* [Qemu-devel] [PATCH 00/51] audio 5.1 patches
@ 2016-01-14 13:45 Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 01/51] qapi: support implicit structs in OptsVisitor Kővágó, Zoltán
` (50 more replies)
0 siblings, 51 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann, Eduardo Habkost
Hi,
Sorry guys, I've disappeared for a while... But anyways, I've updated
my 5.1 audio patches to git master.
About qapi flattening. That didn't really work last time, so now I've
modified OptsVisitor to support both the old way of operation (ignoring
structs) and the new way (requiring fully qualified paths for members
inside structs). Not sure if qapi people would like the idea though...
Otherwise I just updated it to work with master. Haven't really tested
on any OS beside Linux.
Please review.
Kővágó, Zoltán (51):
qapi: support implicit structs in OptsVisitor
qapi: support nested structs in OptsVisitor
qapi: qapi for audio backends
audio: use qapi AudioFormat instead of audfmt_e
audio: -audiodev command line option: documentation
audio: -audiodev command line option basic implementation
alsaaudio: port to -audiodev config
coreaudio: port to -audiodev config
dsoundaudio: port to -audiodev config
noaudio: port to -audiodev config
ossaudio: port to -audiodev config
paaudio: port to -audiodev config
sdlaudio: port to -audiodev config
spiceaudio: port to -audiodev config
wavaudio: port to -audiodev config
audio: -audiodev command line option: cleanup
audio: reduce glob_audio_state usage
audio: basic support for multi backend audio
audio: add audiodev properties to frontends
audio: audiodev= parameters no longer optional when -audiodev present
paaudio: do not move stream when sink/source name is specified
paaudio: properly disconnect streams in fini_*
audio: remove audio_MIN, audio_MAX
audio: do not run each backend in audio_run
paaudio: fix playback glitches
audio: remove read and write pcm_ops
audio: use size_t where makes sense
audio: api for mixeng code free backends
alsaaudio: port to the new audio backend api
coreaudio: port to the new audio backend api
noaudio: port to the new audio backend api
ossaudio: port to the new audio backend api
paaudio: port to the new audio backend api
sdlaudio: port to the new audio backend api
spiceaudio: port to the new audio backend api
wavaudio: port to the new audio backend api
audio: remove remains of the old backend api
audio: unify input and output mixeng buffer management
audio: remove hw->samples, buffer_size_in/out pcm_ops
audio: common rate control code for timer based outputs
audio: split ctl_* functions into enable_* and volume_*
audio: add mixeng option (documentation)
audio: make mixeng optional
paaudio: get/put_buffer functions
audio: support more than two channels in volume setting
audio: replace shift in audio_pcm_info with bytes_per_frame
audio: basic support for multichannel audio
paaudio: channel-map option
usb-audio: do not count on avail bytes actually available
usb-audio: support more than two channels of audio
usbaudio: change playback counters to 64 bit
Makefile | 6 +-
audio/Makefile.objs | 3 +-
audio/alsaaudio.c | 751 +++++----------
audio/audio.c | 1518 ++++++++++++++++---------------
audio/audio.h | 76 +-
audio/audio_int.h | 124 +--
audio/audio_legacy.c | 333 +++++++
audio/audio_pt_int.c | 173 ----
audio/audio_pt_int.h | 22 -
audio/audio_template.h | 99 +-
audio/audio_win_int.c | 18 +-
audio/coreaudio.c | 198 ++--
audio/dsound_template.h | 63 +-
audio/dsoundaudio.c | 449 +++------
audio/mixeng.h | 11 +-
audio/noaudio.c | 98 +-
audio/ossaudio.c | 596 +++++-------
audio/paaudio.c | 881 ++++++++----------
audio/rate_template.h | 2 +-
audio/sdlaudio.c | 183 ++--
audio/spiceaudio.c | 286 ++----
audio/wavaudio.c | 158 +---
audio/wavcapture.c | 8 +-
configure | 5 -
hmp-commands.hx | 11 +-
hmp.c | 2 +-
hw/acpi/core.c | 2 +-
hw/arm/omap2.c | 2 +-
hw/audio/ac97.c | 13 +-
hw/audio/adlib.c | 7 +-
hw/audio/cs4231a.c | 11 +-
hw/audio/es1370.c | 17 +-
hw/audio/gus.c | 9 +-
hw/audio/hda-codec.c | 19 +-
hw/audio/lm4549.c | 6 +-
hw/audio/milkymist-ac97.c | 16 +-
hw/audio/pcspk.c | 5 +-
hw/audio/pl041.c | 1 +
hw/audio/sb16.c | 17 +-
hw/audio/wm8750.c | 14 +-
hw/core/qdev-properties-system.c | 57 ++
hw/input/tsc210x.c | 2 +-
hw/usb/dev-audio.c | 460 ++++++++--
include/hw/qdev-properties.h | 3 +
include/qapi/opts-visitor.h | 2 +-
monitor.c | 12 +-
net/net.c | 2 +-
numa.c | 2 +-
qapi-schema.json | 3 +
qapi/audio.json | 271 ++++++
qapi/opts-visitor.c | 147 ++-
qemu-options.hx | 242 ++++-
tests/qapi-schema/qapi-schema-test.json | 9 +-
tests/qapi-schema/qapi-schema-test.out | 4 +
tests/test-opts-visitor.c | 36 +-
ui/vnc.c | 29 +-
ui/vnc.h | 2 +
vl.c | 13 +-
58 files changed, 3835 insertions(+), 3674 deletions(-)
create mode 100644 audio/audio_legacy.c
delete mode 100644 audio/audio_pt_int.c
delete mode 100644 audio/audio_pt_int.h
create mode 100644 qapi/audio.json
--
2.7.0
^ permalink raw reply [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 01/51] qapi: support implicit structs in OptsVisitor
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 16:43 ` Eric Blake
2016-01-14 13:45 ` [Qemu-devel] [PATCH 02/51] qapi: support nested " Kővágó, Zoltán
` (49 subsequent siblings)
50 siblings, 1 reply; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 UTC (permalink / raw)
To: qemu-devel; +Cc: Michael Roth, Gerd Hoffmann, Markus Armbruster
They are required for flat unions (you still have to allocate the
structs).
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
qapi/opts-visitor.c | 15 +++++++++++++++
1 file changed, 15 insertions(+)
diff --git a/qapi/opts-visitor.c b/qapi/opts-visitor.c
index ef5fb8b..1820d54 100644
--- a/qapi/opts-visitor.c
+++ b/qapi/opts-visitor.c
@@ -149,6 +149,12 @@ opts_start_struct(Visitor *v, void **obj, const char *kind,
}
}
+static void
+opts_start_implicit_struct(Visitor *v, void **obj, size_t size, Error **errp)
+{
+ opts_start_struct(v, obj, NULL, NULL, size, errp);
+}
+
static gboolean
ghr_true(gpointer ign_key, gpointer ign_value, gpointer ign_user_data)
@@ -185,6 +191,12 @@ opts_end_struct(Visitor *v, Error **errp)
ov->fake_id_opt = NULL;
}
+static void
+opts_end_implicit_struct(Visitor *v, Error **errp)
+{
+ opts_end_struct(v, errp);
+}
+
static GQueue *
lookup_distinct(const OptsVisitor *ov, const char *name, Error **errp)
@@ -508,6 +520,9 @@ opts_visitor_new(const QemuOpts *opts)
ov->visitor.start_struct = &opts_start_struct;
ov->visitor.end_struct = &opts_end_struct;
+ ov->visitor.start_implicit_struct = &opts_start_implicit_struct;
+ ov->visitor.end_implicit_struct = &opts_end_implicit_struct;
+
ov->visitor.start_list = &opts_start_list;
ov->visitor.next_list = &opts_next_list;
ov->visitor.end_list = &opts_end_list;
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 02/51] qapi: support nested structs in OptsVisitor
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 01/51] qapi: support implicit structs in OptsVisitor Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 03/51] qapi: qapi for audio backends Kővágó, Zoltán
` (48 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 UTC (permalink / raw)
To: qemu-devel
Cc: Michael Roth, Michael S. Tsirkin, Jason Wang, Markus Armbruster,
Luiz Capitulino, Paolo Bonzini, Gerd Hoffmann,
Kővágó, Zoltán, Igor Mammedov,
Eduardo Habkost
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>
---
hmp.c | 2 +-
hw/acpi/core.c | 2 +-
include/qapi/opts-visitor.h | 2 +-
net/net.c | 2 +-
numa.c | 2 +-
qapi/opts-visitor.c | 132 ++++++++++++++++++++++++++------
tests/qapi-schema/qapi-schema-test.json | 9 ++-
tests/qapi-schema/qapi-schema-test.out | 4 +
tests/test-opts-visitor.c | 36 ++++++++-
vl.c | 2 +-
10 files changed, 163 insertions(+), 30 deletions(-)
diff --git a/hmp.c b/hmp.c
index c2b2c16..3d65529 100644
--- a/hmp.c
+++ b/hmp.c
@@ -1676,7 +1676,7 @@ void hmp_object_add(Monitor *mon, const QDict *qdict)
goto out;
}
- ov = opts_visitor_new(opts);
+ ov = opts_visitor_new(opts, false);
pdict = qdict_clone_shallow(qdict);
visit_start_struct(opts_get_visitor(ov), &dummy, NULL, NULL, 0, &err);
diff --git a/hw/acpi/core.c b/hw/acpi/core.c
index 21e113d..3220ace 100644
--- a/hw/acpi/core.c
+++ b/hw/acpi/core.c
@@ -241,7 +241,7 @@ void acpi_table_add(const QemuOpts *opts, Error **errp)
{
OptsVisitor *ov;
- ov = opts_visitor_new(opts);
+ ov = opts_visitor_new(opts, false);
visit_type_AcpiTableOptions(opts_get_visitor(ov), &hdrs, NULL, &err);
opts_visitor_cleanup(ov);
}
diff --git a/include/qapi/opts-visitor.h b/include/qapi/opts-visitor.h
index fd48c14..8c0f3dc 100644
--- a/include/qapi/opts-visitor.h
+++ b/include/qapi/opts-visitor.h
@@ -30,7 +30,7 @@ typedef struct OptsVisitor OptsVisitor;
* - values below INT64_MIN or LLONG_MIN are rejected,
* - values above INT64_MAX or LLONG_MAX are rejected.
*/
-OptsVisitor *opts_visitor_new(const QemuOpts *opts);
+OptsVisitor *opts_visitor_new(const QemuOpts *opts, bool nested);
void opts_visitor_cleanup(OptsVisitor *nv);
Visitor *opts_get_visitor(OptsVisitor *nv);
diff --git a/net/net.c b/net/net.c
index 87dd356..04dd9fe 100644
--- a/net/net.c
+++ b/net/net.c
@@ -1049,7 +1049,7 @@ int net_client_init(QemuOpts *opts, int is_netdev, Error **errp)
int ret = -1;
{
- OptsVisitor *ov = opts_visitor_new(opts);
+ OptsVisitor *ov = opts_visitor_new(opts, false);
net_visit(opts_get_visitor(ov), is_netdev, &object, &err);
opts_visitor_cleanup(ov);
diff --git a/numa.c b/numa.c
index 1710946..271fbec 100644
--- a/numa.c
+++ b/numa.c
@@ -217,7 +217,7 @@ static int parse_numa(void *opaque, QemuOpts *opts, Error **errp)
Error *err = NULL;
{
- OptsVisitor *ov = opts_visitor_new(opts);
+ OptsVisitor *ov = opts_visitor_new(opts, false);
visit_type_NumaOptions(opts_get_visitor(ov), &object, NULL, &err);
opts_visitor_cleanup(ov);
}
diff --git a/qapi/opts-visitor.c b/qapi/opts-visitor.c
index 1820d54..a21881f 100644
--- a/qapi/opts-visitor.c
+++ b/qapi/opts-visitor.c
@@ -71,6 +71,7 @@ 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
@@ -86,6 +87,9 @@ struct OptsVisitor
* not survive or escape the OptsVisitor object.
*/
QemuOpt *fake_id_opt;
+
+ /* List of field names leading to the current structure. */
+ GQueue *nested_names;
};
@@ -100,6 +104,7 @@ 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) {
@@ -127,6 +132,11 @@ opts_start_struct(Visitor *v, void **obj, const char *kind,
if (obj) {
*obj = g_malloc0(size > 0 ? size : 1);
}
+
+ if (ov->nested_names != NULL) {
+ g_queue_push_tail(ov->nested_names, (gpointer) name);
+ }
+
if (ov->depth++ > 0) {
return;
}
@@ -169,6 +179,10 @@ opts_end_struct(Visitor *v, Error **errp)
OptsVisitor *ov = DO_UPCAST(OptsVisitor, visitor, v);
GQueue *any;
+ if (ov->nested_names != NULL) {
+ g_queue_pop_tail(ov->nested_names);
+ }
+
if (--ov->depth > 0) {
return;
}
@@ -198,15 +212,59 @@ opts_end_implicit_struct(Visitor *v, Error **errp)
}
+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;
}
@@ -218,7 +276,8 @@ opts_start_list(Visitor *v, const char *name, Error **errp)
/* we can't traverse a list in a list */
assert(ov->list_mode == LM_NONE);
- ov->repeated_opts = lookup_distinct(ov, name, errp);
+ assert(ov->repeated_opts == NULL && ov->repeated_name == NULL);
+ ov->repeated_opts = lookup_distinct(ov, name, &ov->repeated_name, errp);
if (ov->repeated_opts != NULL) {
ov->list_mode = LM_STARTED;
}
@@ -254,11 +313,9 @@ opts_next_list(Visitor *v, GenericList **list, Error **errp)
/* range has been completed, fall through in order to pop option */
case LM_IN_PROGRESS: {
- const QemuOpt *opt;
-
- opt = g_queue_pop_head(ov->repeated_opts);
+ g_queue_pop_head(ov->repeated_opts);
if (g_queue_is_empty(ov->repeated_opts)) {
- g_hash_table_remove(ov->unprocessed_opts, opt->name);
+ g_hash_table_remove(ov->unprocessed_opts, ov->repeated_name);
return NULL;
}
link = &(*list)->next;
@@ -284,22 +341,28 @@ opts_end_list(Visitor *v, Error **errp)
ov->list_mode == LM_SIGNED_INTERVAL ||
ov->list_mode == LM_UNSIGNED_INTERVAL);
ov->repeated_opts = NULL;
+
+ g_free(ov->repeated_name);
+ ov->repeated_name = NULL;
+
ov->list_mode = LM_NONE;
}
static const QemuOpt *
-lookup_scalar(const OptsVisitor *ov, const char *name, Error **errp)
+lookup_scalar(const OptsVisitor *ov, const char *name, char** out_key,
+ Error **errp)
{
if (ov->list_mode == LM_NONE) {
GQueue *list;
/* the last occurrence of any QemuOpt takes effect when queried by name
*/
- list = lookup_distinct(ov, name, errp);
+ list = lookup_distinct(ov, name, out_key, errp);
return list ? g_queue_peek_tail(list) : NULL;
}
assert(ov->list_mode == LM_IN_PROGRESS);
+ assert(out_key == NULL || *out_key == NULL);
return g_queue_peek_head(ov->repeated_opts);
}
@@ -321,13 +384,15 @@ opts_type_str(Visitor *v, char **obj, const char *name, Error **errp)
{
OptsVisitor *ov = DO_UPCAST(OptsVisitor, visitor, v);
const QemuOpt *opt;
+ char *key = NULL;
- opt = lookup_scalar(ov, name, errp);
+ opt = lookup_scalar(ov, name, &key, errp);
if (!opt) {
return;
}
*obj = g_strdup(opt->str ? opt->str : "");
- processed(ov, name);
+ processed(ov, key);
+ g_free(key);
}
@@ -337,8 +402,9 @@ opts_type_bool(Visitor *v, bool *obj, const char *name, Error **errp)
{
OptsVisitor *ov = DO_UPCAST(OptsVisitor, visitor, v);
const QemuOpt *opt;
+ char *key = NULL;
- opt = lookup_scalar(ov, name, errp);
+ opt = lookup_scalar(ov, name, &key, errp);
if (!opt) {
return;
}
@@ -355,13 +421,15 @@ opts_type_bool(Visitor *v, bool *obj, const char *name, 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);
}
@@ -373,13 +441,14 @@ opts_type_int(Visitor *v, int64_t *obj, const char *name, Error **errp)
const char *str;
long long val;
char *endptr;
+ char *key = NULL;
if (ov->list_mode == LM_SIGNED_INTERVAL) {
*obj = ov->range_next.s;
return;
}
- opt = lookup_scalar(ov, name, errp);
+ opt = lookup_scalar(ov, name, &key, errp);
if (!opt) {
return;
}
@@ -393,11 +462,13 @@ opts_type_int(Visitor *v, int64_t *obj, const char *name, Error **errp)
if (errno == 0 && endptr > str && INT64_MIN <= val && val <= INT64_MAX) {
if (*endptr == '\0') {
*obj = val;
- processed(ov, name);
+ processed(ov, key);
+ g_free(key);
return;
}
if (*endptr == '-' && ov->list_mode == LM_IN_PROGRESS) {
long long val2;
+ assert(key == NULL);
str = endptr + 1;
val2 = strtoll(str, &endptr, 0);
@@ -418,6 +489,7 @@ opts_type_int(Visitor *v, int64_t *obj, const char *name, 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);
}
@@ -429,13 +501,14 @@ opts_type_uint64(Visitor *v, uint64_t *obj, const char *name, Error **errp)
const char *str;
unsigned long long val;
char *endptr;
+ char *key = NULL;
if (ov->list_mode == LM_UNSIGNED_INTERVAL) {
*obj = ov->range_next.u;
return;
}
- opt = lookup_scalar(ov, name, errp);
+ opt = lookup_scalar(ov, name, &key, errp);
if (!opt) {
return;
}
@@ -447,11 +520,13 @@ opts_type_uint64(Visitor *v, uint64_t *obj, const char *name, Error **errp)
if (parse_uint(str, &val, &endptr, 0) == 0 && val <= UINT64_MAX) {
if (*endptr == '\0') {
*obj = val;
- processed(ov, name);
+ processed(ov, key);
+ g_free(key);
return;
}
if (*endptr == '-' && ov->list_mode == LM_IN_PROGRESS) {
unsigned long long val2;
+ assert(key == NULL);
str = endptr + 1;
if (parse_uint_full(str, &val2, 0) == 0 &&
@@ -470,6 +545,7 @@ opts_type_uint64(Visitor *v, uint64_t *obj, const char *name, 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);
}
@@ -480,8 +556,9 @@ opts_type_size(Visitor *v, uint64_t *obj, const char *name, Error **errp)
const QemuOpt *opt;
int64_t val;
char *endptr;
+ char *key = NULL;
- opt = lookup_scalar(ov, name, errp);
+ opt = lookup_scalar(ov, name, &key, errp);
if (!opt) {
return;
}
@@ -491,11 +568,13 @@ opts_type_size(Visitor *v, uint64_t *obj, const char *name, Error **errp)
if (val < 0 || *endptr) {
error_setg(errp, QERR_INVALID_PARAMETER_VALUE, opt->name,
"a size value representible as a non-negative int64");
+ g_free(key);
return;
}
*obj = val;
- processed(ov, name);
+ processed(ov, key);
+ g_free(key);
}
@@ -506,17 +585,23 @@ opts_optional(Visitor *v, bool *present, const char *name)
/* 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);
}
OptsVisitor *
-opts_visitor_new(const QemuOpts *opts)
+opts_visitor_new(const QemuOpts *opts, bool nested)
{
OptsVisitor *ov;
ov = g_malloc0(sizeof *ov);
+ if (nested) {
+ ov->nested_names = g_queue_new();
+ } else {
+ ov->nested_names = NULL;
+ }
+
ov->visitor.start_struct = &opts_start_struct;
ov->visitor.end_struct = &opts_end_struct;
@@ -560,6 +645,9 @@ opts_visitor_cleanup(OptsVisitor *ov)
if (ov->unprocessed_opts != NULL) {
g_hash_table_destroy(ov->unprocessed_opts);
}
+ if (ov->nested_names != NULL) {
+ g_queue_free(ov->nested_names);
+ }
g_free(ov->fake_id_opt);
g_free(ov);
}
diff --git a/tests/qapi-schema/qapi-schema-test.json b/tests/qapi-schema/qapi-schema-test.json
index 4b89527..bd2bf0f 100644
--- a/tests/qapi-schema/qapi-schema-test.json
+++ b/tests/qapi-schema/qapi-schema-test.json
@@ -125,6 +125,11 @@
'returns': 'int' }
{ 'command': 'guest-sync', 'data': { 'arg': 'any' }, 'returns': 'any' }
+# 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:
#
@@ -138,7 +143,9 @@
'*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 2c546b7..9821063 100644
--- a/tests/qapi-schema/qapi-schema-test.out
+++ b/tests/qapi-schema/qapi-schema-test.out
@@ -153,6 +153,10 @@ object UserDefOptions
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 UserDefOptionsSub
+ member nint: int optional=True
object UserDefTwo
member string0: str optional=False
member dict1: UserDefTwoDict optional=False
diff --git a/tests/test-opts-visitor.c b/tests/test-opts-visitor.c
index 1c753d9..58ca41a 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);
- ov = opts_visitor_new(opts);
+ ov = opts_visitor_new(opts, true);
visit_type_UserDefOptions(opts_get_visitor(ov), &f->userdef, NULL,
&f->err);
opts_visitor_cleanup(ov);
@@ -178,6 +178,34 @@ 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 */
int
@@ -271,6 +299,12 @@ main(int argc, char **argv)
add_test("/visitor/opts/i64/range/2big/full", &expect_fail,
"i64=-0x8000000000000000-0x7fffffffffffffff");
+ /* Test nested structs support */
+ add_test("/visitor/opts/nested/unqualified", &expect_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;
}
diff --git a/vl.c b/vl.c
index 5aaea77..eb7edb0 100644
--- a/vl.c
+++ b/vl.c
@@ -2835,7 +2835,7 @@ static int object_create(void *opaque, QemuOpts *opts, Error **errp)
QDict *pdict;
bool (*type_predicate)(const char *) = opaque;
- ov = opts_visitor_new(opts);
+ ov = opts_visitor_new(opts, false);
pdict = qemu_opts_to_qdict(opts, NULL);
visit_start_struct(opts_get_visitor(ov), &dummy, NULL, NULL, 0, &err);
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 03/51] qapi: qapi for audio backends
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 01/51] qapi: support implicit structs in OptsVisitor Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 02/51] qapi: support nested " Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 16:51 ` Eric Blake
2016-01-14 13:45 ` [Qemu-devel] [PATCH 04/51] audio: use qapi AudioFormat instead of audfmt_e Kővágó, Zoltán
` (47 subsequent siblings)
50 siblings, 1 reply; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann, 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 | 6 +-
qapi-schema.json | 3 +
qapi/audio.json | 263 +++++++++++++++++++++++++++++++++++++++++++++++++++++++
3 files changed, 269 insertions(+), 3 deletions(-)
create mode 100644 qapi/audio.json
diff --git a/Makefile b/Makefile
index d0de2d4..d87c1a9 100644
--- a/Makefile
+++ b/Makefile
@@ -270,9 +270,9 @@ $(SRC_PATH)/qga/qapi-schema.json $(SRC_PATH)/scripts/qapi-commands.py $(qapi-py)
" GEN $@")
qapi-modules = $(SRC_PATH)/qapi-schema.json $(SRC_PATH)/qapi/common.json \
- $(SRC_PATH)/qapi/block.json $(SRC_PATH)/qapi/block-core.json \
- $(SRC_PATH)/qapi/event.json $(SRC_PATH)/qapi/introspect.json \
- $(SRC_PATH)/qapi/crypto.json
+ $(SRC_PATH)/qapi/audio.json $(SRC_PATH)/qapi/block.json \
+ $(SRC_PATH)/qapi/block-core.json $(SRC_PATH)/qapi/event.json \
+ $(SRC_PATH)/qapi/introspect.json $(SRC_PATH)/qapi/crypto.json
qapi-types.c qapi-types.h :\
$(qapi-modules) $(SRC_PATH)/scripts/qapi-types.py $(qapi-py)
diff --git a/qapi-schema.json b/qapi-schema.json
index 2e31733..7b13cfb 100644
--- a/qapi-schema.json
+++ b/qapi-schema.json
@@ -5,6 +5,9 @@
# QAPI common definitions
{ 'include': 'qapi/common.json' }
+# QAPI audio definitions
+{ 'include': 'qapi/audio.json' }
+
# QAPI crypto definitions
{ 'include': 'qapi/crypto.json' }
diff --git a/qapi/audio.json b/qapi/audio.json
new file mode 100644
index 0000000..da0c2df
--- /dev/null
+++ b/qapi/audio.json
@@ -0,0 +1,263 @@
+# -*- 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: 2.6
+##
+{ '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: 2.6
+##
+{ '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: 2.6
+##
+{ '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: 2.6
+##
+{ '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: 2.6
+##
+{ '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: 2.6
+##
+{ '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: 2.6
+##
+{ '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: 2.6
+##
+{ '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: 2.6
+##
+{ 'struct': 'AudiodevWavOptions',
+ 'data': {
+ '*path': 'str' } }
+
+
+##
+# @AudioFormat
+#
+# An enumeration of possible audio formats.
+#
+# Since: 2.6
+##
+{ 'enum': 'AudioFormat',
+ 'data': [ 'u8', 's8', 'u16', 's16', 'u32', 's32' ] }
+
+##
+# @AudiodevDriver
+#
+# An enumeration of possible audio backend drivers.
+#
+# Since: 2.6
+##
+{ '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: 2.6
+##
+{ 'struct': 'AudiodevPerDirectionOptions',
+ 'data': {
+ '*fixed-settings': 'bool',
+ '*frequency': 'int',
+ '*channels': 'int',
+ '*voices': 'int',
+ '*format': 'AudioFormat',
+ '*buffer-len': 'int',
+ '*buffer-count': 'int' } }
+
+##
+# @AudiodevCommonOptions
+#
+# Captures the common configuration 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: 2.6
+##
+{ 'struct': 'AudiodevCommonOptions',
+ 'data': {
+ 'id': 'str',
+ 'driver': 'AudiodevDriver',
+ 'in': 'AudiodevPerDirectionOptions',
+ 'out': 'AudiodevPerDirectionOptions',
+ '*timer-period': 'int' } }
+
+##
+# @AudiodevBackendOptions
+#
+# Options of an audio backend.
+#
+# Since: 2.6
+##
+{ 'union': 'Audiodev',
+ 'base': 'AudiodevCommonOptions',
+ 'discriminator': 'driver',
+ 'data': {
+ 'none': 'AudiodevNoOptions',
+ 'alsa': 'AudiodevAlsaOptions',
+ 'coreaudio': 'AudiodevNoOptions',
+ 'dsound': 'AudiodevDsoundOptions',
+ 'oss': 'AudiodevOssOptions',
+ 'pa': 'AudiodevPaOptions',
+ 'sdl': 'AudiodevNoOptions',
+ 'spice': 'AudiodevNoOptions',
+ 'wav': 'AudiodevWavOptions' } }
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 04/51] audio: use qapi AudioFormat instead of audfmt_e
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (2 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 03/51] qapi: qapi for audio backends Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 05/51] audio: -audiodev command line option: documentation Kővágó, Zoltán
` (46 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 UTC (permalink / raw)
To: qemu-devel; +Cc: Peter Maydell, Michael Walle, open list:ARM, Gerd Hoffmann
I had to include an enum for audio sampling formats into qapi, but that
meant duplicating the audfmt_e enum. This patch replaces audfmt_e and
associated values with the qapi generated AudioFormat enum.
This patch is mostly a search-and-replace, except for switches where the
qapi generated AUDIO_FORMAT_MAX caused problems.
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/alsaaudio.c | 53 ++++++++++++++------------
audio/audio.c | 97 ++++++++++++++++++++++++++---------------------
audio/audio.h | 11 +-----
audio/audio_win_int.c | 18 ++++-----
audio/ossaudio.c | 30 +++++++--------
audio/paaudio.c | 28 +++++++-------
audio/sdlaudio.c | 26 ++++++-------
audio/spiceaudio.c | 4 +-
audio/wavaudio.c | 17 +++++----
audio/wavcapture.c | 2 +-
hw/arm/omap2.c | 2 +-
hw/audio/ac97.c | 2 +-
hw/audio/adlib.c | 2 +-
hw/audio/cs4231a.c | 6 +--
hw/audio/es1370.c | 4 +-
hw/audio/gus.c | 2 +-
hw/audio/hda-codec.c | 18 ++++-----
hw/audio/lm4549.c | 6 +--
hw/audio/milkymist-ac97.c | 2 +-
hw/audio/pcspk.c | 2 +-
hw/audio/sb16.c | 14 +++----
hw/audio/wm8750.c | 4 +-
hw/input/tsc210x.c | 2 +-
hw/usb/dev-audio.c | 2 +-
ui/vnc.c | 14 +++----
25 files changed, 187 insertions(+), 181 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 6315b2d..2b28b99 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -88,7 +88,7 @@ struct alsa_params_req {
struct alsa_params_obt {
int freq;
- audfmt_e fmt;
+ AudioFormat fmt;
int endianness;
int nchannels;
snd_pcm_uframes_t samples;
@@ -295,16 +295,16 @@ 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;
}
@@ -312,7 +312,7 @@ 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;
}
@@ -320,7 +320,7 @@ 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;
}
@@ -328,7 +328,7 @@ 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;
}
@@ -345,58 +345,58 @@ 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:
@@ -639,19 +639,22 @@ 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 5be4b15..334c935 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -75,7 +75,7 @@ static struct {
.settings = {
.freq = 44100,
.nchannels = 2,
- .fmt = AUD_FMT_S16,
+ .fmt = AUDIO_FORMAT_S16,
.endianness = AUDIO_HOST_ENDIANNESS,
}
},
@@ -87,7 +87,7 @@ static struct {
.settings = {
.freq = 44100,
.nchannels = 2,
- .fmt = AUD_FMT_S16,
+ .fmt = AUDIO_FORMAT_S16,
.endianness = AUDIO_HOST_ENDIANNESS,
}
},
@@ -219,58 +219,61 @@ static char *audio_alloc_prefix (const char *s)
return r;
}
-static const char *audio_audfmt_to_string (audfmt_e fmt)
+static const char *audio_audfmt_to_string (AudioFormat fmt)
{
switch (fmt) {
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
return "U8";
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
return "U16";
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
return "S8";
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
return "S16";
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_U32:
return "U32";
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_S32:
return "S32";
+
+ 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",
@@ -280,8 +283,8 @@ static audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval,
}
}
-static audfmt_e audio_get_conf_fmt (const char *envname,
- audfmt_e defval,
+static AudioFormat audio_get_conf_fmt (const char *envname,
+ AudioFormat defval,
int *defaultp)
{
const char *var = getenv (envname);
@@ -384,7 +387,7 @@ static void audio_print_options (const char *prefix,
case AUD_OPT_FMT:
{
- audfmt_e *fmtp = opt->valp;
+ AudioFormat *fmtp = opt->valp;
printf (
"format, %s = %s, (one of: U8 S8 U16 S16 U32 S32)\n",
state,
@@ -471,7 +474,7 @@ static void audio_process_options (const char *prefix,
case AUD_OPT_FMT:
{
- audfmt_e *fmtp = opt->valp;
+ AudioFormat *fmtp = opt->valp;
*fmtp = audio_get_conf_fmt (optname, *fmtp, &def);
}
break;
@@ -502,22 +505,22 @@ static void audio_print_settings (struct audsettings *as)
dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels);
switch (as->fmt) {
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
AUD_log (NULL, "S8");
break;
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
AUD_log (NULL, "U8");
break;
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
AUD_log (NULL, "S16");
break;
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
AUD_log (NULL, "U16");
break;
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_S32:
AUD_log (NULL, "S32");
break;
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_U32:
AUD_log (NULL, "U32");
break;
default:
@@ -548,12 +551,12 @@ static int audio_validate_settings (struct audsettings *as)
invalid |= as->endianness != 0 && as->endianness != 1;
switch (as->fmt) {
- case AUD_FMT_S8:
- case AUD_FMT_U8:
- case AUD_FMT_S16:
- case AUD_FMT_U16:
- case AUD_FMT_S32:
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_S8:
+ case AUDIO_FORMAT_U8:
+ case AUDIO_FORMAT_S16:
+ case AUDIO_FORMAT_U16:
+ case AUDIO_FORMAT_S32:
+ case AUDIO_FORMAT_U32:
break;
default:
invalid = 1;
@@ -569,25 +572,28 @@ static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *a
int bits = 8, sign = 0;
switch (as->fmt) {
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
sign = 1;
/* fall through */
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
break;
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
sign = 1;
/* fall through */
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
bits = 16;
break;
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_S32:
sign = 1;
/* fall through */
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_U32:
bits = 32;
break;
+
+ default:
+ abort();
}
return info->freq == as->freq
&& info->nchannels == as->nchannels
@@ -601,24 +607,27 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
int bits = 8, sign = 0, shift = 0;
switch (as->fmt) {
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
sign = 1;
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
break;
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
sign = 1;
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
bits = 16;
shift = 1;
break;
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_S32:
sign = 1;
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_U32:
bits = 32;
shift = 2;
break;
+
+ default:
+ abort();
}
info->freq = as->freq;
diff --git a/audio/audio.h b/audio/audio.h
index e7ea397..e300511 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -29,15 +29,6 @@
typedef void (*audio_callback_fn) (void *opaque, int avail);
-typedef enum {
- AUD_FMT_U8,
- AUD_FMT_S8,
- AUD_FMT_U16,
- AUD_FMT_S16,
- AUD_FMT_U32,
- AUD_FMT_S32
-} audfmt_e;
-
#ifdef HOST_WORDS_BIGENDIAN
#define AUDIO_HOST_ENDIANNESS 1
#else
@@ -47,7 +38,7 @@ typedef enum {
struct audsettings {
int freq;
int nchannels;
- audfmt_e fmt;
+ AudioFormat fmt;
int endianness;
};
diff --git a/audio/audio_win_int.c b/audio/audio_win_int.c
index e132405..a8cfa77 100644
--- a/audio/audio_win_int.c
+++ b/audio/audio_win_int.c
@@ -23,20 +23,20 @@ int waveformat_from_audio_settings (WAVEFORMATEX *wfx,
wfx->cbSize = 0;
switch (as->fmt) {
- case AUD_FMT_S8:
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_S8:
+ case AUDIO_FORMAT_U8:
wfx->wBitsPerSample = 8;
break;
- case AUD_FMT_S16:
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_S16:
+ case AUDIO_FORMAT_U16:
wfx->wBitsPerSample = 16;
wfx->nAvgBytesPerSec <<= 1;
wfx->nBlockAlign <<= 1;
break;
- case AUD_FMT_S32:
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_S32:
+ case AUDIO_FORMAT_U32:
wfx->wBitsPerSample = 32;
wfx->nAvgBytesPerSec <<= 2;
wfx->nBlockAlign <<= 2;
@@ -84,15 +84,15 @@ int waveformat_to_audio_settings (WAVEFORMATEX *wfx,
switch (wfx->wBitsPerSample) {
case 8:
- as->fmt = AUD_FMT_U8;
+ as->fmt = AUDIO_FORMAT_U8;
break;
case 16:
- as->fmt = AUD_FMT_S16;
+ as->fmt = AUDIO_FORMAT_S16;
break;
case 32:
- as->fmt = AUD_FMT_S32;
+ as->fmt = AUDIO_FORMAT_S32;
break;
default:
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 7dbe333..3ea7d27 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -72,7 +72,7 @@ typedef struct OSSVoiceIn {
struct oss_params {
int freq;
- audfmt_e fmt;
+ AudioFormat fmt;
int nchannels;
int nfrags;
int fragsize;
@@ -150,16 +150,16 @@ static int oss_write (SWVoiceOut *sw, void *buf, int len)
return audio_pcm_sw_write (sw, buf, len);
}
-static int aud_to_ossfmt (audfmt_e fmt, int endianness)
+static int aud_to_ossfmt (AudioFormat fmt, int endianness)
{
switch (fmt) {
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
return AFMT_S8;
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
return AFMT_U8;
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
if (endianness) {
return AFMT_S16_BE;
}
@@ -167,7 +167,7 @@ static int aud_to_ossfmt (audfmt_e fmt, int endianness)
return AFMT_S16_LE;
}
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
if (endianness) {
return AFMT_U16_BE;
}
@@ -184,37 +184,37 @@ static int aud_to_ossfmt (audfmt_e fmt, int endianness)
}
}
-static int oss_to_audfmt (int ossfmt, audfmt_e *fmt, int *endianness)
+static int oss_to_audfmt (int ossfmt, AudioFormat *fmt, int *endianness)
{
switch (ossfmt) {
case AFMT_S8:
*endianness = 0;
- *fmt = AUD_FMT_S8;
+ *fmt = AUDIO_FORMAT_S8;
break;
case AFMT_U8:
*endianness = 0;
- *fmt = AUD_FMT_U8;
+ *fmt = AUDIO_FORMAT_U8;
break;
case AFMT_S16_LE:
*endianness = 0;
- *fmt = AUD_FMT_S16;
+ *fmt = AUDIO_FORMAT_S16;
break;
case AFMT_U16_LE:
*endianness = 0;
- *fmt = AUD_FMT_U16;
+ *fmt = AUDIO_FORMAT_U16;
break;
case AFMT_S16_BE:
*endianness = 1;
- *fmt = AUD_FMT_S16;
+ *fmt = AUDIO_FORMAT_S16;
break;
case AFMT_U16_BE:
*endianness = 1;
- *fmt = AUD_FMT_U16;
+ *fmt = AUDIO_FORMAT_U16;
break;
default:
@@ -502,7 +502,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
int endianness;
int err;
int fd;
- audfmt_e effective_fmt;
+ AudioFormat effective_fmt;
struct audsettings obt_as;
OSSConf *conf = drv_opaque;
@@ -671,7 +671,7 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
int endianness;
int err;
int fd;
- audfmt_e effective_fmt;
+ AudioFormat effective_fmt;
struct audsettings obt_as;
OSSConf *conf = drv_opaque;
diff --git a/audio/paaudio.c b/audio/paaudio.c
index fea6071..cfdbdc6 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -384,21 +384,21 @@ static int qpa_read (SWVoiceIn *sw, void *buf, int len)
return audio_pcm_sw_read (sw, buf, len);
}
-static pa_sample_format_t audfmt_to_pa (audfmt_e afmt, int endianness)
+static pa_sample_format_t audfmt_to_pa (AudioFormat afmt, int endianness)
{
int format;
switch (afmt) {
- case AUD_FMT_S8:
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_S8:
+ case AUDIO_FORMAT_U8:
format = PA_SAMPLE_U8;
break;
- case AUD_FMT_S16:
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_S16:
+ case AUDIO_FORMAT_U16:
format = endianness ? PA_SAMPLE_S16BE : PA_SAMPLE_S16LE;
break;
- case AUD_FMT_S32:
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_S32:
+ case AUDIO_FORMAT_U32:
format = endianness ? PA_SAMPLE_S32BE : PA_SAMPLE_S32LE;
break;
default:
@@ -409,26 +409,26 @@ static pa_sample_format_t audfmt_to_pa (audfmt_e afmt, int endianness)
return format;
}
-static audfmt_e pa_to_audfmt (pa_sample_format_t fmt, int *endianness)
+static AudioFormat pa_to_audfmt (pa_sample_format_t fmt, int *endianness)
{
switch (fmt) {
case PA_SAMPLE_U8:
- return AUD_FMT_U8;
+ return AUDIO_FORMAT_U8;
case PA_SAMPLE_S16BE:
*endianness = 1;
- return AUD_FMT_S16;
+ return AUDIO_FORMAT_S16;
case PA_SAMPLE_S16LE:
*endianness = 0;
- return AUD_FMT_S16;
+ return AUDIO_FORMAT_S16;
case PA_SAMPLE_S32BE:
*endianness = 1;
- return AUD_FMT_S32;
+ return AUDIO_FORMAT_S32;
case PA_SAMPLE_S32LE:
*endianness = 0;
- return AUD_FMT_S32;
+ return AUDIO_FORMAT_S32;
default:
dolog ("Internal logic error: Bad pa_sample_format %d\n", fmt);
- return AUD_FMT_U8;
+ return AUDIO_FORMAT_U8;
}
}
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index 1140f2e..db0f95a 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -115,19 +115,19 @@ static int sdl_unlock_and_post (SDLAudioState *s, const char *forfn)
return sdl_post (s, forfn);
}
-static int aud_to_sdlfmt (audfmt_e fmt)
+static int aud_to_sdlfmt (AudioFormat fmt)
{
switch (fmt) {
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
return AUDIO_S8;
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
return AUDIO_U8;
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
return AUDIO_S16LSB;
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
return AUDIO_U16LSB;
default:
@@ -139,37 +139,37 @@ static int aud_to_sdlfmt (audfmt_e fmt)
}
}
-static int sdl_to_audfmt(int sdlfmt, audfmt_e *fmt, int *endianness)
+static int sdl_to_audfmt(int sdlfmt, AudioFormat *fmt, int *endianness)
{
switch (sdlfmt) {
case AUDIO_S8:
*endianness = 0;
- *fmt = AUD_FMT_S8;
+ *fmt = AUDIO_FORMAT_S8;
break;
case AUDIO_U8:
*endianness = 0;
- *fmt = AUD_FMT_U8;
+ *fmt = AUDIO_FORMAT_U8;
break;
case AUDIO_S16LSB:
*endianness = 0;
- *fmt = AUD_FMT_S16;
+ *fmt = AUDIO_FORMAT_S16;
break;
case AUDIO_U16LSB:
*endianness = 0;
- *fmt = AUD_FMT_U16;
+ *fmt = AUDIO_FORMAT_U16;
break;
case AUDIO_S16MSB:
*endianness = 1;
- *fmt = AUD_FMT_S16;
+ *fmt = AUDIO_FORMAT_S16;
break;
case AUDIO_U16MSB:
*endianness = 1;
- *fmt = AUD_FMT_U16;
+ *fmt = AUDIO_FORMAT_U16;
break;
default:
@@ -341,7 +341,7 @@ static int sdl_init_out(HWVoiceOut *hw, struct audsettings *as,
SDL_AudioSpec req, obt;
int endianness;
int err;
- audfmt_e effective_fmt;
+ AudioFormat effective_fmt;
struct audsettings obt_as;
req.freq = as->freq;
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index 42ae4a4..141fd8d 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -128,7 +128,7 @@ 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);
@@ -256,7 +256,7 @@ static int line_in_init(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
settings.freq = SPICE_INTERFACE_RECORD_FREQ;
#endif
settings.nchannels = SPICE_INTERFACE_RECORD_CHAN;
- settings.fmt = AUD_FMT_S16;
+ settings.fmt = AUDIO_FORMAT_S16;
settings.endianness = AUDIO_HOST_ENDIANNESS;
audio_pcm_init_info (&hw->info, &settings);
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index c586020..81250e6 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -116,20 +116,23 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
stereo = wav_as.nchannels == 2;
switch (wav_as.fmt) {
- case AUD_FMT_S8:
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_S8:
+ case AUDIO_FORMAT_U8:
bits16 = 0;
break;
- case AUD_FMT_S16:
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_S16:
+ case AUDIO_FORMAT_U16:
bits16 = 1;
break;
- case AUD_FMT_S32:
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_S32:
+ case AUDIO_FORMAT_U32:
dolog ("WAVE files can not handle 32bit formats\n");
return -1;
+
+ default:
+ abort();
}
hdr[34] = bits16 ? 0x10 : 0x08;
@@ -224,7 +227,7 @@ static int wav_ctl_out (HWVoiceOut *hw, int cmd, ...)
static WAVConf glob_conf = {
.settings.freq = 44100,
.settings.nchannels = 2,
- .settings.fmt = AUD_FMT_S16,
+ .settings.fmt = AUDIO_FORMAT_S16,
.wav_path = "qemu.wav"
};
diff --git a/audio/wavcapture.c b/audio/wavcapture.c
index 86e9056..798acdd 100644
--- a/audio/wavcapture.c
+++ b/audio/wavcapture.c
@@ -136,7 +136,7 @@ int wav_start_capture (CaptureState *s, const char *path, int freq,
as.freq = freq;
as.nchannels = 1 << stereo;
- as.fmt = bits16 ? AUD_FMT_S16 : AUD_FMT_U8;
+ as.fmt = bits16 ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_U8;
as.endianness = 0;
ops.notify = wav_notify;
diff --git a/hw/arm/omap2.c b/hw/arm/omap2.c
index 98ee19f..f1abf95 100644
--- a/hw/arm/omap2.c
+++ b/hw/arm/omap2.c
@@ -269,7 +269,7 @@ static void omap_eac_format_update(struct omap_eac_s *s)
* does I2S specify it? */
/* All register writes are 16 bits so we we store 16-bit samples
* in the buffers regardless of AGCFR[B8_16] value. */
- fmt.fmt = AUD_FMT_U16;
+ fmt.fmt = AUDIO_FORMAT_U16;
s->codec.in_voice = AUD_open_in(&s->codec.card, s->codec.in_voice,
"eac.codec.in", s, omap_eac_in_cb, &fmt);
diff --git a/hw/audio/ac97.c b/hw/audio/ac97.c
index b173835..fa75f33 100644
--- a/hw/audio/ac97.c
+++ b/hw/audio/ac97.c
@@ -360,7 +360,7 @@ static void open_voice (AC97LinkState *s, int index, int freq)
as.freq = freq;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = 0;
if (freq > 0) {
diff --git a/hw/audio/adlib.c b/hw/audio/adlib.c
index 334935f..ca8de05 100644
--- a/hw/audio/adlib.c
+++ b/hw/audio/adlib.c
@@ -318,7 +318,7 @@ static void adlib_realizefn (DeviceState *dev, Error **errp)
as.freq = s->freq;
as.nchannels = SHIFT;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = AUDIO_HOST_ENDIANNESS;
AUD_register_card ("adlib", &s->card);
diff --git a/hw/audio/cs4231a.c b/hw/audio/cs4231a.c
index f96f561..626a173 100644
--- a/hw/audio/cs4231a.c
+++ b/hw/audio/cs4231a.c
@@ -284,7 +284,7 @@ static void cs_reset_voices (CSState *s, uint32_t val)
switch ((val >> 5) & ((s->dregs[MODE_And_ID] & MODE2) ? 7 : 3)) {
case 0:
- as.fmt = AUD_FMT_U8;
+ as.fmt = AUDIO_FORMAT_U8;
s->shift = as.nchannels == 2;
break;
@@ -294,7 +294,7 @@ static void cs_reset_voices (CSState *s, uint32_t val)
case 3:
s->tab = ALawDecompressTable;
x_law:
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = AUDIO_HOST_ENDIANNESS;
s->shift = as.nchannels == 2;
break;
@@ -302,7 +302,7 @@ static void cs_reset_voices (CSState *s, uint32_t val)
case 6:
as.endianness = 1;
case 2:
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
s->shift = as.nchannels;
break;
diff --git a/hw/audio/es1370.c b/hw/audio/es1370.c
index 592578b..a1c21b8 100644
--- a/hw/audio/es1370.c
+++ b/hw/audio/es1370.c
@@ -409,14 +409,14 @@ 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 e0c8a4e..0e1d15f 100644
--- a/hw/audio/gus.c
+++ b/hw/audio/gus.c
@@ -237,7 +237,7 @@ static void gus_realizefn (DeviceState *dev, Error **errp)
as.freq = s->freq;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = GUS_ENDIANNESS;
s->voice = AUD_open_out (
diff --git a/hw/audio/hda-codec.c b/hw/audio/hda-codec.c
index 3c03ff5..8693b7a 100644
--- a/hw/audio/hda-codec.c
+++ b/hw/audio/hda-codec.c
@@ -97,9 +97,9 @@ static void hda_codec_parse_fmt(uint32_t format, struct audsettings *as)
}
switch (format & AC_FMT_BITS_MASK) {
- case AC_FMT_BITS_8: as->fmt = AUD_FMT_S8; break;
- case AC_FMT_BITS_16: as->fmt = AUD_FMT_S16; break;
- case AC_FMT_BITS_32: as->fmt = AUD_FMT_S32; break;
+ case AC_FMT_BITS_8: as->fmt = AUDIO_FORMAT_S8; break;
+ case AC_FMT_BITS_16: as->fmt = AUDIO_FORMAT_S16; break;
+ case AC_FMT_BITS_32: as->fmt = AUDIO_FORMAT_S32; break;
}
as->nchannels = ((format & AC_FMT_CHAN_MASK) >> AC_FMT_CHAN_SHIFT) + 1;
@@ -128,12 +128,12 @@ static void hda_codec_parse_fmt(uint32_t format, struct audsettings *as)
/* -------------------------------------------------------------------------- */
static const char *fmt2name[] = {
- [ AUD_FMT_U8 ] = "PCM-U8",
- [ AUD_FMT_S8 ] = "PCM-S8",
- [ AUD_FMT_U16 ] = "PCM-U16",
- [ AUD_FMT_S16 ] = "PCM-S16",
- [ AUD_FMT_U32 ] = "PCM-U32",
- [ AUD_FMT_S32 ] = "PCM-S32",
+ [ AUDIO_FORMAT_U8 ] = "PCM-U8",
+ [ AUDIO_FORMAT_S8 ] = "PCM-S8",
+ [ AUDIO_FORMAT_U16 ] = "PCM-U16",
+ [ AUDIO_FORMAT_S16 ] = "PCM-S16",
+ [ AUDIO_FORMAT_U32 ] = "PCM-U32",
+ [ AUDIO_FORMAT_S32 ] = "PCM-S32",
};
typedef struct HDAAudioState HDAAudioState;
diff --git a/hw/audio/lm4549.c b/hw/audio/lm4549.c
index 380ef60..9d4f4b5 100644
--- a/hw/audio/lm4549.c
+++ b/hw/audio/lm4549.c
@@ -185,7 +185,7 @@ void lm4549_write(lm4549_state *s,
struct audsettings as;
as.freq = value;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = 0;
s->voice = AUD_open_out(
@@ -255,7 +255,7 @@ static int lm4549_post_load(void *opaque, int version_id)
struct audsettings as;
as.freq = freq;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = 0;
s->voice = AUD_open_out(
@@ -292,7 +292,7 @@ void lm4549_init(lm4549_state *s, lm4549_callback data_req_cb, void* opaque)
/* Open a default voice */
as.freq = 48000;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = 0;
s->voice = AUD_open_out(
diff --git a/hw/audio/milkymist-ac97.c b/hw/audio/milkymist-ac97.c
index 28f55e8..15169e2 100644
--- a/hw/audio/milkymist-ac97.c
+++ b/hw/audio/milkymist-ac97.c
@@ -297,7 +297,7 @@ static int milkymist_ac97_init(SysBusDevice *dev)
as.freq = 48000;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = 1;
s->voice_in = AUD_open_in(&s->card, s->voice_in,
diff --git a/hw/audio/pcspk.c b/hw/audio/pcspk.c
index 5266fb5..302debf 100644
--- a/hw/audio/pcspk.c
+++ b/hw/audio/pcspk.c
@@ -112,7 +112,7 @@ static void pcspk_callback(void *opaque, int free)
static int pcspk_audio_init(ISABus *bus)
{
PCSpkState *s = pcspk_state;
- struct audsettings as = {PCSPK_SAMPLE_RATE, 1, AUD_FMT_U8, 0};
+ struct audsettings as = {PCSPK_SAMPLE_RATE, 1, AUDIO_FORMAT_U8, 0};
AUD_register_card(s_spk, &s->card);
diff --git a/hw/audio/sb16.c b/hw/audio/sb16.c
index 995435f..163ced0 100644
--- a/hw/audio/sb16.c
+++ b/hw/audio/sb16.c
@@ -61,7 +61,7 @@ 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;
@@ -216,7 +216,7 @@ 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;
@@ -311,18 +311,18 @@ 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;
}
}
@@ -834,7 +834,7 @@ static void legacy_reset (SB16State *s)
as.freq = s->freq;
as.nchannels = 1;
- as.fmt = AUD_FMT_U8;
+ as.fmt = AUDIO_FORMAT_U8;
as.endianness = 0;
s->voice = AUD_open_out (
diff --git a/hw/audio/wm8750.c b/hw/audio/wm8750.c
index b50b331..4c4333c 100644
--- a/hw/audio/wm8750.c
+++ b/hw/audio/wm8750.c
@@ -201,7 +201,7 @@ static void wm8750_set_format(WM8750State *s)
in_fmt.endianness = 0;
in_fmt.nchannels = 2;
in_fmt.freq = s->adc_hz;
- in_fmt.fmt = AUD_FMT_S16;
+ in_fmt.fmt = AUDIO_FORMAT_S16;
s->adc_voice[0] = AUD_open_in(&s->card, s->adc_voice[0],
CODEC ".input1", s, wm8750_audio_in_cb, &in_fmt);
@@ -214,7 +214,7 @@ static void wm8750_set_format(WM8750State *s)
out_fmt.endianness = 0;
out_fmt.nchannels = 2;
out_fmt.freq = s->dac_hz;
- out_fmt.fmt = AUD_FMT_S16;
+ out_fmt.fmt = AUDIO_FORMAT_S16;
s->dac_voice[0] = AUD_open_out(&s->card, s->dac_voice[0],
CODEC ".speaker", s, wm8750_audio_out_cb, &out_fmt);
diff --git a/hw/input/tsc210x.c b/hw/input/tsc210x.c
index 1073bbf..961906f 100644
--- a/hw/input/tsc210x.c
+++ b/hw/input/tsc210x.c
@@ -315,7 +315,7 @@ static void tsc2102_audio_output_update(TSC210xState *s)
fmt.endianness = 0;
fmt.nchannels = 2;
fmt.freq = s->codec.tx_rate;
- fmt.fmt = AUD_FMT_S16;
+ fmt.fmt = AUDIO_FORMAT_S16;
s->dac_voice[0] = AUD_open_out(&s->card, s->dac_voice[0],
"tsc2102.sink", s, (void *) tsc210x_audio_out_cb, &fmt);
diff --git a/hw/usb/dev-audio.c b/hw/usb/dev-audio.c
index 02fb110..9a60569 100644
--- a/hw/usb/dev-audio.c
+++ b/hw/usb/dev-audio.c
@@ -646,7 +646,7 @@ static void usb_audio_realize(USBDevice *dev, Error **errp)
s->out.vol[1] = 240; /* 0 dB */
s->out.as.freq = USBAUDIO_SAMPLE_RATE;
s->out.as.nchannels = 2;
- s->out.as.fmt = AUD_FMT_S16;
+ s->out.as.fmt = AUDIO_FORMAT_S16;
s->out.as.endianness = 0;
streambuf_init(&s->out.buf, s->buffer);
diff --git a/ui/vnc.c b/ui/vnc.c
index 09756cd..2dd269b 100644
--- a/ui/vnc.c
+++ b/ui/vnc.c
@@ -2285,12 +2285,12 @@ 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);
@@ -3015,7 +3015,7 @@ void vnc_init_state(VncState *vs)
vs->as.freq = 44100;
vs->as.nchannels = 2;
- vs->as.fmt = AUD_FMT_S16;
+ vs->as.fmt = AUDIO_FORMAT_S16;
vs->as.endianness = 0;
qemu_mutex_init(&vs->output_mutex);
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 05/51] audio: -audiodev command line option: documentation
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (3 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 04/51] audio: use qapi AudioFormat instead of audfmt_e Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 06/51] audio: -audiodev command line option basic implementation Kővágó, Zoltán
` (45 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 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 215d00d..97f021f 100644
--- a/qemu-options.hx
+++ b/qemu-options.hx
@@ -320,14 +320,230 @@ 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,
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 06/51] audio: -audiodev command line option basic implementation
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (4 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 05/51] audio: -audiodev command line option: documentation Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 07/51] alsaaudio: port to -audiodev config Kővágó, Zoltán
` (44 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 UTC (permalink / raw)
To: qemu-devel; +Cc: Paolo Bonzini, Gerd Hoffmann
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.
