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libretro-o2em/core_audio_mixer.c
jdgleaver 218c9cd2a2
Audio fixes (#55)
2021-11-29 16:50:27 +01:00

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/* Copyright (C) 2010-2020 The RetroArch team
*
* ---------------------------------------------------------------------------------------
* The following license statement only applies to this file (core_audio_mixer.c).
* ---------------------------------------------------------------------------------------
*
* 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.
*/
#ifdef HAVE_CONFIG_H
#include "../../config.h"
#endif
#include <core_audio_mixer.h>
#include <audio/audio_resampler.h>
#ifdef HAVE_RWAV
#include <formats/rwav.h>
#endif
#include <memalign.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>
#ifdef HAVE_STB_VORBIS
#define STB_VORBIS_NO_PUSHDATA_API
#define STB_VORBIS_NO_STDIO
#define STB_VORBIS_NO_CRT
#include <stb/stb_vorbis.h>
#endif
#ifdef HAVE_DR_FLAC
#define DR_FLAC_IMPLEMENTATION
#include <dr/dr_flac.h>
#endif
#ifdef HAVE_DR_MP3
#define DR_MP3_IMPLEMENTATION
#include <retro_assert.h>
#define DRMP3_ASSERT(expression) retro_assert(expression)
#include <dr/dr_mp3.h>
#endif
#ifdef HAVE_IBXM
#include <ibxm/ibxm.h>
#endif
#define CORE_AUDIO_MIXER_MAX_VOICES 1
#define CORE_AUDIO_MIXER_TEMP_BUFFER 8192
struct core_audio_mixer_sound
{
enum core_audio_mixer_type type;
union
{
struct
{
/* wav */
const float* pcm;
unsigned frames;
} wav;
#ifdef HAVE_STB_VORBIS
struct
{
/* ogg */
const void* data;
unsigned size;
} ogg;
#endif
#ifdef HAVE_DR_FLAC
struct
{
/* flac */
const void* data;
unsigned size;
} flac;
#endif
#ifdef HAVE_DR_MP3
struct
{
/* mp */
const void* data;
unsigned size;
} mp3;
#endif
#ifdef HAVE_IBXM
struct
{
/* mod/s3m/xm */
const void* data;
unsigned size;
} mod;
#endif
} types;
};
struct core_audio_mixer_voice
{
union
{
struct
{
unsigned position;
} wav;
#ifdef HAVE_STB_VORBIS
struct
{
stb_vorbis *stream;
void *resampler_data;
const retro_resampler_t *resampler;
float *buffer;
unsigned position;
unsigned samples;
unsigned buf_samples;
float ratio;
} ogg;
#endif
#ifdef HAVE_DR_FLAC
struct
{
float* buffer;
drflac *stream;
void *resampler_data;
const retro_resampler_t *resampler;
unsigned position;
unsigned samples;
unsigned buf_samples;
float ratio;
} flac;
#endif
#ifdef HAVE_DR_MP3
struct
{
drmp3 stream;
void *resampler_data;
const retro_resampler_t *resampler;
float* buffer;
unsigned position;
unsigned samples;
unsigned buf_samples;
float ratio;
} mp3;
#endif
#ifdef HAVE_IBXM
struct
{
int* buffer;
struct replay* stream;
struct module* module;
unsigned position;
unsigned samples;
unsigned buf_samples;
} mod;
#endif
} types;
core_audio_mixer_sound_t *sound;
core_audio_mixer_stop_cb_t stop_cb;
unsigned type;
float volume;
bool repeat;
};
/* TODO/FIXME - static globals */
static struct core_audio_mixer_voice core_s_voices[CORE_AUDIO_MIXER_MAX_VOICES] = {0};
static unsigned core_s_rate = 0;
#ifdef HAVE_RWAV
static bool wav_to_float(const rwav_t* wav, float** pcm, size_t samples_out)
{
size_t i;
/* Allocate on a 16-byte boundary, and pad to a multiple of 16 bytes */
float *f = (float*)memalign_alloc(16,
((samples_out + 15) & ~15) * sizeof(float));
if (!f)
return false;
*pcm = f;
if (wav->bitspersample == 8)
{
float sample = 0.0f;
const uint8_t *u8 = (const uint8_t*)wav->samples;
if (wav->numchannels == 1)
{
for (i = wav->numsamples; i != 0; i--)
{
sample = (float)*u8++ / 255.0f;
sample = sample * 2.0f - 1.0f;
*f++ = sample;
*f++ = sample;
}
}
else if (wav->numchannels == 2)
