From: Marc-André Lureau <[email protected]>
The endianness field used an int to represent a boolean concept, with
0 meaning little-endian and 1 meaning big-endian. This required runtime
validation to reject invalid values and made the code less readable.
Replace with a bool big_endian field that is self-documenting and
type-safe. The compiler now enforces valid values, eliminating the
need for the validation check in audio_validate_settings().
Signed-off-by: Marc-André Lureau <[email protected]>
---
audio/dsound_template.h | 2 +-
include/qemu/audio.h | 2 +-
audio/alsaaudio.c | 8 ++++----
audio/audio-mixeng-be.c | 21 ++++-----------------
audio/jackaudio.c | 4 ++--
audio/ossaudio.c | 22 +++++++++++-----------
audio/paaudio.c | 22 +++++++++++-----------
audio/pwaudio.c | 30 +++++++++++++++---------------
audio/sdlaudio.c | 26 +++++++++++++-------------
audio/sndioaudio.c | 2 +-
audio/spiceaudio.c | 4 ++--
audio/wavaudio.c | 2 +-
audio/wavcapture.c | 2 +-
hw/audio/ac97.c | 2 +-
hw/audio/adlib.c | 2 +-
hw/audio/asc.c | 2 +-
hw/audio/cs4231a.c | 6 +++---
hw/audio/es1370.c | 2 +-
hw/audio/gus.c | 2 +-
hw/audio/lm4549.c | 6 +++---
hw/audio/sb16.c | 8 ++++----
hw/audio/via-ac97.c | 2 +-
hw/audio/virtio-snd.c | 2 +-
hw/audio/wm8750.c | 4 ++--
hw/display/xlnx_dp.c | 2 +-
hw/usb/dev-audio.c | 2 +-
tests/audio/test-audio.c | 2 +-
ui/vnc.c | 2 +-
28 files changed, 90 insertions(+), 103 deletions(-)
diff --git a/audio/dsound_template.h b/audio/dsound_template.h
index af4019bcb34..f9761120875 100644
--- a/audio/dsound_template.h
+++ b/audio/dsound_template.h
@@ -244,7 +244,7 @@ static int dsound_init_out(HWVoiceOut *hw, struct
audsettings *as)
}
ds->first_time = true;
- obt_as.endianness = 0;
+ obt_as.big_endian = false;
audio_pcm_init_info (&hw->info, &obt_as);
if (bc.dwBufferBytes % hw->info.bytes_per_frame) {
diff --git a/include/qemu/audio.h b/include/qemu/audio.h
index a2fbc286eb1..94d81a2a2ec 100644
--- a/include/qemu/audio.h
+++ b/include/qemu/audio.h
@@ -38,7 +38,7 @@ typedef struct audsettings {
int freq;
int nchannels;
AudioFormat fmt;
- int endianness;
+ bool big_endian;
} audsettings;
typedef struct SWVoiceOut SWVoiceOut;
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 3058af58e0b..a606ce0d4cd 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -666,7 +666,7 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings
*as)
struct audsettings obt_as;
Audiodev *dev = hw->s->dev;
- req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
+ req.fmt = aud_to_alsafmt (as->fmt, as->big_endian);
req.freq = as->freq;
req.nchannels = as->nchannels;
@@ -677,7 +677,7 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as)
obt_as.freq = obt.freq;
obt_as.nchannels = obt.nchannels;
obt_as.fmt = obt.fmt;
- obt_as.endianness = obt.endianness;
+ obt_as.big_endian = obt.endianness;
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = obt.samples;
@@ -752,7 +752,7 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings
*as)
struct audsettings obt_as;
Audiodev *dev = hw->s->dev;
- req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
+ req.fmt = aud_to_alsafmt (as->fmt, as->big_endian);
req.freq = as->freq;
req.nchannels = as->nchannels;
