[Qemu-devel] [PATCH v4 00/11] apply volume on client side

[Qemu-devel] [PATCH v4 00/11] apply volume on client side

[Marc-André Lureau]

Hi,

This patch series implements client-side audio volume support. This reduces confusion of guest users when volume control is not effective (because mixemu is disabled or because client-side is muted and can't be unmuted by the guest..)

Instead, the backend is responsible for applying volume giving the guest control over the full range of the client, and avoiding multiple level of volume/mute effects.

Although I was mainly interested in having the Spice audio backend support, I also added PulseAudio backend support (which unfortunately requires full-blown API, even after quick discussion with upstream).

v4:
- code style fixes (only warnings left - mostly spaces)
- dropped the 1 << 32 -1 change
- added QEMU_MIXEMU runtime option, enabled by default
- rfc: removed some deprecated audio options

Marc-André Lureau (11):
  audio: add VOICE_VOLUME ctl
  audio: don't apply volume effect if backend has VOICE_VOLUME_CAP
  hw/ac97: remove USE_MIXER code
  hw/ac97: the volume mask is not only 0x1f
  hw/ac97: add support for volume control
  audio/spice: add support for volume control
  Do not use pa_simple PulseAudio API
  configure: pa_simple is not needed anymore
  Allow controlling volume with PulseAudio backend
  Enable mixemu by default, add runtime option
  audio/rfc: remove PLIVE and PERIOD options

 audio/audio.c          |   64 +++----
 audio/audio_int.h      |    6 +
 audio/audio_template.h |   28 +---
 audio/mixeng.c         |    6 -
 audio/paaudio.c        |  476 +++++++++++++++++++++++++++++++++++++++++++-----
 audio/spiceaudio.c     |   41 ++++
 configure              |   14 +-
 hw/ac97.c              |  140 +++++---------
 hw/hda-audio.c         |    4 -
 9 files changed, 561 insertions(+), 218 deletions(-)

-- 
1.7.7.6

[Qemu-devel] [PATCH v4 01/11] audio: add VOICE_VOLUME ctl

[Marc-André Lureau]

Add a new PCM control operation to update the stream volume on the
audio backend.  The argument given is a SWVoiceOut/SWVoiceIn.

v4:
- verified other backends didn't fail/assert on this new control
  they randomly return 0 or -1, but we ignore return value.

Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
---
 audio/audio.c     |   12 ++++++++++++
 audio/audio_int.h |    1 +
 2 files changed, 13 insertions(+), 0 deletions(-)

diff --git a/audio/audio.c b/audio/audio.c
index 398763f..d76c342 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -2053,17 +2053,29 @@ 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) {
+        HWVoiceOut *hw = sw->hw;
+
         sw->vol.mute = mute;
         sw->vol.l = nominal_volume.l * lvol / 255;
         sw->vol.r = nominal_volume.r * rvol / 255;
+
+        if (hw->pcm_ops->ctl_out) {
+            hw->pcm_ops->ctl_out (hw, VOICE_VOLUME, sw);
+        }
     }
 }
 
 void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
 {
     if (sw) {
+        HWVoiceIn *hw = sw->hw;
+
         sw->vol.mute = mute;
         sw->vol.l = nominal_volume.l * lvol / 255;
         sw->vol.r = nominal_volume.r * rvol / 255;
+
+        if (hw->pcm_ops->ctl_in) {
+            hw->pcm_ops->ctl_in (hw, VOICE_VOLUME, sw);
+        }
     }
 }
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 2003f8b..117f95e 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -231,6 +231,7 @@ void audio_run (const char *msg);
 
 #define VOICE_ENABLE 1
 #define VOICE_DISABLE 2
+#define VOICE_VOLUME 3
 
 static inline int audio_ring_dist (int dst, int src, int len)
 {
-- 
1.7.7.6

[Qemu-devel] [PATCH v4 02/11] audio: don't apply volume effect if backend has VOICE_VOLUME_CAP

[Marc-André Lureau]

If the audio backend is capable of volume control, don't apply
software volume (mixeng_volume ()), but instead, rely on backend
volume control. This will allow guest to have full range volume
control.

Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
---
 audio/audio.c          |    9 +++++++--
 audio/audio_int.h      |    5 +++++
 audio/audio_template.h |    2 ++
 3 files changed, 14 insertions(+), 2 deletions(-)

diff --git a/audio/audio.c b/audio/audio.c
index d76c342..bd9237e 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -957,7 +957,9 @@ int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int size)
         total += isamp;
     }
 
-    mixeng_volume (sw->buf, ret, &sw->vol);
+    if (!(hw->ctl_caps & VOICE_VOLUME_CAP)) {
+        mixeng_volume (sw->buf, ret, &sw->vol);
+    }
 
     sw->clip (buf, sw->buf, ret);
     sw->total_hw_samples_acquired += total;
@@ -1041,7 +1043,10 @@ int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
     swlim = audio_MIN (swlim, samples);
     if (swlim) {
         sw->conv (sw->buf, buf, swlim);
-        mixeng_volume (sw->buf, swlim, &sw->vol);
+
+        if (!(sw->hw->ctl_caps & VOICE_VOLUME_CAP)) {
+            mixeng_volume (sw->buf, swlim, &sw->vol);
+        }
     }
 
     while (swlim) {
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 117f95e..b9b0676 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -82,6 +82,7 @@ typedef struct HWVoiceOut {
     int samples;
     QLIST_HEAD (sw_out_listhead, SWVoiceOut) sw_head;
     QLIST_HEAD (sw_cap_listhead, SWVoiceCap) cap_head;
+    int ctl_caps;
     struct audio_pcm_ops *pcm_ops;
     QLIST_ENTRY (HWVoiceOut) entries;
 } HWVoiceOut;
@@ -101,6 +102,7 @@ typedef struct HWVoiceIn {
 
     int samples;
     QLIST_HEAD (sw_in_listhead, SWVoiceIn) sw_head;
+    int ctl_caps;
     struct audio_pcm_ops *pcm_ops;
     QLIST_ENTRY (HWVoiceIn) entries;
 } HWVoiceIn;
@@ -150,6 +152,7 @@ struct audio_driver {
     int max_voices_in;
     int voice_size_out;
     int voice_size_in;
+    int ctl_caps;
 };
 
 struct audio_pcm_ops {
@@ -233,6 +236,8 @@ void audio_run (const char *msg);
 #define VOICE_DISABLE 2
 #define VOICE_VOLUME 3
 
+#define VOICE_VOLUME_CAP (1 << VOICE_VOLUME)
+
 static inline int audio_ring_dist (int dst, int src, int len)
 {
     return (dst >= src) ? (dst - src) : (len - src + dst);
diff --git a/audio/audio_template.h b/audio/audio_template.h
index e62a713..519432a 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -263,6 +263,8 @@ static HW *glue (audio_pcm_hw_add_new_, TYPE) (struct audsettings *as)
     }
 
     hw->pcm_ops = drv->pcm_ops;
+    hw->ctl_caps = drv->ctl_caps;
+
     QLIST_INIT (&hw->sw_head);
 #ifdef DAC
     QLIST_INIT (&hw->cap_head);
-- 
1.7.7.6

[Qemu-devel] [PATCH v4 03/11] hw/ac97: remove USE_MIXER code

[Marc-André Lureau]

That code doesn't compile. The interesting bits for volume control are
going to be rewritten in the following patch.

Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
---
 hw/ac97.c |  121 -------------------------------------------------------------
 1 files changed, 0 insertions(+), 121 deletions(-)

diff --git a/hw/ac97.c b/hw/ac97.c
index c0fd019..f2804e6 100644
--- a/hw/ac97.c
+++ b/hw/ac97.c
@@ -437,99 +437,6 @@ static void reset_voices (AC97LinkState *s, uint8_t active[LAST_INDEX])
     AUD_set_active_in (s->voice_mc, active[MC_INDEX]);
 }
 
-#ifdef USE_MIXER
-static void set_volume (AC97LinkState *s, int index,
-                        audmixerctl_t mt, uint32_t val)
-{
-    int mute = (val >> MUTE_SHIFT) & 1;
-    uint8_t rvol = VOL_MASK - (val & VOL_MASK);
-    uint8_t lvol = VOL_MASK - ((val >> 8) & VOL_MASK);
-    rvol = 255 * rvol / VOL_MASK;
-    lvol = 255 * lvol / VOL_MASK;
-
-#ifdef SOFT_VOLUME
-    if (index == AC97_Master_Volume_Mute) {
-        AUD_set_volume_out (s->voice_po, mute, lvol, rvol);
-    }
-    else {
-        AUD_set_volume (mt, &mute, &lvol, &rvol);
-    }
-#else
-    AUD_set_volume (mt, &mute, &lvol, &rvol);
-#endif
-
-    rvol = VOL_MASK - ((VOL_MASK * rvol) / 255);
-    lvol = VOL_MASK - ((VOL_MASK * lvol) / 255);
-    mixer_store (s, index, val);
-}
-
-static audrecsource_t ac97_to_aud_record_source (uint8_t i)
-{
-    switch (i) {
-    case REC_MIC:
-        return AUD_REC_MIC;
-
-    case REC_CD:
-        return AUD_REC_CD;
-
-    case REC_VIDEO:
-        return AUD_REC_VIDEO;
-
-    case REC_AUX:
-        return AUD_REC_AUX;
-
-    case REC_LINE_IN:
-        return AUD_REC_LINE_IN;
-
-    case REC_PHONE:
-        return AUD_REC_PHONE;
-
-    default:
-        dolog ("Unknown record source %d, using MIC\n", i);
-        return AUD_REC_MIC;
-    }
-}
-
-static uint8_t aud_to_ac97_record_source (audrecsource_t rs)
-{
-    switch (rs) {
-    case AUD_REC_MIC:
-        return REC_MIC;
-
-    case AUD_REC_CD:
-        return REC_CD;
-
-    case AUD_REC_VIDEO:
-        return REC_VIDEO;
-
-    case AUD_REC_AUX:
-        return REC_AUX;
-
-    case AUD_REC_LINE_IN:
-        return REC_LINE_IN;
-
-    case AUD_REC_PHONE:
-        return REC_PHONE;
-
-    default:
-        dolog ("Unknown audio recording source %d using MIC\n", rs);
-        return REC_MIC;
-    }
-}
-
-static void record_select (AC97LinkState *s, uint32_t val)
-{
-    uint8_t rs = val & REC_MASK;
-    uint8_t ls = (val >> 8) & REC_MASK;
-    audrecsource_t ars = ac97_to_aud_record_source (rs);
-    audrecsource_t als = ac97_to_aud_record_source (ls);
-    AUD_set_record_source (&als, &ars);
-    rs = aud_to_ac97_record_source (ars);
-    ls = aud_to_ac97_record_source (als);
-    mixer_store (s, AC97_Record_Select, rs | (ls << 8));
-}
-#endif
-
 static void mixer_reset (AC97LinkState *s)
 {
     uint8_t active[LAST_INDEX];
@@ -564,12 +471,6 @@ static void mixer_reset (AC97LinkState *s)
     mixer_store (s, AC97_PCM_LR_ADC_Rate         , 0xbb80);
     mixer_store (s, AC97_MIC_ADC_Rate            , 0xbb80);
 
-#ifdef USE_MIXER
-    record_select (s, 0);
-    set_volume (s, AC97_Master_Volume_Mute, AUD_MIXER_VOLUME  , 0x8000);
-    set_volume (s, AC97_PCM_Out_Volume_Mute, AUD_MIXER_PCM    , 0x8808);
-    set_volume (s, AC97_Line_In_Volume_Mute, AUD_MIXER_LINE_IN, 0x8808);
-#endif
     reset_voices (s, active);
 }
 
@@ -628,20 +529,6 @@ static void nam_writew (void *opaque, uint32_t addr, uint32_t val)
         val |= mixer_load (s, index) & 0xf;
         mixer_store (s, index, val);
         break;
-#ifdef USE_MIXER
-    case AC97_Master_Volume_Mute:
-        set_volume (s, index, AUD_MIXER_VOLUME, val);
-        break;
-    case AC97_PCM_Out_Volume_Mute:
-        set_volume (s, index, AUD_MIXER_PCM, val);
-        break;
-    case AC97_Line_In_Volume_Mute:
-        set_volume (s, index, AUD_MIXER_LINE_IN, val);
-        break;
-    case AC97_Record_Select:
-        record_select (s, val);
-        break;
-#endif
     case AC97_Vendor_ID1:
     case AC97_Vendor_ID2:
         dolog ("Attempt to write vendor ID to %#x\n", val);
@@ -1194,14 +1081,6 @@ static int ac97_post_load (void *opaque, int version_id)
     uint8_t active[LAST_INDEX];
     AC97LinkState *s = opaque;
 
-#ifdef USE_MIXER
-    record_select (s, mixer_load (s, AC97_Record_Select));
-#define V_(a, b) set_volume (s, a, b, mixer_load (s, a))
-    V_ (AC97_Master_Volume_Mute, AUD_MIXER_VOLUME);
-    V_ (AC97_PCM_Out_Volume_Mute, AUD_MIXER_PCM);
-    V_ (AC97_Line_In_Volume_Mute, AUD_MIXER_LINE_IN);
-#undef V_
-#endif
     active[PI_INDEX] = !!(s->bm_regs[PI_INDEX].cr & CR_RPBM);
     active[PO_INDEX] = !!(s->bm_regs[PO_INDEX].cr & CR_RPBM);
     active[MC_INDEX] = !!(s->bm_regs[MC_INDEX].cr & CR_RPBM);
-- 
1.7.7.6

[Qemu-devel] [PATCH v4 04/11] hw/ac97: the volume mask is not only 0x1f

[Marc-André Lureau]

It's a case by case (see Table 66. AC ‘97 Baseline Audio Register Map)

Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
---
 hw/ac97.c |    1 -
 1 files changed, 0 insertions(+), 1 deletions(-)

diff --git a/hw/ac97.c b/hw/ac97.c
index f2804e6..f7866ed 100644
--- a/hw/ac97.c
+++ b/hw/ac97.c
@@ -118,7 +118,6 @@ enum {
 #define EACS_VRA 1
 #define EACS_VRM 8
 
-#define VOL_MASK 0x1f
 #define MUTE_SHIFT 15
 
 #define REC_MASK 7
-- 
1.7.7.6

[Qemu-devel] [PATCH v4 05/11] hw/ac97: add support for volume control

[Marc-André Lureau]

Combine output volume with Master and PCM registers values.
Use default values in mixer_reset ().
Set volume on post-load to update backend values.

v4:
- fix some code style

Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
---
 hw/ac97.c |   80 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
 1 files changed, 80 insertions(+), 0 deletions(-)

diff --git a/hw/ac97.c b/hw/ac97.c
index f7866ed..f5ae637 100644
--- a/hw/ac97.c
+++ b/hw/ac97.c
@@ -436,6 +436,64 @@ static void reset_voices (AC97LinkState *s, uint8_t active[LAST_INDEX])
     AUD_set_active_in (s->voice_mc, active[MC_INDEX]);
 }
 
+static void get_volume (uint16_t vol, uint16_t mask, int inverse,
+                        int *mute, uint8_t *lvol, uint8_t *rvol)
+{
+  *mute = (vol >> MUTE_SHIFT) & 1;
+  *rvol = (255 * (vol & mask)) / mask;
+  *lvol = (255 * ((vol >> 8) & mask)) / mask;
+  if (inverse) {
+    *rvol = 255 - *rvol;
+    *lvol = 255 - *lvol;
+  }
+}
+
+static void update_combined_volume_out (AC97LinkState *s)
+{
+    uint8_t lvol, rvol, plvol, prvol;
+    int mute, pmute;
+
+    get_volume (mixer_load (s, AC97_Master_Volume_Mute), 0x3f, 1,
+                &mute, &lvol, &rvol);
+    /* FIXME: should be 1f according to spec */
+    get_volume (mixer_load (s, AC97_PCM_Out_Volume_Mute), 0x3f, 1,
+                &pmute, &plvol, &prvol);
+
+    mute = mute | pmute;
+    lvol = (lvol * plvol) / 255;
+    rvol = (rvol * prvol) / 255;
+
+    AUD_set_volume_out (s->voice_po, mute, lvol, rvol);
+}
+
+static void update_volume_in(AC97LinkState *s)
+{
+    uint8_t lvol, rvol;
+    int mute;
+
+    get_volume (mixer_load (s, AC97_Record_Gain_Mute), 0x0f, 0,
+                &mute, &lvol, &rvol);
+
+    AUD_set_volume_in (s->voice_pi, mute, lvol, rvol);
+}
+
+static void set_volume (AC97LinkState *s, int index, uint32_t val)
+{
+    mixer_store (s, index, val);
+    if (index == AC97_Master_Volume_Mute || index == AC97_PCM_Out_Volume_Mute) {
+        update_combined_volume_out (s);
+    } else if (index == AC97_Record_Gain_Mute) {
+        update_volume_in (s);
+    }
+}
+
+static void record_select (AC97LinkState *s, uint32_t val)
+{
+    uint8_t rs = val & REC_MASK;
+    uint8_t ls = (val >> 8) & REC_MASK;
+    mixer_store (s, AC97_Record_Select, rs | (ls << 8));
+}
+
 static void mixer_reset (AC97LinkState *s)
 {
     uint8_t active[LAST_INDEX];
@@ -470,6 +528,11 @@ static void mixer_reset (AC97LinkState *s)
     mixer_store (s, AC97_PCM_LR_ADC_Rate         , 0xbb80);
     mixer_store (s, AC97_MIC_ADC_Rate            , 0xbb80);
 
