* [Qemu-devel] [PATCH 08/12] qapi: support nested structs in OptsVisitor
2015-06-12 12:33 [Qemu-devel] [PATCH 00/12] -audiodev option Kővágó, Zoltán
` (6 preceding siblings ...)
2015-06-12 12:33 ` [Qemu-devel] [PATCH 07/12] qapi: qapi for audio backends Kővágó, Zoltán
@ 2015-06-12 12:33 ` Kővágó, Zoltán
2015-06-15 8:39 ` Gerd Hoffmann
2015-06-12 12:33 ` [Qemu-devel] [PATCH 09/12] opts: do not print separator before first item in qemu_opts_print Kővágó, Zoltán
` (4 subsequent siblings)
12 siblings, 1 reply; 17+ messages in thread
From: Kővágó, Zoltán @ 2015-06-12 12:33 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
The current OptsVisitor flattens the whole structure, if there are same named
fields under different paths (like `in' and `out' in `Audiodev'), the current
visitor can't cope with them (for example setting `frequency=44100' will set the
in's frequency to 44100 and leave out's frequency unspecified).
This patch fixes it, by the following changes:
1) Specifying just the field name will apply to all fields that has the
specified name (this means it would set both in's and out's frequency to
44100 in the above example).
2) Optionally user can specify the path in the hierarchy. Names are separated by
a dot (e.g. `in.frequency', `foo.bar.something', etc). The user need not
specify the whole path, only the last few components (i.e. `bar.something' is
equivalent to `foo.bar.something' if only `foo' has a `bar' field). This way
1) is just a special case of this when only the last component is specified.
3) In case of an ambiguity (e.g `frequency=44100,in.frequency=8000') the longest
matching (the most specific) path wins (so in this example, in's frequency
would become 8000, because `in.frequency' is more specific that `frequency',
and out's frequency would become 44100, because only `frequency' matches it).
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
qapi/opts-visitor.c | 144 +++++++++++++++++++++++++-------
tests/qapi-schema/qapi-schema-test.json | 9 +-
tests/test-opts-visitor.c | 34 ++++++++
3 files changed, 157 insertions(+), 30 deletions(-)
diff --git a/qapi/opts-visitor.c b/qapi/opts-visitor.c
index f2ad6d7..409d8b7 100644
--- a/qapi/opts-visitor.c
+++ b/qapi/opts-visitor.c
@@ -64,13 +64,14 @@ struct OptsVisitor
/* Non-null iff depth is positive. Each key is a QemuOpt name. Each value
* is a non-empty GQueue, enumerating all QemuOpt occurrences with that
* name. */
- GHashTable *unprocessed_opts;
+ GHashTable *unprocessed_opts, *opts;
/* The list currently being traversed with opts_start_list() /
* opts_next_list(). The list must have a struct element type in the
* schema, with a single mandatory scalar member. */
ListMode list_mode;
GQueue *repeated_opts;
+ char *repeated_name;
/* When parsing a list of repeating options as integers, values of the form
* "a-b", representing a closed interval, are allowed. Elements in the
@@ -86,6 +87,9 @@ struct OptsVisitor
* not survive or escape the OptsVisitor object.
*/
QemuOpt *fake_id_opt;
+
+ /* List of field names leading to the current structure. */
+ GQueue *nested_names;
};
@@ -97,11 +101,12 @@ destroy_list(gpointer list)
static void
-opts_visitor_insert(GHashTable *unprocessed_opts, const QemuOpt *opt)
+opts_visitor_insert(OptsVisitor *ov, const QemuOpt *opt)
{
GQueue *list;
+ assert(opt);
- list = g_hash_table_lookup(unprocessed_opts, opt->name);
+ list = g_hash_table_lookup(ov->opts, opt->name);
if (list == NULL) {
list = g_queue_new();
@@ -109,7 +114,8 @@ opts_visitor_insert(GHashTable *unprocessed_opts, const QemuOpt *opt)
* "key_destroy_func" in opts_start_struct(). Thus cast away key
* const-ness in order to suppress gcc's warning.
*/
- g_hash_table_insert(unprocessed_opts, (gpointer)opt->name, list);
+ g_hash_table_insert(ov->opts, (gpointer)opt->name, list);
+ g_hash_table_insert(ov->unprocessed_opts, (gpointer)opt->name, list);
}
/* Similarly, destroy_list() doesn't call g_queue_free_full(). */
@@ -127,17 +133,27 @@ opts_start_struct(Visitor *v, void **obj, const char *kind,
if (obj) {
*obj = g_malloc0(size > 0 ? size : 1);
}
+
+ /* assuming name is a statically allocated string (or at least it's lifetime
+ * is longer than the visitor's) */
+ if (!name) {
+ name = "";
+ }
+ g_queue_push_tail(ov->nested_names, (gpointer) name);
+
if (ov->depth++ > 0) {
return;
}
- ov->unprocessed_opts = g_hash_table_new_full(&g_str_hash, &g_str_equal,
- NULL, &destroy_list);
+ ov->opts = g_hash_table_new_full(&g_str_hash, &g_str_equal,
+ NULL, &destroy_list);
+ ov->unprocessed_opts = g_hash_table_new(&g_str_hash, &g_str_equal);
+
QTAILQ_FOREACH(opt, &ov->opts_root->head, next) {
/* ensured by qemu-option.c::opts_do_parse() */
assert(strcmp(opt->name, "id") != 0);
- opts_visitor_insert(ov->unprocessed_opts, opt);
+ opts_visitor_insert(ov, opt);
}
if (ov->opts_root->id != NULL) {
@@ -145,7 +161,7 @@ opts_start_struct(Visitor *v, void **obj, const char *kind,
ov->fake_id_opt->name = g_strdup("id");
ov->fake_id_opt->str = g_strdup(ov->opts_root->id);
- opts_visitor_insert(ov->unprocessed_opts, ov->fake_id_opt);
+ opts_visitor_insert(ov, ov->fake_id_opt);
}
}
@@ -163,6 +179,8 @@ opts_end_struct(Visitor *v, Error **errp)
OptsVisitor *ov = DO_UPCAST(OptsVisitor, visitor, v);
GQueue *any;
+ g_queue_pop_tail(ov->nested_names);
+
if (--ov->depth > 0) {
return;
}
@@ -177,6 +195,8 @@ opts_end_struct(Visitor *v, Error **errp)
}
g_hash_table_destroy(ov->unprocessed_opts);
ov->unprocessed_opts = NULL;
+ g_hash_table_destroy(ov->opts);
+ ov->opts = NULL;
if (ov->fake_id_opt) {
g_free(ov->fake_id_opt->name);
g_free(ov->fake_id_opt->str);
@@ -185,16 +205,56 @@ opts_end_struct(Visitor *v, Error **errp)
ov->fake_id_opt = NULL;
}
+static void
+sum_strlen(gpointer data, gpointer user_data)
+{
+ const char *str = data;
+ size_t *sum_len = user_data;
+ *sum_len += strlen(str) + 1;
+}
+
+static void
+append_str(gpointer data, gpointer user_data)
+{
+ strcat(user_data, data);
+ strcat(user_data, ".");
+}
+
+/* lookup a name, trying from the most qualified version (e.g. foo.bar.asd) to
+ * least qualified ones (i.e. foo.bar.asd overrides bar.asd or asd) */
static GQueue *
-lookup_distinct(const OptsVisitor *ov, const char *name, Error **errp)
+lookup_distinct(const OptsVisitor *ov, const char *name, char **out_key,
+ Error **errp)
{
- GQueue *list;
+ GQueue *list = NULL;
+ char *key, *key2;
+ size_t sum_len = strlen(name);
- list = g_hash_table_lookup(ov->unprocessed_opts, name);
+ g_queue_foreach(ov->nested_names, sum_strlen, &sum_len);
+ key = g_malloc(sum_len+1);
+ key[0] = 0;
+ g_queue_foreach(ov->nested_names, append_str, key);
+ strcat(key, name);
+
+ key2 = key;
+ while (*key2) {
+ list = g_hash_table_lookup(ov->opts, key2);
+ if (list) {
+ if (out_key) {
+ *out_key = g_strdup(key2);
+ }
+ break;
+ }
+
+ while (*key2 && *key2++ != '.') {
+ }
+ }
if (!list) {
error_set(errp, QERR_MISSING_PARAMETER, name);
}
+
+ g_free(key);
return list;
}
@@ -206,7 +266,7 @@ opts_start_list(Visitor *v, const char *name, Error **errp)
/* we can't traverse a list in a list */
assert(ov->list_mode == LM_NONE);
- ov->repeated_opts = lookup_distinct(ov, name, errp);
+ ov->repeated_opts = lookup_distinct(ov, name, &ov->repeated_name, errp);
if (ov->repeated_opts != NULL) {
ov->list_mode = LM_STARTED;
}
@@ -242,11 +302,9 @@ opts_next_list(Visitor *v, GenericList **list, Error **errp)
/* range has been completed, fall through in order to pop option */
case LM_IN_PROGRESS: {
- const QemuOpt *opt;
-
- opt = g_queue_pop_head(ov->repeated_opts);
+ g_queue_pop_head(ov->repeated_opts);
if (g_queue_is_empty(ov->repeated_opts)) {
- g_hash_table_remove(ov->unprocessed_opts, opt->name);
+ g_hash_table_remove(ov->unprocessed_opts, ov->repeated_name);
return NULL;
}
link = &(*list)->next;
@@ -272,22 +330,28 @@ opts_end_list(Visitor *v, Error **errp)
ov->list_mode == LM_SIGNED_INTERVAL ||
ov->list_mode == LM_UNSIGNED_INTERVAL);
ov->repeated_opts = NULL;
+
+ g_free(ov->repeated_name);
+ ov->repeated_name = NULL;
+
ov->list_mode = LM_NONE;
}
static const QemuOpt *
-lookup_scalar(const OptsVisitor *ov, const char *name, Error **errp)
+lookup_scalar(const OptsVisitor *ov, const char *name, char** out_key,
+ Error **errp)
{
if (ov->list_mode == LM_NONE) {
GQueue *list;
/* the last occurrence of any QemuOpt takes effect when queried by name
*/
- list = lookup_distinct(ov, name, errp);
+ list = lookup_distinct(ov, name, out_key, errp);
return list ? g_queue_peek_tail(list) : NULL;
}
assert(ov->list_mode == LM_IN_PROGRESS);
+ assert(out_key == NULL || *out_key == NULL);
return g_queue_peek_head(ov->repeated_opts);
}
@@ -309,13 +373,15 @@ opts_type_str(Visitor *v, char **obj, const char *name, Error **errp)
{
OptsVisitor *ov = DO_UPCAST(OptsVisitor, visitor, v);
const QemuOpt *opt;
+ char *key = NULL;
- opt = lookup_scalar(ov, name, errp);
+ opt = lookup_scalar(ov, name, &key, errp);
if (!opt) {
return;
}
*obj = g_strdup(opt->str ? opt->str : "");
- processed(ov, name);
+ processed(ov, key);
+ g_free(key);
}
@@ -325,8 +391,9 @@ opts_type_bool(Visitor *v, bool *obj, const char *name, Error **errp)
{
OptsVisitor *ov = DO_UPCAST(OptsVisitor, visitor, v);
const QemuOpt *opt;
+ char *key = NULL;
- opt = lookup_scalar(ov, name, errp);
+ opt = lookup_scalar(ov, name, &key, errp);
if (!opt) {
return;
}
@@ -343,13 +410,15 @@ opts_type_bool(Visitor *v, bool *obj, const char *name, Error **errp)
} else {
error_set(errp, QERR_INVALID_PARAMETER_VALUE, opt->name,
"on|yes|y|off|no|n");
+ g_free(key);
return;
}
} else {
*obj = true;
}
- processed(ov, name);
+ processed(ov, key);
+ g_free(key);
}
@@ -361,13 +430,14 @@ opts_type_int(Visitor *v, int64_t *obj, const char *name, Error **errp)
const char *str;
long long val;
char *endptr;
+ char *key = NULL;
if (ov->list_mode == LM_SIGNED_INTERVAL) {
*obj = ov->range_next.s;
return;
}
- opt = lookup_scalar(ov, name, errp);
+ opt = lookup_scalar(ov, name, &key, errp);
if (!opt) {
return;
}
@@ -381,11 +451,13 @@ opts_type_int(Visitor *v, int64_t *obj, const char *name, Error **errp)
if (errno == 0 && endptr > str && INT64_MIN <= val && val <= INT64_MAX) {
if (*endptr == '\0') {
*obj = val;
- processed(ov, name);
+ processed(ov, key);
+ g_free(key);
return;
}
if (*endptr == '-' && ov->list_mode == LM_IN_PROGRESS) {
long long val2;
+ assert(key == NULL);
str = endptr + 1;
val2 = strtoll(str, &endptr, 0);
@@ -406,6 +478,7 @@ opts_type_int(Visitor *v, int64_t *obj, const char *name, Error **errp)
error_set(errp, QERR_INVALID_PARAMETER_VALUE, opt->name,
(ov->list_mode == LM_NONE) ? "an int64 value" :
"an int64 value or range");
+ g_free(key);
}
@@ -417,13 +490,14 @@ opts_type_uint64(Visitor *v, uint64_t *obj, const char *name, Error **errp)
const char *str;
unsigned long long val;
char *endptr;
+ char *key = NULL;
if (ov->list_mode == LM_UNSIGNED_INTERVAL) {
*obj = ov->range_next.u;
return;
}
- opt = lookup_scalar(ov, name, errp);
+ opt = lookup_scalar(ov, name, &key, errp);
if (!opt) {
return;
}
@@ -435,11 +509,13 @@ opts_type_uint64(Visitor *v, uint64_t *obj, const char *name, Error **errp)
if (parse_uint(str, &val, &endptr, 0) == 0 && val <= UINT64_MAX) {
if (*endptr == '\0') {
*obj = val;
- processed(ov, name);
+ processed(ov, key);
+ g_free(key);
return;
}
if (*endptr == '-' && ov->list_mode == LM_IN_PROGRESS) {
unsigned long long val2;
+ assert(key == NULL);
str = endptr + 1;
if (parse_uint_full(str, &val2, 0) == 0 &&
@@ -458,6 +534,7 @@ opts_type_uint64(Visitor *v, uint64_t *obj, const char *name, Error **errp)
error_set(errp, QERR_INVALID_PARAMETER_VALUE, opt->name,
(ov->list_mode == LM_NONE) ? "a uint64 value" :
"a uint64 value or range");
+ g_free(key);
}
@@ -468,8 +545,9 @@ opts_type_size(Visitor *v, uint64_t *obj, const char *name, Error **errp)
const QemuOpt *opt;
int64_t val;
char *endptr;
+ char *key = NULL;
- opt = lookup_scalar(ov, name, errp);
+ opt = lookup_scalar(ov, name, &key, errp);
if (!opt) {
return;
}
@@ -479,11 +557,13 @@ opts_type_size(Visitor *v, uint64_t *obj, const char *name, Error **errp)
if (val < 0 || *endptr) {
error_set(errp, QERR_INVALID_PARAMETER_VALUE, opt->name,
"a size value representible as a non-negative int64");
+ g_free(key);
return;
}
*obj = val;
- processed(ov, name);
+ processed(ov, key);
+ g_free(key);
}
@@ -494,7 +574,7 @@ opts_optional(Visitor *v, bool *present, const char *name, Error **errp)
/* we only support a single mandatory scalar field in a list node */
assert(ov->list_mode == LM_NONE);
- *present = (lookup_distinct(ov, name, NULL) != NULL);
+ *present = (lookup_distinct(ov, name, NULL, NULL) != NULL);
}
@@ -505,6 +585,8 @@ opts_visitor_new(const QemuOpts *opts)
ov = g_malloc0(sizeof *ov);
+ ov->nested_names = g_queue_new();
+
ov->visitor.start_struct = &opts_start_struct;
ov->visitor.end_struct = &opts_end_struct;
@@ -545,6 +627,10 @@ opts_visitor_cleanup(OptsVisitor *ov)
if (ov->unprocessed_opts != NULL) {
g_hash_table_destroy(ov->unprocessed_opts);
}
+ if (ov->opts != NULL) {
+ g_hash_table_destroy(ov->opts);
+ }
+ g_queue_free(ov->nested_names);
g_free(ov->fake_id_opt);
g_free(ov);
}
diff --git a/tests/qapi-schema/qapi-schema-test.json b/tests/qapi-schema/qapi-schema-test.json
index c7eaa86..a818eff 100644
--- a/tests/qapi-schema/qapi-schema-test.json
+++ b/tests/qapi-schema/qapi-schema-test.json
@@ -81,6 +81,11 @@
{ 'command': 'user_def_cmd3', 'data': {'a': 'int', '*b': 'int' },
'returns': 'int' }
+# For testing hierarchy support in opts-visitor
+{ 'struct': 'UserDefOptionsSub',
+ 'data': {
+ '*nint': 'int' } }
+
# For testing integer range flattening in opts-visitor. The following schema
# corresponds to the option format:
#
@@ -94,7 +99,9 @@
'*u64' : [ 'uint64' ],
'*u16' : [ 'uint16' ],
'*i64x': 'int' ,
- '*u64x': 'uint64' } }
+ '*u64x': 'uint64' ,
+ 'sub0': 'UserDefOptionsSub',
+ 'sub1': 'UserDefOptionsSub' } }
# testing event
{ 'struct': 'EventStructOne',
diff --git a/tests/test-opts-visitor.c b/tests/test-opts-visitor.c
index ebeee5d..5862c7c 100644
--- a/tests/test-opts-visitor.c
+++ b/tests/test-opts-visitor.c
@@ -177,6 +177,34 @@ expect_u64_max(OptsVisitorFixture *f, gconstpointer test_data)
g_assert(f->userdef->u64->value == UINT64_MAX);
}
+static void
+expect_both(OptsVisitorFixture *f, gconstpointer test_data)
+{
+ expect_ok(f, test_data);
+ g_assert(f->userdef->sub0->has_nint);
+ g_assert(f->userdef->sub0->nint == 13);
+ g_assert(f->userdef->sub1->has_nint);
+ g_assert(f->userdef->sub1->nint == 13);
+}
+
+static void
+expect_sub0(OptsVisitorFixture *f, gconstpointer test_data)
+{
+ expect_ok(f, test_data);
+ g_assert(f->userdef->sub0->has_nint);
+ g_assert(f->userdef->sub0->nint == 13);
+ g_assert(!f->userdef->sub1->has_nint);
+}
+
+static void
+expect_sub1(OptsVisitorFixture *f, gconstpointer test_data)
+{
+ expect_ok(f, test_data);
+ g_assert(!f->userdef->sub0->has_nint);
+ g_assert(f->userdef->sub1->has_nint);
+ g_assert(f->userdef->sub1->nint == 13);
+}
+
/* test cases */
int
@@ -270,6 +298,12 @@ main(int argc, char **argv)
add_test("/visitor/opts/i64/range/2big/full", &expect_fail,
"i64=-0x8000000000000000-0x7fffffffffffffff");
+ /* Test nested structs support */
+ add_test("/visitor/opts/nested/unqualified", &expect_both, "nint=13");
+ add_test("/visitor/opts/nested/both", &expect_both,
+ "sub0.nint=13,sub1.nint=13");
+ add_test("/visitor/opts/nested/sub0", &expect_sub0, "sub0.nint=13");
+ add_test("/visitor/opts/nested/sub1", &expect_sub1, "sub1.nint=13");
g_test_run();
return 0;
}
--
2.4.2
^ permalink raw reply related [flat|nested] 17+ messages in thread* [Qemu-devel] [PATCH 11/12] audio: use qapi AudioFormat instead of audfmt_e
2015-06-12 12:33 [Qemu-devel] [PATCH 00/12] -audiodev option Kővágó, Zoltán
` (9 preceding siblings ...)
