mirror of
https://github.com/libretro/RetroArch.git
synced 2024-11-23 16:09:47 +00:00
fcbd72dbf3
* * Use flags for rarch_state * Get rid of ^M linebreaks in retroarch_types.h * Buildfixes for consoles * (audio driver) use flags instead of bools * (video) Use flags instead of bools * Rewrite input driver state bools into flags
1882 lines
56 KiB
C
1882 lines
56 KiB
C
/**
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* RetroArch - A frontend for libretro.
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* Copyright (C) 2010-2014 - Hans-Kristian Arntzen
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* Copyright (C) 2011-2017 - Daniel De Matteis
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*
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* RetroArch is free software: you can redistribute it and/or modify it under
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* the terms of the GNU General Public License as published by the Free
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* Software Foundation, either version 3 of the License, or (at your option)
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* any later version.
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*
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* RetroArch is distributed in the hope that it will be useful, but WITHOUT
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* ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
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* FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for
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* more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with RetroArch. If not, see <http://www.gnu.org/licenses/>.
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**/
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#include <math.h>
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#include "audio_driver.h"
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#include <string/stdstring.h>
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#include <encodings/utf.h>
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#include <clamping.h>
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#include <retro_assert.h>
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#include <memalign.h>
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#include <audio/conversion/float_to_s16.h>
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#include <audio/conversion/s16_to_float.h>
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#ifdef HAVE_AUDIOMIXER
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#include <audio/audio_mixer.h>
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#include "../tasks/task_audio_mixer.h"
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#endif
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#ifdef HAVE_DSP_FILTER
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#include <audio/dsp_filter.h>
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#endif
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#include <lists/dir_list.h>
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#ifdef HAVE_THREADS
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#include "audio_thread_wrapper.h"
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#endif
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#ifdef HAVE_MENU
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#include "../menu/menu_driver.h"
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#endif
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#ifdef HAVE_NETWORKING
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#include "../network/netplay/netplay.h"
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#endif
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#include "../configuration.h"
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#include "../driver.h"
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#include "../frontend/frontend_driver.h"
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#include "../retroarch.h"
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#include "../list_special.h"
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#include "../file_path_special.h"
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#include "../record/record_driver.h"
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#include "../tasks/task_content.h"
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#include "../verbosity.h"
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#define MENU_SOUND_FORMATS "ogg|mod|xm|s3m|mp3|flac|wav"
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/* Converts decibels to voltage gain. returns voltage gain value. */
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#define DB_TO_GAIN(db) (powf(10.0f, (db) / 20.0f))
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audio_driver_t audio_null = {
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NULL, /* init */
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NULL, /* write */
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NULL, /* stop */
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NULL, /* start */
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NULL, /* alive */
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NULL, /* set_nonblock_state */
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NULL, /* free */
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NULL, /* use_float */
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"null",
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NULL,
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NULL,
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NULL, /* write_avail */
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NULL
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};
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audio_driver_t *audio_drivers[] = {
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#ifdef HAVE_ALSA
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&audio_alsa,
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#if !defined(__QNX__) && !defined(MIYOO) && defined(HAVE_THREADS)
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&audio_alsathread,
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#endif
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#endif
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#ifdef HAVE_TINYALSA
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&audio_tinyalsa,
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#endif
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#if defined(HAVE_AUDIOIO)
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&audio_audioio,
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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_COREAUDIO3
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&audio_coreaudio3,
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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_WASAPI
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&audio_wasapi,
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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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#if defined(__PSL1GHT__) || defined(__PS3__)
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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 WIIU
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&audio_ax,
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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) || defined(ORBIS)
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&audio_psp,
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#endif
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#if defined(PS2)
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&audio_ps2,
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#endif
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#ifdef _3DS
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&audio_ctr_csnd,
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&audio_ctr_dsp,
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#ifdef HAVE_THREADS
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&audio_ctr_dsp_thread,
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#endif
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#endif
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#ifdef SWITCH
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&audio_switch,
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&audio_switch_thread,
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#ifdef HAVE_LIBNX
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&audio_switch_libnx_audren,
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&audio_switch_libnx_audren_thread,
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#endif
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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 audio_driver_state_t audio_driver_st = {0}; /* double alignment */
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/**************************************/
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audio_driver_state_t *audio_state_get_ptr(void)
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{
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return &audio_driver_st;
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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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return char_list_new_special(STRING_LIST_AUDIO_DRIVERS, NULL);
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}
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#ifdef HAVE_TRANSLATE
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/* TODO/FIXME - Doesn't currently work. Fix this. */
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bool audio_driver_is_ai_service_speech_running(void)
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{
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#ifdef HAVE_AUDIOMIXER
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enum audio_mixer_state res = audio_driver_mixer_get_stream_state(10);
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bool ret = (res == AUDIO_STREAM_STATE_NONE) || (res == AUDIO_STREAM_STATE_STOPPED);
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if (!ret)
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return true;
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#endif
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return false;
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}
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#endif
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static enum resampler_quality audio_driver_get_resampler_quality(
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settings_t *settings)
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{
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if (settings)
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return (enum resampler_quality)settings->uints.audio_resampler_quality;
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return RESAMPLER_QUALITY_DONTCARE;
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}
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static bool audio_driver_free_devices_list(void)
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{
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audio_driver_state_t *audio_st = &audio_driver_st;
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if (
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!audio_st->current_audio
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|| !audio_st->current_audio->device_list_free
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|| !audio_st->context_audio_data)
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return false;
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audio_st->current_audio->device_list_free(
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audio_st->context_audio_data,
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audio_st->devices_list);
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audio_st->devices_list = NULL;
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return true;
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}
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#ifdef DEBUG
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static void report_audio_buffer_statistics(void)
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{
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audio_statistics_t audio_stats;
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audio_stats.samples = 0;
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audio_stats.average_buffer_saturation = 0.0f;
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audio_stats.std_deviation_percentage = 0.0f;
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audio_stats.close_to_underrun = 0.0f;
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audio_stats.close_to_blocking = 0.0f;
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if (!audio_compute_buffer_statistics(&audio_stats))
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return;
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RARCH_LOG("[Audio]: Average audio buffer saturation: %.2f %%,"
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" standard deviation (percentage points): %.2f %%.\n"
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"[Audio]: Amount of time spent close to underrun: %.2f %%."
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" Close to blocking: %.2f %%.\n",
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audio_stats.average_buffer_saturation,
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audio_stats.std_deviation_percentage,
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audio_stats.close_to_underrun,
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audio_stats.close_to_blocking);
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}
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#endif
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static void audio_driver_deinit_resampler(void)
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{
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audio_driver_state_t *audio_st = &audio_driver_st;
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if (audio_st->resampler && audio_st->resampler_data)
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audio_st->resampler->free(audio_st->resampler_data);
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audio_st->resampler = NULL;
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audio_st->resampler_data = NULL;
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audio_st->resampler_ident[0] = '\0';
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audio_st->resampler_quality = RESAMPLER_QUALITY_DONTCARE;
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}
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static bool audio_driver_deinit_internal(bool audio_enable)
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{
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audio_driver_state_t *audio_st = &audio_driver_st;
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if ( audio_st->current_audio
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&& audio_st->current_audio->free)
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{
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if (audio_st->context_audio_data)
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audio_st->current_audio->free(audio_st->context_audio_data);
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audio_st->context_audio_data = NULL;
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}
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if (audio_st->output_samples_conv_buf)
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memalign_free(audio_st->output_samples_conv_buf);
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audio_st->output_samples_conv_buf = NULL;
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if (audio_st->input_data)
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memalign_free(audio_st->input_data);
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audio_st->input_data = NULL;
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audio_st->data_ptr = 0;
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#ifdef HAVE_REWIND
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if (audio_st->rewind_buf)
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memalign_free(audio_st->rewind_buf);
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audio_st->rewind_buf = NULL;
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audio_st->rewind_size = 0;
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#endif
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if (!audio_enable)
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{
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audio_st->flags &= ~AUDIO_FLAG_ACTIVE;
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return false;
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}
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audio_driver_deinit_resampler();
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if (audio_st->output_samples_buf)
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memalign_free(audio_st->output_samples_buf);
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audio_st->output_samples_buf = NULL;
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#ifdef HAVE_DSP_FILTER
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audio_driver_dsp_filter_free();
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#endif
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#ifdef DEBUG
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report_audio_buffer_statistics();
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#endif
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return true;
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}
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#ifdef HAVE_AUDIOMIXER
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static void audio_driver_mixer_deinit(void)
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{
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unsigned i;
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audio_driver_st.flags &= ~AUDIO_FLAG_MIXER_ACTIVE;
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for (i = 0; i < AUDIO_MIXER_MAX_SYSTEM_STREAMS; i++)
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{
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audio_driver_mixer_stop_stream(i);
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audio_driver_mixer_remove_stream(i);
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}
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audio_mixer_done();
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}
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#endif
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bool audio_driver_deinit(void)
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{
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settings_t *settings = config_get_ptr();
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#ifdef HAVE_AUDIOMIXER
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audio_driver_mixer_deinit();
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#endif
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audio_driver_free_devices_list();
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return audio_driver_deinit_internal(
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settings->bools.audio_enable);
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}
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bool audio_driver_find_driver(
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void *settings_data,
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const char *prefix,
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bool verbosity_enabled)
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{
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settings_t *settings = (settings_t*)settings_data;
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int i = (int)driver_find_index(
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"audio_driver",
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settings->arrays.audio_driver);
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if (i >= 0)
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audio_driver_st.current_audio = (const audio_driver_t*)
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audio_drivers[i];
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else
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{
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const audio_driver_t *tmp = NULL;
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if (verbosity_enabled)
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{
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unsigned d;
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RARCH_ERR("Couldn't find any %s named \"%s\"\n", prefix,
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settings->arrays.audio_driver);
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RARCH_LOG_OUTPUT("Available %ss are:\n", prefix);
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for (d = 0; audio_drivers[d]; d++)
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{
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if (audio_drivers[d])
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RARCH_LOG_OUTPUT("\t%s\n", audio_drivers[d]->ident);
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}
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RARCH_WARN("Going to default to first %s...\n", prefix);
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}
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tmp = (const audio_driver_t*)audio_drivers[0];
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if (!tmp)
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return false;
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audio_driver_st.current_audio = tmp;
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}
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return true;
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}
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/**
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* audio_driver_flush:
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* @data : pointer to audio buffer.
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* @right : amount of samples to write.
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*
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* Writes audio samples to audio driver. Will first
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* perform DSP processing (if enabled) and resampling.
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**/
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static void audio_driver_flush(
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audio_driver_state_t *audio_st,
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float slowmotion_ratio,
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bool audio_fastforward_mute,
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const int16_t *data, size_t samples,
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bool is_slowmotion, bool is_fastmotion)
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{
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struct resampler_data src_data;
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float audio_volume_gain = (audio_st->mute_enable ||
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(audio_fastforward_mute && is_fastmotion))
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? 0.0f
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: audio_st->volume_gain;
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src_data.data_out = NULL;
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src_data.output_frames = 0;
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convert_s16_to_float(audio_st->input_data, data, samples,
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audio_volume_gain);
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src_data.data_in = audio_st->input_data;
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src_data.input_frames = samples >> 1;
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#ifdef HAVE_DSP_FILTER
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if (audio_st->dsp)
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{
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struct retro_dsp_data dsp_data;
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dsp_data.input = NULL;
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dsp_data.input_frames = 0;
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dsp_data.output = NULL;
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dsp_data.output_frames = 0;
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dsp_data.input = audio_st->input_data;
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dsp_data.input_frames = (unsigned)(samples >> 1);
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retro_dsp_filter_process(audio_st->dsp, &dsp_data);
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if (dsp_data.output)
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{
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src_data.data_in = dsp_data.output;
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src_data.input_frames = dsp_data.output_frames;
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}
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}
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#endif
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src_data.data_out = audio_st->output_samples_buf;
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if (audio_st->flags & AUDIO_FLAG_CONTROL)
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{
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/* Readjust the audio input rate. */
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int half_size = (int)(audio_st->buffer_size / 2);
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int avail =
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(int)audio_st->current_audio->write_avail(
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audio_st->context_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 +
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audio_st->rate_control_delta * direction;
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unsigned write_idx =
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audio_st->free_samples_count++ &
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(AUDIO_BUFFER_FREE_SAMPLES_COUNT - 1);
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audio_st->free_samples_buf[write_idx] = avail;
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audio_st->source_ratio_current =
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audio_st->source_ratio_original * adjust;
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#if 0
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if (verbosity_is_enabled())
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{
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RARCH_LOG_OUTPUT("[Audio]: Audio buffer is %u%% full\n",
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(unsigned)(100 - (avail * 100) /
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audio_st->buffer_size));
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RARCH_LOG_OUTPUT("[Audio]: New rate: %lf, Orig rate: %lf\n",
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audio_st->source_ratio_current,
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audio_st->source_ratio_original);
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}
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#endif
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}
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src_data.ratio = audio_st->source_ratio_current;
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if (is_slowmotion)
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src_data.ratio *= slowmotion_ratio;
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/* Note: Ideally we would divide by the user-configured
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* 'fastforward_ratio' when fast forward is enabled,
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* but in practice this doesn't work:
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* - 'fastforward_ratio' is only a limit. If the host
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* cannot push frames fast enough, the actual ratio
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* will be lower - and crackling will ensue
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* - Most of the time 'fastforward_ratio' will be
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* zero (unlimited)
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* So what we would need to do is measure the time since
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* the last audio flush operation, and calculate a 'real'
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* fast-forward ratio - but this doesn't work either.
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* The measurement is inaccurate and the frame-by-frame
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* fluctuations are too large, so crackling is unavoidable.
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* Since it's going to crackle anyway, there's no point
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* trying to do anything. Just leave the ratio as-is,
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* and hope for the best... */
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audio_st->resampler->process(
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audio_st->resampler_data, &src_data);
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|
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#ifdef HAVE_AUDIOMIXER
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if (audio_st->flags & AUDIO_FLAG_MIXER_ACTIVE)
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{
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bool override = true;
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float mixer_gain = 0.0f;
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bool audio_driver_mixer_mute_enable = audio_st->mixer_mute_enable;
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|
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if (!audio_driver_mixer_mute_enable)
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{
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if (audio_st->mixer_volume_gain == 1.0f)
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override = false;
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mixer_gain = audio_st->mixer_volume_gain;
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}
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audio_mixer_mix(audio_st->output_samples_buf,
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src_data.output_frames, mixer_gain, override);
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}
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#endif
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{
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const void *output_data = audio_st->output_samples_buf;
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unsigned output_frames = (unsigned)src_data.output_frames;
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|
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if (audio_st->flags & AUDIO_FLAG_USE_FLOAT)
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output_frames *= sizeof(float);
|
|
else
|
|
{
|
|
convert_float_to_s16(audio_st->output_samples_conv_buf,
|
|
(const float*)output_data, output_frames * 2);
|
|
|
|
output_data = audio_st->output_samples_conv_buf;
|
|
output_frames *= sizeof(int16_t);
|
|
}
|
|
|
|
audio_st->current_audio->write(audio_st->context_audio_data,
|
|
output_data, output_frames * 2);
|
|
}
|
|
}
|
|
|
|
#ifdef HAVE_AUDIOMIXER
|
|
audio_mixer_stream_t *audio_driver_mixer_get_stream(unsigned i)
|
|
{
|
|
if (i > (AUDIO_MIXER_MAX_SYSTEM_STREAMS-1))
|
|
return NULL;
|
|
return &audio_driver_st.mixer_streams[i];
|
|
}
|
|
|
|
const char *audio_driver_mixer_get_stream_name(unsigned i)
|
|
{
|
|
if (i > (AUDIO_MIXER_MAX_SYSTEM_STREAMS-1))
|
|
return msg_hash_to_str(MENU_ENUM_LABEL_VALUE_NOT_AVAILABLE);
|
|
if (!string_is_empty(audio_driver_st.mixer_streams[i].name))
|
|
return audio_driver_st.mixer_streams[i].name;
|
|
return msg_hash_to_str(MENU_ENUM_LABEL_VALUE_NOT_AVAILABLE);
|
|
}
|
|
|
|
#endif
|
|
|
|
bool audio_driver_init_internal(
|
|
void *settings_data,
|
|
bool audio_cb_inited)
|
|
{
|
|
unsigned new_rate = 0;
|
|
float *samples_buf = NULL;
|
|
settings_t *settings = (settings_t*)settings_data;
|
|
size_t max_bufsamples = AUDIO_CHUNK_SIZE_NONBLOCKING * 2;
|
|
bool audio_enable = settings->bools.audio_enable;
|
|
bool audio_sync = settings->bools.audio_sync;
|
|
bool audio_rate_control = settings->bools.audio_rate_control;
|
|
float slowmotion_ratio = settings->floats.slowmotion_ratio;
|
|
runloop_state_t *runloop_st = runloop_state_get_ptr();
|
|
unsigned audio_latency = (runloop_st->audio_latency > settings->uints.audio_latency) ?
