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audio: -audiodev option, multiple options

PATCH Superseded Kővágó, Zoltán 4 participants 57 messages Aug 6, 2015 – Aug 21, 2015

Subsystems (5)

v1 of 2 Superseded Latest is v2 →

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cover audio: -audiodev option, multiple options

This patch series adds support to multiple audio backends. The first part of the series (until -audiodev command line option) was submitted multiple times, see
http://lists.nongnu.org/archive/html/qemu-devel/2015-06/msg05037.html
and
http://lists.nongnu.org/archive/html/qemu-devel/2015-07/msg00920.html

Afterwards I add support to multiple backends. Audio fronteds gain a new audiodev option to specify the id of the audiodev to use. The audiodev= option is required, unless you use the old environment variable based configuration, in that case it must not used (and you can't use multiple backends).

Finally I also make mixeng usage optional, it can save us some useless format converting when not needed. Also makes easier to support formats currently not supported by qemu (as only the backend has to support it, not the mixeng).

For easier testing pull https://github.com/DirtYiCE/qemu.git tag audio-multi-v1.

Please review.

Kővágó, Zoltán (25):
qapi: support implicit structs in OptsVisitor
qapi: convert NumaOptions into a flat union
net: remove NetLegacy struct
net: use Netdev instead of NetClientOptions in client init
qapi: change Netdev into a flat union
qapi: reorder NetdevBase and Netdev
qapi: qapi for audio backends
qapi: support nested structs in OptsVisitor
audio: use qapi AudioFormat instead of audfmt_e
audio: -audiodev command line option
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 create multiple connections to the same server
paaudio: properly disconnect streams in fini_*
audio: remove gcc specific audio_MIN, audio_MAX
audio: do not run each backend in audio_run
paaudio: fix playback glitches
audio: remove mixeng specific code from backends
audio: common rate control code for timer based outputs
audio: add mixeng option (documentation)
audio: make mixeng optional
paaudio: get/put_buffer functions
audio: split ctl_* functions into enable_* and volume_*

 Makefile                                |    4 +-
 audio/Makefile.objs                     |    3 +-
 audio/alsaaudio.c                       |  739 +++++-----------
 audio/audio.c                           | 1436 +++++++++++++++----------------
 audio/audio.h                           |   67 +-
 audio/audio_int.h                       |  121 +--
 audio/audio_legacy.c                    |  328 +++++++
 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                 |   59 +-
 audio/dsoundaudio.c                     |  444 +++-------
 audio/mixeng.h                          |   11 +-
 audio/noaudio.c                         |   98 +--
 audio/ossaudio.c                        |  589 +++++--------
 audio/paaudio.c                         |  860 ++++++++----------
 audio/rate_template.h                   |    2 +-
 audio/sdlaudio.c                        |  183 ++--
 audio/spiceaudio.c                      |  283 ++----
 audio/wavaudio.c                        |  156 +---
 audio/wavcapture.c                      |    8 +-
 configure                               |    5 -
 hmp-commands.hx                         |   11 +-
 hw/arm/musicpal.c                       |    2 +-
 hw/arm/omap2.c                          |    2 +-
 hw/audio/ac97.c                         |    3 +-
 hw/audio/adlib.c                        |    3 +-
 hw/audio/cs4231a.c                      |    7 +-
 hw/audio/es1370.c                       |   11 +-
 hw/audio/gus.c                          |    3 +-
 hw/audio/hda-codec.c                    |   19 +-
 hw/audio/lm4549.c                       |    6 +-
 hw/audio/milkymist-ac97.c               |    8 +-
 hw/audio/pcspk.c                        |    3 +-
 hw/audio/pl041.c                        |    1 +
 hw/audio/sb16.c                         |   15 +-
 hw/audio/wm8750.c                       |   10 +-
 hw/core/qdev-properties-system.c        |   59 +-
 hw/input/tsc210x.c                      |    2 +-
 hw/net/allwinner_emac.c                 |    2 +-
 hw/net/cadence_gem.c                    |    2 +-
 hw/net/dp8393x.c                        |    2 +-
 hw/net/e1000.c                          |    2 +-
 hw/net/eepro100.c                       |    2 +-
 hw/net/etraxfs_eth.c                    |    2 +-
 hw/net/fsl_etsec/etsec.c                |    2 +-
 hw/net/lan9118.c                        |    2 +-
 hw/net/lance.c                          |    2 +-
 hw/net/mcf_fec.c                        |    2 +-
 hw/net/milkymist-minimac2.c             |    2 +-
 hw/net/mipsnet.c                        |    2 +-
 hw/net/ne2000-isa.c                     |    2 +-
 hw/net/ne2000.c                         |    2 +-
 hw/net/opencores_eth.c                  |    2 +-
 hw/net/pcnet-pci.c                      |    2 +-
 hw/net/rocker/rocker_fp.c               |    2 +-
 hw/net/rtl8139.c                        |    2 +-
 hw/net/smc91c111.c                      |    2 +-
 hw/net/spapr_llan.c                     |    2 +-
 hw/net/stellaris_enet.c                 |    2 +-
 hw/net/vhost_net.c                      |   18 +-
 hw/net/virtio-net.c                     |    6 +-
 hw/net/vmxnet3.c                        |    2 +-
 hw/net/xen_nic.c                        |    2 +-
 hw/net/xgmac.c                          |    2 +-
 hw/net/xilinx_axienet.c                 |    2 +-
 hw/net/xilinx_ethlite.c                 |    2 +-
 hw/usb/dev-audio.c                      |    3 +-
 hw/usb/dev-network.c                    |    2 +-
 include/hw/qdev-properties.h            |    3 +
 include/net/net.h                       |    4 +-
 monitor.c                               |   26 +-
 net/clients.h                           |   20 +-
 net/dump.c                              |    8 +-
 net/hub.c                               |   24 +-
 net/l2tpv3.c                            |    8 +-
 net/net.c                               |  129 ++-
 net/netmap.c                            |    6 +-
 net/slirp.c                             |    8 +-
 net/socket.c                            |   10 +-
 net/tap-win32.c                         |    8 +-
 net/tap.c                               |   28 +-
 net/vde.c                               |    8 +-
 net/vhost-user.c                        |   14 +-
 numa.c                                  |    2 +-
 qapi-schema.json                        |  132 +--
 qapi/audio.json                         |  268 ++++++
 qapi/opts-visitor.c                     |  129 ++-
 qemu-options.hx                         |  237 ++++-
 tests/qapi-schema/qapi-schema-test.json |    9 +-
 tests/test-opts-visitor.c               |   34 +
 ui/vnc.c                                |   29 +-
 ui/vnc.h                                |    2 +
 vl.c                                    |   11 +-
 96 files changed, 3553 insertions(+), 3756 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
Signature · 1 line
2.4.5
#1 lore raw

PATCH 1/25 qapi: support implicit structs in OptsVisitor

They are required for flat unions (you still have to allocate the structs).

---
 qapi/opts-visitor.c | 15 +++++++++++++++
 1 file changed, 15 insertions(+)
qapi/opts-visitor.c +15 -0
index 7ae33b3..aa68814 100644
@@ -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;
Signature · 1 line
2.4.5
#2 lore raw

PATCH 2/25 qapi: convert NumaOptions into a flat union

In reply to Kővágó, Zoltán
---
 numa.c           |  2 +-
 qapi-schema.json | 47 ++++++++++++++++++++++++++++++++++++-----------
 2 files changed, 37 insertions(+), 12 deletions(-)
numa.c +1 -1
index 402804b..376f990 100644
@@ -227,7 +227,7 @@ static int parse_numa(void *opaque, QemuOpts *opts, Error **errp)
     }
 
     switch (object->kind) {
-    case NUMA_OPTIONS_KIND_NODE:
+    case NUMA_DRIVER_NODE:
         numa_node_parse(object->node, opts, &err);
         if (err) {
             goto error;
qapi-schema.json +36 -11

59 lines not on this page · raw message

Signature · 1 line
2.4.5
#3 lore raw

PATCH 3/25 net: remove NetLegacy struct

In reply to Kővágó, Zoltán

NetLegacy is just Netdev with some extra fields (name, vlan) and an optional id. This patch merges the two structs, and net_client_init1 got some extra checks to make sure only accept valid -netdev command lines. This is some extra work, but allows us to uniformly manage both legacy -net and non-legacy -netdev in code.

---
 net/net.c        | 42 +++++++++++++++++++++---------------------
 qapi-schema.json | 30 +++++++++---------------------
 2 files changed, 30 insertions(+), 42 deletions(-)
net/net.c +21 -21
index 28a5597..10fbaca 100644
@@ -911,17 +911,29 @@ static int (* const net_client_init_fun[NET_CLIENT_OPTIONS_KIND_MAX])(
 };
 
 
-static int net_client_init1(const void *object, int is_netdev, Error **errp)
+static int net_client_init1(const Netdev *netdev, int is_netdev, Error **errp)
 {
-    const NetClientOptions *opts;
+    const NetClientOptions *opts = netdev->opts;
     const char *name;
     NetClientState *peer = NULL;
 
     if (is_netdev) {
-        const Netdev *netdev = object;
-        opts = netdev->opts;
         name = netdev->id;
 
+        /* validate -netdev option: has id, no vlan or name */
+        if (!netdev->has_id) {
+            error_setg(errp, QERR_MISSING_PARAMETER, "id");
+            return -1;
+        }
+        if (netdev->has_name) {
+            error_setg(errp, QERR_INVALID_PARAMETER, "name");
+            return -1;
+        }
+        if (netdev->has_vlan) {
+            error_setg(errp, QERR_INVALID_PARAMETER, "vlan");
+            return -1;
+        }
+
         if (opts->kind == NET_CLIENT_OPTIONS_KIND_DUMP ||
             opts->kind == NET_CLIENT_OPTIONS_KIND_NIC ||
             !net_client_init_fun[opts->kind]) {
@@ -930,10 +942,8 @@ static int net_client_init1(const void *object, int is_netdev, Error **errp)
             return -1;
         }
     } else {
-        const NetLegacy *net = object;
-        opts = net->opts;
         /* missing optional values have been initialized to "all bits zero" */
-        name = net->has_id ? net->id : net->name;
+        name = netdev->has_id ? netdev->id : netdev->name;
 
         if (opts->kind == NET_CLIENT_OPTIONS_KIND_NONE) {
             return 0; /* nothing to do */
@@ -954,7 +964,7 @@ static int net_client_init1(const void *object, int is_netdev, Error **errp)
         /* Do not add to a vlan if it's a nic with a netdev= parameter. */
         if (opts->kind != NET_CLIENT_OPTIONS_KIND_NIC ||
             !opts->nic->has_netdev) {
-            peer = net_hub_add_port(net->has_vlan ? net->vlan : 0, NULL);
+            peer = net_hub_add_port(netdev->has_vlan ? netdev->vlan : 0, NULL);
         }
     }
 
@@ -970,26 +980,16 @@ static int net_client_init1(const void *object, int is_netdev, Error **errp)
 }
 
 
-static void net_visit(Visitor *v, int is_netdev, void **object, Error **errp)
-{
-    if (is_netdev) {
-        visit_type_Netdev(v, (Netdev **)object, NULL, errp);
-    } else {
-        visit_type_NetLegacy(v, (NetLegacy **)object, NULL, errp);
-    }
-}
-
-
 int net_client_init(QemuOpts *opts, int is_netdev, Error **errp)
 {
-    void *object = NULL;
+    Netdev *object = NULL;
     Error *err = NULL;
     int ret = -1;
 
     {
         OptsVisitor *ov = opts_visitor_new(opts);
 
-        net_visit(opts_get_visitor(ov), is_netdev, &object, &err);
+        visit_type_Netdev(opts_get_visitor(ov), &object, NULL, &err);
         opts_visitor_cleanup(ov);
     }
 
@@ -1000,7 +1000,7 @@ int net_client_init(QemuOpts *opts, int is_netdev, Error **errp)
     if (object) {
         QapiDeallocVisitor *dv = qapi_dealloc_visitor_new();
 
-        net_visit(qapi_dealloc_get_visitor(dv), is_netdev, &object, NULL);
+        visit_type_Netdev(qapi_dealloc_get_visitor(dv), &object, NULL, NULL);
         qapi_dealloc_visitor_cleanup(dv);
     }
 
qapi-schema.json +9 -21
index 999faa3..8253d0a 100644
@@ -2499,21 +2499,25 @@
     'vhost-user': 'NetdevVhostUserOptions' } }
 
 ##
-# @NetLegacy
+# @Netdev
 #
-# Captures the configuration of a network device; legacy.
+# Captures the configuration of a network device.
 #
-# @vlan: #optional vlan number
+# @vlan: #optional vlan number (legacy, forbidden with -netdev)
 #
-# @id: #optional identifier for monitor commands
+# @id: #optional identifier for monitor commands (required with -netdev)
 #
 # @name: #optional identifier for monitor commands, ignored if @id is present
+#        (legacy, forbidden with -netdev)
 #
 # @opts: device type specific properties (legacy)
 #
 # Since 1.2
+#
+# @id #optional - since 2.5
+# @vlan, @name - since 2.5
 ##
-{ 'struct': 'NetLegacy',
+{ 'struct': 'Netdev',
   'data': {
     '*vlan': 'int32',
     '*id':   'str',
@@ -2521,22 +2525,6 @@
     'opts':  'NetClientOptions' } }
 
 ##
-# @Netdev
-#
-# Captures the configuration of a network device.
-#
-# @id: identifier for monitor commands.
-#
-# @opts: device type specific properties
-#
-# Since 1.2
-##
-{ 'struct': 'Netdev',
-  'data': {
-    'id':   'str',
-    'opts': 'NetClientOptions' } }
-
-##
 # @InetSocketAddress
 #
 # Captures a socket address or address range in the Internet namespace.
Signature · 1 line
2.4.5
#4 lore raw

PATCH 4/25 net: use Netdev instead of NetClientOptions in client init

In reply to Kővágó, Zoltán

This way we no longer need NetClientOptions and can convert Netdev into a flat union.

---
 net/clients.h    | 20 ++++++++++----------
 net/dump.c       |  6 +++---
 net/hub.c        |  6 +++---
 net/l2tpv3.c     |  6 +++---
 net/net.c        | 29 ++++++++++++++---------------
 net/netmap.c     |  4 ++--
 net/slirp.c      |  6 +++---
 net/socket.c     |  6 +++---
 net/tap-win32.c  |  6 +++---
 net/tap.c        | 12 ++++++------
 net/vde.c        |  6 +++---
 net/vhost-user.c |  6 +++---
 12 files changed, 56 insertions(+), 57 deletions(-)
net/clients.h +10 -10
index d47530e..5cae479 100644
@@ -27,39 +27,39 @@
 #include "net/net.h"
 #include "qapi-types.h"
 
-int net_init_dump(const NetClientOptions *opts, const char *name,
+int net_init_dump(const Netdev *netdev, const char *name,
                   NetClientState *peer, Error **errp);
 
 #ifdef CONFIG_SLIRP
-int net_init_slirp(const NetClientOptions *opts, const char *name,
+int net_init_slirp(const Netdev *netdev, const char *name,
                    NetClientState *peer, Error **errp);
 #endif
 
-int net_init_hubport(const NetClientOptions *opts, const char *name,
+int net_init_hubport(const Netdev *netdev, const char *name,
                      NetClientState *peer, Error **errp);
 
-int net_init_socket(const NetClientOptions *opts, const char *name,
+int net_init_socket(const Netdev *netdev, const char *name,
                     NetClientState *peer, Error **errp);
 
-int net_init_tap(const NetClientOptions *opts, const char *name,
+int net_init_tap(const Netdev *netdev, const char *name,
                  NetClientState *peer, Error **errp);
 
-int net_init_bridge(const NetClientOptions *opts, const char *name,
+int net_init_bridge(const Netdev *netdev, const char *name,
                     NetClientState *peer, Error **errp);
 
-int net_init_l2tpv3(const NetClientOptions *opts, const char *name,
+int net_init_l2tpv3(const Netdev *netdev, const char *name,
                     NetClientState *peer, Error **errp);
 #ifdef CONFIG_VDE
-int net_init_vde(const NetClientOptions *opts, const char *name,
+int net_init_vde(const Netdev *netdev, const char *name,
                  NetClientState *peer, Error **errp);
 #endif
 
 #ifdef CONFIG_NETMAP
-int net_init_netmap(const NetClientOptions *opts, const char *name,
+int net_init_netmap(const Netdev *netdev, const char *name,
                     NetClientState *peer, Error **errp);
 #endif
 
-int net_init_vhost_user(const NetClientOptions *opts, const char *name,
+int net_init_vhost_user(const Netdev *netdev, const char *name,
                         NetClientState *peer, Error **errp);
 
 #endif /* QEMU_NET_CLIENTS_H */
net/dump.c +3 -3
index 02c8064..d80fa94 100644
@@ -146,7 +146,7 @@ static int net_dump_init(NetClientState *peer, const char *device,
     return 0;
 }
 
-int net_init_dump(const NetClientOptions *opts, const char *name,
+int net_init_dump(const Netdev *netdev, const char *name,
                   NetClientState *peer, Error **errp)
 {
     int len;
@@ -154,8 +154,8 @@ int net_init_dump(const NetClientOptions *opts, const char *name,
     char def_file[128];
     const NetdevDumpOptions *dump;
 
-    assert(opts->kind == NET_CLIENT_OPTIONS_KIND_DUMP);
-    dump = opts->dump;
+    assert(netdev->opts->kind == NET_CLIENT_OPTIONS_KIND_DUMP);
+    dump = netdev->opts->dump;
 
     assert(peer);
 
net/hub.c +3 -3
index 3047f12..29f65b2 100644
@@ -280,14 +280,14 @@ int net_hub_id_for_client(NetClientState *nc, int *id)
     return 0;
 }
 
-int net_init_hubport(const NetClientOptions *opts, const char *name,
+int net_init_hubport(const Netdev *netdev, const char *name,
                      NetClientState *peer, Error **errp)
 {
     const NetdevHubPortOptions *hubport;
 
-    assert(opts->kind == NET_CLIENT_OPTIONS_KIND_HUBPORT);
+    assert(netdev->opts->kind == NET_CLIENT_OPTIONS_KIND_HUBPORT);
     assert(!peer);
-    hubport = opts->hubport;
+    hubport = netdev->opts->hubport;
 
     net_hub_add_port(hubport->hubid, name);
     return 0;
net/l2tpv3.c +3 -3
index 4f9bcee..d2f8431 100644
@@ -524,7 +524,7 @@ static NetClientInfo net_l2tpv3_info = {
     .cleanup = net_l2tpv3_cleanup,
 };
 
-int net_init_l2tpv3(const NetClientOptions *opts,
+int net_init_l2tpv3(const Netdev *netdev,
                     const char *name,
                     NetClientState *peer, Error **errp)
 {
@@ -545,8 +545,8 @@ int net_init_l2tpv3(const NetClientOptions *opts,
     s->queue_tail = 0;
     s->header_mismatch = false;
 
-    assert(opts->kind == NET_CLIENT_OPTIONS_KIND_L2TPV3);
-    l2tpv3 = opts->l2tpv3;
+    assert(netdev->opts->kind == NET_CLIENT_OPTIONS_KIND_L2TPV3);
+    l2tpv3 = netdev->opts->l2tpv3;
 
     if (l2tpv3->has_ipv6 && l2tpv3->ipv6) {
         s->ipv6 = l2tpv3->ipv6;
net/net.c +14 -15
index 10fbaca..339f188 100644
@@ -813,15 +813,15 @@ int qemu_find_nic_model(NICInfo *nd, const char * const *models,
     return -1;
 }
 
-static int net_init_nic(const NetClientOptions *opts, const char *name,
+static int net_init_nic(const Netdev *netdev, const char *name,
                         NetClientState *peer, Error **errp)
 {
     int idx;
     NICInfo *nd;
     const NetLegacyNicOptions *nic;
 
-    assert(opts->kind == NET_CLIENT_OPTIONS_KIND_NIC);
-    nic = opts->nic;
+    assert(netdev->opts->kind == NET_CLIENT_OPTIONS_KIND_NIC);
+    nic = netdev->opts->nic;
 
     idx = nic_get_free_idx();
     if (idx == -1 || nb_nics >= MAX_NICS) {
@@ -882,7 +882,7 @@ static int net_init_nic(const NetClientOptions *opts, const char *name,
 
 
 static int (* const net_client_init_fun[NET_CLIENT_OPTIONS_KIND_MAX])(
-    const NetClientOptions *opts,
+    const Netdev *netdev,
     const char *name,
     NetClientState *peer, Error **errp) = {
         [NET_CLIENT_OPTIONS_KIND_NIC]       = net_init_nic,
@@ -913,7 +913,6 @@ static int (* const net_client_init_fun[NET_CLIENT_OPTIONS_KIND_MAX])(
 
 static int net_client_init1(const Netdev *netdev, int is_netdev, Error **errp)
 {
-    const NetClientOptions *opts = netdev->opts;
     const char *name;
     NetClientState *peer = NULL;
 
@@ -934,9 +933,9 @@ static int net_client_init1(const Netdev *netdev, int is_netdev, Error **errp)
             return -1;
         }
 
-        if (opts->kind == NET_CLIENT_OPTIONS_KIND_DUMP ||
-            opts->kind == NET_CLIENT_OPTIONS_KIND_NIC ||
-            !net_client_init_fun[opts->kind]) {
+        if (netdev->opts->kind == NET_CLIENT_OPTIONS_KIND_DUMP ||
+            netdev->opts->kind == NET_CLIENT_OPTIONS_KIND_NIC ||
+            !net_client_init_fun[netdev->opts->kind]) {
             error_setg(errp, QERR_INVALID_PARAMETER_VALUE, "type",
                        "a netdev backend type");
             return -1;
@@ -945,16 +944,16 @@ static int net_client_init1(const Netdev *netdev, int is_netdev, Error **errp)
         /* missing optional values have been initialized to "all bits zero" */
         name = netdev->has_id ? netdev->id : netdev->name;
 
-        if (opts->kind == NET_CLIENT_OPTIONS_KIND_NONE) {
+        if (netdev->opts->kind == NET_CLIENT_OPTIONS_KIND_NONE) {
             return 0; /* nothing to do */
         }
-        if (opts->kind == NET_CLIENT_OPTIONS_KIND_HUBPORT) {
+        if (netdev->opts->kind == NET_CLIENT_OPTIONS_KIND_HUBPORT) {
             error_setg(errp, QERR_INVALID_PARAMETER_VALUE, "type",
                        "a net type");
             return -1;
         }
 
-        if (!net_client_init_fun[opts->kind]) {
+        if (!net_client_init_fun[netdev->opts->kind]) {
             error_setg(errp, QERR_INVALID_PARAMETER_VALUE, "type",
                        "a net backend type (maybe it is not compiled "
                        "into this binary)");
@@ -962,17 +961,17 @@ static int net_client_init1(const Netdev *netdev, int is_netdev, Error **errp)
         }
 
         /* Do not add to a vlan if it's a nic with a netdev= parameter. */
-        if (opts->kind != NET_CLIENT_OPTIONS_KIND_NIC ||
-            !opts->nic->has_netdev) {
+        if (netdev->opts->kind != NET_CLIENT_OPTIONS_KIND_NIC ||
+            !netdev->opts->nic->has_netdev) {
             peer = net_hub_add_port(netdev->has_vlan ? netdev->vlan : 0, NULL);
         }
     }
 
-    if (net_client_init_fun[opts->kind](opts, name, peer, errp) < 0) {
+    if (net_client_init_fun[netdev->opts->kind](netdev, name, peer, errp) < 0) {
         /* FIXME drop when all init functions store an Error */
         if (errp && !*errp) {
             error_setg(errp, QERR_DEVICE_INIT_FAILED,
-                       NetClientOptionsKind_lookup[opts->kind]);
+                       NetClientOptionsKind_lookup[netdev->opts->kind]);
         }
         return -1;
     }
net/netmap.c +2 -2
index 508b829..a464618 100644
@@ -435,11 +435,11 @@ static NetClientInfo net_netmap_info = {
  *
  * ... -net netmap,ifname="..."
  */
-int net_init_netmap(const NetClientOptions *opts,
+int net_init_netmap(const Netdev *netdev,
                     const char *name, NetClientState *peer, Error **errp)
 {
     /* FIXME error_setg(errp, ...) on failure */
-    const NetdevNetmapOptions *netmap_opts = opts->netmap;
+    const NetdevNetmapOptions *netmap_opts = netdev->opts->netmap;
     NetClientState *nc;
     NetmapPriv me;
     NetmapState *s;
net/slirp.c +3 -3
index 7657b38..0fc2c52 100644
@@ -736,7 +736,7 @@ static const char **slirp_dnssearch(const StringList *dnsname)
     return ret;
 }
 
-int net_init_slirp(const NetClientOptions *opts, const char *name,
+int net_init_slirp(const Netdev *netdev, const char *name,
                    NetClientState *peer, Error **errp)
 {
     /* FIXME error_setg(errp, ...) on failure */
@@ -746,8 +746,8 @@ int net_init_slirp(const NetClientOptions *opts, const char *name,
     const NetdevUserOptions *user;
     const char **dnssearch;
 
-    assert(opts->kind == NET_CLIENT_OPTIONS_KIND_USER);
-    user = opts->user;
+    assert(netdev->opts->kind == NET_CLIENT_OPTIONS_KIND_USER);
+    user = netdev->opts->user;
 
     vnet = user->has_net ? g_strdup(user->net) :
            user->has_ip  ? g_strdup_printf("%s/24", user->ip) :
net/socket.c +3 -3
index b1e3b1c..75f693c 100644
@@ -699,15 +699,15 @@ static int net_socket_udp_init(NetClientState *peer,
     return 0;
 }
 
-int net_init_socket(const NetClientOptions *opts, const char *name,
+int net_init_socket(const Netdev *netdev, const char *name,
                     NetClientState *peer, Error **errp)
 {
     /* FIXME error_setg(errp, ...) on failure */
     Error *err = NULL;
     const NetdevSocketOptions *sock;
 
-    assert(opts->kind == NET_CLIENT_OPTIONS_KIND_SOCKET);
-    sock = opts->socket;
+    assert(netdev->opts->kind == NET_CLIENT_OPTIONS_KIND_SOCKET);
+    sock = netdev->opts->socket;
 
     if (sock->has_fd + sock->has_listen + sock->has_connect + sock->has_mcast +
         sock->has_udp != 1) {
net/tap-win32.c +3 -3
index 625d53c..acce480 100644
@@ -761,14 +761,14 @@ static int tap_win32_init(NetClientState *peer, const char *model,
     return 0;
 }
 
-int net_init_tap(const NetClientOptions *opts, const char *name,
+int net_init_tap(const Netdev *netdev, const char *name,
                  NetClientState *peer, Error **errp)
 {
     /* FIXME error_setg(errp, ...) on failure */
     const NetdevTapOptions *tap;
 
-    assert(opts->kind == NET_CLIENT_OPTIONS_KIND_TAP);
-    tap = opts->tap;
+    assert(netdev->opts->kind == NET_CLIENT_OPTIONS_KIND_TAP);
+    tap = netdev->opts->tap;
 
     if (!tap->has_ifname) {
         error_report("tap: no interface name");
net/tap.c +6 -6
index bd01590..263f807 100644
@@ -557,7 +557,7 @@ static int net_bridge_run_helper(const char *helper, const char *bridge,
     }
 }
 
-int net_init_bridge(const NetClientOptions *opts, const char *name,
+int net_init_bridge(const Netdev *netdev, const char *name,
                     NetClientState *peer, Error **errp)
 {
     const NetdevBridgeOptions *bridge;
@@ -565,8 +565,8 @@ int net_init_bridge(const NetClientOptions *opts, const char *name,
     TAPState *s;
     int fd, vnet_hdr;
 
-    assert(opts->kind == NET_CLIENT_OPTIONS_KIND_BRIDGE);
-    bridge = opts->bridge;
+    assert(netdev->opts->kind == NET_CLIENT_OPTIONS_KIND_BRIDGE);
+    bridge = netdev->opts->bridge;
 
     helper = bridge->has_helper ? bridge->helper : DEFAULT_BRIDGE_HELPER;
     br     = bridge->has_br     ? bridge->br     : DEFAULT_BRIDGE_INTERFACE;
@@ -716,7 +716,7 @@ static int get_fds(char *str, char *fds[], int max)
     return i;
 }
 
-int net_init_tap(const NetClientOptions *opts, const char *name,
+int net_init_tap(const Netdev *netdev, const char *name,
                  NetClientState *peer, Error **errp)
 {
     const NetdevTapOptions *tap;
@@ -728,8 +728,8 @@ int net_init_tap(const NetClientOptions *opts, const char *name,
     const char *vhostfdname;
     char ifname[128];
 
-    assert(opts->kind == NET_CLIENT_OPTIONS_KIND_TAP);
-    tap = opts->tap;
+    assert(netdev->opts->kind == NET_CLIENT_OPTIONS_KIND_TAP);
+    tap = netdev->opts->tap;
     queues = tap->has_queues ? tap->queues : 1;
     vhostfdname = tap->has_vhostfd ? tap->vhostfd : NULL;
 
net/vde.c +3 -3
index dacaa64..0ac2525 100644
@@ -109,14 +109,14 @@ static int net_vde_init(NetClientState *peer, const char *model,
     return 0;
 }
 
-int net_init_vde(const NetClientOptions *opts, const char *name,
+int net_init_vde(const Netdev *netdev, const char *name,
                  NetClientState *peer, Error **errp)
 {
     /* FIXME error_setg(errp, ...) on failure */
     const NetdevVdeOptions *vde;
 
-    assert(opts->kind == NET_CLIENT_OPTIONS_KIND_VDE);
-    vde = opts->vde;
+    assert(netdev->opts->kind == NET_CLIENT_OPTIONS_KIND_VDE);
+    vde = netdev->opts->vde;
 
     /* missing optional values have been initialized to "all bits zero" */
     if (net_vde_init(peer, "vde", name, vde->sock, vde->port, vde->group,
net/vhost-user.c +3 -3
index 93dcecd..20981a9 100644
@@ -223,14 +223,14 @@ static int net_vhost_check_net(void *opaque, QemuOpts *opts, Error **errp)
     return 0;
 }
 
