audio endianness API changes (malc)

git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@2042 c046a42c-6fe2-441c-8c8c-71466251a162

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
                 );
         }