Merge remote-tracking branch 'remotes/kraxel/tags/audio-20200207-pull-request' into staging

audio: bugfixes, mostly audio backend rewrite fallout

# gpg: Signature made Fri 07 Feb 2020 07:45:44 GMT
# gpg:                using RSA key 4CB6D8EED3E87138
# gpg: Good signature from "Gerd Hoffmann (work) <kraxel@redhat.com>" [full]
# gpg:                 aka "Gerd Hoffmann <gerd@kraxel.org>" [full]
# gpg:                 aka "Gerd Hoffmann (private) <kraxel@gmail.com>" [full]
# Primary key fingerprint: A032 8CFF B93A 17A7 9901  FE7D 4CB6 D8EE D3E8 7138

* remotes/kraxel/tags/audio-20200207-pull-request:
  audio: proper support for float samples in mixeng
  coreaudio: fix coreaudio playback
  audio/dsound: fix invalid parameters error
  audio: audio_generic_get_buffer_in should honor *size
  ossaudio: disable poll mode can't be reached
  ossaudio: prevent SIGSEGV in oss_enable_out
  audio: fix bug 1858488
  audio: prevent SIGSEGV in AUD_get_buffer_size_out
  paaudio: remove unused variables
  audio: fix audio_generic_read
  audio: fix audio_generic_write
  audio/oss: fix buffer pos calculation

Signed-off-by: Peter Maydell <peter.maydell@linaro.org>

15 files changed, 299 insertions, 124 deletions

diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index f37ce1c..a23a5a0 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -307,6 +307,13 @@ static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness)
             return SND_PCM_FORMAT_U32_LE;
         }
 
+    case AUDIO_FORMAT_F32:
+        if (endianness) {
+            return SND_PCM_FORMAT_FLOAT_BE;
+        } else {
+            return SND_PCM_FORMAT_FLOAT_LE;
+        }
+
     default:
         dolog ("Internal logic error: Bad audio format %d\n", fmt);
 #ifdef DEBUG_AUDIO
@@ -370,6 +377,16 @@ static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt,
         *fmt = AUDIO_FORMAT_U32;
         break;
 
+    case SND_PCM_FORMAT_FLOAT_LE:
+        *endianness = 0;
+        *fmt = AUDIO_FORMAT_F32;
+        break;
+
+    case SND_PCM_FORMAT_FLOAT_BE:
+        *endianness = 1;
+        *fmt = AUDIO_FORMAT_F32;
+        break;
+
     default:
         dolog ("Unrecognized audio format %d\n", alsafmt);
         return -1;
@@ -906,6 +923,7 @@ static struct audio_pcm_ops alsa_pcm_ops = {
     .init_out = alsa_init_out,
     .fini_out = alsa_fini_out,
     .write    = alsa_write,
+    .run_buffer_out = audio_generic_run_buffer_out,
     .enable_out = alsa_enable_out,
 
     .init_in  = alsa_init_in,
diff --git a/audio/audio.c b/audio/audio.c
index f63f397..9ac9a20 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -218,6 +218,9 @@ static void audio_print_settings (struct audsettings *as)
     case AUDIO_FORMAT_U32:
         AUD_log (NULL, "U32");
         break;
+    case AUDIO_FORMAT_F32:
+        AUD_log (NULL, "F32");
+        break;
     default:
         AUD_log (NULL, "invalid(%d)", as->fmt);
         break;
@@ -252,6 +255,7 @@ static int audio_validate_settings (struct audsettings *as)
     case AUDIO_FORMAT_U16:
     case AUDIO_FORMAT_S32:
     case AUDIO_FORMAT_U32:
+    case AUDIO_FORMAT_F32:
         break;
     default:
         invalid = 1;
@@ -264,24 +268,28 @@ static int audio_validate_settings (struct audsettings *as)
 
 static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *as)
 {
-    int bits = 8, sign = 0;
+    int bits = 8;
+    bool is_signed = false, is_float = false;
 
     switch (as->fmt) {
     case AUDIO_FORMAT_S8:
-        sign = 1;
+        is_signed = true;
         /* fall through */
     case AUDIO_FORMAT_U8:
         break;
 
     case AUDIO_FORMAT_S16:
-        sign = 1;
+        is_signed = true;
         /* fall through */
     case AUDIO_FORMAT_U16:
         bits = 16;
         break;
 