2015-06-12 12:33 ` [Qemu-devel] [PATCH 10/12] qapi: AllocVisitor Kővágó, Zoltán
@ 2015-06-12 12:33 ` Kővágó, Zoltán
2015-06-12 12:33 ` [Qemu-devel] [PATCH 12/12] audio: -audiodev command line option Kővágó, Zoltán
2015-06-15 9:01 ` [Qemu-devel] [PATCH 00/12] -audiodev option Gerd Hoffmann
12 siblings, 0 replies; 17+ messages in thread
From: Kővágó, Zoltán @ 2015-06-12 12:33 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
I had to include an enum for audio sampling formats into qapi, but that meant
duplicating the audfmt_e enum. This patch replaces audfmt_e and associated
values with the qapi generated AudioFormat enum.
This patch is mostly a search-and-replace, except for switches where the qapi
generated AUDIO_FORMAT_MAX caused problems.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/alsaaudio.c | 53 ++++++++++++++------------
audio/audio.c | 97 ++++++++++++++++++++++++++---------------------
audio/audio.h | 11 +-----
audio/audio_win_int.c | 18 ++++-----
audio/ossaudio.c | 30 +++++++--------
audio/paaudio.c | 28 +++++++-------
audio/sdlaudio.c | 26 ++++++-------
audio/spiceaudio.c | 4 +-
audio/wavaudio.c | 17 +++++----
audio/wavcapture.c | 2 +-
hw/arm/omap2.c | 2 +-
hw/audio/ac97.c | 2 +-
hw/audio/adlib.c | 2 +-
hw/audio/cs4231a.c | 6 +--
hw/audio/es1370.c | 4 +-
hw/audio/gus.c | 2 +-
hw/audio/hda-codec.c | 18 ++++-----
hw/audio/lm4549.c | 6 +--
hw/audio/milkymist-ac97.c | 2 +-
hw/audio/pcspk.c | 2 +-
hw/audio/sb16.c | 14 +++----
hw/audio/wm8750.c | 4 +-
hw/input/tsc210x.c | 2 +-
hw/usb/dev-audio.c | 2 +-
ui/vnc.c | 14 +++----
25 files changed, 187 insertions(+), 181 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index b0a451a..6882638 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -88,7 +88,7 @@ struct alsa_params_req {
struct alsa_params_obt {
int freq;
- audfmt_e fmt;
+ AudioFormat fmt;
int endianness;
int nchannels;
snd_pcm_uframes_t samples;
@@ -307,16 +307,16 @@ static int alsa_write (SWVoiceOut *sw, void *buf, int len)
return audio_pcm_sw_write (sw, buf, len);
}
-static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
+static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness)
{
switch (fmt) {
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
return SND_PCM_FORMAT_S8;
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
return SND_PCM_FORMAT_U8;
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
if (endianness) {
return SND_PCM_FORMAT_S16_BE;
}
@@ -324,7 +324,7 @@ static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
return SND_PCM_FORMAT_S16_LE;
}
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
if (endianness) {
return SND_PCM_FORMAT_U16_BE;
}
@@ -332,7 +332,7 @@ static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
return SND_PCM_FORMAT_U16_LE;
}
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_S32:
if (endianness) {
return SND_PCM_FORMAT_S32_BE;
}
@@ -340,7 +340,7 @@ static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
return SND_PCM_FORMAT_S32_LE;
}
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_U32:
if (endianness) {
return SND_PCM_FORMAT_U32_BE;
}
@@ -357,58 +357,58 @@ static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
}
}
-static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt,
+static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt,
int *endianness)
{
switch (alsafmt) {
case SND_PCM_FORMAT_S8:
*endianness = 0;
- *fmt = AUD_FMT_S8;
+ *fmt = AUDIO_FORMAT_S8;
break;
case SND_PCM_FORMAT_U8:
*endianness = 0;
- *fmt = AUD_FMT_U8;
+ *fmt = AUDIO_FORMAT_U8;
break;
case SND_PCM_FORMAT_S16_LE:
*endianness = 0;
- *fmt = AUD_FMT_S16;
+ *fmt = AUDIO_FORMAT_S16;
break;
case SND_PCM_FORMAT_U16_LE:
*endianness = 0;
- *fmt = AUD_FMT_U16;
+ *fmt = AUDIO_FORMAT_U16;
break;
case SND_PCM_FORMAT_S16_BE:
*endianness = 1;
- *fmt = AUD_FMT_S16;
+ *fmt = AUDIO_FORMAT_S16;
break;
case SND_PCM_FORMAT_U16_BE:
*endianness = 1;
- *fmt = AUD_FMT_U16;
+ *fmt = AUDIO_FORMAT_U16;
break;
case SND_PCM_FORMAT_S32_LE:
*endianness = 0;
- *fmt = AUD_FMT_S32;
+ *fmt = AUDIO_FORMAT_S32;
break;
case SND_PCM_FORMAT_U32_LE:
*endianness = 0;
- *fmt = AUD_FMT_U32;
+ *fmt = AUDIO_FORMAT_U32;
break;
case SND_PCM_FORMAT_S32_BE:
*endianness = 1;
- *fmt = AUD_FMT_S32;
+ *fmt = AUDIO_FORMAT_S32;
break;
case SND_PCM_FORMAT_U32_BE:
*endianness = 1;
- *fmt = AUD_FMT_U32;
+ *fmt = AUDIO_FORMAT_U32;
break;
default:
@@ -651,19 +651,22 @@ static int alsa_open (int in, struct alsa_params_req *req,
bytes_per_sec = freq << (nchannels == 2);
switch (obt->fmt) {
- case AUD_FMT_S8:
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_S8:
+ case AUDIO_FORMAT_U8:
break;
- case AUD_FMT_S16:
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_S16:
+ case AUDIO_FORMAT_U16:
bytes_per_sec <<= 1;
break;
- case AUD_FMT_S32:
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_S32:
+ case AUDIO_FORMAT_U32:
bytes_per_sec <<= 2;
break;
+
+ case AUDIO_FORMAT_MAX:
+ break;
}
threshold = (conf->threshold * bytes_per_sec) / 1000;
diff --git a/audio/audio.c b/audio/audio.c
index 5be4b15..112b57b 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -75,7 +75,7 @@ static struct {
.settings = {
.freq = 44100,
.nchannels = 2,
- .fmt = AUD_FMT_S16,
+ .fmt = AUDIO_FORMAT_S16,
.endianness = AUDIO_HOST_ENDIANNESS,
}
},
@@ -87,7 +87,7 @@ static struct {
.settings = {
.freq = 44100,
.nchannels = 2,
- .fmt = AUD_FMT_S16,
+ .fmt = AUDIO_FORMAT_S16,
.endianness = AUDIO_HOST_ENDIANNESS,
}
},
@@ -219,58 +219,61 @@ static char *audio_alloc_prefix (const char *s)
return r;
}
-static const char *audio_audfmt_to_string (audfmt_e fmt)
+static const char *audio_audfmt_to_string (AudioFormat fmt)
{
switch (fmt) {
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
return "U8";
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
return "U16";
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
return "S8";
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
return "S16";
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_U32:
return "U32";
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_S32:
return "S32";
+
+ case AUDIO_FORMAT_MAX:
+ abort();
}
dolog ("Bogus audfmt %d returning S16\n", fmt);
return "S16";
}
-static audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval,
+static AudioFormat audio_string_to_audfmt (const char *s, AudioFormat defval,
int *defaultp)
{
if (!strcasecmp (s, "u8")) {
*defaultp = 0;
- return AUD_FMT_U8;
+ return AUDIO_FORMAT_U8;
}
else if (!strcasecmp (s, "u16")) {
*defaultp = 0;
- return AUD_FMT_U16;
+ return AUDIO_FORMAT_U16;
}
else if (!strcasecmp (s, "u32")) {
*defaultp = 0;
- return AUD_FMT_U32;
+ return AUDIO_FORMAT_U32;
}
else if (!strcasecmp (s, "s8")) {
*defaultp = 0;
- return AUD_FMT_S8;
+ return AUDIO_FORMAT_S8;
}
else if (!strcasecmp (s, "s16")) {
*defaultp = 0;
- return AUD_FMT_S16;
+ return AUDIO_FORMAT_S16;
}
else if (!strcasecmp (s, "s32")) {
*defaultp = 0;
- return AUD_FMT_S32;
+ return AUDIO_FORMAT_S32;
}
else {
dolog ("Bogus audio format `%s' using %s\n",
@@ -280,8 +283,8 @@ static audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval,
}
}
-static audfmt_e audio_get_conf_fmt (const char *envname,
- audfmt_e defval,
+static AudioFormat audio_get_conf_fmt (const char *envname,
+ AudioFormat defval,
int *defaultp)
{
const char *var = getenv (envname);
@@ -384,7 +387,7 @@ static void audio_print_options (const char *prefix,
case AUD_OPT_FMT:
{
- audfmt_e *fmtp = opt->valp;
+ AudioFormat *fmtp = opt->valp;
printf (
"format, %s = %s, (one of: U8 S8 U16 S16 U32 S32)\n",
state,
@@ -471,7 +474,7 @@ static void audio_process_options (const char *prefix,
case AUD_OPT_FMT:
