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ao_dsound: support 6.1 and 7.1 channel configurations

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Instead of doing the channel reordering manually, use the existing
support in reorder_ch.c.

Untested.
This commit is contained in:
wm4 2012-04-20 04:52:51 +02:00 committed by wm4
parent 37c5c114af
commit 94f72b1e59
1 changed files with 43 additions and 30 deletions
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73
audio/out/ao_dsound.c
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@ -32,9 +32,12 @@
#include <dsound.h>
#include <math.h>
#include <libavutil/avutil.h>
#include "config.h"
#include "audio/format.h"
#include "ao.h"
#include "audio/reorder_ch.h"
#include "audio_out_internal.h"
#include "core/mp_msg.h"
#include "osdep/timer.h"
@ -104,10 +107,25 @@ typedef struct {
#endif
static const int channel_mask[] = {
SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_LOW_FREQUENCY,
SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT,
SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_LOW_FREQUENCY,
SPEAKER_FRONT_LEFT | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT | SPEAKER_LOW_FREQUENCY
/* 1 */ SPEAKER_FRONT_CENTER,
/* 2 */ SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT,
/* 3 */ SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT
| SPEAKER_LOW_FREQUENCY,
/* 4 */ SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT
| SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT,
/* 5 */ SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT
| SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT
| SPEAKER_LOW_FREQUENCY,
/* 6 */ SPEAKER_FRONT_LEFT | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT
| SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT
| SPEAKER_LOW_FREQUENCY,
/* 7 */ SPEAKER_FRONT_LEFT | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT
| SPEAKER_BACK_LEFT | SPEAKER_BACK_CENTER | SPEAKER_BACK_RIGHT
| SPEAKER_LOW_FREQUENCY,
/* 8 */ SPEAKER_FRONT_LEFT | SPEAKER_FRONT_CENTER | SPEAKER_FRONT_RIGHT
| SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT
| SPEAKER_SIDE_LEFT | SPEAKER_SIDE_RIGHT
| SPEAKER_LOW_FREQUENCY,
};
static HINSTANCE hdsound_dll = NULL; ///handle to the dll
@ -329,32 +347,27 @@ static int write_buffer(unsigned char *data, int len)
if (SUCCEEDED(res))
{
if( (ao_data.channels == 6) && !AF_FORMAT_IS_AC3(ao_data.format) ) {
// 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 = af_fmt2bits(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((char*)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((char*)lpvPtr2+(i*ao_data.channels*sampsize)+(chantable[j]*sampsize),data+dwBytes1+(i*ao_data.channels*sampsize)+(j*sampsize),sampsize);
}
}
if (!AF_FORMAT_IS_AC3(ao_data.format)) {
int sampsize = af_fmt2bits(ao_data.format) / 8;
reorder_channel_copy_nch(data,
AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
lpvPtr1,
AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT,
ao_data.channels,
dwBytes1 / sampsize,
sampsize);
if (lpvPtr2 != NULL)
reorder_channel_copy_nch(data + dwBytes1,
AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
lpvPtr2,
AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT,
ao_data.channels,
dwBytes2 / sampsize,
sampsize);
write_offset+=dwBytes1+dwBytes2;
if(write_offset>=buffer_size)write_offset=dwBytes2;
if(write_offset>=buffer_size)
write_offset=dwBytes2;
} else {
// Write to pointers without reordering.
memcpy(lpvPtr1,data,dwBytes1);
@ -432,7 +445,7 @@ static int init(int rate, int channels, int format, int flags)
DSBUFFERDESC dsbdesc;
//check if the channel count and format is supported in general
if (channels > 6) {
if (channels > FF_ARRAY_ELEMS(channel_mask)) {
UninitDirectSound();
mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: 8 channel audio not yet supported\n");
return 0;
@ -490,7 +503,7 @@ static int init(int rate, int channels, int format, int flags)
| DSBCAPS_CTRLVOLUME; /** volume control enabled */
if (channels > 2) {
wformat.dwChannelMask = channel_mask[channels - 3];
wformat.dwChannelMask = channel_mask[channels - 1];
wformat.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
wformat.Samples.wValidBitsPerSample = wformat.Format.wBitsPerSample;
// Needed for 5.1 on emu101k - shit soundblaster