Stop using MA_ASSERT in examples.

This is useful because MA_ASSERT is only defined in the implementation
section of miniaudio.h which can cause issues when people copy/paste
the code and use it in a file that does not have visibility of the
implementation.

Note that there are still more references to implementation-defined
macros, but these have been moved to the public section in the dev-0.12
branch so I'm not bothering to change those just yet.

Public issue https://github.com/mackron/miniaudio/issues/787

examples/custom_backend.c

@@ -180,9 +180,6 @@ static ma_result ma_context_enumerate_devices__sdl(ma_context* pContext, ma_enum
     ma_bool32 cbResult;
     int iDevice;
 
-    MA_ASSERT(pContext != NULL);
-    MA_ASSERT(callback != NULL);
-
     /* Playback */
     if (!isTerminated) {
         int deviceCount = ((MA_PFN_SDL_GetNumAudioDevices)pContextEx->sdl.SDL_GetNumAudioDevices)(0);
@@ -241,8 +238,6 @@ static ma_result ma_context_get_device_info__sdl(ma_context* pContext, ma_device
     const char* pDeviceName;
 #endif
 
-    MA_ASSERT(pContext != NULL);
-
     if (pDeviceID == NULL) {
         if (deviceType == ma_device_type_playback) {
             pDeviceInfo->id.custom.i = 0;
@@ -322,8 +317,6 @@ void ma_audio_callback_capture__sdl(void* pUserData, ma_uint8* pBuffer, int buff
 {
     ma_device_ex* pDeviceEx = (ma_device_ex*)pUserData;
 
-    MA_ASSERT(pDeviceEx != NULL);
-
     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));
 }
 
@@ -331,8 +324,6 @@ void ma_audio_callback_playback__sdl(void* pUserData, ma_uint8* pBuffer, int buf
 {
     ma_device_ex* pDeviceEx = (ma_device_ex*)pUserData;
 
-    MA_ASSERT(pDeviceEx != NULL);
-
     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));
 }
 
@@ -344,9 +335,6 @@ static ma_result ma_device_init_internal__sdl(ma_device_ex* pDeviceEx, const ma_
     const char* pDeviceName;
     int deviceID;
 
-    MA_ASSERT(pDeviceEx  != NULL);
-    MA_ASSERT(pDescriptor != NULL);
-
     /*
     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
@@ -430,8 +418,6 @@ static ma_result ma_device_init__sdl(ma_device* pDevice, const ma_device_config*
     ma_context_ex* pContextEx = (ma_context_ex*)pDevice->pContext;
     ma_result result;
 
-    MA_ASSERT(pDevice != NULL);
-    
     /* 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;
@@ -463,8 +449,6 @@ 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;
 
-    MA_ASSERT(pDevice != NULL);
-
     if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
         ((MA_PFN_SDL_CloseAudioDevice)pContextEx->sdl.SDL_CloseAudioDevice)(pDeviceEx->sdl.deviceIDCapture);
     }
@@ -481,8 +465,6 @@ 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;
 
-    MA_ASSERT(pDevice != NULL);
-
     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);
     }
@@ -499,8 +481,6 @@ 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;
 
-    MA_ASSERT(pDevice != NULL);
-
     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);
     }
@@ -516,8 +496,6 @@ static ma_result ma_context_uninit__sdl(ma_context* pContext)
 {
     ma_context_ex* pContextEx = (ma_context_ex*)pContext;
 
-    MA_ASSERT(pContext != NULL);
-
     ((MA_PFN_SDL_QuitSubSystem)pContextEx->sdl.SDL_QuitSubSystem)(MA_SDL_INIT_AUDIO);
 
     /* Close the handle to the SDL shared object last. */
@@ -545,8 +523,6 @@ static ma_result ma_context_init__sdl(ma_context* pContext, const ma_context_con
 #endif
     };
 
-    MA_ASSERT(pContext != NULL);
-
     (void)pConfig;
 
     /* Check if we have SDL2 installed somewhere. If not it's not usable and we need to abort. */
@@ -641,15 +617,8 @@ Main program starts here.
 
 void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
 {
-    MA_ASSERT(pDevice->playback.channels == DEVICE_CHANNELS);
-
     if (pDevice->type == ma_device_type_playback)  {
-        ma_waveform* pSineWave;
+        ma_waveform_read_pcm_frames((ma_waveform*)pDevice->pUserData, pOutput, frameCount, NULL);
-
-        pSineWave = (ma_waveform*)pDevice->pUserData;
-        MA_ASSERT(pSineWave != NULL);
-
-        ma_waveform_read_pcm_frames(pSineWave, pOutput, frameCount, NULL);
     }
 
     if (pDevice->type == ma_device_type_duplex) {

examples/data_source_chaining.c

@@ -49,7 +49,9 @@ ma_decoder* g_pDecoders;
 
 static ma_data_source* next_callback_tail(ma_data_source* pDataSource)
 {
-    MA_ASSERT(g_decoderCount > 0);  /* <-- We check for this in main() so should never happen. */
+    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

examples/duplex_effect.c

@@ -12,7 +12,7 @@ effect.
 