@@ -763,7 +763,7 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as)
obt_as.freq = obt.freq;
obt_as.nchannels = obt.nchannels;
obt_as.fmt = obt.fmt;
- obt_as.endianness = obt.endianness;
+ obt_as.big_endian = obt.endianness;
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = obt.samples;
diff --git a/audio/audio-mixeng-be.c b/audio/audio-mixeng-be.c
index 146026d0b39..e11c586c827 100644
--- a/audio/audio-mixeng-be.c
+++ b/audio/audio-mixeng-be.c
@@ -135,19 +135,7 @@ static void audio_print_settings (const struct audsettings
*as)
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");
+ AUD_log (NULL, " endianness=%s\n", as->big_endian ? "big" : "little");
}
static int audio_validate_settings (const struct audsettings *as)
@@ -155,7 +143,6 @@ static int audio_validate_settings (const struct
audsettings *as)
int invalid;
invalid = as->nchannels < 1;
- invalid |= as->endianness != 0 && as->endianness != 1;
switch (as->fmt) {
case AUDIO_FORMAT_S8:
@@ -180,7 +167,7 @@ static int audio_pcm_info_eq (struct audio_pcm_info *info,
const struct audsetti
return info->af == as->fmt
&& info->freq == as->freq
&& info->nchannels == as->nchannels
- && info->swap_endianness == (as->endianness != HOST_BIG_ENDIAN);
+ && info->swap_endianness == (as->big_endian != HOST_BIG_ENDIAN);
}
void audio_pcm_init_info (struct audio_pcm_info *info, const struct audsettings *as)
@@ -190,7 +177,7 @@ void audio_pcm_init_info (struct audio_pcm_info *info,
const struct audsettings
info->nchannels = as->nchannels;
info->bytes_per_frame = as->nchannels * audio_format_bits(as->fmt) / 8;
info->bytes_per_second = info->freq * info->bytes_per_frame;
- info->swap_endianness = (as->endianness != HOST_BIG_ENDIAN);
+ info->swap_endianness = (as->big_endian != HOST_BIG_ENDIAN);
}
void audio_pcm_info_clear_buf(struct audio_pcm_info *info, void *buf, int len)
@@ -1797,7 +1784,7 @@ audsettings
audiodev_to_audsettings(AudiodevPerDirectionOptions *pdo)
.freq = pdo->frequency,
.nchannels = pdo->channels,
.fmt = pdo->format,
- .endianness = HOST_BIG_ENDIAN,
+ .big_endian = HOST_BIG_ENDIAN,
};
}
diff --git a/audio/jackaudio.c b/audio/jackaudio.c
index 547672f6d36..58b9d2f04e2 100644
--- a/audio/jackaudio.c
+++ b/audio/jackaudio.c
@@ -531,7 +531,7 @@ static int qjack_init_out(HWVoiceOut *hw, struct
audsettings *as)
.freq = jo->c.freq,
.nchannels = jo->c.nchannels,
.fmt = AUDIO_FORMAT_F32,
- .endianness = 0
+ .big_endian = false
};
audio_pcm_init_info(&hw->info, &os);
@@ -566,7 +566,7 @@ static int qjack_init_in(HWVoiceIn *hw, struct audsettings *as)
.freq = ji->c.freq,
.nchannels = ji->c.nchannels,
.fmt = AUDIO_FORMAT_F32,
- .endianness = 0
+ .big_endian = false
};
audio_pcm_init_info(&hw->info, &is);
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 8388f81343b..0f1c097a666 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -160,36 +160,36 @@ static int aud_to_ossfmt(AudioFormat fmt, bool big_endian)
}
}
-static int oss_to_audfmt (int ossfmt, AudioFormat *fmt, int *endianness)
+static int oss_to_audfmt (int ossfmt, AudioFormat *fmt, bool *big_endian)
{
switch (ossfmt) {
case AFMT_S8:
- *endianness = 0;
+ *big_endian = false;
*fmt = AUDIO_FORMAT_S8;
break;