-int net_init_vhost_user(const NetClientOptions *opts, const char *name,
+int net_init_vhost_user(const Netdev *netdev, const char *name,
                         NetClientState *peer, Error **errp)
 {
     const NetdevVhostUserOptions *vhost_user_opts;
     CharDriverState *chr;
 
-    assert(opts->kind == NET_CLIENT_OPTIONS_KIND_VHOST_USER);
-    vhost_user_opts = opts->vhost_user;
+    assert(netdev->opts->kind == NET_CLIENT_OPTIONS_KIND_VHOST_USER);
+    vhost_user_opts = netdev->opts->vhost_user;
 
     chr = net_vhost_parse_chardev(vhost_user_opts, errp);
     if (!chr) {
Signature · 1 line
2.4.5
#5 lore raw

PATCH 5/25 qapi: change Netdev into a flat union

In reply to Kővágó, Zoltán

Except qapi-schema.json, this patch was geenrated by:

find . -name .git -prune -o -type f \! -name '*~' -print0 | \

  xargs -0 sed -i \
    -e 's/NetClientOptionsKind/NetClientDriver/g' \
    -e 's/NET_CLIENT_OPTIONS_KIND_/NET_CLIENT_DRIVER_/g' \
    -e 's/netdev->opts/netdev/g'
---
 hw/arm/musicpal.c                |  2 +-
 hw/core/qdev-properties-system.c |  2 +-
 hw/net/allwinner_emac.c          |  2 +-
 hw/net/cadence_gem.c             |  2 +-
 hw/net/dp8393x.c                 |  2 +-
 hw/net/e1000.c                   |  2 +-
 hw/net/eepro100.c                |  2 +-
 hw/net/etraxfs_eth.c             |  2 +-
 hw/net/fsl_etsec/etsec.c         |  2 +-
 hw/net/lan9118.c                 |  2 +-
 hw/net/lance.c                   |  2 +-
 hw/net/mcf_fec.c                 |  2 +-
 hw/net/milkymist-minimac2.c      |  2 +-
 hw/net/mipsnet.c                 |  2 +-
 hw/net/ne2000-isa.c              |  2 +-
 hw/net/ne2000.c                  |  2 +-
 hw/net/opencores_eth.c           |  2 +-
 hw/net/pcnet-pci.c               |  2 +-
 hw/net/rocker/rocker_fp.c        |  2 +-
 hw/net/rtl8139.c                 |  2 +-
 hw/net/smc91c111.c               |  2 +-
 hw/net/spapr_llan.c              |  2 +-
 hw/net/stellaris_enet.c          |  2 +-
 hw/net/vhost_net.c               | 18 ++++-----
 hw/net/virtio-net.c              |  6 +--
 hw/net/vmxnet3.c                 |  2 +-
 hw/net/xen_nic.c                 |  2 +-
 hw/net/xgmac.c                   |  2 +-
 hw/net/xilinx_axienet.c          |  2 +-
 hw/net/xilinx_ethlite.c          |  2 +-
 hw/usb/dev-network.c             |  2 +-
 include/net/net.h                |  4 +-
 monitor.c                        | 14 +++----
 net/dump.c                       |  6 +--
 net/hub.c                        | 22 +++++------
 net/l2tpv3.c                     |  6 +--
 net/net.c                        | 84 ++++++++++++++++++++--------------------
 net/netmap.c                     |  4 +-
 net/slirp.c                      |  6 +--
 net/socket.c                     |  8 ++--
 net/tap-win32.c                  |  6 +--
 net/tap.c                        | 24 ++++++------
 net/vde.c                        |  6 +--
 net/vhost-user.c                 | 12 +++---
 qapi-schema.json                 | 36 +++++++++++------
 45 files changed, 166 insertions(+), 154 deletions(-)
hw/arm/musicpal.c +1 -1
index 42f66b3..94bdf5f 100644
@@ -374,7 +374,7 @@ static void eth_cleanup(NetClientState *nc)
 }
 
 static NetClientInfo net_mv88w8618_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_NIC,
+    .type = NET_CLIENT_DRIVER_NIC,
     .size = sizeof(NICState),
     .receive = eth_receive,
     .cleanup = eth_cleanup,
hw/core/qdev-properties-system.c +1 -1
index 921e799..249976e 100644
@@ -229,7 +229,7 @@ static void set_netdev(Object *obj, Visitor *v, void *opaque,
     }
 
     queues = qemu_find_net_clients_except(str, peers,
-                                          NET_CLIENT_OPTIONS_KIND_NIC,
+                                          NET_CLIENT_DRIVER_NIC,
                                           MAX_QUEUE_NUM);
     if (queues == 0) {
         err = -ENOENT;
hw/net/allwinner_emac.c +1 -1
index 0407dee..4fdf824 100644
@@ -422,7 +422,7 @@ static const MemoryRegionOps aw_emac_mem_ops = {
 };
 
 static NetClientInfo net_aw_emac_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_NIC,
+    .type = NET_CLIENT_DRIVER_NIC,
     .size = sizeof(NICState),
     .can_receive = aw_emac_can_receive,
     .receive = aw_emac_receive,
hw/net/cadence_gem.c +1 -1
index 494a346..d74136a 100644
@@ -1175,7 +1175,7 @@ static void gem_set_link(NetClientState *nc)
 }
 
 static NetClientInfo net_gem_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_NIC,
+    .type = NET_CLIENT_DRIVER_NIC,
     .size = sizeof(NICState),
     .can_receive = gem_can_receive,
     .receive = gem_receive,
hw/net/dp8393x.c +1 -1
index ab607e4..fb57900 100644
@@ -810,7 +810,7 @@ static void dp8393x_reset(DeviceState *dev)
 }
 
 static NetClientInfo net_dp83932_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_NIC,
+    .type = NET_CLIENT_DRIVER_NIC,
     .size = sizeof(NICState),
     .can_receive = dp8393x_can_receive,
     .receive = dp8393x_receive,
hw/net/e1000.c +1 -1
index 5c6bcd0..06b5a52 100644
@@ -1515,7 +1515,7 @@ pci_e1000_uninit(PCIDevice *dev)
 }
 
 static NetClientInfo net_e1000_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_NIC,
+    .type = NET_CLIENT_DRIVER_NIC,
     .size = sizeof(NICState),
     .can_receive = e1000_can_receive,
     .receive = e1000_receive,
hw/net/eepro100.c +1 -1
index 60333b7..28552f7 100644
@@ -1832,7 +1832,7 @@ static void pci_nic_uninit(PCIDevice *pci_dev)
 }
 
 static NetClientInfo net_eepro100_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_NIC,
+    .type = NET_CLIENT_DRIVER_NIC,
     .size = sizeof(NICState),
     .receive = nic_receive,
 };
hw/net/etraxfs_eth.c +1 -1
index d600275..f43a170 100644
@@ -577,7 +577,7 @@ static const MemoryRegionOps eth_ops = {
 };
 
 static NetClientInfo net_etraxfs_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_NIC,
+    .type = NET_CLIENT_DRIVER_NIC,
     .size = sizeof(NICState),
     .receive = eth_receive,
     .link_status_changed = eth_set_link,
hw/net/fsl_etsec/etsec.c +1 -1
index 0f5cf44..c936b3b 100644
@@ -369,7 +369,7 @@ static void etsec_set_link_status(NetClientState *nc)
 }
 
 static NetClientInfo net_etsec_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_NIC,
+    .type = NET_CLIENT_DRIVER_NIC,
     .size = sizeof(NICState),
     .receive = etsec_receive,
     .link_status_changed = etsec_set_link_status,
hw/net/lan9118.c +1 -1
index 4f0e840..11bfd23 100644
@@ -1305,7 +1305,7 @@ static const MemoryRegionOps lan9118_16bit_mem_ops = {
 };
 
 static NetClientInfo net_lan9118_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_NIC,
+    .type = NET_CLIENT_DRIVER_NIC,
     .size = sizeof(NICState),
     .receive = lan9118_receive,
     .link_status_changed = lan9118_set_link,
hw/net/lance.c +1 -1
index 780b39d..7940c05 100644
@@ -92,7 +92,7 @@ static const MemoryRegionOps lance_mem_ops = {
 };
 
 static NetClientInfo net_lance_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_NIC,
+    .type = NET_CLIENT_DRIVER_NIC,
     .size = sizeof(NICState),
     .receive = pcnet_receive,
     .link_status_changed = pcnet_set_link_status,
hw/net/mcf_fec.c +1 -1
index 21928f9..5386597 100644
@@ -506,7 +506,7 @@ static const MemoryRegionOps mcf_fec_ops = {
 };
 
 static NetClientInfo net_mcf_fec_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_NIC,
+    .type = NET_CLIENT_DRIVER_NIC,
     .size = sizeof(NICState),
     .receive = mcf_fec_receive,
 };
hw/net/milkymist-minimac2.c +1 -1
index 5d1cf08..aff0c9f 100644
@@ -443,7 +443,7 @@ static void milkymist_minimac2_reset(DeviceState *d)
 }
 
 static NetClientInfo net_milkymist_minimac2_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_NIC,
+    .type = NET_CLIENT_DRIVER_NIC,
     .size = sizeof(NICState),
     .receive = minimac2_rx,
 };
hw/net/mipsnet.c +1 -1
index f261011..b9bf03f 100644
@@ -218,7 +218,7 @@ static const VMStateDescription vmstate_mipsnet = {
 };
 
 static NetClientInfo net_mipsnet_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_NIC,
+    .type = NET_CLIENT_DRIVER_NIC,
     .size = sizeof(NICState),
     .receive = mipsnet_receive,
 };
hw/net/ne2000-isa.c +1 -1
index 17e7199..7da26c9 100644
@@ -42,7 +42,7 @@ typedef struct ISANE2000State {
 } ISANE2000State;
 
 static NetClientInfo net_ne2000_isa_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_NIC,
+    .type = NET_CLIENT_DRIVER_NIC,
     .size = sizeof(NICState),
     .can_receive = ne2000_can_receive,
     .receive = ne2000_receive,
hw/net/ne2000.c +1 -1
index 3492db3..09fa0e3 100644
@@ -703,7 +703,7 @@ void ne2000_setup_io(NE2000State *s, DeviceState *dev, unsigned size)
 }
 
 static NetClientInfo net_ne2000_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_NIC,
+    .type = NET_CLIENT_DRIVER_NIC,
     .size = sizeof(NICState),
     .can_receive = ne2000_can_receive,
     .receive = ne2000_receive,
hw/net/opencores_eth.c +1 -1
index 3642046..4b6407b 100644
@@ -473,7 +473,7 @@ static ssize_t open_eth_receive(NetClientState *nc,
 }
 
 static NetClientInfo net_open_eth_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_NIC,
+    .type = NET_CLIENT_DRIVER_NIC,
     .size = sizeof(NICState),
     .can_receive = open_eth_can_receive,
     .receive = open_eth_receive,
hw/net/pcnet-pci.c +1 -1
index b4d60b8..c04c3a2 100644
@@ -271,7 +271,7 @@ static void pci_pcnet_uninit(PCIDevice *dev)
 }
 
 static NetClientInfo net_pci_pcnet_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_NIC,
+    .type = NET_CLIENT_DRIVER_NIC,
     .size = sizeof(NICState),
     .receive = pcnet_receive,
     .link_status_changed = pcnet_set_link_status,
hw/net/rocker/rocker_fp.c +1 -1
index c693ae5..3f82e30 100644
@@ -166,7 +166,7 @@ static void fp_port_set_link_status(NetClientState *nc)
 }
 
 static NetClientInfo fp_port_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_NIC,
+    .type = NET_CLIENT_DRIVER_NIC,
     .size = sizeof(NICState),
     .receive = fp_port_receive,
     .receive_iov = fp_port_receive_iov,
hw/net/rtl8139.c +1 -1
index edbb61c..974e706 100644
@@ -3462,7 +3462,7 @@ static void rtl8139_set_link_status(NetClientState *nc)
 }
 
 static NetClientInfo net_rtl8139_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_NIC,
+    .type = NET_CLIENT_DRIVER_NIC,
     .size = sizeof(NICState),
     .can_receive = rtl8139_can_receive,
     .receive = rtl8139_receive,
hw/net/smc91c111.c +1 -1
index 74e06e6..4c85eb5 100644
@@ -737,7 +737,7 @@ static const MemoryRegionOps smc91c111_mem_ops = {
 };
 
 static NetClientInfo net_smc91c111_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_NIC,
+    .type = NET_CLIENT_DRIVER_NIC,
     .size = sizeof(NICState),
     .can_receive = smc91c111_can_receive,
     .receive = smc91c111_receive,
hw/net/spapr_llan.c +1 -1
index 1ca5e9c..4ee4b92 100644
@@ -188,7 +188,7 @@ static ssize_t spapr_vlan_receive(NetClientState *nc, const uint8_t *buf,
 }
 
 static NetClientInfo net_spapr_vlan_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_NIC,
+    .type = NET_CLIENT_DRIVER_NIC,
     .size = sizeof(NICState),
     .can_receive = spapr_vlan_can_receive,
     .receive = spapr_vlan_receive,
hw/net/stellaris_enet.c +1 -1
index 21a4773..10290fe 100644
@@ -449,7 +449,7 @@ static void stellaris_enet_reset(stellaris_enet_state *s)
 }
 
 static NetClientInfo net_stellaris_enet_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_NIC,
+    .type = NET_CLIENT_DRIVER_NIC,
     .size = sizeof(NICState),
     .receive = stellaris_enet_receive,
 };
hw/net/vhost_net.c +9 -9
index 5c1d11f..5d24c29 100644
@@ -95,10 +95,10 @@ static const int *vhost_net_get_feature_bits(struct vhost_net *net)
     const int *feature_bits = 0;
 
     switch (net->nc->info->type) {
-    case NET_CLIENT_OPTIONS_KIND_TAP:
+    case NET_CLIENT_DRIVER_TAP:
         feature_bits = kernel_feature_bits;
         break;
-    case NET_CLIENT_OPTIONS_KIND_VHOST_USER:
+    case NET_CLIENT_DRIVER_VHOST_USER:
         feature_bits = user_feature_bits;
         break;
     default:
@@ -125,7 +125,7 @@ void vhost_net_ack_features(struct vhost_net *net, uint64_t features)
 static int vhost_net_get_fd(NetClientState *backend)
 {
     switch (backend->info->type) {
-    case NET_CLIENT_OPTIONS_KIND_TAP:
+    case NET_CLIENT_DRIVER_TAP:
         return tap_get_fd(backend);
     default:
         fprintf(stderr, "vhost-net requires tap backend\n");
@@ -236,7 +236,7 @@ static int vhost_net_start_one(struct vhost_net *net,
         net->nc->info->poll(net->nc, false);
     }
 
-    if (net->nc->info->type == NET_CLIENT_OPTIONS_KIND_TAP) {
+    if (net->nc->info->type == NET_CLIENT_DRIVER_TAP) {
         qemu_set_fd_handler(net->backend, NULL, NULL, NULL);
         file.fd = net->backend;
         for (file.index = 0; file.index < net->dev.nvqs; ++file.index) {
@@ -252,7 +252,7 @@ static int vhost_net_start_one(struct vhost_net *net,
     return 0;
 fail:
     file.fd = -1;
-    if (net->nc->info->type == NET_CLIENT_OPTIONS_KIND_TAP) {
+    if (net->nc->info->type == NET_CLIENT_DRIVER_TAP) {
         while (file.index-- > 0) {
             const VhostOps *vhost_ops = net->dev.vhost_ops;
             int r = vhost_ops->vhost_call(&net->dev, VHOST_NET_SET_BACKEND,
@@ -275,14 +275,14 @@ static void vhost_net_stop_one(struct vhost_net *net,
 {
     struct vhost_vring_file file = { .fd = -1 };
 
-    if (net->nc->info->type == NET_CLIENT_OPTIONS_KIND_TAP) {
+    if (net->nc->info->type == NET_CLIENT_DRIVER_TAP) {
         for (file.index = 0; file.index < net->dev.nvqs; ++file.index) {
             const VhostOps *vhost_ops = net->dev.vhost_ops;
             int r = vhost_ops->vhost_call(&net->dev, VHOST_NET_SET_BACKEND,
                                           &file);
             assert(r >= 0);
         }
-    } else if (net->nc->info->type == NET_CLIENT_OPTIONS_KIND_VHOST_USER) {
+    } else if (net->nc->info->type == NET_CLIENT_DRIVER_VHOST_USER) {
         for (file.index = 0; file.index < net->dev.nvqs; ++file.index) {
             const VhostOps *vhost_ops = net->dev.vhost_ops;
             int r = vhost_ops->vhost_call(&net->dev, VHOST_RESET_OWNER,
@@ -399,10 +399,10 @@ VHostNetState *get_vhost_net(NetClientState *nc)
     }
 
     switch (nc->info->type) {
-    case NET_CLIENT_OPTIONS_KIND_TAP:
+    case NET_CLIENT_DRIVER_TAP:
         vhost_net = tap_get_vhost_net(nc);
         break;
-    case NET_CLIENT_OPTIONS_KIND_VHOST_USER:
+    case NET_CLIENT_DRIVER_VHOST_USER:
         vhost_net = vhost_user_get_vhost_net(nc);
         break;
     default:
hw/net/virtio-net.c +3 -3
index 1510839..ac47bc3 100644
@@ -406,7 +406,7 @@ static int peer_attach(VirtIONet *n, int index)
         return 0;
     }
 
-    if (nc->peer->info->type != NET_CLIENT_OPTIONS_KIND_TAP) {
+    if (nc->peer->info->type != NET_CLIENT_DRIVER_TAP) {
         return 0;
     }
 
@@ -421,7 +421,7 @@ static int peer_detach(VirtIONet *n, int index)
         return 0;
     }
 
-    if (nc->peer->info->type !=  NET_CLIENT_OPTIONS_KIND_TAP) {
+    if (nc->peer->info->type !=  NET_CLIENT_DRIVER_TAP) {
         return 0;
     }
 
@@ -1599,7 +1599,7 @@ static int virtio_net_load_device(VirtIODevice *vdev, QEMUFile *f,
 }
 
 static NetClientInfo net_virtio_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_NIC,
+    .type = NET_CLIENT_DRIVER_NIC,
     .size = sizeof(NICState),
     .can_receive = virtio_net_can_receive,
     .receive = virtio_net_receive,
hw/net/vmxnet3.c +1 -1
index 59b06b8..1772683 100644
@@ -1986,7 +1986,7 @@ static void vmxnet3_set_link_status(NetClientState *nc)
 }
 
 static NetClientInfo net_vmxnet3_info = {
-        .type = NET_CLIENT_OPTIONS_KIND_NIC,
+        .type = NET_CLIENT_DRIVER_NIC,
         .size = sizeof(NICState),
         .can_receive = vmxnet3_can_receive,
         .receive = vmxnet3_receive,
hw/net/xen_nic.c +1 -1
index d7cbfc1..fd2da6a 100644
@@ -280,7 +280,7 @@ static ssize_t net_rx_packet(NetClientState *nc, const uint8_t *buf, size_t size
 /* ------------------------------------------------------------- */
 
 static NetClientInfo net_xen_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_NIC,
+    .type = NET_CLIENT_DRIVER_NIC,
     .size = sizeof(NICState),
     .receive = net_rx_packet,
 };
hw/net/xgmac.c +1 -1
index 15fb681..ccf8a77 100644
@@ -370,7 +370,7 @@ out:
 }
 
 static NetClientInfo net_xgmac_enet_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_NIC,
+    .type = NET_CLIENT_DRIVER_NIC,
     .size = sizeof(NICState),
     .receive = eth_rx,
 };
hw/net/xilinx_axienet.c +1 -1
index d63c423..cc464c9 100644
@@ -933,7 +933,7 @@ xilinx_axienet_data_stream_push(StreamSlave *obj, uint8_t *buf, size_t size)
 }
 
 static NetClientInfo net_xilinx_enet_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_NIC,
+    .type = NET_CLIENT_DRIVER_NIC,
     .size = sizeof(NICState),
     .receive = eth_rx,
 };
hw/net/xilinx_ethlite.c +1 -1
index ad6b553..ab555f6 100644
@@ -214,7 +214,7 @@ static void xilinx_ethlite_reset(DeviceState *dev)
 }
 
 static NetClientInfo net_xilinx_ethlite_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_NIC,
+    .type = NET_CLIENT_DRIVER_NIC,
     .size = sizeof(NICState),
     .can_receive = eth_can_rx,
     .receive = eth_rx,
hw/usb/dev-network.c +1 -1
index 7800cee..97b2c2a 100644
@@ -1330,7 +1330,7 @@ static void usb_net_handle_destroy(USBDevice *dev)
 }
 
 static NetClientInfo net_usbnet_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_NIC,
+    .type = NET_CLIENT_DRIVER_NIC,
     .size = sizeof(NICState),
     .receive = usbnet_receive,
     .cleanup = usbnet_cleanup,
include/net/net.h +2 -2
index 6a6cbef..c0e00ef 100644
@@ -59,7 +59,7 @@ typedef int (SetVnetLE)(NetClientState *, bool);
 typedef int (SetVnetBE)(NetClientState *, bool);
 
 typedef struct NetClientInfo {
-    NetClientOptionsKind type;
+    NetClientDriver type;
     size_t size;
     NetReceive *receive;
     NetReceive *receive_raw;
@@ -104,7 +104,7 @@ typedef struct NICState {
 char *qemu_mac_strdup_printf(const uint8_t *macaddr);
 NetClientState *qemu_find_netdev(const char *id);
 int qemu_find_net_clients_except(const char *id, NetClientState **ncs,
-                                 NetClientOptionsKind type, int max);
+                                 NetClientDriver type, int max);
 NetClientState *qemu_new_net_client(NetClientInfo *info,
                                     NetClientState *peer,
                                     const char *model,
monitor.c +7 -7
index aeea2b5..a40138b 100644
@@ -4188,8 +4188,8 @@ void netdev_add_completion(ReadLineState *rs, int nb_args, const char *str)
     }
     len = strlen(str);
     readline_set_completion_index(rs, len);
-    for (i = 0; NetClientOptionsKind_lookup[i]; i++) {
-        add_completion_option(rs, str, NetClientOptionsKind_lookup[i]);
+    for (i = 0; NetClientDriver_lookup[i]; i++) {
+        add_completion_option(rs, str, NetClientDriver_lookup[i]);
     }
 }
 
@@ -4389,7 +4389,7 @@ void set_link_completion(ReadLineState *rs, int nb_args, const char *str)
         NetClientState *ncs[MAX_QUEUE_NUM];
         int count, i;
         count = qemu_find_net_clients_except(NULL, ncs,
-                                             NET_CLIENT_OPTIONS_KIND_NONE,
+                                             NET_CLIENT_DRIVER_NONE,
                                              MAX_QUEUE_NUM);
         for (i = 0; i < MIN(count, MAX_QUEUE_NUM); i++) {
             const char *name = ncs[i]->name;
@@ -4414,7 +4414,7 @@ void netdev_del_completion(ReadLineState *rs, int nb_args, const char *str)
 
     len = strlen(str);
     readline_set_completion_index(rs, len);
-    count = qemu_find_net_clients_except(NULL, ncs, NET_CLIENT_OPTIONS_KIND_NIC,
+    count = qemu_find_net_clients_except(NULL, ncs, NET_CLIENT_DRIVER_NIC,
                                          MAX_QUEUE_NUM);
     for (i = 0; i < MIN(count, MAX_QUEUE_NUM); i++) {
         QemuOpts *opts;
@@ -4506,7 +4506,7 @@ void host_net_remove_completion(ReadLineState *rs, int nb_args, const char *str)
     readline_set_completion_index(rs, len);
     if (nb_args == 2) {
         count = qemu_find_net_clients_except(NULL, ncs,
-                                             NET_CLIENT_OPTIONS_KIND_NONE,
+                                             NET_CLIENT_DRIVER_NONE,
                                              MAX_QUEUE_NUM);
         for (i = 0; i < MIN(count, MAX_QUEUE_NUM); i++) {
             int id;
@@ -4523,13 +4523,13 @@ void host_net_remove_completion(ReadLineState *rs, int nb_args, const char *str)
         return;
     } else if (nb_args == 3) {
         count = qemu_find_net_clients_except(NULL, ncs,
-                                             NET_CLIENT_OPTIONS_KIND_NIC,
+                                             NET_CLIENT_DRIVER_NIC,
                                              MAX_QUEUE_NUM);
         for (i = 0; i < MIN(count, MAX_QUEUE_NUM); i++) {
             int id;
             const char *name;
 
-            if (ncs[i]->info->type == NET_CLIENT_OPTIONS_KIND_HUBPORT ||
+            if (ncs[i]->info->type == NET_CLIENT_DRIVER_HUBPORT ||
                 net_hub_id_for_client(ncs[i], &id)) {
                 continue;
             }
net/dump.c +3 -3
index d80fa94..e7f77c3 100644
@@ -94,7 +94,7 @@ static void dump_cleanup(NetClientState *nc)
 }
 
 static NetClientInfo net_dump_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_DUMP,
+    .type = NET_CLIENT_DRIVER_DUMP,
     .size = sizeof(DumpState),
     .receive = dump_receive,
     .cleanup = dump_cleanup,
@@ -154,8 +154,8 @@ int net_init_dump(const Netdev *netdev, const char *name,
     char def_file[128];
     const NetdevDumpOptions *dump;
 
-    assert(netdev->opts->kind == NET_CLIENT_OPTIONS_KIND_DUMP);
-    dump = netdev->opts->dump;
+    assert(netdev->kind == NET_CLIENT_DRIVER_DUMP);
+    dump = netdev->dump;
 
     assert(peer);
 
net/hub.c +11 -11
index 29f65b2..671e9ba 100644
@@ -130,7 +130,7 @@ static void net_hub_port_cleanup(NetClientState *nc)
 }
 
 static NetClientInfo net_hub_port_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_HUBPORT,
+    .type = NET_CLIENT_DRIVER_HUBPORT,
     .size = sizeof(NetHubPort),
     .can_receive = net_hub_port_can_receive,
     .receive = net_hub_port_receive,
@@ -265,10 +265,10 @@ int net_hub_id_for_client(NetClientState *nc, int *id)
 {
     NetHubPort *port;
 
-    if (nc->info->type == NET_CLIENT_OPTIONS_KIND_HUBPORT) {
+    if (nc->info->type == NET_CLIENT_DRIVER_HUBPORT) {
         port = DO_UPCAST(NetHubPort, nc, nc);
     } else if (nc->peer != NULL && nc->peer->info->type ==
-            NET_CLIENT_OPTIONS_KIND_HUBPORT) {
+            NET_CLIENT_DRIVER_HUBPORT) {
         port = DO_UPCAST(NetHubPort, nc, nc->peer);
     } else {
         return -ENOENT;
@@ -285,9 +285,9 @@ int net_init_hubport(const Netdev *netdev, const char *name,
 {
     const NetdevHubPortOptions *hubport;
 
-    assert(netdev->opts->kind == NET_CLIENT_OPTIONS_KIND_HUBPORT);
+    assert(netdev->kind == NET_CLIENT_DRIVER_HUBPORT);
     assert(!peer);
-    hubport = netdev->opts->hubport;
+    hubport = netdev->hubport;
 
     net_hub_add_port(hubport->hubid, name);
     return 0;
@@ -314,14 +314,14 @@ void net_hub_check_clients(void)
             }
 
             switch (peer->info->type) {
-            case NET_CLIENT_OPTIONS_KIND_NIC:
+            case NET_CLIENT_DRIVER_NIC:
                 has_nic = 1;
                 break;
-            case NET_CLIENT_OPTIONS_KIND_USER:
-            case NET_CLIENT_OPTIONS_KIND_TAP:
-            case NET_CLIENT_OPTIONS_KIND_SOCKET:
-            case NET_CLIENT_OPTIONS_KIND_VDE:
-            case NET_CLIENT_OPTIONS_KIND_VHOST_USER:
+            case NET_CLIENT_DRIVER_USER:
+            case NET_CLIENT_DRIVER_TAP:
+            case NET_CLIENT_DRIVER_SOCKET:
+            case NET_CLIENT_DRIVER_VDE:
+            case NET_CLIENT_DRIVER_VHOST_USER:
                 has_host_dev = 1;
                 break;
             default:
net/l2tpv3.c +3 -3
index d2f8431..f68cd76 100644
@@ -516,7 +516,7 @@ static void net_l2tpv3_cleanup(NetClientState *nc)
 }
 
 static NetClientInfo net_l2tpv3_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_L2TPV3,
+    .type = NET_CLIENT_DRIVER_L2TPV3,
     .size = sizeof(NetL2TPV3State),
     .receive = net_l2tpv3_receive_dgram,
     .receive_iov = net_l2tpv3_receive_dgram_iov,
@@ -545,8 +545,8 @@ int net_init_l2tpv3(const Netdev *netdev,
     s->queue_tail = 0;
     s->header_mismatch = false;
 
-    assert(netdev->opts->kind == NET_CLIENT_OPTIONS_KIND_L2TPV3);
-    l2tpv3 = netdev->opts->l2tpv3;
+    assert(netdev->kind == NET_CLIENT_DRIVER_L2TPV3);
+    l2tpv3 = netdev->l2tpv3;
 
     if (l2tpv3->has_ipv6 && l2tpv3->ipv6) {
         s->ipv6 = l2tpv3->ipv6;
net/net.c +42 -42

243 lines not on this page · raw message

net/netmap.c +2 -2
index a464618..eff9b8c 100644
@@ -417,7 +417,7 @@ static void netmap_set_offload(NetClientState *nc, int csum, int tso4, int tso6,
 
 /* NetClientInfo methods */
 static NetClientInfo net_netmap_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_NETMAP,
+    .type = NET_CLIENT_DRIVER_NETMAP,
     .size = sizeof(NetmapState),
     .receive = netmap_receive,
     .receive_iov = netmap_receive_iov,
@@ -439,7 +439,7 @@ int net_init_netmap(const Netdev *netdev,
                     const char *name, NetClientState *peer, Error **errp)
 {
     /* FIXME error_setg(errp, ...) on failure */
-    const NetdevNetmapOptions *netmap_opts = netdev->opts->netmap;
+    const NetdevNetmapOptions *netmap_opts = netdev->netmap;
     NetClientState *nc;
     NetmapPriv me;
     NetmapState *s;
net/slirp.c +3 -3
index 0fc2c52..707dca7 100644
@@ -126,7 +126,7 @@ static void net_slirp_cleanup(NetClientState *nc)
 }
 
 static NetClientInfo net_slirp_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_USER,
+    .type = NET_CLIENT_DRIVER_USER,
     .size = sizeof(SlirpState),
     .receive = net_slirp_receive,
     .cleanup = net_slirp_cleanup,
@@ -746,8 +746,8 @@ int net_init_slirp(const Netdev *netdev, const char *name,
     const NetdevUserOptions *user;
     const char **dnssearch;
 