{
for (i = wav->numsamples; i != 0; i--)
{
sample = (float)*u8++ / 255.0f;
sample = sample * 2.0f - 1.0f;
*f++ = sample;
sample = (float)*u8++ / 255.0f;
sample = sample * 2.0f - 1.0f;
*f++ = sample;
}
}
}
else
{
/* TODO/FIXME note to leiradel - can we use audio/conversion/s16_to_float
* functions here? */
float sample = 0.0f;
const int16_t *s16 = (const int16_t*)wav->samples;
if (wav->numchannels == 1)
{
for (i = wav->numsamples; i != 0; i--)
{
sample = (float)((int)*s16++ + 32768) / 65535.0f;
sample = sample * 2.0f - 1.0f;
*f++ = sample;
*f++ = sample;
}
}
else if (wav->numchannels == 2)
{
for (i = wav->numsamples; i != 0; i--)
{
sample = (float)((int)*s16++ + 32768) / 65535.0f;
sample = sample * 2.0f - 1.0f;
*f++ = sample;
sample = (float)((int)*s16++ + 32768) / 65535.0f;
sample = sample * 2.0f - 1.0f;
*f++ = sample;
}
}
}
return true;
}
static bool one_shot_resample(const float* in, size_t samples_in,
unsigned rate, const char *resampler_ident, enum resampler_quality quality,
float** out, size_t* samples_out)
{
struct resampler_data info;
void* data = NULL;
const retro_resampler_t* resampler = NULL;
float ratio = (double)core_s_rate / (double)rate;
if (!retro_resampler_realloc(&data, &resampler,
resampler_ident, quality, ratio))
return false;
/* Allocate on a 16-byte boundary, and pad to a multiple of 16 bytes. We
* add 16 more samples in the formula below just as safeguard, because
* resampler->process sometimes reports more output samples than the
* formula below calculates. Ideally, audio resamplers should have a
* function to return the number of samples they will output given a
* count of input samples. */
*samples_out = (size_t)(samples_in * ratio);
*out = (float*)memalign_alloc(16,
(((*samples_out + 16) + 15) & ~15) * sizeof(float));
if (*out == NULL)
return false;
info.data_in = in;
info.data_out = *out;
info.input_frames = samples_in / 2;
info.output_frames = 0;
info.ratio = ratio;
resampler->process(data, &info);
resampler->free(data);
return true;
}
#endif
void core_audio_mixer_init(unsigned rate)
{
unsigned i;
core_s_rate = rate;
for (i = 0; i < CORE_AUDIO_MIXER_MAX_VOICES; i++)
core_s_voices[i].type = CORE_AUDIO_MIXER_TYPE_NONE;
}
void core_audio_mixer_done(void)
{
unsigned i;
for (i = 0; i < CORE_AUDIO_MIXER_MAX_VOICES; i++)
core_s_voices[i].type = CORE_AUDIO_MIXER_TYPE_NONE;
}
core_audio_mixer_sound_t* core_audio_mixer_load_wav(void *buffer, int32_t size,
const char *resampler_ident, enum resampler_quality quality)
{
#ifdef HAVE_RWAV
/* WAV data */
rwav_t wav;
/* WAV samples converted to float */
float* pcm = NULL;
size_t samples = 0;
/* Result */
core_audio_mixer_sound_t* sound = NULL;
wav.bitspersample = 0;
wav.numchannels = 0;
wav.samplerate = 0;
wav.numsamples = 0;
wav.subchunk2size = 0;
wav.samples = NULL;
if ((rwav_load(&wav, buffer, size)) != RWAV_ITERATE_DONE)
return NULL;
samples = wav.numsamples * 2;
if (!wav_to_float(&wav, &pcm, samples))
return NULL;
if (wav.samplerate != core_s_rate)
{
float* resampled = NULL;
if (!one_shot_resample(pcm, samples, wav.samplerate,
resampler_ident, quality,
&resampled, &samples))
return NULL;
memalign_free((void*)pcm);
pcm = resampled;
}
sound = (core_audio_mixer_sound_t*)calloc(1, sizeof(*sound));
if (!sound)
{
memalign_free((void*)pcm);
return NULL;
}
sound->type = CORE_AUDIO_MIXER_TYPE_WAV;
sound->types.wav.frames = (unsigned)(samples / 2);
sound->types.wav.pcm = pcm;
rwav_free(&wav);
return sound;
#else
return NULL;
#endif
}
core_audio_mixer_sound_t* core_audio_mixer_load_ogg(void *buffer, int32_t size)
