Re: [AD] docs: play_audio_stream clarification

[Eric Botcazou <ebotcazou@xxxxxxxxxx>]

> This function creates a new audio stream and starts playing it. The length
> is the size of each transfer buffer in sample frames, where a sample frame
> is a single sample point (i.e. value) for mono data or a pair of sample
> points for stereo data.

This is better (although I think "sample value" instead of "sample point" 
would be even better) but IMHO not yet crystal clear because...

> The length should normally be (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-&gtvoice as a parameter. The sample data are always in
> unsigned format. For stereo waveforms, the samples are interleaved, left
> first.

...this is confusing.  Above you were talking about "sample frame" and 
"sample point", now you are talking about "sample".

-- 
Eric Botcazou