Hi
On Fri, Feb 13, 2026 at 4:28 PM Mark Cave-Ayland
<[email protected]> wrote:
>
> On 27/01/2026 18:24, [email protected] wrote:
>
> > From: Marc-André Lureau <[email protected]>
> >
> > The audio_pcm_info structure stored three fields (bits, is_signed,
> > is_float) that were always derived from the AudioFormat enum. This
> > redundancy meant the same information was represented twice, with no
> > type-level guarantee that they stayed in sync.
> >
> > Replace these fields with a single AudioFormat field, and add helper
> > functions to extract the derived properties when needed:
> > - audio_format_bits()
> > - audio_format_is_signed()
> > - audio_format_is_float()
> >
> > This improves type safety by making AudioFormat the single source of
> > truth, eliminating the possibility of inconsistent state between the
> > format enum and its derived boolean/integer representations.
> >
> > Signed-off-by: Marc-André Lureau <[email protected]>
> > ---
> > audio/audio_int.h | 4 +-
> > audio/audio_template.h | 12 +--
> > include/qemu/audio.h | 49 +++++++++
> > audio/audio-mixeng-be.c | 218 ++++++++++++----------------------------
> > audio/dbusaudio.c | 12 +--
> > audio/coreaudio.m | 2 +-
> > 6 files changed, 126 insertions(+), 171 deletions(-)
> >
> > diff --git a/audio/audio_int.h b/audio/audio_int.h
> > index 5334c4baad2..dd5f2220d75 100644
> > --- a/audio/audio_int.h
> > +++ b/audio/audio_int.h
> > @@ -45,9 +45,7 @@ struct audio_callback {
> > };
> >
> > struct audio_pcm_info {
> > - int bits;
> > - bool is_signed;
> > - bool is_float;
> > + AudioFormat af;
> > int freq;
> > int nchannels;
> > int bytes_per_frame;
> > diff --git a/audio/audio_template.h b/audio/audio_template.h
> > index 08d60422589..3da91a4782c 100644
> > --- a/audio/audio_template.h
> > +++ b/audio/audio_template.h
> > @@ -173,7 +173,7 @@ static int glue (audio_pcm_sw_init_, TYPE) (
> > sw->empty = true;
> > #endif
> >
> > - if (sw->info.is_float) {
> > + if (audio_format_is_float(hw->info.af)) {
> > #ifdef DAC
> > sw->conv = mixeng_conv_float[sw->info.nchannels == 2]
> > [sw->info.swap_endianness];
> > @@ -188,9 +188,9 @@ static int glue (audio_pcm_sw_init_, TYPE) (
> > sw->clip = mixeng_clip
> > #endif
> > [sw->info.nchannels == 2]
> > - [sw->info.is_signed]
> > + [audio_format_is_signed(hw->info.af)]
> > [sw->info.swap_endianness]
> > - [audio_bits_to_index(sw->info.bits)];
> > + [audio_format_to_index(hw->info.af)];
> > }
> >
> > sw->name = g_strdup (name);
> > @@ -300,7 +300,7 @@ static HW *glue(audio_pcm_hw_add_new_,
> > TYPE)(AudioMixengBackend *s,
> > goto err1;
> > }
> >
> > - if (hw->info.is_float) {
> > + if (audio_format_is_float(hw->info.af)) {
> > #ifdef DAC
> > hw->clip = mixeng_clip_float[hw->info.nchannels == 2]
> > [hw->info.swap_endianness];
> > @@ -315,9 +315,9 @@ static HW *glue(audio_pcm_hw_add_new_,
> > TYPE)(AudioMixengBackend *s,
> > hw->conv = mixeng_conv
> > #endif
> > [hw->info.nchannels == 2]
> > - [hw->info.is_signed]
> > + [audio_format_is_signed(hw->info.af)]
> > [hw->info.swap_endianness]
> > - [audio_bits_to_index(hw->info.bits)];
> > + [audio_format_to_index(hw->info.af)];
> > }
> >
> > glue(audio_pcm_hw_alloc_resources_, TYPE)(hw);
> > diff --git a/include/qemu/audio.h b/include/qemu/audio.h
> > index b6b6ee9b560..a2fbc286eb1 100644
> > --- a/include/qemu/audio.h
> > +++ b/include/qemu/audio.h
> > @@ -185,6 +185,55 @@ bool audio_be_set_dbus_server(AudioBackend *be,
> >
> > const char *audio_application_name(void);
> >
> > +static inline int audio_format_bits(AudioFormat fmt)
>
> Does the inline make a difference here? My understanding is that the
> compiler treats this as a weak hint these days, and generally does the
> right thing by itself.
