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kcm_mixer.c
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kcm_mixer.c
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/**
* Originally digi.c from allegro wiki
* Original authors: KC/Milan
*
* Converted to allegro5 by Ryan Dickie
*/
/* Title: Mixer functions
*/
#include <math.h>
#include <stdio.h>
#include "allegro5/allegro_audio.h"
#include "allegro5/internal/aintern.h"
#include "allegro5/internal/aintern_audio.h"
#include "allegro5/internal/aintern_audio_cfg.h"
ALLEGRO_DEBUG_CHANNEL("audio")
typedef union {
float f32[ALLEGRO_MAX_CHANNELS]; /* max: 7.1 */
int16_t s16[ALLEGRO_MAX_CHANNELS];
void *ptr;
} SAMP_BUF;
static void maybe_lock_mutex(ALLEGRO_MUTEX *mutex)
{
if (mutex) {
al_lock_mutex(mutex);
}
}
static void maybe_unlock_mutex(ALLEGRO_MUTEX *mutex)
{
if (mutex) {
al_unlock_mutex(mutex);
}
}
/* _al_rechannel_matrix:
* This function provides a (temporary!) matrix that can be used to convert
* one channel configuration into another.
*
* Returns a pointer to a statically allocated array.
*/
static float *_al_rechannel_matrix(ALLEGRO_CHANNEL_CONF orig,
ALLEGRO_CHANNEL_CONF target, float gain, float pan)
{
/* Max 7.1 (8 channels) for input and output */
static float mat[ALLEGRO_MAX_CHANNELS][ALLEGRO_MAX_CHANNELS];
size_t dst_chans = al_get_channel_count(target);
size_t src_chans = al_get_channel_count(orig);
size_t i, j;
/* Start with a simple identity matrix */
memset(mat, 0, sizeof(mat));
for (i = 0; i < src_chans && i < dst_chans; i++) {
mat[i][i] = 1.0;
}
/* Multi-channel -> mono conversion (cuts rear/side channels) */
if (dst_chans == 1 && (orig>>4) > 1) {
for (i = 0; i < 2; i++) {
mat[0][i] = 1.0 / sqrt(2.0);
}
/* If the source has a center channel, make sure that's copied 1:1
* (perhaps it should scale the overall output?)
*/
if ((orig >> 4) & 1) {
mat[0][(orig >> 4) - 1] = 1.0;
}
}
/* Center (or mono) -> front l/r conversion */
else if (((orig >> 4) & 1) && !((target >> 4) & 1)) {
mat[0][(orig >> 4) - 1] = 1.0 / sqrt(2.0);
mat[1][(orig >> 4) - 1] = 1.0 / sqrt(2.0);
}
/* Copy LFE */
if ((orig >> 4) != (target >> 4) &&
(orig & 0xF) && (target & 0xF))
{
mat[dst_chans-1][src_chans-1] = 1.0;
}
/* Apply panning, which is supposed to maintain a constant power level.
* I took that to mean we want:
* sqrt(rgain^2 + lgain^2) = 1.0
*/
if (pan != ALLEGRO_AUDIO_PAN_NONE) {
float rgain = sqrt(( pan + 1.0f) / 2.0f);
float lgain = sqrt((-pan + 1.0f) / 2.0f);
/* I dunno what to do about >2 channels, so don't even try for now. */
for (j = 0; j < src_chans; j++) {
mat[0][j] *= lgain;
mat[1][j] *= rgain;
}
}
/* Apply gain */
if (gain != 1.0f) {
for (i = 0; i < dst_chans; i++) {
for (j = 0; j < src_chans; j++) {
mat[i][j] *= gain;
}
}
}
#ifdef DEBUGMODE
{
char debug[1024];
ALLEGRO_DEBUG("sample matrix:\n");
for (i = 0; i < dst_chans; i++) {
strcpy(debug, "");
for (j = 0; j < src_chans; j++) {
sprintf(debug + strlen(debug), " %f", mat[i][j]);
}
ALLEGRO_DEBUG("%s\n", debug);
}
}
#endif
return &mat[0][0];
}
/* _al_kcm_mixer_rejig_sample_matrix:
* Recompute the mixing matrix for a sample attached to a mixer.
* The caller must be holding the mixer mutex.
*/
void _al_kcm_mixer_rejig_sample_matrix(ALLEGRO_MIXER *mixer,
ALLEGRO_SAMPLE_INSTANCE *spl)
{
float *mat;
size_t dst_chans;
size_t src_chans;
size_t i, j;
mat = _al_rechannel_matrix(spl->spl_data.chan_conf,
mixer->ss.spl_data.chan_conf, spl->gain, spl->pan);
dst_chans = al_get_channel_count(mixer->ss.spl_data.chan_conf);
src_chans = al_get_channel_count(spl->spl_data.chan_conf);
if (!spl->matrix)
spl->matrix = al_calloc(1, src_chans * dst_chans * sizeof(float));
for (i = 0; i < dst_chans; i++) {
for (j = 0; j < src_chans; j++) {
spl->matrix[i*src_chans + j] = mat[i*ALLEGRO_MAX_CHANNELS + j];
}
}
}
/* fix_looped_position:
* When a stream loops, this will fix up the position and anything else to
* allow it to safely continue playing as expected. Returns false if it
* should stop being mixed.
*/
static bool fix_looped_position(ALLEGRO_SAMPLE_INSTANCE *spl)
{
bool is_empty;
ALLEGRO_AUDIO_STREAM *stream;
/* Looping! Should be mostly self-explanatory */
switch (spl->loop) {
case ALLEGRO_PLAYMODE_LOOP:
if (spl->loop_end - spl->loop_start != 0) {
if (spl->step > 0) {
while (spl->pos >= spl->loop_end) {
spl->pos -= (spl->loop_end - spl->loop_start);
}
}
else if (spl->step < 0) {
while (spl->pos < spl->loop_start) {
spl->pos += (spl->loop_end - spl->loop_start);
}
}
}
return true;
case ALLEGRO_PLAYMODE_BIDIR:
/* When doing bi-directional looping, you need to do a follow-up
* check for the opposite direction if a loop occurred, otherwise
* you could end up misplaced on small, high-step loops.
*/
if (spl->loop_end - spl->loop_start != 0) {
if (spl->step >= 0) {
check_forward:
if (spl->pos >= spl->loop_end) {
spl->step = -spl->step;
spl->pos = spl->loop_end - (spl->pos - spl->loop_end) - 1;
goto check_backward;
}
}
else {
check_backward:
if (spl->pos < spl->loop_start || spl->pos >= spl->loop_end) {
spl->step = -spl->step;
spl->pos = spl->loop_start + (spl->loop_start - spl->pos);
goto check_forward;
}
}
}
return true;
case ALLEGRO_PLAYMODE_ONCE:
if (spl->pos < spl->spl_data.len && spl->pos >= 0) {
return true;
}
if (spl->step >= 0)
spl->pos = 0;
else
spl->pos = spl->spl_data.len - 1;
spl->is_playing = false;
return false;
case _ALLEGRO_PLAYMODE_STREAM_ONCE:
case _ALLEGRO_PLAYMODE_STREAM_ONEDIR:
stream = (ALLEGRO_AUDIO_STREAM *)spl;
is_empty = false;
while (spl->pos >= spl->spl_data.len && stream->spl.is_playing && !is_empty) {
is_empty = !_al_kcm_refill_stream(stream);
if (is_empty && stream->is_draining) {
stream->spl.is_playing = false;
}
_al_kcm_emit_stream_events(stream);
}
return !(is_empty);
}
ASSERT(false);
return false;
}
#include "kcm_mixer_helpers.inc"
static INLINE int32_t clamp(int32_t val, int32_t min, int32_t max)
{
/* Clamp to min */
val -= min;
val &= (~val) >> 31;
val += min;
/* Clamp to max */
val -= max;
val &= val >> 31;
val += max;
return val;
}
/* Mix as many sample values as possible from the source sample into a mixer
* buffer. Implements stream_reader_t.
