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-rw-r--r--audio/out/ao_alsa.c32
-rw-r--r--audio/out/ao_coreaudio_utils.c3
-rw-r--r--audio/out/ao_dsound.c6
-rw-r--r--audio/out/ao_oss.c58
-rw-r--r--audio/out/ao_pcm.c21
-rw-r--r--audio/out/ao_pulse.c9
-rw-r--r--audio/out/ao_rsound.c42
-rw-r--r--audio/out/ao_sdl.c76
-rw-r--r--audio/out/ao_sndio.c42
-rwxr-xr-xaudio/out/ao_wasapi_utils.c5
10 files changed, 134 insertions, 160 deletions
diff --git a/audio/out/ao_alsa.c b/audio/out/ao_alsa.c
index 60b8338f9d..c344f6f20e 100644
--- a/audio/out/ao_alsa.c
+++ b/audio/out/ao_alsa.c
@@ -38,6 +38,7 @@
#include "options/options.h"
#include "options/m_option.h"
#include "common/msg.h"
+#include "osdep/endian.h"
#define ALSA_PCM_NEW_HW_PARAMS_API
#define ALSA_PCM_NEW_SW_PARAMS_API
@@ -208,22 +209,17 @@ alsa_error:
static const int mp_to_alsa_format[][2] = {
{AF_FORMAT_S8, SND_PCM_FORMAT_S8},
{AF_FORMAT_U8, SND_PCM_FORMAT_U8},
- {AF_FORMAT_U16_LE, SND_PCM_FORMAT_U16_LE},
- {AF_FORMAT_U16_BE, SND_PCM_FORMAT_U16_BE},
- {AF_FORMAT_S16_LE, SND_PCM_FORMAT_S16_LE},
- {AF_FORMAT_S16_BE, SND_PCM_FORMAT_S16_BE},
- {AF_FORMAT_U32_LE, SND_PCM_FORMAT_U32_LE},
- {AF_FORMAT_U32_BE, SND_PCM_FORMAT_U32_BE},
- {AF_FORMAT_S32_LE, SND_PCM_FORMAT_S32_LE},
- {AF_FORMAT_S32_BE, SND_PCM_FORMAT_S32_BE},
- {AF_FORMAT_U24_LE, SND_PCM_FORMAT_U24_3LE},
- {AF_FORMAT_U24_BE, SND_PCM_FORMAT_U24_3BE},
- {AF_FORMAT_S24_LE, SND_PCM_FORMAT_S24_3LE},
- {AF_FORMAT_S24_BE, SND_PCM_FORMAT_S24_3BE},
- {AF_FORMAT_FLOAT_LE, SND_PCM_FORMAT_FLOAT_LE},
- {AF_FORMAT_FLOAT_BE, SND_PCM_FORMAT_FLOAT_BE},
- {AF_FORMAT_AC3, SND_PCM_FORMAT_S16_LE},
- {AF_FORMAT_IEC61937, SND_PCM_FORMAT_S16_LE},
+ {AF_FORMAT_U16, SND_PCM_FORMAT_U16},
+ {AF_FORMAT_S16, SND_PCM_FORMAT_S16},
+ {AF_FORMAT_U32, SND_PCM_FORMAT_U32},
+ {AF_FORMAT_S32, SND_PCM_FORMAT_S32},
+ {AF_FORMAT_U24,
+ MP_SELECT_LE_BE(SND_PCM_FORMAT_U24_3LE, SND_PCM_FORMAT_U24_3BE)},
+ {AF_FORMAT_S24,
+ MP_SELECT_LE_BE(SND_PCM_FORMAT_S24_3LE, SND_PCM_FORMAT_S24_3BE)},
+ {AF_FORMAT_FLOAT, SND_PCM_FORMAT_FLOAT},
+ {AF_FORMAT_AC3, SND_PCM_FORMAT_S16},
+ {AF_FORMAT_IEC61937, SND_PCM_FORMAT_S16},
{AF_FORMAT_MPEG2, SND_PCM_FORMAT_MPEG},
{AF_FORMAT_UNKNOWN, SND_PCM_FORMAT_UNKNOWN},
};
@@ -417,13 +413,13 @@ static int init(struct ao *ao)
if (err < 0) {
MP_INFO(ao, "Format %s is not supported by hardware, trying default.\n",
af_fmt_to_str(ao->format));
- p->alsa_fmt = SND_PCM_FORMAT_S16_LE;
+ p->alsa_fmt = SND_PCM_FORMAT_S16;
if (AF_FORMAT_IS_AC3(ao->format))
ao->format = AF_FORMAT_AC3;
else if (AF_FORMAT_IS_IEC61937(ao->format))
ao->format = AF_FORMAT_IEC61937;
else
- ao->format = AF_FORMAT_S16_LE;
+ ao->format = AF_FORMAT_S16;
}
err = snd_pcm_hw_params_set_format(p->alsa, alsa_hwparams, p->alsa_fmt);
diff --git a/audio/out/ao_coreaudio_utils.c b/audio/out/ao_coreaudio_utils.c
index da06656e9d..b39323ce72 100644
--- a/audio/out/ao_coreaudio_utils.c
+++ b/audio/out/ao_coreaudio_utils.c
@@ -25,6 +25,7 @@
#include "audio/out/ao_coreaudio_utils.h"
#include "audio/out/ao_coreaudio_properties.h"
#include "osdep/timer.h"
+#include "osdep/endian.h"
#include "audio/format.h"
void ca_print_device_list(struct ao *ao)
@@ -142,7 +143,7 @@ void ca_fill_asbd(struct ao *ao, AudioStreamBasicDescription *asbd)
if ((ao->format & AF_FORMAT_SIGN_MASK) == AF_FORMAT_SI)
asbd->mFormatFlags |= kAudioFormatFlagIsSignedInteger;
- if ((ao->format & AF_FORMAT_END_MASK) == AF_FORMAT_BE)
+ if (BYTE_ORDER == BIG_ENDIAN)
asbd->mFormatFlags |= kAudioFormatFlagIsBigEndian;
asbd->mFramesPerPacket = 1;
diff --git a/audio/out/ao_dsound.c b/audio/out/ao_dsound.c
index da1742e077..78b39b17b9 100644
--- a/audio/out/ao_dsound.c
+++ b/audio/out/ao_dsound.c
@@ -392,14 +392,14 @@ static int init(struct ao *ao)
}
switch (format) {
case AF_FORMAT_AC3:
- case AF_FORMAT_S24_LE:
- case AF_FORMAT_S16_LE:
+ case AF_FORMAT_S24:
+ case AF_FORMAT_S16:
case AF_FORMAT_U8:
break;
default:
MP_VERBOSE(ao, "format %s not supported defaulting to Signed 16-bit Little-Endian\n",
af_fmt_to_str(format));
- format = AF_FORMAT_S16_LE;
+ format = AF_FORMAT_S16;
}
//set our audio parameters
ao->samplerate = rate;
diff --git a/audio/out/ao_oss.c b/audio/out/ao_oss.c
index b78f244a7b..de71017432 100644
--- a/audio/out/ao_oss.c
+++ b/audio/out/ao_oss.c
@@ -40,6 +40,7 @@
#include "options/options.h"
#include "common/msg.h"
#include "osdep/timer.h"
+#include "osdep/endian.h"
#if HAVE_SYS_SOUNDCARD_H
#include <sys/soundcard.h>
@@ -89,39 +90,46 @@ static const struct mp_chmap oss_layouts[MP_NUM_CHANNELS + 1] = {
MP_CHMAP8(FL, FR, BL, BR, FC, LFE, SL, SR), // 7.1
};
-static const int format_table[][2] = {
- {AFMT_U8, AF_FORMAT_U8},
- {AFMT_S8, AF_FORMAT_S8},
- {AFMT_U16_LE, AF_FORMAT_U16_LE},
- {AFMT_U16_BE, AF_FORMAT_U16_BE},
- {AFMT_S16_LE, AF_FORMAT_S16_LE},
- {AFMT_S16_BE, AF_FORMAT_S16_BE},
-#ifdef AFMT_S24_PACKED
- {AFMT_S24_PACKED, AF_FORMAT_S24_LE},
+#if !defined(AFMT_S16_NE) && defined(AFMT_S16_LE) && defined(AFMT_S16_BE)
+#define AFMT_S16_NE MP_SELECT_LE_BE(AFMT_S16_LE, AFMT_S16_BE)
#endif
-#ifdef AFMT_U24_LE
- {AFMT_U24_LE, AF_FORMAT_U24_LE},
+
+#if !defined(AFMT_U16_NE) && defined(AFMT_U16_LE) && defined(AFMT_U16_BE)
+#define AFMT_U16_NE MP_SELECT_LE_BE(AFMT_U16_LE, AFMT_U16_BE)
#endif
-#ifdef AFMT_U24_BE
- {AFMT_U24_BE, AF_FORMAT_U24_BE},
+
+#if !defined(AFMT_U24_NE) && defined(AFMT_U24_LE) && defined(AFMT_U24_BE)
+#define AFMT_U24_NE MP_SELECT_LE_BE(AFMT_U24_LE, AFMT_U24_BE)
#endif
-#ifdef AFMT_S24_LE
- {AFMT_S24_LE, AF_FORMAT_S24_LE},
+
+#if !defined(AFMT_S24_NE) && defined(AFMT_S24_LE) && defined(AFMT_S24_BE)
+#define AFMT_S24_NE MP_SELECT_LE_BE(AFMT_S24_LE, AFMT_S24_BE)
#endif
-#ifdef AFMT_S24_BE
- {AFMT_S24_BE, AF_FORMAT_S24_BE},
+
+#if !defined(AFMT_U32_NE) && defined(AFMT_U32_LE) && defined(AFMT_U32_BE)
+#define AFMT_U32MP_SELECT_LE_BE(AFMT_U32_LE, AFMT_U32_BE)
#endif
-#ifdef AFMT_U32_LE
- {AFMT_U32_LE, AF_FORMAT_U32_LE},
+
+#if !defined(AFMT_S32_NE) && defined(AFMT_S32_LE) && defined(AFMT_S32_BE)
+#define AFMT_S32MP_SELECT_LE_BE(AFMT_S32_LE, AFMT_S32_BE)
+#endif
+
+static const int format_table[][2] = {
+ {AFMT_U8, AF_FORMAT_U8},
+ {AFMT_S8, AF_FORMAT_S8},
+ {AFMT_U16_NE, AF_FORMAT_U16},
+ {AFMT_S16_NE, AF_FORMAT_S16},
+#ifdef AFMT_U24_NE
+ {AFMT_U24_NE, AF_FORMAT_U24},
#endif
-#ifdef AFMT_U32_BE
- {AFMT_U32_BE, AF_FORMAT_U32_BE},
+#ifdef AFMT_S24_NE
+ {AFMT_S24_NE, AF_FORMAT_S24},
#endif
-#ifdef AFMT_S32_LE
- {AFMT_S32_LE, AF_FORMAT_S32_LE},
+#ifdef AFMT_U32_NE
+ {AFMT_U32_NE, AF_FORMAT_U32},
#endif
-#ifdef AFMT_S32_BE
- {AFMT_S32_BE, AF_FORMAT_S32_BE},
+#ifdef AFMT_S32_NE
+ {AFMT_S32_NE, AF_FORMAT_S32},
#endif
#ifdef AFMT_FLOAT
{AFMT_FLOAT, AF_FORMAT_FLOAT},
diff --git a/audio/out/ao_pcm.c b/audio/out/ao_pcm.c
index 3c1f46409a..eb089c6c42 100644
--- a/audio/out/ao_pcm.c
+++ b/audio/out/ao_pcm.c
@@ -35,6 +35,7 @@
#include "ao.h"
#include "internal.h"
#include "common/msg.h"
+#include "osdep/endian.h"
#ifdef __MINGW32__
// for GetFileType to detect pipes
@@ -72,8 +73,7 @@ static void fput32le(uint32_t val, FILE *fp)
static void write_wave_header(struct ao *ao, FILE *fp, uint64_t data_length)
{
bool use_waveex = true;
- uint16_t fmt = ao->format == AF_FORMAT_FLOAT_LE ?
