Programming Manual

1. Introduction

miniaudio is a single file library for audio playback and capture. To use it, do the following in one .c file:

#define MINIAUDIO_IMPLEMENTATION
#include "miniaudio.h"

You can do #include "miniaudio.h" in other parts of the program just like any other header.

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.

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.

void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
{
    // In playback mode copy data to pOutput. In capture mode read data from pInput. In full-duplex mode, both
    // pOutput and pInput will be valid and you can move data from pInput into pOutput. Never process more than
    // frameCount frames.
}

int main()
{
    ma_device_config config = ma_device_config_init(ma_device_type_playback);
    config.playback.format   = ma_format_f32;   // Set to ma_format_unknown to use the device's native format.
    config.playback.channels = 2;               // Set to 0 to use the device's native channel count.
    config.sampleRate        = 48000;           // Set to 0 to use the device's native sample rate.
    config.dataCallback      = data_callback;   // This function will be called when miniaudio needs more data.
    config.pUserData         = pMyCustomData;   // Can be accessed from the device object (device.pUserData).

    ma_device device;
    if (ma_device_init(NULL, &config, &device) != MA_SUCCESS) {
        return -1;  // Failed to initialize the device.
    }

    ma_device_start(&device);     // The device is sleeping by default so you'll need to start it manually.

    // Do something here. Probably your program's main loop.

    ma_device_uninit(&device);    // This will stop the device so no need to do that manually.
    return 0;
}

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):

Symbol

Description

Range

ma_format_f32

32-bit floating point

[-1, 1]

ma_format_s16

16-bit signed integer

[-32768, 32767]

ma_format_s24

24-bit signed integer (tightly packed)

[-8388608, 8388607]

ma_format_s32

32-bit signed integer

[-2147483648, 2147483647]

ma_format_u8

8-bit unsigned integer

[0, 255]

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.

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:

ma_device_init()
ma_device_init_ex()
ma_device_uninit()
ma_device_start()
ma_device_stop()

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:

ma_device_config config = ma_device_config_init(ma_device_type_capture);
config.capture.format   = MY_FORMAT;
config.capture.channels = MY_CHANNEL_COUNT;

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).

These are the available device types and how you should handle the buffers in the callback:

Device Type

Callback Behavior

ma_device_type_playback

Write to output buffer, leave input buffer untouched.

ma_device_type_capture

Read from input buffer, leave output buffer untouched.

ma_device_type_duplex

Read from input buffer, write to output buffer.

ma_device_type_loopback

Read from input buffer, leave output buffer untouched.

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:

config.playback.pDeviceID = pMyPlaybackDeviceID;    // Only if requesting a playback or duplex device.
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.

ma_context context;
if (ma_context_init(NULL, 0, NULL, &context) != MA_SUCCESS) {
    // Error.
}

ma_device_info* pPlaybackInfos;
ma_uint32 playbackCount;
ma_device_info* pCaptureInfos;
ma_uint32 captureCount;
if (ma_context_get_devices(&context, &pPlaybackInfos, &playbackCount, &pCaptureInfos, &captureCount) != MA_SUCCESS) {
    // Error.
}

// Loop over each device info and do something with it. Here we just print the name with their index. You may want
// to give the user the opportunity to choose which device they'd prefer.
for (ma_uint32 iDevice = 0; iDevice < playbackCount; iDevice += 1) {
    printf("%d - %s\n", iDevice, pPlaybackInfos[iDevice].name);
}

ma_device_config config = ma_device_config_init(ma_device_type_playback);
config.playback.pDeviceID = &pPlaybackInfos[chosenPlaybackDeviceIndex].id;
config.playback.format    = MY_FORMAT;
config.playback.channels  = MY_CHANNEL_COUNT;
config.sampleRate         = MY_SAMPLE_RATE;
config.dataCallback       = data_callback;
config.pUserData          = pMyCustomData;

ma_device device;
if (ma_device_init(&context, &config, &device) != MA_SUCCESS) {
    // Error
}

...

ma_device_uninit(&device);
ma_context_uninit(&context);

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.

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.

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.

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.