---
audio/Makefile.objs | 2 +-
audio/alsaaudio.c | 2 +-
audio/audio.c | 566 +++++++++++++++++++------------------------------
audio/audio.h | 23 +-
audio/audio_int.h | 6 +-
audio/audio_legacy.c | 202 ++++++++++++++++++
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 | 10 +-
16 files changed, 472 insertions(+), 368 deletions(-)
create mode 100644 audio/audio_legacy.c
diff --git a/audio/Makefile.objs b/audio/Makefile.objs
index 481d1aa..9d8f579 100644
--- a/audio/Makefile.objs
+++ b/audio/Makefile.objs
@@ -1,4 +1,4 @@
-common-obj-y = audio.o noaudio.o wavaudio.o mixeng.o
+common-obj-y = audio.o audio_legacy.o noaudio.o wavaudio.o mixeng.o
common-obj-$(CONFIG_SDL) += sdlaudio.o
common-obj-$(CONFIG_OSS) += ossaudio.o
common-obj-$(CONFIG_SPICE) += spiceaudio.o
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 2b28b99..e545766 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -1126,7 +1126,7 @@ 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 334c935..fcbc543 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -24,7 +24,10 @@
#include "hw/hw.h"
#include "audio.h"
#include "monitor/monitor.h"
+#include "qapi-visit.h"
+#include "qapi/opts-visitor.h"
#include "qemu/timer.h"
+#include "qemu/config-file.h"
#include "sysemu/sysemu.h"
#define AUDIO_CAP "audio"
@@ -42,59 +45,14 @@
The 1st one is the one used by default, that is the reason
that we generate the list.
*/
-static struct audio_driver *drvtab[] = {
+struct audio_driver *drvtab[] = {
#ifdef CONFIG_SPICE
&spice_audio_driver,
#endif
CONFIG_AUDIO_DRIVERS
&no_audio_driver,
- &wav_audio_driver
-};
-
-struct fixed_settings {
- int enabled;
- int nb_voices;
- int greedy;
- struct audsettings settings;
-};
-
-static struct {
- struct fixed_settings fixed_out;
- struct fixed_settings fixed_in;
- union {
- int hertz;
- int64_t ticks;
- } period;
- int try_poll_in;
- int try_poll_out;
-} conf = {
- .fixed_out = { /* DAC fixed settings */
- .enabled = 1,
- .nb_voices = 1,
- .greedy = 1,
- .settings = {
- .freq = 44100,
- .nchannels = 2,
- .fmt = AUDIO_FORMAT_S16,
- .endianness = AUDIO_HOST_ENDIANNESS,
- }
- },
-
- .fixed_in = { /* ADC fixed settings */
- .enabled = 1,
- .nb_voices = 1,
- .greedy = 1,
- .settings = {
- .freq = 44100,
- .nchannels = 2,
- .fmt = AUDIO_FORMAT_S16,
- .endianness = AUDIO_HOST_ENDIANNESS,
- }
- },
-
- .period = { .hertz = 100 },
- .try_poll_in = 1,
- .try_poll_out = 1,
+ &wav_audio_driver,
+ NULL
};
static AudioState glob_audio_state;
@@ -113,9 +71,6 @@ const struct mixeng_volume nominal_volume = {
#ifdef AUDIO_IS_FLAWLESS_AND_NO_CHECKS_ARE_REQURIED
#error No its not
#else
-static void audio_print_options (const char *prefix,
- struct audio_option *opt);
-
int audio_bug (const char *funcname, int cond)
{
if (cond) {
@@ -123,16 +78,9 @@ int audio_bug (const char *funcname, int cond)
AUD_log (NULL, "A bug was just triggered in %s\n", funcname);
if (!shown) {
- struct audio_driver *d;
-
shown = 1;
AUD_log (NULL, "Save all your work and restart without audio\n");
- AUD_log (NULL, "Please send bug report to av1474@comtv.ru\n");
AUD_log (NULL, "I am sorry\n");
- d = glob_audio_state.drv;
- if (d) {
- audio_print_options (d->name, d->options);
- }
}
AUD_log (NULL, "Context:\n");
@@ -194,31 +142,6 @@ void *audio_calloc (const char *funcname, int nmemb, size_t size)
return g_malloc0 (len);
}
-static char *audio_alloc_prefix (const char *s)
-{
- const char qemu_prefix[] = "QEMU_";
- size_t len, i;
- char *r, *u;
-
- if (!s) {
- return NULL;
- }
-
- len = strlen (s);
- r = g_malloc (len + sizeof (qemu_prefix));
-
- u = r + sizeof (qemu_prefix) - 1;
-
- pstrcpy (r, len + sizeof (qemu_prefix), qemu_prefix);
- pstrcat (r, len + sizeof (qemu_prefix), s);
-
- for (i = 0; i < len; ++i) {
- u[i] = qemu_toupper(u[i]);
- }
-
- return r;
-}
-
static const char *audio_audfmt_to_string (AudioFormat fmt)
{
switch (fmt) {
@@ -345,78 +268,6 @@ 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)
{
@@ -1120,7 +971,7 @@ static void audio_reset_timer (AudioState *s)
{
if (audio_is_timer_needed ()) {
timer_mod (s->ts,
- qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + conf.period.ticks);
+ qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + s->period_ticks);
}
else {
timer_del (s->ts);
@@ -1196,7 +1047,7 @@ 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);
}
}
@@ -1241,7 +1092,7 @@ 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);
}
}
@@ -1558,168 +1409,13 @@ 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)
-{
- size_t i;
-
- audio_process_options ("AUDIO", audio_options);
- for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
- struct audio_driver *d = drvtab[i];
- if (d->options) {
- audio_process_options (d->name, d->options);
- }
- }
-
- printf ("Audio options:\n");
- audio_print_options ("AUDIO", audio_options);
- printf ("\n");
-
- printf ("Available drivers:\n");
-
- for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
- struct audio_driver *d = drvtab[i];
-
- printf ("Name: %s\n", d->name);
- printf ("Description: %s\n", d->descr);
-
- audio_pp_nb_voices ("playback", d->max_voices_out);
- audio_pp_nb_voices ("capture", d->max_voices_in);
-
- if (d->options) {
- printf ("Options:\n");
- audio_print_options (d->name, d->options);
- }
- else {
- printf ("No options\n");
- }
- printf ("\n");
- }
-
- printf (
- "Options are settable through environment variables.\n"
- "Example:\n"
-#ifdef _WIN32
- " set QEMU_AUDIO_DRV=wav\n"
- " set QEMU_WAV_PATH=c:\\tune.wav\n"
-#else
- " export QEMU_AUDIO_DRV=wav\n"
- " export QEMU_WAV_PATH=$HOME/tune.wav\n"
- "(for csh replace export with setenv in the above)\n"
-#endif
- " qemu ...\n\n"
- );
-}
-
-static int audio_driver_init (AudioState *s, struct audio_driver *drv)
+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);
@@ -1743,11 +1439,11 @@ 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);
}
@@ -1786,6 +1482,8 @@ static void audio_atexit (void)
if (s->drv) {
s->drv->fini (s->drv_opaque);
}
+
+ qapi_free_Audiodev(s->dev);
}
static const VMStateDescription vmstate_audio = {
@@ -1797,18 +1495,37 @@ static const VMStateDescription vmstate_audio = {
}
};
-static void audio_init (void)
+static Audiodev *parse_option(QemuOpts *opts, Error **errp);
+static int audio_init(Audiodev *dev)
{
size_t i;
int done = 0;
- const char *drvname;
+ const char *drvname = NULL;
VMChangeStateEntry *e;
AudioState *s = &glob_audio_state;
+ QemuOptsList *list = NULL; /* silence gcc warning about uninitialized
+ * variable */
if (s->drv) {
- return;
+ if (dev) {
+ dolog("Cannot create more than one audio backend, sorry\n");
+ qapi_free_Audiodev(dev);
+ }
+ return -1;
}
+ if (dev) {
+ drvname = AudiodevDriver_lookup[dev->driver];
+ } else {
+ audio_handle_legacy_opts();
+ list = qemu_find_opts("audiodev");
+ dev = parse_option(QTAILQ_FIRST(&list->head), &error_abort);
+ if (!dev) {
+ exit(1);
+ }
+ }
+ s->dev = dev;
+
QLIST_INIT (&s->hw_head_out);
QLIST_INIT (&s->hw_head_in);
QLIST_INIT (&s->cap_head);
@@ -1819,10 +1536,8 @@ static void audio_init (void)
hw_error("Could not create audio timer\n");
}
- audio_process_options ("AUDIO", audio_options);
-
- s->nb_hw_voices_out = conf.fixed_out.nb_voices;
- s->nb_hw_voices_in = conf.fixed_in.nb_voices;
+ s->nb_hw_voices_out = dev->out->voices;
+ s->nb_hw_voices_in = dev->in->voices;
if (s->nb_hw_voices_out <= 0) {
dolog ("Bogus number of playback voices %d, setting to 1\n",
@@ -1836,17 +1551,12 @@ static void audio_init (void)
s->nb_hw_voices_in = 0;
}
- {
- int def;
- drvname = audio_get_conf_str ("QEMU_AUDIO_DRV", NULL, &def);
- }
-
if (drvname) {
int found = 0;
- for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
+ for (i = 0; drvtab[i]; i++) {
if (!strcmp (drvname, drvtab[i]->name)) {
- done = !audio_driver_init (s, drvtab[i]);
+ done = !audio_driver_init (s, drvtab[i], dev);
found = 1;
break;
}
@@ -1854,20 +1564,24 @@ static void audio_init (void)
if (!found) {
dolog ("Unknown audio driver `%s'\n", drvname);
- dolog ("Run with -audio-help to list available drivers\n");
}
- }
-
- if (!done) {
- for (i = 0; !done && i < ARRAY_SIZE (drvtab); i++) {
- if (drvtab[i]->can_be_default) {
- done = !audio_driver_init (s, drvtab[i]);
+ } else {
+ for (i = 0; !done && drvtab[i]; i++) {
+ QemuOpts *opts = qemu_opts_find(list, drvtab[i]->name);
+ if (opts) {
+ qapi_free_Audiodev(dev);
+ dev = parse_option(opts, &error_abort);
+ if (!dev) {
+ exit(1);
+ }
+ s->dev = dev;
+ done = !audio_driver_init(s, drvtab[i], dev);
}
}
}
if (!done) {
- done = !audio_driver_init (s, &no_audio_driver);
+ done = !audio_driver_init (s, &no_audio_driver, dev);
if (!done) {
hw_error("Could not initialize audio subsystem\n");
}
@@ -1876,16 +1590,16 @@ static void audio_init (void)
}
}
- if (conf.period.hertz <= 0) {
- if (conf.period.hertz < 0) {
- dolog ("warning: Timer period is negative - %d "
- "treating as zero\n",
- conf.period.hertz);
+ if (dev->timer_period <= 0) {
+ if (dev->timer_period < 0) {
+ dolog ("warning: Timer period is negative - %" PRId64
+ " treating as zero\n",
+ dev->timer_period);
}
- conf.period.ticks = 1;
+ s->period_ticks = 1;
} else {
- conf.period.ticks =
- muldiv64 (1, get_ticks_per_sec (), conf.period.hertz);
+ s->period_ticks =
+ muldiv64(dev->timer_period, get_ticks_per_sec(), 1000000);
}
e = qemu_add_vm_change_state_handler (audio_vm_change_state_handler, s);
@@ -1896,11 +1610,12 @@ static void audio_init (void)
QLIST_INIT (&s->card_head);
vmstate_register (NULL, 0, &vmstate_audio, s);
+ return 0;
}
void AUD_register_card (const char *name, QEMUSoundCard *card)
{
- audio_init ();
+ audio_init(NULL);
card->name = g_strdup (name);
memset (&card->entries, 0, sizeof (card->entries));
QLIST_INSERT_HEAD (&glob_audio_state.card_head, card, entries);
@@ -2070,3 +1785,156 @@ 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;
+ OptsVisitor *ov = opts_visitor_new(opts, true);
+ Audiodev *dev = NULL;
+ visit_type_Audiodev(opts_get_visitor(ov), &dev, NULL, &local_err);
+ opts_visitor_cleanup(ov);
+
+ 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 e300511..177a673 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -24,7 +24,10 @@
#ifndef QEMU_AUDIO_H
#define QEMU_AUDIO_H
+#include <stdarg.h>
#include "config-host.h"
+#include "qapi-types.h"
+#include "qemu/option.h"
#include "qemu/queue.h"
typedef void (*audio_callback_fn) (void *opaque, int avail);
@@ -35,12 +38,21 @@ typedef void (*audio_callback_fn) (void *opaque, int avail);
#define AUDIO_HOST_ENDIANNESS 0
#endif
-struct audsettings {
+typedef struct audsettings {
int freq;
int nchannels;
AudioFormat fmt;
int endianness;
-};
+} audsettings;
+
+audsettings audiodev_to_audsettings(AudiodevPerDirectionOptions *pdo);
+int audioformat_bytes_per_sample(AudioFormat fmt);
+int audio_buffer_frames(AudiodevPerDirectionOptions *pdo,
+ audsettings *as, int def_usecs);
+int audio_buffer_samples(AudiodevPerDirectionOptions *pdo,
+ audsettings *as, int def_usecs);
+int audio_buffer_bytes(AudiodevPerDirectionOptions *pdo,
+ audsettings *as, int def_usecs);
typedef enum {
AUD_CNOTIFY_ENABLE,
@@ -77,10 +89,11 @@ typedef struct QEMUAudioTimeStamp {
uint64_t old_ts;
} QEMUAudioTimeStamp;
+extern QemuOptsList qemu_audiodev_opts;
+
void AUD_vlog (const char *cap, const char *fmt, va_list ap) GCC_FMT_ATTR(2, 0);
void AUD_log (const char *cap, const char *fmt, ...) GCC_FMT_ATTR(2, 3);
-void AUD_help (void);
void AUD_register_card (const char *name, QEMUSoundCard *card);
void AUD_remove_card (QEMUSoundCard *card);
CaptureVoiceOut *AUD_add_capture (
@@ -154,4 +167,8 @@ static inline void *advance (void *p, int incr)
int wav_start_capture (CaptureState *s, const char *path, int freq,
int bits, int nchannels);
+void audio_set_options(void);
+void audio_handle_legacy_opts(void);
+void audio_legacy_help(void);
+
#endif /* audio.h */
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 566df5e..9139f42 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -144,7 +144,7 @@ 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;
@@ -190,6 +190,7 @@ struct SWVoiceCap {
struct AudioState {
struct audio_driver *drv;
+ Audiodev *dev;
void *drv_opaque;
QEMUTimer *ts;
@@ -200,6 +201,7 @@ struct AudioState {
int nb_hw_voices_out;
int nb_hw_voices_in;
int vm_running;
+ int64_t period_ticks;
};
extern struct audio_driver no_audio_driver;
@@ -213,6 +215,8 @@ extern struct audio_driver pa_audio_driver;
extern struct audio_driver spice_audio_driver;
extern const struct mixeng_volume nominal_volume;
+extern struct audio_driver *drvtab[];
+
void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as);
void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len);
diff --git a/audio/audio_legacy.c b/audio/audio_legacy.c
new file mode 100644
index 0000000..8aa0748
--- /dev/null
+++ b/audio/audio_legacy.c
@@ -0,0 +1,202 @@
+#include "audio.h"
+#include "qemu-common.h"
+#include "qemu/config-file.h"
+
+#define AUDIO_CAP "audio-legacy"
+#include "audio_int.h"
+
+typedef enum EnvTransform {
+ ENV_TRANSFORM_NONE,
+ ENV_TRANSFORM_BOOL,
+ ENV_TRANSFORM_FMT,
+ ENV_TRANSFORM_FRAMES_TO_USECS_IN,
+ ENV_TRANSFORM_FRAMES_TO_USECS_OUT,
+ ENV_TRANSFORM_SAMPLES_TO_USECS_IN,
+ ENV_TRANSFORM_SAMPLES_TO_USECS_OUT,
+ ENV_TRANSFORM_BYTES_TO_USECS_IN,
+ ENV_TRANSFORM_BYTES_TO_USECS_OUT,
+ ENV_TRANSFORM_MILLIS_TO_USECS,
+ ENV_TRANSFORM_HZ_TO_USECS,
+} EnvTransform;
+
+typedef struct SimpleEnvMap {
+ const char *name;
+ const char *option;
+ EnvTransform transform;
+} SimpleEnvMap;
+
+SimpleEnvMap global_map[] = {
+ /* DAC/out settings */
+ { "QEMU_AUDIO_DAC_FIXED_SETTINGS", "out.fixed-settings",
+ ENV_TRANSFORM_BOOL },
+ { "QEMU_AUDIO_DAC_FIXED_FREQ", "out.frequency" },
+ { "QEMU_AUDIO_DAC_FIXED_FMT", "out.format", ENV_TRANSFORM_FMT },
+ { "QEMU_AUDIO_DAC_FIXED_CHANNELS", "out.channels" },
+ { "QEMU_AUDIO_DAC_VOICES", "out.voices" },
+
+ /* ADC/in settings */
+ { "QEMU_AUDIO_ADC_FIXED_SETTINGS", "in.fixed-settings",
+ ENV_TRANSFORM_BOOL },
+ { "QEMU_AUDIO_ADC_FIXED_FREQ", "in.frequency" },
+ { "QEMU_AUDIO_ADC_FIXED_FMT", "in.format", ENV_TRANSFORM_FMT },
+ { "QEMU_AUDIO_ADC_FIXED_CHANNELS", "in.channels" },
+ { "QEMU_AUDIO_ADC_VOICES", "in.voices" },
+
+ /* general */
+ { "QEMU_AUDIO_TIMER_PERIOD", "timer-period", ENV_TRANSFORM_HZ_TO_USECS },
+ { /* End of list */ }
+};
+
+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 *drv = getenv("QEMU_AUDIO_DRV");
+
+ if (drv) {
+ legacy_opt(drv);
+ } else {
+ struct audio_driver **drv;
+ for (drv = drvtab; *drv; ++drv) {
+ if ((*drv)->can_be_default) {
+ legacy_opt((*drv)->name);
+ }
+ }
+ }
+}
+
+static int legacy_help_each(void *opaque, QemuOpts *opts, Error **errp)
+{
+ printf("-audiodev ");
+ qemu_opts_print(opts, ",");
+ printf("\n");
+ return 0;
+}
+
+void audio_legacy_help(void)
+{
+ printf("Environment variable based configuration deprecated.\n");
+ printf("Please use the new -audiodev option.\n");
+
+ audio_handle_legacy_opts();
+ printf("\nEquivalent -audiodev to your current environment variables:\n");
+ qemu_opts_foreach(qemu_find_opts("audiodev"), legacy_help_each, NULL, NULL);
+}
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 99b27b2..096b2b3 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -302,8 +302,10 @@ static HW *glue (audio_pcm_hw_add_new_, TYPE) (struct audsettings *as)
static HW *glue (audio_pcm_hw_add_, TYPE) (struct audsettings *as)
{
HW *hw;
+ AudioState *s = &glob_audio_state;
+ AudiodevPerDirectionOptions *pdo = s->dev->TYPE;
- if (glue (conf.fixed_, TYPE).enabled && glue (conf.fixed_, TYPE).greedy) {
+ if (pdo->fixed_settings) {
hw = glue (audio_pcm_hw_add_new_, TYPE) (as);
if (hw) {
return hw;
@@ -331,9 +333,11 @@ static SW *glue (audio_pcm_create_voice_pair_, TYPE) (
SW *sw;
HW *hw;
struct audsettings hw_as;
+ AudioState *s = &glob_audio_state;
+ AudiodevPerDirectionOptions *pdo = s->dev->TYPE;
- if (glue (conf.fixed_, TYPE).enabled) {
- hw_as = glue (conf.fixed_, TYPE).settings;
+ if (pdo->fixed_settings) {
+ hw_as = audiodev_to_audsettings(pdo);
}
else {
hw_as = *as;
@@ -398,6 +402,7 @@ SW *glue (AUD_open_, TYPE) (
)
{
AudioState *s = &glob_audio_state;
+ AudiodevPerDirectionOptions *pdo = s->dev->TYPE;
if (audio_bug (AUDIO_FUNC, !card || !name || !callback_fn || !as)) {
dolog ("card=%p name=%p callback_fn=%p as=%p\n",
@@ -422,7 +427,7 @@ SW *glue (AUD_open_, TYPE) (
return sw;
}
- if (!glue (conf.fixed_, TYPE).enabled && sw) {
+ if (!pdo->fixed_settings && sw) {
glue (AUD_close_, TYPE) (card, sw);
sw = NULL;
}
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index 7150604..f927c0f 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -692,7 +692,7 @@ 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 e9472c1..d447e12 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -782,7 +782,7 @@ 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 50db1f3..108b02d 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -134,7 +134,7 @@ static int no_ctl_in (HWVoiceIn *hw, int cmd, ...)
return 0;
}
-static void *no_audio_init (void)
+static void *no_audio_init (Audiodev *dev)
{
return &no_audio_init;
}
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 3ea7d27..6d64cf2 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -846,7 +846,7 @@ 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 cfdbdc6..dd46fa0 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 db0f95a..580c613 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -391,7 +391,7 @@ 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 141fd8d..e6da29a 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -75,7 +75,7 @@ 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 81250e6..8b0ba91 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -231,7 +231,7 @@ 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 eb7edb0..de47b6b 100644
--- a/vl.c
+++ b/vl.c
@@ -3025,6 +3025,7 @@ int main(int argc, char **argv, char **envp)
qemu_add_opts(&qemu_trace_opts);
qemu_add_opts(&qemu_option_rom_opts);
qemu_add_opts(&qemu_machine_opts);
+ qemu_add_opts(&qemu_audiodev_opts);
qemu_add_opts(&qemu_mem_opts);
qemu_add_opts(&qemu_smp_opts);
qemu_add_opts(&qemu_boot_opts);
@@ -3328,9 +3329,15 @@ 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;
@@ -4533,6 +4540,7 @@ int main(int argc, char **argv, char **envp)
realtime_init();
+ audio_set_options();
audio_init();
cpu_synchronize_all_post_init();
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 07/51] alsaaudio: port to -audiodev config
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (5 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 06/51] audio: -audiodev command line option basic implementation Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 08/51] coreaudio: " Kővágó, Zoltán
` (43 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/alsaaudio.c | 310 ++++++++++++++-------------------------------------
audio/audio_legacy.c | 64 +++++++++++
2 files changed, 148 insertions(+), 226 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index e545766..67060f8 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -22,6 +22,7 @@
* THE SOFTWARE.
