Merge tag 'audio-pull-request' of https://gitlab.com/marcandre.lureau/qemu into staging

Audio patches

- add float sample endianness converters
- various fixes

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# gpg: Signature made Sun 25 May 2025 09:26:09 EDT
# gpg:                using RSA key 87A9BD933F87C606D276F62DDAE8E10975969CE5
# gpg:                issuer "marcandre.lureau@redhat.com"
# gpg: Good signature from "Marc-André Lureau <marcandre.lureau@redhat.com>" [full]
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* tag 'audio-pull-request' of https://gitlab.com/marcandre.lureau/qemu:
  audio: Reset rate control when adding bytes
  alsaaudio: Set try-poll to false by default
  audio: add float sample endianness converters
  audio/mixeng: remove unnecessary pointer type casts
  hw/audio/asc: replace g_malloc0() with g_malloc()
  hw/audio/asc: fix SIGSEGV in asc_realize()
  audio: fix size calculation in AUD_get_buffer_size_out()
  audio: fix SIGSEGV in AUD_get_buffer_size_out()
  tests/functional: use 'none' audio driver for q800 tests

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

11 files changed, 119 insertions, 31 deletions

diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index cacae1ea59..9b6c01c0ef 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -899,7 +899,7 @@ static void alsa_enable_in(HWVoiceIn *hw, bool enable)
 static void alsa_init_per_direction(AudiodevAlsaPerDirectionOptions *apdo)
 {
     if (!apdo->has_try_poll) {
-        apdo->try_poll = true;
+        apdo->try_poll = false;
         apdo->has_try_poll = true;
     }
 }
diff --git a/audio/audio.c b/audio/audio.c
index 41ee11aaad..89f091bc88 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -905,6 +905,14 @@ size_t AUD_read(SWVoiceIn *sw, void *buf, size_t size)
 
 int AUD_get_buffer_size_out(SWVoiceOut *sw)
 {
+    if (!sw) {
+        return 0;
+    }
+
+    if (audio_get_pdo_out(sw->s->dev)->mixing_engine) {
+        return sw->resample_buf.size * sw->info.bytes_per_frame;
+    }
+
     return sw->hw->samples * sw->hw->info.bytes_per_frame;
 }
 
@@ -1884,7 +1892,8 @@ CaptureVoiceOut *AUD_add_capture(
         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];
+            hw->clip = mixeng_clip_float[hw->info.nchannels == 2]
+                [hw->info.swap_endianness];
         } else {
             hw->clip = mixeng_clip
                 [hw->info.nchannels == 2]
@@ -2274,17 +2283,19 @@ size_t audio_rate_peek_bytes(RateCtl *rate, struct audio_pcm_info *info)
     ticks = now - rate->start_ticks;
     bytes = muldiv64(ticks, info->bytes_per_second, NANOSECONDS_PER_SECOND);
     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);
-        frames = 0;
-    }
+    rate->peeked_frames = frames;
 
-    return frames * info->bytes_per_frame;
+    return frames < 0 ? 0 : frames * info->bytes_per_frame;
 }
 
 void audio_rate_add_bytes(RateCtl *rate, size_t bytes_used)
 {
+    if (rate->peeked_frames < 0 || rate->peeked_frames > 65536) {
+        AUD_log(NULL, "Resetting rate control (%" PRId64 " frames)\n",
+                rate->peeked_frames);
+        audio_rate_start(rate);
+    }
+
     rate->bytes_sent += bytes_used;
 }
 
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 2d079d00a2..f78ca05f92 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -255,6 +255,7 @@ const char *audio_application_name(void);
 typedef struct RateCtl {
     int64_t start_ticks;
     int64_t bytes_sent;
+    int64_t peeked_frames;
 } RateCtl;
 
 void audio_rate_start(RateCtl *rate);
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 7ccfec0116..c29d79c443 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -174,9 +174,11 @@ static int glue (audio_pcm_sw_init_, TYPE) (
 
     if (sw->info.is_float) {
 #ifdef DAC
-        sw->conv = mixeng_conv_float[sw->info.nchannels == 2];
+        sw->conv = mixeng_conv_float[sw->info.nchannels == 2]
+            [sw->info.swap_endianness];
 #else
-        sw->clip = mixeng_clip_float[sw->info.nchannels == 2];
+        sw->clip = mixeng_clip_float[sw->info.nchannels == 2]
+            [sw->info.swap_endianness];
 #endif
     } else {
 #ifdef DAC
@@ -303,9 +305,11 @@ static HW *glue(audio_pcm_hw_add_new_, TYPE)(AudioState *s,
 
     if (hw->info.is_float) {
 #ifdef DAC
-        hw->clip = mixeng_clip_float[hw->info.nchannels == 2];
+        hw->clip = mixeng_clip_float[hw->info.nchannels == 2]
+            [hw->info.swap_endianness];
 #else
-        hw->conv = mixeng_conv_float[hw->info.nchannels == 2];
+        hw->conv = mixeng_conv_float[hw->info.nchannels == 2]
+            [hw->info.swap_endianness];
 #endif
     } else {
 #ifdef DAC
diff --git a/audio/mixeng.c b/audio/mixeng.c
index 69f6549224..703ee5448f 100644
--- a/audio/mixeng.c
+++ b/audio/mixeng.c
@@ -283,10 +283,15 @@ static const float float_scale_reciprocal = 1.f / ((int64_t)INT32_MAX + 1);
 #endif
 #endif
 
