[PATCH v3 00/15] audio: improve callback interface for audio frontends

[PATCH v3 00/15] audio: improve callback interface for audio frontends

[Volker Rümelin]

Based-on: <0a4007dc-e11c-f16e-0e21-dbc4e60caa59@t-online.de>
([PATCH v2 00/11] audio: more improvements)

The callback interface for emulated audio devices is strange. The 
callback function has an 'avail' parameter that passes the number of 
bytes that can be written or read. Unfortunately, this value sometimes 
is only an imprecise estimate and the callback functions must check the 
actual bytes written or read. For playback devices, this means that they 
either need a ring buffer or have to write the unwritten bytes again the 
next time. For recording devices, things are a bit easier. They only 
need to continue with the actual number of bytes read.

After this patch series, the 'avail' argument for the -audiodev 
out.mixing-engine=on and in.mixing-engine=on cases is exact. Audio 
frontends only need a linear frame buffer and there's a guarantee they 
can write or read 'avail' bytes.

The -audiodev out.mixing-engine=off case is also mostly accurate. Only 
the D-Bus audio backend is still missing a required function. The 
-audiodev in.mixing-engine=off case always passes a much too large 
'avail' value. I haven't worked on this yet, because there was no reason 
for it so far.

The following logs show the improvements. Not only the audio frontends 
can write or read all needed or available bytes. The same is true for 
the audio backends. For playback, the first six lines in the logs are 
expected. Here you can see how quickly the guest fills the empty 
downstream buffers after playback starts.

QEMU was started with -device ich9-intel-hda,addr=0x1b -device 
hda-duplex,audiodev=audio0 -audiodev 
pa,out.frequency=96000,in.frequency=96000,id=audio0

playback guest 44100Hz => host 96000Hz

unpatched version:
hda_audio_output_cb: to write 8188, written 1704
audio_run_out: free 4458, played 926
hda_audio_output_cb: to write 6488, written 2384
audio_run_out: free 3532, played 1297
hda_audio_output_cb: to write 4104, written 2648
audio_run_out: free 2235, played 1441
audio_run_out: free 794, played 793
audio_run_out: free 897, played 896
audio_run_out: free 831, played 829
...
hda_audio_output_cb: could not write 4 bytes
hda_audio_output_cb: to write 1764, written 1760
audio_run_out: free 960, played 958
...

patched version:
hda_audio_output_cb: to write 8192, written 1620
audio_run_out: free 4458, played 880
hda_audio_output_cb: to write 6576, written 2508
audio_run_out: free 3578, played 1365
hda_audio_output_cb: to write 4068, written 2500
audio_run_out: free 2213, played 1360

record host 96000Hz => guest 44100Hz

unpatched version:
audio_run_in: avail 4458, acquired 4454
audio_run_in: avail 1574, acquired 1572
audio_run_in: avail 766, acquired 764
audio_run_in: avail 1052, acquired 1051
audio_run_in: avail 761, acquired 760
audio_run_in: avail 1123, acquired 1121
...
hda_audio_input_cb: could not read 4 bytes
hda_audio_input_cb: to read 1988, read 1984
audio_run_in: avail 1082, acquired 1080
...

patched version:
(no output)

QEMU was started with -device ich9-intel-hda,addr=0x1b -device 
hda-duplex,audiodev=audio0 -audiodev 
pa,out.frequency=32000,in.frequency=32000,id=audio0

playback guest 44100Hz => host 32000Hz

unpatched version:
hda_audio_output_cb: to write 8188, written 1620
audio_run_out: free 1486, played 294
hda_audio_output_cb: to write 6568, written 2512
audio_run_out: free 1192, played 455
hda_audio_output_cb: to write 4060, written 2504
audio_run_out: free 737, played 455
audio_run_out: free 282, played 281
audio_run_out: free 357, played 356
audio_run_out: free 314, played 313
...
hda_audio_output_cb: could not write 4 bytes
hda_audio_output_cb: to write 1416, written 1412
audio_run_out: free 257, played 256
...

patched version:
hda_audio_output_cb: to write 8192, written 1656
audio_run_out: free 1486, played 300
hda_audio_output_cb: to write 6536, written 2516
audio_run_out: free 1186, played 457
hda_audio_output_cb: to write 4020, written 2540
audio_run_out: free 729, played 460

record host 32000Hz => guest 44100Hz

unpatched version:
audio_run_in: avail 1486, acquired 1485
audio_run_in: avail 272, acquired 271
audio_run_in: avail 366, acquired 365
hda_audio_input_cb: could not read 4 bytes
hda_audio_input_cb: to read 1420, read 1416
audio_run_in: avail 258, acquired 257
audio_run_in: avail 375, acquired 374
hda_audio_input_cb: could not read 4 bytes
hda_audio_input_cb: to read 2056, read 2052
audio_run_in: avail 260, acquired 259
...

patched version:
(no output)

This is the debug code for the logs above.

---snip--
 > --- a/audio/audio.c    2022-12-13 19:14:31.793153558 +0100
 > +++ b/audio/audio.c    2022-12-11 16:24:48.842649711 +0100
 > @@ -1228,6 +1228,10 @@ static void audio_run_out (AudioState *s
 >  #ifdef DEBUG_OUT
 >          dolog("played=%zu\n", played);
 >  #endif
 > +        if (hw_free - played) {
 > +            fprintf(stderr, "%s: free %zu, played %zu\n",
 > +                    __func__, hw_free, played);
 > +        }
 >
 >          if (played) {
 >              hw->ts_helper += played;
 > @@ -1318,6 +1322,7 @@ static void audio_run_in (AudioState *s)
 >              if (sw->active) {
 >                  size_t sw_avail = audio_get_avail(sw);
 >                  size_t avail;
 > +                size_t prev_acquired = sw->total_hw_samples_acquired;
 >
 >                  avail = st_rate_frames_out(sw->rate, sw_avail);
 >                  if (avail > 0) {
 > @@ -1325,6 +1330,11 @@ static void audio_run_in (AudioState *s)
 > sw->callback.fn(sw->callback.opaque,
 >                                      avail * sw->info.bytes_per_frame);
 >                  }
 > +
 > +                if (sw_avail + prev_acquired - 
sw->total_hw_samples_acquired) {
 > +                    fprintf(stderr, "%s: avail %zu, acquired %zu\n", 
__func__,
 > +                            sw_avail, sw->total_hw_samples_acquired 
- prev_acquired);
 > +                }
 >              }
 >          }
 >      }
 > --- a/hw/audio/hda-codec.c    2023-01-04 14:07:31.954304889 +0100
 > +++ b/hw/audio/hda-codec.c    2023-01-04 13:57:47.687320406 +0100
 > @@ -265,20 +265,28 @@ static void hda_audio_input_cb(void *opa
 >      int64_t rpos = st->rpos;
 >
 >      int64_t to_transfer = MIN(B_SIZE - (wpos - rpos), avail);
 > +    unsigned int total_read = 0;
 >
 >      while (to_transfer) {
 >          uint32_t start = (uint32_t) (wpos & B_MASK);
 >          uint32_t chunk = (uint32_t) MIN(B_SIZE - start, to_transfer);
 >          uint32_t read = AUD_read(st->voice.in, st->buf + start, chunk);
 >          wpos += read;
 > +        total_read += read;
 >          to_transfer -= read;
 >          st->wpos += read;
 >          if (chunk != read) {
 > +            fprintf(stderr, "%s: could not read %u bytes\n", __func__,
 > +                    chunk - read);
 >              break;
 >          }
 >      }
 >
 >      hda_timer_sync_adjust(st, -((wpos - rpos) - (B_SIZE >> 1)));
 > +    if (avail != total_read) {
 > +        fprintf(stderr, "%s: to read %d, read %u\n", __func__,
 > +                avail, total_read);
 > +    }
 >  }
 >
 >  static void hda_audio_output_timer(void *opaque)
 > @@ -329,6 +337,7 @@ static void hda_audio_output_cb(void *op
 >      int64_t rpos = st->rpos;
 >
 >      int64_t to_transfer = MIN(wpos - rpos, avail);
 > +    unsigned int total_written = 0;
 >
 >      if (wpos - rpos == B_SIZE) {
 >          /* drop buffer, reset timer adjust */
 > @@ -343,15 +352,22 @@ static void hda_audio_output_cb(void *op
 >          uint32_t start = (uint32_t) (rpos & B_MASK);
 >          uint32_t chunk = (uint32_t) MIN(B_SIZE - start, to_transfer);
 >          uint32_t written = AUD_write(st->voice.out, st->buf + start, 
chunk);
 > +        total_written += written;
 >          rpos += written;
 >          to_transfer -= written;
 >          st->rpos += written;
 >          if (chunk != written) {
 > +            fprintf(stderr, "%s: could not write %u bytes\n", __func__,
 > +                    chunk - written);
 >              break;
 >          }
 >      }
 >
 >      hda_timer_sync_adjust(st, (wpos - rpos) - (B_SIZE >> 1));
 > +    if (avail != total_written) {
 > +        fprintf(stderr, "%s: to write %d, written %u\n", __func__,
 > +                avail, total_written);
 > +    }
 >  }
 >
 >  static void hda_audio_compat_input_cb(void *opaque, int avail)
---snip--

v2:
The patch series was rebased onto [PATCH v2 00/11] audio: more 
improvements. Patch 15/17 (audio: handle leftover audio frame from 
upsampling) and patch 17/17 (audio: remove sw->ratio) needed changes 
because of this.

v3:
Address Marc-André's comments.

