Hi,
Spice clients that are running directly on the host system have
pratcially unlimited bandwidth so to reduce latency allow the user to
configure the buffer_length to a lower value if desired.
While virt-viewer can not take advantage of this, the PureSpice [1]
library used by Looking Glass [2] is able to produce and consume audio
at these rates, combined with the merge request for spice-server [3]
this allows for latencies close to realtime.
[1]https://github.com/gnif/PureSpice
[2]https://github.com/gnif/LookingGlass
[3]https://gitlab.freedesktop.org/spice/spice/-/merge_requests/199
Signed-off-by: Geoffrey McRae<ge...@hostfission.com>
---
audio/spiceaudio.c | 19 ++++++++++++++++---
1 file changed, 16 insertions(+), 3 deletions(-)
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index a8d370fe6f..0c44bbe836 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -76,7 +76,7 @@ static void *spice_audio_init(Audiodev *dev)
if (!using_spice) {
return NULL;
}
- return &spice_audio_init;
+ return dev;
}
static void spice_audio_fini (void *opaque)
@@ -90,6 +90,8 @@ static int line_out_init(HWVoiceOut *hw, struct audsettings
*as,
void *drv_opaque)
{
SpiceVoiceOut *out = container_of (hw, SpiceVoiceOut, hw);
+ Audiodev *dev = (Audiodev *)drv_opaque;
+
struct audsettings settings;
#if SPICE_INTERFACE_PLAYBACK_MAJOR > 1 || SPICE_INTERFACE_PLAYBACK_MINOR >= 3
@@ -102,7 +104,12 @@ static int line_out_init(HWVoiceOut *hw, struct
audsettings *as,
settings.endianness = AUDIO_HOST_ENDIANNESS;
audio_pcm_init_info (&hw->info, &settings);
- hw->samples = LINE_OUT_SAMPLES;
+ if (dev->u.none.out->has_buffer_length) {
+ hw->samples = audio_buffer_samples(dev->u.none.out, &settings, 10000);
hw->samples counts in frames. The buffer is twice as large as expected.
+ hw->samples = audio_buffer_frames(dev->u.none.out, &settings,
10000);
I'm aware the default size of 10000us will not be used, but it's a bad
example because with a default timer-period of 10000us the buffer has to
be a few percent larger than timer-period. Otherwise the emulated audio
device will never catch up if a AUD_write() has been delayed.
+ } else {
+ hw->samples = LINE_OUT_SAMPLES;
+ }
+
out->active = 0;
out->sin.base.sif = &playback_sif.base;
@@ -199,6 +206,7 @@ static void line_out_volume(HWVoiceOut *hw, Volume *vol)
static int line_in_init(HWVoiceIn *hw, struct audsettings *as, void
*drv_opaque)
{
SpiceVoiceIn *in = container_of (hw, SpiceVoiceIn, hw);
+ Audiodev *dev = (Audiodev *)drv_opaque;
struct audsettings settings;
#if SPICE_INTERFACE_RECORD_MAJOR > 2 || SPICE_INTERFACE_RECORD_MINOR >= 3
@@ -211,7 +219,12 @@ static int line_in_init(HWVoiceIn *hw, struct audsettings
*as, void *drv_opaque)
settings.endianness = AUDIO_HOST_ENDIANNESS;
audio_pcm_init_info (&hw->info, &settings);
- hw->samples = LINE_IN_SAMPLES;
+ if (dev->u.none.out->has_buffer_length) {
+ hw->samples = audio_buffer_samples(dev->u.none.in, &settings, 10000);
- hw->samples = audio_buffer_samples(dev->u.none.in, &settings,
10000);
+ hw->samples = audio_buffer_frames(dev->u.none.in, &settings,
10000);
+ } else {
+ hw->samples = LINE_IN_SAMPLES;
+ }
+
in->active = 0;
in->sin.base.sif = &record_sif.base;
Btw. have you seen my "[PATCH 00/15] reduce audio playback latency"
patch series at
https://lists.nongnu.org/archive/html/qemu-devel/2022-01/msg00780.html?
I haven't tested, but I think it's possible to add a buffer_get_free
function to audio/spiceaudio.c. That would eliminate the need to
fine-tune the audio buffer length.
With best regards,
Volker
