Merge remote-tracking branch 'remotes/kraxel/tags/audio-20191018-pull-request' into staging

audio: bugfixes, pa connection and stream naming.
audio: 5.1/7.1 support for alsa, pa and usb-audio.

# gpg: Signature made Fri 18 Oct 2019 08:41:26 BST
# 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-20191018-pull-request:
  paaudio: fix channel order for usb-audio 5.1 and 7.1 streams
  usbaudio: change playback counters to 64 bit
  usb-audio: support more than two channels of audio
  usb-audio: do not count on avail bytes actually available
  audio: basic support for multichannel audio
  audio: replace shift in audio_pcm_info with bytes_per_frame
  audio: support more than two channels in volume setting
  paaudio: get/put_buffer functions
  audio: make mixeng optional
  audio: add mixing-engine option (documentation)
  audio: paaudio: ability to specify stream name
  audio: paaudio: fix connection and stream name
  audio: fix parameter dereference before NULL check
Signed-off-by: NPeter Maydell <peter.maydell@linaro.org>

audio/alsaaudio.c

@@ -493,13 +493,6 @@ static int alsa_open(bool in, struct alsa_params_req *req,
         goto err;
     }
 
-    if (nchannels != 1 && nchannels != 2) {
-        alsa_logerr2 (err, typ,
-                      "Can not handle obtained number of channels %d\n",
-                      nchannels);
-        goto err;
-    }
-
     if (apdo->buffer_length) {
         int dir = 0;
         unsigned int btime = apdo->buffer_length;
@@ -602,7 +595,7 @@ static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
 {
     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
     size_t pos = 0;
-    size_t len_frames = len >> hw->info.shift;
+    size_t len_frames = len / hw->info.bytes_per_frame;
 
     while (len_frames) {
         char *src = advance(buf, pos);
@@ -648,7 +641,7 @@ static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
             }
         }
 
-        pos += written << hw->info.shift;
+        pos += written * hw->info.bytes_per_frame;
         if (written < len_frames) {
             break;
         }
@@ -802,7 +795,8 @@ static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
         void *dst = advance(buf, pos);
         snd_pcm_sframes_t nread;
 
-        nread = snd_pcm_readi(alsa->handle, dst, len >> hw->info.shift);
+        nread = snd_pcm_readi(
+            alsa->handle, dst, len / hw->info.bytes_per_frame);
 
         if (nread <= 0) {
             switch (nread) {
@@ -828,8 +822,8 @@ static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
             }
         }
 
-        pos += nread << hw->info.shift;
-        len -= nread << hw->info.shift;
+        pos += nread * hw->info.bytes_per_frame;
+        len -= nread * hw->info.bytes_per_frame;
     }
 
     return pos;

audio/audio.c

@@ -242,7 +242,7 @@ static int audio_validate_settings (struct audsettings *as)
 {
     int invalid;
 
-    invalid = as->nchannels != 1 && as->nchannels != 2;
+    invalid = as->nchannels < 1;
     invalid |= as->endianness != 0 && as->endianness != 1;
 
     switch (as->fmt) {
@@ -299,12 +299,13 @@ static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *a
 
 void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
 {
-    int bits = 8, sign = 0, shift = 0;
+    int bits = 8, sign = 0, mul;
 
     switch (as->fmt) {
     case AUDIO_FORMAT_S8:
         sign = 1;
     case AUDIO_FORMAT_U8:
+        mul = 1;
         break;
 
     case AUDIO_FORMAT_S16:
@@ -312,7 +313,7 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
         /* fall through */
     case AUDIO_FORMAT_U16:
         bits = 16;
-        shift = 1;
+        mul = 2;
         break;
 
     case AUDIO_FORMAT_S32:
@@ -320,7 +321,7 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
         /* fall through */
     case AUDIO_FORMAT_U32:
         bits = 32;
-        shift = 2;
+        mul = 4;
         break;
 
     default:
@@ -331,9 +332,8 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
     info->bits = bits;
     info->sign = sign;
     info->nchannels = as->nchannels;
-    info->shift = (as->nchannels == 2) + shift;
-    info->align = (1 << info->shift) - 1;
-    info->bytes_per_second = info->freq << info->shift;
+    info->bytes_per_frame = as->nchannels * mul;
+    info->bytes_per_second = info->freq * info->bytes_per_frame;
     info->swap_endianness = (as->endianness != AUDIO_HOST_ENDIANNESS);
 }
 
@@ -344,26 +344,25 @@ void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
     }
 
     if (info->sign) {
-        memset (buf, 0x00, len << info->shift);
+        memset(buf, 0x00, len * info->bytes_per_frame);
     }
     else {
         switch (info->bits) {
         case 8:
-            memset (buf, 0x80, len << info->shift);
+            memset(buf, 0x80, len * info->bytes_per_frame);
             break;
 
         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++) {
+                for (i = 0; i < len * info->nchannels; i++) {
                     p[i] = s;
                 }
             }
@@ -373,14 +372,13 @@ void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
             {
                 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++) {
+                for (i = 0; i < len * info->nchannels; i++) {
                     p[i] = s;
                 }
             }
@@ -558,7 +556,7 @@ static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len)
 
     while (len) {
         st_sample *src = hw->mix_buf->samples + pos;
-        uint8_t *dst = advance(pcm_buf, clipped << hw->info.shift);
+        uint8_t *dst = advance(pcm_buf, clipped * hw->info.bytes_per_frame);
         size_t samples_till_end_of_buf = hw->mix_buf->size - pos;
         size_t samples_to_clip = MIN(len, samples_till_end_of_buf);
 
@@ -607,7 +605,7 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
         return 0;
     }
 
-    samples = size >> sw->info.shift;
+    samples = size / sw->info.bytes_per_frame;
     if (!live) {
         return 0;
     }
@@ -642,7 +640,7 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
 
     sw->clip (buf, sw->buf, ret);
     sw->total_hw_samples_acquired += total;
-    return ret << sw->info.shift;
+    return ret * sw->info.bytes_per_frame;
 }
 