|
|
runloop_st->audio_latency : settings->uints.audio_latency;
|
|
#ifdef HAVE_REWIND
|
|
int16_t *rewind_buf = NULL;
|
|
#endif
|
|
/* Accomodate rewind since at some point we might have two full buffers. */
|
|
size_t outsamples_max = AUDIO_CHUNK_SIZE_NONBLOCKING * 2 * AUDIO_MAX_RATIO * slowmotion_ratio;
|
|
int16_t *conv_buf = (int16_t*)memalign_alloc(64, outsamples_max * sizeof(int16_t));
|
|
float *audio_buf = (float*)memalign_alloc(64, AUDIO_CHUNK_SIZE_NONBLOCKING * 2 * sizeof(float));
|
|
bool verbosity_enabled = verbosity_is_enabled();
|
|
|
|
convert_s16_to_float_init_simd();
|
|
convert_float_to_s16_init_simd();
|
|
|
|
/* Used for recording even if audio isn't enabled. */
|
|
retro_assert(conv_buf != NULL);
|
|
retro_assert(audio_buf != NULL);
|
|
|
|
if (!conv_buf || !audio_buf)
|
|
goto error;
|
|
|
|
memset(audio_buf, 0, AUDIO_CHUNK_SIZE_NONBLOCKING * 2 * sizeof(float));
|
|
|
|
audio_driver_st.input_data = audio_buf;
|
|
audio_driver_st.output_samples_conv_buf = conv_buf;
|
|
audio_driver_st.chunk_block_size = AUDIO_CHUNK_SIZE_BLOCKING;
|
|
audio_driver_st.chunk_nonblock_size = AUDIO_CHUNK_SIZE_NONBLOCKING;
|
|
audio_driver_st.chunk_size = audio_driver_st.chunk_block_size;
|
|
|
|
#ifdef HAVE_REWIND
|
|
/* Needs to be able to hold full content of a full max_bufsamples
|
|
* in addition to its own. */
|
|
rewind_buf = (int16_t*)memalign_alloc(64, max_bufsamples * sizeof(int16_t));
|
|
retro_assert(rewind_buf != NULL);
|
|
|
|
if (!rewind_buf)
|
|
goto error;
|
|
|
|
audio_driver_st.rewind_buf = rewind_buf;
|
|
audio_driver_st.rewind_size = max_bufsamples;
|
|
#endif
|
|
|
|
if (!audio_enable)
|
|
{
|
|
audio_driver_st.flags &= ~AUDIO_FLAG_ACTIVE;
|
|
return false;
|
|
}
|
|
|
|
if (!(audio_driver_find_driver(settings,
|
|
"audio driver", verbosity_enabled)))
|
|
{
|
|
RARCH_ERR("Failed to initialize audio driver.\n");
|
|
return false;
|
|
}
|
|
|
|
if (!audio_driver_st.current_audio || !audio_driver_st.current_audio->init)
|
|
{
|
|
RARCH_ERR("Failed to initialize audio driver. Will continue without audio.\n");
|
|
audio_driver_st.flags &= ~AUDIO_FLAG_ACTIVE;
|
|
return false;
|
|
}
|
|
|
|
#ifdef HAVE_THREADS
|
|
if (audio_cb_inited)
|
|
{
|
|
RARCH_LOG("[Audio]: Starting threaded audio driver ...\n");
|
|
if (!audio_init_thread(
|
|
&audio_driver_st.current_audio,
|
|
&audio_driver_st.context_audio_data,
|
|
*settings->arrays.audio_device
|
|
? settings->arrays.audio_device : NULL,
|
|
settings->uints.audio_output_sample_rate, &new_rate,
|
|
audio_latency,
|
|
settings->uints.audio_block_frames,
|
|
audio_driver_st.current_audio))
|
|
{
|
|
RARCH_ERR("Cannot open threaded audio driver ... Exiting ...\n");
|
|
return false;
|
|
}
|
|
}
|
|
else
|
|
#endif
|
|
{
|
|
audio_driver_st.context_audio_data =
|
|
audio_driver_st.current_audio->init(*settings->arrays.audio_device ?
|
|
settings->arrays.audio_device : NULL,
|
|
settings->uints.audio_output_sample_rate,
|
|
audio_latency,
|
|
settings->uints.audio_block_frames,
|
|
&new_rate);
|
|
}
|
|
|
|
if (new_rate != 0)
|
|
configuration_set_int(settings, settings->uints.audio_output_sample_rate, new_rate);
|
|
|
|
if (!audio_driver_st.context_audio_data)
|
|
{
|
|
RARCH_ERR("Failed to initialize audio driver. Will continue without audio.\n");
|
|
audio_driver_st.flags &= ~AUDIO_FLAG_ACTIVE;
|
|
}
|
|
|
|
audio_driver_st.flags &= ~AUDIO_FLAG_USE_FLOAT;
|
|
if ( (audio_driver_st.flags & AUDIO_FLAG_ACTIVE)
|
|
&& audio_driver_st.current_audio->use_float(
|
|
audio_driver_st.context_audio_data))
|
|
audio_driver_st.flags |= AUDIO_FLAG_USE_FLOAT;
|
|
|
|
if ( !audio_sync
|
|
&& (audio_driver_st.flags & AUDIO_FLAG_ACTIVE))
|
|
{
|
|
if ( (audio_driver_st.flags & AUDIO_FLAG_ACTIVE)
|
|
&& audio_driver_st.context_audio_data)
|
|
audio_driver_st.current_audio->set_nonblock_state(
|
|
audio_driver_st.context_audio_data, true);
|
|
|
|
audio_driver_st.chunk_size =
|
|
audio_driver_st.chunk_nonblock_size;
|
|
}
|
|
|
|
if (audio_driver_st.input <= 0.0f)
|
|
{
|
|
/* Should never happen. */
|
|
RARCH_WARN("[Audio]: Input rate is invalid (%.3f Hz)."
|
|
" Using output rate (%u Hz).\n",
|
|
audio_driver_st.input, settings->uints.audio_output_sample_rate);
|
|
|
|
audio_driver_st.input = settings->uints.audio_output_sample_rate;
|
|
}
|
|
|
|
audio_driver_st.source_ratio_original =
|
|
audio_driver_st.source_ratio_current =
|
|
(double)settings->uints.audio_output_sample_rate / audio_driver_st.input;
|
|
|
|
if (!string_is_empty(settings->arrays.audio_resampler))
|
|
strlcpy(audio_driver_st.resampler_ident,
|
|
settings->arrays.audio_resampler,
|
|
sizeof(audio_driver_st.resampler_ident));
|
|
else
|
|
audio_driver_st.resampler_ident[0] = '\0';
|
|
|
|
audio_driver_st.resampler_quality =
|
|
audio_driver_get_resampler_quality(settings);
|
|
|
|
if (!retro_resampler_realloc(
|
|
&audio_driver_st.resampler_data,
|
|
&audio_driver_st.resampler,
|
|
audio_driver_st.resampler_ident,
|
|
audio_driver_st.resampler_quality,
|
|
audio_driver_st.source_ratio_original))
|
|
{
|
|
RARCH_ERR("Failed to initialize resampler \"%s\".\n",
|
|
audio_driver_st.resampler_ident);
|
|
audio_driver_st.flags &= ~AUDIO_FLAG_ACTIVE;
|
|
}
|
|
|
|
audio_driver_st.data_ptr = 0;
|
|
|
|
retro_assert(settings->uints.audio_output_sample_rate <
|
|
audio_driver_st.input * AUDIO_MAX_RATIO);
|
|
|
|
samples_buf = (float*)memalign_alloc(64, outsamples_max * sizeof(float));
|
|
|
|
retro_assert(samples_buf != NULL);
|
|
|
|
if (!samples_buf)
|
|
goto error;
|
|
|
|
audio_driver_st.output_samples_buf = (float*)samples_buf;
|
|
audio_driver_st.flags &= ~AUDIO_FLAG_CONTROL;
|
|
|
|
if (
|
|
!audio_cb_inited
|
|
&& (audio_driver_st.flags & AUDIO_FLAG_ACTIVE)
|
|
&& (audio_rate_control)
|
|
)
|
|
{
|
|
/* Audio rate control requires write_avail
|
|
* and buffer_size to be implemented. */
|
|
if (audio_driver_st.current_audio->buffer_size)
|
|
{
|
|
audio_driver_st.buffer_size =
|
|
audio_driver_st.current_audio->buffer_size(
|
|
audio_driver_st.context_audio_data);
|
|
audio_driver_st.flags |= AUDIO_FLAG_CONTROL;
|
|
}
|
|
else
|
|
RARCH_WARN("[Audio]: Rate control was desired, but driver does not support needed features.\n");
|
|
}
|
|
|
|
command_event(CMD_EVENT_DSP_FILTER_INIT, NULL);
|
|
|
|
audio_driver_st.free_samples_count = 0;
|
|
|
|
#ifdef HAVE_AUDIOMIXER
|
|
audio_mixer_init(settings->uints.audio_output_sample_rate);
|
|
#endif
|
|
|
|
/* Threaded driver is initially stopped. */
|
|
if (
|
|
(audio_driver_st.flags & AUDIO_FLAG_ACTIVE)
|
|
&& audio_cb_inited
|
|
)
|
|
audio_driver_start(false);
|
|
|
|
return true;
|
|
|
|
error:
|
|
return audio_driver_deinit();
|
|
}
|
|
|
|
void audio_driver_sample(int16_t left, int16_t right)
|
|
{
|
|
audio_driver_state_t *audio_st = &audio_driver_st;
|
|
recording_state_t *recording_st = NULL;
|
|
runloop_state_t *runloop_st = NULL;
|
|
if (audio_st->flags & AUDIO_FLAG_SUSPENDED)
|
|
return;
|
|
audio_st->output_samples_conv_buf[audio_st->data_ptr++] = left;
|
|
audio_st->output_samples_conv_buf[audio_st->data_ptr++] = right;
|
|
|
|
if (audio_st->data_ptr < audio_st->chunk_size)
|
|
return;
|
|
|
|
runloop_st = runloop_state_get_ptr();
|
|
recording_st = recording_state_get_ptr();
|
|
|
|
if ( recording_st->data &&
|
|
recording_st->driver &&
|
|
recording_st->driver->push_audio)
|
|
{
|
|
struct record_audio_data ffemu_data;
|
|
|
|
ffemu_data.data = audio_st->output_samples_conv_buf;
|
|
ffemu_data.frames = audio_st->data_ptr / 2;
|
|
|
|
recording_st->driver->push_audio(recording_st->data, &ffemu_data);
|
|
}
|
|
|
|
if (!( runloop_st->paused
|
|
|| !(audio_st->flags & AUDIO_FLAG_ACTIVE)
|
|
|| !audio_st->output_samples_buf))
|
|
audio_driver_flush(audio_st,
|
|
config_get_ptr()->floats.slowmotion_ratio,
|
|
config_get_ptr()->bools.audio_fastforward_mute,
|
|
audio_st->output_samples_conv_buf,
|
|
audio_st->data_ptr,
|
|
runloop_st->slowmotion,
|
|
runloop_st->fastmotion);
|
|
|
|
audio_st->data_ptr = 0;
|
|
}
|
|
|
|
size_t audio_driver_sample_batch(const int16_t *data, size_t frames)
|
|
{
|
|
size_t frames_remaining = frames;
|
|
recording_state_t *record_st = recording_state_get_ptr();
|
|
runloop_state_t *runloop_st = runloop_state_get_ptr();
|
|
audio_driver_state_t *audio_st = &audio_driver_st;
|
|
|
|
if ((audio_st->flags & AUDIO_FLAG_SUSPENDED) || (frames < 1))
|
|
return frames;
|
|
|
|
/* We want to run this loop at least once, so use a
|
|
* do...while (do...while has only a single conditional
|
|
* jump, as opposed to for and while which have a
|
|
* conditional jump and an unconditional jump). Note,
|
|
* however, that this is only relevant for compilers
|
|
* that are poor at optimisation... */
|
|
do
|
|
{
|
|
size_t frames_to_write =
|
|
(frames_remaining > (AUDIO_CHUNK_SIZE_NONBLOCKING >> 1)) ?
|
|
(AUDIO_CHUNK_SIZE_NONBLOCKING >> 1) : frames_remaining;
|
|
|
|
if ( record_st->data
|
|
&& record_st->driver
|
|
&& record_st->driver->push_audio)
|
|
{
|
|
struct record_audio_data ffemu_data;
|
|
|
|
ffemu_data.data = data;
|
|
ffemu_data.frames = frames_to_write;
|
|
|
|
record_st->driver->push_audio(record_st->data, &ffemu_data);
|
|
}
|
|
|
|
if (!( runloop_st->paused
|
|
|| !(audio_st->flags & AUDIO_FLAG_ACTIVE)
|
|
|| !audio_st->output_samples_buf))
|
|
audio_driver_flush(audio_st,
|
|
config_get_ptr()->floats.slowmotion_ratio,
|
|
config_get_ptr()->bools.audio_fastforward_mute,
|
|
data,
|
|
frames_to_write << 1,
|
|
runloop_st->slowmotion,
|
|
runloop_st->fastmotion);
|
|
|
|
frames_remaining -= frames_to_write;
|
|
data += frames_to_write << 1;
|
|
}
|
|
while (frames_remaining > 0);
|
|
|
|
return frames;
|
|
}
|
|
|
|
#ifdef HAVE_REWIND
|
|
void audio_driver_sample_rewind(int16_t left, int16_t right)
|
|
{
|
|
audio_driver_state_t *audio_st = &audio_driver_st;
|
|
if (audio_st->rewind_ptr == 0)
|
|
return;
|
|
|
|
audio_st->rewind_buf[--audio_st->rewind_ptr] = right;
|
|
audio_st->rewind_buf[--audio_st->rewind_ptr] = left;
|
|
}
|
|
|
|
size_t audio_driver_sample_batch_rewind(
|
|
const int16_t *data, size_t frames)
|
|
{
|
|
size_t i;
|
|
audio_driver_state_t *audio_st = &audio_driver_st;
|
|
size_t samples = frames << 1;
|
|
|
|
for (i = 0; i < samples; i++)
|
|
{
|
|
if (audio_st->rewind_ptr < 1)
|
|
break;
|
|
|
|
audio_st->rewind_buf[--audio_st->rewind_ptr] = data[i];
|
|
}
|
|
|
|
return frames;
|
|
}
|
|
#endif
|
|
|
|
#ifdef HAVE_DSP_FILTER
|
|
void audio_driver_dsp_filter_free(void)
|
|
{
|
|
audio_driver_state_t *audio_st = &audio_driver_st;
|
|
if (audio_st->dsp)
|
|
retro_dsp_filter_free(audio_st->dsp);
|
|
audio_st->dsp = NULL;
|
|
}
|
|
|
|
bool audio_driver_dsp_filter_init(const char *device)
|
|
{
|
|
retro_dsp_filter_t *audio_driver_dsp = NULL;
|
|
struct string_list *plugs = NULL;
|
|
#if defined(HAVE_DYLIB) && !defined(HAVE_FILTERS_BUILTIN)
|
|
char ext_name[32];
|
|
char basedir[256];
|
|
ext_name[0] = '\0';
|
|
fill_pathname_basedir(basedir, device, sizeof(basedir));
|
|
if (!frontend_driver_get_core_extension(ext_name, sizeof(ext_name)))
|
|
return false;
|
|
if (!(plugs = dir_list_new(basedir, ext_name, false, true, false, false)))
|
|
return false;
|
|
#endif
|
|
audio_driver_dsp = retro_dsp_filter_new(
|
|
device, plugs, audio_driver_st.input);
|
|
if (!audio_driver_dsp)
|
|
return false;
|
|
|
|
audio_driver_st.dsp = audio_driver_dsp;
|
|
|
|
return true;
|
|
}
|
|
#endif
|
|
|
|
void audio_driver_set_buffer_size(size_t bufsize)
|
|
{
|
|
audio_driver_st.buffer_size = bufsize;
|
|
}
|
|
|
|
float audio_driver_monitor_adjust_system_rates(
|
|
double input_sample_rate,
|
|
double input_fps,
|
|
float video_refresh_rate,
|
|
unsigned video_swap_interval,
|
|
float audio_max_timing_skew)
|
|
{
|
|
float inp_sample_rate = input_sample_rate;
|
|
float target_video_sync_rate = video_refresh_rate
|
|
/ (float)video_swap_interval;
|
|
float timing_skew =
|
|
fabs(1.0f - input_fps / target_video_sync_rate);
|
|
if (timing_skew <= audio_max_timing_skew)
|
|
return (inp_sample_rate * target_video_sync_rate / input_fps);
|
|
return inp_sample_rate;
|
|
}
|
|
|
|
#ifdef HAVE_REWIND
|
|
void audio_driver_setup_rewind(void)
|
|
{
|
|
unsigned i;
|
|
audio_driver_state_t *audio_st = &audio_driver_st;
|
|
|
|
/* Push audio ready to be played. */
|
|
audio_st->rewind_ptr = audio_st->rewind_size;
|
|
|
|
for (i = 0; i < audio_st->data_ptr; i += 2)
|
|
{
|
|
if (audio_st->rewind_ptr > 0)
|
|
audio_st->rewind_buf[--audio_st->rewind_ptr] =
|
|
audio_st->output_samples_conv_buf[i + 1];
|
|
|
|
if (audio_st->rewind_ptr > 0)
|
|
audio_st->rewind_buf[--audio_st->rewind_ptr] =
|
|
audio_st->output_samples_conv_buf[i + 0];
|
|
}
|
|
|
|
audio_st->data_ptr = 0;
|
|
}
|
|
#endif
|
|
|
|
bool audio_driver_get_devices_list(void **data)
|
|
{
|
|
struct string_list**ptr = (struct string_list**)data;
|
|
if (!ptr)
|
|
return false;
|
|
*ptr = audio_driver_st.devices_list;
|
|
return true;
|
|
}
|
|
|
|
#ifdef HAVE_AUDIOMIXER
|
|
bool audio_driver_mixer_extension_supported(const char *ext)
|
|
{
|
|
unsigned i;
|
|
struct string_list str_list;
|
|
union string_list_elem_attr attr;
|
|
bool ret = false;
|
|
|
|
attr.i = 0;
|
|
if (!string_list_initialize(&str_list))
|
|
return false;
|
|
|
|
#ifdef HAVE_STB_VORBIS
|
|
string_list_append(&str_list, "ogg", attr);
|
|
#endif
|
|
#ifdef HAVE_IBXM
|
|
string_list_append(&str_list, "mod", attr);
|
|
string_list_append(&str_list, "s3m", attr);
|
|
string_list_append(&str_list, "xm", attr);
|
|
#endif
|
|
#ifdef HAVE_DR_FLAC
|
|
string_list_append(&str_list, "flac", attr);
|
|
#endif
|
|
#ifdef HAVE_DR_MP3
|
|
string_list_append(&str_list, "mp3", attr);
|
|
#endif
|
|
string_list_append(&str_list, "wav", attr);
|
|
|
|
for (i = 0; i < str_list.size; i++)
|
|
{
|
|
const char *str_ext = str_list.elems[i].data;
|
|
if (string_is_equal_noncase(str_ext, ext))
|
|
{
|
|
ret = true;
|
|
break;
|
|
}
|
|
}
|
|
|
|
string_list_deinitialize(&str_list);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static int audio_mixer_find_index(
|
|
audio_mixer_sound_t *sound)
|
|
{
|
|
unsigned i;
|
|
|
|
for (i = 0; i < AUDIO_MIXER_MAX_SYSTEM_STREAMS; i++)
|
|
{
|
|
audio_mixer_sound_t *handle = audio_driver_st.mixer_streams[i].handle;
|
|
if (handle == sound)
|
|
return i;
|
|
}
|
|
return -1;
|
|
}
|
|
|
|
static void audio_mixer_play_stop_cb(
|
|
audio_mixer_sound_t *sound, unsigned reason)
|
|
{
|
|
int idx = audio_mixer_find_index(sound);
|
|
|
|
switch (reason)
|
|
{
|
|
case AUDIO_MIXER_SOUND_FINISHED:
|
|
audio_mixer_destroy(sound);
|
|
|
|
if (idx >= 0)
|
|
{
|
|
unsigned i = (unsigned)idx;
|
|
|
|
if (!string_is_empty(audio_driver_st.mixer_streams[i].name))
|
|
free(audio_driver_st.mixer_streams[i].name);
|
|
|
|
audio_driver_st.mixer_streams[i].name = NULL;
|
|
audio_driver_st.mixer_streams[i].state = AUDIO_STREAM_STATE_NONE;
|
|
audio_driver_st.mixer_streams[i].volume = 0.0f;
|
|