+    case AUDIO_FORMAT_F32:
+        is_float = true;
+        /* fall through */
     case AUDIO_FORMAT_S32:
-        sign = 1;
+        is_signed = true;
         /* fall through */
     case AUDIO_FORMAT_U32:
         bits = 32;
@@ -292,33 +300,38 @@ static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *a
     }
     return info->freq == as->freq
         && info->nchannels == as->nchannels
-        && info->sign == sign
+        && info->is_signed == is_signed
+        && info->is_float == is_float
         && info->bits == bits
         && info->swap_endianness == (as->endianness != AUDIO_HOST_ENDIANNESS);
 }
 
 void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
 {
-    int bits = 8, sign = 0, mul;
+    int bits = 8, mul;
+    bool is_signed = false, is_float = false;
 
     switch (as->fmt) {
     case AUDIO_FORMAT_S8:
-        sign = 1;
+        is_signed = true;
         /* fall through */
     case AUDIO_FORMAT_U8:
         mul = 1;
         break;
 
     case AUDIO_FORMAT_S16:
-        sign = 1;
+        is_signed = true;
         /* fall through */
     case AUDIO_FORMAT_U16:
         bits = 16;
         mul = 2;
         break;
 
+    case AUDIO_FORMAT_F32:
+        is_float = true;
+        /* fall through */
     case AUDIO_FORMAT_S32:
-        sign = 1;
+        is_signed = true;
         /* fall through */
     case AUDIO_FORMAT_U32:
         bits = 32;
@@ -331,7 +344,8 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
 
     info->freq = as->freq;
     info->bits = bits;
-    info->sign = sign;
+    info->is_signed = is_signed;
+    info->is_float = is_float;
     info->nchannels = as->nchannels;
     info->bytes_per_frame = as->nchannels * mul;
     info->bytes_per_second = info->freq * info->bytes_per_frame;
@@ -344,7 +358,7 @@ void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
         return;
     }
 
-    if (info->sign) {
+    if (info->is_signed || info->is_float) {
         memset(buf, 0x00, len * info->bytes_per_frame);
     }
     else {
@@ -770,8 +784,9 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
 #ifdef DEBUG_AUDIO
 static void audio_pcm_print_info (const char *cap, struct audio_pcm_info *info)
 {
-    dolog ("%s: bits %d, sign %d, freq %d, nchan %d\n",
-           cap, info->bits, info->sign, info->freq, info->nchannels);
+    dolog("%s: bits %d, sign %d, float %d, freq %d, nchan %d\n",
+          cap, info->bits, info->is_signed, info->is_float, info->freq,
+          info->nchannels);
 }
 #endif
 
@@ -879,9 +894,9 @@ size_t AUD_read(SWVoiceIn *sw, void *buf, size_t size)
     }
 }
 
-int AUD_get_buffer_size_out (SWVoiceOut *sw)
+int AUD_get_buffer_size_out(SWVoiceOut *sw)
 {
-    return sw->hw->mix_buf->size * sw->hw->info.bytes_per_frame;
+    return sw->hw->samples * sw->hw->info.bytes_per_frame;
 }
 
 void AUD_set_active_out (SWVoiceOut *sw, int on)
@@ -1076,10 +1091,8 @@ static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
     while (live) {
         size_t size, decr, proc;
         void *buf = hw->pcm_ops->get_buffer_out(hw, &size);
-        if (!buf) {
-            /* retrying will likely won't help, drop everything. */
-            hw->mix_buf->pos = (hw->mix_buf->pos + live) % hw->mix_buf->size;
-            return clipped + live;
+        if (!buf || size == 0) {
+            break;
         }
 
         decr = MIN(size / hw->info.bytes_per_frame, live);
@@ -1097,6 +1110,10 @@ static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
         }
     }
 
+    if (hw->pcm_ops->run_buffer_out) {
+        hw->pcm_ops->run_buffer_out(hw);
+    }
+
     return clipped;
 }
 
@@ -1403,7 +1420,8 @@ void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size)
     }
     assert(start >= 0 && start < hw->size_emul);
 
-    *size = MIN(hw->pending_emul, hw->size_emul - start);
+    *size = MIN(*size, hw->pending_emul);
+    *size = MIN(*size, hw->size_emul - start);
     return hw->buf_emul + start;
 }
 
@@ -1413,6 +1431,28 @@ void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size)
     hw->pending_emul -= size;
 }
 
+void audio_generic_run_buffer_out(HWVoiceOut *hw)
+{
+    while (hw->pending_emul) {
+        size_t write_len, written;
+        ssize_t start = ((ssize_t) hw->pos_emul) - hw->pending_emul;
+
+        if (start < 0) {
+            start += hw->size_emul;
+        }
+        assert(start >= 0 && start < hw->size_emul);
+
+        write_len = MIN(hw->pending_emul, hw->size_emul - start);
+
+        written = hw->pcm_ops->write(hw, hw->buf_emul + start, write_len);
+        hw->pending_emul -= written;
+
+        if (written < write_len) {
+            break;
+        }
+    }
+}
+
 void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size)
 {
     if (unlikely(!hw->buf_emul)) {
@@ -1428,8 +1468,7 @@ void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size)
     return hw->buf_emul + hw->pos_emul;
 }
 
