audio endianness API changes (malc)

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

4 files changed, 15 insertions, 16 deletions

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
                 );
         }