Merge tag 'kraxel-20221013-pull-request' of https://gitlab.com/kraxel/qemu into staging

pci: cleanup virtio ids.
audio: bugfixes and latency improvements.
misc fixes for hw/display and ui

# -----BEGIN PGP SIGNATURE-----
#
# iQIzBAABCgAdFiEEoDKM/7k6F6eZAf59TLbY7tPocTgFAmNHtYsACgkQTLbY7tPo
# cTjHhg//RDkHbqVSExe+Odw5ISuLu/EXZSHAVjo3KOCUvaj7O2cXi8N7DVfEy5a5
# T3+WSv0v4X6TYSV0PoMb36a11rCuOKzeLZrtEOQeYfG3D1WCVc9gIWMt6omzBC7A
# YQ59P+u19qHD7xD2PP3WRtdcqmsceg1RG+47adX2EnsRZmmu/yJxD72w/Q1kXMuB
# jIzuJU2ZVorYX9y11hnIU3M5pvoX/vjFA+Ib2UGZZdlE3KlUKtJeAtLiZkHfoyd1
# 5janU+PtSU6Z1yVirE7RVz3+IBbfqqEFTkDtMXJucJW/Eod0NHCyo4Q6D64HoiZe
# +JZKkHmuvn8ZUgXMtIOZdH+aOHlaIJzA5SoA2IFxCBVuxn7p4NtPbCRoHHg7gkDh
# BDsq+p/wsdOY06u1txFw9dYy+4tKvWS7+Dxhyme7GT2YUQHrEEG3pzGFmk3PE0Vi
# tEAhmfNRxWzUgIcynQiN/3SnShAI8lANq0SEiiTvqcX7h1TK+cjEYjOTMsjK43nL
# 2W/pgQxJpEPcSs3jgFLnBLk9rUHRNRC+GtMBlwN+Wdc1y17leZHiIinqhHjXuts3
# cJTdv4veeGuJENPIl2rk5JOdvpVtzduDkz+Rzx0mGb+LnAYdK2lBUV5LY9FfdwaK
# 2Bgg02ZYNBz7K2zzFeeV+7b7K/LYOuWkGdzGvKbpqjbefopZmTM=
# =6d/F
# -----END PGP SIGNATURE-----
# gpg: Signature made Thu 13 Oct 2022 02:51:55 EDT
# gpg:                using RSA key A0328CFFB93A17A79901FE7D4CB6D8EED3E87138
# gpg: Good signature from "Gerd Hoffmann (work) <kraxel@redhat.com>" [full]
# gpg:                 aka "Gerd Hoffmann <gerd@kraxel.org>" [full]
# gpg:                 aka "Gerd Hoffmann (private) <kraxel@gmail.com>" [full]
# Primary key fingerprint: A032 8CFF B93A 17A7 9901  FE7D 4CB6 D8EE D3E8 7138

* tag 'kraxel-20221013-pull-request' of https://gitlab.com/kraxel/qemu: (26 commits)
  audio: improve out.voices test
  audio: fix in.voices test
  gtk: Add show_menubar=on|off command line option.
  qemu-edid: Restrict input parameter -d to avoid division by zero
  ui/gtk: Fix the implicit mouse ungrabbing logic
  pci-ids: document modern virtio-pci ids in pci.h too
  pci-ids: drop list of modern virtio devices
  pci-ids: drop PCI_DEVICE_ID_VIRTIO_PMEM
  pci-ids: drop PCI_DEVICE_ID_VIRTIO_MEM
  pci-ids: drop PCI_DEVICE_ID_VIRTIO_IOMMU
  docs: add firmware feature flags
  cirrus_vga: fix potential memory overflow
  ui/gtk-egl: egl context needs to be unbound in the end of gd_egl_switch
  ui/vnc-clipboard: fix integer underflow in vnc_client_cut_text_ext
  audio: prevent an integer overflow in resampling code
  audio: fix sw->buf size for audio recording
  audio: refactor audio_get_avail()
  audio: rename audio_sw_bytes_free()
  audio: swap audio_rate_get_bytes() function parameters
  spiceaudio: update comment
  ...