-size_t audio_generic_put_buffer_out_nowrite(HWVoiceOut *hw, void *buf,
-                                            size_t size)
+size_t audio_generic_put_buffer_out(HWVoiceOut *hw, void *buf, size_t size)
 {
     assert(buf == hw->buf_emul + hw->pos_emul &&
            size + hw->pending_emul <= hw->size_emul);
@@ -1440,35 +1479,6 @@ size_t audio_generic_put_buffer_out_nowrite(HWVoiceOut *hw, void *buf,
     return size;
 }
 
-size_t audio_generic_put_buffer_out(HWVoiceOut *hw, void *buf, size_t size)
-{
-    audio_generic_put_buffer_out_nowrite(hw, buf, size);
-
-    while (hw->pending_emul) {
-        size_t write_len, written;
-        ssize_t start = ((ssize_t) hw->pos_emul) - hw->pending_emul;
-        if (start < 0) {
-            start += hw->size_emul;
-        }
-        assert(start >= 0 && start < hw->size_emul);
-
-        write_len = MIN(hw->pending_emul, hw->size_emul - start);
-
-        written = hw->pcm_ops->write(hw, hw->buf_emul + start, write_len);
-        hw->pending_emul -= written;
-
-        if (written < write_len) {
-            break;
-        }
-    }
-
-    /*
-     * fake we have written everything. non-written data remain in pending_emul,
-     * so we do not have to clip them multiple times
-     */
-    return size;
-}
-
 size_t audio_generic_write(HWVoiceOut *hw, void *buf, size_t size)
 {
     size_t dst_size, copy_size;
@@ -1476,17 +1486,17 @@ size_t audio_generic_write(HWVoiceOut *hw, void *buf, size_t size)
     copy_size = MIN(size, dst_size);
 
     memcpy(dst, buf, copy_size);
-    return hw->pcm_ops->put_buffer_out(hw, buf, copy_size);
+    return hw->pcm_ops->put_buffer_out(hw, dst, copy_size);
 }
 
 size_t audio_generic_read(HWVoiceIn *hw, void *buf, size_t size)
 {
-    size_t dst_size, copy_size;
-    void *dst = hw->pcm_ops->get_buffer_in(hw, &dst_size);
-    copy_size = MIN(size, dst_size);
+    size_t src_size, copy_size;
+    void *src = hw->pcm_ops->get_buffer_in(hw, &src_size);
+    copy_size = MIN(size, src_size);
 
-    memcpy(dst, buf, copy_size);
-    hw->pcm_ops->put_buffer_in(hw, buf, copy_size);
+    memcpy(buf, src, copy_size);
+    hw->pcm_ops->put_buffer_in(hw, src, copy_size);
     return copy_size;
 }
 
@@ -1837,11 +1847,15 @@ CaptureVoiceOut *AUD_add_capture(
 
         cap->buf = g_malloc0_n(hw->mix_buf->size, hw->info.bytes_per_frame);
 
-        hw->clip = mixeng_clip
-            [hw->info.nchannels == 2]
-            [hw->info.sign]
-            [hw->info.swap_endianness]
-            [audio_bits_to_index (hw->info.bits)];
+        if (hw->info.is_float) {
+            hw->clip = mixeng_clip_float[hw->info.nchannels == 2];
+        } else {
+            hw->clip = mixeng_clip
+                [hw->info.nchannels == 2]
+                [hw->info.is_signed]
+                [hw->info.swap_endianness]
+                [audio_bits_to_index(hw->info.bits)];
+        }
 
         QLIST_INSERT_HEAD (&s->cap_head, cap, entries);
         QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
@@ -2080,6 +2094,7 @@ int audioformat_bytes_per_sample(AudioFormat fmt)
 
     case AUDIO_FORMAT_U32:
     case AUDIO_FORMAT_S32:
+    case AUDIO_FORMAT_F32:
         return 4;
 
     case AUDIO_FORMAT__MAX:
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 5ba2078..4775857 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -40,7 +40,8 @@ struct audio_callback {
 
 struct audio_pcm_info {
     int bits;
-    int sign;
+    bool is_signed;
+    bool is_float;
     int freq;
     int nchannels;
     int bytes_per_frame;
@@ -152,6 +153,7 @@ struct audio_pcm_ops {
     int    (*init_out)(HWVoiceOut *hw, audsettings *as, void *drv_opaque);
     void   (*fini_out)(HWVoiceOut *hw);
     size_t (*write)   (HWVoiceOut *hw, void *buf, size_t size);
+    void   (*run_buffer_out)(HWVoiceOut *hw);
     /*
      * get a buffer that after later can be passed to put_buffer_out; optional
      * returns the buffer, and writes it's size to size (in bytes)
@@ -178,10 +180,9 @@ struct audio_pcm_ops {
 
