@@ -255,18 +255,12 @@ This example implements two custom decoders:
-
-
-
+Vorbis via libvorbis
-Vorbis via libvorbis
-
-
-Opus via libopus
-
-A
+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.
@@ -358,8 +352,7 @@ The onInitFile, /* onInitFileW() */
NULL, /* onInitMemory() */
- ma_decoding_backend_uninit__libvorbis,
- ma_decoding_backend_get_channel_map__libvorbis
+ ma_decoding_backend_uninit__libvorbis
};
@@ -435,8 +428,7 @@ The onInitFile, /* onInitFileW() */
NULL, /* onInitMemory() */
- ma_decoding_backend_uninit__libopus,
- ma_decoding_backend_get_channel_map__libopus
+ ma_decoding_backend_uninit__libopus
};
@@ -496,7 +488,7 @@ The onInitFile, /* Initialize the device. */
- result = ma_data_source_get_data_format(&decoder, &format, &channels, &sampleRate);
+ 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);
diff --git a/docs/examples/custom_decoder_engine.html b/docs/examples/custom_decoder_engine.html
new file mode 100644
index 00000000..fbe87b6e
--- /dev/null
+++ b/docs/examples/custom_decoder_engine.html
@@ -0,0 +1,496 @@
+
+
+
+ miniaudio - A single file audio playback and capture library.
+
+
+
+
+
+
+
+
+
+
+
+
+
+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).
+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:
+
+
+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.
+
+#define MA_EXPERIMENTAL__DATA_LOOPING_AND_CHAINING
+#define MINIAUDIO_IMPLEMENTATION
+#include"../miniaudio.h"
+
+#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;
+
+staticma_data_source* next_callback_tail(ma_data_source* pDataSource)
+{
+ MA_ASSERT(g_decoderCount > 0); /* <-- We check for this in main() so should never happen. */
+
+ /*
+ 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];
+}
+
+staticvoid data_callback(ma_device* pDevice, void* pOutput, constvoid* 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;
+
+ if (ma_device_init(NULL, &deviceConfig, &device) != MA_SUCCESS) {
+ printf("Failed to open playback device.\n");
+ result = -1;
+ goto done_decoders;
+ }
+
+ if (ma_device_start(&device) != MA_SUCCESS) {
+ printf("Failed to start playback device.\n");
+ result = -1;
+ 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;
+}
+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.
+
+#define MINIAUDIO_IMPLEMENTATION
+#include"../miniaudio.h"
+#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. */
+
+staticma_waveform g_sourceData; /* The underlying data source of the excite node. */
+static ma_audio_buffer_ref g_exciteData; /* The underlying data source of the source 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, constvoid* pInput, ma_uint32 frameCount)
+{
+ MA_ASSERT(pDevice->capture.format == pDevice->playback.format);
+ MA_ASSERT(pDevice->capture.channels == pDevice->playback.channels);
+
+ /*
+ 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;
+}
+
+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.
+
+#define MINIAUDIO_IMPLEMENTATION
+#include"../miniaudio.h"
+
+#define MAX_DEVICES 2
+#define MAX_SOUNDS 32
+
+void data_callback(ma_device* pDevice, void* pOutput, constvoid* 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) {
+ intc = 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 (;;) {
+ intc = 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]);
+ }
+
+ /*
+ 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;
+}
+
+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.
+
+#define MINIAUDIO_IMPLEMENTATION
+#include"../miniaudio.h"
+
+#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;
+}
+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().
+
+#define MINIAUDIO_IMPLEMENTATION
+#include"../miniaudio.h"
+
+/* 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. */
+
+typedefstruct
+{
+ 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;
+staticint g_soundNodeCount;
+
+void data_callback(ma_device* pDevice, void* pOutput, constvoid* 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.
+ */
+ ma_node_graph_read_pcm_frames(&g_nodeGraph, pOutput, frameCount, NULL);
+
+ (void)pInput; /* 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;
+}
+
+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_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. */
+#define MINIAUDIO_IMPLEMENTATION
+#include"../miniaudio.h"
+
+#ifdef __EMSCRIPTEN__
+#include<emscripten.h>
+
+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
+ 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, constvoid* 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;
+}
+
+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. */
+#define MINIAUDIO_IMPLEMENTATION
+#include"../miniaudio.h"
+
+static ma_resource_manager_data_source g_dataSources[16];
+staticma_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).
+*/
+staticma_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().
+ */
+ 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);
+ } 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);
+
+ 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, pFramesOut, framesToRead, &framesJustRead);
+
+ 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;
+ }
+ }
+
+ 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(pFramesOut, totalFramesRead, ma_format_f32, 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, constvoid* 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;
+
+ 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.
+ */
+ /*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);
+ }
+}
+
+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;
+ 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_RESOURCE_MANAGER_JOB_QUIT) {
+ printf("CUSTOM JOB THREAD TERMINATING VIA MA_RESOURCE_MANAGER_JOB_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 < 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;
+}
+
@@ -262,18 +262,13 @@ decoder straight into ma_data_source_read_p
void data_callback(ma_device* pDevice, void* pOutput, constvoid* pInput, ma_uint32 frameCount)
{
- ma_bool32 isLooping = MA_TRUE;
-
ma_decoder* pDecoder = (ma_decoder*)pDevice->pUserData;
if (pDecoder == NULL) {
return;
}
- /*
- A decoder is a data source which means you can seemlessly plug it into the ma_data_source API. We can therefore take advantage
- of the "loop" parameter of ma_data_source_read_pcm_frames() to handle looping for us.
- */
- ma_data_source_read_pcm_frames(pDecoder, pOutput, frameCount, NULL, isLooping);
+ /* 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;
}
@@ -295,6 +290,12 @@ decoder straight into ma_data_source_read_p
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;
diff --git a/docs/examples/simple_mixing.html b/docs/examples/simple_mixing.html
index 4c1a1baa..571d1bc8 100644
--- a/docs/examples/simple_mixing.html
+++ b/docs/examples/simple_mixing.html
@@ -245,7 +245,7 @@ a.doc-navigation-l4 {
-miniaudio is a single file library for audio playback and capture. To use it, do the following in one .c file:
+miniaudio is a single file library for audio playback and capture. To use it, do the following in
+one .c file:
@@ -266,20 +267,59 @@ You can do #include "miniaudio.h"
-miniaudio uses the concept of a "device" as the abstraction for physical devices. The idea is that you choose a physical device to emit or capture audio from,
-and then move data to/from the device when miniaudio tells you to. Data is delivered to and from devices asynchronously via a callback which you specify when
-initializing the device.
+miniaudio includes both low level and high level APIs. The low level API is good for those who want
+to do all of their mixing themselves and only require a light weight interface to the underlying
+audio device. The high level API is good for those who have complex mixing and effect requirements.
-When initializing the device you first need to configure it. The device configuration allows you to specify things like the format of the data delivered via
-the callback, the size of the internal buffer and the ID of the device you want to emit or capture audio from.
+In miniaudio, objects are transparent structures. Unlike many other libraries, there are no handles
+to opaque objects which means you need to allocate memory for objects yourself. In the examples
+presented in this documentation you will often see objects declared on the stack. You need to be
+careful when translating these examples to your own code so that you don't accidentally declare
+your objects on the stack and then cause them to become invalid once the function returns. In
+addition, you must ensure the memory address of your objects remain the same throughout their
+lifetime. You therefore cannot be making copies of your objects.
-Once you have the device configuration set up you can initialize the device. When initializing a device you need to allocate memory for the device object
-beforehand. This gives the application complete control over how the memory is allocated. In the example below we initialize a playback device on the stack,
-but you could allocate it on the heap if that suits your situation better.
+A config/init pattern is used throughout the entire library. The idea is that you set up a config
+object and pass that into the initialization routine. The advantage to this system is that the
+config object can be initialized with logical defaults and new properties added to it without
+breaking the API. The config object can be allocated on the stack and does not need to be
+maintained after initialization of the corresponding object.
+
+
+
+
+
+
+
+
1.1. Low Level API
+
+The low level API gives you access to the raw audio data of an audio device. It supports playback,
+capture, full-duplex and loopback (WASAPI only). You can enumerate over devices to determine which
+physical device(s) you want to connect to.
+
+
+
+The low level API uses the concept of a "device" as the abstraction for physical devices. The idea
+is that you choose a physical device to emit or capture audio from, and then move data to/from the
+device when miniaudio tells you to. Data is delivered to and from devices asynchronously via a
+callback which you specify when initializing the device.
+
+
+
+When initializing the device you first need to configure it. The device configuration allows you to
+specify things like the format of the data delivered via the callback, the size of the internal
+buffer and the ID of the device you want to emit or capture audio from.
+
+
+
+Once you have the device configuration set up you can initialize the device. When initializing a
+device you need to allocate memory for the device object beforehand. This gives the application
+complete control over how the memory is allocated. In the example below we initialize a playback
+device on the stack, but you could allocate it on the heap if that suits your situation better.
@@ -315,22 +355,29 @@ but you could allocate it on the heap if that suits your situation better.
}
-In the example above, data_callback() is where audio data is written and read from the device. The idea is in playback mode you cause sound to be emitted
-from the speakers by writing audio data to the output buffer (pOutput in the example). In capture mode you read data from the input buffer (pInput) to
-extract sound captured by the microphone. The frameCount parameter tells you how many frames can be written to the output buffer and read from the input
-buffer. A "frame" is one sample for each channel. For example, in a stereo stream (2 channels), one frame is 2 samples: one for the left, one for the right.
-The channel count is defined by the device config. The size in bytes of an individual sample is defined by the sample format which is also specified in the
-device config. Multi-channel audio data is always interleaved, which means the samples for each frame are stored next to each other in memory. For example, in
-a stereo stream the first pair of samples will be the left and right samples for the first frame, the second pair of samples will be the left and right samples
-for the second frame, etc.
+In the example above, data_callback() is where audio data is written and read from the device.
+The idea is in playback mode you cause sound to be emitted from the speakers by writing audio data
+to the output buffer (pOutput in the example). In capture mode you read data from the input
+buffer (pInput) to extract sound captured by the microphone. The frameCount parameter tells you
+how many frames can be written to the output buffer and read from the input buffer. A "frame" is
+one sample for each channel. For example, in a stereo stream (2 channels), one frame is 2
+samples: one for the left, one for the right. The channel count is defined by the device config.
+The size in bytes of an individual sample is defined by the sample format which is also specified
+in the device config. Multi-channel audio data is always interleaved, which means the samples for
+each frame are stored next to each other in memory. For example, in a stereo stream the first pair
+of samples will be the left and right samples for the first frame, the second pair of samples will
+be the left and right samples for the second frame, etc.
-The configuration of the device is defined by the ma_device_config structure. The config object is always initialized with ma_device_config_init(). It's
-important to always initialize the config with this function as it initializes it with logical defaults and ensures your program doesn't break when new members
-are added to the ma_device_config structure. The example above uses a fairly simple and standard device configuration. The call to ma_device_config_init()
-takes a single parameter, which is whether or not the device is a playback, capture, duplex or loopback device (loopback devices are not supported on all
-backends). The config.playback.format member sets the sample format which can be one of the following (all formats are native-endian):
+The configuration of the device is defined by the ma_device_config structure. The config object
+is always initialized with ma_device_config_init(). It's important to always initialize the
+config with this function as it initializes it with logical defaults and ensures your program
+doesn't break when new members are added to the ma_device_config structure. The example above
+uses a fairly simple and standard device configuration. The call to ma_device_config_init() takes
+a single parameter, which is whether or not the device is a playback, capture, duplex or loopback
+device (loopback devices are not supported on all backends). The config.playback.format member
+sets the sample format which can be one of the following (all formats are native-endian):
@@ -402,28 +449,36 @@ ma_format_u8
-The config.playback.channels member sets the number of channels to use with the device. The channel count cannot exceed MA_MAX_CHANNELS. The
-config.sampleRate member sets the sample rate (which must be the same for both playback and capture in full-duplex configurations). This is usually set to
-44100 or 48000, but can be set to anything. It's recommended to keep this between 8000 and 384000, however.
+The config.playback.channels member sets the number of channels to use with the device. The
+channel count cannot exceed MA_MAX_CHANNELS. The config.sampleRate member sets the sample rate
+(which must be the same for both playback and capture in full-duplex configurations). This is
+usually set to 44100 or 48000, but can be set to anything. It's recommended to keep this between
+8000 and 384000, however.
-Note that leaving the format, channel count and/or sample rate at their default values will result in the internal device's native configuration being used
-which is useful if you want to avoid the overhead of miniaudio's automatic data conversion.
+Note that leaving the format, channel count and/or sample rate at their default values will result
+in the internal device's native configuration being used which is useful if you want to avoid the
+overhead of miniaudio's automatic data conversion.
-In addition to the sample format, channel count and sample rate, the data callback and user data pointer are also set via the config. The user data pointer is
-not passed into the callback as a parameter, but is instead set to the pUserData member of ma_device which you can access directly since all miniaudio
-structures are transparent.
+In addition to the sample format, channel count and sample rate, the data callback and user data
+pointer are also set via the config. The user data pointer is not passed into the callback as a
+parameter, but is instead set to the pUserData member of ma_device which you can access
+directly since all miniaudio structures are transparent.
-Initializing the device is done with ma_device_init(). This will return a result code telling you what went wrong, if anything. On success it will return
-MA_SUCCESS. After initialization is complete the device will be in a stopped state. To start it, use ma_device_start(). Uninitializing the device will stop
-it, which is what the example above does, but you can also stop the device with ma_device_stop(). To resume the device simply call ma_device_start() again.
-Note that it's important to never stop or start the device from inside the callback. This will result in a deadlock. Instead you set a variable or signal an
-event indicating that the device needs to stop and handle it in a different thread. The following APIs must never be called inside the callback:
+Initializing the device is done with ma_device_init(). This will return a result code telling you
+what went wrong, if anything. On success it will return MA_SUCCESS. After initialization is
+complete the device will be in a stopped state. To start it, use ma_device_start().
+Uninitializing the device will stop it, which is what the example above does, but you can also stop
+the device with ma_device_stop(). To resume the device simply call ma_device_start() again.
+Note that it's important to never stop or start the device from inside the callback. This will
+result in a deadlock. Instead you set a variable or signal an event indicating that the device
+needs to stop and handle it in a different thread. The following APIs must never be called inside
+the callback:
-You must never try uninitializing and reinitializing a device inside the callback. You must also never try to stop and start it from inside the callback. There
-are a few other things you shouldn't do in the callback depending on your requirements, however this isn't so much a thread-safety thing, but rather a
-real-time processing thing which is beyond the scope of this introduction.
+You must never try uninitializing and reinitializing a device inside the callback. You must also
+never try to stop and start it from inside the callback. There are a few other things you shouldn't
+do in the callback depending on your requirements, however this isn't so much a thread-safety
+thing, but rather a real-time processing thing which is beyond the scope of this introduction.
-The example above demonstrates the initialization of a playback device, but it works exactly the same for capture. All you need to do is change the device type
-from ma_device_type_playback to ma_device_type_capture when setting up the config, like so:
+The example above demonstrates the initialization of a playback device, but it works exactly the
+same for capture. All you need to do is change the device type from ma_device_type_playback to
+ma_device_type_capture when setting up the config, like so:
-In the data callback you just read from the input buffer (pInput in the example above) and leave the output buffer alone (it will be set to NULL when the
-device type is set to ma_device_type_capture).
+In the data callback you just read from the input buffer (pInput in the example above) and leave
+the output buffer alone (it will be set to NULL when the device type is set to
+ma_device_type_capture).
-You will notice in the example above that the sample format and channel count is specified separately for playback and capture. This is to support different
-data formats between the playback and capture devices in a full-duplex system. An example may be that you want to capture audio data as a monaural stream (one
-channel), but output sound to a stereo speaker system. Note that if you use different formats between playback and capture in a full-duplex configuration you
-will need to convert the data yourself. There are functions available to help you do this which will be explained later.
+You will notice in the example above that the sample format and channel count is specified
+separately for playback and capture. This is to support different data formats between the playback
+and capture devices in a full-duplex system. An example may be that you want to capture audio data
+as a monaural stream (one channel), but output sound to a stereo speaker system. Note that if you
+use different formats between playback and capture in a full-duplex configuration you will need to
+convert the data yourself. There are functions available to help you do this which will be
+explained later.
-The example above did not specify a physical device to connect to which means it will use the operating system's default device. If you have multiple physical
-devices connected and you want to use a specific one you will need to specify the device ID in the configuration, like so:
+The example above did not specify a physical device to connect to which means it will use the
+operating system's default device. If you have multiple physical devices connected and you want to
+use a specific one you will need to specify the device ID in the configuration, like so:
@@ -523,10 +585,12 @@ config.playback.pDeviceID = pMyPlaybackDeviceID;
config.capture.pDeviceID = pMyCaptureDeviceID; // Only if requesting a capture, duplex or loopback device.
-To retrieve the device ID you will need to perform device enumeration, however this requires the use of a new concept called the "context". Conceptually
-speaking the context sits above the device. There is one context to many devices. The purpose of the context is to represent the backend at a more global level
-and to perform operations outside the scope of an individual device. Mainly it is used for performing run-time linking against backend libraries, initializing
-backends and enumerating devices. The example below shows how to enumerate devices.
+To retrieve the device ID you will need to perform device enumeration, however this requires the
+use of a new concept called the "context". Conceptually speaking the context sits above the device.
+There is one context to many devices. The purpose of the context is to represent the backend at a
+more global level and to perform operations outside the scope of an individual device. Mainly it is
+used for performing run-time linking against backend libraries, initializing backends and
+enumerating devices. The example below shows how to enumerate devices.
-The first thing we do in this example is initialize a ma_context object with ma_context_init(). The first parameter is a pointer to a list of ma_backend
-values which are used to override the default backend priorities. When this is NULL, as in this example, miniaudio's default priorities are used. The second
-parameter is the number of backends listed in the array pointed to by the first parameter. The third parameter is a pointer to a ma_context_config object
-which can be NULL, in which case defaults are used. The context configuration is used for setting the logging callback, custom memory allocation callbacks,
-user-defined data and some backend-specific configurations.
+The first thing we do in this example is initialize a ma_context object with ma_context_init().
+The first parameter is a pointer to a list of ma_backend values which are used to override the
+default backend priorities. When this is NULL, as in this example, miniaudio's default priorities
+are used. The second parameter is the number of backends listed in the array pointed to by the
+first parameter. The third parameter is a pointer to a ma_context_config object which can be
+NULL, in which case defaults are used. The context configuration is used for setting the logging
+callback, custom memory allocation callbacks, user-defined data and some backend-specific
+configurations.
-Once the context has been initialized you can enumerate devices. In the example above we use the simpler ma_context_get_devices(), however you can also use a
-callback for handling devices by using ma_context_enumerate_devices(). When using ma_context_get_devices() you provide a pointer to a pointer that will,
-upon output, be set to a pointer to a buffer containing a list of ma_device_info structures. You also provide a pointer to an unsigned integer that will
-receive the number of items in the returned buffer. Do not free the returned buffers as their memory is managed internally by miniaudio.
+Once the context has been initialized you can enumerate devices. In the example above we use the
+simpler ma_context_get_devices(), however you can also use a callback for handling devices by
+using ma_context_enumerate_devices(). When using ma_context_get_devices() you provide a pointer
+to a pointer that will, upon output, be set to a pointer to a buffer containing a list of
+ma_device_info structures. You also provide a pointer to an unsigned integer that will receive
+the number of items in the returned buffer. Do not free the returned buffers as their memory is
+managed internally by miniaudio.
