From: Kővágó, Zoltán <dirty.ice.hu@xxxxxxxxx>
I had to include an enum for audio sampling formats into qapi, but that
meant duplicating the audfmt_e enum. This patch replaces audfmt_e and
associated values with the qapi generated AudioFormat enum.
This patch is mostly a search-and-replace, except for switches where the
qapi generated AUDIO_FORMAT_MAX caused problems.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@xxxxxxxxx>
Reviewed-by: Thomas Huth <thuth@xxxxxxxxxx>
Message-id: 01251b2758a1679c66842120b77c0fb46d7d0eaf.1552083282.git.DirtY.iCE.hu@xxxxxxxxx
Signed-off-by: Gerd Hoffmann <kraxel@xxxxxxxxxx>
---
audio/audio.h | 12 +----
audio/alsaaudio.c | 53 +++++++++++----------
audio/audio.c | 97 +++++++++++++++++++++------------------
audio/audio_win_int.c | 18 ++++----
audio/ossaudio.c | 30 ++++++------
audio/paaudio.c | 28 +++++------
audio/sdlaudio.c | 26 +++++------
audio/spiceaudio.c | 4 +-
audio/wavaudio.c | 17 ++++---
audio/wavcapture.c | 2 +-
hw/arm/omap2.c | 2 +-
hw/audio/ac97.c | 2 +-
hw/audio/adlib.c | 2 +-
hw/audio/cs4231a.c | 6 +--
hw/audio/es1370.c | 4 +-
hw/audio/gus.c | 2 +-
hw/audio/hda-codec.c | 18 ++++----
hw/audio/lm4549.c | 6 +--
hw/audio/milkymist-ac97.c | 2 +-
hw/audio/pcspk.c | 2 +-
hw/audio/sb16.c | 14 +++---
hw/audio/wm8750.c | 6 +--
hw/display/xlnx_dp.c | 2 +-
hw/input/tsc210x.c | 2 +-
hw/usb/dev-audio.c | 2 +-
ui/vnc.c | 26 +++++------
26 files changed, 196 insertions(+), 189 deletions(-)
diff --git a/audio/audio.h b/audio/audio.h
index f4339a185e4e..02f29a3b3ef7 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -26,18 +26,10 @@
#define QEMU_AUDIO_H
#include "qemu/queue.h"
+#include "qapi/qapi-types-audio.h"
typedef void (*audio_callback_fn) (void *opaque, int avail);
-typedef enum {
- AUD_FMT_U8,
- AUD_FMT_S8,
- AUD_FMT_U16,
- AUD_FMT_S16,
- AUD_FMT_U32,
- AUD_FMT_S32
-} audfmt_e;
-
#ifdef HOST_WORDS_BIGENDIAN
#define AUDIO_HOST_ENDIANNESS 1
#else
@@ -47,7 +39,7 @@ typedef enum {
struct audsettings {
int freq;
int nchannels;
- audfmt_e fmt;
+ AudioFormat fmt;
int endianness;
};
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 635be73bf495..5bd034267fce 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -87,7 +87,7 @@ struct alsa_params_req {
struct alsa_params_obt {
int freq;
- audfmt_e fmt;
+ AudioFormat fmt;
int endianness;
int nchannels;
snd_pcm_uframes_t samples;
@@ -294,16 +294,16 @@ static int alsa_write (SWVoiceOut *sw, void *buf, int len)
return audio_pcm_sw_write (sw, buf, len);
}
-static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
+static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness)
{
switch (fmt) {
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
return SND_PCM_FORMAT_S8;
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
return SND_PCM_FORMAT_U8;
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
if (endianness) {
return SND_PCM_FORMAT_S16_BE;
}
@@ -311,7 +311,7 @@ static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
return SND_PCM_FORMAT_S16_LE;
}
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
if (endianness) {
return SND_PCM_FORMAT_U16_BE;
}
@@ -319,7 +319,7 @@ static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
return SND_PCM_FORMAT_U16_LE;
}
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_S32:
if (endianness) {
return SND_PCM_FORMAT_S32_BE;
}
@@ -327,7 +327,7 @@ static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
return SND_PCM_FORMAT_S32_LE;
}
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_U32:
if (endianness) {
return SND_PCM_FORMAT_U32_BE;
}
@@ -344,58 +344,58 @@ static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
}
}
-static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt,
+static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt,
int *endianness)
{
switch (alsafmt) {
case SND_PCM_FORMAT_S8:
*endianness = 0;
- *fmt = AUD_FMT_S8;
+ *fmt = AUDIO_FORMAT_S8;
break;
case SND_PCM_FORMAT_U8:
*endianness = 0;
- *fmt = AUD_FMT_U8;
