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audio: proper support for float samples in mixeng

Browse Source 
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
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

17
audio/alsaaudio.c
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;

56
audio/audio.c
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);
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);
@ -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:

3
audio/audio_int.h
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;

41
audio/audio_template.h
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
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);

7
audio/coreaudio.c
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);

92
audio/mixeng.c
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,
int samples)
{
float *in = (float *)src;
#ifndef FLOAT_MIXENG
const float scale = UINT_MAX;
#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;
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,
int samples)
static void conv_natural_float_to_stereo(struct st_sample *dst, const void *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
float *in = (float *)src;
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
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;
while (samples--) {
*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)
{

8
audio/mixeng.h
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,

9
audio/paaudio.c
View File

@ -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;

28
audio/sdlaudio.c
View File

@ -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
qapi/audio.json
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: