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

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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]
# gpg:                 aka "Marc-André Lureau <marcandre.lureau@gmail.com>" [full]
# Primary key fingerprint: 87A9 BD93 3F87 C606 D276  F62D DAE8 E109 7596 9CE5

* 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>
This commit is contained in:
Stefan Hajnoczi
2025-05-25 09:51:07 -04:00
parent 6f388a37e6 2bccabe6df
commit 3c5a5e213e
11 changed files with 119 additions and 31 deletions
Download Patch File Download Diff File Expand all files Collapse all files
audio/alsaaudio.c
+1 -1
View File
@@ -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;
}
}
audio/audio.c
+18 -7
View File
@@ -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;
}
audio/audio_int.h
+1
View File
@@ -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);
audio/audio_template.h
+8 -4
View File
@@ -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
audio/mixeng.c
+73 -10
View File
@@ -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,
audio/mixeng.h
+3 -3
View File
@@ -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,
hw/audio/asc.c
+8 -1
View File
@@ -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 */
qapi/audio.json
+1 -1
View File
@@ -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
##
qemu-options.hx
+2 -2
View File
@@ -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``
tests/functional/test_m68k_q800.py
+2 -1
View File
@@ -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)
tests/functional/test_m68k_replay.py
+2 -1
View File
@@ -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',