+ *fmt = AUDIO_FORMAT_U8;
break;
case SND_PCM_FORMAT_S16_LE:
*endianness = 0;
- *fmt = AUD_FMT_S16;
+ *fmt = AUDIO_FORMAT_S16;
break;
case SND_PCM_FORMAT_U16_LE:
*endianness = 0;
- *fmt = AUD_FMT_U16;
+ *fmt = AUDIO_FORMAT_U16;
break;
case SND_PCM_FORMAT_S16_BE:
*endianness = 1;
- *fmt = AUD_FMT_S16;
+ *fmt = AUDIO_FORMAT_S16;
break;
case SND_PCM_FORMAT_U16_BE:
*endianness = 1;
- *fmt = AUD_FMT_U16;
+ *fmt = AUDIO_FORMAT_U16;
break;
case SND_PCM_FORMAT_S32_LE:
*endianness = 0;
- *fmt = AUD_FMT_S32;
+ *fmt = AUDIO_FORMAT_S32;
break;
case SND_PCM_FORMAT_U32_LE:
*endianness = 0;
- *fmt = AUD_FMT_U32;
+ *fmt = AUDIO_FORMAT_U32;
break;
case SND_PCM_FORMAT_S32_BE:
*endianness = 1;
- *fmt = AUD_FMT_S32;
+ *fmt = AUDIO_FORMAT_S32;
break;
case SND_PCM_FORMAT_U32_BE:
*endianness = 1;
- *fmt = AUD_FMT_U32;
+ *fmt = AUDIO_FORMAT_U32;
break;
default:
@@ -638,19 +638,22 @@ static int alsa_open (int in, struct alsa_params_req *req,
bytes_per_sec = freq << (nchannels == 2);
switch (obt->fmt) {
- case AUD_FMT_S8:
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_S8:
+ case AUDIO_FORMAT_U8:
break;
- case AUD_FMT_S16:
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_S16:
+ case AUDIO_FORMAT_U16:
bytes_per_sec <<= 1;
break;
- case AUD_FMT_S32:
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_S32:
+ case AUDIO_FORMAT_U32:
bytes_per_sec <<= 2;
break;
+
+ default:
+ abort();
}
threshold = (conf->threshold * bytes_per_sec) / 1000;
diff --git a/audio/audio.c b/audio/audio.c
index 909c817103c8..77216e50102e 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -113,7 +113,7 @@ static struct {
.settings = {
.freq = 44100,
.nchannels = 2,
- .fmt = AUD_FMT_S16,
+ .fmt = AUDIO_FORMAT_S16,
.endianness = AUDIO_HOST_ENDIANNESS,
}
},
@@ -125,7 +125,7 @@ static struct {
.settings = {
.freq = 44100,
.nchannels = 2,
- .fmt = AUD_FMT_S16,
+ .fmt = AUDIO_FORMAT_S16,
.endianness = AUDIO_HOST_ENDIANNESS,
}
},
@@ -257,58 +257,61 @@ static char *audio_alloc_prefix (const char *s)
return r;
}
-static const char *audio_audfmt_to_string (audfmt_e fmt)
+static const char *audio_audfmt_to_string (AudioFormat fmt)
{
switch (fmt) {
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
return "U8";
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
return "U16";
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
return "S8";
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
return "S16";
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_U32:
return "U32";
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_S32:
return "S32";
+
+ default:
+ abort();
}
dolog ("Bogus audfmt %d returning S16\n", fmt);
return "S16";
}
-static audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval,
+static AudioFormat audio_string_to_audfmt (const char *s, AudioFormat defval,
int *defaultp)
{
if (!strcasecmp (s, "u8")) {
*defaultp = 0;
- return AUD_FMT_U8;
+ return AUDIO_FORMAT_U8;
}
else if (!strcasecmp (s, "u16")) {
*defaultp = 0;
- return AUD_FMT_U16;
+ return AUDIO_FORMAT_U16;
}
else if (!strcasecmp (s, "u32")) {
*defaultp = 0;
- return AUD_FMT_U32;
+ return AUDIO_FORMAT_U32;
}
else if (!strcasecmp (s, "s8")) {
*defaultp = 0;
- return AUD_FMT_S8;
+ return AUDIO_FORMAT_S8;
}
else if (!strcasecmp (s, "s16")) {
*defaultp = 0;
- return AUD_FMT_S16;
+ return AUDIO_FORMAT_S16;
}
else if (!strcasecmp (s, "s32")) {
*defaultp = 0;
- return AUD_FMT_S32;
+ return AUDIO_FORMAT_S32;
}
else {
dolog ("Bogus audio format `%s' using %s\n",
@@ -318,8 +321,8 @@ static audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval,
}
}
-static audfmt_e audio_get_conf_fmt (const char *envname,
- audfmt_e defval,
+static AudioFormat audio_get_conf_fmt (const char *envname,
+ AudioFormat defval,
int *defaultp)
{
const char *var = getenv (envname);
@@ -421,7 +424,7 @@ static void audio_print_options (const char *prefix,
case AUD_OPT_FMT:
{
- audfmt_e *fmtp = opt->valp;
+ AudioFormat *fmtp = opt->valp;
printf (
"format, %s = %s, (one of: U8 S8 U16 S16 U32 S32)\n",
