auto import from //branches/cupcake/...@130745

19 files changed, 317 insertions, 95 deletions

diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index cc43841..1cc4d6e 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -23,13 +23,13 @@
  * THE SOFTWARE.
  */
 #include <alsa/asoundlib.h>
-#include "vl.h"
+#include "audio.h"
 
 #define AUDIO_CAP "alsa"
 #include "audio_int.h"
 #include <dlfcn.h>
 #include <pthread.h>
-#include "android_debug.h"
+#include "qemu_debug.h"
 
 #define  DEBUG  1
 
@@ -59,6 +59,7 @@
     DYNLINK_FUNC(size_t,snd_pcm_hw_params_sizeof,(void))  \
     DYNLINK_FUNC(int,snd_pcm_hw_params_any,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params))  \
     DYNLINK_FUNC(int,snd_pcm_hw_params_set_access,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_access_t _access))  \
+    DYNLINK_FUNC(int,snd_pcm_hw_params_get_format,(const snd_pcm_hw_params_t *params, snd_pcm_format_t *val)) \
     DYNLINK_FUNC(int,snd_pcm_hw_params_set_format,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_format_t val))  \
     DYNLINK_FUNC(int,snd_pcm_hw_params_set_rate_near,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int *val, int *dir))  \
     DYNLINK_FUNC(int,snd_pcm_hw_params_set_channels_near,(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int *val))  \
@@ -194,7 +195,7 @@ static int alsa_write (SWVoiceOut *sw, void *buf, int len)
     return audio_pcm_sw_write (sw, buf, len);
 }
 
-static int aud_to_alsafmt (audfmt_e fmt)
+static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt)
 {
     switch (fmt) {
     case AUD_FMT_S8:
@@ -209,6 +210,12 @@ static int aud_to_alsafmt (audfmt_e fmt)
     case AUD_FMT_U16:
         return SND_PCM_FORMAT_U16_LE;
 
+    case AUD_FMT_S32:
+        return SND_PCM_FORMAT_S32_LE;
+
+    case AUD_FMT_U32:
+        return SND_PCM_FORMAT_U32_LE;
+
     default:
         dolog ("Internal logic error: Bad audio format %d\n", fmt);
 #ifdef DEBUG_AUDIO
@@ -252,6 +259,26 @@ static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt,
         *fmt = AUD_FMT_U16;
         break;
 
+    case SND_PCM_FORMAT_S32_LE:
+        *endianness = 0;
+        *fmt = AUD_FMT_S32;
+        break;
+
+    case SND_PCM_FORMAT_U32_LE:
+        *endianness = 0;
+        *fmt = AUD_FMT_U32;
+        break;
+
+    case SND_PCM_FORMAT_S32_BE:
+        *endianness = 1;
+        *fmt = AUD_FMT_S32;
+        break;
+
+    case SND_PCM_FORMAT_U32_BE:
+        *endianness = 1;
+        *fmt = AUD_FMT_U32;
+        break;
+
     default:
         dolog ("Unrecognized audio format %d\n", alsafmt);
         return -1;
@@ -468,6 +495,7 @@ static int alsa_open (int in, struct alsa_params_req *req,
     }
 
     err = FF(snd_pcm_hw_params_get_format)(hw_params, &obtfmt);
+    err = FF(snd_pcm_hw_params_get_format) (hw_params, &obtfmt);
     if (err < 0) {
         alsa_logerr2 (err, typ, "Failed to get format\n");
         goto err;
@@ -499,6 +527,11 @@ static int alsa_open (int in, struct alsa_params_req *req,
         case AUD_FMT_U16:
             bytes_per_sec <<= 1;
             break;
+
+        case AUD_FMT_S32:
+        case AUD_FMT_U32:
+            bytes_per_sec <<= 2;
+            break;
         }
 
         threshold = (conf.threshold * bytes_per_sec) / 1000;
diff --git a/audio/audio.c b/audio/audio.c
index 4ba1f7d..5a77dac 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -22,12 +22,17 @@
  * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
  * THE SOFTWARE.
  */
-#include "vl.h"
+#include "hw/hw.h"
+#include "audio.h"
+#include "console.h"
+#include "qemu-timer.h"
+#include "sysemu.h"
 
 #define AUDIO_CAP "audio"
 #include "audio_int.h"
-#include "android_utils.h"
-#include "android.h"
+#include "android/utils/system.h"
+#include "qemu_debug.h"
+#include "android/android.h"
 
 /* #define DEBUG_PLIVE */
 /* #define DEBUG_LIVE */
@@ -161,7 +166,8 @@ static struct {
         {
             44100,              /* freq */
             2,                  /* nchannels */
-            AUD_FMT_S16         /* fmt */
+            AUD_FMT_S16,        /* fmt */
+            AUDIO_HOST_ENDIANNESS
         }
     },
 
