Commit f941aa25 authored by ths's avatar ths

Qemu support for S32 and U32 alsa output, by Vassili Karpov.


git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@2427 c046a42c-6fe2-441c-8c8c-71466251a162
parent b93aebec
......@@ -157,6 +157,12 @@ static int aud_to_alsafmt (audfmt_e fmt)
case AUD_FMT_U16:
return SND_PCM_FORMAT_U16_LE;
case AUD_FMT_S32:
return SND_PCM_FORMAT_S32_LE;
case AUD_FMT_U32:
return SND_PCM_FORMAT_U32_LE;
default:
dolog ("Internal logic error: Bad audio format %d\n", fmt);
#ifdef DEBUG_AUDIO
......@@ -199,6 +205,26 @@ static int alsa_to_audfmt (int alsafmt, audfmt_e *fmt, int *endianness)
*fmt = AUD_FMT_U16;
break;
case SND_PCM_FORMAT_S32_LE:
*endianness = 0;
*fmt = AUD_FMT_S32;
break;
case SND_PCM_FORMAT_U32_LE:
*endianness = 0;
*fmt = AUD_FMT_U32;
break;
case SND_PCM_FORMAT_S32_BE:
*endianness = 1;
*fmt = AUD_FMT_S32;
break;
case SND_PCM_FORMAT_U32_BE:
*endianness = 1;
*fmt = AUD_FMT_U32;
break;
default:
dolog ("Unrecognized audio format %d\n", alsafmt);
return -1;
......
......@@ -80,7 +80,8 @@ static struct {
{
44100, /* freq */
2, /* nchannels */
AUD_FMT_S16 /* fmt */
AUD_FMT_S16, /* fmt */
AUDIO_HOST_ENDIANNESS
}
},
......@@ -91,7 +92,8 @@ static struct {
{
44100, /* freq */
2, /* nchannels */
AUD_FMT_S16 /* fmt */
AUD_FMT_S16, /* fmt */
AUDIO_HOST_ENDIANNESS
}
},
......@@ -166,6 +168,25 @@ int audio_bug (const char *funcname, int cond)
}
#endif
static inline int audio_bits_to_index (int bits)
{
switch (bits) {
case 8:
return 0;
case 16:
return 1;
case 32:
return 2;
default:
audio_bug ("bits_to_index", 1);
AUD_log (NULL, "invalid bits %d\n", bits);
return 0;
}
}
void *audio_calloc (const char *funcname, int nmemb, size_t size)
{
int cond;
......@@ -227,6 +248,12 @@ const char *audio_audfmt_to_string (audfmt_e fmt)
case AUD_FMT_S16:
return "S16";
case AUD_FMT_U32:
return "U32";
case AUD_FMT_S32:
return "S32";
}
dolog ("Bogus audfmt %d returning S16\n", fmt);
......@@ -243,6 +270,10 @@ audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval, int *defaultp)
*defaultp = 0;
return AUD_FMT_U16;
}
else if (!strcasecmp (s, "u32")) {
*defaultp = 0;
return AUD_FMT_U32;
}
else if (!strcasecmp (s, "s8")) {
*defaultp = 0;
return AUD_FMT_S8;
......@@ -251,6 +282,10 @@ audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval, int *defaultp)
*defaultp = 0;
return AUD_FMT_S16;
}
else if (!strcasecmp (s, "s32")) {
*defaultp = 0;
return AUD_FMT_S32;
}
else {
dolog ("Bogus audio format `%s' using %s\n",
s, audio_audfmt_to_string (defval));
......@@ -538,6 +573,8 @@ static int audio_validate_settings (audsettings_t *as)
case AUD_FMT_U8:
case AUD_FMT_S16:
case AUD_FMT_U16:
case AUD_FMT_S32:
case AUD_FMT_U32:
break;
default:
invalid = 1;
......@@ -563,6 +600,12 @@ static int audio_pcm_info_eq (struct audio_pcm_info *info, audsettings_t *as)
case AUD_FMT_U16:
bits = 16;
break;
case AUD_FMT_S32:
sign = 1;
case AUD_FMT_U32:
bits = 32;
break;
}
return info->freq == as->freq
&& info->nchannels == as->nchannels
......@@ -573,7 +616,7 @@ static int audio_pcm_info_eq (struct audio_pcm_info *info, audsettings_t *as)
void audio_pcm_init_info (struct audio_pcm_info *info, audsettings_t *as)
{
int bits = 8, sign = 0;
int bits = 8, sign = 0, shift = 0;
switch (as->fmt) {
case AUD_FMT_S8:
......@@ -585,6 +628,14 @@ void audio_pcm_init_info (struct audio_pcm_info *info, audsettings_t *as)
sign = 1;
case AUD_FMT_U16:
bits = 16;
shift = 1;
break;
case AUD_FMT_S32:
sign = 1;
case AUD_FMT_U32:
bits = 32;
shift = 2;
break;
}
......@@ -592,7 +643,7 @@ void audio_pcm_init_info (struct audio_pcm_info *info, audsettings_t *as)
info->bits = bits;
info->sign = sign;
info->nchannels = as->nchannels;
info->shift = (as->nchannels == 2) + (bits == 16);
info->shift = (as->nchannels == 2) + shift;
info->align = (1 << info->shift) - 1;
info->bytes_per_second = info->freq << info->shift;
info->swap_endianness = (as->endianness != AUDIO_HOST_ENDIANNESS);
......@@ -608,22 +659,49 @@ void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
memset (buf, 0x00, len << info->shift);
}
else {
if (info->bits == 8) {
switch (info->bits) {
case 8:
memset (buf, 0x80, len << info->shift);
}
else {
int i;
uint16_t *p = buf;
int shift = info->nchannels - 1;
short s = INT16_MAX;
break;
if (info->swap_endianness) {
s = bswap16 (s);
case 16:
{
int i;
uint16_t *p = buf;
int shift = info->nchannels - 1;
short s = INT16_MAX;
if (info->swap_endianness) {
s = bswap16 (s);
}
for (i = 0; i < len << shift; i++) {
p[i] = s;
}
}
break;
case 32:
{
int i;
uint32_t *p = buf;
int shift = info->nchannels - 1;
int32_t s = INT32_MAX;
if (info->swap_endianness) {
s = bswap32 (s);
}
for (i = 0; i < len << shift; i++) {
p[i] = s;
for (i = 0; i < len << shift; i++) {
p[i] = s;
}
}
break;
default:
AUD_log (NULL, "audio_pcm_info_clear_buf: invalid bits %d\n",
info->bits);
break;
}
}
}
......@@ -1811,7 +1889,7 @@ CaptureVoiceOut *AUD_add_capture (
[hw->info.nchannels == 2]
[hw->info.sign]
[hw->info.swap_endianness]
[hw->info.bits == 16];
[audio_bits_to_index (hw->info.bits)];
LIST_INSERT_HEAD (&s->cap_head, cap, entries);
LIST_INSERT_HEAD (&cap->cb_head, cb, entries);
......
......@@ -33,7 +33,9 @@ typedef enum {
AUD_FMT_U8,
AUD_FMT_S8,
AUD_FMT_U16,
AUD_FMT_S16
AUD_FMT_S16,
AUD_FMT_U32,
AUD_FMT_S32
} audfmt_e;
#ifdef WORDS_BIGENDIAN
......
......@@ -164,7 +164,7 @@ static int glue (audio_pcm_sw_init_, TYPE) (
[sw->info.nchannels == 2]
[sw->info.sign]
[sw->info.swap_endianness]
[sw->info.bits == 16];
[audio_bits_to_index (sw->info.bits)];
sw->name = qemu_strdup (name);
err = glue (audio_pcm_sw_alloc_resources_, TYPE) (sw);
......@@ -288,7 +288,7 @@ static HW *glue (audio_pcm_hw_add_new_, TYPE) (AudioState *s, audsettings_t *as)
[hw->info.nchannels == 2]
[hw->info.sign]
[hw->info.swap_endianness]
[hw->info.bits == 16];
[audio_bits_to_index (hw->info.bits)];
if (glue (audio_pcm_hw_alloc_resources_, TYPE) (hw)) {
goto err1;
......
