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ibxm: update to 20191214 

- Fix music playback (sample skipping)
- Add more platforms
- Fix Mr. Boom music
pull/11605/head
SimpleTease 2020-11-21 20:30:19 +00:00
parent 175205c1e6
commit 8017ecd2b8
7 changed files with 47 additions and 23 deletions
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1
Makefile.emscripten
View File

@ -35,6 +35,7 @@ HAVE_STATIC_AUDIO_FILTERS = 1
HAVE_STB_FONT = 1
HAVE_CONFIGFILE = 1
HAVE_CHEATS = 1
HAVE_IBXM = 1
MEMORY = 134217728

1
Makefile.libnx
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@ -50,6 +50,7 @@ HAVE_STB_FONT = 1
HAVE_CHEEVOS = 1
HAVE_CHD = 0 # disabled due to static libretro-common and libchdr conflicts between different cores
HAVE_STB_VORBIS = 1
HAVE_IBXM = 1
# RetroArch libnx useful flags
HAVE_THREADS = 1

1
Makefile.switch
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@ -40,6 +40,7 @@ else
HAVE_DYNAMIC = 1
HAVE_PATCH = 1
HAVE_CHEATS = 1
HAVE_IBXM = 1
include Makefile.common
CFLAGS += $(DEF_FLAGS)

2
deps/ibxm/README vendored
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@ -1,5 +1,5 @@
Micromod (c)2017 mumart@gmail.com
Micromod (c)2019 mumart@gmail.com
A good-quality player library for the ProTracker MOD music format
for Javascript (HTML5 Web Audio), Java, ANSI C (SDL) and Pascal (SDL).

38
deps/ibxm/ibxm.c vendored
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@ -4,7 +4,7 @@
#include "ibxm.h"
const char *IBXM_VERSION = "ibxm/ac mod/xm/s3m replay 20170901 (c)mumart@gmail.com";
const char *IBXM_VERSION = "ibxm/ac mod/xm/s3m replay 20191214 (c)mumart@gmail.com";
static const int FP_SHIFT = 15, FP_ONE = 32768, FP_MASK = 32767;
@ -787,8 +787,14 @@ static struct module* module_load_mod( struct data *data, char *message ) {
if( param == 0 && ( effect == 5 || effect == 6 ) ) {
effect -= 2;
}
if( effect == 8 && module->num_channels == 4 ) {
effect = param = 0;
if( effect == 8 ) {
if( module->num_channels == 4 ) {
effect = param = 0;
} else if( param > 128 ) {
param = 128;
} else {
param = ( param * 255 ) >> 7;
}
}
pattern_data[ pat_data_idx + 3 ] = effect;
pattern_data[ pat_data_idx + 4 ] = param;
@ -851,7 +857,7 @@ static struct module* module_load_mod( struct data *data, char *message ) {
}
/* Allocate and initialize a module from the specified data, returns NULL on error.
Message should point to a 64-character buffer to receive error messages. */
Message must point to a 64-character buffer to receive error messages. */
struct module* module_load( struct data *data, char *message ) {
char ascii[ 16 ];
struct module* module;
@ -1427,7 +1433,7 @@ static void channel_row( struct channel *channel, struct note *note ) {
channel_tremolo( channel );
break;
case 0x08: /* Set Panning.*/
channel->panning = ( channel->note.param < 128 ) ? ( channel->note.param << 1 ) : 255;
channel->panning = channel->note.param & 0xFF;
break;
case 0x0A: case 0x84: /* Vol Slide. */
if( channel->note.param > 0 ) {
@ -1901,8 +1907,9 @@ static void downsample( int *buf, int count ) {
}
}
/* Generates audio and returns the number of stereo samples written into mix_buf. */
int replay_get_audio( struct replay *replay, int *mix_buf ) {
/* Generates audio and returns the number of stereo samples written into mix_buf.
Individual channels may be excluded using the mute bitmask. */
int replay_get_audio( struct replay *replay, int *mix_buf, int mute ) {
struct channel *channel;
int idx, num_channels, tick_len = calculate_tick_len( replay->tempo, replay->sample_rate );
/* Clear output buffer. */
@ -1911,12 +1918,25 @@ int replay_get_audio( struct replay *replay, int *mix_buf ) {
num_channels = replay->module->num_channels;
for( idx = 0; idx < num_channels; idx++ ) {
channel = &replay->channels[ idx ];
channel_resample( channel, mix_buf, 0, ( tick_len + 65 ) * 2,
replay->sample_rate * 2, replay->interpolation );
if( !( mute & 1 ) ) {
channel_resample( channel, mix_buf, 0, ( tick_len + 65 ) * 2,
replay->sample_rate * 2, replay->interpolation );
}
channel_update_sample_idx( channel, tick_len * 2, replay->sample_rate * 2 );
mute >>= 1;
}
downsample( mix_buf, tick_len + 64 );
replay_volume_ramp( replay, mix_buf, tick_len );
replay_tick( replay );
return tick_len;
}
/* Returns the currently playing pattern in the sequence.*/
int replay_get_sequence_pos( struct replay *replay ) {
return replay->seq_pos;
}
/* Returns the currently playing row in the pattern. */
int replay_get_row( struct replay *replay ) {
return replay->row;
}

11
deps/ibxm/ibxm.h vendored
View File

@ -48,7 +48,7 @@ struct module {
};
/* Allocate and initialize a module from the specified data, returns NULL on error.
Message should point to a 64-character buffer to receive error messages. */
Message must point to a 64-character buffer to receive error messages. */
struct module* module_load( struct data *data, char *message );
/* Deallocate the specified module. */
void dispose_module( struct module *module );
@ -63,8 +63,13 @@ int replay_calculate_duration( struct replay *replay );
int replay_seek( struct replay *replay, int sample_pos );
/* Set the pattern in the sequence to play. The tempo is reset to the default. */
void replay_set_sequence_pos( struct replay *replay, int pos );
/* Generates audio and returns the number of stereo samples written into mix_buf. */
int replay_get_audio( struct replay *replay, int *mix_buf );
/* Generates audio and returns the number of stereo samples written into mix_buf.
Individual channels may be excluded using the mute bitmask. */
int replay_get_audio( struct replay *replay, int *mix_buf, int mute );
/* Returns the currently playing pattern in the sequence.*/
int replay_get_sequence_pos( struct replay *replay );
/* Returns the currently playing row in the pattern. */
int replay_get_row( struct replay *replay );
/* Returns the length of the output buffer required by replay_get_audio(). */
int calculate_mix_buf_len( int sample_rate );

16
libretro-common/audio/audio_mixer.c
View File

@ -996,13 +996,11 @@ static void audio_mixer_mix_mod(float* buffer, size_t num_frames,
unsigned buf_free = (unsigned)(num_frames * 2);
int* pcm = NULL;
if (voice->types.mod.position == voice->types.mod.samples)
if (voice->types.mod.samples == 0)
{
again:
temp_samples = replay_get_audio(
voice->types.mod.stream, voice->types.mod.buffer );
temp_samples *= 2; /* stereo */
voice->types.mod.stream, voice->types.mod.buffer, 0 ) * 2;
if (temp_samples == 0)
{
@ -1031,10 +1029,9 @@ again:
{
for (i = voice->types.mod.samples; i != 0; i--)
{
samplei = *pcm++ * volume;
samplef = (float)((int)samplei + 32768) / 65535.0f;
samplef = ((float)(*pcm++) + 32768.0f) / 65535.0f;
samplef = samplef * 2.0f - 1.0f;
*buffer++ += samplef;
*buffer++ += samplef * volume;
}
buf_free -= voice->types.mod.samples;
@ -1043,10 +1040,9 @@ again:
for (i = buf_free; i != 0; --i )
{
samplei = *pcm++ * volume;
samplef = (float)((int)samplei + 32768) / 65535.0f;
samplef = ((float)(*pcm++) + 32768.0f) / 65535.0f;
samplef = samplef * 2.0f - 1.0f;
*buffer++ += samplef;
*buffer++ += samplef * volume;
}
voice->types.mod.position += buf_free;