The audio stream functions are for playing digital sounds that are too big
to fit in a regular SAMPLE structure, either because they are huge files
that you want to load in pieces as the data is required, or because you are
doing something clever like generating the waveform on the fly.
This function creates a new audio stream and starts it playing. The
length is the size of each transfer buffer (in samples), which should
normally (but doesn't have to) be a power of two somewhere around 1k in
size. Larger buffers are more efficient and require fewer updates, but
result in more latency between you providing the data and it actually
being played. The bits parameter must be 8 or 16, freq is the sample
rate of the data in Hertz. The vol and pan values use the same 0-255
ranges as the regular sample playing functions. The stereo parameter
should be set to 1 for stereo streams, or 0 otherwise. If you want to
adjust the pitch, volume, or panning of a stream once it is playing,
you can use the regular voice_*() functions with stream->voice as a
parameter. The sample data is always in unsigned format, with stereo
waveforms consisting of alternating left/right samples, left sample
first.
See also:
install_sound,
get_audio_stream_buffer,
stop_audio_stream.
Destroys an audio stream when it is no longer required.
See also:
play_audio_stream.
You must call this function at regular intervals while an audio stream is
playing, to provide the next buffer of sample data (the smaller the
stream buffer size, the more often it must be called). If it returns
NULL, the stream is still playing the previous lot of data, so you don't
need to do anything. If it returns a value, that is the location of the
next buffer to be played, and you should load the appropriate number of
samples (however many you specified when creating the stream) to that
address, for example using an fread() from a disk file. After filling the
buffer with data, call free_audio_stream_buffer() to indicate that the
new data is now valid. Note that this function should not be called from
a timer handler...
See also:
play_audio_stream,
free_audio_stream_buffer.
Call this function after get_audio_stream_buffer() returns a non-NULL
address, to indicate that you have loaded a new block of samples to that
location and the data is now ready to be played.
See also:
get_audio_stream_buffer.
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