The context can be configured via the `pConfig` argument. The config object is initialized with `ma_context_config_init()`. Individual configuration settings
can then be set directly on the structure. See examples below for a usage. Below are the members of the `ma_context_config` object.
can then be set directly on the structure. Below are the members of the `ma_context_config` object.
logCallback
Callback for handling log messages from miniaudio.
The context can be null in which case it uses the default. This is equivalent to passing in a
context that was initialized like so:
A device represents a physical audio device. The idea is you send or receive audio data from the device to either play it back through a speaker, or capture it
from a microphone. Whether or not you should send or receive data from the device (or both) depends on the type of device you are initializing which can be
playback, capture, full-duplex or loopback. (Note that loopback mode is only supported on select backends.) Sending and receiving audio data to and from the
device is done via a callback which is fired by miniaudio at periodic time intervals.
The frequency at which data is deilvered to and from a device depends on the size of it's period which is defined by a buffer size and a period count. The size
of the buffer can be defined in terms of PCM frames or milliseconds, whichever is more convenient. The size of a period is the size of this buffer, divided by
the period count. Generally speaking, the smaller the period, the lower the latency at the expense of higher CPU usage and increased risk of glitching due to
the more frequent and granular data deliver intervals. The size of a period will depend on your requirements, but miniaudio's defaults should work fine for
most scenarios. If you're building a game you should leave this fairly small, whereas if you're building a simple media player you can make it larger. Note
that the period size you request is actually just a hint - miniaudio will tell the backend what you want, but the backend is ultimately responsible for what it
gives you. You cannot assume you will get exactly what you ask for.
When delivering data to and from a device you need to make sure it's in the correct format which you can set through the device configuration. You just set the
format that you want to use and miniaudio will perform all of the necessary conversion for you internally. When delivering data to and from the callback you
can assume the format is the same as what you requested when you initialized the device. See Remarks for more details on miniaudio's data conversion pipeline.
Parameters
----------
pContext (in, optional)
A pointer to the context that owns the device. This can be null, in which case it creates a default context internally.
pConfig (in)
A pointer to the device configuration. Cannot be null. See remarks for details.
pDevice (out)
A pointer to the device object being initialized.
Return Value
------------
MA_SUCCESS if successful; any other error code otherwise.
Thread Safety
-------------
Unsafe. It is not safe to call this function simultaneously for different devices because some backends depend on and mutate global state. The same applies to
calling this at the same time as `ma_device_uninit()`.
Callback Safety
---------------
Unsafe. It is not safe to call this inside any callback.
Remarks
-------
Setting `pContext` to NULL will result in miniaudio creating a default context internally and is equivalent to passing in a context initialized like so:
```c
ma_context_init(NULL, 0, NULL, &context);
```
Do not pass in null for the context if you are needing to open multiple devices. You can,
however, use null when initializing the first device, and then use device.pContext for the
initialization of other devices.
The device's configuration is controlled with pConfig. This allows you to configure the sample
format, channel count, sample rate, etc. Before calling ma_device_init(), you will need to
initialize a ma_device_config object using ma_device_config_init(). You must set the callback in
the device config. Once initialized, the device's config is immutable. If you need to change the
config you will need to initialize a new device.
Passing in 0 to any property in pConfig will force the use of a default value. In the case of
sample format, channel count, sample rate and channel map it will default to the values used by
the backend's internal device. For the size of the buffer you can set bufferSizeInFrames or
bufferSizeInMilliseconds (if both are set it will prioritize bufferSizeInFrames). If both are
set to zero, it will default to MA_BASE_BUFFER_SIZE_IN_MILLISECONDS_LOW_LATENCY or
MA_BASE_BUFFER_SIZE_IN_MILLISECONDS_CONSERVATIVE, depending on whether or not performanceProfile
is set to ma_performance_profile_low_latency or ma_performance_profile_conservative.
If you request exclusive mode and the backend does not support it an error will be returned. For
robustness, you may want to first try initializing the device in exclusive mode, and then fall back
to shared mode if required. Alternatively you can just request shared mode (the default if you
leave it unset in the config) which is the most reliable option. Some backends do not have a
practical way of choosing whether or not the device should be exclusive or not (ALSA, for example)
in which case it just acts as a hint. Unless you have special requirements you should try avoiding
exclusive mode as it's intrusive to the user. Starting with Windows 10, miniaudio will use low-latency
shared mode where possible which may make exclusive mode unnecessary.
When sending or receiving data to/from a device, miniaudio will internally perform a format
conversion to convert between the format specified by pConfig and the format used internally by
the backend. If you pass in NULL for pConfig or 0 for the sample format, channel count,
sample rate _and_ channel map, data transmission will run on an optimized pass-through fast path.
The buffer size should be treated as a hint. miniaudio will try it's best to use exactly what you
ask for, but it may differ. You should not assume the number of frames specified in each call to
the data callback is exactly what you originally specified.
The <periods> property controls how frequently the background thread is woken to check for more
data. It's tied to the buffer size, so as an example, if your buffer size is equivalent to 10
milliseconds and you have 2 periods, the CPU will wake up approximately every 5 milliseconds.
When compiling for UWP you must ensure you call this function on the main UI thread because the
operating system may need to present the user with a message asking for permissions. Please refer
to the official documentation for ActivateAudioInterfaceAsync() for more information.
ALSA Specific: When initializing the default device, requesting shared mode will try using the
"dmix" device for playback and the "dsnoop" device for capture. If these fail it will try falling
back to the "hw" device.
Do not set `pContext` to NULL if you are needing to open multiple devices. You can, however, use NULL when initializing the first device, and then use
device.pContext for the initialization of other devices.
Return Value:
MA_SUCCESS if successful; any other error code otherwise.
The device can be configured via the `pConfig` argument. The config object is initialized with `ma_device_config_init()`. Individual configuration settings can
then be set directly on the structure. Below are the members of the `ma_device_config` object.
Thread Safety: UNSAFE
It is not safe to call this function simultaneously for different devices because some backends
depend on and mutate global state (such as OpenSL|ES). The same applies to calling this at the
same time as ma_device_uninit().
deviceType
Must be `ma_device_type_playback`, `ma_device_type_capture`, `ma_device_type_duplex` of `ma_device_type_loopback`.
Callback Safety: UNSAFE
It is not safe to call this inside any callback.
sampleRate
The sample rate, in hertz. The most common sample rates are 48000 and 44100. Setting this to 0 will use the device's native sample rate.
bufferSizeInFrames
The total size of the buffer in PCM frames. If this is 0, `bufferSizeInMilliseconds` will be used instead. If both are 0 the default buffer size will
be used depending on the selected performance profile. See below for details.
bufferSizeInMilliseconds
The total size of the buffer in milliseconds. If this is 0, `bufferSizeInFrames` will be used instead. If both are 0 the default buffer size will be
used depending on the selected performance profile. See below for details.
periods
The number of periods making up the device's entire buffer. The total buffer size, defined by `bufferSizeInFrames` or `bufferSizeInMilliseconds`, will
be dividied by this number to determine the approximate latency. This is just a hint as backends will be the ones who ultimately decide how your
buffers and periods will be configured.
performanceProfile
A hint to miniaudio as to the performance requirements of your program. Can be either `ma_performance_profile_low_latency` (default) or
`ma_performance_profile_conservative`. This mainly affects the size of default buffers and can usually be left at it's default value.
noPreZeroedOutputBuffer
When set to true, the contents of the output buffer passed into the data callback will be left undefined. When set to false (default), the contents of
the output buffer will be cleared the zero. You can use this to avoid the overhead of zeroing out the buffer if you know can guarantee that your data
callback will write to every sample in the output buffer, or if you are doing your own clearing.
noClip
When set to true, the contents of the output buffer passed into the data callback will be clipped after returning. When set to false (default), the
contents of the output buffer are left alone after returning and it will be left up to the backend itself to decide whether or not the clip. This only
applies when the playback sample format is f32.
dataCallback
The callback to fire whenever data is ready to be delivered to or from the device.
stopCallback
The callback to fire whenever the device has stopped, either explicitly via `ma_device_stop()`, or implicitly due to things like the device being
disconnected.
pUserData
The user data pointer to use with the device. You can access this directly from the device object like `device.pUserData`.
resampling.algorithm
The resampling algorithm to use when miniaudio needs to perform resampling between the rate specified by `sampleRate` and the device's native rate. The
default value is `ma_resample_algorithm_linear`, and the quality can be configured with `resampling.linear.lpfCount`.
resampling.linear.lpfCount
The linear resampler applies a low-pass filter as part of it's procesing for anti-aliasing. This setting controls the quality of the filter. The higher
the value, the better the quality. Setting this to 0 will disable low-pass filtering altogether. The maximum value is `MA_MAX_RESAMPLER_LPF_FILTERS`.
The default value is `min(2, MA_MAX_RESAMPLER_LPF_FILTERS)`.
playback.pDeviceID
A pointer to a `ma_device_id` structure containing the ID of the playback device to initialize. Setting this NULL (default) will use the system's
default playback device. Retrieve the device ID from the `ma_device_info` structure, which can be retrieved using device enumeration.
playback.format
The sample format to use for playback. When set to `ma_format_unknown` the device's native format will be used. This can be retrieved after
initialization from the device object directly with `device.playback.format`.
playback.channels
The number of channels to use for playback. When set to 0 the device's native channel count will be used. This can be retrieved after initialization
from the device object directly with `device.playback.channels`.
playback.channelMap
The channel map to use for playback. When left empty, the device's native channel map will be used. This can be retrieved after initialization from the
device object direct with `device.playback.channelMap`.
playback.shareMode
The preferred share mode to use for playback. Can be either `ma_share_mode_shared` (default) or `ma_share_mode_exclusive`. Note that if you specify
exclusive mode, but it's not supported by the backend, initialization will fail. You can then fall back to shared mode if desired.
playback.pDeviceID
A pointer to a `ma_device_id` structure containing the ID of the playback device to initialize. Setting this NULL (default) will use the system's
default playback device. Retrieve the device ID from the `ma_device_info` structure, which can be retrieved using device enumeration.
capture.format
The sample format to use for capture. When set to `ma_format_unknown` the device's native format will be used. This can be retrieved after
initialization from the device object directly with `device.capture.format`.
capture.channels
The number of channels to use for capture. When set to 0 the device's native channel count will be used. This can be retrieved after initialization
from the device object directly with `device.capture.channels`.
capture.channelMap
The channel map to use for capture. When left empty, the device's native channel map will be used. This can be retrieved after initialization from the
device object direct with `device.capture.channelMap`.
capture.shareMode
The preferred share mode to use for capture. Can be either `ma_share_mode_shared` (default) or `ma_share_mode_exclusive`. Note that if you specify
exclusive mode, but it's not supported by the backend, initialization will fail. You can then fall back to shared mode if desired.
wasapi.noAutoConvertSRC
WASAPI only. When set to true, disables WASAPI's automatic resampling and forces the use of miniaudio's resampler. Defaults to false.
wasapi.noDefaultQualitySRC
WASAPI only. Only used when `wasapi.noAutoConvertSRC` is set to false. When set to true, disables the use of `AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY`.
You should usually leave this set to false, which is the default.
wasapi.noAutoStreamRouting
WASAPI only. When set to true, disables automatic stream routing on the WASAPI backend. Defaults to false.
wasapi.noHardwareOffloading
WASAPI only. When set to true, disables the use of WASAPI's hardware offloading feature. Defaults to false.
alsa.noMMap
ALSA only. When set to true, disables MMap mode. Defaults to false.
pulse.pStreamNamePlayback
PulseAudio only. Sets the stream name for playback.
pulse.pStreamNameCapture
PulseAudio only. Sets the stream name for capture.
Once initialized, the device's config is immutable. If you need to change the config you will need to initialize a new device.
If both `bufferSizeInFrames` and `bufferSizeInMilliseconds` are set to zero, it will default to `MA_BASE_BUFFER_SIZE_IN_MILLISECONDS_LOW_LATENCY` or
`MA_BASE_BUFFER_SIZE_IN_MILLISECONDS_CONSERVATIVE`, depending on whether or not `performanceProfile` is set to `ma_performance_profile_low_latency` or
`ma_performance_profile_conservative`.
If you request exclusive mode and the backend does not support it an error will be returned. For robustness, you may want to first try initializing the device
in exclusive mode, and then fall back to shared mode if required. Alternatively you can just request shared mode (the default if you leave it unset in the
config) which is the most reliable option. Some backends do not have a practical way of choosing whether or not the device should be exclusive or not (ALSA,
for example) in which case it just acts as a hint. Unless you have special requirements you should try avoiding exclusive mode as it's intrusive to the user.
Starting with Windows 10, miniaudio will use low-latency shared mode where possible which may make exclusive mode unnecessary.
After initializing the device it will be in a stopped state. To start it, use `ma_device_start()`.
When sending or receiving data to/from a device, miniaudio will internally perform a format conversion to convert between the format specified by pConfig and
the format used internally by the backend. If you pass in 0 for the sample format, channel count, sample rate _and_ channel map, data transmission will run on
an optimized pass-through fast path. You can retrieve the format, channel count and sample rate by inspecting the `playback/capture.format`,
`playback/capture.channels` and `sampleRate` members of the device object.
When compiling for UWP you must ensure you call this function on the main UI thread because the operating system may need to present the user with a message
asking for permissions. Please refer to the official documentation for ActivateAudioInterfaceAsync() for more information.
ALSA Specific: When initializing the default device, requesting shared mode will try using the "dmix" device for playback and the "dsnoop" device for capture.
If these fail it will try falling back to the "hw" device.