- WAV_ID_FLOAT_PCM : WAV_ID_PCM;
+ uint16_t fmt = ao->format == AF_FORMAT_FLOAT ? WAV_ID_FLOAT_PCM : WAV_ID_PCM;
uint32_t fmt_chunk_size = use_waveex ? 40 : 16;
int bits = af_fmt2bits(ao->format);
@@ -124,16 +124,23 @@ static int init(struct ao *ao)
if (priv->waveheader) {
// WAV files must have one of the following formats
+ // And they don't work in big endian; fixing it would be simple, but
+ // nobody cares.
+ if (BYTE_ORDER == BIG_ENDIAN) {
+ MP_FATAL(ao, "Not supported on big endian.\n");
+ return -1;
+ }
+
switch (ao->format) {
case AF_FORMAT_U8:
- case AF_FORMAT_S16_LE:
- case AF_FORMAT_S24_LE:
- case AF_FORMAT_S32_LE:
- case AF_FORMAT_FLOAT_LE:
+ case AF_FORMAT_S16:
+ case AF_FORMAT_S24:
+ case AF_FORMAT_S32:
+ case AF_FORMAT_FLOAT:
case AF_FORMAT_AC3:
break;
default:
- ao->format = AF_FORMAT_S16_LE;
+ ao->format = AF_FORMAT_S16;
break;
}
}
diff --git a/audio/out/ao_pulse.c b/audio/out/ao_pulse.c
index 790ecb23dc..fa8a6a46be 100644
--- a/audio/out/ao_pulse.c
+++ b/audio/out/ao_pulse.c
@@ -169,12 +169,9 @@ static const struct format_map {
int mp_format;
pa_sample_format_t pa_format;
} format_maps[] = {
- {AF_FORMAT_S16_LE, PA_SAMPLE_S16LE},
- {AF_FORMAT_S16_BE, PA_SAMPLE_S16BE},
- {AF_FORMAT_S32_LE, PA_SAMPLE_S32LE},
- {AF_FORMAT_S32_BE, PA_SAMPLE_S32BE},
- {AF_FORMAT_FLOAT_LE, PA_SAMPLE_FLOAT32LE},
- {AF_FORMAT_FLOAT_BE, PA_SAMPLE_FLOAT32BE},
+ {AF_FORMAT_S16, PA_SAMPLE_S16NE},
+ {AF_FORMAT_S32, PA_SAMPLE_S32NE},
+ {AF_FORMAT_FLOAT, PA_SAMPLE_FLOAT32NE},
{AF_FORMAT_U8, PA_SAMPLE_U8},
{AF_FORMAT_UNKNOWN, 0}
};
diff --git a/audio/out/ao_rsound.c b/audio/out/ao_rsound.c
index e41a89ae37..fe187144a6 100644
--- a/audio/out/ao_rsound.c
+++ b/audio/out/ao_rsound.c
@@ -52,40 +52,26 @@ static int set_format(struct ao *ao)
case AF_FORMAT_S8:
rsd_format = RSD_S8;
break;
- case AF_FORMAT_S16_LE:
- rsd_format = RSD_S16_LE;
+ case AF_FORMAT_S16:
+ rsd_format = RSD_S16_NE;
break;
- case AF_FORMAT_S16_BE:
- rsd_format = RSD_S16_BE;
+ case AF_FORMAT_U16:
+ rsd_format = RSD_U16_NE;
break;
- case AF_FORMAT_U16_LE:
- rsd_format = RSD_U16_LE;
+ case AF_FORMAT_S24:
+ case AF_FORMAT_U24:
+ rsd_format = RSD_S32_NE;
+ ao->format = AF_FORMAT_S32;
break;
- case AF_FORMAT_U16_BE:
- rsd_format = RSD_U16_BE;
+ case AF_FORMAT_S32:
+ rsd_format = RSD_S32_NE;
break;
- case AF_FORMAT_S24_LE:
- case AF_FORMAT_S24_BE:
- case AF_FORMAT_U24_LE:
- case AF_FORMAT_U24_BE:
- rsd_format = RSD_S32_LE;
- ao->format = AF_FORMAT_S32_LE;
- break;
- case AF_FORMAT_S32_LE:
- rsd_format = RSD_S32_LE;
- break;
- case AF_FORMAT_S32_BE:
- rsd_format = RSD_S32_BE;
- break;
- case AF_FORMAT_U32_LE:
- rsd_format = RSD_U32_LE;
- break;
- case AF_FORMAT_U32_BE:
- rsd_format = RSD_U32_BE;
+ case AF_FORMAT_U32:
+ rsd_format = RSD_U32_NE;
break;
default:
- rsd_format = RSD_S16_LE;
- ao->format = AF_FORMAT_S16_LE;
+ rsd_format = RSD_S16_NE;
+ ao->format = AF_FORMAT_S16;
}
return rsd_format;
diff --git a/audio/out/ao_sdl.c b/audio/out/ao_sdl.c
index 9af5db4021..d2362a723e 100644
--- a/audio/out/ao_sdl.c
+++ b/audio/out/ao_sdl.c
@@ -38,6 +38,20 @@ struct priv
float buflen;
};
+static int fmtmap[][2] = {
+ {AF_FORMAT_U8, AUDIO_U8},
+ {AF_FORMAT_S8, AUDIO_S8},
+ {AF_FORMAT_U16, AUDIO_U16SYS},
+ {AF_FORMAT_S16, AUDIO_S16SYS},
+#ifdef AUDIO_S32SYS
+ {AF_FORMAT_S32, AUDIO_S32SYS},
+#endif
+#ifdef AUDIO_F32SYS
+ {AF_FORMAT_FLOAT, AUDIO_F32SYS},
+#endif
+ {0}
+};
+
static void audio_callback(void *userdata, Uint8 *stream, int len)
{
struct ao *ao = userdata;
@@ -104,26 +118,12 @@ static int init(struct ao *ao)
SDL_AudioSpec desired, obtained;
- switch (ao->format) {
- case AF_FORMAT_U8: desired.format = AUDIO_U8; break;
- case AF_FORMAT_S8: desired.format = AUDIO_S8; break;
- case AF_FORMAT_U16_LE: desired.format = AUDIO_U16LSB; break;
- case AF_FORMAT_U16_BE: desired.format = AUDIO_U16MSB; break;
- default:
- case AF_FORMAT_S16_LE: desired.format = AUDIO_S16LSB; break;
- case AF_FORMAT_S16_BE: desired.format = AUDIO_S16MSB; break;
-#ifdef AUDIO_S32LSB
- case AF_FORMAT_S32_LE: desired.format = AUDIO_S32LSB; break;
-#endif
-#ifdef AUDIO_S32MSB
- case AF_FORMAT_S32_BE: desired.format = AUDIO_S32MSB; break;
-#endif
-#ifdef AUDIO_F32LSB
- case AF_FORMAT_FLOAT_LE: desired.format = AUDIO_F32LSB; break;
-#endif
-#ifdef AUDIO_F32MSB
- case AF_FORMAT_FLOAT_BE: desired.format = AUDIO_F32MSB; break;
-#endif
+ desired.format = AUDIO_S16SYS;
+ for (int n = 0; fmtmap[n][0]; n++) {
+ if (ao->format == fmtmap[n][0]) {
+ desired.format = fmtmap[n][1];
+ break;
+ }
}
desired.freq = ao->samplerate;
desired.channels = ao->channels.num;
@@ -156,30 +156,18 @@ static int init(struct ao *ao)
// large, this will help.
ao->device_buffer = 3 * obtained.samples;
- switch (obtained.format) {
- case AUDIO_U8: ao->format = AF_FORMAT_U8; break;
- case AUDIO_S8: ao->format = AF_FORMAT_S8; break;
- case AUDIO_S16LSB: ao->format = AF_FORMAT_S16_LE; break;
- case AUDIO_S16MSB: ao->format = AF_FORMAT_S16_BE; break;
- case AUDIO_U16LSB: ao->format = AF_FORMAT_U16_LE; break;
- case AUDIO_U16MSB: ao->format = AF_FORMAT_U16_BE; break;
-#ifdef AUDIO_S32LSB
- case AUDIO_S32LSB: ao->format = AF_FORMAT_S32_LE; break;
-#endif
-#ifdef AUDIO_S32MSB
- case AUDIO_S32MSB: ao->format = AF_FORMAT_S32_BE; break;
-#endif
-#ifdef AUDIO_F32LSB
- case AUDIO_F32LSB: ao->format = AF_FORMAT_FLOAT_LE; break;
-#endif
-#ifdef AUDIO_F32MSB
- case AUDIO_F32MSB: ao->format = AF_FORMAT_FLOAT_BE; break;
-#endif
- default:
- if (!ao->probing)
- MP_ERR(ao, "could not find matching format\n");
- uninit(ao);
- return -1;
+ ao->format = 0;
+ for (int n = 0; fmtmap[n][0]; n++) {
+ if (obtained.format == fmtmap[n][1]) {
+ ao->format = fmtmap[n][0];
+ break;
+ }
+ }
+ if (!ao->format) {
+ if (!ao->probing)
+ MP_ERR(ao, "could not find matching format\n");
+ uninit(ao);
+ return -1;
}
if (!ao_chmap_sel_get_def(ao, &sel, &ao->channels, obtained.channels)) {
diff --git a/audio/out/ao_sndio.c b/audio/out/ao_sndio.c
index c75027dffc..c1c4ef17ab 100644
--- a/audio/out/ao_sndio.c
+++ b/audio/out/ao_sndio.c
@@ -109,22 +109,16 @@ static int init(struct ao *ao)
struct priv *p = ao->priv;
struct af_to_par {
- int format, bits, sig, le;
+ int format, bits, sig;
} static const af_to_par[] = {
- {AF_FORMAT_U8, 8, 0, 0},
- {AF_FORMAT_S8, 8, 1, 0},
- {AF_FORMAT_U16_LE, 16, 0, 1},
- {AF_FORMAT_U16_BE, 16, 0, 0},
- {AF_FORMAT_S16_LE, 16, 1, 1},
- {AF_FORMAT_S16_BE, 16, 1, 0},
- {AF_FORMAT_U24_LE, 16, 0, 1},
- {AF_FORMAT_U24_BE, 24, 0, 0},
- {AF_FORMAT_S24_LE, 24, 1, 1},
- {AF_FORMAT_S24_BE, 24, 1, 0},
- {AF_FORMAT_U32_LE, 32, 0, 1},
- {AF_FORMAT_U32_BE, 32, 0, 0},
- {AF_FORMAT_S32_LE, 32, 1, 1},
- {AF_FORMAT_S32_BE, 32, 1, 0}
+ {AF_FORMAT_U8, 8, 0},
+ {AF_FORMAT_S8, 8, 1},
+ {AF_FORMAT_U16, 16, 0},
+ {AF_FORMAT_S16, 16, 1},
+ {AF_FORMAT_U24, 16, 0},
+ {AF_FORMAT_S24, 24, 1},
+ {AF_FORMAT_U32, 32, 0},
+ {AF_FORMAT_S32, 32, 1},
}, *ap;
int i;
@@ -149,7 +143,7 @@ static int init(struct ao *ao)
p->par.bits = ap->bits;
p->par.sig = ap->sig;
if (ap->bits > 8)
- p->par.le = ap->le;
+ p->par.le = SIO_LE_NATIVE;
if (ap->bits != SIO_BPS(ap->bits))
p->par.bps = ap->bits / 8;
break;
@@ -175,20 +169,18 @@ static int init(struct ao *ao)
MP_ERR(ao, "couldn't get params\n");
goto error;
}
+ if (p->par.bps > 1 && p->par.le != SIO_LE_NATIVE) {
+ MP_ERR(ao, "swapped endian output not supported\n");
+ goto error;
+ }
if (p->par.bits == 8 && p->par.bps == 1) {
ao->format = p->par.sig ? AF_FORMAT_S8 : AF_FORMAT_U8;
} else if (p->par.bits == 16 && p->par.bps == 2) {
- ao->format = p->par.sig ?
- (p->par.le ? AF_FORMAT_S16_LE : AF_FORMAT_S16_BE) :
- (p->par.le ? AF_FORMAT_U16_LE : AF_FORMAT_U16_BE);
+ ao->format = p->par.sig ? AF_FORMAT_S16 : AF_FORMAT_U16;
} else if ((p->par.bits == 24 || p->par.msb) && p->par.bps == 3) {
- ao->format = p->par.sig ?
- (p->par.le ? AF_FORMAT_S24_LE : AF_FORMAT_S24_BE) :
- (p->par.le ? AF_FORMAT_U24_LE : AF_FORMAT_U24_BE);
+ ao->format = p->par.sig ? AF_FORMAT_S24 : AF_FORMAT_U24;
} else if ((p->par.bits == 32 || p->par.msb) && p->par.bps == 4) {
- ao->format = p->par.sig ?
- (p->par.le ? AF_FORMAT_S32_LE : AF_FORMAT_S32_BE) :
- (p->par.le ? AF_FORMAT_U32_LE : AF_FORMAT_U32_BE);
+ ao->format = p->par.sig ? AF_FORMAT_S32 : AF_FORMAT_U32;
} else {
MP_ERR(ao, "couldn't set format\n");
goto error;
diff --git a/audio/out/ao_wasapi_utils.c b/audio/out/ao_wasapi_utils.c
index afa2ad6b40..c12f3baf61 100755
--- a/audio/out/ao_wasapi_utils.c
+++ b/audio/out/ao_wasapi_utils.c
@@ -292,14 +292,13 @@ static int try_passthrough(struct wasapi_state *state,
union WAVEFMT u;
u.extensible = &wformat;
- MP_VERBOSE(ao, "trying passthrough for %s...\n",
- af_fmt_to_str((ao->format&~AF_FORMAT_END_MASK) | AF_FORMAT_LE));
+ MP_VERBOSE(ao, "trying passthrough for %s...\n", af_fmt_to_str(ao->format));
HRESULT hr = IAudioClient_IsFormatSupported(state->pAudioClient,
state->share_mode,
u.ex, NULL);
if (!FAILED(hr)) {
- ao->format = (ao->format&~AF_FORMAT_END_MASK) | AF_FORMAT_LE;
+ ao->format = ao->format;
state->format = wformat;
return 1;
}
generated by cgit v1.2.3 (git 2.25.1) at 2024-04-27 11:59:20 +0000