diff mbox

[v2,5/6] audio: use qapi AudioFormat instead of audfmt_e

Message ID 9d0c3109b26c78d7be59f54527255f2af81bc68b.1434458391.git.DirtY.iCE.hu@gmail.com
State New
Headers show

Commit Message

=?UTF-8?B?Wm9sdMOhbiBLxZF2w6Fnw7M=?= June 16, 2015, 12:49 p.m. UTC 
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>
---
 audio/alsaaudio.c         | 53 ++++++++++++++------------
 audio/audio.c             | 97 ++++++++++++++++++++++++++---------------------
 audio/audio.h             | 11 +-----
 audio/audio_win_int.c     | 18 ++++-----
 audio/ossaudio.c          | 30 +++++++--------
 audio/paaudio.c           | 28 +++++++-------
 audio/sdlaudio.c          | 26 ++++++-------
 audio/spiceaudio.c        |  4 +-
 audio/wavaudio.c          | 17 +++++----
 audio/wavcapture.c        |  2 +-
 hw/arm/omap2.c            |  2 +-
 hw/audio/ac97.c           |  2 +-
 hw/audio/adlib.c          |  2 +-
 hw/audio/cs4231a.c        |  6 +--
 hw/audio/es1370.c         |  4 +-
 hw/audio/gus.c            |  2 +-
 hw/audio/hda-codec.c      | 18 ++++-----
 hw/audio/lm4549.c         |  6 +--
 hw/audio/milkymist-ac97.c |  2 +-
 hw/audio/pcspk.c          |  2 +-
 hw/audio/sb16.c           | 14 +++----
 hw/audio/wm8750.c         |  4 +-
 hw/input/tsc210x.c        |  2 +-
 hw/usb/dev-audio.c        |  2 +-
 ui/vnc.c                  | 14 +++----
 25 files changed, 187 insertions(+), 181 deletions(-)

Comments

Markus Armbruster June 17, 2015, 8:01 a.m. UTC | #1 
"Kővágó, Zoltán" <dirty.ice.hu@gmail.com> writes:

> 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.
[...]
> diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
> index 6315b2d..4d38f5d 100644
> --- a/audio/alsaaudio.c
> +++ b/audio/alsaaudio.c
[...]
> @@ -639,19 +639,22 @@ 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;
> +
> +        case AUDIO_FORMAT_MAX:
> +            break;

Can this happen?

>          }
>  
>          threshold = (conf->threshold * bytes_per_sec) / 1000;
> diff --git a/audio/audio.c b/audio/audio.c
> index 5be4b15..112b57b 100644
> --- a/audio/audio.c
> +++ b/audio/audio.c
> @@ -75,7 +75,7 @@ static struct {
>          .settings = {
>              .freq = 44100,
>              .nchannels = 2,
> -            .fmt = AUD_FMT_S16,
> +            .fmt = AUDIO_FORMAT_S16,
>              .endianness =  AUDIO_HOST_ENDIANNESS,
>          }
>      },
> @@ -87,7 +87,7 @@ static struct {
>          .settings = {
>              .freq = 44100,
>              .nchannels = 2,
> -            .fmt = AUD_FMT_S16,
> +            .fmt = AUDIO_FORMAT_S16,
>              .endianness = AUDIO_HOST_ENDIANNESS,
>          }
>      },
> @@ -219,58 +219,61 @@ 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";
> +
> +    case AUDIO_FORMAT_MAX:

default: would be more defensive.  Same elsewhere.

> +        abort();
>      }
>  
>      dolog ("Bogus audfmt %d returning S16\n", fmt);
>      return "S16";
>  }
[...]
=?UTF-8?B?Wm9sdMOhbiBLxZF2w6Fnw7M=?= June 17, 2015, 11:05 a.m. UTC | #2 
2015-06-17 10:01 keltezéssel, Markus Armbruster írta:
> "Kővágó, Zoltán" <dirty.ice.hu@gmail.com> writes:
>
>> 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.
> [...]
>> diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
>> index 6315b2d..4d38f5d 100644
>> --- a/audio/alsaaudio.c
>> +++ b/audio/alsaaudio.c
> [...]
>> @@ -639,19 +639,22 @@ 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;
>> +
>> +        case AUDIO_FORMAT_MAX:
>> +            break;
>
> Can this happen?

Not under normal circumstances, but gcc warns otherwise.

