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84
thirdparty/miniaudio-0.11.24/extras/nodes/ma_reverb_node/ma_reverb_node.c vendored Normal file
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#ifndef miniaudio_reverb_node_c
#define miniaudio_reverb_node_c
#define VERBLIB_IMPLEMENTATION
#include "ma_reverb_node.h"
#include <string.h> /* For memset(). */
MA_API ma_reverb_node_config ma_reverb_node_config_init(ma_uint32 channels, ma_uint32 sampleRate)
{
ma_reverb_node_config config;
memset(&config, 0, sizeof(config));
config.nodeConfig = ma_node_config_init(); /* Input and output channels will be set in ma_reverb_node_init(). */
config.channels = channels;
config.sampleRate = sampleRate;
config.roomSize = verblib_initialroom;
config.damping = verblib_initialdamp;
config.width = verblib_initialwidth;
config.wetVolume = verblib_initialwet;
config.dryVolume = verblib_initialdry;
config.mode = verblib_initialmode;
return config;
}
static void ma_reverb_node_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
{
ma_reverb_node* pReverbNode = (ma_reverb_node*)pNode;
(void)pFrameCountIn;
verblib_process(&pReverbNode->reverb, ppFramesIn[0], ppFramesOut[0], *pFrameCountOut);
}
static ma_node_vtable g_ma_reverb_node_vtable =
{
ma_reverb_node_process_pcm_frames,
NULL,
1, /* 1 input bus. */
1, /* 1 output bus. */
MA_NODE_FLAG_CONTINUOUS_PROCESSING /* Reverb requires continuous processing to ensure the tail get's processed. */
};
MA_API ma_result ma_reverb_node_init(ma_node_graph* pNodeGraph, const ma_reverb_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_reverb_node* pReverbNode)
{
ma_result result;
ma_node_config baseConfig;
if (pReverbNode == NULL) {
return MA_INVALID_ARGS;
}
memset(pReverbNode, 0, sizeof(*pReverbNode));
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
if (verblib_initialize(&pReverbNode->reverb, (unsigned long)pConfig->sampleRate, (unsigned int)pConfig->channels) == 0) {
return MA_INVALID_ARGS;
}
baseConfig = pConfig->nodeConfig;
baseConfig.vtable = &g_ma_reverb_node_vtable;
baseConfig.pInputChannels = &pConfig->channels;
baseConfig.pOutputChannels = &pConfig->channels;
result = ma_node_init(pNodeGraph, &baseConfig, pAllocationCallbacks, &pReverbNode->baseNode);
if (result != MA_SUCCESS) {
return result;
}
return MA_SUCCESS;
}
MA_API void ma_reverb_node_uninit(ma_reverb_node* pReverbNode, const ma_allocation_callbacks* pAllocationCallbacks)
{
/* The base node is always uninitialized first. */
ma_node_uninit(pReverbNode, pAllocationCallbacks);
}
#endif /* miniaudio_reverb_node_c */

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thirdparty/miniaudio-0.11.24/extras/nodes/ma_reverb_node/ma_reverb_node.h vendored Normal file
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/* Include ma_reverb_node.h after miniaudio.h */
#ifndef miniaudio_reverb_node_h
#define miniaudio_reverb_node_h
#include "../../../miniaudio.h"
#include "verblib.h"
#ifdef __cplusplus
extern "C" {
#endif
/*
The reverb node has one input and one output.
*/
typedef struct
{
ma_node_config nodeConfig;
ma_uint32 channels; /* The number of channels of the source, which will be the same as the output. Must be 1 or 2. */
ma_uint32 sampleRate;
float roomSize;
float damping;
float width;
float wetVolume;
float dryVolume;
float mode;
} ma_reverb_node_config;
MA_API ma_reverb_node_config ma_reverb_node_config_init(ma_uint32 channels, ma_uint32 sampleRate);
typedef struct
{
ma_node_base baseNode;
verblib reverb;
} ma_reverb_node;
MA_API ma_result ma_reverb_node_init(ma_node_graph* pNodeGraph, const ma_reverb_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_reverb_node* pReverbNode);
MA_API void ma_reverb_node_uninit(ma_reverb_node* pReverbNode, const ma_allocation_callbacks* pAllocationCallbacks);
#ifdef __cplusplus
}
#endif
#endif /* miniaudio_reverb_node_h */

122
thirdparty/miniaudio-0.11.24/extras/nodes/ma_reverb_node/ma_reverb_node_example.c vendored Normal file
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#include "../../../miniaudio.c"
#include "ma_reverb_node.c"
#include <stdio.h>
#define DEVICE_FORMAT ma_format_f32 /* Must always be f32 for this example because the node graph system only works with this. */
#define DEVICE_CHANNELS 1 /* For this example, always set to 1. */
#define DEVICE_SAMPLE_RATE 48000 /* Cannot be less than 22050 for this example. */
static ma_audio_buffer_ref g_dataSupply; /* The underlying data source of the source node. */
static ma_data_source_node g_dataSupplyNode; /* The node that will sit at the root level. Will be reading data from g_dataSupply. */
static ma_reverb_node g_reverbNode; /* The reverb node. */
static ma_node_graph g_nodeGraph; /* The main node graph that we'll be feeding data through. */
void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
{
/*
This example assumes the playback and capture sides use the same format and channel count. The
format must be f32.
