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: Peter Maydell <peter.maydell@linaro.org>
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index cfe4228..f37ce1c 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -493,13 +493,6 @@
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 @@
{
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 @@
}
}
- pos += written << hw->info.shift;
+ pos += written * hw->info.bytes_per_frame;
if (written < len_frames) {
break;
}
@@ -802,7 +795,8 @@
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 @@
}
}
- 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;
diff --git a/audio/audio.c b/audio/audio.c
index 7128ee9..7fc3aa9 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -242,7 +242,7 @@
{
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 @@
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 @@
/* fall through */
case AUDIO_FORMAT_U16:
bits = 16;
- shift = 1;
+ mul = 2;
break;
case AUDIO_FORMAT_S32:
@@ -320,7 +321,7 @@
/* fall through */
case AUDIO_FORMAT_U32:
bits = 32;
- shift = 2;
+ mul = 4;
break;
default:
@@ -331,9 +332,8 @@
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 @@
}
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 @@
{
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 @@
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 @@
return 0;
}
- samples = size >> sw->info.shift;
+ samples = size / sw->info.bytes_per_frame;
if (!live) {
return 0;
}
@@ -642,7 +640,7 @@
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 @@
}
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 @@
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 @@
*/
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 @@
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 @@
#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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
{
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 @@
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 @@
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_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 @@
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 @@
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]
@@ -1842,30 +1892,44 @@
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 @@
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 @@
"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 @@
}
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 @@
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;
}
diff --git a/audio/audio.h b/audio/audio.h
index c74abb8..0db3c7d 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -124,6 +124,16 @@
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,
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 22a703c..5ba2078 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -43,8 +43,7 @@
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 @@
*/
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 @@
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);
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 235d1ac..3287d70 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -78,13 +78,17 @@
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 @@
{
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 @@
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 @@
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 @@
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);
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index 1427c9f..66f0f45 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -440,7 +440,7 @@
}
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 @@
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;
diff --git a/audio/dsound_template.h b/audio/dsound_template.h
index 9f10b68..7a15f91 100644
--- a/audio/dsound_template.h
+++ b/audio/dsound_template.h
@@ -98,8 +98,8 @@
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 @@
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
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index d4a4757..c265c00 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -320,8 +320,8 @@
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",
diff --git a/audio/noaudio.c b/audio/noaudio.c
index ec8a287..ff99b25 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -91,7 +91,7 @@
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;
}
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 0c4451e..c43faee 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -506,16 +506,16 @@
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 @@
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;
diff --git a/audio/paaudio.c b/audio/paaudio.c
index ed31f86..df541a7 100644
--- a/audio/paaudio.c
+++ b/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 @@
} \
} 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 @@
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 @@
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 @@
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 @@
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_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 @@
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 @@
/* 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 @@
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 @@
.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
};
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index 9860f9c..b6b5da4 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -131,7 +131,8 @@
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 @@
}
#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 @@
}
#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);
}
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 47efdc1..e46d834 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -43,14 +43,14 @@
{
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 @@
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) {
diff --git a/hw/usb/dev-audio.c b/hw/usb/dev-audio.c
index ae42e5a..ea604bb 100644
--- a/hw/usb/dev-audio.c
+++ b/hw/usb/dev-audio.c
@@ -37,11 +37,15 @@
#include "desc.h"
#include "audio/audio.h"
+static void usb_audio_reinit(USBDevice *dev, unsigned channels);
+
#define USBAUDIO_VENDOR_NUM 0x46f4 /* CRC16() of "QEMU" */
#define USBAUDIO_PRODUCT_NUM 0x0002
#define DEV_CONFIG_VALUE 1 /* The one and only */
+#define USBAUDIO_MAX_CHANNELS(s) (s->multi ? 8 : 2)
+
/* Descriptor subtypes for AC interfaces */
#define DST_AC_HEADER 1
#define DST_AC_INPUT_TERMINAL 2
@@ -80,6 +84,27 @@
[STRING_REAL_STREAM] = "Audio Output - 48 kHz Stereo",
};
+/*
+ * A USB audio device supports an arbitrary number of alternate
+ * interface settings for each interface. Each corresponds to a block
+ * diagram of parameterized blocks. This can thus refer to things like
+ * number of channels, data rates, or in fact completely different
+ * block diagrams. Alternative setting 0 is always the null block diagram,
+ * which is used by a disabled device.
+ */
+enum usb_audio_altset {
+ ALTSET_OFF = 0x00, /* No endpoint */
+ ALTSET_STEREO = 0x01, /* Single endpoint */
+ ALTSET_51 = 0x02,
+ ALTSET_71 = 0x03,
+};
+
+static unsigned altset_channels[] = {
+ [ALTSET_STEREO] = 2,
+ [ALTSET_51] = 6,
+ [ALTSET_71] = 8,
+};
+
#define U16(x) ((x) & 0xff), (((x) >> 8) & 0xff)
#define U24(x) U16(x), (((x) >> 16) & 0xff)
#define U32(x) U24(x), (((x) >> 24) & 0xff)
@@ -87,7 +112,8 @@
/*
* A Basic Audio Device uses these specific values
*/
-#define USBAUDIO_PACKET_SIZE 192
+#define USBAUDIO_PACKET_SIZE_BASE 96
+#define USBAUDIO_PACKET_SIZE(channels) (USBAUDIO_PACKET_SIZE_BASE * channels)
#define USBAUDIO_SAMPLE_RATE 48000
#define USBAUDIO_PACKET_INTERVAL 1
@@ -121,7 +147,7 @@
0x01, /* u8 bTerminalID */
U16(0x0101), /* u16 wTerminalType */
0x00, /* u8 bAssocTerminal */
- 0x02, /* u16 bNrChannels */
+ 0x02, /* u8 bNrChannels */
U16(0x0003), /* u16 wChannelConfig */
0x00, /* u8 iChannelNames */
STRING_INPUT_TERMINAL, /* u8 iTerminal */
@@ -156,14 +182,14 @@
},
},{
.bInterfaceNumber = 1,
- .bAlternateSetting = 0,
+ .bAlternateSetting = ALTSET_OFF,
.bNumEndpoints = 0,
.bInterfaceClass = USB_CLASS_AUDIO,
.bInterfaceSubClass = USB_SUBCLASS_AUDIO_STREAMING,
.iInterface = STRING_NULL_STREAM,
},{
.bInterfaceNumber = 1,
- .bAlternateSetting = 1,
+ .bAlternateSetting = ALTSET_STEREO,
.bNumEndpoints = 1,
.bInterfaceClass = USB_CLASS_AUDIO,
.bInterfaceSubClass = USB_SUBCLASS_AUDIO_STREAMING,
@@ -199,7 +225,7 @@
{
.bEndpointAddress = USB_DIR_OUT | 0x01,
.bmAttributes = 0x0d,
- .wMaxPacketSize = USBAUDIO_PACKET_SIZE,
+ .wMaxPacketSize = USBAUDIO_PACKET_SIZE(2),
.bInterval = 1,
.is_audio = 1,
/* Stereo Headphone Class-specific
@@ -247,17 +273,274 @@
.str = usb_audio_stringtable,
};
-/*
- * A USB audio device supports an arbitrary number of alternate
- * interface settings for each interface. Each corresponds to a block
- * diagram of parameterized blocks. This can thus refer to things like
- * number of channels, data rates, or in fact completely different
- * block diagrams. Alternative setting 0 is always the null block diagram,
- * which is used by a disabled device.
- */
-enum usb_audio_altset {
- ALTSET_OFF = 0x00, /* No endpoint */
- ALTSET_ON = 0x01, /* Single endpoint */
+/* multi channel compatible desc */
+
+static const USBDescIface desc_iface_multi[] = {
+ {
+ .bInterfaceNumber = 0,
+ .bNumEndpoints = 0,
+ .bInterfaceClass = USB_CLASS_AUDIO,
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL,
+ .bInterfaceProtocol = 0x04,
+ .iInterface = STRING_USBAUDIO_CONTROL,
+ .ndesc = 4,
+ .descs = (USBDescOther[]) {
+ {
+ /* Headphone Class-Specific AC Interface Header Descriptor */
+ .data = (uint8_t[]) {
+ 0x09, /* u8 bLength */
+ USB_DT_CS_INTERFACE, /* u8 bDescriptorType */
+ DST_AC_HEADER, /* u8 bDescriptorSubtype */
+ U16(0x0100), /* u16 bcdADC */
+ U16(0x38), /* u16 wTotalLength */
+ 0x01, /* u8 bInCollection */
+ 0x01, /* u8 baInterfaceNr */
+ }
+ },{
+ /* Generic Stereo Input Terminal ID1 Descriptor */
+ .data = (uint8_t[]) {
+ 0x0c, /* u8 bLength */
+ USB_DT_CS_INTERFACE, /* u8 bDescriptorType */
+ DST_AC_INPUT_TERMINAL, /* u8 bDescriptorSubtype */
+ 0x01, /* u8 bTerminalID */
+ U16(0x0101), /* u16 wTerminalType */
+ 0x00, /* u8 bAssocTerminal */
+ 0x08, /* u8 bNrChannels */
+ U16(0x063f), /* u16 wChannelConfig */
+ 0x00, /* u8 iChannelNames */
+ STRING_INPUT_TERMINAL, /* u8 iTerminal */
+ }
+ },{
+ /* Generic Stereo Feature Unit ID2 Descriptor */
+ .data = (uint8_t[]) {
+ 0x19, /* u8 bLength */
+ USB_DT_CS_INTERFACE, /* u8 bDescriptorType */
+ DST_AC_FEATURE_UNIT, /* u8 bDescriptorSubtype */
+ 0x02, /* u8 bUnitID */
+ 0x01, /* u8 bSourceID */
+ 0x02, /* u8 bControlSize */
+ U16(0x0001), /* u16 bmaControls(0) */
+ U16(0x0002), /* u16 bmaControls(1) */
+ U16(0x0002), /* u16 bmaControls(2) */
+ U16(0x0002), /* u16 bmaControls(3) */
+ U16(0x0002), /* u16 bmaControls(4) */
+ U16(0x0002), /* u16 bmaControls(5) */
+ U16(0x0002), /* u16 bmaControls(6) */
+ U16(0x0002), /* u16 bmaControls(7) */
+ U16(0x0002), /* u16 bmaControls(8) */
+ STRING_FEATURE_UNIT, /* u8 iFeature */
+ }
+ },{
+ /* Headphone Ouptut Terminal ID3 Descriptor */
+ .data = (uint8_t[]) {
+ 0x09, /* u8 bLength */
+ USB_DT_CS_INTERFACE, /* u8 bDescriptorType */
+ DST_AC_OUTPUT_TERMINAL, /* u8 bDescriptorSubtype */
+ 0x03, /* u8 bUnitID */
+ U16(0x0301), /* u16 wTerminalType (SPK) */
+ 0x00, /* u8 bAssocTerminal */
+ 0x02, /* u8 bSourceID */
+ STRING_OUTPUT_TERMINAL, /* u8 iTerminal */
+ }
+ }
+ },
+ },{
+ .bInterfaceNumber = 1,
+ .bAlternateSetting = ALTSET_OFF,
+ .bNumEndpoints = 0,
+ .bInterfaceClass = USB_CLASS_AUDIO,
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIO_STREAMING,
+ .iInterface = STRING_NULL_STREAM,
+ },{
+ .bInterfaceNumber = 1,
+ .bAlternateSetting = ALTSET_STEREO,
+ .bNumEndpoints = 1,
+ .bInterfaceClass = USB_CLASS_AUDIO,
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIO_STREAMING,
+ .iInterface = STRING_REAL_STREAM,
+ .ndesc = 2,
+ .descs = (USBDescOther[]) {
+ {
+ /* Headphone Class-specific AS General Interface Descriptor */
+ .data = (uint8_t[]) {
+ 0x07, /* u8 bLength */
+ USB_DT_CS_INTERFACE, /* u8 bDescriptorType */
+ DST_AS_GENERAL, /* u8 bDescriptorSubtype */
+ 0x01, /* u8 bTerminalLink */
+ 0x00, /* u8 bDelay */
+ 0x01, 0x00, /* u16 wFormatTag */
+ }
+ },{
+ /* Headphone Type I Format Type Descriptor */
+ .data = (uint8_t[]) {
+ 0x0b, /* u8 bLength */
+ USB_DT_CS_INTERFACE, /* u8 bDescriptorType */
+ DST_AS_FORMAT_TYPE, /* u8 bDescriptorSubtype */
+ 0x01, /* u8 bFormatType */
+ 0x02, /* u8 bNrChannels */
+ 0x02, /* u8 bSubFrameSize */
+ 0x10, /* u8 bBitResolution */
+ 0x01, /* u8 bSamFreqType */
+ U24(USBAUDIO_SAMPLE_RATE), /* u24 tSamFreq */
+ }
+ }
+ },
+ .eps = (USBDescEndpoint[]) {
+ {
+ .bEndpointAddress = USB_DIR_OUT | 0x01,
+ .bmAttributes = 0x0d,
+ .wMaxPacketSize = USBAUDIO_PACKET_SIZE(2),
+ .bInterval = 1,
+ .is_audio = 1,
+ /* Stereo Headphone Class-specific
+ AS Audio Data Endpoint Descriptor */
+ .extra = (uint8_t[]) {
+ 0x07, /* u8 bLength */
+ USB_DT_CS_ENDPOINT, /* u8 bDescriptorType */
+ DST_EP_GENERAL, /* u8 bDescriptorSubtype */
+ 0x00, /* u8 bmAttributes */
+ 0x00, /* u8 bLockDelayUnits */
+ U16(0x0000), /* u16 wLockDelay */
+ },
+ },
+ }
+ },{
+ .bInterfaceNumber = 1,
+ .bAlternateSetting = ALTSET_51,
+ .bNumEndpoints = 1,
+ .bInterfaceClass = USB_CLASS_AUDIO,
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIO_STREAMING,
+ .iInterface = STRING_REAL_STREAM,
+ .ndesc = 2,
+ .descs = (USBDescOther[]) {
+ {
+ /* Headphone Class-specific AS General Interface Descriptor */
+ .data = (uint8_t[]) {
+ 0x07, /* u8 bLength */
+ USB_DT_CS_INTERFACE, /* u8 bDescriptorType */
+ DST_AS_GENERAL, /* u8 bDescriptorSubtype */
+ 0x01, /* u8 bTerminalLink */
+ 0x00, /* u8 bDelay */
+ 0x01, 0x00, /* u16 wFormatTag */
+ }
+ },{
+ /* Headphone Type I Format Type Descriptor */
+ .data = (uint8_t[]) {
+ 0x0b, /* u8 bLength */
+ USB_DT_CS_INTERFACE, /* u8 bDescriptorType */
+ DST_AS_FORMAT_TYPE, /* u8 bDescriptorSubtype */
+ 0x01, /* u8 bFormatType */
+ 0x06, /* u8 bNrChannels */
+ 0x02, /* u8 bSubFrameSize */
+ 0x10, /* u8 bBitResolution */
+ 0x01, /* u8 bSamFreqType */
+ U24(USBAUDIO_SAMPLE_RATE), /* u24 tSamFreq */
+ }
+ }
+ },
+ .eps = (USBDescEndpoint[]) {
+ {
+ .bEndpointAddress = USB_DIR_OUT | 0x01,
+ .bmAttributes = 0x0d,
+ .wMaxPacketSize = USBAUDIO_PACKET_SIZE(6),
+ .bInterval = 1,
+ .is_audio = 1,
+ /* Stereo Headphone Class-specific
+ AS Audio Data Endpoint Descriptor */
+ .extra = (uint8_t[]) {
+ 0x07, /* u8 bLength */
+ USB_DT_CS_ENDPOINT, /* u8 bDescriptorType */
+ DST_EP_GENERAL, /* u8 bDescriptorSubtype */
+ 0x00, /* u8 bmAttributes */
+ 0x00, /* u8 bLockDelayUnits */
+ U16(0x0000), /* u16 wLockDelay */
+ },
+ },
+ }
+ },{
+ .bInterfaceNumber = 1,
+ .bAlternateSetting = ALTSET_71,
+ .bNumEndpoints = 1,
+ .bInterfaceClass = USB_CLASS_AUDIO,
+ .bInterfaceSubClass = USB_SUBCLASS_AUDIO_STREAMING,
+ .iInterface = STRING_REAL_STREAM,
+ .ndesc = 2,
+ .descs = (USBDescOther[]) {
+ {
+ /* Headphone Class-specific AS General Interface Descriptor */
+ .data = (uint8_t[]) {
+ 0x07, /* u8 bLength */
+ USB_DT_CS_INTERFACE, /* u8 bDescriptorType */
+ DST_AS_GENERAL, /* u8 bDescriptorSubtype */
+ 0x01, /* u8 bTerminalLink */
+ 0x00, /* u8 bDelay */
+ 0x01, 0x00, /* u16 wFormatTag */
+ }
+ },{
+ /* Headphone Type I Format Type Descriptor */
+ .data = (uint8_t[]) {
+ 0x0b, /* u8 bLength */
+ USB_DT_CS_INTERFACE, /* u8 bDescriptorType */
+ DST_AS_FORMAT_TYPE, /* u8 bDescriptorSubtype */
+ 0x01, /* u8 bFormatType */
+ 0x08, /* u8 bNrChannels */
+ 0x02, /* u8 bSubFrameSize */
+ 0x10, /* u8 bBitResolution */
+ 0x01, /* u8 bSamFreqType */
+ U24(USBAUDIO_SAMPLE_RATE), /* u24 tSamFreq */
+ }
+ }
+ },
+ .eps = (USBDescEndpoint[]) {
+ {
+ .bEndpointAddress = USB_DIR_OUT | 0x01,
+ .bmAttributes = 0x0d,
+ .wMaxPacketSize = USBAUDIO_PACKET_SIZE(8),
+ .bInterval = 1,
+ .is_audio = 1,
+ /* Stereo Headphone Class-specific
+ AS Audio Data Endpoint Descriptor */
+ .extra = (uint8_t[]) {
+ 0x07, /* u8 bLength */
+ USB_DT_CS_ENDPOINT, /* u8 bDescriptorType */
+ DST_EP_GENERAL, /* u8 bDescriptorSubtype */
+ 0x00, /* u8 bmAttributes */
+ 0x00, /* u8 bLockDelayUnits */
+ U16(0x0000), /* u16 wLockDelay */
+ },
+ },
+ }
+ }
+};
+
+static const USBDescDevice desc_device_multi = {
+ .bcdUSB = 0x0100,
+ .bMaxPacketSize0 = 64,
+ .bNumConfigurations = 1,
+ .confs = (USBDescConfig[]) {
+ {
+ .bNumInterfaces = 2,
+ .bConfigurationValue = DEV_CONFIG_VALUE,
+ .iConfiguration = STRING_CONFIG,
+ .bmAttributes = USB_CFG_ATT_ONE | USB_CFG_ATT_SELFPOWER,
+ .bMaxPower = 0x32,
+ .nif = ARRAY_SIZE(desc_iface_multi),
+ .ifs = desc_iface_multi,
+ }
+ },
+};
+
+static const USBDesc desc_audio_multi = {
+ .id = {
+ .idVendor = USBAUDIO_VENDOR_NUM,
+ .idProduct = USBAUDIO_PRODUCT_NUM,
+ .bcdDevice = 0,
+ .iManufacturer = STRING_MANUFACTURER,
+ .iProduct = STRING_PRODUCT,
+ .iSerialNumber = STRING_SERIALNUMBER,
+ },
+ .full = &desc_device_multi,
+ .str = usb_audio_stringtable,
};
/*
@@ -295,15 +578,16 @@
struct streambuf {
uint8_t *data;
- uint32_t size;
- uint32_t prod;
- uint32_t cons;
+ size_t size;
+ uint64_t prod;
+ uint64_t cons;
};
-static void streambuf_init(struct streambuf *buf, uint32_t size)
+static void streambuf_init(struct streambuf *buf, uint32_t size,
+ uint32_t channels)
{
g_free(buf->data);
- buf->size = size - (size % USBAUDIO_PACKET_SIZE);
+ buf->size = size - (size % USBAUDIO_PACKET_SIZE(channels));
buf->data = g_malloc(buf->size);
buf->prod = 0;
buf->cons = 0;
@@ -315,34 +599,37 @@
buf->data = NULL;
}
-static int streambuf_put(struct streambuf *buf, USBPacket *p)
+static int streambuf_put(struct streambuf *buf, USBPacket *p, uint32_t channels)
{
- uint32_t free = buf->size - (buf->prod - buf->cons);
+ int64_t free = buf->size - (buf->prod - buf->cons);
- if (!free) {
+ if (free < USBAUDIO_PACKET_SIZE(channels)) {
return 0;
}
- if (p->iov.size != USBAUDIO_PACKET_SIZE) {
+ if (p->iov.size != USBAUDIO_PACKET_SIZE(channels)) {
return 0;
}
- assert(free >= USBAUDIO_PACKET_SIZE);
+
+ /* can happen if prod overflows */
+ assert(buf->prod % USBAUDIO_PACKET_SIZE(channels) == 0);
usb_packet_copy(p, buf->data + (buf->prod % buf->size),
- USBAUDIO_PACKET_SIZE);
- buf->prod += USBAUDIO_PACKET_SIZE;
- return USBAUDIO_PACKET_SIZE;
+ USBAUDIO_PACKET_SIZE(channels));
+ buf->prod += USBAUDIO_PACKET_SIZE(channels);
+ return USBAUDIO_PACKET_SIZE(channels);
}
-static uint8_t *streambuf_get(struct streambuf *buf)
+static uint8_t *streambuf_get(struct streambuf *buf, size_t *len)
{
- uint32_t used = buf->prod - buf->cons;
+ int64_t used = buf->prod - buf->cons;
uint8_t *data;
- if (!used) {
+ if (used <= 0) {
+ *len = 0;
return NULL;
}
- assert(used >= USBAUDIO_PACKET_SIZE);
data = buf->data + (buf->cons % buf->size);
- buf->cons += USBAUDIO_PACKET_SIZE;
+ *len = MIN(buf->prod - buf->cons,
+ buf->size - (buf->cons % buf->size));
return data;
}
@@ -356,14 +643,15 @@
enum usb_audio_altset altset;
struct audsettings as;
SWVoiceOut *voice;
- bool mute;
- uint8_t vol[2];
+ Volume vol;
struct streambuf buf;
+ uint32_t channels;
} out;
/* properties */
uint32_t debug;
- uint32_t buffer;
+ uint32_t buffer_user, buffer;
+ bool multi;
} USBAudioState;
#define TYPE_USB_AUDIO "usb-audio"
@@ -374,16 +662,21 @@
USBAudioState *s = opaque;
uint8_t *data;
- for (;;) {
- if (avail < USBAUDIO_PACKET_SIZE) {
- return;
- }
- data = streambuf_get(&s->out.buf);
+ while (avail) {
+ size_t written, len;
+
+ data = streambuf_get(&s->out.buf, &len);
if (!data) {
return;
}
- AUD_write(s->out.voice, data, USBAUDIO_PACKET_SIZE);
- avail -= USBAUDIO_PACKET_SIZE;
+
+ written = AUD_write(s->out.voice, data, len);
+ avail -= written;
+ s->out.buf.cons += written;
+
+ if (written < len) {
+ return;
+ }
}
}
@@ -391,10 +684,15 @@
{
switch (altset) {
case ALTSET_OFF:
- streambuf_init(&s->out.buf, s->buffer);
AUD_set_active_out(s->out.voice, false);
break;
- case ALTSET_ON:
+ case ALTSET_STEREO:
+ case ALTSET_51:
+ case ALTSET_71:
+ if (s->out.channels != altset_channels[altset]) {
+ usb_audio_reinit(USB_DEVICE(s), altset_channels[altset]);
+ }
+ streambuf_init(&s->out.buf, s->buffer, s->out.channels);
AUD_set_active_out(s->out.voice, true);
break;
default:
@@ -425,33 +723,33 @@
switch (aid) {
case ATTRIB_ID(MUTE_CONTROL, CR_GET_CUR, 0x0200):
- data[0] = s->out.mute;
+ data[0] = s->out.vol.mute;
ret = 1;
break;
case ATTRIB_ID(VOLUME_CONTROL, CR_GET_CUR, 0x0200):
- if (cn < 2) {
- uint16_t vol = (s->out.vol[cn] * 0x8800 + 127) / 255 + 0x8000;
+ if (cn < USBAUDIO_MAX_CHANNELS(s)) {
+ uint16_t vol = (s->out.vol.vol[cn] * 0x8800 + 127) / 255 + 0x8000;
data[0] = vol;
data[1] = vol >> 8;
ret = 2;
}
break;
case ATTRIB_ID(VOLUME_CONTROL, CR_GET_MIN, 0x0200):
- if (cn < 2) {
+ if (cn < USBAUDIO_MAX_CHANNELS(s)) {
data[0] = 0x01;
data[1] = 0x80;
ret = 2;
}
break;
case ATTRIB_ID(VOLUME_CONTROL, CR_GET_MAX, 0x0200):
- if (cn < 2) {
+ if (cn < USBAUDIO_MAX_CHANNELS(s)) {
data[0] = 0x00;
data[1] = 0x08;
ret = 2;
}
break;
case ATTRIB_ID(VOLUME_CONTROL, CR_GET_RES, 0x0200):
- if (cn < 2) {
+ if (cn < USBAUDIO_MAX_CHANNELS(s)) {
data[0] = 0x88;
data[1] = 0x00;
ret = 2;
@@ -473,16 +771,17 @@
switch (aid) {
case ATTRIB_ID(MUTE_CONTROL, CR_SET_CUR, 0x0200):
- s->out.mute = data[0] & 1;
+ s->out.vol.mute = data[0] & 1;
set_vol = true;
ret = 0;
break;
case ATTRIB_ID(VOLUME_CONTROL, CR_SET_CUR, 0x0200):
- if (cn < 2) {
+ if (cn < USBAUDIO_MAX_CHANNELS(s)) {
uint16_t vol = data[0] + (data[1] << 8);
if (s->debug) {
- fprintf(stderr, "usb-audio: vol %04x\n", (uint16_t)vol);
+ fprintf(stderr, "usb-audio: cn %d vol %04x\n", cn,
+ (uint16_t)vol);
}
vol -= 0x8000;
@@ -491,7 +790,7 @@
vol = 255;
}
- s->out.vol[cn] = vol;
+ s->out.vol.vol[cn] = vol;
set_vol = true;
ret = 0;
}
@@ -500,11 +799,14 @@
if (set_vol) {
if (s->debug) {
- fprintf(stderr, "usb-audio: mute %d, lvol %3d, rvol %3d\n",
- s->out.mute, s->out.vol[0], s->out.vol[1]);
+ int i;
+ fprintf(stderr, "usb-audio: mute %d", s->out.vol.mute);
+ for (i = 0; i < USBAUDIO_MAX_CHANNELS(s); ++i) {
+ fprintf(stderr, ", vol[%d] %3d", i, s->out.vol.vol[i]);
+ }
+ fprintf(stderr, "\n");
}
- AUD_set_volume_out(s->out.voice, s->out.mute,
- s->out.vol[0], s->out.vol[1]);
+ audio_set_volume_out(s->out.voice, &s->out.vol);
}
return ret;
@@ -597,7 +899,7 @@
return;
}
- streambuf_put(&s->out.buf, p);
+ streambuf_put(&s->out.buf, p, s->out.channels);
if (p->actual_length < p->iov.size && s->debug > 1) {
fprintf(stderr, "usb-audio: output overrun (%zd bytes)\n",
p->iov.size - p->actual_length);
@@ -639,6 +941,9 @@
static void usb_audio_realize(USBDevice *dev, Error **errp)
{
USBAudioState *s = USB_AUDIO(dev);
+ int i;
+
+ dev->usb_desc = s->multi ? &desc_audio_multi : &desc_audio;
usb_desc_create_serial(dev);
usb_desc_init(dev);
@@ -646,18 +951,35 @@
AUD_register_card(TYPE_USB_AUDIO, &s->card);
s->out.altset = ALTSET_OFF;
- s->out.mute = false;
- s->out.vol[0] = 240; /* 0 dB */
- s->out.vol[1] = 240; /* 0 dB */
+ s->out.vol.mute = false;
+ for (i = 0; i < USBAUDIO_MAX_CHANNELS(s); ++i) {
+ s->out.vol.vol[i] = 240; /* 0 dB */
+ }
+
+ usb_audio_reinit(dev, 2);
+}
+
+static void usb_audio_reinit(USBDevice *dev, unsigned channels)
+{
+ USBAudioState *s = USB_AUDIO(dev);
+
+ s->out.channels = channels;
+ if (!s->buffer_user) {
+ s->buffer = 32 * USBAUDIO_PACKET_SIZE(s->out.channels);
+ } else {
+ s->buffer = s->buffer_user;
+ }
+
+ s->out.vol.channels = s->out.channels;
s->out.as.freq = USBAUDIO_SAMPLE_RATE;
- s->out.as.nchannels = 2;
+ s->out.as.nchannels = s->out.channels;
s->out.as.fmt = AUDIO_FORMAT_S16;
s->out.as.endianness = 0;
- streambuf_init(&s->out.buf, s->buffer);
+ streambuf_init(&s->out.buf, s->buffer, s->out.channels);
s->out.voice = AUD_open_out(&s->card, s->out.voice, TYPE_USB_AUDIO,
s, output_callback, &s->out.as);
- AUD_set_volume_out(s->out.voice, s->out.mute, s->out.vol[0], s->out.vol[1]);
+ audio_set_volume_out(s->out.voice, &s->out.vol);
AUD_set_active_out(s->out.voice, 0);
}
@@ -669,8 +991,8 @@
static Property usb_audio_properties[] = {
DEFINE_AUDIO_PROPERTIES(USBAudioState, card),
DEFINE_PROP_UINT32("debug", USBAudioState, debug, 0),
- DEFINE_PROP_UINT32("buffer", USBAudioState, buffer,
- 32 * USBAUDIO_PACKET_SIZE),
+ DEFINE_PROP_UINT32("buffer", USBAudioState, buffer_user, 0),
+ DEFINE_PROP_BOOL("multi", USBAudioState, multi, false),
DEFINE_PROP_END_OF_LIST(),
};
@@ -683,7 +1005,6 @@
dc->props = usb_audio_properties;
set_bit(DEVICE_CATEGORY_SOUND, dc->categories);
k->product_desc = "QEMU USB Audio Interface";
- k->usb_desc = &desc_audio;
k->realize = usb_audio_realize;
k->handle_reset = usb_audio_handle_reset;
k->handle_control = usb_audio_handle_control;
diff --git a/qapi/audio.json b/qapi/audio.json
index 9fefdf5..83312b2 100644
--- a/qapi/audio.json
+++ b/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' } }
##
diff --git a/qemu-options.hx b/qemu-options.hx
index 793d70f..996b6fb 100644
--- a/qemu-options.hx
+++ b/qemu-options.hx
@@ -433,6 +433,7 @@
" 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 @@
-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 @@
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