state,
@@ -492,7 +495,7 @@ static void audio_process_options (const char *prefix,
case AUD_OPT_FMT:
{
- audfmt_e *fmtp = opt->valp;
+ AudioFormat *fmtp = opt->valp;
*fmtp = audio_get_conf_fmt (optname, *fmtp, &def);
}
break;
@@ -524,22 +527,22 @@ static void audio_print_settings (struct audsettings *as)
dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels);
switch (as->fmt) {
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
AUD_log (NULL, "S8");
break;
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
AUD_log (NULL, "U8");
break;
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
AUD_log (NULL, "S16");
break;
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
AUD_log (NULL, "U16");
break;
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_S32:
AUD_log (NULL, "S32");
break;
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_U32:
AUD_log (NULL, "U32");
break;
default:
@@ -570,12 +573,12 @@ static int audio_validate_settings (struct audsettings *as)
invalid |= as->endianness != 0 && as->endianness != 1;
switch (as->fmt) {
- case AUD_FMT_S8:
- case AUD_FMT_U8:
- case AUD_FMT_S16:
- case AUD_FMT_U16:
- case AUD_FMT_S32:
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_S8:
+ case AUDIO_FORMAT_U8:
+ case AUDIO_FORMAT_S16:
+ case AUDIO_FORMAT_U16:
+ case AUDIO_FORMAT_S32:
+ case AUDIO_FORMAT_U32:
break;
default:
invalid = 1;
@@ -591,25 +594,28 @@ static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *a
int bits = 8, sign = 0;
switch (as->fmt) {
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
sign = 1;
/* fall through */
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
break;
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
sign = 1;
/* fall through */
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
bits = 16;
break;
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_S32:
sign = 1;
/* fall through */
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_U32:
bits = 32;
break;
+
+ default:
+ abort();
}
return info->freq == as->freq
&& info->nchannels == as->nchannels
@@ -623,24 +629,27 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
int bits = 8, sign = 0, shift = 0;
switch (as->fmt) {
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
sign = 1;
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
break;
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
sign = 1;
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
bits = 16;
shift = 1;
break;
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_S32:
sign = 1;
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_U32:
bits = 32;
shift = 2;
break;
+
+ default:
+ abort();
}
info->freq = as->freq;
diff --git a/audio/audio_win_int.c b/audio/audio_win_int.c
index 6900008d0c47..b938fd667bae 100644
--- a/audio/audio_win_int.c
+++ b/audio/audio_win_int.c
@@ -24,20 +24,20 @@ int waveformat_from_audio_settings (WAVEFORMATEX *wfx,
wfx->cbSize = 0;
switch (as->fmt) {
- case AUD_FMT_S8:
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_S8:
+ case AUDIO_FORMAT_U8:
wfx->wBitsPerSample = 8;
break;
- case AUD_FMT_S16:
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_S16:
+ case AUDIO_FORMAT_U16:
wfx->wBitsPerSample = 16;
wfx->nAvgBytesPerSec <<= 1;
wfx->nBlockAlign <<= 1;
break;
- case AUD_FMT_S32:
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_S32:
+ case AUDIO_FORMAT_U32:
wfx->wBitsPerSample = 32;
wfx->nAvgBytesPerSec <<= 2;
wfx->nBlockAlign <<= 2;
@@ -85,15 +85,15 @@ int waveformat_to_audio_settings (WAVEFORMATEX *wfx,
switch (wfx->wBitsPerSample) {
case 8:
- as->fmt = AUD_FMT_U8;
+ as->fmt = AUDIO_FORMAT_U8;
break;
case 16:
- as->fmt = AUD_FMT_S16;
+ as->fmt = AUDIO_FORMAT_S16;
break;
case 32:
- as->fmt = AUD_FMT_S32;
+ as->fmt = AUDIO_FORMAT_S32;
break;
default:
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 6c69622b4c7e..355e8fbda519 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -70,7 +70,7 @@ typedef struct OSSVoiceIn {
struct oss_params {
int freq;
- audfmt_e fmt;
+ AudioFormat fmt;
int nchannels;
int nfrags;
int fragsize;
@@ -148,16 +148,16 @@ static int oss_write (SWVoiceOut *sw, void *buf, int len)
return audio_pcm_sw_write (sw, buf, len);
}
-static int aud_to_ossfmt (audfmt_e fmt, int endianness)
+static int aud_to_ossfmt (AudioFormat fmt, int endianness)
{
switch (fmt) {
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
return AFMT_S8;
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
return AFMT_U8;
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
if (endianness) {
return AFMT_S16_BE;
}
@@ -165,7 +165,7 @@ static int aud_to_ossfmt (audfmt_e fmt, int endianness)
return AFMT_S16_LE;
}
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
if (endianness) {
return AFMT_U16_BE;
}
@@ -182,37 +182,37 @@ static int aud_to_ossfmt (audfmt_e fmt, int endianness)
}
}
-static int oss_to_audfmt (int ossfmt, audfmt_e *fmt, int *endianness)
+static int oss_to_audfmt (int ossfmt, AudioFormat *fmt, int *endianness)
{
switch (ossfmt) {
case AFMT_S8:
*endianness = 0;
- *fmt = AUD_FMT_S8;
+ *fmt = AUDIO_FORMAT_S8;
break;
case AFMT_U8:
*endianness = 0;
- *fmt = AUD_FMT_U8;
+ *fmt = AUDIO_FORMAT_U8;
break;
case AFMT_S16_LE:
*endianness = 0;
- *fmt = AUD_FMT_S16;
+ *fmt = AUDIO_FORMAT_S16;
break;
case AFMT_U16_LE:
*endianness = 0;
- *fmt = AUD_FMT_U16;
+ *fmt = AUDIO_FORMAT_U16;
break;
case AFMT_S16_BE:
*endianness = 1;
- *fmt = AUD_FMT_S16;
+ *fmt = AUDIO_FORMAT_S16;
break;
case AFMT_U16_BE:
*endianness = 1;
- *fmt = AUD_FMT_U16;
+ *fmt = AUDIO_FORMAT_U16;
break;
default:
@@ -500,7 +500,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
int endianness;
int err;
int fd;
- audfmt_e effective_fmt;
+ AudioFormat effective_fmt;
struct audsettings obt_as;
OSSConf *conf = drv_opaque;
@@ -667,7 +667,7 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
int endianness;
int err;
int fd;
- audfmt_e effective_fmt;
+ AudioFormat effective_fmt;
struct audsettings obt_as;
OSSConf *conf = drv_opaque;
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 6153b908dae1..8246f260a8ac 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -385,21 +385,21 @@ static int qpa_read (SWVoiceIn *sw, void *buf, int len)
return audio_pcm_sw_read (sw, buf, len);
}
-static pa_sample_format_t audfmt_to_pa (audfmt_e afmt, int endianness)
+static pa_sample_format_t audfmt_to_pa (AudioFormat afmt, int endianness)
{
int format;
switch (afmt) {
- case AUD_FMT_S8:
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_S8:
+ case AUDIO_FORMAT_U8:
format = PA_SAMPLE_U8;
break;
- case AUD_FMT_S16:
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_S16:
+ case AUDIO_FORMAT_U16:
format = endianness ? PA_SAMPLE_S16BE : PA_SAMPLE_S16LE;
break;
- case AUD_FMT_S32:
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_S32:
+ case AUDIO_FORMAT_U32:
format = endianness ? PA_SAMPLE_S32BE : PA_SAMPLE_S32LE;
break;
default:
@@ -410,26 +410,26 @@ static pa_sample_format_t audfmt_to_pa (audfmt_e afmt, int endianness)
return format;
}
-static audfmt_e pa_to_audfmt (pa_sample_format_t fmt, int *endianness)
+static AudioFormat pa_to_audfmt (pa_sample_format_t fmt, int *endianness)
{
switch (fmt) {
case PA_SAMPLE_U8:
- return AUD_FMT_U8;
+ return AUDIO_FORMAT_U8;
case PA_SAMPLE_S16BE:
*endianness = 1;
- return AUD_FMT_S16;
+ return AUDIO_FORMAT_S16;
case PA_SAMPLE_S16LE:
*endianness = 0;
- return AUD_FMT_S16;
+ return AUDIO_FORMAT_S16;
case PA_SAMPLE_S32BE:
*endianness = 1;
- return AUD_FMT_S32;
+ return AUDIO_FORMAT_S32;
case PA_SAMPLE_S32LE:
*endianness = 0;
- return AUD_FMT_S32;
+ return AUDIO_FORMAT_S32;
default:
dolog ("Internal logic error: Bad pa_sample_format %d\n", fmt);
- return AUD_FMT_U8;
+ return AUDIO_FORMAT_U8;
}
}
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index f7ee70b15347..4cd4cbaf0043 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -68,19 +68,19 @@ static void GCC_FMT_ATTR (1, 2) sdl_logerr (const char *fmt, ...)
