fix typos

CHANGES.md

@@ -183,7 +183,7 @@ v0.11.7 - 2022-02-06
 
 v0.11.6 - 2022-01-22
 ====================
-* WASAPI: Fix a bug where the device is not stopped when an error occurrs when writing to a playback device.
+* WASAPI: Fix a bug where the device is not stopped when an error occurs when writing to a playback device.
 * PulseAudio: Fix a rare crash due to a division by zero.
 * The node graph can now be used as a node. This allows node graphs to be connected to other node graphs.
 * Fix a crash with high-pass and band-pass filters.
@@ -477,7 +477,7 @@ v0.10.26 - 2020-11-24
 
 v0.10.25 - 2020-11-15
   - PulseAudio: Fix a bug where the stop callback isn't fired.
-  - WebAudio: Fix an error that occurs when Emscripten increases the size of it's heap.
+  - WebAudio: Fix an error that occurs when Emscripten increases the size of its heap.
   - Custom Backends: Change the onContextInit and onDeviceInit callbacks to take a parameter which is a pointer to the config that was
     passed into ma_context_init() and ma_device_init(). This replaces the deviceType parameter of onDeviceInit.
   - Fix compilation warnings on older versions of GCC.

CONTRIBUTING.md

@@ -21,7 +21,7 @@ significant work without talking to me first. If I don't like it, it won't be me
 [Discord](https://discord.gg/9vpqbjU) and [Twitter](https://twitter.com/mackron).
 
 Always base your pull request branch on the "dev" branch. The master branch contains the latest release, which
-means your pull request may not be including the lastest in-development changes which may result in unnecessary
+means your pull request may not be including the latest in-development changes which may result in unnecessary
 conflicts.
 
 I need to review your pull requests before merging. If your pull request is non-trivial, try to break it up into
@@ -63,7 +63,7 @@ not contribute to this project.
 
 Predictable Questions
 ---------------------
-### "Would you consider splitting out [some section of code] into it's own file?"
+### "Would you consider splitting out [some section of code] into its own file?"
 No, the idea is to keep everything in one place. It would be nice in specific cases to split out specific sections
 of the code, such as the resampler, for example. However, this will completely violate one of the major goals of the
 project - to have a complete audio library contained within a single file.

README.md

@@ -180,7 +180,7 @@ Documentation
 Online documentation can be found here: https://miniaud.io/docs/
 
 Documentation can also be found at the top of [miniaudio.h](https://raw.githubusercontent.com/mackron/miniaudio/master/miniaudio.h)
-which is always the most up-to-date and authoritive source of information on how to use miniaudio. All other
+which is always the most up-to-date and authoritative source of information on how to use miniaudio. All other
 documentation is generated from this in-code documentation.
 
 

examples/custom_backend.c

@@ -261,7 +261,7 @@ static ma_result ma_context_get_device_info__sdl(ma_context* pContext, ma_device
     of the device's _actual_ ideal format.
     
     Note: With Emscripten, it looks like non-zero values need to be specified for desiredSpec. Whatever is specified in
-    desiredSpec will be used by SDL since it uses it just does it's own format conversion internally. Therefore, from what
+    desiredSpec will be used by SDL since it uses it just does its own format conversion internally. Therefore, from what
     I can tell, there's no real way to know the device's actual format which means I'm just going to fall back to the full
     range of channels and sample rates on Emscripten builds.
     */

examples/simple_enumeration.c

 /*
 Demonstrates how to enumerate over devices.
 
-Device enumaration requires a `ma_context` object which is initialized with `ma_context_init()`. Conceptually, the
+Device enumeration requires a `ma_context` object which is initialized with `ma_context_init()`. Conceptually, the
 context sits above a device. You can have many devices to one context.
 
 If you use device enumeration, you should explicitly specify the same context you used for enumeration in the call to

examples/simple_mixing.c

@@ -4,7 +4,7 @@ Demonstrates one way to load multiple files and play them all back at the same t
 When mixing multiple sounds together, you should not create multiple devices. Instead you should create only a single
 device and then mix your sounds together which you can do by simply summing their samples together. The simplest way to
 do this is to use floating point samples and use miniaudio's built-in clipper to handling clipping for you. (Clipping
-is when sample are clampled to their minimum and maximum range, which for floating point is -1..1.)
+is when sample are clamped to their minimum and maximum range, which for floating point is -1..1.)
 
 ```
 Usage:   simple_mixing [input file 0] [input file 1] ... [input file n]

extras/nodes/ma_channel_combiner_node/ma_channel_combiner_node_example.c

-/* The channel separtor example also demonstrates how to use the combiner. */
+/* The channel separator example also demonstrates how to use the combiner. */
 #include "../ma_channel_separator_node/ma_channel_separator_node_example.c"

extras/osaudio/osaudio.h

@@ -27,7 +27,7 @@ A complete low-level audio solution requires the following:
     2) The ability to open and close a connection to a device.
     3) The ability to start and stop the device.
     4) The ability to write and read audio data to/from the device.
-    5) The ability to query the device for it's data configuration.
+    5) The ability to query the device for its data configuration.
     6) The ability to notify the application when certain events occur, such as the device being
        stopped, or rerouted.
 
@@ -131,7 +131,7 @@ is an example for how to do this:
     free(info); // The pointer returned by osaudio_enumerate() must be freed with free().
 
 The id structure is just a 256 byte array that uniquely identifies the device. Implementations may
-have different representations for device IDs, and A 256 byte array should accomodates all
+have different representations for device IDs, and A 256 byte array should accommodates all
 device ID representations. Implementations are required to zero-fill unused bytes. The osaudio_id_t
 structure can be copied which makes it suitable for serialization and deserialization in situations
 where you may want to save the device ID to permanent storage so it can be stored in a config file.
@@ -161,7 +161,7 @@ achieved by setting the format, channels and rate to 0. Below is an example:
 
 In addition to the code above, you can explicitly call `osaudio_get_info()` to retrieve the format
 configuration. If you need to know the native configuration before opening the device, you can use
-enumeration. The format, channels and rate will be contined in the first item in the configs array.
+enumeration. The format, channels and rate will be continued in the first item in the configs array.
 
 The examples above all use playback, but the same applies for capture. The only difference is that
 the direction is set to OSAUDIO_INPUT instead of OSAUDIO_OUTPUT.
@@ -268,7 +268,7 @@ typedef int osaudio_result_t;
 #define OSAUDIO_XRUN                           -102   /* An underrun or overrun occurred. Can be returned by osaudio_read() or osaudio_write(). */
 #define OSAUDIO_DEVICE_STOPPED                 -103   /* The device is stopped. Can be returned by osaudio_drain(). It is invalid to call osaudio_drain() on a device that is not running because otherwise it'll get stuck. */
 
-/* Directions. Cannot be combined. Use separate osaudio_t objects for birectional setups. */
+/* Directions. Cannot be combined. Use separate osaudio_t objects for bidirectional setups. */
 typedef int osaudio_direction_t;
 #define OSAUDIO_INPUT                           1
 #define OSAUDIO_OUTPUT                          2
@@ -480,10 +480,10 @@ The code above is equivalent to this:
 On output the config will be filled with the actual configuration. The implementation will perform
 any necessary data conversion between the requested data configuration and the device's native
 configuration. If it cannot, the function will return a OSAUDIO_FORMAT_NOT_SUPPORTED error. In this
-case the caller can decide to reinitialize the device to use it's native configuration and do it's
+case the caller can decide to reinitialize the device to use its native configuration and do its
 own data conversion, or abort if it cannot do so. Use the channel map to determine the ordering of
 your channels. Automatic channel map conversion is not performed - that must be done manually by
-the caller when transfering data to/from the device.
+the caller when transferring data to/from the device.
 
 Close the device with osaudio_close().
 
@@ -494,7 +494,7 @@ osaudio_result_t osaudio_open(osaudio_t* audio, osaudio_config_t* config);
 /*
 Closes a connection to a device.
 
-As soon as this function is called, the device should be considered invalid and unsuable. Do not
+As soon as this function is called, the device should be considered invalid and unusable. Do not
 attempt to use the audio object once this function has been called.
 
 It's invalid to call this while any other function is still running. You can use osaudio_flush() to
@@ -593,7 +593,7 @@ There will be one item in the configs array which will contain the device's curr
 the contents of which will match that of the config that was returned by osaudio_open().
 
 Returns NULL on failure. Do not free the returned pointer. It's up to the implementation to manage
-the meory of this object.
+the memory of this object.
 */
 const osaudio_info_t* osaudio_get_info(osaudio_t audio);
 

extras/osaudio/osaudio_miniaudio.c

@@ -122,7 +122,7 @@ static osaudio_result_t osaudio_determine_miniaudio_backend(ma_backend* pBackend
     /*
     To do this we initialize a dummy device. We allow the caller to make use of this device as an optimization. This is
     only used by osaudio_enumerate_devices() because that can make use of the context from the dummy device rather than
-    having to create it's own. pDummyDevice can be null.
+    having to create its own. pDummyDevice can be null.
     */
     if (pDummyDevice == NULL) {
         pDummyDevice = &dummyDevice;
@@ -135,7 +135,7 @@ static osaudio_result_t osaudio_determine_miniaudio_backend(ma_backend* pBackend
     if (result != MA_SUCCESS || pDummyDevice->pContext->backend == ma_backend_null) {
         /* Failed to open a default playback device. Try capture. */
         if (result == MA_SUCCESS) {
-            /* This means we successfully initialize a device, but it's backend is null. It could be that there's no playback devices attached. Try capture. */
+            /* This means we successfully initialize a device, but its backend is null. It could be that there's no playback devices attached. Try capture. */
             ma_device_uninit(pDummyDevice);
         }
 
@@ -607,7 +607,7 @@ osaudio_result_t osaudio_open(osaudio_t* audio, osaudio_config_t* config)
     }
 
 
-    /* The device object needs to have a it's local info built. We can get the ID and name from miniaudio. */
+    /* The device object needs to have its local info built. We can get the ID and name from miniaudio. */
     result = osaudio_result_from_miniaudio(ma_device_get_info(&(*audio)->device, (*audio)->device.type, &deviceInfo));
     if (result == MA_SUCCESS) {
         memcpy((*audio)->info.id.data, &deviceInfo.id, sizeof((*audio)->info.id.data));
@@ -630,7 +630,7 @@ osaudio_result_t osaudio_open(osaudio_t* audio, osaudio_config_t* config)
         return result;
     }
 
-    /* Now we need a semaphore to control access to the ring buffer to to block read/write when necessary. */
+    /* Now we need a semaphore to control access to the ring buffer to block read/write when necessary. */
     result = osaudio_result_from_miniaudio(ma_semaphore_init((config->direction == OSAUDIO_OUTPUT) ? periodCount : 0, &(*audio)->bufferSemaphore));
     if (result != OSAUDIO_SUCCESS) {
         ma_pcm_rb_uninit(&(*audio)->buffer);

extras/stb_vorbis.c

@@ -5531,7 +5531,7 @@ int stb_vorbis_get_samples_float(stb_vorbis *f, int channels, float **buffer, in
     0.97 - builds under c++ (typecasting, don't use 'class' keyword)
     0.96 - somehow MY 0.95 was right, but the web one was wrong, so here's my 0.95 rereleased as 0.96, fixes a typo in the clamping code
     0.95 - clamping code for 16-bit functions
-    0.94 - not publically released
+    0.94 - not publicly released
     0.93 - fixed all-zero-floor case (was decoding garbage)
     0.92 - fixed a memory leak
     0.91 - conditional compiles to omit parts of the API and the infrastructure to support them: STB_VORBIS_NO_PULLDATA_API, STB_VORBIS_NO_PUSHDATA_API, STB_VORBIS_NO_STDIO, STB_VORBIS_NO_INTEGER_CONVERSION

website/index.html

@@ -104,7 +104,7 @@
 						The resource manager is used for simplifying the hassle of dealing with the loading and management
                         of your audio resources. It will reference count files so they're only loaded once and handles
                         streaming of large audio sources to save on memory. It can even load files asynchronously and
-                        exposes it's job system so you can handle resource management jobs from your existing job
+                        exposes its job system so you can handle resource management jobs from your existing job
                         infrastructure.
 						</div>
 					</td>