gosu 1.4.5.pre1 → 1.4.5

data/dependencies/SDL_sound/libmodplug/sndmix.c

@@ -76,12 +76,14 @@ BOOL CSoundFile_FadeSong(CSoundFile *_this, UINT msec)
 //----------------------------------
 {
 	LONG nsamples = _muldiv(msec, _this->gdwMixingFreq, 1000);
+	LONG nRampLength;
+	UINT noff;
 	if (nsamples <= 0) return FALSE;
 	if (nsamples > 0x100000) nsamples = 0x100000;
 	_this->m_nBufferCount = nsamples;
-	LONG nRampLength = _this->m_nBufferCount;
+	nRampLength = _this->m_nBufferCount;
 	// Ramp everything down
-	for (UINT noff=0; noff < _this->m_nMixChannels; noff++)
+	for (noff=0; noff < _this->m_nMixChannels; noff++)
 	{
 		MODCHANNEL *pramp = &_this->Chn[_this->ChnMix[noff]];
 		if (!pramp) continue;
@@ -128,6 +130,7 @@ UINT CSoundFile_Read(CSoundFile *_this, LPVOID lpDestBuffer, UINT cbBuffer)
 	if (_this->m_dwSongFlags & SONG_ENDREACHED) goto MixDone;
 	while (lRead > 0)
 	{
+		UINT lTotalSampleCount;
 		// Update Channel Data
 		if (!_this->m_nBufferCount)
 		{
@@ -168,7 +171,7 @@ UINT CSoundFile_Read(CSoundFile *_this, LPVOID lpDestBuffer, UINT cbBuffer)
 			X86_MonoFromStereo(_this->MixSoundBuffer, lCount);
 		}
 		nStat++;
-		UINT lTotalSampleCount = lSampleCount;
+		lTotalSampleCount = lSampleCount;
 		// Multichannel
 		if (_this->gnChannels > 2)
 		{
@@ -196,6 +199,10 @@ BOOL CSoundFile_ProcessRow(CSoundFile *_this)
 {
 	if (++_this->m_nTickCount >= _this->m_nMusicSpeed * (_this->m_nPatternDelay+1) + _this->m_nFrameDelay)
 	{
+		MODCHANNEL *pChn;
+		MODCOMMAND *m;
+		UINT nChn;
+
 		_this->m_nPatternDelay = 0;
 		_this->m_nFrameDelay = 0;
 		_this->m_nTickCount = 0;
@@ -216,10 +223,11 @@ BOOL CSoundFile_ProcessRow(CSoundFile *_this)
 						return FALSE;     //never repeat entire song
 					if (!_this->m_nRestartPos)
 					{
+						UINT i;
 						_this->m_nMusicSpeed = _this->m_nDefaultSpeed;
 						_this->m_nMusicTempo = _this->m_nDefaultTempo;
 						_this->m_nGlobalVolume = _this->m_nDefaultGlobalVolume;
-						for (UINT i=0; i<MAX_CHANNELS; i++)
+						for (i=0; i<MAX_CHANNELS; i++)
 						{
 							_this->Chn[i].dwFlags |= CHN_NOTEFADE | CHN_KEYOFF;
 							_this->Chn[i].nFadeOutVol = 0;
@@ -271,9 +279,9 @@ BOOL CSoundFile_ProcessRow(CSoundFile *_this)
 			_this->m_nNextStartRow = 0;
 		}
 		// Reset channel values
-		MODCHANNEL *pChn = _this->Chn;
-		MODCOMMAND *m = _this->Patterns[_this->m_nPattern] + _this->m_nRow * _this->m_nChannels;
-		for (UINT nChn=0; nChn<_this->m_nChannels; pChn++, nChn++, m++)
+		pChn = _this->Chn;
+		m = _this->Patterns[_this->m_nPattern] + _this->m_nRow * _this->m_nChannels;
+		for (nChn=0; nChn<_this->m_nChannels; pChn++, nChn++, m++)
 		{
 			pChn->nRowNote = m->note;
 			pChn->nRowInstr = m->instr;
@@ -311,23 +319,29 @@ BOOL CSoundFile_ProcessRow(CSoundFile *_this)
 BOOL CSoundFile_ReadNote(CSoundFile *_this)
 //-------------------------
 {
+	DWORD nMasterVol;
+	MODCHANNEL *pChn;
+	UINT nChn;
+
 	if (!CSoundFile_ProcessRow(_this)) return FALSE;
 	////////////////////////////////////////////////////////////////////////////////////
 	_this->m_nTotalCount++;
 	if (!_this->m_nMusicTempo) return FALSE;
 	_this->m_nBufferCount = (_this->gdwMixingFreq * 5 * _this->m_nTempoFactor) / (_this->m_nMusicTempo << 8);
 	// Master Volume + Pre-Amplification / Attenuation setup
-	DWORD nMasterVol;
 	{
 		int nchn32 = (_this->m_nChannels < 32) ? _this->m_nChannels : 31;
+		int realmastervol;
+		UINT attenuation;
+		DWORD mastervol;
 		if ((_this->m_nType & MOD_TYPE_IT) && (_this->m_nInstruments) && (nchn32 < 6)) nchn32 = 6;
-		int realmastervol = _this->m_nMasterVolume;
+		realmastervol = _this->m_nMasterVolume;
 		if (realmastervol > 0x80)
 		{
 			realmastervol = 0x80 + ((realmastervol - 0x80) * (nchn32+4)) / 16;
 		}
-		UINT attenuation = (_this->gdwSoundSetup & SNDMIX_AGC) ? PreAmpAGCTable[nchn32>>1] : PreAmpTable[nchn32>>1];
-		DWORD mastervol = (realmastervol * (_this->m_nSongPreAmp + 0x10)) >> 6;
+		attenuation = (_this->gdwSoundSetup & SNDMIX_AGC) ? PreAmpAGCTable[nchn32>>1] : PreAmpTable[nchn32>>1];
+		mastervol = (realmastervol * (_this->m_nSongPreAmp + 0x10)) >> 6;
 		if (mastervol > 0x200) mastervol = 0x200;
 		if ((_this->m_dwSongFlags & SONG_GLOBALFADE) && (_this->m_nGlobalFadeMaxSamples))
 		{
@@ -339,8 +353,8 @@ BOOL CSoundFile_ReadNote(CSoundFile *_this)
 	////////////////////////////////////////////////////////////////////////////////////
 	// Update channels data
 	_this->m_nMixChannels = 0;
-	MODCHANNEL *pChn = _this->Chn;
-	for (UINT nChn=0; nChn<MAX_CHANNELS; nChn++,pChn++)
+	pChn = _this->Chn;
+	for (nChn=0; nChn<MAX_CHANNELS; nChn++,pChn++)
 	{
 		if ((pChn->dwFlags & CHN_NOTEFADE) && (!(pChn->nFadeOutVol|pChn->nRightVol|pChn->nLeftVol)))
 		{
@@ -363,6 +377,8 @@ BOOL CSoundFile_ReadNote(CSoundFile *_this)
 		if ((pChn->nPeriod)	&& (pChn->nLength))
 		{
 			int vol = pChn->nVolume + pChn->nVolSwing;
+			int period, nPeriodFrac;
+			UINT freq, ninc;
 
