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Some cleanups for audio mixer

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This commit is contained in:
twinaphex 2017-02-23 04:00:40 +01:00
parent c8dfdaa5ff
commit 4e0c24acbe
3 changed files with 115 additions and 90 deletions
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110
audio/audio_mixer.c
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@ -38,13 +38,12 @@
#define __forceinline __inline__ __attribute__((__always_inline__,__gnu_inline__)) #define __forceinline __inline__ __attribute__((__always_inline__,__gnu_inline__))
#endif #endif
#include "stb_vorbis.c" #include "../deps/stb/stb_vorbis.h"
/*---------------------------------------------------------------------------*/ /*---------------------------------------------------------------------------*/
#define AUDIO_MIXER_MAX_VOICES 8 #define AUDIO_MIXER_MAX_VOICES 8
#define AUDIO_MIXER_TEMP_OGG_BUFFER 8192 #define AUDIO_MIXER_TEMP_OGG_BUFFER 8192
#define AUDIO_MIXER_TYPE_NONE 0 #define AUDIO_MIXER_TYPE_NONE 0
#define AUDIO_MIXER_TYPE_WAV 1 #define AUDIO_MIXER_TYPE_WAV 1
#define AUDIO_MIXER_TYPE_OGG 2 #define AUDIO_MIXER_TYPE_OGG 2
@ -102,20 +101,21 @@ struct audio_mixer_voice_t
} types; } types;
}; };
static unsigned s_rate;
static audio_mixer_voice_t s_voices[AUDIO_MIXER_MAX_VOICES]; static audio_mixer_voice_t s_voices[AUDIO_MIXER_MAX_VOICES];
static unsigned s_rate = 0;
static bool wav2float(const rwav_t* wav, float** pcm, size_t* samples_out) static bool wav2float(const rwav_t* wav, float** pcm, size_t* samples_out)
{ {
float* f;
float sample;
const uint8_t* u8;
const int16_t* s16;
size_t i; size_t i;
float sample = 0.0f;
const uint8_t* u8 = NULL;
const int16_t* s16 = NULL;
float* f = NULL;
/* Allocate on a 16-byte boundary, and pad to a multiple of 16 bytes */ /* Allocate on a 16-byte boundary, and pad to a multiple of 16 bytes */
*samples_out = wav->numsamples * 2; *samples_out = wav->numsamples * 2;
f = (float*)memalign_alloc(16, ((*samples_out + 15) & ~15) * sizeof(float)); f = (float*)memalign_alloc(16,
((*samples_out + 15) & ~15) * sizeof(float));
if (f == NULL) if (f == NULL)
return false; return false;
@ -184,7 +184,8 @@ static bool wav2float(const rwav_t* wav, float** pcm, size_t* samples_out)
return true; return true;
} }
static bool one_shot_resample(const float* in, size_t samples_in, unsigned rate, float** out, size_t* samples_out) static bool one_shot_resample(const float* in, size_t samples_in,
unsigned rate, float** out, size_t* samples_out)
{ {
void* data = NULL; void* data = NULL;
const retro_resampler_t* resampler = NULL; const retro_resampler_t* resampler = NULL;
@ -196,7 +197,8 @@ static bool one_shot_resample(const float* in, size_t samples_in, unsigned rate,
/* Allocate on a 16-byte boundary, and pad to a multiple of 16 bytes */ /* Allocate on a 16-byte boundary, and pad to a multiple of 16 bytes */
*samples_out = samples_in * ratio; *samples_out = samples_in * ratio;
*out = (float*)memalign_alloc(16, ((*samples_out + 15) & ~15) * sizeof(float)); *out = (float*)memalign_alloc(16,
((*samples_out + 15) & ~15) * sizeof(float));
if (*out == NULL) if (*out == NULL)
return false; return false;
@ -231,17 +233,17 @@ void audio_mixer_done(void)
audio_mixer_sound_t* audio_mixer_load_wav(const char* path) audio_mixer_sound_t* audio_mixer_load_wav(const char* path)
{ {
/* Raw WAV bytes */
void* buffer;
ssize_t size;
/* WAV data */ /* WAV data */
rwav_t wav; rwav_t wav;
/* Raw WAV bytes */
void* buffer = NULL;
ssize_t size = 0;
/* WAV samples converted to float */ /* WAV samples converted to float */
float* pcm; float* pcm = NULL;
float* resampled; float* resampled = NULL;
size_t samples; size_t samples = 0;
/* Result */ /* Result */
audio_mixer_sound_t* sound; audio_mixer_sound_t* sound = NULL;
if (filestream_read_file(path, &buffer, &size) == 0) if (filestream_read_file(path, &buffer, &size) == 0)
return NULL; return NULL;
@ -259,7 +261,8 @@ audio_mixer_sound_t* audio_mixer_load_wav(const char* path)
if (wav.samplerate != s_rate) if (wav.samplerate != s_rate)
{ {
if (!one_shot_resample(pcm, samples, wav.samplerate, &resampled, &samples)) if (!one_shot_resample(pcm, samples,
wav.samplerate, &resampled, &samples))
return NULL; return NULL;
memalign_free((void*)pcm); memalign_free((void*)pcm);
@ -284,9 +287,9 @@ audio_mixer_sound_t* audio_mixer_load_wav(const char* path)
audio_mixer_sound_t* audio_mixer_load_ogg(const char* path) audio_mixer_sound_t* audio_mixer_load_ogg(const char* path)
{ {
void* buffer;
ssize_t size; ssize_t size;
audio_mixer_sound_t* sound; void* buffer = NULL;
audio_mixer_sound_t* sound = NULL;
if (filestream_read_file(path, &buffer, &size) == 0) if (filestream_read_file(path, &buffer, &size) == 0)
return NULL; return NULL;
@ -316,7 +319,9 @@ void audio_mixer_destroy(audio_mixer_sound_t* sound)
free(sound); free(sound);
} }
static bool audio_mixer_play_wav(audio_mixer_sound_t* sound, audio_mixer_voice_t* voice, bool repeat, float volume, audio_mixer_stop_cb_t stop_cb) static bool audio_mixer_play_wav(audio_mixer_sound_t* sound,
audio_mixer_voice_t* voice, bool repeat, float volume,
audio_mixer_stop_cb_t stop_cb)
{ {
voice->type = AUDIO_MIXER_TYPE_WAV; voice->type = AUDIO_MIXER_TYPE_WAV;
voice->repeat = repeat; voice->repeat = repeat;
@ -328,28 +333,34 @@ static bool audio_mixer_play_wav(audio_mixer_sound_t* sound, audio_mixer_voice_t
return true; return true;
} }
static bool audio_mixer_play_ogg(audio_mixer_sound_t* sound, audio_mixer_voice_t* voice, bool repeat, float volume, audio_mixer_stop_cb_t stop_cb) static bool audio_mixer_play_ogg(
audio_mixer_sound_t* sound,
audio_mixer_voice_t* voice,
bool repeat, float volume,
audio_mixer_stop_cb_t stop_cb)
{ {
int res;
stb_vorbis_info info; stb_vorbis_info info;
float ratio; int res = 0;
unsigned samples; float ratio = 0.0f;
unsigned samples = 0;
voice->repeat = repeat; voice->repeat = repeat;
voice->volume = volume; voice->volume = volume;
voice->sound = sound; voice->sound = sound;
voice->stop_cb = stop_cb; voice->stop_cb = stop_cb;
voice->types.ogg.stream = stb_vorbis_open_memory((const unsigned char*)sound->types.ogg.data, sound->types.ogg.size, &res, NULL); voice->types.ogg.stream = stb_vorbis_open_memory(
(const unsigned char*)sound->types.ogg.data,
sound->types.ogg.size, &res, NULL);
if (voice->types.ogg.stream == NULL) if (voice->types.ogg.stream == NULL)
return false; return false;
info = stb_vorbis_get_info(voice->types.ogg.stream); info = stb_vorbis_get_info(voice->types.ogg.stream);
/* Only stereo supported for now */
if (info.channels != 2) if (info.channels != 2)
{ {
/* Only stereo supported for now */
stb_vorbis_close(voice->types.ogg.stream); stb_vorbis_close(voice->types.ogg.stream);
return false; return false;
} }
@ -358,15 +369,18 @@ static bool audio_mixer_play_ogg(audio_mixer_sound_t* sound, audio_mixer_voice_t
{ {
voice->types.ogg.ratio = ratio = (double)s_rate / (double)info.sample_rate; voice->types.ogg.ratio = ratio = (double)s_rate / (double)info.sample_rate;
if (!retro_resampler_realloc(&voice->types.ogg.resampler_data, &voice->types.ogg.resampler, NULL, ratio)) if (!retro_resampler_realloc(&voice->types.ogg.resampler_data,
&voice->types.ogg.resampler, NULL, ratio))
{ {
stb_vorbis_close(voice->types.ogg.stream); stb_vorbis_close(voice->types.ogg.stream);
return false; return false;
} }
} }
samples = voice->types.ogg.buf_samples = (unsigned)(AUDIO_MIXER_TEMP_OGG_BUFFER * ratio); samples =
voice->types.ogg.buffer = (float*)memalign_alloc(16, ((samples + 15) & ~15) * sizeof(float)); voice->types.ogg.buf_samples = (unsigned)(AUDIO_MIXER_TEMP_OGG_BUFFER * ratio);
voice->types.ogg.buffer = (float*)memalign_alloc(16,
((samples + 15) & ~15) * sizeof(float));
if (voice->types.ogg.buffer == NULL) if (voice->types.ogg.buffer == NULL)
{ {
@ -380,10 +394,11 @@ static bool audio_mixer_play_ogg(audio_mixer_sound_t* sound, audio_mixer_voice_t
return true; return true;
} }
audio_mixer_voice_t* audio_mixer_play(audio_mixer_sound_t* sound, bool repeat, float volume, audio_mixer_stop_cb_t stop_cb) audio_mixer_voice_t* audio_mixer_play(audio_mixer_sound_t* sound, bool repeat,
float volume, audio_mixer_stop_cb_t stop_cb)
{ {
unsigned i; unsigned i;
audio_mixer_voice_t* voice; audio_mixer_voice_t* voice = NULL;
bool res = false; bool res = false;
for (i = 0, voice = s_voices; i < AUDIO_MIXER_MAX_VOICES; i++, voice++) for (i = 0, voice = s_voices; i < AUDIO_MIXER_MAX_VOICES; i++, voice++)
@ -399,7 +414,9 @@ audio_mixer_voice_t* audio_mixer_play(audio_mixer_sound_t* sound, bool repeat, f
} }
} }
return res ? voice : NULL; if (res)
return voice;
return NULL;
} }
void audio_mixer_stop(audio_mixer_voice_t* voice) void audio_mixer_stop(audio_mixer_voice_t* voice)
@ -409,12 +426,13 @@ void audio_mixer_stop(audio_mixer_voice_t* voice)
static void mix_wav(float* buffer, size_t num_frames, audio_mixer_voice_t* voice) static void mix_wav(float* buffer, size_t num_frames, audio_mixer_voice_t* voice)
{ {
int i;
unsigned buf_free = num_frames * 2; unsigned buf_free = num_frames * 2;
const audio_mixer_sound_t* sound = voice->sound; const audio_mixer_sound_t* sound = voice->sound;
unsigned pcm_available = sound->types.wav.frames * 2 - voice->types.wav.position; unsigned pcm_available = sound->types.wav.frames
* 2 - voice->types.wav.position;
const float* pcm = sound->types.wav.pcm + voice->types.wav.position; const float* pcm = sound->types.wav.pcm + voice->types.wav.position;
float volume = voice->volume; float volume = voice->volume;
int i;
again: again:
if (pcm_available < buf_free) if (pcm_available < buf_free)
@ -450,19 +468,21 @@ again:
static void mix_ogg(float* buffer, size_t num_frames, audio_mixer_voice_t* voice) static void mix_ogg(float* buffer, size_t num_frames, audio_mixer_voice_t* voice)
{ {
unsigned buf_free = num_frames * 2; int i;
const audio_mixer_sound_t* sound = voice->sound;
float temp_buffer[AUDIO_MIXER_TEMP_OGG_BUFFER]; float temp_buffer[AUDIO_MIXER_TEMP_OGG_BUFFER];
unsigned temp_samples; unsigned buf_free = num_frames * 2;
unsigned temp_samples = 0;
