Commit 021019ff authored by Maarten Lankhorst's avatar Maarten Lankhorst Committed by Alexandre Julliard

dsound: Add an option to mix sound buffers in the mixer again.

parent 5f6ce2de
......@@ -296,7 +296,7 @@ static HRESULT WINAPI IDirectSoundBufferImpl_SetFrequency(
This->freqAdjust = ((DWORD64)This->freq << DSOUND_FREQSHIFT) / This->device->pwfx->nSamplesPerSec;
This->nAvgBytesPerSec = freq * This->pwfx->nBlockAlign;
DSOUND_RecalcFormat(This);
DSOUND_MixToTemporary(This, 0, This->buflen);
DSOUND_MixToTemporary(This, 0, This->buflen, FALSE);
}
RtlReleaseResource(&This->lock);
......@@ -733,9 +733,9 @@ static HRESULT WINAPI IDirectSoundBufferImpl_Unlock(
{
RtlAcquireResourceShared(&iter->lock, TRUE);
if (x1)
DSOUND_MixToTemporary(iter, (DWORD_PTR)p1 - (DWORD_PTR)iter->buffer->memory, x1);
DSOUND_MixToTemporary(iter, (DWORD_PTR)p1 - (DWORD_PTR)iter->buffer->memory, x1, FALSE);
if (x2)
DSOUND_MixToTemporary(iter, 0, x2);
DSOUND_MixToTemporary(iter, 0, x2, FALSE);
RtlReleaseResource(&iter->lock);
}
RtlReleaseResource(&This->device->buffer_list_lock);
......@@ -1223,7 +1223,7 @@ HRESULT IDirectSoundBufferImpl_Duplicate(
dsb->secondary = NULL;
dsb->tmp_buffer = NULL;
DSOUND_RecalcFormat(dsb);
DSOUND_MixToTemporary(dsb, 0, dsb->buflen);
DSOUND_MixToTemporary(dsb, 0, dsb->buflen, FALSE);
/* variable sized struct so calculate size based on format */
size = sizeof(WAVEFORMATEX) + pdsb->pwfx->cbSize;
......
......@@ -57,10 +57,6 @@
WINE_DEFAULT_DEBUG_CHANNEL(dsound);
#define DS_HEL_BUFLEN 0x8000 /* HEL: The buffer length of the emulated buffer */
#define DS_SND_QUEUE_MAX 10 /* max number of fragments to prebuffer, each fragment is approximately 10 ms long */
#define DS_SND_QUEUE_MIN 6 /* If the minimum of prebuffered fragments go below this, forcibly take all locks to prevent underruns */
DirectSoundDevice* DSOUND_renderer[MAXWAVEDRIVERS];
GUID DSOUND_renderer_guids[MAXWAVEDRIVERS];
GUID DSOUND_capture_guids[MAXWAVEDRIVERS];
......@@ -92,10 +88,12 @@ 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 = DS_HEL_BUFLEN;
int ds_snd_queue_max = DS_SND_QUEUE_MAX;
int ds_snd_queue_min = DS_SND_QUEUE_MIN;
int ds_hel_buflen = 32768;
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;
int ds_default_capture = 0;
......@@ -173,43 +171,38 @@ void setup_dsound_options(void)
}
if (!get_config_key( hkey, appkey, "DefaultPlayback", buffer, MAX_PATH ))
ds_default_playback = atoi(buffer);
ds_default_playback = atoi(buffer);
if (!get_config_key( hkey, appkey, "MaxShadowSize", buffer, MAX_PATH ))
ds_snd_shadow_maxsize = atoi(buffer);
if (!get_config_key( hkey, appkey, "DefaultCapture", buffer, MAX_PATH ))
ds_default_capture = atoi(buffer);
ds_default_capture = atoi(buffer);
if (!get_config_key( hkey, appkey, "DefaultSampleRate", buffer, MAX_PATH ))
ds_default_sample_rate = atoi(buffer);
ds_default_sample_rate = atoi(buffer);
if (!get_config_key( hkey, appkey, "DefaultBitsPerSample", buffer, MAX_PATH ))
ds_default_bits_per_sample = atoi(buffer);
ds_default_bits_per_sample = atoi(buffer);
if (appkey) RegCloseKey( appkey );
if (hkey) RegCloseKey( hkey );
if (ds_emuldriver)
WARN("ds_emuldriver = %d (default=0)\n",ds_emuldriver);
if (ds_hel_buflen != DS_HEL_BUFLEN)
WARN("ds_hel_buflen = %d (default=%d)\n",ds_hel_buflen ,DS_HEL_BUFLEN);
if (ds_snd_queue_max != DS_SND_QUEUE_MAX)
WARN("ds_snd_queue_max = %d (default=%d)\n",ds_snd_queue_max ,DS_SND_QUEUE_MAX);
if (ds_snd_queue_min != DS_SND_QUEUE_MIN)
WARN("ds_snd_queue_min = %d (default=%d)\n",ds_snd_queue_min ,DS_SND_QUEUE_MIN);
if (ds_hw_accel != DS_HW_ACCEL_FULL)
WARN("ds_hw_accel = %s (default=Full)\n",
ds_hw_accel==DS_HW_ACCEL_FULL ? "Full" :
ds_hw_accel==DS_HW_ACCEL_STANDARD ? "Standard" :
ds_hw_accel==DS_HW_ACCEL_BASIC ? "Basic" :
ds_hw_accel==DS_HW_ACCEL_EMULATION ? "Emulation" :
"Unknown");
if (ds_default_playback != 0)
WARN("ds_default_playback = %d (default=0)\n",ds_default_playback);
if (ds_default_capture != 0)
WARN("ds_default_capture = %d (default=0)\n",ds_default_playback);
if (ds_default_sample_rate != 44100)
WARN("ds_default_sample_rate = %d (default=44100)\n",ds_default_sample_rate);
if (ds_default_bits_per_sample != 16)
WARN("ds_default_bits_per_sample = %d (default=16)\n",ds_default_bits_per_sample);
TRACE("ds_emuldriver = %d\n", ds_emuldriver);
TRACE("ds_hel_buflen = %d\n", ds_hel_buflen);
TRACE("ds_snd_queue_max = %d\n", ds_snd_queue_max);
TRACE("ds_snd_queue_min = %d\n", ds_snd_queue_min);
TRACE("ds_hw_accel = %s\n",
ds_hw_accel==DS_HW_ACCEL_FULL ? "Full" :
ds_hw_accel==DS_HW_ACCEL_STANDARD ? "Standard" :
ds_hw_accel==DS_HW_ACCEL_BASIC ? "Basic" :
ds_hw_accel==DS_HW_ACCEL_EMULATION ? "Emulation" :
"Unknown");
TRACE("ds_default_playback = %d\n", ds_default_playback);
TRACE("ds_default_capture = %d\n", ds_default_playback);
TRACE("ds_default_sample_rate = %d\n", ds_default_sample_rate);
TRACE("ds_default_bits_per_sample = %d\n", ds_default_bits_per_sample);
TRACE("ds_snd_shadow_maxsize = %d\n", ds_snd_shadow_maxsize);
}
static const char * get_device_id(LPCGUID pGuid)
......
......@@ -35,6 +35,7 @@ extern int ds_emuldriver;
extern int ds_hel_buflen;
extern int ds_snd_queue_max;
extern int ds_snd_queue_min;
extern int ds_snd_shadow_maxsize;
extern int ds_hw_accel;
extern int ds_default_playback;
extern int ds_default_capture;
......@@ -181,7 +182,7 @@ struct IDirectSoundBufferImpl
DSVOLUMEPAN volpan;
DSBUFFERDESC dsbd;
/* used for frequency conversion (PerfectPitch) */
ULONG freqneeded, freqAdjust, freqAcc, freqAccNext;
ULONG freqneeded, freqAdjust, freqAcc, freqAccNext, resampleinmixer;
/* used for mixing */
DWORD primary_mixpos, buf_mixpos, sec_mixpos;
......@@ -449,7 +450,7 @@ void DSOUND_CheckEvent(const IDirectSoundBufferImpl *dsb, DWORD playpos, int len
void DSOUND_RecalcVolPan(PDSVOLUMEPAN volpan);
void DSOUND_AmpFactorToVolPan(PDSVOLUMEPAN volpan);
void DSOUND_RecalcFormat(IDirectSoundBufferImpl *dsb);
void DSOUND_MixToTemporary(const IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD mixlen);
void DSOUND_MixToTemporary(const IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD mixlen, BOOL inmixer);
DWORD DSOUND_secpos_to_bufpos(const IDirectSoundBufferImpl *dsb, DWORD secpos, DWORD secmixpos, DWORD* overshot);
void CALLBACK DSOUND_timer(UINT timerID, UINT msg, DWORD_PTR dwUser, DWORD_PTR dw1, DWORD_PTR dw2);
......
