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-rw-r--r--audio/out/ao_dsound.c83
1 files changed, 15 insertions, 68 deletions
diff --git a/audio/out/ao_dsound.c b/audio/out/ao_dsound.c
index 9ad27306bd..b096dd28db 100644
--- a/audio/out/ao_dsound.c
+++ b/audio/out/ao_dsound.c
@@ -64,26 +64,6 @@ LIBAO_EXTERN(dsound)
static const GUID KSDATAFORMAT_SUBTYPE_PCM = {0x1,0x0000,0x0010, {0x80,0x00,0x00,0xaa,0x00,0x38,0x9b,0x71}};
-#define SPEAKER_FRONT_LEFT 0x1
-#define SPEAKER_FRONT_RIGHT 0x2
-#define SPEAKER_FRONT_CENTER 0x4
-#define SPEAKER_LOW_FREQUENCY 0x8
-#define SPEAKER_BACK_LEFT 0x10
-#define SPEAKER_BACK_RIGHT 0x20
-#define SPEAKER_FRONT_LEFT_OF_CENTER 0x40
-#define SPEAKER_FRONT_RIGHT_OF_CENTER 0x80
-#define SPEAKER_BACK_CENTER 0x100
-#define SPEAKER_SIDE_LEFT 0x200
-#define SPEAKER_SIDE_RIGHT 0x400
-#define SPEAKER_TOP_CENTER 0x800
-#define SPEAKER_TOP_FRONT_LEFT 0x1000
-#define SPEAKER_TOP_FRONT_CENTER 0x2000
-#define SPEAKER_TOP_FRONT_RIGHT 0x4000
-#define SPEAKER_TOP_BACK_LEFT 0x8000
-#define SPEAKER_TOP_BACK_CENTER 0x10000
-#define SPEAKER_TOP_BACK_RIGHT 0x20000
-#define SPEAKER_RESERVED 0x80000000
-
#if 0
#define DSSPEAKER_HEADPHONE 0x00000001
#define DSSPEAKER_MONO 0x00000002
@@ -107,28 +87,6 @@ typedef struct {
} WAVEFORMATEXTENSIBLE, *PWAVEFORMATEXTENSIBLE;
#endif
-static const int channel_mask[] = {
- /* 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
static LPDIRECTSOUND hds = NULL; ///direct sound object
static LPDIRECTSOUNDBUFFER hdspribuf = NULL; ///primary direct sound buffer
@@ -349,22 +307,9 @@ static int write_buffer(unsigned char *data, int len)
if (SUCCEEDED(res))
{
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);
+ memcpy(lpvPtr1, data, dwBytes1);
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);
+ memcpy(lpvPtr2, (char *)data + dwBytes1, dwBytes2);
write_offset+=dwBytes1+dwBytes2;
if(write_offset>=buffer_size)
@@ -432,7 +377,7 @@ static int control(int cmd, void *arg)
\param flags unused
\return 1=success 0=fail
*/
-static int init(int rate, int channels, int format, int flags)
+static int init(int rate, const struct mp_chmap *channels, int format, int flags)
{
int res;
if (!InitDirectSound()) return 0;
@@ -446,14 +391,17 @@ 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 > FF_ARRAY_ELEMS(channel_mask)) {
+ if (ao_data.channels.num > 8) {
+ // More than 8 channels might just work, but needs testing
UninitDirectSound();
- mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: 8 channel audio not yet supported\n");
+ mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: > 8 channel audio not yet supported\n");
return 0;
}
if (AF_FORMAT_IS_AC3(format))
format = AF_FORMAT_AC3_NE;
+ else
+ mp_chmap_reorder_to_waveext(&ao_data.channels);
switch(format){
case AF_FORMAT_AC3_NE:
case AF_FORMAT_S24_LE:
@@ -465,25 +413,24 @@ static int init(int rate, int channels, int format, int flags)
format=AF_FORMAT_S16_LE;
}
//fill global ao_data
- ao_data.channels = channels;
ao_data.samplerate = rate;
ao_data.format = format;
- ao_data.bps = channels * rate * (af_fmt2bits(format)>>3);
+ ao_data.bps = ao_data.channels.num * rate * (af_fmt2bits(format)>>3);
if(ao_data.buffersize==-1) ao_data.buffersize = ao_data.bps; // space for 1 sec
- mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Samplerate:%iHz Channels:%i Format:%s\n", rate, channels, af_fmt2str_short(format));
+ mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Samplerate:%iHz Channels:%i Format:%s\n", rate, ao_data.channels.num, af_fmt2str_short(format));
mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Buffersize:%d bytes (%d msec)\n", ao_data.buffersize, ao_data.buffersize / ao_data.bps * 1000);
//fill waveformatex
ZeroMemory(&wformat, sizeof(WAVEFORMATEXTENSIBLE));
- wformat.Format.cbSize = (channels > 2) ? sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX) : 0;
- wformat.Format.nChannels = channels;
+ wformat.Format.cbSize = (ao_data.channels.num > 2) ? sizeof(WAVEFORMATEXTENSIBLE)-sizeof(WAVEFORMATEX) : 0;
+ wformat.Format.nChannels = ao_data.channels.num;
wformat.Format.nSamplesPerSec = rate;
if (AF_FORMAT_IS_AC3(format)) {
wformat.Format.wFormatTag = WAVE_FORMAT_DOLBY_AC3_SPDIF;
wformat.Format.wBitsPerSample = 16;
wformat.Format.nBlockAlign = 4;
} else {
- wformat.Format.wFormatTag = (channels > 2) ? WAVE_FORMAT_EXTENSIBLE : WAVE_FORMAT_PCM;
+ wformat.Format.wFormatTag = (ao_data.channels.num > 2) ? WAVE_FORMAT_EXTENSIBLE : WAVE_FORMAT_PCM;
wformat.Format.wBitsPerSample = af_fmt2bits(format);
wformat.Format.nBlockAlign = wformat.Format.nChannels * (wformat.Format.wBitsPerSample >> 3);
}
@@ -503,8 +450,8 @@ static int init(int rate, int channels, int format, int flags)
| DSBCAPS_GLOBALFOCUS /** Allows background playing */
| DSBCAPS_CTRLVOLUME; /** volume control enabled */
- if (channels > 2) {
- wformat.dwChannelMask = channel_mask[channels - 1];
+ if (ao_data.channels.num > 2) {
+ wformat.dwChannelMask = mp_chmap_to_waveext(&ao_data.channels);
wformat.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
wformat.Samples.wValidBitsPerSample = wformat.Format.wBitsPerSample;
// Needed for 5.1 on emu101k - shit soundblaster
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-17 10:42:54 +0000