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🛠️🐜 Antkeeper superbuild with dependencies included https://antkeeper.com
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superbuild/modules/openal-soft/examples/alconvolve.c

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/*
* OpenAL Convolution Reverb Example
*
* Copyright (c) 2020 by Chris Robinson <chris.kcat@gmail.com>
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/* This file contains an example for applying convolution reverb to a source. */
#include <assert.h>
#include <inttypes.h>
#include <limits.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include "sndfile.h"
#include "AL/al.h"
#include "AL/alext.h"
#include "common/alhelpers.h"
#ifndef AL_SOFT_convolution_reverb
#define AL_SOFT_convolution_reverb
#define AL_EFFECT_CONVOLUTION_REVERB_SOFT 0xA000
#endif
/* Filter object functions */
static LPALGENFILTERS alGenFilters;
static LPALDELETEFILTERS alDeleteFilters;
static LPALISFILTER alIsFilter;
static LPALFILTERI alFilteri;
static LPALFILTERIV alFilteriv;
static LPALFILTERF alFilterf;
static LPALFILTERFV alFilterfv;
static LPALGETFILTERI alGetFilteri;
static LPALGETFILTERIV alGetFilteriv;
static LPALGETFILTERF alGetFilterf;
static LPALGETFILTERFV alGetFilterfv;
/* Effect object functions */
static LPALGENEFFECTS alGenEffects;
static LPALDELETEEFFECTS alDeleteEffects;
static LPALISEFFECT alIsEffect;
static LPALEFFECTI alEffecti;
static LPALEFFECTIV alEffectiv;
static LPALEFFECTF alEffectf;
static LPALEFFECTFV alEffectfv;
static LPALGETEFFECTI alGetEffecti;
static LPALGETEFFECTIV alGetEffectiv;
static LPALGETEFFECTF alGetEffectf;
static LPALGETEFFECTFV alGetEffectfv;
/* Auxiliary Effect Slot object functions */
static LPALGENAUXILIARYEFFECTSLOTS alGenAuxiliaryEffectSlots;
static LPALDELETEAUXILIARYEFFECTSLOTS alDeleteAuxiliaryEffectSlots;
static LPALISAUXILIARYEFFECTSLOT alIsAuxiliaryEffectSlot;
static LPALAUXILIARYEFFECTSLOTI alAuxiliaryEffectSloti;
static LPALAUXILIARYEFFECTSLOTIV alAuxiliaryEffectSlotiv;
static LPALAUXILIARYEFFECTSLOTF alAuxiliaryEffectSlotf;
static LPALAUXILIARYEFFECTSLOTFV alAuxiliaryEffectSlotfv;
static LPALGETAUXILIARYEFFECTSLOTI alGetAuxiliaryEffectSloti;
static LPALGETAUXILIARYEFFECTSLOTIV alGetAuxiliaryEffectSlotiv;
static LPALGETAUXILIARYEFFECTSLOTF alGetAuxiliaryEffectSlotf;
static LPALGETAUXILIARYEFFECTSLOTFV alGetAuxiliaryEffectSlotfv;
/* This stuff defines a simple streaming player object, the same as alstream.c.
* Comments are removed for brevity, see alstream.c for more details.
*/
#define NUM_BUFFERS 4
#define BUFFER_SAMPLES 8192
typedef struct StreamPlayer {
ALuint buffers[NUM_BUFFERS];
ALuint source;
SNDFILE *sndfile;
SF_INFO sfinfo;
float *membuf;
ALenum format;
} StreamPlayer;
static StreamPlayer *NewPlayer(void)
{
StreamPlayer *player;
player = calloc(1, sizeof(*player));
assert(player != NULL);
alGenBuffers(NUM_BUFFERS, player->buffers);
assert(alGetError() == AL_NO_ERROR && "Could not create buffers");
alGenSources(1, &player->source);
assert(alGetError() == AL_NO_ERROR && "Could not create source");
alSource3i(player->source, AL_POSITION, 0, 0, -1);
alSourcei(player->source, AL_SOURCE_RELATIVE, AL_TRUE);
alSourcei(player->source, AL_ROLLOFF_FACTOR, 0);
assert(alGetError() == AL_NO_ERROR && "Could not set source parameters");
return player;
}
static void ClosePlayerFile(StreamPlayer *player)
{
if(player->sndfile)
sf_close(player->sndfile);
player->sndfile = NULL;
free(player->membuf);
player->membuf = NULL;
}
static void DeletePlayer(StreamPlayer *player)
{
ClosePlayerFile(player);
alDeleteSources(1, &player->source);
alDeleteBuffers(NUM_BUFFERS, player->buffers);
if(alGetError() != AL_NO_ERROR)
fprintf(stderr, "Failed to delete object IDs\n");
memset(player, 0, sizeof(*player));
free(player);
}
static int OpenPlayerFile(StreamPlayer *player, const char *filename)
{
size_t frame_size;
ClosePlayerFile(player);
player->sndfile = sf_open(filename, SFM_READ, &player->sfinfo);
if(!player->sndfile)
{
fprintf(stderr, "Could not open audio in %s: %s\n", filename, sf_strerror(NULL));
return 0;
}
player->format = AL_NONE;
if(player->sfinfo.channels == 1)
player->format = AL_FORMAT_MONO_FLOAT32;
else if(player->sfinfo.channels == 2)
player->format = AL_FORMAT_STEREO_FLOAT32;
else if(player->sfinfo.channels == 6)
player->format = AL_FORMAT_51CHN32;
else if(player->sfinfo.channels == 3)
{
if(sf_command(player->sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
player->format = AL_FORMAT_BFORMAT2D_FLOAT32;
}
else if(player->sfinfo.channels == 4)
{
if(sf_command(player->sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
player->format = AL_FORMAT_BFORMAT3D_FLOAT32;
}
if(!player->format)
{
fprintf(stderr, "Unsupported channel count: %d\n", player->sfinfo.channels);
sf_close(player->sndfile);
player->sndfile = NULL;
return 0;
}
frame_size = (size_t)(BUFFER_SAMPLES * player->sfinfo.channels) * sizeof(float);
player->membuf = malloc(frame_size);
return 1;
}
static int StartPlayer(StreamPlayer *player)
{
ALsizei i;
alSourceRewind(player->source);
alSourcei(player->source, AL_BUFFER, 0);
for(i = 0;i < NUM_BUFFERS;i++)
{
sf_count_t slen = sf_readf_float(player->sndfile, player->membuf, BUFFER_SAMPLES);
if(slen < 1) break;
slen *= player->sfinfo.channels * (sf_count_t)sizeof(float);
alBufferData(player->buffers[i], player->format, player->membuf, (ALsizei)slen,
player->sfinfo.samplerate);
}
if(alGetError() != AL_NO_ERROR)
{
fprintf(stderr, "Error buffering for playback\n");
return 0;
}
alSourceQueueBuffers(player->source, i, player->buffers);
alSourcePlay(player->source);
if(alGetError() != AL_NO_ERROR)
{
fprintf(stderr, "Error starting playback\n");
return 0;
}
return 1;
}
static int UpdatePlayer(StreamPlayer *player)
{
ALint processed, state;
alGetSourcei(player->source, AL_SOURCE_STATE, &state);
alGetSourcei(player->source, AL_BUFFERS_PROCESSED, &processed);
if(alGetError() != AL_NO_ERROR)
{
fprintf(stderr, "Error checking source state\n");
return 0;
}
while(processed > 0)
{
ALuint bufid;
sf_count_t slen;
alSourceUnqueueBuffers(player->source, 1, &bufid);
processed--;
slen = sf_readf_float(player->sndfile, player->membuf, BUFFER_SAMPLES);
if(slen > 0)
{
slen *= player->sfinfo.channels * (sf_count_t)sizeof(float);
alBufferData(bufid, player->format, player->membuf, (ALsizei)slen,
player->sfinfo.samplerate);
alSourceQueueBuffers(player->source, 1, &bufid);
}
if(alGetError() != AL_NO_ERROR)
{
fprintf(stderr, "Error buffering data\n");
return 0;
}
}
if(state != AL_PLAYING && state != AL_PAUSED)
{
ALint queued;
alGetSourcei(player->source, AL_BUFFERS_QUEUED, &queued);
if(queued == 0)
return 0;
alSourcePlay(player->source);
if(alGetError() != AL_NO_ERROR)
{
fprintf(stderr, "Error restarting playback\n");
return 0;
}
}
return 1;
}
/* CreateEffect creates a new OpenAL effect object with a convolution reverb
* type, and returns the new effect ID.
