Version 0.11.25

Public issue https://github.com/mackron/miniaudio/pull/1095

CHANGES.md

@@ -1,3 +1,10 @@
+v0.11.25 - 2026-03-04
+=====================
+* Bug fixes to the WAV decoder.
+* Fixed warnings with the Emscripten build relating to the renaming of of `__EMSCRIPTEN_major/minor/tiny__` macros.
+* Win32: Fixed an error with runtime linking on the UWP build. This is actually a non issue in practice because it would require miniaudio to pass in a DLL name of longer than 2048 characters which it never does.
+
+
 v0.11.24 - 2026-01-17
 =====================
 * Fixed a possible glitch when processing the audio of a `ma_sound` when doing resampling.

extras/miniaudio_split/miniaudio.h

@@ -1,6 +1,6 @@
 /*
 Audio playback and capture library. Choice of public domain or MIT-0. See license statements at the end of this file.
-miniaudio - v0.11.23 - 2025-09-11
+miniaudio - v0.11.25 - 2026-03-04
 
 David Reid - mackron@gmail.com
 
@@ -20,7 +20,7 @@ extern "C" {
 
 #define MA_VERSION_MAJOR    0
 #define MA_VERSION_MINOR    11
-#define MA_VERSION_REVISION 23
+#define MA_VERSION_REVISION 25
 #define MA_VERSION_STRING   MA_XSTRINGIFY(MA_VERSION_MAJOR) "." MA_XSTRINGIFY(MA_VERSION_MINOR) "." MA_XSTRINGIFY(MA_VERSION_REVISION)
 
 #if defined(_MSC_VER) && !defined(__clang__)
@@ -131,7 +131,7 @@ typedef ma_uint16 wchar_t;
 
 
 /* Platform/backend detection. */
-#if defined(_WIN32) || defined(__COSMOPOLITAN__)
+#if defined(_WIN32)
     #define MA_WIN32
     #if defined(MA_FORCE_UWP) || (defined(WINAPI_FAMILY) && ((defined(WINAPI_FAMILY_PC_APP) && WINAPI_FAMILY == WINAPI_FAMILY_PC_APP) || (defined(WINAPI_FAMILY_PHONE_APP) && WINAPI_FAMILY == WINAPI_FAMILY_PHONE_APP)))
         #define MA_WIN32_UWP
@@ -455,9 +455,13 @@ typedef enum
     MA_CHANNEL_AUX_29             = 49,
     MA_CHANNEL_AUX_30             = 50,
     MA_CHANNEL_AUX_31             = 51,
+
+    /* Count. */
+    MA_CHANNEL_POSITION_COUNT,
+
+    /* Aliases. */
     MA_CHANNEL_LEFT               = MA_CHANNEL_FRONT_LEFT,
     MA_CHANNEL_RIGHT              = MA_CHANNEL_FRONT_RIGHT,
-    MA_CHANNEL_POSITION_COUNT     = (MA_CHANNEL_AUX_31 + 1)
 } _ma_channel_position; /* Do not use `_ma_channel_position` directly. Use `ma_channel` instead. */
 
 typedef enum
@@ -2877,16 +2881,12 @@ This section contains the APIs for device playback and capture. Here is where yo
     #if defined(MA_WIN32_DESKTOP)   /* DirectSound and WinMM backends are only supported on desktops. */
         #define MA_SUPPORT_DSOUND
         #define MA_SUPPORT_WINMM
-
-        /* Don't enable JACK here if compiling with Cosmopolitan. It'll be enabled in the Linux section below. */
-        #if !defined(__COSMOPOLITAN__)
-            #define MA_SUPPORT_JACK    /* JACK is technically supported on Windows, but I don't know how many people use it in practice... */
-        #endif
+        #define MA_SUPPORT_JACK     /* JACK is technically supported on Windows, but I don't know how many people use it in practice... */
     #endif
 #endif
 #if defined(MA_UNIX) && !defined(MA_ORBIS) && !defined(MA_PROSPERO)
     #if defined(MA_LINUX)
-        #if !defined(MA_ANDROID) && !defined(__COSMOPOLITAN__)   /* ALSA is not supported on Android. */
+        #if !defined(MA_ANDROID) && !defined(MA_EMSCRIPTEN)   /* ALSA is not supported on Android. */
             #define MA_SUPPORT_ALSA
         #endif
     #endif
@@ -5948,7 +5948,7 @@ Parameters
 ----------
 pBackends (out, optional)
     A pointer to the buffer that will receive the enabled backends. Set to NULL to retrieve the backend count. Setting
-    the capacity of the buffer to `MA_BUFFER_COUNT` will guarantee it's large enough for all backends.
+    the capacity of the buffer to `MA_BACKEND_COUNT` will guarantee it's large enough for all backends.
 
