

MaChorusAnnotation ( ModuleLibrary & mlib )

: ManagedStructureAnnotation("ma_chorus", mlib, "ma_chorus") {

}

};

void dasAudio_chorusInit ( ma_chorus * chorus, float sample_rate, Context * context, LineInfoArg * at ) {

if ( !chorus ) context->throw_error_at(at,"chorus is null");

ma_chorus_init(chorus, sample_rate);

}

void dasAudio_chorusProcess ( ma_chorus * chorus, float * input, float * output, int nSamples, Context * context, LineInfoArg * at ) {

if ( !chorus ) context->throw_error_at(at,"chorus is null");

ma_chorus_process(chorus, input, output, nSamples);

}

void dasAudio_chorusSetConfig ( ma_chorus * chorus, const ma_chorus_config & config, Context * context, LineInfoArg * at ) {

if ( !chorus ) context->throw_error_at(at,"chorus is null");

ma_chorus_set_config(chorus, &config);

}

ma_chorus_config dasAudio_chorusConfigDefault ( ) {

return ma_chorus_config_default();

}

// ─── Effects (bitcrush, waveshaper, djfilter, bandpass) ───

struct MaBitcrushAnnotation : ManagedStructureAnnotation<ma_bitcrush,true,true> {

MaBitcrushAnnotation ( ModuleLibrary & mlib )

: ManagedStructureAnnotation("ma_bitcrush", mlib, "ma_bitcrush") {

addField<DAS_BIND_MANAGED_FIELD(bits)>("bits","bits");

addField<DAS_BIND_MANAGED_FIELD(coarse)>("coarse","coarse");

}

};

struct MaWaveshaperAnnotation : ManagedStructureAnnotation<ma_waveshaper,true,true> {

MaWaveshaperAnnotation ( ModuleLibrary & mlib )

: ManagedStructureAnnotation("ma_waveshaper", mlib, "ma_waveshaper") {

addField<DAS_BIND_MANAGED_FIELD(drive)>("drive","drive");

}

};

struct MaDjfilterAnnotation : ManagedStructureAnnotation<ma_djfilter,true,true> {

MaDjfilterAnnotation ( ModuleLibrary & mlib )

: ManagedStructureAnnotation("ma_djfilter", mlib, "ma_djfilter") {

addField<DAS_BIND_MANAGED_FIELD(position)>("position","position");

addField<DAS_BIND_MANAGED_FIELD(sample_rate)>("sample_rate","sample_rate");

}

};

struct MaBandpassAnnotation : ManagedStructureAnnotation<ma_bandpass,true,true> {

MaBandpassAnnotation ( ModuleLibrary & mlib )

: ManagedStructureAnnotation("ma_bandpass", mlib, "ma_bandpass") {

addField<DAS_BIND_MANAGED_FIELD(freq)>("freq","freq");

addField<DAS_BIND_MANAGED_FIELD(q)>("q","q");

addField<DAS_BIND_MANAGED_FIELD(sample_rate)>("sample_rate","sample_rate");

}

};

void dasAudio_bitcrushInit ( ma_bitcrush * bc, Context * context, LineInfoArg * at ) {

if ( !bc ) context->throw_error_at(at,"bitcrush is null");

ma_bitcrush_init(bc);

}

void dasAudio_bitcrushProcess ( ma_bitcrush * bc, float * buf, int nFrames, Context * context, LineInfoArg * at ) {

if ( !bc ) context->throw_error_at(at,"bitcrush is null");

ma_bitcrush_process(bc, buf, nFrames);

}

void dasAudio_waveshaperInit ( ma_waveshaper * ws, Context * context, LineInfoArg * at ) {

if ( !ws ) context->throw_error_at(at,"waveshaper is null");

ma_waveshaper_init(ws);

}

void dasAudio_waveshaperProcess ( ma_waveshaper * ws, float * buf, int nFrames, Context * context, LineInfoArg * at ) {

if ( !ws ) context->throw_error_at(at,"waveshaper is null");

ma_waveshaper_process(ws, buf, nFrames);

}

void dasAudio_djfilterInit ( ma_djfilter * dj, float sample_rate, Context * context, LineInfoArg * at ) {

if ( !dj ) context->throw_error_at(at,"djfilter is null");

ma_djfilter_init(dj, sample_rate);

}

void dasAudio_djfilterProcess ( ma_djfilter * dj, float * buf, int nFrames, Context * context, LineInfoArg * at ) {

if ( !dj ) context->throw_error_at(at,"djfilter is null");

ma_djfilter_process(dj, buf, nFrames);

}

void dasAudio_bandpassInit ( ma_bandpass * bp, float sample_rate, Context * context, LineInfoArg * at ) {

if ( !bp ) context->throw_error_at(at,"bandpass is null");

ma_bandpass_init(bp, sample_rate);

}

void dasAudio_bandpassSetup ( ma_bandpass * bp, float freq, float q, Context * context, LineInfoArg * at ) {

if ( !bp ) context->throw_error_at(at,"bandpass is null");

ma_bandpass_setup(bp, freq, q);

}

void dasAudio_bandpassProcess ( ma_bandpass * bp, float * buf, int nFrames, Context * context, LineInfoArg * at ) {

if ( !bp ) context->throw_error_at(at,"bandpass is null");

ma_bandpass_process(bp, buf, nFrames);

