* [Qemu-devel] [PATCH v6 01/14] qapi: qapi for audio backends
2019-03-08 22:34 [Qemu-devel] [PATCH v6 00/14] Audio patches Kővágó, Zoltán
@ 2019-03-08 22:34 ` Kővágó, Zoltán
2019-03-09 9:27 ` Markus Armbruster
2019-03-08 22:34 ` [Qemu-devel] [PATCH v6 02/14] audio: use qapi AudioFormat instead of audfmt_e Kővágó, Zoltán
` (13 subsequent siblings)
14 siblings, 1 reply; 18+ messages in thread
From: Kővágó, Zoltán @ 2019-03-08 22:34 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann, Markus Armbruster, Michael Roth, Eric Blake
This patch adds structures into qapi to replace the existing
configuration structures used by audio backends currently. This qapi
will be the base of the -audiodev command line parameter (that replaces
the old environment variables based config).
This is not a 1:1 translation of the old options, I've tried to make
them much more consistent (e.g. almost every backend had an option to
specify buffer size, but the name was different for every backend, and
some backends required usecs, while some other required frames, samples
or bytes). Also tried to reduce the number of abbreviations used by the
config keys.
Some of the more important changes:
* use `in` and `out` instead of `ADC` and `DAC`, as the former is more
user friendly imho
* moved buffer settings into the global setting area (so it's the same
for all backends that support it. Backends that can't change buffer
size will simply ignore them). Also using usecs, as it's probably more
user friendly than samples or bytes.
* try-poll is now an alsa backend specific option (as all other backends
currently ignore it)
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Reviewed-by: Markus Armbruster <armbru@redhat.com>
---
Notes:
Changes from v5:
* documentation fixes
* renamed buffer-len to buffer-length and period-len to period-length
Changes from v4:
* documentation fixes
* renamed pa's source/sink to pa-in/pa-out
* per-direction options changed per Markus Armbruster's comments
Changes from v2:
* update copyright, version numbers
* remove #optional
* per-direction options are now optional (needed for qobject_object_visitor_new_str)
* removed unnecessary AudiodevNoOptions
* changed integers to unsigned
qapi/audio.json | 304 ++++++++++++++++++++++++++++++++++++++++++
qapi/qapi-schema.json | 1 +
qapi/Makefile.objs | 6 +-
3 files changed, 308 insertions(+), 3 deletions(-)
create mode 100644 qapi/audio.json
diff --git a/qapi/audio.json b/qapi/audio.json
new file mode 100644
index 0000000000..97aee37288
--- /dev/null
+++ b/qapi/audio.json
@@ -0,0 +1,304 @@
+# -*- mode: python -*-
+#
+# Copyright (C) 2015-2019 Zoltán Kővágó <DirtY.iCE.hu@gmail.com>
+#
+# This work is licensed under the terms of the GNU GPL, version 2 or later.
+# See the COPYING file in the top-level directory.
+
+##
+# @AudiodevPerDirectionOptions:
+#
+# General audio backend options that are used for both playback and
+# recording.
+#
+# @fixed-settings: use fixed settings for host input/output. When off,
+# frequency, channels and format must not be
+# specified (default true)
+#
+# @frequency: frequency to use when using fixed settings
+# (default 44100)
+#
+# @channels: number of channels when using fixed settings (default 2)
+#
+# @voices: number of voices to use (default 1)
+#
+# @format: sample format to use when using fixed settings
+# (default s16)
+#
+# @buffer-length: the buffer length in microseconds
+#
+# Since: 4.0
+##
+{ 'struct': 'AudiodevPerDirectionOptions',
+ 'data': {
+ '*fixed-settings': 'bool',
+ '*frequency': 'uint32',
+ '*channels': 'uint32',
+ '*voices': 'uint32',
+ '*format': 'AudioFormat',
+ '*buffer-length': 'uint32' } }
+
+##
+# @AudiodevGenericOptions:
+#
+# Generic driver-specific options.
+#
+# @in: options of the capture stream
+#
+# @out: options of the playback stream
+#
+# Since: 4.0
+##
+{ 'struct': 'AudiodevGenericOptions',
+ 'data': {
+ '*in': 'AudiodevPerDirectionOptions',
+ '*out': 'AudiodevPerDirectionOptions' } }
+
+##
+# @AudiodevAlsaPerDirectionOptions:
+#
+# Options of the ALSA backend that are used for both playback and
+# recording.
+#
+# @dev: the name of the ALSA device to use (default 'default')
+#
+# @period-length: the period length in microseconds
+#
+# @try-poll: attempt to use poll mode, falling back to non-polling
+# access on failure (default true)
+#
+# Since: 4.0
+##
+{ 'struct': 'AudiodevAlsaPerDirectionOptions',
+ 'base': 'AudiodevPerDirectionOptions',
+ 'data': {
+ '*dev': 'str',
+ '*period-length': 'uint32',
+ '*try-poll': 'bool' } }
+
+##
+# @AudiodevAlsaOptions:
+#
+# Options of the ALSA audio backend.
+#
+# @in: options of the capture stream
+#
+# @out: options of the playback stream
+#
+# @threshold: set the threshold (in microseconds) when playback starts
+#
+# Since: 4.0
+##
+{ 'struct': 'AudiodevAlsaOptions',
+ 'data': {
+ '*in': 'AudiodevAlsaPerDirectionOptions',
+ '*out': 'AudiodevAlsaPerDirectionOptions',
+ '*threshold': 'uint32' } }
+
+##
+# @AudiodevCoreaudioPerDirectionOptions:
+#
+# Options of the Core Audio backend that are used for both playback and
+# recording.
+#
+# @buffer-count: number of buffers
+#
+# Since: 4.0
+##
+{ 'struct': 'AudiodevCoreaudioPerDirectionOptions',
+ 'base': 'AudiodevPerDirectionOptions',
+ 'data': {
+ '*buffer-count': 'uint32' } }
+
+##
+# @AudiodevCoreaudioOptions:
+#
+# Options of the coreaudio audio backend.
+#
+# @in: options of the capture stream
+#
+# @out: options of the playback stream
+#
+# Since: 4.0
+##
+{ 'struct': 'AudiodevCoreaudioOptions',
+ 'data': {
+ '*in': 'AudiodevCoreaudioPerDirectionOptions',
+ '*out': 'AudiodevCoreaudioPerDirectionOptions' } }
+
+##
+# @AudiodevDsoundOptions:
+#
+# Options of the DirectSound audio backend.
+#
+# @in: options of the capture stream
+#
+# @out: options of the playback stream
+#
+# @latency: add extra latency to playback in microseconds
+# (default 10000)
+#
+# Since: 4.0
+##
+{ 'struct': 'AudiodevDsoundOptions',
+ 'data': {
+ '*in': 'AudiodevPerDirectionOptions',
+ '*out': 'AudiodevPerDirectionOptions',
+ '*latency': 'uint32' } }
+
+##
+# @AudiodevOssPerDirectionOptions:
+#
+# Options of the OSS backend that are used for both playback and
+# recording.
+#
+# @dev: file name of the OSS device (default '/dev/dsp')
+#
+# @buffer-count: number of buffers
+#
+# @try-poll: attempt to use poll mode, falling back to non-polling
+# access on failure (default true)
+#
+# Since: 4.0
+##
+{ 'struct': 'AudiodevOssPerDirectionOptions',
+ 'base': 'AudiodevPerDirectionOptions',
+ 'data': {
+ '*dev': 'str',
+ '*buffer-count': 'uint32',
+ '*try-poll': 'bool' } }
+
+##
+# @AudiodevOssOptions:
+#
+# Options of the OSS audio backend.
+#
+# @in: options of the capture stream
+#
+# @out: options of the playback stream
+#
+# @try-mmap: try using memory-mapped access, falling back to
+# non-memory-mapped access on failure (default true)
+#
+# @exclusive: open device in exclusive mode (vmix won't work)
+# (default false)
+#
+# @dsp-policy: 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: 4.0
+##
+{ 'struct': 'AudiodevOssOptions',
+ 'data': {
+ '*in': 'AudiodevOssPerDirectionOptions',
+ '*out': 'AudiodevOssPerDirectionOptions',
+ '*try-mmap': 'bool',
+ '*exclusive': 'bool',
+ '*dsp-policy': 'uint32' } }
+
+##
+# @AudiodevPaPerDirectionOptions:
+#
+# Options of the Pulseaudio backend that are used for both playback and
+# recording.
+#
+# @name: name of the sink/source to use
+#
+# Since: 4.0
+##
+{ 'struct': 'AudiodevPaPerDirectionOptions',
+ 'base': 'AudiodevPerDirectionOptions',
+ 'data': {
+ '*name': 'str' } }
+
+##
+# @AudiodevPaOptions:
+#
+# Options of the PulseAudio audio backend.
+#
+# @in: options of the capture stream
+#
+# @out: options of the playback stream
+#
+# @server: PulseAudio server address (default: let PulseAudio choose)
+#
+# Since: 4.0
+##
+{ 'struct': 'AudiodevPaOptions',
+ 'data': {
+ '*in': 'AudiodevPaPerDirectionOptions',
+ '*out': 'AudiodevPaPerDirectionOptions',
+ '*server': 'str' } }
+
+##
+# @AudiodevWavOptions:
+#
+# Options of the wav audio backend.
+#
+# @in: options of the capture stream
+#
+# @out: options of the playback stream
+#
+# @path: name of the wav file to record (default 'qemu.wav')
+#
+# Since: 4.0
+##
+{ 'struct': 'AudiodevWavOptions',
+ 'data': {
+ '*in': 'AudiodevPerDirectionOptions',
+ '*out': 'AudiodevPerDirectionOptions',
+ '*path': 'str' } }
+
+
+##
+# @AudioFormat:
+#
+# An enumeration of possible audio formats.
+#
+# Since: 4.0
+##
+{ 'enum': 'AudioFormat',
+ 'data': [ 'u8', 's8', 'u16', 's16', 'u32', 's32' ] }
+
+##
+# @AudiodevDriver:
+#
+# An enumeration of possible audio backend drivers.
+#
+# Since: 4.0
+##
+{ 'enum': 'AudiodevDriver',
+ 'data': [ 'none', 'alsa', 'coreaudio', 'dsound', 'oss', 'pa', 'sdl',
+ 'spice', 'wav' ] }
+
+##
+# @Audiodev:
+#
+# Options of an audio backend.
+#
+# @id: identifier of the backend
+#
+# @driver: the backend driver to use
+#
+# @timer-period: timer period (in microseconds, 0: use lowest possible)
+#
+# Since: 4.0
+##
+{ 'union': 'Audiodev',
+ 'base': {
+ 'id': 'str',
+ 'driver': 'AudiodevDriver',
+ '*timer-period': 'uint32' },
+ 'discriminator': 'driver',
+ 'data': {
+ 'none': 'AudiodevGenericOptions',
+ 'alsa': 'AudiodevAlsaOptions',
+ 'coreaudio': 'AudiodevCoreaudioOptions',
+ 'dsound': 'AudiodevDsoundOptions',
+ 'oss': 'AudiodevOssOptions',
+ 'pa': 'AudiodevPaOptions',
+ 'sdl': 'AudiodevGenericOptions',
+ 'spice': 'AudiodevGenericOptions',
+ 'wav': 'AudiodevWavOptions' } }
diff --git a/qapi/qapi-schema.json b/qapi/qapi-schema.json
index a34899c626..4bd1223637 100644
--- a/qapi/qapi-schema.json
+++ b/qapi/qapi-schema.json
@@ -99,3 +99,4 @@
{ 'include': 'introspect.json' }
{ 'include': 'misc.json' }
{ 'include': 'target.json' }
+{ 'include': 'audio.json' }
diff --git a/qapi/Makefile.objs b/qapi/Makefile.objs
index 77acca0209..729e5185c5 100644
--- a/qapi/Makefile.objs
+++ b/qapi/Makefile.objs
@@ -5,9 +5,9 @@ util-obj-y += opts-visitor.o qapi-clone-visitor.o
util-obj-y += qmp-event.o
util-obj-y += qapi-util.o
-QAPI_COMMON_MODULES = authz block-core block char common crypto introspect
-QAPI_COMMON_MODULES += job migration misc net rdma rocker run-state
-QAPI_COMMON_MODULES += sockets tpm trace transaction ui
+QAPI_COMMON_MODULES = audio authz block-core block char common crypto
+QAPI_COMMON_MODULES += introspect job migration misc net rdma rocker
+QAPI_COMMON_MODULES += run-state sockets tpm trace transaction ui
QAPI_TARGET_MODULES = target
QAPI_MODULES = $(QAPI_COMMON_MODULES) $(QAPI_TARGET_MODULES)
--
2.20.1
^ permalink raw reply related [flat|nested] 18+ messages in thread* Re: [Qemu-devel] [PATCH v6 01/14] qapi: qapi for audio backends
2019-03-08 22:34 ` [Qemu-devel] [PATCH v6 01/14] qapi: qapi for audio backends Kővágó, Zoltán
@ 2019-03-09 9:27 ` Markus Armbruster
2019-03-09 15:28 ` Zoltán Kővágó
0 siblings, 1 reply; 18+ messages in thread
From: Markus Armbruster @ 2019-03-09 9:27 UTC (permalink / raw)
To: Kővágó, Zoltán
Cc: qemu-devel, Michael Roth, Gerd Hoffmann
"Kővágó, Zoltán" <dirty.ice.hu@gmail.com> writes:
> This patch adds structures into qapi to replace the existing
> configuration structures used by audio backends currently. This qapi
> will be the base of the -audiodev command line parameter (that replaces
> the old environment variables based config).
>
> This is not a 1:1 translation of the old options, I've tried to make
> them much more consistent (e.g. almost every backend had an option to
> specify buffer size, but the name was different for every backend, and
> some backends required usecs, while some other required frames, samples
> or bytes). Also tried to reduce the number of abbreviations used by the
> config keys.
>
> Some of the more important changes:
> * use `in` and `out` instead of `ADC` and `DAC`, as the former is more
> user friendly imho
> * moved buffer settings into the global setting area (so it's the same
> for all backends that support it. Backends that can't change buffer
> size will simply ignore them). Also using usecs, as it's probably more
> user friendly than samples or bytes.
> * try-poll is now an alsa backend specific option (as all other backends
> currently ignore it)
>
> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
> Reviewed-by: Markus Armbruster <armbru@redhat.com>
> ---
>
> Notes:
> Changes from v5:
>
> * documentation fixes
> * renamed buffer-len to buffer-length and period-len to period-length
>
> Changes from v4:
>
> * documentation fixes
> * renamed pa's source/sink to pa-in/pa-out
> * per-direction options changed per Markus Armbruster's comments
>
> Changes from v2:
>
> * update copyright, version numbers
> * remove #optional
> * per-direction options are now optional (needed for qobject_object_visitor_new_str)
> * removed unnecessary AudiodevNoOptions
> * changed integers to unsigned
>
> qapi/audio.json | 304 ++++++++++++++++++++++++++++++++++++++++++
> qapi/qapi-schema.json | 1 +
> qapi/Makefile.objs | 6 +-
> 3 files changed, 308 insertions(+), 3 deletions(-)
> create mode 100644 qapi/audio.json
>
> diff --git a/qapi/audio.json b/qapi/audio.json
> new file mode 100644
> index 0000000000..97aee37288
> --- /dev/null
> +++ b/qapi/audio.json
> @@ -0,0 +1,304 @@
> +# -*- mode: python -*-
> +#
> +# Copyright (C) 2015-2019 Zoltán Kővágó <DirtY.iCE.hu@gmail.com>
> +#
> +# This work is licensed under the terms of the GNU GPL, version 2 or later.
> +# See the COPYING file in the top-level directory.
> +
> +##
> +# @AudiodevPerDirectionOptions:
> +#
> +# General audio backend options that are used for both playback and
> +# recording.
> +#
> +# @fixed-settings: use fixed settings for host input/output. When off,
> +# frequency, channels and format must not be
> +# specified (default true)
> +#
> +# @frequency: frequency to use when using fixed settings
> +# (default 44100)
> +#
> +# @channels: number of channels when using fixed settings (default 2)
> +#
> +# @voices: number of voices to use (default 1)
> +#
> +# @format: sample format to use when using fixed settings
> +# (default s16)
> +#
> +# @buffer-length: the buffer length in microseconds
The name buffer-length suggests bytes rather than microseconds. That's
why I suggested @buffer-capacity. Matter of taste, up to you and Gerd.
> +#
> +# Since: 4.0
> +##
> +{ 'struct': 'AudiodevPerDirectionOptions',
> + 'data': {
> + '*fixed-settings': 'bool',
> + '*frequency': 'uint32',
> + '*channels': 'uint32',
> + '*voices': 'uint32',
> + '*format': 'AudioFormat',
> + '*buffer-length': 'uint32' } }
[...]
Regardless:
Reviewed-by: Markus Armbruster <armbru@redhat.com>
^ permalink raw reply [flat|nested] 18+ messages in thread* Re: [Qemu-devel] [PATCH v6 01/14] qapi: qapi for audio backends
2019-03-09 9:27 ` Markus Armbruster
@ 2019-03-09 15:28 ` Zoltán Kővágó
0 siblings, 0 replies; 18+ messages in thread
From: Zoltán Kővágó @ 2019-03-09 15:28 UTC (permalink / raw)
To: Markus Armbruster; +Cc: qemu-devel, Michael Roth, Gerd Hoffmann
On 2019-03-09 10:27, Markus Armbruster wrote:
> "Kővágó, Zoltán" <dirty.ice.hu@gmail.com> writes:
>
>> This patch adds structures into qapi to replace the existing
>> configuration structures used by audio backends currently. This qapi
>> will be the base of the -audiodev command line parameter (that replaces
>> the old environment variables based config).
>>
>> This is not a 1:1 translation of the old options, I've tried to make
>> them much more consistent (e.g. almost every backend had an option to
>> specify buffer size, but the name was different for every backend, and
>> some backends required usecs, while some other required frames, samples
>> or bytes). Also tried to reduce the number of abbreviations used by the
>> config keys.
>>
>> Some of the more important changes:
>> * use `in` and `out` instead of `ADC` and `DAC`, as the former is more
>> user friendly imho
>> * moved buffer settings into the global setting area (so it's the same
>> for all backends that support it. Backends that can't change buffer
>> size will simply ignore them). Also using usecs, as it's probably more
>> user friendly than samples or bytes.
>> * try-poll is now an alsa backend specific option (as all other backends
>> currently ignore it)
>>
>> Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
>> Reviewed-by: Markus Armbruster <armbru@redhat.com>
>> ---
>>
>> Notes:
>> Changes from v5:
>>
>> * documentation fixes
>> * renamed buffer-len to buffer-length and period-len to period-length
>>
>> Changes from v4:
>>
>> * documentation fixes
>> * renamed pa's source/sink to pa-in/pa-out
>> * per-direction options changed per Markus Armbruster's comments
>>
>> Changes from v2:
>>
>> * update copyright, version numbers
>> * remove #optional
>> * per-direction options are now optional (needed for qobject_object_visitor_new_str)
>> * removed unnecessary AudiodevNoOptions
>> * changed integers to unsigned
>>
>> qapi/audio.json | 304 ++++++++++++++++++++++++++++++++++++++++++
>> qapi/qapi-schema.json | 1 +
>> qapi/Makefile.objs | 6 +-
>> 3 files changed, 308 insertions(+), 3 deletions(-)
>> create mode 100644 qapi/audio.json
>>
>> diff --git a/qapi/audio.json b/qapi/audio.json
>> new file mode 100644
>> index 0000000000..97aee37288
>> --- /dev/null
>> +++ b/qapi/audio.json
>> @@ -0,0 +1,304 @@
>> +# -*- mode: python -*-
>> +#
>> +# Copyright (C) 2015-2019 Zoltán Kővágó <DirtY.iCE.hu@gmail.com>
>> +#
>> +# This work is licensed under the terms of the GNU GPL, version 2 or later.
>> +# See the COPYING file in the top-level directory.
>> +
>> +##
>> +# @AudiodevPerDirectionOptions:
>> +#
>> +# General audio backend options that are used for both playback and
>> +# recording.
>> +#
>> +# @fixed-settings: use fixed settings for host input/output. When off,
>> +# frequency, channels and format must not be
>> +# specified (default true)
>> +#
>> +# @frequency: frequency to use when using fixed settings
>> +# (default 44100)
>> +#
>> +# @channels: number of channels when using fixed settings (default 2)
>> +#
>> +# @voices: number of voices to use (default 1)
>> +#
>> +# @format: sample format to use when using fixed settings
>> +# (default s16)
>> +#
>> +# @buffer-length: the buffer length in microseconds
>
> The name buffer-length suggests bytes rather than microseconds. That's
> why I suggested @buffer-capacity. Matter of taste, up to you and Gerd.
@period-length has the same issue. For me @buffer-capacity and
@period-capacity feels a bit weird. I'm not 100% familiar with qapi
conventions, but for me 'size' is what's usually in bytes, and 'length'
can be something else (number of characters in a string, length of a
sound file, ...).
>
>> +#
>> +# Since: 4.0
>> +##
>> +{ 'struct': 'AudiodevPerDirectionOptions',
>> + 'data': {
>> + '*fixed-settings': 'bool',
>> + '*frequency': 'uint32',
>> + '*channels': 'uint32',
>> + '*voices': 'uint32',
>> + '*format': 'AudioFormat',
>> + '*buffer-length': 'uint32' } }
> [...]
>
> Regardless:
> Reviewed-by: Markus Armbruster <armbru@redhat.com>
>
Regards,
Zoltan
^ permalink raw reply [flat|nested] 18+ messages in thread
* [Qemu-devel] [PATCH v6 02/14] audio: use qapi AudioFormat instead of audfmt_e
2019-03-08 22:34 [Qemu-devel] [PATCH v6 00/14] Audio patches Kővágó, Zoltán
2019-03-08 22:34 ` [Qemu-devel] [PATCH v6 01/14] qapi: qapi for audio backends Kővágó, Zoltán
@ 2019-03-08 22:34 ` Kővágó, Zoltán
2019-03-08 22:34 ` [Qemu-devel] [PATCH v6 03/14] audio: -audiodev command line option: documentation Kővágó, Zoltán
` (12 subsequent siblings)
14 siblings, 0 replies; 18+ messages in thread
From: Kővágó, Zoltán @ 2019-03-08 22:34 UTC (permalink / raw)
To: qemu-devel
Cc: Gerd Hoffmann, Thomas Huth, Peter Maydell, Michael Walle,
Alistair Francis, Edgar E. Iglesias, Andrzej Zaborowski,
open list:ARM
I had to include an enum for audio sampling formats into qapi, but that
meant duplicating the audfmt_e enum. This patch replaces audfmt_e and
associated values with the qapi generated AudioFormat enum.
This patch is mostly a search-and-replace, except for switches where the
qapi generated AUDIO_FORMAT_MAX caused problems.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Reviewed-by: Thomas Huth <thuth@redhat.com>
---
audio/audio.h | 12 +----
audio/alsaaudio.c | 53 +++++++++++----------
audio/audio.c | 97 +++++++++++++++++++++------------------
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 | 6 +--
hw/display/xlnx_dp.c | 2 +-
hw/input/tsc210x.c | 2 +-
hw/usb/dev-audio.c | 2 +-
ui/vnc.c | 26 +++++------
26 files changed, 196 insertions(+), 189 deletions(-)
diff --git a/audio/audio.h b/audio/audio.h
index f4339a185e..02f29a3b3e 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -26,18 +26,10 @@
#define QEMU_AUDIO_H
#include "qemu/queue.h"
+#include "qapi/qapi-types-audio.h"
typedef void (*audio_callback_fn) (void *opaque, int avail);
-typedef enum {
- AUD_FMT_U8,
- AUD_FMT_S8,
- AUD_FMT_U16,
- AUD_FMT_S16,
- AUD_FMT_U32,
- AUD_FMT_S32
-} audfmt_e;
-
#ifdef HOST_WORDS_BIGENDIAN
#define AUDIO_HOST_ENDIANNESS 1
#else
@@ -47,7 +39,7 @@ typedef enum {
struct audsettings {
int freq;
int nchannels;
- audfmt_e fmt;
+ AudioFormat fmt;
int endianness;
};
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 635be73bf4..5bd034267f 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -87,7 +87,7 @@ struct alsa_params_req {
struct alsa_params_obt {
int freq;
- audfmt_e fmt;
+ AudioFormat fmt;
int endianness;
int nchannels;
snd_pcm_uframes_t samples;
@@ -294,16 +294,16 @@ static int alsa_write (SWVoiceOut *sw, void *buf, int len)
return audio_pcm_sw_write (sw, buf, len);
}
-static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
+static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness)
{
switch (fmt) {
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
return SND_PCM_FORMAT_S8;
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
return SND_PCM_FORMAT_U8;
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
if (endianness) {
return SND_PCM_FORMAT_S16_BE;
}
@@ -311,7 +311,7 @@ static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
return SND_PCM_FORMAT_S16_LE;
}
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
if (endianness) {
return SND_PCM_FORMAT_U16_BE;
}
@@ -319,7 +319,7 @@ static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
return SND_PCM_FORMAT_U16_LE;
}
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_S32:
if (endianness) {
return SND_PCM_FORMAT_S32_BE;
}
@@ -327,7 +327,7 @@ static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
return SND_PCM_FORMAT_S32_LE;
}
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_U32:
if (endianness) {
return SND_PCM_FORMAT_U32_BE;
}
@@ -344,58 +344,58 @@ static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt, int endianness)
}
}
-static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt,
+static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt,
int *endianness)
{
switch (alsafmt) {
case SND_PCM_FORMAT_S8:
*endianness = 0;
- *fmt = AUD_FMT_S8;
+ *fmt = AUDIO_FORMAT_S8;
break;
case SND_PCM_FORMAT_U8:
*endianness = 0;
- *fmt = AUD_FMT_U8;
+ *fmt = AUDIO_FORMAT_U8;
break;
case SND_PCM_FORMAT_S16_LE:
*endianness = 0;
- *fmt = AUD_FMT_S16;
+ *fmt = AUDIO_FORMAT_S16;
break;
case SND_PCM_FORMAT_U16_LE:
*endianness = 0;
- *fmt = AUD_FMT_U16;
+ *fmt = AUDIO_FORMAT_U16;
break;
case SND_PCM_FORMAT_S16_BE:
*endianness = 1;
- *fmt = AUD_FMT_S16;
+ *fmt = AUDIO_FORMAT_S16;
break;
case SND_PCM_FORMAT_U16_BE:
*endianness = 1;
- *fmt = AUD_FMT_U16;
+ *fmt = AUDIO_FORMAT_U16;
break;
case SND_PCM_FORMAT_S32_LE:
*endianness = 0;
- *fmt = AUD_FMT_S32;
+ *fmt = AUDIO_FORMAT_S32;
break;
case SND_PCM_FORMAT_U32_LE:
*endianness = 0;
- *fmt = AUD_FMT_U32;
+ *fmt = AUDIO_FORMAT_U32;
break;
case SND_PCM_FORMAT_S32_BE:
*endianness = 1;
- *fmt = AUD_FMT_S32;
+ *fmt = AUDIO_FORMAT_S32;
break;
case SND_PCM_FORMAT_U32_BE:
*endianness = 1;
- *fmt = AUD_FMT_U32;
+ *fmt = AUDIO_FORMAT_U32;
break;
default:
@@ -638,19 +638,22 @@ static int alsa_open (int in, struct alsa_params_req *req,
bytes_per_sec = freq << (nchannels == 2);
switch (obt->fmt) {
- case AUD_FMT_S8:
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_S8:
+ case AUDIO_FORMAT_U8:
break;
- case AUD_FMT_S16:
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_S16:
+ case AUDIO_FORMAT_U16:
bytes_per_sec <<= 1;
break;
- case AUD_FMT_S32:
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_S32:
+ case AUDIO_FORMAT_U32:
bytes_per_sec <<= 2;
break;
+
+ default:
+ abort();
}
threshold = (conf->threshold * bytes_per_sec) / 1000;
diff --git a/audio/audio.c b/audio/audio.c
index 909c817103..77216e5010 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -113,7 +113,7 @@ static struct {
.settings = {
.freq = 44100,
.nchannels = 2,
- .fmt = AUD_FMT_S16,
+ .fmt = AUDIO_FORMAT_S16,
.endianness = AUDIO_HOST_ENDIANNESS,
}
},
@@ -125,7 +125,7 @@ static struct {
.settings = {
.freq = 44100,
.nchannels = 2,
- .fmt = AUD_FMT_S16,
+ .fmt = AUDIO_FORMAT_S16,
.endianness = AUDIO_HOST_ENDIANNESS,
}
},
@@ -257,58 +257,61 @@ static char *audio_alloc_prefix (const char *s)
return r;
}
-static const char *audio_audfmt_to_string (audfmt_e fmt)
+static const char *audio_audfmt_to_string (AudioFormat fmt)
{
switch (fmt) {
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
return "U8";
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
return "U16";
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
return "S8";
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
return "S16";
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_U32:
return "U32";
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_S32:
return "S32";
+
+ default:
+ abort();
}
dolog ("Bogus audfmt %d returning S16\n", fmt);
return "S16";
}
-static audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval,
+static AudioFormat audio_string_to_audfmt (const char *s, AudioFormat defval,
int *defaultp)
{
if (!strcasecmp (s, "u8")) {
*defaultp = 0;
- return AUD_FMT_U8;
+ return AUDIO_FORMAT_U8;
}
else if (!strcasecmp (s, "u16")) {
*defaultp = 0;
- return AUD_FMT_U16;
+ return AUDIO_FORMAT_U16;
}
else if (!strcasecmp (s, "u32")) {
*defaultp = 0;
- return AUD_FMT_U32;
+ return AUDIO_FORMAT_U32;
}
else if (!strcasecmp (s, "s8")) {
*defaultp = 0;
- return AUD_FMT_S8;
+ return AUDIO_FORMAT_S8;
}
else if (!strcasecmp (s, "s16")) {
*defaultp = 0;
- return AUD_FMT_S16;
+ return AUDIO_FORMAT_S16;
}
else if (!strcasecmp (s, "s32")) {
*defaultp = 0;
- return AUD_FMT_S32;
+ return AUDIO_FORMAT_S32;
}
else {
dolog ("Bogus audio format `%s' using %s\n",
@@ -318,8 +321,8 @@ static audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval,
}
}
-static audfmt_e audio_get_conf_fmt (const char *envname,
- audfmt_e defval,
+static AudioFormat audio_get_conf_fmt (const char *envname,
+ AudioFormat defval,
int *defaultp)
{
const char *var = getenv (envname);
@@ -421,7 +424,7 @@ static void audio_print_options (const char *prefix,
case AUD_OPT_FMT:
{
- audfmt_e *fmtp = opt->valp;
+ AudioFormat *fmtp = opt->valp;
printf (
"format, %s = %s, (one of: U8 S8 U16 S16 U32 S32)\n",
state,
@@ -492,7 +495,7 @@ static void audio_process_options (const char *prefix,
case AUD_OPT_FMT:
{
- audfmt_e *fmtp = opt->valp;
+ AudioFormat *fmtp = opt->valp;
*fmtp = audio_get_conf_fmt (optname, *fmtp, &def);
}
break;
@@ -524,22 +527,22 @@ static void audio_print_settings (struct audsettings *as)
dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels);
switch (as->fmt) {
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
AUD_log (NULL, "S8");
break;
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
AUD_log (NULL, "U8");
break;
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
AUD_log (NULL, "S16");
break;
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
AUD_log (NULL, "U16");
break;
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_S32:
AUD_log (NULL, "S32");
break;
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_U32:
AUD_log (NULL, "U32");
break;
default:
@@ -570,12 +573,12 @@ static int audio_validate_settings (struct audsettings *as)
invalid |= as->endianness != 0 && as->endianness != 1;
switch (as->fmt) {
- case AUD_FMT_S8:
- case AUD_FMT_U8:
- case AUD_FMT_S16:
- case AUD_FMT_U16:
- case AUD_FMT_S32:
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_S8:
+ case AUDIO_FORMAT_U8:
+ case AUDIO_FORMAT_S16:
+ case AUDIO_FORMAT_U16:
+ case AUDIO_FORMAT_S32:
+ case AUDIO_FORMAT_U32:
break;
default:
invalid = 1;
@@ -591,25 +594,28 @@ static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *a
int bits = 8, sign = 0;
switch (as->fmt) {
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
sign = 1;
/* fall through */
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
break;
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
sign = 1;
/* fall through */
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
bits = 16;
break;
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_S32:
sign = 1;
/* fall through */
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_U32:
bits = 32;
break;
+
+ default:
+ abort();
}
return info->freq == as->freq
&& info->nchannels == as->nchannels
@@ -623,24 +629,27 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
int bits = 8, sign = 0, shift = 0;
switch (as->fmt) {
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
sign = 1;
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
break;
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
sign = 1;
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
bits = 16;
shift = 1;
break;
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_S32:
sign = 1;
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_U32:
bits = 32;
shift = 2;
break;
+
+ default:
+ abort();
}
info->freq = as->freq;
diff --git a/audio/audio_win_int.c b/audio/audio_win_int.c
index 6900008d0c..b938fd667b 100644
--- a/audio/audio_win_int.c
+++ b/audio/audio_win_int.c
@@ -24,20 +24,20 @@ int waveformat_from_audio_settings (WAVEFORMATEX *wfx,
wfx->cbSize = 0;
switch (as->fmt) {
- case AUD_FMT_S8:
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_S8:
+ case AUDIO_FORMAT_U8:
wfx->wBitsPerSample = 8;
break;
- case AUD_FMT_S16:
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_S16:
+ case AUDIO_FORMAT_U16:
wfx->wBitsPerSample = 16;
wfx->nAvgBytesPerSec <<= 1;
wfx->nBlockAlign <<= 1;
break;
- case AUD_FMT_S32:
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_S32:
+ case AUDIO_FORMAT_U32:
wfx->wBitsPerSample = 32;
wfx->nAvgBytesPerSec <<= 2;
wfx->nBlockAlign <<= 2;
@@ -85,15 +85,15 @@ int waveformat_to_audio_settings (WAVEFORMATEX *wfx,
switch (wfx->wBitsPerSample) {
case 8:
- as->fmt = AUD_FMT_U8;
+ as->fmt = AUDIO_FORMAT_U8;
break;
case 16:
- as->fmt = AUD_FMT_S16;
+ as->fmt = AUDIO_FORMAT_S16;
break;
case 32:
- as->fmt = AUD_FMT_S32;
+ as->fmt = AUDIO_FORMAT_S32;
break;
default:
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 6c69622b4c..355e8fbda5 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -70,7 +70,7 @@ typedef struct OSSVoiceIn {
struct oss_params {
int freq;
- audfmt_e fmt;
+ AudioFormat fmt;
int nchannels;
int nfrags;
int fragsize;
@@ -148,16 +148,16 @@ static int oss_write (SWVoiceOut *sw, void *buf, int len)
return audio_pcm_sw_write (sw, buf, len);
}
-static int aud_to_ossfmt (audfmt_e fmt, int endianness)
+static int aud_to_ossfmt (AudioFormat fmt, int endianness)
{
switch (fmt) {
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_S8:
return AFMT_S8;
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_U8:
return AFMT_U8;
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_S16:
if (endianness) {
return AFMT_S16_BE;
}
@@ -165,7 +165,7 @@ static int aud_to_ossfmt (audfmt_e fmt, int endianness)
return AFMT_S16_LE;
}
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_U16:
if (endianness) {
return AFMT_U16_BE;
}
@@ -182,37 +182,37 @@ static int aud_to_ossfmt (audfmt_e fmt, int endianness)
}
}
-static int oss_to_audfmt (int ossfmt, audfmt_e *fmt, int *endianness)
+static int oss_to_audfmt (int ossfmt, AudioFormat *fmt, int *endianness)
{
switch (ossfmt) {
case AFMT_S8:
*endianness = 0;
- *fmt = AUD_FMT_S8;
+ *fmt = AUDIO_FORMAT_S8;
break;
case AFMT_U8:
*endianness = 0;
- *fmt = AUD_FMT_U8;
+ *fmt = AUDIO_FORMAT_U8;
break;
case AFMT_S16_LE:
*endianness = 0;
- *fmt = AUD_FMT_S16;
+ *fmt = AUDIO_FORMAT_S16;
break;
case AFMT_U16_LE:
*endianness = 0;
- *fmt = AUD_FMT_U16;
+ *fmt = AUDIO_FORMAT_U16;
break;
case AFMT_S16_BE:
*endianness = 1;
- *fmt = AUD_FMT_S16;
+ *fmt = AUDIO_FORMAT_S16;
break;
case AFMT_U16_BE:
*endianness = 1;
- *fmt = AUD_FMT_U16;
+ *fmt = AUDIO_FORMAT_U16;
break;
default:
@@ -500,7 +500,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
int endianness;
int err;
int fd;
- audfmt_e effective_fmt;
+ AudioFormat effective_fmt;
struct audsettings obt_as;
OSSConf *conf = drv_opaque;
@@ -667,7 +667,7 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
int endianness;
int err;
int fd;
- audfmt_e effective_fmt;
+ AudioFormat effective_fmt;
struct audsettings obt_as;
OSSConf *conf = drv_opaque;
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 6153b908da..8246f260a8 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -385,21 +385,21 @@ static int qpa_read (SWVoiceIn *sw, void *buf, int len)
return audio_pcm_sw_read (sw, buf, len);
}
-static pa_sample_format_t audfmt_to_pa (audfmt_e afmt, int endianness)
+static pa_sample_format_t audfmt_to_pa (AudioFormat afmt, int endianness)
{
int format;
switch (afmt) {
- case AUD_FMT_S8:
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_S8:
+ case AUDIO_FORMAT_U8:
format = PA_SAMPLE_U8;
break;
- case AUD_FMT_S16:
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_S16:
+ case AUDIO_FORMAT_U16:
format = endianness ? PA_SAMPLE_S16BE : PA_SAMPLE_S16LE;
break;
- case AUD_FMT_S32:
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_S32:
+ case AUDIO_FORMAT_U32:
format = endianness ? PA_SAMPLE_S32BE : PA_SAMPLE_S32LE;
break;
default:
@@ -410,26 +410,26 @@ static pa_sample_format_t audfmt_to_pa (audfmt_e afmt, int endianness)
return format;
}
-static audfmt_e pa_to_audfmt (pa_sample_format_t fmt, int *endianness)
+static AudioFormat pa_to_audfmt (pa_sample_format_t fmt, int *endianness)
{
switch (fmt) {
case PA_SAMPLE_U8:
- return AUD_FMT_U8;
+ return AUDIO_FORMAT_U8;
case PA_SAMPLE_S16BE:
*endianness = 1;
- return AUD_FMT_S16;
+ return AUDIO_FORMAT_S16;
case PA_SAMPLE_S16LE:
*endianness = 0;
- return AUD_FMT_S16;
+ return AUDIO_FORMAT_S16;
case PA_SAMPLE_S32BE:
*endianness = 1;
- return AUD_FMT_S32;
+ return AUDIO_FORMAT_S32;
case PA_SAMPLE_S32LE:
*endianness = 0;
- return AUD_FMT_S32;
+ return AUDIO_FORMAT_S32;
default:
dolog ("Internal logic error: Bad pa_sample_format %d\n", fmt);
- return AUD_FMT_U8;
+ return AUDIO_FORMAT_U8;
}
}
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index f7ee70b153..4cd4cbaf00 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -68,19 +68,19 @@ static void GCC_FMT_ATTR (1, 2) sdl_logerr (const char *fmt, ...)
AUD_log (AUDIO_CAP, "Reason: %s\n", SDL_GetError ());
}
-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:
@@ -92,37 +92,37 @@ 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:
@@ -265,7 +265,7 @@ static int sdl_init_out(HWVoiceOut *hw, struct audsettings *as,
SDL_AudioSpec req, obt;
int endianness;
int err;
- audfmt_e effective_fmt;
+ AudioFormat effective_fmt;
struct audsettings obt_as;
req.freq = as->freq;
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index 6ad0eafbc6..3aeb0cb357 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -130,7 +130,7 @@ static int line_out_init(HWVoiceOut *hw, struct audsettings *as,
settings.freq = SPICE_INTERFACE_PLAYBACK_FREQ;
#endif
settings.nchannels = SPICE_INTERFACE_PLAYBACK_CHAN;
- settings.fmt = AUD_FMT_S16;
+ settings.fmt = AUDIO_FORMAT_S16;
settings.endianness = AUDIO_HOST_ENDIANNESS;
audio_pcm_init_info (&hw->info, &settings);
@@ -258,7 +258,7 @@ static int line_in_init(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
settings.freq = SPICE_INTERFACE_RECORD_FREQ;
#endif
settings.nchannels = SPICE_INTERFACE_RECORD_CHAN;
- settings.fmt = AUD_FMT_S16;
+ settings.fmt = AUDIO_FORMAT_S16;
settings.endianness = AUDIO_HOST_ENDIANNESS;
audio_pcm_init_info (&hw->info, &settings);
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 40adfa30c3..35a614785e 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -117,20 +117,23 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
stereo = wav_as.nchannels == 2;
switch (wav_as.fmt) {
- case AUD_FMT_S8:
- case AUD_FMT_U8:
+ case AUDIO_FORMAT_S8:
+ case AUDIO_FORMAT_U8:
bits16 = 0;
break;
- case AUD_FMT_S16:
- case AUD_FMT_U16:
+ case AUDIO_FORMAT_S16:
+ case AUDIO_FORMAT_U16:
bits16 = 1;
break;
- case AUD_FMT_S32:
- case AUD_FMT_U32:
+ case AUDIO_FORMAT_S32:
+ case AUDIO_FORMAT_U32:
dolog ("WAVE files can not handle 32bit formats\n");
return -1;
+
+ default:
+ abort();
}
hdr[34] = bits16 ? 0x10 : 0x08;
@@ -225,7 +228,7 @@ static int wav_ctl_out (HWVoiceOut *hw, int cmd, ...)
static WAVConf glob_conf = {
.settings.freq = 44100,
.settings.nchannels = 2,
- .settings.fmt = AUD_FMT_S16,
+ .settings.fmt = AUDIO_FORMAT_S16,
.wav_path = "qemu.wav"
};
diff --git a/audio/wavcapture.c b/audio/wavcapture.c
index cd24570aa7..74320dfecc 100644
--- a/audio/wavcapture.c
+++ b/audio/wavcapture.c
@@ -136,7 +136,7 @@ int wav_start_capture (CaptureState *s, const char *path, int freq,
as.freq = freq;
as.nchannels = 1 << stereo;
- as.fmt = bits16 ? AUD_FMT_S16 : AUD_FMT_U8;
+ as.fmt = bits16 ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_U8;
as.endianness = 0;
ops.notify = wav_notify;
diff --git a/hw/arm/omap2.c b/hw/arm/omap2.c
index 94dffb2f57..446223906e 100644
--- a/hw/arm/omap2.c
+++ b/hw/arm/omap2.c
@@ -273,7 +273,7 @@ static void omap_eac_format_update(struct omap_eac_s *s)
* does I2S specify it? */
/* All register writes are 16 bits so we we store 16-bit samples
* in the buffers regardless of AGCFR[B8_16] value. */
- fmt.fmt = AUD_FMT_U16;
+ fmt.fmt = AUDIO_FORMAT_U16;
s->codec.in_voice = AUD_open_in(&s->codec.card, s->codec.in_voice,
"eac.codec.in", s, omap_eac_in_cb, &fmt);
diff --git a/hw/audio/ac97.c b/hw/audio/ac97.c
index d799533aa9..2265622d44 100644
--- a/hw/audio/ac97.c
+++ b/hw/audio/ac97.c
@@ -365,7 +365,7 @@ static void open_voice (AC97LinkState *s, int index, int freq)
as.freq = freq;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = 0;
if (freq > 0) {
diff --git a/hw/audio/adlib.c b/hw/audio/adlib.c
index 97b876c7e0..0957780a3d 100644
--- a/hw/audio/adlib.c
+++ b/hw/audio/adlib.c
@@ -269,7 +269,7 @@ static void adlib_realizefn (DeviceState *dev, Error **errp)
as.freq = s->freq;
as.nchannels = SHIFT;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = AUDIO_HOST_ENDIANNESS;
AUD_register_card ("adlib", &s->card);
diff --git a/hw/audio/cs4231a.c b/hw/audio/cs4231a.c
index 9089dcb47e..62da75eefe 100644
--- a/hw/audio/cs4231a.c
+++ b/hw/audio/cs4231a.c
@@ -288,7 +288,7 @@ static void cs_reset_voices (CSState *s, uint32_t val)
switch ((val >> 5) & ((s->dregs[MODE_And_ID] & MODE2) ? 7 : 3)) {
case 0:
- as.fmt = AUD_FMT_U8;
+ as.fmt = AUDIO_FORMAT_U8;
s->shift = as.nchannels == 2;
break;
@@ -298,7 +298,7 @@ static void cs_reset_voices (CSState *s, uint32_t val)
case 3:
s->tab = ALawDecompressTable;
x_law:
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = AUDIO_HOST_ENDIANNESS;
s->shift = as.nchannels == 2;
break;
@@ -307,7 +307,7 @@ static void cs_reset_voices (CSState *s, uint32_t val)
as.endianness = 1;
/* fall through */
case 2:
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
s->shift = as.nchannels;
break;
diff --git a/hw/audio/es1370.c b/hw/audio/es1370.c
index 97789a0771..a5314d66fd 100644
--- a/hw/audio/es1370.c
+++ b/hw/audio/es1370.c
@@ -414,14 +414,14 @@ static void es1370_update_voices (ES1370State *s, uint32_t ctl, uint32_t sctl)
i,
new_freq,
1 << (new_fmt & 1),
- (new_fmt & 2) ? AUD_FMT_S16 : AUD_FMT_U8,
+ (new_fmt & 2) ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_U8,
d->shift);
if (new_freq) {
struct audsettings as;
as.freq = new_freq;
as.nchannels = 1 << (new_fmt & 1);
- as.fmt = (new_fmt & 2) ? AUD_FMT_S16 : AUD_FMT_U8;
+ as.fmt = (new_fmt & 2) ? AUDIO_FORMAT_S16 : AUDIO_FORMAT_U8;
as.endianness = 0;
if (i == ADC_CHANNEL) {
diff --git a/hw/audio/gus.c b/hw/audio/gus.c
index 8e0b27e0f2..b3e2a7fdd5 100644
--- a/hw/audio/gus.c
+++ b/hw/audio/gus.c
@@ -251,7 +251,7 @@ static void gus_realizefn (DeviceState *dev, Error **errp)
as.freq = s->freq;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = GUS_ENDIANNESS;
s->voice = AUD_open_out (
diff --git a/hw/audio/hda-codec.c b/hw/audio/hda-codec.c
index 617a1c1016..c25bfa38b1 100644
--- a/hw/audio/hda-codec.c
+++ b/hw/audio/hda-codec.c
@@ -99,9 +99,9 @@ static void hda_codec_parse_fmt(uint32_t format, struct audsettings *as)
}
switch (format & AC_FMT_BITS_MASK) {
- case AC_FMT_BITS_8: as->fmt = AUD_FMT_S8; break;
- case AC_FMT_BITS_16: as->fmt = AUD_FMT_S16; break;
- case AC_FMT_BITS_32: as->fmt = AUD_FMT_S32; break;
+ case AC_FMT_BITS_8: as->fmt = AUDIO_FORMAT_S8; break;
+ case AC_FMT_BITS_16: as->fmt = AUDIO_FORMAT_S16; break;
+ case AC_FMT_BITS_32: as->fmt = AUDIO_FORMAT_S32; break;
}
as->nchannels = ((format & AC_FMT_CHAN_MASK) >> AC_FMT_CHAN_SHIFT) + 1;
@@ -134,12 +134,12 @@ static void hda_codec_parse_fmt(uint32_t format, struct audsettings *as)
/* -------------------------------------------------------------------------- */
static const char *fmt2name[] = {
- [ AUD_FMT_U8 ] = "PCM-U8",
- [ AUD_FMT_S8 ] = "PCM-S8",
- [ AUD_FMT_U16 ] = "PCM-U16",
- [ AUD_FMT_S16 ] = "PCM-S16",
- [ AUD_FMT_U32 ] = "PCM-U32",
- [ AUD_FMT_S32 ] = "PCM-S32",
+ [ AUDIO_FORMAT_U8 ] = "PCM-U8",
+ [ AUDIO_FORMAT_S8 ] = "PCM-S8",
+ [ AUDIO_FORMAT_U16 ] = "PCM-U16",
+ [ AUDIO_FORMAT_S16 ] = "PCM-S16",
+ [ AUDIO_FORMAT_U32 ] = "PCM-U32",
+ [ AUDIO_FORMAT_S32 ] = "PCM-S32",
};
typedef struct HDAAudioState HDAAudioState;
diff --git a/hw/audio/lm4549.c b/hw/audio/lm4549.c
index a46f2301af..af8b22b541 100644
--- a/hw/audio/lm4549.c
+++ b/hw/audio/lm4549.c
@@ -185,7 +185,7 @@ void lm4549_write(lm4549_state *s,
struct audsettings as;
as.freq = value;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = 0;
s->voice = AUD_open_out(
@@ -255,7 +255,7 @@ static int lm4549_post_load(void *opaque, int version_id)
struct audsettings as;
as.freq = freq;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = 0;
s->voice = AUD_open_out(
@@ -292,7 +292,7 @@ void lm4549_init(lm4549_state *s, lm4549_callback data_req_cb, void* opaque)
/* Open a default voice */
as.freq = 48000;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = 0;
s->voice = AUD_open_out(
diff --git a/hw/audio/milkymist-ac97.c b/hw/audio/milkymist-ac97.c
index bc8db71ae0..90cce1e6ed 100644
