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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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188
audio/audio_mixer.c
View File

@ -38,16 +38,15 @@
#define __forceinline __inline__ __attribute__((__always_inline__,__gnu_inline__))
#endif
#include "stb_vorbis.c"
#include "../deps/stb/stb_vorbis.h"
/*---------------------------------------------------------------------------*/
#define AUDIO_MIXER_MAX_VOICES 8
#define AUDIO_MIXER_TEMP_OGG_BUFFER 8192
#define AUDIO_MIXER_TYPE_NONE 0
#define AUDIO_MIXER_TYPE_WAV 1
#define AUDIO_MIXER_TYPE_OGG 2
#define AUDIO_MIXER_TYPE_NONE 0
#define AUDIO_MIXER_TYPE_WAV 1
#define AUDIO_MIXER_TYPE_OGG 2
struct audio_mixer_sound_t
{
@ -102,20 +101,21 @@ struct audio_mixer_voice_t
} types;
};
static unsigned s_rate;
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)
{
float* f;
float sample;
const uint8_t* u8;
const int16_t* s16;
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 */
*samples_out = wav->numsamples * 2;
f = (float*)memalign_alloc(16, ((*samples_out + 15) & ~15) * sizeof(float));
*samples_out = wav->numsamples * 2;
f = (float*)memalign_alloc(16,
((*samples_out + 15) & ~15) * sizeof(float));
if (f == NULL)
return false;
@ -132,8 +132,8 @@ static bool wav2float(const rwav_t* wav, float** pcm, size_t* samples_out)
{
sample = (float)*u8++ / 255.0f;
sample = sample * 2.0f - 1.0f;
*f++ = sample;
*f++ = sample;
*f++ = sample;
*f++ = sample;
}
}
else
@ -144,8 +144,8 @@ static bool wav2float(const rwav_t* wav, float** pcm, size_t* samples_out)
{
sample = (float)((int)*s16++ + 32768) / 65535.0f;
sample = sample * 2.0f - 1.0f;
*f++ = sample;
*f++ = sample;
*f++ = sample;
*f++ = sample;
}
}
}
@ -159,10 +159,10 @@ static bool wav2float(const rwav_t* wav, float** pcm, size_t* samples_out)
{
sample = (float)*u8++ / 255.0f;
sample = sample * 2.0f - 1.0f;
*f++ = sample;
*f++ = sample;
sample = (float)*u8++ / 255.0f;
sample = sample * 2.0f - 1.0f;
*f++ = sample;
*f++ = sample;
}
}
else
@ -173,10 +173,10 @@ static bool wav2float(const rwav_t* wav, float** pcm, size_t* samples_out)
{
sample = (float)((int)*s16++ + 32768) / 65535.0f;
sample = sample * 2.0f - 1.0f;
*f++ = sample;
*f++ = sample;
sample = (float)((int)*s16++ + 32768) / 65535.0f;
sample = sample * 2.0f - 1.0f;
*f++ = sample;
*f++ = sample;
}
}
}
@ -184,27 +184,29 @@ static bool wav2float(const rwav_t* wav, float** pcm, size_t* samples_out)
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;
struct resampler_data info = {0};
float ratio = (double)s_rate / (double)rate;
struct resampler_data info = {0};
float ratio = (double)s_rate / (double)rate;
if (!retro_resampler_realloc(&data, &resampler, NULL, ratio))
return false;
/* Allocate on a 16-byte boundary, and pad to a multiple of 16 bytes */
*samples_out = samples_in * ratio;
*out = (float*)memalign_alloc(16, ((*samples_out + 15) & ~15) * sizeof(float));
*samples_out = samples_in * ratio;
*out = (float*)memalign_alloc(16,
((*samples_out + 15) & ~15) * sizeof(float));
if (*out == NULL)
return false;
info.data_in = in;
info.data_out = *out;
info.input_frames = samples_in / 2;
info.ratio = ratio;
info.data_in = in;
info.data_out = *out;
info.input_frames = samples_in / 2;
info.ratio = ratio;
resampler->process(data, &info);
resampler->free(data);
@ -231,17 +233,17 @@ void audio_mixer_done(void)
audio_mixer_sound_t* audio_mixer_load_wav(const char* path)
{
/* Raw WAV bytes */
void* buffer;
ssize_t size;
/* WAV data */
rwav_t wav;
/* Raw WAV bytes */
void* buffer = NULL;
ssize_t size = 0;
/* WAV samples converted to float */
float* pcm;
float* resampled;
size_t samples;
float* pcm = NULL;
float* resampled = NULL;
size_t samples = 0;
/* Result */
audio_mixer_sound_t* sound;
audio_mixer_sound_t* sound = NULL;
if (filestream_read_file(path, &buffer, &size) == 0)
return NULL;
@ -259,7 +261,8 @@ audio_mixer_sound_t* audio_mixer_load_wav(const char* path)
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;
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)
{
void* buffer;
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)
return NULL;
@ -316,40 +319,48 @@ void audio_mixer_destroy(audio_mixer_sound_t* 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->repeat = repeat;
voice->volume = volume;
voice->sound = sound;
voice->stop_cb = stop_cb;
voice->type = AUDIO_MIXER_TYPE_WAV;
voice->repeat = repeat;
voice->volume = volume;
voice->sound = sound;
voice->stop_cb = stop_cb;
voice->types.wav.position = 0;
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;
float ratio;
unsigned samples;
int res = 0;
float ratio = 0.0f;
unsigned samples = 0;
voice->repeat = repeat;
voice->volume = volume;
voice->sound = sound;
voice->stop_cb = stop_cb;
voice->repeat = repeat;
voice->volume = volume;
voice->sound = sound;
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)
return false;
info = stb_vorbis_get_info(voice->types.ogg.stream);
/* Only stereo supported for now */
if (info.channels != 2)
{
/* Only stereo supported for now */
stb_vorbis_close(voice->types.ogg.stream);
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;
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);
return false;
}
}
samples = 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));
samples =
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)
{
@ -380,11 +394,12 @@ static bool audio_mixer_play_ogg(audio_mixer_sound_t* sound, audio_mixer_voice_t
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;
audio_mixer_voice_t* voice;
bool res = false;
audio_mixer_voice_t* voice = NULL;
bool res = false;
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)
@ -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)
{
unsigned buf_free = num_frames * 2;
const 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;
float volume = voice->volume;
int i;
unsigned buf_free = num_frames * 2;
const 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;
float volume = voice->volume;
again:
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)
{
unsigned buf_free = num_frames * 2;
const audio_mixer_sound_t* sound = voice->sound;
float temp_buffer[AUDIO_MIXER_TEMP_OGG_BUFFER];
unsigned temp_samples;
float volume = voice->volume;
struct resampler_data info = {0};
float* pcm;
int i;
float temp_buffer[AUDIO_MIXER_TEMP_OGG_BUFFER];
unsigned buf_free = num_frames * 2;
unsigned temp_samples = 0;
float volume = voice->volume;
struct resampler_data info = {0};
float* pcm = NULL;
const audio_mixer_sound_t* sound = voice->sound;
if (voice->types.ogg.position == voice->types.ogg.samples)
{
again:
temp_samples = stb_vorbis_get_samples_float_interleaved(voice->types.ogg.stream, 2, temp_buffer, AUDIO_MIXER_TEMP_OGG_BUFFER) * 2;
again:
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)
{
@ -484,14 +504,14 @@ static void mix_ogg(float* buffer, size_t num_frames, audio_mixer_voice_t* voice
}
}
info.data_in = temp_buffer;
info.data_out = voice->types.ogg.buffer;
info.data_in = temp_buffer;
info.data_out = voice->types.ogg.buffer;
info.input_frames = temp_samples / 2;
info.ratio = voice->types.ogg.ratio;
info.ratio = voice->types.ogg.ratio;
voice->types.ogg.resampler->process(voice->types.ogg.resampler_data, &info);
voice->types.ogg.position = 0;
voice->types.ogg.samples = voice->types.ogg.buf_samples;
voice->types.ogg.samples = voice->types.ogg.buf_samples;
}
pcm = voice->types.ogg.buffer + voice->types.ogg.position;
@ -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)
{
unsigned i;
audio_mixer_voice_t* voice;
size_t j;
float* sample;
size_t j = 0;
float* sample = NULL;
audio_mixer_voice_t* voice = NULL;
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
#define __AUDIO_MIXER__H
#include <stdint.h>
#include <stddef.h>
#include <boolean.h>
#include <retro_common_api.h>
#include <stdint.h>
RETRO_BEGIN_DECLS

13
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)
g->sorted_order[j] = (uint8) p[j].y;
// precompute the neighbors
for (j=2; j < g->values; ++j) {
int low,hi;
for (j=2; j < g->values; ++j)
{
int low = 0;
int hi = 0;
neighbors(g->Xlist, j, &low,&hi);
g->neighbors[j][0] = low;
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!
static int vorbis_analyze_page(stb_vorbis *f, ProbedPage *z)
{
uint8 header[27], lacing[255];
uint8 lacing[255];
uint8 packet_type[255];
int num_packet, packet_start;
int i,len;
uint32 samples;
uint8 header[27] = {0};
// record where the page starts
z->page_start = stb_vorbis_get_file_offset(f);
@ -5175,8 +5178,8 @@ 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)
{
float **output;
int len = stb_vorbis_get_frame_float(f, NULL, &output);
float **output = {NULL};
int len = stb_vorbis_get_frame_float(f, NULL, &output);
if (len > num_samples) len = num_samples;
if (len)
convert_samples_short(num_c, buffer, 0, f->channels, output, 0, len);