Signed-off-by: Stefan Hajnoczi <stefanha@redhat.com>

9 files changed, 143 insertions, 54 deletions

diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 4a61378cd7..7a2a94cd42 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -602,6 +602,42 @@ static int alsa_open(bool in, struct alsa_params_req *req,
     return -1;
 }
 
+static size_t alsa_buffer_get_free(HWVoiceOut *hw)
+{
+    ALSAVoiceOut *alsa = (ALSAVoiceOut *)hw;
+    snd_pcm_sframes_t avail;
+    size_t alsa_free, generic_free, generic_in_use;
+
+    avail = snd_pcm_avail_update(alsa->handle);
+    if (avail < 0) {
+        if (avail == -EPIPE) {
+            if (!alsa_recover(alsa->handle)) {
+                avail = snd_pcm_avail_update(alsa->handle);
+            }
+        }
+        if (avail < 0) {
+            alsa_logerr(avail,
+                        "Could not obtain number of available frames\n");
+            avail = 0;
+        }
+    }
+
+    alsa_free = avail * hw->info.bytes_per_frame;
+    generic_free = audio_generic_buffer_get_free(hw);
+    generic_in_use = hw->samples * hw->info.bytes_per_frame - generic_free;
+    if (generic_in_use) {
+        /*
+         * This code can only be reached in the unlikely case that
+         * snd_pcm_avail_update() returned a larger number of frames
+         * than snd_pcm_writei() could write. Make sure that all
+         * remaining bytes in the generic buffer can be written.
+         */
+        alsa_free = alsa_free > generic_in_use ? alsa_free - generic_in_use : 0;
+    }
+
+    return alsa_free;
+}
+
 static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
 {
     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
@@ -916,7 +952,7 @@ 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,
+    .buffer_get_free = alsa_buffer_get_free,
     .run_buffer_out = audio_generic_run_buffer_out,
     .enable_out = alsa_enable_out,
 
diff --git a/audio/audio.c b/audio/audio.c
index df6818ed55..cc664271eb 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -986,6 +986,18 @@ 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;
@@ -1002,17 +1014,24 @@ static size_t audio_get_avail (SWVoiceIn *sw)
     }
 
     ldebug (
-        "%s: get_avail live %zu ret %" PRId64 "\n",
+        "%s: get_avail live %zu frontend frames %zu\n",
         SW_NAME (sw),
-        live, (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame
+        live, audio_frontend_frames_in(sw, live)
         );
 
-    return (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame;
+    return live;
 }
 
-static size_t audio_sw_bytes_free(SWVoiceOut *sw, size_t free)
+/**
+ * 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)free << 32) / sw->ratio) * sw->info.bytes_per_frame;
+    return ((int64_t)frames_out << 32) / sw->ratio;
 }
 
 static size_t audio_get_free(SWVoiceOut *sw)
@@ -1034,8 +1053,8 @@ 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 sw_bytes %zu\n",
-          SW_NAME(sw), live, dead, audio_sw_bytes_free(sw, dead));
+    dolog("%s: get_free live %zu dead %zu frontend frames %zu\n",
+          SW_NAME(sw), live, dead, audio_frontend_frames_out(sw, dead));
 #endif
 
     return dead;
@@ -1121,8 +1140,12 @@ static void audio_run_out (AudioState *s)
     HWVoiceOut *hw = NULL;
     SWVoiceOut *sw;
 
-    if (!audio_get_pdo_out(s->dev)->mixing_engine) {
-        while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
+    while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
+        size_t played, live, prev_rpos;
+        size_t hw_free = audio_pcm_hw_get_free(hw);
+        int nb_live;
+
+        if (!audio_get_pdo_out(s->dev)->mixing_engine) {
             /* there is exactly 1 sw for each hw with no mixeng */
             sw = hw->sw_head.lh_first;
 
@@ -1135,16 +1158,16 @@ static void audio_run_out (AudioState *s)
             }
 
             if (sw->active) {
-                sw->callback.fn(sw->callback.opaque, INT_MAX);
+                sw->callback.fn(sw->callback.opaque,
+                                hw_free * sw->info.bytes_per_frame);
             }
-        }
-        return;
-    }
 