 backendCap (in)
     The capacity of the `pBackends` buffer.
@@ -6793,6 +6793,7 @@ typedef struct
     ma_decoding_backend_vtable** ppCustomDecodingBackendVTables;
     ma_uint32 customDecodingBackendCount;
     void* pCustomDecodingBackendUserData;
+    ma_resampler_config resampling;
 } ma_resource_manager_config;
 
 MA_API ma_resource_manager_config ma_resource_manager_config_init(void);
@@ -7120,6 +7121,7 @@ MA_API ma_result ma_node_graph_read_pcm_frames(ma_node_graph* pNodeGraph, void*
 MA_API ma_uint32 ma_node_graph_get_channels(const ma_node_graph* pNodeGraph);
 MA_API ma_uint64 ma_node_graph_get_time(const ma_node_graph* pNodeGraph);
 MA_API ma_result ma_node_graph_set_time(ma_node_graph* pNodeGraph, ma_uint64 globalTime);
+MA_API ma_uint32 ma_node_graph_get_processing_size_in_frames(const ma_node_graph* pNodeGraph);
 
 
 
@@ -7427,6 +7429,7 @@ typedef struct
     ma_bool8 isPitchDisabled;           /* Pitching can be explicitly disabled with MA_SOUND_FLAG_NO_PITCH to optimize processing. */
     ma_bool8 isSpatializationDisabled;  /* Spatialization can be explicitly disabled with MA_SOUND_FLAG_NO_SPATIALIZATION. */
     ma_uint8 pinnedListenerIndex;       /* The index of the listener this node should always use for spatialization. If set to MA_LISTENER_INDEX_CLOSEST the engine will use the closest listener. */
+    ma_resampler_config resampling;
 } ma_engine_node_config;
 
 MA_API ma_engine_node_config ma_engine_node_config_init(ma_engine* pEngine, ma_engine_node_type type, ma_uint32 flags);
@@ -7441,7 +7444,7 @@ typedef struct
     ma_uint32 volumeSmoothTimeInPCMFrames;
     ma_mono_expansion_mode monoExpansionMode;
     ma_fader fader;
-    ma_linear_resampler resampler;                      /* For pitch shift. */
+    ma_resampler resampler;                             /* For pitch shift. */
     ma_spatializer spatializer;
     ma_panner panner;
     ma_gainer volumeGainer;                             /* This will only be used if volumeSmoothTimeInPCMFrames is > 0. */
@@ -7497,6 +7500,7 @@ typedef struct
     ma_uint64 loopPointEndInPCMFrames;
     ma_sound_end_proc endCallback;              /* Fired when the sound reaches the end. Will be fired from the audio thread. Do not restart, uninitialize or otherwise change the state of the sound from here. Instead fire an event or set a variable to indicate to a different thread to change the start of the sound. Will not be fired in response to a scheduled stop with ma_sound_set_stop_time_*(). */
     void* pEndCallbackUserData;
+    ma_resampler_config pitchResampling;
 #ifndef MA_NO_RESOURCE_MANAGER
     ma_resource_manager_pipeline_notifications initNotifications;
 #endif
@@ -7515,7 +7519,10 @@ struct ma_sound
     MA_ATOMIC(4, ma_bool32) atEnd;
     ma_sound_end_proc endCallback;
     void* pEndCallbackUserData;
-    ma_bool8 ownsDataSource;
+    float* pProcessingCache;            /* Will be null if pDataSource is null. */
+    ma_uint32 processingCacheFramesRemaining;
+    ma_uint32 processingCacheCap;
+    ma_bool8 ownsDataSource;    
 