+#define F32_TO_F32S(v) \
+    bswap32((union { uint32_t i; float f; }){ .f = (v) }.i)
+#define F32S_TO_F32(v) \
+    ((union { uint32_t i; float f; }){ .i = bswap32(v) }.f)
+
 static void conv_natural_float_to_mono(struct st_sample *dst, const void *src,
                                        int samples)
 {
-    float *in = (float *)src;
+    const float *in = src;
 
     while (samples--) {
         dst->r = dst->l = CONV_NATURAL_FLOAT(*in++);
@@ -294,10 +299,21 @@ static void conv_natural_float_to_mono(struct st_sample *dst, const void *src,
     }
 }
 
+static void conv_swap_float_to_mono(struct st_sample *dst, const void *src,
+                                    int samples)
+{
+    const uint32_t *in_f32s = src;
+
+    while (samples--) {
+        dst->r = dst->l = CONV_NATURAL_FLOAT(F32S_TO_F32(*in_f32s++));
+        dst++;
+    }
+}
+
 static void conv_natural_float_to_stereo(struct st_sample *dst, const void *src,
                                          int samples)
 {
-    float *in = (float *)src;
+    const float *in = src;
 
     while (samples--) {
         dst->l = CONV_NATURAL_FLOAT(*in++);
@@ -306,15 +322,33 @@ static void conv_natural_float_to_stereo(struct st_sample *dst, const void *src,
     }
 }
 
-t_sample *mixeng_conv_float[2] = {
-    conv_natural_float_to_mono,
-    conv_natural_float_to_stereo,
+static void conv_swap_float_to_stereo(struct st_sample *dst, const void *src,
+                                      int samples)
+{
+    const uint32_t *in_f32s = src;
+
+    while (samples--) {
+        dst->l = CONV_NATURAL_FLOAT(F32S_TO_F32(*in_f32s++));
+        dst->r = CONV_NATURAL_FLOAT(F32S_TO_F32(*in_f32s++));
+        dst++;
+    }
+}
+
+t_sample *mixeng_conv_float[2][2] = {
+    {
+        conv_natural_float_to_mono,
+        conv_swap_float_to_mono,
+    },
+    {
+        conv_natural_float_to_stereo,
+        conv_swap_float_to_stereo,
+    }
 };
 
 static void clip_natural_float_from_mono(void *dst, const struct st_sample *src,
                                          int samples)
 {
-    float *out = (float *)dst;
+    float *out = dst;
 
     while (samples--) {
         *out++ = CLIP_NATURAL_FLOAT(src->l + src->r);
@@ -322,10 +356,21 @@ static void clip_natural_float_from_mono(void *dst, const struct st_sample *src,
     }
 }
 
+static void clip_swap_float_from_mono(void *dst, const struct st_sample *src,
+                                      int samples)
+{
+    uint32_t *out_f32s = dst;
+
+    while (samples--) {
+        *out_f32s++ = F32_TO_F32S(CLIP_NATURAL_FLOAT(src->l + src->r));
+        src++;
+    }
+}
+
 static void clip_natural_float_from_stereo(
     void *dst, const struct st_sample *src, int samples)
 {
-    float *out = (float *)dst;
+    float *out = dst;
 
     while (samples--) {
         *out++ = CLIP_NATURAL_FLOAT(src->l);
@@ -334,9 +379,27 @@ static void clip_natural_float_from_stereo(
     }
 }
 
-f_sample *mixeng_clip_float[2] = {
-    clip_natural_float_from_mono,
-    clip_natural_float_from_stereo,
+static void clip_swap_float_from_stereo(
+    void *dst, const struct st_sample *src, int samples)
+{
+    uint32_t *out_f32s = dst;
+
+    while (samples--) {
+        *out_f32s++ = F32_TO_F32S(CLIP_NATURAL_FLOAT(src->l));
+        *out_f32s++ = F32_TO_F32S(CLIP_NATURAL_FLOAT(src->r));
+        src++;
+    }
+}
+
+f_sample *mixeng_clip_float[2][2] = {
+    {
+        clip_natural_float_from_mono,
+        clip_swap_float_from_mono,
+    },
+    {
+        clip_natural_float_from_stereo,
+        clip_swap_float_from_stereo,
+    }
 };
 
 void audio_sample_to_uint64(const void *samples, int pos,
diff --git a/audio/mixeng.h b/audio/mixeng.h
index a5f56d2c26..ead93ac2f7 100644
--- a/audio/mixeng.h
+++ b/audio/mixeng.h
@@ -42,9 +42,9 @@ typedef void (f_sample) (void *dst, const struct st_sample *src, int samples);
 extern t_sample *mixeng_conv[2][2][2][3];
 extern f_sample *mixeng_clip[2][2][2][3];
 