-    while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
-        size_t played, live, prev_rpos;
-        size_t hw_free = audio_pcm_hw_get_free(hw);
-        int nb_live;
+            if (hw->pcm_ops->run_buffer_out) {
+                hw->pcm_ops->run_buffer_out(hw);
+            }
+
+            continue;
+        }
 
         for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
             if (sw->active) {
@@ -1152,13 +1175,14 @@ static void audio_run_out (AudioState *s)
                 size_t free;
 
                 if (hw_free > sw->total_hw_samples_mixed) {
-                    free = audio_sw_bytes_free(sw,
+                    free = audio_frontend_frames_out(sw,
                         MIN(sw_free, hw_free - sw->total_hw_samples_mixed));
                 } else {
                     free = 0;
                 }
                 if (free > 0) {
-                    sw->callback.fn(sw->callback.opaque, free);
+                    sw->callback.fn(sw->callback.opaque,
+                                    free * sw->info.bytes_per_frame);
                 }
             }
         }
@@ -1297,11 +1321,13 @@ static void audio_run_in (AudioState *s)
             sw->total_hw_samples_acquired -= min;
 
             if (sw->active) {
+                size_t sw_avail = audio_get_avail(sw);
                 size_t avail;
 
-                avail = audio_get_avail (sw);
+                avail = audio_frontend_frames_in(sw, sw_avail);
                 if (avail > 0) {
-                    sw->callback.fn (sw->callback.opaque, avail);
+                    sw->callback.fn(sw->callback.opaque,
+                                    avail * sw->info.bytes_per_frame);
                 }
             }
         }
@@ -1501,10 +1527,6 @@ size_t audio_generic_write(HWVoiceOut *hw, void *buf, size_t size)
         }
     }
 
-    if (hw->pcm_ops->run_buffer_out) {
-        hw->pcm_ops->run_buffer_out(hw);
-    }
-
     return total;
 }
 
@@ -1750,13 +1772,13 @@ static AudioState *audio_init(Audiodev *dev, const char *name)
     s->nb_hw_voices_out = audio_get_pdo_out(dev)->voices;
     s->nb_hw_voices_in = audio_get_pdo_in(dev)->voices;
 
-    if (s->nb_hw_voices_out <= 0) {
+    if (s->nb_hw_voices_out < 1) {
         dolog ("Bogus number of playback voices %d, setting to 1\n",
                s->nb_hw_voices_out);
         s->nb_hw_voices_out = 1;
     }
 
-    if (s->nb_hw_voices_in <= 0) {
+    if (s->nb_hw_voices_in < 0) {
         dolog ("Bogus number of capture voices %d, setting to 0\n",
                s->nb_hw_voices_in);
         s->nb_hw_voices_in = 0;
@@ -2251,26 +2273,39 @@ void audio_rate_start(RateCtl *rate)
     rate->start_ticks = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
 }
 
-size_t audio_rate_get_bytes(struct audio_pcm_info *info, RateCtl *rate,
-                            size_t bytes_avail)
+size_t audio_rate_peek_bytes(RateCtl *rate, struct audio_pcm_info *info)
 {
     int64_t now;
     int64_t ticks;
     int64_t bytes;
-    int64_t samples;
-    size_t ret;
+    int64_t frames;
 
     now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
     ticks = now - rate->start_ticks;
     bytes = muldiv64(ticks, info->bytes_per_second, NANOSECONDS_PER_SECOND);
-    samples = (bytes - rate->bytes_sent) / info->bytes_per_frame;
-    if (samples < 0 || samples > 65536) {
-        AUD_log(NULL, "Resetting rate control (%" PRId64 " samples)\n", samples);
+    frames = (bytes - rate->bytes_sent) / info->bytes_per_frame;
+    if (frames < 0 || frames > 65536) {
+        AUD_log(NULL, "Resetting rate control (%" PRId64 " frames)\n", frames);
         audio_rate_start(rate);
-        samples = 0;
+        frames = 0;
     }
 