     /*
     We're declaring a resource manager data source object here to save us a malloc when loading a
@@ -7573,6 +7580,8 @@ typedef struct
     ma_vfs* pResourceManagerVFS;                    /* A pointer to a pre-allocated VFS object to use with the resource manager. This is ignored if pResourceManager is not NULL. */
     ma_engine_process_proc onProcess;               /* Fired at the end of each call to ma_engine_read_pcm_frames(). For engine's that manage their own internal device (the default configuration), this will be fired from the audio thread, and you do not need to call ma_engine_read_pcm_frames() manually in order to trigger this. */
     void* pProcessUserData;                         /* User data that's passed into onProcess. */
+    ma_resampler_config resourceManagerResampling;  /* The resampling config to use with the resource manager. */
+    ma_resampler_config pitchResampling;            /* The resampling config for the pitch and Doppler effects. You will typically want this to be a fast resampler. For high quality stuff, it's recommended that you pre-resample. */
 } ma_engine_config;
 
 MA_API ma_engine_config ma_engine_config_init(void);
@@ -7602,6 +7611,7 @@ struct ma_engine
     ma_mono_expansion_mode monoExpansionMode;
     ma_engine_process_proc onProcess;
     void* pProcessUserData;
+    ma_resampler_config pitchResamplingConfig;
 };
 
 MA_API ma_result ma_engine_init(const ma_engine_config* pConfig, ma_engine* pEngine);
@@ -7662,8 +7672,12 @@ MA_API ma_engine* ma_sound_get_engine(const ma_sound* pSound);
 MA_API ma_data_source* ma_sound_get_data_source(const ma_sound* pSound);
 MA_API ma_result ma_sound_start(ma_sound* pSound);
 MA_API ma_result ma_sound_stop(ma_sound* pSound);
-MA_API ma_result ma_sound_stop_with_fade_in_pcm_frames(ma_sound* pSound, ma_uint64 fadeLengthInFrames);     /* Will overwrite any scheduled stop and fade. */
-MA_API ma_result ma_sound_stop_with_fade_in_milliseconds(ma_sound* pSound, ma_uint64 fadeLengthInFrames);   /* Will overwrite any scheduled stop and fade. */
+MA_API ma_result ma_sound_stop_with_fade_in_pcm_frames(ma_sound* pSound, ma_uint64 fadeLengthInFrames);     /* Will overwrite any scheduled stop and fade. If you want to restart the sound, first reset it with `ma_sound_reset_stop_time_and_fade()`. There are plans to make this less awkward in the future. */
+MA_API ma_result ma_sound_stop_with_fade_in_milliseconds(ma_sound* pSound, ma_uint64 fadeLengthInFrames);   /* Will overwrite any scheduled stop and fade. If you want to restart the sound, first reset it with `ma_sound_reset_stop_time_and_fade()`. There are plans to make this less awkward in the future. */
+MA_API void ma_sound_reset_start_time(ma_sound* pSound);
+MA_API void ma_sound_reset_stop_time(ma_sound* pSound);
+MA_API void ma_sound_reset_fade(ma_sound* pSound);
+MA_API void ma_sound_reset_stop_time_and_fade(ma_sound* pSound);  /* Resets fades and scheduled stop time. Does not seek back to the start. */
 MA_API void ma_sound_set_volume(ma_sound* pSound, float volume);
 MA_API float ma_sound_get_volume(const ma_sound* pSound);
 MA_API void ma_sound_set_pan(ma_sound* pSound, float pan);
@@ -7825,7 +7839,7 @@ For more information, please refer to <http://unlicense.org/>
 ===============================================================================
 ALTERNATIVE 2 - MIT No Attribution
 ===============================================================================
-Copyright 2025 David Reid
+Copyright 2026 David Reid
 
 Permission is hereby granted, free of charge, to any person obtaining a copy of
 this software and associated documentation files (the "Software"), to deal in

miniaudio.h

@@ -1,6 +1,6 @@
 /*
 Audio playback and capture library. Choice of public domain or MIT-0. See license statements at the end of this file.
-miniaudio - v0.11.24 - 2026-01-17
+miniaudio - v0.11.25 - 2026-03-04
 
 David Reid - mackron@gmail.com
 
@@ -3747,7 +3747,7 @@ extern "C" {
 
 #define MA_VERSION_MAJOR    0
 #define MA_VERSION_MINOR    11
-#define MA_VERSION_REVISION 24
+#define MA_VERSION_REVISION 25
 #define MA_VERSION_STRING   MA_XSTRINGIFY(MA_VERSION_MAJOR) "." MA_XSTRINGIFY(MA_VERSION_MINOR) "." MA_XSTRINGIFY(MA_VERSION_REVISION)
 