*/
#include <alsa/asoundlib.h>
+#include "qapi-visit.h"
#include "qemu-common.h"
#include "qemu/main-loop.h"
#include "audio.h"
@@ -34,28 +35,9 @@
#define AUDIO_CAP "alsa"
#include "audio_int.h"
-typedef struct ALSAConf {
- int size_in_usec_in;
- int size_in_usec_out;
- const char *pcm_name_in;
- const char *pcm_name_out;
- unsigned int buffer_size_in;
- unsigned int period_size_in;
- unsigned int buffer_size_out;
- unsigned int period_size_out;
- unsigned int threshold;
-
- int buffer_size_in_overridden;
- int period_size_in_overridden;
-
- int buffer_size_out_overridden;
- int period_size_out_overridden;
-} ALSAConf;
-
struct pollhlp {
snd_pcm_t *handle;
struct pollfd *pfds;
- ALSAConf *conf;
int count;
int mask;
};
@@ -67,6 +49,7 @@ typedef struct ALSAVoiceOut {
void *pcm_buf;
snd_pcm_t *handle;
struct pollhlp pollhlp;
+ Audiodev *dev;
} ALSAVoiceOut;
typedef struct ALSAVoiceIn {
@@ -74,16 +57,13 @@ typedef struct ALSAVoiceIn {
snd_pcm_t *handle;
void *pcm_buf;
struct pollhlp pollhlp;
+ Audiodev *dev;
} ALSAVoiceIn;
struct alsa_params_req {
int freq;
snd_pcm_format_t fmt;
int nchannels;
- int size_in_usec;
- int override_mask;
- unsigned int buffer_size;
- unsigned int period_size;
};
struct alsa_params_obt {
@@ -409,7 +389,8 @@ static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt,
static void alsa_dump_info (struct alsa_params_req *req,
struct alsa_params_obt *obt,
- snd_pcm_format_t obtfmt)
+ snd_pcm_format_t obtfmt,
+ AudiodevPerDirectionOptions *pdo)
{
dolog ("parameter | requested value | obtained value\n");
dolog ("format | %10d | %10d\n", req->fmt, obtfmt);
@@ -417,8 +398,9 @@ static void alsa_dump_info (struct alsa_params_req *req,
req->nchannels, obt->nchannels);
dolog ("frequency | %10d | %10d\n", req->freq, obt->freq);
dolog ("============================================\n");
- dolog ("requested: buffer size %d period size %d\n",
- req->buffer_size, req->period_size);
+ dolog ("requested: buffer 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);
}
@@ -452,23 +434,25 @@ static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
}
}
-static int alsa_open (int in, struct alsa_params_req *req,
- struct alsa_params_obt *obt, snd_pcm_t **handlep,
- ALSAConf *conf)
+static int alsa_open(bool in, struct alsa_params_req *req,
+ struct alsa_params_obt *obt, snd_pcm_t **handlep,
+ Audiodev *dev)
{
+ AudiodevPerDirectionOptions *pdo = in ? dev->in : dev->out;
+ AudiodevAlsaOptions *aopts = dev->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);
@@ -528,79 +512,49 @@ static int alsa_open (int in, struct alsa_params_req *req,
goto err;
}
- if (req->buffer_size) {
- unsigned long obt;
+ if (pdo->buffer_count) {
+ if (pdo->buffer_len) {
+ int64_t req = pdo->buffer_len * pdo->buffer_count;
- if (size_in_usec) {
int dir = 0;
- unsigned int btime = req->buffer_size;
+ unsigned int btime = req;
- err = snd_pcm_hw_params_set_buffer_time_near (
- handle,
- hw_params,
- &btime,
- &dir
- );
- obt = btime;
- }
- else {
- snd_pcm_uframes_t bsize = req->buffer_size;
+ err = snd_pcm_hw_params_set_buffer_time_near(
+ handle, hw_params, &btime, &dir);
- err = snd_pcm_hw_params_set_buffer_size_near (
- handle,
- hw_params,
- &bsize
- );
- obt = bsize;
- }
- if (err < 0) {
- alsa_logerr2 (err, typ, "Failed to set buffer %s to %d\n",
- size_in_usec ? "time" : "size", req->buffer_size);
- goto err;
- }
+ if (err < 0) {
+ alsa_logerr2(err, typ,
+ "Failed to set buffer time to %" PRId64 "\n",
+ req);
+ goto err;
+ }
- if ((req->override_mask & 2) && (obt - req->buffer_size))
- dolog ("Requested buffer %s %u was rejected, using %lu\n",
- size_in_usec ? "time" : "size", req->buffer_size, obt);
+ if (pdo->has_buffer_count && btime != req) {
+ dolog("Requested buffer time %" PRId64
+ " was rejected, using %u\n", req, btime);
+ }
+ } else {
+ dolog("Can't set buffer-count without buffer-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);
@@ -632,33 +586,10 @@ static int alsa_open (int in, struct alsa_params_req *req,
goto err;
}
- if (!in && conf->threshold) {
- snd_pcm_uframes_t threshold;
- int bytes_per_sec;
-
- bytes_per_sec = freq << (nchannels == 2);
-
- switch (obt->fmt) {
- case AUDIO_FORMAT_S8:
- case AUDIO_FORMAT_U8:
- break;
-
- case AUDIO_FORMAT_S16:
- case AUDIO_FORMAT_U16:
- bytes_per_sec <<= 1;
- break;
-
- case AUDIO_FORMAT_S32:
- case AUDIO_FORMAT_U32:
- bytes_per_sec <<= 2;
- break;
-
- default:
- abort();
- }
-
- threshold = (conf->threshold * bytes_per_sec) / 1000;
- alsa_set_threshold (handle, threshold);
+ if (!in && aopts->has_threshold && aopts->threshold) {
+ struct audsettings as = { .freq = freq };
+ alsa_set_threshold(handle,
+ audio_buffer_frames(pdo, &as, aopts->threshold));
}
obt->nchannels = nchannels;
@@ -671,11 +602,11 @@ static int alsa_open (int in, struct alsa_params_req *req,
obt->nchannels != req->nchannels ||
obt->freq != req->freq) {
dolog ("Audio parameters for %s\n", typ);
- alsa_dump_info (req, obt, obtfmt);
+ alsa_dump_info (req, obt, obtfmt, pdo);
}
#ifdef DEBUG
- alsa_dump_info (req, obt, obtfmt);
+ alsa_dump_info (req, obt, obtfmt, pdo);
#endif
return 0;
@@ -801,19 +732,13 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
struct alsa_params_obt obt;
snd_pcm_t *handle;
struct audsettings obt_as;
- ALSAConf *conf = drv_opaque;
+ Audiodev *dev = drv_opaque;
req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
req.freq = as->freq;
req.nchannels = as->nchannels;
- req.period_size = conf->period_size_out;
- req.buffer_size = conf->buffer_size_out;
- req.size_in_usec = conf->size_in_usec_out;
- req.override_mask =
- (conf->period_size_out_overridden ? 1 : 0) |
- (conf->buffer_size_out_overridden ? 2 : 0);
- if (alsa_open (0, &req, &obt, &handle, conf)) {
+ if (alsa_open (0, &req, &obt, &handle, dev)) {
return -1;
}
@@ -834,7 +759,7 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
}
alsa->handle = handle;
- alsa->pollhlp.conf = conf;
+ alsa->dev = dev;
return 0;
}
@@ -874,16 +799,12 @@ static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl)
static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+ AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->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)) {
@@ -912,19 +833,13 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
struct alsa_params_obt obt;
snd_pcm_t *handle;
struct audsettings obt_as;
- ALSAConf *conf = drv_opaque;
+ Audiodev *dev = drv_opaque;
req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
req.freq = as->freq;
req.nchannels = as->nchannels;
- req.period_size = conf->period_size_in;
- req.buffer_size = conf->buffer_size_in;
- req.size_in_usec = conf->size_in_usec_in;
- req.override_mask =
- (conf->period_size_in_overridden ? 1 : 0) |
- (conf->buffer_size_in_overridden ? 2 : 0);
- if (alsa_open (1, &req, &obt, &handle, conf)) {
+ if (alsa_open (1, &req, &obt, &handle, dev)) {
return -1;
}
@@ -945,7 +860,7 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
}
alsa->handle = handle;
- alsa->pollhlp.conf = conf;
+ alsa->dev = dev;
return 0;
}
@@ -1087,16 +1002,12 @@ static int alsa_read (SWVoiceIn *sw, void *buf, int size)
static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+ AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->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)) {
@@ -1119,88 +1030,36 @@ 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,
@@ -1218,7 +1077,6 @@ static struct audio_pcm_ops alsa_pcm_ops = {
struct audio_driver alsa_audio_driver = {
.name = "alsa",
.descr = "ALSA http://www.alsa-project.org",
- .options = alsa_options,
.init = alsa_audio_init,
.fini = alsa_audio_fini,
.pcm_ops = &alsa_pcm_ops,
diff --git a/audio/audio_legacy.c b/audio/audio_legacy.c
index 8aa0748..bc830c5 100644
--- a/audio/audio_legacy.c
+++ b/audio/audio_legacy.c
@@ -47,6 +47,17 @@ 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;
@@ -157,6 +168,54 @@ 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;
@@ -165,6 +224,11 @@ 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)
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 08/51] coreaudio: port to -audiodev config
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (6 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 07/51] alsaaudio: port to -audiodev config Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 09/51] dsoundaudio: " Kővágó, Zoltán
` (42 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 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 bc830c5..4be972e 100644
--- a/audio/audio_legacy.c
+++ b/audio/audio_legacy.c
@@ -58,6 +58,14 @@ 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;
@@ -228,6 +236,8 @@ 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 f927c0f..46438a2 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -38,11 +38,6 @@
static int isAtexit;
-typedef struct {
- int buffer_frames;
- int nbuffers;
-} CoreaudioConf;
-
typedef struct coreaudioVoiceOut {
HWVoiceOut hw;
pthread_mutex_t mutex;
@@ -514,7 +509,9 @@ 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);
@@ -545,16 +542,17 @@ 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 */
@@ -575,7 +573,8 @@ 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,
@@ -687,41 +686,16 @@ static int coreaudio_ctl_out (HWVoiceOut *hw, int cmd, ...)
return 0;
}
-static CoreaudioConf glob_conf = {
- .buffer_frames = 512,
- .nbuffers = 4,
-};
-
static void *coreaudio_audio_init(Audiodev *dev)
{
- CoreaudioConf *conf = g_malloc(sizeof(CoreaudioConf));
- *conf = glob_conf;
-
atexit(coreaudio_atexit);
- return conf;
+ return dev;
}
static void coreaudio_audio_fini (void *opaque)
{
- g_free(opaque);
}
-static struct audio_option coreaudio_options[] = {
- {
- .name = "BUFFER_SIZE",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.buffer_frames,
- .descr = "Size of the buffer in frames"
- },
- {
- .name = "BUFFER_COUNT",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.nbuffers,
- .descr = "Number of buffers"
- },
- { /* End of list */ }
-};
-
static struct audio_pcm_ops coreaudio_pcm_ops = {
.init_out = coreaudio_init_out,
.fini_out = coreaudio_fini_out,
@@ -733,7 +707,6 @@ static struct audio_pcm_ops coreaudio_pcm_ops = {
struct audio_driver coreaudio_audio_driver = {
.name = "coreaudio",
.descr = "CoreAudio http://developer.apple.com/audio/coreaudio.html",
- .options = coreaudio_options,
.init = coreaudio_audio_init,
.fini = coreaudio_audio_fini,
.pcm_ops = &coreaudio_pcm_ops,
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 09/51] dsoundaudio: port to -audiodev config
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (7 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 08/51] coreaudio: " Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 10/51] noaudio: " Kővágó, Zoltán
` (41 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 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 4be972e..bcf87ff 100644
--- a/audio/audio_legacy.c
+++ b/audio/audio_legacy.c
@@ -66,6 +66,16 @@ 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;
@@ -238,6 +248,8 @@ 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 b439f33..96181ef 100644
--- a/audio/dsound_template.h
+++ b/audio/dsound_template.h
@@ -167,17 +167,18 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
dsound *s = drv_opaque;
WAVEFORMATEX wfx;
struct audsettings obt_as;
- DSoundConf *conf = &s->conf;
#ifdef DSBTYPE_IN
const char *typ = "ADC";
DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;
DSCBUFFERDESC bd;
DSCBCAPS bc;
+ AudiodevPerDirectionOptions *pdo = s->dev->in;
#else
const char *typ = "DAC";
DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
DSBUFFERDESC bd;
DSBCAPS bc;
+ AudiodevPerDirectionOptions *pdo = s->dev->out;
#endif
if (!s->FIELD2) {
@@ -193,8 +194,8 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
memset (&bd, 0, sizeof (bd));
bd.dwSize = sizeof (bd);
bd.lpwfxFormat = &wfx;
+ bd.dwBufferBytes = audio_buffer_bytes(pdo, as, 92880);
#ifdef DSBTYPE_IN
- bd.dwBufferBytes = conf->bufsize_in;
hr = IDirectSoundCapture_CreateCaptureBuffer (
s->dsound_capture,
&bd,
@@ -203,7 +204,6 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
);
#else
bd.dwFlags = DSBCAPS_STICKYFOCUS | DSBCAPS_GETCURRENTPOSITION2;
- bd.dwBufferBytes = conf->bufsize_out;
hr = IDirectSound_CreateSoundBuffer (
s->dsound,
&bd,
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index d447e12..ff72d3f 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -31,6 +31,7 @@
#define AUDIO_CAP "dsound"
#include "audio_int.h"
+#include "qemu/host-utils.h"
#include <windows.h>
#include <mmsystem.h>
@@ -42,16 +43,10 @@
/* #define DEBUG_DSOUND */
typedef struct {
- int bufsize_in;
- int bufsize_out;
- int latency_millis;
-} DSoundConf;
-
-typedef struct {
LPDIRECTSOUND dsound;
LPDIRECTSOUNDCAPTURE dsound_capture;
struct audsettings settings;
- DSoundConf conf;
+ Audiodev *dev;
} dsound;
typedef struct {
@@ -247,9 +242,9 @@ static void GCC_FMT_ATTR (3, 4) dsound_logerr2 (
dsound_log_hresult (hr);
}
-static DWORD millis_to_bytes (struct audio_pcm_info *info, DWORD millis)
+static uint64_t usecs_to_bytes(struct audio_pcm_info *info, uint32_t usecs)
{
- return (millis * info->bytes_per_second) / 1000;
+ return muldiv64(usecs, info->bytes_per_second, 1000000);
}
#ifdef DEBUG_DSOUND
@@ -477,7 +472,7 @@ 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");
@@ -500,14 +495,14 @@ static int dsound_run_out (HWVoiceOut *hw, int live)
len = live << hwshift;
if (ds->first_time) {
- if (conf->latency_millis) {
+ if (dso->latency) {
DWORD cur_blat;
cur_blat = audio_ring_dist (wpos, ppos, bufsize);
ds->first_time = 0;
old_pos = wpos;
old_pos +=
- millis_to_bytes (&hw->info, conf->latency_millis) - cur_blat;
+ usecs_to_bytes(&hw->info, dso->latency) - cur_blat;
old_pos %= bufsize;
old_pos &= ~hw->info.align;
}
@@ -746,12 +741,6 @@ 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;
@@ -787,8 +776,17 @@ 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");
@@ -853,28 +851,6 @@ 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,
@@ -892,7 +868,6 @@ static struct audio_pcm_ops dsound_pcm_ops = {
struct audio_driver dsound_audio_driver = {
.name = "dsound",
.descr = "DirectSound http://wikipedia.org/wiki/DirectSound",
- .options = dsound_options,
.init = dsound_audio_init,
.fini = dsound_audio_fini,
.pcm_ops = &dsound_pcm_ops,
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 10/51] noaudio: port to -audiodev config
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (8 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 09/51] dsoundaudio: " Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 11/51] ossaudio: " Kővágó, Zoltán
` (40 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 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 108b02d..4c94a26 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -161,7 +161,6 @@ static struct audio_pcm_ops no_pcm_ops = {
struct audio_driver no_audio_driver = {
.name = "none",
.descr = "Timer based audio emulation",
- .options = NULL,
.init = no_audio_init,
.fini = no_audio_fini,
.pcm_ops = &no_pcm_ops,
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 11/51] ossaudio: port to -audiodev config
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (9 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 10/51] noaudio: " Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 12/51] paaudio: " Kővágó, Zoltán
` (39 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 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 bcf87ff..3d9f212 100644
--- a/audio/audio_legacy.c
+++ b/audio/audio_legacy.c
@@ -76,6 +76,21 @@ 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;
@@ -250,6 +265,8 @@ 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 6d64cf2..fbf35aa 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -29,6 +29,7 @@
#include "qemu-common.h"
#include "qemu/main-loop.h"
#include "qemu/host-utils.h"
+#include "qapi-visit.h"
#include "audio.h"
#include "trace.h"
@@ -39,16 +40,6 @@
#define USE_DSP_POLICY
#endif
-typedef struct OSSConf {
- int try_mmap;
- int nfrags;
- int fragsize;
- const char *devpath_out;
- const char *devpath_in;
- int exclusive;
- int policy;
-} OSSConf;
-
typedef struct OSSVoiceOut {
HWVoiceOut hw;
void *pcm_buf;
@@ -58,7 +49,7 @@ typedef struct OSSVoiceOut {
int fragsize;
int mmapped;
int pending;
- OSSConf *conf;
+ Audiodev *dev;
} OSSVoiceOut;
typedef struct OSSVoiceIn {
@@ -67,12 +58,12 @@ typedef struct OSSVoiceIn {
int fd;
int nfrags;
int fragsize;
- OSSConf *conf;
+ Audiodev *dev;
} OSSVoiceIn;
struct oss_params {
int freq;
- AudioFormat fmt;
+ int fmt;
int nchannels;
int nfrags;
int fragsize;
@@ -264,19 +255,26 @@ static int oss_get_version (int fd, int *version, const char *typ)
}
#endif
-static int oss_open (int in, struct oss_params *req,
- struct oss_params *obt, int *pfd, OSSConf* conf)
+static int oss_open(int in, struct oss_params *req, audsettings *as,
+ struct oss_params *obt, int *pfd, Audiodev *dev)
{
+ AudiodevOssOptions *oopts = dev->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) {
@@ -287,6 +285,8 @@ static int oss_open (int in, struct oss_params *req,
freq = req->freq;
nchannels = req->nchannels;
fmt = req->fmt;
+ req->nfrags = pdo->has_buffer_count ? pdo->buffer_count : 4;
+ req->fragsize = audio_buffer_bytes(pdo, as, 23220);
if (ioctl (fd, SNDCTL_DSP_SAMPLESIZE, &fmt)) {
oss_logerr2 (errno, typ, "Failed to set sample size %d\n", req->fmt);
@@ -310,18 +310,18 @@ static int oss_open (int in, struct oss_params *req,
}
#ifdef USE_DSP_POLICY
- if (conf->policy >= 0) {
+ if (policy >= 0) {
int version;
if (!oss_get_version (fd, &version, typ)) {
trace_oss_version(version);
if (version >= 0x040000) {
- int policy = conf->policy;
- if (ioctl (fd, SNDCTL_DSP_POLICY, &policy)) {
+ int policy2 = policy;
+ if (ioctl (fd, SNDCTL_DSP_POLICY, &policy2)) {
oss_logerr2 (errno, typ,
"Failed to set timing policy to %d\n",
- conf->policy);
+ policy);
goto err;
}
setfragment = 0;
@@ -504,17 +504,16 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
int fd;
AudioFormat effective_fmt;
struct audsettings obt_as;
- OSSConf *conf = drv_opaque;
+ Audiodev *dev = drv_opaque;
+ AudiodevOssOptions *oopts = dev->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;
}
@@ -541,7 +540,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
oss->mmapped = 0;
- if (conf->try_mmap) {
+ if (oopts->has_try_mmap && oopts->try_mmap) {
oss->pcm_buf = mmap (
NULL,
hw->samples << hw->info.shift,
@@ -601,7 +600,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
}
oss->fd = fd;
- oss->conf = conf;
+ oss->dev = dev;
return 0;
}
@@ -609,16 +608,12 @@ static int oss_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
int trig;
OSSVoiceOut *oss = (OSSVoiceOut *) hw;
+ AudiodevOssPerDirectionOptions *opdo = oss->dev->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) {
@@ -673,16 +668,14 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
int fd;
AudioFormat effective_fmt;
struct audsettings obt_as;
- OSSConf *conf = drv_opaque;
+ Audiodev *dev = drv_opaque;
oss->fd = -1;
req.fmt = aud_to_ossfmt (as->fmt, as->endianness);
req.freq = as->freq;
req.nchannels = as->nchannels;
- req.fragsize = conf->fragsize;
- req.nfrags = conf->nfrags;
- if (oss_open (1, &req, &obt, &fd, conf)) {
+ if (oss_open(1, &req, as, &obt, &fd, dev)) {
return -1;
}
@@ -716,7 +709,7 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
}
oss->fd = fd;
- oss->conf = conf;
+ oss->dev = dev;
return 0;
}
@@ -807,16 +800,12 @@ static int oss_read (SWVoiceIn *sw, void *buf, int size)
static int oss_ctl_in (HWVoiceIn *hw, int cmd, ...)
{
OSSVoiceIn *oss = (OSSVoiceIn *) hw;
+ AudiodevOssPerDirectionOptions *opdo = oss->dev->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);
@@ -836,82 +825,25 @@ static int oss_ctl_in (HWVoiceIn *hw, int cmd, ...)
return 0;
}
-static OSSConf glob_conf = {
- .try_mmap = 0,
- .nfrags = 4,
- .fragsize = 4096,
- .devpath_out = "/dev/dsp",
- .devpath_in = "/dev/dsp",
- .exclusive = 0,
- .policy = 5
-};
-
static void *oss_audio_init(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 wont work)"
- },
-#ifdef USE_DSP_POLICY
- {
- .name = "POLICY",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.policy,
- .descr = "Set the timing policy of the device, -1 to use fragment mode",
- },
-#endif
- { /* End of list */ }
-};
-
static struct audio_pcm_ops oss_pcm_ops = {
.init_out = oss_init_out,
.fini_out = oss_fini_out,
@@ -929,7 +861,6 @@ static struct audio_pcm_ops oss_pcm_ops = {
struct audio_driver oss_audio_driver = {
.name = "oss",
.descr = "OSS http://www.opensound.com",
- .options = oss_options,
.init = oss_audio_init,
.fini = oss_audio_fini,
.pcm_ops = &oss_pcm_ops,
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 12/51] paaudio: port to -audiodev config
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (10 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 11/51] ossaudio: " Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 13/51] sdlaudio: " Kővágó, Zoltán
` (38 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 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 3d9f212..3eab2b9 100644
--- a/audio/audio_legacy.c
+++ b/audio/audio_legacy.c
@@ -91,6 +91,15 @@ 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;
@@ -267,6 +276,8 @@ 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 dd46fa0..d68f6ec 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -1,6 +1,7 @@
/* public domain */
#include "qemu-common.h"
#include "audio.h"
+#include "qapi-visit.h"
#include <pulse/pulseaudio.h>
@@ -9,14 +10,7 @@
#include "audio_pt_int.h"
typedef struct {
- int samples;
- char *server;
- char *sink;
- char *source;
-} PAConf;
-
-typedef struct {
- PAConf conf;
+ Audiodev *dev;
pa_threaded_mainloop *mainloop;
pa_context *context;
} paaudio;
@@ -31,6 +25,7 @@ typedef struct {
void *pcm_buf;
struct audio_pt pt;
paaudio *g;
+ int samples;
} PAVoiceOut;
typedef struct {
@@ -45,6 +40,7 @@ typedef struct {
const void *read_data;
size_t read_index, read_length;
paaudio *g;
+ int samples;
} PAVoiceIn;
static void qpa_audio_fini(void *opaque);
@@ -226,7 +222,7 @@ static void *qpa_thread_out (void *arg)
}
}
- decr = to_mix = audio_MIN (pa->live, pa->g->conf.samples >> 2);
+ decr = to_mix = audio_MIN (pa->live, pa->samples >> 2);
rpos = pa->rpos;
if (audio_pt_unlock (&pa->pt, AUDIO_FUNC)) {
@@ -318,7 +314,7 @@ static void *qpa_thread_in (void *arg)
}
}
- incr = to_grab = audio_MIN (pa->dead, pa->g->conf.samples >> 2);
+ incr = to_grab = audio_MIN (pa->dead, pa->samples >> 2);
wpos = pa->wpos;
if (audio_pt_unlock (&pa->pt, AUDIO_FUNC)) {
@@ -545,6 +541,8 @@ 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;
@@ -565,7 +563,7 @@ 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 */
@@ -577,7 +575,8 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
}
audio_pcm_init_info (&hw->info, &obt_as);
- hw->samples = g->conf.samples;
+ hw->samples = pa->samples = audio_buffer_samples(g->dev->out, &obt_as,
+ 46440);
pa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
pa->rpos = hw->rpos;
if (!pa->pcm_buf) {
@@ -611,6 +610,8 @@ 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;
@@ -622,7 +623,7 @@ 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 */
@@ -634,7 +635,8 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
}
audio_pcm_init_info (&hw->info, &obt_as);
- hw->samples = g->conf.samples;
+ hw->samples = pa->samples = audio_buffer_samples(g->dev->in, &obt_as,
+ 46440);
pa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
pa->wpos = hw->wpos;
if (!pa->pcm_buf) {
@@ -808,14 +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 = {
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,
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 13/51] sdlaudio: port to -audiodev config
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (11 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 12/51] paaudio: " Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 14/51] spiceaudio: " Kővágó, Zoltán
` (37 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 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 3eab2b9..4d8e2b9 100644
--- a/audio/audio_legacy.c
+++ b/audio/audio_legacy.c
@@ -100,6 +100,11 @@ 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;
@@ -278,6 +283,8 @@ 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 580c613..796238a 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -44,18 +44,13 @@ typedef struct SDLVoiceOut {
int decr;
} SDLVoiceOut;
-static struct {
- int nb_samples;
-} conf = {
- .nb_samples = 1024
-};
-
static struct SDLAudioState {
int exit;
SDL_mutex *mutex;
SDL_sem *sem;
int initialized;
bool driver_created;
+ Audiodev *dev;
} glob_sdl;
typedef struct SDLAudioState SDLAudioState;
@@ -347,7 +342,7 @@ static int sdl_init_out(HWVoiceOut *hw, struct audsettings *as,
req.freq = as->freq;
req.format = aud_to_sdlfmt (as->fmt);
req.channels = as->nchannels;
- req.samples = conf.nb_samples;
+ req.samples = audio_buffer_samples(s->dev->out, as, 11610);
req.callback = sdl_callback;
req.userdata = sdl;
@@ -420,6 +415,7 @@ static void *sdl_audio_init(Audiodev *dev)
}
s->driver_created = true;
+ s->dev = dev;
return s;
}
@@ -431,18 +427,9 @@ static void sdl_audio_fini (void *opaque)
SDL_DestroyMutex (s->mutex);
SDL_QuitSubSystem (SDL_INIT_AUDIO);
s->driver_created = false;
+ s->dev = NULL;
}
-static struct audio_option sdl_options[] = {
- {
- .name = "SAMPLES",
- .tag = AUD_OPT_INT,
- .valp = &conf.nb_samples,
- .descr = "Size of SDL buffer in samples"
- },
- { /* End of list */ }
-};
-
static struct audio_pcm_ops sdl_pcm_ops = {
.init_out = sdl_init_out,
.fini_out = sdl_fini_out,
@@ -454,7 +441,6 @@ static struct audio_pcm_ops sdl_pcm_ops = {
struct audio_driver sdl_audio_driver = {
.name = "sdl",
.descr = "SDL http://www.libsdl.org",
- .options = sdl_options,
.init = sdl_audio_init,
.fini = sdl_audio_fini,
.pcm_ops = &sdl_pcm_ops,
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 14/51] spiceaudio: port to -audiodev config
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (12 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 13/51] sdlaudio: " Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 15/51] wavaudio: " Kővágó, Zoltán
` (36 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 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 e6da29a..a9b9a1d 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -371,10 +371,6 @@ 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,
@@ -392,7 +388,6 @@ static struct audio_pcm_ops audio_callbacks = {
struct audio_driver spice_audio_driver = {
.name = "spice",
.descr = "spice audio driver",
- .options = audio_options,
.init = spice_audio_init,
.fini = spice_audio_fini,
.pcm_ops = &audio_callbacks,
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 15/51] wavaudio: port to -audiodev config
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (13 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 14/51] spiceaudio: " Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 16/51] audio: -audiodev command line option: cleanup Kővágó, Zoltán
` (35 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 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 4d8e2b9..2004097 100644
--- a/audio/audio_legacy.c
+++ b/audio/audio_legacy.c
@@ -105,6 +105,14 @@ 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;
@@ -285,6 +293,8 @@ 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 8b0ba91..b1a3b1d 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -23,6 +23,7 @@
*/
#include "hw/hw.h"
#include "qemu/timer.h"
+#include "qapi-visit.h"
#include "audio.h"
#define AUDIO_CAP "wav"
@@ -36,11 +37,6 @@ typedef struct WAVVoiceOut {
int total_samples;
} WAVVoiceOut;
-typedef struct {
- struct audsettings settings;
- const char *wav_path;
-} WAVConf;
-
static int wav_run_out (HWVoiceOut *hw, int live)
{
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
@@ -111,8 +107,10 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
0x02, 0x00, 0x44, 0xac, 0x00, 0x00, 0x10, 0xb1, 0x02, 0x00, 0x04,
0x00, 0x10, 0x00, 0x64, 0x61, 0x74, 0x61, 0x00, 0x00, 0x00, 0x00
};
- WAVConf *conf = drv_opaque;
- struct audsettings wav_as = conf->settings;
+ Audiodev *dev = drv_opaque;
+ AudiodevWavOptions *wopts = dev->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) {
@@ -153,10 +151,10 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
le_store (hdr + 28, hw->info.freq << (bits16 + stereo), 4);
le_store (hdr + 32, 1 << (bits16 + stereo), 2);
- wav->f = fopen (conf->wav_path, "wb");
+ wav->f = fopen(wav_path, "wb");
if (!wav->f) {
dolog ("Failed to open wave file `%s'\nReason: %s\n",
- conf->wav_path, strerror (errno));
+ wav_path, strerror(errno));
g_free (wav->pcm_buf);
wav->pcm_buf = NULL;
return -1;
@@ -224,54 +222,17 @@ static int wav_ctl_out (HWVoiceOut *hw, int cmd, ...)
return 0;
}
-static WAVConf glob_conf = {
- .settings.freq = 44100,
- .settings.nchannels = 2,
- .settings.fmt = AUDIO_FORMAT_S16,
- .wav_path = "qemu.wav"
-};
-
static void *wav_audio_init(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,
@@ -283,7 +244,6 @@ static struct audio_pcm_ops wav_pcm_ops = {
struct audio_driver wav_audio_driver = {
.name = "wav",
.descr = "WAV renderer http://wikipedia.org/wiki/WAV",
- .options = wav_options,
.init = wav_audio_init,
.fini = wav_audio_fini,
.pcm_ops = &wav_pcm_ops,
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 16/51] audio: -audiodev command line option: cleanup
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (14 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 15/51] wavaudio: " Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 17/51] audio: reduce glob_audio_state usage Kővágó, Zoltán
` (34 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Remove no longer needed codes.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio.c | 202 ++----------------------------------------------------
audio/audio_int.h | 17 -----
2 files changed, 4 insertions(+), 215 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index fcbc543..115d41d 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -142,114 +142,6 @@ 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) {
- *defaultp = 0;
- val = atoi (strval);
- return val;
- }
- else {
- *defaultp = 1;
- return defval;
- }
-}
-
-static const char *audio_get_conf_str (const char *key,
- const char *defval,
- int *defaultp)
-{
- const char *val = getenv (key);
- if (!val) {
- *defaultp = 1;
- return defval;
- }
- else {
- *defaultp = 0;
- return val;
- }
-}
-
void AUD_vlog (const char *cap, const char *fmt, va_list ap)
{
if (cap) {
@@ -268,89 +160,6 @@ 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 (AUDIO_FUNC, !prefix)) {
- dolog ("prefix = NULL\n");
- return;
- }
-
- if (audio_bug (AUDIO_FUNC, !opt)) {
- dolog ("opt = NULL\n");
- return;
- }
-
- preflen = strlen (prefix);
-
- for (; opt->name; opt++) {
- size_t len, i;
- int def;
-
- if (!opt->valp) {
- dolog ("Option value pointer for `%s' is not set\n",
- opt->name);
- continue;
- }
-
- len = strlen (opt->name);
- /* len of opt->name + len of prefix + size of qemu_prefix
- * (includes trailing zero) + zero + underscore (on behalf of
- * sizeof) */
- optlen = len + preflen + sizeof (qemu_prefix) + 1;
- optname = g_malloc (optlen);
-
- pstrcpy (optname, optlen, qemu_prefix);
-
- /* copy while upper-casing, including trailing zero */
- for (i = 0; i <= preflen; ++i) {
- optname[i + sizeof (qemu_prefix) - 1] = qemu_toupper(prefix[i]);
- }
- pstrcat (optname, optlen, "_");
- pstrcat (optname, optlen, opt->name);
-
- def = 1;
- switch (opt->tag) {
- case AUD_OPT_BOOL:
- case AUD_OPT_INT:
- {
- int *intp = opt->valp;
- *intp = audio_get_conf_int (optname, *intp, &def);
- }
- break;
-
- case AUD_OPT_FMT:
- {
- AudioFormat *fmtp = opt->valp;
- *fmtp = audio_get_conf_fmt (optname, *fmtp, &def);
- }
- break;
-
- case AUD_OPT_STR:
- {
- const char **strp = opt->valp;
- *strp = audio_get_conf_str (optname, *strp, &def);
- }
- break;
-
- default:
- dolog ("Bad value tag for option `%s' - %d\n",
- optname, opt->tag);
- break;
- }
-
- if (!opt->overriddenp) {
- opt->overriddenp = &opt->overridden;
- }
- *opt->overriddenp = !def;
- g_free (optname);
- }
-}
-
static void audio_print_settings (struct audsettings *as)
{
dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels);
@@ -1047,7 +856,7 @@ void AUD_set_active_out (SWVoiceOut *sw, int on)
if (!hw->enabled) {
hw->enabled = 1;
if (s->vm_running) {
- hw->pcm_ops->ctl_out(hw, VOICE_ENABLE, true /* todo */);
+ hw->pcm_ops->ctl_out(hw, VOICE_ENABLE);
audio_reset_timer (s);
}
}
@@ -1092,7 +901,7 @@ void AUD_set_active_in (SWVoiceIn *sw, int on)
if (!hw->enabled) {
hw->enabled = 1;
if (s->vm_running) {
- hw->pcm_ops->ctl_in(hw, VOICE_ENABLE, true /* todo */);
+ hw->pcm_ops->ctl_in(hw, VOICE_ENABLE);
audio_reset_timer (s);
}
}
@@ -1412,9 +1221,6 @@ 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) {
@@ -1439,11 +1245,11 @@ static void audio_vm_change_state_handler (void *opaque, int running,
s->vm_running = running;
while ((hwo = audio_pcm_hw_find_any_enabled_out (hwo))) {
- hwo->pcm_ops->ctl_out(hwo, op, 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 9139f42..59b2362 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -32,22 +32,6 @@
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;
@@ -143,7 +127,6 @@ struct SWVoiceIn {
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;
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 17/51] audio: reduce glob_audio_state usage
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (15 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 16/51] audio: -audiodev command line option: cleanup Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 18/51] audio: basic support for multi backend audio Kővágó, Zoltán
` (33 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 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 | 41 ++++++++++++++++++++---------------------
audio/audio_int.h | 4 ++++
audio/audio_template.h | 46 ++++++++++++++++++++++------------------------
3 files changed, 46 insertions(+), 45 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index 115d41d..2aa2071 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -367,12 +367,10 @@ 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)) {
@@ -449,7 +447,7 @@ 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);
@@ -762,15 +760,15 @@ static void audio_pcm_print_info (const char *cap, struct audio_pcm_info *info)
/*
* Timer
*/
-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;
@@ -778,7 +776,7 @@ 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 (s->ts,
qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + s->period_ticks);
}
@@ -847,7 +845,7 @@ 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;
@@ -894,7 +892,7 @@ 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) {
@@ -1017,7 +1015,7 @@ 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;
@@ -1121,7 +1119,7 @@ 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, min;
@@ -1224,8 +1222,8 @@ 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;
}
@@ -1244,11 +1242,11 @@ 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);
@@ -1260,7 +1258,7 @@ static void audio_atexit (void)
HWVoiceOut *hwo = NULL;
HWVoiceIn *hwi = NULL;
- while ((hwo = audio_pcm_hw_find_any_out (hwo))) {
+ while ((hwo = audio_pcm_hw_find_any_out(s, hwo))) {
SWVoiceCap *sc;
if (hwo->enabled) {
@@ -1278,7 +1276,7 @@ static void audio_atexit (void)
}
}
- while ((hwi = audio_pcm_hw_find_any_in (hwi))) {
+ while ((hwi = audio_pcm_hw_find_any_in(s, hwi))) {
if (hwi->enabled) {
hwi->pcm_ops->ctl_in (hwi, VOICE_DISABLE);
}
@@ -1459,7 +1457,7 @@ 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;
@@ -1476,6 +1474,7 @@ CaptureVoiceOut *AUD_add_capture (
}
hw = &cap->hw;
+ hw->s = s;
QLIST_INIT (&hw->sw_head);
QLIST_INIT (&cap->cb_head);
@@ -1509,7 +1508,7 @@ CaptureVoiceOut *AUD_add_capture (
QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
hw = NULL;
- while ((hw = audio_pcm_hw_find_any_out (hw))) {
+ while ((hw = audio_pcm_hw_find_any_out(s, hw))) {
audio_attach_capture (hw);
}
return cap;
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 59b2362..101081b 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -51,6 +51,7 @@ struct audio_pcm_info {
typedef struct SWVoiceCap SWVoiceCap;
typedef struct HWVoiceOut {
+ AudioState *s;
int enabled;
int poll_mode;
int pending_disable;
@@ -72,6 +73,7 @@ typedef struct HWVoiceOut {
} HWVoiceOut;
typedef struct HWVoiceIn {
+ AudioState *s;
int enabled;
int poll_mode;
struct audio_pcm_info info;
@@ -93,6 +95,7 @@ typedef struct HWVoiceIn {
struct SWVoiceOut {
QEMUSoundCard *card;
+ AudioState *s;
struct audio_pcm_info info;
t_sample *conv;
int64_t ratio;
@@ -110,6 +113,7 @@ struct SWVoiceOut {
struct SWVoiceIn {
QEMUSoundCard *card;
+ AudioState *s;
int active;
struct audio_pcm_info info;
int64_t ratio;
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 096b2b3..455e50d 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;
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 18/51] audio: basic support for multi backend audio
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (16 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 17/51] audio: reduce glob_audio_state usage Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 19/51] audio: add audiodev properties to frontends Kővágó, Zoltán
` (32 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 UTC (permalink / raw)
To: qemu-devel
Cc: Markus Armbruster, Paolo Bonzini, Gerd Hoffmann, Luiz Capitulino
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 | 98 +++++++++++++++++++++++++++++++++++++-------------
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 ++
vl.c | 3 +-
11 files changed, 129 insertions(+), 39 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index 2aa2071..b435a35 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -55,7 +55,8 @@ struct audio_driver *drvtab[] = {
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,
@@ -1193,11 +1194,14 @@ 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;
@@ -1252,9 +1256,8 @@ static void audio_vm_change_state_handler (void *opaque, int running,
audio_reset_timer (s);
}
-static void audio_atexit (void)
+static void free_audio_state(AudioState *s)
{
- AudioState *s = &glob_audio_state;
HWVoiceOut *hwo = NULL;
HWVoiceIn *hwi = NULL;
@@ -1288,6 +1291,16 @@ static void audio_atexit (void)
}
qapi_free_Audiodev(s->dev);
+ g_free(s);
+}
+
+static void audio_atexit(void)
+{
+ 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 = {
@@ -1300,26 +1313,27 @@ 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;
QemuOptsList *list = NULL; /* silence gcc warning about uninitialized
* variable */
- if (s->drv) {
- if (dev) {
- dolog("Cannot create more than one audio backend, sorry\n");
- qapi_free_Audiodev(dev);
- }
- return -1;
- }
-
if (dev) {
drvname = AudiodevDriver_lookup[dev->driver];
+ } else if (!QTAILQ_EMPTY(&audio_states)) {
+ /* todo: chack for -audiodev we have normal audiodev selection support */
+ return QTAILQ_FIRST(&audio_states);
} else {
audio_handle_legacy_opts();
list = qemu_find_opts("audiodev");
@@ -1328,12 +1342,18 @@ 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_atexit);
+ if (!atexit_registered) {
+ atexit(audio_atexit);
+ atexit_registered = true;
+ }
+ QTAILQ_INSERT_TAIL(&audio_states, s, list);
s->ts = timer_new_ns(QEMU_CLOCK_VIRTUAL, audio_timer, s);
if (!s->ts) {
@@ -1414,15 +1434,18 @@ 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)
@@ -1432,16 +1455,21 @@ 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 = QTAILQ_FIRST(&audio_states);
+ }
+
if (audio_validate_settings (as)) {
dolog ("Invalid settings were passed when trying to add capture\n");
audio_print_settings (as);
@@ -1677,7 +1705,7 @@ 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)
@@ -1743,3 +1771,25 @@ 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 177a673..0085a07 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -80,8 +80,10 @@ 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;
@@ -96,7 +98,8 @@ 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
@@ -164,11 +167,14 @@ 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);
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 /* audio.h */
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 101081b..1d81658 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -189,6 +189,8 @@ struct AudioState {
int nb_hw_voices_in;
int vm_running;
int64_t period_ticks;
+
+ QTAILQ_ENTRY(AudioState) list;
};
extern struct audio_driver no_audio_driver;
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 455e50d..5a3dc90 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 (AUDIO_FUNC, !card || !name || !callback_fn || !as)) {
diff --git a/audio/wavcapture.c b/audio/wavcapture.c
index 798acdd..8b8d3d0 100644
--- a/audio/wavcapture.c
+++ b/audio/wavcapture.c
@@ -104,8 +104,8 @@ 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;
@@ -172,7 +172,7 @@ 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 bb52e4d..cc75baa 100644
--- a/hmp-commands.hx
+++ b/hmp-commands.hx
@@ -757,16 +757,17 @@ 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)",
.mhandler.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 e7e7ae2..a56ef20 100644
--- a/monitor.c
+++ b/monitor.c
@@ -1559,7 +1559,17 @@ 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));
@@ -1567,7 +1577,7 @@ 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 97f021f..184ffd4 100644
--- a/qemu-options.hx
+++ b/qemu-options.hx
@@ -1621,6 +1621,11 @@ everybody else. 'ignore' completely ignores the shared flag and
allows everybody connect unconditionally. Doesn't conform to the rfb
spec but is traditional QEMU behavior.
+@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 2dd269b..dc54e06 100644
--- a/ui/vnc.c
+++ b/ui/vnc.c
@@ -1154,7 +1154,7 @@ 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");
}
@@ -3262,6 +3262,9 @@ static QemuOptsList qemu_vnc_opts = {
},{
.name = "non-adaptive",
.type = QEMU_OPT_BOOL,
+ },{
+ .name = "audiodev",
+ .type = QEMU_OPT_STRING,
},
{ /* end of list */ }
},
@@ -3480,6 +3483,7 @@ void vnc_display_open(const char *id, Error **errp)
#endif
int acl = 0;
int lock_key_sync = 1;
+ const char *audiodev;
if (!vs) {
error_setg(errp, "VNC display not active");
@@ -3723,6 +3727,15 @@ void vnc_display_open(const char *id, Error **errp)
#endif
vs->lock_key_sync = lock_key_sync;
+ audiodev = qemu_opt_get(opts, "audiodev");
+ if (audiodev) {
+ vs->audio_state = audio_state_by_name(audiodev);
+ if (!vs->audio_state) {
+ error_setg(errp, "Audiodev '%s' not found", audiodev);
+ goto fail;
+ }
+ }
+
device_id = qemu_opt_get(opts, "display");
if (device_id) {
DeviceState *dev;
diff --git a/ui/vnc.h b/ui/vnc.h
index a95cc15..21ecfdb 100644
--- a/ui/vnc.h
+++ b/ui/vnc.h
@@ -182,6 +182,8 @@ struct VncDisplay
#ifdef CONFIG_VNC_SASL
VncDisplaySASL sasl;
#endif
+
+ AudioState *audio_state;
};
typedef struct VncTight {
diff --git a/vl.c b/vl.c
index de47b6b..a8e8beb 100644
--- a/vl.c
+++ b/vl.c
@@ -4326,6 +4326,8 @@ int main(int argc, char **argv, char **envp)
exit(0);
}
+ audio_set_options();
+
machine_opts = qemu_get_machine_opts();
if (qemu_opt_foreach(machine_opts, machine_set_property, current_machine,
NULL)) {
@@ -4540,7 +4542,6 @@ int main(int argc, char **argv, char **envp)
realtime_init();
- audio_set_options();
audio_init();
cpu_synchronize_all_post_init();
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 19/51] audio: add audiodev properties to frontends
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (17 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 18/51] audio: basic support for multi backend audio Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 20/51] audio: audiodev= parameters no longer optional when -audiodev present Kővágó, Zoltán
` (31 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 UTC (permalink / raw)
To: qemu-devel; +Cc: Michael Walle, Gerd Hoffmann
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
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
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 0085a07..68545b6 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -177,4 +177,7 @@ 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 /* audio.h */
diff --git a/hw/audio/ac97.c b/hw/audio/ac97.c
index fa75f33..a835cb6 100644
--- a/hw/audio/ac97.c
+++ b/hw/audio/ac97.c
@@ -1393,6 +1393,7 @@ 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 ca8de05..b5202f9 100644
--- a/hw/audio/adlib.c
+++ b/hw/audio/adlib.c
@@ -347,6 +347,7 @@ 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 626a173..43a1a39 100644
--- a/hw/audio/cs4231a.c
+++ b/hw/audio/cs4231a.c
@@ -672,6 +672,7 @@ 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 a1c21b8..614ff19 100644
--- a/hw/audio/es1370.c
+++ b/hw/audio/es1370.c
@@ -1042,6 +1042,11 @@ 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);
@@ -1056,6 +1061,7 @@ static void es1370_class_init (ObjectClass *klass, void *data)
set_bit(DEVICE_CATEGORY_SOUND, dc->categories);
dc->desc = "ENSONIQ AudioPCI ES1370";
dc->vmsd = &vmstate_es1370;
+ dc->props = es1370_properties;
}
static const TypeInfo es1370_info = {
@@ -1072,4 +1078,3 @@ static void es1370_register_types (void)
}
type_init (es1370_register_types)
-
diff --git a/hw/audio/gus.c b/hw/audio/gus.c
index 0e1d15f..e25a033 100644
--- a/hw/audio/gus.c
+++ b/hw/audio/gus.c
@@ -279,6 +279,7 @@ 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 8693b7a..dba3331 100644
--- a/hw/audio/hda-codec.c
+++ b/hw/audio/hda-codec.c
@@ -613,6 +613,7 @@ 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_END_OF_LIST(),
diff --git a/hw/audio/milkymist-ac97.c b/hw/audio/milkymist-ac97.c
index 15169e2..b63fd6f 100644
--- a/hw/audio/milkymist-ac97.c
+++ b/hw/audio/milkymist-ac97.c
@@ -323,6 +323,11 @@ 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);
@@ -331,6 +336,7 @@ static void milkymist_ac97_class_init(ObjectClass *klass, void *data)
k->init = milkymist_ac97_init;
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 302debf..85d6d00 100644
--- a/hw/audio/pcspk.c
+++ b/hw/audio/pcspk.c
@@ -181,6 +181,7 @@ static void pcspk_realizefn(DeviceState *dev, Error **errp)
}
static Property pcspk_properties[] = {
+ DEFINE_AUDIO_PROPERTIES(PCSpkState, card),
DEFINE_PROP_UINT32("iobase", PCSpkState, iobase, -1),
DEFINE_PROP_PTR("pit", PCSpkState, pit),
DEFINE_PROP_END_OF_LIST(),
diff --git a/hw/audio/pl041.c b/hw/audio/pl041.c
index 19982f2..c003b26 100644
--- a/hw/audio/pl041.c
+++ b/hw/audio/pl041.c
@@ -616,6 +616,7 @@ 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 163ced0..92318c7 100644
--- a/hw/audio/sb16.c
+++ b/hw/audio/sb16.c
@@ -1386,6 +1386,7 @@ 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 4c4333c..049ddf9 100644
--- a/hw/audio/wm8750.c
+++ b/hw/audio/wm8750.c
@@ -695,6 +695,11 @@ 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);
@@ -705,6 +710,7 @@ 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 921e799..9dd1a7b 100644
--- a/hw/core/qdev-properties-system.c
+++ b/hw/core/qdev-properties-system.c
@@ -10,6 +10,7 @@
* See the COPYING file in the top-level directory.
*/
+#include "audio/audio.h"
#include "net/net.h"
#include "hw/qdev.h"
#include "qapi/qmp/qerror.h"
@@ -357,6 +358,62 @@ PropertyInfo qdev_prop_vlan = {
.set = set_vlan,
};
+/* --- audiodev --- */
+static void get_audiodev(Object *obj, Visitor *v, void *opaque,
+ const char *name, 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, &p, name, errp);
+ g_free(p);
+}
+
+static void set_audiodev(Object *obj, Visitor *v, void *opaque,
+ const char *name, 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, &str, name, &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);
+}
+
+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 9a60569..deb9032 100644
--- a/hw/usb/dev-audio.c
+++ b/hw/usb/dev-audio.c
@@ -662,6 +662,7 @@ 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 77538a8..edebcfa 100644
--- a/include/hw/qdev-properties.h
+++ b/include/hw/qdev-properties.h
@@ -26,6 +26,7 @@ extern PropertyInfo qdev_prop_vlan;
extern PropertyInfo qdev_prop_pci_devfn;
extern PropertyInfo qdev_prop_blocksize;
extern PropertyInfo qdev_prop_pci_host_devaddr;
+extern PropertyInfo qdev_prop_audiodev;
extern PropertyInfo qdev_prop_arraylen;
#define DEFINE_PROP(_name, _state, _field, _prop, _type) { \
@@ -163,6 +164,8 @@ extern PropertyInfo qdev_prop_arraylen;
DEFINE_PROP_DEFAULT(_n, _s, _f, 0, qdev_prop_blocksize, uint16_t)
#define DEFINE_PROP_PCI_HOST_DEVADDR(_n, _s, _f) \
DEFINE_PROP(_n, _s, _f, qdev_prop_pci_host_devaddr, PCIHostDeviceAddress)
+#define DEFINE_PROP_AUDIODEV(_n, _s, _f) \
+ DEFINE_PROP(_n, _s, _f, qdev_prop_audiodev, QEMUSoundCard)
#define DEFINE_PROP_END_OF_LIST() \
{}
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 20/51] audio: audiodev= parameters no longer optional when -audiodev present
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (18 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 19/51] audio: add audiodev properties to frontends Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 21/51] paaudio: do not move stream when sink/source name is specified Kővágó, Zoltán
` (30 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
Changes from v1:
* (probably) more clear error messages
---
audio/audio.c | 19 +++-
audio/paaudio.c | 297 ++++++++++++++++++++++++++++++++------------------------
2 files changed, 185 insertions(+), 131 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index b435a35..312b955 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -69,6 +69,8 @@ const struct mixeng_volume nominal_volume = {
#endif
};
+static bool legacy_config;
+
#ifdef AUDIO_IS_FLAWLESS_AND_NO_CHECKS_ARE_REQURIED
#error No its not
#else
@@ -1318,7 +1320,7 @@ static Audiodev *parse_option(QemuOpts *opts, Error **errp);
* 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 AudioState *audio_init(Audiodev *dev, const char *name)
{
static bool atexit_registered;
size_t i;
@@ -1332,9 +1334,13 @@ static AudioState *audio_init(Audiodev *dev)
if (dev) {
drvname = AudiodevDriver_lookup[dev->driver];
} else if (!QTAILQ_EMPTY(&audio_states)) {
- /* todo: chack for -audiodev 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_config = true;
audio_handle_legacy_opts();
list = qemu_find_opts("audiodev");
dev = parse_option(QTAILQ_FIRST(&list->head), &error_abort);
@@ -1440,7 +1446,7 @@ 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);
@@ -1466,7 +1472,10 @@ CaptureVoiceOut *AUD_add_capture(
struct capture_callback *cb;
if (!s) {
- /* todo: remove when we have normal audiodev selection support */
+ if (!legacy_config) {
+ dolog("You must specify audiodev when trying to capture\n");
+ goto err0;
+ }
s = QTAILQ_FIRST(&audio_states);
}
@@ -1705,7 +1714,7 @@ 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 d68f6ec..0c9427f 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -9,10 +9,21 @@
#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 {
@@ -43,7 +54,7 @@ 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, ...)
{
@@ -106,11 +117,11 @@ 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;
@@ -119,11 +130,11 @@ 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;
}
@@ -146,53 +157,53 @@ 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;
}
@@ -430,13 +441,13 @@ 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:
@@ -449,14 +460,14 @@ 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:
@@ -467,13 +478,13 @@ 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,
@@ -484,50 +495,48 @@ 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;
}
@@ -543,6 +552,7 @@ 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;
@@ -560,7 +570,7 @@ 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,
@@ -612,6 +622,7 @@ 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;
@@ -620,7 +631,7 @@ 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,
@@ -708,7 +719,7 @@ 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+ */
@@ -728,28 +739,28 @@ 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;
@@ -760,7 +771,7 @@ 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);
@@ -780,123 +791,157 @@ 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);
/* FIXME: use the upcoming "set_source_output_{volume,mute}" */
- op = pa_context_set_source_volume_by_index (g->context,
+ op = pa_context_set_source_volume_by_index(c->context,
pa_stream_get_device_index (pa->stream),
&v, NULL, NULL);
if (!op) {
- qpa_logerr (pa_context_errno (g->context),
+ qpa_logerr (pa_context_errno(c->context),
"set_source_volume() failed\n");
} else {
pa_operation_unref(op);
}
- op = pa_context_set_source_mute_by_index (g->context,
+ op = pa_context_set_source_mute_by_index(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_source_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);
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 21/51] paaudio: do not move stream when sink/source name is specified
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (19 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 20/51] audio: audiodev= parameters no longer optional when -audiodev present Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 22/51] paaudio: properly disconnect streams in fini_* Kővágó, Zoltán
` (29 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 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 0c9427f..53561dd 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -515,6 +515,11 @@ 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 {
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 22/51] paaudio: properly disconnect streams in fini_*
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (20 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 21/51] paaudio: do not move stream when sink/source name is specified Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 23/51] audio: remove audio_MIN, audio_MAX Kővágó, Zoltán
` (28 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
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 | 23 +++++++++++++++++++++--
1 file changed, 21 insertions(+), 2 deletions(-)
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 53561dd..c9b2fcd 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -679,6 +679,25 @@ 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;
@@ -690,7 +709,7 @@ static void qpa_fini_out (HWVoiceOut *hw)
audio_pt_join (&pa->pt, &ret, AUDIO_FUNC);
if (pa->stream) {
- pa_stream_unref (pa->stream);
+ qpa_simple_disconnect(pa->g->conn, pa->stream);
pa->stream = NULL;
}
@@ -710,7 +729,7 @@ static void qpa_fini_in (HWVoiceIn *hw)
audio_pt_join (&pa->pt, &ret, AUDIO_FUNC);
if (pa->stream) {
- pa_stream_unref (pa->stream);
+ qpa_simple_disconnect(pa->g->conn, pa->stream);
pa->stream = NULL;
}
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 23/51] audio: remove audio_MIN, audio_MAX
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (21 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 22/51] paaudio: properly disconnect streams in fini_* Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 24/51] audio: do not run each backend in audio_run Kővágó, Zoltán
` (27 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 UTC (permalink / raw)
To: qemu-devel; +Cc: Michael Walle, Gerd Hoffmann
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/milkymist-ac97.c | 8 ++++----
hw/audio/pcspk.c | 2 +-
hw/audio/sb16.c | 2 +-
hw/audio/wm8750.c | 4 ++--
20 files changed, 62 insertions(+), 79 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 67060f8..e2ad3ba 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -643,7 +643,7 @@ 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) {
@@ -706,7 +706,7 @@ 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);
@@ -924,7 +924,7 @@ 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 312b955..f469db8 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -506,7 +506,7 @@ 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;
@@ -526,14 +526,14 @@ 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);
@@ -587,7 +587,7 @@ 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;
@@ -635,7 +635,7 @@ 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;
}
}
@@ -698,7 +698,7 @@ 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);
@@ -710,7 +710,7 @@ 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;
}
@@ -990,7 +990,7 @@ 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;
@@ -1008,7 +1008,7 @@ 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);
}
@@ -1160,7 +1160,7 @@ 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 68545b6..d85138a 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -150,23 +150,6 @@ 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 46438a2..3df0127 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -420,7 +420,7 @@ 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 ff72d3f..f6e9d01 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -706,7 +706,7 @@ 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 4c94a26..da72e2d 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -49,11 +49,11 @@ 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, get_ticks_per_sec ());
- 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;
}
@@ -108,9 +108,9 @@ static int no_run_in (HWVoiceIn *hw)
muldiv64 (ticks, hw->info.bytes_per_second, get_ticks_per_sec ());
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;
}
@@ -121,7 +121,7 @@ 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 fbf35aa..7bc76c0 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -390,7 +390,7 @@ 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);
@@ -439,7 +439,7 @@ 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);
@@ -458,7 +458,7 @@ 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 c9b2fcd..4206dd0 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -233,7 +233,7 @@ static void *qpa_thread_out (void *arg)
}
}
- decr = to_mix = audio_MIN (pa->live, pa->samples >> 2);
+ decr = to_mix = MIN (pa->live, pa->samples >> 2);
rpos = pa->rpos;
if (audio_pt_unlock (&pa->pt, AUDIO_FUNC)) {
@@ -242,7 +242,7 @@ 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);
@@ -280,7 +280,7 @@ 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;
@@ -325,7 +325,7 @@ static void *qpa_thread_in (void *arg)
}
}
- incr = to_grab = audio_MIN (pa->dead, pa->samples >> 2);
+ incr = to_grab = MIN (pa->dead, pa->samples >> 2);
wpos = pa->wpos;
if (audio_pt_unlock (&pa->pt, AUDIO_FUNC)) {
@@ -334,7 +334,7 @@ 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,
@@ -373,7 +373,7 @@ 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 796238a..180455a 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -260,10 +260,10 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
}
/* 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); */
@@ -306,7 +306,7 @@ 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;
sdl->live = live - decr;
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index a9b9a1d..bc9284a 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -161,20 +161,20 @@ 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;
@@ -292,7 +292,7 @@ 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;
@@ -302,7 +302,7 @@ 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 b1a3b1d..caa9ad2 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -56,12 +56,12 @@ 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 a835cb6..6831524 100644
--- a/hw/audio/ac97.c
+++ b/hw/audio/ac97.c
@@ -958,7 +958,7 @@ 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;
@@ -967,7 +967,7 @@ 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",
@@ -1013,7 +1013,7 @@ 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)
@@ -1034,7 +1034,7 @@ 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;
@@ -1043,7 +1043,7 @@ 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 b5202f9..6c89a70 100644
--- a/hw/audio/adlib.c
+++ b/hw/audio/adlib.c
@@ -218,7 +218,7 @@ 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);
@@ -233,7 +233,7 @@ 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 43a1a39..9c730eb 100644
--- a/hw/audio/cs4231a.c
+++ b/hw/audio/cs4231a.c
@@ -526,7 +526,7 @@ 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);
}
@@ -569,7 +569,7 @@ 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 614ff19..f98a470 100644
--- a/hw/audio/es1370.c
+++ b/hw/audio/es1370.c
@@ -785,7 +785,7 @@ 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;
@@ -794,7 +794,7 @@ 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;
@@ -812,7 +812,7 @@ 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 e25a033..8fd326a 100644
--- a/hw/audio/gus.c
+++ b/hw/audio/gus.c
@@ -110,7 +110,7 @@ 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);
@@ -125,7 +125,7 @@ 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);
@@ -183,7 +183,7 @@ 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 = DMA_get_channel_mode (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/milkymist-ac97.c b/hw/audio/milkymist-ac97.c
index b63fd6f..a32b365 100644
--- a/hw/audio/milkymist-ac97.c
+++ b/hw/audio/milkymist-ac97.c
@@ -182,7 +182,7 @@ 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;
@@ -196,7 +196,7 @@ 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;
@@ -225,7 +225,7 @@ 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;
@@ -239,7 +239,7 @@ 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 85d6d00..131eff1 100644
--- a/hw/audio/pcspk.c
+++ b/hw/audio/pcspk.c
@@ -100,7 +100,7 @@ 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 92318c7..cc87c02 100644
--- a/hw/audio/sb16.c
+++ b/hw/audio/sb16.c
@@ -1147,7 +1147,7 @@ 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 049ddf9..f2bfd80 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);
}
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 24/51] audio: do not run each backend in audio_run
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (22 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 23/51] audio: remove audio_MIN, audio_MAX Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 25/51] paaudio: fix playback glitches Kővágó, Zoltán
` (26 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 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.
