audio: use qapi AudioFormat instead of audfmt_e

I had to include an enum for audio sampling formats into qapi, but that
meant duplicating the audfmt_e enum.  This patch replaces audfmt_e and
associated values with the qapi generated AudioFormat enum.

This patch is mostly a search-and-replace, except for switches where the
qapi generated AUDIO_FORMAT_MAX caused problems.

Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Reviewed-by: Thomas Huth <thuth@redhat.com>
Message-id: 01251b2758a1679c66842120b77c0fb46d7d0eaf.1552083282.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>

11 files changed, 32 insertions, 32 deletions

diff --git a/hw/audio/ac97.c b/hw/audio/ac97.c
index d799533aa9..2265622d44 100644
--- a/hw/audio/ac97.c
+++ b/hw/audio/ac97.c
@@ -365,7 +365,7 @@ static void open_voice (AC97LinkState *s, int index, int freq)
 
     as.freq = freq;
     as.nchannels = 2;
-    as.fmt = AUD_FMT_S16;
+    as.fmt = AUDIO_FORMAT_S16;
     as.endianness = 0;
 
     if (freq > 0) {
diff --git a/hw/audio/adlib.c b/hw/audio/adlib.c
index 97b876c7e0..0957780a3d 100644
--- a/hw/audio/adlib.c
+++ b/hw/audio/adlib.c
@@ -269,7 +269,7 @@ static void adlib_realizefn (DeviceState *dev, Error **errp)
 
     as.freq = s->freq;
     as.nchannels = SHIFT;
-    as.fmt = AUD_FMT_S16;
+    as.fmt = AUDIO_FORMAT_S16;
     as.endianness = AUDIO_HOST_ENDIANNESS;
 
     AUD_register_card ("adlib", &s->card);
diff --git a/hw/audio/cs4231a.c b/hw/audio/cs4231a.c
index 9089dcb47e..62da75eefe 100644
--- a/hw/audio/cs4231a.c
+++ b/hw/audio/cs4231a.c
@@ -288,7 +288,7 @@ static void cs_reset_voices (CSState *s, uint32_t val)
 
     switch ((val >> 5) & ((s->dregs[MODE_And_ID] & MODE2) ? 7 : 3)) {
     case 0:
-        as.fmt = AUD_FMT_U8;
+        as.fmt = AUDIO_FORMAT_U8;
         s->shift = as.nchannels == 2;
         break;
 
@@ -298,7 +298,7 @@ static void cs_reset_voices (CSState *s, uint32_t val)
     case 3:
         s->tab = ALawDecompressTable;
     x_law:
-        as.fmt = AUD_FMT_S16;
+        as.fmt = AUDIO_FORMAT_S16;
         as.endianness = AUDIO_HOST_ENDIANNESS;
         s->shift = as.nchannels == 2;
         break;
@@ -307,7 +307,7 @@ static void cs_reset_voices (CSState *s, uint32_t val)
         as.endianness = 1;
         /* fall through */
     case 2:
-        as.fmt = AUD_FMT_S16;
+        as.fmt = AUDIO_FORMAT_S16;
         s->shift = as.nchannels;
         break;
 
diff --git a/hw/audio/es1370.c b/hw/audio/es1370.c
index 97789a0771..a5314d66fd 100644
--- a/hw/audio/es1370.c
+++ b/hw/audio/es1370.c
@@ -414,14 +414,14 @@ static void es1370_update_voices (ES1370State *s, uint32_t ctl, uint32_t sctl)
                     i,
                     new_freq,
                     1 << (new_fmt & 1),
-                    (new_fmt & 2) ? AUD_FMT_S16 : AUD_FMT_U8,
+                    (new_fmt & 2) ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_U8,
                     d->shift);
             if (new_freq) {
                 struct audsettings as;
 
                 as.freq = new_freq;
                 as.nchannels = 1 << (new_fmt & 1);
-                as.fmt = (new_fmt & 2) ? AUD_FMT_S16 : AUD_FMT_U8;
+                as.fmt = (new_fmt & 2) ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_U8;
                 as.endianness = 0;
 
                 if (i == ADC_CHANNEL) {
diff --git a/hw/audio/gus.c b/hw/audio/gus.c
index 8e0b27e0f2..b3e2a7fdd5 100644
--- a/hw/audio/gus.c
+++ b/hw/audio/gus.c
@@ -251,7 +251,7 @@ static void gus_realizefn (DeviceState *dev, Error **errp)
 
     as.freq = s->freq;
     as.nchannels = 2;
-    as.fmt = AUD_FMT_S16;
+    as.fmt = AUDIO_FORMAT_S16;
     as.endianness = GUS_ENDIANNESS;
 
     s->voice = AUD_open_out (
diff --git a/hw/audio/hda-codec.c b/hw/audio/hda-codec.c
index 617a1c1016..c25bfa38b1 100644
--- a/hw/audio/hda-codec.c
+++ b/hw/audio/hda-codec.c
@@ -99,9 +99,9 @@ static void hda_codec_parse_fmt(uint32_t format, struct audsettings *as)
     }
 
     switch (format & AC_FMT_BITS_MASK) {
-    case AC_FMT_BITS_8:  as->fmt = AUD_FMT_S8;  break;
-    case AC_FMT_BITS_16: as->fmt = AUD_FMT_S16; break;
-    case AC_FMT_BITS_32: as->fmt = AUD_FMT_S32; break;
+    case AC_FMT_BITS_8:  as->fmt = AUDIO_FORMAT_S8;  break;
+    case AC_FMT_BITS_16: as->fmt = AUDIO_FORMAT_S16; break;
+    case AC_FMT_BITS_32: as->fmt = AUDIO_FORMAT_S32; break;
     }
 
     as->nchannels = ((format & AC_FMT_CHAN_MASK) >> AC_FMT_CHAN_SHIFT) + 1;
@@ -134,12 +134,12 @@ static void hda_codec_parse_fmt(uint32_t format, struct audsettings *as)
 /* -------------------------------------------------------------------------- */
 
 static const char *fmt2name[] = {
-    [ AUD_FMT_U8  ] = "PCM-U8",
-    [ AUD_FMT_S8  ] = "PCM-S8",
-    [ AUD_FMT_U16 ] = "PCM-U16",
-    [ AUD_FMT_S16 ] = "PCM-S16",
-    [ AUD_FMT_U32 ] = "PCM-U32",
-    [ AUD_FMT_S32 ] = "PCM-S32",
+    [ AUDIO_FORMAT_U8  ] = "PCM-U8",
+    [ AUDIO_FORMAT_S8  ] = "PCM-S8",
+    [ AUDIO_FORMAT_U16 ] = "PCM-U16",
+    [ AUDIO_FORMAT_S16 ] = "PCM-S16",
+    [ AUDIO_FORMAT_U32 ] = "PCM-U32",
+    [ AUDIO_FORMAT_S32 ] = "PCM-S32",
 };
 
 typedef struct HDAAudioState HDAAudioState;
diff --git a/hw/audio/lm4549.c b/hw/audio/lm4549.c
index a46f2301af..af8b22b541 100644
--- a/hw/audio/lm4549.c
+++ b/hw/audio/lm4549.c
@@ -185,7 +185,7 @@ void lm4549_write(lm4549_state *s,
         struct audsettings as;
         as.freq = value;
         as.nchannels = 2;
-        as.fmt = AUD_FMT_S16;
+        as.fmt = AUDIO_FORMAT_S16;
         as.endianness = 0;
 
         s->voice = AUD_open_out(
@@ -255,7 +255,7 @@ static int lm4549_post_load(void *opaque, int version_id)
     struct audsettings as;
     as.freq = freq;
     as.nchannels = 2;
-    as.fmt = AUD_FMT_S16;
+    as.fmt = AUDIO_FORMAT_S16;
     as.endianness = 0;
 
     s->voice = AUD_open_out(
@@ -292,7 +292,7 @@ void lm4549_init(lm4549_state *s, lm4549_callback data_req_cb, void* opaque)
     /* Open a default voice */
     as.freq = 48000;
     as.nchannels = 2;
-    as.fmt = AUD_FMT_S16;
+    as.fmt = AUDIO_FORMAT_S16;
     as.endianness = 0;
 
     s->voice = AUD_open_out(
diff --git a/hw/audio/milkymist-ac97.c b/hw/audio/milkymist-ac97.c
index bc8db71ae0..90cce1e6ed 100644
--- a/hw/audio/milkymist-ac97.c
+++ b/hw/audio/milkymist-ac97.c
@@ -308,7 +308,7 @@ static void milkymist_ac97_realize(DeviceState *dev, Error **errp)
 
     as.freq = 48000;
     as.nchannels = 2;
-    as.fmt = AUD_FMT_S16;
+    as.fmt = AUDIO_FORMAT_S16;
     as.endianness = 1;
 