Patches 09/17 (audio/mixeng: calculate number of input frames) and 10/17 
(audio: wire up st_rate_frames_in()) were squashed into patch 09/15 
(audio: make playback packet length calculation exact) and the function 
comment of st_rate_frames_in() has more details.

Patches 13/17 (audio/mixeng: calculate number of output frames) and 
14/17 (audio: wire up st_rate_frames_out()) were squashed into patch 
12/15 (audio: make recording packet length calculation exact) and the 
function comment of st_rate_frames_out() has more details.

Volker Rümelin (15):
   audio: change type of mix_buf and conv_buf
   audio: change type and name of the resample buffer
   audio: make the resampling code greedy
   audio: replace the resampling loop in audio_pcm_sw_write()
   audio: remove sw == NULL check
   audio: rename variables in audio_pcm_sw_write()
   audio: don't misuse audio_pcm_sw_write()
   audio: remove unused noop_conv() function
   audio: make playback packet length calculation exact
   audio: replace the resampling loop in audio_pcm_sw_read()
   audio: rename variables in audio_pcm_sw_read()
   audio: make recording packet length calculation exact
   audio: handle leftover audio frame from upsampling
   audio/audio_template: substitute sw->hw with hw
   audio: remove sw->ratio

  audio/audio.c          | 366 +++++++++++++++++++++--------------------
  audio/audio_int.h      |  12 +-
  audio/audio_template.h |  61 ++++---
  audio/mixeng.c         |  80 +++++++++
  audio/mixeng.h         |   2 +
  audio/rate_template.h  |  21 ++-
  6 files changed, 321 insertions(+), 221 deletions(-)

-- 
2.35.3

[PATCH v3 01/15] audio: change type of mix_buf and conv_buf

[Volker Rümelin]

Change the type of mix_buf in struct HWVoiceOut and conv_buf
in struct HWVoiceIn from STSampleBuffer * to STSampleBuffer.
However, a buffer pointer is still needed. For this reason in
struct STSampleBuffer samples[] is changed to *buffer.

This is a preparation for the next patch. The next patch will
add this line, which is not possible with the current struct
STSampleBuffer definition.

+        sw->resample_buf.buffer = hw->mix_buf.buffer + rpos2;

There are no functional changes.

Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
---
 audio/audio.c          | 106 ++++++++++++++++++++---------------------
 audio/audio_int.h      |   6 +--
 audio/audio_template.h |  19 ++++----
 3 files changed, 67 insertions(+), 64 deletions(-)

diff --git a/audio/audio.c b/audio/audio.c
index 772c3cc320..a0b54e4a2e 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -523,8 +523,8 @@ static size_t audio_pcm_hw_find_min_in (HWVoiceIn *hw)
 static size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw)
 {
     size_t live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
-    if (audio_bug(__func__, live > hw->conv_buf->size)) {
-        dolog("live=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
+    if (audio_bug(__func__, live > hw->conv_buf.size)) {
+        dolog("live=%zu hw->conv_buf.size=%zu\n", live, hw->conv_buf.size);
         return 0;
     }
     return live;
@@ -533,13 +533,13 @@ static size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw)
 static size_t audio_pcm_hw_conv_in(HWVoiceIn *hw, void *pcm_buf, size_t samples)
 {
     size_t conv = 0;
-    STSampleBuffer *conv_buf = hw->conv_buf;
+    STSampleBuffer *conv_buf = &hw->conv_buf;
 
     while (samples) {
         uint8_t *src = advance(pcm_buf, conv * hw->info.bytes_per_frame);
         size_t proc = MIN(samples, conv_buf->size - conv_buf->pos);
 
-        hw->conv(conv_buf->samples + conv_buf->pos, src, proc);
+        hw->conv(conv_buf->buffer + conv_buf->pos, src, proc);
         conv_buf->pos = (conv_buf->pos + proc) % conv_buf->size;
         samples -= proc;
         conv += proc;
@@ -561,12 +561,12 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
     if (!live) {
         return 0;
     }
-    if (audio_bug(__func__, live > hw->conv_buf->size)) {
-        dolog("live_in=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
+    if (audio_bug(__func__, live > hw->conv_buf.size)) {
+        dolog("live_in=%zu hw->conv_buf.size=%zu\n", live, hw->conv_buf.size);
         return 0;
     }
 
-    rpos = audio_ring_posb(hw->conv_buf->pos, live, hw->conv_buf->size);
+    rpos = audio_ring_posb(hw->conv_buf.pos, live, hw->conv_buf.size);
 
     samples = size / sw->info.bytes_per_frame;
 
@@ -574,11 +574,11 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
     swlim = MIN (swlim, samples);
 
     while (swlim) {
-        src = hw->conv_buf->samples + rpos;
-        if (hw->conv_buf->pos > rpos) {
-            isamp = hw->conv_buf->pos - rpos;
+        src = hw->conv_buf.buffer + rpos;
+        if (hw->conv_buf.pos > rpos) {
+            isamp = hw->conv_buf.pos - rpos;
         } else {
-            isamp = hw->conv_buf->size - rpos;
+            isamp = hw->conv_buf.size - rpos;
         }
 
         if (!isamp) {
@@ -588,7 +588,7 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
 
         st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
         swlim -= osamp;
-        rpos = (rpos + isamp) % hw->conv_buf->size;
+        rpos = (rpos + isamp) % hw->conv_buf.size;
         dst += osamp;
         ret += osamp;
         total += isamp;
@@ -636,8 +636,8 @@ static size_t audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
     if (nb_live1) {
         size_t live = smin;
 
-        if (audio_bug(__func__, live > hw->mix_buf->size)) {
-            dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size);
+        if (audio_bug(__func__, live > hw->mix_buf.size)) {
+            dolog("live=%zu hw->mix_buf.size=%zu\n", live, hw->mix_buf.size);
             return 0;
         }
         return live;
@@ -654,17 +654,17 @@ static size_t audio_pcm_hw_get_free(HWVoiceOut *hw)
 static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len)
 {
     size_t clipped = 0;
-    size_t pos = hw->mix_buf->pos;
+    size_t pos = hw->mix_buf.pos;
 
     while (len) {
-        st_sample *src = hw->mix_buf->samples + pos;
+        st_sample *src = hw->mix_buf.buffer + pos;
         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_till_end_of_buf = hw->mix_buf.size - pos;
         size_t samples_to_clip = MIN(len, samples_till_end_of_buf);
 
         hw->clip(dst, src, samples_to_clip);
 
-        pos = (pos + samples_to_clip) % hw->mix_buf->size;
+        pos = (pos + samples_to_clip) % hw->mix_buf.size;
         len -= samples_to_clip;
         clipped += samples_to_clip;
     }
@@ -683,11 +683,11 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
         return size;
     }
 
-    hwsamples = sw->hw->mix_buf->size;
+    hwsamples = sw->hw->mix_buf.size;
 
     live = sw->total_hw_samples_mixed;
     if (audio_bug(__func__, live > hwsamples)) {
-        dolog("live=%zu hw->mix_buf->size=%zu\n", live, hwsamples);
+        dolog("live=%zu hw->mix_buf.size=%zu\n", live, hwsamples);
         return 0;
     }
 
@@ -698,7 +698,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
         return 0;
     }
 
-    wpos = (sw->hw->mix_buf->pos + live) % hwsamples;
+    wpos = (sw->hw->mix_buf.pos + live) % hwsamples;
 
     dead = hwsamples - live;
     hw_free = audio_pcm_hw_get_free(sw->hw);
@@ -725,7 +725,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
         st_rate_flow_mix (
             sw->rate,
             sw->buf + pos,
-            sw->hw->mix_buf->samples + wpos,
+            sw->hw->mix_buf.buffer + wpos,
             &isamp,
             &osamp
             );
@@ -989,9 +989,9 @@ static size_t audio_get_avail (SWVoiceIn *sw)
     }
 
     live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
-    if (audio_bug(__func__, live > sw->hw->conv_buf->size)) {
-        dolog("live=%zu sw->hw->conv_buf->size=%zu\n", live,
-              sw->hw->conv_buf->size);
+    if (audio_bug(__func__, live > sw->hw->conv_buf.size)) {
+        dolog("live=%zu sw->hw->conv_buf.size=%zu\n", live,
+              sw->hw->conv_buf.size);
         return 0;
     }
 
@@ -1026,13 +1026,13 @@ static size_t audio_get_free(SWVoiceOut *sw)
 
     live = sw->total_hw_samples_mixed;
 
-    if (audio_bug(__func__, live > sw->hw->mix_buf->size)) {
-        dolog("live=%zu sw->hw->mix_buf->size=%zu\n", live,
-              sw->hw->mix_buf->size);
+    if (audio_bug(__func__, live > sw->hw->mix_buf.size)) {
+        dolog("live=%zu sw->hw->mix_buf.size=%zu\n", live,
+              sw->hw->mix_buf.size);
         return 0;
     }
 
-    dead = sw->hw->mix_buf->size - live;
+    dead = sw->hw->mix_buf.size - live;
 
 #ifdef DEBUG_OUT
     dolog("%s: get_free live %zu dead %zu frontend frames %zu\n",
@@ -1056,12 +1056,12 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
 
             n = samples;
             while (n) {
-                size_t till_end_of_hw = hw->mix_buf->size - rpos2;
+                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.bytes_per_frame;
                 size_t written;
 
-                sw->buf = hw->mix_buf->samples + rpos2;
+                sw->buf = hw->mix_buf.buffer + rpos2;
                 written = audio_pcm_sw_write (sw, NULL, bytes);
                 if (written - bytes) {
                     dolog("Could not mix %zu bytes into a capture "
@@ -1070,14 +1070,14 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
                     break;
                 }
                 n -= to_write;
-                rpos2 = (rpos2 + to_write) % hw->mix_buf->size;
+                rpos2 = (rpos2 + to_write) % hw->mix_buf.size;
             }
         }
     }
 
-    n = MIN(samples, hw->mix_buf->size - rpos);
-    mixeng_clear(hw->mix_buf->samples + rpos, n);
-    mixeng_clear(hw->mix_buf->samples, samples - n);
+    n = MIN(samples, hw->mix_buf.size - rpos);
+    mixeng_clear(hw->mix_buf.buffer + rpos, n);
+    mixeng_clear(hw->mix_buf.buffer, samples - n);
 }
 
 static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
@@ -1103,7 +1103,7 @@ static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
 
         live -= proc;
         clipped += proc;
-        hw->mix_buf->pos = (hw->mix_buf->pos + proc) % hw->mix_buf->size;
+        hw->mix_buf.pos = (hw->mix_buf.pos + proc) % hw->mix_buf.size;
 
         if (proc == 0 || proc < decr) {
             break;
@@ -1174,8 +1174,8 @@ static void audio_run_out (AudioState *s)
             live = 0;
         }
 
-        if (audio_bug(__func__, live > hw->mix_buf->size)) {
-            dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size);
+        if (audio_bug(__func__, live > hw->mix_buf.size)) {
+            dolog("live=%zu hw->mix_buf.size=%zu\n", live, hw->mix_buf.size);
             continue;
         }
 
@@ -1203,13 +1203,13 @@ static void audio_run_out (AudioState *s)
             continue;
         }
 
-        prev_rpos = hw->mix_buf->pos;
+        prev_rpos = hw->mix_buf.pos;
         played = audio_pcm_hw_run_out(hw, live);
         replay_audio_out(&played);
-        if (audio_bug(__func__, hw->mix_buf->pos >= hw->mix_buf->size)) {
-            dolog("hw->mix_buf->pos=%zu hw->mix_buf->size=%zu played=%zu\n",
-                  hw->mix_buf->pos, hw->mix_buf->size, played);
-            hw->mix_buf->pos = 0;
+        if (audio_bug(__func__, hw->mix_buf.pos >= hw->mix_buf.size)) {
+            dolog("hw->mix_buf.pos=%zu hw->mix_buf.size=%zu played=%zu\n",
+                  hw->mix_buf.pos, hw->mix_buf.size, played);
+            hw->mix_buf.pos = 0;
         }
 
 #ifdef DEBUG_OUT
@@ -1290,10 +1290,10 @@ static void audio_run_in (AudioState *s)
 
         if (replay_mode != REPLAY_MODE_PLAY) {
             captured = audio_pcm_hw_run_in(
-                hw, hw->conv_buf->size - audio_pcm_hw_get_live_in(hw));
+                hw, hw->conv_buf.size - audio_pcm_hw_get_live_in(hw));
         }
-        replay_audio_in(&captured, hw->conv_buf->samples, &hw->conv_buf->pos,
-                        hw->conv_buf->size);
+        replay_audio_in(&captured, hw->conv_buf.buffer, &hw->conv_buf.pos,
+                        hw->conv_buf.size);
 
         min = audio_pcm_hw_find_min_in (hw);
         hw->total_samples_captured += captured - min;
@@ -1326,14 +1326,14 @@ static void audio_run_capture (AudioState *s)
         SWVoiceOut *sw;
 
         captured = live = audio_pcm_hw_get_live_out (hw, NULL);
-        rpos = hw->mix_buf->pos;
+        rpos = hw->mix_buf.pos;
         while (live) {
-            size_t left = hw->mix_buf->size - rpos;
+            size_t left = hw->mix_buf.size - rpos;
             size_t to_capture = MIN(live, left);
             struct st_sample *src;
             struct capture_callback *cb;
 
-            src = hw->mix_buf->samples + rpos;
+            src = hw->mix_buf.buffer + rpos;
             hw->clip (cap->buf, src, to_capture);
             mixeng_clear (src, to_capture);
 
@@ -1341,10 +1341,10 @@ static void audio_run_capture (AudioState *s)
                 cb->ops.capture (cb->opaque, cap->buf,
                                  to_capture * hw->info.bytes_per_frame);
             }
-            rpos = (rpos + to_capture) % hw->mix_buf->size;
+            rpos = (rpos + to_capture) % hw->mix_buf.size;
             live -= to_capture;
         }
-        hw->mix_buf->pos = rpos;
+        hw->mix_buf.pos = rpos;
 
         for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
             if (!sw->active && sw->empty) {
@@ -1903,7 +1903,7 @@ CaptureVoiceOut *AUD_add_capture(
 
         audio_pcm_init_info (&hw->info, as);
 
-        cap->buf = g_malloc0_n(hw->mix_buf->size, hw->info.bytes_per_frame);
+        cap->buf = g_malloc0_n(hw->mix_buf.size, hw->info.bytes_per_frame);
 
         if (hw->info.is_float) {
             hw->clip = mixeng_clip_float[hw->info.nchannels == 2];
@@ -1955,7 +1955,7 @@ void AUD_del_capture (CaptureVoiceOut *cap, void *cb_opaque)
                     sw = sw1;
                 }
                 QLIST_REMOVE (cap, entries);
-                g_free (cap->hw.mix_buf);
+                g_free(cap->hw.mix_buf.buffer);
                 g_free (cap->buf);
                 g_free (cap);
             }
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 5028f2354a..061845dcc2 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -58,7 +58,7 @@ typedef struct SWVoiceCap SWVoiceCap;
 
 typedef struct STSampleBuffer {
     size_t pos, size;
-    st_sample samples[];
+    st_sample *buffer;
 } STSampleBuffer;
 
 typedef struct HWVoiceOut {
@@ -71,7 +71,7 @@ typedef struct HWVoiceOut {
     f_sample *clip;
     uint64_t ts_helper;
 
-    STSampleBuffer *mix_buf;
+    STSampleBuffer mix_buf;
     void *buf_emul;
     size_t pos_emul, pending_emul, size_emul;
 
@@ -93,7 +93,7 @@ typedef struct HWVoiceIn {
     size_t total_samples_captured;
     uint64_t ts_helper;
 
-    STSampleBuffer *conv_buf;
+    STSampleBuffer conv_buf;
     void *buf_emul;
     size_t pos_emul, pending_emul, size_emul;
 
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 980e1f4bd0..dd87170cbd 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -71,8 +71,9 @@ static void glue(audio_init_nb_voices_, TYPE)(AudioState *s,
 static void glue (audio_pcm_hw_free_resources_, TYPE) (HW *hw)
 {
     g_free(hw->buf_emul);
-    g_free (HWBUF);
-    HWBUF = NULL;
+    g_free(HWBUF.buffer);
+    HWBUF.buffer = NULL;
+    HWBUF.size = 0;
 }
 
 static void glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw)
@@ -83,10 +84,12 @@ static void glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw)
             dolog("Attempted to allocate empty buffer\n");
         }
 
-        HWBUF = g_malloc0(sizeof(STSampleBuffer) + sizeof(st_sample) * samples);
-        HWBUF->size = samples;
+        HWBUF.buffer = g_new0(st_sample, samples);
+        HWBUF.size = samples;
+        HWBUF.pos = 0;
     } else {
-        HWBUF = NULL;
+        HWBUF.buffer = NULL;
+        HWBUF.size = 0;
     }
 }
 
@@ -111,9 +114,9 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw)
     }
 
 #ifdef DAC
-    samples = ((int64_t) sw->HWBUF->size << 32) / sw->ratio;
+    samples = ((int64_t)sw->HWBUF.size << 32) / sw->ratio;
 #else
-    samples = (int64_t)sw->HWBUF->size * sw->ratio >> 32;
+    samples = (int64_t)sw->HWBUF.size * sw->ratio >> 32;
 #endif
     if (audio_bug(__func__, samples < 0)) {
         dolog("Can not allocate buffer for `%s' (%d samples)\n",
@@ -126,7 +129,7 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw)
         size_t f_fe_min;
 
         /* f_fe_min = ceil(1 [frames] * f_be [Hz] / size_be [frames]) */
-        f_fe_min = (hw->info.freq + HWBUF->size - 1) / HWBUF->size;
+        f_fe_min = (hw->info.freq + HWBUF.size - 1) / HWBUF.size;
         qemu_log_mask(LOG_UNIMP,
                       AUDIO_CAP ": The guest selected a " NAME " sample rate"
                       " of %d Hz for %s. Only sample rates >= %zu Hz are"
-- 
2.35.3

[PATCH v3 02/15] audio: change type and name of the resample buffer

[Volker Rümelin]

Change the type of the resample buffer from struct st_sample *
to STSampleBuffer. Also change the name from buf to resample_buf
for better readability.

The new variables resample_buf.size and resample_buf.pos will be
used after the next patches. There is no functional change.

Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
---
 audio/audio.c          | 15 ++++++++-------
 audio/audio_int.h      |  4 ++--
 audio/audio_template.h | 10 ++++++----
 3 files changed, 16 insertions(+), 13 deletions(-)

diff --git a/audio/audio.c b/audio/audio.c
index a0b54e4a2e..a399147486 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -555,7 +555,7 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
 {
     HWVoiceIn *hw = sw->hw;
     size_t samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
-    struct st_sample *src, *dst = sw->buf;
+    struct st_sample *src, *dst = sw->resample_buf.buffer;
 
     live = hw->total_samples_captured - sw->total_hw_samples_acquired;
     if (!live) {
@@ -595,10 +595,10 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
     }
 
     if (!hw->pcm_ops->volume_in) {
-        mixeng_volume (sw->buf, ret, &sw->vol);
+        mixeng_volume(sw->resample_buf.buffer, ret, &sw->vol);
     }
 
-    sw->clip (buf, sw->buf, ret);
+    sw->clip(buf, sw->resample_buf.buffer, ret);
     sw->total_hw_samples_acquired += total;
     return ret * sw->info.bytes_per_frame;
 }
@@ -706,10 +706,10 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
     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);
+        sw->conv(sw->resample_buf.buffer, buf, samples);
 
         if (!sw->hw->pcm_ops->volume_out) {
-            mixeng_volume(sw->buf, samples, &sw->vol);
+            mixeng_volume(sw->resample_buf.buffer, samples, &sw->vol);
         }
     }
 
@@ -724,7 +724,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
         osamp = blck;
         st_rate_flow_mix (
             sw->rate,
-            sw->buf + pos,
+            sw->resample_buf.buffer + pos,
             sw->hw->mix_buf.buffer + wpos,
             &isamp,
             &osamp
@@ -1061,7 +1061,8 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
                 size_t bytes = to_write * hw->info.bytes_per_frame;
                 size_t written;
 
-                sw->buf = hw->mix_buf.buffer + rpos2;
+                sw->resample_buf.buffer = hw->mix_buf.buffer + rpos2;
+                sw->resample_buf.size = to_write;
                 written = audio_pcm_sw_write (sw, NULL, bytes);
                 if (written - bytes) {
                     dolog("Could not mix %zu bytes into a capture "
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 061845dcc2..8b163e1759 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -109,7 +109,7 @@ struct SWVoiceOut {
     struct audio_pcm_info info;
     t_sample *conv;
     int64_t ratio;
-    struct st_sample *buf;
+    STSampleBuffer resample_buf;
     void *rate;
     size_t total_hw_samples_mixed;
     int active;
@@ -129,7 +129,7 @@ struct SWVoiceIn {
     int64_t ratio;
     void *rate;
     size_t total_hw_samples_acquired;
-    struct st_sample *buf;
+    STSampleBuffer resample_buf;
     f_sample *clip;
     HWVoiceIn *hw;
     char *name;
diff --git a/audio/audio_template.h b/audio/audio_template.h
index dd87170cbd..a0b653f52c 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -95,13 +95,13 @@ static void glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw)
 
 static void glue (audio_pcm_sw_free_resources_, TYPE) (SW *sw)
 {
-    g_free (sw->buf);
+    g_free(sw->resample_buf.buffer);
+    sw->resample_buf.buffer = NULL;
+    sw->resample_buf.size = 0;
 
     if (sw->rate) {
         st_rate_stop (sw->rate);
     }
-
-    sw->buf = NULL;
     sw->rate = NULL;
 }
 
@@ -138,7 +138,9 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw)
         return -1;
     }
 
-    sw->buf = g_new0(st_sample, samples);
+    sw->resample_buf.buffer = g_new0(st_sample, samples);
+    sw->resample_buf.size = samples;
+    sw->resample_buf.pos = 0;
 
 #ifdef DAC
     sw->rate = st_rate_start (sw->info.freq, sw->hw->info.freq);
-- 
2.35.3

[PATCH v3 03/15] audio: make the resampling code greedy

[Volker Rümelin]

Read the maximum possible number of audio frames instead of the
minimum necessary number of frames when the audio stream is
downsampled and the output buffer is limited. This makes the
function symmetrical to upsampling when the input buffer is
limited. The maximum possible number of frames is written here.

With this change it's easier to calculate the exact number of
audio frames the resample function will read or write. These two
functions will be introduced later.

Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Acked-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
---
 audio/rate_template.h | 21 +++++++++++++--------
 1 file changed, 13 insertions(+), 8 deletions(-)

diff --git a/audio/rate_template.h b/audio/rate_template.h
index b432719ebb..6648f0d2e5 100644
--- a/audio/rate_template.h
+++ b/audio/rate_template.h
@@ -40,8 +40,6 @@ void NAME (void *opaque, struct st_sample *ibuf, struct st_sample *obuf,
     int64_t t;
 #endif
 
-    ilast = rate->ilast;
-
     istart = ibuf;
     iend = ibuf + *isamp;
 
@@ -59,15 +57,17 @@ void NAME (void *opaque, struct st_sample *ibuf, struct st_sample *obuf,
         return;
     }
 
-    while (obuf < oend) {
+    /* without input samples, there's nothing to do */
+    if (ibuf >= iend) {
+        *osamp = 0;
+        return;
+    }
 
-        /* Safety catch to make sure we have input samples.  */
-        if (ibuf >= iend) {
-            break;
-        }
+    ilast = rate->ilast;
 
-        /* read as many input samples so that ipos > opos */
+    while (true) {
 
+        /* read as many input samples so that ipos > opos */
         while (rate->ipos <= (rate->opos >> 32)) {
             ilast = *ibuf++;
             rate->ipos++;
@@ -78,6 +78,11 @@ void NAME (void *opaque, struct st_sample *ibuf, struct st_sample *obuf,
             }
         }
 
+        /* make sure that the next output sample can be written */
+        if (obuf >= oend) {
+            break;
+        }
+
         icur = *ibuf;
 
         /* wrap ipos and opos around long before they overflow */
-- 
2.35.3

[PATCH v3 04/15] audio: replace the resampling loop in audio_pcm_sw_write()

[Volker Rümelin]

Replace the resampling loop in audio_pcm_sw_write() with the new
function audio_pcm_sw_resample_out(). Unlike the old resample
loop the new function will try to consume input frames even if
the output buffer is full. This is necessary when downsampling
to avoid reading less audio frames than calculated in advance.
The loop was unrolled to avoid complicated loop control conditions
in this case.

Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Acked-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
---
 audio/audio.c | 63 +++++++++++++++++++++++++++++----------------------
 1 file changed, 36 insertions(+), 27 deletions(-)

diff --git a/audio/audio.c b/audio/audio.c
index a399147486..4412b5fad8 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -673,11 +673,44 @@ static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len)
 /*
  * Soft voice (playback)
  */
+static void audio_pcm_sw_resample_out(SWVoiceOut *sw,
+    size_t frames_in_max, size_t frames_out_max,
+    size_t *total_in, size_t *total_out)
+{
+    HWVoiceOut *hw = sw->hw;
+    struct st_sample *src, *dst;
+    size_t live, wpos, frames_in, frames_out;
+
+    live = sw->total_hw_samples_mixed;
+    wpos = (hw->mix_buf.pos + live) % hw->mix_buf.size;
+
+    /* write to mix_buf from wpos to end of buffer */
+    src = sw->resample_buf.buffer;
+    frames_in = frames_in_max;
+    dst = hw->mix_buf.buffer + wpos;
+    frames_out = MIN(frames_out_max, hw->mix_buf.size - wpos);
+    st_rate_flow_mix(sw->rate, src, dst, &frames_in, &frames_out);
+    wpos += frames_out;
+    *total_in = frames_in;
+    *total_out = frames_out;
+
+    /* write to mix_buf from start of buffer if there are input frames left */
+    if (frames_in_max - frames_in > 0 && wpos == hw->mix_buf.size) {
+        src += frames_in;
+        frames_in = frames_in_max - frames_in;
+        dst = hw->mix_buf.buffer;
+        frames_out = frames_out_max - frames_out;
+        st_rate_flow_mix(sw->rate, src, dst, &frames_in, &frames_out);
+        *total_in += frames_in;
+        *total_out += frames_out;
+    }
+}
+
 static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
 {
-    size_t hwsamples, samples, isamp, osamp, wpos, live, dead, left, blck;
+    size_t hwsamples, samples, live, dead;
     size_t hw_free;
-    size_t ret = 0, pos = 0, total = 0;
+    size_t ret, total;
 
     if (!sw) {
         return size;
@@ -698,8 +731,6 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
         return 0;
     }
 
-    wpos = (sw->hw->mix_buf.pos + live) % hwsamples;
-
     dead = hwsamples - live;
     hw_free = audio_pcm_hw_get_free(sw->hw);
     hw_free = hw_free > live ? hw_free - live : 0;
@@ -713,29 +744,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
         }
     }
 
-    while (samples) {
-        dead = hwsamples - live;
-        left = hwsamples - wpos;
-        blck = MIN (dead, left);
-        if (!blck) {
-            break;
-        }
-        isamp = samples;
-        osamp = blck;
-        st_rate_flow_mix (
-            sw->rate,
-            sw->resample_buf.buffer + pos,
-            sw->hw->mix_buf.buffer + wpos,
-            &isamp,
-            &osamp
-            );
-        ret += isamp;
-        samples -= isamp;
-        pos += isamp;
-        live += osamp;
-        wpos = (wpos + osamp) % hwsamples;
-        total += osamp;
-    }
+    audio_pcm_sw_resample_out(sw, samples, MIN(dead, hw_free), &ret, &total);
 
     sw->total_hw_samples_mixed += total;
     sw->empty = sw->total_hw_samples_mixed == 0;
-- 
2.35.3

[PATCH v3 05/15] audio: remove sw == NULL check

[Volker Rümelin]

All call sites of audio_pcm_sw_write() guarantee that sw is not
NULL. Remove the unnecessary NULL check.

Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
---
 audio/audio.c | 4 ----
 1 file changed, 4 deletions(-)

diff --git a/audio/audio.c b/audio/audio.c
index 4412b5fad8..8f1c0e77b0 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -712,10 +712,6 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
     size_t hw_free;
     size_t ret, total;
 
-    if (!sw) {
-        return size;
-    }
-
     hwsamples = sw->hw->mix_buf.size;
 
     live = sw->total_hw_samples_mixed;
-- 
2.35.3

[PATCH v3 06/15] audio: rename variables in audio_pcm_sw_write()

[Volker Rümelin]

The audio_pcm_sw_write() function uses a lot of very unspecific
variable names. Rename them for better readability.

ret => total_in
total => total_out
size => buf_len
hwsamples => hw->mix_buf.size
samples => frames_in_max

Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
---
 audio/audio.c | 45 ++++++++++++++++++++++-----------------------
 1 file changed, 22 insertions(+), 23 deletions(-)

diff --git a/audio/audio.c b/audio/audio.c
index 8f1c0e77b0..cd10f1ec10 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -706,56 +706,55 @@ static void audio_pcm_sw_resample_out(SWVoiceOut *sw,
     }
 }
 
-static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
+static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t buf_len)
 {
-    size_t hwsamples, samples, live, dead;
-    size_t hw_free;
-    size_t ret, total;
-
-    hwsamples = sw->hw->mix_buf.size;
+    HWVoiceOut *hw = sw->hw;
+    size_t live, dead, hw_free;
+    size_t frames_in_max, total_in, total_out;
 
     live = sw->total_hw_samples_mixed;
-    if (audio_bug(__func__, live > hwsamples)) {
-        dolog("live=%zu hw->mix_buf.size=%zu\n", live, hwsamples);
+    if (audio_bug(__func__, live > hw->mix_buf.size)) {
+        dolog("live=%zu hw->mix_buf.size=%zu\n", live, hw->mix_buf.size);
         return 0;
     }
 
-    if (live == hwsamples) {
+    if (live == hw->mix_buf.size) {
 #ifdef DEBUG_OUT
         dolog ("%s is full %zu\n", sw->name, live);
 #endif
         return 0;
     }
 
-    dead = hwsamples - live;
-    hw_free = audio_pcm_hw_get_free(sw->hw);
+    dead = hw->mix_buf.size - live;
+    hw_free = audio_pcm_hw_get_free(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->resample_buf.buffer, buf, samples);
+    frames_in_max = ((int64_t)MIN(dead, hw_free) << 32) / sw->ratio;
+    frames_in_max = MIN(frames_in_max, buf_len / sw->info.bytes_per_frame);
+    if (frames_in_max) {
+        sw->conv(sw->resample_buf.buffer, buf, frames_in_max);
 
         if (!sw->hw->pcm_ops->volume_out) {
-            mixeng_volume(sw->resample_buf.buffer, samples, &sw->vol);
+            mixeng_volume(sw->resample_buf.buffer, frames_in_max, &sw->vol);
         }
     }
 
-    audio_pcm_sw_resample_out(sw, samples, MIN(dead, hw_free), &ret, &total);
+    audio_pcm_sw_resample_out(sw, frames_in_max, MIN(dead, hw_free),
+                              &total_in, &total_out);
 
-    sw->total_hw_samples_mixed += total;
+    sw->total_hw_samples_mixed += total_out;
     sw->empty = sw->total_hw_samples_mixed == 0;
 
 #ifdef DEBUG_OUT
     dolog (
-        "%s: write size %zu ret %zu total sw %zu\n",
-        SW_NAME (sw),
-        size / sw->info.bytes_per_frame,
-        ret,
+        "%s: write size %zu written %zu total mixed %zu\n",
+        SW_NAME(sw),
+        buf_len / sw->info.bytes_per_frame,
+        total_in,
         sw->total_hw_samples_mixed
         );
 #endif
 
-    return ret * sw->info.bytes_per_frame;
+    return total_in * sw->info.bytes_per_frame;
 }
 
 #ifdef DEBUG_AUDIO
-- 
2.35.3

[PATCH v3 07/15] audio: don't misuse audio_pcm_sw_write()

[Volker Rümelin]

The audio_pcm_sw_write() function is intended to convert a
PCM audio stream to the internal representation, adjust the
volume, and then mix it with the other audio streams with a
possibly changed sample rate in mix_buf. In order for the
audio_capture_mix_and_clear() function to use audio_pcm_sw_write(),
it must bypass the first two tasks of audio_pcm_sw_write().

Since patch "audio: split out the resampling loop in
audio_pcm_sw_write()" this is no longer necessary, because now
the audio_pcm_sw_resample_out() function can be used instead of
audio_pcm_sw_write().

Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Acked-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
---
 audio/audio.c | 29 ++++++++++++++++++-----------
 1 file changed, 18 insertions(+), 11 deletions(-)

diff --git a/audio/audio.c b/audio/audio.c
index cd10f1ec10..44eb7b63b4 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -1056,26 +1056,33 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
 
         for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
             SWVoiceOut *sw = &sc->sw;
-            int rpos2 = rpos;
+            size_t rpos2 = rpos;
 
             n = samples;
             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.bytes_per_frame;
-                size_t written;
+                size_t to_read = MIN(till_end_of_hw, n);
+                size_t live, frames_in, frames_out;
 
                 sw->resample_buf.buffer = hw->mix_buf.buffer + rpos2;
-                sw->resample_buf.size = to_write;
-                written = audio_pcm_sw_write (sw, NULL, bytes);
-                if (written - bytes) {
-                    dolog("Could not mix %zu bytes into a capture "
+                sw->resample_buf.size = to_read;
+                live = sw->total_hw_samples_mixed;
+
+                audio_pcm_sw_resample_out(sw,
+                                          to_read, sw->hw->mix_buf.size - live,
+                                          &frames_in, &frames_out);
+
+                sw->total_hw_samples_mixed += frames_out;
+                sw->empty = sw->total_hw_samples_mixed == 0;
+
+                if (to_read - frames_in) {
+                    dolog("Could not mix %zu frames into a capture "
                           "buffer, mixed %zu\n",
-                          bytes, written);
+                          to_read, frames_in);
                     break;
                 }
-                n -= to_write;
-                rpos2 = (rpos2 + to_write) % hw->mix_buf.size;
+                n -= to_read;
+                rpos2 = (rpos2 + to_read) % hw->mix_buf.size;
             }
         }
     }
-- 
2.35.3

[PATCH v3 08/15] audio: remove unused noop_conv() function

[Volker Rümelin]

The function audio_capture_mix_and_clear() no longer uses
audio_pcm_sw_write() to resample audio frames from one internal
buffer to another. For this reason, the noop_conv() function is
now unused. Remove it.

Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
---
 audio/audio.c | 8 --------
 1 file changed, 8 deletions(-)

diff --git a/audio/audio.c b/audio/audio.c
index 44eb7b63b4..556696b095 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -381,13 +381,6 @@ void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
 /*
  * Capture
  */
-static void noop_conv (struct st_sample *dst, const void *src, int samples)
-{
-    (void) src;
-    (void) dst;
-    (void) samples;
-}
-
 static CaptureVoiceOut *audio_pcm_capture_find_specific(AudioState *s,
                                                         struct audsettings *as)
 {
@@ -485,7 +478,6 @@ static int audio_attach_capture (HWVoiceOut *hw)
         sw->info = hw->info;
         sw->empty = 1;
         sw->active = hw->enabled;
-        sw->conv = noop_conv;
         sw->ratio = ((int64_t) hw_cap->info.freq << 32) / sw->info.freq;
         sw->vol = nominal_volume;
         sw->rate = st_rate_start (sw->info.freq, hw_cap->info.freq);
-- 
2.35.3

[PATCH v3 09/15] audio: make playback packet length calculation exact

[Volker Rümelin]

Introduce the new function st_rate_frames_in() to calculate the
exact number of audio input frames needed to get a given number
of audio output frames. The exact number of frames depends only
on the difference of opos - ipos and the number of output frames.
When downsampling, this function returns the maximum number of
input frames needed.

This new function replaces the audio_frontend_frames_out() function,
which calculated the average number of input frames rounded down
to the nearest integer. Because audio_frontend_frames_out() also
limited the number of input frames to the size of the resample
buffer, st_rate_frames_in() is not a direct replacement and two
additional MIN() functions are needed. One to prevent resample
buffer overflows and one to limit the available bytes for the audio
frontends.

After this patch the audio packet length calculation for playback is
exact. When upsampling, it's still possible that the audio frontends
can't write the last audio frame. This will be fixed later.

Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
---
 audio/audio.c  | 43 ++++++++++++++++++-------------------------
 audio/mixeng.c | 39 +++++++++++++++++++++++++++++++++++++++
 audio/mixeng.h |  1 +
 3 files changed, 58 insertions(+), 25 deletions(-)

diff --git a/audio/audio.c b/audio/audio.c
index 556696b095..e18b5e98c5 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -701,8 +701,8 @@ static void audio_pcm_sw_resample_out(SWVoiceOut *sw,
 static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t buf_len)
 {
     HWVoiceOut *hw = sw->hw;
-    size_t live, dead, hw_free;
-    size_t frames_in_max, total_in, total_out;
+    size_t live, dead, hw_free, sw_max, fe_max;
+    size_t frames_in_max, frames_out_max, total_in, total_out;
 
     live = sw->total_hw_samples_mixed;
     if (audio_bug(__func__, live > hw->mix_buf.size)) {
@@ -720,17 +720,21 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t buf_len)
     dead = hw->mix_buf.size - live;
     hw_free = audio_pcm_hw_get_free(hw);
     hw_free = hw_free > live ? hw_free - live : 0;
-    frames_in_max = ((int64_t)MIN(dead, hw_free) << 32) / sw->ratio;
-    frames_in_max = MIN(frames_in_max, buf_len / sw->info.bytes_per_frame);
-    if (frames_in_max) {
-        sw->conv(sw->resample_buf.buffer, buf, frames_in_max);
+    frames_out_max = MIN(dead, hw_free);
+    sw_max = st_rate_frames_in(sw->rate, frames_out_max);
+    fe_max = MIN(buf_len / sw->info.bytes_per_frame, sw->resample_buf.size);
+    frames_in_max = MIN(sw_max, fe_max);
 
-        if (!sw->hw->pcm_ops->volume_out) {
-            mixeng_volume(sw->resample_buf.buffer, frames_in_max, &sw->vol);
-        }
+    if (!frames_in_max) {
+        return 0;
     }
 
-    audio_pcm_sw_resample_out(sw, frames_in_max, MIN(dead, hw_free),
+    sw->conv(sw->resample_buf.buffer, buf, frames_in_max);
+    if (!sw->hw->pcm_ops->volume_out) {
+        mixeng_volume(sw->resample_buf.buffer, frames_in_max, &sw->vol);
+    }
+
+    audio_pcm_sw_resample_out(sw, frames_in_max, frames_out_max,
                               &total_in, &total_out);
 
     sw->total_hw_samples_mixed += total_out;
@@ -1000,18 +1004,6 @@ static size_t audio_get_avail (SWVoiceIn *sw)
     return live;
 }
 
-/**
- * audio_frontend_frames_out() - returns the number of frames needed to
- * get frames_out frames after resampling
- *
- * @sw: audio playback frontend
- * @frames_out: number of frames
- */
-static size_t audio_frontend_frames_out(SWVoiceOut *sw, size_t frames_out)
-{
-    return ((int64_t)frames_out << 32) / sw->ratio;
-}
-
 static size_t audio_get_free(SWVoiceOut *sw)
 {
     size_t live, dead;
@@ -1031,8 +1023,8 @@ 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 frontend frames %zu\n",
-          SW_NAME(sw), live, dead, audio_frontend_frames_out(sw, dead));
+    dolog("%s: get_free live %zu dead %zu frontend frames %u\n",
+          SW_NAME(sw), live, dead, st_rate_frames_in(sw->rate, dead));
 #endif
 
     return dead;
@@ -1161,12 +1153,13 @@ static void audio_run_out (AudioState *s)
                 size_t free;
 
                 if (hw_free > sw->total_hw_samples_mixed) {
-                    free = audio_frontend_frames_out(sw,
+                    free = st_rate_frames_in(sw->rate,
                         MIN(sw_free, hw_free - sw->total_hw_samples_mixed));
                 } else {
                     free = 0;
                 }
                 if (free > 0) {
+                    free = MIN(free, sw->resample_buf.size);
                     sw->callback.fn(sw->callback.opaque,
                                     free * sw->info.bytes_per_frame);
                 }
diff --git a/audio/mixeng.c b/audio/mixeng.c
index fe454e0725..a24c8c45a7 100644
--- a/audio/mixeng.c
+++ b/audio/mixeng.c
@@ -440,6 +440,45 @@ void st_rate_stop (void *opaque)
     g_free (opaque);
 }
 
+/**
+ * st_rate_frames_in() - returns the number of frames needed to
+ * get frames_out frames after resampling
+ *
+ * @opaque: pointer to struct rate
+ * @frames_out: number of frames
+ *
+ * When downsampling, there may be more than one correct result. In this
+ * case, the function returns the maximum number of input frames needed.
+ */
+uint32_t st_rate_frames_in(void *opaque, uint32_t frames_out)
+{
+    struct rate *rate = opaque;
+    uint64_t opos_start, opos_end;
+    uint32_t ipos_start, ipos_end;
+
+    if (rate->opos_inc == 1ULL << 32) {
+        return frames_out;
+    }
+
+    if (frames_out) {
+        opos_start = rate->opos;
+        ipos_start = rate->ipos;
+    } else {
+        uint64_t offset;
+
+        /* add offset = ceil(opos_inc) to opos and ipos to avoid an underflow */
+        offset = (rate->opos_inc + (1ULL << 32) - 1) & ~((1ULL << 32) - 1);
+        opos_start = rate->opos + offset;
+        ipos_start = rate->ipos + (offset >> 32);
+    }
+    /* last frame written was at opos_start - rate->opos_inc */
+    opos_end = opos_start - rate->opos_inc + rate->opos_inc * frames_out;
+    ipos_end = (opos_end >> 32) + 1;
+
+    /* last frame read was at ipos_start - 1 */
+    return ipos_end + 1 > ipos_start ? ipos_end + 1 - ipos_start : 0;
+}
+
 void mixeng_clear (struct st_sample *buf, int len)
 {
     memset (buf, 0, len * sizeof (struct st_sample));
diff --git a/audio/mixeng.h b/audio/mixeng.h
index 2dcd6df245..64c1e231cc 100644
--- a/audio/mixeng.h
+++ b/audio/mixeng.h
@@ -52,6 +52,7 @@ void st_rate_flow(void *opaque, st_sample *ibuf, st_sample *obuf,
 void st_rate_flow_mix(void *opaque, st_sample *ibuf, st_sample *obuf,
                       size_t *isamp, size_t *osamp);
 void st_rate_stop (void *opaque);
+uint32_t st_rate_frames_in(void *opaque, uint32_t frames_out);
 void mixeng_clear (struct st_sample *buf, int len);
 void mixeng_volume (struct st_sample *buf, int len, struct mixeng_volume *vol);
 
-- 
2.35.3

[PATCH v3 10/15] audio: replace the resampling loop in audio_pcm_sw_read()

[Volker Rümelin]

Replace the resampling loop in audio_pcm_sw_read() with the new
function audio_pcm_sw_resample_in(). Unlike the old resample
loop the new function will try to consume input frames even if
the output buffer is full. This is necessary when downsampling
to avoid reading less audio frames than calculated in advance.
The loop was unrolled to avoid complicated loop control conditions
in this case.

Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Acked-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
---
 audio/audio.c | 59 ++++++++++++++++++++++++++++++---------------------
 1 file changed, 35 insertions(+), 24 deletions(-)

diff --git a/audio/audio.c b/audio/audio.c
index e18b5e98c5..9e9c03a42e 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -543,11 +543,43 @@ static size_t audio_pcm_hw_conv_in(HWVoiceIn *hw, void *pcm_buf, size_t samples)
 /*
  * Soft voice (capture)
  */
+static void audio_pcm_sw_resample_in(SWVoiceIn *sw,
+    size_t frames_in_max, size_t frames_out_max,
+    size_t *total_in, size_t *total_out)
+{
+    HWVoiceIn *hw = sw->hw;
+    struct st_sample *src, *dst;
+    size_t live, rpos, frames_in, frames_out;
+
+    live = hw->total_samples_captured - sw->total_hw_samples_acquired;
+    rpos = audio_ring_posb(hw->conv_buf.pos, live, hw->conv_buf.size);
+
+    /* resample conv_buf from rpos to end of buffer */
+    src = hw->conv_buf.buffer + rpos;
+    frames_in = MIN(frames_in_max, hw->conv_buf.size - rpos);
+    dst = sw->resample_buf.buffer;
+    frames_out = frames_out_max;
+    st_rate_flow(sw->rate, src, dst, &frames_in, &frames_out);
+    rpos += frames_in;
+    *total_in = frames_in;
+    *total_out = frames_out;
+
+    /* resample conv_buf from start of buffer if there are input frames left */
+    if (frames_in_max - frames_in && rpos == hw->conv_buf.size) {
+        src = hw->conv_buf.buffer;
+        frames_in = frames_in_max - frames_in;
+        dst += frames_out;
+        frames_out = frames_out_max - frames_out;
+        st_rate_flow(sw->rate, src, dst, &frames_in, &frames_out);
+        *total_in += frames_in;
+        *total_out += frames_out;
+    }
+}
+
 static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
 {
     HWVoiceIn *hw = sw->hw;
-    size_t samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
-    struct st_sample *src, *dst = sw->resample_buf.buffer;
+    size_t samples, live, ret, swlim, total;
 
     live = hw->total_samples_captured - sw->total_hw_samples_acquired;
     if (!live) {
@@ -558,33 +590,12 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
         return 0;
     }
 
-    rpos = audio_ring_posb(hw->conv_buf.pos, live, hw->conv_buf.size);
-
     samples = size / sw->info.bytes_per_frame;
 
     swlim = (live * sw->ratio) >> 32;
     swlim = MIN (swlim, samples);
 
-    while (swlim) {
-        src = hw->conv_buf.buffer + rpos;
-        if (hw->conv_buf.pos > rpos) {
-            isamp = hw->conv_buf.pos - rpos;
-        } else {
-            isamp = hw->conv_buf.size - rpos;
-        }
-
-        if (!isamp) {
-            break;
-        }
-        osamp = swlim;
-
-        st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
-        swlim -= osamp;
-        rpos = (rpos + isamp) % hw->conv_buf.size;
-        dst += osamp;
-        ret += osamp;
-        total += isamp;
-    }
+    audio_pcm_sw_resample_in(sw, live, swlim, &total, &ret);
 
     if (!hw->pcm_ops->volume_in) {
         mixeng_volume(sw->resample_buf.buffer, ret, &sw->vol);
-- 
2.35.3

[PATCH v3 11/15] audio: rename variables in audio_pcm_sw_read()

[Volker Rümelin]

The audio_pcm_sw_read() function uses a few very unspecific
variable names. Rename them for better readability.