-The ma_device_info structure contains an id member which is the ID you pass to the device config. It also contains the name of the device which is useful
-for presenting a list of devices to the user via the UI.
+The ma_device_info structure contains an id member which is the ID you pass to the device
+config. It also contains the name of the device which is useful for presenting a list of devices
+to the user via the UI.
-When creating your own context you will want to pass it to ma_device_init() when initializing the device. Passing in NULL, like we do in the first example,
-will result in miniaudio creating the context for you, which you don't want to do since you've already created a context. Note that internally the context is
-only tracked by it's pointer which means you must not change the location of the ma_context object. If this is an issue, consider using malloc() to
-allocate memory for the context.
+When creating your own context you will want to pass it to ma_device_init() when initializing the
+device. Passing in NULL, like we do in the first example, will result in miniaudio creating the
+context for you, which you don't want to do since you've already created a context. Note that
+internally the context is only tracked by it's pointer which means you must not change the location
+of the ma_context object. If this is an issue, consider using malloc() to allocate memory for
+the context.
+
+
+
+
+
+
+
+
1.2. High Level API
+
+The high level API consists of three main parts:
+
+
+
+Resource management for loading and streaming sounds.
+
+A node graph for advanced mixing and effect processing.
+
+A high level "engine" that wraps around the resource manager and node graph.
+
+
+
+The resource manager (ma_resource_manager) is used for loading sounds. It supports loading sounds
+fully into memory and also streaming. It will also deal with reference counting for you which
+avoids the same sound being loaded multiple times.
+
+
+
+The node graph is used for mixing and effect processing. The idea is that you connect a number of
+nodes into the graph by connecting each node's outputs to another node's inputs. Each node can
+implement it's own effect. By chaining nodes together, advanced mixing and effect processing can
+be achieved.
+
+
+
+The engine encapsulates both the resource manager and the node graph to create a simple, easy to
+use high level API. The resource manager and node graph APIs are covered in more later sections of
+this manual.
+
+
+
+The code below shows how you can initialize an engine using it's default configuration.
+
+
+This creates an engine instance which will initialize a device internally which you can access with
+ma_engine_get_device(). It will also initialize a resource manager for you which can be accessed
+with ma_engine_get_resource_manager(). The engine itself is a node graph (ma_node_graph) which
+means you can pass a pointer to the engine object into any of the ma_node_graph APIs (with a
+cast). Alternatively, you can use ma_engine_get_node_graph() instead of a cast.
+
+
+
+Note that all objects in miniaudio, including the ma_engine object in the example above, are
+transparent structures. There are no handles to opaque structures in miniaudio which means you need
+to be mindful of how you declare them. In the example above we are declaring it on the stack, but
+this will result in the struct being invalidated once the function encapsulating it returns. If
+allocating the engine on the heap is more appropriate, you can easily do so with a standard call
+to malloc() or whatever heap allocation routine you like:
+
+
+
+
+
+ma_engine* pEngine = malloc(sizeof(*pEngine));
+
+
+The ma_engine API uses the same config/init pattern used all throughout miniaudio. To configure
+an engine, you can fill out a ma_engine_config object and pass it into the first parameter of
+ma_engine_init():
+
+
+This creates an engine instance using a custom config. In this particular example it's showing how
+you can specify a custom resource manager rather than having the engine initialize one internally.
+This is particularly useful if you want to have multiple engine's share the same resource manager.
+
+
+
+The engine must be uninitialized with ma_engine_uninit() when it's no longer needed.
+
+
+
+By default the engine will be started, but nothing will be playing because no sounds have been
+initialized. The easiest but least flexible way of playing a sound is like so:
+
+
+This plays what miniaudio calls an "inline" sound. It plays the sound once, and then puts the
+internal sound up for recycling. The last parameter is used to specify which sound group the sound
+should be associated with which will be explained later. This particular way of playing a sound is
+simple, but lacks flexibility and features. A more flexible way of playing a sound is to first
+initialize a sound:
+
+
+This returns a ma_sound object which represents a single instance of the specified sound file. If
+you want to play the same file multiple times simultaneously, you need to create one sound for each
+instance.
+
+
+
+Sounds should be uninitialized with ma_sound_uninit().
+
+
+
+Sounds are not started by default. Start a sound with ma_sound_start() and stop it with
+ma_sound_stop(). When a sound is stopped, it is not rewound to the start. Use
+ma_sound_seek_to_pcm_frames(&sound, 0) to seek back to the start of a sound. By default, starting
+and stopping sounds happens immediately, but sometimes it might be convenient to schedule the sound
+the be started and/or stopped at a specific time. This can be done with the following functions:
+
+
+The start/stop time needs to be specified based on the absolute timer which is controlled by the
+engine. The current global time time in PCM frames can be retrieved with ma_engine_get_time().
+The engine's global time can be changed with ma_engine_set_time() for synchronization purposes if
+required.
+
+
+
+The third parameter of ma_sound_init_from_file() is a set of flags that control how the sound be
+loaded and a few options on which features should be enabled for that sound. By default, the sound
+is synchronously loaded fully into memory straight from the file system without any kind of
+decoding. If you want to decode the sound before storing it in memory, you need to specify the
+MA_SOUND_FLAG_DECODE flag. This is useful if you want to incur the cost of decoding at an earlier
+stage, such as a loading stage. Without this option, decoding will happen dynamically at mixing
+time which might be too expensive on the audio thread.
+
+
+
+If you want to load the sound asynchronously, you can specify the MA_SOUND_FLAG_ASYNC flag. This
+will result in ma_sound_init_from_file() returning quickly, but the sound will not start playing
+until the sound has had some audio decoded.
+
+
+
+The fourth parameter is a pointer to sound group. A sound group is used as a mechanism to organise
+sounds into groups which have their own effect processing and volume control. An example is a game
+which might have separate groups for sfx, voice and music. Each of these groups have their own
+independent volume control. Use ma_sound_group_init() or ma_sound_group_init_ex() to initialize
+a sound group.
+
+
+
+Sounds and sound groups are nodes in the engine's node graph and can be plugged into any ma_node
+API. This makes it possible to connect sounds and sound groups to effect nodes to produce complex
+effect chains.
+
+
+
+A sound can have it's volume changed with ma_sound_set_volume(). If you prefer decibel volume
+control you can use ma_volume_db_to_linear() to convert from decibel representation to linear.
+
+
+
+Panning and pitching is supported with ma_sound_set_pan() and ma_sound_set_pitch(). If you know
+a sound will never have it's pitch changed with ma_sound_set_pitch() or via the doppler effect,
+you can specify the MA_SOUND_FLAG_NO_PITCH flag when initializing the sound for an optimization.
+
+
+
+By default, sounds and sound groups have spatialization enabled. If you don't ever want to
+spatialize your sounds, initialize the sound with the MA_SOUND_FLAG_NO_SPATIALIZATION flag. The
+spatialization model is fairly simple and is roughly on feature parity with OpenAL. HRTF and
+environmental occlusion are not currently supported, but planned for the future. The supported
+features include:
+
+
+
+Sound and listener positioning and orientation with cones
+
+Attenuation models: none, inverse, linear and exponential
+
+Doppler effect
+
+
+
+Sounds can be faded in and out with ma_sound_set_fade_in_pcm_frames().
+
+
+
+To check if a sound is currently playing, you can use ma_sound_is_playing(). To check if a sound
+is at the end, use ma_sound_at_end(). Looping of a sound can be controlled with
+ma_sound_set_looping(). Use ma_sound_is_looping() to check whether or not the sound is looping.
@@ -606,7 +915,8 @@ allocate memory for the context.
2. Building
-miniaudio should work cleanly out of the box without the need to download or install any dependencies. See below for platform-specific details.
+miniaudio should work cleanly out of the box without the need to download or install any
+dependencies. See below for platform-specific details.
@@ -616,21 +926,32 @@ miniaudio should work cleanly out of the box without the need to download or ins
2.1. Windows
-The Windows build should compile cleanly on all popular compilers without the need to configure any include paths nor link to any libraries.
+The Windows build should compile cleanly on all popular compilers without the need to configure any
+include paths nor link to any libraries.
+
+
+
+The UWP build may require linking to mmdevapi.lib if you get errors about an unresolved external
+symbol for ActivateAudioInterfaceAsync().
+
+
+
2.2. macOS and iOS
-The macOS build should compile cleanly without the need to download any dependencies nor link to any libraries or frameworks. The iOS build needs to be
-compiled as Objective-C and will need to link the relevant frameworks but should compile cleanly out of the box with Xcode. Compiling through the command line
-requires linking to -lpthread and -lm.
+The macOS build should compile cleanly without the need to download any dependencies nor link to
+any libraries or frameworks. The iOS build needs to be compiled as Objective-C and will need to
+link the relevant frameworks but should compile cleanly out of the box with Xcode. Compiling
+through the command line requires linking to -lpthread and -lm.
-Due to the way miniaudio links to frameworks at runtime, your application may not pass Apple's notarization process. To fix this there are two options. The
-first is to use the MA_NO_RUNTIME_LINKING option, like so:
+Due to the way miniaudio links to frameworks at runtime, your application may not pass Apple's
+notarization process. To fix this there are two options. The first is to use the
+MA_NO_RUNTIME_LINKING option, like so:
@@ -643,8 +964,9 @@ first is to use the MA_NO_RUNTIME_LINKING#include"miniaudio.h"
-This will require linking with -framework CoreFoundation -framework CoreAudio -framework AudioUnit. Alternatively, if you would rather keep using runtime
-linking you can add the following to your entitlements.xcent file:
+This will require linking with -framework CoreFoundation -framework CoreAudio -framework AudioUnit.
+Alternatively, if you would rather keep using runtime linking you can add the following to your
+entitlements.xcent file:
@@ -662,34 +984,49 @@ linking you can add the following to your entitlements.xcent file:
2.3. Linux
-The Linux build only requires linking to -ldl, -lpthread and -lm. You do not need any development packages.
+The Linux build only requires linking to -ldl, -lpthread and -lm. You do not need any
+development packages. You may need to link with -latomic if you're compiling for 32-bit ARM.
+
+
+
2.4. BSD
-The BSD build only requires linking to -lpthread and -lm. NetBSD uses audio(4), OpenBSD uses sndio and FreeBSD uses OSS.
+The BSD build only requires linking to -lpthread and -lm. NetBSD uses audio(4), OpenBSD uses
+sndio and FreeBSD uses OSS. You may need to link with -latomic if you're compiling for 32-bit
+ARM.
+
+
+
2.5. Android
-AAudio is the highest priority backend on Android. This should work out of the box without needing any kind of compiler configuration. Support for AAudio
-starts with Android 8 which means older versions will fall back to OpenSL|ES which requires API level 16+.
+AAudio is the highest priority backend on Android. This should work out of the box without needing
+any kind of compiler configuration. Support for AAudio starts with Android 8 which means older
+versions will fall back to OpenSL|ES which requires API level 16+.
+
+
+
+There have been reports that the OpenSL|ES backend fails to initialize on some Android based
+devices due to dlopen() failing to open "libOpenSLES.so". If this happens on your platform
+you'll need to disable run-time linking with MA_NO_RUNTIME_LINKING and link with -lOpenSLES.
-There have been reports that the OpenSL|ES backend fails to initialize on some Android based devices due to dlopen() failing to open "libOpenSLES.so". If
-this happens on your platform you'll need to disable run-time linking with MA_NO_RUNTIME_LINKING and link with -lOpenSLES.
2.6. Emscripten
-The Emscripten build emits Web Audio JavaScript directly and should compile cleanly out of the box. You cannot use -std=c* compiler flags, nor -ansi.
+The Emscripten build emits Web Audio JavaScript directly and should compile cleanly out of the box.
+You cannot use -std=c* compiler flags, nor -ansi.
@@ -1021,9 +1358,9 @@ MA_NO_DEVICE_IO
-Disables playback and recording. This will disable ma_context and
-ma_device APIs. This is useful if you only want to use miniaudio's
-data conversion and/or decoding APIs.
+Disables playback and recording. This will disable ma_context
+and ma_device APIs. This is useful if you only want to use
+miniaudio's data conversion and/or decoding APIs.
@@ -1045,11 +1382,11 @@ MA_NO_THREADING
-Disables the ma_thread, ma_mutex, ma_semaphore and ma_event APIs.
-This option is useful if you only need to use miniaudio for data
-conversion, decoding and/or encoding. Some families of APIs
-require threading which means the following options must also be
-set:
+Disables the ma_thread, ma_mutex, ma_semaphore and
+ma_event APIs. This option is useful if you only need to use
+miniaudio for data conversion, decoding and/or encoding. Some
+families of APIsrequire threading which means the following
+options must also be set:
-Enable processing of MA_LOG_LEVEL_DEBUG messages and printf()
+Enable processing of MA_LOG_LEVEL_DEBUG messages and printf()
output.
@@ -1163,11 +1492,11 @@ MA_DLL
-If set, configures MA_API to either import or export APIs
+If set, configures MA_API to either import or export APIs
depending on whether or not the implementation is being defined.
-If defining the implementation, MA_API will be configured to
+If defining the implementation, MA_API will be configured to
export. Otherwise it will be configured to import. This has no
-effect if MA_API is defined externally.
+effect if MA_API is defined externally.
@@ -1175,41 +1504,48 @@ effect if MA_API is defined externally.
3. Definitions
-This section defines common terms used throughout miniaudio. Unfortunately there is often ambiguity in the use of terms throughout the audio space, so this
-section is intended to clarify how miniaudio uses each term.
+This section defines common terms used throughout miniaudio. Unfortunately there is often ambiguity
+in the use of terms throughout the audio space, so this section is intended to clarify how miniaudio
+uses each term.
3.1. Sample
-A sample is a single unit of audio data. If the sample format is f32, then one sample is one 32-bit floating point number.
+A sample is a single unit of audio data. If the sample format is f32, then one sample is one 32-bit
+floating point number.
3.2. Frame / PCM Frame
-A frame is a group of samples equal to the number of channels. For a stereo stream a frame is 2 samples, a mono frame is 1 sample, a 5.1 surround sound frame
-is 6 samples, etc. The terms "frame" and "PCM frame" are the same thing in miniaudio. Note that this is different to a compressed frame. If ever miniaudio
-needs to refer to a compressed frame, such as a FLAC frame, it will always clarify what it's referring to with something like "FLAC frame".
+A frame is a group of samples equal to the number of channels. For a stereo stream a frame is 2
+samples, a mono frame is 1 sample, a 5.1 surround sound frame is 6 samples, etc. The terms "frame"
+and "PCM frame" are the same thing in miniaudio. Note that this is different to a compressed frame.
+If ever miniaudio needs to refer to a compressed frame, such as a FLAC frame, it will always
+clarify what it's referring to with something like "FLAC frame".
3.3. Channel
-A stream of monaural audio that is emitted from an individual speaker in a speaker system, or received from an individual microphone in a microphone system. A
-stereo stream has two channels (a left channel, and a right channel), a 5.1 surround sound system has 6 channels, etc. Some audio systems refer to a channel as
-a complex audio stream that's mixed with other channels to produce the final mix - this is completely different to miniaudio's use of the term "channel" and
-should not be confused.
+A stream of monaural audio that is emitted from an individual speaker in a speaker system, or
+received from an individual microphone in a microphone system. A stereo stream has two channels (a
+left channel, and a right channel), a 5.1 surround sound system has 6 channels, etc. Some audio
+systems refer to a channel as a complex audio stream that's mixed with other channels to produce
+the final mix - this is completely different to miniaudio's use of the term "channel" and should
+not be confused.
3.4. Sample Rate
-The sample rate in miniaudio is always expressed in Hz, such as 44100, 48000, etc. It's the number of PCM frames that are processed per second.
+The sample rate in miniaudio is always expressed in Hz, such as 44100, 48000, etc. It's the number
+of PCM frames that are processed per second.
@@ -1299,10 +1635,2234 @@ All formats are native-endian.
-
4. Decoding
+
4. Data Sources
-The ma_decoder API is used for reading audio files. Decoders are completely decoupled from devices and can be used independently. The following formats are
-supported:
+The data source abstraction in miniaudio is used for retrieving audio data from some source. A few
+examples include ma_decoder, ma_noise and ma_waveform. You will need to be familiar with data
+sources in order to make sense of some of the higher level concepts in miniaudio.
+
+
+
+The ma_data_source API is a generic interface for reading from a data source. Any object that
+implements the data source interface can be plugged into any ma_data_source function.
+
+
+
+To read data from a data source:
+
+
+
+
+
+ma_result result;
+ma_uint64 framesRead;
+
+result = ma_data_source_read_pcm_frames(pDataSource, pFramesOut, frameCount, &framesRead, loop);
+if (result != MA_SUCCESS) {
+ return result; // Failed to read data from the data source.
+}
+
+
+If you don't need the number of frames that were successfully read you can pass in NULL to the
+pFramesRead parameter. If this returns a value less than the number of frames requested it means
+the end of the file has been reached. MA_AT_END will be returned only when the number of frames
+read is 0.
+
+
+
+When calling any data source function, with the exception of ma_data_source_init() and
+ma_data_source_uninit(), you can pass in any object that implements a data source. For example,
+you could plug in a decoder like so:
+
+
+
+
+
+ma_result result;
+ma_uint64 framesRead;
+ma_decoder decoder; // <-- This would be initialized with ma_decoder_init_*().
+
+result = ma_data_source_read_pcm_frames(&decoder, pFramesOut, frameCount, &framesRead, loop);
+if (result != MA_SUCCESS) {
+ return result; // Failed to read data from the decoder.
+}
+
+
+If you want to seek forward you can pass in NULL to the pFramesOut parameter. Alternatively you
+can use ma_data_source_seek_pcm_frames().
+
+
+You can retrieve the total length of a data source in PCM frames, but note that some data sources
+may not have the notion of a length, such as noise and waveforms, and others may just not have a
+way of determining the length such as some decoders. To retrieve the length:
+
+
+Care should be taken when retrieving the length of a data source where the underlying decoder is
+pulling data from a data stream with an undefined length, such as internet radio or some kind of
+broadcast. If you do this, ma_data_source_get_length_in_pcm_frames() may never return.
+
+
+
+The current position of the cursor in PCM frames can also be retrieved:
+
+
+If you do not need a specific data format property, just pass in NULL to the respective parameter.
+
+
+
+There may be cases where you want to implement something like a sound bank where you only want to
+read data within a certain range of the underlying data. To do this you can use a range:
+
+
+
+
+
+result = ma_data_source_set_range_in_pcm_frames(pDataSource, rangeBegInFrames, rangeEndInFrames);
+if (result != MA_SUCCESS) {
+ return result; // Failed to set the range.
+}
+
+
+This is useful if you have a sound bank where many sounds are stored in the same file and you want
+the data source to only play one of those sub-sounds.
+
+
+
+Custom loop points can also be used with data sources. By default, data sources will loop after
+they reach the end of the data source, but if you need to loop at a specific location, you can do
+the following:
+
+
+
+
+
+result = ma_data_set_loop_point_in_pcm_frames(pDataSource, loopBegInFrames, loopEndInFrames);
+if (result != MA_SUCCESS) {
+ return result; // Failed to set the loop point.
+}
+
+
+The loop point is relative to the current range.
+
+
+
+It's sometimes useful to chain data sources together so that a seamless transition can be achieved.
+To do this, you can use chaining:
+
+
+
+
+
+ma_decoder decoder1;
+ma_decoder decoder2;
+
+// ... initialize decoders with ma_decoder_init_*() ...
+
+result = ma_data_source_set_next(&decoder1, &decoder2);
+if (result != MA_SUCCESS) {
+ return result; // Failed to set the next data source.
+}
+
+result = ma_data_source_read_pcm_frames(&decoder1, pFramesOut, frameCount, pFramesRead, MA_FALSE);
+if (result != MA_SUCCESS) {
+ return result; // Failed to read from the decoder.
+}
+
+
+In the example above we're using decoders. When reading from a chain, you always want to read from
+the top level data source in the chain. In the example above, decoder1 is the top level data
+source in the chain. When decoder1 reaches the end, decoder2 will start seamlessly without any
+gaps.
+
+
+
+Note that the loop parameter is set to false in the example above. When this is set to true, only
+the current data source will be looped. You can loop the entire chain by linking in a loop like so:
+
+
+
+
+
+ma_data_source_set_next(&decoder1, &decoder2); // decoder1 -> decoder2
+ma_data_source_set_next(&decoder2, &decoder1); // decoder2 -> decoder1 (loop back to the start).
+
+
+Note that setting up chaining is not thread safe, so care needs to be taken if you're dynamically
+changing links while the audio thread is in the middle of reading.
+
+
+
+Do not use ma_decoder_seek_to_pcm_frame() as a means to reuse a data source to play multiple
+instances of the same sound simultaneously. Instead, initialize multiple data sources for each
+instance. This can be extremely inefficient depending on the data source and can result in
+glitching due to subtle changes to the state of internal filters.
+
+
+
+
+
+
+
+
4.1. Custom Data Sources
+
+You can implement a custom data source by implementing the functions in ma_data_source_vtable.
+Your custom object must have ma_data_source_base as it's first member:
+
+
+In your initialization routine, you need to call ma_data_source_init() in order to set up the
+base object (ma_data_source_base):
+
+
+
+
+
+staticma_result my_data_source_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
+{
+ // Read data here. Output in the same format returned by my_data_source_get_data_format().
+}
+
+staticma_result my_data_source_seek(ma_data_source* pDataSource, ma_uint64 frameIndex)
+{
+ // Seek to a specific PCM frame here. Return MA_NOT_IMPLEMENTED if seeking is not supported.
+}
+
+staticma_result my_data_source_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
+{
+ // Return the format of the data here.
+}
+
+staticma_result my_data_source_get_cursor(ma_data_source* pDataSource, ma_uint64* pCursor)
+{
+ // Retrieve the current position of the cursor here. Return MA_NOT_IMPLEMENTED and set *pCursor to 0 if there is no notion of a cursor.
+}
+
+staticma_result my_data_source_get_length(ma_data_source* pDataSource, ma_uint64* pLength)
+{
+ // Retrieve the length in PCM frames here. Return MA_NOT_IMPLEMENTED and set *pLength to 0 if there is no notion of a length or if the length is unknown.
+}
+
+static g_my_data_source_vtable =
+{
+ my_data_source_read,
+ my_data_source_seek,
+ my_data_source_get_data_format,
+ my_data_source_get_cursor,
+ my_data_source_get_length
+};
+
+ma_result my_data_source_init(my_data_source* pMyDataSource)
+{
+ ma_result result;
+ ma_data_source_config baseConfig;
+
+ baseConfig = ma_data_source_config_init();
+ baseConfig.vtable = &g_my_data_source_vtable;
+
+ result = ma_data_source_init(&baseConfig, &pMyDataSource->base);
+ if (result != MA_SUCCESS) {
+ return result;
+ }
+
+ // ... do the initialization of your custom data source here ...
+
+ return MA_SUCCESS;
+}
+
+void my_data_source_uninit(my_data_source* pMyDataSource)
+{
+ // ... do the uninitialization of your custom data source here ...
+
+ // You must uninitialize the base data source.
+ ma_data_source_uninit(&pMyDataSource->base);
+}
+
+
+Note that ma_data_source_init() and ma_data_source_uninit() are never called directly outside
+of the custom data source. It's up to the custom data source itself to call these within their own
+init/uninit functions.
+
+
+
+
+
+
+
+
+
+
+
5. Engine
+
+The ma_engine API is a high level API for managing and mixing sounds and effect processing. The
+ma_engine object encapsulates a resource manager and a node graph, both of which will be
+explained in more detail later.
+
+
+
+Sounds are called ma_sound and are created from an engine. Sounds can be associated with a mixing
+group called ma_sound_group which are also created from the engine. Both ma_sound and
+ma_sound_group objects are nodes within the engine's node graph.
+
+
+
+When the engine is initialized, it will normally create a device internally. If you would rather
+manage the device yourself, you can do so and just pass a pointer to it via the engine config when
+you initialize the engine. You can also just use the engine without a device, which again can be
+configured via the engine config.
+
+
+
+The most basic way to initialize the engine is with a default config, like so:
+
+
+This will result in the engine initializing a playback device using the operating system's default
+device. This will be sufficient for many use cases, but if you need more flexibility you'll want to
+configure the engine with an engine config:
+
+
+In the example above we're passing in a pre-initialized device. Since the caller is the one in
+control of the device's data callback, it's their responsibility to manually call
+ma_engine_read_pcm_frames() from inside their data callback:
+
+
+You can also use the engine independent of a device entirely:
+
+
+
+
+
+ma_result result;
+ma_engine engine;
+ma_engine_config engineConfig;
+
+engineConfig = ma_engine_config_init();
+engineConfig.noDevice = MA_TRUE;
+engineConfig.channels = 2; // Must be set when not using a device.
+engineConfig.sampleRate = 48000; // Must be set when not using a device.
+
+result = ma_engine_init(&engineConfig, &engine);
+if (result != MA_SUCCESS) {
+ return result; // Failed to initialize the engine.
+}
+
+
+Note that when you're not using a device, you must set the channel count and sample rate in the
+config or else miniaudio won't know what to use (miniaudio will use the device to determine this
+normally). When not using a device, you need to use ma_engine_read_pcm_frames() to process audio
+data from the engine. This kind of setup is useful if you want to do something like offline
+processing.
+
+
+
+When a sound is loaded it goes through a resource manager. By default the engine will initialize a
+resource manager internally, but you can also specify a pre-initialized resource manager:
+
+
+In this example we are initializing two engines, both of which are sharing the same resource
+manager. This is especially useful for saving memory when loading the same file across multiple
+engines. If you were not to use a shared resource manager, each engine instance would use their own
+which would result in any sounds that are used between both engine's being loaded twice. By using
+a shared resource manager, it would only be loaded once. Using multiple engine's is useful when you
+need to output to multiple playback devices, such as in a local multiplayer game where each player
+is using their own set of headphones.
+
+
+
+By default an engine will be in a started state. To make it so the engine is not automatically
+started you can configure it as such:
+
+
+
+
+
+engineConfig.noAutoStart = MA_TRUE;
+
+// The engine will need to be started manually.
+ma_engine_start(&engine);
+
+// Later on the engine can be stopped with ma_engine_stop().
+ma_engine_stop(&engine);
+
+
+The concept of starting or stopping an engine is only relevant when using the engine with a
+device. Attempting to start or stop an engine that is not associated with a device will result in
+MA_INVALID_OPERATION.
+
+
+
+The master volume of the engine can be controlled with ma_engine_set_volume() which takes a
+linear scale, with 0 resulting in silence and anything above 1 resulting in amplification. If you
+prefer decibel based volume control, use ma_volume_db_to_linear() to convert from dB to linear.
+
+
+
+When a sound is spatialized, it is done so relative to a listener. An engine can be configured to
+have multiple listeners which can be configured via the config:
+
+
+
+
+
+engineConfig.listenerCount = 2;
+
+
+The maximum number of listeners is restricted to MA_ENGINE_MAX_LISTENERS. By default, when a
+sound is spatialized, it will be done so relative to the closest listener. You can also pin a sound
+to a specific listener which will be explained later. Listener's have a position, direction, cone,
+and velocity (for doppler effect). A listener is referenced by an index, the meaning of which is up
+to the caller (the index is 0 based and cannot go beyond the listener count, minus 1). The
+position, direction and velocity are all specified in absolute terms:
+
+
+The engine supports directional attenuation. The listener can have a cone the controls how sound is
+attenuated based on the listener's direction. When a sound is between the inner and outer cones, it
+will be attenuated between 1 and the cone's outer gain:
+
+
+When a sound is inside the inner code, no directional attenuation is applied. When the sound is
+outside of the outer cone, the attenuation will be set to outerGain in the example above. When
+the sound is in between the inner and outer cones, the attenuation will be interpolated between 1
+and the outer gain.
+
+
+
+The engine's coordinate system follows the OpenGL coordinate system where positive X points right,
+positive Y points up and negative Z points forward.
+
+
+
+The simplest and least flexible way to play a sound is like so:
+
+
+This is a "fire and forget" style of function. The engine will manage the ma_sound object
+internally. When the sound finishes playing, it'll be put up for recycling. For more flexibility
+you'll want to initialize a sound object:
+
+
+Sounds need to be uninitialized with ma_sound_uninit().
+
+
+
+The example above loads a sound from a file. If the resource manager has been disabled you will not
+be able to use this function and instead you'll need to initialize a sound directly from a data
+source:
+
+
+Each ma_sound object represents a single instance of the sound. If you want to play the same
+sound multiple times at the same time, you need to initialize a separate ma_sound object.
+
+
+
+For the most flexibility when initializing sounds, use ma_sound_init_ex(). This uses miniaudio's
+standard config/init pattern:
+
+
+
+
+
+ma_sound sound;
+ma_sound_config soundConfig;
+
+soundConfig = ma_sound_config_init();
+soundConfig.pFilePath = NULL; // Set this to load from a file path.
+soundConfig.pDataSource = NULL; // Set this to initialize from an existing data source.
+soundConfig.pInitialAttachment = &someNodeInTheNodeGraph;
+soundConfig.initialAttachmentInputBusIndex = 0;
+soundConfig.channelsIn = 1;
+soundConfig.channelsOut = 0; // Set to 0 to use the engine's native channel count.
+
+result = ma_sound_init_ex(&soundConfig, &sound);
+if (result != MA_SUCCESS) {
+ return result;
+}
+
+
+In the example above, the sound is being initialized without a file nor a data source. This is
+valid, in which case the sound acts as a node in the middle of the node graph. This means you can
+connect other sounds to this sound and allow it to act like a sound group. Indeed, this is exactly
+what a ma_sound_group is.
+
+
+
+When loading a sound, you specify a set of flags that control how the sound is loaded and what
+features are enabled for that sound. When no flags are set, the sound will be fully loaded into
+memory in exactly the same format as how it's stored on the file system. The resource manager will
+allocate a block of memory and then load the file directly into it. When reading audio data, it
+will be decoded dynamically on the fly. In order to save processing time on the audio thread, it
+might be beneficial to pre-decode the sound. You can do this with the MA_SOUND_FLAG_DECODE flag:
+
+
+By default, sounds will be loaded synchronously, meaning ma_sound_init_*() will not return until
+the sound has been fully loaded. If this is prohibitive you can instead load sounds asynchronously
+by specificying the MA_SOUND_FLAG_ASYNC flag:
+
+
+This will result in ma_sound_init_*() returning quickly, but the sound won't yet have been fully
+loaded. When you start the sound, it won't output anything until some sound is available. The sound
+will start outputting audio before the sound has been fully decoded when the MA_SOUND_FLAG_DECODE
+is specified.
+
+
+
+If you need to wait for an asynchronously loaded sound to be fully loaded, you can use a fence. A
+fence in miniaudio is a simple synchronization mechanism which simply blocks until it's internal
+counter hit's zero. You can specify a fence like so:
+
+
+
+
+
+ma_result result;
+ma_fence fence;
+ma_sound sounds[4];
+
+result = ma_fence_init(&fence);
+if (result != MA_SUCCES) {
+ return result;
+}
+
+// Load some sounds asynchronously.
+for (int iSound = 0; iSound < 4; iSound += 1) {
+ ma_sound_init_from_file(&engine, mySoundFilesPaths[iSound], MA_SOUND_FLAG_DECODE | MA_SOUND_FLAG_ASYNC, pGroup, &fence, &sounds[iSound]);
+}
+
+// ... do some other stuff here in the mean time ...
+
+// Wait for all sounds to finish loading.
+ma_fence_wait(&fence);
+
+
+If loading the entire sound into memory is prohibitive, you can also configure the engine to stream
+the audio data:
+
+
+When streaming sounds, 2 seconds worth of audio data is stored in memory. Although it should work
+fine, it's inefficient to use streaming for short sounds. Streaming is useful for things like music
+tracks in games.
+
+
+
+When you initialize a sound, if you specify a sound group the sound will be attached to that group
+automatically. If you set it to NULL, it will be automatically attached to the engine's endpoint.
+If you would instead rather leave the sound unattached by default, you can can specify the
+MA_SOUND_FLAG_NO_DEFAULT_ATTACHMENT flag. This is useful if you want to set up a complex node
+graph.
+
+
+
+Sounds are not started by default. To start a sound, use ma_sound_start(). Stop a sound with
+ma_sound_stop().
+
+
+
+Sounds can have their volume controlled with ma_sound_set_volume() in the same way as the
+engine's master volume.
+
+
+
+Sounds support stereo panning and pitching. Set the pan with ma_sound_set_pan(). Setting the pan
+to 0 will result in an unpanned sound. Setting it to -1 will shift everything to the left, whereas
++1 will shift it to the right. The pitch can be controlled with ma_sound_set_pitch(). A larger
+value will result in a higher pitch. The pitch must be greater than 0.
+
+
+
+The engine supports 3D spatialization of sounds. By default sounds will have spatialization
+enabled, but if a sound does not need to be spatialized it's best to disable it. There are two ways
+to disable spatialization of a sound:
+
+
+
+
+
+// Disable spatialization at initialization time via a flag:
+ma_sound_init_from_file(&engine, "my_sound.wav", MA_SOUND_FLAG_NO_SPATIALIZATION, NULL, NULL, &sound);
+
+// Dynamically disable or enable spatialization post-initialization:
+ma_sound_set_spatialization_enabled(&sound, isSpatializationEnabled);
+
+
+By default sounds will be spatialized based on the closest listener. If a sound should always be
+spatialized relative to a specific listener it can be pinned to one:
+
+
+Like listeners, sounds have a position. By default, the position of a sound is in absolute space,
+but it can be changed to be relative to a listener:
+
+
+Note that relative positioning of a sound only makes sense if there is either only one listener, or
+the sound is pinned to a specific listener. To set the position of a sound:
+
+
+The engine supports different attenuation models which can be configured on a per-sound basis. By
+default the attenuation model is set to ma_attenuation_model_inverse which is the equivalent to
+OpenAL's AL_INVERSE_DISTANCE_CLAMPED. Configure the attenuation model like so:
+
+
+Likewise, in the calculation of attenuation, you can control the minimum and maximum distances for
+the attenuation calculation. This is useful if you want to ensure sounds don't drop below a certain
+volume after the listener moves further away and to have sounds play a maximum volume when the
+listener is within a certain distance:
+
+
+You can fade sounds in and out with ma_sound_set_fade_in_pcm_frames() and
+ma_sound_set_fade_in_milliseconds(). Set the volume to -1 to use the current volume as the
+starting volume:
+
+
+
+
+
+// Fade in over 1 second.
+ma_sound_set_fade_in_milliseconds(&sound, 0, 1, 1000);
+
+// ... sometime later ...
+
+// Fade out over 1 second, starting from the current volume.
+ma_sound_set_fade_in_milliseconds(&sound, -1, 0, 1000);
+
+
+By default sounds will start immediately, but sometimes for timing and synchronization purposes it
+can be useful to schedule a sound to start or stop:
+
+
+
+
+
+// Start the sound in 1 second from now.
+ma_sound_set_start_time_in_pcm_frames(&sound, ma_engine_get_time(&engine) + (ma_engine_get_sample_rate(&engine) * 1));
+
+// Stop the sound in 2 seconds from now.
+ma_sound_set_stop_time_in_pcm_frames(&sound, ma_engine_get_time(&engine) + (ma_engine_get_sample_rate(&engine) * 2));
+
+
+The time is specified in global time which is controlled by the engine. You can get the engine's
+current time with ma_engine_get_time(). The engine's global time is incremented automatically as
+audio data is read, but it can be reset with ma_engine_set_time() in case it needs to be
+resynchronized for some reason.
+
+
+
+To determine whether or not a sound is currently playing, use ma_sound_is_playing(). This will
+take the scheduled start and stop times into account.
+
+
+
+Whether or not a sound should loop can be controlled with ma_sound_set_looping(). Sounds will not
+be looping by default. Use ma_sound_is_looping() to determine whether or not a sound is looping.
+
+
+
+Use ma_sound_at_end() to determine whether or not a sound is currently at the end. For a looping
+sound this should never return true.
+
+
+
+Internally a sound wraps around a data source. Some APIs exist to control the underlying data
+source, mainly for convenience:
+
+
+Sound groups have the same API as sounds, only they are called ma_sound_group, and since they do
+not have any notion of a data source, anything relating to a data source is unavailable.
+
+
+
+Internally, sound data is loaded via the ma_decoder API which means by default in only supports
+file formats that have built-in support in miniaudio. You can extend this to support any kind of
+file format through the use of custom decoders. To do this you'll need to use a self-managed
+resource manager and configure it appropriately. See the "Resource Management" section below for
+details on how to set this up.
+
+
+
+
+
+
+
+
6. Resource Management
+
+Many programs will want to manage sound resources for things such as reference counting and
+streaming. This is supported by miniaudio via the ma_resource_manager API.
+
+
+
+The resource manager is mainly responsible for the following:
+
+
+
+Loading of sound files into memory with reference counting.
+
+Streaming of sound data
+
+
+
+When loading a sound file, the resource manager will give you back a ma_data_source compatible
+object called ma_resource_manager_data_source. This object can be passed into any
+ma_data_source API which is how you can read and seek audio data. When loading a sound file, you
+specify whether or not you want the sound to be fully loaded into memory (and optionally
+pre-decoded) or streamed. When loading into memory, you can also specify whether or not you want
+the data to be loaded asynchronously.
+
+
+
+The example below is how you can initialize a resource manager using it's default configuration:
+
+
+You can configure the format, channels and sample rate of the decoded audio data. By default it
+will use the file's native data format, but you can configure it to use a consistent format. This
+is useful for offloading the cost of data conversion to load time rather than dynamically
+converting at mixing time. To do this, you configure the decoded format, channels and sample rate
+like the code below:
+
+
+In the code above, the resource manager will be configured so that any decoded audio data will be
+pre-converted at load time to the device's native data format. If instead you used defaults and
+the data format of the file did not match the device's data format, you would need to convert the
+data at mixing time which may be prohibitive in high-performance and large scale scenarios like
+games.
+
+
+
+Internally the resource manager uses the ma_decoder API to load sounds. This means by default it
+only supports decoders that are built into miniaudio. It's possible to support additional encoding
+formats through the use of custom decoders. To do so, pass in your ma_decoding_backend_vtable
+vtables into the resource manager config:
+
+
+This system can allow you to support any kind of file format. See the "Decoding" section for
+details on how to implement custom decoders. The miniaudio repository includes examples for Opus
+via libopus and libopusfile and Vorbis via libvorbis and libvorbisfile.
+
+
+
+Asynchronicity is achieved via a job system. When an operation needs to be performed, such as the
+decoding of a page, a job will be posted to a queue which will then be processed by a job thread.
+By default there will be only one job thread running, but this can be configured, like so:
+
+
+By default job threads are managed internally by the resource manager, however you can also self
+manage your job threads if, for example, you want to integrate the job processing into your
+existing job infrastructure, or if you simply don't like the way the resource manager does it. To
+do this, just set the job thread count to 0 and process jobs manually. To process jobs, you first
+need to retrieve a job using ma_resource_manager_next_job() and then process it using
+ma_resource_manager_process_job():
+
+
+
+
+
+config = ma_resource_manager_config_init();
+config.jobThreadCount = 0; // Don't manage any job threads internally.