--- a/hw/audio/milkymist-ac97.c
+++ b/hw/audio/milkymist-ac97.c
@@ -308,7 +308,7 @@ static void milkymist_ac97_realize(DeviceState *dev, Error **errp)
as.freq = 48000;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = 1;
s->voice_in = AUD_open_in(&s->card, s->voice_in,
diff --git a/hw/audio/pcspk.c b/hw/audio/pcspk.c
index b80a62ce90..fdbb4b6e99 100644
--- a/hw/audio/pcspk.c
+++ b/hw/audio/pcspk.c
@@ -162,7 +162,7 @@ static void pcspk_initfn(Object *obj)
static void pcspk_realizefn(DeviceState *dev, Error **errp)
{
- struct audsettings as = {PCSPK_SAMPLE_RATE, 1, AUD_FMT_U8, 0};
+ struct audsettings as = {PCSPK_SAMPLE_RATE, 1, AUDIO_FORMAT_U8, 0};
ISADevice *isadev = ISA_DEVICE(dev);
PCSpkState *s = PC_SPEAKER(dev);
diff --git a/hw/audio/sb16.c b/hw/audio/sb16.c
index c5b9bf79e8..65ea0cd938 100644
--- a/hw/audio/sb16.c
+++ b/hw/audio/sb16.c
@@ -66,7 +66,7 @@ typedef struct SB16State {
int fmt_stereo;
int fmt_signed;
int fmt_bits;
- audfmt_e fmt;
+ AudioFormat fmt;
int dma_auto;
int block_size;
int fifo;
@@ -224,7 +224,7 @@ static void continue_dma8 (SB16State *s)
static void dma_cmd8 (SB16State *s, int mask, int dma_len)
{
- s->fmt = AUD_FMT_U8;
+ s->fmt = AUDIO_FORMAT_U8;
s->use_hdma = 0;
s->fmt_bits = 8;
s->fmt_signed = 0;
@@ -319,18 +319,18 @@ static void dma_cmd (SB16State *s, uint8_t cmd, uint8_t d0, int dma_len)
if (16 == s->fmt_bits) {
if (s->fmt_signed) {
- s->fmt = AUD_FMT_S16;
+ s->fmt = AUDIO_FORMAT_S16;
}
else {
- s->fmt = AUD_FMT_U16;
+ s->fmt = AUDIO_FORMAT_U16;
}
}
else {
if (s->fmt_signed) {
- s->fmt = AUD_FMT_S8;
+ s->fmt = AUDIO_FORMAT_S8;
}
else {
- s->fmt = AUD_FMT_U8;
+ s->fmt = AUDIO_FORMAT_U8;
}
}
@@ -852,7 +852,7 @@ static void legacy_reset (SB16State *s)
as.freq = s->freq;
as.nchannels = 1;
- as.fmt = AUD_FMT_U8;
+ as.fmt = AUDIO_FORMAT_U8;
as.endianness = 0;
s->voice = AUD_open_out (
diff --git a/hw/audio/wm8750.c b/hw/audio/wm8750.c
index 169b006ade..ca0ad73caf 100644
--- a/hw/audio/wm8750.c
+++ b/hw/audio/wm8750.c
@@ -201,7 +201,7 @@ static void wm8750_set_format(WM8750State *s)
in_fmt.endianness = 0;
in_fmt.nchannels = 2;
in_fmt.freq = s->adc_hz;
- in_fmt.fmt = AUD_FMT_S16;
+ in_fmt.fmt = AUDIO_FORMAT_S16;
s->adc_voice[0] = AUD_open_in(&s->card, s->adc_voice[0],
CODEC ".input1", s, wm8750_audio_in_cb, &in_fmt);
@@ -214,7 +214,7 @@ static void wm8750_set_format(WM8750State *s)
out_fmt.endianness = 0;
out_fmt.nchannels = 2;
out_fmt.freq = s->dac_hz;
- out_fmt.fmt = AUD_FMT_S16;
+ out_fmt.fmt = AUDIO_FORMAT_S16;
s->dac_voice[0] = AUD_open_out(&s->card, s->dac_voice[0],
CODEC ".speaker", s, wm8750_audio_out_cb, &out_fmt);
@@ -681,7 +681,7 @@ uint32_t wm8750_adc_dat(void *opaque)
if (s->idx_in >= sizeof(s->data_in)) {
wm8750_in_load(s);
if (s->idx_in >= sizeof(s->data_in)) {
- return 0x80008000; /* silence in AUD_FMT_S16 sample format */
+ return 0x80008000; /* silence in AUDIO_FORMAT_S16 sample format */
}
}
diff --git a/hw/display/xlnx_dp.c b/hw/display/xlnx_dp.c
index cc0f9bc9cc..11b09bd18c 100644
--- a/hw/display/xlnx_dp.c
+++ b/hw/display/xlnx_dp.c
@@ -1260,7 +1260,7 @@ static void xlnx_dp_realize(DeviceState *dev, Error **errp)
as.freq = 44100;
as.nchannels = 2;
- as.fmt = AUD_FMT_S16;
+ as.fmt = AUDIO_FORMAT_S16;
as.endianness = 0;
AUD_register_card("xlnx_dp.audio", &s->aud_card);
diff --git a/hw/input/tsc210x.c b/hw/input/tsc210x.c
index 2eb3cb9518..41731619bb 100644
--- a/hw/input/tsc210x.c
+++ b/hw/input/tsc210x.c
@@ -318,7 +318,7 @@ static void tsc2102_audio_output_update(TSC210xState *s)
fmt.endianness = 0;
fmt.nchannels = 2;
fmt.freq = s->codec.tx_rate;
- fmt.fmt = AUD_FMT_S16;
+ fmt.fmt = AUDIO_FORMAT_S16;
s->dac_voice[0] = AUD_open_out(&s->card, s->dac_voice[0],
"tsc2102.sink", s, (void *) tsc210x_audio_out_cb, &fmt);
diff --git a/hw/usb/dev-audio.c b/hw/usb/dev-audio.c
index 28ac7c5165..c46d5eeb79 100644
--- a/hw/usb/dev-audio.c
+++ b/hw/usb/dev-audio.c
@@ -650,7 +650,7 @@ static void usb_audio_realize(USBDevice *dev, Error **errp)
s->out.vol[1] = 240; /* 0 dB */
s->out.as.freq = USBAUDIO_SAMPLE_RATE;
s->out.as.nchannels = 2;
- s->out.as.fmt = AUD_FMT_S16;
+ s->out.as.fmt = AUDIO_FORMAT_S16;
s->out.as.endianness = 0;
streambuf_init(&s->out.buf, s->buffer);
diff --git a/ui/vnc.c b/ui/vnc.c
index da4a21d4ce..b724e998ba 100644
--- a/ui/vnc.c
+++ b/ui/vnc.c
@@ -1013,16 +1013,16 @@ static void vnc_update_throttle_offset(VncState *vs)
int bps;
switch (vs->as.fmt) {
default:
- case AUD_FMT_U8:
- case AUD_FMT_S8:
+ case AUDIO_FORMAT_U8:
+ case AUDIO_FORMAT_S8:
bps = 1;
break;
- case AUD_FMT_U16:
- case AUD_FMT_S16:
+ case AUDIO_FORMAT_U16:
+ case AUDIO_FORMAT_S16:
bps = 2;
break;
- case AUD_FMT_U32:
- case AUD_FMT_S32:
+ case AUDIO_FORMAT_U32:
+ case AUDIO_FORMAT_S32:
bps = 4;
break;
}
@@ -2369,12 +2369,12 @@ 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);
@@ -3105,7 +3105,7 @@ static void vnc_connect(VncDisplay *vd, QIOChannelSocket *sioc,
vs->as.freq = 44100;
vs->as.nchannels = 2;
- vs->as.fmt = AUD_FMT_S16;
+ vs->as.fmt = AUDIO_FORMAT_S16;
vs->as.endianness = 0;
qemu_mutex_init(&vs->output_mutex);
--
2.20.1
^ permalink raw reply related [flat|nested] 18+ messages in thread* [Qemu-devel] [PATCH v6 03/14] audio: -audiodev command line option: documentation
2019-03-08 22:34 [Qemu-devel] [PATCH v6 00/14] Audio patches Kővágó, Zoltán
2019-03-08 22:34 ` [Qemu-devel] [PATCH v6 01/14] qapi: qapi for audio backends Kővágó, Zoltán
2019-03-08 22:34 ` [Qemu-devel] [PATCH v6 02/14] audio: use qapi AudioFormat instead of audfmt_e Kővágó, Zoltán
@ 2019-03-08 22:34 ` Kővágó, Zoltán
2019-03-08 22:34 ` [Qemu-devel] [PATCH v6 04/14] audio: -audiodev command line option basic implementation Kővágó, Zoltán
` (11 subsequent siblings)
14 siblings, 0 replies; 18+ messages in thread
From: Kővágó, Zoltán @ 2019-03-08 22:34 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann, reviewer:Incompatible changes
This patch adds documentation of an -audiodev command line option, that
deprecates the old QEMU_* environment variables for audio backend
configuration. It's syntax is similar to existing options (-netdev,
-device, etc):
-audiodev driver_name,property=value,...
Although now it's possible to specify multiple -audiodev options on
command line, multiple audio backends are not supported yet.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
Notes:
Changes from v4:
* deprecated QEMU_AUDIO_ env vars
* updated to reflect qapi changes
* added info to qemu-deprecated.texi
qemu-deprecated.texi | 7 ++
qemu-options.hx | 236 ++++++++++++++++++++++++++++++++++++++++++-
2 files changed, 240 insertions(+), 3 deletions(-)
diff --git a/qemu-deprecated.texi b/qemu-deprecated.texi
index 45c57952da..5c07ad4acb 100644
--- a/qemu-deprecated.texi
+++ b/qemu-deprecated.texi
@@ -60,6 +60,13 @@ Support for invalid topologies will be removed, the user must ensure
topologies described with -smp include all possible cpus, i.e.
@math{@var{sockets} * @var{cores} * @var{threads} = @var{maxcpus}}.
+@subsection QEMU_AUDIO_ environment variables and -audio-help (since 4.0)
+
+The ``-audiodev'' argument is now the preferred way to specify audio
+backend settings instead of environment variables. To ease migration to
+the new format, the ``-audiodev-help'' option can be used to convert
+the current values of the environment variables to ``-audiodev'' options.
+
@section QEMU Machine Protocol (QMP) commands
@subsection block-dirty-bitmap-add "autoload" parameter (since 2.12.0)
diff --git a/qemu-options.hx b/qemu-options.hx
index 1cf9aac1fe..52ea13e6a2 100644
--- a/qemu-options.hx
+++ b/qemu-options.hx
@@ -416,14 +416,244 @@ The default is @code{en-us}.
ETEXI
+HXCOMM Deprecated by -audiodev
DEF("audio-help", 0, QEMU_OPTION_audio_help,
- "-audio-help print list of audio drivers and their options\n",
+ "-audio-help show -audiodev equivalent of the currently specified audio settings\n",
QEMU_ARCH_ALL)
STEXI
@item -audio-help
@findex -audio-help
-Will show the audio subsystem help: list of drivers, tunable
-parameters.
+Will show the -audiodev equivalent of the currently specified
+(deprecated) environment variables.
+ETEXI
+
+DEF("audiodev", HAS_ARG, QEMU_OPTION_audiodev,
+ "-audiodev [driver=]driver,id=id[,prop[=value][,...]]\n"
+ " specifies the audio backend to use\n"
+ " id= identifier of the backend\n"
+ " timer-period= timer period in microseconds\n"
+ " in|out.fixed-settings= use fixed settings for host audio\n"
+ " in|out.frequency= frequency to use with fixed settings\n"
+ " in|out.channels= number of channels to use with fixed settings\n"
+ " in|out.format= sample format to use with fixed settings\n"
+ " valid values: s8, s16, s32, u8, u16, u32\n"
+ " in|out.voices= number of voices to use\n"
+ " in|out.buffer-len= length of buffer in microseconds\n"
+ "-audiodev none,id=id,[,prop[=value][,...]]\n"
+ " dummy driver that discards all output\n"
+#ifdef CONFIG_AUDIO_ALSA
+ "-audiodev alsa,id=id[,prop[=value][,...]]\n"
+ " in|out.dev= name of the audio device to use\n"
+ " in|out.period-len= length of period in microseconds\n"
+ " in|out.try-poll= attempt to use poll mode\n"
+ " threshold= threshold (in microseconds) when playback starts\n"
+#endif
+#ifdef CONFIG_AUDIO_COREAUDIO
+ "-audiodev coreaudio,id=id[,prop[=value][,...]]\n"
+ " in|out.buffer-count= number of buffers\n"
+#endif
+#ifdef CONFIG_AUDIO_DSOUND
+ "-audiodev dsound,id=id[,prop[=value][,...]]\n"
+ " latency= add extra latency to playback in microseconds\n"
+#endif
+#ifdef CONFIG_AUDIO_OSS
+ "-audiodev oss,id=id[,prop[=value][,...]]\n"
+ " in|out.dev= path of the audio device to use\n"
+ " in|out.buffer-count= number of buffers\n"
+ " in|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_AUDIO_PA
+ "-audiodev pa,id=id[,prop[=value][,...]]\n"
+ " server= PulseAudio server address\n"
+ " in|out.name= source/sink device name\n"
+#endif
+#ifdef CONFIG_AUDIO_SDL
+ "-audiodev sdl,id=id[,prop[=value][,...]]\n"
+#endif
+#ifdef CONFIG_SPICE
+ "-audiodev spice,id=id[,prop[=value][,...]]\n"
+#endif
+ "-audiodev wav,id=id[,prop[=value][,...]]\n"
+ " path= path of wav file to record\n",
+ QEMU_ARCH_ALL)
+STEXI
+@item -audiodev [driver=]@var{driver},id=@var{id}[,@var{prop}[=@var{value}][,...]]
+@findex -audiodev
+Adds a new audio backend @var{driver} identified by @var{id}. There are
+global and driver specific properties. Some values can be set
+differently for input and output, they're marked with @code{in|out.}.
+You can set the input's property with @code{in.@var{prop}} and the
+output's property with @code{out.@var{prop}}. For example:
+@example
+-audiodev alsa,id=example,in.frequency=44110,out.frequency=8000
+-audiodev alsa,id=example,out.channels=1 # leaves in.channels unspecified
+@end example
+
+Valid global options are:
+
+@table @option
+@item id=@var{identifier}
+Identifies the audio backend.
+
+@item timer-period=@var{period}
+Sets the timer @var{period} used by the audio subsystem in microseconds.
+Default is 10000 (10 ms).
+
+@item in|out.fixed-settings=on|off
+Use fixed settings for host audio. When off, it will change based on
+how the guest opens the sound card. In this case you must not specify
+@var{frequency}, @var{channels} or @var{format}. Default is on.
+
+@item in|out.frequency=@var{frequency}
+Specify the @var{frequency} to use when using @var{fixed-settings}.
+Default is 44100Hz.
+
+@item in|out.channels=@var{channels}
+Specify the number of @var{channels} to use when using
+@var{fixed-settings}. Default is 2 (stereo).
+
+@item in|out.format=@var{format}
+Specify the sample @var{format} to use when using @var{fixed-settings}.
+Valid values are: @code{s8}, @code{s16}, @code{s32}, @code{u8},
+@code{u16}, @code{u32}. Default is @code{s16}.
+
+@item in|out.voices=@var{voices}
+Specify the number of @var{voices} to use. Default is 1.
+
+@item in|out.buffer=@var{usecs}
+Sets the size of the buffer in microseconds.
+
+@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 in|out.dev=@var{device}
+Specify the ALSA @var{device} to use for input and/or output. Default
+is @code{default}.
+
+@item in|out.period-len=@var{usecs}
+Sets the period length in microseconds.
+
+@item in|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.
+
+Core Audio specific options are:
+
+@table @option
+
+@item in|out.buffer-count=@var{count}
+Sets the @var{count} of the buffers.
+
+@end table
+
+@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.
+
+DirectSound specific options are:
+
+@table @option
+
+@item latency=@var{usecs}
+Add extra @var{usecs} microseconds latency to playback. Default is
+10000 (10 ms).
+
+@end table
+
+@item -audiodev oss,id=@var{id}[,@var{prop}[=@var{value}][,...]]
+Creates a backend using OSS. This backend is available on most
+Unix-like systems.
+
+OSS specific options are:
+
+@table @option
+
+@item in|out.dev=@var{device}
+Specify the file name of the OSS @var{device} to use. Default is
+@code{/dev/dsp}.
+
+@item in|out.buffer-count=@var{count}
+Sets the @var{count} of the buffers.
+
+@item in|out.try-poll=on|of
+Attempt to use poll mode with the device. Default is on.
+
+@item try-mmap=on|off
+Try using memory mapped device access. Default is off.
+
+@item exclusive=on|off
+Open the device in exclusive mode (vmix won't work in this case).
+Default is off.
+
+@item dsp-policy=@var{policy}
+Sets the timing policy (between 0 and 10, where smaller number means
+smaller latency but higher CPU usage). Use -1 to use buffer sizes
+specified by @code{buffer} and @code{buffer-count}. This option is
+ignored if you do not have OSS 4. Default is 5.
+
+@end table
+
+@item -audiodev pa,id=@var{id}[,@var{prop}[=@var{value}][,...]]
+Creates a backend using PulseAudio. This backend is available on most
+systems.
+
+PulseAudio specific options are:
+
+@table @option
+
+@item server=@var{server}
+Sets the PulseAudio @var{server} to connect to.
+
+@item in|out.name=@var{sink}
+Use the specified source/sink for recording/playback.
+
+@end table
+
+@item -audiodev sdl,id=@var{id}[,@var{prop}[=@var{value}][,...]]
+Creates a backend using SDL. This backend is available on most systems,
+but you should use your platform's native backend if possible. This
+backend has no backend specific properties.
+
+@item -audiodev spice,id=@var{id}[,@var{prop}[=@var{value}][,...]]
+Creates a backend that sends audio through SPICE. This backend requires
+@code{-spice} and automatically selected in that case, so usually you
+can ignore this option. This backend has no backend specific
+properties.
+
+@item -audiodev wav,id=@var{id}[,@var{prop}[=@var{value}][,...]]
+Creates a backend that writes audio to a WAV file.
+
+Backend specific options are:
+
+@table @option
+
+@item path=@var{path}
+Write recorded audio into the specified file. Default is
+@code{qemu.wav}.
+
+@end table
ETEXI
DEF("soundhw", HAS_ARG, QEMU_OPTION_soundhw,
--
2.20.1
^ permalink raw reply related [flat|nested] 18+ messages in thread* [Qemu-devel] [PATCH v6 04/14] audio: -audiodev command line option basic implementation
2019-03-08 22:34 [Qemu-devel] [PATCH v6 00/14] Audio patches Kővágó, Zoltán
` (2 preceding siblings ...)
2019-03-08 22:34 ` [Qemu-devel] [PATCH v6 03/14] audio: -audiodev command line option: documentation Kővágó, Zoltán
@ 2019-03-08 22:34 ` Kővágó, Zoltán
2019-03-08 22:34 ` [Qemu-devel] [PATCH v6 05/14] alsaaudio: port to -audiodev config Kővágó, Zoltán
` (10 subsequent siblings)
14 siblings, 0 replies; 18+ messages in thread
From: Kővágó, Zoltán @ 2019-03-08 22:34 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann, Paolo Bonzini
Audio drivers now get an Audiodev * as config paramters, instead of the
global audio_option structs. There is some code in audio/audio_legacy.c
that converts the old environment variables to audiodev options (this
way backends do not have to worry about legacy options). It also
contains a replacement of -audio-help, which prints out the equivalent
-audiodev based config of the currently specified environment variables.
Note that backends are not updated and still rely on environment
variables.
Also note that (due to moving try-poll from global to backend specific
option) currently ALSA and OSS will always try poll mode, regardless of
environment variables or -audiodev options.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio.h | 18 +-
audio/audio_int.h | 20 +-
audio/audio_template.h | 42 ++-
audio/alsaaudio.c | 2 +-
audio/audio.c | 630 ++++++++++++++++++-----------------------
audio/audio_legacy.c | 293 +++++++++++++++++++
audio/coreaudio.c | 2 +-
audio/dsoundaudio.c | 2 +-
audio/noaudio.c | 2 +-
audio/ossaudio.c | 2 +-
audio/paaudio.c | 2 +-
audio/sdlaudio.c | 2 +-
audio/spiceaudio.c | 2 +-
audio/wavaudio.c | 2 +-
vl.c | 7 +-
audio/Makefile.objs | 2 +-
16 files changed, 650 insertions(+), 380 deletions(-)
create mode 100644 audio/audio_legacy.c
diff --git a/audio/audio.h b/audio/audio.h
index 02f29a3b3e..64b0f761bc 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -36,12 +36,21 @@ typedef void (*audio_callback_fn) (void *opaque, int avail);
#define AUDIO_HOST_ENDIANNESS 0
#endif
-struct audsettings {
+typedef struct audsettings {
int freq;
int nchannels;
AudioFormat fmt;
int endianness;
-};
+} audsettings;
+
+audsettings audiodev_to_audsettings(AudiodevPerDirectionOptions *pdo);
+int audioformat_bytes_per_sample(AudioFormat fmt);
+int audio_buffer_frames(AudiodevPerDirectionOptions *pdo,
+ audsettings *as, int def_usecs);
+int audio_buffer_samples(AudiodevPerDirectionOptions *pdo,
+ audsettings *as, int def_usecs);
+int audio_buffer_bytes(AudiodevPerDirectionOptions *pdo,
+ audsettings *as, int def_usecs);
typedef enum {
AUD_CNOTIFY_ENABLE,
@@ -81,7 +90,6 @@ typedef struct QEMUAudioTimeStamp {
void AUD_vlog (const char *cap, const char *fmt, va_list ap) GCC_FMT_ATTR(2, 0);
void AUD_log (const char *cap, const char *fmt, ...) GCC_FMT_ATTR(2, 3);
-void AUD_help (void);
void AUD_register_card (const char *name, QEMUSoundCard *card);
void AUD_remove_card (QEMUSoundCard *card);
CaptureVoiceOut *AUD_add_capture (
@@ -163,4 +171,8 @@ void audio_sample_to_uint64(void *samples, int pos,
void audio_sample_from_uint64(void *samples, int pos,
uint64_t left, uint64_t right);
+void audio_parse_option(const char *opt);
+void audio_init_audiodevs(void);
+void audio_legacy_help(void);
+
#endif /* QEMU_AUDIO_H */
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 6c451b995c..7bf5dfc0b5 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -146,7 +146,7 @@ struct audio_driver {
const char *name;
const char *descr;
struct audio_option *options;
- void *(*init) (void);
+ void *(*init) (Audiodev *);
void (*fini) (void *);
struct audio_pcm_ops *pcm_ops;
int can_be_default;
@@ -193,6 +193,7 @@ struct SWVoiceCap {
typedef struct AudioState {
struct audio_driver *drv;
+ Audiodev *dev;
void *drv_opaque;
QEMUTimer *ts;
@@ -203,10 +204,13 @@ typedef struct AudioState {
int nb_hw_voices_out;
int nb_hw_voices_in;
int vm_running;
+ int64_t period_ticks;
} AudioState;
extern const struct mixeng_volume nominal_volume;
+extern const char *audio_prio_list[];
+
void audio_driver_register(audio_driver *drv);
audio_driver *audio_driver_lookup(const char *name);
@@ -248,4 +252,18 @@ static inline int audio_ring_dist (int dst, int src, int len)
#define AUDIO_STRINGIFY_(n) #n
#define AUDIO_STRINGIFY(n) AUDIO_STRINGIFY_(n)
+typedef struct AudiodevListEntry {
+ Audiodev *dev;
+ QSIMPLEQ_ENTRY(AudiodevListEntry) next;
+} AudiodevListEntry;
+
+typedef QSIMPLEQ_HEAD(, AudiodevListEntry) AudiodevListHead;
+AudiodevListHead audio_handle_legacy_opts(void);
+
+void audio_free_audiodev_list(AudiodevListHead *head);
+
+void audio_create_pdos(Audiodev *dev);
+AudiodevPerDirectionOptions *audio_get_pdo_in(Audiodev *dev);
+AudiodevPerDirectionOptions *audio_get_pdo_out(Audiodev *dev);
+
#endif /* QEMU_AUDIO_INT_H */
diff --git a/audio/audio_template.h b/audio/audio_template.h
index 7de227d2d1..1232bb54db 100644
--- a/audio/audio_template.h
+++ b/audio/audio_template.h
@@ -299,11 +299,42 @@ static HW *glue (audio_pcm_hw_add_new_, TYPE) (struct audsettings *as)
return NULL;
}
+AudiodevPerDirectionOptions *glue(audio_get_pdo_, TYPE)(Audiodev *dev)
+{
+ switch (dev->driver) {
+ case AUDIODEV_DRIVER_NONE:
+ return dev->u.none.TYPE;
+ case AUDIODEV_DRIVER_ALSA:
+ return qapi_AudiodevAlsaPerDirectionOptions_base(dev->u.alsa.TYPE);
+ case AUDIODEV_DRIVER_COREAUDIO:
+ return qapi_AudiodevCoreaudioPerDirectionOptions_base(
+ dev->u.coreaudio.TYPE);
+ case AUDIODEV_DRIVER_DSOUND:
+ return dev->u.dsound.TYPE;
+ case AUDIODEV_DRIVER_OSS:
+ return qapi_AudiodevOssPerDirectionOptions_base(dev->u.oss.TYPE);
+ case AUDIODEV_DRIVER_PA:
+ return qapi_AudiodevPaPerDirectionOptions_base(dev->u.pa.TYPE);
+ case AUDIODEV_DRIVER_SDL:
+ return dev->u.sdl.TYPE;
+ case AUDIODEV_DRIVER_SPICE:
+ return dev->u.spice.TYPE;
+ case AUDIODEV_DRIVER_WAV:
+ return dev->u.wav.TYPE;
+
+ case AUDIODEV_DRIVER__MAX:
+ break;
+ }
+ abort();
+}
+
static HW *glue (audio_pcm_hw_add_, TYPE) (struct audsettings *as)
{
HW *hw;
+ AudioState *s = &glob_audio_state;
+ AudiodevPerDirectionOptions *pdo = glue(audio_get_pdo_, TYPE)(s->dev);
- 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 +362,11 @@ static SW *glue (audio_pcm_create_voice_pair_, TYPE) (
SW *sw;
HW *hw;
struct audsettings hw_as;
+ AudioState *s = &glob_audio_state;
+ AudiodevPerDirectionOptions *pdo = glue(audio_get_pdo_, TYPE)(s->dev);
- 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 +431,7 @@ SW *glue (AUD_open_, TYPE) (
)
{
AudioState *s = &glob_audio_state;
+ AudiodevPerDirectionOptions *pdo = glue(audio_get_pdo_, TYPE)(s->dev);
if (audio_bug(__func__, !card || !name || !callback_fn || !as)) {
dolog ("card=%p name=%p callback_fn=%p as=%p\n",
@@ -422,7 +456,7 @@ SW *glue (AUD_open_, TYPE) (
return sw;
}
- if (!glue (conf.fixed_, TYPE).enabled && sw) {
+ if (!pdo->fixed_settings && sw) {
glue (AUD_close_, TYPE) (card, sw);
sw = NULL;
}
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 5bd034267f..8302f3e882 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -1125,7 +1125,7 @@ static ALSAConf glob_conf = {
.pcm_name_in = "default",
};
-static void *alsa_audio_init (void)
+static void *alsa_audio_init(Audiodev *dev)
{
ALSAConf *conf = g_malloc(sizeof(ALSAConf));
*conf = glob_conf;
diff --git a/audio/audio.c b/audio/audio.c
index 77216e5010..79ed360666 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -26,6 +26,9 @@
#include "audio.h"
#include "monitor/monitor.h"
#include "qemu/timer.h"
+#include "qapi/error.h"
+#include "qapi/qobject-input-visitor.h"
+#include "qapi/qapi-visit-audio.h"
#include "sysemu/sysemu.h"
#include "qemu/cutils.h"
#include "sysemu/replay.h"
@@ -46,14 +49,16 @@
The 1st one is the one used by default, that is the reason
that we generate the list.