Reviewed-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/alsaaudio.c | 7 +++++--
audio/audio.c | 17 +++++++----------
audio/audio_int.h | 2 +-
audio/ossaudio.c | 12 ++++++------
4 files changed, 19 insertions(+), 19 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index e2ad3ba..dcc2a09 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -40,6 +40,7 @@ struct pollhlp {
struct pollfd *pfds;
int count;
int mask;
+ AudioState *s;
};
typedef struct ALSAVoiceOut {
@@ -200,11 +201,11 @@ 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:
@@ -758,6 +759,7 @@ 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;
@@ -859,6 +861,7 @@ 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 f469db8..6cafe87 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -790,8 +790,9 @@ static void audio_reset_timer (AudioState *s)
static void audio_timer (void *opaque)
{
- audio_run ("timer");
- audio_reset_timer (opaque);
+ AudioState *s = opaque;
+ audio_run(s, "timer");
+ audio_reset_timer(s);
}
/*
@@ -1194,15 +1195,11 @@ 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 1d81658..c52c656 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 7bc76c0..ae4c452 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -112,28 +112,28 @@ 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)
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 25/51] paaudio: fix playback glitches
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (23 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 24/51] audio: do not run each backend in audio_run Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 26/51] audio: remove read and write pcm_ops Kővágó, Zoltán
` (25 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 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 4206dd0..2d10afd 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -510,10 +510,8 @@ 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 */
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 26/51] audio: remove read and write pcm_ops
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (24 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 25/51] paaudio: fix playback glitches Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 27/51] audio: use size_t where makes sense Kővágó, Zoltán
` (24 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 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 | 14 ++++----------
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(+), 112 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index dcc2a09..a99bc7e 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -271,11 +271,6 @@ 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) {
@@ -997,11 +992,6 @@ 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;
@@ -1067,13 +1057,11 @@ 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 6cafe87..c54bfab 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -567,7 +567,7 @@ 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;
@@ -669,7 +669,7 @@ 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;
@@ -800,8 +800,6 @@ static void audio_timer (void *opaque)
*/
int AUD_write (SWVoiceOut *sw, void *buf, int size)
{
- int bytes;
-
if (!sw) {
/* XXX: Consider options */
return size;
@@ -812,14 +810,11 @@ int AUD_write (SWVoiceOut *sw, void *buf, int size)
return 0;
}
- bytes = sw->hw->pcm_ops->write (sw, buf, size);
- return bytes;
+ return audio_pcm_sw_write(sw, buf, size);
}
int AUD_read (SWVoiceIn *sw, void *buf, int size)
{
- int bytes;
-
if (!sw) {
/* XXX: Consider options */
return size;
@@ -830,8 +825,7 @@ int AUD_read (SWVoiceIn *sw, void *buf, int size)
return 0;
}
- bytes = sw->hw->pcm_ops->read (sw, buf, size);
- return bytes;
+ 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 c52c656..59ccf10 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -146,13 +146,11 @@ 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 @@ extern struct audio_driver *drvtab[];
void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as);
void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len);
-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 3df0127..6820f64 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -496,11 +496,6 @@ 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)
{
@@ -700,7 +695,6 @@ 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 f6e9d01..a80615a 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -453,11 +453,6 @@ 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;
@@ -644,11 +639,6 @@ 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;
@@ -855,13 +845,11 @@ 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 da72e2d..28c72d5 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -58,11 +58,6 @@ 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);
@@ -115,18 +110,6 @@ 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;
@@ -148,13 +131,11 @@ 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 ae4c452..e335b96 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -136,11 +136,6 @@ 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) {
@@ -792,11 +787,6 @@ 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;
@@ -848,13 +838,11 @@ 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 2d10afd..eded171 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -293,11 +293,6 @@ 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)
{
@@ -386,11 +381,6 @@ 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;
@@ -973,13 +963,11 @@ 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 180455a..470e0b9 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -284,11 +284,6 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
/* dolog ("done len=%d\n", len); */
}
-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;
@@ -434,7 +429,6 @@ 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 bc9284a..70bcd22 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -190,11 +190,6 @@ 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);
@@ -323,11 +318,6 @@ 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);
@@ -375,13 +365,11 @@ 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 caa9ad2..930b746 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -81,11 +81,6 @@ 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)
{
@@ -237,7 +232,6 @@ 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,
};
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 27/51] audio: use size_t where makes sense
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (25 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 26/51] audio: remove read and write pcm_ops Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 28/51] audio: api for mixeng code free backends Kővágó, Zoltán
` (23 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/alsaaudio.c | 8 +--
audio/audio.c | 163 ++++++++++++++++++++++++-------------------------
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 | 4 +-
audio/wavaudio.c | 4 +-
11 files changed, 125 insertions(+), 125 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index a99bc7e..94611e8 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -748,8 +748,8 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
alsa->pcm_buf = audio_calloc (AUDIO_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;
}
@@ -850,8 +850,8 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
alsa->pcm_buf = audio_calloc (AUDIO_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 c54bfab..8a167fa 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -499,10 +499,10 @@ 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) {
@@ -512,28 +512,28 @@ 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 (AUDIO_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(AUDIO_FUNC, live > hw->samples)) {
+ dolog("live=%zu 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);
@@ -547,14 +547,14 @@ 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 (AUDIO_FUNC, live < 0 || live > hw->samples)) {
- dolog ("live=%d hw->samples=%d\n", live, hw->samples);
+ dolog("live=%zd samples=%zu\n", live, hw->samples);
return 0;
}
@@ -567,17 +567,17 @@ 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 (AUDIO_FUNC, live < 0 || live > hw->samples)) {
- dolog ("live_in=%d hw->samples=%d\n", live, hw->samples);
+ if (audio_bug(AUDIO_FUNC, live > hw->samples)) {
+ dolog("live_in=%zu samples=%zu\n", live, hw->samples);
return 0;
}
@@ -591,9 +591,9 @@ 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;
}
@@ -602,11 +602,6 @@ static int audio_pcm_sw_read(SWVoiceIn *sw, void *buf, int size)
}
osamp = swlim;
- if (audio_bug (AUDIO_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;
@@ -627,10 +622,10 @@ 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) {
@@ -644,9 +639,9 @@ 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);
@@ -655,10 +650,10 @@ 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 (AUDIO_FUNC, live < 0 || live > hw->samples)) {
- dolog ("live=%d hw->samples=%d\n", live, hw->samples);
+ if (audio_bug(AUDIO_FUNC, live > hw->samples)) {
+ dolog("live=%zu hw->samples=%zu\n", live, hw->samples);
return 0;
}
return live;
@@ -669,10 +664,11 @@ 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;
+ int ret = 0;
+ size_t pos = 0, total = 0;
if (!sw) {
return size;
@@ -681,8 +677,8 @@ 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 (AUDIO_FUNC, live < 0 || live > hwsamples)){
- dolog ("live=%d hw->samples=%d\n", live, hwsamples);
+ if (audio_bug(AUDIO_FUNC, live > hwsamples)){
+ dolog("live=%zu samples=%zu\n", live, hwsamples);
return 0;
}
@@ -798,7 +794,7 @@ 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 */
@@ -813,7 +809,7 @@ 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 */
@@ -922,17 +918,17 @@ 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 (AUDIO_FUNC, live < 0 || live > sw->hw->samples)) {
- dolog ("live=%d sw->hw->samples=%d\n", live, sw->hw->samples);
+ if (audio_bug(AUDIO_FUNC, live > sw->hw->samples)) {
+ dolog("live=%zu samples=%zu\n", live, sw->hw->samples);
return 0;
}
@@ -945,9 +941,9 @@ 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;
@@ -955,8 +951,8 @@ static int audio_get_free (SWVoiceOut *sw)
live = sw->total_hw_samples_mixed;
- if (audio_bug (AUDIO_FUNC, live < 0 || live > sw->hw->samples)) {
- dolog ("live=%d sw->hw->samples=%d\n", live, sw->hw->samples);
+ if (audio_bug(AUDIO_FUNC, live > sw->hw->samples)) {
+ dolog("live=%zu samples=%zu\n", live, sw->hw->samples);
return 0;
}
@@ -971,9 +967,10 @@ 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;
@@ -984,17 +981,17 @@ 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;
@@ -1003,9 +1000,9 @@ 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)
@@ -1014,16 +1011,16 @@ 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 (AUDIO_FUNC, live < 0 || live > hw->samples)) {
- dolog ("live=%d hw->samples=%d\n", live, hw->samples);
+ if (audio_bug(AUDIO_FUNC, live > hw->samples)) {
+ dolog("live=%zu samples=%zu\n", live, hw->samples);
continue;
}
@@ -1056,14 +1053,14 @@ static void audio_run_out (AudioState *s)
prev_rpos = hw->rpos;
played = hw->pcm_ops->run_out (hw, live);
- if (audio_bug (AUDIO_FUNC, hw->rpos >= hw->samples)) {
- dolog ("hw->rpos=%d hw->samples=%d played=%d\n",
- hw->rpos, hw->samples, played);
+ if (audio_bug(AUDIO_FUNC, hw->rpos >= hw->samples)) {
+ dolog("rpos=%zu 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) {
@@ -1078,8 +1075,8 @@ static void audio_run_out (AudioState *s)
}
if (audio_bug (AUDIO_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;
}
@@ -1119,7 +1116,7 @@ static void audio_run_in (AudioState *s)
while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
SWVoiceIn *sw;
- int captured, min;
+ size_t captured, min;
captured = hw->pcm_ops->run_in (hw);
@@ -1131,7 +1128,7 @@ 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) {
@@ -1147,15 +1144,15 @@ 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;
@@ -1178,8 +1175,8 @@ static void audio_run_capture (AudioState *s)
}
if (audio_bug (AUDIO_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;
}
@@ -1511,8 +1508,8 @@ CaptureVoiceOut *AUD_add_capture(
hw->mix_buf = audio_calloc (AUDIO_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;
}
@@ -1521,7 +1518,7 @@ CaptureVoiceOut *AUD_add_capture(
cap->buf = audio_calloc (AUDIO_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 d85138a..a48411a 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -116,7 +116,7 @@ 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);
@@ -137,7 +137,7 @@ 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 59ccf10..a3bf79e 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -59,12 +59,12 @@ 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;
@@ -80,13 +80,13 @@ 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;
@@ -101,7 +101,7 @@ 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;
@@ -118,7 +118,7 @@ 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 @@ extern struct audio_driver *drvtab[];
void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as);
void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len);
-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 5a3dc90..5e28aea 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 (AUDIO_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 (AUDIO_FUNC, hw->samples <= 0)) {
- dolog ("hw->samples=%d\n", hw->samples);
+ dolog("hw->samples=%zu\n", hw->samples);
goto err1;
}
diff --git a/audio/mixeng.h b/audio/mixeng.h
index 9de443b..013d894 100644
--- a/audio/mixeng.h
+++ b/audio/mixeng.h
@@ -24,6 +24,8 @@
#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; };
@@ -32,6 +34,7 @@ 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);
@@ -40,10 +43,10 @@ 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 e335b96..8a73384 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -478,8 +478,8 @@ 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 {
@@ -545,8 +545,8 @@ 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;
@@ -570,8 +570,8 @@ 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);
}
}
}
@@ -585,7 +585,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
);
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
);
@@ -697,8 +697,8 @@ 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 (AUDIO_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 eded171..fb0ab94 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -583,8 +583,8 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
pa->pcm_buf = audio_calloc (AUDIO_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;
}
@@ -644,8 +644,8 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
pa->pcm_buf = audio_calloc (AUDIO_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 bd4b1c7..dc5e021 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 470e0b9..26538c8 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -250,8 +250,8 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
}
if (audio_bug (AUDIO_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 930b746..d49a118 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -136,8 +136,8 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
hw->samples = 1024;
wav->pcm_buf = audio_calloc (AUDIO_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;
}
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 28/51] audio: api for mixeng code free backends
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (26 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 27/51] audio: use size_t where makes sense Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 29/51] alsaaudio: port to the new audio backend api Kővágó, Zoltán
` (22 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio.c | 209 ++++++++++++++++++++++++++++++++++++++++++++++++-
audio/audio_int.h | 41 ++++++++--
audio/audio_template.h | 1 +
3 files changed, 241 insertions(+), 10 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index 8a167fa..0823170 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -544,6 +544,25 @@ 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)
*/
@@ -1005,6 +1024,31 @@ 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;
@@ -1052,7 +1096,11 @@ 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);
+ }
if (audio_bug(AUDIO_FUNC, hw->rpos >= hw->samples)) {
dolog("rpos=%zu samples=%zu played=%zu\n",
hw->rpos, hw->samples, played);
@@ -1110,6 +1158,35 @@ 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;
@@ -1118,7 +1195,12 @@ static void audio_run_in (AudioState *s)
SWVoiceIn *sw;
size_t captured, min;
- 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));
+ }
min = audio_pcm_hw_find_min_in (hw);
hw->total_samples_captured += captured - min;
@@ -1210,12 +1292,135 @@ 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 a3bf79e..469fe5e 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -63,6 +63,8 @@ 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;
@@ -85,6 +87,8 @@ 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;
@@ -143,17 +147,38 @@ 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 5e28aea..f157695 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;
}
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 29/51] alsaaudio: port to the new audio backend api
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (27 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 28/51] audio: api for mixeng code free backends Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 30/51] coreaudio: " Kővágó, Zoltán
` (21 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 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 94611e8..ba54383 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -45,9 +45,6 @@ struct pollhlp {
typedef struct ALSAVoiceOut {
HWVoiceOut hw;
- int wpos;
- int pending;
- void *pcm_buf;
snd_pcm_t *handle;
struct pollhlp pollhlp;
Audiodev *dev;
@@ -56,7 +53,6 @@ typedef struct ALSAVoiceOut {
typedef struct ALSAVoiceIn {
HWVoiceIn hw;
snd_pcm_t *handle;
- void *pcm_buf;
struct pollhlp pollhlp;
Audiodev *dev;
} ALSAVoiceIn;
@@ -611,102 +607,62 @@ 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)
@@ -715,9 +671,6 @@ 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,
@@ -746,14 +699,6 @@ 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 (AUDIO_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;
@@ -848,14 +793,6 @@ 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 (AUDIO_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;
@@ -867,129 +804,48 @@ 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, ...)
@@ -1056,12 +912,12 @@ 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,
};
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 30/51] coreaudio: port to the new audio backend api
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (28 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 29/51] alsaaudio: port to the new audio backend api Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 31/51] noaudio: " Kővágó, Zoltán
` (20 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/coreaudio.c | 130 ++++++++++---------
audio/dsound_template.h | 45 +++----
audio/dsoundaudio.c | 328 ++++++++++++------------------------------------
3 files changed, 171 insertions(+), 332 deletions(-)
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index 6820f64..8d1db98 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -45,9 +45,6 @@ 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
@@ -404,31 +401,29 @@ 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(
@@ -440,19 +435,11 @@ 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;
@@ -460,42 +447,51 @@ 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)
{
@@ -583,6 +579,16 @@ 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) {
@@ -694,7 +700,9 @@ 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
};
diff --git a/audio/dsound_template.h b/audio/dsound_template.h
index 96181ef..ff5a1f8 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 a80615a..364c398 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -52,19 +52,11 @@ 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;
@@ -242,11 +234,6 @@ 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)
{
@@ -311,33 +298,6 @@ 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)
{
@@ -349,7 +309,7 @@ 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,
@@ -453,138 +413,51 @@ 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 (AUDIO_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, ...)
@@ -639,96 +512,49 @@ 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)
@@ -844,12 +670,16 @@ 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
};
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 31/51] noaudio: port to the new audio backend api
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (29 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 30/51] coreaudio: " Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 32/51] ossaudio: " Kővágó, Zoltán
` (19 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 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 28c72d5..c7e47ae 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -38,10 +38,9 @@ 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;
@@ -49,13 +48,9 @@ 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, get_ticks_per_sec ());
- 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)
@@ -89,25 +84,21 @@ 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, get_ticks_per_sec ());
+ 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, get_ticks_per_sec ());
- 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, ...)
@@ -130,12 +121,12 @@ 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
};
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 32/51] ossaudio: port to the new audio backend api
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (30 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 31/51] noaudio: " Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 33/51] paaudio: " Kővágó, Zoltán
` (18 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/ossaudio.c | 289 ++++++++++++++++++++-----------------------------------
1 file changed, 104 insertions(+), 185 deletions(-)
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 8a73384..975adb5 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -42,19 +42,15 @@
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;
@@ -373,97 +369,87 @@ 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)
@@ -474,18 +460,13 @@ 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;
}
}
@@ -536,19 +517,20 @@ 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) {
@@ -568,32 +550,16 @@ 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 (
- AUDIO_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;
@@ -621,7 +587,7 @@ 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 (
@@ -695,13 +661,6 @@ 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 (AUDIO_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;
@@ -713,78 +672,36 @@ 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, ...)
@@ -837,12 +754,14 @@ 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
};
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 33/51] paaudio: port to the new audio backend api
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (31 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 32/51] ossaudio: " Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 34/51] sdlaudio: " Kővágó, Zoltán
` (17 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 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 | 173 ------------------------
audio/audio_pt_int.h | 22 ---
audio/paaudio.c | 372 +++++++--------------------------------------------
configure | 5 -
5 files changed, 45 insertions(+), 528 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 9d8f579..c7216a7 100644
--- a/audio/Makefile.objs
+++ b/audio/Makefile.objs
@@ -6,7 +6,6 @@ common-obj-$(CONFIG_COREAUDIO) += coreaudio.o
common-obj-$(CONFIG_ALSA) += alsaaudio.o
common-obj-$(CONFIG_DSOUND) += dsoundaudio.o
common-obj-$(CONFIG_PA) += paaudio.o
-common-obj-$(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 9a9c306..0000000
--- a/audio/audio_pt_int.c
+++ /dev/null
@@ -1,173 +0,0 @@
-#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, AUDIO_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, AUDIO_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, AUDIO_FUNC);
- }
-
- err1:
- err2 = pthread_mutex_destroy (&p->mutex);
- if (err2) {
- logerr (p, err2, "%s(%s): pthread_mutex_destroy failed", cap, AUDIO_FUNC);
- }
-
- err0:
- logerr (p, err, "%s(%s): %s failed", cap, AUDIO_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, AUDIO_FUNC);
- ret = -1;
- }
-
- err = pthread_mutex_destroy (&p->mutex);
- if (err) {
- logerr (p, err, "%s(%s): pthread_mutex_destroy failed", cap, AUDIO_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, AUDIO_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, AUDIO_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, AUDIO_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, AUDIO_FUNC);
- return -1;
- }
- err = pthread_cond_signal (&p->cond);
- if (err) {
- logerr (p, err, "%s(%s): pthread_cond_signal failed", cap, AUDIO_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, AUDIO_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 0dfff76..0000000
--- 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 /* audio_pt_int.h */
diff --git a/audio/paaudio.c b/audio/paaudio.c
index fb0ab94..556b0fd 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -7,7 +7,6 @@
#define AUDIO_CAP "pulseaudio"
#include "audio_int.h"
-#include "audio_pt_int.h"
typedef struct PAConnection {
char *server;
@@ -28,28 +27,16 @@ 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;
@@ -87,298 +74,96 @@ 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, AUDIO_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, AUDIO_FUNC)) {
- goto exit;
- }
- }
-
- decr = to_mix = MIN (pa->live, pa->samples >> 2);
- rpos = pa->rpos;
-
- if (audio_pt_unlock (&pa->pt, AUDIO_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, AUDIO_FUNC)) {
- return NULL;
- }
-
- pa->rpos = rpos;
- pa->live -= decr;
- pa->decr += decr;
- }
-
- exit:
- audio_pt_unlock (&pa->pt, AUDIO_FUNC);
- return NULL;
-}
-
-static int qpa_run_out (HWVoiceOut *hw, int live)
-{
- int decr;
- PAVoiceOut *pa = (PAVoiceOut *) hw;
-
- if (audio_pt_lock (&pa->pt, AUDIO_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, AUDIO_FUNC);
- }
- else {
- audio_pt_unlock (&pa->pt, AUDIO_FUNC);
- }
- return decr;
-}
-
-/* capture */
-static void *qpa_thread_in (void *arg)
-{
- PAVoiceIn *pa = arg;
- HWVoiceIn *hw = &pa->hw;
-
- if (audio_pt_lock (&pa->pt, AUDIO_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, AUDIO_FUNC)) {
- goto exit;
- }
- }
-
- incr = to_grab = MIN (pa->dead, pa->samples >> 2);
- wpos = pa->wpos;
-
- if (audio_pt_unlock (&pa->pt, AUDIO_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, AUDIO_FUNC)) {
- return NULL;
- }
-
- pa->wpos = wpos;
- pa->dead -= incr;
- pa->incr += incr;
- }
-
- exit:
- audio_pt_unlock (&pa->pt, AUDIO_FUNC);
- return NULL;
-}
-
-static int qpa_run_in (HWVoiceIn *hw)
-{
- int live, incr, dead;
- PAVoiceIn *pa = (PAVoiceIn *) hw;
-
- if (audio_pt_lock (&pa->pt, AUDIO_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, AUDIO_FUNC);
- }
- else {
- audio_pt_unlock (&pa->pt, AUDIO_FUNC);
- }
- return incr;
+ return 0;
}
static pa_sample_format_t audfmt_to_pa (AudioFormat afmt, int endianness)
@@ -466,13 +251,6 @@ 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,
@@ -495,8 +273,6 @@ 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
@@ -580,28 +356,9 @@ 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 (AUDIO_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, AUDIO_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;
}
@@ -641,28 +398,9 @@ 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 (AUDIO_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, AUDIO_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;
}
@@ -688,42 +426,22 @@ 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, AUDIO_FUNC);
- pa->done = 1;
- audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC);
- audio_pt_join (&pa->pt, &ret, AUDIO_FUNC);
-
if (pa->stream) {
qpa_simple_disconnect(pa->g->conn, pa->stream);
pa->stream = NULL;
}
-
- audio_pt_fini (&pa->pt, AUDIO_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, AUDIO_FUNC);
- pa->done = 1;
- audio_pt_unlock_and_signal (&pa->pt, AUDIO_FUNC);
- audio_pt_join (&pa->pt, &ret, AUDIO_FUNC);
-
if (pa->stream) {
qpa_simple_disconnect(pa->g->conn, pa->stream);
pa->stream = NULL;
}
-
- audio_pt_fini (&pa->pt, AUDIO_FUNC);
- g_free (pa->pcm_buf);
- pa->pcm_buf = NULL;
}
static int qpa_ctl_out (HWVoiceOut *hw, int cmd, ...)
@@ -962,12 +680,12 @@ 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 44ac9ab..1c05418 100755
--- a/configure
+++ b/configure
@@ -208,7 +208,6 @@ block_drv_ro_whitelist=""
host_cc="cc"
libs_softmmu=""
libs_tools=""
-audio_pt_int=""
audio_win_int=""
cc_i386=i386-pc-linux-gnu-gcc
libs_qga=""
@@ -2845,7 +2844,6 @@ for drv in $audio_drv_list; do
audio_drv_probe $drv pulse/mainloop.h "-lpulse" \
"pa_mainloop *m = 0; pa_mainloop_free (m); return 0;"
libs_softmmu="-lpulse $libs_softmmu"
- audio_pt_int="yes"
;;
coreaudio)
@@ -4980,9 +4978,6 @@ for drv in $audio_drv_list; do
def=CONFIG_`echo $drv | LC_ALL=C tr '[a-z]' '[A-Z]'`
echo "$def=y" >> $config_host_mak
done
-if test "$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
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 34/51] sdlaudio: port to the new audio backend api
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (32 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 33/51] paaudio: " Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 36/51] wavaudio: " Kővágó, Zoltán
` (16 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/sdlaudio.c | 102 ++++++++++++++++++++++---------------------------------
1 file changed, 41 insertions(+), 61 deletions(-)
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index 26538c8..183ecdd 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -39,9 +39,6 @@
typedef struct SDLVoiceOut {
HWVoiceOut hw;
- int live;
- int rpos;
- int decr;
} SDLVoiceOut;
static struct SDLAudioState {
@@ -230,16 +227,12 @@ 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); */
+ while (len) {
sdl_wait (s, "sdl_callback");
if (s->exit) {
return;
@@ -249,34 +242,23 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
return;
}
- if (audio_bug (AUDIO_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;
- }
+ 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 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;
+ write_len = MIN(MIN(hw->pending_emul, len),
+ hw->size_emul - start);
- /* dolog ("in callback to_mix %d, chunk %d\n", to_mix, chunk); */
- hw->clip (buf, src, chunk);
- sdl->rpos = (sdl->rpos + chunk) % hw->samples;
- to_mix -= chunk;
- buf += chunk << hw->info.shift;
+ 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;
- again:
if (sdl_unlock (s, "sdl_callback")) {
return;
}
@@ -284,37 +266,33 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
/* dolog ("done len=%d\n", len); */
}
-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);
+#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; \
}
- decr = MIN (sdl->decr, live);
- sdl->decr -= decr;
+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)
- sdl->live = live - decr;
- hw->rpos = sdl->rpos;
-
- if (sdl->live > 0) {
- sdl_unlock_and_post (s, "sdl_run_out");
- }
- else {
- sdl_unlock (s, "sdl_run_out");
- }
- return decr;
-}
+#undef SDL_WRAPPER_FUNC
static void sdl_fini_out (HWVoiceOut *hw)
{
@@ -428,7 +406,9 @@ 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,
};
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 36/51] wavaudio: port to the new audio backend api
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (33 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 34/51] sdlaudio: " Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 37/51] audio: remove remains of the old " Kővágó, Zoltán
` (15 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/wavaudio.c | 55 +++++++++----------------------------------------------
1 file changed, 9 insertions(+), 46 deletions(-)
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index d49a118..b289689 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -33,52 +33,27 @@ 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, get_ticks_per_sec ());
-
- 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. */
@@ -134,13 +109,6 @@ 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 (AUDIO_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);
@@ -150,8 +118,6 @@ 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;
}
@@ -205,9 +171,6 @@ 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, ...)
@@ -231,7 +194,7 @@ 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,
};
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 37/51] audio: remove remains of the old backend api
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (34 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 36/51] wavaudio: " Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 38/51] audio: unify input and output mixeng buffer management Kővágó, Zoltán
` (14 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 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 0823170..f7d8d30 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -512,7 +512,7 @@ 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(AUDIO_FUNC, live > hw->samples)) {
@@ -522,29 +522,7 @@ 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;
@@ -1033,7 +1011,7 @@ 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;
@@ -1096,11 +1074,7 @@ 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);
if (audio_bug(AUDIO_FUNC, hw->rpos >= hw->samples)) {
dolog("rpos=%zu samples=%zu played=%zu\n",
hw->rpos, hw->samples, played);
@@ -1195,12 +1169,8 @@ static void audio_run_in (AudioState *s)
SWVoiceIn *sw;
size_t captured, min;
- 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));
min = audio_pcm_hw_find_min_in (hw);
hw->total_samples_captured += captured - min;
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 469fe5e..487867c 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -149,7 +149,6 @@ 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
* returns the buffer, and writes it's size to size (in bytes)
@@ -163,7 +162,6 @@ 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);
@@ -232,11 +230,6 @@ extern struct audio_driver *drvtab[];
void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as);
void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len);
-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);
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 38/51] audio: unify input and output mixeng buffer management
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (35 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 37/51] audio: remove remains of the old " Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 39/51] audio: remove hw->samples, buffer_size_in/out pcm_ops Kővágó, Zoltán
` (13 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio.c | 123 +++++++++++++++++++++++--------------------------
audio/audio_int.h | 12 +++--
audio/audio_template.h | 18 +++-----
3 files changed, 71 insertions(+), 82 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index f7d8d30..987bc35 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -515,8 +515,8 @@ 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(AUDIO_FUNC, live > hw->samples)) {
- dolog("live=%zu samples=%zu\n", live, hw->samples);
+ if (audio_bug(AUDIO_FUNC, live > hw->conv_buf->size)) {
+ dolog("live=%zu samples=%zu\n", live, hw->conv_buf->size);
return 0;
}
return live;
@@ -525,17 +525,17 @@ 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;
}
@@ -550,17 +550,17 @@ 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 (AUDIO_FUNC, live < 0 || live > hw->samples)) {
- dolog("live=%zd samples=%zu\n", live, hw->samples);
+ if (audio_bug(AUDIO_FUNC, live < 0 || live > hw->conv_buf->size)) {
+ dolog("live=%zd samples=%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;
}
}
@@ -570,11 +570,11 @@ 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(AUDIO_FUNC, live > hw->samples)) {
- dolog("live_in=%zu samples=%zu\n", live, hw->samples);
+ if (audio_bug(AUDIO_FUNC, live > hw->conv_buf->size)) {
+ dolog("live_in=%zu samples=%zu\n", live, hw->conv_buf->size);
return 0;
}
@@ -587,11 +587,11 @@ 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) {
@@ -601,7 +601,7 @@ 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;
@@ -649,8 +649,8 @@ static size_t audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
if (nb_live1) {
size_t live = smin;
- if (audio_bug(AUDIO_FUNC, live > hw->samples)) {
- dolog("live=%zu hw->samples=%zu\n", live, hw->samples);
+ if (audio_bug(AUDIO_FUNC, live > hw->mix_buf->size)) {
+ dolog("live=%zu hw->samples=%zu\n", live, hw->mix_buf->size);
return 0;
}
return live;
@@ -671,7 +671,7 @@ 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(AUDIO_FUNC, live > hwsamples)){
@@ -686,7 +686,7 @@ 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;
@@ -712,7 +712,7 @@ 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
);
@@ -823,7 +823,7 @@ 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)
@@ -924,8 +924,8 @@ static size_t audio_get_avail (SWVoiceIn *sw)
}
live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
- if (audio_bug(AUDIO_FUNC, live > sw->hw->samples)) {
- dolog("live=%zu samples=%zu\n", live, sw->hw->samples);
+ if (audio_bug(AUDIO_FUNC, live > sw->hw->conv_buf->size)) {
+ dolog("live=%zu samples=%zu\n", live, sw->hw->conv_buf->size);
return 0;
}
@@ -948,12 +948,12 @@ static size_t audio_get_free(SWVoiceOut *sw)
live = sw->total_hw_samples_mixed;
- if (audio_bug(AUDIO_FUNC, live > sw->hw->samples)) {
- dolog("live=%zu samples=%zu\n", live, sw->hw->samples);
+ if (audio_bug(AUDIO_FUNC, live > sw->hw->mix_buf->size)) {
+ dolog("live=%zu samples=%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",
@@ -978,12 +978,12 @@ 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 "
@@ -992,14 +992,14 @@ 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)
@@ -1017,7 +1017,7 @@ 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;
@@ -1041,8 +1041,8 @@ static void audio_run_out (AudioState *s)
live = 0;
}
- if (audio_bug(AUDIO_FUNC, live > hw->samples)) {
- dolog("live=%zu samples=%zu\n", live, hw->samples);
+ if (audio_bug(AUDIO_FUNC, live > hw->mix_buf->size)) {
+ dolog("live=%zu samples=%zu\n", live, hw->mix_buf->size);
continue;
}
@@ -1073,12 +1073,12 @@ 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);
- if (audio_bug(AUDIO_FUNC, hw->rpos >= hw->samples)) {
+ if (audio_bug(AUDIO_FUNC, hw->mix_buf->pos >= hw->mix_buf->size)) {
dolog("rpos=%zu samples=%zu played=%zu\n",
- hw->rpos, hw->samples, played);
- hw->rpos = 0;
+ hw->mix_buf->pos, hw->mix_buf->size, played);
+ hw->mix_buf->pos = 0;
}
#ifdef DEBUG_OUT
@@ -1135,6 +1135,7 @@ 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;
@@ -1148,10 +1149,10 @@ 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;
@@ -1170,7 +1171,7 @@ static void audio_run_in (AudioState *s)
size_t captured, min;
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));
min = audio_pcm_hw_find_min_in (hw);
hw->total_samples_captured += captured - min;
@@ -1201,14 +1202,14 @@ 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);
@@ -1216,10 +1217,10 @@ 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) {
@@ -1267,7 +1268,7 @@ 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;
@@ -1303,7 +1304,7 @@ 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;
@@ -1678,23 +1679,16 @@ CaptureVoiceOut *AUD_add_capture(
QLIST_INIT (&hw->sw_head);
QLIST_INIT (&cap->cb_head);
- /* XXX find a more elegant way */
- hw->samples = 4096 * 4;
- hw->mix_buf = audio_calloc (AUDIO_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 (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
+ cap->buf = audio_calloc(AUDIO_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;
}
@@ -1715,7 +1709,6 @@ 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 487867c..031f699 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -50,6 +50,11 @@ 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;
@@ -58,11 +63,9 @@ 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;
@@ -82,11 +85,10 @@ 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 f157695..a37aea3 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -76,16 +76,12 @@ 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 (AUDIO_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;
- 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 +100,7 @@ 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 (AUDIO_FUNC, samples, sizeof (struct st_sample));
if (!sw->buf) {
@@ -280,9 +276,7 @@ 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;
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 39/51] audio: remove hw->samples, buffer_size_in/out pcm_ops
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (36 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 38/51] audio: unify input and output mixeng buffer management Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 40/51] audio: common rate control code for timer based outputs Kővágó, Zoltán
` (12 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 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_int.h | 5 +++--
audio/audio_template.h | 16 +++++++---------
audio/coreaudio.c | 10 +++++++++-
audio/dsound_template.h | 8 +++++++-
audio/dsoundaudio.c | 4 ++++
audio/noaudio.c | 2 --
audio/ossaudio.c | 24 ++++++++++++++++++++----
audio/paaudio.c | 21 +++++++++++++++++----
audio/sdlaudio.c | 10 +++++++++-
audio/spiceaudio.c | 14 ++++++++++++--
audio/wavaudio.c | 1 -
12 files changed, 106 insertions(+), 29 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index ba54383..60d0ca5 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -46,6 +46,7 @@ struct pollhlp {
typedef struct ALSAVoiceOut {
HWVoiceOut hw;
snd_pcm_t *handle;
+ size_t samples;
struct pollhlp pollhlp;
Audiodev *dev;
} ALSAVoiceOut;
@@ -53,6 +54,7 @@ typedef struct ALSAVoiceOut {
typedef struct ALSAVoiceIn {
HWVoiceIn hw;
snd_pcm_t *handle;
+ size_t samples;
struct pollhlp pollhlp;
Audiodev *dev;
} ALSAVoiceIn;
@@ -697,7 +699,7 @@ 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;
@@ -705,6 +707,12 @@ 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
@@ -791,7 +799,7 @@ 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;
@@ -799,6 +807,12 @@ 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;
@@ -912,11 +926,13 @@ 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_int.h b/audio/audio_int.h
index 031f699..1250012 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -69,7 +69,6 @@ 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;
@@ -92,7 +91,6 @@ 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;
@@ -152,6 +150,8 @@ 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)
* this is unrelated to the above buffer_size_out function */
@@ -165,6 +165,7 @@ 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 a37aea3..473eac9 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -78,7 +78,12 @@ 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->glue(buffer_size_, TYPE)) {
+ samples = hw->pcm_ops->glue(buffer_size_, TYPE)(hw);
+ } else {
+ samples = 1024; /* todo better default */
+ }
HWBUF = g_malloc0(sizeof(STSampleBuffer) + sizeof(st_sample)*samples);
HWBUF->size = samples;
@@ -261,11 +266,6 @@ static HW *glue(audio_pcm_hw_add_new_, TYPE)(AudioState *s,
goto err0;
}
- if (audio_bug (AUDIO_FUNC, hw->samples <= 0)) {
- dolog("hw->samples=%zu\n", hw->samples);
- goto err1;
- }
-
#ifdef DAC
hw->clip = mixeng_clip
#else
@@ -285,9 +285,7 @@ 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 8d1db98..21a8764 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -45,6 +45,7 @@ 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
@@ -564,7 +565,7 @@ 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 */
@@ -624,6 +625,12 @@ 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;
@@ -701,6 +708,7 @@ 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 ff5a1f8..6a10b67 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 364c398..b0370f0 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -53,12 +53,14 @@ 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)
@@ -671,6 +673,7 @@ 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,
@@ -678,6 +681,7 @@ 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 c7e47ae..6c73a48 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -56,7 +56,6 @@ 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;
}
@@ -75,7 +74,6 @@ 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 975adb5..e036a52 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -46,6 +46,7 @@ typedef struct OSSVoiceOut {
int nfrags;
int fragsize;
int mmapped;
+ size_t samples;
Audiodev *dev;
} OSSVoiceOut;
@@ -54,6 +55,7 @@ typedef struct OSSVoiceIn {
int fd;
int nfrags;
int fragsize;
+ size_t samples;
Audiodev *dev;
} OSSVoiceIn;
@@ -513,11 +515,11 @@ 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,
@@ -565,6 +567,12 @@ 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;
@@ -587,7 +595,7 @@ 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, oss->samples);
trig = PCM_ENABLE_OUTPUT;
if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
oss_logerr (
@@ -660,13 +668,19 @@ 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;
@@ -755,6 +769,7 @@ 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,
@@ -762,6 +777,7 @@ 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 556b0fd..59eca76 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -354,8 +354,7 @@ 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;
@@ -363,6 +362,13 @@ 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;
@@ -396,8 +402,7 @@ 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;
@@ -405,6 +410,12 @@ 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;
@@ -681,11 +692,13 @@ 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 183ecdd..7b32332 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -39,6 +39,7 @@
typedef struct SDLVoiceOut {
HWVoiceOut hw;
+ size_t samples;
} SDLVoiceOut;
static struct SDLAudioState {
@@ -335,7 +336,7 @@ 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;
@@ -343,6 +344,12 @@ 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;
@@ -407,6 +414,7 @@ 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 d508e37..97ab0b3 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -131,7 +131,6 @@ 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;
@@ -142,6 +141,11 @@ 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);
@@ -246,7 +250,6 @@ 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;
@@ -257,6 +260,11 @@ 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);
@@ -327,6 +335,7 @@ 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,
@@ -334,6 +343,7 @@ 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 b289689..823003d 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -108,7 +108,6 @@ 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);
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 40/51] audio: common rate control code for timer based outputs
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (37 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 39/51] audio: remove hw->samples, buffer_size_in/out pcm_ops Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 41/51] audio: split ctl_* functions into enable_* and volume_* Kővágó, Zoltán
` (11 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 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 | 20 ++++++++++----------
5 files changed, 78 insertions(+), 79 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index 987bc35..d79ec38 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -1958,3 +1958,33 @@ 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, get_ticks_per_sec());
+ 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 1250012..16ffa12 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -238,6 +238,15 @@ 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 6c73a48..3cb3a89 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -30,32 +30,26 @@
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, get_ticks_per_sec ());
-
- 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;
}
@@ -66,14 +60,20 @@ 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;
}
@@ -84,25 +84,20 @@ 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, get_ticks_per_sec ());
-
- 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 97ab0b3..c8759fc 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -38,15 +38,10 @@
#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;
@@ -56,7 +51,7 @@ typedef struct SpiceVoiceOut {
typedef struct SpiceVoiceIn {
HWVoiceIn hw;
SpiceRecordInstance sin;
- SpiceRateCtl rate;
+ RateCtl rate;
int active;
} SpiceVoiceIn;
@@ -87,32 +82,6 @@ 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, get_ticks_per_sec ());
- 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,
@@ -156,17 +125,14 @@ 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;
}
@@ -195,7 +161,7 @@ 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:
@@ -275,8 +241,7 @@ 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? */
@@ -298,7 +263,7 @@ 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 823003d..d407a07 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -32,20 +32,15 @@
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, get_ticks_per_sec ());
- 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",
@@ -125,6 +120,8 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
strerror(errno));
return -1;
}
+
+ audio_rate_start(&wav->rate);
return 0;
}
@@ -174,8 +171,11 @@ 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;
}
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 41/51] audio: split ctl_* functions into enable_* and volume_*
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (38 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 40/51] audio: common rate control code for timer based outputs Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 42/51] audio: add mixeng option (documentation) Kővágó, Zoltán
` (10 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 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 | 128 ++++++++++++++++++++-----------------------------
audio/sdlaudio.c | 17 ++-----
audio/spiceaudio.c | 101 ++++++++++++++++----------------------
audio/wavaudio.c | 7 ++-
12 files changed, 214 insertions(+), 311 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 60d0ca5..ad48e80 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -746,34 +746,28 @@ 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)
@@ -862,35 +856,29 @@ 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)
@@ -928,13 +916,13 @@ 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,
};
struct audio_driver alsa_audio_driver = {
diff --git a/audio/audio.c b/audio/audio.c
index d79ec38..7c59748 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -607,7 +607,7 @@ 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);
}
@@ -695,7 +695,7 @@ 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);
}
}
@@ -845,7 +845,9 @@ 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);
}
}
@@ -890,7 +892,9 @@ 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);
}
}
@@ -907,7 +911,9 @@ 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);
+ }
}
}
}
@@ -1053,7 +1059,9 @@ 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);
@@ -1409,15 +1417,18 @@ 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);
}
@@ -1430,8 +1441,8 @@ static void free_audio_state(AudioState *s)
while ((hwo = audio_pcm_hw_find_any_out(s, hwo))) {
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);
@@ -1446,8 +1457,8 @@ static void free_audio_state(AudioState *s)
}
while ((hwi = audio_pcm_hw_find_any_in(s, hwi))) {
- 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);
}
@@ -1763,8 +1774,8 @@ 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);
}
}
}
@@ -1778,8 +1789,8 @@ 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 16ffa12..6f2f309 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -71,7 +71,6 @@ 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;
@@ -92,7 +91,6 @@ 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;
@@ -143,7 +141,6 @@ struct audio_driver {
int max_voices_in;
int voice_size_out;
int voice_size_in;
- int ctl_caps;
};
struct audio_pcm_ops {
@@ -160,7 +157,8 @@ struct audio_pcm_ops {
* 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, ...);
+ 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);
@@ -168,7 +166,8 @@ 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);
@@ -247,12 +246,6 @@ 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 473eac9..762efea 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -256,7 +256,6 @@ 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 21a8764..3d1c355 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -662,13 +662,12 @@ 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);
@@ -676,9 +675,7 @@ 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 (!isAtexit) {
if (isPlaying(core->outputDeviceID)) {
@@ -689,9 +686,7 @@ static int coreaudio_ctl_out (HWVoiceOut *hw, int cmd, ...)
}
}
}
- break;
}
- return 0;
}
static void *coreaudio_audio_init(Audiodev *dev)
@@ -711,7 +706,7 @@ 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
};
struct audio_driver coreaudio_audio_driver = {
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index b0370f0..ea26313 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -362,7 +362,7 @@ 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;
@@ -372,18 +372,17 @@ 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);
@@ -391,28 +390,24 @@ 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)
@@ -462,7 +457,7 @@ 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;
@@ -471,18 +466,17 @@ 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 ?? */
@@ -490,28 +484,24 @@ 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)
@@ -676,7 +666,7 @@ 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,
@@ -684,7 +674,7 @@ 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,
};
struct audio_driver dsound_audio_driver = {
diff --git a/audio/noaudio.c b/audio/noaudio.c
index 3cb3a89..0618b60 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -58,14 +58,13 @@ 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)
@@ -91,14 +90,13 @@ 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)
@@ -115,12 +113,12 @@ 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
};
struct audio_driver no_audio_driver = {
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index e036a52..003b9ac 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -573,59 +573,50 @@ 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, oss->samples);
- 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, oss->samples);
+ 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)
@@ -718,32 +709,25 @@ 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)
@@ -772,13 +756,13 @@ 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
};
struct audio_driver oss_audio_driver = {
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 59eca76..b162c21 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -455,7 +455,7 @@ 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;
@@ -466,48 +466,36 @@ 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;
@@ -518,47 +506,34 @@ 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);
-
- /* FIXME: use the upcoming "set_source_output_{volume,mute}" */
- op = pa_context_set_source_volume_by_index(c->context,
- pa_stream_get_device_index (pa->stream),
- &v, NULL, NULL);
- if (!op) {
- qpa_logerr (pa_context_errno(c->context),
- "set_source_volume() failed\n");
- } else {
- pa_operation_unref(op);
- }
-
- op = pa_context_set_source_mute_by_index(c->context,
- pa_stream_get_index (pa->stream),
- sw->vol.mute, NULL, NULL);
- if (!op) {
- qpa_logerr (pa_context_errno(c->context),
- "set_source_mute() failed\n");
- } else {
- pa_operation_unref (op);
- }
+ /* FIXME: use the upcoming "set_source_output_{volume,mute}" */
+ op = pa_context_set_source_volume_by_index(c->context,
+ pa_stream_get_device_index(pa->stream),
+ &v, NULL, NULL);
+ if (!op) {
+ qpa_logerr(pa_context_errno(c->context),
+ "set_source_volume() failed\n");
+ } else {
+ pa_operation_unref(op);
+ }
- pa_threaded_mainloop_unlock(c->mainloop);
- }
+ op = pa_context_set_source_mute_by_index(c->context,
+ pa_stream_get_index(pa->stream),
+ vol->mute, NULL, NULL);
+ if (!op) {
+ qpa_logerr(pa_context_errno(c->context),
+ "set_source_mute() failed\n");
+ } else {
+ pa_operation_unref(op);
}
- return 0;
+
+ pa_threaded_mainloop_unlock(c->mainloop);
}
/* common */
@@ -693,13 +668,13 @@ 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
};
struct audio_driver pa_audio_driver = {
@@ -713,5 +688,4 @@ 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
};
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index 7b32332..83e38fe 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -350,20 +350,9 @@ 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)
@@ -417,7 +406,7 @@ 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,
};
struct audio_driver sdl_audio_driver = {
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index c8759fc..48ca4e6 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -151,22 +151,20 @@ 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) {
@@ -175,29 +173,21 @@ 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 */
@@ -253,48 +243,38 @@ 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,
@@ -303,13 +283,19 @@ 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
};
struct audio_driver spice_audio_driver = {
@@ -322,9 +308,6 @@ 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 d407a07..b00cc68 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -169,14 +169,13 @@ 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)
@@ -194,7 +193,7 @@ 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,
};
struct audio_driver wav_audio_driver = {
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 42/51] audio: add mixeng option (documentation)
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (39 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 41/51] audio: split ctl_* functions into enable_* and volume_* Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 43/51] audio: make mixeng optional Kővágó, Zoltán
` (9 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann, 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 da0c2df..9bd61de 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 184ffd4..a995fca 100644
--- a/qemu-options.hx
+++ b/qemu-options.hx
@@ -336,6 +336,7 @@ 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"
@@ -404,6 +405,11 @@ 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
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 43/51] audio: make mixeng optional
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (40 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 42/51] audio: add mixeng option (documentation) Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 44/51] paaudio: get/put_buffer functions Kővágó, Zoltán
` (8 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 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 | 22 +++++++++++-----
2 files changed, 79 insertions(+), 13 deletions(-)
diff --git a/audio/audio.c b/audio/audio.c
index 7c59748..ea3219b 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -793,32 +793,46 @@ 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)
@@ -1038,6 +1052,26 @@ 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;
@@ -1174,6 +1208,17 @@ 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, min;
@@ -1654,6 +1699,11 @@ CaptureVoiceOut *AUD_add_capture(
s = QTAILQ_FIRST(&audio_states);
}
+ 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);
@@ -1811,9 +1861,13 @@ 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)) {
@@ -1821,6 +1875,10 @@ 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;
@@ -1832,7 +1890,7 @@ 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 762efea..f3cbc66 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -78,15 +78,19 @@ 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->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->glue(buffer_size_, TYPE)) {
+ samples = hw->pcm_ops->glue(buffer_size_, TYPE)(hw);
+ } else {
+ samples = 1024; /* todo better default */
+ }
+
+ HWBUF = g_malloc0(sizeof(STSampleBuffer) + sizeof(st_sample)*samples);
+ HWBUF->size = samples;
} else {
- samples = 1024; /* todo better default */
+ HWBUF = NULL;
}
-
- HWBUF = g_malloc0(sizeof(STSampleBuffer) + sizeof(st_sample)*samples);
- HWBUF->size = samples;
}
static void glue (audio_pcm_sw_free_resources_, TYPE) (SW *sw)
@@ -105,6 +109,10 @@ 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 (AUDIO_FUNC, samples, sizeof (struct st_sample));
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 44/51] paaudio: get/put_buffer functions
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (41 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 43/51] audio: make mixeng optional Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 45/51] audio: support more than two channels in volume setting Kővágó, Zoltán
` (7 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
This lets us avoid some buffer copying when using mixeng.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/paaudio.c | 83 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++
1 file changed, 83 insertions(+)
diff --git a/audio/paaudio.c b/audio/paaudio.c
index b162c21..4b11f21 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -96,6 +96,59 @@ static inline int PA_STREAM_IS_GOOD(pa_stream_state_t x)
} \
} while (0);
+static void *qpa_get_buffer_in(HWVoiceIn *hw, size_t *size)
+{
+ PAVoiceIn *p = (PAVoiceIn *) hw;
+ PAConnection *c = p->g->conn;
+ int r;
+
+ pa_threaded_mainloop_lock(c->mainloop);
+
+ CHECK_DEAD_GOTO(c, p->stream, unlock_and_fail,
+ "pa_threaded_mainloop_lock failed\n");
+
+ 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");
+ }
+
+ *size = MIN(p->read_length, *size);
+
+ pa_threaded_mainloop_unlock(c->mainloop);
+ return (void *) p->read_data;
+
+unlock_and_fail:
+ pa_threaded_mainloop_unlock(c->mainloop);
+ *size = 0;
+ return NULL;
+}
+
+static void qpa_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size)
+{
+ PAVoiceIn *p = (PAVoiceIn *) hw;
+ PAConnection *c = p->g->conn;
+ int r;
+
+ pa_threaded_mainloop_lock(c->mainloop);
+
+ CHECK_DEAD_GOTO(c, p->stream, unlock,
+ "pa_threaded_mainloop_lock failed\n");
+
+ assert(buf == p->read_data && size <= p->read_length);
+
+ p->read_data += size;
+ p->read_length -= size;
+
+ if (size && !p->read_length) {
+ r = pa_stream_drop(p->stream);
+ CHECK_SUCCESS_GOTO(c, r == 0, unlock, "pa_stream_drop failed\n");
+ }
+
+unlock:
+ pa_threaded_mainloop_unlock(c->mainloop);
+}
+
static size_t qpa_read(HWVoiceIn *hw, void *data, size_t length)
{
PAVoiceIn *p = (PAVoiceIn *) hw;
@@ -134,6 +187,32 @@ unlock_and_fail:
return 0;
}
+static void *qpa_get_buffer_out(HWVoiceOut *hw, size_t *size)
+{
+ PAVoiceOut *p = (PAVoiceOut *) hw;
+ PAConnection *c = p->g->conn;
+ void *ret;
+ int r;
+
+ pa_threaded_mainloop_lock(c->mainloop);
+
+ CHECK_DEAD_GOTO(c, p->stream, unlock_and_fail,
+ "pa_threaded_mainloop_lock failed\n");
+
+ *size = -1;
+ r = pa_stream_begin_write(p->stream, &ret, size);
+ CHECK_SUCCESS_GOTO(c, r >= 0, unlock_and_fail,
+ "pa_stream_begin_write failed\n");
+
+ pa_threaded_mainloop_unlock(c->mainloop);
+ return ret;
+
+unlock_and_fail:
+ pa_threaded_mainloop_unlock(c->mainloop);
+ *size = 0;
+ return NULL;
+}
+
static size_t qpa_write(HWVoiceOut *hw, void *data, size_t length)
{
PAVoiceOut *p = (PAVoiceOut *) hw;
@@ -668,12 +747,16 @@ static struct audio_pcm_ops qpa_pcm_ops = {
.fini_out = qpa_fini_out,
.write = qpa_write,
.buffer_size_out = qpa_buffer_size_out,
+ .get_buffer_out = qpa_get_buffer_out,
+ .put_buffer_out = qpa_write, /* pa handles it */
.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,
+ .get_buffer_in = qpa_get_buffer_in,
+ .put_buffer_in = qpa_put_buffer_in,
.volume_in = qpa_volume_in
};
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 45/51] audio: support more than two channels in volume setting
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (42 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 44/51] paaudio: get/put_buffer functions Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 46/51] audio: replace shift in audio_pcm_info with bytes_per_frame Kővágó, Zoltán
` (6 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 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 ea3219b..7276187 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -1817,30 +1817,42 @@ 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 a48411a..06f15e3 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -127,6 +127,16 @@ 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 6f2f309..8b53c52 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -158,7 +158,7 @@ struct audio_pcm_ops {
* size may be smaller */
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);
@@ -167,7 +167,7 @@ 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 4b11f21..7d5e697 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -534,20 +534,22 @@ 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);
@@ -574,20 +576,22 @@ 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 48ca4e6..833e76f 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -177,13 +177,14 @@ 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);
}
@@ -264,13 +265,14 @@ 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);
}
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 46/51] audio: replace shift in audio_pcm_info with bytes_per_frame
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (43 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 45/51] audio: support more than two channels in volume setting Kővágó, Zoltán
@ 2016-01-14 13:45 ` Kővágó, Zoltán
2016-01-14 13:46 ` [Qemu-devel] [PATCH 47/51] audio: basic support for multichannel audio Kővágó, Zoltán
` (5 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:45 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 ad48e80..5c66f79 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -613,7 +613,7 @@ 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);
@@ -657,7 +657,7 @@ 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;
}
@@ -823,7 +823,7 @@ 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) {
@@ -849,8 +849,8 @@ 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 7276187..1c9bd63 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -267,26 +267,27 @@ 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:
@@ -297,9 +298,8 @@ 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);
}
@@ -310,26 +310,25 @@ 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;
}
}
@@ -339,14 +338,13 @@ 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;
}
}
@@ -529,7 +527,7 @@ 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);
@@ -578,7 +576,7 @@ 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;
}
@@ -613,7 +611,7 @@ 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;
}
/*
@@ -687,7 +685,7 @@ 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;
@@ -731,13 +729,13 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
dolog (
"%s: write size %d ret %d total sw %d\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
@@ -837,7 +835,7 @@ 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)
@@ -952,10 +950,10 @@ 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)
@@ -978,10 +976,11 @@ 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,
@@ -1000,7 +999,7 @@ 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;
@@ -1030,10 +1029,11 @@ 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;
@@ -1181,16 +1181,16 @@ 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);
@@ -1198,7 +1198,7 @@ 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;
@@ -1268,7 +1268,7 @@ 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;
@@ -1321,7 +1321,7 @@ 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;
@@ -1357,7 +1357,7 @@ 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;
@@ -1745,11 +1745,11 @@ CaptureVoiceOut *AUD_add_capture(
audio_pcm_init_info (&hw->info, as);
cap->buf = audio_calloc(AUDIO_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;
}
@@ -2058,14 +2058,14 @@ 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, get_ticks_per_sec());
- 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 8b53c52..0c0289d 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -42,8 +42,7 @@ 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 3d1c355..9ac904d 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -448,7 +448,7 @@ 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) {
@@ -457,7 +457,7 @@ 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 6a10b67..31d356d 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 ea26313..91b1449 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -321,8 +321,8 @@ 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 0618b60..32fbf64 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -86,7 +86,7 @@ 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 003b9ac..79bace0 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -510,16 +510,16 @@ 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,
@@ -654,12 +654,12 @@ 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 833e76f..f42d822 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -131,8 +131,9 @@ 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 b00cc68..96676a4 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -40,14 +40,14 @@ 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;
}
@@ -130,7 +130,7 @@ 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) {
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 47/51] audio: basic support for multichannel audio
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (44 preceding siblings ...)
2016-01-14 13:45 ` [Qemu-devel] [PATCH 46/51] audio: replace shift in audio_pcm_info with bytes_per_frame Kővágó, Zoltán
@ 2016-01-14 13:46 ` Kővágó, Zoltán
2016-01-14 13:46 ` [Qemu-devel] [PATCH 48/51] paaudio: channel-map option Kővágó, Zoltán
` (4 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:46 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 5c66f79..b3cbe6b 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -499,13 +499,6 @@ 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 1c9bd63..f276dc9 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -210,7 +210,7 @@ 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) {
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 48/51] paaudio: channel-map option
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (45 preceding siblings ...)
2016-01-14 13:46 ` [Qemu-devel] [PATCH 47/51] audio: basic support for multichannel audio Kővágó, Zoltán
@ 2016-01-14 13:46 ` Kővágó, Zoltán
2016-01-14 13:46 ` [Qemu-devel] [PATCH 49/51] usb-audio: do not count on avail bytes actually available Kővágó, Zoltán
` (3 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:46 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann, 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 7d5e697..bb9d928 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -336,17 +336,27 @@ 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;
}
@@ -423,7 +433,7 @@ 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
);
@@ -471,7 +481,7 @@ 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 9bd61de..e91de45 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: 2.6
##
{ 'struct': 'AudiodevPaPerDirectionOptions',
'data': {
- '*name': 'str' } }
+ '*name': 'str',
+ '*channel-map': 'str' } }
##
# @AudiodevPaOptions
diff --git a/qemu-options.hx b/qemu-options.hx
index a995fca..33ee3f2 100644
--- a/qemu-options.hx
+++ b/qemu-options.hx
@@ -372,6 +372,7 @@ 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"
@@ -525,6 +526,14 @@ 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}][,...]]
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 49/51] usb-audio: do not count on avail bytes actually available
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (46 preceding siblings ...)
2016-01-14 13:46 ` [Qemu-devel] [PATCH 48/51] paaudio: channel-map option Kővágó, Zoltán
@ 2016-01-14 13:46 ` Kővágó, Zoltán
2016-01-14 13:46 ` [Qemu-devel] [PATCH 51/51] usbaudio: change playback counters to 64 bit Kővágó, Zoltán
` (2 subsequent siblings)
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:46 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 deb9032..8db4a57 100644
--- a/hw/usb/dev-audio.c
+++ b/hw/usb/dev-audio.c
@@ -317,27 +317,28 @@ 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;
}
@@ -369,16 +370,21 @@ 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;
+ }
}
}
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* [Qemu-devel] [PATCH 51/51] usbaudio: change playback counters to 64 bit
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (47 preceding siblings ...)
2016-01-14 13:46 ` [Qemu-devel] [PATCH 49/51] usb-audio: do not count on avail bytes actually available Kővágó, Zoltán
@ 2016-01-14 13:46 ` Kővágó, Zoltán
2016-01-14 14:34 ` [Qemu-devel] [PATCH 00/51] audio 5.1 patches Victor Toso
2016-01-15 13:51 ` Gerd Hoffmann
50 siblings, 0 replies; 56+ messages in thread
From: Kővágó, Zoltán @ 2016-01-14 13:46 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 some 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 7f28996..63f3c4e 100644
--- a/hw/usb/dev-audio.c
+++ b/hw/usb/dev-audio.c
@@ -576,9 +576,9 @@ 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,
@@ -599,12 +599,14 @@ 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);
@@ -613,7 +615,7 @@ 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) {
--
2.7.0
^ permalink raw reply related [flat|nested] 56+ messages in thread
* Re: [Qemu-devel] [PATCH 00/51] audio 5.1 patches
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (48 preceding siblings ...)
2016-01-14 13:46 ` [Qemu-devel] [PATCH 51/51] usbaudio: change playback counters to 64 bit Kővágó, Zoltán
@ 2016-01-14 14:34 ` Victor Toso
2016-01-14 15:30 ` Kővágó Zoltán
2016-01-15 13:51 ` Gerd Hoffmann
50 siblings, 1 reply; 56+ messages in thread
From: Victor Toso @ 2016-01-14 14:34 UTC (permalink / raw)
To: Kővágó, Zoltán
Cc: qemu-devel, Eduardo Habkost, Gerd Hoffmann
Hi,
On Thu, Jan 14, 2016 at 02:45:13PM +0100, Kővágó, Zoltán wrote:
> Hi,
>
> Sorry guys, I've disappeared for a while... But anyways, I've updated
> my 5.1 audio patches to git master.
>
> About qapi flattening. That didn't really work last time, so now I've
> modified OptsVisitor to support both the old way of operation (ignoring
> structs) and the new way (requiring fully qualified paths for members
> inside structs). Not sure if qapi people would like the idea though...
>
> Otherwise I just updated it to work with master. Haven't really tested
> on any OS beside Linux.
>
> Please review.
Patches 35 and 50 are missing
>
>
> Kővágó, Zoltán (51):
> qapi: support implicit structs in OptsVisitor
> qapi: support nested structs in OptsVisitor
> qapi: qapi for audio backends
> audio: use qapi AudioFormat instead of audfmt_e
> audio: -audiodev command line option: documentation
> audio: -audiodev command line option basic implementation
> alsaaudio: port to -audiodev config
> coreaudio: port to -audiodev config
> dsoundaudio: port to -audiodev config
> noaudio: port to -audiodev config
> ossaudio: port to -audiodev config
> paaudio: port to -audiodev config
> sdlaudio: port to -audiodev config
> spiceaudio: port to -audiodev config
> wavaudio: port to -audiodev config
> audio: -audiodev command line option: cleanup
> audio: reduce glob_audio_state usage
> audio: basic support for multi backend audio
> audio: add audiodev properties to frontends
> audio: audiodev= parameters no longer optional when -audiodev present
> paaudio: do not move stream when sink/source name is specified
> paaudio: properly disconnect streams in fini_*
> audio: remove audio_MIN, audio_MAX
> audio: do not run each backend in audio_run
> paaudio: fix playback glitches
> audio: remove read and write pcm_ops
> audio: use size_t where makes sense
> audio: api for mixeng code free backends
> alsaaudio: port to the new audio backend api
> coreaudio: port to the new audio backend api
> noaudio: port to the new audio backend api
> ossaudio: port to the new audio backend api
> paaudio: port to the new audio backend api
> sdlaudio: port to the new audio backend api
> spiceaudio: port to the new audio backend api
this one ^
> wavaudio: port to the new audio backend api
> audio: remove remains of the old backend api
> audio: unify input and output mixeng buffer management
> audio: remove hw->samples, buffer_size_in/out pcm_ops
> audio: common rate control code for timer based outputs
> audio: split ctl_* functions into enable_* and volume_*
> audio: add mixeng option (documentation)
> audio: make mixeng optional
> paaudio: get/put_buffer functions
> audio: support more than two channels in volume setting
> audio: replace shift in audio_pcm_info with bytes_per_frame
> audio: basic support for multichannel audio
> paaudio: channel-map option
> usb-audio: do not count on avail bytes actually available
> usb-audio: support more than two channels of audio
this one ^
toso
> usbaudio: change playback counters to 64 bit
>
> Makefile | 6 +-
> audio/Makefile.objs | 3 +-
> audio/alsaaudio.c | 751 +++++----------
> audio/audio.c | 1518 ++++++++++++++++---------------
> audio/audio.h | 76 +-
> audio/audio_int.h | 124 +--
> audio/audio_legacy.c | 333 +++++++
> audio/audio_pt_int.c | 173 ----
> audio/audio_pt_int.h | 22 -
> audio/audio_template.h | 99 +-
> audio/audio_win_int.c | 18 +-
> audio/coreaudio.c | 198 ++--
> audio/dsound_template.h | 63 +-
> audio/dsoundaudio.c | 449 +++------
> audio/mixeng.h | 11 +-
> audio/noaudio.c | 98 +-
> audio/ossaudio.c | 596 +++++-------
> audio/paaudio.c | 881 ++++++++----------
> audio/rate_template.h | 2 +-
> audio/sdlaudio.c | 183 ++--
> audio/spiceaudio.c | 286 ++----
> audio/wavaudio.c | 158 +---
> audio/wavcapture.c | 8 +-
> configure | 5 -
> hmp-commands.hx | 11 +-
> hmp.c | 2 +-
> hw/acpi/core.c | 2 +-
> hw/arm/omap2.c | 2 +-
> hw/audio/ac97.c | 13 +-
> hw/audio/adlib.c | 7 +-
> hw/audio/cs4231a.c | 11 +-
> hw/audio/es1370.c | 17 +-
> hw/audio/gus.c | 9 +-
> hw/audio/hda-codec.c | 19 +-
> hw/audio/lm4549.c | 6 +-
> hw/audio/milkymist-ac97.c | 16 +-
> hw/audio/pcspk.c | 5 +-
> hw/audio/pl041.c | 1 +
> hw/audio/sb16.c | 17 +-
> hw/audio/wm8750.c | 14 +-
> hw/core/qdev-properties-system.c | 57 ++
> hw/input/tsc210x.c | 2 +-
> hw/usb/dev-audio.c | 460 ++++++++--
> include/hw/qdev-properties.h | 3 +
> include/qapi/opts-visitor.h | 2 +-
> monitor.c | 12 +-
> net/net.c | 2 +-
> numa.c | 2 +-
> qapi-schema.json | 3 +
> qapi/audio.json | 271 ++++++
> qapi/opts-visitor.c | 147 ++-
> qemu-options.hx | 242 ++++-
> tests/qapi-schema/qapi-schema-test.json | 9 +-
> tests/qapi-schema/qapi-schema-test.out | 4 +
> tests/test-opts-visitor.c | 36 +-
> ui/vnc.c | 29 +-
> ui/vnc.h | 2 +
> vl.c | 13 +-
> 58 files changed, 3835 insertions(+), 3674 deletions(-)
> create mode 100644 audio/audio_legacy.c
> delete mode 100644 audio/audio_pt_int.c
> delete mode 100644 audio/audio_pt_int.h
> create mode 100644 qapi/audio.json
>
> --
> 2.7.0
>
>
^ permalink raw reply [flat|nested] 56+ messages in thread
* Re: [Qemu-devel] [PATCH 00/51] audio 5.1 patches
2016-01-14 14:34 ` [Qemu-devel] [PATCH 00/51] audio 5.1 patches Victor Toso
@ 2016-01-14 15:30 ` Kővágó Zoltán
0 siblings, 0 replies; 56+ messages in thread
From: Kővágó Zoltán @ 2016-01-14 15:30 UTC (permalink / raw)
To: Victor Toso; +Cc: qemu-devel, Eduardo Habkost, Gerd Hoffmann
Hi,
2016-01-14 15:34 keltezéssel, Victor Toso írta:
> Hi,
>
> On Thu, Jan 14, 2016 at 02:45:13PM +0100, Kővágó, Zoltán wrote:
>> Hi,
>>
>> Sorry guys, I've disappeared for a while... But anyways, I've updated
>> my 5.1 audio patches to git master.
>>
>> About qapi flattening. That didn't really work last time, so now I've
>> modified OptsVisitor to support both the old way of operation (ignoring
>> structs) and the new way (requiring fully qualified paths for members
>> inside structs). Not sure if qapi people would like the idea though...
>>
>> Otherwise I just updated it to work with master. Haven't really tested
>> on any OS beside Linux.
>>
>> Please review.
>
> Patches 35 and 50 are missing
Hmm, weird. Gmail marks them as delivered. Is there a spamfilter or
something like on the list? I will try to resend them anyway, sorry if
it arrives multiple times.
Zoltan
>
>>
>>
>> Kővágó, Zoltán (51):
>> qapi: support implicit structs in OptsVisitor
>> qapi: support nested structs in OptsVisitor
>> qapi: qapi for audio backends
>> audio: use qapi AudioFormat instead of audfmt_e
>> audio: -audiodev command line option: documentation
>> audio: -audiodev command line option basic implementation
>> alsaaudio: port to -audiodev config
>> coreaudio: port to -audiodev config
>> dsoundaudio: port to -audiodev config
>> noaudio: port to -audiodev config
>> ossaudio: port to -audiodev config
>> paaudio: port to -audiodev config
>> sdlaudio: port to -audiodev config
>> spiceaudio: port to -audiodev config
>> wavaudio: port to -audiodev config
>> audio: -audiodev command line option: cleanup
>> audio: reduce glob_audio_state usage
>> audio: basic support for multi backend audio
>> audio: add audiodev properties to frontends
>> audio: audiodev= parameters no longer optional when -audiodev present
>> paaudio: do not move stream when sink/source name is specified
>> paaudio: properly disconnect streams in fini_*
>> audio: remove audio_MIN, audio_MAX
>> audio: do not run each backend in audio_run
>> paaudio: fix playback glitches
>> audio: remove read and write pcm_ops
>> audio: use size_t where makes sense
>> audio: api for mixeng code free backends
>> alsaaudio: port to the new audio backend api
>> coreaudio: port to the new audio backend api
>> noaudio: port to the new audio backend api
>> ossaudio: port to the new audio backend api
>> paaudio: port to the new audio backend api
>> sdlaudio: port to the new audio backend api
>> spiceaudio: port to the new audio backend api
> this one ^
>> wavaudio: port to the new audio backend api
>> audio: remove remains of the old backend api
>> audio: unify input and output mixeng buffer management
>> audio: remove hw->samples, buffer_size_in/out pcm_ops
>> audio: common rate control code for timer based outputs
>> audio: split ctl_* functions into enable_* and volume_*
>> audio: add mixeng option (documentation)
>> audio: make mixeng optional
>> paaudio: get/put_buffer functions
>> audio: support more than two channels in volume setting
>> audio: replace shift in audio_pcm_info with bytes_per_frame
>> audio: basic support for multichannel audio
>> paaudio: channel-map option
>> usb-audio: do not count on avail bytes actually available
>> usb-audio: support more than two channels of audio
> this one ^
> toso
>> usbaudio: change playback counters to 64 bit
>>
>> Makefile | 6 +-
>> audio/Makefile.objs | 3 +-
>> audio/alsaaudio.c | 751 +++++----------
>> audio/audio.c | 1518 ++++++++++++++++---------------
>> audio/audio.h | 76 +-
>> audio/audio_int.h | 124 +--
>> audio/audio_legacy.c | 333 +++++++
>> audio/audio_pt_int.c | 173 ----
>> audio/audio_pt_int.h | 22 -
>> audio/audio_template.h | 99 +-
>> audio/audio_win_int.c | 18 +-
>> audio/coreaudio.c | 198 ++--
>> audio/dsound_template.h | 63 +-
>> audio/dsoundaudio.c | 449 +++------
>> audio/mixeng.h | 11 +-
>> audio/noaudio.c | 98 +-
>> audio/ossaudio.c | 596 +++++-------
>> audio/paaudio.c | 881 ++++++++----------
>> audio/rate_template.h | 2 +-
>> audio/sdlaudio.c | 183 ++--
>> audio/spiceaudio.c | 286 ++----
>> audio/wavaudio.c | 158 +---
>> audio/wavcapture.c | 8 +-
>> configure | 5 -
>> hmp-commands.hx | 11 +-
>> hmp.c | 2 +-
>> hw/acpi/core.c | 2 +-
>> hw/arm/omap2.c | 2 +-
>> hw/audio/ac97.c | 13 +-
>> hw/audio/adlib.c | 7 +-
>> hw/audio/cs4231a.c | 11 +-
>> hw/audio/es1370.c | 17 +-
>> hw/audio/gus.c | 9 +-
>> hw/audio/hda-codec.c | 19 +-
>> hw/audio/lm4549.c | 6 +-
>> hw/audio/milkymist-ac97.c | 16 +-
>> hw/audio/pcspk.c | 5 +-
>> hw/audio/pl041.c | 1 +
>> hw/audio/sb16.c | 17 +-
>> hw/audio/wm8750.c | 14 +-
>> hw/core/qdev-properties-system.c | 57 ++
>> hw/input/tsc210x.c | 2 +-
>> hw/usb/dev-audio.c | 460 ++++++++--
>> include/hw/qdev-properties.h | 3 +
>> include/qapi/opts-visitor.h | 2 +-
>> monitor.c | 12 +-
>> net/net.c | 2 +-
>> numa.c | 2 +-
>> qapi-schema.json | 3 +
>> qapi/audio.json | 271 ++++++
>> qapi/opts-visitor.c | 147 ++-
>> qemu-options.hx | 242 ++++-
>> tests/qapi-schema/qapi-schema-test.json | 9 +-
>> tests/qapi-schema/qapi-schema-test.out | 4 +
>> tests/test-opts-visitor.c | 36 +-
>> ui/vnc.c | 29 +-
>> ui/vnc.h | 2 +
>> vl.c | 13 +-
>> 58 files changed, 3835 insertions(+), 3674 deletions(-)
>> create mode 100644 audio/audio_legacy.c
>> delete mode 100644 audio/audio_pt_int.c
>> delete mode 100644 audio/audio_pt_int.h
>> create mode 100644 qapi/audio.json
>>
>> --
>> 2.7.0
>>
>>
^ permalink raw reply [flat|nested] 56+ messages in thread
* Re: [Qemu-devel] [PATCH 01/51] qapi: support implicit structs in OptsVisitor
2016-01-14 13:45 ` [Qemu-devel] [PATCH 01/51] qapi: support implicit structs in OptsVisitor Kővágó, Zoltán
@ 2016-01-14 16:43 ` Eric Blake
0 siblings, 0 replies; 56+ messages in thread
From: Eric Blake @ 2016-01-14 16:43 UTC (permalink / raw)
To: Kővágó, Zoltán, qemu-devel
Cc: Markus Armbruster, Michael Roth, Gerd Hoffmann
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On 01/14/2016 06:45 AM, Kővágó, Zoltán wrote:
> They are required for flat unions (you still have to allocate the
> structs).
>
> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
> ---
> qapi/opts-visitor.c | 15 +++++++++++++++
> 1 file changed, 15 insertions(+)
I basically have a modified version of this patch already in my pending
work:
https://lists.gnu.org/archive/html/qemu-devel/2015-12/msg04393.html
but rebased on top of other pending qapi changes.
--
Eric Blake eblake redhat com +1-919-301-3266
Libvirt virtualization library http://libvirt.org
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^ permalink raw reply [flat|nested] 56+ messages in thread
* Re: [Qemu-devel] [PATCH 03/51] qapi: qapi for audio backends
2016-01-14 13:45 ` [Qemu-devel] [PATCH 03/51] qapi: qapi for audio backends Kővágó, Zoltán
@ 2016-01-14 16:51 ` Eric Blake
2016-01-15 13:49 ` Gerd Hoffmann
0 siblings, 1 reply; 56+ messages in thread
From: Eric Blake @ 2016-01-14 16:51 UTC (permalink / raw)
To: Kővágó, Zoltán, qemu-devel
Cc: Gerd Hoffmann, Markus Armbruster
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On 01/14/2016 06:45 AM, 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>
> ---
> +
> +##
> +# @AudiodevNoOptions
> +#
> +# The none, coreaudio, sdl and spice audio backend have no options.
> +#
> +# Since: 2.6
> +##
> +{ 'struct': 'AudiodevNoOptions',
> + 'data': { } }
I've got pending patches that will let us write a flat union as:
... 'data': { 'none': {},
'coreaudio': {},
'foo': 'FooType', ...
but until that lands, your explicit empty-type placeholder is fine.
> +##
> +{ 'struct': 'AudiodevCommonOptions',
> + 'data': {
> + 'id': 'str',
> + 'driver': 'AudiodevDriver',
> + 'in': 'AudiodevPerDirectionOptions',
> + 'out': 'AudiodevPerDirectionOptions',
> + '*timer-period': 'int' } }
> +
> +##
> +# @AudiodevBackendOptions
> +#
> +# Options of an audio backend.
> +#
> +# Since: 2.6
> +##
> +{ 'union': 'Audiodev',
> + 'base': 'AudiodevCommonOptions',
My pending patches will also allow an anonymous base class, so that we
don't have to specify an intermediate AudiodevCommonOptions class. Not
a show-stopper, though.
Overall looks rather sane (I think we've done the bulk of the reviewing
last summer), although I didn't read it closely enough for a R-b tag at
this time.
--
Eric Blake eblake redhat com +1-919-301-3266
Libvirt virtualization library http://libvirt.org
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^ permalink raw reply [flat|nested] 56+ messages in thread
* Re: [Qemu-devel] [PATCH 03/51] qapi: qapi for audio backends
2016-01-14 16:51 ` Eric Blake
@ 2016-01-15 13:49 ` Gerd Hoffmann
0 siblings, 0 replies; 56+ messages in thread
From: Gerd Hoffmann @ 2016-01-15 13:49 UTC (permalink / raw)
To: Eric Blake
Cc: Markus Armbruster, qemu-devel, Kővágó, Zoltán
Hi,
> My pending patches will also allow an anonymous base class, so that we
> don't have to specify an intermediate AudiodevCommonOptions class. Not
> a show-stopper, though.
>
> Overall looks rather sane (I think we've done the bulk of the reviewing
> last summer), although I didn't read it closely enough for a R-b tag at
> this time.
Yes. I was basically waiting for the qapi patches to land first. Which
took quite a bit longer than I expected. v8 posted in December ...
So, what is the status and the plan here?
I still would prefer to have the qapi bits go in first, then merge the
audio work on top of this. But I also don't want wait forever, I'd like
to see the audio bits finally land in qemu 2.6.
cheers,
Gerd
^ permalink raw reply [flat|nested] 56+ messages in thread
* Re: [Qemu-devel] [PATCH 00/51] audio 5.1 patches
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
` (49 preceding siblings ...)
2016-01-14 14:34 ` [Qemu-devel] [PATCH 00/51] audio 5.1 patches Victor Toso
@ 2016-01-15 13:51 ` Gerd Hoffmann
50 siblings, 0 replies; 56+ messages in thread
From: Gerd Hoffmann @ 2016-01-15 13:51 UTC (permalink / raw)
To: Kővágó, Zoltán; +Cc: qemu-devel, Eduardo Habkost
On Do, 2016-01-14 at 14:45 +0100, Kővágó, Zoltán wrote:
> Hi,
>
> Sorry guys, I've disappeared for a while...
No problem. qapi changes still pending anyway ...
I'll try to sort with Eric how we'll handle the qapi bits best.
> But anyways, I've updated
> my 5.1 audio patches to git master.
Thanks, I'll check ASAP (early next week I hope).
cheers,
Gerd
^ permalink raw reply [flat|nested] 56+ messages in thread
end of thread, other threads:[~2016-01-15 13:52 UTC | newest]
Thread overview: 56+ messages (download: mbox.gz follow: Atom feed
-- links below jump to the message on this page --
2016-01-14 13:45 [Qemu-devel] [PATCH 00/51] audio 5.1 patches Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 01/51] qapi: support implicit structs in OptsVisitor Kővágó, Zoltán
2016-01-14 16:43 ` Eric Blake
2016-01-14 13:45 ` [Qemu-devel] [PATCH 02/51] qapi: support nested " Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 03/51] qapi: qapi for audio backends Kővágó, Zoltán
2016-01-14 16:51 ` Eric Blake
2016-01-15 13:49 ` Gerd Hoffmann
2016-01-14 13:45 ` [Qemu-devel] [PATCH 04/51] audio: use qapi AudioFormat instead of audfmt_e Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 05/51] audio: -audiodev command line option: documentation Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 06/51] audio: -audiodev command line option basic implementation Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 07/51] alsaaudio: port to -audiodev config Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 08/51] coreaudio: " Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 09/51] dsoundaudio: " Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 10/51] noaudio: " Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 11/51] ossaudio: " Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 12/51] paaudio: " Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 13/51] sdlaudio: " Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 14/51] spiceaudio: " Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 15/51] wavaudio: " Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 16/51] audio: -audiodev command line option: cleanup Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 17/51] audio: reduce glob_audio_state usage Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 18/51] audio: basic support for multi backend audio Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 19/51] audio: add audiodev properties to frontends Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 20/51] audio: audiodev= parameters no longer optional when -audiodev present Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 21/51] paaudio: do not move stream when sink/source name is specified Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 22/51] paaudio: properly disconnect streams in fini_* Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 23/51] audio: remove audio_MIN, audio_MAX Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 24/51] audio: do not run each backend in audio_run Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 25/51] paaudio: fix playback glitches Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 26/51] audio: remove read and write pcm_ops Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 27/51] audio: use size_t where makes sense Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 28/51] audio: api for mixeng code free backends Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 29/51] alsaaudio: port to the new audio backend api Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 30/51] coreaudio: " Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 31/51] noaudio: " Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 32/51] ossaudio: " Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 33/51] paaudio: " Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 34/51] sdlaudio: " Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 36/51] wavaudio: " Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 37/51] audio: remove remains of the old " Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 38/51] audio: unify input and output mixeng buffer management Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 39/51] audio: remove hw->samples, buffer_size_in/out pcm_ops Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 40/51] audio: common rate control code for timer based outputs Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 41/51] audio: split ctl_* functions into enable_* and volume_* Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 42/51] audio: add mixeng option (documentation) Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 43/51] audio: make mixeng optional Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 44/51] paaudio: get/put_buffer functions Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 45/51] audio: support more than two channels in volume setting Kővágó, Zoltán
2016-01-14 13:45 ` [Qemu-devel] [PATCH 46/51] audio: replace shift in audio_pcm_info with bytes_per_frame Kővágó, Zoltán
2016-01-14 13:46 ` [Qemu-devel] [PATCH 47/51] audio: basic support for multichannel audio Kővágó, Zoltán
2016-01-14 13:46 ` [Qemu-devel] [PATCH 48/51] paaudio: channel-map option Kővágó, Zoltán
2016-01-14 13:46 ` [Qemu-devel] [PATCH 49/51] usb-audio: do not count on avail bytes actually available Kővágó, Zoltán
2016-01-14 13:46 ` [Qemu-devel] [PATCH 51/51] usbaudio: change playback counters to 64 bit Kővágó, Zoltán
2016-01-14 14:34 ` [Qemu-devel] [PATCH 00/51] audio 5.1 patches Victor Toso
2016-01-14 15:30 ` Kővágó Zoltán
2016-01-15 13:51 ` Gerd Hoffmann
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