audio: proper support for float samples in mixeng

This adds proper support for float samples in mixeng by adding a new
audio format for it.

Limitations: only native endianness is supported.  None of the virtual
sound cards support float samples (it looks like most of them only
support 8 and 16 bit, only hda supports 32 bit), it is only used for the
audio backends (i.e. host side).

Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Acked-by: Markus Armbruster <armbru@redhat.com>
Message-id: 8a8b0b5698401b78d3c4c8ec90aef83b95babb06.1580672076.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
This commit is contained in:
Kővágó, Zoltán 2020-02-02 20:38:07 +01:00 committed by Gerd Hoffmann
parent 180b044ffd
commit ed2a4a7941
10 changed files with 182 additions and 81 deletions

View File

@ -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;

View File

@ -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
@ -1832,11 +1847,15 @@ 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];
} else {
hw->clip = mixeng_clip
[hw->info.nchannels == 2]
[hw->info.sign]
[hw->info.is_signed]
[hw->info.swap_endianness]
[audio_bits_to_index (hw->info.bits)];
[audio_bits_to_index(hw->info.bits)];
}
QLIST_INSERT_HEAD (&s->cap_head, cap, entries);
QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
@ -2075,6 +2094,7 @@ 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:

View File

@ -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;

View File

@ -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_float[sw->info.nchannels == 2];
#else
sw->clip = mixeng_clip_float[sw->info.nchannels == 2];
#endif
} else {
#ifdef DAC
sw->conv = mixeng_conv
#else
sw->clip = mixeng_clip
#endif
[sw->info.nchannels == 2]
[sw->info.sign]
[sw->info.is_signed]
[sw->info.swap_endianness]
[audio_bits_to_index (sw->info.bits)];
[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
#else
hw->conv = mixeng_conv
#endif
[hw->info.nchannels == 2]
[hw->info.sign]
[hw->info.is_signed]
[hw->info.swap_endianness]
[audio_bits_to_index (hw->info.bits)];
[audio_bits_to_index(hw->info.bits)];
}
glue(audio_pcm_hw_alloc_resources_, TYPE)(hw);

View File

@ -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);

View File

@ -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,
#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++;
}
}
void clip_natural_float_from_stereo(void *dst, const struct st_sample *src,
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++;
}
}
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)
{

View File

@ -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,

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@ -277,6 +277,9 @@ 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;
@ -302,6 +305,12 @@ 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;

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@ -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;

View File

@ -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: