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[Qemu-devel] [PATCH v3 8/8] audio: -audiodev command line option
|
From: |
Kővágó, Zoltán |
|
Subject: |
[Qemu-devel] [PATCH v3 8/8] audio: -audiodev command line option |
|
Date: |
Thu, 18 Jun 2015 18:43:50 +0200 |
This patch adds an -audiodev command line option, and deprecates the
QEMU_* environment variables for audio backend configuration. It's
syntax is similar to existing options (-netdev, -device, etc): -audiodev
driver_name,property=value,...
Audio drivers now get an Audiodev * as config paramters, instead of the
global audio_option structs. There is some code in audio/audio_legacy.c
that converts the old environment variables to audiodev options (this
way backends do not have to worry about legacy options). It also
contains a replacement of -audio-help, which prints out the equivalent
-audiodev based config of the currently specified environment variables.
Although now it's possible to specify multiple -audiodev options on
command line, multiple audio backends are not supported yet.
Signed-off-by: Kővágó, Zoltán <address@hidden>
---
Changes from v2:
* fixed alsa threshold
* updated to audio qapi changes
Changes from v1:
* updated to everything usecs without suffix
* better rounding when converting between usecs and frames/samples/bytes
* bugfixes in audio_legacy.c
* fixed code broken by qemu_opts changes by Markus
audio/Makefile.objs | 2 +-
audio/alsaaudio.c | 311 ++++++--------------
audio/audio.c | 741 +++++++++++++-----------------------------------
audio/audio.h | 23 +-
audio/audio_int.h | 7 +-
audio/audio_legacy.c | 328 +++++++++++++++++++++
audio/audio_template.h | 13 +-
audio/coreaudio.c | 49 +---
audio/dsound_template.h | 6 +-
audio/dsoundaudio.c | 60 ++--
audio/noaudio.c | 3 +-
audio/ossaudio.c | 154 +++-------
audio/paaudio.c | 80 ++----
audio/sdlaudio.c | 24 +-
audio/spiceaudio.c | 7 +-
audio/wavaudio.c | 60 +---
qemu-options.hx | 230 ++++++++++++++-
vl.c | 9 +-
18 files changed, 997 insertions(+), 1110 deletions(-)
create mode 100644 audio/audio_legacy.c
diff --git a/audio/Makefile.objs b/audio/Makefile.objs
index 481d1aa..9d8f579 100644
--- a/audio/Makefile.objs
+++ b/audio/Makefile.objs
@@ -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-$(CONFIG_SDL) += sdlaudio.o
common-obj-$(CONFIG_OSS) += ossaudio.o
common-obj-$(CONFIG_SPICE) += spiceaudio.o
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 2b28b99..cfe4aec 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -22,6 +22,7 @@
* THE SOFTWARE.
*/
#include <alsa/asoundlib.h>
+#include "qapi-visit.h"
#include "qemu-common.h"
#include "qemu/main-loop.h"
#include "audio.h"
@@ -34,28 +35,9 @@
#define AUDIO_CAP "alsa"
#include "audio_int.h"
-typedef struct ALSAConf {
- int size_in_usec_in;
- int size_in_usec_out;
- const char *pcm_name_in;
- const char *pcm_name_out;
- unsigned int buffer_size_in;
- unsigned int period_size_in;
- unsigned int buffer_size_out;
- unsigned int period_size_out;
- unsigned int threshold;
-
- int buffer_size_in_overridden;
- int period_size_in_overridden;
-
- int buffer_size_out_overridden;
- int period_size_out_overridden;
-} ALSAConf;
-
struct pollhlp {
snd_pcm_t *handle;
struct pollfd *pfds;
- ALSAConf *conf;
int count;
int mask;
};
@@ -67,6 +49,7 @@ typedef struct ALSAVoiceOut {
void *pcm_buf;
snd_pcm_t *handle;
struct pollhlp pollhlp;
+ Audiodev *dev;
} ALSAVoiceOut;
typedef struct ALSAVoiceIn {
@@ -74,16 +57,13 @@ typedef struct ALSAVoiceIn {
snd_pcm_t *handle;
void *pcm_buf;
struct pollhlp pollhlp;
+ Audiodev *dev;
} ALSAVoiceIn;
struct alsa_params_req {
int freq;
snd_pcm_format_t fmt;
int nchannels;
- int size_in_usec;
- int override_mask;
- unsigned int buffer_size;
- unsigned int period_size;
};
struct alsa_params_obt {
@@ -409,7 +389,8 @@ static int alsa_to_audfmt (snd_pcm_format_t alsafmt,
AudioFormat *fmt,
static void alsa_dump_info (struct alsa_params_req *req,
struct alsa_params_obt *obt,
- snd_pcm_format_t obtfmt)
+ snd_pcm_format_t obtfmt,
+ AudiodevPerDirectionOptions *pdo)
{
dolog ("parameter | requested value | obtained value\n");
dolog ("format | %10d | %10d\n", req->fmt, obtfmt);
@@ -417,8 +398,9 @@ static void alsa_dump_info (struct alsa_params_req *req,
req->nchannels, obt->nchannels);
dolog ("frequency | %10d | %10d\n", req->freq, obt->freq);
dolog ("============================================\n");
- dolog ("requested: buffer size %d period size %d\n",
- req->buffer_size, req->period_size);
+ dolog ("requested: buffer size %" PRId64 " buffer count %" PRId64 "\n",
+ pdo->has_buffer_len ? pdo->buffer_len : 0,
+ pdo->has_buffer_len ? pdo->buffer_len : 0);
dolog ("obtained: samples %ld\n", obt->samples);
}
@@ -452,23 +434,25 @@ static void alsa_set_threshold (snd_pcm_t *handle,
snd_pcm_uframes_t threshold)
}
}
-static int alsa_open (int in, struct alsa_params_req *req,
- struct alsa_params_obt *obt, snd_pcm_t **handlep,
- ALSAConf *conf)
+static int alsa_open(bool in, struct alsa_params_req *req,
+ struct alsa_params_obt *obt, snd_pcm_t **handlep,
+ Audiodev *dev)
{
+ AudiodevPerDirectionOptions *pdo = in ? dev->in : dev->out;
+ AudiodevAlsaOptions *aopts = dev->alsa;
+ AudiodevAlsaPerDirectionOptions *apdo =
+ in ? aopts->alsa_in : aopts->alsa_out;
snd_pcm_t *handle;
snd_pcm_hw_params_t *hw_params;
int err;
- int size_in_usec;
unsigned int freq, nchannels;
- const char *pcm_name = in ? conf->pcm_name_in : conf->pcm_name_out;
+ const char *pcm_name = apdo->has_dev ? apdo->dev : "default";
snd_pcm_uframes_t obt_buffer_size;
const char *typ = in ? "ADC" : "DAC";
snd_pcm_format_t obtfmt;
freq = req->freq;
nchannels = req->nchannels;
- size_in_usec = req->size_in_usec;
snd_pcm_hw_params_alloca (&hw_params);
@@ -528,79 +512,49 @@ static int alsa_open (int in, struct alsa_params_req *req,
goto err;
}
- if (req->buffer_size) {
- unsigned long obt;
+ if (pdo->buffer_count) {
+ if (pdo->buffer_len) {
+ int64_t req = pdo->buffer_len * pdo->buffer_count;
- if (size_in_usec) {
int dir = 0;
- unsigned int btime = req->buffer_size;
+ unsigned int btime = req;
- err = snd_pcm_hw_params_set_buffer_time_near (
- handle,
- hw_params,
- &btime,
- &dir
- );
- obt = btime;
- }
- else {
- snd_pcm_uframes_t bsize = req->buffer_size;
+ err = snd_pcm_hw_params_set_buffer_time_near(
+ handle, hw_params, &btime, &dir);
- err = snd_pcm_hw_params_set_buffer_size_near (
- handle,
- hw_params,
- &bsize
- );
- obt = bsize;
- }
- if (err < 0) {
- alsa_logerr2 (err, typ, "Failed to set buffer %s to %d\n",
- size_in_usec ? "time" : "size", req->buffer_size);
- goto err;
- }
+ if (err < 0) {
+ alsa_logerr2(err, typ,
+ "Failed to set buffer time to %" PRId64 "\n",
+ req);
+ goto err;
+ }
- if ((req->override_mask & 2) && (obt - req->buffer_size))
- dolog ("Requested buffer %s %u was rejected, using %lu\n",
- size_in_usec ? "time" : "size", req->buffer_size, obt);
+ if (pdo->has_buffer_count && btime != req) {
+ dolog("Requested buffer time %" PRId64
+ " was rejected, using %u\n", req, btime);
+ }
+ } else {
+ dolog("Can't set buffer_count without buffer_size!\n");
+ }
}
- if (req->period_size) {
- unsigned long obt;
+ if (pdo->buffer_len) {
+ int dir = 0;
+ unsigned int ptime = pdo->buffer_len;
- if (size_in_usec) {
- int dir = 0;
- unsigned int ptime = req->period_size;
-
- err = snd_pcm_hw_params_set_period_time_near (
- handle,
- hw_params,
- &ptime,
- &dir
- );
- obt = ptime;
- }
- else {
- int dir = 0;
- snd_pcm_uframes_t psize = req->period_size;
-
- err = snd_pcm_hw_params_set_period_size_near (
- handle,
- hw_params,
- &psize,
- &dir
- );
- obt = psize;
- }
+ err = snd_pcm_hw_params_set_period_time_near(handle, hw_params, &ptime,
+ &dir);
if (err < 0) {
- alsa_logerr2 (err, typ, "Failed to set period %s to %d\n",
- size_in_usec ? "time" : "size", req->period_size);
+ alsa_logerr2(err, typ, "Failed to set period time to %" PRId64
"\n",
+ pdo->buffer_len);
goto err;
}
- if (((req->override_mask & 1) && (obt - req->period_size)))
- dolog ("Requested period %s %u was rejected, using %lu\n",
- size_in_usec ? "time" : "size", req->period_size, obt);
+ if (pdo->has_buffer_len && ptime != pdo->buffer_len) {
+ dolog("Requested period time %" PRId64 " was rejected, using %d\n",
+ pdo->buffer_len, ptime);
+ }
}
err = snd_pcm_hw_params (handle, hw_params);
@@ -632,33 +586,10 @@ static int alsa_open (int in, struct alsa_params_req *req,
goto err;
}
- if (!in && conf->threshold) {
- snd_pcm_uframes_t threshold;
- int bytes_per_sec;
-
- bytes_per_sec = freq << (nchannels == 2);
-
- switch (obt->fmt) {
- case AUDIO_FORMAT_S8:
- case AUDIO_FORMAT_U8:
- break;
-
- case AUDIO_FORMAT_S16:
- case AUDIO_FORMAT_U16:
- bytes_per_sec <<= 1;
- break;
-
- case AUDIO_FORMAT_S32:
- case AUDIO_FORMAT_U32:
- bytes_per_sec <<= 2;
- break;
-
- default:
- abort();
- }
-
- threshold = (conf->threshold * bytes_per_sec) / 1000;
- alsa_set_threshold (handle, threshold);
+ if (!in && aopts->has_threshold && aopts->threshold) {
+ struct audsettings as = { .freq = freq };
+ alsa_set_threshold(handle,
+ audio_buffer_frames(pdo, &as, aopts->threshold));
}
obt->nchannels = nchannels;
@@ -671,11 +602,11 @@ static int alsa_open (int in, struct alsa_params_req *req,
obt->nchannels != req->nchannels ||
obt->freq != req->freq) {
dolog ("Audio parameters for %s\n", typ);
- alsa_dump_info (req, obt, obtfmt);
+ alsa_dump_info (req, obt, obtfmt, pdo);
}
#ifdef DEBUG
- alsa_dump_info (req, obt, obtfmt);
+ alsa_dump_info (req, obt, obtfmt, pdo);
#endif
return 0;
@@ -801,19 +732,13 @@ static int alsa_init_out(HWVoiceOut *hw, struct
audsettings *as,
struct alsa_params_obt obt;
snd_pcm_t *handle;
struct audsettings obt_as;
- ALSAConf *conf = drv_opaque;
+ Audiodev *dev = drv_opaque;
req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
req.freq = as->freq;
req.nchannels = as->nchannels;
- req.period_size = conf->period_size_out;
- req.buffer_size = conf->buffer_size_out;
- req.size_in_usec = conf->size_in_usec_out;
- req.override_mask =
- (conf->period_size_out_overridden ? 1 : 0) |
- (conf->buffer_size_out_overridden ? 2 : 0);
- if (alsa_open (0, &req, &obt, &handle, conf)) {
+ if (alsa_open (0, &req, &obt, &handle, dev)) {
return -1;
}
@@ -834,7 +759,7 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings
*as,
}
alsa->handle = handle;
- alsa->pollhlp.conf = conf;
+ alsa->dev = dev;
return 0;
}
@@ -874,16 +799,12 @@ static int alsa_voice_ctl (snd_pcm_t *handle, const char
*typ, int ctl)
static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+ AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->alsa->alsa_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 = !apdo->has_try_poll || apdo->try_poll;
ldebug ("enabling voice\n");
if (poll_mode && alsa_poll_out (hw)) {
@@ -912,19 +833,13 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings
*as, void *drv_opaque)
struct alsa_params_obt obt;
snd_pcm_t *handle;
struct audsettings obt_as;
- ALSAConf *conf = drv_opaque;
+ Audiodev *dev = drv_opaque;
req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
req.freq = as->freq;
req.nchannels = as->nchannels;
- req.period_size = conf->period_size_in;
- req.buffer_size = conf->buffer_size_in;
- req.size_in_usec = conf->size_in_usec_in;
- req.override_mask =
- (conf->period_size_in_overridden ? 1 : 0) |
- (conf->buffer_size_in_overridden ? 2 : 0);
- if (alsa_open (1, &req, &obt, &handle, conf)) {
+ if (alsa_open (1, &req, &obt, &handle, dev)) {
return -1;
}
@@ -945,7 +860,7 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings
*as, void *drv_opaque)
}
alsa->handle = handle;
- alsa->pollhlp.conf = conf;
+ alsa->dev = dev;
return 0;
}
@@ -1087,16 +1002,12 @@ static int alsa_read (SWVoiceIn *sw, void *buf, int
size)
static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+ AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->alsa->alsa_in;
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 = !apdo->has_try_poll || apdo->try_poll;
ldebug ("enabling voice\n");
if (poll_mode && alsa_poll_in (hw)) {
@@ -1119,88 +1030,35 @@ static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
return -1;
}
-static ALSAConf glob_conf = {
- .buffer_size_out = 4096,
- .period_size_out = 1024,
- .pcm_name_out = "default",
- .pcm_name_in = "default",
-};
-
-static void *alsa_audio_init (void)
+static void *alsa_audio_init(Audiodev *dev)
{
- ALSAConf *conf = g_malloc(sizeof(ALSAConf));
- *conf = glob_conf;
- return conf;
+ assert(dev->kind == AUDIODEV_DRIVER_ALSA);
+
+ /* need to define them, as otherwise alsa produces no sound
+ * doesn't set has_* so alsa_open can identify it wasn't set by the user */
+ if (!dev->out->has_buffer_count) {
+ dev->out->buffer_count = 4;
+ }
+ if (!dev->out->has_buffer_len) {
+ dev->out->buffer_len = 23219; /* 1024 frames assuming 44100Hz */
+ }
+
+ /* OptsVisitor sets unspecified optional fields to zero, but do not depend
+ * on it... */
+ if (!dev->in->has_buffer_count) {
+ dev->in->buffer_count = 0;
+ }
+ if (!dev->in->has_buffer_len) {
+ dev->in->buffer_len = 0;
+ }
+
+ return dev;
}
static void alsa_audio_fini (void *opaque)
{
- g_free(opaque);
}
-static struct audio_option alsa_options[] = {
- {
- .name = "DAC_SIZE_IN_USEC",
- .tag = AUD_OPT_BOOL,
- .valp = &glob_conf.size_in_usec_out,
- .descr = "DAC period/buffer size in microseconds (otherwise in
frames)"
- },
- {
- .name = "DAC_PERIOD_SIZE",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.period_size_out,
- .descr = "DAC period size (0 to go with system default)",
- .overriddenp = &glob_conf.period_size_out_overridden
- },
- {
- .name = "DAC_BUFFER_SIZE",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.buffer_size_out,
- .descr = "DAC buffer size (0 to go with system default)",
- .overriddenp = &glob_conf.buffer_size_out_overridden
- },
- {
- .name = "ADC_SIZE_IN_USEC",
- .tag = AUD_OPT_BOOL,
- .valp = &glob_conf.size_in_usec_in,
- .descr =
- "ADC period/buffer size in microseconds (otherwise in frames)"
- },
- {
- .name = "ADC_PERIOD_SIZE",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.period_size_in,
- .descr = "ADC period size (0 to go with system default)",
- .overriddenp = &glob_conf.period_size_in_overridden
- },
- {
- .name = "ADC_BUFFER_SIZE",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.buffer_size_in,
- .descr = "ADC buffer size (0 to go with system default)",
- .overriddenp = &glob_conf.buffer_size_in_overridden
- },
- {
- .name = "THRESHOLD",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.threshold,
- .descr = "(undocumented)"
- },
- {
- .name = "DAC_DEV",
- .tag = AUD_OPT_STR,
- .valp = &glob_conf.pcm_name_out,
- .descr = "DAC device name (for instance dmix)"
- },
- {
- .name = "ADC_DEV",
- .tag = AUD_OPT_STR,
- .valp = &glob_conf.pcm_name_in,
- .descr = "ADC device name"
- },
- { /* End of list */ }
-};
-
static struct audio_pcm_ops alsa_pcm_ops = {
.init_out = alsa_init_out,
.fini_out = alsa_fini_out,
@@ -1218,7 +1076,6 @@ static struct audio_pcm_ops alsa_pcm_ops = {
struct audio_driver alsa_audio_driver = {
.name = "alsa",
.descr = "ALSA http://www.alsa-project.org";,
- .options = alsa_options,
.init = alsa_audio_init,
.fini = alsa_audio_fini,
.pcm_ops = &alsa_pcm_ops,
diff --git a/audio/audio.c b/audio/audio.c
index 334c935..76d4b8f 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -24,7 +24,10 @@
#include "hw/hw.h"
#include "audio.h"
#include "monitor/monitor.h"
+#include "qapi-visit.h"
+#include "qapi/opts-visitor.h"
#include "qemu/timer.h"
+#include "qemu/config-file.h"
#include "sysemu/sysemu.h"
#define AUDIO_CAP "audio"
@@ -42,59 +45,14 @@
The 1st one is the one used by default, that is the reason
that we generate the list.
*/
-static struct audio_driver *drvtab[] = {
+struct audio_driver *drvtab[] = {
#ifdef CONFIG_SPICE
&spice_audio_driver,
#endif
CONFIG_AUDIO_DRIVERS
&no_audio_driver,
- &wav_audio_driver
-};
-
-struct fixed_settings {
- int enabled;
- int nb_voices;
- int greedy;
- struct audsettings settings;
-};
-
-static struct {
- struct fixed_settings fixed_out;
- struct fixed_settings fixed_in;
- union {
- int hertz;
- int64_t ticks;
- } period;
- int try_poll_in;
- int try_poll_out;
-} conf = {
- .fixed_out = { /* DAC fixed settings */
- .enabled = 1,
- .nb_voices = 1,
- .greedy = 1,
- .settings = {
- .freq = 44100,
- .nchannels = 2,
- .fmt = AUDIO_FORMAT_S16,
- .endianness = AUDIO_HOST_ENDIANNESS,
- }
- },
-
- .fixed_in = { /* ADC fixed settings */
- .enabled = 1,
- .nb_voices = 1,
- .greedy = 1,
- .settings = {
- .freq = 44100,
- .nchannels = 2,
- .fmt = AUDIO_FORMAT_S16,
- .endianness = AUDIO_HOST_ENDIANNESS,
- }
- },
-
- .period = { .hertz = 100 },
- .try_poll_in = 1,
- .try_poll_out = 1,
+ &wav_audio_driver,
+ NULL
};
static AudioState glob_audio_state;
@@ -113,9 +71,6 @@ const struct mixeng_volume nominal_volume = {
#ifdef AUDIO_IS_FLAWLESS_AND_NO_CHECKS_ARE_REQURIED
#error No its not
#else
-static void audio_print_options (const char *prefix,
- struct audio_option *opt);
-
int audio_bug (const char *funcname, int cond)
{
if (cond) {
@@ -123,16 +78,9 @@ int audio_bug (const char *funcname, int cond)
AUD_log (NULL, "A bug was just triggered in %s\n", funcname);
if (!shown) {
- struct audio_driver *d;
-
shown = 1;
AUD_log (NULL, "Save all your work and restart without audio\n");
- AUD_log (NULL, "Please send bug report to address@hidden");
AUD_log (NULL, "I am sorry\n");
- d = glob_audio_state.drv;
- if (d) {
- audio_print_options (d->name, d->options);
- }
}
AUD_log (NULL, "Context:\n");
@@ -194,139 +142,6 @@ void *audio_calloc (const char *funcname, int nmemb,
size_t size)
return g_malloc0 (len);
}
-static char *audio_alloc_prefix (const char *s)
-{
- const char qemu_prefix[] = "QEMU_";
- size_t len, i;
- char *r, *u;
-
- if (!s) {
- return NULL;
- }
-
- len = strlen (s);
- r = g_malloc (len + sizeof (qemu_prefix));
-
- u = r + sizeof (qemu_prefix) - 1;
-
- pstrcpy (r, len + sizeof (qemu_prefix), qemu_prefix);
- pstrcat (r, len + sizeof (qemu_prefix), s);
-
- for (i = 0; i < len; ++i) {
- u[i] = qemu_toupper(u[i]);
- }
-
- return r;
-}
-
-static const char *audio_audfmt_to_string (AudioFormat fmt)
-{
- switch (fmt) {
- case AUDIO_FORMAT_U8:
- return "U8";
-
- case AUDIO_FORMAT_U16:
- return "U16";
-
- case AUDIO_FORMAT_S8:
- return "S8";
-
- case AUDIO_FORMAT_S16:
- return "S16";
-
- case AUDIO_FORMAT_U32:
- return "U32";
-
- case AUDIO_FORMAT_S32:
- return "S32";
-
- default:
- abort();
- }
-
- dolog ("Bogus audfmt %d returning S16\n", fmt);
- return "S16";
-}
-
-static AudioFormat audio_string_to_audfmt (const char *s, AudioFormat defval,
- int *defaultp)
-{
- if (!strcasecmp (s, "u8")) {
- *defaultp = 0;
- return AUDIO_FORMAT_U8;
- }
- else if (!strcasecmp (s, "u16")) {
- *defaultp = 0;
- return AUDIO_FORMAT_U16;
- }
- else if (!strcasecmp (s, "u32")) {
- *defaultp = 0;
- return AUDIO_FORMAT_U32;
- }
- else if (!strcasecmp (s, "s8")) {
- *defaultp = 0;
- return AUDIO_FORMAT_S8;
- }
- else if (!strcasecmp (s, "s16")) {
- *defaultp = 0;
- return AUDIO_FORMAT_S16;
- }
- else if (!strcasecmp (s, "s32")) {
- *defaultp = 0;
- return AUDIO_FORMAT_S32;
- }
- else {
- dolog ("Bogus audio format `%s' using %s\n",
- s, audio_audfmt_to_string (defval));
- *defaultp = 1;
- return defval;
- }
-}
-
-static AudioFormat audio_get_conf_fmt (const char *envname,
- AudioFormat defval,
- int *defaultp)
-{
- const char *var = getenv (envname);
- if (!var) {
- *defaultp = 1;
- return defval;
- }
- return audio_string_to_audfmt (var, defval, defaultp);
-}
-
-static int audio_get_conf_int (const char *key, int defval, int *defaultp)
-{
- int val;
- char *strval;
-
- strval = getenv (key);
- if (strval) {
- *defaultp = 0;
- val = atoi (strval);
- return val;
- }
- else {
- *defaultp = 1;
- return defval;
- }
-}
-
-static const char *audio_get_conf_str (const char *key,
- const char *defval,
- int *defaultp)
-{
- const char *val = getenv (key);
- if (!val) {
- *defaultp = 1;
- return defval;
- }
- else {
- *defaultp = 0;
- return val;
- }
-}
-
void AUD_vlog (const char *cap, const char *fmt, va_list ap)
{
if (cap) {
@@ -345,161 +160,6 @@ void AUD_log (const char *cap, const char *fmt, ...)
va_end (ap);
}
-static void audio_print_options (const char *prefix,
- struct audio_option *opt)
-{
- char *uprefix;
-
- if (!prefix) {
- dolog ("No prefix specified\n");
- return;
- }
-
- if (!opt) {
- dolog ("No options\n");
- return;
- }
-
- uprefix = audio_alloc_prefix (prefix);
-
- for (; opt->name; opt++) {
- const char *state = "default";
- printf (" %s_%s: ", uprefix, opt->name);
-
- if (opt->overriddenp && *opt->overriddenp) {
- state = "current";
- }
-
- switch (opt->tag) {
- case AUD_OPT_BOOL:
- {
- int *intp = opt->valp;
- printf ("boolean, %s = %d\n", state, *intp ? 1 : 0);
- }
- break;
-
- case AUD_OPT_INT:
- {
- int *intp = opt->valp;
- printf ("integer, %s = %d\n", state, *intp);
- }
- break;
-
- case AUD_OPT_FMT:
- {
- AudioFormat *fmtp = opt->valp;
- printf (
- "format, %s = %s, (one of: U8 S8 U16 S16 U32 S32)\n",
- state,
- audio_audfmt_to_string (*fmtp)
- );
- }
- break;
-
- case AUD_OPT_STR:
- {
- const char **strp = opt->valp;
- printf ("string, %s = %s\n",
- state,
- *strp ? *strp : "(not set)");
- }
- break;
-
- default:
- printf ("???\n");
- dolog ("Bad value tag for option %s_%s %d\n",
- uprefix, opt->name, opt->tag);
- break;
- }
- printf (" %s\n", opt->descr);
- }
-
- g_free (uprefix);
-}
-
-static void audio_process_options (const char *prefix,
- struct audio_option *opt)
-{
- char *optname;
- const char qemu_prefix[] = "QEMU_";
- size_t preflen, optlen;
-
- if (audio_bug (AUDIO_FUNC, !prefix)) {
- dolog ("prefix = NULL\n");
- return;
- }
-
- if (audio_bug (AUDIO_FUNC, !opt)) {
- dolog ("opt = NULL\n");
- return;
- }
-
- preflen = strlen (prefix);
-
- for (; opt->name; opt++) {
- size_t len, i;
- int def;
-
- if (!opt->valp) {
- dolog ("Option value pointer for `%s' is not set\n",
- opt->name);
- continue;
- }
-
- len = strlen (opt->name);
- /* len of opt->name + len of prefix + size of qemu_prefix
- * (includes trailing zero) + zero + underscore (on behalf of
- * sizeof) */
- optlen = len + preflen + sizeof (qemu_prefix) + 1;
- optname = g_malloc (optlen);
-
- pstrcpy (optname, optlen, qemu_prefix);
-
- /* copy while upper-casing, including trailing zero */
- for (i = 0; i <= preflen; ++i) {
- optname[i + sizeof (qemu_prefix) - 1] = qemu_toupper(prefix[i]);
- }
- pstrcat (optname, optlen, "_");
- pstrcat (optname, optlen, opt->name);
-
- def = 1;
- switch (opt->tag) {
- case AUD_OPT_BOOL:
- case AUD_OPT_INT:
- {
- int *intp = opt->valp;
- *intp = audio_get_conf_int (optname, *intp, &def);
- }
- break;
-
- case AUD_OPT_FMT:
- {
- AudioFormat *fmtp = opt->valp;
- *fmtp = audio_get_conf_fmt (optname, *fmtp, &def);
- }
- break;
-
- case AUD_OPT_STR:
- {
- const char **strp = opt->valp;
- *strp = audio_get_conf_str (optname, *strp, &def);
- }
- break;
-
- default:
- dolog ("Bad value tag for option `%s' - %d\n",
- optname, opt->tag);
- break;
- }
-
- if (!opt->overriddenp) {
- opt->overriddenp = &opt->overridden;
- }
- *opt->overriddenp = !def;
- g_free (optname);
- }
-}
-
static void audio_print_settings (struct audsettings *as)
{
dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels);
@@ -1120,7 +780,7 @@ static void audio_reset_timer (AudioState *s)
{
if (audio_is_timer_needed ()) {
timer_mod (s->ts,
- qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + conf.period.ticks);
+ qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + s->period_ticks);
}
else {
timer_del (s->ts);
@@ -1196,7 +856,7 @@ void AUD_set_active_out (SWVoiceOut *sw, int on)
if (!hw->enabled) {
hw->enabled = 1;
if (s->vm_running) {
- hw->pcm_ops->ctl_out (hw, VOICE_ENABLE, conf.try_poll_out);
+ hw->pcm_ops->ctl_out (hw, VOICE_ENABLE);
audio_reset_timer (s);
}
}
@@ -1241,7 +901,7 @@ void AUD_set_active_in (SWVoiceIn *sw, int on)
if (!hw->enabled) {
hw->enabled = 1;
if (s->vm_running) {
- hw->pcm_ops->ctl_in (hw, VOICE_ENABLE, conf.try_poll_in);
+ hw->pcm_ops->ctl_in (hw, VOICE_ENABLE);
audio_reset_timer (s);
}
}
@@ -1558,168 +1218,10 @@ void audio_run (const char *msg)
#endif
}
-static struct audio_option audio_options[] = {
- /* DAC */
- {
- .name = "DAC_FIXED_SETTINGS",
- .tag = AUD_OPT_BOOL,
- .valp = &conf.fixed_out.enabled,
- .descr = "Use fixed settings for host DAC"
- },
- {
- .name = "DAC_FIXED_FREQ",
- .tag = AUD_OPT_INT,
- .valp = &conf.fixed_out.settings.freq,
- .descr = "Frequency for fixed host DAC"
- },
- {
- .name = "DAC_FIXED_FMT",
- .tag = AUD_OPT_FMT,
- .valp = &conf.fixed_out.settings.fmt,
- .descr = "Format for fixed host DAC"
- },
- {
- .name = "DAC_FIXED_CHANNELS",
- .tag = AUD_OPT_INT,
- .valp = &conf.fixed_out.settings.nchannels,
- .descr = "Number of channels for fixed DAC (1 - mono, 2 - stereo)"
- },
- {
- .name = "DAC_VOICES",
