@@ -923,6 +923,7 @@ static struct audio_pcm_ops alsa_pcm_ops = {
.init_out = alsa_init_out,
.fini_out = alsa_fini_out,
.write = alsa_write,
+ .buffer_get_free = audio_generic_buffer_get_free,
.run_buffer_out = audio_generic_run_buffer_out,
.enable_out = alsa_enable_out,
@@ -700,12 +700,18 @@ static size_t audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
return 0;
}
+static size_t audio_pcm_hw_get_free(HWVoiceOut *hw)
+{
+ return hw->pcm_ops->buffer_get_free(hw) / hw->info.bytes_per_frame;
+}
+
/*
* Soft voice (playback)
*/
static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
{
- size_t hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
+ size_t hwsamples, samples, isamp, osamp, wpos, live, dead, left, limit;
+ size_t blck;
size_t ret = 0, pos = 0, total = 0;
if (!sw) {
@@ -728,27 +734,28 @@ 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.bytes_per_frame;
dead = hwsamples - live;
- swlim = ((int64_t) dead << 32) / sw->ratio;
- swlim = MIN (swlim, samples);
- if (swlim) {
- sw->conv (sw->buf, buf, swlim);
+ limit = audio_pcm_hw_get_free(sw->hw);
+ samples = ((int64_t)MIN(dead, limit) << 32) / sw->ratio;
+ limit = size / sw->info.bytes_per_frame;
+ samples = MIN(samples, limit);
+ if (samples) {
+ sw->conv(sw->buf, buf, samples);
if (sw->hw->pcm_ops && !sw->hw->pcm_ops->volume_out) {
- mixeng_volume (sw->buf, swlim, &sw->vol);
+ mixeng_volume(sw->buf, samples, &sw->vol);
}
}
- while (swlim) {
+ while (samples) {
dead = hwsamples - live;
left = hwsamples - wpos;
blck = MIN (dead, left);
if (!blck) {
break;
}
- isamp = swlim;
+ isamp = samples;
osamp = blck;
st_rate_flow_mix (
sw->rate,
@@ -758,7 +765,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
&osamp
);
ret += isamp;
- swlim -= isamp;
+ samples -= isamp;
pos += isamp;
live += osamp;
wpos = (wpos + osamp) % hwsamples;
@@ -1018,6 +1025,11 @@ static size_t audio_get_avail (SWVoiceIn *sw)
return (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame;
}
+static size_t audio_sw_bytes_free(SWVoiceOut *sw, size_t free)
+{
+ return (((int64_t)free << 32) / sw->ratio) * sw->info.bytes_per_frame;
+}
+
static size_t audio_get_free(SWVoiceOut *sw)
{
size_t live, dead;
@@ -1037,13 +1049,11 @@ static size_t audio_get_free(SWVoiceOut *sw)
dead = sw->hw->mix_buf->size - live;
#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.bytes_per_frame);
+ dolog("%s: get_free live %d dead %d sw_bytes %d\n",
+ SW_NAME(sw), live, dead, audio_sw_bytes_free(sw, dead));
#endif
- return (((int64_t) dead << 32) / sw->ratio) * sw->info.bytes_per_frame;
+ return dead;
}
static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
@@ -1180,7 +1190,10 @@ static void audio_run_out (AudioState *s)
if (!live) {
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
if (sw->active) {
- free = audio_get_free (sw);
+ size_t sw_free = audio_get_free(sw);
+ size_t hw_free = audio_pcm_hw_get_free(hw);
+
+ free = audio_sw_bytes_free(sw, MIN(sw_free, hw_free));
if (free > 0) {
sw->callback.fn (sw->callback.opaque, free);
}
@@ -1230,7 +1243,10 @@ static void audio_run_out (AudioState *s)
}
if (sw->active) {
- free = audio_get_free (sw);
+ size_t sw_free = audio_get_free(sw);
+ size_t hw_free = audio_pcm_hw_get_free(hw);
+
+ free = audio_sw_bytes_free(sw, MIN(sw_free, hw_free));
if (free > 0) {
sw->callback.fn (sw->callback.opaque, free);
}
@@ -1437,6 +1453,15 @@ void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size)
hw->pending_emul -= size;
}
+size_t audio_generic_buffer_get_free(HWVoiceOut *hw)
+{
+ if (hw->buf_emul) {
+ return hw->size_emul - hw->pending_emul;
+ } else {
+ return hw->samples * hw->info.bytes_per_frame;
+ }
+}
+
void audio_generic_run_buffer_out(HWVoiceOut *hw)
{
while (hw->pending_emul) {
@@ -1826,6 +1851,14 @@ void AUD_remove_card (QEMUSoundCard *card)
g_free (card->name);
}
+static size_t capture_buffer_get_free(HWVoiceOut *hw)
+{
+ return INT_MAX;
+}
+
+static struct audio_pcm_ops capture_pcm_ops = {
+ .buffer_get_free = capture_buffer_get_free,
+};
CaptureVoiceOut *AUD_add_capture(
AudioState *s,
@@ -1872,6 +1905,7 @@ CaptureVoiceOut *AUD_add_capture(
hw = &cap->hw;
hw->s = s;
+ hw->pcm_ops = &capture_pcm_ops;
QLIST_INIT (&hw->sw_head);
QLIST_INIT (&cap->cb_head);
@@ -154,10 +154,14 @@ struct audio_pcm_ops {
void (*fini_out)(HWVoiceOut *hw);
size_t (*write) (HWVoiceOut *hw, void *buf, size_t size);
void (*run_buffer_out)(HWVoiceOut *hw);
+ /*
+ * Get the free output buffer size. This is an upper limit. The size
+ * returned by function get_buffer_out may be smaller.