*/
static ALuint CreateEffect(void)
{
ALuint effect = 0;
ALenum err;
printf("Using Convolution Reverb\n");
/* Create the effect object and set the convolution reverb effect type. */
alGenEffects(1, &effect);
alEffecti(effect, AL_EFFECT_TYPE, AL_EFFECT_CONVOLUTION_REVERB_SOFT);
/* Check if an error occured, and clean up if so. */
err = alGetError();
if(err != AL_NO_ERROR)
{
fprintf(stderr, "OpenAL error: %s\n", alGetString(err));
if(alIsEffect(effect))
alDeleteEffects(1, &effect);
return 0;
}
return effect;
}
/* LoadBuffer loads the named audio file into an OpenAL buffer object, and
* returns the new buffer ID.
*/
static ALuint LoadSound(const char *filename)
{
const char *namepart;
ALenum err, format;
ALuint buffer;
SNDFILE *sndfile;
SF_INFO sfinfo;
float *membuf;
sf_count_t num_frames;
ALsizei num_bytes;
/* Open the audio file and check that it's usable. */
sndfile = sf_open(filename, SFM_READ, &sfinfo);
if(!sndfile)
{
fprintf(stderr, "Could not open audio in %s: %s\n", filename, sf_strerror(sndfile));
return 0;
}
if(sfinfo.frames < 1 || sfinfo.frames > (sf_count_t)(INT_MAX/sizeof(float))/sfinfo.channels)
{
fprintf(stderr, "Bad sample count in %s (%" PRId64 ")\n", filename, sfinfo.frames);
sf_close(sndfile);
return 0;
}
/* Get the sound format, and figure out the OpenAL format. Use floats since
* impulse responses will usually have more than 16-bit precision.
*/
format = AL_NONE;
if(sfinfo.channels == 1)
format = AL_FORMAT_MONO_FLOAT32;
else if(sfinfo.channels == 2)
format = AL_FORMAT_STEREO_FLOAT32;
else if(sfinfo.channels == 3)
{
if(sf_command(sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
format = AL_FORMAT_BFORMAT2D_FLOAT32;
}
else if(sfinfo.channels == 4)
{
if(sf_command(sndfile, SFC_WAVEX_GET_AMBISONIC, NULL, 0) == SF_AMBISONIC_B_FORMAT)
format = AL_FORMAT_BFORMAT3D_FLOAT32;
}
if(!format)
{
fprintf(stderr, "Unsupported channel count: %d\n", sfinfo.channels);
sf_close(sndfile);
return 0;
}
namepart = strrchr(filename, '/');
if(namepart || (namepart=strrchr(filename, '\\')))
namepart++;
else
namepart = filename;
printf("Loading: %s (%s, %dhz, %" PRId64 " samples / %.2f seconds)\n", namepart,
FormatName(format), sfinfo.samplerate, sfinfo.frames,
(double)sfinfo.frames / sfinfo.samplerate);
fflush(stdout);
/* Decode the whole audio file to a buffer. */
membuf = malloc((size_t)(sfinfo.frames * sfinfo.channels) * sizeof(float));
num_frames = sf_readf_float(sndfile, membuf, sfinfo.frames);
if(num_frames < 1)
{
free(membuf);
sf_close(sndfile);
fprintf(stderr, "Failed to read samples in %s (%" PRId64 ")\n", filename, num_frames);
return 0;
}
num_bytes = (ALsizei)(num_frames * sfinfo.channels) * (ALsizei)sizeof(float);
/* Buffer the audio data into a new buffer object, then free the data and
* close the file.