+    record_select (s, 0);
+    set_volume (s, AC97_Master_Volume_Mute, 0x8000);
+    set_volume (s, AC97_PCM_Out_Volume_Mute, 0x8808);
+    set_volume (s, AC97_Line_In_Volume_Mute, 0x8808);
+
     reset_voices (s, active);
 }
 
@@ -528,6 +591,15 @@ static void nam_writew (void *opaque, uint32_t addr, uint32_t val)
         val |= mixer_load (s, index) & 0xf;
         mixer_store (s, index, val);
         break;
+    case AC97_PCM_Out_Volume_Mute:
+    case AC97_Master_Volume_Mute:
+    case AC97_Record_Gain_Mute:
+    case AC97_Line_In_Volume_Mute:
+        set_volume (s, index, val);
+        break;
+    case AC97_Record_Select:
+        record_select (s, val);
+        break;
     case AC97_Vendor_ID1:
     case AC97_Vendor_ID2:
         dolog ("Attempt to write vendor ID to %#x\n", val);
@@ -1080,6 +1152,14 @@ static int ac97_post_load (void *opaque, int version_id)
     uint8_t active[LAST_INDEX];
     AC97LinkState *s = opaque;
 
+    record_select (s, mixer_load (s, AC97_Record_Select));
+    set_volume (s, AC97_Master_Volume_Mute,
+                mixer_load (s, AC97_Master_Volume_Mute));
+    set_volume (s, AC97_PCM_Out_Volume_Mute,
+                mixer_load (s, AC97_PCM_Out_Volume_Mute));
+    set_volume (s, AC97_Line_In_Volume_Mute,
+                mixer_load (s, AC97_Line_In_Volume_Mute));
+
     active[PI_INDEX] = !!(s->bm_regs[PI_INDEX].cr & CR_RPBM);
     active[PO_INDEX] = !!(s->bm_regs[PO_INDEX].cr & CR_RPBM);
     active[MC_INDEX] = !!(s->bm_regs[MC_INDEX].cr & CR_RPBM);
-- 
1.7.7.6

[Qemu-devel] [PATCH v4 06/11] audio/spice: add support for volume control

[Marc-André Lureau]

Use Spice server volume control API when available.

Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
---
 audio/spiceaudio.c |   41 +++++++++++++++++++++++++++++++++++++++++
 1 files changed, 41 insertions(+), 0 deletions(-)

diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index f972110..6f15591 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -202,7 +202,26 @@ static int line_out_ctl (HWVoiceOut *hw, int cmd, ...)
         }
         spice_server_playback_stop (&out->sin);
         break;
+    case VOICE_VOLUME:
+        {
+#if ((SPICE_INTERFACE_PLAYBACK_MAJOR >= 1) && (SPICE_INTERFACE_PLAYBACK_MINOR >= 2))
+            SWVoiceOut *sw;
+            va_list ap;
+            uint16_t vol[2];
+
+            va_start (ap, cmd);
+            sw = va_arg (ap, SWVoiceOut *);
+            va_end (ap);
+
+            vol[0] = sw->vol.l / ((1ULL << 16) + 1);
+            vol[1] = sw->vol.r / ((1ULL << 16) + 1);
+            spice_server_playback_set_volume (&out->sin, 2, vol);
+            spice_server_playback_set_mute (&out->sin, sw->vol.mute);
+#endif
+            break;
+        }
     }
+
     return 0;
 }
 
@@ -304,7 +323,26 @@ static int line_in_ctl (HWVoiceIn *hw, int cmd, ...)
         in->active = 0;
         spice_server_record_stop (&in->sin);
         break;
+    case VOICE_VOLUME:
+        {
+#if ((SPICE_INTERFACE_RECORD_MAJOR >= 2) && (SPICE_INTERFACE_RECORD_MINOR >= 2))
+            SWVoiceIn *sw;
+            va_list ap;
+            uint16_t vol[2];
+
+            va_start (ap, cmd);
+            sw = va_arg (ap, SWVoiceIn *);
+            va_end (ap);
+
+            vol[0] = sw->vol.l / ((1ULL << 16) + 1);
+            vol[1] = sw->vol.r / ((1ULL << 16) + 1);
+            spice_server_record_set_volume (&in->sin, 2, vol);
+            spice_server_record_set_mute (&in->sin, sw->vol.mute);
+#endif
+            break;
+        }
     }
+
     return 0;
 }
 
@@ -337,6 +375,9 @@ struct audio_driver spice_audio_driver = {
     .max_voices_in  = 1,
     .voice_size_out = sizeof (SpiceVoiceOut),
     .voice_size_in  = sizeof (SpiceVoiceIn),
+#if ((SPICE_INTERFACE_PLAYBACK_MAJOR >= 1) && (SPICE_INTERFACE_PLAYBACK_MINOR >= 2))
+    .ctl_caps       = VOICE_VOLUME_CAP
+#endif
 };
 
 void qemu_spice_audio_init (void)
-- 
1.7.7.6

[Qemu-devel] [PATCH v4 07/11] Do not use pa_simple PulseAudio API

[Marc-André Lureau]

Unfortunately, pa_simple is a limited API which doesn't let us
retrieve the associated pa_stream. It is needed to control the volume
of the stream.

In v4:
- add missing braces

Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
---
 audio/paaudio.c |  377 +++++++++++++++++++++++++++++++++++++++++++++++++------
 1 files changed, 339 insertions(+), 38 deletions(-)

diff --git a/audio/paaudio.c b/audio/paaudio.c
index d1f3912..6f50c1c 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -2,8 +2,7 @@
 #include "qemu-common.h"
 #include "audio.h"
 
-#include <pulse/simple.h>
-#include <pulse/error.h>
+#include <pulse/pulseaudio.h>
 
 #define AUDIO_CAP "pulseaudio"
 #include "audio_int.h"
@@ -15,7 +14,7 @@ typedef struct {
     int live;
     int decr;
     int rpos;
-    pa_simple *s;
+    pa_stream *stream;
     void *pcm_buf;
     struct audio_pt pt;
 } PAVoiceOut;
@@ -26,17 +25,23 @@ typedef struct {
     int dead;
     int incr;
     int wpos;
-    pa_simple *s;
+    pa_stream *stream;
     void *pcm_buf;
     struct audio_pt pt;
+    const void *read_data;
+    size_t read_index, read_length;
 } PAVoiceIn;
 
-static struct {
+typedef struct {
     int samples;
     char *server;
     char *sink;
     char *source;
-} conf = {
+    pa_threaded_mainloop *mainloop;
+    pa_context *context;
+} paaudio;
+
+static paaudio glob_paaudio = {
     .samples = 4096,
 };
 
@@ -51,6 +56,126 @@ static void GCC_FMT_ATTR (2, 3) qpa_logerr (int err, const char *fmt, ...)
     AUD_log (AUDIO_CAP, "Reason: %s\n", pa_strerror (err));
 }
 
