Commit 0036298a authored by Maarten Lankhorst's avatar Maarten Lankhorst Committed by Alexandre Julliard

dsound: Add some comments from earlier patch that makes code a little better understandable.

parent 777ba7da
...@@ -98,6 +98,14 @@ void DSOUND_RecalcFormat(IDirectSoundBufferImpl *dsb) ...@@ -98,6 +98,14 @@ void DSOUND_RecalcFormat(IDirectSoundBufferImpl *dsb)
dsb->writelead = (dsb->freq / 100) * dsb->pwfx->nBlockAlign; dsb->writelead = (dsb->freq / 100) * dsb->pwfx->nBlockAlign;
} }
/**
* Check for application callback requests for when the play position
* reaches certain points.
*
* The offsets that will be triggered will be those between the recorded
* "last played" position for the buffer (i.e. dsb->playpos) and "len" bytes
* beyond that position.
*/
void DSOUND_CheckEvent(IDirectSoundBufferImpl *dsb, int len) void DSOUND_CheckEvent(IDirectSoundBufferImpl *dsb, int len)
{ {
int i; int i;
...@@ -163,6 +171,10 @@ static inline BYTE cvtS16toU8(INT16 s) ...@@ -163,6 +171,10 @@ static inline BYTE cvtS16toU8(INT16 s)
return (s >> 8) ^ (unsigned char)0x80; return (s >> 8) ^ (unsigned char)0x80;
} }
/**
* 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, BYTE *ibuf, BYTE *obuf ) static inline void cp_fields(const IDirectSoundBufferImpl *dsb, BYTE *ibuf, BYTE *obuf )
{ {
DirectSoundDevice * device = dsb->device; DirectSoundDevice * device = dsb->device;
...@@ -209,7 +221,24 @@ static inline void cp_fields(const IDirectSoundBufferImpl *dsb, BYTE *ibuf, BYTE ...@@ -209,7 +221,24 @@ static inline void cp_fields(const IDirectSoundBufferImpl *dsb, BYTE *ibuf, BYTE
} }
} }
/* Now with PerfectPitch (tm) technology */ /**
* Mix at most the given amount of data into the given device buffer from the
* given secondary buffer, starting from the dsb's first currently unmixed
* frame (buf_mixpos), translating frequency (pitch), stereo/mono and
* bits-per-sample. The secondary buffer sample is looped if it is not
* long enough and it is a looping buffer.
* (Doesn't perform any mixing - this is a straight copy operation).
*
* Now with PerfectPitch (tm) technology
*
* dsb = the secondary buffer
* buf = the device buffer
* len = number of bytes to store in the device buffer
*
* Returns: the number of bytes read from the secondary buffer
* (ie. len, adjusted for frequency, number of channels and sample size,
* and limited by buffer length for non-looping buffers)
*/
static INT DSOUND_MixerNorm(IDirectSoundBufferImpl *dsb, BYTE *buf, INT len) static INT DSOUND_MixerNorm(IDirectSoundBufferImpl *dsb, BYTE *buf, INT len)
{ {
INT i, size, ipos, ilen; INT i, size, ipos, ilen;
...@@ -356,6 +385,10 @@ static void DSOUND_MixerVol(IDirectSoundBufferImpl *dsb, BYTE *buf, INT len) ...@@ -356,6 +385,10 @@ static void DSOUND_MixerVol(IDirectSoundBufferImpl *dsb, BYTE *buf, INT len)
} }
} }
/**
* Make sure the device's tmp_buffer is at least the given size. Return a
* pointer to it.
