fixed a little problem with changing devices when initialized our device...

fixed a little problem with changing devices when initialized our device config as duplex; fixed typos

.gitignore

@@ -10,4 +10,5 @@ tests/_build/vc6/
 tests/_build/vc15/
 tools/_build/
 *.vcxproj.user
-.vs/
\ No newline at end of file
+.vs/
+.idea/

miniaudio.h

@@ -1226,7 +1226,7 @@ might be beneficial to pre-decode the sound. You can do this with the `MA_SOUND_
 
 By default, sounds will be loaded synchronously, meaning `ma_sound_init_*()` will not return until
 the sound has been fully loaded. If this is prohibitive you can instead load sounds asynchronously
-by specificying the `MA_SOUND_FLAG_ASYNC` flag:
+by specifying the `MA_SOUND_FLAG_ASYNC` flag:
 
     ```c
     ma_sound_init_from_file(&engine, "my_sound.wav", MA_SOUND_FLAG_DECODE | MA_SOUND_FLAG_ASYNC, pGroup, NULL, &sound);
@@ -1247,7 +1247,7 @@ counter hit's zero. You can specify a fence like so:
     ma_sound sounds[4];
 
     result = ma_fence_init(&fence);
-    if (result != MA_SUCCES) {
+    if (result != MA_SUCCESS) {
         return result;
     }
 
@@ -2028,14 +2028,14 @@ data from the graph:
     ```
 
 When you read audio data, miniaudio starts at the node graph's endpoint node which then pulls in
-data from it's input attachments, which in turn recusively pull in data from their inputs, and so
+data from it's input attachments, which in turn recursively pull in data from their inputs, and so
 on. At the start of the graph there will be some kind of data source node which will have zero
 inputs and will instead read directly from a data source. The base nodes don't literally need to
 read from a `ma_data_source` object, but they will always have some kind of underlying object that
 sources some kind of audio. The `ma_data_source_node` node can be used to read from a
 `ma_data_source`. Data is always in floating-point format and in the number of channels you
 specified when the graph was initialized. The sample rate is defined by the underlying data sources.
-It's up to you to ensure they use a consistent and appropraite sample rate.
+It's up to you to ensure they use a consistent and appropriate sample rate.
 
 The `ma_node` API is designed to allow custom nodes to be implemented with relative ease, but
 miniaudio includes a few stock nodes for common functionality. This is how you would initialize a
@@ -2076,7 +2076,7 @@ another, you do not need to detach first. You can just call `ma_node_attach_outp
 deal with it for you.
 
 Less frequently you may want to create a specialized node. This will be a node where you implement
-your own processing callback to apply a custom effect of some kind. This is similar to initalizing
+your own processing callback to apply a custom effect of some kind. This is similar to initializing
 one of the stock node types, only this time you need to specify a pointer to a vtable containing a
 pointer to the processing function and the number of input and output buses. Example:
 
@@ -2115,7 +2115,7 @@ pointer to the processing function and the number of input and output buses. Exa
     // Each bus needs to have a channel count specified. To do this you need to specify the channel
     // counts in an array and then pass that into the node config.
     ma_uint32 inputChannels[2];     // Equal in size to the number of input channels specified in the vtable.
-    ma_uint32 outputChannels[1];    // Equal in size to the number of output channels specicied in the vtable.
+    ma_uint32 outputChannels[1];    // Equal in size to the number of output channels specified in the vtable.
 
     inputChannels[0]  = channelsIn;
     inputChannels[1]  = channelsIn;
@@ -2393,7 +2393,7 @@ bus and input bus is locked. This locking is specifically for attaching and deta
 different threads and does not affect `ma_node_graph_read_pcm_frames()` in any way. The locking and
 unlocking is mostly self-explanatory, but a slightly less intuitive aspect comes into it when
 considering that iterating over attachments must not break as a result of attaching or detaching a
-node while iteration is occuring.
+node while iteration is occurring.
 
 Attaching and detaching are both quite simple. When an output bus of a node is attached to an input
 bus of another node, it's added to a linked list. Basically, an input bus is a linked list, where
@@ -2459,7 +2459,7 @@ implementation:
     #define MA_NO_FLAC
     ```
 
-Disabling built-in decoding libraries is useful if you use these libraries independantly of the
+Disabling built-in decoding libraries is useful if you use these libraries independently of the
 `ma_decoder` API.
 
 A decoder can be initialized from a file with `ma_decoder_init_file()`, a block of memory with
@@ -2573,11 +2573,11 @@ these are not specified, miniaudio will deal with it for you via a generic imple
 
 When you initialize a custom data source (by implementing the `onInit` function in the vtable) you
 will need to output a pointer to a `ma_data_source` which implements your custom decoder. See the
-section about data sources for details on how to implemen this. Alternatively, see the
+section about data sources for details on how to implement this. Alternatively, see the
 "custom_decoders" example in the miniaudio repository.
 
 The `onInit` function takes a pointer to some callbacks for the purpose of reading raw audio data
-from some abitrary source. You'll use these functions to read from the raw data and perform the
+from some arbitrary source. You'll use these functions to read from the raw data and perform the
 decoding. When you call them, you will pass in the `pReadSeekTellUserData` pointer to the relevant
 parameter.
 
@@ -2728,7 +2728,7 @@ To perform the conversion simply call `ma_channel_converter_process_pcm_frames()
     }
     ```
 
-It is up to the caller to ensure the output buffer is large enough to accomodate the new PCM
+It is up to the caller to ensure the output buffer is large enough to accommodate the new PCM
 frames.
 
 Input and output PCM frames are always interleaved. Deinterleaved layouts are not supported.
@@ -3174,7 +3174,7 @@ you can chain first and second order filters together.
 
 If you need to change the configuration of the filter, but need to maintain the state of internal
 registers you can do so with `ma_lpf_reinit()`. This may be useful if you need to change the sample
-rate and/or cutoff frequency dynamically while maintaing smooth transitions. Note that changing the
+rate and/or cutoff frequency dynamically while maintaining smooth transitions. Note that changing the
 format or channel count after initialization is invalid and will result in an error.
 
 The `ma_lpf` object supports a configurable order, but if you only need a first order filter you
@@ -3496,7 +3496,7 @@ you will want to use. To initialize a ring buffer, do something like the followi
     ```
 
 The `ma_pcm_rb_init()` function takes the sample format and channel count as parameters because
-it's the PCM varient of the ring buffer API. For the regular ring buffer that operates on bytes you
+it's the PCM variant of the ring buffer API. For the regular ring buffer that operates on bytes you
 would call `ma_rb_init()` which leaves these out and just takes the size of the buffer in bytes
 instead of frames. The fourth parameter is an optional pre-allocated buffer and the fifth parameter
 is a pointer to a `ma_allocation_callbacks` structure for custom memory allocation routines.
@@ -3636,8 +3636,8 @@ Some backends have some nuance details you may want to be aware of.
 --------------------
 - If a sound does not require doppler or pitch shifting, consider disabling pitching by
   initializing the sound with the `MA_SOUND_FLAG_NO_PITCH` flag.
-- If a sound does not require spatialization, disable it by initialzing the sound with the
-  `MA_SOUND_FLAG_NO_SPATIALIZATION` flag. It can be renabled again post-initialization with
+- If a sound does not require spatialization, disable it by initializing the sound with the
+  `MA_SOUND_FLAG_NO_SPATIALIZATION` flag. It can be re-enabled again post-initialization with
   `ma_sound_set_spatialization_enabled()`.
 
 
@@ -5286,7 +5286,7 @@ MA_API ma_result ma_resampler_process_pcm_frames(ma_resampler* pResampler, const
 
 
 /*
-Sets the input and output sample sample rate.
+Sets the input and output sample rate.
 */
 MA_API ma_result ma_resampler_set_rate(ma_resampler* pResampler, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut);
 
@@ -6502,7 +6502,7 @@ DEPRECATED. Use ma_device_notification_proc instead.
 The callback for when the device has been stopped.
 
 This will be called when the device is stopped explicitly with `ma_device_stop()` and also called implicitly when the device is stopped through external forces
-such as being unplugged or an internal error occuring.
+such as being unplugged or an internal error occurring.
 
 
 Parameters
@@ -6865,7 +6865,7 @@ If the backend requires absolute flexibility with it's data delivery, it can opt
 which will allow it to implement the logic that will run on the audio thread. This is much more advanced and is completely optional.
 