 			if (vol < 0) vol = 0;
 			if (vol > 256) vol = 256;
@@ -398,8 +414,8 @@ BOOL CSoundFile_ReadNote(CSoundFile *_this)
 			{
 				UINT n = (pChn->nTremorParam >> 4) + (pChn->nTremorParam & 0x0F);
 				UINT ontime = pChn->nTremorParam >> 4;
-				if ((!(_this->m_nType & MOD_TYPE_IT)) || (_this->m_dwSongFlags & SONG_ITOLDEFFECTS)) { n += 2; ontime++; }
 				UINT tremcount = (UINT)pChn->nTremorCount;
+				if ((!(_this->m_nType & MOD_TYPE_IT)) || (_this->m_dwSongFlags & SONG_ITOLDEFFECTS)) { n += 2; ontime++; }
 				if (tremcount >= n) tremcount = 0;
 				if ((_this->m_nTickCount) || (_this->m_nType & (MOD_TYPE_S3M|MOD_TYPE_IT)))
 				{
@@ -420,8 +436,10 @@ BOOL CSoundFile_ReadNote(CSoundFile *_this)
 				if ((pChn->dwFlags & CHN_VOLENV) && (penv->nVolEnv))
 				{
 					int envpos = pChn->nVolEnvPosition;
+					int x1, x2, envvol;
 					UINT pt = penv->nVolEnv - 1;
-					for (UINT i=0; i<(UINT)(penv->nVolEnv-1); i++)
+					UINT i;
+					for (i=0; i<(UINT)(penv->nVolEnv-1); i++)
 					{
 						if (envpos <= penv->VolPoints[i])
 						{
@@ -429,8 +447,7 @@ BOOL CSoundFile_ReadNote(CSoundFile *_this)
 							break;
 						}
 					}
-					int x2 = penv->VolPoints[pt];
-					int x1, envvol;
+					x2 = penv->VolPoints[pt];
 					if (envpos >= x2)
 					{
 						envvol = penv->VolEnv[pt] << 2;
@@ -458,8 +475,10 @@ BOOL CSoundFile_ReadNote(CSoundFile *_this)
 				if ((pChn->dwFlags & CHN_PANENV) && (penv->nPanEnv))
 				{
 					int envpos = pChn->nPanEnvPosition;
+					int x1, x2, y2, envpan, pan;
 					UINT pt = penv->nPanEnv - 1;
-					for (UINT i=0; i<(UINT)(penv->nPanEnv-1); i++)
+					UINT i;
+					for (i=0; i<(UINT)(penv->nPanEnv-1); i++)
 					{
 						if (envpos <= penv->PanPoints[i])
 						{
@@ -467,8 +486,7 @@ BOOL CSoundFile_ReadNote(CSoundFile *_this)
 							break;
 						}
 					}
-					int x2 = penv->PanPoints[pt], y2 = penv->PanEnv[pt];
-					int x1, envpan;
+					x2 = penv->PanPoints[pt], y2 = penv->PanEnv[pt];
 					if (envpos >= x2)
 					{
 						envpan = y2;
@@ -489,7 +507,7 @@ BOOL CSoundFile_ReadNote(CSoundFile *_this)
 					}
 					if (envpan < 0) envpan = 0;
 					if (envpan > 64) envpan = 64;
-					int pan = pChn->nPan;
+					pan = pChn->nPan;
 					if (pan >= 128)
 					{
 						pan += ((envpan - 32) * (256 - pan)) / 32;
@@ -541,7 +559,7 @@ BOOL CSoundFile_ReadNote(CSoundFile *_this)
 				pChn->nRealVolume = _muldiv(vol * _this->m_nGlobalVolume, pChn->nGlobalVol * pChn->nInsVol, 1 << 20);
 			}
 			if (pChn->nPeriod < _this->m_nMinPeriod) pChn->nPeriod = _this->m_nMinPeriod;
-			int period = pChn->nPeriod;
+			period = pChn->nPeriod;
 			if ((pChn->dwFlags & (CHN_GLISSANDO|CHN_PORTAMENTO)) ==	(CHN_GLISSANDO|CHN_PORTAMENTO))
 			{
 				period = CSoundFile_GetPeriodFromNote(_this, CSoundFile_GetNoteFromPeriod(_this, period), pChn->nFineTune, pChn->nC4Speed);
@@ -568,8 +586,10 @@ BOOL CSoundFile_ReadNote(CSoundFile *_this)
 			{
 				INSTRUMENTHEADER *penv = pChn->pHeader;
 				int envpos = pChn->nPitchEnvPosition;
+				int x1, x2, envpitch;
 				UINT pt = penv->nPitchEnv - 1;
-				for (UINT i=0; i<(UINT)(penv->nPitchEnv-1); i++)
+				UINT i;
+				for (i=0; i<(UINT)(penv->nPitchEnv-1); i++)
 				{
 					if (envpos <= penv->PitchPoints[i])
 					{
@@ -577,8 +597,7 @@ BOOL CSoundFile_ReadNote(CSoundFile *_this)
 						break;
 					}
 				}
-				int x2 = penv->PitchPoints[pt];
-				int x1, envpitch;
+				x2 = penv->PitchPoints[pt];
 				if (envpos >= x2)
 				{
 					envpitch = (((int)penv->PitchEnv[pt]) - 32) * 8;
@@ -629,6 +648,7 @@ BOOL CSoundFile_ReadNote(CSoundFile *_this)
 			{
 				UINT vibpos = pChn->nVibratoPos;
 				LONG vdelta;
+				UINT vdepth;
 				switch (pChn->nVibratoType & 0x03)
 				{
 				case 1:
@@ -643,7 +663,7 @@ BOOL CSoundFile_ReadNote(CSoundFile *_this)
 				default:
 					vdelta = ModSinusTable[vibpos];
 				}
-				UINT vdepth = ((_this->m_nType != MOD_TYPE_IT) || (_this->m_dwSongFlags & SONG_ITOLDEFFECTS)) ? 6 : 7;
+				vdepth = ((_this->m_nType != MOD_TYPE_IT) || (_this->m_dwSongFlags & SONG_ITOLDEFFECTS)) ? 6 : 7;
 				vdelta = (vdelta * (int)pChn->nVibratoDepth) >> vdepth;
 				if ((_this->m_dwSongFlags & SONG_LINEARSLIDES) && (_this->m_nType & MOD_TYPE_IT))
 				{
@@ -693,11 +713,12 @@ BOOL CSoundFile_ReadNote(CSoundFile *_this)
 				if (pdelta > 256) pdelta = 256;
 				pChn->nRealPan = pdelta;
 			}
-			int nPeriodFrac = 0;
+			nPeriodFrac = 0;
 			// Instrument Auto-Vibrato
 			if ((pChn->pInstrument) && (pChn->pInstrument->nVibDepth))
 			{
 				MODINSTRUMENT *pins = pChn->pInstrument;
+				int val, n;
 				if (pins->nVibSweep == 0)
 				{
 					pChn->nAutoVibDepth = pins->nVibDepth << 8;
@@ -715,7 +736,6 @@ BOOL CSoundFile_ReadNote(CSoundFile *_this)
 						pChn->nAutoVibDepth = pins->nVibDepth << 8;
 				}
 				pChn->nAutoVibPos += pins->nVibRate;
-				int val;
 				switch(pins->nVibType)
 				{
 				case 4:	// Random
@@ -734,19 +754,20 @@ BOOL CSoundFile_ReadNote(CSoundFile *_this)
 				default:	// Sine
 					val = ft2VibratoTable[pChn->nAutoVibPos & 255];
 				}
-				int n =	((val * pChn->nAutoVibDepth) >> 8);
+				n = ((val * pChn->nAutoVibDepth) >> 8);
 				if (_this->m_nType & MOD_TYPE_IT)
 				{
 					int df1, df2;
+					UINT n1;
 					if (n < 0)
 					{
 						n = -n;
-						UINT n1 = n >> 8;
+						n1 = n >> 8;
 						df1 = LinearSlideUpTable[n1];
 						df2 = LinearSlideUpTable[n1+1];
 					} else
 					{
-						UINT n1 = n >> 8;
+						n1 = n >> 8;
 						df1 = LinearSlideDownTable[n1];
 						df2 = LinearSlideDownTable[n1+1];
 					}
@@ -776,14 +797,14 @@ BOOL CSoundFile_ReadNote(CSoundFile *_this)
 				period = _this->m_nMaxPeriod;
 				nPeriodFrac = 0;
 			}
-			UINT freq = CSoundFile_GetFreqFromPeriod(_this, period, pChn->nC4Speed, nPeriodFrac);
+			freq = CSoundFile_GetFreqFromPeriod(_this, period, pChn->nC4Speed, nPeriodFrac);
 			if ((_this->m_nType & MOD_TYPE_IT) && (freq < 256))
 			{
 				pChn->nFadeOutVol = 0;
 				pChn->dwFlags |= CHN_NOTEFADE;
 				pChn->nRealVolume = 0;
 			}
-			UINT ninc = _muldiv(freq, 0x10000, _this->gdwMixingFreq);
+			ninc = _muldiv(freq, 0x10000, _this->gdwMixingFreq);
 			if ((ninc >= 0xFFB0) && (ninc <= 0x10090)) ninc = 0x10000;
 			if (_this->m_nFreqFactor != 128) ninc = (ninc * _this->m_nFreqFactor) >> 7;
 			if (ninc > 0xFF0000) ninc = 0xFF0000;
@@ -895,6 +916,7 @@ BOOL CSoundFile_ReadNote(CSoundFile *_this)
 			// Adjusting volumes
 			if (_this->gnChannels >= 2)
 			{
+				LONG realvol;
 				int pan = ((int)pChn->nRealPan) - 128;
 				pan *= (int)_this->m_nStereoSeparation;
 				pan /= 128;
@@ -903,7 +925,7 @@ BOOL CSoundFile_ReadNote(CSoundFile *_this)
 				if (pan < 0) pan = 0;
 				if (pan > 256) pan = 256;
 				if (_this->gdwSoundSetup & SNDMIX_REVERSESTEREO) pan = 256 - pan;
-				LONG realvol = (pChn->nRealVolume * kChnMasterVol) >> (8-1);
+				realvol = (pChn->nRealVolume * kChnMasterVol) >> (8-1);
 				if (_this->gdwSoundSetup & SNDMIX_SOFTPANNING)
 				{
 					if (pan < 128)
@@ -1001,7 +1023,8 @@ BOOL CSoundFile_ReadNote(CSoundFile *_this)
 	// Checking Max Mix Channels reached: ordering by volume
 	if ((_this->m_nMixChannels >= _this->m_nMaxMixChannels) && (!(_this->gdwSoundSetup & SNDMIX_DIRECTTODISK)))
 	{
-		for (UINT i=0; i<_this->m_nMixChannels; i++)
+		UINT i=0;
+		for (; i<_this->m_nMixChannels; i++)
 		{
 			UINT j=i;
 			while ((j+1<_this->m_nMixChannels) && (_this->Chn[_this->ChnMix[j]].nRealVolume < _this->Chn[_this->ChnMix[j+1]].nRealVolume))

