fix typos

CHANGES.md

@@ -276,7 +276,7 @@ v0.11.0 - 2021-12-18
   - Add support for disabling denormals on the audio thread.
   - Add a delay/echo effect called ma_delay.
   - Add a stereo pan effect called ma_panner.
-  - Add a spataializer effect called ma_spatializer.
+  - Add a spatializer effect called ma_spatializer.
   - Add support for amplification for device master volume.
   - Remove dependency on MA_MAX_CHANNELS from filters and data conversion.
   - Increase MA_MAX_CHANNELS from 32 to 254.
@@ -862,7 +862,7 @@ v0.9 - 2019-03-06
   - API CHANGE: Add log level to the log callback. New signature:
     - void on_log(ma_context* pContext, ma_device* pDevice, ma_uint32 logLevel, const char* message)
   - API CHANGE: Changes to result codes. Constants have changed and unused codes have been removed. If you're
-    a binding mainainer you will need to update your result code constants.
+    a binding maintainer you will need to update your result code constants.
   - API CHANGE: Change the order of the ma_backend enums to priority order. If you are a binding maintainer, you
     will need to update.
   - API CHANGE: Rename mal_dsp to ma_pcm_converter. All functions have been renamed from mal_dsp_*() to
@@ -971,7 +971,7 @@ v0.8 - 2018-07-05
   - Changed MAL_IMPLEMENTATION to MINI_AL_IMPLEMENTATION for consistency with other libraries. The old
     way is still supported for now, but you should update as it may be removed in the future.
   - API CHANGE: Replace device enumeration APIs. mal_enumerate_devices() has been replaced with
-    mal_context_get_devices(). An additional low-level device enumration API has been introduced called
+    mal_context_get_devices(). An additional low-level device enumeration API has been introduced called
     mal_context_enumerate_devices() which uses a callback to report devices.
   - API CHANGE: Rename mal_get_sample_size_in_bytes() to mal_get_bytes_per_sample() and add
     mal_get_bytes_per_frame().

extras/osaudio/osaudio.h

@@ -66,8 +66,8 @@ blocking, it can be useful to know how many frames can be written/read without b
 achieved with osaudio_get_avail().
 
 Querying the device's configuration is achieved with osaudio_get_info(). This function will return
-a pointer to a osaudio_info_t structure which contains information about the device, most
-importantly it's name and data configuration. The name is important for displaying on a UI, and
+a pointer to an osaudio_info_t structure which contains information about the device, most
+importantly its name and data configuration. The name is important for displaying on a UI, and
 the data configuration is important for knowing how to format your audio data. The osaudio_info_t
 structure will contain an array of osaudio_config_t structures. This will contain one entry, which
 will contain the exact information that was returned in the config structure that was passed to

miniaudio.h

@@ -430,11 +430,11 @@ Sounds and sound groups are nodes in the engine's node graph and can be plugged
 API. This makes it possible to connect sounds and sound groups to effect nodes to produce complex
 effect chains.
 
-A sound can have it's volume changed with `ma_sound_set_volume()`. If you prefer decibel volume
+A sound can have its volume changed with `ma_sound_set_volume()`. If you prefer decibel volume
 control you can use `ma_volume_db_to_linear()` to convert from decibel representation to linear.
 
 Panning and pitching is supported with `ma_sound_set_pan()` and `ma_sound_set_pitch()`. If you know
-a sound will never have it's pitch changed with `ma_sound_set_pitch()` or via the doppler effect,
+a sound will never have its pitch changed with `ma_sound_set_pitch()` or via the doppler effect,
 you can specify the `MA_SOUND_FLAG_NO_PITCH` flag when initializing the sound for an optimization.
 
 By default, sounds and sound groups have spatialization enabled. If you don't ever want to
@@ -10617,7 +10617,7 @@ typedef struct
     void (* onProcess)(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut);
 
     /*
-    A callback for retrieving the number of a input frames that are required to output the
+    A callback for retrieving the number of input frames that are required to output the
     specified number of output frames. You would only want to implement this when the node performs
     resampling. This is optional, even for nodes that perform resampling, but it does offer a
     small reduction in latency as it allows miniaudio to calculate the exact number of input frames
@@ -19514,7 +19514,7 @@ static ma_result ma_device_do_operation__null(ma_device* pDevice, ma_uint32 oper
 