*/
if (pDevice->capture.format != DEVICE_FORMAT || pDevice->playback.format != DEVICE_FORMAT || pDevice->capture.channels != pDevice->playback.channels) {
return;
}
/*
The node graph system is a pulling style of API. At the lowest level of the chain will be a
node acting as a data source for the purpose of delivering the initial audio data. In our case,
the data source is our `pInput` buffer. We need to update the underlying data source so that it
read data from `pInput`.
*/
ma_audio_buffer_ref_set_data(&g_dataSupply, pInput, frameCount);
/* With the source buffer configured we can now read directly from the node graph. */
ma_node_graph_read_pcm_frames(&g_nodeGraph, pOutput, frameCount, NULL);
}
int main(int argc, char** argv)
{
ma_result result;
ma_device_config deviceConfig;
ma_device device;
ma_node_graph_config nodeGraphConfig;
ma_reverb_node_config reverbNodeConfig;
ma_data_source_node_config dataSupplyNodeConfig;
deviceConfig = ma_device_config_init(ma_device_type_duplex);
deviceConfig.capture.pDeviceID = NULL;
deviceConfig.capture.format = DEVICE_FORMAT;
deviceConfig.capture.channels = DEVICE_CHANNELS;
deviceConfig.capture.shareMode = ma_share_mode_shared;
deviceConfig.playback.pDeviceID = NULL;
deviceConfig.playback.format = DEVICE_FORMAT;
deviceConfig.playback.channels = DEVICE_CHANNELS;
deviceConfig.sampleRate = DEVICE_SAMPLE_RATE;
deviceConfig.dataCallback = data_callback;
result = ma_device_init(NULL, &deviceConfig, &device);
if (result != MA_SUCCESS) {
return result;
}
/* Node graph. */
nodeGraphConfig = ma_node_graph_config_init(device.capture.channels);
result = ma_node_graph_init(&nodeGraphConfig, NULL, &g_nodeGraph);
if (result != MA_SUCCESS) {
printf("Failed to initialize node graph.");
goto done0;
}
/* Reverb. Attached straight to the endpoint. */
reverbNodeConfig = ma_reverb_node_config_init(device.capture.channels, device.sampleRate);
result = ma_reverb_node_init(&g_nodeGraph, &reverbNodeConfig, NULL, &g_reverbNode);
if (result != MA_SUCCESS) {
printf("Failed to initialize reverb node.");
goto done1;
}
ma_node_attach_output_bus(&g_reverbNode, 0, ma_node_graph_get_endpoint(&g_nodeGraph), 0);
/* Data supply. Attached to input bus 0 of the reverb node. */
result = ma_audio_buffer_ref_init(device.capture.format, device.capture.channels, NULL, 0, &g_dataSupply);
if (result != MA_SUCCESS) {
printf("Failed to initialize audio buffer for source.");
goto done2;
}
dataSupplyNodeConfig = ma_data_source_node_config_init(&g_dataSupply);
result = ma_data_source_node_init(&g_nodeGraph, &dataSupplyNodeConfig, NULL, &g_dataSupplyNode);
if (result != MA_SUCCESS) {
printf("Failed to initialize source node.");
goto done2;
}
ma_node_attach_output_bus(&g_dataSupplyNode, 0, &g_reverbNode, 0);
/* Now we just start the device and wait for the user to terminate the program. */
ma_device_start(&device);
printf("Press Enter to quit...\n");
getchar();
/* It's important that we stop the device first or else we'll uninitialize the graph from under the device. */
ma_device_stop(&device);
/*done3:*/ ma_data_source_node_uninit(&g_dataSupplyNode, NULL);
done2: ma_reverb_node_uninit(&g_reverbNode, NULL);
done1: ma_node_graph_uninit(&g_nodeGraph, NULL);
done0: ma_device_uninit(&device);
(void)argc;
(void)argv;
return 0;
}

753
thirdparty/miniaudio-0.11.24/extras/nodes/ma_reverb_node/verblib.h vendored Normal file
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/* Reverb Library
* Verblib version 0.5 - 2022-10-25
*
* Philip Bennefall - philip@blastbay.com
*
* See the end of this file for licensing terms.
* This reverb is based on Freeverb, a public domain reverb written by Jezar at Dreampoint.
*
* IMPORTANT: The reverb currently only works with 1 or 2 channels, at sample rates of 22050 HZ and above.
* These restrictions may be lifted in a future version.
*
* USAGE
*
* This is a single-file library. To use it, do something like the following in one .c file.
* #define VERBLIB_IMPLEMENTATION
* #include "verblib.h"
*
* You can then #include this file in other parts of the program as you would with any other header file.
*/
#ifndef VERBLIB_H
#define VERBLIB_H
#ifdef __cplusplus
extern "C" {
#endif
/* COMPILE-TIME OPTIONS */
/* The maximum sample rate that should be supported, specified as a multiple of 44100. */
#ifndef verblib_max_sample_rate_multiplier
#define verblib_max_sample_rate_multiplier 4
#endif
/* The silence threshold which is used when calculating decay time. */
#ifndef verblib_silence_threshold
#define verblib_silence_threshold 80.0 /* In dB (absolute). */
#endif
/* PUBLIC API */
typedef struct verblib verblib;
/* Initialize a verblib structure.
*
* Call this function to initialize the verblib structure.
* Returns nonzero (true) on success or 0 (false) on failure.
* The function will only fail if one or more of the parameters are invalid.
*/
int verblib_initialize ( verblib* verb, unsigned long sample_rate, unsigned int channels );
/* Run the reverb.
*
* Call this function continuously to generate your output.
* output_buffer may be the same pointer as input_buffer if in place processing is desired.
* frames specifies the number of sample frames that should be processed.
*/
void verblib_process ( verblib* verb, const float* input_buffer, float* output_buffer, unsigned long frames );
/* Set the size of the room, between 0.0 and 1.0. */
void verblib_set_room_size ( verblib* verb, float value );
/* Get the size of the room. */
float verblib_get_room_size ( const verblib* verb );
/* Set the amount of damping, between 0.0 and 1.0. */
void verblib_set_damping ( verblib* verb, float value );
/* Get the amount of damping. */
float verblib_get_damping ( const verblib* verb );
/* Set the stereo width of the reverb, between 0.0 and 1.0. */
void verblib_set_width ( verblib* verb, float value );
/* Get the stereo width of the reverb. */
float verblib_get_width ( const verblib* verb );
/* Set the volume of the wet signal, between 0.0 and 1.0. */
void verblib_set_wet ( verblib* verb, float value );
/* Get the volume of the wet signal. */
float verblib_get_wet ( const verblib* verb );
/* Set the volume of the dry signal, between 0.0 and 1.0. */
void verblib_set_dry ( verblib* verb, float value );
/* Get the volume of the dry signal. */
float verblib_get_dry ( const verblib* verb );
/* Set the stereo width of the input signal sent to the reverb, 0.0 or greater.
* Values less than 1.0 narrow the signal, 1.0 sends the input signal unmodified, values greater than 1.0 widen the signal.
*/
void verblib_set_input_width ( verblib* verb, float value );
/* Get the stereo width of the input signal sent to the reverb. */
float verblib_get_input_width ( const verblib* verb );
/* Set the mode of the reverb, where values below 0.5 mean normal and values above mean frozen. */
void verblib_set_mode ( verblib* verb, float value );
/* Get the mode of the reverb. */
float verblib_get_mode ( const verblib* verb );
/* Get the decay time in sample frames based on the current room size setting. */
/* If freeze mode is active, the decay time is infinite and this function returns 0. */
unsigned long verblib_get_decay_time_in_frames ( const verblib* verb );
/* INTERNAL STRUCTURES */
/* Allpass filter */
typedef struct verblib_allpass verblib_allpass;
struct verblib_allpass
{
float* buffer;
float feedback;
int bufsize;
int bufidx;
};
/* Comb filter */
typedef struct verblib_comb verblib_comb;
struct verblib_comb
{
float* buffer;
float feedback;
float filterstore;
float damp1;
float damp2;
int bufsize;
int bufidx;
};
/* Reverb model tuning values */
#define verblib_numcombs 8
#define verblib_numallpasses 4
#define verblib_muted 0.0f
#define verblib_fixedgain 0.015f
#define verblib_scalewet 3.0f
#define verblib_scaledry 2.0f
#define verblib_scaledamp 0.8f
#define verblib_scaleroom 0.28f
#define verblib_offsetroom 0.7f
#define verblib_initialroom 0.5f
#define verblib_initialdamp 0.25f
#define verblib_initialwet 1.0f/verblib_scalewet
#define verblib_initialdry 0.0f
#define verblib_initialwidth 1.0f
#define verblib_initialinputwidth 0.0f
#define verblib_initialmode 0.0f
#define verblib_freezemode 0.5f
#define verblib_stereospread 23
/*
* These values assume 44.1KHz sample rate, but will be verblib_scaled appropriately.
