Allegro Manual: Audio stream routines

# Audio stream routines

- free_audio_stream_buffer — Tells the audio stream player new data can be played.
- get_audio_stream_buffer — Tells you if you need to fill the audiostream or not.
- play_audio_stream — Creates a new audio stream and starts playing it.
- stop_audio_stream — Destroys an audio stream when it is no longer required.

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.

You can think of an AUDIOSTREAM structure as a wrapper around two audio buffers. The first thing you do is fill both buffers with sound data and let Allegro play them. Once the first buffer has been played, the second starts, and Allegro lets you know you have to fill the other one (i.e. graphics double buffering applied to sounds too big to fit into memory).

The implementation of the sound buffers uses normal SAMPLE structures, so you can use all the voice_*() functions to modify the audio streams. Read chapter "Digital sample routines", section "Voice control" for a list of additional functions you can use. Read chapter "Structures and types defined by Allegro" for the internals of the AUDIOSTREAM structure.

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)

Example:

>           /* 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");

Return value: This function returns a pointer to the audio stream or NULL if it could not be created.

> See also: install_sound, get_audio_stream_buffer, stop_audio_stream, AUDIOSTREAM, Voice control.

> Examples using this: exstream.

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

> See also: play_audio_stream.

> Examples using this: exstream.

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). This function should not be called from a timer handler. Example:

>           void *mem_chunk;
>           ...
>           while (TRUE) {
>              ...
>              mem_chunk = get_audio_stream_buffer(buffer);
>              if (mem_chunk != NULL) {
>                 /* Refill the stream buffer. */
>              }
>           }

Return value: 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.

> See also: play_audio_stream, free_audio_stream_buffer.

> Examples using this: exstream.

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. Example:

>           mem_chunk = get_audio_stream_buffer(buffer);
>           if (mem_chunk != NULL) {
>              /* Refill the stream buffer. */
>              ...
>              free_audio_stream_buffer(buffer);
>           }

> See also: get_audio_stream_buffer.

> Examples using this: exstream.

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