*/
static LPBYTE DSOUND_tmpbuffer(DirectSoundDevice *device, DWORD len) static LPBYTE DSOUND_tmpbuffer(DirectSoundDevice *device, DWORD len)
{ {
TRACE("(%p,%d)\n", device, len); TRACE("(%p,%d)\n", device, len);
...@@ -372,6 +405,19 @@ static LPBYTE DSOUND_tmpbuffer(DirectSoundDevice *device, DWORD len) ...@@ -372,6 +405,19 @@ static LPBYTE DSOUND_tmpbuffer(DirectSoundDevice *device, DWORD len)
return device->tmp_buffer; return device->tmp_buffer;
} }
/**
* Mix (at most) the given number of bytes into the given position of the
* device buffer, from the secondary buffer "dsb" (starting at the current
* mix position for that buffer).
*
* Returns the number of bytes actually mixed into the device buffer. This
* will match fraglen unless the end of the secondary buffer is reached
* (and it is not looping).
*
* dsb = the secondary buffer to mix from
* writepos = position (offset) in device buffer to write at
* fraglen = number of bytes to mix
*/
static DWORD DSOUND_MixInBuffer(IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD fraglen) static DWORD DSOUND_MixInBuffer(IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD fraglen)
{ {
INT i, len, ilen, field, todo; INT i, len, ilen, field, todo;
...@@ -381,9 +427,15 @@ static DWORD DSOUND_MixInBuffer(IDirectSoundBufferImpl *dsb, DWORD writepos, DWO ...@@ -381,9 +427,15 @@ static DWORD DSOUND_MixInBuffer(IDirectSoundBufferImpl *dsb, DWORD writepos, DWO
len = fraglen; len = fraglen;
if (!(dsb->playflags & DSBPLAY_LOOPING)) { if (!(dsb->playflags & DSBPLAY_LOOPING)) {
/* This buffer is not looping, so make sure the requested
* length will not take us past the end of the buffer */
int secondary_remainder = dsb->buflen - dsb->buf_mixpos; int secondary_remainder = dsb->buflen - dsb->buf_mixpos;
int adjusted_remainder = MulDiv(dsb->device->pwfx->nAvgBytesPerSec, secondary_remainder, dsb->nAvgBytesPerSec); int adjusted_remainder = MulDiv(dsb->device->pwfx->nAvgBytesPerSec, secondary_remainder, dsb->nAvgBytesPerSec);
assert(adjusted_remainder >= 0); assert(adjusted_remainder >= 0);
/* The adjusted remainder must be at least one sample,
* otherwise we will never reach the end of the
* secondary buffer, as there will perpetually be a
* fractional remainder */
TRACE("secondary_remainder = %d, adjusted_remainder = %d, len = %d\n", secondary_remainder, adjusted_remainder, len); TRACE("secondary_remainder = %d, adjusted_remainder = %d, len = %d\n", secondary_remainder, adjusted_remainder, len);
if (adjusted_remainder < len) { if (adjusted_remainder < len) {
TRACE("clipping len to remainder of secondary buffer\n"); TRACE("clipping len to remainder of secondary buffer\n");
...@@ -404,12 +456,16 @@ static DWORD DSOUND_MixInBuffer(IDirectSoundBufferImpl *dsb, DWORD writepos, DWO ...@@ -404,12 +456,16 @@ static DWORD DSOUND_MixInBuffer(IDirectSoundBufferImpl *dsb, DWORD writepos, DWO
TRACE("MixInBuffer (%p) len = %d, dest = %d\n", dsb, len, writepos); TRACE("MixInBuffer (%p) len = %d, dest = %d\n", dsb, len, writepos);
/* first, copy the data from the DirectSoundBuffer into the temporary
buffer, translating frequency/bits-per-sample/number-of-channels
to match the device settings */
ilen = DSOUND_MixerNorm(dsb, ibuf, len); ilen = DSOUND_MixerNorm(dsb, ibuf, len);
if ((dsb->dsbd.dwFlags & DSBCAPS_CTRLPAN) || if ((dsb->dsbd.dwFlags & DSBCAPS_CTRLPAN) ||
(dsb->dsbd.dwFlags & DSBCAPS_CTRLVOLUME) || (dsb->dsbd.dwFlags & DSBCAPS_CTRLVOLUME) ||
(dsb->dsbd.dwFlags & DSBCAPS_CTRL3D)) (dsb->dsbd.dwFlags & DSBCAPS_CTRL3D))
DSOUND_MixerVol(dsb, ibuf, len); DSOUND_MixerVol(dsb, ibuf, len);
/* Now mix the temporary buffer into the devices main buffer */
if (dsb->device->pwfx->wBitsPerSample == 8) { if (dsb->device->pwfx->wBitsPerSample == 8) {
BYTE *obuf = dsb->device->buffer + writepos; BYTE *obuf = dsb->device->buffer + writepos;
...@@ -667,8 +723,23 @@ void DSOUND_ForceRemix(IDirectSoundBufferImpl *dsb) ...@@ -667,8 +723,23 @@ void DSOUND_ForceRemix(IDirectSoundBufferImpl *dsb)
LeaveCriticalSection(&dsb->lock); LeaveCriticalSection(&dsb->lock);
} }
/**
* Mix some frames from the given secondary buffer "dsb" into the device
* primary buffer.
*
* dsb = the secondary buffer
* playpos = the current play position in the device buffer (primary buffer)
* writepos = the current safe-to-write position in the device buffer
* mixlen = the maximum number of bytes in the primary buffer to mix, from the
* current writepos.
*
* Returns: the number of bytes beyond the writepos that were mixed.
*/
static DWORD DSOUND_MixOne(IDirectSoundBufferImpl *dsb, DWORD playpos, DWORD writepos, DWORD mixlen) static DWORD DSOUND_MixOne(IDirectSoundBufferImpl *dsb, DWORD playpos, DWORD writepos, DWORD mixlen)
{ {
/* The buffer's primary_mixpos may be before or after the the device
* buffer's mixpos, but both must be ahead of writepos. */
DWORD len, slen; DWORD len, slen;
/* determine this buffer's write position */ /* determine this buffer's write position */
DWORD buf_writepos = DSOUND_CalcPlayPosition(dsb, writepos, writepos); DWORD buf_writepos = DSOUND_CalcPlayPosition(dsb, writepos, writepos);
...@@ -823,6 +894,19 @@ post_mix: ...@@ -823,6 +894,19 @@ post_mix:
return slen; return slen;
} }
/**
* For a DirectSoundDevice, go through all the currently playing buffers and
* mix them in to the device buffer.
*
* playpos = the current play position in the primary buffer
* writepos = the current safe-to-write position in the primary buffer
* mixlen = the maximum amount to mix into the primary buffer
* (beyond the current writepos)
* recover = true if the sound device may have been reset and the write
* position in the device buffer changed
*
* Returns: the length beyond the writepos that was mixed to.
*/
static DWORD DSOUND_MixToPrimary(DirectSoundDevice *device, DWORD playpos, DWORD writepos, DWORD mixlen, BOOL recover) static DWORD DSOUND_MixToPrimary(DirectSoundDevice *device, DWORD playpos, DWORD writepos, DWORD mixlen, BOOL recover)
{ {
INT i, len, maxlen = 0; INT i, len, maxlen = 0;
...@@ -941,6 +1025,11 @@ void DSOUND_WaveQueue(DirectSoundDevice *device, DWORD mixq) ...@@ -941,6 +1025,11 @@ void DSOUND_WaveQueue(DirectSoundDevice *device, DWORD mixq)
/* #define SYNC_CALLBACK */ /* #define SYNC_CALLBACK */
/**
* Perform mixing for a Direct Sound device. That is, go through all the
* secondary buffers (the sound bites currently playing) and mix them in
* to the primary buffer (the device buffer).
*/
static void DSOUND_PerformMix(DirectSoundDevice *device) static void DSOUND_PerformMix(DirectSoundDevice *device)
{ {
int nfiller; int nfiller;
......
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