 void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size);
 void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size);
+void audio_generic_run_buffer_out(HWVoiceOut *hw);
 void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size);
 size_t audio_generic_put_buffer_out(HWVoiceOut *hw, void *buf, size_t size);
-size_t audio_generic_put_buffer_out_nowrite(HWVoiceOut *hw, void *buf,
-                                            size_t size);
 size_t audio_generic_write(HWVoiceOut *hw, void *buf, size_t size);
 size_t audio_generic_read(HWVoiceIn *hw, void *buf, size_t size);
 
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 3287d70..7013d30 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -153,15 +153,23 @@ static int glue (audio_pcm_sw_init_, TYPE) (
     sw->ratio = ((int64_t) sw->info.freq << 32) / sw->hw->info.freq;
 #endif
 
+    if (sw->info.is_float) {
 #ifdef DAC
-    sw->conv = mixeng_conv
+        sw->conv = mixeng_conv_float[sw->info.nchannels == 2];
 #else
-    sw->clip = mixeng_clip
+        sw->clip = mixeng_clip_float[sw->info.nchannels == 2];
 #endif
-        [sw->info.nchannels == 2]
-        [sw->info.sign]
-        [sw->info.swap_endianness]
-        [audio_bits_to_index (sw->info.bits)];
+    } else {
+#ifdef DAC
+        sw->conv = mixeng_conv
+#else
+        sw->clip = mixeng_clip
+#endif
+            [sw->info.nchannels == 2]
+            [sw->info.is_signed]
+            [sw->info.swap_endianness]
+            [audio_bits_to_index(sw->info.bits)];
+    }
 
     sw->name = g_strdup (name);
     err = glue (audio_pcm_sw_alloc_resources_, TYPE) (sw);
@@ -276,15 +284,23 @@ static HW *glue(audio_pcm_hw_add_new_, TYPE)(AudioState *s,
         goto err1;
     }
 
+    if (hw->info.is_float) {
 #ifdef DAC
-    hw->clip = mixeng_clip
+        hw->clip = mixeng_clip_float[hw->info.nchannels == 2];
 #else
-    hw->conv = mixeng_conv
+        hw->conv = mixeng_conv_float[hw->info.nchannels == 2];
 #endif
-        [hw->info.nchannels == 2]
-        [hw->info.sign]
-        [hw->info.swap_endianness]
-        [audio_bits_to_index (hw->info.bits)];
+    } else {
+#ifdef DAC
+        hw->clip = mixeng_clip
+#else
+        hw->conv = mixeng_conv
+#endif
+            [hw->info.nchannels == 2]
+            [hw->info.is_signed]
+            [hw->info.swap_endianness]
+            [audio_bits_to_index(hw->info.bits)];
+    }
 
     glue(audio_pcm_hw_alloc_resources_, TYPE)(hw);
 
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index 66f0f45..4b43656 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -411,7 +411,7 @@ static int coreaudio_unlock (coreaudioVoiceOut *core, const char *fn_name)
     }
 COREAUDIO_WRAPPER_FUNC(get_buffer_out, void *, (HWVoiceOut *hw, size_t *size),
                        (hw, size))
-COREAUDIO_WRAPPER_FUNC(put_buffer_out_nowrite, size_t,
+COREAUDIO_WRAPPER_FUNC(put_buffer_out, size_t,
                        (HWVoiceOut *hw, void *buf, size_t size),
                        (hw, buf, size))
 COREAUDIO_WRAPPER_FUNC(write, size_t, (HWVoiceOut *hw, void *buf, size_t size),
@@ -471,20 +471,6 @@ static OSStatus audioDeviceIOProc(
     return 0;
 }
 
-static UInt32 coreaudio_get_flags(struct audio_pcm_info *info,
-                                  struct audsettings *as)
-{
-    UInt32 flags = info->sign ? kAudioFormatFlagIsSignedInteger : 0;
-    if (as->endianness) { /* 0 = little, 1 = big */
-        flags |= kAudioFormatFlagIsBigEndian;
-    }
-
-    if (flags == 0) { /* must not be 0 */
-        flags = kAudioFormatFlagsAreAllClear;
-    }
-    return flags;
-}
-
 static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
                               void *drv_opaque)
 {
@@ -496,6 +482,7 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
     Audiodev *dev = drv_opaque;
     AudiodevCoreaudioPerDirectionOptions *cpdo = dev->u.coreaudio.out;
     int frames;
+    struct audsettings fake_as;
 
