* [Qemu-devel] [PATCH v2 01/52] qapi: support alternates in OptsVisitor
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
@ 2018-12-23 20:51 ` Kővágó, Zoltán
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 02/52] qapi: support nested structs " Kővágó, Zoltán
` (49 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:51 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann, Markus Armbruster, Michael Roth
They are required for flat unions (you still have to allocate the
structs).
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
include/qapi/opts-visitor.h | 7 +++----
qapi/opts-visitor.c | 16 ++++++++++++++++
2 files changed, 19 insertions(+), 4 deletions(-)
diff --git a/include/qapi/opts-visitor.h b/include/qapi/opts-visitor.h
index 9b989e7e08..ca044e3b33 100644
--- a/include/qapi/opts-visitor.h
+++ b/include/qapi/opts-visitor.h
@@ -29,10 +29,9 @@ typedef struct OptsVisitor OptsVisitor;
* - values below INT64_MIN or LLONG_MIN are rejected,
* - values above INT64_MAX or LLONG_MAX are rejected.
*
- * The Opts input visitor does not implement support for visiting QAPI
- * alternates, numbers (other than integers), null, or arbitrary
- * QTypes. It also requires a non-null list argument to
- * visit_start_list().
+ * The Opts input visitor does not implement support for visiting numbers
+ * (other than integers), null, or arbitrary QTypes. It also requires a
+ * non-null list argument to visit_start_list().
*/
Visitor *opts_visitor_new(const QemuOpts *opts);
diff --git a/qapi/opts-visitor.c b/qapi/opts-visitor.c
index 324b197495..4af6043b75 100644
--- a/qapi/opts-visitor.c
+++ b/qapi/opts-visitor.c
@@ -156,6 +156,13 @@ opts_start_struct(Visitor *v, const char *name, void **obj,
}
}
+static void
+opts_start_alternate(Visitor *v, const char *name, GenericAlternate **obj,
+ size_t size, Error **errp)
+{
+ opts_start_struct(v, name, (void **)obj, size, errp);
+}
+
static void
opts_check_struct(Visitor *v, Error **errp)
@@ -198,6 +205,12 @@ opts_end_struct(Visitor *v, void **obj)
ov->fake_id_opt = NULL;
}
+static void
+opts_end_alternate(Visitor *v, void **obj)
+{
+ opts_end_struct(v, obj);
+}
+
static GQueue *
lookup_distinct(const OptsVisitor *ov, const char *name, Error **errp)
@@ -547,6 +560,9 @@ opts_visitor_new(const QemuOpts *opts)
ov->visitor.check_struct = &opts_check_struct;
ov->visitor.end_struct = &opts_end_struct;
+ ov->visitor.start_alternate = &opts_start_alternate;
+ ov->visitor.end_alternate = &opts_end_alternate;
+
ov->visitor.start_list = &opts_start_list;
ov->visitor.next_list = &opts_next_list;
ov->visitor.check_list = &opts_check_list;
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 02/52] qapi: support nested structs in OptsVisitor
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 01/52] qapi: support alternates in OptsVisitor Kővágó, Zoltán
@ 2018-12-23 20:51 ` Kővágó, Zoltán
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 03/52] qapi: qapi for audio backends Kővágó, Zoltán
` (48 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:51 UTC (permalink / raw)
To: qemu-devel
Cc: Gerd Hoffmann, Kővágó,
Zoltán, Michael S. Tsirkin, Igor Mammedov,
Markus Armbruster, Michael Roth, Jason Wang, Eduardo Habkost,
Andreas Färber
From: Kővágó, Zoltán <dirty.ice.hu@gmail.com>
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 always requiring a complete path in case of
nested structs. Fields in the path are separated by dots, similar to C
structs (without pointers), like `in.frequency' or `out.frequency'. You
must provide a full path even in non-ambiguous cases.
To keep backward compatibility, this new feature can be disabled when
creating a new OptsVisitor, in that case it will work identical to
previous versions.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
hw/acpi/core.c | 2 +-
include/qapi/opts-visitor.h | 2 +-
net/net.c | 2 +-
numa.c | 2 +-
qapi/opts-visitor.c | 129 ++++++++++++++++++++----
qom/object_interfaces.c | 2 +-
tests/qapi-schema/qapi-schema-test.json | 9 +-
tests/qapi-schema/qapi-schema-test.out | 4 +
tests/test-opts-visitor.c | 43 +++++++-
9 files changed, 163 insertions(+), 32 deletions(-)
diff --git a/hw/acpi/core.c b/hw/acpi/core.c
index d6f0709691..654508ac13 100644
--- a/hw/acpi/core.c
+++ b/hw/acpi/core.c
@@ -243,7 +243,7 @@ void acpi_table_add(const QemuOpts *opts, Error **errp)
{
Visitor *v;
- v = opts_visitor_new(opts);
+ v = opts_visitor_new(opts, false);
visit_type_AcpiTableOptions(v, NULL, &hdrs, &err);
visit_free(v);
}
diff --git a/include/qapi/opts-visitor.h b/include/qapi/opts-visitor.h
index ca044e3b33..84edb49f7a 100644
--- a/include/qapi/opts-visitor.h
+++ b/include/qapi/opts-visitor.h
@@ -33,6 +33,6 @@ typedef struct OptsVisitor OptsVisitor;
* (other than integers), null, or arbitrary QTypes. It also requires a
* non-null list argument to visit_start_list().
*/
-Visitor *opts_visitor_new(const QemuOpts *opts);
+Visitor *opts_visitor_new(const QemuOpts *opts, bool nested);
#endif
diff --git a/net/net.c b/net/net.c
index 1f7d626197..f5e7d8a6ef 100644
--- a/net/net.c
+++ b/net/net.c
@@ -1106,7 +1106,7 @@ static int net_client_init(QemuOpts *opts, bool is_netdev, Error **errp)
void *object = NULL;
Error *err = NULL;
int ret = -1;
- Visitor *v = opts_visitor_new(opts);
+ Visitor *v = opts_visitor_new(opts, false);
const char *type = qemu_opt_get(opts, "type");
diff --git a/numa.c b/numa.c
index 50ec016013..31a24f750f 100644
--- a/numa.c
+++ b/numa.c
@@ -220,7 +220,7 @@ static int parse_numa(void *opaque, QemuOpts *opts, Error **errp)
NumaOptions *object = NULL;
MachineState *ms = MACHINE(opaque);
Error *err = NULL;
- Visitor *v = opts_visitor_new(opts);
+ Visitor *v = opts_visitor_new(opts, false);
visit_type_NumaOptions(v, NULL, &object, &err);
visit_free(v);
diff --git a/qapi/opts-visitor.c b/qapi/opts-visitor.c
index 4af6043b75..d8af51b16c 100644
--- a/qapi/opts-visitor.c
+++ b/qapi/opts-visitor.c
@@ -72,6 +72,7 @@ struct OptsVisitor
* 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
@@ -87,6 +88,9 @@ 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;
};
@@ -107,6 +111,7 @@ static void
opts_visitor_insert(GHashTable *unprocessed_opts, const QemuOpt *opt)
{
GQueue *list;
+ assert(opt);
list = g_hash_table_lookup(unprocessed_opts, opt->name);
if (list == NULL) {
@@ -134,6 +139,11 @@ opts_start_struct(Visitor *v, const char *name, void **obj,
if (obj) {
*obj = g_malloc0(size);
}
+
+ if (ov->nested_names != NULL) {
+ g_queue_push_tail(ov->nested_names, (gpointer) name);
+ }
+
if (ov->depth++ > 0) {
return;
}
@@ -171,6 +181,10 @@ opts_check_struct(Visitor *v, Error **errp)
GHashTableIter iter;
GQueue *any;
+ if (ov->nested_names != NULL) {
+ g_queue_pop_tail(ov->nested_names);
+ }
+
if (ov->depth > 1) {
return;
}
@@ -212,15 +226,59 @@ opts_end_alternate(Visitor *v, void **obj)
}
+static void
+sum_strlen(gpointer data, gpointer user_data)
+{
+ const char *str = data;
+ size_t *sum_len = user_data;
+
+ if (str) { /* skip NULLs */
+ *sum_len += strlen(str) + 1;
+ }
+}
+
+static void
+append_str(gpointer data, gpointer user_data)
+{
+ const char *str = data;
+ char *concat_str = user_data;
+
+ if (str) {
+ strcat(concat_str, str);
+ strcat(concat_str, ".");
+ }
+}
+
+/* lookup a name, using a fully qualified version */
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;
+
+ if (ov->nested_names != NULL) {
+ size_t sum_len = strlen(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);
+ } else {
+ key = g_strdup(name);
+ }
+
+ list = g_hash_table_lookup(ov->unprocessed_opts, key);
+ if (list && out_key) {
+ *out_key = key;
+ key = NULL;
+ }
- list = g_hash_table_lookup(ov->unprocessed_opts, name);
if (!list) {
error_setg(errp, QERR_MISSING_PARAMETER, name);
}
+
+ g_free(key);
return list;
}
@@ -235,7 +293,8 @@ opts_start_list(Visitor *v, const char *name, GenericList **list, size_t size,
assert(ov->list_mode == LM_NONE);
/* we don't support visits without a list */
assert(list);
- ov->repeated_opts = lookup_distinct(ov, name, errp);
+ assert(ov->repeated_opts == NULL && ov->repeated_name == NULL);
+ ov->repeated_opts = lookup_distinct(ov, name, &ov->repeated_name, errp);
if (ov->repeated_opts) {
ov->list_mode = LM_IN_PROGRESS;
*list = g_malloc0(size);
@@ -266,11 +325,9 @@ opts_next_list(Visitor *v, GenericList *tail, size_t size)
/* 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;
}
break;
@@ -304,22 +361,28 @@ opts_end_list(Visitor *v, void **obj)
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);
}
@@ -341,8 +404,9 @@ opts_type_str(Visitor *v, const char *name, char **obj, Error **errp)
{
OptsVisitor *ov = to_ov(v);
const QemuOpt *opt;
+ char *key = NULL;
- opt = lookup_scalar(ov, name, errp);
+ opt = lookup_scalar(ov, name, &key, errp);
if (!opt) {
*obj = NULL;
return;
@@ -353,7 +417,8 @@ opts_type_str(Visitor *v, const char *name, char **obj, Error **errp)
* valid enum value; this is harmless because tracking what gets
* consumed only matters to visit_end_struct() as the final error
* check if there were no other failures during the visit. */
- processed(ov, name);
+ processed(ov, key);
+ g_free(key);
}
@@ -363,8 +428,9 @@ opts_type_bool(Visitor *v, const char *name, bool *obj, Error **errp)
{
OptsVisitor *ov = to_ov(v);
const QemuOpt *opt;
+ char *key = NULL;
- opt = lookup_scalar(ov, name, errp);
+ opt = lookup_scalar(ov, name, &key, errp);
if (!opt) {
return;
}
@@ -381,13 +447,15 @@ opts_type_bool(Visitor *v, const char *name, bool *obj, Error **errp)
} else {
error_setg(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);
}
@@ -399,13 +467,14 @@ opts_type_int64(Visitor *v, const char *name, int64_t *obj, 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;
}
@@ -419,11 +488,13 @@ opts_type_int64(Visitor *v, const char *name, int64_t *obj, 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);
@@ -444,6 +515,7 @@ opts_type_int64(Visitor *v, const char *name, int64_t *obj, Error **errp)
error_setg(errp, QERR_INVALID_PARAMETER_VALUE, opt->name,
(ov->list_mode == LM_NONE) ? "an int64 value" :
"an int64 value or range");
+ g_free(key);
}
@@ -455,13 +527,14 @@ opts_type_uint64(Visitor *v, const char *name, uint64_t *obj, 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;
}
@@ -473,11 +546,13 @@ opts_type_uint64(Visitor *v, const char *name, uint64_t *obj, 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 &&
@@ -496,6 +571,7 @@ opts_type_uint64(Visitor *v, const char *name, uint64_t *obj, Error **errp)
error_setg(errp, QERR_INVALID_PARAMETER_VALUE, opt->name,
(ov->list_mode == LM_NONE) ? "a uint64 value" :
"a uint64 value or range");
+ g_free(key);
}
@@ -505,8 +581,9 @@ opts_type_size(Visitor *v, const char *name, uint64_t *obj, Error **errp)
OptsVisitor *ov = to_ov(v);
const QemuOpt *opt;
int err;
+ char *key = NULL;
- opt = lookup_scalar(ov, name, errp);
+ opt = lookup_scalar(ov, name, &key, errp);
if (!opt) {
return;
}
@@ -515,10 +592,12 @@ opts_type_size(Visitor *v, const char *name, uint64_t *obj, Error **errp)
if (err < 0) {
error_setg(errp, QERR_INVALID_PARAMETER_VALUE, opt->name,
"a size value");
+ g_free(key);
return;
}
- processed(ov, name);
+ processed(ov, key);
+ g_free(key);
}
@@ -529,7 +608,7 @@ opts_optional(Visitor *v, const char *name, bool *present)
/* 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);
}
@@ -547,7 +626,7 @@ opts_free(Visitor *v)
Visitor *
-opts_visitor_new(const QemuOpts *opts)
+opts_visitor_new(const QemuOpts *opts, bool nested)
{
OptsVisitor *ov;
@@ -556,6 +635,12 @@ opts_visitor_new(const QemuOpts *opts)
ov->visitor.type = VISITOR_INPUT;
+ if (nested) {
+ ov->nested_names = g_queue_new();
+ } else {
+ ov->nested_names = NULL;
+ }
+
ov->visitor.start_struct = &opts_start_struct;
ov->visitor.check_struct = &opts_check_struct;
ov->visitor.end_struct = &opts_end_struct;
diff --git a/qom/object_interfaces.c b/qom/object_interfaces.c
index db85d1eb75..17f995ee31 100644
--- a/qom/object_interfaces.c
+++ b/qom/object_interfaces.c
@@ -119,7 +119,7 @@ Object *user_creatable_add_opts(QemuOpts *opts, Error **errp)
qemu_opts_set_id(opts, NULL);
pdict = qemu_opts_to_qdict(opts, NULL);
- v = opts_visitor_new(opts);
+ v = opts_visitor_new(opts, false);
obj = user_creatable_add_type(type, id, pdict, v, errp);
visit_free(v);
diff --git a/tests/qapi-schema/qapi-schema-test.json b/tests/qapi-schema/qapi-schema-test.json
index cb0857df52..324b9eb40c 100644
--- a/tests/qapi-schema/qapi-schema-test.json
+++ b/tests/qapi-schema/qapi-schema-test.json
@@ -149,6 +149,11 @@
# Smoke test on out-of-band and allow-preconfig-test
{ 'command': 'test-flags-command', 'allow-oob': true, 'allow-preconfig': true }
+# 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:
#
@@ -162,7 +167,9 @@
'*u64' : [ 'uint64' ],
'*u16' : [ 'uint16' ],
'*i64x': 'int' ,
- '*u64x': 'uint64' } }
+ '*u64x': 'uint64' ,
+ 'sub0': 'UserDefOptionsSub',
+ 'sub1': 'UserDefOptionsSub' } }
# testing event
{ 'struct': 'EventStructOne',
diff --git a/tests/qapi-schema/qapi-schema-test.out b/tests/qapi-schema/qapi-schema-test.out
index 9464101d26..3c1b47c7cd 100644
--- a/tests/qapi-schema/qapi-schema-test.out
+++ b/tests/qapi-schema/qapi-schema-test.out
@@ -200,12 +200,16 @@ command boxed-union UserDefNativeListUnion -> None
gen=True success_response=True boxed=True oob=False preconfig=False
command test-flags-command None -> None
gen=True success_response=True boxed=False oob=True preconfig=True
+object UserDefOptionsSub
+ member nint: int optional=True
object UserDefOptions
member i64: intList optional=True
member u64: uint64List optional=True
member u16: uint16List optional=True
member i64x: int optional=True
member u64x: uint64 optional=True
+ member sub0: UserDefOptionsSub optional=False
+ member sub1: UserDefOptionsSub optional=False
object EventStructOne
member struct1: UserDefOne optional=False
member string: str optional=False
diff --git a/tests/test-opts-visitor.c b/tests/test-opts-visitor.c
index 23e897061c..52950dfc4a 100644
--- a/tests/test-opts-visitor.c
+++ b/tests/test-opts-visitor.c
@@ -43,7 +43,7 @@ setup_fixture(OptsVisitorFixture *f, gconstpointer test_data)
NULL);
g_assert(opts != NULL);
- v = opts_visitor_new(opts);
+ v = opts_visitor_new(opts, true);
visit_type_UserDefOptions(v, NULL, &f->userdef, &f->err);
visit_free(v);
qemu_opts_del(opts);
@@ -170,6 +170,34 @@ 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 == 17);
+}
+
+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 */
static void
@@ -184,7 +212,7 @@ test_opts_range_unvisited(void)
opts = qemu_opts_parse(qemu_find_opts("userdef"), "ilist=0-2", false,
&error_abort);
- v = opts_visitor_new(opts);
+ v = opts_visitor_new(opts, false);
visit_start_struct(v, NULL, NULL, 0, &error_abort);
@@ -225,7 +253,7 @@ test_opts_range_beyond(void)
opts = qemu_opts_parse(qemu_find_opts("userdef"), "ilist=0", false,
&error_abort);
- v = opts_visitor_new(opts);
+ v = opts_visitor_new(opts, false);
visit_start_struct(v, NULL, NULL, 0, &error_abort);
@@ -260,7 +288,7 @@ test_opts_dict_unvisited(void)
opts = qemu_opts_parse(qemu_find_opts("userdef"), "i64x=0,bogus=1", false,
&error_abort);
- v = opts_visitor_new(opts);
+ v = opts_visitor_new(opts, false);
visit_type_UserDefOptions(v, NULL, &userdef, &err);
error_free_or_abort(&err);
visit_free(v);
@@ -366,6 +394,13 @@ main(int argc, char **argv)
g_test_add_func("/visitor/opts/dict/unvisited", test_opts_dict_unvisited);
+ /* Test nested structs support */
+ add_test("/visitor/opts/nested/unqualified", &expect_fail, "nint=13");
+ add_test("/visitor/opts/nested/both", &expect_both,
+ "sub0.nint=13,sub1.nint=17");
+ 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;
}
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 03/52] qapi: qapi for audio backends
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 01/52] qapi: support alternates in OptsVisitor Kővágó, Zoltán
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 02/52] qapi: support nested structs " Kővágó, Zoltán
@ 2018-12-23 20:51 ` Kővágó, Zoltán
2019-01-10 2:49 ` Eric Blake
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 04/52] audio: use qapi AudioFormat instead of audfmt_e Kővágó, Zoltán
` (47 subsequent siblings)
50 siblings, 1 reply; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:51 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann, Eric Blake, Markus Armbruster
This patch adds structures into qapi to replace the existing
configuration structures used by audio backends currently. This qapi
will be the base of the -audiodev command line parameter (that replaces
the old environment variables based config).
This is not a 1:1 translation of the old options, I've tried to make
them much more consistent (e.g. almost every backend had an option to
specify buffer size, but the name was different for every backend, and
some backends required usecs, while some other required frames, samples
or bytes). Also tried to reduce the number of abbreviations used by the
config keys.
Some of the more important changes:
* use `in` and `out` instead of `ADC` and `DAC`, as the former is more
user friendly imho
* moved buffer settings into the global setting area (so it's the same
for all backends that support it. Backends that can't change buffer
size will simply ignore them). Also using usecs, as it's probably more
user friendly than samples or bytes.
* try-poll is now an alsa backend specific option (as all other backends
currently ignore it)
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
Makefile.objs | 6 +-
qapi/audio.json | 253 ++++++++++++++++++++++++++++++++++++++++++
qapi/qapi-schema.json | 1 +
3 files changed, 257 insertions(+), 3 deletions(-)
create mode 100644 qapi/audio.json
diff --git a/Makefile.objs b/Makefile.objs
index bc5b8a8442..3f833a70c0 100644
--- a/Makefile.objs
+++ b/Makefile.objs
@@ -1,6 +1,6 @@
-QAPI_MODULES = block-core block char common crypto introspect job migration
-QAPI_MODULES += misc net rdma rocker run-state sockets tpm trace transaction
-QAPI_MODULES += ui
+QAPI_MODULES = audio block-core block char common crypto introspect job
+QAPI_MODULES += migration misc net rdma rocker run-state sockets tpm trace
+QAPI_MODULES += transaction ui
#######################################################################
# Common libraries for tools and emulators
diff --git a/qapi/audio.json b/qapi/audio.json
new file mode 100644
index 0000000000..56d8ce439f
--- /dev/null
+++ b/qapi/audio.json
@@ -0,0 +1,253 @@
+# -*- mode: python -*-
+#
+# Copyright (C) 2015 Zoltán Kővágó <DirtY.iCE.hu@gmail.com>
+#
+# This work is licensed under the terms of the GNU GPL, version 2 or later.
+# See the COPYING file in the top-level directory.
+
+##
+# @AudiodevNoOptions:
+#
+# The none, coreaudio, sdl and spice audio backend have no options.
+#
+# Since: 3.2
+##
+{ 'struct': 'AudiodevNoOptions',
+ 'data': { } }
+
+##
+# @AudiodevAlsaPerDirectionOptions:
+#
+# Options of the alsa backend that are used for both playback and recording.
+#
+# @dev: #optional the name of the alsa device to use (default 'default')
+#
+# @try-poll: #optional attempt to use poll mode, falling back to non polling
+# access on failure (default on)
+#
+# Since: 3.2
+##
+{ 'struct': 'AudiodevAlsaPerDirectionOptions',
+ 'data': {
+ '*dev': 'str',
+ '*try-poll': 'bool' } }
+
+##
+# @AudiodevAlsaOptions:
+#
+# Options of the alsa audio backend.
+#
+# @alsa-in: options of the capture stream
+#
+# @alsa-out: options of the playback stream
+#
+# @threshold: #optional set the threshold (in microseconds) when playback starts
+#
+# Since: 3.2
+##
+{ 'struct': 'AudiodevAlsaOptions',
+ 'data': {
+ 'alsa-in': 'AudiodevAlsaPerDirectionOptions',
+ 'alsa-out': 'AudiodevAlsaPerDirectionOptions',
+ '*threshold': 'int' } }
+
+##
+# @AudiodevDsoundOptions:
+#
+# Options of the dsound audio backend.
+#
+# @latency: #optional add extra latency to playback in microseconds (default
+# 10000)
+#
+# Since: 3.2
+##
+{ 'struct': 'AudiodevDsoundOptions',
+ 'data': {
+ '*latency': 'int' } }
+
+##
+# @AudiodevOssPerDirectionOptions:
+#
+# Options of the oss backend that are used for both playback and recording.
+#
+# @dev: #optional file name of the oss device (default '/dev/dsp')
+#
+# @try-poll: #optional attempt to use poll mode, falling back to non polling
+# access on failure (default on)
+#
+# Since: 3.2
+##
+{ 'struct': 'AudiodevOssPerDirectionOptions',
+ 'data': {
+ '*dev': 'str',
+ '*try-poll': 'bool' } }
+
+##
+# @AudiodevOssOptions:
+#
+# Options of the oss audio backend.
+#
+# @oss-in: options of the capture stream
+#
+# @oss-out: options of the playback stream
+#
+# @try-mmap: #optional try using memory mapped access, falling back to non
+# memory mapped access on failure (default off)
+#
+# @exclusive: #optional open device in exclusive mode (vmix won't work)
+# (default off)
+#
+# @dsp-policy: #optional set the timing policy of the device (between 0 and 10,
+# where smaller number means smaller latency but higher CPU usage)
+# or -1 to use fragment mode (option ignored on some platforms)
+# (default 5)
+#
+# Since: 3.2
+##
+{ 'struct': 'AudiodevOssOptions',
+ 'data': {
+ 'oss-in': 'AudiodevOssPerDirectionOptions',
+ 'oss-out': 'AudiodevOssPerDirectionOptions',
+ '*try-mmap': 'bool',
+ '*exclusive': 'bool',
+ '*dsp-policy': 'int' } }
+
+##
+# @AudiodevPaPerDirectionOptions:
+#
+# Options of the pa backend that are used for both playback and recording.
+#
+# @name: #optional name of the sink/source to use
+#
+# Since: 3.2
+##
+{ 'struct': 'AudiodevPaPerDirectionOptions',
+ 'data': {
+ '*name': 'str' } }
+
+##
+# @AudiodevPaOptions:
+#
+# Options of the pa (PulseAudio) audio backend.
+#
+# @server: #optional PulseAudio server address (default: let PulseAudio choose)
+#
+# @sink: sink specific options
+#
+# @source: source specific options
+#
+# Since: 3.2
+##
+{ 'struct': 'AudiodevPaOptions',
+ 'data': {
+ '*server': 'str',
+ 'sink': 'AudiodevPaPerDirectionOptions',
+ 'source': 'AudiodevPaPerDirectionOptions' } }
+
+##
+# @AudiodevWavOptions:
+#
+# Options of the wav audio backend.
+#
+# @path: #optional name of the wav file to record (default 'qemu.wav')
+#
+# Since: 3.2
+##
+{ 'struct': 'AudiodevWavOptions',
+ 'data': {
+ '*path': 'str' } }
+
+
+##
+# @AudioFormat:
+#
+# An enumeration of possible audio formats.
+#
+# Since: 3.2
+##
+{ 'enum': 'AudioFormat',
+ 'data': [ 'u8', 's8', 'u16', 's16', 'u32', 's32' ] }
+
+##
+# @AudiodevDriver:
+#
+# An enumeration of possible audio backend drivers.
+#
+# Since: 3.2
+##
+{ 'enum': 'AudiodevDriver',
+ 'data': [ 'none', 'alsa', 'coreaudio', 'dsound', 'oss', 'pa', 'sdl', 'spice',
+ 'wav' ] }
+
+##
+# @AudiodevPerDirectionOptions:
+#
+# General audio backend options that are used for both playback and recording.
+#
+# @fixed-settings: #optional use fixed settings for host input/output. When
+# off, frequency, channels and format must not be specified
+# (default on)
+#
+# @frequency: #optional frequency to use when using fixed settings
+# (default 44100)
+#
+# @channels: #optional number of channels when using fixed settings
+# (default 2)
+#
+# @voices: #optional number of voices to use (default 1)
+#
+# @format: #optional sample format to use when using fixed settings
+# (default s16)
+#
+# @buffer-len: #optional the buffer size in microseconds
+#
+# @buffer-count: #optional number of buffers
+#
+# Since: 3.2
+##
+{ 'struct': 'AudiodevPerDirectionOptions',
+ 'data': {
+ '*fixed-settings': 'bool',
+ '*frequency': 'int',
+ '*channels': 'int',
+ '*voices': 'int',
+ '*format': 'AudioFormat',
+ '*buffer-len': 'int',
+ '*buffer-count': 'int' } }
+
+##
+# @Audiodev:
+#
+# Options of an audio backend.
+#
+# @id: identifier of the backend
+#
+# @driver: the backend driver to use
+#
+# @in: options of the capture stream
+#
+# @out: options of the playback stream
+#
+# @timer-period: #optional timer period (in microseconds, 0: use lowest
+# possible)
+#
+# Since: 3.2
+##
+{ 'union': 'Audiodev',
+ 'base': {
+ 'id': 'str',
+ 'driver': 'AudiodevDriver',
+ 'in': 'AudiodevPerDirectionOptions',
+ 'out': 'AudiodevPerDirectionOptions',
+ '*timer-period': 'int' },
+ 'discriminator': 'driver',
+ 'data': {
+ 'none': 'AudiodevNoOptions',
+ 'alsa': 'AudiodevAlsaOptions',
+ 'coreaudio': 'AudiodevNoOptions',
+ 'dsound': 'AudiodevDsoundOptions',
+ 'oss': 'AudiodevOssOptions',
+ 'pa': 'AudiodevPaOptions',
+ 'sdl': 'AudiodevNoOptions',
+ 'spice': 'AudiodevNoOptions',
+ 'wav': 'AudiodevWavOptions' } }
diff --git a/qapi/qapi-schema.json b/qapi/qapi-schema.json
index 3bbdfcee84..d5f9856be7 100644
--- a/qapi/qapi-schema.json
+++ b/qapi/qapi-schema.json
@@ -95,3 +95,4 @@
{ 'include': 'trace.json' }
{ 'include': 'introspect.json' }
{ 'include': 'misc.json' }
+{ 'include': 'audio.json' }
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* Re: [Qemu-devel] [PATCH v2 03/52] qapi: qapi for audio backends
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 03/52] qapi: qapi for audio backends Kővágó, Zoltán
@ 2019-01-10 2:49 ` Eric Blake
0 siblings, 0 replies; 91+ messages in thread
From: Eric Blake @ 2019-01-10 2:49 UTC (permalink / raw)
To: Kővágó, Zoltán, qemu-devel
Cc: Gerd Hoffmann, Markus Armbruster
[-- Attachment #1: Type: text/plain, Size: 4113 bytes --]
On 12/23/18 2:51 PM, Kővágó, Zoltán wrote:
> This patch adds structures into qapi to replace the existing
> configuration structures used by audio backends currently. This qapi
> will be the base of the -audiodev command line parameter (that replaces
> the old environment variables based config).
>
> This is not a 1:1 translation of the old options, I've tried to make
> them much more consistent (e.g. almost every backend had an option to
> specify buffer size, but the name was different for every backend, and
> some backends required usecs, while some other required frames, samples
> or bytes). Also tried to reduce the number of abbreviations used by the
> config keys.
>
> Some of the more important changes:
> * use `in` and `out` instead of `ADC` and `DAC`, as the former is more
> user friendly imho
> * moved buffer settings into the global setting area (so it's the same
> for all backends that support it. Backends that can't change buffer
> size will simply ignore them). Also using usecs, as it's probably more
> user friendly than samples or bytes.
> * try-poll is now an alsa backend specific option (as all other backends
> currently ignore it)
>
> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
> ---
> Makefile.objs | 6 +-
> qapi/audio.json | 253 ++++++++++++++++++++++++++++++++++++++++++
> qapi/qapi-schema.json | 1 +
> 3 files changed, 257 insertions(+), 3 deletions(-)
> create mode 100644 qapi/audio.json
>
> diff --git a/Makefile.objs b/Makefile.objs
> index bc5b8a8442..3f833a70c0 100644
> --- a/Makefile.objs
> +++ b/Makefile.objs
> @@ -1,6 +1,6 @@
> -QAPI_MODULES = block-core block char common crypto introspect job migration
> -QAPI_MODULES += misc net rdma rocker run-state sockets tpm trace transaction
> -QAPI_MODULES += ui
> +QAPI_MODULES = audio block-core block char common crypto introspect job
> +QAPI_MODULES += migration misc net rdma rocker run-state sockets tpm trace
> +QAPI_MODULES += transaction ui
>
> #######################################################################
> # Common libraries for tools and emulators
> diff --git a/qapi/audio.json b/qapi/audio.json
> new file mode 100644
> index 0000000000..56d8ce439f
> --- /dev/null
> +++ b/qapi/audio.json
> @@ -0,0 +1,253 @@
> +# -*- mode: python -*-
> +#
> +# Copyright (C) 2015 Zoltán Kővágó <DirtY.iCE.hu@gmail.com>
Do you want to claim 2015-2019 now? But ultimately what you put is your
call, so don't treat my suggestion as a legal mandate.
> +#
> +# This work is licensed under the terms of the GNU GPL, version 2 or later.
> +# See the COPYING file in the top-level directory.
> +
> +##
> +# @AudiodevNoOptions:
> +#
> +# The none, coreaudio, sdl and spice audio backend have no options.
> +#
> +# Since: 3.2
4.0, now.
> +##
> +{ 'struct': 'AudiodevNoOptions',
> + 'data': { } }
Also, we now have support for empty branches in a flat union, so do you
really need this type, or...
> +
> +##
> +# @AudiodevAlsaPerDirectionOptions:
> +#
> +# Options of the alsa backend that are used for both playback and recording.
> +#
> +# @dev: #optional the name of the alsa device to use (default 'default')
No need to use the '#optional' tag any more; the doc generator now takes
care of that.
> +#
> +# @try-poll: #optional attempt to use poll mode, falling back to non polling
> +# access on failure (default on)
> +#
> +# Since: 3.2
More 4.0 (I'll quit pointing it out)
> +##
> +{ 'union': 'Audiodev',
> + 'base': {
> + 'id': 'str',
> + 'driver': 'AudiodevDriver',
> + 'in': 'AudiodevPerDirectionOptions',
> + 'out': 'AudiodevPerDirectionOptions',
> + '*timer-period': 'int' },
> + 'discriminator': 'driver',
> + 'data': {
> + 'none': 'AudiodevNoOptions',
...you could just omit the lines that don't add anything.
--
Eric Blake, Principal Software Engineer
Red Hat, Inc. +1-919-301-3226
Virtualization: qemu.org | libvirt.org
[-- Attachment #2: OpenPGP digital signature --]
[-- Type: application/pgp-signature, Size: 488 bytes --]
^ permalink raw reply [flat|nested] 91+ messages in thread
* [Qemu-devel] [PATCH v2 04/52] audio: use qapi AudioFormat instead of audfmt_e
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (2 preceding siblings ...)
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 03/52] qapi: qapi for audio backends Kővágó, Zoltán
@ 2018-12-23 20:51 ` Kővágó, Zoltán
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 05/52] audio: -audiodev command line option: documentation Kővágó, Zoltán
` (46 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:51 UTC (permalink / raw)
To: qemu-devel
Cc: Gerd Hoffmann, Peter Maydell, Michael Walle, Alistair Francis,
Edgar E. Iglesias, open list:ARM
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 | 12 +----
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/display/xlnx_dp.c | 2 +-
hw/input/tsc210x.c | 2 +-
hw/usb/dev-audio.c | 2 +-
ui/vnc.c | 26 +++++------
26 files changed, 195 insertions(+), 188 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 635be73bf4..5bd034267f 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -87,7 +87,7 @@ struct alsa_params_req {
struct alsa_params_obt {
int freq;
- audfmt_e fmt;
+ AudioFormat fmt;
int endianness;
int nchannels;
snd_pcm_uframes_t samples;
@@ -294,16 +294,16 @@ static int alsa_write (SWVoiceOut *sw, void *buf, int len)
return audio_pcm_sw_write (sw, buf, len);
}
-static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
+static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness)
{
switch (fmt) {
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
return SND_PCM_FORMAT_S8;
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
return SND_PCM_FORMAT_U8;
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
if (endianness) {
return SND_PCM_FORMAT_S16_BE;
}
@@ -311,7 +311,7 @@ static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
return SND_PCM_FORMAT_S16_LE;
}
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
if (endianness) {
return SND_PCM_FORMAT_U16_BE;
}
@@ -319,7 +319,7 @@ static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
return SND_PCM_FORMAT_U16_LE;
}
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_S32:
if (endianness) {
return SND_PCM_FORMAT_S32_BE;
}
@@ -327,7 +327,7 @@ static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
return SND_PCM_FORMAT_S32_LE;
}
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_U32:
if (endianness) {
return SND_PCM_FORMAT_U32_BE;
}
@@ -344,58 +344,58 @@ static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
}
}
-static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt,
+static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt,
int *endianness)
{
switch (alsafmt) {
case SND_PCM_FORMAT_S8:
*endianness = 0;
- *fmt = AUD_FMT_S8;
+ *fmt = AUDIO_FORMAT_S8;
break;
case SND_PCM_FORMAT_U8:
*endianness = 0;
- *fmt = AUD_FMT_U8;
+ *fmt = AUDIO_FORMAT_U8;
break;
case SND_PCM_FORMAT_S16_LE:
*endianness = 0;
- *fmt = AUD_FMT_S16;
+ *fmt = AUDIO_FORMAT_S16;
break;
case SND_PCM_FORMAT_U16_LE:
*endianness = 0;
- *fmt = AUD_FMT_U16;
+ *fmt = AUDIO_FORMAT_U16;
break;
case SND_PCM_FORMAT_S16_BE:
*endianness = 1;
- *fmt = AUD_FMT_S16;
+ *fmt = AUDIO_FORMAT_S16;
break;
case SND_PCM_FORMAT_U16_BE:
*endianness = 1;
- *fmt = AUD_FMT_U16;
+ *fmt = AUDIO_FORMAT_U16;
break;
case SND_PCM_FORMAT_S32_LE:
*endianness = 0;
- *fmt = AUD_FMT_S32;
+ *fmt = AUDIO_FORMAT_S32;
break;
case SND_PCM_FORMAT_U32_LE:
*endianness = 0;
- *fmt = AUD_FMT_U32;
+ *fmt = AUDIO_FORMAT_U32;
break;
case SND_PCM_FORMAT_S32_BE:
*endianness = 1;
- *fmt = AUD_FMT_S32;
+ *fmt = AUDIO_FORMAT_S32;
break;
case SND_PCM_FORMAT_U32_BE:
*endianness = 1;
- *fmt = AUD_FMT_U32;
+ *fmt = AUDIO_FORMAT_U32;
break;
default:
@@ -638,19 +638,22 @@ static int alsa_open (int in, struct alsa_params_req *req,
bytes_per_sec = freq << (nchannels == 2);
switch (obt->fmt) {
- case AUD_FMT_S8:
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_S8:
+ case AUDIO_FORMAT_U8:
break;
- case AUD_FMT_S16:
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_S16:
+ case AUDIO_FORMAT_U16:
bytes_per_sec <<= 1;
break;
- case AUD_FMT_S32:
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_S32:
+ case AUDIO_FORMAT_U32:
bytes_per_sec <<= 2;
break;
+
+ default:
+ abort();
}
threshold = (conf->threshold * bytes_per_sec) / 1000;
diff --git a/audio/audio.c b/audio/audio.c
index 1ace47f510..96cbd57c37 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -113,7 +113,7 @@ static struct {
.settings = {
.freq = 44100,
.nchannels = 2,
- .fmt = AUD_FMT_S16,
+ .fmt = AUDIO_FORMAT_S16,
.endianness = AUDIO_HOST_ENDIANNESS,
}
},
@@ -125,7 +125,7 @@ static struct {
.settings = {
.freq = 44100,
.nchannels = 2,
- .fmt = AUD_FMT_S16,
+ .fmt = AUDIO_FORMAT_S16,
.endianness = AUDIO_HOST_ENDIANNESS,
}
},
@@ -257,58 +257,61 @@ static char *audio_alloc_prefix (const char *s)
return r;
}
-static const char *audio_audfmt_to_string (audfmt_e fmt)
+static const char *audio_audfmt_to_string (AudioFormat fmt)
{
switch (fmt) {
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
return "U8";
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
return "U16";
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
return "S8";
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
return "S16";
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_U32:
return "U32";
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_S32:
return "S32";
+
+ default:
+ abort();
}
dolog ("Bogus audfmt %d returning S16\n", fmt);
return "S16";
}
-static audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval,
+static AudioFormat audio_string_to_audfmt (const char *s, AudioFormat defval,
int *defaultp)
{
if (!strcasecmp (s, "u8")) {
*defaultp = 0;
- return AUD_FMT_U8;
+ return AUDIO_FORMAT_U8;
}
else if (!strcasecmp (s, "u16")) {
*defaultp = 0;
- return AUD_FMT_U16;
+ return AUDIO_FORMAT_U16;
}
else if (!strcasecmp (s, "u32")) {
*defaultp = 0;
- return AUD_FMT_U32;
+ return AUDIO_FORMAT_U32;
}
else if (!strcasecmp (s, "s8")) {
*defaultp = 0;
- return AUD_FMT_S8;
+ return AUDIO_FORMAT_S8;
}
else if (!strcasecmp (s, "s16")) {
*defaultp = 0;
- return AUD_FMT_S16;
+ return AUDIO_FORMAT_S16;
}
else if (!strcasecmp (s, "s32")) {
*defaultp = 0;
- return AUD_FMT_S32;
+ return AUDIO_FORMAT_S32;
}
else {
dolog ("Bogus audio format `%s' using %s\n",
@@ -318,8 +321,8 @@ static audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval,
}
}
-static audfmt_e audio_get_conf_fmt (const char *envname,
- audfmt_e defval,
+static AudioFormat audio_get_conf_fmt (const char *envname,
+ AudioFormat defval,
int *defaultp)
{
const char *var = getenv (envname);
@@ -421,7 +424,7 @@ static void audio_print_options (const char *prefix,
case AUD_OPT_FMT:
{
- audfmt_e *fmtp = opt->valp;
+ AudioFormat *fmtp = opt->valp;
printf (
"format, %s = %s, (one of: U8 S8 U16 S16 U32 S32)\n",
state,
@@ -508,7 +511,7 @@ 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;
@@ -539,22 +542,22 @@ 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:
@@ -585,12 +588,12 @@ 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;
@@ -606,25 +609,28 @@ static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *a
int bits = 8, sign = 0;
switch (as->fmt) {
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
sign = 1;
/* fall through */
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
break;
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
sign = 1;
/* fall through */
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
bits = 16;
break;
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_S32:
sign = 1;
/* fall through */
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_U32:
bits = 32;
break;
+
+ default:
+ abort();
}
return info->freq == as->freq
&& info->nchannels == as->nchannels
@@ -638,24 +644,27 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
int bits = 8, sign = 0, shift = 0;
switch (as->fmt) {
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
sign = 1;
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
break;
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
sign = 1;
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
bits = 16;
shift = 1;
break;
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_S32:
sign = 1;
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_U32:
bits = 32;
shift = 2;
break;
+
+ default:
+ abort();
}
info->freq = as->freq;
diff --git a/audio/audio.h b/audio/audio.h
index f4339a185e..02f29a3b3e 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -26,18 +26,10 @@
#define QEMU_AUDIO_H
#include "qemu/queue.h"
+#include "qapi/qapi-types-audio.h"
typedef void (*audio_callback_fn) (void *opaque, int avail);
-typedef enum {
- AUD_FMT_U8,
- AUD_FMT_S8,
- AUD_FMT_U16,
- AUD_FMT_S16,
- AUD_FMT_U32,
- AUD_FMT_S32
-} audfmt_e;
-
#ifdef HOST_WORDS_BIGENDIAN
#define AUDIO_HOST_ENDIANNESS 1
#else
@@ -47,7 +39,7 @@ typedef enum {
struct audsettings {
int freq;
int nchannels;
- audfmt_e fmt;
+ AudioFormat fmt;
int endianness;
};
diff --git a/audio/audio_win_int.c b/audio/audio_win_int.c
index 6900008d0c..b938fd667b 100644
--- a/audio/audio_win_int.c
+++ b/audio/audio_win_int.c
@@ -24,20 +24,20 @@ int waveformat_from_audio_settings (WAVEFORMATEX *wfx,
wfx->cbSize = 0;
switch (as->fmt) {
- case AUD_FMT_S8:
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_S8:
+ case AUDIO_FORMAT_U8:
wfx->wBitsPerSample = 8;
break;
- case AUD_FMT_S16:
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_S16:
+ case AUDIO_FORMAT_U16:
wfx->wBitsPerSample = 16;
wfx->nAvgBytesPerSec <<= 1;
wfx->nBlockAlign <<= 1;
break;
- case AUD_FMT_S32:
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_S32:
+ case AUDIO_FORMAT_U32:
wfx->wBitsPerSample = 32;
wfx->nAvgBytesPerSec <<= 2;
wfx->nBlockAlign <<= 2;
@@ -85,15 +85,15 @@ int waveformat_to_audio_settings (WAVEFORMATEX *wfx,
switch (wfx->wBitsPerSample) {
case 8:
- as->fmt = AUD_FMT_U8;
+ as->fmt = AUDIO_FORMAT_U8;
break;
case 16:
- as->fmt = AUD_FMT_S16;
+ as->fmt = AUDIO_FORMAT_S16;
break;
case 32:
- as->fmt = AUD_FMT_S32;
+ as->fmt = AUDIO_FORMAT_S32;
break;
default:
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 6c69622b4c..355e8fbda5 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -70,7 +70,7 @@ typedef struct OSSVoiceIn {
struct oss_params {
int freq;
- audfmt_e fmt;
+ AudioFormat fmt;
int nchannels;
int nfrags;
int fragsize;
@@ -148,16 +148,16 @@ static int oss_write (SWVoiceOut *sw, void *buf, int len)
return audio_pcm_sw_write (sw, buf, len);
}
-static int aud_to_ossfmt (audfmt_e fmt, int endianness)
+static int aud_to_ossfmt (AudioFormat fmt, int endianness)
{
switch (fmt) {
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
return AFMT_S8;
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
return AFMT_U8;
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
if (endianness) {
return AFMT_S16_BE;
}
@@ -165,7 +165,7 @@ static int aud_to_ossfmt (audfmt_e fmt, int endianness)
return AFMT_S16_LE;
}
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
if (endianness) {
return AFMT_U16_BE;
}
@@ -182,37 +182,37 @@ static int aud_to_ossfmt (audfmt_e fmt, int endianness)
}
}
-static int oss_to_audfmt (int ossfmt, audfmt_e *fmt, int *endianness)
+static int oss_to_audfmt (int ossfmt, AudioFormat *fmt, int *endianness)
{
switch (ossfmt) {
case AFMT_S8:
*endianness = 0;
- *fmt = AUD_FMT_S8;
+ *fmt = AUDIO_FORMAT_S8;
break;
case AFMT_U8:
*endianness = 0;
- *fmt = AUD_FMT_U8;
+ *fmt = AUDIO_FORMAT_U8;
break;
case AFMT_S16_LE:
*endianness = 0;
- *fmt = AUD_FMT_S16;
+ *fmt = AUDIO_FORMAT_S16;
break;
case AFMT_U16_LE:
*endianness = 0;
- *fmt = AUD_FMT_U16;
+ *fmt = AUDIO_FORMAT_U16;
break;
case AFMT_S16_BE:
*endianness = 1;
- *fmt = AUD_FMT_S16;
+ *fmt = AUDIO_FORMAT_S16;
break;
case AFMT_U16_BE:
*endianness = 1;
- *fmt = AUD_FMT_U16;
+ *fmt = AUDIO_FORMAT_U16;
break;
default:
@@ -500,7 +500,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
int endianness;
int err;
int fd;
- audfmt_e effective_fmt;
+ AudioFormat effective_fmt;
struct audsettings obt_as;
OSSConf *conf = drv_opaque;
@@ -667,7 +667,7 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
int endianness;
int err;
int fd;
- audfmt_e effective_fmt;
+ AudioFormat effective_fmt;
struct audsettings obt_as;
OSSConf *conf = drv_opaque;
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 4c100bc318..f1f9a741ac 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -385,21 +385,21 @@ static int qpa_read (SWVoiceIn *sw, void *buf, int len)
return audio_pcm_sw_read (sw, buf, len);
}
-static pa_sample_format_t audfmt_to_pa (audfmt_e afmt, int endianness)
+static pa_sample_format_t audfmt_to_pa (AudioFormat afmt, int endianness)
{
int format;
switch (afmt) {
- case AUD_FMT_S8:
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_S8:
+ case AUDIO_FORMAT_U8:
format = PA_SAMPLE_U8;
break;
- case AUD_FMT_S16:
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_S16:
+ case AUDIO_FORMAT_U16:
format = endianness ? PA_SAMPLE_S16BE : PA_SAMPLE_S16LE;
break;
- case AUD_FMT_S32:
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_S32:
+ case AUDIO_FORMAT_U32:
format = endianness ? PA_SAMPLE_S32BE : PA_SAMPLE_S32LE;
break;
default:
@@ -410,26 +410,26 @@ static pa_sample_format_t audfmt_to_pa (audfmt_e afmt, int endianness)
return format;
}
-static audfmt_e pa_to_audfmt (pa_sample_format_t fmt, int *endianness)
+static AudioFormat pa_to_audfmt (pa_sample_format_t fmt, int *endianness)
{
switch (fmt) {
case PA_SAMPLE_U8:
- return AUD_FMT_U8;
+ return AUDIO_FORMAT_U8;
case PA_SAMPLE_S16BE:
*endianness = 1;
- return AUD_FMT_S16;
+ return AUDIO_FORMAT_S16;
case PA_SAMPLE_S16LE:
*endianness = 0;
- return AUD_FMT_S16;
+ return AUDIO_FORMAT_S16;
case PA_SAMPLE_S32BE:
*endianness = 1;
- return AUD_FMT_S32;
+ return AUDIO_FORMAT_S32;
case PA_SAMPLE_S32LE:
*endianness = 0;
- return AUD_FMT_S32;
+ return AUDIO_FORMAT_S32;
default:
dolog ("Internal logic error: Bad pa_sample_format %d\n", fmt);
- return AUD_FMT_U8;
+ return AUDIO_FORMAT_U8;
}
}
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index 9db5ac92bc..aa42ea26bf 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -137,19 +137,19 @@ 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:
@@ -161,37 +161,37 @@ 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:
@@ -386,7 +386,7 @@ 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 6ad0eafbc6..3aeb0cb357 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -130,7 +130,7 @@ static int line_out_init(HWVoiceOut *hw, struct audsettings *as,
settings.freq = SPICE_INTERFACE_PLAYBACK_FREQ;
#endif
settings.nchannels = SPICE_INTERFACE_PLAYBACK_CHAN;
- settings.fmt = AUD_FMT_S16;
+ settings.fmt = AUDIO_FORMAT_S16;
settings.endianness = AUDIO_HOST_ENDIANNESS;
audio_pcm_init_info (&hw->info, &settings);
@@ -258,7 +258,7 @@ static int line_in_init(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
settings.freq = SPICE_INTERFACE_RECORD_FREQ;
#endif
settings.nchannels = SPICE_INTERFACE_RECORD_CHAN;
- settings.fmt = AUD_FMT_S16;
+ settings.fmt = AUDIO_FORMAT_S16;
settings.endianness = AUDIO_HOST_ENDIANNESS;
audio_pcm_init_info (&hw->info, &settings);
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 40adfa30c3..35a614785e 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -117,20 +117,23 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
stereo = wav_as.nchannels == 2;
switch (wav_as.fmt) {
- case AUD_FMT_S8:
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_S8:
+ case AUDIO_FORMAT_U8:
bits16 = 0;
break;
- case AUD_FMT_S16:
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_S16:
+ case AUDIO_FORMAT_U16:
bits16 = 1;
break;
- case AUD_FMT_S32:
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_S32:
+ case AUDIO_FORMAT_U32:
dolog ("WAVE files can not handle 32bit formats\n");
return -1;
+
+ default:
+ abort();
}
hdr[34] = bits16 ? 0x10 : 0x08;
@@ -225,7 +228,7 @@ static int wav_ctl_out (HWVoiceOut *hw, int cmd, ...)
static WAVConf glob_conf = {
.settings.freq = 44100,
.settings.nchannels = 2,
- .settings.fmt = AUD_FMT_S16,
+ .settings.fmt = AUDIO_FORMAT_S16,
.wav_path = "qemu.wav"
};
diff --git a/audio/wavcapture.c b/audio/wavcapture.c
index cf31ed652c..e943565210 100644
--- a/audio/wavcapture.c
+++ b/audio/wavcapture.c
@@ -139,7 +139,7 @@ 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 3c7d1364a9..96e84f6e7e 100644
--- a/hw/arm/omap2.c
+++ b/hw/arm/omap2.c
@@ -273,7 +273,7 @@ static void omap_eac_format_update(struct omap_eac_s *s)
* does I2S specify it? */
/* All register writes are 16 bits so we we store 16-bit samples
* in the buffers regardless of AGCFR[B8_16] value. */
- fmt.fmt = AUD_FMT_U16;
+ fmt.fmt = AUDIO_FORMAT_U16;
s->codec.in_voice = AUD_open_in(&s->codec.card, s->codec.in_voice,
"eac.codec.in", s, omap_eac_in_cb, &fmt);
diff --git a/hw/audio/ac97.c b/hw/audio/ac97.c
index d799533aa9..2265622d44 100644
--- a/hw/audio/ac97.c
+++ b/hw/audio/ac97.c
@@ -365,7 +365,7 @@ static void open_voice (AC97LinkState *s, int index, int freq)
as.freq = freq;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = 0;
if (freq > 0) {
diff --git a/hw/audio/adlib.c b/hw/audio/adlib.c
index 97b876c7e0..0957780a3d 100644
--- a/hw/audio/adlib.c
+++ b/hw/audio/adlib.c
@@ -269,7 +269,7 @@ static void adlib_realizefn (DeviceState *dev, Error **errp)
as.freq = s->freq;
as.nchannels = SHIFT;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = AUDIO_HOST_ENDIANNESS;
AUD_register_card ("adlib", &s->card);
diff --git a/hw/audio/cs4231a.c b/hw/audio/cs4231a.c
index 9089dcb47e..62da75eefe 100644
--- a/hw/audio/cs4231a.c
+++ b/hw/audio/cs4231a.c
@@ -288,7 +288,7 @@ static void cs_reset_voices (CSState *s, uint32_t val)
switch ((val >> 5) & ((s->dregs[MODE_And_ID] & MODE2) ? 7 : 3)) {
case 0:
- as.fmt = AUD_FMT_U8;
+ as.fmt = AUDIO_FORMAT_U8;
s->shift = as.nchannels == 2;
break;
@@ -298,7 +298,7 @@ static void cs_reset_voices (CSState *s, uint32_t val)
case 3:
s->tab = ALawDecompressTable;
x_law:
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = AUDIO_HOST_ENDIANNESS;
s->shift = as.nchannels == 2;
break;
@@ -307,7 +307,7 @@ static void cs_reset_voices (CSState *s, uint32_t val)
as.endianness = 1;
/* fall through */
case 2:
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
s->shift = as.nchannels;
break;
diff --git a/hw/audio/es1370.c b/hw/audio/es1370.c
index 97789a0771..a5314d66fd 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 8e0b27e0f2..b3e2a7fdd5 100644
--- a/hw/audio/gus.c
+++ b/hw/audio/gus.c
@@ -251,7 +251,7 @@ static void gus_realizefn (DeviceState *dev, Error **errp)
as.freq = s->freq;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = GUS_ENDIANNESS;
s->voice = AUD_open_out (
diff --git a/hw/audio/hda-codec.c b/hw/audio/hda-codec.c
index 617a1c1016..c25bfa38b1 100644
--- a/hw/audio/hda-codec.c
+++ b/hw/audio/hda-codec.c
@@ -99,9 +99,9 @@ static void hda_codec_parse_fmt(uint32_t format, struct audsettings *as)
}
switch (format & AC_FMT_BITS_MASK) {
- case AC_FMT_BITS_8: as->fmt = AUD_FMT_S8; break;
- case AC_FMT_BITS_16: as->fmt = AUD_FMT_S16; break;
- case AC_FMT_BITS_32: as->fmt = AUD_FMT_S32; break;
+ case AC_FMT_BITS_8: as->fmt = AUDIO_FORMAT_S8; break;
+ case AC_FMT_BITS_16: as->fmt = AUDIO_FORMAT_S16; break;
+ case AC_FMT_BITS_32: as->fmt = AUDIO_FORMAT_S32; break;
}
as->nchannels = ((format & AC_FMT_CHAN_MASK) >> AC_FMT_CHAN_SHIFT) + 1;
@@ -134,12 +134,12 @@ static void hda_codec_parse_fmt(uint32_t format, struct audsettings *as)
/* -------------------------------------------------------------------------- */
static const char *fmt2name[] = {
- [ AUD_FMT_U8 ] = "PCM-U8",
- [ AUD_FMT_S8 ] = "PCM-S8",
- [ AUD_FMT_U16 ] = "PCM-U16",
- [ AUD_FMT_S16 ] = "PCM-S16",
- [ AUD_FMT_U32 ] = "PCM-U32",
- [ AUD_FMT_S32 ] = "PCM-S32",
+ [ AUDIO_FORMAT_U8 ] = "PCM-U8",
+ [ AUDIO_FORMAT_S8 ] = "PCM-S8",
+ [ AUDIO_FORMAT_U16 ] = "PCM-U16",
+ [ AUDIO_FORMAT_S16 ] = "PCM-S16",
+ [ AUDIO_FORMAT_U32 ] = "PCM-U32",
+ [ AUDIO_FORMAT_S32 ] = "PCM-S32",
};
typedef struct HDAAudioState HDAAudioState;
diff --git a/hw/audio/lm4549.c b/hw/audio/lm4549.c
index a46f2301af..af8b22b541 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 bc8db71ae0..90cce1e6ed 100644
--- a/hw/audio/milkymist-ac97.c
+++ b/hw/audio/milkymist-ac97.c
@@ -308,7 +308,7 @@ static void milkymist_ac97_realize(DeviceState *dev, Error **errp)
as.freq = 48000;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = 1;
s->voice_in = AUD_open_in(&s->card, s->voice_in,
diff --git a/hw/audio/pcspk.c b/hw/audio/pcspk.c
index 908696d483..4a7386a689 100644
--- a/hw/audio/pcspk.c
+++ b/hw/audio/pcspk.c
@@ -114,7 +114,7 @@ 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 c5b9bf79e8..65ea0cd938 100644
--- a/hw/audio/sb16.c
+++ b/hw/audio/sb16.c
@@ -66,7 +66,7 @@ typedef struct SB16State {
int fmt_stereo;
int fmt_signed;
int fmt_bits;
- audfmt_e fmt;
+ AudioFormat fmt;
int dma_auto;
int block_size;
int fifo;
@@ -224,7 +224,7 @@ static void continue_dma8 (SB16State *s)
static void dma_cmd8 (SB16State *s, int mask, int dma_len)
{
- s->fmt = AUD_FMT_U8;
+ s->fmt = AUDIO_FORMAT_U8;
s->use_hdma = 0;
s->fmt_bits = 8;
s->fmt_signed = 0;
@@ -319,18 +319,18 @@ static void dma_cmd (SB16State *s, uint8_t cmd, uint8_t d0, int dma_len)
if (16 == s->fmt_bits) {
if (s->fmt_signed) {
- s->fmt = AUD_FMT_S16;
+ s->fmt = AUDIO_FORMAT_S16;
}
else {
- s->fmt = AUD_FMT_U16;
+ s->fmt = AUDIO_FORMAT_U16;
}
}
else {
if (s->fmt_signed) {
- s->fmt = AUD_FMT_S8;
+ s->fmt = AUDIO_FORMAT_S8;
}
else {
- s->fmt = AUD_FMT_U8;
+ s->fmt = AUDIO_FORMAT_U8;
}
}
@@ -852,7 +852,7 @@ static void legacy_reset (SB16State *s)
as.freq = s->freq;
as.nchannels = 1;
- as.fmt = AUD_FMT_U8;
+ as.fmt = AUDIO_FORMAT_U8;
as.endianness = 0;
s->voice = AUD_open_out (
diff --git a/hw/audio/wm8750.c b/hw/audio/wm8750.c
index f4aa838f62..5d614c0d8b 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/display/xlnx_dp.c b/hw/display/xlnx_dp.c
index cc0f9bc9cc..11b09bd18c 100644
--- a/hw/display/xlnx_dp.c
+++ b/hw/display/xlnx_dp.c
@@ -1260,7 +1260,7 @@ static void xlnx_dp_realize(DeviceState *dev, Error **errp)
as.freq = 44100;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = 0;
AUD_register_card("xlnx_dp.audio", &s->aud_card);
diff --git a/hw/input/tsc210x.c b/hw/input/tsc210x.c
index 1cad57f644..7b826e0e22 100644
--- a/hw/input/tsc210x.c
+++ b/hw/input/tsc210x.c
@@ -318,7 +318,7 @@ static void tsc2102_audio_output_update(TSC210xState *s)
fmt.endianness = 0;
fmt.nchannels = 2;
fmt.freq = s->codec.tx_rate;
- fmt.fmt = AUD_FMT_S16;
+ fmt.fmt = AUDIO_FORMAT_S16;
s->dac_voice[0] = AUD_open_out(&s->card, s->dac_voice[0],
"tsc2102.sink", s, (void *) tsc210x_audio_out_cb, &fmt);
diff --git a/hw/usb/dev-audio.c b/hw/usb/dev-audio.c
index ee43e4914d..b1b9736a19 100644
--- a/hw/usb/dev-audio.c
+++ b/hw/usb/dev-audio.c
@@ -647,7 +647,7 @@ 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 0c1b477425..0e9e9fb63a 100644
--- a/ui/vnc.c
+++ b/ui/vnc.c
@@ -994,16 +994,16 @@ static void vnc_update_throttle_offset(VncState *vs)
int bps;
switch (vs->as.fmt) {
default:
- case AUD_FMT_U8:
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_U8:
+ case AUDIO_FORMAT_S8:
bps = 1;
break;
- case AUD_FMT_U16:
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_U16:
+ case AUDIO_FORMAT_S16:
bps = 2;
break;
- case AUD_FMT_U32:
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_U32:
+ case AUDIO_FORMAT_S32:
bps = 4;
break;
}
@@ -2409,12 +2409,12 @@ 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);
@@ -3145,7 +3145,7 @@ static void vnc_connect(VncDisplay *vd, QIOChannelSocket *sioc,
vs->as.freq = 44100;
vs->as.nchannels = 2;
- vs->as.fmt = AUD_FMT_S16;
+ vs->as.fmt = AUDIO_FORMAT_S16;
vs->as.endianness = 0;
qemu_mutex_init(&vs->output_mutex);
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 05/52] audio: -audiodev command line option: documentation
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (3 preceding siblings ...)
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 04/52] audio: use qapi AudioFormat instead of audfmt_e Kővágó, Zoltán
@ 2018-12-23 20:51 ` Kővágó, Zoltán
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 06/52] audio: -audiodev command line option basic implementation Kővágó, Zoltán
` (45 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:51 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
This patch adds documentation of an -audiodev command line option, that
deprecates the old QEMU_* environment variables for audio backend
configuration. It's syntax is similar to existing options (-netdev,
-device, etc):
-audiodev driver_name,property=value,...
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>
---
qemu-options.hx | 222 +++++++++++++++++++++++++++++++++++++++++++++++-
1 file changed, 219 insertions(+), 3 deletions(-)
diff --git a/qemu-options.hx b/qemu-options.hx
index df42116ecc..3ede6a591d 100644
--- a/qemu-options.hx
+++ b/qemu-options.hx
@@ -416,14 +416,230 @@ 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 currently specified audio settings\n",
QEMU_ARCH_ALL)
STEXI
@item -audio-help
@findex -audio-help
-Will show the audio subsystem help: list of drivers, tunable
-parameters.
+Will show the -audiodev equivalent of the currently specified
+(deprecated) environment variables.
+ETEXI
+
+DEF("audiodev", HAS_ARG, QEMU_OPTION_audiodev,
+ "-audiodev [driver=]driver,id=id[,prop[=value][,...]]\n"
+ " specifies the audio backend to use\n"
+ " id= identifier of the backend\n"
+ " timer-period= timer period in microseconds\n"
+ " in|out.fixed-settings= use fixed settings for host audio\n"
+ " in|out.frequency= frequency to use with fixed settings\n"
+ " in|out.channels= number of channels to use with fixed settings\n"
+ " in|out.format= sample format to use with fixed settings\n"
+ " valid values: s8, s16, s32, u8, u16, u32\n"
+ " in|out.voices= number of voices to use\n"
+ " in|out.buffer-len= size of buffer in microseconds\n"
+ " in|out.buffer-count= number of buffers\n"
+ "-audiodev none,id=id,[,prop[=value][,...]]\n"
+ " dummy driver that discards all output\n"
+#ifdef CONFIG_ALSA
+ "-audiodev alsa,id=id[,prop[=value][,...]]\n"
+ " alsa-in|alsa-out.dev= name of the audio device to use\n"
+ " alsa-in|alsa-out.try-poll= attempt to use poll mode\n"
+ " threshold= threshold (in microseconds) when playback starts\n"
+#endif
+#ifdef CONFIG_COREAUDIO
+ "-audiodev coreaudio,id=id[,prop[=value][,...]]\n"
+#endif
+#ifdef CONFIG_DSOUND
+ "-audiodev dsound,id=id[,prop[=value][,...]]\n"
+ " latency= add extra latency to playback in microseconds\n"
+#endif
+#ifdef CONFIG_OSS
+ "-audiodev oss,id=id[,prop[=value][,...]]\n"
+ " oss-in|oss-out.dev= path of the audio device to use\n"
+ " oss-in|oss-out.try-poll= attempt to use poll mode\n"
+ " try-mmap= try using memory mapped access\n"
+ " exclusive= open device in exclusive mode\n"
+ " dsp-policy= set timing policy (0..10), -1 to use fragment mode\n"
+#endif
+#ifdef CONFIG_PA
+ "-audiodev pa,id=id[,prop[=value][,...]]\n"
+ " server= PulseAudio server address\n"
+ " sink|source.name= sink/source device name\n"
+#endif
+#ifdef CONFIG_SDL
+ "-audiodev sdl,id=id[,prop[=value][,...]]\n"
+#endif
+#ifdef CONFIG_SPICE
+ "-audiodev spice,id=id[,prop[=value][,...]]\n"
+#endif
+ "-audiodev wav,id=id[,prop[=value][,...]]\n"
+ " path= path of wav file to record\n",
+ QEMU_ARCH_ALL)
+STEXI
+@item -audiodev [driver=]@var{driver},id=@var{id}[,@var{prop}[=@var{value}][,...]]
+@findex -audiodev
+Adds a new audio backend @var{driver} identified by @var{id}. There are
+global and driver specific properties. Some values can be set
+differently for input and output, they're marked with @code{in|out.}.
+You can set the input's property with @code{in.@var{prop}} and the
+output's property with @code{out.@var{prop}}. For example:
+@example
+-audiodev alsa,id=example,in.frequency=44110,out.frequency=8000
+-audiodev alsa,id=example,out.channels=1 # leaves in.channels unspecified
+@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 microseconds.
+Default is 10000 (10 ms).
+
+@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. In this case you must not specify
+@var{frequency}, @var{channels} or @var{format}. Default is on.
+
+@item in|out.frequency=@var{frequency}
+Specify the @var{frequency} to use when using @var{fixed-settings}.
+Default is 44100Hz.
+
+@item in|out.channels=@var{channels}
+Specify the number of @var{channels} to use when using
+@var{fixed-settings}. Default is 2 (stereo).
+
+@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}. Default is @code{s16}.
+
+@item in|out.voices=@var{voices}
+Specify the number of @var{voices} to use. Default is 1.
+
+@item in|out.buffer=@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,id=@var{id}[,@var{prop}[=@var{value}][,...]]
+Creates a dummy backend that discards all outputs. This backend has no
+backend specific properties.
+
+@item -audiodev alsa,id=@var{id}[,@var{prop}[=@var{value}][,...]]
+Creates backend using the ALSA. This backend is only available on
+Linux.
+
+ALSA specific options are:
+
+@table @option
+@item alsa-in|alsa-out.dev=@var{device}
+Specify the ALSA @var{device} to use for input and/or output. Default
+is @code{default}.
+
+@item alsa-in|alsa-out.try-poll=on|off
+Attempt to use poll mode with the device. Default is on.
+
+@item threshold=@var{threshold}
+Threshold (in microseconds) when playback starts. Default is 0.
+
+@end table
+
+@item -audiodev coreaudio,id=@var{id}[,@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,id=@var{id}[,@var{prop}[=@var{value}][,...]]
+Creates a backend using Microsoft's DirectSound. This backend is only
+available on Windows and only supports playback.
+
+Backend specific options are:
+
+@table @option
+
+@item latency=@var{usecs}
+Add extra @var{usecs} microseconds latency to playback. Default is
+10000 (10 ms).
+
+@end table
+
+@item -audiodev oss,id=@var{id}[,@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 oss-in|oss-out.dev=@var{device}
+Specify the file name of the OSS @var{device} to use. Default is
+@code{/dev/dsp}.
+
+@item oss-in|oss-out.try-poll=on|of
+Attempt to use poll mode with the device. Default is on.
+
+@item try-mmap=on|off
+Try using memory mapped device access. Default is off.
+
+@item exclusive=on|off
+Open the device in exclusive mode (vmix won't work in this case).
+Default is off.
+
+@item dsp-policy=@var{policy}
+Sets the timing policy (between 0 and 10, where smaller number means
+smaller latency but higher CPU usage). Use -1 to use buffer sizes
+specified by @code{buffer} and @code{buffer-count}. This option is
+ignored if you do not have OSS 4. Default is 5.
+
+@end table
+
+@item -audiodev pa,id=@var{id}[,@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|source.name=@var{sink}
+Use the specified sink/source for playback/recording.
+
+@end table
+
+@item -audiodev sdl,id=@var{id}[,@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,id=@var{id}[,@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,id=@var{id}[,@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. Default is
+@code{qemu.wav}.
+
+@end table
ETEXI
DEF("soundhw", HAS_ARG, QEMU_OPTION_soundhw,
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 06/52] audio: -audiodev command line option basic implementation
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (4 preceding siblings ...)
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 05/52] audio: -audiodev command line option: documentation Kővágó, Zoltán
@ 2018-12-23 20:51 ` Kővágó, Zoltán
2019-01-07 13:13 ` Markus Armbruster
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 07/52] alsaaudio: port to -audiodev config Kővágó, Zoltán
` (44 subsequent siblings)
50 siblings, 1 reply; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:51 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann, Paolo Bonzini
Audio drivers now get an Audiodev * as config paramters, instead of the
global audio_option structs. There is some code in audio/audio_legacy.c
that converts the old environment variables to audiodev options (this
way backends do not have to worry about legacy options). It also
contains a replacement of -audio-help, which prints out the equivalent
-audiodev based config of the currently specified environment variables.
Note that backends are not updated and still rely on environment
variables.
Also note that (due to moving try-poll from global to backend specific
option) currently ALSA and OSS will always try poll mode, regardless of
environment variables or -audiodev options.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/Makefile.objs | 2 +-
audio/alsaaudio.c | 2 +-
audio/audio.c | 589 ++++++++++++++++-------------------------
audio/audio.h | 20 +-
audio/audio_int.h | 6 +-
audio/audio_legacy.c | 211 +++++++++++++++
audio/audio_template.h | 13 +-
audio/coreaudio.c | 2 +-
audio/dsoundaudio.c | 2 +-
audio/noaudio.c | 2 +-
audio/ossaudio.c | 2 +-
audio/paaudio.c | 2 +-
audio/sdlaudio.c | 2 +-
audio/spiceaudio.c | 2 +-
audio/wavaudio.c | 2 +-
vl.c | 11 +-
16 files changed, 491 insertions(+), 379 deletions(-)
create mode 100644 audio/audio_legacy.c
diff --git a/audio/Makefile.objs b/audio/Makefile.objs
index db4fa7f18f..dca87f6347 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-$(CONFIG_SPICE) += spiceaudio.o
common-obj-$(CONFIG_AUDIO_COREAUDIO) += coreaudio.o
common-obj-$(CONFIG_AUDIO_DSOUND) += dsoundaudio.o
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 5bd034267f..8302f3e882 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -1125,7 +1125,7 @@ static ALSAConf glob_conf = {
.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;
diff --git a/audio/audio.c b/audio/audio.c
index 96cbd57c37..e7f25ea84b 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -25,7 +25,12 @@
#include "hw/hw.h"
#include "audio.h"
#include "monitor/monitor.h"
+#include "qapi/opts-visitor.h"
#include "qemu/timer.h"
+#include "qemu/option.h"
+#include "qemu/config-file.h"
+#include "qapi/error.h"
+#include "qapi/qapi-visit-audio.h"
#include "sysemu/sysemu.h"
#include "qemu/cutils.h"
#include "sysemu/replay.h"
@@ -46,11 +51,12 @@
The 1st one is the one used by default, that is the reason
that we generate the list.