*/
buffer = 0;
alGenBuffers(1, &buffer);
alBufferData(buffer, format, membuf, num_bytes, sfinfo.samplerate);
free(membuf);
sf_close(sndfile);
/* Check if an error occured, and clean up if so. */
err = alGetError();
if(err != AL_NO_ERROR)
{
fprintf(stderr, "OpenAL Error: %s\n", alGetString(err));
if(buffer && alIsBuffer(buffer))
alDeleteBuffers(1, &buffer);
return 0;
}
return buffer;
}
int main(int argc, char **argv)
{
ALuint ir_buffer, filter, effect, slot;
StreamPlayer *player;
int i;
/* Print out usage if no arguments were specified */
if(argc < 2)
{
fprintf(stderr, "Usage: %s [-device <name>] <impulse response file> "
"<[-dry | -nodry] filename>...\n", argv[0]);
return 1;
}
argv++; argc--;
if(InitAL(&argv, &argc) != 0)
return 1;
if(!alIsExtensionPresent("AL_SOFTX_convolution_reverb"))
{
CloseAL();
fprintf(stderr, "Error: Convolution revern not supported\n");
return 1;
}
if(argc < 2)
{
CloseAL();
fprintf(stderr, "Error: Missing impulse response or sound files\n");
return 1;
}
/* Define a macro to help load the function pointers. */
#define LOAD_PROC(T, x) ((x) = FUNCTION_CAST(T, alGetProcAddress(#x)))
LOAD_PROC(LPALGENFILTERS, alGenFilters);
LOAD_PROC(LPALDELETEFILTERS, alDeleteFilters);
LOAD_PROC(LPALISFILTER, alIsFilter);
LOAD_PROC(LPALFILTERI, alFilteri);
LOAD_PROC(LPALFILTERIV, alFilteriv);
LOAD_PROC(LPALFILTERF, alFilterf);
LOAD_PROC(LPALFILTERFV, alFilterfv);
LOAD_PROC(LPALGETFILTERI, alGetFilteri);
LOAD_PROC(LPALGETFILTERIV, alGetFilteriv);
LOAD_PROC(LPALGETFILTERF, alGetFilterf);
LOAD_PROC(LPALGETFILTERFV, alGetFilterfv);
LOAD_PROC(LPALGENEFFECTS, alGenEffects);
LOAD_PROC(LPALDELETEEFFECTS, alDeleteEffects);
LOAD_PROC(LPALISEFFECT, alIsEffect);
LOAD_PROC(LPALEFFECTI, alEffecti);
LOAD_PROC(LPALEFFECTIV, alEffectiv);
LOAD_PROC(LPALEFFECTF, alEffectf);
LOAD_PROC(LPALEFFECTFV, alEffectfv);
LOAD_PROC(LPALGETEFFECTI, alGetEffecti);
LOAD_PROC(LPALGETEFFECTIV, alGetEffectiv);
LOAD_PROC(LPALGETEFFECTF, alGetEffectf);
LOAD_PROC(LPALGETEFFECTFV, alGetEffectfv);
LOAD_PROC(LPALGENAUXILIARYEFFECTSLOTS, alGenAuxiliaryEffectSlots);
LOAD_PROC(LPALDELETEAUXILIARYEFFECTSLOTS, alDeleteAuxiliaryEffectSlots);
LOAD_PROC(LPALISAUXILIARYEFFECTSLOT, alIsAuxiliaryEffectSlot);
LOAD_PROC(LPALAUXILIARYEFFECTSLOTI, alAuxiliaryEffectSloti);
LOAD_PROC(LPALAUXILIARYEFFECTSLOTIV, alAuxiliaryEffectSlotiv);
LOAD_PROC(LPALAUXILIARYEFFECTSLOTF, alAuxiliaryEffectSlotf);
LOAD_PROC(LPALAUXILIARYEFFECTSLOTFV, alAuxiliaryEffectSlotfv);
LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTI, alGetAuxiliaryEffectSloti);
LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTIV, alGetAuxiliaryEffectSlotiv);
LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTF, alGetAuxiliaryEffectSlotf);
LOAD_PROC(LPALGETAUXILIARYEFFECTSLOTFV, alGetAuxiliaryEffectSlotfv);
#undef LOAD_PROC
/* Load the reverb into an effect. */
effect = CreateEffect();
if(!effect)
{
CloseAL();
return 1;
}
/* Load the impulse response sound into a buffer. */
ir_buffer = LoadSound(argv[0]);
if(!ir_buffer)
{
alDeleteEffects(1, &effect);
CloseAL();
return 1;
}
/* Create the effect slot object. This is what "plays" an effect on sources
* that connect to it.
*/
slot = 0;
alGenAuxiliaryEffectSlots(1, &slot);
/* Set the impulse response sound buffer on the effect slot. This allows
* effects to access it as needed. In this case, convolution reverb uses it
* as the filter source. NOTE: Unlike the effect object, the buffer *is*
* kept referenced and may not be changed or deleted as long as it's set,
* just like with a source. When another buffer is set, or the effect slot
* is deleted, the buffer reference is released.
*
* The effect slot's gain is reduced because the impulse responses I've
* tested with result in excessively loud reverb. Is that normal? Even with
* this, it seems a bit on the loud side.
*
* Also note: unlike standard or EAX reverb, there is no automatic
* attenuation of a source's reverb response with distance, so the reverb
* will remain full volume regardless of a given sound's distance from the
* listener. You can use a send filter to alter a given source's
* contribution to reverb.
*/
alAuxiliaryEffectSloti(slot, AL_BUFFER, (ALint)ir_buffer);
alAuxiliaryEffectSlotf(slot, AL_EFFECTSLOT_GAIN, 1.0f / 16.0f);
alAuxiliaryEffectSloti(slot, AL_EFFECTSLOT_EFFECT, (ALint)effect);
assert(alGetError()==AL_NO_ERROR && "Failed to set effect slot");
/* Create a filter that can silence the dry path. */
filter = 0;
alGenFilters(1, &filter);
alFilteri(filter, AL_FILTER_TYPE, AL_FILTER_LOWPASS);
alFilterf(filter, AL_LOWPASS_GAIN, 0.0f);
player = NewPlayer();
/* Connect the player's source to the effect slot. */
alSource3i(player->source, AL_AUXILIARY_SEND_FILTER, (ALint)slot, 0, AL_FILTER_NULL);
assert(alGetError()==AL_NO_ERROR && "Failed to setup sound source");
/* Play each file listed on the command line */
for(i = 1;i < argc;i++)
{
const char *namepart;
if(argc-i > 1)
{
if(strcasecmp(argv[i], "-nodry") == 0)
{
alSourcei(player->source, AL_DIRECT_FILTER, (ALint)filter);
++i;
}
else if(strcasecmp(argv[i], "-dry") == 0)
{
alSourcei(player->source, AL_DIRECT_FILTER, AL_FILTER_NULL);
++i;
}
}
if(!OpenPlayerFile(player, argv[i]))
continue;
namepart = strrchr(argv[i], '/');
if(namepart || (namepart=strrchr(argv[i], '\\')))
namepart++;
else
namepart = argv[i];
printf("Playing: %s (%s, %dhz)\n", namepart, FormatName(player->format),
player->sfinfo.samplerate);
fflush(stdout);
if(!StartPlayer(player))
{
ClosePlayerFile(player);
continue;
}
while(UpdatePlayer(player))
al_nssleep(10000000);
ClosePlayerFile(player);
}
printf("Done.\n");
/* All files done. Delete the player and effect resources, and close down
* OpenAL.
*/
DeletePlayer(player);
player = NULL;
alDeleteAuxiliaryEffectSlots(1, &slot);
alDeleteEffects(1, &effect);
alDeleteFilters(1, &filter);
alDeleteBuffers(1, &ir_buffer);
CloseAL();
return 0;
}