-    ret = MIN(samples * info->bytes_per_frame, bytes_avail);
-    rate->bytes_sent += ret;
-    return ret;
+    return frames * info->bytes_per_frame;
+}
+
+void audio_rate_add_bytes(RateCtl *rate, size_t bytes_used)
+{
+    rate->bytes_sent += bytes_used;
+}
+
+size_t audio_rate_get_bytes(RateCtl *rate, struct audio_pcm_info *info,
+                            size_t bytes_avail)
+{
+    size_t bytes;
+
+    bytes = audio_rate_peek_bytes(rate, info);
+    bytes = MIN(bytes, bytes_avail);
+    audio_rate_add_bytes(rate, bytes);
+
+    return bytes;
 }
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 2a6914d2aa..e87ce014a0 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -263,7 +263,9 @@ typedef struct RateCtl {
 } RateCtl;
 
 void audio_rate_start(RateCtl *rate);
-size_t audio_rate_get_bytes(struct audio_pcm_info *info, RateCtl *rate,
+size_t audio_rate_peek_bytes(RateCtl *rate, struct audio_pcm_info *info);
+void audio_rate_add_bytes(RateCtl *rate, size_t bytes_used);
+size_t audio_rate_get_bytes(RateCtl *rate, struct audio_pcm_info *info,
                             size_t bytes_avail);
 
 static inline size_t audio_ring_dist(size_t dst, size_t src, size_t len)
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 98ab557684..720a32e57e 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -110,7 +110,11 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw)
         return 0;
     }
 
+#ifdef DAC
     samples = ((int64_t) sw->HWBUF->size << 32) / sw->ratio;
+#else
+    samples = (int64_t)sw->HWBUF->size * sw->ratio >> 32;
+#endif
 
     sw->buf = audio_calloc(__func__, samples, sizeof(struct st_sample));
     if (!sw->buf) {
diff --git a/audio/dbusaudio.c b/audio/dbusaudio.c
index a3d656d3b0..722df0355e 100644
--- a/audio/dbusaudio.c
+++ b/audio/dbusaudio.c
@@ -82,7 +82,7 @@ static void *dbus_get_buffer_out(HWVoiceOut *hw, size_t *size)
     }
 
     *size = MIN(vo->buf_size - vo->buf_pos, *size);
-    *size = audio_rate_get_bytes(&hw->info, &vo->rate, *size);
+    *size = audio_rate_get_bytes(&vo->rate, &hw->info, *size);
 
     return vo->buf + vo->buf_pos;
 
@@ -343,7 +343,7 @@ dbus_read(HWVoiceIn *hw, void *buf, size_t size)
 
     trace_dbus_audio_read(size);
 
-    /* size = audio_rate_get_bytes(&hw->info, &vo->rate, size); */
+    /* size = audio_rate_get_bytes(&vo->rate, &hw->info, size); */
 
     g_hash_table_iter_init(&iter, da->in_listeners);
     while (g_hash_table_iter_next(&iter, NULL, (void **)&listener)) {
diff --git a/audio/noaudio.c b/audio/noaudio.c
index 84a6bfbb1c..4fdee5adec 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -44,7 +44,7 @@ typedef struct NoVoiceIn {
 static size_t no_write(HWVoiceOut *hw, void *buf, size_t len)
 {
     NoVoiceOut *no = (NoVoiceOut *) hw;
-    return audio_rate_get_bytes(&hw->info, &no->rate, len);
+    return audio_rate_get_bytes(&no->rate, &hw->info, len);
 }
 
 static int no_init_out(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque)
@@ -89,7 +89,7 @@ static void no_fini_in (HWVoiceIn *hw)
 static size_t no_read(HWVoiceIn *hw, void *buf, size_t size)
 {
     NoVoiceIn *no = (NoVoiceIn *) hw;
-    int64_t bytes = audio_rate_get_bytes(&hw->info, &no->rate, size);
+    int64_t bytes = audio_rate_get_bytes(&no->rate, &hw->info, size);
 
     audio_pcm_info_clear_buf(&hw->info, buf, bytes / hw->info.bytes_per_frame);
     return bytes;
diff --git a/audio/rate_template.h b/audio/rate_template.h
index f94c940c61..b432719ebb 100644
--- a/audio/rate_template.h
+++ b/audio/rate_template.h
@@ -72,11 +72,6 @@ void NAME (void *opaque, struct st_sample *ibuf, struct st_sample *obuf,
             ilast = *ibuf++;
             rate->ipos++;
 