If you drop "inline" you get -Werror=unused-function.
>
> > +{
> > + switch (fmt) {
> > + case AUDIO_FORMAT_S8:
> > + case AUDIO_FORMAT_U8:
> > + return 8;
> > +
> > + case AUDIO_FORMAT_S16:
> > + case AUDIO_FORMAT_U16:
> > + return 16;
> > +
> > + case AUDIO_FORMAT_F32:
> > + case AUDIO_FORMAT_S32:
> > + case AUDIO_FORMAT_U32:
> > + return 32;
> > +
> > + case AUDIO_FORMAT__MAX:
> > + break;
> > + }
>
> I'm not sure that AUDIO_FORMAT__MAX is a valid choice here - can you
> drop the explicit case for AUDIO_FORMAT__MAX and then simply have a
> default that calls g_assert_not_reached() instead?
We need to handle all variants, because -Werror=switch. That
>
> > + g_assert_not_reached();
> > +}
> > +
> > +static inline bool audio_format_is_float(AudioFormat fmt)
>
> Same comment here re: inline.
>
> > +{
> > + return fmt == AUDIO_FORMAT_F32;
> > +}
> > +
> > +static inline bool audio_format_is_signed(AudioFormat fmt)
>
> And here.
>
> > +{
> > + switch (fmt) {
> > + case AUDIO_FORMAT_S8:
> > + case AUDIO_FORMAT_S16:
> > + case AUDIO_FORMAT_S32:
> > + case AUDIO_FORMAT_F32:
> > + return true;
> > +
> > + case AUDIO_FORMAT_U8:
> > + case AUDIO_FORMAT_U16:
> > + case AUDIO_FORMAT_U32:
> > + return false;
> > +
> > + case AUDIO_FORMAT__MAX:
>
> Same comment here re: AUDIO_FORMAT__MAX.
>
> > + break;
> > + }
> > +
> > + g_assert_not_reached();
> > +}
> > +
> > #define DEFINE_AUDIO_PROPERTIES(_s, _f) \
> > DEFINE_PROP_AUDIODEV("audiodev", _s, _f)
> >
> > diff --git a/audio/audio-mixeng-be.c b/audio/audio-mixeng-be.c
> > index a0e542754e5..146026d0b39 100644
> > --- a/audio/audio-mixeng-be.c
> > +++ b/audio/audio-mixeng-be.c
> > @@ -62,23 +62,28 @@ int audio_bug (const char *funcname, int cond)
> > return cond;
> > }
> >
> > -static inline int audio_bits_to_index (int bits)
> > +/*
> > + * Convert audio format to mixeng_clip index. Used by audio_pcm_sw_init_
> > and
> > + * audio_mixeng_backend_add_capture()
> > + */
> > +static int audio_format_to_index(AudioFormat af)
> > {
> > - switch (bits) {
> > - case 8:
> > + switch (af) {
> > + case AUDIO_FORMAT_U8:
> > + case AUDIO_FORMAT_S8:
> > return 0;
> > -
> > - case 16:
> > + case AUDIO_FORMAT_U16:
> > + case AUDIO_FORMAT_S16:
> > return 1;
> > -
> > - case 32:
> > + case AUDIO_FORMAT_U32:
> > + case AUDIO_FORMAT_S32:
> > return 2;
> > -
> > - default:
> > - audio_bug ("bits_to_index", 1);
> > - AUD_log (NULL, "invalid bits %d\n", bits);
> > - return 0;
> > + case AUDIO_FORMAT_F32:
> > + case AUDIO_FORMAT__MAX:
>
> Same comment re: AUDIO_FORMAT__MAX.