*
* TYPE is the type of the sample values in the mixer buffer, and
* NEXT_SAMPLE_VALUE must return a buffer of the same type.
*
* Note: Uses Bresenham to keep the precise sample position.
*/
#define BRESENHAM \
do { \
delta = spl->step > 0 ? spl->step : spl->step - spl->step_denom + 1; \
delta /= spl->step_denom; \
delta_error = spl->step - delta * spl->step_denom; \
} while (0)
#define MAKE_MIXER(NAME, NEXT_SAMPLE_VALUE, TYPE) \
static void NAME(void *source, void **vbuf, unsigned int *samples, \
ALLEGRO_AUDIO_DEPTH buffer_depth, size_t dest_maxc) \
{ \
ALLEGRO_SAMPLE_INSTANCE *spl = (ALLEGRO_SAMPLE_INSTANCE *)source; \
TYPE *buf = *vbuf; \
size_t maxc = al_get_channel_count(spl->spl_data.chan_conf); \
size_t samples_l = *samples; \
size_t c; \
int delta, delta_error; \
SAMP_BUF samp_buf; \
\
BRESENHAM; \
\
if (!spl->is_playing) \
return; \
\
while (samples_l > 0) { \
const TYPE *s; \
int old_step = spl->step; \
\
if (!fix_looped_position(spl)) \
return; \
if (old_step != spl->step) { \
BRESENHAM; \
} \
\
/* It might be worth preparing multiple sample values at once. */ \
s = (TYPE *) NEXT_SAMPLE_VALUE(&samp_buf, spl, maxc); \
\
for (c = 0; c < dest_maxc; c++) { \
ALLEGRO_STATIC_ASSERT(kcm_mixer, ALLEGRO_MAX_CHANNELS == 8); \
switch (maxc) { \
case 8: *buf += s[7] * spl->matrix[c*maxc + 7]; \
/* fall through */ \
case 7: *buf += s[6] * spl->matrix[c*maxc + 6]; \
/* fall through */ \
case 6: *buf += s[5] * spl->matrix[c*maxc + 5]; \
/* fall through */ \
case 5: *buf += s[4] * spl->matrix[c*maxc + 4]; \
/* fall through */ \
case 4: *buf += s[3] * spl->matrix[c*maxc + 3]; \
/* fall through */ \
case 3: *buf += s[2] * spl->matrix[c*maxc + 2]; \
/* fall through */ \
case 2: *buf += s[1] * spl->matrix[c*maxc + 1]; \
/* fall through */ \
case 1: *buf += s[0] * spl->matrix[c*maxc + 0]; \
/* fall through */ \
default: break; \
} \
buf++; \
} \
\
spl->pos += delta; \
spl->pos_bresenham_error += delta_error; \
if (spl->pos_bresenham_error >= spl->step_denom) { \
spl->pos++; \
spl->pos_bresenham_error -= spl->step_denom; \
} \
samples_l--; \
} \
fix_looped_position(spl); \
(void)buffer_depth; \
}
MAKE_MIXER(read_to_mixer_point_float_32, point_spl32, float)
MAKE_MIXER(read_to_mixer_linear_float_32, linear_spl32, float)
MAKE_MIXER(read_to_mixer_cubic_float_32, cubic_spl32, float)
MAKE_MIXER(read_to_mixer_point_int16_t_16, point_spl16, int16_t)
MAKE_MIXER(read_to_mixer_linear_int16_t_16, linear_spl16, int16_t)
#undef MAKE_MIXER
/* _al_kcm_mixer_read:
* Mixes the streams attached to the mixer and writes additively to the
* specified buffer (or if *buf is NULL, indicating a voice, convert it and
* set it to the buffer pointer).