 #if defined(_MSC_VER) && !defined(__clang__)
@@ -19358,7 +19358,7 @@ MA_API ma_handle ma_dlopen(ma_log* pLog, const char* filename)
             #else
                 /* *sigh* It appears there is no ANSI version of LoadPackagedLibrary()... */
                 WCHAR filenameW[4096];
-                if (MultiByteToWideChar(CP_UTF8, 0, filename, -1, filenameW, sizeof(filenameW)) == 0) {
+                if (MultiByteToWideChar(CP_UTF8, 0, filename, -1, filenameW, ma_countof(filenameW)) == 0) {
                     handle = NULL;
                 } else {
                     handle = (ma_handle)LoadPackagedLibrary(filenameW, 0);
@@ -41495,18 +41495,37 @@ Web Audio Backend
 #ifdef MA_HAS_WEBAUDIO
 #include <emscripten/emscripten.h>
 
-#if (__EMSCRIPTEN_major__ > 3) || (__EMSCRIPTEN_major__ == 3 && (__EMSCRIPTEN_minor__ > 1 || (__EMSCRIPTEN_minor__ == 1 && __EMSCRIPTEN_tiny__ >= 32)))
+#ifndef MA_EMSCRIPTEN_MAJOR
+    #if defined(__EMSCRIPTEN_MAJOR__)
+        #define MA_EMSCRIPTEN_MAJOR __EMSCRIPTEN_MAJOR__
+    #else
+        #define MA_EMSCRIPTEN_MAJOR __EMSCRIPTEN_major__
+    #endif
+#endif
+#ifndef MA_EMSCRIPTEN_MINOR
+    #if defined(__EMSCRIPTEN_MINOR__)
+        #define MA_EMSCRIPTEN_MINOR __EMSCRIPTEN_MINOR__
+    #else
+        #define MA_EMSCRIPTEN_MINOR __EMSCRIPTEN_minor__
+    #endif
+#endif
+#ifndef MA_EMSCRIPTEN_TINY
+    #if defined(__EMSCRIPTEN_TINY__)
+        #define MA_EMSCRIPTEN_TINY __EMSCRIPTEN_TINY__
+    #else
+        #define MA_EMSCRIPTEN_TINY __EMSCRIPTEN_tiny__
+    #endif
+#endif
+
+#if (MA_EMSCRIPTEN_MAJOR > 3) || (MA_EMSCRIPTEN_MAJOR == 3 && (MA_EMSCRIPTEN_MINOR > 1 || (MA_EMSCRIPTEN_MINOR == 1 && MA_EMSCRIPTEN_TINY >= 32)))
     #include <emscripten/webaudio.h>
     #define MA_SUPPORT_AUDIO_WORKLETS
 
-    #if (__EMSCRIPTEN_major__ > 3) || (__EMSCRIPTEN_major__ == 3 && (__EMSCRIPTEN_minor__ > 1 || (__EMSCRIPTEN_minor__ == 1 && __EMSCRIPTEN_tiny__ >= 70)))
+    #if (MA_EMSCRIPTEN_MAJOR > 3) || (MA_EMSCRIPTEN_MAJOR == 3 && (MA_EMSCRIPTEN_MINOR > 1 || (MA_EMSCRIPTEN_MINOR == 1 && MA_EMSCRIPTEN_TINY >= 70)))
         #define MA_SUPPORT_AUDIO_WORKLETS_VARIABLE_BUFFER_SIZE
     #endif
 #endif
 
-/*
-TODO: Version 0.12: Swap this logic around so that AudioWorklets are used by default. Add MA_NO_AUDIO_WORKLETS.
-*/
 #if defined(MA_ENABLE_AUDIO_WORKLETS) && defined(MA_SUPPORT_AUDIO_WORKLETS)
     #define MA_USE_AUDIO_WORKLETS
 #endif
@@ -59243,6 +59262,10 @@ static ma_result ma_data_source_read_pcm_frames_within_range(ma_data_source* pDa
     ma_uint64 framesRead = 0;
     ma_bool32 loop = ma_data_source_is_looping(pDataSource);
 
+    if (pFramesRead != NULL) {
+        *pFramesRead = 0;
+    }
+
     if (pDataSourceBase == NULL) {
         return MA_AT_END;
     }
@@ -61921,7 +61944,7 @@ extern "C" {
 #define MA_DR_WAV_XSTRINGIFY(x)     MA_DR_WAV_STRINGIFY(x)
 #define MA_DR_WAV_VERSION_MAJOR     0
 #define MA_DR_WAV_VERSION_MINOR     14
-#define MA_DR_WAV_VERSION_REVISION  4
+#define MA_DR_WAV_VERSION_REVISION  5
 #define MA_DR_WAV_VERSION_STRING    MA_DR_WAV_XSTRINGIFY(MA_DR_WAV_VERSION_MAJOR) "." MA_DR_WAV_XSTRINGIFY(MA_DR_WAV_VERSION_MINOR) "." MA_DR_WAV_XSTRINGIFY(MA_DR_WAV_VERSION_REVISION)
 #include <stddef.h>
 #define MA_DR_WAVE_FORMAT_PCM          0x1
@@ -80503,6 +80526,13 @@ MA_PRIVATE ma_uint64 ma_dr_wav__read_smpl_to_metadata_obj(ma_dr_wav__metadata_pa
     MA_DR_WAV_ASSERT(pChunkHeader != NULL);
     if (pMetadata != NULL && bytesJustRead == sizeof(smplHeaderData)) {
         ma_uint32 iSampleLoop;
+        ma_uint32 loopCount;
+        ma_uint32 calculatedLoopCount;
+        loopCount = ma_dr_wav_bytes_to_u32(smplHeaderData + 28);
+        calculatedLoopCount = (pChunkHeader->sizeInBytes - MA_DR_WAV_SMPL_BYTES) / MA_DR_WAV_SMPL_LOOP_BYTES;
+        if (loopCount != calculatedLoopCount) {
+            return totalBytesRead;
+        }
         pMetadata->type                                     = ma_dr_wav_metadata_type_smpl;
         pMetadata->data.smpl.manufacturerId                 = ma_dr_wav_bytes_to_u32(smplHeaderData + 0);
         pMetadata->data.smpl.productId                      = ma_dr_wav_bytes_to_u32(smplHeaderData + 4);
@@ -80513,7 +80543,7 @@ MA_PRIVATE ma_uint64 ma_dr_wav__read_smpl_to_metadata_obj(ma_dr_wav__metadata_pa
         pMetadata->data.smpl.smpteOffset                    = ma_dr_wav_bytes_to_u32(smplHeaderData + 24);
         pMetadata->data.smpl.sampleLoopCount                = ma_dr_wav_bytes_to_u32(smplHeaderData + 28);
         pMetadata->data.smpl.samplerSpecificDataSizeInBytes = ma_dr_wav_bytes_to_u32(smplHeaderData + 32);
-        if (pMetadata->data.smpl.sampleLoopCount == (pChunkHeader->sizeInBytes - MA_DR_WAV_SMPL_BYTES) / MA_DR_WAV_SMPL_LOOP_BYTES) {
+        if (pMetadata->data.smpl.sampleLoopCount == calculatedLoopCount) {
             pMetadata->data.smpl.pLoops = (ma_dr_wav_smpl_loop*)ma_dr_wav__metadata_get_memory(pParser, sizeof(ma_dr_wav_smpl_loop) * pMetadata->data.smpl.sampleLoopCount, MA_DR_WAV_METADATA_ALIGNMENT);
             for (iSampleLoop = 0; iSampleLoop < pMetadata->data.smpl.sampleLoopCount; ++iSampleLoop) {
                 ma_uint8 smplLoopData[MA_DR_WAV_SMPL_LOOP_BYTES];
@@ -80534,6 +80564,8 @@ MA_PRIVATE ma_uint64 ma_dr_wav__read_smpl_to_metadata_obj(ma_dr_wav__metadata_pa
                 MA_DR_WAV_ASSERT(pMetadata->data.smpl.pSamplerSpecificData != NULL);
                 ma_dr_wav__metadata_parser_read(pParser, pMetadata->data.smpl.pSamplerSpecificData, pMetadata->data.smpl.samplerSpecificDataSizeInBytes, &totalBytesRead);
             }
+        } else {
+            MA_DR_WAV_ZERO_OBJECT(&pMetadata->data.smpl);
         }
     }
     return totalBytesRead;
@@ -83149,19 +83181,13 @@ MA_PRIVATE ma_uint64 ma_dr_wav_read_pcm_frames_s16__msadpcm(ma_dr_wav* pWav, ma_
                     newSample0  = ((pWav->msadpcm.prevFrames[0][1] * coeff1Table[pWav->msadpcm.predictor[0]]) + (pWav->msadpcm.prevFrames[0][0] * coeff2Table[pWav->msadpcm.predictor[0]])) >> 8;
                     newSample0 += nibble0 * pWav->msadpcm.delta[0];
                     newSample0  = ma_dr_wav_clamp(newSample0, -32768, 32767);
-                    pWav->msadpcm.delta[0] = (adaptationTable[((nibbles & 0xF0) >> 4)] * pWav->msadpcm.delta[0]) >> 8;
-                    if (pWav->msadpcm.delta[0] < 16) {
-                        pWav->msadpcm.delta[0] = 16;
-                    }
