obt_as.freq = obt.freq;
obt_as.nchannels = obt.nchannels;
obt_as.fmt = effective_fmt;
+ obt_as.endianness = endianness;
- audio_pcm_init_info (
- &hw->info,
- &obt_as,
- audio_need_to_swap_endian (endianness)
- );
+ audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = obt.samples;
alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
obt_as.freq = obt.freq;
obt_as.nchannels = obt.nchannels;
obt_as.fmt = effective_fmt;
+ obt_as.endianness = endianness;
- audio_pcm_init_info (
- &hw->info,
- &obt_as,
- audio_need_to_swap_endian (endianness)
- );
+ audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = obt.samples;
alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
AUD_log (NULL, "invalid(%d)", as->fmt);
break;
}
+ AUD_log (NULL, "endianness=");
+ switch (as->endianness) {
+ case 0:
+ AUD_log (NULL, "little");
+ break;
+ case 1:
+ AUD_log (NULL, "big");
+ break;
+ default:
+ AUD_log (NULL, "invalid");
+ break;
+ }
AUD_log (NULL, "\n");
}
int invalid;
invalid = as->nchannels != 1 && as->nchannels != 2;
+ invalid |= as->endianness != 0 && as->endianness != 1;
switch (as->fmt) {
case AUD_FMT_S8:
}
invalid |= as->freq <= 0;
-
- if (invalid) {
- return -1;
- }
- return 0;
+ return invalid ? -1 : 0;
}
static int audio_pcm_info_eq (struct audio_pcm_info *info, audsettings_t *as)
return info->freq == as->freq
&& info->nchannels == as->nchannels
&& info->sign == sign
- && info->bits == bits;
+ && info->bits == bits
+ && info->swap_endianness == (as->endianness != AUDIO_HOST_ENDIANNESS);
}
-void audio_pcm_init_info (
- struct audio_pcm_info *info,
- audsettings_t *as,
- int swap_endian
- )
+void audio_pcm_init_info (struct audio_pcm_info *info, audsettings_t *as)
{
int bits = 8, sign = 0;
info->shift = (as->nchannels == 2) + (bits == 16);
info->align = (1 << info->shift) - 1;
info->bytes_per_second = info->freq << info->shift;
- info->swap_endian = swap_endian;
+ info->swap_endianness = (as->endianness != AUDIO_HOST_ENDIANNESS);
}
void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
int shift = info->nchannels - 1;
short s = INT16_MAX;
- if (info->swap_endian) {
+ if (info->swap_endianness) {
s = bswap16 (s);
}
static CaptureVoiceOut *audio_pcm_capture_find_specific (
AudioState *s,
- audsettings_t *as,
- int endian
+ audsettings_t *as
)
{
CaptureVoiceOut *cap;
- int swap_endian = audio_need_to_swap_endian (endian);
for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
- if ((cap->hw.info.swap_endian == swap_endian)
- && audio_pcm_info_eq (&cap->hw.info, as)) {
+ if (audio_pcm_info_eq (&cap->hw.info, as)) {
return cap;
}
}
int AUD_add_capture (
AudioState *s,
audsettings_t *as,
- int endian,
struct audio_capture_ops *ops,
void *cb_opaque
)
cb->ops = *ops;
cb->opaque = cb_opaque;
- cap = audio_pcm_capture_find_specific (s, as, endian);
+ cap = audio_pcm_capture_find_specific (s, as);
if (cap) {
LIST_INSERT_HEAD (&cap->cb_head, cb, entries);
return 0;
goto err2;
}
- audio_pcm_init_info (&hw->info, as, endian);
+ audio_pcm_init_info (&hw->info, as);
cap->buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
if (!cap->buf) {
hw->clip = mixeng_clip
[hw->info.nchannels == 2]
[hw->info.sign]
- [hw->info.swap_endian]
+ [hw->info.swap_endianness]
[hw->info.bits == 16];
LIST_INSERT_HEAD (&s->cap_head, cap, entries);
#ifndef QEMU_AUDIO_H
#define QEMU_AUDIO_H
+#include "config.h"
#include "sys-queue.h"
typedef void (*audio_callback_fn_t) (void *opaque, int avail);
AUD_FMT_S16
} audfmt_e;
+#ifdef WORDS_BIGENDIAN
+#define AUDIO_HOST_ENDIANNESS 1
+#else
+#define AUDIO_HOST_ENDIANNESS 0
+#endif
+
typedef struct {
int freq;
int nchannels;
audfmt_e fmt;
+ int endianness;
} audsettings_t;
struct audio_capture_ops {
int AUD_add_capture (
AudioState *s,
audsettings_t *as,
- int endian,
struct audio_capture_ops *ops,
void *opaque
);
const char *name,
void *callback_opaque,
audio_callback_fn_t callback_fn,
- audsettings_t *settings,
- int sw_endian
+ audsettings_t *settings
);
void AUD_close_out (QEMUSoundCard *card, SWVoiceOut *sw);
const char *name,
void *callback_opaque,
audio_callback_fn_t callback_fn,
- audsettings_t *settings,
- int sw_endian
+ audsettings_t *settings
);
void AUD_close_in (QEMUSoundCard *card, SWVoiceIn *sw);
int align;
int shift;
int bytes_per_second;
- int swap_endian;
+ int swap_endianness;
};
typedef struct HWVoiceOut {
extern struct audio_driver dsound_audio_driver;
extern volume_t nominal_volume;
-void audio_pcm_init_info (struct audio_pcm_info *info, audsettings_t *as,
- int swap_endian);
+void audio_pcm_init_info (struct audio_pcm_info *info, audsettings_t *as);
void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len);
int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int len);
return (dst >= src) ? (dst - src) : (len - src + dst);
}
-static inline int audio_need_to_swap_endian (int endianness)
-{
-#ifdef WORDS_BIGENDIAN
- return endianness != 1;
-#else
- return endianness != 0;
-#endif
-}
-
#if defined __GNUC__
#define GCC_ATTR __attribute__ ((__unused__, __format__ (__printf__, 1, 2)))
#define INIT_FIELD(f) . f
SW *sw,
HW *hw,
const char *name,
- audsettings_t *as,
- int endian
+ audsettings_t *as
)
{
int err;
- audio_pcm_init_info (&sw->info, as, audio_need_to_swap_endian (endian));
+ audio_pcm_init_info (&sw->info, as);
sw->hw = hw;
sw->active = 0;
#ifdef DAC
#endif
[sw->info.nchannels == 2]
[sw->info.sign]
- [sw->info.swap_endian]
+ [sw->info.swap_endianness]
[sw->info.bits == 16];
sw->name = qemu_strdup (name);
#endif
[hw->info.nchannels == 2]
[hw->info.sign]
- [hw->info.swap_endian]
+ [hw->info.swap_endianness]
[hw->info.bits == 16];
if (glue (audio_pcm_hw_alloc_resources_, TYPE) (hw)) {
static SW *glue (audio_pcm_create_voice_pair_, TYPE) (
AudioState *s,
const char *sw_name,
- audsettings_t *as,
- int sw_endian
+ audsettings_t *as
)
{
SW *sw;
glue (audio_pcm_hw_add_sw_, TYPE) (hw, sw);
- if (glue (audio_pcm_sw_init_, TYPE) (sw, hw, sw_name, as, sw_endian)) {
+ if (glue (audio_pcm_sw_init_, TYPE) (sw, hw, sw_name, as)) {