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 (sorry) and will need to link the relevant frameworks but should Just Work with Xcode. Compiling through the command line requires linking to -lpthread and -lm.

2.3. Linux

The Linux build only requires linking to -ldl, -lpthread and -lm. You do not need any development packages.

2.4. BSD

The BSD build only requires linking to -lpthread and -lm. NetBSD uses audio(4), OpenBSD uses sndio and FreeBSD uses OSS.

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+.

2.6. Emscripten

The Emscripten build emits Web Audio JavaScript directly and should Just Work without any configuration. You cannot use -std=c* compiler flags, nor -ansi.

2.7. Build Options

#define these options before including miniaudio.h.

Option

Description

MA_NO_WASAPI

Disables the WASAPI backend.

MA_NO_DSOUND

Disables the DirectSound backend.

MA_NO_WINMM

Disables the WinMM backend.

MA_NO_ALSA

Disables the ALSA backend.

MA_NO_PULSEAUDIO

Disables the PulseAudio backend.

MA_NO_JACK

Disables the JACK backend.

MA_NO_COREAUDIO

Disables the Core Audio backend.

MA_NO_SNDIO

Disables the sndio backend.

MA_NO_AUDIO4

Disables the audio(4) backend.

MA_NO_OSS

Disables the OSS backend.

MA_NO_AAUDIO

Disables the AAudio backend.

MA_NO_OPENSL

Disables the OpenSL

MA_NO_WEBAUDIO

Disables the Web Audio backend.

MA_NO_NULL

Disables the null backend.

MA_NO_DECODING

Disables decoding APIs.

MA_NO_ENCODING

Disables encoding APIs.

MA_NO_WAV

Disables the built-in WAV decoder and encoder.

MA_NO_FLAC

Disables the built-in FLAC decoder.

MA_NO_MP3

Disables the built-in MP3 decoder.

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.

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:

MA_NO_DEVICE_IO

MA_NO_GENERATION

Disables generation APIs such a ma_waveform and ma_noise.

MA_NO_SSE2

Disables SSE2 optimizations.

MA_NO_AVX2

Disables AVX2 optimizations.

MA_NO_AVX512

Disables AVX-512 optimizations.

MA_NO_NEON

Disables NEON optimizations.

MA_LOG_LEVEL [level]

Sets the logging level. Set level to one of the following:

MA_LOG_LEVEL_VERBOSE
MA_LOG_LEVEL_INFO
MA_LOG_LEVEL_WARNING
MA_LOG_LEVEL_ERROR

MA_DEBUG_OUTPUT

Enable printf() debug output.

MA_COINIT_VALUE

Windows only. The value to pass to internal calls to CoInitializeEx(). Defaults to COINIT_MULTITHREADED.

MA_API

Controls how public APIs should be decorated. Defaults to extern.

MA_DLL

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 export. Otherwise it will be configured to import. This has no 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.

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.

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".

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.

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.

3.5. Formats

Throughout miniaudio you will see references to different sample formats:

Symbol

Description

Range

ma_format_f32

32-bit floating point

[-1, 1]

ma_format_s16

16-bit signed integer

[-32768, 32767]

ma_format_s24

24-bit signed integer (tightly packed)

[-8388608, 8388607]

ma_format_s32

32-bit signed integer

[-2147483648, 2147483647]

ma_format_u8

8-bit unsigned integer

[0, 255]

All formats are native-endian.

4. Decoding

The ma_decoder API is used for reading audio files. The following formats are supported:

Format

Decoding Backend

Built-In

WAV

dr_wav

Yes

MP3

dr_mp3

Yes

FLAC

dr_flac

Yes

Vorbis

stb_vorbis

No

Vorbis is supported via stb_vorbis which can be enabled by including the header section before the implementation of miniaudio, like the following:

#define STB_VORBIS_HEADER_ONLY
#include "extras/stb_vorbis.c"    // Enables Vorbis decoding.

#define MINIAUDIO_IMPLEMENTATION
#include "miniaudio.h"

// The stb_vorbis implementation must come after the implementation of miniaudio.
#undef STB_VORBIS_HEADER_ONLY
#include "extras/stb_vorbis.c"

A copy of stb_vorbis is included in the "extras" folder in the miniaudio repository (https://github.com/mackron/miniaudio).

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:

#define MA_NO_WAV
#define MA_NO_MP3
#define MA_NO_FLAC

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:

ma_decoder decoder;
ma_result result = ma_decoder_init_file("MySong.mp3", NULL, &decoder);
if (result != MA_SUCCESS) {
    return false;   // An error occurred.
}

...

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:

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.

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:

ma_uint64 framesRead = ma_decoder_read_pcm_frames(pDecoder, pFrames, framesToRead);
if (framesRead < framesToRead) {
    // Reached the end.
}

You can also seek to a specific frame like so:

ma_result result = ma_decoder_seek_to_pcm_frame(pDecoder, targetFrame);
if (result != MA_SUCCESS) {
    return false;   // An error occurred.
}

If you want to loop back to the start, you can simply seek back to the first PCM frame:

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 the _wav, _mp3, etc. varients of the aforementioned initialization APIs:

ma_decoder_init_wav()
ma_decoder_init_mp3()
ma_decoder_init_memory_wav()
ma_decoder_init_memory_mp3()
ma_decoder_init_file_wav()
ma_decoder_init_file_mp3()
etc.

The ma_decoder_init_file() API will try using the file extension to determine which decoding backend to prefer.

5. 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:

#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.

ma_encoder_config config = ma_encoder_config_init(ma_resource_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) {
    // Error
}

...

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:

Enum

Description

ma_resource_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:

framesWritten = ma_encoder_write_pcm_frames(&encoder, pPCMFramesToWrite, framesToWrite);

Encoders must be uninitialized with ma_encoder_uninit().

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.

6.1. Sample Format Conversion

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.

6.1.1. Dithering

Dithering can be set using the ditherMode parameter.

The different dithering modes include the following, in order of efficiency:

Type

Enum Token

None

ma_dither_mode_none

Rectangle

ma_dither_mode_rectangle

Triangle

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:

s16 -> u8
s24 -> u8
s32 -> u8
f32 -> u8
s24 -> s16
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.

6.2. 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.

ma_channel_converter_config config = ma_channel_converter_config_init(
    ma_format,                      // Sample format
    1,                              // Input channels
    NULL,                           // Input channel map
    2,                              // Output channels
    NULL,                           // Output channel map
    ma_channel_mix_mode_default);   // The mixing algorithm to use when combining channels.

result = ma_channel_converter_init(&config, &converter);
if (result != MA_SUCCESS) {
    // Error.
}

To perform the conversion simply call ma_channel_converter_process_pcm_frames() like so:

ma_result result = ma_channel_converter_process_pcm_frames(&converter, pFramesOut, pFramesIn, frameCount);
if (result != MA_SUCCESS) {
    // Error.
}

It is up to the caller to ensure the output buffer is large enough to accomodate the new PCM frames.

Input and output PCM frames are always interleaved. Deinterleaved layouts are not supported.

6.2.1. 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.

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.

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 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:

Name

Description

ma_standard_channel_map_default

Default channel map used by miniaudio. See below.

ma_standard_channel_map_microsoft

Channel map used by Microsoft's bitfield channel maps.

ma_standard_channel_map_alsa

Default ALSA channel map.

ma_standard_channel_map_rfc3551

RFC 3551. Based on AIFF.

ma_standard_channel_map_flac

FLAC channel map.

ma_standard_channel_map_vorbis

Vorbis channel map.

ma_standard_channel_map_sound4

FreeBSD's sound(4).

ma_standard_channel_map_sndio

sndio channel map. http://www.sndio.org/tips.html.

ma_standard_channel_map_webaudio

https://webaudio.github.io/web-audio-api/#ChannelOrdering

Below are the channel maps used by default in miniaudio (ma_standard_channel_map_default):

Channel Count

Mapping

1 (Mono)

0: MA_CHANNEL_MONO

2 (Stereo)

0: MA_CHANNEL_FRONT_LEFT
1: MA_CHANNEL_FRONT_RIGHT

3

0: MA_CHANNEL_FRONT_LEFT
1: MA_CHANNEL_FRONT_RIGHT
2: MA_CHANNEL_FRONT_CENTER

4 (Surround)