+config.flags = MA_RESOURCE_MANAGER_FLAG_NON_BLOCKING; // Optional. Makes ma_resource_manager_next_job() non-blocking.
+
+// ... Initialize your custom job threads ...
+
+void my_custom_job_thread(...)
+{
+ for (;;) {
+ ma_resource_manager_job job;
+ ma_result result = ma_resource_manager_next_job(pMyResourceManager, &job);
+ if (result != MA_SUCCESS) {
+ if (result == MA_NOT_DATA_AVAILABLE) {
+ // No jobs are available. Keep going. Will only get this if the resource manager was initialized
+ // with MA_RESOURCE_MANAGER_FLAG_NON_BLOCKING.
+ continue;
+ } elseif (result == MA_CANCELLED) {
+ // MA_RESOURCE_MANAGER_JOB_QUIT was posted. Exit.
+ break;
+ } else {
+ // Some other error occurred.
+ break;
+ }
+ }
+
+ ma_resource_manager_process_job(pMyResourceManager, &job);
+ }
+}
+
+
+In the example above, the MA_RESOURCE_MANAGER_JOB_QUIT event is the used as the termination
+indicator, but you can use whatever you would like to terminate the thread. The call to
+ma_resource_manager_next_job() is blocking by default, but can be configured to be non-blocking
+by initializing the resource manager with the MA_RESOURCE_MANAGER_FLAG_NON_BLOCKING configuration
+flag. Note that the MA_RESOURCE_MANAGER_JOB_QUIT will never be removed from the job queue. This
+is to give every thread the opportunity to catch the event and terminate naturally.
+
+
+
+When loading a file, it's sometimes convenient to be able to customize how files are opened and
+read instead of using standard fopen(), fclose(), etc. which is what miniaudio will use by
+default. This can be done by setting pVFS member of the resource manager's config:
+
+
+
+
+
+// Initialize your custom VFS object. See documentation for VFS for information on how to do this.
+my_custom_vfs vfs = my_custom_vfs_init();
+
+config = ma_resource_manager_config_init();
+config.pVFS = &vfs;
+
+
+This is particularly useful in programs like games where you want to read straight from an archive
+rather than the normal file system. If you do not specify a custom VFS, the resource manager will
+use the operating system's normal file operations. This is default.
+
+
+
+To load a sound file and create a data source, call ma_resource_manager_data_source_init(). When
+loading a sound you need to specify the file path and options for how the sounds should be loaded.
+By default a sound will be loaded synchronously. The returned data source is owned by the caller
+which means the caller is responsible for the allocation and freeing of the data source. Below is
+an example for initializing a data source:
+
+
+
+
+
+ma_resource_manager_data_source dataSource;
+ma_result result = ma_resource_manager_data_source_init(pResourceManager, pFilePath, flags, &dataSource);
+if (result != MA_SUCCESS) {
+ // Error.
+}
+
+// ...
+
+// A ma_resource_manager_data_source object is compatible with the ma_data_source API. To read data, just call
+// the ma_data_source_read_pcm_frames() like you would with any normal data source.
+result = ma_data_source_read_pcm_frames(&dataSource, pDecodedData, frameCount, &framesRead);
+if (result != MA_SUCCESS) {
+ // Failed to read PCM frames.
+}
+
+// ...
+
+ma_resource_manager_data_source_uninit(pResourceManager, &dataSource);
+
+
+The flags parameter specifies how you want to perform loading of the sound file. It can be a
+combination of the following flags:
+
+
+When no flags are specified (set to 0), the sound will be fully loaded into memory, but not
+decoded, meaning the raw file data will be stored in memory, and then dynamically decoded when
+ma_data_source_read_pcm_frames() is called. To instead decode the audio data before storing it in
+memory, use the MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_DECODE flag. By default, the sound file will
+be loaded synchronously, meaning ma_resource_manager_data_source_init() will only return after
+the entire file has been loaded. This is good for simplicity, but can be prohibitively slow. You
+can instead load the sound asynchronously using the MA_DATA_SOURCE_ASYNC flag. This will result
+in ma_resource_manager_data_source_init() returning quickly, but no data will be returned by
+ma_data_source_read_pcm_frames() until some data is available. When no data is available because
+the asynchronous decoding hasn't caught up, MA_BUSY will be returned by
+ma_data_source_read_pcm_frames().
+
+
+
+For large sounds, it's often prohibitive to store the entire file in memory. To mitigate this, you
+can instead stream audio data which you can do by specifying the
+MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM flag. When streaming, data will be decoded in 1
+second pages. When a new page needs to be decoded, a job will be posted to the job queue and then
+subsequently processed in a job thread.
+
+
+
+For in-memory sounds, reference counting is used to ensure the data is loaded only once. This means
+multiple calls to ma_resource_manager_data_source_init() with the same file path will result in
+the file data only being loaded once. Each call to ma_resource_manager_data_source_init() must be
+matched up with a call to ma_resource_manager_data_source_uninit(). Sometimes it can be useful
+for a program to register self-managed raw audio data and associate it with a file path. Use the
+ma_resource_manager_register_*() and ma_resource_manager_unregister_*() APIs to do this.
+ma_resource_manager_register_decoded_data() is used to associate a pointer to raw, self-managed
+decoded audio data in the specified data format with the specified name. Likewise,
+ma_resource_manager_register_encoded_data() is used to associate a pointer to raw self-managed
+encoded audio data (the raw file data) with the specified name. Note that these names need not be
+actual file paths. When ma_resource_manager_data_source_init() is called (without the
+MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM flag), the resource manager will look for these
+explicitly registered data buffers and, if found, will use it as the backing data for the data
+source. Note that the resource manager does *not* make a copy of this data so it is up to the
+caller to ensure the pointer stays valid for it's lifetime. Use
+ma_resource_manager_unregister_data() to unregister the self-managed data. You can also use
+ma_resource_manager_register_file() and ma_resource_manager_unregister_file() to register and
+unregister a file. It does not make sense to use the MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM
+flag with a self-managed data pointer.
+
+
+
+
+
+
+
+
6.1. Asynchronous Loading and Synchronization
+
+When loading asynchronously, it can be useful to poll whether or not loading has finished. Use
+ma_resource_manager_data_source_result() to determine this. For in-memory sounds, this will
+return MA_SUCCESS when the file has been *entirely* decoded. If the sound is still being decoded,
+MA_BUSY will be returned. Otherwise, some other error code will be returned if the sound failed
+to load. For streaming data sources, MA_SUCCESS will be returned when the first page has been
+decoded and the sound is ready to be played. If the first page is still being decoded, MA_BUSY
+will be returned. Otherwise, some other error code will be returned if the sound failed to load.
+
+
+
+In addition to polling, you can also use a simple synchronization object called a "fence" to wait
+for asynchronously loaded sounds to finish. This is called ma_fence. The advantage to using a
+fence is that it can be used to wait for a group of sounds to finish loading rather than waiting
+for sounds on an individual basis. There are two stages to loading a sound:
+
+
+
+Initialization of the internal decoder; and
+
+Completion of decoding of the file (the file is fully decoded)
+
+
+
+You can specify separate fences for each of the different stages. Waiting for the initialization
+of the internal decoder is important for when you need to know the sample format, channels and
+sample rate of the file.
+
+
+
+The example below shows how you could use a fence when loading a number of sounds:
+
+
+
+
+
+// This fence will be released when all sounds are finished loading entirely.
+ma_fence fence;
+ma_fence_init(&fence);
+
+// This will be passed into the initialization routine for each sound.
+ma_resource_manager_pipeline_notifications notifications = ma_resource_manager_pipeline_notifications_init();
+notifications.done.pFence = &fence;
+
+// Now load a bunch of sounds:
+for (iSound = 0; iSound < soundCount; iSound += 1) {
+ ma_resource_manager_data_source_init(pResourceManager, pSoundFilePaths[iSound], flags, ¬ifications, &pSoundSources[iSound]);
+}
+
+// ... DO SOMETHING ELSE WHILE SOUNDS ARE LOADING ...
+
+// Wait for loading of sounds to finish.
+ma_fence_wait(&fence);
+
+
+In the example above we used a fence for waiting until the entire file has been fully decoded. If
+You only need to wait for the initialization of the internal decoder to complete, you can use the
+init member of the ma_resource_manager_pipeline_notifications object:
+
+
+
+
+
+notifications.init.pFence = &fence;
+
+
+If a fence is not appropriate for your situation, you can instead use a callback that is fired on
+an individual sound basis. This is done in a very similar way to fences:
+
+
+In the example above we just extend the ma_async_notification_callbacks object and pass an
+instantiation into the ma_resource_manager_pipeline_notifications in the same way as we did with
+the fence, only we set pNotification instead of pFence. You can set both of these at the same
+time and they should both work as expected. If using the pNotification system, you need to ensure
+your ma_async_notification_callbacks object stays valid.
+
+
+
+
+
+
+
+
+
+
+
6.2. Resource Manager Implementation Details
+
+Resources are managed in two main ways:
+
+
+
+By storing the entire sound inside an in-memory buffer (referred to as a data buffer)
+
+By streaming audio data on the fly (referred to as a data stream)
+
+
+
+A resource managed data source (ma_resource_manager_data_source) encapsulates a data buffer or
+data stream, depending on whether or not the data source was initialized with the
+MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM flag. If so, it will make use of a
+ma_resource_manager_data_stream object. Otherwise it will use a ma_resource_manager_data_buffer
+object. Both of these objects are data sources which means they can be used with any
+ma_data_source_*() API.
+
+
+
+Another major feature of the resource manager is the ability to asynchronously decode audio files.
+This relieves the audio thread of time-consuming decoding which can negatively affect scalability
+due to the audio thread needing to complete it's work extremely quickly to avoid glitching.
+Asynchronous decoding is achieved through a job system. There is a central multi-producer,
+multi-consumer, fixed-capacity job queue. When some asynchronous work needs to be done, a job is
+posted to the queue which is then read by a job thread. The number of job threads can be
+configured for improved scalability, and job threads can all run in parallel without needing to
+worry about the order of execution (how this is achieved is explained below).
+
+
+
+When a sound is being loaded asynchronously, playback can begin before the sound has been fully
+decoded. This enables the application to start playback of the sound quickly, while at the same
+time allowing to resource manager to keep loading in the background. Since there may be less
+threads than the number of sounds being loaded at a given time, a simple scheduling system is used
+to keep decoding time balanced and fair. The resource manager solves this by splitting decoding
+into chunks called pages. By default, each page is 1 second long. When a page has been decoded, a
+new job will be posted to start decoding the next page. By dividing up decoding into pages, an
+individual sound shouldn't ever delay every other sound from having their first page decoded. Of
+course, when loading many sounds at the same time, there will always be an amount of time required
+to process jobs in the queue so in heavy load situations there will still be some delay. To
+determine if a data source is ready to have some frames read, use
+ma_resource_manager_data_source_get_available_frames(). This will return the number of frames
+available starting from the current position.
+
+
+
+
+
+
+
+
6.2.1. Job Queue
+
+The resource manager uses a job queue which is multi-producer, multi-consumer, and fixed-capacity.
+This job queue is not currently lock-free, and instead uses a spinlock to achieve thread-safety.
+Only a fixed number of jobs can be allocated and inserted into the queue which is done through a
+lock-free data structure for allocating an index into a fixed sized array, with reference counting
+for mitigation of the ABA problem. The reference count is 32-bit.
+
+
+
+For many types of jobs it's important that they execute in a specific order. In these cases, jobs
+are executed serially. For the resource manager, serial execution of jobs is only required on a
+per-object basis (per data buffer or per data stream). Each of these objects stores an execution
+counter. When a job is posted it is associated with an execution counter. When the job is
+processed, it checks if the execution counter of the job equals the execution counter of the
+owning object and if so, processes the job. If the counters are not equal, the job will be posted
+back onto the job queue for later processing. When the job finishes processing the execution order
+of the main object is incremented. This system means the no matter how many job threads are
+executing, decoding of an individual sound will always get processed serially. The advantage to
+having multiple threads comes into play when loading multiple sounds at the same time.
+
+
+
+The resource manager's job queue is not 100% lock-free and will use a spinlock to achieve
+thread-safety for a very small section of code. This is only relevant when the resource manager
+uses more than one job thread. If only using a single job thread, which is the default, the
+lock should never actually wait in practice. The amount of time spent locking should be quite
+short, but it's something to be aware of for those who have pedantic lock-free requirements and
+need to use more than one job thread. There are plans to remove this lock in a future version.
+
+
+
+In addition, posting a job will release a semaphore, which on Win32 is implemented with
+ReleaseSemaphore and on POSIX platforms via a condition variable:
+
+
+Again, this is relevant for those with strict lock-free requirements in the audio thread. To avoid
+this, you can use non-blocking mode (via the MA_RESOURCE_MANAGER_JOB_QUEUE_FLAG_NON_BLOCKING
+flag) and implement your own job processing routine (see the "Resource Manager" section above for
+details on how to do this).
+
+
+
+
+
+
+
+
+
+
+
6.2.2. Data Buffers
+
+When the MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM flag is excluded at initialization time, the
+resource manager will try to load the data into an in-memory data buffer. Before doing so, however,
+it will first check if the specified file is already loaded. If so, it will increment a reference
+counter and just use the already loaded data. This saves both time and memory. When the data buffer
+is uninitialized, the reference counter will be decremented. If the counter hits zero, the file
+will be unloaded. This is a detail to keep in mind because it could result in excessive loading and
+unloading of a sound. For example, the following sequence will result in a file be loaded twice,
+once after the other:
+
+
+A binary search tree (BST) is used for storing data buffers as it has good balance between
+efficiency and simplicity. The key of the BST is a 64-bit hash of the file path that was passed
+into ma_resource_manager_data_source_init(). The advantage of using a hash is that it saves
+memory over storing the entire path, has faster comparisons, and results in a mostly balanced BST
+due to the random nature of the hash. The disadvantage is that file names are case-sensitive. If
+this is an issue, you should normalize your file names to upper- or lower-case before initializing
+your data sources.
+
+
+
+When a sound file has not already been loaded and the MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC
+flag is excluded, the file will be decoded synchronously by the calling thread. There are two
+options for controlling how the audio is stored in the data buffer - encoded or decoded. When the
+MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_DECODE option is excluded, the raw file data will be stored
+in memory. Otherwise the sound will be decoded before storing it in memory. Synchronous loading is
+a very simple and standard process of simply adding an item to the BST, allocating a block of
+memory and then decoding (if MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_DECODE is specified).
+
+
+
+When the MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC flag is specified, loading of the data buffer
+is done asynchronously. In this case, a job is posted to the queue to start loading and then the
+function immediately returns, setting an internal result code to MA_BUSY. This result code is
+returned when the program calls ma_resource_manager_data_source_result(). When decoding has fully
+completed MA_SUCCESS will be returned. This can be used to know if loading has fully completed.
+
+
+
+When loading asynchronously, a single job is posted to the queue of the type
+MA_RESOURCE_MANAGER_JOB_LOAD_DATA_BUFFER_NODE. This involves making a copy of the file path and
+associating it with job. When the job is processed by the job thread, it will first load the file
+using the VFS associated with the resource manager. When using a custom VFS, it's important that it
+be completely thread-safe because it will be used from one or more job threads at the same time.
+Individual files should only ever be accessed by one thread at a time, however. After opening the
+file via the VFS, the job will determine whether or not the file is being decoded. If not, it
+simply allocates a block of memory and loads the raw file contents into it and returns. On the
+other hand, when the file is being decoded, it will first allocate a decoder on the heap and
+initialize it. Then it will check if the length of the file is known. If so it will allocate a
+block of memory to store the decoded output and initialize it to silence. If the size is unknown,
+it will allocate room for one page. After memory has been allocated, the first page will be
+decoded. If the sound is shorter than a page, the result code will be set to MA_SUCCESS and the
+completion event will be signalled and loading is now complete. If, however, there is more to
+decode, a job with the code MA_RESOURCE_MANAGER_JOB_PAGE_DATA_BUFFER_NODE is posted. This job
+will decode the next page and perform the same process if it reaches the end. If there is more to
+decode, the job will post another MA_RESOURCE_MANAGER_JOB_PAGE_DATA_BUFFER_NODE job which will
+keep on happening until the sound has been fully decoded. For sounds of an unknown length, each
+page will be linked together as a linked list. Internally this is implemented via the
+ma_paged_audio_buffer object.
+
+
+
+
+
+
+
+
6.2.3. Data Streams
+
+Data streams only ever store two pages worth of data for each instance. They are most useful for
+large sounds like music tracks in games that would consume too much memory if fully decoded in
+memory. After every frame from a page has been read, a job will be posted to load the next page
+which is done from the VFS.
+
+
+
+For data streams, the MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC flag will determine whether or
+not initialization of the data source waits until the two pages have been decoded. When unset,
+ma_resource_manager_data_source_init() will wait until the two pages have been loaded, otherwise
+it will return immediately.
+
+
+
+When frames are read from a data stream using ma_resource_manager_data_source_read_pcm_frames(),
+MA_BUSY will be returned if there are no frames available. If there are some frames available,
+but less than the number requested, MA_SUCCESS will be returned, but the actual number of frames
+read will be less than the number requested. Due to the asymchronous nature of data streams,
+seeking is also asynchronous. If the data stream is in the middle of a seek, MA_BUSY will be
+returned when trying to read frames.
+
+
+
+When ma_resource_manager_data_source_read_pcm_frames() results in a page getting fully consumed
+a job is posted to load the next page. This will be posted from the same thread that called
+ma_resource_manager_data_source_read_pcm_frames().
+
+
+
+Data streams are uninitialized by posting a job to the queue, but the function won't return until
+that job has been processed. The reason for this is that the caller owns the data stream object and
+therefore miniaudio needs to ensure everything completes before handing back control to the caller.
+Also, if the data stream is uninitialized while pages are in the middle of decoding, they must
+complete before destroying any underlying object and the job system handles this cleanly.
+
+
+
+Note that when a new page needs to be loaded, a job will be posted to the resource manager's job
+thread from the audio thread. You must keep in mind the details mentioned in the "Job Queue"
+section above regarding locking when posting an event if you require a strictly lock-free audio
+thread.
+
+
+
+
+
+
+
+
+
+
+
7. Node Graph
+
+miniaudio's routing infrastructure follows a node graph paradigm. The idea is that you create a
+node whose outputs are attached to inputs of another node, thereby creating a graph. There are
+different types of nodes, with each node in the graph processing input data to produce output,
+which is then fed through the chain. Each node in the graph can apply their own custom effects. At
+the start of the graph will usually be one or more data source nodes which have no inputs, but
+instead pull their data from a data source. At the end of the graph is an endpoint which represents
+the end of the chain and is where the final output is ultimately extracted from.
+
+
+
+Each node has a number of input buses and a number of output buses. An output bus from a node is
+attached to an input bus of another. Multiple nodes can connect their output buses to another
+node's input bus, in which case their outputs will be mixed before processing by the node. Below is
+a diagram that illustrates a hypothetical node graph setup:
+
+
+
+
+
+>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> Data flows left to right >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
+
++---------------+ +-----------------+
+| Data Source 1 =----+ +----------+ +----= Low Pass Filter =----+
++---------------+ | | =----+ +-----------------+ | +----------+
+ +----= Splitter | +----= ENDPOINT |
++---------------+ | | =----+ +-----------------+ | +----------+
+| Data Source 2 =----+ +----------+ +----= Echo / Delay =----+
++---------------+ +-----------------+
+
+
+In the above graph, it starts with two data sources whose outputs are attached to the input of a
+splitter node. It's at this point that the two data sources are mixed. After mixing, the splitter
+performs it's processing routine and produces two outputs which is simply a duplication of the
+input stream. One output is attached to a low pass filter, whereas the other output is attached to
+a echo/delay. The outputs of the the low pass filter and the echo are attached to the endpoint, and
+since they're both connected to the same input but, they'll be mixed.