     /*
     The first thing to do is wait for an operation slot to become available. We only have a single slot for this, but we could extend this later
-    to support queing of operations.
+    to support queuing of operations.
     */
     result = ma_semaphore_wait(&pDevice->null_device.operationSemaphore);
     if (result != MA_SUCCESS) {
@@ -22319,7 +22319,7 @@ static ma_result ma_device_init_internal__wasapi(ma_context* pContext, ma_device
         MA_REFERENCE_TIME bufferDuration = periodDurationInMicroseconds * pData->periodsOut * 10;
 
         /*
-        If the periodicy is too small, Initialize() will fail with AUDCLNT_E_INVALID_DEVICE_PERIOD. In this case we should just keep increasing
+        If the periodicity is too small, Initialize() will fail with AUDCLNT_E_INVALID_DEVICE_PERIOD. In this case we should just keep increasing
         it and trying it again.
         */
         hr = E_FAIL;
@@ -49584,7 +49584,7 @@ MA_API float ma_fader_get_current_volume(const ma_fader* pFader)
     } else if ((ma_uint64)pFader->cursorInFrames >= pFader->lengthInFrames) {   /* Safe case because the < 0 case was checked above. */
         return pFader->volumeEnd;
     } else {
-        /* The cursor is somewhere inside the fading period. We can figure this out with a simple linear interpoluation between volumeBeg and volumeEnd based on our cursor position. */
+        /* The cursor is somewhere inside the fading period. We can figure this out with a simple linear interpolation between volumeBeg and volumeEnd based on our cursor position. */
         return ma_mix_f32_fast(pFader->volumeBeg, pFader->volumeEnd, (ma_uint32)pFader->cursorInFrames / (float)((ma_uint32)pFader->lengthInFrames));    /* Safe cast to uint32 because we clamp it in ma_fader_process_pcm_frames(). */
     }
 }
@@ -49848,7 +49848,7 @@ static void ma_get_default_channel_map_for_spatializer(ma_channel* pChannelMap,
     Special case for stereo. Want to default the left and right speakers to side left and side
     right so that they're facing directly down the X axis rather than slightly forward. Not
     doing this will result in sounds being quieter when behind the listener. This might
-    actually be good for some scenerios, but I don't think it's an appropriate default because
+    actually be good for some scenarios, but I don't think it's an appropriate default because
     it can be a bit unexpected.
     */
     if (channelCount == 2) {
@@ -50486,7 +50486,7 @@ MA_API ma_result ma_spatializer_process_pcm_frames(ma_spatializer* pSpatializer,
         ma_vec3f relativePosNormalized;
         ma_vec3f relativePos;   /* The position relative to the listener. */
         ma_vec3f relativeDir;   /* The direction of the sound, relative to the listener. */
-        ma_vec3f listenerVel;   /* The volocity of the listener. For doppler pitch calculation. */
+        ma_vec3f listenerVel;   /* The velocity of the listener. For doppler pitch calculation. */
         float speedOfSound;
         float distance = 0;
         float gain = 1;
@@ -66785,7 +66785,7 @@ MA_API ma_result ma_noise_set_type(ma_noise* pNoise, ma_noise_type type)
 
     /*
     This function should never have been implemented in the first place. Changing the type dynamically is not
-    supported. Instead you need to uninitialize and reinitiailize a fresh `ma_noise` object. This function
+    supported. Instead you need to uninitialize and reinitialize a fresh `ma_noise` object. This function
     will be removed in version 0.12.
     */
     MA_ASSERT(MA_FALSE);
@@ -70864,7 +70864,7 @@ static ma_result ma_job_process__resource_manager__load_data_buffer(ma_job* pJob
     */
     result = ma_resource_manager_data_buffer_result(pDataBuffer);
     if (result != MA_BUSY) {
-        goto done;  /* <-- This will ensure the exucution pointer is incremented. */
+        goto done;  /* <-- This will ensure the execution pointer is incremented. */
     } else {
         result = MA_SUCCESS;    /* <-- Make sure this is reset. */
     }
@@ -75968,7 +75968,7 @@ MA_API ma_result ma_engine_play_sound_ex(ma_engine* pEngine, const char* pFilePa
         return MA_INVALID_ARGS;
     }
 
-    /* Attach to the endpoint node if nothing is specicied. */
+    /* Attach to the endpoint node if nothing is specified. */
     if (pNode == NULL) {
         pNode = ma_node_graph_get_endpoint(&pEngine->nodeGraph);
         nodeInputBusIndex = 0;