* The values were obtained by listening tests.
*/
#define verblib_combtuningL1 1116
#define verblib_combtuningR1 (1116+verblib_stereospread)
#define verblib_combtuningL2 1188
#define verblib_combtuningR2 (1188+verblib_stereospread)
#define verblib_combtuningL3 1277
#define verblib_combtuningR3 (1277+verblib_stereospread)
#define verblib_combtuningL4 1356
#define verblib_combtuningR4 (1356+verblib_stereospread)
#define verblib_combtuningL5 1422
#define verblib_combtuningR5 (1422+verblib_stereospread)
#define verblib_combtuningL6 1491
#define verblib_combtuningR6 (1491+verblib_stereospread)
#define verblib_combtuningL7 1557
#define verblib_combtuningR7 (1557+verblib_stereospread)
#define verblib_combtuningL8 1617
#define verblib_combtuningR8 (1617+verblib_stereospread)
#define verblib_allpasstuningL1 556
#define verblib_allpasstuningR1 (556+verblib_stereospread)
#define verblib_allpasstuningL2 441
#define verblib_allpasstuningR2 (441+verblib_stereospread)
#define verblib_allpasstuningL3 341
#define verblib_allpasstuningR3 (341+verblib_stereospread)
#define verblib_allpasstuningL4 225
#define verblib_allpasstuningR4 (225+verblib_stereospread)
/* The main reverb structure. This is the structure that you will create an instance of when using the reverb. */
struct verblib
{
unsigned int channels;
float gain;
float roomsize, roomsize1;
float damp, damp1;
float wet, wet1, wet2;
float dry;
float width;
float input_width;
float mode;
/*
* The following are all declared inline
* to remove the need for dynamic allocation.
*/
/* Comb filters */
verblib_comb combL[verblib_numcombs];
verblib_comb combR[verblib_numcombs];
/* Allpass filters */
verblib_allpass allpassL[verblib_numallpasses];
verblib_allpass allpassR[verblib_numallpasses];
/* Buffers for the combs */
float bufcombL1[verblib_combtuningL1* verblib_max_sample_rate_multiplier];
float bufcombR1[verblib_combtuningR1* verblib_max_sample_rate_multiplier];
float bufcombL2[verblib_combtuningL2* verblib_max_sample_rate_multiplier];
float bufcombR2[verblib_combtuningR2* verblib_max_sample_rate_multiplier];
float bufcombL3[verblib_combtuningL3* verblib_max_sample_rate_multiplier];
float bufcombR3[verblib_combtuningR3* verblib_max_sample_rate_multiplier];
float bufcombL4[verblib_combtuningL4* verblib_max_sample_rate_multiplier];
float bufcombR4[verblib_combtuningR4* verblib_max_sample_rate_multiplier];
float bufcombL5[verblib_combtuningL5* verblib_max_sample_rate_multiplier];
float bufcombR5[verblib_combtuningR5* verblib_max_sample_rate_multiplier];
float bufcombL6[verblib_combtuningL6* verblib_max_sample_rate_multiplier];
float bufcombR6[verblib_combtuningR6* verblib_max_sample_rate_multiplier];
float bufcombL7[verblib_combtuningL7* verblib_max_sample_rate_multiplier];
float bufcombR7[verblib_combtuningR7* verblib_max_sample_rate_multiplier];
float bufcombL8[verblib_combtuningL8* verblib_max_sample_rate_multiplier];
float bufcombR8[verblib_combtuningR8* verblib_max_sample_rate_multiplier];
/* Buffers for the allpasses */
float bufallpassL1[verblib_allpasstuningL1* verblib_max_sample_rate_multiplier];
float bufallpassR1[verblib_allpasstuningR1* verblib_max_sample_rate_multiplier];
float bufallpassL2[verblib_allpasstuningL2* verblib_max_sample_rate_multiplier];
float bufallpassR2[verblib_allpasstuningR2* verblib_max_sample_rate_multiplier];
float bufallpassL3[verblib_allpasstuningL3* verblib_max_sample_rate_multiplier];
float bufallpassR3[verblib_allpasstuningR3* verblib_max_sample_rate_multiplier];
float bufallpassL4[verblib_allpasstuningL4* verblib_max_sample_rate_multiplier];
float bufallpassR4[verblib_allpasstuningR4* verblib_max_sample_rate_multiplier];
};
#ifdef __cplusplus
}
#endif
#endif /* VERBLIB_H */
/* IMPLEMENTATION */
#ifdef VERBLIB_IMPLEMENTATION
#include <stddef.h>
#include <math.h>
#ifdef _MSC_VER
#define VERBLIB_INLINE __forceinline