AUD_log (AUDIO_CAP, "Reason: %s\n", SDL_GetError ());
}
-static int aud_to_sdlfmt (audfmt_e fmt)
+static int aud_to_sdlfmt (AudioFormat fmt)
{
switch (fmt) {
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
return AUDIO_S8;
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
return AUDIO_U8;
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
return AUDIO_S16LSB;
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
return AUDIO_U16LSB;
default:
@@ -92,37 +92,37 @@ static int aud_to_sdlfmt (audfmt_e fmt)
}
}
-static int sdl_to_audfmt(int sdlfmt, audfmt_e *fmt, int *endianness)
+static int sdl_to_audfmt(int sdlfmt, AudioFormat *fmt, int *endianness)
{
switch (sdlfmt) {
case AUDIO_S8:
*endianness = 0;
- *fmt = AUD_FMT_S8;
+ *fmt = AUDIO_FORMAT_S8;
break;
case AUDIO_U8:
*endianness = 0;
- *fmt = AUD_FMT_U8;
+ *fmt = AUDIO_FORMAT_U8;
break;
case AUDIO_S16LSB:
*endianness = 0;
- *fmt = AUD_FMT_S16;
+ *fmt = AUDIO_FORMAT_S16;
break;
case AUDIO_U16LSB:
*endianness = 0;
- *fmt = AUD_FMT_U16;
+ *fmt = AUDIO_FORMAT_U16;
break;
case AUDIO_S16MSB:
*endianness = 1;
- *fmt = AUD_FMT_S16;
+ *fmt = AUDIO_FORMAT_S16;
break;
case AUDIO_U16MSB:
*endianness = 1;
- *fmt = AUD_FMT_U16;
+ *fmt = AUDIO_FORMAT_U16;
break;
default:
@@ -265,7 +265,7 @@ static int sdl_init_out(HWVoiceOut *hw, struct audsettings *as,
SDL_AudioSpec req, obt;
int endianness;
int err;
- audfmt_e effective_fmt;
+ AudioFormat effective_fmt;
struct audsettings obt_as;
req.freq = as->freq;
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index 6ad0eafbc67f..3aeb0cb35722 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -130,7 +130,7 @@ static int line_out_init(HWVoiceOut *hw, struct audsettings *as,
settings.freq = SPICE_INTERFACE_PLAYBACK_FREQ;
#endif
settings.nchannels = SPICE_INTERFACE_PLAYBACK_CHAN;
- settings.fmt = AUD_FMT_S16;
+ settings.fmt = AUDIO_FORMAT_S16;
settings.endianness = AUDIO_HOST_ENDIANNESS;
audio_pcm_init_info (&hw->info, &settings);
@@ -258,7 +258,7 @@ static int line_in_init(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
settings.freq = SPICE_INTERFACE_RECORD_FREQ;
#endif
settings.nchannels = SPICE_INTERFACE_RECORD_CHAN;
- settings.fmt = AUD_FMT_S16;
+ settings.fmt = AUDIO_FORMAT_S16;
settings.endianness = AUDIO_HOST_ENDIANNESS;
audio_pcm_init_info (&hw->info, &settings);
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 40adfa30c33a..35a614785ece 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -117,20 +117,23 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
stereo = wav_as.nchannels == 2;
switch (wav_as.fmt) {
- case AUD_FMT_S8:
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_S8:
+ case AUDIO_FORMAT_U8:
bits16 = 0;
break;
- case AUD_FMT_S16:
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_S16:
+ case AUDIO_FORMAT_U16:
bits16 = 1;
break;
- case AUD_FMT_S32:
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_S32:
+ case AUDIO_FORMAT_U32:
dolog ("WAVE files can not handle 32bit formats\n");
return -1;
+
+ default:
+ abort();
}
hdr[34] = bits16 ? 0x10 : 0x08;
@@ -225,7 +228,7 @@ static int wav_ctl_out (HWVoiceOut *hw, int cmd, ...)
static WAVConf glob_conf = {
.settings.freq = 44100,
.settings.nchannels = 2,
- .settings.fmt = AUD_FMT_S16,
+ .settings.fmt = AUDIO_FORMAT_S16,
.wav_path = "qemu.wav"
};
diff --git a/audio/wavcapture.c b/audio/wavcapture.c
index cd24570aa70c..74320dfecc76 100644
--- a/audio/wavcapture.c
+++ b/audio/wavcapture.c
@@ -136,7 +136,7 @@ int wav_start_capture (CaptureState *s, const char *path, int freq,
as.freq = freq;
as.nchannels = 1 << stereo;
- as.fmt = bits16 ? AUD_FMT_S16 : AUD_FMT_U8;
+ as.fmt = bits16 ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_U8;
as.endianness = 0;
