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Update libretro-common

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This commit is contained in:
twinaphex 2018-04-27 23:12:24 +02:00
parent b693d76ebb
commit 1f2ef858ab
3 changed files with 199 additions and 6 deletions
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199
libretro-common/audio/audio_mixer.c
View File

@ -47,12 +47,17 @@
#include <stb/stb_vorbis.h>
#endif
#ifdef HAVE_DR_FLAC
#define DR_FLAC_IMPLEMENTATION
#include <dr/dr_flac.h>
#endif
#ifdef HAVE_IBXM
#include <ibxm/ibxm.h>
#endif
#define AUDIO_MIXER_MAX_VOICES 8
#define AUDIO_MIXER_TEMP_OGG_BUFFER 8192
#define AUDIO_MIXER_TEMP_BUFFER 8192
struct audio_mixer_sound
{
@ -75,6 +80,15 @@ struct audio_mixer_sound
const void* data;
} ogg;
#endif
#ifdef HAVE_DR_FLAC
struct
{
/* flac */
unsigned size;
const void* data;
} flac;
#endif
#ifdef HAVE_IBXM
struct
@ -116,6 +130,20 @@ struct audio_mixer_voice
} ogg;
#endif
#ifdef HAVE_DR_FLAC
struct
{
unsigned position;
unsigned samples;
unsigned buf_samples;
float* buffer;
float ratio;
drflac *stream;
void *resampler_data;
const retro_resampler_t *resampler;
} flac;
#endif
#ifdef HAVE_IBXM
struct
{
@ -343,6 +371,25 @@ audio_mixer_sound_t* audio_mixer_load_ogg(void *buffer, int32_t size)
#endif
}
audio_mixer_sound_t* audio_mixer_load_flac(void *buffer, int32_t size)
{
#ifdef HAVE_DR_FLAC
audio_mixer_sound_t* sound = (audio_mixer_sound_t*)calloc(1, sizeof(*sound));
if (!sound)
return NULL;
sound->type = AUDIO_MIXER_TYPE_FLAC;
sound->types.flac.size = size;
sound->types.flac.data = buffer;
return sound;
#else
return NULL;
#endif
}
audio_mixer_sound_t* audio_mixer_load_mod(void *buffer, int32_t size)
{
#ifdef HAVE_IBXM
@ -386,6 +433,13 @@ void audio_mixer_destroy(audio_mixer_sound_t* sound)
handle = (void*)sound->types.mod.data;
if (handle)
free(handle);
#endif
break;
case AUDIO_MIXER_TYPE_FLAC:
#ifdef HAVE_DR_FLAC
handle = (void*)sound->types.flac.data;
if (handle)
free(handle);
#endif
break;
case AUDIO_MIXER_TYPE_NONE:
@ -436,7 +490,7 @@ static bool audio_mixer_play_ogg(
goto error;
}
samples = (unsigned)(AUDIO_MIXER_TEMP_OGG_BUFFER * ratio);
samples = (unsigned)(AUDIO_MIXER_TEMP_BUFFER * ratio);
ogg_buffer = (float*)memalign_alloc(16,
((samples + 15) & ~15) * sizeof(float));
@ -531,6 +585,61 @@ error:
}
#endif
#ifdef HAV_DR_FLAC
static bool audio_mixer_play_flac(
audio_mixer_sound_t* sound,
audio_mixer_voice_t* voice,
bool repeat, float volume,
audio_mixer_stop_cb_t stop_cb)
{
int res = 0;
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 != s_rate)
{
ratio = (double)s_rate / (double)(dr_flac->sampleRate);
if (!retro_resampler_realloc(&resampler_data,
&resamp, NULL, RESAMPLER_QUALITY_DONTCARE,
ratio))
goto error;
}
samples = (unsigned)(AUDIO_MIXER_TEMP_BUFFER * ratio);
flac_buffer = (float*)memalign_alloc(16,
((samples + 15) & ~15) * sizeof(float));
if (!flac_buffer)
{
resamp->free(resampler_data);
goto error;
}
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
audio_mixer_voice_t* audio_mixer_play(audio_mixer_sound_t* sound, bool repeat,
float volume, audio_mixer_stop_cb_t stop_cb)
{
@ -563,6 +672,11 @@ audio_mixer_voice_t* audio_mixer_play(audio_mixer_sound_t* sound, bool repeat,
case AUDIO_MIXER_TYPE_MOD:
#ifdef HAVE_IBXM
res = audio_mixer_play_mod(sound, voice, repeat, volume, stop_cb);
#endif
break;
case AUDIO_MIXER_TYPE_FLAC:
#ifdef HAVE_DR_FLAC
res = audio_mixer_play_flac(sound, voice, repeat, volume, stop_cb);
#endif
break;
case AUDIO_MIXER_TYPE_NONE:
@ -666,7 +780,7 @@ static void audio_mixer_mix_ogg(float* buffer, size_t num_frames,
{
int i;
struct resampler_data info = { 0 };
float temp_buffer[AUDIO_MIXER_TEMP_OGG_BUFFER] = { 0 };
float temp_buffer[AUDIO_MIXER_TEMP_BUFFER] = { 0 };
unsigned buf_free = (unsigned)(num_frames * 2);
unsigned temp_samples = 0;
float* pcm = NULL;
@ -676,7 +790,7 @@ static void audio_mixer_mix_ogg(float* buffer, size_t num_frames,
again:
temp_samples = stb_vorbis_get_samples_float_interleaved(
voice->types.ogg.stream, 2, temp_buffer,
AUDIO_MIXER_TEMP_OGG_BUFFER) * 2;
AUDIO_MIXER_TEMP_BUFFER) * 2;
if (temp_samples == 0)
{
@ -809,6 +923,78 @@ again:
}
#endif
#ifdef HAVE_DR_FLAC
static void audio_mixer_mix_flac(float* buffer, size_t num_frames,
audio_mixer_voice_t* voice,
float volume)
{
int i;
struct resampler_data info = { 0 };
float temp_buffer[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 = drflac_read_f32( voice->types.flac.stream, AUDIO_MIXER_TEMP_BUFFER, temp_buffer);
if (temp_samples == 0)
{
if (voice->repeat)
{
if (voice->stop_cb)
voice->stop_cb(voice->sound, AUDIO_MIXER_SOUND_REPEATED);
drflac_seek_to_sample(voice->types.flac.stream,0);
goto again;
}
else
{
if (voice->stop_cb)
voice->stop_cb(voice->sound, AUDIO_MIXER_SOUND_FINISHED);
voice->type = 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.ogg.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;
}
else
{
int i;
for (i = buf_free; i != 0; --i )
*buffer++ += *pcm++ * volume;
voice->types.flac.position += buf_free;
voice->types.flac.samples -= buf_free;
}
}
#endif
void audio_mixer_mix(float* buffer, size_t num_frames, float volume_override, bool override)
{
unsigned i;
@ -837,6 +1023,11 @@ void audio_mixer_mix(float* buffer, size_t num_frames, float volume_override, bo
case AUDIO_MIXER_TYPE_MOD:
#ifdef HAVE_IBXM
audio_mixer_mix_mod(buffer, num_frames, voice, volume);
#endif
break;
case AUDIO_MIXER_TYPE_FLAC:
#ifdef HAVE_DR_FLAC
audio_mixer_mix_flac(buffer, num_frames, voice, volume);
#endif
break;
case AUDIO_MIXER_TYPE_NONE:

2
libretro-common/file/file_path.c
View File

@ -392,7 +392,7 @@ char *path_remove_extension(char *path)
return NULL;
if (*last)
*last = '\0';
return last;
return path;
}
/**

4
libretro-common/include/audio/audio_mixer.h
View File

@ -41,7 +41,8 @@ enum audio_mixer_type
AUDIO_MIXER_TYPE_NONE = 0,
AUDIO_MIXER_TYPE_WAV,
AUDIO_MIXER_TYPE_OGG,
AUDIO_MIXER_TYPE_MOD
AUDIO_MIXER_TYPE_MOD,
AUDIO_MIXER_TYPE_FLAC
};
typedef struct audio_mixer_sound audio_mixer_sound_t;
@ -61,6 +62,7 @@ void audio_mixer_done(void);
audio_mixer_sound_t* audio_mixer_load_wav(void *buffer, int32_t size);
audio_mixer_sound_t* audio_mixer_load_ogg(void *buffer, int32_t size);
audio_mixer_sound_t* audio_mixer_load_mod(void *buffer, int32_t size);
audio_mixer_sound_t* audio_mixer_load_flac(void *buffer, int32_t size);
void audio_mixer_destroy(audio_mixer_sound_t* sound);