SoundIoInStream

The size of this struct is not part of the API or ABI.

Members

Variables

bytes_per_frame
int bytes_per_frame;

computed automatically when you call ::soundio_instream_open

bytes_per_sample
int bytes_per_sample;

computed automatically when you call ::soundio_instream_open

device
SoundIoDevice* device;

Populated automatically when you call ::soundio_outstream_create.

error_callback
void function(SoundIoInStream*, int err) error_callback;

Optional callback. err is always SoundIoErrorStreaming. SoundIoErrorStreaming is an unrecoverable error. The stream is in an invalid state and must be destroyed. If you do not supply error_callback, the default callback will print a message to stderr and then abort(). This is called from the SoundIoInStream::read_callback thread context.

format
SoundIoFormat format;

Defaults to #SoundIoFormatFloat32NE, followed by the first one supported.

layout
SoundIoChannelLayout layout;

Defaults to Stereo, if available, followed by the first layout supported.

layout_error
int layout_error;

If setting the channel layout fails for some reason, this field is set to an error code. Possible error codes are: #SoundIoErrorIncompatibleDevice

name
const(char)* name;

Optional: Name of the stream. Defaults to "SoundIoInStream"; PulseAudio uses this for the stream name. JACK uses this for the client name of the client that connects when you open the stream. WASAPI uses this for the session display name. Must not contain a colon (":").

non_terminal_hint
bool non_terminal_hint;

Optional: Hint that this input stream is nonterminal. This is used by JACK and it means that the data received by the stream will be passed on or made available to another stream. Defaults to false.

overflow_callback
void function(SoundIoInStream*) overflow_callback;

This optional callback happens when the sound device buffer is full, yet there is more captured audio to put in it. This is never fired for PulseAudio. This is called from the SoundIoInStream::read_callback thread context.

read_callback
void function(SoundIoInStream*, int frame_count_min, int frame_count_max) read_callback;

In this function call ::soundio_instream_begin_read and ::soundio_instream_end_read as many times as necessary to read at minimum frame_count_min frames and at maximum frame_count_max frames. If you return from read_callback without having read frame_count_min, the frames will be dropped. frame_count_max is how many frames are available to read.

sample_rate
int sample_rate;

Sample rate is the number of frames per second. Defaults to max(sample_rate_min, min(sample_rate_max, 48000))

software_latency
double software_latency;

Ignoring hardware latency, this is the number of seconds it takes for a captured sample to become available for reading. After you call ::soundio_instream_open, this value is replaced with the actual software latency, as near to this value as possible. A higher value means less CPU usage. Defaults to a large value, potentially upwards of 2 seconds. If the device has unknown software latency min and max values, you may still set this, but you might not get the value you requested. For PulseAudio, if you set this value to non-default, it sets PA_STREAM_ADJUST_LATENCY and is the value used for fragsize. For JACK, this value is always equal to SoundIoDevice::software_latency_current

userdata
void* userdata;

Defaults to NULL. Put whatever you want here.

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