Skip to content

W
wine-winehq
Commit f88f3d41 authored by Alexander E. Patrakov's avatar Alexander E. Patrakov Committed by Alexandre Julliard
Browse files
Options

dsound: Replace convert() functions with get() and put().

parent 0195e019
Expand all Hide whitespace changes
Inline Side-by-side
Showing with 50 additions and 39 deletions
dlls/dsound/dsound_private.h
View file @ f88f3d41
......@@ -52,8 +52,10 @@ typedef struct DirectSoundDevice DirectSoundDevice;
typedef struct DirectSoundCaptureDevice DirectSoundCaptureDevice;
/* dsound_convert.h */
typedef void (*bitsconvertfunc)(const void *, void *, UINT, UINT, INT, UINT, UINT);
extern const bitsconvertfunc convertbpp[5][4] DECLSPEC_HIDDEN;
typedef float (*bitsgetfunc)(const IDirectSoundBufferImpl *, DWORD, DWORD);
typedef void (*bitsputfunc)(const IDirectSoundBufferImpl *, DWORD, DWORD, float);
extern const bitsgetfunc getbpp[5] DECLSPEC_HIDDEN;
extern const bitsputfunc putbpp[4] DECLSPEC_HIDDEN;
typedef void (*mixfunc)(const void *, void *, unsigned);
extern const mixfunc mixfunctions[4] DECLSPEC_HIDDEN;
typedef void (*normfunc)(const void *, void *, unsigned);
......@@ -190,8 +192,10 @@ struct IDirectSoundBufferImpl
DS3DBUFFER ds3db_ds3db;
LONG ds3db_lVolume;
BOOL ds3db_need_recalc;
/* IKsPropertySet fields */
bitsconvertfunc convert;
/* Used for bit depth conversion */
bitsgetfunc get;
bitsputfunc put;
struct list entry;
};
......
dlls/dsound/mixer.c
View file @ f88f3d41
......@@ -200,10 +200,8 @@ void DSOUND_RecalcFormat(IDirectSoundBufferImpl *dsb)
dsb->freqAcc = dsb->freqAccNext = 0;
dsb->freqneeded = needresample;
if (ieee)
dsb->convert = convertbpp[4][dsb->device->pwfx->wBitsPerSample/8 - 1];
else
dsb->convert = convertbpp[dsb->pwfx->wBitsPerSample/8 - 1][dsb->device->pwfx->wBitsPerSample/8 - 1];
dsb->get = ieee ? getbpp[4] : getbpp[dsb->pwfx->wBitsPerSample/8 - 1];
dsb->put = putbpp[dsb->device->pwfx->wBitsPerSample/8 - 1];
if (needremix)
{
......@@ -274,37 +272,47 @@ void DSOUND_CheckEvent(const IDirectSoundBufferImpl *dsb, DWORD playpos, int len
* Copy a single frame from the given input buffer to the given output buffer.
* Translate 8 <-> 16 bits and mono <-> stereo
*/
static inline void cp_fields(const IDirectSoundBufferImpl *dsb, const BYTE *ibuf, BYTE *obuf,
static inline void cp_fields(const IDirectSoundBufferImpl *dsb, const BYTE *ibuf,
UINT istride, UINT ostride, UINT count, UINT freqAcc, UINT adj)
{
DirectSoundDevice *device = dsb->device;
INT istep = dsb->pwfx->wBitsPerSample / 8, ostep = device->pwfx->wBitsPerSample / 8;
if (device->pwfx->nChannels == dsb->pwfx->nChannels ||
(device->pwfx->nChannels == 2 && dsb->pwfx->nChannels == 6) ||
(device->pwfx->nChannels == 8 && dsb->pwfx->nChannels == 2) ||
(device->pwfx->nChannels == 6 && dsb->pwfx->nChannels == 2)) {
dsb->convert(ibuf, obuf, istride, ostride, count, freqAcc, adj);
if (device->pwfx->nChannels == 2 || dsb->pwfx->nChannels == 2)
dsb->convert(ibuf + istep, obuf + ostep, istride, ostride, count, freqAcc, adj);
return;
}
if (device->pwfx->nChannels == 1 && dsb->pwfx->nChannels == 2)
{
dsb->convert(ibuf, obuf, istride, ostride, count, freqAcc, adj);
return;
float value;
ULONG adv;
DWORD ipos = dsb->sec_mixpos, opos = 0;
while (count-- > 0) {
if (device->pwfx->nChannels == dsb->pwfx->nChannels ||
(device->pwfx->nChannels == 2 && dsb->pwfx->nChannels == 6) ||
(device->pwfx->nChannels == 8 && dsb->pwfx->nChannels == 2) ||
(device->pwfx->nChannels == 6 && dsb->pwfx->nChannels == 2)) {
value = dsb->get(dsb, ipos, 0);
dsb->put(dsb, opos, 0, value);
if (device->pwfx->nChannels == 2 || dsb->pwfx->nChannels == 2) {
value = dsb->get(dsb, ipos, 1);
dsb->put(dsb, opos, 1, value);
}
}
if (device->pwfx->nChannels == 1 && dsb->pwfx->nChannels == 2)
{
float val = (dsb->get(dsb, ipos, 0) + dsb->get(dsb, ipos, 1)) / 2.;
dsb->put(dsb, opos, 0, val);
}
if (device->pwfx->nChannels == 2 && dsb->pwfx->nChannels == 1)
{
value = dsb->get(dsb, ipos, 0);
dsb->put(dsb, opos, 0, value);
dsb->put(dsb, opos, 1, value);
}
freqAcc += adj;
adv = (freqAcc >> DSOUND_FREQSHIFT);
freqAcc &= (1 << DSOUND_FREQSHIFT) - 1;
ipos += adv * istride;
opos += ostride;
}
if (device->pwfx->nChannels == 2 && dsb->pwfx->nChannels == 1)
{
dsb->convert(ibuf, obuf, istride, ostride, count, freqAcc, adj);
dsb->convert(ibuf, obuf + ostep, istride, ostride, count, freqAcc, adj);
return;
}
WARN("Unable to remap channels: device=%u, buffer=%u\n", device->pwfx->nChannels,
dsb->pwfx->nChannels);
}
/**
......@@ -338,7 +346,7 @@ static inline DWORD DSOUND_BufPtrDiff(DWORD buflen, DWORD ptr1, DWORD ptr2)
static void DSOUND_MixToTemporary(const IDirectSoundBufferImpl *dsb, DWORD tmp_len)
{
INT size;
BYTE *ibp, *obp;
BYTE *ibp;
INT iAdvance = dsb->pwfx->nBlockAlign;
INT oAdvance = dsb->device->pwfx->nBlockAlign;
DWORD freqAcc;
......@@ -352,7 +360,6 @@ static void DSOUND_MixToTemporary(const IDirectSoundBufferImpl *dsb, DWORD tmp_l
else
dsb->device->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, tmp_len);
}
obp = dsb->device->tmp_buffer;
TRACE("(%p, %p)\n", dsb, ibp);
size = tmp_len / oAdvance;
......@@ -362,7 +369,7 @@ static void DSOUND_MixToTemporary(const IDirectSoundBufferImpl *dsb, DWORD tmp_l
TRACE("(%p) Same sample rate %d = primary %d\n", dsb,
dsb->freq, dsb->device->pwfx->nSamplesPerSec);
cp_fields(dsb, ibp, obp, iAdvance, oAdvance, size, 0, 1 << DSOUND_FREQSHIFT);
cp_fields(dsb, ibp, iAdvance, oAdvance, size, 0, 1 << DSOUND_FREQSHIFT);
return;
}
......@@ -372,7 +379,7 @@ static void DSOUND_MixToTemporary(const IDirectSoundBufferImpl *dsb, DWORD tmp_l
DSOUND_secpos_to_bufpos(dsb, dsb->sec_mixpos, dsb->sec_mixpos, &freqAcc);
/* FIXME: Small problem here when we're overwriting buf_mixpos, it then STILL uses old freqAcc, not sure if it matters or not */
cp_fields(dsb, ibp, obp, iAdvance, oAdvance, size, freqAcc, dsb->freqAdjust);
cp_fields(dsb, ibp, iAdvance, oAdvance, size, freqAcc, dsb->freqAdjust);
}
/** Apply volume to the given soundbuffer from (primary) position writepos and length len
......
Markdown is supported
0% or
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment