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Merge remote-tracking branch 'remotes/kraxel/tags/audio-20190312-pull-request' into staging

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audio: introduce -audiodev

# gpg: Signature made Tue 12 Mar 2019 07:12:19 GMT
# gpg:                using RSA key 4CB6D8EED3E87138
# gpg: Good signature from "Gerd Hoffmann (work) <kraxel@redhat.com>" [full]
# gpg:                 aka "Gerd Hoffmann <gerd@kraxel.org>" [full]
# gpg:                 aka "Gerd Hoffmann (private) <kraxel@gmail.com>" [full]
# Primary key fingerprint: A032 8CFF B93A 17A7 9901  FE7D 4CB6 D8EE D3E8 7138

* remotes/kraxel/tags/audio-20190312-pull-request:
  audio: -audiodev command line option: cleanup
  wavaudio: port to -audiodev config
  spiceaudio: port to -audiodev config
  sdlaudio: port to -audiodev config
  paaudio: port to -audiodev config
  ossaudio: port to -audiodev config
  noaudio: port to -audiodev config
  dsoundaudio: port to -audiodev config
  coreaudio: port to -audiodev config
  alsaaudio: port to -audiodev config
  audio: -audiodev command line option basic implementation
  audio: -audiodev command line option: documentation
  audio: use qapi AudioFormat instead of audfmt_e
  qapi: qapi for audio backends

Signed-off-by: Peter Maydell <peter.maydell@linaro.org>

# Conflicts:
#	qemu-deprecated.texi
This commit is contained in:
Peter Maydell
2019-03-12 16:45:13 +00:00
parent 2cb73afa6a 05d2f2a64d
commit cfc3fef6b4
40 changed files with 1835 additions and 1287 deletions
Download Patch File Download Diff File Expand all files Collapse all files
audio/Makefile.objs
+1 -1
View File
@@ -1,4 +1,4 @@
common-obj-y = audio.o noaudio.o wavaudio.o mixeng.o
common-obj-y = audio.o audio_legacy.o noaudio.o wavaudio.o mixeng.o
common-obj-$(CONFIG_SPICE) += spiceaudio.o
common-obj-$(CONFIG_AUDIO_COREAUDIO) += coreaudio.o
common-obj-$(CONFIG_AUDIO_DSOUND) += dsoundaudio.o
audio/alsaaudio.c
+119 -249
View File
File diff suppressed because it is too large Load Diff
audio/audio.c
+299 -560
View File
File diff suppressed because it is too large Load Diff
audio/audio.h
+17 -13
View File
@@ -26,30 +26,31 @@
#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
#define AUDIO_HOST_ENDIANNESS 0
#endif
struct audsettings {
typedef struct audsettings {
int freq;
int nchannels;
audfmt_e fmt;
AudioFormat fmt;
int endianness;
};
} audsettings;
audsettings audiodev_to_audsettings(AudiodevPerDirectionOptions *pdo);
int audioformat_bytes_per_sample(AudioFormat fmt);
int audio_buffer_frames(AudiodevPerDirectionOptions *pdo,
audsettings *as, int def_usecs);
int audio_buffer_samples(AudiodevPerDirectionOptions *pdo,
audsettings *as, int def_usecs);
int audio_buffer_bytes(AudiodevPerDirectionOptions *pdo,
audsettings *as, int def_usecs);
typedef enum {
AUD_CNOTIFY_ENABLE,
@@ -89,7 +90,6 @@ typedef struct QEMUAudioTimeStamp {
void AUD_vlog (const char *cap, const char *fmt, va_list ap) GCC_FMT_ATTR(2, 0);
void AUD_log (const char *cap, const char *fmt, ...) GCC_FMT_ATTR(2, 3);
void AUD_help (void);
void AUD_register_card (const char *name, QEMUSoundCard *card);
void AUD_remove_card (QEMUSoundCard *card);
CaptureVoiceOut *AUD_add_capture (
@@ -171,4 +171,8 @@ void audio_sample_to_uint64(void *samples, int pos,
void audio_sample_from_uint64(void *samples, int pos,
uint64_t left, uint64_t right);
void audio_parse_option(const char *opt);
void audio_init_audiodevs(void);
void audio_legacy_help(void);
#endif /* QEMU_AUDIO_H */
audio/audio_int.h
+19 -18
View File
@@ -33,22 +33,6 @@
struct audio_pcm_ops;
typedef enum {
AUD_OPT_INT,
AUD_OPT_FMT,
AUD_OPT_STR,
AUD_OPT_BOOL
} audio_option_tag_e;
struct audio_option {
const char *name;
audio_option_tag_e tag;
void *valp;
const char *descr;
int *overriddenp;
int overridden;
};
struct audio_callback {
void *opaque;
audio_callback_fn fn;
@@ -145,8 +129,7 @@ typedef struct audio_driver audio_driver;
struct audio_driver {
const char *name;
const char *descr;
struct audio_option *options;
void *(*init) (void);
void *(*init) (Audiodev *);
void (*fini) (void *);
struct audio_pcm_ops *pcm_ops;
int can_be_default;
@@ -193,6 +176,7 @@ struct SWVoiceCap {
typedef struct AudioState {
struct audio_driver *drv;
Audiodev *dev;
void *drv_opaque;
QEMUTimer *ts;
@@ -203,10 +187,13 @@ typedef struct AudioState {
int nb_hw_voices_out;
int nb_hw_voices_in;
int vm_running;
int64_t period_ticks;
} AudioState;
extern const struct mixeng_volume nominal_volume;
extern const char *audio_prio_list[];
void audio_driver_register(audio_driver *drv);
audio_driver *audio_driver_lookup(const char *name);
@@ -248,4 +235,18 @@ static inline int audio_ring_dist (int dst, int src, int len)