{
- audfmt_e *fmtp = opt->valp;
+ AudioFormat *fmtp = opt->valp;
*fmtp = audio_get_conf_fmt (optname, *fmtp, &def);
}
break;
@@ -502,22 +505,22 @@ static void audio_print_settings (struct audsettings *as)
dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels);
switch (as->fmt) {
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
AUD_log (NULL, "S8");
break;
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
AUD_log (NULL, "U8");
break;
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
AUD_log (NULL, "S16");
break;
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
AUD_log (NULL, "U16");
break;
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_S32:
AUD_log (NULL, "S32");
break;
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_U32:
AUD_log (NULL, "U32");
break;
default:
@@ -548,12 +551,12 @@ static int audio_validate_settings (struct audsettings *as)
invalid |= as->endianness != 0 && as->endianness != 1;
switch (as->fmt) {
- case AUD_FMT_S8:
- case AUD_FMT_U8:
- case AUD_FMT_S16:
- case AUD_FMT_U16:
- case AUD_FMT_S32:
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_S8:
+ case AUDIO_FORMAT_U8:
+ case AUDIO_FORMAT_S16:
+ case AUDIO_FORMAT_U16:
+ case AUDIO_FORMAT_S32:
+ case AUDIO_FORMAT_U32:
break;
default:
invalid = 1;
@@ -569,25 +572,28 @@ static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *a
int bits = 8, sign = 0;
switch (as->fmt) {
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
sign = 1;
/* fall through */
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
break;
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
sign = 1;
/* fall through */
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
bits = 16;
break;
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_S32:
sign = 1;
/* fall through */
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_U32:
bits = 32;
break;
+
+ case AUDIO_FORMAT_MAX:
+ abort();
}
return info->freq == as->freq
&& info->nchannels == as->nchannels
@@ -601,24 +607,27 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
int bits = 8, sign = 0, shift = 0;
switch (as->fmt) {
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
sign = 1;
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
break;
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
sign = 1;
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
bits = 16;
shift = 1;
break;
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_S32:
sign = 1;
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_U32:
bits = 32;
shift = 2;
break;
+
+ case AUDIO_FORMAT_MAX:
+ abort();
}
info->freq = as->freq;
diff --git a/audio/audio.h b/audio/audio.h
index e7ea397..e300511 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -29,15 +29,6 @@
typedef void (*audio_callback_fn) (void *opaque, int avail);
-typedef enum {
- AUD_FMT_U8,
- AUD_FMT_S8,
- AUD_FMT_U16,
- AUD_FMT_S16,
- AUD_FMT_U32,
- AUD_FMT_S32
-} audfmt_e;
-
#ifdef HOST_WORDS_BIGENDIAN
#define AUDIO_HOST_ENDIANNESS 1
#else
@@ -47,7 +38,7 @@ typedef enum {
struct audsettings {
int freq;
int nchannels;
- audfmt_e fmt;
+ AudioFormat fmt;
int endianness;
};
diff --git a/audio/audio_win_int.c b/audio/audio_win_int.c
index e132405..a8cfa77 100644
--- a/audio/audio_win_int.c
+++ b/audio/audio_win_int.c
@@ -23,20 +23,20 @@ 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;
@@ -84,15 +84,15 @@ int waveformat_to_audio_settings (WAVEFORMATEX *wfx,
switch (wfx->wBitsPerSample) {
case 8:
- as->fmt = AUD_FMT_U8;
+ as->fmt = AUDIO_FORMAT_U8;
break;
case 16:
- as->fmt = AUD_FMT_S16;
+ as->fmt = AUDIO_FORMAT_S16;
break;
case 32:
- as->fmt = AUD_FMT_S32;
+ as->fmt = AUDIO_FORMAT_S32;
break;
default:
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index d5362ab..4f5bef6 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -72,7 +72,7 @@ typedef struct OSSVoiceIn {
struct oss_params {
int freq;
- audfmt_e fmt;
+ AudioFormat fmt;
int nchannels;
int nfrags;
int fragsize;
@@ -150,16 +150,16 @@ 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;
}
@@ -167,7 +167,7 @@ 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;
}
@@ -184,37 +184,37 @@ 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:
@@ -502,7 +502,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
int endianness;
int err;
int fd;
- audfmt_e effective_fmt;
+ AudioFormat effective_fmt;
struct audsettings obt_as;
OSSConf *conf = drv_opaque;
@@ -670,7 +670,7 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
int endianness;
int err;
int fd;
- audfmt_e effective_fmt;
+ AudioFormat effective_fmt;
struct audsettings obt_as;
OSSConf *conf = drv_opaque;
diff --git a/audio/paaudio.c b/audio/paaudio.c
index fea6071..cfdbdc6 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -384,21 +384,21 @@ 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:
@@ -409,26 +409,26 @@ static pa_sample_format_t audfmt_to_pa (audfmt_e afmt, int endianness)
return format;
}
-static audfmt_e pa_to_audfmt (pa_sample_format_t fmt, int *endianness)
+static AudioFormat pa_to_audfmt (pa_sample_format_t fmt, int *endianness)
{
switch (fmt) {
case PA_SAMPLE_U8:
- return AUD_FMT_U8;
+ return AUDIO_FORMAT_U8;
case PA_SAMPLE_S16BE:
*endianness = 1;
- return AUD_FMT_S16;
+ return AUDIO_FORMAT_S16;
case PA_SAMPLE_S16LE:
*endianness = 0;
- return AUD_FMT_S16;
+ return AUDIO_FORMAT_S16;
case PA_SAMPLE_S32BE:
*endianness = 1;
- return AUD_FMT_S32;
+ return AUDIO_FORMAT_S32;
case PA_SAMPLE_S32LE:
*endianness = 0;
- return AUD_FMT_S32;
+ return AUDIO_FORMAT_S32;
default:
dolog ("Internal logic error: Bad pa_sample_format %d\n", fmt);
- return AUD_FMT_U8;
+ return AUDIO_FORMAT_U8;
}
}
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index 1140f2e..db0f95a 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -115,19 +115,19 @@ static int sdl_unlock_and_post (SDLAudioState *s, const char *forfn)
return sdl_post (s, forfn);
}
-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:
@@ -139,37 +139,37 @@ 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:
@@ -341,7 +341,7 @@ static int sdl_init_out(HWVoiceOut *hw, struct audsettings *as,
SDL_AudioSpec req, obt;
int endianness;
int err;
- audfmt_e effective_fmt;
+ AudioFormat effective_fmt;
struct audsettings obt_as;
req.freq = as->freq;
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index 5c6f726..f556b3b 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -127,7 +127,7 @@ 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);
@@ -255,7 +255,7 @@ static int line_in_init(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
settings.freq = SPICE_INTERFACE_RECORD_FREQ;
#endif
settings.nchannels = SPICE_INTERFACE_RECORD_CHAN;
- settings.fmt = AUD_FMT_S16;
+ settings.fmt = AUDIO_FORMAT_S16;
settings.endianness = AUDIO_HOST_ENDIANNESS;
audio_pcm_init_info (&hw->info, &settings);
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index c586020..62017de 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -116,20 +116,23 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
stereo = wav_as.nchannels == 2;
switch (wav_as.fmt) {
- case AUD_FMT_S8:
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_S8:
+ case AUDIO_FORMAT_U8:
bits16 = 0;
break;
- case AUD_FMT_S16:
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_S16:
+ case AUDIO_FORMAT_U16:
bits16 = 1;
break;
- case AUD_FMT_S32:
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_S32:
+ case AUDIO_FORMAT_U32:
dolog ("WAVE files can not handle 32bit formats\n");
return -1;
+
+ case AUDIO_FORMAT_MAX:
+ abort();
}
hdr[34] = bits16 ? 0x10 : 0x08;
@@ -224,7 +227,7 @@ static int wav_ctl_out (HWVoiceOut *hw, int cmd, ...)
static WAVConf glob_conf = {
.settings.freq = 44100,
.settings.nchannels = 2,
- .settings.fmt = AUD_FMT_S16,
+ .settings.fmt = AUDIO_FORMAT_S16,
.wav_path = "qemu.wav"
};
diff --git a/audio/wavcapture.c b/audio/wavcapture.c
index 6f6d792..b03c244 100644
--- a/audio/wavcapture.c
+++ b/audio/wavcapture.c
@@ -135,7 +135,7 @@ int wav_start_capture (CaptureState *s, const char *path, int freq,
as.freq = freq;
as.nchannels = 1 << stereo;
- as.fmt = bits16 ? AUD_FMT_S16 : AUD_FMT_U8;
+ as.fmt = bits16 ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_U8;
as.endianness = 0;
ops.notify = wav_notify;
diff --git a/hw/arm/omap2.c b/hw/arm/omap2.c
index e39b317..3b14a5d 100644
--- a/hw/arm/omap2.c
+++ b/hw/arm/omap2.c
@@ -269,7 +269,7 @@ static void omap_eac_format_update(struct omap_eac_s *s)
* does I2S specify it? */
/* All register writes are 16 bits so we we store 16-bit samples
* in the buffers regardless of AGCFR[B8_16] value. */
- fmt.fmt = AUD_FMT_U16;
+ fmt.fmt = AUDIO_FORMAT_U16;
s->codec.in_voice = AUD_open_in(&s->codec.card, s->codec.in_voice,
"eac.codec.in", s, omap_eac_in_cb, &fmt);
diff --git a/hw/audio/ac97.c b/hw/audio/ac97.c
index b173835..fa75f33 100644
--- a/hw/audio/ac97.c
+++ b/hw/audio/ac97.c
@@ -360,7 +360,7 @@ static void open_voice (AC97LinkState *s, int index, int freq)
as.freq = freq;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = 0;
if (freq > 0) {
diff --git a/hw/audio/adlib.c b/hw/audio/adlib.c
index 656eb37..f8f0f55 100644
--- a/hw/audio/adlib.c
+++ b/hw/audio/adlib.c
@@ -323,7 +323,7 @@ static void adlib_realizefn (DeviceState *dev, Error **errp)
as.freq = s->freq;
as.nchannels = SHIFT;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = AUDIO_HOST_ENDIANNESS;
AUD_register_card ("adlib", &s->card);
diff --git a/hw/audio/cs4231a.c b/hw/audio/cs4231a.c
index f96f561..626a173 100644
--- a/hw/audio/cs4231a.c
+++ b/hw/audio/cs4231a.c
@@ -284,7 +284,7 @@ static void cs_reset_voices (CSState *s, uint32_t val)
switch ((val >> 5) & ((s->dregs[MODE_And_ID] & MODE2) ? 7 : 3)) {
case 0:
- as.fmt = AUD_FMT_U8;
+ as.fmt = AUDIO_FORMAT_U8;
s->shift = as.nchannels == 2;
break;
@@ -294,7 +294,7 @@ static void cs_reset_voices (CSState *s, uint32_t val)
case 3:
s->tab = ALawDecompressTable;
x_law:
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = AUDIO_HOST_ENDIANNESS;
s->shift = as.nchannels == 2;
break;
@@ -302,7 +302,7 @@ static void cs_reset_voices (CSState *s, uint32_t val)
case 6:
as.endianness = 1;
case 2:
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
s->shift = as.nchannels;
break;
diff --git a/hw/audio/es1370.c b/hw/audio/es1370.c
index 8e7bcf5..f6e74cb 100644
--- a/hw/audio/es1370.c
+++ b/hw/audio/es1370.c
@@ -414,14 +414,14 @@ static void es1370_update_voices (ES1370State *s, uint32_t ctl, uint32_t sctl)
i,
new_freq,
1 << (new_fmt & 1),
- (new_fmt & 2) ? AUD_FMT_S16 : AUD_FMT_U8,
+ (new_fmt & 2) ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_U8,
d->shift);
if (new_freq) {
struct audsettings as;
as.freq = new_freq;
as.nchannels = 1 << (new_fmt & 1);
- as.fmt = (new_fmt & 2) ? AUD_FMT_S16 : AUD_FMT_U8;
+ as.fmt = (new_fmt & 2) ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_U8;
as.endianness = 0;
if (i == ADC_CHANNEL) {
diff --git a/hw/audio/gus.c b/hw/audio/gus.c
index 86223a9..6107824 100644
--- a/hw/audio/gus.c
+++ b/hw/audio/gus.c
@@ -242,7 +242,7 @@ static void gus_realizefn (DeviceState *dev, Error **errp)
as.freq = s->freq;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = GUS_ENDIANNESS;
s->voice = AUD_open_out (
diff --git a/hw/audio/hda-codec.c b/hw/audio/hda-codec.c
index 3c03ff5..8693b7a 100644
--- a/hw/audio/hda-codec.c
+++ b/hw/audio/hda-codec.c
@@ -97,9 +97,9 @@ static void hda_codec_parse_fmt(uint32_t format, struct audsettings *as)
}
switch (format & AC_FMT_BITS_MASK) {
- case AC_FMT_BITS_8: as->fmt = AUD_FMT_S8; break;
- case AC_FMT_BITS_16: as->fmt = AUD_FMT_S16; break;
- case AC_FMT_BITS_32: as->fmt = AUD_FMT_S32; break;
+ case AC_FMT_BITS_8: as->fmt = AUDIO_FORMAT_S8; break;
+ case AC_FMT_BITS_16: as->fmt = AUDIO_FORMAT_S16; break;
+ case AC_FMT_BITS_32: as->fmt = AUDIO_FORMAT_S32; break;
}
as->nchannels = ((format & AC_FMT_CHAN_MASK) >> AC_FMT_CHAN_SHIFT) + 1;
@@ -128,12 +128,12 @@ static void hda_codec_parse_fmt(uint32_t format, struct audsettings *as)
/* -------------------------------------------------------------------------- */
static const char *fmt2name[] = {
- [ AUD_FMT_U8 ] = "PCM-U8",
- [ AUD_FMT_S8 ] = "PCM-S8",
- [ AUD_FMT_U16 ] = "PCM-U16",
- [ AUD_FMT_S16 ] = "PCM-S16",
- [ AUD_FMT_U32 ] = "PCM-U32",
- [ AUD_FMT_S32 ] = "PCM-S32",
+ [ AUDIO_FORMAT_U8 ] = "PCM-U8",
+ [ AUDIO_FORMAT_S8 ] = "PCM-S8",
+ [ AUDIO_FORMAT_U16 ] = "PCM-U16",
+ [ AUDIO_FORMAT_S16 ] = "PCM-S16",
+ [ AUDIO_FORMAT_U32 ] = "PCM-U32",
+ [ AUDIO_FORMAT_S32 ] = "PCM-S32",
};
typedef struct HDAAudioState HDAAudioState;
diff --git a/hw/audio/lm4549.c b/hw/audio/lm4549.c
index 380ef60..9d4f4b5 100644
--- a/hw/audio/lm4549.c
+++ b/hw/audio/lm4549.c
@@ -185,7 +185,7 @@ void lm4549_write(lm4549_state *s,
struct audsettings as;
as.freq = value;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = 0;
s->voice = AUD_open_out(
@@ -255,7 +255,7 @@ static int lm4549_post_load(void *opaque, int version_id)
struct audsettings as;
as.freq = freq;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = 0;
s->voice = AUD_open_out(
@@ -292,7 +292,7 @@ void lm4549_init(lm4549_state *s, lm4549_callback data_req_cb, void* opaque)
/* Open a default voice */
as.freq = 48000;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = 0;
s->voice = AUD_open_out(
diff --git a/hw/audio/milkymist-ac97.c b/hw/audio/milkymist-ac97.c
index 28f55e8..15169e2 100644
--- a/hw/audio/milkymist-ac97.c
+++ b/hw/audio/milkymist-ac97.c
@@ -297,7 +297,7 @@ static int milkymist_ac97_init(SysBusDevice *dev)
as.freq = 48000;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = 1;
s->voice_in = AUD_open_in(&s->card, s->voice_in,
diff --git a/hw/audio/pcspk.c b/hw/audio/pcspk.c
index 5266fb5..302debf 100644
--- a/hw/audio/pcspk.c
+++ b/hw/audio/pcspk.c
@@ -112,7 +112,7 @@ static void pcspk_callback(void *opaque, int free)
static int pcspk_audio_init(ISABus *bus)
{
PCSpkState *s = pcspk_state;
- struct audsettings as = {PCSPK_SAMPLE_RATE, 1, AUD_FMT_U8, 0};
+ struct audsettings as = {PCSPK_SAMPLE_RATE, 1, AUDIO_FORMAT_U8, 0};
AUD_register_card(s_spk, &s->card);
diff --git a/hw/audio/sb16.c b/hw/audio/sb16.c
index b052de5..a159dcc 100644
--- a/hw/audio/sb16.c
+++ b/hw/audio/sb16.c
@@ -66,7 +66,7 @@ typedef struct SB16State {
int fmt_stereo;
int fmt_signed;
int fmt_bits;
- audfmt_e fmt;
+ AudioFormat fmt;
int dma_auto;
int block_size;
int fifo;
@@ -221,7 +221,7 @@ static void continue_dma8 (SB16State *s)
static void dma_cmd8 (SB16State *s, int mask, int dma_len)
{
- s->fmt = AUD_FMT_U8;
+ s->fmt = AUDIO_FORMAT_U8;
s->use_hdma = 0;
s->fmt_bits = 8;
s->fmt_signed = 0;
@@ -316,18 +316,18 @@ static void dma_cmd (SB16State *s, uint8_t cmd, uint8_t d0, int dma_len)
if (16 == s->fmt_bits) {
if (s->fmt_signed) {
- s->fmt = AUD_FMT_S16;
+ s->fmt = AUDIO_FORMAT_S16;
}
else {
- s->fmt = AUD_FMT_U16;
+ s->fmt = AUDIO_FORMAT_U16;
}
}
else {
if (s->fmt_signed) {
- s->fmt = AUD_FMT_S8;
+ s->fmt = AUDIO_FORMAT_S8;
}
else {
- s->fmt = AUD_FMT_U8;
+ s->fmt = AUDIO_FORMAT_U8;
}
}
@@ -839,7 +839,7 @@ static void legacy_reset (SB16State *s)
as.freq = s->freq;
as.nchannels = 1;
- as.fmt = AUD_FMT_U8;
+ as.fmt = AUDIO_FORMAT_U8;
as.endianness = 0;
s->voice = AUD_open_out (
diff --git a/hw/audio/wm8750.c b/hw/audio/wm8750.c
index b50b331..4c4333c 100644
--- a/hw/audio/wm8750.c
+++ b/hw/audio/wm8750.c
@@ -201,7 +201,7 @@ static void wm8750_set_format(WM8750State *s)
in_fmt.endianness = 0;
in_fmt.nchannels = 2;
in_fmt.freq = s->adc_hz;
- in_fmt.fmt = AUD_FMT_S16;
+ in_fmt.fmt = AUDIO_FORMAT_S16;
s->adc_voice[0] = AUD_open_in(&s->card, s->adc_voice[0],
CODEC ".input1", s, wm8750_audio_in_cb, &in_fmt);
@@ -214,7 +214,7 @@ static void wm8750_set_format(WM8750State *s)
out_fmt.endianness = 0;
out_fmt.nchannels = 2;
out_fmt.freq = s->dac_hz;
- out_fmt.fmt = AUD_FMT_S16;
+ out_fmt.fmt = AUDIO_FORMAT_S16;
s->dac_voice[0] = AUD_open_out(&s->card, s->dac_voice[0],
CODEC ".speaker", s, wm8750_audio_out_cb, &out_fmt);
diff --git a/hw/input/tsc210x.c b/hw/input/tsc210x.c
index fae3385..3cf938b 100644
--- a/hw/input/tsc210x.c
+++ b/hw/input/tsc210x.c
@@ -315,7 +315,7 @@ static void tsc2102_audio_output_update(TSC210xState *s)
fmt.endianness = 0;
fmt.nchannels = 2;
fmt.freq = s->codec.tx_rate;
- fmt.fmt = AUD_FMT_S16;
+ fmt.fmt = AUDIO_FORMAT_S16;
s->dac_voice[0] = AUD_open_out(&s->card, s->dac_voice[0],
"tsc2102.sink", s, (void *) tsc210x_audio_out_cb, &fmt);
diff --git a/hw/usb/dev-audio.c b/hw/usb/dev-audio.c
index f092bb8..0171579 100644
--- a/hw/usb/dev-audio.c
+++ b/hw/usb/dev-audio.c
@@ -646,7 +646,7 @@ static void usb_audio_realize(USBDevice *dev, Error **errp)
s->out.vol[1] = 240; /* 0 dB */
s->out.as.freq = USBAUDIO_SAMPLE_RATE;
s->out.as.nchannels = 2;
- s->out.as.fmt = AUD_FMT_S16;
+ s->out.as.fmt = AUDIO_FORMAT_S16;
s->out.as.endianness = 0;
streambuf_init(&s->out.buf, s->buffer);
diff --git a/ui/vnc.c b/ui/vnc.c
index 0c6b5e3..f42ebc2 100644
--- a/ui/vnc.c
+++ b/ui/vnc.c
@@ -2379,12 +2379,12 @@ static int protocol_client_msg(VncState *vs, uint8_t *data, size_t len)
if (len == 4)
return 10;
switch (read_u8(data, 4)) {
- case 0: vs->as.fmt = AUD_FMT_U8; break;
- case 1: vs->as.fmt = AUD_FMT_S8; break;
- case 2: vs->as.fmt = AUD_FMT_U16; break;
- case 3: vs->as.fmt = AUD_FMT_S16; break;
- case 4: vs->as.fmt = AUD_FMT_U32; break;
- case 5: vs->as.fmt = AUD_FMT_S32; break;
+ case 0: vs->as.fmt = AUDIO_FORMAT_U8; break;
+ case 1: vs->as.fmt = AUDIO_FORMAT_S8; break;
+ case 2: vs->as.fmt = AUDIO_FORMAT_U16; break;
+ case 3: vs->as.fmt = AUDIO_FORMAT_S16; break;
+ case 4: vs->as.fmt = AUDIO_FORMAT_U32; break;
+ case 5: vs->as.fmt = AUDIO_FORMAT_S32; break;
default:
VNC_DEBUG("Invalid audio format %d\n", read_u8(data, 4));
vnc_client_error(vs);
@@ -3067,7 +3067,7 @@ void vnc_init_state(VncState *vs)
vs->as.freq = 44100;
vs->as.nchannels = 2;
- vs->as.fmt = AUD_FMT_S16;
+ vs->as.fmt = AUDIO_FORMAT_S16;
vs->as.endianness = 0;
qemu_mutex_init(&vs->output_mutex);
--
2.4.2
^ permalink raw reply related [flat|nested] 17+ messages in thread* [Qemu-devel] [PATCH 12/12] audio: -audiodev command line option
2015-06-12 12:33 [Qemu-devel] [PATCH 00/12] -audiodev option Kővágó, Zoltán
` (10 preceding siblings ...)
2015-06-12 12:33 ` [Qemu-devel] [PATCH 11/12] audio: use qapi AudioFormat instead of audfmt_e Kővágó, Zoltán
@ 2015-06-12 12:33 ` Kővágó, Zoltán
2015-06-15 9:01 ` [Qemu-devel] [PATCH 00/12] -audiodev option Gerd Hoffmann
12 siblings, 0 replies; 17+ messages in thread
From: Kővágó, Zoltán @ 2015-06-12 12:33 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
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, also print out them with -audio-help, to ease
migrating to -audiodev).
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 <DirtY.iCE.hu@gmail.com>
---
audio/Makefile.objs | 2 +-
audio/alsaaudio.c | 284 ++++++------------
audio/audio.c | 745 +++++++++++++-----------------------------------
audio/audio.h | 21 +-
audio/audio_int.h | 7 +-
audio/audio_legacy.c | 319 +++++++++++++++++++++
audio/audio_template.h | 13 +-
audio/coreaudio.c | 49 +---
audio/dsound_template.h | 6 +-
audio/dsoundaudio.c | 56 +---
audio/noaudio.c | 3 +-
audio/ossaudio.c | 155 +++-------
audio/paaudio.c | 81 ++----
audio/sdlaudio.c | 24 +-
audio/spiceaudio.c | 7 +-
audio/wavaudio.c | 61 +---
qemu-options.hx | 218 +++++++++++++-
vl.c | 9 +-
18 files changed, 979 insertions(+), 1081 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 6882638..06230c8 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -22,6 +22,8 @@
* THE SOFTWARE.
*/
#include <alsa/asoundlib.h>
+#include "qapi/alloc-visitor.h"
+#include "qapi-visit.h"
#include "qemu-common.h"
#include "qemu/main-loop.h"
#include "audio.h"
@@ -34,28 +36,9 @@
#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 +50,7 @@ typedef struct ALSAVoiceOut {
void *pcm_buf;
snd_pcm_t *handle;
struct pollhlp pollhlp;
+ Audiodev *dev;
} ALSAVoiceOut;
typedef struct ALSAVoiceIn {
@@ -74,16 +58,13 @@ 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 {
@@ -421,7 +402,8 @@ 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);
@@ -429,8 +411,9 @@ 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_usecs ? pdo->buffer_usecs : 0,
+ pdo->has_buffer_count ? pdo->buffer_count : 0);
dolog ("obtained: samples %ld\n", obt->samples);
}
@@ -464,23 +447,24 @@ 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->opts->alsa;
+ AudiodevAlsaPerDirectionOptions *apdo = in ? aopts->in : aopts->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);
@@ -540,79 +524,49 @@ 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_usecs) {
+ int64_t req = pdo->buffer_usecs * 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_usecs) {
+ int dir = 0;
+ unsigned int ptime = pdo->buffer_usecs;
- 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_usecs);
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_usecs && ptime != pdo->buffer_usecs) {
+ dolog("Requested period time %" PRId64 " was rejected, using %d\n",
+ pdo->buffer_usecs, ptime);
+ }
}
err = snd_pcm_hw_params (handle, hw_params);
@@ -644,7 +598,7 @@ static int alsa_open (int in, struct alsa_params_req *req,
goto err;
}
- if (!in && conf->threshold) {
+ if (!in && aopts->has_threshold && aopts->threshold) {
snd_pcm_uframes_t threshold;
int bytes_per_sec;
@@ -669,7 +623,7 @@ static int alsa_open (int in, struct alsa_params_req *req,
break;
}
- threshold = (conf->threshold * bytes_per_sec) / 1000;
+ threshold = (aopts->threshold * bytes_per_sec) / 1000;
alsa_set_threshold (handle, threshold);
}
@@ -683,11 +637,11 @@ 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;
@@ -813,19 +767,13 @@ 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;
}
@@ -846,7 +794,7 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
}
alsa->handle = handle;
- alsa->pollhlp.conf = conf;
+ alsa->dev = dev;
return 0;
}
@@ -886,16 +834,12 @@ 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->opts->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)) {
@@ -924,19 +868,13 @@ 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;
}
@@ -957,7 +895,7 @@ 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;
}
@@ -1099,16 +1037,12 @@ 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->opts->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)) {
@@ -1131,88 +1065,35 @@ 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->opts->kind == AUDIODEV_BACKEND_OPTIONS_KIND_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_usecs) {
+ dev->out->buffer_usecs = 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_usecs) {
+ dev->in->buffer_usecs = 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,
@@ -1230,7 +1111,6 @@ 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 112b57b..ecd7e6c 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -24,7 +24,11 @@
#include "hw/hw.h"
#include "audio.h"
#include "monitor/monitor.h"
+#include "qapi-visit.h"