#elif defined(__GNUC__)
#if defined(__STRICT_ANSI__)
#define VERBLIB_GNUC_INLINE_HINT __inline__
#else
#define VERBLIB_GNUC_INLINE_HINT inline
#endif
#if (__GNUC__ > 3 || (__GNUC__ == 3 && __GNUC_MINOR__ >= 2)) || defined(__clang__)
#define VERBLIB_INLINE VERBLIB_GNUC_INLINE_HINT __attribute__((always_inline))
#else
#define VERBLIB_INLINE VERBLIB_GNUC_INLINE_HINT
#endif
#elif defined(__WATCOMC__)
#define VERBLIB_INLINE __inline
#else
#define VERBLIB_INLINE
#endif
#define verblib_max(x, y) (((x) > (y)) ? (x) : (y))
#define undenormalise(sample) sample+=1.0f; sample-=1.0f;
/* Allpass filter */
static void verblib_allpass_initialize ( verblib_allpass* allpass, float* buf, int size )
{
allpass->buffer = buf;
allpass->bufsize = size;
allpass->bufidx = 0;
}
static VERBLIB_INLINE float verblib_allpass_process ( verblib_allpass* allpass, float input )
{
float output;
float bufout;
bufout = allpass->buffer[allpass->bufidx];
undenormalise ( bufout );
output = -input + bufout;
allpass->buffer[allpass->bufidx] = input + ( bufout * allpass->feedback );
if ( ++allpass->bufidx >= allpass->bufsize )
{
allpass->bufidx = 0;
}
return output;
}
static void verblib_allpass_mute ( verblib_allpass* allpass )
{
int i;
for ( i = 0; i < allpass->bufsize; i++ )
{
allpass->buffer[i] = 0.0f;
}
}
/* Comb filter */
static void verblib_comb_initialize ( verblib_comb* comb, float* buf, int size )
{
comb->buffer = buf;
comb->bufsize = size;
comb->filterstore = 0.0f;
comb->bufidx = 0;
}
static void verblib_comb_mute ( verblib_comb* comb )
{
int i;
for ( i = 0; i < comb->bufsize; i++ )
{
comb->buffer[i] = 0.0f;
}
}
static void verblib_comb_set_damp ( verblib_comb* comb, float val )
{
comb->damp1 = val;
comb->damp2 = 1.0f - val;
}
static VERBLIB_INLINE float verblib_comb_process ( verblib_comb* comb, float input )
{
float output;
output = comb->buffer[comb->bufidx];
undenormalise ( output );
comb->filterstore = ( output * comb->damp2 ) + ( comb->filterstore * comb->damp1 );
undenormalise ( comb->filterstore );
comb->buffer[comb->bufidx] = input + ( comb->filterstore * comb->feedback );
if ( ++comb->bufidx >= comb->bufsize )
{
comb->bufidx = 0;
}
return output;
}
static void verblib_update ( verblib* verb )
{
/* Recalculate internal values after parameter change. */
int i;
verb->wet1 = verb->wet * ( verb->width / 2.0f + 0.5f );
verb->wet2 = verb->wet * ( ( 1.0f - verb->width ) / 2.0f );
if ( verb->mode >= verblib_freezemode )
{
verb->roomsize1 = 1.0f;
verb->damp1 = 0.0f;
verb->gain = verblib_muted;
}
else
{
verb->roomsize1 = verb->roomsize;
verb->damp1 = verb->damp;
verb->gain = verblib_fixedgain;
}
for ( i = 0; i < verblib_numcombs; i++ )
{
verb->combL[i].feedback = verb->roomsize1;
verb->combR[i].feedback = verb->roomsize1;
verblib_comb_set_damp ( &verb->combL[i], verb->damp1 );
verblib_comb_set_damp ( &verb->combR[i], verb->damp1 );
}
}
static void verblib_mute ( verblib* verb )
{
int i;
if ( verblib_get_mode ( verb ) >= verblib_freezemode )
{
return;
}
for ( i = 0; i < verblib_numcombs; i++ )
{
verblib_comb_mute ( &verb->combL[i] );
verblib_comb_mute ( &verb->combR[i] );
}
for ( i = 0; i < verblib_numallpasses; i++ )
{
verblib_allpass_mute ( &verb->allpassL[i] );
verblib_allpass_mute ( &verb->allpassR[i] );
}
}
static int verblib_get_verblib_scaled_buffer_size ( unsigned long sample_rate, unsigned long value )
{
long double result = ( long double ) sample_rate;
result /= 44100.0;
result = ( ( long double ) value ) * result;
if ( result < 1.0 )
{
result = 1.0;
}
return ( int ) result;
}
int verblib_initialize ( verblib* verb, unsigned long sample_rate, unsigned int channels )
{
int i;
if ( channels != 1 && channels != 2 )
{
return 0; /* Currently supports only 1 or 2 channels. */
}
if ( sample_rate < 22050 )
{
return 0; /* The minimum supported sample rate is 22050 HZ. */
}