+
+
+
+Each input bus must be configured to accept the same number of channels, but the number of channels
+used by input buses can be different to the number of channels for output buses in which case
+miniaudio will automatically convert the input data to the output channel count before processing.
+The number of channels of an output bus of one node must match the channel count of the input bus
+it's attached to. The channel counts cannot be changed after the node has been initialized. If you
+attempt to attach an output bus to an input bus with a different channel count, attachment will
+fail.
+
+
+
+To use a node graph, you first need to initialize a ma_node_graph object. This is essentially a
+container around the entire graph. The ma_node_graph object is required for some thread-safety
+issues which will be explained later. A ma_node_graph object is initialized using miniaudio's
+standard config/init system:
+
+
+
+
+
+ma_node_graph_config nodeGraphConfig = ma_node_graph_config_init(myChannelCount);
+
+result = ma_node_graph_init(&nodeGraphConfig, NULL, &nodeGraph); // Second parameter is a pointer to allocation callbacks.
+if (result != MA_SUCCESS) {
+ // Failed to initialize node graph.
+}
+
+
+When you initialize the node graph, you're specifying the channel count of the endpoint. The
+endpoint is a special node which has one input bus and one output bus, both of which have the
+same channel count, which is specified in the config. Any nodes that connect directly to the
+endpoint must be configured such that their output buses have the same channel count. When you read
+audio data from the node graph, it'll have the channel count you specified in the config. To read
+data from the graph:
+
+
+
+
+
+ma_uint32 framesRead;
+result = ma_node_graph_read_pcm_frames(&nodeGraph, pFramesOut, frameCount, &framesRead);
+if (result != MA_SUCCESS) {
+ // Failed to read data from the node graph.
+}
+
+
+When you read audio data, miniaudio starts at the node graph's endpoint node which then pulls in
+data from it's input attachments, which in turn recusively pull in data from their inputs, and so
+on. At the start of the graph there will be some kind of data source node which will have zero
+inputs and will instead read directly from a data source. The base nodes don't literally need to
+read from a ma_data_source object, but they will always have some kind of underlying object that
+sources some kind of audio. The ma_data_source_node node can be used to read from a
+ma_data_source. Data is always in floating-point format and in the number of channels you
+specified when the graph was initialized. The sample rate is defined by the underlying data sources.
+It's up to you to ensure they use a consistent and appropraite sample rate.
+
+
+
+The ma_node API is designed to allow custom nodes to be implemented with relative ease, but
+miniaudio includes a few stock nodes for common functionality. This is how you would initialize a
+node which reads directly from a data source (ma_data_source_node) which is an example of one
+of the stock nodes that comes with miniaudio:
+
+
+The data source node will use the output channel count to determine the channel count of the output
+bus. There will be 1 output bus and 0 input buses (data will be drawn directly from the data
+source). The data source must output to floating-point (ma_format_f32) or else an error will be
+returned from ma_data_source_node_init().
+
+
+
+By default the node will not be attached to the graph. To do so, use ma_node_attach_output_bus():
+
+
+The code above connects the data source node directly to the endpoint. Since the data source node
+has only a single output bus, the index will always be 0. Likewise, the endpoint only has a single
+input bus which means the input bus index will also always be 0.
+
+
+
+To detach a specific output bus, use ma_node_detach_output_bus(). To detach all output buses, use
+ma_node_detach_all_output_buses(). If you want to just move the output bus from one attachment to
+another, you do not need to detach first. You can just call ma_node_attach_output_bus() and it'll
+deal with it for you.
+
+
+
+Less frequently you may want to create a specialized node. This will be a node where you implement
+your own processing callback to apply a custom effect of some kind. This is similar to initalizing
+one of the stock node types, only this time you need to specify a pointer to a vtable containing a
+pointer to the processing function and the number of input and output buses. Example:
+
+
+
+
+
+staticvoid my_custom_node_process_pcm_frames(ma_node* pNode, constfloat** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
+{
+ // Do some processing of ppFramesIn (one stream of audio data per input bus)
+ constfloat* pFramesIn_0 = ppFramesIn[0]; // Input bus @ index 0.
+ constfloat* pFramesIn_1 = ppFramesIn[1]; // Input bus @ index 1.
+ float* pFramesOut_0 = ppFramesOut[0]; // Output bus @ index 0.
+
+ // Do some processing. On input, pFrameCountIn will be the number of input frames in each
+ // buffer in ppFramesIn and pFrameCountOut will be the capacity of each of the buffers
+ // in ppFramesOut. On output, pFrameCountIn should be set to the number of input frames
+ // your node consumed and pFrameCountOut should be set the number of output frames that
+ // were produced.
+ //
+ // You should process as many frames as you can. If your effect consumes input frames at the
+ // same rate as output frames (always the case, unless you're doing resampling), you need
+ // only look at ppFramesOut and process that exact number of frames. If you're doing
+ // resampling, you'll need to be sure to set both pFrameCountIn and pFrameCountOut
+ // properly.
+}
+
+static ma_node_vtable my_custom_node_vtable =
+{
+ my_custom_node_process_pcm_frames, // The function that will be called process your custom node. This is where you'd implement your effect processing.
+ NULL, // Optional. A callback for calculating the number of input frames that are required to process a specified number of output frames.
+ 2, // 2 input buses.
+ 1, // 1 output bus.
+ 0 // Default flags.
+};
+
+...
+
+// Each bus needs to have a channel count specified. To do this you need to specify the channel
+// counts in an array and then pass that into the node config.
+ma_uint32 inputChannels[2]; // Equal in size to the number of input channels specified in the vtable.
+ma_uint32 outputChannels[1]; // Equal in size to the number of output channels specicied in the vtable.
+
+inputChannels[0] = channelsIn;
+inputChannels[1] = channelsIn;
+outputChannels[0] = channelsOut;
+
+ma_node_config nodeConfig = ma_node_config_init();
+nodeConfig.vtable = &my_custom_node_vtable;
+nodeConfig.pInputChannels = inputChannels;
+nodeConfig.pOutputChannels = outputChannels;
+
+ma_node_base node;
+result = ma_node_init(&nodeGraph, &nodeConfig, NULL, &node);
+if (result != MA_SUCCESS) {
+ // Failed to initialize node.
+}
+
+
+When initializing a custom node, as in the code above, you'll normally just place your vtable in
+static space. The number of input and output buses are specified as part of the vtable. If you need
+a variable number of buses on a per-node bases, the vtable should have the relevant bus count set
+to MA_NODE_BUS_COUNT_UNKNOWN. In this case, the bus count should be set in the node config:
+
+
+
+
+
+static ma_node_vtable my_custom_node_vtable =
+{
+ my_custom_node_process_pcm_frames, // The function that will be called process your custom node. This is where you'd implement your effect processing.
+ NULL, // Optional. A callback for calculating the number of input frames that are required to process a specified number of output frames.
+ MA_NODE_BUS_COUNT_UNKNOWN, // The number of input buses is determined on a per-node basis.
+ 1, // 1 output bus.
+ 0 // Default flags.
+};
+
+...
+
+ma_node_config nodeConfig = ma_node_config_init();
+nodeConfig.vtable = &my_custom_node_vtable;
+nodeConfig.inputBusCount = myBusCount; // <-- Since the vtable specifies MA_NODE_BUS_COUNT_UNKNOWN, the input bus count should be set here.
+nodeConfig.pInputChannels = inputChannels; // <-- Make sure there are nodeConfig.inputBusCount elements in this array.
+nodeConfig.pOutputChannels = outputChannels; // <-- The vtable specifies 1 output bus, so there must be 1 element in this array.
+
+
+In the above example it's important to never set the inputBusCount and outputBusCount members
+to anything other than their defaults if the vtable specifies an explicit count. They can only be
+set if the vtable specifies MA_NODE_BUS_COUNT_UNKNOWN in the relevant bus count.
+
+
+
+Most often you'll want to create a structure to encapsulate your node with some extra data. You
+need to make sure the ma_node_base object is your first member of the structure:
+
+
+
+
+
+typedefstruct
+{
+ ma_node_base base; // <-- Make sure this is always the first member.
+ float someCustomData;
+} my_custom_node;
+
+
+By doing this, your object will be compatible with all ma_node APIs and you can attach it to the
+graph just like any other node.
+
+
+
+In the custom processing callback (my_custom_node_process_pcm_frames() in the example above), the
+number of channels for each bus is what was specified by the config when the node was initialized
+with ma_node_init(). In addition, all attachments to each of the input buses will have been
+pre-mixed by miniaudio. The config allows you to specify different channel counts for each
+individual input and output bus. It's up to the effect to handle it appropriate, and if it can't,
+return an error in it's initialization routine.
+
+
+
+Custom nodes can be assigned some flags to describe their behaviour. These are set via the vtable
+and include the following:
+
+
+
+
+
+
+Flag Name
+
+
+Description
+
+
+
+
+MA_NODE_FLAG_PASSTHROUGH
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+Useful for nodes that do not do any kind of audio
+processing, but are instead used for tracking
+time, handling events, etc. Also used by the
+internal endpoint node. It reads directly from
+the input bus to the output bus. Nodes with this
+flag must have exactly 1 input bus and 1 output
+bus, and both buses must have the same channel
+counts.
+
+
+
+
+MA_NODE_FLAG_CONTINUOUS_PROCESSING
+
+
+
+
+
+
+
+
+
+
+
+Causes the processing callback to be called even
+when no data is available to be read from input
+attachments. This is useful for effects like
+echos where there will be a tail of audio data
+that still needs to be processed even when the
+original data sources have reached their ends.
+
+
+
+
+MA_NODE_FLAG_ALLOW_NULL_INPUT
+
+
+
+
+
+
+
+
+
+
+
+
+Used in conjunction with
+MA_NODE_FLAG_CONTINUOUS_PROCESSING. When this
+is set, the ppFramesIn parameter of the
+processing callback will be set to NULL when
+there are no input frames are available. When
+this is unset, silence will be posted to the
+processing callback.
+
+
+
+
+MA_NODE_FLAG_DIFFERENT_PROCESSING_RATES
+
+
+
+
+
+
+
+Used to tell miniaudio that input and output
+frames are processed at different rates. You
+should set this for any nodes that perform
+resampling.
+
+
+
+
+If you need to make a copy of an audio stream for effect processing you can use a splitter node
+called ma_splitter_node. This takes has 1 input bus and splits the stream into 2 output buses.
+You can use it like this:
+
+
+
+
+
+ma_splitter_node_config splitterNodeConfig = ma_splitter_node_config_init(channelsIn, channelsOut);
+
+ma_splitter_node splitterNode;
+result = ma_splitter_node_init(&nodeGraph, &splitterNodeConfig, NULL, &splitterNode);
+if (result != MA_SUCCESS) {
+ // Failed to create node.
+}
+
+// Attach your output buses to two different input buses (can be on two different nodes).
+ma_node_attach_output_bus(&splitterNode, 0, ma_node_graph_get_endpoint(&nodeGraph), 0); // Attach directly to the endpoint.
+ma_node_attach_output_bus(&splitterNode, 1, &myEffectNode, 0); // Attach to input bus 0 of some effect node.
+
+
+The volume of an output bus can be configured on a per-bus basis:
+
+
+By default the node is in a started state, but since it won't be connected to anything won't
+actually be invoked by the node graph until it's connected. When you stop a node, data will not be
+read from any of it's input connections. You can use this property to stop a group of sounds
+atomically.
+
+
+
+You can configure the initial state of a node in it's config:
+
+
+Note that for the stock specialized nodes, all of their configs will have a nodeConfig member
+which is the config to use with the base node. This is where the initial state can be configured
+for specialized nodes:
+
+
+When using a specialized node like ma_data_source_node or ma_splitter_node, be sure to not
+modify the vtable member of the nodeConfig object.
+
+
+
+
+
+
+
+
7.1. Timing
+
+The node graph supports starting and stopping nodes at scheduled times. This is especially useful
+for data source nodes where you want to get the node set up, but only start playback at a specific
+time. There are two clocks: local and global.
+
+
+
+A local clock is per-node, whereas the global clock is per graph. Scheduling starts and stops can
+only be done based on the global clock because the local clock will not be running while the node
+is stopped. The global clocks advances whenever ma_node_graph_read_pcm_frames() is called. On the
+other hand, the local clock only advances when the node's processing callback is fired, and is
+advanced based on the output frame count.
+
+
+
+To retrieve the global time, use ma_node_graph_get_time(). The global time can be set with
+ma_node_graph_set_time() which might be useful if you want to do seeking on a global timeline.
+Getting and setting the local time is similar. Use ma_node_get_time() to retrieve the local time,
+and ma_node_set_time() to set the local time. The global and local times will be advanced by the
+audio thread, so care should be taken to avoid data races. Ideally you should avoid calling these
+outside of the node processing callbacks which are always run on the audio thread.
+
+
+
+There is basic support for scheduling the starting and stopping of nodes. You can only schedule one
+start and one stop at a time. This is mainly intended for putting nodes into a started or stopped
+state in a frame-exact manner. Without this mechanism, starting and stopping of a node is limited
+to the resolution of a call to ma_node_graph_read_pcm_frames() which would typically be in blocks
+of several milliseconds. The following APIs can be used for scheduling node states:
+
+
+Note that due to the nature of multi-threading the times may not be 100% exact. If this is an
+issue, consider scheduling state changes from within a processing callback. An idea might be to
+have some kind of passthrough trigger node that is used specifically for tracking time and handling
+events.
+
+
+
+
+
+
+
+
+
+
+
7.2. Thread Safety and Locking
+
+When processing audio, it's ideal not to have any kind of locking in the audio thread. Since it's
+expected that ma_node_graph_read_pcm_frames() would be run on the audio thread, it does so
+without the use of any locks. This section discusses the implementation used by miniaudio and goes
+over some of the compromises employed by miniaudio to achieve this goal. Note that the current
+implementation may not be ideal - feedback and critiques are most welcome.
+
+
+
+The node graph API is not *entirely* lock-free. Only ma_node_graph_read_pcm_frames() is expected
+to be lock-free. Attachment, detachment and uninitialization of nodes use locks to simplify the
+implementation, but are crafted in a way such that such locking is not required when reading audio
+data from the graph. Locking in these areas are achieved by means of spinlocks.
+
+
+
+The main complication with keeping ma_node_graph_read_pcm_frames() lock-free stems from the fact
+that a node can be uninitialized, and it's memory potentially freed, while in the middle of being
+processed on the audio thread. There are times when the audio thread will be referencing a node,
+which means the uninitialization process of a node needs to make sure it delays returning until the
+audio thread is finished so that control is not handed back to the caller thereby giving them a
+chance to free the node's memory.
+
+
+
+When the audio thread is processing a node, it does so by reading from each of the output buses of
+the node. In order for a node to process data for one of it's output buses, it needs to read from
+each of it's input buses, and so on an so forth. It follows that once all output buses of a node
+are detached, the node as a whole will be disconnected and no further processing will occur unless
+it's output buses are reattached, which won't be happening when the node is being uninitialized.
+By having ma_node_detach_output_bus() wait until the audio thread is finished with it, we can
+simplify a few things, at the expense of making ma_node_detach_output_bus() a bit slower. By
+doing this, the implementation of ma_node_uninit() becomes trivial - just detach all output
+nodes, followed by each of the attachments to each of it's input nodes, and then do any final clean
+up.
+
+
+
+With the above design, the worst-case scenario is ma_node_detach_output_bus() taking as long as
+it takes to process the output bus being detached. This will happen if it's called at just the
+wrong moment where the audio thread has just iterated it and has just started processing. The
+caller of ma_node_detach_output_bus() will stall until the audio thread is finished, which
+includes the cost of recursively processing it's inputs. This is the biggest compromise made with
+the approach taken by miniaudio for it's lock-free processing system. The cost of detaching nodes
+earlier in the pipeline (data sources, for example) will be cheaper than the cost of detaching
+higher level nodes, such as some kind of final post-processing endpoint. If you need to do mass
+detachments, detach starting from the lowest level nodes and work your way towards the final
+endpoint node (but don't try detaching the node graph's endpoint). If the audio thread is not
+running, detachment will be fast and detachment in any order will be the same. The reason nodes
+need to wait for their input attachments to complete is due to the potential for desyncs between
+data sources. If the node was to terminate processing mid way through processing it's inputs,
+there's a chance that some of the underlying data sources will have been read, but then others not.
+That will then result in a potential desynchronization when detaching and reattaching higher-level
+nodes. A possible solution to this is to have an option when detaching to terminate processing
+before processing all input attachments which should be fairly simple.
+
+
+
+Another compromise, albeit less significant, is locking when attaching and detaching nodes. This
+locking is achieved by means of a spinlock in order to reduce memory overhead. A lock is present
+for each input bus and output bus. When an output bus is connected to an input bus, both the output
+bus and input bus is locked. This locking is specifically for attaching and detaching across
+different threads and does not affect ma_node_graph_read_pcm_frames() in any way. The locking and
+unlocking is mostly self-explanatory, but a slightly less intuitive aspect comes into it when
+considering that iterating over attachments must not break as a result of attaching or detaching a
+node while iteration is occuring.
+
+
+
+Attaching and detaching are both quite simple. When an output bus of a node is attached to an input
+bus of another node, it's added to a linked list. Basically, an input bus is a linked list, where
+each item in the list is and output bus. We have some intentional (and convenient) restrictions on
+what can done with the linked list in order to simplify the implementation. First of all, whenever
+something needs to iterate over the list, it must do so in a forward direction. Backwards iteration
+is not supported. Also, items can only be added to the start of the list.
+
+
+
+The linked list is a doubly-linked list where each item in the list (an output bus) holds a pointer
+to the next item in the list, and another to the previous item. A pointer to the previous item is
+only required for fast detachment of the node - it is never used in iteration. This is an
+important property because it means from the perspective of iteration, attaching and detaching of
+an item can be done with a single atomic assignment. This is exploited by both the attachment and
+detachment process. When attaching the node, the first thing that is done is the setting of the
+local "next" and "previous" pointers of the node. After that, the item is "attached" to the list
+by simply performing an atomic exchange with the head pointer. After that, the node is "attached"
+to the list from the perspective of iteration. Even though the "previous" pointer of the next item
+hasn't yet been set, from the perspective of iteration it's been attached because iteration will
+only be happening in a forward direction which means the "previous" pointer won't actually ever get
+used. The same general process applies to detachment. See ma_node_attach_output_bus() and
+ma_node_detach_output_bus() for the implementation of this mechanism.
+
+
+
+
+
+
+
+
+
+
+
8. Decoding
+
+The ma_decoder API is used for reading audio files. Decoders are completely decoupled from
+devices and can be used independently. The following formats are supported:
@@ -1363,7 +3923,8 @@ No
-Vorbis is supported via stb_vorbis which can be enabled by including the header section before the implementation of miniaudio, like the following:
+Vorbis is supported via stb_vorbis which can be enabled by including the header section before the
+implementation of miniaudio, like the following:
@@ -1384,8 +3945,9 @@ A copy of stb_vorbis is included in the "extras" folder in the miniaud
-Built-in decoders are amalgamated into the implementation section of miniaudio. You can disable the built-in decoders by specifying one or more of the
-following options before the miniaudio implementation:
+Built-in decoders are amalgamated into the implementation section of miniaudio. You can disable the
+built-in decoders by specifying one or more of the following options before the miniaudio
+implementation:
@@ -1396,12 +3958,14 @@ following options before the miniaudio implementation:
#define MA_NO_FLAC
-Disabling built-in decoding libraries is useful if you use these libraries independantly of the ma_decoder API.