data/dependencies/SDL_sound/stb_vorbis.h

@@ -768,11 +768,12 @@ typedef struct
 
 typedef struct
 {
-   uint16 coupling_steps;
+  // https://github.com/nothings/stb/pull/1312
    MappingChannel *chan;
+   uint16 coupling_steps;
    uint8  submaps;
-   uint8  submap_floor[15]; // varies
-   uint8  submap_residue[15]; // varies
+   uint8  submap_floor[16]; // varies
+   uint8  submap_residue[16]; // varies
 } Mapping;
 
 typedef struct
@@ -1275,8 +1276,8 @@ static int vorbis_validate(uint8 *data)
 // called from setup only, once per code book
 // (formula implied by specification)
 //
-// hack: suppress an UBSan error caused by invalid input data.
-// see upstream:  https://github.com/nothings/stb/issues/1168.
+// suppress an UBSan error caused by invalid input data.
+// upstream:  https://github.com/nothings/stb/issues/1168.
 STB_NO_SANITIZE("float-cast-overflow")
 static int lookup1_values(int entries, int dim)
 {
@@ -1777,7 +1778,9 @@ static int codebook_decode_scalar_raw(vorb *f, Codebook *c)
    assert(!c->sparse);
    for (i=0; i < c->entries; ++i) {
       if (c->codeword_lengths[i] == NO_CODE) continue;
-      if (c->codewords[i] == (f->acc & ((1 << c->codeword_lengths[i])-1))) {
+      /* unsigned left shift for 32-bit codewords.
+       * https://github.com/nothings/stb/issues/1168 */
+      if (c->codewords[i] == (f->acc & ((1U << c->codeword_lengths[i])-1))) {
          if (f->valid_bits >= c->codeword_lengths[i]) {
             f->acc >>= c->codeword_lengths[i];
             f->valid_bits -= c->codeword_lengths[i];
@@ -3945,7 +3948,9 @@ static int start_decoder(vorb *f)
             if (values < 0) return error(f, VORBIS_invalid_setup);
             c->lookup_values = (uint32) values;
          } else {
-            c->lookup_values = c->entries * c->dimensions;
+            /* unsigned multiply to suppress (legitimate) warning.
+             * https://github.com/nothings/stb/issues/1168 */
+            c->lookup_values = (unsigned)c->entries * (unsigned)c->dimensions;
          }
          if (c->lookup_values == 0) return error(f, VORBIS_invalid_setup);
          mults = (uint16 *) setup_temp_malloc(f, sizeof(mults[0]) * c->lookup_values);
@@ -5303,8 +5308,8 @@ static int8 channel_position[7][6] =
 #ifndef STB_VORBIS_NO_FAST_SCALED_FLOAT
    typedef union {
       float f;
-      // hack: changed this to unsigned to suppress an UBSan error.
-      // see upstream: https://github.com/nothings/stb/issues/1168.
+      // changed this to unsigned to suppress an UBSan error.
+      // upstream: https://github.com/nothings/stb/issues/1168.
       unsigned int i;
    } float_conv;
    typedef char stb_vorbis_float_size_test[sizeof(float)==4 && sizeof(int) == 4];

data/dependencies/mojoAL/mojoal.c

@@ -470,6 +470,7 @@ SIMDALIGNEDSTRUCT ALsource
     ALfloat cone_outer_gain;
     ALbuffer *buffer;
     SDL_AudioStream *stream;  /* for resampling. */
+    SDL_atomic_t total_queued_buffers;   /* everything queued, playing and processed. AL_BUFFERS_QUEUED value. */
     BufferQueue buffer_queue;
     BufferQueue buffer_queue_processed;
     ALsizei offset;  /* offset in bytes for converted stream! */
@@ -567,7 +568,7 @@ static void queue_new_buffer_items_recursive(BufferQueue *queue, BufferQueueItem
         return;
     }
 