+                    pWav->msadpcm.delta[0] = (ma_int32)ma_dr_wav_clamp(((ma_int64)adaptationTable[((nibbles & 0xF0) >> 4)] * pWav->msadpcm.delta[0]) >> 8, 16, 0x7FFFFFFF);
                     pWav->msadpcm.prevFrames[0][0] = pWav->msadpcm.prevFrames[0][1];
                     pWav->msadpcm.prevFrames[0][1] = newSample0;
                     newSample1  = ((pWav->msadpcm.prevFrames[0][1] * coeff1Table[pWav->msadpcm.predictor[0]]) + (pWav->msadpcm.prevFrames[0][0] * coeff2Table[pWav->msadpcm.predictor[0]])) >> 8;
                     newSample1 += nibble1 * pWav->msadpcm.delta[0];
                     newSample1  = ma_dr_wav_clamp(newSample1, -32768, 32767);
-                    pWav->msadpcm.delta[0] = (adaptationTable[((nibbles & 0x0F) >> 0)] * pWav->msadpcm.delta[0]) >> 8;
-                    if (pWav->msadpcm.delta[0] < 16) {
-                        pWav->msadpcm.delta[0] = 16;
-                    }
+                    pWav->msadpcm.delta[0] = (ma_int32)ma_dr_wav_clamp(((ma_int64)adaptationTable[((nibbles & 0x0F) >> 0)] * pWav->msadpcm.delta[0]) >> 8, 16, 0x7FFFFFFF);
                     pWav->msadpcm.prevFrames[0][0] = pWav->msadpcm.prevFrames[0][1];
                     pWav->msadpcm.prevFrames[0][1] = newSample1;
                     pWav->msadpcm.cachedFrames[2] = newSample0;
@@ -83176,10 +83202,7 @@ MA_PRIVATE ma_uint64 ma_dr_wav_read_pcm_frames_s16__msadpcm(ma_dr_wav* pWav, ma_
                     newSample0  = ((pWav->msadpcm.prevFrames[0][1] * coeff1Table[pWav->msadpcm.predictor[0]]) + (pWav->msadpcm.prevFrames[0][0] * coeff2Table[pWav->msadpcm.predictor[0]])) >> 8;
                     newSample0 += nibble0 * pWav->msadpcm.delta[0];
                     newSample0  = ma_dr_wav_clamp(newSample0, -32768, 32767);
-                    pWav->msadpcm.delta[0] = (adaptationTable[((nibbles & 0xF0) >> 4)] * pWav->msadpcm.delta[0]) >> 8;
-                    if (pWav->msadpcm.delta[0] < 16) {
-                        pWav->msadpcm.delta[0] = 16;
-                    }
+                    pWav->msadpcm.delta[0] = (ma_int32)ma_dr_wav_clamp(((ma_int64)adaptationTable[((nibbles & 0xF0) >> 4)] * pWav->msadpcm.delta[0]) >> 8, 16, 0x7FFFFFFF);
                     pWav->msadpcm.prevFrames[0][0] = pWav->msadpcm.prevFrames[0][1];
                     pWav->msadpcm.prevFrames[0][1] = newSample0;
                     if (pWav->msadpcm.predictor[1] >= ma_dr_wav_countof(coeff1Table) || pWav->msadpcm.predictor[1] >= ma_dr_wav_countof(coeff2Table)) {
@@ -83188,10 +83211,7 @@ MA_PRIVATE ma_uint64 ma_dr_wav_read_pcm_frames_s16__msadpcm(ma_dr_wav* pWav, ma_
                     newSample1  = ((pWav->msadpcm.prevFrames[1][1] * coeff1Table[pWav->msadpcm.predictor[1]]) + (pWav->msadpcm.prevFrames[1][0] * coeff2Table[pWav->msadpcm.predictor[1]])) >> 8;
                     newSample1 += nibble1 * pWav->msadpcm.delta[1];
                     newSample1  = ma_dr_wav_clamp(newSample1, -32768, 32767);
-                    pWav->msadpcm.delta[1] = (adaptationTable[((nibbles & 0x0F) >> 0)] * pWav->msadpcm.delta[1]) >> 8;
-                    if (pWav->msadpcm.delta[1] < 16) {
-                        pWav->msadpcm.delta[1] = 16;
-                    }
+                    pWav->msadpcm.delta[1] = (ma_int32)ma_dr_wav_clamp(((ma_int64)adaptationTable[((nibbles & 0x0F) >> 0)] * pWav->msadpcm.delta[1]) >> 8, 16, 0x7FFFFFFF);
                     pWav->msadpcm.prevFrames[1][0] = pWav->msadpcm.prevFrames[1][1];
                     pWav->msadpcm.prevFrames[1][1] = newSample1;
                     pWav->msadpcm.cachedFrames[2] = newSample0;
@@ -95825,7 +95845,7 @@ For more information, please refer to <http://unlicense.org/>
 ===============================================================================
 ALTERNATIVE 2 - MIT No Attribution
 ===============================================================================
-Copyright 2025 David Reid
+Copyright 2026 David Reid
 
 Permission is hereby granted, free of charge, to any person obtaining a copy of
 this software and associated documentation files (the "Software"), to deal in

website/.webplate/miniaudio-footer.html

@@ -1,9 +1,8 @@
     </div>
     <table style="margin:0 auto; padding:1em 0px; text-align:center;">
         <tr>
-            <td style="vertical-align:center;"><a style="padding:0;" href="https://www.reddit.com/r/miniaudio"><img src="{{ relative-path "img/reddit_white.svg" }}" style="margin:0; padding:0; height:40px; width:40px;"></a></td>
             <td style="vertical-align:center;"><a style="padding:0;" href="https://discord.gg/9vpqbjU"><img src="{{ relative-path "img/Discord-Logo-White.svg" }}" style="padding:0; height:32px; width:32px;"></a></td>
-            <td rel="me" style="vertical-align:center;"><a style="padding:0;" href="https://fosstodon.org/@mackron"><img src="{{ relative-path "img/mastodon_white.svg" }}" style="padding:0; height:24px; width:32px;"></a></td>
+            <td rel="me" style="vertical-align:center;"><a style="padding:0;" href="https://x.com/mackron"><img src="{{ relative-path "img/x_logo.svg" }}" style="padding:0; height:24px; width:24px;"></a></td>
             <td style="vertical-align:center;"><a style="padding:0;" href="https://github.com/mackron/miniaudio"><img src="{{ relative-path "img/github_white.png" }}" style="padding:0; height:24px; width:24px;"></a></td>
         </tr>
     </table>

website/.webplate/miniaudio-header.html

@@ -29,14 +29,11 @@
                         </table>    
                     </td>
                     
-                    <td style="padding:0.1em; width:25%; text-align:right; vertical-align:center;">
-                        <a href="https://www.reddit.com/r/miniaudio"><img src="{{ relative-path "img/reddit_white.svg" }}" style="margin:0; padding:0; height:40px; width:40px;"></a>
-                    </td>
                     <td style="padding:0.1em; width:25%; text-align:right; vertical-align:center;">
                         <a href="https://discord.gg/9vpqbjU"><img src="{{ relative-path "img/Discord-Logo-White.svg" }}" style="margin:0; padding:0; height:32px; width:32px;"></a>
                     </td>
                     <td style="padding:0.1em; width:25%; text-align:right; vertical-align:center;">
-                        <a rel="me" href="https://fosstodon.org/@mackron"><img src="{{ relative-path "img/mastodon_white.svg" }}" style="margin:0; padding:0; height:24px; width:32px;"></a>
+                        <a rel="me" href="https://x.com/mackron"><img src="{{ relative-path "img/x_logo.svg" }}" style="margin:0; padding:0; height:24px; width:24px;"></a>
                     </td>
                     <td style="padding:0.1em; padding-right:1em; width:25%; text-align:right; vertical-align:center;">
                         <a href="https://github.com/mackron/miniaudio"><img src="{{ relative-path "img/github_white.png" }}" style="margin:0; padding:0; height:24px; width:24px;"></a>

website/img/mastodon_white.svg

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website/img/reddit_white.svg

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website/img/x_logo.svg

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