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>

10 files changed, 147 insertions, 140 deletions

diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 635be73bf4..5bd034267f 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -87,7 +87,7 @@ struct alsa_params_req {
 
 struct alsa_params_obt {
     int freq;
-    audfmt_e fmt;
+    AudioFormat fmt;
     int endianness;
     int nchannels;
     snd_pcm_uframes_t samples;
@@ -294,16 +294,16 @@ static int alsa_write (SWVoiceOut *sw, void *buf, int len)
     return audio_pcm_sw_write (sw, buf, len);
 }
 
-static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
+static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness)
 {
     switch (fmt) {
-    case AUD_FMT_S8:
+    case AUDIO_FORMAT_S8:
         return SND_PCM_FORMAT_S8;
 
-    case AUD_FMT_U8:
+    case AUDIO_FORMAT_U8:
         return SND_PCM_FORMAT_U8;
 
-    case AUD_FMT_S16:
+    case AUDIO_FORMAT_S16:
         if (endianness) {
             return SND_PCM_FORMAT_S16_BE;
         }
@@ -311,7 +311,7 @@ static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
             return SND_PCM_FORMAT_S16_LE;
         }
 
-    case AUD_FMT_U16:
+    case AUDIO_FORMAT_U16:
         if (endianness) {
             return SND_PCM_FORMAT_U16_BE;
         }
@@ -319,7 +319,7 @@ static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
             return SND_PCM_FORMAT_U16_LE;
         }
 
-    case AUD_FMT_S32:
+    case AUDIO_FORMAT_S32:
         if (endianness) {
             return SND_PCM_FORMAT_S32_BE;
         }
@@ -327,7 +327,7 @@ static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
             return SND_PCM_FORMAT_S32_LE;
         }
 
-    case AUD_FMT_U32:
+    case AUDIO_FORMAT_U32:
         if (endianness) {
             return SND_PCM_FORMAT_U32_BE;
         }
@@ -344,58 +344,58 @@ static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
     }
 }
 
-static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt,
+static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt,
                            int *endianness)
 {
     switch (alsafmt) {
     case SND_PCM_FORMAT_S8:
         *endianness = 0;
-        *fmt = AUD_FMT_S8;
+        *fmt = AUDIO_FORMAT_S8;
         break;
 
     case SND_PCM_FORMAT_U8:
         *endianness = 0;
-        *fmt = AUD_FMT_U8;
+        *fmt = AUDIO_FORMAT_U8;
         break;
 
     case SND_PCM_FORMAT_S16_LE:
         *endianness = 0;
-        *fmt = AUD_FMT_S16;
+        *fmt = AUDIO_FORMAT_S16;
         break;
 
     case SND_PCM_FORMAT_U16_LE:
         *endianness = 0;
-        *fmt = AUD_FMT_U16;
+        *fmt = AUDIO_FORMAT_U16;
         break;
 
     case SND_PCM_FORMAT_S16_BE:
         *endianness = 1;
-        *fmt = AUD_FMT_S16;
+        *fmt = AUDIO_FORMAT_S16;
         break;
 
     case SND_PCM_FORMAT_U16_BE:
         *endianness = 1;
-        *fmt = AUD_FMT_U16;
+        *fmt = AUDIO_FORMAT_U16;
         break;
 
     case SND_PCM_FORMAT_S32_LE:
         *endianness = 0;
-        *fmt = AUD_FMT_S32;
+        *fmt = AUDIO_FORMAT_S32;
         break;
 
     case SND_PCM_FORMAT_U32_LE:
         *endianness = 0;
-        *fmt = AUD_FMT_U32;
+        *fmt = AUDIO_FORMAT_U32;
         break;
 
     case SND_PCM_FORMAT_S32_BE:
         *endianness = 1;
-        *fmt = AUD_FMT_S32;
+        *fmt = AUDIO_FORMAT_S32;
         break;
 
     case SND_PCM_FORMAT_U32_BE:
         *endianness = 1;
-        *fmt = AUD_FMT_U32;
+        *fmt = AUDIO_FORMAT_U32;
         break;
 
     default:
@@ -638,19 +638,22 @@ static int alsa_open (int in, struct alsa_params_req *req,
         bytes_per_sec = freq << (nchannels == 2);
 
         switch (obt->fmt) {
-        case AUD_FMT_S8:
-        case AUD_FMT_U8:
+        case AUDIO_FORMAT_S8:
+        case AUDIO_FORMAT_U8:
             break;
 
