Initial Commit

thirdparty/miniaudio-0.11.24/examples/build/README.md

@@ -0,0 +1,23 @@
+Examples
+--------
+    gcc ../simple_playback.c -o bin/simple_playback -ldl -lm -lpthread
+    gcc ../simple_playback.c -o bin/simple_playback -ldl -lm -lpthread -Wall -Wextra -Wpedantic -std=c89
+    
+Emscripten
+----------
+On Windows, you need to move into the build and run emsdk_env.bat from a command prompt using an absolute
+path like "C:\emsdk\emsdk_env.bat". Note that PowerShell doesn't work for me for some reason. Examples:
+
+    emcc ../simple_playback_sine.c -o bin/simple_playback_sine.html
+    emcc ../simple_playback_sine.c -o bin/simple_playback_sine.html -s WASM=0 -Wall -Wextra
+    
+To compile with support for Audio Worklets:
+
+    emcc ../simple_playback_sine.c -o bin/simple_playback_sine.html -DMA_ENABLE_AUDIO_WORKLETS -sAUDIO_WORKLET=1 -sWASM_WORKERS=1 -sASYNCIFY
+
+If you output WASM it may not work when running the web page locally. To test you can run with something
+like this:
+
+    emrun ./bin/simple_playback_sine.html
+
+If you want to see stdout on the command line when running from emrun, add `--emrun` to your emcc command.

thirdparty/miniaudio-0.11.24/examples/custom_backend.c

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+/*
+This example show how a custom backend can be implemented.
+
+This implements a full-featured SDL2 backend. It's intentionally built using the same paradigms as the built-in backends in order to make
+it suitable as a solid basis for a custom implementation. The SDL2 backend can be disabled with MA_NO_SDL, exactly like the built-in
+backends. It supports both runtime and compile-time linking and respects the MA_NO_RUNTIME_LINKING option. It also works on Emscripten
+which requires the `-s USE_SDL=2` option.
+
+There may be times where you want to support more than one custom backend. This example has been designed to make it easy to plug-in extra
+custom backends without needing to modify any of the base miniaudio initialization code. A custom context structure is declared called
+`ma_context_ex`. The first member of this structure is a `ma_context` object which allows it to be cast between the two. The same is done
+for devices, which is called `ma_device_ex`. In these structures there is a section for each custom backend, which in this example is just
+SDL. These are only enabled at compile time if `MA_SUPPORT_SDL` is defined, which it always is in this example (you may want to have some
+logic which more intelligently enables or disables SDL support).
+
+To use a custom backend, at a minimum you must set the `custom.onContextInit()` callback in the context config. You do not need to set the
+other callbacks, but if you don't, you must set them in the implementation of the `onContextInit()` callback which is done via an output
+parameter. This is the approach taken by this example because it's the simplest way to support multiple custom backends. The idea is that
+the `onContextInit()` callback is set to a generic "loader", which then calls out to a backend-specific implementation which then sets the
+remaining callbacks if it is successfully initialized.
+
+Custom backends are identified with the `ma_backend_custom` backend type. For the purpose of demonstration, this example only uses the
+`ma_backend_custom` backend type because otherwise the built-in backends would always get chosen first and none of the code for the custom
+backends would actually get hit. By default, the `ma_backend_custom` backend is the lowest priority backend, except for `ma_backend_null`.
+*/
+#include "../miniaudio.c"
+
+#ifdef __EMSCRIPTEN__
+#include <emscripten.h>
+
+void main_loop__em()
+{
+}
+#endif
+
+/* Support SDL on everything. */
+#define MA_SUPPORT_SDL
+
+/*
+Only enable SDL if it's hasn't been explicitly disabled (MA_NO_SDL) or enabled (MA_ENABLE_SDL with
+MA_ENABLE_ONLY_SPECIFIC_BACKENDS) and it's supported at compile time (MA_SUPPORT_SDL).
+*/
+#if defined(MA_SUPPORT_SDL) && !defined(MA_NO_SDL) && (!defined(MA_ENABLE_ONLY_SPECIFIC_BACKENDS) || defined(MA_ENABLE_SDL))
+    #define MA_HAS_SDL
+#endif
+
+
+typedef struct
+{
+    ma_context context; /* Make this the first member so we can cast between ma_context and ma_context_ex. */
+#if defined(MA_SUPPORT_SDL)
+    struct
+    {
+        ma_handle hSDL; /* A handle to the SDL2 shared object. We dynamically load function pointers at runtime so we can avoid linking. */
+        ma_proc SDL_InitSubSystem;
+        ma_proc SDL_QuitSubSystem;
+        ma_proc SDL_GetNumAudioDevices;
+        ma_proc SDL_GetAudioDeviceName;
+        ma_proc SDL_CloseAudioDevice;
+        ma_proc SDL_OpenAudioDevice;
+        ma_proc SDL_PauseAudioDevice;
+    } sdl;
+#endif
+} ma_context_ex;
+
+typedef struct
+{
+    ma_device device;   /* Make this the first member so we can cast between ma_device and ma_device_ex. */
+#if defined(MA_SUPPORT_SDL)
+    struct
+    {
+        int deviceIDPlayback;
+        int deviceIDCapture;
+    } sdl;
+#endif
+} ma_device_ex;
+
+
+
+#if defined(MA_HAS_SDL)
+    /* SDL headers are necessary if using compile-time linking. */
+    #ifdef MA_NO_RUNTIME_LINKING
+        #ifdef __has_include
+            #ifdef MA_EMSCRIPTEN
+                #if !__has_include(<SDL/SDL_audio.h>)
+                    #undef MA_HAS_SDL
+                #endif
+            #else
+                #if !__has_include(<SDL2/SDL_audio.h>)
+                    #undef MA_HAS_SDL
+                #endif
+            #endif
+        #endif
+    #endif
+#endif
+
+
+#if defined(MA_HAS_SDL)
+#define MA_SDL_INIT_AUDIO                       0x00000010
+#define MA_AUDIO_U8                             0x0008
+#define MA_AUDIO_S16                            0x8010
+#define MA_AUDIO_S32                            0x8020
+#define MA_AUDIO_F32                            0x8120
+#define MA_SDL_AUDIO_ALLOW_FREQUENCY_CHANGE     0x00000001
+#define MA_SDL_AUDIO_ALLOW_FORMAT_CHANGE        0x00000002
+#define MA_SDL_AUDIO_ALLOW_CHANNELS_CHANGE      0x00000004
+#define MA_SDL_AUDIO_ALLOW_ANY_CHANGE           (MA_SDL_AUDIO_ALLOW_FREQUENCY_CHANGE | MA_SDL_AUDIO_ALLOW_FORMAT_CHANGE | MA_SDL_AUDIO_ALLOW_CHANNELS_CHANGE)
+
+/* If we are linking at compile time we'll just #include SDL.h. Otherwise we can just redeclare some stuff to avoid the need for development packages to be installed. */
+#ifdef MA_NO_RUNTIME_LINKING
+    #define SDL_MAIN_HANDLED
+    #ifdef MA_EMSCRIPTEN
+        #include <SDL/SDL.h>
+    #else
+        #include <SDL2/SDL.h>
+    #endif
+
+    typedef SDL_AudioCallback   MA_SDL_AudioCallback;
+    typedef SDL_AudioSpec       MA_SDL_AudioSpec;
+    typedef SDL_AudioFormat     MA_SDL_AudioFormat;
+    typedef SDL_AudioDeviceID   MA_SDL_AudioDeviceID;
+#else
+    typedef void (* MA_SDL_AudioCallback)(void* userdata, ma_uint8* stream, int len);
+    typedef ma_uint16 MA_SDL_AudioFormat;
+    typedef ma_uint32 MA_SDL_AudioDeviceID;
+
+    typedef struct MA_SDL_AudioSpec
+    {
+        int freq;
+        MA_SDL_AudioFormat format;
+        ma_uint8 channels;
+        ma_uint8 silence;
+        ma_uint16 samples;
+        ma_uint16 padding;
+        ma_uint32 size;
+        MA_SDL_AudioCallback callback;
+        void* userdata;
+    } MA_SDL_AudioSpec;
+#endif
+
+typedef int                  (* MA_PFN_SDL_InitSubSystem)(ma_uint32 flags);
+typedef void                 (* MA_PFN_SDL_QuitSubSystem)(ma_uint32 flags);
+typedef int                  (* MA_PFN_SDL_GetNumAudioDevices)(int iscapture);
+typedef const char*          (* MA_PFN_SDL_GetAudioDeviceName)(int index, int iscapture);
+typedef void                 (* MA_PFN_SDL_CloseAudioDevice)(MA_SDL_AudioDeviceID dev);
+typedef MA_SDL_AudioDeviceID (* MA_PFN_SDL_OpenAudioDevice)(const char* device, int iscapture, const MA_SDL_AudioSpec* desired, MA_SDL_AudioSpec* obtained, int allowed_changes);
+typedef void                 (* MA_PFN_SDL_PauseAudioDevice)(MA_SDL_AudioDeviceID dev, int pause_on);
+
+MA_SDL_AudioFormat ma_format_to_sdl(ma_format format)
+{
+    switch (format)
+    {
+    case ma_format_unknown: return 0;
+    case ma_format_u8:      return MA_AUDIO_U8;
+    case ma_format_s16:     return MA_AUDIO_S16;
+    case ma_format_s24:     return MA_AUDIO_S32;  /* Closest match. */
+    case ma_format_s32:     return MA_AUDIO_S32;
+    case ma_format_f32:     return MA_AUDIO_F32;
+    default:                return 0;
+    }
+}
+
+ma_format ma_format_from_sdl(MA_SDL_AudioFormat format)
+{
+    switch (format)
+    {
+        case MA_AUDIO_U8:  return ma_format_u8;
+        case MA_AUDIO_S16: return ma_format_s16;
+        case MA_AUDIO_S32: return ma_format_s32;
+        case MA_AUDIO_F32: return ma_format_f32;
+        default:           return ma_format_unknown;
+    }
+}
+
+static ma_result ma_context_enumerate_devices__sdl(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
+{
+    ma_context_ex* pContextEx = (ma_context_ex*)pContext;
+    ma_bool32 isTerminated = MA_FALSE;
+    ma_bool32 cbResult;
+    int iDevice;
+
+    /* Playback */
+    if (!isTerminated) {
+        int deviceCount = ((MA_PFN_SDL_GetNumAudioDevices)pContextEx->sdl.SDL_GetNumAudioDevices)(0);
+        for (iDevice = 0; iDevice < deviceCount; ++iDevice) {
+            ma_device_info deviceInfo;
+            MA_ZERO_OBJECT(&deviceInfo);
+
+            deviceInfo.id.custom.i = iDevice;
+            ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), ((MA_PFN_SDL_GetAudioDeviceName)pContextEx->sdl.SDL_GetAudioDeviceName)(iDevice, 0), (size_t)-1);
+
+            if (iDevice == 0) {
+                deviceInfo.isDefault = MA_TRUE;
+            }
+
+            cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData);
+            if (cbResult == MA_FALSE) {
+                isTerminated = MA_TRUE;
+                break;
+            }
+        }
+    }
+
+    /* Capture */
+    if (!isTerminated) {
+        int deviceCount = ((MA_PFN_SDL_GetNumAudioDevices)pContextEx->sdl.SDL_GetNumAudioDevices)(1);
+        for (iDevice = 0; iDevice < deviceCount; ++iDevice) {
+            ma_device_info deviceInfo;
+            MA_ZERO_OBJECT(&deviceInfo);
+
+            deviceInfo.id.custom.i = iDevice;
+            ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), ((MA_PFN_SDL_GetAudioDeviceName)pContextEx->sdl.SDL_GetAudioDeviceName)(iDevice, 1), (size_t)-1);
+
+            if (iDevice == 0) {
+                deviceInfo.isDefault = MA_TRUE;
+            }
+
+            cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData);
+            if (cbResult == MA_FALSE) {
+                isTerminated = MA_TRUE;
+                break;
+            }
+        }
+    }
+
+    return MA_SUCCESS;
+}
+
+static ma_result ma_context_get_device_info__sdl(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
+{
+    ma_context_ex* pContextEx = (ma_context_ex*)pContext;
+
+#if !defined(__EMSCRIPTEN__)
+    MA_SDL_AudioSpec desiredSpec;
+    MA_SDL_AudioSpec obtainedSpec;
+    MA_SDL_AudioDeviceID tempDeviceID;
+    const char* pDeviceName;
+#endif
+
+    if (pDeviceID == NULL) {
+        if (deviceType == ma_device_type_playback) {
+            pDeviceInfo->id.custom.i = 0;
+            ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
+        } else {
+            pDeviceInfo->id.custom.i = 0;
+            ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
+        }
+    } else {
+        pDeviceInfo->id.custom.i = pDeviceID->custom.i;
+        ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), ((MA_PFN_SDL_GetAudioDeviceName)pContextEx->sdl.SDL_GetAudioDeviceName)(pDeviceID->custom.i, (deviceType == ma_device_type_playback) ? 0 : 1), (size_t)-1);
+    }
+
+    if (pDeviceInfo->id.custom.i == 0) {
+        pDeviceInfo->isDefault = MA_TRUE;
+    }
+
+    /*
+    To get an accurate idea on the backend's native format we need to open the device. Not ideal, but it's the only way. An
+    alternative to this is to report all channel counts, sample rates and formats, but that doesn't offer a good representation
+    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 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.
+    */
+#if defined(__EMSCRIPTEN__)
+    /* Good practice to prioritize the best format first so that the application can use the first data format as their chosen one if desired. */
+    pDeviceInfo->nativeDataFormatCount = 3;
+    pDeviceInfo->nativeDataFormats[0].format     = ma_format_s16;
+    pDeviceInfo->nativeDataFormats[0].channels   = 0;   /* All channel counts supported. */
+    pDeviceInfo->nativeDataFormats[0].sampleRate = 0;   /* All sample rates supported. */
+    pDeviceInfo->nativeDataFormats[0].flags      = 0;
+    pDeviceInfo->nativeDataFormats[1].format     = ma_format_s32;
+    pDeviceInfo->nativeDataFormats[1].channels   = 0;   /* All channel counts supported. */
+    pDeviceInfo->nativeDataFormats[1].sampleRate = 0;   /* All sample rates supported. */
+    pDeviceInfo->nativeDataFormats[1].flags      = 0;
+    pDeviceInfo->nativeDataFormats[2].format     = ma_format_u8;
+    pDeviceInfo->nativeDataFormats[2].channels   = 0;   /* All channel counts supported. */
+    pDeviceInfo->nativeDataFormats[2].sampleRate = 0;   /* All sample rates supported. */
+    pDeviceInfo->nativeDataFormats[2].flags      = 0;
+#else
+    MA_ZERO_MEMORY(&desiredSpec, sizeof(desiredSpec));
+
+    pDeviceName = NULL;
+    if (pDeviceID != NULL) {
+        pDeviceName = ((MA_PFN_SDL_GetAudioDeviceName)pContextEx->sdl.SDL_GetAudioDeviceName)(pDeviceID->custom.i, (deviceType == ma_device_type_playback) ? 0 : 1);
+    }
+
+    tempDeviceID = ((MA_PFN_SDL_OpenAudioDevice)pContextEx->sdl.SDL_OpenAudioDevice)(pDeviceName, (deviceType == ma_device_type_playback) ? 0 : 1, &desiredSpec, &obtainedSpec, MA_SDL_AUDIO_ALLOW_ANY_CHANGE);
+    if (tempDeviceID == 0) {
+        ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "Failed to open SDL device.");
+        return MA_FAILED_TO_OPEN_BACKEND_DEVICE;
+    }
+
+    ((MA_PFN_SDL_CloseAudioDevice)pContextEx->sdl.SDL_CloseAudioDevice)(tempDeviceID);
+
+    /* Only reporting a single native data format. It'll be whatever SDL decides is the best. */
+    pDeviceInfo->nativeDataFormatCount = 1;
+    pDeviceInfo->nativeDataFormats[0].format     = ma_format_from_sdl(obtainedSpec.format);
+    pDeviceInfo->nativeDataFormats[0].channels   = obtainedSpec.channels;
+    pDeviceInfo->nativeDataFormats[0].sampleRate = obtainedSpec.freq;
+    pDeviceInfo->nativeDataFormats[0].flags      = 0;
+
+    /* If miniaudio does not support the format, just use f32 as the native format (SDL will do the necessary conversions for us). */
+    if (pDeviceInfo->nativeDataFormats[0].format == ma_format_unknown) {
+        pDeviceInfo->nativeDataFormats[0].format = ma_format_f32;
+    }
+#endif  /* __EMSCRIPTEN__ */
+
+    return MA_SUCCESS;
+}
+
+
+void ma_audio_callback_capture__sdl(void* pUserData, ma_uint8* pBuffer, int bufferSizeInBytes)
+{
+    ma_device_ex* pDeviceEx = (ma_device_ex*)pUserData;
+
+    ma_device_handle_backend_data_callback((ma_device*)pDeviceEx, NULL, pBuffer, (ma_uint32)bufferSizeInBytes / ma_get_bytes_per_frame(pDeviceEx->device.capture.internalFormat, pDeviceEx->device.capture.internalChannels));
+}
+
+void ma_audio_callback_playback__sdl(void* pUserData, ma_uint8* pBuffer, int bufferSizeInBytes)
+{
+    ma_device_ex* pDeviceEx = (ma_device_ex*)pUserData;
+
+    ma_device_handle_backend_data_callback((ma_device*)pDeviceEx, pBuffer, NULL, (ma_uint32)bufferSizeInBytes / ma_get_bytes_per_frame(pDeviceEx->device.playback.internalFormat, pDeviceEx->device.playback.internalChannels));
+}
+
+static ma_result ma_device_init_internal__sdl(ma_device_ex* pDeviceEx, const ma_device_config* pConfig, ma_device_descriptor* pDescriptor)
+{
+    ma_context_ex* pContextEx = (ma_context_ex*)pDeviceEx->device.pContext;
+    MA_SDL_AudioSpec desiredSpec;
+    MA_SDL_AudioSpec obtainedSpec;
+    const char* pDeviceName;
+    int deviceID;
+
+    /*
+    SDL is a little bit awkward with specifying the buffer size, You need to specify the size of the buffer in frames, but since we may
+    have requested a period size in milliseconds we'll need to convert, which depends on the sample rate. But there's a possibility that
+    the sample rate just set to 0, which indicates that the native sample rate should be used. There's no practical way to calculate this
+    that I can think of right now so I'm just using MA_DEFAULT_SAMPLE_RATE.
+    */
+    if (pDescriptor->sampleRate == 0) {
+        pDescriptor->sampleRate = MA_DEFAULT_SAMPLE_RATE;
+    }
+
+    /*
+    When determining the period size, you need to take defaults into account. This is how the size of the period should be determined.
+
+        1) If periodSizeInFrames is not 0, use periodSizeInFrames; else
+        2) If periodSizeInMilliseconds is not 0, use periodSizeInMilliseconds; else
+        3) If both periodSizeInFrames and periodSizeInMilliseconds is 0, use the backend's default. If the backend does not allow a default
+           buffer size, use a default value of MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_LOW_LATENCY or 
+           MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_CONSERVATIVE depending on the value of pConfig->performanceProfile.
+
+    Note that options 2 and 3 require knowledge of the sample rate in order to convert it to a frame count. You should try to keep the
+    calculation of the period size as accurate as possible, but sometimes it's just not practical so just use whatever you can.
+
+    A helper function called ma_calculate_buffer_size_in_frames_from_descriptor() is available to do all of this for you which is what
+    we'll be using here.
+    */
+    pDescriptor->periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, pDescriptor->sampleRate, pConfig->performanceProfile);
+
+    /* SDL wants the buffer size to be a power of 2 for some reason. */
+    if (pDescriptor->periodSizeInFrames > 32768) {
+        pDescriptor->periodSizeInFrames = 32768;
+    } else {
+        pDescriptor->periodSizeInFrames = ma_next_power_of_2(pDescriptor->periodSizeInFrames);
+    }
+
+
+    /* We now have enough information to set up the device. */
+    MA_ZERO_OBJECT(&desiredSpec);
+    desiredSpec.freq     = (int)pDescriptor->sampleRate;
+    desiredSpec.format   = ma_format_to_sdl(pDescriptor->format);
+    desiredSpec.channels = (ma_uint8)pDescriptor->channels;
+    desiredSpec.samples  = (ma_uint16)pDescriptor->periodSizeInFrames;
+    desiredSpec.callback = (pConfig->deviceType == ma_device_type_capture) ? ma_audio_callback_capture__sdl : ma_audio_callback_playback__sdl;
+    desiredSpec.userdata = pDeviceEx;
+
+    /* We'll fall back to f32 if we don't have an appropriate mapping between SDL and miniaudio. */
+    if (desiredSpec.format == 0) {
+        desiredSpec.format = MA_AUDIO_F32;
+    }
+
+    pDeviceName = NULL;
+    if (pDescriptor->pDeviceID != NULL) {
+        pDeviceName = ((MA_PFN_SDL_GetAudioDeviceName)pContextEx->sdl.SDL_GetAudioDeviceName)(pDescriptor->pDeviceID->custom.i, (pConfig->deviceType == ma_device_type_playback) ? 0 : 1);
+    }
+
+    deviceID = ((MA_PFN_SDL_OpenAudioDevice)pContextEx->sdl.SDL_OpenAudioDevice)(pDeviceName, (pConfig->deviceType == ma_device_type_playback) ? 0 : 1, &desiredSpec, &obtainedSpec, MA_SDL_AUDIO_ALLOW_ANY_CHANGE);
+    if (deviceID == 0) {
+        ma_log_postf(ma_device_get_log((ma_device*)pDeviceEx), MA_LOG_LEVEL_ERROR, "Failed to open SDL2 device.");
+        return MA_FAILED_TO_OPEN_BACKEND_DEVICE;
+    }
+
+    if (pConfig->deviceType == ma_device_type_playback) {
+        pDeviceEx->sdl.deviceIDPlayback = deviceID;
+    } else {
+        pDeviceEx->sdl.deviceIDCapture = deviceID;
+    }
+
+    /* The descriptor needs to be updated with our actual settings. */
+    pDescriptor->format             = ma_format_from_sdl(obtainedSpec.format);
+    pDescriptor->channels           = obtainedSpec.channels;
+    pDescriptor->sampleRate         = (ma_uint32)obtainedSpec.freq;
+    ma_channel_map_init_standard(ma_standard_channel_map_default, pDescriptor->channelMap, ma_countof(pDescriptor->channelMap), pDescriptor->channels);
+    pDescriptor->periodSizeInFrames = obtainedSpec.samples;
+    pDescriptor->periodCount        = 1;    /* SDL doesn't use the notion of period counts, so just set to 1. */
+
+    return MA_SUCCESS;
+}
+
+static ma_result ma_device_init__sdl(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
+{
+    ma_device_ex* pDeviceEx = (ma_device_ex*)pDevice;
+    ma_context_ex* pContextEx = (ma_context_ex*)pDevice->pContext;
+    ma_result result;
+
+    /* SDL does not support loopback mode, so must return MA_DEVICE_TYPE_NOT_SUPPORTED if it's requested. */
+    if (pConfig->deviceType == ma_device_type_loopback) {
+        return MA_DEVICE_TYPE_NOT_SUPPORTED;
+    }
+
+    if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
+        result = ma_device_init_internal__sdl(pDeviceEx, pConfig, pDescriptorCapture);
+        if (result != MA_SUCCESS) {
+            return result;
+        }
+    }
+
+    if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
+        result = ma_device_init_internal__sdl(pDeviceEx, pConfig, pDescriptorPlayback);
+        if (result != MA_SUCCESS) {
+            if (pConfig->deviceType == ma_device_type_duplex) {
+                ((MA_PFN_SDL_CloseAudioDevice)pContextEx->sdl.SDL_CloseAudioDevice)(pDeviceEx->sdl.deviceIDCapture);
+            }
+
+            return result;
+        }
+    }
+
+    return MA_SUCCESS;
+}
+
+static ma_result ma_device_uninit__sdl(ma_device* pDevice)
+{
+    ma_device_ex* pDeviceEx = (ma_device_ex*)pDevice;
+    ma_context_ex* pContextEx = (ma_context_ex*)pDevice->pContext;
+
+    if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+        ((MA_PFN_SDL_CloseAudioDevice)pContextEx->sdl.SDL_CloseAudioDevice)(pDeviceEx->sdl.deviceIDCapture);
+    }
+
+    if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+        ((MA_PFN_SDL_CloseAudioDevice)pContextEx->sdl.SDL_CloseAudioDevice)(pDeviceEx->sdl.deviceIDCapture);
+    }
+
+    return MA_SUCCESS;
+}
+
+static ma_result ma_device_start__sdl(ma_device* pDevice)
+{
+    ma_device_ex* pDeviceEx = (ma_device_ex*)pDevice;
+    ma_context_ex* pContextEx = (ma_context_ex*)pDevice->pContext;
+
+    if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+        ((MA_PFN_SDL_PauseAudioDevice)pContextEx->sdl.SDL_PauseAudioDevice)(pDeviceEx->sdl.deviceIDCapture, 0);
+    }
+
+    if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+        ((MA_PFN_SDL_PauseAudioDevice)pContextEx->sdl.SDL_PauseAudioDevice)(pDeviceEx->sdl.deviceIDPlayback, 0);
+    }
+
+    return MA_SUCCESS;
+}
+
+static ma_result ma_device_stop__sdl(ma_device* pDevice)
+{
+    ma_device_ex* pDeviceEx = (ma_device_ex*)pDevice;
+    ma_context_ex* pContextEx = (ma_context_ex*)pDevice->pContext;
+
+    if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
+        ((MA_PFN_SDL_PauseAudioDevice)pContextEx->sdl.SDL_PauseAudioDevice)(pDeviceEx->sdl.deviceIDCapture, 1);
+    }
+
+    if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
+        ((MA_PFN_SDL_PauseAudioDevice)pContextEx->sdl.SDL_PauseAudioDevice)(pDeviceEx->sdl.deviceIDPlayback, 1);
+    }
+
+    return MA_SUCCESS;
+}
+
+static ma_result ma_context_uninit__sdl(ma_context* pContext)
+{
+    ma_context_ex* pContextEx = (ma_context_ex*)pContext;
+
+    ((MA_PFN_SDL_QuitSubSystem)pContextEx->sdl.SDL_QuitSubSystem)(MA_SDL_INIT_AUDIO);
+
+    /* Close the handle to the SDL shared object last. */
+    ma_dlclose(ma_context_get_log(pContext), pContextEx->sdl.hSDL);
+    pContextEx->sdl.hSDL = NULL;
+
+    return MA_SUCCESS;
+}
+
+static ma_result ma_context_init__sdl(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
+{
+    ma_context_ex* pContextEx = (ma_context_ex*)pContext;
+    int resultSDL;
+    
+#ifndef MA_NO_RUNTIME_LINKING
+    /* We'll use a list of possible shared object names for easier extensibility. */
+    size_t iName;
+    const char* pSDLNames[] = {
+#if defined(_WIN32)
+        "SDL2.dll"
+#elif defined(__APPLE__)
+        "SDL2.framework/SDL2"
+#else
+        "libSDL2-2.0.so.0"
+#endif
+    };
+
+    (void)pConfig;
+
+    /* Check if we have SDL2 installed somewhere. If not it's not usable and we need to abort. */
+    for (iName = 0; iName < ma_countof(pSDLNames); iName += 1) {
+        pContextEx->sdl.hSDL = ma_dlopen(ma_context_get_log(pContext), pSDLNames[iName]);
+        if (pContextEx->sdl.hSDL != NULL) {
+            break;
+        }
+    }
+
+    if (pContextEx->sdl.hSDL == NULL) {
+        return MA_NO_BACKEND;   /* SDL2 could not be loaded. */
+    }
+
+    /* Now that we have the handle to the shared object we can go ahead and load some function pointers. */
+    pContextEx->sdl.SDL_InitSubSystem      = ma_dlsym(ma_context_get_log(pContext), pContextEx->sdl.hSDL, "SDL_InitSubSystem");
+    pContextEx->sdl.SDL_QuitSubSystem      = ma_dlsym(ma_context_get_log(pContext), pContextEx->sdl.hSDL, "SDL_QuitSubSystem");
+    pContextEx->sdl.SDL_GetNumAudioDevices = ma_dlsym(ma_context_get_log(pContext), pContextEx->sdl.hSDL, "SDL_GetNumAudioDevices");
+    pContextEx->sdl.SDL_GetAudioDeviceName = ma_dlsym(ma_context_get_log(pContext), pContextEx->sdl.hSDL, "SDL_GetAudioDeviceName");
+    pContextEx->sdl.SDL_CloseAudioDevice   = ma_dlsym(ma_context_get_log(pContext), pContextEx->sdl.hSDL, "SDL_CloseAudioDevice");
+    pContextEx->sdl.SDL_OpenAudioDevice    = ma_dlsym(ma_context_get_log(pContext), pContextEx->sdl.hSDL, "SDL_OpenAudioDevice");
+    pContextEx->sdl.SDL_PauseAudioDevice   = ma_dlsym(ma_context_get_log(pContext), pContextEx->sdl.hSDL, "SDL_PauseAudioDevice");
+#else
+    pContextEx->sdl.SDL_InitSubSystem      = (ma_proc)SDL_InitSubSystem;
+    pContextEx->sdl.SDL_QuitSubSystem      = (ma_proc)SDL_QuitSubSystem;
+    pContextEx->sdl.SDL_GetNumAudioDevices = (ma_proc)SDL_GetNumAudioDevices;
+    pContextEx->sdl.SDL_GetAudioDeviceName = (ma_proc)SDL_GetAudioDeviceName;
+    pContextEx->sdl.SDL_CloseAudioDevice   = (ma_proc)SDL_CloseAudioDevice;
+    pContextEx->sdl.SDL_OpenAudioDevice    = (ma_proc)SDL_OpenAudioDevice;
+    pContextEx->sdl.SDL_PauseAudioDevice   = (ma_proc)SDL_PauseAudioDevice;
+#endif  /* MA_NO_RUNTIME_LINKING */
+
+    resultSDL = ((MA_PFN_SDL_InitSubSystem)pContextEx->sdl.SDL_InitSubSystem)(MA_SDL_INIT_AUDIO);
+    if (resultSDL != 0) {
+        ma_dlclose(ma_context_get_log(pContext), pContextEx->sdl.hSDL);
+        return MA_ERROR;
+    }
+
+    /*
+    The last step is to make sure the callbacks are set properly in `pCallbacks`. Internally, miniaudio will copy these callbacks into the
+    context object and then use them for then on for calling into our custom backend.
+    */
+    pCallbacks->onContextInit             = ma_context_init__sdl;
+    pCallbacks->onContextUninit           = ma_context_uninit__sdl;
+    pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__sdl;
+    pCallbacks->onContextGetDeviceInfo    = ma_context_get_device_info__sdl;
+    pCallbacks->onDeviceInit              = ma_device_init__sdl;
+    pCallbacks->onDeviceUninit            = ma_device_uninit__sdl;
+    pCallbacks->onDeviceStart             = ma_device_start__sdl;
+    pCallbacks->onDeviceStop              = ma_device_stop__sdl;
+
+    return MA_SUCCESS;
+}
+#endif  /* MA_HAS_SDL */
+
+
+/*
+This is our custom backend "loader". All this does is attempts to initialize our custom backends in the order they are listed. The first
+one to successfully initialize is the one that's chosen. In this example we're just listing them statically, but you can use whatever logic
+you want to handle backend selection.
+
+This is used as the onContextInit() callback in the context config.
+*/
+static ma_result ma_context_init__custom_loader(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
+{
+    ma_result result = MA_NO_BACKEND;
+
+    /* Silence some unused parameter warnings just in case no custom backends are enabled. */
+    (void)pContext;
+    (void)pCallbacks;
+
+    /* SDL. */
+#if !defined(MA_NO_SDL)
+    if (result != MA_SUCCESS) {
+        result = ma_context_init__sdl(pContext, pConfig, pCallbacks);
+    }
+#endif
+
+    /* ... plug in any other custom backends here ... */
+
+    /* If we have a success result we have initialized a backend. Otherwise we need to tell miniaudio about the error so it can skip over our custom backends. */
+    return result;
+}
+
+
+/*
+Main program starts here.
+*/
+#define DEVICE_FORMAT       ma_format_f32
+#define DEVICE_CHANNELS     2
+#define DEVICE_SAMPLE_RATE  48000
+
+void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
+{
+    if (pDevice->type == ma_device_type_playback)  {
+        ma_waveform_read_pcm_frames((ma_waveform*)pDevice->pUserData, pOutput, frameCount, NULL);
+    }
+
+    if (pDevice->type == ma_device_type_duplex) {
+        ma_copy_pcm_frames(pOutput, pInput, frameCount, pDevice->playback.format, pDevice->playback.channels);
+    }
+}
+
+int main(int argc, char** argv)
+{
+    ma_result result;
+    ma_context_config contextConfig;
+    ma_context_ex context;
+    ma_device_config deviceConfig;
+    ma_device_ex device;
+    ma_waveform_config sineWaveConfig;
+    ma_waveform sineWave;
+
+    /*
+    We're just using ma_backend_custom in this example for demonstration purposes, but a more realistic use case would probably want to include
+    other backends as well for robustness.
+    */
+    ma_backend backends[] = {
+        ma_backend_custom
+    };
+
+    /*
+    To implement a custom backend you need to implement the callbacks in the "custom" member of the context config. The only mandatory
+    callback required at this point is the onContextInit() callback. If you do not set the other callbacks, you must set them in
+    onContextInit() by setting them on the `pCallbacks` parameter.
+
+    The way we're doing it in this example enables us to easily plug in multiple custom backends. What we do is set the onContextInit()
+    callback to a generic "loader" function (ma_context_init__custom_loader() in this example), which then calls out to backend-specific
+    context initialization routines, one of which will be for SDL. That way, if for example we wanted to add support for another backend,
+    we don't need to touch this part of the code. Instead we add logic to ma_context_init__custom_loader() to choose the most appropriate
+    custom backend. That will then fill out the other callbacks appropriately.
+    */
+    contextConfig = ma_context_config_init();
+    contextConfig.custom.onContextInit = ma_context_init__custom_loader;
+    
+    result = ma_context_init(backends, sizeof(backends)/sizeof(backends[0]), &contextConfig, (ma_context*)&context);
+    if (result != MA_SUCCESS) {
+        return -1;
+    }
+
+    /* In playback mode we're just going to play a sine wave. */
+    sineWaveConfig = ma_waveform_config_init(DEVICE_FORMAT, DEVICE_CHANNELS, DEVICE_SAMPLE_RATE, ma_waveform_type_sine, 0.2, 220);
+    ma_waveform_init(&sineWaveConfig, &sineWave);
+
+    /* The device is created exactly as per normal. */
+    deviceConfig = ma_device_config_init(ma_device_type_playback);
+    deviceConfig.playback.format   = DEVICE_FORMAT;
+    deviceConfig.playback.channels = DEVICE_CHANNELS;
+    deviceConfig.capture.format    = DEVICE_FORMAT;
+    deviceConfig.capture.channels  = DEVICE_CHANNELS;
+    deviceConfig.sampleRate        = DEVICE_SAMPLE_RATE;
+    deviceConfig.dataCallback      = data_callback;
+    deviceConfig.pUserData         = &sineWave;
+
+    result = ma_device_init((ma_context*)&context, &deviceConfig, (ma_device*)&device);
+    if (result != MA_SUCCESS) {
+        ma_context_uninit((ma_context*)&context);
+        return -1;
+    }
+
+
+    printf("Device Name: %s\n", ((ma_device*)&device)->playback.name);
+
+    if (ma_device_start((ma_device*)&device) != MA_SUCCESS) {
+        ma_device_uninit((ma_device*)&device);
+        ma_context_uninit((ma_context*)&context);
+        return -5;
+    }
+    
+#ifdef __EMSCRIPTEN__
+    emscripten_set_main_loop(main_loop__em, 0, 1);
+#else
+    printf("Press Enter to quit...\n");
+    getchar();
+#endif
+    
+    ma_device_uninit((ma_device*)&device);
+    ma_context_uninit((ma_context*)&context);
+
+    (void)argc;
+    (void)argv;
+
+    return 0;
+}

