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