-        case AUD_FMT_S16:
-        case AUD_FMT_U16:
+        case AUDIO_FORMAT_S16:
+        case AUDIO_FORMAT_U16:
             bytes_per_sec <<= 1;
             break;
 
-        case AUD_FMT_S32:
-        case AUD_FMT_U32:
+        case AUDIO_FORMAT_S32:
+        case AUDIO_FORMAT_U32:
             bytes_per_sec <<= 2;
             break;
+
+        default:
+            abort();
         }
 
         threshold = (conf->threshold * bytes_per_sec) / 1000;
diff --git a/audio/audio.c b/audio/audio.c
index 909c817103..77216e5010 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -113,7 +113,7 @@ static struct {
         .settings = {
             .freq = 44100,
             .nchannels = 2,
-            .fmt = AUD_FMT_S16,
+            .fmt = AUDIO_FORMAT_S16,
             .endianness =  AUDIO_HOST_ENDIANNESS,
         }
     },
@@ -125,7 +125,7 @@ static struct {
         .settings = {
             .freq = 44100,
             .nchannels = 2,
-            .fmt = AUD_FMT_S16,
+            .fmt = AUDIO_FORMAT_S16,
             .endianness = AUDIO_HOST_ENDIANNESS,
         }
     },
@@ -257,58 +257,61 @@ static char *audio_alloc_prefix (const char *s)
     return r;
 }
 
-static const char *audio_audfmt_to_string (audfmt_e fmt)
+static const char *audio_audfmt_to_string (AudioFormat fmt)
 {
     switch (fmt) {
-    case AUD_FMT_U8:
+    case AUDIO_FORMAT_U8:
         return "U8";
 
-    case AUD_FMT_U16:
+    case AUDIO_FORMAT_U16:
         return "U16";
 
-    case AUD_FMT_S8:
+    case AUDIO_FORMAT_S8:
         return "S8";
 
-    case AUD_FMT_S16:
+    case AUDIO_FORMAT_S16:
         return "S16";
 
-    case AUD_FMT_U32:
+    case AUDIO_FORMAT_U32:
         return "U32";
 
-    case AUD_FMT_S32:
+    case AUDIO_FORMAT_S32:
         return "S32";
+
+    default:
+        abort();
     }
 
     dolog ("Bogus audfmt %d returning S16\n", fmt);
     return "S16";
 }
 
-static audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval,
+static AudioFormat audio_string_to_audfmt (const char *s, AudioFormat defval,
                                         int *defaultp)
 {
     if (!strcasecmp (s, "u8")) {
         *defaultp = 0;
-        return AUD_FMT_U8;
+        return AUDIO_FORMAT_U8;
     }
     else if (!strcasecmp (s, "u16")) {
         *defaultp = 0;
-        return AUD_FMT_U16;
+        return AUDIO_FORMAT_U16;
     }
     else if (!strcasecmp (s, "u32")) {
         *defaultp = 0;
-        return AUD_FMT_U32;
+        return AUDIO_FORMAT_U32;
     }
     else if (!strcasecmp (s, "s8")) {
         *defaultp = 0;
-        return AUD_FMT_S8;
+        return AUDIO_FORMAT_S8;
     }
     else if (!strcasecmp (s, "s16")) {
         *defaultp = 0;
-        return AUD_FMT_S16;
+        return AUDIO_FORMAT_S16;
     }
     else if (!strcasecmp (s, "s32")) {
         *defaultp = 0;
-        return AUD_FMT_S32;
+        return AUDIO_FORMAT_S32;
     }
     else {
         dolog ("Bogus audio format `%s' using %s\n",
@@ -318,8 +321,8 @@ static audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval,
     }
 }
 
-static audfmt_e audio_get_conf_fmt (const char *envname,
-                                    audfmt_e defval,
+static AudioFormat audio_get_conf_fmt (const char *envname,
+                                    AudioFormat defval,
                                     int *defaultp)
 {
     const char *var = getenv (envname);
@@ -421,7 +424,7 @@ static void audio_print_options (const char *prefix,
 
         case AUD_OPT_FMT:
             {
-                audfmt_e *fmtp = opt->valp;
+                AudioFormat *fmtp = opt->valp;
                 printf (
                     "format, %s = %s, (one of: U8 S8 U16 S16 U32 S32)\n",
                     state,
@@ -492,7 +495,7 @@ static void audio_process_options (const char *prefix,
 
         case AUD_OPT_FMT:
             {
-                audfmt_e *fmtp = opt->valp;
+                AudioFormat *fmtp = opt->valp;
                 *fmtp = audio_get_conf_fmt (optname, *fmtp, &def);
             }
             break;
@@ -524,22 +527,22 @@ static void audio_print_settings (struct audsettings *as)
     dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels);
 
     switch (as->fmt) {
-    case AUD_FMT_S8:
+    case AUDIO_FORMAT_S8:
         AUD_log (NULL, "S8");
         break;
-    case AUD_FMT_U8:
+    case AUDIO_FORMAT_U8:
         AUD_log (NULL, "U8");
         break;
-    case AUD_FMT_S16:
+    case AUDIO_FORMAT_S16:
         AUD_log (NULL, "S16");
         break;
-    case AUD_FMT_U16:
+    case AUDIO_FORMAT_U16:
         AUD_log (NULL, "U16");
         break;
-    case AUD_FMT_S32:
+    case AUDIO_FORMAT_S32:
         AUD_log (NULL, "S32");
         break;
-    case AUD_FMT_U32:
+    case AUDIO_FORMAT_U32:
         AUD_log (NULL, "U32");
         break;
     default:
@@ -570,12 +573,12 @@ static int audio_validate_settings (struct audsettings *as)
     invalid |= as->endianness != 0 && as->endianness != 1;
 
     switch (as->fmt) {
-    case AUD_FMT_S8:
-    case AUD_FMT_U8:
-    case AUD_FMT_S16:
-    case AUD_FMT_U16:
-    case AUD_FMT_S32:
-    case AUD_FMT_U32:
+    case AUDIO_FORMAT_S8:
+    case AUDIO_FORMAT_U8:
+    case AUDIO_FORMAT_S16:
+    case AUDIO_FORMAT_U16:
+    case AUDIO_FORMAT_S32:
+    case AUDIO_FORMAT_U32:
         break;
     default:
         invalid = 1;
@@ -591,25 +594,28 @@ static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *a
     int bits = 8, sign = 0;
 
     switch (as->fmt) {
-    case AUD_FMT_S8:
+    case AUDIO_FORMAT_S8:
         sign = 1;
         /* fall through */
-    case AUD_FMT_U8:
+    case AUDIO_FORMAT_U8:
         break;
 
-    case AUD_FMT_S16:
+    case AUDIO_FORMAT_S16:
         sign = 1;
         /* fall through */
-    case AUD_FMT_U16:
+    case AUDIO_FORMAT_U16:
         bits = 16;
         break;
 
-    case AUD_FMT_S32:
+    case AUDIO_FORMAT_S32:
         sign = 1;
         /* fall through */
-    case AUD_FMT_U32:
+    case AUDIO_FORMAT_U32:
         bits = 32;
         break;
+
+    default:
+        abort();
     }
     return info->freq == as->freq
         && info->nchannels == as->nchannels
@@ -623,24 +629,27 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
     int bits = 8, sign = 0, shift = 0;
 
     switch (as->fmt) {
-    case AUD_FMT_S8:
+    case AUDIO_FORMAT_S8:
         sign = 1;
-    case AUD_FMT_U8:
+    case AUDIO_FORMAT_U8:
         break;
 
-    case AUD_FMT_S16:
+    case AUDIO_FORMAT_S16:
         sign = 1;
-    case AUD_FMT_U16:
+    case AUDIO_FORMAT_U16:
         bits = 16;
         shift = 1;
         break;
 
-    case AUD_FMT_S32:
+    case AUDIO_FORMAT_S32:
         sign = 1;
-    case AUD_FMT_U32:
+    case AUDIO_FORMAT_U32:
         bits = 32;
         shift = 2;
         break;
+
+    default:
+        abort();
     }
 
     info->freq = as->freq;
diff --git a/audio/audio.h b/audio/audio.h
index f4339a185e..02f29a3b3e 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -26,18 +26,10 @@
 #define QEMU_AUDIO_H
 
 #include "qemu/queue.h"
+#include "qapi/qapi-types-audio.h"
 
 typedef void (*audio_callback_fn) (void *opaque, int avail);
 
-typedef enum {
-    AUD_FMT_U8,
-    AUD_FMT_S8,
-    AUD_FMT_U16,
-    AUD_FMT_S16,
-    AUD_FMT_U32,
-    AUD_FMT_S32
-} audfmt_e;
-
 #ifdef HOST_WORDS_BIGENDIAN
 #define AUDIO_HOST_ENDIANNESS 1
 #else
@@ -47,7 +39,7 @@ typedef enum {
 struct audsettings {
     int freq;
     int nchannels;
-    audfmt_e fmt;
+    AudioFormat fmt;
     int endianness;
 };
 
diff --git a/audio/audio_win_int.c b/audio/audio_win_int.c
index 6900008d0c..b938fd667b 100644
--- a/audio/audio_win_int.c
+++ b/audio/audio_win_int.c
@@ -24,20 +24,20 @@ int waveformat_from_audio_settings (WAVEFORMATEX *wfx,
     wfx->cbSize = 0;
 
     switch (as->fmt) {
-    case AUD_FMT_S8:
-    case AUD_FMT_U8:
+    case AUDIO_FORMAT_S8:
+    case AUDIO_FORMAT_U8:
         wfx->wBitsPerSample = 8;
         break;
 
-    case AUD_FMT_S16:
-    case AUD_FMT_U16:
+    case AUDIO_FORMAT_S16:
+    case AUDIO_FORMAT_U16:
         wfx->wBitsPerSample = 16;
         wfx->nAvgBytesPerSec <<= 1;
         wfx->nBlockAlign <<= 1;
         break;
 
-    case AUD_FMT_S32:
-    case AUD_FMT_U32:
+    case AUDIO_FORMAT_S32:
+    case AUDIO_FORMAT_U32:
         wfx->wBitsPerSample = 32;
         wfx->nAvgBytesPerSec <<= 2;
         wfx->nBlockAlign <<= 2;
@@ -85,15 +85,15 @@ int waveformat_to_audio_settings (WAVEFORMATEX *wfx,
 
     switch (wfx->wBitsPerSample) {
     case 8:
-        as->fmt = AUD_FMT_U8;
+        as->fmt = AUDIO_FORMAT_U8;
         break;
 
     case 16:
-        as->fmt = AUD_FMT_S16;
+        as->fmt = AUDIO_FORMAT_S16;
         break;
 
     case 32:
-        as->fmt = AUD_FMT_S32;
+        as->fmt = AUDIO_FORMAT_S32;
         break;
 
     default:
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 6c69622b4c..355e8fbda5 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -70,7 +70,7 @@ typedef struct OSSVoiceIn {
 
 struct oss_params {
     int freq;
-    audfmt_e fmt;
+    AudioFormat fmt;
     int nchannels;
     int nfrags;
     int fragsize;
@@ -148,16 +148,16 @@ static int oss_write (SWVoiceOut *sw, void *buf, int len)
     return audio_pcm_sw_write (sw, buf, len);
 }
 
-static int aud_to_ossfmt (audfmt_e fmt, int endianness)
+static int aud_to_ossfmt (AudioFormat fmt, int endianness)
 {
     switch (fmt) {
-    case AUD_FMT_S8:
+    case AUDIO_FORMAT_S8:
         return AFMT_S8;
 
-    case AUD_FMT_U8:
+    case AUDIO_FORMAT_U8:
         return AFMT_U8;
 
-    case AUD_FMT_S16:
+    case AUDIO_FORMAT_S16:
         if (endianness) {
             return AFMT_S16_BE;
         }
@@ -165,7 +165,7 @@ static int aud_to_ossfmt (audfmt_e fmt, int endianness)
             return AFMT_S16_LE;
         }
 
-    case AUD_FMT_U16:
+    case AUDIO_FORMAT_U16:
         if (endianness) {
             return AFMT_U16_BE;
         }
@@ -182,37 +182,37 @@ static int aud_to_ossfmt (audfmt_e fmt, int endianness)
     }
 }
 
-static int oss_to_audfmt (int ossfmt, audfmt_e *fmt, int *endianness)
+static int oss_to_audfmt (int ossfmt, AudioFormat *fmt, int *endianness)
 {
     switch (ossfmt) {
     case AFMT_S8:
         *endianness = 0;
-        *fmt = AUD_FMT_S8;
+        *fmt = AUDIO_FORMAT_S8;
         break;
 
     case AFMT_U8:
         *endianness = 0;
-        *fmt = AUD_FMT_U8;
+        *fmt = AUDIO_FORMAT_U8;
         break;
 
     case AFMT_S16_LE:
         *endianness = 0;
-        *fmt = AUD_FMT_S16;
+        *fmt = AUDIO_FORMAT_S16;
         break;
 
     case AFMT_U16_LE:
         *endianness = 0;
-        *fmt = AUD_FMT_U16;
+        *fmt = AUDIO_FORMAT_U16;
         break;
 
     case AFMT_S16_BE:
         *endianness = 1;
-        *fmt = AUD_FMT_S16;
+        *fmt = AUDIO_FORMAT_S16;
         break;
 
     case AFMT_U16_BE:
         *endianness = 1;
-        *fmt = AUD_FMT_U16;
+        *fmt = AUDIO_FORMAT_U16;
         break;
 
     default:
@@ -500,7 +500,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
     int endianness;
     int err;
     int fd;
-    audfmt_e effective_fmt;
+    AudioFormat effective_fmt;
     struct audsettings obt_as;
     OSSConf *conf = drv_opaque;
 
@@ -667,7 +667,7 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
     int endianness;
     int err;
     int fd;
-    audfmt_e effective_fmt;
+    AudioFormat effective_fmt;
     struct audsettings obt_as;
     OSSConf *conf = drv_opaque;
 
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 6153b908da..8246f260a8 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -385,21 +385,21 @@ static int qpa_read (SWVoiceIn *sw, void *buf, int len)
     return audio_pcm_sw_read (sw, buf, len);
 }
 
-static pa_sample_format_t audfmt_to_pa (audfmt_e afmt, int endianness)
+static pa_sample_format_t audfmt_to_pa (AudioFormat afmt, int endianness)
 {
     int format;
 
     switch (afmt) {
-    case AUD_FMT_S8:
-    case AUD_FMT_U8:
+    case AUDIO_FORMAT_S8:
+    case AUDIO_FORMAT_U8:
         format = PA_SAMPLE_U8;
         break;
-    case AUD_FMT_S16:
-    case AUD_FMT_U16:
+    case AUDIO_FORMAT_S16:
+    case AUDIO_FORMAT_U16:
         format = endianness ? PA_SAMPLE_S16BE : PA_SAMPLE_S16LE;
         break;
-    case AUD_FMT_S32:
-    case AUD_FMT_U32:
+    case AUDIO_FORMAT_S32:
+    case AUDIO_FORMAT_U32:
         format = endianness ? PA_SAMPLE_S32BE : PA_SAMPLE_S32LE;
         break;
     default:
@@ -410,26 +410,26 @@ static pa_sample_format_t audfmt_to_pa (audfmt_e afmt, int endianness)
     return format;
 }
 
-static audfmt_e pa_to_audfmt (pa_sample_format_t fmt, int *endianness)
+static AudioFormat pa_to_audfmt (pa_sample_format_t fmt, int *endianness)
 {
     switch (fmt) {
     case PA_SAMPLE_U8:
-        return AUD_FMT_U8;
+        return AUDIO_FORMAT_U8;
     case PA_SAMPLE_S16BE:
         *endianness = 1;
-        return AUD_FMT_S16;
+        return AUDIO_FORMAT_S16;
     case PA_SAMPLE_S16LE:
         *endianness = 0;
-        return AUD_FMT_S16;
+        return AUDIO_FORMAT_S16;
     case PA_SAMPLE_S32BE:
         *endianness = 1;
-        return AUD_FMT_S32;
+        return AUDIO_FORMAT_S32;
     case PA_SAMPLE_S32LE:
         *endianness = 0;
-        return AUD_FMT_S32;
+        return AUDIO_FORMAT_S32;
     default:
         dolog ("Internal logic error: Bad pa_sample_format %d\n", fmt);
-        return AUD_FMT_U8;
+        return AUDIO_FORMAT_U8;
     }
 }
 
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index f7ee70b153..4cd4cbaf00 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -68,19 +68,19 @@ static void GCC_FMT_ATTR (1, 2) sdl_logerr (const char *fmt, ...)
     AUD_log (AUDIO_CAP, "Reason: %s\n", SDL_GetError ());
 }
 
-static int aud_to_sdlfmt (audfmt_e fmt)
+static int aud_to_sdlfmt (AudioFormat fmt)
 {
     switch (fmt) {
-    case AUD_FMT_S8:
+    case AUDIO_FORMAT_S8:
         return AUDIO_S8;
 
-    case AUD_FMT_U8:
+    case AUDIO_FORMAT_U8:
         return AUDIO_U8;
 
-    case AUD_FMT_S16:
+    case AUDIO_FORMAT_S16:
         return AUDIO_S16LSB;
 
-    case AUD_FMT_U16:
+    case AUDIO_FORMAT_U16:
         return AUDIO_U16LSB;
 
     default:
@@ -92,37 +92,37 @@ static int aud_to_sdlfmt (audfmt_e fmt)
     }
 }
 
-static int sdl_to_audfmt(int sdlfmt, audfmt_e *fmt, int *endianness)
+static int sdl_to_audfmt(int sdlfmt, AudioFormat *fmt, int *endianness)
 {
     switch (sdlfmt) {
     case AUDIO_S8:
         *endianness = 0;
-        *fmt = AUD_FMT_S8;
+        *fmt = AUDIO_FORMAT_S8;
         break;
 
     case AUDIO_U8:
         *endianness = 0;
-        *fmt = AUD_FMT_U8;
+        *fmt = AUDIO_FORMAT_U8;
         break;
 
     case AUDIO_S16LSB:
         *endianness = 0;
-        *fmt = AUD_FMT_S16;
+        *fmt = AUDIO_FORMAT_S16;
         break;
 
     case AUDIO_U16LSB:
         *endianness = 0;
-        *fmt = AUD_FMT_U16;
+        *fmt = AUDIO_FORMAT_U16;
         break;
 
     case AUDIO_S16MSB:
         *endianness = 1;
-        *fmt = AUD_FMT_S16;
+        *fmt = AUDIO_FORMAT_S16;
         break;
 
     case AUDIO_U16MSB:
         *endianness = 1;
-        *fmt = AUD_FMT_U16;
+        *fmt = AUDIO_FORMAT_U16;
         break;
 
     default:
@@ -265,7 +265,7 @@ static int sdl_init_out(HWVoiceOut *hw, struct audsettings *as,
     SDL_AudioSpec req, obt;
     int endianness;
     int err;
-    audfmt_e effective_fmt;
+    AudioFormat effective_fmt;
     struct audsettings obt_as;
 
     req.freq = as->freq;
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index 6ad0eafbc6..3aeb0cb357 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -130,7 +130,7 @@ static int line_out_init(HWVoiceOut *hw, struct audsettings *as,
     settings.freq       = SPICE_INTERFACE_PLAYBACK_FREQ;
 #endif
     settings.nchannels  = SPICE_INTERFACE_PLAYBACK_CHAN;
-    settings.fmt        = AUD_FMT_S16;
+    settings.fmt        = AUDIO_FORMAT_S16;
     settings.endianness = AUDIO_HOST_ENDIANNESS;
 
     audio_pcm_init_info (&hw->info, &settings);
@@ -258,7 +258,7 @@ static int line_in_init(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
     settings.freq       = SPICE_INTERFACE_RECORD_FREQ;
 #endif
     settings.nchannels  = SPICE_INTERFACE_RECORD_CHAN;
-    settings.fmt        = AUD_FMT_S16;
+    settings.fmt        = AUDIO_FORMAT_S16;
     settings.endianness = AUDIO_HOST_ENDIANNESS;
 
     audio_pcm_init_info (&hw->info, &settings);
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 40adfa30c3..35a614785e 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -117,20 +117,23 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
 
     stereo = wav_as.nchannels == 2;
     switch (wav_as.fmt) {
-    case AUD_FMT_S8:
-    case AUD_FMT_U8:
+    case AUDIO_FORMAT_S8:
+    case AUDIO_FORMAT_U8:
         bits16 = 0;
         break;
 
-    case AUD_FMT_S16:
-    case AUD_FMT_U16:
+    case AUDIO_FORMAT_S16:
+    case AUDIO_FORMAT_U16:
         bits16 = 1;
         break;
 
-    case AUD_FMT_S32:
-    case AUD_FMT_U32:
+    case AUDIO_FORMAT_S32:
+    case AUDIO_FORMAT_U32:
         dolog ("WAVE files can not handle 32bit formats\n");
         return -1;
+
+    default:
+        abort();
     }
 
     hdr[34] = bits16 ? 0x10 : 0x08;
@@ -225,7 +228,7 @@ static int wav_ctl_out (HWVoiceOut *hw, int cmd, ...)
 static WAVConf glob_conf = {
     .settings.freq      = 44100,
     .settings.nchannels = 2,
-    .settings.fmt       = AUD_FMT_S16,
+    .settings.fmt       = AUDIO_FORMAT_S16,
     .wav_path           = "qemu.wav"
 };
 
diff --git a/audio/wavcapture.c b/audio/wavcapture.c
index cd24570aa7..74320dfecc 100644
--- a/audio/wavcapture.c
+++ b/audio/wavcapture.c
@@ -136,7 +136,7 @@ int wav_start_capture (CaptureState *s, const char *path, int freq,
 
     as.freq = freq;
     as.nchannels = 1 << stereo;
-    as.fmt = bits16 ? AUD_FMT_S16 : AUD_FMT_U8;
+    as.fmt = bits16 ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_U8;
     as.endianness = 0;
 
     ops.notify = wav_notify;