*/
void _al_kcm_mixer_read(void *source, void **buf, unsigned int *samples,
ALLEGRO_AUDIO_DEPTH buffer_depth, size_t dest_maxc)
{
const ALLEGRO_MIXER *mixer;
ALLEGRO_MIXER *m = (ALLEGRO_MIXER *)source;
int maxc = al_get_channel_count(m->ss.spl_data.chan_conf);
int samples_l = *samples;
int i;
if (!m->ss.is_playing)
return;
/* Make sure the mixer buffer is big enough. */
if (m->ss.spl_data.len*maxc < samples_l*maxc) {
al_free(m->ss.spl_data.buffer.ptr);
m->ss.spl_data.buffer.ptr = al_malloc(samples_l*maxc*al_get_audio_depth_size(m->ss.spl_data.depth));
if (!m->ss.spl_data.buffer.ptr) {
_al_set_error(ALLEGRO_GENERIC_ERROR,
"Out of memory allocating mixer buffer");
m->ss.spl_data.len = 0;
return;
}
m->ss.spl_data.len = samples_l;
}
mixer = m;
/* Clear the buffer to silence. */
memset(mixer->ss.spl_data.buffer.ptr, 0, samples_l * maxc * al_get_audio_depth_size(mixer->ss.spl_data.depth));
/* Mix the streams into the mixer buffer. */
for (i = _al_vector_size(&mixer->streams) - 1; i >= 0; i--) {
ALLEGRO_SAMPLE_INSTANCE **slot = _al_vector_ref(&mixer->streams, i);
ALLEGRO_SAMPLE_INSTANCE *spl = *slot;
ASSERT(spl->spl_read);
spl->spl_read(spl, (void **) &mixer->ss.spl_data.buffer.ptr, samples,
m->ss.spl_data.depth, maxc);
}
/* Call the post-processing callback. */
if (mixer->postprocess_callback) {
mixer->postprocess_callback(mixer->ss.spl_data.buffer.ptr,
*samples, mixer->pp_callback_userdata);
}
samples_l *= maxc;
/* Apply the gain if necessary. */
if (mixer->ss.gain != 1.0f) {
float mixer_gain = mixer->ss.gain;
unsigned long i = samples_l;
switch (m->ss.spl_data.depth) {
case ALLEGRO_AUDIO_DEPTH_FLOAT32: {
float *p = mixer->ss.spl_data.buffer.f32;
while (i-- > 0) {
*p++ *= mixer_gain;
}
break;
}
case ALLEGRO_AUDIO_DEPTH_INT16: {
int16_t *p = mixer->ss.spl_data.buffer.s16;
while (i-- > 0) {
*p++ *= mixer_gain;
}
break;
}
case ALLEGRO_AUDIO_DEPTH_INT8:
case ALLEGRO_AUDIO_DEPTH_INT24:
case ALLEGRO_AUDIO_DEPTH_UINT8:
case ALLEGRO_AUDIO_DEPTH_UINT16:
case ALLEGRO_AUDIO_DEPTH_UINT24:
/* Unsupported mixer depths. */
ASSERT(false);
break;
}
}
/* Feeding to a non-voice.
* Currently we only support mixers of the same audio depth doing this.
*/
if (*buf) {
switch (m->ss.spl_data.depth) {
case ALLEGRO_AUDIO_DEPTH_FLOAT32: {
/* We don't need to clamp in the mixer yet. */
float *lbuf = *buf;
float *src = mixer->ss.spl_data.buffer.f32;
while (samples_l-- > 0) {
*lbuf += *src;
lbuf++;
src++;
}
break;
case ALLEGRO_AUDIO_DEPTH_INT16: {
int16_t *lbuf = *buf;
int16_t *src = mixer->ss.spl_data.buffer.s16;
while (samples_l-- > 0) {
int32_t x = *lbuf + *src;
if (x < -32768)
x = -32768;
else if (x > 32767)
x = 32767;
*lbuf = (int16_t)x;
lbuf++;
src++;
}
break;
}
case ALLEGRO_AUDIO_DEPTH_INT8:
case ALLEGRO_AUDIO_DEPTH_INT24:
case ALLEGRO_AUDIO_DEPTH_UINT8:
case ALLEGRO_AUDIO_DEPTH_UINT16:
case ALLEGRO_AUDIO_DEPTH_UINT24:
/* Unsupported mixer depths. */
ASSERT(false);
break;
}
}
return;
}
/* We're feeding to a voice.
* Clamp and convert the mixed data for the voice.