else if ( sample_rate > 44100 * verblib_max_sample_rate_multiplier )
{
return 0; /* The sample rate is too high. */
}
verb->channels = channels;
/* Tie the components to their buffers. */
verblib_comb_initialize ( &verb->combL[0], verb->bufcombL1, verblib_get_verblib_scaled_buffer_size ( sample_rate, verblib_combtuningL1 ) );
verblib_comb_initialize ( &verb->combR[0], verb->bufcombR1, verblib_get_verblib_scaled_buffer_size ( sample_rate, verblib_combtuningR1 ) );
verblib_comb_initialize ( &verb->combL[1], verb->bufcombL2, verblib_get_verblib_scaled_buffer_size ( sample_rate, verblib_combtuningL2 ) );
verblib_comb_initialize ( &verb->combR[1], verb->bufcombR2, verblib_get_verblib_scaled_buffer_size ( sample_rate, verblib_combtuningR2 ) );
verblib_comb_initialize ( &verb->combL[2], verb->bufcombL3, verblib_get_verblib_scaled_buffer_size ( sample_rate, verblib_combtuningL3 ) );
verblib_comb_initialize ( &verb->combR[2], verb->bufcombR3, verblib_get_verblib_scaled_buffer_size ( sample_rate, verblib_combtuningR3 ) );
verblib_comb_initialize ( &verb->combL[3], verb->bufcombL4, verblib_get_verblib_scaled_buffer_size ( sample_rate, verblib_combtuningL4 ) );
verblib_comb_initialize ( &verb->combR[3], verb->bufcombR4, verblib_get_verblib_scaled_buffer_size ( sample_rate, verblib_combtuningR4 ) );
verblib_comb_initialize ( &verb->combL[4], verb->bufcombL5, verblib_get_verblib_scaled_buffer_size ( sample_rate, verblib_combtuningL5 ) );
verblib_comb_initialize ( &verb->combR[4], verb->bufcombR5, verblib_get_verblib_scaled_buffer_size ( sample_rate, verblib_combtuningR5 ) );
verblib_comb_initialize ( &verb->combL[5], verb->bufcombL6, verblib_get_verblib_scaled_buffer_size ( sample_rate, verblib_combtuningL6 ) );
verblib_comb_initialize ( &verb->combR[5], verb->bufcombR6, verblib_get_verblib_scaled_buffer_size ( sample_rate, verblib_combtuningR6 ) );
verblib_comb_initialize ( &verb->combL[6], verb->bufcombL7, verblib_get_verblib_scaled_buffer_size ( sample_rate, verblib_combtuningL7 ) );
verblib_comb_initialize ( &verb->combR[6], verb->bufcombR7, verblib_get_verblib_scaled_buffer_size ( sample_rate, verblib_combtuningR7 ) );
verblib_comb_initialize ( &verb->combL[7], verb->bufcombL8, verblib_get_verblib_scaled_buffer_size ( sample_rate, verblib_combtuningL8 ) );
verblib_comb_initialize ( &verb->combR[7], verb->bufcombR8, verblib_get_verblib_scaled_buffer_size ( sample_rate, verblib_combtuningR8 ) );
verblib_allpass_initialize ( &verb->allpassL[0], verb->bufallpassL1, verblib_get_verblib_scaled_buffer_size ( sample_rate, verblib_allpasstuningL1 ) );
verblib_allpass_initialize ( &verb->allpassR[0], verb->bufallpassR1, verblib_get_verblib_scaled_buffer_size ( sample_rate, verblib_allpasstuningR1 ) );
verblib_allpass_initialize ( &verb->allpassL[1], verb->bufallpassL2, verblib_get_verblib_scaled_buffer_size ( sample_rate, verblib_allpasstuningL2 ) );
verblib_allpass_initialize ( &verb->allpassR[1], verb->bufallpassR2, verblib_get_verblib_scaled_buffer_size ( sample_rate, verblib_allpasstuningR2 ) );
verblib_allpass_initialize ( &verb->allpassL[2], verb->bufallpassL3, verblib_get_verblib_scaled_buffer_size ( sample_rate, verblib_allpasstuningL3 ) );
verblib_allpass_initialize ( &verb->allpassR[2], verb->bufallpassR3, verblib_get_verblib_scaled_buffer_size ( sample_rate, verblib_allpasstuningR3 ) );
verblib_allpass_initialize ( &verb->allpassL[3], verb->bufallpassL4, verblib_get_verblib_scaled_buffer_size ( sample_rate, verblib_allpasstuningL4 ) );
verblib_allpass_initialize ( &verb->allpassR[3], verb->bufallpassR4, verblib_get_verblib_scaled_buffer_size ( sample_rate, verblib_allpasstuningR4 ) );
/* Set default values. */
for ( i = 0; i < verblib_numallpasses; i++ )
{
verb->allpassL[i].feedback = 0.5f;
verb->allpassR[i].feedback = 0.5f;
}
verblib_set_wet ( verb, verblib_initialwet );
verblib_set_room_size ( verb, verblib_initialroom );