-    assert(netdev->opts->kind == NET_CLIENT_OPTIONS_KIND_USER);
-    user = netdev->opts->user;
+    assert(netdev->kind == NET_CLIENT_DRIVER_USER);
+    user = netdev->user;
 
     vnet = user->has_net ? g_strdup(user->net) :
            user->has_ip  ? g_strdup_printf("%s/24", user->ip) :
net/socket.c +4 -4
index 75f693c..feb1337 100644
@@ -346,7 +346,7 @@ static void net_socket_cleanup(NetClientState *nc)
 }
 
 static NetClientInfo net_dgram_socket_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_SOCKET,
+    .type = NET_CLIENT_DRIVER_SOCKET,
     .size = sizeof(NetSocketState),
     .receive = net_socket_receive_dgram,
     .cleanup = net_socket_cleanup,
@@ -429,7 +429,7 @@ static void net_socket_connect(void *opaque)
 }
 
 static NetClientInfo net_socket_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_SOCKET,
+    .type = NET_CLIENT_DRIVER_SOCKET,
     .size = sizeof(NetSocketState),
     .receive = net_socket_receive,
     .cleanup = net_socket_cleanup,
@@ -706,8 +706,8 @@ int net_init_socket(const Netdev *netdev, const char *name,
     Error *err = NULL;
     const NetdevSocketOptions *sock;
 
-    assert(netdev->opts->kind == NET_CLIENT_OPTIONS_KIND_SOCKET);
-    sock = netdev->opts->socket;
+    assert(netdev->kind == NET_CLIENT_DRIVER_SOCKET);
+    sock = netdev->socket;
 
     if (sock->has_fd + sock->has_listen + sock->has_connect + sock->has_mcast +
         sock->has_udp != 1) {
net/tap-win32.c +3 -3
index acce480..3528e4f 100644
@@ -723,7 +723,7 @@ static void tap_set_vnet_hdr_len(NetClientState *nc, int len)
 }
 
 static NetClientInfo net_tap_win32_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_TAP,
+    .type = NET_CLIENT_DRIVER_TAP,
     .size = sizeof(TAPState),
     .receive = tap_receive,
     .cleanup = tap_cleanup,
@@ -767,8 +767,8 @@ int net_init_tap(const Netdev *netdev, const char *name,
     /* FIXME error_setg(errp, ...) on failure */
     const NetdevTapOptions *tap;
 
-    assert(netdev->opts->kind == NET_CLIENT_OPTIONS_KIND_TAP);
-    tap = netdev->opts->tap;
+    assert(netdev->kind == NET_CLIENT_DRIVER_TAP);
+    tap = netdev->tap;
 
     if (!tap->has_ifname) {
         error_report("tap: no interface name");
net/tap.c +12 -12
index 263f807..38a8f2c 100644
@@ -221,7 +221,7 @@ static bool tap_has_ufo(NetClientState *nc)
 {
     TAPState *s = DO_UPCAST(TAPState, nc, nc);
 
-    assert(nc->info->type == NET_CLIENT_OPTIONS_KIND_TAP);
+    assert(nc->info->type == NET_CLIENT_DRIVER_TAP);
 
     return s->has_ufo;
 }
@@ -230,7 +230,7 @@ static bool tap_has_vnet_hdr(NetClientState *nc)
 {
     TAPState *s = DO_UPCAST(TAPState, nc, nc);
 
-    assert(nc->info->type == NET_CLIENT_OPTIONS_KIND_TAP);
+    assert(nc->info->type == NET_CLIENT_DRIVER_TAP);
 
     return !!s->host_vnet_hdr_len;
 }
@@ -239,7 +239,7 @@ static bool tap_has_vnet_hdr_len(NetClientState *nc, int len)
 {
     TAPState *s = DO_UPCAST(TAPState, nc, nc);
 
-    assert(nc->info->type == NET_CLIENT_OPTIONS_KIND_TAP);
+    assert(nc->info->type == NET_CLIENT_DRIVER_TAP);
 
     return !!tap_probe_vnet_hdr_len(s->fd, len);
 }
@@ -248,7 +248,7 @@ static void tap_set_vnet_hdr_len(NetClientState *nc, int len)
 {
     TAPState *s = DO_UPCAST(TAPState, nc, nc);
 
-    assert(nc->info->type == NET_CLIENT_OPTIONS_KIND_TAP);
+    assert(nc->info->type == NET_CLIENT_DRIVER_TAP);
     assert(len == sizeof(struct virtio_net_hdr_mrg_rxbuf) ||
            len == sizeof(struct virtio_net_hdr));
 
@@ -260,7 +260,7 @@ static void tap_using_vnet_hdr(NetClientState *nc, bool using_vnet_hdr)
 {
     TAPState *s = DO_UPCAST(TAPState, nc, nc);
 
-    assert(nc->info->type == NET_CLIENT_OPTIONS_KIND_TAP);
+    assert(nc->info->type == NET_CLIENT_DRIVER_TAP);
     assert(!!s->host_vnet_hdr_len == using_vnet_hdr);
 
     s->using_vnet_hdr = using_vnet_hdr;
@@ -326,14 +326,14 @@ static void tap_poll(NetClientState *nc, bool enable)
 int tap_get_fd(NetClientState *nc)
 {
     TAPState *s = DO_UPCAST(TAPState, nc, nc);
-    assert(nc->info->type == NET_CLIENT_OPTIONS_KIND_TAP);
+    assert(nc->info->type == NET_CLIENT_DRIVER_TAP);
     return s->fd;
 }
 
 /* fd support */
 
 static NetClientInfo net_tap_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_TAP,
+    .type = NET_CLIENT_DRIVER_TAP,
     .size = sizeof(TAPState),
     .receive = tap_receive,
     .receive_raw = tap_receive_raw,
@@ -565,8 +565,8 @@ int net_init_bridge(const Netdev *netdev, const char *name,
     TAPState *s;
     int fd, vnet_hdr;
 
-    assert(netdev->opts->kind == NET_CLIENT_OPTIONS_KIND_BRIDGE);
-    bridge = netdev->opts->bridge;
+    assert(netdev->kind == NET_CLIENT_DRIVER_BRIDGE);
+    bridge = netdev->bridge;
 
     helper = bridge->has_helper ? bridge->helper : DEFAULT_BRIDGE_HELPER;
     br     = bridge->has_br     ? bridge->br     : DEFAULT_BRIDGE_INTERFACE;
@@ -728,8 +728,8 @@ int net_init_tap(const Netdev *netdev, const char *name,
     const char *vhostfdname;
     char ifname[128];
 
-    assert(netdev->opts->kind == NET_CLIENT_OPTIONS_KIND_TAP);
-    tap = netdev->opts->tap;
+    assert(netdev->kind == NET_CLIENT_DRIVER_TAP);
+    tap = netdev->tap;
     queues = tap->has_queues ? tap->queues : 1;
     vhostfdname = tap->has_vhostfd ? tap->vhostfd : NULL;
 
@@ -890,7 +890,7 @@ int net_init_tap(const Netdev *netdev, const char *name,
 VHostNetState *tap_get_vhost_net(NetClientState *nc)
 {
     TAPState *s = DO_UPCAST(TAPState, nc, nc);
-    assert(nc->info->type == NET_CLIENT_OPTIONS_KIND_TAP);
+    assert(nc->info->type == NET_CLIENT_DRIVER_TAP);
     return s->vhost_net;
 }
 
net/vde.c +3 -3
index 0ac2525..585fc23 100644
@@ -68,7 +68,7 @@ static void vde_cleanup(NetClientState *nc)
 }
 
 static NetClientInfo net_vde_info = {
-    .type = NET_CLIENT_OPTIONS_KIND_VDE,
+    .type = NET_CLIENT_DRIVER_VDE,
     .size = sizeof(VDEState),
     .receive = vde_receive,
     .cleanup = vde_cleanup,
@@ -115,8 +115,8 @@ int net_init_vde(const Netdev *netdev, const char *name,
     /* FIXME error_setg(errp, ...) on failure */
     const NetdevVdeOptions *vde;
 
-    assert(netdev->opts->kind == NET_CLIENT_OPTIONS_KIND_VDE);
-    vde = netdev->opts->vde;
+    assert(netdev->kind == NET_CLIENT_DRIVER_VDE);
+    vde = netdev->vde;
 
     /* missing optional values have been initialized to "all bits zero" */
     if (net_vde_init(peer, "vde", name, vde->sock, vde->port, vde->group,
net/vhost-user.c +6 -6
index 20981a9..d504e26 100644
@@ -30,7 +30,7 @@ typedef struct VhostUserChardevProps {
 VHostNetState *vhost_user_get_vhost_net(NetClientState *nc)
 {
     VhostUserState *s = DO_UPCAST(VhostUserState, nc, nc);
-    assert(nc->info->type == NET_CLIENT_OPTIONS_KIND_VHOST_USER);
+    assert(nc->info->type == NET_CLIENT_DRIVER_VHOST_USER);
     return s->vhost_net;
 }
 
@@ -75,20 +75,20 @@ static void vhost_user_cleanup(NetClientState *nc)
 
 static bool vhost_user_has_vnet_hdr(NetClientState *nc)
 {
-    assert(nc->info->type == NET_CLIENT_OPTIONS_KIND_VHOST_USER);
+    assert(nc->info->type == NET_CLIENT_DRIVER_VHOST_USER);
 
     return true;
 }
 
 static bool vhost_user_has_ufo(NetClientState *nc)
 {
-    assert(nc->info->type == NET_CLIENT_OPTIONS_KIND_VHOST_USER);
+    assert(nc->info->type == NET_CLIENT_DRIVER_VHOST_USER);
 
     return true;
 }
 
 static NetClientInfo net_vhost_user_info = {
-        .type = NET_CLIENT_OPTIONS_KIND_VHOST_USER,
+        .type = NET_CLIENT_DRIVER_VHOST_USER,
         .size = sizeof(VhostUserState),
         .cleanup = vhost_user_cleanup,
         .has_vnet_hdr = vhost_user_has_vnet_hdr,
@@ -229,8 +229,8 @@ int net_init_vhost_user(const Netdev *netdev, const char *name,
     const NetdevVhostUserOptions *vhost_user_opts;
     CharDriverState *chr;
 
-    assert(netdev->opts->kind == NET_CLIENT_OPTIONS_KIND_VHOST_USER);
-    vhost_user_opts = netdev->opts->vhost_user;
+    assert(netdev->kind == NET_CLIENT_DRIVER_VHOST_USER);
+    vhost_user_opts = netdev->vhost_user;
 
     chr = net_vhost_parse_chardev(vhost_user_opts, errp);
     if (!chr) {
qapi-schema.json +24 -12
index 8253d0a..fd6b6a2 100644
@@ -2474,16 +2474,31 @@
     '*vhostforce':    'bool' } }
 
 ##
-# @NetClientOptions
+# @NetClientDriver
 #
-# A discriminated record of network device traits.
+# Available netdev drivers.
+#
+# Since 2.5
+##
+{ 'enum': 'NetClientDriver',
+  'data': [ 'none', 'nic', 'user', 'tap', 'l2tpv3', 'socket', 'vde', 'dump',
+            'bridge', 'hubport', 'netmap', 'vhost-user' ] }
+
+##
+# @Netdev
+#
+# Captures the configuration of a network device.
 #
 # Since 1.2
 #
 # 'l2tpv3' - since 2.1
+# @id #optional - since 2.5
+# @vlan, @name - since 2.5
 #
 ##
-{ 'union': 'NetClientOptions',
+{ 'union': 'Netdev',
+  'base': 'NetdevBase',
+  'discriminator': 'type',
   'data': {
     'none':     'NetdevNoneOptions',
     'nic':      'NetLegacyNicOptions',
@@ -2499,9 +2514,9 @@
     'vhost-user': 'NetdevVhostUserOptions' } }
 
 ##
-# @Netdev
+# @NetdevBase
 #
-# Captures the configuration of a network device.
+# Captures the commopn configuration of a network device.
 #
 # @vlan: #optional vlan number (legacy, forbidden with -netdev)
 #
@@ -2510,19 +2525,16 @@
 # @name: #optional identifier for monitor commands, ignored if @id is present
 #        (legacy, forbidden with -netdev)
 #
-# @opts: device type specific properties (legacy)
+# @type: the netdev driver to use
 #
-# Since 1.2
-#
-# @id #optional - since 2.5
-# @vlan, @name - since 2.5
+# Since 2.5
 ##
-{ 'struct': 'Netdev',
+{ 'struct': 'NetdevBase',
   'data': {
     '*vlan': 'int32',
     '*id':   'str',
     '*name': 'str',
-    'opts':  'NetClientOptions' } }
+    'type':  'NetClientDriver' } }
 
 ##
 # @InetSocketAddress
Signature · 1 line
2.4.5
#6 lore raw

Re: [Qemu-devel] [PATCH 05/25] qapi: change Netdev into a flat union

Hi

Quoted 1 line · Zoltán

Except qapi-schema.json, this patch was geenrated by:

generated

Quoted 1 line

+# Captures the commopn configuration of a network device.

common

Signature · 1 line
Marc-André Lureau
#7 lore raw

PATCH 6/25 qapi: reorder NetdevBase and Netdev

In reply to Kővágó, Zoltán

Probably more logical if NetdevBase comes before Netdev. No schematic changes.

---
 qapi-schema.json | 46 +++++++++++++++++++++++-----------------------
 1 file changed, 23 insertions(+), 23 deletions(-)
qapi-schema.json +23 -23
index fd6b6a2..c9d9263 100644
@@ -2485,6 +2485,29 @@
             'bridge', 'hubport', 'netmap', 'vhost-user' ] }
 
 ##
+# @NetdevBase
+#
+# Captures the commopn configuration of a network device.
+#
+# @vlan: #optional vlan number (legacy, forbidden with -netdev)
+#
+# @id: #optional identifier for monitor commands (required with -netdev)
+#
+# @name: #optional identifier for monitor commands, ignored if @id is present
+#        (legacy, forbidden with -netdev)
+#
+# @type: the netdev driver to use
+#
+# Since 2.5
+##
+{ 'struct': 'NetdevBase',
+  'data': {
+    '*vlan': 'int32',
+    '*id':   'str',
+    '*name': 'str',
+    'type':  'NetClientDriver' } }
+
+##
 # @Netdev
 #
 # Captures the configuration of a network device.
@@ -2514,29 +2537,6 @@
     'vhost-user': 'NetdevVhostUserOptions' } }
 
 ##
-# @NetdevBase
-#
-# Captures the commopn configuration of a network device.
-#
-# @vlan: #optional vlan number (legacy, forbidden with -netdev)
-#
-# @id: #optional identifier for monitor commands (required with -netdev)
-#
-# @name: #optional identifier for monitor commands, ignored if @id is present
-#        (legacy, forbidden with -netdev)
-#
-# @type: the netdev driver to use
-#
-# Since 2.5
-##
-{ 'struct': 'NetdevBase',
-  'data': {
-    '*vlan': 'int32',
-    '*id':   'str',
-    '*name': 'str',
-    'type':  'NetClientDriver' } }
-
-##
 # @InetSocketAddress
 #
 # Captures a socket address or address range in the Internet namespace.
Signature · 1 line
2.4.5
#8 lore raw

PATCH 7/25 qapi: qapi for audio backends

In reply to Kővágó, Zoltán

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)

---
 Makefile         |   4 +-
 qapi-schema.json |   3 +
 qapi/audio.json  | 263 +++++++++++++++++++++++++++++++++++++++++++++++++++++++
 3 files changed, 268 insertions(+), 2 deletions(-)
 create mode 100644 qapi/audio.json
Makefile +2 -2
index 340d9c8..bdd0bc6 100644
@@ -262,8 +262,8 @@ $(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/audio.json  $(SRC_PATH)/qapi/block.json \
+               $(SRC_PATH)/qapi/block-core.json $(SRC_PATH)/qapi/event.json
 
 qapi-types.c qapi-types.h :\
 $(qapi-modules) $(SRC_PATH)/scripts/qapi-types.py $(qapi-py)
qapi-schema.json +3 -0
index c9d9263..e9cbe15 100644
@@ -5,6 +5,9 @@
 # QAPI common definitions
 { 'include': 'qapi/common.json' }
 
+# QAPI audio definitions
+{ 'include': 'qapi/audio.json' }
+
 # QAPI block definitions
 { 'include': 'qapi/block.json' }
 
qapi/audio.json new file +263 -0

263 lines not on this page · raw message

Signature · 1 line
2.4.5
#9 lore raw

Re: [Qemu-devel] [PATCH 07/25] qapi: qapi for audio backends

Quoted 99 lines · Zoltán

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)

---
 Makefile         |   4 +-
 qapi-schema.json |   3 +
 qapi/audio.json  | 263 +++++++++++++++++++++++++++++++++++++++++++++++++++++++
 3 files changed, 268 insertions(+), 2 deletions(-)
 create mode 100644 qapi/audio.json
Makefile +2 -2
index 340d9c8..bdd0bc6 100644
@@ -262,8 +262,8 @@ $(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/audio.json  $(SRC_PATH)/qapi/block.json \
+               $(SRC_PATH)/qapi/block-core.json $(SRC_PATH)/qapi/event.json
 
 qapi-types.c qapi-types.h :\
 $(qapi-modules) $(SRC_PATH)/scripts/qapi-types.py $(qapi-py)
qapi-schema.json +3 -0
index c9d9263..e9cbe15 100644
@@ -5,6 +5,9 @@
 # QAPI common definitions
 { 'include': 'qapi/common.json' }
 
+# QAPI audio definitions
+{ 'include': 'qapi/audio.json' }
+
 # QAPI block definitions
 { 'include': 'qapi/block.json' }
 
qapi/audio.json new file +34 -0
new file mode 100644
index 0000000..b57b215
@@ -0,0 +1,263 @@
+# -*- mode: python -*-
+#
+# Copyright (C) 2015 Zoltán Kővágó <[email protected]>
+#
+# 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: 2.5
+##
+{ '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: 2.5
+##
+{ 'struct': 'AudiodevAlsaPerDirectionOptions',
+  'data': {
+    '*dev':      'str',
+    '*try-poll': 'bool' } }
+

qapi/audio.json line 44

Quoted 10 lines
qapi/audio.json new file +10 -0
+##
+# @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 microsecods) when playback starts

microseconds.

Quoted 223 lines
qapi/audio.json new file +219 -1
+#
+# Since: 2.5
+##
+{ '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: 2.5
+##
+{ '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: 2.5
+##
+{ '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: 2.5
+##
+{ '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: 2.5
+##
+{ '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: 2.5
+##
+{ '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: 2.5
+##
+{ 'struct': 'AudiodevWavOptions',
+  'data': {
+    '*path': 'str' } }
+
+
+##
+# @AudioFormat
+#
+# An enumeration of possible audio formats.
+#
+# Since: 2.5
+##
+{ 'enum': 'AudioFormat',
+  'data': [ 'u8', 's8', 'u16', 's16', 'u32', 's32' ] }
+
+##
+# @AudiodevDriver
+#
+# An enumeration of possible audio backend drivers.
+#
+# Since: 2.5
+##
+{ '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: 2.5
+##
+{ '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: 2.5
+##
+{ 'struct': 'AudiodevCommonOptions',
+  'data': {
+    'id':            'str',
+    'driver':        'AudiodevDriver',
+    'in':            'AudiodevPerDirectionOptions',
+    'out':           'AudiodevPerDirectionOptions',
+    '*timer-period': 'int' } }
+
+##
+# @AudiodevBackendOptions
+#
+# Options of an audio backend.
+#
+# Since: 2.5
+##
+{ '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' } }
--

2.4.5

Signature · 1 line
Marc-André Lureau
#10 lore raw

PATCH 8/25 qapi: support nested structs in OptsVisitor

In reply to Kővágó, Zoltán

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' orout.frequency'.

You must provide a full path even in non-ambigous cases. The qapi flattening commits hopefully ensures that this change doesn't create backward compatibility problems.

---
 qapi/opts-visitor.c                     | 114 ++++++++++++++++++++++++++------
 tests/qapi-schema/qapi-schema-test.json |   9 ++-
 tests/test-opts-visitor.c               |  34 ++++++++++
 3 files changed, 135 insertions(+), 22 deletions(-)
qapi/opts-visitor.c +93 -21
index aa68814..ff61d42 100644
@@ -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 +132,9 @@ opts_start_struct(Visitor *v, void **obj, const char *kind,
     if (obj) {
         *obj = g_malloc0(size > 0 ? size : 1);
     }
+
+    g_queue_push_tail(ov->nested_names, (gpointer) name);
+
     if (ov->depth++ > 0) {
         return;
     }
@@ -169,6 +177,8 @@ opts_end_struct(Visitor *v, Error **errp)
     OptsVisitor *ov = DO_UPCAST(OptsVisitor, visitor, v);
     GQueue *any;
 
+    g_queue_pop_tail(ov->nested_names);
+
     if (--ov->depth > 0) {
         return;
     }
@@ -198,15 +208,54 @@ 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;
+    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);
+
+    list = g_hash_table_lookup(ov->unprocessed_opts, key);
+    if (list && out_key) {
+        *out_key = g_strdup(key);
+    }
 
-    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 +267,7 @@ opts_start_list(Visitor *v, const char *name, Error **errp)
 
     /* we can't traverse a list in a list */
     assert(ov->list_mode == LM_NONE);
-    ov->repeated_opts = lookup_distinct(ov, name, errp);
+    ov->repeated_opts = lookup_distinct(ov, name, &ov->repeated_name, errp);
     if (ov->repeated_opts != NULL) {
         ov->list_mode = LM_STARTED;
     }
@@ -254,11 +303,9 @@ 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 +331,28 @@ 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 +374,15 @@ 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 +392,9 @@ 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 +411,15 @@ 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 +431,14 @@ 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 +452,13 @@ 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 +479,7 @@ 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 +491,14 @@ 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 +510,13 @@ 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 +535,7 @@ 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 +546,9 @@ 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 +558,13 @@ 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);
 }
 
 
@@ -506,7 +575,7 @@ opts_optional(Visitor *v, bool *present, const char *name, Error **errp)
 
     /* we only support a single mandatory scalar field in a list node */
     assert(ov->list_mode == LM_NONE);
-    *present = (lookup_distinct(ov, name, NULL) != NULL);
+    *present = (lookup_distinct(ov, name, NULL, NULL) != NULL);
 }
 
 
@@ -517,6 +586,8 @@ opts_visitor_new(const QemuOpts *opts)
 
     ov = g_malloc0(sizeof *ov);
 
+    ov->nested_names = g_queue_new();
+
     ov->visitor.start_struct = &opts_start_struct;
     ov->visitor.end_struct   = &opts_end_struct;
 
@@ -560,6 +631,7 @@ opts_visitor_cleanup(OptsVisitor *ov)
     if (ov->unprocessed_opts != NULL) {
         g_hash_table_destroy(ov->unprocessed_opts);
     }
+    g_queue_free(ov->nested_names);
     g_free(ov->fake_id_opt);
     g_free(ov);
 }
tests/qapi-schema/qapi-schema-test.json +8 -1
index c7eaa86..a818eff 100644
@@ -81,6 +81,11 @@
 { 'command': 'user_def_cmd3', 'data': {'a': 'int', '*b': 'int' },
   'returns': 'int' }
 
+# For testing hierarchy support in opts-visitor
+{ 'struct': 'UserDefOptionsSub',
+  'data': {
+    '*nint': 'int' } }
+
 # For testing integer range flattening in opts-visitor. The following schema
 # corresponds to the option format:
 #
@@ -94,7 +99,9 @@
     '*u64' : [ 'uint64' ],
     '*u16' : [ 'uint16' ],
     '*i64x':   'int'     ,
-    '*u64x':   'uint64'  } }
+    '*u64x':   'uint64'  ,
+    'sub0':    'UserDefOptionsSub',
+    'sub1':    'UserDefOptionsSub' } }
 
 # testing event
 { 'struct': 'EventStructOne',
tests/test-opts-visitor.c +34 -0
index 1c753d9..4393266 100644
@@ -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;
 }
Signature · 1 line
2.4.5
#11 lore raw

Re: [Qemu-devel] [PATCH 08/25] qapi: support nested structs in OptsVisitor

Hi

Quoted 103 lines · Zoltán

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' orout.frequency'.

You must provide a full path even in non-ambigous cases. The qapi flattening commits hopefully ensures that this change doesn't create backward compatibility problems.

---
 qapi/opts-visitor.c                     | 114 ++++++++++++++++++++++++++------
 tests/qapi-schema/qapi-schema-test.json |   9 ++-
 tests/test-opts-visitor.c               |  34 ++++++++++
 3 files changed, 135 insertions(+), 22 deletions(-)
qapi/opts-visitor.c +36 -1
index aa68814..ff61d42 100644
@@ -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 +132,9 @@ opts_start_struct(Visitor *v, void **obj, const char *kind,
     if (obj) {
         *obj = g_malloc0(size > 0 ? size : 1);
     }
+
+    g_queue_push_tail(ov->nested_names, (gpointer) name);
+
     if (ov->depth++ > 0) {
         return;
     }
@@ -169,6 +177,8 @@ opts_end_struct(Visitor *v, Error **errp)
     OptsVisitor *ov = DO_UPCAST(OptsVisitor, visitor, v);
     GQueue *any;
 
+    g_queue_pop_tail(ov->nested_names);
+
     if (--ov->depth > 0) {
         return;
     }
@@ -198,15 +208,54 @@ 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)
 {

qapi/opts-visitor.c opts_end_implicit_struct(Visitor *v, Error **errp) · line 247

Quoted 10 lines
qapi/opts-visitor.c +9 -1
-    GQueue *list;
+    GQueue *list = NULL;
+    char *key;
+    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);

Instead of using a GQueue, I think you could use a GArray, and use g_strjoin() here.

qapi/opts-visitor.c opts_end_implicit_struct(Visitor *v, Error **errp) · line 251

Quoted 4 lines
qapi/opts-visitor.c +4 -0
+
+    list = g_hash_table_lookup(ov->unprocessed_opts, key);
+    if (list && out_key) {
+        *out_key = g_strdup(key);

or just *out_key = key; key = NULL; (g_free accepts NULL)

Quoted 12 lines
qapi/opts-visitor.c +3 -1
+    }
 
-    list = g_hash_table_lookup(ov->unprocessed_opts, name);
     if (!list) {
         error_setg(errp, QERR_MISSING_PARAMETER, name);
     }
+
+    g_free(key);
     return list;
 }
 

qapi/opts-visitor.c opts_start_list(Visitor *v, const char *name, Error **errp) · line 221

Quoted 4 lines
qapi/opts-visitor.c +0 -1
@@ -218,7 +267,7 @@ opts_start_list(Visitor *v, const char *name, Error **errp)
 
     /* we can't traverse a list in a list */
     assert(ov->list_mode == LM_NONE);
-    ov->repeated_opts = lookup_distinct(ov, name, errp);

It would make sense to add an assert(ov->repeated_name == NULL) imho, this could catch potential leaks.

Quoted 242 lines
qapi/opts-visitor.c +41 -17
+    ov->repeated_opts = lookup_distinct(ov, name, &ov->repeated_name, errp);
     if (ov->repeated_opts != NULL) {
         ov->list_mode = LM_STARTED;
     }
@@ -254,11 +303,9 @@ 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 +331,28 @@ 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 +374,15 @@ 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 +392,9 @@ 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 +411,15 @@ 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 +431,14 @@ 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 +452,13 @@ 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 +479,7 @@ 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 +491,14 @@ 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 +510,13 @@ 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 +535,7 @@ 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 +546,9 @@ 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 +558,13 @@ 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);
 }
 
 
@@ -506,7 +575,7 @@ opts_optional(Visitor *v, bool *present, const char *name, Error **errp)
 
     /* we only support a single mandatory scalar field in a list node */
     assert(ov->list_mode == LM_NONE);
-    *present = (lookup_distinct(ov, name, NULL) != NULL);
+    *present = (lookup_distinct(ov, name, NULL, NULL) != NULL);
 }
 
 
@@ -517,6 +586,8 @@ opts_visitor_new(const QemuOpts *opts)
 
     ov = g_malloc0(sizeof *ov);
 
+    ov->nested_names = g_queue_new();
+
     ov->visitor.start_struct = &opts_start_struct;
     ov->visitor.end_struct   = &opts_end_struct;
 
@@ -560,6 +631,7 @@ opts_visitor_cleanup(OptsVisitor *ov)
     if (ov->unprocessed_opts != NULL) {
         g_hash_table_destroy(ov->unprocessed_opts);
     }
+    g_queue_free(ov->nested_names);
     g_free(ov->fake_id_opt);
     g_free(ov);
 }
tests/qapi-schema/qapi-schema-test.json +5 -0
index c7eaa86..a818eff 100644
@@ -81,6 +81,11 @@
 { 'command': 'user_def_cmd3', 'data': {'a': 'int', '*b': 'int' },
   'returns': 'int' }
 
+# For testing hierarchy support in opts-visitor
+{ 'struct': 'UserDefOptionsSub',
+  'data': {
+    '*nint': 'int' } }
+
 # For testing integer range flattening in opts-visitor. The following schema
 # corresponds to the option format:
 #

tests/qapi-schema/qapi-schema-test.json line 103

Quoted 6 lines
tests/qapi-schema/qapi-schema-test.json +2 -1
@@ -94,7 +99,9 @@
     '*u64' : [ 'uint64' ],
     '*u16' : [ 'uint16' ],
     '*i64x':   'int'     ,
-    '*u64x':   'uint64'  } }
+    '*u64x':   'uint64'  ,
+    'sub0':    'UserDefOptionsSub',

make check fails with this patch, you need to update tests/qapi-schema/qapi-schema-test.out.

Quoted 60 lines
tests/qapi-schema/qapi-schema-test.json +1 -0
+    'sub1':    'UserDefOptionsSub' } }
 
 # testing event
 { 'struct': 'EventStructOne',
tests/test-opts-visitor.c +34 -1
index 1c753d9..4393266 100644
@@ -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;
 }
--

2.4.5

Signature · 1 line
Marc-André Lureau
#12 lore raw

Re: [Qemu-devel] [PATCH 08/25] qapi: support nested structs in OptsVisitor

Hi,

2015-08-20 17:55 keltezéssel, Marc-André Lureau írta:
[snip]

Quoted 34 lines
Quoted 34 lines
qapi/opts-visitor.c +30 -2
+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;
+    size_t sum_len = strlen(name);
+
Quoted 8 lines

qapi/opts-visitor.c opts_end_implicit_struct(Visitor *v, Error **errp) · line 247

Quoted 5 lines
qapi/opts-visitor.c +5 -0
+    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);

Instead of using a GQueue, I think you could use a GArray, and use g_strjoin() here.

Unfortunately g_strjoinv won't skip NULL pointers in the array (as it expects a NULL terminated array), but we will get structs with NULL name (the top level struct will be such one). And one can't simply say if (name) { g_array_append_val(...); } in opts_start_struct because we need to know somehow in opts_end_struct if we need to pop or not.

One way is to use NULL pointers because in that case we must pop, but g_strjoinv won't work. An alternate is to only push non-NULL elements into the array and have another bool array where we store if we need to pop or not. Not sure which one is cleaner.

Or maybe if it's guaranteed that only the top-level struct will have a NULL name, we can hack around by only pushing non-NULL elements and ignore empty array when popping.

Zoltan

#13 lore raw

PATCH 9/25 audio: use qapi AudioFormat instead of audfmt_e

In reply to Kővágó, Zoltán

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.

---
 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(-)
audio/alsaaudio.c +28 -25
index 6315b2d..2b28b99 100644
@@ -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;
audio/audio.c +53 -44

230 lines not on this page · raw message

audio/audio.h +1 -10
index e7ea397..e300511 100644
@@ -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;
 };
 
audio/audio_win_int.c +9 -9
index e132405..a8cfa77 100644
@@ -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:
audio/ossaudio.c +15 -15
index 7dbe333..3ea7d27 100644
@@ -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;
 
audio/paaudio.c +14 -14
index fea6071..cfdbdc6 100644
@@ -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;
     }
 }
 
audio/sdlaudio.c +13 -13
index 1140f2e..db0f95a 100644
@@ -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;
audio/spiceaudio.c +2 -2
index 42ae4a4..141fd8d 100644
@@ -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);
audio/wavaudio.c +10 -7
index c586020..81250e6 100644
@@ -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"
 };
 
audio/wavcapture.c +1 -1
index 86e9056..798acdd 100644
@@ -136,7 +136,7 @@ int wav_start_capture (CaptureState *s, const char *path, int freq,
 
     as.freq = freq;
     as.nchannels = 1 << stereo;
-    as.fmt = bits16 ? AUD_FMT_S16 : AUD_FMT_U8;
+    as.fmt = bits16 ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_U8;
     as.endianness = 0;
 
     ops.notify = wav_notify;
hw/arm/omap2.c +1 -1
index e39b317..3b14a5d 100644
@@ -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);
hw/audio/ac97.c +1 -1
index b173835..fa75f33 100644
@@ -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) {
hw/audio/adlib.c +1 -1
index 656eb37..f8f0f55 100644
@@ -323,7 +323,7 @@ static void adlib_realizefn (DeviceState *dev, Error **errp)
 
     as.freq = s->freq;
     as.nchannels = SHIFT;
-    as.fmt = AUD_FMT_S16;
+    as.fmt = AUDIO_FORMAT_S16;
     as.endianness = AUDIO_HOST_ENDIANNESS;
 
     AUD_register_card ("adlib", &s->card);
hw/audio/cs4231a.c +3 -3
index f96f561..626a173 100644
@@ -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;
 
hw/audio/es1370.c +2 -2
index 8e7bcf5..f6e74cb 100644
@@ -414,14 +414,14 @@ static void es1370_update_voices (ES1370State *s, uint32_t ctl, uint32_t sctl)
                     i,
                     new_freq,
                     1 << (new_fmt & 1),
-                    (new_fmt & 2) ? AUD_FMT_S16 : AUD_FMT_U8,
+                    (new_fmt & 2) ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_U8,
                     d->shift);
             if (new_freq) {
                 struct audsettings as;
 
                 as.freq = new_freq;
                 as.nchannels = 1 << (new_fmt & 1);
-                as.fmt = (new_fmt & 2) ? AUD_FMT_S16 : AUD_FMT_U8;
+                as.fmt = (new_fmt & 2) ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_U8;
                 as.endianness = 0;
 
                 if (i == ADC_CHANNEL) {
hw/audio/gus.c +1 -1
index 86223a9..6107824 100644
@@ -242,7 +242,7 @@ static void gus_realizefn (DeviceState *dev, Error **errp)
 
     as.freq = s->freq;
     as.nchannels = 2;
-    as.fmt = AUD_FMT_S16;
+    as.fmt = AUDIO_FORMAT_S16;
     as.endianness = GUS_ENDIANNESS;
 
     s->voice = AUD_open_out (
hw/audio/hda-codec.c +9 -9
index 3c03ff5..8693b7a 100644
@@ -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;
hw/audio/lm4549.c +3 -3
index 380ef60..9d4f4b5 100644
@@ -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(
hw/audio/milkymist-ac97.c +1 -1
index 28f55e8..15169e2 100644
@@ -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,
hw/audio/pcspk.c +1 -1
index 5266fb5..302debf 100644
@@ -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);
 
hw/audio/sb16.c +7 -7
index b052de5..a159dcc 100644
@@ -66,7 +66,7 @@ typedef struct SB16State {
     int fmt_stereo;
     int fmt_signed;
     int fmt_bits;
-    audfmt_e fmt;
+    AudioFormat fmt;
     int dma_auto;
     int block_size;
     int fifo;
@@ -221,7 +221,7 @@ static void continue_dma8 (SB16State *s)
 
 static void dma_cmd8 (SB16State *s, int mask, int dma_len)
 {
-    s->fmt = AUD_FMT_U8;
+    s->fmt = AUDIO_FORMAT_U8;
     s->use_hdma = 0;
     s->fmt_bits = 8;
     s->fmt_signed = 0;
@@ -316,18 +316,18 @@ static void dma_cmd (SB16State *s, uint8_t cmd, uint8_t d0, int dma_len)
 
     if (16 == s->fmt_bits) {
         if (s->fmt_signed) {
-            s->fmt = AUD_FMT_S16;
+            s->fmt = AUDIO_FORMAT_S16;
         }
         else {
-            s->fmt = AUD_FMT_U16;
+            s->fmt = AUDIO_FORMAT_U16;
         }
     }
     else {
         if (s->fmt_signed) {
-            s->fmt = AUD_FMT_S8;
+            s->fmt = AUDIO_FORMAT_S8;
         }
         else {
-            s->fmt = AUD_FMT_U8;
+            s->fmt = AUDIO_FORMAT_U8;
         }
     }
 
@@ -839,7 +839,7 @@ static void legacy_reset (SB16State *s)
 
     as.freq = s->freq;
     as.nchannels = 1;
-    as.fmt = AUD_FMT_U8;
+    as.fmt = AUDIO_FORMAT_U8;
     as.endianness = 0;
 
     s->voice = AUD_open_out (
hw/audio/wm8750.c +2 -2
index b50b331..4c4333c 100644
@@ -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);
hw/input/tsc210x.c +1 -1
index fae3385..3cf938b 100644
@@ -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);
hw/usb/dev-audio.c +1 -1
index f092bb8..0171579 100644
@@ -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);
 
ui/vnc.c +7 -7
index e26973a..febbad6 100644
@@ -2363,12 +2363,12 @@ 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);
@@ -3069,7 +3069,7 @@ 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);
Signature · 1 line
2.4.5
#15 lore raw

PATCH 10/25 audio: -audiodev command line option

In reply to Kővágó, Zoltán

This patch adds an -audiodev command line option, and deprecates the QEMU_* environment variables for audio backend configuration. It's syntax is similar to existing options (-netdev, -device, etc): -audiodev driver_name,property=value,...

Audio drivers now get an Audiodev * as config paramters, instead of the global audio_option structs. There is some code in audio/audio_legacy.c that converts the old environment variables to audiodev options (this way backends do not have to worry about legacy options). It also contains a replacement of -audio-help, which prints out the equivalent -audiodev based config of the currently specified environment variables.

Although now it's possible to specify multiple -audiodev options on command line, multiple audio backends are not supported yet.

---
 audio/Makefile.objs     |   2 +-
 audio/alsaaudio.c       | 311 ++++++--------------
 audio/audio.c           | 760 ++++++++++++++----------------------------------
 audio/audio.h           |  23 +-
 audio/audio_int.h       |  23 +-
 audio/audio_legacy.c    | 328 +++++++++++++++++++++
 audio/audio_template.h  |  13 +-
 audio/coreaudio.c       |  49 +---
 audio/dsound_template.h |   6 +-
 audio/dsoundaudio.c     |  60 ++--
 audio/noaudio.c         |   3 +-
 audio/ossaudio.c        | 155 +++-------
 audio/paaudio.c         |  82 ++----
 audio/sdlaudio.c        |  24 +-
 audio/spiceaudio.c      |   7 +-
 audio/wavaudio.c        |  60 +---
 qemu-options.hx         | 226 +++++++++++++-
 vl.c                    |  10 +-
 18 files changed, 1015 insertions(+), 1127 deletions(-)
 create mode 100644 audio/audio_legacy.c
audio/Makefile.objs +1 -1
index 481d1aa..9d8f579 100644
@@ -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
audio/alsaaudio.c +84 -227

480 lines not on this page · raw message

audio/audio.c +217 -543

909 lines not on this page · raw message

audio/audio.h +20 -3
index e300511..177a673 100644
@@ -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 */
audio/audio_int.h +5 -18
index 566df5e..59b2362 100644
@@ -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;
@@ -143,8 +127,7 @@ struct SWVoiceIn {
 struct audio_driver {
     const char *name;
     const char *descr;
-    struct audio_option *options;
-    void *(*init) (void);
+    void *(*init) (Audiodev *);
     void (*fini) (void *);
     struct audio_pcm_ops *pcm_ops;
     int can_be_default;
@@ -190,6 +173,7 @@ struct SWVoiceCap {
 
 struct AudioState {
     struct audio_driver *drv;
+    Audiodev *dev;
     void *drv_opaque;
 
     QEMUTimer *ts;
@@ -200,6 +184,7 @@ 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 +198,8 @@ 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);
 
audio/audio_legacy.c new file +328 -0

328 lines not on this page · raw message

audio/audio_template.h +9 -4
index 99b27b2..096b2b3 100644
@@ -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;
     }
audio/coreaudio.c +11 -38

100 lines not on this page · raw message

audio/dsound_template.h +3 -3
index b439f33..96181ef 100644
@@ -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,
audio/dsoundaudio.c +17 -43

123 lines not on this page · raw message

audio/noaudio.c +1 -2
index 50db1f3..4c94a26 100644
@@ -134,7 +134,7 @@ static int no_ctl_in (HWVoiceIn *hw, int cmd, ...)
     return 0;
 }
 
-static void *no_audio_init (void)
+static void *no_audio_init (Audiodev *dev)
 {
     return &no_audio_init;
 }
@@ -161,7 +161,6 @@ static struct audio_pcm_ops no_pcm_ops = {
 struct audio_driver no_audio_driver = {
     .name           = "none",
     .descr          = "Timer based audio emulation",
-    .options        = NULL,
     .init           = no_audio_init,
     .fini           = no_audio_fini,
     .pcm_ops        = &no_pcm_ops,
audio/ossaudio.c +43 -112

302 lines not on this page · raw message

audio/paaudio.c +30 -52

185 lines not on this page · raw message

audio/sdlaudio.c +5 -19
index db0f95a..796238a 100644
@@ -44,18 +44,13 @@ typedef struct SDLVoiceOut {
     int decr;
 } SDLVoiceOut;
 
-static struct {
-    int nb_samples;
-} conf = {
-    .nb_samples = 1024
-};
-
 static struct SDLAudioState {
     int exit;
     SDL_mutex *mutex;
     SDL_sem *sem;
     int initialized;
     bool driver_created;
+    Audiodev *dev;
 } glob_sdl;
 typedef struct SDLAudioState SDLAudioState;
 
@@ -347,7 +342,7 @@ static int sdl_init_out(HWVoiceOut *hw, struct audsettings *as,
     req.freq = as->freq;
     req.format = aud_to_sdlfmt (as->fmt);
     req.channels = as->nchannels;
-    req.samples = conf.nb_samples;
+    req.samples = audio_buffer_samples(s->dev->out, as, 11610);
     req.callback = sdl_callback;
     req.userdata = sdl;
 
@@ -391,7 +386,7 @@ static int sdl_ctl_out (HWVoiceOut *hw, int cmd, ...)
     return 0;
 }
 
-static void *sdl_audio_init (void)
+static void *sdl_audio_init(Audiodev *dev)
 {
     SDLAudioState *s = &glob_sdl;
     if (s->driver_created) {
@@ -420,6 +415,7 @@ static void *sdl_audio_init (void)
     }
 
     s->driver_created = true;
+    s->dev = dev;
     return s;
 }
 
@@ -431,18 +427,9 @@ static void sdl_audio_fini (void *opaque)
     SDL_DestroyMutex (s->mutex);
     SDL_QuitSubSystem (SDL_INIT_AUDIO);
     s->driver_created = false;
+    s->dev = NULL;
 }
 
-static struct audio_option sdl_options[] = {
-    {
-        .name  = "SAMPLES",
-        .tag   = AUD_OPT_INT,
-        .valp  = &conf.nb_samples,
-        .descr = "Size of SDL buffer in samples"
-    },
-    { /* End of list */ }
-};
-
 static struct audio_pcm_ops sdl_pcm_ops = {
     .init_out = sdl_init_out,
     .fini_out = sdl_fini_out,
@@ -454,7 +441,6 @@ static struct audio_pcm_ops sdl_pcm_ops = {
 struct audio_driver sdl_audio_driver = {
     .name           = "sdl",
     .descr          = "SDL http://www.libsdl.org",
-    .options        = sdl_options,
     .init           = sdl_audio_init,
     .fini           = sdl_audio_fini,
     .pcm_ops        = &sdl_pcm_ops,
audio/spiceaudio.c +1 -6
index 141fd8d..a9b9a1d 100644
@@ -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;
@@ -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,
audio/wavaudio.c +10 -50

106 lines not on this page · raw message

qemu-options.hx +221 -5

174 lines not on this page · raw message

vl.c +9 -1
index 0adbbd6..f1aa1df 100644

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Signature · 1 line
2.4.5

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#16 lore raw

Re: [Qemu-devel] [PATCH 10/25] audio: -audiodev command line option

Hi

Quoted 2 lines · Zoltán

This patch adds an -audiodev command line option, and deprecates the QEMU_* environment variables for audio backend configuration. It's

"its" or simply "the"

Quoted 9 lines

syntax is similar to existing options (-netdev, -device, etc): -audiodev driver_name,property=value,...

Audio drivers now get an Audiodev * as config paramters, instead of the global audio_option structs. There is some code in audio/audio_legacy.c that converts the old environment variables to audiodev options (this way backends do not have to worry about legacy options). It also contains a replacement of -audio-help, which prints out the equivalent -audiodev based config of the currently specified environment variables.

I guess the option should be deprecated though, perhaps not even visible in -help.

Quoted 18 lines

Although now it's possible to specify multiple -audiodev options on command line, multiple audio backends are not supported yet.

---
 audio/Makefile.objs     |   2 +-
 audio/alsaaudio.c       | 311 ++++++--------------
 audio/audio.c           | 760 ++++++++++++++----------------------------------
 audio/audio.h           |  23 +-
 audio/audio_int.h       |  23 +-
 audio/audio_legacy.c    | 328 +++++++++++++++++++++
 audio/audio_template.h  |  13 +-
 audio/coreaudio.c       |  49 +---
 audio/dsound_template.h |   6 +-
 audio/dsoundaudio.c     |  60 ++--
 audio/noaudio.c         |   3 +-
 audio/ossaudio.c        | 155 +++-------
Quoted 9 lines
 audio/paaudio.c         |  82 ++----
 audio/sdlaudio.c        |  24 +-
 audio/spiceaudio.c      |   7 +-
 audio/wavaudio.c        |  60 +---
 qemu-options.hx         | 226 +++++++++++++-
 vl.c                    |  10 +-
 18 files changed, 1015 insertions(+), 1127 deletions(-)
 create mode 100644 audio/audio_legacy.c

Quite a large patch, perhaps it could be splitted?

Quoted 88 lines
audio/Makefile.objs +1 -1
index 481d1aa..9d8f579 100644
@@ -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
audio/alsaaudio.c +5 -24
index 2b28b99..cfe4aec 100644
@@ -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);

audio/alsaaudio.c static void alsa_dump_info (struct alsa_params_req *req, · line 403

Quoted 8 lines
audio/alsaaudio.c +3 -2
@@ -417,8 +398,9 @@ static void alsa_dump_info (struct alsa_params_req *req,
            req->nchannels, obt->nchannels);
     dolog ("frequency |      %10d |     %10d\n", req->freq, obt->freq);
     dolog ("============================================\n");
-    dolog ("requested: buffer size %d period size %d\n",
-           req->buffer_size, req->period_size);
+    dolog ("requested: buffer size %" PRId64 " buffer count %" PRId64 "\n",
+           pdo->has_buffer_len ? pdo->buffer_len : 0,
+           pdo->has_buffer_len ? pdo->buffer_len : 0);

buffer size & buffer count are both buffer_len here, you should fix this

Quoted 314 lines
audio/alsaaudio.c +55 -136
     dolog ("obtained: samples %ld\n", obt->samples);
 }
 
@@ -452,23 +434,25 @@ static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
     }
 }
 
-static int alsa_open (int in, struct alsa_params_req *req,
-                      struct alsa_params_obt *obt, snd_pcm_t **handlep,
-                      ALSAConf *conf)
+static int alsa_open(bool in, struct alsa_params_req *req,
+                     struct alsa_params_obt *obt, snd_pcm_t **handlep,
+                     Audiodev *dev)
 {
+    AudiodevPerDirectionOptions *pdo = in ? dev->in : dev->out;
+    AudiodevAlsaOptions *aopts = dev->alsa;
+    AudiodevAlsaPerDirectionOptions *apdo =
+        in ? aopts->alsa_in : aopts->alsa_out;
     snd_pcm_t *handle;
     snd_pcm_hw_params_t *hw_params;
     int err;
-    int size_in_usec;
     unsigned int freq, nchannels;
-    const char *pcm_name = in ? conf->pcm_name_in : conf->pcm_name_out;
+    const char *pcm_name = apdo->has_dev ? apdo->dev : "default";
     snd_pcm_uframes_t obt_buffer_size;
     const char *typ = in ? "ADC" : "DAC";
     snd_pcm_format_t obtfmt;
 
     freq = req->freq;
     nchannels = req->nchannels;
-    size_in_usec = req->size_in_usec;
 
     snd_pcm_hw_params_alloca (&hw_params);
 
@@ -528,79 +512,49 @@ static int alsa_open (int in, struct alsa_params_req *req,
         goto err;
     }
 
-    if (req->buffer_size) {
-        unsigned long obt;
+    if (pdo->buffer_count) {
+        if (pdo->buffer_len) {
+            int64_t req = pdo->buffer_len * pdo->buffer_count;
 
-        if (size_in_usec) {
             int dir = 0;
-            unsigned int btime = req->buffer_size;
+            unsigned int btime = req;
 
-            err = snd_pcm_hw_params_set_buffer_time_near (
-                handle,
-                hw_params,
-                &btime,
-                &dir
-                );
-            obt = btime;
-        }
-        else {
-            snd_pcm_uframes_t bsize = req->buffer_size;
+            err = snd_pcm_hw_params_set_buffer_time_near(
+                handle, hw_params, &btime, &dir);
 
-            err = snd_pcm_hw_params_set_buffer_size_near (
-                handle,
-                hw_params,
-                &bsize
-                );
-            obt = bsize;
-        }
-        if (err < 0) {
-            alsa_logerr2 (err, typ, "Failed to set buffer %s to %d\n",
-                          size_in_usec ? "time" : "size", req->buffer_size);
-            goto err;
-        }
+            if (err < 0) {
+                alsa_logerr2(err, typ,
+                             "Failed to set buffer time to %" PRId64 "\n",
+                             req);
+                goto err;
+            }
 
-        if ((req->override_mask & 2) && (obt - req->buffer_size))
-            dolog ("Requested buffer %s %u was rejected, using %lu\n",
-                   size_in_usec ? "time" : "size", req->buffer_size, obt);
+            if (pdo->has_buffer_count && btime != req) {
+                dolog("Requested buffer time %" PRId64
+                      " was rejected, using %u\n", req, btime);
+            }
+        } else {
+            dolog("Can't set buffer_count without buffer_size!\n");
+        }
     }
 
-    if (req->period_size) {
-        unsigned long obt;
+    if (pdo->buffer_len) {
+        int dir = 0;
+        unsigned int ptime = pdo->buffer_len;
 
-        if (size_in_usec) {
-            int dir = 0;
-            unsigned int ptime = req->period_size;
-
-            err = snd_pcm_hw_params_set_period_time_near (
-                handle,
-                hw_params,
-                &ptime,
-                &dir
-                );
-            obt = ptime;
-        }
-        else {
-            int dir = 0;
-            snd_pcm_uframes_t psize = req->period_size;
-
-            err = snd_pcm_hw_params_set_period_size_near (
-                handle,
-                hw_params,
-                &psize,
-                &dir
-                );
-            obt = psize;
-        }
+        err = snd_pcm_hw_params_set_period_time_near(handle, hw_params, &ptime,
+                                                     &dir);
 
         if (err < 0) {
-            alsa_logerr2 (err, typ, "Failed to set period %s to %d\n",
-                          size_in_usec ? "time" : "size", req->period_size);
+            alsa_logerr2(err, typ, "Failed to set period time to %" PRId64 "\n",
+                         pdo->buffer_len);
             goto err;
         }
 
-        if (((req->override_mask & 1) && (obt - req->period_size)))
-            dolog ("Requested period %s %u was rejected, using %lu\n",
-                   size_in_usec ? "time" : "size", req->period_size, obt);
+        if (pdo->has_buffer_len && ptime != pdo->buffer_len) {
+            dolog("Requested period time %" PRId64 " was rejected, using %d\n",
+                  pdo->buffer_len, ptime);
+        }
     }
 
     err = snd_pcm_hw_params (handle, hw_params);
@@ -632,33 +586,10 @@ static int alsa_open (int in, struct alsa_params_req *req,
         goto err;
     }
 
-    if (!in && conf->threshold) {
-        snd_pcm_uframes_t threshold;
-        int bytes_per_sec;
-
-        bytes_per_sec = freq << (nchannels == 2);
-
-        switch (obt->fmt) {
-        case AUDIO_FORMAT_S8:
-        case AUDIO_FORMAT_U8:
-            break;
-
-        case AUDIO_FORMAT_S16:
-        case AUDIO_FORMAT_U16:
-            bytes_per_sec <<= 1;
-            break;
-
-        case AUDIO_FORMAT_S32:
-        case AUDIO_FORMAT_U32:
-            bytes_per_sec <<= 2;
-            break;
-
-        default:
-            abort();
-        }
-
-        threshold = (conf->threshold * bytes_per_sec) / 1000;
-        alsa_set_threshold (handle, threshold);
+    if (!in && aopts->has_threshold && aopts->threshold) {
+        struct audsettings as = { .freq = freq };
+        alsa_set_threshold(handle,
+                           audio_buffer_frames(pdo, &as, aopts->threshold));
     }
 
     obt->nchannels = nchannels;
@@ -671,11 +602,11 @@ static int alsa_open (int in, struct alsa_params_req *req,
          obt->nchannels != req->nchannels ||
          obt->freq != req->freq) {
         dolog ("Audio parameters for %s\n", typ);
-        alsa_dump_info (req, obt, obtfmt);
+        alsa_dump_info (req, obt, obtfmt, pdo);
     }
 
 #ifdef DEBUG
-    alsa_dump_info (req, obt, obtfmt);
+    alsa_dump_info (req, obt, obtfmt, pdo);
 #endif
     return 0;
 
@@ -801,19 +732,13 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
     struct alsa_params_obt obt;
     snd_pcm_t *handle;
     struct audsettings obt_as;
-    ALSAConf *conf = drv_opaque;
+    Audiodev *dev = drv_opaque;
 
     req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
     req.freq = as->freq;
     req.nchannels = as->nchannels;
-    req.period_size = conf->period_size_out;
-    req.buffer_size = conf->buffer_size_out;
-    req.size_in_usec = conf->size_in_usec_out;
-    req.override_mask =
-        (conf->period_size_out_overridden ? 1 : 0) |
-        (conf->buffer_size_out_overridden ? 2 : 0);
 
-    if (alsa_open (0, &req, &obt, &handle, conf)) {
+    if (alsa_open (0, &req, &obt, &handle, dev)) {
         return -1;
     }
 
@@ -834,7 +759,7 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
     }
 
     alsa->handle = handle;
-    alsa->pollhlp.conf = conf;
+    alsa->dev = dev;
     return 0;
 }
 
@@ -874,16 +799,12 @@ static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl)
 static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
 {
     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+    AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->alsa->alsa_out;
 
     switch (cmd) {
     case VOICE_ENABLE:
         {
-            va_list ap;
-            int poll_mode;
-
-            va_start (ap, cmd);
-            poll_mode = va_arg (ap, int);
-            va_end (ap);
+            bool poll_mode = !apdo->has_try_poll || apdo->try_poll;
 
             ldebug ("enabling voice\n");
             if (poll_mode && alsa_poll_out (hw)) {
@@ -912,19 +833,13 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
     struct alsa_params_obt obt;
     snd_pcm_t *handle;
     struct audsettings obt_as;
-    ALSAConf *conf = drv_opaque;
+    Audiodev *dev = drv_opaque;
 
     req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
     req.freq = as->freq;
     req.nchannels = as->nchannels;
-    req.period_size = conf->period_size_in;
-    req.buffer_size = conf->buffer_size_in;
-    req.size_in_usec = conf->size_in_usec_in;
-    req.override_mask =
-        (conf->period_size_in_overridden ? 1 : 0) |
-        (conf->buffer_size_in_overridden ? 2 : 0);
 
-    if (alsa_open (1, &req, &obt, &handle, conf)) {
+    if (alsa_open (1, &req, &obt, &handle, dev)) {
         return -1;
     }
 
@@ -945,7 +860,7 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
     }
 
     alsa->handle = handle;
-    alsa->pollhlp.conf = conf;
+    alsa->dev = dev;
     return 0;
 }
 
@@ -1087,16 +1002,12 @@ static int alsa_read (SWVoiceIn *sw, void *buf, int size)
 static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
 {
     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+    AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->alsa->alsa_in;
 
     switch (cmd) {
     case VOICE_ENABLE:
         {
-            va_list ap;
-            int poll_mode;
-
-            va_start (ap, cmd);
-            poll_mode = va_arg (ap, int);
-            va_end (ap);
+            bool poll_mode = !apdo->has_try_poll || apdo->try_poll;
 
             ldebug ("enabling voice\n");
             if (poll_mode && alsa_poll_in (hw)) {
@@ -1119,88 +1030,35 @@ static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
     return -1;
 }
 
-static ALSAConf glob_conf = {
-    .buffer_size_out = 4096,
-    .period_size_out = 1024,
-    .pcm_name_out = "default",
-    .pcm_name_in = "default",
-};
-
-static void *alsa_audio_init (void)
+static void *alsa_audio_init(Audiodev *dev)
 {
-    ALSAConf *conf = g_malloc(sizeof(ALSAConf));
-    *conf = glob_conf;
-    return conf;

audio/alsaaudio.c static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...) · line 1044

Quoted 10 lines
audio/alsaaudio.c +10 -0
+    assert(dev->kind == AUDIODEV_DRIVER_ALSA);
+
+    /* need to define them, as otherwise alsa produces no sound
+     * doesn't set has_* so alsa_open can identify it wasn't set by the user */
+    if (!dev->out->has_buffer_count) {
+        dev->out->buffer_count = 4;
+    }
+    if (!dev->out->has_buffer_len) {
+        dev->out->buffer_len = 23219; /* 1024 frames assuming 44100Hz */
+    }

how did you compute that? Would be worth leaving that in the code.

Quoted 1,263 lines
audio/alsaaudio.c +11 -65
+
+    /* OptsVisitor sets unspecified optional fields to zero, but do not depend
+     * on it... */
+    if (!dev->in->has_buffer_count) {
+        dev->in->buffer_count = 0;
+    }
+    if (!dev->in->has_buffer_len) {
+        dev->in->buffer_len = 0;
+    }
+
+    return dev;
 }
 
 static void alsa_audio_fini (void *opaque)
 {
-    g_free(opaque);
 }
 
-static struct audio_option alsa_options[] = {
-    {
-        .name        = "DAC_SIZE_IN_USEC",
-        .tag         = AUD_OPT_BOOL,
-        .valp        = &glob_conf.size_in_usec_out,
-        .descr       = "DAC period/buffer size in microseconds (otherwise in frames)"
-    },
-    {
-        .name        = "DAC_PERIOD_SIZE",
-        .tag         = AUD_OPT_INT,
-        .valp        = &glob_conf.period_size_out,
-        .descr       = "DAC period size (0 to go with system default)",
-        .overriddenp = &glob_conf.period_size_out_overridden
-    },
-    {
-        .name        = "DAC_BUFFER_SIZE",
-        .tag         = AUD_OPT_INT,
-        .valp        = &glob_conf.buffer_size_out,
-        .descr       = "DAC buffer size (0 to go with system default)",
-        .overriddenp = &glob_conf.buffer_size_out_overridden
-    },
-    {
-        .name        = "ADC_SIZE_IN_USEC",
-        .tag         = AUD_OPT_BOOL,
-        .valp        = &glob_conf.size_in_usec_in,
-        .descr       =
-        "ADC period/buffer size in microseconds (otherwise in frames)"
-    },
-    {
-        .name        = "ADC_PERIOD_SIZE",
-        .tag         = AUD_OPT_INT,
-        .valp        = &glob_conf.period_size_in,
-        .descr       = "ADC period size (0 to go with system default)",
-        .overriddenp = &glob_conf.period_size_in_overridden
-    },
-    {
-        .name        = "ADC_BUFFER_SIZE",
-        .tag         = AUD_OPT_INT,
-        .valp        = &glob_conf.buffer_size_in,
-        .descr       = "ADC buffer size (0 to go with system default)",
-        .overriddenp = &glob_conf.buffer_size_in_overridden
-    },
-    {
-        .name        = "THRESHOLD",
-        .tag         = AUD_OPT_INT,
-        .valp        = &glob_conf.threshold,
-        .descr       = "(undocumented)"
-    },
-    {
-        .name        = "DAC_DEV",
-        .tag         = AUD_OPT_STR,
-        .valp        = &glob_conf.pcm_name_out,
-        .descr       = "DAC device name (for instance dmix)"
-    },
-    {
-        .name        = "ADC_DEV",
-        .tag         = AUD_OPT_STR,
-        .valp        = &glob_conf.pcm_name_in,
-        .descr       = "ADC device name"
-    },
-    { /* End of list */ }
-};
-
 static struct audio_pcm_ops alsa_pcm_ops = {
     .init_out = alsa_init_out,
     .fini_out = alsa_fini_out,
@@ -1218,7 +1076,6 @@ static struct audio_pcm_ops alsa_pcm_ops = {
 struct audio_driver alsa_audio_driver = {
     .name           = "alsa",
     .descr          = "ALSA http://www.alsa-project.org",
-    .options        = alsa_options,
     .init           = alsa_audio_init,
     .fini           = alsa_audio_fini,
     .pcm_ops        = &alsa_pcm_ops,
audio/audio.c +217 -543
index 334c935..08ac15e 100644
@@ -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 [email protected]\n");
             AUD_log (NULL, "I am sorry\n");
-            d = glob_audio_state.drv;
-            if (d) {
-                audio_print_options (d->name, d->options);
-            }
         }
         AUD_log (NULL, "Context:\n");
 
@@ -194,139 +142,6 @@ void *audio_calloc (const char *funcname, int nmemb, size_t size)
     return g_malloc0 (len);
 }
 
-static char *audio_alloc_prefix (const char *s)
-{
-    const char qemu_prefix[] = "QEMU_";
-    size_t len, i;
-    char *r, *u;
-
-    if (!s) {
-        return NULL;
-    }
-
-    len = strlen (s);
-    r = g_malloc (len + sizeof (qemu_prefix));
-
-    u = r + sizeof (qemu_prefix) - 1;
-
-    pstrcpy (r, len + sizeof (qemu_prefix), qemu_prefix);
-    pstrcat (r, len + sizeof (qemu_prefix), s);
-
-    for (i = 0; i < len; ++i) {
-        u[i] = qemu_toupper(u[i]);
-    }
-
-    return r;
-}
-
-static const char *audio_audfmt_to_string (AudioFormat fmt)
-{
-    switch (fmt) {
-    case AUDIO_FORMAT_U8:
-        return "U8";
-
-    case AUDIO_FORMAT_U16:
-        return "U16";
-
-    case AUDIO_FORMAT_S8:
-        return "S8";
-
-    case AUDIO_FORMAT_S16:
-        return "S16";
-
-    case AUDIO_FORMAT_U32:
-        return "U32";
-
-    case AUDIO_FORMAT_S32:
-        return "S32";
-
-    default:
-        abort();
-    }
-
-    dolog ("Bogus audfmt %d returning S16\n", fmt);
-    return "S16";
-}
-
-static AudioFormat audio_string_to_audfmt (const char *s, AudioFormat defval,
-                                        int *defaultp)
-{
-    if (!strcasecmp (s, "u8")) {
-        *defaultp = 0;
-        return AUDIO_FORMAT_U8;
-    }
-    else if (!strcasecmp (s, "u16")) {
-        *defaultp = 0;
-        return AUDIO_FORMAT_U16;
-    }
-    else if (!strcasecmp (s, "u32")) {
-        *defaultp = 0;
-        return AUDIO_FORMAT_U32;
-    }
-    else if (!strcasecmp (s, "s8")) {
-        *defaultp = 0;
-        return AUDIO_FORMAT_S8;
-    }
-    else if (!strcasecmp (s, "s16")) {
-        *defaultp = 0;
-        return AUDIO_FORMAT_S16;
-    }
-    else if (!strcasecmp (s, "s32")) {
-        *defaultp = 0;
-        return AUDIO_FORMAT_S32;
-    }
-    else {
-        dolog ("Bogus audio format `%s' using %s\n",
-               s, audio_audfmt_to_string (defval));
-        *defaultp = 1;
-        return defval;
-    }
-}
-
-static AudioFormat audio_get_conf_fmt (const char *envname,
-                                    AudioFormat defval,
-                                    int *defaultp)
-{
-    const char *var = getenv (envname);
-    if (!var) {
-        *defaultp = 1;
-        return defval;
-    }
-    return audio_string_to_audfmt (var, defval, defaultp);
-}
-
-static int audio_get_conf_int (const char *key, int defval, int *defaultp)
-{
-    int val;
-    char *strval;
-
-    strval = getenv (key);
-    if (strval) {
-        *defaultp = 0;
-        val = atoi (strval);
-        return val;
-    }
-    else {
-        *defaultp = 1;
-        return defval;
-    }
-}
-
-static const char *audio_get_conf_str (const char *key,
-                                       const char *defval,
-                                       int *defaultp)
-{
-    const char *val = getenv (key);
-    if (!val) {
-        *defaultp = 1;
-        return defval;
-    }
-    else {
-        *defaultp = 0;
-        return val;
-    }
-}
-
 void AUD_vlog (const char *cap, const char *fmt, va_list ap)
 {
     if (cap) {
@@ -345,161 +160,6 @@ void AUD_log (const char *cap, const char *fmt, ...)
     va_end (ap);
 }
 
-static void audio_print_options (const char *prefix,
-                                 struct audio_option *opt)
-{
-    char *uprefix;
-
-    if (!prefix) {
-        dolog ("No prefix specified\n");
-        return;
-    }
-
-    if (!opt) {
-        dolog ("No options\n");
-        return;
-    }
-
-    uprefix = audio_alloc_prefix (prefix);
-
-    for (; opt->name; opt++) {
-        const char *state = "default";
-        printf ("  %s_%s: ", uprefix, opt->name);
-
-        if (opt->overriddenp && *opt->overriddenp) {
-            state = "current";
-        }
-
-        switch (opt->tag) {
-        case AUD_OPT_BOOL:
-            {
-                int *intp = opt->valp;
-                printf ("boolean, %s = %d\n", state, *intp ? 1 : 0);
-            }
-            break;
-
-        case AUD_OPT_INT:
-            {
-                int *intp = opt->valp;
-                printf ("integer, %s = %d\n", state, *intp);
-            }
-            break;
-
-        case AUD_OPT_FMT:
-            {
-                AudioFormat *fmtp = opt->valp;
-                printf (
-                    "format, %s = %s, (one of: U8 S8 U16 S16 U32 S32)\n",
-                    state,
-                    audio_audfmt_to_string (*fmtp)
-                    );
-            }
-            break;
-
-        case AUD_OPT_STR:
-            {
-                const char **strp = opt->valp;
-                printf ("string, %s = %s\n",
-                        state,
-                        *strp ? *strp : "(not set)");
-            }
-            break;
-
-        default:
-            printf ("???\n");
-            dolog ("Bad value tag for option %s_%s %d\n",
-                   uprefix, opt->name, opt->tag);
-            break;
-        }
-        printf ("    %s\n", opt->descr);
-    }
-
-    g_free (uprefix);
-}
-
-static void audio_process_options (const char *prefix,
-                                   struct audio_option *opt)
-{
-    char *optname;
-    const char qemu_prefix[] = "QEMU_";
-    size_t preflen, optlen;
-
-    if (audio_bug (AUDIO_FUNC, !prefix)) {
-        dolog ("prefix = NULL\n");
-        return;
-    }
-
-    if (audio_bug (AUDIO_FUNC, !opt)) {
-        dolog ("opt = NULL\n");
-        return;
-    }
-
-    preflen = strlen (prefix);
-
-    for (; opt->name; opt++) {
-        size_t len, i;
-        int def;
-
-        if (!opt->valp) {
-            dolog ("Option value pointer for `%s' is not set\n",
-                   opt->name);
-            continue;
-        }
-
-        len = strlen (opt->name);
-        /* len of opt->name + len of prefix + size of qemu_prefix
-         * (includes trailing zero) + zero + underscore (on behalf of
-         * sizeof) */
-        optlen = len + preflen + sizeof (qemu_prefix) + 1;
-        optname = g_malloc (optlen);
-
-        pstrcpy (optname, optlen, qemu_prefix);
-
-        /* copy while upper-casing, including trailing zero */
-        for (i = 0; i <= preflen; ++i) {
-            optname[i + sizeof (qemu_prefix) - 1] = qemu_toupper(prefix[i]);
-        }
-        pstrcat (optname, optlen, "_");
-        pstrcat (optname, optlen, opt->name);
-
-        def = 1;
-        switch (opt->tag) {
-        case AUD_OPT_BOOL:
-        case AUD_OPT_INT:
-            {
-                int *intp = opt->valp;
-                *intp = audio_get_conf_int (optname, *intp, &def);
-            }
-            break;
-
-        case AUD_OPT_FMT:
-            {
-                AudioFormat *fmtp = opt->valp;
-                *fmtp = audio_get_conf_fmt (optname, *fmtp, &def);
-            }
-            break;
-
-        case AUD_OPT_STR:
-            {
-                const char **strp = opt->valp;
-                *strp = audio_get_conf_str (optname, *strp, &def);
-            }
-            break;
-
-        default:
-            dolog ("Bad value tag for option `%s' - %d\n",
-                   optname, opt->tag);
-            break;
-        }
-
-        if (!opt->overriddenp) {
-            opt->overriddenp = &opt->overridden;
-        }
-        *opt->overriddenp = !def;
-        g_free (optname);
-    }
-}
-
 static void audio_print_settings (struct audsettings *as)
 {
     dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels);
@@ -1120,7 +780,7 @@ static void audio_reset_timer (AudioState *s)
 {
     if (audio_is_timer_needed ()) {
         timer_mod (s->ts,
-            qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + conf.period.ticks);
+            qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + s->period_ticks);
     }
     else {
         timer_del (s->ts);
@@ -1196,7 +856,7 @@ void AUD_set_active_out (SWVoiceOut *sw, int on)
             if (!hw->enabled) {
                 hw->enabled = 1;
                 if (s->vm_running) {
-                    hw->pcm_ops->ctl_out (hw, VOICE_ENABLE, conf.try_poll_out);
+                    hw->pcm_ops->ctl_out (hw, VOICE_ENABLE);
                     audio_reset_timer (s);
                 }
             }
@@ -1241,7 +901,7 @@ void AUD_set_active_in (SWVoiceIn *sw, int on)
             if (!hw->enabled) {
                 hw->enabled = 1;
                 if (s->vm_running) {
-                    hw->pcm_ops->ctl_in (hw, VOICE_ENABLE, conf.try_poll_in);
+                    hw->pcm_ops->ctl_in (hw, VOICE_ENABLE);
                     audio_reset_timer (s);
                 }
             }
@@ -1558,168 +1218,10 @@ void audio_run (const char *msg)
 #endif
 }
 
-static struct audio_option audio_options[] = {
-    /* DAC */
-    {
-        .name  = "DAC_FIXED_SETTINGS",
-        .tag   = AUD_OPT_BOOL,
-        .valp  = &conf.fixed_out.enabled,
-        .descr = "Use fixed settings for host DAC"
-    },
-    {
-        .name  = "DAC_FIXED_FREQ",
-        .tag   = AUD_OPT_INT,
-        .valp  = &conf.fixed_out.settings.freq,
-        .descr = "Frequency for fixed host DAC"
-    },
-    {
-        .name  = "DAC_FIXED_FMT",
-        .tag   = AUD_OPT_FMT,
-        .valp  = &conf.fixed_out.settings.fmt,
-        .descr = "Format for fixed host DAC"
-    },
-    {
-        .name  = "DAC_FIXED_CHANNELS",
-        .tag   = AUD_OPT_INT,
-        .valp  = &conf.fixed_out.settings.nchannels,
-        .descr = "Number of channels for fixed DAC (1 - mono, 2 - stereo)"
-    },
-    {
-        .name  = "DAC_VOICES",
-        .tag   = AUD_OPT_INT,
-        .valp  = &conf.fixed_out.nb_voices,
-        .descr = "Number of voices for DAC"
-    },
-    {
-        .name  = "DAC_TRY_POLL",
-        .tag   = AUD_OPT_BOOL,
-        .valp  = &conf.try_poll_out,
-        .descr = "Attempt using poll mode for DAC"
-    },
-    /* ADC */
-    {
-        .name  = "ADC_FIXED_SETTINGS",
-        .tag   = AUD_OPT_BOOL,
-        .valp  = &conf.fixed_in.enabled,
-        .descr = "Use fixed settings for host ADC"
-    },
-    {
-        .name  = "ADC_FIXED_FREQ",
-        .tag   = AUD_OPT_INT,
-        .valp  = &conf.fixed_in.settings.freq,
-        .descr = "Frequency for fixed host ADC"
-    },
-    {
-        .name  = "ADC_FIXED_FMT",
-        .tag   = AUD_OPT_FMT,
-        .valp  = &conf.fixed_in.settings.fmt,
-        .descr = "Format for fixed host ADC"
-    },
-    {
-        .name  = "ADC_FIXED_CHANNELS",
-        .tag   = AUD_OPT_INT,
-        .valp  = &conf.fixed_in.settings.nchannels,
-        .descr = "Number of channels for fixed ADC (1 - mono, 2 - stereo)"
-    },
-    {
-        .name  = "ADC_VOICES",
-        .tag   = AUD_OPT_INT,
-        .valp  = &conf.fixed_in.nb_voices,
-        .descr = "Number of voices for ADC"
-    },
-    {
-        .name  = "ADC_TRY_POLL",
-        .tag   = AUD_OPT_BOOL,
-        .valp  = &conf.try_poll_in,
-        .descr = "Attempt using poll mode for ADC"
-    },
-    /* Misc */
-    {
-        .name  = "TIMER_PERIOD",
-        .tag   = AUD_OPT_INT,
-        .valp  = &conf.period.hertz,
-        .descr = "Timer period in HZ (0 - use lowest possible)"
-    },
-    { /* End of list */ }
-};
-
-static void audio_pp_nb_voices (const char *typ, int nb)
+static int audio_driver_init(AudioState *s, struct audio_driver *drv,
+                             Audiodev *dev)
 {
-    switch (nb) {
-    case 0:
-        printf ("Does not support %s\n", typ);
-        break;
-    case 1:
-        printf ("One %s voice\n", typ);
-        break;
-    case INT_MAX:
-        printf ("Theoretically supports many %s voices\n", typ);
-        break;
-    default:
-        printf ("Theoretically supports up to %d %s voices\n", nb, typ);
-        break;
-    }
-
-}
-
-void AUD_help (void)
-{
-    size_t i;
-
-    audio_process_options ("AUDIO", audio_options);
-    for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
-        struct audio_driver *d = drvtab[i];
-        if (d->options) {
-            audio_process_options (d->name, d->options);
-        }
-    }
-
-    printf ("Audio options:\n");
-    audio_print_options ("AUDIO", audio_options);
-    printf ("\n");
-
-    printf ("Available drivers:\n");
-
-    for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
-        struct audio_driver *d = drvtab[i];
-
-        printf ("Name: %s\n", d->name);
-        printf ("Description: %s\n", d->descr);
-
-        audio_pp_nb_voices ("playback", d->max_voices_out);
-        audio_pp_nb_voices ("capture", d->max_voices_in);
-
-        if (d->options) {
-            printf ("Options:\n");
-            audio_print_options (d->name, d->options);
-        }
-        else {
-            printf ("No options\n");
-        }
-        printf ("\n");
-    }
-
-    printf (
-        "Options are settable through environment variables.\n"
-        "Example:\n"
-#ifdef _WIN32
-        "  set QEMU_AUDIO_DRV=wav\n"
-        "  set QEMU_WAV_PATH=c:\\tune.wav\n"
-#else
-        "  export QEMU_AUDIO_DRV=wav\n"
-        "  export QEMU_WAV_PATH=$HOME/tune.wav\n"
-        "(for csh replace export with setenv in the above)\n"
-#endif
-        "  qemu ...\n\n"
-        );
-}
-
-static int audio_driver_init (AudioState *s, struct audio_driver *drv)
-{
-    if (drv->options) {
-        audio_process_options (drv->name, drv->options);
-    }
-    s->drv_opaque = drv->init ();
+    s->drv_opaque = drv->init(dev);
 
     if (s->drv_opaque) {
         audio_init_nb_voices_out (drv);
@@ -1743,11 +1245,11 @@ static void audio_vm_change_state_handler (void *opaque, int running,
 
     s->vm_running = running;
     while ((hwo = audio_pcm_hw_find_any_enabled_out (hwo))) {
-        hwo->pcm_ops->ctl_out (hwo, op, conf.try_poll_out);
+        hwo->pcm_ops->ctl_out (hwo, op);
     }
 
     while ((hwi = audio_pcm_hw_find_any_enabled_in (hwi))) {
-        hwi->pcm_ops->ctl_in (hwi, op, conf.try_poll_in);
+        hwi->pcm_ops->ctl_in (hwi, op);
     }
     audio_reset_timer (s);
 }
@@ -1786,6 +1288,8 @@ static void audio_atexit (void)
     if (s->drv) {
         s->drv->fini (s->drv_opaque);
     }
+
+    qapi_free_Audiodev(s->dev);
 }
 
 static const VMStateDescription vmstate_audio = {
@@ -1797,18 +1301,37 @@ static const VMStateDescription vmstate_audio = {
     }
 };
 
-static void audio_init (void)
+static Audiodev *parse_option(QemuOpts *opts, Error **errp);
+static int audio_init(Audiodev *dev)
 {
     size_t i;
     int done = 0;
-    const char *drvname;
+    const char *drvname = NULL;
     VMChangeStateEntry *e;
     AudioState *s = &glob_audio_state;
+    QemuOptsList *list = NULL; /* silence gcc warning about uninitialized
+                                * variable */
 
     if (s->drv) {
-        return;
+        if (dev) {
+            dolog("Cannot create more than one audio backend, sorry\n");
+            qapi_free_Audiodev(dev);
+        }
+        return -1;
     }
 
+    if (dev) {
+        drvname = AudiodevDriver_lookup[dev->kind];
+    } else {
+        audio_handle_legacy_opts();
+        list = qemu_find_opts("audiodev");
+        dev = parse_option(QTAILQ_FIRST(&list->head), &error_abort);
+        if (!dev) {
+            exit(1);
+        }
+    }
+    s->dev = dev;
+
     QLIST_INIT (&s->hw_head_out);
     QLIST_INIT (&s->hw_head_in);
     QLIST_INIT (&s->cap_head);
@@ -1819,10 +1342,8 @@ static void audio_init (void)
         hw_error("Could not create audio timer\n");
     }
 
-    audio_process_options ("AUDIO", audio_options);
-
-    s->nb_hw_voices_out = conf.fixed_out.nb_voices;
-    s->nb_hw_voices_in = conf.fixed_in.nb_voices;
+    s->nb_hw_voices_out = dev->out->voices;
+    s->nb_hw_voices_in = dev->in->voices;
 
     if (s->nb_hw_voices_out <= 0) {
         dolog ("Bogus number of playback voices %d, setting to 1\n",
@@ -1836,17 +1357,12 @@ static void audio_init (void)
         s->nb_hw_voices_in = 0;
     }
 
-    {
-        int def;
-        drvname = audio_get_conf_str ("QEMU_AUDIO_DRV", NULL, &def);
-    }
-
     if (drvname) {
         int found = 0;
 
-        for (i = 0; i < ARRAY_SIZE (drvtab); i++) {
+        for (i = 0; drvtab[i]; i++) {
             if (!strcmp (drvname, drvtab[i]->name)) {
-                done = !audio_driver_init (s, drvtab[i]);
+                done = !audio_driver_init (s, drvtab[i], dev);
                 found = 1;
                 break;
             }
@@ -1854,20 +1370,24 @@ static void audio_init (void)
 
         if (!found) {
             dolog ("Unknown audio driver `%s'\n", drvname);
-            dolog ("Run with -audio-help to list available drivers\n");
         }
-    }
-
-    if (!done) {
-        for (i = 0; !done && i < ARRAY_SIZE (drvtab); i++) {
-            if (drvtab[i]->can_be_default) {
-                done = !audio_driver_init (s, drvtab[i]);
+    } else {
+        for (i = 0; !done && drvtab[i]; i++) {
+            QemuOpts *opts = qemu_opts_find(list, drvtab[i]->name);
+            if (opts) {
+                qapi_free_Audiodev(dev);
+                dev = parse_option(opts, &error_abort);
+                if (!dev) {
+                    exit(1);
+                }
+                s->dev = dev;
+                done = !audio_driver_init(s, drvtab[i], dev);
             }
         }
     }
 
     if (!done) {
-        done = !audio_driver_init (s, &no_audio_driver);
+        done = !audio_driver_init (s, &no_audio_driver, dev);
         if (!done) {
             hw_error("Could not initialize audio subsystem\n");
         }
@@ -1876,16 +1396,16 @@ static void audio_init (void)
         }
     }
 
-    if (conf.period.hertz <= 0) {
-        if (conf.period.hertz < 0) {
-            dolog ("warning: Timer period is negative - %d "
-                   "treating as zero\n",
-                   conf.period.hertz);
+    if (dev->timer_period <= 0) {
+        if (dev->timer_period < 0) {
+            dolog ("warning: Timer period is negative - %" PRId64
+                   " treating as zero\n",
+                   dev->timer_period);
         }
-        conf.period.ticks = 1;
+        s->period_ticks = 1;
     } else {
-        conf.period.ticks =
-            muldiv64 (1, get_ticks_per_sec (), conf.period.hertz);
+        s->period_ticks =
+            muldiv64(dev->timer_period, get_ticks_per_sec(), 1000000);
     }
 
     e = qemu_add_vm_change_state_handler (audio_vm_change_state_handler, s);
@@ -1896,11 +1416,12 @@ static void audio_init (void)
 
     QLIST_INIT (&s->card_head);
     vmstate_register (NULL, 0, &vmstate_audio, s);
+    return 0;
 }
 
 void AUD_register_card (const char *name, QEMUSoundCard *card)
 {
-    audio_init ();
+    audio_init(NULL);
     card->name = g_strdup (name);
     memset (&card->entries, 0, sizeof (card->entries));
     QLIST_INSERT_HEAD (&glob_audio_state.card_head, card, entries);
@@ -2070,3 +1591,156 @@ 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);
+    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);
+}
audio/audio.h +20 -3
index e300511..177a673 100644
@@ -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 */
audio/audio_int.h +5 -18
index 566df5e..59b2362 100644
@@ -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;
@@ -143,8 +127,7 @@ struct SWVoiceIn {
 struct audio_driver {
     const char *name;
     const char *descr;
-    struct audio_option *options;
-    void *(*init) (void);
+    void *(*init) (Audiodev *);
     void (*fini) (void *);
     struct audio_pcm_ops *pcm_ops;
     int can_be_default;
@@ -190,6 +173,7 @@ struct SWVoiceCap {
 
 struct AudioState {
     struct audio_driver *drv;
+    Audiodev *dev;
     void *drv_opaque;
 
     QEMUTimer *ts;
@@ -200,6 +184,7 @@ 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 +198,8 @@ 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);
 
audio/audio_legacy.c new file +110 -0
new file mode 100644
index 0000000..88f577d
@@ -0,0 +1,328 @@
+#include "audio.h"
+#include "qemu-common.h"
+#include "qemu/config-file.h"
+
+#define AUDIO_CAP "audio-legacy"
+#include "audio_int.h"
+
+typedef enum EnvTransform {
+    ENV_TRANSFORM_NONE,
+    ENV_TRANSFORM_BOOL,
+    ENV_TRANSFORM_FMT,
+    ENV_TRANSFORM_FRAMES_TO_USECS_IN,
+    ENV_TRANSFORM_FRAMES_TO_USECS_OUT,
+    ENV_TRANSFORM_SAMPLES_TO_USECS_IN,
+    ENV_TRANSFORM_SAMPLES_TO_USECS_OUT,
+    ENV_TRANSFORM_BYTES_TO_USECS_IN,
+    ENV_TRANSFORM_BYTES_TO_USECS_OUT,
+    ENV_TRANSFORM_MILLIS_TO_USECS,
+    ENV_TRANSFORM_HZ_TO_USECS,
+} EnvTransform;
+
+typedef struct SimpleEnvMap {
+    const char *name;
+    const char *option;
+    EnvTransform transform;
+} SimpleEnvMap;
+
+SimpleEnvMap global_map[] = {
+    /* DAC/out settings */
+    { "QEMU_AUDIO_DAC_FIXED_SETTINGS", "out.fixed-settings",
+      ENV_TRANSFORM_BOOL },
+    { "QEMU_AUDIO_DAC_FIXED_FREQ", "out.frequency" },
+    { "QEMU_AUDIO_DAC_FIXED_FMT", "out.format", ENV_TRANSFORM_FMT },
+    { "QEMU_AUDIO_DAC_FIXED_CHANNELS", "out.channels" },
+    { "QEMU_AUDIO_DAC_VOICES", "out.voices" },
+
+    /* ADC/in settings */
+    { "QEMU_AUDIO_ADC_FIXED_SETTINGS", "in.fixed-settings",
+      ENV_TRANSFORM_BOOL },
+    { "QEMU_AUDIO_ADC_FIXED_FREQ", "in.frequency" },
+    { "QEMU_AUDIO_ADC_FIXED_FMT", "in.format", ENV_TRANSFORM_FMT },
+    { "QEMU_AUDIO_ADC_FIXED_CHANNELS", "in.channels" },
+    { "QEMU_AUDIO_ADC_VOICES", "in.voices" },
+
+    /* general */
+    { "QEMU_AUDIO_TIMER_PERIOD", "timer-period", ENV_TRANSFORM_HZ_TO_USECS },
+    { /* End of list */ }
+};
+
+SimpleEnvMap alsa_map[] = {
+    { "QEMU_AUDIO_DAC_TRY_POLL", "alsa-out.try-poll", ENV_TRANSFORM_BOOL },
+    { "QEMU_AUDIO_ADC_TRY_POLL", "alsa-in.try-poll", ENV_TRANSFORM_BOOL },
+
+    { "QEMU_ALSA_THRESHOLD", "threshold", ENV_TRANSFORM_MILLIS_TO_USECS },
+    { "QEMU_ALSA_DAC_DEV", "alsa-out.dev" },
+    { "QEMU_ALSA_ADC_DEV", "alsa-in.dev" },
+
+    { /* End of list */ }
+};
+
+SimpleEnvMap coreaudio_map[] = {
+    { "QEMU_COREAUDIO_BUFFER_SIZE", "buffer-len",
+      ENV_TRANSFORM_FRAMES_TO_USECS_OUT },
+    { "QEMU_COREAUDIO_BUFFER_COUNT", "buffer-count" },
+
+    { /* End of list */ }
+};
+
+SimpleEnvMap dsound_map[] = {
+    { "QEMU_DSOUND_LATENCY_MILLIS", "latency", ENV_TRANSFORM_MILLIS_TO_USECS },
+    { "QEMU_DSOUND_BUFSIZE_OUT", "out.buffer-len",
+      ENV_TRANSFORM_BYTES_TO_USECS_OUT },
+    { "QEMU_DSOUND_BUFSIZE_IN", "in.buffer-len",
+      ENV_TRANSFORM_BYTES_TO_USECS_IN },
+
+    { /* End of list */ }
+};
+
+SimpleEnvMap oss_map[] = {
+    { "QEMU_AUDIO_DAC_TRY_POLL", "oss-out.try-poll", ENV_TRANSFORM_BOOL },
+    { "QEMU_AUDIO_ADC_TRY_POLL", "oss-in.try-poll", ENV_TRANSFORM_BOOL },
+
+    { "QEMU_OSS_FRAGSIZE", "buffer-len", ENV_TRANSFORM_BYTES_TO_USECS_OUT },
+    { "QEMU_OSS_NFRAGS", "buffer-count" },
+    { "QEMU_OSS_MMAP", "try-mmap", ENV_TRANSFORM_BOOL },
+    { "QEMU_OSS_DAC_DEV", "oss-out.dev" },
+    { "QEMU_OSS_ADC_DEV", "oss-in.dev" },
+    { "QEMU_OSS_EXCLUSIVE", "exclusive", ENV_TRANSFORM_BOOL },
+    { "QEMU_OSS_POLICY", "dsp-policy" },
+
+    { /* End of list */ }
+};
+
+SimpleEnvMap pa_map[] = {
+    { "QEMU_PA_SAMPLES", "buffer", ENV_TRANSFORM_SAMPLES_TO_USECS_OUT },
+    { "QEMU_PA_SERVER", "server" },
+    { "QEMU_PA_SINK", "sink" },
+    { "QEMU_PA_SOURCE", "source" },
+
+    { /* End of list */ }
+};
+
+SimpleEnvMap sdl_map[] = {
+    { "QEMU_SDL_SAMPLES", "buffer-len", ENV_TRANSFORM_SAMPLES_TO_USECS_OUT },
+    { /* End of list */ }
+};
+
+SimpleEnvMap wav_map[] = {
+    { "QEMU_WAV_FREQUENCY", "out.frequency" },
+    { "QEMU_WAV_FORMAT", "out.format", ENV_TRANSFORM_FMT },

audio/audio_legacy.c line 120

Quoted 10 lines
audio/audio_legacy.c new file +10 -0
+    { "QEMU_WAV_DAC_FIXED_CHANNELS", "out.channels" },
+    { "QEMU_WAV_PATH", "path" },
+    { /* End of list */ }
+};
+
+static unsigned long long toull(const char *str)
+{
+    unsigned long long ret;
+    if (parse_uint_full(str, &ret, 10)) {
+        dolog("Invalid boolean value `%s'\n", str);

integer?

audio/audio_legacy.c line 130

Quoted 10 lines
audio/audio_legacy.c new file +10 -0
+        exit(1);
+    }
+    return ret;
+}
+
+/* non reentrant typesafe or anything, but enough in this small c file */
+static const char *tostr(unsigned long long val)
+{
+    #define LEN ((CHAR_BIT * sizeof(int) - 1) / 3 + 2)
+    static char ret[LEN];

sizeof(int) ?

Quoted 1,322 lines
audio/audio_legacy.c new file +198 -0
+    snprintf(ret, LEN, "%llu", val);
+    return ret;
+}
+
+static uint64_t frames_to_usecs(QemuOpts *opts, uint64_t frames, bool in)
+{
+    const char *opt = in ? "in.frequency" : "out.frequency";
+    uint64_t freq = qemu_opt_get_number(opts, opt, 44100);
+    return (frames * 1000000 + freq/2) / freq;
+}
+
+static uint64_t samples_to_usecs(QemuOpts *opts, uint64_t samples, bool in)
+{
+    const char *opt = in ? "in.channels" : "out.channels";
+    uint64_t channels = qemu_opt_get_number(opts, opt, 2);
+    return frames_to_usecs(opts, samples/channels, in);
+}
+
+static uint64_t bytes_to_usecs(QemuOpts *opts, uint64_t bytes, bool in)
+{
+    const char *opt = in ? "in.format" : "out.format";
+    const char *val = qemu_opt_get(opts, opt);
+    uint64_t bytes_per_sample = (val ? toull(val) : 16) / 8;
+    return samples_to_usecs(opts, bytes * bytes_per_sample, in);
+}
+
+static const char *transform_val(QemuOpts *opts, const char *val,
+                                 EnvTransform transform)
+{
+    switch (transform) {
+    case ENV_TRANSFORM_NONE:
+        return val;
+
+    case ENV_TRANSFORM_BOOL:
+        return toull(val) ? "on" : "off";
+
+    case ENV_TRANSFORM_FMT:
+        if (strcasecmp(val, "u8") == 0) {
+            return "u8";
+        } else if (strcasecmp(val, "u16") == 0) {
+            return "u16";
+        } else if (strcasecmp(val, "u32") == 0) {
+            return "u32";
+        } else if (strcasecmp(val, "s8") == 0) {
+            return "s8";
+        } else if (strcasecmp(val, "s16") == 0) {
+            return "s16";
+        } else if (strcasecmp(val, "s32") == 0) {
+            return "s32";
+        } else {
+            dolog("Invalid audio format `%s'\n", val);
+            exit(1);
+        }
+
+    case ENV_TRANSFORM_FRAMES_TO_USECS_IN:
+        return tostr(frames_to_usecs(opts, toull(val), true));
+    case ENV_TRANSFORM_FRAMES_TO_USECS_OUT:
+        return tostr(frames_to_usecs(opts, toull(val), false));
+
+    case ENV_TRANSFORM_SAMPLES_TO_USECS_IN:
+        return tostr(samples_to_usecs(opts, toull(val), true));
+    case ENV_TRANSFORM_SAMPLES_TO_USECS_OUT:
+        return tostr(samples_to_usecs(opts, toull(val), false));
+
+    case ENV_TRANSFORM_BYTES_TO_USECS_IN:
+        return tostr(bytes_to_usecs(opts, toull(val), true));
+    case ENV_TRANSFORM_BYTES_TO_USECS_OUT:
+        return tostr(bytes_to_usecs(opts, toull(val), false));
+
+    case ENV_TRANSFORM_MILLIS_TO_USECS:
+        return tostr(toull(val) * 1000);
+
+    case ENV_TRANSFORM_HZ_TO_USECS:
+        return tostr(1000000 / toull(val));
+    }
+
+    abort(); /* it's unreachable, gcc */
+}
+
+static void handle_env_opts(QemuOpts *opts, SimpleEnvMap *map)
+{
+    while (map->name) {
+        const char *val = getenv(map->name);
+
+        if (val) {
+            qemu_opt_set(opts, map->option,
+                         transform_val(opts, val, map->transform),
+                         &error_abort);
+        }
+
+        ++map;
+    }
+}
+
+static void handle_alsa_side(QemuOpts *opts, int period, int buffer,
+                             const char *usec_env, const char *period_env,
+                             const char *buffer_env, const char *usec_opt,
+                             const char *count_opt, bool in)
+{
+    char *usec_s, *period_s, *buffer_s;
+    bool usec = false;
+
+    usec_s = getenv(usec_env);
+    if (usec_s) {
+        usec = toull(usec_s);
+    }
+
+    period_s = getenv(period_env);
+    if (period_s) {
+        period = toull(period_s);
+    }
+    if (!usec) {
+        period = frames_to_usecs(opts, period, in);
+    }
+    if (period_s) {
+        qemu_opt_set(opts, usec_opt, tostr(period), &error_abort);
+    }
+
+    buffer_s = getenv(buffer_env);
+    if (buffer_s) {
+        buffer = toull(buffer_s);
+        if (!usec) {
+            buffer = frames_to_usecs(opts, buffer, in);
+        }
+        printf("buffer %d period %d\n", buffer, period);
+        qemu_opt_set(opts, count_opt, tostr((buffer+period/2)/period),
+                     &error_abort);
+    }
+}
+
+static void handle_alsa(QemuOpts *opts)
+{
+    handle_alsa_side(opts, 1024, 4096,
+                     "QEMU_ALSA_DAC_SIZE_IN_USEC", "QEMU_ALSA_DAC_PERIOD_SIZE",
+                     "QEMU_ALSA_DAC_BUFFER_SIZE",
+                     "out.buffer-len", "out.buffer-count", false);
+    handle_alsa_side(opts, 0, 0,
+                     "QEMU_ALSA_ADC_SIZE_IN_USEC", "QEMU_ALSA_ADC_PERIOD_SIZE",
+                     "QEMU_ALSA_ADC_BUFFER_SIZE",
+                     "in.buffer-len", "in.buffer-count", true);
+}
+
+static void legacy_opt(const char *drv)
+{
+    QemuOpts *opts;
+    opts = qemu_opts_create(qemu_find_opts("audiodev"), drv, true,
+                            &error_abort);
+    qemu_opt_set(opts, "driver", drv, &error_abort);
+
+    handle_env_opts(opts, global_map);
+
+    if (strcmp(drv, "alsa") == 0) {
+        handle_env_opts(opts, alsa_map);
+        handle_alsa(opts);
+    } else if (strcmp(drv, "oss") == 0) {
+        handle_env_opts(opts, oss_map);
+    } else if (strcmp(drv, "pa") == 0) {
+        handle_env_opts(opts, pa_map);
+    } else if (strcmp(drv, "sdl") == 0) {
+        handle_env_opts(opts, sdl_map);
+    } else if (strcmp(drv, "wav") == 0) {
+        handle_env_opts(opts, wav_map);
+    }
+}
+
+void audio_handle_legacy_opts(void)
+{
+    const char *drv = getenv("QEMU_AUDIO_DRV");
+
+    if (drv) {
+        legacy_opt(drv);
+    } else {
+        struct audio_driver **drv;
+        for (drv = drvtab; *drv; ++drv) {
+            if ((*drv)->can_be_default) {
+                legacy_opt((*drv)->name);
+            }
+        }
+    }
+}
+
+static int legacy_help_each(void *opaque, QemuOpts *opts, Error **errp)
+{
+    printf("-audiodev ");
+    qemu_opts_print(opts, ",");
+    printf("\n");
+    return 0;
+}
+
+void audio_legacy_help(void)
+{
+    printf("Environment variable based configuration deprecated.\n");
+    printf("Please use the new -audiodev option.\n");
+
+    audio_handle_legacy_opts();
+    printf("\nEquivalent -audiodev to your current environment variables:\n");
+    qemu_opts_foreach(qemu_find_opts("audiodev"), legacy_help_each, NULL, NULL);
+}
audio/audio_template.h +9 -4
index 99b27b2..096b2b3 100644
@@ -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;
     }
audio/coreaudio.c +11 -38
index 6dfd63e..dfa5e79 100644
@@ -34,11 +34,6 @@
 
 static int isAtexit;
 
-typedef struct {
-    int buffer_frames;
-    int nbuffers;
-} CoreaudioConf;
-
 typedef struct coreaudioVoiceOut {
     HWVoiceOut hw;
     pthread_mutex_t mutex;
@@ -292,7 +287,9 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
     int err;
     const char *typ = "playback";
     AudioValueRange frameRange;
-    CoreaudioConf *conf = drv_opaque;
+    Audiodev *dev = drv_opaque;
+    AudiodevPerDirectionOptions *pdo = dev->out;
+    int frames;
 
     /* create mutex */
     err = pthread_mutex_init(&core->mutex, NULL);
@@ -334,16 +331,17 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
         return -1;
     }
 
-    if (frameRange.mMinimum > conf->buffer_frames) {
+    frames = audio_buffer_frames(pdo, as, 11610);
+    if (frameRange.mMinimum > frames) {
         core->audioDevicePropertyBufferFrameSize = (UInt32) frameRange.mMinimum;
         dolog ("warning: Upsizing Buffer Frames to %f\n", frameRange.mMinimum);
     }
-    else if (frameRange.mMaximum < conf->buffer_frames) {
+    else if (frameRange.mMaximum < frames) {
         core->audioDevicePropertyBufferFrameSize = (UInt32) frameRange.mMaximum;
         dolog ("warning: Downsizing Buffer Frames to %f\n", frameRange.mMaximum);
     }
     else {
-        core->audioDevicePropertyBufferFrameSize = conf->buffer_frames;
+        core->audioDevicePropertyBufferFrameSize = frames;
     }
 
     /* set Buffer Frame Size */
@@ -377,7 +375,8 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
                            "Could not get device buffer frame size\n");
         return -1;
     }
-    hw->samples = conf->nbuffers * core->audioDevicePropertyBufferFrameSize;
+    hw->samples = (pdo->has_buffer_count ? pdo->buffer_count : 4) *
+        core->audioDevicePropertyBufferFrameSize;
 
     /* get StreamFormat */
     propertySize = sizeof(core->outputStreamBasicDescription);
@@ -497,41 +496,16 @@ static int coreaudio_ctl_out (HWVoiceOut *hw, int cmd, ...)
     return 0;
 }
 
-static CoreaudioConf glob_conf = {
-    .buffer_frames = 512,
-    .nbuffers = 4,
-};
-
-static void *coreaudio_audio_init (void)
+static void *coreaudio_audio_init(Audiodev *dev)
 {
-    CoreaudioConf *conf = g_malloc(sizeof(CoreaudioConf));
-    *conf = glob_conf;
-
     atexit(coreaudio_atexit);
-    return conf;
+    return dev;
 }
 
 static void coreaudio_audio_fini (void *opaque)
 {
-    g_free(opaque);
 }
 
-static struct audio_option coreaudio_options[] = {
-    {
-        .name  = "BUFFER_SIZE",
-        .tag   = AUD_OPT_INT,
-        .valp  = &glob_conf.buffer_frames,
-        .descr = "Size of the buffer in frames"
-    },
-    {
-        .name  = "BUFFER_COUNT",
-        .tag   = AUD_OPT_INT,
-        .valp  = &glob_conf.nbuffers,
-        .descr = "Number of buffers"
-    },
-    { /* End of list */ }
-};
-
 static struct audio_pcm_ops coreaudio_pcm_ops = {
     .init_out = coreaudio_init_out,
     .fini_out = coreaudio_fini_out,
@@ -543,7 +517,6 @@ static struct audio_pcm_ops coreaudio_pcm_ops = {
 struct audio_driver coreaudio_audio_driver = {
     .name           = "coreaudio",
     .descr          = "CoreAudio http://developer.apple.com/audio/coreaudio.html",
-    .options        = coreaudio_options,
     .init           = coreaudio_audio_init,
     .fini           = coreaudio_audio_fini,
     .pcm_ops        = &coreaudio_pcm_ops,
audio/dsound_template.h +3 -3
index b439f33..96181ef 100644
@@ -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,
audio/dsoundaudio.c +17 -43
index e9472c1..7e7b3f2 100644
@@ -42,16 +42,10 @@
 /* #define DEBUG_DSOUND */
 
 typedef struct {
-    int bufsize_in;
-    int bufsize_out;
-    int latency_millis;
-} DSoundConf;
-
-typedef struct {
     LPDIRECTSOUND dsound;
     LPDIRECTSOUNDCAPTURE dsound_capture;
     struct audsettings settings;
-    DSoundConf conf;
+    Audiodev *dev;
 } dsound;
 
 typedef struct {
@@ -247,9 +241,9 @@ static void GCC_FMT_ATTR (3, 4) dsound_logerr2 (
     dsound_log_hresult (hr);
 }
 
-static DWORD millis_to_bytes (struct audio_pcm_info *info, DWORD millis)
+static uint64_t usecs_to_bytes(struct audio_pcm_info *info, uint32_t usecs)
 {
-    return (millis * info->bytes_per_second) / 1000;
+    return muldiv64(usecs, info->bytes_per_second, 1000000);
 }
 
 #ifdef DEBUG_DSOUND
@@ -477,7 +471,7 @@ static int dsound_run_out (HWVoiceOut *hw, int live)
     LPVOID p1, p2;
     int bufsize;
     dsound *s = ds->s;
-    DSoundConf *conf = &s->conf;
+    AudiodevDsoundOptions *dso = s->dev->dsound;
 
     if (!dsb) {
         dolog ("Attempt to run empty with playback buffer\n");
@@ -500,14 +494,14 @@ static int dsound_run_out (HWVoiceOut *hw, int live)
     len = live << hwshift;
 
     if (ds->first_time) {
-        if (conf->latency_millis) {
+        if (dso->latency) {
             DWORD cur_blat;
 
             cur_blat = audio_ring_dist (wpos, ppos, bufsize);
             ds->first_time = 0;
             old_pos = wpos;
             old_pos +=
-                millis_to_bytes (&hw->info, conf->latency_millis) - cur_blat;
+                usecs_to_bytes(&hw->info, dso->latency) - cur_blat;
             old_pos %= bufsize;
             old_pos &= ~hw->info.align;
         }
@@ -746,12 +740,6 @@ static int dsound_run_in (HWVoiceIn *hw)
     return decr;
 }
 
-static DSoundConf glob_conf = {
-    .bufsize_in         = 16384,
-    .bufsize_out        = 16384,
-    .latency_millis     = 10
-};
-
 static void dsound_audio_fini (void *opaque)
 {
     HRESULT hr;
@@ -782,13 +770,22 @@ static void dsound_audio_fini (void *opaque)
     g_free(s);
 }
 
-static void *dsound_audio_init (void)
+static void *dsound_audio_init(Audiodev *dev)
 {
     int err;
     HRESULT hr;
     dsound *s = g_malloc0(sizeof(dsound));
+    AudiodevDsoundOptions *dso;
+
+    assert(dev->kind == AUDIODEV_DRIVER_DSOUND);
+    s->dev = dev;
+    dso = dev->dsound;
+
+    if (!dso->has_latency) {
+        dso->has_latency = true;
+        dso->latency = 10000; /* 10 ms */
+    }
 
-    s->conf = glob_conf;
     hr = CoInitialize (NULL);
     if (FAILED (hr)) {
         dsound_logerr (hr, "Could not initialize COM\n");
@@ -853,28 +850,6 @@ static void *dsound_audio_init (void)
     return s;
 }
 
-static struct audio_option dsound_options[] = {
-    {
-        .name  = "LATENCY_MILLIS",
-        .tag   = AUD_OPT_INT,
-        .valp  = &glob_conf.latency_millis,
-        .descr = "(undocumented)"
-    },
-    {
-        .name  = "BUFSIZE_OUT",
-        .tag   = AUD_OPT_INT,
-        .valp  = &glob_conf.bufsize_out,
-        .descr = "(undocumented)"
-    },
-    {
-        .name  = "BUFSIZE_IN",
-        .tag   = AUD_OPT_INT,
-        .valp  = &glob_conf.bufsize_in,
-        .descr = "(undocumented)"
-    },
-    { /* End of list */ }
-};
-
 static struct audio_pcm_ops dsound_pcm_ops = {
     .init_out = dsound_init_out,
     .fini_out = dsound_fini_out,
@@ -892,7 +867,6 @@ static struct audio_pcm_ops dsound_pcm_ops = {
 struct audio_driver dsound_audio_driver = {
     .name           = "dsound",
     .descr          = "DirectSound http://wikipedia.org/wiki/DirectSound",
-    .options        = dsound_options,
     .init           = dsound_audio_init,
     .fini           = dsound_audio_fini,
     .pcm_ops        = &dsound_pcm_ops,
audio/noaudio.c +1 -2
index 50db1f3..4c94a26 100644
@@ -134,7 +134,7 @@ static int no_ctl_in (HWVoiceIn *hw, int cmd, ...)
     return 0;
 }
 
-static void *no_audio_init (void)
+static void *no_audio_init (Audiodev *dev)
 {
     return &no_audio_init;
 }
@@ -161,7 +161,6 @@ static struct audio_pcm_ops no_pcm_ops = {
 struct audio_driver no_audio_driver = {
     .name           = "none",
     .descr          = "Timer based audio emulation",
-    .options        = NULL,
     .init           = no_audio_init,
     .fini           = no_audio_fini,
     .pcm_ops        = &no_pcm_ops,
audio/ossaudio.c +43 -112
index 3ea7d27..a5e7f7c 100644
@@ -29,6 +29,7 @@
 #include "qemu-common.h"
 #include "qemu/main-loop.h"
 #include "qemu/host-utils.h"
+#include "qapi-visit.h"
 #include "audio.h"
 #include "trace.h"
 
@@ -39,16 +40,6 @@
 #define USE_DSP_POLICY
 #endif
 
-typedef struct OSSConf {
-    int try_mmap;
-    int nfrags;
-    int fragsize;
-    const char *devpath_out;
-    const char *devpath_in;
-    int exclusive;
-    int policy;
-} OSSConf;
-
 typedef struct OSSVoiceOut {
     HWVoiceOut hw;
     void *pcm_buf;
@@ -58,7 +49,7 @@ typedef struct OSSVoiceOut {
     int fragsize;
     int mmapped;
     int pending;
-    OSSConf *conf;
+    Audiodev *dev;
 } OSSVoiceOut;
 
 typedef struct OSSVoiceIn {
@@ -67,12 +58,12 @@ typedef struct OSSVoiceIn {
     int fd;
     int nfrags;
     int fragsize;
-    OSSConf *conf;
+    Audiodev *dev;
 } OSSVoiceIn;
 
 struct oss_params {
     int freq;
-    AudioFormat fmt;
+    int fmt;
     int nchannels;
     int nfrags;
     int fragsize;
@@ -264,19 +255,26 @@ static int oss_get_version (int fd, int *version, const char *typ)
 }
 #endif
 
-static int oss_open (int in, struct oss_params *req,
-                     struct oss_params *obt, int *pfd, OSSConf* conf)
+static int oss_open(int in, struct oss_params *req, audsettings *as,
+                    struct oss_params *obt, int *pfd, Audiodev *dev)
 {
+    AudiodevOssOptions *oopts = dev->oss;
+    AudiodevOssPerDirectionOptions *opdo = in ? oopts->oss_in : oopts->oss_out;
+    AudiodevPerDirectionOptions *pdo = in ? dev->in : dev->out;
     int fd;
-    int oflags = conf->exclusive ? O_EXCL : 0;
+    int oflags = (oopts->has_exclusive && oopts->exclusive) ? O_EXCL : 0;
     audio_buf_info abinfo;
     int fmt, freq, nchannels;
     int setfragment = 1;
-    const char *dspname = in ? conf->devpath_in : conf->devpath_out;
+    const char *dspname = opdo->has_dev ? opdo->dev : "/dev/dsp";
     const char *typ = in ? "ADC" : "DAC";
+#ifdef USE_DSP_POLICY
+    int policy = oopts->has_dsp_policy ? oopts->dsp_policy : 5;
+#endif
 
     /* Kludge needed to have working mmap on Linux */
-    oflags |= conf->try_mmap ? O_RDWR : (in ? O_RDONLY : O_WRONLY);
+    oflags |= (oopts->has_try_mmap && oopts->try_mmap) ?
+        O_RDWR : (in ? O_RDONLY : O_WRONLY);
 
     fd = open (dspname, oflags | O_NONBLOCK);
     if (-1 == fd) {
@@ -287,6 +285,8 @@ static int oss_open (int in, struct oss_params *req,
     freq = req->freq;
     nchannels = req->nchannels;
     fmt = req->fmt;
+    req->nfrags = pdo->has_buffer_count ? pdo->buffer_count : 4;
+    req->fragsize = audio_buffer_bytes(pdo, as, 23220);
 
     if (ioctl (fd, SNDCTL_DSP_SAMPLESIZE, &fmt)) {
         oss_logerr2 (errno, typ, "Failed to set sample size %d\n", req->fmt);
@@ -310,18 +310,18 @@ static int oss_open (int in, struct oss_params *req,
     }
 
 #ifdef USE_DSP_POLICY
-    if (conf->policy >= 0) {
+    if (policy >= 0) {
         int version;
 
         if (!oss_get_version (fd, &version, typ)) {
             trace_oss_version(version);
 
             if (version >= 0x040000) {
-                int policy = conf->policy;
-                if (ioctl (fd, SNDCTL_DSP_POLICY, &policy)) {
+                int policy2 = policy;
+                if (ioctl (fd, SNDCTL_DSP_POLICY, &policy2)) {
                     oss_logerr2 (errno, typ,
                                  "Failed to set timing policy to %d\n",
-                                 conf->policy);
+                                 policy);
                     goto err;
                 }
                 setfragment = 0;
@@ -504,17 +504,16 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
     int fd;
     AudioFormat effective_fmt;
     struct audsettings obt_as;
-    OSSConf *conf = drv_opaque;
+    Audiodev *dev = drv_opaque;
+    AudiodevOssOptions *oopts = dev->oss;
 
     oss->fd = -1;
 
     req.fmt = aud_to_ossfmt (as->fmt, as->endianness);
     req.freq = as->freq;
     req.nchannels = as->nchannels;
-    req.fragsize = conf->fragsize;
-    req.nfrags = conf->nfrags;
 
-    if (oss_open (0, &req, &obt, &fd, conf)) {
+    if (oss_open(0, &req, as, &obt, &fd, dev)) {
         return -1;
     }
 
@@ -541,7 +540,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
     hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
 
     oss->mmapped = 0;
-    if (conf->try_mmap) {
+    if (oopts->has_try_mmap && oopts->try_mmap) {
         oss->pcm_buf = mmap (
             NULL,
             hw->samples << hw->info.shift,
@@ -601,7 +600,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
     }
 
     oss->fd = fd;
-    oss->conf = conf;
+    oss->dev = dev;
     return 0;
 }
 
@@ -609,16 +608,12 @@ static int oss_ctl_out (HWVoiceOut *hw, int cmd, ...)
 {
     int trig;
     OSSVoiceOut *oss = (OSSVoiceOut *) hw;
+    AudiodevOssPerDirectionOptions *opdo = oss->dev->oss->oss_out;
 
     switch (cmd) {
     case VOICE_ENABLE:
         {
-            va_list ap;
-            int poll_mode;
-
-            va_start (ap, cmd);
-            poll_mode = va_arg (ap, int);
-            va_end (ap);
+            bool poll_mode = !opdo->has_try_poll || opdo->try_poll;
 
             ldebug ("enabling voice\n");
             if (poll_mode) {
@@ -673,16 +668,14 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
     int fd;
     AudioFormat effective_fmt;
     struct audsettings obt_as;
-    OSSConf *conf = drv_opaque;
+    Audiodev *dev = drv_opaque;
 
     oss->fd = -1;
 
     req.fmt = aud_to_ossfmt (as->fmt, as->endianness);
     req.freq = as->freq;
     req.nchannels = as->nchannels;
-    req.fragsize = conf->fragsize;
-    req.nfrags = conf->nfrags;
-    if (oss_open (1, &req, &obt, &fd, conf)) {
+    if (oss_open(1, &req, as, &obt, &fd, dev)) {
         return -1;
     }
 
@@ -716,7 +709,7 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
     }
 
     oss->fd = fd;
-    oss->conf = conf;
+    oss->dev = dev;
     return 0;
 }
 
@@ -807,16 +800,12 @@ static int oss_read (SWVoiceIn *sw, void *buf, int size)
 static int oss_ctl_in (HWVoiceIn *hw, int cmd, ...)
 {
     OSSVoiceIn *oss = (OSSVoiceIn *) hw;
+    AudiodevOssPerDirectionOptions *opdo = oss->dev->oss->oss_out;
 
     switch (cmd) {
     case VOICE_ENABLE:
         {
-            va_list ap;
-            int poll_mode;
-
-            va_start (ap, cmd);
-            poll_mode = va_arg (ap, int);
-            va_end (ap);
+            bool poll_mode = !opdo->has_try_poll || opdo->try_poll;
 
             if (poll_mode) {
                 oss_poll_in (hw);
@@ -836,82 +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 (void)
+static void *oss_audio_init(Audiodev *dev)
 {
-    OSSConf *conf = g_malloc(sizeof(OSSConf));
-    *conf = glob_conf;
+    AudiodevOssOptions *oopts;
+    assert(dev->kind == AUDIODEV_DRIVER_OSS);
 
-    if (access(conf->devpath_in, R_OK | W_OK) < 0 ||
-        access(conf->devpath_out, R_OK | W_OK) < 0) {
-        g_free(conf);
+    oopts = dev->oss;
+    if (access(oopts->oss_in->has_dev ? oopts->oss_in->dev : "/dev/dsp",
+               R_OK | W_OK) < 0 ||
+        access(oopts->oss_out->has_dev ? oopts->oss_out->dev : "/dev/dsp",
+               R_OK | W_OK) < 0) {
         return NULL;
     }
-    return conf;
+    return dev;
 }
 
 static void oss_audio_fini (void *opaque)
 {
-    g_free(opaque);
 }
 
-static struct audio_option oss_options[] = {
-    {
-        .name  = "FRAGSIZE",
-        .tag   = AUD_OPT_INT,
-        .valp  = &glob_conf.fragsize,
-        .descr = "Fragment size in bytes"
-    },
-    {
-        .name  = "NFRAGS",
-        .tag   = AUD_OPT_INT,
-        .valp  = &glob_conf.nfrags,
-        .descr = "Number of fragments"
-    },
-    {
-        .name  = "MMAP",
-        .tag   = AUD_OPT_BOOL,
-        .valp  = &glob_conf.try_mmap,
-        .descr = "Try using memory mapped access"
-    },
-    {
-        .name  = "DAC_DEV",
-        .tag   = AUD_OPT_STR,
-        .valp  = &glob_conf.devpath_out,
-        .descr = "Path to DAC device"
-    },
-    {
-        .name  = "ADC_DEV",
-        .tag   = AUD_OPT_STR,
-        .valp  = &glob_conf.devpath_in,
-        .descr = "Path to ADC device"
-    },
-    {
-        .name  = "EXCLUSIVE",
-        .tag   = AUD_OPT_BOOL,
-        .valp  = &glob_conf.exclusive,
-        .descr = "Open device in exclusive mode (vmix wont work)"
-    },
-#ifdef USE_DSP_POLICY
-    {
-        .name  = "POLICY",
-        .tag   = AUD_OPT_INT,
-        .valp  = &glob_conf.policy,
-        .descr = "Set the timing policy of the device, -1 to use fragment mode",
-    },
-#endif
-    { /* End of list */ }
-};
-
 static struct audio_pcm_ops oss_pcm_ops = {
     .init_out = oss_init_out,
     .fini_out = oss_fini_out,
@@ -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,
audio/paaudio.c +30 -52
index cfdbdc6..a53aaf6 100644
@@ -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->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->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 (void)
+static void *qpa_audio_init(Audiodev *dev)
 {
-    paaudio *g = g_malloc(sizeof(paaudio));
-    g->conf = glob_conf;
+    paaudio *g;
+    AudiodevPaOptions *popts;
+    const char *server;
+
+    assert(dev->kind == AUDIODEV_DRIVER_PA);
+
+    g = g_malloc(sizeof(paaudio));
+    popts = dev->pa;
+    server = popts->has_server ? popts->server : NULL;
+
+    g->dev = dev;
     g->mainloop = NULL;
     g->context = NULL;
 
@@ -825,14 +832,14 @@ static void *qpa_audio_init (void)
     }
 
     g->context = pa_context_new (pa_threaded_mainloop_get_api (g->mainloop),
-                                 g->conf.server);
+                                 server);
     if (!g->context) {
         goto fail;
     }
 
     pa_context_set_state_callback (g->context, context_state_cb, g);
 
-    if (pa_context_connect (g->context, g->conf.server, 0, NULL) < 0) {
+    if (pa_context_connect (g->context, server, 0, NULL) < 0) {
         qpa_logerr (pa_context_errno (g->context),
                     "pa_context_connect() failed\n");
         goto fail;
@@ -895,34 +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,
audio/sdlaudio.c +5 -19
index db0f95a..796238a 100644
@@ -44,18 +44,13 @@ typedef struct SDLVoiceOut {
     int decr;
 } SDLVoiceOut;
 
-static struct {
-    int nb_samples;
-} conf = {
-    .nb_samples = 1024
-};
-
 static struct SDLAudioState {
     int exit;
     SDL_mutex *mutex;
     SDL_sem *sem;
     int initialized;
     bool driver_created;
+    Audiodev *dev;
 } glob_sdl;
 typedef struct SDLAudioState SDLAudioState;
 
@@ -347,7 +342,7 @@ static int sdl_init_out(HWVoiceOut *hw, struct audsettings *as,
     req.freq = as->freq;
     req.format = aud_to_sdlfmt (as->fmt);
     req.channels = as->nchannels;
-    req.samples = conf.nb_samples;
+    req.samples = audio_buffer_samples(s->dev->out, as, 11610);
     req.callback = sdl_callback;
     req.userdata = sdl;
 
@@ -391,7 +386,7 @@ static int sdl_ctl_out (HWVoiceOut *hw, int cmd, ...)
     return 0;
 }
 
-static void *sdl_audio_init (void)
+static void *sdl_audio_init(Audiodev *dev)
 {
     SDLAudioState *s = &glob_sdl;
     if (s->driver_created) {
@@ -420,6 +415,7 @@ static void *sdl_audio_init (void)
     }
 
     s->driver_created = true;
+    s->dev = dev;
     return s;
 }
 
@@ -431,18 +427,9 @@ static void sdl_audio_fini (void *opaque)
     SDL_DestroyMutex (s->mutex);
     SDL_QuitSubSystem (SDL_INIT_AUDIO);
     s->driver_created = false;
+    s->dev = NULL;
 }
 
-static struct audio_option sdl_options[] = {
-    {
-        .name  = "SAMPLES",
-        .tag   = AUD_OPT_INT,
-        .valp  = &conf.nb_samples,
-        .descr = "Size of SDL buffer in samples"
-    },
-    { /* End of list */ }
-};
-
 static struct audio_pcm_ops sdl_pcm_ops = {
     .init_out = sdl_init_out,
     .fini_out = sdl_fini_out,
@@ -454,7 +441,6 @@ static struct audio_pcm_ops sdl_pcm_ops = {
 struct audio_driver sdl_audio_driver = {
     .name           = "sdl",
     .descr          = "SDL http://www.libsdl.org",
-    .options        = sdl_options,
     .init           = sdl_audio_init,
     .fini           = sdl_audio_fini,
     .pcm_ops        = &sdl_pcm_ops,
audio/spiceaudio.c +1 -6
index 141fd8d..a9b9a1d 100644
@@ -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;
@@ -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,
audio/wavaudio.c +10 -50
index 81250e6..1af6d23 100644
@@ -23,6 +23,7 @@
  */
 #include "hw/hw.h"
 #include "qemu/timer.h"
+#include "qapi-visit.h"
 #include "audio.h"
 
 #define AUDIO_CAP "wav"
@@ -36,11 +37,6 @@ typedef struct WAVVoiceOut {
     int total_samples;
 } WAVVoiceOut;
 
-typedef struct {
-    struct audsettings settings;
-    const char *wav_path;
-} WAVConf;
-
 static int wav_run_out (HWVoiceOut *hw, int live)
 {
     WAVVoiceOut *wav = (WAVVoiceOut *) hw;
@@ -111,8 +107,10 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
         0x02, 0x00, 0x44, 0xac, 0x00, 0x00, 0x10, 0xb1, 0x02, 0x00, 0x04,
         0x00, 0x10, 0x00, 0x64, 0x61, 0x74, 0x61, 0x00, 0x00, 0x00, 0x00
     };
-    WAVConf *conf = drv_opaque;
-    struct audsettings wav_as = conf->settings;
+    Audiodev *dev = drv_opaque;
+    AudiodevWavOptions *wopts = dev->wav;
+    struct audsettings wav_as = audiodev_to_audsettings(dev->out);
+    const char *wav_path = wopts->has_path ? wopts->path : "qemu.wav";
 
     stereo = wav_as.nchannels == 2;
     switch (wav_as.fmt) {
@@ -153,10 +151,10 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
     le_store (hdr + 28, hw->info.freq << (bits16 + stereo), 4);
     le_store (hdr + 32, 1 << (bits16 + stereo), 2);
 
-    wav->f = fopen (conf->wav_path, "wb");
+    wav->f = fopen(wav_path, "wb");
     if (!wav->f) {
         dolog ("Failed to open wave file `%s'\nReason: %s\n",
-               conf->wav_path, strerror (errno));
+               wav_path, strerror(errno));
         g_free (wav->pcm_buf);
         wav->pcm_buf = NULL;
         return -1;
@@ -224,54 +222,17 @@ static int wav_ctl_out (HWVoiceOut *hw, int cmd, ...)
     return 0;
 }
 
-static WAVConf glob_conf = {
-    .settings.freq      = 44100,
-    .settings.nchannels = 2,
-    .settings.fmt       = AUDIO_FORMAT_S16,
-    .wav_path           = "qemu.wav"
-};
-
-static void *wav_audio_init (void)
+static void *wav_audio_init(Audiodev *dev)
 {
-    WAVConf *conf = g_malloc(sizeof(WAVConf));
-    *conf = glob_conf;
-    return conf;
+    assert(dev->kind == AUDIODEV_DRIVER_WAV);
+    return dev;
 }
 
 static void wav_audio_fini (void *opaque)
 {
     ldebug ("wav_fini");
-    g_free(opaque);
 }
 
-static struct audio_option wav_options[] = {
-    {
-        .name  = "FREQUENCY",
-        .tag   = AUD_OPT_INT,
-        .valp  = &glob_conf.settings.freq,
-        .descr = "Frequency"
-    },
-    {
-        .name  = "FORMAT",
-        .tag   = AUD_OPT_FMT,
-        .valp  = &glob_conf.settings.fmt,
-        .descr = "Format"
-    },
-    {
-        .name  = "DAC_FIXED_CHANNELS",
-        .tag   = AUD_OPT_INT,
-        .valp  = &glob_conf.settings.nchannels,
-        .descr = "Number of channels (1 - mono, 2 - stereo)"
-    },
-    {
-        .name  = "PATH",
-        .tag   = AUD_OPT_STR,
-        .valp  = &glob_conf.wav_path,
-        .descr = "Path to wave file"
-    },
-    { /* End of list */ }
-};
-
 static struct audio_pcm_ops wav_pcm_ops = {
     .init_out = wav_init_out,
     .fini_out = wav_fini_out,
@@ -283,7 +244,6 @@ static struct audio_pcm_ops wav_pcm_ops = {
 struct audio_driver wav_audio_driver = {
     .name           = "wav",
     .descr          = "WAV renderer http://wikipedia.org/wiki/WAV",
-    .options        = wav_options,
     .init           = wav_audio_init,
     .fini           = wav_audio_fini,
     .pcm_ops        = &wav_pcm_ops,
qemu-options.hx +2 -2
index 77f5853..efd57e6 100644
@@ -185,8 +185,8 @@ Set default value of @var{driver}'s property @var{prop} to @var{value}, e.g.:
 qemu-system-i386 -global ide-drive.physical_block_size=4096 -drive file=file,if=ide,index=0,media=disk
 @end example
 
-In particular, you can use this to set driver properties for devices which are
-created automatically by the machine model. To create a device which is not
+In particular, you can use this to set driver properties for devices which are
+created automatically by the machine model. To create a device which is not
 created automatically and set properties on it, use -@option{device}.
 
 -global @var{driver}.@var{prop}=@var{value} is shorthand for -global

qemu-options.hx The default is @code{en-us}. · line 318

Quoted 7 lines
qemu-options.hx +2 -1
@@ -313,14 +313,230 @@ 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 current audio settings\n",

by current, you mean "old" or "deprecated" perhaps?

Quoted 264 lines
qemu-options.hx +217 -2
     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,in.frequency=44110,out.frequency=8000
+-audiodev alsa,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.
[… more lines in the raw message]

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vl.c +9 -2
index 0adbbd6..f1aa1df 100644

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The rest of this diff is not shown. View the raw message.

Quoted 3 lines

2.4.5

overall looks good to me, but it will require testing.

Tbh, I wonder why qemu has so many audio backends and options... (linking with gstreamer would be just one more lib of 1mb after all, with a lot more flexibility, and it could be made a module..) :)

Signature · 1 line
Marc-André Lureau

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#17 lore raw

PATCH 11/25 audio: reduce glob_audio_state usage

In reply to Kővágó, Zoltán
---
 audio/audio.c          | 41 ++++++++++++++++++++---------------------
 audio/audio_int.h      |  4 ++++
 audio/audio_template.h | 46 ++++++++++++++++++++++------------------------
 3 files changed, 46 insertions(+), 45 deletions(-)
audio/audio.c +20 -21
index 08ac15e..05b24dc 100644
@@ -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;
audio/audio_int.h +4 -0
index 59b2362..101081b 100644
@@ -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;
audio/audio_template.h +22 -24

137 lines not on this page · raw message

Signature · 1 line
2.4.5
#18 lore raw

Re: [Qemu-devel] [PATCH 11/25] audio: reduce glob_audio_state usage

Hi

Quoted 1 line · Zoltán

Fill a bit the commit message, to say simply use AudioState instead. Also explain why you add AudioState *s in struct although it's not used in that commit, saying simply it's going to be used in later commit is enough imho (I guess it is).

Quoted 358 lines
---
 audio/audio.c          | 41 ++++++++++++++++++++---------------------
 audio/audio_int.h      |  4 ++++
 audio/audio_template.h | 46 ++++++++++++++++++++++------------------------
 3 files changed, 46 insertions(+), 45 deletions(-)
audio/audio.c +20 -21
index 08ac15e..05b24dc 100644
@@ -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;
audio/audio_int.h +4 -0
index 59b2362..101081b 100644
@@ -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;
audio/audio_template.h +22 -25
index 096b2b3..455e50d 100644
@@ -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;
--

2.4.5

Signature · 1 line
Marc-André Lureau
#19 lore raw

PATCH 12/25 audio: basic support for multi backend audio

In reply to Kővágó, Zoltán

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.

---
 audio/audio.c          | 95 +++++++++++++++++++++++++++++++++++++-------------
 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, 126 insertions(+), 39 deletions(-)
audio/audio.c +71 -24
index 05b24dc..10b9871 100644
@@ -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 +1313,25 @@ static const VMStateDescription vmstate_audio = {
 };
 
 static Audiodev *parse_option(QemuOpts *opts, Error **errp);
-static int audio_init(Audiodev *dev)
+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 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 */
     if (dev) {
         drvname = AudiodevDriver_lookup[dev->kind];
+    } else if (!QTAILQ_EMPTY(&audio_states)) {
+        return QTAILQ_FIRST(&audio_states);
     } else {
         audio_handle_legacy_opts();
         list = qemu_find_opts("audiodev");
@@ -1328,12 +1340,18 @@ 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 +1432,18 @@ 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 +1453,20 @@ 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 */
+        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 +1702,7 @@ 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 +1768,25 @@ 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 "";
+    }
+}
audio/audio.h +9 -3
index 177a673..0085a07 100644
@@ -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 */
audio/audio_int.h +2 -0
index 101081b..1d81658 100644
@@ -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;
audio/audio_template.h +1 -1
index 455e50d..5a3dc90 100644
@@ -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)) {
audio/wavcapture.c +3 -3
index 798acdd..8b8d3d0 100644
@@ -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;
hmp-commands.hx +6 -5
index d3b7932..9b49dc3 100644
@@ -741,16 +741,17 @@ 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
monitor.c +11 -1
index a40138b..f527753 100644
@@ -1944,7 +1944,17 @@ 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));
 
@@ -1952,7 +1962,7 @@ 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;
qemu-options.hx +5 -0
index efd57e6..60a3563 100644
@@ -1538,6 +1538,11 @@ 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
 
ui/vnc.c +14 -1
index febbad6..feee4ef 100644
@@ -1193,7 +1193,7 @@ 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");
     }
@@ -3295,6 +3295,9 @@ static QemuOptsList qemu_vnc_opts = {
         },{
             .name = "non-adaptive",
             .type = QEMU_OPT_BOOL,
+        },{
+            .name = "audiodev",
+            .type = QEMU_OPT_STRING,
         },
         { /* end of list */ }
     },
@@ -3455,6 +3458,7 @@ void vnc_display_open(const char *id, Error **errp)
     int acl = 0;
 #endif
     int lock_key_sync = 1;
+    const char *audiodev;
 
     if (!vs) {
         error_setg(errp, "VNC display not active");
@@ -3637,6 +3641,15 @@ 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;
ui/vnc.h +2 -0
index 814d720..f73bb94 100644
@@ -187,6 +187,8 @@ struct VncDisplay
 #ifdef CONFIG_VNC_SASL
     VncDisplaySASL sasl;
 #endif
+
+    AudioState *audio_state;
 };
 
 typedef struct VncTight {
vl.c +2 -1
index f1aa1df..30bc8be 100644
@@ -4315,6 +4315,8 @@ int main(int argc, char **argv, char **envp)
         exit(1);
     }
 
+    audio_set_options();
+
     machine_opts = qemu_get_machine_opts();
     if (qemu_opt_foreach(machine_opts, machine_set_property, current_machine,
                          NULL)) {
@@ -4518,7 +4520,6 @@ int main(int argc, char **argv, char **envp)
 
     realtime_init();
 
-    audio_set_options();
     audio_init();
 
     cpu_synchronize_all_post_init();
Signature · 1 line
2.4.5
#20 lore raw

Re: [Qemu-devel] [PATCH 12/25] audio: basic support for multi backend audio

Hi

Quoted 102 lines · Zoltán

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.

---
 audio/audio.c          | 95 +++++++++++++++++++++++++++++++++++++-------------
 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, 126 insertions(+), 39 deletions(-)
audio/audio.c +23 -11
index 05b24dc..10b9871 100644
@@ -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 +1313,25 @@ static const VMStateDescription vmstate_audio = {
 };
 
 static Audiodev *parse_option(QemuOpts *opts, Error **errp);
-static int audio_init(Audiodev *dev)
+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) {

audio/audio.c static const VMStateDescription vmstate_audio = { · line 1330

Quoted 10 lines
audio/audio.c +4 -6
-            dolog("Cannot create more than one audio backend, sorry\n");
-            qapi_free_Audiodev(dev);
-        }
-        return -1;
-    }
-
+    /* 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 */

I think you could place this comment above the function.

Quoted 56 lines
audio/audio.c +17 -5
     if (dev) {
         drvname = AudiodevDriver_lookup[dev->kind];
+    } else if (!QTAILQ_EMPTY(&audio_states)) {
+        return QTAILQ_FIRST(&audio_states);
     } else {
         audio_handle_legacy_opts();
         list = qemu_find_opts("audiodev");
@@ -1328,12 +1340,18 @@ 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 +1432,18 @@ 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 +1453,20 @@ void AUD_remove_card (QEMUSoundCard *card)
 }
 
 
-CaptureVoiceOut *AUD_add_capture (
+CaptureVoiceOut *AUD_add_capture(
+    AudioState *s,

audio/audio.c void AUD_remove_card (QEMUSoundCard *card) · line 1466

Quoted 10 lines
audio/audio.c +1 -1
     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 */

What's the todo here for?

Quoted 196 lines
audio/audio.c +26 -1
+        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 +1702,7 @@ 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 +1768,25 @@ 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 "";
+    }
+}
audio/audio.h +9 -3
index 177a673..0085a07 100644
@@ -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 */
audio/audio_int.h +2 -0
index 101081b..1d81658 100644
@@ -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;
audio/audio_template.h +1 -1
index 455e50d..5a3dc90 100644
@@ -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)) {
audio/wavcapture.c +3 -3
index 798acdd..8b8d3d0 100644
@@ -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;
hmp-commands.hx +6 -5
index d3b7932..9b49dc3 100644
@@ -741,16 +741,17 @@ 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
monitor.c +11 -1
index a40138b..f527753 100644
@@ -1944,7 +1944,17 @@ 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));
 
@@ -1952,7 +1962,7 @@ 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;
qemu-options.hx +0 -0
index efd57e6..60a3563 100644

qemu-options.hx everybody else. 'ignore' completely ignores the shared flag and · line 1544

Quoted 7 lines
qemu-options.hx +4 -0
@@ -1538,6 +1538,11 @@ 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.

That doesn't make much sense to me, vnc (or spice) should be connected to the guest device, not to the audiodev. Same for for the wavcapture.

Quoted 89 lines
qemu-options.hx +1 -0
+
 @end table
 ETEXI
 
ui/vnc.c +14 -1
index febbad6..feee4ef 100644
@@ -1193,7 +1193,7 @@ 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");
     }
@@ -3295,6 +3295,9 @@ static QemuOptsList qemu_vnc_opts = {
         },{
             .name = "non-adaptive",
             .type = QEMU_OPT_BOOL,
+        },{
+            .name = "audiodev",
+            .type = QEMU_OPT_STRING,
         },
         { /* end of list */ }
     },
@@ -3455,6 +3458,7 @@ void vnc_display_open(const char *id, Error **errp)
     int acl = 0;
 #endif
     int lock_key_sync = 1;
+    const char *audiodev;
 
     if (!vs) {
         error_setg(errp, "VNC display not active");
@@ -3637,6 +3641,15 @@ 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;
ui/vnc.h +2 -0
index 814d720..f73bb94 100644
@@ -187,6 +187,8 @@ struct VncDisplay
 #ifdef CONFIG_VNC_SASL
     VncDisplaySASL sasl;
 #endif
+
+    AudioState *audio_state;
 };
 
 typedef struct VncTight {
vl.c +2 -2
index f1aa1df..30bc8be 100644
@@ -4315,6 +4315,8 @@ int main(int argc, char **argv, char **envp)
         exit(1);
     }
 
+    audio_set_options();
+
     machine_opts = qemu_get_machine_opts();
     if (qemu_opt_foreach(machine_opts, machine_set_property, current_machine,
                          NULL)) {
@@ -4518,7 +4520,6 @@ int main(int argc, char **argv, char **envp)
 
     realtime_init();
 
-    audio_set_options();
     audio_init();
 
     cpu_synchronize_all_post_init();
--

2.4.5

Signature · 1 line
Marc-André Lureau
#21 lore raw

Re: [Qemu-devel] [PATCH 12/25] audio: basic support for multi backend audio

Hi

2015-08-20 20:01 keltezéssel, Marc-André Lureau írta:
[snip]

Quoted 8 lines
Quoted 8 lines
qemu-options.hx +0 -0
index efd57e6..60a3563 100644
@@ -1538,6 +1538,11 @@ 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.
 
Quoted 7 lines

qemu-options.hx everybody else. 'ignore' completely ignores the shared flag and · line 1544

Quoted 4 lines
qemu-options.hx +4 -0
+@item audiodev=@var{audiodev}
+
+Use the specified @var{audiodev} when the VNC client requests audio
+transmission.

That doesn't make much sense to me, vnc (or spice) should be connected to the guest device, not to the audiodev. Same for for the wavcapture.

Hmm. Maybe it would make more sense, but it's not how the audio system works. The situation cyrrently (before my patch) is the following: VNC/spice/wavcapture attaches a capture to the global AudioState and starts recording/transmitting the mixed together audio of all connected guest sound cards.

Now with more audiodevs, you have to specify which one you wan't to capture from. If there's multiple sound cards attached to a single audiodev (AudioState), then we will transmit the mix of them. (If there's only one attached though, then we will only transmit that...)

To support capture from a single guest device would require some rewriting of the audio capture logic, in a backward compatible way (which would mean support both capturing form a single card and capturing a mix of all cards. And in the latter case we still need some way to specify audiodev...)

Zoltan

#22 lore raw

Re: [Qemu-devel] [PATCH 12/25] audio: basic support for multi backend audio

Hi

Quoted 6 lines · Kővágó Zoltán

Hmm. Maybe it would make more sense, but it's not how the audio system works. The situation cyrrently (before my patch) is the following:
VNC/spice/wavcapture attaches a capture to the global AudioState and starts recording/transmitting the mixed together audio of all connected guest sound cards.

Ok, thanks for explaining.

Quoted 9 lines

Now with more audiodevs, you have to specify which one you wan't to capture from. If there's multiple sound cards attached to a single audiodev (AudioState), then we will transmit the mix of them. (If there's only one attached though, then we will only transmit that...)

To support capture from a single guest device would require some rewriting of the audio capture logic, in a backward compatible way (which would mean support both capturing form a single card and capturing a mix of all cards. And in the latter case we still need some way to specify audiodev...)

Forget about that for now then :)

thanks

Signature · 1 line
Marc-André Lureau
#23 lore raw

PATCH 13/25 audio: add audiodev properties to frontends

In reply to Kővágó, Zoltán

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

---
 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(-)
audio/audio.h +3 -0
index 0085a07..68545b6 100644
@@ -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 */
hw/audio/ac97.c +1 -0
index fa75f33..a835cb6 100644
@@ -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 (),
 };
hw/audio/adlib.c +1 -0
index f8f0f55..58de8b0 100644
@@ -352,6 +352,7 @@ 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 (),
hw/audio/cs4231a.c +1 -0
index 626a173..43a1a39 100644
@@ -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),
hw/audio/es1370.c +6 -1
index f6e74cb..fe6110d 100644
@@ -1047,6 +1047,11 @@ 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);
@@ -1061,6 +1066,7 @@ 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 = {
@@ -1077,4 +1083,3 @@ static void es1370_register_types (void)
 }
 
 type_init (es1370_register_types)
-
hw/audio/gus.c +1 -0
index 6107824..ee08d50 100644
@@ -284,6 +284,7 @@ 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),
hw/audio/hda-codec.c +1 -0
index 8693b7a..dba3331 100644
@@ -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(),
hw/audio/milkymist-ac97.c +6 -0
index 15169e2..b63fd6f 100644
@@ -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 = {
hw/audio/pcspk.c +1 -0
index 302debf..85d6d00 100644
@@ -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(),
hw/audio/pl041.c +1 -0
index 19982f2..c003b26 100644
@@ -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),
hw/audio/sb16.c +1 -0
index a159dcc..4bbb718 100644
@@ -1391,6 +1391,7 @@ 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),
hw/audio/wm8750.c +6 -0
index 4c4333c..049ddf9 100644
@@ -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 = {
hw/core/qdev-properties-system.c +57 -0

69 lines not on this page · raw message

hw/usb/dev-audio.c +1 -0
index 0171579..48ac992 100644
@@ -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,
                        8 * USBAUDIO_PACKET_SIZE),
include/hw/qdev-properties.h +3 -0
index 77538a8..edebcfa 100644
@@ -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()               \
     {}
Signature · 1 line
2.4.5
Reviewed-by on v1 13/25
#24 lore raw

Re: [Qemu-devel] [PATCH 13/25] audio: add audiodev properties to frontends

looks good,

Quoted 323 lines · Zoltán

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

---
 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(-)
audio/audio.h +3 -0
index 0085a07..68545b6 100644
@@ -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 */
hw/audio/ac97.c +1 -0
index fa75f33..a835cb6 100644
@@ -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 (),
 };
hw/audio/adlib.c +1 -0
index f8f0f55..58de8b0 100644
@@ -352,6 +352,7 @@ 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 (),
hw/audio/cs4231a.c +1 -0
index 626a173..43a1a39 100644
@@ -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),
hw/audio/es1370.c +6 -1
index f6e74cb..fe6110d 100644
@@ -1047,6 +1047,11 @@ 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);
@@ -1061,6 +1066,7 @@ 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 = {
@@ -1077,4 +1083,3 @@ static void es1370_register_types (void)
 }
 
 type_init (es1370_register_types)
-
hw/audio/gus.c +1 -0
index 6107824..ee08d50 100644
@@ -284,6 +284,7 @@ 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),
hw/audio/hda-codec.c +1 -0
index 8693b7a..dba3331 100644
@@ -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(),
hw/audio/milkymist-ac97.c +6 -0
index 15169e2..b63fd6f 100644
@@ -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 = {
hw/audio/pcspk.c +1 -0
index 302debf..85d6d00 100644
@@ -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(),
hw/audio/pl041.c +1 -0
index 19982f2..c003b26 100644
@@ -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),
hw/audio/sb16.c +1 -0
index a159dcc..4bbb718 100644
@@ -1391,6 +1391,7 @@ 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),
hw/audio/wm8750.c +6 -0
index 4c4333c..049ddf9 100644
@@ -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 = {
hw/core/qdev-properties-system.c +57 -0
index 249976e..3807c86 100644
@@ -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)
 {
hw/usb/dev-audio.c +1 -0
index 0171579..48ac992 100644
@@ -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,
                        8 * USBAUDIO_PACKET_SIZE),
include/hw/qdev-properties.h +3 -1
index 77538a8..edebcfa 100644
@@ -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()               \
     {}
--

2.4.5

Signature · 1 line
Marc-André Lureau
#25 lore raw

PATCH 14/25 audio: audiodev= parameters no longer optional when -audiodev present

In reply to Kővágó, Zoltán
---
 audio/audio.c | 13 ++++++++++++-
 1 file changed, 12 insertions(+), 1 deletion(-)
audio/audio.c +12 -1
index 10b9871..7468b94 100644
@@ -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
@@ -1331,8 +1333,13 @@ static AudioState *audio_init(Audiodev *dev)
     if (dev) {
         drvname = AudiodevDriver_lookup[dev->kind];
     } else if (!QTAILQ_EMPTY(&audio_states)) {
+        if (!legacy_config) {
+            dolog("Must specify audiodev when using -audiodev\n");
+            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);
@@ -1463,7 +1470,11 @@ CaptureVoiceOut *AUD_add_capture(
     CaptureVoiceOut *cap;
     struct capture_callback *cb;
 
-    if (!s) { /* todo */
+    if (!s) {
+        if (!legacy_config) {
+            dolog("Must specify audiodev when using -audiodev\n");
+            goto err0;
+        }
         s = QTAILQ_FIRST(&audio_states);
     }
 
Signature · 1 line
2.4.5
Reviewed-by on v1 14/25
#26 lore raw

Re: [Qemu-devel] [PATCH 14/25] audio: audiodev= parameters no longer optional when -audiodev present

Hi

Quoted 33 lines · Zoltán
---
 audio/audio.c | 13 ++++++++++++-
 1 file changed, 12 insertions(+), 1 deletion(-)
audio/audio.c +7 -0
index 10b9871..7468b94 100644
@@ -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
@@ -1331,8 +1333,13 @@ static AudioState *audio_init(Audiodev *dev)
     if (dev) {
         drvname = AudiodevDriver_lookup[dev->kind];
     } else if (!QTAILQ_EMPTY(&audio_states)) {
+        if (!legacy_config) {
+            dolog("Must specify audiodev when using -audiodev\n");
+            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);

audio/audio.c CaptureVoiceOut *AUD_add_capture( · line 1475

Quoted 7 lines
audio/audio.c +3 -1
@@ -1463,7 +1470,11 @@ CaptureVoiceOut *AUD_add_capture(
     CaptureVoiceOut *cap;
     struct capture_callback *cb;
 
-    if (!s) { /* todo */
+    if (!s) {
+        if (!legacy_config) {
+            dolog("Must specify audiodev when using -audiodev\n");

This error message is a bit confusing. I think adding a bit of context could help, you could pass the card name from the AUD_register_card call, and use a sentence like: "You must specify an audiodev= for the device %s".

Quoted 9 lines
audio/audio.c +2 -1
+            goto err0;
+        }
         s = QTAILQ_FIRST(&audio_states);
     }
 
--

2.4.5

looks good otherwise,

Signature · 1 line
Marc-André Lureau
#27 lore raw

PATCH 15/25 paaudio: do not create multiple connections to the same server

In reply to Kővágó, Zoltán
---
 audio/paaudio.c | 301 ++++++++++++++++++++++++++++++++------------------------
 1 file changed, 175 insertions(+), 126 deletions(-)
audio/paaudio.c +175 -126
index a53aaf6..e3b8207 100644
@@ -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 +495,52 @@ 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 (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,
-                                        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 +556,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
     paaudio *g = pa->g = drv_opaque;
     AudiodevPaOptions *popts = g->dev->pa;
     AudiodevPaPerDirectionOptions *ppdo = popts->sink;
+    PAConnection *c = g->conn;
 
     ss.format = audfmt_to_pa (as->fmt, as->endianness);
     ss.channels = as->nchannels;
@@ -560,7 +574,7 @@ 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 +626,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
     paaudio *g = pa->g = drv_opaque;
     AudiodevPaOptions *popts = g->dev->pa;
     AudiodevPaPerDirectionOptions *ppdo = popts->source;
+    PAConnection *c = g->conn;
 
     ss.format = audfmt_to_pa (as->fmt, as->endianness);
     ss.channels = as->nchannels;
@@ -620,7 +635,7 @@ 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 +723,7 @@ 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 +743,28 @@ 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 +775,7 @@ 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 +795,157 @@ 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->kind == AUDIODEV_DRIVER_PA);
 
-    g = g_malloc(sizeof(paaudio));
+    g = g_malloc0(sizeof(paaudio));
     popts = dev->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);
Signature · 1 line
2.4.5
Reviewed-by on v1 15/25
#28 lore raw

Re: [Qemu-devel] [PATCH 15/25] paaudio: do not create multiple connections to the same server

Hi

Quoted 209 lines · Zoltán
---
 audio/paaudio.c | 301 ++++++++++++++++++++++++++++++++------------------------
 1 file changed, 175 insertions(+), 126 deletions(-)
audio/paaudio.c +45 -33
index a53aaf6..e3b8207 100644
@@ -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 +495,52 @@ 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);

audio/paaudio.c static pa_stream *qpa_simple_new ( · line 519

Quoted 10 lines
audio/paaudio.c +10 -0
+
+    flags =
+        PA_STREAM_INTERPOLATE_TIMING
+#ifdef PA_STREAM_ADJUST_LATENCY
+        |PA_STREAM_ADJUST_LATENCY
+#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;

This is unrelated, and I don't think it's justified, imho user should be allowed to move the stream later if needed.

Quoted 356 lines
audio/paaudio.c +120 -94
+    }
 
     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 +556,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
     paaudio *g = pa->g = drv_opaque;
     AudiodevPaOptions *popts = g->dev->pa;
     AudiodevPaPerDirectionOptions *ppdo = popts->sink;
+    PAConnection *c = g->conn;
 
     ss.format = audfmt_to_pa (as->fmt, as->endianness);
     ss.channels = as->nchannels;
@@ -560,7 +574,7 @@ 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 +626,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
     paaudio *g = pa->g = drv_opaque;
     AudiodevPaOptions *popts = g->dev->pa;
     AudiodevPaPerDirectionOptions *ppdo = popts->source;
+    PAConnection *c = g->conn;
 
     ss.format = audfmt_to_pa (as->fmt, as->endianness);
     ss.channels = as->nchannels;
@@ -620,7 +635,7 @@ 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 +723,7 @@ 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 +743,28 @@ 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 +775,7 @@ 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 +795,157 @@ 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->kind == AUDIODEV_DRIVER_PA);
 
-    g = g_malloc(sizeof(paaudio));
+    g = g_malloc0(sizeof(paaudio));
     popts = dev->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);
--
Quoted 3 lines

2.4.5

otherwise

Signature · 1 line
Marc-André Lureau
#29 lore raw

Re: [Qemu-devel] [PATCH 15/25] paaudio: do not create multiple connections to the same server

Hi,

2015-08-20 21:38 keltezéssel, Marc-André Lureau írta:
[snip]

Quoted 4 lines
Quoted 4 lines
audio/paaudio.c +4 -0
+    flags =
+        PA_STREAM_INTERPOLATE_TIMING
+#ifdef PA_STREAM_ADJUST_LATENCY
+        |PA_STREAM_ADJUST_LATENCY
Quoted 8 lines

audio/paaudio.c static pa_stream *qpa_simple_new ( · line 519

Quoted 5 lines
audio/paaudio.c +5 -0
+#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;

This is unrelated, and I don't think it's justified, imho user should be allowed to move the stream later if needed.

True, I will remove it from this commit. But not sure if I want to remove it completely, because it looks like pulseaudio has a tendency to automatically move the stream on connect, even if you explicitly specify a sink/source when connecting. (And the problem is that all streams appear under the name qemu in pulseaudio, so once the user moves one stream, pulseaudio will assume all streams subsequently created by qemu are the same on, and move them too. Maybe we shold add some extra options to the pa backend?)

#30 lore raw

Re: [Qemu-devel] [PATCH 15/25] paaudio: do not create multiple connections to the same server

Quoted 8 lines · Kővágó Zoltán

True, I will remove it from this commit. But not sure if I want to remove it completely, because it looks like pulseaudio has a tendency to automatically move the stream on connect, even if you explicitly specify a sink/source when connecting. (And the problem is that all streams appear under the name qemu in pulseaudio, so once the user moves one stream, pulseaudio will assume all streams subsequently created by qemu are the same on, and move them too. Maybe we shold add some extra options to the pa backend?)

In theory, stream-restore shouldn't move streams if the sink is already specified, see sink_input_new_hook_callback() in PA module-stream-restore.c, but there might be something else applying other policies? Yes, qemu could provide more context to the streams with additional properties, which could help the policy. I think this can be discussed seperately.

Signature · 1 line
Marc-André Lureau
#31 lore raw

PATCH 16/25 paaudio: properly disconnect streams in fini_*

In reply to Kővágó, Zoltán

Currently this needs a workaround due to bug #74624 in pulseaudio.

---
 audio/paaudio.c | 23 +++++++++++++++++++++--
 1 file changed, 21 insertions(+), 2 deletions(-)
audio/paaudio.c +21 -2
index e3b8207..2aee22f 100644
@@ -678,6 +678,25 @@ 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;
@@ -689,7 +708,7 @@ 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;
     }
 
@@ -709,7 +728,7 @@ 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;
     }
 
Signature · 1 line
2.4.5
Reviewed-by on v1 16/25
#32 lore raw

Re: [Qemu-devel] [PATCH 16/25] paaudio: properly disconnect streams in fini_*

Quoted 59 lines · Zoltán

Currently this needs a workaround due to bug #74624 in pulseaudio.

---
 audio/paaudio.c | 23 +++++++++++++++++++++--
 1 file changed, 21 insertions(+), 2 deletions(-)
audio/paaudio.c +21 -3
index e3b8207..2aee22f 100644
@@ -678,6 +678,25 @@ 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;
@@ -689,7 +708,7 @@ 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;
     }
 
@@ -709,7 +728,7 @@ 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;
     }
 
--

2.4.5

Signature · 1 line
Marc-André Lureau
#33 lore raw

PATCH 17/25 audio: remove gcc specific audio_MIN, audio_MAX

In reply to Kővágó, Zoltán

Currently the gcc specific version only evaluates the arguments once, while the generic version evaluates one argument twice, which can cause debugging headaches when an argument has a side effect. This patch at least provides consistent behavior between compilers.

---
 audio/audio.h | 14 --------------
 1 file changed, 14 deletions(-)
audio/audio.h +0 -14
index 68545b6..3a54e17 100644
@@ -150,22 +150,8 @@ 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);
Signature · 1 line
2.4.5
#34 lore raw

Re: [Qemu-devel] [PATCH 17/25] audio: remove gcc specific audio_MIN, audio_MAX

Quoted 3 lines · Zoltán

Currently the gcc specific version only evaluates the arguments once, while the generic version evaluates one argument twice, which can cause debugging headaches when an argument has a side effect.

The answer to that is "don't do that". Do we have macro calls with side effects in the tree?

Quoted 1 line

This patch at least provides consistent behavior between compilers.

Makes sense.

audio/audio.h static inline void *advance (void *p, int incr) · line 168

Quoted 4 lines
audio/audio.h +0 -2
-#else
 #define audio_MIN(a, b) ((a)>(b)?(b):(a))
 #define audio_MAX(a, b) ((a)<(b)?(b):(a))
-#endif

include/qemu/osdep.h already provides MIN/MAX macros.

I think we should either define audio_MIN (and audio_MAX) to those, or simply do s/audio_MIN/MIN/ in audio/*.c

cheers,
Gerd

#35 lore raw

Re: [Qemu-devel] [PATCH 17/25] audio: remove gcc specific audio_MIN, audio_MAX

Quoted 19 lines · Gerd Hoffmann
Quoted 3 lines · Zoltán

Currently the gcc specific version only evaluates the arguments once, while the generic version evaluates one argument twice, which can cause debugging headaches when an argument has a side effect.

The answer to that is "don't do that". Do we have macro calls with side effects in the tree?

Quoted 1 line

This patch at least provides consistent behavior between compilers.

Makes sense.

audio/audio.h static inline void *advance (void *p, int incr) · line 168

Quoted 4 lines
audio/audio.h +0 -2
-#else
 #define audio_MIN(a, b) ((a)>(b)?(b):(a))
 #define audio_MAX(a, b) ((a)<(b)?(b):(a))
-#endif

include/qemu/osdep.h already provides MIN/MAX macros.

Quoted 2 lines

I think we should either define audio_MIN (and audio_MAX) to those, or simply do s/audio_MIN/MIN/ in audio/*.c

My vote is for the latter. Incidentally we already assume both typeof and statement expr support in our compilers, so we could upgrade our local MIN/MAX implementations to use them if we really needed. (We'd have to rename them though, since the system implementation likely does the eval-twice thing.)

A quick grep doesn't show any audio_MIN/MAX which need to avoid multiple-evaluation, though.

thanks
-- PMM

Reviewed-by on v1 17/25
#36 lore raw

Re: [Qemu-devel] [PATCH 17/25] audio: remove gcc specific audio_MIN, audio_MAX

In reply to Kővágó, Zoltán

Hi

Quoted 5 lines · Zoltán

Currently the gcc specific version only evaluates the arguments once, while the generic version evaluates one argument twice, which can cause debugging headaches when an argument has a side effect. This patch at least provides consistent behavior between compilers.

Going this way, you could simply replace audio_MIN/MAX with MIN/MAX (from osdep.h and glib headers)

Quoted 36 lines
---
 audio/audio.h | 14 --------------
 1 file changed, 14 deletions(-)
audio/audio.h +0 -15
index 68545b6..3a54e17 100644
@@ -150,22 +150,8 @@ 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);
--

2.4.5

Signature · 1 line
Marc-André Lureau
#37 lore raw

PATCH 18/25 audio: do not run each backend in audio_run

In reply to Kővágó, Zoltán

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.

---
 audio/alsaaudio.c |  7 +++++--
 audio/audio.c     | 17 +++++++----------
 audio/audio_int.h |  2 +-
 audio/ossaudio.c  | 12 ++++++------
 4 files changed, 19 insertions(+), 19 deletions(-)
audio/alsaaudio.c +5 -2
index cfe4aec..0750d0d 100644
@@ -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;
audio/audio.c +7 -10
index 7468b94..e213deb 100644
@@ -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
     {
audio/audio_int.h +1 -1
index 1d81658..c52c656 100644
@@ -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
audio/ossaudio.c +6 -6
index a5e7f7c..9eae769 100644
@@ -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)
Signature · 1 line
2.4.5
Reviewed-by on v1 18/25
#38 lore raw

Re: [Qemu-devel] [PATCH 18/25] audio: do not run each backend in audio_run

Hi

Quoted 16 lines · Zoltán

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.

---
 audio/alsaaudio.c |  7 +++++--
 audio/audio.c     | 17 +++++++----------
 audio/audio_int.h |  2 +-
 audio/ossaudio.c  | 12 ++++++------
 4 files changed, 19 insertions(+), 19 deletions(-)
audio/alsaaudio.c +0 -0
index cfe4aec..0750d0d 100644

audio/alsaaudio.c struct pollhlp { · line 43

Quoted 4 lines
audio/alsaaudio.c +1 -0
@@ -40,6 +40,7 @@ struct pollhlp {
     struct pollfd *pfds;
     int count;
     int mask;
+    AudioState *s;

Instead of adding that pointer, you could lookup the hw->s using container_of()

Quoted 122 lines
audio/alsaaudio.c +4 -2
 };
 
 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;
audio/audio.c +7 -10
index 7468b94..e213deb 100644
@@ -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
     {
audio/audio_int.h +1 -1
index 1d81658..c52c656 100644
@@ -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
audio/ossaudio.c +6 -7
index a5e7f7c..9eae769 100644
@@ -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)
--
Quoted 3 lines

2.4.5

other than that,

Signature · 1 line
Marc-André Lureau
#39 lore raw

Re: [Qemu-devel] [PATCH 18/25] audio: do not run each backend in audio_run

Hi

2015-08-20 22:49 keltezéssel, Marc-André Lureau írta:

Quoted 14 lines

Hi

Quoted 11 lines · Zoltán

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.

---
  audio/alsaaudio.c |  7 +++++--
  audio/audio.c     | 17 +++++++----------
  audio/audio_int.h |  2 +-
  audio/ossaudio.c  | 12 ++++++------
  4 files changed, 19 insertions(+), 19 deletions(-)
Quoted 11 lines

audio/alsaaudio.c struct pollhlp { · line 43

Quoted 9 lines
audio/alsaaudio.c +1 -0
index cfe4aec..0750d0d 100644
@@ -40,6 +40,7 @@ struct pollhlp {
      struct pollfd *pfds;
      int count;
      int mask;
+    AudioState *s;

Instead of adding that pointer, you could lookup the hw->s using container_of()

I'm afraid that won't work. The pollhlp is used both inside ALSAVoiceIn and ALSAVoiceOut, but they have a common callback.

Quoted 132 lines
Quoted 123 lines
audio/alsaaudio.c +4 -2
  };
 
  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;
audio/audio.c +7 -10
index 7468b94..e213deb 100644
@@ -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
      {
audio/audio_int.h +1 -1
index 1d81658..c52c656 100644
@@ -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
audio/ossaudio.c +6 -7
index a5e7f7c..9eae769 100644
@@ -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)
--
Quoted 3 lines

2.4.5

other than that,

#40 lore raw

PATCH 19/25 paaudio: fix playback glitches

In reply to Kővágó, Zoltán

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.

---
 audio/paaudio.c | 6 ++----
 1 file changed, 2 insertions(+), 4 deletions(-)
audio/paaudio.c +2 -4
index 2aee22f..3990a80 100644
@@ -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 */
         flags |= PA_STREAM_DONT_MOVE;
Signature · 1 line
2.4.5
#41 lore raw

Re: [Qemu-devel] [PATCH 19/25] paaudio: fix playback glitches

Hi

Quoted 7 lines · Zoltán

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.

Afaik, it's not a problem if the audio data is not available right away (live sources): PulseAudio should deal with that. I think it will simply silence and wait (but it could actually be smarter than that?)

PA_STREAM_EARLY_REQUESTS is going to make PA request data more frequently (every minreq). I don't get why that would help here. It's meant for bad behaving applications using usleep and such that aren't fixable, not a solution for audio glitches.

If qemu produces too much data than what PA can consume, it should drop the old data, and only play current audio (unfortunately, that doesn't seem to be the case looking at audio_pcm_sw_write). If qemu produces too little, it can still write whatever it has when it has it.

Could you provide a way to reproduce the glitch you experience or explain more clearly how this flag helps?

thanks

Quoted 27 lines
---
 audio/paaudio.c | 6 ++----
 1 file changed, 2 insertions(+), 4 deletions(-)
audio/paaudio.c +2 -5
index 2aee22f..3990a80 100644
@@ -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 */
         flags |= PA_STREAM_DONT_MOVE;
--

2.4.5

Signature · 1 line
Marc-André Lureau
#42 lore raw

Re: [Qemu-devel] [PATCH 19/25] paaudio: fix playback glitches

2015-08-21 00:08 keltezéssel, Marc-André Lureau írta:

Quoted 11 lines

Hi

Quoted 7 lines · Zoltán

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.

Quoted 3 lines

Afaik, it's not a problem if the audio data is not available right away (live sources): PulseAudio should deal with that. I think it will simply silence and wait (but it could actually be smarter than that?)

Yes, it will simply silence and wait, until more data arrives. The problem is that thing happens, like 10 times each second. (If I watch pacmd list-sink-inputs, I can see that status quickly varies between drained and running, while playing an audio file in the guest)

Quoted 4 lines

PA_STREAM_EARLY_REQUESTS is going to make PA request data more frequently (every minreq). I don't get why that would help here. It's meant for bad behaving applications using usleep and such that aren't fixable, not a solution for audio glitches.

The problem is that normally pulseaudio sends requests for data very late -- assuming that the application will quickly generate the required samples and send them back. The problem is that the audio subsystem only "runs" each 10 ms, so it may take up to 10ms until qemu sends back data after the request, which is apparently too much. With PA_STREAM_EARLY_REQUEST pulseaudio will request data as soon as it has enough free space in it's buffer, and not wait until the last moment.

Maybe tweaking pa buffer settings would help too, I'm not sure. And yes, qemu is pretty much a bad behaving application that, while not using usleep directly, only produces audio every 10ms (by default).

Quoted 9 lines

If qemu produces too much data than what PA can consume, it should drop the old data, and only play current audio (unfortunately, that doesn't seem to be the case looking at audio_pcm_sw_write). If qemu produces too little, it can still write whatever it has when it has it.

Could you provide a way to reproduce the glitch you experience or explain more clearly how this flag helps?

It happens for me with a pretty much a default pulseaudio configuration, even with the stable release. For example day24 from last year qemu advent calendar with pulseaudio backend. (Sometimes it fixes itself after some time, very rarely it work normally, but most cases I just get that constant buffer underflow.)

Quoted 34 lines

thanks

Quoted 27 lines
---
  audio/paaudio.c | 6 ++----
  1 file changed, 2 insertions(+), 4 deletions(-)
audio/paaudio.c +2 -5
index 2aee22f..3990a80 100644
@@ -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 */
          flags |= PA_STREAM_DONT_MOVE;
--

2.4.5

#43 lore raw

PATCH 20/25 audio: remove mixeng specific code from backends

In reply to Kővágó, Zoltán

Backends no longer have to deal with mixeng, they just receive a buffer in the correct sample format, all mixeng logic is now in the audio.c (and mixeng.c). Backends also do not have to deal with soft voices.

Backends now have two way to read/write sound:
* write and read functions: similar to old read/write functions, except that they actually read/write the data to the backend instead of placing it into the mixeng buffer. You no longer need run_in/run_out afterwards.
* get_buffer_out/put_buffer_out: the first function returns a buffer that can hold some audio data. The caller fills this buffer (maybe partially) and calls put_buffer to actually write the data. This way we can save copying the buffer in some cases (for example mmaped audio). Similarly there's get_buffer_in/put_buffer_in for reading.

Backends only have to support one access method, but they can support both if they have efficient implementation for both cases.

Please note that the read/write functions are not used directly in this commit yet.

---

There's one thing I'm not sure about: backends used to return in hw->samples a buffer size, and mixeng allocated that large buffer for itself. The old code pretty much required that the mixeng buffer and the destination buffer have the same size, but it's no longer required. So maybe it would be a better idea to drop buffer_size_* functions (the replacement of old hw->samples) and specify it as a (global) -audiodev option.

Also please double check that coreaudio backend works. I do not have a mac to test.

---
 audio/Makefile.objs     |   1 -
 audio/alsaaudio.c       | 334 ++++++++++++---------------------------
 audio/audio.c           | 412 +++++++++++++++++++++++++++++++++---------------
 audio/audio.h           |   4 +-
 audio/audio_int.h       |  74 +++++----
 audio/audio_pt_int.c    | 173 --------------------
 audio/audio_pt_int.h    |  22 ---
 audio/audio_template.h  |  31 ++--
 audio/coreaudio.c       | 138 ++++++++--------
 audio/dsound_template.h |  53 ++++---
 audio/dsoundaudio.c     | 344 ++++++++++------------------------------
 audio/mixeng.h          |  11 +-
 audio/noaudio.c         |  54 ++-----
 audio/ossaudio.c        | 322 +++++++++++++++----------------------
 audio/paaudio.c         | 401 +++++++---------------------------------------
 audio/rate_template.h   |   2 +-
 audio/sdlaudio.c        | 118 ++++++--------
 audio/spiceaudio.c      | 134 ++++++----------
 audio/wavaudio.c        |  60 +------
 configure               |   5 -
 20 files changed, 938 insertions(+), 1755 deletions(-)
 delete mode 100644 audio/audio_pt_int.c
 delete mode 100644 audio/audio_pt_int.h
audio/Makefile.objs +0 -1
index 9d8f579..c7216a7 100644
@@ -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
 
audio/alsaaudio.c +97 -237

424 lines not on this page · raw message

audio/audio.c +286 -126

708 lines not on this page · raw message

audio/audio.h +2 -2
index 3a54e17..2e3d0db 100644
@@ -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);
 
audio/audio_int.h +45 -29

131 lines not on this page · raw message

audio/audio_pt_int.c deleted +0 -173

173 lines not on this page · raw message

audio/audio_pt_int.h deleted +0 -22
deleted file mode 100644
index 0dfff76..0000000
@@ -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 */
audio/audio_template.h +12 -19
index 5a3dc90..473eac9 100644
@@ -71,20 +71,22 @@ 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;
 }
 
-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 (%d samples)\n",
-               hw->samples);
-        return -1;
+    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 */
     }
 
-    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)
@@ -103,7 +105,7 @@ 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) {
@@ -264,11 +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=%d\n", hw->samples);
-        goto err1;
-    }
-
 #ifdef DAC
     hw->clip = mixeng_clip
 #else
@@ -279,9 +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;
@@ -290,9 +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;
 }
audio/coreaudio.c +74 -64

205 lines not on this page · raw message

audio/dsound_template.h +30 -23

122 lines not on this page · raw message

audio/dsoundaudio.c +83 -261

413 lines not on this page · raw message

audio/mixeng.h +7 -4
index 9de443b..013d894 100644
@@ -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);
audio/noaudio.c +12 -42

103 lines not on this page · raw message

audio/ossaudio.c +123 -199

454 lines not on this page · raw message

audio/paaudio.c +60 -341

125 lines not on this page · raw message

audio/rate_template.h +1 -1
index bd4b1c7..dc5e021 100644

Lines of this file are not shown. View the raw message.

audio/sdlaudio.c +50 -68

118 lines not on this page · raw message

audio/spiceaudio.c +48 -86

134 lines not on this page · raw message

audio/wavaudio.c +8 -52

60 lines not on this page · raw message

configure +0 -5
index cd219d8..16bda1d 100755

Lines of this file are not shown. View the raw message.

The rest of this diff is not shown. View the raw message.

Signature · 1 line
2.4.5

Part of the diff is not shown here. View the raw message.

#44 lore raw

Re: [Qemu-devel] [PATCH 20/25] audio: remove mixeng specific code from backends

Quoted 15 lines · Zoltán

Backends no longer have to deal with mixeng, they just receive a buffer in the correct sample format, all mixeng logic is now in the audio.c (and mixeng.c). Backends also do not have to deal with soft voices.

Backends now have two way to read/write sound:
* write and read functions: similar to old read/write functions, except that they actually read/write the data to the backend instead of placing it into the mixeng buffer. You no longer need run_in/run_out afterwards.
* get_buffer_out/put_buffer_out: the first function returns a buffer that can hold some audio data. The caller fills this buffer (maybe partially) and calls put_buffer to actually write the data. This way we can save copying the buffer in some cases (for example mmaped audio). Similarly there's get_buffer_in/put_buffer_in for reading.

Quoted 2 lines

Backends only have to support one access method, but they can support both if they have efficient implementation for both cases.

Phew, this one became pretty big. Could this be splitted up? The s/int/size_t/ should be easy to separate.

Also having individual patches for individual backends would be nice. Possibly you can first add the new interfaces, & helper functions, then switch backends over one by one, hooking the generic audio_pcm_hw_run_out() function into run_out callback, finally remove the old interfaces and call audio_pcm_hw_run_out directly.

cheers,
Gerd

#45 lore raw

Re: [Qemu-devel] [PATCH 20/25] audio: remove mixeng specific code from backends

Hi

This commit fails to compile with errors such as:

audio/audio.c: In function ‘audio_pcm_hw_run_in’:
audio/audio.c:1143:41: error: ‘struct audio_pcm_info’ has no member named ‘bytes_per_frame’

         size_t size = samples * hw->info.bytes_per_frame;
#46 lore raw

PATCH 21/25 audio: common rate control code for timer based outputs

In reply to Kővágó, Zoltán

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.

---
 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(-)
audio/audio.c +30 -0
index 6ef40e9..94ffd72 100644
@@ -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 = audio_MIN(samples << info->shift, bytes_avail);
+    rate->bytes_sent += ret;
+    return ret;
+}
audio/audio_int.h +9 -0
index 1250012..16ffa12 100644
@@ -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
audio/noaudio.c +22 -27

93 lines not on this page · raw message

audio/spiceaudio.c +7 -42

100 lines not on this page · raw message

audio/wavaudio.c +10 -10
index cabbd76..14afaaf 100644
@@ -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 = audio_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;
 }
 
Signature · 1 line
2.4.5
#47 lore raw

Re: [Qemu-devel] [PATCH 21/25] audio: common rate control code for timer based outputs

Hi

audio/spiceaudio.c static void line_out_fini (HWVoiceOut *hw) · line 137

Quoted 9 lines · Zoltán
audio/spiceaudio.c +2 -4
-    decr = rate_get_samples(&hw->info, &out->rate);
-    decr = audio_MIN(out->fsize - out->fpos, decr);
-
-    *size = decr << 2;
+    *size = audio_rate_get_bytes(&hw->info, &out->rate,
+                                 (out->fsize - out->fpos) << 2);
     return out->frame + out->fpos;
 }

I am not familiar with this "rate" thing, but you are changing the way it works for Spice without explaining why, and that worries me a little.

The rate->bytes_sent is not updated the same way, the result is not always << 2..

I really don't get what this is really doing tbh, some kind of clever buffer limit :)

Signature · 1 line
Marc-André Lureau
#48 lore raw

Re: [Qemu-devel] [PATCH 21/25] audio: common rate control code for timer based outputs

Hi,

2015-08-21 00:37 keltezéssel, Marc-André Lureau írta:

Quoted 17 lines

Hi

audio/spiceaudio.c static void line_out_fini (HWVoiceOut *hw) · line 137

Quoted 9 lines · Zoltán
audio/spiceaudio.c +2 -4
-    decr = rate_get_samples(&hw->info, &out->rate);
-    decr = audio_MIN(out->fsize - out->fpos, decr);
-
-    *size = decr << 2;
+    *size = audio_rate_get_bytes(&hw->info, &out->rate,
+                                 (out->fsize - out->fpos) << 2);
      return out->frame + out->fpos;
  }

I am not familiar with this "rate" thing, but you are changing the way it works for Spice without explaining why, and that worries me a little.

Quoted 6 lines

The rate->bytes_sent is not updated the same way, the result is not always << 2..

I really don't get what this is really doing tbh, some kind of clever buffer limit :)

The old code pretty much ignored if the buffer overflow (which shouldn't happen normally), while the new one doesn't. The problem is that with this new get_buffer/put_buffer thing if we reach the end of the circular buffer, it will produce have the same effect as a buffer overflow, but we need to handle it more gracefully as it will now happen every now and then.

But probably I'll change << 2 to info.shift, to handle more gracefully if the spice stream format changes.

Zoltan

#49 lore raw

PATCH 22/25 audio: add mixeng option (documentation)

In reply to Kővágó, Zoltán

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.

---
 qapi/audio.json | 5 +++++
 qemu-options.hx | 6 ++++++
 2 files changed, 11 insertions(+)
qapi/audio.json +5 -0
index b57b215..0216a10 100644
@@ -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',
qemu-options.hx +6 -0
index 60a3563..bd922bf 100644
@@ -329,6 +329,7 @@ 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"
@@ -397,6 +398,11 @@ 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
Signature · 1 line
2.4.5
#50 lore raw

Re: [Qemu-devel] [PATCH 22/25] audio: add mixeng option (documentation)

Hi

Quoted 14 lines · Zoltán

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

Quoted 5 lines

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.

Since the setting is so complicated, shouldn't it be a OnOffAuto? and let the backend decide what's best for the user?

Quoted 60 lines

However mixeng is not removed, only made optional, so this shouldn't be a big concern.

---
 qapi/audio.json | 5 +++++
 qemu-options.hx | 6 ++++++
 2 files changed, 11 insertions(+)
qapi/audio.json +5 -0
index b57b215..0216a10 100644
@@ -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',
qemu-options.hx +6 -1
index 60a3563..bd922bf 100644
@@ -329,6 +329,7 @@ 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"
@@ -397,6 +398,11 @@ 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
--

2.4.5

Signature · 1 line
Marc-André Lureau
#51 lore raw

Re: [Qemu-devel] [PATCH 22/25] audio: add mixeng option (documentation)

2015-08-21 00:49 keltezéssel, Marc-André Lureau írta:

Quoted 17 lines

Hi

Quoted 14 lines · Zoltán

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

Quoted 8 lines
Quoted 5 lines

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.

Since the setting is so complicated, shouldn't it be a OnOffAuto? and let the backend decide what's best for the user?

Because of the items under disadvantages, the default must be on for backward compatibility. And how would we choose the auto one? For the time being it's only useful in some specific cases (like my upcoming 5.1 support), so we should probably stay with OnOff (adding an auto option is possible later), expecting the user knows what he/she is doing if he/she disables it.

Quoted 64 lines
Quoted 61 lines

However mixeng is not removed, only made optional, so this shouldn't be a big concern.

---
  qapi/audio.json | 5 +++++
  qemu-options.hx | 6 ++++++
  2 files changed, 11 insertions(+)
qapi/audio.json +5 -0
index b57b215..0216a10 100644
@@ -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',
qemu-options.hx +6 -1
index 60a3563..bd922bf 100644
@@ -329,6 +329,7 @@ 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"
@@ -397,6 +398,11 @@ 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
--

2.4.5

#52 lore raw

PATCH 23/25 audio: make mixeng optional

In reply to Kővágó, Zoltán

Implementation of the previously added mixeng option.

---
 audio/audio.c          | 70 +++++++++++++++++++++++++++++++++++++++++++++-----
 audio/audio_template.h | 22 +++++++++++-----
 2 files changed, 79 insertions(+), 13 deletions(-)
audio/audio.c +64 -6
index 94ffd72..ee02b5c 100644
@@ -794,32 +794,46 @@ 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)
@@ -1033,6 +1047,26 @@ 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;
@@ -1167,6 +1201,17 @@ 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;
@@ -1643,6 +1688,11 @@ 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);
@@ -1800,9 +1850,13 @@ 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)) {
@@ -1810,6 +1864,10 @@ 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;
@@ -1821,7 +1879,7 @@ 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;
audio/audio_template.h +15 -7
index 473eac9..942d9a4 100644
@@ -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));
Signature · 1 line
2.4.5
#53 lore raw

Re: [Qemu-devel] [PATCH 23/25] audio: make mixeng optional

Fails to compile:

audio/audio.c: In function ‘audio_run_out’:
audio/audio.c:1058:32: error: ‘struct audio_pcm_ops’ has no member named ‘enable_out’

                 if (hw->pcm_ops->enable_out) {
#54 lore raw

PATCH 24/25 paaudio: get/put_buffer functions

In reply to Kővágó, Zoltán

This lets us avoid some buffer copying when using mixeng.

---
 audio/paaudio.c | 83 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 83 insertions(+)
audio/paaudio.c +83 -0
index a0b17c8..97e12ba 100644
@@ -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 = audio_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;
@@ -692,12 +771,16 @@ 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 */
     .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,
+    .get_buffer_in = qpa_get_buffer_in,
+    .put_buffer_in = qpa_put_buffer_in,
     .ctl_in   = qpa_ctl_in
 };
 
Signature · 1 line
2.4.5
#55 lore raw

PATCH 25/25 audio: split ctl_* functions into enable_* and volume_*

In reply to Kővágó, Zoltán

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.

---
 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(-)
audio/alsaaudio.c +22 -34
index b7ab540..4797863 100644
@@ -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->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->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)
@@ -927,13 +915,13 @@ 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 = {
audio/audio.c +28 -17

117 lines not on this page · raw message

audio/audio_int.h +4 -11
index 16ffa12..6f2f309 100644
@@ -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);
audio/audio_template.h +0 -1
index 942d9a4..f3cbc66 100644
@@ -264,7 +264,6 @@ 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
audio/coreaudio.c +4 -9
index d6c7636..bf35b0b 100644
@@ -473,13 +473,12 @@ 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, audioDeviceIOProc);
@@ -487,9 +486,7 @@ 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)) {
@@ -499,9 +496,7 @@ static int coreaudio_ctl_out (HWVoiceOut *hw, int cmd, ...)
                 }
             }
         }
-        break;
     }
-    return 0;
 }
 
 static void *coreaudio_audio_init(Audiodev *dev)
@@ -521,7 +516,7 @@ 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 = {
audio/dsoundaudio.c +20 -30

140 lines not on this page · raw message

audio/noaudio.c +6 -8
index 3cb3a89..0618b60 100644
@@ -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 = {
audio/ossaudio.c +31 -47

135 lines not on this page · raw message

audio/paaudio.c +51 -77

172 lines not on this page · raw message

audio/sdlaudio.c +3 -14
index ee5ab92..5821341 100644
@@ -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 = {
audio/spiceaudio.c +42 -59

163 lines not on this page · raw message

audio/wavaudio.c +3 -4
index 14afaaf..5b35173 100644
@@ -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 = {
Signature · 1 line
2.4.5
#56 lore raw

Re: [Qemu-devel] [PATCH 25/25] audio: split ctl_* functions into enable_* and volume_*

nice! :)

Looks good, I'll check more thoroughly when you will have an updated series.