*/
-static const char *audio_prio_list[] = {
+const char *audio_prio_list[] = {
"spice",
CONFIG_AUDIO_DRIVERS
"none",
"wav",
+ NULL
};
static QLIST_HEAD(, audio_driver) audio_drivers;
@@ -80,61 +86,6 @@ audio_driver *audio_driver_lookup(const char *name)
return NULL;
}
-static void audio_module_load_all(void)
-{
- int i;
-
- for (i = 0; i < ARRAY_SIZE(audio_prio_list); i++) {
- audio_driver_lookup(audio_prio_list[i]);
- }
-}
-
-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,
-};
-
static AudioState glob_audio_state;
const struct mixeng_volume nominal_volume = {
@@ -151,9 +102,6 @@ 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) {
@@ -161,16 +109,9 @@ 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");
@@ -232,31 +173,6 @@ 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) {
@@ -382,78 +298,6 @@ 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)
{
@@ -1161,11 +1005,11 @@ static void audio_reset_timer (AudioState *s)
{
if (audio_is_timer_needed ()) {
timer_mod_anticipate_ns(s->ts,
- qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + conf.period.ticks);
+ qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + s->period_ticks);
if (!audio_timer_running) {
audio_timer_running = true;
audio_timer_last = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
- trace_audio_timer_start(conf.period.ticks / SCALE_MS);
+ trace_audio_timer_start(s->period_ticks / SCALE_MS);
}
} else {
timer_del(s->ts);
@@ -1179,16 +1023,17 @@ static void audio_reset_timer (AudioState *s)
static void audio_timer (void *opaque)
{
int64_t now, diff;
+ AudioState *s = opaque;
now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
diff = now - audio_timer_last;
- if (diff > conf.period.ticks * 3 / 2) {
+ if (diff > s->period_ticks * 3 / 2) {
trace_audio_timer_delayed(diff / SCALE_MS);
}
audio_timer_last = now;
- audio_run ("timer");
- audio_reset_timer (opaque);
+ audio_run("timer");
+ audio_reset_timer(s);
}
/*
@@ -1248,7 +1093,7 @@ 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, true /* todo */);
audio_reset_timer (s);
}
}
@@ -1293,7 +1138,7 @@ 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, true /* todo */);
audio_reset_timer (s);
}
}
@@ -1614,169 +1459,13 @@ 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)
-{
- 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)
-{
- struct audio_driver *d;
-
- /* make sure we print the help text for modular drivers too */
- audio_module_load_all();
-
- audio_process_options ("AUDIO", audio_options);
- QLIST_FOREACH(d, &audio_drivers, next) {
- 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");
-
- QLIST_FOREACH(d, &audio_drivers, next) {
-
- 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)
+static int audio_driver_init(AudioState *s, struct audio_driver *drv,
+ Audiodev *dev)
{
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);
@@ -1800,11 +1489,11 @@ 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, true /* todo */);
}
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, true /* todo */);
}
audio_reset_timer (s);
}
@@ -1854,6 +1543,11 @@ void audio_cleanup(void)
s->drv->fini (s->drv_opaque);
s->drv = NULL;
}
+
+ if (s->dev) {
+ qapi_free_Audiodev(s->dev);
+ s->dev = NULL;
+ }
}
static const VMStateDescription vmstate_audio = {
@@ -1865,19 +1559,40 @@ static const VMStateDescription vmstate_audio = {
}
};
-static void audio_init (void)
+static Audiodev *parse_option(QemuOpts *opts, Error **errp);
+static int audio_init(Audiodev *dev)
{
size_t i;
int done = 0;
- const char *drvname;
+ const char *drvname = NULL;
VMChangeStateEntry *e;
AudioState *s = &glob_audio_state;
struct audio_driver *driver;
+ /* silence gcc warning about uninitialized variable */
+ QemuOptsList *list = NULL;
if (s->drv) {
- return;
+ if (dev) {
+ dolog("Cannot create more than one audio backend, sorry\n");
+ qapi_free_Audiodev(dev);
+ }
+ return -1;
}
+ if (dev) {
+ /* -audiodev option */
+ drvname = AudiodevDriver_str(dev->driver);
+ } else {
+ /* legacy implicit initialization */
+ audio_handle_legacy_opts();
+ list = qemu_find_opts("audiodev");
+ dev = parse_option(QTAILQ_FIRST(&list->head), &error_abort);
+ if (!dev) {
+ exit(1);
+ }
+ }
+ s->dev = dev;
+
QLIST_INIT (&s->hw_head_out);
QLIST_INIT (&s->hw_head_in);
QLIST_INIT (&s->cap_head);
@@ -1885,10 +1600,8 @@ static void audio_init (void)
s->ts = timer_new_ns(QEMU_CLOCK_VIRTUAL, audio_timer, s);
- 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",
@@ -1902,46 +1615,46 @@ 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) {
driver = audio_driver_lookup(drvname);
if (driver) {
- done = !audio_driver_init(s, driver);
+ done = !audio_driver_init(s, driver, dev);
} else {
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(audio_prio_list); i++) {
+ } else {
+ for (i = 0; !done && audio_prio_list[i]; i++) {
+ QemuOpts *opts = qemu_opts_find(list, audio_prio_list[i]);
driver = audio_driver_lookup(audio_prio_list[i]);
- if (driver && driver->can_be_default) {
- done = !audio_driver_init(s, driver);
+
+ if (driver && opts) {
+ qapi_free_Audiodev(dev);
+ dev = parse_option(opts, &error_abort);
+ if (!dev) {
+ exit(1);
+ }
+ s->dev = dev;
+ done = !audio_driver_init(s, driver, dev);
}
}
}
if (!done) {
driver = audio_driver_lookup("none");
- done = !audio_driver_init(s, driver);
+ done = !audio_driver_init(s, driver, dev);
assert(done);
dolog("warning: Using timer based audio emulation\n");
}
- 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 = NANOSECONDS_PER_SECOND / conf.period.hertz;
+ s->period_ticks = NANOSECONDS_PER_SECOND / dev->timer_period;
}
e = qemu_add_vm_change_state_handler (audio_vm_change_state_handler, s);
@@ -1952,11 +1665,12 @@ 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);
@@ -2127,3 +1841,158 @@ void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
}
}
}
+
+QemuOptsList qemu_audiodev_opts = {
+ .name = "audiodev",
+ .head = QTAILQ_HEAD_INITIALIZER(qemu_audiodev_opts.head),
+ .implied_opt_name = "driver",
+ .desc = {
+ /*
+ * no elements => accept any params
+ * sanity checking will happen later
+ */
+ { /* end of list */ }
+ },
+};
+
+static void validate_per_direction_opts(AudiodevPerDirectionOptions *pdo,
+ Error **errp)
+{
+ if (!pdo->has_fixed_settings) {
+ pdo->has_fixed_settings = true;
+ pdo->fixed_settings = true;
+ }
+ if (!pdo->fixed_settings &&
+ (pdo->has_frequency || pdo->has_channels || pdo->has_format)) {
+ error_setg(errp,
+ "You can't use frequency, channels or format with fixed-settings=off");
+ return;
+ }
+
+ if (!pdo->has_frequency) {
+ pdo->has_frequency = true;
+ pdo->frequency = 44100;
+ }
+ if (!pdo->has_channels) {
+ pdo->has_channels = true;
+ pdo->channels = 2;
+ }
+ if (!pdo->has_voices) {
+ pdo->has_voices = true;
+ pdo->voices = 1;
+ }
+ if (!pdo->has_format) {
+ pdo->has_format = true;
+ pdo->format = AUDIO_FORMAT_S16;
+ }
+}
+
+static Audiodev *parse_option(QemuOpts *opts, Error **errp)
+{
+ Error *local_err = NULL;
+ Visitor *v = opts_visitor_new(opts, true);
+ Audiodev *dev = NULL;
+ visit_type_Audiodev(v, NULL, &dev, &local_err);
+ visit_free(v);
+
+ if (local_err) {
+ goto err2;
+ }
+
+ validate_per_direction_opts(dev->in, &local_err);
+ if (local_err) {
+ goto err;
+ }
+ validate_per_direction_opts(dev->out, &local_err);
+ if (local_err) {
+ goto err;
+ }
+
+ if (!dev->has_timer_period) {
+ dev->has_timer_period = true;
+ dev->timer_period = 10000; /* 100Hz -> 10ms */
+ }
+
+ return dev;
+
+err:
+ qapi_free_Audiodev(dev);
+err2:
+ error_propagate(errp, local_err);
+ return NULL;
+}
+
+static int each_option(void *opaque, QemuOpts *opts, Error **errp)
+{
+ Audiodev *dev = parse_option(opts, errp);
+ if (!dev) {
+ return -1;
+ }
+ return audio_init(dev);
+}
+
+void audio_set_options(void)
+{
+ if (qemu_opts_foreach(qemu_find_opts("audiodev"), each_option, NULL,
+ &error_abort)) {
+ exit(1);
+ }
+}
+
+audsettings audiodev_to_audsettings(AudiodevPerDirectionOptions *pdo)
+{
+ return (audsettings) {
+ .freq = pdo->frequency,
+ .nchannels = pdo->channels,
+ .fmt = pdo->format,
+ .endianness = AUDIO_HOST_ENDIANNESS,
+ };
+}
+
+int audioformat_bytes_per_sample(AudioFormat fmt)
+{
+ switch (fmt) {
+ case AUDIO_FORMAT_U8:
+ case AUDIO_FORMAT_S8:
+ return 1;
+
+ case AUDIO_FORMAT_U16:
+ case AUDIO_FORMAT_S16:
+ return 2;
+
+ case AUDIO_FORMAT_U32:
+ case AUDIO_FORMAT_S32:
+ return 4;
+
+ case AUDIO_FORMAT__MAX:
+ ;
+ }
+ abort();
+}
+
+
+/* frames = freq * usec / 1e6 */
+int audio_buffer_frames(AudiodevPerDirectionOptions *pdo,
+ audsettings *as, int def_usecs)
+{
+ uint64_t usecs = pdo->has_buffer_len ? pdo->buffer_len : def_usecs;
+ return (as->freq * usecs + 500000) / 1000000;
+}
+
+/* samples = channels * frames = channels * freq * usec / 1e6 */
+int audio_buffer_samples(AudiodevPerDirectionOptions *pdo,
+ audsettings *as, int def_usecs)
+{
+ return as->nchannels * audio_buffer_frames(pdo, as, def_usecs);
+}
+
+/*
+ * bytes = bytes_per_sample * samples =
+ * bytes_per_sample * channels * freq * usec / 1e6
+ */
+int audio_buffer_bytes(AudiodevPerDirectionOptions *pdo,
+ audsettings *as, int def_usecs)
+{
+ return audio_buffer_samples(pdo, as, def_usecs) *
+ audioformat_bytes_per_sample(as->fmt);
+}
diff --git a/audio/audio.h b/audio/audio.h
index 02f29a3b3e..7df1b8b249 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -36,12 +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,
@@ -78,10 +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 (
@@ -163,4 +173,8 @@ void audio_sample_to_uint64(void *samples, int pos,
void audio_sample_from_uint64(void *samples, int pos,
uint64_t left, uint64_t right);
+void audio_set_options(void);
+void audio_handle_legacy_opts(void);
+void audio_legacy_help(void);
+
#endif /* QEMU_AUDIO_H */
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 244b454012..24b8793496 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -146,7 +146,7 @@ struct audio_driver {
const char *name;
const char *descr;
struct audio_option *options;
- void *(*init) (void);
+ void *(*init) (Audiodev *);
void (*fini) (void *);
struct audio_pcm_ops *pcm_ops;
int can_be_default;
@@ -193,6 +193,7 @@ struct SWVoiceCap {
struct AudioState {
struct audio_driver *drv;
+ Audiodev *dev;
void *drv_opaque;
QEMUTimer *ts;
@@ -203,10 +204,13 @@ struct AudioState {
int nb_hw_voices_out;
int nb_hw_voices_in;
int vm_running;
+ int64_t period_ticks;
};
extern const struct mixeng_volume nominal_volume;
+extern const char *audio_prio_list[];
+
void audio_driver_register(audio_driver *drv);
audio_driver *audio_driver_lookup(const char *name);
diff --git a/audio/audio_legacy.c b/audio/audio_legacy.c
new file mode 100644
index 0000000000..2242f06a5e
--- /dev/null
+++ b/audio/audio_legacy.c
@@ -0,0 +1,211 @@
+#include "qemu/osdep.h"
+#include "audio.h"
+#include "audio_int.h"
+#include "qemu-common.h"
+#include "qemu/config-file.h"
+#include "qemu/cutils.h"
+#include "qemu/option.h"
+#include "qapi/error.h"
+
+#define AUDIO_CAP "audio-legacy"
+#include "audio_int.h"
+
+typedef enum EnvTransform {
+ ENV_TRANSFORM_NONE,
+ ENV_TRANSFORM_BOOL,
+ ENV_TRANSFORM_FMT,
+ ENV_TRANSFORM_FRAMES_TO_USECS_IN,
+ ENV_TRANSFORM_FRAMES_TO_USECS_OUT,
+ ENV_TRANSFORM_SAMPLES_TO_USECS_IN,
+ ENV_TRANSFORM_SAMPLES_TO_USECS_OUT,
+ ENV_TRANSFORM_BYTES_TO_USECS_IN,
+ ENV_TRANSFORM_BYTES_TO_USECS_OUT,
+ ENV_TRANSFORM_MILLIS_TO_USECS,
+ ENV_TRANSFORM_HZ_TO_USECS,
+} EnvTransform;
+
+typedef struct SimpleEnvMap {
+ const char *name;
+ const char *option;
+ EnvTransform transform;
+} SimpleEnvMap;
+
+SimpleEnvMap global_map[] = {
+ /* DAC/out settings */
+ { "QEMU_AUDIO_DAC_FIXED_SETTINGS", "out.fixed-settings",
+ ENV_TRANSFORM_BOOL },
+ { "QEMU_AUDIO_DAC_FIXED_FREQ", "out.frequency" },
+ { "QEMU_AUDIO_DAC_FIXED_FMT", "out.format", ENV_TRANSFORM_FMT },
+ { "QEMU_AUDIO_DAC_FIXED_CHANNELS", "out.channels" },
+ { "QEMU_AUDIO_DAC_VOICES", "out.voices" },
+
+ /* ADC/in settings */
+ { "QEMU_AUDIO_ADC_FIXED_SETTINGS", "in.fixed-settings",
+ ENV_TRANSFORM_BOOL },
+ { "QEMU_AUDIO_ADC_FIXED_FREQ", "in.frequency" },
+ { "QEMU_AUDIO_ADC_FIXED_FMT", "in.format", ENV_TRANSFORM_FMT },
+ { "QEMU_AUDIO_ADC_FIXED_CHANNELS", "in.channels" },
+ { "QEMU_AUDIO_ADC_VOICES", "in.voices" },
+
+ /* general */
+ { "QEMU_AUDIO_TIMER_PERIOD", "timer-period", ENV_TRANSFORM_HZ_TO_USECS },
+ { /* End of list */ }
+};
+
+static unsigned long long toull(const char *str)
+{
+ unsigned long long ret;
+ if (parse_uint_full(str, &ret, 10)) {
+ dolog("Invalid integer 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)
+{
+ /* max length in decimal possible for an unsigned long long number */
+ #define LEN ((CHAR_BIT * sizeof(unsigned long long) - 1) / 3 + 2)
+ static char ret[LEN];
+ snprintf(ret, LEN, "%llu", val);
+ return ret;
+}
+
+static uint64_t frames_to_usecs(QemuOpts *opts, uint64_t frames, bool in)
+{
+ const char *opt = in ? "in.frequency" : "out.frequency";
+ uint64_t freq = qemu_opt_get_number(opts, opt, 44100);
+ return (frames * 1000000 + freq / 2) / freq;
+}
+
+static uint64_t samples_to_usecs(QemuOpts *opts, uint64_t samples, bool in)
+{
+ const char *opt = in ? "in.channels" : "out.channels";
+ uint64_t channels = qemu_opt_get_number(opts, opt, 2);
+ return frames_to_usecs(opts, samples / channels, in);
+}
+
+static uint64_t bytes_to_usecs(QemuOpts *opts, uint64_t bytes, bool in)
+{
+ const char *opt = in ? "in.format" : "out.format";
+ const char *val = qemu_opt_get(opts, opt);
+ uint64_t bytes_per_sample = (val ? toull(val) : 16) / 8;
+ return samples_to_usecs(opts, bytes * bytes_per_sample, in);
+}
+
+static const char *transform_val(QemuOpts *opts, const char *val,
+ EnvTransform transform)
+{
+ switch (transform) {
+ case ENV_TRANSFORM_NONE:
+ return val;
+
+ case ENV_TRANSFORM_BOOL:
+ return toull(val) ? "on" : "off";
+
+ case ENV_TRANSFORM_FMT:
+ if (strcasecmp(val, "u8") == 0) {
+ return "u8";
+ } else if (strcasecmp(val, "u16") == 0) {
+ return "u16";
+ } else if (strcasecmp(val, "u32") == 0) {
+ return "u32";
+ } else if (strcasecmp(val, "s8") == 0) {
+ return "s8";
+ } else if (strcasecmp(val, "s16") == 0) {
+ return "s16";
+ } else if (strcasecmp(val, "s32") == 0) {
+ return "s32";
+ } else {
+ dolog("Invalid audio format `%s'\n", val);
+ exit(1);
+ }
+
+ case ENV_TRANSFORM_FRAMES_TO_USECS_IN:
+ return tostr(frames_to_usecs(opts, toull(val), true));
+ case ENV_TRANSFORM_FRAMES_TO_USECS_OUT:
+ return tostr(frames_to_usecs(opts, toull(val), false));
+
+ case ENV_TRANSFORM_SAMPLES_TO_USECS_IN:
+ return tostr(samples_to_usecs(opts, toull(val), true));
+ case ENV_TRANSFORM_SAMPLES_TO_USECS_OUT:
+ return tostr(samples_to_usecs(opts, toull(val), false));
+
+ case ENV_TRANSFORM_BYTES_TO_USECS_IN:
+ return tostr(bytes_to_usecs(opts, toull(val), true));
+ case ENV_TRANSFORM_BYTES_TO_USECS_OUT:
+ return tostr(bytes_to_usecs(opts, toull(val), false));
+
+ case ENV_TRANSFORM_MILLIS_TO_USECS:
+ return tostr(toull(val) * 1000);
+
+ case ENV_TRANSFORM_HZ_TO_USECS:
+ return tostr(1000000 / toull(val));
+ }
+
+ abort(); /* it's unreachable, gcc */
+}
+
+static void handle_env_opts(QemuOpts *opts, SimpleEnvMap *map)
+{
+ while (map->name) {
+ const char *val = getenv(map->name);
+
+ if (val) {
+ qemu_opt_set(opts, map->option,
+ transform_val(opts, val, map->transform),
+ &error_abort);
+ }
+
+ ++map;
+ }
+}
+
+static void legacy_opt(const char *drv)
+{
+ QemuOpts *opts;
+ opts = qemu_opts_create(qemu_find_opts("audiodev"), drv, true,
+ &error_abort);
+ qemu_opt_set(opts, "driver", drv, &error_abort);
+
+ handle_env_opts(opts, global_map);
+}
+
+void audio_handle_legacy_opts(void)
+{
+ const char *drvname = getenv("QEMU_AUDIO_DRV");
+
+ if (drvname) {
+ audio_driver *driver = audio_driver_lookup(drvname);
+ if (!driver) {
+ dolog("Unknown audio driver `%s'\n", drvname);
+ }
+ legacy_opt(drvname);
+ } else {
+ for (int i = 0; audio_prio_list[i]; i++) {
+ audio_driver *driver = audio_driver_lookup(audio_prio_list[i]);
+ if (driver && driver->can_be_default) {
+ legacy_opt(driver->name);
+ }
+ }
+ }
+}
+
+static int legacy_help_each(void *opaque, QemuOpts *opts, Error **errp)
+{
+ printf("-audiodev ");
+ qemu_opts_print(opts, ",");
+ printf("\n");
+ return 0;
+}
+
+void audio_legacy_help(void)
+{
+ printf("Environment variable based configuration deprecated.\n");
+ printf("Please use the new -audiodev option.\n");
+
+ audio_handle_legacy_opts();
+ printf("\nEquivalent -audiodev to your current environment variables:\n");
+ qemu_opts_foreach(qemu_find_opts("audiodev"), legacy_help_each, NULL, NULL);
+}
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 7de227d2d1..c1d7207abd 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(__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 638c60b300..7d4225dbee 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -685,7 +685,7 @@ static CoreaudioConf glob_conf = {
.nbuffers = 4,
};
-static void *coreaudio_audio_init (void)
+static void *coreaudio_audio_init(Audiodev *dev)
{
CoreaudioConf *conf = g_malloc(sizeof(CoreaudioConf));
*conf = glob_conf;
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index 3ed73a30d1..02fe777cba 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -783,7 +783,7 @@ 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;
diff --git a/audio/noaudio.c b/audio/noaudio.c
index 1bfebeca7d..2cc274c5e5 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -136,7 +136,7 @@ static int no_ctl_in (HWVoiceIn *hw, int cmd, ...)
return 0;
}
-static void *no_audio_init (void)
+static void *no_audio_init (Audiodev *dev)
{
return &no_audio_init;
}
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 355e8fbda5..e0cadbef29 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -842,7 +842,7 @@ static OSSConf glob_conf = {
.policy = 5
};
-static void *oss_audio_init (void)
+static void *oss_audio_init(Audiodev *dev)
{
OSSConf *conf = g_malloc(sizeof(OSSConf));
*conf = glob_conf;
diff --git a/audio/paaudio.c b/audio/paaudio.c
index f1f9a741ac..0981f010c9 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -812,7 +812,7 @@ 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;
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index aa42ea26bf..097841fde1 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -436,7 +436,7 @@ 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) {
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index 3aeb0cb357..affc3df17f 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -77,7 +77,7 @@ static const SpiceRecordInterface record_sif = {
.base.minor_version = SPICE_INTERFACE_RECORD_MINOR,
};
-static void *spice_audio_init (void)
+static void *spice_audio_init(Audiodev *dev)
{
if (!using_spice) {
return NULL;
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 35a614785e..9eff3555b3 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -232,7 +232,7 @@ static WAVConf glob_conf = {
.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;
diff --git a/vl.c b/vl.c
index 8353d3c718..b5364ffe46 100644
--- a/vl.c
+++ b/vl.c
@@ -3074,6 +3074,7 @@ int main(int argc, char **argv, char **envp)
qemu_add_opts(&qemu_option_rom_opts);
qemu_add_opts(&qemu_machine_opts);
qemu_add_opts(&qemu_accel_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);
@@ -3307,9 +3308,15 @@ 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_noisily(qemu_find_opts("audiodev"),
+ optarg, true)) {
+ exit(1);
+ }
+ break;
case QEMU_OPTION_soundhw:
select_soundhw (optarg);
break;
@@ -4545,6 +4552,8 @@ int main(int argc, char **argv, char **envp)
/* do monitor/qmp handling at preconfig state if requested */
main_loop();
+ audio_set_options();
+
/* from here on runstate is RUN_STATE_PRELAUNCH */
machine_run_board_init(current_machine);
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* Re: [Qemu-devel] [PATCH v2 06/52] audio: -audiodev command line option basic implementation
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 06/52] audio: -audiodev command line option basic implementation Kővágó, Zoltán
@ 2019-01-07 13:13 ` Markus Armbruster
2019-01-07 20:48 ` Zoltán Kővágó
0 siblings, 1 reply; 91+ messages in thread
From: Markus Armbruster @ 2019-01-07 13:13 UTC (permalink / raw)
To: Kővágó, Zoltán
Cc: qemu-devel, Paolo Bonzini, Gerd Hoffmann
"Kővágó, Zoltán" <dirty.ice.hu@gmail.com> writes:
> Audio drivers now get an Audiodev * as config paramters, instead of the
> global audio_option structs. There is some code in audio/audio_legacy.c
> that converts the old environment variables to audiodev options (this
> way backends do not have to worry about legacy options). It also
> contains a replacement of -audio-help, which prints out the equivalent
> -audiodev based config of the currently specified environment variables.
>
> Note that backends are not updated and still rely on environment
> variables.
>
> Also note that (due to moving try-poll from global to backend specific
> option) currently ALSA and OSS will always try poll mode, regardless of
> environment variables or -audiodev options.
>
> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
> ---
[...]
> diff --git a/audio/audio.c b/audio/audio.c
> index 96cbd57c37..e7f25ea84b 100644
> --- a/audio/audio.c
> +++ b/audio/audio.c
[...]
> @@ -2127,3 +1841,158 @@ void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
> }
> }
> }
> +
> +QemuOptsList qemu_audiodev_opts = {
> + .name = "audiodev",
> + .head = QTAILQ_HEAD_INITIALIZER(qemu_audiodev_opts.head),
> + .implied_opt_name = "driver",
> + .desc = {
> + /*
> + * no elements => accept any params
> + * sanity checking will happen later
> + */
> + { /* end of list */ }
> + },
> +};
> +
> +static void validate_per_direction_opts(AudiodevPerDirectionOptions *pdo,
> + Error **errp)
> +{
> + if (!pdo->has_fixed_settings) {
> + pdo->has_fixed_settings = true;
> + pdo->fixed_settings = true;
> + }
> + if (!pdo->fixed_settings &&
> + (pdo->has_frequency || pdo->has_channels || pdo->has_format)) {
> + error_setg(errp,
> + "You can't use frequency, channels or format with fixed-settings=off");
> + return;
> + }
> +
> + if (!pdo->has_frequency) {
> + pdo->has_frequency = true;
> + pdo->frequency = 44100;
> + }
> + if (!pdo->has_channels) {
> + pdo->has_channels = true;
> + pdo->channels = 2;
> + }
> + if (!pdo->has_voices) {
> + pdo->has_voices = true;
> + pdo->voices = 1;
> + }
> + if (!pdo->has_format) {
> + pdo->has_format = true;
> + pdo->format = AUDIO_FORMAT_S16;
> + }
> +}
> +
> +static Audiodev *parse_option(QemuOpts *opts, Error **errp)
> +{
> + Error *local_err = NULL;
> + Visitor *v = opts_visitor_new(opts, true);
> + Audiodev *dev = NULL;
> + visit_type_Audiodev(v, NULL, &dev, &local_err);
> + visit_free(v);
> +
> + if (local_err) {
> + goto err2;
> + }
> +
> + validate_per_direction_opts(dev->in, &local_err);
> + if (local_err) {
> + goto err;
> + }
> + validate_per_direction_opts(dev->out, &local_err);
> + if (local_err) {
> + goto err;
> + }
> +
> + if (!dev->has_timer_period) {
> + dev->has_timer_period = true;
> + dev->timer_period = 10000; /* 100Hz -> 10ms */
> + }
> +
> + return dev;
> +
> +err:
> + qapi_free_Audiodev(dev);
> +err2:
> + error_propagate(errp, local_err);
> + return NULL;
> +}
> +
> +static int each_option(void *opaque, QemuOpts *opts, Error **errp)
> +{
> + Audiodev *dev = parse_option(opts, errp);
> + if (!dev) {
> + return -1;
> + }
> + return audio_init(dev);
> +}
> +
> +void audio_set_options(void)
> +{
> + if (qemu_opts_foreach(qemu_find_opts("audiodev"), each_option, NULL,
> + &error_abort)) {
> + exit(1);
> + }
> +}
[...]
> diff --git a/vl.c b/vl.c
> index 8353d3c718..b5364ffe46 100644
> --- a/vl.c
> +++ b/vl.c
> @@ -3074,6 +3074,7 @@ int main(int argc, char **argv, char **envp)
> qemu_add_opts(&qemu_option_rom_opts);
> qemu_add_opts(&qemu_machine_opts);
> qemu_add_opts(&qemu_accel_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);
> @@ -3307,9 +3308,15 @@ 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_noisily(qemu_find_opts("audiodev"),
> + optarg, true)) {
> + exit(1);
> + }
> + break;
> case QEMU_OPTION_soundhw:
> select_soundhw (optarg);
> break;
> @@ -4545,6 +4552,8 @@ int main(int argc, char **argv, char **envp)
> /* do monitor/qmp handling at preconfig state if requested */
> main_loop();
>
> + audio_set_options();
> +
> /* from here on runstate is RUN_STATE_PRELAUNCH */
> machine_run_board_init(current_machine);
This uses the options visitor, roughly like -netdev. Depends on your
options visitor enhancements. Is there any particular reason not to do
it like -blockdev and -display, with qobject_input_visitor_new_str()?
I'm asking because I'd very much like new code to use
qobject_input_visitor_new_str() rather than the options visitor.
Today, the options visitor looks like an evolutionary dead end. My
command-line qapification (still stuck at the RFC stage, hope to get it
unstuck this year) relies on qobject_input_visitor_new_str(). The goal
is to convert *all* of the command line to it.
I suspect your series uses the options visitor only because back when
you started, qobject_input_visitor_new_str() didn't exist.
^ permalink raw reply [flat|nested] 91+ messages in thread* Re: [Qemu-devel] [PATCH v2 06/52] audio: -audiodev command line option basic implementation
2019-01-07 13:13 ` Markus Armbruster
@ 2019-01-07 20:48 ` Zoltán Kővágó
2019-01-08 3:42 ` Markus Armbruster
2019-01-08 6:06 ` Gerd Hoffmann
0 siblings, 2 replies; 91+ messages in thread
From: Zoltán Kővágó @ 2019-01-07 20:48 UTC (permalink / raw)
To: Markus Armbruster; +Cc: qemu-devel, Paolo Bonzini, Gerd Hoffmann
On 2019-01-07 14:13, Markus Armbruster wrote:
> "Kővágó, Zoltán" <dirty.ice.hu@gmail.com> writes:
>
>> Audio drivers now get an Audiodev * as config paramters, instead of the
>> global audio_option structs. There is some code in audio/audio_legacy.c
>> that converts the old environment variables to audiodev options (this
>> way backends do not have to worry about legacy options). It also
>> contains a replacement of -audio-help, which prints out the equivalent
>> -audiodev based config of the currently specified environment variables.
>>
>> Note that backends are not updated and still rely on environment
>> variables.
>>
>> Also note that (due to moving try-poll from global to backend specific
>> option) currently ALSA and OSS will always try poll mode, regardless of
>> environment variables or -audiodev options.
>>
>> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
>> ---
> [...]
>> diff --git a/audio/audio.c b/audio/audio.c
>> index 96cbd57c37..e7f25ea84b 100644
>> --- a/audio/audio.c
>> +++ b/audio/audio.c
> [...]
>> @@ -2127,3 +1841,158 @@ void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
>> }
>> }
>> }
>> +
>> +QemuOptsList qemu_audiodev_opts = {
>> + .name = "audiodev",
>> + .head = QTAILQ_HEAD_INITIALIZER(qemu_audiodev_opts.head),
>> + .implied_opt_name = "driver",
>> + .desc = {
>> + /*
>> + * no elements => accept any params
>> + * sanity checking will happen later
>> + */
>> + { /* end of list */ }
>> + },
>> +};
>> +
>> +static void validate_per_direction_opts(AudiodevPerDirectionOptions *pdo,
>> + Error **errp)
>> +{
>> + if (!pdo->has_fixed_settings) {
>> + pdo->has_fixed_settings = true;
>> + pdo->fixed_settings = true;
>> + }
>> + if (!pdo->fixed_settings &&
>> + (pdo->has_frequency || pdo->has_channels || pdo->has_format)) {
>> + error_setg(errp,
>> + "You can't use frequency, channels or format with fixed-settings=off");
>> + return;
>> + }
>> +
>> + if (!pdo->has_frequency) {
>> + pdo->has_frequency = true;
>> + pdo->frequency = 44100;
>> + }
>> + if (!pdo->has_channels) {
>> + pdo->has_channels = true;
>> + pdo->channels = 2;
>> + }
>> + if (!pdo->has_voices) {
>> + pdo->has_voices = true;
>> + pdo->voices = 1;
>> + }
>> + if (!pdo->has_format) {
>> + pdo->has_format = true;
>> + pdo->format = AUDIO_FORMAT_S16;
>> + }
>> +}
>> +
>> +static Audiodev *parse_option(QemuOpts *opts, Error **errp)
>> +{
>> + Error *local_err = NULL;
>> + Visitor *v = opts_visitor_new(opts, true);
>> + Audiodev *dev = NULL;
>> + visit_type_Audiodev(v, NULL, &dev, &local_err);
>> + visit_free(v);
>> +
>> + if (local_err) {
>> + goto err2;
>> + }
>> +
>> + validate_per_direction_opts(dev->in, &local_err);
>> + if (local_err) {
>> + goto err;
>> + }
>> + validate_per_direction_opts(dev->out, &local_err);
>> + if (local_err) {
>> + goto err;
>> + }
>> +
>> + if (!dev->has_timer_period) {
>> + dev->has_timer_period = true;
>> + dev->timer_period = 10000; /* 100Hz -> 10ms */
>> + }
>> +
>> + return dev;
>> +
>> +err:
>> + qapi_free_Audiodev(dev);
>> +err2:
>> + error_propagate(errp, local_err);
>> + return NULL;
>> +}
>> +
>> +static int each_option(void *opaque, QemuOpts *opts, Error **errp)
>> +{
>> + Audiodev *dev = parse_option(opts, errp);
>> + if (!dev) {
>> + return -1;
>> + }
>> + return audio_init(dev);
>> +}
>> +
>> +void audio_set_options(void)
>> +{
>> + if (qemu_opts_foreach(qemu_find_opts("audiodev"), each_option, NULL,
>> + &error_abort)) {
>> + exit(1);
>> + }
>> +}
> [...]
>> diff --git a/vl.c b/vl.c
>> index 8353d3c718..b5364ffe46 100644
>> --- a/vl.c
>> +++ b/vl.c
>> @@ -3074,6 +3074,7 @@ int main(int argc, char **argv, char **envp)
>> qemu_add_opts(&qemu_option_rom_opts);
>> qemu_add_opts(&qemu_machine_opts);
>> qemu_add_opts(&qemu_accel_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);
>> @@ -3307,9 +3308,15 @@ 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_noisily(qemu_find_opts("audiodev"),
>> + optarg, true)) {
>> + exit(1);
>> + }
>> + break;
>> case QEMU_OPTION_soundhw:
>> select_soundhw (optarg);
>> break;
>> @@ -4545,6 +4552,8 @@ int main(int argc, char **argv, char **envp)
>> /* do monitor/qmp handling at preconfig state if requested */
>> main_loop();
>>
>> + audio_set_options();
>> +
>> /* from here on runstate is RUN_STATE_PRELAUNCH */
>> machine_run_board_init(current_machine);
>
> This uses the options visitor, roughly like -netdev. Depends on your
> options visitor enhancements. Is there any particular reason not to do
> it like -blockdev and -display, with qobject_input_visitor_new_str()?
> I'm asking because I'd very much like new code to use
> qobject_input_visitor_new_str() rather than the options visitor.
>
> Today, the options visitor looks like an evolutionary dead end. My
> command-line qapification (still stuck at the RFC stage, hope to get it
> unstuck this year) relies on qobject_input_visitor_new_str(). The goal
> is to convert *all* of the command line to it.
>
> I suspect your series uses the options visitor only because back when
> you started, qobject_input_visitor_new_str() didn't exist.
>
Yes, this patch series is a bit old, and at that time this was the best
I could do. I can look into it this (probably only on the weekend
though), it looks like it supports foo.bar=something syntax, so I don't
have to specify a json on the command line...
Regards,
Zoltan
^ permalink raw reply [flat|nested] 91+ messages in thread* Re: [Qemu-devel] [PATCH v2 06/52] audio: -audiodev command line option basic implementation
2019-01-07 20:48 ` Zoltán Kővágó
@ 2019-01-08 3:42 ` Markus Armbruster
2019-01-10 0:13 ` Zoltán Kővágó
2019-01-08 6:06 ` Gerd Hoffmann
1 sibling, 1 reply; 91+ messages in thread
From: Markus Armbruster @ 2019-01-08 3:42 UTC (permalink / raw)
To: Zoltán Kővágó
Cc: Paolo Bonzini, qemu-devel, Gerd Hoffmann
"Zoltán Kővágó" <dirty.ice.hu@gmail.com> writes:
> On 2019-01-07 14:13, Markus Armbruster wrote:
>> "Kővágó, Zoltán" <dirty.ice.hu@gmail.com> writes:
>>
>>> Audio drivers now get an Audiodev * as config paramters, instead of the
>>> global audio_option structs. There is some code in audio/audio_legacy.c
>>> that converts the old environment variables to audiodev options (this
>>> way backends do not have to worry about legacy options). It also
>>> contains a replacement of -audio-help, which prints out the equivalent
>>> -audiodev based config of the currently specified environment variables.
>>>
>>> Note that backends are not updated and still rely on environment
>>> variables.
>>>
>>> Also note that (due to moving try-poll from global to backend specific
>>> option) currently ALSA and OSS will always try poll mode, regardless of
>>> environment variables or -audiodev options.
>>>
>>> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
>>> ---
>> [...]
>>> diff --git a/audio/audio.c b/audio/audio.c
>>> index 96cbd57c37..e7f25ea84b 100644
>>> --- a/audio/audio.c
>>> +++ b/audio/audio.c
>> [...]
>>> @@ -2127,3 +1841,158 @@ void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
>>> }
>>> }
>>> }
>>> +
>>> +QemuOptsList qemu_audiodev_opts = {
>>> + .name = "audiodev",
>>> + .head = QTAILQ_HEAD_INITIALIZER(qemu_audiodev_opts.head),
>>> + .implied_opt_name = "driver",
>>> + .desc = {
>>> + /*
>>> + * no elements => accept any params
>>> + * sanity checking will happen later
>>> + */
>>> + { /* end of list */ }
>>> + },
>>> +};
>>> +
>>> +static void validate_per_direction_opts(AudiodevPerDirectionOptions *pdo,
>>> + Error **errp)
>>> +{
>>> + if (!pdo->has_fixed_settings) {
>>> + pdo->has_fixed_settings = true;
>>> + pdo->fixed_settings = true;
>>> + }
>>> + if (!pdo->fixed_settings &&
>>> + (pdo->has_frequency || pdo->has_channels || pdo->has_format)) {
>>> + error_setg(errp,
>>> + "You can't use frequency, channels or format with fixed-settings=off");
>>> + return;
>>> + }
>>> +
>>> + if (!pdo->has_frequency) {
>>> + pdo->has_frequency = true;
>>> + pdo->frequency = 44100;
>>> + }
>>> + if (!pdo->has_channels) {
>>> + pdo->has_channels = true;
>>> + pdo->channels = 2;
>>> + }
>>> + if (!pdo->has_voices) {
>>> + pdo->has_voices = true;
>>> + pdo->voices = 1;
>>> + }
>>> + if (!pdo->has_format) {
>>> + pdo->has_format = true;
>>> + pdo->format = AUDIO_FORMAT_S16;
>>> + }
>>> +}
>>> +
>>> +static Audiodev *parse_option(QemuOpts *opts, Error **errp)
>>> +{
>>> + Error *local_err = NULL;
>>> + Visitor *v = opts_visitor_new(opts, true);
>>> + Audiodev *dev = NULL;
>>> + visit_type_Audiodev(v, NULL, &dev, &local_err);
>>> + visit_free(v);
>>> +
>>> + if (local_err) {
>>> + goto err2;
>>> + }
>>> +
>>> + validate_per_direction_opts(dev->in, &local_err);
>>> + if (local_err) {
>>> + goto err;
>>> + }
>>> + validate_per_direction_opts(dev->out, &local_err);
>>> + if (local_err) {
>>> + goto err;
>>> + }
>>> +
>>> + if (!dev->has_timer_period) {
>>> + dev->has_timer_period = true;
>>> + dev->timer_period = 10000; /* 100Hz -> 10ms */
>>> + }
>>> +
>>> + return dev;
>>> +
>>> +err:
>>> + qapi_free_Audiodev(dev);
>>> +err2:
>>> + error_propagate(errp, local_err);
>>> + return NULL;
>>> +}
>>> +
>>> +static int each_option(void *opaque, QemuOpts *opts, Error **errp)
>>> +{
>>> + Audiodev *dev = parse_option(opts, errp);
>>> + if (!dev) {
>>> + return -1;
>>> + }
>>> + return audio_init(dev);
>>> +}
>>> +
>>> +void audio_set_options(void)
>>> +{
>>> + if (qemu_opts_foreach(qemu_find_opts("audiodev"), each_option, NULL,
>>> + &error_abort)) {
>>> + exit(1);
>>> + }
>>> +}
>> [...]
>>> diff --git a/vl.c b/vl.c
>>> index 8353d3c718..b5364ffe46 100644
>>> --- a/vl.c
>>> +++ b/vl.c
>>> @@ -3074,6 +3074,7 @@ int main(int argc, char **argv, char **envp)
>>> qemu_add_opts(&qemu_option_rom_opts);
>>> qemu_add_opts(&qemu_machine_opts);
>>> qemu_add_opts(&qemu_accel_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);
>>> @@ -3307,9 +3308,15 @@ 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_noisily(qemu_find_opts("audiodev"),
>>> + optarg, true)) {
>>> + exit(1);
>>> + }
>>> + break;
>>> case QEMU_OPTION_soundhw:
>>> select_soundhw (optarg);
>>> break;
>>> @@ -4545,6 +4552,8 @@ int main(int argc, char **argv, char **envp)
>>> /* do monitor/qmp handling at preconfig state if requested */
>>> main_loop();
>>>
>>> + audio_set_options();
>>> +
>>> /* from here on runstate is RUN_STATE_PRELAUNCH */
>>> machine_run_board_init(current_machine);
>>
>> This uses the options visitor, roughly like -netdev. Depends on your
>> options visitor enhancements. Is there any particular reason not to do
>> it like -blockdev and -display, with qobject_input_visitor_new_str()?
>> I'm asking because I'd very much like new code to use
>> qobject_input_visitor_new_str() rather than the options visitor.
>>
>> Today, the options visitor looks like an evolutionary dead end. My
>> command-line qapification (still stuck at the RFC stage, hope to get it
>> unstuck this year) relies on qobject_input_visitor_new_str(). The goal
>> is to convert *all* of the command line to it.
>>
>> I suspect your series uses the options visitor only because back when
>> you started, qobject_input_visitor_new_str() didn't exist.
>>
>
> Yes, this patch series is a bit old, and at that time this was the best
> I could do. I can look into it this (probably only on the weekend
> though), it looks like it supports foo.bar=something syntax, so I don't
> have to specify a json on the command line...
The dotted key syntax and how it relates to JSON is documented in
util/keyval.c.
If you have further questions on this stuff, please don't hesitate to
ask me.
If you get to the point where you feel your (beefy) series has gotten
stuck on the conversion away from the options visitor, let's discuss how
to best get it unstuck.
Thanks!
^ permalink raw reply [flat|nested] 91+ messages in thread* Re: [Qemu-devel] [PATCH v2 06/52] audio: -audiodev command line option basic implementation
2019-01-08 3:42 ` Markus Armbruster
@ 2019-01-10 0:13 ` Zoltán Kővágó
2019-01-10 7:25 ` Gerd Hoffmann
0 siblings, 1 reply; 91+ messages in thread
From: Zoltán Kővágó @ 2019-01-10 0:13 UTC (permalink / raw)
To: Markus Armbruster; +Cc: Paolo Bonzini, qemu-devel, Gerd Hoffmann
On 2019-01-08 04:42, Markus Armbruster wrote:
> "Zoltán Kővágó" <dirty.ice.hu@gmail.com> writes:
>
>> On 2019-01-07 14:13, Markus Armbruster wrote:
>>> "Kővágó, Zoltán" <dirty.ice.hu@gmail.com> writes:
>>>
>>>> Audio drivers now get an Audiodev * as config paramters, instead of the
>>>> global audio_option structs. There is some code in audio/audio_legacy.c
>>>> that converts the old environment variables to audiodev options (this
>>>> way backends do not have to worry about legacy options). It also
>>>> contains a replacement of -audio-help, which prints out the equivalent
>>>> -audiodev based config of the currently specified environment variables.
>>>>
>>>> Note that backends are not updated and still rely on environment
>>>> variables.
>>>>
>>>> Also note that (due to moving try-poll from global to backend specific
>>>> option) currently ALSA and OSS will always try poll mode, regardless of
>>>> environment variables or -audiodev options.
>>>>
>>>> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
>>>> ---
>>> [...]
>>>> diff --git a/audio/audio.c b/audio/audio.c
>>>> index 96cbd57c37..e7f25ea84b 100644
>>>> --- a/audio/audio.c
>>>> +++ b/audio/audio.c
>>> [...]
>>>> @@ -2127,3 +1841,158 @@ void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
>>>> }
>>>> }
>>>> }
>>>> +
>>>> +QemuOptsList qemu_audiodev_opts = {
>>>> + .name = "audiodev",
>>>> + .head = QTAILQ_HEAD_INITIALIZER(qemu_audiodev_opts.head),
>>>> + .implied_opt_name = "driver",
>>>> + .desc = {
>>>> + /*
>>>> + * no elements => accept any params
>>>> + * sanity checking will happen later
>>>> + */
>>>> + { /* end of list */ }
>>>> + },
>>>> +};
>>>> +
>>>> +static void validate_per_direction_opts(AudiodevPerDirectionOptions *pdo,
>>>> + Error **errp)
>>>> +{
>>>> + if (!pdo->has_fixed_settings) {
>>>> + pdo->has_fixed_settings = true;
>>>> + pdo->fixed_settings = true;
>>>> + }
>>>> + if (!pdo->fixed_settings &&
>>>> + (pdo->has_frequency || pdo->has_channels || pdo->has_format)) {
>>>> + error_setg(errp,
>>>> + "You can't use frequency, channels or format with fixed-settings=off");
>>>> + return;
>>>> + }
>>>> +
>>>> + if (!pdo->has_frequency) {
>>>> + pdo->has_frequency = true;
>>>> + pdo->frequency = 44100;
>>>> + }
>>>> + if (!pdo->has_channels) {
>>>> + pdo->has_channels = true;
>>>> + pdo->channels = 2;
>>>> + }
>>>> + if (!pdo->has_voices) {
>>>> + pdo->has_voices = true;
>>>> + pdo->voices = 1;
>>>> + }
>>>> + if (!pdo->has_format) {
>>>> + pdo->has_format = true;
>>>> + pdo->format = AUDIO_FORMAT_S16;
>>>> + }
>>>> +}
>>>> +
>>>> +static Audiodev *parse_option(QemuOpts *opts, Error **errp)
>>>> +{
>>>> + Error *local_err = NULL;
>>>> + Visitor *v = opts_visitor_new(opts, true);
>>>> + Audiodev *dev = NULL;
>>>> + visit_type_Audiodev(v, NULL, &dev, &local_err);
>>>> + visit_free(v);
>>>> +
>>>> + if (local_err) {
>>>> + goto err2;
>>>> + }
>>>> +
>>>> + validate_per_direction_opts(dev->in, &local_err);
>>>> + if (local_err) {
>>>> + goto err;
>>>> + }
>>>> + validate_per_direction_opts(dev->out, &local_err);
>>>> + if (local_err) {
>>>> + goto err;
>>>> + }
>>>> +
>>>> + if (!dev->has_timer_period) {
>>>> + dev->has_timer_period = true;
>>>> + dev->timer_period = 10000; /* 100Hz -> 10ms */
>>>> + }
>>>> +
>>>> + return dev;
>>>> +
>>>> +err:
>>>> + qapi_free_Audiodev(dev);
>>>> +err2:
>>>> + error_propagate(errp, local_err);
>>>> + return NULL;
>>>> +}
>>>> +
>>>> +static int each_option(void *opaque, QemuOpts *opts, Error **errp)
>>>> +{
>>>> + Audiodev *dev = parse_option(opts, errp);
>>>> + if (!dev) {
>>>> + return -1;
>>>> + }
>>>> + return audio_init(dev);
>>>> +}
>>>> +
>>>> +void audio_set_options(void)
>>>> +{
>>>> + if (qemu_opts_foreach(qemu_find_opts("audiodev"), each_option, NULL,
>>>> + &error_abort)) {
>>>> + exit(1);
>>>> + }
>>>> +}
>>> [...]
>>>> diff --git a/vl.c b/vl.c
>>>> index 8353d3c718..b5364ffe46 100644
>>>> --- a/vl.c
>>>> +++ b/vl.c
>>>> @@ -3074,6 +3074,7 @@ int main(int argc, char **argv, char **envp)
>>>> qemu_add_opts(&qemu_option_rom_opts);
>>>> qemu_add_opts(&qemu_machine_opts);
>>>> qemu_add_opts(&qemu_accel_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);
>>>> @@ -3307,9 +3308,15 @@ 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_noisily(qemu_find_opts("audiodev"),
>>>> + optarg, true)) {
>>>> + exit(1);
>>>> + }
>>>> + break;
>>>> case QEMU_OPTION_soundhw:
>>>> select_soundhw (optarg);
>>>> break;
>>>> @@ -4545,6 +4552,8 @@ int main(int argc, char **argv, char **envp)
>>>> /* do monitor/qmp handling at preconfig state if requested */
>>>> main_loop();
>>>>
>>>> + audio_set_options();
>>>> +
>>>> /* from here on runstate is RUN_STATE_PRELAUNCH */
>>>> machine_run_board_init(current_machine);
>>>
>>> This uses the options visitor, roughly like -netdev. Depends on your
>>> options visitor enhancements. Is there any particular reason not to do
>>> it like -blockdev and -display, with qobject_input_visitor_new_str()?
>>> I'm asking because I'd very much like new code to use
>>> qobject_input_visitor_new_str() rather than the options visitor.
>>>
>>> Today, the options visitor looks like an evolutionary dead end. My
>>> command-line qapification (still stuck at the RFC stage, hope to get it
>>> unstuck this year) relies on qobject_input_visitor_new_str(). The goal
>>> is to convert *all* of the command line to it.
>>>
>>> I suspect your series uses the options visitor only because back when
>>> you started, qobject_input_visitor_new_str() didn't exist.
>>>
>>
>> Yes, this patch series is a bit old, and at that time this was the best
>> I could do. I can look into it this (probably only on the weekend
>> though), it looks like it supports foo.bar=something syntax, so I don't
>> have to specify a json on the command line...
>
> The dotted key syntax and how it relates to JSON is documented in
> util/keyval.c.
>
> If you have further questions on this stuff, please don't hesitate to
> ask me.
>
> If you get to the point where you feel your (beefy) series has gotten
> stuck on the conversion away from the options visitor, let's discuss how
> to best get it unstuck.
>
> Thanks!
>
I briefly looked at converting this single patch, and of course I ran
into a few issues.
The audio_legacy code currently creates a QemuOpts from the environment
variables. This had a nice effect that once I converted the options, I
could print them with qemu_opts_print, helping the user how to convert
the old env variables to -audiodev, since some of the conversions are
not so trivial/straightforward.
I converted them to QDict, which complicated things a bit since now I
have to do proper nesting (until now I could just set key to "foo.bar"
and it worked), especially in the *_to_usecs functions that accessed
previously set options. Also while there's
qobject_input_visitor_new_str, I couldn't find anything that would do
the reverse. I wrote a quick and dirty function that does this, it
appears to work, but unlike QemuOpts, QDict is unordered, and writing
-audiodev id=foo,driver=alsa,rest... is IMHO more logical than writing
them alphabetically. Or maybe it's just me who's too picky and it
doesn't matter.
I was thinking about creating an Audiodev (the qapi type) directly would
be better, then somehow print it with reflection. While this is not a
typical use of qapi, at least qmp_qom_list creates qapi objects
directly, so I assume it's ok. This would save us from parsing
essentially the same data twice, and the code to deal with an Audiodev
is simpler that dealing with the generic QDicts and QObjects, but it
would require some refactoring.
The second problem: with my patched options visitor, nested structs were
required in qapi, the options visitor unconditionally filled them. With
qobject_input_visitor, I have to mark them optional, otherwise the user
have to specify at least one option from the nested structs.
Essentially, if you have something like this:
{ 'struct': 'Inner', 'data': [ '*foo': 'str' ] },
{ 'struct': 'Outer', 'data': [ '*bar': 'Inner' ] }
With the old parser you couldn't specify {}, while with
qobject_input_visitor_new_str, you can's specify { 'bar': {} } without
using JSON. I'm not sure which is the better, but with the old behavior
at least you could mark nested structs as required and not worry about
checking has_x every time.
Now that I wrote all this down, I'm starting to think that
qobject_input_visitor is good, and I should just properly check the
fields (or give them default values). Rubber duck debugging, I think.
Regards,
Zoltan
^ permalink raw reply [flat|nested] 91+ messages in thread* Re: [Qemu-devel] [PATCH v2 06/52] audio: -audiodev command line option basic implementation
2019-01-10 0:13 ` Zoltán Kővágó
@ 2019-01-10 7:25 ` Gerd Hoffmann
2019-01-10 9:40 ` Zoltán Kővágó
0 siblings, 1 reply; 91+ messages in thread
From: Gerd Hoffmann @ 2019-01-10 7:25 UTC (permalink / raw)
To: Zoltán Kővágó
Cc: Markus Armbruster, Paolo Bonzini, qemu-devel
Hi,
> I was thinking about creating an Audiodev (the qapi type) directly would
> be better, then somehow print it with reflection. While this is not a
> typical use of qapi, at least qmp_qom_list creates qapi objects
> directly, so I assume it's ok.
Yes, it's perfectly fine.
> The second problem: with my patched options visitor, nested structs were
> required in qapi, the options visitor unconditionally filled them. With
> qobject_input_visitor, I have to mark them optional, otherwise the user
> have to specify at least one option from the nested structs.
Can we just drop the nesting?
Have a look at DisplayOptions (qapi struct used to store -display
options). It's a union, has some common base fields, and the
discriminator field (type in that case) decides which of the data
branches is used. And on the command line I can do stuff like this:
-display gtk,full-screen=on,zoom-to-fit=on
^^^^^^^^^^^ in struct DisplayGTK
^^^^^^^^^^^ common base field
Audiodev should be able to do the same to support backend-specific
options without nesting.
cheers,
Gerd
^ permalink raw reply [flat|nested] 91+ messages in thread* Re: [Qemu-devel] [PATCH v2 06/52] audio: -audiodev command line option basic implementation
2019-01-10 7:25 ` Gerd Hoffmann
@ 2019-01-10 9:40 ` Zoltán Kővágó
2019-01-10 10:37 ` Gerd Hoffmann
0 siblings, 1 reply; 91+ messages in thread
From: Zoltán Kővágó @ 2019-01-10 9:40 UTC (permalink / raw)
To: Gerd Hoffmann; +Cc: Markus Armbruster, Paolo Bonzini, qemu-devel
On 2019-01-10 08:25, Gerd Hoffmann wrote:
> Hi,
>
>> I was thinking about creating an Audiodev (the qapi type) directly would
>> be better, then somehow print it with reflection. While this is not a
>> typical use of qapi, at least qmp_qom_list creates qapi objects
>> directly, so I assume it's ok.
>
> Yes, it's perfectly fine.
>
>> The second problem: with my patched options visitor, nested structs were
>> required in qapi, the options visitor unconditionally filled them. With
>> qobject_input_visitor, I have to mark them optional, otherwise the user
>> have to specify at least one option from the nested structs.
>
> Can we just drop the nesting?
>
> Have a look at DisplayOptions (qapi struct used to store -display
> options). It's a union, has some common base fields, and the
> discriminator field (type in that case) decides which of the data
> branches is used. And on the command line I can do stuff like this:
>
> -display gtk,full-screen=on,zoom-to-fit=on
> ^^^^^^^^^^^ in struct DisplayGTK
> ^^^^^^^^^^^ common base field
>
> Audiodev should be able to do the same to support backend-specific
> options without nesting.
I think that union thing works, the problem is with input and output
settings, because they're common. Currently you can write in.frequency
and out.frequency, in and out are both AudiodevPerDirectionOptions. The
alternative would be to flatten the whole structure and have
in_frequency, out_frequency and everything else duplicated in Audiodev
Regards,
Zoltan
^ permalink raw reply [flat|nested] 91+ messages in thread
* Re: [Qemu-devel] [PATCH v2 06/52] audio: -audiodev command line option basic implementation
2019-01-10 9:40 ` Zoltán Kővágó
@ 2019-01-10 10:37 ` Gerd Hoffmann
0 siblings, 0 replies; 91+ messages in thread
From: Gerd Hoffmann @ 2019-01-10 10:37 UTC (permalink / raw)
To: Zoltán Kővágó
Cc: Markus Armbruster, Paolo Bonzini, qemu-devel
> > Can we just drop the nesting?
> >
> > Have a look at DisplayOptions (qapi struct used to store -display
> > options). It's a union, has some common base fields, and the
> > discriminator field (type in that case) decides which of the data
> > branches is used. And on the command line I can do stuff like this:
> >
> > -display gtk,full-screen=on,zoom-to-fit=on
> > ^^^^^^^^^^^ in struct DisplayGTK
> > ^^^^^^^^^^^ common base field
> >
> > Audiodev should be able to do the same to support backend-specific
> > options without nesting.
>
> I think that union thing works, the problem is with input and output
> settings, because they're common. Currently you can write in.frequency
> and out.frequency, in and out are both AudiodevPerDirectionOptions.
Ah, right, I remember now.
> The alternative would be to flatten the whole structure and have
> in_frequency, out_frequency and everything else duplicated in Audiodev
Which might be not that bad after all. The parser code is generated.
The audio drivers (where the settings are looked up) have separate code
paths for in and out anyway. So a per-direction struct probably
wouldn't reduce the amout of code in the drivers. Driver-specific
per-direction options are easier to deal with too I think (not sure we
have them though).
Property names should use dashes not underscores btw (i.e.
in-frequency).
cheers,
Gerd
^ permalink raw reply [flat|nested] 91+ messages in thread
* Re: [Qemu-devel] [PATCH v2 06/52] audio: -audiodev command line option basic implementation
2019-01-07 20:48 ` Zoltán Kővágó
2019-01-08 3:42 ` Markus Armbruster
@ 2019-01-08 6:06 ` Gerd Hoffmann
1 sibling, 0 replies; 91+ messages in thread
From: Gerd Hoffmann @ 2019-01-08 6:06 UTC (permalink / raw)
To: Zoltán Kővágó
Cc: Markus Armbruster, qemu-devel, Paolo Bonzini
Hi,
> > I suspect your series uses the options visitor only because back when
> > you started, qobject_input_visitor_new_str() didn't exist.
>
> Yes, this patch series is a bit old, and at that time this was the best
> I could do. I can look into it this (probably only on the weekend
> though), it looks like it supports foo.bar=something syntax, so I don't
> have to specify a json on the command line...
It supports both, i.e. this ...
qemu -display gtk,full-screen=on
... and this ...
qemu -display '{ "type": "gtk", "full-screen": true }'
... has the same effect.
cheers,
Gerd
^ permalink raw reply [flat|nested] 91+ messages in thread
* [Qemu-devel] [PATCH v2 07/52] alsaaudio: port to -audiodev config
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (5 preceding siblings ...)
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 06/52] audio: -audiodev command line option basic implementation Kővágó, Zoltán
@ 2018-12-23 20:51 ` Kővágó, Zoltán
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 08/52] coreaudio: " Kővágó, Zoltán
` (43 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:51 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/alsaaudio.c | 314 ++++++++++++-------------------------------
audio/audio_legacy.c | 64 +++++++++
2 files changed, 152 insertions(+), 226 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 8302f3e882..6f75644538 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -23,6 +23,7 @@
*/
#include "qemu/osdep.h"
#include <alsa/asoundlib.h>
+#include "qapi/opts-visitor.h"
#include "qemu-common.h"
#include "qemu/main-loop.h"
#include "audio.h"
@@ -33,28 +34,9 @@
#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;
};
@@ -66,6 +48,7 @@ typedef struct ALSAVoiceOut {
void *pcm_buf;
snd_pcm_t *handle;
struct pollhlp pollhlp;
+ Audiodev *dev;
} ALSAVoiceOut;
typedef struct ALSAVoiceIn {
@@ -73,16 +56,13 @@ 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 {
@@ -408,7 +388,8 @@ 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);
@@ -416,8 +397,9 @@ 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 len %" PRId64 " buffer count %" PRId64 "\n",
+ pdo->has_buffer_len ? pdo->buffer_len : 0,
+ pdo->has_buffer_count ? pdo->buffer_count : 0);
dolog ("obtained: samples %ld\n", obt->samples);
}
@@ -451,23 +433,25 @@ static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
}
}
-static int alsa_open (int in, struct alsa_params_req *req,
- struct alsa_params_obt *obt, snd_pcm_t **handlep,
- ALSAConf *conf)
+static int alsa_open(bool in, struct alsa_params_req *req,
+ struct alsa_params_obt *obt, snd_pcm_t **handlep,
+ Audiodev *dev)
{
+ AudiodevPerDirectionOptions *pdo = in ? dev->in : dev->out;
+ AudiodevAlsaOptions *aopts = &dev->u.alsa;
+ AudiodevAlsaPerDirectionOptions *apdo =
+ in ? aopts->alsa_in : aopts->alsa_out;
snd_pcm_t *handle;
snd_pcm_hw_params_t *hw_params;
int err;
- int size_in_usec;
unsigned int freq, nchannels;
- const char *pcm_name = in ? conf->pcm_name_in : conf->pcm_name_out;
+ const char *pcm_name = apdo->has_dev ? apdo->dev : "default";
snd_pcm_uframes_t obt_buffer_size;
const char *typ = in ? "ADC" : "DAC";
snd_pcm_format_t obtfmt;
freq = req->freq;
nchannels = req->nchannels;
- size_in_usec = req->size_in_usec;
snd_pcm_hw_params_alloca (&hw_params);
@@ -527,79 +511,49 @@ static int alsa_open (int in, struct alsa_params_req *req,
goto err;
}
- if (req->buffer_size) {
- unsigned long obt;
+ if (pdo->buffer_count) {
+ if (pdo->buffer_len) {
+ int64_t req = pdo->buffer_len * pdo->buffer_count;
- if (size_in_usec) {
int dir = 0;
- unsigned int btime = req->buffer_size;
+ unsigned int btime = req;
- err = snd_pcm_hw_params_set_buffer_time_near (
- handle,
- hw_params,
- &btime,
- &dir
- );
- obt = btime;
- }
- else {
- snd_pcm_uframes_t bsize = req->buffer_size;
+ err = snd_pcm_hw_params_set_buffer_time_near(
+ handle, hw_params, &btime, &dir);
- err = snd_pcm_hw_params_set_buffer_size_near (
- handle,
- hw_params,
- &bsize
- );
- obt = bsize;
- }
- if (err < 0) {
- alsa_logerr2 (err, typ, "Failed to set buffer %s to %d\n",
- size_in_usec ? "time" : "size", req->buffer_size);
- goto err;
- }
+ if (err < 0) {
+ alsa_logerr2(err, typ,
+ "Failed to set buffer time to %" PRId64 "\n",
+ req);
+ goto err;
+ }
- if ((req->override_mask & 2) && (obt - req->buffer_size))
- dolog ("Requested buffer %s %u was rejected, using %lu\n",
- size_in_usec ? "time" : "size", req->buffer_size, obt);
+ if (pdo->has_buffer_count && btime != req) {
+ dolog("Requested buffer time %" PRId64
+ " was rejected, using %u\n", req, btime);
+ }
+ } else {
+ dolog("Can't set buffer-count without buffer-len!\n");
+ }
}
- if (req->period_size) {
- unsigned long obt;
+ if (pdo->buffer_len) {
+ int dir = 0;
+ unsigned int ptime = pdo->buffer_len;
- if (size_in_usec) {
- int dir = 0;
- unsigned int ptime = req->period_size;
-
- err = snd_pcm_hw_params_set_period_time_near (
- handle,
- hw_params,
- &ptime,
- &dir
- );
- obt = ptime;
- }
- else {
- int dir = 0;
- snd_pcm_uframes_t psize = req->period_size;
-
- err = snd_pcm_hw_params_set_period_size_near (
- handle,
- hw_params,
- &psize,
- &dir
- );
- obt = psize;
- }
+ err = snd_pcm_hw_params_set_period_time_near(handle, hw_params, &ptime,
+ &dir);
if (err < 0) {
- alsa_logerr2 (err, typ, "Failed to set period %s to %d\n",
- size_in_usec ? "time" : "size", req->period_size);
+ alsa_logerr2(err, typ, "Failed to set period time to %" PRId64 "\n",
+ pdo->buffer_len);
goto err;
}
- if (((req->override_mask & 1) && (obt - req->period_size)))
- dolog ("Requested period %s %u was rejected, using %lu\n",
- size_in_usec ? "time" : "size", req->period_size, obt);
+ if (pdo->has_buffer_len && ptime != pdo->buffer_len) {
+ dolog("Requested period time %" PRId64 " was rejected, using %d\n",
+ pdo->buffer_len, ptime);
+ }
}
err = snd_pcm_hw_params (handle, hw_params);
@@ -631,33 +585,10 @@ static int alsa_open (int in, struct alsa_params_req *req,
goto err;
}
- if (!in && conf->threshold) {
- snd_pcm_uframes_t threshold;
- int bytes_per_sec;
-
- bytes_per_sec = freq << (nchannels == 2);
-
- switch (obt->fmt) {
- case AUDIO_FORMAT_S8:
- case AUDIO_FORMAT_U8:
- break;
-
- case AUDIO_FORMAT_S16:
- case AUDIO_FORMAT_U16:
- bytes_per_sec <<= 1;
- break;
-
- case AUDIO_FORMAT_S32:
- case AUDIO_FORMAT_U32:
- bytes_per_sec <<= 2;
- break;
-
- default:
- abort();
- }
-
- threshold = (conf->threshold * bytes_per_sec) / 1000;
- alsa_set_threshold (handle, threshold);
+ if (!in && aopts->has_threshold && aopts->threshold) {
+ struct audsettings as = { .freq = freq };
+ alsa_set_threshold(handle,
+ audio_buffer_frames(pdo, &as, aopts->threshold));
}
obt->nchannels = nchannels;
@@ -670,11 +601,11 @@ 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;
@@ -800,19 +731,13 @@ 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;
}
@@ -833,7 +758,7 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
}
alsa->handle = handle;
- alsa->pollhlp.conf = conf;
+ alsa->dev = dev;
return 0;
}
@@ -873,16 +798,12 @@ 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->u.alsa.alsa_out;
switch (cmd) {
case VOICE_ENABLE:
{
- va_list ap;
- int poll_mode;
-
- va_start (ap, cmd);
- poll_mode = va_arg (ap, int);
- va_end (ap);
+ bool poll_mode = !apdo->has_try_poll || apdo->try_poll;
ldebug ("enabling voice\n");
if (poll_mode && alsa_poll_out (hw)) {
@@ -911,19 +832,13 @@ 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;
}
@@ -944,7 +859,7 @@ 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;
}
@@ -1086,16 +1001,12 @@ 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->u.alsa.alsa_in;
switch (cmd) {
case VOICE_ENABLE:
{
- va_list ap;
- int poll_mode;
-
- va_start (ap, cmd);
- poll_mode = va_arg (ap, int);
- va_end (ap);
+ bool poll_mode = !apdo->has_try_poll || apdo->try_poll;
ldebug ("enabling voice\n");
if (poll_mode && alsa_poll_in (hw)) {
@@ -1118,88 +1029,40 @@ 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(Audiodev *dev)
{
- ALSAConf *conf = g_malloc(sizeof(ALSAConf));
- *conf = glob_conf;
- return conf;
+ assert(dev->driver == AUDIODEV_DRIVER_ALSA);
+
+ /*
+ * need to define them, as otherwise alsa produces no sound
+ * doesn't set has_* so alsa_open can identify it wasn't set by the user
+ */
+ if (!dev->out->has_buffer_count) {
+ dev->out->buffer_count = 4;
+ }
+ if (!dev->out->has_buffer_len) {
+ /* 1024 frames assuming 44100Hz */
+ dev->out->buffer_len = 1024 * 1000000 / 44100;
+ }
+
+ /*
+ * OptsVisitor sets unspecified optional fields to zero, but do not depend
+ * on it...
+ */
+ if (!dev->in->has_buffer_count) {
+ dev->in->buffer_count = 0;
+ }
+ if (!dev->in->has_buffer_len) {
+ dev->in->buffer_len = 0;
+ }
+
+ return dev;
}
static void alsa_audio_fini (void *opaque)
{
- g_free(opaque);
}
-static struct audio_option alsa_options[] = {
- {
- .name = "DAC_SIZE_IN_USEC",
- .tag = AUD_OPT_BOOL,
- .valp = &glob_conf.size_in_usec_out,
- .descr = "DAC period/buffer size in microseconds (otherwise in frames)"
- },
- {
- .name = "DAC_PERIOD_SIZE",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.period_size_out,
- .descr = "DAC period size (0 to go with system default)",
- .overriddenp = &glob_conf.period_size_out_overridden
- },
- {
- .name = "DAC_BUFFER_SIZE",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.buffer_size_out,
- .descr = "DAC buffer size (0 to go with system default)",
- .overriddenp = &glob_conf.buffer_size_out_overridden
- },
- {
- .name = "ADC_SIZE_IN_USEC",
- .tag = AUD_OPT_BOOL,
- .valp = &glob_conf.size_in_usec_in,
- .descr =
- "ADC period/buffer size in microseconds (otherwise in frames)"
- },
- {
- .name = "ADC_PERIOD_SIZE",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.period_size_in,
- .descr = "ADC period size (0 to go with system default)",
- .overriddenp = &glob_conf.period_size_in_overridden
- },
- {
- .name = "ADC_BUFFER_SIZE",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.buffer_size_in,
- .descr = "ADC buffer size (0 to go with system default)",
- .overriddenp = &glob_conf.buffer_size_in_overridden
- },
- {
- .name = "THRESHOLD",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.threshold,
- .descr = "(undocumented)"
- },
- {
- .name = "DAC_DEV",
- .tag = AUD_OPT_STR,
- .valp = &glob_conf.pcm_name_out,
- .descr = "DAC device name (for instance dmix)"
- },
- {
- .name = "ADC_DEV",
- .tag = AUD_OPT_STR,
- .valp = &glob_conf.pcm_name_in,
- .descr = "ADC device name"
- },
- { /* End of list */ }
-};
-
static struct audio_pcm_ops alsa_pcm_ops = {
.init_out = alsa_init_out,
.fini_out = alsa_fini_out,
@@ -1217,7 +1080,6 @@ static struct audio_pcm_ops alsa_pcm_ops = {
static 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_legacy.c b/audio/audio_legacy.c
index 2242f06a5e..f91ce0c3e2 100644
--- a/audio/audio_legacy.c
+++ b/audio/audio_legacy.c
@@ -52,6 +52,17 @@ SimpleEnvMap global_map[] = {
{ /* End of list */ }
};
+SimpleEnvMap alsa_map[] = {
+ { "QEMU_AUDIO_DAC_TRY_POLL", "alsa-out.try-poll", ENV_TRANSFORM_BOOL },
+ { "QEMU_AUDIO_ADC_TRY_POLL", "alsa-in.try-poll", ENV_TRANSFORM_BOOL },
+
+ { "QEMU_ALSA_THRESHOLD", "threshold", ENV_TRANSFORM_MILLIS_TO_USECS },
+ { "QEMU_ALSA_DAC_DEV", "alsa-out.dev" },
+ { "QEMU_ALSA_ADC_DEV", "alsa-in.dev" },
+
+ { /* End of list */ }
+};
+
static unsigned long long toull(const char *str)
{
unsigned long long ret;
@@ -162,6 +173,54 @@ static void handle_env_opts(QemuOpts *opts, SimpleEnvMap *map)
}
}
+static void handle_alsa_side(QemuOpts *opts, int period, int buffer,
+ const char *usec_env, const char *period_env,
+ const char *buffer_env, const char *usec_opt,
+ const char *count_opt, bool in)
+{
+ char *usec_s, *period_s, *buffer_s;
+ bool usec = false;
+
+ usec_s = getenv(usec_env);
+ if (usec_s) {
+ usec = toull(usec_s);
+ }
+
+ period_s = getenv(period_env);
+ if (period_s) {
+ period = toull(period_s);
+ }
+ if (!usec) {
+ period = frames_to_usecs(opts, period, in);
+ }
+ if (period_s) {
+ qemu_opt_set(opts, usec_opt, tostr(period), &error_abort);
+ }
+
+ buffer_s = getenv(buffer_env);
+ if (buffer_s) {
+ buffer = toull(buffer_s);
+ if (!usec) {
+ buffer = frames_to_usecs(opts, buffer, in);
+ }
+ printf("buffer %d period %d\n", buffer, period);
+ qemu_opt_set(opts, count_opt, tostr((buffer + period / 2) / period),
+ &error_abort);
+ }
+}
+
+static void handle_alsa(QemuOpts *opts)
+{
+ handle_alsa_side(opts, 1024, 4096,
+ "QEMU_ALSA_DAC_SIZE_IN_USEC", "QEMU_ALSA_DAC_PERIOD_SIZE",
+ "QEMU_ALSA_DAC_BUFFER_SIZE",
+ "out.buffer-len", "out.buffer-count", false);
+ handle_alsa_side(opts, 0, 0,
+ "QEMU_ALSA_ADC_SIZE_IN_USEC", "QEMU_ALSA_ADC_PERIOD_SIZE",
+ "QEMU_ALSA_ADC_BUFFER_SIZE",
+ "in.buffer-len", "in.buffer-count", true);
+}
+
static void legacy_opt(const char *drv)
{
QemuOpts *opts;
@@ -170,6 +229,11 @@ static void legacy_opt(const char *drv)
qemu_opt_set(opts, "driver", drv, &error_abort);
handle_env_opts(opts, global_map);
+
+ if (strcmp(drv, "alsa") == 0) {
+ handle_env_opts(opts, alsa_map);
+ handle_alsa(opts);
+ }
}
void audio_handle_legacy_opts(void)
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 08/52] coreaudio: port to -audiodev config
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (6 preceding siblings ...)
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 07/52] alsaaudio: port to -audiodev config Kővágó, Zoltán
@ 2018-12-23 20:51 ` Kővágó, Zoltán
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 09/52] dsoundaudio: " Kővágó, Zoltán
` (42 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:51 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio_legacy.c | 10 ++++++++++
audio/coreaudio.c | 47 ++++++++++----------------------------------
2 files changed, 20 insertions(+), 37 deletions(-)
diff --git a/audio/audio_legacy.c b/audio/audio_legacy.c
index f91ce0c3e2..42b55035e3 100644
--- a/audio/audio_legacy.c
+++ b/audio/audio_legacy.c
@@ -63,6 +63,14 @@ SimpleEnvMap alsa_map[] = {
{ /* End of list */ }
};
+SimpleEnvMap coreaudio_map[] = {
+ { "QEMU_COREAUDIO_BUFFER_SIZE", "buffer-len",
+ ENV_TRANSFORM_FRAMES_TO_USECS_OUT },
+ { "QEMU_COREAUDIO_BUFFER_COUNT", "buffer-count" },
+
+ { /* End of list */ }
+};
+
static unsigned long long toull(const char *str)
{
unsigned long long ret;
@@ -233,6 +241,8 @@ static void legacy_opt(const char *drv)
if (strcmp(drv, "alsa") == 0) {
handle_env_opts(opts, alsa_map);
handle_alsa(opts);
+ } else if (strcmp(drv, "coreaudio") == 0) {
+ handle_env_opts(opts, coreaudio_map);
}
}
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index 7d4225dbee..e00b8847d7 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -36,11 +36,6 @@
#define MAC_OS_X_VERSION_10_6 1060
#endif
-typedef struct {
- int buffer_frames;
- int nbuffers;
-} CoreaudioConf;
-
typedef struct coreaudioVoiceOut {
HWVoiceOut hw;
pthread_mutex_t mutex;
@@ -507,7 +502,9 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
int err;
const char *typ = "playback";
AudioValueRange frameRange;
- CoreaudioConf *conf = drv_opaque;
+ Audiodev *dev = drv_opaque;
+ AudiodevPerDirectionOptions *pdo = dev->out;
+ int frames;
/* create mutex */
err = pthread_mutex_init(&core->mutex, NULL);
@@ -538,16 +535,17 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
return -1;
}
- if (frameRange.mMinimum > conf->buffer_frames) {
+ frames = audio_buffer_frames(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 */
@@ -568,7 +566,8 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
"Could not get device buffer frame size\n");
return -1;
}
- hw->samples = conf->nbuffers * core->audioDevicePropertyBufferFrameSize;
+ hw->samples = (pdo->has_buffer_count ? pdo->buffer_count : 4) *
+ core->audioDevicePropertyBufferFrameSize;
/* get StreamFormat */
status = coreaudio_get_streamformat(core->outputDeviceID,
@@ -680,40 +679,15 @@ static int coreaudio_ctl_out (HWVoiceOut *hw, int cmd, ...)
return 0;
}
-static CoreaudioConf glob_conf = {
- .buffer_frames = 512,
- .nbuffers = 4,
-};
-
static void *coreaudio_audio_init(Audiodev *dev)
{
- CoreaudioConf *conf = g_malloc(sizeof(CoreaudioConf));
- *conf = glob_conf;
-
- return conf;
+ return dev;
}
static void coreaudio_audio_fini (void *opaque)
{
- g_free(opaque);
}
-static struct audio_option coreaudio_options[] = {
- {
- .name = "BUFFER_SIZE",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.buffer_frames,
- .descr = "Size of the buffer in frames"
- },
- {
- .name = "BUFFER_COUNT",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.nbuffers,
- .descr = "Number of buffers"
- },
- { /* End of list */ }
-};
-
static struct audio_pcm_ops coreaudio_pcm_ops = {
.init_out = coreaudio_init_out,
.fini_out = coreaudio_fini_out,
@@ -725,7 +699,6 @@ static struct audio_pcm_ops coreaudio_pcm_ops = {
static struct audio_driver coreaudio_audio_driver = {
.name = "coreaudio",
.descr = "CoreAudio http://developer.apple.com/audio/coreaudio.html",
- .options = coreaudio_options,
.init = coreaudio_audio_init,
.fini = coreaudio_audio_fini,
.pcm_ops = &coreaudio_pcm_ops,
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 09/52] dsoundaudio: port to -audiodev config
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (7 preceding siblings ...)
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 08/52] coreaudio: " Kővágó, Zoltán
@ 2018-12-23 20:51 ` Kővágó, Zoltán
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 10/52] noaudio: " Kővágó, Zoltán
` (41 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:51 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio_legacy.c | 12 +++++++++
audio/dsound_template.h | 6 ++---
audio/dsoundaudio.c | 59 ++++++++++++-----------------------------
3 files changed, 32 insertions(+), 45 deletions(-)
diff --git a/audio/audio_legacy.c b/audio/audio_legacy.c
index 42b55035e3..48a22c390a 100644
--- a/audio/audio_legacy.c
+++ b/audio/audio_legacy.c
@@ -71,6 +71,16 @@ SimpleEnvMap coreaudio_map[] = {
{ /* End of list */ }
};
+SimpleEnvMap dsound_map[] = {
+ { "QEMU_DSOUND_LATENCY_MILLIS", "latency", ENV_TRANSFORM_MILLIS_TO_USECS },
+ { "QEMU_DSOUND_BUFSIZE_OUT", "out.buffer-len",
+ ENV_TRANSFORM_BYTES_TO_USECS_OUT },
+ { "QEMU_DSOUND_BUFSIZE_IN", "in.buffer-len",
+ ENV_TRANSFORM_BYTES_TO_USECS_IN },
+
+ { /* End of list */ }
+};
+
static unsigned long long toull(const char *str)
{
unsigned long long ret;
@@ -243,6 +253,8 @@ static void legacy_opt(const char *drv)
handle_alsa(opts);
} else if (strcmp(drv, "coreaudio") == 0) {
handle_env_opts(opts, coreaudio_map);
+ } else if (strcmp(drv, "dsound") == 0) {
+ handle_env_opts(opts, dsound_map);
}
}
diff --git a/audio/dsound_template.h b/audio/dsound_template.h
index b439f33f58..96181efb36 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 02fe777cba..a7d04b5033 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -32,6 +32,7 @@
#define AUDIO_CAP "dsound"
#include "audio_int.h"
+#include "qemu/host-utils.h"
#include <windows.h>
#include <mmsystem.h>
@@ -42,17 +43,11 @@
/* #define DEBUG_DSOUND */
-typedef struct {
- int bufsize_in;
- int bufsize_out;
- int latency_millis;
-} DSoundConf;
-
typedef struct {
LPDIRECTSOUND dsound;
LPDIRECTSOUNDCAPTURE dsound_capture;
struct audsettings settings;
- DSoundConf conf;
+ Audiodev *dev;
} dsound;
typedef struct {
@@ -248,9 +243,9 @@ static void GCC_FMT_ATTR (3, 4) dsound_logerr2 (
dsound_log_hresult (hr);
}
-static DWORD millis_to_bytes (struct audio_pcm_info *info, DWORD millis)
+static uint64_t usecs_to_bytes(struct audio_pcm_info *info, uint32_t usecs)
{
- return (millis * info->bytes_per_second) / 1000;
+ return muldiv64(usecs, info->bytes_per_second, 1000000);
}
#ifdef DEBUG_DSOUND
@@ -478,7 +473,7 @@ static int dsound_run_out (HWVoiceOut *hw, int live)
LPVOID p1, p2;
int bufsize;
dsound *s = ds->s;
- DSoundConf *conf = &s->conf;
+ AudiodevDsoundOptions *dso = &s->dev->u.dsound;
if (!dsb) {
dolog ("Attempt to run empty with playback buffer\n");
@@ -501,14 +496,14 @@ static int dsound_run_out (HWVoiceOut *hw, int live)
len = live << hwshift;
if (ds->first_time) {
- if (conf->latency_millis) {
+ if (dso->latency) {
DWORD cur_blat;
cur_blat = audio_ring_dist (wpos, ppos, bufsize);
ds->first_time = 0;
old_pos = wpos;
old_pos +=
- millis_to_bytes (&hw->info, conf->latency_millis) - cur_blat;
+ usecs_to_bytes(&hw->info, dso->latency) - cur_blat;
old_pos %= bufsize;
old_pos &= ~hw->info.align;
}
@@ -747,12 +742,6 @@ static int dsound_run_in (HWVoiceIn *hw)
return decr;
}
-static DSoundConf glob_conf = {
- .bufsize_in = 16384,
- .bufsize_out = 16384,
- .latency_millis = 10
-};
-
static void dsound_audio_fini (void *opaque)
{
HRESULT hr;
@@ -788,8 +777,17 @@ static void *dsound_audio_init(Audiodev *dev)
int err;
HRESULT hr;
dsound *s = g_malloc0(sizeof(dsound));
+ AudiodevDsoundOptions *dso;
+
+ assert(dev->driver == AUDIODEV_DRIVER_DSOUND);
+ s->dev = dev;
+ dso = &dev->u.dsound;
+
+ if (!dso->has_latency) {
+ dso->has_latency = true;
+ dso->latency = 10000; /* 10 ms */
+ }
- s->conf = glob_conf;
hr = CoInitialize (NULL);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not initialize COM\n");
@@ -854,28 +852,6 @@ static void *dsound_audio_init(Audiodev *dev)
return s;
}
-static struct audio_option dsound_options[] = {
- {
- .name = "LATENCY_MILLIS",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.latency_millis,
- .descr = "(undocumented)"
- },
- {
- .name = "BUFSIZE_OUT",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.bufsize_out,
- .descr = "(undocumented)"
- },
- {
- .name = "BUFSIZE_IN",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.bufsize_in,
- .descr = "(undocumented)"
- },
- { /* End of list */ }
-};
-
static struct audio_pcm_ops dsound_pcm_ops = {
.init_out = dsound_init_out,
.fini_out = dsound_fini_out,
@@ -893,7 +869,6 @@ static struct audio_pcm_ops dsound_pcm_ops = {
static struct audio_driver dsound_audio_driver = {
.name = "dsound",
.descr = "DirectSound http://wikipedia.org/wiki/DirectSound",
- .options = dsound_options,
.init = dsound_audio_init,
.fini = dsound_audio_fini,
.pcm_ops = &dsound_pcm_ops,
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 10/52] noaudio: port to -audiodev config
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (8 preceding siblings ...)
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 09/52] dsoundaudio: " Kővágó, Zoltán
@ 2018-12-23 20:51 ` Kővágó, Zoltán
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 11/52] ossaudio: " Kővágó, Zoltán
` (40 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:51 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/noaudio.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/audio/noaudio.c b/audio/noaudio.c
index 2cc274c5e5..f1eb048d80 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -163,7 +163,6 @@ static struct audio_pcm_ops no_pcm_ops = {
static struct audio_driver no_audio_driver = {
.name = "none",
.descr = "Timer based audio emulation",
- .options = NULL,
.init = no_audio_init,
.fini = no_audio_fini,
.pcm_ops = &no_pcm_ops,
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 11/52] ossaudio: port to -audiodev config
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (9 preceding siblings ...)
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 10/52] noaudio: " Kővágó, Zoltán
@ 2018-12-23 20:51 ` Kővágó, Zoltán
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 12/52] paaudio: " Kővágó, Zoltán
` (39 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:51 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio_legacy.c | 17 +++++
audio/ossaudio.c | 153 ++++++++++++-------------------------------
2 files changed, 59 insertions(+), 111 deletions(-)
diff --git a/audio/audio_legacy.c b/audio/audio_legacy.c
index 48a22c390a..f37407586b 100644
--- a/audio/audio_legacy.c
+++ b/audio/audio_legacy.c
@@ -81,6 +81,21 @@ SimpleEnvMap dsound_map[] = {
{ /* End of list */ }
};
+SimpleEnvMap oss_map[] = {
+ { "QEMU_AUDIO_DAC_TRY_POLL", "oss-out.try-poll", ENV_TRANSFORM_BOOL },
+ { "QEMU_AUDIO_ADC_TRY_POLL", "oss-in.try-poll", ENV_TRANSFORM_BOOL },
+
+ { "QEMU_OSS_FRAGSIZE", "buffer-len", ENV_TRANSFORM_BYTES_TO_USECS_OUT },
+ { "QEMU_OSS_NFRAGS", "buffer-count" },
+ { "QEMU_OSS_MMAP", "try-mmap", ENV_TRANSFORM_BOOL },
+ { "QEMU_OSS_DAC_DEV", "oss-out.dev" },
+ { "QEMU_OSS_ADC_DEV", "oss-in.dev" },
+ { "QEMU_OSS_EXCLUSIVE", "exclusive", ENV_TRANSFORM_BOOL },
+ { "QEMU_OSS_POLICY", "dsp-policy" },
+
+ { /* End of list */ }
+};
+
static unsigned long long toull(const char *str)
{
unsigned long long ret;
@@ -255,6 +270,8 @@ static void legacy_opt(const char *drv)
handle_env_opts(opts, coreaudio_map);
} else if (strcmp(drv, "dsound") == 0) {
handle_env_opts(opts, dsound_map);
+ } else if (strcmp(drv, "oss") == 0) {
+ handle_env_opts(opts, oss_map);
}
}
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index e0cadbef29..83ff1d2ebd 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -27,6 +27,7 @@
#include "qemu-common.h"
#include "qemu/main-loop.h"
#include "qemu/host-utils.h"
+#include "qapi/opts-visitor.h"
#include "audio.h"
#include "trace.h"
@@ -37,16 +38,6 @@
#define USE_DSP_POLICY
#endif
-typedef struct OSSConf {
- int try_mmap;
- int nfrags;
- int fragsize;
- const char *devpath_out;
- const char *devpath_in;
- int exclusive;
- int policy;
-} OSSConf;
-
typedef struct OSSVoiceOut {
HWVoiceOut hw;
void *pcm_buf;
@@ -56,7 +47,7 @@ typedef struct OSSVoiceOut {
int fragsize;
int mmapped;
int pending;
- OSSConf *conf;
+ Audiodev *dev;
} OSSVoiceOut;
typedef struct OSSVoiceIn {
@@ -65,12 +56,12 @@ 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;
@@ -262,19 +253,26 @@ 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->u.oss;
+ AudiodevOssPerDirectionOptions *opdo = in ? oopts->oss_in : oopts->oss_out;
+ AudiodevPerDirectionOptions *pdo = in ? dev->in : dev->out;
int fd;
- int oflags = conf->exclusive ? O_EXCL : 0;
+ int oflags = (oopts->has_exclusive && oopts->exclusive) ? O_EXCL : 0;
audio_buf_info abinfo;
int fmt, freq, nchannels;
int setfragment = 1;
- const char *dspname = in ? conf->devpath_in : conf->devpath_out;
+ const char *dspname = opdo->has_dev ? opdo->dev : "/dev/dsp";
const char *typ = in ? "ADC" : "DAC";
+#ifdef USE_DSP_POLICY
+ int policy = oopts->has_dsp_policy ? oopts->dsp_policy : 5;
+#endif
/* Kludge needed to have working mmap on Linux */
- oflags |= conf->try_mmap ? O_RDWR : (in ? O_RDONLY : O_WRONLY);
+ oflags |= (oopts->has_try_mmap && oopts->try_mmap) ?
+ O_RDWR : (in ? O_RDONLY : O_WRONLY);
fd = open (dspname, oflags | O_NONBLOCK);
if (-1 == fd) {
@@ -285,6 +283,8 @@ 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);
@@ -308,18 +308,18 @@ 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;
@@ -502,17 +502,16 @@ 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->u.oss;
oss->fd = -1;
req.fmt = aud_to_ossfmt (as->fmt, as->endianness);
req.freq = as->freq;
req.nchannels = as->nchannels;
- req.fragsize = conf->fragsize;
- req.nfrags = conf->nfrags;
- if (oss_open (0, &req, &obt, &fd, conf)) {
+ if (oss_open(0, &req, as, &obt, &fd, dev)) {
return -1;
}
@@ -539,7 +538,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
oss->mmapped = 0;
- if (conf->try_mmap) {
+ if (oopts->has_try_mmap && oopts->try_mmap) {
oss->pcm_buf = mmap (
NULL,
hw->samples << hw->info.shift,
@@ -597,7 +596,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
}
oss->fd = fd;
- oss->conf = conf;
+ oss->dev = dev;
return 0;
}
@@ -605,16 +604,12 @@ static int oss_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
int trig;
OSSVoiceOut *oss = (OSSVoiceOut *) hw;
+ AudiodevOssPerDirectionOptions *opdo = oss->dev->u.oss.oss_out;
switch (cmd) {
case VOICE_ENABLE:
{
- va_list ap;
- int poll_mode;
-
- va_start (ap, cmd);
- poll_mode = va_arg (ap, int);
- va_end (ap);
+ bool poll_mode = !opdo->has_try_poll || opdo->try_poll;
ldebug ("enabling voice\n");
if (poll_mode) {
@@ -669,16 +664,14 @@ 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;
}
@@ -712,7 +705,7 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
}
oss->fd = fd;
- oss->conf = conf;
+ oss->dev = dev;
return 0;
}
@@ -803,16 +796,12 @@ static int oss_read (SWVoiceIn *sw, void *buf, int size)
static int oss_ctl_in (HWVoiceIn *hw, int cmd, ...)
