izzy/xemu
0
0
Fork 0
mirror of https://github.com/izzy2lost/xemu.git synced 2026-07-06 00:20:22 -07:00
Code Issues Packages Projects Releases Wiki Activity

audio: use qapi AudioFormat instead of audfmt_e

Browse Source 
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@gmail.com>
Reviewed-by: Thomas Huth <thuth@redhat.com>
Message-id: 01251b2758a1679c66842120b77c0fb46d7d0eaf.1552083282.git.DirtY.iCE.hu@gmail.com
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
This commit is contained in:
Kővágó, Zoltán
2019-03-11 10:29:26 +01:00
committed by Gerd Hoffmann
parent 8c3a7d0087
commit 85bc58520c
26 changed files with 196 additions and 189 deletions
Download Patch File Download Diff File Expand all files Collapse all files
audio/alsaaudio.c
+28 -25
View File
@@ -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;
audio/audio.c
+53 -44
View File
@@ -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;
audio/audio.h
+2 -10
View File
@@ -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;
};
audio/audio_win_int.c
+9 -9
View File
@@ -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:
audio/ossaudio.c
+15 -15
View File
@@ -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;
audio/paaudio.c
+14 -14
View File
@@ -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;
}
}
audio/sdlaudio.c
+13 -13
View File
@@ -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;
audio/spiceaudio.c
+2 -2
View File
@@ -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);
audio/wavaudio.c
+10 -7
View File
@@ -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"
};
audio/wavcapture.c
+1 -1
View File
@@ -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;
hw/arm/omap2.c
+1 -1
View File
@@ -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);
hw/audio/ac97.c
+1 -1
View File
@@ -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) {
hw/audio/adlib.c
+1 -1
View File
@@ -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);
hw/audio/cs4231a.c
+3 -3
View File
@@ -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;
hw/audio/es1370.c
+2 -2
View File
@@ -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) {
hw/audio/gus.c
+1 -1
View File
@@ -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 (
hw/audio/hda-codec.c
+9 -9
View File
@@ -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;
hw/audio/lm4549.c
+3 -3
View File
@@ -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(
hw/audio/milkymist-ac97.c
+1 -1
View File
@@ -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,
hw/audio/pcspk.c
+1 -1
View File
@@ -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);

Some files were not shown because too many files have changed in this diff Show More