mirror of
https://github.com/CTCaer/RetroArch.git
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Added an audio mixer
This commit is contained in:
Andre Leiradella
parent
0d945a81d7
commit
7e029a5981
545
audio/audio_mixer.c
Normal file
545
audio/audio_mixer.c
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@ -0,0 +1,545 @@
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/* RetroArch - A frontend for libretro.
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* Copyright (C) 2017 - Andre Leiradella
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*
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* RetroArch is free software: you can redistribute it and/or modify it under the terms
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* of the GNU General Public License as published by the Free Software Found-
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* ation, either version 3 of the License, or (at your option) any later version.
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*
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* RetroArch is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY;
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* without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR
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* PURPOSE. See the GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along with RetroArch.
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* If not, see <http://www.gnu.org/licenses/>.
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*/
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#include "audio_mixer.h"
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#include "audio_driver.h"
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#include <audio/audio_resampler.h>
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#include <streams/file_stream.h>
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#include <formats/rwav.h>
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#include <memalign.h>
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#include <stdlib.h>
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#include <string.h>
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#include <math.h>
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/*---------------------------------------------------------------------------*/
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#define STB_VORBIS_NO_PUSHDATA_API
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#define STB_VORBIS_NO_STDIO
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#define STB_VORBIS_NO_CRT
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#define assert( x )
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#undef NULL
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#ifdef __MINGW32__
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#undef __forceinline
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#define __forceinline __inline__ __attribute__((__always_inline__,__gnu_inline__))
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#endif
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#include "stb_vorbis.c"
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/*---------------------------------------------------------------------------*/
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#define AUDIO_MIXER_MAX_VOICES 8
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#define AUDIO_MIXER_TEMP_OGG_BUFFER 8192
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#define AUDIO_MIXER_TYPE_NONE 0
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#define AUDIO_MIXER_TYPE_WAV 1
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#define AUDIO_MIXER_TYPE_OGG 2
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struct audio_mixer_sound_t
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{
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unsigned type;
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union
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{
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struct
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{
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/* wav */
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unsigned frames;
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const float* pcm;
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} wav;
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struct
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{
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/* ogg */
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unsigned size;
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const void* data;
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} ogg;
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} types;
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};
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struct audio_mixer_voice_t
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{
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unsigned type;
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bool repeat;
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float volume;
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audio_mixer_sound_t* sound;
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audio_mixer_stop_cb_t stop_cb;
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union
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{
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struct
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{
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/* wav */
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unsigned position;
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} wav;
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struct
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{
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/* ogg */
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unsigned position;
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unsigned samples;
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stb_vorbis* stream;
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float* buffer;
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unsigned buf_samples;
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void* resampler_data;
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float ratio;
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const retro_resampler_t* resampler;
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} ogg;
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} types;
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};
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static unsigned s_rate;
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static audio_mixer_voice_t s_voices[AUDIO_MIXER_MAX_VOICES];
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static bool wav2float(const rwav_t* wav, float** pcm, size_t* samples_out)
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{
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float* f;
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float sample;
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const uint8_t* u8;
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const int16_t* s16;
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size_t i;
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/* Allocate on a 16-byte boundary, and pad to a multiple of 16 bytes */
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*samples_out = wav->numsamples * 2;
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f = (float*)memalign_alloc(16, ((*samples_out + 15) & ~15) * sizeof(float));
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if (f == NULL)
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return false;
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*pcm = f;
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if (wav->numchannels == 1)
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{
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if (wav->bitspersample == 8)
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{
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u8 = (const uint8_t*)wav->samples;
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for (i = wav->numsamples; i != 0; i--)
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{
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sample = (float)*u8++ / 255.0f;
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sample = sample * 2.0f - 1.0f;
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*f++ = sample;
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*f++ = sample;