{
#ifdef HAVE_STB_VORBIS
core_audio_mixer_sound_t* sound = (core_audio_mixer_sound_t*)calloc(1, sizeof(*sound));
if (!sound)
return NULL;
sound->type = CORE_AUDIO_MIXER_TYPE_OGG;
sound->types.ogg.size = size;
sound->types.ogg.data = buffer;
return sound;
#else
return NULL;
#endif
}
core_audio_mixer_sound_t* core_audio_mixer_load_flac(void *buffer, int32_t size)
{
#ifdef HAVE_DR_FLAC
core_audio_mixer_sound_t* sound = (core_audio_mixer_sound_t*)calloc(1, sizeof(*sound));
if (!sound)
return NULL;
sound->type = CORE_AUDIO_MIXER_TYPE_FLAC;
sound->types.flac.size = size;
sound->types.flac.data = buffer;
return sound;
#else
return NULL;
#endif
}
core_audio_mixer_sound_t* core_audio_mixer_load_mp3(void *buffer, int32_t size)
{
#ifdef HAVE_DR_MP3
core_audio_mixer_sound_t* sound = (core_audio_mixer_sound_t*)calloc(1, sizeof(*sound));
if (!sound)
return NULL;
sound->type = CORE_AUDIO_MIXER_TYPE_MP3;
sound->types.mp3.size = size;
sound->types.mp3.data = buffer;
return sound;
#else
return NULL;
#endif
}
core_audio_mixer_sound_t* core_audio_mixer_load_mod(void *buffer, int32_t size)
{
#ifdef HAVE_IBXM
core_audio_mixer_sound_t* sound = (core_audio_mixer_sound_t*)calloc(1, sizeof(*sound));
if (!sound)
return NULL;
sound->type = CORE_AUDIO_MIXER_TYPE_MOD;
sound->types.mod.size = size;
sound->types.mod.data = buffer;
return sound;
#else
return NULL;
#endif
}
void core_audio_mixer_destroy(core_audio_mixer_sound_t* sound)
{
void *handle = NULL;
if (!sound)
return;
switch (sound->type)
{
case CORE_AUDIO_MIXER_TYPE_WAV:
handle = (void*)sound->types.wav.pcm;
if (handle)
memalign_free(handle);
break;
case CORE_AUDIO_MIXER_TYPE_OGG:
#ifdef HAVE_STB_VORBIS
handle = (void*)sound->types.ogg.data;
if (handle)
free(handle);
#endif
break;
case CORE_AUDIO_MIXER_TYPE_MOD:
#ifdef HAVE_IBXM
handle = (void*)sound->types.mod.data;
if (handle)
free(handle);
#endif
break;
case CORE_AUDIO_MIXER_TYPE_FLAC:
#ifdef HAVE_DR_FLAC
handle = (void*)sound->types.flac.data;
if (handle)
free(handle);
#endif
break;
case CORE_AUDIO_MIXER_TYPE_MP3:
#ifdef HAVE_DR_MP3
handle = (void*)sound->types.mp3.data;
if (handle)
free(handle);
#endif
break;
case CORE_AUDIO_MIXER_TYPE_NONE:
break;
}
free(sound);
}
static bool core_audio_mixer_play_wav(core_audio_mixer_sound_t* sound,
core_audio_mixer_voice_t* voice, bool repeat, float volume,
core_audio_mixer_stop_cb_t stop_cb)
{
voice->types.wav.position = 0;
return true;
}
#ifdef HAVE_STB_VORBIS
static bool core_audio_mixer_play_ogg(
core_audio_mixer_sound_t* sound,
core_audio_mixer_voice_t* voice,
bool repeat, float volume,
const char *resampler_ident,
enum resampler_quality quality,
core_audio_mixer_stop_cb_t stop_cb)
{
stb_vorbis_info info;
int res = 0;
float ratio = 1.0f;
unsigned samples = 0;
void *ogg_buffer = NULL;
void *resampler_data = NULL;
const retro_resampler_t* resamp = NULL;
stb_vorbis *stb_vorbis = stb_vorbis_open_memory(
(const unsigned char*)sound->types.ogg.data,
sound->types.ogg.size, &res, NULL);
if (!stb_vorbis)
return false;
info = stb_vorbis_get_info(stb_vorbis);
if (info.sample_rate != core_s_rate)
{
ratio = (double)core_s_rate / (double)info.sample_rate;
if (!retro_resampler_realloc(&resampler_data,
&resamp, resampler_ident, quality,
ratio))
goto error;
}
/* Allocate on a 16-byte boundary, and pad to a multiple of 16 bytes. We
* add 16 more samples in the formula below just as safeguard, because
* resampler->process sometimes reports more output samples than the
* formula below calculates. Ideally, audio resamplers should have a
* function to return the number of samples they will output given a
* count of input samples. */
samples = (unsigned)(CORE_AUDIO_MIXER_TEMP_BUFFER * ratio);
ogg_buffer = (float*)memalign_alloc(16,
(((samples + 16) + 15) & ~15) * sizeof(float));
if (!ogg_buffer)
{
if (resamp && resampler_data)
resamp->free(resampler_data);
goto error;
}
/* "system" menu sounds may reuse the same voice without freeing anything first, so do that here if needed */
if (voice->types.ogg.stream)
stb_vorbis_close(voice->types.ogg.stream);
if (voice->types.ogg.resampler && voice->types.ogg.resampler_data)
voice->types.ogg.resampler->free(voice->types.ogg.resampler_data);
if (voice->types.ogg.buffer)
memalign_free(voice->types.ogg.buffer);
voice->types.ogg.resampler = resamp;
voice->types.ogg.resampler_data = resampler_data;
voice->types.ogg.buffer = (float*)ogg_buffer;
voice->types.ogg.buf_samples = samples;
voice->types.ogg.ratio = ratio;
voice->types.ogg.stream = stb_vorbis;
voice->types.ogg.position = 0;
voice->types.ogg.samples = 0;
return true;
error:
stb_vorbis_close(stb_vorbis);
return false;
}
#endif
#ifdef HAVE_IBXM
static bool core_audio_mixer_play_mod(
core_audio_mixer_sound_t* sound,
core_audio_mixer_voice_t* voice,
bool repeat, float volume,
core_audio_mixer_stop_cb_t stop_cb)
{
struct data data;
char message[64];
int buf_samples = 0;
int samples = 0;
void *mod_buffer = NULL;
struct module* module = NULL;
struct replay* replay = NULL;
data.buffer = (char*)sound->types.mod.data;
data.length = sound->types.mod.size;
module = module_load(&data, message);
if (!module)
{
printf("core_audio_mixer_play_mod module_load() failed with error: %s\n", message);
goto error;
}
if (voice->types.mod.module)
dispose_module(voice->types.mod.module);
voice->types.mod.module = module;
replay = new_replay(module, core_s_rate, 1);
if (!replay)
{
printf("core_audio_mixer_play_mod new_replay() failed\n");
goto error;
}
buf_samples = calculate_mix_buf_len(core_s_rate);
mod_buffer = memalign_alloc(16, ((buf_samples + 15) & ~15) * sizeof(int));
if (!mod_buffer)
{
printf("core_audio_mixer_play_mod cannot allocate mod_buffer !\n");
goto error;
}
samples = replay_calculate_duration(replay);
if (!samples)
{
printf("core_audio_mixer_play_mod cannot retrieve duration !\n");
goto error;
}
/* FIXME: stopping and then starting a mod stream will crash here in dispose_replay (ASAN says struct replay is misaligned?) */
if (voice->types.mod.stream)
dispose_replay(voice->types.mod.stream);
if (voice->types.mod.buffer)
memalign_free(voice->types.mod.buffer);
voice->types.mod.buffer = (int*)mod_buffer;
voice->types.mod.buf_samples = buf_samples;
voice->types.mod.stream = replay;
voice->types.mod.position = 0;
voice->types.mod.samples = 0; /* samples; */
return true;
error:
if (mod_buffer)
memalign_free(mod_buffer);
if (module)
dispose_module(module);
return false;
}
#endif
#ifdef HAVE_DR_FLAC
static bool core_audio_mixer_play_flac(
core_audio_mixer_sound_t* sound,
core_audio_mixer_voice_t* voice,
bool repeat, float volume,
const char *resampler_ident,
enum resampler_quality quality,
core_audio_mixer_stop_cb_t stop_cb)
{
float ratio = 1.0f;
unsigned samples = 0;
void *flac_buffer = NULL;
void *resampler_data = NULL;
const retro_resampler_t* resamp = NULL;
drflac *dr_flac = drflac_open_memory((const unsigned char*)sound->types.flac.data,sound->types.flac.size);
if (!dr_flac)
return false;
if (dr_flac->sampleRate != core_s_rate)
{
ratio = (double)core_s_rate / (double)(dr_flac->sampleRate);
if (!retro_resampler_realloc(&resampler_data,
&resamp, resampler_ident, quality,
ratio))
goto error;
}
/* Allocate on a 16-byte boundary, and pad to a multiple of 16 bytes. We
* add 16 more samples in the formula below just as safeguard, because
* resampler->process sometimes reports more output samples than the
* formula below calculates. Ideally, audio resamplers should have a
* function to return the number of samples they will output given a
* count of input samples. */
samples = (unsigned)(CORE_AUDIO_MIXER_TEMP_BUFFER * ratio);
flac_buffer = (float*)memalign_alloc(16,
(((samples + 16) + 15) & ~15) * sizeof(float));
if (!flac_buffer)
{
if (resamp && resamp->free)
resamp->free(resampler_data);
goto error;
}
if (voice->types.flac.stream)
drflac_close(voice->types.flac.stream);
if (voice->types.flac.resampler && voice->types.flac.resampler_data)
voice->types.flac.resampler->free(voice->types.flac.resampler_data);
if (voice->types.flac.buffer)
memalign_free(voice->types.flac.buffer);
voice->types.flac.resampler = resamp;
voice->types.flac.resampler_data = resampler_data;
voice->types.flac.buffer = (float*)flac_buffer;
voice->types.flac.buf_samples = samples;
voice->types.flac.ratio = ratio;
voice->types.flac.stream = dr_flac;
voice->types.flac.position = 0;
voice->types.flac.samples = 0;
return true;
error:
drflac_close(dr_flac);
return false;
}
#endif
#ifdef HAVE_DR_MP3
static bool core_audio_mixer_play_mp3(
core_audio_mixer_sound_t* sound,
core_audio_mixer_voice_t* voice,
bool repeat, float volume,
const char *resampler_ident,
enum resampler_quality quality,
core_audio_mixer_stop_cb_t stop_cb)
{
float ratio = 1.0f;
unsigned samples = 0;
void *mp3_buffer = NULL;
void *resampler_data = NULL;
const retro_resampler_t* resamp = NULL;
bool res;
if (voice->types.mp3.stream.pData)
{
drmp3_uninit(&voice->types.mp3.stream);
memset(&voice->types.mp3.stream, 0, sizeof(voice->types.mp3.stream));
}
res = drmp3_init_memory(&voice->types.mp3.stream, (const unsigned char*)sound->types.mp3.data, sound->types.mp3.size, NULL);
if (!res)
return false;
if (voice->types.mp3.stream.sampleRate != core_s_rate)
{
ratio = (double)core_s_rate / (double)(voice->types.mp3.stream.sampleRate);
if (!retro_resampler_realloc(&resampler_data,
&resamp, resampler_ident, quality,
ratio))
goto error;
}
/* Allocate on a 16-byte boundary, and pad to a multiple of 16 bytes. We
* add 16 more samples in the formula below just as safeguard, because
* resampler->process sometimes reports more output samples than the
* formula below calculates. Ideally, audio resamplers should have a
* function to return the number of samples they will output given a
* count of input samples. */
samples = (unsigned)(CORE_AUDIO_MIXER_TEMP_BUFFER * ratio);
mp3_buffer = (float*)memalign_alloc(16,
(((samples + 16) + 15) & ~15) * sizeof(float));
if (!mp3_buffer)
{
if (resamp && resampler_data)
resamp->free(resampler_data);
goto error;
}
/* "system" menu sounds may reuse the same voice without freeing anything first, so do that here if needed */
if (voice->types.mp3.resampler && voice->types.mp3.resampler_data)
voice->types.mp3.resampler->free(voice->types.mp3.resampler_data);
if (voice->types.mp3.buffer)
memalign_free(voice->types.mp3.buffer);
voice->types.mp3.resampler = resamp;
voice->types.mp3.resampler_data = resampler_data;
voice->types.mp3.buffer = (float*)mp3_buffer;
voice->types.mp3.buf_samples = samples;
voice->types.mp3.ratio = ratio;
voice->types.mp3.position = 0;
voice->types.mp3.samples = 0;
return true;
error:
drmp3_uninit(&voice->types.mp3.stream);
return false;
}
#endif
core_audio_mixer_voice_t* core_audio_mixer_play(core_audio_mixer_sound_t* sound,
