mirror of
https://github.com/CTCaer/RetroArch.git
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883 lines
23 KiB
C
883 lines
23 KiB
C
/* RetroArch - A frontend for libretro.
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* Copyright (C) 2010-2014 - Hans-Kristian Arntzen
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* Copyright (C) 2011-2015 - Daniel De Matteis
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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 <string.h>
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#include <string/string_list.h>
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#include "audio_monitor.h"
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#include "audio_driver.h"
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#include "audio_utils.h"
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#include "audio_thread_wrapper.h"
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#include "../general.h"
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#ifndef AUDIO_BUFFER_FREE_SAMPLES_COUNT
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#define AUDIO_BUFFER_FREE_SAMPLES_COUNT (8 * 1024)
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#endif
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typedef struct audio_driver_input_data
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{
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float *data;
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size_t data_ptr;
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size_t chunk_size;
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size_t nonblock_chunk_size;
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size_t block_chunk_size;
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double src_ratio;
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float in_rate;
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bool use_float;
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float *outsamples;
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int16_t *conv_outsamples;
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int16_t *rewind_buf;
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size_t rewind_ptr;
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size_t rewind_size;
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rarch_dsp_filter_t *dsp;
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bool rate_control;
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double orig_src_ratio;
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size_t driver_buffer_size;
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float volume_gain;
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struct retro_audio_callback audio_callback;
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unsigned buffer_free_samples[AUDIO_BUFFER_FREE_SAMPLES_COUNT];
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uint64_t buffer_free_samples_count;
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} audio_driver_input_data_t;
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static audio_driver_input_data_t audio_data;
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static const audio_driver_t *audio_drivers[] = {
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#ifdef HAVE_ALSA
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&audio_alsa,
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#ifndef __QNX__
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&audio_alsathread,
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#endif
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#endif
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#if defined(HAVE_OSS) || defined(HAVE_OSS_BSD)
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&audio_oss,
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#endif
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#ifdef HAVE_RSOUND
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&audio_rsound,
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#endif
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#ifdef HAVE_COREAUDIO
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&audio_coreaudio,
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#endif
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#ifdef HAVE_AL
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&audio_openal,
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#endif
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#ifdef HAVE_SL
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&audio_opensl,
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#endif
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#ifdef HAVE_ROAR
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&audio_roar,
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#endif
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#ifdef HAVE_JACK
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&audio_jack,
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#endif
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#if defined(HAVE_SDL) || defined(HAVE_SDL2)
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&audio_sdl,
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#endif
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#ifdef HAVE_XAUDIO
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&audio_xa,
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#endif
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#ifdef HAVE_DSOUND
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&audio_dsound,
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#endif
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#ifdef HAVE_PULSE
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&audio_pulse,
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#endif
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#ifdef __CELLOS_LV2__
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&audio_ps3,
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#endif
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#ifdef XENON
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&audio_xenon360,
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#endif
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#ifdef GEKKO
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&audio_gx,
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#endif
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#ifdef EMSCRIPTEN
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&audio_rwebaudio,
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#endif
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#if defined(PSP) || defined(VITA)
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&audio_psp,
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#endif
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#ifdef _3DS
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&audio_ctr,
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#endif
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&audio_null,
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NULL,
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};
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static const audio_driver_t * audio_get_ptr(const driver_t *driver)
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{
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if (driver->audio)
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return driver->audio;
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return NULL;
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}
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/**
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* compute_audio_buffer_statistics:
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*
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* Computes audio buffer statistics.