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}
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}
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else
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{
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s16 = (const int16_t*)wav->samples;
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for (i = wav->numsamples; i != 0; i--)
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{
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sample = (float)((int)*s16++ + 32768) / 65535.0f;
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sample = sample * 2.0f - 1.0f;
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*f++ = sample;
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*f++ = sample;
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}
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}
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}
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else if (wav->numchannels == 2)
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{
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if (wav->bitspersample == 8)
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{
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u8 = (const uint8_t*)wav->samples;
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for (i = wav->numsamples; i != 0; i--)
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{
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sample = (float)*u8++ / 255.0f;
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sample = sample * 2.0f - 1.0f;
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*f++ = sample;
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sample = (float)*u8++ / 255.0f;
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sample = sample * 2.0f - 1.0f;
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*f++ = sample;
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}
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}
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else
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{
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s16 = (const int16_t*)wav->samples;
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for (i = wav->numsamples; i != 0; i--)
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{
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sample = (float)((int)*s16++ + 32768) / 65535.0f;
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sample = sample * 2.0f - 1.0f;
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*f++ = sample;
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sample = (float)((int)*s16++ + 32768) / 65535.0f;
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sample = sample * 2.0f - 1.0f;
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*f++ = sample;
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}
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}
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}
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return true;
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}
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static bool one_shot_resample(const float* in, size_t samples_in, unsigned rate, float** out, size_t* samples_out)
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{
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void* data = NULL;
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const retro_resampler_t* resampler = NULL;
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struct resampler_data info = {0};
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float ratio = (double)s_rate / (double)rate;
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if (!retro_resampler_realloc(&data, &resampler, NULL, ratio))
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return false;
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/* Allocate on a 16-byte boundary, and pad to a multiple of 16 bytes */
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*samples_out = samples_in * ratio;
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*out = (float*)memalign_alloc(16, ((*samples_out + 15) & ~15) * sizeof(float));
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if (*out == NULL)
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return false;
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info.data_in = in;
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info.data_out = *out;
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info.input_frames = samples_in / 2;
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info.ratio = ratio;
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resampler->process(data, &info);
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resampler->free(data);
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return true;
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}
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void audio_mixer_init(unsigned rate)
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{
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unsigned i;
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s_rate = rate;
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for (i = 0; i < AUDIO_MIXER_MAX_VOICES; i++)
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s_voices[i].type = AUDIO_MIXER_TYPE_NONE;
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}
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void audio_mixer_done(void)
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{
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unsigned i;
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for (i = 0; i < AUDIO_MIXER_MAX_VOICES; i++)
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s_voices[i].type = AUDIO_MIXER_TYPE_NONE;
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}
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audio_mixer_sound_t* audio_mixer_load_wav(const char* path)
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{
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/* Raw WAV bytes */
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void* buffer;
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ssize_t size;
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/* WAV data */
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rwav_t wav;
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/* WAV samples converted to float */
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float* pcm;
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float* resampled;
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size_t samples;
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/* Result */
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audio_mixer_sound_t* sound;
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if (filestream_read_file(path, &buffer, &size) == 0)
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return NULL;
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if (rwav_load(&wav, buffer, size) != RWAV_ITERATE_DONE)
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{
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free(buffer);
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return NULL;
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}
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free(buffer);
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if (!wav2float(&wav, &pcm, &samples))
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return NULL;
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if (wav.samplerate != s_rate)
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{
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if (!one_shot_resample(pcm, samples, wav.samplerate, &resampled, &samples))
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return NULL;
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memalign_free((void*)pcm);
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pcm = resampled;
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}
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sound = (audio_mixer_sound_t*)malloc(sizeof(audio_mixer_sound_t));
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if (sound == NULL)
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{
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memalign_free((void*)pcm);
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return NULL;
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}
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sound->type = AUDIO_MIXER_TYPE_WAV;
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sound->types.wav.frames = samples / 2;