......@@ -193,6 +193,8 @@ void DSOUND_RecalcFormat(IDirectSoundBufferImpl *dsb)
dsb->convert = convertbpp[dsb->pwfx->wBitsPerSample/8 - 1][dsb->device->pwfx->wBitsPerSample/8 - 1];
dsb->resampleinmixer = FALSE;
if (needremix)
{
if (needresample)
......@@ -200,8 +202,12 @@ void DSOUND_RecalcFormat(IDirectSoundBufferImpl *dsb)
else
dsb->tmp_buffer_len = dsb->buflen / bAlign * pAlign;
dsb->max_buffer_len = dsb->tmp_buffer_len;
dsb->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, dsb->max_buffer_len);
FillMemory(dsb->tmp_buffer, dsb->tmp_buffer_len, dsb->device->pwfx->wBitsPerSample == 8 ? 128 : 0);
if ((dsb->max_buffer_len <= dsb->device->buflen || dsb->max_buffer_len < ds_snd_shadow_maxsize * 1024 * 1024) && ds_snd_shadow_maxsize >= 0)
dsb->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, dsb->max_buffer_len);
if (dsb->tmp_buffer)
FillMemory(dsb->tmp_buffer, dsb->tmp_buffer_len, dsb->device->pwfx->wBitsPerSample == 8 ? 128 : 0);
else
dsb->resampleinmixer = TRUE;
}
else dsb->max_buffer_len = dsb->tmp_buffer_len = dsb->buflen;
dsb->buf_mixpos = DSOUND_secpos_to_bufpos(dsb, dsb->sec_mixpos, 0, NULL);
......@@ -313,41 +319,52 @@ static inline DWORD DSOUND_BufPtrDiff(DWORD buflen, DWORD ptr1, DWORD ptr2)
* writepos = Starting position of changed buffer
* len = number of bytes to resample from writepos
*
* NOTE: writepos + len <= buflen, This function doesn't loop!
* NOTE: writepos + len <= buflen. When called by mixer, MixOne makes sure of this.
*/
void DSOUND_MixToTemporary(const IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD len)
void DSOUND_MixToTemporary(const IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD len, BOOL inmixer)
{
INT i, size;
BYTE *ibp, *obp, *ibp_begin, *obp_begin;
INT iAdvance = dsb->pwfx->nBlockAlign;
INT oAdvance = dsb->device->pwfx->nBlockAlign;
DWORD freqAcc, target_writepos, overshot;
DWORD freqAcc, target_writepos = 0, overshot, maxlen;
if (!dsb->tmp_buffer)
/* Nothing to do, already ideal format */
/* We resample only when needed */
if ((dsb->tmp_buffer && inmixer) || (!dsb->tmp_buffer && !inmixer) || dsb->resampleinmixer != inmixer)
return;
assert(writepos + len <= dsb->buflen);
if (inmixer && writepos + len < dsb->buflen)
len += dsb->pwfx->nBlockAlign;
maxlen = DSOUND_secpos_to_bufpos(dsb, len, 0, NULL);
ibp = dsb->buffer->memory + writepos;
ibp_begin = dsb->buffer->memory;
obp_begin = dsb->tmp_buffer;
if (!inmixer)
obp_begin = dsb->tmp_buffer;
else if (dsb->device->tmp_buffer_len < maxlen || !dsb->device->tmp_buffer)
{
dsb->device->tmp_buffer_len = maxlen;
if (dsb->device->tmp_buffer)
dsb->device->tmp_buffer = HeapReAlloc(GetProcessHeap(), 0, dsb->device->tmp_buffer, maxlen);
else
dsb->device->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, maxlen);
obp_begin = dsb->device->tmp_buffer;
}
else
obp_begin = dsb->device->tmp_buffer;
TRACE("(%p, %p)\n", dsb, ibp);
/* Check for the best case */
if ((dsb->freq == dsb->device->pwfx->nSamplesPerSec) &&
(dsb->pwfx->wBitsPerSample == dsb->device->pwfx->wBitsPerSample) &&
(dsb->pwfx->nChannels == dsb->device->pwfx->nChannels)) {
obp = dsb->tmp_buffer + writepos;
/* Why would we need a temporary buffer if we do best case? */
FIXME("(%p) Why do we resample for best case??? Bad!!\n", dsb);
CopyMemory(obp, ibp, len);
return;
}
/* Check for same sample rate */
if (dsb->freq == dsb->device->pwfx->nSamplesPerSec) {
TRACE("(%p) Same sample rate %d = primary %d\n", dsb,
dsb->freq, dsb->device->pwfx->nSamplesPerSec);
obp = dsb->tmp_buffer + writepos/iAdvance*oAdvance;
obp = obp_begin;
if (!inmixer)
obp += writepos/iAdvance*oAdvance;
for (i = 0; i < len; i += iAdvance) {
cp_fields(dsb, ibp, obp);
ibp += iAdvance;
......@@ -375,7 +392,10 @@ void DSOUND_MixToTemporary(const IDirectSoundBufferImpl *dsb, DWORD writepos, DW
TRACE("Overshot: %d, freqAcc: %04x\n", overshot, freqAcc);
}
obp = dsb->tmp_buffer + target_writepos;
if (!inmixer)
obp = obp_begin + target_writepos;
else obp = obp_begin;
/* FIXME: Small problem here when we're overwriting buf_mixpos, it then STILL uses old freqAcc, not sure if it matters or not */
while (size > 0) {
cp_fields(dsb, ibp, obp);
......@@ -393,14 +413,17 @@ void DSOUND_MixToTemporary(const IDirectSoundBufferImpl *dsb, DWORD writepos, DW
/** Apply volume to the given soundbuffer from (primary) position writepos and length len
* Returns: NULL if no volume needs to be applied
* or else a memory handle that holds 'len' volume adjusted buffer */
static LPBYTE DSOUND_MixerVol(const IDirectSoundBufferImpl *dsb, DWORD writepos, INT len)
static LPBYTE DSOUND_MixerVol(const IDirectSoundBufferImpl *dsb, INT len)
{
INT i;
BYTE *bpc;
INT16 *bps, *mems;
DWORD vLeft, vRight;
INT nChannels = dsb->device->pwfx->nChannels;
LPBYTE mem = (dsb->tmp_buffer ? dsb->tmp_buffer : dsb->buffer->memory)+writepos;
LPBYTE mem = (dsb->tmp_buffer ? dsb->tmp_buffer : dsb->buffer->memory) + dsb->buf_mixpos;
if (dsb->resampleinmixer)
mem = dsb->device->tmp_buffer;
TRACE("(%p,%d)\n",dsb,len);
TRACE("left = %x, right = %x\n", dsb->volpan.dwTotalLeftAmpFactor,
......@@ -425,12 +448,15 @@ static LPBYTE DSOUND_MixerVol(const IDirectSoundBufferImpl *dsb, DWORD writepos,
if (dsb->device->tmp_buffer_len < len || !dsb->device->tmp_buffer)
{
/* If we just resampled in DSOUND_MixToTemporary, we shouldn't need to resize here */
assert(!dsb->resampleinmixer);
dsb->device->tmp_buffer_len = len;
if (dsb->device->tmp_buffer)
dsb->device->tmp_buffer = HeapReAlloc(GetProcessHeap(), 0, dsb->device->tmp_buffer, len);
else
dsb->device->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, len);
}
bpc = dsb->device->tmp_buffer;
bps = (INT16 *)bpc;
mems = (INT16 *)mem;
......@@ -494,8 +520,13 @@ static DWORD DSOUND_MixInBuffer(IDirectSoundBufferImpl *dsb, DWORD writepos, DWO
len -= len % nBlockAlign; /* data alignment */
}
/* Resample buffer to temporary buffer specifically allocated for this purpose, if needed */
DSOUND_MixToTemporary(dsb, dsb->sec_mixpos, DSOUND_bufpos_to_secpos(dsb, dsb->buf_mixpos+len) - dsb->sec_mixpos, TRUE);
if (dsb->resampleinmixer)
ibuf = dsb->device->tmp_buffer;
/* Apply volume if needed */
volbuf = DSOUND_MixerVol(dsb, dsb->buf_mixpos, len);
volbuf = DSOUND_MixerVol(dsb, len);
if (volbuf)
ibuf = volbuf;
......
......@@ -547,7 +547,7 @@ HRESULT DSOUND_PrimarySetFormat(DirectSoundDevice *device, LPCWAVEFORMATEX wfex,
(*dsb)->freqAdjust = ((DWORD64)(*dsb)->freq << DSOUND_FREQSHIFT) / device->pwfx->nSamplesPerSec;
DSOUND_RecalcFormat((*dsb));
DSOUND_MixToTemporary((*dsb), 0, (*dsb)->buflen);
DSOUND_MixToTemporary((*dsb), 0, (*dsb)->buflen, FALSE);
(*dsb)->primary_mixpos = 0;
RtlReleaseResource(&(*dsb)->lock);
......
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