{
OSSVoiceIn *oss = (OSSVoiceIn *) hw;
+ AudiodevOssPerDirectionOptions *opdo = oss->dev->u.oss.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);
@@ -832,82 +821,25 @@ 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(Audiodev *dev)
{
- OSSConf *conf = g_malloc(sizeof(OSSConf));
- *conf = glob_conf;
+ AudiodevOssOptions *oopts;
+ assert(dev->driver == AUDIODEV_DRIVER_OSS);
- if (access(conf->devpath_in, R_OK | W_OK) < 0 ||
- access(conf->devpath_out, R_OK | W_OK) < 0) {
- g_free(conf);
+ oopts = &dev->u.oss;
+ if (access(oopts->oss_in->has_dev ? oopts->oss_in->dev : "/dev/dsp",
+ R_OK | W_OK) < 0 ||
+ access(oopts->oss_out->has_dev ? oopts->oss_out->dev : "/dev/dsp",
+ R_OK | W_OK) < 0) {
return NULL;
}
- return conf;
+ return dev;
}
static void oss_audio_fini (void *opaque)
{
- g_free(opaque);
}
-static struct audio_option oss_options[] = {
- {
- .name = "FRAGSIZE",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.fragsize,
- .descr = "Fragment size in bytes"
- },
- {
- .name = "NFRAGS",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.nfrags,
- .descr = "Number of fragments"
- },
- {
- .name = "MMAP",
- .tag = AUD_OPT_BOOL,
- .valp = &glob_conf.try_mmap,
- .descr = "Try using memory mapped access"
- },
- {
- .name = "DAC_DEV",
- .tag = AUD_OPT_STR,
- .valp = &glob_conf.devpath_out,
- .descr = "Path to DAC device"
- },
- {
- .name = "ADC_DEV",
- .tag = AUD_OPT_STR,
- .valp = &glob_conf.devpath_in,
- .descr = "Path to ADC device"
- },
- {
- .name = "EXCLUSIVE",
- .tag = AUD_OPT_BOOL,
- .valp = &glob_conf.exclusive,
- .descr = "Open device in exclusive mode (vmix won't work)"
- },
-#ifdef USE_DSP_POLICY
- {
- .name = "POLICY",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.policy,
- .descr = "Set the timing policy of the device, -1 to use fragment mode",
- },
-#endif
- { /* End of list */ }
-};
-
static struct audio_pcm_ops oss_pcm_ops = {
.init_out = oss_init_out,
.fini_out = oss_fini_out,
@@ -925,7 +857,6 @@ static struct audio_pcm_ops oss_pcm_ops = {
static struct audio_driver oss_audio_driver = {
.name = "oss",
.descr = "OSS http://www.opensound.com",
- .options = oss_options,
.init = oss_audio_init,
.fini = oss_audio_fini,
.pcm_ops = &oss_pcm_ops,
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 12/52] paaudio: port to -audiodev config
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (10 preceding siblings ...)
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 11/52] ossaudio: " Kővágó, Zoltán
@ 2018-12-23 20:51 ` Kővágó, Zoltán
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 13/52] sdlaudio: " Kővágó, Zoltán
` (38 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:51 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio_legacy.c | 11 ++++++
audio/paaudio.c | 80 ++++++++++++++++----------------------------
2 files changed, 40 insertions(+), 51 deletions(-)
diff --git a/audio/audio_legacy.c b/audio/audio_legacy.c
index f37407586b..1f99d21803 100644
--- a/audio/audio_legacy.c
+++ b/audio/audio_legacy.c
@@ -96,6 +96,15 @@ SimpleEnvMap oss_map[] = {
{ /* End of list */ }
};
+SimpleEnvMap pa_map[] = {
+ { "QEMU_PA_SAMPLES", "buffer", ENV_TRANSFORM_SAMPLES_TO_USECS_OUT },
+ { "QEMU_PA_SERVER", "server" },
+ { "QEMU_PA_SINK", "sink.name" },
+ { "QEMU_PA_SOURCE", "source.name" },
+
+ { /* End of list */ }
+};
+
static unsigned long long toull(const char *str)
{
unsigned long long ret;
@@ -272,6 +281,8 @@ static void legacy_opt(const char *drv)
handle_env_opts(opts, dsound_map);
} else if (strcmp(drv, "oss") == 0) {
handle_env_opts(opts, oss_map);
+ } else if (strcmp(drv, "pa") == 0) {
+ handle_env_opts(opts, pa_map);
}
}
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 0981f010c9..6de6ed97fc 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -2,6 +2,7 @@
#include "qemu/osdep.h"
#include "qemu-common.h"
#include "audio.h"
+#include "qapi/opts-visitor.h"
#include <pulse/pulseaudio.h>
@@ -10,14 +11,7 @@
#include "audio_pt_int.h"
typedef struct {
- int samples;
- char *server;
- char *sink;
- char *source;
-} PAConf;
-
-typedef struct {
- PAConf conf;
+ Audiodev *dev;
pa_threaded_mainloop *mainloop;
pa_context *context;
} paaudio;
@@ -32,6 +26,7 @@ typedef struct {
void *pcm_buf;
struct audio_pt pt;
paaudio *g;
+ int samples;
} PAVoiceOut;
typedef struct {
@@ -46,6 +41,7 @@ typedef struct {
const void *read_data;
size_t read_index, read_length;
paaudio *g;
+ int samples;
} PAVoiceIn;
static void qpa_audio_fini(void *opaque);
@@ -227,7 +223,7 @@ static void *qpa_thread_out (void *arg)
}
}
- decr = to_mix = audio_MIN(pa->live, pa->g->conf.samples >> 5);
+ decr = to_mix = audio_MIN(pa->live, pa->samples >> 5);
rpos = pa->rpos;
if (audio_pt_unlock(&pa->pt, __func__)) {
@@ -319,7 +315,7 @@ static void *qpa_thread_in (void *arg)
}
}
- incr = to_grab = audio_MIN(pa->dead, pa->g->conf.samples >> 5);
+ incr = to_grab = audio_MIN(pa->dead, pa->samples >> 5);
wpos = pa->wpos;
if (audio_pt_unlock(&pa->pt, __func__)) {
@@ -546,6 +542,8 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
struct audsettings obt_as = *as;
PAVoiceOut *pa = (PAVoiceOut *) hw;
paaudio *g = pa->g = drv_opaque;
+ AudiodevPaOptions *popts = &g->dev->u.pa;
+ AudiodevPaPerDirectionOptions *ppdo = popts->sink;
ss.format = audfmt_to_pa (as->fmt, as->endianness);
ss.channels = as->nchannels;
@@ -566,7 +564,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
g,
"qemu",
PA_STREAM_PLAYBACK,
- g->conf.sink,
+ ppdo->has_name ? ppdo->name : NULL,
&ss,
NULL, /* channel map */
&ba, /* buffering attributes */
@@ -578,7 +576,8 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
}
audio_pcm_init_info (&hw->info, &obt_as);
- hw->samples = g->conf.samples;
+ hw->samples = pa->samples = audio_buffer_samples(g->dev->out, &obt_as,
+ 46440);
pa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
pa->rpos = hw->rpos;
if (!pa->pcm_buf) {
@@ -612,6 +611,8 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
struct audsettings obt_as = *as;
PAVoiceIn *pa = (PAVoiceIn *) hw;
paaudio *g = pa->g = drv_opaque;
+ AudiodevPaOptions *popts = &g->dev->u.pa;
+ AudiodevPaPerDirectionOptions *ppdo = popts->source;
ss.format = audfmt_to_pa (as->fmt, as->endianness);
ss.channels = as->nchannels;
@@ -623,7 +624,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
g,
"qemu",
PA_STREAM_RECORD,
- g->conf.source,
+ ppdo->has_name ? ppdo->name : NULL,
&ss,
NULL, /* channel map */
NULL, /* buffering attributes */
@@ -635,7 +636,8 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
}
audio_pcm_init_info (&hw->info, &obt_as);
- hw->samples = g->conf.samples;
+ hw->samples = pa->samples = audio_buffer_samples(g->dev->in, &obt_as,
+ 46440);
pa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
pa->wpos = hw->wpos;
if (!pa->pcm_buf) {
@@ -808,14 +810,19 @@ static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...)
}
/* common */
-static PAConf glob_conf = {
- .samples = 4096,
-};
-
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->driver == AUDIODEV_DRIVER_PA);
+
+ g = g_malloc(sizeof(paaudio));
+ popts = &dev->u.pa;
+ server = popts->has_server ? popts->server : NULL;
+
+ g->dev = dev;
g->mainloop = NULL;
g->context = NULL;
@@ -825,14 +832,14 @@ static void *qpa_audio_init(Audiodev *dev)
}
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 +902,6 @@ 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 +919,6 @@ static struct audio_pcm_ops qpa_pcm_ops = {
static struct audio_driver pa_audio_driver = {
.name = "pa",
.descr = "http://www.pulseaudio.org/",
- .options = qpa_options,
.init = qpa_audio_init,
.fini = qpa_audio_fini,
.pcm_ops = &qpa_pcm_ops,
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 13/52] sdlaudio: port to -audiodev config
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (11 preceding siblings ...)
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 12/52] paaudio: " Kővágó, Zoltán
@ 2018-12-23 20:51 ` Kővágó, Zoltán
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 14/52] spiceaudio: " Kővágó, Zoltán
` (37 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:51 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio_legacy.c | 7 +++++++
audio/sdlaudio.c | 22 ++++------------------
2 files changed, 11 insertions(+), 18 deletions(-)
diff --git a/audio/audio_legacy.c b/audio/audio_legacy.c
index 1f99d21803..6d95b761b8 100644
--- a/audio/audio_legacy.c
+++ b/audio/audio_legacy.c
@@ -105,6 +105,11 @@ SimpleEnvMap pa_map[] = {
{ /* End of list */ }
};
+SimpleEnvMap sdl_map[] = {
+ { "QEMU_SDL_SAMPLES", "buffer-len", ENV_TRANSFORM_SAMPLES_TO_USECS_OUT },
+ { /* End of list */ }
+};
+
static unsigned long long toull(const char *str)
{
unsigned long long ret;
@@ -283,6 +288,8 @@ static void legacy_opt(const char *drv)
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);
}
}
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index 097841fde1..cf6ac19927 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -49,12 +49,6 @@ typedef struct SDLVoiceOut {
int decr;
} SDLVoiceOut;
-static struct {
- int nb_samples;
-} conf = {
- .nb_samples = 1024
-};
-
static struct SDLAudioState {
int exit;
#if USE_SEMAPHORE
@@ -63,6 +57,7 @@ static struct SDLAudioState {
#endif
int initialized;
bool driver_created;
+ Audiodev *dev;
} glob_sdl;
typedef struct SDLAudioState SDLAudioState;
@@ -392,7 +387,7 @@ 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;
@@ -467,6 +462,7 @@ static void *sdl_audio_init(Audiodev *dev)
#endif
s->driver_created = true;
+ s->dev = dev;
return s;
}
@@ -480,18 +476,9 @@ static void sdl_audio_fini (void *opaque)
#endif
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,
@@ -503,7 +490,6 @@ static struct audio_pcm_ops sdl_pcm_ops = {
static struct audio_driver sdl_audio_driver = {
.name = "sdl",
.descr = "SDL http://www.libsdl.org",
- .options = sdl_options,
.init = sdl_audio_init,
.fini = sdl_audio_fini,
.pcm_ops = &sdl_pcm_ops,
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 14/52] spiceaudio: port to -audiodev config
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (12 preceding siblings ...)
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 13/52] sdlaudio: " Kővágó, Zoltán
@ 2018-12-23 20:51 ` Kővágó, Zoltán
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 15/52] wavaudio: " Kővágó, Zoltán
` (36 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:51 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/spiceaudio.c | 5 -----
1 file changed, 5 deletions(-)
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index affc3df17f..4f7873af5a 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -373,10 +373,6 @@ static int line_in_ctl (HWVoiceIn *hw, int cmd, ...)
return 0;
}
-static struct audio_option audio_options[] = {
- { /* end of list */ },
-};
-
static struct audio_pcm_ops audio_callbacks = {
.init_out = line_out_init,
.fini_out = line_out_fini,
@@ -394,7 +390,6 @@ static struct audio_pcm_ops audio_callbacks = {
static struct audio_driver spice_audio_driver = {
.name = "spice",
.descr = "spice audio driver",
- .options = audio_options,
.init = spice_audio_init,
.fini = spice_audio_fini,
.pcm_ops = &audio_callbacks,
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 15/52] wavaudio: port to -audiodev config
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (13 preceding siblings ...)
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 14/52] spiceaudio: " Kővágó, Zoltán
@ 2018-12-23 20:51 ` Kővágó, Zoltán
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 16/52] audio: -audiodev command line option: cleanup Kővágó, Zoltán
` (35 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:51 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio_legacy.c | 10 ++++++++
audio/wavaudio.c | 58 +++++++-------------------------------------
2 files changed, 19 insertions(+), 49 deletions(-)
diff --git a/audio/audio_legacy.c b/audio/audio_legacy.c
index 6d95b761b8..71bbda04e2 100644
--- a/audio/audio_legacy.c
+++ b/audio/audio_legacy.c
@@ -110,6 +110,14 @@ SimpleEnvMap sdl_map[] = {
{ /* 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;
@@ -290,6 +298,8 @@ static void legacy_opt(const char *drv)
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);
}
}
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 9eff3555b3..214e30ccd9 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -24,6 +24,7 @@
#include "qemu/osdep.h"
#include "qemu/host-utils.h"
#include "qemu/timer.h"
+#include "qapi/opts-visitor.h"
#include "audio.h"
#define AUDIO_CAP "wav"
@@ -37,11 +38,6 @@ typedef struct WAVVoiceOut {
int total_samples;
} WAVVoiceOut;
-typedef struct {
- struct audsettings settings;
- const char *wav_path;
-} WAVConf;
-
static int wav_run_out (HWVoiceOut *hw, int live)
{
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
@@ -112,8 +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->u.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) {
@@ -154,10 +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;
@@ -225,54 +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(Audiodev *dev)
{
- WAVConf *conf = g_malloc(sizeof(WAVConf));
- *conf = glob_conf;
- return conf;
+ assert(dev->driver == AUDIODEV_DRIVER_WAV);
+ return dev;
}
static void wav_audio_fini (void *opaque)
{
ldebug ("wav_fini");
- g_free(opaque);
}
-static struct audio_option wav_options[] = {
- {
- .name = "FREQUENCY",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.settings.freq,
- .descr = "Frequency"
- },
- {
- .name = "FORMAT",
- .tag = AUD_OPT_FMT,
- .valp = &glob_conf.settings.fmt,
- .descr = "Format"
- },
- {
- .name = "DAC_FIXED_CHANNELS",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.settings.nchannels,
- .descr = "Number of channels (1 - mono, 2 - stereo)"
- },
- {
- .name = "PATH",
- .tag = AUD_OPT_STR,
- .valp = &glob_conf.wav_path,
- .descr = "Path to wave file"
- },
- { /* End of list */ }
-};
-
static struct audio_pcm_ops wav_pcm_ops = {
.init_out = wav_init_out,
.fini_out = wav_fini_out,
@@ -284,7 +245,6 @@ static struct audio_pcm_ops wav_pcm_ops = {
static struct audio_driver wav_audio_driver = {
.name = "wav",
.descr = "WAV renderer http://wikipedia.org/wiki/WAV",
- .options = wav_options,
.init = wav_audio_init,
.fini = wav_audio_fini,
.pcm_ops = &wav_pcm_ops,
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 16/52] audio: -audiodev command line option: cleanup
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (14 preceding siblings ...)
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 15/52] wavaudio: " Kővágó, Zoltán
@ 2018-12-23 20:51 ` Kővágó, Zoltán
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 17/52] audio: reduce glob_audio_state usage Kővágó, Zoltán
` (34 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:51 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Remove no longer needed code.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio.c | 201 +---------------------------------------------
audio/audio_int.h | 17 ----
2 files changed, 4 insertions(+), 214 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index e7f25ea84b..0c3f0efd46 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -173,113 +173,6 @@ void *audio_calloc (const char *funcname, int nmemb, size_t size)
return g_malloc0 (len);
}
-static const char *audio_audfmt_to_string (AudioFormat fmt)
-{
- switch (fmt) {
- case AUDIO_FORMAT_U8:
- return "U8";
-
- case AUDIO_FORMAT_U16:
- return "U16";
-
- case AUDIO_FORMAT_S8:
- return "S8";
-
- case AUDIO_FORMAT_S16:
- return "S16";
-
- case AUDIO_FORMAT_U32:
- return "U32";
-
- case AUDIO_FORMAT_S32:
- return "S32";
-
- default:
- abort();
- }
-
- dolog ("Bogus audfmt %d returning S16\n", fmt);
- return "S16";
-}
-
-static AudioFormat audio_string_to_audfmt (const char *s, AudioFormat defval,
- int *defaultp)
-{
- if (!strcasecmp (s, "u8")) {
- *defaultp = 0;
- return AUDIO_FORMAT_U8;
- }
- else if (!strcasecmp (s, "u16")) {
- *defaultp = 0;
- return AUDIO_FORMAT_U16;
- }
- else if (!strcasecmp (s, "u32")) {
- *defaultp = 0;
- return AUDIO_FORMAT_U32;
- }
- else if (!strcasecmp (s, "s8")) {
- *defaultp = 0;
- return AUDIO_FORMAT_S8;
- }
- else if (!strcasecmp (s, "s16")) {
- *defaultp = 0;
- return AUDIO_FORMAT_S16;
- }
- else if (!strcasecmp (s, "s32")) {
- *defaultp = 0;
- return AUDIO_FORMAT_S32;
- }
- else {
- dolog ("Bogus audio format `%s' using %s\n",
- s, audio_audfmt_to_string (defval));
- *defaultp = 1;
- return defval;
- }
-}
-
-static AudioFormat audio_get_conf_fmt (const char *envname,
- AudioFormat defval,
- int *defaultp)
-{
- const char *var = getenv (envname);
- if (!var) {
- *defaultp = 1;
- return defval;
- }
- return audio_string_to_audfmt (var, defval, defaultp);
-}
-
-static int audio_get_conf_int (const char *key, int defval, int *defaultp)
-{
- int val;
- char *strval;
-
- strval = getenv (key);
- if (strval && !qemu_strtoi(strval, NULL, 10, &val)) {
- *defaultp = 0;
- 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) {
@@ -298,89 +191,6 @@ void AUD_log (const char *cap, const char *fmt, ...)
va_end (ap);
}
-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(__func__, !prefix)) {
- dolog ("prefix = NULL\n");
- return;
- }
-
- if (audio_bug(__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);
@@ -1093,7 +903,7 @@ 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, true /* todo */);
+ hw->pcm_ops->ctl_out(hw, VOICE_ENABLE);
audio_reset_timer (s);
}
}
@@ -1138,7 +948,7 @@ 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, true /* todo */);
+ hw->pcm_ops->ctl_in(hw, VOICE_ENABLE);
audio_reset_timer (s);
}
}
@@ -1462,9 +1272,6 @@ void audio_run (const char *msg)
static int audio_driver_init(AudioState *s, struct audio_driver *drv,
Audiodev *dev)
{
- if (drv->options) {
- audio_process_options (drv->name, drv->options);
- }
s->drv_opaque = drv->init(dev);
if (s->drv_opaque) {
@@ -1489,11 +1296,11 @@ 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, true /* todo */);
+ hwo->pcm_ops->ctl_out(hwo, op);
}
while ((hwi = audio_pcm_hw_find_any_enabled_in (hwi))) {
- hwi->pcm_ops->ctl_in(hwi, op, true /* todo */);
+ hwi->pcm_ops->ctl_in(hwi, op);
}
audio_reset_timer (s);
}
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 24b8793496..83fb517b2a 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -33,22 +33,6 @@
struct audio_pcm_ops;
-typedef enum {
- AUD_OPT_INT,
- AUD_OPT_FMT,
- AUD_OPT_STR,
- AUD_OPT_BOOL
-} audio_option_tag_e;
-
-struct audio_option {
- const char *name;
- audio_option_tag_e tag;
- void *valp;
- const char *descr;
- int *overriddenp;
- int overridden;
-};
-
struct audio_callback {
void *opaque;
audio_callback_fn fn;
@@ -145,7 +129,6 @@ typedef struct audio_driver audio_driver;
struct audio_driver {
const char *name;
const char *descr;
- struct audio_option *options;
void *(*init) (Audiodev *);
void (*fini) (void *);
struct audio_pcm_ops *pcm_ops;
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 17/52] audio: reduce glob_audio_state usage
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (15 preceding siblings ...)
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 16/52] audio: -audiodev command line option: cleanup Kővágó, Zoltán
@ 2018-12-23 20:51 ` Kővágó, Zoltán
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 18/52] audio: basic support for multi backend audio Kővágó, Zoltán
` (33 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:51 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Remove glob_audio_state from functions, where possible without breaking
the API. This means that most static functions in audio.c now take an
AudioState pointer instead of implicitly using glob_audio_state. Also
included a pointer in SWVoice*, HWVoice* structs, so that functions
dealing them can know the audio state without having to pass it around
separately.
This is required in order to support multiple simultaneous audio
backends (added in a later commit).
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio.c | 59 +++++++++++++++++++-----------------------
audio/audio_int.h | 7 +++++
audio/audio_template.h | 46 ++++++++++++++++----------------
3 files changed, 56 insertions(+), 56 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index 0c3f0efd46..461aea7a01 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -398,12 +398,10 @@ static void noop_conv (struct st_sample *dst, const void *src, int samples)
(void) samples;
}
-static CaptureVoiceOut *audio_pcm_capture_find_specific (
- struct audsettings *as
- )
+static CaptureVoiceOut *audio_pcm_capture_find_specific(AudioState *s,
+ struct audsettings *as)
{
CaptureVoiceOut *cap;
- AudioState *s = &glob_audio_state;
for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
if (audio_pcm_info_eq (&cap->hw.info, as)) {
@@ -480,7 +478,7 @@ static void audio_detach_capture (HWVoiceOut *hw)
static int audio_attach_capture (HWVoiceOut *hw)
{
- AudioState *s = &glob_audio_state;
+ AudioState *s = hw->s;
CaptureVoiceOut *cap;
audio_detach_capture (hw);
@@ -793,19 +791,15 @@ static void audio_pcm_print_info (const char *cap, struct audio_pcm_info *info)
/*
* Timer
*/
-
-static bool audio_timer_running;
-static uint64_t audio_timer_last;
-
-static int audio_is_timer_needed (void)
+static int audio_is_timer_needed(AudioState *s)
{
HWVoiceIn *hwi = NULL;
HWVoiceOut *hwo = NULL;
- while ((hwo = audio_pcm_hw_find_any_enabled_out (hwo))) {
+ while ((hwo = audio_pcm_hw_find_any_enabled_out(s, hwo))) {
if (!hwo->poll_mode) return 1;
}
- while ((hwi = audio_pcm_hw_find_any_enabled_in (hwi))) {
+ while ((hwi = audio_pcm_hw_find_any_enabled_in(s, hwi))) {
if (!hwi->poll_mode) return 1;
}
return 0;
@@ -813,18 +807,18 @@ static int audio_is_timer_needed (void)
static void audio_reset_timer (AudioState *s)
{
- if (audio_is_timer_needed ()) {
+ if (audio_is_timer_needed(s)) {
timer_mod_anticipate_ns(s->ts,
qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + s->period_ticks);
- if (!audio_timer_running) {
- audio_timer_running = true;
- audio_timer_last = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
+ if (!s->timer_running) {
+ s->timer_running = true;
+ s->timer_last = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
trace_audio_timer_start(s->period_ticks / SCALE_MS);
}
} else {
timer_del(s->ts);
- if (audio_timer_running) {
- audio_timer_running = false;
+ if (s->timer_running) {
+ s->timer_running = false;
trace_audio_timer_stop();
}
}
@@ -836,11 +830,11 @@ static void audio_timer (void *opaque)
AudioState *s = opaque;
now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
- diff = now - audio_timer_last;
+ diff = now - s->timer_last;
if (diff > s->period_ticks * 3 / 2) {
trace_audio_timer_delayed(diff / SCALE_MS);
}
- audio_timer_last = now;
+ s->timer_last = now;
audio_run("timer");
audio_reset_timer(s);
@@ -894,7 +888,7 @@ void AUD_set_active_out (SWVoiceOut *sw, int on)
hw = sw->hw;
if (sw->active != on) {
- AudioState *s = &glob_audio_state;
+ AudioState *s = sw->s;
SWVoiceOut *temp_sw;
SWVoiceCap *sc;
@@ -941,7 +935,7 @@ void AUD_set_active_in (SWVoiceIn *sw, int on)
hw = sw->hw;
if (sw->active != on) {
- AudioState *s = &glob_audio_state;
+ AudioState *s = sw->s;
SWVoiceIn *temp_sw;
if (on) {
@@ -1064,7 +1058,7 @@ static void audio_run_out (AudioState *s)
HWVoiceOut *hw = NULL;
SWVoiceOut *sw;
- while ((hw = audio_pcm_hw_find_any_enabled_out (hw))) {
+ while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
int played;
int live, free, nb_live, cleanup_required, prev_rpos;
@@ -1169,7 +1163,7 @@ static void audio_run_in (AudioState *s)
{
HWVoiceIn *hw = NULL;
- while ((hw = audio_pcm_hw_find_any_enabled_in (hw))) {
+ while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
SWVoiceIn *sw;
int captured = 0, min;
@@ -1275,8 +1269,8 @@ static int audio_driver_init(AudioState *s, struct audio_driver *drv,
s->drv_opaque = drv->init(dev);
if (s->drv_opaque) {
- audio_init_nb_voices_out (drv);
- audio_init_nb_voices_in (drv);
+ audio_init_nb_voices_out(s, drv);
+ audio_init_nb_voices_in(s, drv);
s->drv = drv;
return 0;
}
@@ -1295,11 +1289,11 @@ static void audio_vm_change_state_handler (void *opaque, int running,
int op = running ? VOICE_ENABLE : VOICE_DISABLE;
s->vm_running = running;
- while ((hwo = audio_pcm_hw_find_any_enabled_out (hwo))) {
+ while ((hwo = audio_pcm_hw_find_any_enabled_out(s, hwo))) {
hwo->pcm_ops->ctl_out(hwo, op);
}
- while ((hwi = audio_pcm_hw_find_any_enabled_in (hwi))) {
+ while ((hwi = audio_pcm_hw_find_any_enabled_in(s, hwi))) {
hwi->pcm_ops->ctl_in(hwi, op);
}
audio_reset_timer (s);
@@ -1319,7 +1313,7 @@ void audio_cleanup(void)
HWVoiceIn *hwi, *hwin;
is_cleaning_up = true;
- QLIST_FOREACH_SAFE(hwo, &glob_audio_state.hw_head_out, entries, hwon) {
+ QLIST_FOREACH_SAFE(hwo, &s->hw_head_out, entries, hwon) {
SWVoiceCap *sc;
if (hwo->enabled) {
@@ -1338,7 +1332,7 @@ void audio_cleanup(void)
QLIST_REMOVE(hwo, entries);
}
- QLIST_FOREACH_SAFE(hwi, &glob_audio_state.hw_head_in, entries, hwin) {
+ QLIST_FOREACH_SAFE(hwi, &s->hw_head_in, entries, hwin) {
if (hwi->enabled) {
hwi->pcm_ops->ctl_in (hwi, VOICE_DISABLE);
}
@@ -1515,7 +1509,7 @@ CaptureVoiceOut *AUD_add_capture (
cb->ops = *ops;
cb->opaque = cb_opaque;
- cap = audio_pcm_capture_find_specific (as);
+ cap = audio_pcm_capture_find_specific(s, as);
if (cap) {
QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
return cap;
@@ -1532,6 +1526,7 @@ CaptureVoiceOut *AUD_add_capture (
}
hw = &cap->hw;
+ hw->s = s;
QLIST_INIT (&hw->sw_head);
QLIST_INIT (&cap->cb_head);
@@ -1564,7 +1559,7 @@ CaptureVoiceOut *AUD_add_capture (
QLIST_INSERT_HEAD (&s->cap_head, cap, entries);
QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
- QLIST_FOREACH(hw, &glob_audio_state.hw_head_out, entries) {
+ QLIST_FOREACH(hw, &s->hw_head_out, entries) {
audio_attach_capture (hw);
}
return cap;
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 83fb517b2a..792ec3152a 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -52,6 +52,7 @@ struct audio_pcm_info {
typedef struct SWVoiceCap SWVoiceCap;
typedef struct HWVoiceOut {
+ AudioState *s;
int enabled;
int poll_mode;
int pending_disable;
@@ -73,6 +74,7 @@ typedef struct HWVoiceOut {
} HWVoiceOut;
typedef struct HWVoiceIn {
+ AudioState *s;
int enabled;
int poll_mode;
struct audio_pcm_info info;
@@ -94,6 +96,7 @@ typedef struct HWVoiceIn {
struct SWVoiceOut {
QEMUSoundCard *card;
+ AudioState *s;
struct audio_pcm_info info;
t_sample *conv;
int64_t ratio;
@@ -111,6 +114,7 @@ struct SWVoiceOut {
struct SWVoiceIn {
QEMUSoundCard *card;
+ AudioState *s;
int active;
struct audio_pcm_info info;
int64_t ratio;
@@ -188,6 +192,9 @@ struct AudioState {
int nb_hw_voices_in;
int vm_running;
int64_t period_ticks;
+
+ bool timer_running;
+ uint64_t timer_last;
};
extern const struct mixeng_volume nominal_volume;
diff --git a/audio/audio_template.h b/audio/audio_template.h
index c1d7207abd..ccf6d810f7 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -36,9 +36,9 @@
#define HWBUF hw->conv_buf
#endif
-static void glue (audio_init_nb_voices_, TYPE) (struct audio_driver *drv)
+static void glue(audio_init_nb_voices_, TYPE)(AudioState *s,
+ struct audio_driver *drv)
{
- AudioState *s = &glob_audio_state;
int max_voices = glue (drv->max_voices_, TYPE);
int voice_size = glue (drv->voice_size_, TYPE);
@@ -183,8 +183,8 @@ static void glue (audio_pcm_hw_del_sw_, TYPE) (SW *sw)
static void glue (audio_pcm_hw_gc_, TYPE) (HW **hwp)
{
- AudioState *s = &glob_audio_state;
HW *hw = *hwp;
+ AudioState *s = hw->s;
if (!hw->sw_head.lh_first) {
#ifdef DAC
@@ -199,15 +199,14 @@ static void glue (audio_pcm_hw_gc_, TYPE) (HW **hwp)
}
}
-static HW *glue (audio_pcm_hw_find_any_, TYPE) (HW *hw)
+static HW *glue(audio_pcm_hw_find_any_, TYPE)(AudioState *s, HW *hw)
{
- AudioState *s = &glob_audio_state;
return hw ? hw->entries.le_next : glue (s->hw_head_, TYPE).lh_first;
}
-static HW *glue (audio_pcm_hw_find_any_enabled_, TYPE) (HW *hw)
+static HW *glue(audio_pcm_hw_find_any_enabled_, TYPE)(AudioState *s, HW *hw)
{
- while ((hw = glue (audio_pcm_hw_find_any_, TYPE) (hw))) {
+ while ((hw = glue(audio_pcm_hw_find_any_, TYPE)(s, hw))) {
if (hw->enabled) {
return hw;
}
@@ -215,12 +214,10 @@ static HW *glue (audio_pcm_hw_find_any_enabled_, TYPE) (HW *hw)
return NULL;
}
-static HW *glue (audio_pcm_hw_find_specific_, TYPE) (
- HW *hw,
- struct audsettings *as
- )
+static HW *glue(audio_pcm_hw_find_specific_, TYPE)(AudioState *s, HW *hw,
+ struct audsettings *as)
{
- while ((hw = glue (audio_pcm_hw_find_any_, TYPE) (hw))) {
+ while ((hw = glue(audio_pcm_hw_find_any_, TYPE)(s, hw))) {
if (audio_pcm_info_eq (&hw->info, as)) {
return hw;
}
@@ -228,10 +225,10 @@ static HW *glue (audio_pcm_hw_find_specific_, TYPE) (
return NULL;
}
-static HW *glue (audio_pcm_hw_add_new_, TYPE) (struct audsettings *as)
+static HW *glue(audio_pcm_hw_add_new_, TYPE)(AudioState *s,
+ struct audsettings *as)
{
HW *hw;
- AudioState *s = &glob_audio_state;
struct audio_driver *drv = s->drv;
if (!glue (s->nb_hw_voices_, TYPE)) {
@@ -255,6 +252,7 @@ static HW *glue (audio_pcm_hw_add_new_, TYPE) (struct audsettings *as)
return NULL;
}
+ hw->s = s;
hw->pcm_ops = drv->pcm_ops;
hw->ctl_caps = drv->ctl_caps;
@@ -299,33 +297,33 @@ static HW *glue (audio_pcm_hw_add_new_, TYPE) (struct audsettings *as)
return NULL;
}
-static HW *glue (audio_pcm_hw_add_, TYPE) (struct audsettings *as)
+static HW *glue(audio_pcm_hw_add_, TYPE)(AudioState *s, struct audsettings *as)
{
HW *hw;
- AudioState *s = &glob_audio_state;
AudiodevPerDirectionOptions *pdo = s->dev->TYPE;
if (pdo->fixed_settings) {
- hw = glue (audio_pcm_hw_add_new_, TYPE) (as);
+ hw = glue(audio_pcm_hw_add_new_, TYPE)(s, as);
if (hw) {
return hw;
}
}
- hw = glue (audio_pcm_hw_find_specific_, TYPE) (NULL, as);
+ hw = glue(audio_pcm_hw_find_specific_, TYPE)(s, NULL, as);
if (hw) {
return hw;
}
- hw = glue (audio_pcm_hw_add_new_, TYPE) (as);
+ hw = glue(audio_pcm_hw_add_new_, TYPE)(s, as);
if (hw) {
return hw;
}
- return glue (audio_pcm_hw_find_any_, TYPE) (NULL);
+ return glue(audio_pcm_hw_find_any_, TYPE)(s, NULL);
}
-static SW *glue (audio_pcm_create_voice_pair_, TYPE) (
+static SW *glue(audio_pcm_create_voice_pair_, TYPE)(
+ AudioState *s,
const char *sw_name,
struct audsettings *as
)
@@ -333,7 +331,6 @@ 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 (pdo->fixed_settings) {
@@ -349,8 +346,9 @@ static SW *glue (audio_pcm_create_voice_pair_, TYPE) (
sw_name ? sw_name : "unknown", sizeof (*sw));
goto err1;
}
+ sw->s = s;
- hw = glue (audio_pcm_hw_add_, TYPE) (&hw_as);
+ hw = glue(audio_pcm_hw_add_, TYPE)(s, &hw_as);
if (!hw) {
goto err2;
}
@@ -447,7 +445,7 @@ SW *glue (AUD_open_, TYPE) (
}
}
else {
- sw = glue (audio_pcm_create_voice_pair_, TYPE) (name, as);
+ sw = glue(audio_pcm_create_voice_pair_, TYPE)(s, name, as);
if (!sw) {
dolog ("Failed to create voice `%s'\n", name);
return NULL;
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 18/52] audio: basic support for multi backend audio
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (16 preceding siblings ...)
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 17/52] audio: reduce glob_audio_state usage Kővágó, Zoltán
@ 2018-12-23 20:51 ` Kővágó, Zoltán
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 19/52] audio: add audiodev properties to frontends Kővágó, Zoltán
` (32 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:51 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann, Dr. David Alan Gilbert, Markus Armbruster
Audio functions no longer access glob_audio_state, instead they get an
AudioState as a parameter. This is required in order to support
multiple backends.
glob_audio_state is also gone, and replaced with a tailq so we can store
more than one states.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio.c | 105 +++++++++++++++++++++++++++++++----------
audio/audio.h | 12 +++--
audio/audio_int.h | 2 +
audio/audio_template.h | 2 +-
audio/wavcapture.c | 6 +--
hmp-commands.hx | 11 +++--
monitor.c | 12 ++++-
qemu-options.hx | 5 ++
ui/vnc.c | 15 +++++-
ui/vnc.h | 2 +
10 files changed, 133 insertions(+), 39 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index 461aea7a01..92fa170b46 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -86,7 +86,8 @@ audio_driver *audio_driver_lookup(const char *name)
return NULL;
}
-static AudioState glob_audio_state;
+static QTAILQ_HEAD(AudioStateHead, AudioState) audio_states =
+ QTAILQ_HEAD_INITIALIZER(audio_states);
const struct mixeng_volume nominal_volume = {
.mute = 0,
@@ -1240,11 +1241,14 @@ static void audio_run_capture (AudioState *s)
void audio_run (const char *msg)
{
- AudioState *s = &glob_audio_state;
+ AudioState *s;
+
+ QTAILQ_FOREACH(s, &audio_states, list) {
+ audio_run_out (s);
+ audio_run_in (s);
+ audio_run_capture (s);
+ }
- audio_run_out (s);
- audio_run_in (s);
- audio_run_capture (s);
#ifdef DEBUG_POLL
{
static double prevtime;
@@ -1306,13 +1310,11 @@ bool audio_is_cleaning_up(void)
return is_cleaning_up;
}
-void audio_cleanup(void)
+static void free_audio_state(AudioState *s)
{
- AudioState *s = &glob_audio_state;
HWVoiceOut *hwo, *hwon;
HWVoiceIn *hwi, *hwin;
- is_cleaning_up = true;
QLIST_FOREACH_SAFE(hwo, &s->hw_head_out, entries, hwon) {
SWVoiceCap *sc;
@@ -1349,6 +1351,17 @@ void audio_cleanup(void)
qapi_free_Audiodev(s->dev);
s->dev = NULL;
}
+ g_free(s);
+}
+
+void audio_cleanup(void)
+{
+ is_cleaning_up = true;
+ while (!QTAILQ_EMPTY(&audio_states)) {
+ AudioState *s = QTAILQ_FIRST(&audio_states);
+ QTAILQ_REMOVE(&audio_states, s, list);
+ free_audio_state(s);
+ }
}
static const VMStateDescription vmstate_audio = {
@@ -1361,28 +1374,34 @@ static const VMStateDescription vmstate_audio = {
};
static Audiodev *parse_option(QemuOpts *opts, Error **errp);
-static int audio_init(Audiodev *dev)
+
+/*
+ * if we have dev, this function was called because of an -audiodev argument =>
+ * initialize a new state with it
+ * if dev == NULL => legacy implicit initialization, return the already created
+ * state or create a new one
+ */
+static AudioState *audio_init(Audiodev *dev)
{
+ static bool atexit_registered;
size_t i;
int done = 0;
const char *drvname = NULL;
VMChangeStateEntry *e;
- AudioState *s = &glob_audio_state;
+ AudioState *s;
struct audio_driver *driver;
/* silence gcc warning about uninitialized variable */
QemuOptsList *list = NULL;
- if (s->drv) {
- if (dev) {
- dolog("Cannot create more than one audio backend, sorry\n");
- qapi_free_Audiodev(dev);
- }
- return -1;
- }
-
if (dev) {
/* -audiodev option */
drvname = AudiodevDriver_str(dev->driver);
+ } else if (!QTAILQ_EMPTY(&audio_states)) {
+ /*
+ * todo: check for -audiodev once we have normal audiodev selection
+ * support
+ */
+ return QTAILQ_FIRST(&audio_states);
} else {
/* legacy implicit initialization */
audio_handle_legacy_opts();
@@ -1392,12 +1411,18 @@ static int audio_init(Audiodev *dev)
exit(1);
}
}
+
+ s = g_malloc0(sizeof(AudioState));
s->dev = dev;
QLIST_INIT (&s->hw_head_out);
QLIST_INIT (&s->hw_head_in);
QLIST_INIT (&s->cap_head);
- atexit(audio_cleanup);
+ if (!atexit_registered) {
+ atexit(audio_cleanup);
+ atexit_registered = true;
+ }
+ QTAILQ_INSERT_TAIL(&audio_states, s, list);
s->ts = timer_new_ns(QEMU_CLOCK_VIRTUAL, audio_timer, s);
@@ -1466,15 +1491,18 @@ static int audio_init(Audiodev *dev)
QLIST_INIT (&s->card_head);
vmstate_register (NULL, 0, &vmstate_audio, s);
- return 0;
+ return s;
}
void AUD_register_card (const char *name, QEMUSoundCard *card)
{
- audio_init(NULL);
+ if (!card->state) {
+ card->state = 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);
+ QLIST_INSERT_HEAD (&card->state->card_head, card, entries);
}
void AUD_remove_card (QEMUSoundCard *card)
@@ -1484,16 +1512,21 @@ void AUD_remove_card (QEMUSoundCard *card)
}
-CaptureVoiceOut *AUD_add_capture (
+CaptureVoiceOut *AUD_add_capture(
+ AudioState *s,
struct audsettings *as,
struct audio_capture_ops *ops,
void *cb_opaque
)
{
- AudioState *s = &glob_audio_state;
CaptureVoiceOut *cap;
struct capture_callback *cb;
+ if (!s) {
+ /* todo: remove when we have normal audiodev selection support */
+ s = audio_init(NULL);
+ }
+
if (audio_validate_settings (as)) {
dolog ("Invalid settings were passed when trying to add capture\n");
audio_print_settings (as);
@@ -1730,7 +1763,7 @@ static int each_option(void *opaque, QemuOpts *opts, Error **errp)
if (!dev) {
return -1;
}
- return audio_init(dev);
+ return audio_init(dev) ? 0 : -1;
}
void audio_set_options(void)
@@ -1798,3 +1831,25 @@ int audio_buffer_bytes(AudiodevPerDirectionOptions *pdo,
return audio_buffer_samples(pdo, as, def_usecs) *
audioformat_bytes_per_sample(as->fmt);
}
+
+AudioState *audio_state_by_name(const char *name)
+{
+ AudioState *s;
+ QTAILQ_FOREACH(s, &audio_states, list) {
+ assert(s->dev);
+ if (strcmp(name, s->dev->id) == 0) {
+ return s;
+ }
+ }
+ return NULL;
+}
+
+const char *audio_get_id(QEMUSoundCard *card)
+{
+ if (card->state) {
+ assert(card->state->dev);
+ return card->state->dev->id;
+ } else {
+ return "";
+ }
+}
diff --git a/audio/audio.h b/audio/audio.h
index 7df1b8b249..5e0dcdb93e 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -78,8 +78,10 @@ typedef struct SWVoiceOut SWVoiceOut;
typedef struct CaptureVoiceOut CaptureVoiceOut;
typedef struct SWVoiceIn SWVoiceIn;
+typedef struct AudioState AudioState;
typedef struct QEMUSoundCard {
char *name;
+ AudioState *state;
QLIST_ENTRY (QEMUSoundCard) entries;
} QEMUSoundCard;
@@ -94,7 +96,8 @@ void AUD_log (const char *cap, const char *fmt, ...) GCC_FMT_ATTR(2, 3);
void AUD_register_card (const char *name, QEMUSoundCard *card);
void AUD_remove_card (QEMUSoundCard *card);
-CaptureVoiceOut *AUD_add_capture (
+CaptureVoiceOut *AUD_add_capture(
+ AudioState *s,
struct audsettings *as,
struct audio_capture_ops *ops,
void *opaque
@@ -162,8 +165,8 @@ static inline void *advance (void *p, int incr)
#define audio_MAX(a, b) ((a)<(b)?(b):(a))
#endif
-int wav_start_capture (CaptureState *s, const char *path, int freq,
- int bits, int nchannels);
+int wav_start_capture(AudioState *state, CaptureState *s, const char *path,
+ int freq, int bits, int nchannels);
bool audio_is_cleaning_up(void);
void audio_cleanup(void);
@@ -177,4 +180,7 @@ void audio_set_options(void);
void audio_handle_legacy_opts(void);
void audio_legacy_help(void);
+AudioState *audio_state_by_name(const char *name);
+const char *audio_get_id(QEMUSoundCard *card);
+
#endif /* QEMU_AUDIO_H */
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 792ec3152a..9d9db8c270 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -195,6 +195,8 @@ struct AudioState {
bool timer_running;
uint64_t timer_last;
+
+ QTAILQ_ENTRY(AudioState) list;
};
extern const struct mixeng_volume nominal_volume;
diff --git a/audio/audio_template.h b/audio/audio_template.h
index ccf6d810f7..ce1e5d6559 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -399,7 +399,7 @@ SW *glue (AUD_open_, TYPE) (
struct audsettings *as
)
{
- AudioState *s = &glob_audio_state;
+ AudioState *s = card->state;
AudiodevPerDirectionOptions *pdo = s->dev->TYPE;
if (audio_bug(__func__, !card || !name || !callback_fn || !as)) {
diff --git a/audio/wavcapture.c b/audio/wavcapture.c
index e943565210..88d3141be5 100644
--- a/audio/wavcapture.c
+++ b/audio/wavcapture.c
@@ -107,8 +107,8 @@ static struct capture_ops wav_capture_ops = {
.info = wav_capture_info
};
-int wav_start_capture (CaptureState *s, const char *path, int freq,
- int bits, int nchannels)
+int wav_start_capture(AudioState *state, CaptureState *s, const char *path,
+ int freq, int bits, int nchannels)
{
Monitor *mon = cur_mon;
WAVState *wav;
@@ -175,7 +175,7 @@ int wav_start_capture (CaptureState *s, const char *path, int freq,
goto error_free;
}
- cap = AUD_add_capture (&as, &ops, wav);
+ cap = AUD_add_capture(state, &as, &ops, wav);
if (!cap) {
monitor_printf (mon, "Failed to add audio capture\n");
goto error_free;
diff --git a/hmp-commands.hx b/hmp-commands.hx
index ba71558c25..dc339a5484 100644
--- a/hmp-commands.hx
+++ b/hmp-commands.hx
@@ -796,16 +796,17 @@ ETEXI
{
.name = "wavcapture",
- .args_type = "path:F,freq:i?,bits:i?,nchannels:i?",
- .params = "path [frequency [bits [channels]]]",
+ .args_type = "path:F,freq:i?,bits:i?,nchannels:i?,audiodev:s?",
+ .params = "path [frequency [bits [channels [audiodev]]]]",
.help = "capture audio to a wave file (default frequency=44100 bits=16 channels=2)",
.cmd = hmp_wavcapture,
},
STEXI
-@item wavcapture @var{filename} [@var{frequency} [@var{bits} [@var{channels}]]]
+@item wavcapture @var{filename} [@var{frequency} [@var{bits} [@var{channels} [@var{audiodev}]]]]
@findex wavcapture
-Capture audio into @var{filename}. Using sample rate @var{frequency}
-bits per sample @var{bits} and number of channels @var{channels}.
+Capture audio into @var{filename} from @var{audiodev}. Using sample rate
+@var{frequency} bits per sample @var{bits} and number of channels
+@var{channels}.
Defaults:
@itemize @minus
diff --git a/monitor.c b/monitor.c
index 0ad54d8b11..14e130308f 100644
--- a/monitor.c
+++ b/monitor.c
@@ -2038,7 +2038,17 @@ static void hmp_wavcapture(Monitor *mon, const QDict *qdict)
int bits = qdict_get_try_int(qdict, "bits", -1);
int has_channels = qdict_haskey(qdict, "nchannels");
int nchannels = qdict_get_try_int(qdict, "nchannels", -1);
+ const char *audiodev = qdict_get_try_str(qdict, "audiodev");
CaptureState *s;
+ AudioState *as = NULL;
+
+ if (audiodev) {
+ as = audio_state_by_name(audiodev);
+ if (!as) {
+ monitor_printf(mon, "Invalid audiodev specified\n");
+ return;
+ }
+ }
s = g_malloc0 (sizeof (*s));
@@ -2046,7 +2056,7 @@ static void hmp_wavcapture(Monitor *mon, const QDict *qdict)
bits = has_bits ? bits : 16;
nchannels = has_channels ? nchannels : 2;
- if (wav_start_capture (s, path, freq, bits, nchannels)) {
+ if (wav_start_capture(as, s, path, freq, bits, nchannels)) {
monitor_printf(mon, "Failed to add wave capture\n");
g_free (s);
return;
diff --git a/qemu-options.hx b/qemu-options.hx
index 3ede6a591d..687dd23bf2 100644
--- a/qemu-options.hx
+++ b/qemu-options.hx
@@ -1911,6 +1911,11 @@ can help the device and guest to keep up and not lose events in case
events are arriving in bulk. Possible causes for the latter are flaky
network connections, or scripts for automated testing.
+@item audiodev=@var{audiodev}
+
+Use the specified @var{audiodev} when the VNC client requests audio
+transmission.
+
@end table
ETEXI
diff --git a/ui/vnc.c b/ui/vnc.c
index 0e9e9fb63a..c64ca10de9 100644
--- a/ui/vnc.c
+++ b/ui/vnc.c
@@ -1195,7 +1195,7 @@ static void audio_add(VncState *vs)
ops.destroy = audio_capture_destroy;
ops.capture = audio_capture;
- vs->audio_cap = AUD_add_capture(&vs->as, &ops, vs);
+ vs->audio_cap = AUD_add_capture(vs->vd->audio_state, &vs->as, &ops, vs);
if (!vs->audio_cap) {
error_report("Failed to add audio capture");
}
@@ -3390,6 +3390,9 @@ static QemuOptsList qemu_vnc_opts = {
},{
.name = "non-adaptive",
.type = QEMU_OPT_BOOL,
+ },{
+ .name = "audiodev",
+ .type = QEMU_OPT_STRING,
},
{ /* end of list */ }
},
@@ -3825,6 +3828,7 @@ void vnc_display_open(const char *id, Error **errp)
int acl = 0;
int lock_key_sync = 1;
int key_delay_ms;
+ const char *audiodev;
if (!vd) {
error_setg(errp, "VNC display not active");
@@ -3978,6 +3982,15 @@ void vnc_display_open(const char *id, Error **errp)
vd->ledstate = 0;
vd->key_delay_ms = key_delay_ms;
+ audiodev = qemu_opt_get(opts, "audiodev");
+ if (audiodev) {
+ vd->audio_state = audio_state_by_name(audiodev);
+ if (!vd->audio_state) {
+ error_setg(errp, "Audiodev '%s' not found", audiodev);
+ goto fail;
+ }
+ }
+
device_id = qemu_opt_get(opts, "display");
if (device_id) {
int head = qemu_opt_get_number(opts, "head", 0);
diff --git a/ui/vnc.h b/ui/vnc.h
index a86e0610e8..0c9a2d82e0 100644
--- a/ui/vnc.h
+++ b/ui/vnc.h
@@ -181,6 +181,8 @@ struct VncDisplay
#ifdef CONFIG_VNC_SASL
VncDisplaySASL sasl;
#endif
+
+ AudioState *audio_state;
};
typedef struct VncTight {
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 19/52] audio: add audiodev properties to frontends
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (17 preceding siblings ...)
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 18/52] audio: basic support for multi backend audio Kővágó, Zoltán
@ 2018-12-23 20:51 ` Kővágó, Zoltán
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 20/52] audio: audiodev= parameters no longer optional when -audiodev present Kővágó, Zoltán
` (31 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:51 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann, Michael Walle
Finally add audiodev= options to audio frontends so users can specify
which backend to use when multiple backends exist. Not specifying an
audiodev= option currently causes the first audiodev to be used, this is
fixed in the next commit.
Example usage: -audiodev pa,id=foo -device AC97,audiodev=foo
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio.h | 3 ++
hw/audio/ac97.c | 1 +
hw/audio/adlib.c | 1 +
hw/audio/cs4231a.c | 1 +
hw/audio/es1370.c | 7 +++-
hw/audio/gus.c | 1 +
hw/audio/hda-codec.c | 1 +
hw/audio/milkymist-ac97.c | 6 ++++
hw/audio/pcspk.c | 1 +
hw/audio/pl041.c | 1 +
hw/audio/sb16.c | 1 +
hw/audio/wm8750.c | 6 ++++
hw/core/qdev-properties-system.c | 57 ++++++++++++++++++++++++++++++++
hw/usb/dev-audio.c | 1 +
include/hw/qdev-properties.h | 3 ++
15 files changed, 90 insertions(+), 1 deletion(-)
diff --git a/audio/audio.h b/audio/audio.h
index 5e0dcdb93e..ccfef9e10a 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -183,4 +183,7 @@ void audio_legacy_help(void);
AudioState *audio_state_by_name(const char *name);
const char *audio_get_id(QEMUSoundCard *card);
+#define DEFINE_AUDIO_PROPERTIES(_s, _f) \
+ DEFINE_PROP_AUDIODEV("audiodev", _s, _f)
+
#endif /* QEMU_AUDIO_H */
diff --git a/hw/audio/ac97.c b/hw/audio/ac97.c
index 2265622d44..03e3da4f31 100644
--- a/hw/audio/ac97.c
+++ b/hw/audio/ac97.c
@@ -1408,6 +1408,7 @@ static int ac97_init (PCIBus *bus)
}
static Property ac97_properties[] = {
+ DEFINE_AUDIO_PROPERTIES(AC97LinkState, card),
DEFINE_PROP_UINT32 ("use_broken_id", AC97LinkState, use_broken_id, 0),
DEFINE_PROP_END_OF_LIST (),
};
diff --git a/hw/audio/adlib.c b/hw/audio/adlib.c
index 0957780a3d..0d01cd07c5 100644
--- a/hw/audio/adlib.c
+++ b/hw/audio/adlib.c
@@ -298,6 +298,7 @@ static void adlib_realizefn (DeviceState *dev, Error **errp)
}
static Property adlib_properties[] = {
+ DEFINE_AUDIO_PROPERTIES(AdlibState, card),
DEFINE_PROP_UINT32 ("iobase", AdlibState, port, 0x220),
DEFINE_PROP_UINT32 ("freq", AdlibState, freq, 44100),
DEFINE_PROP_END_OF_LIST (),
diff --git a/hw/audio/cs4231a.c b/hw/audio/cs4231a.c
index 62da75eefe..d25a120b0f 100644
--- a/hw/audio/cs4231a.c
+++ b/hw/audio/cs4231a.c
@@ -687,6 +687,7 @@ static int cs4231a_init (ISABus *bus)
}
static Property cs4231a_properties[] = {
+ DEFINE_AUDIO_PROPERTIES(CSState, card),
DEFINE_PROP_UINT32 ("iobase", CSState, port, 0x534),
DEFINE_PROP_UINT32 ("irq", CSState, irq, 9),
DEFINE_PROP_UINT32 ("dma", CSState, dma, 3),
diff --git a/hw/audio/es1370.c b/hw/audio/es1370.c
index a5314d66fd..8c63912a82 100644
--- a/hw/audio/es1370.c
+++ b/hw/audio/es1370.c
@@ -886,6 +886,11 @@ static int es1370_init (PCIBus *bus)
return 0;
}
+static Property es1370_properties[] = {
+ DEFINE_AUDIO_PROPERTIES(ES1370State, card),
+ DEFINE_PROP_END_OF_LIST(),
+};
+
static void es1370_class_init (ObjectClass *klass, void *data)
{
DeviceClass *dc = DEVICE_CLASS (klass);
@@ -902,6 +907,7 @@ static void es1370_class_init (ObjectClass *klass, void *data)
dc->desc = "ENSONIQ AudioPCI ES1370";
dc->vmsd = &vmstate_es1370;
dc->reset = es1370_on_reset;
+ dc->props = es1370_properties;
}
static const TypeInfo es1370_info = {
@@ -922,4 +928,3 @@ static void es1370_register_types (void)
}
type_init (es1370_register_types)
-
diff --git a/hw/audio/gus.c b/hw/audio/gus.c
index b3e2a7fdd5..dfb74cf0d3 100644
--- a/hw/audio/gus.c
+++ b/hw/audio/gus.c
@@ -295,6 +295,7 @@ static int GUS_init (ISABus *bus)
}
static Property gus_properties[] = {
+ DEFINE_AUDIO_PROPERTIES(GUSState, card),
DEFINE_PROP_UINT32 ("freq", GUSState, freq, 44100),
DEFINE_PROP_UINT32 ("iobase", GUSState, port, 0x240),
DEFINE_PROP_UINT32 ("irq", GUSState, emu.gusirq, 7),
diff --git a/hw/audio/hda-codec.c b/hw/audio/hda-codec.c
index c25bfa38b1..e66103aedf 100644
--- a/hw/audio/hda-codec.c
+++ b/hw/audio/hda-codec.c
@@ -839,6 +839,7 @@ static const VMStateDescription vmstate_hda_audio = {
};
static Property hda_audio_properties[] = {
+ DEFINE_AUDIO_PROPERTIES(HDAAudioState, card),
DEFINE_PROP_UINT32("debug", HDAAudioState, debug, 0),
DEFINE_PROP_BOOL("mixer", HDAAudioState, mixer, true),
DEFINE_PROP_BOOL("use-timer", HDAAudioState, use_timer, true),
diff --git a/hw/audio/milkymist-ac97.c b/hw/audio/milkymist-ac97.c
index 90cce1e6ed..8739cb376a 100644
--- a/hw/audio/milkymist-ac97.c
+++ b/hw/audio/milkymist-ac97.c
@@ -328,6 +328,11 @@ static const VMStateDescription vmstate_milkymist_ac97 = {
}
};
+static Property milkymist_ac97_properties[] = {
+ DEFINE_AUDIO_PROPERTIES(MilkymistAC97State, card),
+ DEFINE_PROP_END_OF_LIST(),
+};
+
static void milkymist_ac97_class_init(ObjectClass *klass, void *data)
{
DeviceClass *dc = DEVICE_CLASS(klass);
@@ -335,6 +340,7 @@ static void milkymist_ac97_class_init(ObjectClass *klass, void *data)
dc->realize = milkymist_ac97_realize;
dc->reset = milkymist_ac97_reset;
dc->vmsd = &vmstate_milkymist_ac97;
+ dc->props = milkymist_ac97_properties;
}
static const TypeInfo milkymist_ac97_info = {
diff --git a/hw/audio/pcspk.c b/hw/audio/pcspk.c
index 4a7386a689..1457cd1ac6 100644
--- a/hw/audio/pcspk.c
+++ b/hw/audio/pcspk.c
@@ -208,6 +208,7 @@ static const VMStateDescription vmstate_spk = {
};
static Property pcspk_properties[] = {
+ DEFINE_AUDIO_PROPERTIES(PCSpkState, card),
DEFINE_PROP_UINT32("iobase", PCSpkState, iobase, -1),
DEFINE_PROP_BOOL("migrate", PCSpkState, migrate, true),
DEFINE_PROP_END_OF_LIST(),
diff --git a/hw/audio/pl041.c b/hw/audio/pl041.c
index c8cc503236..fca1dc8a54 100644
--- a/hw/audio/pl041.c
+++ b/hw/audio/pl041.c
@@ -621,6 +621,7 @@ static const VMStateDescription vmstate_pl041 = {
};
static Property pl041_device_properties[] = {
+ DEFINE_AUDIO_PROPERTIES(PL041State, codec.card),
/* Non-compact FIFO depth property */
DEFINE_PROP_UINT32("nc_fifo_depth", PL041State, fifo_depth,
DEFAULT_FIFO_DEPTH),
diff --git a/hw/audio/sb16.c b/hw/audio/sb16.c
index 65ea0cd938..5a6a880f2e 100644
--- a/hw/audio/sb16.c
+++ b/hw/audio/sb16.c
@@ -1419,6 +1419,7 @@ static int SB16_init (ISABus *bus)
}
static Property sb16_properties[] = {
+ DEFINE_AUDIO_PROPERTIES(SB16State, card),
DEFINE_PROP_UINT32 ("version", SB16State, ver, 0x0405), /* 4.5 */
DEFINE_PROP_UINT32 ("iobase", SB16State, port, 0x220),
DEFINE_PROP_UINT32 ("irq", SB16State, irq, 5),
diff --git a/hw/audio/wm8750.c b/hw/audio/wm8750.c
index 5d614c0d8b..c86fd91f98 100644
--- a/hw/audio/wm8750.c
+++ b/hw/audio/wm8750.c
@@ -700,6 +700,11 @@ void wm8750_set_bclk_in(void *opaque, int new_hz)
wm8750_clk_update(s, 1);
}
+static Property wm8750_properties[] = {
+ DEFINE_AUDIO_PROPERTIES(WM8750State, card),
+ DEFINE_PROP_END_OF_LIST(),
+};
+
static void wm8750_class_init(ObjectClass *klass, void *data)
{
DeviceClass *dc = DEVICE_CLASS(klass);
@@ -710,6 +715,7 @@ static void wm8750_class_init(ObjectClass *klass, void *data)
sc->recv = wm8750_rx;
sc->send = wm8750_tx;
dc->vmsd = &vmstate_wm8750;
+ dc->props = wm8750_properties;
}
static const TypeInfo wm8750_info = {
diff --git a/hw/core/qdev-properties-system.c b/hw/core/qdev-properties-system.c
index b45a7ef54b..4659b6986f 100644
--- a/hw/core/qdev-properties-system.c
+++ b/hw/core/qdev-properties-system.c
@@ -11,6 +11,7 @@
*/
#include "qemu/osdep.h"
+#include "audio/audio.h"
#include "net/net.h"
#include "hw/qdev.h"
#include "qapi/error.h"
@@ -318,6 +319,62 @@ const PropertyInfo qdev_prop_netdev = {
};
+/* --- audiodev --- */
+static void get_audiodev(Object *obj, Visitor *v, const char* name,
+ void *opaque, Error **errp)
+{
+ DeviceState *dev = DEVICE(obj);
+ Property *prop = opaque;
+ QEMUSoundCard *card = qdev_get_prop_ptr(dev, prop);
+ char *p = g_strdup(audio_get_id(card));
+
+ visit_type_str(v, name, &p, errp);
+ g_free(p);
+}
+
+static void set_audiodev(Object *obj, Visitor *v, const char* name,
+ void *opaque, Error **errp)
+{
+ DeviceState *dev = DEVICE(obj);
+ Property *prop = opaque;
+ QEMUSoundCard *card = qdev_get_prop_ptr(dev, prop);
+ AudioState *state;
+ Error *local_err = NULL;
+ int err = 0;
+ char *str;
+
+ if (dev->realized) {
+ qdev_prop_set_after_realize(dev, name, errp);
+ return;
+ }
+
+ visit_type_str(v, name, &str, &local_err);
+ if (local_err) {
+ error_propagate(errp, local_err);
+ return;
+ }
+
+ state = audio_state_by_name(str);
+
+ if (!state) {
+ err = -ENOENT;
+ goto out;
+ }
+ card->state = state;
+
+out:
+ error_set_from_qdev_prop_error(errp, err, dev, prop, str);
+ g_free(str);
+}
+
+const PropertyInfo qdev_prop_audiodev = {
+ .name = "str",
+ .description = "ID of an audiodev to use as a backend",
+ /* release done on shutdown */
+ .get = get_audiodev,
+ .set = set_audiodev,
+};
+
void qdev_prop_set_drive(DeviceState *dev, const char *name,
BlockBackend *value, Error **errp)
{
diff --git a/hw/usb/dev-audio.c b/hw/usb/dev-audio.c
index b1b9736a19..589977d96c 100644
--- a/hw/usb/dev-audio.c
+++ b/hw/usb/dev-audio.c
@@ -663,6 +663,7 @@ static const VMStateDescription vmstate_usb_audio = {
};
static Property usb_audio_properties[] = {
+ DEFINE_AUDIO_PROPERTIES(USBAudioState, card),
DEFINE_PROP_UINT32("debug", USBAudioState, debug, 0),
DEFINE_PROP_UINT32("buffer", USBAudioState, buffer,
32 * USBAUDIO_PACKET_SIZE),
diff --git a/include/hw/qdev-properties.h b/include/hw/qdev-properties.h
index b6758c852e..6156409062 100644
--- a/include/hw/qdev-properties.h
+++ b/include/hw/qdev-properties.h
@@ -34,6 +34,7 @@ extern const PropertyInfo qdev_prop_blocksize;
extern const PropertyInfo qdev_prop_pci_host_devaddr;
extern const PropertyInfo qdev_prop_uuid;
extern const PropertyInfo qdev_prop_arraylen;
+extern const PropertyInfo qdev_prop_audiodev;
extern const PropertyInfo qdev_prop_link;
extern const PropertyInfo qdev_prop_off_auto_pcibar;
extern const PropertyInfo qdev_prop_pcie_link_speed;
@@ -233,6 +234,8 @@ extern const PropertyInfo qdev_prop_pcie_link_width;
+ type_check(QemuUUID, typeof_field(_state, _field)), \
.set_default = true, \
}
+#define DEFINE_PROP_AUDIODEV(_n, _s, _f) \
+ DEFINE_PROP(_n, _s, _f, qdev_prop_audiodev, QEMUSoundCard)
#define DEFINE_PROP_END_OF_LIST() \
{}
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 20/52] audio: audiodev= parameters no longer optional when -audiodev present
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (18 preceding siblings ...)
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 19/52] audio: add audiodev properties to frontends Kővágó, Zoltán
@ 2018-12-23 20:51 ` Kővágó, Zoltán
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 21/52] paaudio: do not move stream when sink/source name is specified Kővágó, Zoltán
` (30 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:51 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
This means you should probably stop using -soundhw (as it doesn't allow
you to specify any options) and add the device manually with -device.
The exception is pcspk, it's currently not possible to manually add it.
To use it with audiodev, use something like this:
-audiodev id=foo,... -global isa-pcspk.audiodev=foo -soundhw pcspk
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio.c | 24 ++--
audio/paaudio.c | 303 +++++++++++++++++++++++++++---------------------
2 files changed, 189 insertions(+), 138 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index 92fa170b46..a7f79f3929 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -100,6 +100,8 @@ const struct mixeng_volume nominal_volume = {
#endif
};
+static bool legacy_config = true;
+
#ifdef AUDIO_IS_FLAWLESS_AND_NO_CHECKS_ARE_REQURIED
#error No its not
#else
@@ -1381,7 +1383,7 @@ static Audiodev *parse_option(QemuOpts *opts, Error **errp);
* if dev == NULL => legacy implicit initialization, return the already created
* state or create a new one
*/
-static AudioState *audio_init(Audiodev *dev)
+static AudioState *audio_init(Audiodev *dev, const char *name)
{
static bool atexit_registered;
size_t i;
@@ -1395,12 +1397,13 @@ static AudioState *audio_init(Audiodev *dev)
if (dev) {
/* -audiodev option */
+ legacy_config = false;
drvname = AudiodevDriver_str(dev->driver);
} else if (!QTAILQ_EMPTY(&audio_states)) {
- /*
- * todo: check for -audiodev once we have normal audiodev selection
- * support
- */
+ if (!legacy_config) {
+ dolog("You must specify an audiodev= for the device %s\n", name);
+ exit(1);
+ }
return QTAILQ_FIRST(&audio_states);
} else {
/* legacy implicit initialization */
@@ -1497,7 +1500,7 @@ static AudioState *audio_init(Audiodev *dev)
void AUD_register_card (const char *name, QEMUSoundCard *card)
{
if (!card->state) {
- card->state = audio_init(NULL);
+ card->state = audio_init(NULL, name);
}
card->name = g_strdup (name);
@@ -1523,8 +1526,11 @@ CaptureVoiceOut *AUD_add_capture(
struct capture_callback *cb;
if (!s) {
- /* todo: remove when we have normal audiodev selection support */
- s = audio_init(NULL);
+ if (!legacy_config) {
+ dolog("You must specify audiodev when trying to capture\n");
+ goto err0;
+ }
+ s = audio_init(NULL, NULL);
}
if (audio_validate_settings (as)) {
@@ -1763,7 +1769,7 @@ static int each_option(void *opaque, QemuOpts *opts, Error **errp)
if (!dev) {
return -1;
}
- return audio_init(dev) ? 0 : -1;
+ return audio_init(dev, NULL) ? 0 : -1;
}
void audio_set_options(void)
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 6de6ed97fc..24b75ca503 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -10,10 +10,21 @@
#include "audio_int.h"
#include "audio_pt_int.h"
-typedef struct {
- Audiodev *dev;
+typedef struct PAConnection {
+ char *server;
+ int refcount;
+ QTAILQ_ENTRY(PAConnection) list;
+
pa_threaded_mainloop *mainloop;
pa_context *context;
+} PAConnection;
+
+static QTAILQ_HEAD(PAConnectionHead, PAConnection) pa_conns =
+ QTAILQ_HEAD_INITIALIZER(pa_conns);
+
+typedef struct {
+ Audiodev *dev;
+ PAConnection *conn;
} paaudio;
typedef struct {
@@ -44,7 +55,7 @@ typedef struct {
int samples;
} PAVoiceIn;
-static void qpa_audio_fini(void *opaque);
+static void qpa_conn_fini(PAConnection *c);
static void GCC_FMT_ATTR (2, 3) qpa_logerr (int err, const char *fmt, ...)
{
@@ -107,11 +118,11 @@ static inline int PA_STREAM_IS_GOOD(pa_stream_state_t x)
static int qpa_simple_read (PAVoiceIn *p, void *data, size_t length, int *rerror)
{
- paaudio *g = p->g;
+ PAConnection *c = p->g->conn;
- pa_threaded_mainloop_lock (g->mainloop);
+ pa_threaded_mainloop_lock(c->mainloop);
- CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
+ CHECK_DEAD_GOTO(c, p->stream, rerror, unlock_and_fail);
while (length > 0) {
size_t l;
@@ -120,11 +131,11 @@ static int qpa_simple_read (PAVoiceIn *p, void *data, size_t length, int *rerror
int r;
r = pa_stream_peek (p->stream, &p->read_data, &p->read_length);
- CHECK_SUCCESS_GOTO (g, rerror, r == 0, unlock_and_fail);
+ CHECK_SUCCESS_GOTO(c, rerror, r == 0, unlock_and_fail);
if (!p->read_data) {
- pa_threaded_mainloop_wait (g->mainloop);
- CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
+ pa_threaded_mainloop_wait(c->mainloop);
+ CHECK_DEAD_GOTO(c, p->stream, rerror, unlock_and_fail);
} else {
p->read_index = 0;
}
@@ -147,53 +158,53 @@ static int qpa_simple_read (PAVoiceIn *p, void *data, size_t length, int *rerror
p->read_length = 0;
p->read_index = 0;
- CHECK_SUCCESS_GOTO (g, rerror, r == 0, unlock_and_fail);
+ CHECK_SUCCESS_GOTO(c, rerror, r == 0, unlock_and_fail);
}
}
- pa_threaded_mainloop_unlock (g->mainloop);
+ pa_threaded_mainloop_unlock(c->mainloop);
return 0;
unlock_and_fail:
- pa_threaded_mainloop_unlock (g->mainloop);
+ pa_threaded_mainloop_unlock(c->mainloop);
return -1;
}
static int qpa_simple_write (PAVoiceOut *p, const void *data, size_t length, int *rerror)
{
- paaudio *g = p->g;
+ PAConnection *c = p->g->conn;
- pa_threaded_mainloop_lock (g->mainloop);
+ pa_threaded_mainloop_lock(c->mainloop);
- CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
+ CHECK_DEAD_GOTO(c, p->stream, rerror, unlock_and_fail);
while (length > 0) {
size_t l;
int r;
while (!(l = pa_stream_writable_size (p->stream))) {
- pa_threaded_mainloop_wait (g->mainloop);
- CHECK_DEAD_GOTO (g, p->stream, rerror, unlock_and_fail);
+ pa_threaded_mainloop_wait(c->mainloop);
+ CHECK_DEAD_GOTO(c, p->stream, rerror, unlock_and_fail);
}
- CHECK_SUCCESS_GOTO (g, rerror, l != (size_t) -1, unlock_and_fail);
+ CHECK_SUCCESS_GOTO(c, rerror, l != (size_t) -1, unlock_and_fail);
if (l > length) {
l = length;
}
r = pa_stream_write (p->stream, data, l, NULL, 0LL, PA_SEEK_RELATIVE);
- CHECK_SUCCESS_GOTO (g, rerror, r >= 0, unlock_and_fail);
+ CHECK_SUCCESS_GOTO(c, rerror, r >= 0, unlock_and_fail);
data = (const uint8_t *) data + l;
length -= l;
}
- pa_threaded_mainloop_unlock (g->mainloop);
+ pa_threaded_mainloop_unlock(c->mainloop);
return 0;
unlock_and_fail:
- pa_threaded_mainloop_unlock (g->mainloop);
+ pa_threaded_mainloop_unlock(c->mainloop);
return -1;
}
@@ -431,13 +442,13 @@ static AudioFormat pa_to_audfmt (pa_sample_format_t fmt, int *endianness)
static void context_state_cb (pa_context *c, void *userdata)
{
- paaudio *g = userdata;
+ PAConnection *conn = userdata;
switch (pa_context_get_state(c)) {
case PA_CONTEXT_READY:
case PA_CONTEXT_TERMINATED:
case PA_CONTEXT_FAILED:
- pa_threaded_mainloop_signal (g->mainloop, 0);
+ pa_threaded_mainloop_signal(conn->mainloop, 0);
break;
case PA_CONTEXT_UNCONNECTED:
@@ -450,14 +461,14 @@ static void context_state_cb (pa_context *c, void *userdata)
static void stream_state_cb (pa_stream *s, void * userdata)
{
- paaudio *g = userdata;
+ PAConnection *c = userdata;
switch (pa_stream_get_state (s)) {
case PA_STREAM_READY:
case PA_STREAM_FAILED:
case PA_STREAM_TERMINATED:
- pa_threaded_mainloop_signal (g->mainloop, 0);
+ pa_threaded_mainloop_signal(c->mainloop, 0);
break;
case PA_STREAM_UNCONNECTED:
@@ -468,13 +479,13 @@ static void stream_state_cb (pa_stream *s, void * userdata)
static void stream_request_cb (pa_stream *s, size_t length, void *userdata)
{
- paaudio *g = userdata;
+ PAConnection *c = userdata;
- pa_threaded_mainloop_signal (g->mainloop, 0);
+ pa_threaded_mainloop_signal(c->mainloop, 0);
}
static pa_stream *qpa_simple_new (
- paaudio *g,
+ PAConnection *c,
const char *name,
pa_stream_direction_t dir,
const char *dev,
@@ -485,50 +496,48 @@ static pa_stream *qpa_simple_new (
{
int r;
pa_stream *stream;
+ pa_stream_flags_t flags;
- pa_threaded_mainloop_lock (g->mainloop);
+ pa_threaded_mainloop_lock(c->mainloop);
- stream = pa_stream_new (g->context, name, ss, map);
+ stream = pa_stream_new(c->context, name, ss, map);
if (!stream) {
goto fail;
}
- pa_stream_set_state_callback (stream, stream_state_cb, g);
- pa_stream_set_read_callback (stream, stream_request_cb, g);
- pa_stream_set_write_callback (stream, stream_request_cb, g);
+ pa_stream_set_state_callback (stream, stream_state_cb, c);
+ pa_stream_set_read_callback (stream, stream_request_cb, c);
+ pa_stream_set_write_callback (stream, stream_request_cb, c);
+
+ flags =
+ PA_STREAM_INTERPOLATE_TIMING
+#ifdef PA_STREAM_ADJUST_LATENCY
+ |PA_STREAM_ADJUST_LATENCY
+#endif
+ |PA_STREAM_AUTO_TIMING_UPDATE;
if (dir == PA_STREAM_PLAYBACK) {
- r = pa_stream_connect_playback (stream, dev, attr,
- PA_STREAM_INTERPOLATE_TIMING
-#ifdef PA_STREAM_ADJUST_LATENCY
- |PA_STREAM_ADJUST_LATENCY
-#endif
- |PA_STREAM_AUTO_TIMING_UPDATE, NULL, NULL);
+ r = pa_stream_connect_playback(stream, dev, attr, flags, NULL, NULL);
} else {
- r = pa_stream_connect_record (stream, dev, attr,
- PA_STREAM_INTERPOLATE_TIMING
-#ifdef PA_STREAM_ADJUST_LATENCY
- |PA_STREAM_ADJUST_LATENCY
-#endif
- |PA_STREAM_AUTO_TIMING_UPDATE);
+ r = pa_stream_connect_record(stream, dev, attr, flags);
}
if (r < 0) {
goto fail;
}
- pa_threaded_mainloop_unlock (g->mainloop);
+ pa_threaded_mainloop_unlock(c->mainloop);
return stream;
fail:
- pa_threaded_mainloop_unlock (g->mainloop);
+ pa_threaded_mainloop_unlock(c->mainloop);
if (stream) {
pa_stream_unref (stream);
}
- *rerror = pa_context_errno (g->context);
+ *rerror = pa_context_errno(c->context);
return NULL;
}
@@ -544,6 +553,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
paaudio *g = pa->g = drv_opaque;
AudiodevPaOptions *popts = &g->dev->u.pa;
AudiodevPaPerDirectionOptions *ppdo = popts->sink;
+ PAConnection *c = g->conn;
ss.format = audfmt_to_pa (as->fmt, as->endianness);
ss.channels = as->nchannels;
@@ -561,7 +571,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);
pa->stream = qpa_simple_new (
- g,
+ c,
"qemu",
PA_STREAM_PLAYBACK,
ppdo->has_name ? ppdo->name : NULL,
@@ -613,6 +623,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
paaudio *g = pa->g = drv_opaque;
AudiodevPaOptions *popts = &g->dev->u.pa;
AudiodevPaPerDirectionOptions *ppdo = popts->source;
+ PAConnection *c = g->conn;
ss.format = audfmt_to_pa (as->fmt, as->endianness);
ss.channels = as->nchannels;
@@ -621,7 +632,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
obt_as.fmt = pa_to_audfmt (ss.format, &obt_as.endianness);
pa->stream = qpa_simple_new (
- g,
+ c,
"qemu",
PA_STREAM_RECORD,
ppdo->has_name ? ppdo->name : NULL,
@@ -709,7 +720,7 @@ static int qpa_ctl_out (HWVoiceOut *hw, int cmd, ...)
PAVoiceOut *pa = (PAVoiceOut *) hw;
pa_operation *op;
pa_cvolume v;
- paaudio *g = pa->g;
+ PAConnection *c = pa->g->conn;
#ifdef PA_CHECK_VERSION /* macro is present in 0.9.16+ */
pa_cvolume_init (&v); /* function is present in 0.9.13+ */
@@ -729,28 +740,28 @@ static int qpa_ctl_out (HWVoiceOut *hw, int cmd, ...)
v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.l) / UINT32_MAX;
v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.r) / UINT32_MAX;
- pa_threaded_mainloop_lock (g->mainloop);
+ pa_threaded_mainloop_lock(c->mainloop);
- op = pa_context_set_sink_input_volume (g->context,
+ op = pa_context_set_sink_input_volume(c->context,
pa_stream_get_index (pa->stream),
&v, NULL, NULL);
if (!op)
- qpa_logerr (pa_context_errno (g->context),
+ qpa_logerr (pa_context_errno(c->context),
"set_sink_input_volume() failed\n");
else
pa_operation_unref (op);
- op = pa_context_set_sink_input_mute (g->context,
+ op = pa_context_set_sink_input_mute(c->context,
pa_stream_get_index (pa->stream),
sw->vol.mute, NULL, NULL);
if (!op) {
- qpa_logerr (pa_context_errno (g->context),
+ qpa_logerr (pa_context_errno(c->context),
"set_sink_input_mute() failed\n");
} else {
pa_operation_unref (op);
}
- pa_threaded_mainloop_unlock (g->mainloop);
+ pa_threaded_mainloop_unlock(c->mainloop);
}
}
return 0;
@@ -761,7 +772,7 @@ static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...)
PAVoiceIn *pa = (PAVoiceIn *) hw;
pa_operation *op;
pa_cvolume v;
- paaudio *g = pa->g;
+ PAConnection *c = pa->g->conn;
#ifdef PA_CHECK_VERSION
pa_cvolume_init (&v);
@@ -781,122 +792,156 @@ static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...)
v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.l) / UINT32_MAX;
v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.r) / UINT32_MAX;
- pa_threaded_mainloop_lock (g->mainloop);
+ pa_threaded_mainloop_lock(c->mainloop);
- op = pa_context_set_source_output_volume (g->context,
- pa_stream_get_index (pa->stream),
+ op = pa_context_set_source_output_volume(c->context,
+ pa_stream_get_index(pa->stream),
&v, NULL, NULL);
if (!op) {
- qpa_logerr (pa_context_errno (g->context),
- "set_source_output_volume() failed\n");
+ qpa_logerr(pa_context_errno(c->context),
+ "set_source_output_volume() failed\n");
} else {
pa_operation_unref(op);
}
- op = pa_context_set_source_output_mute (g->context,
+ op = pa_context_set_source_output_mute(c->context,
pa_stream_get_index (pa->stream),
sw->vol.mute, NULL, NULL);
if (!op) {
- qpa_logerr (pa_context_errno (g->context),
- "set_source_output_mute() failed\n");
+ qpa_logerr(pa_context_errno(c->context),
+ "set_source_output_mute() failed\n");
} else {
pa_operation_unref (op);
}
- pa_threaded_mainloop_unlock (g->mainloop);
+ pa_threaded_mainloop_unlock(c->mainloop);
}
}
return 0;
}
/* common */
+static void *qpa_conn_init(const char *server)
+{
+ PAConnection *c = g_malloc0(sizeof(PAConnection));
+ QTAILQ_INSERT_TAIL(&pa_conns, c, list);
+
+ c->mainloop = pa_threaded_mainloop_new();
+ if (!c->mainloop) {
+ goto fail;
+ }
+
+ c->context = pa_context_new(pa_threaded_mainloop_get_api(c->mainloop),
+ server);
+ if (!c->context) {
+ goto fail;
+ }
+
+ pa_context_set_state_callback(c->context, context_state_cb, c);
+
+ if (pa_context_connect(c->context, server, 0, NULL) < 0) {
+ qpa_logerr(pa_context_errno(c->context),
+ "pa_context_connect() failed\n");
+ goto fail;
+ }
+
+ pa_threaded_mainloop_lock(c->mainloop);
+
+ if (pa_threaded_mainloop_start(c->mainloop) < 0) {
+ goto unlock_and_fail;
+ }
+
+ for (;;) {
+ pa_context_state_t state;
+
+ state = pa_context_get_state(c->context);
+
+ if (state == PA_CONTEXT_READY) {
+ break;
+ }
+
+ if (!PA_CONTEXT_IS_GOOD (state)) {
+ qpa_logerr(pa_context_errno(c->context),
+ "Wrong context state\n");
+ goto unlock_and_fail;
+ }
+
+ /* Wait until the context is ready */
+ pa_threaded_mainloop_wait(c->mainloop);
+ }
+
+ pa_threaded_mainloop_unlock(c->mainloop);
+ return c;
+
+unlock_and_fail:
+ pa_threaded_mainloop_unlock(c->mainloop);
+fail:
+ AUD_log (AUDIO_CAP, "Failed to initialize PA context");
+ qpa_conn_fini(c);
+ return NULL;
+}
+
static void *qpa_audio_init(Audiodev *dev)
{
paaudio *g;
AudiodevPaOptions *popts;
const char *server;
+ PAConnection *c;
assert(dev->driver == AUDIODEV_DRIVER_PA);
- g = g_malloc(sizeof(paaudio));
+ g = g_malloc0(sizeof(paaudio));
popts = &dev->u.pa;
server = popts->has_server ? popts->server : NULL;
g->dev = dev;
- g->mainloop = NULL;
- g->context = NULL;
- g->mainloop = pa_threaded_mainloop_new ();
- if (!g->mainloop) {
- goto fail;
- }
-
- g->context = pa_context_new (pa_threaded_mainloop_get_api (g->mainloop),
- server);
- if (!g->context) {
- goto fail;
- }
-
- pa_context_set_state_callback (g->context, context_state_cb, g);
-
- if (pa_context_connect (g->context, server, 0, NULL) < 0) {
- qpa_logerr (pa_context_errno (g->context),
- "pa_context_connect() failed\n");
- goto fail;
- }
-
- pa_threaded_mainloop_lock (g->mainloop);
-
- if (pa_threaded_mainloop_start (g->mainloop) < 0) {
- goto unlock_and_fail;
- }
-
- for (;;) {
- pa_context_state_t state;
-
- state = pa_context_get_state (g->context);
-
- if (state == PA_CONTEXT_READY) {
+ QTAILQ_FOREACH(c, &pa_conns, list) {
+ if (server == NULL || c->server == NULL ?
