[openal] Programmatic access to sounds played through OpenAL
chris.kcat at gmail.com
Fri Sep 4 12:31:35 EDT 2020
On Thu, 3 Sep 2020 19:27:24 -0700
Karkala Hegde <khegde at usc.edu> wrote:
> Hi Chris,
> Thanks for your answer. I followed the same instructions you
> mentioned and tried writing the samples to a wave file, I assume I
> can create the buffer like so:
> int test_buffer;
> and then populate it using the code you provided.
> You mentioned that if we concatenated all the samples from each frame
> and played at normal speed it would sound the same. How do you
> convert the single dimension buffer array to a 2 channel stereo wav
> file (or two wav files, one for each channel)?
Normally audio is stored with interleaved channels, meaning the samples
for each channel follow consecutively.
buffer | buffer | buffer | buffer | ...
Left-0 | Right-0 | Left-1 | Right-1 | ...
This is how the loopback extension gives the audio. So if you have a
16-bit stereo sound stored in a
then to access sample 'n' of channel 'm', you would do
buffer[n*2 + m]
where '2' can be replaced with the number of channels if different. You
can alternatively declare a multidimensional array as long as the
channel count is fixed:
but that can only work if the channel count is known at compile time.
It won't work correctly if you get mono or other non-stereo data.
To concatenate interleaved audio buffers, just copy all the bytes
from each buffer as-is one after the other. The last channel's last
sample from the first buffer will be followed by the first channel's
first sample from the second buffer, which is correct.
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