-    queue_new_buffer_items_recursive(queue, items->next);
+    queue_new_buffer_items_recursive(queue, (BufferQueueItem*)items->next);
     items->next = NULL;
     if (queue->tail) {
         queue->tail->next = items;
@@ -599,7 +600,7 @@ static void source_mark_all_buffers_processed(ALsource *src)
     while (src->buffer_queue.head) {
         void *ptr;
         BufferQueueItem *item = src->buffer_queue.head;
-        src->buffer_queue.head = item->next;
+        src->buffer_queue.head = (BufferQueueItem*)item->next;
         SDL_AtomicAdd(&src->buffer_queue.num_items, -1);
 
         /* Move it to the processed queue for alSourceUnqueueBuffers() to pick up. */
@@ -619,7 +620,7 @@ static void source_release_buffer_queue(ALCcontext *ctx, ALsource *src)
     obtain_newly_queued_buffers(&src->buffer_queue);
     if (src->buffer_queue.tail != NULL) {
         BufferQueueItem *i;
-        for (i = src->buffer_queue.head; i; i = i->next) {
+        for (i = src->buffer_queue.head; i; i = (BufferQueueItem*)i->next) {
             (void) SDL_AtomicDecRef(&i->buffer->refcount);
         }
         src->buffer_queue.tail->next = ctx->device->playback.buffer_queue_pool;
@@ -631,7 +632,7 @@ static void source_release_buffer_queue(ALCcontext *ctx, ALsource *src)
     obtain_newly_queued_buffers(&src->buffer_queue_processed);
     if (src->buffer_queue_processed.tail != NULL) {
         BufferQueueItem *i;
-        for (i = src->buffer_queue_processed.head; i; i = i->next) {
+        for (i = src->buffer_queue_processed.head; i; i = (BufferQueueItem*)i->next) {
             (void) SDL_AtomicDecRef(&i->buffer->refcount);
         }
         src->buffer_queue_processed.tail->next = ctx->device->playback.buffer_queue_pool;
@@ -764,7 +765,7 @@ ALCboolean alcCloseDevice(ALCdevice *device)
 
     item = device->playback.buffer_queue_pool;
     while (item) {
-        BufferQueueItem *next = item->next;
+        BufferQueueItem *next = (BufferQueueItem*)item->next;
         SDL_free(item);
         item = next;
     }
@@ -1165,11 +1166,11 @@ static void mix_float32_c2_neon(const ALfloat * restrict panning, const float *
 *
 * Permission to use, copy, modify, distribute and sell this software and its
 * documentation for any purpose is hereby granted without fee, provided that
-* the above copyright notice and this license appear in all source copies. 
+* the above copyright notice and this license appear in all source copies.
 * THIS SOFTWARE IS PROVIDED "AS IS" WITHOUT EXPRESS OR IMPLIED WARRANTY OF
 * ANY KIND. See http://www.dspguru.com/wol.htm for more information.
 *
-*****************************************************************************/ 
+*****************************************************************************/
 
 /* FFT routine, (C)1996 S.M.Bernsee. */
 static void pitch_fft(float *fftBuffer, int fftFrameSize, int sign)
@@ -1300,12 +1301,12 @@ static void pitch_shift(ALsource *src, const ALbuffer *buffer, int numSampsToPro
             SDL_memset(state->synmagn, '\0', sizeof (state->synmagn));
             for (k = 0; k <= pitch_framesize2; k++) {
                 index = k*pitchShift;
-                if (index <= pitch_framesize2) { 
+                if (index <= pitch_framesize2) {
                     state->synmagn[index] += state->workspace[2*k];
                     state->synfreq[index] = state->workspace[2*k+1] * pitchShift;
-                } 
+                }
             }
-            
+
             /* ***************** SYNTHESIS ******************* */
             /* this is the synthesis step */
             for (k = 0; k <= pitch_framesize2; k++) {
@@ -1333,7 +1334,7 @@ static void pitch_shift(ALsource *src, const ALbuffer *buffer, int numSampsToPro
                 /* get real and imag part and re-interleave */
                 state->workspace[2*k] = magn*SDL_cos(phase);
                 state->workspace[2*k+1] = magn*SDL_sin(phase);
-            } 
+            }
 
             /* zero negative frequencies */
             for (k = pitch_framesize+2; k < 2*pitch_framesize; k++) state->workspace[k] = 0.;
@@ -1341,7 +1342,7 @@ static void pitch_shift(ALsource *src, const ALbuffer *buffer, int numSampsToPro
             /* do inverse transform */
             pitch_fft(state->workspace, pitch_framesize, 1);
 