     /* create mutex */
     err = pthread_mutex_init(&core->mutex, NULL);
@@ -504,6 +491,9 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
         return -1;
     }
 
+    fake_as = *as;
+    as = &fake_as;
+    as->fmt = AUDIO_FORMAT_F32;
     audio_pcm_init_info (&hw->info, as);
 
     status = coreaudio_get_voice(&core->outputDeviceID);
@@ -572,15 +562,6 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
 
     /* set Samplerate */
     core->outputStreamBasicDescription.mSampleRate = (Float64) as->freq;
-    core->outputStreamBasicDescription.mFormatID = kAudioFormatLinearPCM;
-    core->outputStreamBasicDescription.mFormatFlags =
-        coreaudio_get_flags(&hw->info, as);
-    core->outputStreamBasicDescription.mBytesPerPacket =
-        core->outputStreamBasicDescription.mBytesPerFrame =
-        hw->info.nchannels * hw->info.bits / 8;
-    core->outputStreamBasicDescription.mFramesPerPacket = 1;
-    core->outputStreamBasicDescription.mChannelsPerFrame = hw->info.nchannels;
-    core->outputStreamBasicDescription.mBitsPerChannel = hw->info.bits;
 
     status = coreaudio_set_streamformat(core->outputDeviceID,
                                         &core->outputStreamBasicDescription);
@@ -687,9 +668,12 @@ static void coreaudio_audio_fini (void *opaque)
 static struct audio_pcm_ops coreaudio_pcm_ops = {
     .init_out = coreaudio_init_out,
     .fini_out = coreaudio_fini_out,
+  /* wrapper for audio_generic_write */
     .write    = coreaudio_write,
+  /* wrapper for audio_generic_get_buffer_out */
     .get_buffer_out = coreaudio_get_buffer_out,
-    .put_buffer_out = coreaudio_put_buffer_out_nowrite,
+  /* wrapper for audio_generic_put_buffer_out */
+    .put_buffer_out = coreaudio_put_buffer_out,
     .enable_out = coreaudio_enable_out
 };
 
diff --git a/audio/dsound_template.h b/audio/dsound_template.h
index 7a15f91..9c5ce62 100644
--- a/audio/dsound_template.h
+++ b/audio/dsound_template.h
@@ -244,6 +244,7 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
         goto fail0;
     }
 
+    ds->first_time = true;
     obt_as.endianness = 0;
     audio_pcm_init_info (&hw->info, &obt_as);
 
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index c265c00..bd57082 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -53,12 +53,14 @@ typedef struct {
 typedef struct {
     HWVoiceOut hw;
     LPDIRECTSOUNDBUFFER dsound_buffer;
+    bool first_time;
     dsound *s;
 } DSoundVoiceOut;
 
 typedef struct {
     HWVoiceIn hw;
     LPDIRECTSOUNDCAPTUREBUFFER dsound_capture_buffer;
+    bool first_time;
     dsound *s;
 } DSoundVoiceIn;
 
@@ -414,21 +416,32 @@ static void *dsound_get_buffer_out(HWVoiceOut *hw, size_t *size)
     DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
     LPDIRECTSOUNDBUFFER dsb = ds->dsound_buffer;
     HRESULT hr;
-    DWORD ppos, act_size;
+    DWORD ppos, wpos, act_size;
     size_t req_size;
     int err;
     void *ret;
 
-    hr = IDirectSoundBuffer_GetCurrentPosition(dsb, &ppos, NULL);
+    hr = IDirectSoundBuffer_GetCurrentPosition(
+        dsb, &ppos, ds->first_time ? &wpos : NULL);
     if (FAILED(hr)) {
         dsound_logerr(hr, "Could not get playback buffer position\n");
         *size = 0;
         return NULL;
     }
 
+    if (ds->first_time) {
+        hw->pos_emul = wpos;
+        ds->first_time = false;
+    }
+
     req_size = audio_ring_dist(ppos, hw->pos_emul, hw->size_emul);
     req_size = MIN(req_size, hw->size_emul - hw->pos_emul);
 
+    if (req_size == 0) {
+        *size = 0;
+        return NULL;
+    }
+
     err = dsound_lock_out(dsb, &hw->info, hw->pos_emul, req_size, &ret, NULL,
                           &act_size, NULL, false, ds->s);
     if (err) {
@@ -508,18 +521,24 @@ static void *dsound_get_buffer_in(HWVoiceIn *hw, size_t *size)
     DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;
     LPDIRECTSOUNDCAPTUREBUFFER dscb = ds->dsound_capture_buffer;
     HRESULT hr;
-    DWORD cpos, act_size;
+    DWORD cpos, rpos, act_size;
     size_t req_size;
     int err;
     void *ret;
 