     s->voice_in = AUD_open_in(&s->card, s->voice_in,
diff --git a/hw/audio/pcspk.c b/hw/audio/pcspk.c
index b80a62ce90..fdbb4b6e99 100644
--- a/hw/audio/pcspk.c
+++ b/hw/audio/pcspk.c
@@ -162,7 +162,7 @@ static void pcspk_initfn(Object *obj)
 
 static void pcspk_realizefn(DeviceState *dev, Error **errp)
 {
-    struct audsettings as = {PCSPK_SAMPLE_RATE, 1, AUD_FMT_U8, 0};
+    struct audsettings as = {PCSPK_SAMPLE_RATE, 1, AUDIO_FORMAT_U8, 0};
     ISADevice *isadev = ISA_DEVICE(dev);
     PCSpkState *s = PC_SPEAKER(dev);
 
diff --git a/hw/audio/sb16.c b/hw/audio/sb16.c
index c5b9bf79e8..65ea0cd938 100644
--- a/hw/audio/sb16.c
+++ b/hw/audio/sb16.c
@@ -66,7 +66,7 @@ typedef struct SB16State {
     int fmt_stereo;
     int fmt_signed;
     int fmt_bits;
-    audfmt_e fmt;
+    AudioFormat fmt;
     int dma_auto;
     int block_size;
     int fifo;
@@ -224,7 +224,7 @@ static void continue_dma8 (SB16State *s)
 
 static void dma_cmd8 (SB16State *s, int mask, int dma_len)
 {
-    s->fmt = AUD_FMT_U8;
+    s->fmt = AUDIO_FORMAT_U8;
     s->use_hdma = 0;
     s->fmt_bits = 8;
     s->fmt_signed = 0;
@@ -319,18 +319,18 @@ static void dma_cmd (SB16State *s, uint8_t cmd, uint8_t d0, int dma_len)
 
     if (16 == s->fmt_bits) {
         if (s->fmt_signed) {
-            s->fmt = AUD_FMT_S16;
+            s->fmt = AUDIO_FORMAT_S16;
         }
         else {
-            s->fmt = AUD_FMT_U16;
+            s->fmt = AUDIO_FORMAT_U16;
         }
     }
     else {
         if (s->fmt_signed) {
-            s->fmt = AUD_FMT_S8;
+            s->fmt = AUDIO_FORMAT_S8;
         }
         else {
-            s->fmt = AUD_FMT_U8;
+            s->fmt = AUDIO_FORMAT_U8;
         }
     }
 
@@ -852,7 +852,7 @@ static void legacy_reset (SB16State *s)
 
     as.freq = s->freq;
     as.nchannels = 1;
-    as.fmt = AUD_FMT_U8;
+    as.fmt = AUDIO_FORMAT_U8;
     as.endianness = 0;
 
     s->voice = AUD_open_out (
diff --git a/hw/audio/wm8750.c b/hw/audio/wm8750.c
index 169b006ade..ca0ad73caf 100644
--- a/hw/audio/wm8750.c
+++ b/hw/audio/wm8750.c
@@ -201,7 +201,7 @@ static void wm8750_set_format(WM8750State *s)
     in_fmt.endianness = 0;
     in_fmt.nchannels = 2;
     in_fmt.freq = s->adc_hz;
-    in_fmt.fmt = AUD_FMT_S16;
+    in_fmt.fmt = AUDIO_FORMAT_S16;
 
     s->adc_voice[0] = AUD_open_in(&s->card, s->adc_voice[0],
                     CODEC ".input1", s, wm8750_audio_in_cb, &in_fmt);
@@ -214,7 +214,7 @@ static void wm8750_set_format(WM8750State *s)
     out_fmt.endianness = 0;
     out_fmt.nchannels = 2;
     out_fmt.freq = s->dac_hz;
-    out_fmt.fmt = AUD_FMT_S16;
+    out_fmt.fmt = AUDIO_FORMAT_S16;
 
     s->dac_voice[0] = AUD_open_out(&s->card, s->dac_voice[0],
                     CODEC ".speaker", s, wm8750_audio_out_cb, &out_fmt);
@@ -681,7 +681,7 @@ uint32_t wm8750_adc_dat(void *opaque)
     if (s->idx_in >= sizeof(s->data_in)) {
         wm8750_in_load(s);
         if (s->idx_in >= sizeof(s->data_in)) {
-            return 0x80008000; /* silence in AUD_FMT_S16 sample format */
+            return 0x80008000; /* silence in AUDIO_FORMAT_S16 sample format */
         }
     }