| [AD] audio input latency, changes &new func proposal |
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i've tracked down the audio input buffer size (which effects the latency)
to this piece of code:
from /allegro4.1.11/src/win/wdsinput.c
int digi_directsound_rec_start()
{
...
460: dscBufDesc.dwBufferBytes = dsc_buf_wfx.nAvgBytesPerSec;
...
}
and here, at line 240:
static int get_capture_format_support()
{
...
237: test_wfx->nSamplesPerSec = ds_formats[i].freq;
238: test_wfx->wBitsPerSample = ds_formats[i].bits;
239: test_wfx->nBlockAlign = test_wfx->nChannels *
(test_wfx->wBitsPerSample / 8);
240: test_wfx->nAvgBytesPerSec = test_wfx->nSamplesPerSec *
test_wfx->nBlockAlign;
241: test_wfx->cbSize = 0;
...
}
Q. can the buffer sizes be changed so they return smaller buffers (low
latency) ?
if the buffer size can be adjustable, can i propose a new function
called set_sound_input_buffer_size(size_t i);
aj.