+ server == c->server :
+ strcmp(server, c->server) == 0) {
+ g->conn = c;
break;
}
-
- if (!PA_CONTEXT_IS_GOOD (state)) {
- qpa_logerr (pa_context_errno (g->context),
- "Wrong context state\n");
- goto unlock_and_fail;
- }
-
- /* Wait until the context is ready */
- pa_threaded_mainloop_wait (g->mainloop);
+ }
+ if (!g->conn) {
+ g->conn = qpa_conn_init(server);
+ }
+ if (!g->conn) {
+ g_free(g);
+ return NULL;
}
- pa_threaded_mainloop_unlock (g->mainloop);
-
+ ++g->conn->refcount;
return g;
+}
-unlock_and_fail:
- pa_threaded_mainloop_unlock (g->mainloop);
-fail:
- AUD_log (AUDIO_CAP, "Failed to initialize PA context");
- qpa_audio_fini(g);
- return NULL;
+static void qpa_conn_fini(PAConnection *c)
+{
+ if (c->mainloop) {
+ pa_threaded_mainloop_stop(c->mainloop);
+ }
+
+ if (c->context) {
+ pa_context_disconnect(c->context);
+ pa_context_unref(c->context);
+ }
+
+ if (c->mainloop) {
+ pa_threaded_mainloop_free(c->mainloop);
+ }
+
+ QTAILQ_REMOVE(&pa_conns, c, list);
+ g_free(c);
}
static void qpa_audio_fini (void *opaque)
{
paaudio *g = opaque;
+ PAConnection *c = g->conn;
- if (g->mainloop) {
- pa_threaded_mainloop_stop (g->mainloop);
- }
-
- if (g->context) {
- pa_context_disconnect (g->context);
- pa_context_unref (g->context);
- }
-
- if (g->mainloop) {
- pa_threaded_mainloop_free (g->mainloop);
+ if (--c->refcount == 0) {
+ qpa_conn_fini(c);
}
g_free(g);
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 21/52] paaudio: do not move stream when sink/source name is specified
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (19 preceding siblings ...)
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 20/52] audio: audiodev= parameters no longer optional when -audiodev present Kővágó, Zoltán
@ 2018-12-23 20:51 ` Kővágó, Zoltán
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 22/52] paaudio: properly disconnect streams in fini_* Kővágó, Zoltán
` (29 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:51 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Unless we disable stream moving, pulseaudio can easily move the stream
on connect, effectively ignoring the source/sink specified by the user.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/paaudio.c | 5 +++++
1 file changed, 5 insertions(+)
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 24b75ca503..21ffbf7f6e 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -516,6 +516,11 @@ static pa_stream *qpa_simple_new (
#endif
|PA_STREAM_AUTO_TIMING_UPDATE;
+ if (dev) {
+ /* don't move the stream if the user specified a sink/source */
+ flags |= PA_STREAM_DONT_MOVE;
+ }
+
if (dir == PA_STREAM_PLAYBACK) {
r = pa_stream_connect_playback(stream, dev, attr, flags, NULL, NULL);
} else {
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 22/52] paaudio: properly disconnect streams in fini_*
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (20 preceding siblings ...)
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 21/52] paaudio: do not move stream when sink/source name is specified Kővágó, Zoltán
@ 2018-12-23 20:51 ` Kővágó, Zoltán
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 23/52] audio: remove audio_MIN, audio_MAX Kővágó, Zoltán
` (28 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:51 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann, Marc-André Lureau
Currently this needs a workaround due to bug #74624 in pulseaudio.
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/paaudio.c | 25 +++++++++++++++++++++++--
1 file changed, 23 insertions(+), 2 deletions(-)
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 21ffbf7f6e..fa867a8065 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -680,6 +680,27 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
return -1;
}
+static void qpa_simple_disconnect(PAConnection *c, pa_stream *stream)
+{
+ int err;
+
+ pa_threaded_mainloop_lock(c->mainloop);
+ /*
+ * wait until actually connects. workaround pa bug #74624
+ * https://bugs.freedesktop.org/show_bug.cgi?id=74624
+ */
+ while (pa_stream_get_state(stream) == PA_STREAM_CREATING) {
+ pa_threaded_mainloop_wait(c->mainloop);
+ }
+
+ err = pa_stream_disconnect(stream);
+ if (err != 0) {
+ dolog("Failed to dissconnect! err=%d\n", err);
+ }
+ pa_stream_unref(stream);
+ pa_threaded_mainloop_unlock(c->mainloop);
+}
+
static void qpa_fini_out (HWVoiceOut *hw)
{
void *ret;
@@ -691,7 +712,7 @@ static void qpa_fini_out (HWVoiceOut *hw)
audio_pt_join(&pa->pt, &ret, __func__);
if (pa->stream) {
- pa_stream_unref (pa->stream);
+ qpa_simple_disconnect(pa->g->conn, pa->stream);
pa->stream = NULL;
}
@@ -711,7 +732,7 @@ static void qpa_fini_in (HWVoiceIn *hw)
audio_pt_join(&pa->pt, &ret, __func__);
if (pa->stream) {
- pa_stream_unref (pa->stream);
+ qpa_simple_disconnect(pa->g->conn, pa->stream);
pa->stream = NULL;
}
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 23/52] audio: remove audio_MIN, audio_MAX
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (21 preceding siblings ...)
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 22/52] paaudio: properly disconnect streams in fini_* Kővágó, Zoltán
@ 2018-12-23 20:51 ` Kővágó, Zoltán
2018-12-23 23:49 ` Philippe Mathieu-Daudé
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 24/52] audio: do not run each backend in audio_run Kővágó, Zoltán
` (27 subsequent siblings)
50 siblings, 1 reply; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:51 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann, Michael Walle
There's already a MIN and MAX macro in include/qemu/osdep.h, use them
instead.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
Changes from v1:
* removed audio_MIN, audio_MAX macros
---
audio/alsaaudio.c | 6 +++---
audio/audio.c | 20 ++++++++++----------
audio/audio.h | 17 -----------------
audio/coreaudio.c | 2 +-
audio/dsoundaudio.c | 2 +-
audio/noaudio.c | 10 +++++-----
audio/ossaudio.c | 6 +++---
audio/paaudio.c | 12 ++++++------
audio/sdlaudio.c | 6 +++---
audio/spiceaudio.c | 10 +++++-----
audio/wavaudio.c | 4 ++--
hw/audio/ac97.c | 10 +++++-----
hw/audio/adlib.c | 4 ++--
hw/audio/cs4231a.c | 4 ++--
hw/audio/es1370.c | 6 +++---
hw/audio/gus.c | 6 +++---
hw/audio/hda-codec.c | 16 ++++++++--------
hw/audio/milkymist-ac97.c | 8 ++++----
hw/audio/pcspk.c | 2 +-
hw/audio/sb16.c | 2 +-
hw/audio/wm8750.c | 4 ++--
21 files changed, 70 insertions(+), 87 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 6f75644538..d4bb540cb9 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -642,7 +642,7 @@ static void alsa_write_pending (ALSAVoiceOut *alsa)
while (alsa->pending) {
int left_till_end_samples = hw->samples - alsa->wpos;
- int len = audio_MIN (alsa->pending, left_till_end_samples);
+ int len = MIN (alsa->pending, left_till_end_samples);
char *src = advance (alsa->pcm_buf, alsa->wpos << hw->info.shift);
while (len) {
@@ -705,7 +705,7 @@ static int alsa_run_out (HWVoiceOut *hw, int live)
return 0;
}
- decr = audio_MIN (live, avail);
+ decr = MIN (live, avail);
decr = audio_pcm_hw_clip_out (hw, alsa->pcm_buf, decr, alsa->pending);
alsa->pending += decr;
alsa_write_pending (alsa);
@@ -923,7 +923,7 @@ static int alsa_run_in (HWVoiceIn *hw)
}
}
- decr = audio_MIN (dead, avail);
+ decr = MIN (dead, avail);
if (!decr) {
return 0;
}
diff --git a/audio/audio.c b/audio/audio.c
index a7f79f3929..a32b86a813 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -537,7 +537,7 @@ static int audio_pcm_hw_find_min_in (HWVoiceIn *hw)
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
if (sw->active) {
- m = audio_MIN (m, sw->total_hw_samples_acquired);
+ m = MIN (m, sw->total_hw_samples_acquired);
}
}
return m;
@@ -557,14 +557,14 @@ int audio_pcm_hw_clip_out (HWVoiceOut *hw, void *pcm_buf,
int live, int pending)
{
int left = hw->samples - pending;
- int len = audio_MIN (left, live);
+ int len = MIN (left, live);
int clipped = 0;
while (len) {
struct st_sample *src = hw->mix_buf + hw->rpos;
uint8_t *dst = advance (pcm_buf, hw->rpos << hw->info.shift);
int samples_till_end_of_buf = hw->samples - hw->rpos;
- int samples_to_clip = audio_MIN (len, samples_till_end_of_buf);
+ int samples_to_clip = MIN (len, samples_till_end_of_buf);
hw->clip (dst, src, samples_to_clip);
@@ -618,7 +618,7 @@ int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int size)
}
swlim = (live * sw->ratio) >> 32;
- swlim = audio_MIN (swlim, samples);
+ swlim = MIN (swlim, samples);
while (swlim) {
src = hw->conv_buf + rpos;
@@ -666,7 +666,7 @@ static int audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
if (sw->active || !sw->empty) {
- m = audio_MIN (m, sw->total_hw_samples_mixed);
+ m = MIN (m, sw->total_hw_samples_mixed);
nb_live += 1;
}
}
@@ -729,7 +729,7 @@ int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
dead = hwsamples - live;
swlim = ((int64_t) dead << 32) / sw->ratio;
- swlim = audio_MIN (swlim, samples);
+ swlim = MIN (swlim, samples);
if (swlim) {
sw->conv (sw->buf, buf, swlim);
@@ -741,7 +741,7 @@ int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
while (swlim) {
dead = hwsamples - live;
left = hwsamples - wpos;
- blck = audio_MIN (dead, left);
+ blck = MIN (dead, left);
if (!blck) {
break;
}
@@ -1033,7 +1033,7 @@ static void audio_capture_mix_and_clear (HWVoiceOut *hw, int rpos, int samples)
n = samples;
while (n) {
int till_end_of_hw = hw->samples - rpos2;
- int to_write = audio_MIN (till_end_of_hw, n);
+ int to_write = MIN (till_end_of_hw, n);
int bytes = to_write << hw->info.shift;
int written;
@@ -1051,7 +1051,7 @@ static void audio_capture_mix_and_clear (HWVoiceOut *hw, int rpos, int samples)
}
}
- n = audio_MIN (samples, hw->samples - rpos);
+ n = MIN (samples, hw->samples - rpos);
mixeng_clear (hw->mix_buf + rpos, n);
mixeng_clear (hw->mix_buf, samples - n);
}
@@ -1207,7 +1207,7 @@ static void audio_run_capture (AudioState *s)
rpos = hw->rpos;
while (live) {
int left = hw->samples - rpos;
- int to_capture = audio_MIN (live, left);
+ int to_capture = MIN (live, left);
struct st_sample *src;
struct capture_callback *cb;
diff --git a/audio/audio.h b/audio/audio.h
index ccfef9e10a..bcbe56d639 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -148,23 +148,6 @@ static inline void *advance (void *p, int incr)
return (d + incr);
}
-#ifdef __GNUC__
-#define audio_MIN(a, b) ( __extension__ ({ \
- __typeof (a) ta = a; \
- __typeof (b) tb = b; \
- ((ta)>(tb)?(tb):(ta)); \
-}))
-
-#define audio_MAX(a, b) ( __extension__ ({ \
- __typeof (a) ta = a; \
- __typeof (b) tb = b; \
- ((ta)<(tb)?(tb):(ta)); \
-}))
-#else
-#define audio_MIN(a, b) ((a)>(b)?(b):(a))
-#define audio_MAX(a, b) ((a)<(b)?(b):(a))
-#endif
-
int wav_start_capture(AudioState *state, CaptureState *s, const char *path,
int freq, int bits, int nchannels);
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index e00b8847d7..b6935359ee 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -413,7 +413,7 @@ static int coreaudio_run_out (HWVoiceOut *hw, int live)
core->live);
}
- decr = audio_MIN (core->decr, live);
+ decr = MIN (core->decr, live);
core->decr -= decr;
core->live = live - decr;
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index a7d04b5033..6f074c6f94 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -707,7 +707,7 @@ static int dsound_run_in (HWVoiceIn *hw)
if (!len) {
return 0;
}
- len = audio_MIN (len, dead);
+ len = MIN (len, dead);
err = dsound_lock_in (
dscb,
diff --git a/audio/noaudio.c b/audio/noaudio.c
index f1eb048d80..299ba3d09c 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -51,11 +51,11 @@ static int no_run_out (HWVoiceOut *hw, int live)
now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
ticks = now - no->old_ticks;
bytes = muldiv64(ticks, hw->info.bytes_per_second, NANOSECONDS_PER_SECOND);
- bytes = audio_MIN(bytes, INT_MAX);
+ bytes = MIN(bytes, INT_MAX);
samples = bytes >> hw->info.shift;
no->old_ticks = now;
- decr = audio_MIN (live, samples);
+ decr = MIN (live, samples);
hw->rpos = (hw->rpos + decr) % hw->samples;
return decr;
}
@@ -110,9 +110,9 @@ static int no_run_in (HWVoiceIn *hw)
muldiv64(ticks, hw->info.bytes_per_second, NANOSECONDS_PER_SECOND);
no->old_ticks = now;
- bytes = audio_MIN (bytes, INT_MAX);
+ bytes = MIN (bytes, INT_MAX);
samples = bytes >> hw->info.shift;
- samples = audio_MIN (samples, dead);
+ samples = MIN (samples, dead);
}
return samples;
}
@@ -123,7 +123,7 @@ static int no_read (SWVoiceIn *sw, void *buf, int size)
* useless resampling/mixing */
int samples = size >> sw->info.shift;
int total = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
- int to_clear = audio_MIN (samples, total);
+ int to_clear = MIN (samples, total);
sw->total_hw_samples_acquired += total;
audio_pcm_info_clear_buf (&sw->info, buf, to_clear);
return to_clear << sw->info.shift;
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 83ff1d2ebd..ae06e58d13 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -388,7 +388,7 @@ static void oss_write_pending (OSSVoiceOut *oss)
int samples_written;
ssize_t bytes_written;
int samples_till_end = hw->samples - oss->wpos;
- int samples_to_write = audio_MIN (oss->pending, samples_till_end);
+ int samples_to_write = MIN (oss->pending, samples_till_end);
int bytes_to_write = samples_to_write << hw->info.shift;
void *pcm = advance (oss->pcm_buf, oss->wpos << hw->info.shift);
@@ -437,7 +437,7 @@ static int oss_run_out (HWVoiceOut *hw, int live)
pos = hw->rpos << hw->info.shift;
bytes = audio_ring_dist (cntinfo.ptr, pos, bufsize);
- decr = audio_MIN (bytes >> hw->info.shift, live);
+ decr = MIN (bytes >> hw->info.shift, live);
}
else {
err = ioctl (oss->fd, SNDCTL_DSP_GETOSPACE, &abinfo);
@@ -456,7 +456,7 @@ static int oss_run_out (HWVoiceOut *hw, int live)
return 0;
}
- decr = audio_MIN (abinfo.bytes >> hw->info.shift, live);
+ decr = MIN (abinfo.bytes >> hw->info.shift, live);
if (!decr) {
return 0;
}
diff --git a/audio/paaudio.c b/audio/paaudio.c
index fa867a8065..6ed12851c0 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -234,7 +234,7 @@ static void *qpa_thread_out (void *arg)
}
}
- decr = to_mix = audio_MIN(pa->live, pa->samples >> 5);
+ decr = to_mix = MIN(pa->live, pa->samples >> 5);
rpos = pa->rpos;
if (audio_pt_unlock(&pa->pt, __func__)) {
@@ -243,7 +243,7 @@ static void *qpa_thread_out (void *arg)
while (to_mix) {
int error;
- int chunk = audio_MIN (to_mix, hw->samples - rpos);
+ int chunk = MIN (to_mix, hw->samples - rpos);
struct st_sample *src = hw->mix_buf + rpos;
hw->clip (pa->pcm_buf, src, chunk);
@@ -281,7 +281,7 @@ static int qpa_run_out (HWVoiceOut *hw, int live)
return 0;
}
- decr = audio_MIN (live, pa->decr);
+ decr = MIN (live, pa->decr);
pa->decr -= decr;
pa->live = live - decr;
hw->rpos = pa->rpos;
@@ -326,7 +326,7 @@ static void *qpa_thread_in (void *arg)
}
}
- incr = to_grab = audio_MIN(pa->dead, pa->samples >> 5);
+ incr = to_grab = MIN(pa->dead, pa->samples >> 5);
wpos = pa->wpos;
if (audio_pt_unlock(&pa->pt, __func__)) {
@@ -335,7 +335,7 @@ static void *qpa_thread_in (void *arg)
while (to_grab) {
int error;
- int chunk = audio_MIN (to_grab, hw->samples - wpos);
+ int chunk = MIN (to_grab, hw->samples - wpos);
void *buf = advance (pa->pcm_buf, wpos);
if (qpa_simple_read (pa, buf,
@@ -374,7 +374,7 @@ static int qpa_run_in (HWVoiceIn *hw)
live = audio_pcm_hw_get_live_in (hw);
dead = hw->samples - live;
- incr = audio_MIN (dead, pa->incr);
+ incr = MIN (dead, pa->incr);
pa->incr -= incr;
pa->dead = dead - incr;
hw->wpos = pa->wpos;
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index cf6ac19927..fc61038b77 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -288,10 +288,10 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
#endif
/* dolog ("in callback live=%d\n", live); */
- to_mix = audio_MIN (samples, sdl->live);
+ to_mix = MIN (samples, sdl->live);
decr = to_mix;
while (to_mix) {
- int chunk = audio_MIN (to_mix, hw->samples - hw->rpos);
+ int chunk = MIN (to_mix, hw->samples - hw->rpos);
struct st_sample *src = hw->mix_buf + hw->rpos;
/* dolog ("in callback to_mix %d, chunk %d\n", to_mix, chunk); */
@@ -347,7 +347,7 @@ static int sdl_run_out (HWVoiceOut *hw, int live)
sdl->live);
}
- decr = audio_MIN (sdl->decr, live);
+ decr = MIN (sdl->decr, live);
sdl->decr -= decr;
#if USE_SEMAPHORE
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index 4f7873af5a..b9160991d2 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -163,20 +163,20 @@ static int line_out_run (HWVoiceOut *hw, int live)
}
decr = rate_get_samples (&hw->info, &out->rate);
- decr = audio_MIN (live, decr);
+ decr = MIN (live, decr);
samples = decr;
rpos = hw->rpos;
while (samples) {
int left_till_end_samples = hw->samples - rpos;
- int len = audio_MIN (samples, left_till_end_samples);
+ int len = MIN (samples, left_till_end_samples);
if (!out->frame) {
spice_server_playback_get_buffer (&out->sin, &out->frame, &out->fsize);
out->fpos = out->frame;
}
if (out->frame) {
- len = audio_MIN (len, out->fsize);
+ len = MIN (len, out->fsize);
hw->clip (out->fpos, hw->mix_buf + rpos, len);
out->fsize -= len;
out->fpos += len;
@@ -294,7 +294,7 @@ static int line_in_run (HWVoiceIn *hw)
}
delta_samp = rate_get_samples (&hw->info, &in->rate);
- num_samples = audio_MIN (num_samples, delta_samp);
+ num_samples = MIN (num_samples, delta_samp);
ready = spice_server_record_get_samples (&in->sin, in->samples, num_samples);
samples = in->samples;
@@ -304,7 +304,7 @@ static int line_in_run (HWVoiceIn *hw)
ready = LINE_IN_SAMPLES;
}
- num_samples = audio_MIN (ready, num_samples);
+ num_samples = MIN (ready, num_samples);
if (hw->wpos + num_samples > hw->samples) {
len[0] = hw->samples - hw->wpos;
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 214e30ccd9..723028a466 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -57,12 +57,12 @@ static int wav_run_out (HWVoiceOut *hw, int live)
}
wav->old_ticks = now;
- decr = audio_MIN (live, samples);
+ decr = MIN (live, samples);
samples = decr;
rpos = hw->rpos;
while (samples) {
int left_till_end_samples = hw->samples - rpos;
- int convert_samples = audio_MIN (samples, left_till_end_samples);
+ int convert_samples = MIN (samples, left_till_end_samples);
src = hw->mix_buf + rpos;
dst = advance (wav->pcm_buf, rpos << hw->info.shift);
diff --git a/hw/audio/ac97.c b/hw/audio/ac97.c
index 03e3da4f31..1211541888 100644
--- a/hw/audio/ac97.c
+++ b/hw/audio/ac97.c
@@ -963,7 +963,7 @@ static int write_audio (AC97LinkState *s, AC97BusMasterRegs *r,
uint32_t temp = r->picb << 1;
uint32_t written = 0;
int to_copy = 0;
- temp = audio_MIN (temp, max);
+ temp = MIN (temp, max);
if (!temp) {
*stop = 1;
@@ -972,7 +972,7 @@ static int write_audio (AC97LinkState *s, AC97BusMasterRegs *r,
while (temp) {
int copied;
- to_copy = audio_MIN (temp, sizeof (tmpbuf));
+ to_copy = MIN (temp, sizeof (tmpbuf));
pci_dma_read (&s->dev, addr, tmpbuf, to_copy);
copied = AUD_write (s->voice_po, tmpbuf, to_copy);
dolog ("write_audio max=%x to_copy=%x copied=%x\n",
@@ -1018,7 +1018,7 @@ static void write_bup (AC97LinkState *s, int elapsed)
}
while (elapsed) {
- int temp = audio_MIN (elapsed, sizeof (s->silence));
+ int temp = MIN (elapsed, sizeof (s->silence));
while (temp) {
int copied = AUD_write (s->voice_po, s->silence, temp);
if (!copied)
@@ -1039,7 +1039,7 @@ static int read_audio (AC97LinkState *s, AC97BusMasterRegs *r,
int to_copy = 0;
SWVoiceIn *voice = (r - s->bm_regs) == MC_INDEX ? s->voice_mc : s->voice_pi;
- temp = audio_MIN (temp, max);
+ temp = MIN (temp, max);
if (!temp) {
*stop = 1;
@@ -1048,7 +1048,7 @@ static int read_audio (AC97LinkState *s, AC97BusMasterRegs *r,
while (temp) {
int acquired;
- to_copy = audio_MIN (temp, sizeof (tmpbuf));
+ to_copy = MIN (temp, sizeof (tmpbuf));
acquired = AUD_read (voice, tmpbuf, to_copy);
if (!acquired) {
*stop = 1;
diff --git a/hw/audio/adlib.c b/hw/audio/adlib.c
index 0d01cd07c5..06350abf2c 100644
--- a/hw/audio/adlib.c
+++ b/hw/audio/adlib.c
@@ -194,7 +194,7 @@ static void adlib_callback (void *opaque, int free)
return;
}
- to_play = audio_MIN (s->left, samples);
+ to_play = MIN (s->left, samples);
while (to_play) {
written = write_audio (s, to_play);
@@ -209,7 +209,7 @@ static void adlib_callback (void *opaque, int free)
}
}
- samples = audio_MIN (samples, s->samples - s->pos);
+ samples = MIN (samples, s->samples - s->pos);
if (!samples) {
return;
}
diff --git a/hw/audio/cs4231a.c b/hw/audio/cs4231a.c
index d25a120b0f..70d0947907 100644
--- a/hw/audio/cs4231a.c
+++ b/hw/audio/cs4231a.c
@@ -533,7 +533,7 @@ static int cs_write_audio (CSState *s, int nchan, int dma_pos,
int copied;
size_t to_copy;
- to_copy = audio_MIN (temp, left);
+ to_copy = MIN (temp, left);
if (to_copy > sizeof (tmpbuf)) {
to_copy = sizeof (tmpbuf);
}
@@ -576,7 +576,7 @@ static int cs_dma_read (void *opaque, int nchan, int dma_pos, int dma_len)
till = (s->dregs[Playback_Lower_Base_Count]
| (s->dregs[Playback_Upper_Base_Count] << 8)) << s->shift;
till -= s->transferred;
- copy = audio_MIN (till, copy);
+ copy = MIN (till, copy);
}
if ((copy <= 0) || (dma_len <= 0)) {
diff --git a/hw/audio/es1370.c b/hw/audio/es1370.c
index 8c63912a82..234a05633e 100644
--- a/hw/audio/es1370.c
+++ b/hw/audio/es1370.c
@@ -644,7 +644,7 @@ static void es1370_transfer_audio (ES1370State *s, struct chan *d, int loop_sel,
int size = d->frame_cnt & 0xffff;
int left = ((size - cnt + 1) << 2) + d->leftover;
int transferred = 0;
- int temp = audio_MIN (max, audio_MIN (left, csc_bytes));
+ int temp = MIN (max, MIN (left, csc_bytes));
int index = d - &s->chan[0];
addr += (cnt << 2) + d->leftover;
@@ -653,7 +653,7 @@ static void es1370_transfer_audio (ES1370State *s, struct chan *d, int loop_sel,
while (temp) {
int acquired, to_copy;
- to_copy = audio_MIN ((size_t) temp, sizeof (tmpbuf));
+ to_copy = MIN ((size_t) temp, sizeof (tmpbuf));
acquired = AUD_read (s->adc_voice, tmpbuf, to_copy);
if (!acquired)
break;
@@ -671,7 +671,7 @@ static void es1370_transfer_audio (ES1370State *s, struct chan *d, int loop_sel,
while (temp) {
int copied, to_copy;
- to_copy = audio_MIN ((size_t) temp, sizeof (tmpbuf));
+ to_copy = MIN ((size_t) temp, sizeof (tmpbuf));
pci_dma_read (&s->dev, addr, tmpbuf, to_copy);
copied = AUD_write (voice, tmpbuf, to_copy);
if (!copied)
diff --git a/hw/audio/gus.c b/hw/audio/gus.c
index dfb74cf0d3..2a2a24eacc 100644
--- a/hw/audio/gus.c
+++ b/hw/audio/gus.c
@@ -115,7 +115,7 @@ static void GUS_callback (void *opaque, int free)
GUSState *s = opaque;
samples = free >> s->shift;
- to_play = audio_MIN (samples, s->left);
+ to_play = MIN (samples, s->left);
while (to_play) {
int written = write_audio (s, to_play);
@@ -130,7 +130,7 @@ static void GUS_callback (void *opaque, int free)
net += written;
}
- samples = audio_MIN (samples, s->samples);
+ samples = MIN (samples, s->samples);
if (samples) {
gus_mixvoices (&s->emu, s->freq, samples, s->mixbuf);
@@ -190,7 +190,7 @@ static int GUS_read_DMA (void *opaque, int nchan, int dma_pos, int dma_len)
ldebug ("read DMA %#x %d\n", dma_pos, dma_len);
mode = k->has_autoinitialization(s->isa_dma, s->emu.gusdma);
while (left) {
- int to_copy = audio_MIN ((size_t) left, sizeof (tmpbuf));
+ int to_copy = MIN ((size_t) left, sizeof (tmpbuf));
int copied;
ldebug ("left=%d to_copy=%d pos=%d\n", left, to_copy, pos);
diff --git a/hw/audio/hda-codec.c b/hw/audio/hda-codec.c
index e66103aedf..88106f88b4 100644
--- a/hw/audio/hda-codec.c
+++ b/hw/audio/hda-codec.c
@@ -233,10 +233,10 @@ static void hda_audio_input_timer(void *opaque)
goto out_timer;
}
- int64_t to_transfer = audio_MIN(wpos - rpos, wanted_rpos - rpos);
+ int64_t to_transfer = MIN(wpos - rpos, wanted_rpos - rpos);
while (to_transfer) {
uint32_t start = (rpos & B_MASK);
- uint32_t chunk = audio_MIN(B_SIZE - start, to_transfer);
+ uint32_t chunk = MIN(B_SIZE - start, to_transfer);
int rc = hda_codec_xfer(
&st->state->hda, st->stream, false, st->buf + start, chunk);
if (!rc) {
@@ -261,13 +261,13 @@ static void hda_audio_input_cb(void *opaque, int avail)
int64_t wpos = st->wpos;
int64_t rpos = st->rpos;
- int64_t to_transfer = audio_MIN(B_SIZE - (wpos - rpos), avail);
+ int64_t to_transfer = MIN(B_SIZE - (wpos - rpos), avail);
hda_timer_sync_adjust(st, -((wpos - rpos) + to_transfer - (B_SIZE >> 1)));
while (to_transfer) {
uint32_t start = (uint32_t) (wpos & B_MASK);
- uint32_t chunk = (uint32_t) audio_MIN(B_SIZE - start, to_transfer);
+ uint32_t chunk = (uint32_t) MIN(B_SIZE - start, to_transfer);
uint32_t read = AUD_read(st->voice.in, st->buf + start, chunk);
wpos += read;
to_transfer -= read;
@@ -297,10 +297,10 @@ static void hda_audio_output_timer(void *opaque)
goto out_timer;
}
- int64_t to_transfer = audio_MIN(B_SIZE - (wpos - rpos), wanted_wpos - wpos);
+ int64_t to_transfer = MIN(B_SIZE - (wpos - rpos), wanted_wpos - wpos);
while (to_transfer) {
uint32_t start = (wpos & B_MASK);
- uint32_t chunk = audio_MIN(B_SIZE - start, to_transfer);
+ uint32_t chunk = MIN(B_SIZE - start, to_transfer);
int rc = hda_codec_xfer(
&st->state->hda, st->stream, true, st->buf + start, chunk);
if (!rc) {
@@ -325,7 +325,7 @@ static void hda_audio_output_cb(void *opaque, int avail)
int64_t wpos = st->wpos;
int64_t rpos = st->rpos;
- int64_t to_transfer = audio_MIN(wpos - rpos, avail);
+ int64_t to_transfer = MIN(wpos - rpos, avail);
if (wpos - rpos == B_SIZE) {
/* drop buffer, reset timer adjust */
@@ -340,7 +340,7 @@ static void hda_audio_output_cb(void *opaque, int avail)
while (to_transfer) {
uint32_t start = (uint32_t) (rpos & B_MASK);
- uint32_t chunk = (uint32_t) audio_MIN(B_SIZE - start, to_transfer);
+ uint32_t chunk = (uint32_t) MIN(B_SIZE - start, to_transfer);
uint32_t written = AUD_write(st->voice.out, st->buf + start, chunk);
rpos += written;
to_transfer -= written;
diff --git a/hw/audio/milkymist-ac97.c b/hw/audio/milkymist-ac97.c
index 8739cb376a..1a583fc3b9 100644
--- a/hw/audio/milkymist-ac97.c
+++ b/hw/audio/milkymist-ac97.c
@@ -183,7 +183,7 @@ static void ac97_in_cb(void *opaque, int avail_b)
MilkymistAC97State *s = opaque;
uint8_t buf[4096];
uint32_t remaining = s->regs[R_U_REMAINING];
- int temp = audio_MIN(remaining, avail_b);
+ int temp = MIN(remaining, avail_b);
uint32_t addr = s->regs[R_U_ADDR];
int transferred = 0;
@@ -197,7 +197,7 @@ static void ac97_in_cb(void *opaque, int avail_b)
while (temp) {
int acquired, to_copy;
- to_copy = audio_MIN(temp, sizeof(buf));
+ to_copy = MIN(temp, sizeof(buf));
acquired = AUD_read(s->voice_in, buf, to_copy);
if (!acquired) {
break;
@@ -226,7 +226,7 @@ static void ac97_out_cb(void *opaque, int free_b)
MilkymistAC97State *s = opaque;
uint8_t buf[4096];
uint32_t remaining = s->regs[R_D_REMAINING];
- int temp = audio_MIN(remaining, free_b);
+ int temp = MIN(remaining, free_b);
uint32_t addr = s->regs[R_D_ADDR];
int transferred = 0;
@@ -240,7 +240,7 @@ static void ac97_out_cb(void *opaque, int free_b)
while (temp) {
int copied, to_copy;
- to_copy = audio_MIN(temp, sizeof(buf));
+ to_copy = MIN(temp, sizeof(buf));
cpu_physical_memory_read(addr, buf, to_copy);
copied = AUD_write(s->voice_out, buf, to_copy);
if (!copied) {
diff --git a/hw/audio/pcspk.c b/hw/audio/pcspk.c
index 1457cd1ac6..94530c9265 100644
--- a/hw/audio/pcspk.c
+++ b/hw/audio/pcspk.c
@@ -102,7 +102,7 @@ static void pcspk_callback(void *opaque, int free)
}
while (free > 0) {
- n = audio_MIN(s->samples - s->play_pos, (unsigned int)free);
+ n = MIN(s->samples - s->play_pos, (unsigned int)free);
n = AUD_write(s->voice, &s->sample_buf[s->play_pos], n);
if (!n)
break;
diff --git a/hw/audio/sb16.c b/hw/audio/sb16.c
index 5a6a880f2e..78bf0753bb 100644
--- a/hw/audio/sb16.c
+++ b/hw/audio/sb16.c
@@ -1166,7 +1166,7 @@ static int write_audio (SB16State *s, int nchan, int dma_pos,
int copied;
size_t to_copy;
- to_copy = audio_MIN (temp, left);
+ to_copy = MIN (temp, left);
if (to_copy > sizeof (tmpbuf)) {
to_copy = sizeof (tmpbuf);
}
diff --git a/hw/audio/wm8750.c b/hw/audio/wm8750.c
index c86fd91f98..13337b9b08 100644
--- a/hw/audio/wm8750.c
+++ b/hw/audio/wm8750.c
@@ -68,7 +68,7 @@ static inline void wm8750_in_load(WM8750State *s)
{
if (s->idx_in + s->req_in <= sizeof(s->data_in))
return;
- s->idx_in = audio_MAX(0, (int) sizeof(s->data_in) - s->req_in);
+ s->idx_in = MAX(0, (int) sizeof(s->data_in) - s->req_in);
AUD_read(*s->in[0], s->data_in + s->idx_in,
sizeof(s->data_in) - s->idx_in);
}
@@ -99,7 +99,7 @@ static void wm8750_audio_out_cb(void *opaque, int free_b)
wm8750_out_flush(s);
} else
s->req_out = free_b - s->idx_out;
-
+
s->data_req(s->opaque, s->req_out >> 2, s->req_in >> 2);
}
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* Re: [Qemu-devel] [PATCH v2 23/52] audio: remove audio_MIN, audio_MAX
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 23/52] audio: remove audio_MIN, audio_MAX Kővágó, Zoltán
@ 2018-12-23 23:49 ` Philippe Mathieu-Daudé
2018-12-24 2:16 ` Zoltán Kővágó
0 siblings, 1 reply; 91+ messages in thread
From: Philippe Mathieu-Daudé @ 2018-12-23 23:49 UTC (permalink / raw)
To: Kővágó, Zoltán, qemu-devel, Paolo Bonzini
Cc: Michael Walle, Gerd Hoffmann
Hi Zoltán,
On 12/23/18 9:51 PM, Kővágó, Zoltán wrote:
> There's already a MIN and MAX macro in include/qemu/osdep.h, use them
> instead.
>
> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
>
> ---
>
> Changes from v1:
> * removed audio_MIN, audio_MAX macros
> ---
[...]>
> diff --git a/audio/audio.h b/audio/audio.h
> index ccfef9e10a..bcbe56d639 100644
> --- a/audio/audio.h
> +++ b/audio/audio.h
> @@ -148,23 +148,6 @@ static inline void *advance (void *p, int incr)
> return (d + incr);
> }
>
> -#ifdef __GNUC__
> -#define audio_MIN(a, b) ( __extension__ ({ \
> - __typeof (a) ta = a; \
> - __typeof (b) tb = b; \
> - ((ta)>(tb)?(tb):(ta)); \
> -}))
> -
> -#define audio_MAX(a, b) ( __extension__ ({ \
> - __typeof (a) ta = a; \
> - __typeof (b) tb = b; \
> - ((ta)<(tb)?(tb):(ta)); \
> -}))
> -#else
> -#define audio_MIN(a, b) ((a)>(b)?(b):(a))
> -#define audio_MAX(a, b) ((a)<(b)?(b):(a))
> -#endif
> -
Those MIN/MAX are smarter than the ones in "qemu/osdep.h", I'd keep them
moving them there.
Regards,
Phil.
^ permalink raw reply [flat|nested] 91+ messages in thread* Re: [Qemu-devel] [PATCH v2 23/52] audio: remove audio_MIN, audio_MAX
2018-12-23 23:49 ` Philippe Mathieu-Daudé
@ 2018-12-24 2:16 ` Zoltán Kővágó
2018-12-24 17:16 ` Philippe Mathieu-Daudé
0 siblings, 1 reply; 91+ messages in thread
From: Zoltán Kővágó @ 2018-12-24 2:16 UTC (permalink / raw)
To: Philippe Mathieu-Daudé, qemu-devel, Paolo Bonzini
Cc: Michael Walle, Gerd Hoffmann
Hi Phil,
On 2018-12-24 00:49, Philippe Mathieu-Daudé wrote:
> Hi Zoltán,
>
> On 12/23/18 9:51 PM, Kővágó, Zoltán wrote:
>> There's already a MIN and MAX macro in include/qemu/osdep.h, use them
>> instead.
>>
>> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
>>
>> ---
>>
>> Changes from v1:
>> * removed audio_MIN, audio_MAX macros
>> ---
> [...]>
>> diff --git a/audio/audio.h b/audio/audio.h
>> index ccfef9e10a..bcbe56d639 100644
>> --- a/audio/audio.h
>> +++ b/audio/audio.h
>> @@ -148,23 +148,6 @@ static inline void *advance (void *p, int incr)
>> return (d + incr);
>> }
>>
>> -#ifdef __GNUC__
>> -#define audio_MIN(a, b) ( __extension__ ({ \
>> - __typeof (a) ta = a; \
>> - __typeof (b) tb = b; \
>> - ((ta)>(tb)?(tb):(ta)); \
>> -}))
>> -
>> -#define audio_MAX(a, b) ( __extension__ ({ \
>> - __typeof (a) ta = a; \
>> - __typeof (b) tb = b; \
>> - ((ta)<(tb)?(tb):(ta)); \
>> -}))
>> -#else
>> -#define audio_MIN(a, b) ((a)>(b)?(b):(a))
>> -#define audio_MAX(a, b) ((a)<(b)?(b):(a))
>> -#endif
>> -
>
> Those MIN/MAX are smarter than the ones in "qemu/osdep.h", I'd keep them
> moving them there.
Yes, but only when using gcc (or clang when it emulates gcc).
Unfortunately, they work differently when one of the expressions has
side effects, which is a disaster waiting to happen (when some poor folk
finally tries to compile it with a non-gcc compiler).
Or do we support any compilers beside gcc and clang? Because if not,
sure, do it, just remove the #ifdef and keep only the gcc version.
Regards,
Zoltan
^ permalink raw reply [flat|nested] 91+ messages in thread* Re: [Qemu-devel] [PATCH v2 23/52] audio: remove audio_MIN, audio_MAX
2018-12-24 2:16 ` Zoltán Kővágó
@ 2018-12-24 17:16 ` Philippe Mathieu-Daudé
2018-12-24 20:48 ` Kővágó Zoltán
0 siblings, 1 reply; 91+ messages in thread
From: Philippe Mathieu-Daudé @ 2018-12-24 17:16 UTC (permalink / raw)
To: Zoltán Kővágó, qemu-devel, Paolo Bonzini
Cc: Michael Walle, Gerd Hoffmann
On 12/24/18 3:16 AM, Zoltán Kővágó wrote:
> Hi Phil,
>
> On 2018-12-24 00:49, Philippe Mathieu-Daudé wrote:
>> Hi Zoltán,
>>
>> On 12/23/18 9:51 PM, Kővágó, Zoltán wrote:
>>> There's already a MIN and MAX macro in include/qemu/osdep.h, use them
>>> instead.
>>>
>>> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
>>>
>>> ---
>>>
>>> Changes from v1:
>>> * removed audio_MIN, audio_MAX macros
>>> ---
>> [...]>
>>> diff --git a/audio/audio.h b/audio/audio.h
>>> index ccfef9e10a..bcbe56d639 100644
>>> --- a/audio/audio.h
>>> +++ b/audio/audio.h
>>> @@ -148,23 +148,6 @@ static inline void *advance (void *p, int incr)
>>> return (d + incr);
>>> }
>>>
>>> -#ifdef __GNUC__
>>> -#define audio_MIN(a, b) ( __extension__ ({ \
>>> - __typeof (a) ta = a; \
>>> - __typeof (b) tb = b; \
>>> - ((ta)>(tb)?(tb):(ta)); \
>>> -}))
>>> -
>>> -#define audio_MAX(a, b) ( __extension__ ({ \
>>> - __typeof (a) ta = a; \
>>> - __typeof (b) tb = b; \
>>> - ((ta)<(tb)?(tb):(ta)); \
>>> -}))
>>> -#else
>>> -#define audio_MIN(a, b) ((a)>(b)?(b):(a))
>>> -#define audio_MAX(a, b) ((a)<(b)?(b):(a))
>>> -#endif
>>> -
>>
>> Those MIN/MAX are smarter than the ones in "qemu/osdep.h", I'd keep them
>> moving them there.
>
> Yes, but only when using gcc (or clang when it emulates gcc).
> Unfortunately, they work differently when one of the expressions has
> side effects, which is a disaster waiting to happen (when some poor folk
> finally tries to compile it with a non-gcc compiler).
We already use the typeof extension:
$ git grep typeof|wc -l
145
For MIN/MAX I'd use rather use the __auto_type extension.
> Or do we support any compilers beside gcc and clang? Because if not,
> sure, do it, just remove the #ifdef and keep only the gcc version.
Yes, this is checked by the ./configure script:
1850 # Check whether the compiler matches our minimum requirements:
...
1859 # error You need at least Clang v3.4 to compile QEMU
...
1864 # error You need at least GCC v4.8 to compile QEMU
...
1867 # error You either need GCC or Clang to compiler QEMU
Regards,
Phil.
^ permalink raw reply [flat|nested] 91+ messages in thread* Re: [Qemu-devel] [PATCH v2 23/52] audio: remove audio_MIN, audio_MAX
2018-12-24 17:16 ` Philippe Mathieu-Daudé
@ 2018-12-24 20:48 ` Kővágó Zoltán
2018-12-25 10:40 ` Philippe Mathieu-Daudé
0 siblings, 1 reply; 91+ messages in thread
From: Kővágó Zoltán @ 2018-12-24 20:48 UTC (permalink / raw)
To: Philippe Mathieu-Daudé, qemu-devel, Paolo Bonzini
Cc: Michael Walle, Gerd Hoffmann
On 2018-12-24 18:16, Philippe Mathieu-Daudé wrote:
> On 12/24/18 3:16 AM, Zoltán Kővágó wrote:
>> Hi Phil,
>>
>> On 2018-12-24 00:49, Philippe Mathieu-Daudé wrote:
>>> Hi Zoltán,
>>>
>>> On 12/23/18 9:51 PM, Kővágó, Zoltán wrote:
>>>> There's already a MIN and MAX macro in include/qemu/osdep.h, use them
>>>> instead.
>>>>
>>>> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
>>>>
>>>> ---
>>>>
>>>> Changes from v1:
>>>> * removed audio_MIN, audio_MAX macros
>>>> ---
>>> [...]>
>>>> diff --git a/audio/audio.h b/audio/audio.h
>>>> index ccfef9e10a..bcbe56d639 100644
>>>> --- a/audio/audio.h
>>>> +++ b/audio/audio.h
>>>> @@ -148,23 +148,6 @@ static inline void *advance (void *p, int incr)
>>>> return (d + incr);
>>>> }
>>>>
>>>> -#ifdef __GNUC__
>>>> -#define audio_MIN(a, b) ( __extension__ ({ \
>>>> - __typeof (a) ta = a; \
>>>> - __typeof (b) tb = b; \
>>>> - ((ta)>(tb)?(tb):(ta)); \
>>>> -}))
>>>> -
>>>> -#define audio_MAX(a, b) ( __extension__ ({ \
>>>> - __typeof (a) ta = a; \
>>>> - __typeof (b) tb = b; \
>>>> - ((ta)<(tb)?(tb):(ta)); \
>>>> -}))
>>>> -#else
>>>> -#define audio_MIN(a, b) ((a)>(b)?(b):(a))
>>>> -#define audio_MAX(a, b) ((a)<(b)?(b):(a))
>>>> -#endif
>>>> -
>>>
>>> Those MIN/MAX are smarter than the ones in "qemu/osdep.h", I'd keep them
>>> moving them there.
>>
>> Yes, but only when using gcc (or clang when it emulates gcc).
>> Unfortunately, they work differently when one of the expressions has
>> side effects, which is a disaster waiting to happen (when some poor folk
>> finally tries to compile it with a non-gcc compiler).
>
> We already use the typeof extension:
>
> $ git grep typeof|wc -l
> 145
>
> For MIN/MAX I'd use rather use the __auto_type extension.
>
>> Or do we support any compilers beside gcc and clang? Because if not,
>> sure, do it, just remove the #ifdef and keep only the gcc version.
>
> Yes, this is checked by the ./configure script:
>
> 1850 # Check whether the compiler matches our minimum requirements:
> ...
> 1859 # error You need at least Clang v3.4 to compile QEMU
> ...
> 1864 # error You need at least GCC v4.8 to compile QEMU
> ...
> 1867 # error You either need GCC or Clang to compiler QEMU
Fair enough. I've tried to check the documentation for supported
compilers, but I haven't found any info. Well, I didn't look at the
configure script, I admit that.
This is what I came up with:
#ifndef MIN
#define MIN(a, b) ( __extension__ ({ \
__auto_type _a = (a); \
__auto_type _b = (b); \
_a > _b ? _b : _a; \
}))
#endif
#ifndef MAX
#define MAX(a, b) ( __extension__ ({ \
__auto_type _a = (a); \
__auto_type _b = (b); \
_a < _b ? _b : _a; \
}))
#endif
I changed it a bit, since a and b are the macro parameters, so they are
the variables that we should put into parentheses, and I renamed ta/tb
to _a/_b, this way they're less likely to clobber some other variables.
To be honest, we could probably drop the __extension__ keyword too,
we're not compiling with -pedantic. Other than audio_MIN and audio_MAX,
it only appears in the DO_UPCAST macro.
Regards,
Zoltan
^ permalink raw reply [flat|nested] 91+ messages in thread* Re: [Qemu-devel] [PATCH v2 23/52] audio: remove audio_MIN, audio_MAX
2018-12-24 20:48 ` Kővágó Zoltán
@ 2018-12-25 10:40 ` Philippe Mathieu-Daudé
2018-12-27 12:49 ` Kővágó Zoltán
0 siblings, 1 reply; 91+ messages in thread
From: Philippe Mathieu-Daudé @ 2018-12-25 10:40 UTC (permalink / raw)
To: Kővágó Zoltán, qemu-devel, Paolo Bonzini
Cc: Michael Walle, Gerd Hoffmann
On 12/24/18 9:48 PM, Kővágó Zoltán wrote:
> On 2018-12-24 18:16, Philippe Mathieu-Daudé wrote:
>> On 12/24/18 3:16 AM, Zoltán Kővágó wrote:
>>> Hi Phil,
>>>
>>> On 2018-12-24 00:49, Philippe Mathieu-Daudé wrote:
>>>> Hi Zoltán,
>>>>
>>>> On 12/23/18 9:51 PM, Kővágó, Zoltán wrote:
>>>>> There's already a MIN and MAX macro in include/qemu/osdep.h, use them
>>>>> instead.
>>>>>
>>>>> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
>>>>>
>>>>> ---
>>>>>
>>>>> Changes from v1:
>>>>> * removed audio_MIN, audio_MAX macros
>>>>> ---
>>>> [...]>
>>>>> diff --git a/audio/audio.h b/audio/audio.h
>>>>> index ccfef9e10a..bcbe56d639 100644
>>>>> --- a/audio/audio.h
>>>>> +++ b/audio/audio.h
>>>>> @@ -148,23 +148,6 @@ static inline void *advance (void *p, int incr)
>>>>> return (d + incr);
>>>>> }
>>>>>
>>>>> -#ifdef __GNUC__
>>>>> -#define audio_MIN(a, b) ( __extension__ ({ \
>>>>> - __typeof (a) ta = a; \
>>>>> - __typeof (b) tb = b; \
>>>>> - ((ta)>(tb)?(tb):(ta)); \
>>>>> -}))
>>>>> -
>>>>> -#define audio_MAX(a, b) ( __extension__ ({ \
>>>>> - __typeof (a) ta = a; \
>>>>> - __typeof (b) tb = b; \
>>>>> - ((ta)<(tb)?(tb):(ta)); \
>>>>> -}))
>>>>> -#else
>>>>> -#define audio_MIN(a, b) ((a)>(b)?(b):(a))
>>>>> -#define audio_MAX(a, b) ((a)<(b)?(b):(a))
>>>>> -#endif
>>>>> -
>>>>
>>>> Those MIN/MAX are smarter than the ones in "qemu/osdep.h", I'd keep them
>>>> moving them there.
>>>
>>> Yes, but only when using gcc (or clang when it emulates gcc).
>>> Unfortunately, they work differently when one of the expressions has
>>> side effects, which is a disaster waiting to happen (when some poor folk
>>> finally tries to compile it with a non-gcc compiler).
>>
>> We already use the typeof extension:
>>
>> $ git grep typeof|wc -l
>> 145
>>
>> For MIN/MAX I'd use rather use the __auto_type extension.
>>
>>> Or do we support any compilers beside gcc and clang? Because if not,
>>> sure, do it, just remove the #ifdef and keep only the gcc version.
>>
>> Yes, this is checked by the ./configure script:
>>
>> 1850 # Check whether the compiler matches our minimum requirements:
>> ...
>> 1859 # error You need at least Clang v3.4 to compile QEMU
>> ...
>> 1864 # error You need at least GCC v4.8 to compile QEMU
>> ...
>> 1867 # error You either need GCC or Clang to compiler QEMU
>
> Fair enough. I've tried to check the documentation for supported
> compilers, but I haven't found any info. Well, I didn't look at the
> configure script, I admit that.
You haven't found any info because the minimum requirements are not well
documented on the wiki.
>
> This is what I came up with:
>
> #ifndef MIN
> #define MIN(a, b) ( __extension__ ({ \
> __auto_type _a = (a); \
> __auto_type _b = (b); \
> _a > _b ? _b : _a; \
> }))
> #endif
> #ifndef MAX
> #define MAX(a, b) ( __extension__ ({ \
> __auto_type _a = (a); \
> __auto_type _b = (b); \
> _a < _b ? _b : _a; \
> }))
> #endif
>
>
> I changed it a bit, since a and b are the macro parameters, so they are
> the variables that we should put into parentheses, and I renamed ta/tb
> to _a/_b, this way they're less likely to clobber some other variables.
Sounds good. Now the parentheses around a/b are not necessary.
I wonder however if it wouldn't be cleaner to previously add:
#ifdef MIN
# undef MIN
#endif
and for MAX, rather than only use these macros if undefs.
> To be honest, we could probably drop the __extension__ keyword too,
> we're not compiling with -pedantic. Other than audio_MIN and audio_MAX,
> it only appears in the DO_UPCAST macro.
Fine with me!
If you have this patch ready, you can send it alone out of this series,
no need to resend this whole series for this patch.
Thanks,
Phil.
^ permalink raw reply [flat|nested] 91+ messages in thread* Re: [Qemu-devel] [PATCH v2 23/52] audio: remove audio_MIN, audio_MAX
2018-12-25 10:40 ` Philippe Mathieu-Daudé
@ 2018-12-27 12:49 ` Kővágó Zoltán
2019-01-07 9:54 ` Gerd Hoffmann
0 siblings, 1 reply; 91+ messages in thread
From: Kővágó Zoltán @ 2018-12-27 12:49 UTC (permalink / raw)
To: Philippe Mathieu-Daudé, qemu-devel, Paolo Bonzini
Cc: Michael Walle, Gerd Hoffmann
On 2018-12-25 11:40, Philippe Mathieu-Daudé wrote:
> On 12/24/18 9:48 PM, Kővágó Zoltán wrote:
>> On 2018-12-24 18:16, Philippe Mathieu-Daudé wrote:
>>> On 12/24/18 3:16 AM, Zoltán Kővágó wrote:
>>>> Hi Phil,
>>>>
>>>> On 2018-12-24 00:49, Philippe Mathieu-Daudé wrote:
>>>>> Hi Zoltán,
>>>>>
>>>>> On 12/23/18 9:51 PM, Kővágó, Zoltán wrote:
>>>>>> There's already a MIN and MAX macro in include/qemu/osdep.h, use them
>>>>>> instead.
>>>>>>
>>>>>> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
>>>>>>
>>>>>> ---
>>>>>>
>>>>>> Changes from v1:
>>>>>> * removed audio_MIN, audio_MAX macros
>>>>>> ---
>>>>> [...]>
>>>>>> diff --git a/audio/audio.h b/audio/audio.h
>>>>>> index ccfef9e10a..bcbe56d639 100644
>>>>>> --- a/audio/audio.h
>>>>>> +++ b/audio/audio.h
>>>>>> @@ -148,23 +148,6 @@ static inline void *advance (void *p, int incr)
>>>>>> return (d + incr);
>>>>>> }
>>>>>>
>>>>>> -#ifdef __GNUC__
>>>>>> -#define audio_MIN(a, b) ( __extension__ ({ \
>>>>>> - __typeof (a) ta = a; \
>>>>>> - __typeof (b) tb = b; \
>>>>>> - ((ta)>(tb)?(tb):(ta)); \
>>>>>> -}))
>>>>>> -
>>>>>> -#define audio_MAX(a, b) ( __extension__ ({ \
>>>>>> - __typeof (a) ta = a; \
>>>>>> - __typeof (b) tb = b; \
>>>>>> - ((ta)<(tb)?(tb):(ta)); \
>>>>>> -}))
>>>>>> -#else
>>>>>> -#define audio_MIN(a, b) ((a)>(b)?(b):(a))
>>>>>> -#define audio_MAX(a, b) ((a)<(b)?(b):(a))
>>>>>> -#endif
>>>>>> -
>>>>>
>>>>> Those MIN/MAX are smarter than the ones in "qemu/osdep.h", I'd keep them
>>>>> moving them there.
>>>>
>>>> Yes, but only when using gcc (or clang when it emulates gcc).
>>>> Unfortunately, they work differently when one of the expressions has
>>>> side effects, which is a disaster waiting to happen (when some poor folk
>>>> finally tries to compile it with a non-gcc compiler).
>>>
>>> We already use the typeof extension:
>>>
>>> $ git grep typeof|wc -l
>>> 145
>>>
>>> For MIN/MAX I'd use rather use the __auto_type extension.
>>>
>>>> Or do we support any compilers beside gcc and clang? Because if not,
>>>> sure, do it, just remove the #ifdef and keep only the gcc version.
>>>
>>> Yes, this is checked by the ./configure script:
>>>
>>> 1850 # Check whether the compiler matches our minimum requirements:
>>> ...
>>> 1859 # error You need at least Clang v3.4 to compile QEMU
>>> ...
>>> 1864 # error You need at least GCC v4.8 to compile QEMU
>>> ...
>>> 1867 # error You either need GCC or Clang to compiler QEMU
>>
>> Fair enough. I've tried to check the documentation for supported
>> compilers, but I haven't found any info. Well, I didn't look at the
>> configure script, I admit that.
>
> You haven't found any info because the minimum requirements are not well
> documented on the wiki.
>
>>
>> This is what I came up with:
>>
>> #ifndef MIN
>> #define MIN(a, b) ( __extension__ ({ \
>> __auto_type _a = (a); \
>> __auto_type _b = (b); \
>> _a > _b ? _b : _a; \
>> }))
>> #endif
>> #ifndef MAX
>> #define MAX(a, b) ( __extension__ ({ \
>> __auto_type _a = (a); \
>> __auto_type _b = (b); \
>> _a < _b ? _b : _a; \
>> }))
>> #endif
>>
>>
>> I changed it a bit, since a and b are the macro parameters, so they are
>> the variables that we should put into parentheses, and I renamed ta/tb
>> to _a/_b, this way they're less likely to clobber some other variables.
>
> Sounds good. Now the parentheses around a/b are not necessary.
>
> I wonder however if it wouldn't be cleaner to previously add:
>
> #ifdef MIN
> # undef MIN
> #endif
>
> and for MAX, rather than only use these macros if undefs.
Um, this broke a few things. It also means until now in some cases it
used some random definition of MIN/MAX instead of the one in the header.
/home/dirty_ice/qemu/include/qemu/osdep.h:240:5: error: ‘__auto_type’
used with a bit-field initializer
/home/dirty_ice/qemu/include/qemu/osdep.h:247:35: error: braced-group
within expression allowed only inside a function
The first one is fixable with an explicit cast (ugly but works), but the
second one is more problematic. It means we can't write stuff like
USBPort uports[MAX(MAXPORTS_2, MAXPORTS_3)];
when not in a function. So we either need a dumb version of MIN/MAX, or
scrape the idea altogether.
>
>> To be honest, we could probably drop the __extension__ keyword too,
>> we're not compiling with -pedantic. Other than audio_MIN and audio_MAX,
>> it only appears in the DO_UPCAST macro.
>
> Fine with me!
>
> If you have this patch ready, you can send it alone out of this series,
> no need to resend this whole series for this patch.
There will be adjustmets to other patches too, so I think I'll just
consolidate them.
Regards,
Zoltan
^ permalink raw reply [flat|nested] 91+ messages in thread* Re: [Qemu-devel] [PATCH v2 23/52] audio: remove audio_MIN, audio_MAX
2018-12-27 12:49 ` Kővágó Zoltán
@ 2019-01-07 9:54 ` Gerd Hoffmann
2019-01-07 14:26 ` Eric Blake
0 siblings, 1 reply; 91+ messages in thread
From: Gerd Hoffmann @ 2019-01-07 9:54 UTC (permalink / raw)
To: Kővágó Zoltán
Cc: Philippe Mathieu-Daudé, qemu-devel, Paolo Bonzini, Michael Walle
> The first one is fixable with an explicit cast (ugly but works), but the
> second one is more problematic. It means we can't write stuff like
>
> USBPort uports[MAX(MAXPORTS_2, MAXPORTS_3)];
>
> when not in a function. So we either need a dumb version of MIN/MAX, or
> scrape the idea altogether.
const_max() for stuff which is a compile time constant maybe?
cheers,
Gerd
^ permalink raw reply [flat|nested] 91+ messages in thread
* Re: [Qemu-devel] [PATCH v2 23/52] audio: remove audio_MIN, audio_MAX
2019-01-07 9:54 ` Gerd Hoffmann
@ 2019-01-07 14:26 ` Eric Blake
0 siblings, 0 replies; 91+ messages in thread
From: Eric Blake @ 2019-01-07 14:26 UTC (permalink / raw)
To: Gerd Hoffmann, Kővágó Zoltán
Cc: Paolo Bonzini, Michael Walle, Philippe Mathieu-Daudé, qemu-devel
[-- Attachment #1: Type: text/plain, Size: 761 bytes --]
On 1/7/19 3:54 AM, Gerd Hoffmann wrote:
>> The first one is fixable with an explicit cast (ugly but works), but the
>> second one is more problematic. It means we can't write stuff like
>>
>> USBPort uports[MAX(MAXPORTS_2, MAXPORTS_3)];
>>
>> when not in a function. So we either need a dumb version of MIN/MAX, or
>> scrape the idea altogether.
>
> const_max() for stuff which is a compile time constant maybe?
The conversation on MIN/MAX has been extracted to a new thread:
https://lists.gnu.org/archive/html/qemu-devel/2019-01/msg00727.html
https://lists.gnu.org/archive/html/qemu-devel/2019-01/msg00695.html
--
Eric Blake, Principal Software Engineer
Red Hat, Inc. +1-919-301-3226
Virtualization: qemu.org | libvirt.org
[-- Attachment #2: OpenPGP digital signature --]
[-- Type: application/pgp-signature, Size: 488 bytes --]
^ permalink raw reply [flat|nested] 91+ messages in thread
* [Qemu-devel] [PATCH v2 24/52] audio: do not run each backend in audio_run
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (22 preceding siblings ...)
2018-12-23 20:51 ` [Qemu-devel] [PATCH v2 23/52] audio: remove audio_MIN, audio_MAX Kővágó, Zoltán
@ 2018-12-23 20:52 ` Kővágó, Zoltán
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 25/52] paaudio: fix playback glitches Kővágó, Zoltán
` (26 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:52 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
audio_run is called manually by alsa and oss backends when polling.
In this case only the requesting backend should be run, not all of them.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/alsaaudio.c | 7 +++++--
audio/audio.c | 14 +++++---------
audio/audio_int.h | 2 +-
audio/ossaudio.c | 12 ++++++------
4 files changed, 17 insertions(+), 18 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index d4bb540cb9..ffc6220d7a 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -39,6 +39,7 @@ struct pollhlp {
struct pollfd *pfds;
int count;
int mask;
+ AudioState *s;
};
typedef struct ALSAVoiceOut {
@@ -199,11 +200,11 @@ static void alsa_poll_handler (void *opaque)
break;
case SND_PCM_STATE_PREPARED:
- audio_run ("alsa run (prepared)");
+ audio_run(hlp->s, "alsa run (prepared)");
break;
case SND_PCM_STATE_RUNNING:
- audio_run ("alsa run (running)");
+ audio_run (hlp->s, "alsa run (running)");
break;
default:
@@ -757,6 +758,7 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
return -1;
}
+ alsa->pollhlp.s = hw->s;
alsa->handle = handle;
alsa->dev = dev;
return 0;
@@ -858,6 +860,7 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
return -1;
}
+ alsa->pollhlp.s = hw->s;
alsa->handle = handle;
alsa->dev = dev;
return 0;
diff --git a/audio/audio.c b/audio/audio.c
index a32b86a813..4779061859 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -839,7 +839,7 @@ static void audio_timer (void *opaque)
}
s->timer_last = now;
- audio_run("timer");
+ audio_run(s, "timer");
audio_reset_timer(s);
}
@@ -1241,15 +1241,11 @@ static void audio_run_capture (AudioState *s)
}
}
-void audio_run (const char *msg)
+void audio_run(AudioState *s, const char *msg)
{
- AudioState *s;
-
- QTAILQ_FOREACH(s, &audio_states, list) {
- audio_run_out (s);
- audio_run_in (s);
- audio_run_capture (s);
- }
+ audio_run_out(s);
+ audio_run_in(s);
+ audio_run_capture(s);
#ifdef DEBUG_POLL
{
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 9d9db8c270..9950cea869 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -220,7 +220,7 @@ int audio_pcm_hw_clip_out (HWVoiceOut *hw, void *pcm_buf,
int audio_bug (const char *funcname, int cond);
void *audio_calloc (const char *funcname, int nmemb, size_t size);
-void audio_run (const char *msg);
+void audio_run(AudioState *s, const char *msg);
#define VOICE_ENABLE 1
#define VOICE_DISABLE 2
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index ae06e58d13..fde62c8ce2 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -110,28 +110,28 @@ static void oss_anal_close (int *fdp)
static void oss_helper_poll_out (void *opaque)
{
- (void) opaque;
- audio_run ("oss_poll_out");
+ AudioState *s = opaque;
+ audio_run(s, "oss_poll_out");
}
static void oss_helper_poll_in (void *opaque)
{
- (void) opaque;
- audio_run ("oss_poll_in");
+ AudioState *s = opaque;
+ audio_run(s, "oss_poll_in");
}
static void oss_poll_out (HWVoiceOut *hw)
{
OSSVoiceOut *oss = (OSSVoiceOut *) hw;
- qemu_set_fd_handler (oss->fd, NULL, oss_helper_poll_out, NULL);
+ qemu_set_fd_handler(oss->fd, NULL, oss_helper_poll_out, hw->s);
}
static void oss_poll_in (HWVoiceIn *hw)
{
OSSVoiceIn *oss = (OSSVoiceIn *) hw;
- qemu_set_fd_handler (oss->fd, oss_helper_poll_in, NULL, NULL);
+ qemu_set_fd_handler(oss->fd, oss_helper_poll_in, NULL, hw->s);
}
static int oss_write (SWVoiceOut *sw, void *buf, int len)
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 25/52] paaudio: fix playback glitches
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (23 preceding siblings ...)
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 24/52] audio: do not run each backend in audio_run Kővágó, Zoltán
@ 2018-12-23 20:52 ` Kővágó, Zoltán
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 26/52] audio: remove read and write pcm_ops Kővágó, Zoltán
` (25 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:52 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Pulseaudio normally assumes that when the server wants it, the client
can generate the audio samples and send it right away. Unfortunately
this is not the case with QEMU -- it's up to the emulated system when
does it generate the samples. Buffering the samples and sending them
from a background thread is just a workaround, that doesn't work too
well. Instead enable pa's compatibility support and let pa worry about
the details.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/paaudio.c | 6 ++----
1 file changed, 2 insertions(+), 4 deletions(-)
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 6ed12851c0..9331e630d1 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -511,10 +511,8 @@ static pa_stream *qpa_simple_new (
flags =
PA_STREAM_INTERPOLATE_TIMING
-#ifdef PA_STREAM_ADJUST_LATENCY
- |PA_STREAM_ADJUST_LATENCY
-#endif
- |PA_STREAM_AUTO_TIMING_UPDATE;
+ |PA_STREAM_AUTO_TIMING_UPDATE
+ |PA_STREAM_EARLY_REQUESTS;
if (dev) {
/* don't move the stream if the user specified a sink/source */
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 26/52] audio: remove read and write pcm_ops
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (24 preceding siblings ...)
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 25/52] paaudio: fix playback glitches Kővágó, Zoltán
@ 2018-12-23 20:52 ` Kővágó, Zoltán
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 27/52] audio: use size_t where makes sense Kővágó, Zoltán
` (24 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:52 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
They just called audio_pcm_sw_read/write anyway, so it makes no sense
to have them too. (The noaudio's read is the only exception, but it
should work with the generic code too.)
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/alsaaudio.c | 12 ------------
audio/audio.c | 8 ++++----
audio/audio_int.h | 5 -----
audio/coreaudio.c | 6 ------
audio/dsoundaudio.c | 12 ------------
audio/noaudio.c | 19 -------------------
audio/ossaudio.c | 12 ------------
audio/paaudio.c | 12 ------------
audio/sdlaudio.c | 6 ------
audio/spiceaudio.c | 12 ------------
audio/wavaudio.c | 6 ------
11 files changed, 4 insertions(+), 106 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index ffc6220d7a..19de7d01cb 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -270,11 +270,6 @@ static int alsa_poll_in (HWVoiceIn *hw)
return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLIN);
}
-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 (AudioFormat fmt, int endianness)
{
switch (fmt) {
@@ -996,11 +991,6 @@ static int alsa_run_in (HWVoiceIn *hw)
return read_samples;
}
-static int alsa_read (SWVoiceIn *sw, void *buf, int size)
-{
- return audio_pcm_sw_read (sw, buf, size);
-}
-
static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
@@ -1070,13 +1060,11 @@ static struct audio_pcm_ops alsa_pcm_ops = {
.init_out = alsa_init_out,
.fini_out = alsa_fini_out,
.run_out = alsa_run_out,
- .write = alsa_write,
.ctl_out = alsa_ctl_out,
.init_in = alsa_init_in,
.fini_in = alsa_fini_in,
.run_in = alsa_run_in,
- .read = alsa_read,
.ctl_in = alsa_ctl_in,
};
diff --git a/audio/audio.c b/audio/audio.c
index 4779061859..1ea80ba6a7 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -598,7 +598,7 @@ static int audio_pcm_sw_get_rpos_in (SWVoiceIn *sw)
}
}
-int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int size)
+static int audio_pcm_sw_read(SWVoiceIn *sw, void *buf, int size)
{
HWVoiceIn *hw = sw->hw;
int samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
@@ -700,7 +700,7 @@ static int audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
/*
* Soft voice (playback)
*/
-int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
+static int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
{
int hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
int ret = 0, pos = 0, total = 0;
@@ -858,7 +858,7 @@ int AUD_write (SWVoiceOut *sw, void *buf, int size)
return 0;
}
- return sw->hw->pcm_ops->write(sw, buf, size);
+ return audio_pcm_sw_write(sw, buf, size);
}
int AUD_read (SWVoiceIn *sw, void *buf, int size)
@@ -873,7 +873,7 @@ int AUD_read (SWVoiceIn *sw, void *buf, int size)
return 0;
}
- return sw->hw->pcm_ops->read(sw, buf, size);
+ return audio_pcm_sw_read(sw, buf, size);
}
int AUD_get_buffer_size_out (SWVoiceOut *sw)
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 9950cea869..d269c38465 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -149,13 +149,11 @@ struct audio_pcm_ops {
int (*init_out)(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque);
void (*fini_out)(HWVoiceOut *hw);
int (*run_out) (HWVoiceOut *hw, int live);
- int (*write) (SWVoiceOut *sw, void *buf, int size);
int (*ctl_out) (HWVoiceOut *hw, int cmd, ...);
int (*init_in) (HWVoiceIn *hw, struct audsettings *as, void *drv_opaque);
void (*fini_in) (HWVoiceIn *hw);
int (*run_in) (HWVoiceIn *hw);
- int (*read) (SWVoiceIn *sw, void *buf, int size);
int (*ctl_in) (HWVoiceIn *hw, int cmd, ...);
};
@@ -209,11 +207,8 @@ audio_driver *audio_driver_lookup(const char *name);
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);
-int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int len);
int audio_pcm_hw_get_live_in (HWVoiceIn *hw);
-int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int len);
-
int audio_pcm_hw_clip_out (HWVoiceOut *hw, void *pcm_buf,
int live, int pending);
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index b6935359ee..abbbb45d69 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -489,11 +489,6 @@ static OSStatus audioDeviceIOProc(
return 0;
}
-static int coreaudio_write (SWVoiceOut *sw, void *buf, int len)
-{
- return audio_pcm_sw_write (sw, buf, len);
-}
-
static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
void *drv_opaque)
{
@@ -692,7 +687,6 @@ static struct audio_pcm_ops coreaudio_pcm_ops = {
.init_out = coreaudio_init_out,
.fini_out = coreaudio_fini_out,
.run_out = coreaudio_run_out,
- .write = coreaudio_write,
.ctl_out = coreaudio_ctl_out
};
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index 6f074c6f94..dfcb70c81a 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -454,11 +454,6 @@ static int dsound_ctl_out (HWVoiceOut *hw, int cmd, ...)
return 0;
}
-static int dsound_write (SWVoiceOut *sw, void *buf, int len)
-{
- return audio_pcm_sw_write (sw, buf, len);
-}
-
static int dsound_run_out (HWVoiceOut *hw, int live)
{
int err;
@@ -645,11 +640,6 @@ static int dsound_ctl_in (HWVoiceIn *hw, int cmd, ...)
return 0;
}
-static int dsound_read (SWVoiceIn *sw, void *buf, int len)
-{
- return audio_pcm_sw_read (sw, buf, len);
-}
-
static int dsound_run_in (HWVoiceIn *hw)
{
int err;
@@ -856,13 +846,11 @@ static struct audio_pcm_ops dsound_pcm_ops = {
.init_out = dsound_init_out,
.fini_out = dsound_fini_out,
.run_out = dsound_run_out,
- .write = dsound_write,
.ctl_out = dsound_ctl_out,
.init_in = dsound_init_in,
.fini_in = dsound_fini_in,
.run_in = dsound_run_in,
- .read = dsound_read,
.ctl_in = dsound_ctl_in
};
diff --git a/audio/noaudio.c b/audio/noaudio.c
index 299ba3d09c..c49f2df4b3 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -60,11 +60,6 @@ static int no_run_out (HWVoiceOut *hw, int live)
return decr;
}
-static int no_write (SWVoiceOut *sw, void *buf, int len)
-{
- return audio_pcm_sw_write(sw, buf, len);
-}
-
static int no_init_out(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque)
{
audio_pcm_init_info (&hw->info, as);
@@ -117,18 +112,6 @@ static int no_run_in (HWVoiceIn *hw)
return samples;
}
-static int no_read (SWVoiceIn *sw, void *buf, int size)
-{
- /* use custom code here instead of audio_pcm_sw_read() to avoid
- * useless resampling/mixing */
- int samples = size >> sw->info.shift;
- int total = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
- int to_clear = MIN (samples, total);
- sw->total_hw_samples_acquired += total;
- audio_pcm_info_clear_buf (&sw->info, buf, to_clear);
- return to_clear << sw->info.shift;
-}
-
static int no_ctl_in (HWVoiceIn *hw, int cmd, ...)
{
(void) hw;
@@ -150,13 +133,11 @@ static struct audio_pcm_ops no_pcm_ops = {
.init_out = no_init_out,
.fini_out = no_fini_out,
.run_out = no_run_out,
- .write = no_write,
.ctl_out = no_ctl_out,
.init_in = no_init_in,
.fini_in = no_fini_in,
.run_in = no_run_in,
- .read = no_read,
.ctl_in = no_ctl_in
};
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index fde62c8ce2..70909e5696 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -134,11 +134,6 @@ static void oss_poll_in (HWVoiceIn *hw)
qemu_set_fd_handler(oss->fd, oss_helper_poll_in, NULL, hw->s);
}
-static int oss_write (SWVoiceOut *sw, void *buf, int len)
-{
- return audio_pcm_sw_write (sw, buf, len);
-}
-
static int aud_to_ossfmt (AudioFormat fmt, int endianness)
{
switch (fmt) {
@@ -788,11 +783,6 @@ static int oss_run_in (HWVoiceIn *hw)
return read_samples;
}
-static int oss_read (SWVoiceIn *sw, void *buf, int size)
-{
- return audio_pcm_sw_read (sw, buf, size);
-}
-
static int oss_ctl_in (HWVoiceIn *hw, int cmd, ...)
{
OSSVoiceIn *oss = (OSSVoiceIn *) hw;
@@ -844,13 +834,11 @@ static struct audio_pcm_ops oss_pcm_ops = {
.init_out = oss_init_out,
.fini_out = oss_fini_out,
.run_out = oss_run_out,
- .write = oss_write,
.ctl_out = oss_ctl_out,
.init_in = oss_init_in,
.fini_in = oss_fini_in,
.run_in = oss_run_in,
- .read = oss_read,
.ctl_in = oss_ctl_in
};
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 9331e630d1..6a1919e93b 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -294,11 +294,6 @@ static int qpa_run_out (HWVoiceOut *hw, int live)
return decr;
}
-static int qpa_write (SWVoiceOut *sw, void *buf, int len)
-{
- return audio_pcm_sw_write (sw, buf, len);
-}
-
/* capture */
static void *qpa_thread_in (void *arg)
{
@@ -387,11 +382,6 @@ static int qpa_run_in (HWVoiceIn *hw)
return incr;
}
-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 (AudioFormat afmt, int endianness)
{
int format;
@@ -975,13 +965,11 @@ static struct audio_pcm_ops qpa_pcm_ops = {
.init_out = qpa_init_out,
.fini_out = qpa_fini_out,
.run_out = qpa_run_out,
- .write = qpa_write,
.ctl_out = qpa_ctl_out,
.init_in = qpa_init_in,
.fini_in = qpa_fini_in,
.run_in = qpa_run_in,
- .read = qpa_read,
.ctl_in = qpa_ctl_in
};
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index fc61038b77..ab9166d054 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -325,11 +325,6 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
#endif
}
-static int sdl_write_out (SWVoiceOut *sw, void *buf, int len)
-{
- return audio_pcm_sw_write (sw, buf, len);
-}
-
static int sdl_run_out (HWVoiceOut *hw, int live)
{
int decr;
@@ -483,7 +478,6 @@ static struct audio_pcm_ops sdl_pcm_ops = {
.init_out = sdl_init_out,
.fini_out = sdl_fini_out,
.run_out = sdl_run_out,
- .write = sdl_write_out,
.ctl_out = sdl_ctl_out,
};
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index b9160991d2..f963853ed8 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -192,11 +192,6 @@ static int line_out_run (HWVoiceOut *hw, int live)
return decr;
}
-static int line_out_write (SWVoiceOut *sw, void *buf, int len)
-{
- return audio_pcm_sw_write (sw, buf, len);
-}
-
static int line_out_ctl (HWVoiceOut *hw, int cmd, ...)
{
SpiceVoiceOut *out = container_of (hw, SpiceVoiceOut, hw);
@@ -325,11 +320,6 @@ static int line_in_run (HWVoiceIn *hw)
return num_samples;
}
-static int line_in_read (SWVoiceIn *sw, void *buf, int size)
-{
- return audio_pcm_sw_read (sw, buf, size);
-}
-
static int line_in_ctl (HWVoiceIn *hw, int cmd, ...)
{
SpiceVoiceIn *in = container_of (hw, SpiceVoiceIn, hw);
@@ -377,13 +367,11 @@ static struct audio_pcm_ops audio_callbacks = {
.init_out = line_out_init,
.fini_out = line_out_fini,
.run_out = line_out_run,
- .write = line_out_write,
.ctl_out = line_out_ctl,
.init_in = line_in_init,
.fini_in = line_in_fini,
.run_in = line_in_run,
- .read = line_in_read,
.ctl_in = line_in_ctl,
};
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 723028a466..dda6993fb9 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -82,11 +82,6 @@ static int wav_run_out (HWVoiceOut *hw, int live)
return decr;
}
-static int wav_write_out (SWVoiceOut *sw, void *buf, int len)
-{
- return audio_pcm_sw_write (sw, buf, len);
-}
-
/* VICE code: Store number as little endian. */
static void le_store (uint8_t *buf, uint32_t val, int len)
{
@@ -238,7 +233,6 @@ static struct audio_pcm_ops wav_pcm_ops = {
.init_out = wav_init_out,
.fini_out = wav_fini_out,
.run_out = wav_run_out,
- .write = wav_write_out,
.ctl_out = wav_ctl_out,
};
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 27/52] audio: use size_t where makes sense
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (25 preceding siblings ...)