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sound->types.wav.pcm = pcm;
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rwav_free(&wav);
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return sound;
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}
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audio_mixer_sound_t* audio_mixer_load_ogg(const char* path)
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{
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void* buffer;
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ssize_t size;
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audio_mixer_sound_t* sound;
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if (filestream_read_file(path, &buffer, &size) == 0)
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return NULL;
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sound = (audio_mixer_sound_t*)malloc(sizeof(audio_mixer_sound_t));
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if (sound == NULL)
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{
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free(buffer);
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return NULL;
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}
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sound->type = AUDIO_MIXER_TYPE_OGG;
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sound->types.ogg.size = size;
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sound->types.ogg.data = buffer;
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return sound;
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}
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void audio_mixer_destroy(audio_mixer_sound_t* sound)
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{
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if (sound->type == AUDIO_MIXER_TYPE_WAV)
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memalign_free((void*)sound->types.wav.pcm);
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else if (sound->type == AUDIO_MIXER_TYPE_OGG)
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memalign_free((void*)sound->types.ogg.data);
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free(sound);
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}
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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)
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{
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voice->type = AUDIO_MIXER_TYPE_WAV;
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voice->repeat = repeat;
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voice->volume = volume;
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voice->sound = sound;
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voice->stop_cb = stop_cb;
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voice->types.wav.position = 0;
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return true;
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}
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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)
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{
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int res;
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stb_vorbis_info info;
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float ratio;
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unsigned samples;
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voice->repeat = repeat;
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voice->volume = volume;
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voice->sound = sound;
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voice->stop_cb = stop_cb;
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voice->types.ogg.stream = stb_vorbis_open_memory((const unsigned char*)sound->types.ogg.data, sound->types.ogg.size, &res, NULL);
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if (voice->types.ogg.stream == NULL)
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return false;
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info = stb_vorbis_get_info(voice->types.ogg.stream);
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if (info.channels != 2)
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{
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/* Only stereo supported for now */
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stb_vorbis_close(voice->types.ogg.stream);
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return false;
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}
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if (info.sample_rate != s_rate)
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{
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voice->types.ogg.ratio = ratio = (double)s_rate / (double)info.sample_rate;
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if (!retro_resampler_realloc(&voice->types.ogg.resampler_data, &voice->types.ogg.resampler, NULL, ratio))
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{
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stb_vorbis_close(voice->types.ogg.stream);
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return false;
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}
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}
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samples = voice->types.ogg.buf_samples = (unsigned)(AUDIO_MIXER_TEMP_OGG_BUFFER * ratio);
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voice->types.ogg.buffer = (float*)memalign_alloc(16, ((samples + 15) & ~15) * sizeof(float));
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if (voice->types.ogg.buffer == NULL)
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{
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voice->types.ogg.resampler->free(voice->types.ogg.resampler_data);
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stb_vorbis_close(voice->types.ogg.stream);
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return false;
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}
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voice->type = AUDIO_MIXER_TYPE_OGG;
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voice->types.ogg.position = voice->types.ogg.samples = 0;
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return true;
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}
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audio_mixer_voice_t* audio_mixer_play(audio_mixer_sound_t* sound, bool repeat, float volume, audio_mixer_stop_cb_t stop_cb)
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{
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unsigned i;
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audio_mixer_voice_t* voice;
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bool res = false;
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for (i = 0, voice = s_voices; i < AUDIO_MIXER_MAX_VOICES; i++, voice++)
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{
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if (voice->type == AUDIO_MIXER_TYPE_NONE)
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{
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if (sound->type == AUDIO_MIXER_TYPE_WAV)
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res = audio_mixer_play_wav(sound, voice, repeat, volume, stop_cb);
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else if (sound->type == AUDIO_MIXER_TYPE_OGG)
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res = audio_mixer_play_ogg(sound, voice, repeat, volume, stop_cb);
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break;
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}
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}
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return res ? voice : NULL;
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}
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void audio_mixer_stop(audio_mixer_voice_t* voice)
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{
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if (voice->type == AUDIO_MIXER_TYPE_WAV)
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{
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voice->stop_cb(voice, AUDIO_MIXER_SOUND_STOPPED);
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voice->type = AUDIO_MIXER_TYPE_NONE;
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}
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}
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static void mix_wav(float* buffer, size_t num_frames, audio_mixer_voice_t* voice)
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{
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unsigned buf_free = num_frames * 2;