+#include "qapi/alloc-visitor.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 +46,14 @@
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 +72,6 @@ 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 +79,9 @@ 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 av1474@comtv.ru\n");
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 +143,6 @@ 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";
-
- case AUDIO_FORMAT_MAX:
- 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 +161,6 @@ 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 +781,7 @@ 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 +857,7 @@ 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 +902,7 @@ 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 +1219,10 @@ 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 +1246,11 @@ 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 +1289,8 @@ 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 +1302,36 @@ static const VMStateDescription vmstate_audio = {
}
};
-static void audio_init (void)
+static Audiodev *parse_option(QemuOpts *opts);
+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;
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 = AudiodevBackendOptionsKind_lookup[dev->opts->kind];
+ } else {
+ audio_handle_legacy_opts();
+ list = qemu_find_opts("audiodev");
+ dev = parse_option(QTAILQ_FIRST(&list->head));
+ 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);
+ 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(1, get_ticks_per_sec(), dev->timer_period);
}
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 +1591,141 @@ 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 = "type",
+ .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 *err = NULL;
+ OptsVisitor *ov = opts_visitor_new(opts);
+ Audiodev *dev = NULL;
+ visit_type_Audiodev(opts_get_visitor(ov), &dev, NULL, &err);
+ opts_visitor_cleanup(ov);
+
+ if (err) {
+ error_report_err(err);
+ return NULL;
+ }
+
+ if (!dev->has_id) {
+ dev->has_id = true;
+ dev->id = g_strdup("default");
+ }
+
+ 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 = 100;
+ }
+
+ return dev;
+}
+
+static int each_option(QemuOpts *opts, void *opaque)
+{
+ Audiodev *dev = parse_option(opts);
+ if (!dev) {
+ return -1;
+ }
+ return audio_init(dev);
+}
+
+void audio_set_options(void)
+{
+ if (qemu_opts_foreach(qemu_find_opts("audiodev"), each_option, NULL, 0)) {
+ 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_usecs ? pdo->buffer_usecs : def_usecs;
+ return as->freq * usecs / 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..55d48e9 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -25,6 +25,7 @@
#define QEMU_AUDIO_H
#include "config-host.h"
+#include "qemu-common.h"
#include "qemu/queue.h"
typedef void (*audio_callback_fn) (void *opaque, int avail);
@@ -35,12 +36,21 @@ 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 +87,11 @@ 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 +165,8 @@ 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..571cb6f
--- /dev/null
+++ b/audio/audio_legacy.c
@@ -0,0 +1,319 @@
+#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,
+} 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" },
+ { /* End of list */ }
+};
+
+SimpleEnvMap alsa_map[] = {
+ { "QEMU_AUDIO_DAC_TRY_POLL", "out.try-poll", ENV_TRANSFORM_BOOL },
+ { "QEMU_AUDIO_ADC_TRY_POLL", "in.try-poll", ENV_TRANSFORM_BOOL },
+
+ { "QEMU_ALSA_THRESHOLD", "threshold" },
+ { "QEMU_ALSA_DAC_DEV", "out.dev" },
+ { "QEMU_ALSA_ADC_DEV", "in.dev" },
+
+ { /* End of list */ }
+};
+
+SimpleEnvMap coreaudio_map[] = {
+ { "QEMU_COREAUDIO_BUFFER_SIZE", "buffer-usecs",
+ ENV_TRANSFORM_FRAMES_TO_USECS_OUT },
+ { "QEMU_COREAUDIO_BUFFER_COUNT", "buffer-count" },
+
+ { /* End of list */ }
+};
+
+SimpleEnvMap dsound_map[] = {
+ { "QEMU_DSOUND_LATENCY_MILLIS", "latency-millis" },
+ { "QEMU_DSOUND_BUFSIZE_OUT", "out.buffer-usecs",
+ ENV_TRANSFORM_BYTES_TO_USECS_OUT },
+ { "QEMU_DSOUND_BUFSIZE_IN", "in.buffer-usecs",
+ ENV_TRANSFORM_BYTES_TO_USECS_IN },
+
+ { /* End of list */ }
+};
+
+SimpleEnvMap oss_map[] = {
+ { "QEMU_AUDIO_DAC_TRY_POLL", "out.try-poll", ENV_TRANSFORM_BOOL },
+ { "QEMU_AUDIO_ADC_TRY_POLL", "in.try-poll", ENV_TRANSFORM_BOOL },
+
+ { "QEMU_OSS_FRAGSIZE", "buffer-usecs", ENV_TRANSFORM_BYTES_TO_USECS_OUT },
+ { "QEMU_OSS_NFRAGS", "buffer-count" },
+ { "QEMU_OSS_MMAP", "mmap", ENV_TRANSFORM_BOOL },
+ { "QEMU_OSS_DAC_DEV", "out.dev" },
+ { "QEMU_OSS_ADC_DEV", "in.dev" },
+ { "QEMU_OSS_EXCLUSIVE", "exclusive", ENV_TRANSFORM_BOOL },
+ { "QEMU_OSS_POLICY", "dsp-policy" },
+
+ { /* End of list */ }
+};
+
+SimpleEnvMap pa_map[] = {
+ { "QEMU_PA_SAMPLES", "buffer-usecs", 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-usecs", 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;
+}
+
+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));
+ }
+
+ 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, bool usec, int period, int buffer,
+ const char *period_env, const char *buffer_env,
+ const char *usec_opt, const char *count_opt,
+ const char *freq_opt)
+{
+ char *period_s, *buffer_s;
+
+ period_s = getenv(period_env);
+ if (period_s) {
+ period = toull(period_s);
+ }
+ if (!usec) {
+ period = frames_to_usecs(opts, period, freq_opt);
+ }
+ 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, freq_opt);
+ }
+ qemu_opt_set(opts, count_opt, tostr((buffer+period-1)/period),
+ &error_abort);
+ }
+}
+
+static void handle_alsa(QemuOpts *opts)
+{
+ char *usec_s;
+ bool usec = false;
+
+ usec_s = getenv("DAC_SIZE_IN_USEC");
+ if (usec_s) {
+ usec = toull(usec_s);
+ }
+
+ handle_alsa_side(opts, usec, 1024, 4096,
+ "QEMU_ALSA_DAC_PERIOD_SIZE", "QEMU_ALSA_DAC_BUFFER_SIZE",
+ "out.buffer_usecs", "out.buffer_count", "out.frequency");
+ handle_alsa_side(opts, usec, 0, 0,
+ "QEMU_ALSA_ADC_PERIOD_SIZE", "QEMU_ALSA_ADC_BUFFER_SIZE",
+ "in.buffer_usecs", "in.buffer_count", "in.frequency");
+}
+
+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, "type", 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(QemuOpts *opts, void *opaque)
+{
+ 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, 1);
+}
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..922e431 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 {
@@ -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->opts->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_millis) {
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;
+ millis_to_bytes(&hw->info, dso->latency_millis) - 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->opts->kind == AUDIODEV_BACKEND_OPTIONS_KIND_DSOUND);
+ s->dev = dev;
+ dso = dev->opts->dsound;
+
+ if (!dso->has_latency_millis) {
+ dso->has_latency_millis = true;
+ dso->latency_millis = 10;
+ }
- 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 4f5bef6..4ebb9d1 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -29,6 +29,8 @@
#include "qemu-common.h"
#include "qemu/main-loop.h"
#include "qemu/host-utils.h"
+#include "qapi/alloc-visitor.h"
+#include "qapi-visit.h"
#include "audio.h"
#include "trace.h"
@@ -39,16 +41,6 @@
#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 +50,7 @@ typedef struct OSSVoiceOut {
int fragsize;
int mmapped;
int pending;
- OSSConf *conf;
+ Audiodev *dev;
} OSSVoiceOut;
typedef struct OSSVoiceIn {
@@ -67,12 +59,12 @@ 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 +256,26 @@ 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->opts->oss;
+ AudiodevOssPerDirectionOptions *opdo = in ? oopts->in : oopts->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_mmap && oopts->mmap) ?