>
> > + break;
> > }
> > +
> > + g_assert_not_reached();
> > }
> >
> > void AUD_vlog (const char *cap, const char *fmt, va_list ap)
> > @@ -172,141 +177,68 @@ static int audio_validate_settings (const struct
> > audsettings *as)
> >
> > static int audio_pcm_info_eq (struct audio_pcm_info *info, const struct
> > audsettings *as)
> > {
> > - int bits = 8;
> > - bool is_signed = false, is_float = false;
> > -
> > - switch (as->fmt) {
> > - case AUDIO_FORMAT_S8:
> > - is_signed = true;
> > - /* fall through */
> > - case AUDIO_FORMAT_U8:
> > - break;
> > -
> > - case AUDIO_FORMAT_S16:
> > - is_signed = true;
> > - /* fall through */
> > - case AUDIO_FORMAT_U16:
> > - bits = 16;
> > - break;
> > -
> > - case AUDIO_FORMAT_F32:
> > - is_float = true;
> > - /* fall through */
> > - case AUDIO_FORMAT_S32:
> > - is_signed = true;
> > - /* fall through */
> > - case AUDIO_FORMAT_U32:
> > - bits = 32;
> > - break;
> > -
> > - default:
> > - abort();
> > - }
> > - return info->freq == as->freq
> > + return info->af == as->fmt
> > + && info->freq == as->freq
> > && info->nchannels == as->nchannels
> > - && info->is_signed == is_signed
> > - && info->is_float == is_float
> > - && info->bits == bits
> > && info->swap_endianness == (as->endianness != HOST_BIG_ENDIAN);
> > }
> >
> > void audio_pcm_init_info (struct audio_pcm_info *info, const struct
> > audsettings *as)
> > {
> > - int bits = 8, mul;
> > - bool is_signed = false, is_float = false;
> > -
> > - switch (as->fmt) {
> > - case AUDIO_FORMAT_S8:
> > - is_signed = true;
> > - /* fall through */
> > - case AUDIO_FORMAT_U8:
> > - mul = 1;
> > - break;
> > -
> > - case AUDIO_FORMAT_S16:
> > - is_signed = true;
> > - /* fall through */
> > - case AUDIO_FORMAT_U16:
> > - bits = 16;
> > - mul = 2;
> > - break;
> > -
> > - case AUDIO_FORMAT_F32:
> > - is_float = true;
> > - /* fall through */
> > - case AUDIO_FORMAT_S32:
> > - is_signed = true;
> > - /* fall through */
> > - case AUDIO_FORMAT_U32:
> > - bits = 32;
> > - mul = 4;
> > - break;
> > -
> > - default:
> > - abort();
> > - }
> > -
> > + info->af = as->fmt;
> > info->freq = as->freq;
> > - info->bits = bits;
> > - info->is_signed = is_signed;
> > - info->is_float = is_float;
> > info->nchannels = as->nchannels;
> > - info->bytes_per_frame = as->nchannels * mul;
> > + info->bytes_per_frame = as->nchannels * audio_format_bits(as->fmt) / 8;
> > info->bytes_per_second = info->freq * info->bytes_per_frame;
> > info->swap_endianness = (as->endianness != HOST_BIG_ENDIAN);
> > }
> >
> > -void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int
> > len)
> > +void audio_pcm_info_clear_buf(struct audio_pcm_info *info, void *buf, int
> > len)
> > {
> > if (!len) {
> > return;
> > }
> >
> > - if (info->is_signed || info->is_float) {
> > - memset(buf, 0x00, len * info->bytes_per_frame);
> > - } else {
> > - switch (info->bits) {
> > - case 8:
> > - memset(buf, 0x80, len * info->bytes_per_frame);
> > - break;
> > -
> > - case 16:
> > - {
> > - int i;
> > - uint16_t *p = buf;
> > - short s = INT16_MAX;
> > -
> > - if (info->swap_endianness) {
> > - s = bswap16 (s);
> > - }
> > -
> > - for (i = 0; i < len * info->nchannels; i++) {
> > - p[i] = s;
> > - }
> > - }
> > - break;
> > + switch (info->af) {
> > + case AUDIO_FORMAT_U8:
> > + memset(buf, 0x80, len * info->bytes_per_frame);
>
> This doesn't look right - isn't this the signed version?