-    hr = IDirectSoundCaptureBuffer_GetCurrentPosition(dscb, &cpos, NULL);
+    hr = IDirectSoundCaptureBuffer_GetCurrentPosition(
+        dscb, &cpos, ds->first_time ? &rpos : NULL);
     if (FAILED(hr)) {
         dsound_logerr(hr, "Could not get capture buffer position\n");
         *size = 0;
         return NULL;
     }
 
+    if (ds->first_time) {
+        hw->pos_emul = rpos;
+        ds->first_time = false;
+    }
+
     req_size = audio_ring_dist(cpos, hw->pos_emul, hw->size_emul);
     req_size = MIN(req_size, hw->size_emul - hw->pos_emul);
 
diff --git a/audio/mixeng.c b/audio/mixeng.c
index 2f5ba71..c14b0d8 100644
--- a/audio/mixeng.c
+++ b/audio/mixeng.c
@@ -267,6 +267,76 @@ f_sample *mixeng_clip[2][2][2][3] = {
     }
 };
 
+#ifdef FLOAT_MIXENG
+#define FLOAT_CONV_TO(x) (x)
+#define FLOAT_CONV_FROM(x) (x)
+#else
+static const float float_scale = UINT_MAX;
+#define FLOAT_CONV_TO(x) ((x) * float_scale)
+
+#ifdef RECIPROCAL
+static const float float_scale_reciprocal = 1.f / UINT_MAX;
+#define FLOAT_CONV_FROM(x) ((x) * float_scale_reciprocal)
+#else
+#define FLOAT_CONV_FROM(x) ((x) / float_scale)
+#endif
+#endif
+
+static void conv_natural_float_to_mono(struct st_sample *dst, const void *src,
+                                       int samples)
+{
+    float *in = (float *)src;
+
+    while (samples--) {
+        dst->r = dst->l = FLOAT_CONV_TO(*in++);
+        dst++;
+    }
+}
+
+static void conv_natural_float_to_stereo(struct st_sample *dst, const void *src,
+                                         int samples)
+{
+    float *in = (float *)src;
+
+    while (samples--) {
+        dst->l = FLOAT_CONV_TO(*in++);
+        dst->r = FLOAT_CONV_TO(*in++);
+        dst++;
+    }
+}
+
+t_sample *mixeng_conv_float[2] = {
+    conv_natural_float_to_mono,
+    conv_natural_float_to_stereo,
+};
+
+static void clip_natural_float_from_mono(void *dst, const struct st_sample *src,
+                                         int samples)
+{
+    float *out = (float *)dst;
+
+    while (samples--) {
+        *out++ = FLOAT_CONV_FROM(src->l) + FLOAT_CONV_FROM(src->r);
+        src++;
+    }
+}
+
+static void clip_natural_float_from_stereo(
+    void *dst, const struct st_sample *src, int samples)
+{
+    float *out = (float *)dst;
+
+    while (samples--) {
+        *out++ = FLOAT_CONV_FROM(src->l);
+        *out++ = FLOAT_CONV_FROM(src->r);
+        src++;
+    }
+}
+
+f_sample *mixeng_clip_float[2] = {
+    clip_natural_float_from_mono,
+    clip_natural_float_from_stereo,
+};
 
 void audio_sample_to_uint64(void *samples, int pos,
                             uint64_t *left, uint64_t *right)
diff --git a/audio/mixeng.h b/audio/mixeng.h
index 18e62c7..2dcd6df 100644
--- a/audio/mixeng.h
+++ b/audio/mixeng.h
@@ -38,9 +38,14 @@ typedef struct st_sample st_sample;
 typedef void (t_sample) (struct st_sample *dst, const void *src, int samples);
 typedef void (f_sample) (void *dst, const struct st_sample *src, int samples);
 
+/* indices: [stereo][signed][swap endiannes][8, 16 or 32-bits] */
 extern t_sample *mixeng_conv[2][2][2][3];
 extern f_sample *mixeng_clip[2][2][2][3];
 
+/* indices: [stereo] */
+extern t_sample *mixeng_conv_float[2];
+extern f_sample *mixeng_clip_float[2];
+
 void *st_rate_start (int inrate, int outrate);
 void st_rate_flow(void *opaque, st_sample *ibuf, st_sample *obuf,
                   size_t *isamp, size_t *osamp);
diff --git a/audio/noaudio.c b/audio/noaudio.c
index ff99b25..05798ea 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -118,6 +118,7 @@ static struct audio_pcm_ops no_pcm_ops = {
     .init_out = no_init_out,
     .fini_out = no_fini_out,
     .write    = no_write,
+    .run_buffer_out = audio_generic_run_buffer_out,
     .enable_out = no_enable_out,
 