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*
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**/
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static void compute_audio_buffer_statistics(void)
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{
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unsigned i, low_water_size, high_water_size, avg, stddev;
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float avg_filled, deviation;
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uint64_t accum = 0, accum_var = 0;
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unsigned low_water_count = 0, high_water_count = 0;
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unsigned samples = 0;
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samples = min(audio_data.buffer_free_samples_count,
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AUDIO_BUFFER_FREE_SAMPLES_COUNT);
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if (samples < 3)
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return;
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for (i = 1; i < samples; i++)
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accum += audio_data.buffer_free_samples[i];
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avg = accum / (samples - 1);
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for (i = 1; i < samples; i++)
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{
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int diff = avg - audio_data.buffer_free_samples[i];
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accum_var += diff * diff;
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}
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stddev = (unsigned)sqrt((double)accum_var / (samples - 2));
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avg_filled = 1.0f - (float)avg / audio_data.driver_buffer_size;
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deviation = (float)stddev / audio_data.driver_buffer_size;
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low_water_size = audio_data.driver_buffer_size * 3 / 4;
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high_water_size = audio_data.driver_buffer_size / 4;
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for (i = 1; i < samples; i++)
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{
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if (audio_data.buffer_free_samples[i] >= low_water_size)
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low_water_count++;
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else if (audio_data.buffer_free_samples[i] <= high_water_size)
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high_water_count++;
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}
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RARCH_LOG("Average audio buffer saturation: %.2f %%, standard deviation (percentage points): %.2f %%.\n",
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avg_filled * 100.0, deviation * 100.0);
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RARCH_LOG("Amount of time spent close to underrun: %.2f %%. Close to blocking: %.2f %%.\n",
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(100.0 * low_water_count) / (samples - 1),
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(100.0 * high_water_count) / (samples - 1));
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}
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/**
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* audio_driver_find_handle:
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* @idx : index of driver to get handle to.
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*
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* Returns: handle to audio driver at index. Can be NULL
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* if nothing found.
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**/
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const void *audio_driver_find_handle(int idx)
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{
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const void *drv = audio_drivers[idx];
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if (!drv)
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return NULL;
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return drv;
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}
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/**
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* audio_driver_find_ident:
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* @idx : index of driver to get handle to.
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*
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* Returns: Human-readable identifier of audio driver at index. Can be NULL
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* if nothing found.
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**/
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const char *audio_driver_find_ident(int idx)
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{
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const audio_driver_t *drv = audio_drivers[idx];
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if (!drv)
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return NULL;
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return drv->ident;
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}
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/**
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* config_get_audio_driver_options:
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*
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* Get an enumerated list of all audio driver names, separated by '|'.
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*
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* Returns: string listing of all audio driver names, separated by '|'.
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**/
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const char* config_get_audio_driver_options(void)
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{
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union string_list_elem_attr attr;
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unsigned i;
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char *options = NULL;
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int options_len = 0;
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struct string_list *options_l = string_list_new();
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attr.i = 0;
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if (!options_l)
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return NULL;
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for (i = 0; audio_driver_find_handle(i); i++)
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{
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const char *opt = audio_driver_find_ident(i);
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options_len += strlen(opt) + 1;
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string_list_append(options_l, opt, attr);
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}
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options = (char*)calloc(options_len, sizeof(char));
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if (!options)
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{
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options = NULL;
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goto end;
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}
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string_list_join_concat(options, options_len, options_l, "|");
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end:
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string_list_free(options_l);
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options_l = NULL;
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return options;
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}
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void find_audio_driver(void)
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{
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driver_t *driver = driver_get_ptr();
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settings_t *settings = config_get_ptr();
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int i = find_driver_index("audio_driver", settings->audio.driver);
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if (i >= 0)
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driver->audio = (const audio_driver_t*)audio_driver_find_handle(i);
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else
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{
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unsigned d;