+Disabling built-in decoding libraries is useful if you use these libraries independantly of the
+ma_decoder API.
-A decoder can be initialized from a file with ma_decoder_init_file(), a block of memory with ma_decoder_init_memory(), or from data delivered via callbacks
-with ma_decoder_init(). Here is an example for loading a decoder from a file:
+A decoder can be initialized from a file with ma_decoder_init_file(), a block of memory with
+ma_decoder_init_memory(), or from data delivered via callbacks with ma_decoder_init(). Here is
+an example for loading a decoder from a file:
@@ -1418,8 +3982,9 @@ with ma_decoder_init(). Here is an e
ma_decoder_uninit(&decoder);
-When initializing a decoder, you can optionally pass in a pointer to a ma_decoder_config object (the NULL argument in the example above) which allows you
-to configure the output format, channel count, sample rate and channel map:
+When initializing a decoder, you can optionally pass in a pointer to a ma_decoder_config object
+(the NULL argument in the example above) which allows you to configure the output format, channel
+count, sample rate and channel map:
@@ -1428,12 +3993,14 @@ to configure the output format, channel count, sample rate and channel map:
ma_decoder_config config = ma_decoder_config_init(ma_format_f32, 2, 48000);
-When passing in NULL for decoder config in ma_decoder_init*(), the output format will be the same as that defined by the decoding backend.
+When passing in NULL for decoder config in ma_decoder_init*(), the output format will be the
+same as that defined by the decoding backend.
-Data is read from the decoder as PCM frames. This will return the number of PCM frames actually read. If the return value is less than the requested number of
-PCM frames it means you've reached the end:
+Data is read from the decoder as PCM frames. This will return the number of PCM frames actually
+read. If the return value is less than the requested number of PCM frames it means you've reached
+the end:
@@ -1466,8 +4033,10 @@ If you want to loop back to the start, you can simply seek back to the first PCM
ma_decoder_seek_to_pcm_frame(pDecoder, 0);
-When loading a decoder, miniaudio uses a trial and error technique to find the appropriate decoding backend. This can be unnecessarily inefficient if the type
-is already known. In this case you can use encodingFormat variable in the device config to specify a specific encoding format you want to decode:
+When loading a decoder, miniaudio uses a trial and error technique to find the appropriate decoding
+backend. This can be unnecessarily inefficient if the type is already known. In this case you can
+use encodingFormat variable in the device config to specify a specific encoding format you want
+to decode:
@@ -1480,7 +4049,101 @@ See the ma_encoding_format enum for
-The ma_decoder_init_file() API will try using the file extension to determine which decoding backend to prefer.
+The ma_decoder_init_file() API will try using the file extension to determine which decoding
+backend to prefer.
+
+
+
+
+
+
+
+
8.1. Custom Decoders
+
+It's possible to implement a custom decoder and plug it into miniaudio. This is extremely useful
+when you want to use the ma_decoder API, but need to support an encoding format that's not one of
+the stock formats supported by miniaudio. This can be put to particularly good use when using the
+ma_engine and/or ma_resource_manager APIs because they use ma_decoder internally. If, for
+example, you wanted to support Opus, you can do so with a custom decoder (there if a reference
+Opus decoder in the "extras" folder of the miniaudio repository which uses libopus + libopusfile).
+
+
+
+A custom decoder must implement a data source. A vtable called ma_decoding_backend_vtable needs
+to be implemented which is then passed into the decoder config:
+
+
+There are only two functions that must be implemented - onInit and onUninit. The other
+functions can be implemented for a small optimization for loading from a file path or memory. If
+these are not specified, miniaudio will deal with it for you via a generic implementation.
+
+
+
+When you initialize a custom data source (by implementing the onInit function in the vtable) you
+will need to output a pointer to a ma_data_source which implements your custom decoder. See the
+section about data sources for details on how to implemen this. Alternatively, see the
+"custom_decoders" example in the miniaudio repository.
+
+
+
+The onInit function takes a pointer to some callbacks for the purpose of reading raw audio data
+from some abitrary source. You'll use these functions to read from the raw data and perform the
+decoding. When you call them, you will pass in the pReadSeekTellUserData pointer to the relevant
+parameter.
+
+
+
+The pConfig parameter in onInit can be used to configure the backend if appropriate. It's only
+used as a hint and can be ignored. However, if any of the properties are relevant to your decoder,
+an optimal implementation will handle the relevant properties appropriately.
+
+
+
+If memory allocation is required, it should be done so via the specified allocation callbacks if
+possible (the pAllocationCallbacks parameter).
+
+
+
+If an error occurs when initializing the decoder, you should leave ppBackend unset, or set to
+NULL, and make sure everything is cleaned up appropriately and an appropriate result code returned.
+When multiple custom backends are specified, miniaudio will cycle through the vtables in the order
+they're listed in the array that's passed into the decoder config so it's important that your
+initialization routine is clean.
+
+
+
+When a decoder is uninitialized, the onUninit callback will be fired which will give you an
+opportunity to clean up and internal data.
@@ -1491,10 +4154,11 @@ The ma_decoder_init_file() API will
-
5. Encoding
+
9. Encoding
-The ma_encoding API is used for writing audio files. The only supported output format is WAV which is achieved via dr_wav which is amalgamated into the
-implementation section of miniaudio. This can be disabled by specifying the following option before the implementation of miniaudio:
+The ma_encoding API is used for writing audio files. The only supported output format is WAV
+which is achieved via dr_wav which is amalgamated into the implementation section of miniaudio.
+This can be disabled by specifying the following option before the implementation of miniaudio:
@@ -1503,14 +4167,15 @@ implementation section of miniaudio. This can be disabled by specifying the foll
#define MA_NO_WAV
-An encoder can be initialized to write to a file with ma_encoder_init_file() or from data delivered via callbacks with ma_encoder_init(). Below is an
-example for initializing an encoder to output to a file.
+An encoder can be initialized to write to a file with ma_encoder_init_file() or from data
+delivered via callbacks with ma_encoder_init(). Below is an example for initializing an encoder
+to output to a file.
-ma_encoder_config config = ma_encoder_config_init(ma_resource_format_wav, FORMAT, CHANNELS, SAMPLE_RATE);
+ma_encoder_config config = ma_encoder_config_init(ma_encoding_format_wav, FORMAT, CHANNELS, SAMPLE_RATE);
ma_encoder encoder;
ma_result result = ma_encoder_init_file("my_file.wav", &config, &encoder);
if (result != MA_SUCCESS) {
@@ -1522,8 +4187,9 @@ example for initializing an encoder to output to a file.
ma_encoder_uninit(&encoder);
-When initializing an encoder you must specify a config which is initialized with ma_encoder_config_init(). Here you must specify the file type, the output
-sample format, output channel count and output sample rate. The following file types are supported:
+When initializing an encoder you must specify a config which is initialized with
+ma_encoder_config_init(). Here you must specify the file type, the output sample format, output
+channel count and output sample rate. The following file types are supported:
@@ -1538,15 +4204,17 @@ Description
-ma_resource_format_wav
+ma_encoding_format_wav
WAV
-If the format, channel count or sample rate is not supported by the output file type an error will be returned. The encoder will not perform data conversion so
-you will need to convert it before outputting any audio data. To output audio data, use ma_encoder_write_pcm_frames(), like in the example below:
+If the format, channel count or sample rate is not supported by the output file type an error will
+be returned. The encoder will not perform data conversion so you will need to convert it before
+outputting any audio data. To output audio data, use ma_encoder_write_pcm_frames(), like in the
+example below:
@@ -1563,10 +4231,14 @@ Encoders must be uninitialized with ma_enco
-
6. Data Conversion
-A data conversion API is included with miniaudio which supports the majority of data conversion requirements. This supports conversion between sample formats,
-channel counts (with channel mapping) and sample rates.
+
+
+
10. Data Conversion
+
+A data conversion API is included with miniaudio which supports the majority of data conversion
+requirements. This supports conversion between sample formats, channel counts (with channel
+mapping) and sample rates.
-Conversion between sample formats is achieved with the ma_pcm_*_to_*(), ma_pcm_convert() and ma_convert_pcm_frames_format() APIs. Use ma_pcm_*_to_*()
-to convert between two specific formats. Use ma_pcm_convert() to convert based on a ma_format variable. Use ma_convert_pcm_frames_format() to convert
-PCM frames where you want to specify the frame count and channel count as a variable instead of the total sample count.
+Conversion between sample formats is achieved with the ma_pcm_*_to_*(), ma_pcm_convert() and
+ma_convert_pcm_frames_format() APIs. Use ma_pcm_*_to_*() to convert between two specific
+formats. Use ma_pcm_convert() to convert based on a ma_format variable. Use
+ma_convert_pcm_frames_format() to convert PCM frames where you want to specify the frame count
+and channel count as a variable instead of the total sample count.
@@ -1586,7 +4260,7 @@ PCM frames where you want to specify the frame count and channel count as a vari
-
6.1.1. Dithering
+
10.1.1. Dithering
Dithering can be set using the ditherMode parameter.
@@ -1630,8 +4304,9 @@ ma_dither_mode_triangle
-Note that even if the dither mode is set to something other than ma_dither_mode_none, it will be ignored for conversions where dithering is not needed.
-Dithering is available for the following conversions:
+Note that even if the dither mode is set to something other than ma_dither_mode_none, it will be
+ignored for conversions where dithering is not needed. Dithering is available for the following
+conversions:
@@ -1646,7 +4321,8 @@ s32 -> s16
f32 -> s16
-Note that it is not an error to pass something other than ma_dither_mode_none for conversions where dither is not used. It will just be ignored.
+Note that it is not an error to pass something other than ma_dither_mode_none for conversions where
+dither is not used. It will just be ignored.
@@ -1657,10 +4333,11 @@ Note that it is not an error to pass something other than ma_dither_mode_none fo
-
6.2. Channel Conversion
+
10.2. Channel Conversion
-Channel conversion is used for channel rearrangement and conversion from one channel count to another. The ma_channel_converter API is used for channel
-conversion. Below is an example of initializing a simple channel converter which converts from mono to stereo.
+Channel conversion is used for channel rearrangement and conversion from one channel count to
+another. The ma_channel_converter API is used for channel conversion. Below is an example of
+initializing a simple channel converter which converts from mono to stereo.
@@ -1692,7 +4369,8 @@ To perform the conversion simply call ma_ch
}
-It is up to the caller to ensure the output buffer is large enough to accomodate the new PCM frames.
+It is up to the caller to ensure the output buffer is large enough to accomodate the new PCM
+frames.
@@ -1704,39 +4382,47 @@ Input and output PCM frames are always interleaved. Deinterleaved layouts are no
-
6.2.1. Channel Mapping
+
10.2.1. Channel Mapping
-In addition to converting from one channel count to another, like the example above, the channel converter can also be used to rearrange channels. When
-initializing the channel converter, you can optionally pass in channel maps for both the input and output frames. If the channel counts are the same, and each
-channel map contains the same channel positions with the exception that they're in a different order, a simple shuffling of the channels will be performed. If,
-however, there is not a 1:1 mapping of channel positions, or the channel counts differ, the input channels will be mixed based on a mixing mode which is
-specified when initializing the ma_channel_converter_config object.
+In addition to converting from one channel count to another, like the example above, the channel
+converter can also be used to rearrange channels. When initializing the channel converter, you can
+optionally pass in channel maps for both the input and output frames. If the channel counts are the
+same, and each channel map contains the same channel positions with the exception that they're in
+a different order, a simple shuffling of the channels will be performed. If, however, there is not
+a 1:1 mapping of channel positions, or the channel counts differ, the input channels will be mixed
+based on a mixing mode which is specified when initializing the ma_channel_converter_config
+object.
-When converting from mono to multi-channel, the mono channel is simply copied to each output channel. When going the other way around, the audio of each output
-channel is simply averaged and copied to the mono channel.
+When converting from mono to multi-channel, the mono channel is simply copied to each output
+channel. When going the other way around, the audio of each output channel is simply averaged and
+copied to the mono channel.
-In more complicated cases blending is used. The ma_channel_mix_mode_simple mode will drop excess channels and silence extra channels. For example, converting
-from 4 to 2 channels, the 3rd and 4th channels will be dropped, whereas converting from 2 to 4 channels will put silence into the 3rd and 4th channels.
+In more complicated cases blending is used. The ma_channel_mix_mode_simple mode will drop excess
+channels and silence extra channels. For example, converting from 4 to 2 channels, the 3rd and 4th
+channels will be dropped, whereas converting from 2 to 4 channels will put silence into the 3rd and
+4th channels.
-The ma_channel_mix_mode_rectangle mode uses spacial locality based on a rectangle to compute a simple distribution between input and output. Imagine sitting
-in the middle of a room, with speakers on the walls representing channel positions. The MA_CHANNEL_FRONT_LEFT position can be thought of as being in the corner
-of the front and left walls.
+The ma_channel_mix_mode_rectangle mode uses spacial locality based on a rectangle to compute a
+simple distribution between input and output. Imagine sitting in the middle of a room, with
+speakers on the walls representing channel positions. The MA_CHANNEL_FRONT_LEFT position can be
+thought of as being in the corner of the front and left walls.
-Finally, the ma_channel_mix_mode_custom_weights mode can be used to use custom user-defined weights. Custom weights can be passed in as the last parameter of
+Finally, the ma_channel_mix_mode_custom_weights mode can be used to use custom user-defined
+weights. Custom weights can be passed in as the last parameter of
ma_channel_converter_config_init().
-Predefined channel maps can be retrieved with ma_get_standard_channel_map(). This takes a ma_standard_channel_map enum as it's first parameter, which can
-be one of the following:
+Predefined channel maps can be retrieved with ma_channel_map_init_standard(). This takes a
+ma_standard_channel_map enum as it's first parameter, which can be one of the following:
@@ -1991,9 +4677,10 @@ mapping as the device.
-
6.3. Resampling
+
10.3. Resampling
-Resampling is achieved with the ma_resampler object. To create a resampler object, do something like the following:
+Resampling is achieved with the ma_resampler object. To create a resampler object, do something
+like the following:
@@ -2039,26 +4726,30 @@ The following example shows how data can be processed
// number of output frames written.
-To initialize the resampler you first need to set up a config (ma_resampler_config) with ma_resampler_config_init(). You need to specify the sample format
-you want to use, the number of channels, the input and output sample rate, and the algorithm.
+To initialize the resampler you first need to set up a config (ma_resampler_config) with
+ma_resampler_config_init(). You need to specify the sample format you want to use, the number of
+channels, the input and output sample rate, and the algorithm.
-The sample format can be either ma_format_s16 or ma_format_f32. If you need a different format you will need to perform pre- and post-conversions yourself
-where necessary. Note that the format is the same for both input and output. The format cannot be changed after initialization.
+The sample format can be either ma_format_s16 or ma_format_f32. If you need a different format
+you will need to perform pre- and post-conversions yourself where necessary. Note that the format
+is the same for both input and output. The format cannot be changed after initialization.
-The resampler supports multiple channels and is always interleaved (both input and output). The channel count cannot be changed after initialization.
+The resampler supports multiple channels and is always interleaved (both input and output). The
+channel count cannot be changed after initialization.
-The sample rates can be anything other than zero, and are always specified in hertz. They should be set to something like 44100, etc. The sample rate is the
-only configuration property that can be changed after initialization.
+The sample rates can be anything other than zero, and are always specified in hertz. They should be
+set to something like 44100, etc. The sample rate is the only configuration property that can be
+changed after initialization.
-The miniaudio resampler supports multiple algorithms:
+The miniaudio resampler has built-in support for the following algorithms:
-Because Speex is not public domain it is strictly opt-in and the code is stored in separate files. if you opt-in to the Speex backend you will need to consider
-it's license, the text of which can be found in it's source files in "extras/speex_resampler". Details on how to opt-in to the Speex resampler is explained in
-the Speex Resampler section below.
-
-
-
The algorithm cannot be changed after initialization.
-Processing always happens on a per PCM frame basis and always assumes interleaved input and output. De-interleaved processing is not supported. To process
-frames, use ma_resampler_process_pcm_frames(). On input, this function takes the number of output frames you can fit in the output buffer and the number of
-input frames contained in the input buffer. On output these variables contain the number of output frames that were written to the output buffer and the
-number of input frames that were consumed in the process. You can pass in NULL for the input buffer in which case it will be treated as an infinitely large
-buffer of zeros. The output buffer can also be NULL, in which case the processing will be treated as seek.
+Processing always happens on a per PCM frame basis and always assumes interleaved input and output.
+De-interleaved processing is not supported. To process frames, use
+ma_resampler_process_pcm_frames(). On input, this function takes the number of output frames you
+can fit in the output buffer and the number of input frames contained in the input buffer. On
+output these variables contain the number of output frames that were written to the output buffer
+and the number of input frames that were consumed in the process. You can pass in NULL for the
+input buffer in which case it will be treated as an infinitely large buffer of zeros. The output
+buffer can also be NULL, in which case the processing will be treated as seek.
-The sample rate can be changed dynamically on the fly. You can change this with explicit sample rates with ma_resampler_set_rate() and also with a decimal
-ratio with ma_resampler_set_rate_ratio(). The ratio is in/out.
+The sample rate can be changed dynamically on the fly. You can change this with explicit sample
+rates with ma_resampler_set_rate() and also with a decimal ratio with
+ma_resampler_set_rate_ratio(). The ratio is in/out.
-Sometimes it's useful to know exactly how many input frames will be required to output a specific number of frames. You can calculate this with
-ma_resampler_get_required_input_frame_count(). Likewise, it's sometimes useful to know exactly how many frames would be output given a certain number of
-input frames. You can do this with ma_resampler_get_expected_output_frame_count().
+Sometimes it's useful to know exactly how many input frames will be required to output a specific
+number of frames. You can calculate this with ma_resampler_get_required_input_frame_count().
+Likewise, it's sometimes useful to know exactly how many frames would be output given a certain
+number of input frames. You can do this with ma_resampler_get_expected_output_frame_count().
-Due to the nature of how resampling works, the resampler introduces some latency. This can be retrieved in terms of both the input rate and the output rate
-with ma_resampler_get_input_latency() and ma_resampler_get_output_latency().
+Due to the nature of how resampling works, the resampler introduces some latency. This can be
+retrieved in terms of both the input rate and the output rate with
+ma_resampler_get_input_latency() and ma_resampler_get_output_latency().
@@ -2126,11 +4817,9 @@ with ma_resampler_get_input_latency()
-
6.3.1. Resampling Algorithms
+
10.3.1. Resampling Algorithms
-The choice of resampling algorithm depends on your situation and requirements. The linear resampler is the most efficient and has the least amount of latency,
-but at the expense of poorer quality. The Speex resampler is higher quality, but slower with more latency. It also performs several heap allocations internally
-for memory management.
+The choice of resampling algorithm depends on your situation and requirements.
@@ -2138,28 +4827,29 @@ for memory management.
-
6.3.1.1. Linear Resampling
+
10.3.1.1. Linear Resampling
-The linear resampler is the fastest, but comes at the expense of poorer quality. There is, however, some control over the quality of the linear resampler which
-may make it a suitable option depending on your requirements.
+The linear resampler is the fastest, but comes at the expense of poorer quality. There is, however,
+some control over the quality of the linear resampler which may make it a suitable option depending
+on your requirements.
-The linear resampler performs low-pass filtering before or after downsampling or upsampling, depending on the sample rates you're converting between. When
-decreasing the sample rate, the low-pass filter will be applied before downsampling. When increasing the rate it will be performed after upsampling. By default
-a fourth order low-pass filter will be applied. This can be configured via the lpfOrder configuration variable. Setting this to 0 will disable filtering.
+The linear resampler performs low-pass filtering before or after downsampling or upsampling,
+depending on the sample rates you're converting between. When decreasing the sample rate, the
+low-pass filter will be applied before downsampling. When increasing the rate it will be performed
+after upsampling. By default a fourth order low-pass filter will be applied. This can be configured
+via the lpfOrder configuration variable. Setting this to 0 will disable filtering.
-The low-pass filter has a cutoff frequency which defaults to half the sample rate of the lowest of the input and output sample rates (Nyquist Frequency). This
-can be controlled with the lpfNyquistFactor config variable. This defaults to 1, and should be in the range of 0..1, although a value of 0 does not make
-sense and should be avoided. A value of 1 will use the Nyquist Frequency as the cutoff. A value of 0.5 will use half the Nyquist Frequency as the cutoff, etc.
-Values less than 1 will result in more washed out sound due to more of the higher frequencies being removed. This config variable has no impact on performance
-and is a purely perceptual configuration.
+The low-pass filter has a cutoff frequency which defaults to half the sample rate of the lowest of
+the input and output sample rates (Nyquist Frequency).
-The API for the linear resampler is the same as the main resampler API, only it's called ma_linear_resampler.
+The API for the linear resampler is the same as the main resampler API, only it's called
+ma_linear_resampler.
@@ -2167,36 +4857,58 @@ The API for the linear resampler is the same as the main resampler API, only it&
-
6.3.1.2. Speex Resampling
+
10.3.2. Custom Resamplers
-The Speex resampler is made up of third party code which is released under the BSD license. Because it is licensed differently to miniaudio, which is public
-domain, it is strictly opt-in and all of it's code is stored in separate files. If you opt-in to the Speex resampler you must consider the license text in it's
-source files. To opt-in, you must first #include the following file before the implementation of miniaudio.h:
+You can implement a custom resampler by using the ma_resample_algorithm_custom resampling
+algorithm and setting a vtable in the resampler config:
-Both the header and implementation is contained within the same file. The implementation can be included in your program like so:
+Custom resamplers are useful if the stock algorithms are not appropriate for your use case. You
+need to implement the required functions in ma_resampling_backend_vtable. Note that not all
+functions in the vtable need to be implemented, but if it's possible to implement, they should be.
-
-Note that even if you opt-in to the Speex backend, miniaudio won't use it unless you explicitly ask for it in the respective config of the object you are
-initializing. If you try to use the Speex resampler without opting in, initialization of the ma_resampler object will fail with MA_NO_BACKEND.
+You can use the ma_linear_resampler object for an example on how to implement the vtable. The
+onGetHeapSize callback is used to calculate the size of any internal heap allocation the custom
+resampler will need to make given the supplied config. When you initialize the resampler via the
+onInit callback, you'll be given a pointer to a heap allocation which is where you should store
+the heap allocated data. You should not free this data in onUninit because miniaudio will manage
+it for you.
-The only configuration option to consider with the Speex resampler is the speex.quality config variable. This is a value between 0 and 10, with 0 being
-the fastest with the poorest quality and 10 being the slowest with the highest quality. The default value is 3.
+The onProcess callback is where the actual resampling takes place. On input, pFrameCountIn
+points to a variable containing the number of frames in the pFramesIn buffer and
+pFrameCountOut points to a variable containing the capacity in frames of the pFramesOut buffer.
+On output, pFrameCountIn should be set to the number of input frames that were fully consumed,
+whereas pFrameCountOut should be set to the number of frames that were written to pFramesOut.
+
+
+
+The onSetRate callback is optional and is used for dynamically changing the sample rate. If
+dynamic rate changes are not supported, you can set this callback to NULL.
+
+
+
+The onGetInputLatency and onGetOutputLatency functions are used for retrieving the latency in
+input and output rates respectively. These can be NULL in which case latency calculations will be
+assumed to be NULL.
+
+
+
+The onGetRequiredInputFrameCount callback is used to give miniaudio a hint as to how many input
+frames are required to be available to produce the given number of output frames. Likewise, the
+onGetExpectedOutputFrameCount callback is used to determine how many output frames will be
+produced given the specified number of input frames. miniaudio will use these as a hint, but they
+are optional and can be set to NULL if you're unable to implement them.
@@ -2207,11 +4919,13 @@ the fastest with the poorest quality and 10 being the slowest with the highest q
-
6.4. General Data Conversion
+
10.4. General Data Conversion
-The ma_data_converter API can be used to wrap sample format conversion, channel conversion and resampling into one operation. This is what miniaudio uses
-internally to convert between the format requested when the device was initialized and the format of the backend's native device. The API for general data
-conversion is very similar to the resampling API. Create a ma_data_converter object like this:
+The ma_data_converter API can be used to wrap sample format conversion, channel conversion and
+resampling into one operation. This is what miniaudio uses internally to convert between the format
+requested when the device was initialized and the format of the backend's native device. The API
+for general data conversion is very similar to the resampling API. Create a ma_data_converter
+object like this:
@@ -2233,8 +4947,9 @@ conversion is very similar to the resampling API. Create a ma_data_converter_config_init() to initialize the config, however there's many more properties that can be configured, such as
-channel maps and resampling quality. Something like the following may be more suitable depending on your requirements:
+In the example above we use ma_data_converter_config_init() to initialize the config, however
+there's many more properties that can be configured, such as channel maps and resampling quality.
+Something like the following may be more suitable depending on your requirements:
@@ -2277,33 +4992,42 @@ The following example shows how data can be processed
// of output frames written.
-The data converter supports multiple channels and is always interleaved (both input and output). The channel count cannot be changed after initialization.
+The data converter supports multiple channels and is always interleaved (both input and output).
+The channel count cannot be changed after initialization.
-Sample rates can be anything other than zero, and are always specified in hertz. They should be set to something like 44100, etc. The sample rate is the only
-configuration property that can be changed after initialization, but only if the resampling.allowDynamicSampleRate member of ma_data_converter_config is
-set to MA_TRUE. To change the sample rate, use ma_data_converter_set_rate() or ma_data_converter_set_rate_ratio(). The ratio must be in/out. The
-resampling algorithm cannot be changed after initialization.
+Sample rates can be anything other than zero, and are always specified in hertz. They should be set
+to something like 44100, etc. The sample rate is the only configuration property that can be
+changed after initialization, but only if the resampling.allowDynamicSampleRate member of
+ma_data_converter_config is set to MA_TRUE. To change the sample rate, use
+ma_data_converter_set_rate() or ma_data_converter_set_rate_ratio(). The ratio must be in/out.
+The resampling algorithm cannot be changed after initialization.
-Processing always happens on a per PCM frame basis and always assumes interleaved input and output. De-interleaved processing is not supported. To process
-frames, use ma_data_converter_process_pcm_frames(). On input, this function takes the number of output frames you can fit in the output buffer and the number
-of input frames contained in the input buffer. On output these variables contain the number of output frames that were written to the output buffer and the
-number of input frames that were consumed in the process. You can pass in NULL for the input buffer in which case it will be treated as an infinitely large
-buffer of zeros. The output buffer can also be NULL, in which case the processing will be treated as seek.
+Processing always happens on a per PCM frame basis and always assumes interleaved input and output.
+De-interleaved processing is not supported. To process frames, use
+ma_data_converter_process_pcm_frames(). On input, this function takes the number of output frames
+you can fit in the output buffer and the number of input frames contained in the input buffer. On
+output these variables contain the number of output frames that were written to the output buffer
+and the number of input frames that were consumed in the process. You can pass in NULL for the
+input buffer in which case it will be treated as an infinitely large
+buffer of zeros. The output buffer can also be NULL, in which case the processing will be treated
+as seek.
-Sometimes it's useful to know exactly how many input frames will be required to output a specific number of frames. You can calculate this with
-ma_data_converter_get_required_input_frame_count(). Likewise, it's sometimes useful to know exactly how many frames would be output given a certain number of
-input frames. You can do this with ma_data_converter_get_expected_output_frame_count().
+Sometimes it's useful to know exactly how many input frames will be required to output a specific
+number of frames. You can calculate this with ma_data_converter_get_required_input_frame_count().
+Likewise, it's sometimes useful to know exactly how many frames would be output given a certain
+number of input frames. You can do this with ma_data_converter_get_expected_output_frame_count().
-Due to the nature of how resampling works, the data converter introduces some latency if resampling is required. This can be retrieved in terms of both the
-input rate and the output rate with ma_data_converter_get_input_latency() and ma_data_converter_get_output_latency().
+Due to the nature of how resampling works, the data converter introduces some latency if resampling
+is required. This can be retrieved in terms of both the input rate and the output rate with
+ma_data_converter_get_input_latency() and ma_data_converter_get_output_latency().
@@ -2314,11 +5038,11 @@ input rate and the output rate with ma_data
-
7. Filtering
+
11. Filtering
-
7.1. Biquad Filtering
+
11.1. Biquad Filtering
Biquad filtering is achieved with the ma_biquad API. Example:
@@ -2337,13 +5061,15 @@ Biquad filtering is achieved with the ma_bi
ma_biquad_process_pcm_frames(&biquad, pFramesOut, pFramesIn, frameCount);
-Biquad filtering is implemented using transposed direct form 2. The numerator coefficients are b0, b1 and b2, and the denominator coefficients are a0, a1 and
-a2. The a0 coefficient is required and coefficients must not be pre-normalized.
+Biquad filtering is implemented using transposed direct form 2. The numerator coefficients are b0,
+b1 and b2, and the denominator coefficients are a0, a1 and a2. The a0 coefficient is required and
+coefficients must not be pre-normalized.
-Supported formats are ma_format_s16 and ma_format_f32. If you need to use a different format you need to convert it yourself beforehand. When using
-ma_format_s16 the biquad filter will use fixed point arithmetic. When using ma_format_f32, floating point arithmetic will be used.
+Supported formats are ma_format_s16 and ma_format_f32. If you need to use a different format
+you need to convert it yourself beforehand. When using ma_format_s16 the biquad filter will use
+fixed point arithmetic. When using ma_format_f32, floating point arithmetic will be used.
@@ -2351,7 +5077,8 @@ Input and output frames are always interleaved.
-Filtering can be applied in-place by passing in the same pointer for both the input and output buffers, like so:
+Filtering can be applied in-place by passing in the same pointer for both the input and output
+buffers, like so:
@@ -2360,10 +5087,12 @@ Filtering can be applied in-place by passing in the same pointer for both the in
ma_biquad_process_pcm_frames(&biquad, pMyData, pMyData, frameCount);
-If you need to change the values of the coefficients, but maintain the values in the registers you can do so with ma_biquad_reinit(). This is useful if you
-need to change the properties of the filter while keeping the values of registers valid to avoid glitching. Do not use ma_biquad_init() for this as it will
-do a full initialization which involves clearing the registers to 0. Note that changing the format or channel count after initialization is invalid and will
-result in an error.
+If you need to change the values of the coefficients, but maintain the values in the registers you
+can do so with ma_biquad_reinit(). This is useful if you need to change the properties of the
+filter while keeping the values of registers valid to avoid glitching. Do not use
+ma_biquad_init() for this as it will do a full initialization which involves clearing the
+registers to 0. Note that changing the format or channel count after initialization is invalid and
+will result in an error.
@@ -2371,7 +5100,7 @@ result in an error.
-
7.2. Low-Pass Filtering
+
11.2. Low-Pass Filtering
Low-pass filtering is achieved with the following APIs:
-Supported formats are ma_format_s16 and ma_format_f32. If you need to use a different format you need to convert it yourself beforehand. Input and output
-frames are always interleaved.
+Supported formats are ma_format_s16 and ma_format_f32. If you need to use a different format
+you need to convert it yourself beforehand. Input and output frames are always interleaved.
-Filtering can be applied in-place by passing in the same pointer for both the input and output buffers, like so:
+Filtering can be applied in-place by passing in the same pointer for both the input and output
+buffers, like so:
@@ -2442,7 +5172,8 @@ Filtering can be applied in-place by passing in the same pointer for both the in
ma_lpf_process_pcm_frames(&lpf, pMyData, pMyData, frameCount);
-The maximum filter order is limited to MA_MAX_FILTER_ORDER which is set to 8. If you need more, you can chain first and second order filters together.
+The maximum filter order is limited to MA_MAX_FILTER_ORDER which is set to 8. If you need more,
+you can chain first and second order filters together.
@@ -2453,19 +5184,22 @@ The maximum filter order is limited to MA_M
}
-If you need to change the configuration of the filter, but need to maintain the state of internal registers you can do so with ma_lpf_reinit(). This may be
-useful if you need to change the sample rate and/or cutoff frequency dynamically while maintaing smooth transitions. Note that changing the format or channel
-count after initialization is invalid and will result in an error.
+If you need to change the configuration of the filter, but need to maintain the state of internal
+registers you can do so with ma_lpf_reinit(). This may be useful if you need to change the sample
+rate and/or cutoff frequency dynamically while maintaing smooth transitions. Note that changing the
+format or channel count after initialization is invalid and will result in an error.
-The ma_lpf object supports a configurable order, but if you only need a first order filter you may want to consider using ma_lpf1. Likewise, if you only
-need a second order filter you can use ma_lpf2. The advantage of this is that they're lighter weight and a bit more efficient.
+The ma_lpf object supports a configurable order, but if you only need a first order filter you
+may want to consider using ma_lpf1. Likewise, if you only need a second order filter you can use
+ma_lpf2. The advantage of this is that they're lighter weight and a bit more efficient.
-If an even filter order is specified, a series of second order filters will be processed in a chain. If an odd filter order is specified, a first order filter
-will be applied, followed by a series of second order filters in a chain.
+If an even filter order is specified, a series of second order filters will be processed in a
+chain. If an odd filter order is specified, a first order filter will be applied, followed by a
+series of second order filters in a chain.
@@ -2473,7 +5207,7 @@ will be applied, followed by a series of second order filters in a chain.
-
7.3. High-Pass Filtering
+
11.3. High-Pass Filtering
High-pass filtering is achieved with the following APIs:
@@ -2513,8 +5247,8 @@ High order high-pass filter (Butterworth)
-High-pass filters work exactly the same as low-pass filters, only the APIs are called ma_hpf1, ma_hpf2 and ma_hpf. See example code for low-pass filters
-for example usage.
+High-pass filters work exactly the same as low-pass filters, only the APIs are called ma_hpf1,
+ma_hpf2 and ma_hpf. See example code for low-pass filters for example usage.
@@ -2522,7 +5256,7 @@ for example usage.
-
7.4. Band-Pass Filtering
+
11.4. Band-Pass Filtering
Band-pass filtering is achieved with the following APIs:
@@ -2554,9 +5288,10 @@ High order band-pass filter
-Band-pass filters work exactly the same as low-pass filters, only the APIs are called ma_bpf2 and ma_hpf. See example code for low-pass filters for example
-usage. Note that the order for band-pass filters must be an even number which means there is no first order band-pass filter, unlike low-pass and high-pass
-filters.
+Band-pass filters work exactly the same as low-pass filters, only the APIs are called ma_bpf2 and
+ma_hpf. See example code for low-pass filters for example usage. Note that the order for
+band-pass filters must be an even number which means there is no first order band-pass filter,
+unlike low-pass and high-pass filters.
@@ -2564,7 +5299,7 @@ filters.
-
7.5. Notch Filtering
+
11.5. Notch Filtering
Notch filtering is achieved with the following APIs:
@@ -2590,7 +5325,7 @@ Second order notching filter
-
7.6. Peaking EQ Filtering
+
11.6. Peaking EQ Filtering
Peaking filtering is achieved with the following APIs:
@@ -2616,7 +5351,7 @@ Second order peaking filter
-
7.7. Low Shelf Filtering
+
11.7. Low Shelf Filtering
Low shelf filtering is achieved with the following APIs:
@@ -2640,7 +5375,8 @@ Second order low shelf filter
-Where a high-pass filter is used to eliminate lower frequencies, a low shelf filter can be used to just turn them down rather than eliminate them entirely.
+Where a high-pass filter is used to eliminate lower frequencies, a low shelf filter can be used to
+just turn them down rather than eliminate them entirely.
@@ -2648,7 +5384,7 @@ Where a high-pass filter is used to eliminate lower frequencies, a low shelf fil
-
7.8. High Shelf Filtering
+
11.8. High Shelf Filtering
High shelf filtering is achieved with the following APIs:
@@ -2672,8 +5408,9 @@ Second order high shelf filter
-The high shelf filter has the same API as the low shelf filter, only you would use ma_hishelf instead of ma_loshelf. Where a low shelf filter is used to
-adjust the volume of low frequencies, the high shelf filter does the same thing for high frequencies.
+The high shelf filter has the same API as the low shelf filter, only you would use ma_hishelf
+instead of ma_loshelf. Where a low shelf filter is used to adjust the volume of low frequencies,
+the high shelf filter does the same thing for high frequencies.
@@ -2687,13 +5424,14 @@ adjust the volume of low frequencies, the high shelf filter does the same thing
-
8. Waveform and Noise Generation
+
12. Waveform and Noise Generation
-
8.1. Waveforms
+
12.1. Waveforms
-miniaudio supports generation of sine, square, triangle and sawtooth waveforms. This is achieved with the ma_waveform API. Example:
+miniaudio supports generation of sine, square, triangle and sawtooth waveforms. This is achieved
+with the ma_waveform API. Example:
@@ -2718,13 +5456,14 @@ miniaudio supports generation of sine, square, triangle and sawtooth waveforms.
ma_waveform_read_pcm_frames(&waveform, pOutput, frameCount);
-The amplitude, frequency, type, and sample rate can be changed dynamically with ma_waveform_set_amplitude(), ma_waveform_set_frequency(),
-ma_waveform_set_type(), and ma_waveform_set_sample_rate() respectively.
+The amplitude, frequency, type, and sample rate can be changed dynamically with
+ma_waveform_set_amplitude(), ma_waveform_set_frequency(), ma_waveform_set_type(), and
+ma_waveform_set_sample_rate() respectively.
-You can invert the waveform by setting the amplitude to a negative value. You can use this to control whether or not a sawtooth has a positive or negative
-ramp, for example.
+You can invert the waveform by setting the amplitude to a negative value. You can use this to
+control whether or not a sawtooth has a positive or negative ramp, for example.
@@ -2764,7 +5503,7 @@ ma_waveform_type_sawtooth
-
8.2. Noise
+
12.2. Noise
miniaudio supports generation of white, pink and Brownian noise via the ma_noise API. Example:
@@ -2790,17 +5529,20 @@ miniaudio supports generation of white, pink and Brownian noise via the
-The noise API uses simple LCG random number generation. It supports a custom seed which is useful for things like automated testing requiring reproducibility.
-Setting the seed to zero will default to MA_DEFAULT_LCG_SEED.
+The noise API uses simple LCG random number generation. It supports a custom seed which is useful
+for things like automated testing requiring reproducibility. Setting the seed to zero will default
+to MA_DEFAULT_LCG_SEED.
-The amplitude, seed, and type can be changed dynamically with ma_noise_set_amplitude(), ma_noise_set_seed(), and ma_noise_set_type() respectively.