#define AUDIO_STRINGIFY_(n) #n
#define AUDIO_STRINGIFY(n) AUDIO_STRINGIFY_(n)
typedef struct AudiodevListEntry {
Audiodev *dev;
QSIMPLEQ_ENTRY(AudiodevListEntry) next;
} AudiodevListEntry;
typedef QSIMPLEQ_HEAD(, AudiodevListEntry) AudiodevListHead;
AudiodevListHead audio_handle_legacy_opts(void);
void audio_free_audiodev_list(AudiodevListHead *head);
void audio_create_pdos(Audiodev *dev);
AudiodevPerDirectionOptions *audio_get_pdo_in(Audiodev *dev);
AudiodevPerDirectionOptions *audio_get_pdo_out(Audiodev *dev);
#endif /* QEMU_AUDIO_INT_H */
audio/audio_legacy.c
+544
View File
File diff suppressed because it is too large Load Diff
audio/audio_template.h
+38 -4
View File
@@ -299,11 +299,42 @@ static HW *glue (audio_pcm_hw_add_new_, TYPE) (struct audsettings *as)
return NULL;
}
AudiodevPerDirectionOptions *glue(audio_get_pdo_, TYPE)(Audiodev *dev)
{
switch (dev->driver) {
case AUDIODEV_DRIVER_NONE:
return dev->u.none.TYPE;
case AUDIODEV_DRIVER_ALSA:
return qapi_AudiodevAlsaPerDirectionOptions_base(dev->u.alsa.TYPE);
case AUDIODEV_DRIVER_COREAUDIO:
return qapi_AudiodevCoreaudioPerDirectionOptions_base(
dev->u.coreaudio.TYPE);
case AUDIODEV_DRIVER_DSOUND:
return dev->u.dsound.TYPE;
case AUDIODEV_DRIVER_OSS:
return qapi_AudiodevOssPerDirectionOptions_base(dev->u.oss.TYPE);
case AUDIODEV_DRIVER_PA:
return qapi_AudiodevPaPerDirectionOptions_base(dev->u.pa.TYPE);
case AUDIODEV_DRIVER_SDL:
return dev->u.sdl.TYPE;
case AUDIODEV_DRIVER_SPICE:
return dev->u.spice.TYPE;
case AUDIODEV_DRIVER_WAV:
return dev->u.wav.TYPE;
case AUDIODEV_DRIVER__MAX:
break;
}
abort();
}
static HW *glue (audio_pcm_hw_add_, TYPE) (struct audsettings *as)
{
HW *hw;
AudioState *s = &glob_audio_state;
AudiodevPerDirectionOptions *pdo = glue(audio_get_pdo_, TYPE)(s->dev);
if (glue (conf.fixed_, TYPE).enabled && glue (conf.fixed_, TYPE).greedy) {
if (pdo->fixed_settings) {
hw = glue (audio_pcm_hw_add_new_, TYPE) (as);
if (hw) {
return hw;
@@ -331,9 +362,11 @@ static SW *glue (audio_pcm_create_voice_pair_, TYPE) (
SW *sw;
HW *hw;
struct audsettings hw_as;
AudioState *s = &glob_audio_state;
AudiodevPerDirectionOptions *pdo = glue(audio_get_pdo_, TYPE)(s->dev);
if (glue (conf.fixed_, TYPE).enabled) {
hw_as = glue (conf.fixed_, TYPE).settings;
if (pdo->fixed_settings) {
hw_as = audiodev_to_audsettings(pdo);
}
else {
hw_as = *as;
@@ -398,6 +431,7 @@ SW *glue (AUD_open_, TYPE) (
)
{
AudioState *s = &glob_audio_state;
AudiodevPerDirectionOptions *pdo = glue(audio_get_pdo_, TYPE)(s->dev);
if (audio_bug(__func__, !card || !name || !callback_fn || !as)) {
dolog ("card=%p name=%p callback_fn=%p as=%p\n",
@@ -422,7 +456,7 @@ SW *glue (AUD_open_, TYPE) (
return sw;
}
if (!glue (conf.fixed_, TYPE).enabled && sw) {
if (!pdo->fixed_settings && sw) {
glue (AUD_close_, TYPE) (card, sw);
sw = NULL;
}
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/coreaudio.c
+12 -39
View File
@@ -36,11 +36,6 @@
#define MAC_OS_X_VERSION_10_6 1060
#endif
typedef struct {
int buffer_frames;
int nbuffers;
} CoreaudioConf;
typedef struct coreaudioVoiceOut {
HWVoiceOut hw;
pthread_mutex_t mutex;
@@ -507,7 +502,9 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
int err;
const char *typ = "playback";
AudioValueRange frameRange;
CoreaudioConf *conf = drv_opaque;
Audiodev *dev = drv_opaque;
AudiodevCoreaudioPerDirectionOptions *cpdo = dev->u.coreaudio.out;
int frames;
/* create mutex */
err = pthread_mutex_init(&core->mutex, NULL);
@@ -538,16 +535,17 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
return -1;
}
if (frameRange.mMinimum > conf->buffer_frames) {
frames = audio_buffer_frames(
qapi_AudiodevCoreaudioPerDirectionOptions_base(cpdo), as, 11610);
if (frameRange.mMinimum > frames) {
core->audioDevicePropertyBufferFrameSize = (UInt32) frameRange.mMinimum;
dolog ("warning: Upsizing Buffer Frames to %f\n", frameRange.mMinimum);
}
else if (frameRange.mMaximum < conf->buffer_frames) {
} else if (frameRange.mMaximum < frames) {
core->audioDevicePropertyBufferFrameSize = (UInt32) frameRange.mMaximum;
dolog ("warning: Downsizing Buffer Frames to %f\n", frameRange.mMaximum);
}
else {
core->audioDevicePropertyBufferFrameSize = conf->buffer_frames;
core->audioDevicePropertyBufferFrameSize = frames;
}
/* set Buffer Frame Size */
@@ -568,7 +566,8 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
"Could not get device buffer frame size\n");
return -1;
}
hw->samples = conf->nbuffers * core->audioDevicePropertyBufferFrameSize;
hw->samples = (cpdo->has_buffer_count ? cpdo->buffer_count : 4) *
core->audioDevicePropertyBufferFrameSize;
/* get StreamFormat */
status = coreaudio_get_streamformat(core->outputDeviceID,
@@ -680,40 +679,15 @@ static int coreaudio_ctl_out (HWVoiceOut *hw, int cmd, ...)