*/
*buf = mixer->ss.spl_data.buffer.ptr;
switch (buffer_depth & ~ALLEGRO_AUDIO_DEPTH_UNSIGNED) {
case ALLEGRO_AUDIO_DEPTH_FLOAT32:
/* Do we need to clamp? */
break;
case ALLEGRO_AUDIO_DEPTH_INT24:
switch (mixer->ss.spl_data.depth) {
case ALLEGRO_AUDIO_DEPTH_FLOAT32: {
int32_t off = ((buffer_depth & ALLEGRO_AUDIO_DEPTH_UNSIGNED)
? 0x800000 : 0);
int32_t *lbuf = mixer->ss.spl_data.buffer.s24;
float *src = mixer->ss.spl_data.buffer.f32;
while (samples_l > 0) {
*lbuf = clamp(*(src++) * ((float)0x7FFFFF + 0.5f),
~0x7FFFFF, 0x7FFFFF);
*lbuf += off;
lbuf++;
samples_l--;
}
break;
}
case ALLEGRO_AUDIO_DEPTH_INT16:
/* XXX not yet implemented */
ASSERT(false);
break;
case ALLEGRO_AUDIO_DEPTH_INT8:
case ALLEGRO_AUDIO_DEPTH_INT24:
case ALLEGRO_AUDIO_DEPTH_UINT8:
case ALLEGRO_AUDIO_DEPTH_UINT16:
case ALLEGRO_AUDIO_DEPTH_UINT24:
/* Unsupported mixer depths. */
ASSERT(false);
break;
}
break;
case ALLEGRO_AUDIO_DEPTH_INT16:
switch (mixer->ss.spl_data.depth) {
case ALLEGRO_AUDIO_DEPTH_FLOAT32: {
int16_t off = ((buffer_depth & ALLEGRO_AUDIO_DEPTH_UNSIGNED)
? 0x8000 : 0);
int16_t *lbuf = mixer->ss.spl_data.buffer.s16;
float *src = mixer->ss.spl_data.buffer.f32;
while (samples_l > 0) {
*lbuf = clamp(*(src++) * ((float)0x7FFF + 0.5f), ~0x7FFF, 0x7FFF);
*lbuf += off;
lbuf++;
samples_l--;
}
break;
}
case ALLEGRO_AUDIO_DEPTH_INT16:
/* Handle signedness differences. */
if (buffer_depth != ALLEGRO_AUDIO_DEPTH_INT16) {
int16_t *lbuf = mixer->ss.spl_data.buffer.s16;
while (samples_l > 0) {
*lbuf++ ^= 0x8000;
samples_l--;
}
}
break;
case ALLEGRO_AUDIO_DEPTH_INT8:
case ALLEGRO_AUDIO_DEPTH_INT24:
case ALLEGRO_AUDIO_DEPTH_UINT8:
case ALLEGRO_AUDIO_DEPTH_UINT16:
case ALLEGRO_AUDIO_DEPTH_UINT24:
/* Unsupported mixer depths. */
ASSERT(false);
break;
}
break;
/* Ugh, do we really want to support 8-bit output? */
case ALLEGRO_AUDIO_DEPTH_INT8:
switch (mixer->ss.spl_data.depth) {
case ALLEGRO_AUDIO_DEPTH_FLOAT32: {
int8_t off = ((buffer_depth & ALLEGRO_AUDIO_DEPTH_UNSIGNED)
? 0x80 : 0);
int8_t *lbuf = mixer->ss.spl_data.buffer.s8;
float *src = mixer->ss.spl_data.buffer.f32;
while (samples_l > 0) {
*lbuf = clamp(*(src++) * ((float)0x7F + 0.5f), ~0x7F, 0x7F);
*lbuf += off;
lbuf++;
samples_l--;
}
break;
}
case ALLEGRO_AUDIO_DEPTH_INT16:
/* XXX not yet implemented */
ASSERT(false);
break;
case ALLEGRO_AUDIO_DEPTH_INT8:
case ALLEGRO_AUDIO_DEPTH_INT24:
case ALLEGRO_AUDIO_DEPTH_UINT8:
case ALLEGRO_AUDIO_DEPTH_UINT16:
case ALLEGRO_AUDIO_DEPTH_UINT24:
/* Unsupported mixer depths. */
ASSERT(false);
break;
}
break;
case ALLEGRO_AUDIO_DEPTH_UINT8:
case ALLEGRO_AUDIO_DEPTH_UINT16:
case ALLEGRO_AUDIO_DEPTH_UINT24:
/* Impossible. */
ASSERT(false);
break;
}
(void)dest_maxc;
}
/* Function: al_create_mixer
*/
ALLEGRO_MIXER *al_create_mixer(unsigned int freq,
ALLEGRO_AUDIO_DEPTH depth, ALLEGRO_CHANNEL_CONF chan_conf)
{
ALLEGRO_MIXER *mixer;
int default_mixer_quality = ALLEGRO_MIXER_QUALITY_LINEAR;
const char *p;
/* XXX this is in the wrong place */
p = al_get_config_value(al_get_system_config(), "audio",
"default_mixer_quality");
if (p && p[0] != '\0') {
if (!_al_stricmp(p, "point")) {
ALLEGRO_INFO("Point sampling\n");
default_mixer_quality = ALLEGRO_MIXER_QUALITY_POINT;
}
else if (!_al_stricmp(p, "linear")) {
ALLEGRO_INFO("Linear interpolation\n");
default_mixer_quality = ALLEGRO_MIXER_QUALITY_LINEAR;
}
else if (!_al_stricmp(p, "cubic")) {
ALLEGRO_INFO("Cubic interpolation\n");
default_mixer_quality = ALLEGRO_MIXER_QUALITY_CUBIC;
}
}
if (!freq) {
_al_set_error(ALLEGRO_INVALID_PARAM,
"Attempted to create mixer with no frequency");
return NULL;
}
if (depth != ALLEGRO_AUDIO_DEPTH_FLOAT32 &&
depth != ALLEGRO_AUDIO_DEPTH_INT16) {
_al_set_error(ALLEGRO_INVALID_PARAM, "Unsupported mixer depth");
return NULL;
}
mixer = al_calloc(1, sizeof(ALLEGRO_MIXER));
if (!mixer) {
_al_set_error(ALLEGRO_GENERIC_ERROR,
"Out of memory allocating mixer object");
return NULL;
}
mixer->ss.is_playing = true;
mixer->ss.spl_data.free_buf = true;
mixer->ss.loop = ALLEGRO_PLAYMODE_ONCE;
/* XXX should we have a specific loop mode? */
mixer->ss.gain = 1.0f;
mixer->ss.spl_data.depth = depth;
mixer->ss.spl_data.chan_conf = chan_conf;
mixer->ss.spl_data.frequency = freq;
mixer->ss.is_mixer = true;
mixer->ss.spl_read = NULL;
mixer->quality = default_mixer_quality;
_al_vector_init(&mixer->streams, sizeof(ALLEGRO_SAMPLE_INSTANCE *));
mixer->dtor_item = _al_kcm_register_destructor("mixer", mixer, (void (*)(void *)) al_destroy_mixer);
return mixer;
}
/* Function: al_destroy_mixer
*/
void al_destroy_mixer(ALLEGRO_MIXER *mixer)
{
if (mixer) {
_al_kcm_unregister_destructor(mixer->dtor_item);
_al_kcm_destroy_sample(&mixer->ss, false);
}
}
/* This function is ALLEGRO_MIXER aware */
/* Function: al_attach_sample_instance_to_mixer
*/
bool al_attach_sample_instance_to_mixer(ALLEGRO_SAMPLE_INSTANCE *spl,
ALLEGRO_MIXER *mixer)
{
ALLEGRO_SAMPLE_INSTANCE **slot;
ASSERT(mixer);
ASSERT(spl);
/* Already referenced, do not attach. */
if (spl->parent.u.ptr) {
_al_set_error(ALLEGRO_INVALID_OBJECT,
"Attempted to attach a sample that's already attached");
return false;
}
maybe_lock_mutex(mixer->ss.mutex);
_al_kcm_stream_set_mutex(spl, mixer->ss.mutex);
slot = _al_vector_alloc_back(&mixer->streams);
if (!slot) {
if (mixer->ss.mutex) {
al_unlock_mutex(mixer->ss.mutex);
}
_al_set_error(ALLEGRO_GENERIC_ERROR,
"Out of memory allocating attachment pointers");
return false;
}
(*slot) = spl;
spl->step = (spl->spl_data.frequency) * spl->speed;