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 26/52] audio: remove read and write pcm_ops Kővágó, Zoltán
@ 2018-12-23 20:52 ` Kővágó, Zoltán
2018-12-24 6:19 ` Pavel Dovgalyuk
2018-12-25 11:08 ` Philippe Mathieu-Daudé
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 28/52] audio: api for mixeng code free backends Kővágó, Zoltán
` (23 subsequent siblings)
50 siblings, 2 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:52 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann, Pavel Dovgalyuk, Paolo Bonzini
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/alsaaudio.c | 8 +-
audio/audio.c | 162 ++++++++++++++++++++--------------------
audio/audio.h | 4 +-
audio/audio_int.h | 22 +++---
audio/audio_template.h | 6 +-
audio/mixeng.h | 11 ++-
audio/ossaudio.c | 18 ++---
audio/paaudio.c | 8 +-
audio/rate_template.h | 2 +-
audio/sdlaudio.c | 3 +-
audio/wavaudio.c | 4 +-
include/sysemu/replay.h | 4 +-
replay/replay-audio.c | 16 ++--
13 files changed, 133 insertions(+), 135 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 19de7d01cb..69e7a3868c 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -747,8 +747,8 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
alsa->pcm_buf = audio_calloc(__func__, obt.samples, 1 << hw->info.shift);
if (!alsa->pcm_buf) {
- dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
- hw->samples, 1 << hw->info.shift);
+ dolog("Could not allocate DAC buffer (%zu samples, each %d bytes)\n",
+ hw->samples, 1 << hw->info.shift);
alsa_anal_close1 (&handle);
return -1;
}
@@ -849,8 +849,8 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
alsa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
if (!alsa->pcm_buf) {
- dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
- hw->samples, 1 << hw->info.shift);
+ dolog("Could not allocate ADC buffer (%zu samples, each %d bytes)\n",
+ hw->samples, 1 << hw->info.shift);
alsa_anal_close1 (&handle);
return -1;
}
diff --git a/audio/audio.c b/audio/audio.c
index 1ea80ba6a7..27a8a31a64 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -530,10 +530,10 @@ static int audio_attach_capture (HWVoiceOut *hw)
/*
* Hard voice (capture)
*/
-static int audio_pcm_hw_find_min_in (HWVoiceIn *hw)
+static size_t audio_pcm_hw_find_min_in (HWVoiceIn *hw)
{
SWVoiceIn *sw;
- int m = hw->total_samples_captured;
+ size_t m = hw->total_samples_captured;
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
if (sw->active) {
@@ -543,28 +543,28 @@ static int audio_pcm_hw_find_min_in (HWVoiceIn *hw)
return m;
}
-int audio_pcm_hw_get_live_in (HWVoiceIn *hw)
+size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw)
{
- int live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
- if (audio_bug(__func__, live < 0 || live > hw->samples)) {
- dolog ("live=%d hw->samples=%d\n", live, hw->samples);
+ size_t live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
+ if (audio_bug(__func__, live > hw->samples)) {
+ dolog("live=%zu hw->samples=%zu\n", live, hw->samples);
return 0;
}
return live;
}
-int audio_pcm_hw_clip_out (HWVoiceOut *hw, void *pcm_buf,
- int live, int pending)
+size_t audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf,
+ size_t live, size_t pending)
{
- int left = hw->samples - pending;
- int len = MIN (left, live);
- int clipped = 0;
+ size_t left = hw->samples - pending;
+ size_t len = MIN (left, live);
+ size_t clipped = 0;
while (len) {
struct st_sample *src = hw->mix_buf + hw->rpos;
uint8_t *dst = advance (pcm_buf, hw->rpos << hw->info.shift);
- int samples_till_end_of_buf = hw->samples - hw->rpos;
- int samples_to_clip = MIN (len, samples_till_end_of_buf);
+ size_t samples_till_end_of_buf = hw->samples - hw->rpos;
+ size_t samples_to_clip = MIN (len, samples_till_end_of_buf);
hw->clip (dst, src, samples_to_clip);
@@ -578,14 +578,14 @@ int audio_pcm_hw_clip_out (HWVoiceOut *hw, void *pcm_buf,
/*
* Soft voice (capture)
*/
-static int audio_pcm_sw_get_rpos_in (SWVoiceIn *sw)
+static size_t audio_pcm_sw_get_rpos_in(SWVoiceIn *sw)
{
HWVoiceIn *hw = sw->hw;
- int live = hw->total_samples_captured - sw->total_hw_samples_acquired;
- int rpos;
+ ssize_t live = hw->total_samples_captured - sw->total_hw_samples_acquired;
+ ssize_t rpos;
if (audio_bug(__func__, live < 0 || live > hw->samples)) {
- dolog ("live=%d hw->samples=%d\n", live, hw->samples);
+ dolog("live=%zu hw->samples=%zu\n", live, hw->samples);
return 0;
}
@@ -598,17 +598,17 @@ static int audio_pcm_sw_get_rpos_in (SWVoiceIn *sw)
}
}
-static int audio_pcm_sw_read(SWVoiceIn *sw, void *buf, int size)
+static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
{
HWVoiceIn *hw = sw->hw;
- int samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
+ size_t samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
struct st_sample *src, *dst = sw->buf;
rpos = audio_pcm_sw_get_rpos_in (sw) % hw->samples;
live = hw->total_samples_captured - sw->total_hw_samples_acquired;
- if (audio_bug(__func__, live < 0 || live > hw->samples)) {
- dolog ("live_in=%d hw->samples=%d\n", live, hw->samples);
+ if (audio_bug(__func__, live > hw->samples)) {
+ dolog("live_in=%zu hw->samples=%zu\n", live, hw->samples);
return 0;
}
@@ -622,9 +622,9 @@ static int audio_pcm_sw_read(SWVoiceIn *sw, void *buf, int size)
while (swlim) {
src = hw->conv_buf + rpos;
- isamp = hw->wpos - rpos;
- /* XXX: <= ? */
- if (isamp <= 0) {
+ if (hw->wpos > rpos) {
+ isamp = hw->wpos - rpos;
+ } else {
isamp = hw->samples - rpos;
}
@@ -633,11 +633,6 @@ static int audio_pcm_sw_read(SWVoiceIn *sw, void *buf, int size)
}
osamp = swlim;
- if (audio_bug(__func__, osamp < 0)) {
- dolog ("osamp=%d\n", osamp);
- return 0;
- }
-
st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
swlim -= osamp;
rpos = (rpos + isamp) % hw->samples;
@@ -658,10 +653,10 @@ static int audio_pcm_sw_read(SWVoiceIn *sw, void *buf, int size)
/*
* Hard voice (playback)
*/
-static int audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
+static size_t audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
{
SWVoiceOut *sw;
- int m = INT_MAX;
+ size_t m = SIZE_MAX;
int nb_live = 0;
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
@@ -675,9 +670,9 @@ static int audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
return m;
}
-static int audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
+static size_t audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
{
- int smin;
+ size_t smin;
int nb_live1;
smin = audio_pcm_hw_find_min_out (hw, &nb_live1);
@@ -686,10 +681,10 @@ static int audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
}
if (nb_live1) {
- int live = smin;
+ size_t live = smin;
- if (audio_bug(__func__, live < 0 || live > hw->samples)) {
- dolog ("live=%d hw->samples=%d\n", live, hw->samples);
+ if (audio_bug(__func__, live > hw->samples)) {
+ dolog("live=%zu hw->samples=%zu\n", live, hw->samples);
return 0;
}
return live;
@@ -700,10 +695,10 @@ static int audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
/*
* Soft voice (playback)
*/
-static int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
+static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
{
- int hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
- int ret = 0, pos = 0, total = 0;
+ size_t hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
+ size_t ret = 0, pos = 0, total = 0;
if (!sw) {
return size;
@@ -712,8 +707,8 @@ static int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
hwsamples = sw->hw->samples;
live = sw->total_hw_samples_mixed;
- if (audio_bug(__func__, live < 0 || live > hwsamples)) {
- dolog ("live=%d hw->samples=%d\n", live, hwsamples);
+ if (audio_bug(__func__, live > hwsamples)) {
+ dolog("live=%zu hw->samples=%zu\n", live, hwsamples);
return 0;
}
@@ -767,7 +762,7 @@ static int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
#ifdef DEBUG_OUT
dolog (
- "%s: write size %d ret %d total sw %d\n",
+ "%s: write size %zu ret %zu total sw %zu\n",
SW_NAME (sw),
size >> sw->info.shift,
ret,
@@ -846,7 +841,7 @@ static void audio_timer (void *opaque)
/*
* Public API
*/
-int AUD_write (SWVoiceOut *sw, void *buf, int size)
+size_t AUD_write(SWVoiceOut *sw, void *buf, size_t size)
{
if (!sw) {
/* XXX: Consider options */
@@ -861,7 +856,7 @@ int AUD_write (SWVoiceOut *sw, void *buf, int size)
return audio_pcm_sw_write(sw, buf, size);
}
-int AUD_read (SWVoiceIn *sw, void *buf, int size)
+size_t AUD_read(SWVoiceIn *sw, void *buf, size_t size)
{
if (!sw) {
/* XXX: Consider options */
@@ -970,17 +965,17 @@ void AUD_set_active_in (SWVoiceIn *sw, int on)
}
}
-static int audio_get_avail (SWVoiceIn *sw)
+static size_t audio_get_avail (SWVoiceIn *sw)
{
- int live;
+ size_t live;
if (!sw) {
return 0;
}
live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
- if (audio_bug(__func__, live < 0 || live > sw->hw->samples)) {
- dolog ("live=%d sw->hw->samples=%d\n", live, sw->hw->samples);
+ if (audio_bug(__func__, live > sw->hw->samples)) {
+ dolog("live=%zu sw->hw->samples=%zu\n", live, sw->hw->samples);
return 0;
}
@@ -993,9 +988,9 @@ static int audio_get_avail (SWVoiceIn *sw)
return (((int64_t) live << 32) / sw->ratio) << sw->info.shift;
}
-static int audio_get_free (SWVoiceOut *sw)
+static size_t audio_get_free(SWVoiceOut *sw)
{
- int live, dead;
+ size_t live, dead;
if (!sw) {
return 0;
@@ -1003,8 +998,8 @@ static int audio_get_free (SWVoiceOut *sw)
live = sw->total_hw_samples_mixed;
- if (audio_bug(__func__, live < 0 || live > sw->hw->samples)) {
- dolog ("live=%d sw->hw->samples=%d\n", live, sw->hw->samples);
+ if (audio_bug(__func__, live > sw->hw->samples)) {
+ dolog("live=%zu sw->hw->samples=%zu\n", live, sw->hw->samples);
return 0;
}
@@ -1019,9 +1014,10 @@ static int audio_get_free (SWVoiceOut *sw)
return (((int64_t) dead << 32) / sw->ratio) << sw->info.shift;
}
-static void audio_capture_mix_and_clear (HWVoiceOut *hw, int rpos, int samples)
+static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
+ size_t samples)
{
- int n;
+ size_t n;
if (hw->enabled) {
SWVoiceCap *sc;
@@ -1032,17 +1028,17 @@ static void audio_capture_mix_and_clear (HWVoiceOut *hw, int rpos, int samples)
n = samples;
while (n) {
- int till_end_of_hw = hw->samples - rpos2;
- int to_write = MIN (till_end_of_hw, n);
- int bytes = to_write << hw->info.shift;
- int written;
+ size_t till_end_of_hw = hw->samples - rpos2;
+ size_t to_write = MIN(till_end_of_hw, n);
+ size_t bytes = to_write << hw->info.shift;
+ size_t written;
sw->buf = hw->mix_buf + rpos2;
written = audio_pcm_sw_write (sw, NULL, bytes);
if (written - bytes) {
- dolog ("Could not mix %d bytes into a capture "
- "buffer, mixed %d\n",
- bytes, written);
+ dolog("Could not mix %zu bytes into a capture "
+ "buffer, mixed %zu\n",
+ bytes, written);
break;
}
n -= to_write;
@@ -1051,9 +1047,9 @@ static void audio_capture_mix_and_clear (HWVoiceOut *hw, int rpos, int samples)
}
}
- n = MIN (samples, hw->samples - rpos);
- mixeng_clear (hw->mix_buf + rpos, n);
- mixeng_clear (hw->mix_buf, samples - n);
+ n = MIN(samples, hw->samples - rpos);
+ mixeng_clear(hw->mix_buf + rpos, n);
+ mixeng_clear(hw->mix_buf, samples - n);
}
static void audio_run_out (AudioState *s)
@@ -1062,16 +1058,16 @@ static void audio_run_out (AudioState *s)
SWVoiceOut *sw;
while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
- int played;
- int live, free, nb_live, cleanup_required, prev_rpos;
+ size_t played, live, prev_rpos, free;
+ int nb_live, cleanup_required;
live = audio_pcm_hw_get_live_out (hw, &nb_live);
if (!nb_live) {
live = 0;
}
- if (audio_bug(__func__, live < 0 || live > hw->samples)) {
- dolog ("live=%d hw->samples=%d\n", live, hw->samples);
+ if (audio_bug(__func__, live > hw->samples)) {
+ dolog ("live=%zu hw->samples=%zu\n", live, hw->samples);
continue;
}
@@ -1106,13 +1102,13 @@ static void audio_run_out (AudioState *s)
played = hw->pcm_ops->run_out (hw, live);
replay_audio_out(&played);
if (audio_bug(__func__, hw->rpos >= hw->samples)) {
- dolog ("hw->rpos=%d hw->samples=%d played=%d\n",
- hw->rpos, hw->samples, played);
+ dolog("hw->rpos=%zu hw->samples=%zu played=%zu\n",
+ hw->rpos, hw->samples, played);
hw->rpos = 0;
}
#ifdef DEBUG_OUT
- dolog ("played=%d\n", played);
+ dolog("played=%zu\n", played);
#endif
if (played) {
@@ -1127,8 +1123,8 @@ static void audio_run_out (AudioState *s)
}
if (audio_bug(__func__, played > sw->total_hw_samples_mixed)) {
- dolog ("played=%d sw->total_hw_samples_mixed=%d\n",
- played, sw->total_hw_samples_mixed);
+ dolog("played=%zu sw->total_hw_samples_mixed=%zu\n",
+ played, sw->total_hw_samples_mixed);
played = sw->total_hw_samples_mixed;
}
@@ -1168,7 +1164,7 @@ static void audio_run_in (AudioState *s)
while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
SWVoiceIn *sw;
- int captured = 0, min;
+ size_t captured = 0, min;
if (replay_mode != REPLAY_MODE_PLAY) {
captured = hw->pcm_ops->run_in(hw);
@@ -1183,7 +1179,7 @@ static void audio_run_in (AudioState *s)
sw->total_hw_samples_acquired -= min;
if (sw->active) {
- int avail;
+ size_t avail;
avail = audio_get_avail (sw);
if (avail > 0) {
@@ -1199,15 +1195,15 @@ static void audio_run_capture (AudioState *s)
CaptureVoiceOut *cap;
for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
- int live, rpos, captured;
+ size_t live, rpos, captured;
HWVoiceOut *hw = &cap->hw;
SWVoiceOut *sw;
captured = live = audio_pcm_hw_get_live_out (hw, NULL);
rpos = hw->rpos;
while (live) {
- int left = hw->samples - rpos;
- int to_capture = MIN (live, left);
+ size_t left = hw->samples - rpos;
+ size_t to_capture = MIN(live, left);
struct st_sample *src;
struct capture_callback *cb;
@@ -1230,8 +1226,8 @@ static void audio_run_capture (AudioState *s)
}
if (audio_bug(__func__, captured > sw->total_hw_samples_mixed)) {
- dolog ("captured=%d sw->total_hw_samples_mixed=%d\n",
- captured, sw->total_hw_samples_mixed);
+ dolog("captured=%zu sw->total_hw_samples_mixed=%zu\n",
+ captured, sw->total_hw_samples_mixed);
captured = sw->total_hw_samples_mixed;
}
@@ -1570,8 +1566,8 @@ CaptureVoiceOut *AUD_add_capture(
hw->mix_buf = audio_calloc(__func__, hw->samples,
sizeof(struct st_sample));
if (!hw->mix_buf) {
- dolog ("Could not allocate capture mix buffer (%d samples)\n",
- hw->samples);
+ dolog("Could not allocate capture mix buffer (%zu samples)\n",
+ hw->samples);
goto err2;
}
@@ -1580,7 +1576,7 @@ CaptureVoiceOut *AUD_add_capture(
cap->buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
if (!cap->buf) {
dolog ("Could not allocate capture buffer "
- "(%d samples, each %d bytes)\n",
+ "(%zu samples, each %d bytes)\n",
hw->samples, 1 << hw->info.shift);
goto err3;
}
diff --git a/audio/audio.h b/audio/audio.h
index bcbe56d639..bfb12e7531 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -114,7 +114,7 @@ SWVoiceOut *AUD_open_out (
);
void AUD_close_out (QEMUSoundCard *card, SWVoiceOut *sw);
-int AUD_write (SWVoiceOut *sw, void *pcm_buf, int size);
+size_t AUD_write (SWVoiceOut *sw, void *pcm_buf, size_t size);
int AUD_get_buffer_size_out (SWVoiceOut *sw);
void AUD_set_active_out (SWVoiceOut *sw, int on);
int AUD_is_active_out (SWVoiceOut *sw);
@@ -135,7 +135,7 @@ SWVoiceIn *AUD_open_in (
);
void AUD_close_in (QEMUSoundCard *card, SWVoiceIn *sw);
-int AUD_read (SWVoiceIn *sw, void *pcm_buf, int size);
+size_t AUD_read (SWVoiceIn *sw, void *pcm_buf, size_t size);
void AUD_set_active_in (SWVoiceIn *sw, int on);
int AUD_is_active_in (SWVoiceIn *sw);
diff --git a/audio/audio_int.h b/audio/audio_int.h
index d269c38465..330c465d0b 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -60,12 +60,12 @@ typedef struct HWVoiceOut {
f_sample *clip;
- int rpos;
+ size_t rpos;
uint64_t ts_helper;
struct st_sample *mix_buf;
- int samples;
+ size_t samples;
QLIST_HEAD (sw_out_listhead, SWVoiceOut) sw_head;
QLIST_HEAD (sw_cap_listhead, SWVoiceCap) cap_head;
int ctl_caps;
@@ -81,13 +81,13 @@ typedef struct HWVoiceIn {
t_sample *conv;
- int wpos;
- int total_samples_captured;
+ size_t wpos;
+ size_t total_samples_captured;
uint64_t ts_helper;
struct st_sample *conv_buf;
- int samples;
+ size_t samples;
QLIST_HEAD (sw_in_listhead, SWVoiceIn) sw_head;
int ctl_caps;
struct audio_pcm_ops *pcm_ops;
@@ -102,7 +102,7 @@ struct SWVoiceOut {
int64_t ratio;
struct st_sample *buf;
void *rate;
- int total_hw_samples_mixed;
+ size_t total_hw_samples_mixed;
int active;
int empty;
HWVoiceOut *hw;
@@ -119,7 +119,7 @@ struct SWVoiceIn {
struct audio_pcm_info info;
int64_t ratio;
void *rate;
- int total_hw_samples_acquired;
+ size_t total_hw_samples_acquired;
struct st_sample *buf;
f_sample *clip;
HWVoiceIn *hw;
@@ -207,10 +207,10 @@ audio_driver *audio_driver_lookup(const char *name);
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);
-int audio_pcm_hw_get_live_in (HWVoiceIn *hw);
+size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw);
-int audio_pcm_hw_clip_out (HWVoiceOut *hw, void *pcm_buf,
- int live, int pending);
+size_t audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf,
+ size_t live, size_t pending);
int audio_bug (const char *funcname, int cond);
void *audio_calloc (const char *funcname, int nmemb, size_t size);
@@ -223,7 +223,7 @@ void audio_run(AudioState *s, const char *msg);
#define VOICE_VOLUME_CAP (1 << VOICE_VOLUME)
-static inline int audio_ring_dist (int dst, int src, int len)
+static inline size_t audio_ring_dist(size_t dst, size_t src, size_t len)
{
return (dst >= src) ? (dst - src) : (len - src + dst);
}
diff --git a/audio/audio_template.h b/audio/audio_template.h
index ce1e5d6559..fecbf1a046 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -79,8 +79,8 @@ static int glue (audio_pcm_hw_alloc_resources_, TYPE) (HW *hw)
{
HWBUF = audio_calloc(__func__, hw->samples, sizeof(struct st_sample));
if (!HWBUF) {
- dolog ("Could not allocate " NAME " buffer (%d samples)\n",
- hw->samples);
+ dolog("Could not allocate " NAME " buffer (%zu samples)\n",
+ hw->samples);
return -1;
}
@@ -265,7 +265,7 @@ static HW *glue(audio_pcm_hw_add_new_, TYPE)(AudioState *s,
}
if (audio_bug(__func__, hw->samples <= 0)) {
- dolog ("hw->samples=%d\n", hw->samples);
+ dolog("hw->samples=%zd\n", hw->samples);
goto err1;
}
diff --git a/audio/mixeng.h b/audio/mixeng.h
index b53a5ef99a..2c09ed41e7 100644
--- a/audio/mixeng.h
+++ b/audio/mixeng.h
@@ -25,6 +25,8 @@
#ifndef QEMU_MIXENG_H
#define QEMU_MIXENG_H
+#include <stddef.h>
+
#ifdef FLOAT_MIXENG
typedef float mixeng_real;
struct mixeng_volume { int mute; mixeng_real r; mixeng_real l; };
@@ -33,6 +35,7 @@ struct st_sample { mixeng_real l; mixeng_real r; };
struct mixeng_volume { int mute; int64_t r; int64_t l; };
struct st_sample { int64_t l; int64_t r; };
#endif
+typedef struct st_sample st_sample;
typedef void (t_sample) (struct st_sample *dst, const void *src, int samples);
typedef void (f_sample) (void *dst, const struct st_sample *src, int samples);
@@ -41,10 +44,10 @@ extern t_sample *mixeng_conv[2][2][2][3];
extern f_sample *mixeng_clip[2][2][2][3];
void *st_rate_start (int inrate, int outrate);
-void st_rate_flow (void *opaque, struct st_sample *ibuf, struct st_sample *obuf,
- int *isamp, int *osamp);
-void st_rate_flow_mix (void *opaque, struct st_sample *ibuf, struct st_sample *obuf,
- int *isamp, int *osamp);
+void st_rate_flow(void *opaque, st_sample *ibuf, st_sample *obuf,
+ size_t *isamp, size_t *osamp);
+void st_rate_flow_mix(void *opaque, st_sample *ibuf, st_sample *obuf,
+ size_t *isamp, size_t *osamp);
void st_rate_stop (void *opaque);
void mixeng_clear (struct st_sample *buf, int len);
void mixeng_volume (struct st_sample *buf, int len, struct mixeng_volume *vol);
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 70909e5696..05c7d6f85c 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -476,8 +476,8 @@ static void oss_fini_out (HWVoiceOut *hw)
if (oss->mmapped) {
err = munmap (oss->pcm_buf, hw->samples << hw->info.shift);
if (err) {
- oss_logerr (errno, "Failed to unmap buffer %p, size %d\n",
- oss->pcm_buf, hw->samples << hw->info.shift);
+ oss_logerr(errno, "Failed to unmap buffer %p, size %zu\n",
+ oss->pcm_buf, hw->samples << hw->info.shift);
}
}
else {
@@ -543,8 +543,8 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
0
);
if (oss->pcm_buf == MAP_FAILED) {
- oss_logerr (errno, "Failed to map %d bytes of DAC\n",
- hw->samples << hw->info.shift);
+ oss_logerr(errno, "Failed to map %zu bytes of DAC\n",
+ hw->samples << hw->info.shift);
}
else {
int err;
@@ -568,8 +568,8 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
if (!oss->mmapped) {
err = munmap (oss->pcm_buf, hw->samples << hw->info.shift);
if (err) {
- oss_logerr (errno, "Failed to unmap buffer %p size %d\n",
- oss->pcm_buf, hw->samples << hw->info.shift);
+ oss_logerr(errno, "Failed to unmap buffer %p size %zu\n",
+ oss->pcm_buf, hw->samples << hw->info.shift);
}
}
}
@@ -581,7 +581,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
1 << hw->info.shift);
if (!oss->pcm_buf) {
dolog (
- "Could not allocate DAC buffer (%d samples, each %d bytes)\n",
+ "Could not allocate DAC buffer (%zu samples, each %d bytes)\n",
hw->samples,
1 << hw->info.shift
);
@@ -693,8 +693,8 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
oss->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
if (!oss->pcm_buf) {
- dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
- hw->samples, 1 << hw->info.shift);
+ dolog("Could not allocate ADC buffer (%zu samples, each %d bytes)\n",
+ hw->samples, 1 << hw->info.shift);
oss_anal_close (&fd);
return -1;
}
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 6a1919e93b..251b087a74 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -584,8 +584,8 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
pa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
pa->rpos = hw->rpos;
if (!pa->pcm_buf) {
- dolog ("Could not allocate buffer (%d bytes)\n",
- hw->samples << hw->info.shift);
+ dolog("Could not allocate buffer (%zu bytes)\n",
+ hw->samples << hw->info.shift);
goto fail2;
}
@@ -645,8 +645,8 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
pa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
pa->wpos = hw->wpos;
if (!pa->pcm_buf) {
- dolog ("Could not allocate buffer (%d bytes)\n",
- hw->samples << hw->info.shift);
+ dolog("Could not allocate buffer (%zu bytes)\n",
+ hw->samples << hw->info.shift);
goto fail2;
}
diff --git a/audio/rate_template.h b/audio/rate_template.h
index 6e93588877..f94c940c61 100644
--- a/audio/rate_template.h
+++ b/audio/rate_template.h
@@ -28,7 +28,7 @@
* Return number of samples processed.
*/
void NAME (void *opaque, struct st_sample *ibuf, struct st_sample *obuf,
- int *isamp, int *osamp)
+ size_t *isamp, size_t *osamp)
{
struct rate *rate = opaque;
struct st_sample *istart, *iend;
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index ab9166d054..92da4804c6 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -273,8 +273,7 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
}
if (audio_bug(__func__, sdl->live < 0 || sdl->live > hw->samples)) {
- dolog ("sdl->live=%d hw->samples=%d\n",
- sdl->live, hw->samples);
+ dolog("sdl->live=%d hw->samples=%zu\n", sdl->live, hw->samples);
return;
}
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index dda6993fb9..58300663ff 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -137,8 +137,8 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
hw->samples = 1024;
wav->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
if (!wav->pcm_buf) {
- dolog ("Could not allocate buffer (%d bytes)\n",
- hw->samples << hw->info.shift);
+ dolog("Could not allocate buffer (%zu bytes)\n",
+ hw->samples << hw->info.shift);
return -1;
}
diff --git a/include/sysemu/replay.h b/include/sysemu/replay.h
index 3a7c58e423..5c0a91e44f 100644
--- a/include/sysemu/replay.h
+++ b/include/sysemu/replay.h
@@ -179,9 +179,9 @@ void replay_net_packet_event(ReplayNetState *rns, unsigned flags,
/* Audio */
/*! Saves/restores number of played samples of audio out operation. */
-void replay_audio_out(int *played);
+void replay_audio_out(size_t *played);
/*! Saves/restores recorded samples of audio in operation. */
-void replay_audio_in(int *recorded, void *samples, int *wpos, int size);
+void replay_audio_in(size_t *recorded, void *samples, size_t *wpos, size_t size);
/* VM state operations */
diff --git a/replay/replay-audio.c b/replay/replay-audio.c
index b113836de4..efe1628727 100644
--- a/replay/replay-audio.c
+++ b/replay/replay-audio.c
@@ -16,18 +16,18 @@
#include "sysemu/sysemu.h"
#include "audio/audio.h"
-void replay_audio_out(int *played)
+void replay_audio_out(size_t *played)
{
if (replay_mode == REPLAY_MODE_RECORD) {
g_assert(replay_mutex_locked());
replay_save_instructions();
replay_put_event(EVENT_AUDIO_OUT);
- replay_put_dword(*played);
+ replay_put_qword(*played);
} else if (replay_mode == REPLAY_MODE_PLAY) {
g_assert(replay_mutex_locked());
replay_account_executed_instructions();
if (replay_next_event_is(EVENT_AUDIO_OUT)) {
- *played = replay_get_dword();
+ *played = replay_get_qword();
replay_finish_event();
} else {
error_report("Missing audio out event in the replay log");
@@ -36,7 +36,7 @@ void replay_audio_out(int *played)
}
}
-void replay_audio_in(int *recorded, void *samples, int *wpos, int size)
+void replay_audio_in(size_t *recorded, void *samples, size_t *wpos, size_t size)
{
int pos;
uint64_t left, right;
@@ -44,8 +44,8 @@ void replay_audio_in(int *recorded, void *samples, int *wpos, int size)
g_assert(replay_mutex_locked());
replay_save_instructions();
replay_put_event(EVENT_AUDIO_IN);
- replay_put_dword(*recorded);
- replay_put_dword(*wpos);
+ replay_put_qword(*recorded);
+ replay_put_qword(*wpos);
for (pos = (*wpos - *recorded + size) % size ; pos != *wpos
; pos = (pos + 1) % size) {
audio_sample_to_uint64(samples, pos, &left, &right);
@@ -56,8 +56,8 @@ void replay_audio_in(int *recorded, void *samples, int *wpos, int size)
g_assert(replay_mutex_locked());
replay_account_executed_instructions();
if (replay_next_event_is(EVENT_AUDIO_IN)) {
- *recorded = replay_get_dword();
- *wpos = replay_get_dword();
+ *recorded = replay_get_qword();
+ *wpos = replay_get_qword();
for (pos = (*wpos - *recorded + size) % size ; pos != *wpos
; pos = (pos + 1) % size) {
left = replay_get_qword();
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* Re: [Qemu-devel] [PATCH v2 27/52] audio: use size_t where makes sense
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 27/52] audio: use size_t where makes sense Kővágó, Zoltán
@ 2018-12-24 6:19 ` Pavel Dovgalyuk
2018-12-25 11:08 ` Philippe Mathieu-Daudé
1 sibling, 0 replies; 91+ messages in thread
From: Pavel Dovgalyuk @ 2018-12-24 6:19 UTC (permalink / raw)
To: 'Kővágó, Zoltán', qemu-devel
Cc: 'Gerd Hoffmann', 'Pavel Dovgalyuk',
'Paolo Bonzini'
> From: Kővágó, Zoltán [mailto:dirty.ice.hu@gmail.com]
> diff --git a/replay/replay-audio.c b/replay/replay-audio.c
> index b113836de4..efe1628727 100644
> --- a/replay/replay-audio.c
> +++ b/replay/replay-audio.c
> @@ -16,18 +16,18 @@
> #include "sysemu/sysemu.h"
> #include "audio/audio.h"
>
> -void replay_audio_out(int *played)
> +void replay_audio_out(size_t *played)
> {
> if (replay_mode == REPLAY_MODE_RECORD) {
> g_assert(replay_mutex_locked());
> replay_save_instructions();
> replay_put_event(EVENT_AUDIO_OUT);
> - replay_put_dword(*played);
> + replay_put_qword(*played);
> } else if (replay_mode == REPLAY_MODE_PLAY) {
> g_assert(replay_mutex_locked());
> replay_account_executed_instructions();
> if (replay_next_event_is(EVENT_AUDIO_OUT)) {
> - *played = replay_get_dword();
> + *played = replay_get_qword();
> replay_finish_event();
> } else {
> error_report("Missing audio out event in the replay log");
> @@ -36,7 +36,7 @@ void replay_audio_out(int *played)
> }
> }
>
> -void replay_audio_in(int *recorded, void *samples, int *wpos, int size)
> +void replay_audio_in(size_t *recorded, void *samples, size_t *wpos, size_t size)
> {
> int pos;
> uint64_t left, right;
> @@ -44,8 +44,8 @@ void replay_audio_in(int *recorded, void *samples, int *wpos, int size)
> g_assert(replay_mutex_locked());
> replay_save_instructions();
> replay_put_event(EVENT_AUDIO_IN);
> - replay_put_dword(*recorded);
> - replay_put_dword(*wpos);
> + replay_put_qword(*recorded);
> + replay_put_qword(*wpos);
> for (pos = (*wpos - *recorded + size) % size ; pos != *wpos
> ; pos = (pos + 1) % size) {
> audio_sample_to_uint64(samples, pos, &left, &right);
> @@ -56,8 +56,8 @@ void replay_audio_in(int *recorded, void *samples, int *wpos, int size)
> g_assert(replay_mutex_locked());
> replay_account_executed_instructions();
> if (replay_next_event_is(EVENT_AUDIO_IN)) {
> - *recorded = replay_get_dword();
> - *wpos = replay_get_dword();
> + *recorded = replay_get_qword();
> + *wpos = replay_get_qword();
> for (pos = (*wpos - *recorded + size) % size ; pos != *wpos
> ; pos = (pos + 1) % size) {
> left = replay_get_qword();
Please also increment REPLAY_VERSION in replay.c
It is required because log format is changed and newer version should not
be able reading old logs.
Pavel Dovgalyuk
^ permalink raw reply [flat|nested] 91+ messages in thread* Re: [Qemu-devel] [PATCH v2 27/52] audio: use size_t where makes sense
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 27/52] audio: use size_t where makes sense Kővágó, Zoltán
2018-12-24 6:19 ` Pavel Dovgalyuk
@ 2018-12-25 11:08 ` Philippe Mathieu-Daudé
2019-01-07 9:58 ` Gerd Hoffmann
1 sibling, 1 reply; 91+ messages in thread
From: Philippe Mathieu-Daudé @ 2018-12-25 11:08 UTC (permalink / raw)
To: Kővágó, Zoltán, qemu-devel
Cc: Paolo Bonzini, Gerd Hoffmann, Pavel Dovgalyuk
Hi Zoltán,
On 12/23/18 9:52 PM, Kővágó, Zoltán wrote:
> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
> ---
> audio/alsaaudio.c | 8 +-
> audio/audio.c | 162 ++++++++++++++++++++--------------------
> audio/audio.h | 4 +-
> audio/audio_int.h | 22 +++---
> audio/audio_template.h | 6 +-
> audio/mixeng.h | 11 ++-
> audio/ossaudio.c | 18 ++---
> audio/paaudio.c | 8 +-
> audio/rate_template.h | 2 +-
> audio/sdlaudio.c | 3 +-
> audio/wavaudio.c | 4 +-
> include/sysemu/replay.h | 4 +-
> replay/replay-audio.c | 16 ++--
> 13 files changed, 133 insertions(+), 135 deletions(-)
>
> diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
> index 19de7d01cb..69e7a3868c 100644
> --- a/audio/alsaaudio.c
> +++ b/audio/alsaaudio.c
> @@ -747,8 +747,8 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
>
> alsa->pcm_buf = audio_calloc(__func__, obt.samples, 1 << hw->info.shift);
> if (!alsa->pcm_buf) {
> - dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
> - hw->samples, 1 << hw->info.shift);
> + dolog("Could not allocate DAC buffer (%zu samples, each %d bytes)\n",
> + hw->samples, 1 << hw->info.shift);
> alsa_anal_close1 (&handle);
> return -1;
> }
> @@ -849,8 +849,8 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
>
> alsa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
> if (!alsa->pcm_buf) {
> - dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
> - hw->samples, 1 << hw->info.shift);
> + dolog("Could not allocate ADC buffer (%zu samples, each %d bytes)\n",
> + hw->samples, 1 << hw->info.shift);
> alsa_anal_close1 (&handle);
> return -1;
> }
> diff --git a/audio/audio.c b/audio/audio.c
> index 1ea80ba6a7..27a8a31a64 100644
> --- a/audio/audio.c
> +++ b/audio/audio.c
> @@ -530,10 +530,10 @@ static int audio_attach_capture (HWVoiceOut *hw)
> /*
> * Hard voice (capture)
> */
> -static int audio_pcm_hw_find_min_in (HWVoiceIn *hw)
> +static size_t audio_pcm_hw_find_min_in (HWVoiceIn *hw)
> {
> SWVoiceIn *sw;
> - int m = hw->total_samples_captured;
> + size_t m = hw->total_samples_captured;
>
> for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
> if (sw->active) {
> @@ -543,28 +543,28 @@ static int audio_pcm_hw_find_min_in (HWVoiceIn *hw)
> return m;
> }
>
> -int audio_pcm_hw_get_live_in (HWVoiceIn *hw)
> +size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw)
> {
> - int live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
> - if (audio_bug(__func__, live < 0 || live > hw->samples)) {
> - dolog ("live=%d hw->samples=%d\n", live, hw->samples);
> + size_t live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
> + if (audio_bug(__func__, live > hw->samples)) {
> + dolog("live=%zu hw->samples=%zu\n", live, hw->samples);
> return 0;
> }
> return live;
> }
>
> -int audio_pcm_hw_clip_out (HWVoiceOut *hw, void *pcm_buf,
> - int live, int pending)
> +size_t audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf,
> + size_t live, size_t pending)
> {
> - int left = hw->samples - pending;
> - int len = MIN (left, live);
> - int clipped = 0;
> + size_t left = hw->samples - pending;
> + size_t len = MIN (left, live);
> + size_t clipped = 0;
>
> while (len) {
> struct st_sample *src = hw->mix_buf + hw->rpos;
> uint8_t *dst = advance (pcm_buf, hw->rpos << hw->info.shift);
> - int samples_till_end_of_buf = hw->samples - hw->rpos;
> - int samples_to_clip = MIN (len, samples_till_end_of_buf);
> + size_t samples_till_end_of_buf = hw->samples - hw->rpos;
> + size_t samples_to_clip = MIN (len, samples_till_end_of_buf);
>
> hw->clip (dst, src, samples_to_clip);
>
> @@ -578,14 +578,14 @@ int audio_pcm_hw_clip_out (HWVoiceOut *hw, void *pcm_buf,
> /*
> * Soft voice (capture)
> */
> -static int audio_pcm_sw_get_rpos_in (SWVoiceIn *sw)
> +static size_t audio_pcm_sw_get_rpos_in(SWVoiceIn *sw)
> {
> HWVoiceIn *hw = sw->hw;
> - int live = hw->total_samples_captured - sw->total_hw_samples_acquired;
> - int rpos;
> + ssize_t live = hw->total_samples_captured - sw->total_hw_samples_acquired;
> + ssize_t rpos;
>
> if (audio_bug(__func__, live < 0 || live > hw->samples)) {
> - dolog ("live=%d hw->samples=%d\n", live, hw->samples);
> + dolog("live=%zu hw->samples=%zu\n", live, hw->samples);
> return 0;
> }
>
> @@ -598,17 +598,17 @@ static int audio_pcm_sw_get_rpos_in (SWVoiceIn *sw)
> }
> }
>
> -static int audio_pcm_sw_read(SWVoiceIn *sw, void *buf, int size)
> +static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
> {
> HWVoiceIn *hw = sw->hw;
> - int samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
> + size_t samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
> struct st_sample *src, *dst = sw->buf;
>
> rpos = audio_pcm_sw_get_rpos_in (sw) % hw->samples;
>
> live = hw->total_samples_captured - sw->total_hw_samples_acquired;
> - if (audio_bug(__func__, live < 0 || live > hw->samples)) {
> - dolog ("live_in=%d hw->samples=%d\n", live, hw->samples);
> + if (audio_bug(__func__, live > hw->samples)) {
> + dolog("live_in=%zu hw->samples=%zu\n", live, hw->samples);
> return 0;
> }
>
> @@ -622,9 +622,9 @@ static int audio_pcm_sw_read(SWVoiceIn *sw, void *buf, int size)
>
> while (swlim) {
> src = hw->conv_buf + rpos;
> - isamp = hw->wpos - rpos;
> - /* XXX: <= ? */
> - if (isamp <= 0) {
> + if (hw->wpos > rpos) {
> + isamp = hw->wpos - rpos;
> + } else {
> isamp = hw->samples - rpos;
> }
>
> @@ -633,11 +633,6 @@ static int audio_pcm_sw_read(SWVoiceIn *sw, void *buf, int size)
> }
> osamp = swlim;
>
> - if (audio_bug(__func__, osamp < 0)) {
> - dolog ("osamp=%d\n", osamp);
> - return 0;
> - }
> -
> st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
> swlim -= osamp;
> rpos = (rpos + isamp) % hw->samples;
> @@ -658,10 +653,10 @@ static int audio_pcm_sw_read(SWVoiceIn *sw, void *buf, int size)
> /*
> * Hard voice (playback)
> */
> -static int audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
> +static size_t audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
> {
> SWVoiceOut *sw;
> - int m = INT_MAX;
> + size_t m = SIZE_MAX;
> int nb_live = 0;
>
> for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
> @@ -675,9 +670,9 @@ static int audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
> return m;
> }
>
> -static int audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
> +static size_t audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
> {
> - int smin;
> + size_t smin;
> int nb_live1;
>
> smin = audio_pcm_hw_find_min_out (hw, &nb_live1);
> @@ -686,10 +681,10 @@ static int audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
> }
>
> if (nb_live1) {
> - int live = smin;
> + size_t live = smin;
>
> - if (audio_bug(__func__, live < 0 || live > hw->samples)) {
> - dolog ("live=%d hw->samples=%d\n", live, hw->samples);
> + if (audio_bug(__func__, live > hw->samples)) {
> + dolog("live=%zu hw->samples=%zu\n", live, hw->samples);
> return 0;
> }
> return live;
> @@ -700,10 +695,10 @@ static int audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
> /*
> * Soft voice (playback)
> */
> -static int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
> +static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
> {
> - int hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
> - int ret = 0, pos = 0, total = 0;
> + size_t hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
> + size_t ret = 0, pos = 0, total = 0;
>
> if (!sw) {
> return size;
> @@ -712,8 +707,8 @@ static int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
> hwsamples = sw->hw->samples;
>
> live = sw->total_hw_samples_mixed;
> - if (audio_bug(__func__, live < 0 || live > hwsamples)) {
> - dolog ("live=%d hw->samples=%d\n", live, hwsamples);
> + if (audio_bug(__func__, live > hwsamples)) {
> + dolog("live=%zu hw->samples=%zu\n", live, hwsamples);
> return 0;
> }
>
> @@ -767,7 +762,7 @@ static int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
>
> #ifdef DEBUG_OUT
> dolog (
> - "%s: write size %d ret %d total sw %d\n",
> + "%s: write size %zu ret %zu total sw %zu\n",
> SW_NAME (sw),
> size >> sw->info.shift,
> ret,
> @@ -846,7 +841,7 @@ static void audio_timer (void *opaque)
> /*
> * Public API
> */
> -int AUD_write (SWVoiceOut *sw, void *buf, int size)
> +size_t AUD_write(SWVoiceOut *sw, void *buf, size_t size)
> {
> if (!sw) {
> /* XXX: Consider options */
> @@ -861,7 +856,7 @@ int AUD_write (SWVoiceOut *sw, void *buf, int size)
> return audio_pcm_sw_write(sw, buf, size);
> }
>
> -int AUD_read (SWVoiceIn *sw, void *buf, int size)
> +size_t AUD_read(SWVoiceIn *sw, void *buf, size_t size)
> {
> if (!sw) {
> /* XXX: Consider options */
> @@ -970,17 +965,17 @@ void AUD_set_active_in (SWVoiceIn *sw, int on)
> }
> }
>
> -static int audio_get_avail (SWVoiceIn *sw)
> +static size_t audio_get_avail (SWVoiceIn *sw)
> {
> - int live;
> + size_t live;
>
> if (!sw) {
> return 0;
> }
>
> live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
> - if (audio_bug(__func__, live < 0 || live > sw->hw->samples)) {
> - dolog ("live=%d sw->hw->samples=%d\n", live, sw->hw->samples);
> + if (audio_bug(__func__, live > sw->hw->samples)) {
> + dolog("live=%zu sw->hw->samples=%zu\n", live, sw->hw->samples);
> return 0;
> }
>
> @@ -993,9 +988,9 @@ static int audio_get_avail (SWVoiceIn *sw)
> return (((int64_t) live << 32) / sw->ratio) << sw->info.shift;
> }
>
> -static int audio_get_free (SWVoiceOut *sw)
> +static size_t audio_get_free(SWVoiceOut *sw)
> {
> - int live, dead;
> + size_t live, dead;
>
> if (!sw) {
> return 0;
> @@ -1003,8 +998,8 @@ static int audio_get_free (SWVoiceOut *sw)
>
> live = sw->total_hw_samples_mixed;
>
> - if (audio_bug(__func__, live < 0 || live > sw->hw->samples)) {
> - dolog ("live=%d sw->hw->samples=%d\n", live, sw->hw->samples);
> + if (audio_bug(__func__, live > sw->hw->samples)) {
> + dolog("live=%zu sw->hw->samples=%zu\n", live, sw->hw->samples);
> return 0;
> }
>
> @@ -1019,9 +1014,10 @@ static int audio_get_free (SWVoiceOut *sw)
> return (((int64_t) dead << 32) / sw->ratio) << sw->info.shift;
> }
>
> -static void audio_capture_mix_and_clear (HWVoiceOut *hw, int rpos, int samples)
> +static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
> + size_t samples)
> {
> - int n;
> + size_t n;
>
> if (hw->enabled) {
> SWVoiceCap *sc;
> @@ -1032,17 +1028,17 @@ static void audio_capture_mix_and_clear (HWVoiceOut *hw, int rpos, int samples)
>
> n = samples;
> while (n) {
> - int till_end_of_hw = hw->samples - rpos2;
> - int to_write = MIN (till_end_of_hw, n);
> - int bytes = to_write << hw->info.shift;
> - int written;
> + size_t till_end_of_hw = hw->samples - rpos2;
> + size_t to_write = MIN(till_end_of_hw, n);
> + size_t bytes = to_write << hw->info.shift;
> + size_t written;
>
> sw->buf = hw->mix_buf + rpos2;
> written = audio_pcm_sw_write (sw, NULL, bytes);
> if (written - bytes) {
> - dolog ("Could not mix %d bytes into a capture "
> - "buffer, mixed %d\n",
> - bytes, written);
> + dolog("Could not mix %zu bytes into a capture "
> + "buffer, mixed %zu\n",
> + bytes, written);
> break;
> }
> n -= to_write;
> @@ -1051,9 +1047,9 @@ static void audio_capture_mix_and_clear (HWVoiceOut *hw, int rpos, int samples)
> }
> }
>
> - n = MIN (samples, hw->samples - rpos);
> - mixeng_clear (hw->mix_buf + rpos, n);
> - mixeng_clear (hw->mix_buf, samples - n);
> + n = MIN(samples, hw->samples - rpos);
> + mixeng_clear(hw->mix_buf + rpos, n);
> + mixeng_clear(hw->mix_buf, samples - n);
> }
>
> static void audio_run_out (AudioState *s)
> @@ -1062,16 +1058,16 @@ static void audio_run_out (AudioState *s)
> SWVoiceOut *sw;
>
> while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
> - int played;
> - int live, free, nb_live, cleanup_required, prev_rpos;
> + size_t played, live, prev_rpos, free;
> + int nb_live, cleanup_required;
>
> live = audio_pcm_hw_get_live_out (hw, &nb_live);
> if (!nb_live) {
> live = 0;
> }
>
> - if (audio_bug(__func__, live < 0 || live > hw->samples)) {
> - dolog ("live=%d hw->samples=%d\n", live, hw->samples);
> + if (audio_bug(__func__, live > hw->samples)) {
> + dolog ("live=%zu hw->samples=%zu\n", live, hw->samples);
> continue;
> }
>
> @@ -1106,13 +1102,13 @@ static void audio_run_out (AudioState *s)
> played = hw->pcm_ops->run_out (hw, live);
> replay_audio_out(&played);
> if (audio_bug(__func__, hw->rpos >= hw->samples)) {
> - dolog ("hw->rpos=%d hw->samples=%d played=%d\n",
> - hw->rpos, hw->samples, played);
> + dolog("hw->rpos=%zu hw->samples=%zu played=%zu\n",
> + hw->rpos, hw->samples, played);
> hw->rpos = 0;
> }
>
> #ifdef DEBUG_OUT
> - dolog ("played=%d\n", played);
> + dolog("played=%zu\n", played);
> #endif
>
> if (played) {
> @@ -1127,8 +1123,8 @@ static void audio_run_out (AudioState *s)
> }
>
> if (audio_bug(__func__, played > sw->total_hw_samples_mixed)) {
> - dolog ("played=%d sw->total_hw_samples_mixed=%d\n",
> - played, sw->total_hw_samples_mixed);
> + dolog("played=%zu sw->total_hw_samples_mixed=%zu\n",
> + played, sw->total_hw_samples_mixed);
> played = sw->total_hw_samples_mixed;
> }
>
> @@ -1168,7 +1164,7 @@ static void audio_run_in (AudioState *s)
>
> while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
> SWVoiceIn *sw;
> - int captured = 0, min;
> + size_t captured = 0, min;
>
> if (replay_mode != REPLAY_MODE_PLAY) {
> captured = hw->pcm_ops->run_in(hw);
> @@ -1183,7 +1179,7 @@ static void audio_run_in (AudioState *s)
> sw->total_hw_samples_acquired -= min;
>
> if (sw->active) {
> - int avail;
> + size_t avail;
>
> avail = audio_get_avail (sw);
> if (avail > 0) {
> @@ -1199,15 +1195,15 @@ static void audio_run_capture (AudioState *s)
> CaptureVoiceOut *cap;
>
> for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
> - int live, rpos, captured;
> + size_t live, rpos, captured;
> HWVoiceOut *hw = &cap->hw;
> SWVoiceOut *sw;
>
> captured = live = audio_pcm_hw_get_live_out (hw, NULL);
> rpos = hw->rpos;
> while (live) {
> - int left = hw->samples - rpos;
> - int to_capture = MIN (live, left);
> + size_t left = hw->samples - rpos;
> + size_t to_capture = MIN(live, left);
> struct st_sample *src;
> struct capture_callback *cb;
>
> @@ -1230,8 +1226,8 @@ static void audio_run_capture (AudioState *s)
> }
>
> if (audio_bug(__func__, captured > sw->total_hw_samples_mixed)) {
> - dolog ("captured=%d sw->total_hw_samples_mixed=%d\n",
> - captured, sw->total_hw_samples_mixed);
> + dolog("captured=%zu sw->total_hw_samples_mixed=%zu\n",
> + captured, sw->total_hw_samples_mixed);
> captured = sw->total_hw_samples_mixed;
> }
>
> @@ -1570,8 +1566,8 @@ CaptureVoiceOut *AUD_add_capture(
> hw->mix_buf = audio_calloc(__func__, hw->samples,
> sizeof(struct st_sample));
> if (!hw->mix_buf) {
> - dolog ("Could not allocate capture mix buffer (%d samples)\n",
> - hw->samples);
> + dolog("Could not allocate capture mix buffer (%zu samples)\n",
> + hw->samples);
> goto err2;
> }
>
> @@ -1580,7 +1576,7 @@ CaptureVoiceOut *AUD_add_capture(
> cap->buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
> if (!cap->buf) {
> dolog ("Could not allocate capture buffer "
> - "(%d samples, each %d bytes)\n",
> + "(%zu samples, each %d bytes)\n",
> hw->samples, 1 << hw->info.shift);
> goto err3;
> }
> diff --git a/audio/audio.h b/audio/audio.h
> index bcbe56d639..bfb12e7531 100644
> --- a/audio/audio.h
> +++ b/audio/audio.h
> @@ -114,7 +114,7 @@ SWVoiceOut *AUD_open_out (
> );
>
> void AUD_close_out (QEMUSoundCard *card, SWVoiceOut *sw);
> -int AUD_write (SWVoiceOut *sw, void *pcm_buf, int size);
Can you add short documentation while modifying headers? Such:
/**
* AUD_write:
*
* Returns: the number of bytes written.
*/
> +size_t AUD_write (SWVoiceOut *sw, void *pcm_buf, size_t size);
> int AUD_get_buffer_size_out (SWVoiceOut *sw);
> void AUD_set_active_out (SWVoiceOut *sw, int on);
> int AUD_is_active_out (SWVoiceOut *sw);
> @@ -135,7 +135,7 @@ SWVoiceIn *AUD_open_in (
> );
>
> void AUD_close_in (QEMUSoundCard *card, SWVoiceIn *sw);
> -int AUD_read (SWVoiceIn *sw, void *pcm_buf, int size);
> +size_t AUD_read (SWVoiceIn *sw, void *pcm_buf, size_t size);
> void AUD_set_active_in (SWVoiceIn *sw, int on);
> int AUD_is_active_in (SWVoiceIn *sw);
>
> diff --git a/audio/audio_int.h b/audio/audio_int.h
> index d269c38465..330c465d0b 100644
> --- a/audio/audio_int.h
> +++ b/audio/audio_int.h
> @@ -60,12 +60,12 @@ typedef struct HWVoiceOut {
>
> f_sample *clip;
>
> - int rpos;
> + size_t rpos;
> uint64_t ts_helper;
>
> struct st_sample *mix_buf;
>
> - int samples;
> + size_t samples;
> QLIST_HEAD (sw_out_listhead, SWVoiceOut) sw_head;
> QLIST_HEAD (sw_cap_listhead, SWVoiceCap) cap_head;
> int ctl_caps;
> @@ -81,13 +81,13 @@ typedef struct HWVoiceIn {
>
> t_sample *conv;
>
> - int wpos;
> - int total_samples_captured;
> + size_t wpos;
> + size_t total_samples_captured;
> uint64_t ts_helper;
>
> struct st_sample *conv_buf;
>
> - int samples;
> + size_t samples;
> QLIST_HEAD (sw_in_listhead, SWVoiceIn) sw_head;
> int ctl_caps;
> struct audio_pcm_ops *pcm_ops;
> @@ -102,7 +102,7 @@ struct SWVoiceOut {
> int64_t ratio;
> struct st_sample *buf;
> void *rate;
> - int total_hw_samples_mixed;
> + size_t total_hw_samples_mixed;
> int active;
> int empty;
> HWVoiceOut *hw;
> @@ -119,7 +119,7 @@ struct SWVoiceIn {
> struct audio_pcm_info info;
> int64_t ratio;
> void *rate;
> - int total_hw_samples_acquired;
> + size_t total_hw_samples_acquired;
> struct st_sample *buf;
> f_sample *clip;
> HWVoiceIn *hw;
You forgot to update the prototypes in audio_pcm_ops.
> @@ -207,10 +207,10 @@ audio_driver *audio_driver_lookup(const char *name);
> 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);
>
> -int audio_pcm_hw_get_live_in (HWVoiceIn *hw);
> +size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw);
>
> -int audio_pcm_hw_clip_out (HWVoiceOut *hw, void *pcm_buf,
> - int live, int pending);
> +size_t audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf,
> + size_t live, size_t pending);
>
> int audio_bug (const char *funcname, int cond);
> void *audio_calloc (const char *funcname, int nmemb, size_t size);
> @@ -223,7 +223,7 @@ void audio_run(AudioState *s, const char *msg);
>
> #define VOICE_VOLUME_CAP (1 << VOICE_VOLUME)
>
> -static inline int audio_ring_dist (int dst, int src, int len)
> +static inline size_t audio_ring_dist(size_t dst, size_t src, size_t len)
> {
> return (dst >= src) ? (dst - src) : (len - src + dst);
> }
> diff --git a/audio/audio_template.h b/audio/audio_template.h
> index ce1e5d6559..fecbf1a046 100644
> --- a/audio/audio_template.h
> +++ b/audio/audio_template.h
> @@ -79,8 +79,8 @@ static int glue (audio_pcm_hw_alloc_resources_, TYPE) (HW *hw)
Can you update this function to return a boolean?
> {
> HWBUF = audio_calloc(__func__, hw->samples, sizeof(struct st_sample));
> if (!HWBUF) {
> - dolog ("Could not allocate " NAME " buffer (%d samples)\n",
> - hw->samples);
> + dolog("Could not allocate " NAME " buffer (%zu samples)\n",
> + hw->samples);
> return -1;
> }
>
> @@ -265,7 +265,7 @@ static HW *glue(audio_pcm_hw_add_new_, TYPE)(AudioState *s,
> }
>
> if (audio_bug(__func__, hw->samples <= 0)) {
> - dolog ("hw->samples=%d\n", hw->samples);
> + dolog("hw->samples=%zd\n", hw->samples);
> goto err1;
> }
>
> diff --git a/audio/mixeng.h b/audio/mixeng.h
> index b53a5ef99a..2c09ed41e7 100644
> --- a/audio/mixeng.h
> +++ b/audio/mixeng.h
> @@ -25,6 +25,8 @@
> #ifndef QEMU_MIXENG_H
> #define QEMU_MIXENG_H
>
> +#include <stddef.h>
This shouldn't be necessary, since source files include "qemu/osdep.h".
$ git grep -L qemu/osdep.h audio/*.c
$
> +
> #ifdef FLOAT_MIXENG
> typedef float mixeng_real;
> struct mixeng_volume { int mute; mixeng_real r; mixeng_real l; };
> @@ -33,6 +35,7 @@ struct st_sample { mixeng_real l; mixeng_real r; };
> struct mixeng_volume { int mute; int64_t r; int64_t l; };
> struct st_sample { int64_t l; int64_t r; };
> #endif
> +typedef struct st_sample st_sample;
>
> typedef void (t_sample) (struct st_sample *dst, const void *src, int samples);
> typedef void (f_sample) (void *dst, const struct st_sample *src, int samples);
> @@ -41,10 +44,10 @@ extern t_sample *mixeng_conv[2][2][2][3];
> extern f_sample *mixeng_clip[2][2][2][3];
>
> void *st_rate_start (int inrate, int outrate);
> -void st_rate_flow (void *opaque, struct st_sample *ibuf, struct st_sample *obuf,
> - int *isamp, int *osamp);
> -void st_rate_flow_mix (void *opaque, struct st_sample *ibuf, struct st_sample *obuf,
> - int *isamp, int *osamp);
> +void st_rate_flow(void *opaque, st_sample *ibuf, st_sample *obuf,
> + size_t *isamp, size_t *osamp);
> +void st_rate_flow_mix(void *opaque, st_sample *ibuf, st_sample *obuf,
> + size_t *isamp, size_t *osamp);
> void st_rate_stop (void *opaque);
> void mixeng_clear (struct st_sample *buf, int len);
> void mixeng_volume (struct st_sample *buf, int len, struct mixeng_volume *vol);
> diff --git a/audio/ossaudio.c b/audio/ossaudio.c
> index 70909e5696..05c7d6f85c 100644
> --- a/audio/ossaudio.c
> +++ b/audio/ossaudio.c
> @@ -476,8 +476,8 @@ static void oss_fini_out (HWVoiceOut *hw)
> if (oss->mmapped) {
> err = munmap (oss->pcm_buf, hw->samples << hw->info.shift);
> if (err) {
> - oss_logerr (errno, "Failed to unmap buffer %p, size %d\n",
> - oss->pcm_buf, hw->samples << hw->info.shift);
> + oss_logerr(errno, "Failed to unmap buffer %p, size %zu\n",
> + oss->pcm_buf, hw->samples << hw->info.shift);
> }
> }
> else {
> @@ -543,8 +543,8 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
> 0
> );
> if (oss->pcm_buf == MAP_FAILED) {
> - oss_logerr (errno, "Failed to map %d bytes of DAC\n",
> - hw->samples << hw->info.shift);
> + oss_logerr(errno, "Failed to map %zu bytes of DAC\n",
> + hw->samples << hw->info.shift);
> }
> else {
> int err;
> @@ -568,8 +568,8 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
> if (!oss->mmapped) {
> err = munmap (oss->pcm_buf, hw->samples << hw->info.shift);
> if (err) {
> - oss_logerr (errno, "Failed to unmap buffer %p size %d\n",
> - oss->pcm_buf, hw->samples << hw->info.shift);
> + oss_logerr(errno, "Failed to unmap buffer %p size %zu\n",
> + oss->pcm_buf, hw->samples << hw->info.shift);
> }
> }
> }
> @@ -581,7 +581,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
> 1 << hw->info.shift);
> if (!oss->pcm_buf) {
> dolog (
> - "Could not allocate DAC buffer (%d samples, each %d bytes)\n",
> + "Could not allocate DAC buffer (%zu samples, each %d bytes)\n",
> hw->samples,
> 1 << hw->info.shift
> );
> @@ -693,8 +693,8 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
> hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
> oss->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
> if (!oss->pcm_buf) {
> - dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
> - hw->samples, 1 << hw->info.shift);
> + dolog("Could not allocate ADC buffer (%zu samples, each %d bytes)\n",
> + hw->samples, 1 << hw->info.shift);
> oss_anal_close (&fd);
> return -1;
> }
> diff --git a/audio/paaudio.c b/audio/paaudio.c
> index 6a1919e93b..251b087a74 100644
> --- a/audio/paaudio.c
> +++ b/audio/paaudio.c
> @@ -584,8 +584,8 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
> pa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
> pa->rpos = hw->rpos;
> if (!pa->pcm_buf) {
> - dolog ("Could not allocate buffer (%d bytes)\n",
> - hw->samples << hw->info.shift);
> + dolog("Could not allocate buffer (%zu bytes)\n",
> + hw->samples << hw->info.shift);
> goto fail2;
> }
>
> @@ -645,8 +645,8 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
> pa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
> pa->wpos = hw->wpos;
> if (!pa->pcm_buf) {
> - dolog ("Could not allocate buffer (%d bytes)\n",
> - hw->samples << hw->info.shift);
> + dolog("Could not allocate buffer (%zu bytes)\n",
> + hw->samples << hw->info.shift);
> goto fail2;
> }
>
> diff --git a/audio/rate_template.h b/audio/rate_template.h
> index 6e93588877..f94c940c61 100644
> --- a/audio/rate_template.h
> +++ b/audio/rate_template.h
> @@ -28,7 +28,7 @@
> * Return number of samples processed.
> */
> void NAME (void *opaque, struct st_sample *ibuf, struct st_sample *obuf,
> - int *isamp, int *osamp)
> + size_t *isamp, size_t *osamp)
> {
> struct rate *rate = opaque;
> struct st_sample *istart, *iend;
> diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
> index ab9166d054..92da4804c6 100644
> --- a/audio/sdlaudio.c
> +++ b/audio/sdlaudio.c
> @@ -273,8 +273,7 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
> }
>
> if (audio_bug(__func__, sdl->live < 0 || sdl->live > hw->samples)) {
> - dolog ("sdl->live=%d hw->samples=%d\n",
> - sdl->live, hw->samples);
> + dolog("sdl->live=%d hw->samples=%zu\n", sdl->live, hw->samples);
> return;
> }
>
> diff --git a/audio/wavaudio.c b/audio/wavaudio.c
> index dda6993fb9..58300663ff 100644
> --- a/audio/wavaudio.c
> +++ b/audio/wavaudio.c
> @@ -137,8 +137,8 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
> hw->samples = 1024;
> wav->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
> if (!wav->pcm_buf) {
> - dolog ("Could not allocate buffer (%d bytes)\n",
> - hw->samples << hw->info.shift);
> + dolog("Could not allocate buffer (%zu bytes)\n",
> + hw->samples << hw->info.shift);
> return -1;
> }
>
> diff --git a/include/sysemu/replay.h b/include/sysemu/replay.h
> index 3a7c58e423..5c0a91e44f 100644
> --- a/include/sysemu/replay.h
> +++ b/include/sysemu/replay.h
> @@ -179,9 +179,9 @@ void replay_net_packet_event(ReplayNetState *rns, unsigned flags,
> /* Audio */
>
> /*! Saves/restores number of played samples of audio out operation. */
> -void replay_audio_out(int *played);
> +void replay_audio_out(size_t *played);
> /*! Saves/restores recorded samples of audio in operation. */
> -void replay_audio_in(int *recorded, void *samples, int *wpos, int size);
> +void replay_audio_in(size_t *recorded, void *samples, size_t *wpos, size_t size);
>
> /* VM state operations */
>
> diff --git a/replay/replay-audio.c b/replay/replay-audio.c
> index b113836de4..efe1628727 100644
> --- a/replay/replay-audio.c
> +++ b/replay/replay-audio.c
> @@ -16,18 +16,18 @@
> #include "sysemu/sysemu.h"
> #include "audio/audio.h"
>
> -void replay_audio_out(int *played)
> +void replay_audio_out(size_t *played)
> {
> if (replay_mode == REPLAY_MODE_RECORD) {
> g_assert(replay_mutex_locked());
> replay_save_instructions();
> replay_put_event(EVENT_AUDIO_OUT);
> - replay_put_dword(*played);
> + replay_put_qword(*played);
> } else if (replay_mode == REPLAY_MODE_PLAY) {
> g_assert(replay_mutex_locked());
> replay_account_executed_instructions();
> if (replay_next_event_is(EVENT_AUDIO_OUT)) {
> - *played = replay_get_dword();
> + *played = replay_get_qword();
> replay_finish_event();
> } else {
> error_report("Missing audio out event in the replay log");
> @@ -36,7 +36,7 @@ void replay_audio_out(int *played)
> }
> }
>
> -void replay_audio_in(int *recorded, void *samples, int *wpos, int size)
> +void replay_audio_in(size_t *recorded, void *samples, size_t *wpos, size_t size)
> {
> int pos;
> uint64_t left, right;
> @@ -44,8 +44,8 @@ void replay_audio_in(int *recorded, void *samples, int *wpos, int size)
> g_assert(replay_mutex_locked());
> replay_save_instructions();
> replay_put_event(EVENT_AUDIO_IN);
> - replay_put_dword(*recorded);
> - replay_put_dword(*wpos);
> + replay_put_qword(*recorded);
> + replay_put_qword(*wpos);
> for (pos = (*wpos - *recorded + size) % size ; pos != *wpos
> ; pos = (pos + 1) % size) {
> audio_sample_to_uint64(samples, pos, &left, &right);
> @@ -56,8 +56,8 @@ void replay_audio_in(int *recorded, void *samples, int *wpos, int size)
> g_assert(replay_mutex_locked());
> replay_account_executed_instructions();
> if (replay_next_event_is(EVENT_AUDIO_IN)) {
> - *recorded = replay_get_dword();
> - *wpos = replay_get_dword();
> + *recorded = replay_get_qword();
> + *wpos = replay_get_qword();
> for (pos = (*wpos - *recorded + size) % size ; pos != *wpos
> ; pos = (pos + 1) % size) {
> left = replay_get_qword();
>
Except the audio_pcm_ops prototypes not using size_t, and Pavel comment
about replay version update, this patch looks sane to me.
Returning size_t seems enough (no need to return a ssize_t).
Regards,
Phil.
^ permalink raw reply [flat|nested] 91+ messages in thread* Re: [Qemu-devel] [PATCH v2 27/52] audio: use size_t where makes sense
2018-12-25 11:08 ` Philippe Mathieu-Daudé
@ 2019-01-07 9:58 ` Gerd Hoffmann
0 siblings, 0 replies; 91+ messages in thread
From: Gerd Hoffmann @ 2019-01-07 9:58 UTC (permalink / raw)
To: Philippe Mathieu-Daudé
Cc: Kővágó,
Zoltán, qemu-devel, Paolo Bonzini, Pavel Dovgalyuk
Hi,
> > void AUD_close_out (QEMUSoundCard *card, SWVoiceOut *sw);
> > -int AUD_write (SWVoiceOut *sw, void *pcm_buf, int size);
>
> Can you add short documentation while modifying headers? Such:
>
> /**
> * AUD_write:
> *
> * Returns: the number of bytes written.
> */
While this certainly makes sense I don't think this should be done as
part of this patch series, it is already big enough as-is.
cheers,
Gerd
^ permalink raw reply [flat|nested] 91+ messages in thread
* [Qemu-devel] [PATCH v2 28/52] audio: api for mixeng code free backends
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (26 preceding siblings ...)
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 27/52] audio: use size_t where makes sense Kővágó, Zoltán
@ 2018-12-23 20:52 ` Kővágó, Zoltán
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 29/52] alsaaudio: port to the new audio backend api Kővágó, Zoltán
` (22 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:52 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio.c | 211 ++++++++++++++++++++++++++++++++++++++++-
audio/audio_int.h | 45 +++++++--
audio/audio_template.h | 1 +
3 files changed, 247 insertions(+), 10 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index 27a8a31a64..1897e22444 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -575,6 +575,25 @@ size_t audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf,
return clipped;
}
+static void audio_pcm_hw_clip_out2(HWVoiceOut *hw, void *pcm_buf, size_t len)
+{
+ size_t clipped = 0;
+ size_t pos = hw->rpos;
+
+ while (len) {
+ st_sample *src = hw->mix_buf + pos;
+ uint8_t *dst = advance (pcm_buf, clipped << hw->info.shift);
+ size_t samples_till_end_of_buf = hw->samples - pos;
+ size_t samples_to_clip = MIN(len, samples_till_end_of_buf);
+
+ hw->clip (dst, src, samples_to_clip);
+
+ pos = (pos + samples_to_clip) % hw->samples;
+ len -= samples_to_clip;
+ clipped += samples_to_clip;
+ }
+}
+
/*
* Soft voice (capture)
*/
@@ -1052,6 +1071,31 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
mixeng_clear(hw->mix_buf, samples - n);
}
+static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
+{
+ size_t clipped = 0;
+
+ while (live) {
+ size_t size, decr, proc;
+ void *buf = hw->pcm_ops->get_buffer_out(hw, &size);
+
+ decr = MIN(size >> hw->info.shift, live);
+ audio_pcm_hw_clip_out2(hw, buf, decr);
+ proc = hw->pcm_ops->put_buffer_out(hw, buf, decr << hw->info.shift) >>
+ hw->info.shift;
+
+ live -= proc;
+ clipped += proc;
+ hw->rpos = (hw->rpos + proc) % hw->samples;
+
+ if (proc == 0 || proc < decr) {
+ break;
+ }
+ }
+
+ return clipped;
+}
+
static void audio_run_out (AudioState *s)
{
HWVoiceOut *hw = NULL;
@@ -1099,7 +1143,11 @@ static void audio_run_out (AudioState *s)
}
prev_rpos = hw->rpos;
- played = hw->pcm_ops->run_out (hw, live);
+ if (hw->pcm_ops->run_out) {
+ played = hw->pcm_ops->run_out(hw, live);
+ } else {
+ played = audio_pcm_hw_run_out(hw, live);
+ }
replay_audio_out(&played);
if (audio_bug(__func__, hw->rpos >= hw->samples)) {
dolog("hw->rpos=%zu hw->samples=%zu played=%zu\n",
@@ -1158,6 +1206,35 @@ static void audio_run_out (AudioState *s)
}
}
+static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
+{
+ size_t conv = 0;
+
+ while (samples) {
+ size_t proc;
+ size_t size = samples << hw->info.shift;
+ void *buf = hw->pcm_ops->get_buffer_in(hw, &size);
+
+ assert((size & hw->info.align) == 0);
+ if (size == 0) {
+ hw->pcm_ops->put_buffer_in(hw, buf, size);
+ break;
+ }
+
+ proc = MIN(size >> hw->info.shift,
+ hw->samples - hw->wpos);
+
+ hw->conv(hw->conv_buf + hw->wpos, buf, proc);
+ hw->wpos = (hw->wpos + proc) % hw->samples;
+
+ samples -= proc;
+ conv += proc;
+ hw->pcm_ops->put_buffer_in(hw, buf, proc << hw->info.shift);
+ }
+
+ return conv;
+}
+
static void audio_run_in (AudioState *s)
{
HWVoiceIn *hw = NULL;
@@ -1167,7 +1244,12 @@ static void audio_run_in (AudioState *s)
size_t captured = 0, min;
if (replay_mode != REPLAY_MODE_PLAY) {
- captured = hw->pcm_ops->run_in(hw);
+ if (hw->pcm_ops->run_in) {
+ captured = hw->pcm_ops->run_in(hw);
+ } else {
+ captured = audio_pcm_hw_run_in(
+ hw, hw->samples - audio_pcm_hw_get_live_in(hw));
+ }
}
replay_audio_in(&captured, hw->conv_buf, &hw->wpos, hw->samples);
@@ -1261,12 +1343,137 @@ void audio_run(AudioState *s, const char *msg)
#endif
}
+void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size)
+{
+ ssize_t start;
+
+ if (unlikely(!hw->buf_emul)) {
+ size_t calc_size = hw->samples << hw->info.shift;
+ hw->buf_emul = g_malloc(calc_size);
+ hw->size_emul = calc_size;
+ hw->pos_emul = hw->pending_emul = 0;
+ }
+
+ while (hw->pending_emul < hw->size_emul) {
+ size_t read_len = MIN(hw->size_emul - hw->pos_emul,
+ hw->size_emul - hw->pending_emul);
+ size_t read = hw->pcm_ops->read(hw, hw->buf_emul + hw->pos_emul,
+ read_len);
+ hw->pending_emul += read;
+ if (read < read_len) {
+ break;
+ }
+ }
+
+ start = ((ssize_t) hw->pos_emul) - hw->pending_emul;
+ if (start < 0) {
+ start += hw->size_emul;
+ }
+ assert(start >= 0 && start < hw->size_emul);
+
+ *size = MIN(hw->pending_emul, hw->size_emul - start);
+ return hw->buf_emul + start;
+}
+
+void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size)
+{
+ assert(size <= hw->pending_emul);
+ hw->pending_emul -= size;
+}
+
+void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size)
+{
+ if (unlikely(!hw->buf_emul)) {
+ size_t calc_size = hw->samples << hw->info.shift;
+
+ hw->buf_emul = g_malloc(calc_size);
+ hw->size_emul = calc_size;
+ hw->pos_emul = hw->pending_emul = 0;
+ }
+
+ *size = MIN(hw->size_emul - hw->pending_emul,
+ hw->size_emul - hw->pos_emul);
+ return hw->buf_emul + hw->pos_emul;
+}
+
+size_t audio_generic_put_buffer_out_nowrite(HWVoiceOut *hw, void *buf,
+ size_t size)
+{
+ assert(buf == hw->buf_emul + hw->pos_emul &&
+ size + hw->pending_emul <= hw->size_emul);
+
+ hw->pending_emul += size;
+ hw->pos_emul = (hw->pos_emul + size) % hw->size_emul;
+
+ return size;
+}
+
+size_t audio_generic_put_buffer_out(HWVoiceOut *hw, void *buf, size_t size)
+{
+ audio_generic_put_buffer_out_nowrite(hw, buf, size);
+
+ while (hw->pending_emul) {
+ size_t write_len, written;
+ ssize_t start = ((ssize_t) hw->pos_emul) - hw->pending_emul;
+ if (start < 0) {
+ start += hw->size_emul;
+ }
+ assert(start >= 0 && start < hw->size_emul);
+
+ write_len = MIN(hw->pending_emul, hw->size_emul - start);
+
+ written = hw->pcm_ops->write(hw, hw->buf_emul + start, write_len);
+ hw->pending_emul -= written;
+
+ if (written < write_len) {
+ break;
+ }
+ }
+
+ /*
+ * fake we have written everything. non-written data remain in pending_emul,
+ * so we do not have to clip them multiple times
+ */
+ return size;
+}
+
+size_t audio_generic_write(HWVoiceOut *hw, void *buf, size_t size)
+{
+ size_t dst_size, copy_size;
+ void *dst = hw->pcm_ops->get_buffer_out(hw, &dst_size);
+ copy_size = MIN(size, dst_size);
+
+ memcpy(dst, buf, copy_size);
+ return hw->pcm_ops->put_buffer_out(hw, buf, copy_size);
+}
+
+size_t audio_generic_read(HWVoiceIn *hw, void *buf, size_t size)
+{
+ size_t dst_size, copy_size;
+ void *dst = hw->pcm_ops->get_buffer_in(hw, &dst_size);
+ copy_size = MIN(size, dst_size);
+
+ memcpy(dst, buf, copy_size);
+ hw->pcm_ops->put_buffer_in(hw, buf, copy_size);
+ return copy_size;
+}
+
+
static int audio_driver_init(AudioState *s, struct audio_driver *drv,
Audiodev *dev)
{
s->drv_opaque = drv->init(dev);
if (s->drv_opaque) {
+ if (!drv->pcm_ops->get_buffer_in) {
+ drv->pcm_ops->get_buffer_in = audio_generic_get_buffer_in;
+ drv->pcm_ops->put_buffer_in = audio_generic_put_buffer_in;
+ }
+ if (!drv->pcm_ops->get_buffer_out) {
+ drv->pcm_ops->get_buffer_out = audio_generic_get_buffer_out;
+ drv->pcm_ops->put_buffer_out = audio_generic_put_buffer_out;
+ }
+
audio_init_nb_voices_out(s, drv);
audio_init_nb_voices_in(s, drv);
s->drv = drv;
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 330c465d0b..5c361d9451 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -64,6 +64,8 @@ typedef struct HWVoiceOut {
uint64_t ts_helper;
struct st_sample *mix_buf;
+ void *buf_emul;
+ size_t pos_emul, pending_emul, size_emul;
size_t samples;
QLIST_HEAD (sw_out_listhead, SWVoiceOut) sw_head;
@@ -86,6 +88,8 @@ typedef struct HWVoiceIn {
uint64_t ts_helper;
struct st_sample *conv_buf;
+ void *buf_emul;
+ size_t pos_emul, pending_emul, size_emul;
size_t samples;
QLIST_HEAD (sw_in_listhead, SWVoiceIn) sw_head;
@@ -146,17 +150,42 @@ struct audio_driver {
};
struct audio_pcm_ops {
- int (*init_out)(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque);
- void (*fini_out)(HWVoiceOut *hw);
- int (*run_out) (HWVoiceOut *hw, int live);
- int (*ctl_out) (HWVoiceOut *hw, int cmd, ...);
+ int (*init_out)(HWVoiceOut *hw, audsettings *as, void *drv_opaque);
+ void (*fini_out)(HWVoiceOut *hw);
+ int (*run_out) (HWVoiceOut *hw, int live);
+ size_t (*write) (HWVoiceOut *hw, void *buf, size_t size);
+ /*
+ * get a buffer that after later can be passed to put_buffer_out; optional
+ * returns the buffer, and writes it's size to size (in bytes)
+ * this is unrelated to the above buffer_size_out function
+ */
+ void *(*get_buffer_out)(HWVoiceOut *hw, size_t *size);
+ /*
+ * put back the buffer returned by get_buffer_out; optional
+ * buf must be equal the pointer returned by get_buffer_out,
+ * size may be smaller
+ */
+ size_t (*put_buffer_out)(HWVoiceOut *hw, void *buf, size_t size);
+ int (*ctl_out) (HWVoiceOut *hw, int cmd, ...);
- int (*init_in) (HWVoiceIn *hw, struct audsettings *as, void *drv_opaque);
- void (*fini_in) (HWVoiceIn *hw);
- int (*run_in) (HWVoiceIn *hw);
- int (*ctl_in) (HWVoiceIn *hw, int cmd, ...);
+ int (*init_in) (HWVoiceIn *hw, audsettings *as, void *drv_opaque);
+ void (*fini_in) (HWVoiceIn *hw);
+ int (*run_in) (HWVoiceIn *hw);
+ size_t (*read) (HWVoiceIn *hw, void *buf, size_t size);
+ void *(*get_buffer_in)(HWVoiceIn *hw, size_t *size);
+ void (*put_buffer_in)(HWVoiceIn *hw, void *buf, size_t size);
+ int (*ctl_in) (HWVoiceIn *hw, int cmd, ...);
};
+void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size);
+void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size);
+void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size);
+size_t audio_generic_put_buffer_out(HWVoiceOut *hw, void *buf, size_t size);
+size_t audio_generic_put_buffer_out_nowrite(HWVoiceOut *hw, void *buf,
+ size_t size);
+size_t audio_generic_write(HWVoiceOut *hw, void *buf, size_t size);
+size_t audio_generic_read(HWVoiceIn *hw, void *buf, size_t size);
+
struct capture_callback {
struct audio_capture_ops ops;
void *opaque;
diff --git a/audio/audio_template.h b/audio/audio_template.h
index fecbf1a046..33cdb66d7f 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -71,6 +71,7 @@ static void glue(audio_init_nb_voices_, TYPE)(AudioState *s,
static void glue (audio_pcm_hw_free_resources_, TYPE) (HW *hw)
{
+ g_free(hw->buf_emul);
g_free (HWBUF);
HWBUF = NULL;
}
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 29/52] alsaaudio: port to the new audio backend api
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (27 preceding siblings ...)