*/
-static const char *audio_prio_list[] = {
+const char *audio_prio_list[] = {
"spice",
CONFIG_AUDIO_DRIVERS
"none",
"wav",
+ NULL
};
static QLIST_HEAD(, audio_driver) audio_drivers;
+static AudiodevListHead audiodevs = QSIMPLEQ_HEAD_INITIALIZER(audiodevs);
void audio_driver_register(audio_driver *drv)
{
@@ -80,61 +85,6 @@ audio_driver *audio_driver_lookup(const char *name)
return NULL;
}
-static void audio_module_load_all(void)
-{
- int i;
-
- for (i = 0; i < ARRAY_SIZE(audio_prio_list); i++) {
- audio_driver_lookup(audio_prio_list[i]);
- }
-}
-
-struct fixed_settings {
- int enabled;
- int nb_voices;
- int greedy;
- struct audsettings settings;
-};
-
-static struct {
- struct fixed_settings fixed_out;
- struct fixed_settings fixed_in;
- union {
- int hertz;
- int64_t ticks;
- } period;
- int try_poll_in;
- int try_poll_out;
-} conf = {
- .fixed_out = { /* DAC fixed settings */
- .enabled = 1,
- .nb_voices = 1,
- .greedy = 1,
- .settings = {
- .freq = 44100,
- .nchannels = 2,
- .fmt = AUDIO_FORMAT_S16,
- .endianness = AUDIO_HOST_ENDIANNESS,
- }
- },
-
- .fixed_in = { /* ADC fixed settings */
- .enabled = 1,
- .nb_voices = 1,
- .greedy = 1,
- .settings = {
- .freq = 44100,
- .nchannels = 2,
- .fmt = AUDIO_FORMAT_S16,
- .endianness = AUDIO_HOST_ENDIANNESS,
- }
- },
-
- .period = { .hertz = 100 },
- .try_poll_in = 1,
- .try_poll_out = 1,
-};
-
static AudioState glob_audio_state;
const struct mixeng_volume nominal_volume = {
@@ -151,9 +101,6 @@ const struct mixeng_volume nominal_volume = {
#ifdef AUDIO_IS_FLAWLESS_AND_NO_CHECKS_ARE_REQURIED
#error No its not
#else
-static void audio_print_options (const char *prefix,
- struct audio_option *opt);
-
int audio_bug (const char *funcname, int cond)
{
if (cond) {
@@ -161,16 +108,9 @@ int audio_bug (const char *funcname, int cond)
AUD_log (NULL, "A bug was just triggered in %s\n", funcname);
if (!shown) {
- struct audio_driver *d;
-
shown = 1;
AUD_log (NULL, "Save all your work and restart without audio\n");
- AUD_log (NULL, "Please send bug report to av1474@comtv.ru\n");
AUD_log (NULL, "I am sorry\n");
- d = glob_audio_state.drv;
- if (d) {
- audio_print_options (d->name, d->options);
- }
}
AUD_log (NULL, "Context:\n");
@@ -232,31 +172,6 @@ void *audio_calloc (const char *funcname, int nmemb, size_t size)
return g_malloc0 (len);
}
-static char *audio_alloc_prefix (const char *s)
-{
- const char qemu_prefix[] = "QEMU_";
- size_t len, i;
- char *r, *u;
-
- if (!s) {
- return NULL;
- }
-
- len = strlen (s);
- r = g_malloc (len + sizeof (qemu_prefix));
-
- u = r + sizeof (qemu_prefix) - 1;
-
- pstrcpy (r, len + sizeof (qemu_prefix), qemu_prefix);
- pstrcat (r, len + sizeof (qemu_prefix), s);
-
- for (i = 0; i < len; ++i) {
- u[i] = qemu_toupper(u[i]);
- }
-
- return r;
-}
-
static const char *audio_audfmt_to_string (AudioFormat fmt)
{
switch (fmt) {
@@ -382,78 +297,6 @@ 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)
{
@@ -1141,11 +984,11 @@ static void audio_reset_timer (AudioState *s)
{
if (audio_is_timer_needed ()) {
timer_mod_anticipate_ns(s->ts,
- qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + conf.period.ticks);
+ qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + s->period_ticks);
if (!audio_timer_running) {
audio_timer_running = true;
audio_timer_last = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
- trace_audio_timer_start(conf.period.ticks / SCALE_MS);
+ trace_audio_timer_start(s->period_ticks / SCALE_MS);
}
} else {
timer_del(s->ts);
@@ -1159,16 +1002,17 @@ static void audio_reset_timer (AudioState *s)
static void audio_timer (void *opaque)
{
int64_t now, diff;
+ AudioState *s = opaque;
now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
diff = now - audio_timer_last;
- if (diff > conf.period.ticks * 3 / 2) {
+ if (diff > s->period_ticks * 3 / 2) {
trace_audio_timer_delayed(diff / SCALE_MS);
}
audio_timer_last = now;
- audio_run ("timer");
- audio_reset_timer (opaque);
+ audio_run("timer");
+ audio_reset_timer(s);
}
/*
@@ -1228,7 +1072,7 @@ void AUD_set_active_out (SWVoiceOut *sw, int on)
if (!hw->enabled) {
hw->enabled = 1;
if (s->vm_running) {
- hw->pcm_ops->ctl_out (hw, VOICE_ENABLE, conf.try_poll_out);
+ hw->pcm_ops->ctl_out(hw, VOICE_ENABLE, true);
audio_reset_timer (s);
}
}
@@ -1273,7 +1117,7 @@ void AUD_set_active_in (SWVoiceIn *sw, int on)
if (!hw->enabled) {
hw->enabled = 1;
if (s->vm_running) {
- hw->pcm_ops->ctl_in (hw, VOICE_ENABLE, conf.try_poll_in);
+ hw->pcm_ops->ctl_in(hw, VOICE_ENABLE, true);
audio_reset_timer (s);
}
}
@@ -1594,169 +1438,13 @@ void audio_run (const char *msg)
#endif
}
-static struct audio_option audio_options[] = {
- /* DAC */
- {
- .name = "DAC_FIXED_SETTINGS",
- .tag = AUD_OPT_BOOL,
- .valp = &conf.fixed_out.enabled,
- .descr = "Use fixed settings for host DAC"
- },
- {
- .name = "DAC_FIXED_FREQ",
- .tag = AUD_OPT_INT,
- .valp = &conf.fixed_out.settings.freq,
- .descr = "Frequency for fixed host DAC"
- },
- {
- .name = "DAC_FIXED_FMT",
- .tag = AUD_OPT_FMT,
- .valp = &conf.fixed_out.settings.fmt,
- .descr = "Format for fixed host DAC"
- },
- {
- .name = "DAC_FIXED_CHANNELS",
- .tag = AUD_OPT_INT,
- .valp = &conf.fixed_out.settings.nchannels,
- .descr = "Number of channels for fixed DAC (1 - mono, 2 - stereo)"
- },
- {
- .name = "DAC_VOICES",
- .tag = AUD_OPT_INT,
- .valp = &conf.fixed_out.nb_voices,
- .descr = "Number of voices for DAC"
- },
- {
- .name = "DAC_TRY_POLL",
- .tag = AUD_OPT_BOOL,
- .valp = &conf.try_poll_out,
- .descr = "Attempt using poll mode for DAC"
- },
- /* ADC */
- {
- .name = "ADC_FIXED_SETTINGS",
- .tag = AUD_OPT_BOOL,
- .valp = &conf.fixed_in.enabled,
- .descr = "Use fixed settings for host ADC"
- },
- {
- .name = "ADC_FIXED_FREQ",
- .tag = AUD_OPT_INT,
- .valp = &conf.fixed_in.settings.freq,
- .descr = "Frequency for fixed host ADC"
- },
- {
- .name = "ADC_FIXED_FMT",
- .tag = AUD_OPT_FMT,
- .valp = &conf.fixed_in.settings.fmt,
- .descr = "Format for fixed host ADC"
- },
- {
- .name = "ADC_FIXED_CHANNELS",
- .tag = AUD_OPT_INT,
- .valp = &conf.fixed_in.settings.nchannels,
- .descr = "Number of channels for fixed ADC (1 - mono, 2 - stereo)"
- },
- {
- .name = "ADC_VOICES",
- .tag = AUD_OPT_INT,
- .valp = &conf.fixed_in.nb_voices,
- .descr = "Number of voices for ADC"
- },
- {
- .name = "ADC_TRY_POLL",
- .tag = AUD_OPT_BOOL,
- .valp = &conf.try_poll_in,
- .descr = "Attempt using poll mode for ADC"
- },
- /* Misc */
- {
- .name = "TIMER_PERIOD",
- .tag = AUD_OPT_INT,
- .valp = &conf.period.hertz,
- .descr = "Timer period in HZ (0 - use lowest possible)"
- },
- { /* End of list */ }
-};
-
-static void audio_pp_nb_voices (const char *typ, int nb)
-{
- switch (nb) {
- case 0:
- printf ("Does not support %s\n", typ);
- break;
- case 1:
- printf ("One %s voice\n", typ);
- break;
- case INT_MAX:
- printf ("Theoretically supports many %s voices\n", typ);
- break;
- default:
- printf ("Theoretically supports up to %d %s voices\n", nb, typ);
- break;
- }
-
-}
-
-void AUD_help (void)
-{
- struct audio_driver *d;
-
- /* make sure we print the help text for modular drivers too */
- audio_module_load_all();
-
- audio_process_options ("AUDIO", audio_options);
- QLIST_FOREACH(d, &audio_drivers, next) {
- if (d->options) {
- audio_process_options (d->name, d->options);
- }
- }
-
- printf ("Audio options:\n");
- audio_print_options ("AUDIO", audio_options);
- printf ("\n");
-
- printf ("Available drivers:\n");
-
- QLIST_FOREACH(d, &audio_drivers, next) {
-
- printf ("Name: %s\n", d->name);
- printf ("Description: %s\n", d->descr);
-
- audio_pp_nb_voices ("playback", d->max_voices_out);
- audio_pp_nb_voices ("capture", d->max_voices_in);
-
- if (d->options) {
- printf ("Options:\n");
- audio_print_options (d->name, d->options);
- }
- else {
- printf ("No options\n");
- }
- printf ("\n");
- }
-
- printf (
- "Options are settable through environment variables.\n"
- "Example:\n"
-#ifdef _WIN32
- " set QEMU_AUDIO_DRV=wav\n"
- " set QEMU_WAV_PATH=c:\\tune.wav\n"
-#else
- " export QEMU_AUDIO_DRV=wav\n"
- " export QEMU_WAV_PATH=$HOME/tune.wav\n"
- "(for csh replace export with setenv in the above)\n"
-#endif
- " qemu ...\n\n"
- );
-}
-
-static int audio_driver_init(AudioState *s, struct audio_driver *drv, bool msg)
+static int audio_driver_init(AudioState *s, struct audio_driver *drv,
+ bool msg, Audiodev *dev)
{
if (drv->options) {
audio_process_options (drv->name, drv->options);
}
- s->drv_opaque = drv->init ();
+ s->drv_opaque = drv->init(dev);
if (s->drv_opaque) {
audio_init_nb_voices_out (drv);
@@ -1782,11 +1470,11 @@ static void audio_vm_change_state_handler (void *opaque, int running,
s->vm_running = running;
while ((hwo = audio_pcm_hw_find_any_enabled_out (hwo))) {
- hwo->pcm_ops->ctl_out (hwo, op, conf.try_poll_out);
+ hwo->pcm_ops->ctl_out(hwo, op, true);
}
while ((hwi = audio_pcm_hw_find_any_enabled_in (hwi))) {
- hwi->pcm_ops->ctl_in (hwi, op, conf.try_poll_in);
+ hwi->pcm_ops->ctl_in(hwi, op, true);
}
audio_reset_timer (s);
}
@@ -1836,6 +1524,11 @@ void audio_cleanup(void)
s->drv->fini (s->drv_opaque);
s->drv = NULL;
}
+
+ if (s->dev) {
+ qapi_free_Audiodev(s->dev);
+ s->dev = NULL;
+ }
}
static const VMStateDescription vmstate_audio = {
@@ -1847,19 +1540,58 @@ static const VMStateDescription vmstate_audio = {
}
};
-static void audio_init (void)
+static void audio_validate_opts(Audiodev *dev, Error **errp);
+
+static AudiodevListEntry *audiodev_find(
+ AudiodevListHead *head, const char *drvname)
+{
+ AudiodevListEntry *e;
+ QSIMPLEQ_FOREACH(e, head, next) {
+ if (strcmp(AudiodevDriver_str(e->dev->driver), drvname) == 0) {
+ return e;
+ }
+ }
+
+ return NULL;
+}
+
+static int audio_init(Audiodev *dev)
{
size_t i;
int done = 0;
- const char *drvname;
+ const char *drvname = NULL;
VMChangeStateEntry *e;
AudioState *s = &glob_audio_state;
struct audio_driver *driver;
+ /* silence gcc warning about uninitialized variable */
+ AudiodevListHead head = QSIMPLEQ_HEAD_INITIALIZER(head);
if (s->drv) {
- return;
+ if (dev) {
+ dolog("Cannot create more than one audio backend, sorry\n");
+ qapi_free_Audiodev(dev);
+ }
+ return -1;
}
+ if (dev) {
+ /* -audiodev option */
+ drvname = AudiodevDriver_str(dev->driver);
+ } else {
+ /* legacy implicit initialization */
+ head = audio_handle_legacy_opts();
+ /*
+ * In case of legacy initialization, all Audiodevs in the list will have
+ * the same configuration (except the driver), so it does't matter which
+ * one we chose. We need an Audiodev to set up AudioState before we can
+ * init a driver. Also note that dev at this point is still in the
+ * list.
+ */
+ dev = QSIMPLEQ_FIRST(&head)->dev;
+ audio_validate_opts(dev, &error_abort);
+ }
+ s->dev = dev;
+
QLIST_INIT (&s->hw_head_out);
QLIST_INIT (&s->hw_head_in);
QLIST_INIT (&s->cap_head);
@@ -1867,10 +1599,8 @@ static void audio_init (void)
s->ts = timer_new_ns(QEMU_CLOCK_VIRTUAL, audio_timer, s);
- audio_process_options ("AUDIO", audio_options);
-
- s->nb_hw_voices_out = conf.fixed_out.nb_voices;
- s->nb_hw_voices_in = conf.fixed_in.nb_voices;
+ s->nb_hw_voices_out = audio_get_pdo_out(dev)->voices;
+ s->nb_hw_voices_in = audio_get_pdo_in(dev)->voices;
if (s->nb_hw_voices_out <= 0) {
dolog ("Bogus number of playback voices %d, setting to 1\n",
@@ -1884,46 +1614,42 @@ static void audio_init (void)
s->nb_hw_voices_in = 0;
}
- {
- int def;
- drvname = audio_get_conf_str ("QEMU_AUDIO_DRV", NULL, &def);
- }
-
if (drvname) {
driver = audio_driver_lookup(drvname);
if (driver) {
- done = !audio_driver_init(s, driver, true);
+ done = !audio_driver_init(s, driver, true, dev);
} else {
dolog ("Unknown audio driver `%s'\n", drvname);
- dolog ("Run with -audio-help to list available drivers\n");
}
- }
-
- if (!done) {
- for (i = 0; !done && i < ARRAY_SIZE(audio_prio_list); i++) {
+ } else {
+ for (i = 0; audio_prio_list[i]; i++) {
+ AudiodevListEntry *e = audiodev_find(&head, audio_prio_list[i]);
driver = audio_driver_lookup(audio_prio_list[i]);
- if (driver && driver->can_be_default) {
- done = !audio_driver_init(s, driver, false);
+
+ if (e && driver) {
+ s->dev = dev = e->dev;
+ audio_validate_opts(dev, &error_abort);
+ done = !audio_driver_init(s, driver, false, dev);
+ if (done) {
+ e->dev = NULL;
+ break;
+ }
}
}
}
+ audio_free_audiodev_list(&head);
if (!done) {
driver = audio_driver_lookup("none");
- done = !audio_driver_init(s, driver, false);
+ done = !audio_driver_init(s, driver, false, dev);
assert(done);
dolog("warning: Using timer based audio emulation\n");
}
- if (conf.period.hertz <= 0) {
- if (conf.period.hertz < 0) {
- dolog ("warning: Timer period is negative - %d "
- "treating as zero\n",
- conf.period.hertz);
- }
- conf.period.ticks = 1;
+ if (dev->timer_period <= 0) {
+ s->period_ticks = 1;
} else {
- conf.period.ticks = NANOSECONDS_PER_SECOND / conf.period.hertz;
+ s->period_ticks = NANOSECONDS_PER_SECOND / dev->timer_period;
}
e = qemu_add_vm_change_state_handler (audio_vm_change_state_handler, s);
@@ -1934,11 +1660,22 @@ static void audio_init (void)
QLIST_INIT (&s->card_head);
vmstate_register (NULL, 0, &vmstate_audio, s);
+ return 0;
+}
+
+void audio_free_audiodev_list(AudiodevListHead *head)
+{
+ AudiodevListEntry *e;
+ while ((e = QSIMPLEQ_FIRST(head))) {
+ QSIMPLEQ_REMOVE_HEAD(head, next);
+ qapi_free_Audiodev(e->dev);
+ g_free(e);
+ }
}
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);
@@ -2078,3 +1815,174 @@ void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
}
}
}
+
+void audio_create_pdos(Audiodev *dev)
+{
+ switch (dev->driver) {
+#define CASE(DRIVER, driver, pdo_name) \
+ case AUDIODEV_DRIVER_##DRIVER: \
+ if (!dev->u.driver.has_in) { \
+ dev->u.driver.in = g_malloc0( \
+ sizeof(Audiodev##pdo_name##PerDirectionOptions)); \
+ dev->u.driver.has_in = true; \
+ } \
+ if (!dev->u.driver.has_out) { \
+ dev->u.driver.out = g_malloc0( \
+ sizeof(AudiodevAlsaPerDirectionOptions)); \
+ dev->u.driver.has_out = true; \
+ } \
+ break
+
+ CASE(NONE, none, );
+ CASE(ALSA, alsa, Alsa);
+ CASE(COREAUDIO, coreaudio, Coreaudio);
+ CASE(DSOUND, dsound, );
+ CASE(OSS, oss, Oss);
+ CASE(PA, pa, Pa);
+ CASE(SDL, sdl, );
+ CASE(SPICE, spice, );
+ CASE(WAV, wav, );
+
+ case AUDIODEV_DRIVER__MAX:
+ abort();
+ };
+}
+
+static void audio_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 void audio_validate_opts(Audiodev *dev, Error **errp)
+{
+ Error *err = NULL;
+
+ audio_create_pdos(dev);
+
+ audio_validate_per_direction_opts(audio_get_pdo_in(dev), &err);
+ if (err) {
+ error_propagate(errp, err);
+ return;
+ }
+
+ audio_validate_per_direction_opts(audio_get_pdo_out(dev), &err);
+ if (err) {
+ error_propagate(errp, err);
+ return;
+ }
+
+ if (!dev->has_timer_period) {
+ dev->has_timer_period = true;
+ dev->timer_period = 10000; /* 100Hz -> 10ms */
+ }
+}
+
+void audio_parse_option(const char *opt)
+{
+ AudiodevListEntry *e;
+ Audiodev *dev = NULL;
+
+ Visitor *v = qobject_input_visitor_new_str(opt, "driver", &error_fatal);
+ visit_type_Audiodev(v, NULL, &dev, &error_fatal);
+ visit_free(v);
+
+ audio_validate_opts(dev, &error_fatal);
+
+ e = g_malloc0(sizeof(AudiodevListEntry));
+ e->dev = dev;
+ QSIMPLEQ_INSERT_TAIL(&audiodevs, e, next);
+}
+
+void audio_init_audiodevs(void)
+{
+ AudiodevListEntry *e;
+
+ QSIMPLEQ_FOREACH(e, &audiodevs, next) {
+ audio_init(e->dev);
+ }
+}
+
+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_length ? pdo->buffer_length : def_usecs;
+ return (as->freq * usecs + 500000) / 1000000;
+}
+
+/* samples = channels * frames = channels * freq * usec / 1e6 */
+int audio_buffer_samples(AudiodevPerDirectionOptions *pdo,
+ audsettings *as, int def_usecs)
+{
+ return as->nchannels * audio_buffer_frames(pdo, as, def_usecs);
+}
+
+/*
+ * bytes = bytes_per_sample * samples =
+ * bytes_per_sample * channels * freq * usec / 1e6
+ */
+int audio_buffer_bytes(AudiodevPerDirectionOptions *pdo,
+ audsettings *as, int def_usecs)
+{
+ return audio_buffer_samples(pdo, as, def_usecs) *
+ audioformat_bytes_per_sample(as->fmt);
+}
diff --git a/audio/audio_legacy.c b/audio/audio_legacy.c
new file mode 100644
index 0000000000..ad3b781add
--- /dev/null
+++ b/audio/audio_legacy.c
@@ -0,0 +1,293 @@
+/*
+ * QEMU Audio subsystem: legacy configuration handling
+ *
+ * Copyright (c) 2015-2019 Zoltán Kővágó <DirtY.iCE.hu@gmail.com>
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+#include "qemu/osdep.h"
+#include "audio.h"
+#include "audio_int.h"
+#include "qemu-common.h"
+#include "qemu/cutils.h"
+#include "qapi/error.h"
+#include "qapi/qapi-visit-audio.h"
+#include "qapi/visitor-impl.h"
+
+#define AUDIO_CAP "audio-legacy"
+#include "audio_int.h"
+
+static uint32_t toui32(const char *str)
+{
+ unsigned long long ret;
+ if (parse_uint_full(str, &ret, 10) || ret > UINT32_MAX) {
+ dolog("Invalid integer value `%s'\n", str);
+ exit(1);
+ }
+ return ret;
+}
+
+/* helper functions to convert env variables */
+static void get_bool(const char *env, bool *dst, bool *has_dst)
+{
+ const char *val = getenv(env);
+ if (val) {
+ *dst = toui32(val) != 0;
+ *has_dst = true;
+ }
+}
+
+static void get_int(const char *env, uint32_t *dst, bool *has_dst)
+{
+ const char *val = getenv(env);
+ if (val) {
+ *dst = toui32(val);
+ *has_dst = true;
+ }
+}
+
+static void get_fmt(const char *env, AudioFormat *dst, bool *has_dst)
+{
+ const char *val = getenv(env);
+ if (val) {
+ size_t i;
+ for (i = 0; AudioFormat_lookup.size; ++i) {
+ if (strcasecmp(val, AudioFormat_lookup.array[i]) == 0) {
+ *dst = i;
+ *has_dst = true;
+ return;
+ }
+ }
+
+ dolog("Invalid audio format `%s'\n", val);
+ exit(1);
+ }
+}
+
+/* backend specific functions */
+/* todo */
+
+/* general */
+static void handle_per_direction(
+ AudiodevPerDirectionOptions *pdo, const char *prefix)
+{
+ char buf[64];
+ size_t len = strlen(prefix);
+
+ memcpy(buf, prefix, len);
+ strcpy(buf + len, "FIXED_SETTINGS");
+ get_bool(buf, &pdo->fixed_settings, &pdo->has_fixed_settings);
+
+ strcpy(buf + len, "FIXED_FREQ");
+ get_int(buf, &pdo->frequency, &pdo->has_frequency);
+
+ strcpy(buf + len, "FIXED_FMT");
+ get_fmt(buf, &pdo->format, &pdo->has_format);
+
+ strcpy(buf + len, "FIXED_CHANNELS");
+ get_int(buf, &pdo->channels, &pdo->has_channels);
+
+ strcpy(buf + len, "VOICES");
+ get_int(buf, &pdo->voices, &pdo->has_voices);
+}
+
+static AudiodevListEntry *legacy_opt(const char *drvname)
+{
+ AudiodevListEntry *e = g_malloc0(sizeof(AudiodevListEntry));
+ e->dev = g_malloc0(sizeof(Audiodev));
+ e->dev->id = g_strdup(drvname);
+ e->dev->driver = qapi_enum_parse(
+ &AudiodevDriver_lookup, drvname, -1, &error_abort);
+
+ audio_create_pdos(e->dev);
+
+ handle_per_direction(audio_get_pdo_in(e->dev), "QEMU_AUDIO_ADC_");
+ handle_per_direction(audio_get_pdo_out(e->dev), "QEMU_AUDIO_DAC_");
+
+ get_int("QEMU_AUDIO_TIMER_PERIOD",
+ &e->dev->timer_period, &e->dev->has_timer_period);
+
+ return e;
+}
+
+AudiodevListHead audio_handle_legacy_opts(void)
+{
+ const char *drvname = getenv("QEMU_AUDIO_DRV");
+ AudiodevListHead head = QSIMPLEQ_HEAD_INITIALIZER(head);
+
+ if (drvname) {
+ AudiodevListEntry *e;
+ audio_driver *driver = audio_driver_lookup(drvname);
+ if (!driver) {
+ dolog("Unknown audio driver `%s'\n", drvname);
+ exit(1);
+ }
+ e = legacy_opt(drvname);
+ QSIMPLEQ_INSERT_TAIL(&head, e, next);
+ } else {
+ for (int i = 0; audio_prio_list[i]; i++) {
+ audio_driver *driver = audio_driver_lookup(audio_prio_list[i]);
+ if (driver && driver->can_be_default) {
+ AudiodevListEntry *e = legacy_opt(driver->name);
+ QSIMPLEQ_INSERT_TAIL(&head, e, next);
+ }
+ }
+ if (QSIMPLEQ_EMPTY(&head)) {
+ dolog("Internal error: no default audio driver available\n");
+ exit(1);
+ }
+ }
+
+ return head;
+}
+
+/* visitor to print -audiodev option */
+typedef struct {
+ Visitor visitor;
+
+ bool comma;
+ GList *path;
+} LegacyPrintVisitor;
+
+static void lv_start_struct(Visitor *v, const char *name, void **obj,
+ size_t size, Error **errp)
+{
+ LegacyPrintVisitor *lv = (LegacyPrintVisitor *) v;
+ lv->path = g_list_append(lv->path, g_strdup(name));
+}
+
+static void lv_end_struct(Visitor *v, void **obj)
+{
+ LegacyPrintVisitor *lv = (LegacyPrintVisitor *) v;
+ lv->path = g_list_delete_link(lv->path, g_list_last(lv->path));
+}
+
+static void lv_print_key(Visitor *v, const char *name)
+{
+ GList *e;
+ LegacyPrintVisitor *lv = (LegacyPrintVisitor *) v;
+ if (lv->comma) {
+ putchar(',');
+ } else {
+ lv->comma = true;
+ }
+
+ for (e = lv->path; e; e = e->next) {
+ if (e->data) {
+ printf("%s.", (const char *) e->data);
+ }
+ }
+
+ printf("%s=", name);
+}
+
+static void lv_type_int64(Visitor *v, const char *name, int64_t *obj,
+ Error **errp)
+{
+ lv_print_key(v, name);
+ printf("%" PRIi64, *obj);
+}
+
+static void lv_type_uint64(Visitor *v, const char *name, uint64_t *obj,
+ Error **errp)
+{
+ lv_print_key(v, name);
+ printf("%" PRIu64, *obj);
+}
+
+static void lv_type_bool(Visitor *v, const char *name, bool *obj, Error **errp)
+{
+ lv_print_key(v, name);
+ printf("%s", *obj ? "on" : "off");
+}
+
+static void lv_type_str(Visitor *v, const char *name, char **obj, Error **errp)
+{
+ const char *str = *obj;
+ lv_print_key(v, name);
+
+ while (*str) {
+ if (*str == ',') {
+ putchar(',');
+ }
+ putchar(*str++);
+ }
+}
+
+static void lv_complete(Visitor *v, void *opaque)
+{
+ LegacyPrintVisitor *lv = (LegacyPrintVisitor *) v;
+ assert(lv->path == NULL);
+}
+
+static void lv_free(Visitor *v)
+{
+ LegacyPrintVisitor *lv = (LegacyPrintVisitor *) v;
+
+ g_list_free_full(lv->path, g_free);
+ g_free(lv);
+}
+
+static Visitor *legacy_visitor_new(void)
+{
+ LegacyPrintVisitor *lv = g_malloc0(sizeof(LegacyPrintVisitor));
+
+ lv->visitor.start_struct = lv_start_struct;
+ lv->visitor.end_struct = lv_end_struct;
+ /* lists not supported */
+ lv->visitor.type_int64 = lv_type_int64;
+ lv->visitor.type_uint64 = lv_type_uint64;
+ lv->visitor.type_bool = lv_type_bool;
+ lv->visitor.type_str = lv_type_str;
+
+ lv->visitor.type = VISITOR_OUTPUT;
+ lv->visitor.complete = lv_complete;
+ lv->visitor.free = lv_free;
+
+ return &lv->visitor;
+}
+
+void audio_legacy_help(void)
+{
+ AudiodevListHead head;
+ AudiodevListEntry *e;
+
+ printf("Environment variable based configuration deprecated.\n");
+ printf("Please use the new -audiodev option.\n");
+
+ head = audio_handle_legacy_opts();
+ printf("\nEquivalent -audiodev to your current environment variables:\n");
+ if (!getenv("QEMU_AUDIO_DRV")) {
+ printf("(Since you didn't specify QEMU_AUDIO_DRV, I'll list all "
+ "possibilities)\n");
+ }
+
+ QSIMPLEQ_FOREACH(e, &head, next) {
+ Visitor *v;
+ Audiodev *dev = e->dev;
+ printf("-audiodev ");
+
+ v = legacy_visitor_new();
+ visit_type_Audiodev(v, NULL, &dev, &error_abort);
+ visit_free(v);
+
+ printf("\n");
+ }
+ audio_free_audiodev_list(&head);
+}
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index 638c60b300..7d4225dbee 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -685,7 +685,7 @@ static CoreaudioConf glob_conf = {
.nbuffers = 4,
};
-static void *coreaudio_audio_init (void)
+static void *coreaudio_audio_init(Audiodev *dev)
{
CoreaudioConf *conf = g_malloc(sizeof(CoreaudioConf));
*conf = glob_conf;
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index 3ed73a30d1..02fe777cba 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -783,7 +783,7 @@ static void dsound_audio_fini (void *opaque)
g_free(s);
}
-static void *dsound_audio_init (void)
+static void *dsound_audio_init(Audiodev *dev)
{
int err;
HRESULT hr;
diff --git a/audio/noaudio.c b/audio/noaudio.c
index 1bfebeca7d..79690af1ea 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -136,7 +136,7 @@ static int no_ctl_in (HWVoiceIn *hw, int cmd, ...)
return 0;
}
-static void *no_audio_init (void)
+static void *no_audio_init(Audiodev *dev)
{
return &no_audio_init;
}
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 355e8fbda5..e0cadbef29 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -842,7 +842,7 @@ static OSSConf glob_conf = {
.policy = 5
};
-static void *oss_audio_init (void)
+static void *oss_audio_init(Audiodev *dev)
{
OSSConf *conf = g_malloc(sizeof(OSSConf));
*conf = glob_conf;
diff --git a/audio/paaudio.c b/audio/paaudio.c
index 8246f260a8..d649c58e3d 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -812,7 +812,7 @@ static PAConf glob_conf = {
.samples = 4096,
};
-static void *qpa_audio_init (void)
+static void *qpa_audio_init(Audiodev *dev)
{
if (glob_conf.server == NULL) {
char pidfile[64];
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index 4cd4cbaf00..b9ae506a56 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -315,7 +315,7 @@ static int sdl_ctl_out (HWVoiceOut *hw, int cmd, ...)
return 0;
}
-static void *sdl_audio_init (void)
+static void *sdl_audio_init(Audiodev *dev)
{
SDLAudioState *s = &glob_sdl;
if (s->driver_created) {
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index 3aeb0cb357..affc3df17f 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -77,7 +77,7 @@ static const SpiceRecordInterface record_sif = {
.base.minor_version = SPICE_INTERFACE_RECORD_MINOR,
};
-static void *spice_audio_init (void)
+static void *spice_audio_init(Audiodev *dev)
{
if (!using_spice) {
return NULL;
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 35a614785e..9eff3555b3 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -232,7 +232,7 @@ static WAVConf glob_conf = {
.wav_path = "qemu.wav"
};
-static void *wav_audio_init (void)
+static void *wav_audio_init(Audiodev *dev)
{
WAVConf *conf = g_malloc(sizeof(WAVConf));
*conf = glob_conf;
diff --git a/vl.c b/vl.c
index 4a350de5cd..db04aea7f8 100644
--- a/vl.c
+++ b/vl.c
@@ -3253,9 +3253,12 @@ int main(int argc, char **argv, char **envp)
add_device_config(DEV_BT, optarg);
break;
case QEMU_OPTION_audio_help:
- AUD_help ();
+ audio_legacy_help();
exit (0);
break;
+ case QEMU_OPTION_audiodev:
+ audio_parse_option(optarg);
+ break;
case QEMU_OPTION_soundhw:
select_soundhw (optarg);
break;
@@ -4447,6 +4450,8 @@ int main(int argc, char **argv, char **envp)
/* do monitor/qmp handling at preconfig state if requested */
main_loop();
+ audio_init_audiodevs();
+
/* from here on runstate is RUN_STATE_PRELAUNCH */
machine_run_board_init(current_machine);
diff --git a/audio/Makefile.objs b/audio/Makefile.objs
index db4fa7f18f..dca87f6347 100644
--- a/audio/Makefile.objs
+++ b/audio/Makefile.objs
@@ -1,4 +1,4 @@
-common-obj-y = audio.o noaudio.o wavaudio.o mixeng.o
+common-obj-y = audio.o audio_legacy.o noaudio.o wavaudio.o mixeng.o
common-obj-$(CONFIG_SPICE) += spiceaudio.o
common-obj-$(CONFIG_AUDIO_COREAUDIO) += coreaudio.o
common-obj-$(CONFIG_AUDIO_DSOUND) += dsoundaudio.o
--
2.20.1
^ permalink raw reply related [flat|nested] 18+ messages in thread* [Qemu-devel] [PATCH v6 05/14] alsaaudio: port to -audiodev config
2019-03-08 22:34 [Qemu-devel] [PATCH v6 00/14] Audio patches Kővágó, Zoltán
` (3 preceding siblings ...)
2019-03-08 22:34 ` [Qemu-devel] [PATCH v6 04/14] audio: -audiodev command line option basic implementation Kővágó, Zoltán
@ 2019-03-08 22:34 ` Kővágó, Zoltán
2019-03-08 22:34 ` [Qemu-devel] [PATCH v6 06/14] coreaudio: " Kővágó, Zoltán
` (9 subsequent siblings)
14 siblings, 0 replies; 18+ messages in thread
From: Kővágó, Zoltán @ 2019-03-08 22:34 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/alsaaudio.c | 329 +++++++++++++------------------------------
audio/audio_legacy.c | 84 ++++++++++-
2 files changed, 181 insertions(+), 232 deletions(-)
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index 8302f3e882..49e6884309 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -33,28 +33,9 @@
#define AUDIO_CAP "alsa"
#include "audio_int.h"
-typedef struct ALSAConf {
- int size_in_usec_in;
- int size_in_usec_out;
- const char *pcm_name_in;
- const char *pcm_name_out;
- unsigned int buffer_size_in;
- unsigned int period_size_in;
- unsigned int buffer_size_out;
- unsigned int period_size_out;
- unsigned int threshold;
-
- int buffer_size_in_overridden;
- int period_size_in_overridden;
-
- int buffer_size_out_overridden;
- int period_size_out_overridden;
-} ALSAConf;
-
struct pollhlp {
snd_pcm_t *handle;
struct pollfd *pfds;
- ALSAConf *conf;
int count;
int mask;
};
@@ -66,6 +47,7 @@ typedef struct ALSAVoiceOut {
void *pcm_buf;
snd_pcm_t *handle;
struct pollhlp pollhlp;
+ Audiodev *dev;
} ALSAVoiceOut;
typedef struct ALSAVoiceIn {
@@ -73,16 +55,13 @@ typedef struct ALSAVoiceIn {
snd_pcm_t *handle;
void *pcm_buf;
struct pollhlp pollhlp;
+ Audiodev *dev;
} ALSAVoiceIn;
struct alsa_params_req {
int freq;
snd_pcm_format_t fmt;
int nchannels;
- int size_in_usec;
- int override_mask;
- unsigned int buffer_size;
- unsigned int period_size;
};
struct alsa_params_obt {
@@ -408,17 +387,18 @@ 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,
+ AudiodevAlsaPerDirectionOptions *apdo)
{
- dolog ("parameter | requested value | obtained value\n");
- dolog ("format | %10d | %10d\n", req->fmt, obtfmt);
- dolog ("channels | %10d | %10d\n",
- 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 ("obtained: samples %ld\n", obt->samples);
+ dolog("parameter | requested value | obtained value\n");
+ dolog("format | %10d | %10d\n", req->fmt, obtfmt);
+ dolog("channels | %10d | %10d\n",
+ req->nchannels, obt->nchannels);
+ dolog("frequency | %10d | %10d\n", req->freq, obt->freq);
+ dolog("============================================\n");
+ dolog("requested: buffer len %" PRId32 " period len %" PRId32 "\n",
+ apdo->buffer_length, apdo->period_length);
+ dolog("obtained: samples %ld\n", obt->samples);
}
static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
@@ -451,23 +431,23 @@ 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)
{
+ AudiodevAlsaOptions *aopts = &dev->u.alsa;
+ AudiodevAlsaPerDirectionOptions *apdo = in ? aopts->in : aopts->out;
snd_pcm_t *handle;
snd_pcm_hw_params_t *hw_params;
int err;
- int size_in_usec;
unsigned int freq, nchannels;
- const char *pcm_name = in ? conf->pcm_name_in : conf->pcm_name_out;
+ const char *pcm_name = apdo->has_dev ? apdo->dev : "default";
snd_pcm_uframes_t obt_buffer_size;
const char *typ = in ? "ADC" : "DAC";
snd_pcm_format_t obtfmt;
freq = req->freq;
nchannels = req->nchannels;
- size_in_usec = req->size_in_usec;
snd_pcm_hw_params_alloca (&hw_params);
@@ -527,79 +507,42 @@ static int alsa_open (int in, struct alsa_params_req *req,
goto err;
}
- if (req->buffer_size) {
- unsigned long obt;
+ if (apdo->buffer_length) {
+ int dir = 0;
+ unsigned int btime = apdo->buffer_length;
- if (size_in_usec) {
- int dir = 0;
- unsigned int btime = req->buffer_size;
+ err = snd_pcm_hw_params_set_buffer_time_near(
+ handle, hw_params, &btime, &dir);
- 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_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);
+ alsa_logerr2(err, typ, "Failed to set buffer time to %" PRId32 "\n",
+ apdo->buffer_length);
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 (apdo->has_buffer_length && btime != apdo->buffer_length) {
+ dolog("Requested buffer time %" PRId32
+ " was rejected, using %u\n", apdo->buffer_length, btime);
+ }
}
- if (req->period_size) {
- unsigned long obt;
+ if (apdo->period_length) {
+ int dir = 0;
+ unsigned int ptime = apdo->period_length;
- 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 %" PRId32 "\n",
+ apdo->period_length);
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 (apdo->has_period_length && ptime != apdo->period_length) {
+ dolog("Requested period time %" PRId32 " was rejected, using %d\n",
+ apdo->period_length, ptime);
+ }
}
err = snd_pcm_hw_params (handle, hw_params);
@@ -631,33 +574,12 @@ 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(qapi_AudiodevAlsaPerDirectionOptions_base(apdo),
+ &as, aopts->threshold));
}
obt->nchannels = nchannels;
@@ -670,11 +592,11 @@ 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, apdo);
}
#ifdef DEBUG
- alsa_dump_info (req, obt, obtfmt);
+ alsa_dump_info(req, obt, obtfmt, pdo);
#endif
return 0;
@@ -800,19 +722,13 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
struct alsa_params_obt obt;
snd_pcm_t *handle;
struct audsettings obt_as;
- ALSAConf *conf = drv_opaque;
+ Audiodev *dev = drv_opaque;
req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
req.freq = as->freq;
req.nchannels = as->nchannels;
- req.period_size = conf->period_size_out;
- req.buffer_size = conf->buffer_size_out;
- req.size_in_usec = conf->size_in_usec_out;
- req.override_mask =
- (conf->period_size_out_overridden ? 1 : 0) |
- (conf->buffer_size_out_overridden ? 2 : 0);
- if (alsa_open (0, &req, &obt, &handle, conf)) {
+ if (alsa_open(0, &req, &obt, &handle, dev)) {
return -1;
}
@@ -833,7 +749,7 @@ static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
}
alsa->handle = handle;
- alsa->pollhlp.conf = conf;
+ alsa->dev = dev;
return 0;
}
@@ -873,16 +789,12 @@ static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl)
static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+ AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.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->try_poll;
ldebug ("enabling voice\n");
if (poll_mode && alsa_poll_out (hw)) {
@@ -911,19 +823,13 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
struct alsa_params_obt obt;
snd_pcm_t *handle;
struct audsettings obt_as;
- ALSAConf *conf = drv_opaque;
+ Audiodev *dev = drv_opaque;
req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
req.freq = as->freq;
req.nchannels = as->nchannels;
- req.period_size = conf->period_size_in;
- req.buffer_size = conf->buffer_size_in;
- req.size_in_usec = conf->size_in_usec_in;
- req.override_mask =
- (conf->period_size_in_overridden ? 1 : 0) |
- (conf->buffer_size_in_overridden ? 2 : 0);
- if (alsa_open (1, &req, &obt, &handle, conf)) {
+ if (alsa_open(1, &req, &obt, &handle, dev)) {
return -1;
}
@@ -944,7 +850,7 @@ static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
}
alsa->handle = handle;
- alsa->pollhlp.conf = conf;
+ alsa->dev = dev;
return 0;
}
@@ -1086,16 +992,12 @@ static int alsa_read (SWVoiceIn *sw, void *buf, int size)
static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
{
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+ AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.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->try_poll;
ldebug ("enabling voice\n");
if (poll_mode && alsa_poll_in (hw)) {
@@ -1118,88 +1020,54 @@ 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_init_per_direction(AudiodevAlsaPerDirectionOptions *apdo)
+{
+ if (!apdo->has_try_poll) {
+ apdo->try_poll = true;
+ apdo->has_try_poll = true;
+ }
+}
static void *alsa_audio_init(Audiodev *dev)
{
- ALSAConf *conf = g_malloc(sizeof(ALSAConf));
- *conf = glob_conf;
- return conf;
+ AudiodevAlsaOptions *aopts;
+ assert(dev->driver == AUDIODEV_DRIVER_ALSA);
+
+ aopts = &dev->u.alsa;
+ alsa_init_per_direction(aopts->in);
+ alsa_init_per_direction(aopts->out);
+
+ /*
+ * 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->u.alsa.out->has_period_length) {
+ /* 1024 frames assuming 44100Hz */
+ dev->u.alsa.out->period_length = 1024 * 1000000 / 44100;
+ }
+ if (!dev->u.alsa.out->has_buffer_length) {
+ /* 4096 frames assuming 44100Hz */
+ dev->u.alsa.out->buffer_length = 4096ll * 1000000 / 44100;
+ }
+
+ /*
+ * OptsVisitor sets unspecified optional fields to zero, but do not depend
+ * on it...
+ */
+ if (!dev->u.alsa.in->has_period_length) {
+ dev->u.alsa.in->period_length = 0;
+ }
+ if (!dev->u.alsa.in->has_buffer_length) {
+ dev->u.alsa.in->buffer_length = 0;
+ }
+
+ return dev;
}
static void alsa_audio_fini (void *opaque)
{
- g_free(opaque);
}
-static struct audio_option alsa_options[] = {
- {
- .name = "DAC_SIZE_IN_USEC",
- .tag = AUD_OPT_BOOL,
- .valp = &glob_conf.size_in_usec_out,
- .descr = "DAC period/buffer size in microseconds (otherwise in frames)"
- },
- {
- .name = "DAC_PERIOD_SIZE",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.period_size_out,
- .descr = "DAC period size (0 to go with system default)",
- .overriddenp = &glob_conf.period_size_out_overridden
- },
- {
- .name = "DAC_BUFFER_SIZE",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.buffer_size_out,
- .descr = "DAC buffer size (0 to go with system default)",
- .overriddenp = &glob_conf.buffer_size_out_overridden
- },
- {
- .name = "ADC_SIZE_IN_USEC",
- .tag = AUD_OPT_BOOL,
- .valp = &glob_conf.size_in_usec_in,
- .descr =
- "ADC period/buffer size in microseconds (otherwise in frames)"
- },
- {
- .name = "ADC_PERIOD_SIZE",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.period_size_in,
- .descr = "ADC period size (0 to go with system default)",
- .overriddenp = &glob_conf.period_size_in_overridden
- },
- {
- .name = "ADC_BUFFER_SIZE",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.buffer_size_in,
- .descr = "ADC buffer size (0 to go with system default)",
- .overriddenp = &glob_conf.buffer_size_in_overridden
- },
- {
- .name = "THRESHOLD",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.threshold,
- .descr = "(undocumented)"
- },
- {
- .name = "DAC_DEV",
- .tag = AUD_OPT_STR,
- .valp = &glob_conf.pcm_name_out,
- .descr = "DAC device name (for instance dmix)"
- },
- {
- .name = "ADC_DEV",
- .tag = AUD_OPT_STR,
- .valp = &glob_conf.pcm_name_in,
- .descr = "ADC device name"
- },
- { /* End of list */ }
-};
-
static struct audio_pcm_ops alsa_pcm_ops = {
.init_out = alsa_init_out,
.fini_out = alsa_fini_out,
@@ -1217,7 +1085,6 @@ static struct audio_pcm_ops alsa_pcm_ops = {
static struct audio_driver alsa_audio_driver = {
.name = "alsa",
.descr = "ALSA http://www.alsa-project.org",
- .options = alsa_options,
.init = alsa_audio_init,
.fini = alsa_audio_fini,
.pcm_ops = &alsa_pcm_ops,
diff --git a/audio/audio_legacy.c b/audio/audio_legacy.c
index ad3b781add..f58c8b6aee 100644
--- a/audio/audio_legacy.c
+++ b/audio/audio_legacy.c
@@ -62,6 +62,18 @@ static void get_int(const char *env, uint32_t *dst, bool *has_dst)
}
}
+static void get_str(const char *env, char **dst, bool *has_dst)
+{
+ const char *val = getenv(env);
+ if (val) {
+ if (*has_dst) {
+ g_free(*dst);
+ }
+ *dst = g_strdup(val);
+ *has_dst = true;
+ }
+}
+
static void get_fmt(const char *env, AudioFormat *dst, bool *has_dst)
{
const char *val = getenv(env);
@@ -80,8 +92,69 @@ static void get_fmt(const char *env, AudioFormat *dst, bool *has_dst)
}
}
+
+static void get_millis_to_usecs(const char *env, uint32_t *dst, bool *has_dst)
+{
+ const char *val = getenv(env);
+ if (val) {
+ *dst = toui32(val) * 1000;
+ *has_dst = true;
+ }
+}
+
+static uint32_t frames_to_usecs(uint32_t frames,
+ AudiodevPerDirectionOptions *pdo)
+{
+ uint32_t freq = pdo->has_frequency ? pdo->frequency : 44100;
+ return (frames * 1000000 + freq / 2) / freq;
+}
+
/* backend specific functions */
-/* todo */
+/* ALSA */
+static void handle_alsa_per_direction(
+ AudiodevAlsaPerDirectionOptions *apdo, const char *prefix)
+{
+ char buf[64];
+ size_t len = strlen(prefix);
+ bool size_in_usecs = false;
+ bool dummy;
+
+ memcpy(buf, prefix, len);
+ strcpy(buf + len, "TRY_POLL");
+ get_bool(buf, &apdo->try_poll, &apdo->has_try_poll);
+
+ strcpy(buf + len, "DEV");
+ get_str(buf, &apdo->dev, &apdo->has_dev);
+
+ strcpy(buf + len, "SIZE_IN_USEC");
+ get_bool(buf, &size_in_usecs, &dummy);
+
+ strcpy(buf + len, "PERIOD_SIZE");
+ get_int(buf, &apdo->period_length, &apdo->has_period_length);
+ if (apdo->has_period_length && !size_in_usecs) {
+ apdo->period_length = frames_to_usecs(
+ apdo->period_length,
+ qapi_AudiodevAlsaPerDirectionOptions_base(apdo));
+ }
+
+ strcpy(buf + len, "BUFFER_SIZE");
+ get_int(buf, &apdo->buffer_length, &apdo->has_buffer_length);
+ if (apdo->has_buffer_length && !size_in_usecs) {
+ apdo->buffer_length = frames_to_usecs(
+ apdo->buffer_length,
+ qapi_AudiodevAlsaPerDirectionOptions_base(apdo));
+ }
+}
+
+static void handle_alsa(Audiodev *dev)
+{
+ AudiodevAlsaOptions *aopt = &dev->u.alsa;
+ handle_alsa_per_direction(aopt->in, "QEMU_ALSA_ADC_");
+ handle_alsa_per_direction(aopt->out, "QEMU_ALSA_DAC_");
+
+ get_millis_to_usecs("QEMU_ALSA_THRESHOLD",
+ &aopt->threshold, &aopt->has_threshold);
+}
/* general */
static void handle_per_direction(
@@ -123,6 +196,15 @@ static AudiodevListEntry *legacy_opt(const char *drvname)
get_int("QEMU_AUDIO_TIMER_PERIOD",
&e->dev->timer_period, &e->dev->has_timer_period);
+ switch (e->dev->driver) {
+ case AUDIODEV_DRIVER_ALSA:
+ handle_alsa(e->dev);
+ break;
+
+ default:
+ break;
+ }
+
return e;
}
--
2.20.1
^ permalink raw reply related [flat|nested] 18+ messages in thread* [Qemu-devel] [PATCH v6 06/14] coreaudio: port to -audiodev config
2019-03-08 22:34 [Qemu-devel] [PATCH v6 00/14] Audio patches Kővágó, Zoltán
` (4 preceding siblings ...)
2019-03-08 22:34 ` [Qemu-devel] [PATCH v6 05/14] alsaaudio: port to -audiodev config Kővágó, Zoltán
@ 2019-03-08 22:34 ` Kővágó, Zoltán
2019-03-08 22:34 ` [Qemu-devel] [PATCH v6 07/14] dsoundaudio: " Kővágó, Zoltán
` (8 subsequent siblings)
14 siblings, 0 replies; 18+ messages in thread
From: Kővágó, Zoltán @ 2019-03-08 22:34 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio_legacy.c | 28 +++++++++++++++++++++++++
audio/coreaudio.c | 49 ++++++++++----------------------------------
2 files changed, 39 insertions(+), 38 deletions(-)
diff --git a/audio/audio_legacy.c b/audio/audio_legacy.c
index f58c8b6aee..5f557a2de9 100644
--- a/audio/audio_legacy.c
+++ b/audio/audio_legacy.c
@@ -109,6 +109,17 @@ static uint32_t frames_to_usecs(uint32_t frames,
return (frames * 1000000 + freq / 2) / freq;
}
+
+static void get_frames_to_usecs(const char *env, uint32_t *dst, bool *has_dst,
+ AudiodevPerDirectionOptions *pdo)
+{
+ const char *val = getenv(env);
+ if (val) {
+ *dst = frames_to_usecs(toui32(val), pdo);
+ *has_dst = true;
+ }
+}
+
/* backend specific functions */
/* ALSA */
static void handle_alsa_per_direction(
@@ -156,6 +167,19 @@ static void handle_alsa(Audiodev *dev)
&aopt->threshold, &aopt->has_threshold);
}
+/* coreaudio */
+static void handle_coreaudio(Audiodev *dev)
+{
+ get_frames_to_usecs(
+ "QEMU_COREAUDIO_BUFFER_SIZE",
+ &dev->u.coreaudio.out->buffer_length,
+ &dev->u.coreaudio.out->has_buffer_length,
+ qapi_AudiodevCoreaudioPerDirectionOptions_base(dev->u.coreaudio.out));
+ get_int("QEMU_COREAUDIO_BUFFER_COUNT",
+ &dev->u.coreaudio.out->buffer_count,
+ &dev->u.coreaudio.out->has_buffer_count);
+}
+
/* general */
static void handle_per_direction(
AudiodevPerDirectionOptions *pdo, const char *prefix)
@@ -201,6 +225,10 @@ static AudiodevListEntry *legacy_opt(const char *drvname)
handle_alsa(e->dev);
break;
+ case AUDIODEV_DRIVER_COREAUDIO:
+ handle_coreaudio(e->dev);
+ break;
+
default:
break;
}
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
index 7d4225dbee..1ee43b7d5f 100644
--- a/audio/coreaudio.c
+++ b/audio/coreaudio.c
@@ -36,11 +36,6 @@
#define MAC_OS_X_VERSION_10_6 1060
#endif
-typedef struct {
- int buffer_frames;
- int nbuffers;
-} CoreaudioConf;
-
typedef struct coreaudioVoiceOut {
HWVoiceOut hw;
pthread_mutex_t mutex;
@@ -507,7 +502,9 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
int err;
const char *typ = "playback";
AudioValueRange frameRange;
- CoreaudioConf *conf = drv_opaque;
+ Audiodev *dev = drv_opaque;
+ AudiodevCoreaudioPerDirectionOptions *cpdo = dev->u.coreaudio.out;
+ int frames;
/* create mutex */
err = pthread_mutex_init(&core->mutex, NULL);
@@ -538,16 +535,17 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
return -1;
}
- if (frameRange.mMinimum > conf->buffer_frames) {
+ frames = audio_buffer_frames(
+ qapi_AudiodevCoreaudioPerDirectionOptions_base(cpdo), as, 11610);
+ if (frameRange.mMinimum > frames) {
core->audioDevicePropertyBufferFrameSize = (UInt32) frameRange.mMinimum;
dolog ("warning: Upsizing Buffer Frames to %f\n", frameRange.mMinimum);
- }
- else if (frameRange.mMaximum < conf->buffer_frames) {
+ } else if (frameRange.mMaximum < frames) {
core->audioDevicePropertyBufferFrameSize = (UInt32) frameRange.mMaximum;
dolog ("warning: Downsizing Buffer Frames to %f\n", frameRange.mMaximum);
}
else {
- core->audioDevicePropertyBufferFrameSize = conf->buffer_frames;
+ core->audioDevicePropertyBufferFrameSize = frames;
}
/* set Buffer Frame Size */
@@ -568,7 +566,8 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as,
"Could not get device buffer frame size\n");
return -1;
}
- hw->samples = conf->nbuffers * core->audioDevicePropertyBufferFrameSize;
+ hw->samples = (cpdo->has_buffer_count ? cpdo->buffer_count : 4) *
+ core->audioDevicePropertyBufferFrameSize;
/* get StreamFormat */
status = coreaudio_get_streamformat(core->outputDeviceID,
@@ -680,40 +679,15 @@ static int coreaudio_ctl_out (HWVoiceOut *hw, int cmd, ...)
return 0;
}
-static CoreaudioConf glob_conf = {
- .buffer_frames = 512,
- .nbuffers = 4,
-};
-
static void *coreaudio_audio_init(Audiodev *dev)
{
- CoreaudioConf *conf = g_malloc(sizeof(CoreaudioConf));
- *conf = glob_conf;
-
- return conf;
+ return dev;
}
static void coreaudio_audio_fini (void *opaque)
{
- g_free(opaque);
}
-static struct audio_option coreaudio_options[] = {
- {
- .name = "BUFFER_SIZE",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.buffer_frames,
- .descr = "Size of the buffer in frames"
- },
- {
- .name = "BUFFER_COUNT",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.nbuffers,
- .descr = "Number of buffers"
- },
- { /* End of list */ }
-};
-
static struct audio_pcm_ops coreaudio_pcm_ops = {
.init_out = coreaudio_init_out,
.fini_out = coreaudio_fini_out,
@@ -725,7 +699,6 @@ static struct audio_pcm_ops coreaudio_pcm_ops = {
static struct audio_driver coreaudio_audio_driver = {
.name = "coreaudio",
.descr = "CoreAudio http://developer.apple.com/audio/coreaudio.html",
- .options = coreaudio_options,
.init = coreaudio_audio_init,
.fini = coreaudio_audio_fini,
.pcm_ops = &coreaudio_pcm_ops,
--
2.20.1
^ permalink raw reply related [flat|nested] 18+ messages in thread* [Qemu-devel] [PATCH v6 07/14] dsoundaudio: port to -audiodev config
2019-03-08 22:34 [Qemu-devel] [PATCH v6 00/14] Audio patches Kővágó, Zoltán
` (5 preceding siblings ...)
2019-03-08 22:34 ` [Qemu-devel] [PATCH v6 06/14] coreaudio: " Kővágó, Zoltán
@ 2019-03-08 22:34 ` Kővágó, Zoltán
2019-03-08 22:34 ` [Qemu-devel] [PATCH v6 08/14] noaudio: " Kővágó, Zoltán
` (7 subsequent siblings)
14 siblings, 0 replies; 18+ messages in thread
From: Kővágó, Zoltán @ 2019-03-08 22:34 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/dsound_template.h | 6 ++---
audio/audio_legacy.c | 43 ++++++++++++++++++++++++++++++
audio/dsoundaudio.c | 59 ++++++++++++-----------------------------
3 files changed, 63 insertions(+), 45 deletions(-)
diff --git a/audio/dsound_template.h b/audio/dsound_template.h
index b439f33f58..8ece870c9e 100644
--- a/audio/dsound_template.h
+++ b/audio/dsound_template.h
@@ -167,17 +167,18 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
dsound *s = drv_opaque;
WAVEFORMATEX wfx;
struct audsettings obt_as;
- DSoundConf *conf = &s->conf;
#ifdef DSBTYPE_IN
const char *typ = "ADC";
DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;
DSCBUFFERDESC bd;
DSCBCAPS bc;
+ AudiodevPerDirectionOptions *pdo = s->dev->u.dsound.in;
#else
const char *typ = "DAC";
DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
DSBUFFERDESC bd;
DSBCAPS bc;
+ AudiodevPerDirectionOptions *pdo = s->dev->u.dsound.out;
#endif
if (!s->FIELD2) {
@@ -193,8 +194,8 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
memset (&bd, 0, sizeof (bd));
bd.dwSize = sizeof (bd);
bd.lpwfxFormat = &wfx;
+ bd.dwBufferBytes = audio_buffer_bytes(pdo, as, 92880);
#ifdef DSBTYPE_IN
- bd.dwBufferBytes = conf->bufsize_in;
hr = IDirectSoundCapture_CreateCaptureBuffer (
s->dsound_capture,
&bd,
@@ -203,7 +204,6 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as,
);
#else
bd.dwFlags = DSBCAPS_STICKYFOCUS | DSBCAPS_GETCURRENTPOSITION2;
- bd.dwBufferBytes = conf->bufsize_out;
hr = IDirectSound_CreateSoundBuffer (
s->dsound,
&bd,
diff --git a/audio/audio_legacy.c b/audio/audio_legacy.c
index 5f557a2de9..bf326bd360 100644
--- a/audio/audio_legacy.c
+++ b/audio/audio_legacy.c
@@ -120,6 +120,30 @@ static void get_frames_to_usecs(const char *env, uint32_t *dst, bool *has_dst,
}
}
+static uint32_t samples_to_usecs(uint32_t samples,
+ AudiodevPerDirectionOptions *pdo)
+{
+ uint32_t channels = pdo->has_channels ? pdo->channels : 2;
+ return frames_to_usecs(samples / channels, pdo);
+}
+
+static uint32_t bytes_to_usecs(uint32_t bytes, AudiodevPerDirectionOptions *pdo)
+{
+ AudioFormat fmt = pdo->has_format ? pdo->format : AUDIO_FORMAT_S16;
+ uint32_t bytes_per_sample = audioformat_bytes_per_sample(fmt);
+ return samples_to_usecs(bytes / bytes_per_sample, pdo);
+}
+
+static void get_bytes_to_usecs(const char *env, uint32_t *dst, bool *has_dst,
+ AudiodevPerDirectionOptions *pdo)
+{
+ const char *val = getenv(env);
+ if (val) {
+ *dst = bytes_to_usecs(toui32(val), pdo);
+ *has_dst = true;
+ }
+}
+
/* backend specific functions */
/* ALSA */
static void handle_alsa_per_direction(
@@ -180,6 +204,21 @@ static void handle_coreaudio(Audiodev *dev)
&dev->u.coreaudio.out->has_buffer_count);
}
+/* dsound */
+static void handle_dsound(Audiodev *dev)
+{
+ get_millis_to_usecs("QEMU_DSOUND_LATENCY_MILLIS",
+ &dev->u.dsound.latency, &dev->u.dsound.has_latency);
+ get_bytes_to_usecs("QEMU_DSOUND_BUFSIZE_OUT",
+ &dev->u.dsound.out->buffer_length,
+ &dev->u.dsound.out->has_buffer_length,
+ dev->u.dsound.out);
+ get_bytes_to_usecs("QEMU_DSOUND_BUFSIZE_IN",
+ &dev->u.dsound.in->buffer_length,
+ &dev->u.dsound.in->has_buffer_length,
+ dev->u.dsound.in);
+}
+
/* general */
static void handle_per_direction(
AudiodevPerDirectionOptions *pdo, const char *prefix)
@@ -229,6 +268,10 @@ static AudiodevListEntry *legacy_opt(const char *drvname)
handle_coreaudio(e->dev);
break;
+ case AUDIODEV_DRIVER_DSOUND:
+ handle_dsound(e->dev);
+ break;
+
default:
break;
}
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
index 02fe777cba..a7d04b5033 100644
--- a/audio/dsoundaudio.c
+++ b/audio/dsoundaudio.c
@@ -32,6 +32,7 @@
#define AUDIO_CAP "dsound"
#include "audio_int.h"
+#include "qemu/host-utils.h"
#include <windows.h>
#include <mmsystem.h>
@@ -42,17 +43,11 @@
/* #define DEBUG_DSOUND */
-typedef struct {
- int bufsize_in;
- int bufsize_out;
- int latency_millis;
-} DSoundConf;
-
typedef struct {
LPDIRECTSOUND dsound;
LPDIRECTSOUNDCAPTURE dsound_capture;
struct audsettings settings;
- DSoundConf conf;
+ Audiodev *dev;
} dsound;
typedef struct {
@@ -248,9 +243,9 @@ static void GCC_FMT_ATTR (3, 4) dsound_logerr2 (
dsound_log_hresult (hr);
}
-static DWORD millis_to_bytes (struct audio_pcm_info *info, DWORD millis)
+static uint64_t usecs_to_bytes(struct audio_pcm_info *info, uint32_t usecs)
{
- return (millis * info->bytes_per_second) / 1000;
+ return muldiv64(usecs, info->bytes_per_second, 1000000);
}
#ifdef DEBUG_DSOUND
@@ -478,7 +473,7 @@ static int dsound_run_out (HWVoiceOut *hw, int live)
LPVOID p1, p2;
int bufsize;
dsound *s = ds->s;
- DSoundConf *conf = &s->conf;
+ AudiodevDsoundOptions *dso = &s->dev->u.dsound;
if (!dsb) {
dolog ("Attempt to run empty with playback buffer\n");
@@ -501,14 +496,14 @@ static int dsound_run_out (HWVoiceOut *hw, int live)
len = live << hwshift;
if (ds->first_time) {
- if (conf->latency_millis) {
+ if (dso->latency) {
DWORD cur_blat;
cur_blat = audio_ring_dist (wpos, ppos, bufsize);
ds->first_time = 0;
old_pos = wpos;
old_pos +=
- millis_to_bytes (&hw->info, conf->latency_millis) - cur_blat;
+ usecs_to_bytes(&hw->info, dso->latency) - cur_blat;
old_pos %= bufsize;
old_pos &= ~hw->info.align;
}
@@ -747,12 +742,6 @@ static int dsound_run_in (HWVoiceIn *hw)
return decr;
}
-static DSoundConf glob_conf = {
- .bufsize_in = 16384,
- .bufsize_out = 16384,
- .latency_millis = 10
-};
-
static void dsound_audio_fini (void *opaque)
{
HRESULT hr;
@@ -788,8 +777,17 @@ static void *dsound_audio_init(Audiodev *dev)
int err;
HRESULT hr;
dsound *s = g_malloc0(sizeof(dsound));
+ AudiodevDsoundOptions *dso;
+
+ assert(dev->driver == AUDIODEV_DRIVER_DSOUND);
+ s->dev = dev;
+ dso = &dev->u.dsound;
+
+ if (!dso->has_latency) {
+ dso->has_latency = true;
+ dso->latency = 10000; /* 10 ms */
+ }
- s->conf = glob_conf;
hr = CoInitialize (NULL);
if (FAILED (hr)) {
dsound_logerr (hr, "Could not initialize COM\n");
@@ -854,28 +852,6 @@ static void *dsound_audio_init(Audiodev *dev)
return s;
}
-static struct audio_option dsound_options[] = {
- {
- .name = "LATENCY_MILLIS",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.latency_millis,
- .descr = "(undocumented)"
- },
- {
- .name = "BUFSIZE_OUT",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.bufsize_out,
- .descr = "(undocumented)"
- },
- {
- .name = "BUFSIZE_IN",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.bufsize_in,
- .descr = "(undocumented)"
- },
- { /* End of list */ }
-};
-
static struct audio_pcm_ops dsound_pcm_ops = {
.init_out = dsound_init_out,
.fini_out = dsound_fini_out,
@@ -893,7 +869,6 @@ static struct audio_pcm_ops dsound_pcm_ops = {
static struct audio_driver dsound_audio_driver = {
.name = "dsound",
.descr = "DirectSound http://wikipedia.org/wiki/DirectSound",
- .options = dsound_options,
.init = dsound_audio_init,
.fini = dsound_audio_fini,
.pcm_ops = &dsound_pcm_ops,
--
2.20.1
^ permalink raw reply related [flat|nested] 18+ messages in thread* [Qemu-devel] [PATCH v6 08/14] noaudio: port to -audiodev config
2019-03-08 22:34 [Qemu-devel] [PATCH v6 00/14] Audio patches Kővágó, Zoltán
` (6 preceding siblings ...)
2019-03-08 22:34 ` [Qemu-devel] [PATCH v6 07/14] dsoundaudio: " Kővágó, Zoltán
@ 2019-03-08 22:34 ` Kővágó, Zoltán
2019-03-08 22:34 ` [Qemu-devel] [PATCH v6 09/14] ossaudio: " Kővágó, Zoltán
` (6 subsequent siblings)
14 siblings, 0 replies; 18+ messages in thread
From: Kővágó, Zoltán @ 2019-03-08 22:34 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/noaudio.c | 1 -
1 file changed, 1 deletion(-)
diff --git a/audio/noaudio.c b/audio/noaudio.c
index 79690af1ea..ccc611fc84 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -163,7 +163,6 @@ static struct audio_pcm_ops no_pcm_ops = {
static struct audio_driver no_audio_driver = {
.name = "none",
.descr = "Timer based audio emulation",
- .options = NULL,
.init = no_audio_init,
.fini = no_audio_fini,
.pcm_ops = &no_pcm_ops,
--
2.20.1
^ permalink raw reply related [flat|nested] 18+ messages in thread* [Qemu-devel] [PATCH v6 09/14] ossaudio: port to -audiodev config
2019-03-08 22:34 [Qemu-devel] [PATCH v6 00/14] Audio patches Kővágó, Zoltán
` (7 preceding siblings ...)
2019-03-08 22:34 ` [Qemu-devel] [PATCH v6 08/14] noaudio: " Kővágó, Zoltán
@ 2019-03-08 22:34 ` Kővágó, Zoltán
2019-03-08 22:34 ` [Qemu-devel] [PATCH v6 10/14] paaudio: " Kővágó, Zoltán
` (5 subsequent siblings)
14 siblings, 0 replies; 18+ messages in thread
From: Kővágó, Zoltán @ 2019-03-08 22:34 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio_legacy.c | 32 +++++++++
audio/ossaudio.c | 161 ++++++++++++++-----------------------------
2 files changed, 83 insertions(+), 110 deletions(-)
diff --git a/audio/audio_legacy.c b/audio/audio_legacy.c
index bf326bd360..94f95bbc73 100644
--- a/audio/audio_legacy.c
+++ b/audio/audio_legacy.c
@@ -219,6 +219,34 @@ static void handle_dsound(Audiodev *dev)
dev->u.dsound.in);
}
+/* OSS */
+static void handle_oss_per_direction(
+ AudiodevOssPerDirectionOptions *opdo, const char *try_poll_env,
+ const char *dev_env)
+{
+ get_bool(try_poll_env, &opdo->try_poll, &opdo->has_try_poll);
+ get_str(dev_env, &opdo->dev, &opdo->has_dev);
+
+ get_bytes_to_usecs("QEMU_OSS_FRAGSIZE",
+ &opdo->buffer_length, &opdo->has_buffer_length,
+ qapi_AudiodevOssPerDirectionOptions_base(opdo));
+ get_int("QEMU_OSS_NFRAGS", &opdo->buffer_count,
+ &opdo->has_buffer_count);
+}
+
+static void handle_oss(Audiodev *dev)
+{
+ AudiodevOssOptions *oopt = &dev->u.oss;
+ handle_oss_per_direction(oopt->in, "QEMU_AUDIO_ADC_TRY_POLL",
+ "QEMU_OSS_ADC_DEV");
+ handle_oss_per_direction(oopt->out, "QEMU_AUDIO_DAC_TRY_POLL",
+ "QEMU_OSS_DAC_DEV");
+
+ get_bool("QEMU_OSS_MMAP", &oopt->try_mmap, &oopt->has_try_mmap);
+ get_bool("QEMU_OSS_EXCLUSIVE", &oopt->exclusive, &oopt->has_exclusive);
+ get_int("QEMU_OSS_POLICY", &oopt->dsp_policy, &oopt->has_dsp_policy);
+}
+
/* general */
static void handle_per_direction(
AudiodevPerDirectionOptions *pdo, const char *prefix)
@@ -272,6 +300,10 @@ static AudiodevListEntry *legacy_opt(const char *drvname)
handle_dsound(e->dev);
break;
+ case AUDIODEV_DRIVER_OSS:
+ handle_oss(e->dev);
+ break;
+
default:
break;
}
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index e0cadbef29..fc28981a39 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -37,16 +37,6 @@
#define USE_DSP_POLICY
#endif
-typedef struct OSSConf {
- int try_mmap;
- int nfrags;
- int fragsize;
- const char *devpath_out;
- const char *devpath_in;
- int exclusive;
- int policy;
-} OSSConf;
-
typedef struct OSSVoiceOut {
HWVoiceOut hw;
void *pcm_buf;
@@ -56,7 +46,7 @@ typedef struct OSSVoiceOut {
int fragsize;
int mmapped;
int pending;
- OSSConf *conf;
+ Audiodev *dev;
} OSSVoiceOut;
typedef struct OSSVoiceIn {
@@ -65,12 +55,12 @@ typedef struct OSSVoiceIn {
int fd;
int nfrags;
int fragsize;
- OSSConf *conf;
+ Audiodev *dev;
} OSSVoiceIn;
struct oss_params {
int freq;
- AudioFormat fmt;
+ int fmt;
int nchannels;
int nfrags;
int fragsize;
@@ -262,19 +252,25 @@ static int oss_get_version (int fd, int *version, const char *typ)
}
#endif
-static int oss_open (int in, struct oss_params *req,
- struct oss_params *obt, int *pfd, OSSConf* conf)
+static int oss_open(int in, struct oss_params *req, audsettings *as,
+ struct oss_params *obt, int *pfd, Audiodev *dev)
{
+ AudiodevOssOptions *oopts = &dev->u.oss;
+ AudiodevOssPerDirectionOptions *opdo = in ? oopts->in : oopts->out;
int fd;
- int oflags = conf->exclusive ? O_EXCL : 0;
+ int oflags = (oopts->has_exclusive && oopts->exclusive) ? O_EXCL : 0;
audio_buf_info abinfo;
int fmt, freq, nchannels;
int setfragment = 1;
- const char *dspname = in ? conf->devpath_in : conf->devpath_out;
+ const char *dspname = opdo->has_dev ? opdo->dev : "/dev/dsp";
const char *typ = in ? "ADC" : "DAC";
+#ifdef USE_DSP_POLICY
+ int policy = oopts->has_dsp_policy ? oopts->dsp_policy : 5;
+#endif
/* Kludge needed to have working mmap on Linux */
- oflags |= conf->try_mmap ? O_RDWR : (in ? O_RDONLY : O_WRONLY);
+ oflags |= (oopts->has_try_mmap && oopts->try_mmap) ?
+ O_RDWR : (in ? O_RDONLY : O_WRONLY);
fd = open (dspname, oflags | O_NONBLOCK);
if (-1 == fd) {
@@ -285,6 +281,9 @@ static int oss_open (int in, struct oss_params *req,
freq = req->freq;
nchannels = req->nchannels;
fmt = req->fmt;
+ req->nfrags = opdo->has_buffer_count ? opdo->buffer_count : 4;
+ req->fragsize = audio_buffer_bytes(
+ qapi_AudiodevOssPerDirectionOptions_base(opdo), as, 23220);
if (ioctl (fd, SNDCTL_DSP_SAMPLESIZE, &fmt)) {
oss_logerr2 (errno, typ, "Failed to set sample size %d\n", req->fmt);
@@ -308,18 +307,18 @@ static int oss_open (int in, struct oss_params *req,
}
#ifdef USE_DSP_POLICY
- if (conf->policy >= 0) {
+ if (policy >= 0) {
int version;
if (!oss_get_version (fd, &version, typ)) {
trace_oss_version(version);
if (version >= 0x040000) {
- int policy = conf->policy;
- if (ioctl (fd, SNDCTL_DSP_POLICY, &policy)) {
+ int policy2 = policy;
+ if (ioctl(fd, SNDCTL_DSP_POLICY, &policy2)) {
oss_logerr2 (errno, typ,
"Failed to set timing policy to %d\n",
- conf->policy);
+ policy);
goto err;
}
setfragment = 0;
@@ -502,17 +501,16 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
int fd;
AudioFormat effective_fmt;
struct audsettings obt_as;
- OSSConf *conf = drv_opaque;
+ Audiodev *dev = drv_opaque;
+ AudiodevOssOptions *oopts = &dev->u.oss;
oss->fd = -1;
req.fmt = aud_to_ossfmt (as->fmt, as->endianness);
req.freq = as->freq;
req.nchannels = as->nchannels;
- req.fragsize = conf->fragsize;
- req.nfrags = conf->nfrags;
- if (oss_open (0, &req, &obt, &fd, conf)) {
+ if (oss_open(0, &req, as, &obt, &fd, dev)) {
return -1;
}
@@ -539,7 +537,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
hw->samples = (obt.nfrags * obt.fragsize) >> hw->info.shift;
oss->mmapped = 0;
- if (conf->try_mmap) {
+ if (oopts->has_try_mmap && oopts->try_mmap) {
oss->pcm_buf = mmap (
NULL,
hw->samples << hw->info.shift,
@@ -597,7 +595,7 @@ static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
}
oss->fd = fd;
- oss->conf = conf;
+ oss->dev = dev;
return 0;
}
@@ -605,16 +603,12 @@ static int oss_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
int trig;
OSSVoiceOut *oss = (OSSVoiceOut *) hw;
+ AudiodevOssPerDirectionOptions *opdo = oss->dev->u.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->try_poll;
ldebug ("enabling voice\n");
if (poll_mode) {
@@ -669,16 +663,14 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
int fd;
AudioFormat effective_fmt;
struct audsettings obt_as;
- OSSConf *conf = drv_opaque;
+ Audiodev *dev = drv_opaque;
oss->fd = -1;
req.fmt = aud_to_ossfmt (as->fmt, as->endianness);
req.freq = as->freq;
req.nchannels = as->nchannels;
- req.fragsize = conf->fragsize;
- req.nfrags = conf->nfrags;
- if (oss_open (1, &req, &obt, &fd, conf)) {
+ if (oss_open(1, &req, as, &obt, &fd, dev)) {
return -1;
}
@@ -712,7 +704,7 @@ static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
}
oss->fd = fd;
- oss->conf = conf;
+ oss->dev = dev;
return 0;
}
@@ -803,16 +795,12 @@ static int oss_read (SWVoiceIn *sw, void *buf, int size)
static int oss_ctl_in (HWVoiceIn *hw, int cmd, ...)
{
OSSVoiceIn *oss = (OSSVoiceIn *) hw;
+ AudiodevOssPerDirectionOptions *opdo = oss->dev->u.oss.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->try_poll;
if (poll_mode) {
oss_poll_in (hw);
@@ -832,82 +820,36 @@ 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_init_per_direction(AudiodevOssPerDirectionOptions *opdo)
+{
+ if (!opdo->has_try_poll) {
+ opdo->try_poll = true;
+ opdo->has_try_poll = true;
+ }
+}
static void *oss_audio_init(Audiodev *dev)
{
- OSSConf *conf = g_malloc(sizeof(OSSConf));
- *conf = glob_conf;
+ AudiodevOssOptions *oopts;
+ assert(dev->driver == AUDIODEV_DRIVER_OSS);
- if (access(conf->devpath_in, R_OK | W_OK) < 0 ||
- access(conf->devpath_out, R_OK | W_OK) < 0) {
- g_free(conf);
+ oopts = &dev->u.oss;
+ oss_init_per_direction(oopts->in);
+ oss_init_per_direction(oopts->out);
+
+ if (access(oopts->in->has_dev ? oopts->in->dev : "/dev/dsp",
+ R_OK | W_OK) < 0 ||
+ access(oopts->out->has_dev ? oopts->out->dev : "/dev/dsp",
+ R_OK | W_OK) < 0) {
return NULL;
}
- return conf;
+ return dev;
}
static void oss_audio_fini (void *opaque)
{
- g_free(opaque);
}
-static struct audio_option oss_options[] = {
- {
- .name = "FRAGSIZE",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.fragsize,
- .descr = "Fragment size in bytes"
- },
- {
- .name = "NFRAGS",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.nfrags,
- .descr = "Number of fragments"
- },
- {
- .name = "MMAP",
- .tag = AUD_OPT_BOOL,
- .valp = &glob_conf.try_mmap,
- .descr = "Try using memory mapped access"
- },
- {
- .name = "DAC_DEV",
- .tag = AUD_OPT_STR,
- .valp = &glob_conf.devpath_out,
- .descr = "Path to DAC device"
- },
- {
- .name = "ADC_DEV",
- .tag = AUD_OPT_STR,
- .valp = &glob_conf.devpath_in,
- .descr = "Path to ADC device"
- },
- {
- .name = "EXCLUSIVE",
- .tag = AUD_OPT_BOOL,
- .valp = &glob_conf.exclusive,
- .descr = "Open device in exclusive mode (vmix won't work)"
- },
-#ifdef USE_DSP_POLICY
- {
- .name = "POLICY",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.policy,
- .descr = "Set the timing policy of the device, -1 to use fragment mode",
- },
-#endif
- { /* End of list */ }
-};
-
static struct audio_pcm_ops oss_pcm_ops = {
.init_out = oss_init_out,
.fini_out = oss_fini_out,
@@ -925,7 +867,6 @@ static struct audio_pcm_ops oss_pcm_ops = {
static struct audio_driver oss_audio_driver = {
.name = "oss",
.descr = "OSS http://www.opensound.com",
- .options = oss_options,
.init = oss_audio_init,
.fini = oss_audio_fini,
.pcm_ops = &oss_pcm_ops,
--
2.20.1
^ permalink raw reply related [flat|nested] 18+ messages in thread* [Qemu-devel] [PATCH v6 10/14] paaudio: port to -audiodev config
2019-03-08 22:34 [Qemu-devel] [PATCH v6 00/14] Audio patches Kővágó, Zoltán
` (8 preceding siblings ...)
2019-03-08 22:34 ` [Qemu-devel] [PATCH v6 09/14] ossaudio: " Kővágó, Zoltán
@ 2019-03-08 22:34 ` Kővágó, Zoltán
2019-03-08 22:34 ` [Qemu-devel] [PATCH v6 11/14] sdlaudio: " Kővágó, Zoltán
` (4 subsequent siblings)
14 siblings, 0 replies; 18+ messages in thread
From: Kővágó, Zoltán @ 2019-03-08 22:34 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio_legacy.c | 38 +++++++++++++++++++++
audio/paaudio.c | 81 ++++++++++++++++----------------------------
2 files changed, 67 insertions(+), 52 deletions(-)
diff --git a/audio/audio_legacy.c b/audio/audio_legacy.c
index 94f95bbc73..8d7f1475b5 100644
--- a/audio/audio_legacy.c
+++ b/audio/audio_legacy.c
@@ -127,6 +127,16 @@ static uint32_t samples_to_usecs(uint32_t samples,
return frames_to_usecs(samples / channels, pdo);
}
+static void get_samples_to_usecs(const char *env, uint32_t *dst, bool *has_dst,
+ AudiodevPerDirectionOptions *pdo)
+{
+ const char *val = getenv(env);
+ if (val) {
+ *dst = samples_to_usecs(toui32(val), pdo);
+ *has_dst = true;
+ }
+}
+
static uint32_t bytes_to_usecs(uint32_t bytes, AudiodevPerDirectionOptions *pdo)
{
AudioFormat fmt = pdo->has_format ? pdo->format : AUDIO_FORMAT_S16;
@@ -247,6 +257,30 @@ static void handle_oss(Audiodev *dev)
get_int("QEMU_OSS_POLICY", &oopt->dsp_policy, &oopt->has_dsp_policy);
}
+/* pulseaudio */
+static void handle_pa_per_direction(
+ AudiodevPaPerDirectionOptions *ppdo, const char *env)
+{
+ get_str(env, &ppdo->name, &ppdo->has_name);
+}
+
+static void handle_pa(Audiodev *dev)
+{
+ handle_pa_per_direction(dev->u.pa.in, "QEMU_PA_SOURCE");
+ handle_pa_per_direction(dev->u.pa.out, "QEMU_PA_SINK");
+
+ get_samples_to_usecs(
+ "QEMU_PA_SAMPLES", &dev->u.pa.in->buffer_length,
+ &dev->u.pa.in->has_buffer_length,
+ qapi_AudiodevPaPerDirectionOptions_base(dev->u.pa.in));
+ get_samples_to_usecs(
+ "QEMU_PA_SAMPLES", &dev->u.pa.out->buffer_length,
+ &dev->u.pa.out->has_buffer_length,
+ qapi_AudiodevPaPerDirectionOptions_base(dev->u.pa.out));
+
+ get_str("QEMU_PA_SERVER", &dev->u.pa.server, &dev->u.pa.has_server);
+}
+
/* general */
static void handle_per_direction(
AudiodevPerDirectionOptions *pdo, const char *prefix)
@@ -304,6 +338,10 @@ static AudiodevListEntry *legacy_opt(const char *drvname)
handle_oss(e->dev);
break;
+ case AUDIODEV_DRIVER_PA:
+ handle_pa(e->dev);
+ break;
+
default:
break;
}
diff --git a/audio/paaudio.c b/audio/paaudio.c
index d649c58e3d..5d410ed73f 100644
--- a/audio/paaudio.c
+++ b/audio/paaudio.c
@@ -2,6 +2,7 @@
#include "qemu/osdep.h"
#include "qemu-common.h"
#include "audio.h"
+#include "qapi/opts-visitor.h"
#include <pulse/pulseaudio.h>
@@ -10,14 +11,7 @@
#include "audio_pt_int.h"
typedef struct {
- int samples;
- char *server;
- char *sink;
- char *source;
-} PAConf;
-
-typedef struct {
- PAConf conf;
+ Audiodev *dev;
pa_threaded_mainloop *mainloop;
pa_context *context;
} paaudio;
@@ -32,6 +26,7 @@ typedef struct {
void *pcm_buf;
struct audio_pt pt;
paaudio *g;
+ int samples;
} PAVoiceOut;
typedef struct {
@@ -46,6 +41,7 @@ typedef struct {
const void *read_data;
size_t read_index, read_length;
paaudio *g;
+ int samples;
} PAVoiceIn;
static void qpa_audio_fini(void *opaque);
@@ -227,7 +223,7 @@ static void *qpa_thread_out (void *arg)
}
}
- decr = to_mix = audio_MIN(pa->live, pa->g->conf.samples >> 5);
+ decr = to_mix = audio_MIN(pa->live, pa->samples >> 5);
rpos = pa->rpos;
if (audio_pt_unlock(&pa->pt, __func__)) {
@@ -319,7 +315,7 @@ static void *qpa_thread_in (void *arg)
}
}
- incr = to_grab = audio_MIN(pa->dead, pa->g->conf.samples >> 5);
+ incr = to_grab = audio_MIN(pa->dead, pa->samples >> 5);
wpos = pa->wpos;
if (audio_pt_unlock(&pa->pt, __func__)) {
@@ -546,6 +542,8 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
struct audsettings obt_as = *as;
PAVoiceOut *pa = (PAVoiceOut *) hw;
paaudio *g = pa->g = drv_opaque;
+ AudiodevPaOptions *popts = &g->dev->u.pa;
+ AudiodevPaPerDirectionOptions *ppdo = popts->out;
ss.format = audfmt_to_pa (as->fmt, as->endianness);
ss.channels = as->nchannels;
@@ -566,7 +564,7 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
g,
"qemu",
PA_STREAM_PLAYBACK,
- g->conf.sink,
+ ppdo->has_name ? ppdo->name : NULL,
&ss,
NULL, /* channel map */
&ba, /* buffering attributes */
@@ -578,7 +576,8 @@ static int qpa_init_out(HWVoiceOut *hw, struct audsettings *as,
}
audio_pcm_init_info (&hw->info, &obt_as);
- hw->samples = g->conf.samples;
+ hw->samples = pa->samples = audio_buffer_samples(
+ qapi_AudiodevPaPerDirectionOptions_base(ppdo), &obt_as, 46440);
pa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
pa->rpos = hw->rpos;
if (!pa->pcm_buf) {
@@ -612,6 +611,8 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
struct audsettings obt_as = *as;
PAVoiceIn *pa = (PAVoiceIn *) hw;
paaudio *g = pa->g = drv_opaque;
+ AudiodevPaOptions *popts = &g->dev->u.pa;
+ AudiodevPaPerDirectionOptions *ppdo = popts->in;
ss.format = audfmt_to_pa (as->fmt, as->endianness);
ss.channels = as->nchannels;
@@ -623,7 +624,7 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
g,
"qemu",
PA_STREAM_RECORD,
- g->conf.source,
+ ppdo->has_name ? ppdo->name : NULL,
&ss,
NULL, /* channel map */
NULL, /* buffering attributes */
@@ -635,7 +636,8 @@ static int qpa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
}
audio_pcm_init_info (&hw->info, &obt_as);
- hw->samples = g->conf.samples;
+ hw->samples = pa->samples = audio_buffer_samples(
+ qapi_AudiodevPaPerDirectionOptions_base(ppdo), &obt_as, 46440);
pa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
pa->wpos = hw->wpos;
if (!pa->pcm_buf) {
@@ -808,13 +810,13 @@ static int qpa_ctl_in (HWVoiceIn *hw, int cmd, ...)
}
/* common */
-static PAConf glob_conf = {
- .samples = 4096,
-};
-
static void *qpa_audio_init(Audiodev *dev)
{
- if (glob_conf.server == NULL) {
+ paaudio *g;
+ AudiodevPaOptions *popts = &dev->u.pa;
+ const char *server;
+
+ if (!popts->has_server) {
char pidfile[64];
char *runtime;
struct stat st;
@@ -829,8 +831,12 @@ static void *qpa_audio_init(Audiodev *dev)
}
}
- paaudio *g = g_malloc(sizeof(paaudio));
- g->conf = glob_conf;
+ assert(dev->driver == AUDIODEV_DRIVER_PA);
+
+ g = g_malloc(sizeof(paaudio));
+ server = popts->has_server ? popts->server : NULL;
+
+ g->dev = dev;
g->mainloop = NULL;
g->context = NULL;
@@ -840,14 +846,14 @@ static void *qpa_audio_init(Audiodev *dev)
}
g->context = pa_context_new (pa_threaded_mainloop_get_api (g->mainloop),
- g->conf.server);
+ server);
if (!g->context) {
goto fail;
}
pa_context_set_state_callback (g->context, context_state_cb, g);
- if (pa_context_connect (g->context, g->conf.server, 0, NULL) < 0) {
+ if (pa_context_connect(g->context, server, 0, NULL) < 0) {
qpa_logerr (pa_context_errno (g->context),
"pa_context_connect() failed\n");
goto fail;
@@ -910,34 +916,6 @@ 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,
@@ -955,7 +933,6 @@ static struct audio_pcm_ops qpa_pcm_ops = {
static struct audio_driver pa_audio_driver = {
.name = "pa",
.descr = "http://www.pulseaudio.org/",
- .options = qpa_options,
.init = qpa_audio_init,
.fini = qpa_audio_fini,
.pcm_ops = &qpa_pcm_ops,
--
2.20.1
^ permalink raw reply related [flat|nested] 18+ messages in thread* [Qemu-devel] [PATCH v6 11/14] sdlaudio: port to -audiodev config
2019-03-08 22:34 [Qemu-devel] [PATCH v6 00/14] Audio patches Kővágó, Zoltán
` (9 preceding siblings ...)
2019-03-08 22:34 ` [Qemu-devel] [PATCH v6 10/14] paaudio: " Kővágó, Zoltán
@ 2019-03-08 22:34 ` Kővágó, Zoltán
2019-03-08 22:34 ` [Qemu-devel] [PATCH v6 12/14] spiceaudio: " Kővágó, Zoltán
` (3 subsequent siblings)
14 siblings, 0 replies; 18+ messages in thread
From: Kővágó, Zoltán @ 2019-03-08 22:34 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio_legacy.c | 12 ++++++++++++
audio/sdlaudio.c | 22 ++++------------------
2 files changed, 16 insertions(+), 18 deletions(-)
diff --git a/audio/audio_legacy.c b/audio/audio_legacy.c
index 8d7f1475b5..478cf76d5f 100644
--- a/audio/audio_legacy.c
+++ b/audio/audio_legacy.c
@@ -281,6 +281,14 @@ static void handle_pa(Audiodev *dev)
get_str("QEMU_PA_SERVER", &dev->u.pa.server, &dev->u.pa.has_server);
}
+/* SDL */
+static void handle_sdl(Audiodev *dev)
+{
+ /* SDL is output only */
+ get_samples_to_usecs("QEMU_SDL_SAMPLES", &dev->u.sdl.out->buffer_length,
+ &dev->u.sdl.out->has_buffer_length, dev->u.sdl.out);
+}
+
/* general */
static void handle_per_direction(
AudiodevPerDirectionOptions *pdo, const char *prefix)
@@ -342,6 +350,10 @@ static AudiodevListEntry *legacy_opt(const char *drvname)
handle_pa(e->dev);
break;
+ case AUDIODEV_DRIVER_SDL:
+ handle_sdl(e->dev);
+ break;
+
default:
break;
}
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index b9ae506a56..ff9248ba68 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -44,16 +44,11 @@ typedef struct SDLVoiceOut {
int decr;
} SDLVoiceOut;
-static struct {
- int nb_samples;
-} conf = {
- .nb_samples = 1024
-};
-
static struct SDLAudioState {
int exit;
int initialized;
bool driver_created;
+ Audiodev *dev;
} glob_sdl;
typedef struct SDLAudioState SDLAudioState;
@@ -271,7 +266,7 @@ 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->u.sdl.out, as, 11610);
req.callback = sdl_callback;
req.userdata = sdl;
@@ -329,6 +324,7 @@ static void *sdl_audio_init(Audiodev *dev)
}
s->driver_created = true;
+ s->dev = dev;
return s;
}
@@ -338,18 +334,9 @@ static void sdl_audio_fini (void *opaque)
sdl_close (s);
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,
@@ -361,7 +348,6 @@ static struct audio_pcm_ops sdl_pcm_ops = {
static struct audio_driver sdl_audio_driver = {
.name = "sdl",
.descr = "SDL http://www.libsdl.org",
- .options = sdl_options,
.init = sdl_audio_init,
.fini = sdl_audio_fini,
.pcm_ops = &sdl_pcm_ops,
--
2.20.1
^ permalink raw reply related [flat|nested] 18+ messages in thread* [Qemu-devel] [PATCH v6 12/14] spiceaudio: port to -audiodev config
2019-03-08 22:34 [Qemu-devel] [PATCH v6 00/14] Audio patches Kővágó, Zoltán
` (10 preceding siblings ...)
2019-03-08 22:34 ` [Qemu-devel] [PATCH v6 11/14] sdlaudio: " Kővágó, Zoltán
@ 2019-03-08 22:34 ` Kővágó, Zoltán
2019-03-08 22:34 ` [Qemu-devel] [PATCH v6 13/14] wavaudio: " Kővágó, Zoltán
` (2 subsequent siblings)
14 siblings, 0 replies; 18+ messages in thread
From: Kővágó, Zoltán @ 2019-03-08 22:34 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/spiceaudio.c | 5 -----
1 file changed, 5 deletions(-)
diff --git a/audio/spiceaudio.c b/audio/spiceaudio.c
index affc3df17f..4f7873af5a 100644
--- a/audio/spiceaudio.c
+++ b/audio/spiceaudio.c
@@ -373,10 +373,6 @@ static int line_in_ctl (HWVoiceIn *hw, int cmd, ...)
return 0;
}
-static struct audio_option audio_options[] = {
- { /* end of list */ },
-};
-
static struct audio_pcm_ops audio_callbacks = {
.init_out = line_out_init,
.fini_out = line_out_fini,
@@ -394,7 +390,6 @@ static struct audio_pcm_ops audio_callbacks = {
static struct audio_driver spice_audio_driver = {
.name = "spice",
.descr = "spice audio driver",
- .options = audio_options,
.init = spice_audio_init,
.fini = spice_audio_fini,
.pcm_ops = &audio_callbacks,
--
2.20.1
^ permalink raw reply related [flat|nested] 18+ messages in thread* [Qemu-devel] [PATCH v6 13/14] wavaudio: port to -audiodev config
2019-03-08 22:34 [Qemu-devel] [PATCH v6 00/14] Audio patches Kővágó, Zoltán
` (11 preceding siblings ...)
2019-03-08 22:34 ` [Qemu-devel] [PATCH v6 12/14] spiceaudio: " Kővágó, Zoltán
@ 2019-03-08 22:34 ` Kővágó, Zoltán
2019-03-08 22:34 ` [Qemu-devel] [PATCH v6 14/14] audio: -audiodev command line option: cleanup Kővágó, Zoltán
2019-03-08 22:49 ` [Qemu-devel] [PATCH v6 00/14] Audio patches no-reply
14 siblings, 0 replies; 18+ messages in thread
From: Kővágó, Zoltán @ 2019-03-08 22:34 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio_legacy.c | 16 ++++++++++++
audio/wavaudio.c | 58 +++++++-------------------------------------
2 files changed, 25 insertions(+), 49 deletions(-)
diff --git a/audio/audio_legacy.c b/audio/audio_legacy.c
index 478cf76d5f..6d140119d9 100644
--- a/audio/audio_legacy.c
+++ b/audio/audio_legacy.c
@@ -289,6 +289,18 @@ static void handle_sdl(Audiodev *dev)
&dev->u.sdl.out->has_buffer_length, dev->u.sdl.out);
}
+/* wav */
+static void handle_wav(Audiodev *dev)
+{
+ get_int("QEMU_WAV_FREQUENCY",
+ &dev->u.wav.out->frequency, &dev->u.wav.out->has_frequency);
+ get_fmt("QEMU_WAV_FORMAT", &dev->u.wav.out->format,
+ &dev->u.wav.out->has_format);
+ get_int("QEMU_WAV_DAC_FIXED_CHANNELS",
+ &dev->u.wav.out->channels, &dev->u.wav.out->has_channels);
+ get_str("QEMU_WAV_PATH", &dev->u.wav.path, &dev->u.wav.has_path);
+}
+
/* general */
static void handle_per_direction(
AudiodevPerDirectionOptions *pdo, const char *prefix)
@@ -354,6 +366,10 @@ static AudiodevListEntry *legacy_opt(const char *drvname)
handle_sdl(e->dev);
break;
+ case AUDIODEV_DRIVER_WAV:
+ handle_wav(e->dev);
+ break;
+
default:
break;
}
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 9eff3555b3..8d30f57296 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -24,6 +24,7 @@
#include "qemu/osdep.h"
#include "qemu/host-utils.h"
#include "qemu/timer.h"
+#include "qapi/opts-visitor.h"
#include "audio.h"
#define AUDIO_CAP "wav"
@@ -37,11 +38,6 @@ typedef struct WAVVoiceOut {
int total_samples;
} WAVVoiceOut;
-typedef struct {
- struct audsettings settings;
- const char *wav_path;
-} WAVConf;
-
static int wav_run_out (HWVoiceOut *hw, int live)
{
WAVVoiceOut *wav = (WAVVoiceOut *) hw;
@@ -112,8 +108,10 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
0x02, 0x00, 0x44, 0xac, 0x00, 0x00, 0x10, 0xb1, 0x02, 0x00, 0x04,
0x00, 0x10, 0x00, 0x64, 0x61, 0x74, 0x61, 0x00, 0x00, 0x00, 0x00
};
- WAVConf *conf = drv_opaque;
- struct audsettings wav_as = conf->settings;
+ Audiodev *dev = drv_opaque;
+ AudiodevWavOptions *wopts = &dev->u.wav;
+ struct audsettings wav_as = audiodev_to_audsettings(dev->u.wav.out);
+ const char *wav_path = wopts->has_path ? wopts->path : "qemu.wav";
stereo = wav_as.nchannels == 2;
switch (wav_as.fmt) {
@@ -154,10 +152,10 @@ static int wav_init_out(HWVoiceOut *hw, struct audsettings *as,
le_store (hdr + 28, hw->info.freq << (bits16 + stereo), 4);
le_store (hdr + 32, 1 << (bits16 + stereo), 2);
- wav->f = fopen (conf->wav_path, "wb");
+ wav->f = fopen(wav_path, "wb");
if (!wav->f) {
dolog ("Failed to open wave file `%s'\nReason: %s\n",
- conf->wav_path, strerror (errno));
+ wav_path, strerror(errno));
g_free (wav->pcm_buf);
wav->pcm_buf = NULL;
return -1;
@@ -225,54 +223,17 @@ static int wav_ctl_out (HWVoiceOut *hw, int cmd, ...)
return 0;
}
-static WAVConf glob_conf = {
- .settings.freq = 44100,
- .settings.nchannels = 2,
- .settings.fmt = AUDIO_FORMAT_S16,
- .wav_path = "qemu.wav"
-};
-
static void *wav_audio_init(Audiodev *dev)
{
- WAVConf *conf = g_malloc(sizeof(WAVConf));
- *conf = glob_conf;
- return conf;
+ assert(dev->driver == AUDIODEV_DRIVER_WAV);
+ return dev;
}
static void wav_audio_fini (void *opaque)
{
ldebug ("wav_fini");
- g_free(opaque);
}
-static struct audio_option wav_options[] = {
- {
- .name = "FREQUENCY",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.settings.freq,
- .descr = "Frequency"
- },
- {
- .name = "FORMAT",
- .tag = AUD_OPT_FMT,
- .valp = &glob_conf.settings.fmt,
- .descr = "Format"
- },
- {
- .name = "DAC_FIXED_CHANNELS",
- .tag = AUD_OPT_INT,
- .valp = &glob_conf.settings.nchannels,
- .descr = "Number of channels (1 - mono, 2 - stereo)"
- },
- {
- .name = "PATH",
- .tag = AUD_OPT_STR,
- .valp = &glob_conf.wav_path,
- .descr = "Path to wave file"
- },
- { /* End of list */ }
-};
-
static struct audio_pcm_ops wav_pcm_ops = {
.init_out = wav_init_out,
.fini_out = wav_fini_out,
@@ -284,7 +245,6 @@ static struct audio_pcm_ops wav_pcm_ops = {
static struct audio_driver wav_audio_driver = {
.name = "wav",
.descr = "WAV renderer http://wikipedia.org/wiki/WAV",
- .options = wav_options,
.init = wav_audio_init,
.fini = wav_audio_fini,
.pcm_ops = &wav_pcm_ops,
--
2.20.1
^ permalink raw reply related [flat|nested] 18+ messages in thread* [Qemu-devel] [PATCH v6 14/14] audio: -audiodev command line option: cleanup
2019-03-08 22:34 [Qemu-devel] [PATCH v6 00/14] Audio patches Kővágó, Zoltán
` (12 preceding siblings ...)
2019-03-08 22:34 ` [Qemu-devel] [PATCH v6 13/14] wavaudio: " Kővágó, Zoltán
@ 2019-03-08 22:34 ` Kővágó, Zoltán
2019-03-08 22:49 ` [Qemu-devel] [PATCH v6 00/14] Audio patches no-reply
14 siblings, 0 replies; 18+ messages in thread
From: Kővágó, Zoltán @ 2019-03-08 22:34 UTC (permalink / raw)
To: qemu-devel; +Cc: Gerd Hoffmann
Remove no longer needed code.
Signed-off-by: Kővágó, Zoltán <DirtY.iCE.hu@gmail.com>
---
audio/audio_int.h | 17 -----
audio/audio.c | 186 +---------------------------------------------
2 files changed, 4 insertions(+), 199 deletions(-)
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 7bf5dfc0b5..3f14842709 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -33,22 +33,6 @@
struct audio_pcm_ops;
-typedef enum {
- AUD_OPT_INT,
- AUD_OPT_FMT,
- AUD_OPT_STR,
- AUD_OPT_BOOL
-} audio_option_tag_e;
-
-struct audio_option {
- const char *name;
- audio_option_tag_e tag;
- void *valp;
- const char *descr;
- int *overriddenp;
- int overridden;
-};
-
struct audio_callback {
void *opaque;
audio_callback_fn fn;
@@ -145,7 +129,6 @@ typedef struct audio_driver audio_driver;
struct audio_driver {
const char *name;
const char *descr;
- struct audio_option *options;
void *(*init) (Audiodev *);
void (*fini) (void *);
struct audio_pcm_ops *pcm_ops;
diff --git a/audio/audio.c b/audio/audio.c
index 79ed360666..5fd9a58a80 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -172,113 +172,6 @@ void *audio_calloc (const char *funcname, int nmemb, size_t size)
return g_malloc0 (len);
}
-static const char *audio_audfmt_to_string (AudioFormat fmt)
-{
- switch (fmt) {
- case AUDIO_FORMAT_U8:
- return "U8";
-
- case AUDIO_FORMAT_U16:
- return "U16";
-
- case AUDIO_FORMAT_S8:
- return "S8";
-
- case AUDIO_FORMAT_S16:
- return "S16";
-
- case AUDIO_FORMAT_U32:
- return "U32";
-
- case AUDIO_FORMAT_S32:
- return "S32";
-
- default:
- abort();
- }
-
- dolog ("Bogus audfmt %d returning S16\n", fmt);
- return "S16";
-}
-
-static AudioFormat audio_string_to_audfmt (const char *s, AudioFormat defval,
- int *defaultp)
-{
- if (!strcasecmp (s, "u8")) {
- *defaultp = 0;
- return AUDIO_FORMAT_U8;
- }
- else if (!strcasecmp (s, "u16")) {
- *defaultp = 0;
- return AUDIO_FORMAT_U16;
- }
- else if (!strcasecmp (s, "u32")) {
- *defaultp = 0;
- return AUDIO_FORMAT_U32;
- }
- else if (!strcasecmp (s, "s8")) {
- *defaultp = 0;
- return AUDIO_FORMAT_S8;
- }
- else if (!strcasecmp (s, "s16")) {
- *defaultp = 0;
- return AUDIO_FORMAT_S16;
- }
- else if (!strcasecmp (s, "s32")) {
- *defaultp = 0;
- return AUDIO_FORMAT_S32;
- }
- else {
- dolog ("Bogus audio format `%s' using %s\n",
- s, audio_audfmt_to_string (defval));
- *defaultp = 1;
- return defval;
- }
-}
-
-static AudioFormat audio_get_conf_fmt (const char *envname,
- AudioFormat defval,
- int *defaultp)
-{
- const char *var = getenv (envname);
- if (!var) {
- *defaultp = 1;
- return defval;
- }
- return audio_string_to_audfmt (var, defval, defaultp);
-}
-
-static int audio_get_conf_int (const char *key, int defval, int *defaultp)
-{
- int val;
- char *strval;
-
- strval = getenv (key);
- if (strval && !qemu_strtoi(strval, NULL, 10, &val)) {
- *defaultp = 0;
- return val;
- }
- else {
- *defaultp = 1;
- return defval;
- }
-}
-
-static const char *audio_get_conf_str (const char *key,
- const char *defval,
- int *defaultp)
-{
- const char *val = getenv (key);
- if (!val) {
- *defaultp = 1;
- return defval;
- }
- else {
- *defaultp = 0;
- return val;
- }
-}
-
void AUD_vlog (const char *cap, const char *fmt, va_list ap)
{
if (cap) {
@@ -297,74 +190,6 @@ void AUD_log (const char *cap, const char *fmt, ...)
va_end (ap);
}
-static void audio_process_options (const char *prefix,
- struct audio_option *opt)
-{
- gchar *prefix_upper;
-
- if (audio_bug(__func__, !prefix)) {
- dolog ("prefix = NULL\n");
- return;
- }
-
- if (audio_bug(__func__, !opt)) {
- dolog ("opt = NULL\n");
- return;
- }
-
- prefix_upper = g_utf8_strup(prefix, -1);
-
- for (; opt->name; opt++) {
- char *optname;
- int def;
-
- if (!opt->valp) {
- dolog ("Option value pointer for `%s' is not set\n",
- opt->name);
- continue;
- }
-
- optname = g_strdup_printf("QEMU_%s_%s", prefix_upper, 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);
- }
- g_free(prefix_upper);
-}
-
static void audio_print_settings (struct audsettings *as)
{
dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels);
@@ -1072,7 +897,7 @@ void AUD_set_active_out (SWVoiceOut *sw, int on)
if (!hw->enabled) {
hw->enabled = 1;
if (s->vm_running) {
- hw->pcm_ops->ctl_out(hw, VOICE_ENABLE, true);
+ hw->pcm_ops->ctl_out(hw, VOICE_ENABLE);
audio_reset_timer (s);
}
}
@@ -1117,7 +942,7 @@ void AUD_set_active_in (SWVoiceIn *sw, int on)
if (!hw->enabled) {
hw->enabled = 1;
if (s->vm_running) {
- hw->pcm_ops->ctl_in(hw, VOICE_ENABLE, true);
+ hw->pcm_ops->ctl_in(hw, VOICE_ENABLE);
audio_reset_timer (s);
}
}
@@ -1441,9 +1266,6 @@ void audio_run (const char *msg)
static int audio_driver_init(AudioState *s, struct audio_driver *drv,
bool msg, Audiodev *dev)
{
- if (drv->options) {
- audio_process_options (drv->name, drv->options);
- }
s->drv_opaque = drv->init(dev);
if (s->drv_opaque) {
@@ -1470,11 +1292,11 @@ static void audio_vm_change_state_handler (void *opaque, int running,
s->vm_running = running;
while ((hwo = audio_pcm_hw_find_any_enabled_out (hwo))) {
- hwo->pcm_ops->ctl_out(hwo, op, true);
+ hwo->pcm_ops->ctl_out(hwo, op);
}
while ((hwi = audio_pcm_hw_find_any_enabled_in (hwi))) {
- hwi->pcm_ops->ctl_in(hwi, op, true);
+ hwi->pcm_ops->ctl_in(hwi, op);
}
audio_reset_timer (s);
}
--
2.20.1
^ permalink raw reply related [flat|nested] 18+ messages in thread* Re: [Qemu-devel] [PATCH v6 00/14] Audio patches
2019-03-08 22:34 [Qemu-devel] [PATCH v6 00/14] Audio patches Kővágó, Zoltán
` (13 preceding siblings ...)
2019-03-08 22:34 ` [Qemu-devel] [PATCH v6 14/14] audio: -audiodev command line option: cleanup Kővágó, Zoltán
@ 2019-03-08 22:49 ` no-reply
14 siblings, 0 replies; 18+ messages in thread
From: no-reply @ 2019-03-08 22:49 UTC (permalink / raw)
To: dirty.ice.hu; +Cc: fam, qemu-devel, kraxel
Patchew URL: https://patchew.org/QEMU/cover.1552083282.git.DirtY.iCE.hu@gmail.com/
Hi,
This series seems to have some coding style problems. See output below for
more information:
Type: series
Message-id: cover.1552083282.git.DirtY.iCE.hu@gmail.com
Subject: [Qemu-devel] [PATCH v6 00/14] Audio patches
=== TEST SCRIPT BEGIN ===
#!/bin/bash
git rev-parse base > /dev/null || exit 0
git config --local diff.renamelimit 0
git config --local diff.renames True
git config --local diff.algorithm histogram
./scripts/checkpatch.pl --mailback base..
=== TEST SCRIPT END ===
Updating 3c8cf5a9c21ff8782164d1def7f44bd888713384
From https://github.com/patchew-project/qemu
* [new tag] patchew/cover.1552083282.git.DirtY.iCE.hu@gmail.com -> patchew/cover.1552083282.git.DirtY.iCE.hu@gmail.com
Switched to a new branch 'test'
77a2fc41d3 audio: -audiodev command line option: cleanup
a08295d4a4 wavaudio: port to -audiodev config
2b8db3b1b8 spiceaudio: port to -audiodev config
b13a98214e sdlaudio: port to -audiodev config
f32d4ec566 paaudio: port to -audiodev config
60ef81ebee ossaudio: port to -audiodev config
9361aaae4c noaudio: port to -audiodev config
956b77bf8d dsoundaudio: port to -audiodev config
fd47b30266 coreaudio: port to -audiodev config
b2e94202ed alsaaudio: port to -audiodev config
92e91a2161 audio: -audiodev command line option basic implementation
7597a04f3e audio: -audiodev command line option: documentation
9df02ffc44 audio: use qapi AudioFormat instead of audfmt_e
454e5ebb1c qapi: qapi for audio backends
=== OUTPUT BEGIN ===
1/14 Checking commit 454e5ebb1c11 (qapi: qapi for audio backends)
WARNING: added, moved or deleted file(s), does MAINTAINERS need updating?
#51:
new file mode 100644
total: 0 errors, 1 warnings, 320 lines checked
Patch 1/14 has style problems, please review. If any of these errors
are false positives report them to the maintainer, see
CHECKPATCH in MAINTAINERS.
2/14 Checking commit 9df02ffc443d (audio: use qapi AudioFormat instead of audfmt_e)
ERROR: space prohibited between function name and open parenthesis '('
#36: FILE: audio/alsaaudio.c:297:
+static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness)
ERROR: space prohibited between function name and open parenthesis '('
#84: FILE: audio/alsaaudio.c:347:
+static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt,
ERROR: space prohibited between function name and open parenthesis '('
#205: FILE: audio/audio.c:260:
+static const char *audio_audfmt_to_string (AudioFormat fmt)
ERROR: space prohibited between function name and open parenthesis '('
#241: FILE: audio/audio.c:289:
+static AudioFormat audio_string_to_audfmt (const char *s, AudioFormat defval,
ERROR: space prohibited between function name and open parenthesis '('
#282: FILE: audio/audio.c:324:
+static AudioFormat audio_get_conf_fmt (const char *envname,
ERROR: space prohibited between function name and open parenthesis '('
#523: FILE: audio/ossaudio.c:151:
+static int aud_to_ossfmt (AudioFormat fmt, int endianness)
ERROR: space prohibited between function name and open parenthesis '('
#553: FILE: audio/ossaudio.c:185:
+static int oss_to_audfmt (int ossfmt, AudioFormat *fmt, int *endianness)
ERROR: space prohibited between function name and open parenthesis '('
#620: FILE: audio/paaudio.c:388:
+static pa_sample_format_t audfmt_to_pa (AudioFormat afmt, int endianness)
ERROR: space prohibited between function name and open parenthesis '('
#649: FILE: audio/paaudio.c:413:
+static AudioFormat pa_to_audfmt (pa_sample_format_t fmt, int *endianness)
ERROR: space prohibited between function name and open parenthesis '('
#687: FILE: audio/sdlaudio.c:71:
+static int aud_to_sdlfmt (AudioFormat fmt)
ERROR: trailing statements should be on next line
#954: FILE: hw/audio/hda-codec.c:102:
+ case AC_FMT_BITS_8: as->fmt = AUDIO_FORMAT_S8; break;
ERROR: trailing statements should be on next line
#955: FILE: hw/audio/hda-codec.c:103:
+ case AC_FMT_BITS_16: as->fmt = AUDIO_FORMAT_S16; break;
ERROR: trailing statements should be on next line
#956: FILE: hw/audio/hda-codec.c:104:
+ case AC_FMT_BITS_32: as->fmt = AUDIO_FORMAT_S32; break;
ERROR: space prohibited after that open square bracket '['
#970: FILE: hw/audio/hda-codec.c:137:
+ [ AUDIO_FORMAT_U8 ] = "PCM-U8",
ERROR: space prohibited before that close square bracket ']'
#970: FILE: hw/audio/hda-codec.c:137:
+ [ AUDIO_FORMAT_U8 ] = "PCM-U8",
ERROR: space prohibited after that open square bracket '['
#971: FILE: hw/audio/hda-codec.c:138:
+ [ AUDIO_FORMAT_S8 ] = "PCM-S8",
ERROR: space prohibited before that close square bracket ']'
#971: FILE: hw/audio/hda-codec.c:138:
+ [ AUDIO_FORMAT_S8 ] = "PCM-S8",
ERROR: space prohibited after that open square bracket '['
#972: FILE: hw/audio/hda-codec.c:139:
+ [ AUDIO_FORMAT_U16 ] = "PCM-U16",
ERROR: space prohibited before that close square bracket ']'
#972: FILE: hw/audio/hda-codec.c:139:
+ [ AUDIO_FORMAT_U16 ] = "PCM-U16",
ERROR: space prohibited after that open square bracket '['
#973: FILE: hw/audio/hda-codec.c:140:
+ [ AUDIO_FORMAT_S16 ] = "PCM-S16",
ERROR: space prohibited before that close square bracket ']'
#973: FILE: hw/audio/hda-codec.c:140:
+ [ AUDIO_FORMAT_S16 ] = "PCM-S16",
ERROR: space prohibited after that open square bracket '['
#974: FILE: hw/audio/hda-codec.c:141:
+ [ AUDIO_FORMAT_U32 ] = "PCM-U32",
ERROR: space prohibited before that close square bracket ']'
#974: FILE: hw/audio/hda-codec.c:141:
+ [ AUDIO_FORMAT_U32 ] = "PCM-U32",
ERROR: space prohibited after that open square bracket '['
#975: FILE: hw/audio/hda-codec.c:142:
+ [ AUDIO_FORMAT_S32 ] = "PCM-S32",
ERROR: space prohibited before that close square bracket ']'
#975: FILE: hw/audio/hda-codec.c:142:
+ [ AUDIO_FORMAT_S32 ] = "PCM-S32",
ERROR: trailing statements should be on next line
#1197: FILE: ui/vnc.c:2372:
+ case 0: vs->as.fmt = AUDIO_FORMAT_U8; break;
ERROR: trailing statements should be on next line
#1198: FILE: ui/vnc.c:2373:
+ case 1: vs->as.fmt = AUDIO_FORMAT_S8; break;
ERROR: trailing statements should be on next line
#1199: FILE: ui/vnc.c:2374:
+ case 2: vs->as.fmt = AUDIO_FORMAT_U16; break;
ERROR: trailing statements should be on next line
#1200: FILE: ui/vnc.c:2375:
+ case 3: vs->as.fmt = AUDIO_FORMAT_S16; break;
ERROR: trailing statements should be on next line
#1201: FILE: ui/vnc.c:2376:
+ case 4: vs->as.fmt = AUDIO_FORMAT_U32; break;
ERROR: trailing statements should be on next line
#1202: FILE: ui/vnc.c:2377:
+ case 5: vs->as.fmt = AUDIO_FORMAT_S32; break;
total: 31 errors, 0 warnings, 1028 lines checked
Patch 2/14 has style problems, please review. If any of these errors
are false positives report them to the maintainer, see
CHECKPATCH in MAINTAINERS.
3/14 Checking commit 7597a04f3e28 (audio: -audiodev command line option: documentation)
4/14 Checking commit 92e91a2161ed (audio: -audiodev command line option basic implementation)
ERROR: Macros with multiple statements should be enclosed in a do - while loop
#703: FILE: audio/audio.c:1822:
+#define CASE(DRIVER, driver, pdo_name) \
+ case AUDIODEV_DRIVER_##DRIVER: \
+ if (!dev->u.driver.has_in) { \
+ dev->u.driver.in = g_malloc0( \
+ sizeof(Audiodev##pdo_name##PerDirectionOptions)); \
+ dev->u.driver.has_in = true; \
+ } \
+ if (!dev->u.driver.has_out) { \
+ dev->u.driver.out = g_malloc0( \
+ sizeof(AudiodevAlsaPerDirectionOptions)); \
+ dev->u.driver.has_out = true; \
+ } \
+ break
ERROR: externs should be avoided in .c files
#715: FILE: audio/audio.c:1834:
+ break
ERROR: space prohibited before that close parenthesis ')'
#717: FILE: audio/audio.c:1836:
+ CASE(NONE, none, );
ERROR: space prohibited before that close parenthesis ')'
#720: FILE: audio/audio.c:1839:
+ CASE(DSOUND, dsound, );
ERROR: space prohibited before that close parenthesis ')'
#723: FILE: audio/audio.c:1842:
+ CASE(SDL, sdl, );
ERROR: space prohibited before that close parenthesis ')'
#724: FILE: audio/audio.c:1843:
+ CASE(SPICE, spice, );
ERROR: space prohibited before that close parenthesis ')'
#725: FILE: audio/audio.c:1844:
+ CASE(WAV, wav, );
WARNING: added, moved or deleted file(s), does MAINTAINERS need updating?
#970:
new file mode 100644
total: 7 errors, 1 warnings, 1346 lines checked
Patch 4/14 has style problems, please review. If any of these errors
are false positives report them to the maintainer, see
CHECKPATCH in MAINTAINERS.
5/14 Checking commit b2e94202edce (alsaaudio: port to -audiodev config)
6/14 Checking commit fd47b30266b0 (coreaudio: port to -audiodev config)
7/14 Checking commit 956b77bf8deb (dsoundaudio: port to -audiodev config)
8/14 Checking commit 9361aaae4c0c (noaudio: port to -audiodev config)
9/14 Checking commit 60ef81ebee86 (ossaudio: port to -audiodev config)
10/14 Checking commit f32d4ec5660e (paaudio: port to -audiodev config)
11/14 Checking commit b13a98214e5d (sdlaudio: port to -audiodev config)
12/14 Checking commit 2b8db3b1b8a0 (spiceaudio: port to -audiodev config)
13/14 Checking commit a08295d4a4c6 (wavaudio: port to -audiodev config)
14/14 Checking commit 77a2fc41d3b9 (audio: -audiodev command line option: cleanup)
=== OUTPUT END ===
Test command exited with code: 1
The full log is available at
http://patchew.org/logs/cover.1552083282.git.DirtY.iCE.hu@gmail.com/testing.checkpatch/?type=message.
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