spl->step_denom = mixer->ss.spl_data.frequency;
/* Don't want to be trapped with a step value of 0. */
if (spl->step == 0) {
if (spl->speed > 0.0f)
spl->step = 1;
else
spl->step = -1;
}
/* Set the proper sample stream reader. */
ASSERT(spl->spl_read == NULL);
if (spl->is_mixer) {
spl->spl_read = _al_kcm_mixer_read;
}
else {
switch (mixer->ss.spl_data.depth) {
case ALLEGRO_AUDIO_DEPTH_FLOAT32:
switch (mixer->quality) {
case ALLEGRO_MIXER_QUALITY_POINT:
spl->spl_read = read_to_mixer_point_float_32;
break;
case ALLEGRO_MIXER_QUALITY_LINEAR:
spl->spl_read = read_to_mixer_linear_float_32;
break;
case ALLEGRO_MIXER_QUALITY_CUBIC:
spl->spl_read = read_to_mixer_cubic_float_32;
break;
}
break;
case ALLEGRO_AUDIO_DEPTH_INT16:
switch (mixer->quality) {
case ALLEGRO_MIXER_QUALITY_POINT:
spl->spl_read = read_to_mixer_point_int16_t_16;
break;
case ALLEGRO_MIXER_QUALITY_CUBIC:
ALLEGRO_WARN("Falling back to linear interpolation\n");
/* fallthrough */
case ALLEGRO_MIXER_QUALITY_LINEAR:
spl->spl_read = read_to_mixer_linear_int16_t_16;
break;
}
break;
case ALLEGRO_AUDIO_DEPTH_INT8:
case ALLEGRO_AUDIO_DEPTH_INT24:
case ALLEGRO_AUDIO_DEPTH_UINT8:
case ALLEGRO_AUDIO_DEPTH_UINT16:
case ALLEGRO_AUDIO_DEPTH_UINT24:
/* Unsupported mixer depths. */
ASSERT(false);
break;
}
_al_kcm_mixer_rejig_sample_matrix(mixer, spl);
}
spl->parent.u.mixer = mixer;
spl->parent.is_voice = false;
maybe_unlock_mutex(mixer->ss.mutex);
return true;
}
/* Function: al_attach_audio_stream_to_mixer
*/
bool al_attach_audio_stream_to_mixer(ALLEGRO_AUDIO_STREAM *stream, ALLEGRO_MIXER *mixer)
{
ASSERT(mixer);
ASSERT(stream);
return al_attach_sample_instance_to_mixer(&stream->spl, mixer);
}
/* Function: al_attach_mixer_to_mixer
*/
bool al_attach_mixer_to_mixer(ALLEGRO_MIXER *stream, ALLEGRO_MIXER *mixer)
{
ASSERT(mixer);
ASSERT(stream);
ASSERT(mixer != stream);
if (mixer->ss.spl_data.frequency != stream->ss.spl_data.frequency) {
_al_set_error(ALLEGRO_INVALID_OBJECT,
"Attempted to attach a mixer with different frequencies");
return false;
}
if (mixer->ss.spl_data.depth != stream->ss.spl_data.depth) {
_al_set_error(ALLEGRO_INVALID_OBJECT,
"Mixers of different audio depths cannot be attached to one another");
return false;
}
if (mixer->ss.spl_data.chan_conf != stream->ss.spl_data.chan_conf) {
_al_set_error(ALLEGRO_INVALID_OBJECT,
"Mixers of different channel configurations cannot be attached to one another");
return false;
}
return al_attach_sample_instance_to_mixer(&stream->ss, mixer);
}
/* Function: al_set_mixer_postprocess_callback
*/
bool al_set_mixer_postprocess_callback(ALLEGRO_MIXER *mixer,
void (*pp_callback)(void *buf, unsigned int samples, void *data),
void *pp_callback_userdata)
{