-            /* if ipos overflow, there is  a infinite loop */
-            if (rate->ipos == 0xffffffff) {
-                rate->ipos = 1;
-                rate->opos = rate->opos & 0xffffffff;
-            }
             /* See if we finished the input buffer yet */
             if (ibuf >= iend) {
                 goto the_end;
@@ -85,6 +80,12 @@ void NAME (void *opaque, struct st_sample *ibuf, struct st_sample *obuf,
 
         icur = *ibuf;
 
+        /* wrap ipos and opos around long before they overflow */
+        if (rate->ipos >= 0x10001) {
+            rate->ipos = 1;
+            rate->opos &= 0xffffffff;
+        }
+
         /* interpolate */
 #ifdef FLOAT_MIXENG
 #ifdef RECIPROCAL
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index a8d370fe6f..d17ef1a25e 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -120,6 +120,13 @@ static void line_out_fini (HWVoiceOut *hw)
     spice_server_remove_interface (&out->sin.base);
 }
 
+static size_t line_out_get_free(HWVoiceOut *hw)
+{
+    SpiceVoiceOut *out = container_of(hw, SpiceVoiceOut, hw);
+
+    return audio_rate_peek_bytes(&out->rate, &hw->info);
+}
+
 static void *line_out_get_buffer(HWVoiceOut *hw, size_t *size)
 {
     SpiceVoiceOut *out = container_of(hw, SpiceVoiceOut, hw);
@@ -133,8 +140,6 @@ static void *line_out_get_buffer(HWVoiceOut *hw, size_t *size)
         *size = MIN((out->fsize - out->fpos) << 2, *size);
     }
 
-    *size = audio_rate_get_bytes(&hw->info, &out->rate, *size);
-
     return out->frame + out->fpos;
 }
 
@@ -142,6 +147,8 @@ static size_t line_out_put_buffer(HWVoiceOut *hw, void *buf, size_t size)
 {
     SpiceVoiceOut *out = container_of(hw, SpiceVoiceOut, hw);
 
+    audio_rate_add_bytes(&out->rate, size);
+
     if (buf) {
         assert(buf == out->frame + out->fpos && out->fpos <= out->fsize);
         out->fpos += size >> 2;
@@ -232,10 +239,13 @@ static void line_in_fini (HWVoiceIn *hw)
 static size_t line_in_read(HWVoiceIn *hw, void *buf, size_t len)
 {
     SpiceVoiceIn *in = container_of (hw, SpiceVoiceIn, hw);
-    uint64_t to_read = audio_rate_get_bytes(&hw->info, &in->rate, len) >> 2;
+    uint64_t to_read = audio_rate_get_bytes(&in->rate, &hw->info, len) >> 2;
     size_t ready = spice_server_record_get_samples(&in->sin, buf, to_read);
 
-    /* XXX: do we need this? */
+    /*
+     * If the client didn't send new frames, it most likely disconnected.
+     * Generate silence in this case to avoid a stalled audio stream.
+     */
     if (ready == 0) {
         memset(buf, 0, to_read << 2);
         ready = to_read;
@@ -282,6 +292,7 @@ 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_get_free,
     .get_buffer_out = line_out_get_buffer,
     .put_buffer_out = line_out_put_buffer,
     .enable_out = line_out_enable,
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index ac666335c7..3e1d84db83 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -42,7 +42,7 @@ typedef struct WAVVoiceOut {
 static size_t wav_write_out(HWVoiceOut *hw, void *buf, size_t len)
 {
     WAVVoiceOut *wav = (WAVVoiceOut *) hw;
-    int64_t bytes = audio_rate_get_bytes(&hw->info, &wav->rate, len);
+    int64_t bytes = audio_rate_get_bytes(&wav->rate, &hw->info, len);
     assert(bytes % hw->info.bytes_per_frame == 0);
 
     if (bytes && fwrite(buf, bytes, 1, wav->f) != 1) {