 /*
@@ -715,7 +713,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
     }
 
     wpos = (sw->hw->mix_buf->pos + live) % hwsamples;
-    samples = size >> sw->info.shift;
+    samples = size / sw->info.bytes_per_frame;
 
     dead = hwsamples - live;
     swlim = ((int64_t) dead << 32) / sw->ratio;
@@ -759,13 +757,13 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
     dolog (
         "%s: write size %zu ret %zu total sw %zu\n",
         SW_NAME (sw),
-        size >> sw->info.shift,
+        size / sw->info.bytes_per_frame,
         ret,
         sw->total_hw_samples_mixed
         );
 #endif
 
-    return ret << sw->info.shift;
+    return ret * sw->info.bytes_per_frame;
 }
 
 #ifdef DEBUG_AUDIO
@@ -838,37 +836,51 @@ static void audio_timer (void *opaque)
  */
 size_t AUD_write(SWVoiceOut *sw, void *buf, size_t size)
 {
+    HWVoiceOut *hw;
+
     if (!sw) {
         /* XXX: Consider options */
         return size;
     }
+    hw = sw->hw;
 
-    if (!sw->hw->enabled) {
+    if (!hw->enabled) {
         dolog ("Writing to disabled voice %s\n", SW_NAME (sw));
         return 0;
     }
 
-    return audio_pcm_sw_write(sw, buf, size);
+    if (audio_get_pdo_out(hw->s->dev)->mixing_engine) {
+        return audio_pcm_sw_write(sw, buf, size);
+    } else {
+        return hw->pcm_ops->write(hw, buf, size);
+    }
 }
 
 size_t AUD_read(SWVoiceIn *sw, void *buf, size_t size)
 {
+    HWVoiceIn *hw;
+
     if (!sw) {
         /* XXX: Consider options */
         return size;
     }
+    hw = sw->hw;
 
-    if (!sw->hw->enabled) {
+    if (!hw->enabled) {
         dolog ("Reading from disabled voice %s\n", SW_NAME (sw));
         return 0;
     }
 
-    return audio_pcm_sw_read(sw, buf, size);
+    if (audio_get_pdo_in(hw->s->dev)->mixing_engine) {
+        return audio_pcm_sw_read(sw, buf, size);
+    } else {
+        return hw->pcm_ops->read(hw, buf, size);
+    }
 }
 
 int AUD_get_buffer_size_out (SWVoiceOut *sw)
 {
-    return sw->hw->mix_buf->size << sw->hw->info.shift;
+    return sw->hw->mix_buf->size * sw->hw->info.bytes_per_frame;
 }
 
 void AUD_set_active_out (SWVoiceOut *sw, int on)
@@ -984,10 +996,10 @@ static size_t audio_get_avail (SWVoiceIn *sw)
     ldebug (
         "%s: get_avail live %d ret %" PRId64 "\n",
         SW_NAME (sw),
-        live, (((int64_t) live << 32) / sw->ratio) << sw->info.shift
+        live, (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame
         );
 
-    return (((int64_t) live << 32) / sw->ratio) << sw->info.shift;
+    return (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame;
 }
 
 static size_t audio_get_free(SWVoiceOut *sw)
@@ -1011,10 +1023,11 @@ static size_t audio_get_free(SWVoiceOut *sw)
 #ifdef DEBUG_OUT
     dolog ("%s: get_free live %d dead %d ret %" PRId64 "\n",
            SW_NAME (sw),
-           live, dead, (((int64_t) dead << 32) / sw->ratio) << sw->info.shift);
+           live, dead, (((int64_t) dead << 32) / sw->ratio) *
+           sw->info.bytes_per_frame);
 #endif
 
-    return (((int64_t) dead << 32) / sw->ratio) << sw->info.shift;
+    return (((int64_t) dead << 32) / sw->ratio) * sw->info.bytes_per_frame;
 }
 
 static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
@@ -1033,7 +1046,7 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
             while (n) {
                 size_t till_end_of_hw = hw->mix_buf->size - rpos2;
                 size_t to_write = MIN(till_end_of_hw, n);
-                size_t bytes = to_write << hw->info.shift;
+                size_t bytes = to_write * hw->info.bytes_per_frame;
                 size_t written;
 
                 sw->buf = hw->mix_buf->samples + rpos2;
@@ -1068,10 +1081,11 @@ static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
             return clipped + live;
         }
 
-        decr = MIN(size >> hw->info.shift, live);
+        decr = MIN(size / hw->info.bytes_per_frame, live);
         audio_pcm_hw_clip_out(hw, buf, decr);
-        proc = hw->pcm_ops->put_buffer_out(hw, buf, decr << hw->info.shift) >>
-            hw->info.shift;
+        proc = hw->pcm_ops->put_buffer_out(hw, buf,
+                                           decr * hw->info.bytes_per_frame) /
+            hw->info.bytes_per_frame;
 
         live -= proc;
         clipped += proc;
@@ -1090,6 +1104,26 @@ static void audio_run_out (AudioState *s)
     HWVoiceOut *hw = NULL;
     SWVoiceOut *sw;
 
+    if (!audio_get_pdo_out(s->dev)->mixing_engine) {
+        while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
+            /* there is exactly 1 sw for each hw with no mixeng */
+            sw = hw->sw_head.lh_first;
+
+            if (hw->pending_disable) {
+                hw->enabled = 0;
+                hw->pending_disable = 0;
+                if (hw->pcm_ops->enable_out) {
+                    hw->pcm_ops->enable_out(hw, false);
+                }
+            }
+
+            if (sw->active) {
+                sw->callback.fn(sw->callback.opaque, INT_MAX);
+            }
+        }
+        return;
+    }
+
     while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
         size_t played, live, prev_rpos, free;
         int nb_live, cleanup_required;
@@ -1200,16 +1234,16 @@ static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
 
     while (samples) {
         size_t proc;
-        size_t size = samples << hw->info.shift;
+        size_t size = samples * hw->info.bytes_per_frame;
         void *buf = hw->pcm_ops->get_buffer_in(hw, &size);
 
-        assert((size & hw->info.align) == 0);
+        assert(size % hw->info.bytes_per_frame == 0);
         if (size == 0) {
             hw->pcm_ops->put_buffer_in(hw, buf, size);
             break;
         }
 
-        proc = MIN(size >> hw->info.shift,
+        proc = MIN(size / hw->info.bytes_per_frame,
                    conv_buf->size - conv_buf->pos);
 
         hw->conv(conv_buf->samples + conv_buf->pos, buf, proc);
@@ -1217,7 +1251,7 @@ static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
 
         samples -= proc;
         conv += proc;
-        hw->pcm_ops->put_buffer_in(hw, buf, proc << hw->info.shift);
+        hw->pcm_ops->put_buffer_in(hw, buf, proc * hw->info.bytes_per_frame);
     }
 
     return conv;
@@ -1227,6 +1261,17 @@ static void audio_run_in (AudioState *s)
 {
     HWVoiceIn *hw = NULL;
 
+    if (!audio_get_pdo_in(s->dev)->mixing_engine) {
+        while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
+            /* there is exactly 1 sw for each hw with no mixeng */
+            SWVoiceIn *sw = hw->sw_head.lh_first;
+            if (sw->active) {
+                sw->callback.fn(sw->callback.opaque, INT_MAX);
+            }
+        }
+        return;
+    }
+
     while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
         SWVoiceIn *sw;
         size_t captured = 0, min;
@@ -1280,7 +1325,7 @@ static void audio_run_capture (AudioState *s)
 
             for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
                 cb->ops.capture (cb->opaque, cap->buf,
-                                 to_capture << hw->info.shift);
+                                 to_capture * hw->info.bytes_per_frame);
             }
             rpos = (rpos + to_capture) % hw->mix_buf->size;
             live -= to_capture;
@@ -1333,7 +1378,7 @@ void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size)
     ssize_t start;
 
     if (unlikely(!hw->buf_emul)) {
-        size_t calc_size = hw->conv_buf->size << hw->info.shift;
+        size_t calc_size = hw->conv_buf->size * hw->info.bytes_per_frame;
         hw->buf_emul = g_malloc(calc_size);
         hw->size_emul = calc_size;
         hw->pos_emul = hw->pending_emul = 0;
@@ -1369,7 +1414,7 @@ void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size)
 void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size)
 {
     if (unlikely(!hw->buf_emul)) {
-        size_t calc_size = hw->mix_buf->size << hw->info.shift;
+        size_t calc_size = hw->mix_buf->size * hw->info.bytes_per_frame;
 
         hw->buf_emul = g_malloc(calc_size);
         hw->size_emul = calc_size;
@@ -1751,6 +1796,11 @@ CaptureVoiceOut *AUD_add_capture(
         s = audio_init(NULL, NULL);
     }
 
+    if (!audio_get_pdo_out(s->dev)->mixing_engine) {
+        dolog("Can't capture with mixeng disabled\n");
+        return NULL;
+    }
+
     if (audio_validate_settings (as)) {
         dolog ("Invalid settings were passed when trying to add capture\n");
         audio_print_settings (as);
@@ -1783,7 +1833,7 @@ CaptureVoiceOut *AUD_add_capture(
 
         audio_pcm_init_info (&hw->info, as);
 
-        cap->buf = g_malloc0_n(hw->mix_buf->size, 1 << hw->info.shift);
+        cap->buf = g_malloc0_n(hw->mix_buf->size, hw->info.bytes_per_frame);
 
         hw->clip = mixeng_clip
             [hw->info.nchannels == 2]
@@ -1841,31 +1891,45 @@ void AUD_del_capture (CaptureVoiceOut *cap, void *cb_opaque)
 }
 
 void AUD_set_volume_out (SWVoiceOut *sw, int mute, uint8_t lvol, uint8_t rvol)
+{
+    Volume vol = { .mute = mute, .channels = 2, .vol = { lvol, rvol } };
+    audio_set_volume_out(sw, &vol);
+}
+
+void audio_set_volume_out(SWVoiceOut *sw, Volume *vol)
 {
     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;
+        sw->vol.mute = vol->mute;
+        sw->vol.l = nominal_volume.l * vol->vol[0] / 255;
+        sw->vol.r = nominal_volume.l * vol->vol[vol->channels > 1 ? 1 : 0] /
+            255;
 
         if (hw->pcm_ops->volume_out) {
-            hw->pcm_ops->volume_out(hw, &sw->vol);
+            hw->pcm_ops->volume_out(hw, vol);
         }
     }
 }
 
 void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
+{
+    Volume vol = { .mute = mute, .channels = 2, .vol = { lvol, rvol } };
+    audio_set_volume_in(sw, &vol);
+}
+
+void audio_set_volume_in(SWVoiceIn *sw, Volume *vol)
 {
     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;
+        sw->vol.mute = vol->mute;
+        sw->vol.l = nominal_volume.l * vol->vol[0] / 255;
+        sw->vol.r = nominal_volume.r * vol->vol[vol->channels > 1 ? 1 : 0] /
+            255;
 
         if (hw->pcm_ops->volume_in) {
-            hw->pcm_ops->volume_in(hw, &sw->vol);
+            hw->pcm_ops->volume_in(hw, vol);
         }
     }
 }
@@ -1905,9 +1969,13 @@ void audio_create_pdos(Audiodev *dev)
 static void audio_validate_per_direction_opts(
     AudiodevPerDirectionOptions *pdo, Error **errp)
 {
+    if (!pdo->has_mixing_engine) {
+        pdo->has_mixing_engine = true;
+        pdo->mixing_engine = true;
+    }
     if (!pdo->has_fixed_settings) {
         pdo->has_fixed_settings = true;
-        pdo->fixed_settings = true;
+        pdo->fixed_settings = pdo->mixing_engine;
     }
     if (!pdo->fixed_settings &&
         (pdo->has_frequency || pdo->has_channels || pdo->has_format)) {
@@ -1915,6 +1983,10 @@ static void audio_validate_per_direction_opts(
                    "You can't use frequency, channels or format with fixed-settings=off");
         return;
     }
+    if (!pdo->mixing_engine && pdo->fixed_settings) {
+        error_setg(errp, "You can't use fixed-settings without mixeng");
+        return;
+    }
 
     if (!pdo->has_frequency) {
         pdo->has_frequency = true;
@@ -1926,7 +1998,7 @@ static void audio_validate_per_direction_opts(
     }
     if (!pdo->has_voices) {
         pdo->has_voices = true;
-        pdo->voices = 1;
+        pdo->voices = pdo->mixing_engine ? 1 : INT_MAX;
     }
     if (!pdo->has_format) {
         pdo->has_format = true;
@@ -2081,14 +2153,14 @@ size_t audio_rate_get_bytes(struct audio_pcm_info *info, RateCtl *rate,
     now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
     ticks = now - rate->start_ticks;
     bytes = muldiv64(ticks, info->bytes_per_second, NANOSECONDS_PER_SECOND);
-    samples = (bytes - rate->bytes_sent) >> info->shift;
+    samples = (bytes - rate->bytes_sent) / info->bytes_per_frame;
     if (samples < 0 || samples > 65536) {
         AUD_log(NULL, "Resetting rate control (%" PRId64 " samples)\n", samples);
         audio_rate_start(rate);
         samples = 0;
     }
 
-    ret = MIN(samples << info->shift, bytes_avail);
+    ret = MIN(samples * info->bytes_per_frame, bytes_avail);
     rate->bytes_sent += ret;
     return ret;
 }

audio/audio.h

@@ -124,6 +124,16 @@ 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);
 
+#define AUDIO_MAX_CHANNELS 16
+typedef struct Volume {
+    bool mute;
+    int channels;
+    uint8_t vol[AUDIO_MAX_CHANNELS];
+} Volume;
+
+void audio_set_volume_out(SWVoiceOut *sw, Volume *vol);
+void audio_set_volume_in(SWVoiceIn *sw, Volume *vol);
+
 SWVoiceIn *AUD_open_in (
     QEMUSoundCard *card,
     SWVoiceIn *sw,

audio/audio_int.h

@@ -43,8 +43,7 @@ struct audio_pcm_info {
     int sign;
     int freq;
     int nchannels;
-    int align;
-    int shift;
+    int bytes_per_frame;
     int bytes_per_second;
     int swap_endianness;
 };
@@ -166,7 +165,7 @@ struct audio_pcm_ops {
      */
     size_t (*put_buffer_out)(HWVoiceOut *hw, void *buf, size_t size);
     void   (*enable_out)(HWVoiceOut *hw, bool enable);
-    void   (*volume_out)(HWVoiceOut *hw, struct mixeng_volume *vol);
+    void   (*volume_out)(HWVoiceOut *hw, Volume *vol);
 
     int    (*init_in) (HWVoiceIn *hw, audsettings *as, void *drv_opaque);
     void   (*fini_in) (HWVoiceIn *hw);
@@ -174,7 +173,7 @@ struct audio_pcm_ops {
     void  *(*get_buffer_in)(HWVoiceIn *hw, size_t *size);
     void   (*put_buffer_in)(HWVoiceIn *hw, void *buf, size_t size);
     void   (*enable_in)(HWVoiceIn *hw, bool enable);
-    void   (*volume_in)(HWVoiceIn *hw, struct mixeng_volume *vol);
+    void   (*volume_in)(HWVoiceIn *hw, Volume *vol);
 };
 
 void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size);

audio/audio_template.h

@@ -78,13 +78,17 @@ static void glue (audio_pcm_hw_free_resources_, TYPE) (HW *hw)
 
 static void glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw)
 {
-    size_t samples = hw->samples;
-    if (audio_bug(__func__, samples == 0)) {
-        dolog("Attempted to allocate empty buffer\n");
-    }
+    if (glue(audio_get_pdo_, TYPE)(hw->s->dev)->mixing_engine) {
+        size_t samples = hw->samples;
+        if (audio_bug(__func__, samples == 0)) {
+            dolog("Attempted to allocate empty buffer\n");
+        }
 
-    HWBUF = g_malloc0(sizeof(STSampleBuffer) + sizeof(st_sample) * samples);
-    HWBUF->size = samples;
+        HWBUF = g_malloc0(sizeof(STSampleBuffer) + sizeof(st_sample) * samples);
+        HWBUF->size = samples;
+    } else {
+        HWBUF = NULL;
+    }
 }
 
 static void glue (audio_pcm_sw_free_resources_, TYPE) (SW *sw)
@@ -103,6 +107,10 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw)
 {
     int samples;
 
+    if (!glue(audio_get_pdo_, TYPE)(sw->s->dev)->mixing_engine) {
+        return 0;
+    }
+
     samples = ((int64_t) sw->HWBUF->size << 32) / sw->ratio;
 
     sw->buf = audio_calloc(__func__, samples, sizeof(struct st_sample));
@@ -328,9 +336,9 @@ static HW *glue(audio_pcm_hw_add_, TYPE)(AudioState *s, struct audsettings *as)
     HW *hw;
     AudiodevPerDirectionOptions *pdo = glue(audio_get_pdo_, TYPE)(s->dev);
 
-    if (pdo->fixed_settings) {
+    if (!pdo->mixing_engine || pdo->fixed_settings) {
         hw = glue(audio_pcm_hw_add_new_, TYPE)(s, as);
-        if (hw) {
+        if (!pdo->mixing_engine || hw) {
             return hw;
         }
     }
@@ -425,8 +433,8 @@ SW *glue (AUD_open_, TYPE) (
     struct audsettings *as
     )
 {
-    AudioState *s = card->state;
-    AudiodevPerDirectionOptions *pdo = glue(audio_get_pdo_, TYPE)(s->dev);
+    AudioState *s;
+    AudiodevPerDirectionOptions *pdo;
 
     if (audio_bug(__func__, !card || !name || !callback_fn || !as)) {
         dolog ("card=%p name=%p callback_fn=%p as=%p\n",
@@ -434,6 +442,9 @@ SW *glue (AUD_open_, TYPE) (
         goto fail;
     }
 
+    s = card->state;
+    pdo = glue(audio_get_pdo_, TYPE)(s->dev);
+
     ldebug ("open %s, freq %d, nchannels %d, fmt %d\n",
             name, as->freq, as->nchannels, as->fmt);
 

audio/coreaudio.c

@@ -440,7 +440,7 @@ static OSStatus audioDeviceIOProc(
     }
 
     frameCount = core->audioDevicePropertyBufferFrameSize;
-    pending_frames = hw->pending_emul >> hw->info.shift;
+    pending_frames = hw->pending_emul / hw->info.bytes_per_frame;
 
     /* if there are not enough samples, set signal and return */
     if (pending_frames < frameCount) {
@@ -449,7 +449,7 @@ static OSStatus audioDeviceIOProc(
         return 0;
     }
 
-    len = frameCount << hw->info.shift;
+    len = frameCount * hw->info.bytes_per_frame;
     while (len) {
         size_t write_len;
         ssize_t start = ((ssize_t) hw->pos_emul) - hw->pending_emul;

audio/dsound_template.h

@@ -98,8 +98,8 @@ static int glue (dsound_lock_, TYPE) (
         goto fail;
     }
 
-    if ((p1p && *p1p && (*blen1p & info->align)) ||
-        (p2p && *p2p && (*blen2p & info->align))) {
+    if ((p1p && *p1p && (*blen1p % info->bytes_per_frame)) ||
+        (p2p && *p2p && (*blen2p % info->bytes_per_frame))) {
         dolog("DirectSound returned misaligned buffer %ld %ld\n",
               *blen1p, *blen2p);
         glue(dsound_unlock_, TYPE)(buf, *p1p, p2p ? *p2p : NULL, *blen1p,
@@ -247,14 +247,14 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
     obt_as.endianness = 0;
     audio_pcm_init_info (&hw->info, &obt_as);
 
-    if (bc.dwBufferBytes & hw->info.align) {
+    if (bc.dwBufferBytes % hw->info.bytes_per_frame) {
         dolog (
             "GetCaps returned misaligned buffer size %ld, alignment %d\n",
-            bc.dwBufferBytes, hw->info.align + 1
+            bc.dwBufferBytes, hw->info.bytes_per_frame
             );
     }
     hw->size_emul = bc.dwBufferBytes;
-    hw->samples = bc.dwBufferBytes >> hw->info.shift;
+    hw->samples = bc.dwBufferBytes / hw->info.bytes_per_frame;
     ds->s = s;
 
 #ifdef DEBUG_DSOUND

audio/dsoundaudio.c

@@ -320,8 +320,8 @@ static void dsound_clear_sample (HWVoiceOut *hw, LPDIRECTSOUNDBUFFER dsb,
         return;
     }
 
-    len1 = blen1 >> hw->info.shift;
-    len2 = blen2 >> hw->info.shift;
+    len1 = blen1 / hw->info.bytes_per_frame;
+    len2 = blen2 / hw->info.bytes_per_frame;
 
 #ifdef DEBUG_DSOUND
     dolog ("clear %p,%ld,%ld %p,%ld,%ld\n",

audio/noaudio.c

@@ -91,7 +91,7 @@ static size_t no_read(HWVoiceIn *hw, void *buf, size_t size)
     NoVoiceIn *no = (NoVoiceIn *) hw;
     int64_t bytes = audio_rate_get_bytes(&hw->info, &no->rate, size);
 
-    audio_pcm_info_clear_buf(&hw->info, buf, bytes >> hw->info.shift);
+    audio_pcm_info_clear_buf(&hw->info, buf, bytes / hw->info.bytes_per_frame);
     return bytes;
 }
 

audio/ossaudio.c

@@ -506,16 +506,16 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
     oss->nfrags = obt.nfrags;
     oss->fragsize = obt.fragsize;
 
-    if (obt.nfrags * obt.fragsize & hw->info.align) {
+    if (obt.nfrags * obt.fragsize % hw->info.bytes_per_frame) {
         dolog ("warning: Misaligned DAC buffer, size %d, alignment %d\n",
-               obt.nfrags * obt.fragsize, hw->info.align + 1);
+               obt.nfrags * obt.fragsize, hw->info.bytes_per_frame);
     }
 
-    hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
+    hw->samples = (obt.nfrags * obt.fragsize) / hw->info.bytes_per_frame;
 
     oss->mmapped = 0;
     if (oopts->has_try_mmap && oopts->try_mmap) {
-        hw->size_emul = hw->samples << hw->info.shift;
+        hw->size_emul = hw->samples * hw->info.bytes_per_frame;
         hw->buf_emul = mmap(
             NULL,
             hw->size_emul,
@@ -644,12 +644,12 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
     oss->nfrags = obt.nfrags;
     oss->fragsize = obt.fragsize;
 
-    if (obt.nfrags * obt.fragsize & hw->info.align) {
+    if (obt.nfrags * obt.fragsize % hw->info.bytes_per_frame) {
         dolog ("warning: Misaligned ADC buffer, size %d, alignment %d\n",
-               obt.nfrags * obt.fragsize, hw->info.align + 1);
+               obt.nfrags * obt.fragsize, hw->info.bytes_per_frame);
     }
 
-    hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
+    hw->samples = (obt.nfrags * obt.fragsize) / hw->info.bytes_per_frame;
 
     oss->fd = fd;
     oss->dev = dev;

audio/paaudio.c

@@ -2,6 +2,7 @@
 
 #include "qemu/osdep.h"
 #include "qemu/module.h"
+#include "qemu-common.h"
 #include "audio.h"
 #include "qapi/opts-visitor.h"
 
@@ -98,6 +99,59 @@ static inline int PA_STREAM_IS_GOOD(pa_stream_state_t x)
         }                                                               \
     } while (0)
 
+static void *qpa_get_buffer_in(HWVoiceIn *hw, size_t *size)
+{
+    PAVoiceIn *p = (PAVoiceIn *) hw;
+    PAConnection *c = p->g->conn;
+    int r;
+
+    pa_threaded_mainloop_lock(c->mainloop);
+
+    CHECK_DEAD_GOTO(c, p->stream, unlock_and_fail,
+                    "pa_threaded_mainloop_lock failed\n");
+
+    if (!p->read_length) {
+        r = pa_stream_peek(p->stream, &p->read_data, &p->read_length);
+        CHECK_SUCCESS_GOTO(c, r == 0, unlock_and_fail,
+                           "pa_stream_peek failed\n");
+    }
+
+    *size = MIN(p->read_length, *size);
+
+    pa_threaded_mainloop_unlock(c->mainloop);
+    return (void *) p->read_data;
+
+unlock_and_fail:
+    pa_threaded_mainloop_unlock(c->mainloop);
+    *size = 0;
+    return NULL;
+}
+
+static void qpa_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size)
+{
+    PAVoiceIn *p = (PAVoiceIn *) hw;
+    PAConnection *c = p->g->conn;
+    int r;
+
+    pa_threaded_mainloop_lock(c->mainloop);
+
+    CHECK_DEAD_GOTO(c, p->stream, unlock,
+                    "pa_threaded_mainloop_lock failed\n");
+
+    assert(buf == p->read_data && size <= p->read_length);
+
+    p->read_data += size;
+    p->read_length -= size;
+
+    if (size && !p->read_length) {
+        r = pa_stream_drop(p->stream);
+        CHECK_SUCCESS_GOTO(c, r == 0, unlock, "pa_stream_drop failed\n");
+    }
+
+unlock:
+    pa_threaded_mainloop_unlock(c->mainloop);
+}
+
 static size_t qpa_read(HWVoiceIn *hw, void *data, size_t length)
 {
     PAVoiceIn *p = (PAVoiceIn *) hw;
@@ -136,6 +190,32 @@ unlock_and_fail:
     return 0;
 }
 
+static void *qpa_get_buffer_out(HWVoiceOut *hw, size_t *size)
+{
+    PAVoiceOut *p = (PAVoiceOut *) hw;
+    PAConnection *c = p->g->conn;
+    void *ret;
+    int r;
+
+    pa_threaded_mainloop_lock(c->mainloop);
+
+    CHECK_DEAD_GOTO(c, p->stream, unlock_and_fail,
+                    "pa_threaded_mainloop_lock failed\n");
+
+    *size = -1;
+    r = pa_stream_begin_write(p->stream, &ret, size);
+    CHECK_SUCCESS_GOTO(c, r >= 0, unlock_and_fail,
+                       "pa_stream_begin_write failed\n");
+
+    pa_threaded_mainloop_unlock(c->mainloop);
+    return ret;
+
+unlock_and_fail:
+    pa_threaded_mainloop_unlock(c->mainloop);
+    *size = 0;
+    return NULL;
+}
+
 static size_t qpa_write(HWVoiceOut *hw, void *data, size_t length)
 {
     PAVoiceOut *p = (PAVoiceOut *) hw;
@@ -259,17 +339,59 @@ static pa_stream *qpa_simple_new (
         pa_stream_direction_t dir,
         const char *dev,
         const pa_sample_spec *ss,
-        const pa_channel_map *map,
         const pa_buffer_attr *attr,
         int *rerror)
 {
     int r;
-    pa_stream *stream;
+    pa_stream *stream = NULL;
     pa_stream_flags_t flags;
+    pa_channel_map map;
 
     pa_threaded_mainloop_lock(c->mainloop);
 
-    stream = pa_stream_new(c->context, name, ss, map);
+    pa_channel_map_init(&map);
+    map.channels = ss->channels;
+
+    /*
+     * TODO: This currently expects the only frontend supporting more than 2
+     * channels is the usb-audio.  We will need some means to set channel
+     * order when a new frontend gains multi-channel support.
+     */
+    switch (ss->channels) {
+    case 1:
+        map.map[0] = PA_CHANNEL_POSITION_MONO;
+        break;
+
+    case 2:
+        map.map[0] = PA_CHANNEL_POSITION_LEFT;
+        map.map[1] = PA_CHANNEL_POSITION_RIGHT;
+        break;
+
+    case 6:
+        map.map[0] = PA_CHANNEL_POSITION_FRONT_LEFT;
+        map.map[1] = PA_CHANNEL_POSITION_FRONT_RIGHT;
+        map.map[2] = PA_CHANNEL_POSITION_CENTER;
+        map.map[3] = PA_CHANNEL_POSITION_LFE;
+        map.map[4] = PA_CHANNEL_POSITION_REAR_LEFT;
+        map.map[5] = PA_CHANNEL_POSITION_REAR_RIGHT;
+        break;
+
+    case 8:
+        map.map[0] = PA_CHANNEL_POSITION_FRONT_LEFT;
+        map.map[1] = PA_CHANNEL_POSITION_FRONT_RIGHT;
+        map.map[2] = PA_CHANNEL_POSITION_CENTER;
+        map.map[3] = PA_CHANNEL_POSITION_LFE;
+        map.map[4] = PA_CHANNEL_POSITION_REAR_LEFT;
+        map.map[5] = PA_CHANNEL_POSITION_REAR_RIGHT;
+        map.map[6] = PA_CHANNEL_POSITION_SIDE_LEFT;
+        map.map[7] = PA_CHANNEL_POSITION_SIDE_RIGHT;
+
+    default:
+        dolog("Internal error: unsupported channel count %d\n", ss->channels);
+        goto fail;
+    }
+
+    stream = pa_stream_new(c->context, name, ss, &map);
     if (!stream) {
         goto fail;
     }
@@ -338,11 +460,10 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
 
     pa->stream = qpa_simple_new (
         c,
-        "qemu",
+        ppdo->has_stream_name ? ppdo->stream_name : g->dev->id,
         PA_STREAM_PLAYBACK,
         ppdo->has_name ? ppdo->name : NULL,
         &ss,
-        NULL,                   /* channel map */
         &ba,                    /* buffering attributes */
         &error
         );
@@ -387,11 +508,10 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
 
     pa->stream = qpa_simple_new (
         c,
-        "qemu",
+        ppdo->has_stream_name ? ppdo->stream_name : g->dev->id,
         PA_STREAM_RECORD,
         ppdo->has_name ? ppdo->name : NULL,
         &ss,
-        NULL,                   /* channel map */
         &ba,                    /* buffering attributes */
         &error
         );
@@ -452,20 +572,22 @@ static void qpa_fini_in (HWVoiceIn *hw)
     }
 }
 
-static void qpa_volume_out(HWVoiceOut *hw, struct mixeng_volume *vol)
+static void qpa_volume_out(HWVoiceOut *hw, Volume *vol)
 {
     PAVoiceOut *pa = (PAVoiceOut *) hw;
     pa_operation *op;
     pa_cvolume v;
     PAConnection *c = pa->g->conn;
+    int i;
 
 #ifdef PA_CHECK_VERSION    /* macro is present in 0.9.16+ */
     pa_cvolume_init (&v);  /* function is present in 0.9.13+ */
 #endif
 
-    v.channels = 2;
-    v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->l) / UINT32_MAX;
-    v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->r) / UINT32_MAX;
+    v.channels = vol->channels;
+    for (i = 0; i < vol->channels; ++i) {
+        v.values[i] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->vol[i]) / 255;
+    }
 
     pa_threaded_mainloop_lock(c->mainloop);
 
@@ -492,20 +614,22 @@ static void qpa_volume_out(HWVoiceOut *hw, struct mixeng_volume *vol)
     pa_threaded_mainloop_unlock(c->mainloop);
 }
 
-static void qpa_volume_in(HWVoiceIn *hw, struct mixeng_volume *vol)
+static void qpa_volume_in(HWVoiceIn *hw, Volume *vol)
 {
     PAVoiceIn *pa = (PAVoiceIn *) hw;
     pa_operation *op;
     pa_cvolume v;
     PAConnection *c = pa->g->conn;
+    int i;
 
 #ifdef PA_CHECK_VERSION
     pa_cvolume_init (&v);
 #endif
 
-    v.channels = 2;
-    v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->l) / UINT32_MAX;
-    v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->r) / UINT32_MAX;
+    v.channels = vol->channels;
+    for (i = 0; i < vol->channels; ++i) {
+        v.values[i] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->vol[i]) / 255;
+    }
 
     pa_threaded_mainloop_lock(c->mainloop);
 
@@ -549,6 +673,7 @@ static int qpa_validate_per_direction_opts(Audiodev *dev,
 /* common */
 static void *qpa_conn_init(const char *server)
 {
+    const char *vm_name;
     PAConnection *c = g_malloc0(sizeof(PAConnection));
     QTAILQ_INSERT_TAIL(&pa_conns, c, list);
 
@@ -557,8 +682,9 @@ static void *qpa_conn_init(const char *server)
         goto fail;
     }
 
+    vm_name = qemu_get_vm_name();
     c->context = pa_context_new(pa_threaded_mainloop_get_api(c->mainloop),
-                                server);
+                                vm_name ? vm_name : "qemu");
     if (!c->context) {
         goto fail;
     }
@@ -698,11 +824,15 @@ static struct audio_pcm_ops qpa_pcm_ops = {
     .init_out = qpa_init_out,
     .fini_out = qpa_fini_out,
     .write    = qpa_write,
+    .get_buffer_out = qpa_get_buffer_out,
+    .put_buffer_out = qpa_write, /* pa handles it */
     .volume_out = qpa_volume_out,
 
     .init_in  = qpa_init_in,
     .fini_in  = qpa_fini_in,
     .read     = qpa_read,
+    .get_buffer_in = qpa_get_buffer_in,
+    .put_buffer_in = qpa_put_buffer_in,
     .volume_in = qpa_volume_in
 };
 

audio/spiceaudio.c

@@ -131,7 +131,8 @@ static void *line_out_get_buffer(HWVoiceOut *hw, size_t *size)
 
     if (out->frame) {
         *size = audio_rate_get_bytes(
-            &hw->info, &out->rate, (out->fsize - out->fpos) << hw->info.shift);
+            &hw->info, &out->rate,
+            (out->fsize - out->fpos) * hw->info.bytes_per_frame);
     } else {
         audio_rate_start(&out->rate);
     }
@@ -179,13 +180,14 @@ static void line_out_enable(HWVoiceOut *hw, bool enable)
 }
 
 #if ((SPICE_INTERFACE_PLAYBACK_MAJOR >= 1) && (SPICE_INTERFACE_PLAYBACK_MINOR >= 2))
-static void line_out_volume(HWVoiceOut *hw, struct mixeng_volume *vol)
+static void line_out_volume(HWVoiceOut *hw, Volume *vol)
 {
     SpiceVoiceOut *out = container_of(hw, SpiceVoiceOut, hw);
     uint16_t svol[2];
 