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RARCH_ERR("Couldn't find any audio driver named \"%s\"\n",
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settings->audio.driver);
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RARCH_LOG_OUTPUT("Available audio drivers are:\n");
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for (d = 0; audio_driver_find_handle(d); d++)
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RARCH_LOG_OUTPUT("\t%s\n", audio_driver_find_ident(d));
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RARCH_WARN("Going to default to first audio driver...\n");
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driver->audio = (const audio_driver_t*)audio_driver_find_handle(0);
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if (!driver->audio)
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rarch_fail(1, "find_audio_driver()");
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}
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}
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void uninit_audio(void)
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{
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driver_t *driver = driver_get_ptr();
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settings_t *settings = config_get_ptr();
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if (driver->audio_data && driver->audio)
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driver->audio->free(driver->audio_data);
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if (audio_data.conv_outsamples)
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free(audio_data.conv_outsamples);
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audio_data.conv_outsamples = NULL;
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audio_data.data_ptr = 0;
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if (audio_data.rewind_buf)
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free(audio_data.rewind_buf);
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audio_data.rewind_buf = NULL;
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if (!settings->audio.enable)
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{
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driver->audio_active = false;
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return;
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}
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rarch_resampler_freep(&driver->resampler,
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&driver->resampler_data);
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if (audio_data.audio_callback.callback)
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{
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audio_data.audio_callback.callback = NULL;
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audio_data.audio_callback.set_state = NULL;
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}
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if (audio_data.data)
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free(audio_data.data);
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audio_data.data = NULL;
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if (audio_data.outsamples)
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free(audio_data.outsamples);
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audio_data.outsamples = NULL;
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event_command(EVENT_CMD_DSP_FILTER_DEINIT);
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compute_audio_buffer_statistics();
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}
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void init_audio(void)
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{
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size_t outsamples_max, max_bufsamples = AUDIO_CHUNK_SIZE_NONBLOCKING * 2;
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driver_t *driver = driver_get_ptr();
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settings_t *settings = config_get_ptr();
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audio_convert_init_simd();
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/* Resource leaks will follow if audio is initialized twice. */
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if (driver->audio_data)
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return;
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/* Accomodate rewind since at some point we might have two full buffers. */
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outsamples_max = max_bufsamples * AUDIO_MAX_RATIO *
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settings->slowmotion_ratio;
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/* Used for recording even if audio isn't enabled. */
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rarch_assert(audio_data.conv_outsamples =
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(int16_t*)malloc(outsamples_max * sizeof(int16_t)));
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if (!audio_data.conv_outsamples)
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goto error;
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audio_data.block_chunk_size = AUDIO_CHUNK_SIZE_BLOCKING;
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audio_data.nonblock_chunk_size = AUDIO_CHUNK_SIZE_NONBLOCKING;
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audio_data.chunk_size = audio_data.block_chunk_size;
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/* Needs to be able to hold full content of a full max_bufsamples
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* in addition to its own. */
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rarch_assert(audio_data.rewind_buf = (int16_t*)
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malloc(max_bufsamples * sizeof(int16_t)));
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if (!audio_data.rewind_buf)
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goto error;
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audio_data.rewind_size = max_bufsamples;
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if (!settings->audio.enable)
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{
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driver->audio_active = false;
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return;
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}
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find_audio_driver();
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#ifdef HAVE_THREADS
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if (audio_data.audio_callback.callback)
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{
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RARCH_LOG("Starting threaded audio driver ...\n");
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if (!rarch_threaded_audio_init(&driver->audio, &driver->audio_data,
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*settings->audio.device ? settings->audio.device : NULL,
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settings->audio.out_rate, settings->audio.latency,
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driver->audio))
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{
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RARCH_ERR("Cannot open threaded audio driver ... Exiting ...\n");
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rarch_fail(1, "init_audio()");
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}
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}
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else
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#endif
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{
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driver->audio_data = driver->audio->init(*settings->audio.device ?
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settings->audio.device : NULL,
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settings->audio.out_rate, settings->audio.latency);
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}
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if (!driver->audio_data)
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{
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RARCH_ERR("Failed to initialize audio driver. Will continue without audio.\n");
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driver->audio_active = false;
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}
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audio_data.use_float = false;
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if (driver->audio_active && driver->audio->use_float(driver->audio_data))
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audio_data.use_float = true;
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if (!settings->audio.sync && driver->audio_active)
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{
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event_command(EVENT_CMD_AUDIO_SET_NONBLOCKING_STATE);
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audio_data.chunk_size = audio_data.nonblock_chunk_size;
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}
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if (audio_data.in_rate <= 0.0f)
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{
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/* Should never happen. */
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RARCH_WARN("Input rate is invalid (%.3f Hz). Using output rate (%u Hz).\n",
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audio_data.in_rate, settings->audio.out_rate);
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audio_data.in_rate = settings->audio.out_rate;
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}
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audio_data.orig_src_ratio = audio_data.src_ratio =
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(double)settings->audio.out_rate / audio_data.in_rate;
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if (!rarch_resampler_realloc(&driver->resampler_data,
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&driver->resampler,
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settings->audio.resampler, audio_data.orig_src_ratio))
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{
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RARCH_ERR("Failed to initialize resampler \"%s\".\n",
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settings->audio.resampler);
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driver->audio_active = false;
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}
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rarch_assert(audio_data.data = (float*)
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malloc(max_bufsamples * sizeof(float)));
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if (!audio_data.data)
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goto error;
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audio_data.data_ptr = 0;
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rarch_assert(settings->audio.out_rate <
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audio_data.in_rate * AUDIO_MAX_RATIO);
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rarch_assert(audio_data.outsamples = (float*)
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malloc(outsamples_max * sizeof(float)));
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if (!audio_data.outsamples)
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goto error;
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audio_data.rate_control = false;
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if (!audio_data.audio_callback.callback && driver->audio_active &&
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settings->audio.rate_control)
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{
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/* Audio rate control requires write_avail
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* and buffer_size to be implemented. */
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if (driver->audio->buffer_size)
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{
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audio_data.driver_buffer_size =
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driver->audio->buffer_size(driver->audio_data);
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audio_data.rate_control = true;
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}
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else
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RARCH_WARN("Audio rate control was desired, but driver does not support needed features.\n");
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}
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event_command(EVENT_CMD_DSP_FILTER_INIT);
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audio_data.buffer_free_samples_count = 0;
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if (driver->audio_active && !settings->audio.mute_enable &&
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audio_data.audio_callback.callback)
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{
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/* Threaded driver is initially stopped. */
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driver->audio->start(driver->audio_data);
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}
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return;
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error:
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uninit_audio();
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}
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bool audio_driver_mute_toggle(void)
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{
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driver_t *driver = driver_get_ptr();
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settings_t *settings = config_get_ptr();
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if (!driver->audio_data || !driver->audio_active)
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return false;
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settings->audio.mute_enable = !settings->audio.mute_enable;
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if (settings->audio.mute_enable)
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event_command(EVENT_CMD_AUDIO_STOP);
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else if (!event_command(EVENT_CMD_AUDIO_START))
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{
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driver->audio_active = false;
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return false;
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}
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return true;
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}
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/*
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* audio_driver_readjust_input_rate:
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*
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* Readjust the audio input rate.
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*/
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void audio_driver_readjust_input_rate(void)
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{
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driver_t *driver = driver_get_ptr();
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const audio_driver_t *audio = driver ?
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(const audio_driver_t*)driver->audio : NULL;
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settings_t *settings = config_get_ptr();
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unsigned write_idx = audio_data.buffer_free_samples_count++ &
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(AUDIO_BUFFER_FREE_SAMPLES_COUNT - 1);
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int half_size = audio_data.driver_buffer_size / 2;
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int avail = audio->write_avail(driver->audio_data);
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int delta_mid = avail - half_size;
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double direction = (double)delta_mid / half_size;
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double adjust = 1.0 + settings->audio.rate_control_delta * direction;
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#if 0
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RARCH_LOG_OUTPUT("Audio buffer is %u%% full\n",
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(unsigned)(100 - (avail * 100) / audio_data.driver_buffer_size));
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#endif
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audio_data.buffer_free_samples[write_idx] = avail;
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audio_data.src_ratio = audio_data.orig_src_ratio * adjust;
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#if 0
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RARCH_LOG_OUTPUT("New rate: %lf, Orig rate: %lf\n",
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audio_data.src_ratio, audio_data.orig_src_ratio);
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#endif
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}
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bool audio_driver_alive(void)
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{
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driver_t *driver = driver_get_ptr();
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const audio_driver_t *audio = driver ?
|
|
(const audio_driver_t*)driver->audio : NULL;
|
|
|
|
return audio->alive(driver->audio_data);
|
|
}
|
|
|
|
|
|
bool audio_driver_start(void)
|
|
{
|
|
driver_t *driver = driver_get_ptr();
|
|
const audio_driver_t *audio = audio_get_ptr(driver);
|
|
|
|
return audio->start(driver->audio_data);
|
|
}
|
|
|
|
bool audio_driver_stop(void)
|
|
{
|
|
driver_t *driver = driver_get_ptr();
|
|
const audio_driver_t *audio = audio_get_ptr(driver);
|
|
|
|
return audio->stop(driver->audio_data);
|
|
}
|
|
|
|
void audio_driver_set_nonblock_state(bool toggle)
|
|
{
|
|
driver_t *driver = driver_get_ptr();
|
|
const audio_driver_t *audio = audio_get_ptr(driver);
|
|
|
|
audio->set_nonblock_state(driver->audio_data, toggle);
|
|
}
|
|
|
|
void audio_driver_set_nonblocking_state(bool enable)
|
|
{
|
|
driver_t *driver = driver_get_ptr();
|
|
settings_t *settings = config_get_ptr();
|
|
if (driver->audio_active && driver->audio_data)
|
|
audio_driver_set_nonblock_state(settings->audio.sync ? enable : true);
|
|
|
|
audio_data.chunk_size = enable ? audio_data.nonblock_chunk_size :
|
|
audio_data.block_chunk_size;
|
|
}
|
|
|
|
/**
|
|
* audio_driver_flush:
|
|
* @data : pointer to audio buffer.
|
|
* @right : amount of samples to write.
|
|
*
|
|
* Writes audio samples to audio driver. Will first
|
|
* perform DSP processing (if enabled) and resampling.
|
|
*
|
|
* Returns: true (1) if audio samples were written to the audio
|
|
* driver, false (0) in case of an error.
|
|
**/
|
|
bool audio_driver_flush(const int16_t *data, size_t samples)
|
|
{
|
|
const void *output_data = NULL;
|
|
unsigned output_frames = 0;
|
|
size_t output_size = sizeof(float);
|
|
struct resampler_data src_data = {0};
|
|
struct rarch_dsp_data dsp_data = {0};
|
|
driver_t *driver = driver_get_ptr();
|
|
const audio_driver_t *audio = driver ?
|
|
(const audio_driver_t*)driver->audio : NULL;
|
|
settings_t *settings = config_get_ptr();
|
|
|
|
if (driver->recording_data)
|
|
{
|
|
struct ffemu_audio_data ffemu_data = {0};
|
|
ffemu_data.data = data;
|
|
ffemu_data.frames = samples / 2;
|
|
|
|
if (driver->recording && driver->recording->push_audio)
|
|
driver->recording->push_audio(driver->recording_data, &ffemu_data);
|
|
}
|
|
|
|
if (rarch_main_is_paused() || settings->audio.mute_enable)
|
|
return true;
|
|
if (!driver->audio_active || !audio_data.data)
|
|
return false;
|
|
|
|
RARCH_PERFORMANCE_INIT(audio_convert_s16);
|
|
RARCH_PERFORMANCE_START(audio_convert_s16);
|
|
audio_convert_s16_to_float(audio_data.data, data, samples,
|
|
audio_data.volume_gain);
|
|
RARCH_PERFORMANCE_STOP(audio_convert_s16);
|
|
|
|
src_data.data_in = audio_data.data;
|
|
src_data.input_frames = samples >> 1;
|
|
|
|
dsp_data.input = audio_data.data;
|
|
dsp_data.input_frames = samples >> 1;
|
|
|
|
if (audio_data.dsp)
|
|
{
|
|
RARCH_PERFORMANCE_INIT(audio_dsp);
|
|
RARCH_PERFORMANCE_START(audio_dsp);
|
|
rarch_dsp_filter_process(audio_data.dsp, &dsp_data);
|
|
RARCH_PERFORMANCE_STOP(audio_dsp);
|
|
|
|
if (dsp_data.output)
|
|
{
|
|
src_data.data_in = dsp_data.output;
|
|
src_data.input_frames = dsp_data.output_frames;
|
|
}
|
|
}
|
|
|
|
src_data.data_out = audio_data.outsamples;
|
|
|
|
if (audio_data.rate_control)
|
|
audio_driver_readjust_input_rate();
|
|
|
|
src_data.ratio = audio_data.src_ratio;
|
|
if (rarch_main_is_slowmotion())
|
|
src_data.ratio *= settings->slowmotion_ratio;
|
|
|
|
RARCH_PERFORMANCE_INIT(resampler_proc);
|
|
RARCH_PERFORMANCE_START(resampler_proc);
|
|
rarch_resampler_process(driver->resampler,
|
|
driver->resampler_data, &src_data);
|
|
RARCH_PERFORMANCE_STOP(resampler_proc);
|
|
|
|
output_data = audio_data.outsamples;
|
|
output_frames = src_data.output_frames;
|
|
|
|
if (!audio_data.use_float)
|
|
{
|
|
RARCH_PERFORMANCE_INIT(audio_convert_float);
|
|
RARCH_PERFORMANCE_START(audio_convert_float);
|
|
audio_convert_float_to_s16(audio_data.conv_outsamples,
|
|
(const float*)output_data, output_frames * 2);
|
|
RARCH_PERFORMANCE_STOP(audio_convert_float);
|
|
|
|
output_data = audio_data.conv_outsamples;
|
|
output_size = sizeof(int16_t);
|
|
}
|
|
|
|
if (audio->write(driver->audio_data, output_data, output_frames * output_size * 2) < 0)
|
|
{
|
|
driver->audio_active = false;
|
|
return false;
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
/**
|
|
* audio_driver_sample:
|
|
* @left : value of the left audio channel.
|
|
* @right : value of the right audio channel.
|
|
*
|
|
* Audio sample render callback function.
|
|
**/
|
|
void audio_driver_sample(int16_t left, int16_t right)
|
|
{
|
|
audio_data.conv_outsamples[audio_data.data_ptr++] = left;
|
|
audio_data.conv_outsamples[audio_data.data_ptr++] = right;
|
|
|
|
if (audio_data.data_ptr < audio_data.chunk_size)
|
|
return;
|
|
|
|
audio_driver_flush(audio_data.conv_outsamples, audio_data.data_ptr);
|
|
|
|
audio_data.data_ptr = 0;
|
|
}
|
|
|
|
/**
|
|
* audio_driver_sample_batch:
|
|
* @data : pointer to audio buffer.
|
|
* @frames : amount of audio frames to push.
|
|
*
|
|
* Batched audio sample render callback function.
|
|
*
|
|
* Returns: amount of frames sampled. Will be equal to @frames
|
|
* unless @frames exceeds (AUDIO_CHUNK_SIZE_NONBLOCKING / 2).
|
|
**/
|
|
size_t audio_driver_sample_batch(const int16_t *data, size_t frames)
|
|
{
|
|
if (frames > (AUDIO_CHUNK_SIZE_NONBLOCKING >> 1))
|
|
frames = AUDIO_CHUNK_SIZE_NONBLOCKING >> 1;
|
|
|
|
audio_driver_flush(data, frames << 1);
|
|
|
|
return frames;
|
|
}
|
|
|
|
/**
|
|
* audio_driver_sample_rewind:
|
|
* @left : value of the left audio channel.
|
|
* @right : value of the right audio channel.
|
|
*
|
|
* Audio sample render callback function (rewind version). This callback
|
|
* function will be used instead of audio_driver_sample when rewinding is activated.
|
|
**/
|
|
void audio_driver_sample_rewind(int16_t left, int16_t right)
|
|
{
|
|
audio_data.rewind_buf[--audio_data.rewind_ptr] = right;
|
|
audio_data.rewind_buf[--audio_data.rewind_ptr] = left;
|
|
}
|
|
|
|
/**
|
|
* audio_driver_sample_batch_rewind:
|
|
* @data : pointer to audio buffer.
|
|
* @frames : amount of audio frames to push.
|
|
*
|
|
* Batched audio sample render callback function (rewind version). This callback
|
|
* function will be used instead of audio_driver_sample_batch when rewinding is activated.
|
|
*
|
|
* Returns: amount of frames sampled. Will be equal to @frames
|
|
* unless @frames exceeds (AUDIO_CHUNK_SIZE_NONBLOCKING / 2).
|
|
**/
|
|
size_t audio_driver_sample_batch_rewind(const int16_t *data, size_t frames)
|
|
{
|
|
size_t i;
|
|
size_t samples = frames << 1;
|
|
|
|
for (i = 0; i < samples; i++)
|
|
audio_data.rewind_buf[--audio_data.rewind_ptr] = data[i];
|
|
|
|
return frames;
|
|
}
|
|
|
|
void audio_driver_set_volume_gain(float gain)
|
|
{
|
|
audio_data.volume_gain = gain;
|
|
}
|
|
|
|
void audio_driver_dsp_filter_free(void)
|
|
{
|
|
if (audio_data.dsp)
|
|
rarch_dsp_filter_free(audio_data.dsp);
|
|
audio_data.dsp = NULL;
|
|
}
|
|
|
|
void audio_driver_dsp_filter_init(const char *device)
|
|
{
|
|
audio_data.dsp = rarch_dsp_filter_new(device, audio_data.in_rate);
|
|
if (!audio_data.dsp)
|
|
RARCH_ERR("[DSP]: Failed to initialize DSP filter \"%s\".\n", device);
|
|
}
|
|
|
|
void audio_driver_setup_rewind(void)
|
|
{
|
|
unsigned i;
|
|
|
|
/* Push audio ready to be played. */
|
|
audio_data.rewind_ptr = audio_data.rewind_size;
|
|
|
|
for (i = 0; i < audio_data.data_ptr; i += 2)
|
|
{
|
|
audio_data.rewind_buf[--audio_data.rewind_ptr] =
|
|
audio_data.conv_outsamples[i + 1];
|
|
|
|
audio_data.rewind_buf[--audio_data.rewind_ptr] =
|
|
audio_data.conv_outsamples[i + 0];
|
|
}
|
|
|
|
audio_data.data_ptr = 0;
|
|
}
|
|
|
|
void audio_driver_frame_is_reverse(void)
|
|
{
|
|
/* We just rewound. Flush rewind audio buffer. */
|
|
audio_driver_flush(audio_data.rewind_buf + audio_data.rewind_ptr,
|
|
audio_data.rewind_size - audio_data.rewind_ptr);
|
|
}
|
|
|
|
void audio_monitor_adjust_system_rates(void)
|
|
{
|
|
float timing_skew;
|
|
settings_t *settings = config_get_ptr();
|
|
struct retro_system_av_info *av_info =
|
|
video_viewport_get_system_av_info();
|
|
const struct retro_system_timing *info =
|
|
av_info ? (const struct retro_system_timing*)&av_info->timing : NULL;
|
|
|
|
if (info->sample_rate <= 0.0)
|
|
return;
|
|
|
|
timing_skew = fabs(1.0f - info->fps /
|
|
settings->video.refresh_rate);
|
|
audio_data.in_rate = info->sample_rate;
|
|
|
|
if (timing_skew <= settings->audio.max_timing_skew)
|
|
audio_data.in_rate *= (settings->video.refresh_rate / info->fps);
|
|
|
|
RARCH_LOG("Set audio input rate to: %.2f Hz.\n",
|
|
audio_data.in_rate);
|
|
}
|
|
|
|
/**
|
|
* audio_monitor_set_refresh_rate:
|
|
*
|
|
* Sets audio monitor refresh rate to new value.
|
|
**/
|
|
void audio_monitor_set_refresh_rate(void)
|
|
{
|
|
settings_t *settings = config_get_ptr();
|
|
|
|
double new_src_ratio = (double)settings->audio.out_rate /
|
|
audio_data.in_rate;
|
|
|
|
audio_data.orig_src_ratio = new_src_ratio;
|
|
audio_data.src_ratio = new_src_ratio;
|
|
}
|
|
|
|
void audio_driver_set_buffer_size(size_t bufsize)
|
|
{
|
|
audio_data.driver_buffer_size = bufsize;
|
|
}
|
|
|
|
void audio_driver_set_callback(const void *data)
|
|
{
|
|
const struct retro_audio_callback *cb =
|
|
(const struct retro_audio_callback*)data;
|
|
|
|
if (cb)
|
|
audio_data.audio_callback = *cb;
|
|
}
|
|
|
|
bool audio_driver_has_callback(void)
|
|
{
|
|
return audio_data.audio_callback.callback;
|
|
}
|
|
|
|
void audio_driver_callback(void)
|
|
{
|
|
if (audio_driver_has_callback())
|
|
{
|
|
if (audio_data.audio_callback.callback)
|
|
audio_data.audio_callback.callback();
|
|
}
|
|
}
|
|
|
|
void audio_driver_callback_set_state(bool state)
|
|
{
|
|
if (audio_driver_has_callback())
|
|
{
|
|
if (audio_data.audio_callback.set_state)
|
|
audio_data.audio_callback.set_state(state);
|
|
}
|
|
}
|