bool repeat, float volume,
const char *resampler_ident,
enum resampler_quality quality,
core_audio_mixer_stop_cb_t stop_cb)
{
unsigned i;
bool res = false;
core_audio_mixer_voice_t* voice = core_s_voices;
if (!sound)
return NULL;
for (i = 0; i < CORE_AUDIO_MIXER_MAX_VOICES; i++, voice++)
{
if (voice->type != CORE_AUDIO_MIXER_TYPE_NONE)
continue;
switch (sound->type)
{
case CORE_AUDIO_MIXER_TYPE_WAV:
res = core_audio_mixer_play_wav(sound, voice, repeat, volume, stop_cb);
break;
case CORE_AUDIO_MIXER_TYPE_OGG:
#ifdef HAVE_STB_VORBIS
res = core_audio_mixer_play_ogg(sound, voice, repeat, volume,
resampler_ident, quality, stop_cb);
#endif
break;
case CORE_AUDIO_MIXER_TYPE_MOD:
#ifdef HAVE_IBXM
res = core_audio_mixer_play_mod(sound, voice, repeat, volume, stop_cb);
#endif
break;
case CORE_AUDIO_MIXER_TYPE_FLAC:
#ifdef HAVE_DR_FLAC
res = core_audio_mixer_play_flac(sound, voice, repeat, volume,
resampler_ident, quality, stop_cb);
#endif
break;
case CORE_AUDIO_MIXER_TYPE_MP3:
#ifdef HAVE_DR_MP3
res = core_audio_mixer_play_mp3(sound, voice, repeat, volume,
resampler_ident, quality, stop_cb);
#endif
break;
case CORE_AUDIO_MIXER_TYPE_NONE:
break;
}
break;
}
if (res)
{
voice->type = sound->type;
voice->repeat = repeat;
voice->volume = volume;
voice->sound = sound;
voice->stop_cb = stop_cb;
}
else
voice = NULL;
return voice;
}
void core_audio_mixer_stop(core_audio_mixer_voice_t* voice)
{
core_audio_mixer_stop_cb_t stop_cb = NULL;
core_audio_mixer_sound_t* sound = NULL;
if (voice)
{
stop_cb = voice->stop_cb;
sound = voice->sound;
voice->type = CORE_AUDIO_MIXER_TYPE_NONE;
if (stop_cb)
stop_cb(sound, CORE_AUDIO_MIXER_SOUND_STOPPED);
}
}
static void core_audio_mixer_mix_wav(float* buffer, size_t num_frames,
core_audio_mixer_voice_t* voice,
float volume)
{
int i;
unsigned buf_free = (unsigned)(num_frames * 2);
const core_audio_mixer_sound_t* sound = voice->sound;
unsigned pcm_available = sound->types.wav.frames
* 2 - voice->types.wav.position;
const float* pcm = sound->types.wav.pcm +
voice->types.wav.position;
again:
if (pcm_available < buf_free)
{
for (i = pcm_available; i != 0; i--)
*buffer++ += *pcm++ * volume;
if (voice->repeat)
{
if (voice->stop_cb)
voice->stop_cb(voice->sound, CORE_AUDIO_MIXER_SOUND_REPEATED);
buf_free -= pcm_available;
pcm_available = sound->types.wav.frames * 2;
pcm = sound->types.wav.pcm;
voice->types.wav.position = 0;
goto again;
}
if (voice->stop_cb)
voice->stop_cb(voice->sound, CORE_AUDIO_MIXER_SOUND_FINISHED);
voice->type = CORE_AUDIO_MIXER_TYPE_NONE;
}
else
{
for (i = buf_free; i != 0; i--)
*buffer++ += *pcm++ * volume;
voice->types.wav.position += buf_free;
}
}
#ifdef HAVE_STB_VORBIS
static void core_audio_mixer_mix_ogg(float* buffer, size_t num_frames,
core_audio_mixer_voice_t* voice,
float volume)
{
int i;
float* temp_buffer = NULL;
unsigned buf_free = (unsigned)(num_frames * 2);
unsigned temp_samples = 0;
float* pcm = NULL;
if (voice->types.ogg.position == voice->types.ogg.samples)
{
again:
if (temp_buffer == NULL)
temp_buffer = (float*)malloc(CORE_AUDIO_MIXER_TEMP_BUFFER * sizeof(float));
temp_samples = stb_vorbis_get_samples_float_interleaved(
voice->types.ogg.stream, 2, temp_buffer,
CORE_AUDIO_MIXER_TEMP_BUFFER) * 2;
if (temp_samples == 0)
{
if (voice->repeat)
{
if (voice->stop_cb)
voice->stop_cb(voice->sound, CORE_AUDIO_MIXER_SOUND_REPEATED);
stb_vorbis_seek_start(voice->types.ogg.stream);
goto again;
}
if (voice->stop_cb)
voice->stop_cb(voice->sound, CORE_AUDIO_MIXER_SOUND_FINISHED);
voice->type = CORE_AUDIO_MIXER_TYPE_NONE;
goto cleanup;
}
if (voice->types.ogg.resampler)
{
struct resampler_data info;
info.data_in = temp_buffer;
info.data_out = voice->types.ogg.buffer;
info.input_frames = temp_samples / 2;
info.output_frames = 0;
info.ratio = voice->types.ogg.ratio;
voice->types.ogg.resampler->process(
voice->types.ogg.resampler_data, &info);
}
else
memcpy(voice->types.ogg.buffer, temp_buffer,
temp_samples * sizeof(float));
voice->types.ogg.position = 0;
voice->types.ogg.samples = voice->types.ogg.buf_samples;
}
pcm = voice->types.ogg.buffer + voice->types.ogg.position;
if (voice->types.ogg.samples < buf_free)
{
for (i = voice->types.ogg.samples; i != 0; i--)
*buffer++ += *pcm++ * volume;
buf_free -= voice->types.ogg.samples;
goto again;
}
for (i = buf_free; i != 0; --i )
*buffer++ += *pcm++ * volume;
voice->types.ogg.position += buf_free;
voice->types.ogg.samples -= buf_free;
cleanup:
if (temp_buffer != NULL)
free(temp_buffer);
}
#endif
#ifdef HAVE_IBXM
static void core_audio_mixer_mix_mod(float* buffer, size_t num_frames,
core_audio_mixer_voice_t* voice,
float volume)
{
int i;
float samplef = 0.0f;
unsigned temp_samples = 0;
unsigned buf_free = (unsigned)(num_frames * 2);
int* pcm = NULL;
if (voice->types.mod.samples == 0)
{
again:
temp_samples = replay_get_audio(
voice->types.mod.stream, voice->types.mod.buffer, 0 ) * 2;
if (temp_samples == 0)
{
if (voice->repeat)
{
if (voice->stop_cb)
voice->stop_cb(voice->sound, CORE_AUDIO_MIXER_SOUND_REPEATED);
replay_seek( voice->types.mod.stream, 0);
goto again;
}
if (voice->stop_cb)
voice->stop_cb(voice->sound, CORE_AUDIO_MIXER_SOUND_FINISHED);
voice->type = CORE_AUDIO_MIXER_TYPE_NONE;
return;
}
voice->types.mod.position = 0;
voice->types.mod.samples = temp_samples;
}
pcm = voice->types.mod.buffer + voice->types.mod.position;
if (voice->types.mod.samples < buf_free)
{
for (i = voice->types.mod.samples; i != 0; i--)
{
samplef = ((float)(*pcm++) + 32768.0f) / 65535.0f;
samplef = samplef * 2.0f - 1.0f;
*buffer++ += samplef * volume;
}
buf_free -= voice->types.mod.samples;
goto again;
}
for (i = buf_free; i != 0; --i )
{
samplef = ((float)(*pcm++) + 32768.0f) / 65535.0f;
samplef = samplef * 2.0f - 1.0f;
*buffer++ += samplef * volume;
}
voice->types.mod.position += buf_free;
voice->types.mod.samples -= buf_free;
}
#endif
#ifdef HAVE_DR_FLAC
static void core_audio_mixer_mix_flac(float* buffer, size_t num_frames,
core_audio_mixer_voice_t* voice,
float volume)
{
int i;
struct resampler_data info;
float temp_buffer[CORE_AUDIO_MIXER_TEMP_BUFFER] = { 0 };
unsigned buf_free = (unsigned)(num_frames * 2);
unsigned temp_samples = 0;
float *pcm = NULL;
if (voice->types.flac.position == voice->types.flac.samples)
{
again:
temp_samples = (unsigned)drflac_read_f32( voice->types.flac.stream, CORE_AUDIO_MIXER_TEMP_BUFFER, temp_buffer);
if (temp_samples == 0)
{
if (voice->repeat)
{
if (voice->stop_cb)
voice->stop_cb(voice->sound, CORE_AUDIO_MIXER_SOUND_REPEATED);
drflac_seek_to_sample(voice->types.flac.stream,0);
goto again;
}
if (voice->stop_cb)
voice->stop_cb(voice->sound, CORE_AUDIO_MIXER_SOUND_FINISHED);
voice->type = CORE_AUDIO_MIXER_TYPE_NONE;
return;
}
info.data_in = temp_buffer;
info.data_out = voice->types.flac.buffer;
info.input_frames = temp_samples / 2;
info.output_frames = 0;
info.ratio = voice->types.flac.ratio;
if (voice->types.flac.resampler)
voice->types.flac.resampler->process(
voice->types.flac.resampler_data, &info);
else
memcpy(voice->types.flac.buffer, temp_buffer, temp_samples * sizeof(float));
voice->types.flac.position = 0;