+ */
+ size_t (*buffer_get_free)(HWVoiceOut *hw);
/*
* get a buffer that after later can be passed to put_buffer_out; optional
* returns the buffer, and writes it's size to size (in bytes)
- * this is unrelated to the above buffer_size_out function
*/
void *(*get_buffer_out)(HWVoiceOut *hw, size_t *size);
/*
@@ -181,6 +185,7 @@ struct audio_pcm_ops {
void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size);
void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size);
void audio_generic_run_buffer_out(HWVoiceOut *hw);
+size_t audio_generic_buffer_get_free(HWVoiceOut *hw);
void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size);
size_t audio_generic_put_buffer_out(HWVoiceOut *hw, void *buf, size_t size);
size_t audio_generic_write(HWVoiceOut *hw, void *buf, size_t size);
@@ -409,6 +409,7 @@ static int coreaudio_unlock (coreaudioVoiceOut *core, const char *fn_name)
coreaudio_unlock(core, "coreaudio_" #name); \
return ret; \
}
+COREAUDIO_WRAPPER_FUNC(buffer_get_free, size_t, (HWVoiceOut *hw), (hw))
COREAUDIO_WRAPPER_FUNC(get_buffer_out, void *, (HWVoiceOut *hw, size_t *size),
(hw, size))
COREAUDIO_WRAPPER_FUNC(put_buffer_out, size_t,
@@ -670,6 +671,8 @@ static struct audio_pcm_ops coreaudio_pcm_ops = {
.fini_out = coreaudio_fini_out,
/* wrapper for audio_generic_write */
.write = coreaudio_write,
+ /* wrapper for audio_generic_buffer_get_free */
+ .buffer_get_free = coreaudio_buffer_get_free,
/* wrapper for audio_generic_get_buffer_out */
.get_buffer_out = coreaudio_get_buffer_out,
/* wrapper for audio_generic_put_buffer_out */
@@ -411,6 +411,11 @@ static void dsound_enable_out(HWVoiceOut *hw, bool enable)
}
}
+static size_t dsound_buffer_get_free(HWVoiceOut *hw)
+{
+ return INT_MAX;
+}
+
static void *dsound_get_buffer_out(HWVoiceOut *hw, size_t *size)
{
DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
@@ -686,6 +691,7 @@ static struct audio_pcm_ops dsound_pcm_ops = {
.init_out = dsound_init_out,
.fini_out = dsound_fini_out,
.write = audio_generic_write,
+ .buffer_get_free = dsound_buffer_get_free,
.get_buffer_out = dsound_get_buffer_out,
.put_buffer_out = dsound_put_buffer_out,
.enable_out = dsound_enable_out,
@@ -628,6 +628,7 @@ static struct audio_pcm_ops jack_pcm_ops = {
.init_out = qjack_init_out,
.fini_out = qjack_fini_out,
.write = qjack_write,
+ .buffer_get_free = audio_generic_buffer_get_free,
.run_buffer_out = audio_generic_run_buffer_out,
.enable_out = qjack_enable_out,
@@ -118,6 +118,7 @@ static struct audio_pcm_ops no_pcm_ops = {
.init_out = no_init_out,
.fini_out = no_fini_out,
.write = no_write,
+ .buffer_get_free = audio_generic_buffer_get_free,
.run_buffer_out = audio_generic_run_buffer_out,
.enable_out = no_enable_out,
@@ -391,6 +391,17 @@ static void oss_run_buffer_out(HWVoiceOut *hw)
}
}
+static size_t oss_buffer_get_free(HWVoiceOut *hw)
+{
+ OSSVoiceOut *oss = (OSSVoiceOut *)hw;
+
+ if (oss->mmapped) {
+ return oss_get_available_bytes(oss);
+ } else {
+ return audio_generic_buffer_get_free(hw);
+ }
+}
+
static void *oss_get_buffer_out(HWVoiceOut *hw, size_t *size)
{
OSSVoiceOut *oss = (OSSVoiceOut *) hw;
@@ -754,6 +765,7 @@ static struct audio_pcm_ops oss_pcm_ops = {
.init_out = oss_init_out,
.fini_out = oss_fini_out,
.write = oss_write,
+ .buffer_get_free = oss_buffer_get_free,
.run_buffer_out = oss_run_buffer_out,
.get_buffer_out = oss_get_buffer_out,
.put_buffer_out = oss_put_buffer_out,
@@ -202,6 +202,11 @@ unlock_and_fail:
return 0;
}
+static size_t qpa_buffer_get_free(HWVoiceOut *hw)
+{
+ return INT_MAX;
+}
+
static void *qpa_get_buffer_out(HWVoiceOut *hw, size_t *size)
{
PAVoiceOut *p = (PAVoiceOut *) hw;
@@ -860,6 +865,7 @@ static struct audio_pcm_ops qpa_pcm_ops = {
.init_out = qpa_init_out,
.fini_out = qpa_fini_out,
.write = qpa_write,
+ .buffer_get_free = qpa_buffer_get_free,
.get_buffer_out = qpa_get_buffer_out,
.put_buffer_out = qpa_write, /* pa handles it */
.volume_out = qpa_volume_out,
@@ -253,6 +253,7 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
return ret; \
}
+SDL_WRAPPER_FUNC(buffer_get_free, size_t, (HWVoiceOut *hw), (hw))
SDL_WRAPPER_FUNC(get_buffer_out, void *, (HWVoiceOut *hw, size_t *size),
(hw, size), *size = 0, sdl_unlock)
SDL_WRAPPER_FUNC(put_buffer_out, size_t,
@@ -350,6 +351,8 @@ static struct audio_pcm_ops sdl_pcm_ops = {
.fini_out = sdl_fini_out,
/* wrapper for audio_generic_write */
.write = sdl_write,
+ /* wrapper for audio_generic_buffer_get_free */
+ .buffer_get_free = sdl_buffer_get_free,
/* wrapper for audio_generic_get_buffer_out */
.get_buffer_out = sdl_get_buffer_out,
/* wrapper for audio_generic_put_buffer_out */
@@ -120,6 +120,11 @@ static void line_out_fini (HWVoiceOut *hw)
spice_server_remove_interface (&out->sin.base);
}
+static size_t line_out_buffer_get_free(HWVoiceOut *hw)
+{
+ return INT_MAX;
+}
+
static void *line_out_get_buffer(HWVoiceOut *hw, size_t *size)
{
SpiceVoiceOut *out = container_of(hw, SpiceVoiceOut, hw);
@@ -282,6 +287,7 @@ static struct audio_pcm_ops audio_callbacks = {
.init_out = line_out_init,
.fini_out = line_out_fini,
.write = audio_generic_write,
+ .buffer_get_free = line_out_buffer_get_free,
.get_buffer_out = line_out_get_buffer,
.put_buffer_out = line_out_put_buffer,
.enable_out = line_out_enable,
@@ -197,6 +197,7 @@ static struct audio_pcm_ops wav_pcm_ops = {
.init_out = wav_init_out,
.fini_out = wav_fini_out,
.write = wav_write_out,
+ .buffer_get_free = audio_generic_buffer_get_free,
.run_buffer_out = audio_generic_run_buffer_out,
.enable_out = wav_enable_out,
};
Commit ff095e5231 "audio: api for mixeng code free backends" introduced another FIFO for the audio subsystem with exactly the same size as the mixing-engine FIFO. Most audio backends use this generic FIFO. The generic FIFO used together with the mixing-engine FIFO doubles the audio FIFO size, because that's just two independent FIFOs connected together in series. For audio playback this nearly doubles the playback latency. This patch restores the effective mixing-engine playback buffer size to a pre v4.2.0 size by only accepting the amount of samples for the mixing-engine queue which the downstream queue accepts. Signed-off-by: Volker Rümelin <vr_qemu@t-online.de> --- audio/alsaaudio.c | 1 + audio/audio.c | 68 +++++++++++++++++++++++++++++++++------------ audio/audio_int.h | 7 ++++- audio/coreaudio.c | 3 ++ audio/dsoundaudio.c | 6 ++++ audio/jackaudio.c | 1 + audio/noaudio.c | 1 + audio/ossaudio.c | 12 ++++++++ audio/paaudio.c | 6 ++++ audio/sdlaudio.c | 3 ++ audio/spiceaudio.c | 6 ++++ audio/wavaudio.c | 1 + 12 files changed, 97 insertions(+), 18 deletions(-)