@@ -172,7 +178,8 @@ static struct {
         {
             44100,              /* freq */
             2,                  /* nchannels */
-            AUD_FMT_S16         /* fmt */
+            AUD_FMT_S16,        /* fmt */
+            AUDIO_HOST_ENDIANNESS
         }
     },
 
@@ -189,8 +196,8 @@ volume_t nominal_volume = {
     1.0,
     1.0
 #else
-    UINT_MAX,
-    UINT_MAX
+    1ULL << 32,
+    1ULL << 32
 #endif
 };
 
@@ -247,6 +254,25 @@ int audio_bug (const char *funcname, int cond)
 }
 #endif
 
+static inline int audio_bits_to_index (int bits)
+{
+    switch (bits) {
+    case 8:
+        return 0;
+
+    case 16:
+        return 1;
+
+    case 32:
+        return 2;
+
+    default:
+        audio_bug ("bits_to_index", 1);
+        AUD_log (NULL, "invalid bits %d\n", bits);
+        return 0;
+    }
+}
+
 void *audio_calloc (const char *funcname, int nmemb, size_t size)
 {
     int cond;
@@ -294,7 +320,7 @@ static char *audio_alloc_prefix (const char *s)
     return r;
 }
 
-const char *audio_audfmt_to_string (audfmt_e fmt)
+static const char *audio_audfmt_to_string (audfmt_e fmt)
 {
     switch (fmt) {
     case AUD_FMT_U8:
@@ -308,13 +334,20 @@ const char *audio_audfmt_to_string (audfmt_e fmt)
 
     case AUD_FMT_S16:
         return "S16";
+
+    case AUD_FMT_U32:
+        return "U32";
+
+    case AUD_FMT_S32:
+        return "S32";
     }
 
     dolog ("Bogus audfmt %d returning S16\n", fmt);
     return "S16";
 }
 
-audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval, int *defaultp)
+static audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval,
+                                        int *defaultp)
 {
     if (!strcasecmp (s, "u8")) {
         *defaultp = 0;
@@ -324,6 +357,10 @@ audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval, int *defaultp)
         *defaultp = 0;
         return AUD_FMT_U16;
     }
+    else if (!strcasecmp (s, "u32")) {
+        *defaultp = 0;
+        return AUD_FMT_U32;
+    }
     else if (!strcasecmp (s, "s8")) {
         *defaultp = 0;
         return AUD_FMT_S8;
@@ -332,6 +369,10 @@ audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval, int *defaultp)
         *defaultp = 0;
         return AUD_FMT_S16;
     }
+    else if (!strcasecmp (s, "s32")) {
+        *defaultp = 0;
+        return AUD_FMT_S32;
+    }
     else {
         dolog ("Bogus audio format `%s' using %s\n",
                s, audio_audfmt_to_string (defval));
@@ -439,7 +480,7 @@ static void audio_print_options (const char *prefix,
         const char *state = "default";
         printf ("  %s_%s: ", uprefix, opt->name);
 
-        if (opt->overridenp && *opt->overridenp) {
+        if (opt->overriddenp && *opt->overriddenp) {
             state = "current";
         }
 
@@ -462,7 +503,7 @@ static void audio_print_options (const char *prefix,
             {
                 audfmt_e *fmtp = opt->valp;
                 printf (
-                    "format, %s = %s, (one of: U8 S8 U16 S16)\n",
+                    "format, %s = %s, (one of: U8 S8 U16 S16 U32 S32)\n",
                     state,
                     audio_audfmt_to_string (*fmtp)
                     );
@@ -569,10 +610,10 @@ static void audio_process_options (const char *prefix,
             break;
         }
 
-        if (!opt->overridenp) {
-            opt->overridenp = &opt->overriden;
+        if (!opt->overriddenp) {
+            opt->overriddenp = &opt->overridden;
         }
-        *opt->overridenp = !def;
+        *opt->overriddenp = !def;
         qemu_free (optname);
     }
 }
@@ -626,6 +667,8 @@ static int audio_validate_settings (audsettings_t *as)
     case AUD_FMT_U8:
     case AUD_FMT_S16:
     case AUD_FMT_U16:
+    case AUD_FMT_S32:
+    case AUD_FMT_U32:
         break;
     default:
         invalid = 1;
@@ -651,6 +694,11 @@ static int audio_pcm_info_eq (struct audio_pcm_info *info, audsettings_t *as)
     case AUD_FMT_U16:
         bits = 16;
         break;
+    case AUD_FMT_S32:
+        sign = 1;
+    case AUD_FMT_U32:
+        bits = 32;
+        break;
     }
     return info->freq == as->freq
         && info->nchannels == as->nchannels
@@ -661,7 +709,7 @@ static int audio_pcm_info_eq (struct audio_pcm_info *info, audsettings_t *as)
 
 void audio_pcm_init_info (struct audio_pcm_info *info, audsettings_t *as)
 {
-    int bits = 8, sign = 0;
+    int bits = 8, sign = 0, shift = 0;
 
     switch (as->fmt) {
     case AUD_FMT_S8:
@@ -673,6 +721,14 @@ void audio_pcm_init_info (struct audio_pcm_info *info, audsettings_t *as)
         sign = 1;
     case AUD_FMT_U16:
         bits = 16;
+        shift = 1;
+        break;
+
+    case AUD_FMT_S32:
+        sign = 1;
+    case AUD_FMT_U32:
+        bits = 32;
+        shift = 2;
         break;
     }
 
@@ -680,7 +736,7 @@ void audio_pcm_init_info (struct audio_pcm_info *info, audsettings_t *as)
     info->bits = bits;
     info->sign = sign;
     info->nchannels = as->nchannels;
-    info->shift = (as->nchannels == 2) + (bits == 16);
+    info->shift = (as->nchannels == 2) + shift;
     info->align = (1 << info->shift) - 1;
     info->bytes_per_second = info->freq << info->shift;
     info->swap_endianness = (as->endianness != AUDIO_HOST_ENDIANNESS);
@@ -693,25 +749,52 @@ void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
     }
 
     if (info->sign) {
-        memset (buf, 0x80, len << info->shift);
+        memset (buf, 0x00, len << info->shift);
     }
     else {
-        if (info->bits == 8) {
+        switch (info->bits) {
+        case 8:
             memset (buf, 0x80, len << info->shift);
-        }
-        else {
-            int i;
-            uint16_t *p = buf;
-            int shift = info->nchannels - 1;
-            short s = INT16_MAX;
+            break;
 
-            if (info->swap_endianness) {
-                s = bswap16 (s);
+        case 16:
+            {
+                int i;
+                uint16_t *p = buf;
+                int shift = info->nchannels - 1;
+                short s = INT16_MAX;
+
+                if (info->swap_endianness) {
+                    s = bswap16 (s);
+                }
+
+                for (i = 0; i < len << shift; i++) {
+                    p[i] = s;
+                }
             }
+            break;
 
-            for (i = 0; i < len << shift; i++) {
-                p[i] = s;
+        case 32:
+            {
+                int i;
+                uint32_t *p = buf;
+                int shift = info->nchannels - 1;
+                int32_t s = INT32_MAX;
+
+                if (info->swap_endianness) {
+                    s = bswap32 (s);
+                }
+
+                for (i = 0; i < len << shift; i++) {
+                    p[i] = s;
+                }
             }
+            break;
+
+        default:
+            AUD_log (NULL, "audio_pcm_info_clear_buf: invalid bits %d\n",
+                     info->bits);
+            break;
         }
     }
 }
@@ -1508,7 +1591,7 @@ static void audio_run_capture (AudioState *s)
 
 static void audio_timer (void *opaque)
 {
-    AudioState*     s = opaque;
+    AudioState* s = opaque;
 #if 0
 #define  MAX_DIFFS  1000
     int64_t         now  = qemu_get_clock(vm_clock);
@@ -1586,7 +1669,7 @@ static struct audio_option audio_options[] = {
      "(undocumented)", NULL, 0},
 
     {"LOG_TO_MONITOR", AUD_OPT_BOOL, &conf.log_to_monitor,
-     "print logging messages to montior instead of stderr", NULL, 0},
+     "print logging messages to monitor instead of stderr", NULL, 0},
 
     {NULL, 0, NULL, NULL, NULL, 0}
 };
@@ -2011,7 +2094,7 @@ CaptureVoiceOut *AUD_add_capture (
             [hw->info.nchannels == 2]
             [hw->info.sign]
             [hw->info.swap_endianness]
-            [hw->info.bits == 16];
+            [audio_bits_to_index (hw->info.bits)];
 
         LIST_INSERT_HEAD (&s->cap_head, cap, entries);
         LIST_INSERT_HEAD (&cap->cb_head, cb, entries);
@@ -2045,16 +2128,21 @@ void AUD_del_capture (CaptureVoiceOut *cap, void *cb_opaque)
 
             if (!cap->cb_head.lh_first) {
                 SWVoiceOut *sw = cap->hw.sw_head.lh_first, *sw1;
+
                 while (sw) {
+                    SWVoiceCap *sc = (SWVoiceCap *) sw;
 #ifdef DEBUG_CAPTURE
                     dolog ("freeing %s\n", sw->name);
 #endif
+
                     sw1 = sw->entries.le_next;
                     if (sw->rate) {
                         st_rate_stop (sw->rate);
                         sw->rate = NULL;
                     }
                     LIST_REMOVE (sw, entries);
+                    LIST_REMOVE (sc, entries);
+                    qemu_free (sc);
                     sw = sw1;
                 }
                 LIST_REMOVE (cap, entries);
@@ -2064,3 +2152,21 @@ void AUD_del_capture (CaptureVoiceOut *cap, void *cb_opaque)
         }
     }
 }
+
+void AUD_set_volume_out (SWVoiceOut *sw, int mute, uint8_t lvol, uint8_t rvol)
+{
+    if (sw) {
+        sw->vol.mute = mute;
+        sw->vol.l = nominal_volume.l * lvol / 255;
+        sw->vol.r = nominal_volume.r * rvol / 255;
+    }
+}
+
+void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
+{
+    if (sw) {
+        sw->vol.mute = mute;
+        sw->vol.l = nominal_volume.l * lvol / 255;
+        sw->vol.r = nominal_volume.r * rvol / 255;
+    }
+}
diff --git a/audio/audio.h b/audio/audio.h
index cc7aa04..2c347bf 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -25,6 +25,7 @@
 #define QEMU_AUDIO_H
 
 #include "config.h"
+#include "qemu-common.h"
 #include "sys-queue.h"
 
 typedef void (*audio_callback_fn_t) (void *opaque, int avail);
@@ -33,7 +34,9 @@ typedef enum {
     AUD_FMT_U8,
     AUD_FMT_S8,
     AUD_FMT_U16,
-    AUD_FMT_S16
+    AUD_FMT_S16,
+    AUD_FMT_U32,
+    AUD_FMT_S32
 } audfmt_e;
 
 #ifdef WORDS_BIGENDIAN
@@ -71,7 +74,6 @@ typedef struct CaptureState {
     LIST_ENTRY (CaptureState) entries;
 } CaptureState;
 
-typedef struct AudioState AudioState;
 typedef struct SWVoiceOut SWVoiceOut;
 typedef struct CaptureVoiceOut CaptureVoiceOut;
 typedef struct SWVoiceIn SWVoiceIn;
@@ -125,6 +127,9 @@ int  AUD_is_active_out (SWVoiceOut *sw);
 void     AUD_init_time_stamp_out (SWVoiceOut *sw, QEMUAudioTimeStamp *ts);
 uint64_t AUD_get_elapsed_usec_out (SWVoiceOut *sw, QEMUAudioTimeStamp *ts);
 
+void AUD_set_volume_out (SWVoiceOut *sw, int mute, uint8_t lvol, uint8_t rvol);
+void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol);
+
 SWVoiceIn *AUD_open_in (
     QEMUSoundCard *card,
     SWVoiceIn *sw,
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 099bcbd..6677ec1 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -24,6 +24,8 @@
 #ifndef QEMU_AUDIO_INT_H
 #define QEMU_AUDIO_INT_H
 
+#include "audio/audio.h"
+
 #ifdef CONFIG_COREAUDIO
 #define FLOAT_MIXENG
 /* #define RECIPROCAL */
@@ -44,8 +46,8 @@ struct audio_option {
     audio_option_tag_e tag;
     void *valp;
     const char *descr;
-    int *overridenp;
-    int overriden;
+    int *overriddenp;
+    int overridden;
 };
 
 struct audio_callback {
@@ -206,6 +208,7 @@ extern struct audio_driver esd_audio_driver;
 extern struct audio_driver alsa_audio_driver;
 extern struct audio_driver coreaudio_audio_driver;
 extern struct audio_driver dsound_audio_driver;
+extern struct audio_driver esd_audio_driver;
 extern volume_t nominal_volume;
 
 void audio_pcm_init_info (struct audio_pcm_info *info, audsettings_t *as);
diff --git a/audio/audio_pt_int.c b/audio/audio_pt_int.c
index 3f01762..c753774 100644
--- a/audio/audio_pt_int.c
+++ b/audio/audio_pt_int.c
@@ -1,4 +1,3 @@
-#include "vl.h"
 #include "audio.h"
 
 #define AUDIO_CAP "audio-pt"
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 4575a56..6f8e166 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -164,7 +164,7 @@ static int glue (audio_pcm_sw_init_, TYPE) (
         [sw->info.nchannels == 2]
         [sw->info.sign]
         [sw->info.swap_endianness]
-        [sw->info.bits == 16];
+        [audio_bits_to_index (sw->info.bits)];
 
     sw->name = qemu_strdup (name);
     err = glue (audio_pcm_sw_alloc_resources_, TYPE) (sw);
@@ -293,7 +293,7 @@ static HW *glue (audio_pcm_hw_add_new_, TYPE) (AudioState *s, audsettings_t *as)
         [hw->info.nchannels == 2]
         [hw->info.sign]
         [hw->info.swap_endianness]
-        [hw->info.bits == 16];
+        [audio_bits_to_index (hw->info.bits)];
 
     if (glue (audio_pcm_hw_alloc_resources_, TYPE) (hw)) {
         goto err1;
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index cf3eaf9..f23ebee 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -27,7 +27,7 @@
 #include <string.h>             /* strerror */
 #include <pthread.h>            /* pthread_X */
 