+#define CHECK_SUCCESS_GOTO(c, rerror, expression, label)        \
+    do {                                                        \
+        if (!(expression)) {                                    \
+            if (rerror) {                                       \
+                *(rerror) = pa_context_errno ((c)->context);    \
+            }                                                   \
+            goto label;                                         \
+        }                                                       \
+    } while (0);
+
+#define CHECK_DEAD_GOTO(c, stream, rerror, label)                       \
+    do {                                                                \
+        if (!(c)->context || !PA_CONTEXT_IS_GOOD (pa_context_get_state((c)->context)) || \
+            !(stream) || !PA_STREAM_IS_GOOD (pa_stream_get_state ((stream)))) { \
+            if (((c)->context && pa_context_get_state ((c)->context) == PA_CONTEXT_FAILED) || \
+                ((stream) && pa_stream_get_state ((stream)) == PA_STREAM_FAILED)) { \
+                if (rerror) {                                           \
+                    *(rerror) = pa_context_errno ((c)->context);        \
+                }                                                       \
+            } else {                                                    \
+                if (rerror) {                                           \
+                    *(rerror) = PA_ERR_BADSTATE;                        \
+                }                                                       \
+            }                                                           \
+            goto label;                                                 \
+        }                                                               \
+    } while (0);
+
+static int qpa_simple_read (PAVoiceIn *p, void *data, size_t length, int *rerror)
+{
+    paaudio *g = &glob_paaudio;
+
+    pa_threaded_mainloop_lock (g->mainloop);
+
+    CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
+
+    while (length > 0) {
+        size_t l;
+
+        while (!p->read_data) {
+            int r;
+
+            r = pa_stream_peek (p->stream, &p->read_data, &p->read_length);
+            CHECK_SUCCESS_GOTO (g, rerror, r == 0, unlock_and_fail);
+
+            if (!p->read_data) {
+                pa_threaded_mainloop_wait (g->mainloop);
+                CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
+            } else {
+                p->read_index = 0;
+            }
+        }
+
+        l = p->read_length < length ? p->read_length : length;
+        memcpy (data, (const uint8_t *) p->read_data+p->read_index, l);
+
+        data = (uint8_t *) data + l;
+        length -= l;
+
+        p->read_index += l;
+        p->read_length -= l;
+
+        if (!p->read_length) {
+            int r;
+
+            r = pa_stream_drop (p->stream);
+            p->read_data = NULL;
+            p->read_length = 0;
+            p->read_index = 0;
+
+            CHECK_SUCCESS_GOTO (g, rerror, r == 0, unlock_and_fail);
+        }
+    }
+
+    pa_threaded_mainloop_unlock (g->mainloop);
+    return 0;
+
+unlock_and_fail:
+    pa_threaded_mainloop_unlock (g->mainloop);
+    return -1;
+}
+
+static int qpa_simple_write (PAVoiceOut *p, const void *data, size_t length, int *rerror)
+{
+    paaudio *g = &glob_paaudio;
+
+    pa_threaded_mainloop_lock (g->mainloop);
+
+    CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
+
+    while (length > 0) {
+        size_t l;
+        int r;
+
+        while (!(l = pa_stream_writable_size (p->stream))) {
+            pa_threaded_mainloop_wait (g->mainloop);
+            CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
+        }
+
+        CHECK_SUCCESS_GOTO (g, rerror, l != (size_t) -1, unlock_and_fail);
+
+        if (l > length) {
+            l = length;
+        }
+
+        r = pa_stream_write (p->stream, data, l, NULL, 0LL, PA_SEEK_RELATIVE);
+        CHECK_SUCCESS_GOTO (g, rerror, r >= 0, unlock_and_fail);
+
+        data = (const uint8_t *) data + l;
+        length -= l;
+    }
+
+    pa_threaded_mainloop_unlock (g->mainloop);
+    return 0;
+
+unlock_and_fail:
+    pa_threaded_mainloop_unlock (g->mainloop);
+    return -1;
+}
+
 static void *qpa_thread_out (void *arg)
 {
     PAVoiceOut *pa = arg;
@@ -77,7 +202,7 @@ static void *qpa_thread_out (void *arg)
             }
         }
 
-        decr = to_mix = audio_MIN (pa->live, conf.samples >> 2);
+        decr = to_mix = audio_MIN (pa->live, glob_paaudio.samples >> 2);
         rpos = pa->rpos;
 
         if (audio_pt_unlock (&pa->pt, AUDIO_FUNC)) {
@@ -91,8 +216,8 @@ static void *qpa_thread_out (void *arg)
 
             hw->clip (pa->pcm_buf, src, chunk);
 
-            if (pa_simple_write (pa->s, pa->pcm_buf,
-                                 chunk << hw->info.shift, &error) < 0) {
+            if (qpa_simple_write (pa, pa->pcm_buf,
+                                  chunk << hw->info.shift, &error) < 0) {
                 qpa_logerr (error, "pa_simple_write failed\n");
                 return NULL;
             }
@@ -169,7 +294,7 @@ static void *qpa_thread_in (void *arg)
             }
         }
 
-        incr = to_grab = audio_MIN (pa->dead, conf.samples >> 2);
+        incr = to_grab = audio_MIN (pa->dead, glob_paaudio.samples >> 2);
         wpos = pa->wpos;
 
         if (audio_pt_unlock (&pa->pt, AUDIO_FUNC)) {
@@ -181,8 +306,8 @@ static void *qpa_thread_in (void *arg)
             int chunk = audio_MIN (to_grab, hw->samples - wpos);
             void *buf = advance (pa->pcm_buf, wpos);
 
-            if (pa_simple_read (pa->s, buf,
-                                chunk << hw->info.shift, &error) < 0) {
+            if (qpa_simple_read (pa, buf,
+                                 chunk << hw->info.shift, &error) < 0) {
                 qpa_logerr (error, "pa_simple_read failed\n");
                 return NULL;
             }
@@ -283,6 +408,109 @@ static audfmt_e pa_to_audfmt (pa_sample_format_t fmt, int *endianness)
     }
 }
 
+static void context_state_cb (pa_context *c, void *userdata)
+{
+    paaudio *g = &glob_paaudio;
+
+    switch (pa_context_get_state(c)) {
+    case PA_CONTEXT_READY:
+    case PA_CONTEXT_TERMINATED:
+    case PA_CONTEXT_FAILED:
+        pa_threaded_mainloop_signal (g->mainloop, 0);
+        break;
+
+    case PA_CONTEXT_UNCONNECTED:
+    case PA_CONTEXT_CONNECTING:
+    case PA_CONTEXT_AUTHORIZING:
+    case PA_CONTEXT_SETTING_NAME:
+        break;
+    }
+}
+
+static void stream_state_cb (pa_stream *s, void * userdata)
+{
+    paaudio *g = &glob_paaudio;
+
+    switch (pa_stream_get_state (s)) {
+
+    case PA_STREAM_READY:
+    case PA_STREAM_FAILED:
+    case PA_STREAM_TERMINATED:
+        pa_threaded_mainloop_signal (g->mainloop, 0);
+        break;
+
+    case PA_STREAM_UNCONNECTED:
+    case PA_STREAM_CREATING:
+        break;
+    }
+}
+
+static void stream_request_cb (pa_stream *s, size_t length, void *userdata)
+{
+    paaudio *g = &glob_paaudio;
+
+    pa_threaded_mainloop_signal (g->mainloop, 0);
+}
+
+static pa_stream *qpa_simple_new (
+        const char *server,
+        const char *name,
+        pa_stream_direction_t dir,
+        const char *dev,
+        const char *stream_name,
+        const pa_sample_spec *ss,
+        const pa_channel_map *map,
+        const pa_buffer_attr *attr,
+        int *rerror)
+{
+    paaudio *g = &glob_paaudio;
+    int r;
+    pa_stream *stream;
+
+    pa_threaded_mainloop_lock (g->mainloop);
+
+    stream = pa_stream_new (g->context, name, ss, map);
+    if (!stream) {
+        goto fail;
+    }
+
+    pa_stream_set_state_callback (stream, stream_state_cb, g);
+    pa_stream_set_read_callback (stream, stream_request_cb, g);
+    pa_stream_set_write_callback (stream, stream_request_cb, g);
+
+    if (dir == PA_STREAM_PLAYBACK) {
+        r = pa_stream_connect_playback (stream, dev, attr,
+                                        PA_STREAM_INTERPOLATE_TIMING
+                                        |PA_STREAM_ADJUST_LATENCY
+                                        |PA_STREAM_AUTO_TIMING_UPDATE, NULL, NULL);
+    } else {
+        r = pa_stream_connect_record (stream, dev, attr,
+                                      PA_STREAM_INTERPOLATE_TIMING
+                                      |PA_STREAM_ADJUST_LATENCY
+                                      |PA_STREAM_AUTO_TIMING_UPDATE);
+    }
+
+    if (r < 0) {
+      goto fail;
+    }
+
+    pa_threaded_mainloop_unlock (g->mainloop);
+
+    return stream;
+
+fail:
+    pa_threaded_mainloop_unlock (g->mainloop);
+
+    if (stream) {
+        pa_stream_unref (stream);
+    }
+
+    qpa_logerr (pa_context_errno (g->context),
+                "stream_new() failed\n");
+
+    return NULL;
+}
+
 static int qpa_init_out (HWVoiceOut *hw, struct audsettings *as)
 {
     int error;
@@ -306,24 +534,24 @@ static int qpa_init_out (HWVoiceOut *hw, struct audsettings *as)
 
     obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);
 