......@@ -294,7 +294,6 @@ static int coreaudio_init_out (HWVoiceOut *hw, audsettings_t *as)
coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
UInt32 propertySize;
int err;
int bits = 8;
const char *typ = "playback";
AudioValueRange frameRange;
......@@ -305,10 +304,6 @@ static int coreaudio_init_out (HWVoiceOut *hw, audsettings_t *as)
return -1;
}
if (as->fmt == AUD_FMT_S16 || as->fmt == AUD_FMT_U16) {
bits = 16;
}
audio_pcm_init_info (&hw->info, as);
/* open default output device */
......
......@@ -82,6 +82,7 @@
#undef IN_T
#undef SHIFT
/* Unsigned 16 bit */
#define IN_T uint16_t
#define IN_MIN 0
#define IN_MAX USHRT_MAX
......@@ -101,26 +102,72 @@
#undef IN_T
#undef SHIFT
t_sample *mixeng_conv[2][2][2][2] = {
/* Signed 32 bit */
#define IN_T int32_t
#define IN_MIN INT32_MIN
#define IN_MAX INT32_MAX
#define SIGNED
#define SHIFT 32
#define ENDIAN_CONVERSION natural
#define ENDIAN_CONVERT(v) (v)
#include "mixeng_template.h"
#undef ENDIAN_CONVERT
#undef ENDIAN_CONVERSION
#define ENDIAN_CONVERSION swap
#define ENDIAN_CONVERT(v) bswap32 (v)
#include "mixeng_template.h"
#undef ENDIAN_CONVERT
#undef ENDIAN_CONVERSION
#undef SIGNED
#undef IN_MAX
#undef IN_MIN
#undef IN_T
#undef SHIFT
/* Unsigned 16 bit */
#define IN_T uint32_t
#define IN_MIN 0
#define IN_MAX UINT32_MAX
#define SHIFT 32
#define ENDIAN_CONVERSION natural
#define ENDIAN_CONVERT(v) (v)
#include "mixeng_template.h"
#undef ENDIAN_CONVERT
#undef ENDIAN_CONVERSION
#define ENDIAN_CONVERSION swap
#define ENDIAN_CONVERT(v) bswap32 (v)
#include "mixeng_template.h"
#undef ENDIAN_CONVERT
#undef ENDIAN_CONVERSION
#undef IN_MAX
#undef IN_MIN
#undef IN_T
#undef SHIFT
t_sample *mixeng_conv[2][2][2][3] = {
{
{
{
conv_natural_uint8_t_to_mono,
conv_natural_uint16_t_to_mono
conv_natural_uint16_t_to_mono,
conv_natural_uint32_t_to_mono
},
{
conv_natural_uint8_t_to_mono,
conv_swap_uint16_t_to_mono
conv_swap_uint16_t_to_mono,
conv_swap_uint32_t_to_mono,
}
},
{
{
conv_natural_int8_t_to_mono,
conv_natural_int16_t_to_mono
conv_natural_int16_t_to_mono,
conv_natural_int32_t_to_mono
},
{
conv_natural_int8_t_to_mono,
conv_swap_int16_t_to_mono
conv_swap_int16_t_to_mono,
conv_swap_int32_t_to_mono
}
}
},
......@@ -128,46 +175,54 @@ t_sample *mixeng_conv[2][2][2][2] = {
{
{
conv_natural_uint8_t_to_stereo,
conv_natural_uint16_t_to_stereo
conv_natural_uint16_t_to_stereo,
conv_natural_uint32_t_to_stereo
},
{
conv_natural_uint8_t_to_stereo,
conv_swap_uint16_t_to_stereo
conv_swap_uint16_t_to_stereo,
conv_swap_uint32_t_to_stereo
}
},
{
{
conv_natural_int8_t_to_stereo,
conv_natural_int16_t_to_stereo
conv_natural_int16_t_to_stereo,
conv_natural_int32_t_to_stereo
},
{
conv_natural_int8_t_to_stereo,
conv_swap_int16_t_to_stereo
conv_swap_int16_t_to_stereo,
conv_swap_int32_t_to_stereo,
}
}
}
};
f_sample *mixeng_clip[2][2][2][2] = {
f_sample *mixeng_clip[2][2][2][3] = {
{
{
{
clip_natural_uint8_t_from_mono,
clip_natural_uint16_t_from_mono
clip_natural_uint16_t_from_mono,
clip_natural_uint32_t_from_mono
},
{
clip_natural_uint8_t_from_mono,
clip_swap_uint16_t_from_mono
clip_swap_uint16_t_from_mono,
clip_swap_uint32_t_from_mono
}
},
{
{
clip_natural_int8_t_from_mono,
clip_natural_int16_t_from_mono
clip_natural_int16_t_from_mono,
clip_natural_int32_t_from_mono
},
{
clip_natural_int8_t_from_mono,
clip_swap_int16_t_from_mono
clip_swap_int16_t_from_mono,
clip_swap_int32_t_from_mono
}
}
},
......@@ -175,21 +230,25 @@ f_sample *mixeng_clip[2][2][2][2] = {
{
{
clip_natural_uint8_t_from_stereo,
clip_natural_uint16_t_from_stereo
clip_natural_uint16_t_from_stereo,
clip_natural_uint32_t_from_stereo
},
{
clip_natural_uint8_t_from_stereo,
clip_swap_uint16_t_from_stereo
clip_swap_uint16_t_from_stereo,
clip_swap_uint32_t_from_stereo
}
},
{
{
clip_natural_int8_t_from_stereo,
clip_natural_int16_t_from_stereo
clip_natural_int16_t_from_stereo,
clip_natural_int32_t_from_stereo
},
{
clip_natural_int8_t_from_stereo,
clip_swap_int16_t_from_stereo
clip_swap_int16_t_from_stereo,
clip_swap_int32_t_from_stereo
}
}
}
......
......@@ -37,8 +37,8 @@ typedef void (t_sample) (st_sample_t *dst, const void *src,
int samples, volume_t *vol);
typedef void (f_sample) (void *dst, const st_sample_t *src, int samples);
extern t_sample *mixeng_conv[2][2][2][2];
extern f_sample *mixeng_clip[2][2][2][2];
extern t_sample *mixeng_conv[2][2][2][3];
extern f_sample *mixeng_clip[2][2][2][3];
void *st_rate_start (int inrate, int outrate);
void st_rate_flow (void *opaque, st_sample_t *ibuf, st_sample_t *obuf,
......
......@@ -41,7 +41,8 @@ static struct {
{
44100,
2,
AUD_FMT_S16
AUD_FMT_S16,
AUDIO_HOST_ENDIANNESS
},
"qemu.wav"
};
......@@ -131,6 +132,11 @@ static int wav_init_out (HWVoiceOut *hw, audsettings_t *as)
case AUD_FMT_U16:
bits16 = 1;
break;
case AUD_FMT_S32:
case AUD_FMT_U32:
dolog ("WAVE files can not handle 32bit formats\n");
return -1;
}
hdr[34] = bits16 ? 0x10 : 0x08;
......
......@@ -37,15 +37,15 @@ static void wav_destroy (void *opaque)
if (wav->f) {
le_store (rlen, rifflen, 4);
le_store (dlen, datalen, 4);
qemu_fseek (wav->f, 4, SEEK_SET);
qemu_put_buffer (wav->f, rlen, 4);
qemu_fseek (wav->f, 32, SEEK_CUR);
qemu_put_buffer (wav->f, dlen, 4);
qemu_fclose (wav->f);
}
qemu_free (wav->path);
}
......
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