-#include "vl.h"
+#include "audio.h"
 
 #define AUDIO_CAP "coreaudio"
 #include "audio_int.h"
@@ -628,7 +628,6 @@ coreaudio_run_in (HWVoiceIn *hw)
     if (coreaudio_voice_lock (core, "coreaudio_run_in")) {
         return 0;
     }
-
     D("%s: core.decr=%d core.pos=%d\n", __FUNCTION__, core->decr, core->pos);
     decr        = core->decr;
     core->decr -= decr;
diff --git a/audio/dsound_template.h b/audio/dsound_template.h
index 0896b04..9cc0b9d 100644
--- a/audio/dsound_template.h
+++ b/audio/dsound_template.h
@@ -23,16 +23,20 @@
  */
 #ifdef DSBTYPE_IN
 #define NAME "capture buffer"
+#define NAME2 "DirectSoundCapture"
 #define TYPE in
 #define IFACE IDirectSoundCaptureBuffer
 #define BUFPTR LPDIRECTSOUNDCAPTUREBUFFER
 #define FIELD dsound_capture_buffer
+#define FIELD2 dsound_capture
 #else
 #define NAME "playback buffer"
+#define NAME2 "DirectSound"
 #define TYPE out
 #define IFACE IDirectSoundBuffer
 #define BUFPTR LPDIRECTSOUNDBUFFER
 #define FIELD dsound_buffer
+#define FIELD2 dsound
 #endif
 
 static int glue (dsound_unlock_, TYPE) (
@@ -192,6 +196,11 @@ static int dsound_init_out (HWVoiceOut *hw, audsettings_t *as)
     DSBCAPS bc;
 #endif
 
+    if (!s->FIELD2) {
+        dolog ("Attempt to initialize voice without " NAME2 " object\n");
+        return -1;
+    }
+
     err = waveformat_from_audio_settings (&wfx, as);
     if (err) {
         return -1;
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index 6104acb..8284067 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -26,12 +26,14 @@
  * SEAL 1.07 by Carlos 'pel' Hasan was used as documentation
  */
 
-#include "vl.h"
+#include "audio.h"
 
 #define AUDIO_CAP "dsound"
 #include "audio_int.h"
 
+#define WIN32_LEAN_AND_MEAN
 #include <windows.h>
+#include <mmsystem.h>
 #include <objbase.h>
 #include <dsound.h>
 
@@ -317,23 +319,22 @@ static int waveformat_from_audio_settings (WAVEFORMATEX *wfx, audsettings_t *as)
 
     switch (as->fmt) {
     case AUD_FMT_S8:
-        wfx->wBitsPerSample = 8;
-        break;
-
     case AUD_FMT_U8:
         wfx->wBitsPerSample = 8;
         break;
 
     case AUD_FMT_S16:
+    case AUD_FMT_U16:
         wfx->wBitsPerSample = 16;
         wfx->nAvgBytesPerSec <<= 1;
         wfx->nBlockAlign <<= 1;
         break;
 
-    case AUD_FMT_U16:
-        wfx->wBitsPerSample = 16;
-        wfx->nAvgBytesPerSec <<= 1;
-        wfx->nBlockAlign <<= 1;
+    case AUD_FMT_S32:
+    case AUD_FMT_U32:
+        wfx->wBitsPerSample = 32;
+        wfx->nAvgBytesPerSec <<= 2;
+        wfx->nBlockAlign <<= 2;
         break;
 
     default:
@@ -384,8 +385,13 @@ static int waveformat_to_audio_settings (WAVEFORMATEX *wfx, audsettings_t *as)
         as->fmt = AUD_FMT_S16;
         break;
 
+    case 32:
+        as->fmt = AUD_FMT_S32;
+        break;
+
     default:
-        dolog ("Invalid wave format, bits per sample is not 8 or 16, but %d\n",
+        dolog ("Invalid wave format, bits per sample is not "
+               "8, 16 or 32, but %d\n",
                wfx->wBitsPerSample);
         return -1;
     }
diff --git a/audio/esdaudio.c b/audio/esdaudio.c
index 35d6b45..05c030f 100644
--- a/audio/esdaudio.c
+++ b/audio/esdaudio.c
@@ -24,7 +24,6 @@
  */
 
 #include <esd.h>
-#include "vl.h"
 #include "audio.h"
 #include <signal.h>
 
@@ -33,7 +32,7 @@
 #include "audio_pt_int.h"
 #include <dlfcn.h>
 
-#include "android_debug.h"
+#include "qemu_debug.h"
 
 #define  DEBUG  1
 
@@ -494,14 +493,12 @@ static int qesd_init_in (HWVoiceIn *hw, audsettings_t *as)
         esdfmt |= ESD_BITS16;
         obt_as.fmt = AUD_FMT_S16;
         break;
-#if 0
     case AUD_FMT_S32:
     case AUD_FMT_U32:
         dolog ("Will use 16 instead of 32 bit samples\n");
         esdfmt |= ESD_BITS16;
         obt_as.fmt = AUD_FMT_S16;
         break;
-#endif
     }
     obt_as.endianness = AUDIO_HOST_ENDIANNESS;
 
diff --git a/audio/fmodaudio.c b/audio/fmodaudio.c
index 5875ba1..e230e8b 100644
--- a/audio/fmodaudio.c
+++ b/audio/fmodaudio.c
@@ -23,7 +23,7 @@
  */
 #include <fmod.h>
 #include <fmod_errors.h>
-#include "vl.h"
+#include "audio.h"
 
 #define AUDIO_CAP "fmod"
 #include "audio_int.h"
diff --git a/audio/mixeng.c b/audio/mixeng.c
index 6308d41..34fc6df 100644
--- a/audio/mixeng.c
+++ b/audio/mixeng.c
@@ -22,7 +22,7 @@
  * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
  * THE SOFTWARE.
  */
-#include "vl.h"
+#include "audio.h"
 
 #define AUDIO_CAP "mixeng"
 #include "audio_int.h"
@@ -82,6 +82,7 @@
 #undef IN_T
 #undef SHIFT
 
+/* Unsigned 16 bit */
 #define IN_T uint16_t
 #define IN_MIN 0
 #define IN_MAX USHRT_MAX
@@ -101,26 +102,72 @@
 #undef IN_T
 #undef SHIFT
 
-t_sample *mixeng_conv[2][2][2][2] = {
+/* Signed 32 bit */
+#define IN_T int32_t
+#define IN_MIN INT32_MIN
+#define IN_MAX INT32_MAX
+#define SIGNED
+#define SHIFT 32
+#define ENDIAN_CONVERSION natural
+#define ENDIAN_CONVERT(v) (v)
+#include "mixeng_template.h"
+#undef ENDIAN_CONVERT
+#undef ENDIAN_CONVERSION
+#define ENDIAN_CONVERSION swap
+#define ENDIAN_CONVERT(v) bswap32 (v)
+#include "mixeng_template.h"
+#undef ENDIAN_CONVERT
+#undef ENDIAN_CONVERSION
+#undef SIGNED
+#undef IN_MAX
+#undef IN_MIN
+#undef IN_T
+#undef SHIFT
+
+/* Unsigned 16 bit */
+#define IN_T uint32_t
+#define IN_MIN 0
+#define IN_MAX UINT32_MAX
+#define SHIFT 32
+#define ENDIAN_CONVERSION natural
+#define ENDIAN_CONVERT(v) (v)
+#include "mixeng_template.h"
+#undef ENDIAN_CONVERT
+#undef ENDIAN_CONVERSION
+#define ENDIAN_CONVERSION swap
+#define ENDIAN_CONVERT(v) bswap32 (v)
+#include "mixeng_template.h"
+#undef ENDIAN_CONVERT
+#undef ENDIAN_CONVERSION
+#undef IN_MAX
+#undef IN_MIN
+#undef IN_T
+#undef SHIFT
+
+t_sample *mixeng_conv[2][2][2][3] = {
     {
         {
             {
                 conv_natural_uint8_t_to_mono,
-                conv_natural_uint16_t_to_mono
+                conv_natural_uint16_t_to_mono,
+                conv_natural_uint32_t_to_mono
             },
             {
                 conv_natural_uint8_t_to_mono,
-                conv_swap_uint16_t_to_mono
+                conv_swap_uint16_t_to_mono,
+                conv_swap_uint32_t_to_mono,
             }
         },
         {
             {
                 conv_natural_int8_t_to_mono,
-                conv_natural_int16_t_to_mono
+                conv_natural_int16_t_to_mono,
+                conv_natural_int32_t_to_mono
             },
             {
                 conv_natural_int8_t_to_mono,
-                conv_swap_int16_t_to_mono
+                conv_swap_int16_t_to_mono,
+                conv_swap_int32_t_to_mono
             }
         }
     },
@@ -128,46 +175,54 @@ t_sample *mixeng_conv[2][2][2][2] = {
         {
             {
                 conv_natural_uint8_t_to_stereo,
-                conv_natural_uint16_t_to_stereo
+                conv_natural_uint16_t_to_stereo,
+                conv_natural_uint32_t_to_stereo
             },
             {
                 conv_natural_uint8_t_to_stereo,
-                conv_swap_uint16_t_to_stereo
+                conv_swap_uint16_t_to_stereo,
+                conv_swap_uint32_t_to_stereo
             }
         },
         {
             {
                 conv_natural_int8_t_to_stereo,
-                conv_natural_int16_t_to_stereo
+                conv_natural_int16_t_to_stereo,
+                conv_natural_int32_t_to_stereo
             },
             {
                 conv_natural_int8_t_to_stereo,
-                conv_swap_int16_t_to_stereo
+                conv_swap_int16_t_to_stereo,
+                conv_swap_int32_t_to_stereo,
             }
         }
     }
 };
 
-f_sample *mixeng_clip[2][2][2][2] = {
+f_sample *mixeng_clip[2][2][2][3] = {
     {
         {
             {
                 clip_natural_uint8_t_from_mono,
-                clip_natural_uint16_t_from_mono
+                clip_natural_uint16_t_from_mono,
+                clip_natural_uint32_t_from_mono
             },
             {
                 clip_natural_uint8_t_from_mono,
-                clip_swap_uint16_t_from_mono
+                clip_swap_uint16_t_from_mono,
+                clip_swap_uint32_t_from_mono
             }
         },
         {
             {
                 clip_natural_int8_t_from_mono,
-                clip_natural_int16_t_from_mono
+                clip_natural_int16_t_from_mono,
+                clip_natural_int32_t_from_mono
             },
             {
                 clip_natural_int8_t_from_mono,
-                clip_swap_int16_t_from_mono
+                clip_swap_int16_t_from_mono,
+                clip_swap_int32_t_from_mono
             }
         }
     },
@@ -175,21 +230,25 @@ f_sample *mixeng_clip[2][2][2][2] = {
         {
             {
                 clip_natural_uint8_t_from_stereo,
-                clip_natural_uint16_t_from_stereo
+                clip_natural_uint16_t_from_stereo,
+                clip_natural_uint32_t_from_stereo
             },
             {
                 clip_natural_uint8_t_from_stereo,
-                clip_swap_uint16_t_from_stereo
+                clip_swap_uint16_t_from_stereo,
+                clip_swap_uint32_t_from_stereo
             }
         },
         {
             {
                 clip_natural_int8_t_from_stereo,
-                clip_natural_int16_t_from_stereo
+                clip_natural_int16_t_from_stereo,
+                clip_natural_int32_t_from_stereo
             },
             {
                 clip_natural_int8_t_from_stereo,
-                clip_swap_int16_t_from_stereo
+                clip_swap_int16_t_from_stereo,
+                clip_swap_int32_t_from_stereo
             }
         }
     }
diff --git a/audio/mixeng.h b/audio/mixeng.h
index 9e3bac1..95b68df 100644
--- a/audio/mixeng.h
+++ b/audio/mixeng.h
@@ -37,8 +37,8 @@ typedef void (t_sample) (st_sample_t *dst, const void *src,
                          int samples, volume_t *vol);
 typedef void (f_sample) (void *dst, const st_sample_t *src, int samples);
 
-extern t_sample *mixeng_conv[2][2][2][2];
-extern f_sample *mixeng_clip[2][2][2][2];
+extern t_sample *mixeng_conv[2][2][2][3];
+extern f_sample *mixeng_clip[2][2][2][3];
 
 void *st_rate_start (int inrate, int outrate);
 void st_rate_flow (void *opaque, st_sample_t *ibuf, st_sample_t *obuf,
diff --git a/audio/noaudio.c b/audio/noaudio.c
index 3567e9c..8788a41 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -21,7 +21,7 @@
  * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
  * THE SOFTWARE.
  */
-#include "vl.h"
+#include "qemu-timer.h"
 
 #define AUDIO_CAP "noaudio"
 #include "audio_int.h"
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 17d676a..2ccaade 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -25,7 +25,6 @@
 #include <sys/types.h>
 #include <sys/ioctl.h>
 #include <sys/soundcard.h>
-#include "vl.h"
 
 #define AUDIO_CAP "oss"
 #include "audio_int.h"
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index eb353d7..ea5bccd 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -23,7 +23,6 @@
  */
 #include <SDL.h>
 #include <SDL_thread.h>
-#include "vl.h"
 
 #ifndef _WIN32
 #ifdef __sun__
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 7d4fed1..8a500b9 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -22,10 +22,10 @@
  * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
  * THE SOFTWARE.
  */
-#include "vl.h"
-
 #define AUDIO_CAP "wav"
+#include "qemu-timer.h"
 #include "audio_int.h"
+#include "qemu_file.h"
 
 #define  WAV_AUDIO_IN  1
 
@@ -46,7 +46,8 @@ static struct {
     {
         44100,
         2,
-        AUD_FMT_S16
+        AUD_FMT_S16,
+        0
     },
     "qemu.wav"
 };
@@ -136,6 +137,11 @@ static int wav_out_init (HWVoiceOut *hw, audsettings_t *as)
     case AUD_FMT_U16:
         bits16 = 1;
         break;
+
+    case AUD_FMT_S32:
+    case AUD_FMT_U32:
+        dolog ("WAVE files can not handle 32bit formats\n");
+        return -1;
     }
 
     hdr[34] = bits16 ? 0x10 : 0x08;
@@ -156,7 +162,7 @@ static int wav_out_init (HWVoiceOut *hw, audsettings_t *as)
     le_store (hdr + 28, hw->info.freq << (bits16 + stereo), 4);
     le_store (hdr + 32, 1 << (bits16 + stereo), 2);
 
-    wav->f = fopen (conf_out.wav_path, "wb");
+    wav->f = qemu_fopen (conf_out.wav_path, "wb");
     if (!wav->f) {
         dolog ("Failed to open wave file `%s'\nReason: %s\n",
                conf_out.wav_path, strerror (errno));
@@ -190,7 +196,7 @@ static void wav_out_fini (HWVoiceOut *hw)
     qemu_fseek (wav->f, 32, SEEK_CUR);
     qemu_put_buffer (wav->f, dlen, 4);
 
-    fclose (wav->f);
+    qemu_fclose (wav->f);
     wav->f = NULL;
 
     qemu_free (wav->pcm_buf);
@@ -247,7 +253,7 @@ wav_in_init (HWVoiceIn *hw, audsettings_t *as)
     audsettings_t wav_as = *as;
     int           nchannels, freq, format, bits;
 
-    wav->f = fopen (path, "rb");
+    wav->f = qemu_fopen (path, "rb");
     if (wav->f == NULL) {
         dolog("Failed to open wave file '%s'\nReason: %s\n", path,
               strerror(errno));
@@ -316,7 +322,7 @@ wav_in_init (HWVoiceIn *hw, audsettings_t *as)
     return 0;
 
 Fail:
-    fclose (wav->f);
+    qemu_fclose (wav->f);
     wav->f = NULL;
     return -1;
 }
@@ -330,7 +336,7 @@ static void wav_in_fini (HWVoiceIn *hw)
         return;
     }
 
-    fclose (wav->f);
+    qemu_fclose (wav->f);
     wav->f = NULL;
 
     qemu_free (wav->pcm_buf);
diff --git a/audio/wavcapture.c b/audio/wavcapture.c
index 748b580..d6f733e 100644
--- a/audio/wavcapture.c
+++ b/audio/wavcapture.c
@@ -1,4 +1,6 @@
-#include "vl.h"
+#include "audio/audio.h"
+#include "qemu_file.h"
+#include "console.h"
 
 typedef struct {
     QEMUFile *f;
@@ -46,7 +48,7 @@ static void wav_destroy (void *opaque)
 
     qemu_fseek (wav->f, 32, SEEK_CUR);
     qemu_put_buffer (wav->f, dlen, 4);
-    fclose (wav->f);
+    qemu_fclose (wav->f);
     if (wav->path) {
         qemu_free (wav->path);
     }
@@ -133,7 +135,7 @@ int wav_start_capture (CaptureState *s, const char *path, int freq,
     le_store (hdr + 28, freq << shift, 4);
     le_store (hdr + 32, 1 << shift, 2);
 
-    wav->f = fopen (path, "wb");
+    wav->f = qemu_fopen (path, "wb");
     if (!wav->f) {
         term_printf ("Failed to open wave file `%s'\nReason: %s\n",
                      path, strerror (errno));
@@ -151,6 +153,8 @@ int wav_start_capture (CaptureState *s, const char *path, int freq,
     cap = AUD_add_capture (NULL, &as, &ops, wav);
     if (!cap) {
         term_printf ("Failed to add audio capture\n");
+        qemu_free (wav->path);
+        qemu_fclose (wav->f);
         qemu_free (wav);
         return -1;
     }
diff --git a/audio/winaudio.c b/audio/winaudio.c
index 63cd351..6e8daef 100644
--- a/audio/winaudio.c
+++ b/audio/winaudio.c
@@ -1,7 +1,7 @@
 /*

  * QEMU "simple" Windows audio driver

  *

- * Copyright (c) 2007 The Android Open Source Project
+ * Copyright (c) 2007 The Android Open Source Project

  *

  * Permission is hereby granted, free of charge, to any person obtaining a copy

  * of this software and associated documentation files (the "Software"), to deal

@@ -21,8 +21,6 @@
  * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN

  * THE SOFTWARE.

  */

-#include "vl.h"

-

 #define WIN32_LEAN_AND_MEAN

 #include <windows.h>

 #include <mmsystem.h>