>>           }
>>
>>           threshold = (conf->threshold * bytes_per_sec) / 1000;
>> diff --git a/audio/audio.c b/audio/audio.c
>> index 5be4b15..112b57b 100644
>> --- a/audio/audio.c
>> +++ b/audio/audio.c
>> @@ -75,7 +75,7 @@ static struct {
>>           .settings = {
>>               .freq = 44100,
>>               .nchannels = 2,
>> -            .fmt = AUD_FMT_S16,
>> +            .fmt = AUDIO_FORMAT_S16,
>>               .endianness =  AUDIO_HOST_ENDIANNESS,
>>           }
>>       },
>> @@ -87,7 +87,7 @@ static struct {
>>           .settings = {
>>               .freq = 44100,
>>               .nchannels = 2,
>> -            .fmt = AUD_FMT_S16,
>> +            .fmt = AUDIO_FORMAT_S16,
>>               .endianness = AUDIO_HOST_ENDIANNESS,
>>           }
>>       },
>> @@ -219,58 +219,61 @@ 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";
>> +
>> +    case AUDIO_FORMAT_MAX:
>
> default: would be more defensive.  Same elsewhere.

Ok, I'll change them to default.

>> +        abort();
>>       }
>>
>>       dolog ("Bogus audfmt %d returning S16\n", fmt);
>>       return "S16";
>>   }
> [...]
>
Markus Armbruster June 17, 2015, 11:51 a.m. UTC | #3 
"Kővágó Zoltán" <dirty.ice.hu@gmail.com> writes:

> 2015-06-17 10:01 keltezéssel, Markus Armbruster írta:
>> "Kővágó, Zoltán" <dirty.ice.hu@gmail.com> writes:
>>
>>> 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.
>> [...]
>>> diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
>>> index 6315b2d..4d38f5d 100644
>>> --- a/audio/alsaaudio.c
>>> +++ b/audio/alsaaudio.c
>> [...]
>>> @@ -639,19 +639,22 @@ 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;
>>> +
>>> +        case AUDIO_FORMAT_MAX:
>>> +            break;
>>
>> Can this happen?
>
> Not under normal circumstances, but gcc warns otherwise.

Okay, sounds like another case of "default: abort();" to me :)

[...]
Eric Blake June 17, 2015, 4:01 p.m. UTC | #4 
On 06/17/2015 05:51 AM, Markus Armbruster wrote:

>>>> @@ -639,19 +639,22 @@ 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:

>>>> +
>>>> +        case AUDIO_FORMAT_MAX:
>>>> +            break;
>>>
>>> Can this happen?
>>
>> Not under normal circumstances, but gcc warns otherwise.
> 
> Okay, sounds like another case of "default: abort();" to me :)

gcc has the annoying habit that it can warn you about missing enum
labels in a switch statement, but ONLY if you do not use a 'default:'
label.  Warning about missing labels is nice if you plan to add more
enum values down the road (then the compiler helpfully points out all
the code spots that need to deal with the new value) but with the
drawback that non-enum values bypass the switch completely.  So if you
anticipate this set growing, then s/break/abort()/ is all the more you
need.  On the other hand, if the set is pretty much fixed, then
switching to default: covers more cases including non-enum values
assigned into obt->fmt.
diff mbox

Patch

diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 6315b2d..4d38f5d 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -88,7 +88,7 @@  struct alsa_params_req {
 
 struct alsa_params_obt {
     int freq;
-    audfmt_e fmt;
+    AudioFormat fmt;
     int endianness;
     int nchannels;
     snd_pcm_uframes_t samples;
@@ -295,16 +295,16 @@  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;
         }
@@ -312,7 +312,7 @@  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;
         }
@@ -320,7 +320,7 @@  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;
         }
@@ -328,7 +328,7 @@  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;
         }
@@ -345,58 +345,58 @@  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:
@@ -639,19 +639,22 @@  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;
+
+        case AUDIO_FORMAT_MAX:
+            break;
         }
 
         threshold = (conf->threshold * bytes_per_sec) / 1000;
diff --git a/audio/audio.c b/audio/audio.c
index 5be4b15..112b57b 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -75,7 +75,7 @@  static struct {
         .settings = {
             .freq = 44100,
             .nchannels = 2,
-            .fmt = AUD_FMT_S16,
+            .fmt = AUDIO_FORMAT_S16,
             .endianness =  AUDIO_HOST_ENDIANNESS,
         }
     },
@@ -87,7 +87,7 @@  static struct {
         .settings = {
             .freq = 44100,
             .nchannels = 2,
-            .fmt = AUD_FMT_S16,
+            .fmt = AUDIO_FORMAT_S16,
             .endianness = AUDIO_HOST_ENDIANNESS,
         }
     },
@@ -219,58 +219,61 @@  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";
+
+    case AUDIO_FORMAT_MAX:
+        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",
@@ -280,8 +283,8 @@  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);
@@ -384,7 +387,7 @@  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,
@@ -471,7 +474,7 @@  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;
@@ -502,22 +505,22 @@  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:
@@ -548,12 +551,12 @@  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;
@@ -569,25 +572,28 @@  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;
+
+    case AUDIO_FORMAT_MAX:
+        abort();
     }
     return info->freq == as->freq
         && info->nchannels == as->nchannels
@@ -601,24 +607,27 @@  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;
+
+    case AUDIO_FORMAT_MAX:
+        abort();
     }
 
     info->freq = as->freq;
diff --git a/audio/audio.h b/audio/audio.h
index e7ea397..e300511 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -29,15 +29,6 @@ 
 
 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 +38,7 @@  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 e132405..a8cfa77 100644
--- a/audio/audio_win_int.c
+++ b/audio/audio_win_int.c
@@ -23,20 +23,20 @@  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;
@@ -84,15 +84,15 @@  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 11e76a1..02a3a95 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -72,7 +72,7 @@  typedef struct OSSVoiceIn {
 
 struct oss_params {
     int freq;
-    audfmt_e fmt;
+    AudioFormat fmt;
     int nchannels;
     int nfrags;
     int fragsize;
@@ -150,16 +150,16 @@  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;
         }
@@ -167,7 +167,7 @@  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;
         }
@@ -184,37 +184,37 @@  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:
@@ -502,7 +502,7 @@  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;
 
@@ -671,7 +671,7 @@  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 fea6071..cfdbdc6 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -384,21 +384,21 @@  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:
@@ -409,26 +409,26 @@  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 1140f2e..db0f95a 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -115,19 +115,19 @@  static int sdl_unlock_and_post (SDLAudioState *s, const char *forfn)
     return sdl_post (s, forfn);
 }
 
-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:
@@ -139,37 +139,37 @@  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:
@@ -341,7 +341,7 @@  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 5c6f726..f556b3b 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -127,7 +127,7 @@  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);
@@ -255,7 +255,7 @@  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 c586020..62017de 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -116,20 +116,23 @@  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;
+
+    case AUDIO_FORMAT_MAX:
+        abort();
     }
 
     hdr[34] = bits16 ? 0x10 : 0x08;
@@ -224,7 +227,7 @@  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 6f6d792..b03c244 100644
--- a/audio/wavcapture.c
+++ b/audio/wavcapture.c
@@ -135,7 +135,7 @@  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;
diff --git a/hw/arm/omap2.c b/hw/arm/omap2.c
index e39b317..3b14a5d 100644
--- a/hw/arm/omap2.c
+++ b/hw/arm/omap2.c
@@ -269,7 +269,7 @@  static void omap_eac_format_update(struct omap_eac_s *s)
      * does I2S specify it?  */
     /* All register writes are 16 bits so we we store 16-bit samples
      * in the buffers regardless of AGCFR[B8_16] value.  */
-    fmt.fmt = AUD_FMT_U16;
+    fmt.fmt = AUDIO_FORMAT_U16;
 
     s->codec.in_voice = AUD_open_in(&s->codec.card, s->codec.in_voice,
                     "eac.codec.in", s, omap_eac_in_cb, &fmt);
diff --git a/hw/audio/ac97.c b/hw/audio/ac97.c
index b173835..fa75f33 100644
--- a/hw/audio/ac97.c
+++ b/hw/audio/ac97.c
@@ -360,7 +360,7 @@  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 656eb37..f8f0f55 100644
--- a/hw/audio/adlib.c
+++ b/hw/audio/adlib.c
@@ -323,7 +323,7 @@  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 f96f561..626a173 100644
--- a/hw/audio/cs4231a.c
+++ b/hw/audio/cs4231a.c
@@ -284,7 +284,7 @@  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;
 
@@ -294,7 +294,7 @@  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;
@@ -302,7 +302,7 @@  static void cs_reset_voices (CSState *s, uint32_t val)
     case 6:
         as.endianness = 1;
     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 8e7bcf5..f6e74cb 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 86223a9..6107824 100644
--- a/hw/audio/gus.c
+++ b/hw/audio/gus.c
@@ -242,7 +242,7 @@  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 3c03ff5..8693b7a 100644
--- a/hw/audio/hda-codec.c
+++ b/hw/audio/hda-codec.c
@@ -97,9 +97,9 @@  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;
@@ -128,12 +128,12 @@  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 380ef60..9d4f4b5 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 28f55e8..15169e2 100644
--- a/hw/audio/milkymist-ac97.c
+++ b/hw/audio/milkymist-ac97.c
@@ -297,7 +297,7 @@  static int milkymist_ac97_init(SysBusDevice *dev)
 
     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 5266fb5..302debf 100644
--- a/hw/audio/pcspk.c
+++ b/hw/audio/pcspk.c
@@ -112,7 +112,7 @@  static void pcspk_callback(void *opaque, int free)
 static int pcspk_audio_init(ISABus *bus)
 {
     PCSpkState *s = pcspk_state;
-    struct audsettings as = {PCSPK_SAMPLE_RATE, 1, AUD_FMT_U8, 0};
+    struct audsettings as = {PCSPK_SAMPLE_RATE, 1, AUDIO_FORMAT_U8, 0};
 
     AUD_register_card(s_spk, &s->card);
 
diff --git a/hw/audio/sb16.c b/hw/audio/sb16.c
index b052de5..a159dcc 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;
@@ -221,7 +221,7 @@  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;
@@ -316,18 +316,18 @@  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;
         }
     }
 
@@ -839,7 +839,7 @@  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 b50b331..4c4333c 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);
diff --git a/hw/input/tsc210x.c b/hw/input/tsc210x.c
index fae3385..3cf938b 100644
--- a/hw/input/tsc210x.c
+++ b/hw/input/tsc210x.c
@@ -315,7 +315,7 @@  static void tsc2102_audio_output_update(TSC210xState *s)
     fmt.endianness = 0;
     fmt.nchannels = 2;
     fmt.freq = s->codec.tx_rate;
-    fmt.fmt = AUD_FMT_S16;
+    fmt.fmt = AUDIO_FORMAT_S16;
 
     s->dac_voice[0] = AUD_open_out(&s->card, s->dac_voice[0],
                     "tsc2102.sink", s, (void *) tsc210x_audio_out_cb, &fmt);
diff --git a/hw/usb/dev-audio.c b/hw/usb/dev-audio.c
index f092bb8..0171579 100644
--- a/hw/usb/dev-audio.c
+++ b/hw/usb/dev-audio.c
@@ -646,7 +646,7 @@  static void usb_audio_realize(USBDevice *dev, Error **errp)
     s->out.vol[1]        = 240; /* 0 dB */
     s->out.as.freq       = USBAUDIO_SAMPLE_RATE;
     s->out.as.nchannels  = 2;
-    s->out.as.fmt        = AUD_FMT_S16;
+    s->out.as.fmt        = AUDIO_FORMAT_S16;
     s->out.as.endianness = 0;
     streambuf_init(&s->out.buf, s->buffer);
 
diff --git a/ui/vnc.c b/ui/vnc.c
index 69b605c..42cefb7 100644
--- a/ui/vnc.c
+++ b/ui/vnc.c
@@ -2379,12 +2379,12 @@  static int protocol_client_msg(VncState *vs, uint8_t *data, size_t len)
                 if (len == 4)
                     return 10;
                 switch (read_u8(data, 4)) {
-                case 0: vs->as.fmt = AUD_FMT_U8; break;
-                case 1: vs->as.fmt = AUD_FMT_S8; break;
-                case 2: vs->as.fmt = AUD_FMT_U16; break;
-                case 3: vs->as.fmt = AUD_FMT_S16; break;
-                case 4: vs->as.fmt = AUD_FMT_U32; break;
-                case 5: vs->as.fmt = AUD_FMT_S32; break;
+                case 0: vs->as.fmt = AUDIO_FORMAT_U8; break;
+                case 1: vs->as.fmt = AUDIO_FORMAT_S8; break;
+                case 2: vs->as.fmt = AUDIO_FORMAT_U16; break;
+                case 3: vs->as.fmt = AUDIO_FORMAT_S16; break;
+                case 4: vs->as.fmt = AUDIO_FORMAT_U32; break;
+                case 5: vs->as.fmt = AUDIO_FORMAT_S32; break;
                 default:
                     VNC_DEBUG("Invalid audio format %d\n", read_u8(data, 4));
                     vnc_client_error(vs);
@@ -3065,7 +3065,7 @@  void vnc_init_state(VncState *vs)
 
     vs->as.freq = 44100;
     vs->as.nchannels = 2;
-    vs->as.fmt = AUD_FMT_S16;
+    vs->as.fmt = AUDIO_FORMAT_S16;
     vs->as.endianness = 0;
 
     qemu_mutex_init(&vs->output_mutex);