+The amplitude, seed, and type can be changed dynamically with ma_noise_set_amplitude(),
+ma_noise_set_seed(), and ma_noise_set_type() respectively.
-By default, the noise API will use different values for different channels. So, for example, the left side in a stereo stream will be different to the right
-side. To instead have each channel use the same random value, set the duplicateChannels member of the noise config to true, like so:
+By default, the noise API will use different values for different channels. So, for example, the
+left side in a stereo stream will be different to the right side. To instead have each channel use
+the same random value, set the duplicateChannels member of the noise config to true, like so:
@@ -2840,10 +5582,11 @@ ma_noise_type_brownian
-
9. Audio Buffers
+
13. Audio Buffers
-miniaudio supports reading from a buffer of raw audio data via the ma_audio_buffer API. This can read from memory that's managed by the application, but
-can also handle the memory management for you internally. Memory management is flexible and should support most use cases.
+miniaudio supports reading from a buffer of raw audio data via the ma_audio_buffer API. This can
+read from memory that's managed by the application, but can also handle the memory management for
+you internally. Memory management is flexible and should support most use cases.
@@ -2871,13 +5614,16 @@ result = ma_audio_buffer_init(&config, &buffer);
ma_audio_buffer_uninit(&buffer);
-In the example above, the memory pointed to by pExistingData will *not* be copied and is how an application can do self-managed memory allocation. If you
-would rather make a copy of the data, use ma_audio_buffer_init_copy(). To uninitialize the buffer, use ma_audio_buffer_uninit().
+In the example above, the memory pointed to by pExistingData will *not* be copied and is how an
+application can do self-managed memory allocation. If you would rather make a copy of the data, use
+ma_audio_buffer_init_copy(). To uninitialize the buffer, use ma_audio_buffer_uninit().
-Sometimes it can be convenient to allocate the memory for the ma_audio_buffer structure and the raw audio data in a contiguous block of memory. That is,
-the raw audio data will be located immediately after the ma_audio_buffer structure. To do this, use ma_audio_buffer_alloc_and_init():
+Sometimes it can be convenient to allocate the memory for the ma_audio_buffer structure and the
+raw audio data in a contiguous block of memory. That is, the raw audio data will be located
+immediately after the ma_audio_buffer structure. To do this, use
+ma_audio_buffer_alloc_and_init():
@@ -2901,15 +5647,20 @@ result = ma_audio_buffer_alloc_and_init(&config, &pBuffer);
ma_audio_buffer_uninit_and_free(&buffer);
-If you initialize the buffer with ma_audio_buffer_alloc_and_init() you should uninitialize it with ma_audio_buffer_uninit_and_free(). In the example above,
-the memory pointed to by pExistingData will be copied into the buffer, which is contrary to the behavior of ma_audio_buffer_init().
+If you initialize the buffer with ma_audio_buffer_alloc_and_init() you should uninitialize it
+with ma_audio_buffer_uninit_and_free(). In the example above, the memory pointed to by
+pExistingData will be copied into the buffer, which is contrary to the behavior of
+ma_audio_buffer_init().
-An audio buffer has a playback cursor just like a decoder. As you read frames from the buffer, the cursor moves forward. The last parameter (loop) can be
-used to determine if the buffer should loop. The return value is the number of frames actually read. If this is less than the number of frames requested it
-means the end has been reached. This should never happen if the loop parameter is set to true. If you want to manually loop back to the start, you can do so
-with with ma_audio_buffer_seek_to_pcm_frame(pAudioBuffer, 0). Below is an example for reading data from an audio buffer.
+An audio buffer has a playback cursor just like a decoder. As you read frames from the buffer, the
+cursor moves forward. The last parameter (loop) can be used to determine if the buffer should
+loop. The return value is the number of frames actually read. If this is less than the number of
+frames requested it means the end has been reached. This should never happen if the loop
+parameter is set to true. If you want to manually loop back to the start, you can do so with with
+ma_audio_buffer_seek_to_pcm_frame(pAudioBuffer, 0). Below is an example for reading data from an
+audio buffer.
@@ -2921,8 +5672,8 @@ with with ma_audio_buffer_seek_to_pcm_frame
}
-Sometimes you may want to avoid the cost of data movement between the internal buffer and the output buffer. Instead you can use memory mapping to retrieve a
-pointer to a segment of data:
+Sometimes you may want to avoid the cost of data movement between the internal buffer and the
+output buffer. Instead you can use memory mapping to retrieve a pointer to a segment of data:
@@ -2941,10 +5692,13 @@ pointer to a segment of data:
}
-When you use memory mapping, the read cursor is increment by the frame count passed in to ma_audio_buffer_unmap(). If you decide not to process every frame
-you can pass in a value smaller than the value returned by ma_audio_buffer_map(). The disadvantage to using memory mapping is that it does not handle looping
-for you. You can determine if the buffer is at the end for the purpose of looping with ma_audio_buffer_at_end() or by inspecting the return value of
-ma_audio_buffer_unmap() and checking if it equals MA_AT_END. You should not treat MA_AT_END as an error when returned by ma_audio_buffer_unmap().
+When you use memory mapping, the read cursor is increment by the frame count passed in to
+ma_audio_buffer_unmap(). If you decide not to process every frame you can pass in a value smaller
+than the value returned by ma_audio_buffer_map(). The disadvantage to using memory mapping is
+that it does not handle looping for you. You can determine if the buffer is at the end for the
+purpose of looping with ma_audio_buffer_at_end() or by inspecting the return value of
+ma_audio_buffer_unmap() and checking if it equals MA_AT_END. You should not treat MA_AT_END
+as an error when returned by ma_audio_buffer_unmap().
@@ -2955,20 +5709,24 @@ for you. You can determine if the buffer is at the end for the purpose of loopin
-
10. Ring Buffers
+
14. Ring Buffers
-miniaudio supports lock free (single producer, single consumer) ring buffers which are exposed via the ma_rb and ma_pcm_rb APIs. The ma_rb API operates
-on bytes, whereas the ma_pcm_rb operates on PCM frames. They are otherwise identical as ma_pcm_rb is just a wrapper around ma_rb.
+miniaudio supports lock free (single producer, single consumer) ring buffers which are exposed via
+the ma_rb and ma_pcm_rb APIs. The ma_rb API operates on bytes, whereas the ma_pcm_rb
+operates on PCM frames. They are otherwise identical as ma_pcm_rb is just a wrapper around
+ma_rb.
-Unlike most other APIs in miniaudio, ring buffers support both interleaved and deinterleaved streams. The caller can also allocate their own backing memory for
-the ring buffer to use internally for added flexibility. Otherwise the ring buffer will manage it's internal memory for you.
+Unlike most other APIs in miniaudio, ring buffers support both interleaved and deinterleaved
+streams. The caller can also allocate their own backing memory for the ring buffer to use
+internally for added flexibility. Otherwise the ring buffer will manage it's internal memory for
+you.
-The examples below use the PCM frame variant of the ring buffer since that's most likely the one you will want to use. To initialize a ring buffer, do
-something like the following:
+The examples below use the PCM frame variant of the ring buffer since that's most likely the one
+you will want to use. To initialize a ring buffer, do something like the following:
@@ -2981,49 +5739,63 @@ something like the following:
}
-The ma_pcm_rb_init() function takes the sample format and channel count as parameters because it's the PCM varient of the ring buffer API. For the regular
-ring buffer that operates on bytes you would call ma_rb_init() which leaves these out and just takes the size of the buffer in bytes instead of frames. The
-fourth parameter is an optional pre-allocated buffer and the fifth parameter is a pointer to a ma_allocation_callbacks structure for custom memory allocation
-routines. Passing in NULL for this results in MA_MALLOC() and MA_FREE() being used.
+The ma_pcm_rb_init() function takes the sample format and channel count as parameters because
+it's the PCM varient of the ring buffer API. For the regular ring buffer that operates on bytes you
+would call ma_rb_init() which leaves these out and just takes the size of the buffer in bytes
+instead of frames. The fourth parameter is an optional pre-allocated buffer and the fifth parameter
+is a pointer to a ma_allocation_callbacks structure for custom memory allocation routines.
+Passing in NULL for this results in MA_MALLOC() and MA_FREE() being used.
-Use ma_pcm_rb_init_ex() if you need a deinterleaved buffer. The data for each sub-buffer is offset from each other based on the stride. To manage your
-sub-buffers you can use ma_pcm_rb_get_subbuffer_stride(), ma_pcm_rb_get_subbuffer_offset() and ma_pcm_rb_get_subbuffer_ptr().
+Use ma_pcm_rb_init_ex() if you need a deinterleaved buffer. The data for each sub-buffer is
+offset from each other based on the stride. To manage your sub-buffers you can use
+ma_pcm_rb_get_subbuffer_stride(), ma_pcm_rb_get_subbuffer_offset() and
+ma_pcm_rb_get_subbuffer_ptr().
-Use ma_pcm_rb_acquire_read() and ma_pcm_rb_acquire_write() to retrieve a pointer to a section of the ring buffer. You specify the number of frames you
-need, and on output it will set to what was actually acquired. If the read or write pointer is positioned such that the number of frames requested will require
-a loop, it will be clamped to the end of the buffer. Therefore, the number of frames you're given may be less than the number you requested.
+Use ma_pcm_rb_acquire_read() and ma_pcm_rb_acquire_write() to retrieve a pointer to a section
+of the ring buffer. You specify the number of frames you need, and on output it will set to what
+was actually acquired. If the read or write pointer is positioned such that the number of frames
+requested will require a loop, it will be clamped to the end of the buffer. Therefore, the number
+of frames you're given may be less than the number you requested.
-After calling ma_pcm_rb_acquire_read() or ma_pcm_rb_acquire_write(), you do your work on the buffer and then "commit" it with ma_pcm_rb_commit_read() or
-ma_pcm_rb_commit_write(). This is where the read/write pointers are updated. When you commit you need to pass in the buffer that was returned by the earlier
-call to ma_pcm_rb_acquire_read() or ma_pcm_rb_acquire_write() and is only used for validation. The number of frames passed to ma_pcm_rb_commit_read() and
-ma_pcm_rb_commit_write() is what's used to increment the pointers, and can be less that what was originally requested.
+After calling ma_pcm_rb_acquire_read() or ma_pcm_rb_acquire_write(), you do your work on the
+buffer and then "commit" it with ma_pcm_rb_commit_read() or ma_pcm_rb_commit_write(). This is
+where the read/write pointers are updated. When you commit you need to pass in the buffer that was
+returned by the earlier call to ma_pcm_rb_acquire_read() or ma_pcm_rb_acquire_write() and is
+only used for validation. The number of frames passed to ma_pcm_rb_commit_read() and
+ma_pcm_rb_commit_write() is what's used to increment the pointers, and can be less that what was
+originally requested.
-If you want to correct for drift between the write pointer and the read pointer you can use a combination of ma_pcm_rb_pointer_distance(),
-ma_pcm_rb_seek_read() and ma_pcm_rb_seek_write(). Note that you can only move the pointers forward, and you should only move the read pointer forward via
-the consumer thread, and the write pointer forward by the producer thread. If there is too much space between the pointers, move the read pointer forward. If
+If you want to correct for drift between the write pointer and the read pointer you can use a
+combination of ma_pcm_rb_pointer_distance(), ma_pcm_rb_seek_read() and
+ma_pcm_rb_seek_write(). Note that you can only move the pointers forward, and you should only
+move the read pointer forward via the consumer thread, and the write pointer forward by the
+producer thread. If there is too much space between the pointers, move the read pointer forward. If
there is too little space between the pointers, move the write pointer forward.
-You can use a ring buffer at the byte level instead of the PCM frame level by using the ma_rb API. This is exactly the same, only you will use the ma_rb
-functions instead of ma_pcm_rb and instead of frame counts you will pass around byte counts.
+You can use a ring buffer at the byte level instead of the PCM frame level by using the ma_rb
+API. This is exactly the same, only you will use the ma_rb functions instead of ma_pcm_rb and
+instead of frame counts you will pass around byte counts.
-The maximum size of the buffer in bytes is 0x7FFFFFFF-(MA_SIMD_ALIGNMENT-1) due to the most significant bit being used to encode a loop flag and the internally
-managed buffers always being aligned to MA_SIMD_ALIGNMENT.
+The maximum size of the buffer in bytes is 0x7FFFFFFF-(MA_SIMD_ALIGNMENT-1) due to the most
+significant bit being used to encode a loop flag and the internally managed buffers always being
+aligned to MA_SIMD_ALIGNMENT.
-Note that the ring buffer is only thread safe when used by a single consumer thread and single producer thread.
+Note that the ring buffer is only thread safe when used by a single consumer thread and single
+producer thread.
@@ -3034,7 +5806,7 @@ Note that the ring buffer is only thread safe when used by a single consumer thr
-
11. Backends
+
15. Backends
The following backends are supported by miniaudio.
@@ -3223,52 +5995,60 @@ Some backends have some nuance details you may want to be aware of.
-
11.1. WASAPI
+
15.1. WASAPI
-Low-latency shared mode will be disabled when using an application-defined sample rate which is different to the device's native sample rate. To work around
- this, set wasapi.noAutoConvertSRC to true in the device config. This is due to IAudioClient3_InitializeSharedAudioStream() failing when the
- AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM flag is specified. Setting wasapi.noAutoConvertSRC will result in miniaudio's internal resampler being used instead
- which will in turn enable the use of low-latency shared mode.
-
-
+Low-latency shared mode will be disabled when using an application-defined sample rate which is
+ different to the device's native sample rate. To work around this, set wasapi.noAutoConvertSRC
+ to true in the device config. This is due to IAudioClient3_InitializeSharedAudioStream() failing
+ when the AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM flag is specified. Setting wasapi.noAutoConvertSRC
+ will result in miniaudio's internal resampler being used instead which will in turn enable the
+ use of low-latency shared mode.
-To capture audio on Android, remember to add the RECORD_AUDIO permission to your manifest: <uses-permission android:name="android.permission.RECORD_AUDIO" />
-
+To capture audio on Android, remember to add the RECORD_AUDIO permission to your manifest:
+ <uses-permission android:name="android.permission.RECORD_AUDIO" />
-With OpenSL|ES, only a single ma_context can be active at any given time. This is due to a limitation with OpenSL|ES.
-
+With OpenSL|ES, only a single ma_context can be active at any given time. This is due to a
+ limitation with OpenSL|ES.
-With AAudio, only default devices are enumerated. This is due to AAudio not having an enumeration API (devices are enumerated through Java). You can however
- perform your own device enumeration through Java and then set the ID in the ma_device_id structure (ma_device_id.aaudio) and pass it to ma_device_init().
-
+With AAudio, only default devices are enumerated. This is due to AAudio not having an enumeration
+ API (devices are enumerated through Java). You can however perform your own device enumeration
+ through Java and then set the ID in the ma_device_id structure (ma_device_id.aaudio) and pass it
+ to ma_device_init().
-The backend API will perform resampling where possible. The reason for this as opposed to using miniaudio's built-in resampler is to take advantage of any
- potential device-specific optimizations the driver may implement.
-
-
+The backend API will perform resampling where possible. The reason for this as opposed to using
+ miniaudio's built-in resampler is to take advantage of any potential device-specific
+ optimizations the driver may implement.
-
11.4. UWP
+
+
+
+
15.4. UWP
-UWP only supports default playback and capture devices.
-
+UWP only supports default playback and capture devices.
-UWP requires the Microphone capability to be enabled in the application's manifest (Package.appxmanifest):
-
-
+UWP requires the Microphone capability to be enabled in the application's manifest (Package.appxmanifest):
+
+
+
<Package ...>
...
@@ -3279,23 +6059,21 @@ UWP requires the Microphone capability to be enabled in the application's ma
-
11.5. Web Audio / Emscripten
+
15.5. Web Audio / Emscripten
-You cannot use -std=c* compiler flags, nor -ansi. This only applies to the Emscripten build.
-
+You cannot use -std=c* compiler flags, nor -ansi. This only applies to the Emscripten build.
-The first time a context is initialized it will create a global object called "miniaudio" whose primary purpose is to act as a factory for device objects.
-
+The first time a context is initialized it will create a global object called "miniaudio" whose
+ primary purpose is to act as a factory for device objects.
-Currently the Web Audio backend uses ScriptProcessorNode's, but this may need to change later as they've been deprecated.
-
+Currently the Web Audio backend uses ScriptProcessorNode's, but this may need to change later as
+ they've been deprecated.
-Google has implemented a policy in their browsers that prevent automatic media output without first receiving some kind of user input. The following web page
- has additional details: https://developers.google.com/web/updates/2017/09/autoplay-policy-changes. Starting the device may fail if you try to start playback
- without first handling some kind of user input.
-
-
+Google has implemented a policy in their browsers that prevent automatic media output without
+ first receiving some kind of user input. The following web page has additional details:
+ https://developers.google.com/web/updates/2017/09/autoplay-policy-changes. Starting the device
+ may fail if you try to start playback without first handling some kind of user input.
@@ -3303,32 +6081,56 @@ Google has implemented a policy in their browsers that prevent automatic media o
-
12. Miscellaneous Notes
+
+
+
+
16. Optimization Tips
+
+
+
+
16.1. High Level API
-Automatic stream routing is enabled on a per-backend basis. Support is explicitly enabled for WASAPI and Core Audio, however other backends such as
- PulseAudio may naturally support it, though not all have been tested.
-
+If a sound does not require doppler or pitch shifting, consider disabling pitching by
+ initializing the sound with the MA_SOUND_FLAG_NO_PITCH flag.
-The contents of the output buffer passed into the data callback will always be pre-initialized to silence unless the noPreZeroedOutputBuffer config variable
- in ma_device_config is set to true, in which case it'll be undefined which will require you to write something to the entire buffer.
-
+If a sound does not require spatialization, disable it by initialzing the sound with the
+ MA_SOUND_FLAG_NO_SPATIALIZATION flag. It can be renabled again post-initialization with
+ ma_sound_set_spatialization_enabled().
+
+
+
+
+
+
+
+
+
+
+
17. Miscellaneous Notes
+
-By default miniaudio will automatically clip samples. This only applies when the playback sample format is configured as ma_format_f32. If you are doing
- clipping yourself, you can disable this overhead by setting noClip to true in the device config.
-
+Automatic stream routing is enabled on a per-backend basis. Support is explicitly enabled for
+ WASAPI and Core Audio, however other backends such as PulseAudio may naturally support it, though
+ not all have been tested.
-The sndio backend is currently only enabled on OpenBSD builds.
-
+The contents of the output buffer passed into the data callback will always be pre-initialized to
+ silence unless the noPreZeroedOutputBuffer config variable in ma_device_config is set to true,
+ in which case it'll be undefined which will require you to write something to the entire buffer.
-The audio(4) backend is supported on OpenBSD, but you may need to disable sndiod before you can use it.
-
+By default miniaudio will automatically clip samples. This only applies when the playback sample
+ format is configured as ma_format_f32. If you are doing clipping yourself, you can disable this
+ overhead by setting noClip to true in the device config.
-Note that GCC and Clang requires -msse2, -mavx2, etc. for SIMD optimizations.
-
+Note that GCC and Clang requires -msse2, -mavx2, etc. for SIMD optimizations.
-When compiling with VC6 and earlier, decoding is restricted to files less than 2GB in size. This is due to 64-bit file APIs not being available.
-
+The sndio backend is currently only enabled on OpenBSD builds.
+
+The audio(4) backend is supported on OpenBSD, but you may need to disable sndiod before you can
+ use it.
+
+When compiling with VC6 and earlier, decoding is restricted to files less than 2GB in size. This
+ is due to 64-bit file APIs not being available.
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