voice->types.flac.samples = voice->types.flac.buf_samples;
}
pcm = voice->types.flac.buffer + voice->types.flac.position;
if (voice->types.flac.samples < buf_free)
{
for (i = voice->types.flac.samples; i != 0; i--)
*buffer++ += *pcm++ * volume;
buf_free -= voice->types.flac.samples;
goto again;
}
for (i = buf_free; i != 0; --i )
*buffer++ += *pcm++ * volume;
voice->types.flac.position += buf_free;
voice->types.flac.samples -= buf_free;
}
#endif
#ifdef HAVE_DR_MP3
static void core_audio_mixer_mix_mp3(float* buffer, size_t num_frames,
core_audio_mixer_voice_t* voice,
float volume)
{
int i;
struct resampler_data info;
float temp_buffer[CORE_AUDIO_MIXER_TEMP_BUFFER] = { 0 };
unsigned buf_free = (unsigned)(num_frames * 2);
unsigned temp_samples = 0;
float* pcm = NULL;
if (voice->types.mp3.position == voice->types.mp3.samples)
{
again:
temp_samples = (unsigned)drmp3_read_f32(
&voice->types.mp3.stream,
CORE_AUDIO_MIXER_TEMP_BUFFER / 2, temp_buffer) * 2;
if (temp_samples == 0)
{
if (voice->repeat)
{
if (voice->stop_cb)
voice->stop_cb(voice->sound, CORE_AUDIO_MIXER_SOUND_REPEATED);
drmp3_seek_to_frame(&voice->types.mp3.stream,0);
goto again;
}
if (voice->stop_cb)
voice->stop_cb(voice->sound, CORE_AUDIO_MIXER_SOUND_FINISHED);
voice->type = CORE_AUDIO_MIXER_TYPE_NONE;
return;
}
info.data_in = temp_buffer;
info.data_out = voice->types.mp3.buffer;
info.input_frames = temp_samples / 2;
info.output_frames = 0;
info.ratio = voice->types.mp3.ratio;
if (voice->types.mp3.resampler)
voice->types.mp3.resampler->process(
voice->types.mp3.resampler_data, &info);
else
memcpy(voice->types.mp3.buffer, temp_buffer,
temp_samples * sizeof(float));
voice->types.mp3.position = 0;
voice->types.mp3.samples = voice->types.mp3.buf_samples;
}
pcm = voice->types.mp3.buffer + voice->types.mp3.position;
if (voice->types.mp3.samples < buf_free)
{
for (i = voice->types.mp3.samples; i != 0; i--)
*buffer++ += *pcm++ * volume;
buf_free -= voice->types.mp3.samples;
goto again;
}
for (i = buf_free; i != 0; --i )
*buffer++ += *pcm++ * volume;
voice->types.mp3.position += buf_free;
voice->types.mp3.samples -= buf_free;
}
#endif
void core_audio_mixer_mix(float* buffer, size_t num_frames,
float volume_override, bool override)
{
unsigned i;
size_t j = 0;
float* sample = NULL;
core_audio_mixer_voice_t* voice = core_s_voices;
for (i = 0; i < CORE_AUDIO_MIXER_MAX_VOICES; i++, voice++)
{
float volume = (override) ? volume_override : voice->volume;
switch (voice->type)
{
case CORE_AUDIO_MIXER_TYPE_WAV:
core_audio_mixer_mix_wav(buffer, num_frames, voice, volume);
break;
case CORE_AUDIO_MIXER_TYPE_OGG:
#ifdef HAVE_STB_VORBIS
core_audio_mixer_mix_ogg(buffer, num_frames, voice, volume);
#endif
break;
case CORE_AUDIO_MIXER_TYPE_MOD:
#ifdef HAVE_IBXM
core_audio_mixer_mix_mod(buffer, num_frames, voice, volume);
#endif
break;
case CORE_AUDIO_MIXER_TYPE_FLAC:
#ifdef HAVE_DR_FLAC
core_audio_mixer_mix_flac(buffer, num_frames, voice, volume);
#endif
break;
case CORE_AUDIO_MIXER_TYPE_MP3:
#ifdef HAVE_DR_MP3
core_audio_mixer_mix_mp3(buffer, num_frames, voice, volume);
#endif
break;
case CORE_AUDIO_MIXER_TYPE_NONE:
break;
}
}
for (j = 0, sample = buffer; j < num_frames * 2; j++, sample++)
{
if (*sample < -1.0f)
*sample = -1.0f;
else if (*sample > 1.0f)
*sample = 1.0f;
}
}
float core_audio_mixer_voice_get_volume(core_audio_mixer_voice_t *voice)
{
if (!voice)
return 0.0f;
return voice->volume;
}
void core_audio_mixer_voice_set_volume(core_audio_mixer_voice_t *voice, float val)
{
if (!voice)
return;
voice->volume = val;
}
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