 #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_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. */
@@ -24,8 +24,13 @@ static ma_node_graph       g_nodeGraph;
 
 void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
 {
-    MA_ASSERT(pDevice->capture.format == pDevice->playback.format);
+    /*
-    MA_ASSERT(pDevice->capture.channels == pDevice->playback.channels);
+    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 

examples/node_graph.c

@@ -81,8 +81,6 @@ static int              g_soundNodeCount;
 
 void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
 {
-    MA_ASSERT(pDevice->playback.channels == CHANNELS);
-
     /*
     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.

examples/resource_manager.c

@@ -33,7 +33,6 @@ set, each sound will have their own formats and you'll need to do the necessary
 void main_loop__em(void* pUserData)
 {
     ma_resource_manager* pResourceManager = (ma_resource_manager*)pUserData;
-    MA_ASSERT(pResourceManager != NULL);
 
     /*
     The Emscripten build does not support threading which means we need to process jobs manually. If

examples/resource_manager_advanced.c

@@ -32,8 +32,6 @@ static ma_result ma_data_source_read_pcm_frames_f32_ex(ma_data_source* pDataSour
     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().
     */
-    MA_ASSERT(pDataSource != NULL);
-
     if (dataSourceFormat == ma_format_f32) {
         /* Fast path. No conversion necessary. */
         return ma_data_source_read_pcm_frames(pDataSource, pFramesOut, frameCount, pFramesRead);
@@ -136,10 +134,6 @@ void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uin
     */
     ma_uint32 iDataSource;
 
-    MA_ASSERT(pDevice->playback.format == ma_format_f32);
-
-    (void)pInput;   /* Unused. */
-
     /*
     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.
@@ -150,12 +144,15 @@ void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uin
     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;
-    MA_ASSERT(pResourceManager != NULL);
 
     for (;;) {
         ma_result result;
@@ -191,8 +188,8 @@ static ma_thread_result MA_THREADCALL custom_job_thread(void* pUserData)
         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_RESOURCE_MANAGER_JOB_QUIT) {
+        if (job.toc.breakup.code == MA_JOB_TYPE_QUIT) {
-            printf("CUSTOM JOB THREAD TERMINATING VIA MA_RESOURCE_MANAGER_JOB_QUIT... ");
+            printf("CUSTOM JOB THREAD TERMINATING VIA MA_JOB_TYPE_QUIT... ");
             break;
         }
 

examples/simple_capture.c

@@ -16,10 +16,7 @@ data received by the microphone straight to a WAV file.
 
 void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
 {
-    ma_encoder* pEncoder = (ma_encoder*)pDevice->pUserData;
+    ma_encoder_write_pcm_frames((ma_encoder*)pDevice->pUserData, pInput, frameCount, NULL);
-    MA_ASSERT(pEncoder != NULL);
-
-    ma_encoder_write_pcm_frames(pEncoder, pInput, frameCount, NULL);
 
     (void)pOutput;
 }

examples/simple_duplex.c

@@ -23,8 +23,10 @@ void main_loop__em()
 
 void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
 {
-    MA_ASSERT(pDevice->capture.format == pDevice->playback.format);
+    /* This example assumes the playback and capture sides use the same format and channel count. */
-    MA_ASSERT(pDevice->capture.channels == pDevice->playback.channels);
+    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));

examples/simple_loopback.c

@@ -18,10 +18,7 @@ properties. The output buffer in the callback will be null whereas the input buf
 
 void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
 {
-    ma_encoder* pEncoder = (ma_encoder*)pDevice->pUserData;
+    ma_encoder_write_pcm_frames((ma_encoder*)pDevice->pUserData, pInput, frameCount, NULL);
-    MA_ASSERT(pEncoder != NULL);
-
-    ma_encoder_write_pcm_frames(pEncoder, pInput, frameCount, NULL);
 
     (void)pOutput;
 }

examples/simple_mixing.c

@@ -87,8 +87,7 @@ void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uin
     float* pOutputF32 = (float*)pOutput;
     ma_uint32 iDecoder;
 
-    MA_ASSERT(pDevice->playback.format == SAMPLE_FORMAT);   /* <-- Important for this example. */
+    /* 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);
@@ -107,6 +106,7 @@ void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uin
     }
 
     (void)pInput;
+    (void)pDevice;
 }
 
 int main(int argc, char** argv)

examples/simple_playback_sine.c

@@ -33,14 +33,7 @@ void main_loop__em()
 
 void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
 {
-    ma_waveform* pSineWave;
+    ma_waveform_read_pcm_frames((ma_waveform*)pDevice->pUserData, pOutput, frameCount, NULL);
-
-    MA_ASSERT(pDevice->playback.channels == DEVICE_CHANNELS);
-
-    pSineWave = (ma_waveform*)pDevice->pUserData;
-    MA_ASSERT(pSineWave != NULL);
-
-    ma_waveform_read_pcm_frames(pSineWave, pOutput, frameCount, NULL);
 
     (void)pInput;   /* Unused. */
 }

extras/nodes/ma_delay_node/ma_delay_node_example.c

@@ -15,8 +15,13 @@ static ma_node_graph       g_nodeGraph;         /* The main node graph that we'l
 
 void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
 {
-    MA_ASSERT(pDevice->capture.format == pDevice->playback.format && pDevice->capture.format == ma_format_f32);
+    /*
-    MA_ASSERT(pDevice->capture.channels == pDevice->playback.channels);
+    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 

extras/nodes/ma_reverb_node/ma_reverb_node_example.c

@@ -15,8 +15,13 @@ static ma_node_graph       g_nodeGraph;         /* The main node graph that we'l
 
 void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
 {
-    MA_ASSERT(pDevice->capture.format == pDevice->playback.format && pDevice->capture.format == ma_format_f32);
+    /*
-    MA_ASSERT(pDevice->capture.channels == pDevice->playback.channels);
+    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 

extras/nodes/ma_vocoder_node/ma_vocoder_node_example.c

@@ -24,8 +24,13 @@ static ma_node_graph       g_nodeGraph;
 
 void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
 {
-    MA_ASSERT(pDevice->capture.format == pDevice->playback.format);
+    /*
-    MA_ASSERT(pDevice->capture.channels == pDevice->playback.channels);
+    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