-    pa->s = pa_simple_new (
-        conf.server,
+    pa->stream = qpa_simple_new (
+        glob_paaudio.server,
         "qemu",
         PA_STREAM_PLAYBACK,
-        conf.sink,
+        glob_paaudio.sink,
         "pcm.playback",
         &ss,
         NULL,                   /* channel map */
         &ba,                    /* buffering attributes */
         &error
         );
-    if (!pa->s) {
+    if (!pa->stream) {
         qpa_logerr (error, "pa_simple_new for playback failed\n");
         goto fail1;
     }
 
     audio_pcm_init_info (&hw->info, &obt_as);
-    hw->samples = conf.samples;
+    hw->samples = glob_paaudio.samples;
     pa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
     pa->rpos = hw->rpos;
     if (!pa->pcm_buf) {
@@ -342,8 +570,10 @@ static int qpa_init_out (HWVoiceOut *hw, struct audsettings *as)
     g_free (pa->pcm_buf);
     pa->pcm_buf = NULL;
  fail2:
-    pa_simple_free (pa->s);
-    pa->s = NULL;
+    if (pa->stream) {
+        pa_stream_unref (pa->stream);
+        pa->stream = NULL;
+    }
  fail1:
     return -1;
 }
@@ -361,24 +591,24 @@ static int qpa_init_in (HWVoiceIn *hw, struct audsettings *as)
 
     obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);
 
-    pa->s = pa_simple_new (
-        conf.server,
+    pa->stream = qpa_simple_new (
+        glob_paaudio.server,
         "qemu",
         PA_STREAM_RECORD,
-        conf.source,
+        glob_paaudio.source,
         "pcm.capture",
         &ss,
         NULL,                   /* channel map */
         NULL,                   /* buffering attributes */
         &error
         );
-    if (!pa->s) {
+    if (!pa->stream) {
         qpa_logerr (error, "pa_simple_new for capture failed\n");
         goto fail1;
     }
 
     audio_pcm_init_info (&hw->info, &obt_as);
-    hw->samples = conf.samples;
+    hw->samples = glob_paaudio.samples;
     pa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
     pa->wpos = hw->wpos;
     if (!pa->pcm_buf) {
@@ -397,8 +627,10 @@ static int qpa_init_in (HWVoiceIn *hw, struct audsettings *as)
     g_free (pa->pcm_buf);
     pa->pcm_buf = NULL;
  fail2:
-    pa_simple_free (pa->s);
-    pa->s = NULL;
+    if (pa->stream) {
+        pa_stream_unref (pa->stream);
+        pa->stream = NULL;
+    }
  fail1:
     return -1;
 }
@@ -413,9 +645,9 @@ static void qpa_fini_out (HWVoiceOut *hw)
     audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC);
     audio_pt_join (&pa->pt, &ret, AUDIO_FUNC);
 
-    if (pa->s) {
-        pa_simple_free (pa->s);
-        pa->s = NULL;
+    if (pa->stream) {
+        pa_stream_unref (pa->stream);
+        pa->stream = NULL;
     }
 
     audio_pt_fini (&pa->pt, AUDIO_FUNC);
@@ -433,9 +665,9 @@ static void qpa_fini_in (HWVoiceIn *hw)
     audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC);
     audio_pt_join (&pa->pt, &ret, AUDIO_FUNC);
 
-    if (pa->s) {
-        pa_simple_free (pa->s);
-        pa->s = NULL;
+    if (pa->stream) {
+        pa_stream_unref (pa->stream);
+        pa->stream = NULL;
     }
 
     audio_pt_fini (&pa->pt, AUDIO_FUNC);
@@ -460,37 +692,106 @@ static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...)
 /* common */
 static void *qpa_audio_init (void)
 {
-    return &conf;
+    paaudio *g = &glob_paaudio;
+
+    g->mainloop = pa_threaded_mainloop_new ();
+    if (!g->mainloop) {
+        goto fail;
+    }
+
+    g->context = pa_context_new (pa_threaded_mainloop_get_api (g->mainloop), glob_paaudio.server);
+    if (!g->context) {
+        goto fail;
+    }
+
+    pa_context_set_state_callback (g->context, context_state_cb, g);
+
+    if (pa_context_connect (g->context, glob_paaudio.server, 0, NULL) < 0) {
+        qpa_logerr (pa_context_errno (g->context),
+                    "pa_context_connect() failed\n");
+        goto fail;
+    }
+
+    pa_threaded_mainloop_lock (g->mainloop);
+
+    if (pa_threaded_mainloop_start (g->mainloop) < 0) {
+        goto unlock_and_fail;
+    }
+
+    for (;;) {
+        pa_context_state_t state;
+
+        state = pa_context_get_state (g->context);
+
+        if (state == PA_CONTEXT_READY) {
+            break;
+        }
+
+        if (!PA_CONTEXT_IS_GOOD (state)) {
+            qpa_logerr (pa_context_errno (g->context),
+                        "Wrong context state\n");
+            goto unlock_and_fail;
+        }
+
+        /* Wait until the context is ready */
+        pa_threaded_mainloop_wait (g->mainloop);
+    }
+
+    pa_threaded_mainloop_unlock (g->mainloop);
+
+    return &glob_paaudio;
+
+unlock_and_fail:
+    pa_threaded_mainloop_unlock (g->mainloop);
+fail:
+    AUD_log (AUDIO_CAP, "Failed to initialize PA context");
+    return NULL;
 }
 
 static void qpa_audio_fini (void *opaque)
 {
-    (void) opaque;
+    paaudio *g = opaque;
+
+    if (g->mainloop) {
+        pa_threaded_mainloop_stop (g->mainloop);
+    }
+
+    if (g->context) {
+        pa_context_disconnect (g->context);
+        pa_context_unref (g->context);
+        g->context = NULL;
+    }
+
+    if (g->mainloop) {
+        pa_threaded_mainloop_free (g->mainloop);
+    }
+
+    g->mainloop = NULL;
 }
 
 struct audio_option qpa_options[] = {
     {
         .name  = "SAMPLES",
         .tag   = AUD_OPT_INT,
-        .valp  = &conf.samples,
+        .valp  = &glob_paaudio.samples,
         .descr = "buffer size in samples"
     },
     {
         .name  = "SERVER",
         .tag   = AUD_OPT_STR,
-        .valp  = &conf.server,
+        .valp  = &glob_paaudio.server,
         .descr = "server address"
     },
     {
         .name  = "SINK",
         .tag   = AUD_OPT_STR,
-        .valp  = &conf.sink,
+        .valp  = &glob_paaudio.sink,
         .descr = "sink device name"
     },
     {
         .name  = "SOURCE",
         .tag   = AUD_OPT_STR,
-        .valp  = &conf.source,
+        .valp  = &glob_paaudio.source,
         .descr = "source device name"
     },
     { /* End of list */ }
-- 
1.7.7.6

[Qemu-devel] [PATCH v4 08/11] configure: pa_simple is not needed anymore

[Marc-André Lureau]

Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
---
 configure |    6 +++---
 1 files changed, 3 insertions(+), 3 deletions(-)

diff --git a/configure b/configure
index 39d2b54..160bf63 100755
--- a/configure
+++ b/configure
@@ -1842,9 +1842,9 @@ for drv in $audio_drv_list; do
     ;;
 
     pa)
-    audio_drv_probe $drv pulse/simple.h "-lpulse-simple -lpulse" \
-        "pa_simple *s = 0; pa_simple_free(s); return 0;"
-    libs_softmmu="-lpulse -lpulse-simple $libs_softmmu"
+    audio_drv_probe $drv pulse/mainloop.h "-lpulse" \
+        "pa_mainloop *m = 0; pa_mainloop_free (m); return 0;"
+    libs_softmmu="-lpulse $libs_softmmu"
     audio_pt_int="yes"
     ;;
 
-- 
1.7.7.6

[Qemu-devel] [PATCH v4 09/11] Allow controlling volume with PulseAudio backend

[Marc-André Lureau]

Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
---
 audio/paaudio.c |   99 ++++++++++++++++++++++++++++++++++++++++++++++++++++---
 1 files changed, 94 insertions(+), 5 deletions(-)

diff --git a/audio/paaudio.c b/audio/paaudio.c
index 6f50c1c..e6708d0 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -677,15 +677,103 @@ static void qpa_fini_in (HWVoiceIn *hw)
 
 static int qpa_ctl_out (HWVoiceOut *hw, int cmd, ...)
 {
-    (void) hw;
-    (void) cmd;
+    PAVoiceOut *pa = (PAVoiceOut *) hw;
+    pa_operation *op;
+    pa_cvolume v;
+    paaudio *g = &glob_paaudio;
+
+    pa_cvolume_init (&v);
+
+    switch (cmd) {
+    case VOICE_VOLUME:
+        {
+            SWVoiceOut *sw;
+            va_list ap;
+
+            va_start (ap, cmd);
+            sw = va_arg (ap, SWVoiceOut *);
+            va_end (ap);
+
+            v.channels = 2;
+            v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.l) / UINT32_MAX;
+            v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.r) / UINT32_MAX;
+
+            pa_threaded_mainloop_lock (g->mainloop);
+
+            op = pa_context_set_sink_input_volume (g->context,
+                pa_stream_get_index (pa->stream),
+                &v, NULL, NULL);
+            if (!op)
+                qpa_logerr (pa_context_errno (g->context),
+                            "set_sink_input_volume() failed\n");
+            else
+                pa_operation_unref (op);
+
+            op = pa_context_set_sink_input_mute (g->context,
+                pa_stream_get_index (pa->stream),
+               sw->vol.mute, NULL, NULL);
+            if (!op) {
+                qpa_logerr (pa_context_errno (g->context),
+                            "set_sink_input_mute() failed\n");
+            } else {
+                pa_operation_unref (op);
+            }
+
+            pa_threaded_mainloop_unlock (g->mainloop);
+        }
+    }
     return 0;
 }
 
 static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...)
 {
-    (void) hw;
-    (void) cmd;
+    PAVoiceIn *pa = (PAVoiceIn *) hw;
+    pa_operation *op;
+    pa_cvolume v;
+    paaudio *g = &glob_paaudio;
+
+    pa_cvolume_init (&v);
+
+    switch (cmd) {
+    case VOICE_VOLUME:
+        {
+            SWVoiceIn *sw;
+            va_list ap;
+
+            va_start (ap, cmd);
+            sw = va_arg (ap, SWVoiceIn *);
+            va_end (ap);
+
+            v.channels = 2;
+            v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.l) / UINT32_MAX;
+            v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.r) / UINT32_MAX;
+
+            pa_threaded_mainloop_lock (g->mainloop);
+
+            /* FIXME: use the upcoming "set_source_output_{volume,mute}" */
+            op = pa_context_set_source_volume_by_index (g->context,
+                pa_stream_get_device_index (pa->stream),
+                &v, NULL, NULL);
+            if (!op) {
+                qpa_logerr (pa_context_errno (g->context),
+                            "set_source_volume() failed\n");
+            } else {
+                pa_operation_unref(op);
+            }
+
+            op = pa_context_set_source_mute_by_index (g->context,
+                pa_stream_get_index (pa->stream),
+                sw->vol.mute, NULL, NULL);
+            if (!op) {
+                qpa_logerr (pa_context_errno (g->context),
+                            "set_source_mute() failed\n");
+            } else {
+                pa_operation_unref (op);
+            }
+
+            pa_threaded_mainloop_unlock (g->mainloop);
+        }
+    }
     return 0;
 }
 
@@ -822,5 +910,6 @@ struct audio_driver pa_audio_driver = {
     .max_voices_out = INT_MAX,
     .max_voices_in  = INT_MAX,
     .voice_size_out = sizeof (PAVoiceOut),
-    .voice_size_in  = sizeof (PAVoiceIn)
+    .voice_size_in  = sizeof (PAVoiceIn),
+    .ctl_caps       = VOICE_VOLUME_CAP
 };
-- 
1.7.7.6

[Qemu-devel] [PATCH v4 10/11] Enable mixemu by default, add runtime option

[Marc-André Lureau]

Without MIXEMU, volume control on the guest doesn't work (except when
volume is applied by guest "emulation", in Win7 for example).

Instead rely on backend volume support, or fallback on mixeng if
backend doesn't support volume control, except if mixemu is disabled
with QEMU_MIXEMU=0.

Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
---
 audio/audio.c  |   10 +++++++++-
 audio/mixeng.c |    6 ------
 configure      |    8 --------
 hw/hda-audio.c |    4 ----
 4 files changed, 9 insertions(+), 19 deletions(-)

diff --git a/audio/audio.c b/audio/audio.c
index bd9237e..bb94133 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -70,6 +70,7 @@ static struct {
     int log_to_monitor;
     int try_poll_in;
     int try_poll_out;
+    int mixemu;
 } conf = {
     .fixed_out = { /* DAC fixed settings */
         .enabled = 1,
@@ -100,6 +101,7 @@ static struct {
     .log_to_monitor = 0,
     .try_poll_in = 1,
     .try_poll_out = 1,
+    .mixemu = 1,
 };
 
 static AudioState glob_audio_state;
@@ -1044,7 +1046,7 @@ int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
     if (swlim) {
         sw->conv (sw->buf, buf, swlim);
 
-        if (!(sw->hw->ctl_caps & VOICE_VOLUME_CAP)) {
+        if (!(sw->hw->ctl_caps & VOICE_VOLUME_CAP) && conf.mixemu) {
             mixeng_volume (sw->buf, swlim, &sw->vol);
         }
     }
@@ -1657,6 +1659,12 @@ static struct audio_option audio_options[] = {
         .valp  = &conf.log_to_monitor,
         .descr = "Print logging messages to monitor instead of stderr"
     },
+    {
+        .name  = "MIXEMU",
+        .tag   = AUD_OPT_BOOL,
+        .valp  = &conf.mixemu,
+        .descr = "Enable mixer emulation (1 - enabled, 0 - disabled)"
+    },
     { /* End of list */ }
 };
 
diff --git a/audio/mixeng.c b/audio/mixeng.c
index 5446be6..0b060e3 100644
--- a/audio/mixeng.c
+++ b/audio/mixeng.c
@@ -336,7 +336,6 @@ void mixeng_clear (struct st_sample *buf, int len)
 
 void mixeng_volume (struct st_sample *buf, int len, struct mixeng_volume *vol)
 {
-#ifdef CONFIG_MIXEMU
     if (vol->mute) {
         mixeng_clear (buf, len);
         return;
@@ -352,9 +351,4 @@ void mixeng_volume (struct st_sample *buf, int len, struct mixeng_volume *vol)
 #endif
         buf += 1;
     }
-#else
-    (void) buf;
-    (void) len;
-    (void) vol;
-#endif
 }
diff --git a/configure b/configure
index 160bf63..09fc408 100755
--- a/configure
+++ b/configure
@@ -177,7 +177,6 @@ darwin_user="no"
 bsd_user="no"
 guest_base=""
 uname_release=""
-mixemu="no"
 aix="no"
 blobs="yes"
 pkgversion=""
@@ -770,8 +769,6 @@ for opt do
   ;;
   --enable-nptl) nptl="yes"
   ;;
-  --enable-mixemu) mixemu="yes"
-  ;;
   --disable-linux-aio) linux_aio="no"
   ;;
   --enable-linux-aio) linux_aio="yes"
@@ -1028,7 +1025,6 @@ echo "  --audio-card-list=LIST   set list of emulated audio cards [$audio_card_l
 echo "                           Available cards: $audio_possible_cards"
 echo "  --block-drv-whitelist=L  set block driver whitelist"
 echo "                           (affects only QEMU, not qemu-img)"
-echo "  --enable-mixemu          enable mixer emulation"
 echo "  --disable-xen            disable xen backend driver support"
 echo "  --enable-xen             enable xen backend driver support"
 echo "  --disable-brlapi         disable BrlAPI"
@@ -2885,7 +2881,6 @@ echo "mingw32 support   $mingw32"
 echo "Audio drivers     $audio_drv_list"
 echo "Extra audio cards $audio_card_list"
 echo "Block whitelist   $block_drv_whitelist"
-echo "Mixer emulation   $mixemu"
 echo "VirtFS support    $virtfs"
 echo "VNC support       $vnc"
 if test "$vnc" = "yes" ; then
@@ -3047,9 +3042,6 @@ if test "$audio_win_int" = "yes" ; then
   echo "CONFIG_AUDIO_WIN_INT=y" >> $config_host_mak
 fi
 echo "CONFIG_BDRV_WHITELIST=$block_drv_whitelist" >> $config_host_mak
-if test "$mixemu" = "yes" ; then
-  echo "CONFIG_MIXEMU=y" >> $config_host_mak
-fi
 if test "$vnc" = "yes" ; then
   echo "CONFIG_VNC=y" >> $config_host_mak
 fi
diff --git a/hw/hda-audio.c b/hw/hda-audio.c
index 8995519..75f1402 100644
--- a/hw/hda-audio.c
+++ b/hw/hda-audio.c
@@ -121,15 +121,11 @@ static void hda_codec_parse_fmt(uint32_t format, struct audsettings *as)
 #define QEMU_HDA_AMP_NONE    (0)
 #define QEMU_HDA_AMP_STEPS   0x4a
 
-#ifdef CONFIG_MIXEMU
 #define QEMU_HDA_AMP_CAPS                                               \
     (AC_AMPCAP_MUTE |                                                   \
      (QEMU_HDA_AMP_STEPS << AC_AMPCAP_OFFSET_SHIFT)    |                \
      (QEMU_HDA_AMP_STEPS << AC_AMPCAP_NUM_STEPS_SHIFT) |                \
      (3                  << AC_AMPCAP_STEP_SIZE_SHIFT))
-#else
-#define QEMU_HDA_AMP_CAPS    QEMU_HDA_AMP_NONE
-#endif
 
 /* common: audio output widget */
 static const desc_param common_params_audio_dac[] = {
-- 
1.7.7.6

[Qemu-devel] [PATCH v4 11/11] audio/rfc: remove PLIVE and PERIOD options

[Marc-André Lureau]

- period seems to be unused now
- plive is very obscure and should either be documented or perhaps removed

Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
---
 audio/audio.c          |   35 -----------------------------------
 audio/audio_template.h |   26 --------------------------
 2 files changed, 0 insertions(+), 61 deletions(-)

diff --git a/audio/audio.c b/audio/audio.c
index bb94133..c2e6e15 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -30,7 +30,6 @@
 #define AUDIO_CAP "audio"
 #include "audio_int.h"
 
-/* #define DEBUG_PLIVE */
 /* #define DEBUG_LIVE */
 /* #define DEBUG_OUT */
 /* #define DEBUG_CAPTURE */
@@ -62,11 +61,6 @@ struct fixed_settings {
 static struct {
     struct fixed_settings fixed_out;
     struct fixed_settings fixed_in;
-    union {
-        int hertz;
-        int64_t ticks;
-    } period;
-    int plive;
     int log_to_monitor;
     int try_poll_in;
     int try_poll_out;
@@ -96,8 +90,6 @@ static struct {
         }
     },
 
-    .period = { .hertz = 250 },
-    .plive = 0,
     .log_to_monitor = 0,
     .try_poll_in = 1,
     .try_poll_out = 1,
@@ -1453,9 +1445,6 @@ static void audio_run_out (AudioState *s)
             while (sw) {
                 sw1 = sw->entries.le_next;
                 if (!sw->active && !sw->callback.fn) {
-#ifdef DEBUG_PLIVE
-                    dolog ("Finishing with old voice\n");
-#endif
                     audio_close_out (sw);
                 }
                 sw = sw1;
@@ -1642,18 +1631,6 @@ static struct audio_option audio_options[] = {
     },
     /* Misc */
     {
-        .name  = "TIMER_PERIOD",
-        .tag   = AUD_OPT_INT,
-        .valp  = &conf.period.hertz,
-        .descr = "Timer period in HZ (0 - use lowest possible)"
-    },
-    {
-        .name  = "PLIVE",
-        .tag   = AUD_OPT_BOOL,
-        .valp  = &conf.plive,
-        .descr = "(undocumented)"
-    },
-    {
         .name  = "LOG_TO_MONITOR",
         .tag   = AUD_OPT_BOOL,
         .valp  = &conf.log_to_monitor,
@@ -1898,18 +1875,6 @@ static void audio_init (void)
         }
     }
 
-    if (conf.period.hertz <= 0) {
-        if (conf.period.hertz < 0) {
-            dolog ("warning: Timer period is negative - %d "
-                   "treating as zero\n",
-                   conf.period.hertz);
-        }
-        conf.period.ticks = 1;
-    } else {
-        conf.period.ticks =
-            muldiv64 (1, get_ticks_per_sec (), conf.period.hertz);
-    }
-
     e = qemu_add_vm_change_state_handler (audio_vm_change_state_handler, s);
     if (!e) {
         dolog ("warning: Could not register change state handler\n"
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 519432a..4120afb 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -433,29 +433,6 @@ SW *glue (AUD_open_, TYPE) (
         return sw;
     }
 
-#ifdef DAC
-    if (conf.plive && sw && (!sw->active && !sw->empty)) {
-        live = sw->total_hw_samples_mixed;
-
-#ifdef DEBUG_PLIVE
-        dolog ("Replacing voice %s with %d live samples\n", SW_NAME (sw), live);
-        dolog ("Old %s freq %d, bits %d, channels %d\n",
-               SW_NAME (sw), sw->info.freq, sw->info.bits, sw->info.nchannels);
-        dolog ("New %s freq %d, bits %d, channels %d\n",
-               name,
-               as->freq,
-               (as->fmt == AUD_FMT_S16 || as->fmt == AUD_FMT_U16) ? 16 : 8,
-               as->nchannels);
-#endif
-
-        if (live) {
-            old_sw = sw;
-            old_sw->callback.fn = NULL;
-            sw = NULL;
-        }
-    }
-#endif
-
     if (!glue (conf.fixed_, TYPE).enabled && sw) {
         glue (AUD_close_, TYPE) (card, sw);
         sw = NULL;
@@ -495,9 +472,6 @@ SW *glue (AUD_open_, TYPE) (
             * old_sw->info.bytes_per_second
             / sw->info.bytes_per_second;
 
-#ifdef DEBUG_PLIVE
-        dolog ("Silence will be mixed %d\n", mixed);
-#endif
         sw->total_hw_samples_mixed += mixed;
     }
 #endif
-- 
1.7.7.6

Re: [Qemu-devel] [PATCH v4 05/11] hw/ac97: add support for volume control

[malc]

On Tue, 13 Mar 2012, Marc-Andr? Lureau wrote:

> Combine output volume with Master and PCM registers values.
> Use default values in mixer_reset ().
> Set volume on post-load to update backend values.
> 
> v4:
> - fix some code style

Don't think it fixes anything...

> 
> Signed-off-by: Marc-Andr? Lureau <marcandre.lureau@redhat.com>
> ---
>  hw/ac97.c |   80 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
>  1 files changed, 80 insertions(+), 0 deletions(-)
> 
> diff --git a/hw/ac97.c b/hw/ac97.c
> index f7866ed..f5ae637 100644
> --- a/hw/ac97.c
> +++ b/hw/ac97.c
> @@ -436,6 +436,64 @@ static void reset_voices (AC97LinkState *s, uint8_t active[LAST_INDEX])
>      AUD_set_active_in (s->voice_mc, active[MC_INDEX]);
>  }
>  
> +static void get_volume (uint16_t vol, uint16_t mask, int inverse,
> +                        int *mute, uint8_t *lvol, uint8_t *rvol)
> +{
> +  *mute = (vol >> MUTE_SHIFT) & 1;
> +  *rvol = (255 * (vol & mask)) / mask;
> +  *lvol = (255 * ((vol >> 8) & mask)) / mask;
> +  if (inverse) {
> +    *rvol = 255 - *rvol;
> +    *lvol = 255 - *lvol;

Here.

[..snip..]

-- 
mailto:av1474@comtv.ru

Re: [Qemu-devel] [PATCH v4 11/11] audio/rfc: remove PLIVE and PERIOD options

[malc]

On Tue, 13 Mar 2012, Marc-Andr? Lureau wrote:

> - period seems to be unused now
> - plive is very obscure and should either be documented or perhaps removed

Plive is obscure because the use case was, in any case it's been years
since i've used it so it can, probably, safely go away.

Period on the other hand is unused because i somehow missed the subtle
change of behavior in one of the patches made by, i think, Gerd.

What i know would like to know is this: Jan, back in the day,
advocated for making the mixeng the default (for musicpal), so, Jan -
can you test this series?

[..snip..]