verblib_set_dry ( verb, verblib_initialdry );
verblib_set_damping ( verb, verblib_initialdamp );
verblib_set_width ( verb, verblib_initialwidth );
verblib_set_input_width ( verb, verblib_initialinputwidth );
verblib_set_mode ( verb, verblib_initialmode );
/* The buffers will be full of rubbish - so we MUST mute them. */
verblib_mute ( verb );
return 1;
}
void verblib_process ( verblib* verb, const float* input_buffer, float* output_buffer, unsigned long frames )
{
int i;
float outL, outR, input;
if ( verb->channels == 1 )
{
while ( frames-- > 0 )
{
outL = 0.0f;
input = ( input_buffer[0] * 2.0f ) * verb->gain;
/* Accumulate comb filters in parallel. */
for ( i = 0; i < verblib_numcombs; i++ )
{
outL += verblib_comb_process ( &verb->combL[i], input );
}
/* Feed through allpasses in series. */
for ( i = 0; i < verblib_numallpasses; i++ )
{
outL = verblib_allpass_process ( &verb->allpassL[i], outL );
}
/* Calculate output REPLACING anything already there. */
output_buffer[0] = outL * verb->wet1 + input_buffer[0] * verb->dry;
/* Increment sample pointers. */
++input_buffer;
++output_buffer;
}
}
else if ( verb->channels == 2 )
{
if ( verb->input_width > 0.0f ) /* Stereo input is widened or narrowed. */
{
/*
* The stereo mid/side code is derived from:
* https://www.musicdsp.org/en/latest/Effects/256-stereo-width-control-obtained-via-transfromation-matrix.html
* The description of the code on the above page says:
*
* This work is hereby placed in the public domain for all purposes, including
* use in commercial applications.
*/
const float tmp = 1 / verblib_max ( 1 + verb->input_width, 2 );
const float coef_mid = 1 * tmp;
const float coef_side = verb->input_width * tmp;
while ( frames-- > 0 )
{
const float mid = ( input_buffer[0] + input_buffer[1] ) * coef_mid;
const float side = ( input_buffer[1] - input_buffer[0] ) * coef_side;
const float input_left = ( mid - side ) * ( verb->gain * 2.0f );
const float input_right = ( mid + side ) * ( verb->gain * 2.0f );
outL = outR = 0.0f;
/* Accumulate comb filters in parallel. */
for ( i = 0; i < verblib_numcombs; i++ )
{
outL += verblib_comb_process ( &verb->combL[i], input_left );
outR += verblib_comb_process ( &verb->combR[i], input_right );
}
/* Feed through allpasses in series. */
for ( i = 0; i < verblib_numallpasses; i++ )
{
outL = verblib_allpass_process ( &verb->allpassL[i], outL );
outR = verblib_allpass_process ( &verb->allpassR[i], outR );
}
/* Calculate output REPLACING anything already there. */
output_buffer[0] = outL * verb->wet1 + outR * verb->wet2 + input_buffer[0] * verb->dry;
output_buffer[1] = outR * verb->wet1 + outL * verb->wet2 + input_buffer[1] * verb->dry;
/* Increment sample pointers. */
input_buffer += 2;
output_buffer += 2;
}
}
else /* Stereo input is summed to mono. */
{
while ( frames-- > 0 )
{
outL = outR = 0.0f;
input = ( input_buffer[0] + input_buffer[1] ) * verb->gain;
/* Accumulate comb filters in parallel. */
for ( i = 0; i < verblib_numcombs; i++ )
{
outL += verblib_comb_process ( &verb->combL[i], input );
outR += verblib_comb_process ( &verb->combR[i], input );
}
/* Feed through allpasses in series. */
for ( i = 0; i < verblib_numallpasses; i++ )
{
outL = verblib_allpass_process ( &verb->allpassL[i], outL );
outR = verblib_allpass_process ( &verb->allpassR[i], outR );
}
/* Calculate output REPLACING anything already there. */
output_buffer[0] = outL * verb->wet1 + outR * verb->wet2 + input_buffer[0] * verb->dry;
output_buffer[1] = outR * verb->wet1 + outL * verb->wet2 + input_buffer[1] * verb->dry;
/* Increment sample pointers. */
input_buffer += 2;
output_buffer += 2;
}
}
}
}
void verblib_set_room_size ( verblib* verb, float value )
{
verb->roomsize = ( value * verblib_scaleroom ) + verblib_offsetroom;
verblib_update ( verb );
}
float verblib_get_room_size ( const verblib* verb )
{
return ( verb->roomsize - verblib_offsetroom ) / verblib_scaleroom;
}
void verblib_set_damping ( verblib* verb, float value )
{
verb->damp = value * verblib_scaledamp;
verblib_update ( verb );
}
float verblib_get_damping ( const verblib* verb )
{
return verb->damp / verblib_scaledamp;
}
void verblib_set_wet ( verblib* verb, float value )
{
verb->wet = value * verblib_scalewet;
verblib_update ( verb );
}
float verblib_get_wet ( const verblib* verb )
{
return verb->wet / verblib_scalewet;
}
void verblib_set_dry ( verblib* verb, float value )
{
verb->dry = value * verblib_scaledry;
}
float verblib_get_dry ( const verblib* verb )
{
return verb->dry / verblib_scaledry;
}
void verblib_set_width ( verblib* verb, float value )
{
verb->width = value;
verblib_update ( verb );
}
float verblib_get_width ( const verblib* verb )
{
return verb->width;
}
void verblib_set_input_width ( verblib* verb, float value )
{
verb->input_width = value;
}
float verblib_get_input_width ( const verblib* verb )
{
return verb->input_width;
}
void verblib_set_mode ( verblib* verb, float value )
{
verb->mode = value;
verblib_update ( verb );
}
float verblib_get_mode ( const verblib* verb )
{
if ( verb->mode >= verblib_freezemode )
{
return 1.0f;
}
return 0.0f;
}
unsigned long verblib_get_decay_time_in_frames ( const verblib* verb )
{
double decay;
if ( verb->mode >= verblib_freezemode )
{
return 0; /* Freeze mode creates an infinite decay. */
}
decay = verblib_silence_threshold / fabs ( -20.0 * log ( 1.0 / verb->roomsize1 ) );
decay *= ( double ) ( verb->combR[7].bufsize * 2 );
return ( unsigned long ) decay;
}
#endif /* VERBLIB_IMPLEMENTATION */
/* REVISION HISTORY
*
* Version 0.5 - 2022-10-25
* Added two functions called verblib_set_input_width and verblib_get_input_width.
*
* Version 0.4 - 2021-01-23
* Added a function called verblib_get_decay_time_in_frames.
*
* Version 0.3 - 2021-01-18
* Added support for sample rates of 22050 and above.
*
* Version 0.2 - 2021-01-17
* Added support for processing mono audio.
*
* Version 0.1 - 2021-01-17
* Initial release.
*/
/* LICENSE
This software is available under 2 licenses -- choose whichever you prefer.
------------------------------------------------------------------------------
ALTERNATIVE A - MIT No Attribution License
Copyright (c) 2022 Philip Bennefall
Permission is hereby granted, free of charge, to any person obtaining a copy of
this software and associated documentation files (the "Software"), to deal in
the Software without restriction, including without limitation the rights to
use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
of the Software, and to permit persons to whom the Software is furnished to do
so.
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
SOFTWARE.
------------------------------------------------------------------------------
ALTERNATIVE B - Public Domain (www.unlicense.org)
This is free and unencumbered software released into the public domain.
Anyone is free to copy, modify, publish, use, compile, sell, or distribute this
software, either in source code form or as a compiled binary, for any purpose,
commercial or non-commercial, and by any means.
In jurisdictions that recognize copyright laws, the author or authors of this
software dedicate any and all copyright interest in the software to the public
domain. We make this dedication for the benefit of the public at large and to
the detriment of our heirs and successors. We intend this dedication to be an
overt act of relinquishment in perpetuity of all present and future rights to
this software under copyright law.
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
AUTHORS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN
ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
------------------------------------------------------------------------------
*/