return 0;
}
static CoreaudioConf glob_conf = {
.buffer_frames = 512,
.nbuffers = 4,
};
static void *coreaudio_audio_init (void)
static void *coreaudio_audio_init(Audiodev *dev)
{
CoreaudioConf *conf = g_malloc(sizeof(CoreaudioConf));
*conf = glob_conf;
return conf;
return dev;
}
static void coreaudio_audio_fini (void *opaque)
{
g_free(opaque);
}
static struct audio_option coreaudio_options[] = {
{
.name = "BUFFER_SIZE",
.tag = AUD_OPT_INT,
.valp = &glob_conf.buffer_frames,
.descr = "Size of the buffer in frames"
},
{
.name = "BUFFER_COUNT",
.tag = AUD_OPT_INT,
.valp = &glob_conf.nbuffers,
.descr = "Number of buffers"
},
{ /* End of list */ }
};
static struct audio_pcm_ops coreaudio_pcm_ops = {
.init_out = coreaudio_init_out,
.fini_out = coreaudio_fini_out,
@@ -725,7 +699,6 @@ static struct audio_pcm_ops coreaudio_pcm_ops = {
static struct audio_driver coreaudio_audio_driver = {
.name = "coreaudio",
.descr = "CoreAudio http://developer.apple.com/audio/coreaudio.html",
.options = coreaudio_options,
.init = coreaudio_audio_init,
.fini = coreaudio_audio_fini,
.pcm_ops = &coreaudio_pcm_ops,
audio/dsound_template.h
+3 -3
View File
@@ -167,17 +167,18 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
dsound *s = drv_opaque;
WAVEFORMATEX wfx;
struct audsettings obt_as;
DSoundConf *conf = &s->conf;
#ifdef DSBTYPE_IN
const char *typ = "ADC";
DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;
DSCBUFFERDESC bd;
DSCBCAPS bc;
AudiodevPerDirectionOptions *pdo = s->dev->u.dsound.in;
#else
const char *typ = "DAC";
DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
DSBUFFERDESC bd;
DSBCAPS bc;
AudiodevPerDirectionOptions *pdo = s->dev->u.dsound.out;
#endif
if (!s->FIELD2) {
@@ -193,8 +194,8 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
memset (&bd, 0, sizeof (bd));
bd.dwSize = sizeof (bd);
bd.lpwfxFormat = &wfx;
bd.dwBufferBytes = audio_buffer_bytes(pdo, as, 92880);
#ifdef DSBTYPE_IN
bd.dwBufferBytes = conf->bufsize_in;
hr = IDirectSoundCapture_CreateCaptureBuffer (
s->dsound_capture,
&bd,
@@ -203,7 +204,6 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
);
#else
bd.dwFlags = DSBCAPS_STICKYFOCUS | DSBCAPS_GETCURRENTPOSITION2;
bd.dwBufferBytes = conf->bufsize_out;
hr = IDirectSound_CreateSoundBuffer (
s->dsound,
&bd,
audio/dsoundaudio.c
+18 -43
View File
@@ -32,6 +32,7 @@
#define AUDIO_CAP "dsound"
#include "audio_int.h"
#include "qemu/host-utils.h"
#include <windows.h>
#include <mmsystem.h>
@@ -42,17 +43,11 @@
/* #define DEBUG_DSOUND */
typedef struct {
int bufsize_in;
int bufsize_out;
int latency_millis;
} DSoundConf;
typedef struct {
LPDIRECTSOUND dsound;
LPDIRECTSOUNDCAPTURE dsound_capture;
struct audsettings settings;
DSoundConf conf;
Audiodev *dev;
} dsound;
typedef struct {
@@ -248,9 +243,9 @@ static void GCC_FMT_ATTR (3, 4) dsound_logerr2 (
dsound_log_hresult (hr);
}
static DWORD millis_to_bytes (struct audio_pcm_info *info, DWORD millis)
static uint64_t usecs_to_bytes(struct audio_pcm_info *info, uint32_t usecs)
{
return (millis * info->bytes_per_second) / 1000;
return muldiv64(usecs, info->bytes_per_second, 1000000);
}
#ifdef DEBUG_DSOUND
@@ -478,7 +473,7 @@ static int dsound_run_out (HWVoiceOut *hw, int live)
LPVOID p1, p2;
int bufsize;
dsound *s = ds->s;
DSoundConf *conf = &s->conf;
AudiodevDsoundOptions *dso = &s->dev->u.dsound;
if (!dsb) {
dolog ("Attempt to run empty with playback buffer\n");
@@ -501,14 +496,14 @@ static int dsound_run_out (HWVoiceOut *hw, int live)
len = live << hwshift;
if (ds->first_time) {
if (conf->latency_millis) {
if (dso->latency) {
DWORD cur_blat;
cur_blat = audio_ring_dist (wpos, ppos, bufsize);
ds->first_time = 0;
old_pos = wpos;
old_pos +=
millis_to_bytes (&hw->info, conf->latency_millis) - cur_blat;
usecs_to_bytes(&hw->info, dso->latency) - cur_blat;
old_pos %= bufsize;
old_pos &= ~hw->info.align;
}
@@ -747,12 +742,6 @@ static int dsound_run_in (HWVoiceIn *hw)
return decr;
}
static DSoundConf glob_conf = {