ops.notify = wav_notify;
diff --git a/hw/arm/omap2.c b/hw/arm/omap2.c
index 94dffb2f577c..446223906e47 100644
--- a/hw/arm/omap2.c
+++ b/hw/arm/omap2.c
@@ -273,7 +273,7 @@ static void omap_eac_format_update(struct omap_eac_s *s)
* does I2S specify it? */
/* All register writes are 16 bits so we we store 16-bit samples
* in the buffers regardless of AGCFR[B8_16] value. */
- fmt.fmt = AUD_FMT_U16;
+ fmt.fmt = AUDIO_FORMAT_U16;
s->codec.in_voice = AUD_open_in(&s->codec.card, s->codec.in_voice,
"eac.codec.in", s, omap_eac_in_cb, &fmt);
diff --git a/hw/audio/ac97.c b/hw/audio/ac97.c
index d799533aa991..2265622d44a2 100644
--- a/hw/audio/ac97.c
+++ b/hw/audio/ac97.c
@@ -365,7 +365,7 @@ static void open_voice (AC97LinkState *s, int index, int freq)
as.freq = freq;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = 0;
if (freq > 0) {
diff --git a/hw/audio/adlib.c b/hw/audio/adlib.c
index 97b876c7e08c..0957780a3dc0 100644
--- a/hw/audio/adlib.c
+++ b/hw/audio/adlib.c
@@ -269,7 +269,7 @@ static void adlib_realizefn (DeviceState *dev, Error **errp)
as.freq = s->freq;
as.nchannels = SHIFT;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = AUDIO_HOST_ENDIANNESS;
AUD_register_card ("adlib", &s->card);
diff --git a/hw/audio/cs4231a.c b/hw/audio/cs4231a.c
index 9089dcb47e84..62da75eefebb 100644
--- a/hw/audio/cs4231a.c
+++ b/hw/audio/cs4231a.c
@@ -288,7 +288,7 @@ static void cs_reset_voices (CSState *s, uint32_t val)
switch ((val >> 5) & ((s->dregs[MODE_And_ID] & MODE2) ? 7 : 3)) {
case 0:
- as.fmt = AUD_FMT_U8;
+ as.fmt = AUDIO_FORMAT_U8;
s->shift = as.nchannels == 2;
break;
@@ -298,7 +298,7 @@ static void cs_reset_voices (CSState *s, uint32_t val)
case 3:
s->tab = ALawDecompressTable;
x_law:
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = AUDIO_HOST_ENDIANNESS;
s->shift = as.nchannels == 2;
break;
@@ -307,7 +307,7 @@ static void cs_reset_voices (CSState *s, uint32_t val)
as.endianness = 1;
/* fall through */
case 2:
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
s->shift = as.nchannels;
break;
diff --git a/hw/audio/es1370.c b/hw/audio/es1370.c
index 97789a077196..a5314d66fdc2 100644
--- a/hw/audio/es1370.c
+++ b/hw/audio/es1370.c
@@ -414,14 +414,14 @@ static void es1370_update_voices (ES1370State *s, uint32_t ctl, uint32_t sctl)
i,
new_freq,
1 << (new_fmt & 1),
- (new_fmt & 2) ? AUD_FMT_S16 : AUD_FMT_U8,
+ (new_fmt & 2) ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_U8,
d->shift);
if (new_freq) {
struct audsettings as;
as.freq = new_freq;
as.nchannels = 1 << (new_fmt & 1);
- as.fmt = (new_fmt & 2) ? AUD_FMT_S16 : AUD_FMT_U8;
+ as.fmt = (new_fmt & 2) ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_U8;
as.endianness = 0;
if (i == ADC_CHANNEL) {
diff --git a/hw/audio/gus.c b/hw/audio/gus.c
index 8e0b27e0f237..b3e2a7fdd504 100644
--- a/hw/audio/gus.c
+++ b/hw/audio/gus.c
@@ -251,7 +251,7 @@ static void gus_realizefn (DeviceState *dev, Error **errp)
as.freq = s->freq;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = GUS_ENDIANNESS;
s->voice = AUD_open_out (
diff --git a/hw/audio/hda-codec.c b/hw/audio/hda-codec.c
index 617a1c1016b8..c25bfa38b143 100644
--- a/hw/audio/hda-codec.c
+++ b/hw/audio/hda-codec.c
@@ -99,9 +99,9 @@ static void hda_codec_parse_fmt(uint32_t format, struct audsettings *as)
}
switch (format & AC_FMT_BITS_MASK) {
- case AC_FMT_BITS_8: as->fmt = AUD_FMT_S8; break;
- case AC_FMT_BITS_16: as->fmt = AUD_FMT_S16; break;
- case AC_FMT_BITS_32: as->fmt = AUD_FMT_S32; break;
+ case AC_FMT_BITS_8: as->fmt = AUDIO_FORMAT_S8; break;
+ case AC_FMT_BITS_16: as->fmt = AUDIO_FORMAT_S16; break;
+ case AC_FMT_BITS_32: as->fmt = AUDIO_FORMAT_S32; break;
}
as->nchannels = ((format & AC_FMT_CHAN_MASK) >> AC_FMT_CHAN_SHIFT) + 1;
@@ -134,12 +134,12 @@ static void hda_codec_parse_fmt(uint32_t format, struct audsettings *as)
/* -------------------------------------------------------------------------- */
static const char *fmt2name[] = {
- [ AUD_FMT_U8 ] = "PCM-U8",
- [ AUD_FMT_S8 ] = "PCM-S8",
- [ AUD_FMT_U16 ] = "PCM-U16",
- [ AUD_FMT_S16 ] = "PCM-S16",
- [ AUD_FMT_U32 ] = "PCM-U32",
- [ AUD_FMT_S32 ] = "PCM-S32",
+ [ AUDIO_FORMAT_U8 ] = "PCM-U8",
+ [ AUDIO_FORMAT_S8 ] = "PCM-S8",
+ [ AUDIO_FORMAT_U16 ] = "PCM-U16",
+ [ AUDIO_FORMAT_S16 ] = "PCM-S16",
+ [ AUDIO_FORMAT_U32 ] = "PCM-U32",
+ [ AUDIO_FORMAT_S32 ] = "PCM-S32",
};
typedef struct HDAAudioState HDAAudioState;
diff --git a/hw/audio/lm4549.c b/hw/audio/lm4549.c
index a46f2301af82..af8b22b541e9 100644
--- a/hw/audio/lm4549.c
+++ b/hw/audio/lm4549.c
@@ -185,7 +185,7 @@ void lm4549_write(lm4549_state *s,
struct audsettings as;
as.freq = value;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = 0;
s->voice = AUD_open_out(
@@ -255,7 +255,7 @@ static int lm4549_post_load(void *opaque, int version_id)
struct audsettings as;
as.freq = freq;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = 0;
s->voice = AUD_open_out(
@@ -292,7 +292,7 @@ void lm4549_init(lm4549_state *s, lm4549_callback data_req_cb, void* opaque)
/* Open a default voice */
as.freq = 48000;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = 0;
s->voice = AUD_open_out(
diff --git a/hw/audio/milkymist-ac97.c b/hw/audio/milkymist-ac97.c
index bc8db71ae0dd..90cce1e6eded 100644
--- a/hw/audio/milkymist-ac97.c
+++ b/hw/audio/milkymist-ac97.c
@@ -308,7 +308,7 @@ static void milkymist_ac97_realize(DeviceState *dev, Error **errp)
as.freq = 48000;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = 1;
s->voice_in = AUD_open_in(&s->card, s->voice_in,
diff --git a/hw/audio/pcspk.c b/hw/audio/pcspk.c
index b80a62ce90a4..fdbb4b6e9923 100644
--- a/hw/audio/pcspk.c
+++ b/hw/audio/pcspk.c
@@ -162,7 +162,7 @@ static void pcspk_initfn(Object *obj)
static void pcspk_realizefn(DeviceState *dev, Error **errp)
{
- struct audsettings as = {PCSPK_SAMPLE_RATE, 1, AUD_FMT_U8, 0};
+ struct audsettings as = {PCSPK_SAMPLE_RATE, 1, AUDIO_FORMAT_U8, 0};
ISADevice *isadev = ISA_DEVICE(dev);
PCSpkState *s = PC_SPEAKER(dev);
diff --git a/hw/audio/sb16.c b/hw/audio/sb16.c
index c5b9bf79e819..65ea0cd93832 100644
--- a/hw/audio/sb16.c
+++ b/hw/audio/sb16.c
@@ -66,7 +66,7 @@ typedef struct SB16State {
int fmt_stereo;
int fmt_signed;
int fmt_bits;
- audfmt_e fmt;
+ AudioFormat fmt;
int dma_auto;
int block_size;
int fifo;
@@ -224,7 +224,7 @@ static void continue_dma8 (SB16State *s)
static void dma_cmd8 (SB16State *s, int mask, int dma_len)
{
- s->fmt = AUD_FMT_U8;
+ s->fmt = AUDIO_FORMAT_U8;
s->use_hdma = 0;
s->fmt_bits = 8;
s->fmt_signed = 0;
@@ -319,18 +319,18 @@ static void dma_cmd (SB16State *s, uint8_t cmd, uint8_t d0, int dma_len)
if (16 == s->fmt_bits) {
if (s->fmt_signed) {
- s->fmt = AUD_FMT_S16;
+ s->fmt = AUDIO_FORMAT_S16;
}
else {
- s->fmt = AUD_FMT_U16;
+ s->fmt = AUDIO_FORMAT_U16;
}
}
else {
if (s->fmt_signed) {
- s->fmt = AUD_FMT_S8;
+ s->fmt = AUDIO_FORMAT_S8;
}
else {
- s->fmt = AUD_FMT_U8;
+ s->fmt = AUDIO_FORMAT_U8;
}
}
@@ -852,7 +852,7 @@ static void legacy_reset (SB16State *s)
as.freq = s->freq;
as.nchannels = 1;
- as.fmt = AUD_FMT_U8;
+ as.fmt = AUDIO_FORMAT_U8;
as.endianness = 0;
s->voice = AUD_open_out (
diff --git a/hw/audio/wm8750.c b/hw/audio/wm8750.c
index 169b006ade2d..ca0ad73caf7d 100644
--- a/hw/audio/wm8750.c
+++ b/hw/audio/wm8750.c
@@ -201,7 +201,7 @@ static void wm8750_set_format(WM8750State *s)
in_fmt.endianness = 0;
in_fmt.nchannels = 2;
in_fmt.freq = s->adc_hz;
- in_fmt.fmt = AUD_FMT_S16;