 The audio thread should run data delivery logic in a loop while `ma_device_get_state() == ma_device_state_started` and no errors have been
-encounted. Do not start or stop the device here. That will be handled from outside the `onDeviceDataLoop()` callback.
+encountered. Do not start or stop the device here. That will be handled from outside the `onDeviceDataLoop()` callback.
 
 The invocation of the `onDeviceDataLoop()` callback will be handled by miniaudio. When you start the device, miniaudio will fire this
 callback. When the device is stopped, the `ma_device_get_state() == ma_device_state_started` condition will fail and the loop will be terminated
@@ -8316,7 +8316,7 @@ then be set directly on the structure. Below are the members of the `ma_device_c
         A pointer that will passed to callbacks in pBackendVTable.
 
     resampling.linear.lpfOrder
-        The linear resampler applies a low-pass filter as part of it's procesing for anti-aliasing. This setting controls the order of the filter. The higher
+        The linear resampler applies a low-pass filter as part of it's processing for anti-aliasing. This setting controls the order of the filter. The higher
         the value, the better the quality, in general. Setting this to 0 will disable low-pass filtering altogether. The maximum value is
         `MA_MAX_FILTER_ORDER`. The default value is `min(4, MA_MAX_FILTER_ORDER)`.
 
@@ -9323,7 +9323,7 @@ MA_API ma_bool32 ma_is_loopback_supported(ma_backend backend);
 
 /************************************************************************************************************************************************************
 
-Utiltities
+Utilities
 
 ************************************************************************************************************************************************************/
 
@@ -15516,7 +15516,7 @@ static C89ATOMIC_INLINE void c89atomic_spinlock_unlock(volatile c89atomic_spinlo
 #define MA_ATOMIC_SAFE_TYPE_IMPL(c89TypeExtension, type) \
     static MA_INLINE ma_##type ma_atomic_##type##_get(ma_atomic_##type* x) \
     { \
-        return c89atomic_load_##c89TypeExtension(&x->value); \
+        return (ma_##type) c89atomic_load_##c89TypeExtension(&x->value); \
     } \
     static MA_INLINE void ma_atomic_##type##_set(ma_atomic_##type* x, ma_##type value) \
     { \
@@ -15524,7 +15524,7 @@ static C89ATOMIC_INLINE void c89atomic_spinlock_unlock(volatile c89atomic_spinlo
     } \
     static MA_INLINE ma_##type ma_atomic_##type##_exchange(ma_atomic_##type* x, ma_##type value) \
     { \
-        return c89atomic_exchange_##c89TypeExtension(&x->value, value); \
+        return (ma_##type) c89atomic_exchange_##c89TypeExtension(&x->value, value); \
     } \
     static MA_INLINE ma_bool32 ma_atomic_##type##_compare_exchange(ma_atomic_##type* x, ma_##type* expected, ma_##type desired) \
     { \
@@ -15532,27 +15532,27 @@ static C89ATOMIC_INLINE void c89atomic_spinlock_unlock(volatile c89atomic_spinlo
     } \
     static MA_INLINE ma_##type ma_atomic_##type##_fetch_add(ma_atomic_##type* x, ma_##type y) \
     { \
-        return c89atomic_fetch_add_##c89TypeExtension(&x->value, y); \
+        return (ma_##type) c89atomic_fetch_add_##c89TypeExtension(&x->value, y); \
     } \
     static MA_INLINE ma_##type ma_atomic_##type##_fetch_sub(ma_atomic_##type* x, ma_##type y) \
     { \
-        return c89atomic_fetch_sub_##c89TypeExtension(&x->value, y); \
+        return (ma_##type) c89atomic_fetch_sub_##c89TypeExtension(&x->value, y); \
     } \
     static MA_INLINE ma_##type ma_atomic_##type##_fetch_or(ma_atomic_##type* x, ma_##type y) \
     { \
-        return c89atomic_fetch_or_##c89TypeExtension(&x->value, y); \
+        return (ma_##type) c89atomic_fetch_or_##c89TypeExtension(&x->value, y); \
     } \
     static MA_INLINE ma_##type ma_atomic_##type##_fetch_xor(ma_atomic_##type* x, ma_##type y) \
     { \
-        return c89atomic_fetch_xor_##c89TypeExtension(&x->value, y); \
+        return (ma_##type) c89atomic_fetch_xor_##c89TypeExtension(&x->value, y); \
     } \
     static MA_INLINE ma_##type ma_atomic_##type##_fetch_and(ma_atomic_##type* x, ma_##type y) \
     { \
-        return c89atomic_fetch_and_##c89TypeExtension(&x->value, y); \
+        return (ma_##type) c89atomic_fetch_and_##c89TypeExtension(&x->value, y); \
     } \
     static MA_INLINE ma_##type ma_atomic_##type##_compare_and_swap(ma_atomic_##type* x, ma_##type expected, ma_##type desired) \
     { \
-        return c89atomic_compare_and_swap_##c89TypeExtension(&x->value, expected, desired); \
+        return (ma_##type) c89atomic_compare_and_swap_##c89TypeExtension(&x->value, expected, desired); \
     } \
 