     .init_in  = no_init_in,
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index c43faee..f88d076 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -382,6 +382,15 @@ static size_t oss_get_available_bytes(OSSVoiceOut *oss)
     return audio_ring_dist(cntinfo.ptr, oss->hw.pos_emul, oss->hw.size_emul);
 }
 
+static void oss_run_buffer_out(HWVoiceOut *hw)
+{
+    OSSVoiceOut *oss = (OSSVoiceOut *)hw;
+
+    if (!oss->mmapped) {
+        audio_generic_run_buffer_out(hw);
+    }
+}
+
 static void *oss_get_buffer_out(HWVoiceOut *hw, size_t *size)
 {
     OSSVoiceOut *oss = (OSSVoiceOut *) hw;
@@ -420,7 +429,7 @@ static size_t oss_write(HWVoiceOut *hw, void *buf, size_t len)
             size_t to_copy = MIN(len, hw->size_emul - hw->pos_emul);
             memcpy(hw->buf_emul + hw->pos_emul, buf, to_copy);
 
-            hw->pos_emul = (hw->pos_emul + to_copy) % hw->pos_emul;
+            hw->pos_emul = (hw->pos_emul + to_copy) % hw->size_emul;
             buf += to_copy;
             len -= to_copy;
         }
@@ -570,20 +579,18 @@ static void oss_enable_out(HWVoiceOut *hw, bool enable)
     AudiodevOssPerDirectionOptions *opdo = oss->dev->u.oss.out;
 
     if (enable) {
-        bool poll_mode = opdo->try_poll;
+        hw->poll_mode = opdo->try_poll;
 
         ldebug("enabling voice\n");
-        if (poll_mode) {
+        if (hw->poll_mode) {
             oss_poll_out(hw);
-            poll_mode = 0;
         }
-        hw->poll_mode = poll_mode;
 
         if (!oss->mmapped) {
             return;
         }
 
-        audio_pcm_info_clear_buf(&hw->info, hw->buf_emul, hw->mix_buf->size);
+        audio_pcm_info_clear_buf(&hw->info, hw->buf_emul, hw->samples);
         trig = PCM_ENABLE_OUTPUT;
         if (ioctl(oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
             oss_logerr(errno,
@@ -699,17 +706,15 @@ static void oss_enable_in(HWVoiceIn *hw, bool enable)
     AudiodevOssPerDirectionOptions *opdo = oss->dev->u.oss.out;
 
     if (enable) {
-        bool poll_mode = opdo->try_poll;
+        hw->poll_mode = opdo->try_poll;
 
-        if (poll_mode) {
+        if (hw->poll_mode) {
             oss_poll_in(hw);
-            poll_mode = 0;
         }
-        hw->poll_mode = poll_mode;
     } else {
         if (hw->poll_mode) {
-            hw->poll_mode = 0;
             qemu_set_fd_handler (oss->fd, NULL, NULL, NULL);
+            hw->poll_mode = 0;
         }
     }
 }
@@ -748,6 +753,7 @@ static struct audio_pcm_ops oss_pcm_ops = {
     .init_out = oss_init_out,
     .fini_out = oss_fini_out,
     .write    = oss_write,
+    .run_buffer_out = oss_run_buffer_out,
     .get_buffer_out = oss_get_buffer_out,
     .put_buffer_out = oss_put_buffer_out,
     .enable_out = oss_enable_out,
diff --git a/audio/paaudio.c b/audio/paaudio.c
index dbfe48c..b052084 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -32,7 +32,6 @@ typedef struct {
     HWVoiceOut hw;
     pa_stream *stream;
     paaudio *g;
-    size_t samples;
 } PAVoiceOut;
 
 typedef struct {
@@ -41,7 +40,6 @@ typedef struct {
     const void *read_data;
     size_t read_length;
     paaudio *g;
-    size_t samples;
 } PAVoiceIn;
 
 static void qpa_conn_fini(PAConnection *c);
@@ -279,6 +277,9 @@ static pa_sample_format_t audfmt_to_pa (AudioFormat afmt, int endianness)
     case AUDIO_FORMAT_U32:
         format = endianness ? PA_SAMPLE_S32BE : PA_SAMPLE_S32LE;
         break;
+    case AUDIO_FORMAT_F32:
+        format = endianness ? PA_SAMPLE_FLOAT32BE : PA_SAMPLE_FLOAT32LE;
+        break;
     default:
         dolog ("Internal logic error: Bad audio format %d\n", afmt);
         format = PA_SAMPLE_U8;
@@ -304,6 +305,12 @@ static AudioFormat pa_to_audfmt (pa_sample_format_t fmt, int *endianness)
     case PA_SAMPLE_S32LE:
         *endianness = 0;
         return AUDIO_FORMAT_S32;
+    case PA_SAMPLE_FLOAT32BE:
+        *endianness = 1;
+        return AUDIO_FORMAT_F32;
+    case PA_SAMPLE_FLOAT32LE:
+        *endianness = 0;
+        return AUDIO_FORMAT_F32;
     default:
         dolog ("Internal logic error: Bad pa_sample_format %d\n", fmt);
         return AUDIO_FORMAT_U8;
@@ -488,7 +495,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
     }
 