+ in_fmt.fmt = AUDIO_FORMAT_S16;
s->adc_voice[0] = AUD_open_in(&s->card, s->adc_voice[0],
CODEC ".input1", s, wm8750_audio_in_cb, &in_fmt);
@@ -214,7 +214,7 @@ static void wm8750_set_format(WM8750State *s)
out_fmt.endianness = 0;
out_fmt.nchannels = 2;
out_fmt.freq = s->dac_hz;
- out_fmt.fmt = AUD_FMT_S16;
+ out_fmt.fmt = AUDIO_FORMAT_S16;
s->dac_voice[0] = AUD_open_out(&s->card, s->dac_voice[0],
CODEC ".speaker", s, wm8750_audio_out_cb, &out_fmt);
@@ -681,7 +681,7 @@ uint32_t wm8750_adc_dat(void *opaque)
if (s->idx_in >= sizeof(s->data_in)) {
wm8750_in_load(s);
if (s->idx_in >= sizeof(s->data_in)) {
- return 0x80008000; /* silence in AUD_FMT_S16 sample format */
+ return 0x80008000; /* silence in AUDIO_FORMAT_S16 sample format */
}
}
diff --git a/hw/display/xlnx_dp.c b/hw/display/xlnx_dp.c
index cc0f9bc9cc8b..11b09bd18c19 100644
--- a/hw/display/xlnx_dp.c
+++ b/hw/display/xlnx_dp.c
@@ -1260,7 +1260,7 @@ static void xlnx_dp_realize(DeviceState *dev, Error **errp)
as.freq = 44100;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = 0;
AUD_register_card("xlnx_dp.audio", &s->aud_card);
diff --git a/hw/input/tsc210x.c b/hw/input/tsc210x.c
index 2eb3cb951878..41731619bb6a 100644
--- a/hw/input/tsc210x.c
+++ b/hw/input/tsc210x.c
@@ -318,7 +318,7 @@ static void tsc2102_audio_output_update(TSC210xState *s)
fmt.endianness = 0;
fmt.nchannels = 2;
fmt.freq = s->codec.tx_rate;
- fmt.fmt = AUD_FMT_S16;
+ fmt.fmt = AUDIO_FORMAT_S16;
s->dac_voice[0] = AUD_open_out(&s->card, s->dac_voice[0],
"tsc2102.sink", s, (void *) tsc210x_audio_out_cb, &fmt);
diff --git a/hw/usb/dev-audio.c b/hw/usb/dev-audio.c
index 28ac7c5165ff..c46d5eeb79a3 100644
--- a/hw/usb/dev-audio.c
+++ b/hw/usb/dev-audio.c
@@ -650,7 +650,7 @@ static void usb_audio_realize(USBDevice *dev, Error **errp)
s->out.vol[1] = 240; /* 0 dB */
s->out.as.freq = USBAUDIO_SAMPLE_RATE;
s->out.as.nchannels = 2;
- s->out.as.fmt = AUD_FMT_S16;
+ s->out.as.fmt = AUDIO_FORMAT_S16;
s->out.as.endianness = 0;
streambuf_init(&s->out.buf, s->buffer);
diff --git a/ui/vnc.c b/ui/vnc.c
index da4a21d4ce94..b724e998bab6 100644
--- a/ui/vnc.c
+++ b/ui/vnc.c
@@ -1013,16 +1013,16 @@ static void vnc_update_throttle_offset(VncState *vs)
int bps;
switch (vs->as.fmt) {
default:
- case AUD_FMT_U8:
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_U8:
+ case AUDIO_FORMAT_S8:
bps = 1;
break;
- case AUD_FMT_U16:
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_U16:
+ case AUDIO_FORMAT_S16:
bps = 2;
break;
- case AUD_FMT_U32:
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_U32:
+ case AUDIO_FORMAT_S32:
bps = 4;
break;
}
@@ -2369,12 +2369,12 @@ static int protocol_client_msg(VncState *vs, uint8_t *data, size_t len)
if (len == 4)
return 10;
switch (read_u8(data, 4)) {
- case 0: vs->as.fmt = AUD_FMT_U8; break;
- case 1: vs->as.fmt = AUD_FMT_S8; break;
- case 2: vs->as.fmt = AUD_FMT_U16; break;
- case 3: vs->as.fmt = AUD_FMT_S16; break;
- case 4: vs->as.fmt = AUD_FMT_U32; break;
- case 5: vs->as.fmt = AUD_FMT_S32; break;
+ case 0: vs->as.fmt = AUDIO_FORMAT_U8; break;
+ case 1: vs->as.fmt = AUDIO_FORMAT_S8; break;
+ case 2: vs->as.fmt = AUDIO_FORMAT_U16; break;
+ case 3: vs->as.fmt = AUDIO_FORMAT_S16; break;
+ case 4: vs->as.fmt = AUDIO_FORMAT_U32; break;
+ case 5: vs->as.fmt = AUDIO_FORMAT_S32; break;
default:
VNC_DEBUG("Invalid audio format %d\n", read_u8(data, 4));
vnc_client_error(vs);
@@ -3105,7 +3105,7 @@ static void vnc_connect(VncDisplay *vd, QIOChannelSocket *sioc,
vs->as.freq = 44100;
vs->as.nchannels = 2;
- vs->as.fmt = AUD_FMT_S16;
+ vs->as.fmt = AUDIO_FORMAT_S16;
vs->as.endianness = 0;
qemu_mutex_init(&vs->output_mutex);
--
2.18.1
--
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