-- 
mailto:av1474@comtv.ru

Re: [Qemu-devel] [PATCH v4 05/11] hw/ac97: add support for volume control

[Marc-André Lureau]

On Tue, Mar 13, 2012 at 4:37 PM, malc <av1474@comtv.ru> wrote:
> On Tue, 13 Mar 2012, Marc-Andr? Lureau wrote:
>
>> Combine output volume with Master and PCM registers values.
>> Use default values in mixer_reset ().
>> Set volume on post-load to update backend values.
>>
>> v4:
>> - fix some code style
>
> Don't think it fixes anything...
>
>>
>> Signed-off-by: Marc-Andr? Lureau <marcandre.lureau@redhat.com>
>> ---
>>  hw/ac97.c |   80 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
>>  1 files changed, 80 insertions(+), 0 deletions(-)
>>
>> diff --git a/hw/ac97.c b/hw/ac97.c
>> index f7866ed..f5ae637 100644
>> --- a/hw/ac97.c
>> +++ b/hw/ac97.c
>> @@ -436,6 +436,64 @@ static void reset_voices (AC97LinkState *s, uint8_t active[LAST_INDEX])
>>      AUD_set_active_in (s->voice_mc, active[MC_INDEX]);
>>  }
>>
>> +static void get_volume (uint16_t vol, uint16_t mask, int inverse,
>> +                        int *mute, uint8_t *lvol, uint8_t *rvol)
>> +{
>> +  *mute = (vol >> MUTE_SHIFT) & 1;
>> +  *rvol = (255 * (vol & mask)) / mask;
>> +  *lvol = (255 * ((vol >> 8) & mask)) / mask;
>> +  if (inverse) {
>> +    *rvol = 255 - *rvol;
>> +    *lvol = 255 - *lvol;
>
> Here.
>
> [..snip..]

checkpatch doesn't complain here, I only get 30 warnings:

WARNING: space prohibited between function name and open parenthesis '('

which seems to be the code style in audio/

thanks

-- 
Marc-André Lureau

Re: [Qemu-devel] [PATCH v4 05/11] hw/ac97: add support for volume control

[malc]

[-- Attachment #1: Type: TEXT/PLAIN, Size: 1647 bytes --]

On Tue, 13 Mar 2012, Marc-Andr? Lureau wrote:

> On Tue, Mar 13, 2012 at 4:37 PM, malc <av1474@comtv.ru> wrote:
> > On Tue, 13 Mar 2012, Marc-Andr? Lureau wrote:
> >
> >> Combine output volume with Master and PCM registers values.
> >> Use default values in mixer_reset ().
> >> Set volume on post-load to update backend values.
> >>
> >> v4:
> >> - fix some code style
> >
> > Don't think it fixes anything...
> >
> >>
> >> Signed-off-by: Marc-Andr? Lureau <marcandre.lureau@redhat.com>
> >> ---
> >>  hw/ac97.c |   80 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
> >>  1 files changed, 80 insertions(+), 0 deletions(-)
> >>
> >> diff --git a/hw/ac97.c b/hw/ac97.c
> >> index f7866ed..f5ae637 100644
> >> --- a/hw/ac97.c
> >> +++ b/hw/ac97.c
> >> @@ -436,6 +436,64 @@ static void reset_voices (AC97LinkState *s, uint8_t active[LAST_INDEX])
> >>      AUD_set_active_in (s->voice_mc, active[MC_INDEX]);
> >>  }
> >>
> >> +static void get_volume (uint16_t vol, uint16_t mask, int inverse,
> >> +                        int *mute, uint8_t *lvol, uint8_t *rvol)
> >> +{
> >> +  *mute = (vol >> MUTE_SHIFT) & 1;
> >> +  *rvol = (255 * (vol & mask)) / mask;
> >> +  *lvol = (255 * ((vol >> 8) & mask)) / mask;
> >> +  if (inverse) {
> >> +    *rvol = 255 - *rvol;
> >> +    *lvol = 255 - *lvol;
> >
> > Here.
> >
> > [..snip..]
> 
> checkpatch doesn't complain here, I only get 30 warnings:
> 
> WARNING: space prohibited between function name and open parenthesis '('
> 
> which seems to be the code style in audio/
> 

That's right. But the whole get_volume function is indented with 2
instead of 4 spaces.

-- 
mailto:av1474@comtv.ru

Re: [Qemu-devel] [PATCH v4 05/11] hw/ac97: add support for volume control

[Marc-André Lureau]

On Tue, Mar 13, 2012 at 5:07 PM, malc <av1474@comtv.ru> wrote:
> That's right. But the whole get_volume function is indented with 2
> instead of 4 spaces.

fixed! thanks

-- 
Marc-André Lureau

Re: [Qemu-devel] [PATCH v4 11/11] audio/rfc: remove PLIVE and PERIOD options

[Gerd Hoffmann]

On 03/13/12 16:47, malc wrote:
> On Tue, 13 Mar 2012, Marc-Andr? Lureau wrote:
> 
>> - period seems to be unused now

> Period on the other hand is unused because i somehow missed the subtle
> change of behavior in one of the patches made by, i think, Gerd.

Uhm, which patch?  I think that wasn't intentional, at least I can't
remember intentionally disabling period.  Maybe I missed the subtle
change of behavior too.

I do see the point in having this configurable as this is a cpu overhead
vs. latency tradeoff which one might want to tweak depending on the use
case.

If we keep it we should pass it down to the audio backends I think.
pulseaudio for example uses buffer sizes adjusted for the default period
(see qpa_init_out()), when running with a non-default period pulseaudio
should be able to adjust the buffer sizes accordingly.

cheers,
  Gerd

Re: [Qemu-devel] [PATCH v4 11/11] audio/rfc: remove PLIVE and PERIOD options

[Marc-André Lureau]

Hi

On Wed, Mar 14, 2012 at 10:22 AM, Gerd Hoffmann <kraxel@redhat.com> wrote:
> On 03/13/12 16:47, malc wrote:
>> On Tue, 13 Mar 2012, Marc-Andr? Lureau wrote:
>>
>>> - period seems to be unused now
>
>> Period on the other hand is unused because i somehow missed the subtle
>> change of behavior in one of the patches made by, i think, Gerd.
>
> Uhm, which patch?  I think that wasn't intentional, at least I can't
> remember intentionally disabling period.  Maybe I missed the subtle
> change of behavior too.

Could be:

39deb1e496de81957167daebf5cf5d1fbd5e47c2

> I do see the point in having this configurable as this is a cpu overhead
> vs. latency tradeoff which one might want to tweak depending on the use
> case.

But that would impact a/v sync, or can you report added latency back
to the guest somehow? I imagine audio backend latency should depend on
the configured device buffering/latency, not on an environment tweak.

regards

-- 
Marc-André Lureau

Re: [Qemu-devel] [PATCH v4 11/11] audio/rfc: remove PLIVE and PERIOD options

[Gerd Hoffmann]

On 03/14/12 12:20, Marc-André Lureau wrote:
> Hi
> 
> On Wed, Mar 14, 2012 at 10:22 AM, Gerd Hoffmann <kraxel@redhat.com> wrote:
>> On 03/13/12 16:47, malc wrote:
>>> On Tue, 13 Mar 2012, Marc-Andr? Lureau wrote:
>>>
>>>> - period seems to be unused now
>>
>>> Period on the other hand is unused because i somehow missed the subtle
>>> change of behavior in one of the patches made by, i think, Gerd.
>>
>> Uhm, which patch?  I think that wasn't intentional, at least I can't
>> remember intentionally disabling period.  Maybe I missed the subtle
>> change of behavior too.
> 
> Could be:
> 
> 39deb1e496de81957167daebf5cf5d1fbd5e47c2

Yes, looks like this one is it.

>> I do see the point in having this configurable as this is a cpu overhead
>> vs. latency tradeoff which one might want to tweak depending on the use
>> case.
> 
> But that would impact a/v sync, or can you report added latency back
> to the guest somehow? I imagine audio backend latency should depend on
> the configured device buffering/latency, not on an environment tweak.

Sure, with lower latency you get better a/v sync too.

Guest interfacing is next to impossible I think, simply because real
hardware has no need for that and thus the interfaces simply don't exist
in the hardware we are emulating.  We can try to fix that with a virtio
soundcard which has such interfaces, but it could be this simply shifts
the issue from the driver/hardware interface to the os-kernel/driver
interface.

cheers,
  Gerd

Re: [Qemu-devel] [PATCH v4 11/11] audio/rfc: remove PLIVE and PERIOD options

[Marc-André Lureau]

Hi

> Guest interfacing is next to impossible I think, simply because real
> hardware has no need for that and thus the interfaces simply don't
> exist
> in the hardware we are emulating.  We can try to fix that with a
> virtio
> soundcard which has such interfaces, but it could be this simply
> shifts
> the issue from the driver/hardware interface to the os-kernel/driver
> interface.

btw, the Xen folks just merged a PulseAudio sink/src implementation a few days ago: http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/?id=50a7bf1175eaf07521c00bde8eed2f820e64437f