ret => total_out
total => total_in
size => buf_len
samples => frames_out_max

Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
---
 audio/audio.c | 18 +++++++++---------
 1 file changed, 9 insertions(+), 9 deletions(-)

diff --git a/audio/audio.c b/audio/audio.c
index 9e9c03a42e..22c36d6660 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -576,10 +576,10 @@ static void audio_pcm_sw_resample_in(SWVoiceIn *sw,
     }
 }
 
-static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
+static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t buf_len)
 {
     HWVoiceIn *hw = sw->hw;
-    size_t samples, live, ret, swlim, total;
+    size_t live, frames_out_max, swlim, total_in, total_out;
 
     live = hw->total_samples_captured - sw->total_hw_samples_acquired;
     if (!live) {
@@ -590,20 +590,20 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
         return 0;
     }
 
-    samples = size / sw->info.bytes_per_frame;
+    frames_out_max = buf_len / sw->info.bytes_per_frame;
 
     swlim = (live * sw->ratio) >> 32;
-    swlim = MIN (swlim, samples);
+    swlim = MIN(swlim, frames_out_max);
 
-    audio_pcm_sw_resample_in(sw, live, swlim, &total, &ret);
+    audio_pcm_sw_resample_in(sw, live, swlim, &total_in, &total_out);
 
     if (!hw->pcm_ops->volume_in) {
-        mixeng_volume(sw->resample_buf.buffer, ret, &sw->vol);
+        mixeng_volume(sw->resample_buf.buffer, total_out, &sw->vol);
     }
+    sw->clip(buf, sw->resample_buf.buffer, total_out);
 
-    sw->clip(buf, sw->resample_buf.buffer, ret);
-    sw->total_hw_samples_acquired += total;
-    return ret * sw->info.bytes_per_frame;
+    sw->total_hw_samples_acquired += total_in;
+    return total_out * sw->info.bytes_per_frame;
 }
 
 /*
-- 
2.35.3

[PATCH v3 12/15] audio: make recording packet length calculation exact

[Volker Rümelin]

Introduce the new function st_rate_frames_out() to calculate the
exact number of audio output frames the resampling code can
generate from a given number of audio input frames. When upsampling,
this function returns the maximum number of output frames.

This new function replaces the audio_frontend_frames_in()
function, which calculated the average number of output frames
rounded down to the nearest integer. The audio_frontend_frames_in()
function was additionally used to limit the number of output frames
to the resample buffer size. In audio_pcm_sw_read() the variable
resample_buf.size replaces the open coded audio_frontend_frames_in()
function. In audio_run_in() an additional MIN() function is
necessary.

After this patch the audio packet length calculation for audio
recording is exact.

Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
---
 audio/audio.c  | 29 ++++++++---------------------
 audio/mixeng.c | 41 +++++++++++++++++++++++++++++++++++++++++
 audio/mixeng.h |  1 +
 3 files changed, 50 insertions(+), 21 deletions(-)

diff --git a/audio/audio.c b/audio/audio.c
index 22c36d6660..dad17e59b8 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -579,7 +579,7 @@ static void audio_pcm_sw_resample_in(SWVoiceIn *sw,
 static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t buf_len)
 {
     HWVoiceIn *hw = sw->hw;
-    size_t live, frames_out_max, swlim, total_in, total_out;
+    size_t live, frames_out_max, total_in, total_out;
 
     live = hw->total_samples_captured - sw->total_hw_samples_acquired;
     if (!live) {
@@ -590,12 +590,10 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t buf_len)
         return 0;
     }
 
-    frames_out_max = buf_len / sw->info.bytes_per_frame;
+    frames_out_max = MIN(buf_len / sw->info.bytes_per_frame,
+                         sw->resample_buf.size);
 
-    swlim = (live * sw->ratio) >> 32;
-    swlim = MIN(swlim, frames_out_max);
-
-    audio_pcm_sw_resample_in(sw, live, swlim, &total_in, &total_out);
+    audio_pcm_sw_resample_in(sw, live, frames_out_max, &total_in, &total_out);
 
     if (!hw->pcm_ops->volume_in) {
         mixeng_volume(sw->resample_buf.buffer, total_out, &sw->vol);
@@ -979,18 +977,6 @@ void AUD_set_active_in (SWVoiceIn *sw, int on)
     }
 }
 
-/**
- * audio_frontend_frames_in() - returns the number of frames the resampling
- * code generates from frames_in frames
- *
- * @sw: audio recording frontend
- * @frames_in: number of frames
- */
-static size_t audio_frontend_frames_in(SWVoiceIn *sw, size_t frames_in)
-{
-    return (int64_t)frames_in * sw->ratio >> 32;
-}
-
 static size_t audio_get_avail (SWVoiceIn *sw)
 {
     size_t live;
@@ -1007,9 +993,9 @@ static size_t audio_get_avail (SWVoiceIn *sw)
     }
 
     ldebug (
-        "%s: get_avail live %zu frontend frames %zu\n",
+        "%s: get_avail live %zu frontend frames %u\n",
         SW_NAME (sw),
-        live, audio_frontend_frames_in(sw, live)
+        live, st_rate_frames_out(sw->rate, live)
         );
 
     return live;
@@ -1314,8 +1300,9 @@ static void audio_run_in (AudioState *s)
                 size_t sw_avail = audio_get_avail(sw);
                 size_t avail;
 
-                avail = audio_frontend_frames_in(sw, sw_avail);
+                avail = st_rate_frames_out(sw->rate, sw_avail);
                 if (avail > 0) {
+                    avail = MIN(avail, sw->resample_buf.size);
                     sw->callback.fn(sw->callback.opaque,
                                     avail * sw->info.bytes_per_frame);
                 }
diff --git a/audio/mixeng.c b/audio/mixeng.c
index a24c8c45a7..69f6549224 100644
--- a/audio/mixeng.c
+++ b/audio/mixeng.c
@@ -440,6 +440,47 @@ void st_rate_stop (void *opaque)
     g_free (opaque);
 }
 
+/**
+ * st_rate_frames_out() - returns the number of frames the resampling code
+ * generates from frames_in frames
+ *
+ * @opaque: pointer to struct rate
+ * @frames_in: number of frames
+ *
+ * When upsampling, there may be more than one correct result. In this case,
+ * the function returns the maximum number of output frames the resampling
+ * code can generate.
+ */
+uint32_t st_rate_frames_out(void *opaque, uint32_t frames_in)
+{
+    struct rate *rate = opaque;
+    uint64_t opos_end, opos_delta;
+    uint32_t ipos_end;
+    uint32_t frames_out;
+
+    if (rate->opos_inc == 1ULL << 32) {
+        return frames_in;
+    }
+
+    /* no output frame without at least one input frame */
+    if (!frames_in) {
+        return 0;
+    }
+
+    /* last frame read was at rate->ipos - 1 */
+    ipos_end = rate->ipos - 1 + frames_in;
+    opos_end = (uint64_t)ipos_end << 32;
+
+    /* last frame written was at rate->opos - rate->opos_inc */
+    if (opos_end + rate->opos_inc <= rate->opos) {
+        return 0;
+    }
+    opos_delta = opos_end - rate->opos + rate->opos_inc;
+    frames_out = opos_delta / rate->opos_inc;
+
+    return opos_delta % rate->opos_inc ? frames_out : frames_out - 1;
+}
+
 /**
  * st_rate_frames_in() - returns the number of frames needed to
  * get frames_out frames after resampling
diff --git a/audio/mixeng.h b/audio/mixeng.h
index 64c1e231cc..f9de7cffeb 100644
--- a/audio/mixeng.h
+++ b/audio/mixeng.h
@@ -52,6 +52,7 @@ void st_rate_flow(void *opaque, st_sample *ibuf, st_sample *obuf,
 void st_rate_flow_mix(void *opaque, st_sample *ibuf, st_sample *obuf,
                       size_t *isamp, size_t *osamp);
 void st_rate_stop (void *opaque);
+uint32_t st_rate_frames_out(void *opaque, uint32_t frames_in);
 uint32_t st_rate_frames_in(void *opaque, uint32_t frames_out);
 void mixeng_clear (struct st_sample *buf, int len);
 void mixeng_volume (struct st_sample *buf, int len, struct mixeng_volume *vol);
-- 
2.35.3

[PATCH v3 13/15] audio: handle leftover audio frame from upsampling

[Volker Rümelin]

Upsampling may leave one remaining audio frame in the input
buffer. The emulated audio playback devices are currently
resposible to write this audio frame again in the next write
cycle. Push that task down to audio_pcm_sw_write.

This is another step towards an audio callback interface that
guarantees that when audio frontends are told they can write
n audio frames, they can actually do so.

Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Acked-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
---
 audio/audio.c          | 34 ++++++++++++++++++++++++++++------
 audio/audio_template.h |  6 ++++++
 2 files changed, 34 insertions(+), 6 deletions(-)

diff --git a/audio/audio.c b/audio/audio.c
index dad17e59b8..4836ab8ca8 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -731,16 +731,21 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t buf_len)
     hw_free = hw_free > live ? hw_free - live : 0;
     frames_out_max = MIN(dead, hw_free);
     sw_max = st_rate_frames_in(sw->rate, frames_out_max);
-    fe_max = MIN(buf_len / sw->info.bytes_per_frame, sw->resample_buf.size);
+    fe_max = MIN(buf_len / sw->info.bytes_per_frame + sw->resample_buf.pos,
+                 sw->resample_buf.size);
     frames_in_max = MIN(sw_max, fe_max);
 
     if (!frames_in_max) {
         return 0;
     }
 
-    sw->conv(sw->resample_buf.buffer, buf, frames_in_max);
-    if (!sw->hw->pcm_ops->volume_out) {
-        mixeng_volume(sw->resample_buf.buffer, frames_in_max, &sw->vol);
+    if (frames_in_max > sw->resample_buf.pos) {
+        sw->conv(sw->resample_buf.buffer + sw->resample_buf.pos,
+                 buf, frames_in_max - sw->resample_buf.pos);
+        if (!sw->hw->pcm_ops->volume_out) {
+            mixeng_volume(sw->resample_buf.buffer + sw->resample_buf.pos,
+                          frames_in_max - sw->resample_buf.pos, &sw->vol);
+        }
     }
 
     audio_pcm_sw_resample_out(sw, frames_in_max, frames_out_max,
@@ -749,6 +754,22 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t buf_len)
     sw->total_hw_samples_mixed += total_out;
     sw->empty = sw->total_hw_samples_mixed == 0;
 
+    /*
+     * Upsampling may leave one audio frame in the resample buffer. Decrement
+     * total_in by one if there was a leftover frame from the previous resample
+     * pass in the resample buffer. Increment total_in by one if the current
+     * resample pass left one frame in the resample buffer.
+     */
+    if (frames_in_max - total_in == 1) {
+        /* copy one leftover audio frame to the beginning of the buffer */
+        *sw->resample_buf.buffer = *(sw->resample_buf.buffer + total_in);
+        total_in += 1 - sw->resample_buf.pos;
+        sw->resample_buf.pos = 1;
+    } else if (total_in >= sw->resample_buf.pos) {
+        total_in -= sw->resample_buf.pos;
+        sw->resample_buf.pos = 0;
+    }
+
 #ifdef DEBUG_OUT
     dolog (
         "%s: write size %zu written %zu total mixed %zu\n",
@@ -1155,8 +1176,9 @@ static void audio_run_out (AudioState *s)
                 } else {
                     free = 0;
                 }
-                if (free > 0) {
-                    free = MIN(free, sw->resample_buf.size);
+                if (free > sw->resample_buf.pos) {
+                    free = MIN(free, sw->resample_buf.size)
+                           - sw->resample_buf.pos;
                     sw->callback.fn(sw->callback.opaque,
                                     free * sw->info.bytes_per_frame);
                 }
diff --git a/audio/audio_template.h b/audio/audio_template.h
index a0b653f52c..0d8aab6fad 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -138,6 +138,12 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw)
         return -1;
     }
 
+    /*
+     * Allocate one additional audio frame that is needed for upsampling
+     * if the resample buffer size is small. For large buffer sizes take
+     * care of overflows.
+     */
+    samples = samples < INT_MAX ? samples + 1 : INT_MAX;
     sw->resample_buf.buffer = g_new0(st_sample, samples);
     sw->resample_buf.size = samples;
     sw->resample_buf.pos = 0;
-- 
2.35.3

[PATCH v3 14/15] audio/audio_template: substitute sw->hw with hw

[Volker Rümelin]

Substitute sw->hw with hw in the audio_pcm_sw_alloc_resources_*
functions.

Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
---
 audio/audio_template.h | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

diff --git a/audio/audio_template.h b/audio/audio_template.h
index 0d8aab6fad..7e116426c7 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -107,6 +107,7 @@ static void glue (audio_pcm_sw_free_resources_, TYPE) (SW *sw)
 
 static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw)
 {
+    HW *hw = sw->hw;
     int samples;
 
     if (!glue(audio_get_pdo_, TYPE)(sw->s->dev)->mixing_engine) {
@@ -125,7 +126,6 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw)
     }
 
     if (samples == 0) {
-        HW *hw = sw->hw;
         size_t f_fe_min;
 
         /* f_fe_min = ceil(1 [frames] * f_be [Hz] / size_be [frames]) */
@@ -149,9 +149,9 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw)
     sw->resample_buf.pos = 0;
 
 #ifdef DAC
-    sw->rate = st_rate_start (sw->info.freq, sw->hw->info.freq);
+    sw->rate = st_rate_start(sw->info.freq, hw->info.freq);
 #else
-    sw->rate = st_rate_start (sw->hw->info.freq, sw->info.freq);
+    sw->rate = st_rate_start(hw->info.freq, sw->info.freq);
 #endif
 
     return 0;
-- 
2.35.3

[PATCH v3 15/15] audio: remove sw->ratio

[Volker Rümelin]

Simplify the resample buffer size calculation.

For audio playback we have
sw->ratio = ((int64_t)sw->hw->info.freq << 32) / sw->info.freq;
samples = ((int64_t)sw->HWBUF.size << 32) / sw->ratio;

This can be simplified to
samples = muldiv64(sw->HWBUF.size, sw->info.freq, sw->hw->info.freq);

For audio recording we have
sw->ratio = ((int64_t)sw->info.freq << 32) / sw->hw->info.freq;
samples = (int64_t)sw->HWBUF.size * sw->ratio >> 32;

This can be simplified to
samples = muldiv64(sw->HWBUF.size, sw->info.freq, sw->hw->info.freq);

With hw = sw->hw this becomes in both cases
samples = muldiv64(HWBUF.size, sw->info.freq, hw->info.freq);

Now that sw->ratio is no longer needed, remove sw->ratio.

Acked-by: Mark Cave-Ayland <mark.cave-ayland@ilande.co.uk>
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
---
 audio/audio.c          |  1 -
 audio/audio_int.h      |  2 --
 audio/audio_template.h | 30 +++++++++---------------------
 3 files changed, 9 insertions(+), 24 deletions(-)

diff --git a/audio/audio.c b/audio/audio.c
index 4836ab8ca8..70b096713c 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -478,7 +478,6 @@ static int audio_attach_capture (HWVoiceOut *hw)
         sw->info = hw->info;
         sw->empty = 1;
         sw->active = hw->enabled;
-        sw->ratio = ((int64_t) hw_cap->info.freq << 32) / sw->info.freq;
         sw->vol = nominal_volume;
         sw->rate = st_rate_start (sw->info.freq, hw_cap->info.freq);
         QLIST_INSERT_HEAD (&hw_cap->sw_head, sw, entries);
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 8b163e1759..d51d63f08d 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -108,7 +108,6 @@ struct SWVoiceOut {
     AudioState *s;
     struct audio_pcm_info info;
     t_sample *conv;
-    int64_t ratio;
     STSampleBuffer resample_buf;
     void *rate;
     size_t total_hw_samples_mixed;
@@ -126,7 +125,6 @@ struct SWVoiceIn {
     AudioState *s;
     int active;
     struct audio_pcm_info info;
-    int64_t ratio;
     void *rate;
     size_t total_hw_samples_acquired;
     STSampleBuffer resample_buf;
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 7e116426c7..e42326c20d 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -108,32 +108,23 @@ static void glue (audio_pcm_sw_free_resources_, TYPE) (SW *sw)
 static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw)
 {
     HW *hw = sw->hw;
-    int samples;
+    uint64_t samples;
 
     if (!glue(audio_get_pdo_, TYPE)(sw->s->dev)->mixing_engine) {
         return 0;
     }
 
-#ifdef DAC
-    samples = ((int64_t)sw->HWBUF.size << 32) / sw->ratio;
-#else
-    samples = (int64_t)sw->HWBUF.size * sw->ratio >> 32;
-#endif
-    if (audio_bug(__func__, samples < 0)) {
-        dolog("Can not allocate buffer for `%s' (%d samples)\n",
-              SW_NAME(sw), samples);
-        return -1;
-    }
-
+    samples = muldiv64(HWBUF.size, sw->info.freq, hw->info.freq);
     if (samples == 0) {
-        size_t f_fe_min;
+        uint64_t f_fe_min;
+        uint64_t f_be = (uint32_t)hw->info.freq;
 
         /* f_fe_min = ceil(1 [frames] * f_be [Hz] / size_be [frames]) */
-        f_fe_min = (hw->info.freq + HWBUF.size - 1) / HWBUF.size;
+        f_fe_min = (f_be + HWBUF.size - 1) / HWBUF.size;
         qemu_log_mask(LOG_UNIMP,
                       AUDIO_CAP ": The guest selected a " NAME " sample rate"
-                      " of %d Hz for %s. Only sample rates >= %zu Hz are"
-                      " supported.\n",
+                      " of %d Hz for %s. Only sample rates >= %" PRIu64 " Hz"
+                      " are supported.\n",
                       sw->info.freq, sw->name, f_fe_min);
         return -1;
     }
@@ -141,9 +132,9 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw)
     /*
      * Allocate one additional audio frame that is needed for upsampling
      * if the resample buffer size is small. For large buffer sizes take
-     * care of overflows.
+     * care of overflows and truncation.
      */
-    samples = samples < INT_MAX ? samples + 1 : INT_MAX;
+    samples = samples < SIZE_MAX ? samples + 1 : SIZE_MAX;
     sw->resample_buf.buffer = g_new0(st_sample, samples);
     sw->resample_buf.size = samples;
     sw->resample_buf.pos = 0;
@@ -170,11 +161,8 @@ static int glue (audio_pcm_sw_init_, TYPE) (
     sw->hw = hw;
     sw->active = 0;
 #ifdef DAC
-    sw->ratio = ((int64_t) sw->hw->info.freq << 32) / sw->info.freq;
     sw->total_hw_samples_mixed = 0;
     sw->empty = 1;
-#else
-    sw->ratio = ((int64_t) sw->info.freq << 32) / sw->hw->info.freq;
 #endif
 
     if (sw->info.is_float) {
-- 
2.35.3