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 28/52] audio: api for mixeng code free backends Kővágó, Zoltán
@ 2018-12-23 20:52 ` Kővágó, Zoltán
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 30/52] coreaudio: " Kővágó, Zoltán
` (21 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:52 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/alsaaudio.c | 306 ++++++++++++----------------------------------
1 file changed, 81 insertions(+), 225 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 69e7a3868c..56271b1174 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -44,9 +44,6 @@ struct pollhlp {
typedef struct ALSAVoiceOut {
HWVoiceOut hw;
- int wpos;
- int pending;
- void *pcm_buf;
snd_pcm_t *handle;
struct pollhlp pollhlp;
Audiodev *dev;
@@ -55,7 +52,6 @@ typedef struct ALSAVoiceOut {
typedef struct ALSAVoiceIn {
HWVoiceIn hw;
snd_pcm_t *handle;
- void *pcm_buf;
struct pollhlp pollhlp;
Audiodev *dev;
} ALSAVoiceIn;
@@ -610,102 +606,62 @@ static int alsa_open(bool in, struct alsa_params_req *req,
return -1;
}
-static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
-{
- snd_pcm_sframes_t avail;
-
- avail = snd_pcm_avail_update (handle);
- if (avail < 0) {
- if (avail == -EPIPE) {
- if (!alsa_recover (handle)) {
- avail = snd_pcm_avail_update (handle);
- }
- }
-
- if (avail < 0) {
- alsa_logerr (avail,
- "Could not obtain number of available frames\n");
- return -1;
- }
- }
-
- return avail;
-}
-
-static void alsa_write_pending (ALSAVoiceOut *alsa)
-{
- HWVoiceOut *hw = &alsa->hw;
-
- while (alsa->pending) {
- int left_till_end_samples = hw->samples - alsa->wpos;
- int len = MIN (alsa->pending, left_till_end_samples);
- char *src = advance (alsa->pcm_buf, alsa->wpos << hw->info.shift);
-
- while (len) {
- snd_pcm_sframes_t written;
-
- written = snd_pcm_writei (alsa->handle, src, len);
-
- if (written <= 0) {
- switch (written) {
- case 0:
- trace_alsa_wrote_zero(len);
- return;
-
- case -EPIPE:
- if (alsa_recover (alsa->handle)) {
- alsa_logerr (written, "Failed to write %d frames\n",
- len);
- return;
- }
- trace_alsa_xrun_out();
- continue;
-
- case -ESTRPIPE:
- /* stream is suspended and waiting for an
- application recovery */
- if (alsa_resume (alsa->handle)) {
- alsa_logerr (written, "Failed to write %d frames\n",
- len);
- return;
- }
- trace_alsa_resume_out();
- continue;
-
- case -EAGAIN:
- return;
-
- default:
- alsa_logerr (written, "Failed to write %d frames from %p\n",
- len, src);
- return;
- }
- }
-
- alsa->wpos = (alsa->wpos + written) % hw->samples;
- alsa->pending -= written;
- len -= written;
- }
- }
-}
-
-static int alsa_run_out (HWVoiceOut *hw, int live)
+static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
- int decr;
- snd_pcm_sframes_t avail;
+ size_t pos = 0;
+ size_t len_frames = len >> hw->info.shift;
- avail = alsa_get_avail (alsa->handle);
- if (avail < 0) {
- dolog ("Could not get number of available playback frames\n");
- return 0;
+ while (len_frames) {
+ char *src = advance(buf, pos);
+ snd_pcm_sframes_t written;
+
+ written = snd_pcm_writei(alsa->handle, src, len_frames);
+
+ if (written <= 0) {
+ switch (written) {
+ case 0:
+ trace_alsa_wrote_zero(len_frames);
+ return pos;
+
+ case -EPIPE:
+ if (alsa_recover(alsa->handle)) {
+ alsa_logerr(written, "Failed to write %zu frames\n",
+ len_frames);
+ return pos;
+ }
+ trace_alsa_xrun_out();
+ continue;
+
+ case -ESTRPIPE:
+ /* stream is suspended and waiting for an
+ application recovery */
+ if (alsa_resume(alsa->handle)) {
+ alsa_logerr(written, "Failed to write %zu frames\n",
+ len_frames);
+ return pos;
+ }
+ trace_alsa_resume_out();
+ continue;
+
+ case -EAGAIN:
+ return pos;
+
+ default:
+ alsa_logerr(written, "Failed to write %zu frames from %p\n",
+ len, src);
+ return pos;
+ }
+ }
+
+ pos += written << hw->info.shift;
+ if (written < len_frames) {
+ break;
+ }
+ len_frames -= written;
}
- decr = MIN (live, avail);
- decr = audio_pcm_hw_clip_out (hw, alsa->pcm_buf, decr, alsa->pending);
- alsa->pending += decr;
- alsa_write_pending (alsa);
- return decr;
+ return pos;
}
static void alsa_fini_out (HWVoiceOut *hw)
@@ -714,9 +670,6 @@ static void alsa_fini_out (HWVoiceOut *hw)
ldebug ("alsa_fini\n");
alsa_anal_close (&alsa->handle, &alsa->pollhlp);
-
- g_free(alsa->pcm_buf);
- alsa->pcm_buf = NULL;
}
static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
@@ -745,14 +698,6 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = obt.samples;
- alsa->pcm_buf = audio_calloc(__func__, obt.samples, 1 << hw->info.shift);
- if (!alsa->pcm_buf) {
- dolog("Could not allocate DAC buffer (%zu samples, each %d bytes)\n",
- hw->samples, 1 << hw->info.shift);
- alsa_anal_close1 (&handle);
- return -1;
- }
-
alsa->pollhlp.s = hw->s;
alsa->handle = handle;
alsa->dev = dev;
@@ -847,14 +792,6 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = obt.samples;
- alsa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
- if (!alsa->pcm_buf) {
- dolog("Could not allocate ADC buffer (%zu samples, each %d bytes)\n",
- hw->samples, 1 << hw->info.shift);
- alsa_anal_close1 (&handle);
- return -1;
- }
-
alsa->pollhlp.s = hw->s;
alsa->handle = handle;
alsa->dev = dev;
@@ -866,129 +803,48 @@ static void alsa_fini_in (HWVoiceIn *hw)
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
alsa_anal_close (&alsa->handle, &alsa->pollhlp);
-
- g_free(alsa->pcm_buf);
- alsa->pcm_buf = NULL;
}
-static int alsa_run_in (HWVoiceIn *hw)
+static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
- int hwshift = hw->info.shift;
- int i;
- int live = audio_pcm_hw_get_live_in (hw);
- int dead = hw->samples - live;
- int decr;
- struct {
- int add;
- int len;
- } bufs[2] = {
- { .add = hw->wpos, .len = 0 },
- { .add = 0, .len = 0 }
- };
- snd_pcm_sframes_t avail;
- snd_pcm_uframes_t read_samples = 0;
+ size_t pos = 0;
- if (!dead) {
- return 0;
- }
-
- avail = alsa_get_avail (alsa->handle);
- if (avail < 0) {
- dolog ("Could not get number of captured frames\n");
- return 0;
- }
-
- if (!avail) {
- snd_pcm_state_t state;
-
- state = snd_pcm_state (alsa->handle);
- switch (state) {
- case SND_PCM_STATE_PREPARED:
- avail = hw->samples;
- break;
- case SND_PCM_STATE_SUSPENDED:
- /* stream is suspended and waiting for an application recovery */
- if (alsa_resume (alsa->handle)) {
- dolog ("Failed to resume suspended input stream\n");
- return 0;
- }
- trace_alsa_resume_in();
- break;
- default:
- trace_alsa_no_frames(state);
- return 0;
- }
- }
-
- decr = MIN (dead, avail);
- if (!decr) {
- return 0;
- }
-
- if (hw->wpos + decr > hw->samples) {
- bufs[0].len = (hw->samples - hw->wpos);
- bufs[1].len = (decr - (hw->samples - hw->wpos));
- }
- else {
- bufs[0].len = decr;
- }
-
- for (i = 0; i < 2; ++i) {
- void *src;
- struct st_sample *dst;
+ while (len) {
+ void *dst = advance(buf, pos);
snd_pcm_sframes_t nread;
- snd_pcm_uframes_t len;
- len = bufs[i].len;
+ nread = snd_pcm_readi(alsa->handle, dst, len >> hw->info.shift);
- src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
- dst = hw->conv_buf + bufs[i].add;
+ if (nread <= 0) {
+ switch (nread) {
+ case 0:
+ trace_alsa_read_zero(len);
+ return pos;;
- while (len) {
- nread = snd_pcm_readi (alsa->handle, src, len);
-
- if (nread <= 0) {
- switch (nread) {
- case 0:
- trace_alsa_read_zero(len);
- goto exit;
-
- case -EPIPE:
- if (alsa_recover (alsa->handle)) {
- alsa_logerr (nread, "Failed to read %ld frames\n", len);
- goto exit;
- }
- trace_alsa_xrun_in();
- continue;
-
- case -EAGAIN:
- goto exit;
-
- default:
- alsa_logerr (
- nread,
- "Failed to read %ld frames from %p\n",
- len,
- src
- );
- goto exit;
+ case -EPIPE:
+ if (alsa_recover(alsa->handle)) {
+ alsa_logerr(nread, "Failed to read %zu frames\n", len);
+ return pos;
}
+ trace_alsa_xrun_in();
+ continue;
+
+ case -EAGAIN:
+ return pos;
+
+ default:
+ alsa_logerr(nread, "Failed to read %zu frames to %p\n",
+ len, dst);
+ return pos;;
}
-
- hw->conv (dst, src, nread);
-
- src = advance (src, nread << hwshift);
- dst += nread;
-
- read_samples += nread;
- len -= nread;
}
+
+ pos += nread << hw->info.shift;
+ len -= nread << hw->info.shift;
}
- exit:
- hw->wpos = (hw->wpos + read_samples) % hw->samples;
- return read_samples;
+ return pos;
}
static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
@@ -1059,12 +915,12 @@ static void alsa_audio_fini (void *opaque)
static struct audio_pcm_ops alsa_pcm_ops = {
.init_out = alsa_init_out,
.fini_out = alsa_fini_out,
- .run_out = alsa_run_out,
+ .write = alsa_write,
.ctl_out = alsa_ctl_out,
.init_in = alsa_init_in,
.fini_in = alsa_fini_in,
- .run_in = alsa_run_in,
+ .read = alsa_read,
.ctl_in = alsa_ctl_in,
};
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 30/52] coreaudio: port to the new audio backend api
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (28 preceding siblings ...)
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 29/52] alsaaudio: port to the new audio backend api Kővágó, Zoltán
@ 2018-12-23 20:52 ` Kővágó, Zoltán
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 31/52] dsoundaudio: " Kővágó, Zoltán
` (20 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:52 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/coreaudio.c | 130 ++++++++++++++++++++++++----------------------
1 file changed, 69 insertions(+), 61 deletions(-)
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index abbbb45d69..a532e862dd 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -43,9 +43,6 @@ typedef struct coreaudioVoiceOut {
UInt32 audioDevicePropertyBufferFrameSize;
AudioStreamBasicDescription outputStreamBasicDescription;
AudioDeviceIOProcID ioprocid;
- int live;
- int decr;
- int rpos;
} coreaudioVoiceOut;
#if MAC_OS_X_VERSION_MAX_ALLOWED >= MAC_OS_X_VERSION_10_6
@@ -397,31 +394,29 @@ static int coreaudio_unlock (coreaudioVoiceOut *core, const char *fn_name)
return 0;
}
-static int coreaudio_run_out (HWVoiceOut *hw, int live)
-{
- int decr;
- coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
-
- if (coreaudio_lock (core, "coreaudio_run_out")) {
- return 0;
+#define COREAUDIO_WRAPPER_FUNC(name, ret_type, args_decl, args) \
+ static ret_type glue(coreaudio_, name)args_decl \
+ { \
+ coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw; \
+ ret_type ret; \
+ \
+ if (coreaudio_lock(core, "coreaudio_" #name)) { \
+ return 0; \
+ } \
+ \
+ ret = glue(audio_generic_, name)args; \
+ \
+ coreaudio_unlock(core, "coreaudio_" #name); \
+ return ret; \
}
-
- if (core->decr > live) {
- ldebug ("core->decr %d live %d core->live %d\n",
- core->decr,
- live,
- core->live);
- }
-
- decr = MIN (core->decr, live);
- core->decr -= decr;
-
- core->live = live - decr;
- hw->rpos = core->rpos;
-
- coreaudio_unlock (core, "coreaudio_run_out");
- return decr;
-}
+COREAUDIO_WRAPPER_FUNC(get_buffer_out, void *, (HWVocieOut *hw, size_t *size),
+ (hw, size))
+COREAUDIO_WRAPPER_FUNC(put_buffer_out_nowrite, size_t,
+ (HWVoiceOut *hw, void *buf, size_t size),
+ (hw, buf, size))
+COREAUDIO_WRAPPER_FUNC(write, size_t, (HWVoiceOut *hw, void *buf, size_t size),
+ (hw, buf, size))
+#undef COREAUDIO_WRAPPER_FUNC
/* callback to feed audiooutput buffer */
static OSStatus audioDeviceIOProc(
@@ -433,19 +428,11 @@ static OSStatus audioDeviceIOProc(
const AudioTimeStamp* inOutputTime,
void* hwptr)
{
- UInt32 frame, frameCount;
- float *out = outOutputData->mBuffers[0].mData;
+ UInt32 frameCount, pending_frames;
+ void *out = outOutputData->mBuffers[0].mData;
HWVoiceOut *hw = hwptr;
coreaudioVoiceOut *core = (coreaudioVoiceOut *) hwptr;
- int rpos, live;
- struct st_sample *src;
-#ifndef FLOAT_MIXENG
-#ifdef RECIPROCAL
- const float scale = 1.f / UINT_MAX;
-#else
- const float scale = UINT_MAX;
-#endif
-#endif
+ size_t len;
if (coreaudio_lock (core, "audioDeviceIOProc")) {
inInputTime = 0;
@@ -453,42 +440,51 @@ static OSStatus audioDeviceIOProc(
}
frameCount = core->audioDevicePropertyBufferFrameSize;
- live = core->live;
+ pending_frames = hw->pending_emul >> hw->info.shift;
/* if there are not enough samples, set signal and return */
- if (live < frameCount) {
+ if (pending_frames < frameCount) {
inInputTime = 0;
coreaudio_unlock (core, "audioDeviceIOProc(empty)");
return 0;
}
- rpos = core->rpos;
- src = hw->mix_buf + rpos;
+ len = frameCount << hw->info.shift;
+ while (len) {
+ size_t write_len;
+ ssize_t start = ((ssize_t) hw->pos_emul) - hw->pending_emul;
+ if (start < 0) {
+ start += hw->size_emul;
+ }
+ assert(start >= 0 && start < hw->size_emul);
- /* fill buffer */
- for (frame = 0; frame < frameCount; frame++) {
-#ifdef FLOAT_MIXENG
- *out++ = src[frame].l; /* left channel */
- *out++ = src[frame].r; /* right channel */
-#else
-#ifdef RECIPROCAL
- *out++ = src[frame].l * scale; /* left channel */
- *out++ = src[frame].r * scale; /* right channel */
-#else
- *out++ = src[frame].l / scale; /* left channel */
- *out++ = src[frame].r / scale; /* right channel */
-#endif
-#endif
+ write_len = MIN(MIN(hw->pending_emul, len),
+ hw->size_emul - start);
+
+ memcpy(out, hw->buf_emul + start, write_len);
+ hw->pending_emul -= write_len;
+ len -= write_len;
+ out += write_len;
}
- rpos = (rpos + frameCount) % hw->samples;
- core->decr += frameCount;
- core->rpos = rpos;
-
coreaudio_unlock (core, "audioDeviceIOProc");
return 0;
}
+static UInt32 coreaudio_get_flags(struct audio_pcm_info *info,
+ struct audsettings *as)
+{
+ UInt32 flags = info->sign ? kAudioFormatFlagIsSignedInteger : 0;
+ if (as->endianness) { /* 0 = little, 1 = big */
+ flags |= kAudioFormatFlagIsBigEndian;
+ }
+
+ if (flags == 0) { /* must not be 0 */
+ flags = kAudioFormatFlagsAreAllClear;
+ }
+ return flags;
+}
+
static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
void *drv_opaque)
{
@@ -576,6 +572,16 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
/* set Samplerate */
core->outputStreamBasicDescription.mSampleRate = (Float64) as->freq;
+ core->outputStreamBasicDescription.mFormatID = kAudioFormatLinearPCM;
+ core->outputStreamBasicDescription.mFormatFlags =
+ coreaudio_get_flags(hw->info, as);
+ core->outputStreamBasicDescription.mBytesPerPacket =
+ core->outputStreamBasicDescription.mBytesPerFrame =
+ hw->info.nchannels * hw->info.bits / 8;
+ core->outputStreamBasicDescription.mFramesPerPacket = 1;
+ core->outputStreamBasicDescription.mChannelsPerFrame = hw->info.nchannels;
+ core->outputStreamBasicDescription.mBitsPerChannel = hw->info.bits;
+
status = coreaudio_set_streamformat(core->outputDeviceID,
&core->outputStreamBasicDescription);
if (status != kAudioHardwareNoError) {
@@ -686,7 +692,9 @@ static void coreaudio_audio_fini (void *opaque)
static struct audio_pcm_ops coreaudio_pcm_ops = {
.init_out = coreaudio_init_out,
.fini_out = coreaudio_fini_out,
- .run_out = coreaudio_run_out,
+ .write = coreaudio_write,
+ .get_buffer_out = coreaudio_get_buffer_out,
+ .put_buffer_out = coreaudio_put_buffer_out_nowrite,
.ctl_out = coreaudio_ctl_out
};
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 31/52] dsoundaudio: port to the new audio backend api
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (29 preceding siblings ...)
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 30/52] coreaudio: " Kővágó, Zoltán
@ 2018-12-23 20:52 ` Kővágó, Zoltán
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 32/52] noaudio: " Kővágó, Zoltán
` (19 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:52 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
---
audio/dsound_template.h | 45 +++---
audio/dsoundaudio.c | 328 ++++++++++------------------------------
2 files changed, 102 insertions(+), 271 deletions(-)
diff --git a/audio/dsound_template.h b/audio/dsound_template.h
index 96181efb36..ff5a1f85fd 100644
--- a/audio/dsound_template.h
+++ b/audio/dsound_template.h
@@ -29,6 +29,8 @@
#define BUFPTR LPDIRECTSOUNDCAPTUREBUFFER
#define FIELD dsound_capture_buffer
#define FIELD2 dsound_capture
+#define HWVOICE HWVoiceIn
+#define DSOUNDVOICE DSoundVoiceIn
#else
#define NAME "playback buffer"
#define NAME2 "DirectSound"
@@ -37,6 +39,8 @@
#define BUFPTR LPDIRECTSOUNDBUFFER
#define FIELD dsound_buffer
#define FIELD2 dsound
+#define HWVOICE HWVoiceOut
+#define DSOUNDVOICE DSoundVoiceOut
#endif
static int glue (dsound_unlock_, TYPE) (
@@ -72,8 +76,6 @@ static int glue (dsound_lock_, TYPE) (
)
{
HRESULT hr;
- LPVOID p1 = NULL, p2 = NULL;
- DWORD blen1 = 0, blen2 = 0;
DWORD flag;
#ifdef DSBTYPE_IN
@@ -81,7 +83,7 @@ static int glue (dsound_lock_, TYPE) (
#else
flag = entire ? DSBLOCK_ENTIREBUFFER : 0;
#endif
- hr = glue(IFACE, _Lock)(buf, pos, len, &p1, &blen1, &p2, &blen2, flag);
+ hr = glue(IFACE, _Lock)(buf, pos, len, p1p, blen1p, p2p, blen2p, flag);
if (FAILED (hr)) {
#ifndef DSBTYPE_IN
@@ -96,34 +98,34 @@ static int glue (dsound_lock_, TYPE) (
goto fail;
}
- if ((p1 && (blen1 & info->align)) || (p2 && (blen2 & info->align))) {
+ if ((p1p && *p1p && (*blen1p & info->align)) ||
+ (p2p && *p2p && (*blen2p & info->align))) {
dolog ("DirectSound returned misaligned buffer %ld %ld\n",
- blen1, blen2);
- glue (dsound_unlock_, TYPE) (buf, p1, p2, blen1, blen2);
+ *blen1p, *blen2p);
+ glue (dsound_unlock_, TYPE) (buf, *p1p, p2p ? *p2p : NULL, *blen1p,
+ blen2p ? *blen2p : 0);
goto fail;
}
- if (!p1 && blen1) {
- dolog ("warning: !p1 && blen1=%ld\n", blen1);
- blen1 = 0;
+ if (p1p && !*p1p && *blen1p) {
+ dolog("warning: !p1 && blen1=%ld\n", *blen1p);
+ *blen1p = 0;
}
- if (!p2 && blen2) {
- dolog ("warning: !p2 && blen2=%ld\n", blen2);
- blen2 = 0;
+ if (p2p && !*p2p && *blen2p) {
+ dolog("warning: !p2 && blen2=%ld\n", *blen2p);
+ *blen2p = 0;
}
- *p1p = p1;
- *p2p = p2;
- *blen1p = blen1;
- *blen2p = blen2;
return 0;
fail:
*p1p = NULL - 1;
- *p2p = NULL - 1;
*blen1p = -1;
- *blen2p = -1;
+ if (p2p) {
+ *p2p = NULL - 1;
+ *blen2p = -1;
+ }
return -1;
}
@@ -242,7 +244,6 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
goto fail0;
}
- ds->first_time = 1;
obt_as.endianness = 0;
audio_pcm_init_info (&hw->info, &obt_as);
@@ -252,15 +253,13 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
bc.dwBufferBytes, hw->info.align + 1
);
}
+ hw->size_emul = bc.dwBufferBytes;
hw->samples = bc.dwBufferBytes >> hw->info.shift;
ds->s = s;
#ifdef DEBUG_DSOUND
dolog ("caps %ld, desc %ld\n",
bc.dwBufferBytes, bd.dwBufferBytes);
-
- dolog ("bufsize %d, freq %d, chan %d, fmt %d\n",
- hw->bufsize, settings.freq, settings.nchannels, settings.fmt);
#endif
return 0;
@@ -276,3 +275,5 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
#undef BUFPTR
#undef FIELD
#undef FIELD2
+#undef HWVOICE
+#undef DSOUNDVOICE
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index dfcb70c81a..7b3266aaf3 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -53,19 +53,11 @@ typedef struct {
typedef struct {
HWVoiceOut hw;
LPDIRECTSOUNDBUFFER dsound_buffer;
- DWORD old_pos;
- int first_time;
dsound *s;
-#ifdef DEBUG_DSOUND
- DWORD old_ppos;
- DWORD played;
- DWORD mixed;
-#endif
} DSoundVoiceOut;
typedef struct {
HWVoiceIn hw;
- int first_time;
LPDIRECTSOUNDCAPTUREBUFFER dsound_capture_buffer;
dsound *s;
} DSoundVoiceIn;
@@ -243,11 +235,6 @@ static void GCC_FMT_ATTR (3, 4) dsound_logerr2 (
dsound_log_hresult (hr);
}
-static uint64_t usecs_to_bytes(struct audio_pcm_info *info, uint32_t usecs)
-{
- return muldiv64(usecs, info->bytes_per_second, 1000000);
-}
-
#ifdef DEBUG_DSOUND
static void print_wave_format (WAVEFORMATEX *wfx)
{
@@ -312,33 +299,6 @@ static int dsound_get_status_in (LPDIRECTSOUNDCAPTUREBUFFER dscb,
return 0;
}
-static void dsound_write_sample (HWVoiceOut *hw, uint8_t *dst, int dst_len)
-{
- int src_len1 = dst_len;
- int src_len2 = 0;
- int pos = hw->rpos + dst_len;
- struct st_sample *src1 = hw->mix_buf + hw->rpos;
- struct st_sample *src2 = NULL;
-
- if (pos > hw->samples) {
- src_len1 = hw->samples - hw->rpos;
- src2 = hw->mix_buf;
- src_len2 = dst_len - src_len1;
- pos = src_len2;
- }
-
- if (src_len1) {
- hw->clip (dst, src1, src_len1);
- }
-
- if (src_len2) {
- dst = advance (dst, src_len1 << hw->info.shift);
- hw->clip (dst, src2, src_len2);
- }
-
- hw->rpos = pos % hw->samples;
-}
-
static void dsound_clear_sample (HWVoiceOut *hw, LPDIRECTSOUNDBUFFER dsb,
dsound *s)
{
@@ -350,7 +310,7 @@ static void dsound_clear_sample (HWVoiceOut *hw, LPDIRECTSOUNDBUFFER dsb,
dsb,
&hw->info,
0,
- hw->samples << hw->info.shift,
+ hw->size_emul,
&p1, &p2,
&blen1, &blen2,
1,
@@ -454,138 +414,51 @@ static int dsound_ctl_out (HWVoiceOut *hw, int cmd, ...)
return 0;
}
-static int dsound_run_out (HWVoiceOut *hw, int live)
+static void *dsound_get_buffer_out(HWVoiceOut *hw, size_t *size)
{
- int err;
+ DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
+ LPDIRECTSOUNDBUFFER dsb = ds->dsound_buffer;
HRESULT hr;
+ DWORD ppos, act_size;
+ size_t req_size;
+ int err;
+ void *ret;
+
+ hr = IDirectSoundBuffer_GetCurrentPosition(dsb, &ppos, NULL);
+ if (FAILED(hr)) {
+ dsound_logerr(hr, "Could not get playback buffer position\n");
+ *size = 0;
+ return NULL;
+ }
+
+ req_size = audio_ring_dist(ppos, hw->pos_emul, hw->size_emul);
+ req_size = MIN(req_size, hw->size_emul - hw->pos_emul);
+
+ err = dsound_lock_out(dsb, &hw->info, hw->pos_emul, req_size, &ret, NULL,
+ &act_size, NULL, false, ds->s);
+ if (err) {
+ dolog("Failed to lock buffer\n");
+ *size = 0;
+ return NULL;
+ }
+
+ *size = act_size;
+ return ret;
+}
+
+static size_t dsound_put_buffer_out(HWVoiceOut *hw, void *buf, size_t len)
+{
DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
LPDIRECTSOUNDBUFFER dsb = ds->dsound_buffer;
- int len, hwshift;
- DWORD blen1, blen2;
- DWORD len1, len2;
- DWORD decr;
- DWORD wpos, ppos, old_pos;
- LPVOID p1, p2;
- int bufsize;
- dsound *s = ds->s;
- AudiodevDsoundOptions *dso = &s->dev->u.dsound;
+ int err = dsound_unlock_out(dsb, buf, NULL, len, 0);
- if (!dsb) {
- dolog ("Attempt to run empty with playback buffer\n");
- return 0;
- }
-
- hwshift = hw->info.shift;
- bufsize = hw->samples << hwshift;
-
- hr = IDirectSoundBuffer_GetCurrentPosition (
- dsb,
- &ppos,
- ds->first_time ? &wpos : NULL
- );
- if (FAILED (hr)) {
- dsound_logerr (hr, "Could not get playback buffer position\n");
- return 0;
- }
-
- len = live << hwshift;
-
- if (ds->first_time) {
- if (dso->latency) {
- DWORD cur_blat;
-
- cur_blat = audio_ring_dist (wpos, ppos, bufsize);
- ds->first_time = 0;
- old_pos = wpos;
- old_pos +=
- usecs_to_bytes(&hw->info, dso->latency) - cur_blat;
- old_pos %= bufsize;
- old_pos &= ~hw->info.align;
- }
- else {
- old_pos = wpos;
- }
-#ifdef DEBUG_DSOUND
- ds->played = 0;
- ds->mixed = 0;
-#endif
- }
- else {
- if (ds->old_pos == ppos) {
-#ifdef DEBUG_DSOUND
- dolog ("old_pos == ppos\n");
-#endif
- return 0;
- }
-
-#ifdef DEBUG_DSOUND
- ds->played += audio_ring_dist (ds->old_pos, ppos, hw->bufsize);
-#endif
- old_pos = ds->old_pos;
- }
-
- if ((old_pos < ppos) && ((old_pos + len) > ppos)) {
- len = ppos - old_pos;
- }
- else {
- if ((old_pos > ppos) && ((old_pos + len) > (ppos + bufsize))) {
- len = bufsize - old_pos + ppos;
- }
- }
-
- if (audio_bug(__func__, len < 0 || len > bufsize)) {
- dolog ("len=%d bufsize=%d old_pos=%ld ppos=%ld\n",
- len, bufsize, old_pos, ppos);
- return 0;
- }
-
- len &= ~hw->info.align;
- if (!len) {
- return 0;
- }
-
-#ifdef DEBUG_DSOUND
- ds->old_ppos = ppos;
-#endif
- err = dsound_lock_out (
- dsb,
- &hw->info,
- old_pos,
- len,
- &p1, &p2,
- &blen1, &blen2,
- 0,
- s
- );
if (err) {
+ dolog("Failed to unlock buffer!!\n");
return 0;
}
+ hw->pos_emul = (hw->pos_emul + len) % hw->size_emul;
- len1 = blen1 >> hwshift;
- len2 = blen2 >> hwshift;
- decr = len1 + len2;
-
- if (p1 && len1) {
- dsound_write_sample (hw, p1, len1);
- }
-
- if (p2 && len2) {
- dsound_write_sample (hw, p2, len2);
- }
-
- dsound_unlock_out (dsb, p1, p2, blen1, blen2);
- ds->old_pos = (old_pos + (decr << hwshift)) % bufsize;
-
-#ifdef DEBUG_DSOUND
- ds->mixed += decr << hwshift;
-
- dolog ("played %lu mixed %lu diff %ld sec %f\n",
- ds->played,
- ds->mixed,
- ds->mixed - ds->played,
- abs (ds->mixed - ds->played) / (double) hw->info.bytes_per_second);
-#endif
- return decr;
+ return len;
}
static int dsound_ctl_in (HWVoiceIn *hw, int cmd, ...)
@@ -640,96 +513,49 @@ static int dsound_ctl_in (HWVoiceIn *hw, int cmd, ...)
return 0;
}
-static int dsound_run_in (HWVoiceIn *hw)
+static void *dsound_get_buffer_in(HWVoiceIn *hw, size_t *size)
{
- int err;
+ DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;
+ LPDIRECTSOUNDCAPTUREBUFFER dscb = ds->dsound_capture_buffer;
HRESULT hr;
+ DWORD cpos, act_size;
+ size_t req_size;
+ int err;
+ void *ret;
+
+ hr = IDirectSoundCaptureBuffer_GetCurrentPosition(dscb, &cpos, NULL);
+ if (FAILED(hr)) {
+ dsound_logerr(hr, "Could not get capture buffer position\n");
+ *size = 0;
+ return NULL;
+ }
+
+ req_size = audio_ring_dist(cpos, hw->pos_emul, hw->size_emul);
+ req_size = MIN(req_size, hw->size_emul - hw->pos_emul);
+
+ err = dsound_lock_in(dscb, &hw->info, hw->pos_emul, req_size, &ret, NULL,
+ &act_size, NULL, false, ds->s);
+ if (err) {
+ dolog("Failed to lock buffer\n");
+ *size = 0;
+ return NULL;
+ }
+
+ *size = act_size;
+ return ret;
+}
+
+static void dsound_put_buffer_in(HWVoiceIn *hw, void *buf, size_t len)
+{
DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;
LPDIRECTSOUNDCAPTUREBUFFER dscb = ds->dsound_capture_buffer;
- int live, len, dead;
- DWORD blen1, blen2;
- DWORD len1, len2;
- DWORD decr;
- DWORD cpos, rpos;
- LPVOID p1, p2;
- int hwshift;
- dsound *s = ds->s;
+ int err = dsound_unlock_in(dscb, buf, NULL, len, 0);
- if (!dscb) {
- dolog ("Attempt to run without capture buffer\n");
- return 0;
- }
-
- hwshift = hw->info.shift;
-
- live = audio_pcm_hw_get_live_in (hw);
- dead = hw->samples - live;
- if (!dead) {
- return 0;
- }
-
- hr = IDirectSoundCaptureBuffer_GetCurrentPosition (
- dscb,
- &cpos,
- ds->first_time ? &rpos : NULL
- );
- if (FAILED (hr)) {
- dsound_logerr (hr, "Could not get capture buffer position\n");
- return 0;
- }
-
- if (ds->first_time) {
- ds->first_time = 0;
- if (rpos & hw->info.align) {
- ldebug ("warning: Misaligned capture read position %ld(%d)\n",
- rpos, hw->info.align);
- }
- hw->wpos = rpos >> hwshift;
- }
-
- if (cpos & hw->info.align) {
- ldebug ("warning: Misaligned capture position %ld(%d)\n",
- cpos, hw->info.align);
- }
- cpos >>= hwshift;
-
- len = audio_ring_dist (cpos, hw->wpos, hw->samples);
- if (!len) {
- return 0;
- }
- len = MIN (len, dead);
-
- err = dsound_lock_in (
- dscb,
- &hw->info,
- hw->wpos << hwshift,
- len << hwshift,
- &p1,
- &p2,
- &blen1,
- &blen2,
- 0,
- s
- );
if (err) {
- return 0;
+ dolog("Failed to unlock buffer!!\n");
+ return;
}
-
- len1 = blen1 >> hwshift;
- len2 = blen2 >> hwshift;
- decr = len1 + len2;
-
- if (p1 && len1) {
- hw->conv (hw->conv_buf + hw->wpos, p1, len1);
- }
-
- if (p2 && len2) {
- hw->conv (hw->conv_buf, p2, len2);
- }
-
- dsound_unlock_in (dscb, p1, p2, blen1, blen2);
- hw->wpos = (hw->wpos + decr) % hw->samples;
- return decr;
+ hw->pos_emul = (hw->pos_emul + len) % hw->size_emul;
}
static void dsound_audio_fini (void *opaque)
@@ -845,12 +671,16 @@ static void *dsound_audio_init(Audiodev *dev)
static struct audio_pcm_ops dsound_pcm_ops = {
.init_out = dsound_init_out,
.fini_out = dsound_fini_out,
- .run_out = dsound_run_out,
+ .write = audio_generic_write,
+ .get_buffer_out = dsound_get_buffer_out,
+ .put_buffer_out = dsound_put_buffer_out,
.ctl_out = dsound_ctl_out,
.init_in = dsound_init_in,
.fini_in = dsound_fini_in,
- .run_in = dsound_run_in,
+ .read = audio_generic_read,
+ .get_buffer_in = dsound_get_buffer_in,
+ .put_buffer_in = dsound_put_buffer_in,
.ctl_in = dsound_ctl_in
};
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 32/52] noaudio: port to the new audio backend api
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (30 preceding siblings ...)
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 31/52] dsoundaudio: " Kővágó, Zoltán
@ 2018-12-23 20:52 ` Kővágó, Zoltán
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 33/52] ossaudio: " Kővágó, Zoltán
` (18 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:52 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/noaudio.c | 39 +++++++++++++++------------------------
1 file changed, 15 insertions(+), 24 deletions(-)
diff --git a/audio/noaudio.c b/audio/noaudio.c
index c49f2df4b3..6a3a1c418b 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -40,10 +40,9 @@ typedef struct NoVoiceIn {
int64_t old_ticks;
} NoVoiceIn;
-static int no_run_out (HWVoiceOut *hw, int live)
+static size_t no_write(HWVoiceOut *hw, void *buf, size_t len)
{
NoVoiceOut *no = (NoVoiceOut *) hw;
- int decr, samples;
int64_t now;
int64_t ticks;
int64_t bytes;
@@ -51,13 +50,9 @@ static int no_run_out (HWVoiceOut *hw, int live)
now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
ticks = now - no->old_ticks;
bytes = muldiv64(ticks, hw->info.bytes_per_second, NANOSECONDS_PER_SECOND);
- bytes = MIN(bytes, INT_MAX);
- samples = bytes >> hw->info.shift;
no->old_ticks = now;
- decr = MIN (live, samples);
- hw->rpos = (hw->rpos + decr) % hw->samples;
- return decr;
+ return MIN(len, bytes);
}
static int no_init_out(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque)
@@ -91,25 +86,21 @@ static void no_fini_in (HWVoiceIn *hw)
(void) hw;
}
-static int no_run_in (HWVoiceIn *hw)
+static size_t no_read(HWVoiceIn *hw, void *buf, size_t size)
{
+ size_t to_clear;
NoVoiceIn *no = (NoVoiceIn *) hw;
- int live = audio_pcm_hw_get_live_in (hw);
- int dead = hw->samples - live;
- int samples = 0;
- if (dead) {
- int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
- int64_t ticks = now - no->old_ticks;
- int64_t bytes =
- muldiv64(ticks, hw->info.bytes_per_second, NANOSECONDS_PER_SECOND);
+ int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
+ int64_t ticks = now - no->old_ticks;
+ int64_t bytes =
+ muldiv64 (ticks, hw->info.bytes_per_second, NANOSECONDS_PER_SECOND);
- no->old_ticks = now;
- bytes = MIN (bytes, INT_MAX);
- samples = bytes >> hw->info.shift;
- samples = MIN (samples, dead);
- }
- return samples;
+ no->old_ticks = now;
+ to_clear = MIN(bytes, size);
+
+ audio_pcm_info_clear_buf(&hw->info, buf, to_clear >> hw->info.shift);
+ return to_clear;
}
static int no_ctl_in (HWVoiceIn *hw, int cmd, ...)
@@ -132,12 +123,12 @@ static void no_audio_fini (void *opaque)
static struct audio_pcm_ops no_pcm_ops = {
.init_out = no_init_out,
.fini_out = no_fini_out,
- .run_out = no_run_out,
+ .write = no_write,
.ctl_out = no_ctl_out,
.init_in = no_init_in,
.fini_in = no_fini_in,
- .run_in = no_run_in,
+ .read = no_read,
.ctl_in = no_ctl_in
};
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 33/52] ossaudio: port to the new audio backend api
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (31 preceding siblings ...)
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 32/52] noaudio: " Kővágó, Zoltán
@ 2018-12-23 20:52 ` Kővágó, Zoltán
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 34/52] paaudio: " Kővágó, Zoltán
` (17 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:52 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/ossaudio.c | 287 +++++++++++++++++------------------------------
1 file changed, 104 insertions(+), 183 deletions(-)
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 05c7d6f85c..f1b57c87b4 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -40,19 +40,15 @@
typedef struct OSSVoiceOut {
HWVoiceOut hw;
- void *pcm_buf;
int fd;
- int wpos;
int nfrags;
int fragsize;
int mmapped;
- int pending;
Audiodev *dev;
} OSSVoiceOut;
typedef struct OSSVoiceIn {
HWVoiceIn hw;
- void *pcm_buf;
int fd;
int nfrags;
int fragsize;
@@ -371,97 +367,87 @@ static int oss_open(int in, struct oss_params *req, audsettings *as,
return -1;
}
-static void oss_write_pending (OSSVoiceOut *oss)
+static size_t oss_get_available_bytes(OSSVoiceOut *oss)
{
- HWVoiceOut *hw = &oss->hw;
+ int err;
+ struct count_info cntinfo;
+ assert(oss->mmapped);
+ err = ioctl (oss->fd, SNDCTL_DSP_GETOPTR, &cntinfo);
+ if (err < 0) {
+ oss_logerr(errno, "SNDCTL_DSP_GETOPTR failed\n");
+ return 0;
+ }
+
+ return audio_ring_dist(cntinfo.ptr, oss->hw.pos_emul, oss->hw.size_emul);
+}
+
+static void *oss_get_buffer_out(HWVoiceOut *hw, size_t *size)
+{
+ OSSVoiceOut *oss = (OSSVoiceOut *) hw;
+ if (oss->mmapped) {
+ *size = MIN(oss_get_available_bytes(oss), hw->size_emul - hw->pos_emul);
+ return hw->buf_emul + hw->pos_emul;
+ } else {
+ return audio_generic_get_buffer_out(hw, size);
+ }
+}
+
+static size_t oss_put_buffer_out(HWVoiceOut *hw, void *buf, size_t size)
+{
+ OSSVoiceOut *oss = (OSSVoiceOut *) hw;
if (oss->mmapped) {
- return;
+ assert(buf == hw->buf_emul + hw->pos_emul && size < hw->size_emul);
+
+ hw->pos_emul = (hw->pos_emul + size) % hw->size_emul;
+ return size;
+ } else {
+ return audio_generic_put_buffer_out(hw, buf, size);
+ }
+}
+
+static size_t oss_write(HWVoiceOut *hw, void *buf, size_t len)
+{
+ OSSVoiceOut *oss = (OSSVoiceOut *) hw;
+ size_t pos;
+
+ if (oss->mmapped) {
+ size_t total_len;
+ len = MIN(len, oss_get_available_bytes(oss));
+
+ total_len = len;
+ while (len) {
+ size_t to_copy = MIN(len, hw->size_emul - hw->pos_emul);
+ memcpy(hw->buf_emul + hw->pos_emul, buf, to_copy);
+
+ hw->pos_emul = (hw->pos_emul + to_copy) % hw->pos_emul;
+ buf += to_copy;
+ len -= to_copy;
+ }
+ return total_len;
}
- while (oss->pending) {
- int samples_written;
+ pos = 0;
+ while (len) {
ssize_t bytes_written;
- int samples_till_end = hw->samples - oss->wpos;
- int samples_to_write = MIN (oss->pending, samples_till_end);
- int bytes_to_write = samples_to_write << hw->info.shift;
- void *pcm = advance (oss->pcm_buf, oss->wpos << hw->info.shift);
+ void *pcm = advance(buf, pos);
- bytes_written = write (oss->fd, pcm, bytes_to_write);
+ bytes_written = write(oss->fd, pcm, len);
if (bytes_written < 0) {
if (errno != EAGAIN) {
- oss_logerr (errno, "failed to write %d bytes\n",
- bytes_to_write);
+ oss_logerr(errno, "failed to write %zu bytes\n",
+ len);
}
- break;
- }
-
- if (bytes_written & hw->info.align) {
- dolog ("misaligned write asked for %d, but got %zd\n",
- bytes_to_write, bytes_written);
- return;
+ return pos;
}
- samples_written = bytes_written >> hw->info.shift;
- oss->pending -= samples_written;
- oss->wpos = (oss->wpos + samples_written) % hw->samples;
- if (bytes_written - bytes_to_write) {
+ pos += bytes_written;
+ if (bytes_written < len) {
break;
}
+ len -= bytes_written;
}
-}
-
-static int oss_run_out (HWVoiceOut *hw, int live)
-{
- OSSVoiceOut *oss = (OSSVoiceOut *) hw;
- int err, decr;
- struct audio_buf_info abinfo;
- struct count_info cntinfo;
- int bufsize;
-
- bufsize = hw->samples << hw->info.shift;
-
- if (oss->mmapped) {
- int bytes, pos;
-
- err = ioctl (oss->fd, SNDCTL_DSP_GETOPTR, &cntinfo);
- if (err < 0) {
- oss_logerr (errno, "SNDCTL_DSP_GETOPTR failed\n");
- return 0;
- }
-
- pos = hw->rpos << hw->info.shift;
- bytes = audio_ring_dist (cntinfo.ptr, pos, bufsize);
- decr = MIN (bytes >> hw->info.shift, live);
- }
- else {
- err = ioctl (oss->fd, SNDCTL_DSP_GETOSPACE, &abinfo);
- if (err < 0) {
- oss_logerr (errno, "SNDCTL_DSP_GETOPTR failed\n");
- return 0;
- }
-
- if (abinfo.bytes > bufsize) {
- trace_oss_invalid_available_size(abinfo.bytes, bufsize);
- abinfo.bytes = bufsize;
- }
-
- if (abinfo.bytes < 0) {
- trace_oss_invalid_available_size(abinfo.bytes, bufsize);
- return 0;
- }
-
- decr = MIN (abinfo.bytes >> hw->info.shift, live);
- if (!decr) {
- return 0;
- }
- }
-
- decr = audio_pcm_hw_clip_out (hw, oss->pcm_buf, decr, oss->pending);
- oss->pending += decr;
- oss_write_pending (oss);
-
- return decr;
+ return pos;
}
static void oss_fini_out (HWVoiceOut *hw)
@@ -472,18 +458,13 @@ static void oss_fini_out (HWVoiceOut *hw)
ldebug ("oss_fini\n");
oss_anal_close (&oss->fd);
- if (oss->pcm_buf) {
- if (oss->mmapped) {
- err = munmap (oss->pcm_buf, hw->samples << hw->info.shift);
- if (err) {
- oss_logerr(errno, "Failed to unmap buffer %p, size %zu\n",
- oss->pcm_buf, hw->samples << hw->info.shift);
- }
+ if (oss->mmapped && hw->buf_emul) {
+ err = munmap(hw->buf_emul, hw->size_emul);
+ if (err) {
+ oss_logerr(errno, "Failed to unmap buffer %p, size %zu\n",
+ hw->buf_emul, hw->size_emul);
}
- else {
- g_free (oss->pcm_buf);
- }
- oss->pcm_buf = NULL;
+ hw->buf_emul = NULL;
}
}
@@ -534,19 +515,20 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
oss->mmapped = 0;
if (oopts->has_try_mmap && oopts->try_mmap) {
- oss->pcm_buf = mmap (
+ hw->size_emul = hw->samples << hw->info.shift;
+ hw->buf_emul = mmap (
NULL,
- hw->samples << hw->info.shift,
+ hw->size_emul,
PROT_READ | PROT_WRITE,
MAP_SHARED,
fd,
0
);
- if (oss->pcm_buf == MAP_FAILED) {
+ if (hw->buf_emul == MAP_FAILED) {
oss_logerr(errno, "Failed to map %zu bytes of DAC\n",
- hw->samples << hw->info.shift);
- }
- else {
+ hw->size_emul);
+ hw->buf_emul = NULL;
+ } else {
int err;
int trig = 0;
if (ioctl (fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
@@ -566,30 +548,16 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
}
if (!oss->mmapped) {
- err = munmap (oss->pcm_buf, hw->samples << hw->info.shift);
+ err = munmap(hw->buf_emul, hw->size_emul);
if (err) {
oss_logerr(errno, "Failed to unmap buffer %p size %zu\n",
- oss->pcm_buf, hw->samples << hw->info.shift);
+ hw->buf_emul, hw->size_emul);
}
+ hw->buf_emul = NULL;
}
}
}
- if (!oss->mmapped) {
- oss->pcm_buf = audio_calloc(__func__,
- hw->samples,
- 1 << hw->info.shift);
- if (!oss->pcm_buf) {
- dolog (
- "Could not allocate DAC buffer (%zu samples, each %d bytes)\n",
- hw->samples,
- 1 << hw->info.shift
- );
- oss_anal_close (&fd);
- return -1;
- }
- }
-
oss->fd = fd;
oss->dev = dev;
return 0;
@@ -617,7 +585,7 @@ static int oss_ctl_out (HWVoiceOut *hw, int cmd, ...)
return 0;
}
- audio_pcm_info_clear_buf (&hw->info, oss->pcm_buf, hw->samples);
+ audio_pcm_info_clear_buf(&hw->info, hw->buf_emul, hw->samples);
trig = PCM_ENABLE_OUTPUT;
if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
oss_logerr (
@@ -691,13 +659,6 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
}
hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
- oss->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
- if (!oss->pcm_buf) {
- dolog("Could not allocate ADC buffer (%zu samples, each %d bytes)\n",
- hw->samples, 1 << hw->info.shift);
- oss_anal_close (&fd);
- return -1;
- }
oss->fd = fd;
oss->dev = dev;
@@ -709,78 +670,36 @@ static void oss_fini_in (HWVoiceIn *hw)
OSSVoiceIn *oss = (OSSVoiceIn *) hw;
oss_anal_close (&oss->fd);
-
- g_free(oss->pcm_buf);
- oss->pcm_buf = NULL;
}
-static int oss_run_in (HWVoiceIn *hw)
+static size_t oss_read(HWVoiceIn *hw, void *buf, size_t len)
{
OSSVoiceIn *oss = (OSSVoiceIn *) hw;
- int hwshift = hw->info.shift;
- int i;
- int live = audio_pcm_hw_get_live_in (hw);
- int dead = hw->samples - live;
- size_t read_samples = 0;
- struct {
- int add;
- int len;
- } bufs[2] = {
- { .add = hw->wpos, .len = 0 },
- { .add = 0, .len = 0 }
- };
+ size_t pos = 0;
- if (!dead) {
- return 0;
- }
-
- if (hw->wpos + dead > hw->samples) {
- bufs[0].len = (hw->samples - hw->wpos) << hwshift;
- bufs[1].len = (dead - (hw->samples - hw->wpos)) << hwshift;
- }
- else {
- bufs[0].len = dead << hwshift;
- }
-
- for (i = 0; i < 2; ++i) {
+ while (len) {
ssize_t nread;
- if (bufs[i].len) {
- void *p = advance (oss->pcm_buf, bufs[i].add << hwshift);
- nread = read (oss->fd, p, bufs[i].len);
+ void *dst = advance(buf, pos);
+ nread = read(oss->fd, dst, len);
- if (nread > 0) {
- if (nread & hw->info.align) {
- dolog ("warning: Misaligned read %zd (requested %d), "
- "alignment %d\n", nread, bufs[i].add << hwshift,
- hw->info.align + 1);
- }
- read_samples += nread >> hwshift;
- hw->conv (hw->conv_buf + bufs[i].add, p, nread >> hwshift);
- }
-
- if (bufs[i].len - nread) {
- if (nread == -1) {
- switch (errno) {
- case EINTR:
- case EAGAIN:
- break;
- default:
- oss_logerr (
- errno,
- "Failed to read %d bytes of audio (to %p)\n",
- bufs[i].len, p
- );
- break;
- }
- }
+ if (nread == -1) {
+ switch (errno) {
+ case EINTR:
+ case EAGAIN:
+ break;
+ default:
+ oss_logerr(errno, "Failed to read %zu bytes of audio (to %p)\n",
+ len, dst);
break;
}
}
+
+ pos += nread;
+ len -= nread;
}
- hw->wpos = (hw->wpos + read_samples) % hw->samples;
- return read_samples;
+ return pos;
}
static int oss_ctl_in (HWVoiceIn *hw, int cmd, ...)
@@ -833,12 +752,14 @@ static void oss_audio_fini (void *opaque)
static struct audio_pcm_ops oss_pcm_ops = {
.init_out = oss_init_out,
.fini_out = oss_fini_out,
- .run_out = oss_run_out,
+ .write = oss_write,
+ .get_buffer_out = oss_get_buffer_out,
+ .put_buffer_out = oss_put_buffer_out,
.ctl_out = oss_ctl_out,
.init_in = oss_init_in,
.fini_in = oss_fini_in,
- .run_in = oss_run_in,
+ .read = oss_read,
.ctl_in = oss_ctl_in
};
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 34/52] paaudio: port to the new audio backend api
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (32 preceding siblings ...)
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 33/52] ossaudio: " Kővágó, Zoltán
@ 2018-12-23 20:52 ` Kővágó, Zoltán
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 35/52] sdlaudio: " Kővágó, Zoltán
` (16 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:52 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/Makefile.objs | 1 -
audio/audio_pt_int.c | 174 --------------------
audio/audio_pt_int.h | 22 ---
audio/paaudio.c | 372 ++++++-------------------------------------
configure | 5 -
5 files changed, 45 insertions(+), 529 deletions(-)
delete mode 100644 audio/audio_pt_int.c
delete mode 100644 audio/audio_pt_int.h
diff --git a/audio/Makefile.objs b/audio/Makefile.objs
index dca87f6347..d7490a379f 100644
--- a/audio/Makefile.objs
+++ b/audio/Makefile.objs
@@ -2,7 +2,6 @@ common-obj-y = audio.o audio_legacy.o noaudio.o wavaudio.o mixeng.o
common-obj-$(CONFIG_SPICE) += spiceaudio.o
common-obj-$(CONFIG_AUDIO_COREAUDIO) += coreaudio.o
common-obj-$(CONFIG_AUDIO_DSOUND) += dsoundaudio.o
-common-obj-$(CONFIG_AUDIO_PT_INT) += audio_pt_int.o
common-obj-$(CONFIG_AUDIO_WIN_INT) += audio_win_int.o
common-obj-y += wavcapture.o
diff --git a/audio/audio_pt_int.c b/audio/audio_pt_int.c
deleted file mode 100644
index 3fe56d8514..0000000000
--- a/audio/audio_pt_int.c
+++ /dev/null
@@ -1,174 +0,0 @@
-#include "qemu/osdep.h"
-#include "qemu-common.h"
-#include "audio.h"
-
-#define AUDIO_CAP "audio-pt"
-
-#include "audio_int.h"
-#include "audio_pt_int.h"
-
-static void GCC_FMT_ATTR(3, 4) logerr (struct audio_pt *pt, int err,
- const char *fmt, ...)
-{
- va_list ap;
-
- va_start (ap, fmt);
- AUD_vlog (pt->drv, fmt, ap);
- va_end (ap);
-
- AUD_log (NULL, "\n");
- AUD_log (pt->drv, "Reason: %s\n", strerror (err));
-}
-
-int audio_pt_init (struct audio_pt *p, void *(*func) (void *),
- void *opaque, const char *drv, const char *cap)
-{
- int err, err2;
- const char *efunc;
- sigset_t set, old_set;
-
- p->drv = drv;
-
- err = sigfillset (&set);
- if (err) {
- logerr(p, errno, "%s(%s): sigfillset failed", cap, __func__);
- return -1;
- }
-
- err = pthread_mutex_init (&p->mutex, NULL);
- if (err) {
- efunc = "pthread_mutex_init";
- goto err0;
- }
-
- err = pthread_cond_init (&p->cond, NULL);
- if (err) {
- efunc = "pthread_cond_init";
- goto err1;
- }
-
- err = pthread_sigmask (SIG_BLOCK, &set, &old_set);
- if (err) {
- efunc = "pthread_sigmask";
- goto err2;
- }
-
- err = pthread_create (&p->thread, NULL, func, opaque);
-
- err2 = pthread_sigmask (SIG_SETMASK, &old_set, NULL);
- if (err2) {
- logerr(p, err2, "%s(%s): pthread_sigmask (restore) failed",
- cap, __func__);
- /* We have failed to restore original signal mask, all bets are off,
- so terminate the process */
- exit (EXIT_FAILURE);
- }
-
- if (err) {
- efunc = "pthread_create";
- goto err2;
- }
-
- return 0;
-
- err2:
- err2 = pthread_cond_destroy (&p->cond);
- if (err2) {
- logerr(p, err2, "%s(%s): pthread_cond_destroy failed", cap, __func__);
- }
-
- err1:
- err2 = pthread_mutex_destroy (&p->mutex);
- if (err2) {
- logerr(p, err2, "%s(%s): pthread_mutex_destroy failed", cap, __func__);
- }
-
- err0:
- logerr(p, err, "%s(%s): %s failed", cap, __func__, efunc);
- return -1;
-}
-
-int audio_pt_fini (struct audio_pt *p, const char *cap)
-{
- int err, ret = 0;
-
- err = pthread_cond_destroy (&p->cond);
- if (err) {
- logerr(p, err, "%s(%s): pthread_cond_destroy failed", cap, __func__);
- ret = -1;
- }
-
- err = pthread_mutex_destroy (&p->mutex);
- if (err) {
- logerr(p, err, "%s(%s): pthread_mutex_destroy failed", cap, __func__);
- ret = -1;
- }
- return ret;
-}
-
-int audio_pt_lock (struct audio_pt *p, const char *cap)
-{
- int err;
-
- err = pthread_mutex_lock (&p->mutex);
- if (err) {
- logerr(p, err, "%s(%s): pthread_mutex_lock failed", cap, __func__);
- return -1;
- }
- return 0;
-}
-
-int audio_pt_unlock (struct audio_pt *p, const char *cap)
-{
- int err;
-
- err = pthread_mutex_unlock (&p->mutex);
- if (err) {
- logerr(p, err, "%s(%s): pthread_mutex_unlock failed", cap, __func__);
- return -1;
- }
- return 0;
-}
-
-int audio_pt_wait (struct audio_pt *p, const char *cap)
-{
- int err;
-
- err = pthread_cond_wait (&p->cond, &p->mutex);
- if (err) {
- logerr(p, err, "%s(%s): pthread_cond_wait failed", cap, __func__);
- return -1;
- }
- return 0;
-}
-
-int audio_pt_unlock_and_signal (struct audio_pt *p, const char *cap)
-{
- int err;
-
- err = pthread_mutex_unlock (&p->mutex);
- if (err) {
- logerr(p, err, "%s(%s): pthread_mutex_unlock failed", cap, __func__);
- return -1;
- }
- err = pthread_cond_signal (&p->cond);
- if (err) {
- logerr(p, err, "%s(%s): pthread_cond_signal failed", cap, __func__);
- return -1;
- }
- return 0;
-}
-
-int audio_pt_join (struct audio_pt *p, void **arg, const char *cap)
-{
- int err;
- void *ret;
-
- err = pthread_join (p->thread, &ret);
- if (err) {
- logerr(p, err, "%s(%s): pthread_join failed", cap, __func__);
- return -1;
- }
- *arg = ret;
- return 0;
-}
diff --git a/audio/audio_pt_int.h b/audio/audio_pt_int.h
deleted file mode 100644
index 4c0c15b9af..0000000000
--- a/audio/audio_pt_int.h
+++ /dev/null
@@ -1,22 +0,0 @@
-#ifndef QEMU_AUDIO_PT_INT_H
-#define QEMU_AUDIO_PT_INT_H
-
-#include <pthread.h>
-
-struct audio_pt {
- const char *drv;
- pthread_t thread;
- pthread_cond_t cond;
- pthread_mutex_t mutex;
-};
-
-int audio_pt_init (struct audio_pt *, void *(*) (void *), void *,
- const char *, const char *);
-int audio_pt_fini (struct audio_pt *, const char *);
-int audio_pt_lock (struct audio_pt *, const char *);
-int audio_pt_unlock (struct audio_pt *, const char *);
-int audio_pt_wait (struct audio_pt *, const char *);
-int audio_pt_unlock_and_signal (struct audio_pt *, const char *);
-int audio_pt_join (struct audio_pt *, void **, const char *);
-
-#endif /* QEMU_AUDIO_PT_INT_H */
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 251b087a74..392225c875 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -8,7 +8,6 @@
#define AUDIO_CAP "pulseaudio"
#include "audio_int.h"
-#include "audio_pt_int.h"
typedef struct PAConnection {
char *server;
@@ -29,28 +28,16 @@ typedef struct {
typedef struct {
HWVoiceOut hw;
- int done;
- int live;
- int decr;
- int rpos;
pa_stream *stream;
- void *pcm_buf;
- struct audio_pt pt;
paaudio *g;
int samples;
} PAVoiceOut;
typedef struct {
HWVoiceIn hw;
- int done;
- int dead;
- int incr;
- int wpos;
pa_stream *stream;
- void *pcm_buf;
- struct audio_pt pt;
const void *read_data;
- size_t read_index, read_length;
+ size_t read_length;
paaudio *g;
int samples;
} PAVoiceIn;
@@ -88,298 +75,96 @@ static inline int PA_STREAM_IS_GOOD(pa_stream_state_t x)
}
#endif
-#define CHECK_SUCCESS_GOTO(c, rerror, expression, label) \
+#define CHECK_SUCCESS_GOTO(c, expression, label, msg) \
do { \
if (!(expression)) { \
- if (rerror) { \
- *(rerror) = pa_context_errno ((c)->context); \
- } \
+ qpa_logerr(pa_context_errno((c)->context), msg); \
goto label; \
} \
} while (0)
-#define CHECK_DEAD_GOTO(c, stream, rerror, label) \
+#define CHECK_DEAD_GOTO(c, stream, label, msg) \
do { \
if (!(c)->context || !PA_CONTEXT_IS_GOOD (pa_context_get_state((c)->context)) || \
!(stream) || !PA_STREAM_IS_GOOD (pa_stream_get_state ((stream)))) { \
if (((c)->context && pa_context_get_state ((c)->context) == PA_CONTEXT_FAILED) || \
((stream) && pa_stream_get_state ((stream)) == PA_STREAM_FAILED)) { \
- if (rerror) { \
- *(rerror) = pa_context_errno ((c)->context); \
- } \
+ qpa_logerr(pa_context_errno((c)->context), msg); \
} else { \
- if (rerror) { \
- *(rerror) = PA_ERR_BADSTATE; \
- } \
+ qpa_logerr(PA_ERR_BADSTATE, msg); \
} \
goto label; \
} \
} while (0)
-static int qpa_simple_read (PAVoiceIn *p, void *data, size_t length, int *rerror)
+static size_t qpa_read(HWVoiceIn *hw, void *data, size_t length)
{
+ PAVoiceIn *p = (PAVoiceIn *) hw;
PAConnection *c = p->g->conn;
+ size_t l;
+ int r;
pa_threaded_mainloop_lock(c->mainloop);
- CHECK_DEAD_GOTO(c, p->stream, rerror, unlock_and_fail);
+ CHECK_DEAD_GOTO(c, p->stream, unlock_and_fail,
+ "pa_threaded_mainloop_lock failed\n");
- while (length > 0) {
- size_t l;
-
- while (!p->read_data) {
- int r;
-
- r = pa_stream_peek (p->stream, &p->read_data, &p->read_length);
- CHECK_SUCCESS_GOTO(c, rerror, r == 0, unlock_and_fail);
-
- if (!p->read_data) {
- pa_threaded_mainloop_wait(c->mainloop);
- CHECK_DEAD_GOTO(c, p->stream, rerror, unlock_and_fail);
- } else {
- p->read_index = 0;
- }
- }
-
- l = p->read_length < length ? p->read_length : length;
- memcpy (data, (const uint8_t *) p->read_data+p->read_index, l);
-
- data = (uint8_t *) data + l;
- length -= l;
-
- p->read_index += l;
- p->read_length -= l;
+ if (!p->read_length) {
+ r = pa_stream_peek (p->stream, &p->read_data, &p->read_length);
+ CHECK_SUCCESS_GOTO(c, r == 0, unlock_and_fail,
+ "pa_stream_peek failed\n");
+ }
- if (!p->read_length) {
- int r;
+ l = MIN(p->read_length, length);
+ memcpy(data, p->read_data, l);
- r = pa_stream_drop (p->stream);
- p->read_data = NULL;
- p->read_length = 0;
- p->read_index = 0;
+ p->read_data += l;
+ p->read_length -= l;
- CHECK_SUCCESS_GOTO(c, rerror, r == 0, unlock_and_fail);
- }
+ if (!p->read_length) {
+ r = pa_stream_drop(p->stream);
+ CHECK_SUCCESS_GOTO(c, r == 0, unlock_and_fail,
+ "pa_stream_drop failed\n");
}
pa_threaded_mainloop_unlock(c->mainloop);
- return 0;
+ return l;
unlock_and_fail:
pa_threaded_mainloop_unlock(c->mainloop);
- return -1;
+ return 0;
}
-static int qpa_simple_write (PAVoiceOut *p, const void *data, size_t length, int *rerror)
+static size_t qpa_write(HWVoiceOut *hw, void *data, size_t length)
{
+ PAVoiceOut *p = (PAVoiceOut *) hw;
PAConnection *c = p->g->conn;
+ size_t l;
+ int r;
pa_threaded_mainloop_lock(c->mainloop);
- CHECK_DEAD_GOTO(c, p->stream, rerror, unlock_and_fail);
+ CHECK_DEAD_GOTO(c, p->stream, unlock_and_fail,
+ "pa_threaded_mainloop_lock failed\n");
- while (length > 0) {
- size_t l;
- int r;
+ l = pa_stream_writable_size(p->stream);
- while (!(l = pa_stream_writable_size (p->stream))) {
- pa_threaded_mainloop_wait(c->mainloop);
- CHECK_DEAD_GOTO(c, p->stream, rerror, unlock_and_fail);
- }
+ CHECK_SUCCESS_GOTO(c, l != (size_t) -1, unlock_and_fail,
+ "pa_stream_writable_size failed\n");
- CHECK_SUCCESS_GOTO(c, rerror, l != (size_t) -1, unlock_and_fail);
-
- if (l > length) {
- l = length;
- }
-
- r = pa_stream_write (p->stream, data, l, NULL, 0LL, PA_SEEK_RELATIVE);
- CHECK_SUCCESS_GOTO(c, rerror, r >= 0, unlock_and_fail);
-
- data = (const uint8_t *) data + l;
- length -= l;
+ if (l > length) {
+ l = length;
}
+ r = pa_stream_write(p->stream, data, l, NULL, 0LL, PA_SEEK_RELATIVE);
+ CHECK_SUCCESS_GOTO(c, r >= 0, unlock_and_fail, "pa_stream_write failed\n");
+
pa_threaded_mainloop_unlock(c->mainloop);
- return 0;
+ return l;
unlock_and_fail:
pa_threaded_mainloop_unlock(c->mainloop);
- return -1;
-}
-
-static void *qpa_thread_out (void *arg)
-{
- PAVoiceOut *pa = arg;
- HWVoiceOut *hw = &pa->hw;
-
- if (audio_pt_lock(&pa->pt, __func__)) {
- return NULL;
- }
-
- for (;;) {
- int decr, to_mix, rpos;
-
- for (;;) {
- if (pa->done) {
- goto exit;
- }
-
- if (pa->live > 0) {
- break;
- }
-
- if (audio_pt_wait(&pa->pt, __func__)) {
- goto exit;
- }
- }
-
- decr = to_mix = MIN(pa->live, pa->samples >> 5);
- rpos = pa->rpos;
-
- if (audio_pt_unlock(&pa->pt, __func__)) {
- return NULL;
- }
-
- while (to_mix) {
- int error;
- int chunk = MIN (to_mix, hw->samples - rpos);
- struct st_sample *src = hw->mix_buf + rpos;
-
- hw->clip (pa->pcm_buf, src, chunk);
-
- if (qpa_simple_write (pa, pa->pcm_buf,
- chunk << hw->info.shift, &error) < 0) {
- qpa_logerr (error, "pa_simple_write failed\n");
- return NULL;
- }
-
- rpos = (rpos + chunk) % hw->samples;
- to_mix -= chunk;
- }
-
- if (audio_pt_lock(&pa->pt, __func__)) {
- return NULL;
- }
-
- pa->rpos = rpos;
- pa->live -= decr;
- pa->decr += decr;
- }
-
- exit:
- audio_pt_unlock(&pa->pt, __func__);
- return NULL;
-}
-
-static int qpa_run_out (HWVoiceOut *hw, int live)
-{
- int decr;
- PAVoiceOut *pa = (PAVoiceOut *) hw;
-
- if (audio_pt_lock(&pa->pt, __func__)) {
- return 0;
- }
-
- decr = MIN (live, pa->decr);
- pa->decr -= decr;
- pa->live = live - decr;
- hw->rpos = pa->rpos;
- if (pa->live > 0) {
- audio_pt_unlock_and_signal(&pa->pt, __func__);
- }
- else {
- audio_pt_unlock(&pa->pt, __func__);
- }
- return decr;
-}
-
-/* capture */
-static void *qpa_thread_in (void *arg)
-{
- PAVoiceIn *pa = arg;
- HWVoiceIn *hw = &pa->hw;
-
- if (audio_pt_lock(&pa->pt, __func__)) {
- return NULL;
- }
-
- for (;;) {
- int incr, to_grab, wpos;
-
- for (;;) {
- if (pa->done) {
- goto exit;
- }
-
- if (pa->dead > 0) {
- break;
- }
-
- if (audio_pt_wait(&pa->pt, __func__)) {
- goto exit;
- }
- }
-
- incr = to_grab = MIN(pa->dead, pa->samples >> 5);
- wpos = pa->wpos;
-
- if (audio_pt_unlock(&pa->pt, __func__)) {
- return NULL;
- }
-
- while (to_grab) {
- int error;
- int chunk = MIN (to_grab, hw->samples - wpos);
- void *buf = advance (pa->pcm_buf, wpos);
-
- if (qpa_simple_read (pa, buf,
- chunk << hw->info.shift, &error) < 0) {
- qpa_logerr (error, "pa_simple_read failed\n");
- return NULL;
- }
-
- hw->conv (hw->conv_buf + wpos, buf, chunk);
- wpos = (wpos + chunk) % hw->samples;
- to_grab -= chunk;
- }
-
- if (audio_pt_lock(&pa->pt, __func__)) {
- return NULL;
- }
-
- pa->wpos = wpos;
- pa->dead -= incr;
- pa->incr += incr;
- }
-
- exit:
- audio_pt_unlock(&pa->pt, __func__);
- return NULL;
-}
-
-static int qpa_run_in (HWVoiceIn *hw)
-{
- int live, incr, dead;
- PAVoiceIn *pa = (PAVoiceIn *) hw;
-
- if (audio_pt_lock(&pa->pt, __func__)) {
- return 0;
- }
-
- live = audio_pcm_hw_get_live_in (hw);
- dead = hw->samples - live;
- incr = MIN (dead, pa->incr);
- pa->incr -= incr;
- pa->dead = dead - incr;
- hw->wpos = pa->wpos;
- if (pa->dead > 0) {
- audio_pt_unlock_and_signal(&pa->pt, __func__);
- }
- else {
- audio_pt_unlock(&pa->pt, __func__);
- }
- return incr;
+ return 0;
}
static pa_sample_format_t audfmt_to_pa (AudioFormat afmt, int endianness)
@@ -467,13 +252,6 @@ static void stream_state_cb (pa_stream *s, void * userdata)
}
}
-static void stream_request_cb (pa_stream *s, size_t length, void *userdata)
-{
- PAConnection *c = userdata;
-
- pa_threaded_mainloop_signal(c->mainloop, 0);
-}
-
static pa_stream *qpa_simple_new (
PAConnection *c,
const char *name,
@@ -496,8 +274,6 @@ static pa_stream *qpa_simple_new (
}
pa_stream_set_state_callback (stream, stream_state_cb, c);
- pa_stream_set_read_callback (stream, stream_request_cb, c);
- pa_stream_set_write_callback (stream, stream_request_cb, c);
flags =
PA_STREAM_INTERPOLATE_TIMING
@@ -581,28 +357,9 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = pa->samples = audio_buffer_samples(g->dev->out, &obt_as,
46440);
- pa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
- pa->rpos = hw->rpos;
- if (!pa->pcm_buf) {
- dolog("Could not allocate buffer (%zu bytes)\n",
- hw->samples << hw->info.shift);
- goto fail2;
- }
-
- if (audio_pt_init(&pa->pt, qpa_thread_out, hw, AUDIO_CAP, __func__)) {
- goto fail3;
- }
return 0;
- fail3:
- g_free (pa->pcm_buf);
- pa->pcm_buf = NULL;
- fail2:
- if (pa->stream) {
- pa_stream_unref (pa->stream);
- pa->stream = NULL;
- }
fail1:
return -1;
}
@@ -642,28 +399,9 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = pa->samples = audio_buffer_samples(g->dev->in, &obt_as,
46440);
- pa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
- pa->wpos = hw->wpos;
- if (!pa->pcm_buf) {
- dolog("Could not allocate buffer (%zu bytes)\n",
- hw->samples << hw->info.shift);
- goto fail2;
- }
-
- if (audio_pt_init(&pa->pt, qpa_thread_in, hw, AUDIO_CAP, __func__)) {
- goto fail3;
- }
return 0;
- fail3:
- g_free (pa->pcm_buf);
- pa->pcm_buf = NULL;
- fail2:
- if (pa->stream) {
- pa_stream_unref (pa->stream);
- pa->stream = NULL;
- }
fail1:
return -1;
}
@@ -691,42 +429,22 @@ static void qpa_simple_disconnect(PAConnection *c, pa_stream *stream)
static void qpa_fini_out (HWVoiceOut *hw)
{
- void *ret;
PAVoiceOut *pa = (PAVoiceOut *) hw;
- audio_pt_lock(&pa->pt, __func__);
- pa->done = 1;
- audio_pt_unlock_and_signal(&pa->pt, __func__);
- audio_pt_join(&pa->pt, &ret, __func__);
-
if (pa->stream) {
qpa_simple_disconnect(pa->g->conn, pa->stream);
pa->stream = NULL;
}
-
- audio_pt_fini(&pa->pt, __func__);
- g_free (pa->pcm_buf);
- pa->pcm_buf = NULL;
}
static void qpa_fini_in (HWVoiceIn *hw)
{
- void *ret;
PAVoiceIn *pa = (PAVoiceIn *) hw;
- audio_pt_lock(&pa->pt, __func__);
- pa->done = 1;
- audio_pt_unlock_and_signal(&pa->pt, __func__);
- audio_pt_join(&pa->pt, &ret, __func__);
-
if (pa->stream) {
qpa_simple_disconnect(pa->g->conn, pa->stream);
pa->stream = NULL;
}
-
- audio_pt_fini(&pa->pt, __func__);
- g_free (pa->pcm_buf);
- pa->pcm_buf = NULL;
}
static int qpa_ctl_out (HWVoiceOut *hw, int cmd, ...)
@@ -964,12 +682,12 @@ static void qpa_audio_fini (void *opaque)
static struct audio_pcm_ops qpa_pcm_ops = {
.init_out = qpa_init_out,
.fini_out = qpa_fini_out,
- .run_out = qpa_run_out,
+ .write = qpa_write,
.ctl_out = qpa_ctl_out,
.init_in = qpa_init_in,
.fini_in = qpa_fini_in,
- .run_in = qpa_run_in,
+ .read = qpa_read,
.ctl_in = qpa_ctl_in
};
diff --git a/configure b/configure
index 224d3071ac..872a5efb3b 100755
--- a/configure
+++ b/configure
@@ -286,7 +286,6 @@ block_drv_ro_whitelist=""
host_cc="cc"
libs_softmmu=""
libs_tools=""
-audio_pt_int=""
audio_win_int=""
libs_qga=""
debug_info="yes"
@@ -3339,7 +3338,6 @@ for drv in $audio_drv_list; do
audio_drv_probe $drv pulse/pulseaudio.h "-lpulse" \
"pa_context_set_source_output_volume(NULL, 0, NULL, NULL, NULL); return 0;"
pulse_libs="-lpulse"
- audio_pt_int="yes"
;;
sdl)
@@ -6304,9 +6302,6 @@ echo "PULSE_LIBS=$pulse_libs" >> $config_host_mak
echo "COREAUDIO_LIBS=$coreaudio_libs" >> $config_host_mak
echo "DSOUND_LIBS=$dsound_libs" >> $config_host_mak
echo "OSS_LIBS=$oss_libs" >> $config_host_mak
-if test "$audio_pt_int" = "yes" ; then
- echo "CONFIG_AUDIO_PT_INT=y" >> $config_host_mak
-fi
if test "$audio_win_int" = "yes" ; then
echo "CONFIG_AUDIO_WIN_INT=y" >> $config_host_mak
fi
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 35/52] sdlaudio: port to the new audio backend api
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (33 preceding siblings ...)
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 34/52] paaudio: " Kővágó, Zoltán
@ 2018-12-23 20:52 ` Kővágó, Zoltán
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 36/52] spiceaudio: " Kővágó, Zoltán
` (15 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:52 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
SDL2 is still a big mess, but it's probably not worse than the current
version.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/sdlaudio.c | 136 +++++++++++++++++------------------------------
1 file changed, 50 insertions(+), 86 deletions(-)
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index 92da4804c6..685cbc83b8 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -42,11 +42,6 @@
typedef struct SDLVoiceOut {
HWVoiceOut hw;
- int live;
-#if USE_SEMAPHORE
- int rpos;
-#endif
- int decr;
} SDLVoiceOut;
static struct SDLAudioState {
@@ -252,17 +247,13 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
SDLVoiceOut *sdl = opaque;
SDLAudioState *s = &glob_sdl;
HWVoiceOut *hw = &sdl->hw;
- int samples = len >> hw->info.shift;
if (s->exit) {
return;
}
- while (samples) {
- int to_mix, decr;
-
- /* dolog ("in callback samples=%d\n", samples); */
#if USE_SEMAPHORE
+ while (len) {
sdl_wait (s, "sdl_callback");
if (s->exit) {
return;
@@ -271,95 +262,66 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
if (sdl_lock (s, "sdl_callback")) {
return;
}
-
- if (audio_bug(__func__, sdl->live < 0 || sdl->live > hw->samples)) {
- dolog("sdl->live=%d hw->samples=%zu\n", sdl->live, hw->samples);
- return;
- }
-
- if (!sdl->live) {
- goto again;
- }
-#else
- if (s->exit || !sdl->live) {
- break;
- }
#endif
- /* dolog ("in callback live=%d\n", live); */
- to_mix = MIN (samples, sdl->live);
- decr = to_mix;
- while (to_mix) {
- int chunk = MIN (to_mix, hw->samples - hw->rpos);
- struct st_sample *src = hw->mix_buf + hw->rpos;
+ while (hw->pending_emul && len) {
+ size_t write_len;
+ ssize_t start = ((ssize_t) hw->pos_emul) - hw->pending_emul;
+ if (start < 0) {
+ start += hw->size_emul;
+ }
+ assert(start >= 0 && start < hw->size_emul);
- /* dolog ("in callback to_mix %d, chunk %d\n", to_mix, chunk); */
- hw->clip (buf, src, chunk);
-#if USE_SEMAPHORE
- sdl->rpos = (sdl->rpos + chunk) % hw->samples;
-#else
- hw->rpos = (hw->rpos + chunk) % hw->samples;
-#endif
- to_mix -= chunk;
- buf += chunk << hw->info.shift;
+ write_len = MIN(MIN(hw->pending_emul, len),
+ hw->size_emul - start);
+
+ memcpy(buf, hw->buf_emul + start, write_len);
+ hw->pending_emul -= write_len;
+ len -= write_len;
+ buf += write_len;
}
- samples -= decr;
- sdl->live -= decr;
- sdl->decr += decr;
#if USE_SEMAPHORE
- again:
if (sdl_unlock (s, "sdl_callback")) {
return;
}
-#endif
}
- /* dolog ("done len=%d\n", len); */
-
-#if (SDL_MAJOR_VERSION >= 2)
- /* SDL2 does not clear the remaining buffer for us, so do it on our own */
- if (samples) {
- memset(buf, 0, samples << hw->info.shift);
- }
-#endif
-}
-
-static int sdl_run_out (HWVoiceOut *hw, int live)
-{
- int decr;
- SDLVoiceOut *sdl = (SDLVoiceOut *) hw;
- SDLAudioState *s = &glob_sdl;
-
- if (sdl_lock (s, "sdl_run_out")) {
- return 0;
- }
-
- if (sdl->decr > live) {
- ldebug ("sdl->decr %d live %d sdl->live %d\n",
- sdl->decr,
- live,
- sdl->live);
- }
-
- decr = MIN (sdl->decr, live);
- sdl->decr -= decr;
-
-#if USE_SEMAPHORE
- sdl->live = live - decr;
- hw->rpos = sdl->rpos;
#else
- sdl->live = live;
+ /* clear remaining buffer that we couldn't fill with data */
+ if (len) {
+ memset(buf, 0, len);
+ }
#endif
-
- if (sdl->live > 0) {
- sdl_unlock_and_post (s, "sdl_run_out");
- }
- else {
- sdl_unlock (s, "sdl_run_out");
- }
- return decr;
}
+#define SDL_WRAPPER_FUNC(name, ret_type, args_decl, args, fail, unlock) \
+ static ret_type glue(sdl_, name)args_decl \
+ { \
+ SDLAudioState *s = &glob_sdl; \
+ ret_type ret; \
+ \
+ if (sdl_lock(s, "sdl_" #name)) { \
+ fail; \
+ return 0; /* implicitly casts to NULL */ \
+ } \
+ \
+ ret = glue(audio_generic_, name)args; \
+ \
+ unlock(s, "sdl_" #name); \
+ return ret; \
+ }
+
+SDL_WRAPPER_FUNC(get_buffer_out, void *, (HWVoiceOut *hw, size_t *size),
+ (hw, size), *size = 0, sdl_unlock)
+SDL_WRAPPER_FUNC(put_buffer_out_nowrite, size_t,
+ (HWVoiceOut *hw, void *buf, size_t size), (hw, buf, size),
+ /*nothing*/, sdl_unlock_and_post)
+SDL_WRAPPER_FUNC(write, size_t,
+ (HWVoiceOut *hw, void *buf, size_t size), (hw, buf, size),
+ /* nothing */, sdl_unlock_and_post)
+
+#undef SDL_WRAPPER_FUNC
+
static void sdl_fini_out (HWVoiceOut *hw)
{
(void) hw;
@@ -476,7 +438,9 @@ static void sdl_audio_fini (void *opaque)
static struct audio_pcm_ops sdl_pcm_ops = {
.init_out = sdl_init_out,
.fini_out = sdl_fini_out,
- .run_out = sdl_run_out,
+ .write = sdl_write,
+ .get_buffer_out = sdl_get_buffer_out,
+ .put_buffer_out = sdl_put_buffer_out_nowrite,
.ctl_out = sdl_ctl_out,
};
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 36/52] spiceaudio: port to the new audio backend api
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (34 preceding siblings ...)
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 35/52] sdlaudio: " Kővágó, Zoltán
@ 2018-12-23 20:52 ` Kővágó, Zoltán
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 37/52] wavaudio: " Kővágó, Zoltán
` (14 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:52 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/spiceaudio.c | 112 +++++++++++++++------------------------------
1 file changed, 38 insertions(+), 74 deletions(-)
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index f963853ed8..d1605d3939 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -51,7 +51,7 @@ typedef struct SpiceVoiceOut {
SpiceRateCtl rate;
int active;
uint32_t *frame;
- uint32_t *fpos;
+ uint32_t fpos;
uint32_t fsize;
} SpiceVoiceOut;
@@ -60,7 +60,6 @@ typedef struct SpiceVoiceIn {
SpiceRecordInstance sin;
SpiceRateCtl rate;
int active;
- uint32_t samples[LINE_IN_SAMPLES];
} SpiceVoiceIn;
static const SpicePlaybackInterface playback_sif = {
@@ -152,44 +151,36 @@ static void line_out_fini (HWVoiceOut *hw)
spice_server_remove_interface (&out->sin.base);
}
-static int line_out_run (HWVoiceOut *hw, int live)
+static void *line_out_get_buffer(HWVoiceOut *hw, size_t *size)
{
- SpiceVoiceOut *out = container_of (hw, SpiceVoiceOut, hw);
- int rpos, decr;
- int samples;
+ SpiceVoiceOut *out = container_of(hw, SpiceVoiceOut, hw);
+ size_t decr;
- if (!live) {
- return 0;
+ if (!out->frame) {
+ spice_server_playback_get_buffer(&out->sin, &out->frame, &out->fsize);
+ out->fpos = 0;
}
- decr = rate_get_samples (&hw->info, &out->rate);
- decr = MIN (live, decr);
+ decr = rate_get_samples(&hw->info, &out->rate);
+ decr = MIN(out->fsize - out->fpos, decr);
- samples = decr;
- rpos = hw->rpos;
- while (samples) {
- int left_till_end_samples = hw->samples - rpos;
- int len = MIN (samples, left_till_end_samples);
+ *size = decr << hw->info.shift;
+ return out->frame + out->fpos;
+}
- if (!out->frame) {
- spice_server_playback_get_buffer (&out->sin, &out->frame, &out->fsize);
- out->fpos = out->frame;
- }
- if (out->frame) {
- len = MIN (len, out->fsize);
- hw->clip (out->fpos, hw->mix_buf + rpos, len);
- out->fsize -= len;
- out->fpos += len;
- if (out->fsize == 0) {
- spice_server_playback_put_samples (&out->sin, out->frame);
- out->frame = out->fpos = NULL;
- }
- }
- rpos = (rpos + len) % hw->samples;
- samples -= len;
+static size_t line_out_put_buffer(HWVoiceOut *hw, void *buf, size_t size)
+{
+ SpiceVoiceOut *out = container_of(hw, SpiceVoiceOut, hw);
+
+ out->fpos += size >> 2;
+ assert(buf == out->frame + out->fpos && out->fpos <= out->fsize);
+
+ if (out->fpos == out->fsize) { /* buffer full */
+ spice_server_playback_put_samples(&out->sin, out->frame);
+ out->frame = NULL;
}
- hw->rpos = rpos;
- return decr;
+
+ return size;
}
static int line_out_ctl (HWVoiceOut *hw, int cmd, ...)
@@ -211,9 +202,9 @@ static int line_out_ctl (HWVoiceOut *hw, int cmd, ...)