- .tag = AUD_OPT_INT,
- .valp = &conf.fixed_out.nb_voices,
- .descr = "Number of voices for DAC"
- },
- {
- .name = "DAC_TRY_POLL",
- .tag = AUD_OPT_BOOL,
- .valp = &conf.try_poll_out,
- .descr = "Attempt using poll mode for DAC"
- },
- /* ADC */
- {
- .name = "ADC_FIXED_SETTINGS",
- .tag = AUD_OPT_BOOL,
- .valp = &conf.fixed_in.enabled,
- .descr = "Use fixed settings for host ADC"
- },
- {
- .name = "ADC_FIXED_FREQ",
- .tag = AUD_OPT_INT,
- .valp = &conf.fixed_in.settings.freq,
- .descr = "Frequency for fixed host ADC"
- },
- {
- .name = "ADC_FIXED_FMT",
- .tag = AUD_OPT_FMT,
- .valp = &conf.fixed_in.settings.fmt,
- .descr = "Format for fixed host ADC"
- },
- {
- .name = "ADC_FIXED_CHANNELS",
- .tag = AUD_OPT_INT,
- .valp = &conf.fixed_in.settings.nchannels,
- .descr = "Number of channels for fixed ADC (1 - mono, 2 - stereo)"
- },
- {
- .name = "ADC_VOICES",
- .tag = AUD_OPT_INT,
- .valp = &conf.fixed_in.nb_voices,
- .descr = "Number of voices for ADC"
- },
- {
- .name = "ADC_TRY_POLL",
- .tag = AUD_OPT_BOOL,
- .valp = &conf.try_poll_in,
- .descr = "Attempt using poll mode for ADC"
- },
- /* Misc */
- {
- .name = "TIMER_PERIOD",
- .tag = AUD_OPT_INT,
- .valp = &conf.period.hertz,
- .descr = "Timer period in HZ (0 - use lowest possible)"
- },
- { /* End of list */ }
-};
-
-static void audio_pp_nb_voices (const char *typ, int nb)
+static int audio_driver_init(AudioState *s, struct audio_driver *drv,
+ Audiodev *dev)
{
- switch (nb) {
- case 0:
- printf ("Does not support %s\n", typ);
- break;
- case 1:
- printf ("One %s voice\n", typ);
- break;
- case INT_MAX:
- printf ("Theoretically supports many %s voices\n", typ);
- break;
- default:
- printf ("Theoretically supports up to %d %s voices\n", nb, typ);
- break;
- }
-
-}
-
-void AUD_help (void)
-{
- size_t i;
-
- audio_process_options ("AUDIO", audio_options);
- for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
- struct audio_driver *d = drvtab[i];
- if (d->options) {
- audio_process_options (d->name, d->options);
- }
- }
-
- printf ("Audio options:\n");
- audio_print_options ("AUDIO", audio_options);
- printf ("\n");
-
- printf ("Available drivers:\n");
-
- for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
- struct audio_driver *d = drvtab[i];
-
- printf ("Name: %s\n", d->name);
- printf ("Description: %s\n", d->descr);
-
- audio_pp_nb_voices ("playback", d->max_voices_out);
- audio_pp_nb_voices ("capture", d->max_voices_in);
-
- if (d->options) {
- printf ("Options:\n");
- audio_print_options (d->name, d->options);
- }
- else {
- printf ("No options\n");
- }
- printf ("\n");
- }
-
- printf (
- "Options are settable through environment variables.\n"
- "Example:\n"
-#ifdef _WIN32
- " set QEMU_AUDIO_DRV=wav\n"
- " set QEMU_WAV_PATH=c:\\tune.wav\n"
-#else
- " export QEMU_AUDIO_DRV=wav\n"
- " export QEMU_WAV_PATH=$HOME/tune.wav\n"
- "(for csh replace export with setenv in the above)\n"
-#endif
- " qemu ...\n\n"
- );
-}
-
-static int audio_driver_init (AudioState *s, struct audio_driver *drv)
-{
- if (drv->options) {
- audio_process_options (drv->name, drv->options);
- }
- s->drv_opaque = drv->init ();
+ s->drv_opaque = drv->init(dev);
if (s->drv_opaque) {
audio_init_nb_voices_out (drv);
@@ -1743,11 +1245,11 @@ static void audio_vm_change_state_handler (void
*opaque, int running,
s->vm_running = running;
while ((hwo = audio_pcm_hw_find_any_enabled_out (hwo))) {
- hwo->pcm_ops->ctl_out (hwo, op, conf.try_poll_out);
+ hwo->pcm_ops->ctl_out (hwo, op);
}
while ((hwi = audio_pcm_hw_find_any_enabled_in (hwi))) {
- hwi->pcm_ops->ctl_in (hwi, op, conf.try_poll_in);
+ hwi->pcm_ops->ctl_in (hwi, op);
}
audio_reset_timer (s);
}
@@ -1786,6 +1288,8 @@ static void audio_atexit (void)
if (s->drv) {
s->drv->fini (s->drv_opaque);
}
+
+ qapi_free_Audiodev(s->dev);
}
static const VMStateDescription vmstate_audio = {
@@ -1797,18 +1301,37 @@ static const VMStateDescription vmstate_audio = {
}
};
-static void audio_init (void)
+static Audiodev *parse_option(QemuOpts *opts, Error **errp);
+static int audio_init(Audiodev *dev)
{
size_t i;
int done = 0;
- const char *drvname;
+ const char *drvname = NULL;
VMChangeStateEntry *e;
AudioState *s = &glob_audio_state;
+ QemuOptsList *list = NULL; /* silence gcc warning about uninitialized
+ * variable */
if (s->drv) {
- return;
+ if (dev) {
+ dolog("Cannot create more than one audio backend, sorry\n");
+ qapi_free_Audiodev(dev);
+ }
+ return -1;
}
+ if (dev) {
+ drvname = AudiodevDriver_lookup[dev->kind];
+ } else {
+ audio_handle_legacy_opts();
+ list = qemu_find_opts("audiodev");
+ dev = parse_option(QTAILQ_FIRST(&list->head), &error_abort);
+ if (!dev) {
+ exit(1);
+ }
+ }
+ s->dev = dev;
+
QLIST_INIT (&s->hw_head_out);
QLIST_INIT (&s->hw_head_in);
QLIST_INIT (&s->cap_head);
@@ -1819,10 +1342,8 @@ static void audio_init (void)
hw_error("Could not create audio timer\n");
}
- audio_process_options ("AUDIO", audio_options);
-
- s->nb_hw_voices_out = conf.fixed_out.nb_voices;
- s->nb_hw_voices_in = conf.fixed_in.nb_voices;
+ s->nb_hw_voices_out = dev->out->voices;
+ s->nb_hw_voices_in = dev->in->voices;
if (s->nb_hw_voices_out <= 0) {
dolog ("Bogus number of playback voices %d, setting to 1\n",
@@ -1836,17 +1357,12 @@ static void audio_init (void)
s->nb_hw_voices_in = 0;
}
- {
- int def;
- drvname = audio_get_conf_str ("QEMU_AUDIO_DRV", NULL, &def);
- }
-
if (drvname) {
int found = 0;
- for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
+ for (i = 0; drvtab[i]; i++) {
if (!strcmp (drvname, drvtab[i]->name)) {
- done = !audio_driver_init (s, drvtab[i]);
+ done = !audio_driver_init (s, drvtab[i], dev);
found = 1;
break;
}
@@ -1854,20 +1370,24 @@ static void audio_init (void)
if (!found) {
dolog ("Unknown audio driver `%s'\n", drvname);
- dolog ("Run with -audio-help to list available drivers\n");
}
- }
-
- if (!done) {
- for (i = 0; !done && i < ARRAY_SIZE (drvtab); i++) {
- if (drvtab[i]->can_be_default) {
- done = !audio_driver_init (s, drvtab[i]);
+ } else {
+ for (i = 0; !done && drvtab[i]; i++) {
+ QemuOpts *opts = qemu_opts_find(list, drvtab[i]->name);
+ if (opts) {
+ qapi_free_Audiodev(dev);
+ dev = parse_option(opts, &error_abort);
+ if (!dev) {
+ exit(1);
+ }
+ s->dev = dev;
+ done = !audio_driver_init(s, drvtab[i], dev);
}
}
}
if (!done) {
- done = !audio_driver_init (s, &no_audio_driver);
+ done = !audio_driver_init (s, &no_audio_driver, dev);
if (!done) {
hw_error("Could not initialize audio subsystem\n");
}
@@ -1876,16 +1396,16 @@ static void audio_init (void)
}
}
- if (conf.period.hertz <= 0) {
- if (conf.period.hertz < 0) {
- dolog ("warning: Timer period is negative - %d "
- "treating as zero\n",
- conf.period.hertz);
+ if (dev->timer_period <= 0) {
+ if (dev->timer_period < 0) {
+ dolog ("warning: Timer period is negative - %" PRId64
+ " treating as zero\n",
+ dev->timer_period);
}
- conf.period.ticks = 1;
+ s->period_ticks = 1;
} else {
- conf.period.ticks =
- muldiv64 (1, get_ticks_per_sec (), conf.period.hertz);
+ s->period_ticks =
+ muldiv64(dev->timer_period, get_ticks_per_sec(), 1000000);
}
e = qemu_add_vm_change_state_handler (audio_vm_change_state_handler, s);
@@ -1896,11 +1416,12 @@ static void audio_init (void)
QLIST_INIT (&s->card_head);
vmstate_register (NULL, 0, &vmstate_audio, s);
+ return 0;
}
void AUD_register_card (const char *name, QEMUSoundCard *card)
{
- audio_init ();
+ audio_init(NULL);
card->name = g_strdup (name);
memset (&card->entries, 0, sizeof (card->entries));
QLIST_INSERT_HEAD (&glob_audio_state.card_head, card, entries);
@@ -2070,3 +1591,137 @@ void AUD_set_volume_in (SWVoiceIn *sw, int mute,
uint8_t lvol, uint8_t rvol)
}
}
}
+
+QemuOptsList qemu_audiodev_opts = {
+ .name = "audiodev",
+ .head = QTAILQ_HEAD_INITIALIZER(qemu_audiodev_opts.head),
+ .implied_opt_name = "driver",
+ .desc = {
+ /*
+ * no elements => accept any params
+ * sanity checking will happen later
+ */
+ { /* end of list */ }
+ },
+};
+
+static void set_per_direction_defaults(AudiodevPerDirectionOptions *pdo)
+{
+ if (!pdo->has_fixed_settings) {
+ pdo->has_fixed_settings = true;
+ pdo->fixed_settings = true;
+ }
+ if (!pdo->has_frequency) {
+ pdo->has_frequency = true;
+ pdo->frequency = 44100;
+ }
+ if (!pdo->has_channels) {
+ pdo->has_channels = true;
+ pdo->channels = 2;
+ }
+ if (!pdo->has_voices) {
+ pdo->has_voices = true;
+ pdo->voices = 1;
+ }
+ if (!pdo->has_format) {
+ pdo->has_format = true;
+ pdo->format = AUDIO_FORMAT_S16;
+ }
+}
+
+static Audiodev *parse_option(QemuOpts *opts, Error **errp)
+{
+ Error *local_err = NULL;
+ OptsVisitor *ov = opts_visitor_new(opts);
+ Audiodev *dev = NULL;
+ visit_type_Audiodev(opts_get_visitor(ov), &dev, NULL, &local_err);
+ opts_visitor_cleanup(ov);
+
+ if (local_err) {
+ error_propagate(errp, local_err);
+ return NULL;
+ }
+
+ set_per_direction_defaults(dev->in);
+ set_per_direction_defaults(dev->out);
+
+ if (!dev->has_timer_period) {
+ dev->has_timer_period = true;
+ dev->timer_period = 10000; /* 100Hz -> 10ms */
+ }
+
+ return dev;
+}
+
+static int each_option(void *opaque, QemuOpts *opts, Error **errp)
+{
+ Audiodev *dev = parse_option(opts, errp);
+ if (!dev) {
+ return -1;
+ }
+ return audio_init(dev);
+}
+
+void audio_set_options(void)
+{
+ if (qemu_opts_foreach(qemu_find_opts("audiodev"), each_option, NULL,
+ &error_abort)) {
+ exit(1);
+ }
+}
+
+audsettings audiodev_to_audsettings(AudiodevPerDirectionOptions *pdo)
+{
+ return (audsettings) {
+ .freq = pdo->frequency,
+ .nchannels = pdo->channels,
+ .fmt = pdo->format,
+ .endianness = AUDIO_HOST_ENDIANNESS,
+ };
+}
+
+int audioformat_bytes_per_sample(AudioFormat fmt)
+{
+ switch (fmt) {
+ case AUDIO_FORMAT_U8:
+ case AUDIO_FORMAT_S8:
+ return 1;
+
+ case AUDIO_FORMAT_U16:
+ case AUDIO_FORMAT_S16:
+ return 2;
+
+ case AUDIO_FORMAT_U32:
+ case AUDIO_FORMAT_S32:
+ return 4;
+
+ case AUDIO_FORMAT_MAX:
+ ;
+ }
+ abort();
+}
+
+
+/* frames = freq * usec / 1e6 */
+int audio_buffer_frames(AudiodevPerDirectionOptions *pdo,
+ audsettings *as, int def_usecs)
+{
+ uint64_t usecs = pdo->has_buffer_len ? pdo->buffer_len : def_usecs;
+ return (as->freq * usecs + 500000) / 1000000;
+}
+
+/* samples = channels * frames = channels * freq * usec / 1e6 */
+int audio_buffer_samples(AudiodevPerDirectionOptions *pdo,
+ audsettings *as, int def_usecs)
+{
+ return as->nchannels * audio_buffer_frames(pdo, as, def_usecs);
+}
+
+/* bytes = bytes_per_sample * samples =
+ * bytes_per_sample * channels * freq * usec / 1e6 */
+int audio_buffer_bytes(AudiodevPerDirectionOptions *pdo,
+ audsettings *as, int def_usecs)
+{
+ return audio_buffer_samples(pdo, as, def_usecs) *
+ audioformat_bytes_per_sample(as->fmt);
+}
diff --git a/audio/audio.h b/audio/audio.h
index e300511..177a673 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -24,7 +24,10 @@
#ifndef QEMU_AUDIO_H
#define QEMU_AUDIO_H
+#include <stdarg.h>
#include "config-host.h"
+#include "qapi-types.h"
+#include "qemu/option.h"
#include "qemu/queue.h"
typedef void (*audio_callback_fn) (void *opaque, int avail);
@@ -35,12 +38,21 @@ typedef void (*audio_callback_fn) (void *opaque, int avail);
#define AUDIO_HOST_ENDIANNESS 0
#endif
-struct audsettings {
+typedef struct audsettings {
int freq;
int nchannels;
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,
@@ -77,10 +89,11 @@ typedef struct QEMUAudioTimeStamp {
uint64_t old_ts;
} QEMUAudioTimeStamp;
+extern QemuOptsList qemu_audiodev_opts;
+
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 (
@@ -154,4 +167,8 @@ static inline void *advance (void *p, int incr)
int wav_start_capture (CaptureState *s, const char *path, int freq,
int bits, int nchannels);
+void audio_set_options(void);
+void audio_handle_legacy_opts(void);
+void audio_legacy_help(void);
+
#endif /* audio.h */
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 566df5e..c4539e7 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -143,8 +143,7 @@ struct SWVoiceIn {
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;
@@ -190,6 +189,7 @@ struct SWVoiceCap {
struct AudioState {
struct audio_driver *drv;
+ Audiodev *dev;
void *drv_opaque;
QEMUTimer *ts;
@@ -200,6 +200,7 @@ struct AudioState {
int nb_hw_voices_out;
int nb_hw_voices_in;
int vm_running;
+ int64_t period_ticks;
};
extern struct audio_driver no_audio_driver;
@@ -213,6 +214,8 @@ extern struct audio_driver pa_audio_driver;
extern struct audio_driver spice_audio_driver;
extern const struct mixeng_volume nominal_volume;
+extern struct audio_driver *drvtab[];
+
void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as);
void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int
len);
diff --git a/audio/audio_legacy.c b/audio/audio_legacy.c
new file mode 100644
index 0000000..88f577d
--- /dev/null
+++ b/audio/audio_legacy.c
@@ -0,0 +1,328 @@
+#include "audio.h"
+#include "qemu-common.h"
+#include "qemu/config-file.h"
+
+#define AUDIO_CAP "audio-legacy"
+#include "audio_int.h"
+
+typedef enum EnvTransform {
+ ENV_TRANSFORM_NONE,
+ ENV_TRANSFORM_BOOL,
+ ENV_TRANSFORM_FMT,
+ ENV_TRANSFORM_FRAMES_TO_USECS_IN,
+ ENV_TRANSFORM_FRAMES_TO_USECS_OUT,
+ ENV_TRANSFORM_SAMPLES_TO_USECS_IN,
+ ENV_TRANSFORM_SAMPLES_TO_USECS_OUT,
+ ENV_TRANSFORM_BYTES_TO_USECS_IN,
+ ENV_TRANSFORM_BYTES_TO_USECS_OUT,
+ ENV_TRANSFORM_MILLIS_TO_USECS,
+ ENV_TRANSFORM_HZ_TO_USECS,
+} EnvTransform;
+
+typedef struct SimpleEnvMap {
+ const char *name;
+ const char *option;
+ EnvTransform transform;
+} SimpleEnvMap;
+
+SimpleEnvMap global_map[] = {
+ /* DAC/out settings */
+ { "QEMU_AUDIO_DAC_FIXED_SETTINGS", "out.fixed-settings",
+ ENV_TRANSFORM_BOOL },
+ { "QEMU_AUDIO_DAC_FIXED_FREQ", "out.frequency" },
+ { "QEMU_AUDIO_DAC_FIXED_FMT", "out.format", ENV_TRANSFORM_FMT },
+ { "QEMU_AUDIO_DAC_FIXED_CHANNELS", "out.channels" },
+ { "QEMU_AUDIO_DAC_VOICES", "out.voices" },
+
+ /* ADC/in settings */
+ { "QEMU_AUDIO_ADC_FIXED_SETTINGS", "in.fixed-settings",
+ ENV_TRANSFORM_BOOL },
+ { "QEMU_AUDIO_ADC_FIXED_FREQ", "in.frequency" },
+ { "QEMU_AUDIO_ADC_FIXED_FMT", "in.format", ENV_TRANSFORM_FMT },
+ { "QEMU_AUDIO_ADC_FIXED_CHANNELS", "in.channels" },
+ { "QEMU_AUDIO_ADC_VOICES", "in.voices" },
+
+ /* general */
+ { "QEMU_AUDIO_TIMER_PERIOD", "timer-period", ENV_TRANSFORM_HZ_TO_USECS },
+ { /* End of list */ }
+};
+
+SimpleEnvMap alsa_map[] = {
+ { "QEMU_AUDIO_DAC_TRY_POLL", "alsa-out.try-poll", ENV_TRANSFORM_BOOL },
+ { "QEMU_AUDIO_ADC_TRY_POLL", "alsa-in.try-poll", ENV_TRANSFORM_BOOL },
+
+ { "QEMU_ALSA_THRESHOLD", "threshold", ENV_TRANSFORM_MILLIS_TO_USECS },
+ { "QEMU_ALSA_DAC_DEV", "alsa-out.dev" },
+ { "QEMU_ALSA_ADC_DEV", "alsa-in.dev" },
+
+ { /* End of list */ }
+};
+
+SimpleEnvMap coreaudio_map[] = {
+ { "QEMU_COREAUDIO_BUFFER_SIZE", "buffer-len",
+ ENV_TRANSFORM_FRAMES_TO_USECS_OUT },
+ { "QEMU_COREAUDIO_BUFFER_COUNT", "buffer-count" },
+
+ { /* End of list */ }
+};
+
+SimpleEnvMap dsound_map[] = {
+ { "QEMU_DSOUND_LATENCY_MILLIS", "latency", ENV_TRANSFORM_MILLIS_TO_USECS },
+ { "QEMU_DSOUND_BUFSIZE_OUT", "out.buffer-len",
+ ENV_TRANSFORM_BYTES_TO_USECS_OUT },
+ { "QEMU_DSOUND_BUFSIZE_IN", "in.buffer-len",
+ ENV_TRANSFORM_BYTES_TO_USECS_IN },
+
+ { /* End of list */ }
+};
+
+SimpleEnvMap oss_map[] = {
+ { "QEMU_AUDIO_DAC_TRY_POLL", "oss-out.try-poll", ENV_TRANSFORM_BOOL },
+ { "QEMU_AUDIO_ADC_TRY_POLL", "oss-in.try-poll", ENV_TRANSFORM_BOOL },
+
+ { "QEMU_OSS_FRAGSIZE", "buffer-len", ENV_TRANSFORM_BYTES_TO_USECS_OUT },
+ { "QEMU_OSS_NFRAGS", "buffer-count" },
+ { "QEMU_OSS_MMAP", "try-mmap", ENV_TRANSFORM_BOOL },
+ { "QEMU_OSS_DAC_DEV", "oss-out.dev" },
+ { "QEMU_OSS_ADC_DEV", "oss-in.dev" },
+ { "QEMU_OSS_EXCLUSIVE", "exclusive", ENV_TRANSFORM_BOOL },
+ { "QEMU_OSS_POLICY", "dsp-policy" },
+
+ { /* End of list */ }
+};
+
+SimpleEnvMap pa_map[] = {
+ { "QEMU_PA_SAMPLES", "buffer", ENV_TRANSFORM_SAMPLES_TO_USECS_OUT },
+ { "QEMU_PA_SERVER", "server" },
+ { "QEMU_PA_SINK", "sink" },
+ { "QEMU_PA_SOURCE", "source" },
+
+ { /* End of list */ }
+};
+
+SimpleEnvMap sdl_map[] = {
+ { "QEMU_SDL_SAMPLES", "buffer-len", ENV_TRANSFORM_SAMPLES_TO_USECS_OUT },
+ { /* End of list */ }
+};
+
+SimpleEnvMap wav_map[] = {
+ { "QEMU_WAV_FREQUENCY", "out.frequency" },
+ { "QEMU_WAV_FORMAT", "out.format", ENV_TRANSFORM_FMT },
+ { "QEMU_WAV_DAC_FIXED_CHANNELS", "out.channels" },
+ { "QEMU_WAV_PATH", "path" },
+ { /* End of list */ }
+};
+
+static unsigned long long toull(const char *str)
+{
+ unsigned long long ret;
+ if (parse_uint_full(str, &ret, 10)) {
+ dolog("Invalid boolean value `%s'\n", str);
+ exit(1);
+ }
+ return ret;
+}
+
+/* non reentrant typesafe or anything, but enough in this small c file */
+static const char *tostr(unsigned long long val)
+{
+ #define LEN ((CHAR_BIT * sizeof(int) - 1) / 3 + 2)
+ static char ret[LEN];
+ snprintf(ret, LEN, "%llu", val);
+ return ret;
+}
+
+static uint64_t frames_to_usecs(QemuOpts *opts, uint64_t frames, bool in)
+{
+ const char *opt = in ? "in.frequency" : "out.frequency";
+ uint64_t freq = qemu_opt_get_number(opts, opt, 44100);
+ return (frames * 1000000 + freq/2) / freq;
+}
+
+static uint64_t samples_to_usecs(QemuOpts *opts, uint64_t samples, bool in)
+{
+ const char *opt = in ? "in.channels" : "out.channels";
+ uint64_t channels = qemu_opt_get_number(opts, opt, 2);
+ return frames_to_usecs(opts, samples/channels, in);
+}
+
+static uint64_t bytes_to_usecs(QemuOpts *opts, uint64_t bytes, bool in)
+{
+ const char *opt = in ? "in.format" : "out.format";
+ const char *val = qemu_opt_get(opts, opt);
+ uint64_t bytes_per_sample = (val ? toull(val) : 16) / 8;
+ return samples_to_usecs(opts, bytes * bytes_per_sample, in);
+}
+
+static const char *transform_val(QemuOpts *opts, const char *val,
+ EnvTransform transform)
+{
+ switch (transform) {
+ case ENV_TRANSFORM_NONE:
+ return val;
+
+ case ENV_TRANSFORM_BOOL:
+ return toull(val) ? "on" : "off";
+
+ case ENV_TRANSFORM_FMT:
+ if (strcasecmp(val, "u8") == 0) {
+ return "u8";
+ } else if (strcasecmp(val, "u16") == 0) {
+ return "u16";
+ } else if (strcasecmp(val, "u32") == 0) {
+ return "u32";
+ } else if (strcasecmp(val, "s8") == 0) {
+ return "s8";
+ } else if (strcasecmp(val, "s16") == 0) {
+ return "s16";
+ } else if (strcasecmp(val, "s32") == 0) {
+ return "s32";
+ } else {
+ dolog("Invalid audio format `%s'\n", val);
+ exit(1);
+ }
+
+ case ENV_TRANSFORM_FRAMES_TO_USECS_IN:
