ibxm: update to 20191214

- Fix music playback (sample skipping)
- Add more platforms
- Fix Mr. Boom music

Makefile.emscripten

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

Makefile.libnx

@@ -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

Makefile.switch

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

deps/ibxm/README

@@ -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).

deps/ibxm/ibxm.c

@@ -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;
+}

deps/ibxm/ibxm.h

@@ -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 );
 

libretro-common/audio/audio_mixer.c

@@ -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;