audio: restore mixing-engine playback buffer size

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>
Reviewed-by: Akihiko Odaki <akihiko.odaki@gmail.com>
Message-Id: <20220301191311.26695-10-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>

1 files changed, 52 insertions, 17 deletions

diff --git a/audio/audio.c b/audio/audio.c
index c420a8bd1c..a88572e713 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -663,6 +663,12 @@ 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->pcm_ops->buffer_get_free(hw) :
+            INT_MAX) / hw->info.bytes_per_frame;
+}
+
 static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len)
 {
     size_t clipped = 0;
@@ -687,7 +693,8 @@ static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len)
  */
 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, blck;
+    size_t hw_free;
     size_t ret = 0, pos = 0, total = 0;
 
     if (!sw) {
@@ -710,27 +717,28 @@ 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);
+    hw_free = audio_pcm_hw_get_free(sw->hw);
+    hw_free = hw_free > live ? hw_free - live : 0;
+    samples = ((int64_t)MIN(dead, hw_free) << 32) / sw->ratio;
+    samples = MIN(samples, size / sw->info.bytes_per_frame);
+    if (samples) {
+        sw->conv(sw->buf, buf, samples);
 
         if (!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,
@@ -740,7 +748,7 @@ 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;
@@ -1002,6 +1010,11 @@ 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;
@@ -1021,13 +1034,11 @@ static size_t audio_get_free(SWVoiceOut *sw)
     dead = sw->hw->mix_buf->size - live;
 
 #ifdef DEBUG_OUT
-    dolog ("%s: get_free live %zu dead %zu ret %" PRId64 "\n",
-           SW_NAME (sw),
-           live, dead, (((int64_t) dead << 32) / sw->ratio) *
-           sw->info.bytes_per_frame);
+    dolog("%s: get_free live %zu dead %zu sw_bytes %zu\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,
@@ -1131,12 +1142,21 @@ static void audio_run_out (AudioState *s)
     }
 
     while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
-        size_t played, live, prev_rpos, free;
+        size_t played, live, prev_rpos;
+        size_t hw_free = audio_pcm_hw_get_free(hw);
         int nb_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 free;
+
+                if (hw_free > sw->total_hw_samples_mixed) {
+                    free = audio_sw_bytes_free(sw,
+                        MIN(sw_free, hw_free - sw->total_hw_samples_mixed));
+                } else {
+                    free = 0;
+                }
                 if (free > 0) {
                     sw->callback.fn(sw->callback.opaque, free);
                 }
@@ -1398,6 +1418,15 @@ 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) {
@@ -1445,6 +1474,12 @@ size_t audio_generic_write(HWVoiceOut *hw, void *buf, size_t size)
 {
     size_t total = 0;
 
+    if (hw->pcm_ops->buffer_get_free) {
+        size_t free = hw->pcm_ops->buffer_get_free(hw);
+
+        size = MIN(size, free);
+    }
+
     while (total < size) {
         size_t dst_size = size - total;
         size_t copy_size, proc;