case AFMT_U8:
- *endianness = 0;
+ *big_endian = false;
*fmt = AUDIO_FORMAT_U8;
break;
case AFMT_S16_LE:
- *endianness = 0;
+ *big_endian = false;
*fmt = AUDIO_FORMAT_S16;
break;
case AFMT_U16_LE:
- *endianness = 0;
+ *big_endian = false;
*fmt = AUDIO_FORMAT_U16;
break;
case AFMT_S16_BE:
- *endianness = 1;
+ *big_endian = true;
*fmt = AUDIO_FORMAT_S16;
break;
case AFMT_U16_BE:
- *endianness = 1;
+ *big_endian = true;
*fmt = AUDIO_FORMAT_U16;
break;
@@ -475,7 +475,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as)
oss->fd = -1;
- req.fmt = aud_to_ossfmt (as->fmt, as->endianness);
+ req.fmt = aud_to_ossfmt (as->fmt, as->big_endian);
req.freq = as->freq;
req.nchannels = as->nchannels;
@@ -483,7 +483,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as)
return -1;
}
- err = oss_to_audfmt(obt.fmt, &obt_as.fmt, &obt_as.endianness);
+ err = oss_to_audfmt(obt.fmt, &obt_as.fmt, &obt_as.big_endian);
if (err) {
oss_anal_close (&fd);
return -1;
@@ -601,14 +601,14 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings
*as)
oss->fd = -1;
- req.fmt = aud_to_ossfmt (as->fmt, as->endianness);
+ req.fmt = aud_to_ossfmt (as->fmt, as->big_endian);
req.freq = as->freq;
req.nchannels = as->nchannels;
if (oss_open(1, &req, as, &obt, &fd, dev)) {
return -1;
}
- err = oss_to_audfmt(obt.fmt, &obt_as.fmt, &obt_as.endianness);
+ err = oss_to_audfmt(obt.fmt, &obt_as.fmt, &obt_as.big_endian);
if (err) {
oss_anal_close (&fd);
return -1;
diff --git a/audio/paaudio.c b/audio/paaudio.c
index bc6a8fa67b3..2a745ae38f9 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -352,28 +352,28 @@ static pa_sample_format_t audfmt_to_pa(AudioFormat afmt,
bool big_endian)
return format;
}
-static AudioFormat pa_to_audfmt (pa_sample_format_t fmt, int *endianness)
+static AudioFormat pa_to_audfmt (pa_sample_format_t fmt, bool *big_endian)
{
switch (fmt) {
case PA_SAMPLE_U8:
return AUDIO_FORMAT_U8;
case PA_SAMPLE_S16BE:
- *endianness = 1;
+ *big_endian = true;
return AUDIO_FORMAT_S16;
case PA_SAMPLE_S16LE:
- *endianness = 0;
+ *big_endian = false;
return AUDIO_FORMAT_S16;
case PA_SAMPLE_S32BE:
- *endianness = 1;
+ *big_endian = true;
return AUDIO_FORMAT_S32;
case PA_SAMPLE_S32LE:
- *endianness = 0;
+ *big_endian = false;
return AUDIO_FORMAT_S32;
case PA_SAMPLE_FLOAT32BE:
- *endianness = 1;
+ *big_endian = true;
return AUDIO_FORMAT_F32;
case PA_SAMPLE_FLOAT32LE:
- *endianness = 0;
+ *big_endian = false;
return AUDIO_FORMAT_F32;
default:
error_report("pulseaudio: Internal logic error: Bad pa_sample_format
%d", fmt);
@@ -531,7 +531,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings
*as)
PAConnection *c = apa->conn;
pa->g = apa;
- ss.format = audfmt_to_pa (as->fmt, as->endianness);
+ ss.format = audfmt_to_pa (as->fmt, as->big_endian);
ss.channels = as->nchannels;
ss.rate = as->freq;
@@ -541,7 +541,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as)
ba.maxlength = -1;
ba.prebuf = -1;
- obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);
+ obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.big_endian);
pa->stream = qpa_simple_new (
c,
@@ -582,7 +582,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings
*as)
PAConnection *c = apa->conn;
pa->g = apa;
- ss.format = audfmt_to_pa (as->fmt, as->endianness);
+ ss.format = audfmt_to_pa (as->fmt, as->big_endian);
ss.channels = as->nchannels;
ss.rate = as->freq;
@@ -592,7 +592,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as)
ba.minreq = -1;
ba.prebuf = -1;
- obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);
+ obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.big_endian);
pa->stream = qpa_simple_new (
c,
diff --git a/audio/pwaudio.c b/audio/pwaudio.c
index c30738c16ef..66ca1231541 100644
--- a/audio/pwaudio.c
+++ b/audio/pwaudio.c
@@ -365,7 +365,7 @@ audfmt_to_pw(AudioFormat fmt, bool big_endian)
}
static AudioFormat
-pw_to_audfmt(enum spa_audio_format fmt, int *endianness,
+pw_to_audfmt(enum spa_audio_format fmt, bool *big_endian,
uint32_t *sample_size)
{
switch (fmt) {
@@ -377,43 +377,43 @@ pw_to_audfmt(enum spa_audio_format fmt, int *endianness,
return AUDIO_FORMAT_U8;
case SPA_AUDIO_FORMAT_S16_BE:
*sample_size = 2;
- *endianness = 1;
+ *big_endian = true;
return AUDIO_FORMAT_S16;
case SPA_AUDIO_FORMAT_S16_LE:
*sample_size = 2;
- *endianness = 0;
+ *big_endian = false;
return AUDIO_FORMAT_S16;
case SPA_AUDIO_FORMAT_U16_BE:
*sample_size = 2;
- *endianness = 1;
+ *big_endian = true;
return AUDIO_FORMAT_U16;
case SPA_AUDIO_FORMAT_U16_LE:
*sample_size = 2;
- *endianness = 0;
+ *big_endian = false;
return AUDIO_FORMAT_U16;
case SPA_AUDIO_FORMAT_S32_BE:
*sample_size = 4;
- *endianness = 1;
+ *big_endian = true;
return AUDIO_FORMAT_S32;
case SPA_AUDIO_FORMAT_S32_LE:
*sample_size = 4;
- *endianness = 0;
+ *big_endian = false;
return AUDIO_FORMAT_S32;
case SPA_AUDIO_FORMAT_U32_BE:
*sample_size = 4;
- *endianness = 1;
+ *big_endian = true;
return AUDIO_FORMAT_U32;
case SPA_AUDIO_FORMAT_U32_LE:
*sample_size = 4;
- *endianness = 0;
+ *big_endian = false;
return AUDIO_FORMAT_U32;
case SPA_AUDIO_FORMAT_F32_BE:
*sample_size = 4;
- *endianness = 1;
+ *big_endian = true;
return AUDIO_FORMAT_F32;
case SPA_AUDIO_FORMAT_F32_LE:
*sample_size = 4;
- *endianness = 0;
+ *big_endian = false;
return AUDIO_FORMAT_F32;
default:
*sample_size = 1;
@@ -534,13 +534,13 @@ qpw_init_out(HWVoiceOut *hw, struct audsettings *as)
pw_thread_loop_lock(c->thread_loop);
- v->info.format = audfmt_to_pw(as->fmt, as->endianness);
+ v->info.format = audfmt_to_pw(as->fmt, as->big_endian);
v->info.channels = as->nchannels;
qpw_set_position(as->nchannels, v->info.position);
v->info.rate = as->freq;
obt_as.fmt =
- pw_to_audfmt(v->info.format, &obt_as.endianness, &v->frame_size);
+ pw_to_audfmt(v->info.format, &obt_as.big_endian, &v->frame_size);
v->frame_size *= as->nchannels;
v->req = (uint64_t)AUDIO_MIXENG_BACKEND(c)->dev->timer_period * v->info.rate
@@ -581,13 +581,13 @@ qpw_init_in(HWVoiceIn *hw, struct audsettings *as)
pw_thread_loop_lock(c->thread_loop);
- v->info.format = audfmt_to_pw(as->fmt, as->endianness);
+ v->info.format = audfmt_to_pw(as->fmt, as->big_endian);
v->info.channels = as->nchannels;
qpw_set_position(as->nchannels, v->info.position);
v->info.rate = as->freq;
obt_as.fmt =
- pw_to_audfmt(v->info.format, &obt_as.endianness, &v->frame_size);
+ pw_to_audfmt(v->info.format, &obt_as.big_endian, &v->frame_size);
v->frame_size *= as->nchannels;
/* call the function that creates a new stream for recording */
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index bb667ef9525..b404adbc1e9 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -107,56 +107,56 @@ static int aud_to_sdlfmt (AudioFormat fmt)
}
}
-static int sdl_to_audfmt(int sdlfmt, AudioFormat *fmt, int *endianness)
+static int sdl_to_audfmt(int sdlfmt, AudioFormat *fmt, bool *big_endian)
{
switch (sdlfmt) {
case AUDIO_S8:
- *endianness = 0;
+ *big_endian = false;
*fmt = AUDIO_FORMAT_S8;
break;
case AUDIO_U8:
- *endianness = 0;
+ *big_endian = false;
*fmt = AUDIO_FORMAT_U8;
break;
case AUDIO_S16LSB:
- *endianness = 0;
+ *big_endian = false;
*fmt = AUDIO_FORMAT_S16;
break;
case AUDIO_U16LSB:
- *endianness = 0;
+ *big_endian = false;
*fmt = AUDIO_FORMAT_U16;
break;
case AUDIO_S16MSB:
- *endianness = 1;
+ *big_endian = true;
*fmt = AUDIO_FORMAT_S16;
break;
case AUDIO_U16MSB:
- *endianness = 1;
+ *big_endian = true;
*fmt = AUDIO_FORMAT_U16;
break;
case AUDIO_S32LSB:
- *endianness = 0;
+ *big_endian = false;
*fmt = AUDIO_FORMAT_S32;
break;
case AUDIO_S32MSB:
- *endianness = 1;
+ *big_endian = true;
*fmt = AUDIO_FORMAT_S32;
break;
case AUDIO_F32LSB:
- *endianness = 0;
+ *big_endian = false;
*fmt = AUDIO_FORMAT_F32;
break;
case AUDIO_F32MSB:
- *endianness = 1;
+ *big_endian = true;
*fmt = AUDIO_FORMAT_F32;
break;
@@ -361,7 +361,7 @@ static int sdl_init_out(HWVoiceOut *hw, struct audsettings *as)
return -1;
}
- err = sdl_to_audfmt(obt.format, &obt_as.fmt, &obt_as.endianness);
+ err = sdl_to_audfmt(obt.format, &obt_as.fmt, &obt_as.big_endian);
if (err) {
sdl_close_out(sdl);
return -1;
@@ -417,7 +417,7 @@ static int sdl_init_in(HWVoiceIn *hw, audsettings *as)
return -1;
}
- err = sdl_to_audfmt(obt.format, &obt_as.fmt, &obt_as.endianness);
+ err = sdl_to_audfmt(obt.format, &obt_as.fmt, &obt_as.big_endian);
if (err) {
sdl_close_in(sdl);
return -1;
diff --git a/audio/sndioaudio.c b/audio/sndioaudio.c
index be491895532..69fa9d8ba54 100644
--- a/audio/sndioaudio.c
+++ b/audio/sndioaudio.c
@@ -387,7 +387,7 @@ static int sndio_init(SndioVoice *self,
}
if (req.bits > 8) {
- req.le = as->endianness ? 0 : 1;
+ req.le = as->big_endian ? 0 : 1;
}
req.rate = as->freq;
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index 2547126994d..1b67fe2a540 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -106,7 +106,7 @@ static int line_out_init(HWVoiceOut *hw, struct audsettings
*as)
#endif
settings.nchannels = SPICE_INTERFACE_PLAYBACK_CHAN;
settings.fmt = AUDIO_FORMAT_S16;
- settings.endianness = HOST_BIG_ENDIAN;
+ settings.big_endian = HOST_BIG_ENDIAN;
audio_pcm_init_info (&hw->info, &settings);
hw->samples = LINE_OUT_SAMPLES;
@@ -222,7 +222,7 @@ static int line_in_init(HWVoiceIn *hw, struct audsettings
*as)
#endif
settings.nchannels = SPICE_INTERFACE_RECORD_CHAN;
settings.fmt = AUDIO_FORMAT_S16;
- settings.endianness = HOST_BIG_ENDIAN;
+ settings.big_endian = HOST_BIG_ENDIAN;
audio_pcm_init_info (&hw->info, &settings);
hw->samples = LINE_IN_SAMPLES;
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 5c614ed3b51..217fb9937f8 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -111,7 +111,7 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings
*as)
hdr[34] = bits16 ? 0x10 : 0x08;
- wav_as.endianness = 0;
+ wav_as.big_endian = false;
audio_pcm_init_info (&hw->info, &wav_as);
hw->samples = 1024;
diff --git a/audio/wavcapture.c b/audio/wavcapture.c
index 69aa91e35f6..2dac9461710 100644
--- a/audio/wavcapture.c
+++ b/audio/wavcapture.c
@@ -137,7 +137,7 @@ int wav_start_capture(AudioBackend *state, CaptureState *s,
const char *path,
as.freq = freq;
as.nchannels = 1 << stereo;
as.fmt = bits16 ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_U8;
- as.endianness = 0;
+ as.big_endian = false;
ops.notify = wav_notify;
ops.capture = wav_capture;
diff --git a/hw/audio/ac97.c b/hw/audio/ac97.c
index 5d442b7e067..fd74c249949 100644
--- a/hw/audio/ac97.c
+++ b/hw/audio/ac97.c
@@ -313,7 +313,7 @@ static void open_voice(AC97LinkState *s, int index, int
freq)
as.freq = freq;
as.nchannels = 2;
as.fmt = AUDIO_FORMAT_S16;
- as.endianness = 0;
+ as.big_endian = false;
if (freq > 0) {
s->invalid_freq[index] = 0;
diff --git a/hw/audio/adlib.c b/hw/audio/adlib.c
index ce17e21d5fd..52ee5cb6256 100644
--- a/hw/audio/adlib.c
+++ b/hw/audio/adlib.c
@@ -254,7 +254,7 @@ static void adlib_realizefn (DeviceState *dev, Error **errp)
as.freq = s->freq;
as.nchannels = SHIFT;
as.fmt = AUDIO_FORMAT_S16;
- as.endianness = HOST_BIG_ENDIAN;
+ as.big_endian = HOST_BIG_ENDIAN;
s->voice = audio_be_open_out(
s->audio_be,
diff --git a/hw/audio/asc.c b/hw/audio/asc.c
index 35c7b5750d6..ea59bdde7b8 100644
--- a/hw/audio/asc.c
+++ b/hw/audio/asc.c
@@ -648,7 +648,7 @@ static void asc_realize(DeviceState *dev, Error **errp)
as.freq = ASC_FREQ;
as.nchannels = 2;
as.fmt = AUDIO_FORMAT_U8;
- as.endianness = HOST_BIG_ENDIAN;
+ as.big_endian = HOST_BIG_ENDIAN;
s->voice = audio_be_open_out(s->audio_be, s->voice, "asc.out", s, asc_out_cb,
&as);
diff --git a/hw/audio/cs4231a.c b/hw/audio/cs4231a.c
index e6cae9c988e..c589670e855 100644
--- a/hw/audio/cs4231a.c
+++ b/hw/audio/cs4231a.c
@@ -289,7 +289,7 @@ static void cs_reset_voices (CSState *s, uint32_t val)
}
as.nchannels = (val & (1 << 4)) ? 2 : 1;
- as.endianness = 0;
+ as.big_endian = false;
s->tab = NULL;
switch ((val >> 5) & ((s->dregs[MODE_And_ID] & MODE2) ? 7 : 3)) {
@@ -305,12 +305,12 @@ static void cs_reset_voices (CSState *s, uint32_t val)
s->tab = ALawDecompressTable;
x_law:
as.fmt = AUDIO_FORMAT_S16;
- as.endianness = HOST_BIG_ENDIAN;
+ as.big_endian = HOST_BIG_ENDIAN;
s->shift = as.nchannels == 2;
break;
case 6:
- as.endianness = 1;
+ as.big_endian = true;
/* fall through */
case 2:
as.fmt = AUDIO_FORMAT_S16;
diff --git a/hw/audio/es1370.c b/hw/audio/es1370.c
index e1658393c6a..ca7ad16df49 100644
--- a/hw/audio/es1370.c
+++ b/hw/audio/es1370.c
@@ -407,7 +407,7 @@ static void es1370_update_voices (ES1370State *s, uint32_t
ctl, uint32_t sctl)
as.freq = new_freq;
as.nchannels = 1 << (new_fmt & 1);
as.fmt = (new_fmt & 2) ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_U8;
- as.endianness = 0;
+ as.big_endian = false;
if (i == ADC_CHANNEL) {
s->adc_voice =
diff --git a/hw/audio/gus.c b/hw/audio/gus.c
index 5c2a34c09d2..196c4f72205 100644
--- a/hw/audio/gus.c
+++ b/hw/audio/gus.c
@@ -256,7 +256,7 @@ static void gus_realizefn (DeviceState *dev, Error **errp)
as.freq = s->freq;
as.nchannels = 2;
as.fmt = AUDIO_FORMAT_S16;
- as.endianness = HOST_BIG_ENDIAN;
+ as.big_endian = HOST_BIG_ENDIAN;
s->voice = audio_be_open_out(
s->audio_be,
diff --git a/hw/audio/lm4549.c b/hw/audio/lm4549.c
index 14e15a844ba..a891e975106 100644
--- a/hw/audio/lm4549.c
+++ b/hw/audio/lm4549.c
@@ -202,7 +202,7 @@ void lm4549_write(lm4549_state *s,
as.freq = value;
as.nchannels = 2;
as.fmt = AUDIO_FORMAT_S16;
- as.endianness = 0;
+ as.big_endian = false;
s->voice = audio_be_open_out(
s->audio_be,
@@ -272,7 +272,7 @@ static int lm4549_post_load(void *opaque, int version_id)
as.freq = freq;
as.nchannels = 2;
as.fmt = AUDIO_FORMAT_S16;
- as.endianness = 0;
+ as.big_endian = false;
s->voice = audio_be_open_out(
s->audio_be,
@@ -312,7 +312,7 @@ void lm4549_init(lm4549_state *s, lm4549_callback
data_req_cb, void* opaque,
as.freq = 48000;
as.nchannels = 2;
as.fmt = AUDIO_FORMAT_S16;
- as.endianness = 0;
+ as.big_endian = false;
s->voice = audio_be_open_out(
s->audio_be,
diff --git a/hw/audio/sb16.c b/hw/audio/sb16.c
index c8fc7df8b44..1b5e452a29b 100644
--- a/hw/audio/sb16.c
+++ b/hw/audio/sb16.c
@@ -213,7 +213,7 @@ static void continue_dma8 (SB16State *s)
as.freq = s->freq;
as.nchannels = 1 << s->fmt_stereo;
as.fmt = s->fmt;
- as.endianness = 0;
+ as.big_endian = false;
s->voice = audio_be_open_out(
s->audio_be,
@@ -376,7 +376,7 @@ static void dma_cmd (SB16State *s, uint8_t cmd, uint8_t d0,
int dma_len)
as.freq = s->freq;
as.nchannels = 1 << s->fmt_stereo;
as.fmt = s->fmt;
- as.endianness = 0;
+ as.big_endian = false;
s->voice = audio_be_open_out(
s->audio_be,
@@ -877,7 +877,7 @@ static void legacy_reset (SB16State *s)
as.freq = s->freq;
as.nchannels = 1;
as.fmt = AUDIO_FORMAT_U8;
- as.endianness = 0;
+ as.big_endian = false;
s->voice = audio_be_open_out(
s->audio_be,
@@ -1300,7 +1300,7 @@ static int sb16_post_load (void *opaque, int version_id)
as.freq = s->freq;
as.nchannels = 1 << s->fmt_stereo;
as.fmt = s->fmt;
- as.endianness = 0;
+ as.big_endian = false;
s->voice = audio_be_open_out(
s->audio_be,
diff --git a/hw/audio/via-ac97.c b/hw/audio/via-ac97.c
index 84d137b41a3..9d61283542a 100644
--- a/hw/audio/via-ac97.c
+++ b/hw/audio/via-ac97.c
@@ -237,7 +237,7 @@ static void open_voice_out(ViaAC97State *s)
.freq = CODEC_REG(s, AC97_PCM_Front_DAC_Rate),
.nchannels = s->aur.type & BIT(4) ? 2 : 1,
.fmt = s->aur.type & BIT(5) ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_S8,
- .endianness = 0,
+ .big_endian = false,
};
s->vo = audio_be_open_out(s->audio_be, s->vo, "via-ac97.out", s, out_cb,
&as);
}
diff --git a/hw/audio/virtio-snd.c b/hw/audio/virtio-snd.c
index 89e24c0a8e0..8b949146468 100644
--- a/hw/audio/virtio-snd.c
+++ b/hw/audio/virtio-snd.c
@@ -378,7 +378,7 @@ static void virtio_snd_get_qemu_audsettings(audsettings *as,
as->nchannels = MIN(AUDIO_MAX_CHANNELS, params->channels);
as->fmt = virtio_snd_get_qemu_format(params->format);
as->freq = virtio_snd_get_qemu_freq(params->rate);
- as->endianness = 0; /* Conforming to VIRTIO 1.0: always little endian. */
+ as->big_endian = false; /* Conforming to VIRTIO 1.0: always little endian.
*/
}
/*
diff --git a/hw/audio/wm8750.c b/hw/audio/wm8750.c
index 2a286515b14..e2507b0269a 100644
--- a/hw/audio/wm8750.c
+++ b/hw/audio/wm8750.c
@@ -202,7 +202,7 @@ static void wm8750_set_format(WM8750State *s)
return;
/* Setup input */
- in_fmt.endianness = 0;
+ in_fmt.big_endian = false;
in_fmt.nchannels = 2;
in_fmt.freq = s->adc_hz;
in_fmt.fmt = AUDIO_FORMAT_S16;
@@ -215,7 +215,7 @@ static void wm8750_set_format(WM8750State *s)
CODEC ".input3", s, wm8750_audio_in_cb, &in_fmt);
/* Setup output */
- out_fmt.endianness = 0;
+ out_fmt.big_endian = false;
out_fmt.nchannels = 2;
out_fmt.freq = s->dac_hz;
out_fmt.fmt = AUDIO_FORMAT_S16;
diff --git a/hw/display/xlnx_dp.c b/hw/display/xlnx_dp.c
index 9aa4709b411..7d037b46a35 100644
--- a/hw/display/xlnx_dp.c
+++ b/hw/display/xlnx_dp.c
@@ -1393,7 +1393,7 @@ static void xlnx_dp_realize(DeviceState *dev, Error
**errp)
as.freq = 44100;
as.nchannels = 2;
as.fmt = AUDIO_FORMAT_S16;
- as.endianness = 0;
+ as.big_endian = false;
s->amixer_output_stream = audio_be_open_out(s->audio_be,
s->amixer_output_stream,
diff --git a/hw/usb/dev-audio.c b/hw/usb/dev-audio.c
index 7b758718c12..e18e0a1dfd6 100644
--- a/hw/usb/dev-audio.c
+++ b/hw/usb/dev-audio.c
@@ -975,7 +975,7 @@ static void usb_audio_reinit(USBDevice *dev, unsigned
channels)
s->out.as.freq = USBAUDIO_SAMPLE_RATE;
s->out.as.nchannels = s->out.channels;
s->out.as.fmt = AUDIO_FORMAT_S16;
- s->out.as.endianness = 0;
+ s->out.as.big_endian = false;
streambuf_init(&s->out.buf, s->buffer, s->out.channels);
s->out.voice = audio_be_open_out(s->audio_be, s->out.voice, TYPE_USB_AUDIO,
diff --git a/tests/audio/test-audio.c b/tests/audio/test-audio.c
index e403f11f093..98e77cf542b 100644
--- a/tests/audio/test-audio.c
+++ b/tests/audio/test-audio.c
@@ -49,7 +49,7 @@ static const struct audsettings default_test_settings = {
.freq = SAMPLE_RATE,
.nchannels = CHANNELS,
.fmt = AUDIO_FORMAT_S16,
- .endianness = 0,
+ .big_endian = false,
};
static void dummy_audio_callback(void *opaque, int avail)
diff --git a/ui/vnc.c b/ui/vnc.c
index d56fe2c180e..daf5b01d342 100644
--- a/ui/vnc.c
+++ b/ui/vnc.c
@@ -3372,7 +3372,7 @@ static void vnc_connect(VncDisplay *vd, QIOChannelSocket
*sioc,
vs->as.freq = 44100;
vs->as.nchannels = 2;
vs->as.fmt = AUDIO_FORMAT_S16;
- vs->as.endianness = 0;
+ vs->as.big_endian = false;
qemu_mutex_init(&vs->output_mutex);
vs->bh = qemu_bh_new(vnc_jobs_bh, vs);
