Fixed bug in capture stop. Only current header should be returned to
app.
Make wave fragment size adjustable to guarantee better than 10 ms
latency. Makes a difference for 8 kHz voice apps.
diff --git a/dlls/winmm/wineoss/audio.c b/dlls/winmm/wineoss/audio.c
index 00d47db..7aee153 100644
--- a/dlls/winmm/wineoss/audio.c
+++ b/dlls/winmm/wineoss/audio.c
@@ -375,7 +375,7 @@
error:
close(fd);
return WAVERR_BADFORMAT;
- error2:
+error2:
close(fd);
return MMSYSERR_ERROR;
}
@@ -1534,12 +1534,21 @@
* let's try to fragment the above 64KB (256 * 2^8) */
audio_fragment = 0x01000008;
} else {
- /* shockwave player uses only 4 1k-fragments at a rate of 22050 bytes/sec
- * thus leading to 46ms per fragment, and a turnaround time of 185ms
+ /* A wave device must have a worst case latency of 10 ms so calculate
+ * the largest fragment size less than 10 ms long.
*/
- /* 16 fragments max, 2^10=1024 bytes per fragment */
- audio_fragment = 0x000F000A;
+ int fsize = lpDesc->lpFormat->nAvgBytesPerSec / 100; /* 10 ms chunk */
+ int shift = 0;
+ while ((1 << shift) <= fsize)
+ shift++;
+ shift--;
+ audio_fragment = 0x00100000 + shift; /* 16 fragments of 2^shift */
}
+
+ TRACE("using %d %d byte fragments (%ld ms)\n", audio_fragment >> 16,
+ 1 << (audio_fragment & 0xffff),
+ ((1 << (audio_fragment & 0xffff)) * 1000) / lpDesc->lpFormat->nAvgBytesPerSec);
+
if (wwo->state != WINE_WS_CLOSED) return MMSYSERR_ALLOCATED;
/* we want to be able to mmap() the device, which means it must be opened readable,
* otherwise mmap() will fail (at least under Linux) */
@@ -2794,6 +2803,34 @@
}
break;
case WINE_WM_STOPPING:
+ wwi->state = WINE_WS_STOPPED;
+ wwi->dwTotalRecorded = 0;
+ if (wwi->state != WINE_WS_STOPPED)
+ {
+ if (wwi->ossdev->bTriggerSupport)
+ {
+ /* stop the recording */
+ wwi->ossdev->bInputEnabled = FALSE;
+ enable = getEnables(wwi->ossdev);
+ if (ioctl(wwi->ossdev->fd, SNDCTL_DSP_SETTRIGGER, &enable) < 0) {
+ wwi->ossdev->bInputEnabled = FALSE;
+ ERR("ioctl(%s, SNDCTL_DSP_SETTRIGGER) failed (%s)\n", wwi->ossdev->dev_name, strerror(errno));
+ }
+ }
+ /* return current buffer to app */
+ lpWaveHdr = wwi->lpQueuePtr;
+ if (lpWaveHdr)
+ {
+ LPWAVEHDR lpNext = lpWaveHdr->lpNext;
+ TRACE("stop %p %p\n", lpWaveHdr, lpWaveHdr->lpNext);
+ lpWaveHdr->dwFlags &= ~WHDR_INQUEUE;
+ lpWaveHdr->dwFlags |= WHDR_DONE;
+ widNotifyClient(wwi, WIM_DATA, (DWORD)lpWaveHdr, 0);
+ wwi->lpQueuePtr = lpNext;
+ }
+ }
+ SetEvent(ev);
+ break;
case WINE_WM_RESETTING:
wwi->state = WINE_WS_STOPPED;
wwi->dwTotalRecorded = 0;
@@ -2888,18 +2925,21 @@
* to do hardware playback without hardware capture. */
audio_fragment = wwi->ossdev->audio_fragment;
} else {
- /* This is actually hand tuned to work so that my SB Live:
- * - does not skip
- * - does not buffer too much
- * when sending with the Shoutcast winamp plugin
- */
- /* 15 fragments max, 2^10 = 1024 bytes per fragment */
- audio_fragment = 0x000F000A;
+ /* A wave device must have a worst case latency of 10 ms so calculate
+ * the largest fragment size less than 10 ms long.
+ */
+ int fsize = lpDesc->lpFormat->nAvgBytesPerSec / 100; /* 10 ms chunk */
+ int shift = 0;
+ while ((1 << shift) <= fsize)
+ shift++;
+ shift--;
+ audio_fragment = 0x00100000 + shift; /* 16 fragments of 2^shift */
}
}
- TRACE("using %d %d byte fragments\n", audio_fragment >> 16,
- 1 << (audio_fragment & 0xffff));
+ TRACE("using %d %d byte fragments (%ld ms)\n", audio_fragment >> 16,
+ 1 << (audio_fragment & 0xffff),
+ ((1 << (audio_fragment & 0xffff)) * 1000) / lpDesc->lpFormat->nAvgBytesPerSec);
ret = OSS_OpenDevice(wwi->ossdev, O_RDONLY, &audio_fragment,
1,