audio_driver_st.mixer_streams[i].buf = NULL;
|
|
audio_driver_st.mixer_streams[i].stop_cb = NULL;
|
|
audio_driver_st.mixer_streams[i].handle = NULL;
|
|
audio_driver_st.mixer_streams[i].voice = NULL;
|
|
}
|
|
break;
|
|
case AUDIO_MIXER_SOUND_STOPPED:
|
|
break;
|
|
case AUDIO_MIXER_SOUND_REPEATED:
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void audio_mixer_menu_stop_cb(
|
|
audio_mixer_sound_t *sound, unsigned reason)
|
|
{
|
|
int idx = audio_mixer_find_index(sound);
|
|
|
|
switch (reason)
|
|
{
|
|
case AUDIO_MIXER_SOUND_FINISHED:
|
|
if (idx >= 0)
|
|
{
|
|
unsigned i = (unsigned)idx;
|
|
audio_driver_st.mixer_streams[i].state = AUDIO_STREAM_STATE_STOPPED;
|
|
audio_driver_st.mixer_streams[i].volume = 0.0f;
|
|
}
|
|
break;
|
|
case AUDIO_MIXER_SOUND_STOPPED:
|
|
case AUDIO_MIXER_SOUND_REPEATED:
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void audio_mixer_play_stop_sequential_cb(
|
|
audio_mixer_sound_t *sound, unsigned reason)
|
|
{
|
|
int idx = audio_mixer_find_index(sound);
|
|
|
|
switch (reason)
|
|
{
|
|
case AUDIO_MIXER_SOUND_FINISHED:
|
|
audio_mixer_destroy(sound);
|
|
|
|
if (idx >= 0)
|
|
{
|
|
unsigned i = (unsigned)idx;
|
|
|
|
if (!string_is_empty(audio_driver_st.mixer_streams[i].name))
|
|
free(audio_driver_st.mixer_streams[i].name);
|
|
|
|
if (i < AUDIO_MIXER_MAX_STREAMS)
|
|
audio_driver_st.mixer_streams[i].stream_type = AUDIO_STREAM_TYPE_USER;
|
|
else
|
|
audio_driver_st.mixer_streams[i].stream_type = AUDIO_STREAM_TYPE_SYSTEM;
|
|
|
|
audio_driver_st.mixer_streams[i].name = NULL;
|
|
audio_driver_st.mixer_streams[i].state = AUDIO_STREAM_STATE_NONE;
|
|
audio_driver_st.mixer_streams[i].volume = 0.0f;
|
|
audio_driver_st.mixer_streams[i].buf = NULL;
|
|
audio_driver_st.mixer_streams[i].stop_cb = NULL;
|
|
audio_driver_st.mixer_streams[i].handle = NULL;
|
|
audio_driver_st.mixer_streams[i].voice = NULL;
|
|
|
|
i++;
|
|
|
|
for (; i < AUDIO_MIXER_MAX_SYSTEM_STREAMS; i++)
|
|
{
|
|
if (audio_driver_st.mixer_streams[i].state
|
|
== AUDIO_STREAM_STATE_STOPPED)
|
|
{
|
|
audio_driver_mixer_play_stream_sequential(i);
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
break;
|
|
case AUDIO_MIXER_SOUND_STOPPED:
|
|
case AUDIO_MIXER_SOUND_REPEATED:
|
|
break;
|
|
}
|
|
}
|
|
|
|
static bool audio_driver_mixer_get_free_stream_slot(
|
|
unsigned *id, enum audio_mixer_stream_type type)
|
|
{
|
|
unsigned i = AUDIO_MIXER_MAX_STREAMS;
|
|
unsigned count = AUDIO_MIXER_MAX_SYSTEM_STREAMS;
|
|
|
|
if (type == AUDIO_STREAM_TYPE_USER)
|
|
{
|
|
i = 0;
|
|
count = AUDIO_MIXER_MAX_STREAMS;
|
|
}
|
|
|
|
for (; i < count; i++)
|
|
{
|
|
if (audio_driver_st.mixer_streams[i].state == AUDIO_STREAM_STATE_NONE)
|
|
{
|
|
*id = i;
|
|
return true;
|
|
}
|
|
}
|
|
|
|
return false;
|
|
}
|
|
|
|
bool audio_driver_mixer_add_stream(audio_mixer_stream_params_t *params)
|
|
{
|
|
unsigned free_slot = 0;
|
|
audio_mixer_voice_t *voice = NULL;
|
|
audio_mixer_sound_t *handle = NULL;
|
|
audio_mixer_stop_cb_t stop_cb = audio_mixer_play_stop_cb;
|
|
bool looped = (params->state == AUDIO_STREAM_STATE_PLAYING_LOOPED);
|
|
void *buf = NULL;
|
|
|
|
if (params->stream_type == AUDIO_STREAM_TYPE_NONE)
|
|
return false;
|
|
|
|
switch (params->slot_selection_type)
|
|
{
|
|
case AUDIO_MIXER_SLOT_SELECTION_MANUAL:
|
|
free_slot = params->slot_selection_idx;
|
|
|
|
/* If we are using a manually specified
|
|
* slot, must free any existing stream
|
|
* before assigning the new one */
|
|
audio_driver_mixer_stop_stream(free_slot);
|
|
audio_driver_mixer_remove_stream(free_slot);
|
|
|
|
break;
|
|
case AUDIO_MIXER_SLOT_SELECTION_AUTOMATIC:
|
|
default:
|
|
return audio_driver_mixer_get_free_stream_slot(
|
|
&free_slot, params->stream_type);
|
|
}
|
|
|
|
if (params->state == AUDIO_STREAM_STATE_NONE)
|
|
return false;
|
|
|
|
if (!(buf = malloc(params->bufsize)))
|
|
return false;
|
|
|
|
memcpy(buf, params->buf, params->bufsize);
|
|
|
|
switch (params->type)
|
|
{
|
|
case AUDIO_MIXER_TYPE_WAV:
|
|
handle = audio_mixer_load_wav(buf, (int32_t)params->bufsize,
|
|
audio_driver_st.resampler_ident,
|
|
audio_driver_st.resampler_quality);
|
|
/* WAV is a special case - input buffer is not
|
|
* free()'d when sound playback is complete (it is
|
|
* converted to a PCM buffer, which is free()'d instead),
|
|
* so have to do it here */
|
|
free(buf);
|
|
buf = NULL;
|
|
break;
|
|
case AUDIO_MIXER_TYPE_OGG:
|
|
handle = audio_mixer_load_ogg(buf, (int32_t)params->bufsize);
|
|
break;
|
|
case AUDIO_MIXER_TYPE_MOD:
|
|
handle = audio_mixer_load_mod(buf, (int32_t)params->bufsize);
|
|
break;
|
|
case AUDIO_MIXER_TYPE_FLAC:
|
|
#ifdef HAVE_DR_FLAC
|
|
handle = audio_mixer_load_flac(buf, (int32_t)params->bufsize);
|
|
#endif
|
|
break;
|
|
case AUDIO_MIXER_TYPE_MP3:
|
|
#ifdef HAVE_DR_MP3
|
|
handle = audio_mixer_load_mp3(buf, (int32_t)params->bufsize);
|
|
#endif
|
|
break;
|
|
case AUDIO_MIXER_TYPE_NONE:
|
|
break;
|
|
}
|
|
|
|
if (!handle)
|
|
{
|
|
free(buf);
|
|
return false;
|
|
}
|
|
|
|
switch (params->state)
|
|
{
|
|
case AUDIO_STREAM_STATE_PLAYING_SEQUENTIAL:
|
|
stop_cb = audio_mixer_play_stop_sequential_cb;
|
|
/* fall-through */
|
|
case AUDIO_STREAM_STATE_PLAYING_LOOPED:
|
|
case AUDIO_STREAM_STATE_PLAYING:
|
|
voice = audio_mixer_play(handle, looped, params->volume,
|
|
audio_driver_st.resampler_ident,
|
|
audio_driver_st.resampler_quality, stop_cb);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
audio_driver_st.flags |= AUDIO_FLAG_MIXER_ACTIVE;
|
|
|
|
audio_driver_st.mixer_streams[free_slot].name =
|
|
!string_is_empty(params->basename) ? strdup(params->basename) : NULL;
|
|
audio_driver_st.mixer_streams[free_slot].buf = buf;
|
|
audio_driver_st.mixer_streams[free_slot].handle = handle;
|
|
audio_driver_st.mixer_streams[free_slot].voice = voice;
|
|
audio_driver_st.mixer_streams[free_slot].stream_type = params->stream_type;
|
|
audio_driver_st.mixer_streams[free_slot].type = params->type;
|
|
audio_driver_st.mixer_streams[free_slot].state = params->state;
|
|
audio_driver_st.mixer_streams[free_slot].volume = params->volume;
|
|
audio_driver_st.mixer_streams[free_slot].stop_cb = stop_cb;
|
|
|
|
return true;
|
|
}
|
|
|
|
enum audio_mixer_state audio_driver_mixer_get_stream_state(unsigned i)
|
|
{
|
|
if (i >= AUDIO_MIXER_MAX_SYSTEM_STREAMS)
|
|
return AUDIO_STREAM_STATE_NONE;
|
|
|
|
return audio_driver_st.mixer_streams[i].state;
|
|
}
|
|
|
|
static void audio_driver_mixer_play_stream_internal(
|
|
unsigned i, unsigned type)
|
|
{
|
|
if (i >= AUDIO_MIXER_MAX_SYSTEM_STREAMS)
|
|
return;
|
|
|
|
switch (audio_driver_st.mixer_streams[i].state)
|
|
{
|
|
case AUDIO_STREAM_STATE_STOPPED:
|
|
audio_driver_st.mixer_streams[i].voice =
|
|
audio_mixer_play(audio_driver_st.mixer_streams[i].handle,
|
|
(type == AUDIO_STREAM_STATE_PLAYING_LOOPED) ? true : false,
|
|
1.0f, audio_driver_st.resampler_ident,
|
|
audio_driver_st.resampler_quality,
|
|
audio_driver_st.mixer_streams[i].stop_cb);
|
|
audio_driver_st.mixer_streams[i].state = (enum audio_mixer_state)type;
|
|
break;
|
|
case AUDIO_STREAM_STATE_PLAYING:
|
|
case AUDIO_STREAM_STATE_PLAYING_LOOPED:
|
|
case AUDIO_STREAM_STATE_PLAYING_SEQUENTIAL:
|
|
case AUDIO_STREAM_STATE_NONE:
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void audio_driver_load_menu_bgm_callback(retro_task_t *task,
|
|
void *task_data, void *user_data, const char *error)
|
|
{
|
|
bool contentless = false;
|
|
bool is_inited = false;
|
|
|
|
content_get_status(&contentless, &is_inited);
|
|
|
|
if (!is_inited)
|
|
audio_driver_mixer_play_menu_sound_looped(AUDIO_MIXER_SYSTEM_SLOT_BGM);
|
|
}
|
|
|
|
void audio_driver_load_system_sounds(void)
|
|
{
|
|
char basename_noext[256];
|
|
char sounds_path[PATH_MAX_LENGTH];
|
|
char sounds_fallback_path[PATH_MAX_LENGTH];
|
|
settings_t *settings = config_get_ptr();
|
|
const char *dir_assets = settings->paths.directory_assets;
|
|
const bool audio_enable_menu = settings->bools.audio_enable_menu;
|
|
const bool audio_enable_menu_ok = audio_enable_menu && settings->bools.audio_enable_menu_ok;
|
|
const bool audio_enable_menu_cancel = audio_enable_menu && settings->bools.audio_enable_menu_cancel;
|
|
const bool audio_enable_menu_notice = audio_enable_menu && settings->bools.audio_enable_menu_notice;
|
|
const bool audio_enable_menu_bgm = audio_enable_menu && settings->bools.audio_enable_menu_bgm;
|
|
const bool audio_enable_cheevo_unlock = settings->bools.cheevos_unlock_sound_enable;
|
|
const char *path_ok = NULL;
|
|
const char *path_cancel = NULL;
|
|
const char *path_notice = NULL;
|
|
const char *path_bgm = NULL;
|
|
const char *path_cheevo_unlock = NULL;
|
|
struct string_list *list = NULL;
|
|
struct string_list *list_fallback = NULL;
|
|
unsigned i = 0;
|
|
|
|
if (!audio_enable_menu && !audio_enable_cheevo_unlock)
|
|
goto end;
|
|
|
|
sounds_path[0] = basename_noext[0] ='\0';
|
|
|
|
fill_pathname_join_special(
|
|
sounds_fallback_path,
|
|
dir_assets,
|
|
"sounds",
|
|
sizeof(sounds_fallback_path));
|
|
|
|
fill_pathname_application_special(
|
|
sounds_path,
|
|
sizeof(sounds_path),
|
|
APPLICATION_SPECIAL_DIRECTORY_ASSETS_SOUNDS);
|
|
|
|
list = dir_list_new(sounds_path, MENU_SOUND_FORMATS, false, false, false, false);
|
|
list_fallback = dir_list_new(sounds_fallback_path, MENU_SOUND_FORMATS, false, false, false, false);
|
|
|
|
if (!list)
|
|
{
|
|
list = list_fallback;
|
|
list_fallback = NULL;
|
|
}
|
|
|
|
if (!list || list->size == 0)
|
|
goto end;
|
|
|
|
if (list_fallback && list_fallback->size > 0)
|
|
{
|
|
for (i = 0; i < list_fallback->size; i++)
|
|
{
|
|
if (list->size == 0 || !string_list_find_elem(list, list_fallback->elems[i].data))
|
|
{
|
|
union string_list_elem_attr attr = {0};
|
|
string_list_append(list, list_fallback->elems[i].data, attr);
|
|
}
|
|
}
|
|
}
|
|
|
|
for (i = 0; i < list->size; i++)
|
|
{
|
|
const char *path = list->elems[i].data;
|
|
const char *ext = path_get_extension(path);
|
|
|
|
if (audio_driver_mixer_extension_supported(ext))
|
|
{
|
|
basename_noext[0] = '\0';
|
|
fill_pathname_base(basename_noext, path, sizeof(basename_noext));
|
|
path_remove_extension(basename_noext);
|
|
|
|
if (string_is_equal_noncase(basename_noext, "ok"))
|
|
path_ok = path;
|
|
else if (string_is_equal_noncase(basename_noext, "cancel"))
|
|
path_cancel = path;
|
|
else if (string_is_equal_noncase(basename_noext, "notice"))
|
|
path_notice = path;
|
|
else if (string_is_equal_noncase(basename_noext, "bgm"))
|
|
path_bgm = path;
|
|
else if (string_is_equal_noncase(basename_noext, "unlock"))
|
|
path_cheevo_unlock = path;
|
|
}
|
|
}
|
|
|
|
if (path_ok && audio_enable_menu_ok)
|
|
task_push_audio_mixer_load(path_ok, NULL, NULL, true, AUDIO_MIXER_SLOT_SELECTION_MANUAL, AUDIO_MIXER_SYSTEM_SLOT_OK);
|
|
if (path_cancel && audio_enable_menu_cancel)
|
|
task_push_audio_mixer_load(path_cancel, NULL, NULL, true, AUDIO_MIXER_SLOT_SELECTION_MANUAL, AUDIO_MIXER_SYSTEM_SLOT_CANCEL);
|
|
if (path_notice && audio_enable_menu_notice)
|
|
task_push_audio_mixer_load(path_notice, NULL, NULL, true, AUDIO_MIXER_SLOT_SELECTION_MANUAL, AUDIO_MIXER_SYSTEM_SLOT_NOTICE);
|
|
if (path_bgm && audio_enable_menu_bgm)
|
|
task_push_audio_mixer_load(path_bgm, audio_driver_load_menu_bgm_callback, NULL, true, AUDIO_MIXER_SLOT_SELECTION_MANUAL, AUDIO_MIXER_SYSTEM_SLOT_BGM);
|
|
if (path_cheevo_unlock && audio_enable_cheevo_unlock)
|
|
task_push_audio_mixer_load(path_cheevo_unlock, NULL, NULL, true, AUDIO_MIXER_SLOT_SELECTION_MANUAL, AUDIO_MIXER_SYSTEM_SLOT_ACHIEVEMENT_UNLOCK);
|
|
|
|
end:
|
|
if (list)
|
|
string_list_free(list);
|
|
if (list_fallback)
|
|
string_list_free(list_fallback);
|
|
}
|
|
|
|
void audio_driver_mixer_play_stream(unsigned i)
|
|
{
|
|
audio_driver_st.mixer_streams[i].stop_cb = audio_mixer_play_stop_cb;
|
|
audio_driver_mixer_play_stream_internal(i, AUDIO_STREAM_STATE_PLAYING);
|
|
}
|
|
|
|
void audio_driver_mixer_play_menu_sound_looped(unsigned i)
|
|
{
|
|
audio_driver_st.mixer_streams[i].stop_cb = audio_mixer_menu_stop_cb;
|
|
audio_driver_mixer_play_stream_internal(i, AUDIO_STREAM_STATE_PLAYING_LOOPED);
|
|
}
|
|
|
|
void audio_driver_mixer_play_menu_sound(unsigned i)
|
|
{
|
|
audio_driver_st.mixer_streams[i].stop_cb = audio_mixer_menu_stop_cb;
|
|
audio_driver_mixer_play_stream_internal(i, AUDIO_STREAM_STATE_PLAYING);
|
|
}
|
|
|
|
void audio_driver_mixer_play_stream_looped(unsigned i)
|
|
{
|
|
audio_driver_st.mixer_streams[i].stop_cb = audio_mixer_play_stop_cb;
|
|
audio_driver_mixer_play_stream_internal(i, AUDIO_STREAM_STATE_PLAYING_LOOPED);
|
|
}
|
|
|
|
void audio_driver_mixer_play_stream_sequential(unsigned i)
|
|
{
|
|
audio_driver_st.mixer_streams[i].stop_cb = audio_mixer_play_stop_sequential_cb;
|
|
audio_driver_mixer_play_stream_internal(i, AUDIO_STREAM_STATE_PLAYING_SEQUENTIAL);
|
|
}
|
|
|
|
float audio_driver_mixer_get_stream_volume(unsigned i)
|
|
{
|
|
if (i >= AUDIO_MIXER_MAX_SYSTEM_STREAMS)
|
|
return 0.0f;
|
|
|
|
return audio_driver_st.mixer_streams[i].volume;
|
|
}
|
|
|
|
void audio_driver_mixer_set_stream_volume(unsigned i, float vol)
|
|
{
|
|
audio_mixer_voice_t *voice = NULL;
|
|
|
|
if (i >= AUDIO_MIXER_MAX_SYSTEM_STREAMS)
|
|
return;
|
|
|
|
audio_driver_st.mixer_streams[i].volume = vol;
|
|
|
|
voice =
|
|
audio_driver_st.mixer_streams[i].voice;
|
|
|
|
if (voice)
|
|
audio_mixer_voice_set_volume(voice, DB_TO_GAIN(vol));
|
|
}
|
|
|
|
void audio_driver_mixer_stop_stream(unsigned i)
|
|
{
|
|
if (i >= AUDIO_MIXER_MAX_SYSTEM_STREAMS)
|
|
return;
|
|
|
|
switch (audio_driver_st.mixer_streams[i].state)
|
|
{
|
|
case AUDIO_STREAM_STATE_PLAYING:
|
|
case AUDIO_STREAM_STATE_PLAYING_LOOPED:
|
|
case AUDIO_STREAM_STATE_PLAYING_SEQUENTIAL:
|
|
{
|
|
audio_mixer_voice_t *voice = audio_driver_st.mixer_streams[i].voice;
|
|
|
|
if (voice)
|
|
audio_mixer_stop(voice);
|
|
audio_driver_st.mixer_streams[i].state = AUDIO_STREAM_STATE_STOPPED;
|
|
audio_driver_st.mixer_streams[i].volume = 1.0f;
|
|
}
|
|
break;
|
|
case AUDIO_STREAM_STATE_STOPPED:
|
|
case AUDIO_STREAM_STATE_NONE:
|
|
break;
|
|
}
|
|
}
|
|
|
|
void audio_driver_mixer_remove_stream(unsigned i)
|
|
{
|
|
if (i >= AUDIO_MIXER_MAX_SYSTEM_STREAMS)
|
|
return;
|
|
|
|
switch (audio_driver_st.mixer_streams[i].state)
|
|
{
|
|
case AUDIO_STREAM_STATE_PLAYING:
|
|
case AUDIO_STREAM_STATE_PLAYING_LOOPED:
|
|
case AUDIO_STREAM_STATE_PLAYING_SEQUENTIAL:
|
|
audio_driver_mixer_stop_stream(i);
|
|
/* fall-through */
|
|
case AUDIO_STREAM_STATE_STOPPED:
|
|
{
|
|
audio_mixer_sound_t *handle = audio_driver_st.mixer_streams[i].handle;
|
|
if (handle)
|
|
audio_mixer_destroy(handle);
|
|
|
|
if (!string_is_empty(audio_driver_st.mixer_streams[i].name))
|
|
free(audio_driver_st.mixer_streams[i].name);
|
|
|
|
audio_driver_st.mixer_streams[i].state = AUDIO_STREAM_STATE_NONE;
|
|
audio_driver_st.mixer_streams[i].stop_cb = NULL;
|
|
audio_driver_st.mixer_streams[i].volume = 0.0f;
|
|
audio_driver_st.mixer_streams[i].handle = NULL;
|
|
audio_driver_st.mixer_streams[i].voice = NULL;
|
|
audio_driver_st.mixer_streams[i].name = NULL;
|
|
}
|
|
break;
|
|
case AUDIO_STREAM_STATE_NONE:
|
|
break;
|
|
}
|
|
|
|
}
|
|
|
|
bool audio_driver_mixer_toggle_mute(void)
|
|
{
|
|
audio_driver_st.mixer_mute_enable =
|
|
!audio_driver_st.mixer_mute_enable;
|
|
return true;
|
|
}
|
|
#endif
|
|
|
|
bool audio_driver_enable_callback(void)
|
|
{
|
|
if (!audio_driver_st.callback.callback)
|
|
return false;
|
|
if (audio_driver_st.callback.set_state)
|
|
audio_driver_st.callback.set_state(true);
|
|
return true;
|
|
}
|
|
|
|
bool audio_driver_disable_callback(void)
|
|
{
|
|
if (!audio_driver_st.callback.callback)
|
|
return false;
|
|
|
|
if (audio_driver_st.callback.set_state)
|
|
audio_driver_st.callback.set_state(false);
|
|
return true;
|
|
}
|
|
|
|
bool audio_driver_callback(void)
|
|
{
|
|
settings_t *settings = config_get_ptr();
|
|
bool runloop_paused = runloop_state_get_ptr()->paused;
|
|
#ifdef HAVE_MENU
|
|
#ifdef HAVE_NETWORKING
|
|
bool core_paused = runloop_paused ||
|
|
(settings->bools.menu_pause_libretro && menu_state_get_ptr()->alive &&
|
|
netplay_driver_ctl(RARCH_NETPLAY_CTL_ALLOW_PAUSE, NULL));
|
|
#else
|
|
bool core_paused = runloop_paused ||
|
|
(settings->bools.menu_pause_libretro && menu_state_get_ptr()->alive);
|
|
#endif
|
|
#else
|
|
bool core_paused = runloop_paused;
|
|
#endif
|
|
|
|
if (!audio_driver_st.callback.callback)
|
|
return false;
|
|
|
|
if (!core_paused && audio_driver_st.callback.callback)
|
|
audio_driver_st.callback.callback();
|
|
|
|
return true;
|
|
}
|
|
|
|
bool audio_driver_has_callback(void)
|
|
{
|
|
return audio_driver_st.callback.callback != NULL;
|
|
}
|
|
|
|
static INLINE bool audio_driver_alive(void)
|
|
{
|
|
audio_driver_state_t *audio_st = &audio_driver_st;
|
|
if ( audio_st->current_audio
|
|
&& audio_st->current_audio->alive
|
|
&& audio_st->context_audio_data)
|
|
return audio_st->current_audio->alive(audio_st->context_audio_data);
|
|
return false;
|
|
}
|
|
|
|
bool audio_driver_start(bool is_shutdown)
|
|
{
|
|
audio_driver_state_t *audio_st = &audio_driver_st;
|
|
if (
|
|
!audio_st->current_audio
|
|
|| !audio_st->current_audio->start
|
|
|| !audio_st->context_audio_data)
|
|
goto error;
|
|
if (!audio_st->current_audio->start(
|
|
audio_st->context_audio_data, is_shutdown))
|
|
goto error;
|
|
|
|
return true;
|
|
|
|
error:
|
|
RARCH_ERR("%s\n",
|
|
msg_hash_to_str(MSG_FAILED_TO_START_AUDIO_DRIVER));
|
|
audio_driver_st.flags &= ~AUDIO_FLAG_ACTIVE;
|
|
return false;
|
|
}
|
|
|
|
bool audio_driver_stop(void)
|
|
{
|
|
if ( !audio_driver_st.current_audio
|
|
|| !audio_driver_st.current_audio->stop
|
|
|| !audio_driver_st.context_audio_data
|
|
|| !audio_driver_alive()
|
|
)
|
|
return false;
|
|
return audio_driver_st.current_audio->stop(
|
|
audio_driver_st.context_audio_data);
|
|
}
|
|
|
|
#ifdef HAVE_REWIND
|
|
void audio_driver_frame_is_reverse(void)
|
|
{
|
|
audio_driver_state_t *audio_st = &audio_driver_st;
|
|
recording_state_t *recording_st = recording_state_get_ptr();
|
|
runloop_state_t *runloop_st = runloop_state_get_ptr();
|
|
|
|
/* We just rewound. Flush rewind audio buffer. */
|
|
if ( recording_st->data &&
|
|
recording_st->driver &&
|
|
recording_st->driver->push_audio)
|
|
{
|
|
struct record_audio_data ffemu_data;
|
|
|
|
ffemu_data.data = audio_st->rewind_buf +
|
|
audio_st->rewind_ptr;
|
|
ffemu_data.frames = (audio_st->rewind_size -
|
|
audio_st->rewind_ptr) / 2;
|
|
|
|
recording_st->driver->push_audio(
|
|
recording_st->data,
|
|
&ffemu_data);
|
|
}
|
|
|
|
if (!(
|
|
runloop_st->paused
|
|
|| !(audio_st->flags & AUDIO_FLAG_ACTIVE)
|
|
|| !audio_st->output_samples_buf))
|
|
if (!(audio_st->flags & AUDIO_FLAG_SUSPENDED))
|
|
{
|
|
settings_t *settings = config_get_ptr();
|
|
audio_driver_flush(audio_st,
|
|
settings->floats.slowmotion_ratio,
|
|
settings->bools.audio_fastforward_mute,
|
|
audio_st->rewind_buf +
|
|
audio_st->rewind_ptr,
|
|
audio_st->rewind_size -
|
|
audio_st->rewind_ptr,
|
|
runloop_st->slowmotion,
|
|
runloop_st->fastmotion);
|
|
}
|
|
}
|
|
#endif
|
|
|
|
void audio_set_float(enum audio_action action, float val)
|
|
{
|
|
switch (action)
|
|
{
|
|
case AUDIO_ACTION_VOLUME_GAIN:
|
|
audio_driver_st.volume_gain = DB_TO_GAIN(val);
|
|
break;
|
|
case AUDIO_ACTION_MIXER_VOLUME_GAIN:
|
|
#ifdef HAVE_AUDIOMIXER
|
|
audio_driver_st.mixer_volume_gain = DB_TO_GAIN(val);
|
|
#endif
|
|
break;
|
|
case AUDIO_ACTION_RATE_CONTROL_DELTA:
|
|
audio_driver_st.rate_control_delta = val;
|
|
break;
|
|
case AUDIO_ACTION_NONE:
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
float *audio_get_float_ptr(enum audio_action action)
|
|
{
|
|
switch (action)
|
|
{
|
|
case AUDIO_ACTION_RATE_CONTROL_DELTA:
|
|
return &audio_driver_st.rate_control_delta;
|
|
case AUDIO_ACTION_NONE:
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return NULL;
|
|
}
|
|
|
|
bool *audio_get_bool_ptr(enum audio_action action)
|
|
{
|
|
switch (action)
|
|
{
|
|
case AUDIO_ACTION_MIXER_MUTE_ENABLE:
|
|
#ifdef HAVE_AUDIOMIXER
|
|
return &audio_driver_st.mixer_mute_enable;
|
|
#else
|
|
break;
|
|
#endif
|
|
case AUDIO_ACTION_MUTE_ENABLE:
|
|
return &audio_driver_st.mute_enable;
|
|
case AUDIO_ACTION_NONE:
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return NULL;
|
|
}
|
|
|
|
bool audio_compute_buffer_statistics(audio_statistics_t *stats)
|
|
{
|
|
unsigned i, low_water_size, high_water_size, avg, stddev;
|
|
uint64_t accum = 0;
|
|
uint64_t accum_var = 0;
|
|
unsigned low_water_count = 0;
|
|
unsigned high_water_count = 0;
|
|
audio_driver_state_t *audio_st = &audio_driver_st;
|
|
unsigned samples = MIN(
|
|
(unsigned)audio_st->free_samples_count,
|
|
AUDIO_BUFFER_FREE_SAMPLES_COUNT);
|
|
|
|
if (samples < 3)
|
|
return false;
|
|
|
|
stats->samples = (unsigned)
|
|
audio_st->free_samples_count;
|
|
|
|
#ifdef WARPUP
|
|
/* uint64 to double not implemented, fair chance
|
|
* signed int64 to double doesn't exist either */
|
|
/* https://forums.libretro.com/t/unsupported-platform-help/13903/ */
|
|
(void)stddev;
|
|
#elif defined(_MSC_VER) && _MSC_VER <= 1200
|
|
/* FIXME: error C2520: conversion from unsigned __int64
|
|
* to double not implemented, use signed __int64 */
|
|
(void)stddev;
|
|
#else
|
|
for (i = 1; i < samples; i++)
|
|
accum += audio_st->free_samples_buf[i];
|
|
|
|
avg = (unsigned)accum / (samples - 1);
|
|
|
|
for (i = 1; i < samples; i++)
|
|
{
|
|
int diff = avg - audio_st->free_samples_buf[i];
|
|
accum_var += diff * diff;
|
|
}
|
|
|
|
stddev = (unsigned)
|
|
sqrt((double)accum_var / (samples - 2));
|
|
|
|
stats->average_buffer_saturation = (1.0f - (float)avg
|
|
/ audio_st->buffer_size) * 100.0;
|
|
stats->std_deviation_percentage = ((float)stddev
|
|
/ audio_st->buffer_size) * 100.0;
|
|
#endif
|
|
|
|
low_water_size = (unsigned)(audio_st->buffer_size * 3 / 4);
|
|
high_water_size = (unsigned)(audio_st->buffer_size / 4);
|
|
|
|
for (i = 1; i < samples; i++)
|
|
{
|
|
if (audio_st->free_samples_buf[i] >= low_water_size)
|
|
low_water_count++;
|
|
else if (audio_st->free_samples_buf[i] <= high_water_size)
|
|
high_water_count++;
|
|
}
|
|
|
|
stats->close_to_underrun = (100.0f * low_water_count) / (samples - 1);
|
|
stats->close_to_blocking = (100.0f * high_water_count) / (samples - 1);
|
|
|
|
return true;
|
|
}
|
|
|
|
#ifdef HAVE_MENU
|
|
void audio_driver_menu_sample(void)
|
|
{
|
|
static int16_t samples_buf[1024] = {0};
|
|
settings_t *settings = config_get_ptr();
|
|
video_driver_state_t *video_st = video_state_get_ptr();
|
|
runloop_state_t *runloop_st = runloop_state_get_ptr();
|
|
recording_state_t *recording_st = recording_state_get_ptr();
|
|
struct retro_system_av_info *av_info = &video_st->av_info;
|
|
const struct retro_system_timing *info =
|
|
(const struct retro_system_timing*)&av_info->timing;
|
|
unsigned sample_count = (info->sample_rate / info->fps) * 2;
|
|
audio_driver_state_t *audio_st = &audio_driver_st;
|
|
bool check_flush = !(
|
|
runloop_st->paused
|
|
|| !(audio_st->flags & AUDIO_FLAG_ACTIVE)
|
|
|| !audio_st->output_samples_buf);
|
|
if ((audio_st->flags & AUDIO_FLAG_SUSPENDED))
|
|
check_flush = false;
|
|
|
|
while (sample_count > 1024)
|
|
{
|
|
if ( recording_st->data &&
|
|
recording_st->driver &&
|
|
recording_st->driver->push_audio)
|
|
{
|
|
struct record_audio_data ffemu_data;
|
|
|
|
ffemu_data.data = samples_buf;
|
|
ffemu_data.frames = 1024 / 2;
|
|
|
|
recording_st->driver->push_audio(
|
|
recording_st->data, &ffemu_data);
|
|
}
|
|
if (check_flush)
|
|
audio_driver_flush(audio_st,
|
|
settings->floats.slowmotion_ratio,
|
|
settings->bools.audio_fastforward_mute,
|
|
samples_buf,
|
|
1024,
|
|
runloop_st->slowmotion,
|
|
runloop_st->fastmotion);
|
|
sample_count -= 1024;
|
|
}
|
|
if ( recording_st->data &&
|
|
recording_st->driver &&
|
|
recording_st->driver->push_audio)
|
|
{
|
|
struct record_audio_data ffemu_data;
|
|
|
|
ffemu_data.data = samples_buf;
|
|
ffemu_data.frames = sample_count / 2;
|
|
|
|
recording_st->driver->push_audio(
|
|
recording_st->data, &ffemu_data);
|
|
}
|
|
if (check_flush)
|
|
audio_driver_flush(audio_st,
|
|
settings->floats.slowmotion_ratio,
|
|
settings->bools.audio_fastforward_mute,
|
|
samples_buf,
|
|
sample_count,
|
|
runloop_st->slowmotion,
|
|
runloop_st->fastmotion);
|
|
}
|
|
#endif
|