.bufsize_in = 16384,
.bufsize_out = 16384,
.latency_millis = 10
};
static void dsound_audio_fini (void *opaque)
{
HRESULT hr;
@@ -783,13 +772,22 @@ static void dsound_audio_fini (void *opaque)
g_free(s);
}
static void *dsound_audio_init (void)
static void *dsound_audio_init(Audiodev *dev)
{
int err;
HRESULT hr;
dsound *s = g_malloc0(sizeof(dsound));
AudiodevDsoundOptions *dso;
assert(dev->driver == AUDIODEV_DRIVER_DSOUND);
s->dev = dev;
dso = &dev->u.dsound;
if (!dso->has_latency) {
dso->has_latency = true;
dso->latency = 10000; /* 10 ms */
}
s->conf = glob_conf;
hr = CoInitialize (NULL);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not initialize COM\n");
@@ -854,28 +852,6 @@ static void *dsound_audio_init (void)
return s;
}
static struct audio_option dsound_options[] = {
{
.name = "LATENCY_MILLIS",
.tag = AUD_OPT_INT,
.valp = &glob_conf.latency_millis,
.descr = "(undocumented)"
},
{
.name = "BUFSIZE_OUT",
.tag = AUD_OPT_INT,
.valp = &glob_conf.bufsize_out,
.descr = "(undocumented)"
},
{
.name = "BUFSIZE_IN",
.tag = AUD_OPT_INT,
.valp = &glob_conf.bufsize_in,
.descr = "(undocumented)"
},
{ /* End of list */ }
};
static struct audio_pcm_ops dsound_pcm_ops = {
.init_out = dsound_init_out,
.fini_out = dsound_fini_out,
@@ -893,7 +869,6 @@ static struct audio_pcm_ops dsound_pcm_ops = {
static struct audio_driver dsound_audio_driver = {
.name = "dsound",
.descr = "DirectSound http://wikipedia.org/wiki/DirectSound",
.options = dsound_options,
.init = dsound_audio_init,
.fini = dsound_audio_fini,
.pcm_ops = &dsound_pcm_ops,
audio/noaudio.c
+1 -2
View File
@@ -136,7 +136,7 @@ static int no_ctl_in (HWVoiceIn *hw, int cmd, ...)
return 0;
}
static void *no_audio_init (void)
static void *no_audio_init(Audiodev *dev)
{
return &no_audio_init;
}
@@ -163,7 +163,6 @@ static struct audio_pcm_ops no_pcm_ops = {
static struct audio_driver no_audio_driver = {
.name = "none",
.descr = "Timer based audio emulation",
.options = NULL,
.init = no_audio_init,
.fini = no_audio_fini,
.pcm_ops = &no_pcm_ops,
audio/ossaudio.c
+67 -126
View File
@@ -37,16 +37,6 @@
#define USE_DSP_POLICY
#endif
typedef struct OSSConf {
int try_mmap;
int nfrags;
int fragsize;
const char *devpath_out;
const char *devpath_in;
int exclusive;
int policy;
} OSSConf;
typedef struct OSSVoiceOut {
HWVoiceOut hw;
void *pcm_buf;
@@ -56,7 +46,7 @@ typedef struct OSSVoiceOut {
int fragsize;
int mmapped;
int pending;
OSSConf *conf;
Audiodev *dev;
} OSSVoiceOut;
typedef struct OSSVoiceIn {
@@ -65,12 +55,12 @@ typedef struct OSSVoiceIn {
int fd;
int nfrags;
int fragsize;
OSSConf *conf;
Audiodev *dev;
} OSSVoiceIn;
struct oss_params {
int freq;
audfmt_e fmt;
int fmt;
int nchannels;
int nfrags;
int fragsize;
@@ -148,16 +138,16 @@ 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 +155,7 @@ 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 +172,37 @@ 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:
@@ -262,19 +252,25 @@ static int oss_get_version (int fd, int *version, const char *typ)
}
#endif
static int oss_open (int in, struct oss_params *req,
struct oss_params *obt, int *pfd, OSSConf* conf)
static int oss_open(int in, struct oss_params *req, audsettings *as,
struct oss_params *obt, int *pfd, Audiodev *dev)
{
AudiodevOssOptions *oopts = &dev->u.oss;
AudiodevOssPerDirectionOptions *opdo = in ? oopts->in : oopts->out;
int fd;
int oflags = conf->exclusive ? O_EXCL : 0;
int oflags = (oopts->has_exclusive && oopts->exclusive) ? O_EXCL : 0;
audio_buf_info abinfo;
int fmt, freq, nchannels;
int setfragment = 1;
const char *dspname = in ? conf->devpath_in : conf->devpath_out;
const char *dspname = opdo->has_dev ? opdo->dev : "/dev/dsp";
const char *typ = in ? "ADC" : "DAC";
#ifdef USE_DSP_POLICY
int policy = oopts->has_dsp_policy ? oopts->dsp_policy : 5;
#endif
/* Kludge needed to have working mmap on Linux */
oflags |= conf->try_mmap ? O_RDWR : (in ? O_RDONLY : O_WRONLY);
oflags |= (oopts->has_try_mmap && oopts->try_mmap) ?