thirdparty/miniaudio-0.11.24/examples/custom_decoder.c

@@ -0,0 +1,115 @@
+/*
+Demonstrates how to implement a custom decoder.
+
+This example implements two custom decoders:
+
+  * Vorbis via libvorbis
+  * Opus via libopus
+
+A custom decoder must implement a data source. In this example, the libvorbis data source is called
+`ma_libvorbis` and the Opus data source is called `ma_libopus`. These two objects are compatible
+with the `ma_data_source` APIs and can be taken straight from this example and used in real code.
+
+The custom decoding data sources (`ma_libvorbis` and `ma_libopus` in this example) are connected to
+the decoder via the decoder config (`ma_decoder_config`). You need to implement a vtable for each
+of your custom decoders. See `ma_decoding_backend_vtable` for the functions you need to implement.
+The `onInitFile`, `onInitFileW` and `onInitMemory` functions are optional.
+*/
+#include "../miniaudio.c"
+#include "../extras/decoders/libvorbis/miniaudio_libvorbis.c"
+#include "../extras/decoders/libopus/miniaudio_libopus.c"
+
+#include <stdio.h>
+
+
+void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
+{
+    ma_data_source* pDataSource = (ma_data_source*)pDevice->pUserData;
+    if (pDataSource == NULL) {
+        return;
+    }
+
+    ma_data_source_read_pcm_frames(pDataSource, pOutput, frameCount, NULL);
+
+    (void)pInput;
+}
+
+int main(int argc, char** argv)
+{
+    ma_result result;
+    ma_decoder_config decoderConfig;
+    ma_decoder decoder;
+    ma_device_config deviceConfig;
+    ma_device device;
+    ma_format format;
+    ma_uint32 channels;
+    ma_uint32 sampleRate;
+
+    /*
+    Add your custom backend vtables here. The order in the array defines the order of priority. The
+    vtables will be passed in via the decoder config.
+    */
+    ma_decoding_backend_vtable* pCustomBackendVTables[] =
+    {
+        ma_decoding_backend_libvorbis,
+        ma_decoding_backend_libopus
+    };
+
+
+    if (argc < 2) {
+        printf("No input file.\n");
+        return -1;
+    }
+
+    
+    /* Initialize the decoder. */
+    decoderConfig = ma_decoder_config_init_default();
+    decoderConfig.pCustomBackendUserData = NULL;  /* None of our decoders require user data, so this can be set to null. */
+    decoderConfig.ppCustomBackendVTables = pCustomBackendVTables;
+    decoderConfig.customBackendCount     = sizeof(pCustomBackendVTables) / sizeof(pCustomBackendVTables[0]);
+    
+    result = ma_decoder_init_file(argv[1], &decoderConfig, &decoder);
+    if (result != MA_SUCCESS) {
+        printf("Failed to initialize decoder.");
+        return -1;
+    }
+
+    ma_data_source_set_looping(&decoder, MA_TRUE);
+
+
+    /* Initialize the device. */
+    result = ma_data_source_get_data_format(&decoder, &format, &channels, &sampleRate, NULL, 0);
+    if (result != MA_SUCCESS) {
+        printf("Failed to retrieve decoder data format.");
+        ma_decoder_uninit(&decoder);
+        return -1;
+    }
+
+    deviceConfig = ma_device_config_init(ma_device_type_playback);
+    deviceConfig.playback.format   = format;
+    deviceConfig.playback.channels = channels;
+    deviceConfig.sampleRate        = sampleRate;
+    deviceConfig.dataCallback      = data_callback;
+    deviceConfig.pUserData         = &decoder;
+
+    if (ma_device_init(NULL, &deviceConfig, &device) != MA_SUCCESS) {
+        printf("Failed to open playback device.\n");
+        ma_decoder_uninit(&decoder);
+        return -1;
+    }
+
+    if (ma_device_start(&device) != MA_SUCCESS) {
+        printf("Failed to start playback device.\n");
+        ma_device_uninit(&device);
+        ma_decoder_uninit(&decoder);
+        return -1;
+    }
+
+    printf("Press Enter to quit...");
+    getchar();
+
+    ma_device_uninit(&device);
+    ma_decoder_uninit(&decoder);
+
+    return 0;
+}