}
out->active = 0;
if (out->frame) {
- memset (out->fpos, 0, out->fsize << 2);
+ memset(out->frame + out->fpos, 0, (out->fsize - out->fpos) << 2);
spice_server_playback_put_samples (&out->sin, out->frame);
- out->frame = out->fpos = NULL;
+ out->frame = NULL;
}
spice_server_playback_stop (&out->sin);
break;
@@ -275,49 +266,20 @@ static void line_in_fini (HWVoiceIn *hw)
spice_server_remove_interface (&in->sin.base);
}
-static int line_in_run (HWVoiceIn *hw)
+static size_t line_in_read(HWVoiceIn *hw, void *buf, size_t len)
{
SpiceVoiceIn *in = container_of (hw, SpiceVoiceIn, hw);
- int num_samples;
- int ready;
- int len[2];
- uint64_t delta_samp;
- const uint32_t *samples;
+ uint64_t delta_samp = rate_get_samples(&hw->info, &in->rate);
+ uint64_t to_read = MIN(len >> 2, delta_samp);
+ size_t ready = spice_server_record_get_samples(&in->sin, buf, to_read);
- if (!(num_samples = hw->samples - audio_pcm_hw_get_live_in (hw))) {
- return 0;
- }
-
- delta_samp = rate_get_samples (&hw->info, &in->rate);
- num_samples = MIN (num_samples, delta_samp);
-
- ready = spice_server_record_get_samples (&in->sin, in->samples, num_samples);
- samples = in->samples;
+ /* XXX: do we need this? */
if (ready == 0) {
- static const uint32_t silence[LINE_IN_SAMPLES];
- samples = silence;
- ready = LINE_IN_SAMPLES;
+ memset(buf, 0, to_read << 2);
+ ready = to_read;
}
- num_samples = MIN (ready, num_samples);
-
- if (hw->wpos + num_samples > hw->samples) {
- len[0] = hw->samples - hw->wpos;
- len[1] = num_samples - len[0];
- } else {
- len[0] = num_samples;
- len[1] = 0;
- }
-
- hw->conv (hw->conv_buf + hw->wpos, samples, len[0]);
-
- if (len[1]) {
- hw->conv (hw->conv_buf, samples + len[0], len[1]);
- }
-
- hw->wpos = (hw->wpos + num_samples) % hw->samples;
-
- return num_samples;
+ return ready << 2;
}
static int line_in_ctl (HWVoiceIn *hw, int cmd, ...)
@@ -366,12 +328,14 @@ static int line_in_ctl (HWVoiceIn *hw, int cmd, ...)
static struct audio_pcm_ops audio_callbacks = {
.init_out = line_out_init,
.fini_out = line_out_fini,
- .run_out = line_out_run,
+ .write = audio_generic_write,
+ .get_buffer_out = line_out_get_buffer,
+ .put_buffer_out = line_out_put_buffer,
.ctl_out = line_out_ctl,
.init_in = line_in_init,
.fini_in = line_in_fini,
- .run_in = line_in_run,
+ .read = line_in_read,
.ctl_in = line_in_ctl,
};
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 37/52] wavaudio: port to the new audio backend api
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (35 preceding siblings ...)
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 36/52] spiceaudio: " Kővágó, Zoltán
@ 2018-12-23 20:52 ` Kővágó, Zoltán
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 38/52] audio: remove remains of the old " Kővágó, Zoltán
` (13 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:52 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/wavaudio.c | 54 ++++++++----------------------------------------
1 file changed, 9 insertions(+), 45 deletions(-)
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 58300663ff..31db03aadb 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -34,52 +34,28 @@ typedef struct WAVVoiceOut {
HWVoiceOut hw;
FILE *f;
int64_t old_ticks;
- void *pcm_buf;
int total_samples;
} WAVVoiceOut;
-static int wav_run_out (HWVoiceOut *hw, int live)
+static size_t wav_write_out(HWVoiceOut *hw, void *buf, size_t len)
{
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
- int rpos, decr, samples;
- uint8_t *dst;
- struct st_sample *src;
int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
int64_t ticks = now - wav->old_ticks;
int64_t bytes =
muldiv64(ticks, hw->info.bytes_per_second, NANOSECONDS_PER_SECOND);
- if (bytes > INT_MAX) {
- samples = INT_MAX >> hw->info.shift;
- }
- else {
- samples = bytes >> hw->info.shift;
- }
-
+ bytes = MIN(bytes, len);
+ bytes = bytes >> hw->info.shift << hw->info.shift;
wav->old_ticks = now;
- decr = MIN (live, samples);
- samples = decr;
- rpos = hw->rpos;
- while (samples) {
- int left_till_end_samples = hw->samples - rpos;
- int convert_samples = MIN (samples, left_till_end_samples);
- src = hw->mix_buf + rpos;
- dst = advance (wav->pcm_buf, rpos << hw->info.shift);
-
- hw->clip (dst, src, convert_samples);
- if (fwrite (dst, convert_samples << hw->info.shift, 1, wav->f) != 1) {
- dolog ("wav_run_out: fwrite of %d bytes failed\nReaons: %s\n",
- convert_samples << hw->info.shift, strerror (errno));
- }
-
- rpos = (rpos + convert_samples) % hw->samples;
- samples -= convert_samples;
- wav->total_samples += convert_samples;
+ if (bytes && fwrite(buf, bytes, 1, wav->f) != 1) {
+ dolog("wav_write_out: fwrite of %zu bytes failed\nReaons: %s\n",
+ bytes, strerror(errno));
}
- hw->rpos = rpos;
- return decr;
+ wav->total_samples += bytes >> hw->info.shift;
+ return bytes;
}
/* VICE code: Store number as little endian. */
@@ -135,13 +111,6 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
audio_pcm_init_info (&hw->info, &wav_as);
hw->samples = 1024;
- wav->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
- if (!wav->pcm_buf) {
- dolog("Could not allocate buffer (%zu bytes)\n",
- hw->samples << hw->info.shift);
- return -1;
- }
-
le_store (hdr + 22, hw->info.nchannels, 2);
le_store (hdr + 24, hw->info.freq, 4);
le_store (hdr + 28, hw->info.freq << (bits16 + stereo), 4);
@@ -151,8 +120,6 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
if (!wav->f) {
dolog ("Failed to open wave file `%s'\nReason: %s\n",
wav_path, strerror(errno));
- g_free (wav->pcm_buf);
- wav->pcm_buf = NULL;
return -1;
}
@@ -206,9 +173,6 @@ static void wav_fini_out (HWVoiceOut *hw)
wav->f, strerror (errno));
}
wav->f = NULL;
-
- g_free (wav->pcm_buf);
- wav->pcm_buf = NULL;
}
static int wav_ctl_out (HWVoiceOut *hw, int cmd, ...)
@@ -232,7 +196,7 @@ static void wav_audio_fini (void *opaque)
static struct audio_pcm_ops wav_pcm_ops = {
.init_out = wav_init_out,
.fini_out = wav_fini_out,
- .run_out = wav_run_out,
+ .write = wav_write_out,
.ctl_out = wav_ctl_out,
};
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 38/52] audio: remove remains of the old backend api
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (36 preceding siblings ...)
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 37/52] wavaudio: " Kővágó, Zoltán
@ 2018-12-23 20:52 ` Kővágó, Zoltán
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 39/52] audio: unify input and output mixeng buffer management Kővágó, Zoltán
` (12 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:52 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio.c | 42 ++++++------------------------------------
audio/audio_int.h | 7 -------
2 files changed, 6 insertions(+), 43 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index 1897e22444..4c33e057d7 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -543,7 +543,7 @@ static size_t audio_pcm_hw_find_min_in (HWVoiceIn *hw)
return m;
}
-size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw)
+static size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw)
{
size_t live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
if (audio_bug(__func__, live > hw->samples)) {
@@ -553,29 +553,7 @@ size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw)
return live;
}
-size_t audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf,
- size_t live, size_t pending)
-{
- size_t left = hw->samples - pending;
- size_t len = MIN (left, live);
- size_t clipped = 0;
-
- while (len) {
- struct st_sample *src = hw->mix_buf + hw->rpos;
- uint8_t *dst = advance (pcm_buf, hw->rpos << hw->info.shift);
- size_t samples_till_end_of_buf = hw->samples - hw->rpos;
- size_t samples_to_clip = MIN (len, samples_till_end_of_buf);
-
- hw->clip (dst, src, samples_to_clip);
-
- hw->rpos = (hw->rpos + samples_to_clip) % hw->samples;
- len -= samples_to_clip;
- clipped += samples_to_clip;
- }
- return clipped;
-}
-
-static void audio_pcm_hw_clip_out2(HWVoiceOut *hw, void *pcm_buf, size_t len)
+static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len)
{
size_t clipped = 0;
size_t pos = hw->rpos;
@@ -1080,7 +1058,7 @@ static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
void *buf = hw->pcm_ops->get_buffer_out(hw, &size);
decr = MIN(size >> hw->info.shift, live);
- audio_pcm_hw_clip_out2(hw, buf, decr);
+ audio_pcm_hw_clip_out(hw, buf, decr);
proc = hw->pcm_ops->put_buffer_out(hw, buf, decr << hw->info.shift) >>
hw->info.shift;
@@ -1143,11 +1121,7 @@ static void audio_run_out (AudioState *s)
}
prev_rpos = hw->rpos;
- if (hw->pcm_ops->run_out) {
- played = hw->pcm_ops->run_out(hw, live);
- } else {
- played = audio_pcm_hw_run_out(hw, live);
- }
+ played = audio_pcm_hw_run_out(hw, live);
replay_audio_out(&played);
if (audio_bug(__func__, hw->rpos >= hw->samples)) {
dolog("hw->rpos=%zu hw->samples=%zu played=%zu\n",
@@ -1244,12 +1218,8 @@ static void audio_run_in (AudioState *s)
size_t captured = 0, min;
if (replay_mode != REPLAY_MODE_PLAY) {
- if (hw->pcm_ops->run_in) {
- captured = hw->pcm_ops->run_in(hw);
- } else {
- captured = audio_pcm_hw_run_in(
- hw, hw->samples - audio_pcm_hw_get_live_in(hw));
- }
+ captured = audio_pcm_hw_run_in(
+ hw, hw->samples - audio_pcm_hw_get_live_in(hw));
}
replay_audio_in(&captured, hw->conv_buf, &hw->wpos, hw->samples);
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 5c361d9451..cca7efaa05 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -152,7 +152,6 @@ struct audio_driver {
struct audio_pcm_ops {
int (*init_out)(HWVoiceOut *hw, audsettings *as, void *drv_opaque);
void (*fini_out)(HWVoiceOut *hw);
- int (*run_out) (HWVoiceOut *hw, int live);
size_t (*write) (HWVoiceOut *hw, void *buf, size_t size);
/*
* get a buffer that after later can be passed to put_buffer_out; optional
@@ -170,7 +169,6 @@ struct audio_pcm_ops {
int (*init_in) (HWVoiceIn *hw, audsettings *as, void *drv_opaque);
void (*fini_in) (HWVoiceIn *hw);
- int (*run_in) (HWVoiceIn *hw);
size_t (*read) (HWVoiceIn *hw, void *buf, size_t size);
void *(*get_buffer_in)(HWVoiceIn *hw, size_t *size);
void (*put_buffer_in)(HWVoiceIn *hw, void *buf, size_t size);
@@ -236,11 +234,6 @@ audio_driver *audio_driver_lookup(const char *name);
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);
-size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw);
-
-size_t audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf,
- size_t live, size_t pending);
-
int audio_bug (const char *funcname, int cond);
void *audio_calloc (const char *funcname, int nmemb, size_t size);
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 39/52] audio: unify input and output mixeng buffer management
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (37 preceding siblings ...)
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 38/52] audio: remove remains of the old " Kővágó, Zoltán
@ 2018-12-23 20:52 ` Kővágó, Zoltán
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 40/52] audio: remove hw->samples, buffer_size_in/out pcm_ops Kővágó, Zoltán
` (11 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:52 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Usage notes: hw->samples became hw->{mix,conv}_buf->size, except before
initialization (audio_pcm_hw_alloc_resources_*), hw->samples gives the
initial size of the STSampleBuffer. The next commit tries to fix this
inconsistency.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio.c | 130 ++++++++++++++++++++---------------------
audio/audio_int.h | 12 ++--
audio/audio_template.h | 19 +++---
audio/ossaudio.c | 2 +-
4 files changed, 80 insertions(+), 83 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index 4c33e057d7..f195d8eb95 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -546,8 +546,8 @@ static size_t audio_pcm_hw_find_min_in (HWVoiceIn *hw)
static size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw)
{
size_t live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
- if (audio_bug(__func__, live > hw->samples)) {
- dolog("live=%zu hw->samples=%zu\n", live, hw->samples);
+ if (audio_bug(__func__, live > hw->conv_buf->size)) {
+ dolog("live=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
return 0;
}
return live;
@@ -556,17 +556,17 @@ static size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw)
static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len)
{
size_t clipped = 0;
- size_t pos = hw->rpos;
+ size_t pos = hw->mix_buf->pos;
while (len) {
- st_sample *src = hw->mix_buf + pos;
+ st_sample *src = hw->mix_buf->samples + pos;
uint8_t *dst = advance (pcm_buf, clipped << hw->info.shift);
- size_t samples_till_end_of_buf = hw->samples - pos;
+ size_t samples_till_end_of_buf = hw->mix_buf->size - pos;
size_t samples_to_clip = MIN(len, samples_till_end_of_buf);
hw->clip (dst, src, samples_to_clip);
- pos = (pos + samples_to_clip) % hw->samples;
+ pos = (pos + samples_to_clip) % hw->mix_buf->size;
len -= samples_to_clip;
clipped += samples_to_clip;
}
@@ -581,17 +581,17 @@ static size_t audio_pcm_sw_get_rpos_in(SWVoiceIn *sw)
ssize_t live = hw->total_samples_captured - sw->total_hw_samples_acquired;
ssize_t rpos;
- if (audio_bug(__func__, live < 0 || live > hw->samples)) {
- dolog("live=%zu hw->samples=%zu\n", live, hw->samples);
+ if (audio_bug(__func__, live < 0 || live > hw->conv_buf->size)) {
+ dolog("live=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
return 0;
}
- rpos = hw->wpos - live;
+ rpos = hw->conv_buf->pos - live;
if (rpos >= 0) {
return rpos;
}
else {
- return hw->samples + rpos;
+ return hw->conv_buf->size + rpos;
}
}
@@ -601,11 +601,11 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
size_t samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
struct st_sample *src, *dst = sw->buf;
- rpos = audio_pcm_sw_get_rpos_in (sw) % hw->samples;
+ rpos = audio_pcm_sw_get_rpos_in(sw) % hw->conv_buf->size;
live = hw->total_samples_captured - sw->total_hw_samples_acquired;
- if (audio_bug(__func__, live > hw->samples)) {
- dolog("live_in=%zu hw->samples=%zu\n", live, hw->samples);
+ if (audio_bug(__func__, live > hw->conv_buf->size)) {
+ dolog("live_in=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
return 0;
}
@@ -618,11 +618,11 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
swlim = MIN (swlim, samples);
while (swlim) {
- src = hw->conv_buf + rpos;
- if (hw->wpos > rpos) {
- isamp = hw->wpos - rpos;
+ src = hw->conv_buf->samples + rpos;
+ if (hw->conv_buf->pos > rpos) {
+ isamp = hw->conv_buf->pos - rpos;
} else {
- isamp = hw->samples - rpos;
+ isamp = hw->conv_buf->size - rpos;
}
if (!isamp) {
@@ -632,7 +632,7 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
swlim -= osamp;
- rpos = (rpos + isamp) % hw->samples;
+ rpos = (rpos + isamp) % hw->conv_buf->size;
dst += osamp;
ret += osamp;
total += isamp;
@@ -680,8 +680,8 @@ static size_t audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
if (nb_live1) {
size_t live = smin;
- if (audio_bug(__func__, live > hw->samples)) {
- dolog("live=%zu hw->samples=%zu\n", live, hw->samples);
+ if (audio_bug(__func__, live > hw->mix_buf->size)) {
+ dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size);
return 0;
}
return live;
@@ -701,11 +701,11 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
return size;
}
- hwsamples = sw->hw->samples;
+ hwsamples = sw->hw->mix_buf->size;
live = sw->total_hw_samples_mixed;
if (audio_bug(__func__, live > hwsamples)) {
- dolog("live=%zu hw->samples=%zu\n", live, hwsamples);
+ dolog("live=%zu hw->mix_buf->size=%zu\n", live, hwsamples);
return 0;
}
@@ -716,7 +716,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
return 0;
}
- wpos = (sw->hw->rpos + live) % hwsamples;
+ wpos = (sw->hw->mix_buf->pos + live) % hwsamples;
samples = size >> sw->info.shift;
dead = hwsamples - live;
@@ -742,7 +742,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
st_rate_flow_mix (
sw->rate,
sw->buf + pos,
- sw->hw->mix_buf + wpos,
+ sw->hw->mix_buf->samples + wpos,
&isamp,
&osamp
);
@@ -870,7 +870,7 @@ size_t AUD_read(SWVoiceIn *sw, void *buf, size_t size)
int AUD_get_buffer_size_out (SWVoiceOut *sw)
{
- return sw->hw->samples << sw->hw->info.shift;
+ return sw->hw->mix_buf->size << sw->hw->info.shift;
}
void AUD_set_active_out (SWVoiceOut *sw, int on)
@@ -971,8 +971,9 @@ static size_t audio_get_avail (SWVoiceIn *sw)
}
live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
- if (audio_bug(__func__, live > sw->hw->samples)) {
- dolog("live=%zu sw->hw->samples=%zu\n", live, sw->hw->samples);
+ if (audio_bug(__func__, live > sw->hw->conv_buf->size)) {
+ dolog("live=%zu sw->hw->conv_buf->size=%zu\n", live,
+ sw->hw->conv_buf->size);
return 0;
}
@@ -995,12 +996,13 @@ static size_t audio_get_free(SWVoiceOut *sw)
live = sw->total_hw_samples_mixed;
- if (audio_bug(__func__, live > sw->hw->samples)) {
- dolog("live=%zu sw->hw->samples=%zu\n", live, sw->hw->samples);
+ if (audio_bug(__func__, live > sw->hw->mix_buf->size)) {
+ dolog("live=%zu sw->hw->mix_buf->size=%zu\n", live,
+ sw->hw->mix_buf->size);
return 0;
}
- dead = sw->hw->samples - live;
+ dead = sw->hw->mix_buf->size - live;
#ifdef DEBUG_OUT
dolog ("%s: get_free live %d dead %d ret %" PRId64 "\n",
@@ -1025,12 +1027,12 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
n = samples;
while (n) {
- size_t till_end_of_hw = hw->samples - rpos2;
+ size_t till_end_of_hw = hw->mix_buf->size - rpos2;
size_t to_write = MIN(till_end_of_hw, n);
size_t bytes = to_write << hw->info.shift;
size_t written;
- sw->buf = hw->mix_buf + rpos2;
+ sw->buf = hw->mix_buf->samples + rpos2;
written = audio_pcm_sw_write (sw, NULL, bytes);
if (written - bytes) {
dolog("Could not mix %zu bytes into a capture "
@@ -1039,14 +1041,14 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
break;
}
n -= to_write;
- rpos2 = (rpos2 + to_write) % hw->samples;
+ rpos2 = (rpos2 + to_write) % hw->mix_buf->size;
}
}
}
- n = MIN(samples, hw->samples - rpos);
- mixeng_clear(hw->mix_buf + rpos, n);
- mixeng_clear(hw->mix_buf, samples - n);
+ n = MIN(samples, hw->mix_buf->size - rpos);
+ mixeng_clear(hw->mix_buf->samples + rpos, n);
+ mixeng_clear(hw->mix_buf->samples, samples - n);
}
static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
@@ -1064,7 +1066,7 @@ static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
live -= proc;
clipped += proc;
- hw->rpos = (hw->rpos + proc) % hw->samples;
+ hw->mix_buf->pos = (hw->mix_buf->pos + proc) % hw->mix_buf->size;
if (proc == 0 || proc < decr) {
break;
@@ -1088,8 +1090,8 @@ static void audio_run_out (AudioState *s)
live = 0;
}
- if (audio_bug(__func__, live > hw->samples)) {
- dolog ("live=%zu hw->samples=%zu\n", live, hw->samples);
+ if (audio_bug(__func__, live > hw->mix_buf->size)) {
+ dolog ("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size);
continue;
}
@@ -1120,13 +1122,13 @@ static void audio_run_out (AudioState *s)
continue;
}
- prev_rpos = hw->rpos;
+ prev_rpos = hw->mix_buf->pos;
played = audio_pcm_hw_run_out(hw, live);
replay_audio_out(&played);
- if (audio_bug(__func__, hw->rpos >= hw->samples)) {
- dolog("hw->rpos=%zu hw->samples=%zu played=%zu\n",
- hw->rpos, hw->samples, played);
- hw->rpos = 0;
+ if (audio_bug(__func__, hw->mix_buf->pos >= hw->mix_buf->size)) {
+ dolog("hw->mix_buf->pos=%zu hw->mix_buf->size=%zu played=%zu\n",
+ hw->mix_buf->pos, hw->mix_buf->size, played);
+ hw->mix_buf->pos = 0;
}
#ifdef DEBUG_OUT
@@ -1183,6 +1185,7 @@ static void audio_run_out (AudioState *s)
static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
{
size_t conv = 0;
+ STSampleBuffer *conv_buf = hw->conv_buf;
while (samples) {
size_t proc;
@@ -1196,10 +1199,10 @@ static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
}
proc = MIN(size >> hw->info.shift,
- hw->samples - hw->wpos);
+ conv_buf->size - conv_buf->pos);
- hw->conv(hw->conv_buf + hw->wpos, buf, proc);
- hw->wpos = (hw->wpos + proc) % hw->samples;
+ hw->conv(conv_buf->samples + conv_buf->pos, buf, proc);
+ conv_buf->pos = (conv_buf->pos + proc) % conv_buf->size;
samples -= proc;
conv += proc;
@@ -1219,9 +1222,10 @@ static void audio_run_in (AudioState *s)
if (replay_mode != REPLAY_MODE_PLAY) {
captured = audio_pcm_hw_run_in(
- hw, hw->samples - audio_pcm_hw_get_live_in(hw));
+ hw, hw->conv_buf->size - audio_pcm_hw_get_live_in(hw));
}
- replay_audio_in(&captured, hw->conv_buf, &hw->wpos, hw->samples);
+ replay_audio_in(&captured, hw->conv_buf->samples, &hw->conv_buf->pos,
+ hw->conv_buf->size);
min = audio_pcm_hw_find_min_in (hw);
hw->total_samples_captured += captured - min;
@@ -1252,14 +1256,14 @@ static void audio_run_capture (AudioState *s)
SWVoiceOut *sw;
captured = live = audio_pcm_hw_get_live_out (hw, NULL);
- rpos = hw->rpos;
+ rpos = hw->mix_buf->pos;
while (live) {
- size_t left = hw->samples - rpos;
+ size_t left = hw->mix_buf->size - rpos;
size_t to_capture = MIN(live, left);
struct st_sample *src;
struct capture_callback *cb;
- src = hw->mix_buf + rpos;
+ src = hw->mix_buf->samples + rpos;
hw->clip (cap->buf, src, to_capture);
mixeng_clear (src, to_capture);
@@ -1267,10 +1271,10 @@ static void audio_run_capture (AudioState *s)
cb->ops.capture (cb->opaque, cap->buf,
to_capture << hw->info.shift);
}
- rpos = (rpos + to_capture) % hw->samples;
+ rpos = (rpos + to_capture) % hw->mix_buf->size;
live -= to_capture;
}
- hw->rpos = rpos;
+ hw->mix_buf->pos = rpos;
for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
if (!sw->active && sw->empty) {
@@ -1318,7 +1322,7 @@ void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size)
ssize_t start;
if (unlikely(!hw->buf_emul)) {
- size_t calc_size = hw->samples << hw->info.shift;
+ size_t calc_size = hw->conv_buf->size << hw->info.shift;
hw->buf_emul = g_malloc(calc_size);
hw->size_emul = calc_size;
hw->pos_emul = hw->pending_emul = 0;
@@ -1354,7 +1358,7 @@ void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size)
void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size)
{
if (unlikely(!hw->buf_emul)) {
- size_t calc_size = hw->samples << hw->info.shift;
+ size_t calc_size = hw->mix_buf->size << hw->info.shift;
hw->buf_emul = g_malloc(calc_size);
hw->size_emul = calc_size;
@@ -1740,21 +1744,16 @@ CaptureVoiceOut *AUD_add_capture(
/* XXX find a more elegant way */
hw->samples = 4096 * 4;
- hw->mix_buf = audio_calloc(__func__, hw->samples,
- sizeof(struct st_sample));
- if (!hw->mix_buf) {
- dolog("Could not allocate capture mix buffer (%zu samples)\n",
- hw->samples);
- goto err2;
- }
+ audio_pcm_hw_alloc_resources_out(hw);
audio_pcm_init_info (&hw->info, as);
- cap->buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
+ cap->buf = audio_calloc(__func__, hw->mix_buf->size,
+ 1 << hw->info.shift);
if (!cap->buf) {
dolog ("Could not allocate capture buffer "
"(%zu samples, each %d bytes)\n",
- hw->samples, 1 << hw->info.shift);
+ hw->mix_buf->size, 1 << hw->info.shift);
goto err3;
}
@@ -1774,7 +1773,6 @@ CaptureVoiceOut *AUD_add_capture(
err3:
g_free (cap->hw.mix_buf);
- err2:
g_free (cap);
err1:
g_free (cb);
diff --git a/audio/audio_int.h b/audio/audio_int.h
index cca7efaa05..a5add3c2b8 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -51,6 +51,11 @@ struct audio_pcm_info {
typedef struct SWVoiceCap SWVoiceCap;
+typedef struct STSampleBuffer {
+ size_t pos, size;
+ st_sample samples[];
+} STSampleBuffer;
+
typedef struct HWVoiceOut {
AudioState *s;
int enabled;
@@ -59,11 +64,9 @@ typedef struct HWVoiceOut {
struct audio_pcm_info info;
f_sample *clip;
-
- size_t rpos;
uint64_t ts_helper;
- struct st_sample *mix_buf;
+ STSampleBuffer *mix_buf;
void *buf_emul;
size_t pos_emul, pending_emul, size_emul;
@@ -83,11 +86,10 @@ typedef struct HWVoiceIn {
t_sample *conv;
- size_t wpos;
size_t total_samples_captured;
uint64_t ts_helper;
- struct st_sample *conv_buf;
+ STSampleBuffer *conv_buf;
void *buf_emul;
size_t pos_emul, pending_emul, size_emul;
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 33cdb66d7f..83ffc62183 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -76,16 +76,15 @@ static void glue (audio_pcm_hw_free_resources_, TYPE) (HW *hw)
HWBUF = NULL;
}
-static int glue (audio_pcm_hw_alloc_resources_, TYPE) (HW *hw)
+static void glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw)
{
- HWBUF = audio_calloc(__func__, hw->samples, sizeof(struct st_sample));
- if (!HWBUF) {
- dolog("Could not allocate " NAME " buffer (%zu samples)\n",
- hw->samples);
- return -1;
+ size_t samples = hw->samples;
+ if (audio_bug(__func__, samples == 0)) {
+ dolog("Attempted to allocate empty buffer\n");
}
- return 0;
+ HWBUF = g_malloc0(sizeof(STSampleBuffer) + sizeof(st_sample) * samples);
+ HWBUF->size = samples;
}
static void glue (audio_pcm_sw_free_resources_, TYPE) (SW *sw)
@@ -104,7 +103,7 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw)
{
int samples;
- samples = ((int64_t) sw->hw->samples << 32) / sw->ratio;
+ samples = ((int64_t) sw->HWBUF->size << 32) / sw->ratio;
sw->buf = audio_calloc(__func__, samples, sizeof(struct st_sample));
if (!sw->buf) {
@@ -280,9 +279,7 @@ static HW *glue(audio_pcm_hw_add_new_, TYPE)(AudioState *s,
[hw->info.swap_endianness]
[audio_bits_to_index (hw->info.bits)];
- if (glue (audio_pcm_hw_alloc_resources_, TYPE) (hw)) {
- goto err1;
- }
+ glue(audio_pcm_hw_alloc_resources_, TYPE)(hw);
QLIST_INSERT_HEAD (&s->glue (hw_head_, TYPE), hw, entries);
glue (s->nb_hw_voices_, TYPE) -= 1;
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index f1b57c87b4..08fe047f91 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -585,7 +585,7 @@ static int oss_ctl_out (HWVoiceOut *hw, int cmd, ...)
return 0;
}
- audio_pcm_info_clear_buf(&hw->info, hw->buf_emul, hw->samples);
+ audio_pcm_info_clear_buf(&hw->info, hw->buf_emul, hw->mix_buf->size);
trig = PCM_ENABLE_OUTPUT;
if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
oss_logerr (
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 40/52] audio: remove hw->samples, buffer_size_in/out pcm_ops
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (38 preceding siblings ...)
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 39/52] audio: unify input and output mixeng buffer management Kővágó, Zoltán
@ 2018-12-23 20:52 ` Kővágó, Zoltán
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 41/52] audio: common rate control code for timer based outputs Kővágó, Zoltán
` (10 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:52 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
This patch removes the samples member from HWVoiceIn and HWVoiceOut.
Backends can specify buffer size via the newly added buffer_size_in and
buffer_size_out functions in audio_pcm_ops. They are optional, if not
defined qemu will fall back to some built-in constant.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
Not sure if this is necessary. At first it seemed like a good idea to
have a function so that backends can compute the size on demand when
needed and things like that, but currently it's just a burden. The only
good feature is that it allows a backend to not define a function and
let the audio subsystem choose a default value, but the same could be
achieved by specifying that hw->samples = 0 means use a default value.
So if you guys agree, I'll remove this patch. Maybe add an -audiodev
parameter to change it, overriding whatever the backends supplies.
---
audio/alsaaudio.c | 20 ++++++++++++++++++--
audio/audio.c | 2 --
audio/audio_int.h | 5 +++--
audio/audio_template.h | 24 +++++++++++++++---------
audio/coreaudio.c | 10 +++++++++-
audio/dsound_template.h | 8 +++++++-
audio/dsoundaudio.c | 4 ++++
audio/noaudio.c | 2 --
audio/ossaudio.c | 22 +++++++++++++++++++---
audio/paaudio.c | 21 +++++++++++++++++----
audio/sdlaudio.c | 10 +++++++++-
audio/spiceaudio.c | 14 ++++++++++++--
audio/wavaudio.c | 1 -
13 files changed, 113 insertions(+), 30 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 56271b1174..672803e5c2 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -45,6 +45,7 @@ struct pollhlp {
typedef struct ALSAVoiceOut {
HWVoiceOut hw;
snd_pcm_t *handle;
+ size_t samples;
struct pollhlp pollhlp;
Audiodev *dev;
} ALSAVoiceOut;
@@ -52,6 +53,7 @@ typedef struct ALSAVoiceOut {
typedef struct ALSAVoiceIn {
HWVoiceIn hw;
snd_pcm_t *handle;
+ size_t samples;
struct pollhlp pollhlp;
Audiodev *dev;
} ALSAVoiceIn;
@@ -696,7 +698,7 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
obt_as.endianness = obt.endianness;
audio_pcm_init_info (&hw->info, &obt_as);
- hw->samples = obt.samples;
+ alsa->samples = obt.samples;
alsa->pollhlp.s = hw->s;
alsa->handle = handle;
@@ -704,6 +706,12 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
return 0;
}
+static size_t alsa_buffer_size_out(HWVoiceOut *hw)
+{
+ ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+ return alsa->samples;
+}
+
#define VOICE_CTL_PAUSE 0
#define VOICE_CTL_PREPARE 1
#define VOICE_CTL_START 2
@@ -790,7 +798,7 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
obt_as.endianness = obt.endianness;
audio_pcm_init_info (&hw->info, &obt_as);
- hw->samples = obt.samples;
+ alsa->samples = obt.samples;
alsa->pollhlp.s = hw->s;
alsa->handle = handle;
@@ -798,6 +806,12 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
return 0;
}
+static size_t alsa_buffer_size_in(HWVoiceIn *hw)
+{
+ ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+ return alsa->samples;
+}
+
static void alsa_fini_in (HWVoiceIn *hw)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
@@ -915,11 +929,13 @@ static void alsa_audio_fini (void *opaque)
static struct audio_pcm_ops alsa_pcm_ops = {
.init_out = alsa_init_out,
.fini_out = alsa_fini_out,
+ .buffer_size_out = alsa_buffer_size_out,
.write = alsa_write,
.ctl_out = alsa_ctl_out,
.init_in = alsa_init_in,
.fini_in = alsa_fini_in,
+ .buffer_size_in = alsa_buffer_size_in,
.read = alsa_read,
.ctl_in = alsa_ctl_in,
};
diff --git a/audio/audio.c b/audio/audio.c
index f195d8eb95..7db183b357 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -1742,8 +1742,6 @@ CaptureVoiceOut *AUD_add_capture(
QLIST_INIT (&hw->sw_head);
QLIST_INIT (&cap->cb_head);
- /* XXX find a more elegant way */
- hw->samples = 4096 * 4;
audio_pcm_hw_alloc_resources_out(hw);
audio_pcm_init_info (&hw->info, as);
diff --git a/audio/audio_int.h b/audio/audio_int.h
index a5add3c2b8..598038d999 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -70,7 +70,6 @@ typedef struct HWVoiceOut {
void *buf_emul;
size_t pos_emul, pending_emul, size_emul;
- size_t samples;
QLIST_HEAD (sw_out_listhead, SWVoiceOut) sw_head;
QLIST_HEAD (sw_cap_listhead, SWVoiceCap) cap_head;
int ctl_caps;
@@ -93,7 +92,6 @@ typedef struct HWVoiceIn {
void *buf_emul;
size_t pos_emul, pending_emul, size_emul;
- size_t samples;
QLIST_HEAD (sw_in_listhead, SWVoiceIn) sw_head;
int ctl_caps;
struct audio_pcm_ops *pcm_ops;
@@ -155,6 +153,8 @@ struct audio_pcm_ops {
int (*init_out)(HWVoiceOut *hw, audsettings *as, void *drv_opaque);
void (*fini_out)(HWVoiceOut *hw);
size_t (*write) (HWVoiceOut *hw, void *buf, size_t size);
+ /* get the optimal buffer size in samples; optional */
+ size_t (*buffer_size_out)(HWVoiceOut *hw);
/*
* get a buffer that after later can be passed to put_buffer_out; optional
* returns the buffer, and writes it's size to size (in bytes)
@@ -172,6 +172,7 @@ struct audio_pcm_ops {
int (*init_in) (HWVoiceIn *hw, audsettings *as, void *drv_opaque);
void (*fini_in) (HWVoiceIn *hw);
size_t (*read) (HWVoiceIn *hw, void *buf, size_t size);
+ size_t (*buffer_size_in)(HWVoiceIn *hw);
void *(*get_buffer_in)(HWVoiceIn *hw, size_t *size);
void (*put_buffer_in)(HWVoiceIn *hw, void *buf, size_t size);
int (*ctl_in) (HWVoiceIn *hw, int cmd, ...);
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 83ffc62183..07ce9ce51f 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -78,7 +78,20 @@ static void glue (audio_pcm_hw_free_resources_, TYPE) (HW *hw)
static void glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw)
{
- size_t samples = hw->samples;
+ size_t samples;
+ if (!hw->pcm_ops) {
+ /*
+ * We should only end up here when using wavcapture hmp command (and not
+ * the wavcapture audio backend).
+ * It needs a lot of samples, otherwise you'll end up with "Could not
+ * mix X bytes into a capture buffer" warnings and a garbled capture.
+ */
+ samples = 4096 * 4;
+ } else if (hw->pcm_ops->glue(buffer_size_, TYPE)) {
+ samples = hw->pcm_ops->glue(buffer_size_, TYPE)(hw);
+ } else {
+ samples = 1024; /* todo better default */
+ }
if (audio_bug(__func__, samples == 0)) {
dolog("Attempted to allocate empty buffer\n");
}
@@ -264,11 +277,6 @@ static HW *glue(audio_pcm_hw_add_new_, TYPE)(AudioState *s,
goto err0;
}
- if (audio_bug(__func__, hw->samples <= 0)) {
- dolog("hw->samples=%zd\n", hw->samples);
- goto err1;
- }
-
#ifdef DAC
hw->clip = mixeng_clip
#else
@@ -288,9 +296,7 @@ static HW *glue(audio_pcm_hw_add_new_, TYPE)(AudioState *s,
#endif
return hw;
- err1:
- glue (hw->pcm_ops->fini_, TYPE) (hw);
- err0:
+err0:
g_free (hw);
return NULL;
}
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index a532e862dd..f4210d5784 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -43,6 +43,7 @@ typedef struct coreaudioVoiceOut {
UInt32 audioDevicePropertyBufferFrameSize;
AudioStreamBasicDescription outputStreamBasicDescription;
AudioDeviceIOProcID ioprocid;
+ size_t samples;
} coreaudioVoiceOut;
#if MAC_OS_X_VERSION_MAX_ALLOWED >= MAC_OS_X_VERSION_10_6
@@ -557,7 +558,7 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
"Could not get device buffer frame size\n");
return -1;
}
- hw->samples = (pdo->has_buffer_count ? pdo->buffer_count : 4) *
+ core->samples = (pdo->has_buffer_count ? pdo->buffer_count : 4) *
core->audioDevicePropertyBufferFrameSize;
/* get StreamFormat */
@@ -617,6 +618,12 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
return 0;
}
+static size_t coreaudio_buffer_size_out(HWVoiceOut *hw)
+{
+ coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
+ return core->samples;
+}
+
static void coreaudio_fini_out (HWVoiceOut *hw)
{
OSStatus status;
@@ -693,6 +700,7 @@ static struct audio_pcm_ops coreaudio_pcm_ops = {
.init_out = coreaudio_init_out,
.fini_out = coreaudio_fini_out,
.write = coreaudio_write,
+ .buffer_size_out = coreaudio_buffer_size_out,
.get_buffer_out = coreaudio_get_buffer_out,
.put_buffer_out = coreaudio_put_buffer_out_nowrite,
.ctl_out = coreaudio_ctl_out
diff --git a/audio/dsound_template.h b/audio/dsound_template.h
index ff5a1f85fd..6a10b6751b 100644
--- a/audio/dsound_template.h
+++ b/audio/dsound_template.h
@@ -254,7 +254,7 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
);
}
hw->size_emul = bc.dwBufferBytes;
- hw->samples = bc.dwBufferBytes >> hw->info.shift;
+ ds->samples = bc.dwBufferBytes >> hw->info.shift;
ds->s = s;
#ifdef DEBUG_DSOUND
@@ -268,6 +268,12 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
return -1;
}
+static size_t glue(dsound_buffer_size_, TYPE)(HWVOICE *hw)
+{
+ DSOUNDVOICE *ds = (DSOUNDVOICE *) hw;
+ return ds->samples;
+}
+
#undef NAME
#undef NAME2
#undef TYPE
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index 7b3266aaf3..be6b8d8889 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -54,12 +54,14 @@ typedef struct {
HWVoiceOut hw;
LPDIRECTSOUNDBUFFER dsound_buffer;
dsound *s;
+ size_t samples;
} DSoundVoiceOut;
typedef struct {
HWVoiceIn hw;
LPDIRECTSOUNDCAPTUREBUFFER dsound_capture_buffer;
dsound *s;
+ size_t samples;
} DSoundVoiceIn;
static void dsound_log_hresult (HRESULT hr)
@@ -672,6 +674,7 @@ static struct audio_pcm_ops dsound_pcm_ops = {
.init_out = dsound_init_out,
.fini_out = dsound_fini_out,
.write = audio_generic_write,
+ .buffer_size_out = dsound_buffer_size_out,
.get_buffer_out = dsound_get_buffer_out,
.put_buffer_out = dsound_put_buffer_out,
.ctl_out = dsound_ctl_out,
@@ -679,6 +682,7 @@ static struct audio_pcm_ops dsound_pcm_ops = {
.init_in = dsound_init_in,
.fini_in = dsound_fini_in,
.read = audio_generic_read,
+ .buffer_size_in = dsound_buffer_size_in,
.get_buffer_in = dsound_get_buffer_in,
.put_buffer_in = dsound_put_buffer_in,
.ctl_in = dsound_ctl_in
diff --git a/audio/noaudio.c b/audio/noaudio.c
index 6a3a1c418b..9b1dfb551d 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -58,7 +58,6 @@ static size_t no_write(HWVoiceOut *hw, void *buf, size_t len)
static int no_init_out(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque)
{
audio_pcm_init_info (&hw->info, as);
- hw->samples = 1024;
return 0;
}
@@ -77,7 +76,6 @@ static int no_ctl_out (HWVoiceOut *hw, int cmd, ...)
static int no_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
{
audio_pcm_init_info (&hw->info, as);
- hw->samples = 1024;
return 0;
}
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 08fe047f91..bc34e12de4 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -44,6 +44,7 @@ typedef struct OSSVoiceOut {
int nfrags;
int fragsize;
int mmapped;
+ size_t samples;
Audiodev *dev;
} OSSVoiceOut;
@@ -52,6 +53,7 @@ typedef struct OSSVoiceIn {
int fd;
int nfrags;
int fragsize;
+ size_t samples;
Audiodev *dev;
} OSSVoiceIn;
@@ -511,11 +513,11 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
obt.nfrags * obt.fragsize, hw->info.align + 1);
}
- hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
+ oss->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
oss->mmapped = 0;
if (oopts->has_try_mmap && oopts->try_mmap) {
- hw->size_emul = hw->samples << hw->info.shift;
+ hw->size_emul = oss->samples << hw->info.shift;
hw->buf_emul = mmap (
NULL,
hw->size_emul,
@@ -563,6 +565,12 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
return 0;
}
+static size_t oss_buffer_size_out(HWVoiceOut *hw)
+{
+ OSSVoiceOut *oss = (OSSVoiceOut *) hw;
+ return oss->samples;
+}
+
static int oss_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
int trig;
@@ -658,13 +666,19 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
obt.nfrags * obt.fragsize, hw->info.align + 1);
}
- hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
+ oss->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
oss->fd = fd;
oss->dev = dev;
return 0;
}
+static size_t oss_buffer_size_in(HWVoiceIn *hw)
+{
+ OSSVoiceIn *oss = (OSSVoiceIn *) hw;
+ return oss->samples;
+}
+
static void oss_fini_in (HWVoiceIn *hw)
{
OSSVoiceIn *oss = (OSSVoiceIn *) hw;
@@ -753,6 +767,7 @@ static struct audio_pcm_ops oss_pcm_ops = {
.init_out = oss_init_out,
.fini_out = oss_fini_out,
.write = oss_write,
+ .buffer_size_out = oss_buffer_size_out,
.get_buffer_out = oss_get_buffer_out,
.put_buffer_out = oss_put_buffer_out,
.ctl_out = oss_ctl_out,
@@ -760,6 +775,7 @@ static struct audio_pcm_ops oss_pcm_ops = {
.init_in = oss_init_in,
.fini_in = oss_fini_in,
.read = oss_read,
+ .buffer_size_in = oss_buffer_size_in,
.ctl_in = oss_ctl_in
};
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 392225c875..7cab3cff97 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -355,8 +355,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
}
audio_pcm_init_info (&hw->info, &obt_as);
- hw->samples = pa->samples = audio_buffer_samples(g->dev->out, &obt_as,
- 46440);
+ pa->samples = audio_buffer_samples(g->dev->out, &obt_as, 46440);
return 0;
@@ -364,6 +363,13 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
return -1;
}
+static size_t qpa_buffer_size_out(HWVoiceOut *hw)
+{
+ PAVoiceOut *pa = (PAVoiceOut *) hw;
+ return pa->samples;
+}
+
+
static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
{
int error;
@@ -397,8 +403,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
}
audio_pcm_init_info (&hw->info, &obt_as);
- hw->samples = pa->samples = audio_buffer_samples(g->dev->in, &obt_as,
- 46440);
+ pa->samples = audio_buffer_samples(g->dev->in, &obt_as, 46440);
return 0;
@@ -406,6 +411,12 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
return -1;
}
+static size_t qpa_buffer_size_in(HWVoiceIn *hw)
+{
+ PAVoiceIn *pa = (PAVoiceIn *) hw;
+ return pa->samples;
+}
+
static void qpa_simple_disconnect(PAConnection *c, pa_stream *stream)
{
int err;
@@ -683,11 +694,13 @@ static struct audio_pcm_ops qpa_pcm_ops = {
.init_out = qpa_init_out,
.fini_out = qpa_fini_out,
.write = qpa_write,
+ .buffer_size_out = qpa_buffer_size_out,
.ctl_out = qpa_ctl_out,
.init_in = qpa_init_in,
.fini_in = qpa_fini_in,
.read = qpa_read,
+ .buffer_size_in = qpa_buffer_size_in,
.ctl_in = qpa_ctl_in
};
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index 685cbc83b8..4df35ce31a 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -42,6 +42,7 @@
typedef struct SDLVoiceOut {
HWVoiceOut hw;
+ size_t samples;
} SDLVoiceOut;
static struct SDLAudioState {
@@ -363,7 +364,7 @@ static int sdl_init_out(HWVoiceOut *hw, struct audsettings *as,
obt_as.endianness = endianness;
audio_pcm_init_info (&hw->info, &obt_as);
- hw->samples = obt.samples;
+ sdl->samples = obt.samples;
s->initialized = 1;
s->exit = 0;
@@ -371,6 +372,12 @@ static int sdl_init_out(HWVoiceOut *hw, struct audsettings *as,
return 0;
}
+static size_t sdl_buffer_size_out(HWVoiceOut *hw)
+{
+ SDLVoiceOut *sdl = (SDLVoiceOut *) hw;
+ return sdl->samples;
+}
+
static int sdl_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
(void) hw;
@@ -439,6 +446,7 @@ static struct audio_pcm_ops sdl_pcm_ops = {
.init_out = sdl_init_out,
.fini_out = sdl_fini_out,
.write = sdl_write,
+ .buffer_size_out = sdl_buffer_size_out,
.get_buffer_out = sdl_get_buffer_out,
.put_buffer_out = sdl_put_buffer_out_nowrite,
.ctl_out = sdl_ctl_out,
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index d1605d3939..709245e453 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -133,7 +133,6 @@ static int line_out_init(HWVoiceOut *hw, struct audsettings *as,
settings.endianness = AUDIO_HOST_ENDIANNESS;
audio_pcm_init_info (&hw->info, &settings);
- hw->samples = LINE_OUT_SAMPLES;
out->active = 0;
out->sin.base.sif = &playback_sif.base;
@@ -144,6 +143,11 @@ static int line_out_init(HWVoiceOut *hw, struct audsettings *as,
return 0;
}
+static size_t line_out_buffer_size(HWVoiceOut *hw)
+{
+ return LINE_OUT_SAMPLES;
+}
+
static void line_out_fini (HWVoiceOut *hw)
{
SpiceVoiceOut *out = container_of (hw, SpiceVoiceOut, hw);
@@ -248,7 +252,6 @@ static int line_in_init(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
settings.endianness = AUDIO_HOST_ENDIANNESS;
audio_pcm_init_info (&hw->info, &settings);
- hw->samples = LINE_IN_SAMPLES;
in->active = 0;
in->sin.base.sif = &record_sif.base;
@@ -259,6 +262,11 @@ static int line_in_init(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
return 0;
}
+static size_t line_in_buffer_size(HWVoiceIn *hw)
+{
+ return LINE_IN_SAMPLES;
+}
+
static void line_in_fini (HWVoiceIn *hw)
{
SpiceVoiceIn *in = container_of (hw, SpiceVoiceIn, hw);
@@ -329,6 +337,7 @@ static struct audio_pcm_ops audio_callbacks = {
.init_out = line_out_init,
.fini_out = line_out_fini,
.write = audio_generic_write,
+ .buffer_size_out = line_out_buffer_size,
.get_buffer_out = line_out_get_buffer,
.put_buffer_out = line_out_put_buffer,
.ctl_out = line_out_ctl,
@@ -336,6 +345,7 @@ static struct audio_pcm_ops audio_callbacks = {
.init_in = line_in_init,
.fini_in = line_in_fini,
.read = line_in_read,
+ .buffer_size_in = line_in_buffer_size,
.ctl_in = line_in_ctl,
};
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 31db03aadb..0a0e76d2d9 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -110,7 +110,6 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
wav_as.endianness = 0;
audio_pcm_init_info (&hw->info, &wav_as);
- hw->samples = 1024;
le_store (hdr + 22, hw->info.nchannels, 2);
le_store (hdr + 24, hw->info.freq, 4);
le_store (hdr + 28, hw->info.freq << (bits16 + stereo), 4);
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 41/52] audio: common rate control code for timer based outputs
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (39 preceding siblings ...)
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 40/52] audio: remove hw->samples, buffer_size_in/out pcm_ops Kővágó, Zoltán
@ 2018-12-23 20:52 ` Kővágó, Zoltán
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 42/52] audio: split ctl_* functions into enable_* and volume_* Kővágó, Zoltán
` (9 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:52 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
This commit removes the ad-hoc rate-limiting code from noaudio and
wavaudio, and replaces them with a (slightly modified) code from
spiceaudio. This way multiple write calls (for example when the
circular buffer wraps around) do not cause problems.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio.c | 30 ++++++++++++++++++++++++++++
audio/audio_int.h | 9 +++++++++
audio/noaudio.c | 49 +++++++++++++++++++++-------------------------
audio/spiceaudio.c | 49 +++++++---------------------------------------
audio/wavaudio.c | 21 ++++++++++----------
5 files changed, 78 insertions(+), 80 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index 7db183b357..f89bce3a34 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -2024,3 +2024,33 @@ const char *audio_get_id(QEMUSoundCard *card)
return "";
}
}
+
+void audio_rate_start(RateCtl *rate)
+{
+ memset(rate, 0, sizeof(RateCtl));
+ rate->start_ticks = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
+}
+
+size_t audio_rate_get_bytes(struct audio_pcm_info *info, RateCtl *rate,
+ size_t bytes_avail)
+{
+ int64_t now;
+ int64_t ticks;
+ int64_t bytes;
+ int64_t samples;
+ size_t ret;
+
+ now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
+ ticks = now - rate->start_ticks;
+ bytes = muldiv64(ticks, info->bytes_per_second, NANOSECONDS_PER_SECOND);
+ samples = (bytes - rate->bytes_sent) >> info->shift;
+ if (samples < 0 || samples > 65536) {
+ AUD_log(NULL, "Resetting rate control (%" PRId64 " samples)", samples);
+ audio_rate_start(rate);
+ samples = 0;
+ }
+
+ ret = MIN(samples << info->shift, bytes_avail);
+ rate->bytes_sent += ret;
+ return ret;
+}
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 598038d999..0f0ea3e040 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -242,6 +242,15 @@ void *audio_calloc (const char *funcname, int nmemb, size_t size);
void audio_run(AudioState *s, const char *msg);
+typedef struct RateCtl {
+ int64_t start_ticks;
+ int64_t bytes_sent;
+} RateCtl;
+
+void audio_rate_start(RateCtl *rate);
+size_t audio_rate_get_bytes(struct audio_pcm_info *info, RateCtl *rate,
+ size_t bytes_avail);
+
#define VOICE_ENABLE 1
#define VOICE_DISABLE 2
#define VOICE_VOLUME 3
diff --git a/audio/noaudio.c b/audio/noaudio.c
index 9b1dfb551d..85241612cf 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -32,32 +32,26 @@
typedef struct NoVoiceOut {
HWVoiceOut hw;
- int64_t old_ticks;
+ RateCtl rate;
} NoVoiceOut;
typedef struct NoVoiceIn {
HWVoiceIn hw;
- int64_t old_ticks;
+ RateCtl rate;
} NoVoiceIn;
static size_t no_write(HWVoiceOut *hw, void *buf, size_t len)
{
NoVoiceOut *no = (NoVoiceOut *) hw;
- int64_t now;
- int64_t ticks;
- int64_t bytes;
-
- now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
- ticks = now - no->old_ticks;
- bytes = muldiv64(ticks, hw->info.bytes_per_second, NANOSECONDS_PER_SECOND);
-
- no->old_ticks = now;
- return MIN(len, bytes);
+ return audio_rate_get_bytes(&hw->info, &no->rate, len);
}
static int no_init_out(HWVoiceOut *hw, struct audsettings *as, void *drv_opaque)
{
+ NoVoiceOut *no = (NoVoiceOut *) hw;
+
audio_pcm_init_info (&hw->info, as);
+ audio_rate_start(&no->rate);
return 0;
}
@@ -68,14 +62,20 @@ static void no_fini_out (HWVoiceOut *hw)
static int no_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
- (void) hw;
- (void) cmd;
+ NoVoiceOut *no = (NoVoiceOut *) hw;
+
+ if (cmd == VOICE_ENABLE) {
+ audio_rate_start(&no->rate);
+ }
return 0;
}
static int no_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
{
+ NoVoiceIn *no = (NoVoiceIn *) hw;
+
audio_pcm_init_info (&hw->info, as);
+ audio_rate_start(&no->rate);
return 0;
}
@@ -86,25 +86,20 @@ static void no_fini_in (HWVoiceIn *hw)
static size_t no_read(HWVoiceIn *hw, void *buf, size_t size)
{
- size_t to_clear;
NoVoiceIn *no = (NoVoiceIn *) hw;
+ int64_t bytes = audio_rate_get_bytes(&hw->info, &no->rate, size);
- int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
- int64_t ticks = now - no->old_ticks;
- int64_t bytes =
- muldiv64 (ticks, hw->info.bytes_per_second, NANOSECONDS_PER_SECOND);
-
- no->old_ticks = now;
- to_clear = MIN(bytes, size);
-
- audio_pcm_info_clear_buf(&hw->info, buf, to_clear >> hw->info.shift);
- return to_clear;
+ audio_pcm_info_clear_buf(&hw->info, buf, bytes >> hw->info.shift);
+ return bytes;
}
static int no_ctl_in (HWVoiceIn *hw, int cmd, ...)
{
- (void) hw;
- (void) cmd;
+ NoVoiceIn *no = (NoVoiceIn *) hw;
+
+ if (cmd == VOICE_ENABLE) {
+ audio_rate_start(&no->rate);
+ }
return 0;
}
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index 709245e453..2495866c82 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -40,15 +40,10 @@
#define LINE_IN_SAMPLES (256 * 4)
#endif
-typedef struct SpiceRateCtl {
- int64_t start_ticks;
- int64_t bytes_sent;
-} SpiceRateCtl;
-
typedef struct SpiceVoiceOut {
HWVoiceOut hw;
SpicePlaybackInstance sin;
- SpiceRateCtl rate;
+ RateCtl rate;
int active;
uint32_t *frame;
uint32_t fpos;
@@ -58,7 +53,7 @@ typedef struct SpiceVoiceOut {
typedef struct SpiceVoiceIn {
HWVoiceIn hw;
SpiceRecordInstance sin;
- SpiceRateCtl rate;
+ RateCtl rate;
int active;
} SpiceVoiceIn;
@@ -89,32 +84,6 @@ static void spice_audio_fini (void *opaque)
/* nothing */
}
-static void rate_start (SpiceRateCtl *rate)
-{
- memset (rate, 0, sizeof (*rate));
- rate->start_ticks = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
-}
-
-static int rate_get_samples (struct audio_pcm_info *info, SpiceRateCtl *rate)
-{
- int64_t now;
- int64_t ticks;
- int64_t bytes;
- int64_t samples;
-
- now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
- ticks = now - rate->start_ticks;
- bytes = muldiv64(ticks, info->bytes_per_second, NANOSECONDS_PER_SECOND);
- samples = (bytes - rate->bytes_sent) >> info->shift;
- if (samples < 0 || samples > 65536) {
- error_report("Resetting rate control (%" PRId64 " samples)", samples);
- rate_start(rate);
- samples = 0;
- }
- rate->bytes_sent += samples << info->shift;
- return samples;
-}
-
/* playback */
static int line_out_init(HWVoiceOut *hw, struct audsettings *as,
@@ -158,17 +127,14 @@ static void line_out_fini (HWVoiceOut *hw)
static void *line_out_get_buffer(HWVoiceOut *hw, size_t *size)
{
SpiceVoiceOut *out = container_of(hw, SpiceVoiceOut, hw);
- size_t decr;
if (!out->frame) {
spice_server_playback_get_buffer(&out->sin, &out->frame, &out->fsize);
out->fpos = 0;
}
- decr = rate_get_samples(&hw->info, &out->rate);
- decr = MIN(out->fsize - out->fpos, decr);
-
- *size = decr << hw->info.shift;
+ *size = audio_rate_get_bytes(&hw->info, &out->rate,
+ (out->fsize - out->fpos) << hw->info.shift);
return out->frame + out->fpos;
}
@@ -197,7 +163,7 @@ static int line_out_ctl (HWVoiceOut *hw, int cmd, ...)
break;
}
out->active = 1;
- rate_start (&out->rate);
+ audio_rate_start(&out->rate);
spice_server_playback_start (&out->sin);
break;
case VOICE_DISABLE:
@@ -277,8 +243,7 @@ static void line_in_fini (HWVoiceIn *hw)
static size_t line_in_read(HWVoiceIn *hw, void *buf, size_t len)
{
SpiceVoiceIn *in = container_of (hw, SpiceVoiceIn, hw);
- uint64_t delta_samp = rate_get_samples(&hw->info, &in->rate);
- uint64_t to_read = MIN(len >> 2, delta_samp);
+ uint64_t to_read = audio_rate_get_bytes(&hw->info, &in->rate, len) >> 2;
size_t ready = spice_server_record_get_samples(&in->sin, buf, to_read);
/* XXX: do we need this? */
@@ -300,7 +265,7 @@ static int line_in_ctl (HWVoiceIn *hw, int cmd, ...)
break;
}
in->active = 1;
- rate_start (&in->rate);
+ audio_rate_start(&in->rate);
spice_server_record_start (&in->sin);
break;
case VOICE_DISABLE:
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 0a0e76d2d9..f977d96b9c 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -33,21 +33,15 @@
typedef struct WAVVoiceOut {
HWVoiceOut hw;
FILE *f;
- int64_t old_ticks;
+ RateCtl rate;
int total_samples;
} WAVVoiceOut;
static size_t wav_write_out(HWVoiceOut *hw, void *buf, size_t len)
{
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
- int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
- int64_t ticks = now - wav->old_ticks;
- int64_t bytes =
- muldiv64(ticks, hw->info.bytes_per_second, NANOSECONDS_PER_SECOND);
-
- bytes = MIN(bytes, len);
- bytes = bytes >> hw->info.shift << hw->info.shift;
- wav->old_ticks = now;
+ int64_t bytes = audio_rate_get_bytes(&hw->info, &wav->rate, len);
+ assert(bytes >> hw->info.shift << hw->info.shift == bytes);
if (bytes && fwrite(buf, bytes, 1, wav->f) != 1) {
dolog("wav_write_out: fwrite of %zu bytes failed\nReaons: %s\n",
@@ -127,6 +121,8 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
strerror(errno));
return -1;
}
+
+ audio_rate_start(&wav->rate);
return 0;
}
@@ -176,8 +172,11 @@ static void wav_fini_out (HWVoiceOut *hw)
static int wav_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
- (void) hw;
- (void) cmd;
+ WAVVoiceOut *wav = (WAVVoiceOut *) hw;
+
+ if (cmd == VOICE_ENABLE) {
+ audio_rate_start(&wav->rate);
+ }
return 0;
}
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 42/52] audio: split ctl_* functions into enable_* and volume_*
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (40 preceding siblings ...)
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 41/52] audio: common rate control code for timer based outputs Kővágó, Zoltán
@ 2018-12-23 20:52 ` Kővágó, Zoltán
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 43/52] audio: add mixeng option (documentation) Kővágó, Zoltán
` (8 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:52 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
This way we no longer need vararg functions, improving compile time
error detection. Also now it's possible to check actually what commands
are supported, without needing to manually update ctl_caps.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/alsaaudio.c | 56 +++++++-----------
audio/audio.c | 45 +++++++++------
audio/audio_int.h | 15 ++---
audio/audio_template.h | 1 -
audio/coreaudio.c | 13 ++---
audio/dsoundaudio.c | 50 +++++++---------
audio/noaudio.c | 14 ++---
audio/ossaudio.c | 78 ++++++++++---------------
audio/paaudio.c | 126 ++++++++++++++++-------------------------
audio/sdlaudio.c | 17 +-----
audio/spiceaudio.c | 101 ++++++++++++++-------------------
audio/wavaudio.c | 7 +--
12 files changed, 213 insertions(+), 310 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 672803e5c2..bc097b1f31 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -745,34 +745,28 @@ static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl)
return 0;
}
-static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
+static void alsa_enable_out(HWVoiceOut *hw, bool enable)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.alsa_out;
- switch (cmd) {
- case VOICE_ENABLE:
- {
- bool poll_mode = !apdo->has_try_poll || apdo->try_poll;
+ if (enable) {
+ bool poll_mode = !apdo->has_try_poll || apdo->try_poll;
- ldebug ("enabling voice\n");
- if (poll_mode && alsa_poll_out (hw)) {
- poll_mode = 0;
- }
- hw->poll_mode = poll_mode;
- return alsa_voice_ctl (alsa->handle, "playback", VOICE_CTL_PREPARE);
+ ldebug ("enabling voice\n");
+ if (poll_mode && alsa_poll_out (hw)) {
+ poll_mode = 0;
}
-
- case VOICE_DISABLE:
+ hw->poll_mode = poll_mode;
+ alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PREPARE);
+ } else {
ldebug ("disabling voice\n");
if (hw->poll_mode) {
hw->poll_mode = 0;
alsa_fini_poll (&alsa->pollhlp);
}
- return alsa_voice_ctl (alsa->handle, "playback", VOICE_CTL_PAUSE);
+ alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PAUSE);
}
-
- return -1;
}
static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
@@ -861,35 +855,29 @@ static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
return pos;
}
-static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
+static void alsa_enable_in(HWVoiceIn *hw, bool enable)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.alsa_in;
- switch (cmd) {
- case VOICE_ENABLE:
- {
- bool poll_mode = !apdo->has_try_poll || apdo->try_poll;
+ if (enable) {
+ bool poll_mode = !apdo->has_try_poll || apdo->try_poll;
- ldebug ("enabling voice\n");
- if (poll_mode && alsa_poll_in (hw)) {
- poll_mode = 0;
- }
- hw->poll_mode = poll_mode;
-
- return alsa_voice_ctl (alsa->handle, "capture", VOICE_CTL_START);
+ ldebug ("enabling voice\n");
+ if (poll_mode && alsa_poll_in (hw)) {
+ poll_mode = 0;
}
+ hw->poll_mode = poll_mode;
- case VOICE_DISABLE:
+ alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_START);
+ } else {
ldebug ("disabling voice\n");
if (hw->poll_mode) {
hw->poll_mode = 0;
alsa_fini_poll (&alsa->pollhlp);
}
- return alsa_voice_ctl (alsa->handle, "capture", VOICE_CTL_PAUSE);
+ alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_PAUSE);
}
-
- return -1;
}
static void *alsa_audio_init(Audiodev *dev)
@@ -931,13 +919,13 @@ static struct audio_pcm_ops alsa_pcm_ops = {
.fini_out = alsa_fini_out,
.buffer_size_out = alsa_buffer_size_out,
.write = alsa_write,
- .ctl_out = alsa_ctl_out,
+ .enable_out = alsa_enable_out,
.init_in = alsa_init_in,
.fini_in = alsa_fini_in,
.buffer_size_in = alsa_buffer_size_in,
.read = alsa_read,
- .ctl_in = alsa_ctl_in,
+ .enable_in = alsa_enable_in,
};
static struct audio_driver alsa_audio_driver = {
diff --git a/audio/audio.c b/audio/audio.c
index f89bce3a34..7ba939560a 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -638,7 +638,7 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
total += isamp;
}
- if (!(hw->ctl_caps & VOICE_VOLUME_CAP)) {
+ if (!hw->pcm_ops->volume_in) {
mixeng_volume (sw->buf, ret, &sw->vol);
}
@@ -725,7 +725,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
if (swlim) {
sw->conv (sw->buf, buf, swlim);
- if (!(sw->hw->ctl_caps & VOICE_VOLUME_CAP)) {
+ if (!sw->hw->pcm_ops->volume_out) {
mixeng_volume (sw->buf, swlim, &sw->vol);
}
}
@@ -892,7 +892,9 @@ 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);
+ if (hw->pcm_ops->enable_out) {
+ hw->pcm_ops->enable_out(hw, true);
+ }
audio_reset_timer (s);
}
}
@@ -937,7 +939,9 @@ 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);
+ if (hw->pcm_ops->enable_in) {
+ hw->pcm_ops->enable_in(hw, true);
+ }
audio_reset_timer (s);
}
}
@@ -954,7 +958,9 @@ void AUD_set_active_in (SWVoiceIn *sw, int on)
if (nb_active == 1) {
hw->enabled = 0;
- hw->pcm_ops->ctl_in (hw, VOICE_DISABLE);
+ if (hw->pcm_ops->enable_in) {
+ hw->pcm_ops->enable_in(hw, false);
+ }
}
}
}
@@ -1102,7 +1108,9 @@ static void audio_run_out (AudioState *s)
#endif
hw->enabled = 0;
hw->pending_disable = 0;
- hw->pcm_ops->ctl_out (hw, VOICE_DISABLE);
+ if (hw->pcm_ops->enable_out) {
+ hw->pcm_ops->enable_out(hw, false);
+ }
for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
sc->sw.active = 0;
audio_recalc_and_notify_capture (sc->cap);
@@ -1465,15 +1473,18 @@ static void audio_vm_change_state_handler (void *opaque, int running,
AudioState *s = opaque;
HWVoiceOut *hwo = NULL;
HWVoiceIn *hwi = NULL;
- int op = running ? VOICE_ENABLE : VOICE_DISABLE;
s->vm_running = running;
while ((hwo = audio_pcm_hw_find_any_enabled_out(s, hwo))) {
- hwo->pcm_ops->ctl_out(hwo, op);
+ if (hwo->pcm_ops->enable_out) {
+ hwo->pcm_ops->enable_out(hwo, running);
+ }
}
while ((hwi = audio_pcm_hw_find_any_enabled_in(s, hwi))) {
- hwi->pcm_ops->ctl_in(hwi, op);
+ if (hwi->pcm_ops->enable_in) {
+ hwi->pcm_ops->enable_in(hwi, running);
+ }
}
audio_reset_timer (s);
}
@@ -1493,8 +1504,8 @@ static void free_audio_state(AudioState *s)
QLIST_FOREACH_SAFE(hwo, &s->hw_head_out, entries, hwon) {
SWVoiceCap *sc;
- if (hwo->enabled) {
- hwo->pcm_ops->ctl_out (hwo, VOICE_DISABLE);
+ if (hwo->enabled && hwo->pcm_ops->enable_out) {
+ hwo->pcm_ops->enable_out(hwo, false);
}
hwo->pcm_ops->fini_out (hwo);
@@ -1510,8 +1521,8 @@ static void free_audio_state(AudioState *s)
}
QLIST_FOREACH_SAFE(hwi, &s->hw_head_in, entries, hwin) {
- if (hwi->enabled) {
- hwi->pcm_ops->ctl_in (hwi, VOICE_DISABLE);
+ if (hwi->enabled && hwi->pcm_ops->enable_in) {
+ hwi->pcm_ops->enable_in(hwi, false);
}
hwi->pcm_ops->fini_in (hwi);
QLIST_REMOVE(hwi, entries);
@@ -1827,8 +1838,8 @@ void AUD_set_volume_out (SWVoiceOut *sw, int mute, uint8_t lvol, uint8_t rvol)
sw->vol.l = nominal_volume.l * lvol / 255;
sw->vol.r = nominal_volume.r * rvol / 255;
- if (hw->pcm_ops->ctl_out) {
- hw->pcm_ops->ctl_out (hw, VOICE_VOLUME, sw);
+ if (hw->pcm_ops->volume_out) {
+ hw->pcm_ops->volume_out(hw, &sw->vol);
}
}
}
@@ -1842,8 +1853,8 @@ void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
sw->vol.l = nominal_volume.l * lvol / 255;
sw->vol.r = nominal_volume.r * rvol / 255;
- if (hw->pcm_ops->ctl_in) {
- hw->pcm_ops->ctl_in (hw, VOICE_VOLUME, sw);
+ if (hw->pcm_ops->volume_in) {
+ hw->pcm_ops->volume_in(hw, &sw->vol);
}
}
}
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 0f0ea3e040..a5ead07998 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -72,7 +72,6 @@ typedef struct HWVoiceOut {
QLIST_HEAD (sw_out_listhead, SWVoiceOut) sw_head;
QLIST_HEAD (sw_cap_listhead, SWVoiceCap) cap_head;
- int ctl_caps;
struct audio_pcm_ops *pcm_ops;
QLIST_ENTRY (HWVoiceOut) entries;
} HWVoiceOut;
@@ -93,7 +92,6 @@ typedef struct HWVoiceIn {
size_t pos_emul, pending_emul, size_emul;
QLIST_HEAD (sw_in_listhead, SWVoiceIn) sw_head;
- int ctl_caps;
struct audio_pcm_ops *pcm_ops;
QLIST_ENTRY (HWVoiceIn) entries;
} HWVoiceIn;
@@ -145,7 +143,6 @@ struct audio_driver {
int max_voices_in;
int voice_size_out;
int voice_size_in;
- int ctl_caps;
QLIST_ENTRY(audio_driver) next;
};
@@ -167,7 +164,8 @@ struct audio_pcm_ops {
* size may be smaller
*/
size_t (*put_buffer_out)(HWVoiceOut *hw, void *buf, size_t size);
- int (*ctl_out) (HWVoiceOut *hw, int cmd, ...);
+ void (*enable_out)(HWVoiceOut *hw, bool enable);
+ void (*volume_out)(HWVoiceOut *hw, struct mixeng_volume *vol);
int (*init_in) (HWVoiceIn *hw, audsettings *as, void *drv_opaque);
void (*fini_in) (HWVoiceIn *hw);
@@ -175,7 +173,8 @@ struct audio_pcm_ops {
size_t (*buffer_size_in)(HWVoiceIn *hw);
void *(*get_buffer_in)(HWVoiceIn *hw, size_t *size);
void (*put_buffer_in)(HWVoiceIn *hw, void *buf, size_t size);
- int (*ctl_in) (HWVoiceIn *hw, int cmd, ...);
+ void (*enable_in)(HWVoiceIn *hw, bool enable);
+ void (*volume_in)(HWVoiceIn *hw, struct mixeng_volume *vol);
};
void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size);
@@ -251,12 +250,6 @@ void audio_rate_start(RateCtl *rate);
size_t audio_rate_get_bytes(struct audio_pcm_info *info, RateCtl *rate,
size_t bytes_avail);
-#define VOICE_ENABLE 1
-#define VOICE_DISABLE 2
-#define VOICE_VOLUME 3
-
-#define VOICE_VOLUME_CAP (1 << VOICE_VOLUME)
-
static inline size_t audio_ring_dist(size_t dst, size_t src, size_t len)
{
return (dst >= src) ? (dst - src) : (len - src + dst);
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 07ce9ce51f..9ec565fc48 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -267,7 +267,6 @@ static HW *glue(audio_pcm_hw_add_new_, TYPE)(AudioState *s,
hw->s = s;
hw->pcm_ops = drv->pcm_ops;
- hw->ctl_caps = drv->ctl_caps;
QLIST_INIT (&hw->sw_head);
#ifdef DAC
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index f4210d5784..a3cdbd87ce 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -655,13 +655,12 @@ static void coreaudio_fini_out (HWVoiceOut *hw)
}
}
-static int coreaudio_ctl_out (HWVoiceOut *hw, int cmd, ...)
+static void coreaudio_enable_out (HWVoiceOut *hw, bool enable)
{
OSStatus status;
coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
- switch (cmd) {
- case VOICE_ENABLE:
+ if (enable) {
/* start playback */
if (!isPlaying(core->outputDeviceID)) {
status = AudioDeviceStart(core->outputDeviceID, core->ioprocid);
@@ -669,9 +668,7 @@ static int coreaudio_ctl_out (HWVoiceOut *hw, int cmd, ...)
coreaudio_logerr (status, "Could not resume playback\n");
}
}
- break;
-
- case VOICE_DISABLE:
+ } else {
/* stop playback */
if (!audio_is_cleaning_up()) {
if (isPlaying(core->outputDeviceID)) {
@@ -682,9 +679,7 @@ static int coreaudio_ctl_out (HWVoiceOut *hw, int cmd, ...)
}
}
}
- break;
}
- return 0;
}
static void *coreaudio_audio_init(Audiodev *dev)
@@ -703,7 +698,7 @@ static struct audio_pcm_ops coreaudio_pcm_ops = {
.buffer_size_out = coreaudio_buffer_size_out,
.get_buffer_out = coreaudio_get_buffer_out,
.put_buffer_out = coreaudio_put_buffer_out_nowrite,
- .ctl_out = coreaudio_ctl_out
+ .enable_out = coreaudio_enable_out
};
static struct audio_driver coreaudio_audio_driver = {
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index be6b8d8889..6c10d271fe 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -363,7 +363,7 @@ static int dsound_open (dsound *s)
return 0;
}
-static int dsound_ctl_out (HWVoiceOut *hw, int cmd, ...)
+static void dsound_enable_out(HWVoiceOut *hw, bool enable)
{
HRESULT hr;
DWORD status;
@@ -373,18 +373,17 @@ static int dsound_ctl_out (HWVoiceOut *hw, int cmd, ...)
if (!dsb) {
dolog ("Attempt to control voice without a buffer\n");
- return 0;
+ return;
}
- switch (cmd) {
- case VOICE_ENABLE:
+ if (enable) {
if (dsound_get_status_out (dsb, &status, s)) {
- return -1;
+ return;
}
if (status & DSBSTATUS_PLAYING) {
dolog ("warning: Voice is already playing\n");
- return 0;
+ return;
}
dsound_clear_sample (hw, dsb, s);
@@ -392,28 +391,24 @@ static int dsound_ctl_out (HWVoiceOut *hw, int cmd, ...)
hr = IDirectSoundBuffer_Play (dsb, 0, 0, DSBPLAY_LOOPING);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not start playing buffer\n");
- return -1;
+ return;
}
- break;
-
- case VOICE_DISABLE:
+ } else {
if (dsound_get_status_out (dsb, &status, s)) {
- return -1;
+ return;
}
if (status & DSBSTATUS_PLAYING) {
hr = IDirectSoundBuffer_Stop (dsb);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not stop playing buffer\n");
- return -1;
+ return;
}
}
else {
dolog ("warning: Voice is not playing\n");
}
- break;
}
- return 0;
}
static void *dsound_get_buffer_out(HWVoiceOut *hw, size_t *size)
@@ -463,7 +458,7 @@ static size_t dsound_put_buffer_out(HWVoiceOut *hw, void *buf, size_t len)
return len;
}
-static int dsound_ctl_in (HWVoiceIn *hw, int cmd, ...)
+static void dsound_enable_in(HWVoiceIn *hw, bool enable)
{
HRESULT hr;
DWORD status;
@@ -472,18 +467,17 @@ static int dsound_ctl_in (HWVoiceIn *hw, int cmd, ...)
if (!dscb) {
dolog ("Attempt to control capture voice without a buffer\n");
- return -1;
+ return;
}
- switch (cmd) {
- case VOICE_ENABLE:
+ if (enable) {
if (dsound_get_status_in (dscb, &status)) {
- return -1;
+ return;
}
if (status & DSCBSTATUS_CAPTURING) {
dolog ("warning: Voice is already capturing\n");
- return 0;
+ return;
}
/* clear ?? */
@@ -491,28 +485,24 @@ static int dsound_ctl_in (HWVoiceIn *hw, int cmd, ...)
hr = IDirectSoundCaptureBuffer_Start (dscb, DSCBSTART_LOOPING);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not start capturing\n");
- return -1;
+ return;
}
- break;
-
- case VOICE_DISABLE:
+ } else {
if (dsound_get_status_in (dscb, &status)) {
- return -1;
+ return;
}
if (status & DSCBSTATUS_CAPTURING) {
hr = IDirectSoundCaptureBuffer_Stop (dscb);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not stop capturing\n");
- return -1;
+ return;
}
}
else {
dolog ("warning: Voice is not capturing\n");
}
- break;
}
- return 0;
}
static void *dsound_get_buffer_in(HWVoiceIn *hw, size_t *size)
@@ -677,7 +667,7 @@ static struct audio_pcm_ops dsound_pcm_ops = {
.buffer_size_out = dsound_buffer_size_out,
.get_buffer_out = dsound_get_buffer_out,
.put_buffer_out = dsound_put_buffer_out,
- .ctl_out = dsound_ctl_out,
+ .enable_out = dsound_enable_out,
.init_in = dsound_init_in,
.fini_in = dsound_fini_in,
@@ -685,7 +675,7 @@ static struct audio_pcm_ops dsound_pcm_ops = {
.buffer_size_in = dsound_buffer_size_in,
.get_buffer_in = dsound_get_buffer_in,
.put_buffer_in = dsound_put_buffer_in,
- .ctl_in = dsound_ctl_in
+ .enable_in = dsound_enable_in,
};
static struct audio_driver dsound_audio_driver = {
diff --git a/audio/noaudio.c b/audio/noaudio.c
index 85241612cf..f2938aff66 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -60,14 +60,13 @@ static void no_fini_out (HWVoiceOut *hw)
(void) hw;
}
-static int no_ctl_out (HWVoiceOut *hw, int cmd, ...)
+static void no_enable_out(HWVoiceOut *hw, bool enable)
{
NoVoiceOut *no = (NoVoiceOut *) hw;
- if (cmd == VOICE_ENABLE) {
+ if (enable) {
audio_rate_start(&no->rate);
}
- return 0;
}
static int no_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
@@ -93,14 +92,13 @@ static size_t no_read(HWVoiceIn *hw, void *buf, size_t size)
return bytes;
}
-static int no_ctl_in (HWVoiceIn *hw, int cmd, ...)
+static void no_enable_in(HWVoiceIn *hw, bool enable)
{
NoVoiceIn *no = (NoVoiceIn *) hw;
- if (cmd == VOICE_ENABLE) {
+ if (enable) {
audio_rate_start(&no->rate);
}
- return 0;
}
static void *no_audio_init (Audiodev *dev)
@@ -117,12 +115,12 @@ static struct audio_pcm_ops no_pcm_ops = {
.init_out = no_init_out,
.fini_out = no_fini_out,
.write = no_write,
- .ctl_out = no_ctl_out,
+ .enable_out = no_enable_out,
.init_in = no_init_in,
.fini_in = no_fini_in,
.read = no_read,
- .ctl_in = no_ctl_in
+ .enable_in = no_enable_in
};
static struct audio_driver no_audio_driver = {
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index bc34e12de4..cefc90fe58 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -571,59 +571,50 @@ static size_t oss_buffer_size_out(HWVoiceOut *hw)
return oss->samples;
}
-static int oss_ctl_out (HWVoiceOut *hw, int cmd, ...)
+static void oss_enable_out(HWVoiceOut *hw, bool enable)
{
int trig;
OSSVoiceOut *oss = (OSSVoiceOut *) hw;
AudiodevOssPerDirectionOptions *opdo = oss->dev->u.oss.oss_out;
- switch (cmd) {
- case VOICE_ENABLE:
- {
- bool poll_mode = !opdo->has_try_poll || opdo->try_poll;
+ if (enable) {
+ bool poll_mode = !opdo->has_try_poll || opdo->try_poll;
- ldebug ("enabling voice\n");
- if (poll_mode) {
- oss_poll_out (hw);
- poll_mode = 0;
- }
- hw->poll_mode = poll_mode;
-
- if (!oss->mmapped) {
- return 0;
- }
+ ldebug ("enabling voice\n");
+ if (poll_mode) {
+ oss_poll_out (hw);
+ poll_mode = 0;
+ }
+ hw->poll_mode = poll_mode;
- audio_pcm_info_clear_buf(&hw->info, hw->buf_emul, hw->mix_buf->size);
- trig = PCM_ENABLE_OUTPUT;
- if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
- oss_logerr (
- errno,
- "SNDCTL_DSP_SETTRIGGER PCM_ENABLE_OUTPUT failed\n"
- );
- return -1;
- }
+ if (!oss->mmapped) {
+ return;
}
- break;
- case VOICE_DISABLE:
+ audio_pcm_info_clear_buf(&hw->info, hw->buf_emul, hw->mix_buf->size);
+ trig = PCM_ENABLE_OUTPUT;
+ if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
+ oss_logerr(errno,
+ "SNDCTL_DSP_SETTRIGGER PCM_ENABLE_OUTPUT failed\n");
+ return;
+ }
+ } else {
if (hw->poll_mode) {
qemu_set_fd_handler (oss->fd, NULL, NULL, NULL);
hw->poll_mode = 0;
}
if (!oss->mmapped) {
- return 0;
+ return;
}
ldebug ("disabling voice\n");
trig = 0;
if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
oss_logerr (errno, "SNDCTL_DSP_SETTRIGGER 0 failed\n");
- return -1;
+ return;
}
- break;
}
- return 0;
}
static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
@@ -716,32 +707,25 @@ static size_t oss_read(HWVoiceIn *hw, void *buf, size_t len)
return pos;
}
-static int oss_ctl_in (HWVoiceIn *hw, int cmd, ...)