O_RDWR : (in ? O_RDONLY : O_WRONLY);
fd = open (dspname, oflags | O_NONBLOCK);
if (-1 == fd) {
@@ -285,6 +281,9 @@ static int oss_open (int in, struct oss_params *req,
freq = req->freq;
nchannels = req->nchannels;
fmt = req->fmt;
req->nfrags = opdo->has_buffer_count ? opdo->buffer_count : 4;
req->fragsize = audio_buffer_bytes(
qapi_AudiodevOssPerDirectionOptions_base(opdo), as, 23220);
if (ioctl (fd, SNDCTL_DSP_SAMPLESIZE, &fmt)) {
oss_logerr2 (errno, typ, "Failed to set sample size %d\n", req->fmt);
@@ -308,18 +307,18 @@ static int oss_open (int in, struct oss_params *req,
}
#ifdef USE_DSP_POLICY
if (conf->policy >= 0) {
if (policy >= 0) {
int version;
if (!oss_get_version (fd, &version, typ)) {
trace_oss_version(version);
if (version >= 0x040000) {
int policy = conf->policy;
if (ioctl (fd, SNDCTL_DSP_POLICY, &policy)) {
int policy2 = policy;
if (ioctl(fd, SNDCTL_DSP_POLICY, &policy2)) {
oss_logerr2 (errno, typ,
"Failed to set timing policy to %d\n",
conf->policy);
policy);
goto err;
}
setfragment = 0;
@@ -500,19 +499,18 @@ 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;
Audiodev *dev = drv_opaque;
AudiodevOssOptions *oopts = &dev->u.oss;
oss->fd = -1;
req.fmt = aud_to_ossfmt (as->fmt, as->endianness);
req.freq = as->freq;
req.nchannels = as->nchannels;
req.fragsize = conf->fragsize;
req.nfrags = conf->nfrags;
if (oss_open (0, &req, &obt, &fd, conf)) {
if (oss_open(0, &req, as, &obt, &fd, dev)) {
return -1;
}
@@ -539,7 +537,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
oss->mmapped = 0;
if (conf->try_mmap) {
if (oopts->has_try_mmap && oopts->try_mmap) {
oss->pcm_buf = mmap (
NULL,
hw->samples << hw->info.shift,
@@ -597,7 +595,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
}
oss->fd = fd;
oss->conf = conf;
oss->dev = dev;
return 0;
}
@@ -605,16 +603,12 @@ static int oss_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
int trig;
OSSVoiceOut *oss = (OSSVoiceOut *) hw;
AudiodevOssPerDirectionOptions *opdo = oss->dev->u.oss.out;
switch (cmd) {
case VOICE_ENABLE:
{
va_list ap;
int poll_mode;
va_start (ap, cmd);
poll_mode = va_arg (ap, int);
va_end (ap);
bool poll_mode = opdo->try_poll;
ldebug ("enabling voice\n");
if (poll_mode) {
@@ -667,18 +661,16 @@ 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;
Audiodev *dev = drv_opaque;
oss->fd = -1;
req.fmt = aud_to_ossfmt (as->fmt, as->endianness);
req.freq = as->freq;
req.nchannels = as->nchannels;
req.fragsize = conf->fragsize;
req.nfrags = conf->nfrags;
if (oss_open (1, &req, &obt, &fd, conf)) {
if (oss_open(1, &req, as, &obt, &fd, dev)) {
return -1;
}
@@ -712,7 +704,7 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
}
oss->fd = fd;
oss->conf = conf;
oss->dev = dev;
return 0;
}
@@ -803,16 +795,12 @@ static int oss_read (SWVoiceIn *sw, void *buf, int size)
static int oss_ctl_in (HWVoiceIn *hw, int cmd, ...)
{
OSSVoiceIn *oss = (OSSVoiceIn *) hw;
AudiodevOssPerDirectionOptions *opdo = oss->dev->u.oss.out;
switch (cmd) {
case VOICE_ENABLE:
{
va_list ap;
int poll_mode;
va_start (ap, cmd);
poll_mode = va_arg (ap, int);
va_end (ap);
bool poll_mode = opdo->try_poll;
if (poll_mode) {
oss_poll_in (hw);
@@ -832,82 +820,36 @@ static int oss_ctl_in (HWVoiceIn *hw, int cmd, ...)