0: MA_CHANNEL_FRONT_LEFT
1: MA_CHANNEL_FRONT_RIGHT
2: MA_CHANNEL_FRONT_CENTER
3: MA_CHANNEL_BACK_CENTER

5

0: MA_CHANNEL_FRONT_LEFT
1: MA_CHANNEL_FRONT_RIGHT
2: MA_CHANNEL_FRONT_CENTER
3: MA_CHANNEL_BACK_LEFT
4: MA_CHANNEL_BACK_RIGHT

6 (5.1)

0: MA_CHANNEL_FRONT_LEFT
1: MA_CHANNEL_FRONT_RIGHT
2: MA_CHANNEL_FRONT_CENTER
3: MA_CHANNEL_LFE
4: MA_CHANNEL_SIDE_LEFT
5: MA_CHANNEL_SIDE_RIGHT

7

0: MA_CHANNEL_FRONT_LEFT
1: MA_CHANNEL_FRONT_RIGHT
2: MA_CHANNEL_FRONT_CENTER
3: MA_CHANNEL_LFE
4: MA_CHANNEL_BACK_CENTER
4: MA_CHANNEL_SIDE_LEFT
5: MA_CHANNEL_SIDE_RIGHT

8 (7.1)

0: MA_CHANNEL_FRONT_LEFT
1: MA_CHANNEL_FRONT_RIGHT
2: MA_CHANNEL_FRONT_CENTER
3: MA_CHANNEL_LFE
4: MA_CHANNEL_BACK_LEFT
5: MA_CHANNEL_BACK_RIGHT
6: MA_CHANNEL_SIDE_LEFT
7: MA_CHANNEL_SIDE_RIGHT

Other

All channels set to 0. This is equivalent to the same mapping as the device.

6.3. Resampling

Resampling is achieved with the ma_resampler object. To create a resampler object, do something like the following:

ma_resampler_config config = ma_resampler_config_init(
    ma_format_s16,
    channels,
    sampleRateIn,
    sampleRateOut,
    ma_resample_algorithm_linear);

ma_resampler resampler;
ma_result result = ma_resampler_init(&config, &resampler);
if (result != MA_SUCCESS) {
    // An error occurred...
}

Do the following to uninitialize the resampler:

ma_resampler_uninit(&resampler);

The following example shows how data can be processed

ma_uint64 frameCountIn  = 1000;
ma_uint64 frameCountOut = 2000;
ma_result result = ma_resampler_process_pcm_frames(&resampler, pFramesIn, &frameCountIn, pFramesOut, &frameCountOut);
if (result != MA_SUCCESS) {
    // An error occurred...
}

// At this point, frameCountIn contains the number of input frames that were consumed and frameCountOut contains the
// 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.

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 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:

Algorithm

Enum Token

Linear

ma_resample_algorithm_linear

Speex

ma_resample_algorithm_speex

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.

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().

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().

6.3.1. 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.

6.3.1.1. 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 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 API for the linear resampler is the same as the main resampler API, only it's called ma_linear_resampler.

6.3.1.2. Speex Resampling

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:

#include "extras/speex_resampler/ma_speex_resampler.h"

Both the header and implementation is contained within the same file. The implementation can be included in your program like so:

#define MINIAUDIO_SPEEX_RESAMPLER_IMPLEMENTATION
#include "extras/speex_resampler/ma_speex_resampler.h"

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.

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.

6.4. 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:

ma_data_converter_config config = ma_data_converter_config_init(
    inputFormat,
    outputFormat,
    inputChannels,
    outputChannels,
    inputSampleRate,
    outputSampleRate
);

ma_data_converter converter;
ma_result result = ma_data_converter_init(&config, &converter);
if (result != MA_SUCCESS) {
    // An error occurred...
}

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:

ma_data_converter_config config = ma_data_converter_config_init_default();
config.formatIn = inputFormat;
config.formatOut = outputFormat;
config.channelsIn = inputChannels;
config.channelsOut = outputChannels;
config.sampleRateIn = inputSampleRate;
config.sampleRateOut = outputSampleRate;
ma_get_standard_channel_map(ma_standard_channel_map_flac, config.channelCountIn, config.channelMapIn);
config.resampling.linear.lpfOrder = MA_MAX_FILTER_ORDER;

Do the following to uninitialize the data converter:

ma_data_converter_uninit(&converter);

The following example shows how data can be processed

ma_uint64 frameCountIn  = 1000;
ma_uint64 frameCountOut = 2000;
ma_result result = ma_data_converter_process_pcm_frames(&converter, pFramesIn, &frameCountIn, pFramesOut, &frameCountOut);
if (result != MA_SUCCESS) {
    // An error occurred...
}

// At this point, frameCountIn contains the number of input frames that were consumed and frameCountOut contains the number
// 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.

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.

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().

7. Filtering

7.1. Biquad Filtering

Biquad filtering is achieved with the ma_biquad API. Example:

ma_biquad_config config = ma_biquad_config_init(ma_format_f32, channels, b0, b1, b2, a0, a1, a2);
ma_result result = ma_biquad_init(&config, &biquad);
if (result != MA_SUCCESS) {
    // Error.
}

...

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.

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.

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:

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.

7.2. Low-Pass Filtering

Low-pass filtering is achieved with the following APIs:

API

Description

ma_lpf1

First order low-pass filter

ma_lpf2

Second order low-pass filter

ma_lpf

High order low-pass filter (Butterworth)

Low-pass filter example:

ma_lpf_config config = ma_lpf_config_init(ma_format_f32, channels, sampleRate, cutoffFrequency, order);
ma_result result = ma_lpf_init(&config, &lpf);
if (result != MA_SUCCESS) {
    // Error.
}

...

ma_lpf_process_pcm_frames(&lpf, pFramesOut, pFramesIn, frameCount);

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:

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.

for (iFilter = 0; iFilter < filterCount; iFilter += 1) {
    ma_lpf2_process_pcm_frames(&lpf2[iFilter], pMyData, pMyData, frameCount);
}

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.

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.

7.3. High-Pass Filtering

High-pass filtering is achieved with the following APIs:

API

Description

ma_hpf1

First order high-pass filter

ma_hpf2

Second order high-pass filter

ma_hpf

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.

7.4. Band-Pass Filtering

Band-pass filtering is achieved with the following APIs:

API

Description

ma_bpf2

Second order band-pass filter

ma_bpf

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.

7.5. Notch Filtering

Notch filtering is achieved with the following APIs:

API

Description

ma_notch2

Second order notching filter

7.6. Peaking EQ Filtering

Peaking filtering is achieved with the following APIs:

API

Description

ma_peak2

Second order peaking filter

7.7. Low Shelf Filtering

Low shelf filtering is achieved with the following APIs:

API

Description

ma_loshelf2

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.

7.8. High Shelf Filtering

High shelf filtering is achieved with the following APIs:

API

Description

ma_hishelf2

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.

8. Waveform and Noise Generation

8.1. Waveforms

miniaudio supports generation of sine, square, triangle and sawtooth waveforms. This is achieved with the ma_waveform API. Example:

ma_waveform_config config = ma_waveform_config_init(
    FORMAT,
    CHANNELS,
    SAMPLE_RATE,
    ma_waveform_type_sine,
    amplitude,
    frequency);

ma_waveform waveform;
ma_result result = ma_waveform_init(&config, &waveform);
if (result != MA_SUCCESS) {
    // Error.
}

...

ma_waveform_read_pcm_frames(&waveform, pOutput, frameCount);

The amplitude, frequency and sample rate can be changed dynamically with ma_waveform_set_amplitude(), ma_waveform_set_frequency() 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.

Below are the supported waveform types:

Enum Name

ma_waveform_type_sine

ma_waveform_type_square

ma_waveform_type_triangle

ma_waveform_type_sawtooth

8.2. Noise

miniaudio supports generation of white, pink and Brownian noise via the ma_noise API. Example:

ma_noise_config config = ma_noise_config_init(
    FORMAT,
    CHANNELS,
    ma_noise_type_white,
    SEED,
    amplitude);

ma_noise noise;
ma_result result = ma_noise_init(&config, &noise);
if (result != MA_SUCCESS) {
    // Error.
}

...

ma_noise_read_pcm_frames(&noise, pOutput, frameCount);

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.