+static void oss_enable_in(HWVoiceIn *hw, bool enable)
{
OSSVoiceIn *oss = (OSSVoiceIn *) hw;
AudiodevOssPerDirectionOptions *opdo = oss->dev->u.oss.oss_out;
- switch (cmd) {
- case VOICE_ENABLE:
- {
- bool poll_mode = !opdo->has_try_poll || opdo->try_poll;
+ if (enable) {
+ bool poll_mode = !opdo->has_try_poll || opdo->try_poll;
- if (poll_mode) {
- oss_poll_in (hw);
- poll_mode = 0;
- }
- hw->poll_mode = poll_mode;
+ if (poll_mode) {
+ oss_poll_in (hw);
+ poll_mode = 0;
}
- break;
-
- case VOICE_DISABLE:
+ hw->poll_mode = poll_mode;
+ } else {
if (hw->poll_mode) {
hw->poll_mode = 0;
qemu_set_fd_handler (oss->fd, NULL, NULL, NULL);
}
- break;
}
- return 0;
}
static void *oss_audio_init(Audiodev *dev)
@@ -770,13 +754,13 @@ static struct audio_pcm_ops oss_pcm_ops = {
.buffer_size_out = oss_buffer_size_out,
.get_buffer_out = oss_get_buffer_out,
.put_buffer_out = oss_put_buffer_out,
- .ctl_out = oss_ctl_out,
+ .enable_out = oss_enable_out,
.init_in = oss_init_in,
.fini_in = oss_fini_in,
.read = oss_read,
.buffer_size_in = oss_buffer_size_in,
- .ctl_in = oss_ctl_in
+ .enable_in = oss_enable_in
};
static struct audio_driver oss_audio_driver = {
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 7cab3cff97..6565048ac7 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -458,7 +458,7 @@ static void qpa_fini_in (HWVoiceIn *hw)
}
}
-static int qpa_ctl_out (HWVoiceOut *hw, int cmd, ...)
+static void qpa_volume_out(HWVoiceOut *hw, struct mixeng_volume *vol)
{
PAVoiceOut *pa = (PAVoiceOut *) hw;
pa_operation *op;
@@ -469,48 +469,36 @@ static int qpa_ctl_out (HWVoiceOut *hw, int cmd, ...)
pa_cvolume_init (&v); /* function is present in 0.9.13+ */
#endif
- switch (cmd) {
- case VOICE_VOLUME:
- {
- SWVoiceOut *sw;
- va_list ap;
+ v.channels = 2;
+ v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->l) / UINT32_MAX;
+ v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->r) / UINT32_MAX;
- va_start (ap, cmd);
- sw = va_arg (ap, SWVoiceOut *);
- va_end (ap);
+ pa_threaded_mainloop_lock(c->mainloop);
- v.channels = 2;
- v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.l) / UINT32_MAX;
- v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.r) / UINT32_MAX;
-
- pa_threaded_mainloop_lock(c->mainloop);
-
- op = pa_context_set_sink_input_volume(c->context,
- pa_stream_get_index (pa->stream),
- &v, NULL, NULL);
- if (!op)
- qpa_logerr (pa_context_errno(c->context),
- "set_sink_input_volume() failed\n");
- else
- pa_operation_unref (op);
-
- op = pa_context_set_sink_input_mute(c->context,
- pa_stream_get_index (pa->stream),
- sw->vol.mute, NULL, NULL);
- if (!op) {
- qpa_logerr (pa_context_errno(c->context),
- "set_sink_input_mute() failed\n");
- } else {
- pa_operation_unref (op);
- }
+ op = pa_context_set_sink_input_volume(c->context,
+ pa_stream_get_index(pa->stream),
+ &v, NULL, NULL);
+ if (!op) {
+ qpa_logerr(pa_context_errno(c->context),
+ "set_sink_input_volume() failed\n");
+ } else {
+ pa_operation_unref (op);
+ }
- pa_threaded_mainloop_unlock(c->mainloop);
- }
+ op = pa_context_set_sink_input_mute(c->context,
+ pa_stream_get_index(pa->stream),
+ vol->mute, NULL, NULL);
+ if (!op) {
+ qpa_logerr(pa_context_errno(c->context),
+ "set_sink_input_mute() failed\n");
+ } else {
+ pa_operation_unref(op);
}
- return 0;
+
+ pa_threaded_mainloop_unlock(c->mainloop);
}
-static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...)
+static void qpa_volume_in(HWVoiceIn *hw, struct mixeng_volume *vol)
{
PAVoiceIn *pa = (PAVoiceIn *) hw;
pa_operation *op;
@@ -521,46 +509,33 @@ static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...)
pa_cvolume_init (&v);
#endif
- switch (cmd) {
- case VOICE_VOLUME:
- {
- SWVoiceIn *sw;
- va_list ap;
+ v.channels = 2;
+ v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->l) / UINT32_MAX;
+ v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->r) / UINT32_MAX;
- va_start (ap, cmd);
- sw = va_arg (ap, SWVoiceIn *);
- va_end (ap);
+ pa_threaded_mainloop_lock(c->mainloop);
- v.channels = 2;
- v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.l) / UINT32_MAX;
- v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * sw->vol.r) / UINT32_MAX;
-
- pa_threaded_mainloop_lock(c->mainloop);
-
- op = pa_context_set_source_output_volume(c->context,
- pa_stream_get_index(pa->stream),
- &v, NULL, NULL);
- if (!op) {
- qpa_logerr(pa_context_errno(c->context),
- "set_source_output_volume() failed\n");
- } else {
- pa_operation_unref(op);
- }
-
- op = pa_context_set_source_output_mute(c->context,
- pa_stream_get_index (pa->stream),
- sw->vol.mute, NULL, NULL);
- if (!op) {
- qpa_logerr(pa_context_errno(c->context),
- "set_source_output_mute() failed\n");
- } else {
- pa_operation_unref (op);
- }
+ op = pa_context_set_source_output_volume(c->context,
+ pa_stream_get_index(pa->stream),
+ &v, NULL, NULL);
+ if (!op) {
+ qpa_logerr(pa_context_errno(c->context),
+ "set_source_output_volume() failed\n");
+ } else {
+ pa_operation_unref(op);
+ }
- pa_threaded_mainloop_unlock(c->mainloop);
- }
+ op = pa_context_set_source_output_mute(c->context,
+ pa_stream_get_index(pa->stream),
+ vol->mute, NULL, NULL);
+ if (!op) {
+ qpa_logerr(pa_context_errno(c->context),
+ "set_source_output_mute() failed\n");
+ } else {
+ pa_operation_unref (op);
}
- return 0;
+
+ pa_threaded_mainloop_unlock(c->mainloop);
}
/* common */
@@ -695,13 +670,13 @@ static struct audio_pcm_ops qpa_pcm_ops = {
.fini_out = qpa_fini_out,
.write = qpa_write,
.buffer_size_out = qpa_buffer_size_out,
- .ctl_out = qpa_ctl_out,
+ .volume_out = qpa_volume_out,
.init_in = qpa_init_in,
.fini_in = qpa_fini_in,
.read = qpa_read,
.buffer_size_in = qpa_buffer_size_in,
- .ctl_in = qpa_ctl_in
+ .volume_in = qpa_volume_in
};
static struct audio_driver pa_audio_driver = {
@@ -715,7 +690,6 @@ static struct audio_driver pa_audio_driver = {
.max_voices_in = INT_MAX,
.voice_size_out = sizeof (PAVoiceOut),
.voice_size_in = sizeof (PAVoiceIn),
- .ctl_caps = VOICE_VOLUME_CAP
};
static void register_audio_pa(void)
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index 4df35ce31a..93554df119 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -378,20 +378,9 @@ static size_t sdl_buffer_size_out(HWVoiceOut *hw)
return sdl->samples;
}
-static int sdl_ctl_out (HWVoiceOut *hw, int cmd, ...)
+static void sdl_enable_out(HWVoiceOut *hw, bool enable)
{
- (void) hw;
-
- switch (cmd) {
- case VOICE_ENABLE:
- SDL_PauseAudio (0);
- break;
-
- case VOICE_DISABLE:
- SDL_PauseAudio (1);
- break;
- }
- return 0;
+ SDL_PauseAudio(!enable);
}
static void *sdl_audio_init(Audiodev *dev)
@@ -449,7 +438,7 @@ static struct audio_pcm_ops sdl_pcm_ops = {
.buffer_size_out = sdl_buffer_size_out,
.get_buffer_out = sdl_get_buffer_out,
.put_buffer_out = sdl_put_buffer_out_nowrite,
- .ctl_out = sdl_ctl_out,
+ .enable_out = sdl_enable_out,
};
static struct audio_driver sdl_audio_driver = {
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index 2495866c82..771ecacf27 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -153,22 +153,20 @@ static size_t line_out_put_buffer(HWVoiceOut *hw, void *buf, size_t size)
return size;
}
-static int line_out_ctl (HWVoiceOut *hw, int cmd, ...)
+static void line_out_enable(HWVoiceOut *hw, bool enable)
{
SpiceVoiceOut *out = container_of (hw, SpiceVoiceOut, hw);
- switch (cmd) {
- case VOICE_ENABLE:
+ if (enable) {
if (out->active) {
- break;
+ return;
}
out->active = 1;
audio_rate_start(&out->rate);
spice_server_playback_start (&out->sin);
- break;
- case VOICE_DISABLE:
+ } else {
if (!out->active) {
- break;
+ return;
}
out->active = 0;
if (out->frame) {
@@ -177,29 +175,21 @@ static int line_out_ctl (HWVoiceOut *hw, int cmd, ...)
out->frame = NULL;
}
spice_server_playback_stop (&out->sin);
- break;
- case VOICE_VOLUME:
- {
+ }
+}
+
#if ((SPICE_INTERFACE_PLAYBACK_MAJOR >= 1) && (SPICE_INTERFACE_PLAYBACK_MINOR >= 2))
- SWVoiceOut *sw;
- va_list ap;
- uint16_t vol[2];
+static void line_out_volume(HWVoiceOut *hw, struct mixeng_volume *vol)
+{
+ SpiceVoiceOut *out = container_of(hw, SpiceVoiceOut, hw);
+ uint16_t svol[2];
- va_start (ap, cmd);
- sw = va_arg (ap, SWVoiceOut *);
- va_end (ap);
-
- vol[0] = sw->vol.l / ((1ULL << 16) + 1);
- vol[1] = sw->vol.r / ((1ULL << 16) + 1);
- spice_server_playback_set_volume (&out->sin, 2, vol);
- spice_server_playback_set_mute (&out->sin, sw->vol.mute);
+ svol[0] = vol->l / ((1ULL << 16) + 1);
+ svol[1] = vol->r / ((1ULL << 16) + 1);
+ spice_server_playback_set_volume(&out->sin, 2, svol);
+ spice_server_playback_set_mute(&out->sin, vol->mute);
+}
#endif
- break;
- }
- }
-
- return 0;
-}
/* record */
@@ -255,48 +245,38 @@ static size_t line_in_read(HWVoiceIn *hw, void *buf, size_t len)
return ready << 2;
}
-static int line_in_ctl (HWVoiceIn *hw, int cmd, ...)
+static void line_in_enable(HWVoiceIn *hw, bool enable)
{
SpiceVoiceIn *in = container_of (hw, SpiceVoiceIn, hw);
- switch (cmd) {
- case VOICE_ENABLE:
+ if (enable) {
if (in->active) {
- break;
+ return;
}
in->active = 1;
audio_rate_start(&in->rate);
spice_server_record_start (&in->sin);
- break;
- case VOICE_DISABLE:
+ } else {
if (!in->active) {
- break;
+ return;
}
in->active = 0;
spice_server_record_stop (&in->sin);
- break;
- case VOICE_VOLUME:
- {
+ }
+}
+
#if ((SPICE_INTERFACE_RECORD_MAJOR >= 2) && (SPICE_INTERFACE_RECORD_MINOR >= 2))
- SWVoiceIn *sw;
- va_list ap;
- uint16_t vol[2];
+static void line_in_volume(HWVoiceIn *hw, struct mixeng_volume *vol)
+{
+ SpiceVoiceIn *in = container_of (hw, SpiceVoiceIn, hw);
+ uint16_t svol[2];
- va_start (ap, cmd);
- sw = va_arg (ap, SWVoiceIn *);
- va_end (ap);
-
- vol[0] = sw->vol.l / ((1ULL << 16) + 1);
- vol[1] = sw->vol.r / ((1ULL << 16) + 1);
- spice_server_record_set_volume (&in->sin, 2, vol);
- spice_server_record_set_mute (&in->sin, sw->vol.mute);
+ svol[0] = vol->l / ((1ULL << 16) + 1);
+ svol[1] = vol->r / ((1ULL << 16) + 1);
+ spice_server_record_set_volume(&in->sin, 2, svol);
+ spice_server_record_set_mute(&in->sin, vol->mute);
+}
#endif
- break;
- }
- }
-
- return 0;
-}
static struct audio_pcm_ops audio_callbacks = {
.init_out = line_out_init,
@@ -305,13 +285,19 @@ static struct audio_pcm_ops audio_callbacks = {
.buffer_size_out = line_out_buffer_size,
.get_buffer_out = line_out_get_buffer,
.put_buffer_out = line_out_put_buffer,
- .ctl_out = line_out_ctl,
+ .enable_out = line_out_enable,
+#if ((SPICE_INTERFACE_PLAYBACK_MAJOR >= 1) && (SPICE_INTERFACE_PLAYBACK_MINOR >= 2))
+ .volume_out = line_out_volume,
+#endif
.init_in = line_in_init,
.fini_in = line_in_fini,
.read = line_in_read,
.buffer_size_in = line_in_buffer_size,
- .ctl_in = line_in_ctl,
+ .enable_in = line_in_enable,
+#if ((SPICE_INTERFACE_RECORD_MAJOR >= 2) && (SPICE_INTERFACE_RECORD_MINOR >= 2))
+ .volume_in = line_in_volume,
+#endif
};
static struct audio_driver spice_audio_driver = {
@@ -324,9 +310,6 @@ static struct audio_driver spice_audio_driver = {
.max_voices_in = 1,
.voice_size_out = sizeof (SpiceVoiceOut),
.voice_size_in = sizeof (SpiceVoiceIn),
-#if ((SPICE_INTERFACE_PLAYBACK_MAJOR >= 1) && (SPICE_INTERFACE_PLAYBACK_MINOR >= 2))
- .ctl_caps = VOICE_VOLUME_CAP
-#endif
};
void qemu_spice_audio_init (void)
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index f977d96b9c..672d5fc337 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -170,14 +170,13 @@ static void wav_fini_out (HWVoiceOut *hw)
wav->f = NULL;
}
-static int wav_ctl_out (HWVoiceOut *hw, int cmd, ...)
+static void wav_enable_out(HWVoiceOut *hw, bool enable)
{
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
- if (cmd == VOICE_ENABLE) {
+ if (enable) {
audio_rate_start(&wav->rate);
}
- return 0;
}
static void *wav_audio_init(Audiodev *dev)
@@ -195,7 +194,7 @@ static struct audio_pcm_ops wav_pcm_ops = {
.init_out = wav_init_out,
.fini_out = wav_fini_out,
.write = wav_write_out,
- .ctl_out = wav_ctl_out,
+ .enable_out = wav_enable_out,
};
static struct audio_driver wav_audio_driver = {
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 43/52] audio: add mixeng option (documentation)
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (41 preceding siblings ...)
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 42/52] audio: split ctl_* functions into enable_* and volume_* Kővágó, Zoltán
@ 2018-12-23 20:52 ` Kővágó, Zoltán
2019-01-10 1:43 ` Eric Blake
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 44/52] audio: make mixeng optional Kővágó, Zoltán
` (7 subsequent siblings)
50 siblings, 1 reply; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:52 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann, Eric Blake, Markus Armbruster
This will allow us to disable mixeng when we use a decent backend.
Disabling mixeng have a few advantages:
* we no longer convert the audio output from one format to another, when
the underlying audio system would just convert it to a third format.
We no longer convert, only the underlying system, when needed.
* the underlying system probably has better resampling and sample format
converting methods anyway...
* we may support formats that the mixeng currently does not support (S24
or float samples, more than two channels)
* when using an audio server (like pulseaudio) different sound card
outputs will show up as separate streams, even if we use only one
backend
Disadvantages:
* audio capturing no longer works (wavcapture, and vnc audio extension)
* some backends only support a single playback stream or very picky
about the audio format. In this case we can't disable mixeng.
However mixeng is not removed, only made optional, so this shouldn't be
a big concern.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
qapi/audio.json | 5 +++++
qemu-options.hx | 6 ++++++
2 files changed, 11 insertions(+)
diff --git a/qapi/audio.json b/qapi/audio.json
index 56d8ce439f..180bf207a8 100644
--- a/qapi/audio.json
+++ b/qapi/audio.json
@@ -184,6 +184,10 @@
#
# General audio backend options that are used for both playback and recording.
#
+# @mixeng: #optional use QEMU's mixing engine to mix all streams inside QEMU.
+# When set to off, fixed-settings must be also off. Not every backend
+# compatible with the off setting (default on)
+#
# @fixed-settings: #optional use fixed settings for host input/output. When
# off, frequency, channels and format must not be specified
# (default on)
@@ -207,6 +211,7 @@
##
{ 'struct': 'AudiodevPerDirectionOptions',
'data': {
+ '*mixeng': 'bool',
'*fixed-settings': 'bool',
'*frequency': 'int',
'*channels': 'int',
diff --git a/qemu-options.hx b/qemu-options.hx
index 687dd23bf2..eb65b511fa 100644
--- a/qemu-options.hx
+++ b/qemu-options.hx
@@ -432,6 +432,7 @@ DEF("audiodev", HAS_ARG, QEMU_OPTION_audiodev,
" specifies the audio backend to use\n"
" id= identifier of the backend\n"
" timer-period= timer period in microseconds\n"
+ " in|out.mixeng= use mixeng to mix streams inside QEMU\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"
@@ -500,6 +501,11 @@ Identifies the audio backend.
Sets the timer @var{period} used by the audio subsystem in microseconds.
Default is 10000 (10 ms).
+@item in|out.mixeng=on|off
+Use QEMU's mixing engine to mix all streams inside QEMU. When off,
+@var{fixed-settings} must be off too. Not every backend is fully
+compatible with the off setting. Default is on.
+
@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. In this case you must not specify
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* Re: [Qemu-devel] [PATCH v2 43/52] audio: add mixeng option (documentation)
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 43/52] audio: add mixeng option (documentation) Kővágó, Zoltán
@ 2019-01-10 1:43 ` Eric Blake
2019-01-10 9:12 ` Markus Armbruster
2019-01-16 20:27 ` Zoltán Kővágó
0 siblings, 2 replies; 91+ messages in thread
From: Eric Blake @ 2019-01-10 1:43 UTC (permalink / raw)
To: Kővágó, Zoltán, qemu-devel
Cc: Gerd Hoffmann, Markus Armbruster
[-- Attachment #1: Type: text/plain, Size: 2481 bytes --]
On 12/23/18 2:52 PM, Kővágó, Zoltán wrote:
> This will allow us to disable mixeng when we use a decent backend.
>
> Disabling mixeng have a few advantages:
> * we no longer convert the audio output from one format to another, when
> the underlying audio system would just convert it to a third format.
> We no longer convert, only the underlying system, when needed.
> * the underlying system probably has better resampling and sample format
> converting methods anyway...
> * we may support formats that the mixeng currently does not support (S24
> or float samples, more than two channels)
> * when using an audio server (like pulseaudio) different sound card
> outputs will show up as separate streams, even if we use only one
> backend
>
> Disadvantages:
> * audio capturing no longer works (wavcapture, and vnc audio extension)
> * some backends only support a single playback stream or very picky
> about the audio format. In this case we can't disable mixeng.
>
> However mixeng is not removed, only made optional, so this shouldn't be
> a big concern.
>
> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
> ---
> qapi/audio.json | 5 +++++
> qemu-options.hx | 6 ++++++
> 2 files changed, 11 insertions(+)
>
> diff --git a/qapi/audio.json b/qapi/audio.json
> index 56d8ce439f..180bf207a8 100644
> --- a/qapi/audio.json
> +++ b/qapi/audio.json
> @@ -184,6 +184,10 @@
> #
> # General audio backend options that are used for both playback and recording.
> #
> +# @mixeng: #optional use QEMU's mixing engine to mix all streams inside QEMU.
The #optional tag is no longer necessary (the docs generator figures it
out from the '*mixeng' name below).
> +# When set to off, fixed-settings must be also off. Not every backend
> +# compatible with the off setting (default on)
Missing a '(since 4.0)' tag.
> +#
> # @fixed-settings: #optional use fixed settings for host input/output. When
> # off, frequency, channels and format must not be specified
> # (default on)
> @@ -207,6 +211,7 @@
> ##
> { 'struct': 'AudiodevPerDirectionOptions',
> 'data': {
> + '*mixeng': 'bool',
> '*fixed-settings': 'bool',
> '*frequency': 'int',
> '*channels': 'int',
--
Eric Blake, Principal Software Engineer
Red Hat, Inc. +1-919-301-3226
Virtualization: qemu.org | libvirt.org
[-- Attachment #2: OpenPGP digital signature --]
[-- Type: application/pgp-signature, Size: 488 bytes --]
^ permalink raw reply [flat|nested] 91+ messages in thread* Re: [Qemu-devel] [PATCH v2 43/52] audio: add mixeng option (documentation)
2019-01-10 1:43 ` Eric Blake
@ 2019-01-10 9:12 ` Markus Armbruster
2019-01-16 20:27 ` Zoltán Kővágó
1 sibling, 0 replies; 91+ messages in thread
From: Markus Armbruster @ 2019-01-10 9:12 UTC (permalink / raw)
To: Eric Blake; +Cc: dirty.ice.hu, qemu-devel, Gerd Hoffmann
Eric Blake <eblake@redhat.com> writes:
> On 12/23/18 2:52 PM, Kővágó, Zoltán wrote:
>> This will allow us to disable mixeng when we use a decent backend.
>>
>> Disabling mixeng have a few advantages:
>> * we no longer convert the audio output from one format to another, when
>> the underlying audio system would just convert it to a third format.
>> We no longer convert, only the underlying system, when needed.
>> * the underlying system probably has better resampling and sample format
>> converting methods anyway...
>> * we may support formats that the mixeng currently does not support (S24
>> or float samples, more than two channels)
>> * when using an audio server (like pulseaudio) different sound card
>> outputs will show up as separate streams, even if we use only one
>> backend
>>
>> Disadvantages:
>> * audio capturing no longer works (wavcapture, and vnc audio extension)
>> * some backends only support a single playback stream or very picky
>> about the audio format. In this case we can't disable mixeng.
>>
>> However mixeng is not removed, only made optional, so this shouldn't be
>> a big concern.
>>
>> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
>> ---
>> qapi/audio.json | 5 +++++
>> qemu-options.hx | 6 ++++++
>> 2 files changed, 11 insertions(+)
>>
>> diff --git a/qapi/audio.json b/qapi/audio.json
>> index 56d8ce439f..180bf207a8 100644
>> --- a/qapi/audio.json
>> +++ b/qapi/audio.json
>> @@ -184,6 +184,10 @@
>> #
>> # General audio backend options that are used for both playback and recording.
>> #
>> +# @mixeng: #optional use QEMU's mixing engine to mix all streams inside QEMU.
>
> The #optional tag is no longer necessary (the docs generator figures it
> out from the '*mixeng' name below).
In fact, the qapi generator had special code to detect and reject
#optional tags for some six months (from commit 1d8bda128d2 to
67ab1ce9263), just to ensure patches in flight don't add them. This one
has been in flight for even longer.
[...]
^ permalink raw reply [flat|nested] 91+ messages in thread
* Re: [Qemu-devel] [PATCH v2 43/52] audio: add mixeng option (documentation)
2019-01-10 1:43 ` Eric Blake
2019-01-10 9:12 ` Markus Armbruster
@ 2019-01-16 20:27 ` Zoltán Kővágó
2019-01-16 22:40 ` Eric Blake
1 sibling, 1 reply; 91+ messages in thread
From: Zoltán Kővágó @ 2019-01-16 20:27 UTC (permalink / raw)
To: Eric Blake, qemu-devel; +Cc: Gerd Hoffmann, Markus Armbruster
On 2019-01-10 02:43, Eric Blake wrote:
> On 12/23/18 2:52 PM, Kővágó, Zoltán wrote:
>> This will allow us to disable mixeng when we use a decent backend.
>>
>> Disabling mixeng have a few advantages:
>> * we no longer convert the audio output from one format to another, when
>> the underlying audio system would just convert it to a third format.
>> We no longer convert, only the underlying system, when needed.
>> * the underlying system probably has better resampling and sample format
>> converting methods anyway...
>> * we may support formats that the mixeng currently does not support (S24
>> or float samples, more than two channels)
>> * when using an audio server (like pulseaudio) different sound card
>> outputs will show up as separate streams, even if we use only one
>> backend
>>
>> Disadvantages:
>> * audio capturing no longer works (wavcapture, and vnc audio extension)
>> * some backends only support a single playback stream or very picky
>> about the audio format. In this case we can't disable mixeng.
>>
>> However mixeng is not removed, only made optional, so this shouldn't be
>> a big concern.
>>
>> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
>> ---
>> qapi/audio.json | 5 +++++
>> qemu-options.hx | 6 ++++++
>> 2 files changed, 11 insertions(+)
>>
>> diff --git a/qapi/audio.json b/qapi/audio.json
>> index 56d8ce439f..180bf207a8 100644
>> --- a/qapi/audio.json
>> +++ b/qapi/audio.json
>> @@ -184,6 +184,10 @@
>> #
>> # General audio backend options that are used for both playback and recording.
>> #
>> +# @mixeng: #optional use QEMU's mixing engine to mix all streams inside QEMU.
>
> The #optional tag is no longer necessary (the docs generator figures it
> out from the '*mixeng' name below).
Ok.
>> +# When set to off, fixed-settings must be also off. Not every backend
>> +# compatible with the off setting (default on)
>
> Missing a '(since 4.0)' tag.
The whole AudiodevPerDirectionOptions struct is marked with 'Since:
4.0', so I don't think it's necessary.
>
>> +#
>> # @fixed-settings: #optional use fixed settings for host input/output. When
>> # off, frequency, channels and format must not be specified
>> # (default on)
>> @@ -207,6 +211,7 @@
>> ##
>> { 'struct': 'AudiodevPerDirectionOptions',
>> 'data': {
>> + '*mixeng': 'bool',
>> '*fixed-settings': 'bool',
>> '*frequency': 'int',
>> '*channels': 'int',
>
>
^ permalink raw reply [flat|nested] 91+ messages in thread* Re: [Qemu-devel] [PATCH v2 43/52] audio: add mixeng option (documentation)
2019-01-16 20:27 ` Zoltán Kővágó
@ 2019-01-16 22:40 ` Eric Blake
0 siblings, 0 replies; 91+ messages in thread
From: Eric Blake @ 2019-01-16 22:40 UTC (permalink / raw)
To: Zoltán Kővágó, qemu-devel
Cc: Gerd Hoffmann, Markus Armbruster
[-- Attachment #1: Type: text/plain, Size: 1066 bytes --]
On 1/16/19 2:27 PM, Zoltán Kővágó wrote:
>>> +++ b/qapi/audio.json
>>> @@ -184,6 +184,10 @@
>>> #
>>> # General audio backend options that are used for both playback and recording.
>>> #
>>> +# @mixeng: #optional use QEMU's mixing engine to mix all streams inside QEMU.
>>
>> The #optional tag is no longer necessary (the docs generator figures it
>> out from the '*mixeng' name below).
>
> Ok.
>
>>> +# When set to off, fixed-settings must be also off. Not every backend
>>> +# compatible with the off setting (default on)
>>
>> Missing a '(since 4.0)' tag.
>
> The whole AudiodevPerDirectionOptions struct is marked with 'Since:
> 4.0', so I don't think it's necessary.
Oh right. It's less common to see incremental additions to a type within
a single release, but you are right that the per-member tag is only
needed when an older release exposed the type you are extending.
--
Eric Blake, Principal Software Engineer
Red Hat, Inc. +1-919-301-3226
Virtualization: qemu.org | libvirt.org
[-- Attachment #2: OpenPGP digital signature --]
[-- Type: application/pgp-signature, Size: 488 bytes --]
^ permalink raw reply [flat|nested] 91+ messages in thread
* [Qemu-devel] [PATCH v2 44/52] audio: make mixeng optional
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (42 preceding siblings ...)
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 43/52] audio: add mixeng option (documentation) Kővágó, Zoltán
@ 2018-12-23 20:52 ` Kővágó, Zoltán
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 46/52] audio: support more than two channels in volume setting Kővágó, Zoltán
` (6 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:52 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Implementation of the previously added mixeng option.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio.c | 70 ++++++++++++++++++++++++++++++++++++++----
audio/audio_template.h | 46 ++++++++++++++++-----------
2 files changed, 92 insertions(+), 24 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index 7ba939560a..26ce25989f 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -840,32 +840,46 @@ static void audio_timer (void *opaque)
*/
size_t AUD_write(SWVoiceOut *sw, void *buf, size_t size)
{
+ HWVoiceOut *hw;
+
if (!sw) {
/* XXX: Consider options */
return size;
}
+ hw = sw->hw;
- if (!sw->hw->enabled) {
+ if (!hw->enabled) {
dolog ("Writing to disabled voice %s\n", SW_NAME (sw));
return 0;
}
- return audio_pcm_sw_write(sw, buf, size);
+ if (hw->s->dev->out->mixeng) {
+ return audio_pcm_sw_write(sw, buf, size);
+ } else {
+ return hw->pcm_ops->write(hw, buf, size);
+ }
}
size_t AUD_read(SWVoiceIn *sw, void *buf, size_t size)
{
+ HWVoiceIn *hw;
+
if (!sw) {
/* XXX: Consider options */
return size;
}
+ hw = sw->hw;
- if (!sw->hw->enabled) {
+ if (!hw->enabled) {
dolog ("Reading from disabled voice %s\n", SW_NAME (sw));
return 0;
}
- return audio_pcm_sw_read(sw, buf, size);
+ if (hw->s->dev->in->mixeng) {
+ return audio_pcm_sw_read(sw, buf, size);
+ } else {
+ return hw->pcm_ops->read(hw, buf, size);
+ }
}
int AUD_get_buffer_size_out (SWVoiceOut *sw)
@@ -1087,6 +1101,26 @@ static void audio_run_out (AudioState *s)
HWVoiceOut *hw = NULL;
SWVoiceOut *sw;
+ if (!s->dev->out->mixeng) {
+ while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
+ /* there is exactly 1 sw for each hw with no mixeng */
+ sw = hw->sw_head.lh_first;
+
+ if (hw->pending_disable) {
+ hw->enabled = 0;
+ hw->pending_disable = 0;
+ if (hw->pcm_ops->enable_out) {
+ hw->pcm_ops->enable_out(hw, false);
+ }
+ }
+
+ if (sw->active) {
+ sw->callback.fn(sw->callback.opaque, INT_MAX);
+ }
+ }
+ return;
+ }
+
while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
size_t played, live, prev_rpos, free;
int nb_live, cleanup_required;
@@ -1224,6 +1258,17 @@ static void audio_run_in (AudioState *s)
{
HWVoiceIn *hw = NULL;
+ if (!s->dev->in->mixeng) {
+ while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
+ /* there is exactly 1 sw for each hw with no mixeng */
+ SWVoiceIn *sw = hw->sw_head.lh_first;
+ if (sw->active) {
+ sw->callback.fn(sw->callback.opaque, INT_MAX);
+ }
+ }
+ return;
+ }
+
while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
SWVoiceIn *sw;
size_t captured = 0, min;
@@ -1717,6 +1762,11 @@ CaptureVoiceOut *AUD_add_capture(
s = audio_init(NULL, NULL);
}
+ if (!s->dev->out->mixeng) {
+ dolog("Can't capture with mixeng disabled\n");
+ goto err0;
+ }
+
if (audio_validate_settings (as)) {
dolog ("Invalid settings were passed when trying to add capture\n");
audio_print_settings (as);
@@ -1875,9 +1925,13 @@ QemuOptsList qemu_audiodev_opts = {
static void validate_per_direction_opts(AudiodevPerDirectionOptions *pdo,
Error **errp)
{
+ if (!pdo->has_mixeng) {
+ pdo->has_mixeng = true;
+ pdo->mixeng = true;
+ }
if (!pdo->has_fixed_settings) {
pdo->has_fixed_settings = true;
- pdo->fixed_settings = true;
+ pdo->fixed_settings = pdo->mixeng;
}
if (!pdo->fixed_settings &&
(pdo->has_frequency || pdo->has_channels || pdo->has_format)) {
@@ -1885,6 +1939,10 @@ static void validate_per_direction_opts(AudiodevPerDirectionOptions *pdo,
"You can't use frequency, channels or format with fixed-settings=off");
return;
}
+ if (!pdo->mixeng && pdo->fixed_settings) {
+ error_setg(errp, "You can't use fixed-settings without mixeng");
+ return;
+ }
if (!pdo->has_frequency) {
pdo->has_frequency = true;
@@ -1896,7 +1954,7 @@ static void validate_per_direction_opts(AudiodevPerDirectionOptions *pdo,
}
if (!pdo->has_voices) {
pdo->has_voices = true;
- pdo->voices = 1;
+ pdo->voices = pdo->mixeng ? 1 : INT_MAX;
}
if (!pdo->has_format) {
pdo->has_format = true;
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 9ec565fc48..e5a4d6fa40 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -78,26 +78,32 @@ static void glue (audio_pcm_hw_free_resources_, TYPE) (HW *hw)
static void glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw)
{
- size_t samples;
- if (!hw->pcm_ops) {
- /*
- * We should only end up here when using wavcapture hmp command (and not
- * the wavcapture audio backend).
- * It needs a lot of samples, otherwise you'll end up with "Could not
- * mix X bytes into a capture buffer" warnings and a garbled capture.
- */
- samples = 4096 * 4;
- } else if (hw->pcm_ops->glue(buffer_size_, TYPE)) {
- samples = hw->pcm_ops->glue(buffer_size_, TYPE)(hw);
+ if (hw->s->dev->TYPE->mixeng) {
+ size_t samples;
+ if (!hw->pcm_ops) {
+ /*
+ * We should only end up here when using wavcapture hmp command (and
+ * not the wavcapture audio backend).
+ * It needs a lot of samples, otherwise you'll end up with "Could
+ * not mix X bytes into a capture buffer" warnings and a garbled
+ * capture.
+ */
+ samples = 4096 * 4;
+ } else if (hw->pcm_ops->glue(buffer_size_, TYPE)) {
+ samples = hw->pcm_ops->glue(buffer_size_, TYPE)(hw);
+ } else {
+ samples = 1024; /* todo better default */
+ }
+
+ if (audio_bug(__func__, samples == 0)) {
+ dolog("Attempted to allocate empty buffer\n");
+ }
+
+ HWBUF = g_malloc0(sizeof(STSampleBuffer) + sizeof(st_sample) * samples);
+ HWBUF->size = samples;
} else {
- samples = 1024; /* todo better default */
+ HWBUF = NULL;
}
- if (audio_bug(__func__, samples == 0)) {
- dolog("Attempted to allocate empty buffer\n");
- }
-
- HWBUF = g_malloc0(sizeof(STSampleBuffer) + sizeof(st_sample) * samples);
- HWBUF->size = samples;
}
static void glue (audio_pcm_sw_free_resources_, TYPE) (SW *sw)
@@ -116,6 +122,10 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw)
{
int samples;
+ if (!sw->s->dev->TYPE->mixeng) {
+ return 0;
+ }
+
samples = ((int64_t) sw->HWBUF->size << 32) / sw->ratio;
sw->buf = audio_calloc(__func__, samples, sizeof(struct st_sample));
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 46/52] audio: support more than two channels in volume setting
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (43 preceding siblings ...)
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 44/52] audio: make mixeng optional Kővágó, Zoltán
@ 2018-12-23 20:52 ` Kővágó, Zoltán
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 47/52] audio: replace shift in audio_pcm_info with bytes_per_frame Kővágó, Zoltán
` (5 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:52 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio.c | 28 ++++++++++++++++++++--------
audio/audio.h | 10 ++++++++++
audio/audio_int.h | 4 ++--
audio/paaudio.c | 20 ++++++++++++--------
audio/spiceaudio.c | 14 ++++++++------
5 files changed, 52 insertions(+), 24 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index 26ce25989f..71744fa2c8 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -1880,31 +1880,43 @@ void AUD_del_capture (CaptureVoiceOut *cap, void *cb_opaque)
}
void AUD_set_volume_out (SWVoiceOut *sw, int mute, uint8_t lvol, uint8_t rvol)
+{
+ Volume vol = { .mute = mute, .channels = 2, .vol = { lvol, rvol } };
+ audio_set_volume_out(sw, &vol);
+}
+
+void audio_set_volume_out(SWVoiceOut *sw, Volume *vol)
{
if (sw) {
HWVoiceOut *hw = sw->hw;
- sw->vol.mute = mute;
- sw->vol.l = nominal_volume.l * lvol / 255;
- sw->vol.r = nominal_volume.r * rvol / 255;
+ sw->vol.mute = vol->mute;
+ sw->vol.l = nominal_volume.l * vol->vol[0] / 255;
+ sw->vol.r = nominal_volume.l * vol->vol[vol->channels > 1 ? 1 : 0] / 255;
if (hw->pcm_ops->volume_out) {
- hw->pcm_ops->volume_out(hw, &sw->vol);
+ hw->pcm_ops->volume_out(hw, vol);
}
}
}
void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
+{
+ Volume vol = { .mute = mute, .channels = 2, .vol = { lvol, rvol } };
+ audio_set_volume_in(sw, &vol);
+}
+
+void audio_set_volume_in(SWVoiceIn *sw, Volume *vol)
{
if (sw) {
HWVoiceIn *hw = sw->hw;
- sw->vol.mute = mute;
- sw->vol.l = nominal_volume.l * lvol / 255;
- sw->vol.r = nominal_volume.r * rvol / 255;
+ sw->vol.mute = vol->mute;
+ sw->vol.l = nominal_volume.l * vol->vol[0] / 255;
+ sw->vol.r = nominal_volume.r * vol->vol[vol->channels > 1 ? 1 : 0] / 255;
if (hw->pcm_ops->volume_in) {
- hw->pcm_ops->volume_in(hw, &sw->vol);
+ hw->pcm_ops->volume_in(hw, vol);
}
}
}
diff --git a/audio/audio.h b/audio/audio.h
index bfb12e7531..911365cebd 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -125,6 +125,16 @@ uint64_t AUD_get_elapsed_usec_out (SWVoiceOut *sw, QEMUAudioTimeStamp *ts);
void AUD_set_volume_out (SWVoiceOut *sw, int mute, uint8_t lvol, uint8_t rvol);
void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol);
+#define AUDIO_MAX_CHANNELS 16
+typedef struct Volume {
+ bool mute;
+ int channels;
+ uint8_t vol[AUDIO_MAX_CHANNELS];
+} Volume;
+
+void audio_set_volume_out(SWVoiceOut *sw, Volume *vol);
+void audio_set_volume_in(SWVoiceIn *sw, Volume *vol);
+
SWVoiceIn *AUD_open_in (
QEMUSoundCard *card,
SWVoiceIn *sw,
diff --git a/audio/audio_int.h b/audio/audio_int.h
index a5ead07998..c2f6da5cc2 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -165,7 +165,7 @@ struct audio_pcm_ops {
*/
size_t (*put_buffer_out)(HWVoiceOut *hw, void *buf, size_t size);
void (*enable_out)(HWVoiceOut *hw, bool enable);
- void (*volume_out)(HWVoiceOut *hw, struct mixeng_volume *vol);
+ void (*volume_out)(HWVoiceOut *hw, Volume *vol);
int (*init_in) (HWVoiceIn *hw, audsettings *as, void *drv_opaque);
void (*fini_in) (HWVoiceIn *hw);
@@ -174,7 +174,7 @@ struct audio_pcm_ops {
void *(*get_buffer_in)(HWVoiceIn *hw, size_t *size);
void (*put_buffer_in)(HWVoiceIn *hw, void *buf, size_t size);
void (*enable_in)(HWVoiceIn *hw, bool enable);
- void (*volume_in)(HWVoiceIn *hw, struct mixeng_volume *vol);
+ void (*volume_in)(HWVoiceIn *hw, Volume *vol);
};
void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size);
diff --git a/audio/paaudio.c b/audio/paaudio.c
index e8556e7a9d..3b4b29b38f 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -537,20 +537,22 @@ static void qpa_fini_in (HWVoiceIn *hw)
}
}
-static void qpa_volume_out(HWVoiceOut *hw, struct mixeng_volume *vol)
+static void qpa_volume_out(HWVoiceOut *hw, Volume *vol)
{
PAVoiceOut *pa = (PAVoiceOut *) hw;
pa_operation *op;
pa_cvolume v;
PAConnection *c = pa->g->conn;
+ int i;
#ifdef PA_CHECK_VERSION /* macro is present in 0.9.16+ */
pa_cvolume_init (&v); /* function is present in 0.9.13+ */
#endif
- v.channels = 2;
- v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->l) / UINT32_MAX;
- v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->r) / UINT32_MAX;
+ v.channels = vol->channels;
+ for (i = 0; i < vol->channels; ++i) {
+ v.values[i] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->vol[i]) / 255;
+ }
pa_threaded_mainloop_lock(c->mainloop);
@@ -577,20 +579,22 @@ static void qpa_volume_out(HWVoiceOut *hw, struct mixeng_volume *vol)
pa_threaded_mainloop_unlock(c->mainloop);
}
-static void qpa_volume_in(HWVoiceIn *hw, struct mixeng_volume *vol)
+static void qpa_volume_in(HWVoiceIn *hw, Volume *vol)
{
PAVoiceIn *pa = (PAVoiceIn *) hw;
pa_operation *op;
pa_cvolume v;
PAConnection *c = pa->g->conn;
+ int i;
#ifdef PA_CHECK_VERSION
pa_cvolume_init (&v);
#endif
- v.channels = 2;
- v.values[0] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->l) / UINT32_MAX;
- v.values[1] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->r) / UINT32_MAX;
+ v.channels = vol->channels;
+ for (i = 0; i < vol->channels; ++i) {
+ v.values[i] = ((PA_VOLUME_NORM - PA_VOLUME_MUTED) * vol->vol[i]) / 255;
+ }
pa_threaded_mainloop_lock(c->mainloop);
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index 771ecacf27..b26d10434b 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -179,13 +179,14 @@ static void line_out_enable(HWVoiceOut *hw, bool enable)
}
#if ((SPICE_INTERFACE_PLAYBACK_MAJOR >= 1) && (SPICE_INTERFACE_PLAYBACK_MINOR >= 2))
-static void line_out_volume(HWVoiceOut *hw, struct mixeng_volume *vol)
+static void line_out_volume(HWVoiceOut *hw, Volume *vol)
{
SpiceVoiceOut *out = container_of(hw, SpiceVoiceOut, hw);
uint16_t svol[2];
- svol[0] = vol->l / ((1ULL << 16) + 1);
- svol[1] = vol->r / ((1ULL << 16) + 1);
+ assert(vol->channels == 2);
+ svol[0] = vol->vol[0] * 257;
+ svol[1] = vol->vol[1] * 257;
spice_server_playback_set_volume(&out->sin, 2, svol);
spice_server_playback_set_mute(&out->sin, vol->mute);
}
@@ -266,13 +267,14 @@ static void line_in_enable(HWVoiceIn *hw, bool enable)
}
#if ((SPICE_INTERFACE_RECORD_MAJOR >= 2) && (SPICE_INTERFACE_RECORD_MINOR >= 2))
-static void line_in_volume(HWVoiceIn *hw, struct mixeng_volume *vol)
+static void line_in_volume(HWVoiceIn *hw, Volume *vol)
{
SpiceVoiceIn *in = container_of (hw, SpiceVoiceIn, hw);
uint16_t svol[2];
- svol[0] = vol->l / ((1ULL << 16) + 1);
- svol[1] = vol->r / ((1ULL << 16) + 1);
+ assert(vol->channels == 2);
+ svol[0] = vol->vol[0] * 257;
+ svol[1] = vol->vol[1] * 257;
spice_server_record_set_volume(&in->sin, 2, svol);
spice_server_record_set_mute(&in->sin, vol->mute);
}
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 47/52] audio: replace shift in audio_pcm_info with bytes_per_frame
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (44 preceding siblings ...)
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 46/52] audio: support more than two channels in volume setting Kővágó, Zoltán
@ 2018-12-23 20:52 ` Kővágó, Zoltán
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 48/52] audio: basic support for multichannel audio Kővágó, Zoltán
` (4 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:52 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
The bit shifting trick worked because the number of bytes per frame was
always a power-of-two (since QEMU only supports mono, stereo and 8, 16
and 32 bit samples). But if we want to add support for surround sound,
this no longer holds true.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/alsaaudio.c | 10 +++---
audio/audio.c | 76 ++++++++++++++++++++---------------------
audio/audio_int.h | 3 +-
audio/coreaudio.c | 4 +--
audio/dsound_template.h | 10 +++---
audio/dsoundaudio.c | 4 +--
audio/noaudio.c | 2 +-
audio/ossaudio.c | 14 ++++----
audio/spiceaudio.c | 5 +--
audio/wavaudio.c | 6 ++--
10 files changed, 67 insertions(+), 67 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index bc097b1f31..418b888a57 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -612,7 +612,7 @@ static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
size_t pos = 0;
- size_t len_frames = len >> hw->info.shift;
+ size_t len_frames = len / hw->info.bytes_per_frame;
while (len_frames) {
char *src = advance(buf, pos);
@@ -656,7 +656,7 @@ static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
}
}
- pos += written << hw->info.shift;
+ pos += written * hw->info.bytes_per_frame;
if (written < len_frames) {
break;
}
@@ -822,7 +822,7 @@ static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
void *dst = advance(buf, pos);
snd_pcm_sframes_t nread;
- nread = snd_pcm_readi(alsa->handle, dst, len >> hw->info.shift);
+ nread = snd_pcm_readi(alsa->handle, dst, len / hw->info.bytes_per_frame);
if (nread <= 0) {
switch (nread) {
@@ -848,8 +848,8 @@ static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
}
}
- pos += nread << hw->info.shift;
- len -= nread << hw->info.shift;
+ pos += nread * hw->info.bytes_per_frame;
+ len -= nread * hw->info.bytes_per_frame;
}
return pos;
diff --git a/audio/audio.c b/audio/audio.c
index 71744fa2c8..ee508c1f15 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -298,26 +298,27 @@ static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *a
void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
{
- int bits = 8, sign = 0, shift = 0;
+ int bits = 8, sign = 0, mul;
switch (as->fmt) {
case AUDIO_FORMAT_S8:
sign = 1;
case AUDIO_FORMAT_U8:
+ mul = 1;
break;
case AUDIO_FORMAT_S16:
sign = 1;
case AUDIO_FORMAT_U16:
bits = 16;
- shift = 1;
+ mul = 2;
break;
case AUDIO_FORMAT_S32:
sign = 1;
case AUDIO_FORMAT_U32:
bits = 32;
- shift = 2;
+ mul = 4;
break;
default:
@@ -328,9 +329,8 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
info->bits = bits;
info->sign = sign;
info->nchannels = as->nchannels;
- info->shift = (as->nchannels == 2) + shift;
- info->align = (1 << info->shift) - 1;
- info->bytes_per_second = info->freq << info->shift;
+ info->bytes_per_frame = as->nchannels * mul;
+ info->bytes_per_second = info->freq * info->bytes_per_frame;
info->swap_endianness = (as->endianness != AUDIO_HOST_ENDIANNESS);
}
@@ -341,26 +341,25 @@ void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
}
if (info->sign) {
- memset (buf, 0x00, len << info->shift);
+ memset (buf, 0x00, len * info->bytes_per_frame);
}
else {
switch (info->bits) {
case 8:
- memset (buf, 0x80, len << info->shift);
+ memset (buf, 0x80, len * info->bytes_per_frame);
break;
case 16:
{
int i;
uint16_t *p = buf;
- int shift = info->nchannels - 1;
short s = INT16_MAX;
if (info->swap_endianness) {
s = bswap16 (s);
}
- for (i = 0; i < len << shift; i++) {
+ for (i = 0; i < len * info->nchannels; i++) {
p[i] = s;
}
}
@@ -370,14 +369,13 @@ void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
{
int i;
uint32_t *p = buf;
- int shift = info->nchannels - 1;
int32_t s = INT32_MAX;
if (info->swap_endianness) {
s = bswap32 (s);
}
- for (i = 0; i < len << shift; i++) {
+ for (i = 0; i < len * info->nchannels; i++) {
p[i] = s;
}
}
@@ -560,7 +558,7 @@ static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len)
while (len) {
st_sample *src = hw->mix_buf->samples + pos;
- uint8_t *dst = advance (pcm_buf, clipped << hw->info.shift);
+ uint8_t *dst = advance(pcm_buf, clipped * hw->info.bytes_per_frame);
size_t samples_till_end_of_buf = hw->mix_buf->size - pos;
size_t samples_to_clip = MIN(len, samples_till_end_of_buf);
@@ -609,7 +607,7 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
return 0;
}
- samples = size >> sw->info.shift;
+ samples = size / sw->info.bytes_per_frame;
if (!live) {
return 0;
}
@@ -644,7 +642,7 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
sw->clip (buf, sw->buf, ret);
sw->total_hw_samples_acquired += total;
- return ret << sw->info.shift;
+ return ret * sw->info.bytes_per_frame;
}
/*
@@ -717,7 +715,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
}
wpos = (sw->hw->mix_buf->pos + live) % hwsamples;
- samples = size >> sw->info.shift;
+ samples = size / sw->info.bytes_per_frame;
dead = hwsamples - live;
swlim = ((int64_t) dead << 32) / sw->ratio;
@@ -761,13 +759,13 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
dolog (
"%s: write size %zu ret %zu total sw %zu\n",
SW_NAME (sw),
- size >> sw->info.shift,
+ size / sw->info.bytes_per_frame,
ret,
sw->total_hw_samples_mixed
);
#endif
- return ret << sw->info.shift;
+ return ret * sw->info.bytes_per_frame;
}
#ifdef DEBUG_AUDIO
@@ -884,7 +882,7 @@ size_t AUD_read(SWVoiceIn *sw, void *buf, size_t size)
int AUD_get_buffer_size_out (SWVoiceOut *sw)
{
- return sw->hw->mix_buf->size << sw->hw->info.shift;
+ return sw->hw->mix_buf->size * sw->hw->info.bytes_per_frame;
}
void AUD_set_active_out (SWVoiceOut *sw, int on)
@@ -1000,10 +998,10 @@ static size_t audio_get_avail (SWVoiceIn *sw)
ldebug (
"%s: get_avail live %d ret %" PRId64 "\n",
SW_NAME (sw),
- live, (((int64_t) live << 32) / sw->ratio) << sw->info.shift
+ live, (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame
);
- return (((int64_t) live << 32) / sw->ratio) << sw->info.shift;
+ return (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame;
}
static size_t audio_get_free(SWVoiceOut *sw)
@@ -1027,10 +1025,11 @@ static size_t audio_get_free(SWVoiceOut *sw)
#ifdef DEBUG_OUT
dolog ("%s: get_free live %d dead %d ret %" PRId64 "\n",
SW_NAME (sw),
- live, dead, (((int64_t) dead << 32) / sw->ratio) << sw->info.shift);
+ live, dead, (((int64_t) dead << 32) / sw->ratio) *
+ sw->info.bytes_per_frame);
#endif
- return (((int64_t) dead << 32) / sw->ratio) << sw->info.shift;
+ return (((int64_t) dead << 32) / sw->ratio) * sw->info.bytes_per_frame;
}
static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
@@ -1049,7 +1048,7 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
while (n) {
size_t till_end_of_hw = hw->mix_buf->size - rpos2;
size_t to_write = MIN(till_end_of_hw, n);
- size_t bytes = to_write << hw->info.shift;
+ size_t bytes = to_write * hw->info.bytes_per_frame;
size_t written;
sw->buf = hw->mix_buf->samples + rpos2;
@@ -1079,10 +1078,11 @@ static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
size_t size, decr, proc;
void *buf = hw->pcm_ops->get_buffer_out(hw, &size);
- decr = MIN(size >> hw->info.shift, live);
+ decr = MIN(size / hw->info.bytes_per_frame, live);
audio_pcm_hw_clip_out(hw, buf, decr);
- proc = hw->pcm_ops->put_buffer_out(hw, buf, decr << hw->info.shift) >>
- hw->info.shift;
+ proc = hw->pcm_ops->put_buffer_out(hw, buf,
+ decr * hw->info.bytes_per_frame) /
+ hw->info.bytes_per_frame;
live -= proc;
clipped += proc;
@@ -1231,16 +1231,16 @@ static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
while (samples) {
size_t proc;
- size_t size = samples << hw->info.shift;
+ size_t size = samples * hw->info.bytes_per_frame;
void *buf = hw->pcm_ops->get_buffer_in(hw, &size);
- assert((size & hw->info.align) == 0);
+ assert(size % hw->info.bytes_per_frame == 0);
if (size == 0) {
hw->pcm_ops->put_buffer_in(hw, buf, size);
break;
}
- proc = MIN(size >> hw->info.shift,
+ proc = MIN(size / hw->info.bytes_per_frame,
conv_buf->size - conv_buf->pos);
hw->conv(conv_buf->samples + conv_buf->pos, buf, proc);
@@ -1248,7 +1248,7 @@ static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
samples -= proc;
conv += proc;
- hw->pcm_ops->put_buffer_in(hw, buf, proc << hw->info.shift);
+ hw->pcm_ops->put_buffer_in(hw, buf, proc * hw->info.bytes_per_frame);
}
return conv;
@@ -1322,7 +1322,7 @@ static void audio_run_capture (AudioState *s)
for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
cb->ops.capture (cb->opaque, cap->buf,
- to_capture << hw->info.shift);
+ to_capture * hw->info.bytes_per_frame);
}
rpos = (rpos + to_capture) % hw->mix_buf->size;
live -= to_capture;
@@ -1375,7 +1375,7 @@ void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size)
ssize_t start;
if (unlikely(!hw->buf_emul)) {
- size_t calc_size = hw->conv_buf->size << hw->info.shift;
+ size_t calc_size = hw->conv_buf->size * hw->info.bytes_per_frame;
hw->buf_emul = g_malloc(calc_size);
hw->size_emul = calc_size;
hw->pos_emul = hw->pending_emul = 0;
@@ -1411,7 +1411,7 @@ void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size)
void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size)
{
if (unlikely(!hw->buf_emul)) {
- size_t calc_size = hw->mix_buf->size << hw->info.shift;
+ size_t calc_size = hw->mix_buf->size * hw->info.bytes_per_frame;
hw->buf_emul = g_malloc(calc_size);
hw->size_emul = calc_size;
@@ -1808,11 +1808,11 @@ CaptureVoiceOut *AUD_add_capture(
audio_pcm_init_info (&hw->info, as);
cap->buf = audio_calloc(__func__, hw->mix_buf->size,
- 1 << hw->info.shift);
+ hw->info.bytes_per_frame);
if (!cap->buf) {
dolog ("Could not allocate capture buffer "
"(%zu samples, each %d bytes)\n",
- hw->mix_buf->size, 1 << hw->info.shift);
+ hw->mix_buf->size, hw->info.bytes_per_frame);
goto err3;
}
@@ -2124,14 +2124,14 @@ size_t audio_rate_get_bytes(struct audio_pcm_info *info, RateCtl *rate,
now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
ticks = now - rate->start_ticks;
bytes = muldiv64(ticks, info->bytes_per_second, NANOSECONDS_PER_SECOND);
- samples = (bytes - rate->bytes_sent) >> info->shift;
+ samples = (bytes - rate->bytes_sent) / info->bytes_per_frame;
if (samples < 0 || samples > 65536) {
AUD_log(NULL, "Resetting rate control (%" PRId64 " samples)", samples);
audio_rate_start(rate);
samples = 0;
}
- ret = MIN(samples << info->shift, bytes_avail);
+ ret = MIN(samples * info->bytes_per_frame, bytes_avail);
rate->bytes_sent += ret;
return ret;
}
diff --git a/audio/audio_int.h b/audio/audio_int.h
index c2f6da5cc2..cdd37908e0 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -43,8 +43,7 @@ struct audio_pcm_info {
int sign;
int freq;
int nchannels;
- int align;
- int shift;
+ int bytes_per_frame;
int bytes_per_second;
int swap_endianness;
};
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index a3cdbd87ce..1a835da29d 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -441,7 +441,7 @@ static OSStatus audioDeviceIOProc(
}
frameCount = core->audioDevicePropertyBufferFrameSize;
- pending_frames = hw->pending_emul >> hw->info.shift;
+ pending_frames = hw->pending_emul / hw->info.bytes_per_frame;
/* if there are not enough samples, set signal and return */
if (pending_frames < frameCount) {
@@ -450,7 +450,7 @@ static OSStatus audioDeviceIOProc(
return 0;
}
- len = frameCount << hw->info.shift;
+ len = frameCount * hw->info.bytes_per_frame;
while (len) {
size_t write_len;
ssize_t start = ((ssize_t) hw->pos_emul) - hw->pending_emul;
diff --git a/audio/dsound_template.h b/audio/dsound_template.h
index 6a10b6751b..31d356d084 100644
--- a/audio/dsound_template.h
+++ b/audio/dsound_template.h
@@ -98,8 +98,8 @@ static int glue (dsound_lock_, TYPE) (
goto fail;
}
- if ((p1p && *p1p && (*blen1p & info->align)) ||
- (p2p && *p2p && (*blen2p & info->align))) {
+ if ((p1p && *p1p && (*blen1p % info->bytes_per_frame)) ||
+ (p2p && *p2p && (*blen2p % info->bytes_per_frame))) {
dolog ("DirectSound returned misaligned buffer %ld %ld\n",
*blen1p, *blen2p);
glue (dsound_unlock_, TYPE) (buf, *p1p, p2p ? *p2p : NULL, *blen1p,
@@ -247,14 +247,14 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
obt_as.endianness = 0;
audio_pcm_init_info (&hw->info, &obt_as);
- if (bc.dwBufferBytes & hw->info.align) {
+ if (bc.dwBufferBytes % hw->info.bytes_per_frame) {
dolog (
"GetCaps returned misaligned buffer size %ld, alignment %d\n",
- bc.dwBufferBytes, hw->info.align + 1
+ bc.dwBufferBytes, hw->info.bytes_per_frame
);
}
hw->size_emul = bc.dwBufferBytes;
- ds->samples = bc.dwBufferBytes >> hw->info.shift;
+ ds->samples = bc.dwBufferBytes / hw->info.bytes_per_frame;
ds->s = s;
#ifdef DEBUG_DSOUND
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index 6c10d271fe..81c2092a04 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -322,8 +322,8 @@ static void dsound_clear_sample (HWVoiceOut *hw, LPDIRECTSOUNDBUFFER dsb,
return;
}
- len1 = blen1 >> hw->info.shift;
- len2 = blen2 >> hw->info.shift;
+ len1 = blen1 / hw->info.bytes_per_frame;
+ len2 = blen2 / hw->info.bytes_per_frame;
#ifdef DEBUG_DSOUND
dolog ("clear %p,%ld,%ld %p,%ld,%ld\n",
diff --git a/audio/noaudio.c b/audio/noaudio.c
index f2938aff66..dfeb61c56c 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -88,7 +88,7 @@ static size_t no_read(HWVoiceIn *hw, void *buf, size_t size)
NoVoiceIn *no = (NoVoiceIn *) hw;
int64_t bytes = audio_rate_get_bytes(&hw->info, &no->rate, size);
- audio_pcm_info_clear_buf(&hw->info, buf, bytes >> hw->info.shift);
+ audio_pcm_info_clear_buf(&hw->info, buf, bytes / hw->info.bytes_per_frame);
return bytes;
}
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index cefc90fe58..f456732b5f 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -508,16 +508,16 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
oss->nfrags = obt.nfrags;
oss->fragsize = obt.fragsize;
- if (obt.nfrags * obt.fragsize & hw->info.align) {
+ if (obt.nfrags * obt.fragsize % hw->info.bytes_per_frame) {
dolog ("warning: Misaligned DAC buffer, size %d, alignment %d\n",
- obt.nfrags * obt.fragsize, hw->info.align + 1);
+ obt.nfrags * obt.fragsize, hw->info.bytes_per_frame);
}
- oss->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
+ oss->samples = (obt.nfrags * obt.fragsize) / hw->info.bytes_per_frame;
oss->mmapped = 0;
if (oopts->has_try_mmap && oopts->try_mmap) {
- hw->size_emul = oss->samples << hw->info.shift;
+ hw->size_emul = oss->samples * hw->info.bytes_per_frame;
hw->buf_emul = mmap (
NULL,
hw->size_emul,
@@ -652,12 +652,12 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
oss->nfrags = obt.nfrags;
oss->fragsize = obt.fragsize;
- if (obt.nfrags * obt.fragsize & hw->info.align) {
+ if (obt.nfrags * obt.fragsize % hw->info.bytes_per_frame) {
dolog ("warning: Misaligned ADC buffer, size %d, alignment %d\n",
- obt.nfrags * obt.fragsize, hw->info.align + 1);
+ obt.nfrags * obt.fragsize, hw->info.bytes_per_frame);
}
- oss->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
+ oss->samples = (obt.nfrags * obt.fragsize) / hw->info.bytes_per_frame;
oss->fd = fd;
oss->dev = dev;
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index b26d10434b..a1c80ff92a 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -133,8 +133,9 @@ static void *line_out_get_buffer(HWVoiceOut *hw, size_t *size)
out->fpos = 0;
}
- *size = audio_rate_get_bytes(&hw->info, &out->rate,
- (out->fsize - out->fpos) << hw->info.shift);
+ *size = audio_rate_get_bytes(
+ &hw->info, &out->rate,
+ (out->fsize - out->fpos) * hw->info.bytes_per_frame);
return out->frame + out->fpos;
}
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 672d5fc337..a23f6e710b 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -41,14 +41,14 @@ static size_t wav_write_out(HWVoiceOut *hw, void *buf, size_t len)
{
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
int64_t bytes = audio_rate_get_bytes(&hw->info, &wav->rate, len);
- assert(bytes >> hw->info.shift << hw->info.shift == bytes);
+ assert(bytes % hw->info.bytes_per_frame == 0);
if (bytes && fwrite(buf, bytes, 1, wav->f) != 1) {
dolog("wav_write_out: fwrite of %zu bytes failed\nReaons: %s\n",
bytes, strerror(errno));
}
- wav->total_samples += bytes >> hw->info.shift;
+ wav->total_samples += bytes / hw->info.bytes_per_frame;
return bytes;
}
@@ -131,7 +131,7 @@ static void wav_fini_out (HWVoiceOut *hw)
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
uint8_t rlen[4];
uint8_t dlen[4];
- uint32_t datalen = wav->total_samples << hw->info.shift;
+ uint32_t datalen = wav->total_samples * hw->info.bytes_per_frame;
uint32_t rifflen = datalen + 36;
if (!wav->f) {
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 48/52] audio: basic support for multichannel audio
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (45 preceding siblings ...)
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 47/52] audio: replace shift in audio_pcm_info with bytes_per_frame Kővágó, Zoltán
@ 2018-12-23 20:52 ` Kővágó, Zoltán
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 49/52] paaudio: channel-map option Kővágó, Zoltán
` (3 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:52 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Which currently only means removing some checks. Old code won't require
more than two channels, but new code will need it.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/alsaaudio.c | 7 -------
audio/audio.c | 2 +-
2 files changed, 1 insertion(+), 8 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 418b888a57..063531b743 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -498,13 +498,6 @@ static int alsa_open(bool in, struct alsa_params_req *req,
goto err;
}
- if (nchannels != 1 && nchannels != 2) {
- alsa_logerr2 (err, typ,
- "Can not handle obtained number of channels %d\n",
- nchannels);
- goto err;
- }
-
if (pdo->buffer_count) {
if (pdo->buffer_len) {
int64_t req = pdo->buffer_len * pdo->buffer_count;
diff --git a/audio/audio.c b/audio/audio.c
index ee508c1f15..782e4ed1c7 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -241,7 +241,7 @@ static int audio_validate_settings (struct audsettings *as)
{
int invalid;
- invalid = as->nchannels != 1 && as->nchannels != 2;
+ invalid = as->nchannels < 1;
invalid |= as->endianness != 0 && as->endianness != 1;
switch (as->fmt) {
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 49/52] paaudio: channel-map option
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (46 preceding siblings ...)
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 48/52] audio: basic support for multichannel audio Kővágó, Zoltán
@ 2018-12-23 20:52 ` Kővágó, Zoltán
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 50/52] usb-audio: do not count on avail bytes actually available Kővágó, Zoltán
` (2 subsequent siblings)
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:52 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann, Eric Blake, Markus Armbruster
Add an option to change the channel map used by pulseaudio. If not
specified, falls back to an OSS compatible channel map.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/paaudio.c | 18 ++++++++++++++----
qapi/audio.json | 5 ++++-
qemu-options.hx | 9 +++++++++
3 files changed, 27 insertions(+), 5 deletions(-)
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 3b4b29b38f..f472f25e8f 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -337,17 +337,27 @@ static pa_stream *qpa_simple_new (
pa_stream_direction_t dir,
const char *dev,
const pa_sample_spec *ss,
- const pa_channel_map *map,
+ const char *map,
const pa_buffer_attr *attr,
int *rerror)
{
int r;
pa_stream *stream;
pa_stream_flags_t flags;
+ pa_channel_map pa_map;
pa_threaded_mainloop_lock(c->mainloop);
- stream = pa_stream_new(c->context, name, ss, map);
+ if (map && !pa_channel_map_parse(&pa_map, map)) {
+ dolog("Invalid channel map specified: '%s'\n", map);
+ map = NULL;
+ }
+ if (!map) {
+ pa_channel_map_init_extend(&pa_map, ss->channels,
+ PA_CHANNEL_MAP_OSS);
+ }
+
+ stream = pa_stream_new(c->context, name, ss, &pa_map);
if (!stream) {
goto fail;
}
@@ -424,7 +434,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
PA_STREAM_PLAYBACK,
ppdo->has_name ? ppdo->name : NULL,
&ss,
- NULL, /* channel map */
+ ppdo->has_channel_map ? ppdo->channel_map : NULL,
&ba, /* buffering attributes */
&error
);
@@ -472,7 +482,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
PA_STREAM_RECORD,
ppdo->has_name ? ppdo->name : NULL,
&ss,
- NULL, /* channel map */
+ ppdo->has_channel_map ? ppdo->channel_map : NULL,
NULL, /* buffering attributes */
&error
);
diff --git a/qapi/audio.json b/qapi/audio.json
index 180bf207a8..f3efa4620b 100644
--- a/qapi/audio.json
+++ b/qapi/audio.json
@@ -119,11 +119,14 @@
#
# @name: #optional name of the sink/source to use
#
+# @channel-map: #optional channel map to use (default: OSS compatible map)
+#
# Since: 3.2
##
{ 'struct': 'AudiodevPaPerDirectionOptions',
'data': {
- '*name': 'str' } }
+ '*name': 'str',
+ '*channel-map': 'str' } }
##
# @AudiodevPaOptions:
diff --git a/qemu-options.hx b/qemu-options.hx
index eb65b511fa..d051c52bbb 100644
--- a/qemu-options.hx
+++ b/qemu-options.hx
@@ -468,6 +468,7 @@ DEF("audiodev", HAS_ARG, QEMU_OPTION_audiodev,
"-audiodev pa,id=id[,prop[=value][,...]]\n"
" server= PulseAudio server address\n"
" sink|source.name= sink/source device name\n"
+ " sink|source.channel-map= channel map to use\n"
#endif
#ifdef CONFIG_SDL
"-audiodev sdl,id=id[,prop[=value][,...]]\n"
@@ -621,6 +622,14 @@ Sets the PulseAudio @var{server} to connect to.
@item sink|source.name=@var{sink}
Use the specified sink/source for playback/recording.
+@item sink|source.channel-map=@var{map}
+Use the specified channel map. The default is an OSS compatible
+channel map. Do not forget to escape commas inside the map:
+
+@example
+-audiodev pa,id=example,sink.channel-map=front-left,,front-right
+@end example
+
@end table
@item -audiodev sdl,id=@var{id}[,@var{prop}[=@var{value}][,...]]
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 50/52] usb-audio: do not count on avail bytes actually available
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (47 preceding siblings ...)
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 49/52] paaudio: channel-map option Kővágó, Zoltán
@ 2018-12-23 20:52 ` Kővágó, Zoltán
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 52/52] usbaudio: change playback counters to 64 bit Kővágó, Zoltán
2018-12-25 10:43 ` [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Philippe Mathieu-Daudé
50 siblings, 0 replies; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:52 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
This assumption is no longer true when mixeng is turned off.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
hw/usb/dev-audio.c | 30 ++++++++++++++++++------------
1 file changed, 18 insertions(+), 12 deletions(-)
diff --git a/hw/usb/dev-audio.c b/hw/usb/dev-audio.c
index 589977d96c..510585ad7a 100644
--- a/hw/usb/dev-audio.c
+++ b/hw/usb/dev-audio.c
@@ -318,27 +318,28 @@ static int streambuf_put(struct streambuf *buf, USBPacket *p)
{
uint32_t free = buf->size - (buf->prod - buf->cons);
- if (!free) {
+ if (free < USBAUDIO_PACKET_SIZE) {
return 0;
}
- assert(free >= USBAUDIO_PACKET_SIZE);
+
usb_packet_copy(p, buf->data + (buf->prod % buf->size),
USBAUDIO_PACKET_SIZE);
buf->prod += USBAUDIO_PACKET_SIZE;
return USBAUDIO_PACKET_SIZE;
}
-static uint8_t *streambuf_get(struct streambuf *buf)
+static uint8_t *streambuf_get(struct streambuf *buf, size_t *len)
{
uint32_t used = buf->prod - buf->cons;
uint8_t *data;
if (!used) {
+ *len = 0;
return NULL;
}
- assert(used >= USBAUDIO_PACKET_SIZE);
data = buf->data + (buf->cons % buf->size);
- buf->cons += USBAUDIO_PACKET_SIZE;
+ *len = MIN(buf->prod - buf->cons,
+ buf->size - (buf->cons % buf->size));
return data;
}
@@ -370,16 +371,21 @@ static void output_callback(void *opaque, int avail)
USBAudioState *s = opaque;
uint8_t *data;
- for (;;) {
- if (avail < USBAUDIO_PACKET_SIZE) {
- return;
- }
- data = streambuf_get(&s->out.buf);
+ while (avail) {
+ size_t written, len;
+
+ data = streambuf_get(&s->out.buf, &len);
if (!data) {
return;
}
- AUD_write(s->out.voice, data, USBAUDIO_PACKET_SIZE);
- avail -= USBAUDIO_PACKET_SIZE;
+
+ written = AUD_write(s->out.voice, data, len);
+ avail -= written;
+ s->out.buf.cons += written;
+
+ if (written < len) {
+ return;
+ }
}
}
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* [Qemu-devel] [PATCH v2 52/52] usbaudio: change playback counters to 64 bit
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (48 preceding siblings ...)
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 50/52] usb-audio: do not count on avail bytes actually available Kővágó, Zoltán
@ 2018-12-23 20:52 ` Kővágó, Zoltán
2018-12-25 10:50 ` Philippe Mathieu-Daudé
2018-12-25 10:43 ` [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Philippe Mathieu-Daudé
50 siblings, 1 reply; 91+ messages in thread
From: Kővágó, Zoltán @ 2018-12-23 20:52 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
With stereo playback, they need about 375 minutes of continuous audio
playback to overflow, which is usually not a problem (as stopping and
later resuming playback resets the counters). But with 7.1 audio, they
only need about 95 minutes to overflow.
After the overflow, the buf->prod % USBAUDIO_PACKET_SIZE(channels)
assertion no longer holds true, which will result in overflowing the
buffer. With 64 bit variables, it would take about 762000 years to
overflow.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
hw/usb/dev-audio.c | 12 +++++++-----
1 file changed, 7 insertions(+), 5 deletions(-)
diff --git a/hw/usb/dev-audio.c b/hw/usb/dev-audio.c
index 29475a2b70..45ffc3ebb3 100644
--- a/hw/usb/dev-audio.c
+++ b/hw/usb/dev-audio.c
@@ -577,9 +577,9 @@ static const USBDesc desc_audio_multi = {
struct streambuf {
uint8_t *data;
- uint32_t size;
- uint32_t prod;
- uint32_t cons;
+ size_t size;
+ uint64_t prod;
+ uint64_t cons;
};
static void streambuf_init(struct streambuf *buf, uint32_t size,
@@ -600,12 +600,14 @@ static void streambuf_fini(struct streambuf *buf)
static int streambuf_put(struct streambuf *buf, USBPacket *p, uint32_t channels)
{
- uint32_t free = buf->size - (buf->prod - buf->cons);
+ uint64_t free = buf->size - (buf->prod - buf->cons);
if (free < USBAUDIO_PACKET_SIZE(channels)) {
return 0;
}
+ /* can happen if prod overflows */
+ assert(buf->prod % USBAUDIO_PACKET_SIZE(channels) == 0);
usb_packet_copy(p, buf->data + (buf->prod % buf->size),
USBAUDIO_PACKET_SIZE(channels));
buf->prod += USBAUDIO_PACKET_SIZE(channels);
@@ -614,7 +616,7 @@ static int streambuf_put(struct streambuf *buf, USBPacket *p, uint32_t channels)
static uint8_t *streambuf_get(struct streambuf *buf, size_t *len)
{
- uint32_t used = buf->prod - buf->cons;
+ uint64_t used = buf->prod - buf->cons;
uint8_t *data;
if (!used) {
--
2.20.1
^ permalink raw reply related [flat|nested] 91+ messages in thread* Re: [Qemu-devel] [PATCH v2 52/52] usbaudio: change playback counters to 64 bit
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 52/52] usbaudio: change playback counters to 64 bit Kővágó, Zoltán
@ 2018-12-25 10:50 ` Philippe Mathieu-Daudé
2018-12-27 22:08 ` Kővágó Zoltán
0 siblings, 1 reply; 91+ messages in thread
From: Philippe Mathieu-Daudé @ 2018-12-25 10:50 UTC (permalink / raw)
To: Kővágó, Zoltán, qemu-devel; +Cc: Gerd Hoffmann
On 12/23/18 9:52 PM, Kővágó, Zoltán wrote:
> With stereo playback, they need about 375 minutes of continuous audio
> playback to overflow, which is usually not a problem (as stopping and
> later resuming playback resets the counters). But with 7.1 audio, they
> only need about 95 minutes to overflow.
>
> After the overflow, the buf->prod % USBAUDIO_PACKET_SIZE(channels)
> assertion no longer holds true, which will result in overflowing the
> buffer. With 64 bit variables, it would take about 762000 years to
> overflow.
>
> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
> ---
> hw/usb/dev-audio.c | 12 +++++++-----
> 1 file changed, 7 insertions(+), 5 deletions(-)
>
> diff --git a/hw/usb/dev-audio.c b/hw/usb/dev-audio.c
> index 29475a2b70..45ffc3ebb3 100644
> --- a/hw/usb/dev-audio.c
> +++ b/hw/usb/dev-audio.c
> @@ -577,9 +577,9 @@ static const USBDesc desc_audio_multi = {
>
> struct streambuf {
> uint8_t *data;
> - uint32_t size;
> - uint32_t prod;
> - uint32_t cons;
> + size_t size;
> + uint64_t prod;
> + uint64_t cons;
OK.
> };
>
> static void streambuf_init(struct streambuf *buf, uint32_t size,
> @@ -600,12 +600,14 @@ static void streambuf_fini(struct streambuf *buf)
>
> static int streambuf_put(struct streambuf *buf, USBPacket *p, uint32_t channels)
> {
> - uint32_t free = buf->size - (buf->prod - buf->cons);
> + uint64_t free = buf->size - (buf->prod - buf->cons);
I'd use ssize_t here.
>
> if (free < USBAUDIO_PACKET_SIZE(channels)) {
> return 0;
> }
>
> + /* can happen if prod overflows */
> + assert(buf->prod % USBAUDIO_PACKET_SIZE(channels) == 0);
> usb_packet_copy(p, buf->data + (buf->prod % buf->size),
> USBAUDIO_PACKET_SIZE(channels));
> buf->prod += USBAUDIO_PACKET_SIZE(channels);
> @@ -614,7 +616,7 @@ static int streambuf_put(struct streambuf *buf, USBPacket *p, uint32_t channels)
>
> static uint8_t *streambuf_get(struct streambuf *buf, size_t *len)
> {
> - uint32_t used = buf->prod - buf->cons;
> + uint64_t used = buf->prod - buf->cons;
> uint8_t *data;
>
> if (!used) {
Eventually here:
ssize_t used = buf->prod - buf->cons;
if (used <= 0) {
return NULL;
}
^ permalink raw reply [flat|nested] 91+ messages in thread* Re: [Qemu-devel] [PATCH v2 52/52] usbaudio: change playback counters to 64 bit
2018-12-25 10:50 ` Philippe Mathieu-Daudé
@ 2018-12-27 22:08 ` Kővágó Zoltán
0 siblings, 0 replies; 91+ messages in thread
From: Kővágó Zoltán @ 2018-12-27 22:08 UTC (permalink / raw)
To: Philippe Mathieu-Daudé, qemu-devel; +Cc: Gerd Hoffmann
Hi,
On 2018-12-25 11:50, Philippe Mathieu-Daudé wrote:
> On 12/23/18 9:52 PM, Kővágó, Zoltán wrote:
>> With stereo playback, they need about 375 minutes of continuous audio
>> playback to overflow, which is usually not a problem (as stopping and
>> later resuming playback resets the counters). But with 7.1 audio, they
>> only need about 95 minutes to overflow.
>>
>> After the overflow, the buf->prod % USBAUDIO_PACKET_SIZE(channels)
>> assertion no longer holds true, which will result in overflowing the
>> buffer. With 64 bit variables, it would take about 762000 years to
>> overflow.
>>
>> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
>> ---
>> hw/usb/dev-audio.c | 12 +++++++-----
>> 1 file changed, 7 insertions(+), 5 deletions(-)
>>
>> diff --git a/hw/usb/dev-audio.c b/hw/usb/dev-audio.c
>> index 29475a2b70..45ffc3ebb3 100644
>> --- a/hw/usb/dev-audio.c
>> +++ b/hw/usb/dev-audio.c
>> @@ -577,9 +577,9 @@ static const USBDesc desc_audio_multi = {
>>
>> struct streambuf {
>> uint8_t *data;
>> - uint32_t size;
>> - uint32_t prod;
>> - uint32_t cons;
>> + size_t size;
>> + uint64_t prod;
>> + uint64_t cons;
>
> OK.
>
>> };
>>
>> static void streambuf_init(struct streambuf *buf, uint32_t size,
>> @@ -600,12 +600,14 @@ static void streambuf_fini(struct streambuf *buf)
>>
>> static int streambuf_put(struct streambuf *buf, USBPacket *p, uint32_t channels)
>> {
>> - uint32_t free = buf->size - (buf->prod - buf->cons);
>> + uint64_t free = buf->size - (buf->prod - buf->cons);
>
> I'd use ssize_t here.
>
Hm, I agree with the signed part, but I'm not sure about the size_t
part. Granted, there's a big problem if buf->prod - buf->cons can't be
representated on 32 bits, but still I'd rather use int64_t in this case
and prevent this potential truncation on 32-bit platforms.
>>
>> if (free < USBAUDIO_PACKET_SIZE(channels)) {
>> return 0;
>> }
>>
>> + /* can happen if prod overflows */
>> + assert(buf->prod % USBAUDIO_PACKET_SIZE(channels) == 0);
>> usb_packet_copy(p, buf->data + (buf->prod % buf->size),
>> USBAUDIO_PACKET_SIZE(channels));
>> buf->prod += USBAUDIO_PACKET_SIZE(channels);
>> @@ -614,7 +616,7 @@ static int streambuf_put(struct streambuf *buf, USBPacket *p, uint32_t channels)
>>
>> static uint8_t *streambuf_get(struct streambuf *buf, size_t *len)
>> {
>> - uint32_t used = buf->prod - buf->cons;
>> + uint64_t used = buf->prod - buf->cons;
>> uint8_t *data;
>>
>> if (!used) {
>
> Eventually here:
>
> ssize_t used = buf->prod - buf->cons;
>
> if (used <= 0) {
> return NULL;
> }
>
Regards,
Zoltan
^ permalink raw reply [flat|nested] 91+ messages in thread
* Re: [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches
2018-12-23 20:51 [Qemu-devel] [PATCH v2 00/52] Audio 5.1 patches Kővágó, Zoltán
` (49 preceding siblings ...)
2018-12-23 20:52 ` [Qemu-devel] [PATCH v2 52/52] usbaudio: change playback counters to 64 bit Kővágó, Zoltán
@ 2018-12-25 10:43 ` Philippe Mathieu-Daudé
50 siblings, 0 replies; 91+ messages in thread
From: Philippe Mathieu-Daudé @ 2018-12-25 10:43 UTC (permalink / raw)
To: Kővágó, Zoltán, qemu-devel; +Cc: Gerd Hoffmann
Hi Zoltán,
On 12/23/18 9:51 PM, Kővágó, Zoltán wrote:
> I've updated my audio patchset to the current git master. Other than that not
> much happened since my last update [1], fixed a few small problems that I
> noticed while rebasing my patches.
For a such series with refactors and typedefs changes, I recommend you
to use/setup the scripts/git.orderfile script: Headers are showed first,
then the sources, which makes review easier (no need to constantly scroll).
No need to resend, but if you have to respin, this will be appreciated.
Regards,
Phil.
^ permalink raw reply [flat|nested] 91+ messages in thread