This adds proper support for float samples in mixeng by adding a new
audio format for it.
Limitations: only native endianness is supported.
Signed-off-by: Kővágó, Zoltán <address@hidden>
---
This patch is meant to be applied on top of "[PATCH] coreaudio: fix coreaudio
playback" by Volker Rümelin, available at:
https://lists.nongnu.org/archive/html/qemu-devel/2020-02/msg00114.html
For more information, please refer to that thread.
---
qapi/audio.json | 2 +-
audio/audio_int.h | 3 +-
audio/audio_template.h | 41 ++++++++++++--------
audio/mixeng.h | 8 ++--
audio/alsaaudio.c | 17 ++++++++
audio/audio.c | 56 ++++++++++++++++++---------
audio/coreaudio.c | 7 +---
audio/mixeng.c | 88 ++++++++++++++++++++++++++----------------
audio/paaudio.c | 9 +++++
audio/sdlaudio.c | 28 ++++++++++++++
10 files changed, 180 insertions(+), 79 deletions(-)
diff --git a/qapi/audio.json b/qapi/audio.json
index 83312b2339..d8c507cced 100644
--- a/qapi/audio.json
+++ b/qapi/audio.json
@@ -276,7 +276,7 @@
# Since: 4.0
##
{ 'enum': 'AudioFormat',
- 'data': [ 'u8', 's8', 'u16', 's16', 'u32', 's32' ] }
+ 'data': [ 'u8', 's8', 'u16', 's16', 'u32', 's32', 'f32' ] }
##
# @AudiodevDriver:
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 5ba2078346..cd92e48163 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -40,7 +40,8 @@ struct audio_callback {
struct audio_pcm_info {
int bits;
- int sign;
+ bool is_signed;
+ bool is_float;
int freq;
int nchannels;
int bytes_per_frame;
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 0336d2670c..7013d3041f 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -153,15 +153,23 @@ static int glue (audio_pcm_sw_init_, TYPE) (
sw->ratio = ((int64_t) sw->info.freq << 32) / sw->hw->info.freq;
#endif
+ if (sw->info.is_float) {
#ifdef DAC
- sw->conv = mixeng_conv
+ sw->conv = mixeng_conv_float[sw->info.nchannels == 2];
#else
- sw->clip = mixeng_clip
+ sw->clip = mixeng_clip_float[sw->info.nchannels == 2];
#endif
- [sw->info.nchannels == 2]
- [sw->info.sign]
- [sw->info.swap_endianness]
- [audio_bits_to_index (sw->info.bits)];
+ } else {
+#ifdef DAC
+ sw->conv = mixeng_conv
+#else
+ sw->clip = mixeng_clip
+#endif
+ [sw->info.nchannels == 2]
+ [sw->info.is_signed]
+ [sw->info.swap_endianness]
+ [audio_bits_to_index(sw->info.bits)];
+ }
sw->name = g_strdup (name);
err = glue (audio_pcm_sw_alloc_resources_, TYPE) (sw);
@@ -276,22 +284,23 @@ static HW *glue(audio_pcm_hw_add_new_, TYPE)(AudioState *s,
goto err1;
}
- if (s->dev->driver == AUDIODEV_DRIVER_COREAUDIO) {
+ if (hw->info.is_float) {
#ifdef DAC
- hw->clip = clip_natural_float_from_stereo;
+ hw->clip = mixeng_clip_float[hw->info.nchannels == 2];
#else
- hw->conv = conv_natural_float_to_stereo;
+ hw->conv = mixeng_conv_float[hw->info.nchannels == 2];
#endif
- } else
+ } else {
#ifdef DAC
- hw->clip = mixeng_clip
+ hw->clip = mixeng_clip
#else
- hw->conv = mixeng_conv
+ hw->conv = mixeng_conv
#endif
- [hw->info.nchannels == 2]
- [hw->info.sign]
- [hw->info.swap_endianness]
- [audio_bits_to_index (hw->info.bits)];
+ [hw->info.nchannels == 2]
+ [hw->info.is_signed]
+ [hw->info.swap_endianness]
+ [audio_bits_to_index(hw->info.bits)];
+ }
glue(audio_pcm_hw_alloc_resources_, TYPE)(hw);
diff --git a/audio/mixeng.h b/audio/mixeng.h
index 7ef61763e8..2dcd6df245 100644
--- a/audio/mixeng.h
+++ b/audio/mixeng.h
@@ -38,13 +38,13 @@ typedef struct st_sample st_sample;
typedef void (t_sample) (struct st_sample *dst, const void *src, int samples);
typedef void (f_sample) (void *dst, const struct st_sample *src, int samples);
+/* indices: [stereo][signed][swap endiannes][8, 16 or 32-bits] */
extern t_sample *mixeng_conv[2][2][2][3];
extern f_sample *mixeng_clip[2][2][2][3];
-void conv_natural_float_to_stereo(struct st_sample *dst, const void *src,
- int samples);
-void clip_natural_float_from_stereo(void *dst, const struct st_sample *src,
- int samples);
+/* indices: [stereo] */
+extern t_sample *mixeng_conv_float[2];
+extern f_sample *mixeng_clip_float[2];
void *st_rate_start (int inrate, int outrate);
void st_rate_flow(void *opaque, st_sample *ibuf, st_sample *obuf,
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index f37ce1ce85..768b896a93 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -307,6 +307,13 @@ static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness)
return SND_PCM_FORMAT_U32_LE;
}
+ case AUDIO_FORMAT_F32:
+ if (endianness) {
+ return SND_PCM_FORMAT_FLOAT_BE;
+ } else {
+ return SND_PCM_FORMAT_FLOAT_LE;
+ }
+
default:
dolog ("Internal logic error: Bad audio format %d\n", fmt);
#ifdef DEBUG_AUDIO
@@ -370,6 +377,16 @@ static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt,
*fmt = AUDIO_FORMAT_U32;
break;
+ case SND_PCM_FORMAT_FLOAT_LE:
+ *endianness = 0;
+ *fmt = AUDIO_FORMAT_F32;
+ break;
+
+ case SND_PCM_FORMAT_FLOAT_BE:
+ *endianness = 1;
+ *fmt = AUDIO_FORMAT_F32;
+ break;
+
default:
dolog ("Unrecognized audio format %d\n", alsafmt);
return -1;
diff --git a/audio/audio.c b/audio/audio.c
index f63f39769a..53fdb42ec7 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -218,6 +218,9 @@ static void audio_print_settings (struct audsettings *as)
case AUDIO_FORMAT_U32:
AUD_log (NULL, "U32");
break;
+ case AUDIO_FORMAT_F32:
+ AUD_log (NULL, "F32");
+ break;
default:
AUD_log (NULL, "invalid(%d)", as->fmt);
break;
@@ -252,6 +255,7 @@ static int audio_validate_settings (struct audsettings *as)
case AUDIO_FORMAT_U16:
case AUDIO_FORMAT_S32:
case AUDIO_FORMAT_U32:
+ case AUDIO_FORMAT_F32:
break;
default:
invalid = 1;
@@ -264,24 +268,28 @@ static int audio_validate_settings (struct audsettings *as)
static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *as)
{
- int bits = 8, sign = 0;
+ int bits = 8;
+ bool is_signed = false, is_float = false;
switch (as->fmt) {
case AUDIO_FORMAT_S8:
- sign = 1;
+ is_signed = true;
/* fall through */
case AUDIO_FORMAT_U8:
break;
case AUDIO_FORMAT_S16:
- sign = 1;
+ 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:
- sign = 1;
+ is_signed = true;
/* fall through */
case AUDIO_FORMAT_U32:
bits = 32;
@@ -292,33 +300,38 @@ static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *a
}
return info->freq == as->freq
&& info->nchannels == as->nchannels
- && info->sign == sign
+ && info->is_signed == is_signed
+ && info->is_float == is_float
&& info->bits == bits
&& info->swap_endianness == (as->endianness != AUDIO_HOST_ENDIANNESS);
}
void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
{
- int bits = 8, sign = 0, mul;
+ int bits = 8, mul;
+ bool is_signed = false, is_float = false;
switch (as->fmt) {
case AUDIO_FORMAT_S8:
- sign = 1;
+ is_signed = true;
/* fall through */
case AUDIO_FORMAT_U8:
mul = 1;
break;
case AUDIO_FORMAT_S16:
- sign = 1;
+ 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:
- sign = 1;
+ is_signed = true;
/* fall through */
case AUDIO_FORMAT_U32:
bits = 32;
@@ -331,7 +344,8 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
info->freq = as->freq;
info->bits = bits;
- info->sign = sign;
+ info->is_signed = is_signed;
+ info->is_float = is_float;
info->nchannels = as->nchannels;
info->bytes_per_frame = as->nchannels * mul;
info->bytes_per_second = info->freq * info->bytes_per_frame;
@@ -344,7 +358,7 @@ void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
return;
}
- if (info->sign) {
+ if (info->is_signed || info->is_float) {
memset(buf, 0x00, len * info->bytes_per_frame);
}
else {
@@ -770,8 +784,9 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
#ifdef DEBUG_AUDIO
static void audio_pcm_print_info (const char *cap, struct audio_pcm_info *info)
{
- dolog ("%s: bits %d, sign %d, freq %d, nchan %d\n",
- cap, info->bits, info->sign, info->freq, info->nchannels);
+ dolog("%s: bits %d, sign %d, float %d, freq %d, nchan %d\n",
+ cap, info->bits, info->is_signed, info->is_float, info->freq,
+ info->nchannels);
}
#endif
@@ -1837,11 +1852,15 @@ CaptureVoiceOut *AUD_add_capture(
cap->buf = g_malloc0_n(hw->mix_buf->size, hw->info.bytes_per_frame);
- hw->clip = mixeng_clip
- [hw->info.nchannels == 2]
- [hw->info.sign]
- [hw->info.swap_endianness]
- [audio_bits_to_index (hw->info.bits)];
+ if (hw->info.is_float) {
+ hw->clip = mixeng_clip_float[hw->info.nchannels == 2];
+ } else {
+ hw->clip = mixeng_clip
+ [hw->info.nchannels == 2]
+ [hw->info.is_signed]
+ [hw->info.swap_endianness]
+ [audio_bits_to_index(hw->info.bits)];
+ }
QLIST_INSERT_HEAD (&s->cap_head, cap, entries);
QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
@@ -2080,6 +2099,7 @@ int audioformat_bytes_per_sample(AudioFormat fmt)
case AUDIO_FORMAT_U32:
case AUDIO_FORMAT_S32:
+ case AUDIO_FORMAT_F32:
return 4;
case AUDIO_FORMAT__MAX:
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index 0049db97fa..f1a009610c 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -491,14 +491,9 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
return -1;
}
- /*
- * The canonical audio format for CoreAudio on macOS is float. Currently
- * there is no generic code for AUDIO_FORMAT_F32 in qemu. Here we select
- * AUDIO_FORMAT_S32 instead because only the sample size has to match.
- */
fake_as = *as;
as = &fake_as;
- as->fmt = AUDIO_FORMAT_S32;
+ as->fmt = AUDIO_FORMAT_F32;
audio_pcm_init_info (&hw->info, as);
status = coreaudio_get_voice(&core->outputDeviceID);
diff --git a/audio/mixeng.c b/audio/mixeng.c
index 16b646d48c..c14b0d874c 100644
--- a/audio/mixeng.c
+++ b/audio/mixeng.c
@@ -267,55 +267,77 @@ f_sample *mixeng_clip[2][2][2][3] = {
}
};
-void conv_natural_float_to_stereo(struct st_sample *dst, const void *src,
- int samples)
+#ifdef FLOAT_MIXENG
+#define FLOAT_CONV_TO(x) (x)
+#define FLOAT_CONV_FROM(x) (x)
+#else
+static const float float_scale = UINT_MAX;
+#define FLOAT_CONV_TO(x) ((x) * float_scale)
+
+#ifdef RECIPROCAL
+static const float float_scale_reciprocal = 1.f / UINT_MAX;
+#define FLOAT_CONV_FROM(x) ((x) * float_scale_reciprocal)
+#else
+#define FLOAT_CONV_FROM(x) ((x) / float_scale)
+#endif
+#endif
+
+static void conv_natural_float_to_mono(struct st_sample *dst, const void *src,
+ int samples)
{
float *in = (float *)src;
-#ifndef FLOAT_MIXENG
- const float scale = UINT_MAX;
-#endif
while (samples--) {
-#ifdef FLOAT_MIXENG
- dst->l = *in++;
- dst->r = *in++;
-#else
- dst->l = *in++ * scale;
- dst->r = *in++ * scale;
-#endif
+ dst->r = dst->l = FLOAT_CONV_TO(*in++);
+ dst++;
+ }
+}
+
+static void conv_natural_float_to_stereo(struct st_sample *dst, const void *src,
+ int samples)
+{
+ float *in = (float *)src;
+
+ while (samples--) {
+ dst->l = FLOAT_CONV_TO(*in++);
+ dst->r = FLOAT_CONV_TO(*in++);
dst++;
}
}
-void clip_natural_float_from_stereo(void *dst, const struct st_sample *src,
- int samples)
+t_sample *mixeng_conv_float[2] = {
+ conv_natural_float_to_mono,
+ conv_natural_float_to_stereo,
+};
+
+static void clip_natural_float_from_mono(void *dst, const struct st_sample *src,
+ int samples)
{
float *out = (float *)dst;
-#ifndef FLOAT_MIXENG
-#ifdef RECIPROCAL
- const float scale = 1.f / UINT_MAX;
-#else
- const float scale = UINT_MAX;
-#endif
-#endif
while (samples--) {
-#ifdef FLOAT_MIXENG
- *out++ = src->l;
- *out++ = src->r;
-#else
-#ifdef RECIPROCAL
- *out++ = src->l * scale;
- *out++ = src->r * scale;
-#else
- *out++ = src->l / scale;
- *out++ = src->r / scale;
-#endif
-#endif
+ *out++ = FLOAT_CONV_FROM(src->l) + FLOAT_CONV_FROM(src->r);
+ src++;
+ }
+}
+
+static void clip_natural_float_from_stereo(
+ void *dst, const struct st_sample *src, int samples)
+{
+ float *out = (float *)dst;
+
+ while (samples--) {
+ *out++ = FLOAT_CONV_FROM(src->l);
+ *out++ = FLOAT_CONV_FROM(src->r);
src++;
}
}
+f_sample *mixeng_clip_float[2] = {
+ clip_natural_float_from_mono,
+ clip_natural_float_from_stereo,
+};
+
void audio_sample_to_uint64(void *samples, int pos,
uint64_t *left, uint64_t *right)
{
diff --git a/audio/paaudio.c b/audio/paaudio.c
index dbfe48c03a..1278c5a775 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -279,6 +279,9 @@ static pa_sample_format_t audfmt_to_pa (AudioFormat afmt, int endianness)
case AUDIO_FORMAT_U32:
format = endianness ? PA_SAMPLE_S32BE : PA_SAMPLE_S32LE;
break;
+ case AUDIO_FORMAT_F32:
+ format = endianness ? PA_SAMPLE_FLOAT32BE : PA_SAMPLE_FLOAT32LE;
+ break;
default:
dolog ("Internal logic error: Bad audio format %d\n", afmt);
format = PA_SAMPLE_U8;
@@ -304,6 +307,12 @@ static AudioFormat pa_to_audfmt (pa_sample_format_t fmt, int *endianness)
case PA_SAMPLE_S32LE:
*endianness = 0;
return AUDIO_FORMAT_S32;
+ case PA_SAMPLE_FLOAT32BE:
+ *endianness = 1;
+ return AUDIO_FORMAT_F32;
+ case PA_SAMPLE_FLOAT32LE:
+ *endianness = 0;
+ return AUDIO_FORMAT_F32;
default:
dolog ("Internal logic error: Bad pa_sample_format %d\n", fmt);
return AUDIO_FORMAT_U8;
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index 5c6bcfcb3e..6af1911db9 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -77,6 +77,14 @@ static int aud_to_sdlfmt (AudioFormat fmt)
case AUDIO_FORMAT_U16:
return AUDIO_U16LSB;
+ case AUDIO_FORMAT_S32:
+ return AUDIO_S32LSB;
+
+ /* no unsigned 32-bit support in SDL */
+
+ case AUDIO_FORMAT_F32:
+ return AUDIO_F32LSB;
+
default:
dolog ("Internal logic error: Bad audio format %d\n", fmt);
#ifdef DEBUG_AUDIO
@@ -119,6 +127,26 @@ static int sdl_to_audfmt(int sdlfmt, AudioFormat *fmt, int *endianness)
*fmt = AUDIO_FORMAT_U16;
break;
+ case AUDIO_S32LSB:
+ *endianness = 0;
+ *fmt = AUDIO_FORMAT_S32;
+ break;
+
+ case AUDIO_S32MSB:
+ *endianness = 1;
+ *fmt = AUDIO_FORMAT_S32;
+ break;
+
+ case AUDIO_F32LSB:
+ *endianness = 0;
+ *fmt = AUDIO_FORMAT_F32;
+ break;
+
+ case AUDIO_F32MSB:
+ *endianness = 1;
+ *fmt = AUDIO_FORMAT_F32;
+ break;
+
default:
dolog ("Unrecognized SDL audio format %d\n", sdlfmt);
return -1;
--
2.25.0
Hi,
Best,
Howard