thirdparty/miniaudio-0.11.24/examples/custom_decoder_engine.c

@@ -0,0 +1,79 @@
+/*
+Demonstrates how to implement a custom decoder and use it with the high level API.
+
+This is the same as the custom_decoder example, only it's used with the high level engine API
+rather than the low level decoding API. You can use this to add support for Opus to your games, for
+example (via libopus).
+*/
+#include "../miniaudio.c"
+#include "../extras/decoders/libvorbis/miniaudio_libvorbis.c"
+#include "../extras/decoders/libopus/miniaudio_libopus.c"
+
+#include <stdio.h>
+
+
+int main(int argc, char** argv)
+{
+    ma_result result;
+    ma_resource_manager_config resourceManagerConfig;
+    ma_resource_manager resourceManager;
+    ma_engine_config engineConfig;
+    ma_engine engine;
+
+    /*
+    Add your custom backend vtables here. The order in the array defines the order of priority. The
+    vtables will be passed in to the resource manager config.
+    */
+    ma_decoding_backend_vtable* pCustomBackendVTables[] =
+    {
+        ma_decoding_backend_libvorbis,
+        ma_decoding_backend_libopus
+    };
+
+
+    if (argc < 2) {
+        printf("No input file.\n");
+        return -1;
+    }
+
+
+    /* Using custom decoding backends requires a resource manager. */
+    resourceManagerConfig = ma_resource_manager_config_init();
+    resourceManagerConfig.ppCustomDecodingBackendVTables = pCustomBackendVTables;
+    resourceManagerConfig.customDecodingBackendCount     = sizeof(pCustomBackendVTables) / sizeof(pCustomBackendVTables[0]);
+    resourceManagerConfig.pCustomDecodingBackendUserData = NULL;  /* <-- This will be passed in to the pUserData parameter of each function in the decoding backend vtables. */
+
+    result = ma_resource_manager_init(&resourceManagerConfig, &resourceManager);
+    if (result != MA_SUCCESS) {
+        printf("Failed to initialize resource manager.");
+        return -1;
+    }
+
+
+    /* Once we have a resource manager we can create the engine. */
+    engineConfig = ma_engine_config_init();
+    engineConfig.pResourceManager = &resourceManager;
+
+    result = ma_engine_init(&engineConfig, &engine);
+    if (result != MA_SUCCESS) {
+        printf("Failed to initialize engine.");
+        return -1;
+    }
+
+
+    /* Now we can play our sound. */
+    result = ma_engine_play_sound(&engine, argv[1], NULL);
+    if (result != MA_SUCCESS) {
+        printf("Failed to play sound.");
+        return -1;
+    }
+
+
+    printf("Press Enter to quit...");
+    getchar();
+
+    ma_engine_uninit(&engine);
+    ma_resource_manager_uninit(&resourceManager);
+
+    return 0;
+}

thirdparty/miniaudio-0.11.24/examples/data_source_chaining.c

@@ -0,0 +1,161 @@
+/*
+Demonstrates one way to chain together a number of data sources so they play back seamlessly
+without gaps.
+
+This example uses the chaining system built into the `ma_data_source` API. It will take every sound
+passed onto the command line in order, and then loop back and start again. When looping a chain of
+data sources, you need only link the last data source back to the first one.
+
+To play a chain of data sources, you first need to set up your chain. To set the data source that
+should be played after another, you have two options:
+
+    * Set a pointer to a specific data source
+    * Set a callback that will fire when the next data source needs to be retrieved
+
+The first option is good for simple scenarios. The second option is useful if you need to perform
+some action when the end of a sound is reached. This example will be using both.
+
+When reading data from a chain, you always read from the head data source. Internally miniaudio
+will track a pointer to the data source in the chain that is currently playing. If you don't
+consistently read from the head data source this state will become inconsistent and things won't
+work correctly. When using a chain, this pointer needs to be reset if you need to play the
+chain again from the start:
+
+    ```c
+    ma_data_source_set_current(&headDataSource, &headDataSource);
+    ma_data_source_seek_to_pcm_frame(&headDataSource, 0);
+    ```
+
+The code above is setting the "current" data source in the chain to the head data source, thereby
+starting the chain from the start again. It is also seeking the head data source back to the start
+so that playback starts from the start as expected. You do not need to seek non-head items back to
+the start as miniaudio will do that for you internally.
+*/
+#include "../miniaudio.c"
+
+#include <stdio.h>
+
+/*
+For simplicity, this example requires the device to use floating point samples.
+*/
+#define SAMPLE_FORMAT   ma_format_f32
+#define CHANNEL_COUNT   2
+#define SAMPLE_RATE     48000
+
+ma_uint32   g_decoderCount;
+ma_decoder* g_pDecoders;
+
+static ma_data_source* next_callback_tail(ma_data_source* pDataSource)
+{
+    (void)pDataSource;  /* Unused. */
+
+    if (g_decoderCount > 0) {   /* <-- We check for this in main() so should never happen. */
+        return NULL;
+    }
+
+    /*
+    This will be fired when the last item in the chain has reached the end. In this example we want
+    to loop back to the start, so we need only return a pointer back to the head.
+    */
+    return &g_pDecoders[0];
+}
+
+static void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
+{
+    /*
+    We can just read from the first decoder and miniaudio will resolve the chain for us. Note that
+    if you want to loop the chain, like we're doing in this example, you need to set the `loop`
+    parameter to false, or else only the current data source will be looped.
+    */
+    ma_data_source_read_pcm_frames(&g_pDecoders[0], pOutput, frameCount, NULL);
+
+    /* Unused in this example. */
+    (void)pDevice;
+    (void)pInput;
+}
+
+int main(int argc, char** argv)
+{
+    ma_result result = MA_SUCCESS;
+    ma_uint32 iDecoder;
+    ma_decoder_config decoderConfig;
+    ma_device_config deviceConfig;
+    ma_device device;
+
+    if (argc < 2) {
+        printf("No input files.\n");
+        return -1;
+    }
+
+    g_decoderCount = argc-1;
+    g_pDecoders    = (ma_decoder*)malloc(sizeof(*g_pDecoders) * g_decoderCount);
+
+    /* In this example, all decoders need to have the same output format. */
+    decoderConfig = ma_decoder_config_init(SAMPLE_FORMAT, CHANNEL_COUNT, SAMPLE_RATE);
+    for (iDecoder = 0; iDecoder < g_decoderCount; ++iDecoder) {
+        result = ma_decoder_init_file(argv[1+iDecoder], &decoderConfig, &g_pDecoders[iDecoder]);
+        if (result != MA_SUCCESS) {
+            ma_uint32 iDecoder2;
+            for (iDecoder2 = 0; iDecoder2 < iDecoder; ++iDecoder2) {
+                ma_decoder_uninit(&g_pDecoders[iDecoder2]);
+            }
+            free(g_pDecoders);
+
+            printf("Failed to load %s.\n", argv[1+iDecoder]);
+            return -1;
+        }
+    }
+
+    /*
+    We're going to set up our decoders to run one after the other, but then have the last one loop back
+    to the first one. For demonstration purposes we're going to use the callback method for the last
+    data source.
+    */
+    for (iDecoder = 0; iDecoder < g_decoderCount-1; iDecoder += 1) {
+        ma_data_source_set_next(&g_pDecoders[iDecoder], &g_pDecoders[iDecoder+1]);
+    }
+
+    /*
+    For the last data source we'll loop back to the start, but for demonstration purposes we'll use a
+    callback to determine the next data source in the chain.
+    */
+    ma_data_source_set_next_callback(&g_pDecoders[g_decoderCount-1], next_callback_tail);
+
+
+    /*
+    The data source chain has been established so now we can get the device up and running so we
+    can listen to it.
+    */
+    deviceConfig = ma_device_config_init(ma_device_type_playback);
+    deviceConfig.playback.format   = SAMPLE_FORMAT;
+    deviceConfig.playback.channels = CHANNEL_COUNT;
+    deviceConfig.sampleRate        = SAMPLE_RATE;
+    deviceConfig.dataCallback      = data_callback;
+    deviceConfig.pUserData         = NULL;
+
+    result = ma_device_init(NULL, &deviceConfig, &device);
+    if (result != MA_SUCCESS) {
+        printf("Failed to open playback device.\n");
+        goto done_decoders;
+    }
+
+    result = ma_device_start(&device);
+    if (result != MA_SUCCESS) {
+        printf("Failed to start playback device.\n");
+        goto done;
+    }
+
+    printf("Press Enter to quit...");
+    getchar();
+
+done:
+    ma_device_uninit(&device);
+
+done_decoders:
+    for (iDecoder = 0; iDecoder < g_decoderCount; ++iDecoder) {
+        ma_decoder_uninit(&g_pDecoders[iDecoder]);
+    }
+    free(g_pDecoders);
+
+    return 0;
+}

thirdparty/miniaudio-0.11.24/examples/duplex_effect.c

@@ -0,0 +1,152 @@
+/*
+Demonstrates how to apply an effect to a duplex stream using the node graph system.
+
+This example applies a vocoder effect to the input stream before outputting it. A custom node
+called `ma_vocoder_node` is used to achieve the effect which can be found in the extras folder in
+the miniaudio repository. The vocoder node uses https://github.com/blastbay/voclib to achieve the
+effect.
+*/
+#include "../miniaudio.c"
+#include "../extras/nodes/ma_vocoder_node/ma_vocoder_node.c"
+
+#include <stdio.h>
+
+#define DEVICE_FORMAT      ma_format_f32    /* Must always be f32 for this example because the node graph system only works with this. */
+#define DEVICE_CHANNELS    1                /* For this example, always set to 1. */
+
+static ma_waveform         g_sourceData;    /* The underlying data source of the source node. */
+static ma_audio_buffer_ref g_exciteData;    /* The underlying data source of the excite node. */
+static ma_data_source_node g_sourceNode;    /* A data source node containing the source data we'll be sending through to the vocoder. This will be routed into the first bus of the vocoder node. */
+static ma_data_source_node g_exciteNode;    /* A data source node containing the excite data we'll be sending through to the vocoder. This will be routed into the second bus of the vocoder node. */
+static ma_vocoder_node     g_vocoderNode;   /* The vocoder node. */
+static ma_node_graph       g_nodeGraph;
+
+void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
+{
+    /*
+    This example assumes the playback and capture sides use the same format and channel count. The
+    format must be f32.
+    */
+    if (pDevice->capture.format != DEVICE_FORMAT || pDevice->playback.format != DEVICE_FORMAT || pDevice->capture.channels != pDevice->playback.channels) {
+        return;
+    }
+
+    /*
+    The node graph system is a pulling style of API. At the lowest level of the chain will be a 
+    node acting as a data source for the purpose of delivering the initial audio data. In our case,
+    the data source is our `pInput` buffer. We need to update the underlying data source so that it
+    read data from `pInput`.
+    */
+    ma_audio_buffer_ref_set_data(&g_exciteData, pInput, frameCount);
+
+    /* With the source buffer configured we can now read directly from the node graph. */
+    ma_node_graph_read_pcm_frames(&g_nodeGraph, pOutput, frameCount, NULL);
+}
+
+int main(int argc, char** argv)
+{
+    ma_result result;
+    ma_device_config deviceConfig;
+    ma_device device;
+    ma_node_graph_config nodeGraphConfig;
+    ma_vocoder_node_config vocoderNodeConfig;
+    ma_data_source_node_config sourceNodeConfig;
+    ma_data_source_node_config exciteNodeConfig;
+    ma_waveform_config waveformConfig;
+
+    deviceConfig = ma_device_config_init(ma_device_type_duplex);
+    deviceConfig.capture.pDeviceID  = NULL;
+    deviceConfig.capture.format     = DEVICE_FORMAT;
+    deviceConfig.capture.channels   = DEVICE_CHANNELS;
+    deviceConfig.capture.shareMode  = ma_share_mode_shared;
+    deviceConfig.playback.pDeviceID = NULL;
+    deviceConfig.playback.format    = DEVICE_FORMAT;
+    deviceConfig.playback.channels  = DEVICE_CHANNELS;
+    deviceConfig.dataCallback       = data_callback;
+    result = ma_device_init(NULL, &deviceConfig, &device);
+    if (result != MA_SUCCESS) {
+        return result;
+    }
+
+
+    /* Now we can setup our node graph. */
+    nodeGraphConfig = ma_node_graph_config_init(device.capture.channels);
+
+    result = ma_node_graph_init(&nodeGraphConfig, NULL, &g_nodeGraph);
+    if (result != MA_SUCCESS) {
+        printf("Failed to initialize node graph.");
+        goto done0;
+    }
+
+
+    /* Vocoder. Attached straight to the endpoint. */
+    vocoderNodeConfig = ma_vocoder_node_config_init(device.capture.channels, device.sampleRate);
+
+    result = ma_vocoder_node_init(&g_nodeGraph, &vocoderNodeConfig, NULL, &g_vocoderNode);
+    if (result != MA_SUCCESS) {
+        printf("Failed to initialize vocoder node.");
+        goto done1;
+    }
+
+    ma_node_attach_output_bus(&g_vocoderNode, 0, ma_node_graph_get_endpoint(&g_nodeGraph), 0);
+
+    /* Amplify the volume of the vocoder output because in my testing it is a bit quiet. */
+    ma_node_set_output_bus_volume(&g_vocoderNode, 0, 4);
+
+
+    /* Source/carrier. Attached to input bus 0 of the vocoder node. */
+    waveformConfig = ma_waveform_config_init(device.capture.format, device.capture.channels, device.sampleRate, ma_waveform_type_sawtooth, 1.0, 50);
+
+    result = ma_waveform_init(&waveformConfig, &g_sourceData);
+    if (result != MA_SUCCESS) {
+        printf("Failed to initialize waveform for excite node.");
+        goto done3;
+    }
+
+    sourceNodeConfig = ma_data_source_node_config_init(&g_sourceData);
+
+    result = ma_data_source_node_init(&g_nodeGraph, &sourceNodeConfig, NULL, &g_sourceNode);
+    if (result != MA_SUCCESS) {
+        printf("Failed to initialize excite node.");
+        goto done3;
+    }
+
+    ma_node_attach_output_bus(&g_sourceNode, 0, &g_vocoderNode, 0);
+
+
+    /* Excite/modulator. Attached to input bus 1 of the vocoder node. */
+    result = ma_audio_buffer_ref_init(device.capture.format, device.capture.channels, NULL, 0, &g_exciteData);
+    if (result != MA_SUCCESS) {
+        printf("Failed to initialize audio buffer for source.");
+        goto done2;
+    }
+
+    exciteNodeConfig = ma_data_source_node_config_init(&g_exciteData);
+
+    result = ma_data_source_node_init(&g_nodeGraph, &exciteNodeConfig, NULL, &g_exciteNode);
+    if (result != MA_SUCCESS) {
+        printf("Failed to initialize source node.");
+        goto done2;
+    }
+
+    ma_node_attach_output_bus(&g_exciteNode, 0, &g_vocoderNode, 1);
+
+
+    ma_device_start(&device);
+
+    printf("Press Enter to quit...\n");
+    getchar();
+
+    /* It's important that we stop the device first or else we'll uninitialize the graph from under the device. */
+    ma_device_stop(&device);
+
+/*done4:*/ ma_data_source_node_uninit(&g_exciteNode, NULL);
+done3: ma_data_source_node_uninit(&g_sourceNode, NULL);
+done2: ma_vocoder_node_uninit(&g_vocoderNode, NULL);
+done1: ma_node_graph_uninit(&g_nodeGraph, NULL);
+done0: ma_device_uninit(&device);
+
+    (void)argc;
+    (void)argv;
+    return 0;
+}