ASSERT(mixer);
maybe_lock_mutex(mixer->ss.mutex);
mixer->postprocess_callback = pp_callback;
mixer->pp_callback_userdata = pp_callback_userdata;
maybe_unlock_mutex(mixer->ss.mutex);
return true;
}
/* Function: al_get_mixer_frequency
*/
unsigned int al_get_mixer_frequency(const ALLEGRO_MIXER *mixer)
{
ASSERT(mixer);
return mixer->ss.spl_data.frequency;
}
/* Function: al_get_mixer_channels
*/
ALLEGRO_CHANNEL_CONF al_get_mixer_channels(const ALLEGRO_MIXER *mixer)
{
ASSERT(mixer);
return mixer->ss.spl_data.chan_conf;
}
/* Function: al_get_mixer_depth
*/
ALLEGRO_AUDIO_DEPTH al_get_mixer_depth(const ALLEGRO_MIXER *mixer)
{
ASSERT(mixer);
return mixer->ss.spl_data.depth;
}
/* Function: al_get_mixer_quality
*/
ALLEGRO_MIXER_QUALITY al_get_mixer_quality(const ALLEGRO_MIXER *mixer)
{
ASSERT(mixer);
return mixer->quality;
}
/* Function: al_get_mixer_gain
*/
float al_get_mixer_gain(const ALLEGRO_MIXER *mixer)
{
ASSERT(mixer);
return mixer->ss.gain;
}
/* Function: al_get_mixer_playing
*/
bool al_get_mixer_playing(const ALLEGRO_MIXER *mixer)
{
ASSERT(mixer);
return mixer->ss.is_playing;
}
/* Function: al_get_mixer_attached
*/
bool al_get_mixer_attached(const ALLEGRO_MIXER *mixer)
{
ASSERT(mixer);
return _al_vector_is_nonempty(&mixer->streams);
}
/* Function: al_set_mixer_frequency
*/
bool al_set_mixer_frequency(ALLEGRO_MIXER *mixer, unsigned int val)
{
ASSERT(mixer);
/* You can change the frequency of a mixer as long as it's not attached
* to anything.
*/
if (mixer->ss.parent.u.ptr) {
_al_set_error(ALLEGRO_INVALID_OBJECT,
"Attempted to change the frequency of an attached mixer");
return false;
}
mixer->ss.spl_data.frequency = val;
return true;
}
/* Function: al_set_mixer_quality
*/
bool al_set_mixer_quality(ALLEGRO_MIXER *mixer, ALLEGRO_MIXER_QUALITY new_quality)
{
bool ret;
ASSERT(mixer);
maybe_lock_mutex(mixer->ss.mutex);
if (mixer->quality == new_quality) {
ret = true;
}
else if (_al_vector_size(&mixer->streams) == 0) {
mixer->quality = new_quality;
ret = true;
}
else {
_al_set_error(ALLEGRO_INVALID_OBJECT,
"Attempted to change the quality of a mixer with attachments");
ret = false;
}
maybe_unlock_mutex(mixer->ss.mutex);
return ret;
}
/* Function: al_set_mixer_gain
*/
bool al_set_mixer_gain(ALLEGRO_MIXER *mixer, float new_gain)
{
int i;
ASSERT(mixer);
maybe_lock_mutex(mixer->ss.mutex);
if (mixer->ss.gain != new_gain) {
mixer->ss.gain = new_gain;
for (i = _al_vector_size(&mixer->streams) - 1; i >= 0; i--) {
ALLEGRO_SAMPLE_INSTANCE **slot = _al_vector_ref(&mixer->streams, i);
_al_kcm_mixer_rejig_sample_matrix(mixer, *slot);
}
}
maybe_unlock_mutex(mixer->ss.mutex);
return true;
}
/* Function: al_set_mixer_playing
*/
bool al_set_mixer_playing(ALLEGRO_MIXER *mixer, bool val)