}

// ─── Phaser / Tremolo / Compressor ───

struct MaPhaserAnnotation : ManagedStructureAnnotation<ma_phaser,true,true> {

MaPhaserAnnotation ( ModuleLibrary & mlib )

: ManagedStructureAnnotation("ma_phaser", mlib, "ma_phaser") {

addField<DAS_BIND_MANAGED_FIELD(rate)>("rate","rate");

addField<DAS_BIND_MANAGED_FIELD(depth)>("depth","depth");

addField<DAS_BIND_MANAGED_FIELD(center)>("center","center");

addField<DAS_BIND_MANAGED_FIELD(sweep)>("sweep","sweep");

addField<DAS_BIND_MANAGED_FIELD(sample_rate)>("sample_rate","sample_rate");

}

};

struct MaTremoloAnnotation : ManagedStructureAnnotation<ma_tremolo,true,true> {

MaTremoloAnnotation ( ModuleLibrary & mlib )

: ManagedStructureAnnotation("ma_tremolo", mlib, "ma_tremolo") {

addField<DAS_BIND_MANAGED_FIELD(rate)>("rate","rate");

addField<DAS_BIND_MANAGED_FIELD(depth)>("depth","depth");

addField<DAS_BIND_MANAGED_FIELD(sample_rate)>("sample_rate","sample_rate");

}

};

struct MaCompressorAnnotation : ManagedStructureAnnotation<ma_compressor,true,true> {

MaCompressorAnnotation ( ModuleLibrary & mlib )

: ManagedStructureAnnotation("ma_compressor", mlib, "ma_compressor") {

addField<DAS_BIND_MANAGED_FIELD(threshold_db)>("threshold_db","threshold_db");

addField<DAS_BIND_MANAGED_FIELD(ratio)>("ratio","ratio");

addField<DAS_BIND_MANAGED_FIELD(knee_db)>("knee_db","knee_db");

addField<DAS_BIND_MANAGED_FIELD(attack)>("attack","attack");

addField<DAS_BIND_MANAGED_FIELD(release)>("release","release");

addField<DAS_BIND_MANAGED_FIELD(sample_rate)>("sample_rate","sample_rate");

}

};

void dasAudio_phaserInit ( ma_phaser * ph, float sample_rate, Context * context, LineInfoArg * at ) {

if ( !ph ) context->throw_error_at(at,"phaser is null");

ma_phaser_init(ph, sample_rate);

}

void dasAudio_phaserProcess ( ma_phaser * ph, float * buf, int nFrames, Context * context, LineInfoArg * at ) {

if ( !ph ) context->throw_error_at(at,"phaser is null");

ma_phaser_process(ph, buf, nFrames);

}

void dasAudio_tremoloInit ( ma_tremolo * tr, float sample_rate, Context * context, LineInfoArg * at ) {

if ( !tr ) context->throw_error_at(at,"tremolo is null");

ma_tremolo_init(tr, sample_rate);

}

void dasAudio_tremoloProcess ( ma_tremolo * tr, float * buf, int nFrames, Context * context, LineInfoArg * at ) {

if ( !tr ) context->throw_error_at(at,"tremolo is null");

ma_tremolo_process(tr, buf, nFrames);

}

void dasAudio_compressorInit ( ma_compressor * c, float sample_rate, Context * context, LineInfoArg * at ) {

if ( !c ) context->throw_error_at(at,"compressor is null");

ma_compressor_init(c, sample_rate);

}

void dasAudio_compressorProcess ( ma_compressor * c, float * buf, int nFrames, Context * context, LineInfoArg * at ) {

if ( !c ) context->throw_error_at(at,"compressor is null");

ma_compressor_process(c, buf, nFrames);

}

struct MAHrtfAnnotation : ManagedStructureAnnotation<ma_hrtf> {

MAHrtfAnnotation ( ModuleLibrary & mlib )

: ManagedStructureAnnotation("ma_hrtf", mlib, "ma_hrtf") {

addField<DAS_BIND_MANAGED_FIELD(taps)>("taps","taps");

addField<DAS_BIND_MANAGED_FIELD(azimuth)>("azimuth","azimuth");

addField<DAS_BIND_MANAGED_FIELD(elevation)>("elevation","elevation");

addField<DAS_BIND_MANAGED_FIELD(sampleRate)>("sampleRate","sampleRate");

addField<DAS_BIND_MANAGED_FIELD(leftfip)>("leftfip","leftfip");

addField<DAS_BIND_MANAGED_FIELD(rightfip)>("rightfip","rightfip");

addField<DAS_BIND_MANAGED_FIELD(fft_size)>("fft_size","fft_size");

}

};

void dasAudio_setSampleRate ( I3DL2Reverb * reverb, float rate, Context * context, LineInfoArg * at ) {

if ( !reverb ) context->throw_error_at(at,"reverb is null");

reverb->SetSampleRate(rate);

}

void dasAudio_setProperties ( I3DL2Reverb * reverb, const I3DL2ReverbProperties & props, Context * context, LineInfoArg * at ) {

if ( !reverb ) context->throw_error_at(at,"reverb is null");

reverb->SetReverbProperties(props);

}

void dasAudio_process ( I3DL2Reverb * reverb, float * buffer, float * outBuffer, int nSamples, Context * context, LineInfoArg * at ) {

if ( !reverb ) context->throw_error_at(at,"reverb is null");

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daScript/modules/dasAudio/src/dasAudio.cpp at master · WhyNot135/daScript · GitHub

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