-    svol[0] = vol->l / ((1ULL << 16) + 1);
-    svol[1] = vol->r / ((1ULL << 16) + 1);
+    assert(vol->channels == 2);
+    svol[0] = vol->vol[0] * 257;
+    svol[1] = vol->vol[1] * 257;
     spice_server_playback_set_volume(&out->sin, 2, svol);
     spice_server_playback_set_mute(&out->sin, vol->mute);
 }
@@ -262,13 +264,14 @@ static void line_in_enable(HWVoiceIn *hw, bool enable)
 }
 
 #if ((SPICE_INTERFACE_RECORD_MAJOR >= 2) && (SPICE_INTERFACE_RECORD_MINOR >= 2))
-static void line_in_volume(HWVoiceIn *hw, struct mixeng_volume *vol)
+static void line_in_volume(HWVoiceIn *hw, Volume *vol)
 {
     SpiceVoiceIn *in = container_of(hw, SpiceVoiceIn, hw);
     uint16_t svol[2];
 
-    svol[0] = vol->l / ((1ULL << 16) + 1);
-    svol[1] = vol->r / ((1ULL << 16) + 1);
+    assert(vol->channels == 2);
+    svol[0] = vol->vol[0] * 257;
+    svol[1] = vol->vol[1] * 257;
     spice_server_record_set_volume(&in->sin, 2, svol);
     spice_server_record_set_mute(&in->sin, vol->mute);
 }

audio/wavaudio.c

@@ -43,14 +43,14 @@ static size_t wav_write_out(HWVoiceOut *hw, void *buf, size_t len)
 {
     WAVVoiceOut *wav = (WAVVoiceOut *) hw;
     int64_t bytes = audio_rate_get_bytes(&hw->info, &wav->rate, len);
-    assert(bytes >> hw->info.shift << hw->info.shift == bytes);
+    assert(bytes % hw->info.bytes_per_frame == 0);
 
     if (bytes && fwrite(buf, bytes, 1, wav->f) != 1) {
         dolog("wav_write_out: fwrite of %" PRId64 " bytes failed\nReason: %s\n",
               bytes, strerror(errno));
     }
 
-    wav->total_samples += bytes >> hw->info.shift;
+    wav->total_samples += bytes / hw->info.bytes_per_frame;
     return bytes;
 }
 
@@ -134,7 +134,7 @@ static void wav_fini_out (HWVoiceOut *hw)
     WAVVoiceOut *wav = (WAVVoiceOut *) hw;
     uint8_t rlen[4];
     uint8_t dlen[4];
-    uint32_t datalen = wav->total_samples << hw->info.shift;
+    uint32_t datalen = wav->total_samples * hw->info.bytes_per_frame;
     uint32_t rifflen = datalen + 36;
 
     if (!wav->f) {

qapi/audio.json

@@ -11,6 +11,11 @@
 # General audio backend options that are used for both playback and
 # recording.
 #
+# @mixing-engine: use QEMU's mixing engine to mix all streams inside QEMU and
+#                 convert audio formats when not supported by the backend. When
+#                 set to off, fixed-settings must be also off (default on,
+#                 since 4.2)
+#
 # @fixed-settings: use fixed settings for host input/output. When off,
 #                  frequency, channels and format must not be
 #                  specified (default true)
@@ -31,6 +36,7 @@
 ##
 { 'struct': 'AudiodevPerDirectionOptions',
   'data': {
+    '*mixing-engine':  'bool',
     '*fixed-settings': 'bool',
     '*frequency':      'uint32',
     '*channels':       'uint32',
@@ -206,6 +212,11 @@
 #
 # @name: name of the sink/source to use
 #
+# @stream-name: name of the PulseAudio stream created by qemu.  Can be
+#               used to identify the stream in PulseAudio when you
+#               create multiple PulseAudio devices or run multiple qemu
+#               instances (default: audiodev's id, since 4.2)
+#
 # @latency: latency you want PulseAudio to achieve in microseconds
 #           (default 15000)
 #
@@ -215,6 +226,7 @@
   'base': 'AudiodevPerDirectionOptions',
   'data': {
     '*name': 'str',
+    '*stream-name': 'str',
     '*latency': 'uint32' } }
 
 ##

qemu-options.hx

@@ -433,6 +433,7 @@ DEF("audiodev", HAS_ARG, QEMU_OPTION_audiodev,
     "                specifies the audio backend to use\n"
     "                id= identifier of the backend\n"
     "                timer-period= timer period in microseconds\n"
+    "                in|out.mixing-engine= use mixing engine to mix streams inside QEMU\n"
     "                in|out.fixed-settings= use fixed settings for host audio\n"
     "                in|out.frequency= frequency to use with fixed settings\n"
     "                in|out.channels= number of channels to use with fixed settings\n"
@@ -493,6 +494,10 @@ output's property with @code{out.@var{prop}}. For example:
 -audiodev alsa,id=example,out.channels=1 # leaves in.channels unspecified
 @end example
 
+NOTE: parameter validation is known to be incomplete, in many cases
+specifying an invalid option causes QEMU to print an error message and
+continue emulation without sound.
+
 Valid global options are:
 
 @table @option
@@ -503,6 +508,16 @@ Identifies the audio backend.
 Sets the timer @var{period} used by the audio subsystem in microseconds.
 Default is 10000 (10 ms).
 
+@item in|out.mixing-engine=on|off
+Use QEMU's mixing engine to mix all streams inside QEMU and convert
+audio formats when not supported by the backend.  When off,
+@var{fixed-settings} must be off too.  Note that disabling this option
+means that the selected backend must support multiple streams and the
+audio formats used by the virtual cards, otherwise you'll get no sound.
+It's not recommended to disable this option unless you want to use 5.1
+or 7.1 audio, as mixing engine only supports mono and stereo audio.
+Default is on.
+
 @item in|out.fixed-settings=on|off
 Use fixed settings for host audio.  When off, it will change based on
 how the guest opens the sound card.  In this case you must not specify