     audio_pcm_init_info (&hw->info, &obt_as);
-    hw->samples = pa->samples = audio_buffer_samples(
+    hw->samples = audio_buffer_samples(
         qapi_AudiodevPaPerDirectionOptions_base(ppdo),
         &obt_as, ppdo->buffer_length);
 
@@ -536,7 +543,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
     }
 
     audio_pcm_init_info (&hw->info, &obt_as);
-    hw->samples = pa->samples = audio_buffer_samples(
+    hw->samples = audio_buffer_samples(
         qapi_AudiodevPaPerDirectionOptions_base(ppdo),
         &obt_as, ppdo->buffer_length);
 
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index 5c6bcfc..21b7a04 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -77,6 +77,14 @@ static int aud_to_sdlfmt (AudioFormat fmt)
     case AUDIO_FORMAT_U16:
         return AUDIO_U16LSB;
 
+    case AUDIO_FORMAT_S32:
+        return AUDIO_S32LSB;
+
+    /* no unsigned 32-bit support in SDL */
+
+    case AUDIO_FORMAT_F32:
+        return AUDIO_F32LSB;
+
     default:
         dolog ("Internal logic error: Bad audio format %d\n", fmt);
 #ifdef DEBUG_AUDIO
@@ -119,6 +127,26 @@ static int sdl_to_audfmt(int sdlfmt, AudioFormat *fmt, int *endianness)
         *fmt = AUDIO_FORMAT_U16;
         break;
 
+    case AUDIO_S32LSB:
+        *endianness = 0;
+        *fmt = AUDIO_FORMAT_S32;
+        break;
+
+    case AUDIO_S32MSB:
+        *endianness = 1;
+        *fmt = AUDIO_FORMAT_S32;
+        break;
+
+    case AUDIO_F32LSB:
+        *endianness = 0;
+        *fmt = AUDIO_FORMAT_F32;
+        break;
+
+    case AUDIO_F32MSB:
+        *endianness = 1;
+        *fmt = AUDIO_FORMAT_F32;
+        break;
+
     default:
         dolog ("Unrecognized SDL audio format %d\n", sdlfmt);
         return -1;
@@ -227,7 +255,7 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
 
 SDL_WRAPPER_FUNC(get_buffer_out, void *, (HWVoiceOut *hw, size_t *size),
                  (hw, size), *size = 0, sdl_unlock)
-SDL_WRAPPER_FUNC(put_buffer_out_nowrite, size_t,
+SDL_WRAPPER_FUNC(put_buffer_out, size_t,
                  (HWVoiceOut *hw, void *buf, size_t size), (hw, buf, size),
                  /*nothing*/, sdl_unlock_and_post)
 SDL_WRAPPER_FUNC(write, size_t,
@@ -320,9 +348,12 @@ static void sdl_audio_fini (void *opaque)
 static struct audio_pcm_ops sdl_pcm_ops = {
     .init_out = sdl_init_out,
     .fini_out = sdl_fini_out,
+  /* wrapper for audio_generic_write */
     .write    = sdl_write,
+  /* wrapper for audio_generic_get_buffer_out */
     .get_buffer_out = sdl_get_buffer_out,
-    .put_buffer_out = sdl_put_buffer_out_nowrite,
+  /* wrapper for audio_generic_put_buffer_out */
+    .put_buffer_out = sdl_put_buffer_out,
     .enable_out = sdl_enable_out,
 };
 
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index e46d834..20e6853 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -197,6 +197,7 @@ static struct audio_pcm_ops wav_pcm_ops = {
     .init_out = wav_init_out,
     .fini_out = wav_fini_out,
     .write    = wav_write_out,
+    .run_buffer_out = audio_generic_run_buffer_out,
     .enable_out = wav_enable_out,
 };
 
diff --git a/qapi/audio.json b/qapi/audio.json
index 83312b2..d8c507c 100644
--- a/qapi/audio.json
+++ b/qapi/audio.json
@@ -276,7 +276,7 @@
 # Since: 4.0
 ##
 { 'enum': 'AudioFormat',
-  'data': [ 'u8', 's8', 'u16', 's16', 'u32', 's32' ] }
+  'data': [ 'u8', 's8', 'u16', 's16', 'u32', 's32', 'f32' ] }
 
 ##
 # @AudiodevDriver: