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authorbellard <bellard@c046a42c-6fe2-441c-8c8c-71466251a162>2006-07-04 21:47:22 +0000
committerbellard <bellard@c046a42c-6fe2-441c-8c8c-71466251a162>2006-07-04 21:47:22 +0000
commitd929eba5d47f097302779d55427712c3ceb931ad (patch)
tree14ca7172d2abe2d446f96b885464b17044705d3e
parent219fb125039e175a92aa14684ac688305b5143bd (diff)
audio endianness API changes (malc)
git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@2042 c046a42c-6fe2-441c-8c8c-71466251a162
-rw-r--r--audio/alsaaudio.c14
-rw-r--r--audio/audio.c46
-rw-r--r--audio/audio.h15
-rw-r--r--audio/audio_int.h14
-rw-r--r--audio/audio_template.h21
-rw-r--r--audio/coreaudio.c10
-rw-r--r--audio/dsound_template.h4
-rw-r--r--audio/fmodaudio.c8
-rw-r--r--audio/noaudio.c4
-rw-r--r--audio/ossaudio.c14
-rw-r--r--audio/sdlaudio.c7
-rw-r--r--audio/wavaudio.c3
-rw-r--r--audio/wavcapture.c3
-rw-r--r--hw/adlib.c4
-rw-r--r--hw/es1370.c7
-rw-r--r--hw/pcspk.c4
-rw-r--r--hw/sb16.c16
17 files changed, 85 insertions, 109 deletions
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 2cac396b26..71e5235664 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -662,12 +662,9 @@ static int alsa_init_out (HWVoiceOut *hw, audsettings_t *as)
obt_as.freq = obt.freq;
obt_as.nchannels = obt.nchannels;
obt_as.fmt = effective_fmt;
+ obt_as.endianness = endianness;
- audio_pcm_init_info (
- &hw->info,
- &obt_as,
- audio_need_to_swap_endian (endianness)
- );
+ audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = obt.samples;
alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
@@ -751,12 +748,9 @@ static int alsa_init_in (HWVoiceIn *hw, audsettings_t *as)
obt_as.freq = obt.freq;
obt_as.nchannels = obt.nchannels;
obt_as.fmt = effective_fmt;
+ obt_as.endianness = endianness;
- audio_pcm_init_info (
- &hw->info,
- &obt_as,
- audio_need_to_swap_endian (endianness)
- );
+ audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = obt.samples;
alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
diff --git a/audio/audio.c b/audio/audio.c
index 0de728cc5f..dd86c4d7a7 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -510,6 +510,18 @@ static void audio_print_settings (audsettings_t *as)
AUD_log (NULL, "invalid(%d)", as->fmt);
break;
}
+ AUD_log (NULL, "endianness=");
+ switch (as->endianness) {
+ case 0:
+ AUD_log (NULL, "little");
+ break;
+ case 1:
+ AUD_log (NULL, "big");
+ break;
+ default:
+ AUD_log (NULL, "invalid");
+ break;
+ }
AUD_log (NULL, "\n");
}
@@ -518,6 +530,7 @@ static int audio_validate_settigs (audsettings_t *as)
int invalid;
invalid = as->nchannels != 1 && as->nchannels != 2;
+ invalid |= as->endianness != 0 && as->endianness != 1;
switch (as->fmt) {
case AUD_FMT_S8:
@@ -531,11 +544,7 @@ static int audio_validate_settigs (audsettings_t *as)
}
invalid |= as->freq <= 0;
-
- if (invalid) {
- return -1;
- }
- return 0;
+ return invalid ? -1 : 0;
}
static int audio_pcm_info_eq (struct audio_pcm_info *info, audsettings_t *as)
@@ -557,14 +566,11 @@ static int audio_pcm_info_eq (struct audio_pcm_info *info, audsettings_t *as)
return info->freq == as->freq
&& info->nchannels == as->nchannels
&& info->sign == sign
- && info->bits == bits;
+ && info->bits == bits
+ && info->swap_endianness == (as->endianness != AUDIO_HOST_ENDIANNESS);
}
-void audio_pcm_init_info (
- struct audio_pcm_info *info,
- audsettings_t *as,
- int swap_endian
- )
+void audio_pcm_init_info (struct audio_pcm_info *info, audsettings_t *as)
{
int bits = 8, sign = 0;
@@ -588,7 +594,7 @@ void audio_pcm_init_info (
info->shift = (as->nchannels == 2) + (bits == 16);
info->align = (1 << info->shift) - 1;
info->bytes_per_second = info->freq << info->shift;
- info->swap_endian = swap_endian;
+ info->swap_endianness = (as->endianness != AUDIO_HOST_ENDIANNESS);
}
void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
@@ -610,7 +616,7 @@ void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
int shift = info->nchannels - 1;
short s = INT16_MAX;
- if (info->swap_endian) {
+ if (info->swap_endianness) {
s = bswap16 (s);
}
@@ -635,16 +641,13 @@ static void noop_conv (st_sample_t *dst, const void *src,
static CaptureVoiceOut *audio_pcm_capture_find_specific (
AudioState *s,
- audsettings_t *as,
- int endian
+ audsettings_t *as
)
{
CaptureVoiceOut *cap;
- int swap_endian = audio_need_to_swap_endian (endian);
for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
- if ((cap->hw.info.swap_endian == swap_endian)
- && audio_pcm_info_eq (&cap->hw.info, as)) {
+ if (audio_pcm_info_eq (&cap->hw.info, as)) {
return cap;
}
}
@@ -1697,7 +1700,6 @@ AudioState *AUD_init (void)
int AUD_add_capture (
AudioState *s,
audsettings_t *as,
- int endian,
struct audio_capture_ops *ops,
void *cb_opaque
)
@@ -1725,7 +1727,7 @@ int AUD_add_capture (
cb->ops = *ops;
cb->opaque = cb_opaque;
- cap = audio_pcm_capture_find_specific (s, as, endian);
+ cap = audio_pcm_capture_find_specific (s, as);
if (cap) {
LIST_INSERT_HEAD (&cap->cb_head, cb, entries);
return 0;
@@ -1755,7 +1757,7 @@ int AUD_add_capture (
goto err2;
}
- audio_pcm_init_info (&hw->info, as, endian);
+ audio_pcm_init_info (&hw->info, as);
cap->buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
if (!cap->buf) {
@@ -1768,7 +1770,7 @@ int AUD_add_capture (
hw->clip = mixeng_clip
[hw->info.nchannels == 2]
[hw->info.sign]
- [hw->info.swap_endian]
+ [hw->info.swap_endianness]
[hw->info.bits == 16];
LIST_INSERT_HEAD (&s->cap_head, cap, entries);
diff --git a/audio/audio.h b/audio/audio.h
index 4e1a694d07..14fa3bce96 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -24,6 +24,7 @@
#ifndef QEMU_AUDIO_H
#define QEMU_AUDIO_H
+#include "config.h"
#include "sys-queue.h"
typedef void (*audio_callback_fn_t) (void *opaque, int avail);
@@ -35,10 +36,17 @@ typedef enum {
AUD_FMT_S16
} audfmt_e;
+#ifdef WORDS_BIGENDIAN
+#define AUDIO_HOST_ENDIANNESS 1
+#else
+#define AUDIO_HOST_ENDIANNESS 0
+#endif
+
typedef struct {
int freq;
int nchannels;
audfmt_e fmt;
+ int endianness;
} audsettings_t;
struct audio_capture_ops {
@@ -74,7 +82,6 @@ void AUD_remove_card (QEMUSoundCard *card);
int AUD_add_capture (
AudioState *s,
audsettings_t *as,
- int endian,
struct audio_capture_ops *ops,
void *opaque
);
@@ -85,8 +92,7 @@ SWVoiceOut *AUD_open_out (
const char *name,
void *callback_opaque,
audio_callback_fn_t callback_fn,
- audsettings_t *settings,
- int sw_endian
+ audsettings_t *settings
);
void AUD_close_out (QEMUSoundCard *card, SWVoiceOut *sw);
@@ -104,8 +110,7 @@ SWVoiceIn *AUD_open_in (
const char *name,
void *callback_opaque,
audio_callback_fn_t callback_fn,
- audsettings_t *settings,
- int sw_endian
+ audsettings_t *settings
);
void AUD_close_in (QEMUSoundCard *card, SWVoiceIn *sw);
diff --git a/audio/audio_int.h b/audio/audio_int.h
index c01c16add6..f5dcb2c705 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -61,7 +61,7 @@ struct audio_pcm_info {
int align;
int shift;
int bytes_per_second;
- int swap_endian;
+ int swap_endianness;
};
typedef struct HWVoiceOut {
@@ -198,8 +198,7 @@ extern struct audio_driver coreaudio_audio_driver;
extern struct audio_driver dsound_audio_driver;
extern volume_t nominal_volume;
-void audio_pcm_init_info (struct audio_pcm_info *info, audsettings_t *as,
- int swap_endian);
+void audio_pcm_init_info (struct audio_pcm_info *info, audsettings_t *as);
void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len);
int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int len);
@@ -220,15 +219,6 @@ static inline int audio_ring_dist (int dst, int src, int len)
return (dst >= src) ? (dst - src) : (len - src + dst);
}
-static inline int audio_need_to_swap_endian (int endianness)
-{
-#ifdef WORDS_BIGENDIAN
- return endianness != 1;
-#else
- return endianness != 0;
-#endif
-}
-
#if defined __GNUC__
#define GCC_ATTR __attribute__ ((__unused__, __format__ (__printf__, 1, 2)))
#define INIT_FIELD(f) . f
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 04b30239db..04b47befee 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -140,13 +140,12 @@ static int glue (audio_pcm_sw_init_, TYPE) (
SW *sw,
HW *hw,
const char *name,
- audsettings_t *as,
- int endian
+ audsettings_t *as
)
{
int err;
- audio_pcm_init_info (&sw->info, as, audio_need_to_swap_endian (endian));
+ audio_pcm_init_info (&sw->info, as);
sw->hw = hw;
sw->active = 0;
#ifdef DAC
@@ -164,7 +163,7 @@ static int glue (audio_pcm_sw_init_, TYPE) (
#endif
[sw->info.nchannels == 2]
[sw->info.sign]
- [sw->info.swap_endian]
+ [sw->info.swap_endianness]
[sw->info.bits == 16];
sw->name = qemu_strdup (name);
@@ -288,7 +287,7 @@ static HW *glue (audio_pcm_hw_add_new_, TYPE) (AudioState *s, audsettings_t *as)
#endif
[hw->info.nchannels == 2]
[hw->info.sign]
- [hw->info.swap_endian]
+ [hw->info.swap_endianness]
[hw->info.bits == 16];
if (glue (audio_pcm_hw_alloc_resources_, TYPE) (hw)) {
@@ -336,8 +335,7 @@ static HW *glue (audio_pcm_hw_add_, TYPE) (AudioState *s, audsettings_t *as)
static SW *glue (audio_pcm_create_voice_pair_, TYPE) (
AudioState *s,
const char *sw_name,
- audsettings_t *as,
- int sw_endian
+ audsettings_t *as
)
{
SW *sw;
@@ -365,7 +363,7 @@ static SW *glue (audio_pcm_create_voice_pair_, TYPE) (
glue (audio_pcm_hw_add_sw_, TYPE) (hw, sw);
- if (glue (audio_pcm_sw_init_, TYPE) (sw, hw, sw_name, as, sw_endian)) {
+ if (glue (audio_pcm_sw_init_, TYPE) (sw, hw, sw_name, as)) {
goto err3;
}
@@ -407,8 +405,7 @@ SW *glue (AUD_open_, TYPE) (
const char *name,
void *callback_opaque ,
audio_callback_fn_t callback_fn,
- audsettings_t *as,
- int sw_endian
+ audsettings_t *as
)
{
AudioState *s;
@@ -481,12 +478,12 @@ SW *glue (AUD_open_, TYPE) (
}
glue (audio_pcm_sw_fini_, TYPE) (sw);
- if (glue (audio_pcm_sw_init_, TYPE) (sw, hw, name, as, sw_endian)) {
+ if (glue (audio_pcm_sw_init_, TYPE) (sw, hw, name, as)) {
goto fail;
}
}
else {
- sw = glue (audio_pcm_create_voice_pair_, TYPE) (s, name, as, sw_endian);
+ sw = glue (audio_pcm_create_voice_pair_, TYPE) (s, name, as);
if (!sw) {
dolog ("Failed to create voice `%s'\n", name);
return NULL;
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index 34e416d93a..8512f122b0 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -295,7 +295,6 @@ static int coreaudio_init_out (HWVoiceOut *hw, audsettings_t *as)
UInt32 propertySize;
int err;
int bits = 8;
- int endianess = 0;
const char *typ = "playback";
AudioValueRange frameRange;
@@ -308,16 +307,9 @@ static int coreaudio_init_out (HWVoiceOut *hw, audsettings_t *as)
if (as->fmt == AUD_FMT_S16 || as->fmt == AUD_FMT_U16) {
bits = 16;
- endianess = 1;
}
- audio_pcm_init_info (
- &hw->info,
- as,
- /* Following is irrelevant actually since we do not use
- mixengs clipping routines */
- audio_need_to_swap_endian (endianess)
- );
+ audio_pcm_init_info (&hw->info, as);
/* open default output device */
propertySize = sizeof(core->outputDeviceID);
diff --git a/audio/dsound_template.h b/audio/dsound_template.h
index 96f7cc7fa1..0896b04a03 100644
--- a/audio/dsound_template.h
+++ b/audio/dsound_template.h
@@ -250,8 +250,8 @@ static int dsound_init_out (HWVoiceOut *hw, audsettings_t *as)
}
ds->first_time = 1;
-
- audio_pcm_init_info (&hw->info, &obt_as, audio_need_to_swap_endian (0));
+ obt_as.endianness = 0;
+ audio_pcm_init_info (&hw->info, &obt_as);
if (bc.dwBufferBytes & hw->info.align) {
dolog (
diff --git a/audio/fmodaudio.c b/audio/fmodaudio.c
index 23f267753f..5875ba15e1 100644
--- a/audio/fmodaudio.c
+++ b/audio/fmodaudio.c
@@ -358,6 +358,7 @@ static int fmod_init_out (HWVoiceOut *hw, audsettings_t *as)
{
int bits16, mode, channel;
FMODVoiceOut *fmd = (FMODVoiceOut *) hw;
+ audsettings_t obt_as = *as;
mode = aud_to_fmodfmt (as->fmt, as->nchannels == 2 ? 1 : 0);
fmd->fmod_sample = FSOUND_Sample_Alloc (
@@ -384,7 +385,8 @@ static int fmod_init_out (HWVoiceOut *hw, audsettings_t *as)
fmd->channel = channel;
/* FMOD always operates on little endian frames? */
- audio_pcm_init_info (&hw->info, as, audio_need_to_swap_endian (0));
+ obt_as.endianness = 0;
+ audio_pcm_init_info (&hw->info, &obt_as);
bits16 = (mode & FSOUND_16BITS) != 0;
hw->samples = conf.nb_samples;
return 0;
@@ -418,6 +420,7 @@ static int fmod_init_in (HWVoiceIn *hw, audsettings_t *as)
{
int bits16, mode;
FMODVoiceIn *fmd = (FMODVoiceIn *) hw;
+ audsettings_t obt_as = *as;
if (conf.broken_adc) {
return -1;
@@ -440,7 +443,8 @@ static int fmod_init_in (HWVoiceIn *hw, audsettings_t *as)
}
/* FMOD always operates on little endian frames? */
- audio_pcm_init_info (&hw->info, as, audio_need_to_swap_endian (0));
+ obt_as.endianness = 0;
+ audio_pcm_init_info (&hw->info, &obt_as);
bits16 = (mode & FSOUND_16BITS) != 0;
hw->samples = conf.nb_samples;
return 0;
diff --git a/audio/noaudio.c b/audio/noaudio.c
index 314f6177ad..8fb15a224d 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -68,7 +68,7 @@ static int no_write (SWVoiceOut *sw, void *buf, int len)
static int no_init_out (HWVoiceOut *hw, audsettings_t *as)
{
- audio_pcm_init_info (&hw->info, as, 0);
+ audio_pcm_init_info (&hw->info, as);
hw->samples = 1024;
return 0;
}
@@ -87,7 +87,7 @@ static int no_ctl_out (HWVoiceOut *hw, int cmd, ...)
static int no_init_in (HWVoiceIn *hw, audsettings_t *as)
{
- audio_pcm_init_info (&hw->info, as, 0);
+ audio_pcm_init_info (&hw->info, as);
hw->samples = 1024;
return 0;
}
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 0bdc8eaab1..125e4c8ff7 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -453,12 +453,9 @@ static int oss_init_out (HWVoiceOut *hw, audsettings_t *as)
obt_as.freq = obt.freq;
obt_as.nchannels = obt.nchannels;
obt_as.fmt = effective_fmt;
+ obt_as.endianness = endianness;
- audio_pcm_init_info (
- &hw->info,
- &obt_as,
- audio_need_to_swap_endian (endianness)
- );
+ audio_pcm_init_info (&hw->info, &obt_as);
oss->nfrags = obt.nfrags;
oss->fragsize = obt.fragsize;
@@ -597,12 +594,9 @@ static int oss_init_in (HWVoiceIn *hw, audsettings_t *as)
obt_as.freq = obt.freq;
obt_as.nchannels = obt.nchannels;
obt_as.fmt = effective_fmt;
+ obt_as.endianness = endianness;
- audio_pcm_init_info (
- &hw->info,
- &obt_as,
- audio_need_to_swap_endian (endianness)
- );
+ audio_pcm_init_info (&hw->info, &obt_as);
oss->nfrags = obt.nfrags;
oss->fragsize = obt.fragsize;
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index 9fe212833f..f2a6896a52 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -335,12 +335,9 @@ static int sdl_init_out (HWVoiceOut *hw, audsettings_t *as)
obt_as.freq = obt.freq;
obt_as.nchannels = obt.channels;
obt_as.fmt = effective_fmt;
+ obt_as.endianness = endianess;
- audio_pcm_init_info (
- &hw->info,
- &obt_as,
- audio_need_to_swap_endian (endianess)
- );
+ audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = obt.samples;
s->initialized = 1;
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index ca1e99f4ce..c359fc4f2b 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -135,7 +135,8 @@ static int wav_init_out (HWVoiceOut *hw, audsettings_t *as)
hdr[34] = bits16 ? 0x10 : 0x08;
- audio_pcm_init_info (&hw->info, &wav_as, audio_need_to_swap_endian (0));
+ wav_as.endianness = 0;
+ audio_pcm_init_info (&hw->info, &wav_as);
hw->samples = 1024;
wav->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
diff --git a/audio/wavcapture.c b/audio/wavcapture.c
index 33f04c50b3..7458a5e3e6 100644
--- a/audio/wavcapture.c
+++ b/audio/wavcapture.c
@@ -70,6 +70,7 @@ void wav_capture (const char *path, int freq, int bits16, int stereo)
as.freq = freq;
as.nchannels = 1 << stereo;
as.fmt = bits16 ? AUD_FMT_S16 : AUD_FMT_U8;
+ as.endianness = 0;
ops.state = wav_state_cb;
ops.capture = wav_capture_cb;
@@ -97,5 +98,5 @@ void wav_capture (const char *path, int freq, int bits16, int stereo)
}
qemu_put_buffer (wav->f, hdr, sizeof (hdr));
- AUD_add_capture (NULL, &as, 0, &ops, wav);
+ AUD_add_capture (NULL, &as, &ops, wav);
}
diff --git a/hw/adlib.c b/hw/adlib.c
index f482d1fa84..b47bc3eece 100644
--- a/hw/adlib.c
+++ b/hw/adlib.c
@@ -301,6 +301,7 @@ int Adlib_init (AudioState *audio)
as.freq = conf.freq;
as.nchannels = SHIFT;
as.fmt = AUD_FMT_S16;
+ as.endianness = AUDIO_HOST_ENDIANNESS;
AUD_register_card (audio, "adlib", &s->card);
@@ -310,8 +311,7 @@ int Adlib_init (AudioState *audio)
"adlib",
s,
adlib_callback,
- &as,
- 0 /* XXX: little endian? */
+ &as
);
if (!s->voice) {
Adlib_fini (s);
diff --git a/hw/es1370.c b/hw/es1370.c
index 2aa2db9eb7..0d2d861166 100644
--- a/hw/es1370.c
+++ b/hw/es1370.c
@@ -423,6 +423,7 @@ static void es1370_update_voices (ES1370State *s, uint32_t ctl, uint32_t sctl)
as.freq = new_freq;
as.nchannels = 1 << (new_fmt & 1);
as.fmt = (new_fmt & 2) ? AUD_FMT_S16 : AUD_FMT_U8;
+ as.endianness = 0;
if (i == ADC_CHANNEL) {
s->adc_voice =
@@ -432,8 +433,7 @@ static void es1370_update_voices (ES1370State *s, uint32_t ctl, uint32_t sctl)
"es1370.adc",
s,
es1370_adc_callback,
- &as,
- 0 /* little endian */
+ &as
);
}
else {
@@ -444,8 +444,7 @@ static void es1370_update_voices (ES1370State *s, uint32_t ctl, uint32_t sctl)
i ? "es1370.dac2" : "es1370.dac1",
s,
i ? es1370_dac2_callback : es1370_dac1_callback,
- &as,
- 0 /* litle endian */
+ &as
);
}
}
diff --git a/hw/pcspk.c b/hw/pcspk.c
index 2e30662a25..0d52b31b45 100644
--- a/hw/pcspk.c
+++ b/hw/pcspk.c
@@ -95,7 +95,7 @@ static void pcspk_callback(void *opaque, int free)
int pcspk_audio_init(AudioState *audio)
{
PCSpkState *s = &pcspk_state;
- audsettings_t as = {PCSPK_SAMPLE_RATE, 1, AUD_FMT_U8};
+ audsettings_t as = {PCSPK_SAMPLE_RATE, 1, AUD_FMT_U8, 0};
if (!audio) {
AUD_log(s_spk, "No audio state\n");
@@ -103,7 +103,7 @@ int pcspk_audio_init(AudioState *audio)
}
AUD_register_card(audio, s_spk, &s->card);
- s->voice = AUD_open_out(&s->card, s->voice, s_spk, s, pcspk_callback, &as, 0);
+ s->voice = AUD_open_out(&s->card, s->voice, s_spk, s, pcspk_callback, &as);
if (!s->voice) {
AUD_log(s_spk, "Could not open voice\n");
return -1;
diff --git a/hw/sb16.c b/hw/sb16.c
index 6443a90d03..04325ac031 100644
--- a/hw/sb16.c
+++ b/hw/sb16.c
@@ -203,6 +203,7 @@ static void continue_dma8 (SB16State *s)
as.freq = s->freq;
as.nchannels = 1 << s->fmt_stereo;
as.fmt = s->fmt;
+ as.endianness = 0;
s->voice = AUD_open_out (
&s->card,
@@ -210,8 +211,7 @@ static void continue_dma8 (SB16State *s)
"sb16",
s,
SB_audio_callback,
- &as,
- 0 /* little endian */
+ &as
);
}
@@ -348,6 +348,7 @@ static void dma_cmd (SB16State *s, uint8_t cmd, uint8_t d0, int dma_len)
as.freq = s->freq;
as.nchannels = 1 << s->fmt_stereo;
as.fmt = s->fmt;
+ as.endianness = 0;
s->voice = AUD_open_out (
&s->card,
@@ -355,8 +356,7 @@ static void dma_cmd (SB16State *s, uint8_t cmd, uint8_t d0, int dma_len)
"sb16",
s,
SB_audio_callback,
- &as,
- 0 /* little endian */
+ &as
);
}
@@ -838,6 +838,7 @@ static void legacy_reset (SB16State *s)
as.freq = s->freq;
as.nchannels = 1;
as.fmt = AUD_FMT_U8;
+ as.endianness = 0;
s->voice = AUD_open_out (
&s->card,
@@ -845,8 +846,7 @@ static void legacy_reset (SB16State *s)
"sb16",
s,
SB_audio_callback,
- &as,
- 0 /* little endian */
+ &as
);
/* Not sure about that... */
@@ -1371,6 +1371,7 @@ static int SB_load (QEMUFile *f, void *opaque, int version_id)
as.freq = s->freq;
as.nchannels = 1 << s->fmt_stereo;
as.fmt = s->fmt;
+ as.endianness = 0;
s->voice = AUD_open_out (
&s->card,
@@ -1378,8 +1379,7 @@ static int SB_load (QEMUFile *f, void *opaque, int version_id)
"sb16",
s,
SB_audio_callback,
- &as,
- 0 /* little endian */
+ &as
);
}