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const audio_mixer_sound_t* sound = voice->sound;
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unsigned pcm_available = sound->types.wav.frames * 2 - voice->types.wav.position;
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const float* pcm = sound->types.wav.pcm + voice->types.wav.position;
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float volume = voice->volume;
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int i;
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again:
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if (pcm_available < buf_free)
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{
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for (i = pcm_available; i != 0; i--)
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*buffer++ += *pcm++ * volume;
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if (voice->repeat)
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{
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if (voice->stop_cb)
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voice->stop_cb(voice, AUDIO_MIXER_SOUND_REPEATED);
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buf_free -= pcm_available;
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pcm_available = sound->types.wav.frames * 2;
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pcm = sound->types.wav.pcm;
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voice->types.wav.position = 0;
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goto again;
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}
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if (voice->stop_cb)
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voice->stop_cb(voice, AUDIO_MIXER_SOUND_FINISHED);
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voice->type = AUDIO_MIXER_TYPE_NONE;
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}
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else
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{
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for (i = buf_free; i != 0; i--)
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*buffer++ += *pcm++ * volume;
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voice->types.wav.position += buf_free;
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}
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}
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static void mix_ogg(float* buffer, size_t num_frames, audio_mixer_voice_t* voice)
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{
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unsigned buf_free = num_frames * 2;
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const audio_mixer_sound_t* sound = voice->sound;
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float temp_buffer[AUDIO_MIXER_TEMP_OGG_BUFFER];
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unsigned temp_samples;
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float volume = voice->volume;
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struct resampler_data info = {0};
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float* pcm;
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int i;
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if (voice->types.ogg.position == voice->types.ogg.samples)
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{
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again:
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temp_samples = stb_vorbis_get_samples_float_interleaved(voice->types.ogg.stream, 2, temp_buffer, AUDIO_MIXER_TEMP_OGG_BUFFER) * 2;
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if (temp_samples == 0)
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{
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if (voice->repeat)
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{
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if (voice->stop_cb)
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voice->stop_cb(voice, AUDIO_MIXER_SOUND_REPEATED);
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stb_vorbis_seek_start(voice->types.ogg.stream);
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goto again;
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}
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else
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{
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if (voice->stop_cb)
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voice->stop_cb(voice, AUDIO_MIXER_SOUND_FINISHED);
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voice->types.ogg.resampler->free(voice->types.ogg.resampler_data);
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stb_vorbis_close(voice->types.ogg.stream);
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voice->type = AUDIO_MIXER_TYPE_NONE;
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return;
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}
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}
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info.data_in = temp_buffer;
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info.data_out = voice->types.ogg.buffer;
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info.input_frames = temp_samples / 2;
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info.ratio = voice->types.ogg.ratio;
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|
||||
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;
|
||||
}
|
||||
|
||||
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;
|
||||
}
|
||||
else
|
||||
{
|
||||
for ( int i = buf_free; i != 0; --i )
|
||||
*buffer++ += *pcm++ * volume;
|
||||
|
||||
voice->types.ogg.position += buf_free;
|
||||
voice->types.ogg.samples -= buf_free;
|
||||
}
|
||||
}
|
||||
|
||||
void audio_mixer_mix(float* buffer, size_t num_frames)
|
||||
{
|
||||
unsigned i;
|
||||
audio_mixer_voice_t* voice;
|
||||
size_t j;
|
||||
float* sample;
|
||||
|
||||
for (i = 0, voice = s_voices; i < AUDIO_MIXER_MAX_VOICES; i++, voice++)
|
||||
{
|
||||
if (voice->type == AUDIO_MIXER_TYPE_WAV)
|
||||
mix_wav(buffer, num_frames, voice);
|
||||
else if (voice->type == AUDIO_MIXER_TYPE_OGG)
|
||||
mix_ogg(buffer, num_frames, voice);
|
||||
}
|
||||
|
||||
for (j = 0, sample = buffer; j < num_frames; j++, sample++)
|
||||
{
|
||||
if (*sample < -1.0f)
|
||||
*sample = -1.0f;
|
||||
else if (*sample > 1.0f)
|
||||
*sample = 1.0f;
|
||||
}
|
||||
}
|
||||
50
audio/audio_mixer.h
Normal file
50
audio/audio_mixer.h
Normal file
@ -0,0 +1,50 @@
|
||||
/* RetroArch - A frontend for libretro.
|
||||
* Copyright (C) 2017 - Andre Leiradella
|
||||
*
|
||||
* RetroArch is free software: you can redistribute it and/or modify it under the terms
|
||||
* of the GNU General Public License as published by the Free Software Found-
|
||||
* ation, either version 3 of the License, or (at your option) any later version.
|
||||
*
|
||||
* RetroArch is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY;
|
||||
* without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR
|
||||
* PURPOSE. See the GNU General Public License for more details.
|
||||
*
|
||||
* You should have received a copy of the GNU General Public License along with RetroArch.
|
||||
* If not, see <http://www.gnu.org/licenses/>.
|
||||
*/
|
||||
|
||||
#ifndef __AUDIO_MIXER__H
|
||||
#define __AUDIO_MIXER__H
|
||||
|
||||
#include <boolean.h>
|
||||
#include <retro_common_api.h>
|
||||
#include <stdint.h>
|
||||
|
||||
RETRO_BEGIN_DECLS
|
||||
|
||||
typedef struct audio_mixer_sound_t audio_mixer_sound_t;
|
||||
typedef struct audio_mixer_voice_t audio_mixer_voice_t;
|
||||
|
||||
typedef void (*audio_mixer_stop_cb_t)(audio_mixer_voice_t* voice, unsigned reason);
|
||||
|
||||
/* Reasons passed to the stop callback. */
|
||||
#define AUDIO_MIXER_SOUND_FINISHED 0
|
||||
#define AUDIO_MIXER_SOUND_STOPPED 1
|
||||
#define AUDIO_MIXER_SOUND_REPEATED 2
|
||||
|
||||
void audio_mixer_init(unsigned rate);
|
||||
void audio_mixer_done(void);
|
||||
|
||||
audio_mixer_sound_t* audio_mixer_load_wav(const char* path);
|
||||
audio_mixer_sound_t* audio_mixer_load_ogg(const char* path);
|
||||
|
||||
void audio_mixer_destroy(audio_mixer_sound_t* sound);
|
||||
|
||||
audio_mixer_voice_t* audio_mixer_play(audio_mixer_sound_t* sound, bool repeat, float volume, audio_mixer_stop_cb_t stop_cb);
|
||||
void audio_mixer_stop(audio_mixer_voice_t* voice);
|
||||
|
||||
void audio_mixer_mix(float* buffer, size_t num_frames);
|
||||
|
||||
RETRO_END_DECLS
|
||||
|
||||
#endif
|
||||
5445
audio/stb_vorbis.c
Normal file
5445
audio/stb_vorbis.c
Normal file
File diff suppressed because it is too large
Load Diff
Loading…
Reference in New Issue
Block a user