 #define MA_ATOMIC_SAFE_TYPE_IMPL_PTR(type) \
@@ -39421,6 +39421,14 @@ static ma_result ma_device__post_init_setup(ma_device* pDevice, ma_device_type d
 
     MA_ASSERT(pDevice != NULL);
 
+	if (pDevice->type == ma_device_type_duplex) {
+	    // device is expected to be used as both input/output
+		// without this, deviceType could be playback,
+		// causing the device input cache to be freed
+		// and this will break other code using that cache later
+	    deviceType = ma_device_type_duplex;
+	}
+
     if (deviceType == ma_device_type_capture || deviceType == ma_device_type_duplex || deviceType == ma_device_type_loopback) {
         if (pDevice->capture.format == ma_format_unknown) {
             pDevice->capture.format = pDevice->capture.internalFormat;
@@ -77531,10 +77539,10 @@ DRWAV_PRIVATE drwav_result drwav_wfopen(FILE** ppFile, const wchar_t* pFilePath,
         (void)pAllocationCallbacks;
     }
 #else
-	#if defined(__DJGPP__)
-	{
-	}
-	#else
+    #if defined(__DJGPP__)
+    {
+    }
+    #else
     {
         mbstate_t mbs;
         size_t lenMB;
@@ -77567,7 +77575,7 @@ DRWAV_PRIVATE drwav_result drwav_wfopen(FILE** ppFile, const wchar_t* pFilePath,
         *ppFile = fopen(pFilePathMB, pOpenModeMB);
         drwav__free_from_callbacks(pFilePathMB, pAllocationCallbacks);
     }
-	#endif
+    #endif
     if (*ppFile == NULL) {
         return DRWAV_ERROR;
     }
@@ -85492,10 +85500,10 @@ static drflac_result drflac_wfopen(FILE** ppFile, const wchar_t* pFilePath, cons
         (void)pAllocationCallbacks;
     }
 #else
-	#if defined(__DJGPP__)
-	{
-	}
-	#else
+    #if defined(__DJGPP__)
+    {
+    }
+    #else
     {
         mbstate_t mbs;
         size_t lenMB;
@@ -85528,7 +85536,7 @@ static drflac_result drflac_wfopen(FILE** ppFile, const wchar_t* pFilePath, cons
         *ppFile = fopen(pFilePathMB, pOpenModeMB);
         drflac__free_from_callbacks(pFilePathMB, pAllocationCallbacks);
     }
-	#endif
+    #endif
     if (*ppFile == NULL) {
         return DRFLAC_ERROR;
     }
@@ -90828,10 +90836,10 @@ static drmp3_result drmp3_wfopen(FILE** ppFile, const wchar_t* pFilePath, const
         (void)pAllocationCallbacks;
     }
 #else
-	#if defined(__DJGPP__)
-	{
-	}
-	#else
+    #if defined(__DJGPP__)
+    {
+    }
+    #else
     {
         mbstate_t mbs;
         size_t lenMB;
@@ -90864,7 +90872,7 @@ static drmp3_result drmp3_wfopen(FILE** ppFile, const wchar_t* pFilePath, const
         *ppFile = fopen(pFilePathMB, pOpenModeMB);
         drmp3__free_from_callbacks(pFilePathMB, pAllocationCallbacks);
     }
-	#endif
+    #endif
     if (*ppFile == NULL) {
         return DRMP3_ERROR;
     }