AUDIOSTREAM *play_audio_stream(int len, int bits, int stereo, int freq, int vol, int pan);
This function creates a new audio stream and starts playing it. The length is the size of each transfer buffer in sample frames (not bytes), where a sample frame is a single sample value for mono data or a pair of interleaved sample values (left first) for stereo data. The length should normally be (but doesn't have to be) a power of 2 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 format of the sample data is described in the SAMPLE entry of the "Structures and types defined by Allegro" chapter. The formula to get the size of the buffers in bytes could be:
bytes = length * (bits / 8) * (stereo ? 2 : 1)
/* Create a 22KHz 8bit mono audio stream. */ stream = play_audio_stream(1024, 8, FALSE, 22050, 255, 128); if (!stream) abort_on_error("Error creating audio stream!\n");
This function returns a pointer to the audio stream or NULL if it could not be created.
install_sound(3), get_audio_stream_buffer(3), stop_audio_stream(3), AUDIOSTREAM(3), exstream(3)
AUDIOSTREAM(3), exstream(3), get_audio_stream_buffer(3), install_sound(3), stop_audio_stream(3).