float volume = voice->volume; float volume = voice->volume;
struct resampler_data info = {0}; struct resampler_data info = {0};
float* pcm; float* pcm = NULL;
int i; const audio_mixer_sound_t* sound = voice->sound;
if (voice->types.ogg.position == voice->types.ogg.samples) if (voice->types.ogg.position == voice->types.ogg.samples)
{ {
again: again:
temp_samples = stb_vorbis_get_samples_float_interleaved(voice->types.ogg.stream, 2, temp_buffer, AUDIO_MIXER_TEMP_OGG_BUFFER) * 2; temp_samples = stb_vorbis_get_samples_float_interleaved(
voice->types.ogg.stream, 2, temp_buffer,
AUDIO_MIXER_TEMP_OGG_BUFFER) * 2;
if (temp_samples == 0) if (temp_samples == 0)
{ {
@ -517,9 +537,9 @@ static void mix_ogg(float* buffer, size_t num_frames, audio_mixer_voice_t* voice
void audio_mixer_mix(float* buffer, size_t num_frames) void audio_mixer_mix(float* buffer, size_t num_frames)
{ {
unsigned i; unsigned i;
audio_mixer_voice_t* voice; size_t j = 0;
size_t j; float* sample = NULL;
float* sample; audio_mixer_voice_t* voice = NULL;
for (i = 0, voice = s_voices; i < AUDIO_MIXER_MAX_VOICES; i++, voice++) for (i = 0, voice = s_voices; i < AUDIO_MIXER_MAX_VOICES; i++, voice++)
{ {

4
audio/audio_mixer.h
View File

@ -16,9 +16,11 @@
#ifndef __AUDIO_MIXER__H #ifndef __AUDIO_MIXER__H
#define __AUDIO_MIXER__H #define __AUDIO_MIXER__H
#include <stdint.h>
#include <stddef.h>
#include <boolean.h> #include <boolean.h>
#include <retro_common_api.h> #include <retro_common_api.h>
#include <stdint.h>
RETRO_BEGIN_DECLS RETRO_BEGIN_DECLS

11
audio/stb_vorbis.c → deps/stb/stb_vorbis.h vendored
View File

@ -3935,8 +3935,10 @@ static int start_decoder(vorb *f)
for (j=0; j < g->values; ++j) for (j=0; j < g->values; ++j)
g->sorted_order[j] = (uint8) p[j].y; g->sorted_order[j] = (uint8) p[j].y;
// precompute the neighbors // precompute the neighbors
for (j=2; j < g->values; ++j) { for (j=2; j < g->values; ++j)
int low,hi; {
int low = 0;
int hi = 0;
neighbors(g->Xlist, j, &low,&hi); neighbors(g->Xlist, j, &low,&hi);
g->neighbors[j][0] = low; g->neighbors[j][0] = low;
g->neighbors[j][1] = hi; g->neighbors[j][1] = hi;
@ -4536,11 +4538,12 @@ static uint32 vorbis_find_page(stb_vorbis *f, uint32 *end, uint32 *last)
// doing needless I/O would be crazy! // doing needless I/O would be crazy!
static int vorbis_analyze_page(stb_vorbis *f, ProbedPage *z) static int vorbis_analyze_page(stb_vorbis *f, ProbedPage *z)
{ {
uint8 header[27], lacing[255]; uint8 lacing[255];
uint8 packet_type[255]; uint8 packet_type[255];
int num_packet, packet_start; int num_packet, packet_start;
int i,len; int i,len;
uint32 samples; uint32 samples;
uint8 header[27] = {0};
// record where the page starts // record where the page starts
z->page_start = stb_vorbis_get_file_offset(f); z->page_start = stb_vorbis_get_file_offset(f);
@ -5175,7 +5178,7 @@ static void convert_samples_short(int buf_c, short **buffer, int b_offset, int d
int stb_vorbis_get_frame_short(stb_vorbis *f, int num_c, short **buffer, int num_samples) int stb_vorbis_get_frame_short(stb_vorbis *f, int num_c, short **buffer, int num_samples)
{ {
float **output; float **output = {NULL};
int len = stb_vorbis_get_frame_float(f, NULL, &output); int len = stb_vorbis_get_frame_float(f, NULL, &output);
if (len > num_samples) len = num_samples; if (len > num_samples) len = num_samples;
if (len) if (len)