goto err3;
}
const char *name,
void *callback_opaque ,
audio_callback_fn_t callback_fn,
- audsettings_t *as,
- int sw_endian
+ audsettings_t *as
)
{
AudioState *s;
}
glue (audio_pcm_sw_fini_, TYPE) (sw);
- if (glue (audio_pcm_sw_init_, TYPE) (sw, hw, name, as, sw_endian)) {
+ if (glue (audio_pcm_sw_init_, TYPE) (sw, hw, name, as)) {
goto fail;
}
}
else {
- sw = glue (audio_pcm_create_voice_pair_, TYPE) (s, name, as, sw_endian);
+ sw = glue (audio_pcm_create_voice_pair_, TYPE) (s, name, as);
if (!sw) {
dolog ("Failed to create voice `%s'\n", name);
return NULL;
UInt32 propertySize;
int err;
int bits = 8;
- int endianess = 0;
const char *typ = "playback";
AudioValueRange frameRange;
if (as->fmt == AUD_FMT_S16 || as->fmt == AUD_FMT_U16) {
bits = 16;
- endianess = 1;
}
- audio_pcm_init_info (
- &hw->info,
- as,
- /* Following is irrelevant actually since we do not use
- mixengs clipping routines */
- audio_need_to_swap_endian (endianess)
- );
+ audio_pcm_init_info (&hw->info, as);
/* open default output device */
propertySize = sizeof(core->outputDeviceID);
}
ds->first_time = 1;
-
- audio_pcm_init_info (&hw->info, &obt_as, audio_need_to_swap_endian (0));
+ obt_as.endianness = 0;
+ audio_pcm_init_info (&hw->info, &obt_as);
if (bc.dwBufferBytes & hw->info.align) {
dolog (
{
int bits16, mode, channel;
FMODVoiceOut *fmd = (FMODVoiceOut *) hw;
+ audsettings_t obt_as = *as;
mode = aud_to_fmodfmt (as->fmt, as->nchannels == 2 ? 1 : 0);
fmd->fmod_sample = FSOUND_Sample_Alloc (
fmd->channel = channel;
/* FMOD always operates on little endian frames? */
- audio_pcm_init_info (&hw->info, as, audio_need_to_swap_endian (0));
+ obt_as.endianness = 0;
+ audio_pcm_init_info (&hw->info, &obt_as);
bits16 = (mode & FSOUND_16BITS) != 0;
hw->samples = conf.nb_samples;
return 0;
{
int bits16, mode;
FMODVoiceIn *fmd = (FMODVoiceIn *) hw;
+ audsettings_t obt_as = *as;
if (conf.broken_adc) {
return -1;
}
/* FMOD always operates on little endian frames? */
- audio_pcm_init_info (&hw->info, as, audio_need_to_swap_endian (0));
+ obt_as.endianness = 0;
+ audio_pcm_init_info (&hw->info, &obt_as);
bits16 = (mode & FSOUND_16BITS) != 0;
hw->samples = conf.nb_samples;
return 0;
static int no_init_out (HWVoiceOut *hw, audsettings_t *as)
{
- audio_pcm_init_info (&hw->info, as, 0);
+ audio_pcm_init_info (&hw->info, as);
hw->samples = 1024;
return 0;
}
static int no_init_in (HWVoiceIn *hw, audsettings_t *as)
{
- audio_pcm_init_info (&hw->info, as, 0);
+ audio_pcm_init_info (&hw->info, as);
hw->samples = 1024;
return 0;
}
obt_as.freq = obt.freq;
obt_as.nchannels = obt.nchannels;
obt_as.fmt = effective_fmt;
+ obt_as.endianness = endianness;
- audio_pcm_init_info (
- &hw->info,
- &obt_as,
- audio_need_to_swap_endian (endianness)
- );
+ audio_pcm_init_info (&hw->info, &obt_as);
oss->nfrags = obt.nfrags;
oss->fragsize = obt.fragsize;
obt_as.freq = obt.freq;
obt_as.nchannels = obt.nchannels;
obt_as.fmt = effective_fmt;
+ obt_as.endianness = endianness;
- audio_pcm_init_info (
- &hw->info,
- &obt_as,
- audio_need_to_swap_endian (endianness)
- );
+ audio_pcm_init_info (&hw->info, &obt_as);
oss->nfrags = obt.nfrags;
oss->fragsize = obt.fragsize;
obt_as.freq = obt.freq;
obt_as.nchannels = obt.channels;
obt_as.fmt = effective_fmt;
+ obt_as.endianness = endianess;
- audio_pcm_init_info (
- &hw->info,
- &obt_as,
- audio_need_to_swap_endian (endianess)
- );
+ audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = obt.samples;
s->initialized = 1;
hdr[34] = bits16 ? 0x10 : 0x08;
- audio_pcm_init_info (&hw->info, &wav_as, audio_need_to_swap_endian (0));
+ wav_as.endianness = 0;
+ audio_pcm_init_info (&hw->info, &wav_as);
hw->samples = 1024;
wav->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
as.freq = freq;
as.nchannels = 1 << stereo;
as.fmt = bits16 ? AUD_FMT_S16 : AUD_FMT_U8;
+ as.endianness = 0;
ops.state = wav_state_cb;
ops.capture = wav_capture_cb;
}
qemu_put_buffer (wav->f, hdr, sizeof (hdr));
- AUD_add_capture (NULL, &as, 0, &ops, wav);
+ AUD_add_capture (NULL, &as, &ops, wav);
}
as.freq = conf.freq;
as.nchannels = SHIFT;
as.fmt = AUD_FMT_S16;
+ as.endianness = AUDIO_HOST_ENDIANNESS;
AUD_register_card (audio, "adlib", &s->card);
"adlib",
s,
adlib_callback,
- &as,
- 0 /* XXX: little endian? */
+ &as
);
if (!s->voice) {
Adlib_fini (s);
as.freq = new_freq;
as.nchannels = 1 << (new_fmt & 1);
as.fmt = (new_fmt & 2) ? AUD_FMT_S16 : AUD_FMT_U8;
+ as.endianness = 0;
if (i == ADC_CHANNEL) {
s->adc_voice =
"es1370.adc",
s,
es1370_adc_callback,
- &as,
- 0 /* little endian */
+ &as
);
}
else {
i ? "es1370.dac2" : "es1370.dac1",
s,
i ? es1370_dac2_callback : es1370_dac1_callback,
- &as,
- 0 /* litle endian */
+ &as
);
}
}
int pcspk_audio_init(AudioState *audio)
{
PCSpkState *s = &pcspk_state;
- audsettings_t as = {PCSPK_SAMPLE_RATE, 1, AUD_FMT_U8};
+ audsettings_t as = {PCSPK_SAMPLE_RATE, 1, AUD_FMT_U8, 0};
if (!audio) {
AUD_log(s_spk, "No audio state\n");
}
AUD_register_card(audio, s_spk, &s->card);
- s->voice = AUD_open_out(&s->card, s->voice, s_spk, s, pcspk_callback, &as, 0);
+ s->voice = AUD_open_out(&s->card, s->voice, s_spk, s, pcspk_callback, &as);
if (!s->voice) {
AUD_log(s_spk, "Could not open voice\n");
return -1;
as.freq = s->freq;
as.nchannels = 1 << s->fmt_stereo;
as.fmt = s->fmt;
+ as.endianness = 0;
s->voice = AUD_open_out (
&s->card,
"sb16",
s,
SB_audio_callback,
- &as,
- 0 /* little endian */
+ &as
);
}
as.freq = s->freq;
as.nchannels = 1 << s->fmt_stereo;
as.fmt = s->fmt;
+ as.endianness = 0;
s->voice = AUD_open_out (
&s->card,
"sb16",
s,
SB_audio_callback,
- &as,
- 0 /* little endian */
+ &as
);
}
as.freq = s->freq;
as.nchannels = 1;
as.fmt = AUD_FMT_U8;
+ as.endianness = 0;
s->voice = AUD_open_out (
&s->card,
"sb16",
s,
SB_audio_callback,
- &as,
- 0 /* little endian */
+ &as
);
/* Not sure about that... */
as.freq = s->freq;
as.nchannels = 1 << s->fmt_stereo;
as.fmt = s->fmt;
+ as.endianness = 0;
s->voice = AUD_open_out (
&s->card,
"sb16",
s,
SB_audio_callback,
- &as,
- 0 /* little endian */
+ &as
);
}