+ O_RDWR : (in ? O_RDONLY : O_WRONLY);
fd = open (dspname, oflags | O_NONBLOCK);
if (-1 == fd) {
@@ -287,6 +286,8 @@ 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 +311,18 @@ 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 +505,16 @@ 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->opts->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 +541,7 @@ 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_mmap && oopts->mmap) {
oss->pcm_buf = mmap (
NULL,
hw->samples << hw->info.shift,
@@ -601,7 +601,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
}
oss->fd = fd;
- oss->conf = conf;
+ oss->dev = dev;
return 0;
}
@@ -609,16 +609,12 @@ static int oss_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
int trig;
OSSVoiceOut *oss = (OSSVoiceOut *) hw;
+ AudiodevOssPerDirectionOptions *opdo = oss->dev->opts->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 && oss_poll_out (hw)) {
@@ -672,16 +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;
}
@@ -715,7 +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;
}
@@ -806,16 +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->opts->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)) {
poll_mode = 0;
@@ -834,81 +824,25 @@ 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->opts->kind == AUDIODEV_BACKEND_OPTIONS_KIND_OSS);
- if (access(conf->devpath_in, R_OK | W_OK) < 0 ||
- access(conf->devpath_out, R_OK | W_OK) < 0) {
+ oopts = dev->opts->oss;
+ 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 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,
@@ -926,7 +860,6 @@ 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..60afb04 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -1,6 +1,8 @@
/* public domain */
#include "qemu-common.h"
#include "audio.h"
+#include "qapi/alloc-visitor.h"
+#include "qapi-visit.h"
#include <pulse/pulseaudio.h>
@@ -9,14 +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;
@@ -31,6 +26,7 @@ typedef struct {
void *pcm_buf;
struct audio_pt pt;
paaudio *g;
+ int samples;
} PAVoiceOut;
typedef struct {
@@ -45,6 +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);
@@ -226,7 +223,7 @@ 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 +315,7 @@ 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 +542,7 @@ 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->opts->pa;
ss.format = audfmt_to_pa (as->fmt, as->endianness);
ss.channels = as->nchannels;
@@ -565,7 +563,7 @@ 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 +575,8 @@ 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 +610,7 @@ 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->opts->pa;
ss.format = audfmt_to_pa (as->fmt, as->endianness);
ss.channels = as->nchannels;
@@ -622,7 +622,7 @@ 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 +634,8 @@ 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 +809,19 @@ 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->opts->kind == AUDIODEV_BACKEND_OPTIONS_KIND_PA);
+
+ g = g_malloc(sizeof(paaudio));
+ popts = dev->opts->pa;
+ server = popts->has_server ? popts->server : NULL;
+
+ g->dev = dev;
g->mainloop = NULL;
g->context = NULL;
@@ -825,14 +831,14 @@ 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 +901,6 @@ 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 +918,6 @@ 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 62017de..fa807b9 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -23,6 +23,8 @@
*/
#include "hw/hw.h"
#include "qemu/timer.h"
+#include "qapi/alloc-visitor.h"
+#include "qapi-visit.h"
#include "audio.h"
#define AUDIO_CAP "wav"
@@ -36,11 +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;
@@ -111,8 +108,10 @@ 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->opts->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 +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;
@@ -224,54 +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 = 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->opts->kind == AUDIODEV_BACKEND_OPTIONS_KIND_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 +245,6 @@ 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 1d281f6..e23cddc 100644
--- a/qemu-options.hx
+++ b/qemu-options.hx
@@ -312,14 +312,226 @@ 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[,prop[=value][,...]]\n"
+ " specifies the audio backend to use\n"
+ " id= identifier of the backend\n"
+ " timer-period= timer period in Hz\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-usecs= size of buffer in microseconds\n"
+ " [in.|out.]buffer-count= number of buffers\n"
+ "-audiodev none[,prop[=value][,...]]\n"
+ " dummy driver that discards all output\n"
+#ifdef CONFIG_ALSA
+ "-audiodev alsa[,prop[=value][,...]]\n"
+ " [in.|out]dev= name of the audio device to use\n"
+ " [in.|out]try-poll= attempt to use poll mode\n"
+ " threshold= threshold (in frames) when playback starts\n"
+#endif
+#ifdef CONFIG_COREAUDIO
+ "-audiodev coreaudio[,prop[=value][,...]]\n"
+#endif
+#ifdef CONFIG_DSOUND
+ "-audiodev dsound[,prop[=value][,...]]\n"
+ " latency-millis= add extra latency to playback\n"
+#endif
+#ifdef CONFIG_OSS
+ "-audiodev oss[,prop[=value][,...]]\n"
+ " [in.|out]dev= path of the audio device to use\n"
+ " [in.|out]try-poll= attempt to use poll mode\n"
+ " mmap= try using memory mapped access\n"
+ " exclusive= open device in exclusive mode\n"
+ " dsp-policy= set timing policy, -1 to use fragment mode\n"
+#endif
+#ifdef CONFIG_PA
+ "-audiodev pa[,prop[=value][,...]]\n"
+ " server= PulseAudio server address\n"
+ " sink= sink device name\n"
+ " source= source device name\n"
+#endif
+#ifdef CONFIG_SDL
+ "-audiodev sdl[,prop[=value][,...]]\n"
+#endif
+#ifdef CONFIG_SPICE
+ "-audiodev spice[,prop[=value][,...]]\n"
+#endif
+ "-audiodev wav[,prop[=value][,...]]\n"
+ " path= path of wav file to record\n",
+ QEMU_ARCH_ALL)
+STEXI
+@item -audiodev @var{driver}[,@var{prop}[=@var{value}][,...]]
+@findex -audiodev
+Adds a new audio backend @var{driver}. There are global and driver
+specific properties. Some values can be set differently for input and
+output, they're marked with @code{[in.|out.]}. To only set the input's
+option prefix the property with @code{in.}, to only set the output
+prefix it with @code{out.}, or leave them to specify both. Thus the two
+examples are equal:
+@example
+-audiodev alsa,in.frequency=44110,out.frequency=44110
+-audiodev alsa,frequency=44100
+@end example
+
+Valid global options are:
+
+@table @option
+@item id=@var{identifier}
+Identifies the audio backend.
+
+@item timer-period=@var{period}
+Sets the timer @var{period} used by the audio subsystem, in Hz.
+
+@item [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.
+
+@item [in.|out.]frequency=@var{frequency}
+Specify the @var{frequency} to use when using @var{fixed-settings}.
+
+@item [in.|out.]channels=@var{channels}
+Specify the number of @var{channels} to use when using
+@var{fixed-settings}.
+
+@item [in.|out.]format=@var{format}
+Specify the sample @var{format} to use when using @var{fixed-settings}.
+Valid values are: @code{s8}, @code{s16}, @code{s32}, @code{u8},
+@code{u16}, @code{u32}.
+
+@item [in.|out.]voices=@var{voices}
+Specify the number of @var{voices} to use.
+
+@item [in.|out.]buffer-usecs=@var{usecs}
+Sets the size of the buffer in microseconds.
+
+@item [in.|out.]buffer-count=@var{count}
+Sets the @var{count} of the buffers.
+
+@end table
+
+@item -audiodev none[,@var{prop}[=@var{value}][,...]]
+Creates a dummy backend that discards all outputs. This backend has no
+backend specific properties.
+
+@item -audiodev alsa[,@var{prop}[=@var{value}][,...]]
+Creates backend using the ALSA. This backend is only available on
+Linux.
+
+ALSA specific options are:
+
+@table @option
+@item [in.|out.]dev=@var{device}
+Specify the ALSA @var{device} to use for input and/or output.
+
+@item [in.|out.]try-poll=on|of
+Attempt to use poll mode with the device.
+
+@item threshold=@var{threshold}
+Threshold (in frames) when playback starts.
+
+@end table
+
+@item -audiodev coreaudio[,@var{prop}[=@var{value}][,...]]
+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.
+
+@item -audiodev dsound[,@var{prop}[=@var{value}][,...]]
+Creates a backend using Microsof's DirectSound. This backend is only
+available on Windows and only supports playback.
+
+Backend specific options are:
+
+@table @option
+
+@item latency-millis=@var{millis}
+Add extra @var{millis} milliseconds latency to playback.
+
+@end table
+
+@item -audiodev oss[,@var{prop}[=@var{value}][,...]]
+Creates a backend using OSS. This backend is available on most
+Unix-like systems.
+
+OSS specific options are:
+
+@table @option
+
+@item [in.|out.]dev=@var{path}
+Specify the @var{path} of the OSS device to use.
+
+@item [in.|out.]try-poll=on|of
+Attempt to use poll mode with the device.
+
+@item mmap=on|off
+Try using memory mapped device access.
+
+@item exclusive=on|off
+Open the device in exclusive mode (vmix won't work in this case).
+
+@item dsp-policy=@var{policy}
+Sets the timing policy (between 0 and 10). Use -1 to use buffer sizes
+specified by @code{buffer-usecs} and @code{buffer-count}. This option
+is ignored if you do not have OSS 4.
+
+@end table
+
+@item -audiodev pa[,@var{prop}[=@var{value}][,...]]
+Creates a backend using PulseAudio. This backend is available on most
+systems.
+
+PulseAudio specific options are:
+
+@table @option
+
+@item server=@var{server}
+Sets the PulseAudio @var{server} to connect to.
+
+@item sink=@var{sink}
+Use the specified @var{sink} for playback.
+
+@item source=@var{source}
+Use the specified @var{source} for recording.
+
+@end table
+
+@item -audiodev sdl[,@var{prop}[=@var{value}][,...]]
+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.
+
+@item -audiodev spice[,@var{prop}[=@var{value}][,...]]
+Creates a backend that sends audio through SPICE. This backend requires
+@code{-spice} and automatically selected in that case, so usually you
+can ignore this option. This backend has no backend specific
+properties.
+
+@item -audiodev wav[,@var{prop}[=@var{value}][,...]]
+Creates a backend that writes audio to a WAV file.
+
+Backend specific options are:
+
+@table @option
+
+@item path=@var{path}
+Write recorded audio into the specified file.
+
+@end table
ETEXI
DEF("soundhw", HAS_ARG, QEMU_OPTION_soundhw,
diff --git a/vl.c b/vl.c
index 9542095..10a0aee 100644
--- a/vl.c
+++ b/vl.c
@@ -2860,6 +2860,7 @@ 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);
@@ -3157,9 +3158,14 @@ 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;
@@ -4326,6 +4332,7 @@ int main(int argc, char **argv, char **envp)
realtime_init();
+ audio_set_options();
audio_init();
cpu_synchronize_all_post_init();
--
2.4.2
^ permalink raw reply related [flat|nested] 17+ messages in thread