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:

config.duplicateChannels = MA_TRUE;

Below are the supported noise types.

Enum Name

ma_noise_type_white

ma_noise_type_pink

ma_noise_type_brownian

9. 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.

Audio buffers are initialised using the standard configuration system used everywhere in miniaudio:

ma_audio_buffer_config config = ma_audio_buffer_config_init(
    format,
    channels,
    sizeInFrames,
    pExistingData,
    &allocationCallbacks);

ma_audio_buffer buffer;
result = ma_audio_buffer_init(&config, &buffer);
if (result != MA_SUCCESS) {
    // Error.
}

...

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().

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():

ma_audio_buffer_config config = ma_audio_buffer_config_init(
    format,
    channels,
    sizeInFrames,
    pExistingData,
    &allocationCallbacks);

ma_audio_buffer* pBuffer
result = ma_audio_buffer_alloc_and_init(&config, &pBuffer);
if (result != MA_SUCCESS) {
    // Error
}

...

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().

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.

ma_uint64 framesRead = ma_audio_buffer_read_pcm_frames(pAudioBuffer, pFramesOut, desiredFrameCount, isLooping);
if (framesRead < desiredFrameCount) {
    // If not looping, this means the end has been reached. This should never happen in looping mode with valid input.
}

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:

void* pMappedFrames;
ma_uint64 frameCount = frameCountToTryMapping;
ma_result result = ma_audio_buffer_map(pAudioBuffer, &pMappedFrames, &frameCount);
if (result == MA_SUCCESS) {
    // Map was successful. The value in frameCount will be how many frames were _actually_ mapped, which may be
    // less due to the end of the buffer being reached.
    ma_copy_pcm_frames(pFramesOut, pMappedFrames, frameCount, pAudioBuffer->format, pAudioBuffer->channels);

    // You must unmap the buffer.
    ma_audio_buffer_unmap(pAudioBuffer, frameCount);
}

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().

10. 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.

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:

ma_pcm_rb rb;
ma_result result = ma_pcm_rb_init(FORMAT, CHANNELS, BUFFER_SIZE_IN_FRAMES, NULL, NULL, &rb);
if (result != MA_SUCCESS) {
    // Error
}

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_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.

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.

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.

11. Backends

The following backends are supported by miniaudio.

Name

Enum Name

Supported Operating Systems

WASAPI

ma_backend_wasapi

Windows Vista+

DirectSound

ma_backend_dsound

Windows XP+

WinMM

ma_backend_winmm

Windows XP+ (may work on older versions, but untested)

Core Audio

ma_backend_coreaudio

macOS, iOS

ALSA

ma_backend_alsa

Linux

PulseAudio

ma_backend_pulseaudio

Cross Platform (disabled on Windows, BSD and Android)

JACK

ma_backend_jack

Cross Platform (disabled on BSD and Android)

sndio

ma_backend_sndio

OpenBSD

audio(4)

ma_backend_audio4

NetBSD, OpenBSD

OSS

ma_backend_oss

FreeBSD

AAudio

ma_backend_aaudio

Android 8+

OpenSL ES

ma_backend_opensl

Android (API level 16+)

Web Audio

ma_backend_webaudio

Web (via Emscripten)

Null

ma_backend_null

Cross Platform (not used on Web)

Some backends have some nuance details you may want to be aware of.

11.1. 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.

11.2. PulseAudio

11.3. Android

  • 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 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.

11.4. UWP

  • UWP only supports default playback and capture devices.
  • UWP requires the Microphone capability to be enabled in the application's manifest (Package.appxmanifest):
<Package ...>
    ...
    <Capabilities>
        <DeviceCapability Name="microphone" />
    </Capabilities>
</Package>

11.5. Web Audio / Emscripten

  • 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.
  • 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.

12. Miscellaneous Notes

  • 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 contents of the output buffer passed into the data callback will always be pre-initialized to zero 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.
  • 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.
  • 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.
  • 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.
Copyright © 2020 David Reid
Developed by David Reid - mackron@gmail.com