+ return tostr(frames_to_usecs(opts, toull(val), true));
+ case ENV_TRANSFORM_FRAMES_TO_USECS_OUT:
+ return tostr(frames_to_usecs(opts, toull(val), false));
+
+ case ENV_TRANSFORM_SAMPLES_TO_USECS_IN:
+ return tostr(samples_to_usecs(opts, toull(val), true));
+ case ENV_TRANSFORM_SAMPLES_TO_USECS_OUT:
+ return tostr(samples_to_usecs(opts, toull(val), false));
+
+ case ENV_TRANSFORM_BYTES_TO_USECS_IN:
+ return tostr(bytes_to_usecs(opts, toull(val), true));
+ case ENV_TRANSFORM_BYTES_TO_USECS_OUT:
+ return tostr(bytes_to_usecs(opts, toull(val), false));
+
+ case ENV_TRANSFORM_MILLIS_TO_USECS:
+ return tostr(toull(val) * 1000);
+
+ case ENV_TRANSFORM_HZ_TO_USECS:
+ return tostr(1000000 / toull(val));
+ }
+
+ abort(); /* it's unreachable, gcc */
+}
+
+static void handle_env_opts(QemuOpts *opts, SimpleEnvMap *map)
+{
+ while (map->name) {
+ const char *val = getenv(map->name);
+
+ if (val) {
+ qemu_opt_set(opts, map->option,
+ transform_val(opts, val, map->transform),
+ &error_abort);
+ }
+
+ ++map;
+ }
+}
+
+static void handle_alsa_side(QemuOpts *opts, int period, int buffer,
+ const char *usec_env, const char *period_env,
+ const char *buffer_env, const char *usec_opt,
+ const char *count_opt, bool in)
+{
+ char *usec_s, *period_s, *buffer_s;
+ bool usec = false;
+
+ usec_s = getenv(usec_env);
+ if (usec_s) {
+ usec = toull(usec_s);
+ }
+
+ period_s = getenv(period_env);
+ if (period_s) {
+ period = toull(period_s);
+ }
+ if (!usec) {
+ period = frames_to_usecs(opts, period, in);
+ }
+ if (period_s) {
+ qemu_opt_set(opts, usec_opt, tostr(period), &error_abort);
+ }
+
+ buffer_s = getenv(buffer_env);
+ if (buffer_s) {
+ buffer = toull(buffer_s);
+ if (!usec) {
+ buffer = frames_to_usecs(opts, buffer, in);
+ }
+ printf("buffer %d period %d\n", buffer, period);
+ qemu_opt_set(opts, count_opt, tostr((buffer+period/2)/period),
+ &error_abort);
+ }
+}
+
+static void handle_alsa(QemuOpts *opts)
+{
+ handle_alsa_side(opts, 1024, 4096,
+ "QEMU_ALSA_DAC_SIZE_IN_USEC", "QEMU_ALSA_DAC_PERIOD_SIZE",
+ "QEMU_ALSA_DAC_BUFFER_SIZE",
+ "out.buffer-len", "out.buffer-count", false);
+ handle_alsa_side(opts, 0, 0,
+ "QEMU_ALSA_ADC_SIZE_IN_USEC", "QEMU_ALSA_ADC_PERIOD_SIZE",
+ "QEMU_ALSA_ADC_BUFFER_SIZE",
+ "in.buffer-len", "in.buffer-count", true);
+}
+
+static void legacy_opt(const char *drv)
+{
+ QemuOpts *opts;
+ opts = qemu_opts_create(qemu_find_opts("audiodev"), drv, true,
+ &error_abort);
+ qemu_opt_set(opts, "driver", drv, &error_abort);
+
+ handle_env_opts(opts, global_map);
+
+ if (strcmp(drv, "alsa") == 0) {
+ handle_env_opts(opts, alsa_map);
+ handle_alsa(opts);
+ } else if (strcmp(drv, "oss") == 0) {
+ handle_env_opts(opts, oss_map);
+ } else if (strcmp(drv, "pa") == 0) {
+ handle_env_opts(opts, pa_map);
+ } else if (strcmp(drv, "sdl") == 0) {
+ handle_env_opts(opts, sdl_map);
+ } else if (strcmp(drv, "wav") == 0) {
+ handle_env_opts(opts, wav_map);
+ }
+}
+
+void audio_handle_legacy_opts(void)
+{
+ const char *drv = getenv("QEMU_AUDIO_DRV");
+
+ if (drv) {
+ legacy_opt(drv);
+ } else {
+ struct audio_driver **drv;
+ for (drv = drvtab; *drv; ++drv) {
+ if ((*drv)->can_be_default) {
+ legacy_opt((*drv)->name);
+ }
+ }
+ }
+}
+
+static int legacy_help_each(void *opaque, QemuOpts *opts, Error **errp)
+{
+ printf("-audiodev ");
+ qemu_opts_print(opts, ",");
+ printf("\n");
+ return 0;
+}
+
+void audio_legacy_help(void)
+{
+ printf("Environment variable based configuration deprecated.\n");
+ printf("Please use the new -audiodev option.\n");
+
+ audio_handle_legacy_opts();
+ printf("\nEquivalent -audiodev to your current environment variables:\n");
+ qemu_opts_foreach(qemu_find_opts("audiodev"), legacy_help_each, NULL,
NULL);
+}
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 99b27b2..096b2b3 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -302,8 +302,10 @@ static HW *glue (audio_pcm_hw_add_new_, TYPE) (struct
audsettings *as)
static HW *glue (audio_pcm_hw_add_, TYPE) (struct audsettings *as)
{
HW *hw;
+ AudioState *s = &glob_audio_state;
+ AudiodevPerDirectionOptions *pdo = s->dev->TYPE;
- 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 +333,11 @@ static SW *glue (audio_pcm_create_voice_pair_, TYPE) (
SW *sw;
HW *hw;
struct audsettings hw_as;
+ AudioState *s = &glob_audio_state;
+ AudiodevPerDirectionOptions *pdo = s->dev->TYPE;
- 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 +402,7 @@ SW *glue (AUD_open_, TYPE) (
)
{
AudioState *s = &glob_audio_state;
+ AudiodevPerDirectionOptions *pdo = s->dev->TYPE;
if (audio_bug (AUDIO_FUNC, !card || !name || !callback_fn || !as)) {
dolog ("card=%p name=%p callback_fn=%p as=%p\n",
@@ -422,7 +427,7 @@ 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;
}
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index 6dfd63e..dfa5e79 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -34,11 +34,6 @@
static int isAtexit;
-typedef struct {
- int buffer_frames;
- int nbuffers;
-} CoreaudioConf;
-
typedef struct coreaudioVoiceOut {
HWVoiceOut hw;
pthread_mutex_t mutex;
@@ -292,7 +287,9 @@ 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;
+ AudiodevPerDirectionOptions *pdo = dev->out;
+ int frames;
/* create mutex */
err = pthread_mutex_init(&core->mutex, NULL);
@@ -334,16 +331,17 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct
audsettings *as,
return -1;
}
- if (frameRange.mMinimum > conf->buffer_frames) {
+ frames = audio_buffer_frames(pdo, 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 */
@@ -377,7 +375,8 @@ 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 = (pdo->has_buffer_count ? pdo->buffer_count : 4) *
+ core->audioDevicePropertyBufferFrameSize;
/* get StreamFormat */
propertySize = sizeof(core->outputStreamBasicDescription);
@@ -497,41 +496,16 @@ 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;
-
atexit(coreaudio_atexit);
- 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,
@@ -543,7 +517,6 @@ static struct audio_pcm_ops coreaudio_pcm_ops = {
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,
diff --git a/audio/dsound_template.h b/audio/dsound_template.h
index b439f33..96181ef 100644
--- a/audio/dsound_template.h
+++ b/audio/dsound_template.h
@@ -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->in;
#else
const char *typ = "DAC";
DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
DSBUFFERDESC bd;
DSBCAPS bc;
+ AudiodevPerDirectionOptions *pdo = s->dev->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,
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index e9472c1..7e7b3f2 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -42,16 +42,10 @@
/* #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 {
@@ -247,9 +241,9 @@ 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
@@ -477,7 +471,7 @@ 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->dsound;
if (!dsb) {
dolog ("Attempt to run empty with playback buffer\n");
@@ -500,14 +494,14 @@ 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;
}
@@ -746,12 +740,6 @@ 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;
@@ -782,13 +770,22 @@ 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->kind == AUDIODEV_DRIVER_DSOUND);
+ s->dev = dev;
+ dso = dev->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");
@@ -853,28 +850,6 @@ 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,
@@ -892,7 +867,6 @@ static struct audio_pcm_ops dsound_pcm_ops = {
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,
diff --git a/audio/noaudio.c b/audio/noaudio.c
index 50db1f3..4c94a26 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -134,7 +134,7 @@ 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;
}
@@ -161,7 +161,6 @@ static struct audio_pcm_ops no_pcm_ops = {
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,
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 02a3a95..a5e7f7c 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -29,6 +29,7 @@
#include "qemu-common.h"
#include "qemu/main-loop.h"
#include "qemu/host-utils.h"
+#include "qapi-visit.h"
#include "audio.h"
#include "trace.h"
@@ -39,16 +40,6 @@
#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;
@@ -58,7 +49,7 @@ typedef struct OSSVoiceOut {
int fragsize;
int mmapped;
int pending;
- OSSConf *conf;
+ Audiodev *dev;
} OSSVoiceOut;
typedef struct OSSVoiceIn {
@@ -67,12 +58,12 @@ typedef struct OSSVoiceIn {
int fd;
int nfrags;
int fragsize;
- OSSConf *conf;
+ Audiodev *dev;
} OSSVoiceIn;
struct oss_params {
int freq;
- AudioFormat fmt;
+ int fmt;
int nchannels;
int nfrags;
int fragsize;
@@ -264,19 +255,26 @@ 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->oss;
+ AudiodevOssPerDirectionOptions *opdo = in ? oopts->oss_in : oopts->oss_out;
+ AudiodevPerDirectionOptions *pdo = in ? dev->in : dev->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) {
@@ -287,6 +285,8 @@ static int oss_open (int in, struct oss_params *req,
freq = req->freq;
nchannels = req->nchannels;
fmt = req->fmt;
+ req->nfrags = pdo->has_buffer_count ? pdo->buffer_count : 4;
+ req->fragsize = audio_buffer_bytes(pdo, as, 23220);
if (ioctl (fd, SNDCTL_DSP_SAMPLESIZE, &fmt)) {
oss_logerr2 (errno, typ, "Failed to set sample size %d\n", req->fmt);
@@ -310,18 +310,18 @@ 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;
@@ -504,17 +504,16 @@ static int oss_init_out(HWVoiceOut *hw, struct
audsettings *as,
int fd;
AudioFormat effective_fmt;
struct audsettings obt_as;
- OSSConf *conf = drv_opaque;
+ Audiodev *dev = drv_opaque;
+ AudiodevOssOptions *oopts = dev->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;
}
@@ -541,7 +540,7 @@ 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,
@@ -601,7 +600,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings
*as,
}
oss->fd = fd;
- oss->conf = conf;
+ oss->dev = dev;
return 0;
}
@@ -609,16 +608,12 @@ static int oss_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
int trig;
OSSVoiceOut *oss = (OSSVoiceOut *) hw;
+ AudiodevOssPerDirectionOptions *opdo = oss->dev->oss->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->has_try_poll || opdo->try_poll;
ldebug ("enabling voice\n");
if (poll_mode) {
@@ -673,16 +668,14 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings
*as, void *drv_opaque)
int fd;
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;
}
@@ -716,7 +709,7 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings
*as, void *drv_opaque)
}
oss->fd = fd;
- oss->conf = conf;
+ oss->dev = dev;
return 0;
}
@@ -807,16 +800,12 @@ 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->oss->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->has_try_poll || opdo->try_poll;
if (poll_mode) {
oss_poll_in (hw);
@@ -836,81 +825,25 @@ 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_audio_init(Audiodev *dev)
{
- OSSConf *conf = g_malloc(sizeof(OSSConf));
- *conf = glob_conf;
+ AudiodevOssOptions *oopts;
+ assert(dev->kind == AUDIODEV_DRIVER_OSS);
- if (access(conf->devpath_in, R_OK | W_OK) < 0 ||
- access(conf->devpath_out, R_OK | W_OK) < 0) {
+ oopts = dev->oss;
+ if (access(oopts->oss_in->has_dev ? oopts->oss_in->dev : "/dev/dsp",
+ R_OK | W_OK) < 0 ||
+ access(oopts->oss_out->has_dev ? oopts->oss_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 wont 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,
@@ -928,7 +861,6 @@ static struct audio_pcm_ops oss_pcm_ops = {
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,
diff --git a/audio/paaudio.c b/audio/paaudio.c
index cfdbdc6..a9a6a5c 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -1,6 +1,7 @@
/* public domain */
#include "qemu-common.h"
#include "audio.h"
+#include "qapi-visit.h"
#include <pulse/pulseaudio.h>
@@ -9,14 +10,7 @@
#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;
@@ -31,6 +25,7 @@ typedef struct {
void *pcm_buf;
struct audio_pt pt;
paaudio *g;
+ int samples;
} PAVoiceOut;
typedef struct {
@@ -45,6 +40,7 @@ typedef struct {
const void *read_data;
size_t read_index, read_length;
paaudio *g;
+ int samples;
} PAVoiceIn;
static void qpa_audio_fini(void *opaque);
@@ -226,7 +222,7 @@ static void *qpa_thread_out (void *arg)
}
}
- decr = to_mix = audio_MIN (pa->live, pa->g->conf.samples >> 2);
+ decr = to_mix = audio_MIN (pa->live, pa->samples >> 2);
rpos = pa->rpos;
if (audio_pt_unlock (&pa->pt, AUDIO_FUNC)) {
@@ -318,7 +314,7 @@ static void *qpa_thread_in (void *arg)
}
}
- incr = to_grab = audio_MIN (pa->dead, pa->g->conf.samples >> 2);
+ incr = to_grab = audio_MIN (pa->dead, pa->samples >> 2);
wpos = pa->wpos;
if (audio_pt_unlock (&pa->pt, AUDIO_FUNC)) {
@@ -545,6 +541,7 @@ 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->pa;
ss.format = audfmt_to_pa (as->fmt, as->endianness);
ss.channels = as->nchannels;
@@ -565,7 +562,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings
*as,
g,
"qemu",
PA_STREAM_PLAYBACK,
- g->conf.sink,
+ popts->has_sink ? popts->sink : NULL,
&ss,
NULL, /* channel map */
&ba, /* buffering attributes */
@@ -577,7 +574,8 @@ 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(g->dev->out, &obt_as,
+ 46440);
pa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
pa->rpos = hw->rpos;
if (!pa->pcm_buf) {
@@ -611,6 +609,7 @@ 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->pa;
ss.format = audfmt_to_pa (as->fmt, as->endianness);
ss.channels = as->nchannels;
@@ -622,7 +621,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings
*as, void *drv_opaque)
g,
"qemu",
PA_STREAM_RECORD,
- g->conf.source,
+ popts->has_source ? popts->source : NULL,
&ss,
NULL, /* channel map */
NULL, /* buffering attributes */
@@ -634,7 +633,8 @@ 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(g->dev->in, &obt_as,
+ 46440);
pa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
pa->wpos = hw->wpos;
if (!pa->pcm_buf) {
@@ -808,14 +808,19 @@ 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)
{
- paaudio *g = g_malloc(sizeof(paaudio));
- g->conf = glob_conf;
+ paaudio *g;
+ AudiodevPaOptions *popts;
+ const char *server;
+
+ assert(dev->kind == AUDIODEV_DRIVER_PA);
+
+ g = g_malloc(sizeof(paaudio));
+ popts = dev->pa;
+ server = popts->has_server ? popts->server : NULL;
+
+ g->dev = dev;
g->mainloop = NULL;
g->context = NULL;
@@ -825,14 +830,14 @@ 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;
@@ -895,34 +900,6 @@ 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,
@@ -940,7 +917,6 @@ static struct audio_pcm_ops qpa_pcm_ops = {
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,
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index db0f95a..796238a 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -44,18 +44,13 @@ typedef struct SDLVoiceOut {
int decr;
} SDLVoiceOut;
-static struct {
- int nb_samples;
-} conf = {
- .nb_samples = 1024
-};
-
static struct SDLAudioState {
int exit;
SDL_mutex *mutex;
SDL_sem *sem;
int initialized;
bool driver_created;
+ Audiodev *dev;
} glob_sdl;
typedef struct SDLAudioState SDLAudioState;
@@ -347,7 +342,7 @@ static int sdl_init_out(HWVoiceOut *hw, struct audsettings
*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->out, as, 11610);
req.callback = sdl_callback;
req.userdata = sdl;
@@ -391,7 +386,7 @@ 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) {
@@ -420,6 +415,7 @@ static void *sdl_audio_init (void)
}
s->driver_created = true;
+ s->dev = dev;
return s;
}
@@ -431,18 +427,9 @@ static void sdl_audio_fini (void *opaque)
SDL_DestroyMutex (s->mutex);
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,
@@ -454,7 +441,6 @@ static struct audio_pcm_ops sdl_pcm_ops = {
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,
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index f556b3b..441fbcb 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -74,7 +74,7 @@ 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;
@@ -370,10 +370,6 @@ 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,
@@ -391,7 +387,6 @@ static struct audio_pcm_ops audio_callbacks = {
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,
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 81250e6..1af6d23 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -23,6 +23,7 @@
*/
#include "hw/hw.h"
#include "qemu/timer.h"
+#include "qapi-visit.h"
#include "audio.h"
#define AUDIO_CAP "wav"
@@ -36,11 +37,6 @@ 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;
@@ -111,8 +107,10 @@ 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->wav;
+ struct audsettings wav_as = audiodev_to_audsettings(dev->out);
+ const char *wav_path = wopts->has_path ? wopts->path : "qemu.wav";
stereo = wav_as.nchannels == 2;
switch (wav_as.fmt) {
@@ -153,10 +151,10 @@ 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;
@@ -224,54 +222,17 @@ static int wav_ctl_out (HWVoiceOut *hw, int cmd, ...)
return 0;
}
-static WAVConf glob_conf = {
- .settings.freq = 44100,
- .settings.nchannels = 2,
- .settings.fmt = AUDIO_FORMAT_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->kind == 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,
@@ -283,7 +244,6 @@ static struct audio_pcm_ops wav_pcm_ops = {
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,
diff --git a/qemu-options.hx b/qemu-options.hx
index 5438f98..2ce03c4 100644
--- a/qemu-options.hx
+++ b/qemu-options.hx
@@ -185,8 +185,8 @@ Set default value of @var{driver}'s property @var{prop} to
@var{value}, e.g.:
qemu-system-i386 -global ide-drive.physical_block_size=4096 -drive
file=file,if=ide,index=0,media=disk
@end example
-In particular, you can use this to set driver properties for devices which are
-created automatically by the machine model. To create a device which is not
+In particular, you can use this to set driver properties for devices which are
+created automatically by the machine model. To create a device which is not
created automatically and set properties on it, use address@hidden
The two syntaxes are equivalent. The longer one works for drivers whose name
@@ -312,14 +312,234 @@ The default is @code{en-us}.
ETEXI
+HXCOMM Deprecated by -audiodev
DEF("audio-help", 0, QEMU_OPTION_audio_help,
- "-audio-help print list of audio drivers and their options\n",
+ "-audio-help show -audiodev equivalent of the current audio
settings\n",
QEMU_ARCH_ALL)
STEXI
@item -audio-help
@findex -audio-help
-Will show the audio subsystem help: list of drivers, tunable
-parameters.
+Will show the -audiodev equivalent of the currently specified
+(deprecated) environment variables.
+ETEXI
+
+DEF("audiodev", HAS_ARG, QEMU_OPTION_audiodev,
+ "-audiodev [driver=]driver,id=id[,prop[=value][,...]]\n"
+ " specifies the audio backend to use\n"
+ " id= identifier of the backend\n"
+ " timer-period= timer period in microseconds\n"
+ " in|out.fixed-settings= use fixed settings for host
audio\n"
+ " in|out.frequency= frequency to use with fixed settings\n"
+ " in|out.channels= number of channels to use with fixed
settings\n"
+ " in|out.format= sample format to use with fixed settings\n"
+ " valid values: s8, s16, s32, u8, u16, u32\n"
+ " in|out.voices= number of voices to use\n"
+ " in|out.buffer-len= size of buffer in microseconds\n"
+ " in|out.buffer-count= number of buffers\n"
+ "-audiodev none,id=id,[,prop[=value][,...]]\n"
+ " dummy driver that discards all output\n"
+#ifdef CONFIG_ALSA
+ "-audiodev alsa,id=id[,prop[=value][,...]]\n"
+ " alsa-in|alsa-out.dev= name of the audio device to use\n"
+ " alsa-in|alsa-out.try-poll= attempt to use poll mode\n"
+ " threshold= threshold (in microseconds) when playback
starts\n"
+#endif
+#ifdef CONFIG_COREAUDIO
+ "-audiodev coreaudio,id=id[,prop[=value][,...]]\n"
+#endif
+#ifdef CONFIG_DSOUND
+ "-audiodev dsound,id=id[,prop[=value][,...]]\n"
+ " latency= add extra latency to playback in microseconds\n"
+#endif
+#ifdef CONFIG_OSS
+ "-audiodev oss,id=id[,prop[=value][,...]]\n"
+ " oss-in|oss-out.dev= path of the audio device to use\n"
+ " oss-in|oss-out.try-poll= attempt to use poll mode\n"
+ " try-mmap= try using memory mapped access\n"
+ " exclusive= open device in exclusive mode\n"
+ " dsp-policy= set timing policy (0..10), -1 to use fragment
mode\n"
+#endif
+#ifdef CONFIG_PA
+ "-audiodev pa,id=id[,prop[=value][,...]]\n"
+ " server= PulseAudio server address\n"
+ " sink= sink device name\n"
+ " source= source device name\n"
+#endif
+#ifdef CONFIG_SDL
+ "-audiodev sdl,id=id[,prop[=value][,...]]\n"
+#endif
+#ifdef CONFIG_SPICE
+ "-audiodev spice,id=id[,prop[=value][,...]]\n"
+#endif
+ "-audiodev wav,id=id[,prop[=value][,...]]\n"
+ " path= path of wav file to record\n",
+ QEMU_ARCH_ALL)
+STEXI
address@hidden -audiodev address@hidden,address@hidden,@address@hidden,...]]
address@hidden -audiodev
+Adds a new audio backend @var{driver} identified by @var{id}. There are
+global and driver specific properties. Some values can be set
+differently for input and output, they're marked with @code{in|out.}.
+You can set the input's property with @address@hidden and the
+output's property with @address@hidden For example:
address@hidden
+-audiodev alsa,in.frequency=44110,out.frequency=8000
+-audiodev alsa,out.channels=1 # leaves in.channels unspecified
address@hidden example
+
+Valid global options are:
+
address@hidden @option
address@hidden address@hidden
+Identifies the audio backend.
+
address@hidden address@hidden
+Sets the timer @var{period} used by the audio subsystem in microseconds.
+Default is 10000 (10 ms).
+
address@hidden in|out.fixed-settings=on|off
+Use fixed settings for host audio. When off, it will change based on
+how the guest opens the sound card. In this case you must not specify
address@hidden, @var{channels} or @var{format}. Default is on.
+
address@hidden in|address@hidden
+Specify the @var{frequency} to use when using @var{fixed-settings}.
+Default is 44100Hz.
+
address@hidden in|address@hidden
+Specify the number of @var{channels} to use when using
address@hidden Default is 2 (stereo).
+
address@hidden in|address@hidden
+Specify the sample @var{format} to use when using @var{fixed-settings}.
+Valid values are: @code{s8}, @code{s16}, @code{s32}, @code{u8},
address@hidden, @code{u32}. Default is @code{s16}.
+
address@hidden in|address@hidden
+Specify the number of @var{voices} to use. Default is 1.
+
address@hidden in|address@hidden
+Sets the size of the buffer in microseconds.
+
address@hidden in|address@hidden
+Sets the @var{count} of the buffers.
+
address@hidden table
+
address@hidden -audiodev none,address@hidden,@address@hidden,...]]
+Creates a dummy backend that discards all outputs. This backend has no
+backend specific properties.
+
address@hidden -audiodev alsa,address@hidden,@address@hidden,...]]
+Creates backend using the ALSA. This backend is only available on
+Linux.
+
+ALSA specific options are:
+
address@hidden @option
address@hidden alsa-in|address@hidden
+Specify the ALSA @var{device} to use for input and/or output. Default
+is @code{default}.
+
address@hidden alsa-in|alsa-out.try-poll=on|off
+Attempt to use poll mode with the device. Default is on.
+
address@hidden address@hidden
+Threshold (in microseconds) when playback starts. Default is 0.
+
address@hidden table
+
address@hidden -audiodev coreaudio,address@hidden,@address@hidden,...]]
+Creates a backend using Apple's Core Audio. This backend is only
+available on Mac OS and only supports playback. This backend has no
+backend specific properties.
+
address@hidden -audiodev dsound,address@hidden,@address@hidden,...]]
+Creates a backend using Microsoft's DirectSound. This backend is only
+available on Windows and only supports playback.
+
+Backend specific options are:
+
address@hidden @option
+
address@hidden address@hidden
+Add extra @var{usecs} microseconds latency to playback. Default is
+10000 (10 ms).
+
address@hidden table
+
address@hidden -audiodev oss,address@hidden,@address@hidden,...]]
+Creates a backend using OSS. This backend is available on most
+Unix-like systems.
+
+OSS specific options are:
+
address@hidden @option
+
address@hidden oss-in|address@hidden
+Specify the file name of the OSS @var{device} to use. Default is
address@hidden/dev/dsp}.
+
address@hidden oss-in|oss-out.try-poll=on|of
+Attempt to use poll mode with the device. Default is on.
+
address@hidden try-mmap=on|off
+Try using memory mapped device access. Default is off.
+
address@hidden exclusive=on|off
+Open the device in exclusive mode (vmix won't work in this case).
+Default is off.
+
address@hidden address@hidden
+Sets the timing policy (between 0 and 10, where smaller number means
+smaller latency but higher CPU usage). Use -1 to use buffer sizes
+specified by @code{buffer} and @code{buffer-count}. This option is
+ignored if you do not have OSS 4. Default is 5.
+
address@hidden table
+
address@hidden -audiodev pa,address@hidden,@address@hidden,...]]
+Creates a backend using PulseAudio. This backend is available on most
+systems.
+
+PulseAudio specific options are:
+
address@hidden @option
+
address@hidden address@hidden
+Sets the PulseAudio @var{server} to connect to.
+
address@hidden address@hidden
+Use the specified @var{sink} for playback.
+
address@hidden address@hidden
+Use the specified @var{source} for recording.
+
address@hidden table
+
address@hidden -audiodev sdl,address@hidden,@address@hidden,...]]
+Creates a backend using SDL. This backend is available on most systems,
+but you should use your platform's native backend if possible. This
+backend has no backend specific properties.
+
address@hidden -audiodev spice,address@hidden,@address@hidden,...]]
+Creates a backend that sends audio through SPICE. This backend requires
address@hidden and automatically selected in that case, so usually you
+can ignore this option. This backend has no backend specific
+properties.
+
address@hidden -audiodev wav,address@hidden,@address@hidden,...]]
+Creates a backend that writes audio to a WAV file.
+
+Backend specific options are:
+
address@hidden @option
+
address@hidden address@hidden
+Write recorded audio into the specified file. Default is
address@hidden
+
address@hidden table
ETEXI
DEF("soundhw", HAS_ARG, QEMU_OPTION_soundhw,
diff --git a/vl.c b/vl.c
index 2201e27..c3d6ef9 100644
--- a/vl.c
+++ b/vl.c
@@ -2873,6 +2873,7 @@ int main(int argc, char **argv, char **envp)
qemu_add_opts(&qemu_trace_opts);
qemu_add_opts(&qemu_option_rom_opts);
qemu_add_opts(&qemu_machine_opts);
+ qemu_add_opts(&qemu_audiodev_opts);
qemu_add_opts(&qemu_mem_opts);
qemu_add_opts(&qemu_smp_opts);
qemu_add_opts(&qemu_boot_opts);
@@ -3170,9 +3171,14 @@ int main(int argc, char **argv, char **envp)
add_device_config(DEV_BT, optarg);
break;
case QEMU_OPTION_audio_help:
- AUD_help ();
+ audio_legacy_help();
exit (0);
break;
+ case QEMU_OPTION_audiodev:
+ if (!qemu_opts_parse(qemu_find_opts("audiodev"), optarg, 1)) {
+ exit(1);
+ }
+ break;
case QEMU_OPTION_soundhw:
select_soundhw (optarg);
break;
@@ -4339,6 +4345,7 @@ int main(int argc, char **argv, char **envp)
realtime_init();
+ audio_set_options();
audio_init();
cpu_synchronize_all_post_init();
--
2.4.3
- [Qemu-devel] [PATCH v3 4/8] qapi: qapi for audio backends, (continued)
- [Qemu-devel] [PATCH v3 4/8] qapi: qapi for audio backends, Kővágó, Zoltán, 2015/06/18
- [Qemu-devel] [PATCH v3 1/8] qapi: support implicit structs in OptsVisitor, Kővágó, Zoltán, 2015/06/18
- [Qemu-devel] [PATCH v3 6/8] opts: produce valid command line in qemu_opts_print, Kővágó, Zoltán, 2015/06/18
- [Qemu-devel] [PATCH v3 5/8] qapi: support nested structs in OptsVisitor, Kővágó, Zoltán, 2015/06/18
- [Qemu-devel] [PATCH v3 7/8] audio: use qapi AudioFormat instead of audfmt_e, Kővágó, Zoltán, 2015/06/18
- [Qemu-devel] [PATCH v3 3/8] qapi: change Netdev and NetLegacy into a flat union, Kővágó, Zoltán, 2015/06/18
- [Qemu-devel] [PATCH v3 8/8] audio: -audiodev command line option,
Kővágó, Zoltán <=