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channel reorder patch by Florian Dietrich <flodt8 at yahoo.de>

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git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@13675 b3059339-0415-0410-9bf9-f77b7e298cf2
This commit is contained in:
faust3 2004-10-18 19:23:13 +00:00
parent a594e93e86
commit ae746ad8e8
1 changed files with 40 additions and 10 deletions
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50
libao2/ao_dsound.c
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@ -253,15 +253,6 @@ static int write_buffer(unsigned char *data, int len)
LPVOID lpvPtr2;
DWORD dwBytes2;
DWORD play_offset;
int space;
// make sure we have enough space to write data
IDirectSoundBuffer_GetCurrentPosition(hdsbuf,&play_offset,NULL);
space=buffer_size-(write_offset-play_offset);
if(space > buffer_size)space -= buffer_size; // write_offset < play_offset
if(space < len) len = space;
// Lock the buffer
res = IDirectSoundBuffer_Lock(hdsbuf,write_offset, len, &lpvPtr1, &dwBytes1, &lpvPtr2, &dwBytes2, 0);
// If the buffer was lost, restore and retry lock.
@ -274,11 +265,39 @@ static int write_buffer(unsigned char *data, int len)
if (SUCCEEDED(res))
{
// Write to pointers.
if( (ao_data.channels == 6) && (ao_data.format!=AFMT_AC3) ) {
// reorder channels while writing to pointers.
// it's this easy because buffer size and len are always
// aligned to multiples of channels*bytespersample
// there's probably some room for speed improvements here
const int chantable[6] = {0, 1, 4, 5, 2, 3}; // reorder "matrix"
int i, j;
int numsamp,sampsize;
sampsize = audio_out_format_bits(ao_data.format)>>3; // bytes per sample
numsamp = dwBytes1 / (ao_data.channels * sampsize); // number of samples for each channel in this buffer
for( i = 0; i < numsamp; i++ ) for( j = 0; j < ao_data.channels; j++ ) {
memcpy(lpvPtr1+(i*ao_data.channels*sampsize)+(chantable[j]*sampsize),data+(i*ao_data.channels*sampsize)+(j*sampsize),sampsize);
}
if (NULL != lpvPtr2 )
{
numsamp = dwBytes2 / (ao_data.channels * sampsize);
for( i = 0; i < numsamp; i++ ) for( j = 0; j < ao_data.channels; j++ ) {
memcpy(lpvPtr2+(i*ao_data.channels*sampsize)+(chantable[j]*sampsize),data+dwBytes1+(i*ao_data.channels*sampsize)+(j*sampsize),sampsize);
}
}
write_offset+=dwBytes1+dwBytes2;
if(write_offset>=buffer_size)write_offset=dwBytes2;
} else {
// Write to pointers without reordering.
memcpy(lpvPtr1,data,dwBytes1);
if (NULL != lpvPtr2 )memcpy(lpvPtr2,data+dwBytes1,dwBytes2);
write_offset+=dwBytes1+dwBytes2;
if(write_offset>=buffer_size)write_offset=dwBytes2;
}
// Release the data back to DirectSound.
res = IDirectSoundBuffer_Unlock(hdsbuf,lpvPtr1,dwBytes1,lpvPtr2,dwBytes2);
@ -409,6 +428,8 @@ static int init(int rate, int channels, int format, int flags)
res = IDirectSoundBuffer_SetFormat( hdspribuf, (WAVEFORMATEX *)&wformat );
if ( res != DS_OK ) mp_msg(MSGT_AO, MSGL_WARN,"ao_dsound: cannot set primary buffer format (%s), using standard setting (bad quality)", dserr2str(res));
mp_msg(MSGT_AO, MSGL_V, "ao_dsound: primary buffer created\n");
// now create the stream buffer
res = IDirectSound_CreateSoundBuffer(hds, &dsbdesc, &hdsbuf, NULL);
@ -497,6 +518,15 @@ static int get_space()
*/
static int play(void* data, int len, int flags)
{
DWORD play_offset;
int space;
// make sure we have enough space to write data
IDirectSoundBuffer_GetCurrentPosition(hdsbuf,&play_offset,NULL);
space=buffer_size-(write_offset-play_offset);
if(space > buffer_size)space -= buffer_size; // write_offset < play_offset
if(space < len) len = space;
len = (len / ao_data.outburst) * ao_data.outburst;
return write_buffer(data, len);
}