return 0;
}
static OSSConf glob_conf = {
.try_mmap = 0,
.nfrags = 4,
.fragsize = 4096,
.devpath_out = "/dev/dsp",
.devpath_in = "/dev/dsp",
.exclusive = 0,
.policy = 5
};
static void *oss_audio_init (void)
static void oss_init_per_direction(AudiodevOssPerDirectionOptions *opdo)
{
OSSConf *conf = g_malloc(sizeof(OSSConf));
*conf = glob_conf;
if (!opdo->has_try_poll) {
opdo->try_poll = true;
opdo->has_try_poll = true;
}
}
if (access(conf->devpath_in, R_OK | W_OK) < 0 ||
access(conf->devpath_out, R_OK | W_OK) < 0) {
g_free(conf);
static void *oss_audio_init(Audiodev *dev)
{
AudiodevOssOptions *oopts;
assert(dev->driver == AUDIODEV_DRIVER_OSS);
oopts = &dev->u.oss;
oss_init_per_direction(oopts->in);
oss_init_per_direction(oopts->out);
if (access(oopts->in->has_dev ? oopts->in->dev : "/dev/dsp",
R_OK | W_OK) < 0 ||
access(oopts->out->has_dev ? oopts->out->dev : "/dev/dsp",
R_OK | W_OK) < 0) {
return NULL;
}
return conf;
return dev;
}
static void oss_audio_fini (void *opaque)
{
g_free(opaque);
}
static struct audio_option oss_options[] = {
{
.name = "FRAGSIZE",
.tag = AUD_OPT_INT,
.valp = &glob_conf.fragsize,
.descr = "Fragment size in bytes"
},
{
.name = "NFRAGS",
.tag = AUD_OPT_INT,
.valp = &glob_conf.nfrags,
.descr = "Number of fragments"
},
{
.name = "MMAP",
.tag = AUD_OPT_BOOL,
.valp = &glob_conf.try_mmap,
.descr = "Try using memory mapped access"
},
{
.name = "DAC_DEV",
.tag = AUD_OPT_STR,
.valp = &glob_conf.devpath_out,
.descr = "Path to DAC device"
},
{
.name = "ADC_DEV",
.tag = AUD_OPT_STR,
.valp = &glob_conf.devpath_in,
.descr = "Path to ADC device"
},
{
.name = "EXCLUSIVE",
.tag = AUD_OPT_BOOL,
.valp = &glob_conf.exclusive,
.descr = "Open device in exclusive mode (vmix won't work)"
},
#ifdef USE_DSP_POLICY
{
.name = "POLICY",
.tag = AUD_OPT_INT,
.valp = &glob_conf.policy,
.descr = "Set the timing policy of the device, -1 to use fragment mode",
},
#endif
{ /* End of list */ }
};
static struct audio_pcm_ops oss_pcm_ops = {
.init_out = oss_init_out,
.fini_out = oss_fini_out,
@@ -925,7 +867,6 @@ static struct audio_pcm_ops oss_pcm_ops = {
static struct audio_driver oss_audio_driver = {
.name = "oss",
.descr = "OSS http://www.opensound.com",
.options = oss_options,
.init = oss_audio_init,
.fini = oss_audio_fini,
.pcm_ops = &oss_pcm_ops,
audio/paaudio.c
+44 -67
View File
@@ -2,6 +2,7 @@
#include "qemu/osdep.h"
#include "qemu-common.h"
#include "audio.h"
#include "qapi/opts-visitor.h"
#include <pulse/pulseaudio.h>
@@ -10,14 +11,7 @@
#include "audio_pt_int.h"
typedef struct {
int samples;
char *server;
char *sink;
char *source;
} PAConf;
typedef struct {
PAConf conf;
Audiodev *dev;
pa_threaded_mainloop *mainloop;
pa_context *context;
} paaudio;
@@ -32,6 +26,7 @@ typedef struct {
void *pcm_buf;
struct audio_pt pt;
paaudio *g;
int samples;
} PAVoiceOut;
typedef struct {
@@ -46,6 +41,7 @@ typedef struct {
const void *read_data;
size_t read_index, read_length;
paaudio *g;
int samples;
} PAVoiceIn;
static void qpa_audio_fini(void *opaque);
@@ -227,7 +223,7 @@ static void *qpa_thread_out (void *arg)
}
}
decr = to_mix = audio_MIN(pa->live, pa->g->conf.samples >> 5);
decr = to_mix = audio_MIN(pa->live, pa->samples >> 5);
rpos = pa->rpos;
if (audio_pt_unlock(&pa->pt, __func__)) {
@@ -319,7 +315,7 @@ static void *qpa_thread_in (void *arg)
}
}
incr = to_grab = audio_MIN(pa->dead, pa->g->conf.samples >> 5);
incr = to_grab = audio_MIN(pa->dead, pa->samples >> 5);
wpos = pa->wpos;
if (audio_pt_unlock(&pa->pt, __func__)) {
@@ -385,21 +381,21 @@ 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 +406,26 @@ 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;
}
}
@@ -546,6 +542,8 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
struct audsettings obt_as = *as;
PAVoiceOut *pa = (PAVoiceOut *) hw;
paaudio *g = pa->g = drv_opaque;
AudiodevPaOptions *popts = &g->dev->u.pa;
AudiodevPaPerDirectionOptions *ppdo = popts->out;
ss.format = audfmt_to_pa (as->fmt, as->endianness);
ss.channels = as->nchannels;
@@ -566,7 +564,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
g,
"qemu",
PA_STREAM_PLAYBACK,
g->conf.sink,
ppdo->has_name ? ppdo->name : NULL,
&ss,
NULL, /* channel map */
&ba, /* buffering attributes */
@@ -578,7 +576,8 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
}
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = g->conf.samples;
hw->samples = pa->samples = audio_buffer_samples(
qapi_AudiodevPaPerDirectionOptions_base(ppdo), &obt_as, 46440);
pa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
pa->rpos = hw->rpos;
if (!pa->pcm_buf) {
@@ -612,6 +611,8 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
struct audsettings obt_as = *as;
PAVoiceIn *pa = (PAVoiceIn *) hw;
paaudio *g = pa->g = drv_opaque;
AudiodevPaOptions *popts = &g->dev->u.pa;
AudiodevPaPerDirectionOptions *ppdo = popts->in;
ss.format = audfmt_to_pa (as->fmt, as->endianness);
ss.channels = as->nchannels;
@@ -623,7 +624,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
g,
"qemu",
PA_STREAM_RECORD,
g->conf.source,
ppdo->has_name ? ppdo->name : NULL,
&ss,
NULL, /* channel map */
NULL, /* buffering attributes */
@@ -635,7 +636,8 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
}
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = g->conf.samples;
hw->samples = pa->samples = audio_buffer_samples(
qapi_AudiodevPaPerDirectionOptions_base(ppdo), &obt_as, 46440);
pa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
pa->wpos = hw->wpos;
if (!pa->pcm_buf) {
@@ -808,13 +810,13 @@ static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...)
