dsound: Correct amount of buffers used for playing.

Fixes regression caused by ce06de4208
Amount of time per buffer was doubled, but amount of buffers wasn't 
halved, so latency was doubled.
This commit is contained in:
Maarten Lankhorst 2009-01-01 13:24:53 +01:00 committed by Alexandre Julliard
parent 02229896be
commit 94c620cf95
2 changed files with 3 additions and 3 deletions

View File

@ -1140,7 +1140,7 @@ IDirectSoundCaptureBufferImpl_Start(
if (device->buffer) {
int c;
DWORD blocksize = 4 * DSOUND_fraglen(device->pwfx->nSamplesPerSec, device->pwfx->nBlockAlign);
DWORD blocksize = DSOUND_fraglen(device->pwfx->nSamplesPerSec, device->pwfx->nBlockAlign);
device->nrofpwaves = device->buflen / blocksize + !!(device->buflen % blocksize);
TRACE("nrofpwaves=%d\n", device->nrofpwaves);

View File

@ -91,8 +91,8 @@ HRESULT mmErr(UINT err)
/* All default settings, you most likely don't want to touch these, see wiki on UsefulRegistryKeys */
int ds_emuldriver = 0;
int ds_hel_buflen = 32768 * 2;
int ds_snd_queue_max = 20;
int ds_snd_queue_min = 14;
int ds_snd_queue_max = 10;
int ds_snd_queue_min = 6;
int ds_snd_shadow_maxsize = 2;
int ds_hw_accel = DS_HW_ACCEL_FULL;
int ds_default_playback = 0;