No,
>
> > + break;
> > + case AUDIO_FORMAT_U16: {
> > + int i;
> > + uint16_t *p = buf;
> > + short s = INT16_MAX;
> >
> > - case 32:
> > - {
> > - int i;
> > - uint32_t *p = buf;
> > - int32_t s = INT32_MAX;
> > + if (info->swap_endianness) {
> > + s = bswap16(s);
> > + }
> >
> > - if (info->swap_endianness) {
> > - s = bswap32 (s);
> > - }
> > + for (i = 0; i < len * info->nchannels; i++) {
> > + p[i] = s;
> > + }
>
> I think this is signed too?
>
> > + break;
> > + }
> > + case AUDIO_FORMAT_U32: {
> > + int i;
> > + uint32_t *p = buf;
> > + int32_t s = INT32_MAX;
> >
> > - for (i = 0; i < len * info->nchannels; i++) {
> > - p[i] = s;
> > - }
> > - }
> > - break;
> > + if (info->swap_endianness) {
> > + s = bswap32(s);
> > + }
> >
> > - default:
> > - AUD_log (NULL, "audio_pcm_info_clear_buf: invalid bits %d\n",
> > - info->bits);
> > - break;
> > + for (i = 0; i < len * info->nchannels; i++) {
> > + p[i] = s;
> > }
>
> And also here.
>
> > + break;
> > + }
> > + case AUDIO_FORMAT_S8:
> > + case AUDIO_FORMAT_S16:
> > + case AUDIO_FORMAT_S32:
> > + case AUDIO_FORMAT_F32:
> > + memset(buf, 0x00, len * info->bytes_per_frame);
>
> ... and this the unsigned version? I think they've been swapped.
no, unsigned PCM use mid-range value for zero.
>
> > + break;
> > + case AUDIO_FORMAT__MAX:
> > + g_assert_not_reached();
>
> This looks more like I would expect, although possibly replace the case
> with a default? I expect that another assert() would have tripped by now
> with real usage if AUDIO_FORMAT__MAX were specified, but I think it
> would be good to be consistent.
Unfortunately, the compiler complains for non-void functions if you
don't finish the function with the g_assert_not_reached().
>
> > }
> > }
> >
> > @@ -719,8 +651,8 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void
> > *buf, size_t buf_len)
> > #ifdef DEBUG_AUDIO
> > static void audio_pcm_print_info (const char *cap, struct audio_pcm_info
> > *info)
> > {
> > - dolog("%s: bits %d, sign %d, float %d, freq %d, nchan %d\n",
> > - cap, info->bits, info->is_signed, info->is_float, info->freq,
> > + dolog("%s: %s, freq %d, nchan %d\n",
> > + cap, AudioFormat_str(info->af), info->freq,
> > info->nchannels);
>
> More dolog. Presumably this gets converted to error_report() later?