-            /* do windowing and add to output accumulator */ 
+            /* do windowing and add to output accumulator */
             for(k=0; k < pitch_framesize; k++) {
                 window = -.5*SDL_cos(2.*M_PI*(double)k/(double)pitch_framesize)+.5;
                 state->outputaccum[k] += 2.*window*state->workspace[2*k]/(pitch_framesize2*osamp);
@@ -1466,7 +1467,7 @@ static ALCboolean mix_source_buffer_queue(ALCcontext *ctx, ALsource *src, Buffer
     while ((len > 0) && (mix_source_buffer(ctx, src, queue, &stream, &len))) {
         /* Finished this buffer! */
         BufferQueueItem *item = queue;
-        BufferQueueItem *next = queue ? queue->next : NULL;
+        BufferQueueItem *next = queue ? (BufferQueueItem*)queue->next : NULL;
         void *ptr;
 
         if (queue) {
@@ -2569,6 +2570,16 @@ static void _alcGetIntegerv(ALCdevice *device, const ALCenum param, const ALCsiz
             *values = OPENAL_VERSION_MINOR;
             return;
 
+        case ALC_FREQUENCY:
+            if (!device) {
+                *values = 0;
+                set_alc_error(device, ALC_INVALID_DEVICE);
+                return;
+            }
+
+            *values = device->frequency;
+            return;
+
         default: break;
     }
 
@@ -3589,6 +3600,7 @@ static void _alGenSources(const ALsizei n, ALuint *names)
 
         SDL_zerop(src);
         SDL_AtomicSet(&src->state, AL_INITIAL);
+        SDL_AtomicSet(&src->total_queued_buffers, 0);
         src->name = names[i];
         src->type = AL_UNDETERMINED;
         src->recalc = AL_TRUE;
@@ -3906,7 +3918,7 @@ static void _alGetSourcefv(const ALuint name, const ALenum param, ALfloat *value
         case AL_REFERENCE_DISTANCE: *values = src->reference_distance; break;
         case AL_ROLLOFF_FACTOR: *values = src->rolloff_factor; break;
         case AL_MAX_DISTANCE: *values = src->max_distance; break;
-        case AL_PITCH: source_set_pitch(ctx, src, *values); break;
+        case AL_PITCH: *values = src->pitch; break;
         case AL_CONE_INNER_ANGLE: *values = src->cone_inner_angle; break;
         case AL_CONE_OUTER_ANGLE: *values = src->cone_outer_angle; break;
         case AL_CONE_OUTER_GAIN:  *values = src->cone_outer_gain; break;
@@ -3973,8 +3985,7 @@ static void _alGetSourceiv(const ALuint name, const ALenum param, ALint *values)
         case AL_SOURCE_STATE: *values = (ALint) SDL_AtomicGet(&src->state); break;
         case AL_SOURCE_TYPE: *values = (ALint) src->type; break;
         case AL_BUFFER: *values = (ALint) (src->buffer ? src->buffer->name : 0); break;
-        /* !!! FIXME: AL_BUFFERS_QUEUED is the total number of buffers pending, playing, and processed, so this is wrong. It might also have to be 1 if there's a static buffer, but I'm not sure. */
-        case AL_BUFFERS_QUEUED: *values = (ALint) SDL_AtomicGet(&src->buffer_queue.num_items); break;
+        case AL_BUFFERS_QUEUED: *values = (ALint) SDL_AtomicGet(&src->total_queued_buffers); break;
         case AL_BUFFERS_PROCESSED: *values = (ALint) SDL_AtomicGet(&src->buffer_queue_processed.num_items); break;
         case AL_SOURCE_RELATIVE: *values = (ALint) src->source_relative; break;
         case AL_LOOPING: *values = (ALint) src->looping; break;
@@ -4134,7 +4145,7 @@ static void source_play(ALCcontext *ctx, const ALsizei n, const ALuint *names)
        with it, as nothing else owns the pointers in that list. */
     do {
         ptr = SDL_AtomicGetPtr(&ctx->playlist_todo);
-        todoend->next = ptr;
+        todoend->next = (SourcePlayTodo*)ptr;
     } while (!SDL_AtomicCASPtr(&ctx->playlist_todo, ptr, todo.next));
 }
 
@@ -4162,6 +4173,9 @@ static void source_stop(ALCcontext *ctx, const ALuint name)
             }
             SDL_AtomicSet(&src->state, AL_STOPPED);
             source_mark_all_buffers_processed(src);
+            if (src->stream) {
+                SDL_AudioStreamClear(src->stream);
+            }
             if (must_lock) {
                 SDL_UnlockMutex(ctx->source_lock);
             }
@@ -4228,9 +4242,10 @@ static void source_set_offset(ALsource *src, ALenum param, ALfloat value)
     if (!ctx) {
         set_al_error(ctx, AL_INVALID_OPERATION);
         return;
-    }
-
-    if (src->type == AL_STREAMING) {
+    } else if (src->type == AL_UNDETERMINED) {  /* no buffer to seek in */
+        set_al_error(ctx, AL_INVALID_OPERATION);
+        return;
+    } else if (src->type == AL_STREAMING) {
         FIXME("set_offset for streaming sources not implemented");
         return;
     }
@@ -4347,7 +4362,7 @@ static void _alSourceQueueBuffers(const ALuint name, const ALsizei nb, const ALu
 
         item = ctx->device->playback.buffer_queue_pool;
         if (item) {
-            ctx->device->playback.buffer_queue_pool = item->next;
+            ctx->device->playback.buffer_queue_pool = (BufferQueueItem*)item->next;
         } else {  /* allocate a new item */
             item = (BufferQueueItem *) SDL_calloc(1, sizeof (BufferQueueItem));
             if (!item) {
@@ -4400,7 +4415,7 @@ static void _alSourceQueueBuffers(const ALuint name, const ALsizei nb, const ALu
         if (queue) {
             /* Drop our claim on any buffers we planned to queue. */
             BufferQueueItem *item;
-            for (item = queue; item != NULL; item = item->next) {
+            for (item = queue; item != NULL; item = (BufferQueueItem*)item->next) {
                 if (item->buffer) {
                     (void) SDL_AtomicDecRef(&item->buffer->refcount);
                 }
@@ -4446,6 +4461,7 @@ static void _alSourceQueueBuffers(const ALuint name, const ALsizei nb, const ALu
         SDL_AtomicSetPtr(&queueend->next, ptr);
     } while (!SDL_AtomicCASPtr(&src->buffer_queue.just_queued, ptr, queue));
 
+    SDL_AtomicAdd(&src->total_queued_buffers, (int) nb);
     SDL_AtomicAdd(&src->buffer_queue.num_items, (int) nb);
 }
 ENTRYPOINTVOID(alSourceQueueBuffers,(ALuint name, ALsizei nb, const ALuint *bufnames),(name,nb,bufnames))
@@ -4477,6 +4493,7 @@ static void _alSourceUnqueueBuffers(const ALuint name, const ALsizei nb, ALuint
     }
 
     SDL_AtomicAdd(&src->buffer_queue_processed.num_items, -((int) nb));
+    SDL_AtomicAdd(&src->total_queued_buffers, -((int) nb));
 
     obtain_newly_queued_buffers(&src->buffer_queue_processed);
 
@@ -4484,7 +4501,7 @@ static void _alSourceUnqueueBuffers(const ALuint name, const ALsizei nb, ALuint
     for (i = 0; i < nb; i++) {
         /* buffer_queue_processed.num_items said list was long enough. */
         SDL_assert(item != NULL);
-        item = item->next;
+        item = (BufferQueueItem*)item->next;
     }
     src->buffer_queue_processed.head = item;
     if (!item) {
@@ -4498,7 +4515,7 @@ static void _alSourceUnqueueBuffers(const ALuint name, const ALsizei nb, ALuint
         }
         bufnames[i] = item->buffer ? item->buffer->name : 0;
         queueend = item;
-        item = item->next;
+        item = (BufferQueueItem*)item->next;
     }
 
     /* put the whole new queue back in the pool for reuse later. */

data/dependencies/utf8proc/utf8proc.c

@@ -101,7 +101,7 @@ UTF8PROC_DLLEXPORT const char *utf8proc_version(void) {
 }
 
 UTF8PROC_DLLEXPORT const char *utf8proc_unicode_version(void) {
-  return "14.0.0";
+  return "15.0.0";
 }
 
 UTF8PROC_DLLEXPORT const char *utf8proc_errmsg(utf8proc_ssize_t errcode) {

data/dependencies/utf8proc/utf8proc.h

@@ -71,7 +71,7 @@
 /** The MAJOR version number (increased when backwards API compatibility is broken). */
 #define UTF8PROC_VERSION_MAJOR 2
 /** The MINOR version number (increased when new functionality is added in a backwards-compatible manner). */
-#define UTF8PROC_VERSION_MINOR 7
+#define UTF8PROC_VERSION_MINOR 8
 /** The PATCH version (increased for fixes that do not change the API). */
 #define UTF8PROC_VERSION_PATCH 0
 /** @} */