}
/* common */
static PAConf glob_conf = {
.samples = 4096,
};
static void *qpa_audio_init (void)
static void *qpa_audio_init(Audiodev *dev)
{
if (glob_conf.server == NULL) {
paaudio *g;
AudiodevPaOptions *popts = &dev->u.pa;
const char *server;
if (!popts->has_server) {
char pidfile[64];
char *runtime;
struct stat st;
@@ -829,8 +831,12 @@ static void *qpa_audio_init (void)
}
}
paaudio *g = g_malloc(sizeof(paaudio));
g->conf = glob_conf;
assert(dev->driver == AUDIODEV_DRIVER_PA);
g = g_malloc(sizeof(paaudio));
server = popts->has_server ? popts->server : NULL;
g->dev = dev;
g->mainloop = NULL;
g->context = NULL;
@@ -840,14 +846,14 @@ static void *qpa_audio_init (void)
}
g->context = pa_context_new (pa_threaded_mainloop_get_api (g->mainloop),
g->conf.server);
server);
if (!g->context) {
goto fail;
}
pa_context_set_state_callback (g->context, context_state_cb, g);
if (pa_context_connect (g->context, g->conf.server, 0, NULL) < 0) {
if (pa_context_connect(g->context, server, 0, NULL) < 0) {
qpa_logerr (pa_context_errno (g->context),
"pa_context_connect() failed\n");
goto fail;
@@ -910,34 +916,6 @@ static void qpa_audio_fini (void *opaque)
g_free(g);
}
struct audio_option qpa_options[] = {
{
.name = "SAMPLES",
.tag = AUD_OPT_INT,
.valp = &glob_conf.samples,
.descr = "buffer size in samples"
},
{
.name = "SERVER",
.tag = AUD_OPT_STR,
.valp = &glob_conf.server,
.descr = "server address"
},
{
.name = "SINK",
.tag = AUD_OPT_STR,
.valp = &glob_conf.sink,
.descr = "sink device name"
},
{
.name = "SOURCE",
.tag = AUD_OPT_STR,
.valp = &glob_conf.source,
.descr = "source device name"
},
{ /* End of list */ }
};
static struct audio_pcm_ops qpa_pcm_ops = {
.init_out = qpa_init_out,
.fini_out = qpa_fini_out,
@@ -955,7 +933,6 @@ static struct audio_pcm_ops qpa_pcm_ops = {
static struct audio_driver pa_audio_driver = {
.name = "pa",
.descr = "http://www.pulseaudio.org/",
.options = qpa_options,
.init = qpa_audio_init,
.fini = qpa_audio_fini,
.pcm_ops = &qpa_pcm_ops,
audio/sdlaudio.c
+18 -32
View File
@@ -44,16 +44,11 @@ typedef struct SDLVoiceOut {
int decr;
} SDLVoiceOut;
static struct {
int nb_samples;
} conf = {
.nb_samples = 1024
};
static struct SDLAudioState {
int exit;
int initialized;
bool driver_created;
Audiodev *dev;
} glob_sdl;
typedef struct SDLAudioState SDLAudioState;
@@ -68,19 +63,19 @@ 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 +87,37 @@ 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:
@@ -265,13 +260,13 @@ 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;
req.format = aud_to_sdlfmt (as->fmt);
req.channels = as->nchannels;
req.samples = conf.nb_samples;
req.samples = audio_buffer_samples(s->dev->u.sdl.out, as, 11610);
req.callback = sdl_callback;
req.userdata = sdl;
@@ -315,7 +310,7 @@ static int sdl_ctl_out (HWVoiceOut *hw, int cmd, ...)
return 0;
}
static void *sdl_audio_init (void)
static void *sdl_audio_init(Audiodev *dev)
{
SDLAudioState *s = &glob_sdl;
if (s->driver_created) {
@@ -329,6 +324,7 @@ static void *sdl_audio_init (void)
}
s->driver_created = true;
s->dev = dev;
return s;
}
@@ -338,18 +334,9 @@ static void sdl_audio_fini (void *opaque)
sdl_close (s);
SDL_QuitSubSystem (SDL_INIT_AUDIO);
s->driver_created = false;
s->dev = NULL;
}
static struct audio_option sdl_options[] = {
{
.name = "SAMPLES",
.tag = AUD_OPT_INT,
.valp = &conf.nb_samples,
.descr = "Size of SDL buffer in samples"
},
{ /* End of list */ }
};
static struct audio_pcm_ops sdl_pcm_ops = {
.init_out = sdl_init_out,
.fini_out = sdl_fini_out,
@@ -361,7 +348,6 @@ static struct audio_pcm_ops sdl_pcm_ops = {
static struct audio_driver sdl_audio_driver = {
.name = "sdl",
.descr = "SDL http://www.libsdl.org",
.options = sdl_options,
.init = sdl_audio_init,
.fini = sdl_audio_fini,
.pcm_ops = &sdl_pcm_ops,
audio/spiceaudio.c
+3 -8
View File
@@ -77,7 +77,7 @@ static const SpiceRecordInterface record_sif = {
.base.minor_version = SPICE_INTERFACE_RECORD_MINOR,
};
static void *spice_audio_init (void)
static void *spice_audio_init(Audiodev *dev)
{
if (!using_spice) {
return NULL;
@@ -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);
@@ -373,10 +373,6 @@ static int line_in_ctl (HWVoiceIn *hw, int cmd, ...)
return 0;
}
static struct audio_option audio_options[] = {
{ /* end of list */ },
};
static struct audio_pcm_ops audio_callbacks = {
.init_out = line_out_init,
.fini_out = line_out_fini,
@@ -394,7 +390,6 @@ static struct audio_pcm_ops audio_callbacks = {
static struct audio_driver spice_audio_driver = {
.name = "spice",
.descr = "spice audio driver",
.options = audio_options,
.init = spice_audio_init,
.fini = spice_audio_fini,
.pcm_ops = &audio_callbacks,
audio/wavaudio.c
+19 -56
View File
@@ -24,6 +24,7 @@
#include "qemu/osdep.h"
#include "qemu/host-utils.h"
#include "qemu/timer.h"
#include "qapi/opts-visitor.h"
#include "audio.h"
#define AUDIO_CAP "wav"
@@ -37,11 +38,6 @@ typedef struct WAVVoiceOut {
int total_samples;
} WAVVoiceOut;
typedef struct {
struct audsettings settings;
const char *wav_path;
} WAVConf;
static int wav_run_out (HWVoiceOut *hw, int live)
{
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
@@ -112,25 +108,30 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
0x02, 0x00, 0x44, 0xac, 0x00, 0x00, 0x10, 0xb1, 0x02, 0x00, 0x04,
0x00, 0x10, 0x00, 0x64, 0x61, 0x74, 0x61, 0x00, 0x00, 0x00, 0x00
};
WAVConf *conf = drv_opaque;
struct audsettings wav_as = conf->settings;
Audiodev *dev = drv_opaque;
AudiodevWavOptions *wopts = &dev->u.wav;
struct audsettings wav_as = audiodev_to_audsettings(dev->u.wav.out);
const char *wav_path = wopts->has_path ? wopts->path : "qemu.wav";
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;
@@ -151,10 +152,10 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
le_store (hdr + 28, hw->info.freq << (bits16 + stereo), 4);
le_store (hdr + 32, 1 << (bits16 + stereo), 2);
wav->f = fopen (conf->wav_path, "wb");
wav->f = fopen(wav_path, "wb");
if (!wav->f) {
dolog ("Failed to open wave file `%s'\nReason: %s\n",
conf->wav_path, strerror (errno));
wav_path, strerror(errno));
g_free (wav->pcm_buf);
wav->pcm_buf = NULL;
return -1;
@@ -222,54 +223,17 @@ static int wav_ctl_out (HWVoiceOut *hw, int cmd, ...)
return 0;
}
static WAVConf glob_conf = {
.settings.freq = 44100,
.settings.nchannels = 2,
.settings.fmt = AUD_FMT_S16,
.wav_path = "qemu.wav"
};
static void *wav_audio_init (void)
static void *wav_audio_init(Audiodev *dev)
{
WAVConf *conf = g_malloc(sizeof(WAVConf));
*conf = glob_conf;
return conf;
assert(dev->driver == AUDIODEV_DRIVER_WAV);
return dev;
}
static void wav_audio_fini (void *opaque)
{
ldebug ("wav_fini");
g_free(opaque);
}
static struct audio_option wav_options[] = {
{
.name = "FREQUENCY",
.tag = AUD_OPT_INT,
.valp = &glob_conf.settings.freq,
.descr = "Frequency"
},
{
.name = "FORMAT",
.tag = AUD_OPT_FMT,
.valp = &glob_conf.settings.fmt,
.descr = "Format"
},
{
.name = "DAC_FIXED_CHANNELS",
.tag = AUD_OPT_INT,
.valp = &glob_conf.settings.nchannels,
.descr = "Number of channels (1 - mono, 2 - stereo)"
},
{
.name = "PATH",
.tag = AUD_OPT_STR,
.valp = &glob_conf.wav_path,
.descr = "Path to wave file"
},
{ /* End of list */ }
};
static struct audio_pcm_ops wav_pcm_ops = {
.init_out = wav_init_out,
.fini_out = wav_fini_out,
@@ -281,7 +245,6 @@ static struct audio_pcm_ops wav_pcm_ops = {
static struct audio_driver wav_audio_driver = {
.name = "wav",
.descr = "WAV renderer http://wikipedia.org/wiki/WAV",
.options = wav_options,
.init = wav_audio_init,
.fini = wav_audio_fini,
.pcm_ops = &wav_pcm_ops,
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) {

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