>
> > }
> > #endif
> > @@ -1759,15 +1691,15 @@ static CaptureVoiceOut
> > *audio_mixeng_backend_add_capture(
> >
> > cap->buf = g_malloc0_n(hw->mix_buf.size,
> > hw->info.bytes_per_frame);
> >
> > - if (hw->info.is_float) {
> > + if (audio_format_is_float(hw->info.af)) {
> > hw->clip = mixeng_clip_float[hw->info.nchannels == 2]
> > [hw->info.swap_endianness];
> > } else {
> > hw->clip = mixeng_clip
> > [hw->info.nchannels == 2]
> > - [hw->info.is_signed]
> > + [audio_format_is_signed(hw->info.af)]
> > [hw->info.swap_endianness]
> > - [audio_bits_to_index(hw->info.bits)];
> > + [audio_format_to_index(hw->info.af)];
> > }
> >
> > QLIST_INSERT_HEAD (&s->cap_head, cap, entries);
> > @@ -1869,29 +1801,6 @@ audsettings
> > audiodev_to_audsettings(AudiodevPerDirectionOptions *pdo)
> > };
> > }
> >
> > -int audioformat_bytes_per_sample(AudioFormat fmt)
> > -{
> > - switch (fmt) {
> > - case AUDIO_FORMAT_U8:
> > - case AUDIO_FORMAT_S8:
> > - return 1;
> > -
> > - case AUDIO_FORMAT_U16:
> > - case AUDIO_FORMAT_S16:
> > - return 2;
> > -
> > - case AUDIO_FORMAT_U32:
> > - case AUDIO_FORMAT_S32:
> > - case AUDIO_FORMAT_F32:
> > - return 4;
> > -
> > - case AUDIO_FORMAT__MAX:
> > - ;
> > - }
> > - abort();
> > -}
> > -
> > -
> > /* frames = freq * usec / 1e6 */
> > int audio_buffer_frames(AudiodevPerDirectionOptions *pdo,
> > audsettings *as, int def_usecs)
> > @@ -1914,8 +1823,7 @@ int audio_buffer_samples(AudiodevPerDirectionOptions
> > *pdo,
> > int audio_buffer_bytes(AudiodevPerDirectionOptions *pdo,
> > audsettings *as, int def_usecs)
> > {
> > - return audio_buffer_samples(pdo, as, def_usecs) *
> > - audioformat_bytes_per_sample(as->fmt);
> > + return audio_buffer_samples(pdo, as, def_usecs) *
> > audio_format_bits(as->fmt) / 8;
> > }
> >
> > void audio_rate_start(RateCtl *rate)
> > diff --git a/audio/dbusaudio.c b/audio/dbusaudio.c
> > index e284542b2dd..72d6194033b 100644
> > --- a/audio/dbusaudio.c
> > +++ b/audio/dbusaudio.c
> > @@ -147,9 +147,9 @@ dbus_init_out_listener(QemuDBusDisplay1AudioOutListener
> > *listener,
> > qemu_dbus_display1_audio_out_listener_call_init(
> > listener,
> > (uintptr_t)hw,
> > - hw->info.bits,
> > - hw->info.is_signed,
> > - hw->info.is_float,
> > + audio_format_bits(hw->info.af),
> > + audio_format_is_signed(hw->info.af),
> > + audio_format_is_float(hw->info.af),
> > hw->info.freq,
> > hw->info.nchannels,
> > hw->info.bytes_per_frame,
> > @@ -273,9 +273,9 @@ dbus_init_in_listener(QemuDBusDisplay1AudioInListener
> > *listener, HWVoiceIn *hw)
> > qemu_dbus_display1_audio_in_listener_call_init(
> > listener,
> > (uintptr_t)hw,
> > - hw->info.bits,
> > - hw->info.is_signed,
> > - hw->info.is_float,
> > + audio_format_bits(hw->info.af),
> > + audio_format_is_signed(hw->info.af),
> > + audio_format_is_float(hw->info.af),
> > hw->info.freq,
> > hw->info.nchannels,
> > hw->info.bytes_per_frame,
> > diff --git a/audio/coreaudio.m b/audio/coreaudio.m
> > index 40d7986b1d7..08bab353831 100644
> > --- a/audio/coreaudio.m
> > +++ b/audio/coreaudio.m
> > @@ -359,7 +359,7 @@ static OSStatus init_out_device(coreaudioVoiceOut *core)
> > AudioValueRange frameRange;
> >
> > AudioStreamBasicDescription streamBasicDescription = {
> > - .mBitsPerChannel = core->hw.info.bits,
> > + .mBitsPerChannel = audio_format_bits(core->hw.info.af),
> > .mBytesPerFrame = core->hw.info.bytes_per_frame,
> > .mBytesPerPacket = core->hw.info.bytes_per_frame,
> > .mChannelsPerFrame = core->hw.info.nchannels,
>
>
> ATB,
>
> Mark.
>
