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audio.c
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audio.c
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/*
* Audio output
*
* Copyright (C) 2016
* Sandor Zsuga (Jubatian)
* Uzem (the base of CUzeBox) is copyright (C)
* David Etherton,
* Eric Anderton,
* Alec Bourque (Uze),
* Filipe Rinaldi,
* Sandor Zsuga (Jubatian),
* Matt Pandina (Artcfox)
*
* This program is free software: you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation, either version 3 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program. If not, see <http://www.gnu.org/licenses/>.
*/
#include "audio.h"
/* Uzebox's audio frequency (that is, as the emulator does it, the line
** frequency of NTSC signal, in hertz) */
#define AUDIO_LFREQ 15734U
/* Whole for the audio sample increment, bits */
#define AUDIO_INC_W 17U
/* Audio output buffer size, must be a power of 2 (48KHz samples) */
#ifndef __EMSCRIPTEN__
#define AUDIO_OUT_SIZE 2048U
#else
#define AUDIO_OUT_SIZE 1024U
#endif
/* Preferred buffer filledness. Good value depends on output buffer size.
** (A 48KHz output sample roughly equals 3 Uzebox samples, the fill must be
** at least the consumed Uzebox sample count to have a chance of smooth
** playback) */
#ifndef __EMSCRIPTEN__
#define AUDIO_FILL ((AUDIO_OUT_SIZE / 3U) * 2U)
#else
#define AUDIO_FILL ((AUDIO_OUT_SIZE / 3U) * 8U)
#endif
/* Ring buffer size, must be a power of 2 (15.7KHz Uzebox samples) */
#ifndef __EMSCRIPTEN__
#define AUDIO_BUF_SIZE (AUDIO_OUT_SIZE * 2U)
#else
#define AUDIO_BUF_SIZE (AUDIO_OUT_SIZE * 8U)
#endif
/* Emscripten notes: The total buffer size intentionally is roughly two times
** larger than the size used for native compiles, but balanced differently,
** maintaining a longer buffer within the emulator, requesting a rather small
** one for output. Normally this wouldn't be optimal (if the callback can't
** happen in time, skips occur), but Emscripten has some pecularity about
** lags, larger output buffers steeply (NOT propotionally) increasing audio
** lag. */
/* Ring buffer of samples */
static uint8 audio_buf[AUDIO_BUF_SIZE];
static auint audio_buf_r;
static auint audio_buf_w;
/* Audio device */
static auint audio_dev = 0U;
/* Calculated increment fraction, depends on hardware audio frequency */
static auint audio_inc_p = 0U;
/* Read pointer increment fraction for audio output (used to scale frequency) */
static auint audio_inc;
/* Current pointer fraction */
static auint audio_frac;
/* Previous remaining samples counts for averaging */
static auint audio_pbrem[31];
/* Previous remaining samples average count for differential correction */
static auint audio_pbrav;
/* First run after reset mark (to reset in sync) */
static boole audio_frun;
/* Frequency scaling state: enabled or disabled */
static boole audio_fs_isena = TRUE;
/*
** Audio callback
*/
void audio_callback(void* dummy, Uint8* stream, int len)
{
auint i;
auint brem;
auint brav;
auint bras;
auint xinc;
/* If this is the first run, init */
if (audio_frun){
audio_buf_r = 0U;
audio_buf_w = AUDIO_FILL;
audio_inc = audio_inc_p;
audio_frac = 0U;
audio_pbrav = audio_buf_w;
for (i = 0U; i < 31U; i++){ audio_pbrem[i] = audio_buf_w; }
memset(&(audio_buf[0]), 0x80U, sizeof(audio_buf));
audio_frun = FALSE;
}
/* Calculate averaging buffer of remaining samples. */
brem = (audio_buf_w - audio_buf_r) & (AUDIO_BUF_SIZE - 1U);
brav = brem;
bras = brem;
for (i = 0U; i < 31U; i++){ brav += audio_pbrem[i]; }
for (i = 0U; i < 7U; i++){ bras += audio_pbrem[i]; }
brav = (brav + 15U) >> 5;
bras = (bras + 3U) >> 3;
/* Frequency scaling */
if (audio_fs_isena){
/* Propotional */
if (brav < AUDIO_FILL){
if (audio_inc > (((audio_inc_p) * 50U) / 100U)){ /* Allow slow down to 50% (for too slow machines) */
audio_inc --;
}
}else if (brav > AUDIO_FILL){
if (audio_inc < (((audio_inc_p) * 105U) / 100U)){
audio_inc ++;
}
}else{}
/* Differential */
if (brav < audio_pbrav){
audio_inc --;
}else if (brav > audio_pbrav){
audio_inc ++;
}else{}
}
/* Produce a temporary increment to push the buffer's filledness towards
** the ideal point faster by a short term average. Too large buffers are
** drained faster, possibly caused by event congestion (many events firing
** at once such as after a load burst). */
xinc = audio_inc;
if (bras < AUDIO_FILL){
xinc -= (AUDIO_FILL - bras) >> 3;
if (bras < ((AUDIO_FILL * 7U) / 8U)){
xinc -= (((AUDIO_FILL * 7U) / 8U) - bras) >> 1;
}
}else{
xinc += (bras - AUDIO_FILL) >> 3;
if (bras > ((AUDIO_FILL * 9U) / 8U)){
xinc += (bras - ((AUDIO_FILL * 9U) / 8U));
}
if (bras > ((AUDIO_FILL * 3U) / 2U)){
xinc += (bras - ((AUDIO_FILL * 3U) / 2U)) << 3;
}
}
/* Sample output */
for (i = 0U; i < len; i++){
if (audio_buf_w == audio_buf_r){
stream[i] = audio_buf[(audio_buf_r - 1U) & (AUDIO_BUF_SIZE - 1U)];
}else{
stream[i] = audio_buf[audio_buf_r];
audio_frac += xinc;
if (audio_frac >= (1U << AUDIO_INC_W)){
audio_frac &= ((1U << AUDIO_INC_W) - 1U);
audio_buf_r = (audio_buf_r + 1U) & (AUDIO_BUF_SIZE - 1U);
}
}
}
/* Finalize scaling state */
for (i = 30U; i != 0U; i--){ audio_pbrem[i] = audio_pbrem[i - 1U]; }
audio_pbrem[0] = brem;
audio_pbrav = brav;
}
#ifndef USE_SDL1
#ifdef TARGET_WINDOWS_MINGW
/* On Windows try to resolve WASAPI bugs with SDL2 by prioritizing DirectSound
** if it is available. Call before initializing audio subsystem. Note that the
** emulator can use WASAPI, but on some systems / driver combinations it was
** reported to be buggy. */
static void audio_wasapi_workaround(void)
{
auint didx = SDL_GetNumAudioDrivers();
boole hasds = FALSE;
const char* drvname;
const char* dsound = "directsound";
auint dslen = strlen(dsound);
while (didx != 0U){
/* Find out whether directsound is actually available since if it isn't,
** setting the environment variable to use it will cause audio
** initialization to fail. */
didx --;
drvname = SDL_GetAudioDriver(didx);
if (drvname != NULL){
if (strncmp(dsound, drvname, dslen) == 0){
hasds = TRUE;
}
}
}
if (hasds){
/* If it is available, set environment var. to use it, but only unless it is
** already set to something else (no overwrite), so it is still possible to
** select the audio driver externally. */
SDL_setenv("SDL_AUDIODRIVER", dsound, 0);
}
}
#endif
#endif
/*
** Attempts to initialize audio. If it returns false, no sound will be
** generated.
*/
boole audio_init(void)
{
SDL_AudioSpec desired;
#ifndef USE_SDL1
SDL_AudioSpec have;
#endif
if (audio_dev == 0U){
memset(&desired, 0, sizeof(desired));
desired.freq = 48000U;
desired.format = AUDIO_U8;
desired.callback = audio_callback;
desired.channels = 1U;
desired.samples = AUDIO_OUT_SIZE;
audio_frun = TRUE;
#ifdef USE_SDL1
SDL_InitSubSystem(SDL_INIT_AUDIO);
audio_dev = 1U;
if (SDL_OpenAudio(&desired, NULL) < 0){ audio_dev = 0U; }
if (audio_dev != 0U){
SDL_PauseAudio(1);
audio_inc_p = (((auint)(AUDIO_LFREQ) << AUDIO_INC_W) / 48000U);
}
#else
#ifdef TARGET_WINDOWS_MINGW
/* On Windows SDL2 builds, attempt WASAPI workaround. */
audio_wasapi_workaround();
#endif
SDL_InitSubSystem(SDL_INIT_AUDIO);
audio_dev = SDL_OpenAudioDevice(NULL, 0, &desired, &have, SDL_AUDIO_ALLOW_FREQUENCY_CHANGE);
if (audio_dev != 0U){
SDL_PauseAudioDevice(audio_dev, 1);
audio_inc_p = (((auint)(AUDIO_LFREQ) << AUDIO_INC_W) / have.freq);
}
#endif
}
return (audio_dev != 0U);
}
/*
** Resets audio, call it along with the start of emulation.
*/
void audio_reset(void)
{
audio_frun = TRUE;
#ifdef USE_SDL1
if (audio_dev != 0U){ SDL_PauseAudio(0); }
#else
if (audio_dev != 0U){ SDL_PauseAudioDevice(audio_dev, 0); }
#endif
}
/*
** Tears down audio (if initialization succeed)
*/
void audio_quit(void)
{
if (audio_dev != 0U){
#ifdef USE_SDL1
SDL_CloseAudio();
#else
SDL_CloseAudioDevice(audio_dev);
#endif
audio_dev = 0U;
}
}
/*
** Send a frame (unsigned 8 bit samples) to the audio device. If NULL is
** passed, then silence will be added.
*/
void audio_sendframe(uint8 const* samples, auint len)
{
auint i;
auint brem = (audio_buf_r - audio_buf_w) & (AUDIO_BUF_SIZE - 1U);
if ( (brem != 0U) &&
(brem < len) ){ return; } /* Buffer full */
if (samples != NULL){
for (i = 0U; i < len; i++){
audio_buf[audio_buf_w] = samples[i];
audio_buf_w = (audio_buf_w + 1U) & (AUDIO_BUF_SIZE - 1U);
}
}else{
for (i = 0U; i < len; i++){
audio_buf[audio_buf_w] = 0x80U;
audio_buf_w = (audio_buf_w + 1U) & (AUDIO_BUF_SIZE - 1U);
}
}
}
/*
** Returns current long-term audio frequency.
*/
auint audio_getfreq(void)
{
return (48000U * audio_inc) >> AUDIO_INC_W;
}
/*
** Enables or disables frequency scaling. By default frequency scaling is
** enabled. When disabled, the long term PD controller is fixed at whatever
** frequency it determined last.
*/
void audio_freqscale_ena(boole ena)
{
audio_fs_isena = ena;
}