# Audio stream routines

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- free_audio_stream_buffer
- get_audio_stream_buffer
- play_audio_stream
- stop_audio_stream

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.

AUDIOSTREAM *play_audio_stream(int len, bits, stereo, freq, vol, pan);
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.

void stop_audio_stream(AUDIOSTREAM *stream);
Destroys an audio stream when it is no longer required.

void *get_audio_stream_buffer(AUDIOSTREAM *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...

void free_audio_stream_buffer(AUDIOSTREAM *stream);
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.

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