thirdparty/miniaudio-0.11.24/examples/engine_advanced.c

@@ -0,0 +1,250 @@
+/*
+This example demonstrates some of the advanced features of the high level engine API.
+
+The following features are demonstrated:
+
+  * Initialization of the engine from a pre-initialized device.
+  * Self-managed resource managers.
+  * Multiple engines with a shared resource manager.
+  * Creation and management of `ma_sound` objects.
+
+This example will play the sound that's passed in on the command line.
+
+Using a shared resource manager, as we do in this example, is useful for when you want to user
+multiple engines so that you can output to multiple playback devices simultaneoulys. An example
+might be a local co-op multiplayer game where each player has their own headphones.
+*/
+#include "../miniaudio.c"
+
+#define MAX_DEVICES 2
+#define MAX_SOUNDS  32
+
+void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
+{
+    (void)pInput;
+
+    /*
+    Since we're managing the underlying device ourselves, we need to read from the engine directly.
+    To do this we need access to the `ma_engine` object which we passed in to the user data. One
+    advantage of this is that you could do your own audio processing in addition to the engine's
+    standard processing.
+    */
+    ma_engine_read_pcm_frames((ma_engine*)pDevice->pUserData, pOutput, frameCount, NULL);
+}
+
+int main(int argc, char** argv)
+{
+    ma_result result;
+    ma_context context;
+    ma_resource_manager_config resourceManagerConfig;
+    ma_resource_manager resourceManager;
+    ma_engine engines[MAX_DEVICES];
+    ma_device devices[MAX_DEVICES];
+    ma_uint32 engineCount = 0;
+    ma_uint32 iEngine;
+    ma_device_info* pPlaybackDeviceInfos;
+    ma_uint32 playbackDeviceCount;
+    ma_uint32 iAvailableDevice;
+    ma_uint32 iChosenDevice;
+    ma_sound sounds[MAX_SOUNDS];
+    ma_uint32 soundCount;
+    ma_uint32 iSound;
+
+    if (argc < 2) {
+        printf("No input file.");
+        return -1;
+    }
+
+
+    /*
+    We are going to be initializing multiple engines. In order to save on memory usage we can use a self managed
+    resource manager so we can share a single resource manager across multiple engines.
+    */
+    resourceManagerConfig = ma_resource_manager_config_init();
+    resourceManagerConfig.decodedFormat     = ma_format_f32;    /* ma_format_f32 should almost always be used as that's what the engine (and most everything else) uses for mixing. */
+    resourceManagerConfig.decodedChannels   = 0;                /* Setting the channel count to 0 will cause sounds to use their native channel count. */
+    resourceManagerConfig.decodedSampleRate = 48000;            /* Using a consistent sample rate is useful for avoiding expensive resampling in the audio thread. This will result in resampling being performed by the loading thread(s). */
+
+    result = ma_resource_manager_init(&resourceManagerConfig, &resourceManager);
+    if (result != MA_SUCCESS) {
+        printf("Failed to initialize resource manager.");
+        return -1;
+    }
+
+
+    /* We're going to want a context so we can enumerate our playback devices. */
+    result = ma_context_init(NULL, 0, NULL, &context);
+    if (result != MA_SUCCESS) {
+        printf("Failed to initialize context.");
+        return -1;
+    }
+
+    /*
+    Now that we have a context we will want to enumerate over each device so we can display them to the user and give
+    them a chance to select the output devices they want to use.
+    */
+    result = ma_context_get_devices(&context, &pPlaybackDeviceInfos, &playbackDeviceCount, NULL, NULL);
+    if (result != MA_SUCCESS) {
+        printf("Failed to enumerate playback devices.");
+        ma_context_uninit(&context);
+        return -1;
+    }
+
+
+    /* We have our devices, so now we want to get the user to select the devices they want to output to. */
+    engineCount = 0;
+
+    for (iChosenDevice = 0; iChosenDevice < MAX_DEVICES; iChosenDevice += 1) {
+        int c = 0;
+        for (;;) {
+            printf("Select playback device %d ([%d - %d], Q to quit):\n", iChosenDevice+1, 0, ma_min((int)playbackDeviceCount, 9));
+
+            for (iAvailableDevice = 0; iAvailableDevice < playbackDeviceCount; iAvailableDevice += 1) {
+                printf("    %d: %s\n", iAvailableDevice, pPlaybackDeviceInfos[iAvailableDevice].name);
+            }
+
+            for (;;) {
+                c = getchar();
+                if (c != '\n') {
+                    break;
+                }
+            }
+        
+            if (c == 'q' || c == 'Q') {
+                return 0;   /* User aborted. */
+            }
+
+            if (c >= '0' && c <= '9') {
+                c -= '0';
+
+                if (c < (int)playbackDeviceCount) {
+                    ma_device_config deviceConfig;
+                    ma_engine_config engineConfig;
+
+                    /*
+                    Create the device first before the engine. We'll specify the device in the engine's config. This is optional. When a device is
+                    not pre-initialized the engine will create one for you internally. The device does not need to be started here - the engine will
+                    do that for us in `ma_engine_start()`. The device's format is derived from the resource manager, but can be whatever you want.
+                    It's useful to keep the format consistent with the resource manager to avoid data conversions costs in the audio callback. In
+                    this example we're using the resource manager's sample format and sample rate, but leaving the channel count set to the device's
+                    native channels. You can use whatever format/channels/rate you like.
+                    */
+                    deviceConfig = ma_device_config_init(ma_device_type_playback);
+                    deviceConfig.playback.pDeviceID = &pPlaybackDeviceInfos[c].id;
+                    deviceConfig.playback.format    = resourceManager.config.decodedFormat;
+                    deviceConfig.playback.channels  = 0;
+                    deviceConfig.sampleRate         = resourceManager.config.decodedSampleRate;
+                    deviceConfig.dataCallback       = data_callback;
+                    deviceConfig.pUserData          = &engines[engineCount];
+
+                    result = ma_device_init(&context, &deviceConfig, &devices[engineCount]);
+                    if (result != MA_SUCCESS) {
+                        printf("Failed to initialize device for %s.\n", pPlaybackDeviceInfos[c].name);
+                        return -1;
+                    }
+
+                    /* Now that we have the device we can initialize the engine. The device is passed into the engine's config. */
+                    engineConfig = ma_engine_config_init();
+                    engineConfig.pDevice          = &devices[engineCount];
+                    engineConfig.pResourceManager = &resourceManager;
+                    engineConfig.noAutoStart      = MA_TRUE;    /* Don't start the engine by default - we'll do that manually below. */
+
+                    result = ma_engine_init(&engineConfig, &engines[engineCount]);
+                    if (result != MA_SUCCESS) {
+                        printf("Failed to initialize engine for %s.\n", pPlaybackDeviceInfos[c].name);
+                        ma_device_uninit(&devices[engineCount]);
+                        return -1;
+                    }
+                    
+                    engineCount += 1;
+                    break;
+                } else {
+                    printf("Invalid device number.\n");
+                }
+            } else {
+                printf("Invalid device number.\n");
+            }
+        }
+
+        printf("Device %d: %s\n", iChosenDevice+1, pPlaybackDeviceInfos[c].name);
+    }
+
+
+    /* We should now have our engine's initialized. We can now start them. */
+    for (iEngine = 0; iEngine < engineCount; iEngine += 1) {
+        result = ma_engine_start(&engines[iEngine]);
+        if (result != MA_SUCCESS) {
+            printf("WARNING: Failed to start engine %d.\n", iEngine);
+        }
+    }
+
+
+    /*
+    At this point our engine's are running and outputting nothing but silence. To get them playing something we'll need
+    some sounds. In this example we're just using one sound per engine, but you can create as many as you like. Since
+    we're using a shared resource manager, the sound data will only be loaded once. This is how you would implement
+    multiple listeners.
+    */
+    soundCount = 0;
+    for (iEngine = 0; iEngine < engineCount; iEngine += 1) {
+        /* Just one sound per engine in this example. We're going to be loading this asynchronously. */
+        result = ma_sound_init_from_file(&engines[iEngine], argv[1], MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_DECODE | MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC | MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM, NULL, NULL, &sounds[iEngine]);
+        if (result != MA_SUCCESS) {
+            printf("WARNING: Failed to load sound \"%s\"", argv[1]);
+            break;
+        }
+
+        /*
+        The sound can be started as soon as ma_sound_init_from_file() returns, even for sounds that are initialized
+        with MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC. The sound will start playing while it's being loaded. Note that if the
+        asynchronous loading process cannot keep up with the rate at which you try reading you'll end up glitching.
+        If this is an issue, you need to not load sounds asynchronously.
+        */
+        result = ma_sound_start(&sounds[iEngine]);
+        if (result != MA_SUCCESS) {
+            printf("WARNING: Failed to start sound.");
+        }
+
+        soundCount += 1;
+    }
+
+
+    printf("Press Enter to quit...");
+    getchar();
+    
+    for (;;) {
+        int c = getchar();
+        if (c == '\n') {
+            break;
+        }
+    }
+
+    /* Teardown. */
+
+    /* The application owns the `ma_sound` object which means you're responsible for uninitializing them. */
+    for (iSound = 0; iSound < soundCount; iSound += 1) {
+        ma_sound_uninit(&sounds[iSound]);
+    }
+
+    /* We can now uninitialize each engine. */
+    for (iEngine = 0; iEngine < engineCount; iEngine += 1) {
+        ma_engine_uninit(&engines[iEngine]);
+
+        /*
+        The engine has been uninitialized so now lets uninitialize the device. Do this first to ensure we don't
+        uninitialize the resource manager from under the device while the data callback is running.
+        */
+        ma_device_uninit(&devices[iEngine]);
+    }
+
+    /* The context can only be uninitialized after the devices. */
+    ma_context_uninit(&context);
+
+    /*
+    Do the resource manager last. This way we can guarantee the data callbacks of each device aren't trying to access
+    and data managed by the resource manager.
+    */
+    ma_resource_manager_uninit(&resourceManager);
+
+    return 0;
+}

thirdparty/miniaudio-0.11.24/examples/engine_effects.c

@@ -0,0 +1,103 @@
+/*
+Demonstrates how to apply an effect to sounds using the high level engine API.
+
+This example will load a file from the command line and apply an echo/delay effect to it. It will
+show you how to manage `ma_sound` objects and how to insert an effect into the graph.
+
+The `ma_engine` object is a node graph and is compatible with the `ma_node_graph` API. The
+`ma_sound` object is a node within the node and is compatible with the `ma_node` API. This means
+that applying an effect is as simple as inserting an effect node into the graph and plugging in the
+sound's output into the effect's input. See the Node Graph example for how to use the node graph.
+
+This example is playing only a single sound at a time which means only a single `ma_sound` object
+it being used. If you want to play multiple sounds at the same time, even if they're for the same
+sound file, you need multiple `ma_sound` objects.
+*/
+#include "../miniaudio.c"
+
+#define DELAY_IN_SECONDS    0.2f
+#define DECAY               0.25f   /* Volume falloff for each echo. */
+
+static ma_engine g_engine;
+static ma_sound g_sound;            /* This example will play only a single sound at once, so we only need one `ma_sound` object. */
+static ma_delay_node g_delayNode;   /* The echo effect is achieved using a delay node. */
+
+int main(int argc, char** argv)
+{
+    ma_result result;
+
+    if (argc < 2) {
+        printf("No input file.");
+        return -1;
+    }
+
+    /* The engine needs to be initialized first. */
+    result = ma_engine_init(NULL, &g_engine);
+    if (result != MA_SUCCESS) {
+        printf("Failed to initialize audio engine.");
+        return -1;
+    }
+
+
+    /*
+    We'll build our graph starting from the end so initialize the delay node now. The output of
+    this node will be connected straight to the output. You could also attach it to a sound group
+    or any other node that accepts an input.
+
+    Creating a node requires a pointer to the node graph that owns it. The engine itself is a node
+    graph. In the code below we can get a pointer to the node graph with `ma_engine_get_node_graph()`
+    or we could simple cast the engine to a ma_node_graph* like so:
+    
+        (ma_node_graph*)&g_engine
+
+    The endpoint of the graph can be retrieved with `ma_engine_get_endpoint()`.
+    */
+    {
+        ma_delay_node_config delayNodeConfig;
+        ma_uint32 channels;
+        ma_uint32 sampleRate;
+
+        channels   = ma_engine_get_channels(&g_engine);
+        sampleRate = ma_engine_get_sample_rate(&g_engine);
+
+        delayNodeConfig = ma_delay_node_config_init(channels, sampleRate, (ma_uint32)(sampleRate * DELAY_IN_SECONDS), DECAY);
+
+        result = ma_delay_node_init(ma_engine_get_node_graph(&g_engine), &delayNodeConfig, NULL, &g_delayNode);
+        if (result != MA_SUCCESS) {
+            printf("Failed to initialize delay node.");
+            return -1;
+        }
+
+        /* Connect the output of the delay node to the input of the endpoint. */
+        ma_node_attach_output_bus(&g_delayNode, 0, ma_engine_get_endpoint(&g_engine), 0);
+    }
+
+
+    /* Now we can load the sound and connect it to the delay node. */
+    {
+        result = ma_sound_init_from_file(&g_engine, argv[1], 0, NULL, NULL, &g_sound);
+        if (result != MA_SUCCESS) {
+            printf("Failed to initialize sound \"%s\".", argv[1]);
+            return -1;
+        }
+
+        /* Connect the output of the sound to the input of the effect. */
+        ma_node_attach_output_bus(&g_sound, 0, &g_delayNode, 0);
+
+        /*
+        Start the sound after it's applied to the sound. Otherwise there could be a scenario where
+        the very first part of it is read before the attachment to the effect is made.
+        */
+        ma_sound_start(&g_sound);
+    }
+
+
+    printf("Press Enter to quit...");
+    getchar();
+
+    ma_sound_uninit(&g_sound);
+    ma_delay_node_uninit(&g_delayNode, NULL);
+    ma_engine_uninit(&g_engine);
+
+    return 0;
+}

thirdparty/miniaudio-0.11.24/examples/engine_hello_world.c

@@ -0,0 +1,34 @@
+/*
+This example demonstrates how to initialize an audio engine and play a sound.
+
+This will play the sound specified on the command line.
+*/
+#include "../miniaudio.c"
+
+#include <stdio.h>
+
+int main(int argc, char** argv)
+{
+    ma_result result;
+    ma_engine engine;
+
+    if (argc < 2) {
+        printf("No input file.");
+        return -1;
+    }
+
+    result = ma_engine_init(NULL, &engine);
+    if (result != MA_SUCCESS) {
+        printf("Failed to initialize audio engine.");
+        return -1;
+    }
+
+    ma_engine_play_sound(&engine, argv[1], NULL);
+
+    printf("Press Enter to quit...");
+    getchar();
+
+    ma_engine_uninit(&engine);
+
+    return 0;
+}

thirdparty/miniaudio-0.11.24/examples/engine_sdl.c

@@ -0,0 +1,137 @@
+/*
+Shows how to use the high level engine API with SDL.
+
+By default, miniaudio's engine API will initialize a device internally for audio output. You can
+instead use the engine independently of a device. To show this off, this example will use SDL for
+audio output instead of miniaudio.
+
+This example will load the sound specified on the command line and rotate it around the listener's
+head.
+*/
+#define MA_NO_DEVICE_IO /* <-- Disables the `ma_device` API. We don't need that in this example since SDL will be doing that part for us. */
+#include "../miniaudio.c"
+
+#define SDL_MAIN_HANDLED
+#include <SDL2/SDL.h>    /* Change this to your include location. Might be <SDL.h>. */
+
+#define CHANNELS    2               /* Must be stereo for this example. */
+#define SAMPLE_RATE 48000
+
+static ma_engine g_engine;
+static ma_sound g_sound;            /* This example will play only a single sound at once, so we only need one `ma_sound` object. */
+
+void data_callback(void* pUserData, ma_uint8* pBuffer, int bufferSizeInBytes)
+{
+    ma_uint32 bufferSizeInFrames;
+
+    (void)pUserData;
+
+    /* Reading is just a matter of reading straight from the engine. */
+    bufferSizeInFrames = (ma_uint32)bufferSizeInBytes / ma_get_bytes_per_frame(ma_format_f32, ma_engine_get_channels(&g_engine));
+    ma_engine_read_pcm_frames(&g_engine, pBuffer, bufferSizeInFrames, NULL);
+}
+
+int main(int argc, char** argv)
+{
+    ma_result result;
+    ma_engine_config engineConfig;
+    SDL_AudioSpec desiredSpec;
+    SDL_AudioSpec obtainedSpec;
+    SDL_AudioDeviceID deviceID;
+
+    if (argc < 2) {
+        printf("No input file.");
+        return -1;
+    }
+
+    /*
+    We'll initialize the engine first for the purpose of the example, but since the engine and SDL
+    are independent of each other you can initialize them in any order. You need only make sure the
+    channel count and sample rates are consistent between the two.
+
+    When initializing the engine it's important to make sure we don't initialize a device
+    internally because we want SDL to be dealing with that for us instead.
+    */
+    engineConfig = ma_engine_config_init();
+    engineConfig.noDevice   = MA_TRUE;      /* <-- Make sure this is set so that no device is created (we'll deal with that ourselves). */
+    engineConfig.channels   = CHANNELS;
+    engineConfig.sampleRate = SAMPLE_RATE;
+
+    result = ma_engine_init(&engineConfig, &g_engine);
+    if (result != MA_SUCCESS) {
+        printf("Failed to initialize audio engine.");
+        return -1;
+    }
+
+    /* Now load our sound. */
+    result = ma_sound_init_from_file(&g_engine, argv[1], 0, NULL, NULL, &g_sound);
+    if (result != MA_SUCCESS) {
+        printf("Failed to initialize sound.");
+        return -1;
+    }
+
+    /* Loop the sound so we can continuously hear it. */
+    ma_sound_set_looping(&g_sound, MA_TRUE);
+
+    /*
+    The sound will not be started by default, so start it now. We won't hear anything until the SDL
+    audio device has been opened and started.
+    */
+    ma_sound_start(&g_sound);
+
+
+    /*
+    Now that we have the engine and sound we can initialize SDL. This could have also been done
+    first before the engine and sound.
+    */
+    if (SDL_InitSubSystem(SDL_INIT_AUDIO) != 0) {
+        printf("Failed to initialize SDL sub-system.");
+        return -1;
+    }
+
+    MA_ZERO_OBJECT(&desiredSpec);
+    desiredSpec.freq     = ma_engine_get_sample_rate(&g_engine);
+    desiredSpec.format   = AUDIO_F32;
+    desiredSpec.channels = ma_engine_get_channels(&g_engine);
+    desiredSpec.samples  = 512;
+    desiredSpec.callback = data_callback;
+    desiredSpec.userdata = NULL;
+
+    deviceID = SDL_OpenAudioDevice(NULL, 0, &desiredSpec, &obtainedSpec, SDL_AUDIO_ALLOW_ANY_CHANGE);
+    if (deviceID == 0) {
+        printf("Failed to open SDL audio device.");
+        return -1;
+    }
+
+    /* Start playback. */
+    SDL_PauseAudioDevice(deviceID, 0);
+
+#if 1
+    {
+        /* We'll move the sound around the listener which we'll leave at the origin. */
+        float stepAngle = 0.002f;
+        float angle = 0;
+        float distance = 2;
+
+        for (;;) {
+            double x = ma_cosd(angle) - ma_sind(angle);
+            double y = ma_sind(angle) + ma_cosd(angle);
+
+            ma_sound_set_position(&g_sound, (float)x * distance, 0, (float)y * distance);
+
+            angle += stepAngle;
+            ma_sleep(1);
+        }
+    }
+#else
+    printf("Press Enter to quit...");
+    getchar();
+#endif
+
+    ma_sound_uninit(&g_sound);
+    ma_engine_uninit(&g_engine);
+    SDL_CloseAudioDevice(deviceID);
+    SDL_QuitSubSystem(SDL_INIT_AUDIO);
+
+    return 0;
+}

thirdparty/miniaudio-0.11.24/examples/engine_steamaudio.c

@@ -0,0 +1,448 @@
+/*
+Demonstrates integration of Steam Audio with miniaudio's engine API.
+
+In this example a HRTF effect from Steam Audio will be applied. To do this a custom node will be
+implemented which uses Steam Audio's IPLBinauralEffect and IPLHRTF objects.
+
+By implementing this as a node, it can be plugged into any position within the graph. The output
+channel count of this node is always stereo.
+
+Steam Audio requires fixed sized processing, the size of which must be specified at initialization
+time of the IPLBinauralEffect and IPLHRTF objects. To ensure miniaudio and Steam Audio are
+consistent, you must set the period size in the engine config to be consistent with the frame size
+you specify in your IPLAudioSettings object. If for some reason you want the period size of the
+engine to be different to that of your Steam Audio configuration, you'll need to implement a sort
+of buffering solution to your node.
+*/
+#include "../miniaudio.c"
+
+#include <stdint.h> /* Required for uint32_t which is used by STEAMAUDIO_VERSION, and a random use of uint8_t. If there's a Steam Audio maintainer reading this, that needs to be fixed to use IPLuint32 and IPLuint8. */
+
+/* Need to silence some warnings from the Steam Audio headers. */
+#if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
+    #pragma GCC diagnostic push
+    #pragma GCC diagnostic ignored "-Wlong-long"
+    #pragma GCC diagnostic ignored "-Wpedantic"
+#endif
+#include <phonon.h> /* Steam Audio */
+#if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
+    #pragma GCC diagnostic pop
+#endif
+
+#define FORMAT      ma_format_f32   /* Must be floating point. */
+#define CHANNELS    2               /* Must be stereo for this example. */
+#define SAMPLE_RATE 48000
+
+
+static ma_result ma_result_from_IPLerror(IPLerror error)
+{
+    switch (error)
+    {
+        case IPL_STATUS_SUCCESS:     return MA_SUCCESS;
+        case IPL_STATUS_OUTOFMEMORY: return MA_OUT_OF_MEMORY;
+        case IPL_STATUS_INITIALIZATION:
+        case IPL_STATUS_FAILURE:
+        default: return MA_ERROR;
+    }
+}
+
+
+typedef struct
+{
+    ma_node_config nodeConfig;
+    ma_uint32 channelsIn;
+    IPLAudioSettings iplAudioSettings;
+    IPLContext iplContext;
+    IPLHRTF iplHRTF;   /* There is one HRTF object to many binaural effect objects. */
+} ma_steamaudio_binaural_node_config;
+
+MA_API ma_steamaudio_binaural_node_config ma_steamaudio_binaural_node_config_init(ma_uint32 channelsIn, IPLAudioSettings iplAudioSettings, IPLContext iplContext, IPLHRTF iplHRTF);
+
+
+typedef struct
+{
+    ma_node_base baseNode;
+    IPLAudioSettings iplAudioSettings;
+    IPLContext iplContext;
+    IPLHRTF iplHRTF;
+    IPLBinauralEffect iplEffect;
+    ma_vec3f direction;
+    float* ppBuffersIn[2];      /* Each buffer is an offset of _pHeap. */
+    float* ppBuffersOut[2];     /* Each buffer is an offset of _pHeap. */
+    void* _pHeap;
+} ma_steamaudio_binaural_node;
+
+MA_API ma_result ma_steamaudio_binaural_node_init(ma_node_graph* pNodeGraph, const ma_steamaudio_binaural_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_steamaudio_binaural_node* pBinauralNode);
+MA_API void ma_steamaudio_binaural_node_uninit(ma_steamaudio_binaural_node* pBinauralNode, const ma_allocation_callbacks* pAllocationCallbacks);
+MA_API ma_result ma_steamaudio_binaural_node_set_direction(ma_steamaudio_binaural_node* pBinauralNode, float x, float y, float z);
+
+
+MA_API ma_steamaudio_binaural_node_config ma_steamaudio_binaural_node_config_init(ma_uint32 channelsIn, IPLAudioSettings iplAudioSettings, IPLContext iplContext, IPLHRTF iplHRTF)
+{
+    ma_steamaudio_binaural_node_config config;
+
+    MA_ZERO_OBJECT(&config);
+    config.nodeConfig       = ma_node_config_init();
+    config.channelsIn       = channelsIn;
+    config.iplAudioSettings = iplAudioSettings;
+    config.iplContext       = iplContext;
+    config.iplHRTF          = iplHRTF;
+
+    return config;
+}
+
+
+static void ma_steamaudio_binaural_node_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
+{
+    ma_steamaudio_binaural_node* pBinauralNode = (ma_steamaudio_binaural_node*)pNode;
+    IPLBinauralEffectParams binauralParams;
+    IPLAudioBuffer inputBufferDesc;
+    IPLAudioBuffer outputBufferDesc;
+    ma_uint32 totalFramesToProcess = *pFrameCountOut;
+    ma_uint32 totalFramesProcessed = 0;
+
+    MA_ZERO_OBJECT(&binauralParams);
+    binauralParams.direction.x   = pBinauralNode->direction.x;
+    binauralParams.direction.y   = pBinauralNode->direction.y;
+    binauralParams.direction.z   = pBinauralNode->direction.z;
+    binauralParams.interpolation = IPL_HRTFINTERPOLATION_NEAREST;
+    binauralParams.spatialBlend  = 1.0f;
+    binauralParams.hrtf          = pBinauralNode->iplHRTF;
+
+    inputBufferDesc.numChannels = (IPLint32)ma_node_get_input_channels(pNode, 0);
+
+    /* We'll run this in a loop just in case our deinterleaved buffers are too small. */
+    outputBufferDesc.numSamples  = pBinauralNode->iplAudioSettings.frameSize;
+    outputBufferDesc.numChannels = 2;
+    outputBufferDesc.data        = pBinauralNode->ppBuffersOut;
+
+    while (totalFramesProcessed < totalFramesToProcess) {
+        ma_uint32 framesToProcessThisIteration = totalFramesToProcess - totalFramesProcessed;
+        if (framesToProcessThisIteration > (ma_uint32)pBinauralNode->iplAudioSettings.frameSize) {
+            framesToProcessThisIteration = (ma_uint32)pBinauralNode->iplAudioSettings.frameSize;
+        }
+
+        if (inputBufferDesc.numChannels == 1) {
+            /* Fast path. No need for deinterleaving since it's a mono stream. */
+            pBinauralNode->ppBuffersIn[0] = (float*)ma_offset_pcm_frames_const_ptr_f32(ppFramesIn[0], totalFramesProcessed, 1);
+        } else {
+            /* Slow path. Need to deinterleave the input data. */
+            ma_deinterleave_pcm_frames(ma_format_f32, inputBufferDesc.numChannels, framesToProcessThisIteration, ma_offset_pcm_frames_const_ptr_f32(ppFramesIn[0], totalFramesProcessed, inputBufferDesc.numChannels), (void**)&pBinauralNode->ppBuffersIn[0]);
+        }
+
+        inputBufferDesc.data       = pBinauralNode->ppBuffersIn;
+        inputBufferDesc.numSamples = (IPLint32)framesToProcessThisIteration;
+
+        /* Apply the effect. */
+        iplBinauralEffectApply(pBinauralNode->iplEffect, &binauralParams, &inputBufferDesc, &outputBufferDesc);
+
+        /* Interleave straight into the output buffer. */
+        ma_interleave_pcm_frames(ma_format_f32, 2, framesToProcessThisIteration, (const void**)&pBinauralNode->ppBuffersOut[0], ma_offset_pcm_frames_ptr_f32(ppFramesOut[0], totalFramesProcessed, 2));
+
+        /* Advance. */
+        totalFramesProcessed += framesToProcessThisIteration;
+    }
+
+    (void)pFrameCountIn;    /* Unused. */
+}
+
+static ma_node_vtable g_ma_steamaudio_binaural_node_vtable =
+{
+    ma_steamaudio_binaural_node_process_pcm_frames,
+    NULL,
+    1,  /* 1 input channel. */
+    1,  /* 1 output channel. */
+    0
+};
+
+MA_API ma_result ma_steamaudio_binaural_node_init(ma_node_graph* pNodeGraph, const ma_steamaudio_binaural_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_steamaudio_binaural_node* pBinauralNode)
+{
+    ma_result result;
+    ma_node_config baseConfig;
+    ma_uint32 channelsIn;
+    ma_uint32 channelsOut;
+    IPLBinauralEffectSettings iplBinauralEffectSettings;
+    size_t heapSizeInBytes;
+
+    if (pBinauralNode == NULL) {
+        return MA_INVALID_ARGS;
+    }
+
+    MA_ZERO_OBJECT(pBinauralNode);
+
+    if (pConfig == NULL || pConfig->iplAudioSettings.frameSize == 0 || pConfig->iplContext == NULL || pConfig->iplHRTF == NULL) {
+        return MA_INVALID_ARGS;
+    }
+
+    /* Steam Audio only supports mono and stereo input. */
+    if (pConfig->channelsIn < 1 || pConfig->channelsIn > 2) {
+        return MA_INVALID_ARGS;
+    }
+
+    channelsIn  = pConfig->channelsIn;
+    channelsOut = 2;    /* Always stereo output. */
+
+    baseConfig = ma_node_config_init();
+    baseConfig.vtable          = &g_ma_steamaudio_binaural_node_vtable;
+    baseConfig.pInputChannels  = &channelsIn;
+    baseConfig.pOutputChannels = &channelsOut;
+    result = ma_node_init(pNodeGraph, &baseConfig, pAllocationCallbacks, &pBinauralNode->baseNode);
+    if (result != MA_SUCCESS) {
+        return result;
+    }
+
+    pBinauralNode->iplAudioSettings = pConfig->iplAudioSettings;
+    pBinauralNode->iplContext       = pConfig->iplContext;
+    pBinauralNode->iplHRTF          = pConfig->iplHRTF;
+
+    MA_ZERO_OBJECT(&iplBinauralEffectSettings);
+    iplBinauralEffectSettings.hrtf = pBinauralNode->iplHRTF;
+
+    result = ma_result_from_IPLerror(iplBinauralEffectCreate(pBinauralNode->iplContext, &pBinauralNode->iplAudioSettings, &iplBinauralEffectSettings, &pBinauralNode->iplEffect));
+    if (result != MA_SUCCESS) {
+        ma_node_uninit(&pBinauralNode->baseNode, pAllocationCallbacks);
+        return result;
+    }
+
+    heapSizeInBytes = 0;
+
+    /*
+    Unfortunately Steam Audio uses deinterleaved buffers for everything so we'll need to use some
+    intermediary buffers. We'll allocate one big buffer on the heap and then use offsets. We'll
+    use the frame size from the IPLAudioSettings structure as a basis for the size of the buffer.
+    */
+    heapSizeInBytes += sizeof(float) * channelsOut * pBinauralNode->iplAudioSettings.frameSize; /* Output buffer. */
+    heapSizeInBytes += sizeof(float) * channelsIn  * pBinauralNode->iplAudioSettings.frameSize; /* Input buffer. */
+
+    pBinauralNode->_pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
+    if (pBinauralNode->_pHeap == NULL) {
+        iplBinauralEffectRelease(&pBinauralNode->iplEffect);
+        ma_node_uninit(&pBinauralNode->baseNode, pAllocationCallbacks);
+        return MA_OUT_OF_MEMORY;
+    }
+
+    pBinauralNode->ppBuffersOut[0] = (float*)pBinauralNode->_pHeap;
+    pBinauralNode->ppBuffersOut[1] = (float*)ma_offset_ptr(pBinauralNode->_pHeap, sizeof(float) * pBinauralNode->iplAudioSettings.frameSize);
+
+    {
+        ma_uint32 iChannelIn;
+        for (iChannelIn = 0; iChannelIn < channelsIn; iChannelIn += 1) {
+            pBinauralNode->ppBuffersIn[iChannelIn] = (float*)ma_offset_ptr(pBinauralNode->_pHeap, sizeof(float) * pBinauralNode->iplAudioSettings.frameSize * (channelsOut + iChannelIn));
+        }
+    }
+
+    return MA_SUCCESS;
+}
+
+MA_API void ma_steamaudio_binaural_node_uninit(ma_steamaudio_binaural_node* pBinauralNode, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+    if (pBinauralNode == NULL) {
+        return;
+    }
+
+    /* The base node is always uninitialized first. */
+    ma_node_uninit(&pBinauralNode->baseNode, pAllocationCallbacks);
+
+    /*
+    The Steam Audio objects are deleted after the base node. This ensures the base node is removed from the graph
+    first to ensure these objects aren't getting used by the audio thread.
+    */
+    iplBinauralEffectRelease(&pBinauralNode->iplEffect);
+    ma_free(pBinauralNode->_pHeap, pAllocationCallbacks);
+}
+
+MA_API ma_result ma_steamaudio_binaural_node_set_direction(ma_steamaudio_binaural_node* pBinauralNode, float x, float y, float z)
+{
+    if (pBinauralNode == NULL) {
+        return MA_INVALID_ARGS;
+    }
+
+    pBinauralNode->direction.x = x;
+    pBinauralNode->direction.y = y;
+    pBinauralNode->direction.z = z;
+
+    return MA_SUCCESS;
+}
+
+
+
+
+static ma_engine g_engine;
+static ma_sound g_sound;            /* This example will play only a single sound at once, so we only need one `ma_sound` object. */
+static ma_steamaudio_binaural_node g_binauralNode;   /* The echo effect is achieved using a delay node. */
+
+int main(int argc, char** argv)
+{
+    ma_result result;
+    ma_engine_config engineConfig;
+    IPLAudioSettings iplAudioSettings;
+    IPLContextSettings iplContextSettings;
+    IPLContext iplContext;
+    IPLHRTFSettings iplHRTFSettings;
+    IPLHRTF iplHRTF;
+
+    if (argc < 2) {
+        printf("No input file.");
+        return -1;
+    }
+
+    /* The engine needs to be initialized first. */
+    engineConfig = ma_engine_config_init();
+    engineConfig.channels   = CHANNELS;
+    engineConfig.sampleRate = SAMPLE_RATE;
+
+    /*
+    Steam Audio requires processing in fixed sized chunks. Setting the period size in the engine config will
+    ensure our updates happen in predicably sized chunks as required by Steam Audio.
+
+    Note that the configuration of Steam Audio below (IPLAudioSettings) will use this variable to specify the
+    update size to ensure it remains consistent.
+    */
+    engineConfig.periodSizeInFrames = 256;
+
+    result = ma_engine_init(&engineConfig, &g_engine);
+    if (result != MA_SUCCESS) {
+        printf("Failed to initialize audio engine.");
+        return -1;
+    }
+
+    /*
+    Now that we have the engine we can initialize the Steam Audio objects.
+    */
+    MA_ZERO_OBJECT(&iplAudioSettings);
+    iplAudioSettings.samplingRate = ma_engine_get_sample_rate(&g_engine);
+
+    /*
+    If there's any Steam Audio developers reading this, why is the frame size needed? This needs to
+    be documented. If this is for some kind of buffer management with FFT or something, then this
+    need not be exposed to the public API. There should be no need for the public API to require a
+    fixed sized update.
+
+    It's important that this be set to the periodSizeInFrames specified in the engine config above.
+    This ensures updates on both the miniaudio side and the Steam Audio side are consistent.
+    */
+    iplAudioSettings.frameSize = engineConfig.periodSizeInFrames;
+
+
+    /* IPLContext */
+    MA_ZERO_OBJECT(&iplContextSettings);
+    iplContextSettings.version = STEAMAUDIO_VERSION;
+    
+    result = ma_result_from_IPLerror(iplContextCreate(&iplContextSettings, &iplContext));
+    if (result != MA_SUCCESS) {
+        ma_engine_uninit(&g_engine);
+        return result;
+    }
+
+
+    /* IPLHRTF */
+    MA_ZERO_OBJECT(&iplHRTFSettings);
+    iplHRTFSettings.type   = IPL_HRTFTYPE_DEFAULT;
+    iplHRTFSettings.volume = 1;
+
+    result = ma_result_from_IPLerror(iplHRTFCreate(iplContext, &iplAudioSettings, &iplHRTFSettings, &iplHRTF));
+    if (result != MA_SUCCESS) {
+        iplContextRelease(&iplContext);
+        ma_engine_uninit(&g_engine);
+        return result;
+    }
+
+
+    /*
+    The binaural node will need to know the input channel count of the sound so we'll need to load
+    the sound first. We'll initialize this such that it'll be initially detached from the graph.
+    It will be attached to the graph after the binaural node is initialized.
+    */
+    {
+        ma_sound_config soundConfig;
+
+        soundConfig = ma_sound_config_init();
+        soundConfig.pFilePath   = argv[1];
+        soundConfig.flags       = MA_SOUND_FLAG_NO_DEFAULT_ATTACHMENT;  /* We'll attach this to the graph later. */
+
+        result = ma_sound_init_ex(&g_engine, &soundConfig, &g_sound);
+        if (result != MA_SUCCESS) {
+            return result;
+        }
+
+        /* We'll let the Steam Audio binaural effect do the directional attenuation for us. */
+        ma_sound_set_directional_attenuation_factor(&g_sound, 0);
+
+        /* Loop the sound so we can get a continuous sound. */
+        ma_sound_set_looping(&g_sound, MA_TRUE);
+    }
+
+
+    /*
+    We'll build our graph starting from the end so initialize the binaural node now. The output of
+    this node will be connected straight to the output. You could also attach it to a sound group
+    or any other node that accepts an input.
+
+    Creating a node requires a pointer to the node graph that owns it. The engine itself is a node
+    graph. In the code below we can get a pointer to the node graph with `ma_engine_get_node_graph()`
+    or we could simple cast the engine to a ma_node_graph* like so:
+    
+        (ma_node_graph*)&g_engine
+
+    The endpoint of the graph can be retrieved with `ma_engine_get_endpoint()`.
+    */
+    {
+        ma_steamaudio_binaural_node_config binauralNodeConfig;
+
+        /*
+        For this example we're just using the engine's channel count, but a more optimal solution
+        might be to set this to mono if the source data is also mono.
+        */
+        binauralNodeConfig = ma_steamaudio_binaural_node_config_init(CHANNELS, iplAudioSettings, iplContext, iplHRTF);
+
+        result = ma_steamaudio_binaural_node_init(ma_engine_get_node_graph(&g_engine), &binauralNodeConfig, NULL, &g_binauralNode);
+        if (result != MA_SUCCESS) {
+            printf("Failed to initialize binaural node.");
+            return -1;
+        }
+
+        /* Connect the output of the delay node to the input of the endpoint. */
+        ma_node_attach_output_bus(&g_binauralNode, 0, ma_engine_get_endpoint(&g_engine), 0);
+    }
+
+
+    /* We can now wire up the sound to the binaural node and start it. */
+    ma_node_attach_output_bus(&g_sound, 0, &g_binauralNode, 0);
+    ma_sound_start(&g_sound);
+
+#if 1
+    {
+        /*
+        We'll move the sound around the listener which we'll leave at the origin. We'll then get
+        the direction to the listener and update the binaural node appropriately.
+        */
+        float stepAngle = 0.002f;
+        float angle = 0;
+        float distance = 2;
+
+        for (;;) {
+            double x = ma_cosd(angle) - ma_sind(angle);
+            double y = ma_sind(angle) + ma_cosd(angle);
+            ma_vec3f direction;
+
+            ma_sound_set_position(&g_sound, (float)x * distance, 0, (float)y * distance);
+            direction = ma_sound_get_direction_to_listener(&g_sound);
+
+            /* Update the direction of the sound. */
+            ma_steamaudio_binaural_node_set_direction(&g_binauralNode, direction.x, direction.y, direction.z);
+            angle += stepAngle;
+
+            ma_sleep(1);
+        }
+    }
+#else
+    printf("Press Enter to quit...");
+    getchar();
+#endif
+
+    ma_sound_uninit(&g_sound);
+    ma_steamaudio_binaural_node_uninit(&g_binauralNode, NULL);
+    ma_engine_uninit(&g_engine);
+
+    return 0;
+}

thirdparty/miniaudio-0.11.24/examples/hilo_interop.c

@@ -0,0 +1,146 @@
+/*
+Demonstrates interop between the high-level and the low-level API.
+
+In this example we are using `ma_device` (the low-level API) to capture data from the microphone
+which we then play back through the engine as a sound. We use a ring buffer to act as the data
+source for the sound.
+
+This is just a very basic example to show the general idea on how this might be achieved. In
+this example a ring buffer is being used as the intermediary data source, but you can use anything
+that works best for your situation. So long as the data is captured from the microphone, and then
+delivered to the sound (via a data source), you should be good to go.
+
+A more robust example would probably not want to use a ring buffer directly as the data source.
+Instead you would probably want to do a custom data source that handles underruns and overruns of
+the ring buffer and deals with desyncs between capture and playback. In the future this example
+may be updated to make use of a more advanced data source that handles all of this.
+*/
+#include "../miniaudio.c"
+
+static ma_pcm_rb rb;
+static ma_device device;
+static ma_engine engine;
+static ma_sound sound;  /* The sound will be the playback of the capture side. */
+
+void capture_data_callback(ma_device* pDevice, void* pFramesOut, const void* pFramesIn, ma_uint32 frameCount)
+{
+    ma_result result;
+    ma_uint32 framesWritten;
+
+    (void)pFramesOut;
+
+    /* We need to write to the ring buffer. Need to do this in a loop. */
+    framesWritten = 0;
+    while (framesWritten < frameCount) {
+        void* pMappedBuffer;
+        ma_uint32 framesToWrite = frameCount - framesWritten;
+
+        result = ma_pcm_rb_acquire_write(&rb, &framesToWrite, &pMappedBuffer);
+        if (result != MA_SUCCESS) {
+            break;
+        }
+
+        if (framesToWrite == 0) {
+            break;
+        }
+
+        /* Copy the data from the capture buffer to the ring buffer. */
+        ma_copy_pcm_frames(pMappedBuffer, ma_offset_pcm_frames_const_ptr_f32((const float*)pFramesIn, framesWritten, pDevice->capture.channels), framesToWrite, pDevice->capture.format, pDevice->capture.channels);
+
+        result = ma_pcm_rb_commit_write(&rb, framesToWrite);
+        if (result != MA_SUCCESS) {
+            break;
+        }
+
+        framesWritten += framesToWrite;
+    }
+}
+
+int main(int argc, char** argv)
+{
+    ma_result result;
+    ma_device_config deviceConfig;
+
+    /*
+    The first thing we'll do is set up the capture side. There are two parts to this. The first is
+    the device itself, and the other is the ring buffer. It doesn't matter what order we initialize
+    these in, so long as the ring buffer is created before the device is started so that the
+    callback can be guaranteed to have a valid destination. We'll initialize the device first, and
+    then use the format, channels and sample rate to initialize the ring buffer.
+
+    It's important that the sample format of the device is set to f32 because that's what the engine
+    uses internally.
+    */
+
+    /* Initialize the capture device. */
+    deviceConfig = ma_device_config_init(ma_device_type_capture);
+    deviceConfig.capture.format = ma_format_f32;
+    deviceConfig.dataCallback = capture_data_callback;
+
+    result = ma_device_init(NULL, &deviceConfig, &device);
+    if (result != MA_SUCCESS) {
+        printf("Failed to initialize capture device.");
+        return -1;
+    }
+
+    /* Initialize the ring buffer. */
+    result = ma_pcm_rb_init(device.capture.format, device.capture.channels, device.capture.internalPeriodSizeInFrames * 5, NULL, NULL, &rb);
+    if (result != MA_SUCCESS) {
+        printf("Failed to initialize the ring buffer.");
+        return -1;
+    }
+
+    /*
+    Ring buffers don't require a sample rate for their normal operation, but we can associate it
+    with a sample rate. We'll want to do this so the engine can resample if necessary.
+    */
+    ma_pcm_rb_set_sample_rate(&rb, device.sampleRate);
+
+
+
+    /*
+    At this point the capture side is set up and we can now set up the playback side. Here we are
+    using `ma_engine` and linking the captured data to a sound so it can be manipulated just like
+    any other sound in the world.
+
+    Note that we have not yet started the capture device. Since the captured data is tied to a
+    sound, we'll link the starting and stopping of the capture device to the starting and stopping
+    of the sound.
+    */
+
+    /* We'll get the engine up and running before we start the capture device. */
+    result = ma_engine_init(NULL, &engine);
+    if (result != MA_SUCCESS) {
+        printf("Failed to initialize the engine.");
+        return -1;
+    }
+
+    /*
+    We can now create our sound. This is created from a data source, which in this example is a
+    ring buffer. The capture side will be writing data into the ring buffer, whereas the sound
+    will be reading from it.
+    */
+    result = ma_sound_init_from_data_source(&engine, &rb, 0, NULL, &sound);
+    if (result != MA_SUCCESS) {
+        printf("Failed to initialize the sound.");
+        return -1;
+    }
+
+    /* Link the starting of the device and sound together. */
+    ma_device_start(&device);
+    ma_sound_start(&sound);
+
+
+    printf("Press Enter to quit...\n");
+    getchar();
+
+    ma_sound_uninit(&sound);
+    ma_engine_uninit(&engine);
+    ma_device_uninit(&device);
+    ma_pcm_rb_uninit(&rb);
+
+
+    (void)argc;
+    (void)argv;
+    return 0;
+}

thirdparty/miniaudio-0.11.24/examples/node_graph.c

@@ -0,0 +1,248 @@
+/*
+This example shows how to use the node graph system.
+
+The node graph system can be used for doing complex mixing and effect processing. The idea is that
+you have a number of nodes that are connected to each other to form a graph. At the end of the
+graph is an endpoint which all nodes eventually connect to.
+
+A node is used to do some kind of processing on zero or more input streams and produce one or more
+output streams. Each node can have a number of inputs and outputs. Each of these is called a bus in
+miniaudio. Some nodes, particularly data source nodes, have no inputs and instead generate their
+outputs dynamically. All nodes will have at least one output or else it'll be disconnected from the
+graph and will never get processed. Each output bus of a node will be connected to an input bus of
+another node, but they don't all need to connect to the same input node. For example, a splitter
+node has 1 input bus and 2 output buses and is used to duplicate a signal. You could then branch
+off and have one output bus connected to one input node and the other connected to a different
+input node, and then have two different effects process for each of the duplicated branches.
+
+Any number of output buses can be connected to an input bus in which case the output buses will be
+mixed before processing by the input node. This is how you would achieve the mixing part of the
+node graph.
+
+This example will be using the following node graph set up:
+
+    ```
+    >>>>>>>>>>>>>>>>>>>>>>>>>>>>>> Data flows left to right >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
+
+    +---------------+                              +-----------------+
+    | Data Source 1 =----+    +----------+    +----= Low Pass Filter =----+
+    +---------------+    |    |          =----+    +-----------------+    |    +----------+
+                         +----= Splitter |                                +----= ENDPOINT |
+    +---------------+    |    |          =----+    +-----------------+    |    +----------+
+    | Data Source 2 =----+    +----------+    +----=  Echo / Delay   =----+
+    +---------------+                              +-----------------+
+    ```
+
+This does not represent a realistic real-world scenario, but it demonstrates how to make use of
+mixing, multiple outputs and multiple effects.
+
+The data source nodes are connected to the input of the splitter. They'll be mixed before being
+processed by the splitter. The splitter has two output buses. In the graph above, one bus will be
+routed to a low pass filter, whereas the other bus will be routed to an echo effect. Then, the
+outputs of these two effects will be connected to the input bus of the endpoint. Because both of
+the outputs are connected to the same input bus, they'll be mixed at that point.
+
+The two data sources at the start of the graph have no inputs. They'll instead generate their
+output by reading from a data source. The data source in this case will be one `ma_decoder` for
+each input file specified on the command line.
+
+You can also control the volume of an output bus. In this example, we set the volumes of the low
+pass and echo effects so that one of them becomes more obvious than the other.
+
+When you want to read from the graph, you simply call `ma_node_graph_read_pcm_frames()`.
+*/
+#include "../miniaudio.c"
+
+/* Data Format */
+#define FORMAT              ma_format_f32   /* Must always be f32. */
+#define CHANNELS            2
+#define SAMPLE_RATE         48000
+
+/* Effect Properties */
+#define LPF_BIAS            0.9f    /* Higher values means more bias towards the low pass filter (the low pass filter will be more audible). Lower values means more bias towards the echo. Must be between 0 and 1. */
+#define LPF_CUTOFF_FACTOR   80      /* High values = more filter. */
+#define LPF_ORDER           8
+#define DELAY_IN_SECONDS    0.2f
+#define DECAY               0.5f    /* Volume falloff for each echo. */
+
+typedef struct
+{
+    ma_data_source_node node;   /* If you make this the first member, you can pass a pointer to this struct into any `ma_node_*` API and it will "Just Work". */
+    ma_decoder decoder;
+} sound_node;
+
+static ma_node_graph    g_nodeGraph;
+static ma_lpf_node      g_lpfNode;
+static ma_delay_node    g_delayNode;
+static ma_splitter_node g_splitterNode;
+static sound_node*      g_pSoundNodes;
+static int              g_soundNodeCount;
+
+void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
+{
+    /*
+    Hearing the output of the node graph is as easy as reading straight into the output buffer. You just need to
+    make sure you use a consistent data format or else you'll need to do your own conversion.
+    */
+    ma_node_graph_read_pcm_frames(&g_nodeGraph, pOutput, frameCount, NULL);
+
+    (void)pInput;   /* Unused. */
+    (void)pDevice;  /* Unused. */
+}
+
+int main(int argc, char** argv)
+{
+    int iarg;
+    ma_result result;
+
+    /* We'll set up our nodes starting from the end and working our way back to the start. We'll need to set up the graph first. */
+    {
+        ma_node_graph_config nodeGraphConfig = ma_node_graph_config_init(CHANNELS);
+
+        result = ma_node_graph_init(&nodeGraphConfig, NULL, &g_nodeGraph);
+        if (result != MA_SUCCESS) {
+            printf("ERROR: Failed to initialize node graph.");
+            return -1;
+        }
+    }
+
+
+    /* Low Pass Filter. */
+    {
+        ma_lpf_node_config lpfNodeConfig = ma_lpf_node_config_init(CHANNELS, SAMPLE_RATE, SAMPLE_RATE / LPF_CUTOFF_FACTOR, LPF_ORDER);
+
+        result = ma_lpf_node_init(&g_nodeGraph, &lpfNodeConfig, NULL, &g_lpfNode);
+        if (result != MA_SUCCESS) {
+            printf("ERROR: Failed to initialize low pass filter node.");
+            return -1;
+        }
+
+        /* Connect the output bus of the low pass filter node to the input bus of the endpoint. */
+        ma_node_attach_output_bus(&g_lpfNode, 0, ma_node_graph_get_endpoint(&g_nodeGraph), 0);
+
+        /* Set the volume of the low pass filter to make it more of less impactful. */
+        ma_node_set_output_bus_volume(&g_lpfNode, 0, LPF_BIAS);
+    }
+
+
+    /* Echo / Delay. */
+    {
+        ma_delay_node_config delayNodeConfig = ma_delay_node_config_init(CHANNELS, SAMPLE_RATE, (ma_uint32)(SAMPLE_RATE * DELAY_IN_SECONDS), DECAY);
+
+        result = ma_delay_node_init(&g_nodeGraph, &delayNodeConfig, NULL, &g_delayNode);
+        if (result != MA_SUCCESS) {
+            printf("ERROR: Failed to initialize delay node.");
+            return -1;
+        }
+
+        /* Connect the output bus of the delay node to the input bus of the endpoint. */
+        ma_node_attach_output_bus(&g_delayNode, 0, ma_node_graph_get_endpoint(&g_nodeGraph), 0);
+
+        /* Set the volume of the delay filter to make it more of less impactful. */
+        ma_node_set_output_bus_volume(&g_delayNode, 0, 1 - LPF_BIAS);
+    }
+
+
+    /* Splitter. */
+    {
+        ma_splitter_node_config splitterNodeConfig = ma_splitter_node_config_init(CHANNELS);
+
+        result = ma_splitter_node_init(&g_nodeGraph, &splitterNodeConfig, NULL, &g_splitterNode);
+        if (result != MA_SUCCESS) {
+            printf("ERROR: Failed to initialize splitter node.");
+            return -1;
+        }
+
+        /* Connect output bus 0 to the input bus of the low pass filter node, and output bus 1 to the input bus of the delay node. */
+        ma_node_attach_output_bus(&g_splitterNode, 0, &g_lpfNode,   0);
+        ma_node_attach_output_bus(&g_splitterNode, 1, &g_delayNode, 0);
+    }
+
+
+    /* Data sources. Ignore any that cannot be loaded. */
+    g_pSoundNodes = (sound_node*)ma_malloc(sizeof(*g_pSoundNodes) * argc-1, NULL);
+    if (g_pSoundNodes == NULL) {
+        printf("Failed to allocate memory for sounds.");
+        return -1;
+    }
+
+    g_soundNodeCount = 0;
+    for (iarg = 1; iarg < argc; iarg += 1) {
+        ma_decoder_config decoderConfig = ma_decoder_config_init(FORMAT, CHANNELS, SAMPLE_RATE);
+
+        result = ma_decoder_init_file(argv[iarg], &decoderConfig, &g_pSoundNodes[g_soundNodeCount].decoder);
+        if (result == MA_SUCCESS) {
+            ma_data_source_node_config dataSourceNodeConfig = ma_data_source_node_config_init(&g_pSoundNodes[g_soundNodeCount].decoder);
+
+            result = ma_data_source_node_init(&g_nodeGraph, &dataSourceNodeConfig, NULL, &g_pSoundNodes[g_soundNodeCount].node);
+            if (result == MA_SUCCESS) {
+                /* The data source node has been created successfully. Attach it to the splitter. */
+                ma_node_attach_output_bus(&g_pSoundNodes[g_soundNodeCount].node, 0, &g_splitterNode, 0);
+                g_soundNodeCount += 1;
+            } else {
+                printf("WARNING: Failed to init data source node for sound \"%s\". Ignoring.", argv[iarg]);
+                ma_decoder_uninit(&g_pSoundNodes[g_soundNodeCount].decoder);
+            }
+        } else {
+            printf("WARNING: Failed to load sound \"%s\". Ignoring.", argv[iarg]);
+        }
+    }
+
+    /* Everything has been initialized successfully so now we can set up a playback device so we can listen to the result. */
+    {
+        ma_device_config deviceConfig;
+        ma_device device;
+
+        deviceConfig = ma_device_config_init(ma_device_type_playback);
+        deviceConfig.playback.format   = FORMAT;
+        deviceConfig.playback.channels = CHANNELS;
+        deviceConfig.sampleRate        = SAMPLE_RATE;
+        deviceConfig.dataCallback      = data_callback;
+        deviceConfig.pUserData         = NULL;
+
+        result = ma_device_init(NULL, &deviceConfig, &device);
+        if (result != MA_SUCCESS) {
+            printf("ERROR: Failed to initialize device.");
+            goto cleanup_graph;
+        }
+
+        result = ma_device_start(&device);
+        if (result != MA_SUCCESS) {
+            ma_device_uninit(&device);
+            goto cleanup_graph;
+        }
+
+        printf("Press Enter to quit...\n");
+        getchar();
+
+        /* We're done. Clean up the device. */
+        ma_device_uninit(&device);
+    }
+
+
+cleanup_graph:
+    {
+        /* It's good practice to tear down the graph from the lowest level nodes first. */
+        int iSound;
+
+        /* Sounds. */
+        for (iSound = 0; iSound < g_soundNodeCount; iSound += 1) {
+            ma_data_source_node_uninit(&g_pSoundNodes[iSound].node, NULL);
+            ma_decoder_uninit(&g_pSoundNodes[iSound].decoder);
+        }
+
+        /* Splitter. */
+        ma_splitter_node_uninit(&g_splitterNode, NULL);
+        
+        /* Echo / Delay */
+        ma_delay_node_uninit(&g_delayNode, NULL);
+
+        /* Low Pass Filter */
+        ma_lpf_node_uninit(&g_lpfNode, NULL);
+
+        /* Node Graph */
+        ma_node_graph_uninit(&g_nodeGraph, NULL);
+    }    
+
+    return 0;
+}

thirdparty/miniaudio-0.11.24/examples/resource_manager.c

@@ -0,0 +1,149 @@
+/*
+Demonstrates how you can use the resource manager to manage loaded sounds.
+
+This example loads the first sound specified on the command line via the resource manager and then plays it using the
+low level API.
+
+You can control whether or not you want to load the sound asynchronously and whether or not you want to store the data
+in-memory or stream it. When storing the sound in-memory you can also control whether or not it is decoded. To do this,
+specify a combination of the following options in `ma_resource_manager_data_source_init()`:
+
+    * MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC  - Load asynchronously.
+    * MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_DECODE - Store the sound in-memory in uncompressed/decoded format.
+    * MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM - Stream the sound from disk rather than storing entirely in memory. Useful for music.
+
+The object returned by the resource manager is just a standard data source which means it can be plugged into any of
+`ma_data_source_*()` APIs just like any other data source and it should just work.
+
+Internally, there's a background thread that's used to process jobs and enable asynchronicity. By default there is only
+a single job thread, but this can be configured in the resource manager config. You can also implement your own threads
+for processing jobs. That is more advanced, and beyond the scope of this example.
+
+When you initialize a resource manager you can specify the sample format, channels and sample rate to use when reading
+data from the data source. This means the resource manager will ensure all sounds will have a standard format. When not
+set, each sound will have their own formats and you'll need to do the necessary data conversion yourself.
+*/
+#define MA_NO_ENGINE        /* We're intentionally not using the ma_engine API here. */
+#include "../miniaudio.c"
+
+#ifdef __EMSCRIPTEN__
+#include <emscripten.h>
+
+void main_loop__em(void* pUserData)
+{
+    ma_resource_manager* pResourceManager = (ma_resource_manager*)pUserData;
+
+    /*
+    The Emscripten build does not support threading which means we need to process jobs manually. If
+    there are no jobs needing to be processed this will return immediately with MA_NO_DATA_AVAILABLE.
+    */
+    ma_resource_manager_process_next_job(pResourceManager);
+}
+#endif
+
+void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
+{
+    ma_data_source_read_pcm_frames((ma_data_source*)pDevice->pUserData, pOutput, frameCount, NULL);
+
+    (void)pInput;
+}
+
+int main(int argc, char** argv)
+{
+    ma_result result;
+    ma_device_config deviceConfig;
+    ma_device device;
+    ma_resource_manager_config resourceManagerConfig;
+    ma_resource_manager resourceManager;
+    ma_resource_manager_data_source dataSource;
+
+    if (argc < 2) {
+        printf("No input file.");
+        return -1;
+    }
+
+
+    /* We'll initialize the device first. */
+    deviceConfig = ma_device_config_init(ma_device_type_playback);
+    deviceConfig.dataCallback = data_callback;
+    deviceConfig.pUserData    = &dataSource;    /* <-- We'll be reading from this in the data callback. */
+
+    result = ma_device_init(NULL, &deviceConfig, &device);
+    if (result != MA_SUCCESS) {
+        printf("Failed to initialize device.");
+        return -1;
+    }
+
+
+    /*
+    We have the device so now we want to initialize the resource manager. We'll use the resource manager to load a
+    sound based on the command line.
+    */
+    resourceManagerConfig = ma_resource_manager_config_init();
+    resourceManagerConfig.decodedFormat     = device.playback.format;
+    resourceManagerConfig.decodedChannels   = device.playback.channels;
+    resourceManagerConfig.decodedSampleRate = device.sampleRate;
+
+    /*
+    We're not supporting threading with Emscripten so go ahead and disable threading. It's important
+    that we set the appropriate flag and also the job thread count to 0.
+    */
+#ifdef __EMSCRIPTEN__
+    resourceManagerConfig.flags |= MA_RESOURCE_MANAGER_FLAG_NO_THREADING;
+    resourceManagerConfig.jobThreadCount = 0;
+#endif
+
+    result = ma_resource_manager_init(&resourceManagerConfig, &resourceManager);
+    if (result != MA_SUCCESS) {
+        ma_device_uninit(&device);
+        printf("Failed to initialize the resource manager.");
+        return -1;
+    }
+
+    /* Now that we have a resource manager we can load a sound. */
+    result = ma_resource_manager_data_source_init(
+        &resourceManager,
+        argv[1],
+        MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_DECODE | MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC | MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM,
+        NULL,   /* Async notification. */
+        &dataSource);
+    if (result != MA_SUCCESS) {
+        printf("Failed to load sound \"%s\".", argv[1]);
+        return -1;
+    }
+
+    /* In this example we'll enable looping. */
+    ma_data_source_set_looping(&dataSource, MA_TRUE);
+
+
+    /* Now that we have a sound we can start the device. */
+    result = ma_device_start(&device);
+    if (result != MA_SUCCESS) {
+        ma_device_uninit(&device);
+        printf("Failed to start device.");
+        return -1;
+    }
+
+#ifdef __EMSCRIPTEN__
+    emscripten_set_main_loop_arg(main_loop__em, &resourceManager, 0, 1);
+#else
+    printf("Press Enter to quit...\n");
+    getchar();
+#endif
+
+    /* Teardown. */
+
+    /* Uninitialize the device first to ensure the data callback is stopped and doesn't try to access any data. */
+    ma_device_uninit(&device);
+
+    /*
+    Before uninitializing the resource manager we need to uninitialize every data source. The data source is owned by
+    the caller which means you're responsible for uninitializing it.
+    */
+    ma_resource_manager_data_source_uninit(&dataSource);
+
+    /* Uninitialize the resource manager after each data source. */
+    ma_resource_manager_uninit(&resourceManager);
+
+    return 0;
+}

thirdparty/miniaudio-0.11.24/examples/resource_manager_advanced.c

@@ -0,0 +1,336 @@
+/*
+Demonstrates how you can use the resource manager to manage loaded sounds.
+
+The resource manager can be used to create a data source whose resources are managed internally by miniaudio. The data
+sources can then be read just like any other data source such as decoders and audio buffers.
+
+In this example we use the resource manager independently of the `ma_engine` API so that we can demonstrate how it can
+be used by itself without getting it confused with `ma_engine`.
+
+The main feature of the resource manager is the ability to decode and stream audio data asynchronously. Asynchronicity
+is achieved with a job system. The resource manager will issue jobs which are processed by a configurable number of job
+threads. You can also implement your own custom job threads which this example also demonstrates.
+
+In this example we show how you can create a data source, mix them with other data sources, configure the number of job
+threads to manage internally and how to implement your own custom job thread.
+*/
+#define MA_NO_ENGINE        /* We're intentionally not using the ma_engine API here. */
+#include "../miniaudio.c"
+
+static ma_resource_manager_data_source g_dataSources[16];
+static ma_uint32                       g_dataSourceCount;
+
+
+/*
+TODO: Consider putting these public functions in miniaudio.h. Will depend on ma_mix_pcm_frames_f32()
+being merged into miniaudio.h (it's currently in miniaudio_engine.h).
+*/
+static ma_result ma_data_source_read_pcm_frames_f32_ex(ma_data_source* pDataSource, float* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead, ma_format dataSourceFormat, ma_uint32 dataSourceChannels)
+{
+    /*
+    This function is intended to be used when the format and channel count of the data source is
+    known beforehand. The idea is to avoid overhead due to redundant calls to ma_data_source_get_data_format().
+    */
+    if (dataSourceFormat == ma_format_f32) {
+        /* Fast path. No conversion necessary. */
+        return ma_data_source_read_pcm_frames(pDataSource, pFramesOut, frameCount, pFramesRead);
+    } else {
+        /* Slow path. Conversion necessary. */
+        ma_result result;
+        ma_uint64 totalFramesRead;
+        ma_uint8 temp[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
+        ma_uint64 tempCapInFrames = sizeof(temp) / ma_get_bytes_per_frame(dataSourceFormat, dataSourceChannels);
+
+        if (pFramesRead != NULL) {
+            *pFramesRead = 0;
+        }
+        
+        totalFramesRead = 0;
+        while (totalFramesRead < frameCount) {
+            ma_uint64 framesJustRead;
+            ma_uint64 framesToRead = frameCount - totalFramesRead;
+            if (framesToRead > tempCapInFrames) {
+                framesToRead = tempCapInFrames;
+            }
+
+            result = ma_data_source_read_pcm_frames(pDataSource, temp, framesToRead, &framesJustRead);
+            if (result != MA_SUCCESS) {
+                break;
+            }
+
+            ma_convert_pcm_frames_format(ma_offset_pcm_frames_ptr_f32(pFramesOut, totalFramesRead, dataSourceChannels), ma_format_f32, temp, dataSourceFormat, framesJustRead, dataSourceChannels, ma_dither_mode_none);
+            totalFramesRead += framesJustRead;
+
+            if (result != MA_SUCCESS) {
+                break;
+            }
+        }
+
+        if (pFramesRead != NULL) {
+            *pFramesRead = totalFramesRead;
+        }
+
+        return MA_SUCCESS;
+    }
+}
+
+MA_API ma_result ma_data_source_read_pcm_frames_f32(ma_data_source* pDataSource, float* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
+{
+    ma_result result;
+    ma_format format;
+    ma_uint32 channels;
+
+    result = ma_data_source_get_data_format(pDataSource, &format, &channels, NULL, NULL, 0);
+    if (result != MA_SUCCESS) {
+        return result;  /* Failed to retrieve the data format of the data source. */
+    }
+
+    return ma_data_source_read_pcm_frames_f32_ex(pDataSource, pFramesOut, frameCount, pFramesRead, format, channels);
+}
+
+MA_API ma_result ma_data_source_read_pcm_frames_and_mix_f32(ma_data_source* pDataSource, float* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead, float volume)
+{
+    ma_result result;
+    ma_format format;
+    ma_uint32 channels;
+    ma_uint64 totalFramesRead;
+
+    if (pFramesRead != NULL) {
+        *pFramesRead = 0;
+    }
+
+    if (pDataSource == NULL) {
+        return MA_INVALID_ARGS;
+    }
+
+    result = ma_data_source_get_data_format(pDataSource, &format, &channels, NULL, NULL, 0);
+    if (result != MA_SUCCESS) {
+        return result;  /* Failed to retrieve the data format of the data source. */
+    }
+
+    totalFramesRead = 0;
+    while (totalFramesRead < frameCount) {
+        float temp[MA_DATA_CONVERTER_STACK_BUFFER_SIZE/sizeof(float)];
+        ma_uint64 tempCapInFrames = ma_countof(temp) / channels;
+        ma_uint64 framesJustRead;
+        ma_uint64 framesToRead = frameCount - totalFramesRead;
+        if (framesToRead > tempCapInFrames) {
+            framesToRead = tempCapInFrames;
+        }
+        
+        result = ma_data_source_read_pcm_frames_f32_ex(pDataSource, temp, framesToRead, &framesJustRead, format, channels);
+
+        ma_mix_pcm_frames_f32(ma_offset_pcm_frames_ptr_f32(pFramesOut, totalFramesRead, channels), temp, framesJustRead, channels, volume);
+        totalFramesRead += framesJustRead;
+
+        if (result != MA_SUCCESS) {
+            break;
+        }
+    }
+
+    if (pFramesRead != NULL) {
+        *pFramesRead = totalFramesRead;
+    }
+
+    return MA_SUCCESS;
+}
+
+
+void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
+{
+    /*
+    In this example we're just going to play our data sources layered on top of each other. This
+    assumes the device's format is f32 and that the buffer is not pre-silenced.
+    */
+    ma_uint32 iDataSource;
+
+    /*
+    If the device was configured with noPreSilencedOutputBuffer then you would need to silence the
+    buffer here, or make sure the first data source to be mixed is copied rather than mixed.
+    */
+    /*ma_silence_pcm_frames(pOutput, frameCount, ma_format_f32, pDevice->playback.channels);*/
+
+    /* For each sound, mix as much data as we can. */
+    for (iDataSource = 0; iDataSource < g_dataSourceCount; iDataSource += 1) {
+        ma_data_source_read_pcm_frames_and_mix_f32(&g_dataSources[iDataSource], (float*)pOutput, frameCount, NULL, /* volume = */1);
+    }
+
+    /* Unused. */
+    (void)pInput;
+    (void)pDevice;
+}
+
+static ma_thread_result MA_THREADCALL custom_job_thread(void* pUserData)
+{
+    ma_resource_manager* pResourceManager = (ma_resource_manager*)pUserData;
+
+    for (;;) {
+        ma_result result;
+        ma_resource_manager_job job;
+
+        /*
+        Retrieve a job from the queue first. This defines what it is you're about to do. By default this will be
+        blocking. You can initialize the resource manager with MA_RESOURCE_MANAGER_FLAG_NON_BLOCKING to not block in
+        which case MA_NO_DATA_AVAILABLE will be returned if no jobs are available.
+
+        When the quit job is returned (MA_RESOURCE_MANAGER_JOB_QUIT), the return value will always be MA_CANCELLED. If you don't want
+        to check the return value (you should), you can instead check if the job code is MA_RESOURCE_MANAGER_JOB_QUIT and use that
+        instead.
+        */
+        result = ma_resource_manager_next_job(pResourceManager, &job);
+        if (result != MA_SUCCESS) {
+            if (result == MA_CANCELLED) {
+                printf("CUSTOM JOB THREAD TERMINATING VIA MA_CANCELLED... ");
+            } else {
+                printf("CUSTOM JOB THREAD ERROR: %s. TERMINATING... ", ma_result_description(result));
+            }
+
+            break;
+        }
+
+        /*
+        Terminate if we got a quit message. You don't need to terminate like this, but's a bit more robust. You can
+        just use a global variable or something similar if it's easier for your particular situation. The quit job
+        remains in the queue and will continue to be returned by future calls to ma_resource_manager_next_job(). The
+        reason for this is to give every job thread visibility to the quit job so they have a chance to exit.
+
+        We won't actually be hitting this code because the call above will return MA_CANCELLED when the MA_RESOURCE_MANAGER_JOB_QUIT
+        event is received which means the `result != MA_SUCCESS` logic above will catch it. If you do not check the
+        return value of ma_resource_manager_next_job() you will want to check for MA_RESOURCE_MANAGER_JOB_QUIT like the code below.
+        */
+        if (job.toc.breakup.code == MA_JOB_TYPE_QUIT) {
+            printf("CUSTOM JOB THREAD TERMINATING VIA MA_JOB_TYPE_QUIT... ");
+            break;
+        }
+
+        /* Call ma_resource_manager_process_job() to actually do the work to process the job. */
+        printf("PROCESSING IN CUSTOM JOB THREAD: %d\n", job.toc.breakup.code);
+        ma_resource_manager_process_job(pResourceManager, &job);
+    }
+
+    printf("TERMINATED\n");
+    return (ma_thread_result)0;
+}
+
+int main(int argc, char** argv)
+{
+    ma_result result;
+    ma_device_config deviceConfig;
+    ma_device device;
+    ma_resource_manager_config resourceManagerConfig;
+    ma_resource_manager resourceManager;
+    ma_thread jobThread;
+    int iFile;
+
+    deviceConfig = ma_device_config_init(ma_device_type_playback);
+    deviceConfig.playback.format = ma_format_f32;
+    deviceConfig.dataCallback    = data_callback;
+    deviceConfig.pUserData       = NULL;
+
+    result = ma_device_init(NULL, &deviceConfig, &device);
+    if (result != MA_SUCCESS) {
+        printf("Failed to initialize device.");
+        return -1;
+    }
+
+
+    /* We can start the device before loading any sounds. We'll just end up outputting silence. */
+    result = ma_device_start(&device);
+    if (result != MA_SUCCESS) {
+        ma_device_uninit(&device);
+        printf("Failed to start device.");
+        return -1;
+    }
+
+
+    /*
+    We have the device so now we want to initialize the resource manager. We'll use the resource manager to load some
+    sounds based on the command line.
+    */
+    resourceManagerConfig = ma_resource_manager_config_init();
+
+    /*
+    We'll set a standard decoding format to save us to processing time at mixing time. If you're wanting to use
+    spatialization with your decoded sounds, you may want to consider leaving this as 0 to ensure the file's native
+    channel count is used so you can do proper spatialization.
+    */
+    resourceManagerConfig.decodedFormat     = device.playback.format;
+    resourceManagerConfig.decodedChannels   = device.playback.channels;
+    resourceManagerConfig.decodedSampleRate = device.sampleRate;
+
+    /* The number of job threads to be managed internally. Set this to 0 if you want to self-manage your job threads */
+    resourceManagerConfig.jobThreadCount = 4;
+
+    result = ma_resource_manager_init(&resourceManagerConfig, &resourceManager);
+    if (result != MA_SUCCESS) {
+        ma_device_uninit(&device);
+        printf("Failed to initialize the resource manager.");
+        return -1;
+    }
+
+    /*
+    Now that we have a resource manager we can set up our custom job thread. This is optional. Normally when doing
+    self-managed job threads you would set the internal job thread count to zero. We're doing both internal and
+    self-managed job threads in this example just for demonstration purposes.
+    */
+    ma_thread_create(&jobThread, ma_thread_priority_default, 0, custom_job_thread, &resourceManager, NULL);
+
+    /* Create each data source from the resource manager. Note that the caller is the owner. */
+    for (iFile = 0; iFile < (int)ma_countof(g_dataSources) && iFile < argc-1; iFile += 1) {
+        result = ma_resource_manager_data_source_init(
+            &resourceManager,
+            argv[iFile+1],
+            MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_DECODE | MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC /*| MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM*/,
+            NULL,   /* Async notification. */
+            &g_dataSources[iFile]);
+
+        if (result != MA_SUCCESS) {
+            break;
+        }
+
+        /* Use looping in this example. */
+        ma_data_source_set_looping(&g_dataSources[iFile], MA_TRUE);
+
+        g_dataSourceCount += 1;
+    }
+
+    printf("Press Enter to quit...");
+    getchar();
+
+
+    /* Teardown. */
+
+    /*
+    Uninitialize the device first to ensure the data callback is stopped and doesn't try to access
+    any data.
+    */
+    ma_device_uninit(&device);
+
+    /*
+    Our data sources need to be explicitly uninitialized. ma_resource_manager_uninit() will not do
+    it for us. This needs to be done before posting the quit event and uninitializing the resource
+    manager or else we'll get stuck in a deadlock because ma_resource_manager_data_source_uninit()
+    will be waiting for the job thread(s) to finish work, which will never happen because they were
+    just terminated.
+    */
+    for (iFile = 0; (size_t)iFile < g_dataSourceCount; iFile += 1) {
+        ma_resource_manager_data_source_uninit(&g_dataSources[iFile]);
+    }
+
+    /*
+    Before uninitializing the resource manager we need to make sure a quit event has been posted to
+    ensure we can get out of our custom thread. The call to ma_resource_manager_uninit() will also
+    do this, but we need to call it explicitly so that our self-managed thread can exit naturally.
+    You only need to post a quit job if you're using that as the exit indicator. You can instead
+    use whatever variable you want to terminate your job thread, but since this example is using a
+    quit job we need to post one. Note that you don't need to do this if you're not managing your
+    own threads - ma_resource_manager_uninit() alone will suffice in that case.
+    */
+    ma_resource_manager_post_job_quit(&resourceManager);
+    ma_thread_wait(&jobThread); /* Wait for the custom job thread to finish so it doesn't try to access any data. */
+    
+    /* Uninitialize the resource manager after each data source. */
+    ma_resource_manager_uninit(&resourceManager);
+
+    return 0;
+}

thirdparty/miniaudio-0.11.24/examples/simple_capture.c

@@ -0,0 +1,70 @@
+/*
+Demonstrates how to capture data from a microphone using the low-level API.
+
+This example simply captures data from your default microphone until you press Enter. The output is saved to the file
+specified on the command line.
+
+Capturing works in a very similar way to playback. The only difference is the direction of data movement. Instead of
+the application sending data to the device, the device will send data to the application. This example just writes the
+data received by the microphone straight to a WAV file.
+*/
+#include "../miniaudio.c"
+
+#include <stdlib.h>
+#include <stdio.h>
+
+void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
+{
+    ma_encoder_write_pcm_frames((ma_encoder*)pDevice->pUserData, pInput, frameCount, NULL);
+
+    (void)pOutput;
+}
+
+int main(int argc, char** argv)
+{
+    ma_result result;
+    ma_encoder_config encoderConfig;
+    ma_encoder encoder;
+    ma_device_config deviceConfig;
+    ma_device device;
+
+    if (argc < 2) {
+        printf("No output file.\n");
+        return -1;
+    }
+
+    encoderConfig = ma_encoder_config_init(ma_encoding_format_wav, ma_format_f32, 2, 44100);
+
+    if (ma_encoder_init_file(argv[1], &encoderConfig, &encoder) != MA_SUCCESS) {
+        printf("Failed to initialize output file.\n");
+        return -1;
+    }
+
+    deviceConfig = ma_device_config_init(ma_device_type_capture);
+    deviceConfig.capture.format   = encoder.config.format;
+    deviceConfig.capture.channels = encoder.config.channels;
+    deviceConfig.sampleRate       = encoder.config.sampleRate;
+    deviceConfig.dataCallback     = data_callback;
+    deviceConfig.pUserData        = &encoder;
+
+    result = ma_device_init(NULL, &deviceConfig, &device);
+    if (result != MA_SUCCESS) {
+        printf("Failed to initialize capture device.\n");
+        return -2;
+    }
+
+    result = ma_device_start(&device);
+    if (result != MA_SUCCESS) {
+        ma_device_uninit(&device);
+        printf("Failed to start device.\n");
+        return -3;
+    }
+
+    printf("Press Enter to stop recording...\n");
+    getchar();
+    
+    ma_device_uninit(&device);
+    ma_encoder_uninit(&encoder);
+
+    return 0;
+}

thirdparty/miniaudio-0.11.24/examples/simple_duplex.c

@@ -0,0 +1,72 @@
+/*
+Demonstrates duplex mode which is where data is captured from a microphone and then output to a speaker device.
+
+This example captures audio from the default microphone and then outputs it straight to the default playback device
+without any kind of modification. If you wanted to, you could also apply filters and effects to the input stream
+before outputting to the playback device.
+
+Note that the microphone and playback device must run in lockstep. Any kind of timing deviation will result in audible
+glitching which the backend may not be able to recover from. For this reason, miniaudio forces you to use the same
+sample rate for both capture and playback. If internally the native sample rates differ, miniaudio will perform the
+sample rate conversion for you automatically.
+*/
+#include "../miniaudio.c"
+
+#include <stdio.h>
+
+#ifdef __EMSCRIPTEN__
+void main_loop__em()
+{
+}
+#endif
+
+void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
+{
+    /* This example assumes the playback and capture sides use the same format and channel count. */
+    if (pDevice->capture.format != pDevice->playback.format || pDevice->capture.channels != pDevice->playback.channels) {
+        return;
+    }
+
+    /* In this example the format and channel count are the same for both input and output which means we can just memcpy(). */
+    MA_COPY_MEMORY(pOutput, pInput, frameCount * ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels));
+}
+
+int main(int argc, char** argv)
+{
+    ma_result result;
+    ma_device_config deviceConfig;
+    ma_device device;
+
+    deviceConfig = ma_device_config_init(ma_device_type_duplex);
+    deviceConfig.capture.pDeviceID  = NULL;
+    deviceConfig.capture.format     = ma_format_s16;
+    deviceConfig.capture.channels   = 2;
+    deviceConfig.capture.shareMode  = ma_share_mode_shared;
+    deviceConfig.playback.pDeviceID = NULL;
+    deviceConfig.playback.format    = ma_format_s16;
+    deviceConfig.playback.channels  = 2;
+    deviceConfig.dataCallback       = data_callback;
+    result = ma_device_init(NULL, &deviceConfig, &device);
+    if (result != MA_SUCCESS) {
+        return result;
+    }
+
+#ifdef __EMSCRIPTEN__
+    getchar();
+#endif
+
+    ma_device_start(&device);
+
+#ifdef __EMSCRIPTEN__
+    emscripten_set_main_loop(main_loop__em, 0, 1);
+#else
+    printf("Press Enter to quit...\n");
+    getchar();
+#endif
+
+    ma_device_uninit(&device);
+
+    (void)argc;
+    (void)argv;
+    return 0;
+}

thirdparty/miniaudio-0.11.24/examples/simple_enumeration.c

@@ -0,0 +1,53 @@
+/*
+Demonstrates how to enumerate over devices.
+
+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
+`ma_device_init()` when you initialize your devices.
+*/
+#include "../miniaudio.c"
+
+#include <stdio.h>
+
+int main(int argc, char** argv)
+{
+    ma_result result;
+    ma_context context;
+    ma_device_info* pPlaybackDeviceInfos;
+    ma_uint32 playbackDeviceCount;
+    ma_device_info* pCaptureDeviceInfos;
+    ma_uint32 captureDeviceCount;
+    ma_uint32 iDevice;
+
+    if (ma_context_init(NULL, 0, NULL, &context) != MA_SUCCESS) {
+        printf("Failed to initialize context.\n");
+        return -2;
+    }
+
+    result = ma_context_get_devices(&context, &pPlaybackDeviceInfos, &playbackDeviceCount, &pCaptureDeviceInfos, &captureDeviceCount);
+    if (result != MA_SUCCESS) {
+        printf("Failed to retrieve device information.\n");
+        return -3;
+    }
+
+    printf("Playback Devices\n");
+    for (iDevice = 0; iDevice < playbackDeviceCount; ++iDevice) {
+        printf("    %u: %s\n", iDevice, pPlaybackDeviceInfos[iDevice].name);
+    }
+
+    printf("\n");
+
+    printf("Capture Devices\n");
+    for (iDevice = 0; iDevice < captureDeviceCount; ++iDevice) {
+        printf("    %u: %s\n", iDevice, pCaptureDeviceInfos[iDevice].name);
+    }
+
+
+    ma_context_uninit(&context);
+
+    (void)argc;
+    (void)argv;
+    return 0;
+}

thirdparty/miniaudio-0.11.24/examples/simple_loopback.c

@@ -0,0 +1,78 @@
+/*
+Demonstrates how to implement loopback recording.
+
+This example simply captures data from your default playback device until you press Enter. The output is saved to the
+file specified on the command line.
+
+Loopback mode is when you record audio that is played from a given speaker. It is only supported on WASAPI, but can be
+used indirectly with PulseAudio by choosing the appropriate loopback device after enumeration.
+
+To use loopback mode you just need to set the device type to ma_device_type_loopback and set the capture device config
+properties. The output buffer in the callback will be null whereas the input buffer will be valid.
+*/
+#include "../miniaudio.c"
+
+#include <stdlib.h>
+#include <stdio.h>
+
+void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
+{
+    ma_encoder_write_pcm_frames((ma_encoder*)pDevice->pUserData, pInput, frameCount, NULL);
+
+    (void)pOutput;
+}
+
+int main(int argc, char** argv)
+{
+    ma_result result;
+    ma_encoder_config encoderConfig;
+    ma_encoder encoder;
+    ma_device_config deviceConfig;
+    ma_device device;
+
+    /* Loopback mode is currently only supported on WASAPI. */
+    ma_backend backends[] = {
+        ma_backend_wasapi
+    };
+
+    if (argc < 2) {
+        printf("No output file.\n");
+        return -1;
+    }
+
+    encoderConfig = ma_encoder_config_init(ma_encoding_format_wav, ma_format_f32, 2, 44100);
+
+    if (ma_encoder_init_file(argv[1], &encoderConfig, &encoder) != MA_SUCCESS) {
+        printf("Failed to initialize output file.\n");
+        return -1;
+    }
+
+    deviceConfig = ma_device_config_init(ma_device_type_loopback);
+    deviceConfig.capture.pDeviceID = NULL; /* Use default device for this example. Set this to the ID of a _playback_ device if you want to capture from a specific device. */
+    deviceConfig.capture.format    = encoder.config.format;
+    deviceConfig.capture.channels  = encoder.config.channels;
+    deviceConfig.sampleRate        = encoder.config.sampleRate;
+    deviceConfig.dataCallback      = data_callback;
+    deviceConfig.pUserData         = &encoder;
+
+    result = ma_device_init_ex(backends, sizeof(backends)/sizeof(backends[0]), NULL, &deviceConfig, &device);
+    if (result != MA_SUCCESS) {
+        printf("Failed to initialize loopback device.\n");
+        return -2;
+    }
+
+    result = ma_device_start(&device);
+    if (result != MA_SUCCESS) {
+        ma_device_uninit(&device);
+        printf("Failed to start device.\n");
+        return -3;
+    }
+
+    printf("Press Enter to stop recording...\n");
+    getchar();
+    
+    ma_device_uninit(&device);
+    ma_encoder_uninit(&encoder);
+
+    return 0;
+}

thirdparty/miniaudio-0.11.24/examples/simple_looping.c

@@ -0,0 +1,75 @@
+/*
+Shows one way to handle looping of a sound.
+
+This example uses a decoder as the data source. Decoders can be used with the `ma_data_source` API which, conveniently,
+supports looping via the `ma_data_source_read_pcm_frames()` API. To use it, all you need to do is pass a pointer to the
+decoder straight into `ma_data_source_read_pcm_frames()` and it will just work.
+*/
+#include "../miniaudio.c"
+
+#include <stdio.h>
+
+void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
+{
+    ma_decoder* pDecoder = (ma_decoder*)pDevice->pUserData;
+    if (pDecoder == NULL) {
+        return;
+    }
+
+    /* Reading PCM frames will loop based on what we specified when called ma_data_source_set_looping(). */
+    ma_data_source_read_pcm_frames(pDecoder, pOutput, frameCount, NULL);
+
+    (void)pInput;
+}
+
+int main(int argc, char** argv)
+{
+    ma_result result;
+    ma_decoder decoder;
+    ma_device_config deviceConfig;
+    ma_device device;
+
+    if (argc < 2) {
+        printf("No input file.\n");
+        return -1;
+    }
+
+    result = ma_decoder_init_file(argv[1], NULL, &decoder);
+    if (result != MA_SUCCESS) {
+        return -2;
+    }
+
+    /*
+    A decoder is a data source which means we just use ma_data_source_set_looping() to set the
+    looping state. We will read data using ma_data_source_read_pcm_frames() in the data callback.
+    */
+    ma_data_source_set_looping(&decoder, MA_TRUE);
+
+    deviceConfig = ma_device_config_init(ma_device_type_playback);
+    deviceConfig.playback.format   = decoder.outputFormat;
+    deviceConfig.playback.channels = decoder.outputChannels;
+    deviceConfig.sampleRate        = decoder.outputSampleRate;
+    deviceConfig.dataCallback      = data_callback;
+    deviceConfig.pUserData         = &decoder;
+
+    if (ma_device_init(NULL, &deviceConfig, &device) != MA_SUCCESS) {
+        printf("Failed to open playback device.\n");
+        ma_decoder_uninit(&decoder);
+        return -3;
+    }
+
+    if (ma_device_start(&device) != MA_SUCCESS) {
+        printf("Failed to start playback device.\n");
+        ma_device_uninit(&device);
+        ma_decoder_uninit(&decoder);
+        return -4;
+    }
+
+    printf("Press Enter to quit...");
+    getchar();
+
+    ma_device_uninit(&device);
+    ma_decoder_uninit(&decoder);
+
+    return 0;
+}

thirdparty/miniaudio-0.11.24/examples/simple_mixing.c

@@ -0,0 +1,198 @@
+/*
+Demonstrates one way to load multiple files and play them all back at the same time.
+
+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 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]
+Example: simple_mixing file1.wav file2.flac
+```
+*/
+#include "../miniaudio.c"
+
+#include <stdio.h>
+
+/*
+For simplicity, this example requires the device to use floating point samples.
+*/
+#define SAMPLE_FORMAT   ma_format_f32
+#define CHANNEL_COUNT   2
+#define SAMPLE_RATE     48000
+
+ma_uint32   g_decoderCount;
+ma_decoder* g_pDecoders;
+ma_bool32*  g_pDecodersAtEnd;
+
+ma_event g_stopEvent; /* <-- Signaled by the audio thread, waited on by the main thread. */
+
+ma_bool32 are_all_decoders_at_end(void)
+{
+    ma_uint32 iDecoder;
+    for (iDecoder = 0; iDecoder < g_decoderCount; ++iDecoder) {
+        if (g_pDecodersAtEnd[iDecoder] == MA_FALSE) {
+            return MA_FALSE;
+        }
+    }
+
+    return MA_TRUE;
+}
+
+ma_uint32 read_and_mix_pcm_frames_f32(ma_decoder* pDecoder, float* pOutputF32, ma_uint32 frameCount)
+{
+    /*
+    The way mixing works is that we just read into a temporary buffer, then take the contents of that buffer and mix it with the
+    contents of the output buffer by simply adding the samples together. You could also clip the samples to -1..+1, but I'm not
+    doing that in this example.
+    */
+    ma_result result;
+    float temp[4096];
+    ma_uint32 tempCapInFrames = ma_countof(temp) / CHANNEL_COUNT;
+    ma_uint32 totalFramesRead = 0;
+
+    while (totalFramesRead < frameCount) {
+        ma_uint64 iSample;
+        ma_uint64 framesReadThisIteration;
+        ma_uint32 totalFramesRemaining = frameCount - totalFramesRead;
+        ma_uint32 framesToReadThisIteration = tempCapInFrames;
+        if (framesToReadThisIteration > totalFramesRemaining) {
+            framesToReadThisIteration = totalFramesRemaining;
+        }
+
+        result = ma_decoder_read_pcm_frames(pDecoder, temp, framesToReadThisIteration, &framesReadThisIteration);
+        if (result != MA_SUCCESS || framesReadThisIteration == 0) {
+            break;
+        }
+
+        /* Mix the frames together. */
+        for (iSample = 0; iSample < framesReadThisIteration*CHANNEL_COUNT; ++iSample) {
+            pOutputF32[totalFramesRead*CHANNEL_COUNT + iSample] += temp[iSample];
+        }
+
+        totalFramesRead += (ma_uint32)framesReadThisIteration;
+
+        if (framesReadThisIteration < (ma_uint32)framesToReadThisIteration) {
+            break;  /* Reached EOF. */
+        }
+    }
+    
+    return totalFramesRead;
+}
+
+void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
+{
+    float* pOutputF32 = (float*)pOutput;
+    ma_uint32 iDecoder;
+
+    /* This example assumes the device was configured to use ma_format_f32. */
+    for (iDecoder = 0; iDecoder < g_decoderCount; ++iDecoder) {
+        if (!g_pDecodersAtEnd[iDecoder]) {
+            ma_uint32 framesRead = read_and_mix_pcm_frames_f32(&g_pDecoders[iDecoder], pOutputF32, frameCount);
+            if (framesRead < frameCount) {
+                g_pDecodersAtEnd[iDecoder] = MA_TRUE;
+            }
+        }
+    }
+
+    /*
+    If at the end all of our decoders are at the end we need to stop. We cannot stop the device in the callback. Instead we need to
+    signal an event to indicate that it's stopped. The main thread will be waiting on the event, after which it will stop the device.
+    */
+    if (are_all_decoders_at_end()) {
+        ma_event_signal(&g_stopEvent);
+    }
+
+    (void)pInput;
+    (void)pDevice;
+}
+
+int main(int argc, char** argv)
+{
+    ma_result result;
+    ma_decoder_config decoderConfig;
+    ma_device_config deviceConfig;
+    ma_device device;
+    ma_uint32 iDecoder;
+
+    if (argc < 2) {
+        printf("No input files.\n");
+        return -1;
+    }
+
+    g_decoderCount   = argc-1;
+    g_pDecoders      = (ma_decoder*)malloc(sizeof(*g_pDecoders)      * g_decoderCount);
+    g_pDecodersAtEnd = (ma_bool32*) malloc(sizeof(*g_pDecodersAtEnd) * g_decoderCount);
+
+    /* In this example, all decoders need to have the same output format. */
+    decoderConfig = ma_decoder_config_init(SAMPLE_FORMAT, CHANNEL_COUNT, SAMPLE_RATE);
+    for (iDecoder = 0; iDecoder < g_decoderCount; ++iDecoder) {
+        result = ma_decoder_init_file(argv[1+iDecoder], &decoderConfig, &g_pDecoders[iDecoder]);
+        if (result != MA_SUCCESS) {
+            ma_uint32 iDecoder2;
+            for (iDecoder2 = 0; iDecoder2 < iDecoder; ++iDecoder2) {
+                ma_decoder_uninit(&g_pDecoders[iDecoder2]);
+            }
+            free(g_pDecoders);
+            free(g_pDecodersAtEnd);
+
+            printf("Failed to load %s.\n", argv[1+iDecoder]);
+            return -3;
+        }
+        g_pDecodersAtEnd[iDecoder] = MA_FALSE;
+    }
+
+    /* Create only a single device. The decoders will be mixed together in the callback. In this example the data format needs to be the same as the decoders. */
+    deviceConfig = ma_device_config_init(ma_device_type_playback);
+    deviceConfig.playback.format   = SAMPLE_FORMAT;
+    deviceConfig.playback.channels = CHANNEL_COUNT;
+    deviceConfig.sampleRate        = SAMPLE_RATE;
+    deviceConfig.dataCallback      = data_callback;
+    deviceConfig.pUserData         = NULL;
+
+    if (ma_device_init(NULL, &deviceConfig, &device) != MA_SUCCESS) {
+        for (iDecoder = 0; iDecoder < g_decoderCount; ++iDecoder) {
+            ma_decoder_uninit(&g_pDecoders[iDecoder]);
+        }
+        free(g_pDecoders);
+        free(g_pDecodersAtEnd);
+
+        printf("Failed to open playback device.\n");
+        return -3;
+    }
+
+    /*
+    We can't stop in the audio thread so we instead need to use an event. We wait on this thread in the main thread, and signal it in the audio thread. This
+    needs to be done before starting the device. We need a context to initialize the event, which we can get from the device. Alternatively you can initialize
+    a context separately, but we don't need to do that for this example.
+    */
+    ma_event_init(&g_stopEvent);
+
+    /* Now we start playback and wait for the audio thread to tell us to stop. */
+    if (ma_device_start(&device) != MA_SUCCESS) {
+        ma_device_uninit(&device);
+        for (iDecoder = 0; iDecoder < g_decoderCount; ++iDecoder) {
+            ma_decoder_uninit(&g_pDecoders[iDecoder]);
+        }
+        free(g_pDecoders);
+        free(g_pDecodersAtEnd);
+
+        printf("Failed to start playback device.\n");
+        return -4;
+    }
+
+    printf("Waiting for playback to complete...\n");
+    ma_event_wait(&g_stopEvent);
+    
+    /* Getting here means the audio thread has signaled that the device should be stopped. */
+    ma_device_uninit(&device);
+    
+    for (iDecoder = 0; iDecoder < g_decoderCount; ++iDecoder) {
+        ma_decoder_uninit(&g_pDecoders[iDecoder]);
+    }
+    free(g_pDecoders);
+    free(g_pDecodersAtEnd);
+
+    return 0;
+}

thirdparty/miniaudio-0.11.24/examples/simple_playback.c

@@ -0,0 +1,74 @@
+/*
+Demonstrates how to load a sound file and play it back using the low-level API.
+
+The low-level API uses a callback to deliver audio between the application and miniaudio for playback or recording. When
+in playback mode, as in this example, the application sends raw audio data to miniaudio which is then played back through
+the default playback device as defined by the operating system.
+
+This example uses the `ma_decoder` API to load a sound and play it back. The decoder is entirely decoupled from the
+device and can be used independently of it. This example only plays back a single sound file, but it's possible to play
+back multiple files by simple loading multiple decoders and mixing them (do not create multiple devices to do this). See
+the simple_mixing example for how best to do this.
+*/
+#include "../miniaudio.c"
+
+#include <stdio.h>
+
+void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
+{
+    ma_decoder* pDecoder = (ma_decoder*)pDevice->pUserData;
+    if (pDecoder == NULL) {
+        return;
+    }
+
+    ma_decoder_read_pcm_frames(pDecoder, pOutput, frameCount, NULL);
+
+    (void)pInput;
+}
+
+int main(int argc, char** argv)
+{
+    ma_result result;
+    ma_decoder decoder;
+    ma_device_config deviceConfig;
+    ma_device device;
+
+    if (argc < 2) {
+        printf("No input file.\n");
+        return -1;
+    }
+
+    result = ma_decoder_init_file(argv[1], NULL, &decoder);
+    if (result != MA_SUCCESS) {
+        printf("Could not load file: %s\n", argv[1]);
+        return -2;
+    }
+
+    deviceConfig = ma_device_config_init(ma_device_type_playback);
+    deviceConfig.playback.format   = decoder.outputFormat;
+    deviceConfig.playback.channels = decoder.outputChannels;
+    deviceConfig.sampleRate        = decoder.outputSampleRate;
+    deviceConfig.dataCallback      = data_callback;
+    deviceConfig.pUserData         = &decoder;
+
+    if (ma_device_init(NULL, &deviceConfig, &device) != MA_SUCCESS) {
+        printf("Failed to open playback device.\n");
+        ma_decoder_uninit(&decoder);
+        return -3;
+    }
+
+    if (ma_device_start(&device) != MA_SUCCESS) {
+        printf("Failed to start playback device.\n");
+        ma_device_uninit(&device);
+        ma_decoder_uninit(&decoder);
+        return -4;
+    }
+
+    printf("Press Enter to quit...");
+    getchar();
+
+    ma_device_uninit(&device);
+    ma_decoder_uninit(&decoder);
+
+    return 0;
+}

thirdparty/miniaudio-0.11.24/examples/simple_playback_sine.c

@@ -0,0 +1,83 @@
+/*
+Demonstrates playback of a sine wave.
+
+Since all this example is doing is playing back a sine wave, we can disable decoding (and encoding) which will slightly
+reduce the size of the executable. This is done with the `MA_NO_DECODING` and `MA_NO_ENCODING` options.
+
+The generation of sine wave is achieved via the `ma_waveform` API. A waveform is a data source which means it can be
+seamlessly plugged into the `ma_data_source_*()` family of APIs as well.
+
+A waveform is initialized using the standard config/init pattern used throughout all of miniaudio. Frames are read via
+the `ma_waveform_read_pcm_frames()` API.
+
+This example works with Emscripten.
+*/
+#define MA_NO_DECODING
+#define MA_NO_ENCODING
+#include "../miniaudio.c"
+
+#include <stdio.h>
+
+#ifdef __EMSCRIPTEN__
+#include <emscripten.h>
+
+void main_loop__em()
+{
+}
+#endif
+
+#define DEVICE_FORMAT       ma_format_f32
+#define DEVICE_CHANNELS     2
+#define DEVICE_SAMPLE_RATE  48000
+
+void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
+{
+    ma_waveform_read_pcm_frames((ma_waveform*)pDevice->pUserData, pOutput, frameCount, NULL);
+
+    (void)pInput;   /* Unused. */
+}
+
+int main(int argc, char** argv)
+{
+    ma_waveform sineWave;
+    ma_device_config deviceConfig;
+    ma_device device;
+    ma_waveform_config sineWaveConfig;
+
+    deviceConfig = ma_device_config_init(ma_device_type_playback);
+    deviceConfig.playback.format   = DEVICE_FORMAT;
+    deviceConfig.playback.channels = DEVICE_CHANNELS;
+    deviceConfig.sampleRate        = DEVICE_SAMPLE_RATE;
+    deviceConfig.dataCallback      = data_callback;
+    deviceConfig.pUserData         = &sineWave;
+
+    if (ma_device_init(NULL, &deviceConfig, &device) != MA_SUCCESS) {
+        printf("Failed to open playback device.\n");
+        return -4;
+    }
+
+    printf("Device Name: %s\n", device.playback.name);
+
+    sineWaveConfig = ma_waveform_config_init(device.playback.format, device.playback.channels, device.sampleRate, ma_waveform_type_sine, 0.2, 220);
+    ma_waveform_init(&sineWaveConfig, &sineWave);
+
+    if (ma_device_start(&device) != MA_SUCCESS) {
+        printf("Failed to start playback device.\n");
+        ma_device_uninit(&device);
+        return -5;
+    }
+    
+#ifdef __EMSCRIPTEN__
+    emscripten_set_main_loop(main_loop__em, 0, 1);
+#else
+    printf("Press Enter to quit...\n");
+    getchar();
+#endif
+    
+    ma_device_uninit(&device);
+    ma_waveform_uninit(&sineWave);  /* Uninitialize the waveform after the device so we don't pull it from under the device while it's being reference in the data callback. */
+    
+    (void)argc;
+    (void)argv;
+    return 0;
+}

thirdparty/miniaudio-0.11.24/examples/simple_spatialization.c

@@ -0,0 +1,85 @@
+/*
+Demonstrates how to do basic spatialization via the high level API.
+
+You can position and orientate sounds to create a simple spatialization effect. This example shows
+how to do this.
+
+In addition to positioning sounds, there is the concept of a listener. This can also be positioned
+and orientated to help with spatialization.
+
+This example only covers the basics to get your started. See the documentation for more detailed
+information on the available features.
+
+To use this example, pass in the path of a sound as the first argument. The sound will be
+positioned in front of the listener, while the listener rotates on the the spot to create an
+orbiting effect. Terminate the program with Ctrl+C.
+*/
+#include "../miniaudio.c"
+
+#include <stdio.h>
+#include <math.h>   /* For sinf() and cosf() */
+
+/* Silence warning about unreachable code for MSVC. */
+#ifdef _MSC_VER
+    #pragma warning(push)
+    #pragma warning(disable: 4702)
+#endif
+
+int main(int argc, char** argv)
+{
+    ma_result result;
+    ma_engine engine;
+    ma_sound sound;
+    float listenerAngle = 0;
+
+    if (argc < 2) {
+        printf("No input file.\n");
+        return -1;
+    }
+
+    result = ma_engine_init(NULL, &engine);
+    if (result != MA_SUCCESS) {
+        printf("Failed to initialize engine.\n");
+        return -1;
+    }
+
+    result = ma_sound_init_from_file(&engine, argv[1], 0, NULL, NULL, &sound);
+    if (result != MA_SUCCESS) {
+        printf("Failed to load sound: %s\n", argv[1]);
+        ma_engine_uninit(&engine);
+        return -1;
+    }
+
+    /* This sets the position of the sound. miniaudio follows the same coordinate system as OpenGL, where -Z is forward. */
+    ma_sound_set_position(&sound, 0, 0, -1);
+
+    /*
+    This sets the position of the listener. The second parameter is the listener index. If you have only a single listener, which is
+    most likely, just use 0. The position defaults to (0,0,0).
+    */
+    ma_engine_listener_set_position(&engine, 0, 0, 0, 0);
+
+
+    /* Sounds are stopped by default. We'll start it once initial parameters have been setup. */
+    ma_sound_start(&sound);
+
+
+    /* Rotate the listener on the spot to create an orbiting effect. */
+    for (;;) {
+        listenerAngle += 0.01f;
+        ma_engine_listener_set_direction(&engine, 0, (float)sin(listenerAngle), 0, (float)cos(listenerAngle));
+
+        ma_sleep(1);
+    }
+
+
+    /* Won't actually get here, but do this to tear down. */
+    ma_sound_uninit(&sound);
+    ma_engine_uninit(&engine);
+
+    return 0;
+}
+
+#ifdef _MSC_VER
+    #pragma warning(pop)
+#endif