-/* indices: [stereo] */
-extern t_sample *mixeng_conv_float[2];
-extern f_sample *mixeng_clip_float[2];
+/* indices: [stereo][swap endianness] */
+extern t_sample *mixeng_conv_float[2][2];
+extern f_sample *mixeng_clip_float[2][2];
 
 void *st_rate_start (int inrate, int outrate);
 void st_rate_flow(void *opaque, st_sample *ibuf, st_sample *obuf,
diff --git a/hw/audio/asc.c b/hw/audio/asc.c
index 18382ccf6a..edd42d6d91 100644
--- a/hw/audio/asc.c
+++ b/hw/audio/asc.c
@@ -12,6 +12,7 @@
 
 #include "qemu/osdep.h"
 #include "qemu/timer.h"
+#include "qapi/error.h"
 #include "hw/sysbus.h"
 #include "hw/irq.h"
 #include "audio/audio.h"
@@ -653,11 +654,17 @@ static void asc_realize(DeviceState *dev, Error **errp)
 
     s->voice = AUD_open_out(&s->card, s->voice, "asc.out", s, asc_out_cb,
                             &as);
+    if (!s->voice) {
+        AUD_remove_card(&s->card);
+        error_setg(errp, "Initializing audio stream failed");
+        return;
+    }
+
     s->shift = 1;
     s->samples = AUD_get_buffer_size_out(s->voice) >> s->shift;
     s->mixbuf = g_malloc0(s->samples << s->shift);
 
-    s->silentbuf = g_malloc0(s->samples << s->shift);
+    s->silentbuf = g_malloc(s->samples << s->shift);
     memset(s->silentbuf, 0x80, s->samples << s->shift);
 
     /* Add easc registers if required */
diff --git a/qapi/audio.json b/qapi/audio.json
index dd5a58d13e..49633cf317 100644
--- a/qapi/audio.json
+++ b/qapi/audio.json
@@ -96,7 +96,7 @@
 # @period-length: the period length in microseconds
 #
 # @try-poll: attempt to use poll mode, falling back to non-polling
-#     access on failure (default true)
+#     access on failure (default false)
 #
 # Since: 4.0
 ##
diff --git a/qemu-options.hx b/qemu-options.hx
index aab53bcfe8..7eb8e02b4b 100644
--- a/qemu-options.hx
+++ b/qemu-options.hx
@@ -965,7 +965,7 @@ SRST
         Sets the period length in microseconds.
 
     ``in|out.try-poll=on|off``
-        Attempt to use poll mode with the device. Default is on.
+        Attempt to use poll mode with the device. Default is off.
 
     ``threshold=threshold``
         Threshold (in microseconds) when playback starts. Default is 0.
@@ -1002,7 +1002,7 @@ SRST
     ``in|out.buffer-count=count``
         Sets the count of the buffers.
 
-    ``in|out.try-poll=on|of``
+    ``in|out.try-poll=on|off``
         Attempt to use poll mode with the device. Default is on.
 
     ``try-mmap=on|off``
diff --git a/tests/functional/test_m68k_q800.py b/tests/functional/test_m68k_q800.py
index 400b7aeb5d..b3e655346c 100755
--- a/tests/functional/test_m68k_q800.py
+++ b/tests/functional/test_m68k_q800.py
@@ -25,7 +25,8 @@ class Q800MachineTest(LinuxKernelTest):
         kernel_command_line = (self.KERNEL_COMMON_COMMAND_LINE +
                                'console=ttyS0 vga=off')
         self.vm.add_args('-kernel', kernel_path,
-                         '-append', kernel_command_line)
+                         '-append', kernel_command_line,
+                         '-audio', 'none')
         self.vm.launch()
         console_pattern = 'Kernel command line: %s' % kernel_command_line
         self.wait_for_console_pattern(console_pattern)
diff --git a/tests/functional/test_m68k_replay.py b/tests/functional/test_m68k_replay.py
index 18c1db539c..213d6ae07e 100755
--- a/tests/functional/test_m68k_replay.py
+++ b/tests/functional/test_m68k_replay.py
@@ -24,7 +24,8 @@ class M68kReplay(ReplayKernelBase):
         kernel_command_line = (self.KERNEL_COMMON_COMMAND_LINE +
                                'console=ttyS0 vga=off')
         console_pattern = 'No filesystem could mount root'
-        self.run_rr(kernel_path, kernel_command_line, console_pattern)
+        self.run_rr(kernel_path, kernel_command_line, console_pattern,
+                    args=('-audio', 'none'))
 
     ASSET_MCF5208 = Asset(
        'https://qemu-advcal.gitlab.io/qac-best-of-multiarch/download/day07.tar.xz',